| 00:00.00 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 00:00.09 | Shaun2222 | i just want to install the default player :) guess theirs not one.. |
| 00:00.11 | sleepy_one | Shaun2222, theoretically you can use any mp3 or audio player than can ouput the audio in raw format |
| 00:00.20 | Dream_WEaver | ie. You made the necessary changes to support the lack of mpg123? |
| 00:01.05 | Shaun2222 | Dream_WEaver: just [default] mode=files directory=/var/lib/asterisk/mohmp3 |
| 00:01.58 | Dream_WEaver | What OS shaun? |
| 00:02.05 | Shaun2222 | centos 4.x |
| 00:02.24 | Dream_WEaver | I would just go ahead and try sleepy_one's way :) |
| 00:02.32 | Dream_WEaver | There is an example in the musiconhold.conf |
| 00:03.37 | *** join/#asterisk zigman (i=zigman@irc.zigman.de) |
| 00:03.55 | sleepy_one | Shaun2222, I can PM you madplay install instructions it's easy :-) |
| 00:03.57 | Dream_WEaver | I don't know about resource intense though... |
| 00:03.59 | Dream_WEaver | root 1708 0.0 0.2 5504 2764 ? S 06:11 0:00 mpg123 -q -s --mono -r 8000 -f 8192 2unlimited - No Limit.mp3 gwen stefani- halla back girl.mp3 Gunther |
| 00:04.03 | Dream_WEaver | Taking nothing here. |
| 00:05.29 | Shaun2222 | i wonder if the problem is that i didnt convert the mp3's using lame with format_mp3? |
| 00:06.35 | Shaun2222 | ok |
| 00:06.37 | Shaun2222 | now it's working |
| 00:06.49 | Shaun2222 | kind of stupid... |
| 00:07.09 | Shaun2222 | i had to rebuild the format_mp3 because it was built in the future... |
| 00:07.39 | *** join/#asterisk Navire (n=navire@201.19.170.6) |
| 00:08.23 | Navire | iax work with client peer to peer? |
| 00:08.47 | *** join/#asterisk Denmark (n=fake@62.242.24.182) |
| 00:12.26 | Darwin35 | will asterisk operate my garbage disposal ? |
| 00:13.11 | sleepy_one | Navire, I may be wrong but I don't think so I believe you need Client0 <-> IAX2 <-> * <-> IAX2 <-> Client1 or Client0 <-> IAX2 <-> * <-> ALMOST_ANY_PROTO <-> Client1 |
| 00:14.16 | sleepy_one | Navire, I believe you could replace * with any IAX2 compatible server in the middle. I suppose you could add IAX2 capabilities to any client but I do not know if any such clients already exist |
| 00:14.50 | sleepy_one | Darwin35, of course it will!!! :-D |
| 00:15.06 | Darwin35 | tell me how |
| 00:15.17 | Darwin35 | what do ineed |
| 00:15.32 | ast_gittl | i m hiring asterisk enginner specilized in predictive dialer |
| 00:15.38 | Darwin35 | other then kram flipping the switch |
| 00:17.01 | ast_gittl | so any body interesting ? |
| 00:17.07 | ast_gittl | i m hiring asterisk enginner specilized in predictive dialer |
| 00:17.29 | Strom_C | ast_gittl, please stop asking the same thing repeatedly |
| 00:17.36 | ast_gittl | ok sir |
| 00:17.38 | ast_gittl | let me change |
| 00:17.41 | ManxPower | ast_gittl, nope. bad karma |
| 00:17.49 | ast_gittl | do you want to get a job of asterisk engineer ? |
| 00:17.53 | ast_gittl | now better ? |
| 00:18.07 | Strom_C | ast_gittl, I think you missed the point |
| 00:18.11 | Strom_C | this is not the place |
| 00:18.30 | Strom_C | go post on a job listings board |
| 00:18.40 | ast_gittl | i see |
| 00:18.44 | ast_gittl | is this asterisk room? |
| 00:18.48 | Dovid | yes it is |
| 00:18.51 | Dovid | FOR HELP |
| 00:18.52 | ast_gittl | i wana get help on predictive dialer but no one helping |
| 00:18.56 | ast_gittl | EXACTLY |
| 00:18.58 | ast_gittl | i need HELP |
| 00:19.01 | Dovid | u want help installing ? |
| 00:19.14 | ManxPower | ast_gittl, nobody wants the bad karma |
| 00:19.17 | Strom_C | ugh, a story-changer |
| 00:19.44 | ManxPower | Really, posting on a job board is your best bet. |
| 00:20.17 | ast_gittl | ok |
| 00:20.24 | ast_gittl | so no one is the expert here?? |
| 00:20.32 | Shaun2222 | ast_gittl: with the way you acted a little while ago i doubt anybody will help you.... |
| 00:20.51 | Strom_C | ast_gittl, go away |
| 00:21.06 | Navire | <PROTECTED> |
| 00:21.41 | Hmmhesays | a predictive dialers |
| 00:21.57 | Hmmhesays | so you can call me during dinner |
| 00:22.10 | Strom_C | telemarketers are the scum of the earth |
| 00:22.23 | Strom_C | especially those who want to abuse a wonderful thing like asterisk to do it :) |
| 00:22.23 | ManxPower | ast_gittl, There are a few taboos here, in this Asterisk IRC culture. One of them is predictive dialers because they can be a tool used for one of the horrors of the world -- telemarketing. Another taboo is the IPP G729 libs. |
| 00:22.36 | Strom_C | that's like the seventh level of telemarketing hell |
| 00:22.50 | Denmark | I wonder if Digium has USB fsx cards supported by GNU/Linux? |
| 00:22.53 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 00:23.00 | Hmmhesays | even though setting up asterisk to do predictive dialing would be cake |
| 00:23.06 | Dovid | no they dont denmark |
| 00:23.12 | bkw__ | Hmmhesays, NO IT IS NOT |
| 00:23.15 | bkw__ | what are you smokin? |
| 00:23.16 | Denmark | Dovid : A pitty :) |
| 00:23.23 | Hmmhesays | nothing fancy |
| 00:23.30 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:23.30 | Dovid | although there are other companies that do |
| 00:23.47 | Denmark | Dovid : You have any recommendations? |
| 00:23.56 | Dovid | do search |
| 00:23.58 | Dovid | on google |
| 00:24.05 | Denmark | ok, thanks |
| 00:24.11 | bkw__ | GOOGLE KNOWS ALL |
| 00:24.31 | Navire | Somebody know ClientA IAX2 direct with ClientB IAX2. Is possible? |
| 00:24.32 | Beirdo | and what it doesn't know today, it likely will tomorrow |
| 00:24.37 | Hmmhesays | little edgy today bkw__? |
| 00:24.46 | ast_gittl | ok guys |
| 00:24.50 | ast_gittl | any body good |
| 00:24.56 | bkw__ | Hmmhesays, been on a plane for 6 hours |
| 00:24.57 | bkw__ | :P |
| 00:25.11 | Denmark | ast_gittl : Nah .. only the bad and the ugly ones left .. |
| 00:25.38 | Beirdo | bkw_, that would make anyone fed up if not downright grumpy, you're doin fine |
| 00:25.41 | Strom_C | ast_gittl, I might consider doing it if you give me a really awesome blowjob |
| 00:25.46 | *** join/#asterisk vopi (n=kkk@202.139.199.21) |
| 00:26.03 | vopi | hello |
| 00:26.59 | ManxPower | Strom_C, please don't tease the breeders? |
| 00:27.55 | Strom_C | ManxPower, oh come on, it's hard enough to even give me what i consider a mediocre blowjob, so it's almost guaranteed that I won't get "really awesome" from ast_gittl |
| 00:27.57 | ast_gittl | hi |
| 00:28.02 | ManxPower | bkw_, I read it a year or so ago. It basically says that you can use the IPP libs for non-commercial stuff, but it's up to you to get the correct patent licensing. Only intel could get away with that. |
| 00:28.08 | ast_gittl | well |
| 00:28.32 | Strom_C | and even then, it only might make me consider it |
| 00:28.43 | Strom_C | it's just a first step, you see |
| 00:29.39 | Denmark | Dovid : Found none. Guess I will wait till Digium gets some. The idea is to connect the card to my router, and use it as ata. |
| 00:29.59 | Dovid | email the asterisk users list |
| 00:30.01 | ast_gittl | well Strom and Manx Power |
| 00:30.08 | ast_gittl | do not worry |
| 00:30.12 | ast_gittl | you will be happy |
| 00:30.19 | ast_gittl | just provide me complete predictive dialer solution |
| 00:30.28 | Beirdo | oh piss off |
| 00:30.42 | *** part/#asterisk Navire (n=navire@201.19.170.6) |
| 00:30.46 | Strom_C | ast_gittl, no no, you must provide sexual favors FIRST |
| 00:31.02 | file | Strommy Boy! behave yourself |
| 00:31.04 | ast_gittl | ok |
| 00:31.06 | *** join/#asterisk michaelo (n=michaelo@adsl-153-11-203.gsp.bellsouth.net) |
| 00:31.14 | ast_gittl | i can fuck you if you want i m a male |
| 00:31.26 | Hmmhesays | someone kick him |
| 00:31.46 | *** mode/#asterisk [+b *!*n=zxc786@202.59.90.*] by file |
| 00:31.46 | *** kick/#asterisk [ast_gittl!n=jcolp@mctnnbsa24w-142167060049.pppoe-dynamic.nb.aliant.net] by file (file) |
| 00:31.52 | Strom_C | yay |
| 00:31.55 | Beirdo | ahhh, that works better |
| 00:32.16 | file | I will tolerate behavior from the regulars for a short time, but new people acting that way... nope |
| 00:32.34 | *** join/#asterisk jpeeler (n=jpeeler@c-71-226-105-137.hsd1.sc.comcast.net) |
| 00:32.36 | ManxPower | file, I feel the same way. |
| 00:32.44 | nain | hi |
| 00:32.46 | file | have to earn my respect and friendship |
| 00:32.46 | alephcom | I'm talking to him in another window that's going to get closed soon. :-) |
| 00:32.49 | sleepy_one | file, thank you! |
| 00:33.41 | Hmmhesays | good file |
| 00:34.07 | *** kick/#asterisk [Hmmhesays!n=jcolp@mctnnbsa24w-142167060049.pppoe-dynamic.nb.aliant.net] by file (file) |
| 00:34.09 | file | :D |
| 00:34.15 | Strom_C | hahaha |
| 00:34.19 | Strom_C | DO NOT PET THE FILE |
| 00:34.31 | Strom_C | (/) |
| 00:34.43 | file | only if you pay the petting fee |
| 00:34.55 | Strom_C | is that the regular petting fee or the heavy petting fee? |
| 00:34.57 | *** join/#asterisk Hmmhesays (n=Neg@72.24.105.126) |
| 00:35.02 | file | Strom_C: both! |
| 00:35.06 | Strom_C | ooooooo |
| 00:35.12 | Hmmhesays | lol, so that coming a mile away |
| 00:35.16 | Hmmhesays | *saw even |
| 00:35.42 | sleepy_one | file, do you have a minute to answer an * question? |
| 00:35.50 | file | sleepy_one: ask a question in here and it may get answered |
| 00:37.23 | sleepy_one | file, thanks! the question is has anyone encountered a situation where you lose audio in asterisk after 3min on a zap channel ? running * 1.2.6 zaptel 1.2.5 and * SVN and zaptel SVN as of yesterday with Digium hardware |
| 00:37.32 | Denmark | The requested URL /pipermail/asterisk-users/2002-October.txt.gz was not found on this server. |
| 00:37.38 | file | I have not, but I don't use zaptel hardware |
| 00:37.56 | Denmark | oh well .. google may have cache. :) |
| 00:38.10 | Denmark | nzh |
| 00:38.23 | Hmmhesays | pri? |
| 00:38.49 | sleepy_one | file, any similar problems with non-zaptel hardware? I'm trying to determine what might be causing the problem |
| 00:39.12 | file | sleepy_one: mailing lists are your best bet... or hang around here, but does it only happen with zaptel hardware? |
| 00:39.18 | Hmmhesays | i have the problem in pri gateways that aren't set to take the clock from the telco |
| 00:39.49 | sleepy_one | file, as far as I know it only happens on a TDM400p with 4 FXOs on BT in the UK |
| 00:39.58 | file | did you call support? |
| 00:40.03 | Hmmhesays | arugh fscking BT, I hate them |
| 00:40.13 | sleepy_one | file, spent hours on the phone, yes |
| 00:40.20 | file | dejavu |
| 00:40.30 | Hmmhesays | do a 3 minute milliwatt test |
| 00:40.47 | sleepy_one | file, I am not sure if it is the zaptel hardware or BT or a strange combination thereof |
| 00:41.48 | shmaltz | how can I make sure that .wav and not .WAV is sent as the attachment in voicemail.conf? |
| 00:42.13 | sleepy_one | or the PC hardware itself... it's very strange with FXS_LS audio is gone after about 3min with FXS_KS it hangs up after about 3min it doesn't even work with FXS_GS |
| 00:43.17 | Strom_C | why in gods name would you be provisioning the port as groundstart if it's not a groundstart line? |
| 00:43.19 | Hmmhesays | sleepy_one do a 3 minute milliwatt test |
| 00:43.22 | sleepy_one | there are at least 2 or 3 other people in the UK that are experiencing similar problems |
| 00:43.40 | Hmmhesays | see if it's hanging up on some freaky silence detect |
| 00:47.53 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 00:49.05 | sleepy_one | I have tried both busydetect and callprogress on and off together and separately, zttool, zttest, fxs_ls fxs_ks fxs_gs ( didn't work ) and everything else I could think of |
| 00:50.09 | Strom_C | sleepy_one, you realize what a groundstart line is, right? |
| 00:51.58 | sleepy_one | Strom_C, I tried just about everything I could try just in case I knew GS wasn't gonna work but I tried anyway |
| 00:52.12 | Strom_C | ok |
| 00:52.29 | Strom_C | just making sure you're not wildly just trying everything because "IT DOESNT WORK" |
| 00:53.41 | *** join/#asterisk Strom_M (n=strom@66.159.243.59) |
| 00:55.46 | *** join/#asterisk Jon335 (n=Jon335@unaffiliated/jon335) |
| 00:56.18 | IOscanner | anyone know if there is a way to pass caller id info for inbound calls from SPA-3000? |
| 00:56.37 | Hmmhesays | inbound pstn->ip? |
| 00:57.23 | IOscanner | My guess is Dial Plan 1: would need to pass the callerid info, but not sure what |
| 00:57.25 | *** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM) |
| 00:57.53 | Hmmhesays | good luck generating callerid to an fxo port |
| 00:57.54 | *** join/#asterisk infinii (i=infinii@adsl-68-78-84-149.dsl.milwwi.ameritech.net) |
| 00:58.17 | IOscanner | it works find for digium stuff |
| 00:58.34 | Hmmhesays | from asterisk-->pstn? |
| 00:59.12 | IOscanner | it is using username of the device, so it should be able to the SPA to pass the callerid instead of the username used to connect to asterisk for the trunk |
| 01:00.19 | Hmmhesays | callerid="Hookers and Blow" <9003574888> |
| 01:00.23 | Hmmhesays | in your sip.conf |
| 01:00.52 | *** join/#asterisk cybertheq (n=infra@216-251-177-106.ips.cpinternet.com) |
| 01:01.14 | IOscanner | that will not work with the SPA-3000 I am trying get the SPA to pass the callerid of the inbound call from the fxo port on the SPA passing the call to asterisk via a trunk |
| 01:01.44 | Hmmhesays | that doesn't make much sense |
| 01:02.01 | *** join/#asterisk talljon84 (n=talljon8@66-168-63-104.dhcp.mdsn.wi.charter.com) |
| 01:02.47 | IOscanner | why I have a SIpura ATA that has an FXO port on it. It is registered to asterisk via a trunk. when I receive a call on the FXO port the SPA passes the call to the asterisk trunk |
| 01:03.06 | IOscanner | it then used the defined dial plan to call the extension, voicemail, IVR I define. |
| 01:03.14 | IOscanner | it works fine, just not passing callerid info |
| 01:04.15 | Hmmhesays | there is likely a setting in the spa to enable callerid detection |
| 01:04.17 | FlyboySR22 | We are getting ready to purchase a PRI card for our Asterisk system and I am trying to decide between the Digium card and the Sangoma card - anyone have any expirence with these cards...? |
| 01:04.40 | Hmmhesays | normally i use external gateways |
| 01:04.54 | IOscanner | Digium works fine, Sangoma works fine and the new openvox card works fine |
| 01:05.05 | IOscanner | openvox cards is half the cost of either card |
| 01:05.06 | FlyboySR22 | any preference between them ..? |
| 01:05.17 | FlyboySR22 | the Digium and the Sangoma appear to cost about the same |
| 01:05.19 | Strom_M | FlyboySR22, the digium card will be a little easier to configure |
| 01:05.25 | sleepy_one | I have tried digium T1 cards and they work fine, the analog cards work too but analog sux |
| 01:05.34 | IOscanner | No since they all work the same I am currently using the openvox cards |
| 01:05.48 | FlyboySR22 | openvox... |
| 01:05.51 | FlyboySR22 | openvox.com |
| 01:06.00 | IOscanner | just don't buy the single port fxo cards from openvox they don't work very well and don't support callerid |
| 01:06.09 | IOscanner | openvox.com.cn I think |
| 01:06.30 | sleepy_one | in my experience the Digium T1 cards are a LOT better than their analog counterparts |
| 01:06.50 | FlyboySR22 | I have a few of the digium analog cards and have never had any issues with them |
| 01:06.52 | Hmmhesays | that rings true of pretty much any analog/digital counterpart |
| 01:07.10 | sleepy_one | yes indeed |
| 01:07.11 | Hmmhesays | its the nature of the beast |
| 01:07.24 | cybertheq | *** |
| 01:07.34 | sleepy_one | analog tends to suck regardless of manufacturer |
| 01:07.36 | cybertheq | Does anyone have ADSI (screen phone) experience? |
| 01:07.43 | FlyboySR22 | Does the openvox D110P support PRIs |
| 01:07.45 | FlyboySR22 | ? |
| 01:08.05 | sleepy_one | I have also tried expensive channel banks and they sucked too ( $1,000 and up ) |
| 01:08.07 | FlyboySR22 | nevermind, I see the info on their sitre about the PRI support |
| 01:08.10 | FlyboySR22 | :-) |
| 01:08.16 | *** join/#asterisk pdunkel (n=pdunkel@213.235.192.27) |
| 01:09.16 | IOscanner | yes |
| 01:09.21 | IOscanner | and split data |
| 01:09.34 | IOscanner | I have one doing 6 pri and the rest data |
| 01:09.52 | pdunkel | Hi there |
| 01:10.04 | IOscanner | they seem to support the cards fine via e-mail and called me once when I was having a problem with a card |
| 01:10.04 | pdunkel | <PROTECTED> |
| 01:10.10 | sleepy_one | cybertheq, No, but this might help http://voip-info.org/wiki/ search for adsi |
| 01:10.21 | pdunkel | Is this a good time to ask or are you guys busy? |
| 01:10.28 | cybertheq | ***thanks, been there*** have current woes *** |
| 01:10.33 | Shaun2222 | if i do exten => s,1,xxxx and then s,200,xxxx will asterisk see that exten 2-199 dont exist and jump to 200 or will it just dump out? |
| 01:10.46 | IOscanner | I can't seem to get them to call me or comment back about the crappy single fxo port cards I bought and they don't work |
| 01:10.51 | FlyboySR22 | IOscanner, you have 6 of the D110P in a single box..? |
| 01:11.20 | shmaltz | how can I make sure that .wav and not .WAV is sent as the attachment in voicemail.conf? |
| 01:11.20 | Denmark | S100U |
| 01:11.29 | FlyboySR22 | I was figuring if I got the Digium card, at least I could get support on it :-) |
| 01:12.05 | IOscanner | no two I have 5 single fxo cards that are crap I am trying to return to them. Other then that I have the 4 port fxo and fxs cards from them and they work fine you just have to add a w before dialing out. |
| 01:12.09 | IOscanner | other then that they work fine. |
| 01:12.24 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 01:12.37 | *** join/#asterisk AnonMon (n=j00nix@MTL-HSE-ppp195158.qc.sympatico.ca) |
| 01:12.44 | FlyboySR22 | good deal, thanks much for the info !! It is appreciated. |
| 01:12.52 | IOscanner | the hardest part is to make sure you have a good zapata.conf and zaptel.conf file to support the PRI/data if you plan to use it |
| 01:13.06 | IOscanner | besure you have a kernel that has hdlc support. |
| 01:13.07 | Denmark | Dovid : S100U .. but it has been discontinued. |
| 01:13.11 | AnonMon | guys, you know if * can utilize FXO ports in one of those SIP gateways? |
| 01:13.17 | Shaun2222 | anybody know why i'm seeign this error? http://pastebin.com/645305 |
| 01:13.18 | Dovid | hmm |
| 01:13.22 | IOscanner | I have found a few versions that don't work well and have problems before 2.6.8 |
| 01:13.26 | Dovid | Denmark: did u email the list ? |
| 01:13.34 | IOscanner | hdlc was foobar before 2.6.8 kernel |
| 01:13.39 | Denmark | Dovid : nah.. can't really email from this box. |
| 01:13.51 | Dovid | u subscribed to the list ? |
| 01:14.01 | Denmark | Dovid : Nope |
| 01:14.06 | *** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
| 01:14.07 | Dovid | so subscribe |
| 01:14.15 | Denmark | yeah |
| 01:14.19 | Dovid | Shaun2222: whast the problem ? |
| 01:14.25 | Denmark | I will when I get onother box. |
| 01:14.28 | Dovid | ok |
| 01:14.33 | Dovid | some one there can help u |
| 01:14.39 | Shaun2222 | Dovid: those errors... |
| 01:14.48 | Shaun2222 | Dovid: are you dovid bender? |
| 01:14.55 | Dovid | huh ? |
| 01:15.02 | Denmark | Dovid : It seems Digium used to do a S100U that did what I wanted, and there was a module to support it too. |
| 01:15.02 | Shaun2222 | just wondering.. |
| 01:15.17 | Dovid | Denmark: go look on ebay |
| 01:15.18 | AnonMon | anybody? |
| 01:15.22 | Denmark | Dovid : most other fxs to USB seems to be skype only. |
| 01:15.29 | pdunkel | Hi AnonMon |
| 01:15.31 | Denmark | ahh of course |
| 01:15.37 | AnonMon | hello pdunkel |
| 01:15.41 | Dovid | si know the list had info a few months back |
| 01:15.44 | Shaun2222 | Dovid: i want those errors to go away... |
| 01:15.44 | Denmark | get a used card :) |
| 01:15.45 | file[laptop] | The S100U never existed, nope, seriously, uh huh ... yeah |
| 01:15.57 | file[laptop] | okay it did, it just sucked |
| 01:16.05 | Denmark | file[laptop] : How did it suck? |
| 01:16.09 | pdunkel | I just asked these guys "I am thinking of a new app and wanted to have a short check whether anyone else thinks this might be worth it or whether I'm waisting my time." |
| 01:16.10 | file[laptop] | way before my time though |
| 01:16.13 | *** join/#asterisk Yasheee (n=yashy@mail.yashy.com) |
| 01:16.15 | AnonMon | I am wondering if one could utilize FXO port found in a sip gateway via * |
| 01:16.15 | file[laptop] | Denmark: didn't work well |
| 01:16.22 | pdunkel | No answer yet, guess they are busy. |
| 01:16.43 | pdunkel | How about you? |
| 01:17.11 | IOscanner | pdunkel: I have cisco phones so I wrote an xml app to do that |
| 01:17.12 | Yasheee | Anyone have a URL for using Monitor() when the telephone plugged into the X100P card is in use? (outbound and inbound calls) |
| 01:17.13 | Denmark | file[laptop] : I guess Digium gave up on USB after that? |
| 01:17.18 | AnonMon | I am no1. n00b at this, don't know nothing |
| 01:17.20 | *** part/#asterisk talljon84 (n=talljon8@66-168-63-104.dhcp.mdsn.wi.charter.com) |
| 01:17.34 | pdunkel | IOScanner: to do what? |
| 01:18.21 | file[laptop] | Denmark: aye |
| 01:19.18 | IOscanner | weather, directory, stocks, zip codes, customer database, addresses etc |
| 01:19.35 | IOscanner | all feed from web or mysql |
| 01:20.10 | *** join/#asterisk cced (n=dev2003@222.33.36.205) |
| 01:20.18 | pdunkel | Nice, so why exactly are you telling me? I was thinking more along the lines of roaming users within asterisk. |
| 01:20.45 | cced | who is familar with chan_zap or chan_sip? |
| 01:21.12 | pdunkel | However I'm very glad you have written so much. Maybe you have written that as well 9.9 |
| 01:21.18 | IOscanner | I thought you wanted to write a weather program |
| 01:22.12 | ljam | ~centosbug |
| 01:22.14 | jbot | centosbug is probably a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. |
| 01:22.24 | IOscanner | I think there is already an AGI program to do weather check by calling an extension that is pointed to the AGI script |
| 01:22.39 | AnonMon | guys, let me ask you this |
| 01:22.40 | pdunkel | How did you get that idea? Well anyway, I was thinking along the lines of a user dialing an extension, entering their id and pin. From then on the device they called from can be dialed using DialUser(<userid>) until they deregister. |
| 01:22.55 | AnonMon | could one utilize the FXO port on the SPA-3000 via *? |
| 01:23.05 | cced | who chan_zap get data from zaptel?thannks |
| 01:23.33 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 01:23.36 | AnonMon | say for instance, * connects to SPA3000 via SIP then makes calls over its FXO port |
| 01:23.53 | shmaltz | how can I make sure that .wav and not .WAV is sent as the attachment in voicemail.conf? |
| 01:24.14 | Strom_M | shmaltz, what difference does it make? |
| 01:24.42 | shmaltz | Strom_M, becuase most Windoz machines cant' play back the WAV one |
| 01:24.56 | Strom_M | shmaltz, show me your voicemail.conf |
| 01:24.59 | Strom_M | pastebin |
| 01:25.00 | pdunkel | shmaltz: I don't think you can. You might check app_voicemail.c, But I thisnk it does it lowercase, in which case it might be your MTA, |
| 01:25.22 | AnonMon | no takers? |
| 01:25.28 | shmaltz | pdunke, it's not the case that bothers me, but the format |
| 01:25.31 | Strom_M | AnonMon, yes you can |
| 01:25.44 | shmaltz | Strom_M, one min |
| 01:25.54 | Yasheee | shmaltz: format=wav49|gsm|wav ; have you tried removing the ones you don't want? |
| 01:26.15 | shmaltz | Yasheee, I'm not sure which one I want |
| 01:26.39 | Yasheee | try each, one at a time then? |
| 01:26.43 | shmaltz | the one app_voicemail sends is in gsm 6.10 format according to MS recorder |
| 01:27.00 | shmaltz | Yasheee, I know, but it's not realy an option at the moment |
| 01:27.15 | Yasheee | <shrug> |
| 01:27.33 | pdunkel | Shmaltz: You want wav, windows doesn't play gsm and wav49 is newer. wav is your vanilla windows wav format. |
| 01:27.34 | Yasheee | Anyone have a URL for using Monitor() when the telephone plugged into the X100P card is in use? (outbound and inbound calls) |
| 01:27.37 | *** join/#asterisk vopi (n=kkk@202.139.197.33) |
| 01:27.52 | AnonMon | Strom_M, thank you dear sir, now would I be able to route calls coming through PSTN port on the SPA-3000 to *? |
| 01:28.17 | Strom_M | AnonMon, sounds like you need to read THE BOOK |
| 01:28.20 | Strom_M | ~thebook |
| 01:28.21 | jbot | thebook is, like, Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 01:28.24 | Strom_M | ~docs |
| 01:28.26 | jbot | extra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 01:28.40 | shmaltz | pdunkel, but voicemail doesn't send the good wav one, there are 2 wav files created: .wav and .WAV, I want .wav, and app_voicemail is sending .WAV |
| 01:28.48 | AnonMon | Strom_M, rtfm of SPA3000? |
| 01:28.50 | Hmmhesays | AnonMon: I'll help you out for a little bit of cash, i need some bar money tonight |
| 01:29.22 | cced | who want to help me? |
| 01:29.30 | pdunkel | shmaltz: as yashee said, remove the unneccessary formats from voicemail.conf (format=wav) should do it! |
| 01:29.48 | AnonMon | Hmmhesays, ya think I would adopt this weird route if I had the $? I would gone with a decent channel bank or digium cards. Dont ya figure? |
| 01:30.02 | Hmmhesays | 25 bucks' |
| 01:30.05 | Dream_WEaver | slePP sucks |
| 01:30.06 | shmaltz | pdunkel, well I cant' realy try that at the moment, but I guess if/when I have the time I will try it |
| 01:30.07 | Dream_WEaver | oops |
| 01:30.08 | Dream_WEaver | ;) |
| 01:30.18 | *** join/#asterisk tuxinator_linux (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 01:30.27 | AnonMon | Hmmhesays, sorry, can't afford it |
| 01:30.37 | Hmmhesays | ohwell, worth a shot |
| 01:31.49 | cced | I want to know how to work chan_zap chan_sip libpri zaptel |
| 01:31.58 | cced | no document about this |
| 01:32.06 | IOscanner | Okay got SPA-3000 to pass callerid you have to delay answering the call for 3 seconds to get it time to pass the callerid info |
| 01:32.25 | Strom_M | IOscanner, duh |
| 01:32.25 | shmaltz | Strom_M, here: http://pastebin.ca/48466 |
| 01:32.57 | Strom_M | shmaltz, yeah, try just format=wav |
| 01:33.35 | shmaltz | thanks guys, Strom_M, pdunkel, and Yasheee |
| 01:33.44 | Hmmhesays | AnonMon sipura's FAQ on the spa-3000 has pretty much everything you need to know |
| 01:34.51 | terrapen | anyone know how to reset a polycom IP600 that has bad config information stored? |
| 01:35.07 | terrapen | i can't even get to the main application where i do the 4+6+8+* |
| 01:35.12 | Strom_M | terrapen, i like using a sledgehammer |
| 01:35.29 | terrapen | it loads sip.ld, the processes configuration, the displays error 0x10100, then reboots |
| 01:36.10 | terrapen | i need a "hold these keys down and then power on" kind of reset |
| 01:36.38 | AnonMon | Hmmhesays, I guess I needed to be pointed in the RTFM line ;) |
| 01:37.08 | Hmmhesays | ~rtfm |
| 01:37.10 | jbot | rtfm is, like, Read The F*cking Manual (TM) |
| 01:37.25 | Darwin35 | ~google |
| 01:37.26 | jbot | [google] a search engine found at http://www.google.com/ |
| 01:37.35 | AnonMon | haha, infobot |
| 01:37.37 | AnonMon | jbot, shutup |
| 01:37.39 | jbot | yes master, I'll STFU |
| 01:37.39 | Darwin35 | ~wiki |
| 01:38.05 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 01:38.12 | Darwin35 | ~docs |
| 01:38.13 | jbot | hmm... docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 01:38.21 | sleepy_one | ~while(1) { read($TFM); } |
| 01:38.43 | Darwin35 | ~sex jbot |
| 01:38.52 | AnonMon | jbot, good boy |
| 01:38.52 | jbot | AnonMon: aw, gee |
| 01:38.53 | Yasheee | Having 50 documents doesn't make RTFM very simple |
| 01:39.03 | Darwin35 | ~sex |
| 01:39.06 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/ |
| 01:40.24 | sleepy_one | I LOVE sEx |
| 01:40.27 | Darwin35 | yum death by garlic mashpatato's |
| 01:40.33 | Strom_M | jbot, phone |
| 01:40.35 | jbot | methinks phone is warbling while I'm updating the flash from blob...it's amusing now, it's like it knows I'm erasing its brain. Mwa ha ha ha. |
| 01:40.43 | sleepy_one | yum groupinstall asterisk |
| 01:40.59 | Denmark | I am installing asterisk right now. |
| 01:41.06 | Denmark | (debian unstable) |
| 01:41.07 | Darwin35 | ~jbot |
| 01:41.08 | jbot | [jbot] only marginally useful at best, or a silly little bugger |
| 01:41.18 | terrapen | sigh |
| 01:41.20 | terrapen | damn you polycom |
| 01:41.25 | terrapen | it SHOULDN'T be this hard. |
| 01:41.47 | Darwin35 | polycom licks bullballs |
| 01:42.04 | Nugget | and that's making you hard? |
| 01:42.15 | Denmark | Darwin35 : What about their phones? |
| 01:42.18 | Hmmhesays | and here we go.weeeeeeeeeeeee |
| 01:42.33 | Darwin35 | asterisk giving me head makes me hard |
| 01:42.45 | Strom_M | ~catsex |
| 01:42.49 | *** part/#asterisk naturalblue (n=Administ@87.192.100.109) |
| 01:42.50 | Hmmhesays | you got yours hooked up to a fleshlight too huh? |
| 01:43.24 | Strom_M | jbot, catsex is also dead hookers |
| 01:43.25 | jbot | Strom_M: okay |
| 01:43.34 | Darwin35 | my asteerisk box runs on 9v |
| 01:43.41 | Strom_M | ~catsex |
| 01:43.42 | jbot | extra, extra, read all about it, catsex is dead hookers |
| 01:43.47 | sleepy_one | ok, let's stop messing about now ;-) |
| 01:43.49 | Strom_M | muahahaha |
| 01:44.22 | Hmmhesays | yeah right, its night time |
| 01:44.34 | Hmmhesays | this is when the fun starts |
| 01:44.42 | sleepy_one | in other news .... there be MOH here |
| 01:44.49 | sleepy_one | I just love MAD |
| 01:44.52 | sleepy_one | ~mad |
| 01:44.54 | jbot | mad is probably Rob Leslie's MPEG Audio Decoder, GPL, at http://www.mars.org/home/rob/proj/mpeg/ |
| 01:45.07 | sleepy_one | jbot, thank you :-) |
| 01:45.07 | jbot | sleepy_one: sure thing |
| 01:45.16 | *** join/#asterisk RednckRay (i=ray@cpe-65-189-198-222.neo.res.rr.com) |
| 01:45.35 | RednckRay | Hello ? |
| 01:45.45 | sleepy_one | hi |
| 01:46.14 | Darwin35 | nobody here but us cyborgs |
| 01:46.15 | tuxinator_linux | anything more descriptive than 'hello ?' ? |
| 01:46.17 | sleepy_one | MAD just works |
| 01:46.23 | RednckRay | Sorry. The nickserv registration thing threw me a bit there. |
| 01:46.56 | sleepy_one | jbot, I prefer mad.sf.net ;-) |
| 01:46.57 | jbot | You prefer mad.sf.net ;-)? |
| 01:47.20 | RednckRay | Thanks Hmmhesays. Now I've got Gretchen Wilson singing in my head. |
| 01:47.29 | Hmmhesays | could be worse, at least she's hot |
| 01:47.49 | RednckRay | Yes she is. |
| 01:48.51 | RednckRay | Size large crowd for so little chattin. Must be all the cyborgs. |
| 01:48.56 | *** part/#asterisk pdunkel (n=pdunkel@213.235.192.27) |
| 01:49.07 | RednckRay | Anybody here use Digium's IAXy ? |
| 01:49.20 | sleepy_one | yeppers that's a roger |
| 01:49.38 | Denmark | hmm .. I have problems installing asterisk. |
| 01:49.54 | Denmark | Debian unstable .. seems to be circular dependency. |
| 01:49.57 | sleepy_one | RednckRay, Yes, I have used many IAXy's |
| 01:50.15 | RednckRay | Sweet. About how long doe it take that there IAXy to give you the joyous ringing sounds after you finish dialing ? |
| 01:50.16 | sleepy_one | Denmark, y not install from src? |
| 01:50.51 | Denmark | sleepy_one : Habit, I guess. Debian packages are usually easier to get running, and often has a nice config. |
| 01:51.12 | sleepy_one | RednckRay, it doesn't always it depends sometimes I get no ringing sounds |
| 01:52.29 | sleepy_one | Denmark, I usually for pkg in in (pkg0 pkg1 pkg2 .... pkgN) do; wget $pkg; tar xzvf $pkg; cd $pkg; make clean; make install; |
| 01:52.52 | Darwin35 | hokers get you fresh hot hokers here |
| 01:52.56 | sleepy_one | s/in// |
| 01:53.10 | Hmmhesays | i use Debian stable, a lot |
| 01:53.17 | RednckRay | Well, the reason I ask is that I'm currently using an AG168 with the IAX firmware and I spent a couple days trying to figure out why it wasn't working. As it turned out, it worked a bit more slowly than I had expected. About 20 seconds more slowly. |
| 01:53.43 | sleepy_one | Denmark, that would work for * for other packages you have to throw ./configure in there ;-) |
| 01:53.51 | Denmark | sleepy_one : nice |
| 01:53.54 | Denmark | :) |
| 01:55.28 | Darwin35 | asterisk+sangooma on fbsd rocks a |
| 01:55.31 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 01:55.48 | sleepy_one | Denmark, with some minor modifications |
| 01:56.11 | sleepy_one | RednckRay, it takes 20sec to dial? |
| 01:56.46 | RednckRay | Well, sometimes it's closer to 15 seconds. |
| 01:57.02 | [av]bani | \o> |
| 01:57.04 | [av]bani | <o/ |
| 01:57.49 | RednckRay | Being the slacker than I am, I'm fine with 20 seconds of delay. The boss seems to think that may hamper productivity a bit though. So I was thinking maybe the IAXy's might be a better choice. |
| 01:58.51 | Hmmhesays | 20 seconds before the iaxy dials out? |
| 01:59.26 | RednckRay | No...it's 20 seconds before my AG168 dials out. I'm asking what the same delay on an IAXy is. |
| 01:59.48 | Hmmhesays | most ata's are 3-5 second timeouts |
| 01:59.52 | |omni| | check the dialplan on the handset (or adapter as it may be) |
| 02:00.04 | |omni| | you may just need to adjust the dialplan so it knows when you're done punching buttons |
| 02:00.18 | Hmmhesays | set your interdigit timeout |
| 02:00.26 | |omni| | I had to change diaplan on my ciscos and the sipura |
| 02:00.37 | RednckRay | The dialplan is set to "none" on the AG168. |
| 02:00.45 | |omni| | that maybe the issue |
| 02:00.51 | Hmmhesays | there should be a digit timeout, or an interdigit timeout |
| 02:00.58 | |omni| | you probably just have no dialplan and a 15 sec timeout |
| 02:01.07 | Denmark | hmm.. turns out I have installed asterisk. |
| 02:01.22 | Denmark | Is there a GUI or something? |
| 02:01.37 | Hmmhesays | not unless you haven't amp/freepbx installed |
| 02:01.41 | sleepy_one | asterisk -r -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvc |
| 02:01.48 | sleepy_one | after safe_asterisk |
| 02:02.43 | sleepy_one | Denmark, safe_asterisk; asterisk -r |
| 02:03.04 | sleepy_one | Denmark, unless asterisk is already running in which case you just want asterisk -r |
| 02:03.23 | Darwin35 | asterisk -vvvvvvgr |
| 02:03.28 | tuxinator_linux | why safe? |
| 02:03.35 | Hmmhesays | if you installed it with debian packages, it won't start by itself |
| 02:03.40 | Darwin35 | it restarts when it dies |
| 02:04.08 | Denmark | Hmmhesays : I did edit the /etc/default/asterisk to allow it to run. |
| 02:04.17 | tuxinator_linux | never had it die yet.. guess I would have figured that one out later |
| 02:04.28 | Hmmhesays | ./etc/init.d/asterisk start |
| 02:04.46 | [av]bani | http://www.wacotrib.com/news/content/news/stories/2006/04/06/04062006wacbillnye.html?cxtype=rss&cxsvc=7&cxcat=11 |
| 02:04.49 | [av]bani | bill nye <3 |
| 02:06.43 | theorem_ | curious .. |
| 02:06.55 | sleepy_one | tuxinator_linux, * doesn't usually die but it has died on me b4 |
| 02:07.02 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 02:07.36 | theorem_ | If I do: exten => _9XXXXXXX,1,Dial(IAX2/provider/123${EXTEN:1}) |
| 02:07.58 | theorem_ | what happens to the 8th digit if I so choose to dial it ? |
| 02:08.06 | theorem_ | well, 9th actually .. |
| 02:08.11 | [av]bani | http://finance.yahoo.com/q/bc?t=5y&s=RHAT&l=on&z=m&q=l&c=msft |
| 02:08.12 | [av]bani | lol? |
| 02:08.57 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 02:09.13 | AvoidingDeadlock | I AM I AM |
| 02:09.15 | X-Rob | theorem_, it won't match |
| 02:09.20 | Denmark | I got the cli |
| 02:09.29 | sleepy_one | cli > help |
| 02:09.31 | X-Rob | if you dial 9111111111 it won't match 9XXXXXXXX |
| 02:09.54 | theorem_ | X-Rob - ok, so what happens then if nothing I have matches it ? |
| 02:10.01 | theorem_ | does it take a close match, or ? |
| 02:10.08 | X-Rob | it doesn't do anything |
| 02:10.11 | [TK]D-Fender | no match = refused |
| 02:10.16 | X-Rob | '.' means 'match anything' |
| 02:10.19 | [TK]D-Fender | 404 |
| 02:10.27 | X-Rob | so you want _9., |
| 02:10.39 | theorem_ | I see |
| 02:10.50 | |omni| | you guys know of any IAX ATAs other than the IAXy? |
| 02:11.06 | theorem_ | whern the PSTN gets 11 digits though ... what happens there ? I assume they strip it to 10 ? |
| 02:11.07 | X-Rob | omni, there are PA1688 based ATA's |
| 02:11.12 | [TK]D-Fender | |omni| : Gnet has one and there is another... |
| 02:11.13 | Denmark | I installed destar .. but.. |
| 02:11.14 | *** join/#asterisk AvoidingDeadlock (n=Avoiding@c-68-32-112-142.hsd1.md.comcast.net) |
| 02:11.18 | X-Rob | and PA1688 support IAX |
| 02:11.21 | |omni| | any decent ones? |
| 02:11.29 | [TK]D-Fender | |omni| : nope. |
| 02:11.29 | |omni| | okay..digging |
| 02:11.39 | Denmark | http:/localhost/ ?? |
| 02:11.46 | [TK]D-Fender | Pa168X = shitty. |
| 02:13.35 | theorem_ | X-Rob - I think I bugged you before (sorry) but I forgot if you recommended a decent SIP Soft phone for Win* -- if not, could you reccommend ? |
| 02:13.36 | |omni| | okay..and question two... |
| 02:13.44 | |omni| | valid alternatives to TE110P ? |
| 02:14.03 | [TK]D-Fender | theorem_ : X-Pro / eyebeam |
| 02:14.03 | |omni| | just looking for options...I've got Digium hardware in two boxes here with no issues |
| 02:14.06 | Hmmhesays | external gateway |
| 02:14.18 | theorem_ | Thaks Fender, will try those |
| 02:14.18 | ljam | hrmmm.... does openser not use that GUI to control the proxy anymore? |
| 02:14.18 | [TK]D-Fender | |omni| : Sangoma A101 |
| 02:14.29 | [TK]D-Fender | theorem_ : Neither are FREE however |
| 02:14.34 | theorem_ | oh :-/ |
| 02:14.36 | Hmmhesays | serWeb was for ser |
| 02:14.44 | [TK]D-Fender | theorem_ : thats the problem... |
| 02:14.51 | theorem_ | I forgot that condition , any free ones ? |
| 02:14.58 | X-Rob | there's a funky free IAX one |
| 02:15.05 | theorem_ | "funky" ? |
| 02:15.06 | X-Rob | I've got that one the lappys |
| 02:15.07 | X-Rob | funky |
| 02:15.10 | [TK]D-Fender | theorem_ : not that I'd associate with "good" |
| 02:15.19 | theorem_ | I see |
| 02:15.20 | sleepy_one | cya all l8r :-) |
| 02:15.21 | X-Rob | idefisk |
| 02:15.34 | ljam | Hmmhesays: so am I suppose to use the DB solution and control the users and auth there? |
| 02:15.36 | theorem_ | s/had/head |
| 02:15.38 | [TK]D-Fender | idefisk = fugly... but sorta gets the job done... it crashes on me... |
| 02:15.42 | |omni| | cool |
| 02:16.24 | theorem_ | I suppose I could give it a whirl, just wanna test before I buy more hardware |
| 02:16.48 | [TK]D-Fender | theorem_ : if you just need something quick, grab X-Lite |
| 02:17.08 | theorem_ | win* ? |
| 02:17.12 | [TK]D-Fender | theorem_ : but its lacks the commercial codec and transfer/conference. |
| 02:17.16 | [TK]D-Fender | theorem_ : yes. |
| 02:17.35 | theorem_ | ok, I'll try that too |
| 02:17.39 | [TK]D-Fender | Linux has Ekiga now which is just cool... |
| 02:18.02 | [TK]D-Fender | voice & video with all the basic functions that X-Lite lacks... |
| 02:18.04 | *** join/#asterisk Dovid (n=Dovid@HFA62-0-168-76.bb.netvision.net.il) |
| 02:18.38 | Denmark | Hmm.. there has to be docs on how to start |
| 02:18.39 | [TK]D-Fender | That'd be something to get if a Win32 version is to be released |
| 02:18.43 | Denmark | ~docs |
| 02:18.44 | jbot | extra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 02:18.49 | *** join/#asterisk Tili (n=Tili@61.140.191.13) |
| 02:18.56 | [TK]D-Fender | Denmark : ... |
| 02:18.59 | [TK]D-Fender | ~thebook |
| 02:19.00 | jbot | i guess thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 02:19.04 | theorem_ | Denmark - sarcasm ? |
| 02:19.40 | Denmark | theorem_ : Nah ... just wondering how this "destar" is supposed to work. |
| 02:20.08 | ljam | very carefully |
| 02:22.39 | theorem_ | X-Lite seems to make Bike parts too ... |
| 02:22.58 | Denmark | oh ... asterisk at home is a live cd or something? |
| 02:23.38 | RednckRay | They're Counterpath now. xten.com or some such |
| 02:24.05 | RednckRay | For the X-Lite softphone that is. |
| 02:24.49 | *** join/#asterisk somegeek_ (i=levin@unaffiliated/somegeek) |
| 02:25.16 | tuxinator_linux | [TK]D-Fender: why would you want at win32 version? |
| 02:25.29 | X-Rob | Denmark, not a live CD. It installs a full blown * system on your pc. Nuking anything on your hard drive |
| 02:26.31 | RednckRay | The asterisk@home folks do have a vmware image too. |
| 02:26.45 | *** join/#asterisk exten123 (n=exten@60.49.6.190) |
| 02:26.50 | [TK]D-Fender | tuxinator_linux : So that there would be an Open viable alternative to eyeBeam and remove most other suggestions from the playing field |
| 02:27.04 | *** join/#asterisk lilo_ (i=levin@freenode/staff/pdpc.levin) |
| 02:27.40 | tuxinator_linux | ~win32 |
| 02:27.41 | jbot | rumour has it, win32 is a stupid idea for asterisk |
| 02:27.53 | Denmark | ok |
| 02:27.59 | tuxinator_linux | jbot doesn't seem to agree |
| 02:28.01 | Denmark | http://localhost:8080 |
| 02:28.01 | theorem_ | cygwin ? |
| 02:28.13 | [TK]D-Fender | tuxinator_linux : who said anything about * for win32? I'ma talking SIP clients.... |
| 02:28.20 | *** join/#asterisk Pegger (n=peg@pool-68-163-155-240.bos.east.verizon.net) |
| 02:28.23 | tuxinator_linux | ahh |
| 02:28.25 | exten123 | some TDM400P issue here. My card have 2 FXO and 2 FXS modules install. after I load the zaptel drive the LED at the back of the card only 2 have the light. What happen to the card? |
| 02:28.37 | tuxinator_linux | excuess me.. I shouldn't join in the middle, sorry |
| 02:28.58 | [TK]D-Fender | exten123 : did you plug the molex connector to the card? |
| 02:29.16 | X-Rob | exten123, you have a broken card. |
| 02:29.25 | X-Rob | send it back |
| 02:29.41 | tuxinator_linux | X-Rob: ouch, but deserved |
| 02:29.58 | X-Rob | tuxinator_linux 8) |
| 02:30.03 | X-Rob | ~win32 |
| 02:30.05 | jbot | [win32] a stupid idea for asterisk |
| 02:30.07 | X-Rob | heh |
| 02:30.09 | X-Rob | I love that |
| 02:30.37 | X-Rob | jbot, or win32 is actually a stupid idea for anything critical |
| 02:30.38 | jbot | I think you lost me on that one, X-Rob |
| 02:30.44 | tuxinator_linux | it's about as good of an idea as using win32 as a server |
| 02:31.02 | X-Rob | jbot, win32 is a stupid idea for asterisk, and actually a stupid idea for anything critical |
| 02:31.03 | jbot | ...but win32 is already something else... |
| 02:31.06 | tuxinator_linux | but this isn't a win32 bashing channel |
| 02:31.09 | X-Rob | jbot, no, win32 is a stupid idea for asterisk, and actually a stupid idea for anything critical |
| 02:31.11 | jbot | okay, X-Rob |
| 02:31.15 | X-Rob | ~win32 |
| 02:31.16 | jbot | from memory, win32 is a stupid idea for asterisk, and actually a stupid idea for anything critical |
| 02:31.16 | exten123 | [TK]D-Fender, you mean 12V power to card right, I got |
| 02:31.19 | [av]bani | http://forums.somethingawful.com/showthread.php?s=&threadid=1844470&perpage=40&pagenumber=1 |
| 02:32.00 | exten123 | X-Rob, I think so, |
| 02:32.14 | *** part/#asterisk exten123 (n=exten@60.49.6.190) |
| 02:32.17 | riddlebox | hrmm anyone got an idea why I just recorded 4 prompts but my agi script doesnt play them, I can use Playback() in extensions.conf and they work? |
| 02:32.26 | X-Rob | exten123, sorry. It's broken. It could also mean that the daughter cards aren't plugged in correctly |
| 02:32.38 | *** join/#asterisk exten123 (n=exten@60.49.6.190) |
| 02:36.01 | *** part/#asterisk RednckRay (i=ray@cpe-65-189-198-222.neo.res.rr.com) |
| 02:36.24 | Denmark | ~phone |
| 02:36.25 | jbot | rumour has it, phone is warbling while I'm updating the flash from blob...it's amusing now, it's like it knows I'm erasing its brain. Mwa ha ha ha. |
| 02:37.17 | Denmark | I assume phones are like customer accounts? |
| 02:37.46 | *** part/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 02:37.47 | *** part/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 02:37.52 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 02:38.09 | [TK]D-Fender | exten123 : TRY REVERSING THEM |
| 02:43.53 | *** join/#asterisk mattwj2005 (n=Matt@user-12l3lqm.cable.mindspring.com) |
| 02:45.26 | Denmark | weee! |
| 02:45.34 | Denmark | I guess I have asterisk running! |
| 02:45.39 | De_Mon | doesn't asterisk have an app to convert a string to numbers? |
| 02:45.41 | Tili | what is the difference between Jitter Delay and Frames Sent Per Second. I want abit more technical perspective |
| 02:45.57 | De_Mon | like IF($EXTEN == 'HELLO') |
| 02:46.02 | mattwj2005 | I got telasip running :) |
| 02:46.26 | VeNoMouS_ | who here knows bout progress_ind on pots? |
| 02:46.41 | VeNoMouS_ | cisco related |
| 02:47.35 | mattwj2005 | anyone know anything about meetme? |
| 02:47.51 | mattwj2005 | if not, what do you guys think of Meet me 2? |
| 02:47.51 | Yasheee | anyone have a good (URL?) example for someone not using SIP? I just want 3 extensions with an X100P card. I have one extension which is taking messages, I just want to add the option for all 3 I've created. |
| 02:50.10 | De_Mon | mattwj2005 where do i get meetme2? |
| 02:50.10 | [TK]D-Fender | Yasheee : What are these 3 extensions coming in under? |
| 02:50.33 | mattwj2005 | just a sec I'll check |
| 02:50.45 | X-Rob | Yasheee, exactly the same as Sip, except instead of dialling SIP/123, you dial Zap/1 (or 2 or 3) |
| 02:50.51 | mattwj2005 | right now I am watching dl.tv on dialup :P |
| 02:50.51 | Yasheee | [TK]D-Fender: in voicemail.conf I have 1, 2 and 3 defined. Maybe I should pastebin my extensions.conf |
| 02:51.20 | mattwj2005 | I mean on a P3 800 |
| 02:51.53 | [TK]D-Fender | Yasheee : Ok, what you're saying makes them just "virtual" is there anything more tangable about these "extensions" or are you referring to IVR menu options that just lead to different mailboxes, etc? |
| 02:52.10 | riddlebox | this is so wierd, in my agi script, I can go 3/4 the way through, then it will not play any of my recordings? |
| 02:52.55 | mattwj2005 | crap it is a broken link |
| 02:52.58 | De_Mon | mattwj2005 what did you want to know about meetme? |
| 02:53.09 | Yasheee | http://www.devrandom.org/p/12 is how I have it setup right now |
| 02:53.43 | Pegger | humm is only voip where like dns, one server goes down the phone just tries to connect to the next in line |
| 02:53.57 | Yasheee | when I call it just says no one is at extension 1 and lets me leave a message. I'd like the option to leave a message at one or two, otherwise goto 3 |
| 02:54.02 | mattwj2005 | how to get a conference setup |
| 02:54.13 | Yasheee | just going off various online examples, take it easy on my current config :) |
| 02:54.16 | mattwj2005 | I want to create a conference room |
| 02:54.34 | X-Rob | Yasheee, that's because you have s,6,Voicemail(u1) |
| 02:54.34 | Yasheee | mattwj2005: meetme I believe it's called if you search "asterisk meetme" |
| 02:54.53 | mattwj2005 | oh I have been trying to get it install |
| 02:55.02 | mattwj2005 | just a second....I'll show you my errors |
| 02:55.34 | X-Rob | peraps you mean to have t,1,Voicemail(u1) (timeout) |
| 02:55.34 | X-Rob | ? |
| 02:55.41 | X-Rob | oh you've got that |
| 02:55.45 | X-Rob | just take out the s,6 |
| 02:55.54 | exten123 | Why my chan_zap.so cannot be load, I have configure my zapata.conf |
| 02:56.10 | exten123 | it show setup_zap: Ignoring signalling |
| 02:56.35 | X-Rob | Yasheee, you want s,6 to be 'WaitExten(10)' |
| 02:56.39 | X-Rob | or something like that |
| 02:56.56 | Yasheee | X-Rob: sweet thanks. Now I guess I just need a message at s,6 specifying there are 3 extensions and who is at each? |
| 02:57.14 | X-Rob | You've got background asterisk-recording |
| 02:57.27 | X-Rob | that's your message saying 'push 1 for 1, 2 for 2, 3 for 3, or don't do anything for 1' |
| 02:57.42 | mattwj2005 | <PROTECTED> |
| 02:57.42 | mattwj2005 | <PROTECTED> |
| 02:57.42 | mattwj2005 | <PROTECTED> |
| 02:57.42 | mattwj2005 | <PROTECTED> |
| 02:57.42 | mattwj2005 | <PROTECTED> |
| 02:57.43 | mattwj2005 | <PROTECTED> |
| 02:57.45 | mattwj2005 | <PROTECTED> |
| 02:57.47 | mattwj2005 | <PROTECTED> |
| 02:57.49 | mattwj2005 | <PROTECTED> |
| 02:57.51 | mattwj2005 | Apr 6 21:57:05 WARNING[14337]: chan_zap.c:1069 zt_open: Unable to open '/dev/zap/pseudo': No such device |
| 02:57.53 | mattwj2005 | Apr 6 21:57:05 ERROR[14337]: chan_zap.c:7643 chandup: Unable to dup channel: No such device |
| 02:57.55 | mattwj2005 | Apr 6 21:57:05 WARNING[14337]: app_meetme.c:461 build_conf: Unable to open pseudo channel - trying device |
| 02:57.57 | mattwj2005 | Apr 6 21:57:05 WARNING[14337]: app_meetme.c:464 build_conf: Unable to open pseudo device |
| 02:57.59 | mattwj2005 | <PROTECTED> |
| 02:58.01 | mattwj2005 | <PROTECTED> |
| 02:58.03 | mattwj2005 | <PROTECTED> |
| 02:58.04 | X-Rob | yay for ignoring flooders |
| 02:58.05 | mattwj2005 | <PROTECTED> |
| 02:58.07 | mattwj2005 | sorry guys :( |
| 02:58.09 | mattwj2005 | I didn't mean to flood the channel |
| 02:58.11 | JunK-Y | mattwj2005: stop flooding the channel, use pastebin.ca |
| 02:58.18 | mattwj2005 | sorry |
| 02:58.19 | JunK-Y | u need a zaptel timer to use meetme. |
| 02:59.04 | mattwj2005 | how do I use pastebin? |
| 02:59.06 | Yasheee | X-Rob: all that it's saying now is a pause, and then "The person at extension 3 is unavailable" |
| 02:59.16 | JunK-Y | u go at www.pastebin.ca |
| 02:59.21 | JunK-Y | ~pb |
| 02:59.22 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 02:59.28 | X-Rob | Yasheee, do you not have a file 'asterisk-recording'? |
| 02:59.29 | Yasheee | nothing about extension 1 or 2 |
| 03:00.15 | Yasheee | # locate asterisk-recording |
| 03:00.15 | mattwj2005 | http://pastebin.ca/48473 |
| 03:00.16 | Yasheee | # |
| 03:00.22 | X-Rob | So why are you trying to play it then? |
| 03:00.27 | Yasheee | strange, it seemed to work when I only had the one extension |
| 03:00.55 | X-Rob | exten => s,5,BackGround(asterisk-recording) |
| 03:00.59 | X-Rob | ^^ why is that there? |
| 03:01.01 | Yasheee | well, it did work when only one, so I just left it hoping it did work.. I didn't look for the file inititally, it just worked |
| 03:01.16 | X-Rob | look in /var/lib/asterisk/sounds |
| 03:01.21 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 03:01.21 | Yasheee | it worked before, so I left it |
| 03:01.26 | X-Rob | there's a whole pile of sounds there |
| 03:01.27 | mattwj2005 | it didn't find asterisk-recording |
| 03:01.39 | mattwj2005 | I am doing a updatedb |
| 03:02.35 | Yasheee | it's /usr/share/asterisk/sounds on debian, and not showing that filename. Is it possible that is macro that calls other files? (i.e it owuld be a different asterisk-recording if there are different numbers of extensions?) |
| 03:02.43 | *** join/#asterisk Math` (n=Math_@modemcable148.4-81-70.mc.videotron.ca) |
| 03:02.55 | *** join/#asterisk w32 (n=234@adsl-70-224-88-120.dsl.sbndin.ameritech.net) |
| 03:03.09 | X-Rob | when you do a background, don't put the extension on it (eg, those have .gsm on 'em.. Just use 'Background(press-1&or&press-2&or&press-3) |
| 03:04.07 | X-Rob | Yasheee, here's a hint I think you're missing. 'show application applicationame' - eg, 'show application Background' or 'Show application dial' |
| 03:04.13 | X-Rob | at the asterisk console |
| 03:04.30 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167060049.pppoe-dynamic.nb.aliant.net) |
| 03:05.49 | Denmark | ok |
| 03:05.58 | Denmark | something still needs tweaking |
| 03:06.49 | Yasheee | ahh sweet.. thanks. very useful |
| 03:07.15 | Yasheee | is there an easy way to listen to the plethora of gsm files at the shell? like mpg123 or something? |
| 03:07.28 | Denmark | yes |
| 03:07.44 | X-Rob | Yasheee, there's a 'sounds.txt' file in /usr/src/asterisk |
| 03:07.57 | X-Rob | that tells you what's in each file |
| 03:08.42 | Yasheee | ./usr/share/doc/asterisk-sounds-main/sounds.txt.gz on debian |
| 03:08.47 | Yasheee | thanks though! handy |
| 03:08.59 | X-Rob | Yasheee, try to avoid using disto-specific asterisk's |
| 03:09.06 | X-Rob | get the source, compile it yourself. |
| 03:09.09 | X-Rob | then we can help you |
| 03:09.45 | Yasheee | well I've found anything path related you've mentioned, so not to worry. package management is my preferred (lazy man) approach |
| 03:10.02 | X-Rob | and that's _not_ the sounds.txt I was talking about. |
| 03:10.11 | X-Rob | there's sounds that come with asterisk, and sounds that come with asterisk-sounds |
| 03:10.22 | X-Rob | both have a sounds.txt file |
| 03:10.54 | Yasheee | yikes, I eat my words then, as that's the only asterisk related sounds.txt I have |
| 03:11.42 | X-Rob | first one in asterisk's sounds.txt is agent-alreadyon.gsm |
| 03:11.51 | X-Rob | oops |
| 03:11.53 | X-Rob | my fault |
| 03:11.58 | X-Rob | it's sounds.txt and sounds-extra.txt |
| 03:12.02 | X-Rob | serves me right for not checking |
| 03:12.12 | *** join/#asterisk CowboyKitty (n=CowboyKi@adsl-67-66-188-100.dsl.rcsntx.swbell.net) |
| 03:12.45 | Yasheee | apologizing for helping a newb hasn't hadn't a chance to wade through the plethora of docs? hardly required |
| 03:15.13 | Yasheee | okay just found an issue.. I had someone call me, but I didn't get it until asterisk picked up.. so it defaulted to ext3 as planned, and I left a quick message and hit #.. message saved and sent. But then it seemed to endlessly loop saying ext3 was unavailable and to leave a message |
| 03:16.23 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 03:16.44 | mattwj2005 | so what is a zap channel? |
| 03:17.24 | Denmark | extension - is that the number to dial? |
| 03:17.38 | X-Rob | Yasheee, post your current extensions.conf |
| 03:17.55 | mattwj2005 | just don't cut and paste it here.... |
| 03:18.00 | mattwj2005 | big no no |
| 03:18.15 | Yasheee | X-Rob: same as before, just removed the s,6 line |
| 03:18.15 | mattwj2005 | once again...sorry guys |
| 03:18.30 | Yasheee | http://www.devrandom.org/p/12 |
| 03:18.50 | X-Rob | ok |
| 03:19.03 | X-Rob | so you don't have a t,2 |
| 03:19.15 | X-Rob | (which shouldn't matter, I think it just goes to 3 skipping non-existing ones) |
| 03:19.22 | Yasheee | @#^&* |
| 03:19.26 | Yasheee | oh |
| 03:19.28 | mattwj2005 | so once I install zaptel....I have to recompile my kernel? |
| 03:19.32 | Yasheee | which makes sense |
| 03:19.33 | [TK]D-Fender | <PROTECTED> |
| 03:19.42 | X-Rob | Yasheee, asterisk console, 'set debug 4' and 'set verbose 4' |
| 03:19.47 | X-Rob | then call into it |
| 03:19.48 | [TK]D-Fender | mattwj2005 : No, |
| 03:19.51 | X-Rob | pastebin the output |
| 03:20.00 | mattwj2005 | okay |
| 03:20.20 | Yasheee | [TK]D-Fender: ;4200 => 9855,Mark Spencer,markster@linux-support.net,mypager@digium.com,attach=no|serveremail=myaddy@digium.com|tz=central|maxmsg=10 |
| 03:20.26 | Yasheee | is an example in voicemail.conf |
| 03:20.38 | [TK]D-Fender | Thats different.. not in my experience... |
| 03:21.26 | mattwj2005 | http://pastebin.ca/48477 |
| 03:22.27 | [TK]D-Fender | Yasheee : And your s,6 leaves a voicemail automatically, and then you can exit with # and end up in the menu only to leave another VM... |
| 03:22.49 | Yasheee | X-Rob: very bizarre.. it did hangup that time. I'll pastebin anyways.. |
| 03:23.13 | X-Rob | I also noticed you've spelt 'VoiceMailMain' wrong on 8500 |
| 03:23.22 | [TK]D-Fender | mattwj2005 : You need tor ecompile zaptel and enable ZTDUMMY. there is a line to be uncommented out and a kernel module to load. |
| 03:23.39 | X-Rob | [TK]D-Fender, yes, it's a bit of a ballsup |
| 03:23.44 | Yasheee | [TK]D-Fender: so multiple email addresses does work, we both got t hem. |
| 03:23.47 | X-Rob | but I think he's getting the hang of dialplans now |
| 03:24.39 | [TK]D-Fender | X-Rob : Funny... I don't see a single Dial in there ;) its an answering machine with multiple boxes :) |
| 03:24.44 | theorem_ | D-Fender -- ahhh ... I think I have to load that too .. modprobe ztdummy ? what config line ? |
| 03:24.54 | X-Rob | [TK]D-Fender, that's what he's setting it up to be, I think |
| 03:25.06 | [TK]D-Fender | theorem_ : Don't recall.... check it out on the wiki.. it'll tell you where to go... |
| 03:25.11 | theorem_ | ok |
| 03:25.20 | theorem_ | ~docs |
| 03:25.22 | jbot | methinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 03:25.22 | [TK]D-Fender | X-Rob : then the module isn't loaded... |
| 03:25.30 | mattwj2005 | do you know where I have to uncomment and what module to load? |
| 03:25.37 | X-Rob | [TK]D-Fender, wrong name |
| 03:25.43 | theorem_ | lol |
| 03:25.44 | Yasheee | All I'm looking for is a glorified mailbox at this stage.. as well as an option to record all inbound and outbound calls silently. |
| 03:26.03 | X-Rob | Yasheee, well that means you need to have the calls come in and go out |
| 03:26.16 | X-Rob | you can't just 'tap' a line |
| 03:26.24 | *** join/#asterisk somegeek (i=levin@unaffiliated/somegeek) |
| 03:26.29 | theorem_ | ...unless you're with the FBI ... |
| 03:26.29 | [TK]D-Fender | Yasheee : can't really do outbound like that right now... it'll never know when to start spying... |
| 03:26.37 | Yasheee | At this point the mailbox seems to be working, I just have add a vm message that specifies who is at each extenstion and add it to the Background |
| 03:26.48 | [TK]D-Fender | X-Rob : you can if its your own (usually and depending where you are) |
| 03:26.55 | X-Rob | [TK]D-Fender, I meant asterisk can't |
| 03:27.00 | X-Rob | not 'you as a person, under the law] |
| 03:27.05 | Yasheee | In Canada, only one party has to agree to the tapping |
| 03:27.18 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 03:27.32 | theorem_ | in the US, nobody does :( |
| 03:27.34 | Yasheee | so I need FXS to handle outbound tapping? |
| 03:27.36 | [TK]D-Fender | Yasheee : I think it would be healthy if you stopped using the word "extension". That is somewhat reserved for the concept of a uniquely number telephone device that will ring. |
| 03:28.16 | [TK]D-Fender | Yasheee : you need to have * KNOW when a call is being placed. the only realistic way to do that is for your phones to pass THROUGH *. |
| 03:28.21 | Yasheee | [TK]D-Fender: fair enough, each unique VM box |
| 03:28.59 | Denmark | [TK]D-Fender : So if your phone number is "1-800-astguru" thats your extension? |
| 03:29.12 | Yasheee | Is there a way to check VM remotely? As well, if I pickup a landline in my house can I check it? |
| 03:29.26 | [TK]D-Fender | Denmark : No, thats just a PSTN phone #. |
| 03:29.50 | [TK]D-Fender | Yasheee : not with just an x100.... |
| 03:30.02 | Yasheee | These are the two questions from betterhalf@example.com who I'm trying to convince we need a PBX in the house :) |
| 03:30.12 | mattwj2005 | will doing a make zttool help? |
| 03:30.18 | Yasheee | as the existing telco offers this |
| 03:30.26 | [TK]D-Fender | 1-800-astguru would jsut be a phone #. What happens when you dial it and where it goes is another matter... |
| 03:30.31 | Yasheee | [TK]D-Fender: can I do either of them? |
| 03:30.32 | Tili | hey IAX2 on asterisk side had de-jitter mechanism |
| 03:30.46 | *** join/#asterisk FlyboySR22 (n=rsears@sdtc.ar01.f2-40.host2.1.americanis.net) |
| 03:31.01 | X-Rob | Yasheee, 'show application voicemailmain' |
| 03:31.19 | Denmark | [TK]D-Fender : So whats an extension? If I add phones to the asterisk, the extension is not the phonenumber? |
| 03:31.22 | [TK]D-Fender | Yasheee : there are many ways to make the experience more "transparent" so that it doesn't FEEL like you're using a pbx. |
| 03:31.52 | [TK]D-Fender | Denmark : Typically an "extension" is a phone (soft, hard, or other) attached to a PBX. |
| 03:32.32 | [TK]D-Fender | Denmark : So if you were to connect a SIP phone to * and provide a means for it to dial out and/or be called by some means, then it would be an extension. |
| 03:32.43 | theorem_ | you have to edit zaptel.conf in /etc |
| 03:32.46 | theorem_ | fyi |
| 03:32.53 | justinu | 3 DIGIT AREA CODE-3 DIGIT OFFICE CODE-4 DIGIT EXTENSION |
| 03:32.57 | justinu | NPA-NXX-XXXX |
| 03:33.38 | [TK]D-Fender | actually IIRC the 2nd set of 3 digits is better referred to as the "exchange" |
| 03:34.03 | justinu | office code, or exchange |
| 03:34.14 | justinu | central office, branch exchange |
| 03:34.48 | justinu | the romanian word for it is "chentrala telefonica" which I thought sounded neat |
| 03:36.01 | justinu | in the bell system, they were also called class 5 offices |
| 03:36.24 | Yasheee | X-Rob: that's why that was there. Perfect! thanks so much |
| 03:36.28 | Denmark | [TK]D-Fender : OK .. lets say I just setup asterisk. I want to call asterisk from a softphone (linphone) to listen to the current time. I have added "say time" as extension "95". |
| 03:36.38 | Yasheee | [TK]D-Fender: that's what I'm shooting for :) |
| 03:36.58 | Denmark | [TK]D-Fender : Now I add a phone also.. say extension "12345" |
| 03:37.08 | Yasheee | I have the www ui setup for handling messages for when she gets home.. something the teclo doesn't offer :) |
| 03:37.53 | Yasheee | not being able to check messages from a local landline is the only issue I can see thus far, and that's because I don't have headsets yet to sell her on SIP |
| 03:38.02 | [TK]D-Fender | Denmark : Thats more like an IVR choice that leads to an internal feature... a bit of a misnomer. Think of an extension as something that has to be seperately wired into the system |
| 03:38.47 | [TK]D-Fender | Yasheee : well you have it set up so you can check from outside your home, but not internally. you might use a soft-phone for that but you'd need to check it from a computer. |
| 03:38.50 | Denmark | [TK]D-Fender : Well... I have to test the installation somehow. |
| 03:39.47 | [TK]D-Fender | Yasheee : an idea for you would be to get an ATA, plug your whole house onto it and use it as 1 extensions. incoming calls would be answered by the X100P immediately and would Dial your ATA. on no-answer/busy you'd let them leave a VM. |
| 03:40.21 | [TK]D-Fender | Denmark : Yeah using a soft-phone to "test-drive" * is fine. nothing wrong with that, its just that it'd suck to use it as a real phone full-time |
| 03:40.54 | [TK]D-Fender | Yasheee : just get an ATA like the SPA-2002 |
| 03:44.36 | [av]bani | ick |
| 03:45.49 | X-Rob | ~centosbug |
| 03:45.50 | jbot | centosbug is, like, a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. |
| 03:46.03 | Yasheee | that's a pretty big and ugly chassis, I hope there are many bits inside :) thanks for the lead, will shop around. |
| 03:46.15 | *** join/#asterisk Denmark (n=fake@62.242.24.182) |
| 03:46.29 | Denmark | when I call the number "sip:95@localhost", I get "Not acceptable here". |
| 03:46.42 | X-Rob | Denmark, you don't have a compatible codec |
| 03:46.44 | X-Rob | 'allow=all' |
| 03:46.59 | theorem_ | X-Rob - is there a log for asterisk I can check to see when SIP clients attempt to authenticate ? I seem to be having a problem authenticating with the soft phone. |
| 03:47.00 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 03:47.08 | X-Rob | theorem_, /var/log/asterisk/full |
| 03:47.12 | theorem_ | thanks |
| 03:48.08 | theorem_ | hmm, full ? |
| 03:48.45 | X-Rob | if it doesn't exist, enable it in /etc/asterisk/logger.conf |
| 03:48.51 | theorem_ | ok |
| 03:49.11 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 03:49.17 | *** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 03:49.57 | exten123 | I got a problem when inbound call to asterisk (auto attendant) inside dialplan. when the caller party handup immediate asterisk still continue playing the audio file until it reach the WaitExten statement. What function that I can use to resolve this issue? |
| 03:50.07 | Denmark | X-Rob : You're right! Now it seems to work somewhat! Thanks! |
| 03:50.19 | X-Rob | *amazing* |
| 03:51.17 | [TK]D-Fender | exten123 : What interface is the call coming in under? |
| 03:51.33 | exten123 | [TK]D-Fender,Zap |
| 03:51.37 | *** join/#asterisk bmg505 (n=leon@dsl-146-1-166.telkomadsl.co.za) |
| 03:52.42 | [av]bani | \o/ |
| 03:52.44 | [TK]D-Fender | <PROTECTED> |
| 03:52.59 | [av]bani | one of many |
| 03:53.23 | exten123 | [TK]D-Fender, That mean cannot solve it? |
| 03:53.51 | [TK]D-Fender | exten123 : read up on the WIKI to see how you can tweak it, but don't expect miracles.... |
| 03:54.55 | exten123 | [TK]D-Fender, it's digium card got this kind of problem only or others compatiable card also like that? |
| 03:55.43 | Denmark | ok sleepy time or something. :) |
| 03:57.40 | [TK]D-Fender | exten123 : its a weakness of the technology being used. you may need to ask your telco to do a polarity inversion or "open-loop" disconnect indication if avaiable. Like I said, go ready the wiki on "disconnect supervision" |
| 03:58.05 | theorem_ | ooo I got it to register :) |
| 03:58.35 | *** join/#asterisk chris_ast (n=Administ@59.93.56.163) |
| 03:59.22 | theorem_ | Apr 6 23:56:14 DEBUG[4294] pbx.c: Function result is 'DOCUMENTATION' |
| 03:59.26 | theorem_ | haha :) |
| 04:02.03 | exten123 | [TK]D-Fender, Thanks I has read it. |
| 04:03.21 | mattwj2005 | meet me sucks :( |
| 04:03.30 | *** join/#asterisk veto_ (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 04:04.08 | exten123 | anothers question when I do testing. why when I call from Zap interface the caller need to wait a number of ring before it's connected |
| 04:04.24 | X-Rob | exten123, it's waiting for caller ID to be delivered. |
| 04:04.32 | [TK]D-Fender | exten123 : becuase its waiting for the caller-ID to come in. |
| 04:04.52 | mattwj2005 | will Asterisk @ home work on a p100? |
| 04:06.15 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 04:09.29 | chris_ast | Whenever I transfer a call I want the connection to wait until the # key is pressed to complete the call? Please tell me how can I do this |
| 04:10.06 | chris_ast | theorem_, exten123,[TK]D-Fender: please help |
| 04:10.56 | [TK]D-Fender | <PROTECTED> |
| 04:11.10 | chris_ast | I foind dial(${TRUNK}c/number,20,r) but did not work |
| 04:11.17 | [TK]D-Fender | chris_ast : Is that on a Zap interface? |
| 04:11.27 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 04:11.30 | PakiPenguin_ | morning |
| 04:11.33 | *** join/#asterisk vopi (n=kkk@202.139.210.238) |
| 04:11.42 | [TK]D-Fender | chris_ast : you need to add one of tT to do # transfers |
| 04:12.03 | chris_ast | I am trying for follow me only not a opure transfer |
| 04:12.28 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 04:12.39 | chris_ast | I want to display a ring to the caller until # is pressed on the cell phone |
| 04:13.31 | chris_ast | [TK]D-Fender: It is not on zap |
| 04:14.44 | Math` | chris_ast: you're trying to do smth like "Press # to accept the fees" ? |
| 04:14.49 | *** join/#asterisk somegeek_ (i=levin@unaffiliated/somegeek) |
| 04:17.36 | chris_ast | math. what I am trying is |
| 04:18.07 | chris_ast | http://www.voip-info.org/wiki-Asterisk+Tips+follow+me just see this if possible, when I get a incoming call |
| 04:18.10 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 04:18.24 | chris_ast | I will dial out a number for say 10secs |
| 04:19.07 | [TK]D-Fender | chris_ast : What kind of interface is your phone on? |
| 04:19.11 | inv_Arp | any page that explains the revised dialplan syntax for 1.2.x? |
| 04:19.25 | tsume | somegeek_: people must really hate you :P |
| 04:19.30 | chris_ast | later I play a music and when I divert to next numberI want to diisplay a ring to the caller until # is pressed on the cell phone. |
| 04:19.39 | tsume | lilo: I seen the idiot who used your nick, heh. |
| 04:19.42 | tsume | lilo: the nerve |
| 04:20.56 | chris_ast | <PROTECTED> |
| 04:21.14 | [TK]D-Fender | chris_ast : So what kind of phone is it? |
| 04:21.46 | Math` | what the hell is "ANC" in the pap2's regional settings |
| 04:21.55 | chris_ast | I am using only softphones DIAX, Express talk |
| 04:22.15 | [TK]D-Fender | chris_ast : And they don't have an internal transfer capability? |
| 04:23.52 | chris_ast | I am executing priorities and there I am doing this not from phone |
| 04:24.34 | chris_ast | <PROTECTED> |
| 04:25.38 | chris_ast | when callee presses # it shud start ringing for caller |
| 04:26.29 | chris_ast | I found in link above using option 'c' with dial but it is not working |
| 04:31.50 | exten123 | how can we dial sip callgroup? |
| 04:32.37 | *** join/#asterisk argos73 (n=mike@cpe-24-93-184-116.neo.res.rr.com) |
| 04:35.43 | *** join/#asterisk forao (n=dfasdfs@ool-4354d60d.dyn.optonline.net) |
| 04:44.07 | VeNoMouS_ | exten123 : define callgroup? |
| 04:44.13 | VeNoMouS_ | u do mean queue? |
| 04:44.14 | VeNoMouS_ | agents? |
| 04:44.21 | VeNoMouS_ | random sip numbers u have |
| 04:44.23 | VeNoMouS_ | ... |
| 04:44.56 | inv_Arp | any page that explains the revised dialplan syntax for 1.2.x? |
| 04:45.23 | VeNoMouS_ | inv_Arp : invest in an english for dummys book first |
| 04:46.15 | inv_Arp | ok first person today on my /ignore list |
| 04:46.29 | Math` | yeah /ignore is useful |
| 04:46.34 | VeNoMouS_ | oh gn0 how will i handle that |
| 04:46.52 | VeNoMouS_ | like i give a fuck if im on your ignore |
| 04:46.59 | VeNoMouS_ | stop speaking retarded english |
| 04:50.47 | nick125 | VeNoMouS_: remember, Non ex transverso sed deorsum |
| 04:51.03 | VeNoMouS_ | parla italiano si? |
| 04:51.10 | VeNoMouS_ | e el spaino? |
| 04:51.18 | nick125 | http://en.wikipedia.org/w/index.php?title=BOFH&oldid=46881987 |
| 04:51.37 | VeNoMouS_ | lol |
| 04:51.43 | VeNoMouS_ | chupa moi cazzo punta |
| 04:51.45 | X-Rob | VeNoMouS_, I hope you ph33r inv_Arp's l33t ignore sk1llz. |
| 04:51.51 | VeNoMouS_ | X-Rob oh i do |
| 04:51.59 | X-Rob | 8) |
| 04:52.23 | X-Rob | Although, on his side, there wasn't anything _wrong_ with what he said. |
| 04:52.26 | X-Rob | just a stupid question. |
| 04:52.28 | X-Rob | sorry |
| 04:52.33 | X-Rob | no such thing as a stupid question |
| 04:52.35 | X-Rob | only stupid people. |
| 04:52.38 | VeNoMouS_ | fuck man this shit is pissing me off |
| 04:52.42 | VeNoMouS_ | fucking cunt arse cisco |
| 04:52.49 | X-Rob | VeNoMouS_, what's up with it? |
| 04:52.52 | X-Rob | <-- ccie |
| 04:53.01 | wasim | VeNoMouS_: thats an oxymoron |
| 04:53.37 | VeNoMouS_ | X-Rob : problem with the tx -> alert on a ringing for t.37 |
| 04:54.01 | X-Rob | VeNoMouS_, fuck. You really have to ask me shit I don't know about, don't you 8) |
| 04:54.03 | VeNoMouS_ | since carrier that is inboound is using na instead of ituu, it requires a alert |
| 04:54.21 | VeNoMouS_ | so ive done progress_ind alert setup 8 |
| 04:54.35 | VeNoMouS_ | but yea its not going through for some reason on the pots |
| 04:55.09 | VeNoMouS_ | lol im cc nothing yet i work with cisco shit all day long |
| 04:55.23 | VeNoMouS_ | and cisco nz gets us to do shit for them |
| 04:55.37 | X-Rob | heh |
| 04:55.45 | VeNoMouS_ | <PROTECTED> |
| 04:55.50 | VeNoMouS_ | fucking cunt arse telecom |
| 04:55.52 | X-Rob | nick125, nothing. Apart from 'hello' [1 second pause] |
| 04:55.56 | VeNoMouS_ | requring a ringing signal |
| 04:55.57 | X-Rob | 'anyone there?' |
| 04:56.12 | X-Rob | VeNoMouS_, hmm. |
| 04:56.13 | CrashHD | i-t---wwiiii----ll---saaaaay------I don -----tt work to-----we-----llll |
| 04:56.20 | X-Rob | I'd suggest kicking it in the crotch. |
| 04:56.26 | X-Rob | but that's not all that useful |
| 04:56.29 | X-Rob | I admit 8) |
| 04:57.48 | *** join/#asterisk websae (i=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
| 04:58.00 | nick125 | X-Rob: hehe, I should try it..actually, I will! |
| 04:58.11 | *** join/#asterisk Tili (i=Tili@61.140.191.13) |
| 04:58.23 | X-Rob | nick125, it works, just latency kills the natural flow of the conversation. |
| 04:59.14 | websae | anyone here do a lot of call traffic through SIP or IAX trunks ? curious of quality and reliability you have been having...anyone? |
| 04:59.28 | *** join/#asterisk AvoidingDeadlock (n=Avoiding@c-68-32-112-142.hsd1.md.comcast.net) |
| 04:59.59 | VeNoMouS_ | lol X-Rob |
| 05:00.00 | VeNoMouS_ | check this |
| 05:00.01 | VeNoMouS_ | <PROTECTED> |
| 05:00.06 | VeNoMouS_ | WLG-CORE01(config-dial-peer)#progress_ind ? |
| 05:00.06 | VeNoMouS_ | % Unrecognized command |
| 05:00.06 | VeNoMouS_ | WLG-CORE01(config-dial-peer)#progress_ind |
| 05:00.12 | X-Rob | bahahaha |
| 05:00.14 | *** join/#asterisk nick125_lappy (n=nick@unaffiliated/nick125) |
| 05:00.19 | VeNoMouS_ | thats in a mmoip dial-peer |
| 05:00.30 | nick125_lappy | I'm connected over my GPRS connection |
| 05:00.45 | nick125_lappy | this should be fun.. |
| 05:00.50 | X-Rob | nick125_lappy, I think you'll find that the throughput kills it there, rather than the latency |
| 05:00.52 | websae | through Sprint? |
| 05:00.54 | websae | or what? |
| 05:01.02 | nick125_lappy | websae: tmobile..sprint is cdma |
| 05:01.16 | asterboy | seen [TK]D-Fender |
| 05:01.23 | websae | that's right, i forgot |
| 05:01.39 | asterboy | websae what was the price on those GXP-2000s? |
| 05:02.24 | nick125_lappy | Apr 6 16:02:00 localhost MCE: The hardware reports a non fatal, correctable incident occurred on CPU 0. |
| 05:02.26 | X-Rob | asterboy, get a SPA941 |
| 05:02.28 | nick125_lappy | Apr 6 16:02:00 localhost Bank 1: 9400000000000151 |
| 05:02.31 | nick125_lappy | ^^ why doesnt that sound good? |
| 05:02.39 | X-Rob | gxp's are OK, but SPA941's kick their arse, and they're about the same price |
| 05:03.18 | asterboy | nice |
| 05:03.23 | asterboy | ~spa941 |
| 05:04.21 | asterboy | jbot, spa941 is an affordable, feature rich IP business phone shown here: http://www.zingotel.com/online/en/business/SPA941?PHPSESSID=6603853f4081dffad6966eab01b162a7 |
| 05:04.23 | jbot | okay, asterboy |
| 05:04.59 | [av]bani | X-Rob: depends on what you're looking for. 941 has no poe, no 10/100, and no backlight |
| 05:05.08 | [av]bani | no dual ether |
| 05:05.17 | Math` | asterboy: you should rip out the PHPSESSID to make a nicer url |
| 05:05.17 | X-Rob | no backlight is a bit of a bitch |
| 05:05.24 | [av]bani | gxp has 7 speeddial+blf |
| 05:05.38 | [av]bani | linksys really wasted an opportunity with the 942 |
| 05:05.43 | [av]bani | its S H I T |
| 05:06.15 | asterboy | Math' what is the PHPSESSID? (PHP Session ID)? |
| 05:06.24 | Math` | yup |
| 05:06.30 | nick125_lappy | websae: whats up? |
| 05:06.36 | asterboy | Where do I take that out? |
| 05:06.51 | Math` | http://www.zingotel.com/online/en/business/SPA941?PHPSESSID=6603853f4081dffad6966eab01b162a7 becomes http://www.zingotel.com/online/en/business/SPA941 |
| 05:06.56 | inv_Arp | asterboy: use cookies... |
| 05:06.57 | asterboy | got it |
| 05:07.11 | asterboy | ya, interesting. |
| 05:07.20 | asterboy | I hate cookies. |
| 05:07.32 | Math` | whats the diff between the 941 and 942? |
| 05:07.34 | asterboy | s/hate/ate/g |
| 05:07.42 | asterboy | :P |
| 05:08.11 | asterboy | I really did eat some cookies ust before getting on. |
| 05:08.20 | asterboy | in case anybody needed to know that. |
| 05:08.51 | X-Rob | why didn't you just go s/h//? |
| 05:08.55 | X-Rob | I hate cookies |
| 05:08.57 | X-Rob | s/h// |
| 05:08.59 | X-Rob | see? |
| 05:09.13 | asterboy | that is far too efficient! |
| 05:09.26 | Math` | lol |
| 05:09.29 | X-Rob | s/see/Well, You could have an entire conversation this way/ |
| 05:09.35 | X-Rob | sorta |
| 05:09.47 | X-Rob | s/sorta/but you'd still need to give it an input each time I thinkg/ |
| 05:10.07 | X-Rob | s/sorta/or can you do it twice?/ |
| 05:10.14 | X-Rob | s/sorta/hah, you can!/ |
| 05:12.08 | asterboy | those SPAs are not cheap: http://cgi.ebay.com/LINKSYS-SPA-941-SPA941-2-4-Line-VOIP-SIP-Phone-NEW_W0QQitemZ9700822410QQcategoryZ61840QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 05:12.26 | asterboy | how do you strip that line? |
| 05:12.42 | asterboy | I'd rather get a Polycom IP500 for that price. |
| 05:12.51 | argos73 | heh - jbot needs to have an "annoyance" variable worked in somewhere... after a bit of abuse, it should start getting mouthy... :) |
| 05:13.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:15.34 | Gamercjm | Has anyone here had any luck installing Asterisk on RedHat Enterprise 4 |
| 05:16.41 | mogorman | asterisk has very few deps |
| 05:16.48 | asterboy | Can't get my Buddy Watch working. |
| 05:16.53 | mogorman | it should work fine on any distro |
| 05:17.27 | Gamercjm | I heard there are some special script i might need to run |
| 05:17.33 | asterboy | Something is wrong in my Dial Plan, can someone take a peek and give some insight? |
| 05:17.36 | asterboy | http://pastebin.ca/48393 |
| 05:19.11 | asterboy | Can *not* call other extensions. |
| 05:19.23 | asterboy | So, I'm sure it is related. |
| 05:23.49 | asterboy | wow, I sure know how to clear a room. |
| 05:26.48 | drray | I like how it took us 10 minutes to get asterisk working with the PRI, then 3 hours to get the mitel to settle down |
| 05:27.16 | *** join/#asterisk terracon (n=tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
| 05:27.19 | wasim | :) |
| 05:28.42 | [av]bani | ~phones |
| 05:28.43 | jbot | i heard phones is at http://bani.anime.net/phones/ |
| 05:28.44 | [av]bani | jbot: ~phones |
| 05:29.24 | [av]bani | 941 isnt such a good deal |
| 05:29.52 | theorem_ | "Timeout, but no rule 't' in context 'default' " |
| 05:30.06 | [av]bani | the 901 is ghettophone! |
| 05:31.48 | asterboy | ~ghettophone |
| 05:32.14 | asterboy | jbot, ghettophone is a phone that is crap...for example, the 901 |
| 05:32.15 | jbot | okay, asterboy |
| 05:32.29 | [av]bani | yay! |
| 05:32.35 | asterboy | lol |
| 05:32.50 | asterboy | nope 941 does not look like a good deal |
| 05:34.34 | asterboy | why can't I dial another sip extension? |
| 05:34.57 | asterboy | what am I missing in this dial plan? |
| 05:35.02 | asterboy | http://pastebin.ca/48393 |
| 05:36.54 | asterboy | must be because I'm forcing it to always use ZAP channels or something. |
| 05:37.38 | Math` | do you like.... create one context per phone? |
| 05:37.56 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 05:39.46 | asterboy | Math`, yes |
| 05:40.11 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 05:40.30 | asterboy | well, actually these are 6 line Polycoms, so I've been using the first three lines to setup as extensions or contexts. |
| 05:40.47 | Math` | why? |
| 05:41.00 | Math` | why not creating a context named "outgoing" |
| 05:41.04 | Math` | and point all your phones to |
| 05:41.10 | Math` | (it) |
| 05:41.38 | asterboy | yes that is most likely the problem...I need change the relationship there. |
| 05:41.49 | *** join/#asterisk Laggy_McGee (n=jchadwic@pool-71-245-122-77.cmdnnj.fios.verizon.net) |
| 05:43.09 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.222) |
| 05:43.14 | asterboy | the idea orginally, was to setup each extension with its own line. |
| 05:43.34 | asterboy | The to get transfers to work, I had to group trunk anyway. |
| 05:49.00 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 05:50.03 | Laggy_McGee | I just set up my asterisk server and I'm getting phasing... |
| 05:50.14 | Laggy_McGee | Can someone point me to the cause and solution of this? |
| 05:50.34 | asterboy | what is phasing? |
| 05:50.45 | bsdfreak | heh |
| 05:50.57 | drray | echo? |
| 05:51.21 | *** join/#asterisk Creperum (n=ilya@mail.tex.kiev.ua) |
| 05:51.50 | asterboy | something can be out of phase. |
| 05:51.53 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 05:52.05 | asterboy | so I imagine phasing is an amplification. |
| 05:52.36 | asterboy | out of phase is to demuxing as phasing is to multiplexing? |
| 05:53.00 | X-Rob | I just set up _my_ asterisk server, and I'm getting fluxions! |
| 05:54.02 | drray | I bet he just has echo |
| 05:54.07 | asterboy | jbot, phasing is the process of ensuring that both sending and receiving facsimile machines start at the same position on the page. |
| 05:54.09 | jbot | asterboy: okay |
| 05:54.19 | X-Rob | ~echo |
| 05:54.21 | jbot | i heard echo is Displays the given arguments on the screen. Syntax: echo (arg1) (arg2) ..(argN). Where arg1 through argN are the arguments to echo. Example: echo "Hello World" displays the string "Hello World". "Why echo occurs" at http://lists.digium.com/pipermail/asterisk-users/2005-February/088794.html |
| 05:54.31 | drray | oh, I might be wrong then |
| 05:54.42 | asterboy | lol, your WAAAY off! |
| 05:54.49 | X-Rob | jbot, no, echo is normal and a natrual part of Voice over IP. |
| 05:54.50 | jbot | X-Rob: okay |
| 05:54.56 | X-Rob | *snigger* |
| 05:54.59 | asterboy | lol |
| 05:55.10 | X-Rob | jbot, no, echo is normal and a natural part of Voice over IP. |
| 05:55.11 | jbot | X-Rob: okay |
| 05:55.14 | X-Rob | ~echo |
| 05:55.15 | jbot | rumour has it, echo is normal and a natural part of Voice over IP. |
| 05:55.27 | X-Rob | jbot, no, echo is normal and a natural part of Voice over IP. What? You don't like it? Nah. It's great! |
| 05:55.28 | jbot | okay, X-Rob |
| 05:55.45 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 05:56.12 | X-Rob | jbot, no, echo is normal and a natural part of Voice over IP. What? You don't like it? Nah. It's great! You really don't want it? Well, read http://lists.digium.com/pipermail/asterisk-users/2005-February/088794.html then. |
| 05:56.13 | jbot | X-Rob: okay |
| 05:56.38 | X-Rob | jbot, echo is also echo, echo ..echo... cho..o... .. |
| 05:56.40 | jbot | okay, X-Rob |
| 05:56.43 | X-Rob | ~echo |
| 05:56.45 | jbot | well, echo is normal and a natural part of Voice over IP. What? You don't like it? Nah. It's great! You really don't want it? Well, read http://lists.digium.com/pipermail/asterisk-users/2005-February/088794.html then. echo, echo ..echo... cho..o... .. |
| 05:56.52 | asterboy | echo is a FEATURE of * |
| 05:57.09 | asterboy | so people with ADHD can have a feedback of their conversation. |
| 05:57.22 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-130.claranet.co.uk) |
| 05:57.34 | mogorman | right asterboy .................... |
| 05:57.57 | asterboy | :P |
| 05:58.30 | X-Rob | jbot, echo is also It also helps ADHD people keep track of their conversations. It's all good! |
| 05:58.31 | jbot | X-Rob: okay |
| 05:58.35 | X-Rob | ~echo |
| 05:58.37 | jbot | well, echo is normal and a natural part of Voice over IP. What? You don't like it? Nah. It's great! You really don't want it? Well, read http://lists.digium.com/pipermail/asterisk-users/2005-February/088794.html then. echo, echo ..echo... cho..o... .. It also helps ADHD people keep track of their conversations. It's all good! |
| 05:58.47 | asterboy | LOL |
| 06:01.27 | X-Rob | bastard! |
| 06:01.36 | X-Rob | You killed X-Rob! You Bastard! |
| 06:02.31 | *** join/#asterisk linlin (i=linlin@c-67-184-231-154.hsd1.il.comcast.net) |
| 06:03.10 | VeNoMouS_ | YES!! |
| 06:03.16 | VeNoMouS_ | he died! |
| 06:06.22 | Laggy_McGee | I am having wicked echo |
| 06:06.27 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.222) |
| 06:06.41 | Laggy_McGee | like a CD skipping |
| 06:06.44 | Laggy_McGee | only worse |
| 06:06.55 | nick125_lappy | possible network latency? |
| 06:07.05 | Laggy_McGee | it's a local server |
| 06:07.14 | nick125_lappy | what codec? |
| 06:07.20 | Laggy_McGee | GSM, I assume |
| 06:07.30 | Laggy_McGee | I'm using a basically unmodified confgi |
| 06:07.54 | Laggy_McGee | allow=gsm is the first line on the sip.conf for the device |
| 06:07.56 | asterboy | interrupts? |
| 06:08.06 | Laggy_McGee | > |
| 06:08.07 | Laggy_McGee | ? |
| 06:08.15 | asterboy | Is this a zaptel install? |
| 06:08.46 | Laggy_McGee | yes |
| 06:08.56 | asterboy | and your using sip phones? |
| 06:09.12 | Laggy_McGee | Yes. A SIP softphone. |
| 06:09.25 | Laggy_McGee | Local server, softphone on my laptop, dialing the 1000 test number |
| 06:09.41 | asterboy | ok, so no fxs equipment to test with. |
| 06:09.43 | Laggy_McGee | No. |
| 06:09.48 | Laggy_McGee | Not even fxo |
| 06:09.51 | Laggy_McGee | pure software |
| 06:09.59 | Laggy_McGee | brand new install |
| 06:10.31 | *** join/#asterisk diLLec (n=dillec@ns3.wireless4u.de) |
| 06:11.14 | asterboy | ztoh, ok so no zaptel in use then |
| 06:11.16 | X-Rob | Doesn't 1000 call digium? |
| 06:12.00 | asterboy | well we just had some great additions to jbot's echo database |
| 06:12.00 | X-Rob | no |
| 06:12.02 | asterboy | ~echo |
| 06:12.04 | jbot | i guess echo is normal and a natural part of Voice over IP. What? You don't like it? Nah. It's great! You really don't want it? Well, read http://lists.digium.com/pipermail/asterisk-users/2005-February/088794.html then. echo, echo ..echo... cho..o... .. It also helps ADHD people keep track of their conversations. It's all good! |
| 06:12.23 | X-Rob | Laggy_McGee, dude, you have an issue. |
| 06:12.33 | X-Rob | either not enough cpu, or not enough network capacity. |
| 06:12.54 | X-Rob | being that it'll run happily on a pii/333 or higher, I think your network is broken. |
| 06:12.59 | asterboy | run top |
| 06:13.02 | X-Rob | (unless you're trying to run it on a p1 or something) |
| 06:13.09 | asterboy | and make a call to see if the CPU is bound |
| 06:13.19 | Laggy_McGee | X-Rob: EPIA - 1000 ghz |
| 06:13.21 | *** join/#asterisk mtnbkr (n=mtnbkr@c-67-165-9-234.hsd1.ct.comcast.net) |
| 06:13.27 | Laggy_McGee | er... you know |
| 06:13.33 | asterboy | memory? |
| 06:13.35 | X-Rob | Laggy_McGee, I use them _all the time_ |
| 06:13.40 | asterboy | disk space? |
| 06:13.41 | X-Rob | they're great machines |
| 06:13.43 | X-Rob | what distro? |
| 06:13.48 | Laggy_McGee | X-Rob: hell yeah! :) |
| 06:14.02 | Laggy_McGee | X-Rob: Gentoo (is there any other?) :) |
| 06:14.10 | X-Rob | Well, yes. Plenty. |
| 06:14.17 | X-Rob | Asterisk *heart* CentOS/WhiteBox/RHEL |
| 06:14.23 | X-Rob | what kernel? |
| 06:14.23 | Laggy_McGee | plenty of disk space... though I must mention it's a laptop hard drive |
| 06:14.32 | Laggy_McGee | 2.6.15, I think |
| 06:15.04 | X-Rob | Hmm, should be OK. I find 2.6.14.7 is stable as, but I haven't looked at 16, which has new funky timer features for ztdummy. |
| 06:15.06 | Laggy_McGee | asterboy: 512, I think... I'll check |
| 06:15.48 | Laggy_McGee | 2.6.14, the Gentoo hardened kernel |
| 06:15.59 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 06:16.16 | Laggy_McGee | asterboy: yes, 512 |
| 06:16.20 | asterboy | lots |
| 06:16.30 | Laggy_McGee | indeed |
| 06:16.43 | Laggy_McGee | what's the easiest way to test my network latency? |
| 06:16.49 | nick125_lappy | ping |
| 06:16.51 | X-Rob | Laggy_McGee, 'ping -f your-phones-ip' |
| 06:16.51 | asterboy | open up cli and watch sip debug |
| 06:16.55 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 06:17.01 | X-Rob | if you get a shitload of .'s there a problem with you rnetwork or phone |
| 06:17.36 | Laggy_McGee | asterboy: "chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call" |
| 06:18.03 | nick125_lappy | Laggy_McGee: try what X-Rob suggested.. |
| 06:18.52 | Laggy_McGee | X-Rob: how long is it supposed to take? |
| 06:19.08 | X-Rob | Laggy_McGee, it's ping-flooding |
| 06:19.12 | Laggy_McGee | only one . but it's taking a while |
| 06:19.12 | X-Rob | hit ^C |
| 06:19.44 | X-Rob | it's not a network issue |
| 06:19.50 | Laggy_McGee | PING toast.jesschadwick.com (192.168.0.2) 56(84) bytes of data. |
| 06:19.50 | Laggy_McGee | . |
| 06:19.50 | Laggy_McGee | --- toast.jesschadwick.com ping statistics --- |
| 06:19.52 | Laggy_McGee | 7054 packets transmitted, 7053 received, 0% packet loss, time 53680ms |
| 06:20.20 | X-Rob | if you had a bad network link, it would be ouputting a . for every lost packet |
| 06:20.27 | X-Rob | (Eg, try ping -f 1.1.1.1) |
| 06:20.28 | Laggy_McGee | ah |
| 06:20.45 | asterboy | but 53000ms is pretty high |
| 06:20.47 | X-Rob | no |
| 06:20.51 | X-Rob | that's the total run time |
| 06:20.58 | asterboy | oh shit ya |
| 06:21.02 | drray | cumlative |
| 06:21.07 | drray | or however you spell that |
| 06:21.16 | asterboy | p0rn! |
| 06:21.19 | Laggy_McGee | X-Rob: ah, I see |
| 06:21.28 | X-Rob | Laggy_McGee, has to be a kernel issue. |
| 06:21.34 | X-Rob | or possibly compler? |
| 06:21.35 | X-Rob | dunno |
| 06:21.39 | X-Rob | something's broken |
| 06:21.45 | X-Rob | try downloading the source fresh |
| 06:21.52 | X-Rob | from asterisk.org |
| 06:21.55 | X-Rob | clean 2.6.14.7 kernel |
| 06:21.57 | X-Rob | etc |
| 06:23.27 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:23.33 | kmilitzer | Morning all |
| 06:24.32 | asterboy | the retrans_pkt sounds like a key to debugging. |
| 06:24.52 | asterboy | why is sip retransmitting to maximum retries? |
| 06:25.28 | asterboy | are you traversing NAT? |
| 06:26.32 | Laggy_McGee | Yes |
| 06:27.03 | asterboy | make sure your ports are free and that sip.cfg has proper settings. |
| 06:27.06 | Laggy_McGee | My clients are in the 192.168.1.0/24 subnet and the server is 192.168.0.2, seperated by a router |
| 06:27.12 | asterboy | maybe time to paste your sip.conf in pastybin |
| 06:27.18 | Laggy_McGee | everything is open right now |
| 06:27.24 | asterboy | DMZ? |
| 06:27.39 | Laggy_McGee | Yes. the * server is in the DMZ |
| 06:27.45 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 06:27.58 | asterboy | might want to try it outside the router for shits and giggles. |
| 06:28.15 | Laggy_McGee | "outside the router" = on the same subnet? |
| 06:28.25 | X-Rob | points to asterboy for asking the right questions |
| 06:28.27 | X-Rob | Laggy_McGee, yes |
| 06:28.35 | X-Rob | have it working on the same physical network first |
| 06:28.41 | X-Rob | then move it |
| 06:28.45 | X-Rob | and fix what's breaking it |
| 06:33.32 | *** join/#asterisk somegeek (i=levin@unaffiliated/somegeek) |
| 06:45.35 | asterboy | ok, night all...thanks X-Rob for the push to make an outgoing context, I'll get back to my Bible and sstudy the basics again, then pick ur brain. |
| 07:00.45 | *** join/#asterisk Ansonmus (n=ahaeser@a213-84-26-148.adsl.xs4all.nl) |
| 07:01.15 | Ansonmus | Hello all, please can anyone look at this discussen and how to come to a solution? http://sourceforge.net/forum/forum.php?thread_id=1437025&forum_id=420324 |
| 07:03.24 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 07:07.58 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 07:07.58 | *** mode/#asterisk [+o denon] by ChanServ |
| 07:08.00 | X-Rob | Ansonmus, hassle Klaus? |
| 07:12.01 | *** join/#asterisk VEVE (n=VEVE3@ip56509b95.direct-adsl.nl) |
| 07:14.55 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 07:19.02 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@bzq-88-155-180-17.red.bezeqint.net) |
| 07:20.27 | Gamercjm | SIP doesnt seem to work from an external computer(outside the local network) how to I enable external? |
| 07:22.51 | chris_ast | whenever I call someone from extensions.conf I want to actually consider call as accepted only when the called user presses some key say #, please tell me how can I do this |
| 07:23.10 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:23.27 | chris_ast | Ansonmus, Gamercjm, psk: any idea on my question |
| 07:24.20 | Gamercjm | so they have to press # to accept the call? |
| 07:24.24 | Gamercjm | i dont get it |
| 07:24.49 | chris_ast | any key for example I used # |
| 07:25.05 | Gamercjm | so they have to press 'a button' to accept the call? |
| 07:25.14 | chris_ast | exactly |
| 07:25.25 | Gamercjm | umm |
| 07:25.29 | chris_ast | some input must be there from them |
| 07:25.47 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 07:25.53 | Gamercjm | well just dont answer the call :/ |
| 07:26.25 | chris_ast | bcoz even if they are in out of coverage area the message played by network is considered as accepted call right |
| 07:26.57 | PoWeRKiLL | Hi |
| 07:27.02 | chris_ast | RoyK, please help. My question is whenever I call someone from extensions.conf I want to actually consider call as accepted only when the called user presses some key say #, please tell me how can I do this |
| 07:27.48 | chris_ast | Gamercjm: did u get my question or am I still confusing u? |
| 07:27.58 | PoWeRKiLL | I have a silly bug sometime when receiving call from my zap interface and dialing a sip device I anwser on my sip device and for the caller coming from the zap interface it's still ringing any idea ? |
| 07:28.58 | Gamercjm | Im trying to work on a problem right now, cant go into details on someone else atm |
| 07:29.37 | chris_ast | whatz ur problem? |
| 07:29.52 | Gamercjm | (Gamercjm) SIP doesnt seem to work from an external computer(outside the local network) how to I enable external? |
| 07:30.10 | Gamercjm | so i can use like a softphone from an external computer |
| 07:30.22 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:30.32 | chris_ast | which softphone are u using |
| 07:30.52 | chris_ast | what is the server name u r giving |
| 07:33.51 | *** join/#asterisk powerchip (n=powerchi@197.80-202-229.nextgentel.com) |
| 07:34.25 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 07:34.51 | RoyK | chris_ast: I have no idea whatsoever |
| 07:36.02 | powerchip | I have a problem , random I call some I not hear they but they can hear me , but is ranom , not all time :S |
| 07:36.13 | powerchip | what can I do? |
| 07:36.47 | Corydon76-home | Isolate the problem |
| 07:36.58 | *** join/#asterisk exten123 (n=exten@60.49.6.190) |
| 07:37.04 | powerchip | how can I fix it? |
| 07:37.13 | exten123 | How to setup SIP NAT? |
| 07:37.50 | powerchip | I not have nat on:S |
| 07:38.50 | powerchip | firewall is off too |
| 07:38.51 | exten123 | powerchip, why? |
| 07:39.19 | powerchip | I ony use a firewall on linux |
| 07:39.22 | exten123 | powerchip, I have set my router to route port 5060 and rtp port to my asterisk but seen like cannot |
| 07:40.10 | powerchip | but me ruternot like rtp |
| 07:40.32 | powerchip | it is 510i |
| 07:41.23 | powerchip | Speed Touch |
| 07:42.41 | powerchip | so I have a Default Server to the asterisk box |
| 07:43.22 | powerchip | exten123: You know how fix it? |
| 07:45.02 | exten123 | powerchip,can teach me? I still new in this sector |
| 07:45.41 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 07:46.16 | powerchip | exten123: hm ok , try take off teh firewall |
| 07:46.21 | powerchip | the* |
| 07:46.34 | powerchip | it u have on on you ruter |
| 07:46.37 | exten123 | powerchip, I have make a range of port for my rtp does this work? |
| 07:47.01 | powerchip | exten123: not work for me , mabye for u |
| 07:47.27 | powerchip | some firewall not like rtp |
| 07:47.36 | powerchip | rtp go faste |
| 07:47.37 | exten123 | powerchip, how can take of firewall I stay in a building I use there internet. |
| 07:48.02 | powerchip | exten123: telenet you ruter |
| 07:48.05 | *** join/#asterisk KriS83 (n=KriS@212.202.141.92) |
| 07:48.20 | exten123 | powerchip, rtp is using UDP protocol right? |
| 07:48.23 | powerchip | Corydon76-home: how can I fix Isolate? |
| 07:48.31 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 07:49.30 | powerchip | 5060 (UDP og TCP) |
| 07:49.30 | powerchip | 16300-16700 (UDP) |
| 07:50.22 | *** join/#asterisk UnderMine (n=paddy@host81-134-63-109.in-addr.btopenworld.com) |
| 07:50.39 | UnderMine | morning |
| 07:50.43 | Syrus_ | look sip.conf and rtp.conf , u can fix ports |
| 07:52.36 | powerchip | Syrus_: hey can u help me? |
| 07:52.36 | exten123 | in sip.conf got this localnet what the purpose? set my local lan ip range? |
| 07:52.51 | UnderMine | anyone know anything about configuring ISDN2 in the uk? |
| 07:57.53 | UnderMine | on a AWM C2 i have got capi detecting the card and apperently working |
| 07:58.40 | UnderMine | but none channels can pick up or dial they just stay offline |
| 07:59.19 | *** join/#asterisk thomas____ (n=thomas_@ALyon-110-1-8-85.w80-14.abo.wanadoo.fr) |
| 07:59.21 | thomas____ | hello |
| 07:59.32 | UnderMine | morning |
| 07:59.42 | thomas____ | is there a special command in asterisk to purge master.csv file, or can i delete it by hand ? |
| 08:02.01 | UnderMine | any attempts to dial any of the ISDN DDIs gives me number unobtainable |
| 08:02.09 | Syrus_ | powerchip, maybe what's your pb ? |
| 08:02.27 | powerchip | random I call some I not hear they but they can hear me , but is ranom , not all time :S |
| 08:02.38 | powerchip | random* |
| 08:03.42 | Syrus_ | powerchip, call into your lan or over the net ? |
| 08:04.29 | powerchip | sip --> atserisk --> ata -> phone |
| 08:04.46 | powerchip | asterisk * |
| 08:04.49 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 08:05.00 | powerchip | the asterisk server is on lan |
| 08:05.33 | Syrus_ | ata and sip are on the same lan ? |
| 08:06.08 | powerchip | yes |
| 08:06.44 | Syrus_ | don't redirect ports 5060 and rtp ports on lan |
| 08:06.47 | powerchip | the sip server are from a noway comany ip24 |
| 08:06.54 | powerchip | norway |
| 08:06.56 | *** part/#asterisk terracon (n=tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
| 08:07.38 | Shaun2222 | it seams that if read() times out that it throw's a warning in asterisk |
| 08:07.41 | Shaun2222 | anyway to correct that |
| 08:08.46 | powerchip | ok |
| 08:09.43 | powerchip | how can I fix it? |
| 08:10.00 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 08:10.23 | puzzled | morning all |
| 08:10.31 | powerchip | I get a error then I debug |
| 08:10.35 | powerchip | morrnig |
| 08:10.40 | Syrus_ | look your firewall config, no redirect ports for rtp |
| 08:12.01 | powerchip | Call Leg/Transaction Does Not Exit " Bck from 10.0.0.9 |
| 08:12.21 | powerchip | I have it of now :S |
| 08:12.27 | powerchip | off* |
| 08:15.15 | *** join/#asterisk af_ (n=af@ip-164-94.sn2.eutelia.it) |
| 08:16.57 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 08:18.35 | UnderMine | ok need to find out which protocol DSS1, CT1, VN3, AUSTEL, 5ESS or NI1 the UK uses. Any clues? |
| 08:19.05 | *** join/#asterisk vopi (n=kkk@202.139.204.141) |
| 08:22.46 | puzzled | UnderMine: afaik euroisdn and in SS7 land their own BTNUP version last time I had to deal with it |
| 08:28.58 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:33.54 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 08:37.30 | *** join/#asterisk wiseguy_ (n=chivilis@infospalvos.lt) |
| 08:37.30 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 08:37.36 | wiseguy_ | hellow |
| 08:37.44 | wiseguy_ | im trying to install eicon diva card |
| 08:37.57 | wiseguy_ | when starting asterisk i get error - CAPI not installed, CAPI disabled! |
| 08:38.42 | Strom_M | solution: install CAPI |
| 08:39.05 | RoyK | ~seen zoa |
| 08:39.08 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 6d 13h 25m 21s ago, saying: 'toi tu il passe compose'. |
| 08:39.20 | wiseguy_ | Strom_M: capi ? |
| 08:39.41 | *** join/#asterisk Tili (i=Tili@61.140.191.13) |
| 08:43.45 | wiseguy_ | may chan_capi? i have installed it |
| 08:51.10 | UnderMine | Any ideas on debugging CAPI |
| 08:54.20 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:56.06 | *** join/#asterisk shiznatix (n=shiznati@213-35-236-110-dsl.end.estpak.ee) |
| 08:57.46 | shiznatix | hello. I am trying to fax with spandsp and the fax machine will ring but when it trys to establish a connection it fails and the call is hung up. Any help? |
| 09:01.06 | chris_ast | Strom_M,jbot,wiseguy_:Whenever I call someone from extensions.conf I want to actually consider call as accepted only when the called user presses some key say # (any key), please tell me how can I do that? |
| 09:01.27 | Strom_M | why? |
| 09:01.36 | Strom_M | what's wrong with good old supervision? |
| 09:02.00 | chris_ast | bcoz even if they are in out of coverage area the message played by network is considered as accepted call right? |
| 09:02.03 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:03.40 | vopi | hi all |
| 09:03.43 | chris_ast | Actually trying to similar to something given here http://www.voip-info.org/wiki-Asterisk+Tips+follow+me but do not how to use trunk and option c there |
| 09:07.47 | puzzled | shiznatix: don´t use spandsp version 0.3 with asterisk version 1.2.x |
| 09:08.15 | *** join/#asterisk redcap1 (i=redcap@xs3.xs4all.nl) |
| 09:09.13 | Strom_M | chris_ast, no |
| 09:09.27 | Strom_M | if you receive an error recording, the recording does not supervise |
| 09:09.43 | Strom_M | if it goes to voicemail, then yes, the call will supervise |
| 09:10.08 | Strom_M | but error recordings are not supposed to pass supervision back to the calling party |
| 09:10.50 | chris_ast | so we go to next priority |
| 09:11.05 | Strom_M | what? |
| 09:11.21 | Strom_M | do you understand what answer supervision is, chris_ast? |
| 09:11.40 | chris_ast | I did not |
| 09:11.45 | Strom_M | sigh |
| 09:13.49 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 09:14.03 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 09:14.13 | Strom_M | hello Ariel_ |
| 09:15.41 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 09:16.21 | *** join/#asterisk HuntrCkr (n=huntrckr@196.38.235.2) |
| 09:16.55 | HuntrCkr | i need some help |
| 09:17.44 | HuntrCkr | we currently have an old analogue pabx system in place |
| 09:18.03 | HuntrCkr | with analogue phones all over for the extensions |
| 09:18.17 | *** join/#asterisk k31th (n=keith@87.117.194.66) |
| 09:18.38 | shiznatix | puzzled, why can't i use 0.3 with asterisk 1.2.4? |
| 09:18.45 | Strom_M | HuntrCkr, ok |
| 09:18.51 | Strom_M | what do you need help with? |
| 09:19.12 | puzzled | shiznatix: because the author of spandsp said so |
| 09:19.18 | HuntrCkr | now, we are looking at either upgrading the pabx only, and getting TDM2400 cards for all the extensions, or upgrading everything to IP/SIP style phones |
| 09:19.49 | puzzled | HuntrCkr: get SIP phones. analog sucks :) |
| 09:19.52 | HuntrCkr | what hardware would one require in the asterisk servers to get the IP/SIP phones connected |
| 09:19.52 | Strom_M | HuntrCkr, how many stations are hanging off your pabx? |
| 09:20.08 | shiznatix | puzzled, ok no problems though, i am using spandsp-0.0.2 |
| 09:20.20 | Strom_M | HuntrCkr, to connect the IP phones, all you need is an ethernet interface |
| 09:20.21 | *** join/#asterisk Bambr (n=Bambr@213-35-236-110-dsl.end.estpak.ee) |
| 09:20.28 | HuntrCkr | Strom_M: 3 office, 30 exts, 20 exts, and 12 exts |
| 09:20.43 | HuntrCkr | Strom_M: incoming lines, 8, 8, 4 |
| 09:20.45 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 09:20.46 | Strom_M | three different offices? |
| 09:20.52 | HuntrCkr | yes |
| 09:20.54 | Strom_M | ok |
| 09:21.08 | Strom_M | what I would recommend is this: |
| 09:21.26 | Strom_M | rip out your analog lines and replace them with PRI |
| 09:21.54 | Strom_M | if you want IP phones, go for IP phones if there's a reliable enough data network to support them |
| 09:22.18 | HuntrCkr | hehe... ok Strom_M... slight problem though... |
| 09:22.30 | Strom_M | otherwise, since you're doing the PRI thing anyway, get a multi-span T1 card and some channel banks and hook your existing analog phones into the channel banks |
| 09:23.03 | HuntrCkr | I am in South Africa... don't have much choice about the incoming lines... analogue, and no other choices really |
| 09:23.28 | Strom_M | Telkom doesn't do ISDN in Johannesburg? |
| 09:23.30 | Strom_M | that's surprising |
| 09:24.17 | HuntrCkr | it does Strom_M, but it isn't cost effective at all... and the network quality is very poor |
| 09:24.31 | HuntrCkr | the 3 offices will be interlinked via VPN over ADSL |
| 09:24.43 | Strom_M | where are the three offices, out of curiosity? |
| 09:25.17 | HuntrCkr | Vereeniging( 80km south of Jhb about), Pretoria and Rustenburg |
| 09:25.28 | Strom_M | hm ok |
| 09:25.59 | Strom_M | do your offices have a data network that's fast enough and low-latency enough to support IP phones? |
| 09:26.22 | Strom_M | and is your network easy to set up such that data traffic from your PCs won't clobber your voip traffic? |
| 09:26.47 | HuntrCkr | not sure what that would be, but we have a pretty low load across a 100Mb Switched network in all three offices |
| 09:27.01 | Strom_M | alright |
| 09:27.18 | Strom_M | you may be able to get away with IP phones |
| 09:27.24 | HuntrCkr | very easy on the voip side... have Linux servers for gateways with QoS setup at all three offices |
| 09:27.27 | shiznatix | Does anyone know what the error "Call failed to go through, reason 3" means? |
| 09:27.29 | Strom_M | though the cost per-station will be lower if you keep your analog sets |
| 09:28.35 | Strom_M | dammit HuntrCkr, you've got me craving ostrich biltong now. I can't get decent ostrich biltong in los angeles :) |
| 09:29.02 | HuntrCkr | hmmm... thats what i was wondering... the TDM2400 cards for all the extensions add quite a bit to the cost, and was wondering what cards are required in the server to bring in the IP phones, if any |
| 09:29.18 | Strom_M | HuntrCkr, nothing but an ethernet interface |
| 09:29.25 | Strom_M | but then you're buying new phones for everyone |
| 09:29.59 | HuntrCkr | lol... i would be so glad if I could throw out this crap... half of them's 0 buttons don't even work properly anymore due to continuos use |
| 09:30.18 | Strom_M | so its roughly US$100 per station for the TDM2400, or US$150-US$300 per station for a good quality IP phone |
| 09:30.24 | Strom_M | ah heh |
| 09:30.40 | Strom_M | one install I did kept their analog phones, but they've got good quality phones |
| 09:32.15 | Strom_M | if you're going to be replacing the analog sets, then its more cost-effective to just go straight to IP sets |
| 09:33.32 | HuntrCkr | i figure the cost may be more for the IP sets, but hoping by cutting out all the FXO modules i need for the analogue extensions, i can get away with a slightly better phone for everyone |
| 09:33.48 | Strom_M | yeah |
| 09:34.10 | Strom_M | the cost really is in the modules |
| 09:34.45 | chris_ast | Strom_M: pls excuse for my ignorance, please tell me whether the call is considered as answered or not if network plays out of coverage message |
| 09:35.23 | Strom_M | chris_ast, if the network does not pass answer supervision back to your end when it plays the recording (as it shouldn't) then the call is not considered answered |
| 09:35.59 | chris_ast | Strom_M: ok :) |
| 09:36.03 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 09:36.35 | Strom_M | dammit, I have no idea where I can go to get good pizza by the slice at 2:30 AM |
| 09:37.03 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 09:38.23 | Splat | if you have a setup where you have multiple incoming lines and depending what line a call comes in on it gets answered differently.. what handsets would you look at? currently I'm tossing up between snom's and polycoms.. from what I understand the snoms have indicator lights to tell you which line is ringing where the polycom doesn't.. but I haven't had a chance to play with either a snom or a polycom.. (I only know not to use grandstre |
| 09:38.23 | Splat | am phones.. heh) |
| 09:39.05 | Strom_M | Splat, my cisco has 6 line appearances, and I use that feature to tell me how to answer my calls |
| 09:40.19 | Splat | do they interface well with asterisk? and I'm not sure I want to think about the price tag for around 9 handsets.. hehe |
| 09:40.25 | fourcheeze | Splat: you can do this with 1 line and munging the cli |
| 09:41.24 | fourcheeze | Splat: but certainly snoms, ciscos and polycoms are all capable |
| 09:42.57 | Splat | fourcheeze: they have 6 incoming analog lines.. so it's basically just routing based on which line the call is on.. but multiple people in each ring group.. I figure using a multi line phone will mean I just have the call goto a different line.. but I was told the polycom doesn't exactly tell you which line is ringing.. (I may be mistaken as I haven't actually had a chance to even see one up close yet) |
| 09:43.36 | fourcheeze | Splat: certainly the polycom 600 has a light next to each line |
| 09:43.41 | fourcheeze | which rings when the line rings |
| 09:43.54 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 09:44.00 | fourcheeze | so you'll need 6*9 sip accounts |
| 09:44.07 | Splat | so if I go for the 601's they'd do the job? |
| 09:44.10 | fourcheeze | to do it that way |
| 09:44.29 | Splat | fourcheeze: on a local asterisk box that's easy enough to do.. heh |
| 09:44.30 | fourcheeze | I think any multi-line phone at the price will do what you want, but it's not how I would do it |
| 09:44.55 | Splat | fourcheeze: how would you do it? |
| 09:45.01 | fourcheeze | well, it's not exactly scalable this way |
| 09:45.05 | fourcheeze | lets say they get another 6 lines |
| 09:45.12 | fourcheeze | and another 6 etc etc |
| 09:45.22 | fourcheeze | all I do is append something to the CLI |
| 09:45.39 | fourcheeze | so if it was 012345678 it becomes 012345678 [line1] |
| 09:46.05 | fourcheeze | then it's all on the display too |
| 09:46.14 | fourcheeze | rather than having to find the light |
| 09:46.16 | Splat | I can't see them getting more lines without more staff.. but yes I see what you're saying |
| 09:46.28 | fourcheeze | well more staff makes it even worse |
| 09:46.40 | fourcheeze | you might have a need for 12*18 sip accounts just for 18 staff |
| 09:47.00 | fourcheeze | not so much a problem to configure but it makes maintenance harder |
| 09:47.03 | Splat | every line wouldn't need an active sip connection would it? |
| 09:47.28 | fourcheeze | with asterisk every line needs at least one unique sip connection |
| 09:47.38 | fourcheeze | otherwise another phone will steal the registration |
| 09:48.03 | Splat | you should be able to have a phone that supports 6 lines but only have 2 currently setup until such time as you need more lines going to the phone.. |
| 09:48.18 | fourcheeze | oh sure |
| 09:48.32 | fourcheeze | but you can't use the same 2 for everyone |
| 09:48.40 | Splat | I know that |
| 09:48.47 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 09:48.53 | fourcheeze | ok, you can with sip proxies generally |
| 09:48.57 | fourcheeze | however |
| 09:49.05 | fourcheeze | that's just what I would do |
| 09:49.19 | fourcheeze | 1 account per phone unique to that phone and munge the CLI |
| 09:49.29 | fourcheeze | ring whichever combination of phones you want for each incoming line |
| 09:50.43 | Splat | hrmm |
| 09:51.29 | UnderMine | BRI in the uk...aarrggh |
| 09:51.38 | Modcuts | i have an issue where that using the line "exten => s,1,Macro(vm,402)" to send all calls after a time check to go to divert to vm but even though the directory has a recorded msg it plays the preset women? |
| 09:51.41 | Splat | well I really doubt I'd be putting in any single line phones.. I haven't even seen any decent single line phones.. the polycom 301 as 2 lines and wouldn't be suitable anyway.. would want atleast the 501s.. |
| 09:52.06 | fourcheeze | Splat: sure, use a nice phone |
| 09:52.07 | Strom_M | Modcuts, pastebin your macro please |
| 09:52.16 | Strom_M | ~pastebin |
| 09:52.19 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste |
| 09:53.07 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 09:55.40 | Modcuts | Storm_m: I thing is before i reseted the box and changed the ip it worked fine the marco works it's just it plays the women saying "the person at exten 401" instead of a the "prerecorded .wav" |
| 09:56.18 | Strom_M | Modcuts, pastebin the macro, please. |
| 09:59.23 | Modcuts | yep ok |
| 10:04.30 | *** join/#asterisk exten123 (n=exten@60.49.6.190) |
| 10:07.01 | *** join/#asterisk FlyboySR22 (n=rsears@sdtc.ar01.f2-40.host2.1.americanis.net) |
| 10:09.57 | *** join/#asterisk jserve (i=anwi73@p54BC960D.dip0.t-ipconnect.de) |
| 10:10.00 | jserve | Hi all |
| 10:10.43 | *** join/#asterisk andrebarbosa (n=andrebar@62.48.215.190) |
| 10:11.16 | andrebarbosa | anyone knows what happen to the lists.digium.com? most of the links are dead? :s |
| 10:11.48 | exten123 | guy for SIP 5060 port and RTP port both should be UDP right? |
| 10:12.11 | Strom_M | 5060 is TCP AFAIK |
| 10:12.38 | andrebarbosa | nop |
| 10:12.42 | andrebarbosa | 5060 is udp also |
| 10:13.03 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 10:13.11 | andrebarbosa | asterisk doesnt implement sip over tcp |
| 10:13.47 | Mavvie | doesn't support. |
| 10:14.01 | andrebarbosa | :) |
| 10:14.03 | exten123 | andrebarbosa, that mean my firewall should open udp rather then TCP right? |
| 10:14.17 | andrebarbosa | ya |
| 10:14.23 | exten123 | Strom_M, What you eman AFAIK? |
| 10:14.27 | RoyK | ~seen zoa |
| 10:14.35 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 6d 15h 48s ago, saying: 'toi tu il passe compose'. |
| 10:15.01 | exten123 | it's some others SIP 5060 run at TCP? |
| 10:16.53 | Strom_M | exten123, "As Far As I Know" |
| 10:18.05 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:19.20 | exten123 | Strom_M, Sorry asking stupid question. |
| 10:20.36 | exten123 | do you ppl CDR name as master.csv or you guy also got individual extensions.cvs file. Last time my asterisk only got master.csv file but now got individual. What setting make this happen? |
| 10:34.02 | x86 | exten123: i recommend you use MySQL for CDR |
| 10:34.11 | x86 | exten123: it's much easier to handle :) |
| 10:34.50 | Splat | I get given different csv's for each account code.. as well as the master.csv |
| 10:40.17 | *** join/#asterisk haikumore (n=haikumor@62-14-65-56.inversas.jazztel.es) |
| 10:45.10 | exten123 | Splat, why sometime is individual but sometime seperate? |
| 10:45.36 | exten123 | x86, it's diff to setup connection cdr to use MySql |
| 10:45.38 | chris_ast | For supporting 100 simultaneous calls what is the appropriate hardware we need? |
| 10:46.35 | chris_ast | Strom_M: :) your comments are needed again |
| 10:46.44 | Strom_M | ok |
| 10:47.02 | Splat | exten123: it's using billing codes |
| 10:48.23 | bsdfreak | <PROTECTED> |
| 10:48.49 | *** join/#asterisk Winkie (n=urmom@cpc2-stre1-6-0-cust119.bagu.cable.ntl.com) |
| 10:48.51 | Winkie | damn you freenode |
| 10:49.34 | wiseguy_ | damn you Winkie |
| 10:49.34 | wiseguy_ | ;-) |
| 10:51.19 | exten123 | got to go, thanks guy see you again. |
| 10:55.18 | *** join/#asterisk serverline (n=serverli@p54BA4551.dip0.t-ipconnect.de) |
| 11:03.31 | *** join/#asterisk vlrk (n=vlrk@59.93.78.141) |
| 11:03.44 | vlrk | has any body tried sipura provisioning |
| 11:04.28 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 11:05.40 | *** join/#asterisk Raffounz (n=raffounz@217.9.64.169) |
| 11:05.49 | Raffounz | good morning |
| 11:05.56 | Raffounz | does anyone know something about jingle's integration in asterisk? |
| 11:06.49 | *** join/#asterisk Greek-B0y (n=grb@193.220.93.162) |
| 11:07.10 | Greek-B0y | does anybody know of a VoIP provider that uses Asterisk for their sip server? |
| 11:08.20 | Rawplayer | Greek-B0y: does it matter? |
| 11:08.57 | *** join/#asterisk rsherwin (n=grimreap@203.177.13.60) |
| 11:09.09 | KriS83 | Could somebody explain to me exacly what a SER does? I don't quite get it? |
| 11:10.29 | rsherwin | after installing asterisk... how do i test if its installed correctly? |
| 11:10.59 | Strom_M | rsherwin: you start it up? |
| 11:11.18 | Greek-B0y | Rawplayer, not really. I'm just trying to determine if VoIP providers use Asterisk |
| 11:12.05 | rsherwin | Strom_M.... on the orielly book it mentioned " # modprobe zaptel |
| 11:12.06 | rsherwin | " |
| 11:12.10 | vlrk | has any body tried sipura provisioning |
| 11:12.15 | Strom_M | rsherwin, are you installing zaptel? |
| 11:12.31 | rsherwin | i already dun that |
| 11:12.36 | rsherwin | done |
| 11:12.38 | Strom_M | so now you start asterisk |
| 11:13.25 | rsherwin | i never started it... i was hoping after "make samples"... i can test if its installed properly |
| 11:13.48 | Strom_M | no, you have to start it u |
| 11:13.50 | Strom_M | er, up |
| 11:14.08 | Strom_M | if it starts up and doesn't die, it's installed correctly |
| 11:15.14 | rsherwin | ./asterisk <--- no params? |
| 11:15.37 | Winkie | -vvvc |
| 11:15.48 | Winkie | anyone got much experience with chan_agent and the manager API |
| 11:15.55 | Winkie | i'm trying to find a way to link agent calls to their Local/ counterparts |
| 11:16.24 | drray | uhm, modprobe zaptel.. ztcfg -vv, then start asterisk |
| 11:16.33 | drray | after you configure it |
| 11:17.29 | rsherwin | where is modprobe located? |
| 11:17.35 | redcap1 | I configured zaptel, but I think I have some issues on the S0 bus now :( |
| 11:17.36 | Winkie | /sbin/modprobe |
| 11:18.07 | redcap1 | sometimes I get a no service available message when I call my number ... |
| 11:18.36 | redcap1 | I'm getting these messages: 2 received TEI check request for TEI = 81 |
| 11:20.55 | rsherwin | hmmmm.. after running modprobe zaptel.... no running deamon when running ps -ax...right? |
| 11:21.05 | drray | lsmod |
| 11:21.17 | redcap1 | rsherwin: correct |
| 11:21.33 | rsherwin | where is lsmod? |
| 11:21.39 | drray | /sbin/lsmod |
| 11:21.52 | Strom_M | rsherwin, isn't all this in your path already? |
| 11:22.02 | Strom_M | or are you running ultra-barebones-linux? :) |
| 11:22.08 | redcap1 | :D |
| 11:22.20 | drray | ultar-newbie-linuxuser |
| 11:22.22 | rsherwin | this box giving me haedaches.... |
| 11:22.38 | redcap1 | hmm, more errors on my bristuff debug output: Apr 7 15:20:52 WARNING[28293]: chan_zap.c:8206 zt_pri_error: 2 TEI remove TEI = 127 |
| 11:22.41 | Strom_M | rsherwin, debian is your friend ;) |
| 11:22.58 | rsherwin | .... well this is a RH EL AS v4 Update 1 |
| 11:23.08 | rsherwin | i'm sort of BSD |
| 11:23.15 | drray | if he is having issues with RH, then Debian is not the answer |
| 11:23.54 | Strom_M | or perhaps Ubuntu, from the tribal African word meaning "I can't configure Red Hat!" |
| 11:24.11 | drray | hey.. I use ununtu |
| 11:24.15 | drray | er |
| 11:24.18 | Strom_M | i kid, i kid ;) |
| 11:24.22 | kmilitzer | I was just wondering (well actually I came across this before but never had time to check it) what exactley pbx_gtkconsole and/or kdeconsole does ... I can find no info what it does exactley ... |
| 11:24.25 | drray | on my laptop |
| 11:25.18 | rsherwin | [root@boses sbin]# ./lsmod | grep zaptel |
| 11:25.19 | rsherwin | zaptel 196740 0 |
| 11:25.19 | rsherwin | crc_ccitt 6081 1 zaptel |
| 11:25.22 | rsherwin | right? |
| 11:25.50 | Strom_M | rsherwin, do you have a zaptel card? |
| 11:25.55 | rsherwin | nope |
| 11:26.01 | drray | oops |
| 11:26.05 | Strom_M | ok, so you can skip the whole zaptel thing |
| 11:26.15 | drray | I guess I should have axed that |
| 11:26.18 | Strom_M | yes |
| 11:26.24 | Strom_M | important questions these are :) |
| 11:26.42 | rsherwin | yup... actually its just for learning it |
| 11:26.56 | rsherwin | since sooner i'll be getting the card |
| 11:26.59 | drray | well, he was axing about compiling zaptel, so axing if he had a zaptel card might have been snarky |
| 11:27.12 | Ikarus | Are there any configurables on the echo cancelling stuff (like selecting an algorithm, etc) other then the learn time ? |
| 11:27.29 | rsherwin | my intention for now was only SIP |
| 11:27.57 | *** join/#asterisk QbY (n=Kelvin@cm-12-146-225-110.dhcp.geo-sc.southerncoastalcable.net) |
| 11:27.59 | drray | actually, someone else was.. |
| 11:28.27 | QbY | anyone have any suggestion for queue reporting? calls answered, by whom, time wait, etc..... |
| 11:28.40 | rsherwin | so from the samples config i can run asterisk? |
| 11:28.47 | Strom_M | rsherwin, not really |
| 11:28.57 | Strom_M | you must modify the samples to match what you're actually doing |
| 11:29.08 | drray | you need a mickey mouse sip.conf and a basic extensions.conf |
| 11:29.08 | Strom_M | it seems you've already got the o'reilly book |
| 11:29.47 | rsherwin | yup i got the book |
| 11:30.06 | Ahrimanes | QbY: show queue <queue> on the cli? |
| 11:30.48 | rsherwin | will that sample not going to start asterisk... just for me to see it running? |
| 11:30.54 | QbY | Ahrimanes: No, more as in historical reporting.. |
| 11:31.09 | rsherwin | then as i move on... edit configs to my liking |
| 11:31.21 | Ahrimanes | QbY: ah.. hm dont know any of the call center solutions out there myself unfortunately |
| 11:31.22 | drray | I don't think you need samples |
| 11:31.27 | drray | (IMO) |
| 11:31.33 | stoffell | Ikarus, look in zconfig.h if you use zaptel |
| 11:31.39 | rsherwin | hmmmmmm |
| 11:32.00 | Ikarus | Ah well, I just discovered the voice quality with echo cancelling disabled is higher in most cases here |
| 11:32.26 | rsherwin | 2668 pts/1 S 0:00 bash |
| 11:32.28 | rsherwin | <PROTECTED> |
| 11:32.28 | rsherwin | <PROTECTED> |
| 11:32.33 | rsherwin | is that a good sign? |
| 11:32.39 | stoffell | Ikarus, you can also try ECHO_CAN_MG2 canceller |
| 11:32.40 | drray | yes now asterisk -r |
| 11:32.45 | redcap1 | yep, now do asterisk -r |
| 11:32.48 | redcap1 | :) |
| 11:32.54 | drray | then set verbose 10 |
| 11:33.26 | rsherwin | Connected to Asterisk 1.2.6 currently running on boses (pid = 2746) |
| 11:33.27 | rsherwin | boses*CLI> |
| 11:33.31 | rsherwin | very nice |
| 11:33.35 | drray | show channels |
| 11:33.46 | drray | now go edit sip.conf and extensions.conf |
| 11:34.03 | Ikarus | stoffell: yes, but if it works better without any cancelling I am not going to mess with recompiling the piece of shit (I am sorry, but the build system and source code of asterisk aswell as the config system, while better then commercial solutions still stink) |
| 11:34.17 | rsherwin | ok.... that is next |
| 11:34.23 | stoffell | Ikarus, indeed, if not needed, don't use, you're correct |
| 11:34.24 | Rawplayer | for what should i look when i want voip clients to dail to other phone numbers over my voip isp |
| 11:34.28 | rsherwin | thanx |
| 11:34.33 | drray | get your sip phone registered |
| 11:34.47 | *** join/#asterisk fulgas (n=fulgas@209.8.233.244) |
| 11:34.47 | Rawplayer | i can call to the other softphone's in my lan |
| 11:34.58 | Rawplayer | but not to external numbers |
| 11:35.01 | Rawplayer | it gets a 404 |
| 11:35.33 | Strom_M | Rawplayer, do you have dialplan rules for outbound calls? |
| 11:36.01 | Rawplayer | nope |
| 11:36.07 | Rawplayer | but thats what i needed to know |
| 11:36.10 | Rawplayer | thanks |
| 11:36.39 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 11:37.35 | drray | it appears turning off the timing has stopped my T1 from respanning every 10 minutes |
| 11:38.04 | drray | I was getting -- B-channel 0/7 successfully restarted on span 4 all damn day |
| 11:38.09 | Ikarus | stoffell: I just dug up a doc and it seems Asterisk's echo cancelling was actually interfering with that of the telco (or cellphone), need to remember that |
| 11:40.22 | *** join/#asterisk iDunno (i=brettp@miranda.sommitrealweird.co.uk) |
| 11:42.29 | *** join/#asterisk hatamen (n=hatamen@59.191.63.24) |
| 11:46.08 | stoffell | Ikarus, okay, cool... good to know that is. thanks |
| 11:46.38 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 11:47.36 | key2 | !seen kran |
| 11:47.39 | key2 | !seen kram |
| 11:48.22 | mut | ~seen kram |
| 11:48.25 | jbot | kram <n=mark@pdpc/sponsor/digium/kram> was last seen on IRC in channel #asterisk, 4d 8h 41m 46s ago, saying: 'oh most certainly :)'. |
| 11:48.27 | mut | uh |
| 11:48.32 | mut | hm |
| 11:48.49 | *** part/#asterisk KriS83 (n=KriS@212.202.141.92) |
| 11:52.49 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 11:57.12 | *** join/#asterisk yuta-vcnet (i=yuta-vcn@212.118.246.50) |
| 11:57.33 | yuta-vcnet | hi all, has someone tried Asterisk - vTiger integration? |
| 11:57.49 | *** join/#asterisk pavelnedved (n=afso7523@freedom.rd.francetelecom.com) |
| 11:58.27 | rsherwin | is there a test configuration of extensions.conf and sip.com? |
| 11:58.36 | rsherwin | sip.conf |
| 11:58.53 | *** join/#asterisk Jon335 (n=Jon335@unaffiliated/jon335) |
| 12:01.19 | inv_Arp | any page that explains the revised dialplan syntax for 1.2.x? |
| 12:03.46 | drray | the wiki has barebones |
| 12:04.30 | *** part/#asterisk pavelnedved (n=afso7523@freedom.rd.francetelecom.com) |
| 12:06.25 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
| 12:09.24 | FlyboySR22 | Good Morning Everyone |
| 12:09.44 | *** join/#asterisk coppice (n=chatzill@72.168.17.210.dyn.pacific.net.hk) |
| 12:12.03 | backblue | yuta-vcnet: what its the objective? |
| 12:12.37 | rsherwin | my sip phone cant connect... any pointer? |
| 12:12.55 | Strom_M | rsherwin, did you put the correct settings in sip.conf? |
| 12:13.05 | Strom_M | did you configure your sip phone the same way? |
| 12:13.09 | Strom_M | did you reload the configuration? |
| 12:13.16 | *** join/#asterisk Thazza (n=me@229.9.233.220.exetel.com.au) |
| 12:13.20 | Thazza | hey all |
| 12:13.59 | rsherwin | its a simple sip.conf |
| 12:14.03 | rsherwin | [root@boses sbin]# cat /etc/asterisk/sip.conf |
| 12:14.03 | rsherwin | [general] |
| 12:14.03 | rsherwin | context=default |
| 12:14.03 | rsherwin | srvlookup=yes |
| 12:14.03 | rsherwin | [sdc] |
| 12:14.03 | rsherwin | type=friend |
| 12:14.05 | rsherwin | secret=welcome |
| 12:14.07 | rsherwin | qualify=yes ; Qualify peer is no more than 2000 ms away |
| 12:14.09 | rsherwin | nat=no ; This phone is not natted |
| 12:14.10 | Thazza | what does this mean? |
| 12:14.11 | rsherwin | host=dynamic ; This device registers with us |
| 12:14.11 | Strom_M | rsherwin, SHUT UP |
| 12:14.12 | Thazza | handle_response_invite: Forbidden - wrong password on authentication for INVITE t |
| 12:14.13 | rsherwin | canreinvite=no ; Asterisk by default tries to redirect |
| 12:14.15 | rsherwin | context=internal ; the internal context controls what we can do |
| 12:14.16 | Strom_M | rsherwin, pastebin |
| 12:14.19 | Ahrimanes | eek nopaste or something please |
| 12:14.23 | Strom_M | ~pastebin |
| 12:14.25 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste |
| 12:14.28 | rsherwin | oooppsssss |
| 12:14.32 | rsherwin | sorry about that |
| 12:14.33 | Strom_M | rsherwin, DO NOT FLOOD THE CHANNEL AGAIN |
| 12:14.47 | Thazza | what does this mean? |
| 12:14.49 | Thazza | handle_response_invite: Forbidden - wrong password on authentication for INVITE t |
| 12:15.03 | Strom_M | Thazza, it means you have the wrong password |
| 12:15.04 | Strom_M | durh |
| 12:15.07 | drray | so any clue what would cause a B-channel to restart? my PRI provider says it should not do it.. it does not cut off the call or anything |
| 12:15.16 | *** join/#asterisk zignig (n=zignig@the.interthingy.com) |
| 12:15.34 | Thazza | Strom_M: yhea that is asip account i have, and i can recive calls via that sip account, just not make them. |
| 12:15.47 | Thazza | Strom_M: any ideas on what could be wrong? |
| 12:15.55 | rsherwin | to start with... is the sip.conf correct? |
| 12:15.59 | Strom_M | Thazza, make sure you didnt fuck up the password somehwere in your conf |
| 12:16.01 | zignig | just got FWD working on my asterisk box working, w00t. :) |
| 12:16.05 | Strom_M | rsherwin, pastebin it and I'll look |
| 12:16.19 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-96-30.telkomadsl.co.za) |
| 12:16.21 | Thazza | Strom_M: It has registered fine with the same password. |
| 12:16.36 | Strom_M | Thazza, sigh. pastebin your sip.conf |
| 12:17.24 | *** join/#asterisk saftsack (n=saftsack@p54A7CF86.dip.t-dialin.net) |
| 12:17.34 | tamp4x | if an asterisk machine trunked to you is going out g729 , dtmf rfc2833, and your asterisk machine apsses their calls out to a provider, but is ulaw and inband, will the dtmf conversion work properly? or does it need to be the same mode all the way through |
| 12:18.26 | zignig | tamp4x: what is it terminating onto ? |
| 12:18.57 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 12:18.58 | rsherwin | Storm_M.... http://pastebin.com/645907 |
| 12:19.09 | tamp4x | an acme sbc |
| 12:19.30 | zignig | tamp4x: a what ? :/ |
| 12:19.35 | tamp4x | sbc |
| 12:19.44 | Thazza | Strom_M: would prefer not to.. as it is split.. i will do a pastebin of the main settings for that account. |
| 12:20.42 | Strom_M | rsherwin, ok, and what settings do you have on your sip phone? |
| 12:21.35 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:21.50 | *** join/#asterisk KriS|work (n=KriS@isperoni.de) |
| 12:22.02 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 12:22.11 | rsherwin | on the sip phone... i just edit the.. enable=yes; display name = SDC; username = sdc; password = welcome; domain = ip-of-asterisk; sip proxy = ip-of-asterisk... right? |
| 12:22.18 | tamp4x | ?? |
| 12:22.24 | Thazza | Strom_M: http://pastebin.com/645910 |
| 12:22.40 | Thazza | Strom_M: does not include regsiter statment |
| 12:23.05 | zignig | tamp4x: what is an sbc ? |
| 12:23.25 | Strom_M | rsherwin, which sip phone are you using? |
| 12:23.29 | tamp4x | if you dont know you cant answer my question |
| 12:23.36 | tamp4x | session border controller |
| 12:24.08 | rsherwin | x-pro |
| 12:24.11 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 12:24.28 | Strom_M | rsherwin, asterisk is running, right? |
| 12:24.36 | rsherwin | yup |
| 12:24.38 | *** join/#asterisk x3me (n=x3me@201.11.226.31) |
| 12:24.39 | x3me | hi |
| 12:24.47 | Strom_M | did you reload your sip configuration or restart asterisk? |
| 12:24.53 | rsherwin | Connected to Asterisk 1.2.6 currently running on boses (pid = 2684) |
| 12:25.01 | rsherwin | both ... |
| 12:25.07 | *** part/#asterisk QbY (n=Kelvin@cm-12-146-225-110.dhcp.geo-sc.southerncoastalcable.net) |
| 12:25.23 | x3me | i can put an mp3 in "stand by" on my asterisk server and serve options to the caller ? |
| 12:25.36 | x3me | for example: "Dial X for blablabla" |
| 12:25.54 | x3me | and then the caller selects X, asterisk calls the specific extension |
| 12:26.07 | Strom_M | rsherwin, so...the phone isnt registering? |
| 12:26.17 | Strom_M | x3me, that's extremely easy to do |
| 12:26.30 | Strom_M | x3me, but you want to use wav files, not mp3s |
| 12:26.39 | x3me | ah, ok |
| 12:26.48 | x3me | did you known some documentation about this? |
| 12:27.04 | Strom_M | ~docs |
| 12:27.11 | jbot | docs is probably probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 12:27.28 | Strom_M | ~thebook |
| 12:27.32 | jbot | thebook is probably Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 12:27.58 | x3me | let me see |
| 12:28.40 | coppice | i thought thebook was something the police threw at serious offenders :-\ |
| 12:28.49 | Thazza | Strom_M: Any ideas on my bug? |
| 12:28.55 | zignig | coppice: no , that's the riot act. |
| 12:28.58 | zignig | coppice: :) |
| 12:29.06 | Strom_M | Thazza, kvetch at your SIP provider |
| 12:30.40 | Thazza | Strom_M: Thats what i was thinkning.. just can't work out why incomming work., it is just outgoing. |
| 12:30.54 | znoG | guys, silly question, yes, but the connector that plugs into a TDM2400 card would be a RJ21 male or female? |
| 12:31.10 | Thazza | Strom_M: Will give them a call in the morning if it is not fixed. It is 10:30pm at night here. |
| 12:31.23 | coppice | zignig: i thought they read them the riot act, or throw the book at them. either way they end up unconcious |
| 12:31.39 | Strom_M | znoG, standard 50-pin amphenol connector |
| 12:31.41 | zignig | coppice: ah yes, you're right. |
| 12:31.48 | Thazza | Strom_M: thanks.. have to run now. |
| 12:31.49 | Strom_M | znoG, go to Graybar or something :) |
| 12:31.56 | znoG | Strom_M: yep but i need the male or female? |
| 12:32.01 | Strom_M | the male |
| 12:32.03 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 12:32.07 | znoG | ok, thanks :) |
| 12:33.36 | SpaceBass | I have a CID question....there is no way to change the caller ID text is there? I know some providers allow setcallerID |
| 12:33.55 | Strom_M | SpaceBass, which country? |
| 12:34.00 | SpaceBass | US |
| 12:34.01 | SpaceBass | sorry |
| 12:34.12 | Strom_M | SpaceBass, IAX connection, or PRI? |
| 12:34.24 | SpaceBass | I was under the impression it was basically a reverse lookup...so not possable |
| 12:34.25 | SpaceBass | IAX |
| 12:34.49 | Strom_M | SpaceBass, yes, the calling party name is a CNAM lookup on the terminating end of the connection |
| 12:35.02 | *** join/#asterisk rsherwin (i=sample@203.177.13.60) |
| 12:35.13 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 12:35.29 | SpaceBass | so thats a 'yes you can change the number but not the name' ? |
| 12:35.42 | Strom_M | yyes |
| 12:35.47 | SpaceBass | thought so |
| 12:35.58 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 12:35.59 | xheliox | Hey all, using the latest version of Asterisk & Zaptel, with the TE210P, everything works perfect, except when I dial outbound over the PRI, when the call is answered, the 10 dtmf digits I used to dial the person is played back to the person on the called side. |
| 12:36.03 | rsherwin | Strom_M.... any clue as to why my sip phone login timeout? |
| 12:36.08 | xheliox | Any suggestions as to why that might be happening? |
| 12:36.16 | SpaceBass | damn...so I cant have the cisco 7912 in my master bathroom call out saying "Master Bathroom" :) |
| 12:36.33 | xheliox | Though the person who dialed cannot hear them. |
| 12:37.18 | Strom_M | rsherwin, is it just that the phone isnt registering? |
| 12:37.31 | [TK]D-Fender | xheliox : sounds like you're using an unusual mode of T1, not standard PRI signalling. |
| 12:37.33 | rsherwin | nope |
| 12:37.45 | x3me | Strom_M, what "name" the article of this subject maybe have ? |
| 12:37.50 | rsherwin | it says on the console ... Login Timed Out |
| 12:37.56 | x3me | "How to make an call center?" heh |
| 12:37.58 | Strom_M | rsherwin, paste the exact line |
| 12:38.02 | [TK]D-Fender | xheliox : pastebin your zaptel & zapata |
| 12:38.14 | xheliox | Stand by. |
| 12:38.23 | Strom_M | x3me, learn how to use contexts and the playback command |
| 12:38.41 | x3me | ok |
| 12:38.51 | *** part/#asterisk zignig (n=zignig@the.interthingy.com) |
| 12:38.57 | *** join/#asterisk thomas____ (n=thomas_@ALyon-110-1-8-85.w80-14.abo.wanadoo.fr) |
| 12:38.58 | thomas____ | hello |
| 12:39.01 | rsherwin | it says on the console ... Login Timed Out. Contact Network Admin |
| 12:39.08 | Strom_M | the asterisk console? |
| 12:39.20 | rsherwin | sip phone (x-pro) |
| 12:39.29 | Strom_M | try a different softphone |
| 12:39.32 | thomas____ | is there a way to know which extensions are having a call ? |
| 12:39.39 | rsherwin | ok |
| 12:39.43 | Strom_M | thomas____, show channels |
| 12:39.46 | thomas____ | i want to make a php script that displays all extensions that are free or in call |
| 12:40.10 | rsherwin | like what soft phone easy to configure? |
| 12:40.13 | thomas____ | yes but i want to extract this information from a script, is there an API or a socket that can give me this information |
| 12:40.20 | xheliox | [TK]D-Fender: http://pastebin.ca/48514 (zapata.conf) -- http://pastebin.ca/48516 (zaptel.conf) |
| 12:40.24 | Strom_M | thomas____, look up "manager interface" |
| 12:40.29 | [TK]D-Fender | thomas____ : Go read up on "Astrerisk Presence" on the WIKI. |
| 12:40.39 | thomas____ | thanks |
| 12:40.48 | *** join/#asterisk iDunno (i=brettp@miranda.sommitrealweird.co.uk) |
| 12:40.55 | Strom_M | rsherwin, sjphone perhaps? |
| 12:41.13 | rsherwin | ok..i'll look for that |
| 12:41.17 | tamp4x | if an asterisk machine trunked to you is going out g729 , dtmf rfc2833, and your asterisk machine apsses their calls out to a provider, but is ulaw and inband, will the dtmf conversion work properly? or does it need to be the same mode all the way through |
| 12:42.08 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-96-30.telkomadsl.co.za) |
| 12:43.04 | [TK]D-Fender | xheliox : you don't seem to have a switchtype defined, and you should clear out the unneccesay junk from there. Your switchtype should like be either national, or ni2 |
| 12:43.38 | [TK]D-Fender | tamp4x : * is a B2BUA and will convert side A to side B any way it has to. |
| 12:44.32 | xheliox | [TK]D-Fender: I'm fairly certain when I set it to national, which it is, it did not fix the issue. And when I turn on debugging, it detects the switch type as national, regardless of the setting. |
| 12:45.05 | Strom_M | xheliox, NI1 or NI2? |
| 12:45.55 | xheliox | [TK]D-Fender: Line 62, switchtype=national |
| 12:45.57 | xheliox | It's set. |
| 12:46.07 | xheliox | And it's using ni2, Strom. |
| 12:46.26 | *** join/#asterisk Thazza (n=me@229.9.233.220.exetel.com.au) |
| 12:46.38 | [TK]D-Fender | xheliox : hmmm, maybe somethings wrong on telco side... |
| 12:46.58 | xheliox | [TK]D-Fender: That's where I'm pointing the finger, but I wanted to check that with the experts. :) |
| 12:47.06 | *** join/#asterisk theHub (n=karlhubn@69.177.93.20) |
| 12:47.07 | [TK]D-Fender | xheliox : What you described sounds like a voice T1 using DTMF for DIDs. |
| 12:47.24 | [TK]D-Fender | xheliox : Even though your side looks more like a "normal" PRI |
| 12:47.41 | xheliox | Fair enough. I'll mention that to the tech when I talk to him. |
| 12:49.43 | xheliox | Thanks. :) |
| 12:51.46 | SplasPood | Hrm, it looks like milliwatt test lines for various POTS-style telcos are hard to come by |
| 12:52.12 | Strom_M | "pots-style telcos" |
| 12:52.20 | Strom_M | I'm asphyxiating from laughter |
| 12:52.27 | SplasPood | :) |
| 12:52.41 | SplasPood | yea man, they roll pots-style |
| 12:53.06 | *** part/#asterisk hatamen (n=hatamen@59.191.63.24) |
| 12:53.11 | rsherwin | Cannot dial phone number in current service configuration <------- |
| 12:53.20 | thomas____ | is there a simple software that can connect ton 5038 port (manager interface) to fetch informations about free extensions ? |
| 12:53.27 | rsherwin | thats from my sjphone |
| 12:53.44 | *** join/#asterisk vopi (n=kkk@202.139.208.242) |
| 12:54.12 | vopi | hi alls |
| 12:54.16 | *** join/#asterisk hatamen (n=hatamen@59.191.63.24) |
| 12:54.36 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 12:55.28 | UnderMine | hi |
| 12:56.51 | *** part/#asterisk hatamen (n=hatamen@59.191.63.24) |
| 12:56.57 | Winkie | guys i could REALLY do with some agent help if people are around |
| 12:57.08 | rsherwin | Strom_M... what info is need on the sjphone? |
| 12:57.14 | Winkie | i need to link together the Local / SIP section of an agentcallback with the Zap/Agent side |
| 12:57.18 | Winkie | there seems to be no linking event at all |
| 12:57.21 | mut | anyone ever used perl to ssh? |
| 12:57.27 | Strom_M | rsherwin, use the "sip proxy" setting |
| 12:57.32 | UnderMine | any ideas why a avm c2 would not appear to detect a connected line |
| 12:57.33 | *** join/#asterisk jack999 (n=jack@kbhn-vbrg-sr0-vl204-006.perspektivbredband.net) |
| 12:57.45 | Strom_M | rsherwin, you would really do well to step through the o'reilly asterisk book tutorial |
| 12:57.47 | Strom_M | ~thebook |
| 12:57.52 | jbot | from memory, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 12:58.28 | k31th | I was thinking, is there any thing out there that stores phonebooks for asterisk ? i.e i look on my website and i want to call ben, so i click bens name and it will dial MY phone then when i pick up it will dial ben ?? |
| 12:59.54 | [TK]D-Fender | k31th : yes, there are, go to the WIKI and look under "Asterisk GUI" and I believe you'll find info on some dialer apps out there |
| 12:59.59 | thomas____ | i don t find any API in asterisk manager that can give me all free extensions. how can i get the extensions status ? |
| 13:00.12 | k31th | [TK]D-Fender: thank you |
| 13:01.56 | Winkie | mut: oh hey, you got crypt installed? |
| 13:01.56 | [TK]D-Fender | thomas____ :I just told you to go look up "Asterisk Presence" on the WIKI... what you're looking for is there... |
| 13:02.03 | mut | yea |
| 13:02.07 | mut | i got it from some othe src |
| 13:02.13 | mut | soulcage or something like that |
| 13:02.18 | Winkie | oh, I PMd you a source too |
| 13:02.21 | mut | o |
| 13:02.22 | k31th | wat platform do you guys run asterisk on ? |
| 13:02.24 | Winkie | 17:53.44 <Winkie> http://pavleck.com/tech-support/tech-support-activestate-activeperl-ppm-error-error-ppd-for-crypt-desppd-could-not-be-found/ |
| 13:02.31 | Winkie | k31th: arch linux |
| 13:02.36 | k31th | distro wise ? centos / debian / gentoo ? |
| 13:02.44 | Winkie | arch |
| 13:02.45 | Strom_M | debian |
| 13:02.48 | mut | http://www.soulcage.net/ppds/?S=A |
| 13:02.48 | FlyboySR22 | k31th, Gentoo |
| 13:02.54 | coppice | the platforms I use are the floor and a large window sill |
| 13:03.03 | [TK]D-Fender | Slackware over here..... |
| 13:03.16 | [TK]D-Fender | k31th : You can run it on just about anything if you have the proper libs installed.... |
| 13:03.28 | k31th | yes i find alot of people use debian and gentoo these days. |
| 13:03.43 | k31th | [TK]D-Fender: yeah, i belive there is a centos package |
| 13:03.49 | austinnichols101 | centos here |
| 13:03.51 | [TK]D-Fender | k31th : Try and pick something "standard" (like a top 10 distro from distrowatch.com) and make sure to install the normal dev stuff... |
| 13:03.54 | k31th | infact there must be AAH runs on centos |
| 13:04.02 | coppice | I find a lot of people use Windows these days, but that's no recommendation |
| 13:04.06 | Winkie | mut: so what's your SSHing issue? |
| 13:04.16 | k31th | coppice: lol... |
| 13:04.23 | mut | Can't connect to 65.111.222.6, port 22: Unknown error at |
| 13:04.26 | mut | unknown |
| 13:04.26 | mut | heh |
| 13:04.28 | [TK]D-Fender | k31th : and AVOID packaging for *.... always just download from Digium's FTP. Packaging is good for supporting apps like MySQL, and others... |
| 13:04.38 | mut | on trying to create the connection |
| 13:05.27 | rsherwin | af the sip phone console say... sip: registered as sip:sip@123.123.123.123 ... is that ok? |
| 13:05.48 | bsdfreak | http://dojo.breakbeatninja.com:7777 <- tune in |
| 13:07.00 | Winkie | mut: that's interesting |
| 13:07.01 | k31th | [TK]D-Fender: i like to use packages to avoid breaking things when i upgrade ? I gues gentoo would keep this tidy as i can use there ebuild ? |
| 13:07.04 | Winkie | mut: you got your code? |
| 13:07.15 | mut | '80004005' |
| 13:07.26 | VEVE | I get errors with compiling zaptel-1.2.5 on a 2.6 Centos kernel. I know that this is RedHat bug. Who knows the work around?? |
| 13:07.46 | Winkie | mut: is that a pastebin id or something? |
| 13:07.50 | Winkie | VEVE: what's the error? |
| 13:08.13 | VEVE | i hve to check winkie |
| 13:08.26 | mut | http://pastebin.ca/48521 |
| 13:08.36 | k31th | VEVE: install DAG repos and install it with yum ? |
| 13:08.39 | [TK]D-Fender | k31th : Lets try this again... *avoid packaging Asterisk*..... |
| 13:08.47 | k31th | :D |
| 13:08.50 | k31th | ok |
| 13:08.56 | k31th | [TK]D-Fender: why tho ? |
| 13:09.04 | thomas____ | [TK]D-Fender, i didn t find anything on "asterisk presence". but if i type "show channels" in the asterisk prompt, i get what i want. I want to see this from a socket. |
| 13:09.22 | Winkie | mut: show me your code too? |
| 13:09.26 | [TK]D-Fender | k31th : Packagers can screw up and its just not worth it and may not always be compatible with your install... compiling it yourself lets it adapt to your system and has never "broken" anything for me. |
| 13:09.42 | mut | <PROTECTED> |
| 13:09.42 | mut | <PROTECTED> |
| 13:10.11 | k31th | [TK]D-Fender: wat about a gentoo ebuild ? |
| 13:10.13 | [TK]D-Fender | thomas____ : Try looking for "hints" ..... these are special entries in your extensions.conf that allow you to track the in-use status of extensions.... |
| 13:10.26 | [TK]D-Fender | k31th : I wouldn't.... |
| 13:10.30 | VEVE | DAG Repos ?????, I m not a linux expert, sorry |
| 13:11.13 | Winkie | mut: that's worrying that it just errors like that |
| 13:11.32 | coppice | Repo Man |
| 13:11.49 | austinnichols101 | download and compile yourself is definitely the way to go |
| 13:11.54 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 13:12.08 | austinnichols101 | I give it a 1.5 on the difficulty meter |
| 13:12.29 | Ariel_ | Morning folks |
| 13:12.29 | coppice | which is calibrated from 0.0 to 1.0 |
| 13:12.35 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 13:12.46 | *** join/#asterisk voiphil (n=chatzill@lego.emarketsouth.com) |
| 13:12.48 | rsherwin | Strom_M..... its working now... |
| 13:12.48 | austinnichols101 | yet regularly pegs out at 9 |
| 13:12.58 | rsherwin | i need a good dialplan for the test |
| 13:13.07 | Strom_M | rsherwin, write one |
| 13:13.19 | x3me | if i have an default extension called "s", how i call it ?! |
| 13:13.25 | x3me | i'm using softphones X-Lite |
| 13:13.32 | x3me | with an SIP extension configured |
| 13:13.39 | k31th | VEVE: well aparntly im wrong in suggesting that, perhaps you should try installing from source |
| 13:13.40 | voiphil | you don't, it's the default for a context |
| 13:13.53 | x3me | right, but how can i hear the messages ? |
| 13:14.22 | Strom_M | x3me, like i said, YOU NEED TO RTFM |
| 13:14.25 | UnderMine | avm c2 appear alive but unresponsive |
| 13:14.27 | voiphil | s is not a literal extension it is the default, kinda like x in algebra |
| 13:14.38 | Strom_M | ~thebook |
| 13:14.39 | jbot | somebody said thebook was Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 13:14.40 | Ariel_ | x3me, you send a call to that context that starts with the exten => s,1,bhah |
| 13:14.40 | VEVE | k31th what do you mean with installing from source. |
| 13:15.07 | Strom_M | x3me, http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:15.13 | Strom_M | READ THE BOOK |
| 13:15.40 | voiphil | x3me listen to Strom, that book is truly excellent and even better, it's excellent and free |
| 13:15.50 | mut | well |
| 13:15.56 | mut | lemme try at command line |
| 13:16.26 | HuntrCkr | Strom_M: Thanx for the help earlier... Just had a meeting with the MD, and the whole project seems to be a go with the IP phones... |
| 13:16.37 | Strom_M | HuntrCkr, cool |
| 13:16.39 | Strom_M | congrats |
| 13:16.46 | mut | eh same thign |
| 13:16.47 | UnderMine | capi sees it but it is perminently offline any ideas? |
| 13:17.34 | HuntrCkr | anyways, time to go home here :) |
| 13:17.48 | HuntrCkr | cheers guys... be seeing a lot of you soon :) |
| 13:18.47 | Strom_M | please god no - I've got the mentos jingle stuck in my head |
| 13:20.16 | *** join/#asterisk noky (n=Noky@200.69.211.18) |
| 13:20.19 | noky | hi |
| 13:20.46 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 13:21.10 | mut | ideas Winkie? |
| 13:21.20 | Winkie | mut: all i can think is check google, it's a little bizarre |
| 13:21.32 | mut | yea google isn't givin me much |
| 13:21.36 | Winkie | perhaps install a windows ssh client and use Net::SSH instead of the perl reimplimentation? |
| 13:22.45 | *** join/#asterisk coppice (n=chatzill@229.198.17.210.dyn.pacific.net.hk) |
| 13:23.05 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
| 13:23.40 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 13:23.55 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool149-149.nas31.salt-lake-city1.ut.us.da.qwest.net) |
| 13:24.28 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:25.08 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:25.08 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:28.37 | voiphil | I seem to be having a problem with hangup detection on a couple of extensions, in your experience is the <bye> getting lost and is there a way to detect, say, silence on the line for x number of seconds and then do a soft hangup at the pbx? |
| 13:29.12 | Strom_M | voiphil, the answer is to order far-end disconnect supervision on your analog lines |
| 13:29.18 | *** join/#asterisk AvoidingDeadlock (n=Avoiding@c-68-32-112-142.hsd1.md.comcast.net) |
| 13:30.05 | voiphil | Unfortunatel Strom that is not the case, these are SIP extensions, and my PBX trunks are all sip lines to my IP/TDM gateway (quintum cms) |
| 13:30.29 | Strom_M | voiphil, then something is wrong with your signaling |
| 13:30.56 | *** join/#asterisk bkw__ (n=bkw_@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 13:30.57 | noky | http://pastebin.com/645994 |
| 13:31.06 | noky | what is this "warnings"? |
| 13:31.18 | noky | is a queue call |
| 13:31.22 | voiphil | to be more specific, I think it may be a wiring issue, it happens to the same two stations consistently, we have switched IP phones both different models and different phones of the same model. |
| 13:31.47 | [TK]D-Fender | noky : I'd make sure your MP3's don't have ID3 tags... |
| 13:31.48 | voiphil | Also tried different logical extensions at the same physical port, same problem |
| 13:32.21 | noky | could be, i will check this |
| 13:32.22 | noky | thanks |
| 13:33.55 | voiphil | So I suppose that what I am really asking is for a kludge, since the <bye> seems to be getting lost somewhere in the network, is there a way to supervise those particular extensions (similar to absolute timeout) for silence detection and perform a soft hangup when the timer is reached? |
| 13:38.43 | noky | yes it is [TK]D-Fender thanks |
| 13:38.52 | *** join/#asterisk Lino` (i=Lino@i577BCC6C.versanet.de) |
| 13:39.48 | *** join/#asterisk fjean (n=fjean@201009180124.user.veloxzone.com.br) |
| 13:40.08 | fjean | hi guys ! |
| 13:41.01 | fjean | I need an IAX2 provider with good rates to Brazil that accepts paypal....it's king of urgent, any hint ? :- ) |
| 13:41.30 | Ariel_ | fjean, have you tried voipjet |
| 13:42.03 | fjean | ariel, yes but there cell is too expensive... |
| 13:42.37 | fjean | actually its the only one I used up to know... |
| 13:43.03 | fjean | :-) |
| 13:43.09 | voiphil | fjean, you will find that cell terminations are going to be expensive from everyone as the portion from the landline pop to the cell pop is charged by the cellular company |
| 13:43.57 | *** part/#asterisk chris_ast (n=Administ@59.93.56.163) |
| 13:43.58 | fjean | cool |
| 13:44.31 | Strom_M | brazil! |
| 13:44.39 | Strom_M | where hearts were entertaining June! |
| 13:44.53 | fjean | I have a couple here in Brazil for half the price but you hear half the words as well :-) |
| 13:44.55 | Strom_M | (anything to get that goddamned Mentos jingle out of my head) |
| 13:45.42 | fjean | samba ? |
| 13:45.53 | Strom_M | ? |
| 13:46.00 | VEVE | Help, Please can anybody explain how i can change the linux-2.6 symlink to /lib/modules/2.6.5.138/build in order to get zaptel compiled. |
| 13:46.13 | Hmmhesays | I use voipjet quite heavily |
| 13:46.14 | Strom_M | VEVE, install kernel headers |
| 13:46.36 | fjean | hhhesays - me too as a backup for proper.. |
| 13:46.38 | VEVE | strom_m: i do not get it |
| 13:47.26 | Strom_M | im not awake enough for this |
| 13:47.58 | SplasPood | Does anyone know if it's possible to have members of a queue that reside on cell phones and somehow when sending calls to them require they hit a key to accept (so calls don't end up in voicemail) before connecting the call? |
| 13:48.07 | Hmmhesays | what do you use for a primary route |
| 13:48.25 | fjean | mmhesays - providers here in brazil... |
| 13:49.07 | fjean | mmmhesays - but not for the quality, its mostly for the cost of it |
| 13:49.56 | wasim | SplasPood: ackcall |
| 13:50.18 | voiphil | SplasPood, not on the asterisk side, that sort of call acceptance would have to be offered by the cell provider |
| 13:51.05 | [TK]D-Fender | wasim : isn't ackcall restricted to those DIRECTLY looged into the queue by AgentLogin? |
| 13:51.28 | Hmmhesays | i have some customers in brazil |
| 13:52.20 | fjean | cool |
| 13:52.24 | *** part/#asterisk hensema (n=erik@scrat.hensema.net) |
| 13:53.10 | wasim | [TK]D-Fender: correct |
| 13:53.23 | Hmmhesays | still trying to get him to send me some brazillian liqour |
| 13:53.49 | [TK]D-Fender | wasim : no good for Cell phones.. I'm sure they don't want to stay on the phone sucking up batteries just waiting and being charged for air-time as well.... not really a viable idea in the vast majority of cases... |
| 13:54.20 | voiphil | Strom_M: Fresh and full of life (sorry, couldn't resist) |
| 13:54.32 | Strom_M | AHHHHHHHH DAMN YOU |
| 13:54.34 | SplasPood | [TK]D-Fender: Yea, exactly.. |
| 13:55.06 | wasim | SplasPood: i guess you could .call file it out |
| 13:55.21 | wasim | SplasPood: but it'd require some major surgery to get the call from the queue to a call file |
| 13:55.54 | mut | fouind the problem Winkie |
| 13:55.58 | mut | firewall :P |
| 13:56.23 | Hmmhesays | firewalls are fun |
| 13:56.50 | Hmmhesays | especially when you have a braindead admin |
| 13:56.54 | fjean | mmmhesays, you can send me an email if you want to check www . sunfone . com . br in the contacts, Ill get it |
| 13:57.02 | SplasPood | wasim: so much for allowing people on cell phones to be part of the support queue |
| 13:57.03 | mut | well i was trying other crap, got a msg with IO::Socket:Connect in it |
| 13:57.05 | fjean | sory have to go, bye for now |
| 13:57.06 | mut | and it just clicked |
| 13:57.10 | *** part/#asterisk fjean (n=fjean@201009180124.user.veloxzone.com.br) |
| 13:57.20 | mut | oh yea, firewall won't let ssh out on that server |
| 13:57.23 | mut | :P |
| 13:57.39 | mut | new problem, won't accepot my login now |
| 13:57.53 | *** part/#asterisk voiphil (n=chatzill@lego.emarketsouth.com) |
| 13:58.01 | SplasPood | wasim: Hrm.. Dial() can call a macro before the lines are bridged... I could add Local/ channels to the queue... and have the macro prompt the called party for input.. |
| 13:58.43 | SplasPood | cause then I could return MACRO_RESULT CONGESTION if they don't and I'm guessing the call would remain in queue? |
| 13:59.30 | [TK]D-Fender | SplasPood : I'm not sure which side of the call gets the macro though.... might work otherwise... |
| 14:00.13 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:01.46 | SplasPood | [TK]D-Fender: according to show application Dial, the called leg |
| 14:02.24 | noky | can i autologin a agents without dial the extension that have the 'AgentCallbackLogin' ? |
| 14:03.21 | Hmmhesays | noky, why were you spamming me yesterday |
| 14:03.48 | [TK]D-Fender | SplasPood : Sounds viable then... dump in an ivr with a timeout on it. |
| 14:03.55 | noky | ? |
| 14:04.50 | UnderMine | Anyone understand ISDN2/BRI? |
| 14:06.17 | [TK]D-Fender | Hmmhesays : Spent a few minutes learning some Kim Mitchell oldies.... great stuff to "rock out" to... |
| 14:06.29 | Hmmhesays | [TK]D-Fender, might as well go for a soda |
| 14:06.37 | [TK]D-Fender | Nobody hurt... |
| 14:07.21 | eric_hill | UnderMine: understand what? |
| 14:07.26 | Hmmhesays | haha, I saw a band play that not to long ago |
| 14:07.41 | *** join/#asterisk Creperum (n=ilya@mail.tex.kiev.ua) |
| 14:07.42 | Hmmhesays | i was working on lady picture show by stp last night |
| 14:07.58 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:08.25 | [TK]D-Fender | Hmmhesays : Just got through the into and first bit of that one, but failed to find tabs for "Rock'n'Roll Duty" and don't ahve the MP3's |
| 14:09.22 | Hmmhesays | I might have it |
| 14:11.53 | Hmmhesays | ahh thats a negative |
| 14:12.13 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:12.33 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 14:13.34 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 14:13.51 | noky | can i autologin a agents without dial the extension that have the 'AgentCallbackLogin' ? |
| 14:14.42 | eric_hill | noky: I'm working on this very thing: exten => _*223,1,Macro(automatic_acd_logon_logoff|test_queue|SIP/${CUT(CALLERID(num),-,1)}) |
| 14:15.06 | eric_hill | noky: in the macro... exten => s,1,AddQueueMember(${ARG1}|${ARG2}) |
| 14:15.08 | *** join/#asterisk nothinman (i=shakey@adan81.neoplus.adsl.tpnet.pl) |
| 14:15.13 | nothinman | hey, ho. |
| 14:15.43 | Hmmhesays | WTF was that noky, i should pastebin that |
| 14:16.20 | noky | is it work ? |
| 14:16.28 | *** join/#asterisk brodiem (i=0@cpe-66-69-222-36.austin.res.rr.com) |
| 14:16.32 | nothinman | quick question ;) how can I handle "i" extension? I mean if someone presses incorrect key -> do nothing. It is just disconnecting me now. If I add exten => i,1,NoOp(blah) it doesn't work as it should... |
| 14:17.33 | brodiem | nothinman, what cmd are you using that would point to exten => i if incorrect key is pressed? |
| 14:18.22 | *** join/#asterisk zaf (n=zaf@wsip-68-228-9-79.br.br.cox.net) |
| 14:18.50 | Ariel_ | nothinman, exten => i,1,goto(s,1) or where you want it to go. NoOp does thing but display it on the cli screen |
| 14:19.09 | nothinman | if the key is correct, it executes goto(), and it does work. But if I press for example 7, which has not been defined, it disconnects. And I don't want asterisk to do anything. But just don't disconnect me. |
| 14:19.10 | taec | Is there a way through the API or another standard method to get the number of voicemails in a persons mailbox? Or do I need to physically go on disk to do that? |
| 14:19.23 | [TK]D-Fender | nothinman : have it go like this "exten => i,1,Gotot(s,5)" or whatever the last line of "s" is |
| 14:19.43 | nothinman | it's a bit complicated ivr menu... |
| 14:19.55 | Hmmhesays | lets go |
| 14:19.59 | [TK]D-Fender | nothinman : I doubt its that bad.... |
| 14:20.09 | [TK]D-Fender | nothinman : pastebin it and show us. |
| 14:20.12 | [TK]D-Fender | ~pb |
| 14:20.14 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 14:20.14 | brodiem | taec, show voicemail users |
| 14:20.20 | nothinman | I can paste few lines... |
| 14:20.22 | nothinman | [my-ivr-51-3] |
| 14:20.22 | nothinman | <PROTECTED> |
| 14:20.22 | nothinman | <PROTECTED> |
| 14:20.22 | nothinman | <PROTECTED> |
| 14:20.22 | nothinman | <PROTECTED> |
| 14:20.23 | nothinman | <PROTECTED> |
| 14:20.25 | nothinman | <PROTECTED> |
| 14:20.27 | nothinman | <PROTECTED> |
| 14:20.29 | nothinman | <PROTECTED> |
| 14:20.31 | nothinman | <PROTECTED> |
| 14:20.33 | nothinman | <PROTECTED> |
| 14:20.35 | nothinman | <PROTECTED> |
| 14:20.37 | nothinman | <PROTECTED> |
| 14:20.39 | nothinman | <PROTECTED> |
| 14:20.54 | shiznatix | sombody kill nothinman |
| 14:20.56 | Hmmhesays | ~pb |
| 14:20.57 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 14:21.04 | hackeron | all of a sudden, asterisk stopped detecting call pickup, I start talking to person, after 90 seconds I see Nobody picked up in 90000ms and asterisk hangs up. Any ideas? |
| 14:21.18 | [TK]D-Fender | PASTEBIN! |
| 14:21.23 | nothinman | shiznatix? |
| 14:21.25 | hackeron | I'm using zapata with verizon analog lines |
| 14:21.29 | hackeron | and it was all fine yesterday |
| 14:21.41 | taec | brodiem, Nah, MailboxCount is what I was looking for ... through the API. |
| 14:21.42 | Hmmhesays | zaptel isn't detecting the answer |
| 14:21.46 | taec | brodiem, thanks though :) |
| 14:21.48 | hackeron | its a TDM400 card |
| 14:22.07 | [TK]D-Fender | nothinman : that IVR just goes in circles and doesn't allow input.... |
| 14:22.42 | nothinman | [TK]D-Fender: sorry man, it does work ;) |
| 14:22.51 | [TK]D-Fender | nothinman : s,10 just loops right back and preclude input... nice idea on trying to limit the # of failed appemts, but thats not QUITE the way... |
| 14:22.59 | nothinman | [TK]D-Fender: it repeats option 3 times and disconnects |
| 14:23.23 | nothinman | eem |
| 14:23.26 | [TK]D-Fender | nothinman : I wish I could see HOW that can work... here, I'll show you the more appropriate way... |
| 14:23.27 | nothinman | so what is the way? |
| 14:23.39 | triple-e | Hey |
| 14:24.01 | triple-e | anyone know how to print to the CLI from an AGI script |
| 14:24.08 | triple-e | standard print won't do it |
| 14:24.17 | [TK]D-Fender | nothinman : http://pastebin.ca/48532 |
| 14:25.20 | Hmmhesays | stdout |
| 14:25.28 | brettnem | hey all, having a weird asterisk problem |
| 14:25.29 | nothinman | [TK]D-Fender: it's almost the same ;) |
| 14:25.40 | brettnem | all of a sudden asterisk isn't responding to anything.. I've restarted it a few times |
| 14:25.58 | nothinman | [TK]D-Fender: it really does work, I've got about 400 different options now, and it does the work. |
| 14:25.59 | Hmmhesays | if you're using the safe_asterisk script, kill that first then kill asterisk |
| 14:26.03 | triple-e | Hmmhesays: isnt that the same as a print statement |
| 14:26.12 | brettnem | tethereal shows a lot of packets coming in, but asterisk never replies.. any ideas??! |
| 14:26.22 | nothinman | [TK]D-Fender: for now, the problem is handling invalid keys/options ("i" extension) |
| 14:26.25 | Hmmhesays | will if print doesn't work and stdout does, then no it's not |
| 14:26.27 | [TK]D-Fender | nothinman : close, but I still don't see how yours doesn't just infinitely loop like it look like it does |
| 14:26.59 | fourcheeze | brettnem: anything on the console? |
| 14:27.02 | [TK]D-Fender | nothinman : Well you see in mine how I have a counter on failed attempts... just try it my way :) |
| 14:27.17 | brettnem | fourcheeze: just shows a bunch of unreachable peers.. every now and then 1 shows up.. |
| 14:27.26 | brettnem | fourcheeze: but showing about 150 down right now |
| 14:27.44 | brettnem | why would asterisk simply not respond.. |
| 14:27.54 | brettnem | I manually unloaded chan_sip.so and reloaded it and didn't see any errors. |
| 14:28.09 | fourcheeze | brettnem: netstat -anup |
| 14:28.24 | fourcheeze | got any ports there? |
| 14:28.37 | brettnem | lots |
| 14:28.47 | fourcheeze | ok, but got your asterisk ones? |
| 14:28.48 | brettnem | asterisk is on 24 of them |
| 14:29.05 | brettnem | I do see asteisk on 5060 |
| 14:29.12 | fourcheeze | hmmm |
| 14:29.15 | Hmmhesays | reboot |
| 14:29.26 | brettnem | reboot?! |
| 14:29.30 | brettnem | like power off/on? |
| 14:29.35 | nothinman | [TK]D-Fender: it doesn't loop infinitely ;) try to analyze it ;) |
| 14:29.37 | fourcheeze | shutdown -r now |
| 14:29.46 | brettnem | why should it need a reboot? |
| 14:29.50 | fourcheeze | it shouldn't |
| 14:29.58 | brettnem | um |
| 14:30.10 | fourcheeze | but at least it's doing something while you're thinking :-) |
| 14:30.16 | brettnem | ok, good point.. |
| 14:30.33 | brettnem | oh.. :( |
| 14:30.37 | brettnem | it's been up for 377 days! |
| 14:30.44 | fourcheeze | :-) |
| 14:30.54 | nothinman | [TK]D-Fender: anyway, in your example if they press wrong key they are returned to the begining. I want it to take no action |
| 14:31.21 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 14:31.22 | x3me | hi |
| 14:31.26 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-96-30.telkomadsl.co.za) |
| 14:31.37 | x3me | i want to listen the full message before choose an option |
| 14:31.47 | brettnem | I so don't want to reboot it.. |
| 14:31.48 | x3me | i try to put an Wait(2) in extensions.conf but isn't works |
| 14:31.57 | x3me | i'm using Backgroud to play the sound |
| 14:32.04 | brettnem | :( |
| 14:32.04 | brettnem | 248 sip peers [10 online , 238 offline] |
| 14:32.08 | x3me | and if the caller digit "0" he dials to an another extension |
| 14:32.09 | brettnem | what the heck |
| 14:32.13 | brettnem | I can ping all those peers. |
| 14:32.20 | brettnem | and I see them all trying to register.. |
| 14:32.48 | fourcheeze | brettnem: where are your peers stored? |
| 14:33.01 | brettnem | WAHOO! I got it |
| 14:33.08 | brettnem | what the heck? |
| 14:33.16 | brettnem | peers stored? it's all sip.conf |
| 14:33.22 | fourcheeze | ok |
| 14:33.24 | fourcheeze | what was it? |
| 14:33.27 | brettnem | I restarted the network on the box.. why would that do it? |
| 14:33.34 | [TK]D-Fender | nothinman : then just change the goto to s,8 |
| 14:33.54 | brettnem | .. /etc/init.d/network restart |
| 14:34.05 | fourcheeze | did you reach some system limit? |
| 14:34.33 | brettnem | <big shrug> |
| 14:34.35 | [TK]D-Fender | nothinman : jumps to a neutral line and lets "s" re-terminate and start the timouts again |
| 14:34.39 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
| 14:34.52 | brettnem | it hard to really debug when you got that many phones down.. just got to get it working again.. and so now I don't know what happened.. |
| 14:34.54 | brettnem | you know.. |
| 14:35.02 | brettnem | I did see something weird in the netstat list.. |
| 14:35.03 | fourcheeze | yeah |
| 14:35.09 | fourcheeze | ok |
| 14:35.12 | fourcheeze | what was that? |
| 14:35.16 | brettnem | I saw asterisk on 5060 twice on the same PID |
| 14:35.24 | brettnem | same IP |
| 14:35.27 | fourcheeze | interesting |
| 14:35.28 | brettnem | well 0.0.0.0 on both |
| 14:35.39 | brettnem | yeah.. maybe a hung process was remaining bound to the port? |
| 14:35.42 | brettnem | freaky |
| 14:35.43 | fourcheeze | yeah |
| 14:35.46 | fourcheeze | sounds like that |
| 14:35.49 | fourcheeze | I had something similar once |
| 14:35.57 | fourcheeze | an asterisk that wouldn't die |
| 14:36.00 | brettnem | 377 days.. I had no idea.. nice |
| 14:36.07 | brettnem | it died just fine.. which was weird |
| 14:36.24 | fourcheeze | maybe it was waiting for something to happen on the network |
| 14:36.33 | fourcheeze | and restarting the network gave it a little kick |
| 14:36.38 | brettnem | heh, it was happening.. over 200 peers trying to register at once.. |
| 14:36.55 | fourcheeze | jooi what spec system is that? |
| 14:37.17 | brettnem | hmm |
| 14:37.27 | brettnem | nothing special really.. I'm all G711 and all SIP |
| 14:37.31 | fourcheeze | right |
| 14:37.42 | fourcheeze | what's your usual number of concurrent calls? |
| 14:37.57 | brettnem | Intel(R) Pentium(R) 4 CPU 3.20GHz |
| 14:38.05 | brettnem | like maybe 20 |
| 14:38.15 | fourcheeze | and how much load do you get? |
| 14:38.17 | nothinman | [TK]D-Fender: yeah, that would be the best idea. I've just done it. Cheers. |
| 14:38.24 | brettnem | fourcheeze: next to nothing.. |
| 14:38.27 | fourcheeze | cool |
| 14:38.36 | brettnem | yeah.. I usually don't have that many registered.. |
| 14:38.51 | brettnem | 96 registered right now |
| 14:38.52 | nothinman | btw anyone knows where I can buy 8x FXO card for asterisk? |
| 14:39.07 | fourcheeze | brettnem: did asterisk stop responding before or after you restarted it? |
| 14:39.14 | brettnem | before.. |
| 14:39.21 | fourcheeze | bit weird then |
| 14:39.25 | brettnem | it wasn't responding.. so I restarted it. |
| 14:39.28 | UnderMine | eric_hill: I have a AVM C2 which appears to be configured but does not work |
| 14:41.36 | eric_hill | UnderMine: how do you know if it's configured? Do you get channels in asterisk? |
| 14:42.26 | UnderMine | eric_hill: Contr1: 2 B channels total, 2 B channels free. |
| 14:42.40 | UnderMine | eric_hill: Contr2: 2 B channels total, 2 B channels free. |
| 14:43.04 | UnderMine | kisdnwatch shows the same but says they are both offline |
| 14:45.23 | UnderMine | eric_hill: when I try to use the channel i get CAPI INFO 0x3302: Protocol error layer 2 |
| 14:46.32 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-67.rockynet.com) |
| 14:46.58 | eric_hill | UnderMine: Is this the same problem? http://groups.google.com/group/Asterisk-users/browse_thread/thread/f46e4a265102600a/fe1d7796f2f06bfd%23fe1d7796f2f06bfd |
| 14:47.10 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 14:47.44 | brodiem | anyone know how to generate a UserEvent NOT using dialplans? |
| 14:48.29 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 14:50.27 | *** join/#asterisk stoffell (n=stoffell@d51A582C0.access.telenet.be) |
| 14:50.32 | UnderMine | eric_hill: looks similar will have good look at it. |
| 14:51.01 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:51.15 | ManxPower | brodiem, how do you want to generate a UserEvent? |
| 14:53.47 | *** join/#asterisk nitram (i=foo@superblob.com) |
| 14:54.15 | *** join/#asterisk Gunnar (n=gunnar@212.130.108.131) |
| 14:54.24 | Hmmhesays | telepathy? |
| 14:56.37 | *** join/#asterisk Ekushey (i=blackmor@fedora/Ekushey) |
| 14:57.28 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-96-30.telkomadsl.co.za) |
| 14:58.17 | brodiem | ManxPower, preferably with PHP, basically I need to be able to create it on-demand, without needing to make a call, so CLI is even fine |
| 14:59.23 | *** part/#asterisk Ekushey (i=blackmor@fedora/Ekushey) |
| 14:59.30 | brodiem | ManxPower, I want to change how DND is turned on/off. With the snom phones, you can specify an action URL when the DND button is pressed. I need the URL to the action URL to be able to send a UserEvent so that FOP recognizes the ext went on DND |
| 15:00.27 | ManxPower | brodiem, I have no suggestions |
| 15:00.42 | *** part/#asterisk bkw_ (n=Avoiding@c-68-32-112-142.hsd1.md.comcast.net) |
| 15:00.58 | [TK]D-Fender | brodiem : Why avoid the dialplan entirely then? just have the CLI issue a local channel call with no interactivity about it besides issueing the UserEvent |
| 15:01.32 | *** join/#asterisk ldnblk (n=ldnblk@host217-34-122-211.in-addr.btopenworld.com) |
| 15:02.39 | brodiem | [TK]D-Fender, I had a little bit of confusion with that. If I create a call file coming from the extension turning DND on/off (i.e. SIP/XXX), then won't SIP/XXX ring when that channel gets answered to connect them together? |
| 15:03.59 | Modcuts | Does anyone know where the unavailble vm message is kept? |
| 15:04.01 | [TK]D-Fender | brodiem : don't call the phone, just something like "Local/123@dndtoggle" which won't actually dial anything. |
| 15:04.18 | [TK]D-Fender | and just issue your userevent and hangup;. |
| 15:04.31 | brodiem | hmm |
| 15:04.48 | *** part/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 15:06.02 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 15:06.02 | brodiem | FOP receives its notification using: "UserEvent(ASTDB|Family: dnd^State: On)". Does anyone know how it knows which Ext to associate that with? I'm just wondering if it'll still work if the source channel isn't the actual SIP channel I need to toggle DND for |
| 15:06.44 | *** part/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:07.47 | *** join/#asterisk salviadud (n=ralfalfa@201.135.13.124) |
| 15:08.28 | brodiem | or maybe if it receives the userevent it looks in the the DND DB |
| 15:08.34 | brodiem | which I cna specifically set |
| 15:08.58 | *** join/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
| 15:10.04 | noky | how to set a condition to dialpeer in a extensions.conf ? like: if (${CALLERID} == 12) Goto ext1 else Goto ext2; |
| 15:10.13 | [TK]D-Fender | yup, that sounds like the way... |
| 15:11.21 | triple-e | noky: gotoif |
| 15:11.38 | *** join/#asterisk antoniofcano (n=antoniof@5.Red-80-32-90.staticIP.rima-tde.net) |
| 15:12.38 | UnderMine | eric_hill: definately an improvement |
| 15:12.42 | noky | thanks triple-e |
| 15:12.46 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 15:13.18 | UnderMine | eric_hill: ISDN2: Call CAPI/ISDN2/b07970532659-5 with B3 overlap (pres=0x00, ton=0x00) |
| 15:13.54 | UnderMine | now getting an engaged tone when I try to call out |
| 15:14.02 | *** part/#asterisk antoniofcano (n=antoniof@5.Red-80-32-90.staticIP.rima-tde.net) |
| 15:15.46 | fourcheeze | Hmmhesays: you alive today? |
| 15:16.41 | eric_hill | what does "engaged tone" mean? Dial tone? |
| 15:17.15 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
| 15:17.20 | Katty | hi lads. |
| 15:17.27 | eric_hill | Katty! |
| 15:17.47 | eric_hill | hrm... doesn't quite sound the same as "Norm!" |
| 15:18.07 | fourcheeze | eric_hill: engaged is like a busy tone, but you're planning to make it permanent |
| 15:18.24 | iDunno | morning Katty |
| 15:18.26 | Katty | eric_hill: uhh, hi. |
| 15:18.40 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 15:19.07 | fourcheeze | euchh, all this hugging |
| 15:19.25 | SplasPood | Does anyone know if it's possible to include other files via extensions.ael |
| 15:19.39 | fourcheeze | thanks for that |
| 15:19.42 | *** join/#asterisk Defraz (i=t0tal@tim.mychoice.cc) |
| 15:20.08 | Hmmhesays | i had taco hell last night |
| 15:20.21 | *** join/#asterisk devarus (i=Robert@S0106000d610ee3bb.va.shawcable.net) |
| 15:20.26 | Katty | Hmmhesays: did it make ya sick? |
| 15:20.28 | key2 | taco bell or taco hell |
| 15:20.32 | SpaceBass | hey- no one brought me tacos! |
| 15:20.51 | SpaceBass | (of course I was on a plane) |
| 15:21.05 | wasim | and don't have a taco bell this side of the suez ... |
| 15:21.10 | fourcheeze | Hmmhesays: I've been thinking about you SER/asterisk setup |
| 15:21.13 | Katty | SpaceBass: did you /ask/ for tacos |
| 15:21.17 | fourcheeze | Hmmhesays: can I just get some stuff clear in my head? |
| 15:21.38 | Hmmhesays | such as? |
| 15:21.44 | eric_hill | SpaceBass: eating tacos in an enclosed airplane has GOT to break at least one law... |
| 15:21.47 | fourcheeze | ok, all users register with SER |
| 15:21.58 | fourcheeze | all other requests get redirected via 302 to an asterisk? |
| 15:22.04 | Hmmhesays | yes |
| 15:22.05 | SpaceBass | eric_hill, I;d agree |
| 15:22.05 | [TK]D-Fender | Hmmhesays : thats the annoying thing about Quebec... no Taco Bell here.. even though it is kida shitty, at least its a change of pace from your typical burger joint... |
| 15:22.21 | Hmmhesays | [TK]D-Fender yeah, usually its an after bar stop for me |
| 15:22.27 | SpaceBass | eric_hill, that's what the curtin b/t first class and coach is for...keeping coach smells out :D |
| 15:22.31 | fourcheeze | Hmmhesays: so then when asterisk wants to find someone it always calls them at SER |
| 15:22.41 | Hmmhesays | yes |
| 15:22.42 | eric_hill | lol |
| 15:22.54 | Hmmhesays | its a basic redirect setup |
| 15:23.02 | UnderMine | eric_hill:strange when i dial out i then get the interupted dial tone. if i then dial the same number without hanging up it works.? |
| 15:23.03 | fourcheeze | how does * know that the incoming call is a redirect from SER? |
| 15:23.06 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 15:23.10 | astra^^ | hello all |
| 15:23.22 | Katty | Hmmhesays: we're having a concert down here! |
| 15:23.36 | *** part/#asterisk mattwj2005 (n=Matt@user-12l3lqm.cable.mindspring.com) |
| 15:23.37 | Katty | Hmmhesays: april 21st |
| 15:23.51 | Hmmhesays | fourcheeze cause you set an incoming context for the ser domain |
| 15:23.54 | Hmmhesays | Katty oh reawlly? |
| 15:23.56 | Katty | Hmmhesays: and i /might/ be conned into doing accompany vocals. |
| 15:24.07 | astra^^ | is ther any network converter which convertts from a anolog fax machine to voip on a avaya network? |
| 15:24.14 | Katty | Hmmhesays: they're looking for a 'cranberries' type sound almost |
| 15:24.18 | Hmmhesays | cool |
| 15:24.18 | fourcheeze | Hmmhesays: ok, so the incoming to * appears to come from SER? |
| 15:24.35 | Hmmhesays | oh no, that song is going to be stuck in my head now |
| 15:24.39 | fourcheeze | Hmmhesays: how do you handle trunks? Do you dial our from * or from SER? |
| 15:24.50 | astra^^ | Fax ->analog to voip N/w converter? |
| 15:24.58 | Hmmhesays | not sure what you mean by that |
| 15:25.23 | astra^^ | we are having a fax machine .ie analog |
| 15:25.33 | fourcheeze | Hmmhesays: ok user wants to dial out, does SER pass that on to * or does it do a 302 to the outbound proxy? |
| 15:25.48 | astra^^ | and we want to make it compatible with the voip network |
| 15:25.56 | astra^^ | ie avaya network |
| 15:26.04 | Hmmhesays | fourcheeze, 302 to asterisk then off to a termination gateway |
| 15:26.11 | fourcheeze | ok, cool |
| 15:26.14 | fourcheeze | I think I'm going to have a go at this |
| 15:26.18 | fourcheeze | however |
| 15:26.20 | astra^^ | is there any converter like one we have for analog phones |
| 15:26.26 | astra^^ | ATA's |
| 15:26.41 | Hmmhesays | you would use the same ata's |
| 15:27.16 | fourcheeze | Hmmhesays: so it's only registrations and then requests back to clients via SER that are not met with a 302 |
| 15:27.24 | Hmmhesays | bingo |
| 15:27.28 | fourcheeze | sounds easy |
| 15:27.34 | IronHelix | astra i think you want an ata with t.38 fax |
| 15:27.41 | Hmmhesays | fourcheeze this is a simple setup, take care of shit nat issues with linksys's running openvpn |
| 15:27.42 | IronHelix | sadly as far as i know asterisk doesnt support t.38 yet |
| 15:27.55 | salviadud | would there be a way to write a java app for iax2 on a cell phone. so i don't need to waste anymore "cell phone" minutes? |
| 15:28.03 | fourcheeze | Hmmhesays: how do you find SER handles nat compared to * ? |
| 15:28.17 | salviadud | or i'm i screwed, and would have to settle for SIP? |
| 15:28.24 | *** join/#asterisk gammacoder (n=chatzill@64-132-192-33.gen.twtelecom.net) |
| 15:28.37 | IronHelix | salviadud- sure, if you have a good phone (windows mobile, maybe symbian or blackberry) and more importantly enough cell bandwidth to amke the call |
| 15:28.46 | *** join/#asterisk heka (n=heka@82.114.68.124) |
| 15:28.53 | IronHelix | you'd probably need at least EDGE data if not 3g |
| 15:29.25 | salviadud | how bout a nokia 6101? |
| 15:29.26 | IronHelix | however iax vs sip isnt the problem, bandwidth and latency are |
| 15:29.26 | astra^^ | hmm... |
| 15:29.53 | salviadud | so you're saying that only bandwith is limiting... |
| 15:29.59 | Hmmhesays | fourcheeze: i don't |
| 15:30.01 | IronHelix | theoretically yes |
| 15:30.19 | astra^^ | IronHelix:am already having a ATA .. so it can be used for he fax machine as well right? |
| 15:30.21 | fourcheeze | Hmmhesays: what does that mean? |
| 15:30.22 | salviadud | i would say that registering to * might be a pain |
| 15:30.24 | astra^^ | i am not using asterisk |
| 15:30.32 | Hmmhesays | fourcheeze i have yet to use ser with nat |
| 15:30.35 | fourcheeze | ahh ok |
| 15:30.36 | astra^^ | it is connected to avaiya network |
| 15:30.44 | IronHelix | salvia also your cell provider may block voip |
| 15:31.42 | IronHelix | astra- hmm, dunno about avaya or any specifics... what i can say tho is that the 'right' way to send fax over voip is T.38. T.38 has the ATA decode the fax transmission tones and send the raw data, rather than the audio of the tones. The ATA on the other end remodulates these into the other fax machine |
| 15:31.57 | salviadud | if they do! i kill them |
| 15:32.17 | hackeron | anyone? all of a sudden, asterisk stopped detecting call pickup, I call someone through an anlog line, start talking to a person, after 90 seconds I see Nobody picked up in 90000ms in the asterisk console and asterisk hangs up. Any ideas? |
| 15:32.25 | *** join/#asterisk TripleF555 (n=TripleF5@modemcable084.12-201-24.mc.videotron.ca) |
| 15:32.34 | IronHelix | the other, more common way, is to set codec to g.711 ulaw/alaw and hope everything works |
| 15:32.36 | astra^^ | hmmm thanx iron |
| 15:32.47 | IronHelix | np astra |
| 15:32.50 | TripleF555 | hey ghuys.. anyone know if we got the mesages for.. ( if one call the 800 # but theres a local exchange for him.. ) that would tell him so.,. |
| 15:32.56 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:33.12 | TripleF555 | Like please call XXX-1234 from your calling area |
| 15:33.15 | TripleF555 | or something like it |
| 15:33.31 | IronHelix | hmmm |
| 15:33.49 | IronHelix | i think that could be implemented either by you or your provider |
| 15:34.12 | IronHelix | not exactly sure tho |
| 15:34.42 | heka | any ide on how to replace back the grandstream firmware if it has fail upgrading? |
| 15:34.45 | IronHelix | if you did it, just do a dialplan logic (if areacode = my area code) then goto (context that tells them to call xxx1234) else goto (ivr) |
| 15:34.57 | TripleF555 | hmm |
| 15:35.01 | TripleF555 | got alot of numbers |
| 15:35.04 | TripleF555 | maybe a macro ? |
| 15:35.09 | IronHelix | heka- depends on how bad it is. i think theres an ip addy the phones sometimes default to... |
| 15:35.14 | TripleF555 | macro ( pass the callerid . and etc |
| 15:35.35 | IronHelix | yeah macro might make it easy, have the macro do nothing but figure out if they're local or not and then return something |
| 15:36.30 | heka | IronHelix: it fails to boot! |
| 15:36.37 | TripleF555 | nah it will return the correct # to dial |
| 15:36.47 | IronHelix | when you power it heka, what happens? |
| 15:36.51 | IronHelix | ohh ic you have many local numbers |
| 15:37.08 | TripleF555 | as in .. function checkphone($callerid,$called) |
| 15:37.09 | heka | <PROTECTED> |
| 15:38.00 | IronHelix | hmmm... does the LED flash at all? |
| 15:38.43 | heka | flashes 3 times and then stops for a while then flashes again |
| 15:39.25 | IronHelix | ahh, are you running firmware 1.0.1.x? or have you upgraded to 1.0.2.x / 1.1.x |
| 15:39.26 | *** join/#asterisk Kernel_Core (n=I@193.251.135.118) |
| 15:39.27 | Kernel_Core | hi all |
| 15:39.43 | SpaceBass | my parents get their t1 today |
| 15:39.45 | Kernel_Core | anybody here configured sangoma (E3,T3) Card for asterisk ? |
| 15:39.45 | SpaceBass | fun fun |
| 15:40.03 | IronHelix | in 1.0.1.x, blank screen with always on LED is booting, the sometimes flashing is when it tries to get something from tftp |
| 15:40.37 | heka | IronHelix: I was downloading the firmware from gratissip TFTP |
| 15:40.46 | [TK]D-Fender | Kernel_Core : Sangoma doesn't have a channelized E/T3 yet, only data... |
| 15:41.02 | [TK]D-Fender | Kernel_Core : Its coming from what I hear though... |
| 15:41.04 | IronHelix | did you try just letting it sit for like 20min? |
| 15:41.06 | Kernel_Core | or anybody used more than 1 (4 E1/T1) Card per server ? |
| 15:41.35 | heka | IronHelix: it has about 15 minutes now |
| 15:41.43 | heka | it have* :P |
| 15:41.46 | mut | trying to delete a directory on a linux box from a windows box/asp page, any suggestions on methods? samba mount and delete or something like that.. |
| 15:42.08 | [TK]D-Fender | Kernel_Core : Typically not a great idea for the load that would place on it. |
| 15:42.09 | Kernel_Core | [TK]D-Fender:it is too bad if their Product isnot channelized ... |
| 15:42.12 | IronHelix | and its still going? still solid,w ith a blink every now and then? |
| 15:42.26 | heka | IronHelix: yes |
| 15:42.47 | heka | IronHelix: I have upgrade other phones too but never took that long |
| 15:42.53 | Kernel_Core | [TK]D-Fender: 4 E1 for incomming and 4 E1 for outgoing traffic... isn't it possible ? |
| 15:43.00 | [TK]D-Fender | mut : Use Putty & a script to SSH in and do it. |
| 15:43.19 | eric_hill | Anybody know where I can find documentation on the "information element"? Console message is "!! Unknown IE 49 (cs4, Unknown Information Element)" |
| 15:43.27 | IronHelix | agreed, i'd let it sit for a while tho, just in case it works itself out... if its not up at +20 or +30min then kill it and try booting it w/out network |
| 15:43.37 | [TK]D-Fender | Kernel_Core :Sure it is, but if * has to do much processing on it it might choke down a server... |
| 15:43.37 | eric_hill | It doesn't seem to hurt anything, but I'd like to fix it if possible. |
| 15:43.59 | Kernel_Core | [TK]D-Fender: what do you suggest for HW ? at least ... |
| 15:44.20 | Kernel_Core | [TK]D-Fender: and which card you suggest ? Sangoma or Digium ? |
| 15:44.24 | [TK]D-Fender | Kernel_Core : Look at "asterisk dimensioning" on the WIKI for some tips |
| 15:44.35 | heka | IronHelix: Im talking about second boot after putting the tftp server, in the first time i let it for about 4 hours |
| 15:44.39 | *** part/#asterisk TripleF555 (n=TripleF5@modemcable084.12-201-24.mc.videotron.ca) |
| 15:44.56 | heka | then rebooted it again. without/with network cable and the same |
| 15:45.12 | heka | hapens |
| 15:45.39 | IronHelix | hmmm |
| 15:45.59 | heka | IronHelix: also I cant ping it. |
| 15:46.02 | IronHelix | what firmware rev were you trying to load on it? |
| 15:46.37 | heka | REV 2.0 |
| 15:47.11 | heka | IronHelix: is there any way to reset to factory settings so the tftp server will be removed? |
| 15:47.31 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 15:48.05 | IronHelix | there is no 2.0... theres the 'old' 1.0.1.x, and the 'new' 1.0.2.x and 1.1.0.x |
| 15:48.23 | IronHelix | heka the way to restore factory defaults requires accessing the phone |
| 15:49.03 | IronHelix | you either scroll to reset, then type in the phone's MAC, then hit select (old) or select factory reset from the phone's menu (new) |
| 15:49.22 | heka | IronHelix: you mean for the program version? it is 1.0.8.16 |
| 15:49.25 | IronHelix | i think there may be a thing where the phone defaults to some failsafe IP address but i dont know what it is |
| 15:55.19 | *** join/#asterisk gammacoder (n=chatzill@64-132-192-33.gen.twtelecom.net) |
| 15:57.29 | *** join/#asterisk hgaillac (n=Harry@32.12.119-80.rev.gaoland.net) |
| 15:58.05 | hgaillac | hello how can i configure in order to 407 message are not sent back |
| 15:58.59 | noky | unconditional forwarding is only need a extension? |
| 15:59.07 | noky | dialpeer ? |
| 16:01.14 | Katty | Hmmhesays: is you /is/, or is you /ain't/ mah luh zahn yuh *hee* |
| 16:03.21 | Rawplayer | RE |
| 16:03.42 | Hmmhesays | what what what? |
| 16:04.25 | Katty | Hmmhesays: my gay friends are so amusing |
| 16:04.31 | Katty | Hmmhesays: they're /real/ fancy |
| 16:05.14 | devarus | by adding 'accountcode=blahblah' to sip.conf i get an account code in cdrs for outbound traffic,,,, |
| 16:05.28 | devarus | is there a way to assign an accountcode to an inbound DID? |
| 16:06.27 | devarus | i am using asterisk 1.2.4 |
| 16:06.49 | Katty | bkw__: say that in chicago |
| 16:06.57 | Katty | bkw__: i'll giggle for 10 minutes |
| 16:07.07 | Rawplayer | giggle for pr0n! |
| 16:07.22 | *** join/#asterisk cybertheq (n=infra@216-251-177-106.ips.cpinternet.com) |
| 16:07.33 | Katty | pron is boring |
| 16:08.34 | cybertheq | hello. Anyone here have experience with ADSI and Quicknet FXS? |
| 16:09.22 | iDunno | tron is much better. |
| 16:09.35 | Katty | pretty boys are the best. |
| 16:09.43 | Katty | well no i lied |
| 16:09.46 | Katty | pretty girls are the best |
| 16:10.29 | iDunno | I'll go along with that. |
| 16:10.36 | iDunno | pretty girls are the best indeed :) |
| 16:12.05 | Yasheee | Is there a way to listen to a .gsm file at the shell? (i.e. mpg123, or cat /dev/audio) |
| 16:12.47 | *** join/#asterisk riddlebox (n=blah@24-171-40-167.dhcp.stls.mo.charter.com) |
| 16:12.54 | Hmmhesays | I hate making websites |
| 16:13.36 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 16:17.05 | noky | i trying to forward a call unconditionally... i have a dude... i must to configurate something about DTMF in sip.conf ? because i only have information in wiki about the extensions.... |
| 16:17.33 | noky | in a conversation i dial a extensions.. but ast* doesn't nothing |
| 16:17.59 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 16:18.03 | salviadud | pretty girls are evil sometimes |
| 16:18.17 | salviadud | i was dating this girl... she was pretty |
| 16:18.31 | salviadud | and damn, she was nasty |
| 16:18.59 | salviadud | i'd be playing nintendo, then she'd fart in my face |
| 16:19.12 | salviadud | i miss her... |
| 16:19.24 | noky | and |
| 16:19.25 | noky | apperas |
| 16:19.27 | noky | appears |
| 16:19.28 | noky | 2006-04-07 12:43:48 WARNING[5052]: app_db.c:179 put_exec: This application has been deprecated, please use the ${DB(family/key)} function instead. |
| 16:20.09 | salviadud | post your extensions.conf on pastebin |
| 16:20.40 | noky | oks |
| 16:20.49 | andrebarbosa | anyone knows when will be release asterisk stable with chan_misdn 0.3.0 stable release today? |
| 16:20.58 | *** join/#asterisk alexis101 (n=alexis@70.54.204.92) |
| 16:21.11 | noky | http://pastebin.com/646325 |
| 16:21.51 | *** part/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
| 16:22.02 | [TK]D-Fender | noky : DBGet & DBPut are OLD. go checkout "asterisk functions" on the WIKI |
| 16:22.08 | noky | ok |
| 16:22.09 | noky | thanks |
| 16:22.26 | Yasheee | Is there a way to listen to a .gsm file at the shell? (i.e. mpg123, or cat /dev/audio) |
| 16:22.30 | alexis101 | i guys i have a little problem ... my custumer ask me to make hiddens option in a menu if the caller enter a secert digits sequence ... |
| 16:22.59 | salviadud | really? that's great |
| 16:23.02 | [TK]D-Fender | alexis101 : Easy.. just shove it in and don't announe it :) |
| 16:23.06 | salviadud | i'd do it with Background |
| 16:23.20 | salviadud | exactly, don't announce it |
| 16:23.34 | mocker | Or have them enter it and then put them to an Authenticate() app... |
| 16:23.58 | Yasheee | alexis101: exten => 88888,1,DoWhatever |
| 16:24.28 | GerbilNut | not 88888, people with fat fingers will get their accidently |
| 16:24.57 | salviadud | god i'm bored |
| 16:25.25 | salviadud | i hate workin' with a bunch of squares |
| 16:25.30 | alexis101 | is that possible ? |
| 16:25.34 | salviadud | i'd be prankcalling narconon right now |
| 16:25.38 | *** join/#asterisk TripleF555 (n=TripleF5@modemcable084.12-201-24.mc.videotron.ca) |
| 16:25.40 | TripleF555 | [app_rxfax.so]Apr 7 12:23:49 WARNING[53101]: loader.c:723 __load_resource: cannot load /usr/local/lib/asterisk/modules/app_rxfax.so /usr/local/lib/asterisk/modules/app_rxfax.so: Undefined symbol "t30_get_far_ident" |
| 16:25.41 | TripleF555 | Apr 7 12:23:49 WARNING[53101]: loader.c:733 __load_resource: /usr/local/lib/asterisk/modules/app_rxfax.so: Undefined symbol "t30_get_far_ident" |
| 16:25.43 | TripleF555 | anyone ? |
| 16:25.49 | TripleF555 | 1.2.1 with pre 20 |
| 16:25.51 | TripleF555 | spandsp |
| 16:25.59 | *** join/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 16:26.00 | salviadud | try 1.2.6 |
| 16:26.05 | TripleF555 | i mean 1.2.6 |
| 16:26.06 | TripleF555 | lol |
| 16:26.08 | TripleF555 | svn |
| 16:26.16 | salviadud | ohh. well then |
| 16:26.23 | TripleF555 | show version |
| 16:26.23 | TripleF555 | Asterisk SVN-trunk-r18090M |
| 16:26.57 | TripleF555 | anyone have it for freebsd |
| 16:27.11 | alexis101 | anyone have an idea ?? |
| 16:27.11 | TripleF555 | the whole 9 yards.. dspandsp and rx tx and makefile |
| 16:27.12 | lzhang | hey guys, when asterisk asks "please enter a new extension" when doing agent logins for queues, how does it know which extensions there are? |
| 16:27.22 | TripleF555 | from context |
| 16:27.30 | TripleF555 | queues has a context |
| 16:27.34 | lzhang | I'm using realtime for extensions and I think I need to add a switch somewhere so it will recognize |
| 16:27.34 | TripleF555 | so uses from there |
| 16:27.52 | TripleF555 | yeah had same problem.. make a link.. |
| 16:28.05 | TripleF555 | ex..1234 ,1,goto(elsewhere,s,1, |
| 16:28.07 | TripleF555 | etc |
| 16:28.18 | TripleF555 | so it will jump out of local queue context to elsewhere for that agent |
| 16:28.22 | TripleF555 | not sure how i did it |
| 16:29.58 | lzhang | I'm not sure I follow you |
| 16:30.05 | TripleF555 | me neither |
| 16:30.06 | TripleF555 | lol |
| 16:30.18 | lzhang | where should I goto? another context? |
| 16:30.20 | TripleF555 | realtime is hard to implement |
| 16:30.22 | Katty | Hmmhesays: what shall i lunch upon? |
| 16:32.51 | salviadud | katty, eat tacos |
| 16:33.07 | *** join/#asterisk redcap1 (n=phez@redcap.xs4all.nl) |
| 16:33.28 | Hmmhesays | Katty: something vegetably |
| 16:33.45 | TripleF555 | tacos eat kathy |
| 16:35.08 | *** join/#asterisk kakadu (n=blubb@p54B8D2ED.dip.t-dialin.net) |
| 16:35.13 | TripleF555 | app_rxfax.c:283: error: structure has no member named `logging' |
| 16:35.23 | TripleF555 | i get this with svn |
| 16:35.32 | TripleF555 | if i comment out then on module load it crashes * |
| 16:35.49 | lzhang | exten=> 2001,1,AgentCallbackLogin(||${CALLERIDNUM}@mycontext) what does the CALLERIDNUM@mycontext do? |
| 16:36.56 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:37.40 | kakadu | hi |
| 16:37.41 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-175-120.lsanca.fios.verizon.net) |
| 16:37.55 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 16:38.14 | kakadu | i have a question concerning zaptel config for a ISDN Phone connected via HFC Card ... |
| 16:38.26 | TripleF555 | oh well time to try *@home |
| 16:39.26 | kakadu | nobody? |
| 16:39.46 | kakadu | well I will keep crawling the web ... :-) |
| 16:39.54 | redcap1 | kakadu: so do I :) |
| 16:40.06 | TripleF555 | surfing |
| 16:40.16 | TripleF555 | crawling was in the mid 90's when all dialups |
| 16:40.25 | TripleF555 | ;) now we rock on 16 Mbs cable |
| 16:40.39 | redcap1 | :D |
| 16:40.46 | redcap1 | 12Mbit |
| 16:40.54 | santoshr | while dialing a ques consisting of agents after the first time out, the call is flowing into status no-answer instead of ringin the next agent |
| 16:40.55 | kakadu | well its more 2 MBit Cable with me but thats fast enough |
| 16:41.07 | santoshr | why would this happen |
| 16:41.10 | *** join/#asterisk op3r (n=op3r@202.71.189.66) |
| 16:41.13 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
| 16:41.13 | op3r | any one here? |
| 16:41.16 | gbodemantv | hello all |
| 16:41.24 | salviadud | back then, doom was a great game |
| 16:41.30 | [TK]D-Fender | lzhang : Go read the WIKI page on AgentCallbackLogin. You clearly haven't gone over it enough. |
| 16:41.36 | gbodemantv | so the question of the day is ...who is using realtime queues? |
| 16:41.47 | [TK]D-Fender | lzhang : And while you're at it, look up "Asterisk variables" too... |
| 16:41.54 | op3r | [TK]D-Fender: can you check if I did something wrong here http://pastebin.com/646366 ? |
| 16:41.56 | redcap1 | I have also some problems with both my ISDN cards (hfc) |
| 16:42.10 | op3r | can anyone please check this http://pastebin.com/646366 and tell me whats wrong with it? |
| 16:42.14 | santoshr | it works properly if call flow based oin dial status is disabled ? |
| 16:42.21 | redcap1 | The one connected to my telco crashes my s0 bus (sometimes) |
| 16:42.30 | [TK]D-Fender | op3r : I don't debug AMP, and I doubt you'll find anyone else here for that either... |
| 16:42.41 | redcap1 | and the other one (TE mode) does nothing :( |
| 16:42.57 | lzhang | will do [TK]D-Fender |
| 16:43.02 | op3r | [TK]D-Fender: its not an amp. its an installer for asterisk with amp |
| 16:43.08 | santoshr | while dialing a ques consisting of agents after the first time out, the call is flowing into status no-answer instead of ringin the next agent .. : it works properly if call flow based oin dial status is disabled ? |
| 16:44.42 | santoshr | [TK]D-Fender: .. any clues dude |
| 16:44.47 | [TK]D-Fender | op3r : Same damn thing to me... |
| 16:44.59 | TripleF555 | think i can push a ast@home + normal svn on same box ? |
| 16:45.11 | [TK]D-Fender | santoshr : Verify your queue Strategy |
| 16:45.12 | TripleF555 | got only one fbsd box .. |
| 16:45.40 | salviadud | amp sux |
| 16:45.45 | salviadud | end of story |
| 16:45.54 | santoshr | [TK]D-Fender: even with ringall, after the first timeout it is going in the no-answer dial status |
| 16:46.16 | op3r | :( |
| 16:46.30 | salviadud | op3r, you still have hope |
| 16:46.33 | salviadud | learn asterisk! |
| 16:46.45 | salviadud | don't let fancy nancy guis fool you |
| 16:46.51 | [TK]D-Fender | santoshr : And what happens then? |
| 16:47.16 | op3r | salviadud: I know! I am trying to do a script to auto install asterisk though :( |
| 16:47.31 | salviadud | really? |
| 16:47.37 | [TK]D-Fender | santoshr : pastebin how you call the queue, and the queues.conf after it. |
| 16:47.41 | salviadud | make;make install ? |
| 16:47.48 | santoshr | ok [TK]D-Fender |
| 16:48.07 | op3r | salviadud: nope I am trying to install the dependencies first |
| 16:48.08 | salviadud | that would work for me... |
| 16:48.11 | noky | i correct my error [TK]D-Fender ... but how can i transfer a call ? |
| 16:48.15 | noky | must i press flash in the phone ? |
| 16:48.22 | op3r | and setting up mysql, apache and stuff |
| 16:48.23 | salviadud | what dependencies? |
| 16:48.23 | op3r | :( |
| 16:48.25 | salviadud | lipbri? |
| 16:48.58 | op3r | salviadud: http://pastebin.com/646366 |
| 16:49.00 | op3r | :( |
| 16:49.30 | noky | i have the dialpeer |
| 16:49.50 | noky | i dial in my phone, but don't nothing |
| 16:49.52 | salviadud | is this debian? |
| 16:49.59 | *** part/#asterisk ldnblk (n=ldnblk@host217-34-122-211.in-addr.btopenworld.com) |
| 16:50.02 | [TK]D-Fender | noky : What error, and what kind of ATA/phone are you using? |
| 16:50.05 | gbodemantv | any good resources that anyone knows to look for realtime configs |
| 16:50.15 | op3r | salviadud: centos |
| 16:50.17 | gbodemantv | have the tables created |
| 16:50.19 | op3r | yep debian |
| 16:50.22 | Qwell[] | gbodemantv: the files in the doc/ directory |
| 16:50.27 | gbodemantv | just trying to figure what to put in queues.conf |
| 16:51.07 | noky | i'm using a panasonic phone connected to a gateway sip |
| 16:51.25 | salviadud | i'm sorry op3r |
| 16:51.31 | salviadud | i can't help you out with amp |
| 16:51.42 | santoshr | [TK]D-Fender: http://sial.org/pbot/16654. .. can u please have a look at this. |
| 16:52.00 | gbodemantv | qwell : Does not seem to have info on queues in there |
| 16:52.05 | op3r | salviadud: i can remove the amp but I also need the LAMP |
| 16:52.33 | [TK]D-Fender | santoshr : Can't open the page, use pastebin please... |
| 16:52.37 | [TK]D-Fender | ~pb |
| 16:52.39 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 16:52.39 | mut | like wtf |
| 16:52.46 | mut | do all women try to make you feel sorry for them or something |
| 16:52.54 | salviadud | guilty |
| 16:52.59 | salviadud | some women do |
| 16:53.03 | mut | well |
| 16:53.08 | salviadud | they will try to get attention that way |
| 16:53.15 | mut | my gf is always "no i don't have the money for this or that" |
| 16:53.30 | mut | and ppl buy her crap all the time |
| 16:53.30 | salviadud | tell her "get a job, lazy biatch" |
| 16:53.36 | MooingLemur | translation: "give me money"? |
| 16:53.39 | mut | she has a job |
| 16:53.40 | santoshr | [TK]D-Fender: http://pastebin.com/646392 |
| 16:54.00 | Hmmhesays | [TK]D-Fender i'm gonna pm you |
| 16:54.01 | mut | someone asks "do you want to order anything for lunch" |
| 16:54.04 | [TK]D-Fender | mut : I've gotr a friend with a GF like that.... she's completely worthless.. none to bright, on gov't aid, and a burnout... |
| 16:54.07 | mut | instead of saying no |
| 16:54.17 | mut | she says "well i don't have any money, i've have to put it on my credit card" |
| 16:54.30 | salviadud | Fender, is that girl from Greenbo Alabama? |
| 16:54.45 | MooingLemur | is it too much to ask to carry $20 at all times? :P |
| 16:54.52 | Qwell[] | MooingLemur: yes |
| 16:54.53 | mut | .. |
| 16:54.54 | mut | yea |
| 16:55.00 | mut | i never have money on my wallet |
| 16:55.03 | nothinman | time to go |
| 16:55.05 | mut | $10 at most |
| 16:55.09 | nothinman | goodbye everyone |
| 16:55.11 | [TK]D-Fender | santoshr : You can NOT put an "Answer" in that macro or it will not redistribute the call! |
| 16:55.16 | salviadud | mut, where's the money then? |
| 16:55.18 | mut | but i don't try to make ppl feel guilty i have no money, i just tell them no |
| 16:55.19 | nothinman | [TK]D-Fender: thanks for help man :) |
| 16:55.20 | Qwell[] | because the moment you spend any of that emergency money...you no longer have $20 in your pocket |
| 16:55.31 | Qwell[] | which means you can never carry $20 at all times...unless you never spend it |
| 16:55.34 | [TK]D-Fender | santoshr : no "Playback, no Answer, NOTHING. Just Dial, and variable logic. |
| 16:55.37 | Qwell[] | which makes it useless to be in your pocket |
| 16:55.43 | [TK]D-Fender | nothinman : What did I help you with again? |
| 16:55.44 | mut | i use my debit card for everything |
| 16:55.46 | [TK]D-Fender | :) |
| 16:55.51 | mut | if i can't buy it with my debit card i don't get it |
| 16:55.54 | [TK]D-Fender | mut : The only Plastic I like... |
| 16:56.01 | salviadud | mut, you gotta check out if it's cheaper |
| 16:56.12 | [TK]D-Fender | mut : CC is good for phone orders and bills.... |
| 16:56.16 | salviadud | some credit card companies add a cost for using a debit card |
| 16:56.17 | mut | i have no costs incurred using it that way |
| 16:56.22 | MooingLemur | well, the idea was to replace part of that $20 at the next convenient opportunity, such as at a grocery store checkout |
| 16:56.34 | nothinman | [TK]D-Fender: I don't remember. I was about 2000 asterisk config lines ago ;) but thanks ;) |
| 16:56.37 | Qwell[] | MooingLemur: The way the question was worded, it's either impossible or pointless |
| 16:56.48 | [TK]D-Fender | nothinman : All good :) |
| 16:56.49 | MooingLemur | you took it too literally :P |
| 16:56.54 | Qwell[] | of course I did |
| 16:56.56 | mut | i have like $120 a month in spending cash |
| 16:57.04 | mut | toward gas and whatever else i'de need |
| 16:57.12 | nothinman | see ya! |
| 16:57.19 | santoshr | [TK]D-Fender: but without playback.. how.. tell tht not available |
| 16:57.39 | [TK]D-Fender | mut : Much the same... floating cash for the small stuff, debit for evrything else CC for the places that bitch about debit... |
| 16:58.02 | MooingLemur | carrying a small amount of cash seems like a good idea. Some places surcharge for small plastic purchases |
| 16:58.09 | Qwell[] | 35c... |
| 16:58.17 | Qwell[] | That's hardly noticable |
| 16:58.20 | [TK]D-Fender | santoshr : The fact you're in a queue tells you that you'll have to wait for someone. You simply CANNOT do an answer or the call will not redistribute. |
| 16:58.46 | [TK]D-Fender | santoshr : And don't even DREAM about calling the Queue again from within an exten like that... you'll nuke all your agents. |
| 16:59.20 | MooingLemur | that could be a 20% surcharge on the product :P |
| 16:59.31 | gbodemantv | qwell: any idea what needs to be in the queues.conf file when setting up realtime queues |
| 16:59.32 | gbodemantv | ? |
| 16:59.40 | Qwell[] | MooingLemur: Then that means it's only $2 |
| 16:59.48 | Qwell[] | which is 10% of the $20 |
| 16:59.48 | *** join/#asterisk kelvin^8^ (n=kelvin@60.49.70.199) |
| 16:59.57 | gbodemantv | queues.conf file |
| 16:59.57 | gbodemantv | [general] |
| 16:59.57 | gbodemantv | realtime_family = family_name,family_name1,.. |
| 17:00.08 | gbodemantv | is what the wiki says |
| 17:00.13 | Qwell[] | gbodemantv: no, that goes elsewhere |
| 17:00.28 | santoshr | [TK]D-Fender: calling the ques fmor within an extension ?? dint get wht u r trin to say |
| 17:00.35 | MooingLemur | the idea is to carry cash so you don't annoy coworkers when they offer to get you lunch :P |
| 17:00.35 | Qwell[] | extconfig.conf |
| 17:00.36 | kelvin^8^ | where can i find a installation guide for asterisk? i just need simple voip for internal office use no pstn |
| 17:00.36 | MooingLemur | hehe |
| 17:00.48 | Qwell[] | MooingLemur: make them pay |
| 17:00.51 | gbodemantv | so is the wiki wrong? |
| 17:00.56 | [TK]D-Fender | santoshr : never mind then... just get rid of the playbacks, and answers. |
| 17:01.10 | Qwell[] | gbodemantv: no, you aren't reading it right I'd bet |
| 17:01.28 | *** part/#asterisk theHub (n=karlhubn@69.177.93.20) |
| 17:01.41 | Qwell[] | gbodemantv: and in that situation...I'd just walk to the ATM 20 ft away :P |
| 17:01.44 | Hmmhesays | ~docs |
| 17:01.45 | jbot | [docs] probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:01.52 | santoshr | but the the stdexten is the macro which is dailni all my extensions.. i would need to playbck unavailability |
| 17:02.01 | Hmmhesays | there you go kelvin^8^ otherwise you can hire one of us to do it |
| 17:04.28 | *** join/#asterisk X-Gen (n=x-gen@dsl-145-224-51.telkomadsl.co.za) |
| 17:05.23 | hgaillac | Hello is ther a solution in order to asterisk does no sent back 407 message proxy authenticatio |
| 17:05.34 | op3r | kelvin^8^: try this http://pastebin.com/646391 |
| 17:05.36 | op3r | :D |
| 17:05.41 | santoshr | [TK]D-Fender: this same setup is working properly with normal sip extens and not agents.. the ques redistribute and the stratg properly |
| 17:05.42 | op3r | and tell me what went wrong? |
| 17:05.51 | kelvin^8^ | thank i reading it |
| 17:05.56 | IronHelix | hgaillac- remove the secret= statement from the sip.conf entry and it should not require authentication |
| 17:07.35 | *** join/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
| 17:08.06 | santoshr | [TK]D-Fender: u around ? |
| 17:08.11 | *** join/#asterisk tdonahue-laptop (n=tdonahue@231.212.118.66.brainstorminternet.net) |
| 17:08.17 | tdonahue-laptop | morning all |
| 17:08.31 | Hmmhesays | yo ho ho |
| 17:08.45 | TripleF555 | morn hmm |
| 17:09.00 | Hmmhesays | hola |
| 17:10.12 | tdonahue-laptop | i'm having some trouble with cisco 7960 phones and nat. outbound calls work, but whenever i dial extension to extension I get an error "Unable to create channel of type 'SIP' (cause 3 - No route to destination)" |
| 17:10.20 | hgaillac | Ironhelix: secret is not set in extension !? |
| 17:10.50 | IronHelix | hmmm, is it getting rejected when it tries to register? |
| 17:11.20 | tdonahue-laptop | anyone have any thoughts? |
| 17:11.36 | IronHelix | tdonahue-laptop- sounds like a network error |
| 17:12.08 | IronHelix | i think whats happening is the cisco phones arent correctly using STUN |
| 17:12.15 | IronHelix | or they arent putting the right IP in their SDP |
| 17:12.16 | tdonahue-laptop | traceroute and pings work both way... |
| 17:12.29 | tdonahue-laptop | ok, i'll take a look at the SDP |
| 17:13.02 | IronHelix | i think they're telling your * box to setup the channel to 192.168.whatever which asterisk can't route to |
| 17:14.03 | IronHelix | bbl |
| 17:15.00 | mut | 10% surcharge would be rediculous |
| 17:15.31 | mut | and i use my bank to pay my bills, online bill pay i can send a check to anyone i want free |
| 17:16.26 | mut | but i only bring home.. $1100 a month or so |
| 17:16.47 | Qwell[] | "online bill pay"? :D |
| 17:16.51 | Qwell[] | Wonder what bank THAT could be |
| 17:17.15 | mut | heh |
| 17:17.32 | Qwell[] | the one I work for, likely :p |
| 17:17.46 | mut | citizens.. |
| 17:17.52 | Qwell[] | or not |
| 17:18.02 | mut | i like it |
| 17:18.12 | mut | pay everything with it |
| 17:18.29 | mut | i've only used 4 checks from my book in 3 yrs |
| 17:21.20 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 17:21.45 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:23.04 | Katty | mishehu: ! |
| 17:24.45 | Hmmhesays | lol |
| 17:24.52 | Katty | >.< |
| 17:25.03 | Qwell[] | he kicked you? |
| 17:25.10 | Qwell[] | big meanie |
| 17:25.21 | Katty | Hmmhesays: i found the most adorable picture of $boy this morning. |
| 17:25.23 | Katty | Hmmhesays: wanna see? |
| 17:25.30 | Hmmhesays | sure |
| 17:26.17 | Katty | Hmmhesays: cute huh ^_^ |
| 17:27.04 | hgaillac | HOW CAN I DISABLE 407 PROXY AUTHENTICATION |
| 17:27.48 | Qwell[] | Katty: his volume potentiometer was wired in backwards |
| 17:27.52 | Qwell[] | You just turned him up ;/ |
| 17:27.56 | Katty | :<< |
| 17:28.10 | Hmmhesays | insecure=very |
| 17:28.24 | *** part/#asterisk kelvin^8^ (n=kelvin@60.49.70.199) |
| 17:29.05 | hgaillac | Hmmhesays: i' ve done it even insecure=invite |
| 17:29.38 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:30.33 | kakadu | reload ==> setup_zap: ignoring overlapdial - what can I do? |
| 17:32.37 | Yasheee | can someone take a look at my simple VM setup and see if there is a better way to implement it? http://www.devrandom.org/p/14 |
| 17:36.06 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 17:36.27 | *** join/#asterisk nain (n=nain@202.59.90.178) |
| 17:36.36 | nain | hi |
| 17:37.35 | SplasPood | Anyone happen to know why I might be getting this when calling out via nufone (SIP) ... It was working up until recently.. Got SIP response 500 "I'm terribly sorry, server error occurred (1/SL)" back from 66.225.202.76 |
| 17:37.41 | [TK]D-Fender | santoshr : Make another macro for your queues then. |
| 17:39.00 | [TK]D-Fender | Yasheee : Well there is no s,6 for your goto to land on, but besides that you really should hangup after going to voicemailmain.... |
| 17:39.02 | brettnem | hey can anyone recommend a good cell phone you can ssh from other than the TREO? :) |
| 17:39.22 | *** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
| 17:39.30 | _Thor | Hello everyone |
| 17:39.40 | [TK]D-Fender | brettnem : Maybe a UTStarcom PPC 6700? |
| 17:39.49 | brettnem | Do you use it? |
| 17:40.01 | SplasPood | brettnem: Blackberry |
| 17:40.20 | brettnem | I was thinking about the blackberrys... hmm |
| 17:40.41 | brettnem | the LG VX9800 looks interesting.. |
| 17:40.55 | [TK]D-Fender | brettnem : Not yet, I'm preparing to demo one. Its a win2003 CE based system |
| 17:41.11 | [TK]D-Fender | brettnem : I'd bet there is a compatible SSH client for it somewhere... |
| 17:41.11 | *** join/#asterisk somegeek_ (i=levin@unaffiliated/somegeek) |
| 17:41.20 | brettnem | right |
| 17:41.31 | _Thor | I am having problems with a northstar hearing the dtmf tones from a linksys box, anybody has seen this before? |
| 17:41.36 | op3r | can anyone help me debug this script? |
| 17:41.45 | op3r | http://pastebin.com/646391 |
| 17:41.52 | brettnem | [TK]D-Fender: how much is that phone? |
| 17:42.09 | [TK]D-Fender | brettnem : alone I'm not sure, $400CDN and dropping |
| 17:42.19 | [TK]D-Fender | with some sort of plan... |
| 17:42.29 | Math` | is the motorola cn-260 out yet? |
| 17:42.35 | Math` | the gsm+wifi one |
| 17:42.37 | brettnem | hmm 400 cdn eh? |
| 17:42.43 | [TK]D-Fender | op3r : Forget asking generically here for that script, use the mailing lists.... |
| 17:42.44 | brettnem | yes, gsm+wifi sounds interesting. :) |
| 17:42.52 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 17:43.06 | Math` | brettnem: the best is you can hot-swap calls when you get in-range of your wifi |
| 17:43.08 | [TK]D-Fender | brettnem : supposedly... and yeah, thats why I'm interested... hybrid VoIP/cell use |
| 17:43.22 | brettnem | cool |
| 17:43.28 | brettnem | I think the treo can do that too |
| 17:43.29 | Math` | [TK]D-Fender: you checked motorola's solution? |
| 17:43.36 | brettnem | earthlink has some compelling deals |
| 17:43.46 | [TK]D-Fender | Math` : nope, I'm only beginning my research into viable models. |
| 17:44.15 | Yasheee | [TK]D-Fender: I noticed that myself after posting.. I replaced the Goto with Hangup |
| 17:44.18 | _Thor | dtmf experts anyone? |
| 17:44.44 | brettnem | woah: http://www.110220volts.com/Merchant2/merchant.mvc?Screen=PROD&Product_Code=MPX-300 |
| 17:45.00 | tdonahue-laptop | oops... might help if i acually tried having Dial contact the correct SIP account |
| 17:45.03 | brettnem | well there's a phone with everything |
| 17:45.33 | [TK]D-Fender | Yasheee : other than that... yeah you turned a PC into a $25 digital answering machine :) |
| 17:45.52 | mut | $1800 tho? |
| 17:45.58 | brettnem | yeah, crazy eh? |
| 17:46.16 | mut | a pda-phone is cheaper and does just as much or more |
| 17:46.18 | hgaillac | no way to disable 407 proxy authentication !? |
| 17:46.21 | brettnem | why can't they just make a bluetooth, speakerphone, wifi phone with ssh capability.. that's all I need |
| 17:46.32 | brettnem | for cheap |
| 17:46.34 | brettnem | :D |
| 17:47.12 | file | hgaillac: you make a peer entry with the IP address that the INVITE is going to come from, and set insecure=very and make sure the user portion in the From header does not match a user entry - then it'll authenticate on IP address and not send back 407 |
| 17:47.25 | mut | tho the dual flip out feature is cool |
| 17:47.35 | [TK]D-Fender | brettnem : the 6700 does it all except having a confirmed SSH client, but thats almost assured already so.... it does everything else for sure. |
| 17:48.00 | brettnem | yeah, I think it might still be too pricy. Considering a treo 650 is like $99 ;) |
| 17:48.19 | brettnem | of course, I think you have to donate a child to get that deal |
| 17:48.20 | [TK]D-Fender | brettnem : Also good for using bluetooth as a link to it as a modem over EVDO |
| 17:48.28 | brettnem | yeah.. nice |
| 17:48.43 | brettnem | http://store.treocentral.com/content/smartphones/index.htm |
| 17:48.54 | brettnem | <sigh> |
| 17:49.09 | brettnem | Not sure if I should just get the Treo 650.. or get something newer.. |
| 17:50.02 | SplasPood | Can someone tell me if the examples @ http://www.voip-info.org/wiki/view/Asterisk+AEL are correct for setting up contexts+extens.. it seems to be taking my extens and converting them into a Set() |
| 17:51.12 | [TK]D-Fender | SplasPood : AEL = waste of time... no real point yet |
| 17:51.32 | SplasPood | [TK]: Other than I prefer it's syntax over straight up extensions.conf |
| 17:52.05 | brettnem | so the treo 650 doesn't support bluetooth? grrr |
| 17:52.19 | _Thor | for the record, I found the solution to my dtmf problem... I changed the dtmf transfer method to inband. thanks |
| 17:52.31 | brettnem | oh maybe it does.. hmm |
| 17:53.36 | *** join/#asterisk bkw_ (n=brian@c-68-32-112-142.hsd1.md.comcast.net) |
| 17:54.05 | vader-- | what are the recommended sizes for partitions when using asterisk? |
| 17:54.18 | [av]bani | ael is silly, they should have used lua instead |
| 17:54.21 | vader-- | sizes for /, swap. /tmp, /var, /usr, /home |
| 17:54.26 | vader-- | i have a 136gig drive |
| 17:55.27 | *** join/#asterisk ketanp (n=ketanp@67.132.43.2) |
| 17:55.30 | *** join/#asterisk aze (n=lucky@ACayenne-101-1-11-34.w81-248.abo.wanadoo.fr) |
| 17:55.36 | aze | hi all |
| 17:55.44 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 17:57.58 | [TK]D-Fender | vader-- : Depends... I just use one giant happy partition for everything except swap :) |
| 17:58.13 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
| 17:59.11 | *** part/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:00.44 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:03.40 | santoshr | but [TK]D-Fender how to isolate the Dial part , i mean ques are executing dial which is in the std-exten macro. ques are being dialed from a diferent macro. |
| 18:05.49 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 18:06.04 | *** join/#asterisk lorinc (n=ang@caracas-3132.adsl.interware.hu) |
| 18:07.01 | asterboy | I haev my hint setup for Buddy Watch/Asterisk Presence, however 'sip show subscriptions' does *not* show any subscribers. :-( |
| 18:07.07 | asterboy | exten => 711,hint,SIP/Home2 |
| 18:07.08 | asterboy | exten => 711,1,Dial(SIP/Home2,90,twTW) |
| 18:08.00 | ljam | [TK]D-Fender: you ever seen an Asterisk box send something like OPTIONS sip:mydomain.com SIP/2.0 then of course Asterisk looks for mydomain.com as the user in default (domain ) -- which of course doesn't exist since sip:mydomain.com is not a real URI |
| 18:08.12 | [av]bani | http://www.tenjo.de/ |
| 18:08.15 | [av]bani | those wacky germans |
| 18:08.20 | asterboy | Guess the subcribe portion is a function of the SIP phones, specifically, my Polycom IP600. |
| 18:08.31 | asterboy | Anyone else have Buddy Watch working with Polycom phones? |
| 18:08.54 | SplasPood | aster: I do, however there's a 7 exten limit currently |
| 18:09.24 | asterboy | right now my limit is at zero. :-( |
| 18:09.29 | SplasPood | aster: you need to add the extens you want to watch to the directory on the phone you wish to watch from |
| 18:09.33 | SplasPood | and set <bw>1</bw> |
| 18:09.36 | SplasPood | if I recall |
| 18:09.42 | asterboy | yes that has been done. |
| 18:09.50 | SplasPood | you enable presence support on that phone? |
| 18:09.56 | asterboy | what do you have for the <cn> tag? |
| 18:10.31 | SplasPood | I don't |
| 18:10.35 | asterboy | err...<ct> tag |
| 18:10.39 | SplasPood | <PROTECTED> |
| 18:10.40 | SplasPood | <PROTECTED> |
| 18:10.40 | SplasPood | <PROTECTED> |
| 18:10.40 | SplasPood | <PROTECTED> |
| 18:10.40 | SplasPood | <PROTECTED> |
| 18:10.40 | SplasPood | <PROTECTED> |
| 18:10.42 | SplasPood | as an example |
| 18:10.53 | asterboy | ah |
| 18:11.08 | asterboy | I have: |
| 18:11.09 | asterboy | <ct>sip:Home2@192.168.1.8</ct> |
| 18:11.22 | SplasPood | that's prolly your issue... |
| 18:11.23 | asterboy | Thinking it was the sip url. |
| 18:11.30 | SplasPood | I forget what exactly <ct> is tho |
| 18:11.35 | asterboy | so it's the hint extenstion |
| 18:11.40 | SwK[Work] | anyone gotta good source for 1U IBM Xseries servers? |
| 18:11.48 | asterboy | <ct> is the ConTact |
| 18:11.54 | asterboy | as in the number to dial |
| 18:12.02 | asterboy | that is why I thought SIP URL. |
| 18:12.02 | SplasPood | SwK: My employer buys lots of them for our customers (managed/unmanaged servers) |
| 18:12.44 | [av]bani | http://www.michaeldashow.com/zoom/zoom_wageslave.html |
| 18:12.52 | asterboy | So I need <ct>711</ct> |
| 18:12.52 | asterboy | exten => 711,hint,SIP/Home2 |
| 18:13.31 | SplasPood | yea |
| 18:14.10 | hgaillac | file: sorry it's not clear |
| 18:14.42 | [TK]D-Fender | asterboy : do you have presence enabled in your sip.cfg? |
| 18:14.57 | asterboy | checking |
| 18:15.35 | *** join/#asterisk chino[server] (n=daquino@e82-103-128-114s.easyspeedy.com) |
| 18:15.37 | *** mode/#asterisk [+o file[laptop]] by file |
| 18:15.52 | asterboy | <feature feature.1.name="presence" feature.1.enabled="0" |
| 18:15.57 | SplasPood | ahh |
| 18:16.00 | SplasPood | thats your problem |
| 18:16.02 | SplasPood | all the way. |
| 18:16.04 | [TK]D-Fender | asterboy : BAD.. should be "1" |
| 18:16.34 | chino[server] | can i hook up my asterisk server to a ethernet switch and then just hook up ip telephones to the switch instead of buying a pci card with multiple ports ? |
| 18:16.53 | SplasPood | chino[server]: yea |
| 18:17.00 | [TK]D-Fender | chino[server] : Thats the idea.... |
| 18:17.06 | SplasPood | chino[server]: the PCI cards are for analog interfaces |
| 18:17.12 | SplasPood | chino[server]: POTS phones |
| 18:17.20 | SplasPood | and some digital interfaces |
| 18:17.21 | SplasPood | such as PRI |
| 18:17.31 | chino[server] | does my asterisk server need some special device attached to it or can i simple point my asterisk server to the ip over my voip provider and send my credentials ? |
| 18:17.39 | SplasPood | IP phones are the way to go |
| 18:17.42 | *** join/#asterisk vopi (n=kkk@202.139.209.194) |
| 18:17.43 | SplasPood | chino[server]: the latter is correct |
| 18:17.44 | asterboy | that is why my messaging was not working either probably. |
| 18:17.53 | noky | 2006-04-07 14:39:56 NOTICE[5534]: rtp.c:330 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. ClientIP: 200.123.139.81 |
| 18:17.53 | SplasPood | chino[server]: No other hardware is needed for a pure SIP environment |
| 18:17.57 | noky | what is that ? |
| 18:17.58 | *** part/#asterisk TripleF555 (n=TripleF5@modemcable084.12-201-24.mc.videotron.ca) |
| 18:18.11 | SplasPood | noky: Concealment Noise, asterisk does not support it.. |
| 18:18.17 | *** join/#asterisk ComputerWarm (n=dan@HS196-230-97.nt.net) |
| 18:18.24 | chino[server] | ok what about if i have an old meridian network setup right now can i still keep my old meridian phones ? |
| 18:18.26 | ComputerWarm | hello all is there a support channel for gnugk? |
| 18:18.32 | noky | must support |
| 18:18.42 | SplasPood | noky: eh? |
| 18:18.47 | [TK]D-Fender | chino[server] : You'd need a gateway like a Citel or Intel one.. pricey.... |
| 18:18.56 | chino[server] | so a gateway is a pci card ? |
| 18:19.16 | SplasPood | chino: could be a box with ethernet into it that speaks SIP to asterisk |
| 18:19.25 | [TK]D-Fender | chino[server] : no, its a 1U rack device that has an amphenol connector & RJ45. it converts them to SIP. |
| 18:19.32 | ComputerWarm | i will take that as no... thanks anyway |
| 18:19.35 | chino[server] | SplasPood: which question was that in refrence too ? |
| 18:19.47 | SplasPood | ComputerWarm: I'd take it as a "no one knows" |
| 18:19.57 | SplasPood | chino: [14:18] chino[server]so a gateway is a pci card ? |
| 18:19.57 | ComputerWarm | SplasPood ya same here |
| 18:19.58 | chino[server] | [TK]D-Fender: wow how much does that go for ? so the ip phones them selves talk sip ? |
| 18:20.10 | SplasPood | chino[server]: yes |
| 18:20.25 | chino[server] | i thought sip was just for setting up the route |
| 18:20.39 | SplasPood | Well the phones speak SIP+RTP |
| 18:20.44 | SplasPood | RTP for the audio stream |
| 18:20.44 | chino[server] | yes cool! |
| 18:20.49 | chino[server] | what about video ? |
| 18:21.03 | SplasPood | Well if you had some sorta SIP video phone, but I dunno of any offhand |
| 18:21.05 | [TK]D-Fender | chino[server] : SIP is the VoIP protocol you should be using, and you plug your Nortel phones into the gatway and it will convert it to SIP out its ethernet port. |
| 18:21.13 | chino[server] | so how much does the cheapest ip phones go for that support all the features of the pbx ? |
| 18:21.13 | [TK]D-Fender | chino[server] : $2350USD +/- |
| 18:21.19 | [TK]D-Fender | chino[server] : for 24 ports |
| 18:21.42 | asterboy | $83 for the GXP-2000 |
| 18:21.51 | [TK]D-Fender | chino[server] : depends on the quailty. Could be <$100 or >$400 depending on your needs & quality. |
| 18:22.07 | [TK]D-Fender | chino[server] : I strongly suggest Polycom for any business use. |
| 18:22.08 | chino[server] | [TK]D-Fender: yes but tehy alos have wierd codes that you would use to change features and i think only the meridian pbx understands them ? or did asterisk also build ina compatibility for those feature signals ? |
| 18:22.09 | asterboy | Go for the Cisco with the full color screen |
| 18:22.21 | chino[server] | whats polycom ? |
| 18:22.22 | *** join/#asterisk kisu (n=daniel@cielkisu.tb.as8758.net) |
| 18:22.33 | brettnem | anyone know how the blackberry 7520 compares to the treo 650?? :) |
| 18:22.45 | [TK]D-Fender | chino[server] : The gateway fakes being the pbx completely and controls all aspects of the phone. Don't forget that digital phones are DUMB... |
| 18:23.07 | chino[server] | yes they are all dumb right they just send commands to the pbx ? |
| 18:23.11 | [TK]D-Fender | chino[server] : Polycom is a manufacturer of quality SIP hardphones. |
| 18:23.16 | file | DUMB |
| 18:23.21 | ComputerWarm | anyone here have knowledge they would like to share on getting gnugk working with a radius server? |
| 18:23.24 | chino[server] | so what about wireless voip ? |
| 18:23.33 | [TK]D-Fender | chino[server] : exactly, and this gateway allows you to assign the phones buttons to SIP phone features. |
| 18:23.36 | vopi | nufone |
| 18:23.55 | chino[server] | so im better of just getting all new ip phones and throwing away the old meridian system right ? |
| 18:23.56 | file | I'm working on my roar! oh I just can't wait to be king |
| 18:23.59 | [TK]D-Fender | chino[server] : Wireless = suck right now. Typically you'd use an ATA and a "normal" cordless phone (if you know whats good for you) |
| 18:24.10 | [TK]D-Fender | chino[server] : If at all possible, yes |
| 18:24.11 | chino[server] | ATA ? |
| 18:24.12 | file | noone saying do this, noone saying be there, noone saying stop that, noone saying see here! |
| 18:24.30 | Strom_M | oh god, not the lion king |
| 18:24.43 | [TK]D-Fender | chino[server] : Analog Telephone Adapter. converts an analog phone to digital/VoIP |
| 18:24.49 | file | Strom_M: oh I just can't wait to be king |
| 18:24.55 | Strom_M | i had successfully kept that thing out my conscious mind for a decade |
| 18:25.08 | ljam | </sarcasm> |
| 18:25.11 | Katty | Hmmhesays: go look at bif naked's legs. |
| 18:25.15 | chino[server] | [TK]D-Fender: thats the best way your saying ? isn't getting ip phones better ? and you just told me that getting converters is costly... |
| 18:25.24 | file | ljam: hush you |
| 18:25.29 | Katty | file: from the top! |
| 18:25.36 | Katty | file: everybody look left! |
| 18:25.42 | Katty | file: everybody look right! |
| 18:25.45 | hgaillac | HOW CAN I DISABLE 407 PROXY AUTHENTICATION !!!! |
| 18:25.51 | chino[server] | [TK]D-Fender: oh im sory your talking about cordless |
| 18:25.55 | file | hgaillac: okay stop doing that... I told you |
| 18:25.58 | Strom_M | hgaillac, by not screaming about it |
| 18:26.01 | chino[server] | yes but what features would hooking up a regular phone support ? |
| 18:26.04 | file | hgaillac: you make a peer entry with the IP address that the INVITE is going to come from, and set insecure=very and make sure the user portion in the From header does not match a user entry - then it'll authenticate on IP address and not send back 407 |
| 18:26.40 | justinu|laptop | insecure=very |
| 18:26.40 | ljam | hgaillac: ummm you DON'T -- thats how SIP works |
| 18:26.49 | chino[server] | so how much do these voip phones go for ? |
| 18:27.00 | file | ljam: not... totally |
| 18:27.11 | file | it's how authentication happens when it's needed to be precise |
| 18:27.22 | ljam | file: right :) I like to auth, and not just assume :) |
| 18:27.34 | *** join/#asterisk sysdebug (n=sysdebug@200.250.222.8) |
| 18:27.36 | file | ljam: he's probably getting a call from a provider or something |
| 18:27.59 | file | Katty: everybody look left, everybody look right, everywhere you look I'm standing spotlight |
| 18:28.01 | file | not yet! |
| 18:28.10 | Katty | teehee! |
| 18:28.15 | asterboy | yipppie! |
| 18:28.22 | Katty | i caaan see what's happening! |
| 18:28.23 | asterboy | have Buddy Watch working. |
| 18:28.27 | Katty | and they don't have a CLUE |
| 18:28.40 | asterboy | BUT you have to put the hint in the context you want to watch. |
| 18:28.41 | [TK]D-Fender | chino[server] : Using analog phones would depend on the ATA you use. Lets say you can do everything you might imagin with a full-featured analog line + transfer calls. so call-waiting, 3-way conference, transfer, hold (maybe). |
| 18:28.42 | file | Katty: Hakuna Matata! |
| 18:28.54 | file | it means no worries, for the rest of your days... |
| 18:28.55 | asterboy | Enable presence in sip.cfg for polycom phones. |
| 18:29.03 | file | it's a problem free... philosophy! hakuna matata! |
| 18:29.08 | [TK]D-Fender | chino[server] : on a SIP hard phone you can do all that without using flash + stupid * codes and the like. |
| 18:29.10 | asterboy | AND add to contact directory with bw enabled. |
| 18:29.10 | chino[server] | [TK]D-Fender: do the low end voip phones support all the features of the pbx ? |
| 18:29.17 | ljam | file: Hakuna Matata?! I was just saying that the other day too |
| 18:29.20 | [TK]D-Fender | chino[server] : Typically. |
| 18:29.28 | chino[server] | [TK]D-Fender: how much do they go for ? |
| 18:29.31 | file | ljam: :D |
| 18:29.45 | asterboy | good god...the wiki on Buddy Watch is useless: http://www.voip-info.org/wiki/view/Asterisk+presence |
| 18:29.46 | asterboy | Need a major rewrite. |
| 18:29.52 | Strom_M | taaaaake oooooon meeeeeeeee |
| 18:29.53 | ljam | asterboy: learn how it works then write something good |
| 18:30.11 | [TK]D-Fender | chino[server] : Depends where you are and what kind of quality you want, as well as the features (extras). |
| 18:30.12 | asterboy | I will |
| 18:30.20 | ComputerWarm | does anyone know of any good examples on getting a radius server working with gnugk? |
| 18:30.24 | chino[server] | [TK]D-Fender: is there any in particular that asterisk works best with and should i go through asterisk to purchase them ? |
| 18:30.50 | Strom_M | "go through asterisk: |
| 18:30.55 | Strom_M | I can see that now |
| 18:30.58 | Katty | haaakunnnnnaaaaaaaa matata |
| 18:31.00 | Katty | do do be do |
| 18:31.03 | Cherebrum | asterboy: we just got full presence and subsription working with asterisk and openpbx today |
| 18:31.08 | Strom_M | asterisk*CLI> iax2 buy phone |
| 18:31.08 | Katty | dooooschbilildydodbedilydo |
| 18:31.12 | file | Strom_C: you could send the call through Asterisk to place a phone order |
| 18:31.12 | Katty | aweeeeeeeeee dodomatataa |
| 18:31.15 | [TK]D-Fender | asterboy : the A@H handbook has a really good guide to it actually... |
| 18:31.28 | [TK]D-Fender | chino[server] : Where are you located? |
| 18:31.30 | Cherebrum | asterboy: the plan is to release the code as GPL as soon as we have all the bugs worked out |
| 18:31.31 | Katty | against popular belief, that was /not/ a hairball |
| 18:31.32 | ljam | Katty: lol |
| 18:31.33 | asterboy | ah, I'll have to check out the A@H guide. |
| 18:31.37 | Qwell[] | file: Don't be silly. We all know Asterisk can't do phones |
| 18:31.51 | file | Qwell[]: :( |
| 18:31.58 | chino[server] | [TK]D-Fender: what about collisions i mean a switch is good for talking between extensions in the internal network but what happens when every externsion is trying to talk to the outside world over one tiny rj45 and then also over a dsl line which is used by the pc's for the internet.... is there enough bandwidth to support the phone calls ? |
| 18:32.01 | [TK]D-Fender | chino[server] : Asterisk isn't a company, and its creators, Digium do not sell phones, just telecom cards.... |
| 18:32.02 | Strom_M | yes, asterisk is only for steak-over-IP |
| 18:32.03 | chino[server] | [TK]D-Fender: im located in jersey |
| 18:32.06 | Katty | file: let's dance! |
| 18:32.07 | asterboy | Cherebrum, is that different than the current method of presence |
| 18:32.09 | asterboy | ?? |
| 18:32.09 | Katty | file: on a /log/ |
| 18:32.18 | Katty | or better yet, we can do the moveitmoveit song |
| 18:32.18 | file | Katty: realllllllllllly? |
| 18:32.20 | Katty | i like to moveitmoveit |
| 18:32.23 | Katty | she like to moveitmoveit |
| 18:32.25 | Katty | we like to |
| 18:32.25 | Katty | MOVE IT |
| 18:32.53 | [TK]D-Fender | chino[server] : depends on a lot of things... collisions isn't really a normal issue, overall bandwith can be and you never mentioned anything about using the public internet for any of this yet... |
| 18:32.59 | Cherebrum | asterboy: we're putting openser in between the phones and asterisk and then passing the subscribe and notify messages relating to presence to a perl app that processes them |
| 18:33.01 | file | I can see what's happening and they don't have a clue, they'll fall in love and here's the bottom line - our trios down the tube |
| 18:33.21 | Cherebrum | asterboy: and the perl module can connect to the asterisk management console |
| 18:33.26 | [TK]D-Fender | chino[server] : Then I'd suggest Polycom IP 501's for anyone needing a general purpose phone with speakerphone, IP301's for all the lower end users |
| 18:33.33 | chino[server] | [TK]D-Fender: yes i mean right now my meridian system is using a regular phone line for our calls why cant broad band support 5 phones ? |
| 18:33.37 | asterboy | Cherebrum, but if the call presence already works, isn't that a bit overkill? |
| 18:33.39 | Cherebrum | so you can toggle presence from an app such as a agentlogin or whatever you want |
| 18:33.52 | Katty | i dedicate this to file http://video.google.com/videoplay?docid=-8042715548064436211 |
| 18:33.57 | Cherebrum | asterboy: It doesn't indicate if an agent is logged into a queue |
| 18:34.04 | [TK]D-Fender | chino[server] : You never said if you wanted * to use your analog lines or go VoIP over the internet for PSTN termination... |
| 18:34.10 | Strom_M | chino[server], because voice calls are 64kbps plus overhead and your standard DSL only has 256kbps uplink |
| 18:34.14 | chino[server] | [TK]D-Fender: but what type of wierd symptoms would we see from bandwidth issues ? |
| 18:34.16 | Cherebrum | asterboy: and this is just one step away from having bridged line appearances via SIP-B |
| 18:34.27 | [TK]D-Fender | chino[server] : Dropped calls, stutter, etc. |
| 18:34.28 | asterboy | ooooo nice |
| 18:34.43 | Cherebrum | asterboy: and asterisk only does two states for presence.. on the phone or off the phone... polycom supports 7 states |
| 18:34.54 | chino[server] | so then how does the meridian with a regular phone line support 5 phones at once ? |
| 18:35.05 | asterboy | but with SIP-B there won't be a limit of 7 |
| 18:35.14 | Strom_M | chino[server], what do you mean "regular phone line" |
| 18:35.19 | Cherebrum | the limit of 7 is a limitation of asterisk not the polycom |
| 18:35.25 | [TK]D-Fender | Cherebrum : No, asterisk supports multiple status' for presence..... |
| 18:35.31 | file | Cherebrum: that depends. |
| 18:35.34 | Cherebrum | stkn: Since when? |
| 18:35.39 | Katty | since 1904. |
| 18:35.40 | Cherebrum | er [TK]: since when? |
| 18:35.45 | chino[server] | i believe a phone line is plugged into the meridian and thats the only connection from the pstn just a regular phone line not some T1 line |
| 18:35.51 | file | Cherebrum: by using the method we support, Polycom can only handle 7 |
| 18:35.52 | [TK]D-Fender | Cherebrum : No, that is reverse.. it is POLYCOM that has the limit which they are extending in the next firmware... |
| 18:36.12 | Strom_M | chino[server], you're probably wrong. Regular POTS lines can only support one call at any given time :) |
| 18:36.16 | Cherebrum | hmm.. I was told different |
| 18:36.30 | [TK]D-Fender | chino[server] : You are missing some core telephony / VoIP concepts.... |
| 18:36.46 | chino[server] | [TK]D-Fender: why is that ? |
| 18:36.57 | [TK]D-Fender | Cherebrum : AFAIK always... |
| 18:37.21 | file | Polycom used to only support 5, then they upped it to 7 |
| 18:37.29 | [TK]D-Fender | chino[server] : You are mixing points at which VoIP is even used in your PBX layout. |
| 18:37.47 | chino[server] | [TK]D-Fender: well voip is new to me sorry |
| 18:37.54 | Cherebrum | the polycom sets can actually subscribe directly to each other when you use a proxy server and asterisk doesn't need to touch the messages |
| 18:37.56 | lzhang | I'm trying to set up a queue and I'm getting WARNING[13984] app_queue.c: Unable to join queue 'pickupq' |
| 18:38.06 | malverian | Ugh... |
| 18:38.08 | lzhang | it's busying when I'm calling the queue extension |
| 18:38.11 | [TK]D-Fender | chino[server] : Do you expect to use your existing 5 ANALOG lines with * as a PBX and SIP hardphones for your users? |
| 18:38.13 | malverian | Anyone here actually use AstManProxy ? |
| 18:38.21 | Cherebrum | malverian: yes |
| 18:38.36 | malverian | Cherebrum, You know that it doesn't forward the Asterisk banner on to connected clients? |
| 18:38.53 | malverian | This breaks the Java AsteriskManager library. |
| 18:38.53 | Cherebrum | malverian: yes.. we patched it to do that |
| 18:38.59 | chino[server] | [TK]D-Fender: i would either backwards adapt the 5 analog meridian phones or setup 5 new voip phones |
| 18:39.01 | Cherebrum | malverian: exactly |
| 18:39.02 | malverian | Cherebrum, Mind sharing? |
| 18:39.14 | malverian | I also have patched it to allow filtering of certain packets... |
| 18:39.26 | malverian | astmanproxy.conf: filter = link, unlink, newexten |
| 18:39.31 | malverian | Which I'm also willing to share :) |
| 18:40.19 | chino[server] | [TK]D-Fender: but from what you told me before the prices im not going to backwards adapt those meridian phones i would just buy new voip phones and use a ethernet switch |
| 18:40.31 | malverian | The structure of the program makes it difficult to send anything that is not a real "packet" |
| 18:41.16 | [TK]D-Fender | chino[server] : Ok, lets start this over. How many phones & lines does your current system have? |
| 18:41.17 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 18:41.29 | chino[server] | 5 |
| 18:41.44 | chino[server] | well actually 6 im sorry |
| 18:42.09 | chino[server] | but i doubt hte 6th will be needed because it was just some stupid answering machine |
| 18:42.18 | chino[server] | i know with asteriskj we just use the hd to save voice mails too |
| 18:43.56 | [TK]D-Fender | chino[server] : FAILURE... lets try again... how manay lines to the PHONE COMPANY come into your system right now? And THEN, how many total PHONES (enxtensions) |
| 18:43.56 | chino[server] | sorry if i sound confused its just my nature to ask many questions even if i know the answers |
| 18:44.37 | chino[server] | ah yes we have 6 extensions and im not sure about the external line im trying to call my boss right nwo but i believe it was just a single external line plugged into the meridian |
| 18:45.05 | znoG | [TK]D-Fender: you've made a cable with RJ21 connector before, haven't you? |
| 18:45.26 | chino[server] | wtf is a rj21 |
| 18:45.36 | asterboy | so * will support bridged line appearances in the next version release? |
| 18:45.56 | file | asterboy: in 1.4? no |
| 18:46.02 | CrashHD | lol |
| 18:46.03 | CrashHD | file |
| 18:46.04 | CrashHD | another one |
| 18:46.12 | ljam | bridged line appear! :) |
| 18:46.22 | asterboy | lol |
| 18:46.23 | [TK]D-Fender | znoG : Nope, I've always just had an interconnector company do that or me (BIX) |
| 18:46.24 | file | ljam: Segmentation fault. |
| 18:46.37 | Katty | i'll seg /your/ fault in a minute. |
| 18:46.39 | ljam | file: inconcievable! |
| 18:46.46 | asterboy | I did hit backspace for that but said fuck it. |
| 18:46.46 | CrashHD | file what would it take consulting/time/money wise to do the bridge line appearances |
| 18:46.55 | chino[server] | [TK]D-Fender: sorry am i being a pain |
| 18:46.56 | file | CrashHD: talk to Digium |
| 18:46.59 | znoG | i bought a RJ21 connector, i have my multi-pair cable (25 pairs) and I don't get how I will be able to tell which cables belong to each pair. In other words, if I connect up the 25 pairs to the RJ21, how do I tell on the other end of the cable which pair is which? |
| 18:47.01 | ljam | Katty: you're sending mixed signals :) |
| 18:47.15 | justinu|laptop | they should be color coded |
| 18:47.18 | Katty | s/ljam/captainobvious/ |
| 18:47.24 | [TK]D-Fender | BRB.... |
| 18:47.39 | [TK]D-Fender | ljam : As you wiiiiiisssshhh! |
| 18:47.52 | Katty | dwedded piwate wobberts! |
| 18:48.09 | justinu|laptop | znoG: http://www.levitonvoicedata.com/learning/pairingdiagram.asp |
| 18:48.16 | justinu|laptop | 2 seconds searching google gave me that link |
| 18:48.23 | znoG | justinu|laptop: i sweeeear i googled for this |
| 18:48.27 | Katty | i think i'm going to name my next cat princess buttercup |
| 18:48.32 | file | when it comes to fixing prices, there are lots of tricks he knows! how it all increases only bits and pieces |
| 18:48.38 | Katty | unless it's black, inwhich case i'm naming it tarball |
| 18:48.50 | asterboy | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg130884.html |
| 18:49.02 | Katty | file: tarball's a fine name for a kitty, don't you think? |
| 18:49.11 | file | Katty: indeed |
| 18:49.12 | asterboy | "we are |
| 18:49.14 | asterboy | > very seriously considering doing it in the near future" |
| 18:49.15 | Strom_M | Katty, I always wanted to name a cat "inky" |
| 18:49.21 | Katty | Strom_M: that's cute too! |
| 18:49.36 | Strom_M | but then I discovered I'm allergic to cats, so I bestowed the name on my thinkpad instead |
| 18:49.42 | asterboy | seems to suggest 1.4 will have it. |
| 18:49.45 | znoG | justinu|laptop: thanks mate :) i googled for the wrong thing, by the look of it. Should have taken RJ21 out of the string |
| 18:49.52 | asterboy | may be wishful thinking. |
| 18:49.54 | Katty | :< |
| 18:50.05 | Katty | i don't think inky's a very good laptop name |
| 18:50.11 | Katty | mine's named Morgana |
| 18:50.20 | file | asterboy: it's wishful thinking |
| 18:50.23 | asterboy | lol |
| 18:50.28 | file | asterboy: 1.4 is due out in a few months |
| 18:50.35 | Katty | it's a pretty, girly, curvacious laptop. |
| 18:50.38 | file | and we're in architectural freeze right now |
| 18:50.39 | znoG | justinu|laptop: now, the next problem. Once I've wired all that up to the RJ21, on the other end of the cable there are many blue cables, many orange cables, etc... so how to tell which is which? |
| 18:50.42 | Katty | else i'd have named it sebastian |
| 18:50.42 | asterboy | but without bla |
| 18:51.07 | justinu|laptop | znog: fyi, my search stringwas "rj21x telco color coding" |
| 18:51.29 | asterboy | This is why my dial plan was so hard for me at first. I was trying to setup the lines the same way you would a bla |
| 18:51.37 | justinu|laptop | znog: the color coding should follow the standard laid out in the previous link... blue/white pairs with white/blue |
| 18:51.46 | Katty | why is changing time left as a root thing? |
| 18:51.54 | file | see, Asterisk isn't a key system... as much as people want it to be (at least right now it isn't) |
| 18:52.11 | Cherebrum | Katty: my old roommate had a cat named Steve. Which is an odd name for a cat. imagine saying, "I'm petting Steve." |
| 18:52.15 | asterboy | ya, I'm just starting to figure that out. |
| 18:52.28 | file | it's a PBX. |
| 18:52.29 | asterboy | too bad it took some head banging on desk to get there. |
| 18:52.32 | [TK]D-Fender | chino[server] : PM |
| 18:52.35 | Katty | Cherebrum: steve likes his tail pulled. |
| 18:52.46 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 18:53.17 | asterboy | It gets close with the Flash Operator Panel, in terms of management. |
| 18:53.28 | Katty | FOP is pretty nice when it works (= |
| 18:53.39 | asterboy | Looks fantastic |
| 18:53.48 | Katty | sometimes it deadlocks the server tho |
| 18:53.51 | Katty | i dunno why |
| 18:54.03 | asterboy | FOP would be nice on a CISCO phone with the big color display. |
| 18:54.31 | asterboy | A phone with a TOUCH screen and FOP would kick ass. |
| 18:54.48 | Katty | kicking ass isn't all that fun |
| 18:54.58 | asterboy | especially when they just shit themselves. |
| 18:55.07 | Katty | gosh |
| 18:55.09 | file | asterboy: you reminded me of something I need to do for BLA |
| 18:55.09 | CrashHD | stinky |
| 18:55.19 | CrashHD | BLA! |
| 18:55.21 | asterboy | good...get that going. |
| 18:55.40 | asterboy | All I hear is BLA, BLA, BLA |
| 18:55.49 | Katty | file: explain to me why changing time is a root thing |
| 18:55.53 | CrashHD | the only system I could find that implemented BLA was that new shitty linksys phone system |
| 18:55.54 | CrashHD | hah |
| 18:56.09 | malverian | Katty, Because it affects your entire system. |
| 18:56.19 | file | CrashHD: which one? |
| 18:56.22 | malverian | Katty, Some programs are time-dependent, and could do specific things based on the time. |
| 18:56.32 | CrashHD | file: the SLA 9000 |
| 18:56.38 | Katty | oh. |
| 18:56.40 | Katty | like cron jobs? |
| 18:56.43 | asterboy | ah |
| 18:56.44 | justinu|laptop | i want to see the SIP-B standard |
| 18:56.45 | malverian | Katty, Exactly. |
| 18:56.47 | justinu|laptop | how does it work? |
| 18:56.48 | Katty | ooh |
| 18:56.50 | Katty | that'd be skeery |
| 18:57.00 | CrashHD | file: spa9000 |
| 18:57.19 | file | CrashHD: now - you realize... that it's Asterisk? |
| 18:57.38 | CrashHD | CrashHD: I wasn't sure, I saw mention of asterisk for voicemail |
| 18:57.41 | CrashHD | File: |
| 18:57.42 | CrashHD | lol |
| 18:57.45 | CrashHD | talking to myself |
| 18:57.46 | CrashHD | gosh |
| 18:57.52 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 18:57.54 | file | talking to yourself is a mental condition, don't make me have you commited |
| 18:57.54 | Katty | loler. |
| 18:58.02 | asterboy | http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1139414817110&pagename=Linksys%2FCommon%2FVisitorWrapper |
| 18:58.11 | Katty | lolerskates. |
| 18:58.25 | CrashHD | file what about taking their code under the GPL? |
| 18:58.43 | file | CrashHD: http://voxilla.com/name-News-article-sid-173.html read that |
| 18:59.17 | brettnem | woah, spa9000? |
| 18:59.58 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 19:00.16 | asterboy | great, now were going to have just as many versions of * floating around as we have in linux distros |
| 19:00.27 | hgaillac | file: I set a peer entry insecure=very what's the ip address of the invite |
| 19:00.28 | asterboy | I'll have to make a web page called: * watch |
| 19:00.40 | CrashHD | file: it says their changes are getting integrated into asterisk |
| 19:00.54 | asterboy | "their" version of * |
| 19:01.02 | CrashHD | ahhh |
| 19:01.09 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 19:01.14 | asterboy | hence the need for an * watch |
| 19:01.16 | file | hgaillac: I don't know, I'm not your provider or whoever is sending you the call |
| 19:01.18 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 19:01.34 | Cherebrum | CrashHD: Are you talking about OpenPBX? |
| 19:01.51 | file | he's talking about the SPA9000 |
| 19:01.55 | asterboy | but...they do say that * will integrate those changes. |
| 19:02.11 | chino[server] | [TK]D-Fender: PM |
| 19:02.13 | asterboy | not sure how they qualify that. |
| 19:02.33 | CrashHD | can't their code be reused and integrated into the asterisk trunk? |
| 19:02.37 | CrashHD | under the GPL? |
| 19:03.11 | CrashHD | if they are using asterisk |
| 19:03.17 | JonR800 | asterisk is gpl? |
| 19:03.21 | CrashHD | ohh |
| 19:03.22 | CrashHD | guess not |
| 19:03.29 | CrashHD | didn't know asterisk wasn't gpl |
| 19:03.49 | justinu|laptop | hey JonR800 |
| 19:03.54 | rpm | exten => 715011,hint,SIP/1501a&SIP/1501b |
| 19:03.54 | rpm | exten => 71501,1,Macro(tl-simple-standard,715011) |
| 19:03.55 | justinu|laptop | ltns, how you been? |
| 19:03.55 | JonR800 | hey man |
| 19:04.00 | rpm | anyone know what that hint doesn't work? |
| 19:04.28 | chino[server] | [TK]D-Fender: are you registered ? |
| 19:04.34 | Hmmhesays | Bueller |
| 19:05.02 | asterboy | cause you have and extra "1" in the hint's extension number? |
| 19:05.39 | asterboy | or your missing a "1" at the end of the Macro extension |
| 19:05.41 | hgaillac | file: I use ser as outbound sip proxy so i want to send sip:info@xxx.xxx.xxx.xxx to asterisk where info is a context with type=peer insecure very |
| 19:06.38 | MRH2 | and ,71501,1 ... 71501,2 |
| 19:08.24 | *** join/#asterisk FlyboySR22 (n=rsears@sdtc.ar01.f2-40.host2.1.americanis.net) |
| 19:08.26 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 19:08.35 | *** join/#asterisk utsl (i=nhawkins@earth.v2business.com) |
| 19:10.10 | znoG | justinu|laptop: heh, the whole problem was that i didn't realize the cable is actually twisted, even if it's not very twisted. |
| 19:10.21 | znoG | just slightly but enough to tell apart from others |
| 19:10.54 | justinu|laptop | so it doesn't have the correct color coding on the wires? |
| 19:10.56 | brettnem | maybe it's STP "slightly twisted pair" |
| 19:11.13 | mishehu | Standard Temperature Pressure ? |
| 19:11.14 | mishehu | heh |
| 19:11.20 | utsl | STP=shielded |
| 19:11.25 | brettnem | I know.. : |
| 19:11.27 | brettnem | :P |
| 19:11.34 | brettnem | or Signal Transfer Point |
| 19:11.40 | brettnem | Stone Temple Pilots |
| 19:11.57 | mishehu | Solid Transferrable Plutonium |
| 19:12.08 | Katty | yay, mishehu |
| 19:12.15 | mishehu | Some Trip Purposefully |
| 19:12.25 | Katty | porpoisefully |
| 19:12.25 | mishehu | Katty: I can't make you fit in that acronym ;-) |
| 19:12.46 | Katty | details, details |
| 19:12.50 | Katty | don't confuse me with the facts! |
| 19:13.30 | mishehu | I'd like to see someone come up with what my nickname would stand for if it was an acronym. |
| 19:13.33 | brettnem | http://www.mit.edu/people/mjbauer/ihtfp.html |
| 19:13.45 | brettnem | IHTFP! |
| 19:13.58 | justinu|laptop | MISter HEfty HUge? |
| 19:14.04 | justinu|laptop | sorry... it just came to mind |
| 19:14.38 | Katty | microscopic internal super hot electric heliotope univisims. |
| 19:15.02 | Katty | i don't think those last two are really words, but they sound /real/ good |
| 19:15.04 | Cherebrum | This fucking place hates you, too. |
| 19:15.11 | file | hgaillac: so send it to a clearing context that matches everything, and then use a Goto to send it to the specified context |
| 19:15.12 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 19:15.16 | Katty | Cherebrum: simmer down. |
| 19:15.40 | Cherebrum | Katty: read the RFC.. that's the proper responce |
| 19:16.07 | brettnem | Cherebrum: no, that's TFPHY |
| 19:16.13 | *** join/#asterisk test34 (i=1000@unaffiliated/test34) |
| 19:16.23 | Cherebrum | :p |
| 19:17.01 | Katty | this person isn't an aussie |
| 19:17.42 | mishehu | Katty: doesn't make sense though... |
| 19:17.59 | hgaillac | file: [info]exten => info,1,Answer()exten => info,n,Dial(Sip/84,10)exten => info,n,Dial(Sip/85,10)exten => info,n,Hangup |
| 19:18.02 | mishehu | a friend once upon a time came up with a funny one involving eskimos, but I can't remember what it was. |
| 19:18.26 | Katty | mishehu: i never make sense. |
| 19:20.14 | znoG | justinu|laptop: yep it does, i just didn't realize they were twisted (a little) so i couldn't understand how I was going to tell apart the different blues, whites, etc. |
| 19:24.31 | *** join/#asterisk afrosheen (n=test@txprotoa2.august.net) |
| 19:24.43 | afrosheen | has anyone seen any weirdness with 'missing' voicemails in 1.2.5 |
| 19:26.33 | CoffeeIV_ | is there a special way you have to specify the single-digit extension 0 in the dial plan ? |
| 19:26.34 | afrosheen | wow I guess it's lunchtime |
| 19:27.10 | *** join/#asterisk mattwj2005 (n=Matt@user-12l3lqm.cable.mindspring.com) |
| 19:27.33 | [TK]D-Fender | CoffeeIV_ : Yeah... exten => 0,1,DoSomethingUsefulHere() |
| 19:27.57 | mattwj2005 | hey guys I have a license question for you |
| 19:28.31 | utsl | CoffeeIV_: except when it needs to be exten => o,1,etc. |
| 19:28.39 | mattwj2005 | my ata supports G.729 and so does my service provider |
| 19:28.42 | CoffeeIV_ | D-Fender: that's exactly what isn't working for me -- it hangs up when I dial 0 |
| 19:28.53 | CoffeeIV_ | i'll try "o" |
| 19:28.55 | [TK]D-Fender | CoffeeIV_ : Pastebin your dialplan... |
| 19:28.56 | [TK]D-Fender | ~pb |
| 19:28.57 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 19:28.57 | mattwj2005 | so can I legally use it? |
| 19:29.11 | [TK]D-Fender | CoffeeIV_ : No, not "o" thats reserved... |
| 19:29.13 | afrosheen | mattwj2005: as long as both ends have a valid license it should work |
| 19:29.27 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 19:29.50 | mattwj2005 | sweet! 8kps communications here we come :-D |
| 19:29.51 | Qwell[] | afrosheen: You won't be able to use voicemail or anything, unless Asterisk has a license |
| 19:29.55 | Qwell[] | erm |
| 19:29.57 | Qwell[] | mattwj2005: |
| 19:30.10 | afrosheen | hence why I said 'both ends' |
| 19:30.32 | Qwell[] | Unless asterisk has a license, you can only do passthrough |
| 19:30.32 | afrosheen | his provider has one, now he needs one |
| 19:30.43 | afrosheen | it's what, $10 or $5US per license? |
| 19:30.47 | Qwell[] | $10 |
| 19:30.53 | afrosheen | a bargain at twice the price |
| 19:31.05 | mattwj2005 | true but I am just using my asterisk box as a throughput device for my service provider |
| 19:31.22 | Qwell[] | then sure, it'll work fine |
| 19:31.35 | *** join/#asterisk cdtrips (n=cdtrips@zentality.v2business.com) |
| 19:32.42 | afrosheen | as long as you're not doing voicemail and whatnot with * |
| 19:33.50 | shido6 | can I use PCIe with TDM cards |
| 19:34.03 | Qwell[] | pcix |
| 19:34.29 | shido6 | does anyone make a PCIe TDM card? |
| 19:34.39 | Qwell[] | not that I've seen... |
| 19:34.52 | afrosheen | what'd be the point, a pcie slot is for high bandwidth video cards |
| 19:34.57 | afrosheen | 64bit pci, yeah |
| 19:35.01 | afrosheen | I can see that |
| 19:35.04 | Qwell[] | afrosheen: some boxes don't have pci |
| 19:35.21 | Qwell[] | but most (all?) digium pci cards should work in a pcix slot |
| 19:35.22 | afrosheen | I've never seen a real server without pci for the last...7 years or more |
| 19:36.18 | Yasheee | When I have an asterisk voicemail, is there a way to have the phone in my X100P card "blink" like it does when there are messages at the telco? |
| 19:36.29 | Qwell[] | Yasheee: sure |
| 19:36.49 | Qwell[] | wait |
| 19:36.51 | afrosheen | Yasheee: I think that feature is called 'hint' |
| 19:37.00 | Qwell[] | x100p is fxo |
| 19:37.07 | afrosheen | ooooh |
| 19:37.13 | file | the port on the X100P where you plug in a phone is passthru |
| 19:37.16 | *** part/#asterisk kakadu (n=blubb@p54B8D2ED.dip.t-dialin.net) |
| 19:37.16 | file | Asterisk doesn't even know it's there |
| 19:37.22 | CoffeeIV_ | does WaitExten() not handle zero or something ? |
| 19:37.38 | Yasheee | oh okay |
| 19:37.43 | Yasheee | so I'd need an fxs to do so to all the phones in my house? |
| 19:37.45 | afrosheen | CoffeeIV_: I'm pretty sure 0 is reserved, I remember seeing that in the code and in the docs, I could be wrong though |
| 19:38.58 | *** join/#asterisk Bob27 (n=a@modemcable050.60-203-24.mc.videotron.ca) |
| 19:39.23 | Bob27 | anybody home? |
| 19:39.37 | Strom_M | no |
| 19:39.39 | Strom_M | we're all dead |
| 19:39.41 | Strom_M | sorry |
| 19:39.43 | Strom_M | go home |
| 19:39.46 | Strom_M | live your life |
| 19:39.50 | Strom_M | spend time with your children |
| 19:39.54 | Bob27 | saddddddddd |
| 19:40.16 | Bob27 | it makes me wondering what is my purpose in life |
| 19:40.46 | Bob27 | dont make me cry |
| 19:40.53 | Bob27 | now are you there? |
| 19:42.05 | Bob27 | dring dring? |
| 19:42.18 | fourcheeze | Bob27: are you waiting for a message from above or something? |
| 19:42.21 | Bob27 | I wonder if there is a way to install asterisk@home over a network? |
| 19:42.29 | *** join/#asterisk Vahram (n=nospam@83.139.4.112) |
| 19:42.42 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:42.47 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-92.apple.com) |
| 19:42.48 | fourcheeze | Bob27: ask on #freepbx |
| 19:42.52 | Bob27 | ok thanks |
| 19:43.07 | afrosheen | why bug the amp people with @home questions |
| 19:43.18 | afrosheen | they won't know |
| 19:43.26 | Strom_M | afrosheen, who cares - it's all the same cesspool |
| 19:43.31 | afrosheen | ouch |
| 19:43.38 | fourcheeze | Bob27: ok don't ask there |
| 19:43.52 | Bob27 | ok |
| 19:44.04 | fourcheeze | ask on #bzflag ;-) |
| 19:44.06 | afrosheen | my five bucks is on 'can't install @home over a network, it requires physical presence just like every other linux install' |
| 19:44.30 | Yasheee | I have: exten => s,1,Wait(4); Is there a way to specify that is someone picks up a telephone in that 4 seconds, to Monitor() insteading of going to the s,2,Voicemail sequence? |
| 19:44.34 | fourcheeze | you can install debian over a network |
| 19:44.34 | Strom_M | my six seventy-five is on 'why the hell do you want @home anyway?' |
| 19:44.34 | Corydon-w | If you email the developer of A@H with about 15 emails, spaced 2 minutes apart, he'll be sure to help you, then. |
| 19:44.43 | Yasheee | s/that is/that if/ |
| 19:45.06 | Yasheee | lol crazy |
| 19:45.11 | afrosheen | fourcheeze: yeah but you gotta sit there and boot from something to begin with right? |
| 19:45.30 | afrosheen | Corydon-w: lol |
| 19:46.07 | fourcheeze | afrosheen: I think by "install over network" doesn't he mean that he doesn't want to download all the media? |
| 19:46.07 | fourcheeze | that's how I took it |
| 19:46.12 | afrosheen | fourcheeze: afaik it's an ISO that you burn and install from |
| 19:46.15 | Bob27 | Strom_M: cause I'm not a linux wizz and appreciate all the packaging of the @home version but unfortunately I have only ssh access to the server |
| 19:46.21 | afrosheen | I could be wrong but I don't follow that |
| 19:46.41 | Strom_M | Bob27, you don't need to be a linux whiz |
| 19:46.45 | Qwell[] | Bob27: No, if you only have ssh access, it isn't possible |
| 19:46.45 | fourcheeze | Bob27: what's running on your server? |
| 19:46.51 | fourcheeze | which distro? |
| 19:46.52 | justinu|laptop | Bob27: if you're serious about learning asterisk, you'd do well to learn linux too |
| 19:46.53 | afrosheen | it's not 'whiz', it's 'wizard' |
| 19:46.54 | Bob27 | nothing yet I am planning |
| 19:47.10 | afrosheen | with the pointy hats and wands |
| 19:47.12 | fourcheeze | Bob27: it's not at all hard to install asterisk |
| 19:47.13 | Bob27 | I installed the @home version on my dev PC and I begin to learn |
| 19:47.13 | Strom_M | liunx whiz - it's like cheese whiz, but open source |
| 19:47.13 | justinu|laptop | it's really not all that difficult |
| 19:47.30 | justinu|laptop | what's the best liveCD distro of linux? |
| 19:47.33 | justinu|laptop | or liveDVD |
| 19:47.39 | fourcheeze | gentoo :-) |
| 19:48.02 | fourcheeze | gentoo live cd: everything you need to build a linux system :-) |
| 19:48.34 | afrosheen | kororaa |
| 19:48.35 | afrosheen | :) |
| 19:48.45 | Bob27 | I knew linux now I'm an evil Microsoft disciple |
| 19:48.49 | afrosheen | it's the xgl eye candy master |
| 19:49.02 | fourcheeze | justinu|laptop: if you want gui stuff try ubuntu/kubuntu livecd |
| 19:49.08 | fourcheeze | Bob27: return to the fold |
| 19:49.25 | fourcheeze | you know if makes sense |
| 19:49.46 | Bob27 | :) |
| 19:49.50 | justinu|laptop | yeah... i just wanna see where these distros are at before I commit to installing on my laptop |
| 19:50.25 | afrosheen | justinu|laptop: actually if you're shooting for a multimedia-gui laptop, try PCLinuxOS also |
| 19:50.28 | lzhang | how do I use realtime extensions with AgentCallbackLogin? |
| 19:50.43 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-92.apple.com) |
| 19:50.44 | lzhang | I'm not sure what to put into the context for AgentCallbackLogin |
| 19:50.58 | Bob27 | anyway, thanks guy |
| 19:51.00 | justinu|laptop | PCLinuxOS? ok.. i'll take a looksee |
| 19:51.12 | Bob27 | +s |
| 19:51.19 | *** part/#asterisk Bob27 (n=a@modemcable050.60-203-24.mc.videotron.ca) |
| 19:52.38 | terrapen | any polycom owners here? |
| 19:52.42 | afrosheen | <PROTECTED> |
| 19:52.47 | terrapen | why the hell are my SNTP settings not taking |
| 19:52.55 | afrosheen | polycom is a-1 golden choice for #asterisk |
| 19:52.55 | terrapen | my clock is still flashing! |
| 19:53.04 | terrapen | it's like the VCR clock problem of the 1980s |
| 19:53.19 | triple-e | terrapen: your dhcp server isn't setting the ntp setting |
| 19:53.28 | afrosheen | he's absolutely right |
| 19:53.37 | triple-e | so its refering to the config |
| 19:53.38 | terrapen | yeah, it's not....but I provided an SNTP setting in sip.cnf |
| 19:53.39 | afrosheen | dhcp has to feed that setting |
| 19:53.53 | afrosheen | terrapen: do you have access to the dns server? |
| 19:53.59 | terrapen | <SNTP tcpIpApp.sntp.resyncPeriod="3600" tcpIpApp.sntp.address="131.107.1.10" tcpIpApp.sntp.gmtOffset="-25200" ... |
| 19:54.00 | triple-e | that it is downloading from the config |
| 19:54.17 | terrapen | afro, you mean the DHCP server? |
| 19:54.27 | terrapen | Yes, I do. But I was hoping to not mess with it. |
| 19:54.37 | justinu|laptop | terrapen: try packet sniffing the phone, see why it fails to talk to with the NTP server |
| 19:54.59 | afrosheen | it's not a big change, I can show you what our microsoft dhcp server feeds our phones |
| 19:55.05 | justinu|laptop | or perhaps increase the logging level of the phone itself, it might say something useful |
| 19:55.07 | terrapen | afro, yeah, I know, it's easy. but again, i'm just wondering why the sip.cnf setting is not taking |
| 19:55.10 | triple-e | terrapen: look for this |
| 19:55.11 | triple-e | ipmid.cfg: <SNTP tcpIpApp.sntp.resyncPeriod="86400" tcpIpApp.sntp.address="0.us.pool.ntp.org" |
| 19:55.32 | triple-e | sip.cfg: <SNTP tcpIpApp.sntp.resyncPeriod="86400" tcpIpApp.sntp.address="0.us.pool.ntp.org" |
| 19:56.00 | afrosheen | terrapen: just set it on the server, won't hurt anythin |
| 19:56.22 | terrapen | i just pasted my cfg: |
| 19:56.22 | terrapen | <SNTP tcpIpApp.sntp.resyncPeriod="3600" tcpIpApp.sntp.address="131.107.1.10" tcpIpApp.sntp.gmtOffset="-25200" |
| 19:56.28 | terrapen | Yes, I know that its easy to do on the DHCP server |
| 19:56.35 | terrapen | again, I want to make sip.cfg work as it should |
| 19:57.16 | terrapen | our DHCP server is on our Cisco |
| 19:57.21 | terrapen | not something I really want to mess with. |
| 19:57.35 | afrosheen | I don't blame you |
| 19:57.39 | *** join/#asterisk trex005 (n=Travis@oh-65-40-131-234.sta.sprint-hsd.net) |
| 19:58.17 | triple-e | look in your ipmid.cfg whats the sntp address there ? |
| 19:58.59 | justinu|laptop | ipmid.cfg is deprecated |
| 19:59.13 | justinu|laptop | and he just pasted the line twice |
| 19:59.24 | terrapen | my phone is not looking at ipmid.cfg |
| 19:59.41 | terrapen | <APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="x101/x101.cfg, sip.cfg" MISC_FILES="" LOG_FILE_DIRECTORY="x101" OVERRIDES_DIRECTORY="x101" CONTACTS_DIRECTORY="x101"/> |
| 19:59.50 | trex005 | hello folks...... I'm looking at picking up "Asterisk: The Future of Telephony" is that my best way to get a foundational understanding? |
| 20:00.29 | afrosheen | ~pb |
| 20:00.31 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 20:00.43 | terrapen | i suppose i could set the SNTP server on the phone |
| 20:00.46 | terrapen | but that's cheating |
| 20:00.53 | terrapen | must....make.....work |
| 20:01.01 | trex005 | is that a no? |
| 20:01.04 | afrosheen | terrapen: here's a working config for california |
| 20:01.05 | afrosheen | http://pastebin.com/646726 |
| 20:01.11 | terrapen | trex, try it and see if it works for you |
| 20:02.20 | trex005 | terrapen: I'm looking for concentrated in depth learning.... is that possible? :) |
| 20:02.42 | justinu|laptop | the book is a good star trex |
| 20:02.43 | terrapen | http://pastebin.com/646733 |
| 20:02.49 | terrapen | see, my server is valid. |
| 20:03.38 | terrapen | http://pastebin.com/646734 |
| 20:03.43 | terrapen | that's my config bit, too |
| 20:04.23 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 20:04.23 | terrapen | INTERESTING |
| 20:04.27 | terrapen | i think i just figured it out |
| 20:04.31 | terrapen | stupid firewall! |
| 20:04.39 | terrapen | phone is on a different subnet |
| 20:04.51 | terrapen | and for some reason, we're blocking NTP |
| 20:05.14 | *** join/#asterisk kavit (n=kavit@210.10.211.58) |
| 20:06.30 | kavit | hello all, I have been looking to read up on DUNDi and theres not much out there on the web apart from a page on voip-info.org, does anyone have any links/tutorials for DUNDi ? |
| 20:07.13 | *** join/#asterisk rene- (n=rene-@201.137.74.112) |
| 20:07.32 | Rawplayer | what is wrong with this? |
| 20:07.34 | Rawplayer | exten => _XXXXXXX,1,Dial,SIP/outgoing/0031[b]45[/b]${EXTEN} |
| 20:07.34 | Rawplayer | exten => _XXXXXXX,2,Congestion |
| 20:07.45 | Rawplayer | our country code is 0031 |
| 20:07.56 | Rawplayer | or local code is 045 |
| 20:08.02 | ManxPower | kavit, dundi.com ? |
| 20:08.19 | ManxPower | Rawplayer, it's wrong. |
| 20:08.21 | Rawplayer | when someone calls a phonenumber that is local then asterisk should add that to that number |
| 20:08.35 | ManxPower | [b] would be wrong so would [/b] |
| 20:08.44 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 20:08.53 | Rawplayer | so when someone calls 5847393 on his softphone asterisk should add 003145 in front of that number |
| 20:08.53 | rene- | hello, i have gotten two avaya ip phones, i believe they run h323, the models are 4610 and 5610, i have heard that asterisk supports woomera, ooh323 and the old h323 implementation, i might be lucky and this can be reflashed to sip, id appreciate some advice about how to interconnect those avaya phones to * |
| 20:09.02 | ManxPower | [av]bani, b, /, ] are not valid phone number digits |
| 20:09.30 | ManxPower | Rawplayer, [, b, /, ] are not valid phone number digits |
| 20:10.23 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 20:10.23 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.2.6 and Zaptel 1.2.5 Released! (March 27, 2006) -=- http://www.asterisk.org/ -=- AMP/FreePBX users should join #freepbx for support |
| 20:10.25 | ManxPower | perhaps you want Dial(SIP/outgoing/003145${EXTEN}) |
| 20:10.35 | Rawplayer | maybe hmm |
| 20:10.47 | kavit | ManxPower: it has a technical spec of DUNDi protocol and I am reading through it, I was also interested in learning how to configure the service for asterisk. I guess I should have phrased my question better. |
| 20:11.07 | ManxPower | kavit, nothing in /path/to/src/asterisk/docs |
| 20:11.11 | ManxPower | for Dundi? |
| 20:11.31 | afrosheen | dundi sounds like a curry restaurant |
| 20:11.53 | kavit | ManxPower: I will have a look, Google is often my first and last resort |
| 20:11.53 | wasim | mmmh spices |
| 20:12.31 | mcf3782 | BellSouth has managed to configure my POTS line to not deliver CallerID info. I've discovered that I can temporarily "fix" it by dialing "*82" prior a call. What's the safest way for me to add the ability to dial that to my dial plan? |
| 20:12.58 | [av]bani | rene-: you should be able to get sip for 4610 |
| 20:13.09 | ManxPower | mcf3782, put *82w in front of your ${EXTEN} in your Dial line |
| 20:13.44 | rene- | [av]bani: cool |
| 20:14.16 | mcf3782 | Ahhh! 'w'. That's what I've been missing. |
| 20:14.22 | mcf3782 | Thanks, ManxPower. :) |
| 20:14.58 | ManxPower | mcf3782, w means wait .5 sec |
| 20:15.54 | mcf3782 | Yep. I see that in the docs now that you've mentioned it... I think I've run into the 'forrest/trees' problem. Sometimes all it takes is another pair of eyes. |
| 20:15.56 | *** join/#asterisk apardo (n=apardo@87.218.45.206) |
| 20:16.02 | ManxPower | my example only works on ANALOG FXO ports, BTW. |
| 20:16.40 | mcf3782 | And since it's getting close to 1700EDT here in Atlanta; it's getting very close to time for a good cold pitcher of hard cider, which should definitely improve things. :) |
| 20:17.32 | rene- | [av]bani: any word on SIP compatability for the 5610 sets? |
| 20:18.35 | *** join/#asterisk trimi` (i=Whatt@62.162.243.36) |
| 20:18.38 | *** join/#asterisk ToTo (n=ToTo@host129-167.pool872.interbusiness.it) |
| 20:18.50 | mcf3782 | hehe |
| 20:19.02 | *** join/#asterisk Derkommissar (n=Alberto@adsl-153-47-91.mia.bellsouth.net) |
| 20:19.23 | trimi` | Hey PPL is there any AGI script for asterisk, that will accept calls only from a list of number, and all numbers outside of list will be blocked ? |
| 20:19.42 | *** part/#asterisk utsl (i=nhawkins@earth.v2business.com) |
| 20:20.04 | afrosheen | what's PPL |
| 20:20.23 | wasim | Pigeon Packet Loop |
| 20:20.27 | afrosheen | gotcha |
| 20:20.43 | afrosheen | I heard fog will cause packet collisions |
| 20:20.48 | shido6 | does PPp stand for Pigeon Packet Poop ? |
| 20:20.53 | rene- | trimi: you can do it yourself quite easily and in the dialplan and this would make the thing faster to run, i see two options internal asterisk DB or using mysql queries from the dialplan |
| 20:21.22 | trimi` | <rene-> im begginer |
| 20:21.29 | trimi` | i tried with Gotoif |
| 20:21.44 | trimi` | but everytime i want to add new number i have to edit extensions.conf |
| 20:21.48 | trimi` | its not a solution |
| 20:22.00 | rene- | the easier will be asterisk DB, it is really easy, i would define an extension to add numbers to the database, which is accomplished tru dbput |
| 20:22.13 | brettnem | eq asterisk db |
| 20:22.15 | brettnem | eww |
| 20:22.25 | rene- | you should also have an extension to delete numbers from the database |
| 20:22.52 | lzhang | I'm trying to hotdesk with AgentLogin, how can I achieve this? |
| 20:29.58 | *** join/#asterisk ToTo (n=ToTo@host129-167.pool872.interbusiness.it) |
| 20:31.00 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-96-30.telkomadsl.co.za) |
| 20:31.22 | SplasPood | hrm... You can have a Background() and sub extens for the menu options within a macro, no? |
| 20:32.23 | hgaillac | Anybody could explain me how to disable 407 prxy authentication ? |
| 20:36.27 | *** join/#asterisk kend (n=chatzill@host-64-65-199-187.man.choiceone.net) |
| 20:36.49 | *** part/#asterisk trex005 (n=Travis@oh-65-40-131-234.sta.sprint-hsd.net) |
| 20:39.04 | SplasPood | Didn't someone answer that question before? |
| 20:39.59 | mcf3782 | i know it was terribly campy and not very well written, but I liked the show. :) |
| 20:40.05 | mcf3782 | oops |
| 20:40.07 | mcf3782 | wrong window |
| 20:41.13 | gbodemantv | hello all |
| 20:41.28 | gbodemantv | so I have got my realtime queue to work |
| 20:41.54 | gbodemantv | but it seems that it only notices changes to the queue with a reload |
| 20:41.58 | gbodemantv | any way around this |
| 20:41.59 | gbodemantv | ?? |
| 20:43.19 | Katty | SplasPood: i think that person has just been asking that question every 15 minutes or something and ignoring all help |
| 20:43.32 | justinu|laptop | lol |
| 20:43.34 | justinu|laptop | katty's right |
| 20:45.26 | *** join/#asterisk Gunnar (n=gunnar@212.130.108.131) |
| 20:45.34 | docelm0 | Say can one tell me an easy way to know if asterisk is in the RTP stream? |
| 20:48.06 | Yasheee | I have: exten => s,1,Wait(4); Is there a way to specify that is someone picks up a telephone in that 4 seconds, to Monitor() insteading of going to the s,2,Voicemail sequence? |
| 20:49.47 | tzanger | Yasheee: the dial macro option? |
| 20:51.47 | mattwj2005 | so what is the easiest way to install me in Debian linux? |
| 20:51.52 | mattwj2005 | *meetme |
| 20:52.05 | mattwj2005 | should I try a test release? |
| 20:53.14 | rpm | exten => s,n,GotoIf($[ "${CALLERIDNAME}" : "Autowakeup" ]?noDnd:guestCheck) - does that not work anymore in asterisk? |
| 20:53.29 | Yasheee | tzanger: Dial()? This is an X100P card |
| 20:53.48 | Qwell[] | rpm: it never did |
| 20:53.52 | [av]bani | \o/ |
| 20:54.01 | Qwell[] | : != = |
| 20:54.18 | Qwell[] | The $[] will always return the same value... |
| 20:54.40 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 20:55.13 | generalhan | whats up all |
| 20:55.47 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcd.mn.charter.com) |
| 20:56.07 | SpaceBass | work...sucking...thats whats up :) |
| 20:56.17 | generalhan | thats ALWAYS whats up ! lol |
| 20:56.31 | Hmmhesays | of course |
| 20:56.32 | SpaceBass | lol...so true |
| 20:56.51 | Yasheee | for me, it sucks not working :) |
| 20:56.59 | generalhan | lol |
| 20:57.00 | SpaceBass | hey Hmmhesays ...how are you? got a gig this weekend? |
| 20:57.20 | Hmmhesays | no gigs this weekend, just some drinking |
| 20:57.23 | Katty | he's always got a gig |
| 20:57.25 | Hmmhesays | motorcycle riding, guitar playing |
| 20:57.26 | Katty | he practically /is/ the gig |
| 20:57.29 | SpaceBass | i was looking forward to unwinding outside with a beer and music |
| 20:57.32 | Katty | if by gig i mean amusing |
| 20:57.35 | generalhan | im having some issues with a TDM40B if someone could take a look at these errors/warnings for me i would really appreciate it :: http://generalhan.pastebin.ca/48559 |
| 20:57.42 | Hmmhesays | i'm trying to think up a good url name |
| 20:57.44 | SpaceBass | lol @ Katty |
| 20:57.48 | Hmmhesays | voipfister isn't going to cut it |
| 20:58.00 | SpaceBass | Hmmhesays, isn't drinking great? i mean I really love it....too much |
| 20:58.10 | SpaceBass | voipfister! THAT IS AWESOME |
| 20:58.28 | Hmmhesays | lol |
| 20:58.34 | Hmmhesays | I enjoy teh booze |
| 20:58.44 | SpaceBass | "Fingerbang your asterisk into the future with Voipfister" ... there's your catch phrase |
| 20:59.00 | Hmmhesays | LOL |
| 20:59.06 | Hmmhesays | wow, just, wow |
| 20:59.09 | tzanger | SpaceBass: oh dear |
| 20:59.20 | afrosheen | bahahah |
| 20:59.21 | SpaceBass | too graphic? sorry...was thinking about that southpark episode |
| 20:59.29 | gbodemantv | Qwell: any tips on the realtime queues needing a reload |
| 20:59.33 | Hmmhesays | lol, no that was awesome |
| 21:00.08 | *** join/#asterisk skyboy (n=skyboy@72.18.13.34) |
| 21:00.18 | skyboy | hello. Ist possible to seperate the signaling portion and rtp streams (ie media streams) on two seperate interfaces in Asterisk? |
| 21:01.08 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-92.apple.com) |
| 21:03.54 | afrosheen | skyboy: what for |
| 21:03.57 | De_Mon | is there an ODBC version of the MYSQL command? |
| 21:04.21 | docelm0 | Anyone with an idea on the RTP question? How can I tell if asterisk is in the RTP stream? |
| 21:04.25 | rpm | Qwell[]: how could i rewrite that $[ "${CALLERIDNAME}" : "Autowakeup" ]? I have a external application which creates a call with the CID of Autowakeup |
| 21:04.32 | mattwj2005 | so what is the difference between a 2.4 kernel or a 2.6 kernel in regards to meetme installation ease? |
| 21:04.47 | docelm0 | Cause it says its trying to bridge.... |
| 21:04.58 | mattwj2005 | does 2.6 have the built in modules? |
| 21:05.04 | generalhan | can some one please help me out with my TDM40B issue ? if you look at the pastebin it looks like everything is set up ok on the ztcfg -vvvvvvv but still i get the errors at the bottom of the screen |
| 21:05.04 | generalhan | http://generalhan.pastebin.ca/48568 |
| 21:05.19 | afrosheen | generalhan: did you haxor that power for tdm yet? |
| 21:05.35 | SpaceBass | i was going to tackle doing user based outbound routes today....time just got away |
| 21:05.36 | generalhan | yes |
| 21:05.45 | generalhan | afrosheen: and now im having this problem ... http://generalhan.pastebin.ca/48568 |
| 21:06.16 | generalhan | it wont load the stupid FXS ports for my fax machines |
| 21:06.48 | afrosheen | generalhan: so it's lighting the ports? |
| 21:06.52 | generalhan | yep |
| 21:07.10 | justinu|laptop | docelm0: type "rtp debug" |
| 21:07.15 | generalhan | all ports are lit up, the ztcfg -vvvvv shows it all good, zap show status in the CLI shows it all good yet nothing |
| 21:07.17 | skyboy | afrosheen: I have an intermediary server that the signaling porition needs to talk to and be dedicated. the "media" portion will go elsewhere.. :) |
| 21:08.02 | docelm0 | damnit.. |
| 21:08.07 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 21:08.18 | docelm0 | Well screw that idea.. Its in.. How do I force it to stay out of the stream? |
| 21:08.45 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 21:08.48 | afrosheen | skyboy: oh, I thought you were dealing with double nat or something |
| 21:09.04 | docelm0 | I need asterisk to act like SER basically.. I dont know SER very well so I am using asterisk for this.. I have canreinvite=yes turned on.. |
| 21:09.28 | justinu|laptop | i'm sure you know that asterisk isn't a SIP Proxy |
| 21:09.32 | justinu|laptop | it's a B2BUA |
| 21:09.52 | docelm0 | I know.. Im trying to make it one.. |
| 21:09.58 | Katty | docelm0: file's all about SER |
| 21:10.06 | docelm0 | ohh FILE! |
| 21:10.10 | justinu|laptop | it's like trying to force a square peg into a round hole |
| 21:10.11 | Katty | he's SERific |
| 21:10.16 | justinu|laptop | you're best served reading up on SER |
| 21:10.19 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 21:10.19 | file | what |
| 21:10.27 | docelm0 | I cant find any decient documentation |
| 21:10.37 | justinu|laptop | RFC3261 is a good start |
| 21:10.38 | Katty | file: your brain is licky. |
| 21:10.40 | file | Asterisk can't act as SER. |
| 21:10.45 | file | SER is a SIP proxy |
| 21:10.47 | docelm0 | I know this |
| 21:10.47 | file | Asterisk is a B2BUA |
| 21:10.52 | justinu|laptop | echo... echo |
| 21:10.53 | justinu|laptop | echo |
| 21:11.00 | docelm0 | ECHO ECHO! |
| 21:11.01 | Nugget | you can make asterisk behave as a proxy, but it isn't fun and I don't recommend it. |
| 21:11.13 | file | I did something to make it sort of act as a proxy... |
| 21:11.18 | file | passthru SDP, really evil |
| 21:11.22 | justinu|laptop | heh |
| 21:11.47 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 21:12.59 | justinu|laptop | encrypt |
| 21:13.04 | *** part/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 21:13.19 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 21:13.29 | *** part/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 21:13.46 | Samoied | anyone recommands a news site for asterisk? |
| 21:14.15 | mattwj2005 | 2.4 or 2.6 better for asterisk? |
| 21:14.16 | Katty | justinu|laptop: is you /is/, or is you /ain't/ bitiful |
| 21:14.19 | file | Samoied: Asterisk Daily News, http://www.sineapps.com/news.php |
| 21:14.44 | Samoied | file: thanks |
| 21:14.52 | Katty | how is asterisk like algebra? (joke) |
| 21:15.01 | De_Mon | whats the deal with app_odbcexec? does it work in 1.2.6? |
| 21:15.24 | Katty | think about it! |
| 21:15.27 | gbodemantv | anybody know about realtime queues |
| 21:15.28 | Katty | it's /real/ easy if you're good at math! |
| 21:15.29 | mattwj2005 | you have to put 2 and 2 together? :P |
| 21:15.29 | Katty | and asterisk |
| 21:15.33 | Katty | no |
| 21:15.40 | skyboy | asterisk=algebra because you stick a negative under a square root and its all imaginary....with no support. |
| 21:15.41 | Katty | they both use the FOIL method |
| 21:15.43 | gbodemantv | and why I have to reload after a change? |
| 21:18.13 | *** join/#asterisk Jon335 (n=Jon335@unaffiliated/jon335) |
| 21:18.24 | gbodemantv | so very confused |
| 21:19.08 | op3r | how do you access svn on a centos? |
| 21:19.38 | Hmmhesays | haha |
| 21:20.15 | ljam | op3r: yum install subversion |
| 21:23.09 | gbodemantv | qwell: Can you help me |
| 21:23.12 | Katty | no one got my well thought out little pun |
| 21:23.42 | rene- | in spanish a pun can be funny but not always |
| 21:24.40 | rene- | a pun is the written sound of air leaving the buttocks |
| 21:25.07 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 21:25.42 | *** part/#asterisk mattwj2005 (n=Matt@user-12l3lqm.cable.mindspring.com) |
| 21:27.11 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool149-149.nas31.salt-lake-city1.ut.us.da.qwest.net) |
| 21:29.27 | *** join/#asterisk mattwj2005 (n=Matt@user-12l3lqm.cable.mindspring.com) |
| 21:29.43 | *** part/#asterisk mattwj2005 (n=Matt@user-12l3lqm.cable.mindspring.com) |
| 21:30.46 | gbodemantv | has anyone had this issue? |
| 21:32.53 | rene- | is not a big deal gbodemantv: just issue a system call for asterisk -rtx'reload' whenever you do any change, is what other commercial gui vendors for asterisk do (e.g. Signate) |
| 21:36.06 | afrosheen | gbodemantv: that's how * works..reload the config files after changes |
| 21:37.49 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 21:38.22 | *** join/#asterisk tdonahue-laptop (n=tdonahue@231.212.118.66.brainstorminternet.net) |
| 21:38.28 | brodiem | What's a better status viewer than FOP? |
| 21:38.36 | brodiem | terrible bugs... |
| 21:38.56 | Katty | a human |
| 21:38.57 | tdonahue-laptop | brodiem, i use the panel from asteriskgurus.com |
| 21:39.28 | brodiem | tdonahue-laptop, I get a godaddy parked page at that domain |
| 21:39.29 | Katty | fop does like to deadlock :< |
| 21:40.00 | brodiem | Katty, it keeps reporting extensions still on the phone after they hangup and never fixes itself unless you kill op_server, at which point it loses all agent login, DND status, etc |
| 21:40.11 | afrosheen | brodiem: have you kept up with FOP? |
| 21:40.16 | *** join/#asterisk nite (n=nite@gateway.digium.com) |
| 21:40.19 | brodiem | yep |
| 21:40.25 | brodiem | .25 |
| 21:40.44 | afrosheen | I follow the FOP list and you should probably post to it if you're having issues |
| 21:40.57 | tdonahue-laptop | brodiem, sorry, http://www.asteriskguru.com/tools/switchboard.php |
| 21:42.32 | brodiem | tdonahue-laptop, does it show status by Agent/XXXX instead of just SIP/XXXX? |
| 21:42.53 | tdonahue-laptop | umm... i think there is a pane for the Agent status |
| 21:43.01 | tdonahue-laptop | on the right hand side |
| 21:43.48 | tdonahue-laptop | has anyone had any issues getting Playback and Voicemail actually play to the phones? |
| 21:43.56 | tdonahue-laptop | i'm having trouble on a new 1.2 install |
| 21:43.59 | afrosheen | tdonahue-laptop: YES! |
| 21:44.17 | afrosheen | tdonahue-laptop: since 1.2.5 we've had weirdness with voicemail, thought I was alone on this |
| 21:44.28 | tdonahue-laptop | afrosheen, have you found any resolution? |
| 21:44.41 | afrosheen | our phones are showing the voicemail indicator before the vm is even finished recording |
| 21:44.41 | key2 | !seen kram |
| 21:44.47 | shaun222 | bahh, anybody have a gmail invite they can shoot me.. |
| 21:44.51 | justinu | yes |
| 21:44.53 | afrosheen | shaun222: yeah |
| 21:45.02 | Yasheee | no, I'm out of invites |
| 21:45.11 | key2 | I do |
| 21:45.15 | Yasheee | you only get what.. 100/day? |
| 21:45.20 | justinu | heh |
| 21:45.27 | shaun222 | afrosheen: just pm'd you my email |
| 21:45.49 | shaun222 | thanks |
| 21:45.54 | afrosheen | welcome |
| 21:46.09 | justinu | is afrosheen anything like soul glow? |
| 21:46.13 | Hmmhesays | yay my form emails me now |
| 21:46.17 | Hmmhesays | schweet |
| 21:46.49 | afrosheen | see Coming To America for a reference |
| 21:47.15 | justinu | i remember soul glow from coming to america |
| 21:47.18 | justinu | but not afrosheen |
| 21:47.39 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 21:48.24 | tdonahue-laptop | no one has any problems with Playback not actually playing the audio? |
| 21:49.29 | gbodemantv | rene: it seems that the realtime queues aren't reallt "realtime" if I have to reloa |
| 21:49.37 | Strom_C | tdonahue-laptop, what format is the file in? |
| 21:49.37 | gbodemantv | is there any other way around this? |
| 21:49.40 | afrosheen | afrosheen is an analog to the equally fictional soul glow |
| 21:49.58 | afrosheen | tdonahue-laptop: yes, that's what mine does sometimes |
| 21:50.08 | tdonahue-laptop | whatever ships with asterisk and asterisk-sounds |
| 21:50.09 | afrosheen | it doesn't play the caller id, timestamp or anything |
| 21:50.29 | afrosheen | just skips the message right? |
| 21:50.45 | tdonahue-laptop | they are just the builtin files |
| 21:50.50 | justinu | ok, just making sure I understand the reference |
| 21:51.00 | tdonahue-laptop | afrosheen, i'm not even getting the "Login" sound |
| 21:51.06 | tdonahue-laptop | errr... prompt |
| 21:51.16 | *** join/#asterisk inv_arp[work] (i=junya@adsl-11-225-195.mia.bellsouth.net) |
| 21:51.23 | afrosheen | tdonahue-laptop: from *98 or what |
| 21:51.33 | afrosheen | *98+ext should get you in |
| 21:51.44 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 21:52.10 | tdonahue-laptop | VoicemailMain does work through, if I just type my extension in it works, so the server is getting my button presses |
| 21:52.19 | tdonahue-laptop | i just don't hear anything coming from the server |
| 21:52.35 | *** part/#asterisk serverline (n=serverli@p54BA4551.dip0.t-ipconnect.de) |
| 21:52.40 | afrosheen | one way audio maybe? |
| 21:52.44 | afrosheen | nat involved or no |
| 21:52.57 | tdonahue-laptop | yes, but regular calls work fine |
| 21:53.06 | afrosheen | interesting |
| 21:53.18 | tdonahue-laptop | extension to extension with canreinvite=no and trunked calls to our providor |
| 21:53.25 | [av]bani | it's soul glo |
| 21:53.43 | afrosheen | what happens if you tail -f /var/log/asterisk/full while you reproduce the issue |
| 21:53.55 | afrosheen | gj [av]bani |
| 21:53.59 | Hmmhesays | friday friday friday, time for the happy fun show to start |
| 21:54.01 | Hmmhesays | weeeeeeeeeee |
| 21:54.17 | justinu | [av]bani: fuck you, fuck you and fuck you too! |
| 21:54.26 | [av]bani | http://letyoursoulglo.ytmnd.com/ |
| 21:54.41 | Hmmhesays | what the hell is up with noky, has anyone else been spammed by him? |
| 21:54.44 | justinu | yeah |
| 21:54.56 | justinu | after I told him to stop PMing me unless he's paying, it stopped |
| 21:55.15 | *** join/#asterisk voip_pd2200 (n=voip_pd2@202.59.90.180) |
| 21:55.15 | Hmmhesays | yeah |
| 21:55.18 | voip_pd2200 | hi |
| 21:55.19 | [av]bani | :D |
| 21:55.38 | Hmmhesays | i should register voipfister |
| 21:55.58 | Strom_C | hahahahahaha |
| 21:56.05 | tdonahue-laptop | is there a pastbot i should use? |
| 21:56.10 | Strom_C | i wanted to register voipfuck but someone took it |
| 21:56.11 | tdonahue-laptop | errr.. pastebot |
| 21:56.12 | Strom_C | ~pb |
| 21:56.15 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 21:56.15 | voip_pd2200 | can anybody help in setting up astguiclient |
| 21:57.11 | Hmmhesays | maybe is helpidontknow.org too long? |
| 21:57.22 | Strom_C | i dont thinknk so |
| 21:57.22 | voip_pd2200 | what is this side hmm? |
| 21:57.37 | Hmmhesays | um, left? |
| 21:58.05 | tdonahue-laptop | my output from /var/log/asterisk/full is at http://pastebin.com/646927 |
| 21:58.06 | voip_pd2200 | i m selling a pd for $100 , anybody need it? |
| 21:58.14 | tdonahue-laptop | i don't see anything out of the ordinary though |
| 21:58.23 | Hmmhesays | ok, time to commence the friday drunk |
| 21:58.26 | Hmmhesays | i'm out peeps |
| 21:58.30 | voip_pd2200 | ok |
| 21:58.38 | voip_pd2200 | you can get out hmmm |
| 21:58.40 | afrosheen | is orly.com registered yet |
| 21:58.48 | voip_pd2200 | i can tell you |
| 21:58.49 | tdonahue-laptop | later Hmmhesays |
| 21:58.49 | afrosheen | or yarly |
| 21:58.51 | voip_pd2200 | also i can register domains |
| 21:59.11 | *** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 21:59.43 | voip_pd2200 | does anybody have customized astguiclient Predictive Dialer? |
| 21:59.50 | Hmmhesays | not this guy again |
| 21:59.56 | Hmmhesays | didn't you get kicked yesterday? |
| 22:00.11 | voip_pd2200 | how many times you can kick me? |
| 22:00.15 | voip_pd2200 | will you block all ips of this world? |
| 22:00.25 | Hmmhesays | ok i'm really out of here now |
| 22:00.28 | voip_pd2200 | i m serious so you should be serious too |
| 22:00.40 | afrosheen | lol foreign trolls |
| 22:00.50 | voip_pd2200 | also i have fake ip identity software |
| 22:00.56 | voip_pd2200 | so plz beaware next time kicking me out |
| 22:00.59 | voip_pd2200 | i m here to have help |
| 22:01.04 | voip_pd2200 | understand? |
| 22:01.10 | tdonahue-laptop | on one has any ideas what could be causing my one way audio with Playback? |
| 22:01.11 | afrosheen | yeah but nobody owes that to you though |
| 22:01.21 | afrosheen | this isn't paid support |
| 22:01.30 | nick125 | that doesn't mean anyone is obligated to give you help.. |
| 22:01.30 | voip_pd2200 | well |
| 22:01.32 | voip_pd2200 | i know |
| 22:01.37 | voip_pd2200 | if you not i not need help |
| 22:01.39 | afrosheen | so don't come in here with that attitude |
| 22:01.46 | voip_pd2200 | mr afrosheen |
| 22:01.54 | voip_pd2200 | tell me any domain you have and i can have it rejected |
| 22:01.55 | voip_pd2200 | lol |
| 22:01.57 | voip_pd2200 | i m registar |
| 22:02.02 | voip_pd2200 | so beaware from me |
| 22:02.05 | voip_pd2200 | i am VOIP carrier |
| 22:02.06 | voip_pd2200 | understand? |
| 22:02.10 | nick125 | don't threaten us. |
| 22:02.11 | afrosheen | do I care? no. |
| 22:02.15 | tdonahue-laptop | voip_pd2200, congratulations, i'm a voip carrier too |
| 22:02.15 | voip_pd2200 | lol |
| 22:02.18 | voip_pd2200 | i m not threating |
| 22:02.19 | voip_pd2200 | nice |
| 22:02.22 | voip_pd2200 | nice to meet you laptop |
| 22:02.29 | voip_pd2200 | see i m not here for a war |
| 22:02.29 | *** mode/#asterisk [+b *!n=voip_pd2@202.59.90.*] by file |
| 22:02.30 | *** kick/#asterisk [voip_pd2200!n=jcolp@mctnnbsa24w-142167060049.pppoe-dynamic.nb.aliant.net] by file (file) |
| 22:02.47 | file | You may now return to your regular conversations. |
| 22:02.48 | afrosheen | yay |
| 22:02.49 | Strom_C | <3 |
| 22:02.52 | *** join/#asterisk NewSole (n=dave@d226-108-46.home.cgocable.net) |
| 22:03.17 | NewSole | can someone please let me know why the h232 crashes |
| 22:03.35 | kink0 | NewSole, that is what I would to know too |
| 22:04.13 | kink0 | I never had crashes at asterisk, until add h323, then crashes with too many open files, even I set a very high value for ulimit -n |
| 22:05.53 | justinu | hey, that voip_pd2200 guy was entertaining |
| 22:05.55 | justinu | understand? |
| 22:07.20 | gbodemantv | is there a realtime channel? |
| 22:07.33 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 22:07.56 | justinu | kink0: so there's a file descriptor leak somewhere in your h323 module |
| 22:08.12 | X-Rob | that's impressive. |
| 22:08.18 | X-Rob | file descriptor leaks are hard to do 8) |
| 22:08.19 | afrosheen | lol @ justinu |
| 22:08.22 | justinu | heh |
| 22:08.32 | afrosheen | I hold a button for EMP bomb, understand? |
| 22:08.43 | justinu | afrosheen: do I care? no. |
| 22:08.47 | justinu | ;) |
| 22:08.54 | afrosheen | lol |
| 22:08.57 | kink0 | justini I used as documentations recomends, and at least I have not over two crashes a day |
| 22:09.15 | nick125 | I was soo scared..he is a VOIP Carrier!!! OMG, soo much fear!! |
| 22:09.40 | justinu | kink0: what changes did you make? maybe it'll help NewSole |
| 22:10.07 | brodiem | tdonahue-laptop, just checked out that switchboard app...know of any others? It doesn't seem to be geared for display in a call centre environment |
| 22:10.27 | kink0 | justinu:NewSolo : I just compiled/install openh323/pwlib, and compile h323 in asterisk. Then configure the h323.conf and start asterisk, that is all. |
| 22:10.30 | *** join/#asterisk kavit (n=kavit@210.10.216.105) |
| 22:10.58 | kink0 | NewSole, your asterisk crashes with any error ? when calls ? without calls in progress ? what is your scenary ? |
| 22:11.09 | justinu | scenario, kink0 |
| 22:11.14 | justinu | not scenary |
| 22:11.14 | kink0 | mine crashes when a lot ( about 1 per second ) is entering. |
| 22:11.25 | NewSole | ok this is the funny part..... |
| 22:11.48 | kink0 | justinu: sorry... english is not my mother language, I request from you some understanding for my mistakes :) |
| 22:11.57 | justinu | i understand, i'm just giving you a tip |
| 22:12.11 | *** join/#asterisk Igg-man (n=kc0itq@rrcs-67-53-20-210.west.biz.rr.com) |
| 22:12.12 | brodiem | Anyone know what's a better status viewer than FOP? |
| 22:12.13 | NewSole | if running using "asterisk -cgvvvr" it crashes |
| 22:12.16 | NewSole | but |
| 22:12.24 | NewSole | if running using "asterisk -cgvvv" it does not |
| 22:12.32 | justinu | theres no point in running -cr |
| 22:12.45 | kink0 | hmmmmm -c and -r ?? |
| 22:13.41 | kink0 | I only use -cvvv ... not -r, well only -r when I need to use a yet running asterisk instance. |
| 22:13.52 | justinu | yes, that is correct |
| 22:14.07 | NewSole | nope just crashed |
| 22:14.20 | NewSole | using asterisk -cgvv |
| 22:14.27 | kink0 | NewSole, and if you remove h323 support then does not crashs ? |
| 22:14.39 | NewSole | nope |
| 22:15.04 | kink0 | NewSole, why -c and r at same call ? |
| 22:15.09 | NewSole | but our provider stop using sip and only accepts h323 |
| 22:15.32 | kink0 | NewSole, yes, was just for testing if was a h323 issue |
| 22:15.59 | *** join/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br) |
| 22:16.10 | _Paulo_ | hi |
| 22:16.22 | NewSole | we were running fine today... but was forced to use h323 today |
| 22:17.16 | gbodemantv | I can't really find any more info on Realtime queues and reload |
| 22:17.28 | _Paulo_ | I got a grandstream 286 |
| 22:17.37 | kink0 | NewSole, my experience is since I added h323 support, when I got a lot of h323 calls, then crashes, but can be up for many hours, even days, if does not arrives a lot of fast h323 connections |
| 22:17.46 | gbodemantv | is there any way I can add queue memberes and not do a reload |
| 22:17.46 | X-Rob | NewSole, use a different provider. |
| 22:17.54 | _Paulo_ | what should I put at the autodial field in the setup? |
| 22:18.13 | gbodemantv | just seems like it is the same as adding them to queues.conf |
| 22:18.24 | kink0 | X-Rob, do you think is a h323 related problem ? |
| 22:18.25 | NewSole | ya kind of hard.... we have a large ballance on current provider.... and no refunds |
| 22:18.45 | X-Rob | kink0, Ooh, I don't know. I just know that H323 gives you CANCER. |
| 22:18.54 | kink0 | NewSole, in my case, is hard, is not a provider, is a client who sends me h323 |
| 22:19.25 | NewSole | kink0 |
| 22:19.33 | NewSole | do you have realtime installed |
| 22:19.46 | kink0 | and I was thinking about to install some h323->SIP gateway between my client and me |
| 22:19.52 | X-Rob | NewSole, of course you can get a refund. It was working, they stopped it working. |
| 22:20.25 | kink0 | NewSole, yes. |
| 22:20.32 | NewSole | ya but lets get a refund friday night.... and find a new provider over week end |
| 22:20.34 | drray | I think a letter from lawyer would get you a refund |
| 22:20.49 | X-Rob | NewSole, that's not hard. There's thousands of providers. |
| 22:21.12 | X-Rob | Oh, if only voip_pd2200 was here. HE's a voip provider! |
| 22:21.15 | NewSole | ok... thnx kink0... I found problem and now to get solution |
| 22:21.22 | kink0 | NewSole, how much calls are you sending per second ? |
| 22:21.32 | NewSole | 3-400 at once |
| 22:22.07 | kink0 | and crashes even you send less than lets say... 10 calls per second ? |
| 22:22.28 | NewSole | yup but not all are h323 |
| 22:22.42 | gbodemantv | newsole: yeah realtime installed |
| 22:23.00 | gbodemantv | newsole: all users and phones in realtime |
| 22:23.20 | gbodemantv | just enabled queues and it works, just needing to do a reload after i add queue memebers |
| 22:23.26 | NewSole | both times it crashed..... I did a bittrace on core's |
| 22:23.32 | gbodemantv | which seems like I should not have to do it |
| 22:23.35 | justinu|laptop | yeah, let voip_pd2200 back in here to straighten these guys out |
| 22:23.40 | justinu|laptop | understand? |
| 22:23.41 | NewSole | and crashed on realtime querery |
| 22:24.38 | kink0 | but realtime queryring is near the same for h323 and SIP, right ? |
| 22:26.31 | NewSole | not sure... but it crash point on both times are in realtime |
| 22:26.44 | NewSole | has to be somthing in the module |
| 22:27.42 | kink0 | are you ussing the module compiled from asterisk/h323 ? or third part module ? |
| 22:28.00 | kink0 | I am ussing from asterisk , from v 1.2.5 |
| 22:28.11 | _Paulo_ | someone can help me with a grandstream 286? |
| 22:28.24 | _Paulo_ | Im not familiar with SIP |
| 22:28.28 | NewSole | asterisk/h323 |
| 22:28.35 | justinu|laptop | _Paulo_: what's up? |
| 22:29.57 | kink0 | NewSole, can I private you to paste my chan_h323 module depencendy and compare with yours ? |
| 22:30.15 | kink0 | or use pastebin ... is just about 15 lines |
| 22:30.30 | ketanp | can anyone get to www.voxilla.com? |
| 22:31.18 | NewSole | yup kink |
| 22:32.04 | _Paulo_ | justinu, hi |
| 22:32.26 | _Paulo_ | i got the phone registered. |
| 22:32.44 | *** part/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
| 22:32.51 | *** join/#asterisk Hyper_Eye (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 22:32.54 | _Paulo_ | but i cant place a call. |
| 22:33.12 | gbodemantv | so frustrated |
| 22:33.27 | gbodemantv | keep on having to do a reload |
| 22:33.42 | _Paulo_ | i put "1" at the "Offhook Auto-Dial" field |
| 22:33.46 | gbodemantv | I am trying to set it up so that managers can modify their own queues through a web interface |
| 22:33.57 | gbodemantv | but if I have to do a reload every time... |
| 22:34.03 | tdonahue-laptop | well, i've verified my audio issue with both a Polycom phone and outside the firewall, so its got to be something to do with 1.2.4 |
| 22:34.07 | tdonahue-laptop | err.. 1.2.6 |
| 22:34.57 | _Paulo_ | i turned sip debug on |
| 22:35.05 | *** part/#asterisk Hyper_Eye (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 22:35.11 | *** join/#asterisk Derkommissar (n=Alberto@adsl-153-47-91.mia.bellsouth.net) |
| 22:35.50 | _Paulo_ | I see To: <sip:1@bridge-tic.braslink.com>;tag=as77f15d6a |
| 22:35.54 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 22:36.03 | _Paulo_ | among debug messages. |
| 22:36.09 | _Paulo_ | is this right? |
| 22:38.05 | tdonahue-laptop | _Paulo_, 1 is probably what the Grandstream seems to be dialing |
| 22:38.17 | tdonahue-laptop | try putting the number that you want to have dialed |
| 22:38.28 | _Paulo_ | tdonahue-laptop, I have a context for this user |
| 22:39.18 | _Paulo_ | tdonahue-laptop, why extension 1 is not ringing? |
| 22:39.18 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 22:39.18 | tdonahue-laptop | _Paulo_, what are you trying to do? are you trying to have it dial a number as soon as you pick the phone up |
| 22:39.20 | tdonahue-laptop | ? |
| 22:39.44 | _Paulo_ | tdonahue-laptop, I have to setup this ata to my boss |
| 22:40.13 | _Paulo_ | tdonahue-laptop, he would like to pick up the phone and get on our pbx. |
| 22:40.21 | TedC | how stable is trunk these days? |
| 22:40.29 | *** join/#asterisk Derkommissar (n=Alberto@adsl-153-47-91.mia.bellsouth.net) |
| 22:40.44 | TedC | I need t.38, and I was having transfer problems with the 1.2 t.38 patch/ |
| 22:41.18 | _Paulo_ | tdonahue-laptop, I have this on my boss context: |
| 22:41.24 | *** join/#asterisk NewSole (n=dave@d226-108-46.home.cgocable.net) |
| 22:41.24 | _Paulo_ | tdonahue-laptop, exten => _X.,n,DISA(no-password|local) |
| 22:41.43 | tainted- | anyone dealt with ISPs blocking/throttling VoIP? |
| 22:41.45 | justinu|laptop | if he's got an ATA, why would he need DISA? |
| 22:42.47 | _Paulo_ | justinu|laptop, Im lost here. |
| 22:42.57 | justinu|laptop | ok, just set up a normal entry in sip.conf for him |
| 22:43.07 | justinu|laptop | put him in the same context you have your other PBX users in |
| 22:43.14 | justinu|laptop | it's very simple |
| 22:43.26 | _Paulo_ | justinu|laptop, dont use * as a PBX |
| 22:43.34 | _Paulo_ | justinu|laptop, I use it as a fax server |
| 22:43.48 | _Paulo_ | justinu|laptop, but my boss travels a lot |
| 22:43.50 | justinu|laptop | so what is your PBX then? |
| 22:44.14 | _Paulo_ | justinu|laptop, and long distance calls are still very expensive in Brazil. |
| 22:44.19 | justinu|laptop | i understand |
| 22:44.40 | Derkommissar | Anyone here lives in Miami fla ? |
| 22:44.54 | _Paulo_ | my boss lives |
| 22:45.00 | Derkommissar | Cool |
| 22:45.06 | key2 | what would be the cheapest SIP provider for middle east call ? Does someone know ? |
| 22:45.55 | Derkommissar | but im looking for someone to consult on an asterisk instalation |
| 22:46.00 | Derkommissar | i dont have the time for it |
| 22:46.00 | _Paulo_ | justinu|laptop, he wants to plug the ata anyware and place calls to Sao Paulo City as a local call. |
| 22:46.10 | *** part/#asterisk chino[server] (n=daquino@e82-103-128-114s.easyspeedy.com) |
| 22:46.22 | justinu|laptop | _Paulo_: that's fine... how does your asterisk machine makes calls in Sao Paulo? |
| 22:46.45 | Derkommissar | _Paulo_ a couple of lines and a nice digium card would allow you to do that |
| 22:47.27 | afrosheen | paulo: with the unpredictable nature of 'mystery networks' that he'd plug the ata into, I'd go with an iAxy |
| 22:47.34 | _Paulo_ | Derkommissar, I have 2 E1s on this box, 60 channels, 550 DIDs |
| 22:47.41 | Derkommissar | Cool |
| 22:47.47 | Derkommissar | WOW that is nice |
| 22:47.50 | justinu|laptop | so put him in whatever context those E1s are in |
| 22:47.57 | Derkommissar | can i buy a couple of DID's from you? |
| 22:48.04 | _Paulo_ | Derkommissar, But they are being used for faxing. |
| 22:48.10 | Derkommissar | or just use the command GOTO |
| 22:48.19 | justinu|laptop | yeah, lots of different ways |
| 22:48.43 | _Paulo_ | Why Im not geting the DISA tone? |
| 22:48.47 | afrosheen | iax is good at cutting through nat, firewalls, etc |
| 22:48.55 | *** join/#asterisk nagl (n=nagl@86.59.54.238) |
| 22:49.08 | Derkommissar | 55.,1,GOTO(whereverthet1'sareat,1) |
| 22:49.11 | _Paulo_ | can I paste my sip debug? |
| 22:49.24 | Derkommissar | paste it in private on me |
| 22:49.28 | Derkommissar | or in pastebin.com |
| 22:50.02 | _Paulo_ | thanks! |
| 22:53.24 | *** join/#asterisk terrapen (n=cjs@166.70.183.108) |
| 22:53.36 | tainted- | are there any ATAs that support VPNs? |
| 22:53.44 | tainted- | ATAs or IP phones |
| 22:58.55 | afrosheen | doubtful |
| 22:59.07 | afrosheen | although cisco IP phones may have some kind of vpn support |
| 22:59.21 | afrosheen | vpn = overhead and overhead is bad for voip streams |
| 22:59.26 | tainted- | well |
| 22:59.37 | tainted- | there were some articles that said vpns actually help voip |
| 22:59.52 | tainted- | although ur right.. it does generate more overhead |
| 23:00.28 | tainted- | but it might become a necessity if isps start blocking voip |
| 23:01.32 | afrosheen | ssip is coming soon |
| 23:01.37 | afrosheen | that should help a little |
| 23:01.46 | tainted- | isn't there srtp already |
| 23:02.40 | afrosheen | no idea |
| 23:02.42 | tainted- | http://www.networkworld.com/reviews/2006/022006-ssl-voip-test.html |
| 23:02.45 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-92.apple.com) |
| 23:03.04 | tdonahue-laptop | anyone have any ideas why i am getting the one way audio from Playback and VoicemailMain? This happens both behind and not behind nat |
| 23:03.08 | *** join/#asterisk Denmark (n=fake@62.242.24.182) |
| 23:03.59 | *** part/#asterisk tdonahue-laptop (n=tdonahue@231.212.118.66.brainstorminternet.net) |
| 23:04.08 | *** join/#asterisk tdonahue-laptop (n=tdonahue@231.212.118.66.brainstorminternet.net) |
| 23:04.48 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-92.apple.com) |
| 23:06.59 | jeffgus | hi all |
| 23:07.08 | jeffgus | before i email the asterisk-user mailing list |
| 23:07.12 | jeffgus | i thought i'd ask in here |
| 23:07.28 | jeffgus | does anyone have suggests for building an asterisk box with 100 analog ports? |
| 23:07.30 | Denmark | Hi jeffgus |
| 23:07.47 | jeffgus | i thought just 5 channel banks would be fine |
| 23:08.11 | jeffgus | but i want to make sure that having 2 T1 boards will work fine |
| 23:08.17 | jeffgus | with timing and all |
| 23:09.26 | Derkommissar | that would work |
| 23:09.31 | Derkommissar | get good hardware |
| 23:09.34 | Derkommissar | and lots of ram |
| 23:09.43 | jeffgus | 4gig's of ram |
| 23:09.50 | jeffgus | would that be enough? |
| 23:10.06 | Derkommissar | yea sounds bigger than my first hardrive |
| 23:10.10 | Derkommissar | LOL |
| 23:10.13 | jeffgus | heh |
| 23:10.15 | justinu|laptop | 2 quad span boards? |
| 23:10.22 | tdonahue-laptop | jeffgus, we have 2 4-port cards with as many as 8 channelbanks on a couple servers |
| 23:10.22 | Derkommissar | Maybe Dual XEON |
| 23:10.23 | jeffgus | yeah |
| 23:10.26 | Denmark | my first harddrive was 40MB, I guess.. |
| 23:10.37 | justinu|laptop | 40 meg here also |
| 23:10.43 | justinu|laptop | seagate MFM drive |
| 23:10.48 | jeffgus | tdonahue-laptop, how is the performance with regards to call load? |
| 23:10.49 | Derkommissar | same as me |
| 23:10.58 | justinu|laptop | DOS 3.2 couldn't format more than 32meg at a time either |
| 23:11.02 | Denmark | justinu: RLL Conner |
| 23:11.28 | justinu|laptop | i actually re low-level formatted that drive later as an RLL |
| 23:11.32 | justinu|laptop | to get more space :) |
| 23:11.33 | tdonahue-laptop | they aren't heavilly used, but it doesn't seem to be bad |
| 23:11.51 | Denmark | justnulling2: Mine was IDE |
| 23:11.54 | jeffgus | the quad port card will bridge calls between ports, correct? |
| 23:12.03 | jeffgus | but not bridge between cards |
| 23:12.07 | jeffgus | correct? |
| 23:12.09 | Denmark | justnulling2: I missed the lowlevel fun. ;-) |
| 23:12.16 | justinu|laptop | jeffgus: can do both |
| 23:12.49 | jeffgus | justinu|laptop, oh! it can? so it should have low impact then |
| 23:13.25 | justinu|laptop | well, i dunno about the performance impact of switching a call across two different cards vs. both calls being on the same card |
| 23:13.42 | jeffgus | justinu|laptop, ah, ok |
| 23:13.58 | justinu|laptop | did someone indicate that performance would be better in one scenario vs. another? |
| 23:14.26 | gbodemantv | is there a way to just reload queues |
| 23:14.27 | jeffgus | justinu|laptop, nope, it just seemed logical that a quad port card could do that, but i wasn't sure about between cards |
| 23:14.31 | gbodemantv | not do a full asterisk reload |
| 23:14.33 | gbodemantv | ??? |
| 23:14.50 | jeffgus | it seems to me that the overall impact should be low for the system |
| 23:14.57 | jeffgus | no codec conversion |
| 23:15.01 | justinu|laptop | jeffgus: you might ask digium about this... but i would think it would be the same, regardless |
| 23:15.04 | justinu|laptop | due to the way zaptel works |
| 23:15.09 | Derkommissar | i have around 4 e1's of traffic with 8,000 sip terminals on a dual xeon PC. the pc cpu and memory usage doesnt go beond 50% ever |
| 23:15.11 | jeffgus | and outgoing calls will go over a pri link to the PSTN |
| 23:15.23 | Derkommissar | but i dont use any cards in that server |
| 23:15.48 | jeffgus | Derkommissar, no cards? how do you connect the e1's into the server then? |
| 23:16.17 | Derkommissar | i have an cisco as5400 to terminate to the PSTN |
| 23:16.18 | jeffgus | Derkommissar, how much RAM is in that machine? |
| 23:16.27 | jeffgus | Derkommissar, ah! okay |
| 23:16.28 | Derkommissar | or i terminate on a voip provider |
| 23:16.33 | Derkommissar | 2 gigs |
| 23:16.49 | jeffgus | ok |
| 23:16.51 | jeffgus | sounds good then |
| 23:16.56 | Derkommissar | one sec |
| 23:17.03 | jeffgus | this is a high rise condo building |
| 23:17.05 | gbodemantv | what does a -rtx reload accomplish |
| 23:17.06 | gbodemantv | ? |
| 23:17.19 | Derkommissar | 8465 sip peers [1455 online , 7010 offline] |
| 23:17.32 | jeffgus | over all call load should be low since it's mainly people calling the main desk, hotel style |
| 23:17.41 | Derkommissar | not much traffic rigth now |
| 23:17.45 | Derkommissar | 107 active calls |
| 23:18.01 | jeffgus | 107 call and what type of load? |
| 23:18.06 | Derkommissar | Mem: 1034364k total, 896716k used, 137648k free, 129396k buffers |
| 23:18.09 | jeffgus | any codec conversion? |
| 23:18.26 | Derkommissar | Cpu(s): 38.3% us, 8.1% sy, 0.0% ni, 0.0% id, 0.0% wa, 0.3% hi, 3.3% si |
| 23:18.43 | Derkommissar | yes some calls from g729 to ULAW |
| 23:18.46 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 23:19.13 | jeffgus | Derkommissar, ah, interesting. so your load could be lower if they were all ulaw |
| 23:19.19 | *** join/#asterisk dlynes (n=dlynes@S010600c09f9a0fc4.vc.shawcable.net) |
| 23:19.22 | Derkommissar | yes |
| 23:19.47 | Derkommissar | and for every call i run 2 perl agi's scripts for routing |
| 23:20.25 | jeffgus | is the as5400 a better way to agregate a bunch of channel banks like i need to do? |
| 23:20.48 | Derkommissar | depends... digium cards are good and cheap |
| 23:21.03 | Derkommissar | a 5400 would run you about 25,000 |
| 23:21.08 | jeffgus | woah |
| 23:21.09 | jeffgus | ok |
| 23:21.10 | jeffgus | heheh |
| 23:21.39 | jeffgus | from your description i should be fine even if i had to handle 200 analog lines |
| 23:21.45 | Derkommissar | i did it because i had no other choice |
| 23:21.50 | jeffgus | as long as they are all 64kbit |
| 23:21.54 | Derkommissar | my e1's where ss7 |
| 23:22.09 | jeffgus | oh, ok, that makes sense |
| 23:22.13 | Derkommissar | so i had to use the as5400 along with a cisco PGW |
| 23:22.33 | Derkommissar | if it would have been ISDN i would have probably done it with digium or sangoma cards |
| 23:23.11 | jeffgus | ok |
| 23:23.16 | jeffgus | this is good info |
| 23:23.32 | jeffgus | it sounds like i don't have to be worried about the configuration then |
| 23:24.10 | Derkommissar | i would say so |
| 23:24.11 | jeffgus | Derkommissar, so you are a ViOP provider then? |
| 23:24.19 | jeffgus | that is why you needed the ss7 link? |
| 23:24.20 | Derkommissar | dont use any modules you dont have to |
| 23:24.24 | Derkommissar | correct |
| 23:24.31 | justinu | there was another voip provider here earlier... he was pretty cool |
| 23:25.26 | jeffgus | the customer was asking me if residents could have their own phone numbers routed though the asterisk box |
| 23:25.51 | Derkommissar | yea you can |
| 23:25.55 | Derkommissar | you can port them over |
| 23:25.59 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 23:26.09 | jeffgus | Derkommissar, is it difficult to work with a phone company to do that? |
| 23:26.15 | justinu | jeffgus: are these 200 analog lines for stations, or trunks? |
| 23:26.16 | jeffgus | i've never had to ask for that from them |
| 23:26.23 | Derkommissar | no |
| 23:26.28 | Derkommissar | it takes 10 days |
| 23:26.32 | jeffgus | justinu, 200 analog lines going to condos in a high rise building |
| 23:26.35 | justinu | ah |
| 23:26.43 | Derkommissar | and depends on your provider |
| 23:26.48 | justinu | delivering DSL over those lines also? |
| 23:26.57 | jeffgus | Derkommissar, no ss7 required, correct? the phone company sends proper callerid info? |
| 23:27.28 | justinu | jeffgus: considering it's residential usage, I think you'll experience fairly low volume |
| 23:27.32 | Derkommissar | no |
| 23:27.34 | jeffgus | justinu, no, the building hasn't decided on internet access. options are Ethernet or cable modems |
| 23:27.47 | justinu | you could also deliver DSL over the existing telco wiring |
| 23:27.48 | jeffgus | justinu, yes |
| 23:27.49 | Derkommissar | you can just port the numbers to a pri |
| 23:27.55 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 23:28.03 | Qwell[] | justinu-- |
| 23:28.14 | Qwell[] | buying and setting up a dslam == :( |
| 23:28.15 | *** join/#asterisk op3r (n=op3r@202.71.189.66) |
| 23:28.18 | justinu | is it that rough? |
| 23:28.22 | Qwell[] | dunno |
| 23:28.27 | Qwell[] | probably not, heh |
| 23:28.33 | jeffgus | Derkommissar, that sounds fairly easy then. i'll let them know that it can be an option. |
| 23:28.37 | Qwell[] | You would have to deal with people bitching though... |
| 23:28.37 | justinu | yeah - i've got some contacts working for a company that does exactly that |
| 23:28.40 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-92.apple.com) |
| 23:28.42 | op3r | one question, How can you automatically start up asterisk and mysql everytime the server starts? |
| 23:28.43 | Qwell[] | at least ethernet is easy |
| 23:28.48 | justinu | piggybacks DSL onto existing telco wiring |
| 23:28.59 | justinu | the DSLAMs aren't even that expensive |
| 23:29.01 | jeffgus | justinu, the company i'm working with is a private cable tv operator for high rise condos |
| 23:29.10 | Derkommissar | just add safe_asterisk to your startup script |
| 23:29.11 | justinu | cool |
| 23:29.12 | jeffgus | this main bread and butter is CATV |
| 23:29.22 | justinu | what city? |
| 23:29.32 | jeffgus | SoCal |
| 23:29.40 | jeffgus | Westside and download Los Angeles |
| 23:29.42 | justinu | company isn't polaris, is it? |
| 23:29.45 | jeffgus | nope |
| 23:29.46 | Derkommissar | Cool |
| 23:29.51 | justinu | ok |
| 23:29.55 | Derkommissar | Download Los Angeles ? |
| 23:29.56 | jeffgus | justinu, British American |
| 23:29.59 | justinu | polaris is more business oriented |
| 23:30.01 | Derkommissar | LOL |
| 23:30.10 | jeffgus | Derkommissar, you know them? |
| 23:30.15 | tainted- | jeffgus are you in LA? |
| 23:30.19 | jeffgus | yes |
| 23:30.19 | justinu | but they offer a similar package |
| 23:30.23 | Derkommissar | No it just sounds cool |
| 23:30.26 | Derkommissar | :-) |
| 23:30.28 | justinu | i'm from LA too |
| 23:30.30 | tainted- | is the Linux User Group any good in LA? |
| 23:30.32 | justinu | from/in |
| 23:30.37 | tainted- | i've been wanting to attend |
| 23:30.44 | Qwell[] | tainted-: there are several... |
| 23:30.45 | jeffgus | Derkommissar, the founder is from the UK |
| 23:30.47 | Derkommissar | since we are saying where we are from,,, Im from Miami |
| 23:30.49 | jeffgus | :) |
| 23:30.50 | justinu | i think strom_c runs one |
| 23:31.02 | tainted- | Qwell i thought you were in Canada |
| 23:31.05 | Derkommissar | i just went to VON canada |
| 23:31.05 | Qwell[] | no |
| 23:31.07 | justinu | lol |
| 23:31.17 | Derkommissar | in toronto they have an Asterisk User Group |
| 23:31.30 | tainted- | I wish there was an Asterisk User Group in LA |
| 23:31.30 | Qwell[] | tainted-: irvnca? meh |
| 23:31.31 | jeffgus | the big LUG is great, but i usually go to the much smaller SFVLUG |
| 23:31.33 | Derkommissar | :-/ I wonder why there isnt one here. |
| 23:31.48 | jeffgus | tainted-, we're trying to get one started as a SIG of the SFVLUG |
| 23:32.04 | tainted- | i'm in east LA.. the DSLAN is in irvine i guess |
| 23:32.14 | Qwell[] | That's one hell of a run |
| 23:32.21 | justinu | heh |
| 23:32.22 | tainted- | surprised me as well |
| 23:32.27 | justinu | they do that now |
| 23:32.34 | tainted- | i would love to get together and talk asterisk / biz |
| 23:32.45 | Qwell[] | tainted-: start one, you'll have... |
| 23:32.49 | justinu | dslam is in your local CO tho |
| 23:32.50 | Qwell[] | about 6 people there :p |
| 23:32.52 | justinu | not in irvine |
| 23:33.00 | Denmark | What do I loose if I delete the astdb every time asterisk is restarted? |
| 23:33.07 | Qwell[] | Denmark: The astdb |
| 23:33.09 | justinu | heh |
| 23:33.11 | tainted- | 6 peopls is a lot |
| 23:33.13 | justinu | sip registrations, etc. |
| 23:33.23 | Denmark | Qwell: Hehe .. sharp and fast. :-) |
| 23:33.24 | tainted- | u guys interested in a asterisk user group? |
| 23:33.27 | Qwell[] | tainted-: half of us might even bring beer :P |
| 23:33.36 | jeffgus | tainted-, check the #sfvlug channel on here. hopefully the AsteriskSIG will have meeting dates soon |
| 23:33.37 | justinu | depends on where it is |
| 23:33.39 | tainted- | wow all this time i thought u were a world away |
| 23:33.46 | tainted- | Qwell |
| 23:33.50 | Derkommissar | huh ! |
| 23:33.52 | Qwell[] | tainted-: It's IRC...I might as well be |
| 23:34.01 | Qwell[] | tainted-: actually, funny story... |
| 23:34.17 | Qwell[] | tainted-: There is a guy on a channel I frequent on efnet...lives literally around the corner from me |
| 23:34.21 | Qwell[] | in a channel of only 40 people |
| 23:34.53 | tainted- | lol |
| 23:35.04 | Qwell[] | saw his wsco.charter.net one day, and kinda wtf'd |
| 23:35.07 | *** join/#asterisk iq|mobile (n=iq@71-38-73-211.omah.qwest.net) |
| 23:35.14 | Denmark | Qwell: Is it #corner_of_east_and_west_street ? |
| 23:35.22 | Denmark | :o) |
| 23:35.24 | jeffgus | Derkommissar, thanks again for the detailed info. that really helped give me an idea on server requirements. it seems the voip-info wiki is a little pesimistic about requirments |
| 23:35.28 | Qwell[] | Denmark: yeah! How'd you know!? |
| 23:35.33 | Denmark | hehe :) |
| 23:35.39 | tainted- | Qwell where are you located? |
| 23:35.40 | Qwell[] | I deserved that one |
| 23:35.44 | Qwell[] | tainted-: west covina |
| 23:35.55 | tainted- | oh wow.. that's 10 mins away |
| 23:35.57 | tainted- | i'm in rosemead |
| 23:35.58 | jeffgus | hehe |
| 23:36.08 | jeffgus | i'm currently in Culver City right now |
| 23:36.09 | Qwell[] | tainted-: I'm currently sitting in rosemead... :p |
| 23:36.15 | jeffgus | but live in the SF Valley |
| 23:36.16 | Denmark | Pulver? |
| 23:36.18 | tainted- | oh wow haha |
| 23:36.23 | Qwell[] | 10fwy @ rosemead |
| 23:36.31 | tainted- | near the denny's? |
| 23:36.34 | Qwell[] | well, I guess it's technically el monte |
| 23:36.42 | tainted- | jeffgus santa clarita? |
| 23:36.47 | tainted- | jeffgus palmdale? |
| 23:36.48 | Qwell[] | umm...where is the denny's? Is that on ...wtf street do I work on? |
| 23:36.50 | jeffgus | nope, near northridge |
| 23:36.52 | Denmark | ~astdb |
| 23:36.54 | Qwell[] | Flair |
| 23:36.55 | jeffgus | city called Mission Hills |
| 23:37.03 | jeffgus | right were the 5 and 118 meet |
| 23:37.14 | Qwell[] | tainted-: if so, yes |
| 23:37.15 | tainted- | oh yea, near the BFE |
| 23:37.20 | Qwell[] | BFE? |
| 23:37.43 | tainted- | Qwell i know where you're at.. there's a jackinthebox there |
| 23:37.52 | tainted- | and some small office parks.. a few high rises |
| 23:37.55 | Qwell[] | tainted-: yes, I'm across the street from that. Two big bank buildings... |
| 23:38.05 | tainted- | u work for a voip company? |
| 23:38.10 | Qwell[] | No, a bank |
| 23:38.29 | Denmark | Joke? |
| 23:38.31 | [av]bani | Qwell[] is evil |
| 23:38.34 | tainted- | oh |
| 23:38.34 | Qwell[] | [av]bani: very |
| 23:38.38 | tainted- | that's a surprise |
| 23:38.40 | mog_work | Qwell, cant be evil |
| 23:38.45 | Qwell[] | mog_work: I can't? |
| 23:38.48 | tainted- | justinu|laptop where are you at? |
| 23:38.52 | [av]bani | and Qwell[] sucks because he hasnt given me chan_skinny yet :/ |
| 23:38.55 | mog_work | i will not live under an evil dictator |
| 23:39.00 | Qwell[] | [av]bani: ...hello :P |
| 23:39.02 | Qwell[] | 6589 |
| 23:39.09 | [av]bani | O RLY |
| 23:39.10 | Qwell[] | 6859 rather |
| 23:39.13 | [av]bani | LIES |
| 23:39.16 | [av]bani | DECEIT |
| 23:39.22 | Qwell[] | http://bugs.digium.com/view.php?id=6859 |
| 23:39.24 | Qwell[] | Go test it |
| 23:39.41 | mog_work | commit it qwell |
| 23:39.42 | [av]bani | dynamic blf? |
| 23:39.44 | mog_work | you know you want to |
| 23:39.44 | Qwell[] | mog_work: heh |
| 23:39.52 | Qwell[] | [av]bani: I need to support blf first :p |
| 23:39.58 | mog_work | do it cause i told you to. |
| 23:39.59 | Qwell[] | mog_work: I should. |
| 23:40.01 | Qwell[] | k |
| 23:40.29 | [av]bani | applies to 1.2.6 ? |
| 23:40.43 | Qwell[] | no |
| 23:41.15 | [av]bani | hmm |
| 23:41.22 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 23:41.24 | Qwell[] | well, maybe |
| 23:41.26 | [av]bani | is it possible to turn on the backlight on 7970 when the phone rings? |
| 23:41.27 | Qwell[] | if you ignore whitespace |
| 23:41.39 | Qwell[] | [av]bani: I don't know...possibly |
| 23:41.59 | [av]bani | i also would like to prevent the phone from obscuring the speeddial names when i pick the handset up :P |
| 23:42.15 | [av]bani | the dial window is huuuuuuuuge |
| 23:42.18 | Qwell[] | [av]bani: I don't think that's possible. The call plane is part of the firmware |
| 23:42.23 | *** join/#asterisk exten123 (n=exten@202.133.101.88) |
| 23:42.26 | [av]bani | there must be a setting somewhere |
| 23:42.29 | [av]bani | or hax the firmware :/ |
| 23:42.32 | Qwell[] | heh |
| 23:42.34 | Qwell[] | hex |
| 23:42.59 | [av]bani | in order to speeddial i have to speeddial, which puts it on speakerphone, then pick the handset up |
| 23:43.03 | Denmark | I must sleep |
| 23:43.07 | [av]bani | if i want to pick the handset up first, i cant see the names :/ |
| 23:43.48 | [av]bani | also bad for transfers, cant see speeddials |
| 23:43.59 | [av]bani | transfer should close the call plane |
| 23:44.06 | [av]bani | so you can f'n see things |
| 23:44.07 | Denmark | [av]bani : I wish someone would make an IP-phone with free software BIOS, GPL like asterisk. |
| 23:44.17 | [av]bani | Denmark: you mean like pa168x ? |
| 23:44.18 | exten123 | can we set group sip extensions so that can dial by group like what ZAP got? |
| 23:44.19 | mog_work | umm ssnom ......... |
| 23:44.25 | mog_work | snom runs linux |
| 23:44.28 | mog_work | and has oss stuff |
| 23:44.29 | [av]bani | mog_work: its not gpl |
| 23:44.30 | Qwell[] | mog_work: snom firmware is open source? |
| 23:44.39 | mog_work | snom firmware runs linux and can be reflashed |
| 23:44.40 | Denmark | [av]bani : pa168x? |
| 23:44.56 | mog_work | they have it all on their site |
| 23:45.00 | [av]bani | mog_work: the hardware is undocumented, there's a stillborn project attempting to hack it, but so far got nowhere |
| 23:45.04 | [av]bani | mog_work: no |
| 23:45.07 | *** join/#asterisk coolhp (n=crap@modemcable240.139-203-24.mc.videotron.ca) |
| 23:45.09 | [av]bani | they have the kernel, that's about it |
| 23:45.11 | [av]bani | no drivers |
| 23:45.14 | [av]bani | no code |
| 23:45.15 | [av]bani | no nothing |
| 23:45.32 | mog_work | kernel drivers for everything |
| 23:45.38 | [av]bani | you have to reverse engineer the binaries to figure out the hw |
| 23:45.39 | Denmark | well |
| 23:45.48 | coolhp | Would anyone happen to know what this means ? Unable to create channel of type 'SIP' (cause 58 - Bearer capability not available) |
| 23:45.57 | Denmark | you need userspace stuff also .. that too needs to be free. |
| 23:46.23 | Denmark | coolhp : Just a guess.. have you enabled the codecs? |
| 23:46.26 | [av]bani | http://www.openhardphone.org/ <- snom openfirmware project |
| 23:46.28 | [av]bani | it's dead |
| 23:46.49 | Denmark | so pa168x? |
| 23:46.50 | coolhp | Denmark : I have... at least in on that specific sip user's context. |
| 23:46.55 | [av]bani | no activity because there's insufficient info to make a real firmware for it |
| 23:47.09 | [av]bani | Denmark: pa168x is the only true opensource hardphone out there right now |
| 23:47.25 | Denmark | [av]bani : How is hardware quality? |
| 23:47.32 | [av]bani | Denmark: cheap chinese crap mostly |
| 23:47.36 | Denmark | darn |
| 23:47.45 | Denmark | sound? |
| 23:47.52 | coolhp | But it might be related to :Asked to get a channel of unsupported format ilbc while capability is unknown |
| 23:48.00 | coolhp | Which is even more confusing. |
| 23:48.00 | PakiPenguin_ | hi |
| 23:48.01 | coolhp | LOL |
| 23:48.05 | PakiPenguin_ | anyone here uses astribank? |
| 23:48.15 | PakiPenguin_ | i am having tons of trouble :( configuring it |
| 23:48.21 | PakiPenguin_ | actually getting it to be detected |
| 23:48.48 | Denmark | [av]bani : Is the sound quality worse than glx2000? |
| 23:49.26 | coolhp | Denmark... I think I may have gound the issue. |
| 23:49.33 | coolhp | But yet I am unsure of how to resolve it. |
| 23:49.47 | coolhp | Should asterisk be able to transcode from ilbc to ulaw ? |
| 23:50.01 | Denmark | coolhp : I only use ulaw so far. |
| 23:51.13 | Denmark | [av]bani : Have you tried the phone yourself? |
| 23:52.05 | NewSole | is there a real life |
| 23:52.33 | Denmark | NewSole : Sure, but its surreal. |
| 23:52.56 | NewSole | is there a real life h323 and asterisk... and not just fanitcy |
| 23:53.19 | tainted- | NewSole michael! |
| 23:53.27 | NewSole | yes |
| 23:53.29 | tainted- | NewSole how are the negotiations going |
| 23:53.50 | NewSole | still going |
| 23:53.50 | Denmark | It seems that pa168x even comes in an iax2 version.. |
| 23:54.42 | tainted- | NewSole expect them to release an in-house version before those talks end :D |
| 23:55.09 | x86 | what does everyone think of my website design: http://www.shellshark.net/ |
| 23:55.10 | NewSole | already running beta now |
| 23:55.29 | tainted- | really! |
| 23:55.31 | Denmark | http://www.aredfox.com/esolutions.htm |
| 23:55.37 | dlynes | coolhp: yes |
| 23:55.44 | NewSole | yup |
| 23:56.19 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 23:56.46 | dlynes | coolhp: What is the environment within which you're having codec problems? |
| 23:57.20 | dlynes | [hC]: You're from Vancouver? |
| 23:57.34 | [hC] | yeah, how did you know? |
| 23:57.35 | [hC] | :) |
| 23:57.51 | [hC] | metrobridge, i guess. |
| 23:57.52 | dlynes | Metrobridge networks :) |
| 23:58.19 | [hC] | how do you know metrobridge? |
| 23:58.19 | tainted- | can u guys get DIDs in 604? |
| 23:58.25 | dlynes | Yeah, easily |
| 23:58.35 | dlynes | I've got 484 and 628 cos |
| 23:58.35 | [hC] | I have DID's in 604 |
| 23:59.15 | tainted- | [hC] how much? |
| 23:59.18 | dlynes | [hC]: Are you looking for any extra work on asterisk? |
| 23:59.26 | [hC] | here's the deal |
| 23:59.33 | [hC] | I work for metrobridge as a routing engineer |
| 23:59.40 | [hC] | i also share office space with them |
| 23:59.43 | dlynes | [hC]: ah...guess not, then :) |
| 23:59.45 | [hC] | for my company, voxter communications |
| 23:59.47 | [hC] | however! |
| 23:59.53 | [hC] | my company (voxter) is looking for skilled asterisk people |