| 00:00.10 | drray | compile zaptel? |
| 00:00.14 | generalhan | do you have it ? |
| 00:00.26 | generalhan | and what version zaptel and asterisk ? |
| 00:00.36 | braniff | asterisk 1.2.6 |
| 00:00.54 | generalhan | this is your first time compiling asterisk ? |
| 00:01.26 | braniff | i didn't compile it. i just let the package manager install it on fedora core 5 (yum install asterisk) |
| 00:01.38 | generalhan | you should go to asterisk.org and get the newest versions of each Asterisk Zaptel Addons and Sounds. then compile each |
| 00:01.43 | braniff | ok |
| 00:01.52 | generalhan | im not sure what versions of zaptel you need with each version of asterisk |
| 00:02.00 | generalhan | im just helping out cause its a bit slow in here ! lol |
| 00:02.13 | braniff | there was that security update anyways that i need... |
| 00:02.25 | generalhan | braniff: just type 'wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.9.1.tar.gz' |
| 00:02.45 | generalhan | and 'wget http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.6.tar.gz' |
| 00:02.45 | braniff | will that get all the dependencies ? |
| 00:02.50 | braniff | ah |
| 00:02.54 | *** join/#asterisk Splat (n=Splat@220-253-134-28.TAS.netspace.net.au) |
| 00:03.10 | generalhan | well the dependencies you should be ok with since you had them installed via yum and FC5 |
| 00:03.18 | braniff | right |
| 00:04.03 | generalhan | so get those 2 files and compile both ... then do 'modprobe zaptel'; 'modprobe ztdummy'; 'safe_asterisk' |
| 00:04.11 | braniff | ok |
| 00:04.16 | generalhan | and see how that fairs for you |
| 00:04.25 | braniff | thanks generalhan ! |
| 00:04.31 | generalhan | at least if it doesnt work ... by that time someone MUCH smarter than i am will be here to help you ! lol |
| 00:04.42 | braniff | heh |
| 00:04.51 | generalhan | braniff: no prob ... and good luck |
| 00:07.08 | generalhan | wow i hope i know what im talking about ... for his sake ! lol |
| 00:07.14 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 00:08.10 | *** join/#asterisk Twister (n=bob@71-213-209-229.sxcy.qwest.net) |
| 00:08.20 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
| 00:08.22 | *** join/#asterisk DJ-Pyro_ (n=DJ-Pyro@lan-gw.brevient.net) |
| 00:08.41 | Twister | has anyone had any luck with integrating asterisk with 3com nbx (specifically using asterisk as a client on the pbx) |
| 00:08.46 | knarfly | Need to test my setup. Can anyone with an FWDNET account give me a call? |
| 00:09.08 | Twister | although it would be nifty to connect the nbx phones to asterisk :) |
| 00:10.10 | puzzled | Twister: the nbx needs to do sip for starters. does it do that these days? |
| 00:10.50 | Twister | ya |
| 00:11.08 | puzzled | amazing, they have seen the light |
| 00:11.12 | Twister | but i think the phone has to contact the nbx to download the firmware for it each time its booted |
| 00:11.28 | Twister | so in that respect its still propritary |
| 00:11.37 | puzzled | ah yes, there was something fishy about their supposedly open standards support |
| 00:11.42 | Twister | lol ya |
| 00:11.45 | puzzled | what an idiots |
| 00:11.50 | Twister | amen to that |
| 00:12.08 | Twister | its a neat system in itself..just sux to try to add anyting non-3com to it |
| 00:12.33 | JoseBravo | <PROTECTED> |
| 00:13.23 | Twister | most likely not, are you sure thats the name of the channel? |
| 00:15.12 | *** join/#asterisk a1fa (n=x64@24.144.58.223) |
| 00:15.14 | a1fa | yo yo |
| 00:15.36 | puzzled | romeo |
| 00:15.40 | a1fa | anybody know how to merge this new voice changer patch |
| 00:15.47 | a1fa | puzzled: no. i hate alfa romeo |
| 00:15.52 | a1fa | the shittiest car ever |
| 00:16.14 | puzzled | blasphemy! |
| 00:16.54 | a1fa | no |
| 00:17.37 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:18.52 | puzzled | anyone that uses cdr_mysql and does not want to loose cdrs when mysql dies might want to apply the patch from 7274 |
| 00:19.23 | puzzled | a1fa: which voice changer patch? |
| 00:19.33 | mountainm2k | interesting? what's the patch make it do instead? |
| 00:19.42 | a1fa | the new one that is ready for merger with the svn tree on asterisk.org |
| 00:19.43 | mountainm2k | heh, I trust mysql more than I trust Asterisk at this point in time |
| 00:20.18 | a1fa | puzzled: this one http://bugs.digium.com/view.php?id=6944&nbn=10 |
| 00:20.26 | puzzled | mountainm2k: it spools the cdrs when mysql dies which it currently does not |
| 00:20.34 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 00:21.03 | puzzled | a1fa: that is quite a funny app. I'm going to have a lot of fun with that (if it works) |
| 00:21.21 | a1fa | it works |
| 00:21.25 | a1fa | i am running an older version |
| 00:21.34 | a1fa | the problem is.. this is an svn patch |
| 00:21.46 | a1fa | what ever the fuck that means |
| 00:23.07 | puzzled | that it is meant to be applied to svn trunk |
| 00:23.27 | a1fa | well, yeah |
| 00:23.34 | a1fa | when i try to patch it, i get some errors |
| 00:23.41 | a1fa | and it creates wrong files |
| 00:24.21 | a1fa | i am patching against 1.2.9.1 |
| 00:25.26 | puzzled | a1fa: won't work. the patch is made for the upcoming 1.4 |
| 00:26.01 | a1fa | Hunk #1 FAILED at 44. Hunk #2 FAILED at 71. 2 out of 2 hunks FAILED -- saving rejects to file Makefile.rej |
| 00:26.02 | a1fa | yeah |
| 00:26.09 | a1fa | i guess I am going to use the old version |
| 00:26.17 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 00:26.20 | TheCops | Hi |
| 00:26.28 | mountainm2k | hi |
| 00:26.30 | rene- | pinchi asterisk |
| 00:27.37 | puzzled | Twister: search the asterisk-users list. There are messages from ppl that have successfully hooked up * to an NBX via a T1 link |
| 00:27.58 | rene- | what will 1.4 bring to the table? |
| 00:28.15 | a1fa | rene-: voice changer |
| 00:28.18 | TheCops | Someone know how to make appear the HoldEvent in asterisk API manager? |
| 00:28.27 | a1fa | built in voice changer |
| 00:28.33 | puzzled | Twister: Oct. 4 2005 subject: Question about 3Com(r) 3101 Basic Phone |
| 00:28.57 | rene- | voice changer as in i can talk like a really old guy or a little girl? |
| 00:30.44 | a1fa | yes |
| 00:32.25 | a1fa | Connected to Asterisk SVN-tag-1.2.9.1-r32601M |
| 00:32.32 | a1fa | UNREACHABLE |
| 00:32.33 | a1fa | lol |
| 00:33.03 | a1fa | so fucking lame |
| 00:33.04 | a1fa | i hate nat |
| 00:33.26 | *** join/#asterisk TheSin (n=TheSin@iphost-64-56-130-194.edm.wiband.net) |
| 00:33.28 | mountainm2k | why can't SVN work though NAT??? It's just a client-server protocol... |
| 00:33.34 | mountainm2k | Whoops, that was offtopic, sorry... |
| 00:33.53 | TheSin | is there a good way to debug AGI stuff |
| 00:34.00 | feld | anyone know of a sip phone for Windows CE? |
| 00:34.07 | TheSin | app-fax AGI isn't working for me and I can't see why it's dieing |
| 00:34.11 | mountainm2k | agi debug help any? |
| 00:34.30 | *** join/#asterisk CrlMan (n=CrlMan@unaffiliated/crlman2) |
| 00:35.15 | TheSin | thanks mountainm2k |
| 00:35.52 | TheSin | didn't even know that existed gonna try it out :D |
| 00:37.42 | sparkleytone | our current phone setup is an analog PBX that goes to analog phones over rj45 cable...is it possible to use the card from the devkit to have asterisk route multiple extensions from one cable? |
| 00:41.16 | nextime | Hi. I have an extension to autologon agents with callbacklogin, like this: |
| 00:41.16 | nextime | exten => _*XXXX,2,Dial(Local/${EXTEN:1}@localchanneldefs/n,,D(9999#)) |
| 00:41.33 | nextime | and in localchandefs |
| 00:41.33 | nextime | exten => _XXXX,1,AgentCallbackLogin(${EXTEN}||${CALLERID(num)}@operatoricb) |
| 00:41.45 | nextime | of course, the pass for all agents is 9999 |
| 00:42.09 | nextime | but it seem that AgentCallbackLogin never receive the DTMF from the Dial command |
| 00:42.13 | nextime | any idea? |
| 00:43.00 | feld | ~pastebin |
| 00:43.01 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 00:43.06 | feld | nextime: ^^^ |
| 00:43.51 | nextime | feld : 2 lines is "flooding channel?" |
| 00:44.00 | feld | i see 7 lines |
| 00:44.09 | nextime | feld : only 2 are a paste |
| 00:44.16 | feld | you could have had one pastebin link and one paragraph explaining. |
| 00:44.46 | nextime | feld : ok, so, from now we can chat on pastebin, irc is useless |
| 00:44.52 | feld | instead you had incomplete sentences scattered in there |
| 00:45.18 | nextime | come on guy, it's only a simple question, not a verbose debug or a complete config file |
| 00:45.35 | nextime | with your "pastebin suggest" you have flooded more than me :) |
| 00:46.10 | feld | for example nextime your last 3 messages could have been combined into one. no i didnt flood more than you. i've only spoken in this channel 6 times now. you've 10 times. |
| 00:46.48 | justinu|laptop | ~seen dylnes_office |
| 00:46.59 | jbot | i haven't seen 'dylnes_office', justinu|laptop |
| 00:47.06 | justinu|laptop | ~seen dlynes_office |
| 00:47.10 | jbot | dlynes_office is currently on #asterisk (1h 12m 59s), last said: 'same here, but then again, i don't use queues or agents, either'. |
| 00:47.31 | justinu|laptop | dlynes_office: success http://jabber.tardcat.com/photos/photos.cgi?album=irc&res=1024&image=1150418744.jpg |
| 00:47.35 | justinu|laptop | thanks man! |
| 00:47.57 | justinu|laptop | laterz |
| 00:48.09 | TheSin | could this be messing up my agi script? |
| 00:48.09 | TheSin | AGI Rx << Jun 15 18:47:16 WARNING[4297]: asterisk.c:806 ast_set_priority: Unable to set normal priority |
| 00:48.45 | knarfly | Need to test my setup. Can anyone with an FWDNET account give me a call? |
| 00:48.55 | nextime | feld : i can't belive that you are so hard, come on, i'm not in irc from 1 day, i'm not write one word a line, i'm not flood with a complete file here, i'm only ask a question, if you don't want answer to me, simply don't answer at all, i don't need a "irc guru" to explain how i have to write on irc. |
| 00:49.42 | sparkleytone | welcome to IRC. |
| 00:49.50 | nextime | anyway, no more word about this. Anyone have idea of what can generate my problem? |
| 00:50.16 | feld | nextime: thanks for using a paragraph lol. dude, my point is this: if you write your deal in several lines and nobody answers it's much harder to repeat later. instead you could have a link and a paragraph to repeat in 15 mins instead of 7 lines again. and no, I dont have your answer :( |
| 00:51.12 | nextime | feld : i'm not a kiddie that need to repeat a question every 15 min. I ask only once, and i go to debug or to rtfm to solve my problem |
| 00:51.34 | *** join/#asterisk Eggplant (n=none@dsl-216-155-214-007.cascadeaccess.com) |
| 00:52.22 | feld | nextime: should IRC be used as a last resort? after you did your RTFMing and debugging? =/ I thought IRC support channels like this were for the "extra help", not because you don't want to do your troubleshooting and read the vast documentation. |
| 00:52.28 | feld | *shouldnt |
| 00:56.06 | nextime | feld : i think that irc channel is to chat, maybe to help when i can, and to ask for a hint ( NOT to solve a problem for me, only to have a hint ) after you do some test that seem to not work correctly, after read some doc, after read some debug, and after that i ask on irc, sometime not even, and if anyone can tell me a usefull hint i'm happy, if no one say me something that help me, i go to debug more and more. Please, don't think that |
| 00:57.11 | feld | im not here to fight. i'm obviously being an ass. ignore me. i've had to fight with a DOS based PoS system for 9 hours today. it wasn't good. |
| 00:59.12 | nextime | feld : today was a long day to me too, and here is 3:00 am, so you can imagine how tired i'm no. Anyway, now i really need to go to solve my problem, so, ctrl+f2, go to shell, and debug. Bye. |
| 00:59.33 | feld | goodbye, goodluck, and goodnight! :) |
| 01:00.14 | nextime | thanks :) |
| 01:01.33 | feld | anytime. see you nextime, nextime. :) |
| 01:01.54 | feld | i have to relax and play some GRAW now. |
| 01:02.27 | *** part/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 01:05.23 | *** join/#asterisk sevard (n=sev@24-179-179-47.dhcp.dlth.mn.charter.com) |
| 01:05.39 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 01:08.47 | *** join/#asterisk anonymouz666 (n=lynx@ns2.redetaho.com.br) |
| 01:12.33 | *** join/#asterisk MacWeenie (n=chatzill@82-35-73-28.cable.ubr02.dals.blueyonder.co.uk) |
| 01:15.52 | *** join/#asterisk DrkShdw (n=scorpio@fl-209-26-20-205.sta.sprint-hsd.net) |
| 01:20.30 | knarfly | Okay time to watch the Maverick whip the Heat....Asterisk will have to wait |
| 01:23.37 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:23.52 | *** part/#asterisk sevard (n=sev@24-179-179-47.dhcp.dlth.mn.charter.com) |
| 01:24.56 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 01:27.41 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 01:32.22 | feld | file: when you go AFK do you /nick file_not_found ? |
| 01:32.31 | file | I'm always here :D |
| 01:32.53 | file | if I'm not here, civilization as we know it is gone |
| 01:33.11 | feld | true |
| 01:33.16 | feld | i need my file(s) |
| 01:35.20 | file | I don't wanna be recovered! |
| 01:36.59 | dlynes_home | [TK]D-Fender: btw...I sorta have the sangoma a200d up and running under 2.6.15.5 now :) |
| 01:37.12 | [TK]D-Fender | "sorta"? |
| 01:37.16 | dlynes_home | [TK]D-Fender: the only issue is that it doesn't seem to detect hangups |
| 01:37.32 | dlynes_home | [TK]D-Fender: and fxotune doesn't work with it |
| 01:37.46 | file | [TK]D-Fender: you know what? you should see how much a BRI would cost in your area just for kicks |
| 01:38.11 | dlynes_home | file: the telco will swear up and down several times over that they don't offer BRI's anymore |
| 01:38.27 | file | dlynes_home: that's why I said "just for kicks" as it would be an adventure |
| 01:38.30 | dlynes_home | only after you persist and ask them like 20 times will they finally admit that they sell them :) |
| 01:39.22 | dlynes_home | btw |
| 01:39.26 | dlynes_home | can you get dsl over isdn? |
| 01:40.38 | file | yes |
| 01:41.14 | file | only operates at 144Kbps though |
| 01:41.18 | dlynes_home | oh |
| 01:41.19 | dlynes_home | that sucks |
| 01:42.51 | [TK]D-Fender | file $100/mo last I checked, more like 60$ now I suspect |
| 01:43.12 | file | how much is.. ugh... analog? |
| 01:43.31 | dlynes_home | $35-60, depending on whether it's residential, or business |
| 01:43.46 | dlynes_home | and where in the country you're getting it |
| 01:43.55 | file | ha, my telco doesn't even advertise analog as an option for business anymore |
| 01:44.02 | dlynes_home | eh? |
| 01:44.04 | file | they do centrex centrex centrex |
| 01:44.14 | dlynes_home | how the fuck do you do adsl on centrex? |
| 01:44.21 | dlynes_home | or fax for that matter? |
| 01:44.25 | [TK]D-Fender | file : Centrex IS analog... just with faked wannabe features nobody uses |
| 01:44.40 | dlynes_home | i thought centrex was nortel digital? |
| 01:44.50 | file | dlynes_home: so did I |
| 01:44.54 | [TK]D-Fender | dlynes_home : Nope... its telco based PBX (sorta) |
| 01:45.12 | dlynes_home | yeah, which is usually an oversized option11, no? |
| 01:45.50 | file | $22 for an analog line |
| 01:45.57 | dlynes_home | damn cheap |
| 01:45.57 | [TK]D-Fender | Don't know the back-end.... just had centrex lines on my old 8x24 PBX... on diff was not having dial 9 suppression at first so having to prefix on outbound. |
| 01:46.07 | znoG_ | just wondering, for the people here who implement Asterisk systems, who do you mainly target them at? looking at boxed PBX prices, a small office with, say, 8 extensions and 2-3 lines who don't necessarily need to do any VoIP calls obviously prefer the cost of a boxed PBX to a PC to run Asterisk on, plus digium hardware, plus ATAs or more digium hardware for the extensions, etc etc... i'm all for Asterisk, of course, but I use it for lots of ov |
| 01:46.18 | znoG_ | and, I know all too well Asterisk is very powerful compared to a US$300 PBX |
| 01:46.19 | dlynes_home | but i'm not gonna move to new brunswick just cause phone lines are $22 instead of $35, either |
| 01:46.52 | dlynes_home | [TK]D-Fender: you also have to use meridian handsets, too |
| 01:46.57 | dlynes_home | [TK]D-Fender: which are digital handsets |
| 01:47.24 | file | I'm trying to get a business price |
| 01:47.32 | [TK]D-Fender | dlynes_home : No. I plug 10$ cheapo phones on those lines jsut the same. ANALOG <- |
| 01:47.57 | dlynes_home | [TK]D-Fender: hrm...maybe your telco implements it differently? |
| 01:48.05 | dlynes_home | [TK]D-Fender: with telus you have to have a meridian handset |
| 01:48.25 | [TK]D-Fender | dlynes_home : :/ |
| 01:49.21 | file | so $48.95/mth for centrex, 3 call managements features, 3 additional features, whtie and yellow page listings, 120 minutes of long distance to Canada, a toll-free number, and calling card |
| 01:50.03 | [TK]D-Fender | dlynes_home : You know "Meridian" aren't all "digital" phones... its an "umbrealla" branding that does include pure analog devices as well... I've never heard of getting a Digitral PHONE signal thrown on a line like that anywhere... |
| 01:50.06 | dlynes_home | znoG: our system is competitive with keysystems starting at about 3 lines (assuming we don't do PoE), and about 4 phones |
| 01:50.13 | dlynes_home | [TK]D-Fender: i know that |
| 01:50.18 | file | they make this so... cheap |
| 01:50.36 | [TK]D-Fender | file : While implies analog + hook-flash activated BS you can't use with a real PBX, nor would you WANT to. |
| 01:50.46 | dlynes_home | [TK]D-Fender: we've got like 20 boxes full of analog meridian handsets, too |
| 01:51.02 | dlynes_home | [TK]D-Fender: maybe i should've specified Meridian 1 |
| 01:51.07 | [TK]D-Fender | dlynes_home : CRAPTASTIC! |
| 01:51.09 | *** join/#asterisk pigpen2 (n=mark@209.159.234.250) |
| 01:51.20 | file | yeah they want you to rent or buy a meridian or a vista phone |
| 01:51.36 | drray | lease, with 0 the first year |
| 01:51.37 | dlynes_home | yeah...vistas are all analog, i think |
| 01:51.57 | znoG | dlynes_home: considering that keysystems (i presume that's what boxed PBXs are called) are about $300-400 for a 4 line 8 extension system.. how do you compete? |
| 01:51.59 | *** join/#asterisk philth|work (n=ceac2822@d38-179-126.home1.cgocable.net) |
| 01:52.04 | znoG | dlynes_home: the PC for Asterisk alone costs that much |
| 01:52.32 | dlynes_home | znoG: add on voicemail, handsets, and a ups and you'll see that value is much higher than $300-400 |
| 01:52.35 | drray | you compete by offering faxing, printing |
| 01:52.40 | drray | voicemail |
| 01:52.42 | znoG | dlynes_home: true, didn't think about voicemail |
| 01:52.44 | drray | and free suppor the first year |
| 01:52.58 | file | so an extra business line is $41/mth |
| 01:53.15 | znoG | dlynes_home: what do you normally sell in terms of FXS for a 8 extension asterisk system? |
| 01:53.17 | dlynes_home | znoG: not to mention support contracts for asterisk are much lower because you don't have to ding them with 1/2 hour travel time, fuel surcharge, minimum one hour charge, ... |
| 01:53.37 | drray | unless you can get away with that... :) |
| 01:53.50 | dlynes_home | we can get away with that |
| 01:53.55 | dlynes_home | but the thing is |
| 01:53.56 | [TK]D-Fender | * = expansion. IVR, Queues (a fortune with key systems), etc... |
| 01:54.10 | dlynes_home | the customer would rather not pay for it, if they have another solution |
| 01:54.18 | [TK]D-Fender | znoG : The "key system" is practically DEAD. |
| 01:54.26 | dlynes_home | extra features are much cheaper on asterisk than on a keysystem, too |
| 01:54.27 | file | I wish it was |
| 01:54.35 | drray | I do however think there is room for a pic based PBX system |
| 01:54.54 | dlynes_home | but if you want 1 or 2 phone lines, no voicemail, and 3 or 4 extensions |
| 01:55.08 | dlynes_home | obviously asterisk won't be able to hold a stick to a keysystem |
| 01:55.24 | znoG | yeah, that's what I mean |
| 01:55.24 | znoG | depends on features |
| 01:55.38 | dlynes_home | yeah...those customers we still sell nortel or panasonic keysystems to |
| 01:55.42 | drray | but a keysystem can't let you have a local extension at home |
| 01:55.44 | dlynes_home | everyone else, we try to sell them on asterisk |
| 01:55.47 | drray | or trunk |
| 01:56.03 | TheCops | Someone know how to see when a SIP channel put another SIP channel on hold via API Manager?! |
| 01:56.14 | dlynes_home | znoG: for an 8 extnesion asterisk system, we usually don't offer fxs |
| 01:56.15 | TheCops | I dont see it via event |
| 01:56.31 | dlynes_home | znoG: what the hell for? no need for fxs, for the most part |
| 01:56.59 | dlynes_home | znoG: unless you're adding voip lines to an existing keysystem |
| 01:57.28 | znoG | dlynes_home: right, but how do you connect extensions to Asterisk, then? |
| 01:57.32 | znoG | dlynes_home: using VoIP phones? |
| 01:57.44 | dlynes_home | exactly |
| 01:57.55 | DrkShdw | VoIP phones, or an ata adapter |
| 01:58.12 | [TK]D-Fender | dlynes_home : I can get a used P3 plenty sufficient for * for $100 tops, add 2 X100P @ $15ea ($130), add 2 SPA-2002 @ 70$ ($270), and 4 analog Phones @ $20 ea ($350). THERE. a complete PBX INCLUDING VM and the works, and the phones to boot.... |
| 01:58.18 | dlynes_home | voip phones...ata adapters suck, cause you're stuck trying to do all the regular pbx features on an analog phone |
| 01:58.53 | dlynes_home | [TK]D-Fender: and? Would you actually use that POS? |
| 01:59.06 | drray | or warranty it? |
| 01:59.10 | znoG | dlynes_home: yeah, agreed, plus VoIP phones have come down heaps in price, almost not worth using ATAs |
| 01:59.14 | dlynes_home | yeah...or what drray said :P |
| 01:59.15 | [TK]D-Fender | dlynes_home : Knowing * was packing it and I could do IVR, queues, VoIP routing, SURE. |
| 01:59.19 | znoG | dlynes_home: out of curiosity whats your VoIP phone of choice? |
| 01:59.20 | drray | what you are paying for is service |
| 01:59.25 | TheCops | Hi [TK]D-Fender |
| 01:59.27 | drray | not the OSS technology |
| 01:59.46 | [TK]D-Fender | dlynes_home : Yeah, I'm spoiled and would SPLURGE on Polycom hands down, but really how many calls can you juggle at a time, or NEED to? |
| 01:59.50 | dlynes_home | [TK]D-Fender: so the x100p cards would just be decorations? |
| 01:59.56 | file | [TK]D-Fender: 42 |
| 02:00.13 | Qwell | file: only? nub |
| 02:00.20 | file | :\ |
| 02:00.29 | [TK]D-Fender | dlynes_home : I kinda thought they'd take in 2 lines :) |
| 02:00.54 | dlynes_home | [TK]D-Fender: heh...and you trust them? |
| 02:00.56 | [TK]D-Fender | dlynes_home : I have set up plenty of systems using them, and yes they're cheap crap, but they can work... |
| 02:01.11 | dlynes_home | [TK]D-Fender: especially two in one machine? |
| 02:01.16 | [TK]D-Fender | dlynes_home : Not as much as I would an A200d, but I'm trying to compete ;) |
| 02:01.41 | [TK]D-Fender | dlynes_home : I set up a system with 2 in it for a client of mine 2 months ago |
| 02:01.44 | dlynes_home | i was thinking more like an a200, not an a200d |
| 02:01.55 | [TK]D-Fender | dlynes_home : Works just fine, including SpanDSP faxing. |
| 02:02.07 | dlynes_home | ah...nice |
| 02:02.13 | dlynes_home | i've never gotten spandsp to work properly |
| 02:02.14 | dlynes_home | but then again |
| 02:02.21 | [TK]D-Fender | dlynes_home : You asked what I'd prefer over an X100P :) I go for GOLD |
| 02:02.28 | dlynes_home | i had a bum machine or a bum te405p |
| 02:02.32 | dlynes_home | not sure which it was |
| 02:02.36 | znoG | i just got iaxmodem going with hylafax, worked just fine.. |
| 02:03.22 | [TK]D-Fender | I run 50 fax DID's on my PRi at work... people love it. |
| 02:03.34 | dlynes_home | damn |
| 02:03.41 | dlynes_home | well, i'll get a chance to try it soon |
| 02:03.53 | dlynes_home | if i get it working on the pri |
| 02:03.58 | dlynes_home | my boss will be ecstatic |
| 02:04.18 | dlynes_home | for him, that's more important than voip |
| 02:04.37 | philth|work | Quick question, With a Sipura 3000 is there anyway to use asterisk to control it to pick up the line? |
| 02:04.44 | [TK]D-Fender | dlynes_home : For me VoIP is just how I get my Polycom's to talk to * :) |
| 02:04.44 | TheCops | znoG, hylafax own:) |
| 02:04.46 | philth|work | And monitor said line |
| 02:04.47 | dlynes_home | ewww |
| 02:04.51 | dlynes_home | that's nasty |
| 02:04.59 | dlynes_home | you see that dog kennel on ctv news from quebec? |
| 02:05.20 | [TK]D-Fender | dlynes_home : No cable/satelite :) I'm TV-Free :) |
| 02:05.28 | dlynes_home | ah |
| 02:05.28 | Qwell | OTA! |
| 02:06.10 | [TK]D-Fender | I have a friend who downloads EVERYTHING and my * serves double duty with my 52" HDTV for HTPC usage :) |
| 02:06.32 | philth|work | With a sipura 3000 is there a way with asterisk to have it pick up the line and monitor it? |
| 02:06.34 | *** join/#asterisk d-tech (n=dtc@72.245.233.107) |
| 02:06.55 | [TK]D-Fender | philth|work : Yes. |
| 02:07.07 | [TK]D-Fender | philth|work : Dail a null number against it. |
| 02:07.12 | [TK]D-Fender | Dial* |
| 02:07.32 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 02:07.34 | [TK]D-Fender | philth|work : However there is that fact of them being able to HEAR you. |
| 02:08.16 | [TK]D-Fender | philth|work : So use an analog phone and mute it and hope they don't notice the "pop" and voltage drop setting off their "line in use" light |
| 02:08.20 | [TK]D-Fender | :) |
| 02:09.35 | Sponge_bob | anyone here use cisco as their gateway? |
| 02:12.46 | Sponge_bob | what i'm getting to is i try to run a meetme conference and when two-three people are on it is ok. once i get a few more people some don't hear others 'in a tunnel' and some do. I tried to turn the echocancelation, echocancelaltionwhenbridged and echotraining on. anything else i can do? |
| 02:15.39 | dlynes_home | dooed |
| 02:15.39 | dlynes_home | us dollar is only worth $1.10cdn now |
| 02:15.39 | znoG | how much is a Sangoma A200 word? |
| 02:15.39 | dlynes_home | It's gonna be less than $1Cdn by the end of the summer |
| 02:15.39 | znoG | s/word/worth |
| 02:15.39 | dlynes_home | znoG: around the same as a tdm400p, retail |
| 02:15.49 | znoG | and works better than the tdm400p? |
| 02:16.09 | dlynes_home | no idea..>I've never used the tdm400p |
| 02:16.11 | *** join/#asterisk d-tech (n=dtc@72.245.233.107) |
| 02:16.21 | philth|work | [TK]D-Fender: Thanks |
| 02:16.22 | dlynes_home | [TK]D-Fender seems to think so, though |
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| 02:19.11 | [TK]D-Fender | The A200 has some interesting stats that let it scale between the TDM400 and TDM2400 for tartget market, and on the technological side offer pretty solid performance. |
| 02:19.36 | Dandan | re |
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| 02:21.29 | znoG | [TK]D-Fender: yea, i just read' that you can keep adding them as you need to.. whereas with the TDM400 you have a maximum of 4 ports and the TDM2400 a maximum of 24 |
| 02:21.40 | dlynes_home | bill gates is quitting in two years!!!! |
| 02:21.42 | dlynes_home | yay! |
| 02:21.57 | drray | it'll still have his thumbprint on it |
| 02:22.09 | dlynes_home | probably, but he's stepping down as ceo/chairman |
| 02:22.29 | drray | won't he still be chief technology poobah |
| 02:22.36 | znoG | [TK]D-Fender: how do they actually link up though? are they just PCI cards and use their own bus? |
| 02:22.40 | dlynes_home | nope |
| 02:22.48 | dlynes_home | he hasn't had that title for quite some time |
| 02:22.57 | znoG | i think he may just have enough money to retire happily |
| 02:22.57 | znoG | :) |
| 02:23.08 | dlynes_home | you think? |
| 02:23.12 | drray | I dunno |
| 02:23.21 | dlynes_home | i'd think he'd have enough money to retire miserably |
| 02:23.26 | drray | :) |
| 02:23.27 | hads | Anyone understand fxotune more than me and is able to tell me why I always get 1=5,0,0,0,0,0,0,0,0 (fxotune from trunk BTW) |
| 02:23.31 | dlynes_home | he's got so much money he won't know what to do with his retirement |
| 02:23.34 | drray | I don't get the gates hate |
| 02:24.02 | [TK]D-Fender | znoG : Well the A200 occupies a single PCI SLOT, but expands using multiple PCI *backplanes*. So its scales differrently, and very well in between, while offering full G.68-2002 HWEC where the TDM400 offers none, and the TDM2400 only G.68-2000 |
| 02:24.05 | dlynes_home | that's not gates hate |
| 02:24.11 | dlynes_home | that's gates pity |
| 02:24.19 | drray | not you personally, just in general |
| 02:24.34 | DrkShdw | I'm with you, drray. the man revolutionized the industry. but it's pure hatred for him that basically started the OSS movement. so, I'm all for it :P |
| 02:24.34 | dlynes_home | i feel sorry for rich people |
| 02:24.39 | dlynes_home | they don't know how to enjoy life |
| 02:25.03 | *** join/#asterisk withaynoti (n=withayno@c-68-49-245-183.hsd1.dc.comcast.net) |
| 02:25.05 | drray | well, if you are not happy poor, then you are not going to be happy rich |
| 02:25.33 | Sponge_bob | anyone here have experience with meetme echos? |
| 02:25.35 | dlynes_home | DrkShdw: i think that was more derision, than hatred |
| 02:27.00 | *** join/#asterisk los415 (n=los415@sfca-office.corp.race.com) |
| 02:27.51 | znoG | [TK]D-Fender: what exactly is G.68-2002? |
| 02:28.11 | *** join/#asterisk d-tech (n=dtc@72.245.233.107) |
| 02:28.36 | withaynoti | i realize that this is the wrong forum for this kinda question, but i was wondering if anyone knew of another channel where i could get some advice on choosing a VoIP provider in the US |
| 02:29.01 | withaynoti | (you probably get a million of these questions a day) |
| 02:29.02 | dlynes_home | withaynoti: try www.calltermination.com |
| 02:29.07 | [TK]D-Fender | znoG : 128 ms EC on all channels, not an incrementally dropping like the original TDM2400 VPM. |
| 02:29.27 | dlynes_home | withaynoti: it's a huge list of providers, carriers, terminators, originators |
| 02:29.38 | [TK]D-Fender | znoG : And I typo'd.. thats G.168-2002 |
| 02:29.51 | dlynes_home | withaynoti: some of them may not offer service to you...verify with them that they will sell retail |
| 02:30.02 | dlynes_home | withaynoti: some of them only do wholesale |
| 02:30.26 | withaynoti | oy. that IS a big list |
| 02:30.41 | withaynoti | dlynes_home: thanks for the tip |
| 02:30.44 | dlynes_home | np |
| 02:39.08 | withaynoti | also, does anyone happen to know of a small & fairly cheap asterisk appliance available? small office...only 2 users. |
| 02:39.36 | *** join/#asterisk tlowe_ (n=tlowe@bgp.terrorist.net) |
| 02:40.28 | *** join/#asterisk pigpen2 (n=mark@209.159.234.250) |
| 02:40.29 | [TK]D-Fender | withaynoti : 2 Phones, 2 lines? |
| 02:40.38 | withaynoti | right |
| 02:40.58 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 02:42.05 | [TK]D-Fender | withaynoti : 100$ used P3 + (2 x X100P) = $130. |
| 02:42.23 | [TK]D-Fender | withaynoti : And an SPA-2002 for $70 for your 2 phones |
| 02:43.48 | withaynoti | yeah...i just thought there might be something available on the market that was fairly cheap. kinda like linksys's new toy. |
| 02:44.51 | [TK]D-Fender | withaynoti : Whats the price on this "new toy"? Last I checked is enables VM's, but does not STORE them itself (USB drive), and doesn't offer *'s power. |
| 02:46.36 | withaynoti | [TK]D-Fender: absolutely true. which is why i was hoping for something w/ a bit more power to it w/ asterisk pre-installed |
| 02:47.28 | [TK]D-Fender | withaynoti : That system would just fine.... as for preinstalled... hey, its only an hour or two to install a distro, slap * onto it and walk away... |
| 02:47.35 | dlynes_home | withaynoti: there's a guy selling a fully configured asterisk server on ebay for about $65USD |
| 02:47.59 | dlynes_home | withaynoti: he sells them regularly, too |
| 02:48.00 | drray | Hell, I bet you could find someone to do it for you for $50 or so |
| 02:48.32 | [TK]D-Fender | dlynes_home : I did STRETCH that figure.. I can get a P3 thats usable for $75 CAD :) hat is in itself a full PBX, just no implication of supporting actual LINES by default. |
| 02:48.38 | znoG | [TK]D-Fender: nice, will definately consider the Sangoma cards |
| 02:49.35 | philth|work | Another Quick Questions here, When I dial on the ATA portion of the SPA-3000 there is a long delay before the number is dialed, is this with asterisk or is the SPA buffering numbers until it thinks that I am done dialing? |
| 02:50.03 | znoG | philth|work: you need to configure the SPA dialplan |
| 02:51.33 | philth|work | znoG: Oh? This is specific to the SPA or is it in * documentation? |
| 02:57.12 | *** join/#asterisk tpak (n=tpak@c-67-190-182-213.hsd1.co.comcast.net) |
| 02:58.19 | tpak | howdy - anyone used sipp lately with rtp? |
| 02:58.58 | znoG | philth|work: specific to the SPA, haven't seen any other ATAs with dialplans |
| 02:59.24 | znoG | philth|work: you will find it in the SPA documentation as an Appendix "Dial Plan" |
| 03:01.02 | philth|work | I found a wiki with an explination on how these work. I had to do something like this to get the outside line to immediatly ring a voip line. |
| 03:01.56 | philth|work | I don't have it in front of me but it was something like (<:99>S0) and it called "99" within it's context. |
| 03:02.23 | philth|work | Is there a smoother way to forward all calls from the pstn line a specific dialplan on *? |
| 03:03.41 | drray | is there a way to assign lables to Zap/Channels? so that when it says Hungup 'Zap/43-1' it could say "room 123"? |
| 03:03.52 | drray | in verbose mode |
| 03:05.43 | riddlebox | how would I setup a mailbox so that you can press * or # at the greeting to get into voicemail? |
| 03:06.49 | *** join/#asterisk caribou7 (n=Jack@24-236-221-158.dhcp.aldl.mi.charter.com) |
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| 03:06.59 | Twister | i hate 3com! |
| 03:07.20 | [TK]D-Fender | riddlebox : Look at the "a" standard extension.... |
| 03:07.46 | caribou7 | Quick question, has anything been done to change the behavior of the MP3player command in Asterisk? |
| 03:08.33 | caribou7 | I ask because every place I used that command is broken after an upgrade. |
| 03:08.36 | philth|work | znoG: Thanks for your help :) |
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| 03:13.48 | riddlebox | [TK]D-Fender, would this be how it would be used, exten =>123,3,Voicemail(123,a:VoiceMailMain())? |
| 03:15.31 | feld | how long does fxotune take to run? |
| 03:16.16 | Sponge_bob | feld: what's fxotune? |
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| 03:22.32 | [TK]D-Fender | riddlebox : No. Go read up on the * standard extensions "h,a,s,fax, etc....) |
| 03:23.02 | Qwell | i and t are valid too... |
| 03:23.03 | Qwell | so |
| 03:23.09 | Qwell | i,t,h,a,s,fax |
| 03:23.10 | Qwell | heh |
| 03:23.24 | Qwell | or... |
| 03:23.29 | Qwell | fax,s,h,i,a,t |
| 03:27.21 | tpak | anyone know what aprameter to pass to sipp to tell it which pcapplay file to use?? |
| 03:27.31 | Qwell | tpak: yeah, one sec |
| 03:28.07 | Qwell | -sf /path/to/pcap.xml |
| 03:28.39 | tpak | Qwell: thanks - where did you find that? I've been looking under all the wrong rocks... |
| 03:28.46 | Qwell | ./sipp |
| 03:29.45 | tpak | ahhh - that's the scenario file - do I have to set up a custom scenario to use a pcap file? |
| 03:29.52 | Qwell | yes |
| 03:30.05 | Qwell | but you can dump an existing scenario, with -sd I believe |
| 03:30.16 | tpak | yup - will do |
| 03:30.22 | Qwell | or download one.. |
| 03:30.36 | Qwell | the sipp/pcapplay faq has a few |
| 03:31.47 | tpak | download one where is that faq? on the sipp wiki? |
| 03:31.52 | Qwell | yeah |
| 03:32.26 | tpak | thanks - trying to get an outbound dialer call recording scenario going under load... much fun |
| 03:34.41 | riddlebox | [TK]D-Fender, this is what I have found http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 03:37.18 | [TK]D-Fender | riddlebox : Congratulations, your answer is the first entry on that page... |
| 03:37.42 | riddlebox | yeah it says a: |
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| 03:39.28 | callisgn | hey does anyone need any ram? |
| 03:39.34 | Qwell | callisgn: umm..sure? |
| 03:47.18 | Sponge_bob | callisgn: sure why not |
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| 03:59.32 | CrashHD | how can I force asterisk to exit with a non 0 value for testing? |
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| 04:04.54 | Nugget | # echo "!#/bin/sh exit 1" > /usr/bin/asterisk |
| 04:05.38 | CrashHD | ahh |
| 04:05.44 | CrashHD | I didn't think about that |
| 04:06.03 | CrashHD | anyway to actually make the real asterisk prog exit non 0/ |
| 04:06.04 | CrashHD | ? |
| 04:07.17 | znoG | editing the source? :) |
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| 04:14.01 | CrashHD | heh |
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| 04:15.34 | *** join/#asterisk DataCompBoy (n=datacomp@217.8.236.1) |
| 04:15.50 | DataCompBoy | Hello, people! :) |
| 04:16.34 | DataCompBoy | I can't google anything useful... is there asterisk management interface bindings for erlang available? |
| 04:16.41 | Qwell | erlang? |
| 04:16.47 | jsaunders | If sip (ilbc) in -> sip (g723) out had calls dropping about 10% of the time, well into the conversation, would you attribute it to the network connection or more likely a faulty codec or something along those lines? I am thinking even with hiccups and bumps in a dsl connection, the worst you would experience would be delay and degraded audio quality, shouldn't drop the call I wouldn't think. However, if a codec had a hiccup, t |
| 04:16.47 | jsaunders | hat could mess up the rtp no? Hmm. Much to ponder. |
| 04:18.30 | DataCompBoy | also, what state of MGCP module for asterisk to act as media gateway? still only "wanted"? |
| 04:18.52 | DataCompBoy | on wiki bounty from 2004 year:) |
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| 04:50.47 | m_a_g_o | good evening... I need to recompile say.c so * supports the spanish syntax for two different sound folders. I tried setting the language to the folder names and sounds are taken correctly, but the syntax used is english... could someone please guide me to recompile after making the modifications? =) |
| 04:50.47 | *** join/#asterisk zepmantra (i=wahhh@203.215.100.96) |
| 04:51.20 | m_a_g_o | gcc ... ? |
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| 04:55.11 | alephcom | m_a_g_o=Max? |
| 04:55.21 | m_a_g_o | yes =) |
| 04:55.30 | alephcom | ok, I won't go ask Max then. :-) |
| 04:55.36 | Telamon | Is there a way to determine if an extension exists in a given context? IE, ExtensionExists(default,104)? The only thing I've been able to come up with is Gosub-ing to it and then put in an i,1,return |
| 04:56.18 | m_a_g_o | not much progress, still where I left |
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| 05:01.06 | CunningPike | Telamon: That would be how I would do it...... |
| 05:01.39 | CunningPike | ~seen dlynes_home |
| 05:01.49 | jbot | dlynes_home is currently on #asterisk (22h 9m 46s). Has said a total of 255 messages. Is idling for 2h 13m 50s, last said: 'withaynoti: he sells them regularly, too'. |
| 05:01.59 | CunningPike | ~seen dlynes_office |
| 05:02.01 | jbot | dlynes_office is currently on #asterisk (5h 27m 50s), last said: 'same here, but then again, i don't use queues or agents, either'. |
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| 05:10.50 | dlynes_home | CunningPike: i'm back now |
| 05:10.56 | dlynes_home | CunningPike: just had to run to guildford and back |
| 05:11.03 | dlynes_home | CunningPike: what's up?' |
| 05:11.20 | CunningPike | dlynes_home: Just wanted to thank you for your efforts on behalf of a Vancouver AUG |
| 05:11.26 | dlynes_home | ah |
| 05:11.26 | dlynes_home | heh |
| 05:11.46 | dlynes_home | Any of those guys that responded were people that had responded previously? |
| 05:12.01 | CunningPike | dlynes_home: No, I don't believe so |
| 05:14.10 | dlynes_home | Do you have enough yet? I'm guessing no? |
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| 05:37.59 | stephane_ | jour |
| 05:38.10 | dlynes_home | nuit |
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| 05:45.44 | NewSole | hmmm |
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| 06:08.30 | Beighto | Anybody have a minute to help me troubleshoot my conferencing in AAH? |
| 06:09.21 | Beighto | ok |
| 06:09.24 | *** join/#asterisk newsmafia (n=newsmafi@wsip-70-166-5-130.sd.sd.cox.net) |
| 06:09.58 | dlynes_home | Beighto: freepbx config files are just so much different from the default config files that we aren't able to help much with them in this channel |
| 06:10.33 | Beighto | I switched over to #freepbx, thanks |
| 06:12.27 | CunningPike | dlynes_home: |
| 06:12.52 | CunningPike | dlynes_home: We have about a half dozen right now - not really enough, I don't think.... |
| 06:13.13 | dlynes_home | yeah...that's what i figured |
| 06:13.26 | CunningPike | dlynes_home: Give it time - I'll ask again after the summer |
| 06:13.28 | CunningPike | dlynes_home: |
| 06:13.36 | CunningPike | dlynes_home: Off to bed now.... ttyl |
| 06:16.22 | *** join/#asterisk lorinc (n=ang@caracas-3689.adsl.interware.hu) |
| 06:18.49 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 06:19.46 | *** join/#asterisk oej (n=oej@213.115.215.54) |
| 06:20.32 | Beighto | While I am here, if one is serving multiple area codes and would like each area code to be able to dial a number as if it were local and attach the appropriate area code in asterisk automatically, is this possible on a single server? |
| 06:21.12 | dlynes_home | Beighto: huh? you mean do seven digit dialing? |
| 06:21.12 | x86 | sure |
| 06:21.45 | Beighto | yes, 7 digit dialing and automatically add the correct area code according to the context |
| 06:22.00 | dlynes_home | Beighto: oh...like if they're in the 708 area code, and they dial seven digits, automatically add the 1708? |
| 06:22.05 | Beighto | and any ideas how one would do that? In the extensions.conf ? |
| 06:22.06 | x86 | Beighto: make a phantom extension where people can dial and enter their area code, then in your dialplan, capture 7 digit dialing and query astdb to see if there is a 3 digit code for the given caller ID |
| 06:22.06 | cjk | hi, is anyone here using batch=yes in cdr.conf? if so, has anyone tested to enable very small batches (size less than 10) and a time of less than 30 seconds? |
| 06:24.20 | Beighto | Thats a bit out of my experience at this point x86, but I will hold on to that idea as I learn more. Do you have any other suggestions, or is that basically a solid way of handling it? |
| 06:25.17 | Beighto | so, dlynes_home I would like to add the 1708 to the 7 digit dial, and then a 1515 to another 7 digit dial under a different context |
| 06:25.47 | x86 | Beighto: it's either that or implement an AGI :) |
| 06:25.59 | dlynes_home | Beighto: if you want an easier way of doing it, but that requires more work, set up different outgoing contexts for each area code |
| 06:26.03 | x86 | Beighto: i do it with an AGI, just because I want to use a MySQL backend and not astdb |
| 06:26.14 | dlynes_home | Beighto: then set up a variable for their area codes for that outgoing context |
| 06:26.28 | dlynes_home | Beighto: before you go into the outgoing context |
| 06:26.56 | dlynes_home | Beighto: then if they do a 7 digit dial, prepend that area code to the 7 digits |
| 06:27.37 | Beighto | what is the command you guys are using to direct a message to me? |
| 06:27.57 | dlynes_home | huh? |
| 06:28.04 | dlynes_home | waht are you talking about? |
| 06:28.27 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43.pool8172.interbusiness.it) |
| 06:28.27 | Beighto | the irc command to direct the messag |
| 06:28.37 | Beighto | dlynes_home>Beighto: then if they do a 7 digit dial, prepend that area code to the 7 digits |
| 06:29.09 | dlynes_home | i think you mean highlighting it |
| 06:29.21 | dlynes_home | your irc client does that automatically when your name appears in the line |
| 06:29.43 | dlynes_home | Beighto: highlighted |
| 06:29.48 | dlynes_home | not highlighted |
| 06:29.57 | Beighto | ahh |
| 06:29.59 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:30.08 | Beighto | dlynes_home: highlighted? |
| 06:30.18 | dlynes_home | correct |
| 06:30.30 | dlynes_home | or beighto is highlighted beighto |
| 06:30.41 | Beighto | cool |
| 06:31.36 | Beighto | x86: I'm not up to learn another programming language, I have to learn the asterisk language first. So I think AGI is out for now. |
| 06:31.56 | kmilitzer | Morning everyone ... anyone going to astricon berlin on monday? |
| 06:32.08 | oej | absolutely not |
| 06:32.14 | oej | But morning anyway |
| 06:32.36 | kmilitzer | oej: I know that you are not going ;) |
| 06:34.07 | Beighto | dlynes_home I just finished reading all that stuff you sent me about the outgoing contexts, and that is more along my level at the moment and I will give that a try. Thank you, maybe I can try x86's method once I have your way figured out completely. |
| 06:34.23 | dlynes_home | Beighto: well, that's what i figured |
| 06:34.30 | dlynes_home | Beighto: mine's more work, but less learning curve |
| 06:36.11 | Beighto | exactly, do any of you have any recommendations on reading material. So far I am reading Orielly's Asterisk Future of Telephony and the Asteriskguru tutorials |
| 06:36.52 | dlynes_home | ~docs |
| 06:36.59 | jbot | docs is probably probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 06:36.59 | dlynes_home | ~wikis |
| 06:37.01 | jbot | methinks wikis is http://www.voip-info.org |
| 06:38.17 | Beighto | good stuff, thanks... I'll try and spend more time in the channel and hopefully help some people that are noobier than me. |
| 06:39.09 | *** join/#asterisk pjchilds (n=pjchilds@pdpc/supporter/student/pjchilds) |
| 06:40.18 | dlynes_home | Beighto: well, what goes around, comes around, as they say |
| 06:41.05 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 06:41.59 | *** join/#asterisk nexxstar (n=nexstar@ip68-111-77-138.oc.oc.cox.net) |
| 06:42.15 | nexxstar | when voicemails are sent to email as attachment, the files cant be played, what would cause this? |
| 06:44.03 | dlynes_home | nexxstar: maybe you're sending them as gsm attachments, and your email client doesn't know how to play gsm files? |
| 06:44.44 | nexxstar | let me check |
| 06:45.55 | nexxstar | what other file types can i send them as and where is that changeable? |
| 06:46.08 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 06:46.18 | nexxstar | they are sending as .wav49 |
| 06:46.59 | dlynes_home | nexxstar: well, if you know what format they're in, i'm sure you know where it's changeable |
| 06:47.23 | nexxstar | im helping someone with this, not my actual box |
| 06:47.25 | dlynes_home | nexxstar: take a look at the sample voicemail.conf file that gets installed as part of the original install when you do make samples |
| 06:47.30 | nexxstar | would wav49 be ok though? |
| 06:47.40 | dlynes_home | nexxstar: that'll tell you what formats are available |
| 06:47.45 | dlynes_home | nexxstar: should be, yeah |
| 06:47.55 | nexxstar | the atachment get sent |
| 06:48.14 | dlynes_home | yeah...wav49 is microsoft wav file format |
| 06:48.23 | nexxstar | u go to open it.. on any of the computers and it goes to open and shows the window loading then nothing |
| 06:48.30 | nexxstar | but some random times it will play |
| 06:48.39 | nexxstar | 1 out of 30 tries |
| 06:48.42 | dlynes_home | nexxstar: maybe you've got a crappy default wav file player then |
| 06:48.55 | dlynes_home | or maybe it's an empty wav file |
| 06:49.09 | dlynes_home | i suspect the latter |
| 06:49.12 | nexxstar | but when it plays, it plays |
| 06:49.18 | nexxstar | not like its playing nothing |
| 06:49.26 | dlynes_home | i.e. someone hung up without leaving a message on the ones that don't load |
| 06:49.47 | dlynes_home | or you're saying for one particular attachment |
| 06:49.53 | dlynes_home | sometimes it'll play, and sometimes it won't? |
| 06:49.58 | *** join/#asterisk LH-euhost (n=LH-euhos@pD9E1E34D.dip0.t-ipconnect.de) |
| 06:49.59 | nexxstar | yes |
| 06:50.04 | nexxstar | sometimes yes, sometimes no |
| 06:50.12 | dlynes_home | then that's something screwed up on the windows machine that's trying to open it |
| 06:50.13 | nexxstar | we called and left a bunch of voicemails |
| 06:50.17 | nexxstar | on different phones |
| 06:50.24 | nexxstar | all the wav files do it |
| 06:50.27 | nexxstar | none will play |
| 06:50.32 | nexxstar | then one randomly will |
| 06:50.35 | *** join/#asterisk aclark78 (n=aclark@cpe-70-116-103-202.houston.res.rr.com) |
| 06:50.44 | dlynes_home | tell them to reinstall windows on that machine then |
| 06:50.50 | nexxstar | wev tried on 10different computers |
| 06:50.59 | nexxstar | even our own |
| 06:54.02 | dlynes_home | but say you leave one voicemail |
| 06:54.18 | dlynes_home | you open that one email ten times, and maybe one out of those ten files, that wav file will play? |
| 06:54.27 | dlynes_home | s/files/times/ |
| 06:54.30 | nexxstar | yes |
| 06:54.34 | nexxstar | but more like 30 times |
| 06:54.53 | *** join/#asterisk Shoragan (n=shoragan@134.169.175.72) |
| 06:54.55 | dlynes_home | but not 30 different voicemails right? the same voicemail? |
| 06:55.00 | nexxstar | yes |
| 06:55.14 | nexxstar | and tried different voicemails on different machines |
| 06:55.22 | dlynes_home | have you tried a different email client? |
| 06:55.50 | nexxstar | im told it may be a broken link problem |
| 06:55.57 | nexxstar | yes, tried different email clients |
| 06:55.59 | dlynes_home | broken link? |
| 06:56.05 | dlynes_home | it should be a file attached to the message |
| 06:56.23 | dlynes_home | can you forward me the email? |
| 06:56.36 | nexxstar | it is, but it wont let you download it eather, only sometimes |
| 06:56.46 | nexxstar | yes, you can forward it, same issue |
| 06:56.52 | dlynes_home | try forwarding it to me |
| 06:56.58 | dlynes_home | dlynes@cwazy.co.uk |
| 06:57.06 | nexxstar | 1 sec |
| 06:57.08 | dlynes_home | I'll take a look at it |
| 06:57.15 | dlynes_home | Maybe there's something wrong with the email |
| 06:57.21 | dlynes_home | Which version of Asterisk are you using? |
| 06:57.27 | dlynes_home | And what mail server are you using? |
| 06:57.33 | nexxstar | exchange |
| 06:57.43 | dlynes_home | on the asterisk box? |
| 06:57.46 | nexxstar | no |
| 06:57.51 | nexxstar | windows server |
| 06:57.51 | dlynes_home | exchange doesn't run on linux |
| 06:57.52 | nexxstar | for email |
| 06:58.06 | nexxstar | Version 2.1beta1 |
| 06:58.08 | dlynes_home | You've got a mail server you're running on the asterisk box too |
| 06:58.15 | dlynes_home | and it talks to your exchange box |
| 06:58.21 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 06:58.30 | dlynes_home | which mail server are you running on the asterisk box? |
| 06:58.46 | dlynes_home | is it sendmail? postfix? qmail? |
| 06:59.24 | nexxstar | sorry |
| 06:59.28 | nexxstar | gmail |
| 06:59.38 | dlynes_home | gmail is a webmail client |
| 06:59.43 | dlynes_home | what's yoru mail server? |
| 07:00.05 | nexxstar | let me check, il still a little new at this |
| 07:00.09 | nexxstar | 1 sec |
| 07:00.27 | DrkShdw | nexxstar: try telneting to your asterisk machine on port 25. it'll prolly tell you what it is. |
| 07:00.27 | Nugget | telnet is eeeeeeevil! |
| 07:00.39 | nexxstar | i can ssh |
| 07:00.48 | DrkShdw | don't ssh. telnet |
| 07:01.07 | nexxstar | i can get to teh seb inerface remotly |
| 07:01.11 | nexxstar | web |
| 07:01.12 | aclark78 | anyone have any SIP phone recommendations, or brands to avoid? |
| 07:01.19 | dlynes_home | DrkShdw: not if he's only using it as an mua, and not as an mta |
| 07:01.19 | nexxstar | get polycom 501 |
| 07:01.29 | nexxstar | ordered 500, not one was bad |
| 07:01.44 | aclark78 | i was looking at those, is there speakerphone good? |
| 07:01.49 | nexxstar | dont get cisco, sucks, and too much $ |
| 07:01.54 | nexxstar | linksys sucks too |
| 07:01.56 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 07:02.05 | nexxstar | dude, audio is sooo nice on it |
| 07:02.12 | DrkShdw | dlynes_home: hence the word "try".. many standard linux installs still have sendmail open on 25 still |
| 07:02.41 | dlynes_home | nexxstar: ls -al `which sendmail` to find out which mailer you're using |
| 07:02.57 | nexxstar | i can telnet into it |
| 07:02.59 | nexxstar | cant |
| 07:03.37 | dlynes_home | nexxstar: if it's not a symbolic link, you're using sendmail; otherwise, you're using whatever it's a symbolic link to |
| 07:04.01 | dlynes_home | or if it's a really small size, it might be a shell script wrapper for another mailer |
| 07:04.41 | nexxstar | ls -al `which sendmail |
| 07:04.43 | nexxstar | no dir |
| 07:05.02 | dlynes_home | no dir? huh? |
| 07:05.09 | dlynes_home | what's dir? |
| 07:05.31 | nexxstar | ls: which sendmail: No such file or directory |
| 07:05.48 | dlynes_home | yeah |
| 07:05.59 | *** join/#asterisk X-Rob (n=rob@CPE-60-231-85-52.qld.bigpond.net.au) |
| 07:06.01 | aclark78 | you have to put which sendmail in backticks, the ` next to the 1 on US keyboards, not the ' |
| 07:06.03 | dlynes_home | you didn't copy/paste what i said...that's why |
| 07:06.13 | dlynes_home | ls -al `which sendmail` |
| 07:06.19 | nexxstar | i did that |
| 07:06.27 | nexxstar | here... |
| 07:06.28 | dlynes_home | note the first __and__ final backticks |
| 07:06.34 | nexxstar | ya |
| 07:06.35 | nexxstar | i know |
| 07:06.50 | dlynes_home | then try it like this: ls -al $(which sendmail) |
| 07:06.57 | nexxstar | ltasterisk1:~# ls -al 'which sendmail' |
| 07:07.00 | X-Rob | more stupid people, dlynes_home ? |
| 07:07.10 | X-Rob | YAY |
| 07:07.11 | nexxstar | k, 1 sec |
| 07:07.12 | dlynes_home | nexxstar: those are forward quotes, not backquotes |
| 07:07.14 | X-Rob | indeed there are |
| 07:07.28 | X-Rob | ` is not ' |
| 07:07.57 | nexxstar | lrwxrwxrwx 1 root root 5 2006-05-09 23:11 /usr/sbin/sendmail -> exim4 |
| 07:08.02 | X-Rob | yay |
| 07:08.08 | dlynes_home | nexxstar: so you're using exim for your mailer then |
| 07:08.12 | dlynes_home | not exchange |
| 07:08.31 | nexxstar | well its exchange for the windows server |
| 07:08.34 | dlynes_home | X-Rob: you know of any issues with exim mangling attachments? |
| 07:08.36 | nexxstar | but that doesent matter |
| 07:08.38 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 07:08.56 | X-Rob | lilo, mmmm, not that jumps out at me. How mangled is it? |
| 07:08.58 | X-Rob | uh |
| 07:09.07 | X-Rob | s/lilo/dlynes_home/ |
| 07:09.12 | nexxstar | cant download attachments |
| 07:09.12 | dlynes_home | lol |
| 07:09.14 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 07:09.17 | *** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-14-3.w86-207.abo.wanadoo.fr) |
| 07:09.20 | X-Rob | dunno how I managed that. |
| 07:09.22 | dlynes_home | nexxstar: have you forwarded that voicemail to me yet? |
| 07:09.25 | nexxstar | files will play 1 out of 30 tries |
| 07:09.28 | nexxstar | very random |
| 07:09.39 | X-Rob | use postfix. lt's l33t. |
| 07:09.43 | nexxstar | i cant get into the companys email unless im there, i tried |
| 07:09.47 | X-Rob | with a capital 3! |
| 07:09.50 | X-Rob | two of them, even. |
| 07:09.58 | nexxstar | how do i change that |
| 07:10.00 | dlynes_home | X-Rob: i doubt he knows how to set up postfix...he didn't even know what mailer he was using |
| 07:10.12 | X-Rob | nexxstar, which distro are you using? |
| 07:10.19 | nexxstar | i can get the info to someone that may know |
| 07:10.23 | nexxstar | debian |
| 07:10.31 | X-Rob | ooh. |
| 07:10.34 | X-Rob | buggered if I know. |
| 07:10.40 | X-Rob | 'yum install postfix' works on centos 8) |
| 07:10.57 | nexxstar | apt-get install postfix |
| 07:10.59 | dlynes_home | omfg |
| 07:11.09 | dlynes_home | what a stupid waste of money this movie was |
| 07:11.18 | X-Rob | dlynes_home, bloodrayne? |
| 07:11.28 | dlynes_home | Mr. Accident Movie |
| 07:11.41 | dlynes_home | I think it was so low, it didn't even rank as 1/2 star |
| 07:12.22 | dlynes_home | It's about some clown that uncovers a plan to sell nicotine laced eggs |
| 07:12.33 | dlynes_home | starring yahoo serious whoever the hell that is |
| 07:12.48 | X-Rob | no way |
| 07:12.54 | X-Rob | he's an australian actor |
| 07:13.04 | X-Rob | was in young einstein |
| 07:13.08 | dlynes_home | ah |
| 07:13.16 | X-Rob | in which he discovered how to split the atom to make beer foamy. |
| 07:13.49 | dlynes_home | sounds like another dumb movie |
| 07:14.57 | X-Rob | oh yes |
| 07:15.05 | X-Rob | he had an affair with madame curie, too |
| 07:15.08 | X-Rob | (in the movie) |
| 07:15.38 | dlynes_home | cool |
| 07:15.41 | dlynes_home | was she a dog? |
| 07:15.55 | X-Rob | nah |
| 07:15.56 | X-Rob | hawt. |
| 07:16.02 | X-Rob | radioactive, but hawt. |
| 07:16.04 | dlynes_home | nexxstar: i'm still waiting for that email |
| 07:16.10 | *** join/#asterisk arcraig (n=arcraig@c-67-181-157-7.hsd1.ca.comcast.net) |
| 07:16.23 | nexxstar | i cant get into the company email |
| 07:16.27 | nexxstar | i have to wait till i get there |
| 07:17.22 | dlynes_home | nexxstar: you can't make asterisk email the voicemail somewhere else? |
| 07:18.35 | dlynes_home | nexxstar: i.e. edit voicemail.conf, change the email address for the extension, do a reload at the asterisk cli, leave a voicemail, edit voicemail.conf, change the email address back to the original email address, reload from the asterisk cli again |
| 07:19.14 | nexxstar | i could, waiting to see if this post fix is going to work first |
| 07:20.23 | nexxstar | cli? |
| 07:20.34 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 07:20.39 | aclark78 | command line interface / console |
| 07:20.54 | nexxstar | what is the reload command? |
| 07:21.34 | *** join/#asterisk Gabriel25 (n=gabe@user-12ld5f7.cable.mindspring.com) |
| 07:21.38 | Gabriel25 | fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf |
| 07:22.22 | Gabriel25 | <PROTECTED> |
| 07:22.23 | Gabriel25 | <PROTECTED> |
| 07:22.33 | Gabriel25 | what can be the problem ? |
| 07:22.44 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 07:22.59 | caribou7 | X-Rob, you heard of anyone having problems with the MP3player command in the latest TrixBox? |
| 07:23.41 | X-Rob | Gabriel25, above that, it'll actually try a Dial - you need to look in /var/log/asterisk/full to find out why the Dial failed |
| 07:23.51 | caribou7 | (I do not mean Music on Hold, that works fine) |
| 07:24.05 | X-Rob | caribou7, what command then? |
| 07:24.17 | caribou7 | MP3player |
| 07:24.57 | X-Rob | [Description] |
| 07:24.58 | X-Rob | <PROTECTED> |
| 07:24.58 | X-Rob | which typically would be a filename or a URL. User can exit by pressing |
| 07:24.58 | X-Rob | any key on the dialpad, or by hanging up. |
| 07:24.58 | X-Rob | wpm4l-gw*CLI> |
| 07:25.09 | X-Rob | being that mpg123 isn't installed on trixbox, that's probably the reason |
| 07:25.22 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 07:25.36 | caribou7 | *21,5,MP3player(/tmp/foo.mp3) |
| 07:25.54 | caribou7 | It isn't? How do they play Music on Hold then? |
| 07:27.00 | X-Rob | format_mp3 |
| 07:27.07 | caribou7 | -bash: mpg123: command not found |
| 07:27.10 | X-Rob | bingo |
| 07:27.18 | caribou7 | Hmmmm |
| 07:28.06 | caribou7 | Is format_mp3 a command that will play MP3's or is that something deeper? |
| 07:29.57 | caribou7 | Looks like you are busy but thanks, you have at least answered the mystery! |
| 07:33.00 | *** join/#asterisk tparcina (n=tparcina@lns02-0309.dsl.iskon.hr) |
| 07:33.07 | tparcina | good morning group! |
| 07:34.27 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
| 07:35.29 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
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| 07:43.09 | *** part/#asterisk aclark78 (n=aclark@cpe-70-116-103-202.houston.res.rr.com) |
| 07:43.49 | *** join/#asterisk michael-i (n=michael@141.41.38.58) |
| 07:46.30 | *** join/#asterisk af_ (n=af@ip-164-240.sn2.eutelia.it) |
| 07:46.38 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
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| 07:54.38 | tparcina | there is houge discussion going on here :)) |
| 07:55.25 | tparcina | is Internet down and I don't know about it? :))) |
| 07:56.39 | *** join/#asterisk loopt (n=pt@gw1.sanyo.hu) |
| 07:57.59 | *** join/#asterisk Gabriel25 (n=gabe@user-12ld5f7.cable.mindspring.com) |
| 07:58.10 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 08:00.04 | tparcina | questin, has anybody worked with A, B and C caller ID num? |
| 08:01.12 | tparcina | As far as i know that should be A-ID of caller; B-ID of called number; C - ID of redirected number. |
| 08:03.01 | tparcina | now i have problem, when I put forward on my phone, asterisk in CDR writes wrong data (or maybee right data, but unusefull to me). I would like that asterisk in cdr writes data from which phone redirection was made instead of caller ID of number that was redirected |
| 08:03.07 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 08:03.23 | tparcina | i hope that someone has understand what is the problem... |
| 08:03.41 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:04.26 | X-Rob | tparcina, you can't do it - I logged a bug about that the other day. The short answer is 'no' and the long answer is 'really, no. Do your own CDR's' |
| 08:05.12 | *** join/#asterisk michael-i (n=michael@141.41.38.58) |
| 08:05.47 | *** join/#asterisk jpeeler (n=jpeeler4@host81-154-150-237.range81-154.btcentralplus.com) |
| 08:11.28 | *** join/#asterisk swytch (n=ezcall@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 08:20.05 | *** join/#asterisk Sonderblade (n=mah@host-213.131.147.169.addr.tdcsong.se) |
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| 08:37.00 | tparcina | X-Rob, that is the second thing that i have heard that asterisk can't do it... |
| 08:37.34 | tparcina | X-Rob: at least for this one asterisk can do in on another way, so it isn't big problem... :) |
| 08:38.42 | tparcina | X-Rob: so, have you made you own cdr's? is it posible to put your own crd's to database using ODBC drivers? |
| 08:39.09 | X-Rob | tparcina, you can do it any way you want - using DeadAGI, or the mysql application |
| 08:43.38 | tparcina | X-Rob: yes, by default asterisk saves default data do database, how to make him to save my CDR's (instead of default) to database? |
| 08:44.02 | X-Rob | you have to write your own CDR stuff, using an AGI or mysql. |
| 08:44.03 | *** join/#asterisk gfox (n=vince@calvix/staff/gfox) |
| 08:44.04 | X-Rob | (as I said) |
| 08:46.01 | gfox | hi |
| 08:46.38 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 08:46.59 | *** join/#asterisk basty (n=basty@212.218.65.201) |
| 08:47.01 | basty | Hi |
| 08:48.18 | tparcina | X-Rob: sorry, didn't understand that that was the answer on my question, i thouth that was for something else... thank you |
| 08:48.58 | gfox | do you know where I can find some hints about asterisk exit status (return code) values and significations |
| 08:49.09 | gfox | rapidly checking the code I can see most of the time the error status is 1 |
| 08:49.12 | basty | I have setup a queue - it works perfect. Now if I use my Snom to forward incomming calls to an external number, i receive "-- Got SIP response 480 "Do Not Disturb" back from 172.16.10.54" and I run into the specified queue instead of the call-forwarding to the external number. Anyone could give me a hand? |
| 08:52.00 | basty | my extensions.conf is: exten => _249XXX,1,SetMusicOnHold(default) - exten => _249XXX,2,Dial(SIP/basty,10) - exten => _249XXX,3,Answer() - exten => _249XXX,4,Queue(empfangs-queue|t|||40) - exten => _249XXX,5,VoiceMail(u130) - exten => _249XXX,6,PlayBack(vm-goodbye) - exten => _249XXX,7,Wait(2) - exten => _249XXX,8,Hangup() |
| 08:54.01 | bon | anyone round willing to help me with extensions & pattern matching? :) |
| 08:58.26 | qdk | bon: just aask! |
| 08:58.30 | qdk | -a |
| 08:59.35 | bon | ok:) |
| 09:00.06 | bon | qdk: http://pastebin.com/712527 |
| 09:00.15 | bon | my question to that extension is |
| 09:00.31 | bon | if i dial 3300XXXX it matches the first entry and dials correctly |
| 09:00.48 | bon | BUT if i dial XX3300XXXX it matches not the 2nd but the 3rd |
| 09:00.52 | bon | am i missing a brain? :) |
| 09:02.25 | qdk | what are you dialing exactly. |
| 09:02.31 | *** join/#asterisk Eggplant (n=none@dsl-216-155-214-007.cascadeaccess.com) |
| 09:03.00 | bon | 023300XXXX |
| 09:03.47 | qdk | bon: _0 is more specific than _X |
| 09:04.04 | bon | well but that shouldn't influence it, no? |
| 09:04.35 | bon | what i would like to achieve would be <anything_preceeding_it>3300XXXX -> first context & anything_else -> 2nd |
| 09:04.38 | qdk | bon: the rule is most specific win... but i have read something about a bug in the rule, so im not 100% sure. |
| 09:05.35 | qdk | bon: then remove the _0 extentions? |
| 09:06.20 | bon | the 3rd one you mean? |
| 09:06.33 | bon | where would all the rest fall then? |
| 09:12.38 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 09:18.20 | *** join/#asterisk LokeshIndian (n=lokesh_k@estrela.nortenet.pt) |
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| 09:30.10 | *** join/#asterisk oej (n=oej@213.115.215.54) |
| 09:36.26 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 09:36.29 | Bert- | hello |
| 09:36.42 | bon | hey |
| 09:37.19 | qdk | bon: yes, that would be a problem. |
| 09:38.53 | bon | hm |
| 09:38.53 | bon | ok |
| 09:38.56 | bon | so i found out |
| 09:39.03 | bon | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting |
| 09:39.54 | *** join/#asterisk pa (n=paolo@unaffiliated/pa) |
| 09:39.55 | bon | but anyway, if i have _.3300XXXX and _0. and i dial 0023300XXXX which pattern does it match? |
| 09:43.44 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 09:44.26 | qdk | bon: im guessing _0 |
| 09:46.53 | bon | qdk: hm but that is in the included context, it should dial the first pattern it matches which is _.3300XXXX |
| 09:47.05 | bon | which actually doesn't happen :) |
| 09:47.50 | qdk | bon: no, the rule is most specific |
| 09:47.54 | qdk | not first match |
| 09:48.03 | bon | argh |
| 09:48.12 | bon | so how do i achieve what i need? :) |
| 09:48.21 | Bert- | well, I have a little issue about NAT and one way audio on asterisk |
| 09:48.36 | qdk | bon: and i most specific "dont work" then i think its last match, |
| 09:48.46 | Bert- | Can I only put asterisk in DMZ ? |
| 09:48.56 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 09:48.57 | Bert- | to avoid nat issues ? |
| 09:49.06 | qdk | Bert-: yes, that is mostly due to NAT misconfiguration. |
| 09:49.33 | qdk | Bert-: DMZ is also behind NAT. |
| 09:50.02 | Bert- | I totally agree |
| 09:50.03 | Bert- | is a good tutorial about that somewhere ? |
| 09:52.31 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 09:52.32 | Bert- | ... |
| 09:53.11 | bon | qdk: quite misunderstood your suggestion, sorry :) |
| 09:53.26 | qdk | Bert-: try google? not really an * question. |
| 09:54.42 | qdk | Bert-: its just a matter of portforwaring the correct ports to the * behind the NAT... look in your manual for your NAT-device to see how its done. |
| 09:55.00 | Bert- | yes |
| 09:55.02 | Bert- | the way is |
| 09:55.08 | Bert- | I already done that |
| 09:55.23 | Bert- | I routed port 5060 UDP 8000 UDP and rtp range UDP 10000-200000 |
| 09:55.35 | Bert- | I setup public IP address |
| 09:55.51 | Bert- | I'll find what's wrong :) |
| 09:57.45 | Bert- | hmmm |
| 09:59.19 | *** join/#asterisk rkr245 (n=ravi@81.21.33.35) |
| 09:59.30 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 09:59.47 | Bert- | f*ck*ing connction ... |
| 10:00.02 | Bert- | every time people call on our pstn phone, we lost synchro |
| 10:00.25 | Bert- | our national telecom provider is still unable to fix that |
| 10:00.26 | Bert- | ... |
| 10:00.39 | *** part/#asterisk rkr245 (n=ravi@81.21.33.35) |
| 10:01.41 | *** part/#asterisk tdi (n=tdi@reykin.pozman.pl) |
| 10:01.41 | qdk | Bert-: telnet the to port through the nat and check with tcpdump if any data gets through. |
| 10:01.42 | Nugget | telnet is eeeeeeevil! |
| 10:01.53 | qdk | telnet to the* |
| 10:06.36 | *** join/#asterisk jeffik (n=Jeff@kns223.NetSurf.Net) |
| 10:10.53 | Bert- | I'll try by using ethereal instead of telnet :) |
| 10:11.33 | qdk | Bert-: i hope you mean instead of tcpdump. |
| 10:12.21 | Bert- | lol |
| 10:12.23 | Bert- | yep sorry |
| 10:20.57 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 10:27.34 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 10:28.41 | Bert- | the way is |
| 10:28.53 | Bert- | I have one way audio but I can hear called party |
| 10:29.00 | Bert- | so nat is ok, as I receive voice packets |
| 10:29.05 | Bert- | but called party can't hear me |
| 10:29.23 | Bert- | am I wrong ? |
| 10:30.06 | sternn | Is called party behind their own NAT? |
| 10:30.44 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net) |
| 10:31.44 | *** join/#asterisk CrlMan (n=CrlMan@unaffiliated/crlman2) |
| 10:32.51 | *** join/#asterisk tparcina (n=tparcina@lns02-0309.dsl.iskon.hr) |
| 10:33.39 | LH-euhost | hello, i am doing a callback with * and i have a problem that asterisk does not hang up my initial call. |
| 10:33.39 | LH-euhost | after i make the callback i make a hangup, but it does not hangup actually. any hints? exten => 498092320/0173351,3,Hangup |
| 10:33.48 | Bert- | I don't now |
| 10:33.50 | Bert- | know |
| 10:33.56 | Bert- | it is a sip provider |
| 10:34.49 | tparcina | LH-euhost, what is this / in 498092320/0173351 for? |
| 10:36.44 | tparcina | does anybody have more informations about "promiscredir" in sip.conf? |
| 10:38.36 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
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| 10:46.59 | tzanger | tparcina: he's doing CID-based routing |
| 11:04.21 | tparcina | tzanger, do you have any link where i could reade more about it? |
| 11:05.16 | tzanger | yeah, the asterisk handbook draft, the asterisk book that was published last year, voip-info.org... this isn't secret stuff, but you are required to do your homework |
| 11:08.32 | *** join/#asterisk sep (n=sep@217.17.211.40) |
| 11:10.06 | sep | i once read a guide/into/doc on how to configure asterisk so it could stand between telco and old propritary PBX on the isdn interfaces, but i have lost the url and can not seam to locate it again in google. does anyone know of a doc describing such a setup ? |
| 11:14.29 | tzanger | I do that now with PRI |
| 11:14.54 | *** join/#asterisk Gimp- (n=Gimp@c-69-136-138-107.hsd1.mi.comcast.net) |
| 11:15.00 | tzanger | you havw two interfaces, one to the telco where you are the network termination, and one that you plug into the PBX where you're loop termination |
| 11:15.14 | sep | tzanger, do you have a doc/url for that setup ? |
| 11:15.23 | tzanger | (interface facing telco is the CPE side, interface facing the pBX is the CO side) |
| 11:15.28 | tzanger | sep: there isn't anythign much to it |
| 11:15.52 | tzanger | you receive a call on the telco side and Dial(Zap/g2/${EXTEN}) on the PBX side |
| 11:16.06 | tzanger | you receive a call on the PBX side and Dial(Zap/g1/${EXTEN}) on the telco side |
| 11:16.22 | sep | i guess ill give it a shot |
| 11:18.14 | tzanger | we're here if you need help |
| 11:18.17 | Gimp- | hey, im looking to set up an asterisk box but im not sure what hardware will best suit my needs. i use vonage and i basically just want to set up voicemail for a single phone. could anyone help me out? ;) |
| 11:18.39 | drray | you just make the asterisk box look like the telco and the pbx wont know |
| 11:18.39 | tzanger | vonage and asteriskd on't get along |
| 11:19.08 | Gimp- | theres no way to get it to work with vonage? :( |
| 11:19.22 | tzanger | sure there is |
| 11:19.25 | tzanger | it's just suboptimal |
| 11:19.31 | Gimp- | how so? |
| 11:19.36 | tzanger | you take an FXO card and plug it into the vonage box |
| 11:19.39 | drray | ata to fxo |
| 11:19.50 | tzanger | you take an FXS card and plug it into a regular phone |
| 11:20.06 | tzanger | or yes, an ATA will work too |
| 11:21.05 | Gimp- | ah |
| 11:21.19 | Gimp- | how much do you think it would cost to buy those two cards? |
| 11:21.29 | drray | I personally don't get why you would use vonage (or skype) with asterisk |
| 11:22.05 | Gimp- | well i save alot of money using vonage but their voicemail system is horrible |
| 11:22.23 | tzanger | Gimp-: you'll probably save more with another provide |
| 11:22.26 | tzanger | er provider |
| 11:22.35 | drray | minutes are cheap |
| 11:22.39 | tzanger | there are plenty who offer rates comparable to vonage |
| 11:23.05 | tzanger | and then you don't have to waste your money trying to connect through them by going through a double A-D on your side |
| 11:23.16 | drray | or being locked out of your own hardware |
| 11:23.42 | Gimp- | yeah.. i don't really like their router |
| 11:23.45 | drray | I mean don't get me wrong, you have someone to bitch to if you use vonage |
| 11:24.22 | Gimp- | do you have any suggestions for service providers that are the same price with unlimited calling anywhere in the us? |
| 11:24.22 | drray | but there is huge sweat equity to be achieved with asterisk and a sip/iaxy provider |
| 11:25.13 | drray | I get all my minutes from my PRI, so I don't, but there are tons of them on the wiki |
| 11:25.46 | Gimp- | k. ill check it out |
| 11:26.33 | drray | the neat thing is you can pick 2 providers, that way if one is sucking you can fall over to the other one |
| 11:27.17 | Gimp- | im a little short on money atm |
| 11:29.17 | drray | there were sip outbound providers you could hook up with for a buck or so via paypal, at .02 a minute that will get you 30 minutes |
| 11:29.47 | drray | I do not know if that is still the case |
| 11:29.52 | drray | but I can't see how it would not be |
| 11:29.54 | tparcina | i have newer user bash script's so i need to ask you is it possible. i need to connect to ftp server, enter username and password, upload a file, and exit from ftp program. can i doo all that with bash script or i need to use something else? |
| 11:30.24 | tparcina | the problem might be because i don't know how bash script executes when another program (ftp) starts |
| 11:30.30 | tzanger | bash and expect may be able to do it |
| 11:30.44 | tzanger | or use something like ncftp where you can specify login and even commands on the command line |
| 11:30.57 | drray | or wput |
| 11:31.05 | tparcina | what is wput? |
| 11:31.20 | tzanger | opposite of wget |
| 11:31.25 | drray | the opposite of wget http://en.wikipedia.org/wiki/Wput |
| 11:31.45 | qdk | tparcina: ncftpput is in the ncftp package. |
| 11:31.53 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 11:32.02 | drray | just keep in mind that anyone who can read your script can pwn your ftp server |
| 11:32.10 | tparcina | ok, thank you bouth. now i need to do some readings |
| 11:35.35 | *** join/#asterisk P-NuT (n=P-Nut@CPE-60-227-84-159.nsw.bigpond.net.au) |
| 11:35.50 | *** join/#asterisk McLazarus (n=mcallist@pool-72-78-136-117.phlapa.east.verizon.net) |
| 11:36.30 | Gimp- | http://www.digitnetworks.com/store/product_info.php?cPath=25&products_id=31 |
| 11:36.50 | Gimp- | would that work for me? |
| 11:37.03 | *** join/#asterisk P-NuT (n=P-Nut@CPE-60-227-84-159.nsw.bigpond.net.au) |
| 11:40.50 | drray | I'd get an iaxy, personally |
| 11:41.41 | drray | but yeah, that would work |
| 11:42.08 | Gimp- | i just need something cheap that will still get the job done |
| 11:42.51 | Gimp- | btw.. thanks for all your help :) |
| 11:42.56 | drray | sure |
| 11:43.02 | drray | help someone else later |
| 11:43.17 | Gimp- | always do when im able :) |
| 11:44.08 | Gimp- | hmm.. one more thing. i read something about a windows version of asterisk. does that actually exist? |
| 11:45.03 | drray | it does, I'd do yate before I did it though |
| 11:45.22 | drray | asterisk with fedora core 4 is easy |
| 11:46.08 | drray | you just need disk one of fedora |
| 11:46.19 | drray | not all 4 or however many |
| 11:47.00 | drray | I'd run asterisk off a live cd before I ran it under windows |
| 11:47.06 | Gimp- | the only reason i ask is because id rather not spend the money on a hard drive for my old pc atm |
| 11:47.48 | Gimp- | and this one is on 24/7 and i only restart it once every 3-4 weeks and dont notice much performance loss |
| 11:48.01 | drray | has openwrt gotten the fxs ports working? |
| 11:48.07 | *** join/#asterisk aze (n=aze@ACayenne-101-1-3-12.w81-248.abo.wanadoo.fr) |
| 11:48.17 | Gimp- | but im guessing the windows version has lots of problems associated with it like most windows apps |
| 11:48.26 | Curus | Someone made a Linux-on-Windows image with asterisk on it |
| 11:48.43 | Curus | Astwind |
| 11:48.54 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 11:49.22 | Curus | It's from 2004 though, so a bit useless I guess |
| 11:50.33 | Curus | www.asteriskwin32.com looks better, but they're on 10.0.0.10 |
| 11:50.46 | tzanger | whee |
| 11:51.23 | tzanger | drray: which WRT has FXS ports? Or are you talking about the 'P' series I think it is? |
| 11:51.39 | drray | the pap ones |
| 11:51.59 | *** join/#asterisk coppice (n=chatzill@44.199.17.210.dyn.pacific.net.hk) |
| 11:52.05 | luke-jr_ | drray: WRTP54G is just recently supported for the CPU |
| 11:52.22 | drray | and the fxs ports? |
| 11:52.28 | luke-jr_ | drray: I went in the channel for it, and I couldn't get any straight answers |
| 11:52.36 | luke-jr_ | for example, they deny existence of any SIP daemon |
| 11:52.53 | luke-jr_ | From my limited research, the FXS ports use binary drivers |
| 11:53.02 | Curus | #openwrt is the only channel on Freenode I've managed to get banned on |
| 11:53.12 | drray | so no wrapper either? |
| 11:53.13 | luke-jr_ | Curus: try #wrtp54g |
| 11:53.19 | luke-jr_ | drray: wrapper? |
| 11:53.40 | drray | you can wrap binaries, I've never done it |
| 11:53.52 | luke-jr_ | what do you mean by wrap it? |
| 11:53.57 | drray | the same way you can use a windows binary in linux |
| 11:53.59 | drray | for drivers |
| 11:54.07 | luke-jr_ | shrug |
| 11:54.10 | luke-jr_ | I'm not sure I care much |
| 11:54.12 | luke-jr_ | either way, it's illegal |
| 11:54.27 | luke-jr_ | I sure would like to get a WRTP54G running Asterisk tho |
| 11:54.36 | drray | How is it illegal? |
| 11:54.50 | luke-jr_ | the GPL does not permit binary-only licenses to link |
| 11:55.02 | Curus | You can violate the GPL as mcuh as you want as long as you don't distribute the result |
| 11:55.09 | Curus | GPL respects your privacy |
| 11:55.14 | luke-jr_ | Curus: Linksys obviously distributes the result |
| 11:55.14 | drray | personal use is key |
| 11:55.32 | Curus | Commercial use is fine too, as long as you don't give it to someone else |
| 11:56.04 | drray | right now I am using an openwrt router and an iaxy to drive a payphone at a location |
| 11:56.15 | Curus | Build a new google based on gazillions of WRT's with binary drivers in a mesh network or something |
| 11:56.15 | drray | I'd love to just have the openwrt box there instead |
| 11:56.38 | luke-jr_ | Linksys distributes binary modules linking to Linux-- clearly and blatantly a GPL violation |
| 11:57.41 | tparcina | drray, fzanger, i have used wput, it works fine. it's possible securty hole, so you can't use it everywhere, but it works |
| 11:58.00 | tparcina | thank you guys |
| 11:58.11 | drray | you're welcome |
| 11:58.30 | drray | it's 100% a security hole |
| 11:58.32 | drray | :) |
| 11:58.53 | drray | so protect your ftp server accordingly |
| 11:58.55 | luke-jr_ | aboslutely no potential for a so-called 'grey area' |
| 11:59.45 | drray | risk threshold |
| 12:00.20 | tparcina | yes, i will protect my ftp |
| 12:00.44 | drray | I'd guess just making your ftp user, write only |
| 12:00.54 | drray | and then copying it out once it shows up |
| 12:01.14 | drray | and non root, non sudo |
| 12:02.02 | CrlMan | hi people... i new to asterisk universe... googling form fxo, more preciselly, wcfxo module, i found some interest things... my trouble is make it loads any softmodem as fxo... i have one wich is that sl1900 chip... after serious crashs here i found the exact place where it hangs in source code (in wcfxo_initialize(wc) under last outb before clear interrupts)... can someone help-me?? |
| 12:02.05 | drray | even then watch it like a hawk |
| 12:02.31 | drray | CrlMan, are you sharing an IRQ? |
| 12:02.52 | *** join/#asterisk esculapio__ (n=ESCulapi@200.88.44.66) |
| 12:03.06 | CrlMan | no... its alone... do you know if it will work with this smartlink chip? |
| 12:03.33 | drray | I do not know, I used a digium provide x100p card |
| 12:03.40 | CrlMan | <PROTECTED> |
| 12:03.40 | CrlMan | , wcfxo |
| 12:04.02 | CrlMan | what is this tregion? do u know? |
| 12:04.19 | *** join/#asterisk groogs (n=greg@d38-54-164.commercial1.cgocable.net) |
| 12:05.18 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 12:05.41 | CrlMan | drray, do u understand that card initialization performed by wcfxo_initialize (that outbs)? |
| 12:06.08 | drray | CrlMan I hear what you are saying but that is outside my skillset |
| 12:06.19 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 12:06.33 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.173) |
| 12:06.37 | drray | is it possible that tregions is part of pciregions? |
| 12:06.46 | drray | http://www.xml.com/ldd/chapter/book/ch15.html |
| 12:07.26 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 12:09.20 | *** join/#asterisk znoG (n=gs@205-17-235-201.fibertel.com.ar) |
| 12:10.31 | CrlMan | i reading that one... my system hang when it does the the outb of readdma, the end of readdma addr... |
| 12:11.32 | drray | that leads me to believe that you have a resource conflict, or the driver is not talking to the chipset properlly(it's not compaitble) |
| 12:11.45 | drray | but that is my uneducated opinion |
| 12:12.10 | *** join/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
| 12:14.37 | coppice | CrlMan: have you been modifying the wcfxo driver, because it doesn't normally work with an sl1900 |
| 12:15.29 | CrlMan | yes. |
| 12:15.35 | trixter | coppice: does it normally work anywhere? :P |
| 12:16.02 | CrlMan | no big mods... only add that pci ids... |
| 12:16.10 | coppice | it works with the one intel chip it was developed for. it works very well too, unless you have a bad card |
| 12:16.42 | coppice | CrlMan: only adding PCI IDs == good way to crash your machine, bad way to achieve anything useful |
| 12:16.52 | *** join/#asterisk aze_ (n=aze@ACayenne-101-1-7-153.w81-248.abo.wanadoo.fr) |
| 12:18.00 | CrlMan | coppice, ... should be... i was believing that chips are a standard... |
| 12:19.33 | CrlMan | the intel 537pg-pu works like x1000 right? .... why? |
| 12:21.13 | drray | you understand that you are spending time troubleshooting something you could buy for $9 |
| 12:21.59 | *** join/#asterisk oej (n=oej@213.115.215.54) |
| 12:22.35 | CrlMan | i'm learning... not only to make it works... but to understand all process involved... |
| 12:22.43 | drray | :) |
| 12:22.56 | CrlMan | :) |
| 12:23.25 | drray | I'm not saying you should not do it, it is your time to do with as you see fit. |
| 12:23.30 | CrlMan | when it works if dont like it will be stay here like ever: stopped! :D |
| 12:23.44 | CrlMan | ok... |
| 12:24.57 | CrlMan | i'm just tring to make a more reliable code... and improve some tech to reach a standard. |
| 12:26.26 | tparcina | has anybody used cisco call manager expres - ccme? i need to conect it with asterisk. and i have, and it works just fine, but only wht g711 codec :)) so, i would like to use gsm codec. does anybody have any clue on this one? |
| 12:31.06 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:31.52 | tparcina | hi fender |
| 12:33.34 | [TK]D-Fender | y0 |
| 12:33.41 | *** join/#asterisk X-Rob (n=rob@CPE-60-231-85-52.qld.bigpond.net.au) |
| 12:35.39 | *** join/#asterisk pigpen2 (n=mark@clinlab.com) |
| 12:36.27 | *** join/#asterisk mrtwister (n=manopulu@107.250.broadband5.iol.cz) |
| 12:36.58 | tparcina | Fender, are you from california (or i have change you with d-lynes :))? |
| 12:38.13 | *** join/#asterisk tainted_ (n=point@adsl-71-129-39-158.dsl.irvnca.pacbell.net) |
| 12:43.14 | coppice | CrlMan: an intel 537 doesn't work *like* an x100p. it *is* the chip on an x100p card |
| 12:43.42 | coppice | CrlMan: there is no standard for modem cards. There was one, called AMR, but it died |
| 12:46.31 | *** join/#asterisk toooms (n=tooms@203.57.129.6) |
| 12:46.32 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:47.41 | toooms | hi, im getting really bad voice quality using a cisco 2621 with a pri in it and terminating the calls on an asterisk machine, basically just using moh and listening to it if you make noise down the line you hear the other side cut out any ideas as to what this could be? |
| 12:48.32 | *** join/#asterisk Lino` (n=Lino@i577BC252.versanet.de) |
| 12:50.38 | tparcina | tooms, question - you have conected asterisk and ccme? |
| 12:52.04 | toooms | tparcina; dont think so, im not sure what ccme is |
| 12:53.04 | trelane_ | ~centos |
| 12:53.10 | jbot | i guess centos is better than Fedora Core except for that silly bug, see ~centosbug for details |
| 12:53.10 | *** join/#asterisk morex (i=morex@host86-137-21-78.range86-137.btcentralplus.com) |
| 12:53.11 | trelane_ | infobot? |
| 12:53.13 | morex | Hey there |
| 12:53.17 | trelane_ | ~centosbug |
| 12:53.18 | jbot | somebody said centosbug was a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 12:53.28 | morex | Does anybody know of a way of bridging two existing calls? |
| 12:53.39 | morex | I've been using MeetMe, but this won't allow SIP reinvites... |
| 12:53.45 | tparcina | tooms, ccme = cisco call maganer express |
| 12:54.07 | *** join/#asterisk BertZ (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 12:54.10 | BertZ | hi again |
| 12:54.15 | BertZ | I found something strange |
| 12:54.30 | tparcina | BertZ, no you didn't |
| 12:54.40 | tparcina | :) |
| 12:54.42 | BertZ | :p |
| 12:54.44 | toooms | tparcina; nope .. i've just got a pri connected to my cisco, my asterisk machine plugs directly into the cisco, the cisco talks sip to asterisk |
| 12:54.57 | tparcina | ok, that's it |
| 12:55.13 | tparcina | tooms, can you tell me what codec do you use to connect two of them? |
| 12:55.38 | toooms | tparcina; g711alaw and g711ulaw seem to be the only 2 i can get working |
| 12:55.54 | tparcina | tooms, yes that's my problem |
| 12:56.32 | BertZ | well the way is when I call someone, he answer, but he don't hear me. If I put the call on hold, then off hold, all is okay, called party can hear me |
| 12:56.50 | BertZ | oops sorry for my english ... |
| 12:57.11 | BertZ | is a softphone issue ? |
| 12:57.19 | BertZ | never had this kind of issues before ... |
| 12:57.22 | tparcina | toooms, have you tried g729 codec? |
| 12:57.43 | toooms | tparcina; yerp |
| 12:57.50 | toooms | tparcina; my asterisk machine rejects the call setup claiming it cant support g729 |
| 12:58.07 | trelane_ | how the feck do you typo rw_lock in spinlock.h and not notice? |
| 12:58.47 | tparcina | toooms, what doesn't work for you on cisco, maybe my colegue can help you |
| 13:00.17 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 13:00.21 | [TK]D-Fender | coppice : Don't forget CMR also ;) (though I'm sure we'd like to) |
| 13:00.50 | coppice | that was the expanded version of AMR, wasn't it? |
| 13:00.53 | [TK]D-Fender | tparcina : I'm from Quebec, and dlynes is from Brittish Columbia, so wrong on both counts :) |
| 13:01.09 | coppice | AMR was a good idea technically, but a commercially naive idea |
| 13:01.26 | [TK]D-Fender | coppice : Different function set. AMR had "audio" |
| 13:01.43 | [TK]D-Fender | VERY naive...and cheesy as hell. |
| 13:02.03 | *** part/#asterisk morex (i=morex@host86-137-21-78.range86-137.btcentralplus.com) |
| 13:02.05 | coppice | AMR linked AC97 with an FXO socket. I think CMR did that plus a few more things |
| 13:02.13 | coppice | why cheesy? |
| 13:02.20 | [TK]D-Fender | toooms : You paid for an activation code for G729 licenses? |
| 13:02.32 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.239.195.Dial1.SanJose1.Level3.net) |
| 13:02.37 | toooms | [TK]D-Fender; nope |
| 13:03.07 | Katty | hi lads. |
| 13:03.19 | [TK]D-Fender | coppice : Puny ass little shit to circumvent having even a cheap PCI modem. Create a slot to save 10$. Total waste. Actually considering the volume sold it wasn't any more cost effective to the user and still wasted a slot... |
| 13:03.27 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:03.31 | [TK]D-Fender | toooms : Well then no G729 for you. Its not a free codec. |
| 13:04.02 | toooms | [TK]D-Fender; cheers.. thats ok .. im happy with g711u/alaw |
| 13:04.37 | [TK]D-Fender | toooms : Ok, but that was the reason. You should use G711 if you can afford the bandwidth. |
| 13:05.10 | coppice | D-Fender: that was not its purpose. its purpose was to keep the approved parts on a separate card, so approvable cards could be made for each country. snag was it made the card so low value, nobody had the incentive to go through approvals with it |
| 13:05.12 | toooms | [TK]D-Fender; my issue is im getting crap voice quality between a cisco and the asterisk machine |
| 13:06.13 | [TK]D-Fender | coppice : Ok, thats the backstory I missed... I think the best description of my attitude towards it was "dismissal" |
| 13:06.34 | [TK]D-Fender | toooms : Even on g711? its a local connection right? Not over the net? |
| 13:07.17 | toooms | [TK]D-Fender; dialing from a bog standard pstn line, via the isdn into the cisco and through to the asterisk machine |
| 13:08.15 | coppice | D-Fender: a little while later the approvals in some key countries loosened up, making the socket in many laptops approvable in most majors markets with some software tweak, and kicking off the Zapata 1 work :-) |
| 13:08.22 | [TK]D-Fender | toooms : So G711 is from Cisco > * all local LAN then? No reason for poor quality on the codec's part... |
| 13:08.28 | *** join/#asterisk af_ (n=af@ip-164-240.sn2.eutelia.it) |
| 13:08.50 | [TK]D-Fender | coppice : Ok, the last one buys it some forgiveness then :) |
| 13:09.05 | toooms | [TK]D-Fender; correct.. i dont think it's the codec either |
| 13:09.16 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 13:09.36 | coppice | i think its those gremlins. maybe steven spielberg can help |
| 13:09.37 | [TK]D-Fender | toooms : What telco? BT? |
| 13:09.38 | toooms | [TK]D-Fender; dialling over the lan from a sip handset into the asterisk machine is fine, it's just calls that come off the pstn via a pri into the cisco that are bad quality |
| 13:10.13 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 13:10.23 | [TK]D-Fender | toooms : I was about to blame your "bog standard" PSTN line, but thats not even YOUR line, thats just one you're calling in THROUGH for testing isn't it? |
| 13:10.25 | Bert- | hmm connction issues ... |
| 13:10.40 | Bert- | well is a way to completely desactivate hold option on asterisk plz ? |
| 13:11.04 | Bert- | or is it a x-lite known bug ? |
| 13:11.41 | toooms | [TK]D-Fender; yerp |
| 13:12.22 | toooms | [TK]D-Fender; even just dialling from my mobile phone is giving me the same results |
| 13:12.40 | *** join/#asterisk jpeeler (n=jpeeler4@host81-154-150-237.range81-154.btcentralplus.com) |
| 13:14.07 | [TK]D-Fender | toooms : So its the PRI that crap... hmmm |
| 13:15.04 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 13:15.42 | *** join/#asterisk Icheb (n=icheb@sebsoft.xs4all.nl) |
| 13:16.42 | *** join/#asterisk acehunky (n=chat_jok@221.128.139.22) |
| 13:16.45 | acehunky | hello .. |
| 13:16.49 | toooms | [TK]D-Fender; whats going wrong is that I can hear the audio from the remote end of the call, but when i make a noise from my end you can hear the remote end get cut out of the audio |
| 13:16.59 | acehunky | i want to know .. how can i download the latest unstable asterisk code ? |
| 13:17.02 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 13:18.14 | *** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com) |
| 13:18.30 | tparcina | tooms, have you connected sccp softphone (or hardphone) to cisco and then called sip phone on asterisk? |
| 13:19.01 | toooms | tparcina; dont think i can with sip |
| 13:19.17 | Gimp- | does anyone know if its possible to run two operating systems at once and what i would use to do so? |
| 13:19.33 | tparcina | tooms, why you woudn't be able to do that? |
| 13:19.52 | [TK]D-Fender | toooms : That sounds like a really crappy attempt at echo cancellation... |
| 13:20.17 | [TK]D-Fender | acehunky : Pick any release you want! ;) |
| 13:20.50 | toooms | [TK]D-Fender; i've disabled all of that in my cisco |
| 13:20.51 | [TK]D-Fender | acehunky : On a more serious note, SVN download instruction are on the Wiki, and asterisk.org |
| 13:21.03 | [TK]D-Fender | toooms : Dunno... something is doing it... |
| 13:21.04 | tparcina | i think we head that problem, i just need to know are calls good when he calls from phone (that is connected to cisco) to phone (that is conected to *) |
| 13:21.28 | *** join/#asterisk bkervaski (n=bkervask@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 13:22.07 | bkervaski | Hi all. I'm trying to debug spandsp. I have the |debug option in my extensions.conf, but there is no output in the asterisk cli. I've issued the debug level 6 command inside the cli, no luck. Any help would be greatly appreciated. Thanks! |
| 13:22.34 | toooms | [TK]D-Fender; does asterisk include any sort of support for echo cancellation? |
| 13:25.08 | jpeeler | hey guys does Primary D-Channel on span x down represent a misconfiguration or a line issue/ |
| 13:26.14 | acehunky | i wanna know more on Asterisk 1.4 .. thats the reason i was asking of svn checkout .. does 1.4 exist in any form ? |
| 13:27.55 | nextime | when a caller join a queue, if the caller talk with an agent and the agent hangup, is there a way to continue in the dialplan? It seem that the agent hangup cause the hangup even for the caller channel |
| 13:28.33 | brad_mssw | toooms: http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation |
| 13:29.49 | toooms | brad_mssw; thanks |
| 13:30.32 | X-Rob | GOOOAL |
| 13:32.19 | bkervaski | Hi all. I'm trying to debug spandsp. I have the |debug option in my extensions.conf, but there is no output in the asterisk cli. I've issued the debug level 6 command inside the cli, no luck. Any help would be greatly appreciated. Thanks! |
| 13:32.24 | [TK]D-Fender | toooms : not on VoIP channels |
| 13:33.01 | [TK]D-Fender | acehunky : No, we're at 1.2 and SVN is 1.3 waiting to become 1.4 final |
| 13:33.23 | [TK]D-Fender | bkervaski : Please don't spam messages like that, we heard you the first time and its still on screen... |
| 13:33.47 | *** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 13:35.49 | bkervaski | It's not on my screen ?? That's a pretty short sighted comment.. everybody has different screen resolutions, font sizes, etc. |
| 13:36.35 | *** join/#asterisk aze (n=aze@ACayenne-101-1-8-138.w81-248.abo.wanadoo.fr) |
| 13:38.37 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 13:39.19 | *** join/#asterisk jrprado (n=jrprado@200.138.117.119) |
| 13:40.22 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 13:46.30 | *** join/#asterisk af_ (n=af@ip-164-240.sn2.eutelia.it) |
| 13:47.00 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 13:47.34 | feld_ | how bad is it going to be for a client to test my asterisk to PSTN from china? I'm in the USA. |
| 13:47.55 | feld_ | anything he should expect? is the extra latency going to destroy his call or add some unwanted echo or what? |
| 13:48.06 | jpeeler | Does Primary D-Channel on span x down represent a misconfiguration or a line issue? |
| 13:48.28 | [TK]D-Fender | feld : Should be survivable |
| 13:48.35 | tzanger | jpeeler: could be anything |
| 13:48.43 | tparcina | feld, everything depends on you-him internet connection. it could be great, it could be useless |
| 13:49.11 | coppice | most VoIP gets blocked in and out of china |
| 13:49.17 | [TK]D-Fender | jpeeler : Probably you card failed to synch. kill *, and restart it again a few times if it doesn't come up then its time to start debugging. Try a loopback on your card and see if it goes green. |
| 13:49.28 | feld_ | this could seal the deal if it works. I'm afraid it is going to be a disaster due to shitty connections to China anyway lol |
| 13:49.45 | mrtwister | question about asterisk realtime. after creation of new peer in database, should i do asterisk reload or it automatically will be updated? |
| 13:50.07 | mrtwister | correction - i talk about sip friend/peer/user |
| 13:50.14 | tparcina | coppice, you are joking, right? |
| 13:50.15 | jpeeler | [TK]D-Fender, I don't know how to do a loopback, but I'll try restarting * a few times |
| 13:50.32 | coppice | tpcarina: joking about what? |
| 13:50.43 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.173) |
| 13:50.49 | tparcina | about calls get blocket in and out of china |
| 13:51.26 | feld_ | China's firewall is going to screw us over isnt it |
| 13:51.59 | coppice | nope. broadband within china is generally very fast. internationally the speed is a bit variable - can be great, can be slow. they block SIP, H.323 and RTP traffic, though |
| 13:52.26 | coppice | people in china tend to be looking for non-standard VoIP protocols |
| 13:52.26 | jpeeler | [TK]D-Fender, how would I go about doing a loopback test? |
| 13:52.48 | feld_ | yeah but China's whole internet is controlled. that's what I was worrying about. |
| 13:53.46 | coppice | in southern china ADSL is wonderful. 100% of the time it runs at full speed. you never see congestion :-) |
| 13:55.46 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.173) |
| 13:55.59 | [TK]D-Fender | jpeeler : Go read up on how to make a loopback dongle. You'll need an RJ45 head and a small piece of Cat5 to strip for 4 wires. |
| 13:56.50 | jpeeler | [TK]D-Fender, is this just a test to see if the card is good? The card was just working and I have a feeling it has more to do with the kernel upgrade I just did. |
| 13:57.13 | jpeeler | [TK]D-Fender, at least if the line isn't faulty... |
| 13:57.16 | coppice | I think E1 cards should be supplied with a loopback lead. for the 50 cents it costs, the reduced support problems make it a bargain |
| 13:57.16 | [TK]D-Fender | jpeeler : Tests the card and its signalling and readiness |
| 13:57.18 | *** join/#asterisk P-NuT (n=P-Nut@CPE-60-227-93-75.nsw.bigpond.net.au) |
| 13:57.28 | *** join/#asterisk kpettit (n=keith@adsl-70-241-74-116.dsl.hstntx.swbell.net) |
| 13:57.40 | [TK]D-Fender | coppice : Mass produces I'm sure its more like .05$ tops ;) |
| 13:57.45 | kpettit | had a queue question. Is 7777 some sort of system extension? |
| 13:58.14 | DrkShdw | 7777 replicates an inbound call to asterisk |
| 13:58.16 | kpettit | I'm seeing all sorts of traffic go to a Sip/7777 but that dosen't exist in extensions.conf or sip.conf |
| 13:58.17 | [TK]D-Fender | kpettit : Nothin hard coded anywhere I've heard of... maybe you could elaborate... |
| 13:58.28 | coppice | D-Fender I was working on US rates. around here it would certainlu be cheaper :-) |
| 13:58.43 | kpettit | DrkShdw, what do you mean replicates? |
| 13:58.47 | [TK]D-Fender | kpettit : And queues.conf? Agents.conf? Maybe somone punched it into AgentCallbackLogin? |
| 13:58.57 | [TK]D-Fender | coppice : Fair enough. |
| 13:59.18 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 13:59.44 | kpettit | [TK]D-Fender, that's what I was qurious about. Maybe aagent fat fingering something |
| 13:59.55 | DrkShdw | I mean it simulates an incoming call on the trunk.. from an extension |
| 13:59.58 | P-NuT | hey guys.. |
| 14:00.05 | kpettit | so is that normal. |
| 14:00.38 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 14:00.54 | P-NuT | If I wanted to get the incoming zap/1 channel to call a group of extensions, where would I be looking to do that? |
| 14:01.09 | [TK]D-Fender | P-NuT : "show application dial" |
| 14:01.21 | DrkShdw | It's useful if for example you setup an IVR.. and all of a sudden it stops working for inbound calls. dial 7777 from an extension, and see if IVR is working. |
| 14:01.44 | feld_ | P-NuT: extensions.conf |
| 14:01.53 | P-NuT | [TK]D-Fender: thanks. |
| 14:02.23 | kpettit | [TK]D-Fender, Wouldn't the agent login error out if they put in a exten that dosen't exist? not to mention a sip peer that dosent exist |
| 14:02.27 | feld_ | mkay off to a job site. |
| 14:02.46 | kpettit | DrkShdw, So will it show that for all the queue calls, or is there some special circumstand that casues that? |
| 14:04.06 | [TK]D-Fender | kpettit : It doesn't validate, and it depends if you use a catch-all like _x. to grab call-outs to reps... that could assume its given legit #'s. |
| 14:04.09 | DrkShdw | That question is kind of ambiguous.. what is it doing, or not doing? |
| 14:04.37 | [TK]D-Fender | kpettit : So that could cause an assumed valid dial like "SIP/${EXTEN}) to show up. |
| 14:05.14 | [TK]D-Fender | kpettit : Look before the dial to seew what agent it was calling. |
| 14:05.31 | [TK]D-Fender | kpettit : And how about some CLI output in a pastebin? ;) |
| 14:05.35 | [TK]D-Fender | ~pb |
| 14:05.36 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 14:05.42 | P-NuT | So, is it cheating to get everything I want going on freepbx and then take the /etc/asterisk config files and work out asterisk without a gui from that? |
| 14:05.46 | Meaty | -- |
| 14:06.15 | Hmmhesays | bah I wish you could steal headers out of glibc and use them in uclibc |
| 14:06.18 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 14:07.01 | toooms | ~cisco |
| 14:07.02 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks! |
| 14:07.04 | *** join/#asterisk faberk (n=faberk@host104-228.pool80181.interbusiness.it) |
| 14:07.05 | coppice | doesn't uclibc already have a proper set of header? |
| 14:07.19 | [TK]D-Fender | P-NuT : What you're asking for is a SINGLE dial line in extensions.conf. You clearly aren't reading the instructions for it... |
| 14:09.03 | kpettit | [TK]D-Fender, I think you were right. Looks like agent was loggin in as 7777\ |
| 14:09.09 | kpettit | ughh, gotta love that. |
| 14:09.20 | *** join/#asterisk JackEStorm (n=thinkthi@ip68-225-72-125.no.no.cox.net) |
| 14:09.41 | *** part/#asterisk tparcina (n=tparcina@lns02-0309.dsl.iskon.hr) |
| 14:10.46 | *** join/#asterisk Meaty-Wrk (n=cp_simbu@office.abi.ca) |
| 14:10.47 | nextime | is there a way to continue in the dialplan after a caller exit from a queue ( not for timeout, but after get bridged with an agent that hangup the call ) |
| 14:10.57 | Hmmhesays | everything that could possibly go wrong this morning went wrong, no alarm clock, ran out of gas, left my wallet at home |
| 14:11.26 | *** join/#asterisk Iam8up|lpy (n=iam8up@cpe-24-210-253-66.woh.res.rr.com) |
| 14:11.57 | Iam8up|lpy | can anyone tell me where the voicemail waves are stored |
| 14:12.09 | mutilator | in the voicemail ocean ofcourse |
| 14:12.18 | Iam8up|lpy | huh? |
| 14:12.29 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.192.10) |
| 14:12.33 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 14:12.42 | mutilator | /var/spool/asterisk/voicemail/ |
| 14:12.50 | mutilator | try that |
| 14:12.54 | Iam8up|lpy | mutilator - thanks |
| 14:14.10 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 14:14.28 | P-NuT | [TK]D-Fender: k thanks mate. |
| 14:14.44 | *** join/#asterisk RoyK (n=roy@ti211210a080-0454.bb.online.no) |
| 14:17.18 | *** join/#asterisk QbY_ (n=Kelvin@cm-64-221-171-241.dhcp.southerncoastalcable.net) |
| 16:19.05 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 16:19.05 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.9.1 and 1.0.11.1 released with a critical security fix for chan_iax2, please upgrade immediately (June 6, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
| 16:19.32 | liran_ | brif8: dont know that cisco but maybe its missing it's software image so it's booting it up, waiting for an image to be transfered over xmodem or something and when it times-out it reboots. |
| 16:20.36 | brif8 | liran_: nope it was working just fine now it doesn't. no matter where I connect it it won't complete the boot cycle. no matter what I connect after it the other 7960s work fine. so it's definately phone related |
| 16:22.53 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 16:23.03 | generalhan | whats up all !!? |
| 16:23.36 | mountainm2k | nuffin |
| 16:25.16 | generalhan | slow in here today |
| 16:25.29 | generalhan | jbot, hows it going ?? |
| 16:25.42 | InHisName | Are dialplans different on diff mfgr ATA to get same results ? |
| 16:26.18 | InHisName | Telco AC211 seems diff dialplan from sipuras |
| 16:26.43 | charles___ | InHisName: dial plan ? |
| 16:27.03 | liran_ | brif8: sorry i cant help, dont know that cisco |
| 16:27.13 | generalhan | which cisco ? |
| 16:27.33 | charles___ | brif8: do you have an tftp image for that phone boot ? |
| 16:28.05 | generalhan | ~seen brif8 |
| 16:28.21 | jbot | brif8 is currently on #asterisk. Has said a total of 1 messages. Is idling for 7m 45s, last said: 'liran_: nope it was working just fine now it doesn't. no matter where I connect it it won't complete the boot cycle. no matter what I connect after it the other 7960s work fine. so it's definately phone related'. |
| 16:28.41 | *** join/#asterisk tRSS (n=tRSS@193.220.221.2) |
| 16:28.47 | brif8 | generalhan: cisco 7960 |
| 16:28.49 | *** part/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
| 16:28.52 | tRSS | hey everyone! :) |
| 16:29.11 | charles___ | brif8: how do you do the provisioning to the phone |
| 16:29.17 | generalhan | brif8: what is the issue i just got in .. i use 15 7960s in the office here |
| 16:29.22 | charles___ | brif8: each phone different mac, different config file |
| 16:29.23 | brif8 | charles__: yes I have an tftp image. as I said the phone was working fine then next day just does this. |
| 16:29.46 | tRSS | when I put someone on hold, I can see from the CLI that MoH starts but instantly ends, but I don't see any errors, although the caller is still on hold, except that there is no music? |
| 16:29.53 | charles___ | brif8: sniff your tftp and see what's going on |
| 16:30.33 | brif8 | charles__: yes each phone has it's own <mac_address>.cnf file and they all connect using DHCP to get an IP address and then tftp to get the image and then use SIP to connect with * sip.conf |
| 16:30.33 | charles___ | brif8: tcpdump -i eth1 ether dst 00:03:7B:DD:78:75 or ether src 00:03:7B:DD:78:75 -nnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnn -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv -X |
| 16:31.25 | charles___ | brif8: I've got issues of phones unable to update themselves or can't parse cnf files, watch the tftp, look for FILE NOT FOUND |
| 16:31.41 | brif8 | generalhan: I have a 7960 that suddently (starting yesterday : was working for weeks before that) does a constant power boot. powers up, shows configuring IP and then powers up again, and repeats this over and over ..... over.... |
| 16:31.52 | InHisName | charles___ I have |xxxxxx|xxxxxx| plus more but I cant get out to * beyond 3 digits. |
| 16:32.23 | generalhan | brif8: i had issues like that when the firmware version wasnt typed in correctly in sip.conf and OSX97.txt |
| 16:32.28 | charles___ | InHisName: digit timeout, what number are you dialing ? |
| 16:32.34 | brif8 | charles___: ok I'll check for File not found, I'm not even sure it's getting that far. it displays configuring IP for such a short time and then reboots |
| 16:33.02 | tRSS | someone help me with Music on Hold?? |
| 16:33.25 | brif8 | O79XX.TXT has |
| 16:33.26 | brif8 | P0S3-05-3-00 |
| 16:33.29 | charles___ | InHisName: try X.T |
| 16:33.51 | generalhan | and what about in the conf file for the phone ? |
| 16:33.55 | InHisName | FWD #'s are 5 & 6 digits plus prefix of "7" and a few 3 digt ones. |
| 16:34.05 | generalhan | i just use the sip.conf ... but i know people like to put them in the idividual files too |
| 16:34.49 | brif8 | generalham: it exists and is readable etc.. |
| 16:34.49 | charles___ | InHisName: didn't got that |
| 16:34.50 | InHisName | x = digit, .= one or more digits ? and what is T = ? |
| 16:34.56 | generalhan | s/sip.conf/SIPDefault.cnh |
| 16:35.02 | charles___ | InHisName: T timeout |
| 16:35.15 | charles___ | InHisName: will always wait for timeout or # |
| 16:35.51 | brif8 | SIPDefault has image_version: "P0S3-05-3-00" |
| 16:36.03 | charles___ | brif8: try getting new firmware |
| 16:36.05 | InHisName | I searched the admin setting s for a timeout value to change could not find any.. Will try the string anyway. |
| 16:36.07 | brif8 | as I say the other phones work just fine |
| 16:36.21 | charles___ | brif8: maybe firmware error, so if you update firmware will reflash |
| 16:36.24 | generalhan | i would definately upgrade anyway |
| 16:36.34 | generalhan | 5-3-00 is an OLD version |
| 16:36.54 | generalhan | im using 8-3-00 and most of the issues have been resolved already |
| 16:36.55 | *** join/#asterisk Laerte (n=bho@217.221.36.10) |
| 16:36.57 | charles___ | InHisName: different phones different settings, did you tried leaving blank ? it should work and wait for # |
| 16:37.22 | generalhan | they have a little MWI issue to fix but other than that, this is the best ver. ive had so far |
| 16:37.32 | charles___ | brif8: if it doesn't work after reflash, trow to the garbage |
| 16:37.36 | brif8 | let me check what files to does ask for on the tftp and then I'll know how far it gets. I'm not sure it's getting that far to even check about firmware versions |
| 16:37.57 | charles___ | brif8: it will get firmware as the first step |
| 16:38.03 | charles___ | brif8: DHCP and then FIRMWARE |
| 16:38.43 | InHisName | charles___ blank, eh? I'll try that next. |
| 16:39.04 | brif8 | ok I'll check brb |
| 16:39.07 | *** part/#asterisk brif8 (n=Administ@ttienterprises.org) |
| 16:39.48 | [TK]D-Fender | Funny Cisco firmware starts with P0S ... you'd think its a SIGN or something ;) |
| 16:39.55 | generalhan | yea |
| 16:40.02 | generalhan | i thought that a couple of upgrades ago ! lol |
| 16:42.36 | InHisName | charles___ thanks! (x.T) did it. Is there a reference for all the terms in dial plans ? |
| 16:43.26 | CunningPike | tRSS: What's the problem? |
| 16:43.52 | *** join/#asterisk angom_w (n=angom@red-corp-201.143.59.99.telnor.net) |
| 16:43.53 | *** part/#asterisk caribou7 (n=Jack@24-236-221-158.dhcp.aldl.mi.charter.com) |
| 16:45.11 | tRSS | <PROTECTED> |
| 16:45.28 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-20-4.w193-253.abo.wanadoo.fr) |
| 16:45.32 | jhiver | hi guys |
| 16:45.32 | tRSS | and I don't see any errors in the CLI althouth the verbosity is set to 10 or so |
| 16:45.38 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.239.195.Dial1.SanJose1.Level3.net) |
| 16:45.55 | jhiver | I need to interconnect an asterisk box with a softswitch which needs a h323 gatekeeper |
| 16:46.06 | jhiver | have you got any ideas on how to achieve this? |
| 16:46.14 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 16:46.54 | generalhan | tRSS: has it been working fine in the past or is this a new instal ? |
| 16:47.15 | [TK]D-Fender | tRSS : Pastebin your musiconhold.conf and veryify if there are indeed files in the stated folder in a format * can access. |
| 16:47.18 | charles___ | InHisName: that's not a dialplan |
| 16:47.24 | charles___ | InHisName: it's digittimeout |
| 16:47.43 | tRSS | [TK]D-Fender: give me a minute to pastebin |
| 16:47.43 | [TK]D-Fender | charles___ : It is a dialplan, just a rather simple one. |
| 16:48.06 | charles___ | jhiver: h323 on asterisk -> softswitch |
| 16:48.09 | tRSS | ~pastebin |
| 16:48.11 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 16:48.44 | charles___ | [TK]D-Fender: all that it manages is digits timeout, sending the call or waiting for #, okay I will call dialplan from now |
| 16:49.06 | generalhan | ~[TK]D-Fender |
| 16:49.16 | generalhan | aww |
| 16:50.23 | jhiver | charles___, asterisk is not a gatekeeper, is it? |
| 16:50.35 | generalhan | jbot: no |
| 16:50.37 | jbot | YES |
| 16:50.51 | [TK]D-Fender | charles___ : The fact that it will accept jsut about anything is irrelevent. Its a pattern. Just a "global" one :) I use that on my Polycom's for instance and just let * deal with everything. |
| 16:51.01 | charles___ | [TK]D-Fender: some equipment call it digit map |
| 16:51.01 | generalhan | jbot: [TK]D-Fender rocks the casbah !!! |
| 16:51.07 | [TK]D-Fender | :) |
| 16:51.10 | tRSS | [TK]D-Fender: here is the pastebin: http://pastebin.ca/66230 |
| 16:51.26 | generalhan | i cant get jbot to do anything |
| 16:51.42 | charles___ | [TK]D-Fender: what does TK means ? |
| 16:51.45 | [TK]D-Fender | tRSS : Go confirm that you have the proper verion of MPG123 runnig (only version 0.59r is acceptable) |
| 16:51.54 | [TK]D-Fender | charles___ : Old clan name.... |
| 16:51.54 | generalhan | jbot: [TK]D-Fender is rockin' the casbah !!! |
| 16:51.56 | jbot | generalhan: okay |
| 16:52.00 | tRSS | generalhan: we are moving from a desktop * machine to a server machine. I am configuring everything on it all over |
| 16:52.01 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:52.13 | generalhan | did you install mpg123 ? |
| 16:52.25 | [TK]D-Fender | generalhan : I was setting him up for that ;) |
| 16:52.28 | generalhan | lol |
| 16:52.31 | generalhan | sorry |
| 16:52.51 | tRSS | [TK]D-Fender: I don't have mpg123, but I thought with Asterisk 1.2.XX, mpg123 was not required anymore |
| 16:52.54 | [TK]D-Fender | generalhan : I like to give people a lot of rope. All they want in fact. Then I give it a good yank :D |
| 16:53.02 | generalhan | ~[TK]D-Fender |
| 16:53.04 | jbot | [[tk]d-fender] rockin' the casbah !!! |
| 16:53.12 | generalhan | ?? |
| 16:53.21 | [TK]D-Fender | tRSS : It isn't, but your setup TELLS it to use MPG123. You need to use "mode=files" for Native MoH |
| 16:53.48 | [TK]D-Fender | tRSS : And you need to have installed asterisk-addons if you want it to support MP3 format files. |
| 16:54.17 | generalhan | Well i gotta go find a way to crawl around in the ceiling to get a line from my TDM to the conference room for the new 2W i just ordered |
| 16:54.19 | tRSS | [TK]D-Fender: that explains everything. i have asterisk-addons installed. so what does default do? |
| 16:54.47 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
| 16:55.15 | tRSS | i mean mode=quietmp3, that is the default mode |
| 16:55.18 | [TK]D-Fender | tRSS : its not the context, its the MODE line. |
| 16:55.32 | charles___ | [TK]D-Fender: the knights ? |
| 16:55.38 | [TK]D-Fender | tRSS : That tells it to call MPG123 with a specifi decode style to lower the volume |
| 16:55.57 | [TK]D-Fender | charles___ : Nope... far off. |
| 16:56.05 | tRSS | Gotcha! [TK]D-Fender does rock the casbah. lol :) |
| 16:56.23 | generalhan | ~[TK]D-Fender |
| 16:56.25 | jbot | from memory, [tk]d-fender is rockin' the casbah !!! |
| 16:56.32 | generalhan | HAHA ! there we go !~ |
| 16:57.07 | iq | ~asterisk |
| 16:57.08 | jbot | asterisk is the best free PBX in the world. |
| 16:57.28 | generalhan | jbot: you are wrong |
| 16:57.30 | jbot | generalhan: what are you talking about? |
| 16:57.44 | generalhan | jbot: asterisk is the best PBX period |
| 16:57.45 | jbot | ...but asterisk is already something else... |
| 16:57.49 | tRSS | [TK]D-Fender: now it works like a charm. Thank you so much. btw, what is [TK]? ;) |
| 16:57.57 | Qwell[] | jbot: lock asterisk |
| 16:57.59 | Qwell[] | :D |
| 16:58.02 | generalhan | lol |
| 16:58.04 | a1fa | lol |
| 16:58.09 | a1fa | cock fight again |
| 16:58.11 | tRSS | jbot: asterisk is something else? |
| 16:58.13 | jbot | ...but asterisk is already something else... |
| 16:58.16 | a1fa | jbot: die and dont come back |
| 16:58.18 | jbot | ACTION takes two shots to the head, murders and dont come back then crumples to the ground, lifeless... |
| 16:58.27 | tRSS | lmao!!!! |
| 16:58.41 | tRSS | jbot: you know what? you are one funny bot. |
| 16:58.43 | jbot | tRSS: what are you talking about? |
| 16:59.03 | [TK]D-Fender | tRSS : My old Action:Half-Life clan |
| 16:59.04 | Qwell[] | ~lart tRSS |
| 16:59.13 | tRSS | jbot: i am talking about you |
| 16:59.14 | jbot | it is my pleasure to meet you, talking about you |
| 17:00.12 | tRSS | [TK]D-Fender: tRSS is my Counter-Strike handle. Now its been ages since I last played counter-strike |
| 17:01.39 | [TK]D-Fender | tRSS : I was an AHL God.... My clanmates and I spent a week alternating at #1 world-wide on GameSpy back in the day. |
| 17:01.42 | *** join/#asterisk surfdue (n=surfdue@unaffiliated/surfdue) |
| 17:02.38 | surfdue | I have Dial(SIP/200,30|m) how can I change this to dial my number? |
| 17:02.48 | surfdue | my PSNT number |
| 17:02.51 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 17:03.31 | tRSS | [TK]D-Fender: now those were the days ;) |
| 17:04.22 | surfdue | please anyone is it Dial(PSNT/1<number>,30|m) |
| 17:04.44 | [TK]D-Fender | surfdue : You really need to ge read the book.... |
| 17:04.46 | [TK]D-Fender | ~book |
| 17:04.47 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:05.00 | surfdue | tk this is an emergency I will read it I cant do it now! |
| 17:05.00 | surfdue | plz |
| 17:05.17 | surfdue | If I cant make my phone ring my cell I will loose all my domains |
| 17:05.20 | surfdue | :'( |
| 17:05.22 | Qwell[] | [TK]D-Fender: u help me be the next vonage? |
| 17:05.51 | [TK]D-Fender | surfdue : If you don't know how to use the PSTN throught whatever equipment you have there I doubt its even set up right to work if we DID show you a sample dialplan line... |
| 17:06.08 | [TK]D-Fender | Qwell : I'm sure you can feil for Chapter 11 all by yourself ;) |
| 17:06.09 | surfdue | it is tk |
| 17:06.12 | [TK]D-Fender | file* |
| 17:06.27 | surfdue | its setup to dial a sip line atm but the sip isnt working, so I need it to go to the cell |
| 17:06.30 | [TK]D-Fender | file : Not you.. typo correction :) |
| 17:06.43 | [TK]D-Fender | surfdue : Through what? |
| 17:06.44 | Qwell[] | file: * |
| 17:06.45 | *** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
| 17:07.00 | surfdue | [TK]D-Fender, extensions.conf |
| 17:08.19 | [TK]D-Fender | surfdue : Wrong answer. HOW do you intend to call your cell? |
| 17:08.39 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 17:09.07 | [TK]D-Fender | file : Bug tracker is looking pretty clean, is this the prep-work for 1.4? |
| 17:09.15 | Qwell[] | clean...ha |
| 17:09.38 | surfdue | [TK]D-Fender, through asterisk |
| 17:09.59 | file | [TK]D-Fender: clean... interesting word to describe it |
| 17:10.09 | [TK]D-Fender | surfdue : You are clearly not understanding. WHAT TECHNOLOGY ARE YOU GOING TO USE TO DIAL YOUR DAMNED PHONE?! |
| 17:10.21 | Qwell[] | [TK]D-Fender: teh intarweb! |
| 17:10.24 | surfdue | [TK]D-Fender, asterisk |
| 17:10.34 | surfdue | [TK]D-Fender, i dont have zap or anything setup |
| 17:10.44 | surfdue | [TK]D-Fender, I w ant to route the call to asterisk which will then call |
| 17:10.49 | [TK]D-Fender | surfdue : So what is going to allow you to dial your phone? |
| 17:11.03 | surfdue | [TK]D-Fender, im confused at your question? |
| 17:11.29 | [TK]D-Fender | surfdue : What HARDWARE allows you to call a phone? You have a SIP phone.. FINE. What is going to Allow you to call a PSTN number though? |
| 17:11.45 | surfdue | [TK]D-Fender, through asterlink my provider |
| 17:11.58 | surfdue | [TK]D-Fender, they have the hardware its routed through my asterisk |
| 17:12.05 | [TK]D-Fender | surfdue : Geez... that was a hard answer to beat out of you... |
| 17:12.20 | surfdue | [TK]D-Fender, sorry :noob: |
| 17:12.25 | surfdue | ;) |
| 17:12.49 | [TK]D-Fender | surfdue : You saying you don't know how to dial out using them? Sounds like you should already be set up for it... |
| 17:13.30 | surfdue | this is my current extensions.conf http://host41.pastebin.com/713336 |
| 17:13.59 | surfdue | [TK]D-Fender, I think when dialing out he set it to use sip by defualt so it wont understand what the PSNT is, can we change this and how? |
| 17:14.45 | file | SIP/yourcellphone@asterlink instead of SIP/200 |
| 17:14.51 | surfdue | ok |
| 17:14.53 | [TK]D-Fender | exten => _1NXXNXXXXXX,2,Dial(SIP/${EXTEN}@asterlink) |
| 17:14.55 | surfdue | thank you |
| 17:15.08 | [TK]D-Fender | surfdue : I think you are a litle blind if you don't know how to use your provider... |
| 17:15.15 | surfdue | i am |
| 17:15.16 | surfdue | :) |
| 17:15.22 | [TK]D-Fender | surfdue : Wake up and look at what you're doing... |
| 17:15.28 | surfdue | question does asterisk require a restart after config changes? |
| 17:15.45 | Qwell[] | CunningPike: That's advil |
| 17:16.06 | CunningPike | Qwell[]: Ah - that's the problem |
| 17:16.08 | file | crap I was washing my sheets, I wonder if they are done yet |
| 17:16.21 | Qwell[] | file: nope! |
| 17:16.37 | CunningPike | file: TMI |
| 17:16.38 | Qwell[] | CunningPike: ...that's a flintstones chewable vitamin! |
| 17:16.39 | CunningPike | ;) |
| 17:16.44 | CunningPike | Qwell[]: Crap |
| 17:16.48 | [TK]D-Fender | CunningPike : I want the blue pill Morpheus! |
| 17:17.09 | CunningPike | [TK]D-Fender: Viagra? |
| 17:17.25 | CunningPike | ~viagra |
| 17:17.30 | jbot | somebody said viagra was the nickname for the Woody Tech Support Crew |
| 17:17.39 | CunningPike | jbot cracks me up |
| 17:18.23 | file | into the dryer they go |
| 17:19.16 | InHisName | voicemail crashes * when it tries to play unavailable msg. |
| 17:19.38 | surfdue | file can you check your pm please |
| 17:19.46 | [TK]D-Fender | InHisName : Any, or a specific box only? |
| 17:19.48 | InHisName | CLI is dead and I found rebooting only gets it going again. |
| 17:20.03 | InHisName | 350mhz pentium |
| 17:20.29 | [TK]D-Fender | InHisName : I meant ANY "unavailable" message in your system, or one VM box's one specifically? |
| 17:20.36 | InHisName | never crashed before today. |
| 17:20.55 | [TK]D-Fender | InHisName : WRONG ANSWER. Please pay attention to the questions being asked. |
| 17:20.55 | InHisName | in sip show regisgtry / peers ? |
| 17:21.12 | [TK]D-Fender | InHisName : PAY ATTENTION TO THE QUESTION <-------------------- |
| 17:21.18 | generalhan | lol |
| 17:21.20 | file | [TK]D-Fender: wow, you should be a high school teacher |
| 17:21.25 | surfdue | [TK]D-Fender, is there any reason even with exten => 8006170141,1,Dial(SIP/14074967041@asterlink,30|m) |
| 17:21.26 | surfdue | <PROTECTED> |
| 17:21.41 | [TK]D-Fender | file : No.. I wouldn't be allowed to smack them around..... |
| 17:22.13 | generalhan | [TK]D-Fender: yea i dont think i could do it ... just wanting to smack 'em around and knowing that you cant |
| 17:22.16 | InHisName | I call 1021 from 1023 , rings until timeout and the voicemail is SUPPOSED to play u1021 messages. Crashes instead. |
| 17:22.18 | file | keep in mind that Asterlink does voicemail, so chances are the call isn't even hittng your box |
| 17:22.35 | [TK]D-Fender | surfdue : Did you do a SIP debug on it? What error messages does it tell you? Why is the sky blue? Who killed J.R.? |
| 17:22.42 | surfdue | file, I was thinking that but xachen said its not :| |
| 17:22.55 | generalhan | haha Dallas |
| 17:23.05 | file | do a sip debug like [TK]D-Fender said |
| 17:23.08 | surfdue | file, I think your right |
| 17:23.12 | *** join/#asterisk thock (n=thock@63.133.144.2) |
| 17:23.18 | generalhan | ~[TK]D-Fender |
| 17:23.20 | jbot | rumour has it, [tk]d-fender is rockin' the casbah !!! |
| 17:23.26 | InHisName | OK, wait 1/2 hour wile I crash it and see what sip debug grabs before it dies. |
| 17:25.00 | surfdue | file: http://host41.pastebin.com/713359 |
| 17:25.40 | InHisName | says scheduling voice mail for NOANSWER and less than 20 lines is dead. Posting the lot on postbin. |
| 17:26.13 | surfdue | file: can you reach asterlink.com ? |
| 17:26.33 | file | personally? no |
| 17:26.56 | [TK]D-Fender | SIP/2.0 401 Unauthorized |
| 17:27.24 | [TK]D-Fender | SIP/2.0 404 Not Found |
| 17:27.27 | surfdue | [TK]D-Fender, I noticed that but I get emails from asterlink when I leave a message |
| 17:27.38 | [TK]D-Fender | # |
| 17:27.38 | [TK]D-Fender | From: "4072930335" <sip:4072930335@66.250.69.15>;tag=as787b191c |
| 17:27.38 | [TK]D-Fender | # |
| 17:27.38 | [TK]D-Fender | To: <sip:8006170141@host41.com>;tag=as67bb4bec |
| 17:27.46 | [TK]D-Fender | Stupid PB filler lines... |
| 17:28.03 | surfdue | ya xachen setup asterlink with a dialplan not a normal register :| |
| 17:28.08 | [TK]D-Fender | surfdue : Go work out your config with them.... |
| 17:28.12 | surfdue | k |
| 17:28.27 | *** join/#asterisk EinsteinTaylor (n=dtaylor@216.243.100.29) |
| 17:28.29 | surfdue | [TK]D-Fender, can you tell me if you can reach asterlink.com I cant? |
| 17:29.08 | EinsteinTaylor | asterlink.com came up ok here |
| 17:29.25 | surfdue | darrn :( |
| 17:29.37 | EinsteinTaylor | here being new mexico...so that may or may not help u |
| 17:29.44 | surfdue | :P |
| 17:30.00 | [TK]D-Fender | surfdue : Have you tried PHONING them? |
| 17:30.11 | surfdue | there is no ping response |
| 17:30.14 | surfdue | I dont have the phone number |
| 17:30.17 | surfdue | its on that page |
| 17:30.29 | EinsteinTaylor | 1-877-2787565 |
| 17:30.30 | surfdue | [TK]D-Fender, do you have it? EinsteinTaylor can you give me the phone number? |
| 17:30.32 | surfdue | thank you |
| 17:30.34 | EinsteinTaylor | there ya go |
| 17:30.35 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 17:30.40 | *** join/#asterisk tanvalmg (n=mguevarr@67-134-234-194.dia.static.qwest.net) |
| 17:30.44 | EinsteinTaylor | np |
| 17:30.52 | InHisName | [TK]D-Fender I have postbined the debug out: http://pastebin.ca/66244 on the crashing voicemail. |
| 17:30.53 | surfdue | :) |
| 17:31.15 | EinsteinTaylor | any sip guru's in here got a few minutes? |
| 17:31.42 | dlynes_home | [TK]D-Fender: have you encountered a problem where if you do Wait(2), Answer(), Hangup() on the a200 cards that the remote end doesn't detect the hangup, and it keeps ringing the line over and over again? |
| 17:31.49 | file | EinsteinTaylor: ask. |
| 17:32.15 | tanvalmg | Hi, we tried to install a TDM400P (w/ 3 FXO) in an IBM Netfinity 4000R, but the card can not be detected at boot time. Any thoughts on what could be wrong? Thanks! |
| 17:32.32 | file | is it PCI 2.2? |
| 17:32.34 | mountainm2k | <PROTECTED> |
| 17:32.36 | [TK]D-Fender | InHisName : That wasn't at all what I asked... |
| 17:32.50 | EinsteinTaylor | i have two asterisk servers...one is production and has a ISDN pri card in it..i'm setting up a secondary asterisk server for testing different features without disrupting our customers voicemail...i want all incoming and outgoing calls to relay through my production server with the pri card in it |
| 17:32.51 | [TK]D-Fender | dlynes_home : Nope.... |
| 17:32.54 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:32.54 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:32.57 | tRSS | is there a way to find out the DID that was called to in asterisk, since calls are coming to another sip box, but get forwarded to asterisk box's queue? |
| 17:33.02 | *** join/#asterisk notjason (n=jordan@ool-457183bb.dyn.optonline.net) |
| 17:33.03 | EinsteinTaylor | i've been thinking maybe i need the IAX setup but i'm not getting it |
| 17:33.04 | tanvalmg | montainm2k: Yes, the zaptel modules are loaded wctdm, etc... |
| 17:33.08 | dlynes_home | [TK]D-Fender: ok, thanks |
| 17:33.32 | dlynes_home | [TK]D-Fender: i'll try it on the pri too (using a digium te405p) |
| 17:33.38 | dlynes_home | [TK]D-Fender: maybe it happens there, as well |
| 17:33.48 | [TK]D-Fender | dlynes_home : PRI = disconnect supervision all the time. |
| 17:33.56 | mountainm2k | <PROTECTED> |
| 17:33.57 | dlynes_home | ah |
| 17:34.07 | [TK]D-Fender | dlynes_home : An advantage of the tech |
| 17:34.09 | InHisName | [TK]D-Fender Since I haven't answered your question, then I need to see it repeated fresh. |
| 17:34.21 | tRSS | [TK]D-Fender: can you help me with my problem stated above. |
| 17:34.23 | dlynes_home | [TK]D-Fender: well, is there a reverse option for disconnect supervision? |
| 17:34.31 | tanvalmg | montainm2k: it shows ZT_CHANCONFIG failed |
| 17:34.34 | dlynes_home | [TK]D-Fender: i.e. force my end to reverse the lines on hangup? |
| 17:34.37 | [TK]D-Fender | InHisName : Its it only box 1234's message that it crashes on, or ALL messages? |
| 17:34.42 | dlynes_home | erm reverse polarity i mean? |
| 17:34.58 | file | tanvalmg: if you do dmesg does it show that the TDM board was detected? |
| 17:35.00 | [TK]D-Fender | dlynes_home : You don't tell the telco the call is over, its the reverse. |
| 17:35.04 | jpeeler | anybody know of any special configuration required for SIP based instant messaging? |
| 17:35.11 | file | jpeeler: we don't support it |
| 17:35.12 | [TK]D-Fender | dlynes_home : Make sure they do a cut or polarity reversal on it. |
| 17:35.24 | [TK]D-Fender | jpeeler : * does not support SIP IM's yet |
| 17:35.25 | dlynes_home | [TK]D-Fender: when I'm hanging up the call, not when the remote end hangs up the call |
| 17:35.27 | InHisName | box 1021 & 1023 & 1024 crash it so far. |
| 17:35.35 | tanvalmg | file: no it doesn't...even with lspci |
| 17:35.36 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 17:35.42 | [TK]D-Fender | dlynes_home : When you hang up, its should simply disconnect... |
| 17:35.52 | jpeeler | [TK]D-Fender, even though the message makes it to the console? |
| 17:35.54 | [TK]D-Fender | InHisName : Only the unavail? What about busy? |
| 17:35.55 | file | tanvalmg: then it sounds like your motherboard is not PCI 2.2 compliant, so the TDM board won't show up |
| 17:35.58 | dlynes_home | [TK]D-Fender: i suspect maybe asterisk or the sangoma driver isn't reversing the polarity for long enough |
| 17:36.09 | InHisName | I don't have a 1234 extension anymore. |
| 17:36.13 | [TK]D-Fender | jpeeler : it shows up in console as "FYI", its not passed on. |
| 17:36.23 | InHisName | I can triy the busy in a bit when I reboot. |
| 17:36.28 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 17:36.37 | [TK]D-Fender | InHisName : Try to isolate the problem.... |
| 17:36.46 | tanvalmg | file: That's what I thought too...would it really not show up if the motherboard is not PCI 2.2? |
| 17:36.55 | funxion | does anyone here use a2billing |
| 17:36.56 | file | tanvalmg: aye |
| 17:37.01 | vader-- | ok they brought up my T1 |
| 17:37.04 | vader-- | y |
| 17:37.04 | vader-- | a |
| 17:37.14 | vader-- | outbound calling working, inbound calling works |
| 17:37.31 | tRSS | i need some help, please. |
| 17:37.31 | file | bringing down T1s is much more fun |
| 17:37.31 | tRSS | :) |
| 17:37.45 | jpeeler | [TK]D-Fender, glad to know I wasn't doing anything wrong. Is instant messaging being planned or is it something asterisk is not concerned about? |
| 17:37.59 | EinsteinTaylor | file: was the clear enough? |
| 17:38.13 | [TK]D-Fender | jpeeler : I *think* Aterisk 1.4 may have some more support for it... |
| 17:38.26 | file | jpeeler: it's being planned |
| 17:38.28 | tanvalmg | file: thanks! Will the PCI version show in dmesg? |
| 17:38.41 | file | tanvalmg: dunno, best thing is to check your mobo manual |
| 17:38.57 | EinsteinTaylor | tanvalmg: it might...u'll be better off to pipe the dmesg to less...if it does show it it will probaby be in the top half |
| 17:39.53 | tanvalmg | file: thats my problem...lost all manuals..can not find it even in IBM site since it is discontinued already |
| 17:40.35 | TripleFFFF | <PROTECTED> |
| 17:40.35 | TripleFFFF | <PROTECTED> |
| 17:40.58 | tanvalmg | EinstenTaylor: What i see is "PCI BIOS Revision 2.10" ...not sure if that's it |
| 17:40.59 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:41.15 | TripleFFFF | actually i see - Playing 'revolution/nova_sales_service' (language 'fr') |
| 17:41.34 | paolob | Guys, hi all of you! What's this: "WARNING[13257]: chan_sip.c:1066 __sip_xmit: sip_xmit of 0x8141238 (len 483) to 67.55.159.160:5060 returned -1: Bad file descriptor" ? |
| 17:41.35 | TripleFFFF | so .. but its english !!!.. if i get the file and play it its french |
| 17:41.35 | [TK]D-Fender | TripleFFFF : it'd need to be in revolution/fr/nova_sales_service |
| 17:41.37 | TripleFFFF | what could be wrong |
| 17:41.42 | TripleFFFF | oh |
| 17:41.44 | TripleFFFF | ki |
| 17:41.47 | paolob | is it a hardware problem ? |
| 17:41.51 | TripleFFFF | well logger should tell me file not found |
| 17:41.56 | [TK]D-Fender | TripleFFFF : reveille-toi! |
| 17:42.07 | TripleFFFF | jessai mec |
| 17:42.21 | [TK]D-Fender | TripleFFFF : No, it TRIES to find it in french, and if it fails, THEN it plays the default WITHOUT any warning. |
| 17:42.33 | [TK]D-Fender | TripleFFFF : Its a "best efforts" based system. |
| 17:42.33 | tRSS | I have call come to another SIP (Zultys) box, that call will be forwarded to my asterisk box and my asterisk box is suppose to check the DID on which the call came to, and then assign it to a queue. how can I achieve this in my extensions.conf. I am not sure, what application (cmd) i have to use to check the DID that is forwarded to my *. |
| 17:42.50 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 17:43.04 | TripleFFFF | worked |
| 17:43.17 | TripleFFFF | merci fender |
| 17:43.20 | TripleFFFF | gracias |
| 17:43.22 | [TK]D-Fender | TripleFFFF : Using the new ones from June Wallack? |
| 17:43.24 | TripleFFFF | choung way |
| 17:43.31 | TripleFFFF | yeah |
| 17:43.36 | TripleFFFF | also made a custom ivr for a client |
| 17:43.41 | TripleFFFF | actually made2 last night |
| 17:43.55 | TripleFFFF | till i saw i wasnt billing on ivr's |
| 17:43.56 | InHisName | [TK]D-Fender: The busy message crashed it also for 1021 to 1023 while 1023 busy. |
| 17:44.00 | TripleFFFF | had to rewrite drtool cdrlib |
| 17:44.10 | [TK]D-Fender | TripleFFFF : cool... I'm still running on Prival's half-baked ones, and hear that Claude is due to have the new ones timing adjusted... |
| 17:44.25 | [TK]D-Fender | TripleFFFF : I'm told the inter-word timing is a little "jittery" |
| 17:44.41 | TripleFFFF | yeah on the voicemail |
| 17:44.42 | TripleFFFF | but hey |
| 17:44.58 | [TK]D-Fender | InHisName : try trashing the sound files and see if its the app thats to blame, and not the sound file. |
| 17:45.10 | TripleFFFF | i got frenchie clients that cant undestant basic englsh .. they get "same thing" at english mcdonalds.. and they wodner why its always a different menu |
| 17:45.19 | [TK]D-Fender | TripleFFFF : SAme here.... |
| 17:45.32 | TripleFFFF | tought id get a laugh of that one |
| 17:45.40 | InHisName | [TK]D-Fender: I started crashing when setup entries in voicemail.conf and later found my directions missed the "addmailbox" steps. I have done that now, but still crashes. |
| 17:46.03 | [TK]D-Fender | InHisName : Sounds like its time to scrap it all and compile fresh maybe... |
| 17:46.20 | InHisName | Hmmhesays, hold for a while I do that. |
| 17:46.40 | [TK]D-Fender | TripleFFFF : I've got to update mine sometime soon... the girl who did mine is a little high-pitched and she gets on my nerves ;) |
| 17:46.55 | *** join/#asterisk vechers (n=svecher@64.61.117.139) |
| 17:47.01 | TripleFFFF | oh |
| 17:47.02 | *** part/#asterisk vechers (n=svecher@64.61.117.139) |
| 17:47.03 | TripleFFFF | yeah |
| 17:47.05 | TripleFFFF | check these 2 |
| 17:47.07 | TripleFFFF | 888 411 9757 |
| 17:47.10 | TripleFFFF | that our lines |
| 17:47.18 | TripleFFFF | and then .. tell me how bad she is |
| 17:47.21 | TripleFFFF | shes from voice123 |
| 17:47.24 | TripleFFFF | took first one up |
| 17:47.25 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:47.26 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:47.31 | TripleFFFF | then try. a ivr i did last night |
| 17:47.39 | TripleFFFF | 8002324085 |
| 17:47.43 | TripleFFFF | thats alot better |
| 17:47.46 | [TK]D-Fender | TripleFFFF : Crap quality and starts only in english... |
| 17:47.48 | TripleFFFF | ill have her full time i think |
| 17:48.03 | Dandan | re a;; |
| 17:48.04 | Dandan | re all |
| 17:48.08 | TripleFFFF | yeah i had to trx to g729 for whatever reason ill remake in gsm 8k mno |
| 17:48.10 | Dandan | I am sick and tired of that PRI~ |
| 17:48.12 | TripleFFFF | 8002324085 |
| 17:48.30 | TripleFFFF | dand gibve it to me ill amnage it |
| 17:48.42 | Dandan | TripleFFFF: I can't get it to work :/ |
| 17:48.49 | TripleFFFF | hehe |
| 17:49.01 | TripleFFFF | they only in english btw |
| 17:49.04 | TripleFFFF | no time for frenchie on that |
| 17:49.14 | TripleFFFF | anfd my voice is.. lol first attempts for a client |
| 17:49.17 | TripleFFFF | 800 725 4309 |
| 17:49.34 | TripleFFFF | got a oood voice but did that in 5 minutes |
| 17:49.35 | Dandan | TripleFFFF: http://lists.digium.com/pipermail/asterisk-users/2006-June/155866.html |
| 17:50.33 | TripleFFFF | hehe you having fun ? |
| 17:50.39 | [TK]D-Fender | TripleFFFF : Good voice for an interrogator.... |
| 17:50.43 | TripleFFFF | yeah |
| 17:50.47 | TripleFFFF | im military |
| 17:50.49 | [TK]D-Fender | :D |
| 17:50.50 | TripleFFFF | used to be |
| 17:51.00 | TripleFFFF | so i get to the point.. PRESS 1 sodlier |
| 17:51.04 | [TK]D-Fender | And you need a jitter-buffer ;) |
| 17:51.08 | TripleFFFF | press 9 and do does pushups |
| 17:51.11 | TripleFFFF | im sip |
| 17:51.49 | TripleFFFF | actually that last one ..elwoh.. |
| 17:51.52 | TripleFFFF | was a bit messy |
| 17:51.59 | TripleFFFF | had to use 4 contetxts+ labels |
| 17:52.11 | TripleFFFF | since i had to use 1 for exten on differnet occasions and different subsub menus |
| 17:52.16 | TripleFFFF | all realtime |
| 17:52.25 | TripleFFFF | + all crap cdr crap |
| 17:52.25 | [TK]D-Fender | Realtime? overkill for most things... |
| 17:52.29 | TripleFFFF | nah |
| 17:52.30 | TripleFFFF | i need it |
| 17:52.35 | TripleFFFF | im building a new backend |
| 17:52.38 | TripleFFFF | you wont believe it |
| 17:52.51 | TripleFFFF | actually rewrote most of it |
| 17:52.57 | TripleFFFF | cdrtool is worst thing |
| 17:55.02 | funxion | does anyone here use a2billing |
| 17:56.21 | *** join/#asterisk camelon (n=chiardon@200.71.58.39) |
| 17:59.00 | paolob | guys, any help with this error: " WARNING[13257]: chan_sip.c:1066 __sip_xmit: sip_xmit of 0x8143da8 (len 471) to 83.138.130.145:5060 returned -1: Bad file descriptor" ? thank you? It comes out continuosly, and doesn't permit asterisk to end the reloading process |
| 17:59.07 | camelon | Hello! |
| 17:59.23 | TripleFFFF | hmm |
| 17:59.29 | TripleFFFF | permissions on filesystem it seems |
| 17:59.31 | Dandan | I NEED TO MAKE THIS ERROR GO AWAY! |
| 18:00.46 | [TK]D-Fender | Dandan : Why is it going through Cisco first? Have you tried with it in-line? |
| 18:00.54 | *** join/#asterisk asterboy (n=kevin@S01060204ee2b6007.ed.shawcable.net) |
| 18:03.33 | asterboy | show warrenty says I have no warranty! |
| 18:03.38 | asterboy | I want my money back! |
| 18:03.45 | Strom_C | maybe because you're misspelling warranty? |
| 18:03.49 | asterboy | doh |
| 18:04.06 | asterboy | well once any way |
| 18:04.30 | dlynes_office | paolob: which version of asterisk? |
| 18:04.39 | asterboy | Finally, have a TDM card that is actually working. |
| 18:04.46 | dlynes_office | asterboy: which one? |
| 18:04.55 | dlynes_office | x100p? |
| 18:05.09 | asterboy | That TDM400 which was giving me no end of grief |
| 18:05.15 | paolob | dlynes_home, 1.2.7.1 |
| 18:05.16 | dlynes_office | ah |
| 18:05.30 | asterboy | Had to stip the kernel just short of ...well...nothing |
| 18:05.55 | asterboy | Now zttest gives me above the 99.75%. |
| 18:06.06 | dlynes_office | wow...I guess the sip channel driver changed huge between 1.2.8 and 1.2.7.1 |
| 18:06.10 | asterboy | otherwise you get all kinds of intermittent problems |
| 18:06.56 | notjason | Can anyone answer some questions for a telecom and Unix veteran who's trying to get a better handle on *? |
| 18:07.13 | kaz0358 | go for it, not jason |
| 18:07.30 | asterboy | but your a veteran...you have already been in the trenches. |
| 18:07.34 | dlynes_office | paolob: I'm guessing someone slipped a bug into chan_sip.c, whereby they forgot to reinitialize a socket variable after they were finished with it |
| 18:07.48 | dlynes_office | paolob: Try upgrading to 1.2.9.1 to see if that bug gets fixed |
| 18:08.25 | paolob | dlynes_home, the weird thing is that it began to me today, and I it wasn't upgrading asterisk |
| 18:08.40 | notjason | * seems to be a pretty capable switch and voice server, but how does it do on PBX-type "feature phone" features? For example, busy lamp fields, parking, bridged appearances, off hook announce, whisper page, etc, etc.? |
| 18:08.44 | dlynes_office | paolob: yeah...the bug was probably there all along, but you only encountered it now |
| 18:09.18 | asterboy | ~recipies |
| 18:09.20 | paolob | dlynes_home, but why didn't it appear before...? I have been running asterisk for a week... |
| 18:09.22 | Qwell[] | file: time for an overly technical answer :p |
| 18:09.28 | paolob | without any problem |
| 18:09.29 | file | what, for what? |
| 18:09.50 | dlynes_office | Qwell[]: file wrote chan_sip? |
| 18:10.02 | Qwell[] | no, notjason asked how things worked ;) |
| 18:10.02 | file | no, I didn't |
| 18:10.46 | file | paolob: some of those are key system features which aren't implemented (bridged line appearances for example) and some are things that depend on the phone in order to be able to do it (off hook announce) |
| 18:11.06 | dlynes_office | paolob: some bugs only crop up when very particular circumstances exist |
| 18:11.07 | Qwell[] | file: wrong one :D |
| 18:11.13 | file | er notjason |
| 18:11.22 | asterboy | what the heck is the url of the asterisk recipes site? |
| 18:11.26 | asterboy | ~docs |
| 18:11.27 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 18:11.27 | dlynes_office | Qwell[]: give him a break...he hasn't had his bottle of Schooner yet this morning :) |
| 18:11.29 | dlynes_office | ~recipes |
| 18:11.30 | jbot | i hope you have some good ones ;), or but if you don't, go here for some good ones: http://astrecipes.net/ |
| 18:11.35 | Qwell[] | dlynes_office: He's had three! |
| 18:11.36 | asterboy | ah thanks! |
| 18:11.37 | notjason | When you say "key system features" aren't supported, is that a design issue, as in "Asterisk doesn't do key system things"? Or just a current limitation? |
| 18:11.49 | Qwell[] | notjason: design choice, I'd say |
| 18:12.03 | Qwell[] | there are other more "proper" ways to do some of those things |
| 18:12.03 | dlynes_office | notjason: a lot of that's a limitation of the phones available, too |
| 18:12.06 | file | notjason: well Asterisk was designed to be a PBX, not a key system... we realize people want those and we have ideas... but nothing implemented exactly to do a key system |
| 18:12.24 | *** join/#asterisk justinu|laptop (n=justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 18:12.49 | file | Nugget: I want sweet&sour :( |
| 18:12.52 | Qwell[] | well, now THAT's ironic! |
| 18:12.58 | [TK]D-Fender | notjason : Certain things * can do, some can be done with some trickery, others it will do SOON, others it simply won't. |
| 18:13.00 | dlynes_office | notjason: for example, some sip phones do call announce, but it's nothing you can control from the pbx, the announcement is generated internally, and offhand, i don't know of any sip phones that'll do paging |
| 18:13.05 | notjason | I do hear the "* is not a key system" argument alot, but there are many large PBX platforms (Avaya Definity or CM, for example) that are considered PBXs and also have those features. |
| 18:13.32 | file | notjason: well one of the other problems is that they control the phones in use |
| 18:13.34 | notjason | Obviously, any features would need phone support, I guess that goes without saying. |
| 18:13.37 | [TK]D-Fender | notjason : Ask yourself the real featureset you require, and HOW you can operate, not just how you're used to operating. |
| 18:13.41 | dlynes_office | notjason: by paging, i mean paging through the phone's internal speaker, not through a loudspeaker |
| 18:13.52 | dlynes_office | notjason: asterisk can do paging through a horn or a loudspeaker already |
| 18:13.55 | Qwell[] | file: I shall bring 3 muffins, and a "thing" of sweet & sour |
| 18:13.56 | *** part/#asterisk amdtech (n=amd011@ab1-1-246.shsu.edu) |
| 18:14.05 | notjason | I also meant station set paging (as opposed to loudpseaker) |
| 18:14.12 | [TK]D-Fender | dlynes_home : Many phones with auto-answer capabilities can do paging... |
| 18:14.20 | file | the other tricky thing is we have to be protocol independent on this all... or at least try to be |
| 18:14.22 | dlynes_office | [TK]D-Fender: ah |
| 18:14.26 | [TK]D-Fender | dlynes_home : Including Polycom's of course ;) |
| 18:14.37 | dlynes_office | [TK]D-Fender: but it's either autoanswer, or not autoanswer |
| 18:14.37 | justinu|laptop | atheros wifi driver for linux doesn't have any power management features whatsoever :( |
| 18:14.50 | Qwell[] | skinny can do autoanswer paging :D |
| 18:15.01 | dlynes_office | [TK]D-Fender: with a keysystem, paging and calling are two entirely different things, and they can work together |
| 18:15.09 | *** join/#asterisk amdtech (n=amd011@ab1-1-246.shsu.edu) |
| 18:15.26 | [TK]D-Fender | dlynes_home : With Polycom auto-answer, paging = calling w/ conditions attached :) |
| 18:15.49 | dlynes_office | notjason: and i'm guessing whisper page is just a low volume phone page? |
| 18:16.03 | dlynes_office | notjason: i've never heard of that feature on a keysystem |
| 18:16.21 | file | notjason: look what you started |
| 18:16.40 | [TK]D-Fender | dlynes_home : Whisper page I believe is a directly force-answer on the phone. |
| 18:16.48 | dlynes_office | [TK]D-Fender: so you can tell polycom to autoanswer or to ring the phone normally when you call it? |
| 18:16.51 | [TK]D-Fender | dlynes_home : Which is also doable. |
| 18:16.52 | notjason | [TK]D-Fender: Yeah, I know there are many ways to get what one wants out of a "phone system", and with larger applications, things like bridged appearances give way to parking, etc. I was more interested in whether Asterisk would even have the capabilities to be a viable replacement in "small" applications like homes and small offices, which often have "squared system" behaviors, even when they also offer "PBX" features |
| 18:17.00 | [TK]D-Fender | dlynes_office : Correct. |
| 18:17.01 | *** join/#asterisk [vmwarez]dotcom (n=jjones@216.147.224.254) |
| 18:17.20 | dlynes_office | [TK]D-Fender: i'm guessing that's something you'd have to modify some chan_sip code for? |
| 18:17.35 | [TK]D-Fender | notjason : Bridged (Shared) call appearance support is due for * 1.4. |
| 18:17.42 | notjason | "Whisper page" is like station set paging, but the paging comes in through the handset when the phone is in use and can only be heard by the paged party |
| 18:17.49 | dlynes_office | [TK]D-Fender: but only partial support |
| 18:17.49 | [TK]D-Fender | dlynes_home : Nop, just use SIPAddHeader. |
| 18:17.52 | dlynes_office | ah |
| 18:18.02 | notjason | To be fair, whisper page is one of the least used features ever :) |
| 18:18.03 | *** join/#asterisk Beighto (n=chatzill@64.160.113.130) |
| 18:18.09 | paolob | dlynes_home, the problem of the "bad file descriptor" begins when asterisk reads sip_notify.conf |
| 18:18.19 | [TK]D-Fender | notjason : AHH.. umm.. thats pretty mucha "HELL NO" idea for * :) |
| 18:18.50 | vader-- | hmmm asterisk recognizes the caller id name but for some reason my cisco 7940G phones only show the number |
| 18:18.50 | dlynes_office | [TK]D-Fender: yeah...making coffee at 6am automatically is probably a higher priority |
| 18:18.56 | vader-- | any ways to get the name on them? |
| 18:19.00 | [TK]D-Fender | notjason : SIP is not at all designed for that concept and it'd be into NSA territory to implement, and the get busted for ;) |
| 18:19.08 | dlynes_office | vader--: smash them with a sledgehammer? |
| 18:19.14 | [TK]D-Fender | dlynes_office : I'll give you my code for that later ;) |
| 18:19.18 | notjason | It also seems to me that a lot of the fancy phone system behaviors I'm talking about are probably in opposition to the architecture of SIP, since they require switch control over the phone's UI. |
| 18:19.25 | notjason | Yeah |
| 18:19.46 | dlynes_office | lol |
| 18:19.51 | file | Qwell[]: might want to get that cough checked |
| 18:19.54 | [TK]D-Fender | notjason : Indeed. So are these questions merely "hypothetical" or are you considering an actual * implementation? |
| 18:19.57 | Qwell[] | file: ;) |
| 18:19.58 | dlynes_office | heh....Qwell's always plugging chan_skinny :) |
| 18:20.05 | Qwell[] | but, yeah...skinny CAN do that |
| 18:20.07 | dlynes_office | Even if it is buggy as all hell :) |
| 18:20.09 | Qwell[] | theoretically, heh |
| 18:20.16 | [TK]D-Fender | dlynes_home : And I always plug Polycom, and thus there is balance ;) |
| 18:20.17 | Qwell[] | dlynes_office: damn straight...and it's going into trunk tonight! |
| 18:20.23 | notjason | I'm somewhat surprised by that response (re SIP being a "hell no" feature, considering Asterisk's great conferencing capability. |
| 18:20.47 | Qwell[] | notjason: like I said...things can be implemented in different ways |
| 18:20.48 | [TK]D-Fender | notjason : I meant "whisper page" is a HELL NO. Not SIP.. |
| 18:21.09 | notjason | Sorry, I mean "whisper page on SIP" |
| 18:21.18 | [TK]D-Fender | notjason : I can imagine a way, but it'd be WAY out of the spec and have to be impleneted in the core of * and chan_sip. UGLY to say the least. |
| 18:21.35 | notjason | I could see that, architecturally |
| 18:21.40 | [TK]D-Fender | notjason : And people just don't care about that feature as you said ;) |
| 18:21.46 | notjason | Indeed |
| 18:21.48 | dlynes_office | [TK]D-Fender: have you taken a look at app_dial.c or app_voicemail.c? |
| 18:21.51 | [TK]D-Fender | notjason : So ARE you looking at a real * install? |
| 18:21.53 | Qwell[] | [TK]D-Fender: nah, that could be done in dialplan, I bet |
| 18:21.55 | dlynes_office | [TK]D-Fender: those are pretty damned ugly, too :) |
| 18:22.03 | [TK]D-Fender | dlynes_home : Once, and only briefly |
| 18:22.08 | notjason | Real, but not important. Just a personal setup at a few locations for now. |
| 18:22.15 | dlynes_office | and then you got a migraine from looking at it? |
| 18:22.36 | notjason | But I was asked recently by a friend who actually sells PBX systems about the viability of using Asterisk for residential and small office installs |
| 18:22.50 | dlynes_office | notjason: quite viable |
| 18:22.52 | [TK]D-Fender | notjason : VERY worthwhile option, and can meet "key" systems in price on almost the lowest end and KILL it with anything more interesting, including VM / IVR, etc. |
| 18:23.01 | notjason | (The question was prompted by the total failure of Avaya's new "tiny" one-X platform to be even remotely useful) |
| 18:23.10 | dlynes_office | notjason: I've installed quite a few asterisk systems instead of keysystems |
| 18:23.20 | [TK]D-Fender | notjason : Most of us here run it at work, and I run it at home, and consult all around.. I'd say * is quite capable :) |
| 18:23.21 | dlynes_office | notjason: I'm replacing a panasonic keysystem with one next week |
| 18:23.38 | notjason | But if you can't do bridged appearances, I don't see it being a great replacement for residential phone systems, for example. |
| 18:23.49 | dlynes_office | residential? |
| 18:23.55 | Strom_C | "phone systems"? |
| 18:23.57 | Strom_C | haha |
| 18:24.00 | notjason | Yes, you can get people to use parking instead, but when you tell them that, they see it as a disadvantage |
| 18:24.02 | Qwell[] | why would a residential pbx need bridged appearances? |
| 18:24.04 | Strom_C | residences don't have phone systems |
| 18:24.07 | [TK]D-Fender | notjason : I almost got an Avaya IP Office solution shoved down my throat but *'s option cost HALF as much and offered better growth potential and destroys it on features. |
| 18:24.08 | dlynes_office | how do you get residential and phone systems in the same sentence? |
| 18:24.16 | notjason | You don't know my friend's clients :) |
| 18:24.21 | Strom_C | residences have a half dozen phones on the same copper loop |
| 18:24.24 | Dr-Linux | Jun 16 11:06:38 WARNING[31264]: chan_zap.c:6105 ss_thread: CallerID returned with error on channel 'Zap/2-1' << what does it mean? |
| 18:24.46 | notjason | Okay, but IP Office (despite being total crap) has a bunch of features that * doesn't. |
| 18:24.48 | *** join/#asterisk cp5 (n=cp5@rtr3-t1.fonality.com) |
| 18:24.51 | dlynes_office | Dr-Linux: it means either your gains are too low, or you didn't wait the requisite 2 rings for caller id |
| 18:24.53 | [TK]D-Fender | Dr-Linux : Means it couldn't read it properly (noisy line maybe) |
| 18:24.56 | cp5 | anyone ever run into Event 18 on analog lines? |
| 18:24.58 | dlynes_office | Dr-Linux: before answering it |
| 18:24.59 | notjason | I know a _lot_ about IP Office (OH THE PAIN!) |
| 18:25.06 | dlynes_office | notjason: avaya? |
| 18:25.14 | [TK]D-Fender | notjason : Yes a few small features, but * added SO mcuh more by contrast... |
| 18:25.16 | notjason | Yeah. |
| 18:25.25 | dlynes_office | notjason: i don't know if you noticed or not |
| 18:25.29 | TripleFFFF | anyway top see real time codec trasnaltion ? |
| 18:25.34 | dlynes_office | notjason: but avaya is also what? about $20K? |
| 18:26.21 | notjason | I think what Asterisk really has to offer "small business" installations over IP Office is cost. Aside from that, * doesn't give them anything they want (which is a shame, but true) |
| 18:26.21 | Dr-Linux | dlynes_home: it not always come, but some time i see this warning on CLI |
| 18:26.21 | [TK]D-Fender | dlynes_home : My quote for an Avaya was about $45K. * cost me $25K, and I tossed $5000 on a supported box, and 26 IP 600's instead of cheap shit phones.... |
| 18:26.25 | dlynes_office | Dr-Linux: your gain probably needs to be increased a little bit then |
| 18:26.30 | Strom_C | notjason: not true. I've been able to meet all of my clients' small office needs with asterisk |
| 18:26.42 | [TK]D-Fender | dlynes_home : Avaya is a choice to make some suit happy who is name-dropping-happy. |
| 18:26.45 | notjason | Avaya is a total ripoff, of course, but only when compared to *. Compared to its "closed" competition, IP Office, for example, is "more money, but you get more", to most customers. |
| 18:26.48 | Strom_C | notjason: you have to separate out perceived need from actual need |
| 18:26.54 | dlynes_office | notjason: I know 0 small offices that could even afford avaya ip office |
| 18:27.03 | Dr-Linux | dlynes_home: well, when i increase the gain, then it creates echo .. currently voice is just FINE. |
| 18:27.13 | dlynes_office | notjason: I'm guessing a small office to you is an office with 100 staff? |
| 18:27.16 | amdtech | Strom_C: that hits the nail right on the head |
| 18:27.27 | amdtech | problem is most people think they know what they need |
| 18:27.46 | Strom_C | if people really knew what they needed, then all of us consultants would be out of a job :) |
| 18:27.49 | notjason | A small office, for this argument, is 15-100 staff. |
| 18:27.51 | amdtech | exactly |
| 18:27.57 | mountainm2k | Strom_C: LOL |
| 18:27.59 | [TK]D-Fender | amdtech : And they think they need things the way they did it before... it took people very little time here to adapt to not having "lines" on their phones.... |
| 18:28.26 | dlynes_office | notjason: tell them how much moolah they're going to be saving, and they'll stop asking for all the stupid features they don't need |
| 18:28.26 | amdtech | no kidding... same here... we've had "rollover" lines on our phones for years... it took me 2 months to convince them to try call waiting |
| 18:28.54 | Strom_C | amdtech: I just do multi-appearance hunting on multiline IP phones |
| 18:29.07 | Strom_C | amdtech: it's like five lines of code in extensions.conf :) |
| 18:29.14 | amdtech | yep :) |
| 18:29.16 | dlynes_office | And I already have something similar to bridged line appearance |
| 18:29.18 | notjason | I think the problem that comes up when selling * in a small business setting is that the customer is also being pitched by other vendors who are offering the same capabilities at a higher price, but also saying "and you won't have to change how you work; you can keep doing 'pick up Bob on line 3'" |
| 18:29.37 | mountainm2k | Strom_C: is it even that many? using Grandstream I didn't need to add anything -- or maybe I did and don't know it |
| 18:29.38 | dlynes_office | If you have a decent multi line phone, you can already see the line ringing on multiple phones |
| 18:29.40 | [TK]D-Fender | notjason : Valid scale... True that Key/hybrid systems can match * is one thing and offer features that * can't because of the integration, * DOES still offer you a lot thats NOT feasable with key systems. Video phones? Soon to Be Video voicemail? IVR/Queues/Voip capability right away? |
| 18:29.40 | mountainm2k | (newb) |
| 18:29.43 | dlynes_office | and see which phone it is |
| 18:29.46 | dlynes_office | erm which line it is |
| 18:29.47 | notjason | dlynes_office: What's that? |
| 18:30.02 | dlynes_office | notjason: just buy a decent phone, not some cheap piece of crap chinese phone |
| 18:30.20 | [TK]D-Fender | notjason : and with parking you can say "pick up bob on parking 3" just as easy... |
| 18:30.22 | Strom_C | notjason: so you ask them "do you really want to spend an extra five figures on not having to spend a day or two getting used to the new system?" |
| 18:30.31 | dlynes_office | notjason: I know Aastra supports it for sure, Grandstream GXP2000 supports it, and I'm sure Polycom supports it |
| 18:30.33 | *** join/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal) |
| 18:30.50 | Heimidal | how do you upgrade Asterisk? I've tried following directions I found online but 'show version' still reports the old version |
| 18:30.56 | notjason | As the integrator, I find * to be quite compelling, what with it's in-built voice server abilities (conf, IVR, VM, etc.), but it doesn't help the sell. |
| 18:31.03 | [TK]D-Fender | amdtech : You don't have roll-over? |
| 18:31.05 | Qwell[] | Heimidal: Did you run make install? Did you restart asterisk? |
| 18:31.08 | Strom_C | notjason: then you're a lousy salesman |
| 18:31.15 | Qwell[] | Strom_C: agreed |
| 18:31.25 | notjason | If you say so, Strom_C |
| 18:31.37 | amdtech | [TK]D-Fender: their version of rollover was basically a two line phone... if the first line was busy, the second line rang |
| 18:31.44 | Qwell[] | notjason: many of us have been doing this for years, and making a decent living from it... |
| 18:31.45 | dlynes_office | notjason: Asterisk has a lot more going for it than IP Office, imho |
| 18:31.53 | [TK]D-Fender | Then again Qwell still tries to sell us SCCP/Skinny and we still won't hear of it ;) |
| 18:31.57 | amdtech | [TK]D-Fender: so i told them i'm not going to support that since the phones have call waiting, plus they get a speed dial button now |
| 18:32.00 | Qwell[] | ;) |
| 18:32.05 | Qwell[] | [TK]D-Fender: Just wait... |
| 18:32.13 | Heimidal | Qwell[]: man, I feel like an idiot.. didn't restart it :P |
| 18:32.23 | [TK]D-Fender | amdtech : Ok, so the lack of roll-over is your CHOICE to offer them and extra feature then. |
| 18:32.23 | asterboy | I'm getting serious echo when listening to music on hold. Seems to be picking up anything in my background and trying to overlay it on output which of course creates grabled sound. If I mute the mic of course it stops that behavior. |
| 18:32.38 | [TK]D-Fender | Qwell : I wont hold my breath ;) |
| 18:32.47 | asterboy | Any suggestions beside disabling music on hold? |
| 18:32.50 | dlynes_office | asterboy: get rid of the cheap phone and replace it with something better |
| 18:32.51 | notjason | I know several business owners who were pitched * solutions by one vendor only to have a proprietary vendor pitch them another system that they found more compelling (not that I think that's a good thing; just telling what I know) |
| 18:32.57 | Qwell[] | [TK]D-Fender: It's going in to trunk this weekend...ready or not :P |
| 18:32.58 | asterboy | Polycom? |
| 18:33.09 | [TK]D-Fender | Qwell : LATTER ;) |
| 18:33.10 | amdtech | [TK]D-Fender: you could say that |
| 18:33.14 | dlynes_office | asterboy: ask polycom to replace the handset then, maybe |
| 18:33.24 | asterboy | musint have it setup correctly. |
| 18:33.31 | dlynes_office | asterboy: or you might have echo disabled on your tdm card |
| 18:33.39 | dlynes_office | asterboy: or gains not set correctly |
| 18:33.39 | *** join/#asterisk saftsack (n=saftsack@p54A7ED94.dip.t-dialin.net) |
| 18:33.41 | asterboy | happens on any of the polys via handset or speaker |
| 18:33.44 | [TK]D-Fender | amdtech : I do the same. I'm running Polycom IP 600's here with 3 line keys for rolling, and 3 for speed-dials/presence, etc. |
| 18:33.56 | asterboy | maybe gains to much? |
| 18:33.59 | amdtech | notjason: you really just need to know which nerves to hit :) asterisk doesn't have a large enough install base to sell as well |
| 18:34.01 | dlynes_office | asterboy: ah...yeah...it's something to do with your echo cancellation settings then, or your gains |
| 18:34.10 | notjason | So maybe this isn't the place to ask, but is there another open project that's a better fit to what I'm seeking? |
| 18:34.19 | dlynes_office | asterboy: it only happens on analog calls, right? |
| 18:34.26 | notjason | amdtech: That's probably true |
| 18:34.29 | amdtech | [TK]D-Fender: yeah, we're replacing a ccm, so i'm having to make do with the cisco's |
| 18:34.33 | Strom_C | notjason: yes, it's "improve your ability to sell the product to your clients" |
| 18:34.47 | notjason | Huh? |
| 18:34.48 | [TK]D-Fender | notjason : * simply lacks name recognition. It does make it harder to sell since you don't have 10 tons of printed material to present and woo them with since functionality is very implementation specific. |
| 18:35.05 | Qwell[] | [TK]D-Fender: bah! print the wiki |
| 18:35.06 | [TK]D-Fender | amdtech : Cisco's are great phone, though finnicky with SIP. |
| 18:35.23 | dlynes_office | notjason: set up a demo unit, bring it into their office, and demo it |
| 18:35.23 | amdtech | [TK]D-Fender: you know, at first, it was a pain... but they're extremely stable for us |
| 18:35.41 | amdtech | [TK]D-Fender: our entire infrastructure is cisco, so management is a breeze (switches and on the server) |
| 18:35.43 | [TK]D-Fender | Qwell[] : More proof of your clearly superior marketing skillz (wit'a'z!)\ |
| 18:35.46 | dlynes_office | notjason: show them some of the features asterisk can do that ip office can't |
| 18:35.54 | Beighto | speaking of phones, has anybody tried any Snom phones? |
| 18:36.01 | [TK]D-Fender | amdtech : Can work if you know what you're doing. |
| 18:36.05 | dlynes_office | notjason: such as having home extensions, or having their cell phone as an extension, ... |
| 18:36.14 | Heimidal | is there any way to make sure asterisk knows where mpg123 is? it's not playing hold music but I know I have everything setup right |
| 18:36.15 | notjason | [TK]D-Fender: Maybe, but I think it's more than just a marketing gap. I don't thing Asterisk is deficient when compared to its closed competition, but it is "different" in what it is able to do. I think Asterisk will get much stronger when it combines its current advantages with the (very small number of) missing abilities that are decision factors for customers. |
| 18:36.24 | Strom_C | notjason: or the incredible ease of moves, adds, and changes |
| 18:36.28 | amdtech | i think the biggest selling point for asterisk is that if you can think of it, there's a pretty good chance that you/someone can program it in |
| 18:36.46 | [TK]D-Fender | Beighto : SNOM = flakey ass firmware, poor LCD usability. Only viable in certain Euro markets... |
| 18:36.57 | dlynes_office | notjason: also, the changeup costs with asterisk are significantly lower to the customer because you don't need to do a site visit |
| 18:37.00 | [TK]D-Fender | notjason : I agree for the most-part.... |
| 18:37.03 | justinu|laptop | unless you want really flexible outbound dialing, asterisk can probably do it |
| 18:37.07 | dlynes_office | notjason: you can just go in through an ssh tunnel |
| 18:37.08 | notjason | I don't really do IP Office systems; I do Definity / Call Manager. It's much harder to win against Definity. |
| 18:37.18 | mountainm2k | [TK]D-Fender: how about the Grandstream? I prettu much bought what was cheap |
| 18:37.28 | [TK]D-Fender | notjason : But looking at realist phone system REQUIREMENT, I'd say * eats up over 90% of corporate needs rather nicely. |
| 18:37.36 | mountainm2k | probably going to get a IP301 to try out as well tho |
| 18:37.39 | [TK]D-Fender | mountainm2k : FORGET GRANDSUCK |
| 18:37.45 | mountainm2k | heh |
| 18:37.46 | dlynes_office | hahahaha |
| 18:38.07 | dlynes_office | I don't understand why people go for grandstream |
| 18:38.07 | mountainm2k | So you're going to tell me to buy the Polycom? |
| 18:38.12 | Beighto | [TK]D-Fender : So, what do you like besides Cisco? |
| 18:38.13 | [TK]D-Fender | notjason : Depends.... yes Definity is top end, but the price tag is exorbitant. If cost is no issue, sure, go ahead.... |
| 18:38.17 | notjason | In terms of the kind of things it tries to do, Asterisk is much more a Definity killer than an IP Office killer (substitute your favorite non-Avaya platforms for those, of course) |
| 18:38.20 | [TK]D-Fender | Beighto : Polycom > ALL. |
| 18:38.21 | mountainm2k | I know why I went with them -- cheap |
| 18:38.22 | dlynes_office | It's only what? about $30 cheaper than polycom? and the customer's never heard of grandstream |
| 18:38.40 | mountainm2k | I loved Definity, way back... |
| 18:38.43 | notjason | Definity is wicked expensive -- it really is obscene |
| 18:39.10 | [TK]D-Fender | notjason : BCM is a flakey pile of shit as well... so * has a great place in the market... |
| 18:39.23 | [TK]D-Fender | notjason : Just none of the means to promote/validate itself. |
| 18:39.30 | dlynes_office | notjason: you can still tromp on a Nortel CICS, or a Panasonic KX-TDA30, too |
| 18:39.33 | mountainm2k | heh, if only the documentation were as good as teh software... |
| 18:39.34 | asterboy | Would be nice to dial an extension and listen to music over the phone |
| 18:39.35 | notjason | But Definity offers, basically, point-for-point what Asterisk can do (though * is far cheaper and non obnoxiously proprietary) and then adds a lot of convenience features |
| 18:39.45 | [TK]D-Fender | dlynes_home : CICS?! *barf* |
| 18:39.47 | notjason | (I can't believe I'm defending Definity. What's wrong with me?) |
| 18:39.50 | mountainm2k | For a newB it's tough to get started... It's getting better, but it's still not great |
| 18:39.51 | asterboy | should be a way to do that with the dial plan. |
| 18:40.03 | dlynes_office | [TK]D-Fender: Most of our customers are using CICS |
| 18:40.18 | dlynes_office | [TK]D-Fender: some of our newer ones are using KX-TDA30's |
| 18:40.19 | [TK]D-Fender | asterboy : "show application musiconhold) |
| 18:40.24 | Beighto | I could do Polycom, price isn't horrible, looks reasonable |
| 18:40.31 | [TK]D-Fender | dlynes_office : Poor, poor saps... |
| 18:40.34 | dlynes_office | Those are the two most popular keysystems in Vancouver, right now |
| 18:40.40 | [TK]D-Fender | Beighto : Very solid phones.... |
| 18:40.51 | [TK]D-Fender | dlynes_home : Much the same here... I'm aiming to change that ;) |
| 18:40.56 | dlynes_office | same here |
| 18:41.03 | dlynes_office | We're converting a lot of them over to Asterisk |
| 18:41.14 | *** join/#asterisk jonnysupersonic (n=jonny@dsl-165-169-195.telkomadsl.co.za) |
| 18:41.19 | camelon | I'm done an asterisk reload and suddenly one of two channel banks don´t working . . .wich could be the most probaly reazon . . .hardware o software? |
| 18:41.24 | Beighto | [TK]D-Fender are you even happy with their bottom of the line phones? Looks like they go for about $135 |
| 18:41.29 | dlynes_office | but as far as price point competition goes |
| 18:41.34 | Qwell[] | camelon: my guess is a typo |
| 18:41.37 | notjason | So, what will the bridged appearance capability in 1.4 look like? |
| 18:41.43 | dlynes_office | CICS and KX-TDA30 are much harder to beat than Avaya IP Office |
| 18:41.46 | Qwell[] | notjason: like this |
| 18:41.51 | Qwell[] | | - | - | |
| 18:41.54 | notjason | I'm concerned because I feel like SIP is really incompatible with stuff like that |
| 18:41.58 | camelon | qwell . .thanks . .typo?? |
| 18:42.00 | notjason | Thanks, Qwell |
| 18:42.06 | Qwell[] | notjason: no problem |
| 18:42.21 | *** join/#asterisk rollot (n=rollotom@c-68-37-168-10.hsd1.pa.comcast.net) |
| 18:42.46 | [TK]D-Fender | Beighto : I won all of them. Depends on your needs as to which is best for you. Need/want PoE? What kind of call volume? Need Speakerphone? |
| 18:42.57 | [TK]D-Fender | s/won/own/ |
| 18:43.03 | rollot | Anyone know what's going on with NuFone? |
| 18:43.05 | dlynes_office | need full duplex speakerphone? |
| 18:43.18 | notjason | I'm really eager to see this stuff because I'd love to get this friend's company (which is a pretty sizable Avaya dealer for which I consult on "open" projects) away from proprietary and into * (or something else that's open) |
| 18:43.24 | dlynes_office | polycom's only half duplex, remember |
| 18:43.25 | asterboy | thanks TK |
| 18:43.29 | dlynes_office | erm polycom 301 i mean |
| 18:43.36 | Beighto | good to know |
| 18:43.45 | [TK]D-Fender | dlynes_home : Only the 301 is half, the 430/501/601 are all full-duplex |
| 18:43.56 | dlynes_office | [TK]D-Fender: that's what i said |
| 18:44.11 | Heimidal | is there any way to debug hold music? |
| 18:44.17 | *** part/#asterisk surfdue (n=surfdue@unaffiliated/surfdue) |
| 18:44.18 | notjason | Should I be looking beyond SIP for this? Is there a system that works better for this stuff and is supported by Asterisk? |
| 18:44.19 | [TK]D-Fender | dlynes_home : My response was more throrogh than your quicky correction :) |
| 18:44.21 | Beighto | I wish someone would make a descent IAX phone |
| 18:44.32 | camelon | qwell .. .about my writing???sorry! |
| 18:44.55 | mountainm2k | Beighto: here here |
| 18:45.15 | mountainm2k | would be best for home office phones |
| 18:45.39 | [TK]D-Fender | Heimidal : Maybe you should try describing the problem. |
| 18:46.18 | [TK]D-Fender | notjason : Maybe SCCP, but then again your just replacing SIP issues with SCCP ones and validing the * core VS cCCM. |
| 18:46.20 | asterboy | works nice. |
| 18:46.32 | Strom_C | notjason: Some people seem to go balls-crazy over H.323; Qwell has also been working on some awesome SCCP stuff that's going into the development version of asterisk |
| 18:47.02 | asterboy | now I need to program my Polycom's [Services] button to show me a list of streaming radio stations that I can select from. |
| 18:47.41 | [TK]D-Fender | asterboy : Doable (on a 60x) |
| 18:47.53 | kaz0358 | heh.. i finally got registering 16 accounts from a single asterisk box working again. whatever their upgrade was yesterday, it was causing SIP 600 errors--too frequent reguests. anyway, it is really a hack. basically i increased the defaultexpiry by a couple of minutes and then modified chan_sip.c to set the expirary to max expirary - some random amount. it would almost be better to add in some extra option to control how fast register attempts |
| 18:47.56 | Beighto | I had a home office that had PPPoA and couldn't get my crappy grandstrem through to the server |
| 18:47.57 | notjason | I know even less about the specifics of SCCP and H.323 than I do about SIP, but I do see that many "traditional" phone systems that support IP "feature phones" are using H.323 (and of course Cisco using SCCP for the same purpose) |
| 18:47.57 | kaz0358 | are made to a particular host. |
| 18:47.58 | Heimidal | I've set up hold music to play when I press 4. It worked before I upgraded to 1.2.9.1, but now it no longer works. |
| 18:48.15 | asterboy | I have [Services] on my 600 set to give me this menu: |
| 18:48.17 | Heimidal | in the console, it shows it start playing and then stopping |
| 18:48.23 | asterboy | --> Digg.com Feed |
| 18:48.27 | notjason | Is SCCP truly "open", or is it a matter of playing catch-up every time Cisco puts out some new software? |
| 18:48.29 | asterboy | --> Weather |
| 18:48.32 | kaz0358 | ohh.. i should have mentioned that i was getting the sip 600 error registering with free world dialup |
| 18:48.36 | asterboy | --> Calendar |
| 18:48.45 | asterboy | ---> Family Pictures |
| 18:48.53 | *** join/#asterisk dsully (i=daniel@electricrain.com) |
| 18:48.53 | asterboy | works so sweet |
| 18:49.05 | notjason | asterboy: That sounds very cool. |
| 18:49.19 | asterboy | ya, my wife really likes the Weather button |
| 18:49.39 | dsully | hi there. I just moved my asterisk box (Debian) from one data center to another, and switched my PRI service over. I'm now getting errors on the d-channel: |
| 18:49.49 | dsully | Jun 16 11:45:08 NOTICE[1547]: chan_zap.c:8171 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 18:49.50 | notjason | That capability is an _enormous_ selling point in high-end residential applications. |
| 18:49.57 | dsully | Jun 16 11:46:05 NOTICE[1547]: chan_zap.c:8171 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 18:50.02 | notjason | And we sell, um, "a lot" of those :) |
| 18:50.09 | asterboy | residential? |
| 18:50.21 | notjason | Yes. A.K.A. "rich people" |
| 18:50.28 | asterboy | ah |
| 18:50.37 | Strom_C | dsully: same telco providing the PRI? |
| 18:50.37 | asterboy | Where I live there are only cheap fuckers |
| 18:50.38 | dsully | Googling shows trying to increase the jitterbuffer (didn't work). Or suggests an interrupt problem. Disks are scsi, asterisk is the only thing the box is running. |
| 18:50.43 | dsully | Strom_C: yes. |
| 18:50.49 | asterboy | WalFart shoppers |
| 18:50.50 | Strom_C | dsully: same switch? |
| 18:50.59 | notjason | They all have Partner, Legend, Magix, or IP Office, which does what they want, but is waay to big for them |
| 18:51.01 | dlynes_office | notjason: h323, sip, iax2, mgcp, skinny (sccp), ... |
| 18:51.01 | dlynes_office | notjason: the Nortel BCM protocol is partially implemented, too |
| 18:51.01 | dlynes_office | notjason: but by and large, sip has the most number of features |
| 18:51.09 | kaz0358 | conceptially, does anyone concur with the idea that if you have a LOT of register entries in the sip.conf file, that it should stagger them out of a time interval when asterisk is first starting up or you do a sip reload? |
| 18:51.10 | dsully | yes, same switch type. |
| 18:51.20 | Strom_C | dsully: is it the same physical switch |
| 18:51.32 | kaz0358 | and that the interval should be specified somewhere in the sip.conf file |
| 18:51.32 | dsully | that I don't know.. I can ask XO. |
| 18:51.36 | Strom_C | dsully: you do that |
| 18:51.52 | dsully | why/how would that affect the PRI service? |
| 18:51.56 | Strom_C | dsully: if you havent changed anything on your end, but the PRI itself has changed, then that points to problems with the PRI |
| 18:52.11 | dsully | right - they're looking into it, but not finding any errors on their end. |
| 18:52.12 | asterboy | hmmm, my musiconhold is timing out |
| 18:52.13 | notjason | Reading some stuff on the Web, it looks like being able to assign park/unpark to feature buttons with working lights does 90% of what would be needed for our customers. |
| 18:52.23 | notjason | And, almost more importantly, what I want for my house :) |
| 18:52.24 | dsully | Wanted to check to see if I missed anything that could be software related. |
| 18:52.39 | *** join/#asterisk jonnysupersonic (n=jonny@dsl-165-169-195.telkomadsl.co.za) |
| 18:52.42 | Strom_C | dsully: have you modified your asterisk box AT ALL in the course of the move? |
| 18:52.58 | notjason | That handles everything except being able to have someone join you on an existing call by hitting a lit "line" button. |
| 18:53.01 | asterboy | is there a setting to tell musiconhold to play indefinately |
| 18:53.02 | asterboy | ? |
| 18:53.33 | dsully | Strom_C: Just the eth0 network config. Everything else is exactly the same. |
| 18:53.49 | Strom_C | dsully: find out if XO has your circuit coming off the same switch as before |
| 18:54.12 | rollot | has anyone else gotten the $60 to port toll fre did's for NuFone phone call from their CSR? I'm hesitant to pay (they seemed anxious that i pay for ALL of my #'s). is there another provider that anyone knows that offers similar features and supports LNP? |
| 18:54.20 | dsully | Ok. Somehow I doubt it is. About 15 miles between the old location and new location. |
| 18:54.33 | Strom_C | dsully: what's the ISDN switchtype? |
| 18:54.43 | dsully | national |
| 18:54.49 | Strom_C | yeah, that tells me nothing |
| 18:55.01 | Strom_C | figure out where the hell the circuit is coming from :) |
| 18:55.06 | dsully | heh. sorry. I will find out on their end when they call me back. |
| 18:55.32 | [TK]D-Fender | dsully : Make sure your PRI card is on its own IRQ... |
| 18:55.47 | asterboy | ah, I think it's the Polycom...must be the timeout if you don't press anything. |
| 18:55.58 | [TK]D-Fender | dsully : Sounds like a serious clocking failure. pastebin "cat /proc/interrupts" and "dmesg" |
| 18:55.59 | [TK]D-Fender | ~pb |
| 18:56.01 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 18:57.15 | Heimidal | http://rafb.net/paste/results/3lmGFD18.html |
| 18:57.23 | camelon | after doing some asterisk reloading one of two channel banks is not working . . . but after a total system halt and channel banks off .. . althe system is working . . . where y must look for the cause? hardware or softgware?? |
| 18:57.28 | Heimidal | that is what happens when I make a call within the office and place the call on hold |
| 18:57.42 | Heimidal | no music ever plays |
| 18:58.06 | dsully | [TK]D-Fender: http://pastebin.com/713562 |
| 18:58.16 | [TK]D-Fender | Heimidal : Pastebin your musiconhold.conf |
| 18:58.38 | Beighto | dsully: just out of curiosity what city are you located in with XO? |
| 18:58.52 | [TK]D-Fender | dsully : What model of server? |
| 18:59.17 | dsully | dell poweredge850 |
| 18:59.24 | dsully | Mountain View, CA |
| 18:59.38 | dsully | (this was all working before the move, fyi, for 9+ months) |
| 19:00.08 | dsully | and sorry - not scsi disks, sata disks. |
| 19:00.11 | [TK]D-Fender | Intel(R) PRO/1000 Network Driver - version 6.0.60-k2 |
| 19:00.30 | [TK]D-Fender | dsully : This typically implies the e1000 driver which has known issues with Digium cards... |
| 19:00.33 | Beighto | dsully: okay, was curious because I will be colocating my servers to XO in Roseville, CA. They offer a sip connection as well, rather than traditional PRI. |
| 19:01.19 | dsully | yeah - should I try the old e100 driver? or..? |
| 19:02.01 | dsully | oh, hmm.. nm. that driver is loaded, but the interface in there is a tg3 |
| 19:02.38 | [TK]D-Fender | dsully : Disable that NIC COMPLETELY |
| 19:02.46 | funxion | anyone familiar with this error ==> Call to undefined function mysql_connect() |
| 19:02.49 | dsully | there is no e1000 nic. |
| 19:02.54 | [TK]D-Fender | dsully : And unload the driver moudle. |
| 19:02.59 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
| 19:03.21 | dlynes_office | dsully: is it a broadcom chipset? |
| 19:03.25 | dsully | it's built into the kernel. I can rebuild without it built in, but doubt that would make much of a difference. |
| 19:03.33 | dsully | yes, i believe the tg3 is a broadcom chipset. |
| 19:03.48 | dlynes_office | yeah...i've had no end of problems with that chipset, too |
| 19:04.43 | [TK]D-Fender | dsully : Digium cards can be rather picky about these things... |
| 19:04.45 | dlynes_office | dsully: I don't think the problem is so much with the dual 1Gb ethernet card, as it is with the broadcom chipset, myself |
| 19:05.13 | dsully | Ok.. I'm still wondering why this issue came up now, when everything was working before. |
| 19:05.19 | dlynes_office | dsully: i've got my dual port eepro1000 card disabled, no drivers loaded, and i still have issues with my digium card |
| 19:05.40 | dsully | You would think they'd fix those issues.. very common eth cards. |
| 19:05.49 | dlynes_office | dsully: mine was fine for about 1-1/2 years before it started acting up |
| 19:06.05 | dsully | If I need to run to Frys and pick up a eth card - what is a recommended chipset/card? |
| 19:06.19 | dlynes_office | dsully: well, like i said |
| 19:06.35 | dlynes_office | dsully: i added in two new network cards (both 3com's) and i've still got issues |
| 19:07.25 | dlynes_office | and 3com cards are pretty much the best cards on the market afaik |
| 19:07.50 | lunk | 3com did invent ethernet |
| 19:08.03 | dsully | yeah, but their Vortex cards were flakey. |
| 19:08.09 | lunk | wt! |
| 19:08.14 | dsully | and Bob Metcalfe has long since left. |
| 19:08.19 | lunk | 905b-tx for the win |
| 19:08.30 | dlynes_office | vortex? you mean the 905b? |
| 19:08.53 | dlynes_office | or was it 905c? |
| 19:08.55 | Dandan | is there a way to send email notification through an external smtp server"? |
| 19:08.56 | dsully | probably. it's been a while. the e100/e1000 cards have been rock solid for me in every other application. |
| 19:09.15 | *** join/#asterisk Samoied (n=Samoied@201.41.254.10) |
| 19:09.43 | dlynes_office | i've only had a problem with one of those 3com cards |
| 19:09.49 | dlynes_office | and it was because of a cold solder joint |
| 19:09.54 | *** join/#asterisk cardiffit (n=sb@cpc1-pnwn1-0-0-cust445.cdif.cable.ntl.com) |
| 19:10.34 | cardiffit | Is there anyone here that has got caller id workin on NTL in the UK??? |
| 19:10.41 | *** join/#asterisk surfdue (n=surfdue@unaffiliated/surfdue) |
| 19:11.15 | surfdue | ok im getting these errors http://host41.pastebin.com/713554 it succesfully connects to my asterisk now but gets a 404 while what I can see trying to read the extensions config? |
| 19:11.23 | *** join/#asterisk xenoterracide (n=xeno@h69-129-148-178.69-129.unk.tds.net) |
| 19:11.27 | Heimidal | [TK]D-Fender: http://rafb.net/paste/results/iDDNFl83.html |
| 19:11.28 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 19:11.49 | [TK]D-Fender | Heimidal : What version of MPG123 are you running? |
| 19:11.52 | surfdue | To: <sip:8006170141@host41.com>;tag=as3d5a0e0e SIP/2.0 404 Not Found << seems to me like its trying to find an sip entry that looks like that no? |
| 19:11.57 | file | surfdue: ahum |
| 19:11.58 | file | # |
| 19:11.58 | file | Looking for 8006170141 in default (domain host41.com) |
| 19:11.58 | file | # |
| 19:12.00 | file | Jun 16 14:50:42 NOTICE[9136]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'default' |
| 19:12.09 | file | there, read that - use your brain - it's right in front of you |
| 19:12.13 | Heimidal | 0.59r |
| 19:12.14 | surfdue | file, I noticed that |
| 19:12.16 | dlynes_office | # |
| 19:12.17 | dlynes_office | Looking for 8006170141 in default (domain host41.com) |
| 19:12.17 | dlynes_office | # |
| 19:12.17 | dlynes_office | Jun 16 14:50:42 NOTICE[9136]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'default' |
| 19:12.17 | dlynes_office | # |
| 19:12.20 | surfdue | file, what should I use? |
| 19:12.21 | dlynes_office | yeah. |
| 19:12.34 | cardiffit | Is there anyone here that has got caller id workin on NTL in the UK??? |
| 19:12.35 | surfdue | or where is the docs for context? |
| 19:12.35 | thock | Question to someone who can answer: I have a sangoma a104d and a A200. (TI and Analog), is there any way to have zaptel/zapata configure both devices within one conf? |
| 19:13.23 | Dandan | is there a way to send email notification through an external smtp server? |
| 19:13.31 | funxion | anyone familiar with this error ==> Call to undefined function mysql_connect() |
| 19:13.32 | [TK]D-Fender | surfdue : Stop with the constant nagging for stuff you should have read up on AGES ago and go read THE BOOK. |
| 19:13.33 | [TK]D-Fender | ~book |
| 19:13.36 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:13.37 | surfdue | extensions.conf has no context in general should I add it? and forsay incoming? |
| 19:14.02 | [TK]D-Fender | thock : "one conf?" |
| 19:14.12 | thock | well, within the zaptel and zapata.conf's, respectively |
| 19:14.25 | [TK]D-Fender | thock : Yes, you configure them all in the same 2 files. |
| 19:14.27 | thock | i.e, sangoma suggests doing FXO/FXS first, then any sort of T1 |
| 19:14.28 | Heimidal | [TK]D-Fender: would the version I'm using be a problem? |
| 19:14.31 | thock | is that done by Groups? |
| 19:14.37 | thock | like, Signalling = blah |
| 19:14.40 | thock | group = 1/2/3/ etc? |
| 19:14.45 | [TK]D-Fender | Heimidal : Extremely possible. What are you running? |
| 19:14.49 | dlynes_office | thock: remember the a200 reserves 24 channels |
| 19:14.52 | surfdue | Jun 16 15:14:19 NOTICE[16241]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'default' where is this in? |
| 19:14.58 | Heimidal | [TK]D-Fender: 0.59r |
| 19:15.03 | dlynes_office | thock: so your t1 will start at 25, not 5 |
| 19:15.07 | thock | funxion: MySQL isnt running or asterisk can't login to it. |
| 19:15.21 | [TK]D-Fender | thock, no, your channel range will change. The A200 if detected first will be the first 24 and push everything else UP. Or the reverse if thats the case. |
| 19:15.23 | surfdue | and why is it looking for defualt? |
| 19:15.28 | dlynes_office | surfdue: it's not in anywhere, because you haven't written it yet |
| 19:15.34 | liran_ | i'm reading up "Asterisk - The future of telephony" by oreilly and it's worth every penny. great book. |
| 19:15.35 | dlynes_office | surfdue: because you told it to |
| 19:15.42 | funxion | thock asterisk loggs the cdr to the mysql database that output id from php agi |
| 19:15.43 | [TK]D-Fender | Heimidal : Got MP3's in that folder you mentioned? No ID3 tags on them? (a no-no) |
| 19:15.54 | surfdue | dlynes_office, where did I the sip.conf is set to context=incoming |
| 19:15.58 | dlynes_office | surfdue: are you using freepbx/amp/trixbox or asterisk@home? |
| 19:16.00 | xenoterracide | I know these are stupid questions. and I should read the book. I have the oreilly one here that I'm going to go through but that will take longer than I want. one if I update the manager.conf do I have to restart asterisk for it to load? 2 does asterisk have a web configuration utility. if so what port |
| 19:16.02 | Heimidal | [TK]D-Fender: how would I remove the ID3 tags? |
| 19:16.10 | surfdue | dlynes_office, im using asterisk pbx |
| 19:16.13 | thock | [TK]D-Fender: i wonder if the digital channels are still 24, would that be the case? It's only 1 line. |
| 19:16.32 | [TK]D-Fender | Heimidal : Pick your MP3 tool of choice... |
| 19:16.38 | dlynes_office | surfdue: yeah, and you probably overrode that context with your sip extension |
| 19:16.48 | [TK]D-Fender | thock : Your bchan = channels.... |
| 19:16.55 | dlynes_office | surfdue: or you have something where you didn't define an context |
| 19:17.04 | dlynes_office | surfdue: and if you don't specify one, it'll use default |
| 19:17.08 | thock | yes, but would one analog line take up 24 of those channels? |
| 19:17.09 | Katty | hi lovables. |
| 19:17.21 | dlynes_office | thock: yes |
| 19:17.24 | kaz0358 | is there a way to just reload extensions.conf without reloading everything else? |
| 19:17.25 | surfdue | dlynes_office, in extensions.conf i even definded one |
| 19:17.44 | surfdue | dlynes_office, i dont understand whats up with that |
| 19:17.44 | dlynes_office | thock: it'll only use the first channel, but hte card itself takes up 24 chanels |
| 19:17.44 | thock | Okay |
| 19:17.48 | thock | so just start the PRI channels at 25 then |
| 19:17.49 | dlynes_office | kaz0358: extensions reload |
| 19:17.49 | thock | simple enough |
| 19:17.51 | thock | Thanks :) |
| 19:17.57 | dlynes_office | thock: correct |
| 19:18.05 | kaz0358 | thanks |
| 19:18.11 | Katty | well don't say hi all at once ;) |
| 19:18.13 | dlynes_office | thock: it's somethign a lot of people overlook when they've got more than one sangoma card installed |
| 19:18.19 | dlynes_office | Katty: HI!!!! |
| 19:18.22 | Katty | :> |
| 19:18.37 | surfdue | dlynes_office, I guess im looking to what do I need to do to fix the error do I need to change the [incoming] to [defualt]? |
| 19:18.46 | Katty | :< |
| 19:19.08 | dlynes_office | surfdue: no...that won't fix the problem, that'll mask it |
| 19:19.23 | dlynes_office | surfdue: you need to fix your dialplan or your sip.conf file |
| 19:19.28 | Katty | file, deary, where are you |
| 19:19.33 | surfdue | dlynes_office, k |
| 19:19.34 | dlynes_office | surfdue: or both |
| 19:19.50 | dlynes_office | surfdue: they're buggy as all hell |
| 19:20.31 | surfdue | dlynes_office, http://host41.pastebin.com/713606 the context incoming is located in extensions.conf |
| 19:20.42 | surfdue | dlynes_office, is this not correct? |
| 19:21.06 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 19:21.24 | dlynes_office | surfdue: yeah, it's fine |
| 19:21.36 | dlynes_office | surfdue: i'm guessing that's not your whole sip.conf file though |
| 19:21.46 | surfdue | nop |
| 19:21.46 | surfdue | lol |
| 19:21.53 | *** part/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 19:21.54 | dlynes_office | surfdue: because you're kinda missing phones |
| 19:22.04 | [TK]D-Fender | thock : The A200 takes up 24 channels because its based on their AFT T1 fram and is expandable to 24 ports. |
| 19:22.10 | dlynes_office | surfdue: not to mention phone lines |
| 19:22.13 | surfdue | lol |
| 19:22.19 | surfdue | 1 sec ill give you the hole thing |
| 19:22.33 | dlynes_office | surfdue: scrub the passwords before you post it |
| 19:22.50 | dlynes_office | surfdue: and while you're at it, post your extensions.conf so i have the WHOLE picture |
| 19:22.58 | surfdue | ok |
| 19:23.00 | [TK]D-Fender | dlynes_home : "Not worth saving" |
| 19:23.10 | dlynes_office | [TK]D-Fender: lol |
| 19:23.11 | [TK]D-Fender | dlynes_home : I'd have asked already... |
| 19:23.25 | surfdue | in the sip.conf in nano how do I replace |
| 19:23.27 | surfdue | is it possible? |
| 19:23.34 | dlynes_office | surfdue: try asking on #nano |
| 19:24.06 | dlynes_office | surfdue: or use an editor that you understand |
| 19:24.08 | surfdue | j |
| 19:24.10 | surfdue | k |
| 19:24.14 | dlynes_office | surfdue: like maybe notepad |
| 19:25.13 | dlynes_office | [TK]D-Fender: come on...he's not as bad as that other guy |
| 19:25.20 | dlynes_office | ~kernel20 |
| 19:25.23 | jbot | [kernel20] an annoying user that is allergic to reading documentation. |
| 19:25.35 | surfdue | ok host41.com/sip.conf and host41.com/ext.conf |
| 19:25.43 | [TK]D-Fender | dlynes_office : No, kernel20 can LEARN. I've proved it.... there is *NOBODY* home here.... |
| 19:25.58 | dlynes_office | kernel20 can learn? |
| 19:26.02 | dlynes_office | you're sure about that? |
| 19:26.04 | surfdue | [TK]D-Fender, please dont say anything to me if you cant say anything nice? |
| 19:26.12 | [TK]D-Fender | dlynes_office : Yes, I managed to get through... it took some work, but I did... |
| 19:26.40 | dlynes_office | surfdue: ummmm |
| 19:26.49 | dlynes_office | surfdue: i thought you said you weren't using context=default anywhere? |
| 19:26.52 | [TK]D-Fender | surfdue : I was saying it TO you :) And semi-sorry, but you show no signs of ANY effort to read and learn at all. |
| 19:27.02 | surfdue | tk i did |
| 19:27.03 | dlynes_office | surfdue: you need to put your glasses on and read your own config files |
| 19:27.07 | dlynes_office | surfdue: and don't pm me |
| 19:27.46 | dlynes_office | anything you need help with, if someone helps you in the channel, others might be able to learn from it |
| 19:27.51 | *** part/#asterisk notjason (n=jordan@ool-457183bb.dyn.optonline.net) |
| 19:27.58 | [TK]D-Fender | dlynes_office : Want to repeat the PM'd link? |
| 19:28.03 | dlynes_office | surfdue: the general rule of etiquette is that if you want to pm someone, ask first |
| 19:28.04 | surfdue | should that be incoming or internal? |
| 19:28.12 | [TK]D-Fender | dlynes_office : I'd like to see what was overlooked... |
| 19:28.36 | dlynes_office | [TK]D-Fender: what pm'd link? |
| 19:28.44 | [TK]D-Fender | Katty : Thanks.. I can use all I can get these days... |
| 19:28.50 | Katty | (= |
| 19:29.03 | dlynes_office | surfdue: on which one? |
| 19:29.06 | [TK]D-Fender | dlynes_office : I missed the "full" sip.conf he was supposed to post, though that's what he sent |
| 19:29.12 | surfdue | the asterlink switches |
| 19:29.16 | Katty | anyone polycom certified? |
| 19:29.25 | [TK]D-Fender | Katty : I may as well be, why? :) |
| 19:29.26 | Katty | cause if ya are, we need to chat. |
| 19:29.26 | dlynes_office | <surfdue> ok host41.com/sip.conf and host41.com/ext.conf |
| 19:29.36 | Katty | [TK]D-Fender: my company wants me to get certified |
| 19:29.37 | surfdue | sip.conf |
| 19:29.43 | surfdue | the ony ones with context=defualt |
| 19:29.45 | dlynes_office | Katty: i'm going through certification atm |
| 19:29.45 | Katty | [TK]D-Fender: i need to know teh process. |
| 19:29.46 | surfdue | defualt* |
| 19:29.53 | Katty | dlynes_office: where do i start? |
| 19:30.00 | dlynes_office | Katty: their website |
| 19:30.05 | Katty | dlynes_office: )= |
| 19:30.07 | [TK]D-Fender | Katty : Multiple-choice on-line test you can apparently pass without even OWNING one :) |
| 19:30.07 | dlynes_office | Katty: There's a place on there to apply |
| 19:30.10 | Katty | dlynes_office: i went and looked there. |
| 19:30.14 | Katty | dlynes_office: but did not see anything. |
| 19:30.17 | Katty | [TK]D-Fender: :> |
| 19:30.21 | dlynes_office | Katty: there's three different levels, too |
| 19:30.28 | [TK]D-Fender | Katty : A friend of mine did it apparently. |
| 19:30.33 | dlynes_office | Katty: yeah...you don't need to own one to pass |
| 19:30.36 | [TK]D-Fender | Katty : Call them up... |
| 19:30.37 | Katty | i'll go look on their website again |
| 19:30.54 | [TK]D-Fender | Katty : The most important on any website is their PHONE NUMBER :) |
| 19:31.00 | [TK]D-Fender | ^info |
| 19:31.23 | Katty | kay |
| 19:31.43 | dlynes_office | surfdue: anyways, if you want anyone coming in through your outbound legs to make free phone calls through your system, be my guest and set their contexts to internal, also |
| 19:31.52 | surfdue | dlynes_office, no error now jsut 404s |
| 19:32.07 | *** join/#asterisk notjason (n=notjason@ool-457183bb.dyn.optonline.net) |
| 19:32.26 | surfdue | bbl |
| 19:33.39 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:34.58 | dlynes_office | Katty: Partners Menu -> Channel Certification -> Become a Partner |
| 19:35.44 | Katty | :>>>>>>> |
| 19:35.46 | Katty | *boingboing* |
| 19:36.01 | dlynes_office | And I guess they have more than three certs now |
| 19:39.11 | [TK]D-Fender | dlynes_office : I was prepping for VoIP certs |
| 19:39.54 | dlynes_office | yeah...that's what we're going to be prepping for, too |
| 19:40.02 | dlynes_office | We've been on their list for over a year now |
| 19:40.10 | dlynes_office | Just haven't had the bloody time to become certified |
| 19:40.20 | dlynes_office | Kinda sucks running an interconnect with only two people |
| 19:40.33 | Katty | meesha |
| 19:40.34 | dlynes_office | One's the telecom guy, and the other's the computer guy |
| 19:40.41 | dlynes_office | I'm the computer guy |
| 19:40.42 | Katty | that's a lotta certs |
| 19:42.59 | *** part/#asterisk xenoterracide (n=xeno@h69-129-148-178.69-129.unk.tds.net) |
| 19:43.42 | funxion | so noone uses a2billing |
| 19:43.55 | [TK]D-Fender | funxion : Very few people here... |
| 19:43.57 | dlynes_office | funxion: people use it |
| 19:44.02 | dlynes_office | funxion: just not many on here |
| 19:44.14 | funxion | anyone here good with php agi |
| 19:44.18 | dlynes_office | funxion: probably better off to ask on the asterisk-users list |
| 19:44.26 | dlynes_office | ~lists |
| 19:44.33 | dlynes_office | ~list |
| 19:44.37 | jbot | one warez list being sent |
| 19:44.47 | dlynes_office | ~mailinglist |
| 19:44.49 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 19:46.20 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 19:47.02 | *** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
| 19:47.51 | tekati | ANYONE: I am using Broadvoice now and they managed to let my DID get LNP'ed to a different carrier and it has taken them over a week already and it is still not fixed. Does ANYONE know of a different VoIP provider Asterisk friendly that has some sort of Unlimited calling? |
| 19:48.24 | tekati | Or at least very reasonable. I use this for my home phone line. |
| 19:50.03 | funxion | anyone seen this beofre |
| 19:50.04 | funxion | AGI Tx >> 510 Invalid or unknown command |
| 19:50.04 | funxion | AGI Rx << Fatal error: Call to undefined function mysql_connect() |
| 19:50.26 | Katty | twisted[asteria]: you around? |
| 19:51.18 | MikeJ[Laptop] | hi Katty! |
| 19:51.24 | Katty | hey hun |
| 19:51.37 | MikeJ[Laptop] | wassup these days.. long time no see |
| 19:52.03 | Katty | been pretty busy with work....never at home at night. |
| 19:52.17 | Katty | doin that social thing, i spose. |
| 19:52.19 | Katty | how's you? |
| 19:52.32 | MikeJ[Laptop] | good |
| 19:52.49 | MikeJ[Laptop] | busy.. work too much, catch flack from the wife... |
| 19:52.53 | MikeJ[Laptop] | same old same old.. |
| 19:53.22 | surfdue | ok heres the latest http://host41.pastebin.com/713688 |
| 19:53.27 | Katty | excellent. |
| 19:53.38 | surfdue | for some reason still 404 but gladly no context error :) |
| 19:53.39 | MikeJ[Laptop] | indeed |
| 19:55.02 | Katty | are you going to the chicago thing in august, mike? |
| 19:55.23 | MikeJ[Laptop] | surfdue, looks like the extension doesn't exist |
| 19:55.28 | surfdue | hmm |
| 19:55.33 | surfdue | what should it look like? |
| 19:55.42 | Katty | bkw_: and i need to talk to you when you come back. |
| 19:55.52 | Katty | bkw_: no rush, deary |
| 19:55.58 | MikeJ[Laptop] | it should be ext 8006170141 in incoming |
| 19:55.59 | surfdue | its under [incoming] exten => _1NXXNXXXXXX,1,Dial(SIP/200,30|m) |
| 19:56.10 | surfdue | err |
| 19:56.11 | surfdue | sorry |
| 19:56.12 | surfdue | lol |
| 19:56.14 | Vorondil | tekati: we use teliax at work, and it's been pretty good. very asterisk friendly |
| 19:56.27 | MikeJ[Laptop] | Katty, you may want to PM him or IM him.. he's been running around crazy today |
| 19:56.39 | Katty | eh, i'll just call him then |
| 19:56.43 | tekati | Thank you will check out their website and see. |
| 19:56.46 | surfdue | ty |
| 19:56.47 | surfdue | lol |
| 19:56.53 | MikeJ[Laptop] | surfdue, no 1... |
| 19:57.12 | MikeJ[Laptop] | calls coming in with just 10 digit with no leading 1 |
| 19:58.04 | surfdue | :P |
| 19:58.06 | surfdue | got it thanks |
| 19:58.15 | MikeJ[Laptop] | :) |
| 19:58.27 | *** join/#asterisk caloi (i=caloi@69.67.251.96) |
| 19:58.37 | *** part/#asterisk [vmwarez]dotcom (n=jjones@216.147.224.254) |
| 19:59.09 | [TK]D-Fender | 404 |
| 19:59.21 | caloi | Anyone has any experience building a call queue on Asterisk, with remote agents (remote = not on asterisk server) |
| 19:59.28 | [TK]D-Fender | people keep thinking its LYING... |
| 19:59.38 | MikeJ[Laptop] | heh |
| 19:59.48 | caloi | my thought is, the 'remote agents' could log in to the call queue, say over the pstn or a cell phone |
| 19:59.56 | [TK]D-Fender | caloi : Possible, but painful |
| 20:00.06 | caloi | and when a call comes into the AST queue, the call is sent to the agent |
| 20:00.19 | MikeJ[Laptop] | caloi, you can do that with onhook agaents, or do offhook agents and have it call em through local too |
| 20:00.35 | [TK]D-Fender | caloi : Not really.. once a call to the outside is placed it is considered "answered" immediately and the call won't redistribute. |
| 20:00.36 | surfdue | ok thanks mike, next problem is the sip wont ring my pap2 heres the context let me look at it before you yell at me that im stupid [TK]D-Fender http://host41.pastebin.com/713701 |
| 20:00.43 | *** part/#asterisk rollot (n=rollotom@c-68-37-168-10.hsd1.pa.comcast.net) |
| 20:00.45 | MooingLemur | all agents are remote, it's just that you may be using Zap instead of IAX or SIP |
| 20:00.46 | [TK]D-Fender | caloi : Plus the problem of Cell phone VM taking the line... |
| 20:01.24 | surfdue | could that music on hold error stop the hole process? |
| 20:01.53 | tekati | Vorondil: How is the service with those folks? I frequently experience sound issues with Broadvoice. They even have a local number in my area at Teliax to so that is great news. |
| 20:02.02 | caloi | we have a 'hosted solution' which ties Mitel phones (using the minet protocol) to a server on our end, no function exists for call queueing and i'd like to use the AST to provide that service, but as TK said, it seems like there is no way to maintain "presence" of an agent |
| 20:02.09 | caloi | we could disable VM on the lines, |
| 20:02.20 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 20:02.59 | stephane_ | soir le monde |
| 20:03.20 | MooingLemur | I've disabled cell phone voicemail myself for similar reasons (not for queueing, but for simultaneous ringing) |
| 20:03.37 | caloi | Woud it be possible to create a queue that rings each off-asterisk line 3 times then rolls to the next? if the first line was busy, it would roll to the next immediatly? |
| 20:03.57 | caloi | in theroy it sounds possible to me --- but in reality it may be difficult |
| 20:04.36 | [TK]D-Fender | surfdue : I see it looks like it says its dialing, what happens on it? |
| 20:04.44 | [TK]D-Fender | And pastebin your sip.conf again (complete) |
| 20:04.48 | caloi | an 'on-hook' agent is one who dials into the AST and then waits for a call, while they remain off hook right? |
| 20:04.56 | [TK]D-Fender | stephane_ : Salut. |
| 20:05.37 | Gabriel25 | to activate music on hold for linksys PAP2A-na I have to add MOH Server: ip from my pbx box |
| 20:07.08 | [TK]D-Fender | Gabriel25 : Never heard that before... |
| 20:07.19 | surfdue | [TK]D-Fender, it seems to just keep ringing? host41.com/sip.conf |
| 20:07.32 | [TK]D-Fender | surfdue : Does it ACTUALLY ring? |
| 20:07.39 | surfdue | no |
| 20:07.42 | dlynes_office | [TK]D-Fender: does stephane_ ever say anything besides jour, matin, or soir le monde? :P |
| 20:07.47 | surfdue | well when im calling the number yes |
| 20:07.51 | surfdue | but the pap2 no |
| 20:08.30 | [TK]D-Fender | surfdue : So the PAP2 seems to "think" its ringing, but doesn't? |
| 20:08.51 | surfdue | yes |
| 20:08.51 | surfdue | lol |
| 20:09.06 | [TK]D-Fender | surfdue : Make sure you didn't disable the ringer on the actual phone then, and make sure that you plugged it into the right port on it. |
| 20:09.16 | surfdue | it is |
| 20:09.18 | surfdue | its all right |
| 20:09.21 | [TK]D-Fender | since the PAP2 is a dual port device. |
| 20:09.33 | [TK]D-Fender | verify the ringer some other way |
| 20:10.01 | [TK]D-Fender | surfdue : And when the CLI says it should be ringing, try just picking up... |
| 20:10.01 | dlynes_office | surfdue: another thing I would do, just to simplify things, is to strip your configuration down to one phone/ata port, and one outgoing line |
| 20:10.09 | [TK]D-Fender | surfdue : Even if you don't hear it. |
| 20:10.14 | Dr-Linux | http://www.digitnetworks.com/store/product_info.php?products_id=33 |
| 20:10.24 | surfdue | it works now |
| 20:10.24 | surfdue | lol |
| 20:10.28 | dlynes_office | surfdue: then it'll be much easier to pinpoint your issues, and troubleshoot |
| 20:10.37 | Dr-Linux | i wanna buy digium FXO card (4 port) |
| 20:10.37 | surfdue | it seems line was ringer was on but line 2 was not |
| 20:10.43 | Dr-Linux | is the above link fine? |
| 20:10.46 | surfdue | thats odd they both connect to the same asterisk :P |
| 20:10.49 | surfdue | thank you |
| 20:11.17 | dlynes_office | Dr-Linux: yep...the link works just fine |
| 20:11.52 | Dr-Linux | dlynes_home: are you reffer any other? |
| 20:12.22 | dlynes_office | www.voipsupply.com? |
| 20:12.55 | camelon | <dlynes_office> ~mailinglist |
| 20:12.55 | camelon | <jbot> Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 20:12.55 | camelon | * t |
| 20:13.12 | mitcheloc | or better - www.voiplink.com |
| 20:13.15 | MikeJ[Laptop] | jbot abuse |
| 20:13.18 | jbot | extra, extra, read all about it, abuse is http://www.rfc-ignorant.org/policy-abuse.php or RFC 2142 or echo Shut your festering gob, you tit! | mail abuse |
| 20:13.18 | dlynes_office | camelon: ? |
| 20:13.31 | MikeJ[Laptop] | heh |
| 20:14.29 | camelon | SORRY!!a little error . . . I apologize evryone! |
| 20:15.13 | Dr-Linux | from where i should buy Digium FXO 4 port card? |
| 20:15.31 | dlynes_office | Dr-Linux: from a store that sells them |
| 20:15.31 | MikeJ[Laptop] | from the guy with the best price? |
| 20:15.32 | [TK]D-Fender | Dr-Linux : www.voipsupply.com or www.atacomm.com would be good places in the US |
| 20:15.44 | MikeJ[Laptop] | ebay? |
| 20:15.50 | Gabriel25 | how can I setup music on hold with linksys PAP2T ? |
| 20:16.08 | dlynes_office | MikeJ[Laptop]: problem with a lot of ebay shops is that they won't ship outside the US |
| 20:16.09 | Dr-Linux | dlynes_office: @ :@ :@ :@ |
| 20:16.11 | [TK]D-Fender | MikeJ[Laptop] : Leave a molar and the Digium Faerie will leave you one! |
| 20:16.18 | Dr-Linux | [TK]D-Fender: thanks |
| 20:16.20 | MikeJ[Laptop] | exactly |
| 20:16.30 | Dr-Linux | MikeJ[Laptop]: http://www.digiumcards.com/digium_cards_fxo.html how about this? |
| 20:16.38 | MikeJ[Laptop] | ummm |
| 20:16.40 | [TK]D-Fender | Gabriel25 : You're not supposed to need a MoH server... it "just works" as is... |
| 20:16.53 | Dr-Linux | i'm already using 3 digium cards with same modules |
| 20:16.54 | MikeJ[Laptop] | Dr-Linux, I dunno.. how about it? |
| 20:16.55 | *** join/#asterisk bblack (n=mbarb@38-118-35-20.rapiddsl.net) |
| 20:17.00 | [TK]D-Fender | Gabriel25 : Got MoH on other phones in your setup and not the PAP2? |
| 20:17.01 | bblack | hi guys. I have a tdm400p and running asterisk trunk. When I'm ringing a phone over a fxs port the tdm400p doesn't detect when I pick up the phone. I'll put it back, it keeps ringing... any thoughts? (ps works well with 1.2.9.1) |
| 20:17.14 | Dr-Linux | MikeJ[Laptop]: the first one card on the link above |
| 20:17.28 | *** join/#asterisk feld_ (n=feld@ruc-mwt-gw-1.ruc.mwt.net) |
| 20:17.38 | [TK]D-Fender | Dr-Linux : just use one of the 2 I gave you... reputable places... |
| 20:17.50 | dlynes_office | Dr-Linux: you said you already have three |
| 20:17.59 | Dr-Linux | [TK]D-Fender: ok great |
| 20:17.59 | dlynes_office | Dr-Linux: why not just order from the same place again? |
| 20:18.13 | *** join/#asterisk rickyO (n=ricky@newly.osser.se) |
| 20:18.21 | Dr-Linux | dlynes_home: i hve 3 and have some T1 cards, but needs more for more setups |
| 20:18.31 | rickyO | Hi! |
| 20:18.35 | dlynes_office | Dr-Linux: but what was wrong with your last supplier? |
| 20:19.22 | rickyO | Does anyone have any experience connecting a legacy PBX to asterisk? |
| 20:19.30 | Dr-Linux | dlynes_office: well, always look for the best |
| 20:19.38 | dlynes_office | shurg |
| 20:19.48 | [TK]D-Fender | rickyO : What do you have in mind? |
| 20:19.57 | Dr-Linux | dlynes_office: i bought those cards when i don't know even about asterisk , neither about this place |
| 20:20.05 | dlynes_office | ic |
| 20:20.36 | rickyO | Well, I've got an * with two E1 cards, both are working and I've got most of the stuff going as it was before putting * inbetween out E1 and the PBX |
| 20:20.41 | Dr-Linux | [TK]D-Fender << this is guy who configured and install my first card. that's why i respect him much |
| 20:20.53 | syzygybsd | [TK]D-Fender: didn't you know, vauge questions always get exactly the answer you want |
| 20:20.56 | dlynes_office | lol |
| 20:20.57 | feld_ | Dr-Linux: he's a genius |
| 20:21.30 | rickyO | The only problem I've seen so far is that if I lift the receiver and ask for a line I get the line but I am only able to press 1 digit... |
| 20:21.32 | Dr-Linux | he helps without any mean |
| 20:21.44 | rickyO | Then * thinks I'm finished. |
| 20:22.00 | syzygybsd | rickyO: paste your extensions.conf somewhere and give me the link |
| 20:22.06 | [TK]D-Fender | rickyO : E1 Channel bank? |
| 20:22.10 | rickyO | When i enter the full number on the phone and then press "send number" it works. |
| 20:22.36 | [TK]D-Fender | rickyO : Sounds like a phone dial-plan error... what are you using? |
| 20:23.13 | rickyO | No, 2 Digium Wildcard TE110P T1/E1 Cards, 1 connected to the E1, the other to a DATAX E141 (E1 to 8 BRI converter) |
| 20:23.53 | syzygybsd | my guess is it is _X.,1,Dial(Zap/g1/${EXTEN}) |
| 20:24.00 | rickyO | I've got a DORO PBX with 3 BRI-lines... |
| 20:24.05 | [TK]D-Fender | rickyO : BRI phones? |
| 20:24.13 | Dr-Linux | http://www.voipsupply.com/product_info.php?products_id=294&searchid=19687 |
| 20:24.14 | syzygybsd | paste your extensions.conf so we can view it |
| 20:24.19 | Dr-Linux | this looks fine to me |
| 20:24.38 | [TK]D-Fender | Dr-Linux : Thats what you asked for... |
| 20:25.19 | MikeJ[Laptop] | Hmmhesays, ping |
| 20:25.23 | Hmmhesays | yo |
| 20:25.25 | Hmmhesays | what up |
| 20:25.30 | MikeJ[Laptop] | hey.. you still in fargo |
| 20:25.31 | Dr-Linux | [TK]D-Fender: yes, that one i need for new Pakistan setup |
| 20:25.46 | Hmmhesays | MikeJ[Laptop] |
| 20:25.47 | Hmmhesays | yeah |
| 20:25.59 | MikeJ[Laptop] | how far is perham, mn from you/ |
| 20:26.06 | Dr-Linux | anybody is using Cisco 7910 ip phone? |
| 20:26.10 | MikeJ[Laptop] | hr or 2? |
| 20:26.10 | rickyO | Ok, there's a lot of old "shit" in there as well, check under remote abd pbx contexts : http://pastebin.com/713741 |
| 20:26.30 | rickyO | abd=and |
| 20:26.44 | camelon | please help!. . .two times this day that a one of my two channelbanks suddenly stop working with yellow alarma in it. . . .someguide please |
| 20:27.12 | Dr-Linux | sorry cisco 7920 phone |
| 20:27.15 | [TK]D-Fender | rickyO : What context are your phones in? |
| 20:27.19 | Qwell[] | Dr-Linux: ask me on Monday |
| 20:27.31 | Qwell[] | well..Tuesday |
| 20:27.46 | Qwell[] | Dr-Linux: one is being overnighted to me, by somebody who wants it working right in chan_skinny :) |
| 20:28.06 | rickyO | the phones are digital connected to the PBX (group g1). |
| 20:28.07 | dlynes_office | Qwell[]: ah...people are finally starting to donate to the cause? |
| 20:28.15 | Qwell[] | dlynes_office: loan to the cause |
| 20:28.21 | dlynes_office | Qwell[]: ah...bummer :) |
| 20:28.22 | syzygybsd | rickyO: I don't know if this will help at all but try adding a:: exten => s,1,SetTimeout(digit=5) |
| 20:28.24 | [TK]D-Fender | rickyO : Pastebin their config please. |
| 20:28.27 | Dr-Linux | Qwell[]: i'm already using some 7940/60 |
| 20:28.37 | Qwell[] | Dr-Linux: What do you need to know about the 7920? |
| 20:29.03 | Dr-Linux | Qwell[]: i just wanna confirm , is the 7920 works the same way, and in the same way i'd need to load SIP firmware on it? |
| 20:29.04 | dlynes_office | Qwell[]: are you at least getting a bounty for your trouble? |
| 20:29.10 | camelon | dlynes . . .you that had some digium car picky problems could give me some guide?? |
| 20:29.18 | dlynes_office | camelon: ? |
| 20:29.19 | Qwell[] | dlynes_office: no.. |
| 20:29.21 | [TK]D-Fender | rickyO : I do suspect right off that the BRI phones are using some sort of "early dial" feature that incrementally checks a dial string for validity and is being cut short because of it.... |
| 20:29.25 | dlynes_office | Qwell[]: damn |
| 20:29.37 | Qwell[] | Dr-Linux: I don't know if the 7920 has sip firmware |
| 20:29.49 | rickyO | The it does work if I remove the * machine... |
| 20:29.51 | Dr-Linux | Qwell[]: when i install new Cisco 7940/60 , first i load SIP 6.3 and i upgrade it to SIP 7.X latest version |
| 20:30.04 | rickyO | And connect the Datax directly to the E1... |
| 20:30.13 | Dr-Linux | Qwell[]: i just wanna know, if i can do same on cisco 7920 |
| 20:30.21 | Qwell[] | Dr-Linux: I'm only going to be testing on sccp anyways, so.. |
| 20:30.30 | dlynes_office | camelon: if you're having a problem like that, it usually helps to see some logs and that kinda thing |
| 20:30.47 | [TK]D-Fender | rickyO : Could be that the PBX is better equiped to deal with its signalling... since its not a SIP phone you may not ahve a "dial-plan" per-se, but rather an issue with the phones interface style VS *. |
| 20:30.58 | Dr-Linux | Qwell[]: i need it with SIP firmware, i asked here maybe someone already got it to work |
| 20:31.17 | [TK]D-Fender | Dr-Linux : WiFi Phones SUCK. All of them... I wouldn't bother... |
| 20:31.33 | [TK]D-Fender | Dr-Linux : Use ATA's w/ good analog phones... |
| 20:31.35 | Qwell[] | [TK]D-Fender: I don't know... |
| 20:31.55 | rickyO | Ok, do you think there is a way to "stop" * from trying to dial directly with the first digit pressed? |
| 20:32.16 | rickyO | I get -- Extension '0' in context 'remote' from '' does not exist. Rejecting call on channel 0/31, span 1 |
| 20:32.16 | charles___ | hey guys |
| 20:32.21 | *** join/#asterisk swytch (n=ezcall@d83-177-224-198.cust.tele2.fr) |
| 20:32.24 | Dr-Linux | [TK]D-Fender: but already my BOSS purchased cisco 7920 , now he wants me to load sip firmware on it. :S |
| 20:32.24 | [TK]D-Fender | rickyO : Dunno... this is a challenging one... I am not the best to advise on this. |
| 20:32.26 | charles___ | anyone had experienced the GXP-2000 from grandstream ? |
| 20:32.39 | [TK]D-Fender | rickyO : but please do confirm which context your phones are using.... |
| 20:33.07 | rickyO | Well dialing out goes through remote and diallin through pbx (all set up in zapata.conf) |
| 20:33.16 | *** join/#asterisk jadaz87 (i=jadaz87@silenceisdefeat.org) |
| 20:33.17 | [TK]D-Fender | Dr-Linux : sounds like Quebec politics "First we declare independance! Then we vote on it! Any questions?!" |
| 20:33.22 | jadaz87 | hello everyone |
| 20:33.34 | jadaz87 | where did the a@h documentation go? |
| 20:33.43 | Qwell[] | jadaz87: to the bit bucket |
| 20:33.54 | rickyO | Hmm.... |
| 20:33.54 | [TK]D-Fender | jadaz87 : Straight to hell, not passing "GO" or collecting 200$ :) |
| 20:34.15 | Dr-Linux | [TK]D-Fender: :S |
| 20:34.29 | jadaz87 | Qwell[]: do you know what the default login credentials are for remote system administration? |
| 20:34.36 | Qwell[] | ~amp |
| 20:34.37 | jbot | somebody said amp was NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 20:34.50 | [TK]D-Fender | Dr-Linux : Advise him that "buying first and expecting success to follow" = failure. |
| 20:35.26 | Dr-Linux | nice words |
| 20:35.29 | Katty | what're you peoples doin tonight? |
| 20:35.37 | Katty | i've a perfectly good evening to fill up with various frolicings |
| 20:35.46 | Qwell[] | Katty: Fixing skinny. Wanna help? |
| 20:35.51 | [TK]D-Fender | Katty : Pool, beer, women. No particular order :) |
| 20:35.53 | Nivex | I intend to go swimming. |
| 20:36.10 | Katty | hmm, swimming.. |
| 20:36.14 | Nivex | [TK]D-Fender: two outta three ain't bad :) |
| 20:36.15 | Katty | [TK]D-Fender: i like where you're going. |
| 20:36.19 | generalhan | i need the smallest person here to come help me fish a lost wire out from the ceiling ... any takers ?? |
| 20:36.26 | Katty | [TK]D-Fender: well, the billiards anyway ;) |
| 20:36.26 | dlynes_office | probably just ogling women, myself |
| 20:36.28 | syzygybsd | [TK]D-Fender: if at first you don't succeed, redefine success |
| 20:36.36 | Hmmhesays | MikeJ[Laptop]: perham is about 55 miles |
| 20:36.42 | swytch | hi. i have calls coming in from a sip proxy to *, and going out again to the same sip proxy. and i see that just before hangup, * calls 'h' on the sip proxy. anyone knows what this could be? |
| 20:36.44 | dlynes_office | generalhan: lol |
| 20:36.46 | Katty | Hmmhesays: now why couldn't you be out here? |
| 20:36.48 | dlynes_office | generalhan: you and me both |
| 20:36.53 | [TK]D-Fender | syzygybsd : "If at first you don't succeed, skydiving may not be for you" |
| 20:36.54 | Hmmhesays | out where? |
| 20:36.56 | Katty | Hmmhesays: we could have a great friday night. |
| 20:36.56 | dlynes_office | generalhan: just lost a glow rod in a wall last week :( |
| 20:36.58 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 20:37.05 | Katty | Hmmhesays: there's a place you'd love down here called the Poor House |
| 20:37.08 | Hmmhesays | tonight i'll be about 20 miles from perham |
| 20:37.16 | Katty | perham? |
| 20:37.23 | Katty | is that in iowa? ;) |
| 20:37.25 | generalhan | dlynes_home: this is a 200+ year old building (all brick) and i got a wire stuck somewhere on my last pull string ... so i can cut and start over ... this is my last shot at it |
| 20:37.39 | Hmmhesays | <MikeJ[Laptop]> how far is perham, mn from you/ |
| 20:37.44 | dlynes_office | generalhan: fun fun fun |
| 20:37.48 | dlynes_office | generalhan: and no conduit, eh? |
| 20:38.02 | generalhan | conduit on the outside of the walls ... but not in the ceiling |
| 20:38.03 | file | can't sleep, codecs will transcode me! |
| 20:38.19 | swytch | i have a really simple config, and then wonders why * comes up with this call to 'h' (wich really takes place -- i trace the sip) as i dont have any 'h' extension anywhere. |
| 20:38.22 | dlynes_office | ceiling tiles on the ceiling though? |
| 20:38.24 | Qwell[] | [TK]D-Fender: good luck... |
| 20:38.40 | generalhan | dlynes_home: OMG that would be soooooooo easy ... no its all one big peice of sheet rock |
| 20:38.46 | dlynes_office | hahahahaha |
| 20:38.55 | *** join/#asterisk hohum (n=dcorbe@mail.interceltelecoms.com) |
| 20:38.57 | hohum | ser |
| 20:38.58 | rickyO | [TK]D-Fender: Thanks for your help so far... Do you have any ideas on who might be able to give me some help? |
| 20:39.11 | Hmmhesays | hohm what about ser |
| 20:39.11 | Qwell[] | generalhan: at least it isn't 18 inches of cement, every 10 feet or so |
| 20:39.16 | dlynes_office | yeah...almost everywhere we go, it's ceiling tiles and gypsum wallboard walls |
| 20:39.23 | generalhan | the worst part is that the place i need to crawl into is in between truss studs ... and the sheet rock wont hold me on its own .. so i really dont know what to do |
| 20:39.33 | [TK]D-Fender | rickyO : I'm still waiting for the answer to my question :) What context do your phones use? |
| 20:40.03 | dlynes_office | but we've had the odd place where it's 12 inches of concrete wall we have to drill through, or 48 inches of concrete floor we have to drill through |
| 20:40.06 | *** join/#asterisk roche (n=roche@crsj-dc1-fw001.accuhosting.com) |
| 20:40.10 | generalhan | Qwell[]: actually ... replace cement with bricks and thats about it ... they have added onto this building soo many times in the past 200 years ... its freaking crazy |
| 20:40.16 | Qwell[] | heh |
| 20:40.39 | dlynes_office | i don't even think this city has a single building that's 200 years old :) |
| 20:40.59 | dlynes_office | I think the oldest is probably 120 or 130 years old |
| 20:41.03 | generalhan | i think this is the oldest one in this town ... it was like an oldschool hotel/saloon back in the day |
| 20:41.10 | rickyO | [TK]D-Fender: For dialin to the legacy pbx I use [pbx] the legacy pbx uses [remote] to dial out, all configured in zapata.conf. |
| 20:41.24 | generalhan | they have old paintings and B&W pics hanging everywhere |
| 20:41.24 | dlynes_office | generalhan: yeah...i live in one of the newest cities in north america, though |
| 20:41.40 | generalhan | dlynes_office: where ? |
| 20:41.44 | dlynes_office | generalhan: as soon as a building gets to be about 30 or 40 years old, someone'll tear it down and build a new one |
| 20:41.45 | *** join/#asterisk hohum (n=dcorbe@12.195.58.236) |
| 20:41.51 | dlynes_office | generalhan: vancouver, canada |
| 20:42.02 | dlynes_office | nobody here likes old houses |
| 20:42.17 | dlynes_office | and there's no heritage legislation here, either |
| 20:42.22 | swytch | anyone would know why asterisk would emit a sip call to h@the_proxy (with same call ident as main call) just before hangup ? |
| 20:42.30 | dlynes_office | so nothing to stop them from tearing down a hundred year old structure |
| 20:42.31 | generalhan | see i really need to get to canada and meet everyone ... why are all you smart tech people from Canada ? |
| 20:42.43 | Qwell[] | swytch: because it's supposed to? |
| 20:42.44 | dlynes_office | Cause Vancouver kicks ass :) |
| 20:42.47 | generalhan | lol |
| 20:43.00 | dlynes_office | Lotsa super cute asian chicks here :) |
| 20:43.03 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 20:43.11 | swytch | Qwell[]: oh? |
| 20:43.12 | generalhan | ive only been to Montreal when i was in NY going to college ... nice place to drink but we didnt explore at all |
| 20:43.15 | [TK]D-Fender | rickyO : I was asking about your phones directly connected through that PRI/BRI adapter... |
| 20:43.20 | Katty | file: come to cape! |
| 20:43.28 | Katty | file: we can..hmm. |
| 20:43.31 | Katty | file: feed ducks! |
| 20:43.33 | Qwell[] | Katty: he isn't allowed out of his cage... |
| 20:43.35 | Qwell[] | he's been bad |
| 20:43.36 | Katty | file: and chase the college girls around |
| 20:43.46 | dlynes_office | Katty: i don't even think he's allowed to leave home yet |
| 20:43.48 | Qwell[] | alright, I guess you can have him |
| 20:43.49 | Katty | file: actually, the collage girls aren't here anymore. they all went home |
| 20:43.50 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 20:43.54 | file | Katty: :( |
| 20:43.57 | Qwell[] | oh, then nope, back in the cage |
| 20:44.02 | [TK]D-Fender | dlynes_office : Not true.. I've met him :) |
| 20:44.06 | Katty | file: we could watch the time warp at the university center tonight! |
| 20:44.09 | Katty | file: participation is required. |
| 20:44.09 | dlynes_office | [TK]D-Fender: lol |
| 20:44.15 | rickyO | [TK]D-Fender: Well, the sip-phone I have works as always, it's the phone using the legacy PBX that has this problem. |
| 20:44.44 | swytch | Qwell[]: the whole story is that i have a really simple config, and wonders why * comes up with this call to 'h' (wich really takes place -- i trace the sip) as i dont have any 'h' extension anywhere. |
| 20:44.51 | Katty | file: :> |
| 20:44.51 | dlynes_office | ever notice how when someone asks for help |
| 20:44.52 | nortex | Slightly OT but I have 70 some odd Polycom phones that fail to add entries to their directory, the phone says "Bust, try later" Has naybody seen this? |
| 20:44.55 | dlynes_office | and you go to help them |
| 20:44.59 | dlynes_office | they've gone afk? |
| 20:45.06 | Katty | dlynes_office: that's me ;) |
| 20:45.08 | generalhan | s/come/some |
| 20:45.12 | Katty | dlynes_office: actually, that's /not/ true. |
| 20:45.13 | dlynes_office | Katty: heh |
| 20:45.22 | Katty | dlynes_office: usually i've just pmed [TK]D-Fender |
| 20:45.23 | [TK]D-Fender | rickyO : Ok, I'm just stumped... not tech I'm really familiar with.,... |
| 20:45.26 | Katty | or asked twisted[asteria] |
| 20:45.33 | Katty | i rarely ask in channel anymore |
| 20:45.36 | dlynes_office | Katty: twisted's always afk |
| 20:45.40 | Katty | dlynes_office: not true |
| 20:45.41 | [TK]D-Fender | nortex : I have... usually a file permissions issue. |
| 20:45.54 | Katty | dlynes_office: even if he was, i could call him |
| 20:46.01 | dlynes_office | ah |
| 20:46.16 | dlynes_office | yeah...i've only been on this channel a short time |
| 20:46.20 | rickyO | [TK]D-Fender: Well, I set it up for the first time today myself... |
| 20:46.24 | swytch | Qwell[]: asterisk also calls the caller on the sip proxy. this behaviour does not takes place for calls coming from GW to another GW. Is asterisk supposed to do these calls to 'h' on the sip proxy. |
| 20:46.26 | dlynes_office | but i've already met two people in person from this channel |
| 20:46.26 | Katty | i'm not sure how long i've been here... |
| 20:46.34 | Katty | more than a year, me thinks |
| 20:46.43 | dlynes_office | about 2 months for me |
| 20:46.45 | dlynes_office | maybe not even that long |
| 20:46.46 | *** join/#asterisk pixiesfr (n=pixiesfr@highfun.net4.nerim.net) |
| 20:46.46 | Katty | Hmmhesays: have any idea how long i've been here? |
| 20:46.53 | Hmmhesays | No idea |
| 20:46.55 | Hmmhesays | where are you? |
| 20:46.57 | Qwell[] | Katty: 3 months |
| 20:47.05 | Qwell[] | give or take 2 years? |
| 20:47.07 | Katty | Qwell[]: that's not true. |
| 20:47.12 | generalhan | seriously ... anyone in the southwestern USA that wants to crawl around in the ceiling for me ... i cant pay you ... but ill sit and watch !! haha |
| 20:47.16 | Katty | Hmmhesays: err, this channel |
| 20:47.18 | pixiesfr | hi, can someone help me on TE110P |
| 20:47.23 | rickyO | [TK]D-Fender: It really didn't take that long (and no reconfiguration of the legacy pbx has been done), I was on my way home when I lifted the receiver instead of just enter the number to dial... |
| 20:47.27 | Hmmhesays | I remember one of the first days |
| 20:47.37 | rickyO | [TK]D-Fender: ... and it didn't work... |
| 20:48.02 | Hmmhesays | it was in fall of 2004 |
| 20:48.08 | Katty | damn. |
| 20:48.12 | Katty | longer thani thought |
| 20:48.20 | [TK]D-Fender | rickyO : Hrm :/ Ok, wish I knew more..... |
| 20:48.30 | [TK]D-Fender | Ok, I'm going home.. back in 10 mins ;) |
| 20:48.34 | Katty | cya |
| 20:48.34 | rickyO | [TK]D-Fender: Me too, anyway, thanks for your time... |
| 20:48.42 | [TK]D-Fender | rickyO : np. |
| 20:48.48 | pixiesfr | Unable to dialout with TE110P |
| 20:48.50 | swytch | Qwell[]: is it supposed to according to some sip RFC ? (where can i find out more) |
| 20:50.17 | mountainm2k | are PCI-Express slots backward compatible (IE can I put a TDM400P or a TE110P in one)? |
| 20:50.17 | dlynes_office | bbiab...just got make a quick errand |
| 20:50.22 | dlynes_office | mountainm2k: pci express, or pci-x? |
| 20:50.29 | mountainm2k | Express |
| 20:50.32 | dlynes_office | no idea |
| 20:50.36 | mountainm2k | X is a tiny little thing, it looks like |
| 20:50.41 | dlynes_office | pci-x is backwards compatible though |
| 20:50.48 | dlynes_office | but express is a different animl |
| 20:50.54 | mountainm2k | I'm examinging the inards of a Dell PowerEdge 1800 |
| 20:51.37 | Katty | are they dusty? |
| 20:51.39 | liran_ | Congestion() plays a busy-tone and hang up the call or do i need a hungup() directive after it? |
| 20:51.44 | mountainm2k | there are two long green slots, two small black ones, one 32bit (5V) and one 64bit (doesn't say, thinking 3.3v) slots |
| 20:51.58 | mountainm2k | No, I'm assuming the machine is brand new -- since it's in a photo on Dell's site |
| 20:52.41 | mountainm2k | the green ones look like they're backward compatible -- are they X or Express? |
| 20:53.21 | vader-- | do any of you guys have a good international number i could dial for testing? |
| 20:53.31 | Qwell[] | vader--: try the pope? |
| 20:53.34 | vader-- | hehe |
| 20:53.36 | mountainm2k | LOL |
| 20:53.45 | Qwell[] | or inmarsat? |
| 20:53.49 | *** part/#asterisk praet (n=praet@wsip-68-15-32-50.ri.ri.cox.net) |
| 20:54.06 | *** part/#asterisk swytch (n=ezcall@d83-177-224-198.cust.tele2.fr) |
| 20:54.28 | *** join/#asterisk tpak (n=tpak@69-162-129-127.clspco.adelphia.net) |
| 20:54.47 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 20:55.00 | MooingLemur | liran_: depends on the channel, but if it's not answered, Congestion() signals the calling party and the dialplan terminates there. If it's answered, I'm not sure how long it'll run before either hanging up or moving to the next priority. Try it and find out :P |
| 20:55.39 | liran_ | i see |
| 20:55.43 | *** join/#asterisk Seyr (n=Seyr@cpe-67-10-139-141.houston.res.rr.com) |
| 20:55.51 | *** part/#asterisk QbY_ (n=Kelvin@cm-64-221-171-241.dhcp.southerncoastalcable.net) |
| 20:56.02 | Seyr | Does anyone still user SER + Asterisk? It looks like the SER project is dead |
| 20:56.02 | *** join/#asterisk pdunkel (n=pdunkel@213.235.231.189) |
| 20:56.36 | pixiesfr | does anyone can tell why I could dialout ? |
| 20:56.45 | pixiesfr | does anyone can tell why I couldn't dialout ? |
| 20:57.11 | *** part/#asterisk tpak (n=tpak@69-162-129-127.clspco.adelphia.net) |
| 20:58.00 | *** part/#asterisk dsully (i=daniel@electricrain.com) |
| 20:59.42 | *** join/#asterisk dorphalsig (n=chiardon@200.71.58.39) |
| 21:00.08 | *** join/#asterisk mopri (n=jjohn@201.192.107.57) |
| 21:01.32 | mopri | hi, is it posible to add something to put something like this? callerid=asreceived <"9"+asreceived> something like that.. i know obviously it won't work like that |
| 21:01.50 | Qwell[] | mopri: no, but it can be done in dialplan |
| 21:06.12 | carrar | set it to a variable |
| 21:06.13 | dorphalsig | Hello. |
| 21:06.28 | carrar | then set callerid to 9variable |
| 21:06.39 | *** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 21:07.01 | Qwell[] | callerid will already be in the var.. |
| 21:07.03 | dorphalsig | I have this weird message in my logs.... |
| 21:07.32 | dorphalsig | "Didnt finish Caller-ID Spill Cancelling." |
| 21:07.34 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:08.14 | dorphalsig | and my system is crashing |
| 21:08.42 | dorphalsig | (actually,half of it) |
| 21:09.09 | *** part/#asterisk Seyr (n=Seyr@cpe-67-10-139-141.houston.res.rr.com) |
| 21:10.36 | carrar | ,Set(CALLERID(number)=9${CALLERIDNUM}) |
| 21:10.40 | carrar | that should work |
| 21:11.15 | Qwell[] | carrar: why use CALLERID(number) and CALLERIDNUM? |
| 21:11.24 | carrar | spice it up! |
| 21:11.26 | Qwell[] | both are the same thing, but the latter is deprecated |
| 21:11.47 | generalhan | Anyone in here with a 7960 that the MWI will NOT go away ... even if there arent any messages ??? i need to figure out whats wrong with this thing |
| 21:14.26 | dorphalsig | I have this weird message in my logs...."Didnt finish Caller-ID Spill Cancelling." and half my system is crashing |
| 21:14.48 | dorphalsig | what can that mean? |
| 21:16.28 | *** join/#asterisk kolpi9a (i=kolpi9a@c-68-60-13-69.hsd1.ga.comcast.net) |
| 21:20.22 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 21:21.21 | [TK]D-Fender | . |
| 21:25.30 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 21:25.42 | robin_sz | meep? |
| 21:27.36 | Katty | meeper. |
| 21:29.13 | *** join/#asterisk RF_MIA (n=mw1@ip67-93-229-222.z229-93-67.customer.algx.net) |
| 21:30.33 | rickyO | [TK]D-Fender: I've gotten a bit further... |
| 21:31.35 | rickyO | [TK]D-Fender: I put this http://pastebin.com/713880 in instead of the old [remote]. |
| 21:34.46 | [TK]D-Fender | rickyO : Shouldn't work at all... |
| 21:34.52 | generalhan | lol |
| 21:35.23 | rickyO | [TK]D-Fender: It does... I still loose the first number when doing DTMF dialing, but block send works as before... |
| 21:35.39 | rickyO | number=digit... |
| 21:36.00 | [TK]D-Fender | :/ |
| 21:36.02 | generalhan | STUPID MWI ... some one make this red light DIE !!!!!! |
| 21:38.02 | rickyO | [TK]D-Fender: It seems the PBX sends 1 digit after I choose to open a line.... That digit is lost, then I can enter the number and asterisk receives it as DTMF. |
| 21:38.47 | rickyO | If I enter the full number and then send it from the phone it uses the last "block" extension instead. |
| 21:39.00 | groogs | i'm tired of echo with my TDM400P .. what should i replace it with? |
| 21:39.08 | *** join/#asterisk X-Rob (n=rob@CPE-60-231-85-52.qld.bigpond.net.au) |
| 21:39.10 | Qwell[] | groogs: proper zap configs |
| 21:39.13 | generalhan | lol |
| 21:39.14 | Qwell[] | and, that right there is the man to ask |
| 21:39.20 | Qwell[] | X-Rob: ;) |
| 21:39.28 | groogs | Qwell[]: i have tried, and tried, and tried. my lines seem to change or something |
| 21:39.42 | Qwell[] | ask X-Rob...I'm sure he'll help you out |
| 21:40.02 | generalhan | Qwell[]: oh yeah thats cool ... as soon as he comes in toss him right in the fire ! |
| 21:40.03 | nortex | [TK]D-Fender, On that Polycom question about directories, is the phone trying to edit the file on the ftp server? |
| 21:40.27 | X-Rob | Geez, I only just woke up 8) |
| 21:40.30 | X-Rob | Wozzup, groogs? |
| 21:40.36 | [TK]D-Fender | nortex : Yup... resyncing local changes |
| 21:40.57 | groogs | X-Rob: never managed to 100% get rid of echo on my tdm400p. i was just going to buy something else |
| 21:40.57 | generalhan | groogs: first things first you should prolly pastebin your zapata.conf and zaptel.conf files |
| 21:41.17 | groogs | but then you guys are also going to make me upgrade ;) |
| 21:41.20 | X-Rob | groogs, mine's pretty much perfect now |
| 21:41.30 | groogs | i'm still on asterisk 1.0.2 i think |
| 21:41.32 | X-Rob | that's using zaptel-trunk tho |
| 21:41.36 | Qwell[] | ... |
| 21:41.38 | groogs | heh |
| 21:41.43 | X-Rob | Well, duh 8 |
| 21:41.44 | X-Rob | ) |
| 21:41.45 | generalhan | groogs: if you update .... that could solve ALL your iisues |
| 21:41.50 | groogs | yeah maybe it's time |
| 21:41.53 | generalhan | why are you so opposed |
| 21:41.54 | Qwell[] | there are MUCH better echo cans now |
| 21:42.11 | groogs | its running some version of amp between 007 and 009 |
| 21:42.41 | groogs | basically, i got it to work and just didnt want to touch it.. ;) |
| 21:42.46 | generalhan | lol |
| 21:42.52 | Qwell[] | sounds like AMP to me |
| 21:42.58 | groogs | oh. of course, it still does this "TDM PCI Master Abort" thing and locks up the box |
| 21:43.21 | *** join/#asterisk d-tech (n=dtc@72.245.233.107) |
| 21:43.44 | groogs | Qwell[]: hey, amp is fine. (note, me and X-Rob are both dev's.. though i haven't done any work in a while) |
| 21:43.51 | generalhan | ok well groogs question is loaded and complex ... will someone please tell me how to solve the MWI issue on my 7960s |
| 21:43.52 | groogs | :) |
| 21:44.12 | mopri | hi i have asterisk v1.0.9, and in the extensions i have a 928 extension, and a _9., but if i dial 928XXXX, it will go to 928 instead of 928XXXX, can i set a digittimeout in v1.0.9? |
| 21:44.18 | groogs | generalhan: with freepbx? |
| 21:44.37 | generalhan | groogs: i wouldnt be here if that were the case |
| 21:44.55 | generalhan | ive seen the post like 1000 times when people come in here for that |
| 21:44.58 | generalhan | ~amp |
| 21:45.00 | jbot | amp is probably NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 21:45.04 | generalhan | ~freepbx |
| 21:45.06 | jbot | i heard freepbx is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 21:45.38 | groogs | yeah i know , and #freepbx gets the same thing with centOS and other questions because of A@H |
| 21:45.48 | generalhan | yea |
| 21:46.17 | generalhan | it just seems dumb to me that only 2 out of 15 7960s are doing this ... im gonna put some electrical tape over the light i sweart |
| 21:46.29 | generalhan | s/sweart/swear |
| 21:46.50 | groogs | has anyone ever seen a box do that "tdm pci master abort" thing? It basically freezes, and that message scrolls across the console forever. does it every 2 to 30 days |
| 21:47.10 | groogs | generalhan: is there anything in /var/spool/asterisk/vm/default/* ? |
| 21:47.16 | groogs | (or whatever your path is) |
| 21:47.22 | generalhan | nope no messages |
| 21:47.27 | groogs | generalhan: IIRC, any .txt file in that directory makes the MWI turn on |
| 21:47.36 | generalhan | yea |
| 21:47.37 | generalhan | i checked |
| 21:47.39 | generalhan | there is nothing |
| 21:47.51 | groogs | any hidden files? |
| 21:47.53 | groogs | ls -la |
| 21:47.54 | generalhan | i even deleted the "inbox" "old" and "work" message folders completely |
| 21:48.02 | generalhan | ls -al shows nothing |
| 21:48.20 | X-Rob | Get rid of the Cisco's, use Snom's. |
| 21:48.24 | groogs | try creating msg0000.txt, wait a minute, then delete it |
| 21:48.25 | X-Rob | *nods firmly* |
| 21:48.34 | thock | anyone know of a few things i can do to cancel out some echo/ |
| 21:48.47 | generalhan | X-Rob: thanks man ... now that ive spent 1000's of dollars on these ones ! lol |
| 21:48.52 | X-Rob | groogs, now you mention it, I do remember seeing that years ago. |
| 21:48.58 | jhiver | lads, any experience with Asterisk + H.323 ? I've read the wikis but the sample h323.conf files are pretty sparse |
| 21:48.59 | X-Rob | (The PCI master abort thing) |
| 21:49.24 | jhiver | there's only one, and it's for oh323, and it's doesn't explain anything :( |
| 21:49.32 | groogs | actually theres a bit more on google now, compared to a year ago.. i'll look thruogh |
| 21:52.53 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
| 21:53.48 | X-Rob | anyway |
| 21:53.50 | X-Rob | time for me to go |
| 21:53.52 | X-Rob | home. w00t. |
| 21:57.19 | generalhan | Well crap ... so i switched phones with the owner so he wouldnt have the red light anymore and i would ... nope ... even with my phone he has the red light on his line and i still dont have it with his old phone |
| 21:57.22 | generalhan | this sux |
| 21:57.46 | groogs | generalhan: SIP? |
| 21:57.50 | generalhan | yes |
| 21:57.52 | generalhan | 8-3-00 |
| 21:58.34 | groogs | generalhan: you'll probably have to unplug all other phones to do this, but go into the console and type "sip debug" and then plug the phone in |
| 21:58.41 | Bullseye_Network | generalhan: try creating a new mailbox for him |
| 21:59.00 | Bullseye_Network | backup the existing IF he has saved messages. |
| 21:59.09 | generalhan | nope no messages |
| 21:59.10 | Bullseye_Network | and create it like new |
| 21:59.21 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-54-60.dsl.tul2ok.sbcglobal.net) |
| 21:59.55 | *** join/#asterisk Mattwj2005 (n=Matt@user-12l3n74.cable.mindspring.com) |
| 22:00.06 | generalhan | if i have the format as wav in voicemail.conf does that mean that * will play the busy.wav to the caller when they get to VM ? |
| 22:00.19 | Mattwj2005 | hey guys....I am getting a weird error...any ideas? Jun 16 16:58:42 WARNING[26647]: chan_sip.c:1066 __sip_xmit: sip_xmit of 0x8169f90 (len 395) to xx.xx.xx.xx:yyyy returned -1: Bad file descriptor |
| 22:00.57 | generalhan | because im using "format=wav49|gsm|wav" should i just have one in there ? |
| 22:01.43 | Bullseye_Network | generalhan: its fine that way |
| 22:01.55 | generalhan | im just going though all the options ... |
| 22:02.01 | groogs | and thats not going to affect MWI anyawy |
| 22:02.05 | generalhan | cause if i make him a new VM folder my macros wont work anymore |
| 22:02.23 | groogs | ? |
| 22:02.56 | generalhan | like if i make his VM box 9000 instead of 7004 then my macro that is set to call 7004 then goto u7004 wont work anymore |
| 22:03.31 | groogs | i think he meant recreate the directory |
| 22:03.39 | Bullseye_Network | Just delete the mailbox. and recreate it as the same number. |
| 22:04.02 | Bullseye_Network | Remove it from the voicemail.conf and reload and readd it like it was brand new |
| 22:04.20 | Mattwj2005 | any ideas what that error means? |
| 22:04.29 | surfdue | question asterisk i notice has alot of sound files, is there a way I can with them create a push 1 for support etc? |
| 22:04.40 | surfdue | I mean I think there is the recorded files to create that :P |
| 22:05.00 | Qwell[] | surfdue: sure |
| 22:05.35 | groogs | surfdue: if it's for a business, it honestly sounds better to just record your own |
| 22:05.36 | surfdue | qwell any advice in creating them? exten => 1,1,Playback(you-reached-sales-office) |
| 22:05.36 | surfdue | <PROTECTED> |
| 22:05.43 | Qwell[] | with & |
| 22:05.48 | surfdue | ok thank you |
| 22:06.00 | groogs | even if all hte sounds are there, the tone will sound weird |
| 22:06.16 | Qwell[] | groogs: no more so than other IVRs |
| 22:06.27 | Qwell[] | especially that hoe Claire... |
| 22:06.41 | Qwell[] | god I hate Claire |
| 22:06.44 | groogs | it sounds better if you just record the whole thing as one file |
| 22:06.49 | Qwell[] | groogs: well, sure |
| 22:06.57 | surfdue | where are the asterisk sounds at again? |
| 22:08.16 | Bullseye_Network | surfdue: I just create a mailbox and call and leave a message as the recording then I copy the gsm file into the /var/lib/asterisk/sounds directory and call them directly |
| 22:08.34 | Bullseye_Network | Just rename them of coarse |
| 22:08.35 | Qwell[] | Bullseye_Network: app_record is far better suited for that.. |
| 22:08.54 | surfdue | k |
| 22:08.56 | Bullseye_Network | that works too. |
| 22:09.08 | groogs | Bullseye_Network: yeah, setup an extension like *99 to run playback(beep) record(temp-recording) or whatever |
| 22:09.15 | Qwell[] | but, I doubt you sound anything like Allison :P |
| 22:09.22 | generalhan | lol |
| 22:09.48 | Mattwj2005 | any ideas anyone? |
| 22:09.49 | Mattwj2005 | :-s |
| 22:10.45 | Qwell[] | Mattwj2005: just one, but it's patent pending |
| 22:10.55 | Mattwj2005 | lol :P |
| 22:10.58 | Bullseye_Network | Mattwj2005: are all phones doing it or just a specific one? |
| 22:11.22 | Mattwj2005 | I only have one ATA |
| 22:11.38 | Mattwj2005 | it has two lines on it though |
| 22:11.49 | dlynes_office | Mattwj2005: Was it you under a different nick asking about that earlier? |
| 22:12.35 | Mattwj2005 | mine names are aways Mattwj then followed by a number |
| 22:12.50 | Mattwj2005 | usually 2005 sometimes it gives me 2006, etc |
| 22:12.56 | dlynes_office | Mattwj2005: ah...guess it wasn't you then...someone else was having the same problem |
| 22:13.09 | dlynes_office | Mattwj2005: What version of asterisk are you using? |
| 22:13.11 | surfdue | ok ok question, i am running the voice after no activity I want to repeat the menu and then hangup? |
| 22:13.13 | surfdue | possible eh? |
| 22:13.20 | dlynes_office | Mattwj2005: 1.2.7.1? |
| 22:13.21 | Qwell[] | ~docs |
| 22:13.26 | jbot | docs is probably probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 22:13.27 | Mattwj2005 | 1.2.9.1 |
| 22:13.30 | Qwell[] | surfdue: read time |
| 22:13.35 | surfdue | k thank you |
| 22:13.42 | surfdue | where it at :P |
| 22:13.49 | surfdue | k got it |
| 22:13.50 | dlynes_office | Mattwj2005: ah, ok...looks like that bug might still be in there, then..the other guy was using 1.2.7.1 and I suggested upgrading to 1.2.9.1 |
| 22:14.09 | dlynes_office | Mattwj2005: you get spammed with the error when you try to shut down, right? |
| 22:14.21 | Mattwj2005 | yeah definitely |
| 22:14.23 | *** join/#asterisk brc_ (n=brc_@pdpc/supporter/basic/brc) |
| 22:14.32 | surfdue | qwell do i really need a zaptel timer? |
| 22:14.33 | Mattwj2005 | it is continous |
| 22:14.37 | dlynes_office | Mattwj2005: Yeah...I'd report it as a bug then |
| 22:14.39 | Qwell[] | surfdue: you tell me |
| 22:14.48 | Qwell[] | Do you have a need for it? |
| 22:15.09 | surfdue | Qwell[], :) oh I know what to do :) |
| 22:15.10 | *** part/#asterisk RF_MIA (n=mw1@ip67-93-229-222.z229-93-67.customer.algx.net) |
| 22:15.11 | dlynes_office | Mattwj2005: I believe it's probably a socket that was previously closed, but someone forgot to reassign it a value of -1, or that specific code is not checking to see if the socket is closed already |
| 22:15.15 | surfdue | Qwell[], i can do a wait |
| 22:15.44 | Mattwj2005 | where is the official bugzilla? |
| 22:15.51 | dlynes_office | Mattwj2005: bugs.digium.com |
| 22:15.52 | Qwell[] | bugs.digium.com |
| 22:16.07 | dlynes_office | Mattwj2005: when you report the bug |
| 22:16.18 | dlynes_office | Mattwj2005: report it as not working on both 1.2.7.1 and 1.2.9.1 |
| 22:16.41 | dlynes_office | Mattwj2005: one sec, and i'll get the irc nick of the other individual that was having the problem |
| 22:17.44 | *** join/#asterisk chombe (i=chombe@213.137.115.223) |
| 22:17.46 | Mattwj2005 | okay will do |
| 22:18.29 | dlynes_office | Jun 16 09:43:40 <paolob> guys, any help with this error: " WARNING[13257]: chan_sip.c:1066 __sip_xmit: sip_xmit of 0x8143da8 (len 471) to 83.138.130.145:5060 returned -1: Bad file descriptor" ? thank you? It comes out continuosly, and doesn't permit asterisk to end the reloading process |
| 22:18.38 | dlynes_office | Paste that into your bug report, too |
| 22:18.42 | dlynes_office | And it's for 1.2.7.1 |
| 22:20.24 | dlynes_office | Mattwj2005: when does it start for you? |
| 22:20.31 | dlynes_office | Mattwj2005: when you read sip_notify.conf? |
| 22:21.59 | Mattwj2005 | I think when it starts app_externalivr.so |
| 22:22.16 | dlynes_office | ok, for him, he believes it was probably when he read sip_notify.conf |
| 22:22.25 | dlynes_office | So you might want to include that in the bug report, too |
| 22:22.39 | dlynes_office | That for you it seems to be app_externalivr.so and for him it seemed to be sip_notify.conf |
| 22:23.01 | dlynes_office | iow, when you're reporting a bug, just try to provide as much information as possible |
| 22:23.18 | dlynes_office | the more information you've got, the greater the chances of it getting fixed |
| 22:23.40 | dlynes_office | if you can provide a small example system that's able to reproduce the error over and over again, it's even more helpful |
| 22:24.04 | Mattwj2005 | [app_externalivr.so]Jun 16 17:21:16 WARNING[26921]: chan_sip.c:1066 __sip_xmit: sip_xmit of 0x8185320 (len 394) to xx.xx.xx.xx:yyyy returned -1: Bad file descriptor |
| 22:24.10 | Mattwj2005 | is the first time I see it |
| 22:24.27 | dlynes_office | In paolob's case, he was previously running with no problems...he just started running into the problem today |
| 22:27.48 | Mattwj2005 | what do you mean by a "sample system"? |
| 22:29.48 | dlynes_office | a simple extensions.conf file, sip.conf file...whatever the minimum configuration is you can set up, to reproduce the problem |
| 22:30.29 | dlynes_office | in your case, it's probably not possible, because you probably wouldn't know where to begin |
| 22:30.44 | dlynes_office | Just mentioning it, so you keep it mind for the future |
| 22:30.46 | *** join/#asterisk knarfly (n=Knarf980@c-69-180-98-189.hsd1.fl.comcast.net) |
| 22:30.53 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 22:31.09 | dlynes_office | If you've got a small test case, it's almost guaranteed the bug will get fixed fast |
| 22:31.37 | dlynes_office | If you don't have a small test case, it all depends on how well the troubleshooting team is able to track down the real problem |
| 22:31.48 | knarfly | I'm having trouble with my setup. Can anyone with an FWDNET account give me a test call? |
| 22:31.58 | Mattwj2005 | okay |
| 22:31.59 | *** join/#asterisk timscott (n=a@d198-53-23-18.abhsia.telus.net) |
| 22:32.15 | dlynes_office | ewwwwww |
| 22:32.16 | dlynes_office | telus |
| 22:32.31 | knarfly | Mattwj2005: Call me at 775806 |
| 22:32.41 | Mattwj2005 | sure |
| 22:33.12 | timscott | telus phj33r. |
| 22:33.19 | dlynes_office | heh |
| 22:33.34 | *** join/#asterisk znoG (n=gs@205-17-235-201.fibertel.com.ar) |
| 22:36.00 | CunningPike | What is the correct syntax for the number in Set(CALLERID(num) - is it just the straight number? |
| 22:36.13 | dlynes_office | yeah...no quotes |
| 22:36.25 | CunningPike | dlynes_office: And no <> either |
| 22:36.26 | dlynes_office | Set(CALLERID(num)=666) |
| 22:36.41 | carrar | Set(CALLERID(number)=2068675309 |
| 22:36.46 | CunningPike | dlynes_office: So why has it stopped working then :S |
| 22:36.55 | dlynes_office | CunningPike: and num and number will both work |
| 22:36.57 | carrar | ) |
| 22:37.09 | dlynes_office | CunningPike: no idea? |
| 22:37.18 | dlynes_office | CunningPike: let's see your extensions.conf line? |
| 22:37.23 | CunningPike | dlynes_office: Me either - I think our card is sending crap |
| 22:37.35 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 22:37.44 | dlynes_office | CunningPike: it's for an outgoing pri call? |
| 22:37.48 | carrar | is it scottish? |
| 22:38.00 | dlynes_office | carrar: no, it's irish |
| 22:38.03 | carrar | haha |
| 22:38.12 | dlynes_office | carrar: i'm quite serious |
| 22:38.27 | CunningPike | dlynes_office: Yes - same syntax works on one card and not on the other - and the trunk from asterisk -> nortel stop working this morning, too. I think the two are related |
| 22:38.47 | CunningPike | dlynes_office: I feel a restart coming on....... |
| 22:38.49 | dlynes_office | CunningPike: have you done a pri intense debug to see what q.931 is reporting? |
| 22:38.59 | dlynes_office | CunningPike: i.e. for the outgoing caller id? |
| 22:39.40 | CunningPike | dlynes_office: Yes - it displays the CID fine, but gets Protocol error back from the Nortel. The Nortel doesn't see any number come across so hangs up the call |
| 22:39.49 | CunningPike | dlynes_office: I think it's buggered |
| 22:40.00 | dlynes_office | maybe it's nortel that's buggered |
| 22:40.18 | CunningPike | dlynes_office: Luckily we have another trunk to the Nortel, so I switched the routing to that |
| 22:40.44 | dlynes_office | CunningPike: you guys all work 9-5 anyways |
| 22:41.03 | dlynes_office | CunningPike: just set up a cron schedule for 4am -> asterisk -rx "restart when convenient" |
| 22:41.07 | CunningPike | dlynes_office: Could be - but that in conjunction with our CID problems makes me wonder about the Zap card |
| 22:41.33 | CunningPike | dlynes_office: Yup - I'm hoping I don't have to reload zaptel :( |
| 22:41.39 | dlynes_office | hehe |
| 22:44.04 | dlynes_office | have you tried it on a sangoma yet? |
| 22:44.23 | CunningPike | dlynes_office: Nope |
| 22:44.32 | CunningPike | dlynes_office: Not going to, either |
| 22:44.35 | CunningPike | :D |
| 22:45.24 | Dr-Linux | what does this mean? >> chan_sip.c:6275 check_auth: stale nonce received from '128<sip:128@70.43.66.123>' |
| 22:45.30 | CunningPike | dlynes_office: This is right, right? exten => s,n(DNVTelus),Set(CALLERID(num)=604990${CALLERID(num)}) |
| 22:45.55 | dlynes_office | CunningPike: looks like it |
| 22:46.04 | CunningPike | dlynes_office: The CLI shows it up just fine |
| 22:46.12 | dlynes_office | CunningPike: try this: |
| 22:46.39 | CunningPike | dlynes_office: Invisible ink? |
| 22:46.52 | dlynes_office | CunningPike: exten => s,n(DNVTelus),Noop(${CALLERID(num)}) ; exten => s,n(DNVTelus),Set(CALLERID(num)=604990${CALLERID(num)}) |
| 22:47.05 | dlynes_office | that way you can see what it is before it tries calling it |
| 22:47.23 | CunningPike | dlynes_office: Ya - did that - it looks just fine |
| 22:47.33 | dlynes_office | and then maybe do a Noop(${CALLERID(num)}) after it as well |
| 22:47.40 | CunningPike | dlynes_office: That too :D |
| 22:47.44 | dlynes_office | ah |
| 22:47.49 | dlynes_office | then it's something funky with your pri |
| 22:48.10 | dlynes_office | q.931 is reporting the callerid as what it should be, right? |
| 22:49.58 | *** part/#asterisk angom_w (n=angom@red-corp-201.143.59.99.telnor.net) |
| 22:51.20 | dlynes_office | if q.931 isn't reporting it properly, it could be a problem with libpri or your driver |
| 22:51.27 | dlynes_office | more than likely libpri though |
| 22:54.43 | groogs | * keeps telling me "The previous reload command didn't finish yet" .. but it's been a few mins now.. is there a way to see what is keeping it? |
| 22:55.17 | dlynes_office | groogs: what's your verbose level set to? |
| 22:55.38 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 22:56.05 | groogs | dlynes_office: 9 |
| 22:56.14 | dlynes_office | and you typed 'reload'? |
| 22:56.25 | groogs | yep. now actually it does nothing |
| 22:56.42 | dlynes_office | did you issue a restart xxx xxxxxx previously? |
| 22:56.44 | groogs | in fact, no commands on CLI are responding |
| 22:57.32 | dlynes_office | groogs: did you? |
| 22:57.51 | groogs | no |
| 22:58.02 | dlynes_office | how about an moh reload? |
| 22:58.28 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 22:58.41 | dlynes_office | and what version of asterisk are you running? |
| 22:58.42 | groogs | just a sec, i'll see if i can reliably reproduce this (it's happened a couple times now, requiring a total restart of asterisk) |
| 22:59.05 | groogs | dlynes_office: lastest, downloaded+compiled an hour ago |
| 22:59.15 | dlynes_office | groogs: and what specifically is the latest? |
| 22:59.57 | Bullseye_Network | voicemail has been acting funny lately... It appears to be taking messages but the tmp file is not created and never copied over to the INBOX. It work fine for a while and then I have to RELOAD to get it working again. |
| 23:00.29 | dlynes_office | Bullseye_Network: just let it run its course...it's practicing to become a comedian |
| 23:00.37 | dlynes_office | Bullseye_Network: that's why it's called comedian voicemail system |
| 23:00.56 | groogs | 1.2.9.1 |
| 23:01.04 | Bullseye_Network | It worked fine until i upgraded to 1.2.9.1 |
| 23:01.43 | dlynes_office | i guess nobody on here has a sense of humor :) |
| 23:01.52 | Bullseye_Network | nope |
| 23:01.55 | Bullseye_Network | :P |
| 23:02.01 | CunningPike | ~lart dlynes_office |
| 23:02.11 | dlynes_office | ~kill CunningPike |
| 23:02.18 | jbot | ACTION shoots a super-inverse pseudophoton gun at CunningPike |
| 23:02.18 | CunningPike | Bastard |
| 23:02.29 | CunningPike | Haha - missed!! |
| 23:02.39 | Bullseye_Network | lol |
| 23:02.43 | dlynes_office | ~spam CunningPike |
| 23:02.44 | jbot | you should get a hotmail account to spam CunningPike |
| 23:03.17 | CunningPike | Bugger - twisted my ankle |
| 23:05.39 | thock | question |
| 23:05.45 | thock | for some reason, when i run ztcfg -vvv, i get 50 channels |
| 23:05.47 | thock | two T1 lines |
| 23:05.49 | thock | everything is fine |
| 23:05.49 | thock | but |
| 23:05.55 | thock | when trying to dial the main DID for the secondary span |
| 23:05.58 | thock | i get "the number is not in service" |
| 23:06.12 | thock | and when physically changing the ports on the a104d they're on, only the first span restarts |
| 23:06.14 | Qwell[] | 50 channels on 2 T1s? |
| 23:06.20 | Qwell[] | That's...like...not right |
| 23:06.26 | thock | yes i don't know |
| 23:06.31 | thock | 24x2 + 2 dchans? |
| 23:06.34 | Qwell[] | no.. |
| 23:06.41 | Qwell[] | 23*2 + 2 |
| 23:06.57 | Qwell[] | which is probably the problem ;) |
| 23:07.52 | Bullseye_Network | bchan=1-23 dchan=24 bchan=25-47 dchan=48 |
| 23:08.18 | thock | no |
| 23:08.21 | thock | to FXO analog ports |
| 23:08.24 | thock | a200 and a a104 |
| 23:08.25 | thock | sorry |
| 23:08.26 | thock | my mistake |
| 23:08.36 | Dandan | :/ |
| 23:09.00 | thock | really though, if you have 48 channels, but only half of them are being recognized, even though zt says they're all perfectly fine and running fine |
| 23:09.03 | thock | what could cause this? |
| 23:09.05 | thock | i'm stumped |
| 23:09.06 | Dandan | PRI Bearer capabilities: http://lists.digium.com/pipermail/asterisk-users/2006-June/155866.html |
| 23:09.09 | thock | even my telco's NOC is stumped |
| 23:09.18 | Dandan | can anyone help me? |
| 23:09.31 | Qwell[] | thock: probably screwed up your D channels somewhere.. |
| 23:09.50 | thock | hm |
| 23:09.59 | Qwell[] | 24 * 24 != 50 |
| 23:10.08 | Dandan | My NOC says that my asterisk sends wrong bearer capabilities |
| 23:10.13 | Qwell[] | heh...24 + 24 even |
| 23:10.15 | Dandan | and that's the reason we can't connect |
| 23:10.35 | thock | Qwell: 24+24+2 = 50 |
| 23:10.40 | Dandan | 24^2 != 24+24 :D |
| 23:10.50 | Qwell[] | thock: yeah...and a T1 doesn't HAVE 25 channels |
| 23:11.17 | Dandan | 2x E-1 = 60 though :) |
| 23:11.33 | thock | no, really? |
| 23:11.35 | thock | i know it doesn't |
| 23:11.40 | thock | 23 channels + one dchan |
| 23:11.42 | thock | right? |
| 23:11.44 | thock | or am i horribly wrong |
| 23:11.55 | Qwell[] | yes |
| 23:12.06 | Qwell[] | so where are you getting 50 from? |
| 23:12.13 | thock | you said "yes" |
| 23:12.24 | thock | as though i was reading your mind as to which that yes was the answer for |
| 23:12.31 | thock | yes i am correct or yes i am wrong |
| 23:12.36 | CunningPike | thock: A single PRI span is 23B+D = 24 channels total |
| 23:12.44 | thock | okay |
| 23:12.46 | thock | sorry |
| 23:12.53 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 23:12.54 | thock | i don't mean to be a dick like that, i'm just frustrated |
| 23:12.59 | thock | my telco's noc are a bunch of retards anyway |
| 23:13.41 | thock | signalling = fxs_ks |
| 23:13.43 | thock | group = 0 |
| 23:13.45 | thock | channel = 3-4 |
| 23:13.47 | thock | signalling = pri_cpe |
| 23:13.50 | thock | group = 1 |
| 23:13.51 | thock | channel => 25-47 |
| 23:13.53 | thock | signalling = pri_cpe |
| 23:13.54 | thock | i'm so very sorry, didn't mean to do that |
| 23:13.55 | thock | group = 2 |
| 23:13.56 | dlynes_office | ~pb |
| 23:13.59 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 23:13.59 | thock | channel => 49-71 |
| 23:14.03 | thock | yes i know |
| 23:14.05 | thock | i'm sorry |
| 23:14.08 | thock | had the wrong window up |
| 23:14.10 | thock | sorry :( |
| 23:14.28 | CunningPike | ~lart thock |
| 23:14.43 | dlynes_office | finally an appropriate lart :) |
| 23:14.47 | CunningPike | :D |
| 23:15.05 | CunningPike | What does 'lart' mean, anyway |
| 23:15.13 | thock | Live Action ReTards |
| 23:15.13 | dlynes_office | ~wiki lart |
| 23:15.14 | thock | ? |
| 23:15.30 | dlynes_office | luser it is :) |
| 23:15.35 | thock | ~lart jbot |
| 23:15.41 | *** part/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 23:15.43 | thock | :( |
| 23:15.46 | dlynes_office | thock: nice try :) |
| 23:15.49 | Bullseye_Network | lol |
| 23:15.59 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 23:16.05 | Bullseye_Network | Thats great |
| 23:16.11 | dlynes_office | now, that's funny :) |
| 23:16.25 | dlynes_office | glad to see jbot's got a sense of humor :) |
| 23:16.38 | CunningPike | My favorite jbot encounter was the chap who thought it was a real person |
| 23:16.47 | CunningPike | Started arguing with it |
| 23:16.47 | dlynes_office | are you serious? |
| 23:16.50 | Qwell[] | CunningPike: ...which time? |
| 23:17.00 | CunningPike | A few days ago - it was pretty funny |
| 23:17.15 | Bullseye_Network | ~laugh |
| 23:17.18 | jbot | ACTION rolls around on the floor laughing |
| 23:17.19 | Bullseye_Network | oh well |
| 23:17.22 | CunningPike | I eventually put him out of his misery |
| 23:17.25 | Bullseye_Network | oh there it is |
| 23:17.44 | *** join/#asterisk NewSole (n=dave@d226-105-226.home.cgocable.net) |
| 23:19.03 | thock | hey |
| 23:19.12 | thock | when you define spans in zaptel.conf |
| 23:19.18 | thock | and lets say that you have them configured as span 1 and span 2 |
| 23:19.21 | thock | would these two lines work? |
| 23:19.27 | thock | span = 1,1,0,esf,b8zs |
| 23:19.28 | thock | span = 2,2,0,esf,b8zs |
| 23:19.35 | CunningPike | thock: Yes |
| 23:19.37 | thock | or should the second be 2,1,0,esf,b8zs? |
| 23:19.43 | thock | because it's giving me an error when i run ztcfg |
| 23:19.54 | CunningPike | thock: No - only one span can be the primary timing source |
| 23:20.06 | CunningPike | thock: What error are you getting? |
| 23:20.10 | thock | so 2,2,0,esf,etc should be right/ |
| 23:20.15 | thock | 50 channels configured. |
| 23:20.15 | thock | ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 23:20.25 | thock | they were originally 2 and 3 |
| 23:20.37 | CunningPike | pastebin your zaptel and zapata.conf |
| 23:20.42 | thock | k |
| 23:20.42 | thock | sec |
| 23:21.35 | thock | http://pastebin.ca/66407 |
| 23:21.49 | CunningPike | dlynes_office: I think that what q.931 says it's sending and what it is actually sending are different |
| 23:22.31 | thock | that zaptel.conf is what's giving me the sludge in ztcfg -vvv |
| 23:22.52 | CunningPike | thock: OK - you need to put your channel defs for each span on separate lines - hang while I correct it |
| 23:22.56 | thock | sure |
| 23:22.58 | thock | Thanks :D |
| 23:23.02 | CunningPike | s/hang/hang on/ |
| 23:23.42 | NewSole | Ok... Last Chance.... VegaStream 400 with 4 PRI T1/E1 up for grabs $3000.00 (BRAND NEW) |
| 23:24.01 | dlynes_office | CunningPike: you have any good suggestions for getting rid of crack addicts? |
| 23:24.23 | dlynes_office | CunningPike: had a couple of those losers smoking a pipe outside our back door this morning |
| 23:24.54 | CunningPike | dlynes_office: Ask jbot |
| 23:25.01 | NewSole | http://www.voipsupply.com/product_info.php?products_id=465 |
| 23:25.20 | dlynes_office | jbot: crack addicts? |
| 23:25.29 | jbot | ACTION cracks addicts 's rock in half and then proceeds to shatter the halves into tiny pieces |
| 23:25.54 | *** join/#asterisk Mattwj2005 (n=Matt@user-12l3n74.cable.mindspring.com) |
| 23:26.01 | Bullseye_Network | wouldnt fxsks start 1-2 then the Ts 3-50 and 52-?? |
| 23:26.12 | Mattwj2005 | I submitted the bug report |
| 23:26.14 | thock | CunningPike: when you said to define the channels for each span on separate lines, did you mean going "bchan = 25-47" then the dchan of 48, \n "bchan = 49-71" then the dchan for 72? |
| 23:26.24 | thock | Bullseye_Network: the A200 has the FXO card installed in the wrong slot |
| 23:26.24 | thock | hehe |
| 23:26.27 | thock | that's why |
| 23:26.34 | CunningPike | thock: http://pastebin.ca/66408 |
| 23:26.59 | thock | you only changed zaptel, yes? |
| 23:27.14 | CunningPike | thock: Yes - your zapata looks OK at first glance |
| 23:27.17 | Bullseye_Network | but asterisk would still see it as 1 and 2 |
| 23:28.09 | dlynes_office | jbot: crack pipe? |
| 23:28.11 | jbot | ACTION cracks pipe 's rock in half and then proceeds to shatter the halves into tiny pieces |
| 23:28.26 | thock | okay |
| 23:28.27 | thock | i'll try this |
| 23:28.28 | thock | thanks |
| 23:28.43 | thock | same error |
| 23:28.48 | thock | should i reboot or restart anything? |
| 23:28.54 | dlynes_office | jbot: bong? |
| 23:28.55 | jbot | methinks bong is a device for the consumption of drugs. Normally cannabis. The cannabis is burned on a wire mesh, and the resulting smoke is bubbled through water, to cool it and remove many harsh elements, before it is inhaled by the drug 'user.' |
| 23:29.40 | Bullseye_Network | jbot: linux? |
| 23:29.42 | jbot | i guess linux is a steaming pile of groganpaste |
| 23:29.44 | thock | failed on span 1 again |
| 23:29.48 | dlynes_office | lol |
| 23:30.12 | dlynes_office | Mattwj2005: what's the bug number? |
| 23:30.41 | Mattwj2005 | 0007381 |
| 23:31.05 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 23:31.14 | CunningPike | thock: What's the error message? |
| 23:31.22 | CunningPike | thock: Did you restart zaptel? |
| 23:31.26 | thock | hm |
| 23:31.30 | Mattwj2005 | Jun 16 18:22:05 WARNING[4860]: chan_sip.c:1066 __sip_xmit: sip_xmit of 0x81539d0 (len 477) to xx.xx.xx.xx:yyyy returned -1: Bad file descriptor |
| 23:31.30 | Mattwj2005 | Jun 16 18:22:05 NOTICE[4860]: chan_sip.c:11396 sip_poke_noanswer: Peer 'telasip-gw' is now UNREACHABLE! Last qualify: 0 |
| 23:31.30 | Mattwj2005 | Jun 16 18:22:05 WARNING[4860]: chan_sip.c:1066 __sip_xmit: sip_xmit of0x8166300 (len 395) to xx.xx.xx.xx:yyyy returned -1: Bad file descriptor |
| 23:31.32 | thock | I restarted wanrouter |
| 23:31.38 | thock | which i think restarts zaptel |
| 23:31.44 | thock | not sure how to restart zaptel manually either.. really. |
| 23:31.45 | thock | But |
| 23:31.47 | *** join/#asterisk CoffeeIV (i=rgr@cpe-70-112-100-20.austin.res.rr.com) |
| 23:31.52 | thock | the same error as before, ZT_CHANCONF |
| 23:31.58 | thock | failed on span 1 |
| 23:32.02 | thock | invalid arg |
| 23:32.05 | thock | but |
| 23:32.06 | thock | if i change it to |
| 23:32.21 | thock | http://pastebin.ca/66409 |
| 23:32.25 | thock | ztcfg shuts up and works |
| 23:32.31 | thock | which is strange |
| 23:32.32 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 23:32.36 | thock | and actually not tha tstrange now that i think of it |
| 23:32.46 | thock | becuase analog is first, T1/E1's are secondary with sangoma hardware |
| 23:32.46 | thock | like |
| 23:32.51 | thock | you have to define the FXO/FXS stuff first |
| 23:32.56 | thock | (hence span 1) |
| 23:32.59 | thock | then span 2 and 3 are the T1's |
| 23:32.59 | CoffeeIV | anybody use www.ipcomms.net with Asterisk ? they advertise IAX -- does it work OK ? |
| 23:33.01 | thock | so that makes some sense |
| 23:33.14 | thock | any idea what the first three numbers are in that equasion? |
| 23:33.19 | thock | span number, something, something? |
| 23:33.37 | CunningPike | thock: Span, timing source priority, line length |
| 23:33.51 | thock | Length? strange. |
| 23:33.58 | thock | Is it supposed to be 0, then? |
| 23:34.01 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 23:34.12 | thock | hm |
| 23:34.15 | CunningPike | thock: It's a buildout number - 0 covers up to 133 feet |
| 23:34.15 | thock | i wonder if that timing is the problem |
| 23:34.21 | thock | ah |
| 23:34.28 | thock | so the second number, timing source |
| 23:34.35 | thock | setting to 1 i'm assuming is hardware based, yes? |
| 23:34.37 | Bullseye_Network | rmmod zaptelhttp://pastebin.ca/66411 |
| 23:34.39 | Bullseye_Network | ops |
| 23:34.40 | Bullseye_Network | http://pastebin.ca/66411 |
| 23:34.48 | Bullseye_Network | thats my zaptel.conf at one location |
| 23:34.48 | riddlebox | [TK]D-Fender, you got a minute? |
| 23:34.50 | Bullseye_Network | its just 1 T1 |
| 23:34.57 | Bullseye_Network | but I have the Ts first |
| 23:34.57 | CunningPike | thock: You have some other hardware? FXS stuff? |
| 23:35.09 | thock | I've got one A200 which is an FXO card with 2 channels |
| 23:35.17 | thock | and a single A104D which has 4 T1/E1 slots |
| 23:35.23 | thock | two of which are used by T1's, a PRI and an LDT1 |
| 23:35.49 | thock | damn, no manpages for zaptel |
| 23:36.03 | dlynes_office | wow...down to 197 bugs |
| 23:36.20 | CunningPike | thock: I'm not sure how to integrate FXS stuff with PRI stuff in zaptel - maybe someone else knows... |
| 23:36.39 | CunningPike | If the way you have it now works, run with it |
| 23:37.04 | thock | that's the thing |
| 23:37.06 | thock | it doesn't work :P |
| 23:37.07 | NewSole | Ok... Last Chance.... VegaStream 400 with 4 PRI T1/E1 up for grabs $3000.00 (BRAND NEW SHOWN ON) http://www.voipsupply.com/product_info.php?products_id=465 |
| 23:37.08 | thock | it starts up |
| 23:37.08 | Mattwj2005 | hey question about submitting a bug ticket |
| 23:37.15 | thock | but the second LDT1 doesn't accept incoming/outgoing calls |
| 23:37.20 | thock | Thanks for the help mang |
| 23:37.22 | thock | appricated :) |
| 23:37.51 | CunningPike | thock: OK - np |
| 23:37.58 | dlynes_office | woah |
| 23:37.58 | dlynes_office | cool |
| 23:38.08 | dlynes_office | g.722 passthrough is in |
| 23:38.26 | Mattwj2005 | will they let me know what they find? Also, are they any free prizes....you know like a free copy of the source code or something :-P |
| 23:38.38 | dlynes_office | heh |
| 23:38.46 | dlynes_office | Mattwj2005: Just put a monitor on the bug |
| 23:38.55 | dlynes_office | Mattwj2005: then any time the status changes, you'll be emailed |
| 23:39.10 | dlynes_office | including whenever someone adds a comment |
| 23:39.38 | riddlebox | can someone explain to me how to setup when a mailbox greeting is played you can press * or # to acess VoicemailMain()? |
| 23:40.25 | Mattwj2005 | that is cool...thanks |
| 23:40.46 | dlynes_office | cool |
| 23:40.50 | thock | riddle |
| 23:40.52 | thock | read the wiki |
| 23:40.55 | dlynes_office | g.722 passthrough support is slated for 1.4 |
| 23:40.56 | thock | it's pretty detailed. |
| 23:41.03 | thock | ~wiki |
| 23:41.38 | *** join/#asterisk flynux (n=flynux@2a01:38:0:0:0:0:0:1) |
| 23:42.08 | *** join/#asterisk MissNeBuN (n=gabe@user-12ld5f7.cable.mindspring.com) |
| 23:42.17 | MissNeBuN | hi |
| 23:42.30 | MissNeBuN | when dial *77 i can`t record |
| 23:42.58 | MissNeBuN | or if I deal *65 or *99 or *65 |
| 23:43.06 | MissNeBuN | what can be the problem ? |
| 23:45.02 | eipi | why could be (my mistake) that * is not resolving a peer correctly, i do "sip show peers" and it shows my externip and no the peer ip |
| 23:45.21 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 23:45.21 | *** mode/#asterisk [+o russellb] by ChanServ |
| 23:48.34 | eipi | and status is UNREACHABLE |
| 23:53.16 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 23:55.41 | riddlebox | thock, I got it, just had to read further into it |
| 23:57.59 | thock | riddlebox: cool |
