| 00:02.55 | NERvOus | kimosabe: not sure if I pasted everything you asked: http://pastebin.ca/3177 |
| 00:04.28 | kimosabe | i need xtensions.conf context |
| 00:04.52 | NERvOus | full extensions.conf ? |
| 00:04.53 | NERvOus | ok |
| 00:05.32 | NERvOus | the sip entry is inside the [default] context |
| 00:05.52 | kimosabe | nervous i have exactlly this and it works http://voxilla.com/modules.php?op=modload&name=News&file=article&sid=39 |
| 00:07.31 | *** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de) |
| 00:07.49 | NERvOus | callerid="SPA1" <2201> |
| 00:08.01 | kimosabe | chill i lookin at it |
| 00:08.14 | NERvOus | Since our SPA-2000 is only talking locally to our asterisk machine, |
| 00:08.14 | NERvOus | ; special consideration for NAT (Network Address Translation) is not needed. |
| 00:08.43 | NERvOus | there are no sip phones on public ip addresses |
| 00:08.46 | NERvOus | they are local |
| 00:08.48 | NERvOus | that's why it works |
| 00:08.54 | kimosabe | nervous look at the context you have no context on sip.con for sips |
| 00:08.57 | kimosabe | context=home |
| 00:09.10 | kimosabe | context= what ever context ias in extensions |
| 00:09.17 | kimosabe | i didnt see extensions.conf |
| 00:10.26 | kimosabe | nervous paste bin extensions context |
| 00:10.35 | NERvOus | ok |
| 00:11.07 | PTG123 | anyone here use an iaxy? |
| 00:11.14 | PTG123 | or an iax softphone? |
| 00:11.32 | *** join/#asterisk D1ng0 (~dingo@130.205.8.67.cfl.rr.com) |
| 00:12.43 | kimosabe | does any one a good sip software that is open source |
| 00:12.45 | NERvOus | http://pastebin.ca/3178 |
| 00:13.04 | *** join/#asterisk mindCrime (~mindCrime@rrcs-24-106-188-6.se.biz.rr.com) |
| 00:13.20 | NERvOus | I'm using linphone as a sip client |
| 00:13.23 | Druken | PTG123: apparently my hardphone will do iax2 |
| 00:13.26 | Druken | but i use sip |
| 00:13.40 | PTG123 | really: which one is it? |
| 00:13.46 | PTG123 | kimosabe: you mean a softphone? |
| 00:14.03 | kimosabe | yes |
| 00:14.05 | Druken | PTG123: the VST-1000 from pcphoneline.com |
| 00:14.11 | PTG123 | there isn't one, why not iax2? |
| 00:14.14 | Druken | kinda a crappy phone... but it works |
| 00:14.21 | kimosabe | can i change setings on it to make it work with my server |
| 00:14.35 | PTG123 | kimosabe: you can do that without source, what do you mean? |
| 00:14.46 | PTG123 | yah i don't even see the need for a hardphone myself :) |
| 00:15.00 | PTG123 | i'll stick with me $70 usb headset and a good softphone |
| 00:15.10 | kimosabe | i have only used sipuras but i want software becuase i just sold my last sip |
| 00:15.10 | Druken | PTG123: honestly... i would never use a computer and headset |
| 00:15.26 | PTG123 | why is that druken? |
| 00:15.29 | kimosabe | need 5 day solution |
| 00:15.35 | PTG123 | kimsodabe: so you want one just to use, then use x-lite |
| 00:15.45 | Druken | if the computer crashes, so does your phone call |
| 00:15.46 | kimosabe | thanks |
| 00:15.51 | kimosabe | oki |
| 00:16.19 | PTG123 | druekn: true, had that happen twice, but in general my pc never crashes |
| 00:16.25 | zno | Drunken: the satisfaction of hanging up on someone is not the same as with a hard phone either |
| 00:16.28 | *** join/#asterisk Kevorkian (~chris@ool-18b9b958.dyn.optonline.net) |
| 00:16.42 | PTG123 | i love headsets though, and the quality of a computer headset is much better then that of a phone in general |
| 00:16.54 | Druken | zno: exactly |
| 00:17.23 | PTG123 | you could slam down your hands on the keyboard |
| 00:17.25 | PTG123 | that may be more fun too |
| 00:17.35 | Druken | doubt it.. hehehe |
| 00:17.43 | Druken | i dun have a single softphone... |
| 00:18.00 | Druken | all my stuff is hardware.... |
| 00:18.01 | PTG123 | www.xten.com kimosabe |
| 00:18.10 | PTG123 | i use my phone around 5 hours a day |
| 00:18.15 | *** join/#asterisk Yoda-BZH (~yoda-bzh@pro.game.admin.at.yoda-bzh.net) |
| 00:18.15 | PTG123 | i like the softphone :) |
| 00:18.52 | Druken | my phones ring from 8-6 daily |
| 00:18.56 | MrBelvedr | heh |
| 00:18.56 | zno | we need some wifi sip headsets on the market |
| 00:19.14 | Druken | we need some wifi cell phones :) |
| 00:19.28 | arrgh | that don't suck |
| 00:19.43 | PTG123 | maybe a little windows ce phone |
| 00:19.46 | PTG123 | w/ wifi |
| 00:19.56 | Druken | i'd setup a wifi network if i could take the phone around with me... |
| 00:20.00 | arrgh | there is for the iPaq |
| 00:20.06 | kimosabe | nervous |
| 00:20.09 | zno | PTG123: can't you already do that with a pocketpc? |
| 00:20.14 | kimosabe | u still there how many sip u have |
| 00:20.16 | Druken | i'm waiting for this new thing to come out... that apparently can do wireless hi-speed |
| 00:20.18 | zno | compactflash card |
| 00:20.24 | PTG123 | yah never tried th ough, i am not even sure if my ipaq has a mic input jack |
| 00:20.32 | Druken | then i'll be buying the first SIP call phone based on that |
| 00:20.42 | Druken | S/call/cell |
| 00:21.16 | PTG123 | i can't figure out why my iax client isn't get the ring event |
| 00:21.36 | Druken | is it setup as a friend? |
| 00:21.54 | PTG123 | yes |
| 00:21.58 | NERvOus | kimosabe: 1 for now |
| 00:22.00 | PTG123 | i think my actual program is rejecting it |
| 00:22.03 | NERvOus | kimosabe: trying to call 600 |
| 00:22.50 | arrgh | really |
| 00:22.59 | Druken | you have 1 sip phone... |
| 00:23.04 | arrgh | a tad bit more widespread wifi coverage (depending on where you live) |
| 00:23.11 | arrgh | coupled with a decent wifi phone |
| 00:23.15 | *** join/#asterisk Jammy (~jammy@24.244.183.75) |
| 00:23.16 | Druken | so how exactly are you going to make a call with only 1 phone? |
| 00:23.20 | kimosabe | nervous here you go http://www.pastebin.com/131387 |
| 00:23.36 | NERvOus | drray: I call 600 |
| 00:23.39 | kimosabe | hehehe cant make a call with one soft phone |
| 00:23.42 | arrgh | coupled with a local or inexpensive pstn provider, could *really* cut into the LEC's business |
| 00:23.42 | NERvOus | Druken: I call 600 |
| 00:23.50 | Druken | and that does? |
| 00:23.54 | NERvOus | and do an echo test |
| 00:24.02 | NERvOus | you talk, you hear you back |
| 00:24.07 | NERvOus | I tested it with IAX |
| 00:24.10 | NERvOus | and it works |
| 00:24.14 | NERvOus | the problem is with SIP |
| 00:25.08 | NERvOus | http://pastebin.ca/3178 --> lines 133 to 136 |
| 00:25.34 | NERvOus | kimosabe: that link doesn't help |
| 00:26.22 | Druken | NERvOus: in your sip.conf do you have a context = demo ? |
| 00:26.46 | NERvOus | no context=default |
| 00:27.36 | Druken | ok, i see that |
| 00:27.38 | Druken | however |
| 00:27.38 | NERvOus | [default] includes [demo] |
| 00:27.45 | Druken | it's in the wrong damn place :) |
| 00:27.52 | Druken | put it under your account |
| 00:27.55 | NERvOus | ok |
| 00:27.56 | Druken | not under general |
| 00:28.16 | NERvOus | perfect! |
| 00:28.19 | NERvOus | thanks a lot! |
| 00:28.26 | Druken | tee hee |
| 00:28.54 | NERvOus | I need to re-read the doc, I thought that settings underl [general] were applied to each account |
| 00:29.10 | NERvOus | sort of global settings |
| 00:29.25 | Druken | no... general and global are totally diffrent |
| 00:29.47 | Druken | but.. it's always good practice to have a context line in every account |
| 00:29.57 | NERvOus | k |
| 00:30.10 | Druken | allows for much better control and security |
| 00:30.23 | *** join/#asterisk bonbon-home (~happy@81-86-185-223.dsl.pipex.com) |
| 00:31.23 | bonbon-home | there was a cvs update to chan_iax2.c where the log message is "make sure read/write format are setup properly". Anyone know what issue this fix addresses/ |
| 00:31.24 | bonbon-home | ? |
| 00:32.16 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 00:34.03 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 00:35.21 | *** part/#asterisk NERvOus (nervous@host43-119.pool80116.interbusiness.it) |
| 00:35.36 | *** join/#asterisk owh (~onno@OptusSatelliteServices.22bjc76f09.optus.net.au) |
| 00:38.14 | owh | bkw_: Sorry to bother, but I am unable to locate the patch you speak of for the polling frequency for the register command. |
| 00:39.21 | owh | kram: Sorry, can't make it to the meeting, I'm in Australia today :-) |
| 00:40.07 | dudewhere | I'm using -v1-0-12/13/04-18:46:23 and I keep getting errors: CallerID returned with error on channel 'Zap and Got event 17 (Polarity Reversal)... Is it my vesion? What else can I be doing wrong? I'm using a TDM400p |
| 00:41.28 | ManxPower | dudewhere: Your line is weird or you set zapata callerid for polarity reverasl |
| 00:41.35 | PTG123 | anyone here us iaxclient? :) |
| 00:42.54 | *** join/#asterisk owh (~onno@OptusSatelliteServices.22bjc76f09.optus.net.au) |
| 00:43.14 | *** join/#asterisk asterisksip (~asterisks@me-lewiston-pw-110.mint.adelphia.net) |
| 00:44.21 | *** join/#asterisk quickmoney (~jfu2808@CPE00a0c5e1b8b3-CM013010000950.cpe.net.cable.rogers.com) |
| 00:44.21 | dudewhere | ManxPower: Assuming I set my callerid for polarity reversal? What can I do to fix it? |
| 00:44.52 | ManxPower | /etc/asterisk/zapata.conf |
| 00:47.22 | *** join/#asterisk tzanger (~tzanger@165.154.13.35) |
| 00:47.23 | *** join/#asterisk sivana (~richard@northbay-pppoe-32.vianet.ca) |
| 00:51.35 | twisted | THIS IS ABSOLUTE BULLSHIT |
| 00:51.48 | PTG123 | what? |
| 00:52.00 | twisted | http://www.astricon.net/training/dcap.shtml |
| 00:52.44 | srt | lol |
| 00:52.46 | twisted | the ONLY way to be considered certified is to either pay over $3000 US or to attend astricon and pay for the damn test |
| 00:52.47 | *** join/#asterisk mofu (~eric@pcp04232064pcs.plyntv01.mi.comcast.net) |
| 00:52.53 | twisted | ABSOLUTE BULLSHIT |
| 00:52.58 | PTG123 | hahaha |
| 00:52.59 | PTG123 | haha |
| 00:53.00 | srt | who had that idea? |
| 00:53.09 | twisted | good ole steve sokol and olle |
| 00:53.15 | twisted | but digium is in on it too |
| 00:53.18 | PTG123 | why do you need to be certified? |
| 00:53.30 | *** join/#asterisk snewpy (~markl@203-217-38-118.dyn.iinet.net.au) |
| 00:53.30 | twisted | i have not had a single problem with anything that has been done with this until now |
| 00:53.32 | denon | because the cert costs money |
| 00:53.53 | PTG123 | heh |
| 00:53.58 | PTG123 | thats pretty unbelievable :) |
| 00:54.03 | twisted | denon, the money goes to steve and olle |
| 00:54.04 | srt | i dont think that will work well |
| 00:54.20 | NormAst | <PROTECTED> |
| 00:54.24 | twisted | This is almost enough to make me pack up and fucking walk out on the project |
| 00:54.26 | denon | twisted: so why is digium supporting it? |
| 00:54.36 | *** part/#asterisk darby_t (~tom@dne202.neoplus.adsl.tpnet.pl) |
| 00:54.41 | PTG123 | it sort of goes against the whole opensource concept :) |
| 00:54.46 | twisted | sort of?! |
| 00:54.51 | owh | Has anyone here had some experience with high cost high latency links for iax2 trunks? |
| 00:54.51 | denon | so email mark |
| 00:55.17 | PTG123 | heheh |
| 00:55.23 | twisted | this means, as a contributer, and someone who has been involved with this project for a long time now, i'm not considered good enough unless i get a certification |
| 00:55.51 | *** join/#asterisk blackdog (mwotton@pill01.orchestra.cse.unsw.EDU.AU) |
| 00:55.51 | twisted | who determined that steve sokol and olle are good enough to train anyone and certify them anyway? |
| 00:55.53 | PTG123 | well i am sure they would give in and give you one for free :) But its the others that will be left in the cold :) |
| 00:55.55 | sumasuma | ~twisted |
| 00:55.56 | jbot | it has been said that twisted is twisted@indigent-networks.com, but you can paypal him at toastido@toastido.net |
| 00:55.59 | srt | that depends on who believes in this certification stuff |
| 00:56.11 | twisted | srt, and in the buisness world, that's a BIG DEAL |
| 00:56.17 | owh | twisted: I don't know you or your contribution, but it sounds like you're upset. Perhaps taking a big deep breath and asking whomever instigated the idea... |
| 00:56.32 | twisted | owh, fuck that.. |
| 00:56.38 | twisted | owh, talking gets nowhere in this project |
| 00:56.46 | owh | twisted :-) |
| 00:56.47 | PTG123 | get rialed up :) |
| 00:56.48 | twisted | i've been involved long enough to know that |
| 00:57.17 | PTG123 | you could fork off like xfree ;) |
| 00:57.21 | srt | i dont know of any certs for open source software that anybody in the business wold cares about |
| 00:57.23 | owh | heh |
| 00:57.32 | owh | srt: Give it time. |
| 00:57.47 | srt | all this redhat/... cert stuff didnt work well |
| 00:58.01 | srt | why should this be different with digium |
| 00:58.02 | file[laptop] | twisted" you're irritated too? |
| 00:58.16 | file[laptop] | let's go stab people |
| 00:58.34 | snewpy | twisted: unautoignore me so i can send you a private message :) |
| 01:00.04 | PTG123 | you could forge something out of an asterisk server you could use for stabbing |
| 01:00.11 | owh | Anyone played with VoIP over two-way satellite around here? |
| 01:01.33 | CompWizrd | owh: ouch. |
| 01:01.44 | mofu | sorry missed it, what are they trying to limit to "Certified" techs, contributing back to main project? |
| 01:01.49 | owh | CompWizrd: What makes you say that :-) |
| 01:01.52 | CompWizrd | latency |
| 01:02.02 | CompWizrd | actually, i noticed my cell phone has crazy latency as well |
| 01:02.07 | PTG123 | its a certification, like you saying you are cisco certified |
| 01:02.08 | owh | CompWizrd: Hey, 800-1200ms isn't that bad :-) |
| 01:02.17 | *** join/#asterisk evilbuny (~evilbuny@192-172-93-202.dsl.nbdsl.net) |
| 01:02.35 | owh | CompWizrd: Besides, I'm mobile, so I can have connectivity anywhere :-) |
| 01:03.52 | CompWizrd | i wonder if there's a way to get VoIP on my blackberry 7510 |
| 01:03.57 | CompWizrd | since data costs me nothing, but voice does |
| 01:04.18 | owh | CompWizrd: Install Debian on it and you can play... |
| 01:04.36 | CompWizrd | i don't think i can put debian on that, can i? |
| 01:05.07 | CompWizrd | either way, i need the phone for work |
| 01:05.14 | CompWizrd | now if there was a java voip thingy, i could run that |
| 01:05.25 | owh | CompWizrd: Let work pay for it... |
| 01:05.30 | CompWizrd | yeah, work does pay for it |
| 01:05.33 | CompWizrd | they got me it a few months ago |
| 01:05.43 | CompWizrd | i'm deaf, so the email/etc is very handy to have |
| 01:05.46 | owh | CompWizrd: I think there is a Java VoIP thingy, but I forget its name. |
| 01:05.51 | CompWizrd | and makes it far easier to get ahold of me |
| 01:06.04 | *** join/#asterisk menger (~menger@static-88.243.240.220.dsl.comindico.com.au) |
| 01:06.07 | owh | CompWizrd: So, what point is there to having VoIP on your blackberry then? |
| 01:06.18 | CompWizrd | owh: save my company money, and b) see if it can be done ;) |
| 01:06.33 | owh | CompWizrd: But if you cannot hear it, what is the point? |
| 01:06.42 | CompWizrd | oh, i see your point there.. i'm not completely deaf |
| 01:06.51 | CompWizrd | 50db loss at 250-1000hz, 130+ db loss after that |
| 01:06.58 | owh | CompWizrd: You're better of getting some voice recognition stuff happening and making it into email :-) |
| 01:07.11 | CompWizrd | yes, that's something i'd love to see done |
| 01:07.15 | owh | CompWizrd: SO IF I SHOUT, you can hear me better :-) |
| 01:07.29 | CompWizrd | where it'd packet up the voice that it hears, and it sends it off to a server over the data link, the server turns it to text, and sends it back on my phone |
| 01:07.48 | CompWizrd | that would be a killer app for the deaf |
| 01:08.08 | owh | CompWizrd: Well, IBM has released their voice recognition code, so it will happen some day. |
| 01:08.20 | dca | i'm getting a lot of these warnings "Dec 19 18:06:56 WARNING[20095]: chan_iax2.c:5051 socket_read: Error: Resource temporarily unavailable" any ideas? |
| 01:08.36 | dca | also, i just install a t100p and now i have not audio... |
| 01:08.39 | CompWizrd | my employer would pay a lot of money to have something work perfectly like that |
| 01:08.41 | owh | dca: No idea, but is your connection intermittent? |
| 01:08.46 | twisted | i think, if there isn't a resolution within 10 buisness days |
| 01:08.48 | twisted | i'm folding. |
| 01:08.49 | CompWizrd | or put it on a heads up display that i wear or something |
| 01:09.05 | *** join/#asterisk bofh42 (~bofh42@pD9EC34DF.dip.t-dialin.net) |
| 01:09.08 | dca | owh, no it's a dedicated 100mb line |
| 01:09.28 | owh | twisted, there is of course that other event happening within that time-line, so perhaps it might take a little longer... |
| 01:09.35 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
| 01:09.42 | twisted | owh, what other event? |
| 01:09.47 | harryvv | Anyone here running debian? |
| 01:09.47 | owh | dca: As I said, dunno, but I'm just throwing ideas at you... |
| 01:09.57 | owh | twisted, you know, December 25th etc. |
| 01:09.58 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 01:10.03 | owh | harryvv: Might be :-) |
| 01:10.03 | twisted | owh, ten buisness days |
| 01:10.06 | twisted | BUISNESS DAYS |
| 01:10.10 | twisted | not ten calendar days |
| 01:10.39 | owh | twisted, my point is that every company deals with that date differently. Some stop on Wednesday, others work through the whole lot. |
| 01:10.54 | twisted | owh, stop nit picking and go home |
| 01:10.57 | harryvv | twisted, what sources.list do you use to download *? |
| 01:11.00 | owh | harryvv: Ok, I give in - yes, I'm running debian. |
| 01:11.04 | *** join/#asterisk PBXtech (~nik@65-103-224-165.slkc.qwest.net) |
| 01:11.10 | twisted | harryvv, i don't run shit distro |
| 01:11.27 | twisted | harryvv, i download * from cvs |
| 01:11.28 | owh | twisted, I'm not, I'm just trying to assist in expectation management :-) |
| 01:11.30 | mofu | harryvv: its in normal unstable apt sources |
| 01:11.40 | PBXtech | why am i getting this error: Found no matching peer or user for '65.103.xx.xx:64568' when i set it to "friend" it works fine fails on "peer" |
| 01:11.42 | ManxPower | Everyone with a working asterisk downloads it from CVS |
| 01:11.53 | mofu | not sure about stable or testing |
| 01:11.53 | harryvv | twisted never used cvs |
| 01:12.43 | owh | mofu: You have a point there, lemmie look where mine came from... |
| 01:13.00 | PBXtech | and i get these also: SIP/2.0 481 Call Leg Does Not Exist |
| 01:13.41 | owh | mofu: I'm using 1.0.2-2 from testing I believe... |
| 01:13.54 | mofu | * 1.0.2 in Testin and Unstable, 0.1.11-3 in Stable . . . |
| 01:14.54 | *** join/#asterisk wazzasp (~d@81.196.252.98) |
| 01:15.15 | wazzasp | hello ppl |
| 01:15.49 | dca | anyone know what this means? Dec 19 18:06:56 WARNING[20095]: chan_iax2.c:5051 socket_read: Error: Resource temporarily unavailable |
| 01:16.09 | CompWizrd | i http://www.backports.org/debian/dists/stable/asterisk/ for asterisk under debian stable, as well |
| 01:16.11 | bkw_ | kram dude you're fuckin up |
| 01:16.18 | bkw_ | BOYCOTT!!! |
| 01:16.24 | CompWizrd | 1.0.1-0 |
| 01:16.26 | bkw_ | I AM NOT PAYING TO BE DIGIUM certified |
| 01:16.30 | bkw_ | thats pure bullshit |
| 01:16.31 | bkw_ | PURE |
| 01:16.48 | owh | bkw_: You should talk to twisted about that :-) |
| 01:16.54 | bkw_ | don't worry |
| 01:16.59 | bkw_ | I have already voiced my opinion |
| 01:17.01 | bkw_ | to the people |
| 01:17.03 | wazzasp | im trying to make an extension to call me back...need help |
| 01:17.28 | ManxPower | wazzasp: You mean like the one on the WFFS? |
| 01:17.31 | owh | bkw_: I looked for the patch you suggested for the register polling frequency, but my search abilities must not include finding things. |
| 01:17.32 | jradford | what exactly are we supposed to boycott? |
| 01:17.35 | ManxPower | Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ |
| 01:17.39 | bkw_ | owh let me see |
| 01:17.42 | ManxPower | look in the downloads/abandones-stuff |
| 01:18.19 | bkw_ | owh http://bugs.digium.com/bug_view_page.php?bug_id=0003082 |
| 01:18.33 | owh | jradford: I think we need to boycott M&M's, but only red ones. |
| 01:18.36 | owh | bkw_: Tah |
| 01:18.39 | wazzasp | what is WFFs :) |
| 01:18.54 | wazzasp | lol |
| 01:18.54 | jradford | owh: I'll gladly do so if you can show me something better :) |
| 01:19.18 | owh | jradford: Well the yellow ones work well. |
| 01:19.23 | ManxPower | wazzasp: The World Famous Fnords Site. It should be in your bookmarks |
| 01:19.30 | jradford | but they are not equal to the red ones |
| 01:19.49 | ManxPower | If it's not then you can bookmark the list of URLS that says "BOOKMARK THEM!" next to them |
| 01:19.55 | dca | peculiar problem, i just installed a t100p on a debian box with 2.6 kernel, moprobed successfully, but it cut off the audio. then i modprobe -r and all is well....what in the world is that about? |
| 01:20.05 | wazzasp | oky let me have a look..im always using voip-info.org |
| 01:20.24 | hmodes | err, oy |
| 01:20.37 | ManxPower | wazzasp: Notice the callback stuff is in the abandoned-stuff directory...so don't ask me questions about it. |
| 01:20.54 | wazzasp | ok thx |
| 01:20.56 | wazzasp | :) |
| 01:21.03 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-60.client.comcast.net) |
| 01:22.39 | owh | jradford: I can supply some paint remover and some red paint if you like. I'll throw in a small brush. |
| 01:22.54 | jradford | well thats it then, I'll give up the red m&m's |
| 01:23.01 | mofu | owh, where do I send the red ones for a refund? |
| 01:23.15 | owh | mofu: I'd contact the supplier. |
| 01:23.58 | owh | bkw_: I looked at that, it looks to be a sip register, not a trunk register. I might be completely missing the point and ignorant, so be gentle :-) |
| 01:25.02 | bkw_ | owh thats for sip |
| 01:25.03 | bkw_ | not iax |
| 01:25.38 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
| 01:25.48 | owh | But the register that is causing me grief is in iax.conf, so am I just being dumb? |
| 01:27.04 | ManxPower | bkw_: He's using the debian version of Asterisk |
| 01:27.48 | owh | ManxPower: Does that make a difference (other than the version)? |
| 01:28.15 | ManxPower | owh: We have no idea what changes the packager did. |
| 01:28.40 | ManxPower | There is at least one issue with a packager naming one of the asterisk packages the same as a TOTALLY different branch in CVS |
| 01:28.46 | NormAst | Why do I get Ring Requested on channel 0/1 already in use on span 1. Hangup owner? Anyone |
| 01:29.09 | owh | kram: Sorry, can't make it to the meeting, I'm in Australia today. |
| 01:29.14 | kram | sokay |
| 01:29.40 | owh | ManxPower: The debian change-log just moves a few files around AFAIK. |
| 01:30.00 | owh | ManxPower: Are you talking about gtk-console.so perhaps? |
| 01:30.06 | ManxPower | Specifically the Debian asterisk-sounds package is NOT the asterisk-sounds in CVS. |
| 01:30.49 | owh | Understand, that would piss people off :-) |
| 01:31.28 | ManxPower | some poor sod wasted a couple of days because of that. |
| 01:31.38 | owh | ManxPower: I can just imagine. |
| 01:32.06 | harryvv | owh what card do you run |
| 01:32.18 | owh | ManxPower: I had a similar affair with gtk-console, the solution was to download the debian voip maintainers list and search through the archive. |
| 01:32.26 | asterisksip | anyone connect their asterisk box to a h323 service? |
| 01:32.27 | owh | harryvv: None, only ATA-186's around here. |
| 01:32.38 | harryvv | k |
| 01:32.47 | owh | ManxPower: Suffice to say, I lodged a bug report :-) |
| 01:34.18 | wazzasp | ManxPower: i whant the extension to hangup imidiatly adn to call me back, withouth any promts.. |
| 01:34.38 | *** join/#asterisk Kevorkian (~chris@ool-18b9b958.dyn.optonline.net) |
| 01:35.11 | *** join/#asterisk owh (~onno@OptusSatelliteServices.22bjc76f09.optus.net.au) |
| 01:35.29 | owh | ManxPower: While this is likely not your problem, can you give me a specific example of a missing sound that is in the current CVS version and not in Debian, or point me to a list of the sounds? |
| 01:35.37 | ManxPower | wazzasp: Then I guess you have some programming to do. |
| 01:35.44 | wazzasp | yeah |
| 01:35.45 | wazzasp | :) |
| 01:35.48 | wazzasp | lol |
| 01:35.49 | wazzasp | thx |
| 01:36.28 | ManxPower | owh: The debian maintainer took all the sound files out of Asterisk ansd put them in a package named asterisk-sounds. HOWEVER, there is already an asterisk-sounds CVS for additional sounds not part of the asterisk download |
| 01:37.19 | owh | The package is now called asterisk-sounds-main, so I suppose there could be made a case for asterisk-sounds-extra if I understand what you are telling me. |
| 01:37.34 | ManxPower | owh: Yes. |
| 01:37.59 | ManxPower | A perfect example of a packager doing stuff nobody expected. |
| 01:38.36 | owh | I understand. |
| 01:38.56 | ManxPower | The ONLY software I don't use RPMS for is Asterisk and Spamassassin, since both change so fast that no package will be up to date. |
| 01:39.07 | *** join/#asterisk drumkilla (~russelb@pcp03785394pcs.mplsnt01.sc.comcast.net) |
| 01:39.07 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 01:39.12 | ManxPower | And each package is the first version of the package. No time to fix packaging bugs |
| 01:39.25 | wazzasp | dose enyone used sox aplication to convert to .gsm files? i have monster voices when i use taht aplication |
| 01:39.26 | wazzasp | lol |
| 01:39.39 | twisted | drumkilla, wassabi! |
| 01:39.40 | jradford | monster voices rule! |
| 01:39.42 | PTG123 | monster voices is a feature not a b ug :) |
| 01:39.51 | drumkilla | twisted: hey bud |
| 01:39.52 | owh | ManxPower: Are we talking about the same thing. Debian stable is ancient, but Debian testing and Debian unstable are at 1.0.2 |
| 01:40.07 | jradford | wazzasp: did you follow the directions in the wiki for converting sounds? |
| 01:40.23 | ManxPower | owh: 1.0.3 was released a couple of weeks ago and has massive numbers of bug fixes. |
| 01:40.30 | owh | ManxPower: Would you consider the missing sounds a bug or a wishlist? |
| 01:40.33 | ManxPower | Un faxt post 1.0.3 CVS already has lots of bug fixes. |
| 01:40.39 | wazzasp | i guess not..i just used teh instruction witch came with the software |
| 01:40.40 | wazzasp | :) |
| 01:40.46 | ManxPower | owh: The "missing sounds" are just wrongly named. |
| 01:41.12 | owh | My understanding from what you wrote is that they are not available at all... |
| 01:41.55 | ManxPower | It would be like a packager calling firefox the package "mozilla-mail" |
| 01:42.42 | PTG123 | oh the fighting as began about the cert on the mailing list :) |
| 01:43.04 | tessier_ | Go bkw! |
| 01:43.38 | owh | But if I understand correctly, you're saying that the release version of Asterisk contains the sounds as part of the release and that the Debian version contains a separate package for the sounds. In addition, there are extra sounds that are available for Asterisk that are not packaged. Is that what you are saying? |
| 01:43.51 | tessier_ | Never use the debian version of anything. |
| 01:44.00 | tessier_ | It is likely to be way out of date and way fsckd up. |
| 01:44.23 | owh | Hmm, tessier_, that blanket statement does not make sense to me. |
| 01:44.26 | *** join/#asterisk bprice20 (~brandon@alb-69-200-20-166.nycap.rr.com) |
| 01:44.33 | tessier_ | owh: It has been my experience. |
| 01:44.54 | bprice20 | is someone willing to help me with what I believe is an isuue with my iax.conf |
| 01:44.55 | tessier_ | If it is a mature stable project that is a standard part of a Linux distro the debian package is probably ok |
| 01:45.07 | owh | tessier_: Fortunately, that has not been mine. I installed Asterisk on Friday and was making calls on Saturday. |
| 01:45.08 | tessier_ | But if it is something like asterisk get it from asterisk cvs |
| 01:45.27 | tessier_ | You could make calls with a version of asterisk from two years ago. |
| 01:46.22 | owh | tessier_: My install was: apt-get install asterisk |
| 01:46.48 | bprice20 | I can place a call via my asterisk server, but the clients get no sound |
| 01:46.52 | owh | tessier_: I added my two ATA-186's to the config files ans was good to go. |
| 01:46.53 | bprice20 | I get Accepting AUTHENTICATED call from 69.200.31.34, requested format = 6, actual format = 6 |
| 01:46.53 | bprice20 | <PROTECTED> |
| 01:46.53 | bprice20 | <PROTECTED> |
| 01:47.18 | bprice20 | yet no audio from the phone connected to an iaxy or an iax2 client firefly |
| 01:47.30 | tessier_ | bprice20: NAT involved anywhere? |
| 01:47.48 | jradford | bprice: do you have the volume turned up? |
| 01:48.05 | *** join/#asterisk owh (~onno@OptusSatelliteServices.22bjc76f09.optus.net.au) |
| 01:48.39 | bprice20 | tessier no |
| 01:48.46 | owh | bprice20: The hitchikers guide talks about answering every call first, did you? <- Note, that I've been playing with Asterisk for a grand total of 48 hours. |
| 01:49.40 | jradford | 48 hours and you tried to change my M&M colors? |
| 01:50.01 | owh | jradford: I've been playing with M&M's for much longer :-) |
| 01:50.36 | owh | ManxPower: Did I understand you correctly? |
| 01:50.46 | jradford | well then I'll trust your judgement :) |
| 01:50.47 | bprice20 | owh I basically set up an extension that does this: |
| 01:50.50 | bprice20 | exten => 9999,1,VoicemailMain |
| 01:50.50 | bprice20 | exten => 9999,2,Hangup |
| 01:50.56 | bprice20 | thats correct right? |
| 01:51.12 | owh | Shouldn't there be an answer in there first? |
| 01:51.34 | *** join/#asterisk owh (~onno@OptusSatelliteServices.22bjc76f09.optus.net.au) |
| 01:51.45 | owh | FSK, I wish it would stop doing that... |
| 01:51.53 | owh | bprice20: Shouldn't there be an answer in there first? |
| 01:52.04 | bprice20 | Its shows from the cli that it plays accepts the call and plays vm-login |
| 01:52.27 | owh | bprice20: My reading suggests that the audio goes nowhere until the call is answered. |
| 01:53.14 | bprice20 | ok I added an answer trying it now |
| 01:54.03 | owh | Hmm, the example configs don't actually have an answer for the voice mail extension at 8500... |
| 01:54.15 | bprice20 | mine didn't either |
| 01:54.21 | bprice20 | i didn't think it was needed |
| 01:54.28 | bprice20 | that didn't work anyway |
| 01:54.31 | owh | As jradford already observed, owh knows nothing about M&M's... |
| 01:54.41 | owh | Uh, Asterisk :-) |
| 01:55.19 | owh | bprice20: Does 8500 still work? |
| 01:55.29 | jradford | bprice: did you start with the demo before you started making changes to your configs? |
| 01:56.09 | bprice20 | same problem |
| 01:56.18 | owh | On the phone.... |
| 01:56.22 | jradford | I only use IAX for inbound, however something looks funny here: bprice20> -- Executing VoiceMailMain("IAX2/firefly@firefly/5", "") in new stack |
| 01:57.06 | bprice20 | I firefly is the user i created for use in testing |
| 01:57.12 | bprice20 | the iax2 client firefly |
| 01:57.24 | bprice20 | how SHOULD that look |
| 01:57.40 | jradford | have you tried to use Playback(soundfile) to see if it plays a sound to your client? |
| 01:57.40 | bprice20 | I basically created a friend firefly w/ extension 5003 |
| 01:58.16 | bprice20 | yes I tries Playback(goodbye) & hangup |
| 01:58.19 | bprice20 | still no sound |
| 01:58.31 | bprice20 | tried (oops) |
| 01:59.09 | bprice20 | exten => #,1,Playback(demo-thanks) --doesn't play sound |
| 01:59.12 | jradford | if you've got a minute I'll go grab firefly and see if I can make it work real quick on my already running system |
| 01:59.22 | bprice20 | it plays according to the cli but no audio on the phone |
| 01:59.34 | bprice20 | I'd be muy gratefull |
| 02:02.20 | bprice20 | make sure you set qualify=no for that extesion in iax.conf according to voip viki |
| 02:02.23 | bprice20 | wiki |
| 02:02.46 | bprice20 | [firefly] |
| 02:02.46 | bprice20 | type=friend |
| 02:02.46 | bprice20 | host=dynamic |
| 02:02.46 | bprice20 | secret=coblp2 |
| 02:02.46 | bprice20 | context=default |
| 02:02.46 | bprice20 | regexten=5003 |
| 02:02.48 | bprice20 | callerid="Brandon IAX Client"<(555) 555=5555> |
| 02:02.50 | bprice20 | trunk=no |
| 02:02.54 | bprice20 | qualify=no |
| 02:02.56 | bprice20 | sample |
| 02:05.55 | harryvv | is the tiget jet a digium x100p clone? |
| 02:08.48 | *** join/#asterisk IQ (~iq@70-57-182-12.omah.qwest.net) |
| 02:10.32 | nestAr | harryvv: some of them, yes. |
| 02:10.37 | nestAr | mine shows up as such |
| 02:11.39 | JerJer | grr |
| 02:11.47 | JerJer | trunk=no and qualify=no are the defaults |
| 02:11.50 | JerJer | so why both? |
| 02:11.59 | JerJer | er |
| 02:12.18 | bprice20 | dunno went off of a sample config |
| 02:14.27 | Gushi | Anyone can tell me what "cause 0" is in this sense? |
| 02:14.28 | Gushi | Dec 19 18:14:08 NOTICE[3873]: app_dial.c:803 dial_exec: Unable to create channel of type 'Zap' (cause 0) |
| 02:15.46 | blackdog | Is anyone working on another scripting interface to Asterisk? |
| 02:15.47 | harryvv | netstr, okay guy sent me what was advertised as a digium x100p and well dont know if its the same. it has asterisk documentation. |
| 02:16.00 | nestAr | probably some shady clone guy |
| 02:16.08 | nestAr | does the wcfxo module load? |
| 02:16.23 | PTG123 | blackdog: why od you ask? |
| 02:16.50 | harryvv | nestAr, whats the command? |
| 02:17.27 | bprice20 | nestAr modprobe wcfxo |
| 02:17.53 | blackdog | PTG123: because manually encoding priorities feels like coding in assembler. |
| 02:18.26 | blackdog | and i'd probably hack up something (however dodgy) rather than turn myself into a human compiler. :) |
| 02:18.44 | PTG123 | heh thats why i installed realtime, and made my own interface for automating it all :) |
| 02:19.09 | harryvv | bprice well that said cannot locate module wcfxo |
| 02:19.38 | nestAr | harryvv: sounds like you need to compile zaptel |
| 02:19.46 | nestAr | read the documentation |
| 02:19.52 | blackdog | PTG123: realtime? |
| 02:19.52 | nestAr | it's pretty easy |
| 02:20.52 | PTG123 | yes it load everything from a db live, no more conf files |
| 02:21.23 | blackdog | what's the advantage? |
| 02:21.37 | PTG123 | no reloading for changes for starters |
| 02:21.43 | *** join/#asterisk CaMeLL (~dave@alb-66-65-202-87.nycap.rr.com) |
| 02:22.09 | blackdog | reloading asterisk? It can't read its config files on the fly? |
| 02:22.18 | bprice20 | 20 bucks viap paypal in the morning to whomever can help me get audio working on my asterisk box |
| 02:22.22 | bprice20 | blackdog |
| 02:22.32 | bprice20 | yes it can run reload from an asterisk cli |
| 02:22.50 | harryvv | nestar, yea just apt-get installed zaptel but at the end it said make dev does not understand zaptel do it your self and it spit out the command lines plus some scripts. I guess i need to do those by hand. |
| 02:23.21 | nestAr | uhhh |
| 02:23.22 | nestAr | i dunno |
| 02:23.28 | nestAr | i compiled it |
| 02:23.30 | bprice20 | no takers on the 20 paypal bucks |
| 02:23.31 | nestAr | didn't install packaged |
| 02:23.34 | bprice20 | 30? |
| 02:23.42 | asterisksip | anyone connect their * box to providers via h323? |
| 02:23.44 | NormAst | 20 bucks... Free money.. |
| 02:24.10 | PTG123 | blackdog go back to sleep |
| 02:24.11 | PTG123 | :) |
| 02:24.32 | NormAst | bprice20: you can send me paypal bucks... :) |
| 02:24.36 | bprice20 | No I need help getting the now broken audio working, on my asterisk box |
| 02:25.06 | bprice20 | its showing the audio playing from the cli but its not actually playing |
| 02:25.16 | bprice20 | NormAst: can you help me |
| 02:25.16 | NormAst | Behind nat? |
| 02:25.25 | blackdog | PTG123: in the middle of the day? i'm lazy, but not _that_ lazy... |
| 02:25.27 | bprice20 | no neither the server no the client are |
| 02:25.53 | bprice20 | I'm using iax2 anyway i though that was nat OK'ed |
| 02:26.00 | jradford | bprice: I used the default settings shown in the wiki and it works perfectly 1st time. |
| 02:26.31 | bprice20 | you know what I'm gonna bacup my config and try the default settings from the wiki --again |
| 02:28.42 | harryvv | nestar, you run debian? |
| 02:28.47 | nestAr | yea |
| 02:29.13 | nestAr | my * box sez it's only 16 degrees outside.. |
| 02:29.15 | nestAr | BRRRRR |
| 02:29.56 | menger | is 17 here LOL |
| 02:30.15 | nestAr | cel? |
| 02:30.21 | nestAr | it's farenheit here |
| 02:30.22 | nestAr | :X |
| 02:30.31 | menger | cel |
| 02:30.36 | nestAr | that's like -1902389238 in cel |
| 02:31.37 | harryvv | you must be back east. |
| 02:31.54 | nestAr | yeah.. i'm in kentucky |
| 02:34.01 | *** part/#asterisk mblodget-home (~mblodget@c24.241.224.118.mad.wi.charter.com) |
| 02:35.29 | *** join/#asterisk crimd (~crimd@66-65-89-219.nyc.rr.com) |
| 02:36.39 | *** join/#asterisk jayden (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 02:37.27 | ZX81 | what is: == No one is available to answer at this time (1:0/0/0) ? |
| 02:37.43 | ZX81 | this is calling through a remote box of mine |
| 02:38.02 | zigman | thats cvs head right ? |
| 02:38.14 | *** join/#asterisk Moc (~mochouina@modemcable212.49-80-70.mc.videotron.ca) |
| 02:39.35 | zigman | at this time (%d:%d/%d/%d)\n", numlines, numbusy, numcongestion, numnochan); |
| 02:40.03 | zigman | use the source luke :P |
| 02:43.14 | *** join/#asterisk robert_wwl (~robert@pcp517130pcs.nash01.tn.comcast.net) |
| 02:44.00 | asterisksip | anyone with a SIPURA box wanna do some testing? need to make sure my * box allows connections outsdie of my network |
| 02:44.14 | ZX81 | :-) ta zigman |
| 02:44.40 | ZX81 | zigman: but surely that equates to one line, none busy... |
| 02:44.42 | ZX81 | :-) |
| 02:45.14 | PTG123 | man the scuttlebut was overwith before it started :) |
| 02:45.15 | PTG123 | heh |
| 02:45.22 | ZX81 | :-) |
| 02:45.55 | *** join/#asterisk crimd (~crimd@66-65-89-219.nyc.rr.com) |
| 02:47.28 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.215) |
| 02:47.33 | *** join/#asterisk ScottPBX (~email@ool-435776a6.dyn.optonline.net) |
| 02:47.57 | kimosabe | i just downloade x-lite softphone but i need one that i can modify the sip seting on and the server setings so that it can log in to my server |
| 02:47.57 | ScottPBX | hi. I'm kinda new at Voip can I ask you folks a stupid question? |
| 02:48.18 | kimosabe | scottpbs shure |
| 02:48.22 | PTG123 | can't be any stupider then the last :) |
| 02:48.37 | PTG123 | kim: do you not know how to set up a sip account? |
| 02:48.44 | kimosabe | shure |
| 02:48.47 | bprice20 | what does app_voicemail.c:3311 vm_execmain: Couldn't read username: mean when trying to run: |
| 02:48.50 | bprice20 | exten => 9999,1,VoicemailMain |
| 02:48.50 | bprice20 | exten => 9999,2,Hangup |
| 02:49.04 | ScottPBX | let's say I wanted to connect VOIP to PSTN...I'm doing this on Linux, Fedora core 2...is asterisk a solution I should consider? Is it even possible to do VOIP to PSTN with no providers? |
| 02:49.06 | PTG123 | well you don't need to modify anything, xlite asks for login pwrd and server address |
| 02:49.07 | kimosabe | ptg123 what do u need 2 know |
| 02:49.19 | *** join/#asterisk Hymie (hymie@L8R.NET) |
| 02:49.22 | kimosabe | scot yes |
| 02:49.27 | PTG123 | scott: yes you can do that no problem |
| 02:49.33 | bprice20 | ScottPBX you would probably need to get a t1 or some t1's or a t3 |
| 02:49.34 | PTG123 | scott; you just need a compatible card to do it |
| 02:49.35 | mofu | ScottPBX: you just need an FXO adapter and * |
| 02:49.42 | ScottPBX | We have covad |
| 02:49.44 | ScottPBX | dsl |
| 02:49.50 | kimosabe | scot a x100p card would also do |
| 02:49.54 | PTG123 | Scott: and normal analog phone line? |
| 02:50.02 | ScottPBX | right we have a landline |
| 02:50.05 | ScottPBX | analog. |
| 02:50.13 | ScottPBX | this would be separate. |
| 02:50.18 | ScottPBX | for long distance mostly. |
| 02:50.26 | kimosabe | scott what exally do u wish 2 implement inconection of branch offices |
| 02:50.34 | PTG123 | scott: yes you can do it, any reason your not thinking of just porting your phone to a voip provdier? |
| 02:50.47 | ScottPBX | yes, they don't want to do that. |
| 02:50.57 | PTG123 | just one phone line scott? |
| 02:51.12 | ScottPBX | they want to make a voip pbx, but I had read VOIP to PSTN couldn't be done without a provider. |
| 02:51.17 | ScottPBX | they want to skip the provider. |
| 02:51.23 | ScottPBX | yes one phone line. |
| 02:51.24 | kimosabe | scott just state what u are trying 2 acomplish |
| 02:51.55 | PTG123 | ok so you just need an fxo provider |
| 02:52.08 | ScottPBX | okay. |
| 02:52.32 | PTG123 | er card |
| 02:52.34 | bkw_ | oh I forgot about this http://bugs.digium.com/bug_view_page.php?bug_id=0001449 |
| 02:52.59 | file[laptop] | creepy |
| 02:53.26 | PTG123 | hey whats the c command to set a variable you can have the extensions.conf read? |
| 02:53.40 | ScottPBX | alright...fxo isn't it. I'd like to call a regular phone in brazil from a VOIP setup in the office. |
| 02:53.44 | ScottPBX | without a provider. |
| 02:53.56 | PTG123 | scott: yes it is it :) |
| 02:54.35 | ScottPBX | lol now I'm more confused. |
| 02:54.51 | NormAst | bkw_: do you know what Ring Requested on channel 0/1 already in use on span 1. Hang up owner is? |
| 02:55.05 | NormAst | After I get these it stops taking calls |
| 02:55.13 | *** join/#asterisk _E|nyPRI (E_nste_N@S0106000bdb97681e.wp.shawcable.net) |
| 02:56.17 | PTG123 | http://web.nulogictech.com/product_info.php?products_id=41&osCsid=d6ad65714ec77a80e924d0ed77bb90f0 |
| 02:56.19 | PTG123 | ah he left |
| 02:56.49 | crimd | he seemed slightly confused... :) |
| 02:56.55 | kimosabe | ptg123 do you know a good soft phone software that is free and that i can change seting on so that i can point it to my asterisk server |
| 02:57.02 | NormAst | bwk: On yea no 15 minutes answer here eh.. The hard stuff. |
| 02:57.15 | PTG123 | kimo: XLITE :) |
| 02:57.28 | PTG123 | it lets you change what you need |
| 02:57.32 | PTG123 | did you download from xten.com |
| 02:57.35 | kimosabe | i just downleaded it from earth link but i cant change setings on it |
| 02:57.43 | PTG123 | thats earthlinks version |
| 02:57.45 | PTG123 | hard coded for them |
| 02:57.49 | PTG123 | uninistall and go to xten.com |
| 02:57.50 | PTG123 | and install |
| 02:57.52 | crimd | I have the same question as Kimosabe, good linux business quality soft phone for linux? anyone? :) |
| 02:57.54 | kimosabe | oki |
| 02:58.10 | cypromis | business quality != softphone |
| 02:58.27 | PTG123 | actually softphones are higher quality, so you don't know what your talking about :) |
| 02:58.40 | crimd | Soft phones can be really good. |
| 02:58.43 | PTG123 | xpro running 729 with a high quality usb headset is gonna blow away most phones |
| 02:59.17 | kimosabe | ptg123 yes its the same site i donloaded from is there a link in particular |
| 02:59.20 | cypromis | hmmm maybe ulaw land phones |
| 02:59.51 | PTG123 | how are you going to get any higher quality then a direct digital to digital conversion? from a pc, which has 1000 times the processing power of a ip phone? |
| 03:00.17 | crimd | in any event the linux soft phone offerings seem rather weak, any suggestions? I would just run X-Pro with crossover but I want to export this to a thin client. |
| 03:00.18 | jradford | umm speakers and microphones are still analog :) |
| 03:00.20 | cypromis | how are you going to guarantee that the softphone has enough of that pc ? |
| 03:00.21 | PTG123 | http://builds.xten.net/download-public.php?download=1135 |
| 03:00.23 | PTG123 | download that |
| 03:00.41 | kimosabe | thanks man |
| 03:00.56 | PTG123 | crimd: iaxcomm isn't bad, however the new iaxclient has some bugs.. so there was some latency when i tried it.. however i am told it may not be on linux |
| 03:02.19 | crimd | PTG123: I tried that, its not quite up there, I really need something similar to an avaya softphone except on linux. IAX would be great though, no problems there :) |
| 03:02.30 | *** join/#asterisk Shido6 (~shido@d57-87-253.home.cgocable.net) |
| 03:02.40 | PTG123 | what didn't you like about iaxcomm? |
| 03:02.45 | Hymie | hmm.. does anyone know if callprogress = yes has any options, so that I can have it ignore the first 5 seconds of dialing.. then look for ringtones? |
| 03:02.55 | *** join/#asterisk IQ (~iq@70-57-182-12.omah.qwest.net) |
| 03:03.15 | crimd | Also anyone have any luck compiling asterisk on PPC hardware? I had some problems on a JS20 blade with the gsm codec. |
| 03:03.44 | ZX81 | Hymie: callprogress only works in USA |
| 03:03.50 | ZX81 | and even then hardly |
| 03:03.54 | Hymie | it works well here |
| 03:03.58 | ZX81 | sweet |
| 03:04.01 | ZX81 | here==? |