00:00.24 | *** join/#asterisk tzafrir (~tzafrir@62.90.10.53) [NETSPLIT VICTIM] |
00:00.31 | *** join/#asterisk aggelos (~aggelos@egate.eleven.de) [NETSPLIT VICTIM] |
00:00.40 | *** join/#asterisk dg1nsw (~schulte@gate.sympat.de) [NETSPLIT VICTIM] |
00:02.29 | tuxinator_linux | brc__: cool video's |
00:02.47 | brc__ | PTG123, eh? |
00:02.57 | brc__ | exten is a channel variable |
00:03.03 | *** join/#asterisk BigCanOfTuna (~chatzilla@dsl-macn-66-18-205-30-cgy.nucleus.com) |
00:03.04 | brc__ | so it'll stay with the channel wherever it goes |
00:04.04 | BigCanOfTuna | Is there an API for asterisk that will allow me to issue commands to it via another programming language such as Python or Ruby? |
00:04.16 | brc__ | well there's a long answer |
00:04.18 | brc__ | and a short answer |
00:04.57 | hermie | ...and the short answer is AGI |
00:05.06 | brc__ | the long answer is it depends |
00:05.17 | brc__ | on what you mean by 'issue commands' |
00:05.35 | harryvv | hi brc |
00:05.36 | hermie | I was thinking the long answer was "pick a better programming language" :-) |
00:05.41 | brc__ | greets harry |
00:05.59 | brc__ | ruby rulez |
00:06.07 | mrgoby | yeah, like php |
00:06.11 | brc__ | EWW |
00:06.18 | hermie | mrgoby: no, that'd be even worse |
00:06.31 | mrgoby | :-) |
00:06.52 | hermie | at least python and ruby were meant to be run |
00:06.52 | brc__ | keep that where it belongs, short web scripts |
00:06.52 | brc__ | anyway |
00:06.52 | brc__ | BigCanOfTuna, did you disappear? |
00:06.55 | BigCanOfTuna | brc_: sorry, in the background. |
00:07.03 | *** part/#asterisk GaryH (~ghawkins@gromit.garysoft.co.uk) |
00:07.08 | brc__ | if you'd like to give a short description of what you want to do maybe somebody can answer the question |
00:07.28 | harryvv | brc, I had a bad experaince last night...hard drive was cycling over and over last night..ie failing. It was my asterisk system. |
00:07.28 | brc__ | my as in home? or production somewhere |
00:07.43 | harryvv | it was my test box running our phones. |
00:07.49 | brc__ | ah |
00:07.49 | BigCanOfTuna | I am planning on creating a web interface to issue command to my asterisk server. |
00:07.52 | harryvv | luckily it was internal. |
00:07.53 | brc__ | had backups? |
00:08.02 | tuxinator_linux | harryvv: I had what I thought was a hard drive problem but it was that my TDP400P was sharing and IRQ |
00:08.07 | harryvv | yes, have the confs on the windows machine but not current. |
00:08.07 | brc__ | BigCanOfTuna, gotta be more specific buddy |
00:08.27 | brc__ | manager will *probably* do what you want |
00:08.29 | mrgoby | uuuum |
00:08.43 | BigCanOfTuna | brc__: I would like to invoke certain tasks, such as giving it a number, and asterisk calls it. |
00:08.44 | mrgoby | any reasy why my asterisk seg faults when i have an mp3 in my mohmp3 dir ? |
00:08.44 | tuxinator_linux | harryvv: What do you mean by cycling? |
00:08.54 | brc__ | you can list channels, redirect em, create em, etc |
00:09.03 | harryvv | tux, no this is possibly a hard drive failure. Got this message "hda :dma timer_expiry dma stats = 0x21 I tried to fsck the drive and did not work. |
00:09.09 | mrgoby | s/reasy/reason/g |
00:09.22 | tuxinator_linux | harryvv: I got a very simular error |
00:09.26 | BigCanOfTuna | brc__: Isn't AGI asterisk calling external scripts? |
00:09.34 | tuxinator_linux | harryvv: except mine was 0X24 |
00:09.34 | brc__ | yes, agi is not what you want |
00:09.37 | mrgoby | mpg123 must be screwed up |
00:09.39 | *** join/#asterisk dcb (~dcb@CPE-60-231-180-246.vic.bigpond.net.au) |
00:09.42 | harryvv | tux, as if it was rebooting over and over but the system was running..my wife was on the phone with voip running but the hard drive kept cycling. |
00:09.43 | BigCanOfTuna | brc__: I want external scripts to call the asterisk api. |
00:09.47 | brc__ | there is a bit of info on the manager interface on the wiki, ti's what you want |
00:09.48 | mrgoby | i bet they used the default gentoo moh |
00:10.06 | BigCanOfTuna | brc__: thanks. |
00:10.25 | brc__ | well you can't exactly call the asterisk api unfortuantly.. |
00:10.30 | *** part/#asterisk dcb (~dcb@CPE-60-231-180-246.vic.bigpond.net.au) |
00:10.35 | mrgoby | i saw there is a make target for mpg123 now... is that recommended ? |
00:10.46 | brc__ | yes |
00:10.57 | brc__ | *highly* |
00:11.03 | mrgoby | does it install itself ? |
00:11.06 | brc__ | yes |
05:12.03 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
05:12.03 | *** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.6 Released || Dev Conf 1PM CST MARCH 10th -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf Aug 3rd - 5th |
05:12.05 | brc_ | like he said, for the fun of it |
05:12.19 | holycow | ehe :) fair enough |
05:12.57 | Tr0j4N | I guess Asterisk doesn't like it when you service zaptel stop while it's running. Is a Kernel panic bad?? |
05:13.03 | sivana | in a bottle?.... Brilliant! |
05:13.41 | MikeJ[Jayden] | so I can say... asterisk runs on windows, some other guy did it, I just was testing... it has some serious issues with fork, cuz of the way cygwin does it |
05:13.47 | rhollan | OT: Grr... wife wants me to go get groceries.... She should be grateful I made Drunkard Noodles for dinner. |
05:13.48 | brc_ | UNPOSSIBLE! |
05:13.51 | MikeJ[Jayden] | but you can make calls |
05:13.58 | holycow | oh so it doesn't run on windows |
05:14.03 | holycow | it runs on cygwin |
05:14.04 | brc_ | yes it does |
05:14.04 | holycow | *nod* |
05:14.18 | brc_ | that is different from using colinux |
05:14.27 | holycow | not by much |
05:14.34 | geekster | can anyone help me out with an extension issue that im having ? |
05:14.35 | MikeJ[Jayden] | sure it is. |
05:14.45 | brc_ | yes it is holycow |
05:15.02 | rhollan | CYGWIN is a commpatibility layer over Win32, IIRC, not not a virtual machine emulator. |
05:18.28 | erwinism | MikeJ[Jayden] i cant find manual how to add a user on the links you said |
05:18.39 | *** join/#asterisk pimpwell (~pimpwell@ool-44c6ab45.dyn.optonline.net) |
05:18.52 | brc_ | erwinism, a 'user'? |
05:19.07 | brc_ | ~asterisk docs project |
05:19.11 | brc_ | ~asterisk docs |
05:19.13 | jbot | rumour has it, asterisk documentation project is at http://asteriskdocs.org |
05:19.22 | brc_ | read |
05:19.43 | *** join/#asterisk Damin_Mobile (~pocketirc@252.sub-70-214-4.myvzw.com) |
05:19.50 | erwinism | brc ok |
05:19.58 | pimpwell | good evening, was wondering about the .call file a little bit and in my situation what are my choices. I place the .call file in the outbound directory through FTP, anyway to program the .call file to somehow get me a response back on the calls status wether its busy, no one answered or terminated successfully? |
05:20.18 | Damin_Mobile | The eagle has landed! |
05:20.28 | pimpwell | the asterisk server is not local to me |
05:20.35 | Damin_Mobile | Where is the beer? |
05:20.47 | pimpwell | I got 2 22oz of corona in the fridge |
05:20.56 | pimpwell | take em :o |
05:20.57 | Tr0j4N | when i run service zaptel start it tells me kobject_register failed for t1xxp (-17) |
05:22.41 | pimpwell | can the .call file do anything special like send an email or something for return report |
05:22.42 | rhollan | I've got some Pyramid IPA and Foster's (damn wife.... ugh) in me fridge. :-) |
05:23.42 | MikeJ[Jayden] | damin.. you in the bathroom on IRC again :) |
05:24.21 | pimpwell | can I treat the .call file like PERL? |
05:24.40 | pimpwell | and add a conditional statement in there or things of that nature? |
05:26.31 | erwinism | brc, im sorry i couldnt find any articles ont he http://asteriskdocs.org related to adding a user. maybe you could direct me to the link. |
05:26.31 | JonR800 | pimpwell, my suggestion.. use the manager API or use an AGI |
05:27.10 | Damin_Mobile | Mikej; no in a plane |
05:27.14 | JonR800 | pimpwell: either use the call file to connect the user to the AGI... or user the manager API to initiate the calll |
05:27.33 | pimpwell | the box isnt on my end |
05:27.34 | MikeJ[Jayden] | hdhd |
05:27.37 | MikeJ[Jayden] | hehe |
05:27.37 | pimpwell | it belongs to someone else |
05:27.46 | JonR800 | k, then use the AGI method |
05:27.50 | pimpwell | what I need is to FTP my .call file to their box |
05:28.10 | pimpwell | and be able to check status on each call |
05:28.12 | scythelx | pimpwell: well... you wont get any response from asterisk on the status of the call if you do that |
05:28.29 | pimpwell | everycall gets logged right? |
05:28.48 | pimpwell | I figure through CDR |
05:28.50 | scythelx | pimpwell: yea.. but you said you dont have access to the box.. unless you wanted to ftp the log files back and parse them |
05:29.02 | pimpwell | ya, is it 1 log file? |
05:29.05 | pimpwell | 1 giant one |
05:29.10 | pimpwell | or 1 for each call |
05:29.17 | scythelx | 1 giant one |
05:29.27 | JonR800 | depends how you set it up.. |
05:29.32 | holycow | i'll be off for a while, i just dropped by to give a shout out to any asterisk devs that might around |
05:29.34 | scythelx | y does it have to be ftp anyways |
05:29.38 | holycow | and anyone else in the community |
05:29.40 | *** join/#asterisk The_Ball (~alex@dsl-73.131.240.220.lns02-wick-bne.dsl.comindico.com.au) |
05:29.45 | *** join/#asterisk hemant (hemant@220.226.49.235) |
05:29.46 | holycow | :) this is one rocking tool |
05:29.49 | The_Ball | what port's need to be open for iax2? |
05:29.58 | JerJer[mobile] | udp port 4569 |
05:30.08 | pimpwell | ftp because it's not my box, I just make the calls. |
05:30.34 | JonR800 | what exactly are you trying to accomplish? why do you need a status? |
05:30.44 | pimpwell | similar to wake up calls |
05:30.51 | pimpwell | my users need to know if the calls went out |
05:30.55 | pimpwell | and how many more are left |
05:31.04 | pimpwell | they just cant click a button and be like okay, im done. |
05:31.07 | scythelx | well.. just have the owner of the box give you a manager interface login and your set... |
05:31.23 | pimpwell | I am doign this all throuhg php |
05:31.26 | JonR800 | this is a programming problem.. not an asterisk problem. |
05:31.27 | scythelx | so.. |
05:31.33 | pimpwell | jon: its both man |
05:31.37 | JonR800 | not really |
05:31.40 | The_Ball | JerJer[mobile], ah! udp, no worries |
05:31.53 | scythelx | google -> phpagi |
05:31.56 | JonR800 | the mechanisms to do this are there on the asterisk end.. you just need to tie it together |
05:32.13 | pimpwell | I was just thinking the .call file is more then just a file |
05:32.15 | JerJer[mobile] | looks like asterisk made slashdot again |
05:32.33 | MikeJ[Jayden] | link? |
05:32.42 | scythelx | nah it wont parse dynamic arguments as far as i know |
05:32.44 | JonR800 | pimpwell: nope |
05:33.02 | scythelx | your best bet is the manager interface... seriously |
05:33.12 | JerJer[mobile] | www.slashdot.org |
05:33.16 | MikeJ[Jayden] | http://it.slashdot.org/it/05/03/06/1945210.shtml?tid=126&tid=218 |
05:33.22 | MikeJ[Jayden] | thanks....smartass |
05:35.35 | *** join/#asterisk ethzer0 (~ethzer0@d141-238-51.home.cgocable.net) |
05:36.17 | scythelx | pimpwell: |
05:36.20 | scythelx | oops |
05:36.23 | scythelx | pimpwell: http://www.voip-info.org/wiki-Asterisk+tips+Wake-Up+Call+PHP |
05:36.36 | scythelx | its alreadly built for you.. just get a manager login... managers.conf |
05:38.13 | JonR800 | i wrote the ugly perl portion of that |
05:38.25 | pimpwell | I know I read that |
05:38.30 | pimpwell | but, I dont have the bandwidth |
05:38.33 | pimpwell | I need to use someone elses box |
05:38.36 | The_Ball | is the bindaddr= nessesary in iax.conf? |
05:38.50 | scythelx | pimpwell: it does.. thru the manager interface... its like a telnet session.... |
05:38.52 | pimpwell | second, I am not doing wake up calls, it's just something my system could do |
05:38.56 | JonR800 | you don't need the manager interface |
05:39.01 | MikeJ[Jayden] | ball, only if you need it |
05:39.02 | JonR800 | ahh |
05:39.26 | pimpwell | I will be using php to run my system, telnetting in wont help me |
05:39.37 | rhollan | well, i gotta go... later ... been fun |
05:39.41 | JonR800 | the wake up call app uses call files |
05:39.45 | pimpwell | I wont be sitting there checking the status of everycall manually |
05:39.45 | *** part/#asterisk hemant (hemant@220.226.49.235) |
05:39.49 | JonR800 | and what would you need bandwidth for pimpwell? |
05:40.01 | pimpwell | for 6:00am when 100 people want to wake up |
05:40.02 | scythelx | JonR800: ah thats right |
05:40.06 | pimpwell | at 6:00am |
05:40.09 | joshua_ | OK thanks guys. |
05:40.10 | *** part/#asterisk joshua_ (joshua@cl-303.ams-04.nl.sixxs.net) |
05:40.55 | *** join/#asterisk j0 (~dan@S010600105a04ed8d.va.shawcable.net) |
05:41.00 | JonR800 | pimpwell: i don't get it.. why would that use bandwidth.. this box is local to where you're offering the services right? |
05:41.07 | JonR800 | you just wanted to remotely monitor |
05:41.10 | pimpwell | no, it isn't |
05:41.13 | JonR800 | oh |
05:41.19 | pimpwell | it's a remote box, I FTP the .call file to |
05:41.28 | pimpwell | drop it in the outbound. |
05:41.28 | JerJer[mobile] | ugg |
05:41.36 | JerJer[mobile] | talk about insecure |
05:41.38 | JonR800 | sorry i thought it was on a customer's premises. |
05:41.43 | brc_ | SFTP |
05:41.44 | JerJer[mobile] | and bloated |
05:41.48 | JerJer[mobile] | SCP |
05:41.56 | pimpwell | talk about people trying to rip me off for bandwidth usage |
05:42.01 | brc_ | ssh+ftp |
05:42.05 | erwinism | do i have to enable the RTP port range on my firewall? |
05:42.45 | pimpwell | 30 calls simultaneously, 10$ a month 1.2cents a minute |
05:42.47 | JerJer[mobile] | depends on your firewall config |
05:42.51 | pimpwell | is what Im paying. |
05:43.01 | JerJer[mobile] | pimpwell: you are getting cornholed |
05:43.04 | pimpwell | ahaha |
05:43.10 | pimpwell | not really. |
05:43.14 | erwinism | JerJer[mobile] how can i enable it? |
05:43.25 | scythelx | pimpwell: yea u are... |
05:43.26 | Damin_Mobile | JerJer: You arent in san jose. |
05:44.13 | pimpwell | he will also hold my sound files |
05:44.32 | scythelx | he has to if your ftping your .call files in |
05:44.36 | pimpwell | so I dont have to collocate for 150 a month or whatever a certain someone in here was going to charge me to throw my box |
05:44.38 | MikeJ[Jayden] | pimpwell.. I'll hold your sound files :) |
05:44.58 | JonR800 | i'll hold something else. |
05:45.14 | pimpwell | <enter dick joke here> |
05:45.41 | JonR800 | i thought i had that covered.. |
05:45.41 | MikeJ[Jayden] | speaking of moose jokes, where is bkw |
05:45.41 | JonR800 | doh |
05:45.56 | brc_ | von |
05:46.06 | JerJer[mobile] | Damin_Mobile: no |
05:46.07 | MikeJ[Jayden] | y, I know... |
05:46.32 | JerJer[mobile] | pimpwell: if you just want to make a phone call, no you don't have to collocate a box |
05:46.42 | pimpwell | I dont want to make a phone call |
05:46.50 | pimpwell | I have a system, a web based system |
05:46.58 | JerJer[mobile] | ok and the problem is? |
05:47.15 | MikeJ[Jayden] | well... if you don't want to make phone calls, then this is probably the wrong place to be chatting :) |
05:47.17 | pimpwell | using a remote asterisk box makes checkign the status of each call a pain |
05:47.28 | scythelx | he wants to ftp .call files in and get the status of the call.. but doesnt want to use the manger interfcae because hes concerned about bandwidth? |
05:47.50 | JonR800 | have a script on the server side monitor it for you |
05:48.04 | pimpwell | scythelx: manager interface requires manually login? |
05:48.13 | *** join/#asterisk potter (~hq28@202.58.252.14) |
05:48.34 | MikeJ[Jayden] | you could progamatically communicate w/ manager, it is just text |
05:48.36 | scythelx | pimpwell: php script will log in for you, send the dial commands, and return the status |
05:48.42 | scythelx | all back to your web based system |
05:48.44 | JonR800 | MikeJ[Jayden]: bingo. |
05:48.44 | JerJer[mobile] | pimpwell: what is the wrong with the manager interface? |
05:48.57 | MikeJ[Jayden] | B-I-N-G-O :) |
05:49.01 | pimpwell | hmm |
05:49.06 | newpers | how many simultaneous phone calls can asterisk handle? or is that dependent upon the machine specs? |
05:49.06 | scythelx | pimp: its like a telnet interface its all text |
05:49.08 | JerJer[mobile] | newpers: all of them |
05:49.15 | MikeJ[Jayden] | JerJer... many things, but that is a diff story :) |
05:49.18 | harryvv | man installing asterisk on a FC3 system is being a pain. |
05:49.28 | harryvv | when its a 64 bit system |
05:49.36 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
05:49.37 | JerJer[mobile] | 64 bit system won't help asterisk |
05:49.49 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
05:49.54 | harryvv | jerjer well i know but its there and thats what my system is. |
05:50.06 | JerJer[mobile] | get a real system then |
05:50.18 | harryvv | jer, its a opteron system as real as it gets. |
05:50.21 | JerJer[mobile] | um no |
05:50.43 | harryvv | um yes for a graphics workstation it is. My asterisk hd on the other system failed last night. |
05:50.43 | Primer | opteron > *, except when your shit's not too 64 bit friendly |
05:50.55 | Primer | in which case, just boot a 32 bit kernel and it's still a very fast box |
05:51.01 | scythelx | pimpwell: http://lists.digium.com/pipermail/asterisk-users/2003-November/025595.html |
05:51.21 | scythelx | pimpwell: example php script to connect, then u send your dial commands |
05:51.46 | harryvv | anyone by chance know what rpm restorecon command belongs to. |
05:51.53 | pimpwell | all I know is if my business works and asterisk makes my life easier I am donating so fukin much to whoever runs and maintains this beast |
05:52.07 | pimpwell | would that be digium? |
05:52.14 | JonR800 | that'd be me |
05:52.20 | scythelx | me too :) |
05:52.30 | pimpwell | john you cant hold my money if your already holding my.. |
05:52.34 | pimpwell | :o |
05:52.45 | pimpwell | got your hands full there |
05:52.46 | JonR800 | i have two hands |
05:52.50 | pimpwell | ahah |
05:53.00 | JonR800 | just leave the money on the dresser then |
05:53.27 | JerJer[mobile] | pimpwell: its your system, why should someone else run and mantain it? |
05:53.40 | pimpwell | no, I mean the asterisk community |
05:53.45 | pimpwell | it has to have a donate button somewhere |
05:53.55 | pimpwell | most open source do |
05:54.10 | JonR800 | www.digium.com.. you could buy some hardware or support packages |
05:54.31 | pimpwell | once I get enough customers I get my own lines |
05:54.36 | pimpwell | and I will need the hardware |
05:54.44 | scythelx | pimpwell: buy us all some g729 licenses |
05:55.11 | JonR800 | buy me a T1 |
05:55.56 | mrgoby | is there a way to grab a channel sitting on a waitforresponse and connect it to a meetme ? i want to be able to have someone holding outside a conf basically and transfered in exactly when i want |
05:56.04 | pimpwell | probably need to get one of you to consult me once that happens |
05:56.11 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
05:56.25 | Sedorox | Does asterisk have a problem with SMP boxes? |
05:56.30 | JerJer[mobile] | Sedorox: no |
05:56.41 | Sedorox | hmmm |
05:56.56 | Sedorox | I have a FBSD5.3 box... SMP... and about 5 days uptime for asterisk |
05:56.58 | JerJer[mobile] | mrgoby: waitforresponse? |
05:57.01 | Sedorox | I do a "stop now" |
05:57.05 | Sedorox | it locks the entire box |
05:57.09 | JerJer[mobile] | run Linux then |
05:57.34 | Sedorox | can't.. its a colo box... hard to get to it.. and the person who owns the box prefers fbsd |
05:57.42 | JerJer[mobile] | then get your own box |
05:57.45 | JerJer[mobile] | and run Linux |
05:58.05 | mrgoby | hehe |
05:58.11 | Sedorox | hrm |
05:58.21 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
05:58.21 | JonR800 | im sure that's just what you want to hear Sedorox |
05:58.59 | Sedorox | well I know fbsd isn't 'supported'.. but I just mainly wanna know if anyone else has run into problems like this.. |
05:59.08 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
05:59.12 | MikeJ[Jayden] | JerJer.. you know anyone he can colo a linux * box w/ ? |
05:59.31 | Sedorox | I know 2 places I can colo cheap at... |
05:59.33 | JerJer[mobile] | sure |
05:59.36 | Sedorox | but that isn't the point |
05:59.39 | MikeJ[Jayden] | hehe |
05:59.42 | Sedorox | :-p |
05:59.44 | JonR800 | Sedorox: another fine answer, have the box owner boot it into a non SMP kernel :-P |
05:59.51 | mrgoby | will Manager command redirect work for this ? |
06:00.00 | MikeJ[Jayden] | ok.. gnight all |
06:00.01 | Sedorox | the box owner sits 3 hrs away from it :-p |
06:00.04 | Sedorox | night |
06:00.04 | JerJer[mobile] | unlike others, I don't advertize for myself |
06:00.15 | MikeJ[Jayden] | I know.. |
06:00.16 | mrgoby | nufone is great ! |
06:00.21 | MikeJ[Jayden] | hey does tho |
06:00.22 | mrgoby | there, i did it for you jerjer |
06:00.24 | MikeJ[Jayden] | he does |
06:01.13 | mrgoby | will redirect work for this? can i have someone sitting on a waitforresponse, then redirect to a conference room ? basically allowing me to put them in the conf whenever i want ? |
06:01.20 | MikeJ[Jayden] | JerJer... what building is your southfield data center in? If it is in the galeria, I may have a customer for you |
06:01.38 | JerJer[mobile] | not in galeria |
06:01.42 | MikeJ[Jayden] | k |
06:01.53 | MikeJ[Jayden] | ok.. nighty time. |
06:02.28 | mrgoby | sorry, waitforresponse == responsetimeout |
06:02.44 | JerJer[mobile] | mrgoby: write an app |
06:03.01 | mrgoby | an app that transfers the call ? |
06:03.23 | mrgoby | i wont need to write one if i can do it through the management interface |
06:03.50 | JerJer[mobile] | just sittting in a responsetimeout is very hackish |
06:04.17 | mrgoby | well, it is for an art project and not a production system :0) so hacks are okay in this case |
06:04.33 | JerJer[mobile] | so like are you trying to make certain users wait until a 'conference moderator' joins? |
06:04.59 | erwinism | hello, what does this means? NOTICE[11825]: chan_sip.c:7681 handle_request: Registration from 'xlite <sip:111@192.168.2.200>' failed for '192.168.2.105' |
06:05.11 | JerJer[mobile] | means registration failed |
06:05.17 | mrgoby | basically.... i'm making it to where people move around in contexts, but if one person finds a certain extension, then all others get pulled into the meetme with them |
06:05.19 | erwinism | JerJer[mobile] how can i fix this? |
06:05.37 | JerJer[mobile] | erwinism: send the proper username and secret |
06:05.56 | JerJer[mobile] | mrgoby: mmkay |
06:06.12 | mrgoby | so, i need to be able to yank people into the meetme at any given time |
06:06.15 | mrgoby | basically |
06:06.31 | erwinism | JerJer[mobile] i did already. maybe i missed something on the sip.conf ? |
06:06.32 | JerJer[mobile] | nasty |
06:06.35 | mrgoby | can i do that with Action: Redirect ? |
06:06.45 | JerJer[mobile] | sounds like a job for redrect |
06:06.51 | mrgoby | coolness |
06:07.04 | JerJer[mobile] | but its not gong to be fun |
06:07.11 | mrgoby | why come ? |
06:07.42 | JerJer[mobile] | how do you plan on figuring out which calls to transfer into the meetme? |
06:07.51 | mrgoby | all zap channels |
06:08.01 | mrgoby | only 4 |
06:08.25 | *** join/#asterisk ryguillian (~ryguillia@c-24-12-96-52.client.comcast.net) |
06:08.46 | harryvv | jerjer also this is going to be the backup asterisk system once i get the original up and running again. |
06:10.33 | *** join/#asterisk Damin_Mobile (~pocketirc@26.sub-70-214-24.myvzw.com) |
06:10.34 | JerJer[mobile] | i'm sorry |
06:10.47 | Damin_Mobile | Bastards lost my luggage! |
06:10.50 | JerJer[mobile] | lol |
06:10.55 | JerJer[mobile] | that sucks |
06:11.33 | harryvv | damin, should have cut off your ancle tracker and put it in the luggage :) |
06:11.58 | Damin_Mobile | They think it might come in on the eleven fifteen fligh in an hour or so. |
06:12.26 | JerJer[mobile] | famous last words |
06:12.37 | brc_ | Damin_Mobile! |
06:12.40 | brc_ | that sucks |
06:13.34 | Damin_Mobile | Whenever you hear the phrase 'unexpected plane change' you can be sure that your luggage wont make the switch with you. |
06:14.11 | erwinism | JerJer[mobile], ok the slite is already logged in, buy how come it says "Your number is: xlite1 " ?? |
06:14.28 | The_Ball | Does this look correct for iaxtel? exten => _1700NXXXXXX,1,Dial(IAX2/< my username >:< my password >@iaxtel.com/${EXTEN}@iaxtel) |
06:15.33 | The_Ball | i get a Mar 7 16:08:47 WARNING[20420]: chan_iax2.c:1477 attempt_transmit: Max retries exceeded to host 69.73.19.178 on IAX2/69.73.19.178:4569/3 (type = 6, subclass = 1, ts=3, seqno=0) message |
06:15.41 | JerJer[mobile] | erwinism: no idea what your talkin about |
06:15.56 | JerJer[mobile] | is iaxtel up? |
06:16.05 | erwinism | jerjer, how can i set a user's number? im using SIP |
06:16.13 | The_Ball | JerJer[mobile], what do you mean? |
06:16.31 | JerJer[mobile] | iaxtel has never been reliable |
06:17.54 | JerJer[mobile] | erwinism: set the username and secret in your softphone |
06:18.10 | *** join/#asterisk tecnico (~tecnico@user-24-236-123-31.knology.net) |
06:18.51 | erwinism | jerjer yes i did, my username is: xlite1 and secret is: xlite1 |
06:19.05 | erwinism | and is already logged in |
06:19.18 | JerJer[mobile] | then what is the problem? |
06:19.36 | erwinism | now, how can the other softphone call me? how would i know my number? |
06:19.42 | JerJer[mobile] | you have to make one |
06:19.45 | JerJer[mobile] | in asterisk |
06:19.48 | JerJer[mobile] | extensions.conf |
06:19.52 | erwinism | oh! |
06:20.04 | erwinism | ok, i will look for it. thanks :D |
06:20.05 | JerJer[mobile] | i think i see a light bulb turn on |
06:20.17 | *** join/#asterisk bsdfreak (ninja@enterthebass.com) |
06:21.01 | erwinism | hehe |
06:21.43 | Damin_Mobile | So im sitting in this place called Martini Monkey 's drinking beer. |
06:21.49 | JerJer[mobile] | hell yeah |
06:21.53 | JerJer[mobile] | waiting on ur luggage? |
06:22.22 | *** join/#asterisk witten (~witten@D-128-208-60-207.dhcp4.washington.edu) |
06:22.32 | Damin_Mobile | So I ask the bartender for another Sierra Nevada pale ale.. |
06:22.38 | Beirdo | man, setting up a meetme conference was easier than I expected |
06:23.11 | Damin_Mobile | He gives me a bud light and a shot of Crown Royal... |
06:23.15 | Damin_Mobile | WTF? |
06:23.48 | Damin_Mobile | A its free.. |
06:23.49 | Beirdo | I'll take the Crown Royal, but the bud light? blech |
06:23.54 | Damin_Mobile | <PROTECTED> |
06:25.11 | Beirdo | if that arsehole calls at 5am again today, I'll be mega-pissy |
06:25.42 | *** part/#asterisk witten (~witten@D-128-208-60-207.dhcp4.washington.edu) |
06:25.47 | Damin_Mobile | bkw_ has arrived! |
06:26.14 | Damin_Mobile | Alright..Gotta sign off now... |
06:26.18 | Damin_Mobile | Later... |
06:27.45 | *** join/#asterisk mrgoby (~mrgoby@141.211.162.97) |
06:29.47 | JerJer[mobile] | yeah smells like its time to signoff as well |
06:32.09 | *** join/#asterisk zoa (zoa@142.131.189.23) |
06:32.12 | zoa | hi there |
06:32.15 | zoa | live from von |
06:32.21 | zoa | or at least live from san jose |
06:32.38 | newpers | so, for if i had a 1 voip line -> asterisk with three extensions. I could only have 1 person on an extension at a time? |
06:33.54 | newpers | In otherwords, I couldn't have all three extensions in use at once? |
06:36.37 | erwinism | hello, where can i edit my PBX voice prompts? |
06:36.43 | erwinism | it gives me sample voice |
06:36.46 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
06:36.50 | erwinism | i just installed asterisk |
06:37.00 | mrgoby | ~docs |
06:37.01 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
06:37.16 | *** join/#asterisk thepdakid (~vtrandal@c-24-8-106-135.client.comcast.net) |
06:38.34 | thepdakid | any use packet8 |
06:39.04 | newpers | or is that only for pots? |
06:39.26 | thepdakid | Anyone use Packet8.net for phone service |
06:39.29 | thepdakid | ? |
06:39.46 | thepdakid | IP over cable modem |
06:40.12 | godsmoke | thepdakid: what do you want to know? |
06:40.20 | harryvv | zoa how has von been today |
06:40.43 | harryvv | time to reboot. |
06:41.23 | thepdakid | does it work as well as vonage? And if I am going to setup a asterisk server what service should I have? |
06:41.51 | godsmoke | thepdakid: "work as well as vonage" -- not sure what you're referring to specifically |
06:42.32 | godsmoke | vonage has a completely different target audience than packet8 -- their services are both voip, but vonage focuses on home-user needs |
06:44.14 | newpers | for this setup: 2 extensions set up on asterisk with one vonage line. When I'm on extension one and someone calls, there's calls for extension 2, they get a busy signal? |
06:45.42 | *** join/#asterisk geekster (~Klenert@fw.telehouse.com) |
06:47.30 | PTG123 | anyone know how to make it if the person calls their own number, to not ask them for their password? |
06:47.33 | PTG123 | when checking voicemail |
06:48.01 | *** join/#asterisk cc (~cc@byte.fedora) |
06:49.59 | thepdakid | Anyone use Packet8.net for phone service? Any trouble with audio dropouts? |
06:51.39 | thepdakid | ? |
06:51.55 | Inv_arp | thepdakid: packet8 works with *? |
06:52.24 | thepdakid | I don't know but it's not SIP |
06:54.14 | *** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com) |
06:54.33 | thepdakid | vonage uses SIP and seems to work better than Packet8 |
06:55.38 | TheEmperor | hi guys, can anyone help with musiconhold? |
06:56.04 | TheEmperor | it doesn't seem to play the .mp3 file i put into /var/lib/asterisk/mohmp3 |
06:56.22 | TheEmperor | even when i have already edited musiconhold.conf |
06:56.54 | erwinism | hahahahahahahaha at last i already installed asterisk!!! |
06:57.04 | *** join/#asterisk shaZwaz (~adnans@203.81.196.167) |
06:57.08 | justinnnn | anyone no why txfax/spandsp (any versions) crash asterisk |
06:57.11 | justinnnn | after it sends a fax ? |
06:57.21 | geekster | TheEmperor, do you see this when you run "ps aux" |
06:57.22 | geekster | oot 83227 0.0 0.6 3260 2824 ?? I 12:28AM 0:00.03 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river. |
06:57.23 | geekster | root 83226 0.0 1.0 5428 4916 ?? I 12:28AM 0:01.91 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river. |
06:57.42 | *** part/#asterisk shaZwaz (~adnans@203.81.196.167) |
06:57.58 | *** join/#asterisk shaZwaz (~adnans@203.81.196.167) |
06:58.18 | geekster | has anyone gotten wakupcall to work / |
06:58.40 | shaZwaz | hi all |
06:58.41 | TheEmperor | geekster: ? |
06:58.51 | geekster | yes |
06:58.55 | justinnnn | anyone txfax ???? |
06:58.59 | justinnnn | rxfax works cool |
06:59.49 | TheEmperor | geekster: i don't get those error messages |
06:59.49 | shaZwaz | justinnnn, I tried it a few weeks ago , gets stuck |
07:00.03 | geekster | those are not error message, that is the process list |
07:00.42 | geekster | mpg123 should be running to play musiconhold. |
07:00.43 | TheEmperor | geekster: yes, i see those there |
07:01.39 | TheEmperor | root 2919 0.0 0.1 4248 1148 tty1 S 14:51 0:00 mpg123 -q -s --mo |
07:01.39 | TheEmperor | root 2922 0.0 0.0 3772 540 tty1 S 14:51 0:00 mpg123 -q -s --mo |
07:02.08 | geekster | whats your musiconhold.conf file look like ? |
07:02.34 | TheEmperor | [classes] |
07:02.35 | TheEmperor | default => mp3:/var/lib/asterisk/mohmp3/ |
07:02.35 | TheEmperor | ;loud => mp3:/var/lib/asterisk/mohmp3 |
07:02.35 | TheEmperor | ;random => quietmp3:/var/lib/asterisk/mohmp3,-z |
07:02.35 | TheEmperor | beat => quietmp3:/var/lib/asterisk/mohmp3/beat |
07:02.43 | *** join/#asterisk Insanity5 (~feaw@204.134.196.33) |
07:03.04 | geekster | try this: default => quietmp3:/usr/local/share/asterisk/mohmp3 |
07:03.26 | TheEmperor | ok.. |
07:03.36 | geekster | remove the other crap for now. |
07:04.41 | Insanity5 | Is there a recommended solution for deploying asterisk in a mission critical environment? I want, perhaps in an idealistic sense to be able to have the user simply power cycle the machine if something goes wrong. Points of failure, including fans and hard disks should be minimized. Outside of using server grade hardware, are there any solutions to minimize points of failure (hard disks, fans, etc). |
07:05.26 | Insanity5 | And the issue here is not the size of deployment, but rather that the geek (me) who is setting it up will be many miles away. If I could approach the reliability of say a traditional panasonic / lucent system, that would be ideal. |
07:05.45 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
07:07.54 | TheEmperor | geekster: there is no asterisk directory in /usr/local/share/ |
07:08.53 | geekster | make it match your setup, that is where mine is |
07:08.57 | TheEmperor | oh ok |
07:09.00 | TheEmperor | duh :) |
07:09.48 | justinnnn | grr stupid txfax |
07:09.51 | justinnnn | does it actualy work ? |
07:09.55 | TheEmperor | geekster: it works! :) |
07:10.02 | TheEmperor | geekster: thanks! |
07:10.18 | geekster | yup |
07:10.19 | geekster | np |
07:10.36 | *** join/#asterisk Goshen (Goshen@c-67-172-238-57.client.comcast.net) |
07:10.41 | TheEmperor | how come it didn't work the last time? |
07:10.47 | Trionnis | anyone aware of outgoing voice issues with broadvoice? |
07:10.55 | Trionnis | people tell me I keep "fading out" |
07:11.08 | Trionnis | I hear incoming audio just fine, however |
07:12.06 | Goshen | someone said something about 4 hours ago about problems with broadvoice...don't remember who it was |
07:14.05 | Trionnis | k |
07:14.06 | Trionnis | thakns |
07:14.08 | Trionnis | er |
07:14.09 | Trionnis | thanks |
07:14.10 | Trionnis | :) |
07:14.21 | Insanity5 | Can you use stanaphone with asterisk or no? |
07:14.21 | Inv_arp | Insanity5: set it up like u would any other mission critical system... |
07:15.07 | Inv_arp | Insanity5: iyes |
07:15.57 | TheEmperor | geekster: how do i turn the volume down on musiconhold? |
07:15.57 | justinnnn | ppls |
07:16.00 | justinnnn | wat does this mean ? |
07:16.00 | justinnnn | #define MAX_BLOCK_SIZE 240 |
07:16.03 | justinnnn | in app_txfax.c ? |
07:16.18 | geekster | EMP; that i'm not sure about. |
07:16.26 | Inv_arp | TheEmperor: resample the music at a lower volume... any music editor |
07:16.31 | geekster | what type of phone do you have. |
07:17.15 | TheEmperor | just using normal office phones |
07:17.19 | Insanity5 | Inv_arp - I know... There's just something about computers tha says it won't be working 15 years from now without someone touching it :) |
07:17.28 | TheEmperor | Inv_arp: any recomendations? |
07:18.02 | Inv_arp | TheEmperor: audacity even sox can do that must must be converted to wav first |
07:18.09 | Insanity5 | Inv_arp - The reason I asked about stanaphone is because I thought there was some special feature that your voip provider must support to produce a working system. I can't remember anymore, I'm about to set my first one up though :) |
07:19.02 | TheEmperor | Inv_arp: ok |
07:21.20 | Goshen | who has an enum164.org entry that I can call to test my enum lookup? I tested it working for my entry |
07:22.16 | *** join/#asterisk par (par@trackbugz.antisecurity.net) |
07:26.23 | Mavvie | goshen 61293353018 |
07:26.38 | Goshen | Calling |
07:26.51 | Goshen | wait..what kind of number is that? |
07:27.01 | Mavvie | it's my number. |
07:27.15 | Goshen | 6 is the international prefix? |
07:27.18 | Mavvie | no, 61 |
07:27.28 | par | ozzie |
07:27.36 | Mavvie | oi! |
07:27.40 | *** join/#asterisk Eight (~blake@12-205-155-39.client.mchsi.com) |
07:27.45 | Mavvie | my wife would be proud of me. |
07:28.20 | Goshen | so I guess I dial 01161293353018 |
07:28.40 | Mavvie | yes |
07:29.02 | Goshen | confusing my sipura :) |
07:29.04 | Mavvie | I won't pick up the phone. |
07:29.09 | Mavvie | to save you money. |
07:29.21 | Goshen | if you have an enum lookup it should go direct... |
07:29.54 | dfunnell | Hi all, can anyone help with a dial-out problem I am having? Using CAPI * is dialling as soon as a pattern is matched, which makes it difficult with variable length numbers (such as mobile numbers, etc.) Using exten => _100.,1,Dial,CAPI/470:${EXTEN:1} for example dials out on 00n (where n is the fourth digit dialled after '100'). V desperate, will pay in beer. |
07:30.02 | Goshen | confusing my dialplan too |
07:30.08 | Mavvie | :-) |
07:33.16 | PTG123 | anyone know of a way to know the name of the sip account dialing the extension? |
07:34.31 | jontow | AGI might do it (?) |
07:34.39 | GrimStone | there any solution to "got a response on a call we don't know of" problems with Broadvoice outgoing ? |
07:36.22 | *** join/#asterisk jmhunter (~jmhunter@64.77.199.223) |
07:36.22 | *** mode/#asterisk [+o jmhunter] by ChanServ |
07:36.29 | jmhunter | whats up bitches |
07:36.37 | PTG123 | well i want to do it in the extension line |
07:36.40 | PTG123 | in extensions.conf |
07:36.50 | geekster | has anyone gotten wakupcall working ? |
07:37.04 | Goshen | Mavvie: SIP/0293353018@barnet.com.au :) |
07:37.09 | jmhunter | I NEED A VON PASS |
07:37.10 | jmhunter | NOW |
07:37.12 | jmhunter | please |
07:37.22 | Goshen | Mavvie: == No one is available to answer at this time |
07:37.37 | jmhunter | ~ seen brc_ |
07:37.38 | jbot | brc_ is currently on #asterisk. Has said a total of 14 messages. Is idling for 1h 24m 58s |
07:37.58 | brc_ | ~seen jmhunter |
07:37.59 | jbot | jmhunter is currently on #asterisk (1m 37s). Has said a total of 5 messages. Is idling for 22s |
07:38.11 | jmhunter | whats up bitch |
07:38.20 | brc_ | your pm window broken?' |
07:38.22 | Mavvie | Goshen: didn't see a call coming in. |
07:39.02 | *** join/#asterisk Tarox (someone@pD9E7BF13.dip.t-dialin.net) |
07:40.09 | Mavvie | Goshen: are you sure it calls here? |
07:40.13 | Mavvie | to here? |
07:40.37 | Goshen | I think so..I pasted all of the messages in query window |
07:40.44 | Goshen | I will try again with sip debug on |
07:41.02 | jmhunter | ~seen kram |
07:41.03 | jbot | kram <~mark@kram.digium.sponsor.pdpc> was last seen on IRC in channel #asterisk, 2d 4h 4m 27s ago, saying: 'oh :)'. |
07:41.14 | jmhunter | fucking shit |
07:41.32 | jmhunter | where are you mother fuckers |
07:41.37 | jmhunter | ill deop myself for that |
07:42.48 | Mavvie | Goshen: still: I don't see any packets here |
07:42.55 | *** mode/#asterisk [+o brc_] by jmhunter |
07:43.02 | brc_ | 0_0 |
07:43.21 | rvhi | is there a way to find out how many agents on the phone in ACD? |
07:43.34 | jmhunter | kram |
07:43.36 | jmhunter | bkw |
07:43.38 | jmhunter | twisted |
07:43.40 | jmhunter | FUUUUCK |
07:45.03 | Goshen | Mavvie: firewall? |
07:45.08 | Mavvie | Goshen: none here. |
07:45.25 | jmhunter | fuuuck where is everyone |
07:45.29 | jmhunter | someone should deop me |
07:45.33 | jmhunter | brc hit me |
07:45.36 | brc_ | yeah they should |
07:45.41 | Mavvie | --- #asterisk :You need to be a channel operator to do that |
07:45.45 | Mavvie | can't tell that I didn't try |
07:46.07 | *** kick/#asterisk [jmhunter!~brian@brc.base.supporter.pdpc] by brc_ (#asterisk /me hands jm some soap) |
07:46.11 | Mavvie | Goshen: can you ping tardis.barnet.com.au ? |
07:46.20 | *** join/#asterisk jmhunter (~jmhunter@64.77.199.223) |
07:46.20 | *** mode/#asterisk [+o jmhunter] by ChanServ |
07:47.02 | Goshen | 64 bytes from tardis.barnet.com.au (202.83.176.38): icmp_seq=1 ttl=45 time=189 ms |
07:47.03 | *** mode/#asterisk [+brc +q!*@*] by jmhunter |
07:47.19 | brc_ | fumble fingers |
07:47.23 | Goshen | that isn't the machine I am calling though |
07:47.31 | *** mode/#asterisk [+brc -q!*@*] by jmhunter |
07:47.38 | brc_ | dude |
07:47.39 | brc_ | haha |
07:47.45 | *** mode/#asterisk [+brc +q!*@*] by jmhunter |
07:47.45 | *** join/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net) |
07:47.49 | Qwell | he needs practice. :p |
07:47.51 | brc_ | hey there dr |
07:47.54 | jmhunter | no one can here u |
07:47.56 | brc_ | DrRighteous, at von? |
07:48.02 | *** mode/#asterisk [+brc -q!*@*] by jmhunter |
07:48.06 | Goshen | Mavvie: 64 bytes from tim.barnet.com.au (202.83.176.33): icmp_seq=4 ttl=45 time=205 ms |
07:48.09 | DrRighteous | I wish |
07:48.14 | dfunnell | Did I mention I will pay in beer? Help! |
07:48.16 | DrRighteous | But I sent file to CON |
07:48.18 | DrRighteous | VON |
07:48.18 | Qwell | /unban +q!*@*, /mode #asterisk -rc |
07:48.20 | Mavvie | Goshen: "I don't know" |
07:48.20 | Qwell | or something, heh |
07:48.28 | brc_ | no, it's supposed to be +rc |
07:48.37 | *** mode/#asterisk [-b +q!*@*] by brc_ |
07:48.42 | jmhunter | her Dr Rightheous.. u hook me up with a von pass.. ill hook u up with OPS |
07:49.09 | DrRighteous | haha, file's an employee... and he knows how to beg |
07:49.16 | *** join/#asterisk footnote (~jhicks@67.141.135.121) |
07:49.47 | GrimStone | thats so Rightheous.. |
07:49.53 | footnote | "he's dead jim" |
07:50.13 | jmhunter | jacob not jm |
07:50.15 | jmhunter | blah |
07:50.20 | jmhunter | jim |
07:50.24 | footnote | hrm |
07:50.36 | footnote | i don't remember ole jake |
07:50.50 | jmhunter | i think i was around before u |
07:50.57 | footnote | doubt it :) |
07:51.00 | jmhunter | im an old timer |
07:51.11 | footnote | my grandson born today's name is jacob! |
07:51.26 | footnote | jacob tyler hicks |
07:51.29 | Mavvie | congratulations. |
07:51.46 | footnote | yeah, he's gonna be one heckuva good flyfisherman |
07:52.05 | *** mode/#asterisk [+o DrRighteous] by jmhunter |
07:52.56 | jmhunter | o ok.. well i was here mostly during summer |
07:53.19 | Goshen | Mavvie: phone ring? |
07:55.29 | Goshen | Mavvie: sip call didn't work so it fell back to dialing with voipuser.org, it looked like your phone rang, did it? |
07:55.51 | Mavvie | Goshen: aha, but that wasn't via the sip phone :-) |
07:56.14 | *** join/#asterisk kks (~kks@203.115.210.253) |
07:56.31 | Goshen | yea, I can't reach your sip phone...can you? |
07:56.37 | Mavvie | yes. |
07:56.53 | Goshen | anyone else have a enum164.org entry that I can try calling? |
07:57.08 | *** join/#asterisk HitTop (~Miranda@HSE-Toronto-ppp286299.sympatico.ca) |
07:58.24 | footnote | hrm |
07:59.54 | *** join/#asterisk Duckbizkit (~DMAN@ip-216-97-163-53.valornet.com) |
08:00.02 | Duckbizkit | sweet |
08:00.10 | jmhunter | In search of VON passes |
08:00.12 | jmhunter | anyone |
08:00.32 | tuxinator_linux | jmhunter: Nope, but I'll be at Meet * |
08:00.51 | tuxinator_linux | I'm in search of VON passes |
08:01.36 | Insanity5 | How many people here use voip for their business? I am kind of weary of switching over from POTS, simply because of the possibly downtime from the provider and possible issues with SBC dsl. |
08:01.52 | tuxinator_linux | I wouldn't do it over DSL |
08:02.00 | tuxinator_linux | for a business at least |
08:02.11 | tuxinator_linux | unless you only do one call at a time |
08:02.15 | *** mode/#asterisk [-b -q!*@*] by jmhunter |
08:02.21 | Duckbizkit | i've got a tough VM question....any takers? |
08:02.34 | tuxinator_linux | spit it out |
08:02.48 | footnote | VM is hard, let's go shopping. |
08:03.03 | tuxinator_linux | VM like in Voice Mail? |
08:03.07 | Qwell | footnote: I'm in. I need new clothes. |
08:03.20 | Goshen | Insanity5: the other problem with dropping your POTS is loosing your phone directory listing |
08:03.23 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
08:03.42 | footnote | vlad the impaler works for atmel now |
08:03.44 | footnote | kewl |
08:03.54 | Insanity5 | Goshen - Can't you port it over? |
08:03.55 | tuxinator_linux | People that don't use ILECS loose the phone directory listing anyways |
08:03.55 | Duckbizkit | if i call VoiceMailMain(s204) , it works fine and goes straight to VM. if i call VoiceMailMain(s6233239933), it prompts for the login/pass |
08:04.05 | Insanity5 | Goshen - The goones with the phonebook will gladly sell you any overpriced listing you want, I'd assume. |
08:04.18 | Duckbizkit | it seems like these 10 digit numbers i'm passing to VM are just messing it up |
08:04.23 | Goshen | Insanity5: of course...but how about directory assistance? |
08:04.25 | tuxinator_linux | That's what I had to do, buy an add in the phone book |
08:04.28 | Duckbizkit | because everything eles works |
08:04.31 | Duckbizkit | *else |
08:04.43 | tuxinator_linux | never thought about directory assistance |
08:04.48 | tuxinator_linux | don't they have a phone book? |
08:04.54 | Goshen | Inanity5: also let me know when you find a VOIP provider that does number portability on US numbers... |
08:05.07 | Goshen | Nufone does for their local area code only |
08:05.29 | Insanity5 | Gos - Vonage? |
08:05.45 | Goshen | Vonage doesn't allow Asterisk so they are out |
08:05.49 | Goshen | Gos? |
08:05.53 | Qwell | Goshen: broadvoice supposedly does. or, will soon |
08:06.07 | GrimStone | BroadVoice worked fine until Saturday |
08:06.15 | GrimStone | then they screwed it up |
08:06.16 | jmhunter | bv has as of like june |
08:06.25 | Goshen | Qwell: too many people in here complaining today about Broadvoice being down this weekend |
08:06.27 | Insanity5 | Goshen - AT&T VOIP? |
08:06.29 | Qwell | dunno, the site said "coming soon" |
08:06.41 | Goshen | Think AT&T is going to allow Asterisk? |
08:06.52 | GrimStone | there seems to be some strange issue with asterisk , and BroadVoice presently |
08:07.02 | Insanity5 | Goshen - Why not? |
08:07.08 | Qwell | Are there any providers that do everything most people want? heh |
08:07.12 | Duckbizkit | so any ideas? |
08:07.26 | Goshen | because it is too open |
08:07.38 | GrimStone | keep getting this on outgoing calls - "Got a response on a call we dont know about. " |
08:07.50 | Qwell | GrimStone: weird nat issue? |
08:08.02 | Insanity5 | Goshen - What does Asterisk require, that stanaphone and broadvoice supplies, that vonage won't? |
08:08.03 | GrimStone | i don't have NAT .. thats the strange thing |
08:08.08 | Qwell | ie; the response looks like its coming from your router or something? |
08:08.24 | Qwell | GrimStone: That would be tricky then, yeah. heh |
08:08.33 | GrimStone | if i set nat=yes , i can hear the voice data , but those messages still come up in asterisk |
08:08.34 | Goshen | Insanity5: allow you to connect your Asterisk server to them... |
08:08.38 | *** join/#asterisk iceyp (~icepick@max.unix.co.nz) |
08:08.52 | Goshen | I mean you could go VOIP to FXS, then FXO card to Asterisk server...but who wants that |
08:08.53 | Qwell | GrimStone: Your box has a external IP? |
08:08.58 | GrimStone | and asterisk seems to think the call is still not connected |
08:09.03 | Qwell | Goshen: With vonage, or at&T? |
08:09.12 | Goshen | Qwell, right |
08:09.15 | iceyp | can someone test 08449865089 from the UK and tell me if it connects u to an IVR plz, my telephone provider and voip providers dont have PSTN in the UK |
08:09.24 | Qwell | I mean, were you specifying one in particular? |
08:09.27 | GrimStone | Qwell: yes , and broadvoice's subnet has access to the right ports |
08:09.49 | Qwell | GrimStone: odd. got me |
08:09.51 | Insanity5 | Goshen - I mean though, what is so special about asterisk as a SIP client compared to any generic SIP device? How would they know? |
08:09.57 | Goshen | no, just in general foe closed voip providers |
08:10.15 | Goshen | Insanity5: because their configurations are secret |
08:10.18 | GrimStone | Qwell: and with nat=yes when the call gets over it shows up as NO ANSWER |
08:10.19 | Qwell | Insanity5: You underestimate them |
08:10.44 | ta[i]nted | how do i delimit SetVar: in a call file? |
08:10.50 | Duckbizkit | :) |
08:10.51 | Goshen | Because when you place a call it says ASTERISK in the call ;) |
08:11.02 | ta[i]nted | SetVar: foo=bar pee=poo |
08:11.03 | Qwell | Goshen: to be fair, that can be changed, right? |
08:11.16 | Qwell | Duckbizkit: Doesn't voicemailmain take you to the login always? |
08:11.20 | Qwell | I haven't rtfm'd lately |
08:11.22 | Insanity5 | Goshen - OLutside of any possible "referred" type value Asterisk passes along, I see no technical reason why it would not work, provided you obtained the SIP login and pass. |
08:11.32 | Duckbizkit | not if you put the s in front of the mailbox |
08:11.46 | Qwell | ahh |
08:11.59 | Qwell | Duckbizkit: and the 10 digit one is valid in voicemail.conf? |
08:12.03 | *** join/#asterisk memic (~memic@195.135.160.215) |
08:12.29 | Duckbizkit | well we're running realtime |
08:12.35 | Duckbizkit | but yes it's valid, i have left several VMs |
08:12.43 | Goshen | Insanity5: they don't give you your login and pass...and I am not interested in hacking a provider...rather go with one that supports * |
08:13.03 | Qwell | Goshen++ |
08:13.36 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
08:13.45 | Insanity5 | Goshen - Is it possible to configure a setup where say, if you don't answer your home phone (voip), asterisk forwards it on out another voip channel to your cell? Or is this just a bad idea all around for bandwidth and latence/delays. Is this best handeled at the provider level? |
08:13.46 | Qwell | Duckbizkit: realtime includes voicemail.conf? |
08:13.56 | Qwell | Duckbizkit: I've never really looked into it. I think I might sometime. |
08:14.06 | Qwell | Insanity5: sure its possible |
08:14.11 | Duckbizkit | yeah you still need voicemail.conf when you run realtime |
08:14.15 | Duckbizkit | for all the other config stuff |
08:14.16 | atmel | footnote, what? |
08:14.20 | Duckbizkit | realtime just moves off the users to the DB |
08:14.27 | Qwell | Insanity5: I personally do it at home with my fwd account. |
08:14.30 | *** join/#asterisk iceyp (~icepick@max.unix.co.nz) |
08:14.31 | Goshen | Insanity5: better at provider, then if not you can at your * box, you can do anything with * |
08:14.32 | iceyp | meh |
08:14.33 | atmel | ;) |
08:14.39 | atmel | eu vin sa te mananc |
08:14.41 | iceyp | anyone try that number for me from the uk? |
08:14.42 | atmel | muahaha |
08:14.57 | Qwell | Duckbizkit: oh, I see. |
08:15.01 | Goshen | iceyp: try #asterisk-uk |
08:15.11 | Goshen | I got someone in there to dial mine to test... |
08:15.25 | Insanity5 | Goshen - I mean, incomign call over voip from broadvoice, forwaded out to stanaphone to your cell making three trips across the country and back, and on a cell none the less. Does this smell as bad as I think? Or is it quite bareable? |
08:15.52 | Qwell | Insanity5: sounds like me connecting to work with the vpn! |
08:15.58 | Goshen | depends on your connection |
08:16.05 | Insanity5 | Qwell - lol |
08:16.07 | iceyp | Goshen thanks |
08:16.15 | Insanity5 | Goshen - 200kbit upstream, downstream's nto an issue |
08:16.21 | Qwell | Insanity5: CA to IA, back to CA (1 city away from me), out to IA, back to CA, then out to MN for files |
08:16.29 | Goshen | I am really looking forward to 3-13-05...thats the day my connection goes to 6mb/s down 768/kb/s up |
08:16.33 | moonwick | Insanity5: it's worse than you think |
08:16.41 | Qwell | Goshen: which provider? |
08:16.49 | Insanity5 | Sounds like speakeasy |
08:16.50 | Goshen | Comcast + $10 speed upgrade |
08:16.51 | jmhunter | ~seen pfn |
08:16.52 | jbot | pfn <500@adsl-69-107-210-254.dsl.pltn13.pacbell.net> was last seen on IRC in channel #asterisk, 13d 6h 23m 15s ago, saying: 'only with iax'. |
08:16.54 | Qwell | ahh |
08:16.57 | moonwick | latency is a very real problem with VoIP |
08:17.11 | Insanity5 | moonwick - Why does it sound bad, upstream bandwdith or flat out latency? |
08:17.21 | Qwell | latency, no doubt |
08:17.25 | Insanity5 | The cell phone will add it's own fair share. |
08:17.42 | Insanity5 | moonwick - Now, arguably, does a POT line or voip line have more latency? |
08:18.07 | moonwick | that's not a very good question, in all honesty |
08:18.34 | *** join/#asterisk mithro (~tim@dsl1-83.gw1.adl1.airnet.com.au) |
08:18.39 | Insanity5 | moonwick - It probably isn't :( |
08:18.40 | moonwick | but depending on the setup, VoIP is going to be much more prone to latency |
08:18.48 | *** join/#asterisk djin (~djin@62.58.40.196) |
08:18.56 | Insanity5 | I've found the voice quality is sometimes better on voip though. |
08:19.14 | mithro | hi! i'm after some information for FXO/FXS stuff which is compatible with australia |
08:19.41 | Insanity5 | moonwick - Would it be a lot less noticeable if I found a local voip provider where latency was say 40 ms instead of 80? I mean, arguably 40+40 = 80, and 80+80 = 160, but who knwos how the provider is going to backhaul it. |
08:19.53 | moonwick | Insanity5: yeah, that can make a considerable difference |
08:19.57 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
08:20.02 | Duckbizkit | well i guess i'm just fuckered |
08:20.02 | Duckbizkit | heh |
08:20.14 | Insanity5 | moonwick - I mean 80ms is acceptable for voip, but the double latency from a two directional link can add up. |
08:20.23 | Insanity5 | moonwick - Phone lines = $36/month out here :(. It's going. |
08:20.25 | moonwick | the human ear starts noticing latency even as low as 200ms |
08:20.41 | moonwick | and when you consider round trip IP latency, and codec latency... |
08:20.42 | moonwick | it all adds up |
08:20.47 | mithro | people have said on the internet they have had problems with the normal X100 cards |
08:20.51 | Insanity5 | moonwick - What is the typical cell phone latency? |
08:21.02 | moonwick | depends on the network |
08:21.05 | Insanity5 | sprintpcs |
08:21.14 | Insanity5 | I've noticed the newer phones compress the hell out of the audio more too. |
08:21.19 | Insanity5 | cdma / 1.9ghz |
08:21.32 | moonwick | I dunno, but my GSM phone seems to have pretty negligible latency |
08:21.38 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
08:21.40 | moonwick | it's not as easy to measure as it is with voip |
08:21.54 | Insanity5 | Nope, but I can tell when two cells are talking. |
08:22.14 | Insanity5 | I ping ~100ms to sip.broadvoice.com, they a bad idea? |
08:22.19 | *** join/#asterisk af_ (~af@ip-148-227.sn1.eutelia.it) |
08:22.38 | moonwick | 100ms is probably tolerable, but it also depends on what you plan to do with those calls once they hit your PBX |
08:22.46 | ta[i]nted | anyone here use call files? |
08:22.55 | moonwick | I've got my asterisk box set up to ring my cell phone whenever I receive a call |
08:23.06 | Duckbizkit | ta[i]nted, i've used them to annoy my friends |
08:23.12 | Qwell | moonwick: Just ring it, or forward entirely? |
08:23.13 | Goshen | moonwick : can I see that part of your dialplan? |
08:23.20 | ta[i]nted | Duckbizkit u ever use the SetVar in a call file? |
08:23.24 | Qwell | Goshen: yeah, I was gonna ask the same |
08:23.27 | moonwick | Qwell: ring it, along with a couple of SIP phones |
08:23.27 | Goshen | I was thinking about doing the same thing |
08:23.37 | ta[i]nted | that's easy |
08:23.40 | moonwick | sec |
08:23.46 | Qwell | oh, like Dial(Zap/1&SIP/blah) ? |
08:23.53 | moonwick | exactly |
08:23.54 | ta[i]nted | exten s,1,Dial(SIP/cellphone&IAX/foo) |
08:23.58 | Qwell | oh, o |
08:23.59 | Qwell | k |
08:24.13 | Duckbizkit | nope never used SetVar |
08:24.24 | Insanity5 | moonwick - I'd Asterisk to pick it up, say "connecting your call" nextel style, perhaps play some geeky hold music, while it simultaounsly rings my cell and home (SIP connected) phone. It would also be acceptable to place 3 rings on the cell and then forward it to the cell. |
08:24.43 | moonwick | I have problems with that because I can't get SIP to reliably cut my asterisk box out of the path when a call is answered by the cell phone |
08:24.46 | tzafrir | good morning |
08:24.49 | Insanity5 | The last line has an obvious type-o :) |
08:24.52 | *** join/#asterisk newpers (newpers@ip24-56-8-180.ph.ph.cox.net) |
08:24.55 | Qwell | moonwick: That doesn't actually take up any minutes on the cell, unless you answer, huh? |
08:25.02 | moonwick | correct |
08:25.08 | Qwell | interesting |
08:25.21 | mithro | so anyone know people using Asterisk in australia? |
08:25.29 | Insanity5 | moonwick - I suppose that would require suffient outgoing lines, or SIP connected phones. |
08:25.32 | Goshen | moonwick: would still like to see that part of your dialplan |
08:25.33 | moonwick | it would work well, if I could find a way to reliably get my asterisk box to pull itself out of the media path for calls answered by the cell |
08:25.51 | moonwick | Goshen: just look at what quell wrote |
08:26.09 | moonwick | dial(SIP/homephone&IAX/provider/<cell#>) |
08:26.35 | Goshen | so you are sending it out over voip |
08:26.56 | moonwick | yeah |
08:27.07 | Qwell | That kinda sucks though, since it basically charges you twice |
08:27.12 | Goshen | what if you added a Tt on the end? |
08:27.14 | newpers | I want to get asterisk up an running on 3 extensions and one line coming from nufone.net's voip service. I'm trying to figure out exactly what I need without using softphones. Is this correct: voip service, asterisk & server, 3 sip phones or 3 iaxy converters and 3 analog phones? |
08:27.35 | moonwick | Qwell: I use voicepulse for the incoming DID, which is flat rate |
08:27.45 | Qwell | ahh |
08:27.48 | Goshen | would that allow the call to bridge? or if you specified bridging in your iax.conf? |
08:27.49 | newpers | Is there anything else? And would it be cheaper to use sip phones over iaxy |
08:27.51 | moonwick | but consider that I'm in austin, and voicepulse's server is in NY |
08:27.54 | Insanity5 | stanaphone is free incoming too, no monthly fee. Caveat: New york # |
08:28.02 | Qwell | newpers: you won't need sip phones if you use an iaxy |
08:28.08 | Qwell | erm, nevermind! misread that |
08:28.14 | moonwick | if someone from Austin calls my DID and I answer from the cell, it gets to travel to and from NY twice. |
08:28.24 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-12-87.d4.club-internet.fr) |
08:28.24 | Insanity5 | Plus cell phone latency... |
08:28.26 | Qwell | newpers: It can vary greatly. If you want an expensive SIP phone, sure, it'll cost more. |
08:28.33 | newpers | which would you recommend? |
08:28.33 | Qwell | but if you want a cheapy grandstream, it may be cheaper |
08:28.38 | Insanity5 | moonwick - How does it sound? |
08:28.41 | *** join/#asterisk Damin_Mobile (~pocketirc@64.sub-70-214-30.myvzw.com) |
08:28.43 | moonwick | like crap |
08:28.48 | Qwell | newpers: I would recommend you do a little research, honestly. |
08:28.55 | Qwell | find out which method would be better for you |
08:29.04 | Insanity5 | moonwick - Better than a cell to cell call? |
08:29.05 | newpers | I didn't know there was a difference |
08:29.09 | moonwick | SIP is supposedly capable of optimizing extraneous hosts out of the media path, but I've had a lot of trouble getting that to work reliably |
08:29.15 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
08:29.31 | *** join/#asterisk pbxjunkie (~stormtroo@videocomputer.gr) |
08:31.24 | Insanity5 | moonwick - No problem with using SIP to connect to an office asterisk server via SIP over the internet and "work from home"? |
08:31.42 | Zeeek | I do that every day |
08:31.44 | Insanity5 | And just be say, an extension? |
08:32.02 | Goshen | I had to use IAX to get to my office... |
08:32.05 | jmhunter | ~seen twisted[work] |
08:32.06 | jbot | twisted[work] is currently on #asterisk |
08:32.11 | Goshen | office is double(tripple) nat |
08:32.18 | Insanity5 | Goshen - Yuck :) |
08:32.19 | Insanity5 | Hehe. |
08:32.21 | Goshen | * server is NAT |
08:32.24 | moonwick | Insanity5: depends on how much latency there is between home and the office |
08:32.30 | Insanity5 | moonwick - 60ms. |
08:32.33 | Goshen | so it was just ugly...IAXy made it right through |
08:32.48 | Qwell | thats alot of hops! |
08:32.56 | Insanity5 | moonwick - Only 128kbit available upstream though. |
08:32.57 | moonwick | if the office is connected to the PSTN from there, that's probably no big deal |
08:33.04 | Insanity5 | moonwick - Yes, it's pstn. |
08:33.13 | moonwick | yeah, no big deal, I imagine |
08:33.38 | Qwell | it sucks, because our firewall at work is far too restrictive |
08:33.51 | Insanity5 | Qwell - Probably for good cuase :) |
08:33.53 | moonwick | 128kbps is plenty for VoIP, but you'll need some sort of QoS |
08:34.00 | Qwell | Insanity5: Just because I work for a bank... |
08:34.01 | Qwell | :p |
08:34.10 | Insanity5 | Qwell - Understandable :) |
08:34.19 | Zeeek | some offices subscribe to http filtering services. If a site talks about subject "a" they arbitrarily put it in the list |
08:34.20 | Qwell | we get 80, and only http |
08:34.33 | shaZwaz | is there a cracked version of Eyebeam ? |
08:34.35 | Insanity5 | Qwell - You're lucky for that. |
08:34.40 | Insanity5 | moonwick - The bset I can do is a hacked-up linksys router firmware w/ qos. |
08:34.54 | Insanity5 | moonwick - However, there's only websurfing on the other side. But it could get ugly quick :0 |
08:35.24 | Insanity5 | Is it best to make it so your fax line never has to touch the * box? |
08:35.31 | moonwick | once I got around to setting up an old pentium with openbsd and altq, I became far more satisfied with voip |
08:36.50 | *** join/#asterisk mitcheloc (~mitchel@69-169-28-46.anhmca.adelphia.net) |
08:36.59 | Insanity5 | Why did they discontinue the X100P cards? |
08:37.06 | Insanity5 | I know they're $6.99 on ebay, but still |
08:37.10 | Insanity5 | There's no analog solutions left. |
08:37.22 | mitcheloc | $6.99 for a clone no? |
08:37.28 | Qwell | for the clone, yeah |
08:37.29 | mitcheloc | the tdm400b cards are analog |
08:37.34 | mitcheloc | not the real thing though |
08:38.16 | Insanity5 | Anything wrong with this? http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=44940&item=6748462326&rd=1&ssPageName=WDVW |
08:38.16 | footnote | hrm. altivec based codecs |
08:38.39 | Qwell | Insanity5: its a clone |
08:38.50 | Insanity5 | Does it matter? you can't get the real thing anymore anyways. |
08:38.54 | Insanity5 | Is there any quality loss? |
08:39.59 | brc_ | yes |
08:40.07 | Insanity5 | brc_ - Yes to which one? :) |
08:40.20 | brc_ | last |
08:40.30 | Insanity5 | Just inferior components? |
08:40.49 | *** join/#asterisk NoCAT (NoCAT@c-24-9-32-2.client.comcast.net) |
08:41.07 | NoCAT | anyone alive in here? |
08:41.17 | Zeeek | barely |
08:41.54 | *** join/#asterisk andi2 (~andi@212.88.172.176) |
08:41.56 | mitcheloc | nah |
08:42.00 | mitcheloc | they are intel pci modems |
08:42.04 | mitcheloc | with the same chipset |
08:42.31 | Insanity5 | Is there anything wrong with using a fleabay card? |
08:42.33 | mitcheloc | i bought one too a while ago, but never used it, i realized i didn't have a use and jumped straight to a t100p =), now i've figured a use for it though |
08:42.46 | mitcheloc | yep, your not supporting digium |
08:43.02 | mitcheloc | i didn't know anything about asterisk then, i would not buy a clone again |
08:43.09 | Zeeek | have you guys ever needed digium support? |
08:43.14 | mitcheloc | many times |
08:43.20 | Zeeek | how is it? |
08:43.22 | moonwick | once |
08:43.28 | mitcheloc | pretty good, they do try and help you |
08:43.30 | Insanity5 | mitcheloc - They don't offer a single card anymore, and I don't want to fork out for a 4 channel. |
08:43.34 | mitcheloc | and will call in other people when you have problems |
08:43.44 | mitcheloc | * that they can't solve |
08:43.56 | Zeeek | I have a very minor but I emailed. We'll see what happens |
08:44.48 | Zeeek | If those $10 cards really work, it'd be a grea deal to set up for soho and home use |
08:45.09 | Zeeek | the ad/listing emphasies asterisk |
08:45.23 | mitcheloc | ugh, /me found a very annoying bug in asterisk |
08:46.24 | Insanity5 | Are quad T-1's worth anything? I have an old clarent Quad T-1 gateway that has such cards in pci form. |
08:46.52 | Insanity5 | I've been meaning to put the whole box up on ebay... I got it for less than 20 at a bankruptcy auction just to play with it :P |
08:47.26 | mitcheloc | sounds like expensive hardware |
08:48.06 | Insanity5 | mitcheloc - It was, they paid something like $10,000 according to the invoice. However, I know it's worth pennies now... It's some massive voip gateway solution for 96 channels off some custom nt4 software. |
08:48.19 | Insanity5 | Voip wasn't exactly in its heyday in 1995 either :P |
08:50.30 | Insanity5 | mitcheloc - They look like this, except they have quad T-1 and soem custom hardware codec solution cards that this one doesn't: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61839&item=5754686552&rd=1&ssPageName=WDVW |
08:51.23 | Insanity5 | I think I'll sell it... that's the first one I've seen that sold over the last 6 months :P |
08:51.43 | drumkilla | me being file |
08:51.48 | tuxinator_linux | <PROTECTED> |
08:52.16 | NoCAT | what $10 cards? |
08:52.56 | Insanity5 | NoCAT - http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=44940&item=6748462326&rd=1&ssPageName=WDVW |
08:53.33 | Insanity5 | NoCAT - Probably clones / firmware hacks / who knows. If they work, it's what I need. I can't find a single port card on diginum's website. |
08:54.12 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
08:54.12 | *** mode/#asterisk [+o bkw_] by ChanServ |
08:54.16 | bkw_ | yo yo yo |
08:54.18 | bkw_ | wasabi |
08:54.24 | drumkilla | yoyo yo yo!!!!!!!!!!! |
08:54.28 | drumkilla | VON! |
08:54.54 | Qwell | bkw_: sushi? |
08:55.10 | *** join/#asterisk file (~file@251.134.218.209.transedge.com) |
08:55.11 | tuxinator_linux | Wuzzup bkw_ |
08:55.27 | file | omg Becky! |
08:55.37 | drumkilla | file!!!!!! |
08:55.42 | file | drumkilla!!! |
08:55.49 | tuxinator_linux | What about me ? |
08:55.55 | Qwell | file&drumkilla! |
08:56.31 | tuxinator_linux | file: look at her butt, its like, sooooo big |
08:56.35 | TheEmperor | hi, can anyone tell me why my voicemails don't get sent to me as an email? |
08:56.44 | TheEmperor | do i need to install a webserver on my * box? |
08:56.52 | tuxinator_linux | sendmail |
08:56.59 | Insanity5 | What happens if you connect more than one device at once with a voip provider? DO they both rings at once? Does the voip provider freak out? |
08:57.17 | TheEmperor | tuxinator_linux: sendmail? |
08:58.00 | Qwell | Insanity5: depends on if they support multiple devices |
08:58.17 | file | omg Becky, what is it?!? |
08:58.20 | bkw_ | GOD LOVE SQL INJECTIONS |
08:58.20 | bkw_ | har har har |
08:58.20 | bkw_ | NEXT!!! |
08:58.21 | Qwell | (it would likely ring them both, if they're both registered) |
08:58.37 | tuxinator_linux | TheEmperor: You need a program to send mail out with. Hold on... |
08:58.40 | NoCAT | how does a voice t1 work? do you have 24 phone numbers or does asterisk work the numbers out? |
08:58.45 | TheEmperor | tuxinator_linux: oh... |
08:58.53 | Insanity5 | Qwell - Is the call quality greater for a voip-to-voip call than pots-to-pots? I know in house cisco stuff sounds nice. |
08:58.57 | Qwell | NoCAT: DIDs aren't (they can be) associated to a specific port. |
08:59.07 | Qwell | Insanity5: got me |
08:59.12 | Insanity5 | NoCAT - Normally, you'll use it as a hunt group. |
08:59.23 | Insanity5 | NoCAT - T1's give you lots of fleixibility. |
08:59.47 | NoCAT | how does the number assigning work? can you have more then 24 phone numbers? |
08:59.55 | Qwell | NoCAT: Don't think of a port on a T1 as a phone number. Its simply a line. You can set it up for incoming or outgoing (or both). Then you get as many phone numbers as you want. |
09:00.04 | Qwell | You can get 1, or you can get 500. It doesn't matter. |
09:00.15 | NoCAT | thats amazing.. |
09:00.19 | Qwell | but |
09:00.26 | Qwell | each port, can only have ONE active call |
09:00.42 | Qwell | ie; if you have 1 T1 with 500 phone numbers, you can still only get 24 calls. |
09:00.59 | tuxinator_linux | Qwell: You should clarify that is a ISDN PRI over a T1 circuit |
09:00.59 | Qwell | (and that DOES include outgoing as well) |
09:01.05 | Qwell | tuxinator_linux: umm, ok |
09:01.08 | Insanity5 | NoCAT - And you can split half with data and voice :) And other fun stuff. |
09:01.09 | Qwell | what he said |
09:01.18 | Qwell | in other words, rtfw! |
09:01.22 | tuxinator_linux | I was confused for a long time |
09:01.44 | Insanity5 | Do any voip providers still let you mess with callerid liek you can with a T1/pri? :) |
09:01.57 | *** join/#asterisk soulz- (~soulz@host-137-132-45-54.imcb.nus.edu.sg) |
09:01.59 | soulz- | hello all |
09:02.04 | *** part/#asterisk pbxjunkie (~stormtroo@videocomputer.gr) |
09:02.21 | tuxinator_linux | TheEmperor: http://voip-info.org/wiki-Asterisk+config+voicemail.conf |
09:02.34 | NoCAT | can you get numbers from your local phone company? lets say i have 12+ phonelines now, at $30 each. and i wanted to get a t1, would i be able to keep numbers i currently have? |
09:02.36 | tuxinator_linux | take a look at "mailcmd" |
09:02.48 | tuxinator_linux | NoCAT: don't see why now |
09:02.50 | Qwell | NoCAT: not sure. maybe you can port them |
09:02.50 | tuxinator_linux | not |
09:03.02 | Qwell | DIDs are dirt cheap from what I hear, too |
09:03.03 | tuxinator_linux | NoCAT: That is what I plan to do |
09:03.03 | TheEmperor | tuxinator_linux: ok thanks |
09:03.19 | Qwell | ie; you sure as hell aren't going to pay $30, or even $3. Maybe something like $.30 |
09:03.25 | tuxinator_linux | $2/month for 20 numbers or something like that |
09:03.26 | NoCAT | qwest is currently my provider |
09:03.29 | Qwell | the T1 is still going to cost you a bit though, of course |
09:03.34 | Insanity5 | NoCAT - If they're terminated in the safe local c/o, liekly. |
09:03.34 | Qwell | its completely seperate |
09:03.35 | NoCAT | service provider |
09:03.40 | tuxinator_linux | You will pay about 550 for the T1 |
09:03.58 | Qwell | it varies greatly from state to state (and probably city to city) |
09:04.02 | tuxinator_linux | but that is 23 voice channels and one control (D) |
09:04.03 | Qwell | and of course, between providers |
09:04.22 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
09:04.46 | Qwell | sometimes the pricepoint is at like 10-15 lines, right? |
09:04.48 | Insanity5 | What do people do for extra phone lines as I'm sure any * user finds a need for them. Free voip provider? |
09:04.54 | TheEmperor | tuxinator_linux: so do i need to install a send out mail program as well? |
09:04.55 | tuxinator_linux | I am getting ready to purchace my first PRI, does anyone recommend any providers? |
09:05.08 | tuxinator_linux | TheEmperor: Yes |
09:05.14 | Insanity5 | Qwell - There's also logistical, business, and management decisions. Do you WANT to manage 16 individual analog lines? |
09:05.17 | Faithful | Hi all |
09:05.17 | tuxinator_linux | TheEmperor: It looks like it |
09:05.18 | TheEmperor | tuxinator_linux: what do you recommend? |
09:05.22 | Qwell | Insanity5: of course |
09:05.32 | tuxinator_linux | TheEmperor: sendmail is fine |
09:05.51 | tuxinator_linux | TheEmperor: Better to start with that as * expects it. |
09:05.52 | TheEmperor | tuxinator_linux: so i have to configure mailcmd, the line to send out using sendmail? |
09:05.55 | Qwell | I'd just have the whole thing sent into *, heh |
09:06.00 | Insanity5 | Qwell - No, I meant 16 individual phone jacks, phone wires, and copper pairs. a T-1 is one circuit. Not to mention they usually come with a higher SLA. |
09:06.13 | Qwell | yeah, I know what you meant |
09:06.19 | Faithful | Got my USB Bluetooth adapter today... I'm ready for blue tooth extensions ;) |
09:06.30 | TheEmperor | tuxinator_linux: oh, not really, I think all I have to do is to install sendmail and get it working and that will work right? |
09:06.32 | Insanity5 | Qwell - A T-1 is digitally controlled, clarity, instant dialing, flexibility, ability to forge caller id to your main number, it's there. |
09:06.48 | soulz- | Mar 7 17:00:25 NOTICE[258]: chan_sip.c:7539 handle_response: Failed to authenticate on INVITE to '"Dinesh" <sip:xxxxx@sip.broadvoice.com>;tag=as498b8f00' |
09:06.49 | Qwell | Insanity5: yeah, I've been well learned lately |
09:06.57 | soulz- | hello all, i seem to have get this lately on bv |
09:07.06 | soulz- | does anyone else have the same problem? |
09:07.07 | Insanity5 | Qwell - Now if you had 8 bri circuits, it might be another story. But still. |
09:07.12 | Qwell | hmm, bv really has been having problems this week, eh? |
09:07.24 | soulz- | yeah qwell |
09:07.27 | Faithful | soulz-: have you made the broadvoice patches? |
09:07.28 | Qwell | Insanity5: Is there a reason to get 8 BRI vs a PRI? |
09:07.52 | soulz- | faithful: CVS-HEAD-02/17/05-02:39:03 its a new cvs |
09:07.52 | Qwell | bri is generally what, 2 lines? |
09:07.55 | tuxinator_linux | TheEmperor: You will have to configure * to use it |
09:07.58 | Faithful | Qwell: if you only have a bri |
09:08.14 | Faithful | soulz-: but you still need to patch for bv |
09:08.15 | TheEmperor | tuxinator_linux: in the voicemail.conf? |
09:08.16 | Qwell | Faithful: availability you mean? |
09:08.35 | Faithful | PRI & BRI are different services |
09:08.36 | tuxinator_linux | TheEmperor: I have not done it personally, but from looking at the wiki page, yes |
09:08.39 | soulz- | faithful: i heard that the patch is not needed for the new ones as i read on the wiki |
09:08.39 | Qwell | somebody mind explaining something to me? I'm still slightly lacking right here... |
09:08.48 | soulz- | it was working perfect until 3 days ago |
09:08.49 | tuxinator_linux | BRI terminates differently |
09:08.49 | footnote | Faithful: They don't have bluetooth here in georgia. |
09:08.50 | mitcheloc | set canreinvite = no? |
09:08.50 | GMsoft | hey guys, anyone have a working config to make asterisk authenticate with gnugk using SimplePasswordAuth ? |
09:09.15 | soulz- | and i didn't touch anything |
09:09.15 | Qwell | When you get a PRI, what does the telco do for cabling? |
09:09.15 | footnote | We've got some GREEN teeth |
09:09.15 | footnote | no blue ones |
09:09.15 | Qwell | Do they actually have to send a new line to you, or? |
09:09.15 | Faithful | soulz-: check the source... |
09:09.17 | Insanity5 | Qwell - Well, it's better than 16 analog lines. It depends on the cost and availability. |
09:09.29 | Faithful | I had to patch when I set it up recently |
09:09.41 | Insanity5 | Qwell - BRI = ISDN = digital = 2 lines generally. |
09:09.42 | soulz- | where did u do it from faithful |
09:10.03 | Insanity5 | Qwell - Can also be used for 64k data on demand per channel. |
09:10.03 | Qwell | right |
09:10.08 | soulz- | http://edvina.net/broadvoice/broadvoicesip.txt ?? |
09:10.14 | Insanity5 | Qwell - It's a digitally controlled line, there's some advantages there. |
09:10.17 | Qwell | see above about my cabling question? |
09:10.19 | tuxinator_linux | ~BRI |
09:10.20 | jbot | extra, extra, read all about it, bri is the Basic Rate Interface , an ISDN access interface type composed of two B-channels each at 64 kbps and one D-channel at 16kbps (2B+D). |
09:10.23 | Faithful | soulz-: bv's asterisk setup page |
09:10.28 | tuxinator_linux | ~PRI |
09:10.29 | jbot | it has been said that pri is Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc. |
09:10.35 | Qwell | oh, so BRI still has a D, I see |
09:10.38 | Faithful | soulz-: I had to hand patch it ... |
09:10.42 | tuxinator_linux | Qwell: both do |
09:10.50 | Qwell | yeah, I knew a PRI did |
09:11.11 | NoCAT | can i resell acess to my t1 with voip legally? |
09:11.19 | footnote | Ok, for extra points, why doesn't E1 total 32 slots? |
09:11.28 | mitcheloc | if it's a business line |
09:11.40 | Insanity5 | Qwell - D sends dialing/call waiting/etc data to the telco. |
09:11.50 | Qwell | Insanity5: Yeah, I know. :) |
09:11.50 | Insanity5 | NoCAT - Yes, if ytou can make money at it :) |
09:11.53 | mitcheloc | * i think, not sure, but i look at it like you can if it's business internet, but you can't if it's a residential internet line |
09:11.55 | Qwell | I'm not a complete newb |
09:12.16 | soulz- | faithful: emm, thanks |
09:12.21 | Insanity5 | ISP's use PRI's for dialin lines. |
09:12.47 | Qwell | You know, its funny |
09:12.50 | Insanity5 | ISDN was supposed to eb in every household at the turn of the 90's... never happened. It was to be the analog replacement. |
09:13.06 | mitcheloc | could someone evaluate this for me? i've been stuck on it since last night |
09:13.07 | mitcheloc | exten => 7145154091, 3, GotoIf(${ALERT} = "<http://127.0.0.1/Bellcore-dr3>" ? 100 : 4); |
09:13.09 | Qwell | several months ago, I thought a phone number could only have 1 call at a time. I always wondered how ISPs worked in that regard, when they give you a local DID |
09:13.21 | footnote | Insanity5: hayes almost went out of business the first time over ISDN not taking off |
09:13.21 | mitcheloc | i'm trying to make that work...just a simple goto if |
09:13.33 | Faithful | Insanity5: but DSL/VoIP will be... |
09:13.35 | mitcheloc | but it don't work...heh |
09:13.39 | Qwell | footnote: So, why doesn't it equal 32? |
09:13.41 | footnote | And that was when they OWNED the modem market |
09:13.43 | tuxinator_linux | Qwell: Hunt groups are the other option to DID |
09:13.55 | Insanity5 | footnote - And then they did anyways. |
09:14.12 | footnote | Qwell: slot 0 overhead |
09:14.19 | Insanity5 | Technology is amazing... I mean the POT has been around since before WWII. |
09:14.20 | Qwell | So...when you get a PRI, does it go over standard copper, or do they need to run new cabling to you? |
09:14.21 | Qwell | footnote: ahh |
09:14.34 | tuxinator_linux | Qwell: T1 circut |
09:14.38 | Insanity5 | Qwell - Copper. Sometimes a new wire from the junction box, capacity permitting. |
09:14.46 | Qwell | What is the junction box? |
09:14.57 | Insanity5 | Qwell - Look outside. Carefully. It's close by. |
09:14.58 | footnote | Qwell: oops, slot 16 |
09:14.59 | Qwell | where, I should say |
09:15.02 | tuxinator_linux | The box outsite your house, or any box on the way to the CO |
09:15.12 | Qwell | little green guy? |
09:15.12 | footnote | it toggles from overhead to CCS i think |
09:15.21 | footnote | heck, i'd have to go look now :) |
09:15.45 | Insanity5 | Qwell - Typically disquised that way, yes. Sometimes it can be cable/electric though. |
09:15.55 | Insanity5 | Qwell - If you're in a very ritzy area, you might have to dig. |
09:16.01 | Insanity5 | IE: Underground :) |
09:16.09 | Qwell | and who usually pays for that? |
09:16.14 | Qwell | not the telco, I would imagine |
09:16.19 | Insanity5 | They do, with contract. |
09:16.21 | Insanity5 | Generally. |
09:16.23 | Qwell | oh |
09:16.31 | Qwell | large contract, I assume? |
09:16.37 | Insanity5 | 12 months at least. |
09:16.39 | tuxinator_linux | Qwell: When I got my T1 to my house many years back, they had to dig up my yard and put a box in my backyard because there wasn't enough capacity |
09:16.40 | Qwell | Seems that only 2-3 years wouldn't cover the cost of digging |
09:16.42 | Insanity5 | Depends on what you want to pay and the company. |
09:16.52 | Duckbizkit | <PROTECTED> |
09:16.52 | footnote | Dreaded words: "trenching required" |
09:16.57 | footnote | esp PacHell |
09:17.04 | Insanity5 | Qwell - A) It's an investment in the infracstructure in teh area B) Sooner or later, it will happen. |
09:17.05 | Qwell | What if you have fiber in your area? |
09:17.08 | Duckbizkit | i took one drink and about spit it across the room. i added about 1/4 milk to it and it STILL tasted worse than any straight coffee i've ever had. |
09:17.11 | Qwell | Insanity5: good point |
09:17.16 | Insanity5 | Qwell - Out of your league. |
09:17.23 | Insanity5 | Qwell - And if you're in a busienss district, it's probably out there. |
09:17.26 | Qwell | Insanity5: well no, I mean... |
09:17.35 | Qwell | if there is fiber under you, will they use that instead of copper? |
09:17.41 | tuxinator_linux | Qwell: nope |
09:17.45 | Insanity5 | tuxinator_linux - T-1 to residencial areas usually get through into the penalty box. |
09:17.47 | Qwell | there are several areas around here where they laid fiber recently |
09:17.53 | Insanity5 | Qwell - Can not, will not, no need. |
09:17.57 | Qwell | oh, ok |
09:18.06 | footnote | you can all use my global mesh network |
09:18.10 | Insanity5 | HEhe. |
09:18.20 | footnote | for a fee |
09:18.28 | tuxinator_linux | Insanity5: It was when I was running a web hosting business out of my room. |
09:18.34 | Insanity5 | tuxinator_linux - Business house of the house eh? |
09:18.37 | mitcheloc | i say we make a wireless network from house to house, one fixed antenna on every house in the world, then we can let voip takeover! |
09:18.43 | Insanity5 | tuxinator_linux - Now it's a $99 box at rackshack.net... hehe |
09:18.58 | tuxinator_linux | Yep |
09:18.59 | footnote | mitcheloc: call me when you get the routing schema worked out, i'll invest. |
09:19.16 | Insanity5 | tuxinator_linux - It's amazing how bandwdith dropped in price. |
09:19.18 | Qwell | mitcheloc: 172.023 lng, 109.492 lat, 500' > sealevel |
09:19.19 | footnote | don't violate any patents plz k thx |
09:19.22 | Qwell | mitcheloc: toss me a link |
09:19.22 | mitcheloc | heh screw routing, a simpe linksys 4 port will do the trick |
09:19.28 | footnote | uhm |
09:19.34 | footnote | nevermind :) |
09:19.45 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
09:19.56 | mitcheloc | footnote: i was just kidding ;) |
09:19.57 | tuxinator_linux | Insanity5: Yep, I'm getting a full T1 for $420/month through Netifice |
09:20.03 | Insanity5 | tuxinator_linux - I remember when I threw up a game server on a T-1 in 1998, it was filled instantly due to demand. I put one up in 2005 and there's like 50 million servers and 5 million players. |
09:20.04 | footnote | nugget must be getting old. |
09:20.12 | Insanity5 | tuxinator_linux - Still using the T-1? |
09:20.15 | tuxinator_linux | Insanity5: THey have a great SLA |
09:20.20 | tuxinator_linux | to the house? |
09:20.23 | Insanity5 | yes |
09:20.23 | footnote | I can remember him hax0ring away until 6AM |
09:20.30 | tuxinator_linux | Insanity5: Nope, long gone |
09:20.33 | Insanity5 | Or for webhosting for any crazy reason. |
09:20.44 | Insanity5 | webhosting needs to be moved to the bandwidth :P |
09:21.16 | Insanity5 | Bandwidth is about three things: Location, location, location. |
09:21.20 | tuxinator_linux | yep |
09:21.25 | mitcheloc | full t1s? they are like $289 or something |
09:21.29 | tuxinator_linux | Next to fiber is nice |
09:21.29 | NoCAT | tuxinator there are still people who will pay to have private access to game servers |
09:21.34 | Insanity5 | mitcheloc - Minus the pipe. |
09:21.38 | footnote | move next to a fire station or hospital. |
09:21.45 | tuxinator_linux | NoCAT: oh ya? |
09:21.53 | footnote | NOT a police station. |
09:21.57 | mitcheloc | Insanity5: not sure...i think my work pays only $289 total |
09:21.57 | Qwell | footnote: firestations have bandwidth? |
09:22.00 | Insanity5 | lol |
09:22.04 | mitcheloc | * ex-work =) (just quit) |
09:22.08 | footnote | they might smell the pot... |
09:22.12 | NoCAT | not many, but i know some a few groups. |
09:22.30 | Insanity5 | NoCAT - They do, and thats' why there's a proliferation of game servers. I remember when it took more than paying $40/month to get a gameserver... namely it was bandwidth that was the issue. |
09:22.30 | footnote | Qwell: they're usually connected and have power outage privileges |
09:22.32 | tuxinator_linux | There are so many cheep T1 providers with crappy sevice |
09:22.35 | tuxinator_linux | like XO |
09:22.39 | Qwell | footnote: power outage privs? |
09:22.48 | footnote | rolling blackout immunity |
09:22.48 | Qwell | like, first back up? |
09:22.51 | Qwell | ahh |
09:22.51 | NoCAT | does qwest use interleaving on their t1 lines? |
09:23.06 | Insanity5 | NoCAT - dsl, yes, minimum 60ms ping back. |
09:23.13 | Insanity5 | There's a lot of half-duplex t-1's on the market today too. |
09:23.16 | Insanity5 | 2 wire crap. |
09:23.28 | footnote | huh? |
09:23.28 | tuxinator_linux | eww |
09:23.35 | Qwell | footnote: come to think of it, when I lived at my moms, we never got hit with a rolling blackout |
09:23.45 | Qwell | less then a block away from the fire station |
09:23.51 | footnote | Qwell: california? |
09:23.53 | Qwell | yeah |
09:23.58 | Insanity5 | Qwell - Sometimes they'll just tag the circuit. |
09:24.02 | footnote | yep, that was it then |
09:24.06 | Qwell | hmm |
09:24.10 | Insanity5 | tuxinator_linux - Eww on what, teh two wire or latency? hehe |
09:24.13 | Qwell | interesting, never realized that |
09:24.16 | Insanity5 | tuxinator_linux - Is level3 bandwidth good stuff? |
09:24.20 | footnote | urm |
09:24.22 | tuxinator_linux | half duplex |
09:24.24 | Qwell | I doubt I'm close enough now. ;/ |
09:24.26 | footnote | it's not T1 |
09:24.27 | NoCAT | blackouts were fake |
09:24.31 | Insanity5 | tuxinator_linux - Ya, it's nasty. |
09:24.43 | footnote | NoCAT: yeah, cheney ken lay, et al |
09:24.43 | tuxinator_linux | Insanity5: Level3 looks good |
09:24.44 | mitcheloc | Insanity5: are you looking for hosting a server? |
09:24.53 | Insanity5 | Have one hosted on :) |
09:24.57 | Insanity5 | mitcheloc - Why? |
09:24.59 | Insanity5 | hehe |
09:25.04 | NoCAT | can't forget arnold |
09:25.09 | footnote | ah-nald |
09:25.11 | NoCAT | arnold for president..haha |
09:25.12 | mitcheloc | oh, well www.calpop.com has the best bandwith, and worst reliability, just in case you were looking =p |
09:25.36 | Qwell | mitcheloc: let me guess, ex-work? :p |
09:25.36 | Insanity5 | Well I can pull 20megabit... that's way more than I need. |
09:25.40 | tuxinator_linux | mitcheloc: like cox, 5M d/l when its working |
09:25.42 | footnote | I've already seceded my apartment. |
09:25.52 | Insanity5 | I'll push 4gb a month, if I'm lucky. |
09:25.58 | mitcheloc | i used to volunteer there, but the ex-work was somewhere else |
09:26.01 | Qwell | ahh |
09:26.05 | footnote | Sovereign State of Footnote. |
09:26.12 | Qwell | so, ex-volunteer-work |
09:26.19 | tuxinator_linux | Kitty just laied a nasty poo, can't breathe |
09:26.26 | Qwell | footnote: How near is that to the Federation of Qwell? |
09:26.28 | *** join/#asterisk Cresl1n_ (~matt@68.159.151.148) |
09:26.30 | mitcheloc | yea, but hey it was a great learning experience |
09:26.32 | Insanity5 | tuxinator_linux - Better clean up the mess. |
09:26.35 | Cresl1n_ | hey b's |
09:26.36 | Cresl1n_ | :-) |
09:26.39 | footnote | Qwell: we can work out a trade agreement. |
09:26.40 | mitcheloc | i learned so much in one year that it's just crazy |
09:26.46 | tuxinator_linux | Insanity5: Litter maid, have to wait 10 minutes |
09:26.58 | Insanity5 | Go hit the button! |
09:26.59 | Insanity5 | lol |
09:27.01 | footnote | Qwell: have your people call my people |
09:27.01 | Qwell | footnote: I'm not really one for international treaties |
09:27.03 | Insanity5 | unplug-plug it in |
09:27.04 | Insanity5 | hehe |
09:27.06 | tuxinator_linux | Insanity5: No button |
09:27.17 | Insanity5 | There has got to be a manual function. |
09:27.21 | tuxinator_linux | Insanity5: That would work |
09:27.27 | Cresl1n_ | exit |
09:27.41 | tuxinator_linux | ah man, what did she eat |
09:27.45 | footnote | Qwell: Ok, well I'll see you at the economic summit then |
09:27.53 | Insanity5 | Whatever you feed her / forgot to put away. |
09:27.55 | *** join/#asterisk ptblank (~MURDER1@68-169-176-29.lmdaca.adelphia.net) |
09:28.02 | tuxinator_linux | Any of you going to Meet *? |
09:28.09 | NoCAT | if you can only have 24 simultaneous calls on a voice t1. does running voip allow you to recieve more calls? |
09:28.19 | Qwell | NoCAT: If its not over the T1, sure |
09:28.23 | tuxinator_linux | NoCAT: 23 calls |
09:28.29 | NoCAT | sorry |
09:28.31 | NoCAT | 23 |
09:28.34 | footnote | hehe, back in the IBM mainframe days we called the '*' a 'splat' |
09:28.35 | jmhunter | kram |
09:28.39 | Insanity5 | NoCAT - If they call your voip # |
09:28.41 | Insanity5 | lol |
09:28.53 | tuxinator_linux | NoCAT: VoIP can compress more calls in |
09:29.02 | soulz- | dial_exec_full: Unable to create channel of type 'SIP' |
09:29.04 | mitcheloc | anyone here use an ibm laptop?, i'm lookin for a new one, and they caught my eye |
09:29.06 | soulz- | what does this error mean? |
09:29.09 | Insanity5 | tuxinator_linux - Ih ave it when backbone providers forget to set a reverse dns, or leave an outdate reverse dns on a route of a company that they existed as many years ago. |
09:29.11 | tuxinator_linux | NoCAT: but is it worth it? |
09:29.20 | Insanity5 | mitcheloc - They're nice, durable, full features, lightweight, and EXPENSIVE. |
09:29.37 | Insanity5 | tuxinator_linux - "ih ave it" = "I have it" :) |
09:29.39 | Qwell | a decent thinkpad can run $2k easy |
09:29.40 | mitcheloc | true, but worth the money? |
09:29.43 | footnote | soulz-: after making a few calls? |
09:29.43 | Insanity5 | GRRR - hate it :) |
09:29.46 | Insanity5 | Can't flipped type |
09:29.47 | Insanity5 | lol |
09:29.48 | *** join/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr) |
09:29.52 | mitcheloc | i was looking at the x40 |
09:29.52 | soulz- | footnote: yeah |
09:29.56 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
09:29.58 | tuxinator_linux | I like my toshiba |
09:30.01 | mitcheloc | they are going to have the new one in april come with the fingerpring scanner =) |
09:30.05 | Insanity5 | t41 ot 42 or whatever it is looks nice. |
09:30.07 | mitcheloc | *fingerprint |
09:30.14 | Qwell | I might like my Dell...if they would have sent me a freaking power cable |
09:30.15 | footnote | soulz-: sounds sorta like something is hanging and not releasing channels? |
09:30.16 | soulz- | footnote: i have this error after i updated my cvs and also updated the invite on bv |
09:30.26 | tuxinator_linux | I'm not a fan of Dells |
09:30.28 | NoCAT | tuxinator what do you mean is it worth it? i'm not sure how it works exactly. it can compress more calls but your still limited to the number of calls you can recieve by the amount of availble lines on the t1 |
09:30.33 | Insanity5 | tuxinator_linux - How is time warner b/w? |
09:30.34 | footnote | doesn't sound good |
09:30.42 | soulz- | tell me about it:P |
09:30.51 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
09:30.58 | tuxinator_linux | Insanity5: What about Time Warner? |
09:31.00 | footnote | i haven't figured out gdb usage yet |
09:31.08 | footnote | haven't tried |
09:31.18 | tuxinator_linux | NoCAT: You have to worry about sound quality and latency |
09:31.19 | Qwell | NoCAT: If its going from your voip provider, to your * box, and out to internal extensions, it won't hit the T1 at all |
09:31.28 | footnote | a nifty set of * gdb macros would be a good thing |
09:31.31 | NoCAT | tuxinator even on a lan? |
09:31.34 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
09:31.40 | tuxinator_linux | NoCAT: with a PRI, you dont' have those problems |
09:31.46 | NoCAT | what is a pri? |
09:31.51 | Qwell | ~pri |
09:31.52 | jbot | somebody said pri was Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc. |
09:32.14 | Insanity5 | tuxinator_linux - How is time warners bandwidth? |
09:32.16 | tuxinator_linux | NoCAT: on a lan, you use SIP, AIX, etc |
09:32.20 | footnote | NoCAT: complicated stuff the phone company created to prevent competition |
09:32.34 | tuxinator_linux | Insanity5: Haven't lived in a Timer Warner area |
09:32.54 | Insanity5 | tuxinator_linux - Sombody has to have a rate the bandwidth provider page... lol |
09:33.15 | Qwell | Insanity5: like broadbandreports.com? |
09:33.21 | tuxinator_linux | Insanity5: I was thinking about doing a rate a VOICE/INTERNET/ETC provider page |
09:33.22 | mitcheloc | yep www.webhostingtalk.com |
09:33.25 | Insanity5 | Qwell - Yes, but for major backbone providers. |
09:33.29 | Qwell | oh |
09:33.42 | Insanity5 | level3/qwest/uunet/etc... |
09:33.45 | Qwell | Insanity5: simply put, they all suck :p |
09:33.47 | mitcheloc | wht is where the serious people go and talk about the stuff |
09:33.50 | Qwell | ^^overgeneralization |
09:33.53 | tuxinator_linux | UUNET network is nice |
09:33.59 | tuxinator_linux | Used it while using MCI |
09:34.16 | tuxinator_linux | Genuity was bought by Netifice |
09:34.27 | Insanity5 | genuity godo or bad? |
09:34.30 | tuxinator_linux | good |
09:34.39 | Insanity5 | Used to have them before level3 |
09:34.44 | Insanity5 | In the whole transition mess. |
09:34.57 | tuxinator_linux | Insanity5: Who did you have? |
09:35.20 | mitcheloc | if you want real solid bandiwth though, you have to go straight to a provider, like pacbell, att, something like that |
09:35.20 | tuxinator_linux | I had PSInet to the house |
09:35.22 | mitcheloc | not a reseller |
09:35.26 | Qwell | Genuity ha[sd] good bandwidth, right? |
09:35.37 | Insanity5 | tuxinator_linux - Genuity direct for b/w in a chicago datacenter, ran our own transport back. |
09:35.39 | tuxinator_linux | mitcheloc: I dont' think that is entirely true |
09:35.43 | Insanity5 | tuxinator_linux - Then level3./bbnplanet |
09:35.57 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
09:36.07 | *** part/#asterisk Duckbizkit (~DMAN@ip-216-97-163-53.valornet.com) |
09:36.09 | mitcheloc | i don't know, i think it's pretty accurate, resellers add one more point of possible failure |
09:36.11 | tuxinator_linux | level3 owns a lot of fiber |
09:36.21 | Insanity5 | Yes. |
09:36.29 | tuxinator_linux | mitcheloc: but they get the ILEC to be more responsive |
09:36.29 | Insanity5 | I wonder if latency will ever get lower, or if speed of light is the limit. |
09:36.33 | Insanity5 | Gah :) |
09:36.44 | Qwell | Insanity5: well, it is fiber. ;] |
09:36.48 | Insanity5 | hehe |
09:36.49 | tuxinator_linux | multiplexing light |
09:37.03 | Insanity5 | Yes... Never becoming any faster... |
09:37.03 | mitcheloc | sweet!!! just finished setting up my door, a $2/month broadvoice # lets me in/out of my front door =) |
09:37.08 | tuxinator_linux | various bandwidths of light |
09:37.17 | Qwell | its also going through a bunch of silicon, isn't it? |
09:37.22 | mitcheloc | next stop is my car, i hate keys! |
09:37.23 | Insanity5 | But chicago > LA over timewarner taking 75 ms is high. They should have cross-country down to 40-50ms. |
09:37.30 | *** part/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr) |
09:37.33 | Insanity5 | mitcheloc - LOL, you geek. |
09:37.49 | tuxinator_linux | Ahhh, kitty poo is gone |
09:37.51 | Insanity5 | mitcheloc - Any authentication, or just call it? hehe. |
09:37.54 | mitcheloc | heh, the worst part is that i have 6 roommates and i forced them all into using their cell phones to get into the house =) |
09:38.05 | Qwell | wtf |
09:38.09 | Qwell | what if bv is down? :P |
09:38.11 | Insanity5 | mitcheloc - Better carry a spare battery. |
09:38.14 | mitcheloc | callerid auth, and yes i know it's fakeable, but you have to know the number and the numbers to fake too |
09:38.18 | tuxinator_linux | mitcheloc: weirdo, he he |
09:38.23 | *** join/#asterisk RoyK (~roy@80.239.107.121) |
09:38.35 | mitcheloc | i have a back up number too that i was using for a while now, it's off a t1 in another city |
09:38.46 | mitcheloc | but just switched it to BV cause it answers faster |
09:38.46 | Qwell | I hate to ask... |
09:38.49 | Insanity5 | Better still hide a key. |
09:38.51 | Qwell | but what if both are down? |
09:38.59 | mitcheloc | then they can't come in the house, too bad |
09:39.02 | tuxinator_linux | He could just break a window |
09:39.04 | Qwell | heh |
09:39.10 | Insanity5 | Rock? :) |
09:39.11 | Qwell | or turn the knob manually |
09:39.17 | tuxinator_linux | Qwell: too much work |
09:39.22 | Qwell | Insanity5: cellphone would already be in hand |
09:39.25 | mitcheloc | nope, the knob is fixed, i used an allen wrench to make quick work on that |
09:39.27 | Qwell | cellphone would be a good weapon |
09:39.31 | Insanity5 | You should rig it so the door springs open :) |
09:39.38 | mitcheloc | heh ;) |
09:39.42 | Insanity5 | Qwell - Waste of money, cmon now, that'd cost more than a new window. |
09:39.45 | mitcheloc | i don't need no backup! |
09:39.46 | NoCAT | does asterisk do sms over internet to mobile phones? |
09:39.49 | tuxinator_linux | does it great you when you open the door |
09:39.59 | tuxinator_linux | NoCAT: maybe |
09:40.03 | Qwell | "Hello Tom, breakfast is ready." |
09:40.03 | Insanity5 | NoCAT - Could probably rig it too. |
09:40.07 | mitcheloc | yea it goes BEEP then a pause, about 8 seconds then allison says "THANK YOU" |
09:40.08 | mitcheloc | and hangs up |
09:40.11 | tuxinator_linux | NoCAT: I think there is a wikie for it |
09:40.15 | Insanity5 | usually @messaging.sprintpcs.com or @yourprovider.com does the trick. |
09:40.34 | Insanity5 | mitcheloc - Waht hardware did you use? |
09:40.42 | mitcheloc | mmm i went the expensive route |
09:40.45 | Insanity5 | Not true sms, but a gateway over. |
09:41.06 | mitcheloc | a rabbit + simple relay circuite, a little c programming, and a heavy duty strike from smarthome.com |
09:41.15 | mitcheloc | oh and a lot of time installing the damn thing |
09:41.15 | NoCAT | yeah i'm sorry i shoudln't ask questions i can easiy research.. been reading alot lately. |
09:41.18 | mitcheloc | that takes forever |
09:41.25 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
09:41.33 | Insanity5 | People here are VERY forgiving :) |
09:41.43 | tuxinator_linux | ~rtfw |
09:41.44 | jbot | methinks rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
09:41.54 | Qwell | mitcheloc: rabbit with, or without hamster wheel? |
09:41.54 | tuxinator_linux | no forgivness for you |
09:42.13 | tuxinator_linux | just kidding NoCAT |
09:42.19 | mitcheloc | rabbit without hamster wheel =), mines a smart rabbit, plug her into the lan and access her from wherever |
09:42.25 | Insanity5 | Does anybody have a really neat routing system set up they want to showcase? lol |
09:42.25 | Qwell | nice |
09:42.43 | Qwell | Insanity5: voip, or lan? |
09:42.49 | Insanity5 | Qwell - Any. |
09:42.57 | Qwell | I have a crappy routing system. :p |
09:43.01 | mitcheloc | i have a gps, it routes me well |
09:43.03 | tuxinator_linux | hmm, I'm going to go steal some cookies, be back in a minute |
09:43.07 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com) |
09:43.09 | Insanity5 | there's got to be a good reposity of good voice samples (hint: sexy female voice) somewhere :) |
09:43.19 | Qwell | sparcstation with a quad sbus ethernet card - 4 different subnets |
09:43.22 | mitcheloc | heh...and i thought he meant browser cookies for a second |
09:43.29 | moonwick | do what I do. have your girlfriend record the samples. |
09:43.34 | mitcheloc | Insanity5: your welcome to pay my gf ;) |
09:43.47 | Insanity5 | lol |
09:43.51 | Qwell | mitcheloc: dating allison, are we? |
09:43.54 | mitcheloc | Insanity5: she does a good job, and i have a good phone for her to use to record. |
09:43.58 | mitcheloc | bah! |
09:44.02 | Insanity5 | Automated systems must have females answering :) |
09:44.04 | mitcheloc | ** eww no thanks |
09:44.16 | moonwick | I've actually found that the perfect microphone for recording sounds for asterisk is just a pair of headphones. |
09:44.17 | Qwell | I need sleep |
09:44.18 | Insanity5 | mitcheloc - I figured a headset would be best... or .wav files |
09:44.25 | mitcheloc | yea, it's the only relief we have from not seeing girls while sitting infront of the computer debugging asterisk heh |
09:44.46 | mitcheloc | i prefer a cisco 7960 for recording |
09:44.57 | Qwell | moonwick: heh, you just reminded me of a funny story |
09:44.59 | Insanity5 | Do you have * pick up everything even before it rings (HOLD, CONENCTING CALL), does it get rid of telemarketers, etc? |
09:45.20 | mitcheloc | telemarketers are fun to talk to, why wouldn't you want them? |
09:45.24 | Qwell | moonwick: verizon shut me off once, but I still had DSL. I had to use dialpad.com(back when they were free) to call verizon to get my phone back. |
09:45.34 | Qwell | moonwick: But, I didn't have a mic, and I owned one pair of headphones |
09:45.57 | mitcheloc | record a track that goes "uh huh.......mmm.....yea.... that sounds good..... maybe.....mmm.....sorry...can you repeat that?" |
09:46.00 | Qwell | I would nod to my friend, and he would switch from mic to speaker port...ugh |
09:46.00 | mitcheloc | then transfer them there |
09:46.14 | Insanity5 | mitcheloc - LOL, do you do that? |
09:46.17 | tuxinator_linux | <PROTECTED> |
09:46.20 | mitcheloc | no, but i should |
09:46.23 | Insanity5 | mitcheloc - I would love to place them in a satiracle hold loop. |
09:46.43 | moonwick | heh |
09:47.43 | tuxinator_linux | mitcheloc: How did it workout? |
09:47.49 | mitcheloc | well......if you screw up the asterisk box, you can't connect to it, cause you lose the t1 connection |
09:47.56 | mitcheloc | other then that it works great |
09:48.17 | tuxinator_linux | I thought about that |
09:48.27 | mitcheloc | i did that |
09:48.33 | tuxinator_linux | That is why I am getting two T1 circuits and and PRI |
09:48.36 | mitcheloc | big big pain |
09:48.40 | Insanity5 | mitcheloc - call 1-866-836-0971 - lol pick an option |
09:48.45 | tuxinator_linux | two T1 for BGP |
09:48.55 | mitcheloc | by integrated i mean half hdlc data and half phone lines |
09:49.01 | *** join/#asterisk Bonbon (~bonbon@83.146.53.34) |
09:49.22 | *** join/#asterisk benno2 (~benno2@host31-15.pool80182.interbusiness.it) |
09:49.30 | tuxinator_linux | mitcheloc: I know, that is what I have right now through XO, and it never stays up |
09:49.42 | Bonbon | has anyone done any tapi integration with xlite? |
09:49.45 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
09:49.49 | footnote | boy, this is a really scary pic of tux |
09:49.57 | footnote | http://www.satanic.org |
09:49.57 | mitcheloc | scratchy |
09:49.58 | Zeeek | hey drumkilla? |
09:50.00 | Insanity5 | mitcheloc- I like option 8 :) |
09:50.09 | mitcheloc | heh |
09:50.16 | Insanity5 | mitcheloc - Did you call it? |
09:50.19 | tuxinator_linux | I called it to |
09:50.24 | file | ha |
09:50.28 | mitcheloc | hey my gf should do your voices, cause that chick hurts my ears |
09:50.33 | footnote | mmm brownies |
09:50.34 | mitcheloc | whats with the monkeys? |
09:50.37 | Insanity5 | It's not mine :) |
09:50.38 | Insanity5 | rofl |
09:50.51 | Insanity5 | Came from this: http://216.239.63.104/search?q=cache:7WPkq6zWSqQJ:www.careerbuilder.com/JobSeeker/Jobs/JobDetails.aspx%3FJob_DID%3DJ8F1576JD0Y1PM5Q4F2+poo+careerbuilder&hl=en |
09:50.55 | file | mitcheloc: ah yes your bugnote... it's because asterisk does blocking DNS lookups, no way not to at the moment - so make sure you have stable DNS servers |
09:50.56 | mitcheloc | thats a real good quality call, what do you use? |
09:51.07 | *** join/#asterisk visik7 (~ciao@visik7.user) |
09:51.17 | Insanity5 | here's their webistE: http://www.yeknominc.com/ |
09:51.19 | Insanity5 | lol |
09:51.41 | mitcheloc | file: oh i'm, i'm sorry, then you knew about it, i definately use more stable dns servers, just didn't know if anyone knew about this |
09:51.47 | *** join/#asterisk TheJudge (JTR@209-203-52-3.network.ods.co.za) |
09:51.57 | mitcheloc | * will use |
09:51.59 | Insanity5 | mitcheloc - It's not mine... I was just remembered it from a while back. |
09:52.04 | benno2 | question: is it possible to define an extension that if it gets called, the called party (not the caller) first hears a short message and then the call gets connected. that way the called person can know where the call comes from. (distinctive ring is not an option because the extension could be redirect on a cellphone) |
09:52.23 | mitcheloc | file: i guess mostly it was just it was hard to figure out what was the source of my problems, perhaps a comment in the notice to point users the right way |
09:52.40 | TheJudge | hello all |
09:52.48 | TheJudge | what sould I set my jitter size to |
09:52.49 | TheJudge | ? |
09:52.53 | TheJudge | or how does it work |
09:52.57 | Insanity5 | mitcheloc - Was your comment about the good voice from the yakmov industries joke? |
09:53.14 | file | everyone else is going to sleep, I should do the same I suppose |
09:53.16 | tuxinator_linux | I hate being on hold |
09:53.24 | mitcheloc | Insanity5: no, just the quality of the call |
09:53.25 | tuxinator_linux | the monkeys are interesting though |
09:53.25 | *** join/#asterisk RoyKa (~roy@80.239.107.80) |
09:53.37 | file | bkw is dead to the world |
09:53.40 | file | twisted is the same |
09:53.43 | file | only me left |
09:53.45 | Insanity5 | tuxinator_linux - lol. It's almost a game. |
09:53.51 | *** join/#asterisk chaven (~cbalzac@evil.maas-biolabs.net) |
09:53.57 | Insanity5 | tuxinator_linux - They even have tv ads on their fake website :P |
09:54.03 | mitcheloc | file: we can keep you company ;) |
09:54.09 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
09:54.12 | file | hehe |
09:54.20 | TheJudge | I am using broadvoice |
09:54.28 | TheJudge | how do you work out jitter |
09:54.29 | TheJudge | ? |
09:54.36 | Insanity5 | TUX - I want something like that to transfer a telemarket too though. |
09:54.39 | mitcheloc | i bet they are resting for von tomorrow |
09:54.44 | footnote | TheJudge: empirically or mathematically? |
09:54.54 | TheJudge | lol |
09:54.57 | TheJudge | either |
09:55.01 | TheJudge | what does it do ? |
09:55.03 | *** part/#asterisk jmhunter (~jmhunter@64.77.199.223) |
09:55.04 | footnote | well, is it working? |
09:55.16 | TheJudge | call is jittery ? |
09:55.22 | TheJudge | like static on the line ? |
09:55.27 | footnote | pops like slips |
09:55.36 | tuxinator_linux | Insanity5: Me too |
09:56.15 | RoyKa | footnote: using what? sip? |
09:56.15 | tuxinator_linux | Those videos were on during the superbowl |
09:56.35 | mitcheloc | what was the purpose of the website then? |
09:56.54 | footnote | a "slip" happens when your amount of jitter is larger than the buffered amount of time |
09:56.59 | footnote | or something like that |
09:57.02 | TheJudge | ok |
09:57.12 | footnote | so, make it twice as big |
09:57.16 | footnote | see what happens |
09:57.21 | Insanity5 | tuxinator_linux - They also placed job descriptions like "flung poo collector" thorughout their site :P |
09:57.29 | footnote | i haven't actually done this for asterisk yet :) |
09:57.32 | TheJudge | I have cisco router, how can I get it to prio sip trafffic ? |
09:57.34 | mitcheloc | footnote: whats the link? |
09:57.41 | Insanity5 | TheJudge - By port? |
09:57.54 | file | oh no bkw is snoring |
09:58.00 | footnote | TheJudge: paypal Insanity5 he's got Cisco scoop |
09:58.01 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
09:58.15 | Insanity5 | rofl |
09:58.24 | Insanity5 | Cisco is just a PITA. |
09:58.29 | footnote | Insanity5: i expect a kickback this time dammit |
09:58.38 | Insanity5 | hehe |
09:58.46 | TheJudge | cool, but what port ? |
09:58.54 | TheJudge | does it not choose a random port ? |
09:58.55 | Qwell | file: Now seems to be the perfect time to mess with him |
09:59.07 | Qwell | ok, bed...for real this time |
09:59.14 | file | lol |
09:59.14 | *** join/#asterisk mithro (~tim@dsl1-83.gw1.adl1.airnet.com.au) |
09:59.38 | tuxinator_linux | Qwell: what are going to do with this 'bed'? |
09:59.39 | Insanity5 | TheJudge - Use it based on server port. At least with most, you can't due true QoS on the application level, but you can infer that voip SIP traffic will stay on it's port. Add it to an access list, and give it priority. |
09:59.44 | tuxinator_linux | ~bed |
09:59.45 | jbot | bed is, like, a thing programmers have never heard of, ask me about shower |
09:59.55 | footnote | mithro: is it true that vegemite makes good axle grease? |
10:00.03 | mitcheloc | ~shower with me |
10:00.05 | tuxinator_linux | footnote: sure do |
10:00.07 | benno2 | ok found the solution, cmd dial: A(x): Play an announcement (x.gsm) to the called party. |
10:00.08 | benno2 | :) |
10:00.10 | Insanity5 | And I'm assuming you're talking about a router. |
10:00.11 | TheJudge | tahnks |
10:00.14 | TheJudge | yes |
10:00.15 | mitcheloc | mmm jbot didn't like that |
10:00.17 | mitcheloc | ~shower |
10:00.18 | jbot | extra, extra, read all about it, shower is man using one hand in a very usefull way |
10:00.31 | mitcheloc | ~hand |
10:00.32 | TheJudge | so eg port 41000 and give it priority |
10:00.41 | RoyKa | ~lart mitcheloc |
10:00.41 | mitcheloc | ~useful? |
10:00.53 | mitcheloc | ~thank you? |
10:00.53 | jbot | pas de quoi, mitcheloc |
10:01.05 | mitcheloc | ~mitcheloc |
10:01.08 | Insanity5 | yes, and you'll have to classify "everything else" in another access group. Cisco is quirky, but you set two numbers and it'll get say, 7 packets of one to 1 packet of another. |
10:01.24 | tuxinator_linux | jbot habla espanol? |
10:01.38 | footnote | jbot not messican |
10:01.49 | Insanity5 | TheJudge - If you enver a newer IOS, try looking at Weighter fair queueing. |
10:01.57 | mitcheloc | status |
10:02.01 | file | I should plug my laptop in |
10:02.21 | tuxinator_linux | file: Good Idea |
10:02.23 | TheJudge | thanks |
10:02.29 | file | and sleep |
10:02.32 | footnote | I'm finally updating my PowerBook to 10.2 |
10:02.35 | tuxinator_linux | ~sleep |
10:02.36 | jbot | from memory, sleep is overrated, and a poor substitute for caffeine |
10:02.45 | tuxinator_linux | ~mac |
10:02.46 | jbot | mac is, like, the best computer ever |
10:02.56 | tuxinator_linux | hmmm |
10:03.05 | footnote | jbot is a genius! |
10:03.19 | Insanity5 | TheJudge - Start by making two access control lists, everythign else, and the voip traffic and go from there. |
10:03.24 | tuxinator_linux | One silly button on the mouse?? |
10:03.24 | mitcheloc | jbot is smart |
10:03.25 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
10:03.27 | mitcheloc | jbot: what is 60 mph in angstroms per femtosecond? |
10:03.29 | jbot | mitcheloc: I think you lost me on that one |
10:03.31 | TheJudge | ok eg |
10:03.32 | mitcheloc | ~what is 60 mph in angstroms per femtosecond |
10:03.34 | jbot | I think you lost me on that one, mitcheloc |
10:03.39 | TheJudge | match iup all all in one |
10:03.41 | mitcheloc | he answered me in private! |
10:03.42 | footnote | tuxinator_linux: mainly because it's BSD based and not derivative of linux |
10:03.43 | Insanity5 | TheJudge - Also just enablign "fair-queue" can sometimes help a bit. |
10:03.45 | mitcheloc | now he won't do it in public =/ |
10:03.46 | TheJudge | and voip in the other |
10:04.03 | mitcheloc | lets start off simple |
10:04.06 | Insanity5 | TheJudge - YOu got it... do it by port number or server ip. |
10:04.09 | mitcheloc | jbot: what is 2+2? |
10:04.10 | jbot | 4 |
10:04.13 | mitcheloc | good boy |
10:04.21 | mitcheloc | jbot: now what is 2*4? |
10:04.23 | jbot | mitcheloc: okay |
10:04.23 | TheJudge | do you know what port broadvoice use ? |
10:04.30 | mitcheloc | jbot: WHAT IS 2*4? |
10:04.32 | jbot | mitcheloc: what are you talking about? |
10:04.36 | riksta | hahaha |
10:04.40 | mitcheloc | jbot: 2*4? |
10:04.44 | footnote | uh oh, it's in eliza mode |
10:04.50 | mitcheloc | damnit! YOU MECHANICAL USELESS MONSTER! |
10:04.57 | mitcheloc | jbot: byte me! |
10:04.59 | jbot | /me bytes mitcheloc |
10:04.59 | tuxinator_linux | ~cry |
10:05.00 | jbot | ACTION cries and sobs until he nearly drownds in his own tears |
10:05.00 | Insanity5 | TheJudge - It's SIP. google it. |
10:05.05 | mitcheloc | noo! |
10:05.18 | riksta | mitcheloc: if it's any help...i think the answer is 8 ;) |
10:05.25 | mitcheloc | nickometer bkw |
10:05.30 | tuxinator_linux | ~kill jbot |
10:05.33 | jbot | ACTION shoots a ionized fluxproton gun at jbot |
10:05.42 | Insanity5 | TheJudge - You may also just be able to associate the class map with the protocol instead of relying on an access list... but I seriously doubt it. |
10:05.46 | mitcheloc | riksta: thanks, but i'm tring to teach jbot, he will learn! |
10:05.48 | tuxinator_linux | not a very smart monster |
10:05.56 | footnote | oh, kill the messenger! |
10:05.58 | footnote | hehe |
10:06.03 | Insanity5 | TheJudge - If you go the WFQ route. |
10:06.07 | TheJudge | yea |
10:06.08 | TheJudge | ? |
10:06.09 | mitcheloc | theres got to be something useful on this page so that we can destroy jbot!!!! https://jbot.dev.java.net/jbot-user-guide.html |
10:06.17 | TheJudge | thanks |
10:06.19 | *** part/#asterisk chaven (~cbalzac@evil.maas-biolabs.net) |
10:06.26 | footnote | mitcheloc: oh, it's a java bot |
10:06.35 | footnote | it'll self destruct, be patient |
10:06.50 | Insanity5 | TheJudge - But I don't think the router will allow SIP as a protocl. I seriously don't know. |
10:07.02 | tuxinator_linux | ~bullshit |
10:07.03 | jbot | from memory, bullshit is If you want to speak bullshit, please go to #debian.bullshit. sdf dflkj Linux sucks sfg yo momma dfg #debian.bullshit |
10:07.07 | mithro | footnote: dunno, i'd rather eat axle grease then vegemite |
10:07.12 | footnote | hahahah |
10:07.25 | Insanity5 | TheJudge - If you have neough, you may jsut want to dedicate enough bandwidth soely to the VOIP cause. IF not, let the router handle availability. |
10:07.30 | mitcheloc | insult footnote |
10:07.31 | footnote | sounds like jim dixon must have set jbot up |
10:07.33 | mitcheloc | ~insult footnote |
10:07.39 | footnote | haha |
10:07.50 | mitcheloc | ~insult jbot |
10:07.53 | *** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
10:08.03 | footnote | ~rape mitcheloc |
10:08.07 | footnote | eep |
10:08.15 | tuxinator_linux | ~qos |
10:08.16 | jbot | extra, extra, read all about it, qos is Quality of Service, a great source of information is located @ http://www.lartc.org |
10:08.16 | mitcheloc | ~ebonify footnote |
10:08.17 | footnote | hehe |
10:08.18 | *** join/#asterisk The_Ball (~alex@static-100.35.240.220.dsl.comindico.com.au) |
10:08.27 | mitcheloc | ~ebonify |
10:08.36 | footnote | Slap mah fro! |
10:08.44 | Insanity5 | TheJudge - create acl -- ie access list permit udp all all range (sip port start here) (sip port end here). check the syntax of the all all part, it's probaby wrong. |
10:08.44 | tuxinator_linux | ~ebonify I need to get some sleep pretty soon |
10:08.58 | The_Ball | Hi, can anyone help me with a newbie dialplan question, here is my five line dialplan: http://channels.debian.net/paste/322 how can i make the phone start to ring at the very begining? |
10:08.58 | footnote | hrm jive | festival for asterisk |
10:09.02 | mitcheloc | i need to a get uh some uh sleep uh soonah |
10:09.23 | mitcheloc | nevermind, *sleep=sex |
10:09.31 | Insanity5 | TheJudge - Create class maps that encompass each access list -- class map class1^M Match Access-group 101^M (^M= enter). |
10:09.48 | tuxinator_linux | ~time |
10:09.49 | jbot | rumour has it, time is 1 dimensional, or everlasting |
10:10.08 | footnote | ~quantum mechanics |
10:10.11 | jbot | methinks quantum mechanics is the devil's arithmetic |
10:10.11 | mitcheloc | jbot: time is the 4th dimension stupid! |
10:10.13 | jbot | ...but time is already something else... |
10:10.32 | mitcheloc | jbot: NO it's not! |
10:10.42 | mitcheloc | jbot: time is change |
10:10.43 | jbot | ...but time is already something else... |
10:10.43 | tuxinator_linux | jbot is a dork |
10:10.49 | footnote | ~particle physics |
10:10.53 | mitcheloc | jbot: change is the only constant |
10:10.59 | mitcheloc | YES IT DOES! |
10:11.09 | tuxinator_linux | jbot, he is right |
10:11.11 | jbot | ...but he is already something else... |
10:11.23 | Insanity5 | TheJudge - Easy way? flat out limit all to less than max b/w, and give voip suffient b/.w - policy map policy1^M (now you have a policy map). class class1^M (class1 we just made, voip traffic) bandwidth 80000^M (80kbit) Queue-limit 30 (play with this value). exit. Repeate for class two. Pray it works. |
10:11.27 | tuxinator_linux | jobi: you want to fight about it |
10:11.45 | tuxinator_linux | jbot: you want to fight about it? |
10:12.07 | mitcheloc | jbot: i'm going to sleep, and i shall kill you in my dreams, good bye |
10:12.08 | Insanity5 | TheJudge - IF you want to get into actually prioritizing and droppign traffic... well good luck it's beyond most of my abilities. Start reading here: http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120t/120t5/cbwfq.htm |
10:12.14 | tuxinator_linux | Insanity5: What lanauge are you speaking? |
10:12.27 | Insanity5 | tuxinator_linux - Cisco :0 |
10:12.29 | Insanity5 | IOs |
10:12.34 | tuxinator_linux | Ewww |
10:12.38 | mitcheloc | Insanity5: qos on a linux router? |
10:12.45 | TheJudge | ta |
10:12.45 | Insanity5 | mitcheloc - No... Cisco IOs |
10:12.49 | footnote | I saw the source for IOS when I was at Cisco. |
10:13.02 | mitcheloc | Insanity5: - No... i mean, USE a linux router for qos |
10:13.02 | footnote | Which is why I'll never buy a Cisco product. |
10:13.06 | Insanity5 | footnote - Now it's on your local peer2peer network lol |
10:13.11 | footnote | nope |
10:13.16 | Insanity5 | mitcheloc - What a concept... I know... he wants to use his cisco. |
10:13.21 | Insanity5 | footnote - An old version is. |
10:13.22 | tuxinator_linux | I have had the best luck with Cisco routers |
10:13.33 | footnote | It's the ugliest piece of crap you'll ever look at |
10:13.43 | Insanity5 | footnote - I beleive you, but it sure is stable. |
10:13.50 | Insanity5 | Very draconian at times. |
10:13.51 | footnote | well |
10:14.02 | footnote | on some products |
10:14.23 | footnote | Avoid powerpc based cisco products |
10:14.30 | mitcheloc | i bought a dlink router...or was it netgear |
10:14.36 | mitcheloc | 10/100/1000 16 port switch |
10:14.46 | mitcheloc | managed, with qos =) |
10:14.49 | footnote | powerpc-based==acquisition |
10:14.50 | mitcheloc | i'll be here thursday |
10:14.59 | Insanity5 | mitcheloc - Yes, with an easy to use gui, lol |
10:15.03 | mitcheloc | thats going in my house, overkill? heh! |
10:15.07 | footnote | mips base, "native" cisco |
10:15.11 | mitcheloc | yep, webmanagement =) |
10:15.51 | Insanity5 | mitcheloc - wrt54g linksys route with hacked firmware here does the job as good as any: http://wrt54g.thermoman.de/ |
10:16.33 | mitcheloc | yea but it's not a gigabit switch |
10:17.20 | footnote | mitcheloc: cmon, you need gigabit for pr0n? |
10:17.24 | mitcheloc | what WOULD be nice mmm..../me drools, can you use the wrt54g as a wireless router and trigger in out pins somehow on the ports? so i can turn on and off relays via a cheap ~$50 single board computer! =) |
10:17.26 | Insanity5 | mitcheloc - No, it's consumer junk lol. but it works. |
10:17.40 | riksta | mitcheloc: yeah the wrt54g is good |
10:17.42 | mitcheloc | yes, for porn |
10:18.02 | Insanity5 | mitcheloc - Take a look at these: http://www.parallax.com/html_pages/products/basicstamps/basic_stamps.asp |
10:18.07 | Insanity5 | mitcheloc - Little computer for those jbos. |
10:18.08 | riksta | personally, i made my own access point |
10:18.08 | Insanity5 | jobs |
10:18.12 | mitcheloc | i have to be able to stream porn to every cat5 jack in my house |
10:18.15 | mitcheloc | * at the same time |
10:18.25 | Insanity5 | hehe |
10:18.30 | Insanity5 | In your VR world. |
10:18.34 | mitcheloc | nonono not LAN aware! |
10:18.36 | footnote | riksta: really? |
10:18.54 | riksta | footnote: yeah, if you buy a decent wireless card, you can put it in "master" mode :) |
10:18.54 | mitcheloc | riksta: pci wireless card + linux router |
10:18.55 | djin | pbx.c:1280 pbx_extension_helper: No application 'SerVar' |
10:18.58 | footnote | oh |
10:19.04 | djin | did I miss something that changed? |
10:19.09 | mitcheloc | djin: setvar not servar |
10:19.13 | riksta | footnote: then you just set a wep key, and an IP and run a dhcp server |
10:19.19 | djin | hahha |
10:19.22 | footnote | gotit |
10:19.40 | footnote | riksta: i thought you laid out your own board or something :) |
10:19.42 | Zeeek | djin don't be so hardon yourself |
10:19.43 | mitcheloc | Insanity5: what do you know about cheap computers that i can plug into my lan, that aren't computers =)? |
10:19.47 | riksta | footnote: naw |
10:20.04 | Insanity5 | mitcheloc - Get some flash disk contraption if that's what your'e really after. |
10:20.18 | mitcheloc | *cheap*!! |
10:20.28 | Insanity5 | mitcheloc - You can interface with one of those stamps via serial. Just jack a couple pair of wire off your ethernet wiring. |
10:20.36 | Insanity5 | Stamps are cheap. |
10:20.39 | footnote | rabbit |
10:20.42 | footnote | z80 like |
10:20.43 | footnote | tcp |
10:20.52 | mitcheloc | yea i use the rabbit |
10:20.59 | mitcheloc | but i was hoping for another cheaper alternative |
10:21.03 | djin | Well Zeeks. Such a stupid mistake, but overlooked because it's in a longer list of Setvar's. |
10:21.05 | footnote | weird c-like thingy eh? |
10:21.05 | Insanity5 | They'rel ike $20 |
10:21.06 | mitcheloc | rabbits are my friends =) |
10:21.07 | mithro | anyone in Australia using a FXO/FXS stuff? |
10:21.20 | djin | Zeeks = Zeeek, sorry. |
10:21.20 | Insanity5 | mitcheloc - I have piles of pentium 2 computers.... pay shipping or you haul... $20 |
10:21.29 | Insanity5 | computers are free lol |
10:21.33 | Zeeek | There are so many Zeeeks |
10:21.34 | mitcheloc | Insanity5: so stamp + wrt54g??? that would be sweet |
10:21.41 | footnote | hrm |
10:21.45 | mitcheloc | *** let me change that, cheap and SMALL |
10:21.46 | Insanity5 | Probably could interface by serial :P |
10:21.51 | footnote | pic basic |
10:21.59 | Insanity5 | and they're easy adn painless to code. |
10:22.32 | Insanity5 | mit - There http://www.rs485.com/pespsx3.html |
10:22.47 | Insanity5 | ethernet > serial controller thingy. |
10:22.51 | mitcheloc | ooooooh /drool |
10:22.59 | footnote | I wish I could get a little TMS320 on a credit card board |
10:23.06 | footnote | cheep |
10:23.07 | mitcheloc | cost? |
10:23.25 | Insanity5 | Dont' know, probably a couple hundred. |
10:23.30 | mitcheloc | $300!!! |
10:23.35 | footnote | sheesh |
10:23.43 | mitcheloc | NO THANKS! rabbit, i'll keep |
10:23.45 | footnote | you can get a terminal server cheaer |
10:23.50 | footnote | er cheeper |
10:23.55 | Insanity5 | mitcheloc - But somebody has to make one somewher.e |
10:24.07 | footnote | it's for legacy retrofits |
10:24.13 | Insanity5 | It wasj ust the I'm feelin lucky button on google.com with ethernet and serial entered in. |
10:24.16 | Insanity5 | Anyways goodnight :) |
10:24.20 | footnote | like a DMS100 |
10:24.25 | mitcheloc | good night |
10:24.38 | mitcheloc | terminal server cheaper? |
10:24.46 | mitcheloc | thin client you mean? |
10:25.01 | footnote | naw a junker terminal server on ebay :) |
10:25.01 | Insanity5 | mitcheloc - http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=11175&item=5756115258&rd=1 |
10:25.12 | ta[i]nted | anyone here have DID through BV? |
10:25.21 | Insanity5 | mitcheloc - better yet, try this one: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=11175&item=5757661249&rd=1 |
10:25.26 | Zeeek | BV has had problems with asterisk recently |
10:25.28 | Insanity5 | erial To Ethernet Interface - Network Everything!! |
10:25.34 | mitcheloc | what does that do? |
10:25.35 | ta[i]nted | Zeeek do you have DID through BV? |
10:25.48 | Zeeek | no but a lot of discussion here about recent issues fwiw |
10:26.05 | *** join/#asterisk pratik (~root@202.149.48.214) |
10:26.05 | Insanity5 | mitcheloc - Ethernet to serial with documentation. Only $30 BIN to boot. |
10:26.09 | mitcheloc | i'm going to googlerape "terminal server" tomorrow and learn it all |
10:26.11 | mitcheloc | $30 is cheap |
10:26.15 | Zeeek | someone even asked to put a link the title |
10:26.16 | footnote | hrm, is #asterisk logged? |
10:26.31 | mitcheloc | my irc client logs it ;) |
10:26.37 | Bonbon | does anyone use the low-end cisco phones with * ? |
10:26.40 | Zeeek | for BV stuff check the mailing list |
10:26.41 | Insanity5 | footnote - Why? lol |
10:26.45 | mitcheloc | i'm sure most peoples do too |
10:26.52 | RoyKa | hmmmm. what's the cheapest digium shite I can use for a timing source? |
10:26.57 | mitcheloc | Bonbon: lowend =? |
10:26.58 | footnote | Insanity5: for searching |
10:27.05 | Insanity5 | footnote- ahh, ok. |
10:27.07 | pratik | hello everyone |
10:27.09 | footnote | that's the trouble with IRC |
10:27.16 | footnote | it just goes *poof* |
10:27.20 | RoyKa | ~lart Zeeek |
10:27.29 | footnote | mailing lists are better for important stuff |
10:27.30 | Insanity5 | Ya, go punch your nick (if unique) in google, it's fun :) |
10:27.32 | Bonbon | mitcheloc - 7905 / 7910 |
10:27.47 | mitcheloc | yea, they work perfect |
10:27.51 | mitcheloc | crappy thing is they don't have mics |
10:28.09 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
10:28.10 | RoyK | hmmmmmm |
10:28.11 | Insanity5 | bye :) |
10:28.12 | pratik | i am constantly getting this error "Mar 7 16:01:06 WARNING[4164]: chan_sip.c:728 retrans_pkt: Maximum retries exceeded on call 376a6eb97378753c1f38252867c4273f@127.0.0.1 for seqno 121 (Critical Request) |
10:28.13 | pratik | " |
10:28.16 | Insanity5 | mitcheloc - It's $30... just play with it |
10:28.30 | riksta | pratik: looks like it isn't registering |
10:28.53 | pratik | but not registering with what |
10:29.11 | RoyK | Anyone here doing large numbers of SIP traffic with asterisk? |
10:29.13 | Bonbon | mitcheloc - thanks. so you mean there is no speakerphone? |
10:29.25 | mitcheloc | there is a speaker |
10:29.27 | mitcheloc | but no microphone |
10:29.30 | Bonbon | RoyK: what's a "large number" |
10:29.38 | Bonbon | mitcheloc: so we can only monitor? |
10:29.42 | mitcheloc | yes |
10:29.45 | RoyK | 100 concurrent calls or so? |
10:29.47 | riksta | mitcheloc: are you sure about that, i thought there was a mic |
10:30.01 | RoyK | over WAN |
10:30.46 | pratik | what should i do? |
10:30.57 | Bonbon | Royk: yes, we do that |
10:31.03 | riksta | pratik: do you have a sip provider |
10:31.16 | mitcheloc | riksta: maybe your right, maybe it's the 7912..., i'll go look under my bed, i've got one there |
10:32.17 | pratik | yes, i tried registering with my own server |
10:32.39 | mitcheloc | yep yep, it all came back to me now, 7905 has one cat5 jack and 7910 has pass through, but same model, and no microphones |
10:32.50 | mitcheloc | and i don't think there is a 7912..not sure what i was thinking there |
10:32.52 | pratik | but then i removed it from the registration |
10:33.11 | riksta | mitcheloc: hehe...i thought so :) |
10:33.16 | pratik | i.e i removed it from the sip.conf and extensions.conf |
10:33.49 | footnote | hrm |
10:33.53 | footnote | m4 |
10:34.28 | pratik | and when i see the "sip show peers" it says that the status to that server is unmonitored |
10:34.33 | *** join/#asterisk Ubuz (~momo@CBL217-132-84-235.bb.netvision.net.il) |
10:35.35 | mitcheloc | ok wow i'm tired, good night |
10:35.57 | Ubuz | help needed - I installed the latest stable version as instructed in the web page, installed everything and now asterisk will not load with an error: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_dundi.so: undefined symbol: ast_config_load |
10:38.05 | pratik | riksta:what is the possible solution to this |
10:38.19 | riksta | depends what you are trying to do |
10:38.31 | riksta | if you took out the register, you need to reload |
10:38.33 | The_Ball | Hi, sorry to nag but can anyone help me with a newbie dialplan question, here is my five line dialplan: http://channels.debian.net/paste/322 how can i make the phone start to ring at the very begining? |
10:38.57 | pratik | riksta: ya i reloaded it several times |
10:39.45 | pratik | if you want i can paste the sip.conf and the extensions.conf in the pastebin |
10:40.01 | *** join/#asterisk TheJudge (JTR@209-203-52-3.network.ods.co.za) |
10:40.24 | TheJudge | anyone here from South Africa ? |
10:40.35 | riksta | yeah, im a bit busy, but paste it there, and maybe someone else can have a look at it before i do |
10:40.57 | pratik | ok i will |
10:43.06 | footnote | TheJudge: No but I just sent some stuff there :) |
10:43.31 | footnote | for MTN |
10:44.26 | pratik | ya i have pasted it in pastebin.ca/6949 |
10:44.38 | TheJudge | :0 |
10:44.46 | TheJudge | they trying to get voip running ? |
10:44.59 | TheJudge | I hear the are batteling with there edge device etc ? |
10:45.03 | footnote | it wasn't for voip directly but yes |
10:45.19 | pratik | if any one can have a look plz let me know what is the error |
10:45.31 | footnote | TheJudge: I'm only involved with the message store :) |
10:45.36 | TheJudge | ok |
10:45.43 | footnote | "It's not my fault sir I swear!" |
10:45.45 | footnote | hehe |
10:45.46 | pratik | even my fwd calls are not going |
10:45.53 | TheJudge | what do ppl think of broadvoice ? |
10:46.03 | TheJudge | anyone else know of good voip terminatiors ? |
10:46.18 | pratik | where as when i do "sip show peers" it shows that it is registered |
10:47.48 | footnote | voip is proof customers will tolerate "one nines" :) |
10:48.36 | TheJudge | huh ? |
10:48.49 | footnote | 9x.xxx% uptime :) |
10:49.11 | TheJudge | lol |
10:49.12 | TheJudge | :) |
10:49.25 | TheJudge | more like 1000000 |
10:49.34 | TheJudge | voip bring new meaning to uptime ? |
10:49.38 | TheJudge | in South Africa |
10:49.43 | footnote | hehe |
10:49.49 | riksta | footnote: i don't get it |
10:50.02 | Delvar | its ileagel isnt it? |
10:50.21 | TheJudge | what voip in South Africa ? it was legalised 2 feb 2005 |
10:50.28 | pratik | riksta:did u go through it |
10:50.38 | Delvar | TheJudge: ah didnt know that |
10:50.46 | footnote | riksta: well, five nines is what they pound old school telecom developers with |
10:50.58 | footnote | so testing is a BIG deal |
10:51.01 | riksta | footnote: oh, is that an american thing? |
10:51.06 | footnote | yeah |
10:51.10 | riksta | ahh, im UK |
10:51.10 | footnote | it is pure suck |
10:51.17 | footnote | it's mostly marketing |
10:51.27 | footnote | with a lot of red tape for developers |
10:51.37 | footnote | it's impossible to get updates out, etc |
10:51.57 | pratik | i have pastd it in www.pastebin.ca/6949 |
10:52.46 | TheJudge | trying to setup voip in sa is such a mess |
10:53.02 | TheJudge | there is not enough bandwidth ! ! |
10:53.22 | Zeeek | pratik I can't find your question: what is the problem? |
10:53.22 | footnote | TheJudge: you surf? |
10:53.25 | TheJudge | yea |
10:53.31 | footnote | sharks |
10:53.36 | TheJudge | not a problem |
10:53.43 | TheJudge | also dive instructure |
10:53.50 | TheJudge | love diving with sharks |
10:53.52 | footnote | Where are you in SA? |
10:53.59 | TheJudge | live in JHB |
10:54.06 | pratik | i am continuosly getting an error"Mar 7 16:27:04 WARNING[4164]: chan_sip.c:728 retrans_pkt: Maximum retries exceeded on call 376a6eb97378753c1f38252867c4273f@127.0.0.1 for seqno 198 (Critical Request) |
10:54.07 | pratik | " |
10:54.09 | TheJudge | but got house in Capetwon and Durban |
10:54.19 | footnote | so summer is winding down there |
10:54.30 | Zeeek | that means that a sip peer is unreachable |
10:54.37 | Zeeek | usually |
10:54.56 | TheJudge | slowly yea, but in winter it is still warm in durban temp is around 23 c and water about 19 c |
10:55.01 | Zeeek | one way to check is to remove all the register = statements and put them back one by one |
10:55.02 | TheJudge | so its all good ! |
10:55.12 | footnote | TheJudge: I used to live in Huntington Beach, CA USA, we had a lot of folks from SA there |
10:55.16 | TheJudge | where you from |
10:55.26 | pratik | and the when i did "sip show peers" so to that particular server it said request sent |
10:55.30 | footnote | I'm in Dawsonville Georgia now. |
10:55.39 | footnote | (home of Bill Elliot!) |
10:55.41 | footnote | hehe |
10:56.03 | footnote | TheJudge: Bill Elliot == redneck race car driver |
10:56.18 | Zeeek | pratik nothing to with anything, but decide whetheer you want nat=yes and leave one line commented or not |
10:56.28 | pratik | so froom the sip.conf and extensions.conf i removed the particular data regarding |
10:56.39 | footnote | ok, time for bed before the sun comes up |
10:57.01 | Zeeek | pratik can you call fwd ? |
10:57.11 | pratik | yes i need nat=yes because our network is behind NAT |
10:57.22 | pratik | no but it shows that it is registered |
10:57.29 | Zeeek | yes but you have it two times, once commented out - get rid of one |
10:57.43 | pratik | ok i'll do it |
10:57.50 | pratik | just a minute |
10:57.55 | Zeeek | comment out the fwd register and see if the sip message stops |
10:58.20 | Zeeek | comment this line out of sip.conf : register => 607191:mypassword@fwd.pulver.com/607191 |
10:59.50 | Zeeek | pratik also you do NOT want dtmfmode=inband |
11:00.45 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
11:01.06 | Zeeek | pratik this is all I have in sip.conf: http://pastebin.ca/6950 |
11:01.27 | mesi | pratik/zeeek: My FWD authentication fails all the time :-( |
11:01.45 | Zeeek | SIP or IAX? |
11:02.06 | mesi | SIP |
11:02.22 | *** join/#asterisk meppl (~mephisto@pD95421C1.dip.t-dialin.net) |
11:02.22 | pratik | ok zeeek , thanks for that i'll change it but why are the fwd calls not going |
11:02.22 | mesi | I just changed my pass and now authentication fails all the time. :-( |
11:02.35 | meppl | guten morgen |
11:02.42 | Zeeek | mesi: change it back! |
11:02.52 | mesi | meppl: Morning. I think it is english here, not german ;-) |
11:03.04 | mesi | zeeek: I can't! Somebody stole it! |
11:03.11 | meppl | good morning mesi |
11:03.12 | Zeeek | mesi get a new account |
11:03.21 | mesi | Zeeek: But I love my no, 434240! |
11:03.30 | meppl | mesi, i did "/amsg guten morgen" ;) |
11:03.31 | mesi | Ok, I'll try to change it back. |
11:03.34 | Zeeek | I can get you a 2XXXXX number |
11:03.36 | meppl | mesi, thats the problem ;) |
11:04.19 | mesi | meppl: ah, ok. So you are from germany? |
11:05.08 | meppl | yes |
11:05.12 | TheJudge | does any one know of another service provide another the broad voice who support diffrent codecs ? such as 729a or 711 |
11:05.13 | TheJudge | ? |
11:05.20 | mesi | This really sucks! I have got several password problems. With FWD, ENUM and the like :-( Seems to be a bad time! |
11:05.30 | mesi | meppl: Me too. |
11:06.07 | *** join/#asterisk ptblank (~MURDER1@68-169-176-29.lmdaca.adelphia.net) |
11:06.18 | tuxinator_linux | You ever get a mood where you just don't wanna code? |
11:06.22 | tuxinator_linux | I'm in that mod |
11:06.24 | tuxinator_linux | mood |
11:06.41 | mesi | tuxinator: I know that. |
11:06.54 | tuxinator_linux | And I love coding |
11:07.18 | tuxinator_linux | but I need to get it done. I have added about 10 lines in the last 5 hours |
11:07.43 | mesi | Tuxinator: That sounds bad. |
11:08.06 | meppl | mesi, german speaking people are in germany, austria, switzerland, luxemburg and some parts of africa |
11:08.07 | tuxinator_linux | starting to pick up, but I will need to sleep soon |
11:08.13 | meppl | mesi, oh, okay |
11:10.01 | geekster | anyone using asterisk to peer with sipphone.com ? |
11:10.06 | mesi | Oh dear! this is really bad! I can only use the stolen FWD password with asterisk, not a newly set one ;-( |
11:10.29 | mesi | geekster: I would like to. |
11:10.43 | pratik | zeeek: i have checked that, can you paste your extensions.conf |
11:10.57 | geekster | mesi: Its not working for me to well |
11:11.06 | mesi | meppl: nevertheless, you are from germany :-) |
11:11.07 | tuxinator_linux | mesi: sad part is that it only took 5 minutes to do what I wanted to do |
11:11.17 | shido6 | pastebin.ca |
11:11.24 | Zeeek | pratik I did that above |
11:11.58 | mesi | tuxinator: And why would you have to get this done? Is it for your Job? |
11:12.35 | pratik | Zeek: ya but that was only the sip.conf , i want to see your extensions.conf |
11:12.46 | soulz- | hello all |
11:12.49 | pratik | Zeeek: if you dont mind |
11:13.05 | soulz- | Mar 7 19:06:38 NOTICE[868]: app_dial.c:936 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) |
11:13.05 | geekster | hey there soulz |
11:13.09 | soulz- | i am getting this error message |
11:13.10 | geekster | hows it goin ? |
11:13.13 | soulz- | can anyone help? |
11:13.16 | soulz- | hey geekster |
11:14.01 | geekster | soulz-: I've never seen this error, anyone else want to take a stab ? |
11:14.20 | tuxinator_linux | mesi: I own a couple companies and I am working a few large projects for one of them, including, writing the whole patient records and managed system and * |
11:14.23 | soulz- | weird ehh? i never seen as well |
11:15.01 | geekster | soulz-: what actions are you doing that it causes tihs error ? |
11:15.02 | tuxinator_linux | mesi: I stay busy |
11:15.17 | pratik | zeeek: i want to know where am i going wrong |
11:15.25 | soulz- | geekster: i am using bv, and they just implemented this new thing, called invite auth |
11:15.48 | soulz- | and i patched it to the current cvs, and this error comes up |
11:15.56 | Zeeek | pratik - I posted my config once and I'm not on pastebin anymore - look above and find it |
11:16.17 | tih | 'cause 3' is "no route to destination". |
11:16.45 | mesi | tuxinator: so, than it is really really bad! |
11:16.46 | soulz- | tih: exten => _2.,1,Dial(SIP/broadvoice/${EXTEN:1}) |
11:16.55 | Zeeek | hello tih |
11:16.59 | soulz- | tih: this is what causes the cause 3 error |
11:17.00 | tih | soulz-: it doesn't know how to reach broadvoice. |
11:17.18 | pratik | zeeek but on 6950 u have only pasted your sip.conf |
11:17.19 | soulz- | tih: emmm, okay |
11:17.33 | mesi | Has anybody ever changed the fwd password and still registered with asterisk by changing sip.conf? It doesn't work for me here. |
11:17.35 | tih | Check your definition for broadvoice -- it's probably a 'peer' entry in your sip.conf, right? What host are you telling it it's on? |
11:17.57 | tuxinator_linux | mesi: I never did like working for people. I have fun, and isn't that what your suppose to do in life? |
11:18.27 | Zeeek | you like working in general? |
11:18.37 | tuxinator_linux | yes |
11:18.44 | tuxinator_linux | I enjoy creating |
11:18.45 | Zeeek | that's a step, then :) |
11:18.49 | mesi | tuxinator: Yes, that's what life is all about!! |
11:18.54 | Zeeek | ah, reating. An ahhtist |
11:18.59 | tuxinator_linux | writing programs is a power trip |
11:19.01 | soulz- | tih: i just use type = friend |
11:19.22 | tih | soulz-: ok, and host = ? |
11:19.33 | mesi | Does Asterisk store fwd passwords somewhere else than in sip.conf where I configured it? |
11:19.33 | soulz- | i changed to peer also same thing |
11:19.34 | Zeeek | for BV issues, check the mailing list - there have been problems lately specific to asterisk |
11:19.49 | Zeeek | several people were discussing this yesterday |
11:19.50 | soulz- | host = sip.broadvoice.com |
11:19.59 | soulz- | zeek: i did try it |
11:20.01 | Zeeek | everyone is having problems |
11:20.06 | soulz- | but my error is not the same |
11:20.19 | Zeeek | dod you see the wiki? |
11:20.20 | tih | soulz-: and can you ping sip.broadvoice.com from your machine? |
11:20.39 | soulz- | yup |
11:20.41 | soulz- | can do so |
11:20.53 | Zeeek | [there's been so much chatter about Broadvoice in about the last week] |
11:21.16 | mesi | Zeek: so broadvoice is not working so well? |
11:21.52 | soulz- | zeek: tell me about it, they just implemented something |
11:21.58 | tih | Hm. Then probably what Zeeek says - or you might want to debug a bit, as in: study the actual SIP packets being sent and received. |
11:21.58 | soulz- | without letting the users know |
11:22.43 | soulz- | Mar 7 19:22:22 NOTICE[969]: app_dial.c:936 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) |
11:22.48 | soulz- | this is bugging me |
11:23.06 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
11:23.24 | Zeeek | we use a customer's weather service which had a procedure for ftp stats. They just changed it with no notice and our cron job went on blithely downloading the same file every day until someone noticed the temperature never changed :) |
11:23.38 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
11:24.35 | pratik | <PROTECTED> |
11:25.05 | pratik | linksys(PAP2) is same as sipura |
11:25.38 | mesi | I get it now! FWD password for web interface and sip aren't the same :-( |
11:26.03 | Mavvie | why is there a SIPCALLID, a SIPUSERAGENT and a SIPDOMAIN, but no SIPUSERNAME? |
11:26.27 | Zeeek | mesi, uh ya! |
11:26.29 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
11:27.04 | mesi | Zeek: So how am I expected to change this stupid SIP pass?! ;-) |
11:28.47 | Zeeek | there is a page somewhere |
11:28.55 | Zeeek | but why not use IAX while you're at it? |
11:29.08 | Zeeek | ooops gotta go |
11:29.11 | Zeeek | bye all |
11:29.20 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
11:29.33 | pratik | <PROTECTED> |
11:29.59 | mesi | Zeek: You know, I use IAX with iaxtel. I want to do SIP, too. :-) |
11:30.08 | mesi | Oh, gone. |
11:30.22 | soulz- | pratik: is voip legal in india? |
11:30.56 | *** join/#asterisk pranav (pranav@202-149-48-214.broadband.isp.exatt.net) |
11:31.16 | pratik | soulz:yes it is valid but only for outgoing calls |
11:31.25 | pranav | hi |
11:32.03 | mesi | Hi pranav. |
11:32.16 | pranav | hi |
11:32.26 | soulz- | pratik: ok |
11:32.34 | *** join/#asterisk asteriskforuk (~vircuser@i-195-137-59-254.freedom2surf.net) |
11:32.48 | asteriskforuk | hello |
11:32.54 | asteriskforuk | ne1 awake? |
11:32.58 | benno2 | could it be that with some 1.4.x budgetone firmware versions the call often gets hung up after 40-60secs ? |
11:33.26 | pratik | soulz: can yv tell me i was using sipura device with my asterisk , can i replace it by sipura router |
11:33.30 | asteriskforuk | ne1 help with setting up agents? |
11:33.42 | asteriskforuk | i've got a call queue and it works ok.. now want to try agent logins etc |
11:33.54 | *** join/#asterisk MuppetMaster (~muppetmas@a82-92-73-185.adsl.xs4all.nl) |
11:35.40 | soulz- | pratik: i have never used sipura sorry dude |
11:37.02 | MuppetMaster | Hellol |
11:37.05 | MuppetMaster | What is the question on Sipura? |
11:37.13 | pratik | ok fine, see i have paswted my sip.conf and extensions.conf in the pastebin.ca/6949 |
11:37.38 | pratik | if you have time just go through it, my fwd calls are not going |
11:38.09 | *** join/#asterisk TheEmperor (TheEmperor@218.111.51.46) |
11:38.55 | shaZwaz | anyone know where I can get a G723 license ? |
11:39.47 | tuxinator_linux | what about G729? |
11:40.03 | soulz- | shazwar: u don't need a licence i think |
11:40.11 | *** join/#asterisk Ubuz (~momo@CBL217-132-84-235.bb.netvision.net.il) |
11:40.17 | shaZwaz | I wnat more compression |
11:40.21 | soulz- | just use allow=g723 |
11:40.37 | shaZwaz | to save my bandwidth |
11:40.47 | shaZwaz | anyone tried speex ? |
11:40.52 | soulz- | use gsm or ilbc |
11:40.59 | Ubuz | help needed - I installed a stable version of asterisk over an existing vesion, and now asterisk won't load any modules. any idea why? |
11:41.05 | shaZwaz | I am already using 729 |
11:41.13 | soulz- | works great on a dialup:) |
11:41.47 | shaZwaz | Ubuz, what does the CLI read ? |
11:41.49 | *** join/#asterisk Lethargicclown (~chatzilla@ool-18bee80e.dyn.optonline.net) |
11:41.55 | pratik | can anyone tell me why my fwd calls are not going |
11:42.05 | soulz- | iax2 debug |
11:42.12 | RoyK | arg |
11:42.13 | soulz- | adios |
11:42.14 | RoyK | Mar 7 13:09:14 WARNING[3243]: chan_zap.c:9615 setup_zap: Ignoring switchtype |
11:42.14 | RoyK | Mar 7 13:09:14 ERROR[3243]: chan_zap.c:9436 setup_zap: Unknown signalling method 'pri_cpe' |
11:42.15 | RoyK | wtf? |
11:42.22 | Ubuz | shaZwaz: I cannot load the CLI, it stops with an error |
11:42.35 | shaZwaz | what does that say ? |
11:43.00 | shaZwaz | RoyK, is that 0.6 ? |
11:43.18 | Ubuz | loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_dundi.so: undefined symbol: ast_config_load |
11:43.29 | RoyK | shaZwaz: huh? 1.0.6 |
11:43.30 | Ubuz | loader.c:440 load_modules: Loading module pbx_dundi.so failed! |
11:43.53 | RoyK | er |
11:43.54 | RoyK | fuck |
11:43.56 | RoyK | 1.0.3 |
11:44.21 | shaZwaz | Ubuz, go to modules.conf and put noload = pbx_dundi.so |
11:44.47 | Ubuz | shzWaz: If I do that the error appears on the next module |
11:44.49 | ta[i]nted | god damn it |
11:44.52 | ta[i]nted | i'm addicted to * |
11:44.54 | pratik | can anyone tell me why my fwd calls are not going |
11:44.57 | ta[i]nted | :( |
11:45.09 | shaZwaz | Ubuz, may be u didn't compile it right |
11:46.43 | *** join/#asterisk Inferna (~sasha@194.158.51.171) |
11:46.47 | Inferna | hello |
11:47.01 | shaZwaz | do a make clean ; make install on all src directories |
11:47.10 | RoyK | http://pastebin.ca/6951 |
11:47.14 | Inferna | can somebody help me with one strange thing? |
11:47.14 | RoyK | can someone help me here? |
11:47.21 | shaZwaz | and see if there are any compilation errors |
11:47.29 | RoyK | shaZwaz: that's 1.0.6, and it just won't load |
11:47.35 | RoyK | asterisk 1.0.6 as well |
11:47.36 | Ubuz | shaZwaz: did it several times, no errors |
11:47.43 | shaZwaz | RoyK, do a rmmod on the cards |
11:47.45 | Inferna | <PROTECTED> |
11:47.46 | RoyK | the config files are identical to those of another server |
11:47.52 | RoyK | shaZwaz: tried that as well |
11:48.03 | Inferna | and then i have 2 dublicate calls from h323 |
11:48.08 | RoyK | ztcfg -vvvvvv show everything's fine |
11:48.24 | asteriskforuk | IAX ... ne1 know what ports to open on a firewall? and which iax phone works well? |
11:48.26 | RoyK | trying a rebooot... |
11:48.29 | shaZwaz | check u zapata.conf then |
11:48.33 | shaZwaz | zaptel.conf |
11:48.52 | RoyK | shaZwaz: both of them have been copied from a production server with the same te410p card installed, same hardware |
11:48.58 | RoyK | same suse 9.1 distro |
11:49.07 | RoyK | a little older kernel |
11:49.30 | shaZwaz | 2.4 ! |
11:49.42 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
11:49.45 | RoyK | production server runs 2.6.8.1. test server 2.6.11 |
11:50.07 | shaZwaz | did u compile with make2.6 |
11:50.26 | shaZwaz | zaptel that is |
11:50.28 | RoyK | make linux26 |
11:50.33 | shaZwaz | yup |
11:50.44 | RoyK | shaZwaz: otherwise the modules wouldn't load, obviously...... |
11:50.49 | Inferna | is any guru here? |
11:50.53 | RoyK | ~lart shaZwaz |
11:50.56 | Inferna | i have the problem with queue |
11:50.59 | Inferna | plz help me |
11:51.01 | ta[i]nted | BV sucks balls |
11:51.18 | RoyK | Inferna: noone can help you unless you say WHAT is wrong |
11:51.19 | Lethargicclown | Should I coninue looking to see if asterisk works on cygwin, Or should I give up? |
11:51.29 | pratik | ok fine, see i have paswted my sip.conf and extensions.conf in the pastebin.ca/6949 |
11:51.31 | shaZwaz | may be the configs are bad |
11:51.37 | RoyK | Lethargicclown: it might work, but it'll suck |
11:51.44 | gambolputty | Ubuz: Did you delete the modules of the prior * version? |
11:51.49 | pratik | can anyone check why my fwd calls are not going |
11:51.50 | RoyK | shaZwaz: they work on another server |
11:51.52 | shaZwaz | u must have missed something |
11:51.53 | RoyK | same fscking files |
11:52.37 | shaZwaz | otherwise use the same source too |
11:53.04 | Inferna | <PROTECTED> |
11:53.16 | pratik | gambolputty:can you help me? |
11:53.32 | gambolputty | with? |
11:53.42 | pratik | my fwd calls are not going |
11:53.56 | gambolputty | you use iax or sip to connect to fwd? |
11:54.01 | pratik | sip |
11:54.04 | Ubuz | gambolputty: How can I delete the modules of the older version? |
11:54.46 | gambolputty | Ubuz: The first thing I do is stop the current instance of *. |
11:54.55 | gambolputty | Then unload zaptel and ztdummy |
11:55.08 | gambolputty | Delete modules in /usr/lib/asterisk |
11:55.23 | gambolputty | then recompile another * version as needed |
11:55.36 | gambolputty | modprobe ztdummy |
11:55.44 | gambolputty | and make sure * works with asterisk -vvvvgc |
11:55.45 | Ubuz | gambolputty: I deleted the modules, removed everything under /var/lib/asterisk |
11:55.54 | pratik | gambolputty: ihave pasted my sip.conf and extensions.conf in thge pastebin.ca/6949 |
11:55.55 | Ubuz | then I recompiled and reinstalled the modules |
11:56.03 | Ubuz | there's not problem with zaptel, it works fine |
11:56.12 | gambolputty | ok |
11:56.19 | RoyK | fuck |
11:56.27 | gambolputty | pratik: Is your * box behind a firewall? |
11:57.46 | gambolputty | Ubuz: What distro of Linux are you using? |
11:58.11 | RoyK | shaZwaz: anyway, asterisk should not fscking tell me pri_cpe is unknown signalling meth |
11:58.26 | Ubuz | gambolputty: SuSE 9.1 |
11:58.46 | pratik | gambolputty:yes |
11:59.07 | pratik | gambolputty: does that affect |
11:59.08 | gambolputty | pratik: Use IAX instead to connect to fwd. |
11:59.55 | pratik | ok IAX i am not very much familiar with, i'll have to read the documents |
11:59.58 | mesi | gambolputty: I use iax now. But anyway I'd like to change my stolen SIP password! |
12:00.12 | mesi | pratik: http://www.freeworlddialup.com/content/view/full/1501 |
12:00.35 | gambolputty | Ubuz: I use Fedora Core 2 |
12:00.47 | gambolputty | I have to be careful when kernel versions change |
12:00.49 | mesi | Quite easy. And it uses the password you use for the FWD webpage. Anyway, it seems you cannot change the SIP password :-( |
12:00.49 | pratik | gambolputty: but see tell me to make a call i am using a sipura phone, will that worki with the IAX |
12:01.07 | gambolputty | A Sipura device does with with * |
12:01.16 | gambolputty | and then * connects to fwd using IAX |
12:01.25 | pratik | gambolputty:and sipura phone is a sip phone |
12:01.28 | mesi | pratik: is your phone connected to asterisk? Then it will work. Asterisk can do sip to the phone, but be connected via IAX to FWD. |
12:01.28 | gambolputty | All Sipura devices speak Sip |
12:01.38 | shaZwaz | RoyK, pridialplan=unknown |
12:01.38 | shaZwaz | <PROTECTED> |
12:01.38 | shaZwaz | <PROTECTED> |
12:01.45 | Ubuz | gambolputty: Everything worked, but I saw some bugs in asterisk, so I decided to install the latest stable version, and since then it won't work |
12:02.11 | gambolputty | Maybe try cvs head version? |
12:02.32 | pratik | mesi:yes my sipura phone and the asterisk are connectedd in the same network |
12:02.45 | shaZwaz | RoyK, u sure its pri_cpe .. I mean spelled it correct ? |
12:03.07 | *** join/#asterisk jks (~jks@0x503e4c12.arcnxx4.adsl-dhcp.tele.dk) |
12:03.09 | mesi | pratik: So look at the webpage I pasted. Your sippura will work with asterisk being connected to FWD via IAX. |
12:03.24 | mithro | anyone know if the X100P cards work with caller ID in australia? |
12:03.26 | pratik | ok ya i'll go through it |
12:04.15 | shaZwaz | Why is it ignoring the switchtype ...something wrong there may be try switchtype=euroisdn |
12:04.55 | jks | how do I set the mapped address on a SNOM IP-Phone? (I've got 1:1 NAT) |
12:05.59 | Ubuz | gambolputty: I got the latest CVS version and it doesn't work too. You think the problem is with the kernel? |
12:06.28 | shaZwaz | Ubuz, revert to older to verify |
12:07.13 | Ubuz | shaZwaz: Revert to older what and how do I get it? |
12:07.54 | shaZwaz | u overwrote it didn't u ! |
12:08.21 | shaZwaz | check cvs.digium.com |
12:08.44 | The_Ball | how can i make my asterisk server accept "calls" from any anonymous users on any ip? |
12:08.51 | Ubuz | I am copying a working version from another computer |
12:09.08 | mesi | Anybody for a call? My FWD is 434240 :-) |
12:10.26 | mesi | Who called? :-) |
12:10.31 | mesi | That was only one ring. :-\ |
12:10.42 | shaZwaz | me |
12:10.45 | shaZwaz | :) |
12:10.48 | mesi | ShaZwaz ;-) |
12:11.11 | mesi | Even my hardphones ring, though they aren't internetphones. I connected them to asterisk ;-) |
12:11.23 | mesi | That's funny. |
12:11.52 | shaZwaz | u must be ringing them in the dialplan |
12:12.01 | shaZwaz | where the call is landing |
12:14.07 | mesi | I do. The phone rang, but you obviously hung up immediatly :-\ |
12:14.29 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
12:15.06 | FaithX | Hey guys, where can I get some usefull info on bristuff? |
12:15.37 | RoyK | ~lart shaZwaz for not listening |
12:17.45 | *** join/#asterisk Jer1326 (~Jer@rdu57-251-152.nc.rr.com) |
12:18.23 | jks | anyone got a snom IP phone? |
12:19.33 | Jer1326 | can someone offer me a sugestion as to why atxfer doesnt work for me? |
12:22.52 | mesi | Are all theses sip-providers interconnected? Like sipphone and FWD? |
12:23.16 | Jer1326 | some are some arent |
12:24.04 | sambal | Jer1326: which phone? |
12:24.15 | mesi | Jer: So I have to register my asterisk to several of them to be able to call every member, right? |
12:25.21 | Jer1326 | yep |
12:26.00 | gambolputty | jks: I have a snom 190 ip phone |
12:26.06 | Ubuz | ok, the latest version in the CVS doesn't work well for me. I installed CVS-HEAD-03/07/05 and it works |
12:26.16 | jks | gambolputty: have you tried using it with 1:1 NAT? |
12:26.17 | Jer1326 | sambal: my cell.... calling in using IAX via teliax |
12:26.28 | jks | gambolputty: it keeps using its internal IP in the SIP messages, no matter what I configure :-/ |
12:26.41 | gambolputty | What do you mean by 1:1 nat? |
12:26.57 | Jer1326 | even if i set atxfer => *4 or whatever it still says transfer when i hit # :/ |
12:27.07 | jks | gambolputty: The internal IP corresponds directly to one external IP |
12:27.10 | Makenshi | Morning gambolputty :> how are you these days? |
12:27.17 | gambolputty | fine |
12:28.25 | mesi | Is it advisable to register with sipphone.com? |
12:28.25 | gambolputty | my snom phone connects with * |
12:28.25 | gambolputty | are you trying to make a sip call to somewhere? |
12:28.29 | jks | gambolputty: I'm just trying to get it to connect to my asterisk server |
12:28.39 | mesi | Me? Well, I just want to be able to call many people and they should be able to call me :-) |
12:28.55 | jks | gambolputty: from the SIP trace I can see that it puts its 10.x.x.x address in the SIP messages, instead of its external IP... that's what I want to change |
12:29.06 | gambolputty | jks: post your sip.conf on pastebin.com |
12:29.33 | jks | gambolputty: it's the phone's settings I want to change? |
12:29.40 | gambolputty | not sure |
12:29.47 | jks | it is :-) |
12:29.47 | gambolputty | lets make the phone register with * first |
12:30.02 | jks | gambolputty: well, the reason it doesn't register is because the phone sends off it's internal IP |
12:30.19 | jks | gambolputty: my sip.conf is pretty standard... I have host=dynamic and nat=yes |
12:30.26 | RoyK | ~seen zoa |
12:31.07 | jbot | zoa <zoa@142.131.189.23> was last seen on IRC in channel #asterisk, 5h 58m 46s ago, saying: 'or at least live from san jose'. |
12:31.07 | gambolputty | that's probably what it should do when it registers with * |
12:31.07 | jks | gambolputty: (and it works perfectly if I put the phone on the external IP directly) |
12:31.07 | gambolputty | is * behind nat? |
12:31.08 | jks | gambolputty: I want the phone to send it's external IP now that it got one |
12:32.42 | gambolputty | why you want to have the snom phone send and external ip? |
12:32.50 | shaZwaz | RoyK, what happend ? |
12:33.00 | jks | gambolputty: my asterisk server is outside my nat |
12:33.05 | gambolputty | ok |
12:33.07 | jks | gambolputty: it can contact the phone only on the external ip |
12:33.11 | jks | gambolputty: so that's why |
12:33.23 | jks | gambolputty: I'm using 1:1 nat, so the phone is the only thing on that external ip |
12:33.24 | pimpwell | anyone have a manager intterface I can check out |
12:33.25 | shaZwaz | ~wakeup RoyK |
12:34.28 | jbot | ACTION silently aproaches RoyK, who's sleeping (zZzZZZzzZZ, Ronc !!! ronc!), gets off his pants and shoots a noisy fart ... PUBFBFBFBBBFFF!!! |
12:34.28 | pimpwell | so I can just get a look at what it's like |
12:34.28 | gambolputty | not sure, both my * box and snom are behind nat on the same subnet |
12:34.28 | gambolputty | sip seems to work best if two devices are on the same subnet |
12:34.28 | RoyK | shaZwaz: I had forgot to compile libpri before compiling asterisk...... |
12:34.29 | RoyK | recompiling asterisk worked |
12:34.29 | shaZwaz | :) |
12:34.33 | shaZwaz | I told u must be missing somthing |
12:35.00 | RoyK | ~lart shaZwaz |
12:35.10 | shaZwaz | ~kill RoyK |
12:36.09 | jbot | ACTION shoots a hyper-charged neutrino gun at RoyK |
12:36.09 | RoyK | shaZwaz: you were just babbling general idiocity |
12:36.09 | jks | gambolputty: yes, but I haven't got them on the same subnet... it's not possible for me. |
12:36.09 | shaZwaz | I told u u were missing something |
12:36.09 | gambolputty | sip and nat have problems getting along |
12:36.09 | shaZwaz | personally I never used te400p cards |
12:36.16 | jks | gambolputty: well, I'm not using "normal" nat |
12:36.28 | gambolputty | ok |
12:36.32 | jks | gambolputty: this should be very easy to get working, if I just could get the phone to send out the right IP |
12:36.34 | shaZwaz | since everything looked fine with the config .... |
12:36.46 | jks | gambolputty: you don't know how to tell the phone which IP it should think it has? |
12:37.00 | gambolputty | not in that manner |
12:37.04 | gambolputty | try different settings |
12:37.04 | jks | gambolputty: okay |
12:37.06 | shaZwaz | I was sure u missed some step and there u were ...missed compiling libpri |
12:37.08 | jks | gambolputty: I have. |
12:37.09 | gambolputty | use the internal sip trace function |
12:37.23 | jks | gambolputty: it's a normal setting on every phone I've come across... I just can't find it on this phone. |
12:37.28 | jks | gambolputty: I am using the internal sip trace function.... |
12:37.32 | gambolputty | ok |
12:37.49 | shaZwaz | well anyway I accept that I am not a * guru |
12:39.03 | Inferna | how to make ring call progress tone insteed of default music in queue? |
12:40.23 | Delvar | anyone know of a free outbound proxy server software? |
12:41.05 | shaZwaz | software ? |
12:41.13 | moonwick | squid? |
12:41.45 | shaZwaz | what u mean by outbound proxy server software ? |
12:44.30 | shaZwaz | ~ping |
12:45.16 | jbot | pong |
12:45.16 | Delvar | sorry |
12:45.16 | Delvar | for SIP/RTP |
12:45.16 | RoyK | Delvar: SER? |
12:45.16 | Delvar | Ser does rtp? |
12:45.16 | RoyK | no |
12:45.16 | Delvar | i thought it was SIP only |
12:45.16 | RoyK | a proxy doesn't do RTP |
12:45.18 | RoyK | proxy does SIP and not RTP |
12:45.23 | shaZwaz | well SIP can't so anything without it ? |
12:45.30 | RoyK | Delvar: a gateway does RTP |
12:46.01 | Delvar | RoyK: sorry my bad terminolayg, i want a free gateway then :) ... not asterisk |
12:46.17 | shaZwaz | to use for ur own ? |
12:46.22 | Delvar | yes |
12:46.39 | shaZwaz | over to u RoyK |
12:47.24 | RoyK | Delvar: aefirion, sipx, yate |
12:47.25 | gambolputty | Delvar: What are you trying to do with a SIP proxy? |
12:47.32 | RoyK | Delvar: or, of course, asterisk |
12:47.52 | shaZwaz | RoyK, u ever tried Aferion ? |
12:48.07 | RoyK | it's basically the same as asterisk atm |
12:48.09 | RoyK | it's a fork |
12:48.21 | RoyK | but it'll change quite dramatically over the next two months or so |
12:48.30 | shaZwaz | what about its SIP capabilities ? |
12:48.32 | Delvar | RoyK: i need an outbound proxy to proxy RTP trafic to get arround nat issues without STUN/port forwarding... |
12:48.57 | RoyK | shaZwaz: right now, same as asterisk, but it'll use OPAL for sip/h323 instead of the crap in * |
12:49.07 | Delvar | RoyK: preferably used alongside SER |
12:49.36 | shaZwaz | guess have to wait then |
12:49.39 | RoyK | Delvar: we use plain asterisk with SIP with the server on an official IP, and it works for thousands of SIP clients |
12:50.29 | shaZwaz | RoyK, anyother choice for saving bandwidth other than using 729 ? |
12:50.34 | *** join/#asterisk wizard2000 (~dang@fubar.arcbox.com) |
12:50.42 | RoyK | shaZwaz: speex? ilbc? |
12:50.51 | wizard2000 | hi guys, having a few problems with zaptel and FC3, anyone got a sec to help? |
12:50.54 | RoyK | g.723.1.... |
12:50.58 | shaZwaz | speex is proprietry ? |
12:51.00 | Delvar | RoyK: so thats a no then?.. |
12:51.06 | RoyK | shaZwaz: speex is very, very open |
12:51.20 | shaZwaz | gime the link will u |
12:51.25 | RoyK | Delvar: I only say it works for us with asterisk |
12:51.33 | RoyK | ~google speex |
12:51.35 | RoyK | ~speex |
12:51.47 | RoyK | ~lart shaZwaz |
12:52.57 | shaZwaz | RoyK, have u tested it on a productions server ? |
12:53.03 | Delvar | RoyK: i know but asterisk doesnt scale too well, especialy when its getting a lot of registration requests a the same time. and to be honest id refer to use something more dedicated like SER to proxy the rtp. |
12:54.18 | *** join/#asterisk pilif (~pilif@mail.sensational.ch) |
12:54.19 | RoyK | shaZwaz: no, but a lot of people do |
12:54.24 | shaZwaz | I mean any issue with that? |
12:54.51 | RoyK | Delvar: then just use SER..... |
12:54.56 | pilif | I'm looking for a good, understandable tutorial on how to configure asterisk |
12:55.03 | RoyK | SER gets an INVITE, and tells client 'you send that RTP there' |
12:55.07 | pilif | are there any current web-resources? |
12:55.14 | RoyK | ~docs |
12:55.24 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
12:55.57 | Jer1326 | ugh i just broke my *...recompiled to lastest CVS and not i get a bus error ...arrgh |
12:56.09 | Jer1326 | it wont even start :( |
12:56.12 | Delvar | RoyK: SER does NOT do RTP :)... hence why i was asking |
12:56.28 | ta[i]nted | i see a bad moon risin~ |
12:57.25 | Jer1326 | can someone help with a simple compile question...? |
12:57.27 | RoyK | Delvar: SER proxies RTP, but never gateways it |
12:57.39 | RoyK | Delvar: read 'asterisk at large' on the wiki |
12:57.49 | RoyK | http://www.voip-info.org/wiki-Asterisk+at+large |
12:58.13 | Delvar | RoyK: ok thanks, ill read about it, i was jsut told SER wouldnt do it, maybe it will. |
12:58.16 | *** join/#asterisk markak (~markak@ndn-165-143-245.telkomadsl.co.za) |
12:58.47 | wizard2000 | Mar 7 12:58:37 sip wait_for_sysfs[2908]: either wait_for_sysfs (udev 039) needs an update to handle the device '/class/zaptel/zap1' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to <linux-hotplug-devel@lists.sourceforge.net> |
12:58.51 | wizard2000 | any ideas people? |
13:02.06 | markak | hi all quick question. if one wanted to for example allow extensions .conf to pull phone number from a mysql database and use it for the dial line is it easy to do. i.e we deal with 1200 + branches accross the country. each branch has a unique branch number. could i place a rule that when our operators dial eg 8 followed by the branch number let say 832 asterisk could query the mysqldb for the phone number and create the dialplan exten => |
13:05.12 | markak | anyone have an idea ? |
13:05.39 | MuppetMaster | markak: Yes, this is fairly straightforward. |
13:06.03 | MuppetMaster | markak: You could either use an AGI to launch an external app to do the lookup, or you could use the built in MySQL lookup function. |
13:06.42 | MuppetMaster | markak: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MYSQL#comments |
13:06.43 | Darwin35 | anyone know if kram made it to ca |
13:07.06 | markak | thanks guys |
13:07.09 | *** part/#asterisk markak (~markak@ndn-165-143-245.telkomadsl.co.za) |
13:07.33 | Darwin35 | ok now the fun of festival |
13:07.38 | Darwin35 | and sphinx |
13:08.05 | Jer1326 | does sphinx realllly work? |
13:08.10 | pilif | can I use the Fritz! USB-Adaptor for my first experiments? I've such a device lying around here and don't want to buy anything just to learn if it'd work |
13:08.58 | Darwin35 | dont know |
13:09.10 | Darwin35 | but going to test and see |
13:09.52 | Lethargicclown | Any special instructions for compiling this in cygwin? |
13:09.59 | Darwin35 | ? |
13:10.53 | Lethargicclown | Just hoping i can test this in cygwin before I dedicate a machine to it |
13:11.03 | Darwin35 | ok I am using 300 megs of a 500 meg cf for fbsd and asterisk thats nt bad |
13:11.30 | Darwin35 | nt/not |
13:12.05 | Darwin35 | just have to get festival and a few agi apps |
13:12.05 | Darwin35 | on it now |
13:12.17 | Jer1326 | i'm intrested to see how well it works |
13:13.12 | Darwin35 | so far it works execpt for the g729 issue I ave to take up with kram |
13:13.23 | Darwin35 | my tdm card works |
13:13.31 | Darwin35 | 4 port fxs |
13:13.36 | Jer1326 | ahh |
13:13.46 | Jer1326 | can someone help me with a segfault? |
13:14.25 | Darwin35 | cut abd paste in pastebin and show it |
13:14.39 | Darwin35 | see what can be done |
13:14.55 | Jer1326 | you want the output of gdb? |
13:15.31 | Darwin35 | yeah pu tit in there also |
13:16.44 | Darwin35 | oops I said tit |
13:16.46 | Darwin35 | hheehe |
13:17.09 | Jer1326 | lol |
13:17.50 | Darwin35 | man I wish my new credit card would get here |
13:18.14 | Darwin35 | I need a new sip phone |
13:19.41 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
13:20.00 | Jer1326 | http://pastebin.ca/6955 |
13:21.38 | Darwin35 | ok looking now |
13:23.22 | Darwin35 | this linux or ffbsd |
13:23.27 | Jer1326 | fbsd |
13:23.36 | Darwin35 | I know that the mailing list had a workaround in it |
13:23.41 | Darwin35 | hhold on |
13:23.48 | Jer1326 | alright |
13:23.55 | The_Ball | is there any catchall extention? like [incoming] \ exten => *,1,Answer() and so on? |
13:24.08 | Jer1326 | isnt it s? |
13:24.18 | The_Ball | oh, is it |
13:24.26 | Inferna | <PROTECTED> |
13:24.56 | The_Ball | i got an error when a iax user tried to dial in to a non existant extention |
13:25.00 | The_Ball | in that context |
13:25.12 | Darwin35 | jer check pvt window |
13:25.15 | Inferna | the_ball: check context |
13:25.17 | The_Ball | Mar 7 22:43:59 NOTICE[8677]: chan_iax2.c:5461 socket_read: Rejected connect attempt from 203.49.132.59, request '80001234@incoming' does not exist |
13:26.34 | MuppetMaster | The_Ball: Yes, it is the 's' extension exten => s,1,Dial.... |
13:26.56 | Darwin35 | it means yo have not setup the user inthe iax.conf and it cant point to a context it cant find |
13:27.27 | The_Ball | hmm, ok |
13:27.54 | Inferna | anybody had problem with freezed calls in the queue? |
13:28.56 | *** join/#asterisk theogeor (theogeor@212.118.246.50) |
13:29.09 | *** join/#asterisk ToyMan (~stuq@204-8-82-238.webjogger.net) |
13:29.47 | theogeor | Hi everybody.... I am looking for help on configuring * and connect it with an MD110 and a TLU76/1 card |
13:30.04 | theogeor | anybody who would like to help :) |
13:30.36 | Darwin35 | its a gothic scottish morning |
13:31.16 | *** join/#asterisk seong (~seong@219.95.129.15) |
13:31.24 | theogeor | I am using an * box with TE410 card |
13:31.28 | theogeor | any ideas ? |
13:31.37 | Inferna | small question, can X-server be the problem of bad music on hold playing? |
13:33.19 | theogeor | Anybody alive in this channel ? |
13:33.23 | theogeor | :( |
13:35.30 | theogeor | Hello anybody here ? |
13:37.27 | Grooby | ....zzzzZZZZ |
13:41.02 | Darwin35 | no that mpg123 |
13:41.37 | *** join/#asterisk hellop (~LeeHarvey@cpe-70-93-41-67.hawaii.res.rr.com) |
13:41.54 | hellop | hi |
13:42.09 | hellop | Hey I just saw the recent slashdot post about Asterisk PBX. |
13:42.20 | hellop | gjgj |
13:42.55 | hellop | So.. I have a question: How much for a commercial, non-asterisk, pbx? 10-20, 50-100, 100-200 lines. |
13:43.24 | hellop | I'm doing an English paper on Asterisk... |
13:43.26 | hellop | for scool |
13:43.42 | hellop | scewl |
13:44.51 | theogeor | Hello anybody who has connected * on an MD110 PBX through TLU76/1 card ? |
13:44.53 | hellop | Man, I've done so many google searches trying to find some prices.. $2499 for some hardware. What about $60000 for 100 line system? |
13:49.14 | *** join/#asterisk boch (~as24@200.59.172.98) |
13:49.21 | Inferna | Darwin35: how can i fix this mopg123? |
13:49.59 | Darwin35 | mpg123 is junk now days look at res_mp3 in the addons |
13:50.44 | Darwin35 | or you can deinstall the ver of mpg123 you have and build the one that * supports |
13:50.47 | Inferna | Darwin35: look let me describe you one thing |
13:50.56 | Darwin35 | in the asterisk src dir type make mpg123 |
13:51.01 | Inferna | Inferna: for SIP calls muscoc is ok |
13:51.01 | hellop | So.. I have a question: How much for a commercial, non-asterisk, pbx? 10-20, 50-100, 100-200 lines. |
13:51.09 | hellop | How about just a guess? |
13:51.10 | Inferna | Inferna: i get this problem only for incoming h.323 calls |
13:51.30 | Darwin35 | I doont deal with h.323 |
13:51.44 | Inferna | <Darwin35> it's not silence supresion problem |
13:51.56 | Inferna | Darwin: becausel voice promts are very good |
13:52.05 | Inferna | Darwin35: only music played by mpg123 |
13:52.09 | Inferna | is suxx |
13:52.12 | Darwin35 | yes |
13:52.24 | CleanerX | can someone do we a favour and do a ns lookup for me? |
13:52.42 | Darwin35 | mpg123 -50r is a pain |
13:52.51 | Inferna | Darwin: i am using the latest one |
13:53.01 | Darwin35 | you will have problems |
13:53.26 | Darwin35 | thats why there is res_mp3 in the asterisk_addons |
13:53.42 | Inferna | Darwin: so i should try res_mp3? |
13:54.25 | Darwin35 | that or uniinstall the mpg123 you have installed |
13:54.40 | Darwin35 | and cd to your asterisk src dir and type make mpg123 |
13:54.57 | Darwin35 | it will pull the last working ver of mpg123 that worked with asterisk |
13:55.03 | Darwin35 | and build and install it |
13:55.16 | CleanerX | can someone do we a favour and do a ns lookup for me? |
13:55.36 | hellop | Doesn't anyone have any idea how much a traditional PBX costs? |
13:55.38 | Darwin35 | I hope we either fix mpg123 soon or drop it |
13:56.38 | Inferna | Ddarwin: i don't see res_mp3 in asterisk-addons |
13:56.52 | *** join/#asterisk fac_ (faceoff@devel.acdbddh.eu.org) |
13:56.55 | fac_ | hello |
13:57.50 | Darwin35 | it might be res_mpg123 |
13:58.33 | Jer1326 | CleanerX what do you need looked up |
13:59.30 | *** join/#asterisk santiago (~santiago@63.245.86.95) |
13:59.39 | theogeor | Hello anybody who has connected * on an MD110 PBX through TLU76/1 card ? |
14:02.24 | Inferna | Darwin: format_mp3, res_perl and mysql |
14:02.30 | Inferna | that's what i can see there |
14:02.34 | Inferna | maybe format_mp3? |
14:03.34 | Darwin35 | think thats it |
14:04.03 | Darwin35 | but if you want to use mpg123 install the one asterisk uses |
14:04.11 | Darwin35 | in your asterisk src dir |
14:05.07 | *** part/#asterisk mogorman (~mogorman@146.229.176.173) |
14:05.40 | *** join/#asterisk kant (~bernd@207.42.191.67) |
14:09.55 | Darwin35 | ok 1.0.6 is about patchd for ports |
14:10.07 | *** part/#asterisk pilif (~pilif@mail.sensational.ch) |
14:10.12 | wizard2000 | when i pick up my handset (plugged in to a TDM) i do not get a dialtone but constant beeping... any ideas? |
14:10.15 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34) |
14:10.21 | wizard2000 | i can dial that phone from a sip extenstion mind |
14:10.31 | Darwin35 | you have vm |
14:10.45 | Darwin35 | its called a studer tne |
14:10.55 | hellop | This is a PBX channel. Can anyone tell me how much 100 line PBX systems cost? |
14:10.58 | Darwin35 | but if your not getting tone at all |
14:11.01 | hellop | Darwin, you know. come on |
14:11.26 | wizard2000 | Darwin35: was that directed at me? |
14:11.44 | Darwin35 | hellop depends on function and hardware |
14:11.48 | Darwin35 | wizard yes |
14:11.56 | Darwin35 | wizard pvt me |
14:12.27 | hellop | Darwin35 standard function, basic. |
14:13.07 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
14:13.33 | mikegrb | hellop: ten million dollars |
14:14.03 | hellop | Really, I'm just looking for some examples of some traditional PBX systems... |
14:14.18 | Jer1326 | hellop: why? |
14:14.43 | hellop | Hundreds of business get started each month, they need PBX, what's the estimated cost? |
14:14.57 | Jer1326 | <1000 |
14:15.06 | Darwin35 | well nec start at 10 grand and go up |
14:15.10 | Darwin35 | thats without vm |
14:15.12 | hellop | Jer1326, Doing a writing assignment for my English 209 Business writing class. |
14:15.28 | Darwin35 | and thats for 50 extensions |
14:15.32 | kant | hellop: With *, you should need to worry about hardware. |
14:16.18 | hellop | kant, so, I'm trying to do a comparisan between Asterisk and non-asterisk. So, I'm trying to find high-priced real world example. |
14:17.09 | Darwin35 | <PROTECTED> |
14:17.34 | hellop | Darwin35, I do searches for stuff like "nex pbx price" or "100 line pbx price" and I find only stuff like $795 two-line pbx. |
14:17.53 | hellop | How about 100 lines pbx price? |
14:18.12 | Darwin35 | I have never priced a 100 unit system |
14:18.28 | Darwin35 | let look in my pricing book for units I use to sell |
14:18.31 | kant | Call Nortel and ask them for a quote on their Norstar system. |
14:18.48 | hellop | Darwin35, You said $10,000 for 50 extensions. That is just the unit, add 10,000 for 50 Cisco 7940's right? |
14:18.49 | drsperm | If you are in the US, check with intertel |
14:18.53 | Darwin35 | Merlin 75 line with out VM start at 9 grand |
14:19.30 | Darwin35 | I have not delt with cisco phones yet |
14:19.50 | drsperm | You could always check out Cisco...that is what is in peoples face pretty heavy now. |
14:20.04 | *** join/#asterisk inspired (mikael@omicron.wwis.net) |
14:20.05 | kant | The phones will cost you more than the PBX in the end... |
14:20.05 | drsperm | Cisco = $$ |
14:20.06 | Darwin35 | merlin 75 line with vm starts at 15,000 |
14:20.20 | hellop | ok, Nortel, Intertel, Merlin.. but you guys are not talking total cost right? Just the unit, then you have to pay consultation fee for setup, right? |
14:20.38 | *** join/#asterisk hajekd (~hajekd@mail.idoox.com) |
14:20.45 | kant | And maintainance fees... |
14:20.52 | dfunnell | Hi all, can anyone help with a dial-out problem I am having? Using CAPI * is dialling as soon as a pattern is matched, which makes it difficult with variable length numbers (such as mobile numbers, etc.) Using exten => _100.,1,Dial,CAPI/470:${EXTEN:1} for example dials out on 00n (where n is the fourth digit dialled). V desperate, will pay in beer. |
14:21.04 | drsperm | I know that the hourly rate for Intertel is $225/hr... |
14:21.08 | *** join/#asterisk hajekd (~hajekd@mail.idoox.com) |
14:21.17 | drsperm | Most linux guys charge between $125 - $180/hr |
14:21.18 | hellop | I'm saying, you are building $50,000,000 manufacturing plant. You need 100 phones. Whats the total cost to install? |
14:21.30 | Darwin35 | unlike asterisk most pbx systems dont come wiith vm built in |
14:21.37 | Darwin35 | so its not a fair pricing |
14:21.39 | hellop | vm? |
14:21.42 | hellop | voicemail |
14:21.45 | Darwin35 | voicemail |
14:21.47 | kant | yes |
14:21.59 | hellop | Well, I mean vm included. |
14:22.00 | Darwin35 | vm is a addon |
14:22.04 | hellop | Total phone system. |
14:22.46 | drsperm | Whenever you get numbers from people like us on line, remember to ask where we are located.... |
14:22.48 | hellop | I worked for a company, they built the thing in a couple months... 100 line phone system installed turnkey. I didn't even notice. We never had to call anyone for help. |
14:22.52 | drsperm | I am in SA, TX, USA |
14:23.08 | hellop | Gave us a stack of books... how to add extensions, voicemail... must have cost alot. |
14:24.16 | hellop | "The thing" built the huge manufacturing plant in a few months.. |
14:24.25 | drsperm | dfunnell: sorry I didn't reply...I have no clue yet... |
14:24.33 | drsperm | linux = veteran |
14:24.38 | drsperm | asterisk = newbie |
14:24.56 | hellop | channels? |
14:25.11 | hellop | huh? |
14:25.33 | FaithX | hey the docs on bristuff are pretty sparse |
14:25.36 | dfunnell | drsperm: no worries, think it has everyone stumped - can't seem to find a resolution for this one anywhere. |
14:26.07 | dfunnell | Should have seen asterisk specialist who came out and had a look yesterday - went home a broken man ;-) |
14:26.09 | hellop | ok, but thanks for the company names.. I guess I'll call them for a quote. |
14:26.15 | drsperm | dfunnell: good luck. catch me in a month, maybe I can help. |
14:26.55 | FaithX | drsperm, do you run a bank or something? |
14:27.08 | drsperm | Why would you say that? |
14:27.34 | dfunnell | Any other takers? Did I mention I will pay in beer? |
14:27.46 | drsperm | oh...sorry I was slow to catch on... |
14:27.53 | MikeJ[Jayden] | what's the issue? |
14:28.41 | hellop | dfunnell, many it sounds like your problem is a simple regex problem, could be fixed in a couple keystrokes. |
14:28.47 | dfunnell | MikeJ[Jayden]: Can't stop * using CAPI from dialling as soon as pattern is matched. |
14:28.49 | tzanger | dammit linux1394.org has been down for a while now |
14:29.01 | dfunnell | hellop: Please go on! |
14:29.03 | hellop | many=man sorry |
14:29.25 | MikeJ[Jayden] | ummm... it's suposed to dial as soon as patern is matched.. although I have no idea on capi |
14:29.31 | hellop | dfunnell, oh.. so Yeah, you need to find in the source code, where it is matching the thing, and change it. |
14:30.03 | djin | dfunnell, is your example the actual line in extensions? |
14:30.35 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
14:30.44 | hellop | dfunnell, I have no idea what you are talking about, but if you tell me what you are trying to match, I could make you a regular expression to match it. |
14:30.55 | dfunnell | MikeJ[Jayden]: Yeah, but using exten => _100.,1,Dial,CAPI/470:${EXTEN:1} it starts dialling on the fourth digit and fails miserably |
14:31.30 | *** join/#asterisk eipi (~eipi@100-172-114-200.fibertel.com.ar) |
14:31.53 | djin | dfunnel, why not use _100XXXXXXXXX.,1,Dial for example? |
14:31.59 | RoyK | hm |
14:32.04 | dfunnell | Means that dialling a US number, for example 100195551234, is impossible, as * tries to dial 001 (leading 1 dropped). |
14:32.12 | RoyK | when do the digium guys wake up? |
14:32.57 | *** join/#asterisk mesi (~player@dsl-082-083-143-045.arcor-ip.net) |
14:33.15 | *** join/#asterisk oej (~oej@63.83.135.35) |
14:33.20 | mesi | Hi oej |
14:33.21 | MikeJ[Jayden] | welll the leading 1 is dropped because you have :1 after ext. |
14:33.23 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
14:33.26 | RoyK | hej.... |
14:33.34 | dfunnell | djin: Problem is numbers are of variable length. Example - mobile numbers in NZ can be 6 or 7 digits long. Using _102XXXXXXX works fine for 6 digit numbers (021 and 6 digits, for ex.), but not with 7 digits. |
14:33.35 | Zeeek | Hej RoyK |
14:33.40 | djin | dfunnell, did you read my remark? |
14:33.51 | djin | Hi Zeeek. |
14:34.04 | RoyK | how can I debug a te410/pri? |
14:34.07 | djin | oops, dfunnel, sorry :( |
14:34.14 | oej | Hej - hello from San Jose |
14:34.26 | Zeeek | I have a server in San José |
14:34.34 | dfunnell | djin: No worries, keep the ideas coming |
14:34.36 | dfunnell | ! |
14:34.37 | Zeeek | say hi |
14:34.37 | MikeJ[Jayden] | hey oej.. |
14:34.41 | mesi | My asterisk is registered with sipphone.com, but whenever I call it I just get an unavailable message. The message I speak then, is mailed to me. Why am I unavailable on sipphone.com? |
14:34.53 | djin | dfunnel, add a _102XXXXXXXX (extra X) then? |
14:34.53 | FaithX | dfunnell, just chuck a few more rules in... |
14:34.58 | Zeeek | mesi did you get FWD working? |
14:35.00 | MikeJ[Jayden] | and is this only on capi calls, or is it no matter what you point it too? |
14:35.30 | MikeJ[Jayden] | if you make it NoOp(${EXTEN}) it does the same thing, right? |
14:35.36 | dfunnell | hellop: deeply uneasy about messing with the source code. Not sure I'd even know what I am looking for, let alone how to change it. |
14:35.46 | drsperm | I have 2 questions for all: |
14:35.46 | drsperm | 1. What sip providers would you reccomend for use in Texas,USA |
14:35.46 | drsperm | 2. Is there any way to pull my vonage numbers into asterisk? I read a posting that it was possible and found a config...I was wondering that everyone thought? |
14:35.54 | mesi | Zeeek: Yes! Thanks. The point was, that it doesn't change this stupid Sip-password. IAX works fine and the password can even be changed. |
14:35.54 | djin | you could create more lines matching variable lengths. |
14:36.09 | dfunnell | MikeJ[Jayden]: Only a problem using CAPI, but this is all I am using for dialling out, so am not using pattern matching for anything else. |
14:36.10 | *** join/#asterisk Darkar (~alex@m174.net81-66-29.noos.fr) |
14:36.12 | Zeeek | mesi aha. So IRC rules, yes? |
14:36.13 | MikeJ[Jayden] | vonage softphone works with asterisk, the ATA's no |
14:36.19 | Darkar | › System Statistics: OS: Windows XP 5.01.2600Service Pack 2 on AMD Athlon(tm) XP 2400+ (AuthenticAMD) Ram: 511MB total, 297MB in use (58%) GPU: 128MB WinFast A340 Uptime: 5hrs 38mins 13secs Hdd's (4): 35.3GB/335GB |
14:36.31 | MikeJ[Jayden] | dfullel see ^^ about noop |
14:36.34 | mesi | Zeeek: IRC rules anywayz. Since about 5 years :-) |
14:37.06 | drsperm | MikeJ[Jayden]: ok...softphone...ie: eliminate my ata and go straight to the asterisk box? |
14:37.08 | *** join/#asterisk faccione (~as@host54-54.pool212171.interbusiness.it) |
14:37.09 | Essobi | Heh. |
14:37.17 | faccione | hiya |
14:37.23 | dfunnell | MikeJ[Jayden]: Not sure about NoOp(${EXTEN}). Sorry for the stupid question, but what is ^^? |
14:37.26 | Essobi | I feel ashamed thinking about how many years I've been logged on to IRC. |
14:37.46 | Lethargicclown | You're saying if I have Vonage then I can only use soft phones? |
14:37.53 | Essobi | dfunnell http://www.voip-info.org |
14:38.15 | Jer1326 | Lethargicclown: * works with vonage if you have a softphone acct |
14:38.20 | faccione | does anybody speaks italian here ? |
14:38.25 | Moc | ^^ mean look at the previous lines |
14:38.46 | drsperm | ok..so I take it there is a difference between a normal vonage account and a softphone account? |
14:38.49 | dfunnell | djin: Tried that (different lines with different lengths of 'XXX's', but * tries to dial as soon as it reaches the shortest match (so only works if number is as long as shortest match). |
14:39.00 | Jer1326 | no softphone is an ADDON to a normal account |
14:39.07 | Essobi | Or /whois Jer1326 |
14:39.09 | Essobi | lol |
14:39.11 | *** join/#asterisk fugitivo (~ajf@201.255.107.93) |
14:39.19 | drsperm | Jer1326: ah. |
14:39.24 | Essobi | Jerjer hiding out, or do we got a new jeremy? :) |
14:39.27 | Lethargicclown | What VOIP provider should I use for *? |
14:39.37 | Essobi | ;) |
14:39.43 | Essobi | I wouldn't admit to it either. |
14:39.48 | Jer1326 | lol |
14:39.49 | dfunnell | Moc: oh, of course ;-) |
14:39.53 | Essobi | Lethargicclown Use Jerjers. |
14:39.54 | *** join/#asterisk ACiDV (~joel@iteckGW.infoteck.qc.ca) |
14:40.16 | Essobi | what's the name of that? Umm. damnit it's too early.. I need coffeeeee. |
14:40.24 | Jer1326 | nuphone? |
14:40.30 | Essobi | Roger, roger. |
14:40.32 | Zeeek | is everyoing on the planet having trouble logging in to free SIP accounts today or what? |
14:40.35 | hellop | dfunnell: don't be scared of messing with the source... you compile asterisk from the source. You might break it, you might fix it... you might learn alot. |
14:40.38 | Essobi | Lethargicclown Use nuphone. |
14:40.40 | FaithX | dfunnell, Is this something that just started happening or did you upgrade? |
14:40.45 | faccione | Gentlemen, question, could be stupid, but i didn't found a solution around. How can I set asterisk to dial ONLY some international prefix and reject all the other ? |
14:40.46 | dfunnell | MikeJ[Jayden]: Are you suggesting NoOp(${EXTEN}) before the dial? |
14:40.51 | Essobi | hellop RTFS is my motto. |
14:41.01 | hellop | Zeeek, maybe the telcos have blocked the ports. |
14:41.04 | ACiDV | I have 2 servers linked with IAX, if I set delayreject=1, I cannot receive call from only 1 side. if I set delayreject=0 all work on both sides, it's normal ? :D |
14:41.16 | Essobi | faccione http://www.voip-info.org |
14:41.23 | Essobi | Search for extension matching |
14:41.26 | Jer1326 | faccione just edit the dialplan |
14:41.27 | Essobi | in the dial plan |
14:41.37 | Zeeek | I have a new application I'm releasing the first day of next month |
14:41.38 | faccione | Essobi i didn't found a solution there. |
14:41.41 | Essobi | _01001234.... |
14:41.43 | dfunnell | Essobi: have spent many hours trawling voip-info.org without success. Def. appreciate any specific URL's you can recommend. |
14:41.45 | Essobi | You didn't look hard enough. |
14:41.55 | Essobi | dfunnell use the google search |
14:41.58 | faccione | does not work |
14:41.59 | Jer1326 | Zeeek what does it do? |
14:42.00 | Zeeek | Ir's called EetMe() and doesn't take ANY argunments |
14:42.08 | Zeeek | muhahahah |
14:42.20 | Essobi | anything you see in a dial plan... use "cmd dial" to lookup dial for instance. |
14:42.24 | hellop | Essobi, RTFS |
14:42.29 | Essobi | :) |
14:42.36 | Essobi | RTFS baby.. RTFS |
14:42.42 | dfunnell | FaithX: New box, so new problem. |
14:42.42 | Essobi | I do it everyday. |
14:42.47 | Jer1326 | whats RTFS? |
14:42.55 | Essobi | READ THE FUCKING SOURCE |
14:42.58 | Essobi | Maha |
14:43.00 | Jer1326 | hahahahahahahahahahahahahahahaahah |
14:43.08 | Zeeek | Roll to Floor Slowly |
14:43.16 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com) |
14:43.18 | Essobi | Use the Source, Luke. |
14:43.28 | Zeeek | May the SOurce... |
14:43.28 | dfunnell | Essobi: If I bothered looking properly then how could I get the chance to dazzle you all with my blazing ignorance? |
14:43.33 | Essobi | Hey Zeeek ... You going to that cook out? |
14:43.45 | faccione | thanks |
14:43.45 | Zeeek | ~cookout |
14:43.52 | Essobi | dfunnell Write that EatMe app zeek was talking about to he couldn't use that joke anymore. |
14:43.57 | Zeeek | ~jbot cookout |
14:44.06 | Essobi | Mehe.. I'll leave that joke along. |
14:44.12 | elric | does asterisk support 64 bit processors? |
14:44.16 | Essobi | Sure. |
14:44.36 | Essobi | alone even |
14:45.09 | Zeeek | what to do what to do? |
14:45.29 | Zeeek | I just got a heart rending letter from a Dr in Nigeria who wants to give me all his money |
14:45.40 | drsperm | Zeeek: count me in. |
14:45.40 | Inferna | can anybody help me with format_mp3, i loaded it, made proper musiconhold.conf configuration, but cannot hear any music on hold in queue |
14:45.46 | roamer323 | 64 bit processor - 4 terabyte of dialing plan - individual permit/deny to every dialable number in the universe - yeah ! :-) |
14:46.05 | epoch | Zeeek: wow, sounds like a worthy cause... think he could use an extra couple grand? |
14:46.12 | FaithX | Inferna, you need mpg123 |
14:46.23 | FaithX | not mpg321 |
14:46.25 | dfunnell | hellop: Sure, may learn a lot, but need to get this puppy installed v soon (like sometime last week!), so a bit worried I'll spend fruitless hours trawling through source I don't understand. Surely someone has done this before without messing with the code. |
14:46.29 | roamer323 | a reload takes nine days on a 16 processor SMP system. |
14:46.39 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
14:46.46 | ManxPower | ~docs |
14:46.47 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
14:46.49 | tzanger | morn ManxPower |
14:46.54 | drsperm | ok...so vonage is out...so which provider should I look at for businesses (without the $$ of a DS1)? |
14:46.57 | ManxPower | 'morning tzanger |
14:46.58 | Makenshi | roamer323, surely you would use realtime for that |
14:47.04 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
14:47.05 | tzanger | drsperm: in or out |
14:47.17 | Inferna | <FaithX> i am using format_mp3 |
14:47.18 | drsperm | of what? |
14:47.20 | ManxPower | drsperm: I don't believe ANY VoIP company is good enough for a business enviroment. |
14:47.27 | tzanger | ManxPower: I disagree |
14:47.38 | drsperm | ManxPower: sorry...mainly for testing.... |
14:47.39 | tzanger | with a good network connection it's no different than traditional POTS |
14:47.55 | tzanger | drsperm: are you looking for long distance termination or someone to route calls to you? |
14:47.58 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
14:47.58 | ManxPower | Even if the VoIP company is good enough, you still have the Big Bad Internet to cause problems. |
14:48.02 | FaithX | Inferna, do you have mpg123 installed? |
14:48.18 | drsperm | here is my topology: * box resides behind 200MB ethernet uplink to internet backbone... |
14:48.20 | roamer323 | Makenshi - sure, and a cluster of mySQL servers chucking away continously :-) |
14:48.23 | tzanger | ManxPower: agreed, but it's really not *that* bad, and use a second cheap connection from another provider for redundancy |
14:48.29 | drsperm | phones reside at business class cable modems... |
14:48.33 | drsperm | pots or sip? |
14:48.43 | tzanger | drsperm: I don't care about your topology, are you looking for origination or termination service? |
14:48.48 | drsperm | both. |
14:48.51 | tzanger | ok |
14:49.07 | drsperm | I can do pots..but I heard I can pick up an echo.... |
14:49.10 | tzanger | termination I use nufone with great success. No, they're not the absolute cheapest but then again they've never gone down for me either |
14:49.13 | drsperm | ...possibly from a bad source... |
14:49.17 | tzanger | drsperm: you can pick up echo anywhere |
14:49.27 | drsperm | k |
14:49.42 | tzanger | for origination it seems much more fractured, there are lots of providers who will provide DIDs but I have not heard of rock-solid reliabiliyt from ANY of them |
14:49.57 | tzanger | I mean just look at the livevoip/broadvoice/sixtel posts on -users |
14:50.02 | tzanger | those are probably the big 3 |
14:50.11 | tzanger | I don't use origination myself though |
14:50.12 | ManxPower | tzafrir: teliax too. |
14:50.36 | tzanger | ManxPower: I haven't heard anything about teliax... they might be worth investigating since nobody's openly bashing them (maybe their shit works?) |
14:50.37 | drsperm | ok..so what you are saying is that for getting started and testing, use an existing pots...then if possible move to a pri or such.... |
14:50.45 | drsperm | ^^if afforadable... |
14:50.47 | Jer1326 | i use telax |
14:50.49 | Jer1326 | teliax |
14:50.49 | tzanger | drsperm: you can still get echo on a PRI (I had it) |
14:50.55 | drsperm | k...thanks... |
14:50.57 | tzanger | Jer1326: how is it for origination |
14:51.15 | Jer1326 | its great i love it...but i cant reach them when i need them |
14:51.19 | *** join/#asterisk gabb0 (~gabb0@indo1.indosoft.unb.ca) |
14:51.35 | tzanger | Jer1326: ever have your DID stop working? How many minutes a month do you originate from them |
14:51.35 | gabb0 | hello |
14:51.36 | Lethargicclown | Anybody have any information on broadvoice and *? |
14:51.48 | tzanger | Lethargicclown: it's all over -users and the wiki. I don't recommend them |
14:51.53 | Jer1326 | i do close to 5000min/month and no more DID has never died yet |
14:51.55 | hellop | echo is the mind killer |
14:51.58 | `Sauron | LEthargicclown: If you follow the voip-info.org pages, it works great. |
14:51.59 | tzanger | Jer1326: very nice |
14:52.08 | tzanger | I will have to check them out |
14:52.15 | gabb0 | Just wondering if anyone has any experience using the tdd agi application or any other method for tty |
14:52.19 | tzanger | we do about 5kmin/mo termination and 10kmin/mo origination |
14:52.20 | hellop | no serious, my customers won't tolerate echo... |
14:52.38 | tzanger | hellop: yes echo sucks |
14:52.46 | Jer1326 | they offered me extra channels free too :) |
14:52.47 | hellop | drsperm, why is vonage out? |
14:52.50 | tzanger | hellop: if it's THAT much of a problem, spend the $1000 and get a hardware T1 echo can, jeez |
14:52.56 | theogeor | Hello anybody who has connected * on an MD110 PBX through TLU76/1 card ? |
14:53.11 | tzanger | hellop: personally zap's builitn ecoh can seems to work well so long as you really tweak it |
14:53.40 | Lethargicclown | what's this about echo? |
14:53.47 | tzanger | what's this about echo? :-) |
14:53.49 | ManxPower | NNNNOOOOOO!!!!! Capital One Bank is going to aquire the local bank I opened an account with THREE DAYS ago. |
14:53.57 | drsperm | hellop: from what I understand softphone only...which is ok..but 500 min is out... |
14:53.57 | tzanger | ManxPower: that sucks |
14:54.13 | hellop | tzanger, I'm in hawaii.. hope it's feasible. |
14:54.22 | tzanger | hellop: hope what is feasable |
14:54.33 | ManxPower | tzanger: I picked that specific bank because they are a regional bank with a good rep for services and decent customer service. |
14:54.37 | *** join/#asterisk channan (~channan9@66.180.121.185) |
14:54.46 | tzanger | ManxPower: well close it out and tell them why |
14:55.21 | ManxPower | tzanger: I'm rather tempted. |
14:55.28 | hellop | tzanger, I hope that when I setup this Bugetone phone, I can make vonage calls over the Pacific Ocean with a quality that is marketable. |
14:55.49 | tzanger | tempted, shmempted, tell them that they've got 6 months to prove that their customer service wont' take a bath with the takeover |
14:56.14 | mesi | Can somebody try to call me on SipPhone? No 17476014869 |
14:56.19 | Jer1326 | hellop you cant resell vonage service.... |
14:56.21 | ManxPower | tzanger: Honestly, I wasn't planning on having the account for more than 12 months anyway. |
14:56.23 | channan | hello - anyone uses Grandstream Budgetone 100 series IP phone model? I bought the 100 model, everything's working ok except the speak phone volume really bad. Would the 101 or 102 be any better? thanks |
14:56.28 | drsperm | so is Broadvoice cool? |
14:56.28 | mesi | I am afraid when calling myself, the line is busy and thus I am unavailable :-( |
14:56.39 | `Sauron | drsperm: Haven't had any problems here |
14:56.42 | hellop | Jer1326, oh. No ersellers contracts? |
14:56.47 | Zeeek | channan - return the phone if the volume is bad |
14:56.49 | Lethargicclown | mesi: it's busy |
14:56.50 | drsperm | `Sauron: thanks... |
14:57.01 | hellop | Jer1326, sides, not reselling, installing their service for a client. |
14:57.05 | mesi | lethargicclown: That's strangge! :-( |
14:57.08 | drsperm | This channel is too active for a Monday morning... |
14:57.15 | `Sauron | Only thing I'm waiting for is for them to port my number.. |
14:57.16 | channan | drsperm- broadvoice is cool. I used it since Christmas and had no problem, calling everywhere in US and Europe |
14:57.18 | `Sauron | dum di dum |
14:57.19 | `Sauron | yawn |
14:57.23 | ManxPower | drsperm: We blame Slashdot. |
14:57.33 | `Sauron | Oh, right. |
14:57.38 | `Sauron | the * article |
14:57.42 | Zeeek | join #asterix |
14:57.46 | `Sauron | I saw it, and bookmarked the page |
14:57.46 | Lethargicclown | Yep, slashdot |
14:57.46 | *** join/#asterisk mrgoby (~mrgoby@defactowireless.org) |
14:57.49 | `Sauron | figured I'll read it later |
14:58.01 | channan | the only thing I don't really like is it's hard to get tek support (even though I did not need it, just want to see how responsive its support is) |
14:58.09 | tzanger | hellop: you're using the cheapest of cheap -- you might want to ebay yourself a nice cisco 7960 or polycom just to compare |
14:58.21 | tzanger | drsperm: you should see it when bkw's in his element |
14:58.31 | drsperm | ah... |
14:58.31 | `Sauron | channan: with BV? You just have to wait a while. ONce you get a guy on the line, they're good |
14:58.37 | Zeeek | Please note that when Newsweek or International Herald Tribune does a big story on voIP they talk about Jeff Pulver and Vonage, never mention asterisk. |
14:58.45 | dfunnell | MikeJ[Jayden]: Tried NoOp(${EXTEN}) before Dial, but * also does NoOp as soon as pattern is matched and then moves to next line (i.e. then tries to dial). Wondering if this isn't a CAPI problem after all? |
14:58.46 | tzanger | Zeeek: mindshare |
14:58.52 | Zeeek | I guess it's because of the media machines these companies have |
14:59.00 | Zeeek | mindshare? |
14:59.02 | hellop | tzafrir, 7940's on the way |
14:59.10 | drsperm | tzanger: you just brought up something...I just ordered some Polycom 500's...r they good (vs. Cisco) |
14:59.17 | par | its all about the benjamins |
14:59.27 | Zeeek | you have to have a mind to share and that lets me out |
14:59.33 | tzanger | Zeeek: exactly -- asterisk is much less noticeable becuase they're not spewing their name everwhere and coining phrases like "I got voiped" |
14:59.41 | tzanger | drsperm: they are supposed to be very nice |
14:59.46 | ManxPower | Cisco: You don't get SIP firmware with it, you don't get a power supply with it. Polycom: You get SIP firmware with it, you get a power supply with it. You pick. |
14:59.47 | drsperm | cool... |
14:59.49 | tzanger | I don't do SIP so I cant' say from experience |
14:59.55 | Zeeek | Duuuude... I got Ass terisked |
15:00.00 | tzanger | hahahaha |
15:00.03 | Jer1326 | hahaha |
15:00.03 | hellop | tzanger, I was just wondering if the long distance to Hawaii, would make the VOIP unusable due to echo... |
15:00.05 | tzanger | dude, your ass got tricked |
15:00.07 | drsperm | That is what I heard...Cisco is very $$ too.. |
15:00.12 | Zeeek | kiss my .... |
15:00.27 | ManxPower | hellop: Location doesn't matter. Network latency and jitter matter. |
15:00.38 | tzanger | hellop: only if your hybrids are shit or the handset is so fucked up (mechanically) that all your earpiece audio gets received by the mic |
15:00.52 | channan | Zeeek-I bought in ebay and can't return (did not notice the prob until I needed :( ). are you saying that you don't have that problem? Perhaps mine is just a defective unit? |
15:00.55 | Zeeek | The Internat'l Herald Trib has "email this article... AIM this article... IM this article" |
15:01.08 | tzanger | what's the difference between AIM and IM |
15:01.08 | tzanger | ? |
15:01.16 | dfunnell | Essobi: Searched again for resolution, including looking at cmd Dial, haven't found anything new to help. |
15:01.26 | Zeeek | channan I'd say so because [inspite of all the guys that diss the GS line] the phone works GREAT! (for $75) |
15:01.30 | *** join/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net) |
15:01.35 | MuppetMaster | AIM is specific to AOL while IM is a broader term for AOL, MSN, Yahoo, Jabber, SIMPLE, etc. |
15:01.36 | tzanger | yes for the money it is not a bad phone |
15:01.43 | Zeeek | it looks a little wacky and has clown buttons |
15:01.50 | Zeeek | but the speaker ROCKS! |
15:01.53 | tzanger | the white ones look like ass but the lback ones are nice |
15:01.58 | drsperm | anyone heard of a telco called "Topaz"...I hear they have good $$ on site to site DS1's |
15:02.03 | Zeeek | somehow I could onlmy order black ones |
15:02.08 | tzanger | Zeeek: if you think the gs speakerphone rocks you have never been on a good conference call with a speakerphone |
15:02.11 | Zeeek | I have a 101 and a 102 |
15:02.29 | tzanger | the polycom (non voip) speakerphones are bar none, hands down the *best* |
15:02.31 | Zeeek | well I don't need a $2000 sound point, thas fo sho' |
15:02.38 | Lethargicclown | Nothing's better then my half duplex speaker phone! |
15:02.45 | tzanger | Lethargicclown: yeah, no echo there! :-) |
15:02.46 | Lethargicclown | for only $120 |
15:02.48 | Zeeek | but I am getting an ip500 so I'll see what that's up to next |
15:03.02 | par | who is the cheapest CLEC offering T1? |
15:03.28 | drsperm | par: site to site? |
15:03.32 | Zeeek | also ordered on of them IAX chinses ones "just to see" |
15:03.32 | par | yepo |
15:03.42 | drsperm | par: State? City? |
15:04.06 | par | no sorry instead.. not site to site |
15:04.20 | drsperm | k...where? |
15:04.25 | par | texas |
15:04.27 | drsperm | 4 voice i take it... |
15:04.30 | drsperm | city? |
15:04.35 | par | no not a PRI |
15:04.42 | drsperm | k...city? |
15:04.43 | par | integrated T1 |
15:04.46 | par | San Antonio |
15:04.52 | channan | drsperm-cisco7960 quality is really good. I owned a few. the only thing is it's pretty hard to set up the first time. My friend said it sounds a lot better than caling thru my analog home phone. the price is steep though |
15:04.56 | ManxPower | par: nobody can answer that. |
15:05.03 | ManxPower | par: You need to do the research for your location. |
15:05.21 | drsperm | par: I can...I live in SA....and own an ISP in sa... :) |
15:05.30 | Zeeek | It's Katty!!!! http://www.mindsay.com/network/kat1587 |
15:07.33 | *** join/#asterisk wazquis (~akv@lnxbx.dk) |
15:07.58 | wazquis | ain't it possible to run asterisk without a soundcard? i cannot start asterisk up on my laptop |
15:08.04 | Zeeek | sure it is |
15:08.16 | Zeeek | don need no stinking sound card |
15:08.23 | wazquis | i tried noload => alsa.conf and oss.conf |
15:08.34 | Zeeek | what error do you get? |
15:08.37 | wazquis | .......Ouch ... error while writing audio data: : Broken pipe |
15:08.40 | ManxPower | wazquis: You mean noload => chan_alsa.so and chan_oss.so, right? |
15:08.43 | Zeeek | before that |
15:08.54 | ManxPower | wazquis: That error is from mpg123 |
15:08.56 | MuppetMaster | wazquis: It is possible to run Asterisk with a soundcard. |
15:09.03 | Zeeek | or not. |
15:09.08 | wazquis | ManxPower, oh... |
15:09.11 | wazquis | i'll check |
15:09.25 | kant | When using the 'switch =>...' statement to forward a call to another * server, will CDRs be written to the forwarded * server? |
15:09.41 | wazquis | ManxPower, yes... chan_alsa.so and chan_oss.so |
15:09.45 | ManxPower | kant: Only for the portion of the call that the remote server handles. |
15:10.37 | hellop | tks all |
15:10.42 | Zeeek | Question: A call comes in on ZAP with CID. You do NOT answer it. Is it then possible to call back on the same line if the caller hung up? Without a .call file ? |
15:11.02 | ManxPower | Zeeek: Yes. |
15:11.06 | kant | So if I forward a call to an * server with all the PSTN hookups, that * will have the CDRs corresponding to the calls made through it? |
15:11.22 | ManxPower | kant: Yes. |
15:11.29 | wazquis | noload=>chan_alsa.so |
15:11.34 | wazquis | ........Mar 7 16:10:33 WARNING[1077138752]: chan_skinny.c:2584 reload_config: Unable to get our IP address, Skinny disabled |
15:11.35 | Zeeek | Manx I tried it in the h extension but no matter what I did it wouldn't work |
15:11.38 | wazquis | the only error i get now.. |
15:11.45 | Zeeek | do I need to lmiberate the channel, destroy it? |
15:11.49 | wazquis | (and the mpg123 thingie..) |
15:11.59 | kant | And the originating * will have CDRs corresponding to the 'switch =>'? |
15:12.04 | ManxPower | wazquis: Unless you are using the Skinny protocol, that is a harmless message. |
15:12.18 | Zeeek | not an error but a WARNING |
15:12.28 | ManxPower | Zeeek: You mean AUTOMATICALLY call them back? No. |
15:12.36 | Zeeek | heh, amusing |
15:12.54 | Zeeek | ya, I can call them back on my cell too if I write the number down :) |
15:13.09 | wazquis | ManxPower, okay... but the asterisk ain't starting up...the last message is the one you said was from mpg123 |
15:13.12 | ManxPower | Why not use the call back function of your phone? |
15:13.24 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
15:13.30 | Zeeek | Manx what I did was create an app that waits 2 seconds and then copies a .call file to outgouing |
15:14.42 | *** join/#asterisk linagee (~linagee@netblock-66-245-227-114.dslextreme.com) |
15:14.52 | linagee | cool. they have a user friendly asterisk iso now. :) |
15:15.17 | dfunnell | Essobi: Come on, Essobi, you talk big about not searching hard enough, but it doesn't sound like you have any more idea than I do re. finding an answer to this one! |
15:17.15 | Zeeek | Manx - it's not a phone that's being called it's a DID from the USA |
15:17.24 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
15:17.45 | Zeeek | so the .call file works fine, it was just a thought about changing the channel dynamically |
15:18.04 | RoyK | anyone that knows what to do if the te410p doesn't get/put any interrupts? |
15:18.14 | Zeeek | DIDcomingn in to asterisk I mean, needing in fact to call out SIP to a phone elsewhere |
15:18.46 | ManxPower | RoyK, Put it in a system without the incompatable Intel chipset. |
15:18.49 | *** part/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
15:19.06 | RoyK | ManxPower: well, I have another box with bloody identical hardware and there it workks |
15:19.11 | RoyK | s/kks/ks |
15:19.59 | ManxPower | RoyK, It's a problem with a specific Intel Chipset. |
15:20.14 | techie | Another day in the world of packetized voice. |
15:21.12 | *** join/#asterisk MasterYoda (~mnicholso@dhcp-155.digium.com) |
15:21.21 | ManxPower | RoyK, see the mailing list archive. I can't find the page on digium's web site that talks about it. |
15:22.22 | *** join/#asterisk tull (~danka@wwwcache2.livjm.ac.uk) |
15:22.28 | tull | hello |
15:22.41 | tull | is anyone using sipura 2000 or spa3000? |
15:23.00 | ast_freak | sipura 2000 |
15:23.07 | tull | may I pvt? |
15:23.33 | ast_freak | sure |
15:23.38 | *** join/#asterisk viLeR (1000@ip-33-7.telesat.com.co) |
15:24.02 | wazquis | ManxPower, the asterisk doesn't start up... i if it those two aint problems..i cannot see what is going wrong |
15:24.45 | *** join/#asterisk hemant (hemant@220.226.25.97) |
15:24.47 | ManxPower | wazquis, noload => chan_skinny.so in /etc/asterisk/modules.conf and rename /etc/asterisk/musiconhold.conf to some other name. |
15:26.41 | Zeeek | wazquis what you should do is pastebin the console output, several lines before it stops |
15:26.47 | Zeeek | ~pastebin |
15:26.49 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
15:26.59 | kant | Finally I get multi-homed connection. |
15:27.04 | RoyK | ManxPower: you sure? http://www.voip-info.org/wiki-Asterisk+TE410p+No+Interrupts |
15:27.19 | ManxPower | RoyK, Therre it is. |
15:27.36 | *** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl) |
15:27.43 | Zeeek | How common is use of IAX (not IAX2) today? |
15:27.56 | ManxPower | RoyK, That's not the page I was thinking of. |
15:28.00 | RoyK | ManxPower: ok |
15:28.04 | ManxPower | Zeeek, nobody uses IAX |
15:28.08 | tzanger | hahhaa |
15:28.35 | Zeeek | I'm asking because of the zillion pages on the web that show the wrong port |
15:28.53 | Zeeek | [general] |
15:28.53 | Zeeek | port=5036 ; What port to use |
15:29.13 | ManxPower | Zeeek, Asterisk will ignore that in modern versions. |
15:29.28 | Zeeek | [general]If shit like that is NOT true, we should all get rid of it everywhere we can |
15:29.43 | Zeeek | it's BAD to even show stuff like that |
15:29.51 | Zeeek | needless waste (a little like my posts here) |
15:30.47 | RoyK | ManxPower: how come? |
15:30.52 | Catalyst4Change | Reform Asterisk Social Security now |
15:31.15 | wazquis | ManxPower, ok...i get no errors now... but it just stops after initializing... http://pastebin.com/250517 |
15:31.31 | *** join/#asterisk jalsot_ (~tamas@abacus.eworldcom.hu) |
15:31.44 | MasterYoda | Zeeek: what about port = 5036? for iax? |
15:31.58 | Zeeek | it's obviously not right for IAX2 |
15:32.07 | MasterYoda | Zeeek: well iax is 4569 |
15:32.56 | mesi | What can be the reason that I am unavailable on sipphone.com if I am registered and can make calls and the firewall is open? |
15:32.57 | *** join/#asterisk file (~file@251.134.218.209.transedge.com) |
15:32.58 | RoyK | ManxPower: but then how do you explain that the card works in identical box? |
15:33.34 | tzanger | RoyK: easy, the identical box isn't |
15:33.37 | ManxPower | RoyK, I can't. |
15:33.41 | tzanger | RoyK: or the card got damaged in transit |
15:33.45 | dfunnell | All - pretty sure that my problem is * related and not CAPI related. |
15:33.46 | tzanger | RoyK: or the motherboard has a bad port |
15:33.47 | dfunnell | * tries to do whatever (run macro, dial, etc.) as soon as it hits '.' in _102. or similar. Is this normal? Any way of getting it to wait for user to dial whole number? |
15:33.48 | ManxPower | tzanger, Well that is the OBVIOUS answer. |
15:33.50 | tzanger | RoyK: any number of reasons |
15:34.04 | tzanger | RoyK: or the BIOS isn't set to identical settings (I have personally seen this) |
15:34.07 | ManxPower | dfunnell, no. |
15:34.23 | MasterYoda | dfunnell: dial faster.... |
15:34.31 | tzanger | MasterYoda: :-) |
15:34.35 | ManxPower | dfunnell, unless you do NOT know the number of digits. Of course if you are using a SIP phone this doesn't apply. |
15:35.08 | dfunnell | MasterYoda: ;-) |
15:35.21 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
15:35.21 | *** mode/#asterisk [+o anthm] by ChanServ |
15:35.46 | dfunnell | ManxPower: Bugger. That is the whole problem - don't know number of digits that make up number. |
15:37.18 | RoyK | ManxPower: http://karlsbakk.net/asdf.txt |
15:37.25 | RoyK | ManxPower: that's a simple lspci |
15:37.37 | *** join/#asterisk Jas_Williams (~Jason@host81-155-66-178.range81-155.btcentralplus.com) |
15:37.43 | dfunnell | ManxPower: Are you saying that, for all of it's features, * doesn't provide a (relatively straightforward) way to support variable-length numbers? |
15:39.02 | *** join/#asterisk xorol (~x@213.219.182.88.fixedpower.by.edpnet.be) |
15:39.04 | xorol | ello |
15:39.11 | ManxPower | dfunnell, If you don't know the number of digits (and only if you don't know the number of digits) then you have to use "." in your pattern. Of course, if you are using VoIP then this does not apply. |
15:39.15 | xorol | i got a little question .. |
15:39.23 | jontow | i have a feeling we're gonna see that '@home' nonsense an awful lot in the next couple days ;) |
15:39.35 | linagee | jontow: lol |
15:39.38 | ManxPower | dfunnell, no, I'm saying that most people that think they need variable length numbers don't actually ned them. |
15:39.48 | linagee | jontow: naw. i was in here prior to @home stuff. |
15:40.07 | xorol | i'm trying to transfer a call by using dial(channel,30,t) but i can't transfer the call afterwards (im using WELLTECH 1501 (fxs gateway) as hardware) |
15:40.16 | ManxPower | jontow, Thank Dog that I'll be online much less over the next few days. |
15:40.21 | dfunnell | ManxPower: Please go on. I'm only having problems dialling out, where I don't have the ability to control number length. |
15:40.21 | jontow | hahah |
15:40.21 | xorol | it won't react by sending the # ... |
15:40.34 | linagee | jontow: personally i don't like @home stuff. ;-) make users buy phones, not backend systems! :) |
15:40.40 | *** join/#asterisk Smythe (~Smythe@spock.cbcag.edu) |
15:40.42 | ManxPower | Pinhole, And your AGI is run twice for each call? |
15:40.48 | dfunnell | ManxPower: When I use '.' it tries to dial as soon as it reaches the '.' (i.e. doesn't wait for the whole number) |
15:41.09 | ManxPower | dfunnell, You never told me what device you arre using to dial. |
15:41.14 | jontow | personally .. i like ramps for learning curves, but eh.. honestly sometimes you need to just climb the ladder :) |
15:41.22 | Pinhole | ManxPower, yup. sometimes more than that. |
15:41.33 | ManxPower | Pinhole, THAT is the problem with _. |
15:41.44 | dfunnell | ManxPower: example dial string: exten => _102.,1,Dial,CAPI/470,${CALLERIDNUM},${EXTEN:1} |
15:42.12 | ManxPower | dfunnell, Are you going to make me start saying naughty words to get you to tell me what phone you are using? |
15:42.28 | *** join/#asterisk Ubuz (~momo@DSL212-235-37-117.bb.netvision.net.il) |
15:42.31 | dfunnell | ManxPower: Doesn't only seem to be problem with CAPI, if I try and run macro of function (such as NoOp) it happens straight away too. |
15:42.45 | dfunnell | ManxPower: Sorry, SIP phones - Granstream |
15:42.51 | dfunnell | Budgetone 100 |
15:43.09 | ManxPower | dfunnell, Your phone collects the whole number, then sends the entire dialed number to Asterisk. |
15:43.19 | linagee | much moreso than traditional four lettered words |
15:43.37 | dfunnell | ManxPower: Give me a couple of secs to try something out. Dont' go anywhere! |
15:43.46 | ManxPower | dfunnell, So look at the phone. Specifically Early Dial does not work on those phones. |
15:44.11 | linagee | ManxPower: unless of course you tell your phone to send numbers as soon as they're dialed or something weird and unstandard. (grandstream phones can do that at least) |
15:44.14 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
15:44.14 | *** mode/#asterisk [+o bkw_] by ChanServ |
15:44.34 | ManxPower | linagee, Yes, but that feature does not work in any phone I've heard of. |
15:44.41 | linagee | ManxPower: probably not. ;-) |
15:44.54 | *** join/#asterisk Bentley (~rbc@S01060080c8135e6a.cg.shawcable.net) |
15:45.06 | *** join/#asterisk jalsot_ (~tamas@abacus.eworldcom.hu) |
15:45.19 | dfunnell | ManxPower: So you think I need to turn early dial off? If so then do you know any way of avoiding a delay between user dialling internal extension and it actually doing anything? |
15:45.22 | RoyK | ManxPower: see http://karlsbakk.net/te410p/. the difference between the two lspci -vvv output is two lines |
15:45.42 | jontow | dfunnell; with the grandstreams, you can 'tune' the timeout period for dialing.. |
15:45.45 | jontow | its in the web GUI :) |
15:45.48 | ManxPower | dfunnell, You use a non-piece-of-crap phone |
15:46.01 | jontow | thats about all you've got for options though.. |
15:46.46 | ManxPower | dfunnell, mid and high end phones let you put a dialplan in the phone. |
15:46.56 | dfunnell | jontow: Ok, thanks. Also have 4 x analogue phones plugged into Zap's, am I going to have same problem with those? |
15:47.04 | jontow | no clue |
15:47.11 | jontow | i've yet to use ATAs in any way shape or form.. |
15:47.12 | ManxPower | dfunnell, But if you use . in your dialplan you will have to wait until DigitTimeout before the call will happen. |
15:47.13 | jontow | <PROTECTED> |
15:47.14 | dfunnell | ManxPower: I see. Thanks for your help. Trying now. |
15:47.16 | mesi | Everything on sipphone works for me except that I can't be called on my number. |
15:47.25 | jontow | mesi; so setup your extension.. |
15:47.32 | *** join/#asterisk trimi` (Pharrel@62.162.232.119) |
15:47.35 | ManxPower | dfunnell, zap devices send each digit as it is dialed. |
15:47.49 | dfunnell | ManxPower: Thanks ManxPower. Do you mean DigitTimeout in phone? |
15:47.52 | *** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net) |
15:47.57 | jontow | no, DigitTimeout in * |
15:48.06 | ManxPower | dfunnell, for zap, digittimeout in * |
15:48.20 | ManxPower | for sip, the digit timeout on the phone |
15:48.25 | MikeJ[Jayden] | Manx, you should get paid for this... |
15:48.34 | MikeJ[Jayden] | somone paypal Manx some cash |
15:48.34 | dfunnell | ManxPower: Oh, I see! |
15:48.36 | ManxPower | MikeJ[Jayden], too bad NOBODY else feels that way. |
15:48.57 | dfunnell | ManxPower: I've already offered to pay in beer... when are you next in NZ? |
15:49.15 | MikeJ[Jayden] | who here apretiates all of ManxPower's help? |
15:49.59 | mesi | jontow: I set it up. It doesn't seem to work. In sip.conf I say context=extern-sip, so that this context would be the default for incoming sip calls. And the register line names /1747 at the end. So this extension should be called, right? |
15:50.16 | jontow | manxpower; i've always felt that way.. but i can't stand paypal :( |
15:50.27 | jontow | you helped me repeatedly in subtle ways when i was first picking this stuff up ;) |
15:50.30 | dfunnell | MikeJ[Jayden]: Let me answer that once I see if it is of any use ;) |
15:50.38 | ManxPower | see my /away message |
15:51.33 | MikeJ[Jayden] | so, you wana go to europe? |
15:51.34 | ManxPower | jontow, I'm alwways most proud of my insults. 8-) |
15:51.39 | ManxPower | MikeJ[Jayden], Yes. |
15:51.39 | jontow | heheh |
15:51.51 | Zeeek | GUYS This is it ! the breakthough! |
15:51.54 | jontow | i wouldn't mind europe.. :) |
15:51.59 | jontow | i think it'd be a hell of a good time |
15:52.01 | ManxPower | MikeJ[Jayden], I would prefer Benelux, but most any place there would be OK |
15:52.04 | Zeeek | The KILLER VOIP APP not kidding |
15:52.04 | jontow | at least until i get arrested for something stupid :o |
15:52.04 | visik7 | european software patents has been approved |
15:52.15 | BuckRogers | <PROTECTED> |
15:52.44 | ManxPower | To everyone that I've helped: Find me a job in Europe. |
15:52.51 | jontow | mesi; thats a little out of context.. pun intended.. |
15:53.08 | Zeeek | All voIP providers need to arrange credit accounts. When we want to pay someone that helped us, we just dial a number at the provider and Allison will arrange the cross billing! What do you think? |
15:53.17 | jontow | manx; i'd be looking.. but im looking for a new job for me.. my employer(s) are a joke :/ |
15:53.33 | Zeeek | Manx want to do PHP, C and web design in Paris? |
15:53.35 | mesi | jontow: What is pun? |
15:53.36 | ManxPower | jontow, My entire COUNTRY is becoiming a joke. 8-) |
15:53.46 | ManxPower | Zeeek, I am not a programmer. |
15:53.48 | jontow | ;) you're in the USA like me, hey? |
15:53.57 | ManxPower | jontow, Yup. |
15:54.08 | jontow | yeah.. agreed in full. |
15:54.18 | Zeeek | Manxpower there are a lot of startupasterisk based businesses here in Paris |
15:54.36 | ManxPower | jontow, Not just That Bush Thing. It all started in the 1980's |
15:54.40 | jontow | if you find that position in europe.. want an apprentice? my gf too :P |
15:54.41 | techie | Paris, nice. |
15:54.58 | jontow | i know .. i was born in the 80s, of course it was ajoke :) what with the spandex, big hair, and showboat-rock-singers |
15:55.04 | Zeeek | ManxPower, is the word "cheese-eatin surender mnkeys" in your vocabulary? If it is, forget it! |
15:55.16 | *** part/#asterisk kant (~bernd@207.42.191.67) |
15:55.16 | ManxPower | Zeeek, Unfortunatly, I'm a VERY VERY bad programmer. I can do simple scripts in perl, PHP, C, but I just can't do big projects. |
15:55.29 | *** join/#asterisk Rival (~rival@66.177.249.219) |
15:55.36 | Zeeek | Asterisk based starups Manx, I'll arrange it for you |
15:55.45 | ManxPower | Zeeek, I'm not all that fond of what little French culture I've seen/heard of, but I don't hate the french. |
15:55.53 | Zeeek | ok, thazt's good |
15:56.10 | jontow | I like the French.. but I'll be honest.. I'm just a sucker for the wine. |
15:56.22 | Zeeek | because Amer'ican culture isn't much to be crazy about either :) To each his Zone |
15:56.27 | jontow | the language is fun too :) especially once you're all hopped up on the wine. |
15:56.30 | ManxPower | I have to major issues with working in Europe. 1) I'm not a citizen of the EU, so there's lots of paperwork 2) Like most americans, I only speak English. |
15:56.53 | Zeeek | Both can be fixed Manx - why do you want to work in Europe? |
15:56.59 | xorol | manx : belgian speak the BEST english in the EU (besides UK ofcourse .. try me ) |
15:57.06 | ManxPower | Zeeek, I want to get out of the USA on (maybe) a perm basis. |
15:57.17 | Zeeek | Canada isn"t good enuf? |
15:57.20 | ManxPower | xorol, Well, my preference would be a job somewhere in Benelux. |
15:57.33 | ManxPower | Zeeek, Canada would just be a stepping stone to Europe. |
15:57.34 | xorol | manxpower : we are hiring :) |
15:57.52 | xorol | manxpower : we are even located in brussels |
15:57.56 | ManxPower | xorol, Please tell me you are hireing for something other than a programmer. |
15:57.58 | Rival | is there a problem with http://asteriskathome.sourceforge.net/ |
15:58.00 | Zeeek | Manx - the language is not a problem in this business and you surely can network here and in other places to find a job |
15:58.02 | Rival | i cant connect |
15:58.04 | ManxPower | xorol, I may be in Burssels this summer. |
15:58.17 | xorol | Manx:: just pass by .. we'll have a lunch in the Hilton |
15:58.18 | Rival | is there a secondary link anyone would be kind enough to msg me with? |
15:58.27 | ManxPower | xorol, your e-mail address? |
15:58.49 | ManxPower | xorol, If I get a job offer, I won't wait until summer. 8-) |
15:58.59 | ManxPower | ~docs |
15:59.00 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
15:59.12 | *** join/#asterisk kswtch (~killswitc@213.146.107.241) |
15:59.16 | Zeeek | ManxPower where can we find a recent CV of you ? |
15:59.19 | ManxPower | I want to go to Europe this summer for VON and Astricon. |
15:59.31 | ManxPower | Zeeek, I have to finish making one. |
15:59.50 | ManxPower | xorol and zeek, /msg me your e-mail addresses. |
16:00.15 | RoyK | ManxPower: did you see the lspci outputs? |
16:00.35 | ManxPower | RoyK, No. I can't really help with hardware issues at that level. |
16:00.59 | RoyK | ManxPower: the point is, they're equal |
16:01.24 | Zeeek | ManxPower send one to me here: ManxPower-RMRAsterisk@sneakemail.com |
16:01.32 | ManxPower | RoyK, The only thing I know about no-interrupts is what I read on the mailing list. |
16:01.39 | ManxPower | Zeeek, Does that address expire? |
16:01.48 | nestAr | wtf |
16:01.51 | Zeeek | no but I'll have to kill it if I start getting crap |
16:01.58 | nestAr | Polycom says you can't reprogram the softkeys |
16:02.00 | ManxPower | Zeeek, Of course. |
16:02.00 | Zeeek | Sneakemail rulez! |
16:02.23 | Zeeek | should be fine for a few weeks minimum |
16:02.52 | nestAr | argh |
16:02.53 | Zeeek | Also Max, take a look at the guru.com site and see if stuff is open in the areas that interest you (both geo and work wise) |
16:03.02 | Zeeek | Max/Manx/ |
16:03.05 | nestAr | i'm starting to regret buying these phones. |
16:03.16 | Zeeek | which phones nestAr ? |
16:03.21 | nestAr | Polycom's |
16:03.29 | Zeeek | OH? |
16:03.36 | nestAr | what's the point of having soft buttons |
16:03.40 | nestAr | if you can't change them |
16:03.40 | Zeeek | I haven't ordered yet, what's wrong? |
16:03.57 | Zeeek | none of the docs mention this? |
16:04.12 | nestAr | there's a section about reprogramming a whole key |
16:04.23 | nestAr | doesn't exactly make a lot of sense to me |
16:04.24 | dfunnell | ManxPower: Thanks Manx, it all works, can't believe it was that easy! |
16:04.32 | nestAr | so i inquire with tech support |
16:04.35 | *** join/#asterisk gdb (~cbell@circe.inetdb.com) |
16:04.36 | nestAr | i get a one line reply.. |
16:04.41 | nestAr | "Voip engineering has not allowed the softkeys to be programmed. |
16:04.41 | nestAr | " |
16:04.56 | ManxPower | nestAr, Polycom does NOT do end user support. |
16:05.02 | ManxPower | nestAr, they lie. |
16:05.27 | dfunnell | ManxPower: Where do I send the beer? |
16:05.38 | ManxPower | I managed to program a softkey on a Polycom about 6 months ago. Decided that it was more trouble than I had time to deal with at that time, since it was a "cool thing" and not a "required thing" |
16:05.40 | *** join/#asterisk waszi (waszi@vorlon.icpnet.pl) |
16:05.51 | ManxPower | dfunnell, Can you paypal it to me? *grin* |
16:06.22 | nestAr | fucking cocksucking bastards is what they are. |
16:06.31 | Zeeek | Polycom guys, what docs do you have? I noticed about 5 different PDF available in open download? |
16:06.39 | ManxPower | nestAr, I talked to at least one other person that did it at the time, the FUCTION of the key is set in one place, the LABEL for the key is done in the language localication stuff. |
16:07.05 | *** join/#asterisk eKo1 (~bernd@207.42.191.67) |
16:07.15 | ManxPower | Zeeek, I use the Polycom Admin Soundpoint IP SIP document. |
16:07.38 | Zeeek | and that has enuf tech data to program if you have the time? |
16:08.02 | ManxPower | Zeeek, That and a LOT of work. It's a reference guide for people that already know how to do it. |
16:08.30 | Zeeek | so in short, when we get this PayManx() app ruinning, you'll be able to do it? |
16:08.44 | ManxPower | LOL! |
16:09.09 | Zeeek | did you see what I said above? Providers like nufone could give us a number to call and this would credit your account (with them or for cach) |
16:09.12 | MuppetMaster | Anyone on here having problems with MOH after upgrading to v1.0.6? http://voxilla.com/forum-viewtopic-t-2586.html |
16:09.22 | Zeeek | totally doable by and for the asterisk community |
16:09.36 | Zeeek | Muppet ya, it's broken |
16:09.49 | ManxPower | MuppetMaster, Yes, 1.0.6 has a MoH bug. get CVS -r v1-0 or wait for 1.0.7 |
16:09.53 | *** join/#asterisk jalsot_ (~tamas@abacus.eworldcom.hu) |
16:09.53 | MuppetMaster | Okay, is there already a bug report. |
16:09.53 | Zeeek | http://willypick.mindsay.com/?entry=16 |
16:10.09 | Zeeek | ans a patch |
16:10.26 | mesi | Can somebody please call 17476014869 again and tell me what happens? |
16:10.31 | mesi | on sipphone. |
16:11.07 | Zeeek | You you call a number like Pay-Manx on nufone and nufone moves credit from your account to the payee's - ya see? |
16:11.14 | Zeeek | this is what the Japanese do with cellphones |
16:11.24 | Zeeek | bypass PayPal |
16:11.46 | Zeeek | instant notification - the payee can then help you |
16:11.54 | Zeeek | it isn't as dumb as it sounds |
16:12.17 | Zeeek | all stays in the community |
16:12.23 | Zeeek | anyone out there? |
16:12.47 | ManxPower | Zeeek, It's a cool idea, seems like it would require a LOT of work not related to Asterisk. |
16:12.48 | JerJer[mobile] | nope |
16:12.51 | JerJer[mobile] | just us bots |
16:12.51 | kswtch | "We're sorry, the user bla bla has not setup voicemail bla bla " |
16:12.53 | kswtch | @ Zeeek |
16:13.36 | Zeeek | just as I convinced my partner to use the SIP unlimited channel, ever rpovider is UNREACHABLE |
16:13.43 | MuppetMaster | Zeeek: Where may I find the bug report for MOH that contains details on the patch? |
16:13.49 | kswtch | oh it was mesi who requested the call... |
16:13.56 | MuppetMaster | Zeek: Can't seem to find it on Mantis with MOH/music on hold/etc. |
16:14.12 | Zeeek | just a sec I was there yesterday |
16:14.43 | Zeeek | <PROTECTED> |
16:15.01 | Zeeek | in fact download THIS: http://bugs.digium.com/file_download.php?file_id=5032&type=bug |
16:15.53 | Zeeek | Manx a lot of shit requires a lot of work not related to Asterisk |
16:16.10 | MuppetMaster | Thanks! |
16:16.10 | Zeeek | one thing is sure - PayPal, as good an idea as it is, SUCKS |
16:16.23 | jontow | so does adobe acrobat reader for unix |
16:16.27 | Zeeek | they fucking spam me every time I touch anything PayPal related |
16:16.55 | Zeeek | Adobe in genral sucks and thaey also are world class spammers and home callers - as good as Real Networks |
16:17.04 | nestAr | i don't get any paypal spam |
16:17.06 | nestAr | :dunno: |
16:17.24 | `Sauron | I think paypal has slowly crept up far enough in my bayes filter that they're getting filtered out |
16:17.25 | eKo1 | I get paypal spamn. |
16:17.36 | *** join/#asterisk xpasha (~pavel@217.30.252.68) |
16:17.43 | `Sauron | which has the added side effect of also filtering legit paypal email (payment confirmations, etc) |
16:17.47 | xpasha | hi |
16:17.51 | eKo1 | Don't know why since I never use it. |
16:18.02 | xpasha | anybody used zaphfc from bristuff? |
16:18.14 | Zeeek | They keep sending me stuff about how my life would be better if I got Verified(tm) |
16:18.26 | `Sauron | hum |
16:18.30 | `Sauron | yeah |
16:18.32 | `Sauron | they do that to me too |
16:18.40 | `Sauron | it's cuz they want access to your bank account |
16:18.42 | Zeeek | Every purchase |
16:18.43 | shaZwaz | verified ? |
16:18.56 | Zeeek | I did that once - too scaray |
16:18.57 | shaZwaz | oh |
16:19.01 | xpasha | Mar 7 17:18:16 localhost kernel: zaphfc: bchan rx fifo not enough bytes to receive! (z1=528, z2=527, wanted 8 got 2), probably a buffer overrun. |
16:19.03 | `Sauron | yeah |
16:19.05 | xpasha | what is the shit? |
16:19.14 | `Sauron | they want you to associate your paypal account with a bank account |
16:19.21 | `Sauron | so they can do ACH payments/withdrawals |
16:19.24 | Zeeek | PLUS I tried to pay using my normal debit card at two sites yesterday and the payment was refused |
16:19.54 | `Sauron | I did finally break down and associate one of my low-limit credit cards with my paypal account |
16:19.55 | *** join/#asterisk alerque (~alerque@bear.ouraynet.com) |
16:20.04 | `Sauron | that way they can't screw me out of too much money |
16:21.10 | *** join/#asterisk beta3 (~dan@dan2.active.supporter.pdpc) |
16:21.27 | eKo1 | No such switch 'IAX' <--- I always get this when doing a switch => IAX/... |
16:21.33 | beta3 | how do I do sutter tone based on channels |
16:21.40 | beta3 | every time I set it seems to affect globally |
16:22.00 | *** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com) |
16:22.01 | Rival | didnt ebay buy out paypal? |
16:22.04 | eKo1 | It's as if IAX doesn't exist... |
16:22.06 | `Sauron | yeah |
16:22.22 | Trionnis | heh |
16:22.29 | Trionnis | those guys are rich mofo's too |
16:23.28 | beta3 | ??? |
16:23.38 | Trionnis | the guys that started paypal |
16:23.51 | Trionnis | started it here at UIUC Beckman |
16:24.03 | Trionnis | now they're rolling around town in H2's |
16:24.10 | Trionnis | lol |
16:24.20 | Rival | ah they earned it |
16:24.24 | Rival | paypal was a good idea |
16:24.33 | Trionnis | yes, it was |
16:24.47 | jontow | potentially bad implementation though :) |
16:24.59 | Trionnis | can't say I'm real keen on thier business methods |
16:25.00 | nestAr | i replied to polycom that I believed they were lying to me.. |
16:25.10 | nestAr | i feel better now |
16:25.29 | Rival | true but same can be said for any uber succesfull business |
16:25.30 | wolfson | other than the fact that regulators in multiple states are after them for being an unauthoirzed bank |
16:26.07 | ManxPower | I only run reasonably small amount of money thru Paypal. |
16:26.14 | Zeeek | which is why my idea of paying thru our trusted providers si a good one :) |
16:26.18 | beta3 | PayPal is fraudulent |
16:26.32 | nestAr | i love paypal |
16:26.34 | Zeeek | trusted being the key word here |
16:26.35 | nestAr | i use it all the time |
16:26.42 | Zeeek | PooPal |
16:26.49 | Nugget | I use paypal, but I sure don't trust it. |
16:26.59 | beta3 | ManxPower: do you have any idea on how to get stutter tone for differing voicemail boxes on different channels |
16:26.59 | Nugget | and it's a very expensive service |
16:27.11 | Zeeek | the results are in, we all use it but onlt one person trusts it |
16:27.13 | BuckRogers | paypal is used for the tranfer of money that are directed at websites that promote islamic extremism |
16:27.17 | jontow | i don't use it :) |
16:27.18 | jontow | i never have |
16:27.32 | BuckRogers | donations to them are accepted by paypal |
16:27.55 | nestAr | i have no reason to not trust it |
16:27.55 | ManxPower | beta3, mailbox=mailbosnumber@thecontextinvoicemailconf |
16:27.55 | nestAr | never had a problem with their service |
16:27.55 | Zeeek | Buck so are Citibanks and every other important financial institution |
16:27.58 | beta3 | ManxPower: thats not the issue, its the problem of getting different mailboxes for different zaptel channels |
16:27.59 | ManxPower | you can have multiple mailbosnumber@thecontextinvoicemailconf seperated by a , |
16:28.00 | jontow | I had an account once.. only because they said they'd give me $10 for free if i did.. and all i had to do to get it was request that they mail me a check.. i requested said check, it never came, i never actually used the account due to that :) |
16:28.00 | BuckRogers | i have never had a problem with their service either |
16:28.17 | BuckRogers | yes i know zeeek |
16:28.18 | jontow | i figured if they wouldn't come through on such a simple offer, then handing them my money wasn't going to be a beneficial thing for me. |
16:28.19 | Nugget | the plural of "anecdote" is not "data". |
16:28.23 | jontow | that was .. years ago |
16:28.28 | BuckRogers | we have a major problem |
16:28.44 | BuckRogers | im not trying to single them out |
16:28.51 | ManxPower | beta3, I don't really understand your issue. |
16:28.54 | Nugget | there are numerous stories of paypal freezing accounts due to misunderstanding and never returning the money. |
16:29.14 | `Sauron | Nugget: Which is why I have never verified my account |
16:29.18 | Nugget | paypal is perfectly reliable until they decide you're doing something bad, and once that happens you're screwed. |
16:29.19 | BuckRogers | yeah and where does that money go when frozen |
16:29.29 | beta3 | ManxPower: everytime I set mailbox= in zapata.conf, it sets it globally, I want to set it for each fxs |
16:29.30 | Nugget | even if you're not doing something bad |
16:29.52 | cbachman | Nugget... that's perfectly true. My understanding is that there is one (1!) individual who decides what is bad |
16:29.59 | BuckRogers | i got realtime astrisk woking very good right now |
16:30.02 | ManxPower | beta3, you need to define each channel seperatly, then define the mailbox= right above the channel => for that channel |
16:30.12 | beta3 | ManxPower: I've alreayd done that |
16:30.13 | BuckRogers | much better then my english |
16:30.23 | ManxPower | beta3, *shrug* It works for everyone else. |
16:30.32 | ManxPower | beta3, put your zapata.conf on pastebin.ca |
16:31.09 | dfunnell | Fellow geeks - general question this time. (Sorry about newbie question, but CAPI documentation is wafer thin). |
16:31.11 | dfunnell | I've got 4 x BRI in my * machine (and am using CAPI) and I want * to try dialling each available CAPI channel when dialling out (i.e. move from one to the next if the first one is busy). Anyone know the correct syntax for doing so? Can you group CAPI channels like you can group Zaps? |
16:31.53 | ManxPower | dfunnell, I don't know for sure, but using group= like in zaptel is the best way, and I would ASSUME chan_capi would support group= |
16:32.11 | BuckRogers | has any one experimented with php and mysql for web configuation of user accounts |
16:32.21 | ManxPower | dfunnell, You should ask on the mailing list if you don't find the answer here |
16:32.56 | BuckRogers | such as turning on and off features like anonomus call rejection |
16:33.00 | BuckRogers | from a website |
16:33.17 | Trionnis | how else would you do web configuration? |
16:33.21 | Trionnis | ;) |
16:33.31 | JerJer[mobile] | vi index.html |
16:33.36 | ManxPower | BuckRogers, it seems that most people that have done that consider it an advantage in competing and so don't release the scripts. |
16:33.52 | BuckRogers | Trionnis: i have no idea other then what i stated do you? |
16:34.00 | dfunnell | ManxPower: Thanks. Do you mean CAPI mailing list? Also having problem where, if a trunk is in use, incoming calls ring and ring, but * doesn't seem to recognise and answer ringing line. |
16:34.08 | Trionnis | that was a smartass crack about your last line of text |
16:34.15 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net) |
16:34.24 | Trionnis | I really wouldn't know... I'm just here for lame comic relief |
16:34.27 | BuckRogers | right on bro |
16:34.30 | ManxPower | dfunnell, Unfortunatly I've never used CAPI, so I can only give you very general advice about it. |
16:35.06 | Trionnis | although I agree with ManxPower... it's likely that you won't find one "prefabbed" really |
16:35.30 | Trionnis | I'm willing to help ya write one if you GPL it ;) |
16:35.38 | BuckRogers | we are working on it now |
16:35.55 | *** join/#asterisk algorithmn (~na@ool-18bce89c.dyn.optonline.net) |
16:35.56 | beta3 | ManxPower: its too big |
16:36.08 | BuckRogers | i think we will keep it private also |
16:36.15 | Trionnis | of course you will |
16:36.16 | BuckRogers | and sell it to the highest bidder |
16:36.17 | Nugget | ewwww. GPL is ucky. :) |
16:36.18 | BuckRogers | j/k |
16:36.25 | BuckRogers | no one will pay for that |
16:36.28 | ManxPower | The way I PLAN on doing this when the time comes is use PHP to provide the interface, MySQL as the backend (since something more powerful is not needed) and then AGI in the dialplan to query the settings and set the channel variables, then let the dialplan do what needs to be done via the channel variables. Who knows what I'll ACTUALLY do when the time comes. |
16:36.32 | Trionnis | ;) |
16:36.44 | JerJer[mobile] | agi is not desired or needed |
16:37.05 | JerJer[mobile] | but you will figure this out once you write one |
16:37.09 | Nugget | That sounds like a sane approach, ManxPower. |
16:37.30 | Nugget | (well, except for the mysql part -- naturally :) |
16:37.36 | Trionnis | lol |
16:37.40 | ManxPower | Nugget, As JerJer points out AGI could be an issue, but since I'll never have more than 100 users, I dont think it will be an issue. |
16:37.40 | BuckRogers | jer jer what do u use instead of agi |
16:37.41 | BuckRogers | ? |
16:38.13 | ManxPower | Nugget, It's a perfect app for MySQL. i.e. lots of reads, not many writes. |
16:38.18 | algorithmn | whats the deal my buck of rogers? |
16:38.22 | Nugget | I know, i was just trolling. |
16:38.38 | BuckRogers | cold chilling, holding it down my man |
16:38.40 | *** part/#asterisk Inferna (~sasha@194.158.51.171) |
16:38.51 | Nugget | although I'd always suggest postgresql over mysql I don't honestly mean to suggest that mysql is a /bad/ choice. it's just not the best choice. |
16:38.55 | algorithmn | word... thanks for the wiskey |
16:39.05 | BuckRogers | no problem |
16:39.16 | ManxPower | Nugget, People forget "the right tool for the job" maxim. |
16:39.16 | algorithmn | it really put some punch in the mornin coffee |
16:39.18 | eKo1 | How well do you think AGI scales when you have over 100 users? |
16:39.24 | Nugget | mysql has a host of problems that come with it -- but anyone willing to deal with those shortcomings is clearly free to choose mysql. |
16:39.31 | tzanger | I'll come right out and say it's a bad choice |
16:39.36 | *** join/#asterisk zapa (zapa@200.77.110.182) |
16:39.36 | ManxPower | eKo1, AGI has quite a bit of overhead to start up. |
16:39.47 | tzanger | if you only need somewhere to throw your data, use sqlite or db2/3/4 or even a flat file... |
16:39.55 | BuckRogers | nugget what type of problems that woudl affect asterisk |
16:39.55 | BuckRogers | ? |
16:40.02 | tzanger | or if you use php, by all means use mysql :-) |
16:40.03 | eKo1 | ManxPower: So building a module would be the 'faster' choice? |
16:40.09 | ManxPower | tzafrir, sqlite is also something I've considered. |
16:40.13 | tzanger | I really like sqlite |
16:40.19 | tzanger | it's small, portable and works well |
16:40.20 | ManxPower | eKo1, by orders of magnitude |
16:40.27 | tzanger | I just wish it had some network connectivity :-) |
16:40.51 | Nugget | BuckRogers: using mysql means never being certain that the data you're reading is what you wrote. |
16:40.58 | Nugget | it's fast, though |
16:41.05 | tzanger | Nugget: correct, but it's not THAT fast |
16:41.11 | eKo1 | But making AGIs are orders of magnitude easier than modules. |
16:41.14 | tzanger | compare it to pg for realworld use, they're neck and neck |
16:41.19 | tzafrir | ManxPower, it is even something I have considered. My latest * packages have sqlite support |
16:41.24 | tzanger | it's like saying your P4/3.02 is faster than your P4/3.00 |
16:41.31 | tzanger | sure it's faster but it's not noticeable |
16:41.39 | BuckRogers | Nugget i dont understand? |
16:41.46 | tzanger | Postgres < 7.x did suck for speed |
16:41.52 | tzanger | but 7.4.3 and the 8.x betas are insane |
16:41.56 | BuckRogers | what i read is not what i wrote? Mysql injection? |
16:42.08 | BuckRogers | thats a hacker tool right |
16:42.15 | Nugget | mysql is designed to silently change the data you insert if it doesn't fit. |
16:42.16 | tzanger | BuckRogers: MySQL will take it upon itself to make your data fit, instead of tleling you the data won't fit |
16:42.37 | tzanger | it's one of *the* most *assinine* "features" for a DB I've ever ever run across |
16:42.57 | BuckRogers | so make sure your inserts are of a certian size? |
16:43.04 | algorithmn | buckrogers: i woulnd;t worry about injection. i'll help u put in some detection into the php scripts w/admin e-mail notification |
16:43.04 | tzanger | BuckRogers: no it's worse than that |
16:43.10 | tzanger | BuckRogers: create a table with an integer column |
16:43.17 | tzanger | now insert "mary had a little lamb" into that column |
16:43.22 | tzanger | mysql will take it and not even blink. |
16:43.41 | algorithmn | does it drop it or convert ascii->int |
16:43.41 | BuckRogers | ok thats good though right |
16:43.45 | tzanger | ??! |
16:43.50 | tzanger | BuckRogers: step AWAY from the keyboard |
16:43.50 | Nugget | no, that's terrible. |
16:43.54 | *** join/#asterisk Damin_Mobile (~pocketirc@72.sub-70-214-30.myvzw.com) |
16:44.05 | BuckRogers | ahh letters where numbers should be |
16:44.07 | ManxPower | tzanger, Some people would say that you need to validate your data before sending it to your database. |
16:44.07 | BuckRogers | got cha |
16:44.19 | *** join/#asterisk skrusty (muad@217.79.111.73) |
16:44.24 | skrusty | afternoon |
16:44.27 | ManxPower | I don't disagree with that. |
16:44.29 | BuckRogers | good point ManxPower |
16:44.30 | Nugget | ManxPower: most people would agree that the database is the sanest place to do that validation. |
16:44.33 | tzanger | ManxPower: fuck that -- the DB is there to store the data I give it. If it can't, it should throw back an error, not silently mangle it |
16:44.38 | algorithmn | programming convention can help alliviate that, just more time wated i guess |
16:44.39 | Nugget | but mysql makes such validation impossible. |
16:44.39 | tzanger | ManxPower: besides |
16:44.52 | tzanger | ManxPower: if the fucking db has CONSTRAINTS and it doesn't listen to them, why does it have them in the first place?! |
16:44.56 | ManxPower | But honestly, the choice of database can be changed pretty easily, especially if you do it before you start coding. |
16:45.20 | tzanger | ManxPower: if you code correctly (SQL92/95 compliant queries) it sure helps |
16:45.30 | ManxPower | tzanger, The Sweeds have a twisted sense of humor? |
16:45.33 | sivana | what's a good ATA that has dual ethernet (computer, connection)? |
16:45.35 | eKo1 | and/or use odbc |
16:45.52 | BuckRogers | sivana : grandstream.com |
16:46.02 | BuckRogers | nice equipment |
16:46.06 | tzanger | eKo1: ODBC is *A* solution but it's not necessarily a good one :-) |
16:46.15 | tzanger | BuckRogers: CHEAP equipment, not necessarily good :-) |
16:46.16 | *** join/#asterisk Makenshi (~makenshi@2001:630:1c0:2001:280:c8ff:fee2:921f) |
16:46.25 | Nugget | imagine if asterisk just dialed a random number if you tried to call a number which was not specified in the dialplan. would the solution be to just "always make sure you dial the right number?" |
16:46.40 | sivana | just until the MixBox is done mfg :) |
16:46.40 | BuckRogers | really i had better luck with them then sipura |
16:46.55 | tzanger | sivana: shhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhaddap... :-0 |
16:46.57 | tzanger | er :-) |
16:47.04 | BuckRogers | sipura gsm does not work correctly i emailed support over 3 weeks ago with no respones |
16:47.21 | Nugget | would the solution involve modifying every possible soft and hard phone out there to validate that the dialed number was one that the dialplan covered? |
16:47.21 | sivana | <PROTECTED> |
16:47.30 | tzanger | Nugget: asterisk isn't actually a good comparison... I can say "pollywaddledoodle=yes" in iax.conf and * will happily ignore it |
16:47.39 | sivana | Anyone used the SPA-2100? |
16:47.42 | Trionnis | anyone know if there's some docs floating around about preconfiguring xlite before installed? |
16:47.53 | BuckRogers | yes we have one |
16:47.59 | Trionnis | I'd like to set up a buddy of mine on my * server, but I'm *not* walking him through setting it up |
16:48.00 | Nugget | true enough. I was more using the dialplan specifically as the comparison. |
16:48.05 | sivana | you like? |
16:48.05 | BuckRogers | i would rater have the 2000 |
16:48.06 | tzanger | ~google preconfigure xlite asterisk |
16:48.14 | tzanger | ... odd |
16:48.17 | sivana | BuckRogers: oh... why? |
16:48.20 | Trionnis | already did that :) |
16:48.20 | *** join/#asterisk [Outcast] (~knoppix@h0006259a2649.ne.client2.attbi.com) |
16:48.24 | tzanger | Nugget: :-) |
16:48.41 | BuckRogers | just poor firm ware |
16:49.08 | BuckRogers | i think they try to promis more then they give |
16:49.17 | *** part/#asterisk Smythe (~Smythe@spock.cbcag.edu) |
16:51.00 | mesi | I just cannot configure this stupid sipphone thingy. Everything's just like with sip to fwd, but still receiving calls from sipphone doesn't work. |
16:51.54 | shido6 | mesi |
16:51.57 | shido6 | dont freak out |
16:51.59 | mesi | shido: HI! |
16:52.04 | shido6 | whats up? |
16:52.21 | mesi | shido: I try to receive calls form siphphone, but it wouldn't work. |
16:52.37 | shido6 | ok |
16:52.40 | shido6 | how are you trying? |
16:52.42 | shido6 | show me what u got |
16:52.47 | mesi | shido: though everything is fine, I am registered, the web page sipphone.com knows my ip address, I can make calls... |
16:52.47 | shido6 | at pastebin.ca (sip.conf) |
16:52.59 | shido6 | making calls is easy |
16:53.06 | shido6 | receiving calls ...esp if ur nat'd can be a problem |
16:53.09 | shido6 | but you can fix it |
16:53.22 | mesi | I don't do nat with fwd and still it works fine! |
16:53.32 | shido6 | ok show me what u got |
16:53.34 | Trionnis | ~google preconfigure xlite |
16:54.02 | Trionnis | erf |
16:54.06 | mesi | shido: never used pastebin, but I'll try... |
16:54.42 | BuckRogers | you might as well program in pastel |
16:54.45 | *** join/#asterisk crash3m (crash3m@crash3m.user) |
16:54.55 | MikeJ[Jayden] | ~pastebin |
16:54.56 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
16:54.58 | ManxPower | tzanger, Once I get the coppice bounty out of the way I may issue a bounty to generate an error/warning for invalid .conf entries. |
16:55.12 | shido6 | hey ManxPower |
16:55.19 | ManxPower | hello shaZwaz |
16:55.19 | tzanger | what's coppice's bounty |
16:55.23 | ManxPower | ..er..hello shido6 |
16:55.26 | shido6 | someone asked me about AS5300s and ppp dialup with the Quad span cards, you ever set that up? |
16:55.40 | *** part/#asterisk crash3m (crash3m@crash3m.user) |
16:55.41 | ManxPower | tzanger, The stuck channels problem that he fixed, but he's STILL not collected it. |
16:55.42 | shido6 | i've set up hdlc and data/voice T's |
16:55.44 | shido6 | but never ppp |
16:55.48 | shaZwaz | howdy ManxPower |
16:55.55 | Trionnis | ~google preconfigure sip phone |
16:56.04 | Trionnis | erf |
16:56.06 | ManxPower | shido6, no, since that requires an ISDN DATA connection and not a modem connection. |
16:56.24 | mesi | shido: I have pasted my sip.conf there. |
16:56.26 | tzanger | ManxPower: oh yeah uh... I fixed that... yeah... |
16:56.35 | Trionnis | ok, enough google spamming.. guess I'll open a browser |
16:56.37 | Trionnis | :) |
16:56.37 | mesi | shidO: into "mesi: sip.conf" |
16:56.45 | ManxPower | tzanger, Hmm? |
16:56.59 | ManxPower | tzanger, I thought you helped with the E&M/Wink problem? |
16:57.17 | mesi | shido: My extensions.conf is quite huge! |
16:57.19 | ManxPower | the stuck channels problem was a different issue 8-) |
16:57.58 | *** join/#asterisk marshall (~test@S0106000f66563988.wp.shawcable.net) |
16:58.28 | Trionnis | eek! |
16:58.38 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
16:58.38 | Trionnis | no suppression? |
16:59.23 | shido6 | just show me the extension u have given the sip provider access to |
16:59.24 | *** part/#asterisk oej (~oej@63.83.135.35) |
16:59.45 | Darwin35 | they had surpression but the damage is done |
16:59.52 | Darwin35 | it was in a back corner |
16:59.55 | Trionnis | :( |
17:00.01 | tzanger | ManxPower: I'm joking, I'm just trying to collect his bounty |
17:00.28 | Trionnis | sounds like it wasn't properly installed |
17:00.49 | ManxPower | tzanger, LOL! |
17:00.49 | Trionnis | I'd think that there should be 100% coverage of machine areas |
17:01.04 | Trionnis | sorry to hear that man... that really sucks |
17:01.16 | Darwin35 | its a old building refurbished |
17:01.29 | Darwin35 | built in 18 96 |
17:01.37 | Trionnis | eek |
17:01.38 | shido6 | mesi, whats the pastebin.ca url they gave you? |
17:01.38 | Darwin35 | new wiring |
17:01.44 | Darwin35 | new lights |
17:01.50 | Trionnis | doesn't sound like the best place for a DC |
17:01.58 | Darwin35 | but the server room was never properly setup |
17:01.59 | *** join/#asterisk Blackvel (~blackvel@dsl-213-023-034-235.arcor-ip.net) |
17:02.20 | Darwin35 | well now I get to have my way with the offic |
17:02.27 | Darwin35 | eand redo the server room |
17:02.31 | Trionnis | any clues as to what started it? |
17:02.47 | Darwin35 | not yet they wont let us in the area yet |
17:02.51 | inspired | what the hell? suddenly one of my peers have stopped appearing in "iax2 show peers", however it is still in iax.conf. nothing was changed in any config file. does anyone know what's wrong? the system is in production |
17:03.06 | Darwin35 | I have a feeling a non grounded power strip |
17:03.10 | tzanger | inspired: interesting |
17:03.14 | mesi | shido: http://pastebin.ca/6965 is my extensions.conf |
17:03.20 | tzanger | pastebin their entry and mangle hte password/username |
17:03.28 | mesi | <PROTECTED> |
17:03.29 | Trionnis | non.... grounded? |
17:03.34 | Trionnis | in a server room?? |
17:03.35 | inspired | ok tzanger |
17:03.56 | Trionnis | wow |
17:03.58 | Darwin35 | there were to stripps missing the grounding pins |
17:04.00 | Trionnis | that's..... |
17:04.10 | Trionnis | incredibly farking stupid |
17:04.11 | Blackvel | so who did fix that broadvoice problem for incoming voicemail instead of asterisk call yet? |
17:04.12 | Darwin35 | I had put dont use on them but have a feeling they got used |
17:04.13 | Trionnis | =) |
17:04.16 | *** join/#asterisk Tarox (someone@pD9E79ED7.dip.t-dialin.net) |
17:04.24 | inspired | http://pastebin.ca/6966 |
17:04.30 | inspired | check it tzanger |
17:04.36 | Trionnis | shoulda pitched 'em |
17:04.50 | Darwin35 | 1 is the broadvoice issue |
17:05.00 | tzanger | inspired: and sip show peers doesn't show briiz? |
17:05.03 | Darwin35 | 2 is the mpg123 issues with * |
17:05.06 | Trionnis | there's a broadvoice issue? |
17:05.17 | Sedorox | whats wrong with mpg123? |
17:05.20 | Blackvel | darkskiez: i am tspeaking to someone |
17:05.24 | Trionnis | I had some screwy stuff last night, but I've not noticed anything so far today |
17:05.24 | ManxPower | Darwin35, A fire usually gets management to listen to MIS 8-) |
17:05.25 | Darwin35 | mpg123 sucks |
17:05.28 | inspired | no, it's not a sip peer |
17:05.28 | Sedorox | lol |
17:05.31 | Blackvel | and he says, your 1.0.5 fix does not fix that problems :) |
17:05.38 | Trionnis | hahaah, very true ManxPower |
17:05.58 | *** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com) |
17:06.02 | Darwin35 | I am looking into the 1.0.6 issue with broadvoice and * |
17:06.09 | Blackvel | Darwin35 i mean |
17:06.13 | Trionnis | ahh, I haven't upgraded yet |
17:06.15 | Darwin35 | give me time to figure it out |
17:06.17 | Blackvel | but he now got 1.0.5 :) |
17:06.19 | Trionnis | seems like it's good I didn't |
17:06.21 | Trionnis | ;) |
17:06.37 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
17:06.39 | inspired | tzanger: this is scary. users are trying to call out but our termination partner is gone from iax2 show peers |
17:06.44 | Blackvel | seems to work from time to time, but not always, so broadvoice sends callers instead to asterisk to BV voicemail :) |
17:06.48 | tzanger | uh |
17:06.53 | Darwin35 | I have to put everything on the back burner and go to a meeting about this fire |
17:06.55 | tzanger | inspired: you showed me a sip peer did you not? |
17:06.58 | Darwin35 | I will return |
17:07.03 | Trionnis | ok |
17:07.10 | Trionnis | good luck Darwin |
17:07.17 | Blackvel | i am not sure if I could understand your issues very well darwin and if was the problem you where working on |
17:07.18 | Trionnis | hopefully they'll listen |
17:07.19 | Blackvel | fire? |
17:07.19 | ManxPower | Darwin35, TURN OFF THE BURNER FIRST! 8-) |
17:07.20 | Blackvel | uff |
17:07.25 | Darwin35 | hahaha |
17:07.27 | Trionnis | oooh |
17:07.30 | Trionnis | pwn! |
17:07.31 | Trionnis | lol |
17:07.47 | inspired | tzanger: no, it's an iax2 peer |
17:07.57 | tzanger | inspired: ahh |
17:08.02 | inspired | and it's been working fine for a long time |
17:08.07 | tzanger | odd |
17:08.08 | Blackvel | what? |
17:08.08 | inspired | but now it disappeared |
17:08.20 | tzanger | inspired: there's not a space in front of it or anything |
17:08.24 | Trionnis | uh, stupid question, but have you made sure the service didn't go down? |
17:08.24 | *** join/#asterisk Dibbler (~Dibbler@snaddy.plus.com) |
17:08.38 | Trionnis | sometimes it's the simple things ;) |
17:09.08 | inspired | tzanger: in front of it? |
17:09.08 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
17:09.08 | Darwin35 | Manx take over for me for awhile |
17:09.08 | Darwin35 | back in 30 min |
17:09.08 | inspired | no, everything is aligned to the left |
17:09.09 | tzanger | i..e "[briiz]" not " [briiz]" |
17:09.28 | tzanger | inspired: just for shits and giggles, add a "qualify=500" to it and reload |
17:09.32 | *** join/#asterisk sob0l (~peter@uo166.internetdsl.tpnet.pl) |
17:09.52 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
17:10.10 | inspired | ok |
17:11.50 | inspired | and another thing. asterisk has started having SHITTY responsiveness when it comes to issuing commands on the CLI. a simple "reload" takes 15-30 seconds before anything happens |
17:11.57 | inspired | a stop gracefully has the same problem |
17:12.26 | shaZwaz | just intalled speex ..looks fine thou |
17:13.19 | *** join/#asterisk alerque (~alerque@onyx.ouraynet.com) |
17:13.20 | *** part/#asterisk alerque (~alerque@onyx.ouraynet.com) |
17:13.21 | jontow | inspired; look at 'top' .. is mpg123 going nuts? |
17:13.26 | *** join/#asterisk alerque (~alerque@onyx.ouraynet.com) |
17:13.29 | inspired | mpg123 is not even installed |
17:13.34 | inspired | load 0.02 |
17:14.16 | *** join/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk) |
17:14.19 | blitzrage | ok... fuck this guy |
17:14.21 | blitzrage | http://techdatapros.com/asterisk/ |
17:14.46 | Zeeek | hey Blitz |
17:14.48 | inspired | just restarted asterisk. takes fucking one minute before it even loads any modules |
17:15.12 | blitzrage | "Digium X100P FXO card which can be purchased off eBay for $6.95" |
17:15.23 | blitzrage | obviously a clone, and we'll probably have a million newbs asking how to get it to work |
17:15.31 | JerJer[mobile] | Caveat Emptor |
17:15.46 | blitzrage | anyways... thats my rant, I'm going back to the productive IRC channels |
17:15.54 | blitzrage | Zeeek: btw - yo! |
17:15.54 | *** part/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk) |
17:15.56 | Nugget | heh |
17:15.56 | Trionnis | would help if they weren't advertised that way |
17:16.43 | Trionnis | could just be a case of simple ignorance :) |
17:20.08 | inspired | tzanger: qualify=500 didn't help at all |
17:22.26 | inspired | going to test it |
17:22.37 | inspired | uhm, just received a fresh iax2 phone. |
17:22.42 | inspired | will make a review |
17:22.51 | inspired | at least it looks good :D |
17:23.11 | Sedorox | is it in production? |
17:23.35 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
17:23.35 | inspired | yep, it's an ATCOM something |
17:23.39 | inspired | got it from iaxtalk.com |
17:23.57 | inspired | the lcd is even flippable! |
17:23.58 | inspired | :D |
17:24.03 | Sedorox | hmmm |
17:24.15 | Zeeek | inspired how you like phone ? |
17:24.42 | inspired | haven't tried it yet. |
17:24.50 | Sedorox | http://www.iaxtalk.com/product_info.php?cPath=1&products_id=36&osCsid=d9391e67327e4075a742ca43ea1f92db |
17:24.51 | Sedorox | ?? |
17:24.54 | shmaltz | <PROTECTED> |
17:25.03 | inspired | ATCOM |
17:25.15 | inspired | will volume import these to EU if it works good |
17:25.33 | inspired | it certainly doesn't look like a grandstream |
17:25.35 | Blackvel | who got iaxtel? |
17:25.36 | inspired | and it's even cheaper! |
17:25.45 | Blackvel | and is open minded for a call test? |
17:26.01 | inspired | Sedorox: yep, except I have the two port one |
17:26.10 | Sedorox | inspired: was gonna say that.. cheaper then a grandstream.. and better looking |
17:26.11 | Sedorox | ah ok |
17:26.12 | Zeeek | inspired $30-$50 shipping |
17:26.19 | Zeeek | add ^^^^^ |
17:26.21 | *** part/#asterisk alerque (~alerque@onyx.ouraynet.com) |
17:26.21 | Zeeek | from China |
17:26.28 | *** join/#asterisk jalsot_ (~tamas@abacus.eworldcom.hu) |
17:26.33 | inspired | still cheaper than buying a grandstream in EU |
17:26.36 | *** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net) |
17:26.47 | Zeeek | buying anything here is way more expensive |
17:26.47 | inspired | plus if I import many of them, I will have cheaper shipping |
17:26.52 | Zeeek | yes |
17:27.00 | Zeeek | and you could get a lot of us to group an order |
17:27.10 | harryvv | zeek you at von now |
17:27.16 | inspired | yep |
17:27.17 | Zeeek | except I just ordered one from the States |
17:27.22 | *** join/#asterisk X-Gen (~x-gen@rrba-146-120-223.telkomadsl.co.za) |
17:27.25 | BuckRogers | what is the codec support for the iaxtel? |
17:27.27 | inspired | the same model? |
17:27.32 | Zeeek | no I'm at Von's doing the grocery shopping |
17:28.09 | harryvv | zeeek :) how many voip products do you have in your grocery cart now? |
17:28.09 | Sedorox | I wanna know how those ATA's from that company work... |
17:28.42 | BuckRogers | AT-320EE VoIP phone, what codecs does it support? |
17:28.57 | inspired | BuckRogers: Audio codec G.711,G.723,G.729,GSM |
17:29.10 | inspired | Sedorox: I also bought the ATA. will test that too |
17:29.12 | BuckRogers | Inspired thats soilid |
17:29.24 | BuckRogers | I luv my gsm |
17:29.31 | inspired | yep, gsm rocks |
17:29.39 | BuckRogers | nice and tight and compact |
17:29.46 | BuckRogers | sounds great |
17:29.52 | BuckRogers | little overhead |
17:29.52 | Sedorox | inspired: please let me know about that!!! |
17:29.58 | inspired | fucking shit. I can't add new users to iax.conf |
17:30.05 | Sedorox | ? |
17:30.07 | inspired | what the fuck happened?! |
17:30.19 | inspired | one peer disappeared and I can't even add new ones |
17:30.36 | inspired | exit |
17:30.41 | inspired | oops. hehe |
17:31.01 | harryvv | just a little oversight |
17:33.05 | jontow | woohoo.. getting my pair of T100P cards back today :P |
17:33.11 | *** join/#asterisk j0 (dan@S010600095b00a5a9.vc.shawcable.net) |
17:36.36 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-221-15.dsl.scarlet.be) |
17:37.08 | *** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
17:38.23 | inspired | now that kicks ASS. the phone has a 100 mbit switch |
17:38.26 | inspired | and is 69$ |
17:38.32 | inspired | will upgrade to iax firmware now |
17:38.43 | Sedorox | lol |
17:38.51 | Sedorox | oh.. doesn't come with the iax firmware, huh? |
17:39.13 | inspired | no, sadly :( |
17:39.21 | Sedorox | but they include it on a CD or something? |
17:39.57 | inspired | on the web |
17:40.43 | Sedorox | hmmm |
17:40.43 | Sedorox | ok |
17:41.21 | tzanger | inspired: odd |
17:41.29 | tzanger | asterisk is otherwise working fine?? |
17:41.49 | Sedorox | bbiab |
17:42.02 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
17:43.25 | *** join/#asterisk Damin_Mobile (~pocketirc@90.sub-70-214-0.myvzw.com) |
17:44.45 | dontmsgme | nn |
17:44.52 | opus___ | hello |
17:44.57 | opus___ | is there a way to encrypt sip calls |
17:44.57 | dontmsgme | Hi |
17:45.18 | opus___ | hello dont |
17:46.02 | *** join/#asterisk Conductor (~thomas@62.8.240.132) |
17:46.21 | Conductor | hi all. i have some problems with the webinterface of meetme2... |
17:46.41 | Darwin35 | ok I have a neww as*hole and so does the whole office |
17:46.44 | Conductor | everything is ok, but when i click on Listen or Kick nothing seems to happen |
17:46.54 | *** join/#asterisk beta3 (~dan@dan2.active.supporter.pdpc) |
17:47.06 | Trionnis | damn Darwin35... :( |
17:47.07 | beta3 | I'm having major echo issues on calls coming from the fxo |
17:47.08 | opus___ | where is the web interface code for meetme2 , curious |
17:47.11 | beta3 | any suggestions |
17:47.30 | Trionnis | isn't it great when people choose to piss and moan instead of addressing the issue? |
17:47.43 | Trionnis | (yes, that's sarcasm) |
17:47.46 | Conductor | opus___, http://www.areski.net/asterisk-meetme/about.php |
17:47.49 | opus___ | beta3 have you contacted the manufacture, is the cable to long? |
17:48.01 | beta3 | opus___: this is a digium card |
17:48.07 | opus___ | conductor - thanks |
17:48.07 | Darwin35 | yes |
17:48.17 | Darwin35 | so now they gave me my own project |
17:48.21 | beta3 | opus___: I hooked it up at the dmark of the house, so its shorter than the phone system originally for the home |
17:48.28 | Trionnis | hopefully things will come together properly now? |
17:48.35 | Darwin35 | my 17 boxes are to be moved to my office |
17:48.40 | Damin_Mobile | Darwin: What happened? |
17:48.50 | Darwin35 | we had a small fire |
17:48.58 | Trionnis | ah |
17:48.58 | Darwin35 | lost 11 servers |
17:49.02 | Trionnis | so you're getting space heaters |
17:49.04 | Conductor | hasn't anyone ever tried meetme2? |
17:49.06 | Trionnis | that's nice of them |
17:49.26 | Darwin35 | most where old old 21264 dec alpha servers |
17:49.39 | opus___ | condcutor - i'll check it out later tonight if you see me again i'll let you know if i had the same problem.. what was the problem you are having again? |
17:49.40 | Trionnis | wow... people still use those? |
17:49.43 | Trionnis | ;) |
17:49.53 | Damin_Mobile | Insurance shoulld help |
17:50.13 | beta3 | does anybody know how to get rid of echo on the fxo interfaces |
17:50.26 | Conductor | opus___, everything is ok, but when i click on Listen or Kick nothing seems to happen |
17:50.40 | Conductor | opus___, you need to apply a patch to the source code.. |
17:51.11 | Damin_Mobile | beta3: Enable echo cancellation. look at the wiki. |
17:51.39 | opus___ | hmmm |
17:51.43 | beta3 | Damin_Mobile: thats already done |
17:51.48 | Darwin35 | yeah insurance will cover |
17:51.55 | BuckRogers | Conductor that website looks nice very promising how does it work, good so/so bad? |
17:52.14 | harryvv | darwin was it the ps in one of the servers that caught fire |
17:52.16 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
17:52.27 | Damin_Mobile | Darwin; prolly get a decent payout. |
17:53.41 | Darwin35 | I wish I was getting money out of it |
17:53.45 | Trionnis | yeah, make sure you point out how hard it is to replace those "highly specialized" servers |
17:53.48 | Trionnis | ;) |
17:54.14 | harryvv | Do most standard PBX units have a onboard ups mounted in the same plastic enclosure? |
17:54.37 | epoch | harryvv: some of them have batteries yeah |
17:54.39 | Darwin35 | but now my project will get a boost |
17:54.42 | harryvv | Was looking for a nortel networks unit and wondered because of its large size |
17:55.03 | Trionnis | hi harry :) |
17:55.09 | Trionnis | how ya doin? |
17:55.12 | Damin_Mobile | New servers! |
17:55.24 | harryvv | epoch, whats the typical Ah rating of those batteries? |
17:55.34 | Trionnis | Damin_Mobile: that's like xmas to an IT geek ;) |
17:55.41 | harryvv | hi Trionnis |
17:55.41 | Darwin35 | I just got new servers for this project of setting up asterisk pbx systems |
17:55.46 | epoch | harryvv: hell if I know :) |
17:56.23 | Darwin35 | I need someone to go over my new extensions.conf file |
17:56.23 | Conductor | BuckRogers, doesnt work very well. must apply patch first, cant use postgres and even then it doesnt work correctly |
17:56.24 | harryvv | I guess the best way is to ask a ex pbx technican here. |
17:56.30 | *** join/#asterisk Tarox (someone@pD9E7BAF5.dip.t-dialin.net) |
17:56.33 | Conductor | cu |
17:56.36 | Conductor | \quit |
17:57.11 | Trionnis | harryvv: some do, some don't |
17:57.38 | harryvv | Trionnis, is that at the request of the customer ? |
17:57.43 | Trionnis | I ran an NEC ivs-2000 a while back that used a golf cart battery |
17:57.47 | Trionnis | separate |
17:57.51 | harryvv | k |
17:57.59 | Trionnis | but I've also seen panasonics with small ones built in |
17:58.00 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
17:58.12 | SexyKen | Hey guys -- does Asterisk support Extension Mobility? |
17:58.27 | Trionnis | I think some of the nortels, et al have space for one, but you have to get the "kit" to use it |
17:58.29 | Darwin35 | I need someone to go over my new extensions.conf file and tell me what is wrong |
17:58.31 | harryvv | I would feel more comfortable selling a unit with a ups onboard. |
17:58.40 | Trionnis | likely cheaper and easier to just get a second ups |
17:58.43 | Darwin35 | I think for the most part its fine |
17:58.44 | Trionnis | er |
17:58.48 | Trionnis | addon ups, that is |
17:59.03 | harryvv | darwin whats the problem |
17:59.16 | *** join/#asterisk cool4ever2 (~craeck@mail.innovate-it.ch) |
17:59.18 | Darwin35 | not all the functions seem to get called |
17:59.31 | harryvv | are thay all using sip |
17:59.32 | Darwin35 | so I just went over it |
17:59.37 | Darwin35 | yes |
17:59.42 | ManxPower | ~docs |
17:59.43 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
17:59.46 | harryvv | turn on sip debug |
17:59.55 | trimi` | any1 have used calling card application in asterist? if yes which is the best? any help ? |
17:59.56 | harryvv | if it comes to that |
17:59.57 | harryvv | :) |
18:00.01 | Darwin35 | i have its not showing enough info |
18:00.28 | harryvv | well then it sounds like its a misconfig or not enough configuration information. |
18:00.52 | trimi` | any1 have used calling card application in asterist? if yes which is the best? any help ? |
18:00.54 | Darwin35 | thats why I want some one to read the file and see if they see what I am missing |
18:01.01 | Darwin35 | yes |
18:01.04 | Darwin35 | BKW does |
18:01.15 | Trionnis | Darwin35: stick it on pastebin.ca |
18:01.28 | Darwin35 | ok |
18:01.29 | Trionnis | many eyes will catch things quicker ;) |
18:01.42 | SexyKen | Does anyone here know what Extension Mobility is? |
18:01.58 | shaZwaz | ok guys see u later |
18:02.13 | harryvv | never heard of it unless its just a extension thats directed to a cell phone number which I have done. |
18:02.32 | SexyKen | http://www.voip-info.org/tiki-index.php?page=PBX%20Extension%20Mobility |
18:02.36 | SexyKen | I just can't find any other info on it. |
18:02.55 | trimi` | any1 have used calling card application in asterist? if yes which is the best? any help ? |
18:03.56 | harryvv | sexy, just sign up for a voip service and make a extention that logs into that service and configure the line to dial your cell phone. |
18:04.19 | SexyKen | •harryvv• You're not at all close to what I need done. |
18:04.32 | SexyKen | harry - Apparently you use Asterisk on a personal level. |
18:04.33 | Darwin35 | http://pastebin.ca/6970 |
18:04.37 | harryvv | okay then your needs are different |
18:05.39 | Darwin35 | I know there ar emore functions out there |
18:05.42 | Zeeek | can you transfer an incoming call to meetme and hangup? |
18:05.47 | Darwin35 | but this is what I have |
18:05.56 | Darwin35 | yes |
18:06.02 | Trionnis | looking now Darwin35 |
18:06.44 | machinehd | With a 7960 can you set it up so it can see if other extensions are currently in use? Or are the 6 lines completely dedicated to that phone? |
18:07.09 | ManxPower | machinehd, I believe the Cisco SIP does not support shared call appearances. |
18:07.17 | ManxPower | Use Flash Operator Panel if you need that. |
18:07.52 | *** part/#asterisk Bentley (~rbc@S01060080c8135e6a.cg.shawcable.net) |
18:07.55 | *** join/#asterisk denon (denon@66.207.128.103) |
18:07.55 | *** mode/#asterisk [+o denon] by ChanServ |
18:08.20 | machinehd | ManxPower, thanks, I've been using FOP. Do you know of any phone that can do that? Secretaries seem to like the phones where they can see what lines are in use |
18:08.26 | trimi` | any1 have used calling card application in asterisk? if yes which is the best? any help ? |
18:08.30 | Zeeek | If I tranfer to meetme and hang up, both channels are hung up |
18:08.48 | ManxPower | machinehd, see the mailing list archives. |
18:08.51 | ManxPower | ~mailinglist |
18:08.52 | jbot | rumour has it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. |
18:09.01 | Trionnis | ya know trimi`.... if you keep that up, you're gonna get people honked off at you |
18:09.23 | ManxPower | machinehd, At least the SNOMs, maybe others support it. Asterisk's shared line appearance is not all that good and there are like 4 standards for doing that feature. |
18:09.48 | Darwin35 | SNOM can lick my crack those thieves |
18:10.02 | Darwin35 | sorry sore spot |
18:10.21 | ManxPower | Trionnis, Yes they will. Personally I don't usually answer questons from people using "words" like "any1", "r", "u", etc. |
18:10.44 | Trionnis | heh |
18:10.51 | modulus_ | manx, y not? |
18:11.03 | *** join/#asterisk neko2 (~neko@212.200.132.7) |
18:11.39 | Darwin35 | Tri the biggest part in that is the calling card function |
18:12.21 | Darwin35 | the only part any one gave me |
18:12.24 | Trionnis | ok, I just got to that part |
18:12.28 | Trionnis | lemme look |
18:12.35 | Trionnis | so far it's nicely done |
18:12.38 | Trionnis | commented well |
18:12.39 | Darwin35 | I wrote 98 % my self |
18:12.39 | Trionnis | ;) |
18:12.53 | Darwin35 | I am a stickler for that |
18:13.07 | Trionnis | good habit to have :) |
18:13.25 | ManxPower | modulus_, Because if they are too damn lazy to type real words, then I'm too damn lazy to help them. |
18:13.34 | modulus_ | manx, i c |
18:13.40 | Darwin35 | when this is done I am going to post it in the wiki pages as a very full robust extensions file |
18:13.45 | Darwin35 | maybe |
18:14.06 | Trionnis | hahah |
18:14.14 | Trionnis | nicely played modulus_ |
18:14.15 | Trionnis | ;) |
18:14.42 | *** join/#asterisk iceyp (~icepick@max.unix.co.nz) |
18:14.52 | iceyp | anyone from nufone.net here? |
18:15.06 | iceyp | shido6 ? |
18:15.38 | Darwin35 | rollover I have not yet tested I wrote that today |
18:16.04 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
18:17.10 | opus___ | is there a reason I see a million " chan_sip.c:901 __sip_ack: Stopping retransmission on '5f614b1a1193a64e2a41f50c0acfc4b6@192.168.2.168' of Request 102: Found |
18:17.15 | Trionnis | so far that calling care stuff looks decent |
18:17.22 | opus___ | lines when my 3+ SJphone implementation? |
18:17.33 | opus___ | is that a bad message |
18:17.34 | Trionnis | I wouldn't sign off on it without testing of course, but it *looks* right |
18:18.01 | ManxPower | opus___, stop running in debug mode. |
18:18.01 | Darwin35 | brb hold a min |
18:18.06 | *** join/#asterisk boch (~as24@200.59.172.98) |
18:18.29 | Trionnis | k |
18:18.34 | shido6 | aroo? |
18:18.37 | iceyp | opus___ I get them for missing UDP packets all the time |
18:18.37 | shido6 | whats up? |
18:18.51 | iceyp | or invalid DTMF tone |
18:18.53 | iceyp | shido6 umm |
18:19.10 | Darwin35 | back |
18:19.13 | iceyp | any reason I can dial this number +448449865089 from nufone? |
18:19.19 | Darwin35 | Tri PVt me if you have time |
18:19.44 | ManxPower | iceyp, 011448449865089 |
18:20.00 | ManxPower | And you have to make sure you are authorized by Nufone to make international calls |
18:20.02 | iceyp | yeah, i have a setting which makes me use 00 for intl |
18:20.14 | iceyp | allows me to make calls to other countrys |
18:20.25 | ManxPower | iceyp, no, in the USA you need 011 as the international prefix |
18:20.27 | iceyp | just having problems with a few ranges |
18:20.44 | iceyp | ManxPower i have a dialplan |
18:20.45 | iceyp | mkay |
18:20.51 | ManxPower | iceyp, Are you sure they are not Premium numbers? |
18:21.04 | iceyp | what u mean |
18:21.09 | iceyp | some countries can dial them |
18:21.15 | tzanger | "premium numbers" ?? |
18:21.23 | RoyK | premium arsehole? |
18:21.33 | ManxPower | tzanger, like 900 and 976 numbers in the USa. |
18:21.36 | tzanger | Premium Poontang |
18:21.41 | devel | is the sqlite cdr support "complete"? |
18:21.41 | RoyK | premium wannabee :) |
18:21.43 | tzanger | ManxPower: ahh okay |
18:21.46 | Trionnis | only $4.95 a minute! |
18:21.48 | Zeeek | the 900 PayManx number? |
18:21.49 | Trionnis | :) |
18:21.54 | iceyp | nah they not |
18:22.01 | Zeeek | hey that's it - I got it now |
18:22.03 | iceyp | 3-5c per min |
18:22.07 | iceyp | cant member it |
18:22.09 | Zeeek | support DID |
18:22.10 | *** join/#asterisk Bentley (~rbc@S01060080c8135e6a.cg.shawcable.net) |
18:22.11 | iceyp | 3-5p even |
18:22.17 | *** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk) |
18:22.19 | iceyp | its like the 0870 number |
18:22.25 | Trionnis | wow iceyp... that went *way* over your head, didn't it? |
18:22.30 | Trionnis | =) |
18:22.47 | Zeeek | shido6 in fact I had a problem today dialing an 800 number |
18:23.08 | Zeeek | it went through with other provierds but not nf |
18:23.56 | shido6 | u have a callerid set you're sending? |
18:23.57 | Zeeek | anyone use nufone that can try this 800? |
18:23.57 | tzanger | Zeeek: what was your outgoing callerid set to |
18:23.57 | shido6 | err |
18:24.01 | shido6 | u have to send a callerid |
18:24.08 | tzanger | Zeeek: give me the 800# |
18:24.09 | *** join/#asterisk Goshen (~Goshen@70-57-80-147.slkc.qwest.net) |
18:24.19 | Zeeek | I went thriough this last time with JerJer: callerid is 866nnnnn |
18:24.37 | tzanger | Zeeek: some 800#s will NOT accept calls from 800#s |
18:24.41 | Zeeek | Wells Fargo bank: (auto ans) 800 742 4932 |
18:25.03 | Darwin35 | ok time to setup a 976 sex line with asterisk |
18:25.07 | Zeeek | I seem to rememnber that it worked at one time |
18:25.08 | Goshen | thats odd...I am trying to call this number 18887713493, when it goes over FWD(IAX), command line shows it connecting, but I don't hear anything, when I change it to dial out over ENUM it goes over a sip server |
18:25.10 | tzanger | I found this out -- it's not a nufone problem, it's the far-end problem, if I set my outoging callerid to my 800# I couldn't get through, but dialing with a different (non-tf) 800# went fine |
18:25.13 | Darwin35 | setup a bunch of conf rooms |
18:25.16 | Goshen | and its the same problem, I don't hear any audio |
18:25.25 | Goshen | other toll free numbers have been working fine today |
18:25.42 | Zeeek | tzanger did you try it thru nufone? |
18:25.42 | SexyKen | Does asterisk support fucking extension mobility or extension roaming |
18:25.52 | tzanger | Goshen: FWD's a free service... make sure you're getting your money's worth |
18:25.54 | Zeeek | fucking extensions are banned |
18:25.58 | tzanger | Zeeek: hold |
18:25.59 | Blackvel | who uses iaxtel here? |
18:26.10 | SexyKen | I use iaxtel |
18:26.12 | stevekstevek | fucking extensions -- setting up a 900# service? |
18:26.20 | Zeeek | eggs actly |
18:26.22 | SexyKen | oh wait |
18:26.23 | SexyKen | no i dont |
18:26.25 | SexyKen | I use tel iax |
18:26.38 | Goshen | tzanger: the point is that it isn't working with two providers... |
18:26.43 | Goshen | guess I will try nufone now |
18:26.47 | SexyKen | NO! |
18:26.49 | SexyKen | Dont use nufone |
18:26.51 | SexyKen | They suck. |
18:27.03 | *** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net) |
18:27.08 | mishehu | sucking can be good. depends on in what context |
18:27.15 | Goshen | Sexy: keep your sucking and fu***ing to yourself ;) |
18:27.23 | SexyKen | Bend over bitch. |
18:27.27 | SexyKen | I'll show you what it's all about. |
18:27.31 | Zeeek | tzanger it just kept ringing for me |
18:27.33 | Trionnis | I'd imagine the "fucking extensions" might have a counterpart that would involve sucking |
18:27.36 | Darwin35 | ok back to topic |
18:27.45 | tzanger | Zeeek: works just fine |
18:27.46 | Darwin35 | take the sextalk to #moosepenis |
18:27.49 | jontow | neat.. i have 5 * PBX's connected via IAX2 to a 'central switch' running freebsd/* |
18:27.52 | Zeeek | actually it makes sense that a bank would be leery of 866 numbers |
18:27.56 | jontow | the 'switch' is routing calls between the PBXs :) |
18:27.57 | Zeeek | oh? |
18:28.01 | Zeeek | let me try again... |
18:28.05 | tzanger | IAX2/myusername@nufone-1/18007424932||g |
18:28.22 | tzanger | funky guitar music and slow sexy voice "welcome to wells fargo" |
18:28.54 | SexyKen | •jontow• I'm interested in talking to you regarding some developmen.t |
18:29.15 | opus___ | there was a number that if you called on a mpx200 smartphone it would emit some tone that caused the phone not able to input dtmf |
18:29.15 | Juggie | are you paying? :P |
18:29.23 | Zeeek | tzanger: calling @NuFone/18007424932 |
18:29.36 | Zeeek | my CID 866nnnnnnn |
18:29.38 | tzanger | Zeeek: I call the same number with a SetCIDNum(8005062688) in front of it and I get congestion |
18:30.16 | Zeeek | so 866 would give that too? |
18:30.19 | tzanger | Zeeek: as I said, it depends entirely on what the far-end 800# is set to accept |
18:30.22 | Zeeek | I don't get congestion, it just rings and rings |
18:30.29 | *** join/#asterisk viLeR (1000@ip-33-7.telesat.com.co) |
18:30.41 | Zeeek | drops thru to voipjet and connects right away |
18:30.48 | tzanger | Zeeek: change your callerid and see what it does |
18:30.55 | Zeeek | k |
18:31.02 | *** join/#asterisk mixi (~mixi@pD9EE1CBE.dip.t-dialin.net) |
18:32.50 | Zeeek | tzanger - that's hilarious - ok, understood |
18:33.07 | tzanger | Zeeek: I ran into this with a customer |
18:33.08 | tzanger | same issue |
18:33.18 | tzanger | solution was to set the outgoing callerdi to a non-800# when dialing 800#s |
18:36.06 | Zeeek | I thought that was impossible since |
18:36.15 | Trionnis | ~amp |
18:36.16 | jbot | [amp] an Audio MPEG Player. [non-free] |
18:36.19 | Trionnis | erf |
18:36.22 | Trionnis | ~AMP |
18:36.22 | jbot | [amp] an Audio MPEG Player. [non-free] |
18:36.26 | Trionnis | grrr |
18:36.43 | ManxPower | Trionnis, Do it again! Maybe jbot will change it's answer! |
18:36.43 | Trionnis | ~google asterisk management portal |
18:36.53 | Trionnis | lol |
18:36.54 | Trionnis | :P |
18:37.42 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
18:38.03 | Zeeek | ~seen Sideways |
18:38.04 | jbot | Zeeek: i haven't seen 'sideways' |
18:38.04 | Goshen | What is this? |
18:38.05 | Goshen | Mar 7 11:33:13 WARNING[21547]: chan_zap.c:5653 handle_init_event: Detected alar |
18:38.05 | Goshen | m on channel 1: Red Alarm |
18:38.14 | Zeeek | you should, it won all the oscars |
18:38.32 | ManxPower | Goshen, that means the telco line went away |
18:38.49 | Goshen | odd |
18:38.57 | Goshen | ManxPower: thanks |
18:39.40 | Trionnis | the POTS line died?? but according to that commercial I saw from SBC, that doesn't happen!! |
18:39.47 | Trionnis | ./sarcasm |
18:41.10 | *** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
18:46.46 | thepdakid | SexyKen, please explain what you mean by extension mobility. |
18:46.49 | Goshen | hmm, perhaps my generic x100p is acting up |
18:47.27 | Zeeek | give Digium support a call |
18:47.38 | Zeeek | not |
18:47.41 | Goshen | lol, not |
18:49.42 | opus___ | hey |
18:51.27 | Zeeek | so with all theis testing of free providers maybe someone has come up with a way to route a call from FWD to sipgate to gossiptel to CallUk to sipphone to mytcom.it to like2phone and back? |
18:51.48 | Zeeek | lag about 1800ms |
18:52.13 | jontow | thats a lot of call processign.. :) |
18:52.17 | Trionnis | wow |
18:52.31 | Trionnis | reminds me of my ... er "a friend's" blueboxing days |
18:54.06 | modulus_ | k3wl! 4n 31337 ph0n3 h4x0r! |
18:54.15 | Zeeek | so getting back to my meetme transfer problem: I'd like to be able to start a conference from a customer I call, send him to meetme and then go there my self on a lowly Grandstream. Is that possible (GS doesn't do actualtransfers) |
18:54.15 | Trionnis | laf |
18:54.18 | Trionnis | ;) |
18:54.27 | modulus_ | :P |
18:54.35 | *** join/#asterisk spackle (~spackle@209.234.83.19) |
18:54.38 | Trionnis | it was fun to call through all that, and hear like 3-5 seconds of lag in the audio |
18:55.04 | Zeeek | I got that on FWD in the old days |
18:55.14 | Zeeek | It was like being on the moon |
18:55.22 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
18:55.37 | Trionnis | yup |
18:56.02 | spackle | Trionnis: just joined, & I'm curous about what you were calling though. |
18:56.22 | spackle | er, through. |
18:56.47 | spackle | Playing with tin cans and string again? |
18:57.02 | Zeeek | old POTS blueboxing |
18:57.06 | *** join/#asterisk iceyp (~icepick@firewall.unix.co.nz) |
18:57.22 | iceyp | hey guys, how do you run asterisk in the most possible debug mode... asterisk -vvvvvvc ? |
18:57.45 | iceyp | I want more than that, i.e. a call that hits the pabx but doesnt have right context, or extension dont exist etc |
18:57.56 | Trionnis | add a few more v's |
18:58.20 | Ash | iceyp: just enable debugging from the cli |
18:58.25 | iceyp | when someone calls my UK number, i was getting a 'sip-inbound' doesn't exist in my logs, so i added the context and the extension, but now i get no logs |
18:58.36 | Ash | does asterisk have a way of piping out debug output elsewhere? |
18:58.46 | iceyp | so i dont know if i ment to do 004487 or 4487 or 87.. as the extension |
18:58.51 | Ash | other than, say, running it behind 'script' |
18:58.59 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
19:00.35 | opus___ | whoah i've got 1008ms lag to broadvoice |
19:00.39 | Delvar | it can spit out most info to log files... |
19:02.32 | opus___ | why is asterisk continously 'auto destroying call' and 'stopping retransmission' for sip clients |
19:03.25 | modulus_ | for every sprinkle i find, i shall kill you. - stewie |
19:03.42 | harryvv | opus, does it destroy the call when you hang up |
19:03.54 | Zeeek | how can I put a caller into a conference room? |
19:03.57 | Blackvel | opus___: brvoice? |
19:04.00 | Blackvel | does it work again? |
19:04.06 | modulus_ | broadvoice sucks |
19:04.09 | modulus_ | broadvoice sucks donkey balls |
19:04.10 | Blackvel | why |
19:04.12 | opus___ | harryvv - good question |
19:04.13 | Blackvel | tell me |
19:04.17 | harryvv | :) |
19:04.17 | opus___ | modulus - who do you recommend |
19:04.32 | modulus_ | i recommend getting your own pstn<-> voip gateway |
19:04.44 | opus___ | Uhh, I think I'll stick with $7 dollars a month |
19:04.53 | modulus_ | bastard |
19:04.54 | Blackvel | opus: $7? where? |
19:05.00 | Trionnis | hahaha |
19:05.04 | Trionnis | 7? |
19:05.07 | modulus_ | 7 bob |
19:05.07 | opus___ | broadvoice.com |
19:05.10 | Blackvel | modulus_: you didnt answer the question :) |
19:05.14 | modulus_ | i sure did |
19:05.19 | Blackvel | it sucks |
19:05.20 | Blackvel | but why? |
19:05.22 | modulus_ | [07-Mar:11:04 modulus_] i recommend getting your own pstn<-> voip gateway |
19:05.27 | Trionnis | you can get a level3 DID from sipmedia with unlim incoming and 60 outbound for 4.95 |
19:05.31 | Blackvel | (not for business use) |
19:05.31 | modulus_ | oh b/c _ALL_ voip providers suck |
19:05.40 | Blackvel | hmm |
19:05.41 | Blackvel | i see |
19:05.47 | Blackvel | you are against everything |
19:05.47 | modulus_ | trionnis, how's sipmedia? |
19:05.47 | Blackvel | eheh |
19:05.53 | Trionnis | so far no complaints |
19:05.55 | opus___ | trionnis -- how is there uptime / latency |
19:05.57 | Trionnis | no outages |
19:06.01 | modulus_ | quality? |
19:06.08 | Blackvel | opus: do you have BV working right now with *? |
19:06.11 | opus___ | broadvoice seems to go down all the time although my configuration is kinda shaky |
19:06.11 | Trionnis | latency is low, but I'm coming from a coloed box in a Savvis DC |
19:06.15 | opus___ | blackvel yes |
19:06.20 | Trionnis | quality is fine |
19:06.26 | Trionnis | ulaw only tho :-/ |
19:06.28 | modulus_ | trionnis, do they support iax? |
19:06.30 | modulus_ | ulaw is fine |
19:06.31 | Trionnis | not yet |
19:06.34 | modulus_ | b/w is not an issue for me |
19:06.39 | Blackvel | opus___: let me call you on your BV pstn |
19:06.47 | Trionnis | I talked to their engineer, and he says it's coming in the next month or so |
19:06.50 | Blackvel | and I want to see if BV sends me to your voicemail |
19:06.57 | Trionnis | that seems to be the usual BS line, so I'm leery |
19:07.05 | opus___ | blackvel - no it goes to my asterisk voice tree right now |
19:07.16 | Blackvel | can you show me? |
19:07.27 | harryvv | What would cause this when starting * Unable to open '/dev/zap/channel': No such file or dir. |
19:07.27 | Blackvel | a well known contact of me has a big problem with it rightnow |
19:07.41 | Blackvel | BV doesnt work as expected, does silly stuff like voicemail instead of ringing asterisk |
19:07.47 | Blackvel | how come it does work for you? |
19:07.51 | Trionnis | harryvv: "modprobe zaptel" |
19:07.55 | modulus_ | trionnis, hah! |
19:08.03 | harryvv | tri, all the modules are installed. |
19:08.11 | modulus_ | it's _always_ "coming soon" |
19:08.18 | Trionnis | yep |
19:08.24 | opus___ | shit this irc script sucks |
19:08.39 | Trionnis | although I have to admit they've been good about the other stuff they've added |
19:08.46 | Trionnis | like I said, time will tell |
19:08.52 | modulus_ | 60 outbound min. cost 4.95? |
19:08.56 | Trionnis | ya |
19:08.57 | harryvv | Tri, the directory does not exist. I had this resolved in the past but my asterisk hd failed and dont recall the steps to resolve it. I do suspect udev has to be linkes to /dev/zap/channel? |
19:09.00 | Trionnis | 2.5c after |
19:09.13 | Trionnis | I think, lemme look |
19:09.14 | modulus_ | nationwide? |
19:09.16 | Trionnis | ya |
19:09.22 | modulus_ | interesting |
19:09.40 | *** join/#asterisk mbranca (~matteo@host-84-222-6-8.cust-adsl.tiscali.it) |
19:09.45 | harryvv | Tri do a ls -il on /dev/zap/channel |
19:09.48 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
19:10.04 | Trionnis | https://www.myphonecompany.com/download/MyDevice%20Form.PDF |
19:10.13 | Trionnis | that's the signup form for their "MyDevice" plan |
19:10.23 | Trionnis | the plans themselves aren't listed on the site |
19:10.27 | Darwin35 | http://pastebin.ca/6970 |
19:10.34 | Trionnis | 1 sec harryvv |
19:10.37 | harryvv | k |
19:10.53 | *** part/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
19:10.58 | harryvv | I think udev is where the config information is being hardlinked to. |
19:11.05 | harryvv | or from |
19:11.20 | RoyK | Anyone that knows if Chris Hozian is here? |
19:11.42 | Zeeek | Who wants to know? |
19:11.45 | Nugget | anyone! |
19:11.59 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
19:12.02 | Zeeek | I don't know anyone |
19:12.13 | RoyK | digium support guy |
19:12.53 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
19:13.00 | Zeeek | So, in order to call my client and transfer him to a conference, I really need to call the conference on my phone and then call him using a call file that dumps him in the conference, yes? |
19:13.27 | Zeeek | Thats' the only way I coan think of that is technology and phone independent |
19:13.53 | RoyK | Zeeek: unplug the cable and plug it into the 'conference' switch |
19:13.55 | RoyK | simple |
19:14.00 | RoyK | ~lart Zeeek |
19:14.05 | Zeeek | ~lart RoyK |
19:14.19 | Zeeek | ewww |
19:14.24 | Zeeek | green blood |
19:14.31 | RoyK | :) |
19:14.32 | RoyK | blue! |
19:14.43 | Zeeek | lots of incest in those countries, huh? |
19:15.27 | tzanger | oh dear |
19:15.47 | *** join/#asterisk gi0ffe (~giofe@200.121.60.8) |
19:16.00 | RoyK | Zeeek: blue blood is a phrase used for royal people |
19:16.10 | Zeeek | so if I dial say *90123456789 and generate a call file in an app right before dialing Meetme i'm cool, huh? huh? |
19:16.26 | Zeeek | And we all know that royals are all related |
19:16.39 | Zeeek | or didn't you know that RoyK? Check out the UK Royals |
19:16.42 | RoyK | Zeeek: you'll prolly need to do more than that to be cool |
19:16.48 | Zeeek | Look at those ears for example |
19:16.49 | RoyK | :) |
19:16.50 | RoyK | hehehehehe |
19:17.02 | Zeeek | no I think I WILL be cool |
19:17.12 | Zeeek | at least in the context of calling a conference |
19:17.14 | RoyK | I thought perhaps you'd missed the phrase |
19:17.24 | Zeeek | ??? |
19:17.30 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
19:17.46 | KalD|Work | Is there anything out there that supports dynamic conference w/ expire times for asterisk? |
19:17.49 | Zeeek | phase modulation? Far superior to Amplitude |
19:17.49 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
19:17.57 | Zeeek | Whoa dude... |
19:18.32 | Zeeek | I think there is a flag like LAST_PERSON_TURN_OUT_LIGHTS or something |
19:18.46 | *** join/#asterisk Luhiwu (~marsosa@200.63.89.248) |
19:18.53 | KalD|Work | lol that'd be cool |
19:19.17 | PTG123 | hey |
19:19.20 | PTG123 | can someone test a sip account for me? |
19:19.28 | PTG123 | i am not getting audio, and want to see if its me locally |
19:19.29 | PTG123 | or my server |
19:20.15 | *** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
19:20.40 | modulus_ | trionnis, myphonecompany.com ? |
19:21.08 | Zeeek | why not call FWD echo or something? |
19:21.15 | Zeeek | PTG |
19:21.21 | Luhiwu | hi all, anyone knows where can I find the expected telephone number lenght for each country? |
19:21.25 | Zeeek | that's what those things are for |
19:21.32 | PTG123 | zeek: well its def not working fo rme |
19:21.39 | PTG123 | zeek: i can't even hear voicemail prompts |
19:21.45 | Zeeek | Luhiwu - there isn't an easy way to do that |
19:21.55 | Zeeek | because some countries have variable lenghs |
19:22.12 | Zeeek | PTG123 are you using NAT and what phone? |
19:22.23 | Luhiwu | Zeeek: ok, but is there any list for the fixed lenghs countries? |
19:22.28 | opus___ | is a Pentium III 700mhz with 128 megs of ram to slow for asterisk? |
19:22.32 | PTG123 | zeek: yes and xpro |
19:22.35 | PTG123 | tried from several computers |
19:22.45 | topping | opus___: no |
19:22.48 | Zeeek | opus_ not enuf RAM, no CPU problem tho |
19:22.56 | opus___ | awesome |
19:22.59 | *** join/#asterisk Damin_Mobile (~pocketirc@158.sub-70-214-16.myvzw.com) |
19:23.02 | sambal | does someone know why i get this error when i try to register on a sip account? |
19:23.02 | sambal | Got SIP response 403 "This domain is not served here" back from |
19:23.10 | sambal | what can be wrong? |
19:23.16 | opus___ | whats your fromdoain= |
19:23.18 | Zeeek | PTG did you turn ON transmit silence in X-Lite/Por |
19:23.27 | sambal | opus___: nothing ;) |
19:23.30 | Damin_Mobile | sambal: you sip domain is wrong |
19:23.30 | opus___ | hehe |
19:23.42 | PTG123 | zeek yah |
19:23.42 | jks | anybody using siproxd? |
19:23.42 | PTG123 | i hear nothing |
19:23.45 | opus___ | try fromdomain= same as host |
19:23.53 | Zeeek | what's your NAT forwarding situation? |
19:24.19 | *** join/#asterisk adorah (~jack@80.179.34.21.forward.012.net.il) |
19:25.08 | adorah | I have sip audio problems with remote users: anyboday can help? |
19:26.34 | Moc____ | hi all |
19:26.42 | adorah | hi hi |
19:26.43 | tzanger | afternoon Moc____ |
19:26.53 | adorah | I have sip audio problems with remote users: anyboday can help? |
19:27.07 | tzanger | adorah: two things. |
19:27.18 | spackle | Moc, are you working on the Nortel protocol? |
19:27.20 | tzanger | adorah: first, not getting an answer within 60 seconds is not a good reason to repeat |
19:27.20 | adorah | listen.. |
19:27.21 | PTG123 | adorah: join the club :) |
19:27.31 | tzanger | adorah: second, you need to ask better questions |
19:27.35 | adorah | :) |
19:27.39 | adorah | suach as? |
19:27.42 | tzanger | ~google how to ask smart questions |
19:27.44 | Moc____ | spackle: not rightnow, I know 2 person does |
19:27.53 | tzanger | that first link is perfect |
19:28.15 | *** part/#asterisk gi0ffe (~giofe@200.121.60.8) |
19:28.20 | spackle | Moc, Just going to offer some encouragement if you were. |
19:28.31 | Moc____ | ;) |
19:28.31 | adorah | Thx..I hope it is not a link for how to ask smart questions:) |
19:28.42 | tzanger | adorah: it is exactly that |
19:28.47 | tzanger | we can't help you unless you ask good questions |
19:28.51 | adorah | LOL |
19:28.55 | Damin_Mobile | Dude... this conference ROCKS! |
19:29.00 | tzanger | and "sip audio problems with remote users" is *not* a smart question |
19:29.07 | tzanger | Damin_Mobile: which conf is that |
19:29.20 | tzanger | topping: who me? |
19:29.22 | tzanger | I am not trolling |
19:29.24 | PTG123 | you know |
19:29.30 | PTG123 | i think i found my problem, its not choosing a codec |
19:29.36 | topping | tzanger: definitely not you |
19:29.36 | tzanger | he wants help, and he needs to be able to communicate effectively to receive it |
19:29.38 | PTG123 | the question is, why isn't asterisk logging errors about it |
19:29.50 | modulus_ | isn't there a def. codec? |
19:29.51 | Damin_Mobile | Von... |
19:29.55 | Zeeek | PTG usually you see that on CLI |
19:30.05 | opus___ | damin - what type of conference setup do you have |
19:30.05 | adorah | Inded I get no error msg at all.. |
19:30.06 | tzanger | Damin_Mobile: ahhhh I thought you were listening to a conf :-) |
19:30.07 | Zeeek | FWD is ulaw only IIRC |
19:30.13 | PTG123 | Zeeek: you would htink :) |
19:30.18 | modulus_ | hi zeeeeeeeeeeeeeeeeek |
19:30.19 | Damin_Mobile | opus; the works.. |
19:30.24 | PTG123 | <PROTECTED> |
19:30.25 | PTG123 | <PROTECTED> |
19:30.25 | PTG123 | <PROTECTED> |
19:30.25 | PTG123 | <PROTECTED> |
19:30.25 | PTG123 | <PROTECTED> |
19:30.25 | PTG123 | <PROTECTED> |
19:30.26 | PTG123 | <PROTECTED> |
19:30.28 | PTG123 | <PROTECTED> |
19:30.29 | Trionnis | modulus_: I'm here... bumped the scroll wheel |
19:30.30 | PTG123 | <PROTECTED> |
19:30.31 | Trionnis | ACK |
19:30.32 | PTG123 | <PROTECTED> |
19:30.33 | Trionnis | STOP |
19:30.34 | PTG123 | <PROTECTED> |
19:30.36 | PTG123 | <PROTECTED> |
19:30.38 | PTG123 | <PROTECTED> |
19:30.40 | PTG123 | <PROTECTED> |
19:30.41 | Trionnis | ... |
19:30.42 | PTG123 | all i see |
19:30.44 | PTG123 | you see any codec probs? |
19:30.45 | modulus_ | someone's got a slow connection |
19:30.46 | topping | PTG123: http://rafb.net/paste |
19:30.49 | nestAr | lol |
19:30.49 | Trionnis | ~lart PTG123 |
19:31.05 | tzanger | that's not good enough jbot |
19:31.09 | tzanger | ~lart PTG123 |
19:31.15 | Trionnis | much better |
19:31.19 | PTG123 | blah blah |
19:31.21 | Trionnis | sheesh |
19:31.22 | modulus_ | jbot halberd? |
19:31.24 | PTG123 | byt anyhow see any issues? |
19:31.25 | PTG123 | am i blind? :) |
19:31.29 | tzanger | PTG123: it seelected ulaw |
19:31.40 | Trionnis | modulus_: yes, that's sipmedia's "consumer" brand |
19:31.43 | PTG123 | no it selected ulaw between my box and teliax |
19:31.46 | opus___ | does sip show peers show that its reachable? |
19:31.57 | Trionnis | sorry, bumped my scroll wheel |
19:32.01 | Trionnis | didn't see your question |
19:32.11 | modulus_ | trionnis, as opposed to non-consumer? |
19:32.23 | Trionnis | yeah, they do bulk accounts |
19:32.31 | modulus_ | sipmedia is bulk? |
19:32.36 | Trionnis | don't know much more than that really, just know it exists |
19:32.43 | Blackvel | does anyone use iaxtel? i want to the the out/in dailing |
19:33.50 | opus___ | shesh |
19:33.58 | opus___ | $30 activation free plus +$10 shipping |
19:34.23 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
19:34.29 | Trionnis | opus___: ? |
19:34.37 | opus___ | myphonecompany .. is that sipmedia? |
19:34.41 | Trionnis | yes |
19:34.50 | Trionnis | look in the upper right hand corner of the main page |
19:34.59 | roamer323 | hey - anyone from VON online? |
19:34.59 | modulus_ | trionnis, you do any calling card stuffs? |
19:35.00 | PTG123 | yes and they suck :) |
19:35.05 | opus___ | I don't immediately see the $5 dollar plan, is this a special deal |
19:35.05 | PTG123 | myphonecompany sucks |
19:35.10 | Trionnis | I pasted the link |
19:35.20 | Trionnis | https://www.myphonecompany.com/download/MyDevice%20Form.PDF |
19:35.24 | Trionnis | that's the signup form |
19:35.28 | Trionnis | for the "MyDevice" plan |
19:35.32 | opus___ | thanks a bunch |
19:35.38 | Trionnis | welcome |
19:35.48 | Trionnis | it's only $10 activ. |
19:35.51 | opus___ | Wait, if myphonecompany.com sucks and broadvoice.com sucks, who else has a deal $<7 dollars that doesn't suck? :) |
19:36.01 | modulus_ | opus___, enter the voip |
19:36.03 | adorah | anyone can explain the meaning of that msg: |
19:36.07 | adorah | Mar 7 21:05:02 WARNING[1804]: chan_sip.c:755 retrans_pkt: Maximum retries exceeded on call 6ed142327c9d4c7733c7e4710ad452c8@192.168.2.102 for seqno 102 (Critical Request) |
19:36.25 | Trionnis | packet loss would be the first guess |
19:36.28 | anthm | means network error |
19:36.29 | opus___ | firewall |
19:36.34 | *** join/#asterisk asjoyner (~asjoyner@dargo.trilug.org) |
19:36.35 | anthm | cant ping the other end |
19:36.44 | PTG123 | opus: all i know is they don't respond to most of their support requests, you can't get through to them on the phone, etc |
19:36.48 | adorah | no firewall along the way but router/nat.. |
19:36.51 | PTG123 | i ported my #s from them |
19:36.53 | RaYmAn-Bx | I get a bunch of those as well..without any problems |
19:37.05 | opus___ | adorah try nat=yes combinations ? |
19:37.06 | modulus_ | who wants to look at some agi code? |
19:37.07 | Trionnis | I'll refute that |
19:37.10 | asjoyner | Someone correct me if I'm wrong, but you can't do traditional PBX "line presences" with Cisco 7960 phones w/ SIP image talking to Asterisk |
19:37.14 | adorah | I did NAT=yes |
19:37.19 | roamer323 | it is impossible to compare voip provider unless everyone is calling the same number, from the same geographic location :-( |
19:37.25 | modulus_ | asterisk::agi is kinda sucky |
19:37.27 | Trionnis | I just talked to the engineer about a porting situation less than an hour ago |
19:37.33 | Trionnis | answered on the 3rd ring |
19:37.33 | opus___ | adorah - for [general] ? |
19:37.39 | opus___ | adorah - do you have externip ? |
19:38.01 | adorah | nope havn't done for [general] wilco..thx |
19:38.27 | adorah | what the hack is externip? |
19:38.38 | modulus_ | opposite of internip? |
19:38.46 | modulus_ | i'm a fuckin' genius |
19:38.49 | *** join/#asterisk marshall (~test@S0106000f66563988.wp.shawcable.net) |
19:38.56 | Trionnis | ironically enough, your typo is accurate |
19:39.04 | opus___ | sets the outgoing address in NAT |
19:39.05 | adorah | oh.yes I've read smthg about it..will try.. |
19:39.09 | Trionnis | it's a hack to IPv4 to make NAT traversal work |
19:39.11 | Trionnis | ;) |
19:39.36 | modulus_ | trionnis what do you use * for? |
19:39.42 | Trionnis | lots of stuff |
19:39.47 | Trionnis | company incoming calls |
19:39.50 | opus___ | i got a system to work that was double natted while still on an external IP for the internet with nat=yes and externip=localarea_ip_number |
19:39.55 | modulus_ | trionnis, office stuffs? |
19:39.57 | Trionnis | bluetooth presence to do auto transfer to my cell |
19:39.59 | Trionnis | yeah |
19:40.10 | modulus_ | trionnis, anything prepaid? |
19:40.26 | Trionnis | also have a shoutcast radio show that connects to a meetme conference for a site I manage |
19:40.34 | Trionnis | prepaid? |
19:40.36 | Trionnis | don't follow |
19:40.42 | opus___ | couldn't get mp3player to play shoutcast yet... |
19:40.50 | Trionnis | well, icecast, actually |
19:40.57 | Trionnis | .ogg stream |
19:40.59 | opus___ | icecast? is that a program |
19:41.04 | modulus_ | trionnis, calling card apps? |
19:41.06 | Trionnis | www.icecast.org |
19:41.07 | BoRiS | marshall!!!!! |
19:41.07 | Trionnis | ah |
19:41.08 | Trionnis | no |
19:41.10 | Trionnis | I don't |
19:41.24 | modulus_ | i wrote a perl agi script to deduct from a balance per outgoing call |
19:41.31 | modulus_ | the problem is agi kinda sucks |
19:41.32 | modulus_ | it's slow |
19:41.39 | modulus_ | so is perl |
19:41.48 | BoRiS | modulus_: What did you expect? :-p |
19:41.49 | modulus_ | and asterisk::agi sucks |
19:41.50 | Trionnis | php > all |
19:41.52 | Trionnis | ;) |
19:42.14 | modulus_ | asterisk::agi == more unneccessary layers of abstraction |
19:42.18 | adorah | point is I can dial from remote sip to Zap or sip and it ring the other hand but can't extablish voice both was |
19:42.36 | adorah | both ways.. |
19:42.49 | opus___ | adorah -- sounds like a firewall is blocking RTP .. just a guess |
19:42.50 | Trionnis | firewall issue, most likely adorah |
19:42.54 | modulus_ | anthm, it's not blasphemey |
19:42.56 | *** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net) |
19:43.02 | modulus_ | anthm, perl has it's many uses |
19:43.03 | Trionnis | make sure you have the ports forwarded properly |
19:43.05 | opus___ | unblock udp ports 1024-50000 ? |
19:43.07 | adorah | I disabled firewalls both sides |
19:43.11 | modulus_ | anthm, not when performance is an issue though |
19:43.23 | modulus_ | don't get me wrong i'm a perler |
19:43.49 | opus___ | adorah - i think there is a bug also in asterisk with nat=yes where you have to actually 'stop now' and restart the whole process |
19:44.03 | *** join/#asterisk Goshen (Goshen@c-67-172-238-57.client.comcast.net) |
19:44.17 | adorah | give me dime for every time I've done it..:) |
19:44.25 | Goshen | when I got that alarm 1 red from my zap line... |
19:44.34 | opus___ | sip show peers shows the user as reachable? |
19:44.35 | Goshen | I picked up the PTSN phone, and sure enough it was dead |
19:44.40 | adorah | yup |
19:44.46 | opus___ | it rings, no sound |
19:44.49 | opus___ | on either end? |
19:44.50 | Goshen | so I went down the street to the interface box, and two guys were there working on it |
19:44.53 | adorah | indeed |
19:44.54 | Goshen | so they fixed it |
19:45.05 | adorah | ring no sound |
19:45.24 | opus___ | i'd run a tcpdump on it |
19:45.32 | opus___ | see if rtp packets are going by successfully |
19:45.44 | adorah | and if mot? |
19:45.44 | opus___ | and if so, i would experiment with turning off all codecs and enabling them one by one |
19:46.24 | adorah | I use xlite btw..may be there is the problem? |
19:46.25 | opus___ | adorah run that by again, mot? |
19:46.34 | *** join/#asterisk gr8nash (~basketoju@mamabear.si-forest.com) |
19:46.35 | opus___ | dunno, never used it. I use sjphone |
19:46.43 | opus___ | .. heh, try sjphone |
19:46.55 | spackle | I helped somebody today who had a problem with xlite. |
19:47.04 | opus___ | for some reason i hear xlite is evil |
19:47.07 | BuckRogers | i got xlite, its nice |
19:47.08 | adorah | the dtmdtf is inband in sjphone? |
19:47.13 | spackle | SIP implemenation is off a little in xlite. |
19:47.17 | BuckRogers | why its free great for testing |
19:47.26 | *** join/#asterisk [Outcast] (~knoppix@h0006259a2649.ne.client2.attbi.com) |
19:47.41 | opus___ | adorah - dunno |
19:47.49 | BuckRogers | spackle dont forget they want you to buy the full version |
19:48.00 | adorah | sjphone is free too..here ip phones r very expensive yet |
19:48.22 | spackle | BuckRogers: I've never used it. I used firefly for softphone. |
19:48.22 | Nugget | everyone has differing ideas on what "very expensive" means. |
19:48.25 | scrubb | can anyone tell me whatis the biggest cause of: WARNING[1102212400]: PRI: !! Got reject for frame 4, retransmitting frame 6 now, updating n_are! |
19:48.32 | *** join/#asterisk nani707 (~nene@nat-183.sjc1.globix.net) |
19:48.35 | opus___ | does anybody use an IP phone that they swear by... I support about 60 polycom IP500 and have to RMA about one a week... |
19:48.36 | nani707 | hi everybody |
19:48.39 | adorah | I think over 200$ is very expensive |
19:48.58 | *** join/#asterisk lucca (~lucca@export.accela.net) |
19:49.27 | nani707 | anybody can tell how two trunk SIP outgoing lines , as we do with Zap |
19:49.37 | opus___ | oh yeah, if you want to buy a cheap IP500 the company I rma'em to is selling them on ebay for $80/pop |
19:49.46 | Nugget | heh |
19:49.47 | Juggie | opus___, i've had no problems with the mitel5055 |
19:49.52 | Nugget | pre-abused phones. |
19:49.54 | spackle | opus__: that's scary, are they phone abusers? |
19:49.55 | Juggie | but the 5220's are out now |
19:49.56 | opus___ | give me the MAC and i'll tell you why I rma'd it :) |
19:50.01 | Nugget | haha |
19:50.19 | adorah | I know how it is on ebay..just that import is not easy..local regulations.. |
19:50.21 | opus___ | juggie - thanks, I'll check it out |
19:50.23 | Nugget | we should set up a carfax-style phonefax site for phone MACs. |
19:50.29 | Juggie | cisco's have problems withg the pins in the handset |
19:50.30 | adorah | too much trouble and paperwork |
19:50.38 | Juggie | they allways come loose so you cant hear anything on the headset |
19:51.07 | opus___ | polycom has serious problems with their headset port as well, weird. |
19:51.22 | Juggie | the 5055/5220 i've had no problems with. other then the 5055 locks up if you dont send keep alive packets from the sip server... so set qualify=yes |
19:51.27 | opus___ | last week one person blew up 4 phones in one day |
19:51.28 | adorah | Did anyone try the IAX sf from Sokol? |
19:53.02 | opus___ | mitel 5055 is around $350? |
19:53.47 | Juggie | i think we get them for like 250$ cdn |
19:53.52 | opus___ | oh |
19:53.56 | spackle | opus__: was he feeding them coffee? |
19:54.28 | modulus_ | app_dial.c:510 wait_for_answer: Unable to forward frame |
19:54.30 | opus___ | nope, I think it was either bad power or the guy was turning the headset amp up _so_ high that it blew a circuit |
19:54.34 | modulus_ | almost as useful as BSOD |
19:54.54 | nani707 | hi, i need to trunk two sip outbound lines as single trunk (if possible load balance), there are instructions for zap channels , how to do it with sip |
19:54.56 | *** join/#asterisk newpers (newpers@ip24-56-8-180.ph.ph.cox.net) |
19:55.38 | Juggie | nani707, multiple providers? |
19:55.52 | SexyKen | Hey guys I need a web app that will show me queue info |
19:55.54 | nani707 | same provider juggie |
19:55.57 | SexyKen | Is this hard to develop? |
19:56.12 | Juggie | nani707, then it shoudn't really be an issue, unless you are using two different accounts? |
19:56.22 | nani707 | yes two different accounts |
19:56.32 | spackle | Um, SexyKen, have you looked at Flash operator panel? |
19:56.38 | nani707 | sip/1001 sip/1002 outbound lines for example |
19:56.38 | Juggie | write some code in your dialplan to count the number of active calls per account. |
19:56.43 | Juggie | use global variables. |
19:56.58 | Juggie | which ever account is lower, dial with that |
19:57.17 | adorah | Well enough stupid questions 4 1 day. Thx opus..will try.. |
19:57.25 | nani707 | i have used availchan , this worked but if i have 50 lines i am afraid it takes too much time |
19:57.27 | Juggie | or you can just use random to select the sip to use |
19:57.40 | *** part/#asterisk adorah (~jack@80.179.34.21.forward.012.net.il) |
19:57.43 | modulus_ | Mar 7 12:05:23 WARNING[16904]: Unable to forward frame |
19:57.47 | modulus_ | wtf IS that? |
19:58.05 | Juggie | nani707, sip has no knowledge of number of lines, its up to your provider to determine how many calls it wants to host for you |
19:58.06 | *** join/#asterisk j0 (dan@S010600095b00a5a9.vc.shawcable.net) |
19:58.44 | nani707 | yes, i am using broadvoice , he provides only one per sip |
19:59.11 | nani707 | i read somewhere like zap/g2 which can roundrobin within a t1 , can we do same with sip |
19:59.47 | Juggie | nani707, then that is easy |
20:00.02 | Juggie | just track of the sip line is in use or ont |
20:00.03 | nani707 | but zap is for analog lines only right |
20:00.10 | nani707 | oh! |
20:00.14 | Juggie | zap is for pstn, not necessairly analog |
20:00.20 | Juggie | *of=if |
20:00.22 | nani707 | what if i have 100 lines |
20:00.38 | Juggie | well if you have 100 broadvoice accounts that would be rather painful no? |
20:00.39 | nani707 | can i use availchan still, it may take time right |
20:01.03 | Juggie | i'm not famaliar with availchan & sip |
20:01.17 | nani707 | may be not broadvoice, but some other providers |
20:01.24 | modulus_ | wow that agi was pretty damn fast |
20:01.29 | Juggie | also, you can just try the providers in order |
20:01.35 | nani707 | ok |
20:01.39 | Nugget | chanisavail() can take multiple targets -- it's not like you'd need to make 100 calls to that application. |
20:01.43 | Juggie | i mean, if you have an active call, it will reject the second one? |
20:01.53 | Nugget | but I fear there is probably a more fundamental misunderstanding at work here |
20:01.55 | Juggie | then you just move to trying the next provider |
20:02.11 | nani707 | got it juggie |
20:02.40 | nani707 | may be some kind of macro may do it |
20:02.50 | Juggie | you will just have to do a Dial and trap for the error that means call rejected.... |
20:02.57 | nani707 | yeah |
20:03.09 | newpers | do you all se the cisco 7920 wireless ip phone ever dropping it's price from $750 to ~$200? |
20:03.32 | Juggie | that would be easiest.... thats what i do when people dial 4 digit numbers on my sip phones., |
20:03.34 | Juggie | i try it via sip |
20:03.38 | Juggie | if it fails, i dial pri |
20:03.49 | Juggie | i'm on an internal pri so i can call 4 digits on it. |
20:03.50 | nani707 | ok |
20:03.57 | nani707 | ok |
20:04.12 | nani707 | thanks juggie and nugget |
20:04.19 | *** join/#asterisk Damin_Mobile (~pocketirc@99.sub-70-214-6.myvzw.com) |
20:04.33 | *** join/#asterisk hajekd (~hajekd@21.208.65.212.contactel.net) |
20:05.01 | nani707 | this asterisk is really interesting, works the way it is supposed to |
20:05.13 | modulus_ | no it isn't |
20:05.14 | modulus_ | voip sucdks |
20:05.17 | modulus_ | sucks* |
20:05.28 | Damin_Mobile | Yeah... Unlike Windows! |
20:05.32 | opus___ | is there any way to seemlessly crypto sip calls... |
20:05.34 | Nugget | all software sucks. |
20:05.40 | Qwell | Nugget: people too |
20:05.48 | nani707 | there is no software which has no bugs |
20:05.55 | Qwell | except mine |
20:05.59 | nani707 | factor of good better best |
20:06.08 | Nugget | no software is best. |
20:06.25 | *** join/#asterisk nel (~oeo@199.75.106.11) |
20:06.28 | modulus_ | my prepaid agi software is the best. |
20:06.33 | nel | hello |
20:06.35 | nani707 | and what is it? |
20:06.39 | PTG123 | my software is usually pretty bug free :) |
20:06.50 | modulus_ | my software follows the unix philosophy |
20:06.54 | modulus_ | [07-Mar:12:06] *boch!~as24@200.59.172.98* hey |
20:06.57 | nani707 | don wory ptg, when u put load on it u will see bugs popping up |
20:07.03 | nel | I'm having problems uploading sip.dl file on Polycom Soundpoint 500 anybody using those type of phones? |
20:07.09 | modulus_ | hi boch! |
20:07.15 | PTG123 | nani: i am use to writing software that does millions of requests per day :) load is my specialty |
20:07.16 | boch | modulus_ hi |
20:07.17 | nani707 | i saw this areskiCC , it';s cool |
20:07.28 | nani707 | wow PTG |
20:07.31 | PTG123 | right now my big problem is fighting asterisk bugs |
20:07.52 | nani707 | may be write your own Asterisk |
20:07.52 | boch | modulus_ is your agi in perl? |
20:07.57 | modulus_ | boch, it sure is |
20:08.17 | PTG123 | nani707: thought about it, but alot of work :) |
20:08.32 | PTG123 | nani707: i think i am gonna start by redoing chan_sip.c |
20:08.39 | opus___ | nel - got it to work, didn't make any changes to the polycom files |
20:08.43 | nani707 | see!!!, the one who wrote Core is not a idiot |
20:08.58 | boch | modulus_ can u giveme an example of comunicating with * in perl? |
20:09.05 | opus___ | nel -used the web interface to configure the phone for SIP and for REGISTRATION, make sure you have the IP in both |
20:09.13 | modulus_ | boch, sure hold on |
20:09.24 | opus___ | nel - i used the 1.3.1 rom updated from ftp |
20:09.25 | boch | modulus_ :D |
20:09.38 | modulus_ | ummm i'm gonna go smoke |
20:09.41 | modulus_ | i'll be right back |
20:09.44 | nani707 | k mod |
20:10.05 | modulus_ | then i'll show you my perl |
20:10.16 | Nugget | I quit smoking 7 years, 10 months, 23 hours, 10 minutes, and 16 seconds ago. During that time, I would have smoked 62,922 cigarettes. (That's like smoking a 2.98 mile-long cigarette) By quitting, I've saved $11,011.35! I've avoided inhaling 1.64 kg of tar, 100 grams of nicotine, and 1.01 kg of carbon monoxide. |
20:10.29 | Nugget | ^ my perl :) |
20:10.31 | Trionnis | oh boy |
20:10.37 | Trionnis | er |
20:10.48 | opus___ | PTG123 - what type of bugs |
20:12.30 | PTG123 | opus: my latest one is a codec pproblem :) |
20:12.33 | shido6 | boink |
20:12.45 | boch | modulus_ ok, np |
20:12.47 | nani707 | codec , is it possible |
20:13.00 | PTG123 | opus: calls just weren't sneding audio, yet there were no errors in the cli.. finally i realized they weren't selecting any codec |
20:13.29 | PTG123 | it seems the allow/deny codec stuff is flakey |
20:13.35 | PTG123 | it worked before i went to bed which is sad :) |
20:13.39 | PTG123 | i woke up to no calls working |
20:14.13 | nani707 | oh no |
20:14.31 | PTG123 | disallow all |
20:14.34 | PTG123 | allow g729 |
20:14.37 | PTG123 | didn't work :) |
20:14.43 | Pinhole | lots of stuff works when you go to bed with *. I find that if asterisk restarts when I go to bed, more stuff works in the morning. |
20:14.48 | opus___ | seems very straight forward, like, you would have to put effort into messing up that functionality:) |
20:14.49 | nani707 | sometimes i see unknown in show sip channels, also internal addresses on show sip channels |
20:15.03 | opus___ | pinhole - i have that same problem:) |
20:15.04 | PTG123 | well allow all |
20:15.06 | PTG123 | fixed it |
20:15.06 | PTG123 | heh |
20:15.09 | PTG123 | with nothing else |
20:15.16 | PTG123 | i don't care enough to restrict codecs anyhow |
20:17.38 | opus___ | i think thats to get around buggy sip implemenations anyway |
20:17.40 | *** join/#asterisk tigger42 (~tigger42@p54A2536F.dip.t-dialin.net) |
20:17.46 | opus___ | disallow all |
20:17.52 | opus___ | allow the_one_that_works |
20:18.02 | PTG123 | opus___: i think chan_sip just needs to be redone.. i am gonna put that on my list |
20:18.09 | PTG123 | althought i hear someone else may be doing that |
20:18.35 | opus___ | this ircII script seriously sucks |
20:18.44 | PTG123 | i already replaced all the sip user and peer stuff with my own code |
20:18.52 | opus___ | really |
20:19.09 | PTG123 | yes works great :) |
20:19.22 | PTG123 | its more intelligent too, voicemail is now automatic, etc |
20:19.22 | opus___ | what were the problems with the origional stack? |
20:19.32 | opus___ | you just wanted extra features? |
20:19.33 | PTG123 | they were slow pigs :) |
20:19.45 | PTG123 | well i didn't like all the string queries and link list passing of realtime |
20:19.46 | opus___ | i notice that |
20:19.48 | eKo1 | You mean the code is ugly and slow? |
20:19.48 | PTG123 | and i needed a realtime db |
20:19.52 | PTG123 | hah yes |
20:20.07 | eKo1 | Hmm...I'll have to take a look at it. |
20:20.08 | PTG123 | mine is all hash based in memory, synched to disk.. with a remote server for adding accounts |
20:20.25 | opus___ | do you use odbc? |
20:20.37 | PTG123 | no i use my own db.. sort of like berkeleydb |
20:20.45 | PTG123 | and my own client/server sofware for ifacing with it |
20:21.26 | tigger42 | hi all!, before i start to ask silly questions: is there a tutorial to read for following setup: isdn line, isdn capi card, isdn incoming/outgoing calls gated to/from PCs via SIP, no outgoing VoIP Traffic? |
20:21.48 | opus___ | have you searched voip-info.org? |
20:21.55 | opus___ | cd /usr/src/ |
20:22.22 | tigger42 | opos: thanks, will look there. |
20:22.23 | MikeJ[Jayden] | anyone here using head and using IAX trunking? |
20:22.41 | devel | so where are the sqlite database schema creation scripts then? |
20:22.45 | opus___ | ./channels/chan_sip.c |
20:24.10 | devel | ah, i see it "auto creates" the table (not sure i care for that behaviour) |
20:24.31 | opus___ | uhhh |
20:24.37 | opus___ | <PROTECTED> |
20:24.37 | opus___ | nds FROM sipfriends WHERE ipaddr=\"%s\" AND port=\"%d\"", ast_inet_ntoa(iabuf, sizeof(iabuf), sin->sin_addr), ntohs(s |
20:24.37 | opus___ | in->sin_port)); |
20:24.43 | opus___ | mysql functions? |
20:25.02 | opus___ | where ipaddr=\""; drop database master; |
20:25.02 | opus___ | hehe |
20:26.08 | *** join/#asterisk Skysky (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
20:26.09 | *** join/#asterisk marc324 (~marc32344@64-34-29-65.dsl.teksavvy.com) |
20:26.32 | Skysky | hi, can someone suggest me a cheap 1800 provider pls? |
20:26.47 | *** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net) |
20:27.06 | nestAr | iax.cc |
20:27.55 | *** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
20:28.01 | harryvv | what process during the asterisk install makes /dev/zap/channel ? is it make install26 while in /usr/src/zaptel ? |
20:28.19 | harryvv | linux26 I mean |
20:28.52 | tzanger | harryvv: it doesn't do that in 2.6 |
20:28.55 | tzanger | it uses udev |
20:28.57 | tzanger | read README.udev |
20:29.03 | hajekd | how many calls can asterisk handle on pIII/800 MHz? |
20:29.05 | Trionnis | not completely true |
20:29.07 | Skysky | nestAr: im from canada. i wonder if iax.cc do service for canada |
20:29.15 | Trionnis | I'm running a CentOS 3 box that doesn't use udev |
20:29.18 | tzanger | Skysky: no |
20:29.21 | MikeJ[Jayden] | hajekd, depends |
20:29.28 | aminorex | hajekd: about 20 or 30 |
20:29.35 | nestAr | my suggestion is to move. |
20:29.38 | nestAr | but that's just me |
20:29.41 | Trionnis | 2.6.10-grsec kernel |
20:30.32 | Juggie | does anyone still have a copy of x-web (from xten) they dont offer it as a download anymore. |
20:30.48 | Skysky | tzanger: by not serving |
20:30.57 | tzanger | Skysky: huh? |
20:31.11 | Skysky | sry.. i hit enter by mistake |
20:31.32 | harryvv | tzanger yea i did and modified the permission in /etc/udev/permissions.f/50-udev.permissions and does not seem to resolve the permission issue. if thats what the problem is. |
20:31.45 | tzanger | harryvv: are you running as non-root? |
20:31.46 | Skysky | i was just asking . by saying not serving, does it mean that ppl from canada won't be able to reach me by dialing my specific 1800 # |
20:31.53 | harryvv | no running as su - |
20:32.11 | tzanger | Skysky: by not serviing I mean that they will have to subject you to something on the line of a 10c/min tarriff (international 800#) |
20:32.20 | tzanger | Skysky: it's not iax.cc who imposes that, it is the gov. |
20:32.26 | tzanger | harryvv: that's not hte issue then |
20:32.36 | Skysky | oic~ |
20:32.37 | Skysky | thx |
20:32.40 | tzanger | did you add the lines to the udev.rules that *create* the dev structure for it? |
20:32.54 | *** join/#asterisk Corydon-w (tan@vcchgate.vcch01.springfield.tn.us.vcch.net) |
20:33.23 | harryvv | tzanger you mean 50-udev.permissions ? |
20:33.36 | Darwin35 | re |
20:33.36 | tzanger | no |
20:33.38 | tzanger | that's the permissions |
20:33.45 | tzanger | I mean the udev rules themseelves |
20:33.46 | Darwin35 | the t-3 in my rack is stp |
20:33.54 | Darwin35 | boxes back online |
20:34.05 | tzanger | harryvv: |
20:34.06 | tzanger | I mean |
20:34.07 | MikeJ[Jayden] | harryvv, there is a doc on the wiki for centos4, the udev.rules lines you need are in there |
20:34.08 | tzanger | KERNEL="zapctl", NAME="zap/ctl" |
20:34.08 | tzanger | KERNEL="zaptimer", NAME="zap/timer" |
20:34.08 | tzanger | KERNEL="zapchannel", NAME="zap/channel" |
20:34.10 | tzanger | KERNEL="zappseudo", NAME="zap/pseudo" |
20:34.13 | tzanger | KERNEL="zap[0-9]*", NAME="zap/%n" |
20:34.15 | tzanger | in the udev.rules file |
20:34.16 | Darwin35 | everyone here is fired go homre |
20:34.21 | MikeJ[Jayden] | or you could just copy those :) |
20:34.26 | Nugget | I am home. |
20:34.32 | harryvv | i dont see one but do have a 50-udev.rules |
20:34.38 | tzanger | harryvv: add 'em there then |
20:34.41 | harryvv | k |
20:34.59 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
20:35.28 | Juggie | anyone have x-web still? |
20:35.44 | harryvv | tzanger, looks like it was already taken care of. |
20:35.46 | MikeJ[Jayden] | Juggie, is ther ean echo in here? |
20:36.25 | tzanger | harryvv: ok |
20:36.37 | tzanger | harryvv: and /dev/zap/ctl (and timer and channel and so on) were all created? |
20:36.49 | harryvv | no |
20:36.58 | Moc____ | what voip provider do you people recommend ? |
20:37.00 | tzanger | then it wasn't already done :-) |
20:37.01 | eKo1 | Registration timeouts are usually a signed of network problems right? |
20:37.07 | tzanger | harryvv: or you're not running udev |
20:37.11 | Moc____ | maybe there is one I havent tryed ;) |
20:37.13 | tzanger | harryvv: or you haven't loaded the zaptel drivres |
20:37.16 | wolfson | is it normal for out of state calls to just get a CIDNAME of "VA CALL", etc... I assume the telco is trying to save money on lookups |
20:37.19 | MikeJ[Jayden] | MOC! |
20:37.19 | tzanger | Moc____: nufone |
20:37.27 | harryvv | sorry some of them were. Channel was not created |
20:37.39 | tzanger | harryvv: your udev rules file must be fucked up then |
20:37.46 | harryvv | zaptel drivers are loading on boot and have been verified with ztcfg |
20:38.03 | tzanger | harryvv: then your udev rules aren't right |
20:38.09 | tzanger | or udev is br0ked |
20:38.32 | *** join/#asterisk andrew_un (~andrew@h-67-102-251-250.nycmny83.covad.net) |
20:39.02 | harryvv | tzanger, here is whats at the bottom of 50-rules.d KERNEL="zapctl", NAME="zap/ctl" |
20:39.02 | harryvv | KERNEL="zaptimer",NAME="zap/timer" |
20:39.02 | harryvv | KERNEL="zapchannel",NAME="zap/pseudo" |
20:39.02 | harryvv | KERNEL="zap[0-9]*",NAME="zap/%n" |
20:39.30 | harryvv | mm |
20:40.33 | Moc____ | tzanger: Im gaving problem with nufone .. |
20:40.39 | eKo1 | I just know I have network problems somewhere. I'm getting a lot of: Maximum retries exceeded on call... |
20:41.15 | tzanger | harryvv: read README.udev and follow it precisely |
20:41.16 | Moc____ | hi MikeJ |
20:41.19 | tzanger | Moc____: what problems? |
20:41.35 | harryvv | already read but will look again. |
20:41.48 | Moc____ | audio aint good, I get echo, I also get call that audio work only in 1 direction |
20:42.18 | tzanger | Moc____: odd, I do 5kmin/mo with them without *any* issue |
20:42.38 | Moc____ | everything my routing drop on nufone, Im sure to receive a complain |
20:42.48 | tzanger | Moc____: huh? |
20:43.02 | MikeJ[Jayden] | tzanger, where are you? |
20:43.12 | tzanger | MikeJ[Jayden]: midwestern ontario, canada |
20:43.21 | Moc____ | everytime my routing make me use nufone, I get a call sayign qualify aint good |
20:43.24 | opus___ | anyone use scandsp? |
20:43.25 | MikeJ[Jayden] | o... your one of those :) |
20:44.00 | Moc____ | I get the same bad qualify (exact same problem) using Voiceconduits |
20:44.01 | MikeJ[Jayden] | opus, no, no one uses scandsp, some use spandsp |
20:44.15 | Moc____ | asterlink seem alot better |
20:46.48 | Luhiwu | anyone knows who can sell local numbers in USA & Europe? |
20:47.11 | jks | anyone knows of an easy way to get conference call working? (without timing hardware) |
20:47.24 | SexyKen | Hey guys |
20:47.39 | Luhiwu | jks, i've read there is a module for zaptel timing based on usb |
20:47.46 | tzanger | Moc____: that is strange, what's your traceroute to switch-1 look like |
20:47.46 | SexyKen | I keep getting this problem in Asterisk. Agent 1304 is logged into the queue, yet he's not getting the calls. Instead Asterisk CLI shows this: |
20:47.47 | SexyKen | <PROTECTED> |
20:47.47 | SexyKen | <PROTECTED> |
20:47.47 | SexyKen | <PROTECTED> |
20:47.51 | jks | Luhiwu: do you know it's name? |
20:48.01 | jks | Luhiwu: I've tried app_conference, but it's very alpha quality |
20:48.02 | Luhiwu | jks: looking for it |
20:48.23 | *** join/#asterisk iceyp (~icepick@max.unix.co.nz) |
20:48.33 | andrew_un | Hi All! any one can help with tons of "Uhhuh. NMI received for unknown reason 21..." stuff? |
20:48.49 | nel | I'm having problems uploading sip.dl file on Polycom Soundpoint 500 anybody using those type of phones? |
20:48.57 | Luhiwu | jks: http://www.voip-info.org/wiki-Asterisk+timer |
20:49.03 | *** join/#asterisk santiago (~santiago@63.245.86.95) |
20:49.06 | iceyp | hey guys, creating a meeting room or conf room without a password, is it required to have zaptel drivers or are their other ways |
20:49.06 | harryvv | vonage is really pushing there marketing up here in bc canada. |
20:49.20 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
20:49.47 | jks | iceyp: try app_conference |
20:49.54 | *** join/#asterisk Televoip (~Televoip@HSE-Ottawa-ppp3493161.sympatico.ca) |
20:49.55 | harryvv | see it on tv on the radio in newspaper flyers. |
20:50.49 | *** part/#asterisk Televoip (~Televoip@HSE-Ottawa-ppp3493161.sympatico.ca) |
20:52.35 | *** join/#asterisk Spacebar (~stingray@stingr.net) |
20:52.54 | *** join/#asterisk nomercious (~nomerciou@HSE-Ottawa-ppp3493161.sympatico.ca) |
20:53.28 | opus___ | do you need to have the asterisk zaptel dummy timer installed for pure VOIP? |
20:53.37 | Jer1326 | <PROTECTED> |
20:53.40 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net) |
20:54.14 | nel | I'm having problems uploading sip.dl file on Polycom Soundpoint 500 anybody using those type of phones? |
20:54.17 | RoyK | opus___: depends what you do |
20:54.29 | RoyK | opus___: for SIP, non-trunking IAX etc, no need |
20:54.46 | *** join/#asterisk ckruetze (~ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
20:56.45 | iceyp | jks to transfer a call it also requires zaptel? |
20:56.58 | jks | iceyp: no |
20:57.02 | PTG123 | what is the diff between the cisco 7960 and 7960G? |
20:57.18 | iceyp | where is the cheapest location to get a 7960? |
20:57.31 | PTG123 | i'll tell you what if you tell me my answer :) |
20:57.43 | iceyp | lol i dont know :( |
20:57.47 | iceyp | i can ask google tho |
20:57.58 | PTG123 | you want new or used? |
20:58.07 | iceyp | either |
20:58.14 | iceyp | ebay too expesnive and difficult to get em |
20:58.15 | Nugget | the 7960G has pictures instead of words on the buttons. |
20:58.26 | RoyK | ~seen dsfr |
20:58.39 | jbot | dsfr is currently on #asterisk |
20:58.43 | RoyK | ~seen beer |
20:58.44 | jbot | beer <ExUser@h-67-101-28-114.nycmny83.dynamic.covad.net> was last seen on IRC in channel #kde, 20d 18h 13m 26s ago, saying: 'to work from my home pc'. |
20:59.01 | tuxinator_linux | PTG123: the G model is 1000M |
20:59.38 | Nugget | what does "1000M" mean? |
20:59.48 | tuxinator_linux | Gig Ethernet |
21:01.03 | tuxinator_linux | PTG123: Sorry I a was wrong |
21:01.28 | fac_ | any idea to generate some uniqueid |
21:01.34 | fac_ | maybe sessionid? on webservice? |
21:01.43 | fac_ | this is for temporaly filename |
21:02.00 | shido6 | isnt there a timestamp.agi |
21:02.05 | shido6 | or something out there somewhere |
21:03.06 | Nugget | tuxinator_linux: I don't believe that is correct. |
21:03.21 | tuxinator_linux | Nugget: tuxinator_linux: PTG123: Sorry I a was wrong |
21:03.32 | iceyp | fsck i didnt find out where to get the phone from |
21:03.39 | Nugget | afaik, the only two differences between the 7960 and 7960G are the pictures/words on the buttons and the wiring needed to do PoE. |
21:03.42 | *** join/#asterisk denon (denon@synapse.subneural.net) |
21:03.42 | *** mode/#asterisk [+o denon] by ChanServ |
21:03.47 | Sedorox | iceyp: voipsupply.com? |
21:04.01 | iceyp | Sedorox they the cheapest? |
21:04.10 | Nugget | the "G" certainly has no correlation to gigabit. |
21:04.15 | Sedorox | dunno |
21:04.16 | Nugget | (if that's what you were guessing) |
21:04.16 | Luhiwu | anyone knows who can sell bulk local numbers in USA & Europe? |
21:04.23 | Sedorox | brb |
21:04.41 | Nugget | the "G" in 7960G is for "Global" |
21:05.23 | tuxinator_linux | Nugget: I got mixed up with the '-GE' on some models |
21:07.11 | tuxinator_linux | Man, I overslept, but like 4 hours |
21:07.34 | tuxinator_linux | oh well |
21:08.42 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
21:08.55 | *** join/#asterisk davidleib (reik@85-250-7-229.bb.netvision.net.il) |
21:10.03 | *** join/#asterisk BigCanOfTuna (~chatzilla@dsl-macn-66-18-205-30-cgy.nucleus.com) |
21:12.32 | BigCanOfTuna | I would like to have my asterisk server call a PSTN number and present a dialtone to the user who picks up. Anyone have an idea how this might be done through the manger api? |
21:12.48 | opus___ | so, whats the best asterisk/database setup (stable) around now? any good instructions? |
21:13.21 | Sedorox | BigCanOfTuna: there is a wiki about how to have asterisk do callbacks |
21:13.32 | Sedorox | probably what you would want... |
21:13.39 | BigCanOfTuna | Sedorox: great, I'll have a look, thanks. |
21:13.55 | Sedorox | yup.. I'm not sure where exactly it is.. but I've seen it thrown in there somewhere |
21:13.56 | Sedorox | lol |
21:14.42 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
21:15.34 | Goshen | does anyone here have personal experice faxing with VOIP? |
21:16.06 | opus___ | goshen - i was looking into building scandsp but in the current cvs head it fails |
21:16.07 | nel | anybody can give me a hand configuring polycom ip soundpoint 500 for asterisk? |
21:16.33 | eKo1 | Hmm....Having * behind a multihomed nat router is trouble I tell you. |
21:16.35 | opus___ | nel - sure |
21:16.48 | nel | opus___: thanks |
21:16.51 | opus___ | eKo1 - yup i got it to work |
21:17.02 | opus___ | nel - use the 1.3.1 rom |
21:17.04 | nel | I'm having problems uploading the sip.ld file from the asterisk wki |
21:17.08 | nel | 1.3.1 rom |
21:17.10 | nel | ok |
21:17.16 | nel | just that file? |
21:17.20 | opus___ | Umm, |
21:17.31 | eKo1 | Somehow, the load balancing is fucking up my sip registrations. |
21:18.00 | opus___ | nel - on your ftp server have this archive unziped http://www.freedomphones.net/polycom/files/SoundPointIP_SIP_1_3_1.zip |
21:18.05 | Goshen | opus: I tried but mine failed too..I posted a bug report on it, and he said you have to edit the makefile or something like that |
21:18.07 | opus___ | don't make any changes to any of the files |
21:18.25 | opus___ | nel - then, have an entry in your sip.conf |
21:18.54 | iceyp | ok found somewhere cheaper... http://ecoustics.pricegrabber.com/search_getprod.php/masterid=559911 |
21:19.03 | iceyp | $296 for 7960G |
21:19.25 | nel | opus___: let me try that, thanks.,..I tried with some of roms on that page and I have problems uploading the sip.ld file |
21:19.25 | opus___ | nat=yes \ type=friend \ host=dynamic \ dtmfmode=inband \ progressinband=no \ progressinband=never \ qualify=yes |
21:19.27 | nel | let me try again |
21:19.34 | nel | thanks!! |
21:19.36 | tzanger | Moc____: nufone does all their own TDM termination -- if you place a call to nufone it is terminated to TDM by them, not handed off ot a third party VOIP provider |
21:19.40 | opus___ | nel - the trick is to first get it working using the web interface |
21:19.49 | nel | I see |
21:19.54 | nel | I have an old firmware installed |
21:19.56 | nel | from 2002 |
21:20.00 | opus___ | then, making changes:) |
21:20.07 | nel | I don't know if that's not letting me upload the new rom |
21:20.14 | *** join/#asterisk e-guy (~e-guy@host162-48.pool8252.interbusiness.it) |
21:20.16 | opus___ | what ftp server are you running? |
21:20.21 | nel | wu-ftpd |
21:20.28 | opus___ | thats your problem |
21:20.36 | nel | :O |
21:20.38 | opus___ | the word is wu-ftpd doesn't work |
21:20.43 | opus___ | vsftpd does |
21:20.52 | nel | thanks I'm gonna try that |
21:20.56 | nel | thank opus!!!!!!!!! |
21:20.57 | nel | :D:D:D |
21:21.24 | e-guy | any coder with experience in iax out there? |
21:22.12 | *** join/#asterisk SleepyCow (SleepyCow@wnpgmb09dc1-89-6.dynamic.mts.net) |
21:22.17 | nel | brb |
21:22.18 | nel | bye |
21:24.28 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
21:24.30 | SleepyCow | Hello all. I am the technical director of a small, non-profit radio station, and would like to implement VOIP telphony. Would anyone be able to take the time to answer a few questions regarding a odd configuration that we would have to implement? |
21:24.39 | *** part/#asterisk santiago (~santiago@63.245.86.95) |
21:24.46 | opus___ | sleepycow sure |
21:25.01 | shido6 | shoot, SleepyCow |
21:25.04 | SleepyCow | Fantastic |
21:25.22 | SleepyCow | Currently, the radio station has a total of eight phone lines, seven of which I would like to replace with VOIP |
21:25.30 | RoyK | ~seen dsfr |
21:25.32 | jbot | dsfr is currently on #asterisk |
21:25.42 | SleepyCow | The eighth is a leased line to our transmitter site, so it has to stay with the telco |
21:25.45 | tuxinator_linux | ~seen me |
21:25.47 | jbot | me <~daniel@tc26.chem.vu.nl> was last seen on IRC in channel #debian, 5d 3h 43m 54s ago, saying: 'can anyone help with getting an external monitor to run?'. |
21:25.55 | RoyK | --- [dsfr] idle 46:33:30, signon: Fri Mar 4 17:55:30 .......... OMG |
21:26.21 | tuxinator_linux | ~seen tuxinator_linux |
21:26.22 | jbot | tuxinator_linux is currently on #asterisk. Has said a total of 155 messages. Is idling for 1s |
21:26.29 | RoyK | ~lart jbot |
21:26.32 | Sedorox | ~seen Sedorox |
21:26.33 | jbot | sedorox is currently on #asterisk (32m 53s). Has said a total of 8 messages. Is idling for 1s |
21:26.37 | SleepyCow | We have plenty of bandwidth, and have an all purpouse box we use as a router, file server, DB server, etc. It is a Sempron 2600+ with 256ram, 2x 80gb RAID 1, dual ethernet |
21:26.43 | RoyK | ~lart Sedorox |
21:26.49 | SleepyCow | And it is currently running debian, 2.4 kernal i think |
21:26.54 | SleepyCow | Anyhow, |
21:27.03 | Sedorox | funny.. as I hate my access cladd |
21:27.21 | SleepyCow | Three of the phone lines are used for the office. They currently have a 474 Prefix and opperate as follows: |
21:27.53 | SleepyCow | 1 Line is a rotary (7027) that if 7027 is busy it rings the second line (6588) and if thats busy dumps straight to voice mail |
21:28.09 | SleepyCow | the third line is (6518), not a rotary, if busy staright to voice mail |
21:28.26 | SleepyCow | two phone in the office, one with 7027 & 6588, one with 6518 & 7027 |
21:28.36 | SleepyCow | we also have 4 POTS lines, hooked to our radio equipment |
21:28.39 | shido6 | 5 phones so far, right? |
21:28.45 | shido6 | 4 pots lines from the telco |
21:28.56 | SleepyCow | 2 phones on 3 lines, all from a PBX, prefix 474 |
21:29.08 | SleepyCow | 0 phones (radio equipment) on 4 pots lines, prefix 269 |
21:29.47 | SleepyCow | the POTS lines opperate as follows: The main line (8636) is a rotary for two more lines. If all three are busy they just busy out. |
21:30.07 | SleepyCow | There is a fouth line we use as a VIP line for outgoing calls only, generally (8929 i think) |
21:30.16 | SleepyCow | so here is what I need to know: |
21:30.34 | *** join/#asterisk zyke (~zakforeve@84.45.132.117) |
21:30.51 | SleepyCow | #1: Can Asterisk (or any VOIP solution, for that matter) handle 'rotary' configurations where one phone line will drop through to another of busy |
21:31.00 | shido6 | yes |
21:31.05 | opus___ | is rotary an euroopean word for queue? |
21:31.08 | SleepyCow | #2: How can i interface 4 pots devices to VOIP wihtout buying 4 seperate SIP to POTS converters |
21:31.11 | shido6 | rotary, analog |
21:31.13 | SleepyCow | no no |
21:31.18 | SleepyCow | rotary means, say you have 2 lines |
21:31.19 | shido6 | pulse phones |
21:31.21 | tuxinator_linux | pulse |
21:31.24 | SleepyCow | NOT PULSE |
21:31.26 | shido6 | ok |
21:31.29 | Blackvel | hehe |
21:31.31 | SleepyCow | 555-1111 and 555-1112 |
21:31.35 | SleepyCow | i am talking on 555-1111 |
21:31.41 | tuxinator_linux | call hunt? |
21:31.48 | SleepyCow | you call 555-1111. It is buys, but instead of giving you busy it ring 1112 |
21:31.51 | shido6 | yes |
21:31.52 | *** join/#asterisk jterrero (~jterrero@mcse-irc.isys-networks.com) |
21:31.52 | opus___ | you mean a work queue, workgroup, rotary to us means grandma's phone |
21:31.53 | SleepyCow | call hunt might be a word for it yes |
21:31.57 | shido6 | u can do that in Asterisk |
21:32.00 | shido6 | yes you can, SleepyCow |
21:32.02 | Juggie | thats called a huntgroup |
21:32.03 | Juggie | very simple |
21:32.09 | tuxinator_linux | In US is is called 'call hunting' |
21:32.16 | SleepyCow | sorry thats what my telco calls it belive iit or not (MTS Alstream) |
21:32.27 | SleepyCow | ANYHow... |
21:32.27 | opus___ | oh yeah, call hunting != queue. what am i smoking. |
21:32.32 | terrapen | anyone using a rotary phone with an IAXy |
21:32.33 | Juggie | none the less, yes you can call all the phones in order |
21:32.53 | SleepyCow | OK, i figured the 'call hunting' wouldnt be a problem |
21:32.59 | terrapen | i'm wondering if the IAXy translates pulse to tone |
21:33.06 | SleepyCow | to do it, I need a VOIP account / phone number for each, right? |
21:33.08 | terrapen | so i can use a rotary phone to navigate IVRs |
21:33.14 | *** join/#asterisk jedaustin (~chatzilla@140.198.4.225) |
21:33.25 | SleepyCow | terrapen: Best best is to build/buy a pulse to DTMF converter |
21:33.28 | opus___ | terrapen - yeah, i really would like to do that |
21:34.09 | SleepyCow | Anyhow, so now the radio station stuff is all POTS only (The office phones can be replaced with SIP phones) |
21:34.20 | SleepyCow | how do I interface 4 pots lines to VOIP without buying 4 seperate converters? |
21:34.28 | SleepyCow | is there a reasonably priced PCI card that will do it? |
21:35.15 | *** join/#asterisk yaaar (~chatzilla@lifebook.tranquility.net) |
21:35.20 | yaaar | word |
21:35.34 | *** join/#asterisk file[Digium] (~jcolp@66.199.241.90) |
21:35.36 | file[Digium] | hello everyone |
21:35.45 | SleepyCow | Greetings & Saultations |
21:35.59 | sivana | blinking green/amber on a Cisco switch means what? |
21:36.00 | Sedorox | you sucj |
21:36.02 | Sedorox | suck |
21:36.07 | opus___ | sleepycow - you could outsource all the lines to a voip provider fairly cheap. I don't have any experience with the hardware part |
21:36.14 | yaaar | anybody using asterisk@home? And perhaps know why after installation it sits at "GRUB loading stage 2" for like 5-10 minutes before just giving up and leaving me at a "grub>" prompt? |
21:36.31 | Sedorox | something is messed up on the filesystem |
21:36.35 | opus___ | yaaar - wrong cpu for kernel |
21:36.45 | SleepyCow | Okay. Another question. here in canada, are there any VOIP providers that work with asterix, that are cheap, and can get local 204 area code? |
21:36.56 | opus___ | sleepycow chances are yes |
21:36.57 | SleepyCow | my research says no (vonage doesnt work with asterisk right?) |
21:36.57 | Sedorox | grub has nothing to do with kernel and CPU |
21:37.02 | yaaar | opus___: i don't know...it's a bog-standard PII-550 |
21:37.08 | yaaar | wait...make that III |
21:37.17 | Blackvel | tried already to reinstall grub into MBR? |
21:37.27 | opus___ | sedorox - the kernel won't load after stage 2 if its for pentium III and you run it on pentium II |
21:37.53 | Sedorox | ummmm... ookkk.. never heard of that |
21:37.55 | opus___ | yaaar dunno then man maybe sedorox is right about the filesystem |
21:38.01 | yaaar | Blackvel: not yet. It has only done it once, and I just ctrl-alt-del and it's still sitting a "GRUB loading stage 2" still |
21:38.14 | *** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net) |
21:38.19 | Sedorox | SleepyCow: in the wiki there are voip providers for North Amaerica, and have canada and US |
21:38.29 | SleepyCow | Been there allready |
21:38.32 | CoaxD | Is there an option in voicemail.conf to ONLY email a NOTIFICATION of voicemail to the email address specified? (i.e. not the attachment too.) If not, I'll hack the source.. |
21:38.38 | Sedorox | then how can you say no? |
21:38.39 | yaaar | opus___: like I said, the box is P-III ....the doc says you need at least a pII-300 |
21:38.43 | file[Digium] | hey JerJEr |
21:38.58 | file[Digium] | Russell is scaring me, he wants to demonstrate festival |
21:39.11 | SleepyCow | Also, seeing as I dont need any extensions or anything. Just straight voice mail on a few lines, is asterisk the right solution? perhaps we can use somehitng simpler or?? |
21:39.19 | CoaxD | reason being, I wanna specify the email address of a cellphone so I get paged when there's voicemail.. But obviously, the attachment is gonna be extraneous.. |
21:39.32 | *** join/#asterisk Skysky (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
21:39.37 | file[Digium] | bbl |
21:39.39 | file[Digium] | gotta setup stuff |
21:39.46 | Blackvel | n8 |
21:40.11 | CoaxD | duh. pager-email. nevermind |
21:40.19 | opus___ | sleepycow - yes esp if you are non-profit and don't want to buy a $5k system |
21:43.47 | SleepyCow | why would we spend $5k |
21:43.58 | SleepyCow | i figure all i need are 2 sip phones and perhaps some pots adapters |
21:44.47 | SleepyCow | or am I missing something |
21:44.48 | *** join/#asterisk Darwin35 (~Darin@c-24-3-241-22.client.comcast.net) |
21:45.11 | gr8nash | hi all.. can anyone recomend a good VOIP company . ie broadvoice or nuphone. etc? |
21:45.20 | gr8nash | im in the US |
21:46.22 | eKo1 | at&t |
21:46.37 | *** join/#asterisk alegh (~ag11@OL12-112.fibertel.com.ar) |
21:46.37 | jedaustin | I was going to ask the same question.. which VOIP provider is the best solution for asterisk |
21:47.08 | SleepyCow | i would like to find any VOIP provider that will work with Asterisk that has (204) area code |
21:47.09 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
21:47.10 | *** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net) |
21:47.12 | *** join/#asterisk miguellinux (~miguellin@200.47.223.190) |
21:47.25 | eKo1 | Check the wiki people. |
21:47.39 | SleepyCow | I did |
21:47.41 | *** join/#asterisk nel (~oeo@199.75.106.64) |
21:47.55 | opus___ | I wonder if I could sit here all day and broker sip connections:) |
21:48.13 | SleepyCow | hey opus if you think you can sell me service be my guest :) |
21:48.19 | Darwin35 | well broadvoice I am having issues with |
21:48.26 | nel | opus___:Hi again |
21:48.41 | jedaustin | I've heard that Nuphone offers IAX termination, but but other providers offer unlimited options |
21:48.43 | opus___ | sleepycow - I use broadvoice. Setup an account for $7 dollars and experiment |
21:48.49 | opus___ | nel any success? :) |
21:48.50 | gr8nash | Darwin35 what problems? |
21:48.52 | Darwin35 | 3 weeks now and no one from thier tech suppor has responded to my request for a phone call |
21:49.00 | gr8nash | ahh |
21:49.01 | nel | I downloaded 1.3.1 |
21:49.03 | xkev | can I check out from cvs code from a certain point in time? I started modifying crap without making a copy |
21:49.03 | jedaustin | Darwin35: Thats not good |
21:49.03 | Darwin35 | calls not rining threw |
21:49.07 | nel | and used vsftpd |
21:49.11 | nel | but still having the same problem |
21:49.14 | Darwin35 | dialing out and getting nothing but dead air |
21:49.17 | nel | "error saving application sip.ld |
21:49.45 | nel | 0307082947|so |4|01|---------- Initial log entry ---------- |
21:49.46 | nel | 0307082947|cfg |4|01|Initial log entry. |
21:49.46 | nel | 0307082947|cdp |4|01|Initial log entry |
21:49.46 | nel | 0307082947|cdp |5|01|CDP is DISABLED. CDP not detected at boot. |
21:49.46 | nel | 0307082947|cdp |5|01|802.1Q/VLAN tagging is DISABLED. |
21:49.46 | nel | 0307082947|so |3|01|Target: Orion, Board=2345-11500-001 Rev=A, IP=172.16.1.21 |
21:49.48 | nel | 0307082947|so |3|01|Target: Build=2.0.2 30-Apr-02 16:33 |
21:49.50 | nel | 0307082947|so |3|01|Target: BootBlock=1.0.1 15-Jan-02 17:14 |
21:49.52 | nel | 0307082947|so |3|01|Serial: Part number=2345-11500-001;Revision=1; |
21:49.54 | nel | 0307082947|app1 |4|01|Initial log entry. |
21:49.57 | Darwin35 | caller id not working call waiting not working |
21:49.58 | nel | 0307082947|app1 |4|01|Could not initialize resolver library with server 0.0.0.0 and domain . |
21:50.00 | nel | 0307082948|so |3|01|Link status is Net up, PC down. |
21:50.02 | nel | 0307083003|app1 |4|01|Could not load time from (0.0.0.0) repeatedly. Bad sign. |
21:50.04 | nel | 0307083003|app1 |3|01|Bootline: eim(0,0)bootHost:flash e=172.16.1.21:ffffff00 h=172.16.1.6 g=172.16.1.1 u=polycom pw=password tn=CircaIP |
21:50.06 | Darwin35 | nel |
21:50.08 | nel | 0307083021|cfg |3|01|Attempting to use 000000000000.cfg. |
21:50.09 | nel | 0307083022|cfg |4|01|File is 6089998, which is bigger than file system.!! |
21:50.11 | nel | 0307083022|app1 |6|01|Error in saving application. |
21:50.11 | jedaustin | nel: I read theres a patch on http://www.broadvoice.com/support_install_asterisk.html |
21:50.11 | Darwin35 | ~pastebin |
21:50.13 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
21:50.15 | jontow | yikes man |
21:50.20 | opus___ | nel - oh shit |
21:50.26 | spackle | make it stop |
21:50.43 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
21:50.44 | opus___ | nel -- did you wipe out the entire directory in the ftp server before extracting 1.3.1 |
21:50.45 | opus___ | ? |
21:50.56 | nel | mmm |
21:51.00 | nel | yes |
21:51.06 | nel | well I have a backup folder |
21:51.10 | nel | let me wipe everything |
21:51.11 | nel | except |
21:51.17 | nel | the files from the SIP 1.3.1 |
21:51.18 | SexyKen | Anyone know why the Flash Operator Panel would show everyone as available when we're really on calls? |
21:51.28 | opus___ | does the phone come up eventually? |
21:51.30 | jontow | not updating? |
21:51.47 | SexyKen | •jontow• But when I shut down op_server.pl it knows. |
21:51.49 | Darwin35 | yes |
21:51.54 | nel | jedaustin: I will try the patch |
21:51.55 | tigger42 | could someone point me to a tutorial on how to setup asterisk as an isdn gateway using a isdn4linux card so all PCs with SIP software can call out using ISDN lines? |
21:51.56 | jontow | sexyken; look at the CLI and see if a user is logging into the manager interface when you restart op_server |
21:51.59 | terrapen | im not sure that i want to use a pulse-to-tone converter |
21:52.01 | zyke | any one using the port sipura ? i can't make concurrent calls with it .. does it need particular setting in asterisk? |
21:52.03 | terrapen | between the phone and the IAXy |
21:52.09 | zyke | 2 port sipura? |
21:52.10 | terrapen | but i guess i won't have much of a choice |
21:52.26 | SexyKen | jontow - nope I dont |
21:52.45 | Darwin35 | broadvoice is pissing me off. |
21:52.50 | jontow | ok, thats a potential problem.. look at op_server.cfg |
21:53.07 | SexyKen | Ah there it goes. |
21:53.16 | *** join/#asterisk Skysky (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
21:53.20 | nel | jedaustin: I think the patch is not related to Polycom phones? |
21:53.32 | SexyKen | jontow - Do you know what the little arrow above my queues means? |
21:53.36 | nel | is there anyway to erase everything on the polycom phone? |
21:53.36 | opus___ | nel - i didn't need a patch .. hmm |
21:53.37 | jontow | double click on it |
21:53.40 | jontow | its your queue data |
21:53.41 | jontow | :) |
21:53.50 | nel | maybe there is not enough free space |
21:53.52 | jontow | your phones should get it too once there is call history availble |
21:54.02 | jontow | as well as conferences and zaptel channels |
21:54.21 | opus___ | nel - on the wiki they talk about the problem you are having.. but its about version 2.5 or something, thats why i asked if you wiped the directory |
21:54.28 | jedaustin | Right.. patch related to asterisk/Broadvoice |
21:54.30 | MikeJ[Jayden] | anyone doing call waiting? |
21:55.01 | SexyKen | jontow - Interesting I must have done something wrong then...cuz it shows no one in the queues. |
21:55.07 | SexyKen | And still no one on the phones |
21:55.11 | SexyKen | But the manager is logged in. |
21:55.14 | jontow | weird :) |
21:55.21 | opus___ | nel -- when I had that problem I tried 1.3.1 and it worked. Eventually ( I think 10 minutes) it just booted.. also the polycom IP500 phone I have is actually a rebranded one from Shoreline called the sp500 I dunno if it has extra memory |
21:55.34 | MikeJ[Jayden] | I have not done anything with call waiting but think I want to with the analog phones at the house, just not sure quite how it works with the devices Ihave |
21:55.37 | nel | I see |
21:55.41 | nel | :( |
21:55.54 | *** part/#asterisk nomercious (~nomerciou@HSE-Ottawa-ppp3493161.sympatico.ca) |
21:55.57 | opus___ | umm |
21:56.15 | opus___ | If you know vxworks you can telnet into the phone and initiate a reflash |
21:56.19 | opus___ | :) dunno about that |
21:56.29 | opus___ | same OS that fucked up the mars rover:) |
21:56.39 | Darwin35 | http://pastebin.ca/6978 |
21:57.22 | nel | I will try to telnet the phone |
21:57.31 | wolfson | that was a design issue not an problem in vxworks, they actually would likely have not fixed it, if it was not running vxworks |
21:57.46 | Beirdo | opus___: it wasn't the OS that fucked the rover |
21:57.54 | wolfson | bad things happen when you run out of of hd and swap space |
21:57.56 | Beirdo | it was crappy software design |
21:58.02 | Darwin35 | that extensions.conf should kepp you all happy major functionality |
21:58.03 | Beirdo | priority inversion |
21:58.05 | jedaustin | Can asterisk be set to call 3 numbers simultaneously and transfer an inbound call to the first that picks up? |
21:58.08 | *** join/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca) |
21:58.22 | terrapen | wouldn't it be cool if Asterisk could do Pulse ---> DTMF translation internally? |
21:58.28 | terrapen | without the use of special hardware? |
21:58.35 | MikeJ[Jayden] | jedaustin, yes, just use dial, and seperate with , |
21:58.39 | Darwin35 | it can I thought |
21:58.46 | Darwin35 | I thought they fixed it |
21:58.53 | terrapen | how? |
21:59.08 | Darwin35 | should be in the wiki |
21:59.16 | terrapen | http://www.sandman.com/images/superdial.jpg |
21:59.24 | terrapen | that damned thing is 50$US |
21:59.27 | Darwin35 | it might need a tdm card |
21:59.35 | terrapen | http://www.voip-info.org/wiki-Dial+Pulse+to+Touchtone+DTMF+Converters |
22:00.17 | MikeJ[Jayden] | not , .... & sorry |
22:00.33 | SexyKen | jontow you here? |
22:00.39 | terrapen | it does dialing on a ZAP channel |
22:00.54 | terrapen | but I don't think it can take an IAX2 channel and convert pulse to DTMF |
22:00.58 | terrapen | but maybe i am wrong |
22:00.59 | denon | so what the heck still uses pulse anyway? |
22:01.03 | terrapen | i don't understand how it would work |
22:01.07 | *** join/#asterisk xorol (~x@d51532D42.access.telenet.be) |
22:01.08 | xorol | ello |
22:01.09 | terrapen | denon, me and my crazy-ass phones |
22:01.14 | *** join/#asterisk ScythelX (Fleb@pc-24-181-176-72.sbi.ct.charter.com) |
22:01.15 | xorol | anyone using welltech 1501 or 3502 ? |
22:01.18 | terrapen | denon, i want to put a rotary payphone in my room |
22:01.18 | denon | old antiques or somethin? |
22:01.28 | denon | ah yeah, figured that had to be the case |
22:01.30 | Darwin35 | lol |
22:01.37 | terrapen | i like 1970s phones |
22:01.41 | terrapen | esp. Western Electric |
22:01.54 | eKo1 | Pulse is cool. I remember dialing 911 using nothing but the hook button. |
22:02.06 | Darwin35 | I have a batphone when I pick itup it dial only 1 extension |
22:02.24 | *** join/#asterisk Damin_Mobile (~pocketirc@3.sub-70-214-7.myvzw.com) |
22:02.30 | ScythelX | hello all I'm setting up voicemail to read messages from the database in blob format - I have the table setup now do I edit extconfig.conf to point asterisk to start using the database to store its information? |
22:02.35 | nestAr | lol @ batphone |
22:02.53 | nestAr | do you keep a glass globe over it? |
22:02.57 | Darwin35 | they have them for sale on ebay |
22:03.13 | Darwin35 | its the redphone with a sticker that says batfone |
22:03.18 | nestAr | lol |
22:03.32 | terrapen | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=6158793968&ssPageName=STRK:MEWA:IT |
22:03.37 | terrapen | thinking about that for my bedroom |
22:03.43 | terrapen | but that guy's shipping is a rip-off |
22:03.50 | terrapen | so i may go with a full-on payphone |
22:04.15 | ScythelX | thats sweet |
22:04.23 | terrapen | isn't it. |
22:04.25 | xorol | anyone using welltech 1501 or 3502 ? i got transfer problems |
22:04.33 | Darwin35 | I am working on a touchscreenn phone with video |
22:04.44 | terrapen | and that jackass wants to sell you the mounting bracket, too |
22:04.50 | terrapen | which came with the damned phone |
22:04.51 | ScythelX | isnt the cisco 7970 touch screen? |
22:05.21 | terrapen | i *have* to have a bell stand for this |
22:05.29 | Darwin35 | those blue phones are for callingcard and creditcard and collect calls only |
22:05.37 | terrapen | http://www.payphoneoutlet.com/9833.html |
22:05.44 | terrapen | that's what i want in my room |
22:05.51 | terrapen | darwin: or Asterisk :) |
22:05.52 | SexyKen | Anyone here use Flash OPerator Panel? |
22:06.04 | Darwin35 | yeah |
22:06.23 | Darwin35 | I want a old British Pay Fone |
22:06.33 | SexyKen | When someone is on a call, shoulnd't it show that via the panel? |
22:06.35 | terrapen | that damned pedestal is $268 |
22:06.37 | Darwin35 | and the call box they caame in |
22:06.43 | KalD|Work | anyone ever have an IAXY box just lock up after an hour? |
22:06.45 | terrapen | so, add $100 for the phone |
22:07.05 | terrapen | kald: i hope not. haven't tested mine that much yet |
22:07.45 | Darwin35 | 1950's Princess Retro-style Telephone |
22:07.45 | Darwin35 | Regular price: $89.95Sale price: $59.49 |
22:08.12 | KalD|Work | terrapen, I was in a conference and the box just went dead - I pulled the network cable and the two lights on the cat5 jack were solid - and the off-hook like was on even after I unplugged the phone. I had to pull power for 30 seconds to get it to come back |
22:08.56 | terrapen | damn. |
22:10.03 | *** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com) |
22:10.03 | xorol | anyone using welltech 1501 or 3502 ? |
22:10.09 | Darwin35 | ok thats it I am going to develop a sip pay phone |
22:10.09 | xorol | srry |
22:10.41 | Darwin35 | that can use creditcards calling cards and cash |
22:10.45 | TheBear | ok trying to install asterisk on gentoo kernel 2.6. On 'modprobe zaptel' I get in /var/log/messages "zaptel: Unknown symbol crc_ccitt_table" any ideas ? |
22:10.52 | *** part/#asterisk davidl (reik@85-250-7-229.bb.netvision.net.il) |
22:13.30 | terrapen | which would be cooler in an apartment: |
22:13.35 | terrapen | a regular payphone |
22:13.39 | terrapen | or a coinless payphone |
22:13.40 | SleepyCow | Quick Q: If three poeple on three phones want to make three seperate outgoing calls, all form the same location/network, I need 3 voip accounts, or just one? |
22:13.41 | *** join/#asterisk Beave (~beave@vistech.org) |
22:13.50 | SleepyCow | terrapen: Regualr pay phone |
22:14.02 | SleepyCow | terrapen: Preferable rotary dial 3 slot |
22:14.10 | terrapen | i dont like the 3-slots |
22:14.19 | SleepyCow | Damn young whipersnapper |
22:14.19 | terrapen | i'm more of a fan of 1970s era |
22:14.22 | SleepyCow | ;) |
22:14.25 | Darwin35 | http://www.payphoneoutlet.com/19paclmopain.html |
22:14.51 | terrapen | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=985&item=6156299182&rd=1&ssPageName=WDVW |
22:14.56 | eKo1 | terrapen: So you have an afro? |
22:14.56 | *** join/#asterisk Martz (Martz_test@62.3.201.11) |
22:14.57 | terrapen | some lucky bastard go that |
22:15.00 | SleepyCow | So anyone have an awesome answer to my basic question? |
22:15.29 | terrapen | WOW |
22:15.30 | terrapen | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=38038&item=6511839802&rd=1&ssPageName=WDVW |
22:15.32 | terrapen | look at that |
22:15.35 | terrapen | lucky bastard |
22:15.46 | Darwin35 | question whats your question you old fart of a sleepy cow |
22:15.47 | SleepyCow | thats a POS |
22:15.50 | spackle | SleepyCow, if you have IAX trunking you would only need one account. |
22:16.09 | SleepyCow | spackle, elaborate please - can I get that (cheaply) from a VOIP provider? |
22:16.20 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
22:16.20 | SleepyCow | and run Asterisk at my end on our multipurpose linux box? |
22:16.26 | ScythelX | hello all I'm setting up voicemail to read messages from the database in blob format - I have the table setup now do I edit extconfig.conf to point asterisk to start using the database to store its information? odbcstorage => odbc,mysql1,voicemessages isnt workin |
22:17.17 | eKo1 | Are you using head? |
22:17.21 | ScythelX | yeah |
22:17.38 | spackle | SleepyCow, I use, Ahem, Nufone for ouot bound calls, and they allow trunking. I can make multiple outbound calls on the same account. I don't think it is very expensive, but that is subjective. |
22:18.59 | jedaustin | Other than nuphone, what voip providers allow trunking? |
22:19.39 | SexyKen | Anyone know why agent stuff wont work with flash operator panel? |
22:20.11 | SleepyCow | spackle: How about multiple inbound on the same number. Possible? |
22:21.13 | spackle | Sleepycow, I think that depends on your provider. I haven't gotten that far myself yet. ;-) |
22:21.18 | SleepyCow | anyone know if voip.net is useable with Asterisk? |
22:22.32 | nel | this polycom phone is frustrating me |
22:22.42 | nel | I have tried with all the SIP.zip files in http://www.freedomphones.net/polycom/files/ |
22:22.45 | nel | none of them works |
22:23.56 | Darwin35 | asterisk did this to me |
22:23.58 | Darwin35 | asterisk did this to me |
22:24.19 | jedaustin | Darwin35: download asterisk@home.. start over |
22:24.31 | SexyKen | Asterisk wont show agents logging into the queue what's up with that |
22:25.48 | Darwin35 | asterisk@home doees not like my mini-itx board |
22:25.55 | *** join/#asterisk MatsK (~NNSCRIPT@8.80-202-60.nextgentel.com) |
22:26.01 | jedaustin | It's likely Linux |
22:26.13 | gr8nash | jedaustin i just installed @home and my x-lite client says "Discovered Port Restricted Cone NAT Firewall" |
22:26.39 | gr8nash | i dont have a firewall on my box.. and there is none between me and @home.. unless @home has one? |
22:26.46 | jedaustin | gr8nash: are you NATing behind a router? |
22:27.10 | gr8nash | im behind one at work..but both boxes are on the same side of the firewall |
22:27.18 | Darwin35 | I am just working on the fbsd port of * updateing the patches and the ports |
22:28.22 | Darwin35 | and I have to go salvage a bad kernel now |
22:28.35 | gr8nash | NAT: 65.75.199.122 its grabbing my external address when all i want to do is connect to the local astrix box.. heh.. =/ |
22:29.00 | ScythelX | freebsd doesnt support zaptel yet does it? |
22:29.03 | SexyKen | Doesn't anyone know a damn thing about flash operator panel? |
22:29.06 | Jer1326 | yep it do3es |
22:29.16 | nel | opus__: I used SoundPoint_IP_bootrom_2.3.0_w-SIP.zip |
22:29.16 | ScythelX | really |
22:29.20 | nel | and it seems it worked |
22:29.22 | ScythelX | even ztdummy? |
22:29.23 | nel | is that enough??? |
22:29.25 | jedaustin | what's the web site address for nuphone? |
22:29.25 | Jer1326 | yes |
22:29.31 | ScythelX | thats awesome |
22:29.36 | Jer1326 | i use ztdummy all the time |
22:29.50 | ScythelX | how stable is it on freebsd |
22:30.06 | ScythelX | because I wanted to run it on freebsd a while back and couldnt because of the use of zaptel wasnt there |
22:30.06 | *** join/#asterisk ruied (~a@85.138.10.212) |
22:30.09 | Jer1326 | havent had it crash my box yet in 2 mths |
22:30.20 | Darwin35 | jer1326 greetings |
22:30.22 | ScythelX | well thats just awesome |
22:30.24 | opus___ | nel - really |
22:30.29 | Jer1326 | hiya darwin :) |
22:30.32 | ScythelX | definately switching it to my freebsd box |
22:30.48 | Darwin35 | give me a day or 2 before you do |
22:30.56 | Darwin35 | I am working on the ports tree ver |
22:30.56 | Jer1326 | why? |
22:31.07 | Darwin35 | updating to 1.0.6 and a few new patches |
22:31.08 | Jer1326 | please dont make it havew a millon deps |
22:31.17 | Jer1326 | but i need CVS :) |
22:31.30 | *** part/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca) |
22:31.32 | ScythelX | Jer1326: are you running CVS head as well on the bsd box |
22:31.33 | Darwin35 | why |
22:31.39 | Jer1326 | yep |
22:31.47 | Jer1326 | atxfer anit in 1.0.6 |
22:31.52 | ScythelX | Jer1326: that just made my day |
22:32.01 | Darwin35 | then your installing like a linux box and not how we do on fbsd |
22:32.02 | Jer1326 | i'm glad i can make someone happy :) |
22:32.26 | Jer1326 | no i modded my make files to fit |
22:32.34 | ScythelX | hopefully it will run ok in a production environment |
22:32.48 | spackle | SexyKen, go read the FOP docs in the configs and on the web page, what you want to know is there. getting upset isn't a good way to ask for help. |
22:33.02 | Jer1326 | well this server handles 25 concun chans no problem |
22:33.04 | Darwin35 | well I will make a head ver of the port |
22:33.11 | Jer1326 | THANKYOU!!!! :)) |
22:33.15 | Darwin35 | but keeping it patched will be hard |
22:33.20 | Jer1326 | yea... |
22:33.45 | Darwin35 | do you use the patches we have in the asterisk port now |
22:34.04 | gr8nash | anyone know if @home comes installed with a firewall? |
22:34.10 | Jer1326 | me? no. i roll my own |
22:34.23 | Darwin35 | they need to also make asterisk-sounds a part of * |
22:34.29 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
22:34.44 | Darwin35 | tired of manualing moving files |
22:34.48 | Darwin35 | but thats me |
22:34.57 | xorol | anyone using welltech 1501 or 3502 ? i got transfer problems |
22:36.32 | Darwin35 | http://pastebin.ca/6978 |
22:36.45 | Darwin35 | there you go jer go read |
22:37.37 | *** part/#asterisk nel (~oeo@199.75.106.64) |
22:39.05 | modulus_ | hi |
22:41.07 | Darwin35 | they need the asterisk to make a asterisk-head.tar.gz |
22:41.18 | Darwin35 | every night at midnight |
22:41.24 | Darwin35 | and 6 am |
22:41.29 | Darwin35 | and 12pm |
22:41.38 | Darwin35 | and6 pm |
22:41.56 | Darwin35 | that way you get the best head at that tiime asterisk has |
22:42.05 | Darwin35 | sorry that sound s bad |
22:42.21 | MikeJ[Jayden] | darwin, do you have bandwidth? |
22:42.34 | MikeJ[Jayden] | set it up and post it. |
22:42.36 | Darwin35 | no I am on dsl |
22:43.06 | Darwin35 | or atleast once a day |
22:43.19 | Darwin35 | its hard to make a pport pull cvs |
22:46.29 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
22:55.00 | *** join/#asterisk [Paul] (~paul@80.100.33.108) |
22:55.16 | [Paul] | wow |
22:56.18 | *** join/#asterisk ruiner (ruiner@ruiner.netslacking.net) |
22:58.32 | ruiner | ok, i'm a real newb here and don't want to sound dumb, but i'm going to anyway. i've got an asterisk box setup, have a pots line connected to an fxo (i think) card in a cisco 3640, which is routing all calls to my asterisk box via sip. i can call the number and asterisk picks up with the demo sound, but dialing numbers does nothing, i can't do the echo test by typing 600 and can't figure out why. can someone point me in the right direction? |
23:00.00 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
23:00.00 | *** mode/#asterisk [+o bkw_] by ChanServ |
23:00.20 | [Paul] | is your phone set to send DTMF tones? |
23:00.21 | jontow | woi.. you've got a full blown cisco 3640 for a single POTS line/ :) |
23:00.41 | jontow | why not just a single $75 FXO card in the asterisk box? |
23:01.01 | ruiner | work for an isp, we're just using hardware we have laying around to mess around with this stuff |
23:01.04 | jontow | aha :) |
23:01.07 | jontow | cool, then |
23:01.11 | ruiner | as far as dtmf tones, as far as i know it is |
23:01.13 | ruiner | it's not pulse dialing |
23:01.14 | jontow | learning experiences are good |
23:01.27 | jontow | so is dinner.. which is where i think i'm headed :D |
23:01.30 | ruiner | yeah, we want to eventually roll out VoIP for broadband customers |
23:01.49 | ruiner | flat fee to dial any city we provide to |
23:02.13 | ruiner | heh, ok |
23:05.17 | ruiner | is it possible my router is blocking those tones somehow? |
23:06.39 | [Paul] | i have no idea |
23:06.43 | [Paul] | newbe myself :) |
23:07.26 | [Paul] | i'm stuck on transfering calls |
23:08.04 | Jer1326 | what is the problem with it? |
23:08.10 | Jer1326 | paul: |
23:08.52 | [Paul] | it does work with internal extensions, but does not with an external isdn number |
23:09.11 | jedaustin | is nupone a us voip provider? |
23:09.16 | jedaustin | er nuphone |
23:09.18 | [Paul] | that's for unsupervised calls |
23:09.31 | [Paul] | - transfers |
23:09.38 | *** join/#asterisk stustu (~stustu@fluffy.fatburen.org) |
23:09.49 | [Paul] | don't know how to initiate a supervised transfer at all |
23:09.57 | Jer1326 | you need to be running cvs for that |
23:10.02 | Nugget | <-- requires supervision |
23:10.21 | mikegrb | You so do. |
23:10.34 | jedaustin | What's supervised call transfer |
23:10.45 | [Paul] | i'm kind of a linux-newbe |
23:10.45 | mikegrb | when your boss is listening in |
23:11.10 | [Paul] | i've installed cvs with DSELECT |
23:11.24 | Jer1326 | put t in your dialpan |
23:11.28 | KalD|Work | anyone have the proper sox arguments to convert .wav to .gsm? |
23:11.28 | Jer1326 | dialplan |
23:11.44 | Jer1326 | sox -r 8000 -c 1 file.wav file.gsm |
23:11.59 | KalD|Work | Jer1326, thx |
23:12.30 | [Paul] | and i have this annoying echo with SIP |
23:12.37 | *** join/#asterisk justinnnn (~dsf@solid.mpa.net.au) |
23:12.38 | justinnnn | hey |
23:12.44 | justinnnn | anyone wana make some money helping me with txfax :) ? |
23:12.56 | [Paul] | but apparently there's nothing to do about that |
23:13.35 | Jer1326 | what do you nede justinnnn |
23:14.19 | [Paul] | however i've hot asterisk version 1.0.5. that includes the supervised transfers, doesn't it? |
23:14.33 | Jer1326 | no |
23:14.41 | [Paul] | :( |
23:16.57 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
23:17.18 | *** join/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
23:17.37 | *** join/#asterisk impressmenicely (impressmen@host-66-81-63-177.rev.o1.com) |
23:18.18 | redder86 | Thanks for all the fish? |
23:18.37 | mikegrb | So long. |
23:18.47 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
23:19.00 | *** part/#asterisk eKo1 (~bernd@207.42.191.67) |
23:19.34 | Darwin35 | fish I will take to big mouth bass and a side of troout with lobster tail to finish it off |
23:19.36 | [Paul] | i'm doing a MAKE UPDATE |
23:19.42 | [Paul] | that's the idea? |
23:20.00 | stustu | A question about CVS and Asterisk versions: Are all the 1.0.x releases limited to bug fixes, and all new functionality in CVS? Are there any know plans as for when a 1.1 or 2.0 release is up with more new stuff included? How stable is the CVS verison? |
23:20.07 | mikegrb | Todays fish is trout a la crem. |
23:20.30 | KalD|Work | Jer1326, what can i do if it plays back at half the speed? |
23:20.30 | Darwin35 | ok |
23:20.58 | Jer1326 | KalD|Work let me check my syntax hang on |
23:21.05 | devel | hey, is there any way to "clear" a SetGroup? i did a 'SetGroup(${CALLERIDNUM})', and the first call works as expected, but just keeps incrementing, so fails every call after that (even after hangup) |
23:21.58 | redder86 | stustu: in theory the 1.0.x releases are limited to bug fixes only |
23:21.59 | KalD|Work | Jer1326, I got it - |
23:22.06 | Jer1326 | good |
23:22.11 | KalD|Work | Jer1326, put your args inbetween the file names |
23:22.42 | redder86 | stustu: but "bugfix" is usually a very vague term around here. |
23:22.48 | Jer1326 | that works too |
23:22.53 | redder86 | stustu: one person's bug is another's feature |
23:22.53 | *** join/#asterisk gezick (gezick@sartre.ispvip.biz) |
23:23.01 | *** part/#asterisk gezick (gezick@sartre.ispvip.biz) |
23:23.14 | *** join/#asterisk gezick (gezick@sartre.ispvip.biz) |
23:23.25 | [Paul] | i get the following message: cvs [update aborted]: connect to cvs.digium.com(66.225.202.81):2401 failed: Connection timed out |
23:23.32 | redder86 | stustu: some people run CVS HEAD in production environments... so it's gotta be somewhat stable, although I'm sure you have to be more cautious than using CVS 1_0 |
23:23.57 | gezick | could someone point me to an explanation of sip uri's? |
23:24.04 | gezick | trying to figure out what Registration from '<sip:192.168.0.61@192.168.0.61:5060>' failed for '192.168.0.79' means |
23:24.04 | stustu | redder86: Looking at the CVS logs, it seems that the branch was created about half a year ago... I'm running FreeBSD, so I started out with the port (1.0.5 now). |
23:25.37 | stustu | The FreeBSD port contains quite a few patches... But maybe the CVS HEAD does not need them any more? |
23:26.06 | Darwin35 | I am working on that right now |
23:26.13 | dwC- | does anyone know what the best way is to go about having a inbound call from my pstn get a prompt to enter a CIDnum then another prompt to dialout to a IAX channel with that CIDnum set? |
23:26.25 | Darwin35 | it neeeds a few of the patche like the patch:Makefile |
23:26.35 | Darwin35 | to set the paths right |
23:26.40 | Jer1326 | privacymgr |
23:26.44 | Darwin35 | give me time guys |
23:26.54 | Darwin35 | what abot it |
23:27.06 | Darwin35 | jbot sex now |
23:27.22 | Darwin35 | jbot is gone |
23:27.28 | Darwin35 | booo whooo |
23:27.35 | sivana | ~seen normast |
23:27.37 | jbot | normast <HydraIRC@Ottawa-HSE-ppp4119108.sympatico.ca> was last seen on IRC in channel #asterisk, 4d 17h 40m 28s ago, saying: 'juggie: Thanks keep it in mind.'. |
23:27.50 | Darwin35 | ~jobot sex |
23:28.01 | Darwin35 | ~jbot sex |
23:28.03 | jbot | I'm pregnant |
23:28.07 | Darwin35 | hehhee |
23:28.08 | mikegrb | Darwin35: stop playing with the box |
23:28.09 | redder86 | lovely |
23:28.10 | Jer1326 | lol |
23:28.45 | gezick | what does chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 488a7d7843dccfd23f547998336f1644@192.168.0.61 for seqno 102 (Non-critical Request) mean? |
23:28.51 | Darwin35 | patching head is not easy |
23:29.16 | Darwin35 | where is kram |
23:29.20 | tuxinator_linux | VON |
23:29.29 | redder86 | gezick: asterisk gave up trying to place that call? |
23:29.31 | Darwin35 | we need a daily tar.gz of cvs-head |
23:29.42 | dwC- | hrmm |
23:29.45 | Darwin35 | so this port I am making will work |
23:29.48 | stustu | I tried typing "make" on a FreeBSD 5.3 system right now. At least it did compile, but linking complains about "__use_ast_pthread_create_instead__" missing... |
23:29.59 | gezick | redder86: why is it trying to call it... is it because i put a defaultip entry in that extension? |
23:30.05 | Darwin35 | there is a patch |
23:30.11 | Darwin35 | its ion the mailing list |
23:30.12 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
23:30.18 | JerJer[mobile] | don't use freebsd |
23:30.25 | opus___ | gazick - i get that message like every second when no activity is going on as well, can't figure it out |
23:30.28 | redder86 | gezick: I don't know |
23:30.36 | Darwin35 | I am going to add the patch |
23:30.41 | *** part/#asterisk newpers (newpers@ip24-56-8-180.ph.ph.cox.net) |
23:30.50 | gezick | i'm also getting Mar 7 17:30:06 NOTICE[10937]: chan_sip.c:7681 handle_request: Registration from '<sip:192.168.0.61@192.168.0.61:5060>' failed for '192.168.0.79' |
23:30.52 | gezick | all the time |
23:30.54 | redder86 | gezick: I don't think that the IP address should worry you. |
23:31.08 | Darwin35 | stop putting fbsd down I do alot to keep asterisk ported |
23:31.12 | gezick | where 192.168.0.61 is my asterisk server, and 192.168.0.79 is my polycom phone |
23:31.25 | Jer1326 | i had no problem with compiling while it was patched |
23:31.29 | Jer1326 | once |
23:31.29 | redder86 | gezick: well, that's just saying that you have a context messed up in your sip.conf or the SIP client is misconfigured |
23:31.45 | justinnnn | any txfax ppls ? |
23:31.53 | redder86 | txfax: ewe |
23:32.05 | justinnnn | wat do u use ? |
23:32.10 | redder86 | hylafax |
23:32.11 | gezick | is the context before or after the @? |
23:32.23 | redder86 | gezick: in the [brackets] |
23:32.51 | opus___ | i got this message continously for ever |
23:32.53 | opus___ | Mar 7 16:41:08 DEBUG[3529]: chan_sip.c:832 __sip_autodestruct: Auto destroying call 'AE6616BF-0097-4261-B58E-C4403B3CEEF5@192.168.1.101' |
23:32.54 | ruiner | ah, I think I found my problem. In case anyone else is interested, if your Cisco router is taking your calls and passing it to your asterisk box, you need to have a fairly recent version of IOS for DTMF tones to be sent through SIP |
23:33.18 | Darwin35 | http://pastebin.ca/6983 there is the utils.c patch |
23:33.23 | *** part/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
23:33.58 | *** join/#asterisk Dibbler (~Dibbler@snaddy.plus.com) |
23:34.47 | opus___ | anybody have an idea what that means? |
23:35.23 | Darwin35 | for fbsd |
23:35.43 | opus___ | i'm on lamex |
23:36.08 | `Sauron | opus: If you think it's so lame, then quit using it. |
23:36.11 | `Sauron | Duh |
23:36.19 | opus___ | sometimes being lame is cool |
23:37.49 | modulus_ | so you're like |
23:37.52 | modulus_ | really REALLY cool? |
23:38.04 | ruiner | roffle |
23:38.35 | opus___ | yeah, my refigurator runs the latest 2.6.11.1 kernel |
23:38.46 | *** join/#asterisk laloo (~laloo@042.142-60-66.FTTH-SWI.surewest.net) |
23:39.25 | stustu | Darwin35: Thanks! I got the latest CVS through the compiler now... I probably still need to remake with a patched Makefile in order to have the paths right... Would you know where to find a patch for that as well? |
23:40.17 | *** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com) |
23:40.23 | ManxPower | ~docs |
23:40.24 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
23:40.52 | laloo | Guys. Can someone help me? I just upgraded asterisk to version 1.0.6. I was running some old dev version earlier. Now, when I try to run asterisk, it comes back with an error, "/usr/lib/asterisk/modules/pbx_dundi.so: undefined symbol: pbx_substitute_variables_varshead" |
23:40.59 | KalD|Work | how do you conditionally strip the last digit in a number? i.e. if it is # discard it |
23:41.21 | ManxPower | Every day I hate MS Windows more. |
23:41.35 | tuxinator_linux | ManxPower: You too? |
23:42.07 | ManxPower | tuxinator_linux, Silly me, I thought I could save my roaming profile to a USB drive. Unfortunatly my "roaming profile" is almost 200 megs. |
23:42.09 | Nugget | that's how I feel about linux. |
23:42.23 | opus___ | anyone know of a cheap supplier of heatsets (mic/headphone jack based)? |
23:42.25 | ManxPower | Nugget, Heritic |
23:42.43 | Nugget | I'd probably feel that way about windows, too, but thankfully my job doesn't require me to use it. |
23:42.48 | ManxPower | And I may have gotten my first spyware infestation |
23:42.52 | tuxinator_linux | ManxPower: Unneccisarily huge |
23:42.52 | Nugget | oof |
23:42.55 | KalD|Work | ManxPower, but as MS would tell you - drive space is cheap - buy a bigger usb drive.. losers! |
23:43.19 | Nugget | all I do with Windows is play Day of Defeat. |
23:43.22 | Nugget | and run Quicken |
23:43.24 | KalD|Work | ManxPower, they dont get the point - just because you have 1TB doesn't mean fill it |
23:43.35 | tuxinator_linux | KalD|Work: Cheap is more expensive than free |
23:43.36 | ManxPower | I run windows on my laptop, that's all. |
23:43.52 | tuxinator_linux | I am moving my laptops to linux |
23:43.59 | tuxinator_linux | well dual boot |
23:44.13 | ManxPower | tuxinator_linux, I could do that, but I'd get mad all the time because my /home dir was not sync'd. |
23:44.32 | KalD|Work | tuxinator_linux, exactly why I run linux =) I can get 'cheap' copies of MS software but I still pay for it in terms of time and just pure screaming at the top of my lungs about how much I hate it over and over... or I can just run linux and be happy =) |
23:45.23 | tuxinator_linux | I just think it is silly that a program with so many bugs can cost so much and be so popular |
23:45.53 | tuxinator_linux | I do spend more time maintaining Windows machines that my linux ones |
23:45.55 | Nugget | it's popular because it does what people want, unlike linux which does what developers want. |
23:46.08 | tuxinator_linux | Nugget: true...ist |
23:46.11 | tuxinator_linux | ish |
23:46.40 | Nugget | my beef with linux is that it suffers from trying to do everything -- it ends up doing all things in mediocrity and no things well. |
23:46.44 | tuxinator_linux | You should hear my wife yell at Word when she has to use it at school |
23:46.45 | ruiner | linux can be a pain as well...especially when installing new hardware. with windows (usually) it just works |
23:46.49 | Nugget | it just isn't very good at things. |
23:47.01 | ruiner | but i do prefer linux over windows because i have a lot more control over things |
23:47.02 | Nugget | it's just adequate at doing just about everything |
23:47.14 | ruiner | but i use windows so i can play games |
23:47.26 | Darwin35 | ? |
23:47.28 | tuxinator_linux | Nugget: good at being stable, and being a server, but you're right not a perfect desktop OS yet |
23:47.33 | Nugget | I disagree. |
23:47.42 | Nugget | for stability or servers there are much better solutions. |
23:47.45 | *** part/#asterisk laloo (~laloo@042.142-60-66.FTTH-SWI.surewest.net) |
23:48.07 | tuxinator_linux | ruiner: I have not had any problews with installing hardware on linux, kudzu is nice |
23:48.07 | Darwin35 | /usr/ports/net/asterisk/files |
23:48.15 | Nugget | and much of the push to make linux more appropriate for desktop use are coming at the direct expense of stability and server tasks. |
23:48.23 | Darwin35 | thats all the current fbsd patches |
23:48.35 | ManxPower | tuxinator_linux, You kow who P. T. Barnum was, right? |
23:48.48 | tuxinator_linux | ManxPower: sorry |
23:49.11 | ManxPower | tuxinator_linux, he was am american showman. He is the one that said "A sucker is born every minute." |
23:49.28 | tuxinator_linux | ManxPower: Sounds like a cool guy |
23:49.33 | dwC- | is there any way to use DISA but without a dialtone, just dead air while waiting for user input? |
23:50.09 | stustu | Darwin35: Ok, I guess it will be possible to apply patch-Makefile to the CVS head Makefile. I thought that maybe a diff relative to a newer Makefile is floating around somewhere... |
23:50.14 | ManxPower | He is also the guy that opened a "freak show" type of traveling museum. It was so popular that people were not leaving quickly enough, so he put up a sign "This Way To The Egress", people followed the sign, found themselves outside and had to pay again to get back in. |
23:50.32 | tuxinator_linux | that's funny |
23:51.31 | ManxPower | tuxinator_linux, *nod* He's one of my minor heros. |
23:51.35 | Darwin35 | I just use th eports and now help maintain it |
23:51.53 | Darwin35 | it all works execpt for I have to update bri now |
23:53.33 | *** part/#asterisk impressmenicely (impressmen@host-66-81-63-177.rev.o1.com) |
23:53.46 | ruiner | later gents and gentettes |
23:55.17 | stustu | I really have a hard time finding out what's in the bristuff patch. Can I read about that somewhere (except examining the patch itself?) |
23:55.24 | tuxinator_linux | ManxPower: http://home.nycap.rr.com/useless/barnum/ |
23:56.25 | gr8nash | anyone have a page for VOIP providers.. i must have missed it on the wikki |
23:56.30 | gr8nash | im in the US |
23:57.01 | ManxPower | tuxinator_linux, Cool. |
23:57.21 | Lethargicclown | gr8nash, broadvoice looks good to me, but i would wait for an answer from someone more experienced then myself |
23:57.30 | SleepyCow | Hey guys. Does one get better sound quality from a PCI FXS card (Like the wild 4 port) than from an ethernet to pots bridge? |
23:57.51 | mishehu | SleepyCow: Yes. |
23:58.00 | mishehu | SleepyCow: No. |
23:58.05 | gr8nash | Lethargicclown they looked ok to me to..but alot of people have not-so-nice of things to say about them |
23:58.07 | SleepyCow | mu? |
23:58.10 | mishehu | SleepyCow: I guess the answer is "maybe" |
23:58.16 | mishehu | the kow says "mu" |
23:58.23 | SleepyCow | can you elaboratE? |
23:58.31 | SleepyCow | Cows say moo, kow says 'mu' |
23:58.40 | opus___ | yes |
23:58.41 | opus___ | no |
23:58.42 | opus___ | :) |
23:58.51 | modulus_ | jbot dogcow? |
23:58.52 | jbot | MOOOFF!! |
23:59.19 | opus___ | "Yeah, we're actually ... we don't have any passengers on board, so we decided to have a little fun and come up here," |
23:59.22 | opus___ | great idea |
23:59.24 | SleepyCow | Seriously, is there any advantage to using the 4 port fxs card right in the asterix box versus external pots to ethernet bridges? |
23:59.39 | JerJer[mobile] | yes |
23:59.42 | mishehu | jbot mog |
23:59.43 | SleepyCow | ? |