| 00:00.03 | UnixDog | its now alot quieter |
| 00:00.09 | *** join/#asterisk jayeola (n=user@78.146.246.174) |
| 00:02.13 | PC_Clone | does anyone happen to know what causes and "Write to 39 failed: Unknown error 500" ? It brings down my PRI after it |
| 00:02.36 | drmessano | jblack: ARE you HERE sir? |
| 00:02.42 | *** join/#asterisk RoyK (n=roy@ip-18-22-149-91.dialup.ice.no) |
| 00:02.56 | drmessano | Lazy bastard.. WAKE UP |
| 00:02.57 | drmessano | errr |
| 00:03.09 | drmessano | s/lazy bastard/hey man/ |
| 00:03.22 | drmessano | bah |
| 00:04.07 | hmmhesays | ooooooh I need a dirty woman |
| 00:05.11 | b1ch0 | any advice? or tip, i am rounding over this issue since last week |
| 00:05.26 | b1ch0 | and dont know what to do |
| 00:05.32 | drmessano | I think I need to get drunk and fix my Gizmo problem |
| 00:06.49 | [TK]D-Fender | b1ch0, Here's some advise : forget about G.723 |
| 00:07.19 | b1ch0 | ok it doesnt matter, i can use 729 |
| 00:07.22 | [TK]D-Fender | drmessano, Ironic that inebriation is usually the START of projects like Gizmo :) |
| 00:07.35 | hmmhesays | <PROTECTED> |
| 00:07.40 | [TK]D-Fender | b1ch0, then feel free to license and run that |
| 00:07.46 | b1ch0 | but i want tio try using diferent codecs on remote branch |
| 00:08.08 | b1ch0 | diferent than 711 |
| 00:08.19 | *** part/#asterisk RoyK (n=roy@ip-18-22-149-91.dialup.ice.no) |
| 00:08.24 | *** join/#asterisk rcahilig (n=test@203.115.187.98) |
| 00:08.26 | PC_Clone | hmmhesays: thats cause this version of asterisk does not support the receiving of a dirty woman by fax |
| 00:08.29 | drmessano | LOL |
| 00:08.47 | drmessano | I work better when I am drunk |
| 00:09.12 | [TK]D-Fender | PC_Clone, Mine's in-transit from Irkutsk as we speak ;) |
| 00:09.45 | b1ch0 | and on remote branch phones i only can chose : 711, 722, 723 and 729 |
| 00:09.54 | [TK]D-Fender | b1ch0, Doesn't matter. |
| 00:10.14 | rcahilig | Guys I have problem compiling zaptel-1.2.9 in centos 4.4, the error is make[1]: Leaving directory '/usr/src/kernels/2.6.9-67.0.1.EL-smp-i686' |
| 00:10.21 | drmessano | I would hate to send a woman by fax over Voip.. When it cuts off in the middle, you have a 50/50 shot at getting the good half |
| 00:10.23 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 00:10.23 | PC_Clone | so no one knows what the heck a "Write to 39" is? |
| 00:10.34 | b1ch0 | i know , i know, but just want to test it |
| 00:10.36 | PC_Clone | drmessano: either half can be the good half |
| 00:10.51 | drmessano | The path with the mouth is never the good half |
| 00:10.53 | *** join/#asterisk Bosse (i=bosse@miraculix.klykken.com) |
| 00:11.06 | PC_Clone | it's when it's faxed in landscape that you have a problem |
| 00:11.23 | drmessano | Unless you get a misfeed and only get the middle.. then, you're gold |
| 00:11.53 | b1ch0 | or asterisk is no able to handle diferent codec at the same time ? |
| 00:12.06 | [TK]D-Fender | b1ch0, Sure it can. |
| 00:12.10 | Bosse | i have a problem with internal calls on Asterisk 1.4.10 + FreePBX 2.3.1 - i've pasted sip debug and extensions on http://pastebin.ca/885553. outgoing calls to provider are ok, as well as incoming DID routing to voicemail, but no internal calls or routing to them. |
| 00:12.21 | drmessano | I've always said that I could design the perfect woman, but she would starve to death due to her lack of a mouth |
| 00:12.22 | Bosse | the calls seem to get a "603 declined" by * without packets even reaching the internal destination |
| 00:12.31 | [TK]D-Fender | Bosse, FreePBX is NOT supported here, please use their channel |
| 00:12.38 | Bosse | [TK]D-Fender: ok |
| 00:13.41 | b1ch0 | or maybe it is because i am trying to call from iax client to a sip one ? |
| 00:14.37 | lunaphyte | what is the difference between friend and peer, in sip.conf? |
| 00:14.38 | [TK]D-Fender | b1ch0, You've shown us nothing. Don't expect any advise until you do. |
| 00:14.38 | Shaun2222 | is there a simple changelog for trunk? |
| 00:14.38 | [TK]D-Fender | lunaphyte : at this point not much. |
| 00:14.38 | Shaun2222 | looking for something global, not file specific. |
| 00:15.27 | b1ch0 | what do you want to see ? |
| 00:15.39 | b1ch0 | iax, sip conf ? |
| 00:15.51 | b1ch0 | i dont know what to do |
| 00:16.34 | b1ch0 | and just want to try if this transcoding work |
| 00:16.43 | [TK]D-Fender | b1ch0, Show the CLI of the call failingwith SIP & IAX2 debug enabled, your sip.conf and iax.conf entries for everything relevant |
| 00:17.08 | b1ch0 | how do i enable debug ? |
| 00:17.16 | rcahilig | I have problem compiling zaptel-1.2.9 in centos 4.4, the error is make[1]: Leaving directory '/usr/src/kernels/2.6.9-67.0.1.EL-smp-i686' |
| 00:17.21 | [TK]D-Fender | b1ch0, "sip debug" , "iax2 debug" |
| 00:17.37 | b1ch0 | ():-) |
| 00:17.44 | [TK]D-Fender | rcahilig, That isn't the error, thats just the last thing it says as it leaves |
| 00:18.00 | [TK]D-Fender | rcahilig, and that is an ancient version. |
| 00:18.59 | rcahilig | <PROTECTED> |
| 00:19.20 | lunaphyte | [TK]D-Fender: i was just reading that that was the case with user and peer as of a while ago - friend and peer are following a similar path? |
| 00:20.16 | rcahilig | <PROTECTED> |
| 00:20.56 | [TK]D-Fender | rcahilig, Wake up time... pastebin that WHOLE DAMN THING. |
| 00:21.19 | [TK]D-Fender | lunaphyte : friend used to mean user & peer |
| 00:22.37 | tzafrir | rcahilig, please try a newer version... |
| 00:22.40 | lunaphyte | these days, what might make someone choose peer vs. friend (or vice versa) in a config? |
| 00:23.02 | tzafrir | problems with newer kernels were resolved in later versions of Zaptel |
| 00:23.12 | [TK]D-Fender | lunaphyte : These days there's pretty much only "peer" |
| 00:23.18 | lunaphyte | ok |
| 00:23.21 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 00:23.21 | [TK]D-Fender | lunaphyte : Don't waste time neurosing over it. |
| 00:23.28 | Mahmoud | hey guys |
| 00:23.39 | lunaphyte | that's enough background for me, yeah. |
| 00:24.04 | *** join/#asterisk UnixDog (n=unixdog@ppp-69-238-217-105.dsl.irvnca.pacbell.net) |
| 00:24.55 | Mahmoud | any device, like SPA, but that links voip to GSM network instead? |
| 00:25.32 | Inssomniak | can anyone recommend a softphone for linux? |
| 00:26.01 | *** join/#asterisk thevoipforum_org (n=hex@87-194-8-43.bethere.co.uk) |
| 00:26.06 | [TK]D-Fender | Inssomniak, Ekiga |
| 00:26.22 | rcahilig | [TK]D-Fender: http://pastebin.com/m6e040fe8 |
| 00:26.22 | [TK]D-Fender | Mahmoud, Easily Googleable unde "GSM SIP Gateway" |
| 00:27.25 | [TK]D-Fender | rcahilig, Trash your zaptel folder, re-extract it, rebuild. |
| 00:27.26 | Mahmoud | [TK]D-Fender, heh thanks and sorry for insulting your intellegence |
| 00:28.09 | [TK]D-Fender | Mahmoud, no, thats not insulting my intelligence, that just failing to try the simple little things you should always try doing first. |
| 00:29.05 | [TK]D-Fender | Mahmoud, And Every link before I scroll is exactly what you're looking for... http://www.google.ca/search?hl=en&q=GSM+SIP+Gateway&btnG=Google+Search&meta= |
| 00:29.09 | Mahmoud | [TK]D-Fender, i wanted to know recommended/popular ones |
| 00:29.31 | *** part/#asterisk rcahilig (n=test@203.115.187.98) |
| 00:29.35 | [TK]D-Fender | Mahmoud, few people use them, and fewer report back of those who come here. |
| 00:30.29 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 00:32.08 | tzafrir | Iamnacho, ekiga or twinkle. |
| 00:33.15 | *** join/#asterisk teknoprep (n=chrisraw@unaffiliated/teknoprep) |
| 00:36.05 | UnixDog | hey |
| 00:39.16 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-7f010c69dc746f06) |
| 00:40.28 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 00:40.49 | b1ch0 | TK http://pastebin.com/m58c43327 |
| 00:42.08 | [TK]D-Fender | b1ch0, Line 8: actual format = g723, you should NOT BE USING G.723 |
| 00:42.37 | [TK]D-Fender | b1ch0, Second, there is no SIP debug in there. |
| 00:42.46 | [TK]D-Fender | b1ch0, Third FreePBX is not supported here. |
| 00:43.00 | drmessano | Is there a fourth? |
| 00:43.22 | CrashSys | Vote for d-fender! |
| 00:43.24 | [TK]D-Fender | drmessano, No, good things come in 3's like "judge, jury, and executioner" |
| 00:43.30 | drmessano | lol |
| 00:44.25 | drmessano | I love how you can tell someone GUESSED at peer settings for a provider or service and somehow it worked |
| 00:44.59 | *** join/#asterisk r0d3nt (n=astrutt@foster.stonedcoder.org) |
| 00:45.10 | drmessano | fromuser, username, authuser, user |
| 00:45.19 | drmessano | secret, password |
| 00:45.23 | drmessano | All in one |
| 00:45.41 | drmessano | reinvite=no |
| 00:45.42 | drmessano | lol |
| 00:47.12 | real0ne | i nedd a guid to install and configure asterisk on debian |
| 00:47.36 | drmessano | ~book |
| 00:47.36 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 00:48.01 | Mahmoud | lol http://youtube.com/watch?v=xgZKjJt-TkU |
| 00:49.11 | plik | real0ne: make notes of what you do to get it working, and then there'll be a guide for the next person that needs one :) |
| 00:49.34 | real0ne | ok |
| 00:49.36 | real0ne | thanks |
| 00:50.12 | real0ne | i want just a documentation |
| 00:50.29 | real0ne | how to install asterisk and configure it |
| 00:50.37 | [TK]D-Fender | real0ne, the "book" IS documentation. |
| 00:50.49 | [TK]D-Fender | real0ne, You've asked before and the answer isn;t changing. |
| 00:51.05 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 00:51.05 | real0ne | i can't bye it :P |
| 00:51.24 | Shaun2222 | real0ne: go search google for debian asterisk howto |
| 00:51.27 | Shaun2222 | sure you'll find somthing |
| 00:51.40 | real0ne | yes |
| 00:51.42 | real0ne | ok |
| 00:51.47 | real0ne | ^^ |
| 00:51.53 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 00:52.01 | drmessano | HAHAH... Watching Law & Order.. they're doing a traceroute (TraceROOT) on a machine and the IP resolved to... 392.163.1.104 |
| 00:52.12 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 00:52.13 | Shaun2222 | not really that hard to install... download source, `./configure && make && make install` for the most part... |
| 00:52.15 | plik | lol |
| 00:52.19 | lunaphyte | oooh! ipv6! |
| 00:52.26 | drmessano | No no |
| 00:52.28 | drmessano | ipv5 |
| 00:52.32 | lunaphyte | oh - ipv5 then. |
| 00:52.36 | lunaphyte | ha! |
| 00:52.40 | Shaun2222 | real0ne: http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Debian |
| 00:53.17 | drmessano | Set up a DHCP server on 999.999.999.999 and assign the next available IP... the internet will "tilt" |
| 00:53.36 | drmessano | and sadly.. only someone who remembers pinball will appreciate that joke |
| 00:53.37 | drmessano | lol |
| 00:53.43 | *** join/#asterisk [Outcast] (n=outcast@203-114-166-26.eth.sta.inspire.net.nz) |
| 00:53.51 | lunaphyte | you mean tilt like when you bang on a pinball machine? |
| 00:54.10 | [TK]D-Fender | real0ne, DOWNLOAD the PDF or read the HTML |
| 00:54.19 | real0ne | hehe |
| 00:54.29 | real0ne | ok [TK] |
| 00:54.45 | drmessano | very good lunaphyte |
| 00:54.48 | teknoprep | whats up with www.asterisknow.org being down ? |
| 00:55.11 | lunaphyte | i'm messing with an spa3012, trying to get it to pick up the phone when it rings and then pass the call into the sip phones registered to asterisk. i have it picking up the call, but i think i'm missing something when it comes to passing the call to asterisk. |
| 00:55.14 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1167864374.dsl.bell.ca) |
| 00:55.24 | lunaphyte | drmessano: thank you sir. |
| 00:56.09 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1167864374.dsl.bell.ca) |
| 00:57.11 | *** join/#asterisk MaliutaWrk (n=nikolai@kiev.lusan.id.au) |
| 01:00.41 | [TK]D-Fender | lunaphyte : www.voxilla.com <- go read the forums on how to set it up |
| 01:03.23 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 01:04.03 | hmmhesays | i wish rxfax gave me more debugging info than that |
| 01:06.00 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 01:08.56 | drmessano | 9 out of 11 people chose not to vote for Guiliani |
| 01:09.00 | drmessano | irony |
| 01:09.21 | scooby2 | lol |
| 01:10.14 | [TK]D-Fender | drmessano, one less horrifically wrong choice for the ballot |
| 01:10.28 | drmessano | indeed |
| 01:11.24 | lunaphyte | i'll admit i have a bit of a soft spot for giuliani. |
| 01:11.24 | lunaphyte | oh, wow - he dropped out. |
| 01:11.24 | drmessano | Guiliani lied more about 9/11 than Bush did.. |
| 01:11.42 | drmessano | He also set into motion the reason why the NYFD had shitty two-way radios on 9/11 |
| 01:12.10 | drmessano | So... good riddance |
| 01:12.25 | [TK]D-Fender | lunaphyte : I have a soft spot for Giuliani as well.. its right above the twigs I use to cover the gaping maw of the alligator pit I want him to walk over :D |
| 01:12.35 | drmessano | heh |
| 01:12.40 | lunaphyte | yeesh. |
| 01:12.44 | lunaphyte | :) |
| 01:12.45 | [TK]D-Fender | <- not one to mince words |
| 01:13.04 | hmmhesays | well its not even getting past negotiating speed |
| 01:13.12 | hmmhesays | can I set rxfax to a certain speed somehow? |
| 01:13.33 | drmessano | I thought you were going to say something about the soft spot you tap with the spoon to open up his "3 minute egg" shaped head after taking it out of the pot |
| 01:14.08 | drmessano | Ever time I see him I craze soft boiled eggs |
| 01:14.15 | drmessano | Everytime I see him I craze soft boiled eggs |
| 01:14.20 | drmessano | crave too |
| 01:14.23 | drmessano | Damn, forget it |
| 01:14.25 | mjoyce | heh |
| 01:14.30 | drmessano | I give up on typing, forever |
| 01:15.12 | drmessano | "Mr Guiliani, how do you feel about health care?" "9/11" |
| 01:15.18 | drmessano | Ass |
| 01:15.37 | b1ch0 | well it is not my fault if best tecnician are here and if i am not using web interface to work with this fucking pbx |
| 01:15.39 | drmessano | Billary in 08! |
| 01:15.51 | lunaphyte | omg gtfo. |
| 01:15.57 | lunaphyte | no way in hell. |
| 01:16.04 | *** join/#asterisk anthm (n=anthm@70-9-24-37.area4.spcsdns.net) |
| 01:16.04 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:16.14 | drmessano | It's either Billary or Hillbilly.. your choice |
| 01:16.18 | b1ch0 | that is why i am directly working on config files and asking in this group |
| 01:16.20 | [TK]D-Fender | drmessano, you forgot the noun and the verb in RG's answer ! |
| 01:16.32 | drmessano | lol |
| 01:16.53 | b1ch0 | if you can feel better, i can format pbx and only install asterisk .. |
| 01:16.58 | drmessano | b1ch0 |
| 01:16.59 | lunaphyte | well, it's a bit removed from any rationale, but i was living in nyc at the time, and i think it probably comes from all the emotion that goes along with that. |
| 01:17.10 | drmessano | If you have a box with FreePBX |
| 01:17.15 | drmessano | and are editing config files |
| 01:17.19 | drmessano | ITS STILL FREEPBX |
| 01:17.29 | b1ch0 | it is not my faul if " office they are using asterisk framework with freepbx |
| 01:17.35 | lunaphyte | but, i concur wholeheartedly - i was tired of every other word coming out of his mouth being 9/11. |
| 01:17.45 | drmessano | I can eat a ham sandwich in a Pinto and I am still not driving it to work! |
| 01:17.51 | *** join/#asterisk RoyK (n=roy@ip-87-8-149-91.dialup.ice.no) |
| 01:18.07 | [TK]D-Fender | b1ch0, You didn't provide even half of what was requested after I even had to tell you to enable sip debug, you didn't disable G.723 like you were told, and you're using a GUI that isn't supported here. If you're looking for sympathy, the well runs dry... |
| 01:18.48 | drmessano | Besides, if you told me in #freePBX that were hand editing a config file created by FreePBX, I would /ignore you |
| 01:19.18 | drmessano | ~YFAL |
| 01:19.37 | drmessano | jbot: YFAL = You Fail At Life |
| 01:19.47 | lunaphyte | hmm. well that _kind_ of worked. now when i call in, i get a dial tone.. |
| 01:19.53 | [TK]D-Fender | ALYAD : All Your Dreams Are Dead |
| 01:19.56 | lunaphyte | seems a bit less than secure. |
| 01:20.33 | [TK]D-Fender | lunaphyte : You need to set your FXO dialplan on the SPA to immediately throw the call over to *. |
| 01:20.58 | [TK]D-Fender | lunaphyte :looks something like "<s0:....yadda yadda |
| 01:21.01 | b1ch0 | it seems that is a bad day, still try and be back |
| 01:21.20 | b1ch0 | again ... here there are best people with best knoleadge |
| 01:21.36 | *** join/#asterisk osiris (n=osiris@c-71-205-29-230.hsd1.mi.comcast.net) |
| 01:22.47 | drmessano | Today must be upside down config day |
| 01:22.52 | *** join/#asterisk tripps (n=ss@72.20.150.196) |
| 01:24.07 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:25.34 | *** join/#asterisk egypcio (n=vinicius@unaffiliated/egypcio) |
| 01:26.01 | *** join/#asterisk BeeBuu (n=beebuu@219.132.190.48) |
| 01:27.58 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 01:30.49 | ZX81 | <PROTECTED> |
| 01:34.41 | *** join/#asterisk husimon (n=nhuisman@aeko.IfA.Hawaii.Edu) |
| 01:35.23 | husimon | does anyone know of good place to get logos for cisco phones ? |
| 01:36.05 | husimon | i found one with a penguin and asterisk logoish but it's pretty poor quality |
| 01:36.51 | Nugget | http://macnugget.org/stuff/asterisk-cow-real.bmp <-- cow logo |
| 01:37.29 | husimon | lol |
| 01:37.31 | husimon | nice |
| 01:37.39 | Nugget | ]:8) |
| 01:37.45 | husimon | why a cow? |
| 01:37.51 | Nugget | Cows are cool! |
| 01:38.21 | husimon | i think maybe i'll try and edit the logo from asterisk.org |
| 01:39.36 | jblack | drmessano: Right here |
| 01:43.46 | plik | husimon: onr of the cisco pages on voip-info.org has detail of the suggested size and the corect hex code for the 2 shades of grey they can handle (other than #000000 & #ffffff) |
| 01:43.56 | plik | /s/onr/one/ |
| 01:46.07 | DrkShadow | Is there a way to take _any_ incoming call from a given trunk and send it to a context's s extension? I'm getting now, call from 'number' to extension 'number' rejected because extension not found. |
| 01:46.24 | jblack | woot. I got my itad assignment |
| 01:47.54 | drmessano | jblack |
| 01:47.58 | drmessano | DONT MOVE |
| 01:47.58 | jblack | drmessano: |
| 01:48.09 | SteveTotaro | ~ronpaul |
| 01:48.15 | drmessano | Sipphone.. inbound.. you got it working? |
| 01:48.35 | jblack | drmessano: I believe so. I don't have anyone on sipphone to call me. I know outbound works |
| 01:48.43 | jblack | my other inbound sips work |
| 01:48.52 | drmessano | May I see your config, minus the sneaky squirrel stuff? :) |
| 01:48.54 | SteveTotaro | hey i finally figured out what was wrong with my ODBC setup |
| 01:48.58 | jblack | sure |
| 01:49.05 | SteveTotaro | damn whitespace |
| 01:50.06 | *** join/#asterisk tasterisk (n=Keyser@rrcs-24-73-85-186.se.biz.rr.com) |
| 01:51.48 | jblack | drmessano: http://pastebin.ca/885705 |
| 01:53.00 | jblack | You're welcome to call me at 1747 276 8946 if you want to test |
| 01:53.08 | Mahmoud | it seems that spa is not the best in its field |
| 01:53.26 | drmessano | Dear god.. could it be I have too much in there, even for my slimmed down config? brb |
| 01:54.07 | jblack | I did chop a couple things out of general, like forcing a certain codec. |
| 01:54.34 | jblack | I disallow all and only allow gsm+ulaw, allow guest, autokill=yes, canreinvite=yes by default |
| 01:55.57 | jblack | do you want my outgoing? |
| 01:56.15 | drmessano | in is my problem.. lemme see if I can call you |
| 01:56.21 | jblack | k |
| 01:56.32 | jblack | watching the console |
| 01:56.49 | drmessano | unavailable, and not Allison |
| 01:57.12 | jblack | pardon, sipphone |
| 01:57.25 | jblack | Can you dial 2225551234 ? |
| 01:57.28 | drmessano | Theres something VERY missing here |
| 01:57.33 | drmessano | uh |
| 01:57.42 | jblack | via sipphone |
| 01:57.58 | jblack | 222 XXX XXXX are individual chatlines |
| 01:58.40 | husimon | god damnit stupid gimp, i scale my image and it changes the fraking colors! |
| 01:58.49 | husimon | ruined all my logo work! |
| 01:59.55 | husimon | no interpolation for you gimp |
| 02:00.01 | DrkShadow | try "undo" |
| 02:00.44 | DrkShadow | but.. if you don't do interpolation.. it's gonna come out looking like crap, probably. |
| 02:01.32 | husimon | yeah but i only want 4 colors |
| 02:01.42 | DrkShadow | .... |
| 02:01.49 | husimon | you can't have more then 4 |
| 02:02.10 | DrkShadow | I'd interpolate at 24-bit and then decrease color depth. |
| 02:02.11 | *** join/#asterisk bsdwarrior (i=bashguar@24.229.241.237.res-cmts.sm.ptd.net) |
| 02:02.17 | bsdwarrior | tkd-fender |
| 02:03.46 | bsdwarrior | after I put a user in a queue I want to get the digit they press exten => 7,n,Read(key||1) but that dont work |
| 02:04.33 | *** part/#asterisk tasterisk (n=Keyser@rrcs-24-73-85-186.se.biz.rr.com) |
| 02:05.17 | drmessano | same |
| 02:05.23 | jblack | perhaps its me. |
| 02:05.28 | jblack | I didn't see a thing over here. |
| 02:05.36 | drmessano | I am gonna try something.. hang on |
| 02:05.43 | drmessano | I am seeing the same.. |
| 02:05.51 | drmessano | Nothing here on inbound |
| 02:06.06 | jblack | I got allison when I called you |
| 02:06.22 | *** join/#asterisk tasterisk (n=Keyser@rrcs-24-73-85-186.se.biz.rr.com) |
| 02:06.51 | jblack | I left you a message. |
| 02:07.17 | drmessano | Gizmo sent me the VM |
| 02:07.32 | jblack | hmmm. |
| 02:07.33 | drmessano | I think thats Alicia, not Allison |
| 02:07.34 | drmessano | lol |
| 02:07.36 | jblack | I'll be back in 5. |
| 02:07.38 | drmessano | ok |
| 02:07.44 | jblack | Yeah, it's kinda-allison. LIke a cheap knock off |
| 02:07.55 | jblack | some floozy that I could never love. |
| 02:08.01 | drmessano | lol |
| 02:08.33 | *** join/#asterisk jake[work] (n=jakep@pool-71-175-120-208.phlapa.east.verizon.net) |
| 02:09.50 | PC_Clone | anyone know where they keep the secret decoder ring for error message numbers? |
| 02:10.21 | joe | google? |
| 02:10.25 | jake[work] | my asterisk box is sending packets with tos 0xB8 and I get them at the destination (a T1 circuit) as tos 0x18. Is this common? |
| 02:10.35 | PC_Clone | and let me say, I googled and opened up a few boxes of cracker jack already |
| 02:11.00 | PC_Clone | jake[work]: check to make sure the router isn't rewriting them |
| 02:11.14 | jake[work] | the packet cap is from the router |
| 02:11.22 | jake[work] | (on the wan side) |
| 02:11.46 | jblack | drmessano: You can try sip:jblack@mercury.linuxguru.net |
| 02:11.54 | PC_Clone | is it * ----- [router] -------t1--------[router] ----- lan ? |
| 02:12.09 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 02:12.10 | jake[work] | yes |
| 02:12.20 | jake[work] | (but t1 is to the internet |
| 02:12.38 | jblack | That way we can verify my sip isn't borked, and that'll be the verification you want that sipphone isn't sending out sip. |
| 02:12.59 | PC_Clone | which router are the packets being captured at? |
| 02:13.08 | jake[work] | the T1 |
| 02:13.12 | jake[work] | sorry |
| 02:13.16 | PC_Clone | they're both on the t1 |
| 02:13.18 | jake[work] | the service location (not PBX) |
| 02:13.39 | *** join/#asterisk anthm (n=anthm@68-29-144-6.area4.spcsdns.net) |
| 02:13.39 | *** mode/#asterisk [+o anthm] by ChanServ |
| 02:13.46 | PC_Clone | is the B8 setting the EF ? |
| 02:13.51 | jake[work] | yep |
| 02:13.52 | PC_Clone | can't remember off the top of my head |
| 02:14.08 | PC_Clone | only root can set EF on linux |
| 02:14.10 | jake[work] | did a tcpdump on asterisk and it is setting it |
| 02:14.26 | PC_Clone | ya it will show you that on the box I think |
| 02:14.43 | PC_Clone | but once it goes through the layers of the os it gets stripped if I remember correctly |
| 02:14.48 | jake[work] | oh |
| 02:14.50 | PC_Clone | can you sniff outside of the box |
| 02:14.55 | jake[work] | um |
| 02:15.05 | jake[work] | i can try that |
| 02:15.05 | PC_Clone | do you control the router closest to it? |
| 02:15.12 | jake[work] | no - it's in a datacenter |
| 02:15.26 | jake[work] | but I can go to another computer in the datacenter |
| 02:15.43 | PC_Clone | do you have a second nic in that computer? |
| 02:15.57 | jake[work] | no |
| 02:16.00 | PC_Clone | ya |
| 02:16.08 | jake[work] | but np - i can test from another pc |
| 02:16.10 | PC_Clone | prob gonna have to do it off another computer |
| 02:16.16 | bsdwarrior | im trying to test this digium card, with patlooptest, I get the error (Error 4989): Unexpected result, 127 != 128, 1 bytes since last error. |
| 02:16.21 | PC_Clone | you very well could be setting it |
| 02:16.27 | PC_Clone | and it could be leaving with it |
| 02:16.31 | jake[work] | have you ever run into ISPs filtering tos? |
| 02:16.36 | PC_Clone | but then it could be being stripped by the router |
| 02:16.47 | PC_Clone | never done ToS outside a private line |
| 02:16.50 | jake[work] | ok |
| 02:17.06 | PC_Clone | mainly because most ISPs won't use it |
| 02:17.16 | PC_Clone | they could care less if you say your traffic is important |
| 02:17.32 | jake[work] | i'm paying for a T1! they should care |
| 02:17.38 | jake[work] | :o) |
| 02:17.43 | PC_Clone | yes, and they will on that T1 |
| 02:17.52 | PC_Clone | but your t1 is meeting up with fiber somewhere |
| 02:17.57 | PC_Clone | and then they don't care anymore |
| 02:17.58 | PC_Clone | :) |
| 02:18.01 | jake[work] | true |
| 02:18.28 | PC_Clone | do you have a site to site VPN to the datacenter? |
| 02:18.37 | jake[work] | nope - it's just internet |
| 02:18.55 | PC_Clone | ya, your ToS won't really mean anything |
| 02:19.06 | jake[work] | hmm |
| 02:19.11 | PC_Clone | you can tell the datacenter for their router to use it |
| 02:19.19 | PC_Clone | which then will make sure YO |
| 02:19.20 | PC_Clone | UR |
| 02:19.38 | PC_Clone | t1 considers it higher priority |
| 02:19.58 | PC_Clone | and then use it on your office T1 router so it too does |
| 02:20.09 | jake[work] | my router (at the service location) vendor tells me I need to get B8 up to there or the data traffic won't get priority |
| 02:20.12 | PC_Clone | but if there is a bottleneck somewhere upstream from either, your ToS won't matter prob |
| 02:20.22 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-da3b46c1c235a39f) |
| 02:20.26 | jake[work] | i got a sonicwall 1260 |
| 02:20.52 | PC_Clone | don't get me wrong, it's still usefull |
| 02:20.55 | DrkShadow | When I dial in, I get "Call from '<number>' to extension '<number>' rejected because extension not found." How can I figure out what context it's looking in? I have exten => _X.,1,Goto(...) but it's not executing. |
| 02:21.02 | jake[work] | correction or the voip traffic won't get priority |
| 02:21.32 | PC_Clone | well, they should be able to use other things as well |
| 02:21.46 | PC_Clone | you don't HAVE to use the EF bit to have HP |
| 02:21.47 | jake[work] | really? |
| 02:21.52 | PC_Clone | ya |
| 02:21.55 | jake[work] | hmm |
| 02:22.05 | jake[work] | not what the tier 3 tech said :o) |
| 02:22.30 | jake[work] | suppose the connection gets flooded with a huge d/l, voip traffic can still get priority? |
| 02:22.36 | PC_Clone | one sec |
| 02:22.41 | PC_Clone | let me check what I have |
| 02:22.48 | Mahmoud | any device that hosts GSM network from its own without the need of SIM cards? |
| 02:22.56 | Mahmoud | for home use |
| 02:23.15 | bsdwarrior | anyone around to help me with patlooptest ? |
| 02:23.18 | Mahmoud | the gsm-sip gateway i found needs a SIM card, which means i need to pay my SP money as well |
| 02:23.34 | PC_Clone | jake[work]: I use 0x68 b/c I don't run as root |
| 02:23.36 | PC_Clone | so no EF |
| 02:23.40 | jblack | sheesh. Are no-cost sip providers legally required to have a fundamental & fatal flaw? |
| 02:23.48 | jake[work] | hmm |
| 02:23.51 | PC_Clone | then I told my router to take 68 as the highest priority |
| 02:23.53 | drmessano | "Alicia" again |
| 02:23.57 | drmessano | PAP2 time |
| 02:24.15 | PC_Clone | and I actually also told it to rewrite it with the EF bit, but it really doesn't matter |
| 02:24.17 | jake[work] | and no prob even with huge d/l flooding the connection? |
| 02:24.22 | PC_Clone | since it's already in the router |
| 02:24.30 | PC_Clone | it's a private t1 |
| 02:24.34 | jake[work] | oh :o) |
| 02:24.42 | PC_Clone | but |
| 02:24.54 | PC_Clone | nothing else has higher priority |
| 02:24.58 | PC_Clone | so, doesn't matter |
| 02:25.13 | PC_Clone | one sec |
| 02:25.15 | PC_Clone | let me get you a doc |
| 02:25.20 | jake[work] | yeah - i was seeing http traffic using the override bit |
| 02:25.38 | jake[work] | which knocked the voip right out |
| 02:25.59 | jake[work] | 0x20 i belive |
| 02:26.00 | PC_Clone | who gave http a higher priority then voip on the router? |
| 02:26.22 | jake[work] | the source (in this case xdrive.com) |
| 02:26.28 | jake[work] | just d/l a file |
| 02:27.06 | PC_Clone | http://www.cisco.com/warp/public/105/dscpvalues.html |
| 02:27.23 | PC_Clone | the values are really just "classes" |
| 02:27.37 | PC_Clone | then you take the class and assign it a priority on the routers and switches |
| 02:27.41 | b1ch0 | here i am, back again .... thanks to drmessano |
| 02:28.17 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 02:28.22 | husimon | Well I created an ok logo, i think some pixel based editing could improve it but : http://kolea.ifa.hawaii.edu/~nhuisman/logo.bmp |
| 02:28.37 | jake[work] | does it matter though if all traffic arrives at the office router? VOIP,web, etc Will tos really fix the call quality issues? |
| 02:28.59 | PC_Clone | here's how it works on the routers |
| 02:29.17 | PC_Clone | imagine a series of tubes to quote the great senator from alaska |
| 02:29.22 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1088937109.dsl.bell.ca) |
| 02:29.23 | husimon | lol |
| 02:29.23 | jake[work] | ok |
| 02:29.26 | b1ch0 | then, if i need a codec (aka 729) i have to install and add it as module ... but if it is just a library .. how can system knows how many simultaneous chennels i can use |
| 02:29.39 | PC_Clone | it says "ok, I can fit x packets into this tube" |
| 02:29.50 | PC_Clone | and I have x+5 that want int |
| 02:29.59 | jake[work] | ok |
| 02:30.06 | PC_Clone | well I take the higer priority ones first |
| 02:30.19 | PC_Clone | then, the other router should do the same thing |
| 02:30.34 | jake[work] | ok |
| 02:30.35 | PC_Clone | you can only control what you're sending to the internet |
| 02:30.40 | PC_Clone | not what you're getting from it |
| 02:30.47 | jake[work] | so should be able to select on IP also |
| 02:30.50 | husimon | which is usually the bottleneck though |
| 02:30.51 | husimon | so that's good |
| 02:31.02 | jake[work] | and only let in what I want |
| 02:31.04 | jake[work] | ? |
| 02:31.09 | husimon | depends what your router lets you set |
| 02:31.09 | PC_Clone | let out what you want out |
| 02:31.26 | PC_Clone | tell your routers....expecially at the datacenter |
| 02:31.32 | PC_Clone | VoIP goes firswt |
| 02:31.38 | Mahmoud | i guess d/c... |
| 02:31.39 | jake[work] | the datacenter one only is for the pbx |
| 02:31.46 | PC_Clone | ??????? |
| 02:31.48 | PC_Clone | why |
| 02:31.48 | jake[work] | no issue there |
| 02:31.52 | Mahmoud | does any one know of any device to host gsm networ for home use? |
| 02:32.12 | PC_Clone | Mahmoud: someone announced one at CES |
| 02:32.22 | *** join/#asterisk St1ckm4n (i=St1ckm4n@75.145.72.133) |
| 02:32.23 | Mahmoud | i want to connect to my own GSM network in home |
| 02:32.38 | jake[work] | i can get the tos is being sent at the datacenter (still need to confirm) |
| 02:32.43 | PC_Clone | go search a round a bit for wireless home phone with cellular |
| 02:33.01 | jake[work] | i that was not english |
| 02:33.03 | PC_Clone | whatever ToS value is being sent by * |
| 02:33.04 | Mahmoud | PC_Clone, k |
| 02:33.05 | jake[work] | uck |
| 02:33.11 | PC_Clone | the router can make it the highest |
| 02:33.13 | *** join/#asterisk tasterisk (n=keyser30@rrcs-24-73-85-186.se.biz.rr.com) |
| 02:33.28 | jake[work] | rthe rotuer at the dc? |
| 02:33.32 | PC_Clone | a |
| 02:33.33 | PC_Clone | ya |
| 02:33.34 | jake[work] | ok |
| 02:33.35 | tasterisk | Can anyone help with using call files? |
| 02:33.45 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 02:33.49 | jake[work] | if that is working... |
| 02:33.55 | PC_Clone | they're groups more then prioirty values |
| 02:34.05 | bsdwarrior | tasterisk are you familar with php or perl? |
| 02:34.10 | jake[work] | i can test to my home pc and see if it is sending it |
| 02:34.19 | jake[work] | (asuming my isp isn't filtering too) |
| 02:34.41 | PC_Clone | it doesn't matter if they go out over the internet |
| 02:34.55 | tasterisk | Yes both but I'm trying to use a call file to help understand outbound calling first. How to use the right parms will help with using php and or perl. |
| 02:35.00 | jake[work] | i know the the asterisk box receives B8 from my phone at home |
| 02:35.16 | jake[work] | (over the Internet) |
| 02:35.28 | PC_Clone | ya, but it doesn't matter |
| 02:35.32 | jake[work] | so at least inbound to the asterisk box we're good |
| 02:35.36 | tasterisk | I just need to know what valuse to use for Channel, extension, context, etc. |
| 02:35.47 | jake[work] | right |
| 02:35.49 | PC_Clone | your router at home is the only one that should care about that B8 |
| 02:36.12 | PC_Clone | the router at the DC should care about the * DSCP value |
| 02:36.15 | jake[work] | but i should see it come in - (at home)? |
| 02:36.24 | PC_Clone | and the router at the office should care about the DSCP value from the LAN |
| 02:36.25 | jake[work] | assuming all is correct |
| 02:36.35 | jake[work] | come in B8 that is |
| 02:36.42 | PC_Clone | you may |
| 02:36.45 | PC_Clone | you might now |
| 02:36.46 | bsdwarrior | tasterisk - http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 02:36.48 | PC_Clone | err not |
| 02:37.03 | jake[work] | but if i do - sure sign all is well from the dc |
| 02:37.07 | jake[work] | ? |
| 02:37.11 | PC_Clone | because once your DC router knows its important, nothing else is going to anyway |
| 02:37.23 | PC_Clone | nope |
| 02:37.44 | jake[work] | cause it's over the internet |
| 02:37.51 | PC_Clone | b/c you could be sending B8 and your DC router could go "great, you think you're important, but I don't care." |
| 02:38.01 | jake[work] | ok |
| 02:38.14 | PC_Clone | your routers need to work together with the dscp emitting device |
| 02:38.40 | jake[work] | the only problem seems to be when someone at the office d/l a large file |
| 02:38.53 | PC_Clone | and you will always have that problem |
| 02:38.58 | jake[work] | aargh |
| 02:39.28 | PC_Clone | well unless your ISP is nice enough that they will do QoS for you |
| 02:39.32 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 02:39.55 | jake[work] | i can't say - hey web tcp traffic - wait for voip? |
| 02:40.06 | jake[work] | in the office router |
| 02:40.15 | PC_Clone | nope because your ISP has the big pipes |
| 02:40.17 | jake[work] | er yield to voip |
| 02:40.21 | PC_Clone | you have the small one |
| 02:40.21 | tasterisk | I've seen the page on voip-info, I have broadvoice setup. I have a sip phone installed. I can make a call using the sip phone. |
| 02:40.27 | jameswf-home | prime candidate foe censorship http://www.trixbox.org/forums/trixbox-forums/help/fonality-phone-server-problems |
| 02:40.45 | PC_Clone | so =====>--------------|router |
| 02:40.57 | PC_Clone | think about where you would want to "limit" what gets in |
| 02:41.14 | PC_Clone | at the > not the router |
| 02:41.23 | rabelais | when executing a pause, am I supposed to inline the w just like any other number? e.g. SIP/outcontext/*82w2125551234 ? |
| 02:41.32 | jake[work] | ok - then i'm screwed |
| 02:41.37 | tasterisk | I have a sip context [Bv_Out], I have a context which come with freebpx [custom-wakeupext] I have it setup to place outbound calls with a 9 prefix. |
| 02:41.46 | PC_Clone | who's your ISP? |
| 02:41.51 | PC_Clone | at the office? |
| 02:41.57 | jake[work] | cox |
| 02:42.07 | PC_Clone | as in cable? |
| 02:42.14 | jake[work] | no - it's t1 servic |
| 02:42.14 | tasterisk | Channel: SIP/BV_Out, Contect: custom-wakeupext, Extension: 98135551212 |
| 02:42.23 | jake[work] | but yes the cable company |
| 02:42.35 | PC_Clone | call up the tech service team, they might be willing to |
| 02:42.37 | tasterisk | I move the file into the outgoing directory. |
| 02:42.42 | PC_Clone | you don't know if you don't ask |
| 02:42.58 | PC_Clone | ask if they can do QoS on your inbound traffic |
| 02:43.03 | jake[work] | already have a ticket in for that. just was wondering if there were any alternatives |
| 02:43.20 | jake[work] | (on the office router) |
| 02:43.42 | PC_Clone | not really, the traffic has already clogged the tube |
| 02:43.54 | PC_Clone | you'd just be discarding it, and more would be coming |
| 02:44.15 | jake[work] | right. that makes sense |
| 02:44.25 | jake[work] | thank you for explaing this! |
| 02:44.36 | PC_Clone | I suggest keeping a 2x4 under your desk |
| 02:44.41 | jake[work] | private line is the way to go then |
| 02:44.46 | PC_Clone | when you see who's using the traffic, use it |
| 02:44.47 | jake[work] | haha |
| 02:44.53 | PC_Clone | then your phones will work |
| 02:45.00 | jake[work] | they're in CA I'm in PA - need a long 2x4 |
| 02:45.20 | PC_Clone | you COULD have a vpn to the datacenter |
| 02:45.22 | husimon | 2x4 in the form of a remote power strip |
| 02:45.27 | PC_Clone | for ALL traffic |
| 02:45.41 | PC_Clone | then all web traffic would go through that and you could prioritize it |
| 02:45.43 | jake[work] | sure - but would that get priority over their web? |
| 02:45.46 | jake[work] | oh |
| 02:45.48 | jake[work] | i see |
| 02:45.52 | jake[work] | hmm |
| 02:45.52 | PC_Clone | but your datacenter bill would go up a lot |
| 02:46.02 | jake[work] | i got 1000 GB |
| 02:46.13 | PC_Clone | trasnfer or bust speed? |
| 02:46.17 | jake[work] | transfer |
| 02:46.23 | PC_Clone | a tb a month free? |
| 02:46.34 | jake[work] | $60 |
| 02:46.44 | PC_Clone | who's your datacenter? |
| 02:46.48 | jake[work] | superb.net |
| 02:46.52 | PC_Clone | I might colo there then :) |
| 02:46.55 | husimon | shit standard cable gives you like 500gb a mo right? |
| 02:46.58 | jake[work] | they are pretty good |
| 02:47.06 | PC_Clone | ok |
| 02:47.20 | PC_Clone | well then have a VPN to the dc for all traffic |
| 02:47.24 | angryuser | what about ovh hosting? |
| 02:47.39 | jake[work] | yeah - i can try that :o) |
| 02:47.41 | PC_Clone | then route the inet out through the dc |
| 02:47.45 | jake[work] | yep - i get you |
| 02:47.53 | jake[work] | that makes sense |
| 02:47.58 | PC_Clone | now your calls are secure too |
| 02:48.05 | jake[work] | sidfe benefit |
| 02:48.24 | PC_Clone | do you have a sonicwall at the office too? |
| 02:48.36 | jake[work] | that's the only place i have it |
| 02:48.44 | jake[work] | the dc - who knows |
| 02:48.52 | PC_Clone | see if they can put one there |
| 02:48.57 | PC_Clone | do a site to site with those |
| 02:49.01 | jake[work] | for addl $ - i'm sure |
| 02:49.04 | PC_Clone | sure |
| 02:49.17 | angryuser | hey compare it to ovh hosting ;) |
| 02:49.29 | jake[work] | i can do sonic wall to asterisk box though |
| 02:49.30 | husimon | wonder how much that will slow your traffic down |
| 02:49.32 | jake[work] | ovh? |
| 02:49.34 | husimon | going over vpn |
| 02:49.51 | angryuser | ovh.net |
| 02:49.57 | angryuser | ovh.com ;) |
| 02:50.12 | PC_Clone | jake[work]: do you have spare U's |
| 02:50.13 | PC_Clone | ? |
| 02:50.17 | PC_Clone | at the colo? |
| 02:50.38 | husimon | ovh.com isn't english |
| 02:50.39 | jake[work] | it's $60 per server - they own the server |
| 02:50.55 | jake[work] | i can do as many as i want |
| 02:50.56 | *** join/#asterisk techie (n=techie@adsl-76-214-8-235.dsl.lsan03.sbcglobal.net) |
| 02:50.58 | PC_Clone | oh, you're doing dedicated server |
| 02:51.01 | jake[work] | yes |
| 02:51.12 | angryuser | the cheapest server i have ever saw 20€/M 100mbit inlimited |
| 02:51.20 | angryuser | unlimited |
| 02:51.29 | jake[work] | i'd be nervous about the unlimited deal |
| 02:51.37 | jake[work] | it's never unlimited :o) |
| 02:51.44 | husimon | yeah |
| 02:51.53 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-9b2bd580f5ebf2ee) |
| 02:51.55 | PC_Clone | jake[work]: hmmm you might be able to setup a software vpn on the server |
| 02:51.58 | angryuser | i had 4 TB of traffic once |
| 02:52.02 | PC_Clone | like openswan or something |
| 02:52.13 | husimon | oh yeah you didn't read that small clause in the agreement that says once you go over 10tb you pay $10 a kilobit? |
| 02:52.14 | jake[work] | yep - exactly what I'm thining. |
| 02:52.19 | PC_Clone | but you'd need two nic's I'm sure to route the traffic |
| 02:52.29 | carrar | or the clause that says you might have congestion |
| 02:52.31 | PC_Clone | maybe not |
| 02:52.32 | jake[work] | i can get another put in the box |
| 02:52.39 | jake[work] | i do have 2 ips |
| 02:52.39 | angryuser | no it is unlimited, no hidden stones |
| 02:52.50 | PC_Clone | well you might be all set then |
| 02:53.03 | jake[work] | that would be so cool! |
| 02:53.20 | husimon | so why aren't the asterisk servers located at the place the phones are? |
| 02:53.50 | angryuser | when you have 5 offices, easy to manage |
| 02:53.55 | jake[work] | that wouldn't buy me anything - the data would still take priority |
| 02:54.09 | husimon | oh, you run more then asterisk on the server? |
| 02:54.13 | jake[work] | (even if it were at the location) |
| 02:54.31 | jake[work] | nope - just asterisk |
| 02:54.49 | jake[work] | but it's not at the location cause i have 4 offices |
| 02:54.49 | angryuser | <husimon> no imagine a company with 5 offices all over country |
| 02:54.57 | PC_Clone | jake[work]: they have a $90 linux firewall |
| 02:55.01 | husimon | ok i understand |
| 02:55.05 | PC_Clone | ask them if that will do site to site |
| 02:55.05 | jake[work] | :o) |
| 02:55.07 | PC_Clone | vpn |
| 02:55.18 | jake[work] | yeah - my bets are on |
| 02:55.39 | jake[work] | prob not |
| 02:55.44 | PC_Clone | man, I should consult for a living |
| 02:55.48 | jake[work] | you should |
| 02:55.50 | angryuser | vpn could work, but not for centrailized config |
| 02:56.01 | jake[work] | i do |
| 02:56.06 | PC_Clone | so do I |
| 02:56.10 | PC_Clone | :) |
| 02:56.19 | jake[work] | but my installs are IVR based in a controlled environment |
| 02:56.24 | angryuser | i prefer 4 * server, so if inet go down>>>>E1/T1 |
| 02:56.37 | jake[work] | tie line to PBX |
| 02:56.38 | *** join/#asterisk robbiet480 (n=robbiet4@c-98-207-66-39.hsd1.ca.comcast.net) |
| 02:57.04 | robbiet480 | hey no one is answering in the gui room. can anyone help me. my question isnt strictly gui either. it involves manager |
| 02:57.04 | *** join/#asterisk PepOSX (n=angeldav@190.72.148.225) |
| 02:57.06 | jake[work] | goin over the internet is scary |
| 02:57.07 | PC_Clone | angryuser: huh? |
| 02:57.22 | husimon | hmm |
| 02:57.23 | PC_Clone | jake[work]: ya that's why I don't do it |
| 02:57.29 | husimon | any idea why I could call voicemail and not hear anything? |
| 02:57.38 | jake[work] | i was "helping a friend" |
| 02:57.41 | PC_Clone | my clients won't do SIP most of the time outside the office for that very reason |
| 02:57.45 | husimon | yet it says i'm connecting and playing the correct fie |
| 02:57.46 | jake[work] | then he merged with a CA office |
| 02:57.46 | husimon | file |
| 02:57.52 | husimon | i just migrated from 1.2 -> 1.4 |
| 02:58.11 | angryuser | <PC_Clone> ? |
| 02:58.14 | PC_Clone | I usually do PRIs and POTS installs for asterisk |
| 02:58.23 | jake[work] | right - much easier |
| 02:58.27 | PC_Clone | angryuser: i prefer 4 * server, so if inet go down>>>>E1/T1 |
| 02:58.35 | robbiet480 | hey no one is answering in the gui room. can anyone help me. my question isnt strictly gui either. it involves manager |
| 02:58.37 | PC_Clone | I didn't understand what you mean |
| 02:58.38 | PC_Clone | t |
| 02:58.47 | jake[work] | <robbiet480> - ask |
| 02:58.51 | robbiet480 | ok |
| 02:58.59 | robbiet480 | i am getting the following error when starting up |
| 02:59.01 | robbiet480 | http.c: Unable to bind http server to (ip):8088: Cannot assign requested address |
| 02:59.10 | robbiet480 | its starting as root |
| 02:59.12 | jake[work] | sounds like the addy is in use |
| 02:59.16 | robbiet480 | but its not |
| 02:59.22 | robbiet480 | the box is the only one with it |
| 02:59.33 | PC_Clone | no, something on the box is using port 8088 |
| 02:59.41 | robbiet480 | wait |
| 02:59.46 | robbiet480 | could it be xend? |
| 02:59.53 | jameswf-home | ~paypal |
| 02:59.53 | jbot | paypal is, like, at http://www.paypalsucks.com and http://www.paypal.com |
| 02:59.55 | robbiet480 | wait no that dosent make sense |
| 03:00.04 | robbiet480 | because this is a VPS |
| 03:00.07 | jake[work] | netstat has an option to see the processes - i think |
| 03:00.11 | robbiet480 | kk |
| 03:00.19 | PC_Clone | they're control panel might run on it |
| 03:00.24 | husimon | where can I turn on debuging to determine why I can't hear the voicemail main when I call voicemail? |
| 03:00.29 | husimon | i used to be able to hear it fine with 1.2 |
| 03:01.04 | PC_Clone | so, no one has found my secret decoder ring yet? |
| 03:01.05 | angryuser | <PC_Clone> in case of internet failure, vpn is good to interconnect different * servers, you forget nat problems and all, as i sait i prefer to do not use 100% sip installs, people need to call out if internet goes down, so in that case we pass with normal telco |
| 03:01.06 | PC_Clone | boooooo |
| 03:01.17 | robbiet480 | jake[work]: could it be selinux? |
| 03:01.20 | husimon | PC_Clone, what secret encoder ring? |
| 03:01.20 | jake[work] | <husimon> sounds like NAT issues - |
| 03:01.24 | angryuser | *said damn keyboard |
| 03:01.35 | PC_Clone | for the error numbers I'm getting in my logs :) |
| 03:01.36 | husimon | jake[work], except it's on the same network 1.2 was on, and there is no nat |
| 03:01.44 | husimon | PC_Clone, what software/hardware? |
| 03:01.47 | robbiet480 | ~selinux |
| 03:01.47 | jbot | well, selinux is the NSA's port to Linux of the FLASK security Architectur, called Security Enhanced Linux. |
| 03:01.56 | robbiet480 | ~asterisk |
| 03:01.56 | jbot | methinks asterisk is the best free PBX in the world, or #asterisk on irc.freenode.net, or http://www.asterisk.org |
| 03:01.59 | robbiet480 | hehe |
| 03:02.01 | robbiet480 | anyway |
| 03:02.02 | robbiet480 | jake? |
| 03:02.06 | jake[work] | netstat -p |
| 03:02.13 | angryuser | what do you use for vpn? proprietary or openvpn? |
| 03:02.22 | husimon | unless for some reason this box installed with a firewall when the last one didn't. |
| 03:02.29 | husimon | nope nothing |
| 03:02.30 | robbiet480 | nope jake |
| 03:02.33 | robbiet480 | not running |
| 03:02.37 | jake[work] | netstat -pa |
| 03:02.41 | PC_Clone | asterisk with a TE120P |
| 03:03.09 | robbiet480 | udp 0 0 *:mgcp-callagent *:* 1978/asterisk |
| 03:03.09 | robbiet480 | udp 0 0 *:4520 *:* 1978/asterisk |
| 03:03.09 | robbiet480 | udp 0 0 *:sip *:* 1978/asterisk |
| 03:03.09 | robbiet480 | udp 0 0 *:iax *:* 1978/asterisk |
| 03:03.19 | jake[work] | ~pastebin |
| 03:03.19 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 03:03.31 | robbiet480 | sorry |
| 03:03.34 | jake[work] | np |
| 03:03.35 | robbiet480 | hold on |
| 03:03.52 | robbiet480 | http://pastebin.com/m17d65430 |
| 03:03.53 | PC_Clone | husimon: asterisk with a TE120 |
| 03:04.02 | jake[work] | even better - telnet localhost 8088 |
| 03:04.03 | Nugget | telnet is eeeeeeevil! |
| 03:04.19 | husimon | PC_Clone, ah nothing that would ring a bell, other then googling it. |
| 03:04.22 | SteveTotaro | what exactly does the ms figure in sip show peers measure? |
| 03:04.31 | PC_Clone | <PROTECTED> |
| 03:04.32 | robbiet480 | http://pastebin.com/m727a1477 |
| 03:04.36 | PC_Clone | that is SOOOOOO descriptive |
| 03:05.06 | jake[work] | refused - ok |
| 03:05.24 | robbiet480 | [Jan 30 14:09:28] WARNING[25324] manager.c: Unable to bind socket: Cannot assign requested address |
| 03:05.42 | jake[work] | did you try chaning the port |
| 03:05.44 | jake[work] | ? |
| 03:05.46 | robbiet480 | i can |
| 03:05.47 | PC_Clone | they might not let you bind to that port |
| 03:05.49 | robbiet480 | hold on |
| 03:05.52 | SteveTotaro | sip2 xx.xx.xx.xx 5060 OK (339 ms) |
| 03:06.01 | PC_Clone | or they might not let you run services that bind to ports |
| 03:06.04 | robbiet480 | pc_clone its my own dedicated server |
| 03:06.05 | husimon | PC_Clone, might be a hardware problem |
| 03:06.05 | SteveTotaro | what is 339ms exactly? |
| 03:06.20 | robbiet480 | its my own dedicated server that i am selling vps space on |
| 03:06.24 | PC_Clone | husimon: any shot it's IRQ? |
| 03:06.30 | robbiet480 | (http://www.surephire.com |
| 03:06.31 | robbiet480 | ) |
| 03:06.36 | SteveTotaro | if i ping the ip it takes ~10ms |
| 03:06.46 | PC_Clone | husimon: I've gotten it pretty well seperated (APIC/smp_affinity) |
| 03:06.50 | SteveTotaro | so why does sip show peers say 339ms |
| 03:07.03 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 03:07.20 | husimon | PC_Clone, you could do a quick grep in the source code and try to see what prints that error |
| 03:07.20 | jameswf-home | the moment of truth is on... this show rocks |
| 03:07.33 | husimon | PC_Clone, might not be hard to find out what causes the error |
| 03:07.35 | PC_Clone | jameswf-home: what channel |
| 03:07.43 | husimon | what show? |
| 03:07.54 | robbiet480 | jake im posting the new netstat -pa |
| 03:07.57 | robbiet480 | standby |
| 03:08.01 | jake[work] | ok |
| 03:08.03 | SteveTotaro | yes, what is the moment of truth, like cheaters? |
| 03:08.10 | robbiet480 | http://pastebin.com/m35a45ff5 |
| 03:08.25 | jameswf-home | fox |
| 03:08.27 | robbiet480 | so the manager port is getting through |
| 03:08.37 | robbiet480 | its just the http |
| 03:08.39 | jake[work] | right |
| 03:08.47 | jake[work] | manager is on 5038 |
| 03:09.33 | robbiet480 | nvm |
| 03:09.33 | jameswf-home | a show with a lie detecter and fd up questions |
| 03:09.33 | robbiet480 | manager isnt getting through |
| 03:09.33 | robbiet480 | telnet says connection refused |
| 03:09.33 | robbiet480 | lemme try quick running a webserver on 80 |
| 03:09.37 | husimon | obviously they find people who have dirt to hide |
| 03:09.46 | PC_Clone | jameswf-home: I have news on |
| 03:09.47 | husimon | otherwise they would lose money real quick |
| 03:09.48 | SteveTotaro | i gotta see this |
| 03:09.58 | jameswf-home | well im on arizona time |
| 03:10.09 | PC_Clone | damn you |
| 03:10.21 | SteveTotaro | hrm, guess i can find it on youtube later |
| 03:10.27 | PC_Clone | torrents |
| 03:10.42 | PC_Clone | torrents -> usb hard drive -> xbox 360 |
| 03:10.58 | jameswf-home | piracy is a sin worth 5 hail mary's |
| 03:11.00 | husimon | so i just read that their testing isn't correct because you can't ask shock questions in a lie detector test, it would cause false values |
| 03:11.11 | jake[work] | back to church |
| 03:11.14 | PC_Clone | jameswf-home: good thing I'm not christan |
| 03:11.53 | robbiet480 | craps |
| 03:12.01 | robbiet480 | @jake apache cant get through |
| 03:12.15 | jake[work] | can't bind either |
| 03:12.19 | robbiet480 | could it be selinux? |
| 03:12.26 | jake[work] | refused doesn't sound right |
| 03:12.27 | robbiet480 | ~selinux |
| 03:12.27 | jbot | selinux is probably the NSA's port to Linux of the FLASK security Architectur, called Security Enhanced Linux. |
| 03:12.34 | robbiet480 | well the exact term is |
| 03:12.40 | PC_Clone | they might just block it since it's VPS and not dedicagted |
| 03:12.42 | robbiet480 | ya |
| 03:12.45 | robbiet480 | refused |
| 03:12.52 | robbiet480 | pc_clone |
| 03:12.55 | robbiet480 | it is dedicated |
| 03:12.59 | robbiet480 | i host with softlayer |
| 03:12.59 | PC_Clone | you could REALLY screw with the rst of the box if you can run your OWN ports |
| 03:13.02 | jake[work] | i would think not responding |
| 03:13.16 | robbiet480 | PC_Clone: its my own dedicated server and I run xen on top of it |
| 03:13.22 | robbiet480 | and I sell VPSes |
| 03:13.25 | PC_Clone | gotcha |
| 03:13.29 | robbiet480 | (http://www.surephire.com) |
| 03:13.49 | PC_Clone | dude, how old are you? |
| 03:13.50 | PC_Clone | :) |
| 03:13.53 | robbiet480 | 16 |
| 03:13.54 | robbiet480 | :D |
| 03:13.58 | jake[work] | damn - i feel old |
| 03:14.06 | robbiet480 | some call me a entrepreneur |
| 03:14.16 | PC_Clone | haha...I remember when I was that age selling hosting |
| 03:14.17 | PC_Clone | :) |
| 03:14.19 | robbiet480 | tommorow you will see asterisk all over facebook! |
| 03:14.28 | PC_Clone | now VoIP, that's balls |
| 03:14.29 | robbiet480 | we will be on facebook and the top 3 podshow podcasts coming soon |
| 03:14.50 | robbiet480 | i hope to throw some press digium/asterisks way |
| 03:14.58 | robbiet480 | anyway back to the problem at hand |
| 03:15.07 | PC_Clone | if I see you all over facebook I'm canceling my account :) |
| 03:15.16 | robbiet480 | jake should i go talk to softlayer (my hoster) |
| 03:15.20 | robbiet480 | you will PC_Clone |
| 03:15.27 | robbiet480 | we just bought 1000 impressions |
| 03:15.36 | robbiet480 | and we are expandning to 1 mil |
| 03:15.48 | robbiet480 | i see a lot of potential in asterisk/voip solutions |
| 03:15.52 | PC_Clone | what's the OS on the server? |
| 03:16.00 | robbiet480 | centos base/centos guest |
| 03:16.21 | robbiet480 | in fact |
| 03:16.24 | robbiet480 | PC_Clone |
| 03:16.27 | PC_Clone | see if there is anything in the secure log |
| 03:16.29 | robbiet480 | ill give you a promo code for preorder |
| 03:16.32 | robbiet480 | okdokie |
| 03:16.52 | jake[work] | <robbiet480> sounds like something else is using that port - it shouldn't be refused |
| 03:16.59 | robbiet480 | im checking |
| 03:17.07 | robbiet480 | actually |
| 03:17.10 | robbiet480 | all ports arent binding |
| 03:17.15 | robbiet480 | 22 isnt binding either |
| 03:18.00 | robbiet480 | ok |
| 03:18.05 | robbiet480 | im gonna give softlayer a call |
| 03:18.57 | robbiet480 | i love having softlayer |
| 03:19.00 | robbiet480 | best host ever |
| 03:19.03 | robbiet480 | i reccomend |
| 03:19.16 | robbiet480 | 24/7/365 support |
| 03:19.17 | plik | them, over you? |
| 03:19.21 | robbiet480 | rofl |
| 03:19.25 | robbiet480 | i mean for big stuff |
| 03:19.26 | plik | good business model ;) |
| 03:19.44 | robbiet480 | hold on |
| 03:19.46 | robbiet480 | ill build a code |
| 03:20.44 | jake[work] | i'm gonna try this vpn out. tnx again pc |
| 03:20.55 | PC_Clone | jake[work]: no problem |
| 03:20.58 | PC_Clone | good luck |
| 03:22.20 | PC_Clone | anyone know a good lowcost colo while we're on these topics? |
| 03:22.33 | *** join/#asterisk PiranhaP (n=piranha@74.197.39.229) |
| 03:22.38 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 03:23.36 | robbiet480 | if anyone wants it |
| 03:23.37 | robbiet480 | http://surephire.com/order/?code=ASTERISK01 |
| 03:23.45 | robbiet480 | code for early entry |
| 03:26.45 | PiranhaP | Is it only Zaptel trunks that pass DID information? |
| 03:27.07 | PiranhaP | or can it be passed via SIP or IAX2? |
| 03:27.54 | PC_Clone | PiranhaP: I think SIP and IAX trunks can as well |
| 03:27.58 | PC_Clone | depends on the provider |
| 03:29.24 | PiranhaP | The reason I ask is that I'm setup with a PSTN gateway that I have two DIDs setup with (CallWithUs.com...highly recommended, btw). I am using trixbox to setup the inbound routes, but it doesn't seem to respect my routing. I assume the DID isn't being passed. Where can I tell? |
| 03:30.09 | PC_Clone | do a sip debug |
| 03:30.12 | jameswf-home | ~trixbox |
| 03:30.12 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 03:30.40 | jameswf-home | my expierience is call withus is cheap but not good inbound |
| 03:30.52 | angryuser | Unlimited Bandwidth (100 Mbps Uplink) $ 2000 month lol <robbiet480> your hoster not really cheap look at this http://www.ovh.com/fr/particulier/produits/xeonquadcore.xml 333€/m 2x1 gbit unlimited xeon |
| 03:31.37 | robbiet480 | actually |
| 03:31.42 | robbiet480 | mine is cheap |
| 03:31.43 | PiranhaP | jameswf-home: I have had a couple minor hiccups, but they seem to be good. Then again, I don't call myself too often. |
| 03:32.05 | robbiet480 | i got dual core xeon procs with 250 GB HD 4 GB RAM 10 Mbit Port with VPN for 170 a month |
| 03:32.15 | robbiet480 | dual core at 4 GhZ per cord |
| 03:32.16 | robbiet480 | core* |
| 03:32.28 | PiranhaP | jameswf-home: And I posted here instead of #freepbx because I figured it was more of a general trunk question. |
| 03:32.32 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:32.32 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:32.48 | PiranhaP | and was curious where asterisk logs the DID |
| 03:33.00 | PiranhaP | PC_Clone: I will give that a try |
| 03:34.00 | drmessano | Anyone here using Gizmo/Sipphone with *? |
| 03:34.02 | robbiet480 | and angryuser i dont need unlmited bandwidth |
| 03:34.43 | drmessano | Got a PAP2 working PERFECTLY on it, but can't get inbound to work.. |
| 03:34.49 | drmessano | ..on asterisk |
| 03:34.56 | angryuser | 170 http://www.ovh.com/fr/particulier/produits/mg.xml 149€ 1 gbit unlimited |
| 03:35.18 | angryuser | look at config |
| 03:35.42 | angryuser | it's a quadcore q6600 |
| 03:36.26 | Inssomniak | so I would need an "FXO" card to interface my POTS line to asterisk? |
| 03:37.03 | ManxPower | ~fxofxs |
| 03:37.04 | jbot | it has been said that fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 03:37.24 | Inssomniak | thx |
| 03:37.43 | Inssomniak | I need just a single port.. X100P ok? |
| 03:37.51 | angryuser | <robbiet480> 10 mbit v 1 gbit feel the diff ;) |
| 03:37.56 | robbiet480 | true |
| 03:38.04 | robbiet480 | but i do fine on 10mbit |
| 03:38.14 | J4k3 | Inssomniak: they work ok optimally, if they don't like your line or your system doesn't like the card, no. |
| 03:38.17 | J4k3 | ;) |
| 03:38.19 | robbiet480 | and gbit for 300 more a month. i dont want it |
| 03:38.55 | *** join/#asterisk rcasterisk (n=gfdgfd@201-213-210-25.net.prima.net.ar) |
| 03:38.56 | angryuser | i dont want you change you hoster ;) but anyway if one day you need it, dont overpay |
| 03:39.04 | robbiet480 | true |
| 03:39.20 | J4k3 | wow, is it my imagination or is that colo really cheap? |
| 03:39.30 | J4k3 | MY BITS ARE FROM FRANCE |
| 03:39.47 | kyron | J4k3, they will stink |
| 03:40.10 | angryuser | when it's cheap, it dont |
| 03:40.10 | PC_Clone | don't move it to france...they're just cover it with cheese |
| 03:40.50 | Inssomniak | J4k3, if it doesnt like my line? well, its a standard issue north american dial tone, and, as for it not liking my system? as in my motherboard or PCI bus? |
| 03:41.07 | angryuser | yep all querrel cheeseVhamburgers :) |
| 03:41.15 | angryuser | old |
| 03:42.06 | jameswf-home | they should ask... have you ever fantisized about having sex with a goat |
| 03:42.35 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 03:42.54 | angryuser | only if it looks like my girlfriend |
| 03:42.59 | angryuser | <PROTECTED> |
| 03:43.17 | jblack | tmi |
| 03:43.18 | PC_Clone | angryuser: you're girlfriend looks like a goat? |
| 03:43.23 | jameswf-home | your girlfriend looks like a goat? |
| 03:43.27 | St1ckm4n | if I made changes to my zapata.conf shouldn't a "module reload" apply those changes? |
| 03:43.33 | jameswf-home | doh |
| 03:43.42 | angryuser | no look it from other side |
| 03:43.52 | *** join/#asterisk acidst0rm (i=acid@69.42.217.130) |
| 03:43.54 | jameswf-home | St1ckm4n: not always |
| 03:44.03 | PC_Clone | your girlfriend looks like a goats ass? |
| 03:44.34 | angryuser | anyway allvays better than a clone ;) |
| 03:45.24 | PC_Clone | :) |
| 03:45.39 | PC_Clone | how does an irq conflict manifest itself? |
| 03:45.47 | St1ckm4n | jameswf: I changed my callwaiting setting to no but it hasn't had any effect |
| 03:45.52 | PC_Clone | in asterisk when you have a PRI? |
| 03:47.13 | angryuser | before i go to see all prices of ovh, if you dont understand french http://www.ovh.com/fr/particulier/produits/offres_dedies.xml |
| 03:48.26 | PC_Clone | ~irq |
| 03:48.26 | jbot | irq is probably (Interrupt request) This is a gateway for devices to request time from the processor to do their jobs. For instance, every time you hit a key on your keyboard an interrupt is generated on the keyboard IRQ. This is mainly only a concern for PC users. |
| 03:48.41 | ManxPower | PC_Clone: HDLC Abort or FCS Erors |
| 03:48.54 | PC_Clone | ManxPower: FCS? |
| 03:48.59 | PC_Clone | frame control? |
| 03:50.32 | BeeBuu | anyone using Adminsparadise here? |
| 03:51.11 | drmessano | AP is dead |
| 03:51.27 | BeeBuu | dead? |
| 03:51.37 | drmessano | I heard it's dead.. gone |
| 03:51.40 | kyron | PC_Clone, cat /proc/interrupts |
| 03:51.58 | BeeBuu | drmessano: are you sure? |
| 03:52.11 | PC_Clone | here is the log http://pastebin.com/d3c56bf1a |
| 03:53.00 | PC_Clone | http://pastebin.com/d481a8a2f |
| 03:53.10 | PC_Clone | cat /proc/inturrupts |
| 03:53.26 | drmessano | BeeBuu, all signs point to "Dead" |
| 03:54.21 | BeeBuu | ... |
| 03:56.24 | BeeBuu | drmessano: any suggestion about FAX system with asterisk? |
| 03:56.27 | PiranhaP | PC_Clone: I wasn't able to see the DID info in the 'iax2 debug'. But looking in /var/log/asterisk/full, I see something that makes me think the DID may be a mixture of my IAX account name and the DID. |
| 03:57.25 | PC_Clone | PiranhaP: they don't have an example config for you to use on their site? |
| 04:00.38 | PiranhaP | PC_Clone: They have a lot of examples for getting the trunk set up, but no info that I can find on the format of the DID. |
| 04:00.59 | PC_Clone | I would email tech support and ask how the DID comes in |
| 04:01.15 | PC_Clone | for instance, on a PRI I deal with it only sends 4 |
| 04:01.19 | PC_Clone | but some send 7 |
| 04:01.27 | PC_Clone | so they could use anything |
| 04:03.02 | angryuser | normally you can see it even on cli......if no actions are defined for that DID"s |
| 04:03.06 | PiranhaP | PC_Clone: I see. That's why I wish I could just find it in a log. But their support is very responsive, even enough for my impatient self :) |
| 04:03.42 | PC_Clone | PiranhaP: you pay for their service, including support. USE it and don't waste YOUR time |
| 04:04.50 | PiranhaP | PC_Clone: Right, right |
| 04:05.19 | PC_Clone | kyron: did you get a chance to look at all that? |
| 04:06.56 | *** join/#asterisk bkruse_home (n=kruz@76.73.154.120) |
| 04:06.56 | *** mode/#asterisk [+o bkruse_home] by ChanServ |
| 04:23.13 | *** join/#asterisk mikebee (n=jcrum@66-193-179-217.static.twtelecom.net) |
| 04:24.32 | mikebee | Hi, anyone around that can answer a question about a codec_g729a.so problem I've been having on 1.4.17? |
| 04:27.13 | *** part/#asterisk techie (n=techie@adsl-76-214-8-235.dsl.lsan03.sbcglobal.net) |
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| 04:28.30 | *** mode/#asterisk [+o bkruse_home] by ChanServ |
| 04:31.26 | *** join/#asterisk angryuser (i=nononon@df01t2-212-194-194-206.d4.club-internet.fr) |
| 04:32.41 | *** join/#asterisk sergey (n=sergey@91.189.233.66) |
| 04:32.41 | jameswf-home | ~ask |
| 04:32.42 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 04:38.15 | PC_Clone | jameswf-home: have you ever had IRQ issues in your time? |
| 04:38.43 | *** join/#asterisk teknoprep (n=chrisraw@unaffiliated/teknoprep) |
| 04:40.40 | jameswf-home | yes |
| 04:40.51 | jameswf-home | what kernel are you on |
| 04:41.11 | PC_Clone | 2.6.9.something....hold on |
| 04:41.27 | *** join/#asterisk CrashSys (n=kumba@62-209.187-72.tampabay.res.rr.com) |
| 04:41.27 | PC_Clone | <PROTECTED> |
| 04:41.55 | jameswf-home | you should get off trixbox or upgrade to 2.4 |
| 04:42.01 | PC_Clone | here is the log of something that happens a few times a week 2.6.9-34.0.2.ELsmp |
| 04:42.04 | PC_Clone | woops |
| 04:42.08 | CrashSys | Get off of Trixbox?!?!? Blaspheme! |
| 04:42.29 | PC_Clone | http://pastebin.com/d3c56bf1a |
| 04:42.41 | PC_Clone | jameswf-home: not my call |
| 04:42.56 | PC_Clone | I support another company....they have what they have |
| 04:42.59 | jameswf-home | well 2.6.9-34 is whats killing you |
| 04:43.17 | jameswf-home | you should be on 2.6.18+ |
| 04:43.45 | PC_Clone | how so? |
| 04:43.53 | drmessano | Trixbox is like training wheels without a bike |
| 04:44.08 | jameswf-home | how so? upgrade |
| 04:44.25 | PC_Clone | no, I mean why is that kernel causing issues that are fixed in 2.6.18 |
| 04:44.52 | jameswf-home | 2.6.9 has no apic support and handles IRQ's like michael jackson in a preschool |
| 04:46.11 | PC_Clone | ha |
| 04:47.04 | jameswf-home | and |
| 04:47.08 | jameswf-home | ~trixbox |
| 04:47.08 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 04:47.58 | PC_Clone | fine :) |
| 04:50.30 | *** join/#asterisk techie (n=techie@adsl-76-214-8-235.dsl.lsan03.sbcglobal.net) |
| 04:50.39 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-ba1710199e3411d4) |
| 04:57.28 | robbiet480 | nighty night |
| 04:57.35 | robbiet480 | thx for the help PC_Clone |
| 04:57.42 | robbiet480 | and jake if he is still around |
| 04:57.43 | PC_Clone | no prob |
| 04:57.59 | robbiet480 | www.surephire.com/order/?code=ASTERISK01 |
| 04:58.00 | robbiet480 | :D |
| 04:58.01 | robbiet480 | cyz |
| 04:59.41 | husimon | I'm having an issue with playing back audio, i'm not on a nat, where do I look first? |
| 05:00.31 | husimon | http://pastebin.com/m34c8a17e |
| 05:00.38 | husimon | it says it is playing back but I hear nothing |
| 05:00.52 | husimon | I previously was on 1.2 and everything worked fine, then I moved to 1.4 and now no audio |
| 05:04.47 | husimon | weird |
| 05:04.51 | husimon | unloading ztdummy fixed it |
| 05:09.36 | *** join/#asterisk metfan2007 (n=metfan20@201.103.91.58) |
| 05:09.43 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 05:09.57 | metfan2007 | Hi all!!! How can I change the language for Local channel??? |
| 05:10.24 | metfan2007 | I need to do it without changing anything in extensions.conf |
| 05:10.39 | metfan2007 | something like sip.conf, zapata.conf, etc. |
| 05:10.50 | husimon | playback language? |
| 05:11.00 | metfan2007 | es |
| 05:11.08 | metfan2007 | spanish |
| 05:11.27 | metfan2007 | I have set language=es in sip.conf |
| 05:11.39 | husimon | did you download the spanish language pack? |
| 05:11.51 | metfan2007 | but when I use Dial(Local/....blahblah it changes to english |
| 05:11.56 | metfan2007 | yep |
| 05:12.18 | husimon | hmm not sure i'm new to asterisk too |
| 05:12.21 | metfan2007 | using Dial(SIP/.... everything works fine |
| 05:12.36 | metfan2007 | anyone else? |
| 05:13.45 | husimon | is there an external config file for Local? |
| 05:15.06 | metfan2007 | if you find it tell me xD |
| 05:16.21 | husimon | do you really need to use Local/? |
| 05:16.28 | husimon | that's for what? changing contexts? |
| 05:16.47 | husimon | and why can't you change it in extensions.conf? |
| 05:19.05 | husimon | shouldn't : exten => 5000,1,VoicemailMain(${CALLERIDNUM}) : get the extension and pass it as the username? |
| 05:20.34 | husimon | oh maybe the callerid has text in it |
| 05:22.05 | metfan2007 | In fact I need to use Local channel, so I'll be searching for an answer |
| 05:22.10 | metfan2007 | thanks anyway |
| 05:32.04 | metfan2007 | any body has used VICIDial? |
| 05:33.01 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 05:33.22 | CrashSys | What about vicidial? |
| 05:44.05 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 05:44.57 | husimon | odd |
| 05:45.14 | husimon | any idea why if i call my voicemail it doesn't pass the number from callerid along? |
| 05:45.32 | husimon | it says "mailbox" and then if I let it go for about 7 seconds it asks for password |
| 05:45.45 | husimon | seems like a pretty long delay |
| 05:47.23 | metfan2007 | I have some issues with Vicidial, It does not show the customer info on the agent screen.... the phones rings ok, the Asterisk CLI is Ok, but simply it does not show the info |
| 05:47.27 | metfan2007 | any idea? |
| 05:52.07 | obnauticus | Where do i configure voicemail -> Email server settings? |
| 05:52.20 | obnauticus | i know voicemail.conf, but what configuration options |
| 05:53.27 | obnauticus | Nevermind :\ |
| 05:53.50 | husimon | basically the default will work with you adding your entries under [default] |
| 05:53.54 | husimon | then you can figure out exactly what you need |
| 05:53.56 | husimon | http://www.voip-info.org/tiki-index.php?page=Asterisk+VoiceMail |
| 05:54.47 | husimon | i don't understand why my voicemail isn't auto logging in |
| 05:54.53 | husimon | at least to the extension level |
| 05:56.28 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:58.10 | obnauticus | thanks husimon |
| 05:58.23 | *** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290) |
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| 06:25.42 | hmmhesays | so iaxmodem uses spandsp? |
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| 07:26.54 | metfan2007 | it is possible to avoid Local channel to change the language setting? |
| 07:27.51 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 07:28.11 | BeeBuu | is trixbox come with any CRM system? |
| 07:28.36 | BBHoss | this aint the place to ask about trixbox bro |
| 07:28.45 | angryuser | ${LANGUAGE} |
| 07:28.56 | *** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp) |
| 07:29.09 | BBHoss | but yes, it comes with SugarCRM, just don't come looking here for support |
| 07:29.18 | angryuser | Predefined Channel Variables <metfan2007> |
| 07:30.05 | angryuser | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetLanguage <metfan2007> |
| 07:30.08 | metfan2007 | angryuser, but how do you configure de Local channel default language?? there is no language.conf |
| 07:30.12 | metfan2007 | sorry, local.conf |
| 07:30.27 | BeeBuu | thanks,BBHoss,i just want to know that. |
| 07:30.47 | BBHoss | BeeBuu, you can ask questions in #freepbx and #trixbox |
| 07:30.49 | metfan2007 | angryuser, that's the issue, I don't want to modify extensions.conf |
| 07:31.01 | BBHoss | tough |
| 07:31.10 | angryuser | http://www.voip-info.org/wiki/view/Asterisk+multi-language |
| 07:31.16 | BBHoss | metfan2007, thats the only way i know of |
| 07:31.17 | BeeBuu | i got that. |
| 07:31.35 | angryuser | The default language for a particular channel is set in that channel's configuration file. Look for a setting like: |
| 07:31.35 | angryuser | <PROTECTED> |
| 07:32.24 | metfan2007 | angryuser, I know that, but for the "Local" channel type there is not configuration file... |
| 07:32.50 | angryuser | onli in dialplan thne |
| 07:33.07 | angryuser | pff only at dialplan then |
| 07:34.57 | angryuser | you want to have a multi language ivr? |
| 07:35.57 | sergee | morning comrades |
| 07:36.37 | sergee | angryuser: SetLanguage is deprecated |
| 07:36.47 | *** join/#asterisk CVirus (n=GoD@82.201.178.194) |
| 07:37.18 | sergee | angryuser: metfan2007: SET(CHANNEL(language)=es) |
| 07:37.27 | angryuser | i dont need it anyway ;) |
| 07:37.59 | angryuser | but thx for info good to know |
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| 07:43.38 | *** join/#asterisk amazen720 (n=amazen@ool-182c45a9.dyn.optonline.net) |
| 07:45.05 | amazen720 | Hey guys I need some help |
| 07:45.50 | amazen720 | I have just installed FreePBX 2.3.1.3 trixbox CE and the music on hold doesnt play the MP3 |
| 07:46.30 | drmessano | lol |
| 07:46.33 | drmessano | #trixbox |
| 07:47.02 | amazen720 | I just read that this version of trixbox left out MP3 compatibility, is this true |
| 07:47.19 | drmessano | This is not the #trixbox support channel |
| 07:47.28 | amazen720 | okay I'm sorry wrong room |
| 07:47.32 | angryuser | in your place i would reencode the mp3 |
| 07:48.09 | amazen720 | reencode it to what format |
| 07:48.43 | angryuser | any compatible by * gsm, or wav |
| 07:49.16 | amazen720 | I tried .wav doesn't work either |
| 07:50.47 | angryuser | wav can be different ;) http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk |
| 07:50.55 | sergee | wav is container |
| 07:51.12 | sergee | recode it to ulaw, 64 signed 8 bit mono |
| 07:52.04 | amazen720 | okay I'll try that |
| 07:55.36 | amazen720 | sergee: what would be the best program to re-encode wav to ulaw or gsm? |
| 07:56.28 | *** join/#asterisk ^shark_ (n=^shark_@217.194.147.193) |
| 07:56.36 | sergee | amazen720: ulaw |
| 07:58.31 | amazen720 | sergee: I'm kinda a newbie, how do I use ulaw to convert the .wav or mp3 file |
| 07:59.02 | sergee | i believe you need to read man page for sox utility |
| 08:00.04 | amazen720 | sergee; I see their is a program called sox, I think I'll try that |
| 08:04.40 | angryuser | <PROTECTED> |
| 08:10.21 | *** join/#asterisk tparcina (n=tparcina@cisco10.fesb.hr) |
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| 08:15.07 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 08:15.07 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk, -Addons, Libpri 1.6.0-beta2 (2008/01/28), Asterisk 1.4.18-rc3 (2008/01/30), *-Addons 1.4.5 (2007/12/1), Zaptel 1.4.8 (2008/01/14), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 08:15.38 | *** join/#asterisk l0verb0y (i=l0verb0y@210.1.137.41) |
| 08:20.53 | tparcina | hi channel! |
| 08:22.27 | tparcina | I need to configure asterisk to make outgoing calls, and once when called party picks up I need to play them message. and no, I won't spam people! |
| 08:22.35 | BBHoss | yeah right |
| 08:22.39 | BBHoss | whats it for then |
| 08:23.09 | tparcina | it's for one tourist agency which needs to contact people who's apartmant's it's ranting |
| 08:23.35 | BBHoss | look at call files, in the ~book |
| 08:23.40 | BBHoss | ~book |
| 08:23.41 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 08:23.57 | tparcina | once when someone asks that agency for apartman, they'll put 10 numbers to call, and they will tell (press some number) are they free to rent or not |
| 08:24.07 | BBHoss | then build a php script that will makes the call files from a list of numbers |
| 08:24.14 | tzafrir | tparcina, but if you're calling out via an analog line be aware that a call is considered answered immediately |
| 08:24.31 | tparcina | I knwo about call files, but I didn't know that call file can play message |
| 08:24.39 | BBHoss | uhh yeah |
| 08:24.49 | BBHoss | they can do anything, they just call people and you tell them what to do |
| 08:25.29 | tzafrir | And besides, I just wonder if you call there to rant about those aparments. That would be making a good use of Asterisk |
| 08:25.45 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 08:25.48 | tzafrir | (Sorry, couldn't resist) |
| 08:25.48 | BBHoss | business is business... :) |
| 08:26.09 | tparcina | BBHoss: yes, php script will generate call files. i guess I have, once again, to check what call file can do :) |
| 08:26.27 | tparcina | tzafrir: thank you, I'll have that in mind |
| 08:26.50 | tzafrir | tparcina, there are a number of ways to generate calls. There's also the Originate manager event |
| 08:26.52 | tparcina | BBHoss: do you have any example code for that? |
| 08:27.05 | tzafrir | And as of 1.4 there's also the "originate" command from the CLI |
| 08:27.07 | BBHoss | basically you can call someone and use the Playback application to playback a message, or in your case, you would pass them into a context which would be an IVR |
| 08:28.10 | tparcina | tzafrir: sorry, I didn't understant this (good use of Asterisk) |
| 08:28.57 | tzafrir | What trunk do you call through? |
| 08:29.04 | tzafrir | zap? sip? |
| 08:29.10 | BBHoss | tzafrir, yeah you could probably use the asterisk -ax "originate blah blah" in a shell script to do it |
| 08:29.26 | tzafrir | -rx |
| 08:29.39 | tparcina | tzafrir: which one do you recomend, using a call file or Originate manager event? will both do job just right? |
| 08:29.44 | BBHoss | yeah, my bad :) |
| 08:29.51 | BBHoss | yeah |
| 08:30.04 | tzafrir | All of them do basically the same thing |
| 08:30.21 | tzafrir | a call file gives you slightly more options, I think |
| 08:30.35 | tzafrir | But may be less convinient |
| 08:30.56 | tzafrir | manager may be handy if you already have a web interface and such |
| 08:31.02 | tparcina | BBHoss: it won't be IVR, it would be simple AA |
| 08:31.10 | BBHoss | yeah because you'd have to generate them from a file of numbers |
| 08:31.20 | tparcina | tzafrir: I'll call through SIP lines |
| 08:31.23 | tzafrir | If you just want to run something from cron, a CLI command may be the simplest way |
| 08:32.03 | tzafrir | originate SIP/peername/number extension extension@context |
| 08:32.27 | BBHoss | anybody know where to get a dvorak mechanical-switch keyboard? |
| 08:32.50 | tzafrir | with extension@context - land it in a nice place in your dialplan - play a message and maybe even a small IVR |
| 08:33.05 | *** join/#asterisk chesty (n=chesty@chesterton.id.au) |
| 08:34.09 | tparcina | tzafrir: it can't be from cron, because the workers should have to put what numbers should be called and when. so web interface is most convinient |
| 08:35.32 | tzafrir | In that case, a manager interface may be the most straight forward manner |
| 08:35.36 | tparcina | tzafrir: thank you, I didn't know it's simmple as that :) |
| 08:39.25 | *** join/#asterisk af_ (n=getsmart@88-149-240-211.dynamic.ngi.it) |
| 08:45.38 | *** join/#asterisk rvhi (n=chatzill@udp186710uds.hawaiiantel.net) |
| 08:46.29 | rvhi | hi we are going to buy a fax server, any good ones? we don't want to build our own, just want to buy one for both inbound and outbound faxing for our customers |
| 08:50.15 | l0verb0y | does anyone know how to get sox to decode mp3s? |
| 08:50.46 | eharris | is there any way to read a star (*) keypress as a menu option in extensions.conf? |
| 08:53.02 | BBHoss | * |
| 08:54.48 | *** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp) |
| 08:58.43 | *** join/#asterisk lrk (n=lrk@213.145.102.1) |
| 08:59.24 | FlatFoot | morning all |
| 08:59.58 | FlatFoot | tzafrir: do you ever sleep ? |
| 09:01.22 | eharris | * seems to be considered an invalid extension, and doesn't work with WaitExten |
| 09:01.36 | *** join/#asterisk Sajjad_Ali_Musht (n=Sajjad_A@octroi.enst-bretagne.fr) |
| 09:03.40 | *** join/#asterisk ferksy (n=abc@p50867EC9.dip.t-dialin.net) |
| 09:09.16 | tzafrir | occasionally |
| 09:09.45 | FlatFoot | tzafrir: could swear you were in here talking last night bout 10pm uk |
| 09:14.01 | BBHoss | METH IS FUN<--------- |
| 09:15.33 | FlatFoot | BBHoss you on that stuff again ! |
| 09:15.48 | BBHoss | fortunately, yes |
| 09:19.40 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:20.17 | *** join/#asterisk saftsack (n=oliver@p4FC75E9F.dip.t-dialin.net) |
| 09:20.32 | saftsack | scripts/Makefile.build:46: *** CFLAGS was changed in "/usr/src/asterisk/bristuff-0.3.0-PRE-1y-p/zaptel-1.2.22.1/Makefile". Fix it to use EXTRA_CFLAGS. Schluss. |
| 09:20.37 | saftsack | has someone an idea? |
| 09:22.18 | tzafrir | saftsack, hmm.... building bristuff on kernel 2.6.24? |
| 09:22.34 | saftsack | yes |
| 09:22.46 | tzafrir | I commited a fix for that in zaptel svn, but it's a bit delicate |
| 09:22.51 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 09:23.05 | saftsack | do you have a link to this patch? |
| 09:23.30 | tzafrir | not sure exactly, because there were a number of changes |
| 09:23.44 | tzafrir | It involved moving some stuff out of the makefile |
| 09:23.54 | tzafrir | And even some renames and such |
| 09:24.37 | saftsack | would be very nice if you would look because i dont know for what i should search |
| 09:25.02 | *** part/#asterisk goodmove (n=yves@206.48.58.100) |
| 09:25.23 | tzafrir | maybe use zaptel svn (svn co http://svn.digium.com/zaptel/branches/1.2 zaptel-1.2) and slightly edit download.sh accordingly |
| 09:28.22 | saftsack | yes or i will try the 1.4 branch from bristuff now ;) |
| 09:28.34 | saftsack | with the cause 18 patch. should run too. |
| 09:28.40 | tzafrir | you'll basically need to do the same changes |
| 09:28.51 | tzafrir | What hardware do you have? |
| 09:31.06 | tzafrir | What do you mean by "cause 18 patch"? |
| 09:31.19 | saftsack | do you get german? |
| 09:31.27 | saftsack | http://www.ip-phone-forum.de/showthread.php?t=157107 |
| 09:33.27 | saftsack | but same error here |
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| 09:39.07 | saftsack | tzafrir, still here? |
| 09:39.18 | tzafrir | yes |
| 09:40.51 | saftsack | 1.4.8 and 1.2.23 standalone fails too |
| 09:41.23 | saftsack | trying svn now |
| 09:45.49 | tzafrir | saftsack, I'm not familiar with that issue. Trying to figure it from the thread via babelfish |
| 09:46.09 | saftsack | svn works |
| 09:48.20 | tzafrir | there's a zip file in that thread , but it seems that getting it requires registration. And I really don't feel like registering to yet another forum... |
| 09:49.44 | saftsack | i can download it for you |
| 09:50.00 | saftsack | eehhmm with what do i have to replace readv and writev elelments? |
| 09:50.05 | saftsack | something with aio, or? |
| 09:50.35 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 09:55.50 | saftsack | dunno the real problem ;) but i now that the svn works with all patches |
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| 10:19.44 | yang | Can I install asterisknow on top of asterisk 1.4 ? I would like to test the GUI for configuration files |
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| 10:21.23 | dandre | hello, |
| 10:21.52 | dandre | Is there any free tapi driver that works for asterisk 1.4 ? |
| 10:22.51 | mosty | dandre, what is tapi? |
| 10:23.37 | dandre | This allow any windows application (ie outlook) to dial a phone number |
| 10:23.38 | yang | dandre: There is some tapi, whcih works in 1.2 allready |
| 10:23.54 | yang | dandre: but don't ask me how its configured |
| 10:24.11 | mosty | dandre, wouldn't it be simpler to just use a sip software phone? |
| 10:24.14 | dandre | I have tried asttapi and siptapi but with no uccess |
| 10:24.51 | dandre | from outlook, right click on a contact then dial |
| 10:25.07 | mosty | dandre, there are sip plugins for outlook that can do that |
| 10:25.14 | dandre | and I want to use my desktop phone not a soft phone |
| 10:25.44 | mosty | dandre, get an ATA, or a real sip phone? :) |
| 10:26.00 | mosty | i bet you will find this solution much better supported |
| 10:26.32 | dandre | yes I do have a real phone but I want to make this phone dial the number from outlook |
| 10:26.53 | mosty | what kind of phone is it? |
| 10:27.11 | dandre | sip, iax or zap |
| 10:27.39 | dandre | no matter , I want a global solution for all my phones |
| 10:28.24 | mosty | i think you can get "click to call" plugins for outlook that will ring your phone and once you pickup, they will ring the destination |
| 10:29.04 | BBHoss | heres one, requires AMI |
| 10:29.05 | BBHoss | http://www.thirdlane.com/products/call-companion |
| 10:29.21 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 10:29.48 | BBHoss | fscking expensive |
| 10:30.31 | mosty | can't be the only one out there though |
| 10:31.05 | yang | dandre: http://www.voip-info.org/wiki/view/Asterisk+TAPI |
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| 10:33.19 | dandre | yang: ok but I haven't got those working correctly |
| 10:34.36 | yang | well with some tricks you should be able...becouse where I work we got it working with 1.2.x |
| 10:35.44 | patrick-- | hey everyone.can those cards supply my ISDN phones with power? http://www.pbx-shop.de/product_info.php/info/p482_Sangoma---A500-14-Port-S-T-BRI.html |
| 10:36.18 | mosty | patrick--, look on sangoma's wiki? |
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| 10:39.30 | Agrajag- | has anyone actually managed to get a decent enough festival setup for production use? |
| 10:41.25 | cy3o3 | So is RTP fully supported in * these days? |
| 10:41.53 | mosty | cy3o3: what part of rtp do you think is missing? |
| 10:43.17 | cy3o3 | guess I need to research RTP |
| 10:43.24 | JT | :o |
| 10:44.13 | mosty | cy3o3: well SIP uses RTP, and I can make SIP calls just fine... |
| 10:44.26 | cy3o3 | yeah was just reading that heh |
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| 10:47.10 | *** part/#asterisk VijayG (n=vijay@58.68.47.109) |
| 10:52.24 | cy3o3 | I'm just looking at this http://farsight.freedesktop.org/wiki/ |
| 10:52.32 | cy3o3 | wondering how that'd play with * |
| 10:54.42 | mvanbaak | looks cool |
| 10:54.44 | mosty | i'm not sure, i've never tried video conferencing with asterisk |
| 10:56.41 | mvanbaak | me neither, dont have the hardware to test it |
| 10:57.09 | BBHoss | heh the app_conference supports video CONFERENCES! |
| 10:58.03 | dandre | yang: the problem with this solution is that a user defined in manager.conf with call privilege as stated can issue originate command in ANY context. I want to set the context as in sip.conf |
| 11:00.00 | BBHoss | counterpath eyebeam supports outlook-integration |
| 11:01.13 | BBHoss | then you could set it up to do call forwarding to your ip phone if necassaru |
| 11:02.58 | BBHoss | actually it works differently, so it might not work like you want it |
| 11:04.22 | BBHoss | dandre, also here http://www.callingcircles.com/ |
| 11:06.26 | dandre | ok I'll look at these products |
| 11:06.38 | BBHoss | i think the second one will work better |
| 11:06.53 | BBHoss | it uses TAPI, i don't think counterpaths soultuion does |
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| 11:23.36 | Modcuts | Morning... i'm upgrading from 1.2 to 1.4 and i wanted to know if i need to put anything in cdr.conf other then enabled=yes if i'm using a mysql cdr |
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| 11:25.03 | BBHoss | Modcuts, look at the UPGRADE txt file, it will tell you what you need, or SHOULD |
| 11:25.53 | Modcuts | i did have a look and it still didn't seem to be working properly.. will take another look |
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| 12:06.00 | windback | Does exist any plugin to restric the call based in a password for each user?? |
| 12:06.11 | windback | *restrict |
| 12:06.32 | mvanbaak | Authenticate() |
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| 12:15.27 | Azam | hi , can i dial a call to another asterisk server at a specific context |
| 12:16.28 | Azam | i mean i want to do something like this Dial(SIP/123@192.168.0.1{context}) |
| 12:16.38 | Azam | somehow i want to define the context as well |
| 12:17.18 | Azam | actually the other server has this extension 123 but it is in another context |
| 12:17.21 | Azam | please help |
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| 12:19.18 | __freedom__lover | hi guys, i need to consult a cdr record in a h extension, but in asterisk 1.2.13 cdr is recorded just after h extensions |
| 12:19.38 | __freedom__lover | someone knows a way to do that? |
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| 12:25.51 | Azam | How we can include context in our realtime configuration? How to add inlude=>local to include local context in distance call context? Is there any way to put that line in our extension table? |
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| 12:26.39 | __freedom__lover | Azam: no, but i have a solution for you |
| 12:27.30 | __freedom__lover | Azam: if you want manage that in a php or cli interface.. |
| 12:28.21 | Azam | __freedom__lover: i dont have tht :( |
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| 12:46.55 | hmmhesays | coppice: you were right about rxfax, I can't get it to successfully receive a fax at all. |
| 12:47.14 | coppice | does txfax send them OK? |
| 12:47.36 | hmmhesays | haven't tried yet, I will shortly though |
| 12:48.22 | coppice | txfax should work OK. it uses very little floating point, when built for a blackfin - assuming you are using a recent version |
| 12:49.16 | hmmhesays | I'm using the version out of agx-addons on sourceforge |
| 12:49.51 | hmmhesays | I didn't see http://www.pastebin.ca/886090 is the output from the blackfin |
| 12:50.21 | hmmhesays | whoa two sentences there. I didn't seen any tx or rxfax apps on soft-switch.org anymore |
| 12:51.09 | Modcuts | whats the best way to see if a module has loaded or not? |
| 12:51.23 | hmmhesays | show modules? |
| 12:51.53 | coppice | I was referring to a recent spandsp, not the apps. those are just tiny shims to interface spandsp to asterisk |
| 12:53.02 | tzafrir | module show ... |
| 12:53.13 | tzafrir | or unload it and load it |
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| 12:57.14 | *** join/#asterisk xezz (n=elleel@trust-it.gr) |
| 12:58.57 | xezz | hello, i have a linksys spa 962, is it possible to lock/unlock the phone with a combination hit os something ? i dont want others to make calls without permission.. |
| 12:59.21 | sergee | xezz: yes it is |
| 12:59.37 | sergee | xezz: keywords: hot desk dialplan |
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| 13:01.48 | hmmhesays | oh thats great with pc's reboot themselves |
| 13:02.26 | hmmhesays | Jan 31 13:00:01] WARNING[763]: app_txfax.c:408 txfax_exec: Transmission loop error |
| 13:03.44 | hmmhesays | coppice is there a version of txfax I should be trying beside the agx one on sourceforge? |
| 13:05.41 | coppice | that one on sourceforge, with the latest spandsp from soft-switch.org should work OK |
| 13:07.13 | hmmhesays | I was having trouble with that on my i386 machine, kept getting transmission loop errors, with agx addons 1.4.3 and spandsp 0.0.4pre16 |
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| 13:07.59 | lirakis_work | does anyone know if you can get the saydigits or similar functions to work in spanish or another language? |
| 13:08.29 | hmmhesays | set the LANUAGE variable before you call your app |
| 13:08.34 | lirakis_work | .. or rather.. id like to have them be multi-lingual.. i need both english and spanish.. so i cant just wipe out the sound files |
| 13:08.52 | lirakis_work | hmmhesays: nice thaks |
| 13:08.54 | lirakis_work | *thanks |
| 13:08.58 | hmmhesays | np |
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| 13:14.35 | sergee | xezz: |
| 13:14.38 | sergee | ~book |
| 13:14.47 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 13:17.21 | sergee | xezz: public chat please |
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| 13:21.26 | ferksy | ~fax |
| 13:21.27 | jbot | Well, apperantly the fax was concieved of by Napoleon Bonaparte. He commissioned a system of devices that could transmit a traced image electrically over telegraph lines to a remote device that would redraw the image identically. |
| 13:21.40 | *** join/#asterisk Victor_Yure (n=Victor_Y@unaffiliated/victoryure/x-837844) |
| 13:21.48 | anonymouz666 | hahahaha |
| 13:22.05 | lirakis_work | hmmhesays: do i have to download another soundfile pack to get spanish digit read back? i set properly, but it reads back in english and gives me the following cli warning http://pastebin.ca/886113 |
| 13:23.57 | *** join/#asterisk Modcuts (n=Bob@ppbackup.gotadsl.co.uk) |
| 13:24.39 | Modcuts | tzafrir:when i try to load it says it's "load_resource: Module 'cdr_addon_mysql.so' already exists" but when i got to reload it doesn't exist...(when you tab complete) |
| 13:26.12 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:29.13 | robl^ | hey [TK]D-Fender! |
| 13:29.34 | *** join/#asterisk aleff (n=alef@c9063318.virtua.com.br) |
| 13:30.27 | hmmhesays | lirakis_work, of course |
| 13:30.51 | lirakis_work | hmmhesays: im trying to find out the sound pack to d/l :\ it seems there are a lot or random ones... |
| 13:30.52 | aleff | hi. i'm using users.conf in asterisk 1.4 but it seems to ignore the context entry.. when I call from a sip softphone it always look into the "default" context instead of the context defined in users.conf |
| 13:31.10 | lirakis_work | hmmhesays: but nothing that looks like its from digium |
| 13:32.16 | hmmhesays | http://downloads.digium.com/pub/telephony/sounds/ |
| 13:32.22 | lirakis_work | hmmhesays: when you set the language, does it make asterisk look for a different directory? like es/ or does it look for a different sound file name.. like es-1.gsm |
| 13:32.24 | hmmhesays | links followed straight from asterisk.org |
| 13:32.42 | lirakis_work | hmmhesays: guess i should stay away from voip-info .. wiki atrophy |
| 13:32.46 | aleff | can anybody help me? |
| 13:32.47 | hmmhesays | indeed |
| 13:33.04 | robl^ | I think its time for a NEW wiki |
| 13:33.04 | hmmhesays | obviously it is not matching your peer aleff |
| 13:33.25 | [TK]D-Fender | aleff: Pastebin "sip show peer [yourpeer]", its config, and the complete CLI output of your call from beginning to end with SIP DEBUG enabled. |
| 13:33.27 | [TK]D-Fender | ~pb |
| 13:33.27 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:33.38 | aleff | hmmhesays i enabled sip debug and I always get "Looking for 6878 in default (domain 10.0.0.7) |
| 13:33.51 | aleff | no matter what I put in users.conf as context |
| 13:34.09 | hmmhesays | app_txfax.c:408 txfax_exec: Transmission loop error <-- txfax does not work on any of the machines (2) i've just installed it on |
| 13:34.15 | [TK]D-Fender | aleff: That message doesn't tell us what SIP peer is being matched. Please provide what I have requested |
| 13:34.36 | aleff | ok |
| 13:35.27 | lirakis_work | hmmhesays: so .. ive got the sound pack .. but it has all the exact same file names as whats in my current /var/lib/asterisk/sounds directory... i dont want to overwrite them... |
| 13:35.55 | hmmhesays | i'm guessing you didn't download the es soundpack |
| 13:36.05 | hmmhesays | did you build asterisk from source? 1.2 or 1.4? |
| 13:36.16 | lirakis_work | hmmhesays: asterisk-core-sounds-es-ulaw-current.tar.gz |
| 13:36.22 | lirakis_work | hmmhesays: source.. and this is 1.2 |
| 13:37.12 | hmmhesays | create a folder called ES and put them in there |
| 13:37.27 | hmmhesays | when you set your language variable it will look in the folder you set |
| 13:37.28 | lirakis_work | hmmhesays: okay.. thats what i thought maybe |
| 13:37.54 | hmmhesays | bah on to hylafax and iaxmodem, argh |
| 13:38.00 | lirakis_work | hmmhesays: thanks for your help .. ill work on it |
| 13:38.31 | hmmhesays | No problem |
| 13:38.35 | tzanger | what's the generally-accepted good ATA for 1fxs/1fxo or 2fxs/1fxo? Is there a sipura device to handle that? |
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| 13:38.50 | hmmhesays | hey tzanger |
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| 13:39.12 | tzanger | I'd love a router that had wifi, 1-2fxs, 1 fxo and ran linux (like a wrt54+voip) but no idea if such a beast exists |
| 13:39.15 | [TK]D-Fender | tzanger: SPA-3102 |
| 13:39.23 | tzanger | 3102 eh? I'll go look that up |
| 13:39.27 | tzanger | [TK]D-Fender: whereabouts in montreal are you? |
| 13:39.34 | coppice | hmmhesays: hylafax and iaxmodem aren't going to run on your blackfin |
| 13:39.41 | [TK]D-Fender | tzanger: Work in St-Laurent, live in Pierrefonds |
| 13:39.49 | hmmhesays | I wasn't going to try hylafax |
| 13:39.54 | tzanger | [TK]D-Fender: heh that didn't help me at all... I'll be working in Verdun |
| 13:39.57 | hmmhesays | not on the blackfin itself |
| 13:39.57 | tzanger | coppice: good morning |
| 13:40.11 | aleff | [TK]D-Fender http://pastebin.ca/886131 |
| 13:40.14 | aleff | here it is |
| 13:40.15 | [TK]D-Fender | tzanger: Verdun is about a 20 min drive |
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| 13:40.33 | tzanger | [TK]D-Fender: ahh okay that's not so bad. I won't have a car but we should meet up for a drink next week |
| 13:41.20 | tzanger | [TK]D-Fender: any idea what a cab costs from the airport to verdun area? I ain't paying $200+ for a rental for a week |
| 13:41.24 | hmmhesays | coppice I'm kind of stuck here, I can't find any info about the app_txfax.c:408 txfax_exec: Transmission loop error |
| 13:41.43 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 13:41.47 | [TK]D-Fender | tzanger: Depends how stuck in traffic you're going to be. |
| 13:41.50 | hmmhesays | its doing the same thing on my i386 machines as the blackfin |
| 13:42.09 | [TK]D-Fender | tzanger: And Hwy 20 can get bad at prime time |
| 13:42.23 | tzanger | good to know |
| 13:42.48 | hmmhesays | aleff, Found no matching peer or user for '10.0.5.16:5072' |
| 13:43.20 | tzanger | nice, 3102 is 2FXS+1FXO, perfecto. |
| 13:43.24 | aleff | that's the problem.. when I uncomment the [default] context everything works... |
| 13:43.45 | aleff | i already commented that in sip.conf |
| 13:43.51 | aleff | reloaded, restarted |
| 13:44.00 | tzanger | coppice: I'm finding that I have to learn some DSP theory in order to do tone generation on blackfin... I'm looking at indications.conf as a guide but I'm wondering if you have something a little more... approachable? |
| 13:44.29 | [TK]D-Fender | tzanger: 3102 = 1FXS, 1 FXO |
| 13:44.36 | [TK]D-Fender | tzanger: & router |
| 13:44.49 | hmmhesays | try changing your type = user once and see if that works |
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| 13:45.19 | aleff | hm |
| 13:45.20 | aleff | ok |
| 13:45.28 | tzanger | [TK]D-Fender: 3102 datasheet says "2 service lines, 1 FXO connection" |
| 13:45.35 | hmmhesays | aleff, if it does that will probably break something else |
| 13:45.51 | cpm | tzanger, dave rowetel sells turn key ata/blackfin boxes to play with. |
| 13:46.02 | tzanger | cpm: yeah I'm familliar with those |
| 13:46.05 | cpm | k |
| 13:46.07 | [TK]D-Fender | tzanger: that means 2 Reg's IIRC. Its got 1 FXS |
| 13:46.18 | aleff | hmmhesays actually the entries in users.conf was created by the gui.. there's no type= |
| 13:46.22 | aleff | im gonna try anyway |
| 13:46.23 | tzanger | cpm: I have a blackfin dev board I write drivers for high-density telephony on it |
| 13:46.26 | tzanger | [TK]D-Fender: ahh okay |
| 13:46.28 | [TK]D-Fender | tzanger: Ot implying call-waiting for a 2nd call |
| 13:46.38 | cpm | cool |
| 13:46.38 | tzanger | [TK]D-Fender: gotcha... misleading datahseet :-) |
| 13:46.46 | coppice | tzanger: DDS is probably the best way to generate tones in that sort of environment |
| 13:46.48 | hmmhesays | aleff, no idea then, asterisk isn't finding a user to match a conext to so it is going into the default context |
| 13:47.35 | aleff | hmmhesays but how can it allow me to register? heh |
| 13:47.41 | aleff | with the password etc |
| 13:47.52 | aleff | users.conf seems buggy. |
| 13:47.56 | hmmhesays | because a register is not an invite |
| 13:48.17 | drmessano | If I have a provider sending me calls, and my debug is showing the TO as To: <sip:17475551212@63.80.235.198:5060>;tag=as77fe963b |
| 13:48.26 | hmmhesays | so coppice basically what you are saying is I'm out of luck? |
| 13:48.29 | drmessano | 63.blah being THEIR proxy |
| 13:48.50 | [TK]D-Fender | aleff: What exactly are you entering into Ekiga's Dial line? |
| 13:48.53 | drmessano | How do I get * to not ignore them? |
| 13:49.02 | aleff | [TK]D-Fender sip:6878 |
| 13:49.10 | [TK]D-Fender | aleff: Try removing the prefix |
| 13:49.42 | hmmhesays | my ekiga does not like that |
| 13:50.08 | aleff | [TK]D-Fender same thing. it only works when I have a [default] context |
| 13:50.28 | drmessano | That seems to be the issue with Gizmo.. I see a lot of my other peers sending me calls to did@myip or did@myhostname, but Gizmo sends them to did@THEIRproxy |
| 13:50.34 | aleff | if I include = internal into [default] it works |
| 13:50.39 | drmessano | Wondering if its that simple |
| 13:50.56 | coppice | hmmhesays: do you know where that transmission loop error comes from? it isn't from spandsp. does it come from app_rxfax? |
| 13:51.01 | aleff | but then I wont be able to use my dial plans while creating users through the gui |
| 13:51.25 | hmmhesays | coppice, this comes on the transmit side only |
| 13:51.27 | *** join/#asterisk ifnotwhynot (n=davidh@c1-211-5.rrba.isadsl.co.za) |
| 13:51.35 | ifnotwhynot | hi there |
| 13:51.36 | [TK]D-Fender | aleff: Try manually commenting out your users.conf entry and make a sip.conf one and see how that fares |
| 13:51.37 | hmmhesays | I can paste you everything that outputs on the console |
| 13:51.46 | coppice | ok, then. it is in app_txfax? |
| 13:51.51 | aleff | [TK]D-Fender works fine. :) |
| 13:51.54 | aleff | i tried that |
| 13:52.09 | hmmhesays | seems to be |
| 13:52.10 | [TK]D-Fender | alexcf: Time for my regularly scheduled.... |
| 13:52.10 | aleff | the problem is that users.conf seems to ignore the context entry |
| 13:52.10 | hmmhesays | [Jan 31 13:06:16] WARNING[771]: app_txfax.c:408 txfax_exec: Transmission loop error |
| 13:52.22 | [TK]D-Fender | users.conf = flaming pile of shit/ |
| 13:52.34 | hmmhesays | as for app_rxfax it always fails with the same reason |
| 13:52.49 | ifnotwhynot | aleff are u trying to receive o rtransmit faxes? |
| 13:53.02 | *** join/#asterisk codejunky (n=jan@cl-2718.ham-01.de.sixxs.net) |
| 13:53.04 | ifnotwhynot | sorry got the tail end of conversation |
| 13:53.16 | aleff | ifnotwhynot there's nothing related to fax :p |
| 13:53.34 | aleff | [TK]D-Fender well, the problem is that the gui only manage users.conf |
| 13:53.49 | drmessano | Guess i'll keep looking |
| 13:53.50 | hmmhesays | don't user the gui? or try with a different client |
| 13:54.29 | aleff | but what hmmhesays said makes sense |
| 13:54.35 | ifnotwhynot | whats the problem? drmessano? |
| 13:54.37 | [TK]D-Fender | alexcf: Yes, definitely try another client |
| 13:55.01 | aleff | it's probably unable to find the user when it's in users.conf, and sip calls go into the [default] context |
| 13:55.15 | hmmhesays | aleff, no probably there, that is what is happening |
| 13:55.23 | drmessano | Not recieving calls from Gizmo.. I noticed to SIP TO is mydid@theirproxy vs the expected mydid@myip .. I suspect that's the problem |
| 13:55.43 | aleff | the strange thing is that I can register myself when the user entry is in users.conf but not in sip.conf |
| 13:55.45 | drmessano | Not sure what needs to be relaxed for * to not ignore it |
| 13:56.02 | aleff | but when I call, it doesn't find the user in sip.conf and place the call into the [default] context |
| 13:56.06 | aleff | too buggy. |
| 13:56.13 | hmmhesays | drmessano, need a sip debug to tell you that |
| 13:56.16 | lirakis_work | hmmhesays: .. hmm extracted eveything into /var/lib/asterisk/sounds/ES and did a reload but i get the same warning about sound file not existing |
| 13:56.44 | drmessano | I just told you whats in it.. Everything else is normal except the sip to |
| 13:56.49 | drmessano | I pasted it up above |
| 13:57.22 | tzanger | [TK]D-Fender: wrtp54g looks like it might be closer to what I'm after (sans FXO though) |
| 13:57.57 | hmmhesays | in 1.4 --> Set(CHANNEL(language)=ES) |
| 13:58.07 | hmmhesays | before you call whatever you're using to playback the audio |
| 13:58.10 | [TK]D-Fender | tzanger: Well I guess it depends what you need. |
| 13:58.16 | tzanger | yeah I'm kind of torn :-) |
| 13:58.23 | tzanger | I'm getting rid of the home * box I think |
| 13:58.40 | hmmhesays | tzanger, use one of your blackfins :D |
| 13:59.03 | tzanger | hmmhesays: the thought crossed my mind |
| 13:59.24 | drmessano | http://www.pastebin.ca/886143 |
| 13:59.31 | drmessano | Thats the relevant bit |
| 13:59.36 | ifnotwhynot | drmessano: do you use sip to register with thme? |
| 13:59.42 | lirakis_work | hmmhesays: i do set that... "$agi->exec("SET LANGUAGE()=$lang");" and it seems to take as shown in the cli paste.. http://pastebin.ca/886113 |
| 14:00.04 | lirakis_work | hmmhesays: it has to be caps though? .. maybe thats it |
| 14:00.18 | hmmhesays | yes file name are case sensitive |
| 14:00.43 | drmessano | It's registered fine, and I can make calls |
| 14:01.00 | ifnotwhynot | lirakis_work pastebin link not working |
| 14:01.22 | drmessano | Obviously it's sending me something on inbound |
| 14:01.47 | hmmhesays | ifnotwhynot, it worked for me, he was setting LANGUAGE to something invalid |
| 14:01.51 | drmessano | Nothing set up different than I haven't set up for 20 other sip peers/providers |
| 14:01.52 | ifnotwhynot | lirakis_work: so you have problems with incoming? |
| 14:01.59 | drmessano | jblack is having the same issue |
| 14:02.32 | ifnotwhynot | on your register string are you pointing it to an extension? |
| 14:02.37 | drmessano | I've been tearing through debug and noticed that all the other inbound calls have some form of mydid@myip .. |
| 14:02.44 | lirakis_work | hmmhesays ifnotwhynot: same issue ... http://pastebin.ca/886146 ifnotwhynot: im trying to get spanish playback of digits |
| 14:02.51 | drmessano | I've tried it both ways |
| 14:03.06 | lirakis_work | hmmhesays ifnotwhynot: it still says them in english.. and gives me that warning |
| 14:03.44 | hmmhesays | are you sure the file is in the ~asterisk/sounds/ES directory? |
| 14:03.46 | ifnotwhynot | did you load the spanish sound adon? |
| 14:03.57 | [TK]D-Fender | lirakis_work: the path you wrote before showed "ES " and not"es", also you need a "digits/es" as well |
| 14:03.59 | hmmhesays | ifnotwhynot, he's in 1.2 |
| 14:04.06 | drmessano | I can load the config in a PAP2 and it works great |
| 14:04.13 | drmessano | Nothing special at all |
| 14:04.14 | lirakis_work | [TK]D-Fender: i changed it |
| 14:04.19 | hmmhesays | [TK]D-Fender, its not ES/digits? |
| 14:04.26 | [TK]D-Fender | hmmhesays: Correct |
| 14:04.51 | hmmhesays | [TK]D-Fender, I thought all the file paths were the same except one level deeper |
| 14:05.00 | lirakis_work | [TK]D-Fender: [root@pbx ES]# pwd /var/lib/asterisk/sounds/ES |
| 14:05.07 | [TK]D-Fender | *'s standard language layout : (place * normally looks)/lang. |
| 14:05.11 | drmessano | Outgoing works.. and no matter how stupidly simple or relaxed I allow inbound, it still burps |
| 14:05.28 | hmmhesays | drmessano, you haven't posted any relevant debugging information |
| 14:05.44 | [TK]D-Fender | lirakis_work: /var/lib/asterisk/sounds/es @ /var/lib/asterisk/sounds/digits/es |
| 14:05.47 | drmessano | What i posted wasnt relevant? |
| 14:05.56 | lirakis_work | ifnotwhynot: i installed asterisk-core-sounds-es-ulaw-current.tar.gz into /var/lib/asterisk/sounds/ES |
| 14:06.01 | [TK]D-Fender | lirakis_work: not /var/lib/asterisk/sounds/es/digits |
| 14:06.14 | lirakis_work | [TK]D-Fender: hmm .. okay |
| 14:06.41 | lirakis_work | [TK]D-Fender: i got a whole core sounds language pack though.. i guess.. i have to move each "chunk" into its own ES subdirectory |
| 14:06.52 | lirakis_work | [TK]D-Fender: like digits/ES |
| 14:06.53 | hmmhesays | that is a stupid way of doing things |
| 14:07.18 | hmmhesays | they should have just put the whole pack one level deeper |
| 14:07.45 | ifnotwhynot | drmessano: try something like this seems to work for me register => username:password@sipprovider.com/1000 meaning it will register me whith their server and allow incoming calls to my registered account to point to exten 1000======do this for your extension.cong exten => 1000,1,Playback(vm-goodbye) to test |
| 14:08.01 | drmessano | tried all that |
| 14:08.02 | [TK]D-Fender | hmmhesays: Agreed. Then again, so is assuming english in the first place. the dir structure should be completely uniform and separate and english itself in a dedicated folder jsut like any other. |
| 14:08.29 | *** join/#asterisk mltlnx (n=mltlnx@64.3.170.41.ptr.us.xo.net) |
| 14:08.59 | lirakis_work | [TK]D-Fender: okay .. now i copeid the spanish digits folder to /var/lib/asterisk/sounds/digits/ES |
| 14:09.01 | hmmhesays | [TK]D-Fender, I agree |
| 14:09.01 | ifnotwhynot | past your sip.conf |
| 14:09.02 | lirakis_work | trying... |
| 14:09.28 | lirakis_work | rarg! |
| 14:09.37 | lirakis_work | exact same warning |
| 14:09.50 | lirakis_work | and .. its saying digits in english |
| 14:09.51 | hmmhesays | turn on debugging |
| 14:09.58 | hmmhesays | in logger.conf |
| 14:10.32 | *** join/#asterisk tinloaf (n=tinloaf@d90-134-72-61.cust.tele2.de) |
| 14:11.34 | tinloaf | hey. i think i got something wrong. i want my asterisk to connect mit too "foo@bar.com" via SIP when some phone dials "sfoo@bar.com". is the following extension-line OK? |
| 14:11.35 | lirakis_work | hmmhesays [TK]D-Fender: .. paste with agi debug on http://pastebin.ca/886155 |
| 14:11.42 | tinloaf | exten => _s.,1,Dial(SIP/${EXTEN:2},30) |
| 14:12.15 | hmmhesays | I don't remember if you turn on debugging if it will give you the full file path that it is looking for |
| 14:12.30 | lirakis_work | hmmhesays: okay .. ill turn it on in logger.conf too |
| 14:13.22 | *** join/#asterisk mltlnx (n=mltlnx@64.3.170.41.ptr.us.xo.net) |
| 14:13.46 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:13.49 | [TK]D-Fender | lirakis_work: do you see /var/lib/asterisk/sounds/digits/es/and ? |
| 14:14.24 | *** join/#asterisk mikebee (n=jcrum@66-193-179-217.static.twtelecom.net) |
| 14:14.53 | lirakis_work | [TK]D-Fender: no, that file doesnt exist, but its playing all the other digits in english too :\ |
| 14:15.08 | hmmhesays | oh boy britany has gone crazy again |
| 14:15.10 | hmmhesays | wee haw |
| 14:15.12 | [TK]D-Fender | lirakis_work: did you fix your path to lower-case? |
| 14:15.30 | lirakis_work | [TK]D-Fender: no .. i thought that was arbitrary as long as you are consitent |
| 14:15.42 | [TK]D-Fender | lirakis_work: No, case always matters. |
| 14:16.10 | *** join/#asterisk moellerdk (n=chatzill@4806ds1-hl.0.fullrate.dk) |
| 14:16.48 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:16.58 | lirakis_work | [TK]D-Fender: .. okay.. changing path to es .. |
| 14:17.04 | *** join/#asterisk freezey (n=freezey@gw.mypublisher.com) |
| 14:17.12 | [TK]D-Fender | lirakis_work: paths <- |
| 14:17.20 | hmmhesays | you should be able to use a uppercase path |
| 14:19.05 | *** join/#asterisk macbrody (n=chatzill@217-162-14-126.static.cablecom.ch) |
| 14:19.10 | lirakis_work | [TK]D-Fender: alright.. im getting some where now |
| 14:19.17 | lirakis_work | [TK]D-Fender: thanks |
| 14:19.57 | [TK]D-Fender | hmmhesays: this isn't Windows. You can't get away with that much illiteracy :p |
| 14:20.18 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 14:20.45 | bsdwarrior | how do you read a key press while from a user in a queue ? read() doesnt seem to work |
| 14:20.54 | tzafrir | How can I tell that asterisk has finished loading and registering everything? |
| 14:21.18 | tzafrir | This is something that takes a number of seconds. |
| 14:21.27 | mosty | tzafrir, from a script? |
| 14:21.34 | tzafrir | Currently I need to add 'sleep 5' to one of my scripts |
| 14:21.56 | tzafrir | Otherwise it sometimes gets to asterisk befire asterisk registers some CLI commands |
| 14:22.23 | mosty | tzafrir, how are you reloading? |
| 14:22.29 | defswork | I've been landed with helping a company get an asterisk install working again after they rejigged their network and broke everything - I've got asterisk talking to their sip peer ok and outgoing calls work but no incoming |
| 14:22.47 | defswork | they swear they've not done anything on the firewall - is anything needed ? |
| 14:23.03 | [TK]D-Fender | bsdwarrior: Read the sample queues.conf, its all in there. |
| 14:23.26 | tzafrir | mosty: stopping and later starting |
| 14:23.48 | [TK]D-Fender | tzafrir : I need to add "sleep 10hr" to my life... |
| 14:24.06 | mosty | tzafrir, does asterisk -rx 'reload foo' return once it's finished loading? |
| 14:24.22 | tzafrir | mosty, no. It returns immediately |
| 14:24.27 | freezey | [TK]D-Fender: ok so let me see if i can reword this now =] 5 days later right? haha... so... i currently have an avaya system inplace and what i want to do is replace my avaya system with asterisk... i have the PC i have the system setup.. but now what hardware do i need to make this asterisk system take over avaya? |
| 14:24.29 | *** join/#asterisk lemanal (n=lemanal@cpe-066-026-085-055.nc.res.rr.com) |
| 14:24.58 | tzafrir | But I'm talking about a real stop and a full start |
| 14:25.31 | bsdwarrior | tkd-fender , I fixed that periodic announce btw |
| 14:26.18 | hmmhesays | [TK]D-Fender, haha yeah I'm doing something else at the moment here |
| 14:26.42 | mosty | tzafrir, perhaps there's a manager event? |
| 14:27.11 | tzafrir | yuck. This means I need to actually define a manager user. |
| 14:27.21 | tzafrir | Which makes the scirpt much less generic |
| 14:28.51 | bsdwarrior | tkd-fender there is no mention of how to read I.E. the user pressed 1 while in the queue |
| 14:30.03 | [TK]D-Fender | bsdwarrior : Yes, there is. |
| 14:30.19 | [TK]D-Fender | bsdwarrior: its explicit and I've pointed countless others to it as well. |
| 14:30.49 | moellerdk | Hey everyone! Does anyone know how to originate a call from the CLI? I'd like to call a number and connect it to let's say the MusicOnHold application. |
| 14:31.09 | [TK]D-Fender | freezey: What Exact harware are you looking to interface with * (connection type and models please). |
| 14:31.36 | [TK]D-Fender | moellerdk: Any reason not to jsut use a soft-phone? |
| 14:31.48 | bsdwarrior | tkd-fender , intresting, set context =somecontext |
| 14:32.20 | [TK]D-Fender | bsdwarrior: You'd almost think it was hidden in the big print... |
| 14:32.31 | moellerdk | [TK]D-Fender: Absolutely... I need to originate 30 calls just for a test. |
| 14:32.45 | [TK]D-Fender | moellerdk: Yikes. |
| 14:33.04 | [TK]D-Fender | moellerdk: looking to load-test MoH? |
| 14:33.26 | bsdwarrior | tkd-fender forgive me as a learn. |
| 14:33.34 | davevg-btwtech | tzafrir, maybe "core show uptime" will help you? |
| 14:33.42 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 14:33.49 | moellerdk | [TK]D-Fender: Nope... actually I look to load test another application - MoH was just an example :-) |
| 14:34.16 | tzafrir | I guess I can try a poll loop |
| 14:34.44 | [TK]D-Fender | moellerdk: I can't think of a viable way to do this from CLI itself, but you could use AMI Originate, or Call-files for this. |
| 14:35.43 | tzafrir | But I wonder if there's any wimple thing I could wait on, and that would only return after asterisk has fully initialized |
| 14:36.09 | moellerdk | [TK]D-Fender: Should be quite simple. I just can't figure the parameters out... Otherwise it's just a simple Up-Arrow, Enter, Up-Arrow, Enter thing. I'll originate them all to the same number. |
| 14:36.37 | freezey | [TK]D-Fender: basically i am looking to use asterisk as my main handler for all the extensions and voicemail system etc etc... so i want it to replace the avaya system |
| 14:37.06 | plik | moellerdk: see the book chapert on call files, the just a simple shell script to move it over 30 times |
| 14:37.11 | [TK]D-Fender | freezey: I asked for precise details on the HARDWARE and LINES you are looking to connect to *. |
| 14:37.29 | freezey | ok i will get precise details |
| 14:37.33 | freezey | let me go take a look |
| 14:37.57 | [TK]D-Fender | freezey: Funny thats what I asked for days ago, asked again now, and even had to repeatmyself in this session.... |
| 14:38.49 | freezey | haha sorry man |
| 14:38.53 | davevg-btwtech | tzafrir, can look at the verbose log for "Asterisk Ready"? That is the last thing it says when I restarted my * instance |
| 14:39.18 | tzafrir | davevg-btwtech, unles I catch the previous one |
| 14:39.28 | mpwizard | Hi, how do I match *15# in extensions.conf? |
| 14:39.58 | davevg-btwtech | tzafrir, it is timestamped ie "[Jan 31 09:37:41] VERBOSE[6290] logger.c: Asterisk Ready." |
| 14:40.07 | Qwell | tzafrir: how about an option to asterisk -r which blocks until asterisk is finished loading? |
| 14:40.17 | moellerdk | plik: Hmm... So absolutely nobody knows the syntax of the originate command in the CLI? |
| 14:40.27 | tzafrir | Qwell, that would be very nice |
| 14:40.28 | Qwell | moellerdk: help originate |
| 14:40.31 | [TK]D-Fender | mpwizard: exten *15#,1,NoOp(Yippy-kai-yay....) |
| 14:40.38 | Qwell | tzafrir: shouldn't be hard to add, I wouldn't think |
| 14:40.39 | [TK]D-Fender | mpwizard: exten => *15#,1,NoOp(Yippy-kai-yay....) |
| 14:41.01 | moellerdk | Qwell: Sure... I know that. But I can't make it work :-) |
| 14:41.17 | plik | well somebody must, but I've only built my frst one a couple of weeks ago and am still a newbie, so I was just offering a workable solution :) |
| 14:41.57 | [TK]D-Fender | moellerdk: "help originate" |
| 14:42.02 | plik | a script to place 30 calls is far more elegant than "up arrow & enter" again and agaain |
| 14:42.09 | mpwizard | [TK]D-Fender: Tried that, it does not match... Getting 404. |
| 14:42.17 | *** join/#asterisk techie (n=techie@adsl-76-214-8-235.dsl.lsan03.sbcglobal.net) |
| 14:42.22 | [TK]D-Fender | mpwizard: then your phone isn't sending what you think it is. |
| 14:42.26 | *** part/#asterisk techie (n=techie@adsl-76-214-8-235.dsl.lsan03.sbcglobal.net) |
| 14:42.34 | [TK]D-Fender | mpwizard: Pastebin & sip debug are your friends. |
| 14:42.37 | moellerdk | plik: Come on... It's a one time thing right now :-) |
| 14:43.20 | plik | its 2 lines of bash and a call file that you can prety much copy and paste & just change the exten no |
| 14:43.34 | plik | I know which I'd find quickest |
| 14:44.26 | mpwizard | [TK]D-Fender: :-) sip debug says Looking for *15# in default ... I will paste the sip debug at pastebin. |
| 14:44.40 | *** join/#asterisk cjk (n=ldidelot@195.26.5.145) |
| 14:45.07 | [TK]D-Fender | mpwizard: and your dialplan, and "dialplan show default" |
| 14:46.18 | mpwizard | [TK]D-Fender: Okidoki... |
| 14:46.26 | plik | you could even just copy the call file to your system and do mv <file> /var/spool/asterisk/outgoing and up arrow that 30 times if you're that keen on human interaction |
| 14:47.16 | cjk | hi, i do the following, dial(SIP/USER); Answer(), dial(SIP/User) but with musiconhold.... problem is that the my user gets 2 missed calls, or 2 popups on the softphone or when he refuses the call the first time, the call comes back. any idea, hint or solution? |
| 14:47.19 | mosty | i'm having some trouble with codec negotiation between asterisk 1.2.13 and 1.4.17 using iax, the user and peer both only allow ulaw, what could be wrong? http://paste.debian.net/48126 |
| 14:47.24 | plik | theres a sample.call file in the source, as well as an example in the book, so whats the problem |
| 14:47.48 | mosty | cjk, answer the first time |
| 14:48.18 | mosty | cjk, if you dial twice, and they don't answer, that is two missed calls |
| 14:48.33 | [TK]D-Fender | mosty: Next time -< IAX2 DEBUG |
| 14:49.48 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:49.48 | cjk | mosty: well the person that calls my user should first hear a normal ring tone, and then musiconhold, how can i do this within one call? |
| 14:50.17 | mosty | [TK]D-Fender, sure, one moment |
| 14:50.30 | mosty | cjk, put some ringing sound at the start of your music? |
| 14:50.41 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
| 14:50.56 | cjk | mosty: well yeah, good joke |
| 14:51.30 | [TK]D-Fender | cjk: wHY ARE YOU CALLING TWICE? |
| 14:51.36 | [TK]D-Fender | darn caps... |
| 14:51.49 | mosty | cjk, if you dial twice, it's two calls, simple as that |
| 14:52.18 | cjk | [TK]D-Fender: i do not want to answer the call immediatley as it costs money to the calling person..... but at some point after 30 seconds i want to answer it and play some MOH. |
| 14:52.21 | *** join/#asterisk RoyK (n=roy@ip-58-8-149-91.dialup.ice.no) |
| 14:52.44 | cjk | so i need a "modifed" dial application that can first do normal rinong and then siwthc to moh without calling my sip user twice |
| 14:53.01 | [TK]D-Fender | cjk: No way to do that while maintaining the initial call. |
| 14:53.18 | [TK]D-Fender | cjk: Live with this in some other way becasue * doesn't off what you're asking for. |
| 14:53.51 | cjk | [TK]D-Fender: well but you confirm that its technically possible if i modify the DIAL application? |
| 14:53.52 | plik | oh, perhaps I misunderstood then |
| 14:54.22 | [TK]D-Fender | cjk: WELL if you want to change the source, SURE, you could invent your own app to do it, but this is extreme |
| 14:54.48 | [TK]D-Fender | cjk: And you're more than welcome to go for it. |
| 14:55.07 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 14:55.22 | cjk | [TK]D-Fender: i guess :) ok thanks for the hep |
| 14:55.23 | cjk | help |
| 14:56.59 | [TK]D-Fender | less |
| 14:57.09 | [TK]D-Fender | I love these little word association games... |
| 14:57.35 | mosty | [TK]D-Fender, would you mind msg'ing me your private pastebin? the iax2 debug info has some personal info in it |
| 14:58.19 | mpwizard | Has anyone experienced choppy sound when using background to play an alaw file bigger than 16kb? Using 1.2.24 |
| 15:01.13 | mikebee | Anyone around that can answer a question about codec_g729a.so and compatibility for Asterisk 1.4.17? |
| 15:03.36 | jake[work] | rtcp doesn't seem to return any stats - all zeros... any trick to make this work? |
| 15:03.42 | mosty | mikebee, don't ask to ask, just ask |
| 15:04.30 | *** join/#asterisk mltlnx (n=mltlnx@rrcs-24-105-157-191.nyc.biz.rr.com) |
| 15:04.47 | *** join/#asterisk hendrixski (n=hendrixs@cpe-74-65-1-222.rochester.res.rr.com) |
| 15:06.03 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:06.24 | agx | anyone have experience of connecting asterisk to an Ericcson PaBX via H323? |
| 15:06.52 | mikebee | mosty: heh, sorry. I'm using version 33 of codec_g729a.so from Digium, which is the newest I can find on their site anyway. I'm getting the following when I load it with 1.4.17 |
| 15:07.14 | mikebee | "[Jan 29 09:41:45] WARNING[13132]: loader.c:620 inspect_module: Module 'codec_g729a.so' was not compiled against a recent version of Asterisk and may cause instability." |
| 15:07.30 | mosty | mikebee, yeah i get that too, it's just a warning |
| 15:07.30 | mikebee | "[Jan 29 09:41:45] WARNING[13132]: loader.c:620 inspect_module: Module 'codec_g729a.so' was not compiled against a recent version of Asterisk and may cause instability." |
| 15:07.38 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:07.57 | mikebee | the problem is, I think it actually is causing instability |
| 15:08.40 | file | [TK]D-Fender: I just want... |
| 15:08.49 | mikebee | 1.4.17 was dying on occasion |
| 15:09.16 | mosty | mikebee, how much load was asterisk under? |
| 15:09.59 | mikebee | so I moved back to 1.4.15 with no problems. The reason I think the g729 codec was the problem is once I had to kill -9 asterisk, I tried an immediate restart, and it hung on loading modules.conf |
| 15:10.17 | mikebee | around 4 calls were going on at the time |
| 15:10.39 | mikebee | so not much (3Ghz Xeon, 5GB RAM) |
| 15:10.57 | mosty | 4 calls, that should be easy to reproduce on a test box, perhaps you should do so and submit a bug report |
| 15:11.17 | file | mikebee: Please contact Digium support |
| 15:12.32 | mikebee | I might try to replicate it with astertest or something. It doesn't happen very often, probably once or twice a day |
| 15:12.42 | *** part/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 15:12.51 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 15:12.51 | *** mode/#asterisk [+o file] by ChanServ |
| 15:13.25 | mosty | file, i have an asterisk 1.2 box with multiple iax accounts setup to make calls to a single asterisk 1.4 box, and asterisk 1.2 seems to be sending the wrong username, do you know why that might be? |
| 15:13.48 | file | without seeing an iax2 debug and console output I can't answer that |
| 15:14.00 | mikebee | file: you mean contact them thru bugs.digium.com? I didn't have much luck finding their contact info on www.digium.com |
| 15:14.34 | file | mikebee: http://www.digium.com/en/company/contact.php |
| 15:14.38 | mosty | file, mind if i /msg you a url? the log has personal info in it |
| 15:14.50 | file | mosty: fine |
| 15:15.34 | mikebee | thanks guys |
| 15:15.40 | file | mosty: you misspelled username |
| 15:15.48 | file | you put "usename" instead |
| 15:15.58 | mikebee | I'll see if I can open a bug and also contact digium on this issue. |
| 15:16.11 | mosty | file, argh! thanks |
| 15:16.19 | file | mikebee: please don't open one on Mantis... it's not the place for it |
| 15:16.32 | file | (if it's an issue with the g729 codec) |
| 15:17.22 | mikebee | file: really? k. I'll work it with just Digium then |
| 15:27.14 | ifnotwhynot | hi there need to know how application musicon hold applies to inclomming calls, i see u use setmusiconhold(default) on incoming of first extension how does asterisk know to use musiconhold when transfering a call or call hold is there other instanses where one can link to using the musiconhold feature, on legacy pbx's we have obe moh source(being a pc playing prerecorded scripts or a standalone cd player looping a sound file) i dont quite |
| 15:27.22 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:28.04 | *** join/#asterisk mltlnx (n=mltlnx@rrcs-24-105-157-191.nyc.biz.rr.com) |
| 15:31.03 | ifnotwhynot | my typing is terrible got used to the mouse on windows, new learning curve on asterisk and linux, no wonder my configs file never seem to work the first time! |
| 15:31.27 | ifnotwhynot | is this chanel active? |
| 15:32.31 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:32.47 | mosty | ifnotwhynot, http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 15:33.56 | *** join/#asterisk ManxPower (n=manxpowe@19.sub-75-202-0.myvzw.com) |
| 15:35.42 | *** join/#asterisk Victor_Yure (n=Victor_Y@unaffiliated/victoryure/x-837844) |
| 15:35.54 | ifnotwhynot | setins:11,2,5,demon,0010110,name_&:user_6007, {demon_func??_&&restart safe_asterisk//??config_extension,g729open{func_load:/?g729_for_all_channels/?ZAP/0-60} |
| 15:36.04 | ifnotwhynot | sorry wrong screen |
| 15:36.12 | ifnotwhynot | thx mosty |
| 15:41.42 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 15:43.02 | jameswf | boing |
| 15:43.25 | Weetos | boing too |
| 16:04.38 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:04.44 | *** join/#asterisk truz_`24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com) |
| 16:05.40 | jameswf | was that a tumble weed? |
| 16:06.17 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 16:07.08 | bsdwarrior | once I sent a call with the manager, how can I lookup the status of the call in cdr ? |
| 16:07.53 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 16:09.24 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 16:10.02 | bsdwarrior | can I set the userfield to be a unique string then look it up by that later ? |
| 16:10.07 | waKKu | hi... if someone can help me, see: http://pastebin.ca/886298 - I'm trying to use Pickup Group with IAX at asterisk 1.4.15 ... with no sucess ... any idea ? |
| 16:11.33 | UnixDog | bsdwarrior, it depends on what crd interface you use |
| 16:11.44 | UnixDog | are you using a sql interface ? |
| 16:13.56 | bsdwarrior | cdr is in postgres, yes |
| 16:14.09 | bsdwarrior | I dont know any other way to find out if the call was completed ,etc. |
| 16:15.53 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:15.53 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:15.53 | [TK]D-Fender | bsdwarrior: Set a unique variable in your Originate, then scan active channels for that var. |
| 16:16.20 | *** join/#asterisk b11d (n=no@234-200-29-134.hcc.mnscu.edu) |
| 16:16.25 | b11d | hello lads |
| 16:16.42 | *** join/#asterisk marloshouse (n=markpetr@pool-96-226-12-133.dllstx.fios.verizon.net) |
| 16:17.16 | [TK]D-Fender | bsdwarrior: And CDR will tell you only when its actually finished |
| 16:17.23 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:17.27 | b11d | CunningPike! |
| 16:17.28 | b11d | TK! |
| 16:17.33 | b11d | bonjour lads! |
| 16:17.40 | marloshouse | i am trying to work with my digital receptionist to re-record some of my prompts and it's not working. I dial *77 to make a recording and then hang up. Then I hit *99 and it just plays a "dong" and I don't hear anything. Am I missing a procedure? |
| 16:17.55 | *** part/#asterisk corramor (n=corramor@216.207.245.1) |
| 16:18.21 | bsdwarrior | tkd-fender, thanks, what do you mean by scan the active channels? will I see the var with show channels ? |
| 16:18.56 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
| 16:19.02 | lmadsen | b11d: bonjour mes amis |
| 16:19.08 | ZaVoid | morning |
| 16:19.14 | b11d | haha! |
| 16:19.30 | freezey | [TK]D-Fender: ok so sorry took so long but ok... so back there i have a Optimum Light Path Patch Panel... an IP400 Digiatl Station and a Phone station and a IP 406 Office |
| 16:21.54 | [TK]D-Fender | freezey: I asked what kind of connection your equipment and lines used. |
| 16:22.11 | jameswf | marloshouse: what distro are you using |
| 16:22.19 | freezey | [TK]D-Fender: t1 |
| 16:22.20 | [TK]D-Fender | bsdwarrior: Yes you can drill a single channel and see its vars. |
| 16:22.27 | *** join/#asterisk MindTheGap (n=MindTheG@c9503f78.bhz.virtua.com.br) |
| 16:22.33 | marloshouse | jameswf, it's a very old asterisk-at-home |
| 16:22.37 | b11d | anyone here running Asterisk 1.4.x and a Rhino CB box sucessfully? |
| 16:23.41 | jameswf | b11d: Yes |
| 16:24.02 | b11d | |I get a dialtone on each channel, but I cant call channel to channel. I get dead silence.. any tips? |
| 16:24.17 | b11d | asterisk shows the channels are bridged, but no audio is passed. |
| 16:24.22 | marloshouse | jameswf, Asterisk CVS-v1-0-03/11/05-09:00:15, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <markster@digium.com> |
| 16:24.41 | marloshouse | i'm just wondering if i'm missing some obvious "im done recording my prompt" button |
| 16:24.46 | marloshouse | rather than just hanging up |
| 16:25.21 | [TK]D-Fender | freezey: What phone model? And you're using a T1 to the Telco? |
| 16:25.42 | jameswf | the channelbank is just a modem (it modulaes and demodulates analog to digital) there is likely an issue with your asterisk config, you can do an auto t1 to ensure the hannelbank and asterisk match up but if you have dial tone and can break dial tone it is probably good |
| 16:26.05 | *** join/#asterisk tristanbob (n=tristanr@oalug/member/tristanbob) |
| 16:26.16 | jameswf | jbot: tell marloshouse about trixbox |
| 16:26.16 | *** join/#asterisk corramor (n=corramor@216.207.245.1) |
| 16:26.57 | marloshouse | jameswf, thank you |
| 16:27.03 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 16:27.07 | [TK]D-Fender | marloshouse: Its your dialplan.... Did you add a playback for your recording extension? |
| 16:27.21 | UnixDog | whats a dial plan |
| 16:27.24 | freezey | [TK]D-Fender: yeah the t1 comes in through the boxes that verizon put in with those cards in them... and how do i find out what model? |
| 16:27.28 | UnixDog | where can I find the dial plan |
| 16:27.28 | [TK]D-Fender | marloshouse: Ah, A@H... nevermind. |
| 16:27.44 | UnixDog | when is atserisk going to have a full featured dial plan |
| 16:28.07 | jameswf | UnixDog: You have to buy one if you put money in my paypal Ill sell you one |
| 16:28.08 | UnixDog | where do I find the full vertical dp |
| 16:28.19 | marloshouse | [TK]D-Fender, what? |
| 16:28.22 | UnixDog | lol |
| 16:28.23 | b11d | jameswf.. yeah.. I dont know what could be wrong with my asterisk config though. Everything looks good to me.. |
| 16:28.42 | jameswf | b11d: watch the cli |
| 16:28.53 | UnixDog | asterisk -vvvvvvvvvvgr |
| 16:28.54 | b11d | I am.. it all looks good. |
| 16:28.57 | UnixDog | watch the output |
| 16:29.00 | b11d | i have debug and verbose set.. |
| 16:29.04 | UnixDog | watch to see where it fails |
| 16:29.32 | UnixDog | /var/log/asterisk/messages |
| 16:29.41 | UnixDog | see it it has any errors in it |
| 16:30.05 | outtolunc | grr |
| 16:30.09 | waKKu | does someone that can help me with pickupgroup, see: http://pastebin.ca/886298 - I'm trying to use Pickup Group with IAX at asterisk 1.4.15 ... with no sucess ... any idea ? |
| 16:30.21 | jameswf | ~ron paul |
| 16:30.22 | jbot | ZOMG RONPAULAPPLEUBUNTU FOR PRESIDENT |
| 16:30.44 | b11d | no errors in messages for me.. |
| 16:30.53 | *** join/#asterisk weazahl (n=jeremy@adsl-76-230-116-17.dsl.ksc2mo.sbcglobal.net) |
| 16:30.59 | plik | his luch will probably get soggy in the shower... |
| 16:31.00 | jameswf | did you try an autoT1 |
| 16:31.19 | b11d | yeah a few times.. |
| 16:31.35 | b11d | I can sucessfully call from one Zap channel to another on the same CB, but just no audio.. :( |
| 16:31.37 | UnixDog | what type of channelbank ? |
| 16:31.42 | b11d | that tells me the T1 works.. |
| 16:31.46 | b11d | its a CB-24, all FXS. |
| 16:31.50 | jameswf | UnixDog: its ours |
| 16:32.08 | bsdwarrior | Set(CDR(userfield)=1234) doesnt work |
| 16:32.12 | jameswf | What is your span line |
| 16:32.15 | UnixDog | since when does digium have a cb |
| 16:32.25 | b11d | span=2,0,0,esf,b8zs |
| 16:32.26 | jameswf | << is Rhino not Digium |
| 16:32.33 | b11d | also tried span=2,1,0,esf,b8zs |
| 16:32.34 | UnixDog | rhino |
| 16:32.36 | UnixDog | ok |
| 16:32.41 | jameswf | for giggles try 2,1,0 |
| 16:32.45 | b11d | I did.. |
| 16:32.47 | jameswf | doh |
| 16:33.02 | UnixDog | the company who after 9 months still has no bsd driver |
| 16:33.05 | [TK]D-Fender | bsdwarrior: I said set a VARIABLE. CDR doesn't get committed until the call has ended. |
| 16:33.08 | marloshouse | jameswf, i have been wanting to upgrade this thing anyway, and asterisk-at-home was just easy at that time - what would you recommend now? |
| 16:33.19 | jameswf | Elastix |
| 16:33.25 | [TK]D-Fender | bsdwarrior: So set a CHANNEL VARIABLE at the same time you set the userfield. |
| 16:34.03 | b11d | ztmonitor shows the audio being sent but no incoming audio.. which matches up with the problem im seeing. |
| 16:34.24 | jameswf | We have BSD drivers for a few products but we release based on quality not peer preasure :) |
| 16:34.27 | [TK]D-Fender | freezey: You never answered what phone model..... |
| 16:34.33 | UnixDog | ok and your only using g711/ulaw and gsm for soundfile transcoding |
| 16:34.48 | [TK]D-Fender | jameswf: Or the lack of peers in your bracket ;) |
| 16:34.51 | UnixDog | your not trying to use any other codec |
| 16:35.25 | *** join/#asterisk joelsolanki (i=joelsola@220.224.103.226) |
| 16:35.38 | b11d | I am not setting any codec anywhere.. so i assume g711 |
| 16:35.44 | UnixDog | and watch's [TK]D-Fender slip slide away |
| 16:35.52 | jameswf | If you have ringback the audio path is there |
| 16:35.59 | weazahl | will a dual xeon 2GHz w/ 2GB happily handle 40 users? 16 trunks total? i imagine so. |
| 16:36.15 | jameswf | can you call fxs to trunk.... |
| 16:36.19 | UnixDog | weazahl, thats over kill but yes |
| 16:36.23 | b11d | yeah.. as soon as I see "native bridgeing Zap/28-1 and Zap/30-1" it is silent.. |
| 16:36.35 | jameswf | lubricant and carpet dont mix... ask your mom :) |
| 16:36.48 | weazahl | thats what i though. about how much would that handle? |
| 16:36.48 | b11d | this is a test, closed system.. its just asterisk -> CB with two phones on the CB. |
| 16:36.59 | [TK]D-Fender | weazahl: How unqualified... you haven't meantioned any of the things that actually place a load or or described your 'trunks", etc |
| 16:37.07 | UnixDog | crisco and carpet still make for a slipper setup |
| 16:37.59 | *** join/#asterisk teknoprep (n=chrisraw@unaffiliated/teknoprep) |
| 16:38.14 | b11d | i was going to assign a SIP phone to this test system to see if SIP <-> analog worked but havent yet |
| 16:38.21 | jameswf | http://xkcd.com/330/ << Crisco |
| 16:38.27 | weazahl | all digital termination. using an external multiport ata. converting ulaw to gsm. |
| 16:38.38 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 16:38.50 | [TK]D-Fender | weazahl: That description is just plain broken... try again. |
| 16:39.22 | jameswf | b11d pastebin your dial plan |
| 16:39.34 | b11d | ok.. give me a minute. |
| 16:39.45 | weazahl | it was a general question. not a request for a quote or anything. lighten up |
| 16:40.08 | UnixDog | thats just not right |
| 16:40.13 | jameswf | :) |
| 16:40.36 | jameswf | << avid reader of xkcd |
| 16:40.42 | UnixDog | and how many people at rhino have fallen pray to it |
| 16:41.09 | UnixDog | and how many pieces of equipment ended up in the mix |
| 16:41.34 | b11d | jameswf.. http://pastebin.ca/886349 |
| 16:41.52 | bsdwarrior | tkd-fender this is what im doing, keep in mind this from a php script. http://pastebin.com/d6499a328 |
| 16:43.09 | [TK]D-Fender | weazahl: Well I would like to give you a proper answer to your wanting to validate your system specs. Details are kinda required for that. |
| 16:43.15 | UnixDog | bsdwarrior, are you trying to watch a livecall or pull data from a cdr ? |
| 16:43.22 | [TK]D-Fender | weazahl: Can you elaborate on the link a bit more? |
| 16:43.55 | joelsolanki | Hi room |
| 16:43.59 | joelsolanki | 75.126.46.186 1206315 1b99bf83596 00102/00000 unkn No Tx: ACK |
| 16:44.00 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:44.02 | UnixDog | weazahl, 20 to 40 phones 15 concurrent sip calls to the real world |
| 16:44.03 | [TK]D-Fender | b11d: pastebin your zapata.conf |
| 16:44.07 | joelsolanki | sip show channels shows about output |
| 16:44.07 | b11d | am doing it now TK. |
| 16:44.11 | bsdwarrior | unixdog, im running a php script that gets numbers from the db and calls them via the manager. I want to then lookup if the call is completed in cdr so I can flag my db so they arent called again |
| 16:44.14 | UnixDog | p3 500 256 megs ram |
| 16:44.21 | joelsolanki | this is a stale call |
| 16:44.29 | joelsolanki | call is already disconnected |
| 16:44.45 | weazahl | UnixDog: thats about it |
| 16:45.14 | [TK]D-Fender | bsdwarrior: #8 = You can't set a function like this. #7=not correct syntax. |
| 16:45.22 | b11d | zaptel and zapata: http://pastebin.ca/886357 |
| 16:45.22 | joelsolanki | but still i can see in sip show channels |
| 16:45.22 | joelsolanki | i notice that this type of calls are having wrong cdr duration. |
| 16:45.22 | joelsolanki | means a very high usage in seconds. |
| 16:45.22 | joelsolanki | is that so ? |
| 16:45.48 | joelsolanki | actually i m using asterisk for 1 client for wholesale. and we see for some calls it is billing wrongly. |
| 16:46.04 | UnixDog | weazahl, but ig you need any transcoding other then ulaw/alaw/gsm its going to need more power |
| 16:46.14 | joelsolanki | so we found that in sip show channels calls which are stale is a problem |
| 16:46.23 | UnixDog | if you plan to use g729 then a p3 800 would be needed |
| 16:46.23 | joelsolanki | any experienced notes plz ? |
| 16:46.43 | UnixDog | with atleast 512 egs ram |
| 16:47.03 | UnixDog | but that also depends on if your using a gui or not |
| 16:47.13 | [TK]D-Fender | b11d: You shouldn't have span 1 & 2 both as primary. Also is this only on bridged calls? AKA, SIP>ZAP & ZAP>SIP work? |
| 16:47.24 | UnixDog | and a sql db on the same system |
| 16:47.44 | b11d | havent tried SIP with this yet TK.. just ZAP<->ZAP at this point. |
| 16:47.57 | b11d | was planning on trying that later though |
| 16:48.08 | [TK]D-Fender | b11d: Time to pin it down then. Do separate testing. |
| 16:48.21 | [TK]D-Fender | b11d: That deserves to be now. |
| 16:48.28 | b11d | ok.. I figured :) |
| 16:48.29 | b11d | am on it. |
| 16:48.35 | [TK]D-Fender | b11d: If Audio doesn't work with just, forget bridged. |
| 16:48.48 | jake4pres | can anyone help me with an asterisk 1.6 SIP issue - debug at http://pastebin.com/m141dd35a |
| 16:48.58 | UnixDog | weazahl, is your nick reged ? |
| 16:49.06 | bsdwarrior | whats the correct syntax |
| 16:49.08 | weazahl | UnixDog: yeah, i know the transcoding takes power, could have all the phones in ulaw. but that would reduce quality of the in house calls. that system should be able to handle 15 calls |
| 16:49.15 | weazahl | yes it is |
| 16:49.26 | UnixDog | ok |
| 16:49.38 | *** join/#asterisk RoyK (n=roy@ip-85-4-149-91.dialup.ice.no) |
| 16:49.54 | UnixDog | so inhouse ulaw should be fine with a qos router |
| 16:50.54 | jm|laptop | hmm |
| 16:51.03 | jm|laptop | you can tell when a call was purely VOIP |
| 16:51.32 | jm|laptop | I just rang a supplier who I'm pretty sure use VOIP and perhaps because of our suppliers I don't think it even hit the PSTN |
| 16:52.40 | agx | Does the Junghans 2 BRI card works with chan_misdn? |
| 16:55.05 | drmessano | hmmm |
| 16:55.28 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:55.53 | drmessano | Does anyone have a working Gizmo/Sipphone config? |
| 16:56.17 | jameswf | b11d: I wanter your dial plan but for giggles try switching your signalling to ks |
| 16:56.56 | b11d | jameswf.. i did that already too.. no dice :) |
| 16:57.07 | b11d | i posted my extensions.conf too james.. |
| 16:57.35 | jameswf | sorry was away didnt scroll up |
| 17:00.12 | jameswf | why dont you add exten => 3333,1,answer exten => 333,n,echo and see if you get an echo |
| 17:00.26 | jameswf | s/333/3333/ |
| 17:00.35 | jameswf | damnit |
| 17:00.38 | drmessano | lol |
| 17:00.42 | b11d | ok.. will do. |
| 17:00.43 | b11d | one sec. |
| 17:00.43 | jameswf | you know what i meanr |
| 17:01.49 | b11d | it echos.. any way to delay the echo so it is more aparent? |
| 17:02.03 | freezey | [TK]D-Fender: oh i asked how do i find out the phone model? |
| 17:02.22 | tzafrir | drmessano, search for sipphone.com in voip-info.org |
| 17:02.22 | [TK]D-Fender | freezey: ........ |
| 17:02.37 | *** join/#asterisk myiagy (n=Jose@200.215.59.133) |
| 17:02.38 | freezey | [TK]D-Fender: you asked what the phone model is |
| 17:02.51 | [TK]D-Fender | freezey: Di i really even have to say it? |
| 17:02.56 | *** join/#asterisk Cyorxamp (n=Cyorxamp@212.57.232.254) |
| 17:03.10 | freezey | [TK]D-Fender: i guess so ha |
| 17:03.42 | [TK]D-Fender | freezey: You walk over, pick the ^@%#ing phone up, flip it over and read the damn model off the bottom sticker! |
| 17:03.44 | jameswf | you could record wait playback but thats more complicated... the issue is in asterisk and bridging as the audio paths are all there eliminating the channelbank and cards as an issue |
| 17:03.54 | freezey | [TK]D-Fender: oh like that :P |
| 17:04.10 | freezey | [TK]D-Fender: 5410 |
| 17:04.28 | b11d | agreed |
| 17:04.33 | [TK]D-Fender | I've squeezed blood out of stones considerably easier than this |
| 17:04.36 | b11d | im installing a SIP phone now to check that out.. |
| 17:06.05 | [TK]D-Fender | freezey: Ok, your entire Avaya system is officially useless to Asterisk and will have to be scrapped in its entirety |
| 17:06.21 | FlatFoot | [TK]D-Fender chill man :p |
| 17:06.35 | freezey | [TK]D-Fender: why is that?> |
| 17:06.36 | drmessano | [TK]D-Fender, you got a minute to look at a debug for me? |
| 17:06.39 | [TK]D-Fender | FlatFoot: Perfectly chill, there was no anger in that last statement at all. |
| 17:06.52 | [TK]D-Fender | FlatFoot: the one before was just seething with bile however :) |
| 17:07.08 | kyron | Trick question: the 320 poly. I have has a neat feature of automatically calling the "missed call" number that I am presently viewing, problem is, it attempts to call without pre-pending the number with a 9 (to dial out) is this something I have to do in the poly config or is it a "hack" I have to perform on the * side (me thinks it's the poly, hacking the * side means literally eliminating the 9 to dial out) |
| 17:07.21 | FlatFoot | [TK]D-Fender what you been eating ? shouldn't have bile |
| 17:07.39 | *** join/#asterisk Vec (n=Vec@87-194-2-194.bethere.co.uk) |
| 17:07.44 | [TK]D-Fender | freezey: Because there is no way to connect your phones, or other Avaya gear to *. Citel's gateway (about the only thing out there), does not support the 5400 series phones. |
| 17:07.54 | [TK]D-Fender | freezey: those are digital handsets. |
| 17:08.02 | [TK]D-Fender | freezey: * can't talk to those |
| 17:08.05 | FlatFoot | kyron why not just add a 9 to the incoming callerid |
| 17:08.23 | FlatFoot | thats what we do for the numpties in the office |
| 17:08.30 | [TK]D-Fender | kyron: No, remove that silly prefix from your * dialplan. |
| 17:08.31 | kyron | FlatFoot, ah, ok, that's in the "hack *" category ;) |
| 17:09.00 | kyron | [TK]D-Fender, and make all outgoing calls matched on a 10 digit pattern? |
| 17:09.15 | kyron | [TK]D-Fender, that was the solution I was thinking about... |
| 17:09.21 | [TK]D-Fender | kyron: All of my dialplans are auto-adaptive 7-10-11 digit |
| 17:09.23 | FlatFoot | unfortunately all our old phone systems used a nine to get out and people have got used to it |
| 17:09.37 | drmessano | I guess not lol |
| 17:09.50 | kyron | FlatFoot, well, it's not my case so I'll be more than happy to scrap it ;) |
| 17:10.02 | *** join/#asterisk the_5th_wheel (n=edd@dsl-242-161-240.telkomadsl.co.za) |
| 17:10.03 | FlatFoot | kyron scrap it if you can |
| 17:10.14 | [TK]D-Fender | FlatFoot: Nothing a little judicious tasering won't solve (or make more entertaining) |
| 17:10.24 | the_5th_wheel | Jan 31 19:06:37 WARNING[11518]: chan_zap.c:9166 pri_dchannel: Ring requested on channel 0/1 already in use on span 7. Hanging up owner. <-- can someone inform me on what this means? |
| 17:10.31 | the_5th_wheel | it seems i cant accept any phonecalls |
| 17:10.39 | FlatFoot | it's all right for you lot over the pond , with all these wonderful gadgets |
| 17:11.04 | Vec | I have a Sipura 3000, which is behind NAT connecting to a Asterisk Box which has a public ip, I am able to make calls, and sometimes recieve calls, keep getting the following registration problems http://pastebin.ca/886377, does anyone have any idea, why this is happening ? |
| 17:11.10 | [TK]D-Fender | FlatFoot: Fine... a cricket bat is a perfectly viable substitute :) |
| 17:11.11 | *** join/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 17:11.21 | kyron | so, basically: NXXNXXXXXX |
| 17:11.34 | FlatFoot | [TK]D-Fender lol , running to the pavillion for one now |
| 17:11.36 | kyron | though this one looks...uhm...stoo-pid |
| 17:12.05 | drmessano | http://www.pastebin.ca/886376 <--- If anyone wants to have a look.. Can't get inbound working with gizmo/sipphone |
| 17:12.52 | [TK]D-Fender | drmessano: Ignoring this INVITE request <- whee! |
| 17:13.07 | kyron | actually not that stupid... the present typer is the dumb one... pff ;) |
| 17:13.18 | freezey | [TK]D-Fender: ok so whats your suggestion? |
| 17:13.21 | drmessano | Ok.... I got that much |
| 17:13.22 | [TK]D-Fender | drmessano: [sipphone-rw] insecure=port,invite |
| 17:13.35 | [TK]D-Fender | freezey: Sell your old system and plan a complete new one |
| 17:13.37 | freezey | [TK]D-Fender: say i get rid of avaya system i can completely implenet asterisk but does that mean i need all new phones? |
| 17:13.44 | drmessano | I tried that :( |
| 17:14.01 | drmessano | One of the first things I tried, actually |
| 17:14.02 | [TK]D-Fender | freezey: Yes, like I said earlier, NONE of your old hardware is usable by *. |
| 17:14.22 | [TK]D-Fender | drmessano: pastebin your peer entry |
| 17:14.34 | freezey | [TK]D-Fender: brb i gotta run and do somethin but i got a few more questions |
| 17:15.16 | Qwell | the avaya phones can do SIP, can't they? or are they old ones? |
| 17:15.30 | Qwell | oh, digital handsets...meh |
| 17:15.40 | [TK]D-Fender | Qwell: Crap-tastic! |
| 17:15.53 | Qwell | ebay! |
| 17:15.56 | freezey | hmm |
| 17:16.12 | Qwell | could probably get a bit for them, from some sucker^Wbody |
| 17:17.13 | drmessano | http://www.pastebin.ca/886393 |
| 17:17.59 | Vec | I have a Sipura 3000 which is behind NAT connecting to a Asterisk box in front of NAT, it keeps deregistering every 60 seconds and then registers again a few seconds later, (from the perspective of the asterisk box), please see http://pastebin.ca/886377, any ideas ? |
| 17:18.30 | [TK]D-Fender | drmessano: type=peer |
| 17:18.41 | [TK]D-Fender | Vec: pastebin its sip.conf entries |
| 17:18.43 | lirakis_work | Vec: you have more than one device with the same "extension" ? |
| 17:19.01 | [TK]D-Fender | lirakis_work: Not the problem per PB. |
| 17:20.30 | kyron | [TK]D-Fender, care to share patterns? |
| 17:21.02 | [TK]D-Fender | kyron: You already ahve the 1st.... do I need to hand-hold you for the other 2? :) |
| 17:21.04 | lirakis_work | [TK]D-Fender: mhmm... thanks for that multi language help earlier.. thats a wierd way to layout the sound structure... and im going to restructure my application so that it doesnt have the same issues that * has with multi-lingual support |
| 17:21.18 | drmessano | Still no go |
| 17:21.22 | Vec | lirakis_work : definately not more than 1 device with the same extension, here is the sip.conf entry http://pastebin.ca/886377 |
| 17:21.36 | Vec | [TK]D-Fender : http://pastebin.ca/886377 sip.conf entry, thanks |
| 17:22.00 | kyron | [TK]D-Fender, noooo... well, my supplier has specific provisions about 1-800 # (that is the only case where I have to send out the call with the 1) |
| 17:22.18 | lirakis_work | Vec: thats not sip.conf ... check your link |
| 17:22.19 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 17:22.21 | Vec | [TK]D-Fender : sorry thats it http://pastebin.ca/886398 |
| 17:22.30 | drmessano | jblack and I have been pounding this out with both our boxes for a few days.. toggled peer types, insecure, add authuser, fromdomain and all sorts of unnecessary crap that others posted in forums that "worked" |
| 17:22.53 | drmessano | Fire up a PAP2 with the config and it works great |
| 17:23.05 | kyron | [TK]D-Fender, are your hands soft? |
| 17:23.11 | drmessano | I have a bunch of other working peers |
| 17:23.20 | drmessano | Just something about Gizmo |
| 17:23.25 | [TK]D-Fender | Vec: Looks fine... I think either the network connection and/or router blows. What model are they using? |
| 17:23.26 | lirakis_work | Vec: it [TK] wanted the WHOLE sip.conf |
| 17:23.48 | [TK]D-Fender | lirakis_work: No, I didn't |
| 17:24.12 | [TK]D-Fender | lirakis_work: And it looks fine. config checks out, its routing/router at this point. |
| 17:24.14 | plik | drmessano: got port 5004 open for udp both ways? |
| 17:24.23 | drmessano | Why? |
| 17:24.27 | lirakis_work | [TK]D-Fender: what were you looking for (curious) in the peer entry? |
| 17:24.43 | plik | dunno, apparently something wierd with gizmo |
| 17:24.45 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 17:24.51 | drmessano | Thats with their CLIENT |
| 17:24.56 | [TK]D-Fender | kyron: My adaptive goes out my PRi for 10 as 11 except 514/450, All others force 1. |
| 17:24.57 | plik | drmessano: according to http://www.voip-info.org/wiki/view/Asterisk+settings+Gizmo |
| 17:25.00 | Vec | [TK]D-Fender : I don't think its the connection if a ping the asterisk server from where I am, I get no packet loss, for a long period of time, and the registration problems seem to occur too consistantly (about every 60 seconds) for it to be a network problem) ? |
| 17:25.02 | drmessano | Like I said, I used a PAP2 and it worked great |
| 17:25.11 | [TK]D-Fender | lirakis_work: Qualify, NAT. |
| 17:25.16 | drmessano | With no open ports to it |
| 17:25.26 | lirakis_work | [TK]D-Fender: hmm |
| 17:25.29 | plik | ok , just sharing what I read. |
| 17:25.34 | drmessano | I appreciate it |
| 17:25.40 | drmessano | It shouldnt be this hard.. at all |
| 17:25.49 | *** join/#asterisk tangel (i=tangel@208.239.77.234) |
| 17:25.53 | [TK]D-Fender | Vec: Could be a staringht NAT map. The router could be mangling the shit out of things. Dlick, Cisco PIX, and others are notorious for this |
| 17:25.55 | plik | I'll be ready to try it in a few days, so we can all struggle together then |
| 17:25.59 | tangel | hello. |
| 17:26.04 | tangel | does anyone know if the digium cards work in Windows? |
| 17:26.04 | [TK]D-Fender | D-Link* |
| 17:26.07 | lirakis_work | Vec: where is your server at ? not locally? |
| 17:26.13 | Qwell | tangel: no, and neither does Asterisk |
| 17:26.14 | tangel | i have a TDM400P |
| 17:26.21 | drmessano | You can join jblack and I in #fuckgizmo if we're still at it in a few days |
| 17:26.22 | [TK]D-Fender | tangel: No drivers for it. Sangoma's do. |
| 17:26.23 | tangel | oh.. i noticed an asterisk port to windows |
| 17:26.26 | tangel | does it not work? |
| 17:26.28 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 17:26.28 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:26.29 | plik | heh OK |
| 17:26.33 | Qwell | tangel: no, it's from 2005, and it didn't even work then |
| 17:26.41 | tangel | ok.. good info |
| 17:26.42 | Qwell | install Linux, and move on |
| 17:26.50 | tangel | what do people recommend for a small possibly fanless asterisk machine? |
| 17:27.00 | Qwell | tangel: Digium's asterisk appliance :D |
| 17:27.02 | drmessano | I find it hard to believe that a PAP2 will work with something that Asterisk wont.. |
| 17:27.02 | tangel | i have linux on an old athlon machine that makes too much noise |
| 17:27.12 | drmessano | So its some BS with Gizmo |
| 17:27.13 | tangel | hmm. i was hoping something for a couple hundred bucks.. does that fit? |
| 17:27.14 | Qwell | if not, there are things like the soekris (sp) boards |
| 17:27.16 | lirakis_work | Vec: 400-500 ms is a long delay |
| 17:27.26 | tangel | true.. my friend has a soekris machine |
| 17:27.33 | Qwell | those are nice |
| 17:27.35 | tangel | it would be nice if i could use the digium board |
| 17:27.42 | Qwell | I thought they had PCI? |
| 17:27.54 | Qwell | I know there are some nano/micro-ata boards that have pci |
| 17:27.54 | tangel | not sure.. maybe it does. |
| 17:28.06 | kyron | [TK]D-Fender, ah, ok, you're supplying your own... |
| 17:28.07 | tangel | yea.. i'll probably shoot for a shuttle or moving the machine out of my office |
| 17:28.09 | Qwell | ata? atx. |
| 17:28.12 | tangel | anyway, thanks for the great info |
| 17:28.20 | [TK]D-Fender | drmessano: try type=peer & insecure=very just because |
| 17:28.31 | [TK]D-Fender | kyron: I add a 1 for LD call-backs, etc. |
| 17:28.52 | [TK]D-Fender | kyron: My users go to their Polycom missed calls screen and call back with impunity. |
| 17:29.17 | kyron | [TK]D-Fender, you mean you add the 1 on inbound LD calls...correct? |
| 17:29.33 | kyron | [TK]D-Fender, ah those evil users...bend THEM...not the system! |
| 17:29.34 | kyron | :P |
| 17:29.41 | [TK]D-Fender | kyron: No, I add it to what I literally dial based on the pattern that is received |
| 17:29.42 | drmessano | ok |
| 17:29.48 | Vec | [TK]D-Fender, lirakis_work : the ATA is in London, UK, the asterisk server is in South Africa, hence the long latency, there is no packet loss, the router in South Africa is a Planet.com.tw router in the UK its a SpeedTouch ADSL2+ router, pretty confident its not a problem at the IP layer. |
| 17:30.05 | [TK]D-Fender | kyron: using "9" BS... thats just... well BS. |
| 17:30.07 | kyron | [TK]D-Fender, ok, you filter local Vs LD NXX |
| 17:30.13 | kyron | [TK]D-Fender, agreed |
| 17:30.16 | [TK]D-Fender | kyron: Pick up the phone, dial , and the PBX does the rest |
| 17:30.26 | kyron | totally |
| 17:30.52 | FlatFoot | Vec get rid of the SpeedTouch , change to a draytek |
| 17:31.03 | [TK]D-Fender | kyron: Like I said I pas 10 digit 450/514 as-is and allow them to do 11 as they wish. Non 450/514 gets expanded to 11 digit consistently. |
| 17:31.10 | kyron | the only chill I have about all this is managing the 911 calls..I mean, yes it's documented and all...it's just that it can't be conveniently tested... |
| 17:31.21 | Qwell | kyron: sure it can |
| 17:31.28 | Vec | FlatFoot : why don't u like the Speedtouch ? |
| 17:31.36 | kyron | Qwell, O_o... |
| 17:31.42 | Qwell | dial 911, explain very briefly that it is not an emergency, and if they can verify that your information is showing correctly. |
| 17:31.46 | drmessano | Same crap |
| 17:32.00 | FlatFoot | Vec no they are a bit slow passing data , the draytek does not get in the way |
| 17:32.01 | Qwell | just...don't...do it more than once :p |
| 17:32.13 | kyron | Qwell, hmmm...I won't get fla....ah... yeah..._once_ |
| 17:32.25 | Qwell | how else are you going to test it? |
| 17:32.30 | plik | I'm inclined to agree that draytek is likely to be better than Speedtuch, but have no hard evidence yet |
| 17:32.35 | Qwell | they expect those calls, just go about it correctly |
| 17:32.58 | FlatFoot | see ya all a bit l8r |
| 17:33.06 | plik | ta ta |
| 17:33.55 | b11d | OK.. I just tested SIP<->ZAP and it works! |
| 17:34.08 | b11d | ZAP<->ZAP continues to not work.. |
| 17:36.13 | b11d | kyron.. just call your local police department and ask for the number for your 911 dispatch center, call that number and arrange a test. I do it once a year here.. |
| 17:37.29 | Qwell | b11d: that's no fun |
| 17:37.34 | kyron | b11d, yeah, I was thinking of something along that line. For the moment I still have a "hard" line but on the implementation, I intend on going full VoIP...then again... we might keep one hard line for faxes...so maybe I could get one FXO and pass the 911 calls through that line... |
| 17:37.44 | kyron | Qwell, I-ky-youu |
| 17:38.04 | [TK]D-Fender | b11d: ok, next psycho test : loop the all between a local channel <--- |
| 17:38.22 | [TK]D-Fender | call* |
| 17:38.27 | hendrixski | when a caller enters the queue, before they are connected to a member is there anything that already plays a sound file for them? |
| 17:38.38 | [TK]D-Fender | b11d: dont forget the "/n" to leave * in the way |
| 17:38.43 | b11d | hmm.. not sure how to do that TK.. let me work on that.. |
| 17:39.02 | [TK]D-Fender | hendrixski: MoH, Period annouce, yes |
| 17:39.44 | hendrixski | [TK]D-Fender, MoH stands for Music on Hold, yes? |
| 17:39.57 | [TK]D-Fender | hendrixski: Yes |
| 17:41.00 | b11d | bbl all.. gotta go wake up the wife. |
| 17:41.10 | [TK]D-Fender | b11d|bbl: http://pastebin.ca/886419 |
| 17:41.19 | b11d|bbl | ok. |
| 17:41.32 | b11d|bbl | thanks :) |
| 17:42.54 | b11d|bbl | dialing 1111 from 2222 still results in no audio with that TK.. |
| 17:46.58 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:51.06 | b11d|bbl | ok I gotta run for a bit.. will be back after lunch. Thanks for your help! |
| 17:52.39 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:52.41 | kyron | Wow... freePBX configs _are_ convoluted... the price of genericity I guess.. |
| 17:53.04 | kyron | no wonder the machine specs are...uhm...GHz++ |
| 17:54.41 | Qwell | kyron: no, really, it has nothing to do with "genericity (I like that word, btw)".. it's just done poorly.. |
| 17:55.28 | kyron | Qwell, LOL, you mean the whole mish-mash of macros isn't really requiredP :P |
| 17:55.45 | Qwell | no.. |
| 17:55.55 | Qwell | I mean, yes, that's what I mean - no it isn't required.. |
| 17:56.23 | ZaVoid | can someone point me to a good place on iax messaging call signaling that is? so i can read up on it |
| 17:56.32 | kyron | hehehe...not surprised. Guess rebuilding my system from trix to plain * will water down the configs to a single simple file :P |
| 17:56.41 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583384.dsl.bell.ca) |
| 17:57.58 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 17:58.01 | Uatec | hi there |
| 17:58.20 | Uatec | how can i replace a digit in a variable in my dialplan? |
| 17:58.25 | Uatec | what i mean is, i have ${EXTEN} |
| 17:58.41 | Uatec | and i want to replace all "#" in it with "." |
| 17:59.03 | bkruse | use ${REGEX} |
| 17:59.37 | bkruse | core show function REGEX |
| 18:00.13 | bkruse | so set(EXTEN=${REGEX("reg exp", ${EXTEN})}) should work |
| 18:01.00 | plik | thats awesome:) |
| 18:01.17 | RoyK | or rather Set(NEWEXT=... |
| 18:01.17 | plik | could be really good fun on april fools day |
| 18:02.28 | kyron | hmmmm...REGEX...goodies :P |
| 18:05.31 | outtolunc | that should give him a heart attack, that is unless you already do sex chatting <G> |
| 18:06.35 | *** join/#asterisk mltlnx (n=mltlnx@rrcs-24-105-157-191.nyc.biz.rr.com) |
| 18:07.12 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-75-194-159.bflony.east.verizon.net) |
| 18:09.33 | freezey | [TK]D-Fender: wheres a list of models of avay that asterisk works with?> cause if i have to i will buy a new phone system |
| 18:10.16 | plik | outtolunc: or you could have "no need for a credit card as you've already been billed by our premium rate system" |
| 18:10.58 | *** join/#asterisk Simon-- (n=sim@staff-nat.netnation.com) |
| 18:12.08 | [TK]D-Fender | freezey: Ley me put it this way : www.citel.com makes an adapter taht at **$3100** can let you use 24 phones its is compatible with with *. The cost of 24 new Polycom IP 330 = $2760 + $350 for a PoE Switch = $3110 for a setup that you are MUCH better off with. That doesn't account for your having to have COMPATIBLE handsets for the Citel either. |
| 18:12.32 | hendrixski | in asterisk, monitor refers to recording, right? |
| 18:12.34 | [TK]D-Fender | freezey: that Avaya junk should be disposed of as quickly as possible. |
| 18:13.04 | [TK]D-Fender | freezey: it is a complete dead-end and you will only waste time trying to avoid this imminently practical truth. |
| 18:13.09 | [TK]D-Fender | hendrixski: Yes |
| 18:13.32 | Simon-- | anybody aware of bugs in TE410P (1st Gen) that cause outbound-direction clicking and eventually bad enough noise that even d-chan data gets corrupted and calls get hung up? this appears to be occurring on multiple boxes for us now, and appears to be load-related (zttool reports no lost interrupts, and outbound (towards telco) only) |
| 18:13.46 | freezey | [TK]D-Fender: dam... my old system i had it running with avaya Definity PBX |
| 18:13.51 | hendrixski | [TK]D-Fender, thanks again, and do you know if that is fired up before or after a caller is connected to a member? |
| 18:14.19 | [TK]D-Fender | hendrixski: Not sure of exact timing. |
| 18:14.41 | *** join/#asterisk CVirus (n=GoD@196.205.192.211) |
| 18:14.45 | freezey | [TK]D-Fender: one of the people i set it up with actually held a presentation in las vegas about it |
| 18:15.05 | hendrixski | [TK]D-Fender, that's Ok... I'd figure it out if I knew what "channels" referred to :-/ |
| 18:15.23 | [TK]D-Fender | hendrixski: Not kow what "channels" are is usually a shoot-on-sight offense :) |
| 18:15.57 | hendrixski | [TK]D-Fender, it's my first time opening up asterisk, I'm sure the bullets will miss me :-) |
| 18:16.54 | hendrixski | from having googled it before I gather that it's something to do with connecting with devices, but I'm not sure if I'm "getting" it |
| 18:17.06 | plik | uhoh, not the NewbBlaster9000 !!!! |
| 18:17.38 | kyron | Poly Q: if i define a dialplan option in a phone's config, will it be _merged_ with the sip.cfg file or will it _replace_ the one in sip.cfg. The question is specifically related to dialplan.digitmap (wondering if I have to copy over the one defined in sip.cfg or not) |
| 18:17.58 | [TK]D-Fender | kyron: Should replace. |
| 18:18.06 | [TK]D-Fender | kyron: Tag for tag should be override |
| 18:18.24 | kyron | [TK]D-Fender, thanks, btw, NewbBlaster9000 is so yesterday, get the real thing, like Doom's BFG :P |
| 18:19.16 | freezey | [TK]D-Fender: hey i wanted to thank you for your help |
| 18:19.23 | freezey | [TK]D-Fender: and patience =P |
| 18:19.30 | plik | the only irc-weapon thats horrendously out of date is mircs trout |
| 18:19.54 | [TK]D-Fender | ~fish plik |
| 18:19.54 | jbot | ACTION slaps plik around with a large trout |
| 18:19.54 | kyron | plik, LOL, that's soooo BBSish ;) |
| 18:20.00 | kyron | LOOOOOOOOOOLLLLL |
| 18:20.03 | kyron | it Lives! |
| 18:20.14 | [TK]D-Fender | ~jbot |
| 18:20.15 | jbot | [jbot] a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch |
| 18:20.54 | kyron | [TK]D-Fender, a nagging question, is it TK as in the TK programming or what? |
| 18:21.28 | [TK]D-Fender | kyron: No, my antiquated clan nick from over 5 years ago... |
| 18:21.44 | kyron | 5...antiquated O_o...eh |
| 18:22.22 | Vec | Why would asterisk think a SIP phone is UNREACHABLE when the SIP phone is able to make outgoing calls, and the SIP phone thinks it is registered (The SIP phone is behind NAT)? |
| 18:23.13 | [TK]D-Fender | Vec: because inbound mapping fails. |
| 18:23.42 | Vec | [TK]D-Fender : any idea how to fix that, I definately think that is my problem ? |
| 18:23.44 | [TK]D-Fender | Vec: I can ignore you talking to me and you think I'm not there, but that doesn't stop me from speaking to you and you suddenly realizing "OH HAI KTHX!" |
| 18:24.02 | [TK]D-Fender | Vec: as i said its either the route, or the router. |
| 18:24.55 | Vec | [TK]D-Fender : I don't think its the router, maybe the router, would port forwarding on the NATed where the phone is help ? |
| 18:25.51 | [TK]D-Fender | Vec: No. What you could do is turn off qualify, enable NAT keep-alive on the device (which should have been off thus far), and see if that helps |
| 18:26.41 | Vec | [TK]D-Fender : k thanks, I have tried NAT keep-alive, no and off on the phone, will now try with qualify off |
| 18:28.39 | *** join/#asterisk guillote_GNU (n=guillote@host63.201-253-22.telecom.net.ar) |
| 18:28.39 | *** join/#asterisk qdk (n=qdk@87.63.229.126) |
| 18:30.34 | plik | ~book |
| 18:30.35 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 18:30.40 | Vec | [TK]D-Fender : with most of my phones behind NAT the port in 'sip show peers' is 5060, but with my phone outside the LAN the port always changes at the moment its 57941, any idea why that happens ? Is that port the port the phone connects to the asterisk box on, or the port the asterisk box connects to the phone on ? |
| 18:30.48 | *** part/#asterisk Simon-- (n=sim@staff-nat.netnation.com) |
| 18:30.53 | jameswf | jbot play dead |
| 18:30.53 | jbot | why don't you jameswf |
| 18:31.23 | hendrixski | jbot, tell me about channels |
| 18:31.26 | [TK]D-Fender | Vec: do you have multiple phones at that site? |
| 18:31.34 | [TK]D-Fender | hendrixski: ... |
| 18:31.36 | [TK]D-Fender | ~book |
| 18:31.37 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 18:31.38 | jameswf | jbot sit |
| 18:31.38 | jbot | ACTION sits |
| 18:31.38 | [TK]D-Fender | ^^^^^^^^^ |
| 18:31.44 | jameswf | jbot stay |
| 18:31.45 | jbot | woof |
| 18:31.46 | [TK]D-Fender | ~areyouadog ? |
| 18:31.47 | jbot | Bark! Bark! |
| 18:31.52 | jameswf | jbot good boy |
| 18:31.52 | jbot | aw, gee, jameswf |
| 18:31.58 | [TK]D-Fender | ~botsnack |
| 18:31.58 | jbot | [TK]D-Fender: aw, gee |
| 18:32.03 | Vec | [TK]D-Fender : yes, about 50 on the LAN, 1 outside. |
| 18:32.10 | [TK]D-Fender | jameswf: jbot loves me best :p |
| 18:32.23 | [TK]D-Fender | Vec: I'm talking behind that 1 router |
| 18:32.23 | jameswf | jbot who do you love best |
| 18:32.45 | jameswf | jbot who do you love best is <reply> jameswf |
| 18:32.47 | jbot | okay, jameswf |
| 18:32.47 | [TK]D-Fender | ~8ball is jameswf barking up the wrong tree? |
| 18:32.47 | jbot | Unsure. |
| 18:32.47 | Vec | [TK]D-Fender : no |
| 18:32.51 | jameswf | jbot who do you love best |
| 18:32.52 | jbot | jameswf |
| 18:32.55 | jameswf | :) |
| 18:32.56 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 18:33.27 | [TK]D-Fender | lies! |
| 18:33.48 | [T]ank | i have had music on hold working for ever and suddenly it is not working. I do not have the option of restarting asterisk to fix it. I have tried an moh reload and a reload res_musiconhold.so and nothing seems to affect it. |
| 18:33.56 | *** join/#asterisk Goldfisch (n=gturnqui@158-147-54-92.harris.com) |
| 18:34.00 | [T]ank | is there something on linux that maybe failed that I can check for? |
| 18:36.10 | jameswf | have you tried reload app_mp3.so |
| 18:36.44 | [T]ank | trying now |
| 18:37.08 | [T]ank | dont have mp3.so |
| 18:37.52 | [T]ank | should I? |
| 18:38.17 | jameswf | what are your moh files |
| 18:38.25 | jameswf | should have asked first eh |
| 18:38.28 | [T]ank | wav |
| 18:38.32 | jameswf | oh nm |
| 18:39.29 | jameswf | what do your full logs say |
| 18:40.24 | [T]ank | full logs you mean /var/log/asterisk/messages? |
| 18:40.35 | jameswf | sure lets go eith that |
| 18:40.43 | *** join/#asterisk timeshell (n=Khoja@gw.lusi.on.ca) |
| 18:40.59 | timeshell | Greetings |
| 18:41.57 | jameswf | also does moh show files match up with your context in moh show classes |
| 18:42.18 | *** join/#asterisk trippss (n=ss@72.20.150.196) |
| 18:43.01 | jameswf | hold music should be a tt-monkeys loop... that should reduce hold times |
| 18:44.42 | [T]ank | yeah the moh show files is correct. |
| 18:45.22 | jameswf | do you have prompts and other audio |
| 18:45.27 | [T]ank | yes |
| 18:47.12 | *** join/#asterisk AndyGraybeal (i=minDlabs@node212.36.251.72.1dial.com) |
| 18:49.04 | *** part/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 18:56.50 | plik | whats the FLA you guys use for a voip provider? |
| 18:56.52 | plik | itsp? |
| 19:01.23 | *** join/#asterisk anarcat (n=anarcat@pdpc/supporter/active/anarcat) |
| 19:01.34 | Qwell | FLA? |
| 19:01.35 | anarcat | hi |
| 19:01.41 | Qwell | oh |
| 19:01.42 | Qwell | ~itsp |
| 19:01.43 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 19:01.43 | Qwell | yeah |
| 19:02.08 | mvanbaak | ~itsplist-us |
| 19:02.09 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, or http://www.jnctn.com, or http://www.bandwidth.com |
| 19:04.42 | hmmhesays | ~seen agx |
| 19:05.13 | jbot | agx is currently on #asterisk-dev (1h 12m 29s) #asterisk-bugs (1h 12m 29s), last said: 'Qwell, ok'. |
| 19:05.37 | anarcat | [OT] i'm looking for a full duplex (possibly voip) device for conference rooms, any good suggestions? |
| 19:05.48 | timeshell | Qwell: How do you remove phantom Service Providers that were entered in a previous version of the gui; the ones that still show up as a provider in the Calling Rules section but no longer appear in the Service Providers section? |
| 19:07.20 | AndyGraybeal | anarcat: thare's that polycom thing for like $9k |
| 19:07.28 | anarcat | whoa |
| 19:07.29 | anarcat | 9k?? |
| 19:07.41 | anarcat | i'm more in the 200$ price range here, so i guess i'd have to go analog? |
| 19:07.46 | AndyGraybeal | yea, lemme look it up |
| 19:07.55 | AndyGraybeal | usd |
| 19:07.57 | Qwell | anarcat: why not...use asterisk? |
| 19:08.32 | anarcat | Qwell: well, sure, we use asterisk, but we need people sitting at a conference room to be able to chat with someone in argentina (we're in canada) without echo and good sound quality |
| 19:08.39 | anarcat | which is why i said this was off-topic :) |
| 19:09.08 | Qwell | so you just need phones, or what? |
| 19:09.15 | anarcat | yes. |
| 19:09.22 | Qwell | the polycom conference phones are very good |
| 19:09.23 | anarcat | i figured people here would have experience with that |
| 19:09.27 | anarcat | yeah... i heard |
| 19:09.28 | Qwell | they're probably in the 300-400 range though |
| 19:09.33 | Qwell | not really sure |
| 19:09.36 | anarcat | i heard they're very expensive too ;) |
| 19:09.57 | Qwell | anarcat: would you like me to go grab the model number of them? |
| 19:10.06 | Qwell | so you can price it and such |
| 19:10.15 | *** join/#asterisk af_ (n=getsmart@88-149-240-211.dynamic.ngi.it) |
| 19:10.43 | anarcat | Qwell: i'm looking at this page now: http://www.polycom.com/usa/en/products/voice/small_medium_conference_room/soundstation2.html |
| 19:11.00 | anarcat | cheapest looks like 500$ |
| 19:11.03 | anarcat | maybe i should go used... |
| 19:11.07 | Qwell | are those the SIP ones? |
| 19:11.27 | anarcat | i think the 500$ is POTS |
| 19:11.41 | Qwell | I think it's the SoundStation IP 4000 |
| 19:11.46 | Qwell | http://www.polycom.com/usa/en/products/voice/small_medium_conference_room/soundstation_ip4000.html |
| 19:13.17 | jameswf | I use to work for the co-founder of polycom tht guy is a genius |
| 19:13.48 | anarcat | interesting: http://store.polycom.com/searchresults.aspx?dept_id=51 |
| 19:13.51 | anarcat | 300$! |
| 19:14.22 | Qwell | wow, the soundstation ip 4000 is like $700 |
| 19:14.30 | anarcat | sweet |
| 19:15.52 | anarcat | lots of stuff on ebay |
| 19:15.59 | anarcat | thanks the tip, people |
| 19:16.45 | mjoyce | the phone is awesome though |
| 19:16.53 | Qwell | mjoyce: agreed |
| 19:16.59 | Qwell | mjoyce: did you see my message yesterday? |
| 19:17.19 | mjoyce | likely not, this screen is only set for a buffer of like 500 |
| 19:17.43 | [TK]D-Fender | anarcat: http://www.polycom.com/usa/en/products/voice/small_medium_conference_room/soundstation2w.html |
| 19:17.57 | [TK]D-Fender | anarcat: I use that + ATA here |
| 19:17.59 | mjoyce | missed it :( |
| 19:18.29 | anarcat | [TK]D-Fender: and does it work^ :) |
| 19:18.31 | anarcat | ? |
| 19:18.37 | mjoyce | id use the IP 601 / 650 if i couldn't afford the 4000, the speakerphones on the 6xx's are legit |
| 19:18.38 | [TK]D-Fender | anarcat: Quite well |
| 19:18.42 | [TK]D-Fender | anarcat: And portable |
| 19:18.46 | *** join/#asterisk ManxPower (n=manxpowe@19.sub-75-202-0.myvzw.com) |
| 19:19.22 | anarcat | you mean it's wireless? is that 802.11[abg] wifi? |
| 19:19.43 | [TK]D-Fender | anarcat: No, wireless analog |
| 19:19.51 | [TK]D-Fender | anarcat: I said with ATA here |
| 19:20.02 | [TK]D-Fender | anarcat: Which means if they don't have * at some point, its reusable too |
| 19:20.12 | anarcat | yeah, pretty cool |
| 19:20.21 | anarcat | although I fail to see the point in the analog wireless here... |
| 19:20.28 | mikebee | funny, I was just configuring a IP soundstation 4k this second, heh |
| 19:20.43 | mikebee | switched back to this screen and saw that, weird |
| 19:21.12 | [TK]D-Fender | anarcat: Means you can bring it to another conference room, and have it up and running without wires trailing all over the place. |
| 19:21.28 | *** join/#asterisk GrumpManAtWork (n=meanderi@pool-72-78-137-186.phlapa.east.verizon.net) |
| 19:21.46 | anarcat | [TK]D-Fender: pretty cool :) |
| 19:22.40 | *** join/#asterisk grandpapadot (n=null@mail.heavylogic.com) |
| 19:23.41 | grandpapadot | Is there any magic to getting Asterisk to take advantage of more than a single CPU. I have a dual-dual core (4 x cores) system and Asterisk, the only user-land app running, seems to hover on CPU0, using between 8-12% on average. My other cores stay at 0%. |
| 19:24.03 | grandpapadot | Asterisk 1.2 |
| 19:24.24 | bkruse | [TK]D-Fender: test mokoiax released, and asterisk package on moko :] |
| 19:24.45 | [TK]D-Fender | bkruse: Cool... If only I didn't sign on with Bell for 3 more years... |
| 19:24.50 | [TK]D-Fender | bkruse: CDMA <- |
| 19:24.52 | twisted | mokoiax? |
| 19:24.56 | [TK]D-Fender | bkruse: Price was right... |
| 19:25.19 | bkruse | twisted: iax client for openmoko |
| 19:25.23 | twisted | ahh |
| 19:25.27 | bkruse | [TK]D-Fender: aww :[ its really slick |
| 19:25.36 | bkruse | qtopia on the neo1973 also |
| 19:25.42 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 19:25.47 | bkruse | twisted: how ya been? |
| 19:25.52 | twisted | busy |
| 19:25.57 | *** join/#asterisk jlar (n=chatzill@santana.office-ww.wideideas.net) |
| 19:26.01 | [TK]D-Fender | bkruse: I know, but i couldn't jsutify the costs, and thats raw data. I have unlimited internet on mine for $37+tax all inclusive |
| 19:26.34 | jlar | ~jbot |
| 19:26.35 | jbot | somebody said jbot was a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch |
| 19:26.49 | RoyK | http://www.darwinawards.com/darwin/darwin2007-18.html |
| 19:26.57 | jameswf | ping tzafrir |
| 19:27.08 | tzafrir | פ×מע |
| 19:27.11 | tzafrir | pong |
| 19:27.15 | bkruse | [TK]D-Fender: nice |
| 19:28.57 | RoyK | tzafrir: wtf was that language? |
| 19:28.59 | RoyK | greek? |
| 19:29.05 | mikebee | grandpapadot: take a look at http://www.voip-info.org/wiki-Asterisk+dimensioning , but my guess is you probably just have a single thread that is taking up that much cpu |
| 19:29.30 | plik | Thanks Qwell (delayed as I stepped out for a while) |
| 19:29.41 | Qwell | plik: you're welcome - I forgot what for though :) |
| 19:29.49 | tzafrir | RoyK, Hebrew. |
| 19:29.50 | plik | itsp :) |
| 19:30.07 | grandpapadot | mikebee: Thanks. |
| 19:30.14 | RoyK | tzafrir: oh. you're israeli? |
| 19:30.53 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:31.12 | tzafrir | RoyK, yes |
| 19:31.17 | RoyK | ok |
| 19:31.52 | RoyK | I thought you were from the US |
| 19:33.38 | jlar | seen hax |
| 19:33.49 | *** join/#asterisk drmessano-LT (n=nonya@207.230.140.240) |
| 19:33.55 | jlar | ~seen hax |
| 19:34.03 | jbot | hax is currently on #asterisk, last said: 'jlar: well, i guess that's a pretty good idea... i was really just more interested in having record keeping smtp deliveries, so i don't lose voicemail'. |
| 19:34.14 | grandpapadot | mikebee: Yea, I've seen that. Asterisk doesn't appear to be forking anything and I don't know if it even will. I'm running multiple codecs and about 200 peers, rouoghtly 20-30 calls at any given time. The server is holding up extremely well, no issues, just something I noticed and was wondering if there was anything specific at compile-time or launch-time that would make asterisk thread better. I'm running Ubuntu 6.06LTS and/or Debian. |
| 19:37.08 | *** join/#asterisk ^Migs^ (n=Migs@205.122.50.234) |
| 19:37.21 | mvanbaak | anyone know what this is for: [ Context 'app_dial_gosub_virtual_context' created by 'app_dial' ] |
| 19:38.20 | file | grandpapadot: nope... |
| 19:38.23 | Corydon76-vcch | It's for app_dial |
| 19:39.58 | Corydon76-vcch | mvanbaak: more specifically, it's for use with the U() option of Dial |
| 19:40.06 | robl^ | what's app_dial for? |
| 19:40.19 | *** part/#asterisk anarcat (n=anarcat@pdpc/supporter/active/anarcat) |
| 19:40.36 | grandpapadot | If g729a.so would fork I would be happy.. But I'm not skilled enough to take it on myself.. |
| 19:41.09 | ManxPower | fork? |
| 19:41.27 | drmessano-LT | ~fork |
| 19:41.28 | jbot | spoon! |
| 19:41.32 | drmessano-LT | exactly |
| 19:41.38 | mikebee | grandpapadot: since you're running 1.2, I'm not sure how much multithreading it handles. My understanding however is that some parts of Asterisk multithread, but some do not. |
| 19:41.57 | ManxPower | grandpapadot: Asterisk does not generally "fork" It uses threads, which on some versions of PS shows as a process. |
| 19:42.03 | file | grandpapadot: why? Asterisk does use threads, and the translations of audio using g729 happen in different threads... so the system should distribute those |
| 19:42.18 | robl^ | spork! |
| 19:42.47 | mikebee | grandpapadot: you should be able to do ps -aux | grep asterisk and get a sense of how many threads are running. |
| 19:42.53 | mvanbaak | Corydon76-vcch: thanks |
| 19:43.05 | mvanbaak | found it |
| 19:43.08 | grandpapadot | It's running a single thread. |
| 19:43.11 | ManxPower | mikebee: many versions of ps do NOT show threads. |
| 19:43.16 | mjoyce | ps will show forks() but not threads |
| 19:43.22 | file | it's not running a single thread. |
| 19:43.27 | ManxPower | mjoyce: that depends on the PS |
| 19:43.31 | file | if your Asterisk is running and works it is not using a single thread |
| 19:43.39 | grandpapadot | ps -aux | grep asterisk reports a single process |
| 19:43.46 | grandpapadot | file: Yea, I mean process |
| 19:44.00 | ZaVoid | So anyone got a good reference point for IAX Signaling so i can start reading up on it? |
| 19:44.16 | mikebee | ManxPower: really? odd. Guess I'm only familiar with RH and SuSE. Anything else is out of my realm. |
| 19:44.19 | ManxPower | grandpapadot: try ps -eLf |
| 19:44.27 | ZaVoid | ps -aef | grep asterisk is myfavorite |
| 19:44.29 | RoyK | grandpapadot: killall -9 asterisk :P |
| 19:44.32 | ZaVoid | along with killall asterisk |
| 19:44.39 | ZaVoid | or -9 too :) |
| 19:44.44 | mikebee | heh ZaVoid |
| 19:45.04 | grandpapadot | file: I would think, in a multi-threaded application, that the threads would distribute themselves amongs the available cpu cores exposed by the kernel by the schduler. So my thought was that I missed something at compile time. |
| 19:45.08 | ZaVoid | and kill -9 safe_asterisk |
| 19:45.57 | mjoyce | ps aux | grep asterisk | grep -v grep | awk {' print $2 '} | xargs kill |
| 19:46.01 | ManxPower | grandpapadot: try " ps -eLf" |
| 19:46.07 | ZaVoid | reboot now |
| 19:46.22 | mikebee | halt :P |
| 19:46.27 | grandpapadot | Like 30 processes (threads?) So I wonder why they are all on CPU0? |
| 19:46.29 | ManxPower | grandpapadot: do you see more asterisk entries now? |
| 19:46.36 | mjoyce | grandpapadot - i don't think the linux scheduler is that good, i think it balances by PID |
| 19:46.42 | *** join/#asterisk lemanal (n=lemanal@cpe-066-026-085-055.nc.res.rr.com) |
| 19:46.45 | ZaVoid | hey ManxPower you got any good reference i can read up on for iax signaling? |
| 19:46.58 | ManxPower | mjoyce: As threads are usually all the same PID, I doubt that is the case. |
| 19:47.06 | ManxPower | ZaVoid: The IAX2 proto-RFC. |
| 19:47.22 | ManxPower | grandpapadot: cat /proc/interrupts |
| 19:47.23 | ZaVoid | proto rfc? |
| 19:47.27 | *** join/#asterisk mintee (n=mintee@72-165-177-90.dia.static.qwest.net) |
| 19:47.27 | ZaVoid | does it have a rfc #? |
| 19:47.33 | ManxPower | ZaVoid: ask file |
| 19:47.48 | file | I know it had a draft... |
| 19:47.57 | ZaVoid | hi file |
| 19:48.02 | ManxPower | the draft is what I was thinking of |
| 19:48.03 | BBHoss | iax2 is draft |
| 19:48.04 | *** join/#asterisk pgarcia (n=pedgarci@201.72.232.157) |
| 19:48.13 | ZaVoid | right |
| 19:48.16 | BBHoss | probably always will be... |
| 19:48.32 | file | I don't know where it is but a Google can tell you |
| 19:48.44 | grandpapadot | mjoyce: So it would have to be a fork() to get threads to another CPU? odd |
| 19:49.00 | ZaVoid | yeah i omitted the draft part |
| 19:49.01 | ZaVoid | http://tools.ietf.org/id/draft-ietf-enum-iax-02.txt |
| 19:49.03 | ZaVoid | there it is thanks guys |
| 19:49.17 | grandpapadot | You know, it's hard to complain, the software is obviously great at what it does. |
| 19:49.23 | grandpapadot | And the price is right. |
| 19:49.45 | ManxPower | grandpapadot: I assume you are running an SMP kernel? |
| 19:50.22 | BBHoss | i know a few years ago, asterisk wouldnt run on dual core dual proc machines |
| 19:50.30 | grandpapadot | ManxPower: Yep. |
| 19:50.34 | BBHoss | well, it wouldnt run on the second proc |
| 19:51.00 | ManxPower | BBHoss: that is complete and utter bullshit. |
| 19:51.17 | BBHoss | see for yourself: |
| 19:51.23 | BBHoss | http://lists.digium.com/pipermail/asterisk-users/2006-March/142186.html |
| 19:51.39 | BBHoss | read the quote |
| 19:52.10 | ZaVoid | so file.. when i have a device registered via iax to an asterisk box.. and i'm routing his calls to a sip termination endpoint.. that media would have to be routed through the asterisk box regardless right? |
| 19:52.12 | BBHoss | more: http://lists.digium.com/pipermail/asterisk-users/2006-March/142186.html |
| 19:52.15 | ManxPower | BBHoss: you mean the part where someone claims the exact same thing you are claiming? |
| 19:52.17 | drmessano-LT | Go to the next message |
| 19:52.21 | drmessano-LT | http://lists.digium.com/pipermail/asterisk-users/2006-March/142275.html |
| 19:52.25 | file | ZaVoid: ...yes |
| 19:52.31 | ZaVoid | ok thought so |
| 19:52.37 | ManxPower | let me know when you found the message you are looking for. |
| 19:52.38 | BBHoss | but these articles are from 06, so i don't know if they are still accurate |
| 19:52.54 | grandpapadot | ManxPower: Do you run multi-core asterisk implementations and see the load being distribute? It might just be the normal m.o. |
| 19:52.58 | drmessano-LT | BBHoss, read the one I posted |
| 19:53.23 | ManxPower | You understand that the multi-threaded application or multi-proccess application's SMP, etc stuff is all handled by the OS, right? The application has nothing to do with hit. |
| 19:53.43 | grandpapadot | ManxPower: Clearly, which is why I asked the question to begin with. |
| 19:54.10 | mintee | heh |
| 19:54.10 | BBHoss | drmessano-LT, hmm |
| 19:54.11 | grandpapadot | ManxPower: It didn't make sense to me to have 15% average on CPU0 which the others averaged < .1% |
| 19:54.17 | drmessano-LT | Finding one post that says "Asterisk is made of cookies" doesn't make it so |
| 19:54.34 | ZaVoid | ahh i found the rifght draft now http://www.tools.ietf.org/html/draft-guy-iax-03 |
| 19:54.44 | ManxPower | drmessano's link is a good one. |
| 19:55.20 | ManxPower | grandpapadot: did you read the message drmessano-LT linked to? |
| 19:55.21 | grandpapadot | ManxPower: I know it won't be equal, but someone distrubted. Again, thinking it was just a compile-time problem. Maybe specifing ARCH=686 would make it better or something, dunno, I'm clueless. |
| 19:55.29 | BBHoss | im not arguing against it, it was just something i stumbled across |
| 19:55.31 | drmessano-LT | ManxPower: It was hard, I clicked "Next Message" from the one he posted |
| 19:55.40 | drmessano-LT | Wow.. how that works |
| 19:56.12 | ManxPower | drmessano: he should have done that before claiming "asterisk didn't work on multiproc multicore systems" |
| 19:56.13 | drmessano-LT | I always go a few ahead and a few back when google returns a thread from somewhere.. saves time later arguing :) |
| 19:57.04 | grandpapadot | Yep, that's what I would assume (The post) which takes me back to something's wrong with either my OS or my Asterisk deployment on these servers. |
| 19:57.07 | ManxPower | drmessano-LT: I do that before I cite a message as a source. |
| 19:57.17 | drmessano-LT | Don't argue Wikipedia in court either.. OOPS, I just changed the article on you |
| 19:57.46 | ManxPower | grandpapadot: or perhaps your asterisk is spending a lot of time doing some of the few things that are not done in separate threads. |
| 19:58.23 | ManxPower | There HAVE been some issues with HyperThreading. |
| 19:58.37 | grandpapadot | Well, we're transcoding a lot, a lot of g729, recording (mixmonitor) calls, a lot of switching between contexts, I'd think we're doing a lot of different things all the time. |
| 19:58.37 | drmessano-LT | ManxPower: I get frustrated when I refer someone to something and have to argue for 10 minutes that this wasn't the FIRST instance I came across, it was the 5th or 20th... but seeing how most people Google, I try to be patient :) |
| 19:58.39 | ManxPower | But as anyone with a brain knows, Hyperthreading is not SMP, dual core, or dual proc |
| 19:58.48 | BBHoss | ah 1.6 is now using hash tables for for SIP? sweet |
| 19:59.37 | grandpapadot | This is a dual-dual core system (quade core) opterons, smp kernel, googled the heck out of it, was looking for insight in channel. |
| 20:00.10 | ManxPower | grandpapadot: make sure the utilities you are using do not have issues with SMP. |
| 20:00.26 | BBHoss | grandpapadot, i read something about a program called oprofile, you might want to check it out |
| 20:00.33 | ManxPower | I recall some version of ifconfig that always showed frame errors on a specific type of card. |
| 20:00.39 | *** join/#asterisk beek (n=klinebl@65.211.106.243) |
| 20:00.49 | ManxPower | there were no frame errors, but ifconfig program wa lying |
| 20:00.55 | grandpapadot | thanks, all. |
| 20:01.31 | drmessano-LT | Windows 3.1 only sees one CPU :( |
| 20:01.49 | BBHoss | on noez no flying toasters! |
| 20:01.51 | *** join/#asterisk zapp-branigan (n=zapp_bra@9.218.216.87.static.jazztel.es) |
| 20:02.19 | grandpapadot | drmessano-LT: eh? |
| 20:02.20 | drmessano-LT | After Dark.. almost as cool as watching Asterisk debug |
| 20:02.27 | BBHoss | lol |
| 20:02.44 | drmessano-LT | If you get really sto..... eh nevermind |
| 20:02.45 | BBHoss | gnome-screensaver has one, but its not as cool as the original |
| 20:03.00 | zapp-branigan | hi one question can be more than 1 iax2 peer in the same adsl (ip behind a nat) |
| 20:03.10 | BBHoss | well, if you got electricsheep you'd really be screwed |
| 20:03.44 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:04.15 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 20:04.36 | ZPertee | wil voip work satellite internet. satellite is my only option as I am in a rural area |
| 20:04.46 | grandpapadot | drmessano-LT: a single post is hardly gospel, I figured I would get some answers from the experience in the channel. |
| 20:05.09 | BBHoss | ZPertee, depends on network conditions, but its not really ideal |
| 20:05.12 | timeshell | Quality would depend on bandwidth and latency |
| 20:05.20 | drmessano-LT | grandpapadot: I'm with ya, man |
| 20:05.20 | grandpapadot | ZPertee: Latency would kill it. |
| 20:05.34 | plik | jbot: itsplist-uk is UK based ITSps include www.voiptalk.org/ www.voipon.co.uk/ www.gradwell.com/ and a few other tinpot companies you can dig up with google. |
| 20:05.34 | jbot | plik: okay |
| 20:05.52 | BBHoss | i've heard that satellite has gotten a bit better now though |
| 20:06.31 | BBHoss | ZPertee, what kind of results do you get from this:http://myvoipspeed.visualware.com/ test |
| 20:06.37 | grandpapadot | ZPertee: 600-1200ms latency, you would be talking over each other. You would have to "Hey, did you see Lost on TV last night, over. <pause> Yes, over.<pause>", etc |
| 20:06.42 | *** join/#asterisk angryuser[A] (i=nononon@df01t2-212-195-198-128.d4.club-internet.fr) |
| 20:07.26 | ZPertee | ok. I'll have try that link |
| 20:08.22 | ManxPower | 600ms - 1200ms is pretty typical for satellite |
| 20:08.36 | ManxPower | at least for consumer stuff |
| 20:09.55 | BBHoss | yeah i have a buddy out in NM that has a microwave uplink at 100Mbit :) |
| 20:10.12 | ManxPower | microwave and satellite are the same1 |
| 20:10.32 | [T]ank | i am trying to do a 468* salute on a polycom 501. I am trying to reset it to factory default because It is no longer able to reach the ftp server it is pointed to. It does not seem to be doing a factory reset, it still comes up and give an error that it cannot load its configs. Is there a different way to factory reset this if I cannot fully boot the phone? |
| 20:12.02 | *** join/#asterisk CVirus (n=GoD@196.205.192.211) |
| 20:12.04 | mjoyce | i think its #468 |
| 20:12.17 | drmessano-LT | Microwave and satellite are the same only if the path is 200 or so miles :) |
| 20:13.15 | [T]ank | #468 is not doing anything. *468 resets it just not fully |
| 20:14.26 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 20:14.33 | mocker | Anyone having any issues w/ Teliax today? |
| 20:15.14 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-71-252.socal.res.rr.com) |
| 20:17.10 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 20:19.22 | ManxPower | [T]ank: 468* does not change the bootrom settings, but it changes everything else. |
| 20:19.42 | [T]ank | ManxPower: is there anything that does change the bootrom? |
| 20:19.50 | ManxPower | [T]ank: are you sure it's not getting it's TFTP server via DHCP? |
| 20:19.59 | [T]ank | yeah. |
| 20:20.01 | ManxPower | [T]ank: go into the bootrom and change it. |
| 20:20.09 | ManxPower | "setup" option on the FIRST screen |
| 20:20.12 | [T]ank | ok |
| 20:20.46 | [T]ank | so if i set it to an ip address of 0.0.0.0, shouldnt it just boot up without trying ftp? |
| 20:20.54 | [T]ank | that is what I have tried to do at least. |
| 20:21.02 | [T]ank | is there something different to do? |
| 20:22.16 | ManxPower | [T]ank: you would have to read the ADMIN MANUAL |
| 20:23.37 | De_Mon | There. I know have a working substitute.agi perl script that'll do global search/replace. bleh |
| 20:23.44 | De_Mon | s/k// |
| 20:23.55 | *** part/#asterisk pgarcia (n=pedgarci@201.72.232.157) |
| 20:25.51 | *** join/#asterisk saftsack (n=saftsack@p4FC75E9F.dip.t-dialin.net) |
| 20:30.17 | *** join/#asterisk riksta (n=rick@rhamnett.plus.com) |
| 20:30.26 | freezey | [TK]D-Fender: is it just the 5400 series phones it doesnt work with or does it not work with all the hardware that i have?? |
| 20:30.49 | *** join/#asterisk mltlnx (n=mltlnx@209.10.153.194) |
| 20:31.57 | zapp-branigan | hi one question can be more than 1 iax2 peer in the same adsl (behind router) or the same port of 4569 can't be shared ? |
| 20:32.15 | *** join/#asterisk fiXXXerMet (n=kjohnson@dsl092-156-002.wdc2.dsl.speakeasy.net) |
| 20:32.54 | riksta | Hi, does anyone have experience in setting up the latest a2billing? V1.3.2 Stable? I've followed the howto to the letter (twice for good measure) on a clean Xen domU install, and when the AGI runs (which i can see that it does in the console) it just hangs up after 4 seconds, I don't get the expected "please enter your pin number" |
| 20:32.59 | zapp-branigan | i see that some times the asterisk give the peers another port who can i do this ? |
| 20:33.13 | riksta | I also turned up the debug level to 3 and i get nothing extra |
| 20:36.37 | [TK]D-Fender | freezey: All of it |
| 20:37.22 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 20:37.38 | [TK]D-Fender | riksta: GUI's are not supported here. Please go to their channel |
| 20:38.03 | tzanger | groovy, I'm confirmed to be in montreal now next week |
| 20:38.04 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 20:38.14 | [TK]D-Fender | tzanger: How long? |
| 20:38.30 | tzanger | unfortunately the motel I'm staying at has people who speak very poor english, and my french is equally if not more poor |
| 20:39.02 | tzanger | [TK]D-Fender: arriving 2330 this monday, depart 1700 friday |
| 20:39.07 | tzanger | then back again the week after most likely |
| 20:39.14 | tzanger | not sure after that |
| 20:39.36 | tzanger | hello man |
| 20:39.53 | tzanger | ManxPower even |
| 20:40.53 | [TK]D-Fender | tzanger: Cool. Definietly cab-worthy that late for arrival. |
| 20:41.10 | [TK]D-Fender | tzanger: e-mail your contact info and we'll pick a night. |
| 20:42.27 | tzanger | [TK]D-Fender: indeed. I won't rent a car for $200 for the week, that's insane just to go back and forth between the hotel and the job site |
| 20:43.24 | riksta | [TK]D-Fender: cool |
| 20:44.03 | [TK]D-Fender | Actually where is the job relative to the hotel? |
| 20:44.30 | tzanger | [TK]D-Fender: the job is on that little island near verdun (commerce road?) and the hotel is in brossard I think |
| 20:44.45 | [TK]D-Fender | Brossard!??! |
| 20:44.49 | [TK]D-Fender | tzayou NEED a car then |
| 20:44.50 | hmmhesays | ok this blackfin is only answering my zap calls some of the time |
| 20:45.34 | tzanger | yep, 8700 boulevard marie-victorin in brossard |
| 20:45.47 | robl^ | hmmhesays: maybe it only answers them when they are ringing? ;-) |
| 20:45.58 | tzanger | touchtunes is at 3 place du commerce in verdun |
| 20:46.28 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:46.33 | tzanger | 15 minute drive google's saying |
| 20:46.46 | [TK]D-Fender | tzanger: OMG no. |
| 20:47.04 | [TK]D-Fender | tzanger: Lying sack of shit you have no idea how bad the inbound bridge is in the morning... |
| 20:47.10 | [TK]D-Fender | (Google, not you :p |
| 20:47.18 | tzanger | uh oh |
| 20:47.50 | [TK]D-Fender | tzanger: What ^%$#tard put you up off-island? |
| 20:47.57 | tzanger | *coughs* me |
| 20:48.03 | tzanger | I was looking for cheap hotels |
| 20:48.09 | [TK]D-Fender | tzanger: It'll cost you! :p |
| 20:48.35 | tzanger | oh dear |
| 20:48.48 | [TK]D-Fender | Can you correct? |
| 20:49.04 | tzanger | well I will see... I might be able to but I haven't been able to find a similar-rate hotel |
| 20:49.05 | [TK]D-Fender | or have they smashed your card 10 ways from tuesday already? :) |
| 20:49.10 | tzanger | hahaha no no |
| 20:49.14 | drmessano-LT | If the inbound bridge is bad as the inbound calls from gizmo, I would guess "complete gridlock" |
| 20:49.23 | [TK]D-Fender | tzanger: Hunt fast. |
| 20:49.52 | zapp-branigan | hi one question can be more than 1 iax2 peer in the same adsl (behind router) or the same port of 4569 can't be shared ? |
| 20:49.53 | tzanger | it's that nasty eh? |
| 20:50.18 | ManxPower | zapp-branigan: the source port is different anyway, so it should not matter |
| 20:50.28 | [TK]D-Fender | tzanger: It can add an hour to your day easily |
| 20:50.36 | *** join/#asterisk DarWin_vcch (n=daryl@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 20:52.03 | J4k3 | helicopters are king |
| 20:52.13 | J4k3 | land transportation is for suckers! |
| 20:52.48 | zapp-branigan | ManxPower then more than one peer can be behind a router ? |
| 20:52.49 | tzanger | [TK]D-Fender: yikes |
| 20:53.43 | ManxPower | zapp-branigan: there SHOULD not be a problem with that, but we use all SIP for the past 2 years |
| 20:54.00 | hmmhesays | robl^, haha |
| 20:54.06 | hmmhesays | no it seems it is not dropping the line when it should |
| 20:55.28 | drmessano-LT | black helicopter FTFW |
| 20:55.30 | zapp-branigan | ManxPower is better sip tan Iax ? what is the reason for use SIp ? |
| 20:55.45 | zapp-branigan | using sip you need rtp |
| 20:56.13 | tzanger | [TK]D-Fender: it looks like I'm screwed eithe rway... I have to cross a bridge to get to ile des soeurs no matter whether I come from brossard or from quebec |
| 20:56.19 | [TK]D-Fender | tzanger: Yeah, thats 15minutes at midnight. |
| 20:56.31 | tzanger | yuck |
| 20:56.38 | [TK]D-Fender | tzanger: Yup. |
| 20:57.01 | [TK]D-Fender | tzanger: You are better off with a car then a cab if you want to stay there. |
| 20:57.07 | [TK]D-Fender | MUCH |
| 20:57.11 | drmessano-LT | [TK]D-Fender: thanks for your help earlier, BTW.. Still not working, but when I get home I am gonna try sacrificing a few goats |
| 20:57.19 | tzanger | ok if I'm screwed from Brossard and screwed from Montreal, what's the difference? |
| 20:57.26 | [TK]D-Fender | drmessano-LT: You gain wisdom child! |
| 20:57.34 | ManxPower | zapp-branigan: we had issues with IAX2. |
| 20:57.48 | [TK]D-Fender | tzanger: No... you're screwed beause you have to go from one to the other. |
| 20:57.58 | ManxPower | also, assuming no nat is incolved, you could, in theory do reinvites between phones connected to two different asterisk servers |
| 20:58.04 | tzanger | well no, what I'm saying is that the destination is on the island |
| 20:58.20 | tzanger | it *looks* like I have to take a bridge to get there from Montreal, and still have to take a bridge to get there from Brossard |
| 20:58.25 | [TK]D-Fender | tzanger: its not the destination, its the fact that bridge is BETWEEN them :) |
| 20:58.53 | [TK]D-Fender | tzanger: So yeah, it'll cost you plenty of time,a nd for sure car rental will be better than cab |
| 20:59.08 | tzanger | wow, that sucks balls |
| 20:59.25 | [TK]D-Fender | tzanger: Find something close, hit up public transit for a week |
| 20:59.35 | zapp-branigan | then ManxPower the good of the iax is the reinvites but what was the problem of the Iax ? |
| 20:59.40 | tzanger | that *is* about as close as I can get |
| 20:59.46 | tzanger | at least according to google |
| 20:59.47 | [TK]D-Fender | tzanger: e-mail me the specifics and I'll see if I can find you something tonight |
| 20:59.58 | [TK]D-Fender | google = SMRT |
| 21:00.20 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 21:02.39 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) |
| 21:02.55 | tzanger | I can find all kinds of nice fancy expensive hotels there for sure heh |
| 21:03.18 | J4k3 | priceline! william shatner is never wrong! |
| 21:04.55 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:04.55 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:05.28 | drmessano-LT | PRICELINE NEGOTIATOR! |
| 21:05.53 | J4k3 | genitalia annihilator! |
| 21:05.54 | freezey | [TK]D-Fender: ok so if i go out and i buy all cisco VOIP phones.... what else would i need from there? |
| 21:06.02 | drmessano-LT | Shatner got a room for me with 2 free ho's and they paid ME $10 to stay there! |
| 21:06.24 | J4k3 | freezey: some bridge devices to go from cisco's f'd up sip implementation to asterisk? :D |
| 21:06.53 | freezey | J4k3: haha this just sucks that my entire avaya system is not compatible |
| 21:07.08 | Qwell | freezey: buy Polycom |
| 21:07.09 | Qwell | not Cisco |
| 21:07.21 | Qwell | you will be much happier overall |
| 21:07.54 | drmessano-LT | Avaya isn't Y2K compliant.. meaning they haven't implemented anything that the rest of the world hasn't been doing since 2000 |
| 21:07.59 | Daviey | buy Linksys, Polycom, Aastra - just not cisco branded stuff please |
| 21:08.18 | hmmhesays | tzanger you ever have a problem with one of these blackfins not freeing its fxo channel? |
| 21:08.28 | freezey | Ok so i go out and i buy all polycom phones for my office... after that what else is going to be needed? |
| 21:08.31 | hmmhesays | I'm getting all kinds of crazy errors, using the xorcom firmware |
| 21:08.34 | freezey | am i going to need a VOIP provider? |
| 21:08.38 | freezey | what kind of switch will i need? |
| 21:08.49 | tzafrir | hmmhesays, like what? |
| 21:09.01 | tzafrir | what version of zaptel? |
| 21:09.09 | J4k3 | freezey: you're going to need a PC and either an itsp or trunks (preferably not analog ones) from your local telco. |
| 21:09.15 | ManxPower | freezey: that all depends on how you design your system |
| 21:09.16 | tzanger | hmmhesays: I've never used Rowe's hardare |
| 21:09.17 | tzanger | er hardware |
| 21:09.27 | tzanger | all of my stuff is high density (500+ ports) |
| 21:09.30 | freezey | well right now i have these dam avaya trunks |
| 21:09.38 | tzanger | actually that's untrue, the lowest is 288 ports |
| 21:09.51 | Qwell | tzanger: why 288? |
| 21:09.56 | Qwell | oh, nevermind |
| 21:10.01 | hmmhesays | gotcha |
| 21:10.02 | Qwell | 12 T1s? |
| 21:10.04 | Qwell | 24* |
| 21:10.06 | Qwell | 12* |
| 21:10.12 | hmmhesays | um, whatever zaptel version goes with asterisk 1.4.13 |
| 21:10.16 | [TK]D-Fender | freezey: Stop. NOW. You have no clue about what you have, or What kind of solution would be best. You need to sit down with a consultant. |
| 21:10.33 | ManxPower | freezey: analog or digital avaya trunks? You don't even know enough to ask the right questions at this point. |
| 21:10.36 | mjoyce | 500 calls on asterisk is scary :( |
| 21:10.44 | freezey | i am just talking seeing what would be the best isnt that the point of these channels? TK? |
| 21:10.54 | tzanger | Qwell: maybe, that's just how the system was... 4 extensions per card, 8 cards per shelf, 9 shelves |
| 21:10.55 | freezey | Digital |
| 21:10.58 | J4k3 | mjoyce: less scary than 5000 POTS lines on a DMS-10 :) |
| 21:11.03 | Qwell | tzanger: ahh |
| 21:11.07 | ManxPower | freezey: not really. This channel is learn bout asterisk? |
| 21:11.14 | mjoyce | J4k3: h4w |
| 21:11.16 | Qwell | tzanger: that your "magic" "reused" box? |
| 21:11.27 | ManxPower | freezey: and you know what my next question is, of course. |
| 21:11.43 | freezey | nope |
| 21:11.45 | freezey | ha |
| 21:11.47 | tzafrir | hmmhesays, what errors do you see? what is the output of lsusb? |
| 21:11.51 | J4k3 | mjoyce: you laugh, I live with it every day! :E |
| 21:11.52 | ManxPower | what KIND of digital lines? |
| 21:11.59 | freezey | t1 |
| 21:12.02 | ManxPower | what SIGNALLING |
| 21:12.03 | hmmhesays | tzafrir, one sec i'm pastebinning |
| 21:12.06 | ManxPower | So, not PRI then? |
| 21:12.16 | freezey | yeah this i am unsure of |
| 21:12.23 | mjoyce | ManxPower: im guessing ami / e_m |
| 21:12.24 | ManxPower | freezey: then you need to stop until you are sure. |
| 21:12.26 | mjoyce | lawlz! |
| 21:12.41 | freezey | alright.. the system was inplace before i got here |
| 21:12.42 | tzanger | Qwell: one of them. the other is a nortel box, and there's anothe rone yet I haven't got to |
| 21:12.48 | freezey | so i guess i can call the company and find out exactly what we have |
| 21:12.49 | Qwell | fun |
| 21:12.51 | ManxPower | freezey: Here, drink this bottle with a skill and crossbones on it. I THINK it's water. |
| 21:12.57 | freezey | ha |
| 21:13.00 | mjoyce | freezey ask for a 'cut sheet' from your carrier |
| 21:13.05 | Qwell | tzanger: you gonna end up open sourcing the stuff you've done? |
| 21:13.14 | robl^ | tzanger: what kind of nortel? Norstar? BCM? Meridian? |
| 21:13.16 | Qwell | people would be jumping all over it, I'm sure |
| 21:13.31 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 21:14.27 | drmessano-LT | ManxPower: was "skill" a typo or a freudian slip? :) |
| 21:14.34 | freezey | mjoyce: the cut sheet will have all the info i am looking for? that will assist in answering some of these questions? |
| 21:14.42 | ManxPower | drmessano-LT: could it be both? |
| 21:15.06 | ManxPower | freezey: a cut sheet is what any PBX person will ask for when they spec out a PBX |
| 21:15.13 | freezey | ok cool |
| 21:15.16 | mjoyce | freezey: yes, it should have the following ifnormation that is crucial for setting up asterisk with a t1: Signaling, Encoding, Framing, (Switchtype if ISDN) and what channels are in use |
| 21:15.16 | freezey | let me give them a ring |
| 21:15.33 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 21:15.41 | freezey | the person before me just didnt leave any information on anything here |
| 21:15.42 | robl^ | freezey: yes.. its the documentation of all things the tech needs to program a PBX |
| 21:15.47 | mjoyce | freezey: pray that you have signaling: pri encoding: b8zs framing:esf |
| 21:15.47 | freezey | k |
| 21:15.48 | drmessano-LT | ManxPower: Sure :) |
| 21:16.47 | hmmhesays | tzafrir, http://www.pastebin.ca/886692 |
| 21:16.48 | tzanger | Qwell: hoping to, not sure |
| 21:17.00 | tzanger | robl^: unfortunately it's the old old old old nortel stuff right now M1 I think it's called |
| 21:17.25 | tzanger | I hope to take that to both Meridian and MICS though |
| 21:17.43 | robl^ | Meridian 1? |
| 21:17.45 | hmmhesays | after the first call into the fxo port it hangs |
| 21:17.48 | tzafrir | hmmhesays, try using: module unload chan_zap.so |
| 21:17.58 | tzafrir | and then: module load chan_zap.so |
| 21:18.03 | tzafrir | instead of zap restart |
| 21:18.17 | hmmhesays | [Jan 31 21:18:02] WARNING[1099]: loader.c:472 ast_unload_resource: Unloading embedded modules is not supported. |
| 21:18.47 | tzafrir | run zap restart again, maybe? |
| 21:20.26 | tzanger | heh maybe I should just rent a shithole apartment in iles des soeurs |
| 21:20.37 | hmmhesays | when the call comes in also it just drops out of nowhere, I noticed in the debug there is a polarity reversal |
| 21:21.12 | *** join/#asterisk VitoCorleon (n=Owner@rrcs-76-79-244-73.west.biz.rr.com) |
| 21:21.14 | VitoCorleon | hey guys |
| 21:21.52 | tzafrir | right, this is something resolved in 1.4.8 |
| 21:22.04 | x86 | hmmhesays: polarity on tip and ring POTS doesn't really ever matter, usually |
| 21:22.14 | VitoCorleon | How do i make it when a incoming service provider gets a call forwards to an outside number? ( im using asterisk now) |
| 21:22.21 | [TK]D-Fender | x86: cONSIDERING RINGING IS ac, NOT REALLY... |
| 21:22.34 | x86 | [TK]D-Fender: DONT TASE ME BRO! |
| 21:22.45 | [TK]D-Fender | VitoCorleon: Please visit their support channel. GUI's are not supported here |
| 21:22.58 | hmmhesays | unless you're using polarity reversal for answer/disconnect supervision |
| 21:23.00 | tzafrir | hmmhesays, hmmm.... disable polarity reversal support for now. or try building from a newer copy, if you can |
| 21:23.12 | VitoCorleon | it seems like they are not supported in there channels also, no one is alive in there channels |
| 21:24.02 | hmmhesays | hmm how do I disable that? |
| 21:24.16 | tzafrir | no, it's a bug in the polarity detection. The driver emits a false event. It has already been fixed in svn.digium.com . And IIRC also in latest firmware |
| 21:24.17 | [TK]D-Fender | ok, BBL |
| 21:24.35 | drmessano-LT | Wow.. I just grabbed a cup from the water cooler.. the slogan for the company that provides our water... "Water you drinking?" |
| 21:24.47 | drmessano-LT | "Not sure if it's water or not, actually" |
| 21:26.32 | drmessano-LT | x86, polarity usually only matters if the Delorean takes you back to 1963 |
| 21:27.36 | J4k3 | depends on your POTS, too |
| 21:27.50 | J4k3 | not all POTS is proper 2-wire high/low |
| 21:28.03 | jm|laptop | hmm |
| 21:28.11 | J4k3 | I've seen POTS where one side of the pair is bonded to ground |
| 21:28.13 | jm|laptop | how do I SET two variables in a .call file? |
| 21:28.19 | Daviey | 4 wire POTS ftl |
| 21:28.36 | J4k3 | if you're going to waste 4 wires, hook up a T1/E1 :P |
| 21:28.50 | *** part/#asterisk RoyK (n=roy@ip-85-4-149-91.dialup.ice.no) |
| 21:28.52 | JT | all POTS has one side of the pair bonded to ground |
| 21:28.53 | drmessano-LT | lol |
| 21:29.02 | JT | and most T1s and E1s are delivered over 2 wire SHDSL these days |
| 21:29.22 | JT | on POTS the positive side is bonded to ground |
| 21:29.31 | JT | hence why exchange battery voltage is -48VDC |
| 21:29.35 | J4k3 | JT: actually, the POTS to my house floats +30/-30, unless something is very wrong. |
| 21:29.37 | JT | with reference to ground |
| 21:29.38 | J4k3 | (on hook) |
| 21:29.49 | JT | J4k3: sounds like something is wrong |
| 21:30.01 | Daviey | on our POTS, the provider can't make their mind up which is +ve and -ve :) |
| 21:30.06 | JT | it should be -48VDC with reference to earth, nominally |
| 21:30.10 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 21:30.10 | J4k3 | JT: when one side is bonded to ground, you get oodles of 60 hz hum. |
| 21:30.22 | mjoyce | i have seen low voltage like that, it is usually when the line is bridged through a fax machine or something like that |
| 21:30.28 | JT | it's meant to be bonded to ground at only one place, the exchange |
| 21:30.51 | JT | bonding positive to ground reduces corrosion of the copper cables |
| 21:31.05 | drmessano-LT | or a fax switch, mjoyce.. they love 20 or 30 volts on hook lol |
| 21:31.09 | Daviey | Just make sure whilst wiring a socket you don't put a wire in your mouth, when someone rings.. I've done that :) |
| 21:31.22 | mjoyce | 200v, ouch! |
| 21:31.22 | Qwell | Daviey: same...sucked |
| 21:31.27 | Daviey | :) |
| 21:31.29 | Qwell | mjoyce: 90ish |
| 21:31.34 | Daviey | fizz fizz, fizz fizz |
| 21:31.40 | J4k3 | Daviey: I was rewiring a dialup ISP due to lightning damage... 20+ POTS lines ringing at once and I slipped and put my hand in the punchdown block |
| 21:31.52 | J4k3 | that *sucked* |
| 21:31.57 | JT | mjoyce: 90VAC @16-20Hz |
| 21:31.59 | JT | not 200v |
| 21:32.06 | x86 | J4k3: i've done that myself ;) |
| 21:32.06 | drmessano-LT | Daviey.. My first wiring experience as a 12 yr old was cut short due to a sister with friends ringing our house every 5 minutes |
| 21:32.07 | mjoyce | word, i always thought it was 200 |
| 21:32.20 | x86 | J4k3: that's when the punchblock "bites" you ;) |
| 21:32.24 | Qwell | 90 is with boostringer, isn't it? usually 45? |
| 21:32.28 | JT | no |
| 21:32.31 | JT | 90VAC is standard |
| 21:32.34 | Qwell | oh |
| 21:32.36 | jblack | hallo |
| 21:32.36 | drmessano-LT | 90 is standard |
| 21:32.38 | drmessano-LT | 90/20 |
| 21:32.43 | JT | sure telephony cards put out pissweak voltages ;) |
| 21:32.45 | VitoCorleon | how can i make an extension that will forward all of its calls to outside number? |
| 21:32.46 | JT | and ATAs too |
| 21:32.47 | J4k3 | x86: it definetly bit, and spit me out |
| 21:33.07 | x86 | heh |
| 21:33.18 | J4k3 | T/R voltages here are always at least 60v |
| 21:33.18 | x86 | good thing it let you go :P |
| 21:33.32 | Daviey | VitoCorleon: It won't be an extension.. just an entry in extensions.conf |
| 21:33.38 | J4k3 | but I'm only 8kft from the remote... I'm sure at 30kft its a little closer to 48 ;) |
| 21:33.42 | drmessano-LT | A neon ring indicator won't flash over until it hits about 75v.. and you really need 85 or so to get them to work |
| 21:33.53 | VitoCorleon | oh and how can i get a sample of the text of the entry? |
| 21:34.03 | J4k3 | plus, how else do you dry out your brand new battery bank except by slamming too much voltage toward it? |
| 21:34.17 | Daviey | VitoCorleon: Look for how you'd normally place an outgoing call :) |
| 21:34.26 | drmessano-LT | A PAP2 on 70v default won't flicker a neon ringer |
| 21:34.54 | Daviey | VitoCorleon: then change the expression from XXX. etc to 500,1 => |
| 21:35.12 | VitoCorleon | exten=_9XXXXXXX!,1,Macro(trunkdial,${trunk_1}/1818${EXTEN:0}) ---- would this work? |
| 21:35.59 | J4k3 | drmessano-LT: that'd explain why the old desk phone I used on a PAP2 sounded like it was dying when it rang. |
| 21:36.01 | Daviey | VitoCorleon: yeah, but change ${EXTEN:0} to the real number |
| 21:36.05 | drmessano-LT | lol |
| 21:36.12 | drmessano-LT | thats part of it |
| 21:36.18 | VitoCorleon | Daviey, how about this -- exten=Macro(trunkdial,${trunk_1}/18182649160) |
| 21:36.21 | drmessano-LT | They're also a little weak on bell ringers |
| 21:36.28 | drmessano-LT | They can handle 5 or 6 electronic ones |
| 21:36.34 | J4k3 | yeah |
| 21:36.35 | drmessano-LT | But one bell will strangle it |
| 21:36.57 | drmessano-LT | If you set them at 90v they do a bit better |
| 21:37.04 | drmessano-LT | 70v is just teh suxor |
| 21:37.46 | Daviey | VitoCorleon: yeah, but you need to change the _9XXXXXXX! into 500, if you want 500 to forward to that |
| 21:38.03 | VitoCorleon | ah! k |
| 21:39.05 | VitoCorleon | Daviey, would this make all calls from that trunk forward http://pastebin.ca/886716 |
| 21:41.02 | GrumpManAtWork | J4k3: solution = iphone ;) |
| 21:41.19 | J4k3 | GrumpManAtWork: unluckily GSM isn't usable in north america, unless you never leave major cities. |
| 21:41.25 | J4k3 | or stay on the most major of highways |
| 21:41.44 | J4k3 | otherwise AT&T's coverage is pretty much overpriced hotspot coverage. |
| 21:42.06 | Daviey | VitoCorleon: suck it and see :) |
| 21:42.23 | VitoCorleon | lol |
| 21:42.41 | J4k3 | plus, I've gotten kinda used to actually having reasonably fast internet to my phone... EDGE lost its cool factor in 2002 or so |
| 21:42.46 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:42.52 | freezey | mjoyce: ok i got the cut sheet from my service provider |
| 21:43.02 | Daviey | was EDGE ever cool? |
| 21:43.15 | freezey | mjoyce: now i have a clearer vision of what i am looking at |
| 21:43.18 | J4k3 | I dunno, it was glorified multilink PPP over GPRS as far as I could ever tell :P |
| 21:43.36 | J4k3 | at least thats what it ran like :P |
| 21:43.45 | J4k3 | SHOTGUN CELLULAR TECHNOLOGIES |
| 21:43.47 | aashi_france | i have a question, can ASTERISK be used as session border controller ??? |
| 21:43.47 | Daviey | J4k3: I say that, but any wireless internet here is a rip off compared to you guys |
| 21:44.12 | J4k3 | Daviey: eh... I pay about $10 USD/GB for my cellular data |
| 21:44.48 | jameswf | EDGE sucks |
| 21:44.49 | J4k3 | $45/month (above my contractually required voice plan), but I only get 5GB/month with that |
| 21:44.56 | J4k3 | Verizon has a nasty quota. |
| 21:45.32 | J4k3 | but, if you want to rid yourself of verizon, all you have to do is download >5GB in a month |
| 21:45.57 | lirakis_work | later all |
| 21:46.00 | *** part/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 21:46.01 | jameswf | thats alot of porn for a cell phone |
| 21:46.04 | Qwell | UnixDog: see ya |
| 21:46.14 | jameswf | lol bye |
| 21:46.24 | J4k3 | I use like 50MB of data on the phone per month, tops |
| 21:46.29 | J4k3 | and thats usually windows live messenger voice chatting |
| 21:46.44 | J4k3 | (which I'm uninstalling, I suspect its part of the stability problem with my phone) |
| 21:46.54 | J4k3 | the rest is tethering |
| 21:47.10 | jameswf | blackberry uses alot of data I have unlimited so dont know what I use I imagine streaming xm is pretty intensive |
| 21:47.30 | J4k3 | I wanna say XM itself only uses like 64kbit/channel |
| 21:47.38 | drmessano-LT | Asterisk has pretty well pissed me off too.. Open Source is overrated! Screw you guys, i'm going to Wal Mart and getting a HelloKitty Skype phone! |
| 21:47.41 | J4k3 | (it sounds like xing-ripped mp3s to me...) |
| 21:47.43 | jameswf | I want youtube on the blackberry damnit |
| 21:47.48 | Daviey | J4k3: ~$30 for 3Gb 3G PC dongle here |
| 21:47.57 | jameswf | hello kitty porn? |
| 21:47.58 | Daviey | J4k3: contract p/m |
| 21:48.06 | J4k3 | hrm, about the same |
| 21:48.24 | *** join/#asterisk jsmith (n=jsmith@72.21.36.138) |
| 21:48.24 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 21:48.40 | jameswf | ~hellokitty |
| 21:48.43 | jameswf | ~hello kitty |
| 21:48.44 | jbot | ACTION stands up and shouts "HELLO kitty!" |
| 21:48.50 | J4k3 | ~smellyourkitty |
| 21:49.03 | jameswf | ~smll your kitty |
| 21:49.06 | jameswf | ~smell your kitty |
| 21:49.09 | Daviey | J4k3: I should get one, but i'm too damn cheap.. It's rare for me to not be in wifi distance of a friendly network anyway |
| 21:49.16 | jameswf | ~stupid bot |
| 21:49.17 | jbot | Stupid human. |
| 21:49.28 | drmessano-LT | I want a Skype phone built into the side of a lunchbox... then, I will pwn |
| 21:49.30 | VitoCorleon | Daviey, ok i did it :) but how can i make it play a ringing sound while its forwarding? |
| 21:49.37 | timeshell | I have the same issue as is posted here... http://www.tek-tips.com/viewthread.cfm?qid=1409052&page=1 |
| 21:49.45 | timeshell | Can anyone enlighten me? |
| 21:49.49 | J4k3 | Daviey: yeah... I'm tempted to cancel the data service on this and just pay for some cable internets at my girlfriend's house. |
| 21:50.03 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 21:50.05 | J4k3 | her neighbors all have crappy DSL... and when I say crappy, I mean 512k-at-best crappy. |
| 21:50.13 | jameswf | ~bevis |
| 21:50.22 | jsmith | timeshell: Set the "timeout" setting in your queue configuration to something less than the Dial() timeout |
| 21:50.26 | J4k3 | the neighbor that had cable, was paying for 10mbit, would get at least 6mbit, and they moved :( |
| 21:50.30 | drmessano-LT | ~beavis |
| 21:50.31 | jbot | Mail poller for Enlightenment. URL: http://saturn5.locnet.net/epplets/ |
| 21:50.41 | [T]ank | i am changing the path in musiconhold.conf to a new location. then i do a moh reload at the cli. then i do a moh show classes. The old path is still in there. |
| 21:50.43 | [T]ank | what am i missing? |
| 21:50.46 | *** join/#asterisk sob0l (n=sobol@devel4.net) |
| 21:50.55 | jsmith | timeshell: In that example, the timeout was 60 seconds. That means "Try to ring the agent's phone for 60 seconds before giving up and moving on to another agent" |
| 21:50.56 | J4k3 | I should get some 'business' bandwidth and start a neighborhood hotspot. |
| 21:50.56 | J4k3 | haha |
| 21:51.02 | J4k3 | big nasty meraki mesh |
| 21:51.10 | *** join/#asterisk niekie (i=niek@bergnetworks.com) |
| 21:51.14 | Daviey | VitoCorleon: use Ringing() :) |
| 21:51.28 | VitoCorleon | exten=Ringing() ? |
| 21:51.28 | jsmith | [T]ank: That's the second report of that I've heard today... if it continues, I'd file a bug |
| 21:51.31 | jameswf | I have an open AP that goes to a honey pot.... all dns resolves to some fat porn |
| 21:51.40 | niekie | jameswf: eww.. |
| 21:51.42 | Daviey | VitoCorleon: http://www.voip-info.org/wiki-Asterisk+cmd+Ringing |
| 21:51.45 | jameswf | :) |
| 21:51.49 | timeshell | jsmith: I'm doing a ringall. |
| 21:51.52 | drmessano-LT | HAHA |
| 21:51.55 | timeshell | jsmith: Same apply? |
| 21:51.59 | *** join/#asterisk CVirus (n=GoD@196.205.192.211) |
| 21:52.02 | jsmith | timeshell: Of course! |
| 21:52.02 | drmessano-LT | jameswf: I did the SAME thing to my neighbors |
| 21:52.06 | niekie | jameswf: nah, set up upside-downternet. |
| 21:52.09 | niekie | It's way more fu. |
| 21:52.12 | niekie | fun* |
| 21:52.17 | jameswf | thats what you get for connecting to linksys |
| 21:52.18 | [T]ank | jsmith: it was me that reported the first bug. I am seeing it on a different server now. |
| 21:52.26 | J4k3 | upside down fat pr0n |
| 21:52.27 | drmessano-LT | It was awesome.. I have a real estate office behind my apt building.. I got at least 20 of them on DHCP |
| 21:52.30 | jsmith | timeshell: Or add a different extension for your callback agents that doesn't go to voicemail |
| 21:52.33 | Daviey | upside down goatse |
| 21:52.38 | niekie | jameswf: you have one of those LinkSys WRT54G's? |
| 21:52.43 | drmessano-LT | I just looked in the leases and grinned |
| 21:52.49 | *** join/#asterisk mltlnx (n=mltlnx@209.10.153.194) |
| 21:52.55 | niekie | They're nice. |
| 21:52.57 | jameswf | the honeypot is a BEFW11s4 |
| 21:53.02 | niekie | Ah. |
| 21:53.02 | drmessano-LT | HA |
| 21:53.03 | drmessano-LT | Same here |
| 21:53.13 | drmessano-LT | Good for nothing else |
| 21:53.18 | jameswf | exactly |
| 21:53.32 | sob0l | hi |
| 21:53.33 | drmessano-LT | I have 4 of them.. a little 500mhz box did the trick |
| 21:53.48 | jameswf | anything UDP makes it lock up like a small boy at a priest convention |
| 21:53.56 | niekie | Anyway, my goals with Asterisk for today have been completed :) (set up a support queue indicator to show if tech support is available on a website) |
| 21:54.00 | J4k3 | dialup losers always try to connect to it, then shove in their email address and password for their dialup. |
| 21:54.04 | drmessano-LT | Yep.. they had serious UDP issues |
| 21:54.05 | husimon | how do I echo a variable during a call |
| 21:54.11 | husimon | ie, echo $CALLERIDNUM |
| 21:54.14 | sob0l | what would be the difference between those two _X./*,1,app and _X.,1,app? |
| 21:54.19 | jblack | drmessano-LT: How did the sipphone thing go? |
| 21:54.23 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:54.25 | drmessano-LT | Well |
| 21:54.39 | drmessano-LT | [TK]D-Fender looked at it some, and got as far as we did :) |
| 21:54.40 | [T]ank | husimon: you just want to echo it on the CLI> screen? or are you doing phpagi? |
| 21:54.46 | husimon | just to the cli |
| 21:54.50 | jblack | hurh. |
| 21:54.53 | niekie | Use a Noop? |
| 21:54.53 | drmessano-LT | I spent hours tweaking |
| 21:55.06 | [T]ank | NoOp() |
| 21:55.16 | husimon | it doesn't appear that there is anything in $CALLERIDNUM because i'm calling voicemailmain with it as an argument and it won't skip past the mailbox |
| 21:55.18 | [T]ank | NoOp($CALLERIDNUM) |
| 21:55.25 | husimon | yet if I wait 8 seconds it then skips |
| 21:55.36 | husimon | [T]ank, ok yeah I have that now |
| 21:56.03 | *** join/#asterisk stuff1 (n=user@mail.win-ent.com) |
| 21:56.09 | JT | [T]ank: well and truly deprecated syntax |
| 21:56.11 | husimon | the callerid works fine on other phones, see the person calling |
| 21:56.22 | JT | husimon: do not use ${CALLERIDNUM} |
| 21:56.28 | JT | $CALLERID(num)} |
| 21:56.29 | husimon | JT, ok, what should I use? |
| 21:56.34 | stuff1 | hey i am trying to compile asterisk 1.4 in a chroot env, it finds my zaptel headers, but it doesn't seem to compile chan_zap |
| 21:56.40 | stuff1 | is there a reason for this? |
| 21:57.33 | husimon | hmm |
| 21:57.43 | husimon | is calleridnum totally wrong, or just an old syntax |
| 21:58.06 | hmmhesays | well polarity reversal doesn't seem to be enabled |
| 21:58.19 | [TK]D-Fender | husimon, 1.0 syntx |
| 21:58.27 | husimon | ok that makes sense |
| 21:58.28 | husimon | thanks |
| 21:58.58 | husimon | is there a reference that lists all the syntax changes, or do I need to just find them along the way by reading the newer command references. |
| 21:59.30 | freezey | [TK]D-Fender: got a cut sheet with all my info for my setup... |
| 22:01.09 | husimon | ? |
| 22:01.34 | stuff1 | anyone? |
| 22:01.43 | *** join/#asterisk bmg505 (n=leon@196.209.181.41) |
| 22:02.05 | [T]ank | JT: sorry... which syntax were you talking about. The one in my question or my asnwer to the NoOp? |
| 22:02.46 | bsdwarrior | Set: PassedInfo=1234 in a call file is not going into the cdr |
| 22:02.55 | JT | [T]ank: scroll up. |
| 22:03.01 | JT | [T]ank: i said exactly what was deprecated |
| 22:03.41 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 22:03.51 | *** join/#asterisk linux_galore (n=Richard@dsl-220-253-76-44.NSW.netspace.net.au) |
| 22:03.52 | [T]ank | oh, sorry... I did not mean to imply that I was giving the correct syntax, merely the NoOp() command was all I meant to provide. Thanks for pointing that out. You are correct. |
| 22:04.56 | linux_galore | one question, If I were to design an VoIP telephone that needs to work with Asterisk (automatic recognition on the network) were would I find the information to do this |
| 22:05.03 | *** join/#asterisk jmls (n=jmls@81.138.42.77) |
| 22:05.14 | linux_galore | ? |
| 22:05.27 | [TK]D-Fender | linux_galore, Auto recognition of what? |
| 22:05.40 | husimon | i think you'd still need to write a program that runs on the asterisk server for provisioning |
| 22:05.48 | husimon | then have it scan the network for your voip phone signature |
| 22:05.58 | jmls | If I were to put * in as a pure isdn voice-recorder (30 channels max), using MixMonitor (and potentially chan_spy) what sort of spec machine would I be looking at ? |
| 22:05.59 | husimon | or use dhcp |
| 22:06.13 | linux_galore | [TK]D-Fender: the type of telephone it is and what resources the device needs |
| 22:06.24 | husimon | asterisk won't automatically setup the sip.conf and extensions unless you scripted that. |
| 22:06.27 | stuff1 | what about taking the bonjour and using that |
| 22:06.27 | stuff1 | ? |
| 22:06.37 | Qwell | stuff1: using it how? |
| 22:06.47 | [TK]D-Fender | linux_galore, What is doing the "recognizing here? |
| 22:06.48 | Qwell | Do you want every phone to be able to connect to your server? very unlikely |
| 22:06.56 | stuff1 | well bonjour is setup to advertise and discover services |
| 22:06.57 | Qwell | rogue phones are bad |
| 22:07.03 | linux_galore | well I was thinking bonjour or zeroconf (both are pretty much the same really) |
| 22:07.08 | husimon | yeah you still want to have an allowed mac list |
| 22:07.22 | Qwell | macs can be spoofed |
| 22:07.24 | stuff1 | linux_galore: that what i wa suggesting |
| 22:07.51 | husimon | Qwell, plus of course a password. |
| 22:08.05 | tzanger | [TK]D-Fender: getting from rue rene levesque to ile les soeurs... any better? |
| 22:08.08 | linux_galore | what we are trying to do is create a telephone that plugs in and just works with minimal stuffing around |
| 22:09.21 | husimon | linux_galore, I think you could just write a few scripts and some daemons and use existing phones. |
| 22:09.26 | linux_galore | the phone works fine on CISCO,Avaya IP/NEC IP 2000 etc systems but we havent done much work in regards to asterisk |
| 22:09.27 | [TK]D-Fender | tzanger, Avoid bridges if you know whats good for you. |
| 22:09.41 | tzanger | ... how can you avoid a bridge when you're traveling to an island? |
| 22:09.42 | GrumpManAtWork | boat ? |
| 22:09.48 | tzanger | heh |
| 22:09.50 | [TK]D-Fender | tzanger, You are working on-island... STAY on-island |
| 22:09.58 | linux_galore | but does asterisk have anything that assists it to inter operate with freshly connected telephones |
| 22:10.06 | [TK]D-Fender | linux_galore, No |
| 22:10.15 | tzanger | [TK]D-Fender: I haven't found a hotel on island yet.. didn't know that was a possibiloity... will investigate that now |
| 22:10.20 | [TK]D-Fender | linux_galore, You don't seem to have a grasp of what * is. |
| 22:10.33 | linux_galore | [TK]D-Fender: so basically your stuck with what the OS provides |
| 22:10.48 | [TK]D-Fender | tzanger, We were the friggen capital of Canada FFS! You know our population? Huge. |
| 22:11.16 | [TK]D-Fender | tzanger, I know a dozen withing 15 minutes of each other. |
| 22:11.34 | tzanger | *on* the island? |
| 22:11.57 | bsdwarrior | tkd-fender, any idea what im doing wrong here. I cant set userfield in the cdr table. Set: userfield=test |
| 22:12.21 | linux_galore | [TK]D-Fender: so your saying asterisk doesnt need to know what is connected and magically does voice services management ? |
| 22:12.52 | [TK]D-Fender | tzanger, Within even a tiny corner. |
| 22:13.18 | tzanger | [TK]D-Fender: daaaaaaaaamn. ok, I will concede to your superior knowledge of montreal and guitar skills. :-) |
| 22:13.22 | Daviey | linux_galore: I'm certain people have made it work through other methods.. but thats secret sauce i think :) |
| 22:13.25 | [TK]D-Fender | linux_galore, No, I'm saying NOTHING is automated in Asterisk PERIOD, and you'd better get reading before dreaming of integration that does not exist. |
| 22:14.02 | linux_galore | [TK]D-Fender: so its manualised still ? |
| 22:14.12 | Daviey | linux_galore: it is possible, just not with Asterisk directly |
| 22:14.59 | *** join/#asterisk AndyGraybeal_ (i=minDlabs@node191.37.251.72.1dial.com) |
| 22:15.13 | [TK]D-Fender | linux_galore, Stop and go read the BOOK |
| 22:15.14 | [TK]D-Fender | ~book |
| 22:15.15 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 22:15.24 | [TK]D-Fender | linux_galore, You clearly have no idea what you're dealing with. |
| 22:15.26 | linux_galore | Daviey: what Im finding is asterisk has nothing and has to use some glue scripts and zeroconf/dns/bonjour/name_your_poison to manage network resources and how they connect |
| 22:15.38 | tzanger | asterisk is powered from magic pixie dust |
| 22:16.01 | Daviey | linux_galore: and that is a problem becayse? |
| 22:16.05 | Daviey | because? |
| 22:16.31 | *** join/#asterisk Docfxit (n=none@ip-64-32-143-214.lax.megapath.net) |
| 22:16.47 | linux_galore | Daviey: when your comminting a few millions to a VoIP telephone the lack of stated standards for a particular product is a big problem |
| 22:17.08 | Daviey | linux_galore: then don't use it :) |
| 22:17.52 | Daviey | linux_galore: seriously, this isn't an * problem |
| 22:17.56 | linux_galore | Daviey: Im trying to sell open source solution to the pen pushers and your response is of little help |
| 22:18.03 | linux_galore | solutions* |
| 22:18.22 | tzanger | oh dear |
| 22:18.40 | Daviey | linux_galore: Having extensions automagically work is possible |
| 22:18.43 | Docfxit | Does anyone know for iBeam softphone if the Account# and pin come from iBeam or from the Asterisk box? |
| 22:18.57 | tzanger | linux_galore: if you've got that kind of coin to throw at this, you can certainly hire some decent contractors to help you do exactly what you want. I believe [TK]D-Fender here has been helping greatly, why not hire him for his expertise? |
| 22:19.38 | linux_galore | Daviey: I cant just tell them to shove 101 standards into a product with spare resources measures in K (due to costs) they will think Im on crack |
| 22:21.41 | Daviey | linux_galore: I'm not sure what else to say.. you wanted to know if you can have extensions work out of the box with zero manual confs, right? I'm telling you it is |
| 22:22.15 | Daviey | just not using * for that part |
| 22:22.29 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:22.29 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:22.34 | [TK]D-Fender | linux_galore, There is no "auto-anything" in *. |
| 22:22.52 | Daviey | well auto attendant :) |
| 22:22.52 | jameswf | my * auto crashes sometimes |
| 22:23.03 | linux_galore | Daviey: we have zeroconf implemented (maybe bonnjour in the long run) but Im wondering if asterisk work better with a set model or has its own defined model like cisco etc |
| 22:24.09 | Daviey | linux_galore: Asterisk has flat files or database as standards.. Doing auto-conf is entirely seperate |
| 22:25.00 | Daviey | It's like asking apache to auto conf virtual hosts when a domain is pointed towards it.. not Apache's problem |
| 22:25.08 | jameswf | you either take * in all its glory and configure it by hand or cripple it to a simple pbx and use a built solution such as elastix |
| 22:26.58 | husimon | linux_galore, seems like the people investing money in this didn't do their homework. |
| 22:27.00 | tzanger | auto-crashes, hahaha |
| 22:27.07 | hmmhesays | does anyone have the ability to send me an int'l fax? |
| 22:27.10 | linux_galore | jameswf: Im trying to make it so anyone offering a Asterisk solution can easily connect a VoIP telephone product we make |
| 22:27.33 | husimon | linux_galore, that's pretty much impossible with the interface asterisk has. |
| 22:27.42 | Qwell | linux_galore: feel free to write a module for zeroconf, and submit it to bugs.digium.com |
| 22:27.45 | Daviey | linux_galore: Yes, it's possible - geez |
| 22:27.54 | tzanger | hmmhesays: not me, sorry |
| 22:27.59 | *** join/#asterisk zobia (n=laurashr@222.212.64.63) |
| 22:28.05 | hmmhesays | tzafrir, aren't you in israel? |
| 22:28.12 | linux_galore | Daviey: yes I need the documentation to hand the engineer then |
| 22:28.19 | tzafrir | hmmhesays, yes |
| 22:28.21 | hmmhesays | I don't have any pstn lines here haha I can't send a fax any way other than IP |
| 22:28.33 | hmmhesays | my blackfin is in tel aviv |
| 22:28.43 | tzanger | linux_galore: so write the software requirements spec, hand it to the engineer and you'll get your autoconf |
| 22:28.51 | Daviey | linux_galore: I don't think anyone will want to do your job for you :) |
| 22:29.02 | zobia | hello everyone. i got a deadlock issue and coredump. but only can trace a little can not go further who can help me out? |
| 22:29.18 | x86 | zobia: what version of asterisk? |
| 22:29.22 | jameswf | zobia: what * version |
| 22:29.32 | jameswf | ~deadlock |
| 22:29.35 | zobia | hi jameswf:1.2.24 |
| 22:29.38 | x86 | jameswf: jynx ;) |
| 22:29.50 | mjoyce | zobia - let me guess, it was stuck on the channel structure! |
| 22:29.50 | zobia | yes. deadlock |
| 22:29.58 | zobia | yes. channel |
| 22:30.01 | linux_galore | Daviey: we are already writing kernel patches for Linux we dont want to write a whole VoIP inter operation standard just for a telephone |
| 22:30.01 | zobia | sccp channel |
| 22:30.18 | husimon | linux_galore, what are the kernell patches for? |
| 22:30.27 | linux_galore | husimon: security |
| 22:30.57 | Daviey | linux_galore: you need to google to see how others have done it. |
| 22:31.05 | husimon | linux_galore, is your phone running linux? |
| 22:31.16 | husimon | i don't get it |
| 22:31.22 | Daviey | it's not a fscking VOIP inter operation standard |
| 22:31.37 | linux_galore | husimon: no its a variant of TRON |
| 22:31.44 | zobia | any one can help or i need to put the core file somewhere to let you guys take a look. |
| 22:31.45 | mjoyce | you could easily auto-provisionon any network that is using option 66 |
| 22:31.50 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 22:32.07 | mjoyce | scan the incomming dhcp requests and drop config files on the fly... |
| 22:32.09 | Daviey | mjoyce: i would hope he knows that |
| 22:32.36 | husimon | i dunno how far he wants to auto-configure |
| 22:32.47 | jameswf | my router uses option 666 i think its possesed |
| 22:32.56 | mjoyce | heh |
| 22:33.03 | linux_galore | husimon: the world is full of idiots and your trying to sell telephones, welcome to my world |
| 22:33.10 | Daviey | jameswf: does it have PiXiE's aswell? |
| 22:33.24 | husimon | linux_galore, so what standard does this phone use? sip? |
| 22:33.30 | florz | could someone please kick thevoipforum_org for spamming? |
| 22:33.34 | zobia | jameswf: mjoyce do you have any ideas? |
| 22:33.34 | mjoyce | linux_galore i would suggest selling a solution where you ship the phones pre-configured |
| 22:33.43 | zobia | it said CCP/6012-00000051' |
| 22:33.43 | zobia | Jan 31 13:11:31 DEBUG[3687] channel.c: Avoiding initial deadlock for 'SC |
| 22:33.43 | linux_galore | husimon: mainly sip with lots of licenses codecs |
| 22:33.50 | mjoyce | zobia when it comes to deadlocks? don't use queues. heh. |
| 22:33.52 | jameswf | if you cater to idiots you will attract them |
| 22:33.52 | linux_galore | licensed* |
| 22:34.05 | mjoyce | almost all of the deadlocks i have seen are in the masquerade <sic?> code |
| 22:34.10 | zobia | yes. i use queue for inbound. but not for outbound . |
| 22:34.11 | mjoyce | its messy shit |
| 22:34.11 | Daviey | linux_galore: which ones.. G.729 and what else? |
| 22:34.28 | zobia | but i don't know if it happened when inbound come or outbound |
| 22:34.33 | mjoyce | increasing system performance is a good way to avoid deadlocks |
| 22:34.35 | *** part/#asterisk jmls (n=jmls@81.138.42.77) |
| 22:34.49 | Daviey | mjoyce: thats a nasty solution :) |
| 22:34.58 | zobia | when i use gdb for the core dump the i do find something like qlog() has stuck |
| 22:34.58 | freezey | mjoyce: got the sheet |
| 22:35.13 | mjoyce | i had a customer once that had like 200 queues, lowering the amount of queues helped lower the load on all the inbound queue checks |
| 22:35.23 | mjoyce | helping to avoid deadlocks |
| 22:35.44 | zobia | but i only have 1 queue |
| 22:35.45 | husimon | is it possible to copy astdb databases to another asterisk server and have it work? |
| 22:36.07 | zobia | mjoyce . are you sure it's queue problem? |
| 22:36.10 | linux_galore | Daviey: H.324, ADSL, SKINNY and maybe MGCP and some weird cisco ones and NEC stuff |
| 22:36.28 | zobia | mjoyce: qlog() also belongs to queue? |
| 22:36.31 | mjoyce | like i said, most deadlocks i have seen are in queue code |
| 22:36.50 | jameswf | zobia: what channel type |
| 22:36.55 | husimon | who here does asterisk ha? and how do you go about it psync? |
| 22:36.56 | mjoyce | yeah, qlog is the handle to the queue log file iirc :) |
| 22:36.57 | linux_galore | Daviey: Im jus wondering if adding zeroconf would cover Asterisk |
| 22:37.00 | zobia | jameswf it's sccp |
| 22:37.01 | linux_galore | just* |
| 22:37.54 | Daviey | linux_galore: have you actually ran * yet? |
| 22:38.23 | linux_galore | Daviey: not yet, they want to know what is needed on the telephone before they even go there |
| 22:39.06 | Daviey | linux_galore: installing * on an old pc (even 486) wouldn't hurt.. that is honestly the best R&D |
| 22:39.11 | zobia | jameswf : can you help me? |
| 22:39.12 | jameswf | zobia: have you checked https://developer.berlios.de/bugs/?group_id=4020 |
| 22:39.27 | zobia | jameswf: let me check |
| 22:39.34 | linux_galore | Daviey: so just dhcp and zeroconf you thing should be fine ? |
| 22:39.44 | linux_galore | think* |
| 22:39.48 | Daviey | linux_galore: then buy a _handful_ of phones and learn how to auto-prov' them |
| 22:39.54 | freezey | mjoyce: they gave me about 3 pages of info with all of my stuff on it |
| 22:40.02 | Inssomniak | so asterisk doesnt necessarily have to do voip? It can also work just as well with analog lines? |
| 22:40.10 | linux_galore | Daviey: lol, we can make one in a day |
| 22:40.14 | x86 | Inssomniak: sure can |
| 22:40.17 | x86 | Inssomniak: TDM too |
| 22:40.21 | Daviey | linux_galore: for such a large project - i reckon they could afford $700 USD buying the popular handsets |
| 22:40.23 | Inssomniak | what is TDM? |
| 22:40.34 | x86 | Inssomniak: time division multiplexing |
| 22:40.48 | x86 | Inssomniak: used for such things as T1's |
| 22:40.58 | tzanger | Daviey: my first * box was a P90 with no MMX support |
| 22:40.58 | zobia | jameswf: i did not read this before .thanks . let me read for a while. |
| 22:41.01 | jameswf | wow chan sccp shouldntwork at the rate its been developed how would it keep up |
| 22:41.05 | linux_galore | Daviey: we have samples of others already and the product we have pretty much supports all the obvious protocols but not zeroconf |
| 22:41.08 | Daviey | tzanger: cheap jack :) |
| 22:41.10 | Inssomniak | this is cool, and I can have a different IVR for analog lines vs. a incoming voip call? |
| 22:41.12 | tzanger | it took about 500ms to transcode 1ms of ilbc, IIRC :-) |
| 22:41.27 | x86 | jameswf: chan_sccp is _crap_ |
| 22:41.39 | *** join/#asterisk mltlnx (n=mltlnx@209.10.153.194) |
| 22:41.42 | x86 | tzanger: ilbc == fail |
| 22:41.43 | mjoyce | Inssomniak: the internet sucks |
| 22:41.51 | mjoyce | Inssomniak: as such, voip over the internet sucks |
| 22:41.56 | mjoyce | (sometimes) |
| 22:41.59 | x86 | Inssomniak: sure, you can do anything with asterisk |
| 22:42.13 | Inssomniak | voip sucks? heh |
| 22:42.16 | Daviey | asterisk even made me a coffee just now |
| 22:42.19 | x86 | Inssomniak: you could have a different IVR for multiple VoIP connections as well |
| 22:42.22 | mjoyce | that isn't waht i said |
| 22:42.22 | tzanger | x86: yep, but it was just to see :-) |
| 22:42.30 | mjoyce | <mjoyce> Inssomniak: the internet sucks |
| 22:42.32 | linux_galore | Daviey: I just need to confirm what I suspect and that is asterisk really doesnt care about the connection stuff that is OS level and can be anything you want |
| 22:42.42 | Daviey | yeah |
| 22:42.45 | husimon | linux_galore, shouldn't reading the manual tell you that? |
| 22:42.48 | Inssomniak | but say, voip over a private cable TV network is OK! :) |
| 22:43.00 | mjoyce | Inssomniak: cable modems suck |
| 22:43.08 | x86 | tzanger: try doing g729 to ilbc on a p90 >;-) |
| 22:43.11 | linux_galore | husimon: it doesnt really define that side of things well |
| 22:43.13 | Inssomniak | so, what is "good" then heh? |
| 22:43.21 | husimon | mjoyce, they are a lot better then dialup.. |
| 22:43.22 | husimon | hehe |
| 22:43.23 | Daviey | linux_galore: think of * as what Apache is to the www world.. end of the day, just an application |
| 22:43.42 | mjoyce | Inssomniak: your little motorolla surfboard doesn't have very good caches, it shares inbound and outboudn IO on one cache - all of the context switching required to pass rtp traffic in and out crushes those things at just a few concurrent calls |
| 22:43.45 | husimon | linux_galore, If I were you i'd setup an asterisk install, then you will know exactly how it works |
| 22:44.11 | Inssomniak | heh dont look at me man.. I have DSL.. |
| 22:44.14 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 22:44.16 | tzanger | x86: please, I didn't want a fire hazard in my basement |
| 22:44.18 | zobia | jameswf: hello again. i read those but still no idea how to resolve it. |
| 22:44.24 | Daviey | husimon: i suggested that.. but he didn't seem keen - easier to ask lazyweb |
| 22:44.41 | jameswf | zobia: my recomendation dont use chan_sccp |
| 22:45.02 | husimon | zobia what was your problem? |
| 22:45.15 | zobia | jameswf : use sip? |
| 22:45.20 | husimon | and you said you had asterisk 1.2 |
| 22:45.21 | husimon | ? |
| 22:45.24 | zobia | husimon : my problem is deadlock |
| 22:45.33 | husimon | i've heard the chan_sccp in 1.4 is a lot better |
| 22:45.36 | husimon | you should try upgrading |
| 22:45.41 | zobia | husimon: sccp deadlock |
| 22:45.44 | husimon | i coudln't get sccp to work at all |
| 22:45.44 | tzanger | coppice: I'm looking at asterisk's indications.conf to see how it's going about generating arbitary frequency sine waves in an attempt to do the same myself on the blackfin. Now my dsp-fu is rather weak, and I'm finding the code hard to follow. Do you have any resources I might be able to use to better undrestand how people generate sinewaves of arbitrary frequencies like this? |
| 22:45.45 | husimon | in 1.2 |
| 22:45.47 | mjoyce | chan_sccp didn't cause your deadlock |
| 22:45.50 | husimon | it would crash my asterisk over and over |
| 22:45.53 | mjoyce | im betting app_queue or app_agent |
| 22:46.15 | BBHoss | is it possible for me to recieve audio from my provider on a different IP than I send my audio to? SIP |
| 22:46.17 | russellb | mjoyce: you're just speculating ... and there is no such thing as app_agent |
| 22:46.25 | mjoyce | BBHoss: reinvite |
| 22:46.37 | husimon | russellb, oo caught red handed. |
| 22:46.52 | russellb | zobia: what version are you running? |
| 22:46.56 | BBHoss | well i'm getting one-way audio, what should i set canreinvite to? |
| 22:46.56 | husimon | 1.2.x |
| 22:47.01 | zobia | if it's queue 's problem. is there any patch for it? |
| 22:47.06 | russellb | zobia: if you're running 1.4, we have some built in deadlock debugging that makes it easy to fix |
| 22:47.17 | zobia | @russellb:i am using 1.2.24 |
| 22:47.41 | russellb | zobia: ok, well I would _strongly_ suggest moving to asterisk 1.4. then, if you have a deadlock, i would be happy to look at it and fix it. |
| 22:48.02 | coppice | tzanger: if I remember rightly, * is using a 2nd order resonator to generate tones. That's only a few lines of code. its quick and simple, and OK for pure tones. If you want to modulate it can have problems, which is why I suggested DDS yesterday. That is a very generic and totally stable way to generate tones |
| 22:48.11 | zobia | @russellb:ok. thanks i will upgrade it. it seems the only way to make easier |
| 22:48.24 | russellb | zobia: yep ... |
| 22:48.35 | russellb | zobia: hop in to #asterisk-bugs if you need assistance with debugging a deadlock on 1.4 |
| 22:49.15 | zobia | @russellb: i can find you in #asterisk-bugs? |
| 22:49.27 | russellb | zobia: yep, and a bunch of other developers that can help. |
| 22:49.30 | BBHoss | ~nat |
| 22:49.31 | jbot | i guess nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 22:50.10 | zobia | @russellb: thanks a lot. |
| 22:50.16 | russellb | zobia: you're welcome |
| 22:50.39 | zobia | @russellb: no one can debug on 1.2.24 now? |
| 22:50.57 | russellb | zobia: the 1.2 series is no longer officially supported |
| 22:51.02 | *** part/#asterisk jsmith (n=jsmith@72.21.36.138) |
| 22:51.06 | *** join/#asterisk tecnico (n=tecnico@user-24-214-56-217.knology.net) |
| 22:51.07 | russellb | so, no, not from me, or any of the other main asterisk developers. |
| 22:51.27 | tzanger | coppice: hmm, ok |
| 22:51.32 | tzanger | I only need simple tones |
| 22:51.44 | zobia | @russellb: i see, thanks |
| 22:51.56 | coppice | then the few lines of code in * should be all you need |
| 22:52.38 | Inssomniak | I need an IVR if I want a voice mail setup that is like "press 1 for dons voicemail", "press 2 for joe's voicemail", right? " |
| 22:53.54 | tzanger | coppice: ok, I pared out all of the config file parsing, it really does seem to come down to a very simple routine |
| 22:53.58 | BBHoss | heh, didnt have my damn nat setup right :) |
| 22:54.22 | tzanger | bah the good stuff's hiding behind ieee logins |
| 22:55.08 | coppice | tzanger: the essential kernel of most DSP is just a few lines of code :-) |
| 22:55.23 | tzanger | coppice: the trick lies in knowing which! |
| 22:55.50 | Inssomniak | Im gonna have so much fun with this |
| 22:56.20 | tzanger | oh weird |
| 22:56.51 | tzanger | so goertzel is used to detect a number of tones at the same time... you excite the same goertzel filters in order to oscillate?? |
| 23:00.33 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 23:01.24 | *** part/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 23:03.34 | *** join/#asterisk Robba (n=rob@203.56.181.15) |
| 23:04.12 | Inssomniak | I know with asterisk ppl tell me it does everything.. but.. a call comes in, customer selects extension, that doesnt answer, can asterisk call a cell phone on another line? and ultimately if that doesnt answer after a few rings, THEN prompt to leave a message? |
| 23:04.25 | BBHoss | Inssomniak, yeah |
| 23:04.29 | Inssomniak | woah... |
| 23:04.38 | BBHoss | that would actually be very simple |
| 23:05.02 | Robba | Hi guys |
| 23:05.08 | BBHoss | the only key is to pickup before the cell phones vm picks up |
| 23:05.18 | Robba | havin a bit of an issue with a TE122P |
| 23:05.29 | Inssomniak | BBHoss, well in my case the cell phones dont have voicemail :) |
| 23:05.38 | BBHoss | great |
| 23:07.02 | BBHoss | Inssomniak, first, build and IVR/AA with an option that Dials the cell, like this: exten => 3,1,Dial(SIP/12567775555@siptrunk,15) the 15 is a 15 second timeout |
| 23:07.21 | BBHoss | then just manage the timeout to send to voicemail, like they do in the book |
| 23:07.33 | BBHoss | or ideally build a macro for the vm |
| 23:09.56 | husimon | what does this pattern match? "_*41*X." *41*<a digit><anything> ? |
| 23:10.21 | Qwell | husimon: 1 or more anythings at the end |
| 23:10.34 | husimon | the prefix is *41* though eh |
| 23:11.53 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 23:13.24 | Inssomniak | last question I promise for a while, a FAX comes in on my analog line, asterisk can take the fax, then send it out one of my FXS lines with a different ringtone that my fax could answer? |
| 23:13.55 | *** join/#asterisk urden (i=tumis@gateway/tor/x-b17b791016436380) |
| 23:15.14 | urden | hey, i just had an excellent thought. why don't people route VOIP traffic over wifi? that would be super awesome! just think of all the cable people could avoid stringing |
| 23:17.25 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 23:17.44 | fakhir | ~wifisip |
| 23:17.45 | jbot | [~wifisip] Please refer to ~wifivoip for info on this. |
| 23:18.15 | urden | whats sip |
| 23:18.39 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 23:18.39 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:18.39 | russellb | sip is a big pile of sip |
| 23:18.50 | JT | ~wifivoip |
| 23:18.50 | jbot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 23:19.00 | urden | that's like saying pis is a big pile of pis |
| 23:19.10 | husimon | exactly |
| 23:19.17 | JT | voip over wifi tends to suck |
| 23:19.20 | urden | i don't know what NAT, ATA, DECT are either |
| 23:19.28 | JT | well time to google |
| 23:19.35 | JT | you should at least know what NAT is |
| 23:19.47 | urden | maybe i should just try to stay out of other people's affairs |
| 23:19.59 | husimon | urden, there are wifi voip phones now |
| 23:20.13 | urden | but they suck? hard? |
| 23:20.17 | urden | all of them? |
| 23:20.18 | husimon | urden, DECT is another form of wifi it's just not 802.xxx |
| 23:20.22 | urden | some only slightly less than others? |
| 23:20.28 | urden | ahh i see |
| 23:20.29 | husimon | depends how you use them |
| 23:20.44 | urden | well there was this one episode of beavis and butthead |
| 23:20.50 | *** part/#asterisk urden (i=tumis@gateway/tor/x-b17b791016436380) |
| 23:24.27 | BBHoss | husimon, DECT is not wifi, or like it |
| 23:25.30 | husimon | BBHoss, all i'm saying is you get the same result in that you don't use wires. |
| 23:25.54 | BBHoss | well using a 27mhz analog cordless phone doesn't use wires either |
| 23:27.24 | hmmhesays | anyone ever seen wireshark not log something because you are pushing a lot of data through the ethernet interface? |
| 23:27.44 | husimon | ok so I shouldn't have used the word wifi, but wireless phone. |
| 23:28.06 | BBHoss | yeah, dect is just so much better than anything else out there |
| 23:28.32 | husimon | so i've heard |
| 23:28.50 | hmmhesays | that or tcp dump, these guys are claiming that tcpdump is missing packets written to the hard drive |
| 23:29.00 | *** join/#asterisk nny_1 (n=Scott@64.20.141.61.dyn-e-pool14.pool.hargray.net) |
| 23:29.19 | husimon | although it is possible to use dect with a data network |
| 23:29.26 | BBHoss | uhh yeah |
| 23:29.36 | BBHoss | but you don't transmit data over DECT |
| 23:29.39 | husimon | you can |
| 23:29.41 | husimon | is what i'm saying |
| 23:29.48 | BBHoss | why would you though |
| 23:30.00 | hmmhesays | I've never heard of tcp dump dropping data beause of a slow hard drive |
| 23:30.06 | BBHoss | its not designed for data anyways, its designed for voice |
| 23:30.36 | husimon | yeah i'm just mentioning this after reading the wiki entry and the bottom of it has a section on dect with an added standard that works with data |
| 23:30.47 | husimon | but I don't think it ever got anywhere |
| 23:30.50 | *** part/#asterisk zobia (n=laurashr@222.212.64.63) |
| 23:31.10 | BBHoss | yeah the only reason DECT is catching on in the US is because it is so huge in Europe |
| 23:31.57 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:36.09 | tzanger | DECT rox yer sox |
| 23:36.29 | tzanger | I just wish there was a US DECT phone with bluetooth |
| 23:36.31 | tzanger | god that's annoying |
| 23:37.01 | coppice | you want a cordless cordless phone? |
| 23:37.06 | tzanger | yes, dammit |
| 23:37.19 | husimon | without using an ata? |
| 23:37.23 | ManxPower | The answer is WITHIN, grasshopper. |
| 23:37.46 | husimon | yeah I don't quite understand why there are no 2/5ghz handsets that are sip |
| 23:37.52 | husimon | i don't really want to setup a dect netowkr |
| 23:38.14 | tzanger | husimon: that wouldn't be a bad idea for hte home, but I guess slapping a 1-port FXS ATA on the back of the base station isn't tha tbig a deal |
| 23:38.16 | *** part/#asterisk beek (n=klinebl@65.211.106.243) |
| 23:38.28 | husimon | husimon, yeah i'm just going to use an ata |
| 23:38.38 | husimon | then the users can pick what ever the hell they want if they like wireless |
| 23:38.46 | husimon | plus it's probably way cheaper |
| 23:39.06 | tzanger | I'm trying to find a wireless router + 2FXS + FXO that will run openwrt... I get very close with the WRTP54GS, but no FXO |
| 23:39.13 | husimon | laugh i chatted to myself. |
| 23:39.19 | tzanger | well for small offices I don't see the point of DECT |
| 23:39.51 | tzanger | I've installed it in 40k sq m factoies and you want it to be able to roam |
| 23:39.51 | husimon | wrt rox |
| 23:40.27 | husimon | 40 square kilometer factorY? |
| 23:40.34 | husimon | wtf are you building :P |
| 23:40.41 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk, -Addons, Libpri 1.6.0-beta2 (2008/01/28), Asterisk 1.4.18-rc4 (2008/01/31), *-Addons 1.4.5 (2007/12/1), Zaptel 1.4.8 (2008/01/14), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 23:40.50 | tzanger | the aastra ones are nice becuase they take regular old nimh AAA batteries, but they fucked up the headset jack placement so you can't charge with a headset plugged in |
| 23:40.53 | tzanger | husimon: that's not what I said |
| 23:41.17 | tzanger | 40k sq m != 40 square km |
| 23:41.21 | tzanger | unless my math is REALLY wrong |
| 23:41.32 | husimon | sorry 40,000 square miles ? |
| 23:42.03 | husimon | hehe |
| 23:44.07 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 23:44.45 | tzanger | haha |
| 23:44.48 | tzanger | I switched channels |
| 23:44.59 | tzanger | 18:44 < tzanger> bah I am mistaken |
| 23:44.59 | tzanger | 18:44 < tzanger> 40000 sq ft not 40000 sq km |
| 23:44.59 | tzanger | 18:44 < tzanger> er sq m |
| 23:45.00 | tzanger | 18:44 < tzanger> fuck |
| 23:45.00 | tzanger | 18:44 < tzanger> 40000 ft2 not 40000 m2 |
| 23:45.02 | tzanger | 18:44 < tzanger> there that's better |
| 23:45.04 | tzanger | 18:44 < tzanger> haha that's only a factor of 10 off |
| 23:45.06 | tzanger | 18:45 < tzanger> although I've been in 40k m2 factories (engine plants) |
| 23:45.44 | tzanger | ok now that is funny |
| 23:45.53 | tzanger | go to google, type in "find chuck norris" and click "I'm feeling lucky" |
| 23:47.43 | krdian_ | tzanger: rotfl |
| 23:49.43 | lunaphyte | i've been screwing around with a linksys spa3102, setting it up to answer pstn calls and pass them through asterisk to an ip phone - i've managed to get it to function, but i'm not quite sure i've gone about it in the sanest manner. |
| 23:49.52 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.105.226) |
| 23:51.28 | lunaphyte | i have the 3102 configured w/ pstn-to-voip gateway enabled, and a pstn answer delay of 1 sec, with a "caller default dial plan" of (S0 <:1160> ). |
| 23:52.06 | *** join/#asterisk mosty (n=mostyn@ppp191-34.static.internode.on.net) |
| 23:53.14 | lunaphyte | i have this entry in sip.conf : http://rafb.net/p/sk1bel12.html |
| 23:53.50 | husimon | tzanger, yeah was wondering what you meant :P fun stuff. |
| 23:56.46 | ManxPower | lunaphyte" there is no sane manner with that device for PSTN passthru |
| 23:56.56 | lunaphyte | here's what i've put in extensions.conf : http://rafb.net/p/tvlyx433.html |
| 23:57.20 | lunaphyte | no? |
