| 00:00.11 | AgiNamu | Qwell, well, each user CAN sign up. The thing is, each user must be registered somehow |
| 00:00.12 | AgiNamu | we don't magically detect their location :) |
| 00:00.19 | Qwell | no, of course not |
| 00:00.21 | AgiNamu | We offer a website, and we'll also have a SOAP interface |
| 00:00.29 | Qwell | but, it is a seperate account with you, or? |
| 00:00.37 | AgiNamu | so you could write a program to export your db to us |
| 00:00.58 | AgiNamu | You get one master account, then you authenticate to us |
| 00:00.59 | memmo | anyone know of a voip offering support for asterisk and Toronto (416) DIDs ? |
| 00:01.03 | AgiNamu | and tell us who the user is etc. |
| 00:01.07 | AgiNamu | as well as which DID |
| 00:01.43 | Qwell | "Our server will respond with the 10-digit PSTN number to call" I assume thats unique per provider or something? |
| 00:01.58 | AgiNamu | well, it works in combo with the 10-digit + caller ID |
| 00:02.20 | AgiNamu | So, our gateway might tell you to dial 123, caller id 456 |
| 00:02.23 | AgiNamu | that'll route you |
| 00:02.35 | AgiNamu | We're going to write an asterisk app that will set things up |
| 00:02.47 | AgiNamu | and just leave it to you to add 2 lines to the dialplan to Dial(...) |
| 00:03.22 | opus__ | does redirecting ports for IAX2 work well? I just created a monster.. behind nat behind nat |
| 00:03.35 | AgiNamu | opus, yea, IAX2 doesnt seem to have any issues |
| 00:03.38 | AgiNamu | I've done multi-nat |
| 00:03.43 | opus__ | cool |
| 00:03.50 | *** join/#asterisk lpires (~lpires@200.243.188.2) |
| 00:04.07 | terrapen | wow |
| 00:04.14 | Qwell | AgiNamu: I'm looking at this pricing...kinda sucks for per minute providers, who have users that register and do nothing with their account |
| 00:04.17 | AgiNamu | Qwell, the 10-digit redirect could be a "random" number as far as you are concerned. |
| 00:04.17 | terrapen | setting format=wav fixes the problem |
| 00:04.20 | terrapen | (in voicemail.conf) |
| 00:04.25 | terrapen | not sure why wav49 would be broken |
| 00:04.30 | terrapen | maybe its a -HEAD thing |
| 00:04.31 | AgiNamu | Qwell, well, if they provide a DID, they have to |
| 00:04.42 | AgiNamu | but Intrado isn't exactly a cheap solution |
| 00:04.53 | Qwell | This whole thing is gonna suck for end users |
| 00:05.05 | cochi | mh. any comments yet on if format=foo|bar had changing behaviour? |
| 00:05.07 | AgiNamu | Qwell, why do you say that? |
| 00:05.17 | Qwell | AgiNamu: I use less then $1 some months. |
| 00:05.19 | terrapen | qwell, but it will really suck for an end-user with an emergency and no 911 service |
| 00:05.31 | AgiNamu | you get a DID for less than a $? |
| 00:05.31 | Qwell | terrapen: I'll use my cell |
| 00:05.40 | Qwell | AgiNamu: tollfree, yes |
| 00:05.51 | terrapen | i was over at my buddies two months ago, cleaning our weapons |
| 00:06.00 | AgiNamu | oh hmm... i don't know if Tollfree numbers are required. I guess so :P |
| 00:06.02 | terrapen | and he had an accidental discharge and shot himself in the femur |
| 00:06.09 | terrapen | thank god he had 911 service |
| 00:06.15 | *** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net) |
| 00:06.16 | AgiNamu | Well, I'm not the FCC. I think they should have a bit more reasonable laws |
| 00:06.21 | terrapen | because you can't get cell service very well here |
| 00:06.31 | Pete_Largo | where is that zaptel autoconfiguration tool located on the web? |
| 00:06.33 | Qwell | terrapen: 911 always roams, doesn't it? |
| 00:06.38 | AgiNamu | but it's a LOT cheaper than paying the access fee + redudant frame relay + T1 + PRIs, etc. etc. |
| 00:06.47 | terrapen | qwell, not if you dont have cell service |
| 00:06.54 | AgiNamu | Even at our 10K DID pricing at *93* cents, it's much cheaper with us. |
| 00:07.13 | terrapen | he lives in a big concrete and steel building and he probably would be dead right now if i had to leave him bleeding to go call 911 |
| 00:07.28 | Qwell | terrapen: well, for that, sure, its great having it |
| 00:07.29 | AgiNamu | yea, I've never had good luck with cells inside buildings :( |
| 00:07.39 | blitzrage | terrapen: you were cleaning loaded guns? |
| 00:07.49 | terrapen | blitz, i wasn't |
| 00:07.50 | AgiNamu | Qwell, I guess that tollfree providers shouldn't offer outgoing service then |
| 00:07.58 | AgiNamu | that'd get them out of the FCC mandate |
| 00:08.06 | Qwell | AgiNamu: Then I would need a second provider, which is just stupid |
| 00:08.07 | terrapen | for some reason, he didn't drop his magazine before clearing |
| 00:08.11 | blitzrage | terrapen: sheesh, I've been cleaning guns since I was 5, and even I know to check that the gun isn't loaded |
| 00:08.13 | terrapen | and he chambered a round |
| 00:08.16 | Qwell | and, if they DID do that, they'd charge me more for my DID |
| 00:08.24 | terrapen | i have no idea why it happened |
| 00:08.25 | blitzrage | terrapen: that sucks... |
| 00:08.26 | Qwell | no matter how you slice it, I as a customer, get kinda screwed. |
| 00:08.30 | terrapen | he's been around guns all his life, too |
| 00:08.30 | AgiNamu | Qwell, not a second provider, a second service. |
| 00:08.35 | blitzrage | terrapen: someone wasn't respecting their guns |
| 00:08.38 | AgiNamu | if you have an outgoing-only service, you dont pay |
| 00:08.46 | AgiNamu | if you have incoming-only, you obviously dont pay |
| 00:08.53 | terrapen | qwell, i'm a pay-as-you-go customer, too |
| 00:08.58 | AgiNamu | So, providers just need to get a bit more creative |
| 00:09.00 | AgiNamu | it's not that hard. |
| 00:09.02 | terrapen | but i dont mind paying a buck or two each month for 911 |
| 00:09.03 | Qwell | if you have both with the same provider, you're using whats called a "loophole" |
| 00:09.12 | Qwell | quite looked down upon |
| 00:09.13 | terrapen | hell, i think POTS users have to pay a 911 fee anyway |
| 00:09.19 | terrapen | at least here in san antonio, tx they do |
| 00:09.31 | AgiNamu | it might be a "loophole", or it could be completely legit |
| 00:09.35 | blitzrage | terrapen: same here in ON, Canada |
| 00:09.42 | Qwell | its most certainly a loophole |
| 00:09.55 | AgiNamu | if you have a device that receives calls just to upload info, it obviously doesn't need 911 service |
| 00:10.00 | Qwell | one that will likely be closed quite quickly |
| 00:10.03 | AgiNamu | if you have another device that just places calls |
| 00:10.05 | terrapen | is there a table that lists the actual 911 number for each areacode+prefix? |
| 00:10.08 | AgiNamu | they are completely separate services |
| 00:10.35 | AgiNamu | Well, im just trying to be helpful :\ |
| 00:10.43 | terrapen | hrmmm |
| 00:10.58 | AgiNamu | Anyways, I just wanted to put that out, see what you guys thought. |
| 00:11.06 | terrapen | its interesting agi |
| 00:11.10 | AgiNamu | you can email me at mgg@dash911.com |
| 00:11.19 | Qwell | You all owe me...I'm subsidizing your 911 fee. :p |
| 00:11.25 | terrapen | anyone want to help me debug wav49 writing? |
| 00:11.25 | AgiNamu | We looked at a lot of stuff before doing this. Inc. that company that does stuff with OnStar |
| 00:11.42 | terrapen | i'm trying to figure out why Asterisk is writing apparently munged wav49 format files |
| 00:11.52 | AgiNamu | but AFAIk, Only intrado has deals with the ILECs and access to the selective routers + access to ALI |
| 00:11.57 | AgiNamu | so, thanks |
| 00:11.58 | AgiNamu | I gotta go now |
| 00:12.08 | irv999 | is there any way for a phone to list all calls parked on asterisk? |
| 00:13.03 | Nivex | that would be a fun way to hijack a call :) |
| 00:13.43 | opus__ | qwell -- loopholes are there for a reason |
| 00:14.00 | Qwell | opus__: So, I'm gonna have to buy two ATAs? |
| 00:14.11 | Qwell | nah, its stupid, and gonna cost me money... |
| 00:14.19 | AgiNamu | yea, well, thank your government then |
| 00:14.30 | AgiNamu | I'm just tyring to make it less expensive than it'd otherwise be for your ITSP |
| 00:14.40 | Qwell | AgiNamu: nothing against you at all |
| 00:14.46 | opus__ | i'm starting my own 911 service. i'm going to have my own fire, amb, and police |
| 00:14.46 | AgiNamu | :) |
| 00:14.57 | Qwell | its just, now that I'm seeing how the pricing structure will likely work... |
| 00:15.00 | Qwell | I'm pissed |
| 00:15.08 | AgiNamu | well, ther'es a possibility of per-call pricing |
| 00:15.14 | AgiNamu | But that's not an option right now |
| 00:15.32 | opus__ | nobody ever calls 911 and if they do they use there cellphone. |
| 00:15.38 | Qwell | opus__: exactly! |
| 00:15.42 | *** join/#asterisk silug (~steve@customer34.wisperisp.net) |
| 00:15.55 | Qwell | honestly...who is stupid enough to trust A) Their ISP, B) Their provider, C) Their providers provider |
| 00:15.55 | AgiNamu | opus__, from stats I've looked at, each subscriber will call once a year, approx. |
| 00:15.59 | Qwell | its absolutely rediculous |
| 00:16.07 | AgiNamu | The same people who fine for a nipple? |
| 00:16.12 | *** join/#asterisk JohnJacob (~JohnJacob@pcp0011543387pcs.mainf01.in.comcast.net) |
| 00:16.13 | *** join/#asterisk mmlj4 (~looseduk@ip68-14-124-25.no.no.cox.net) |
| 00:16.24 | opus__ | what pisses me off is that when i call 911 from my cellphone, the phone's GPS sends them my coordiantes. Now when I load up a debugger on my MPX200 smartphone, it saids there is no GPS. I can't even use my own fucking hardware |
| 00:16.59 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
| 00:17.08 | opus__ | the only reason to call 911 over voip is to start a law suit. like the wendy's chilli finger lady |
| 00:17.12 | Qwell | this might be a better idea, if they forced ISPs who have voip users to have 100% uptime |
| 00:17.20 | AgiNamu | opus__, you sure you're getting GPS? |
| 00:17.23 | Qwell | Then I'll buy it, sure |
| 00:17.25 | AgiNamu | GPS won't work well in buildings at all... |
| 00:17.39 | AgiNamu | And of course, a lot of GPS devices give you +/- 25metre resolution |
| 00:17.49 | AgiNamu | so in anywhere dense.... |
| 00:17.52 | opus__ | aginamu - yeah if i call 911 and say 'fuck you', 10 cops will show up no matter where I am. they send delta force i sware |
| 00:17.58 | AgiNamu | lol |
| 00:18.12 | terrapen | :q |
| 00:18.34 | AgiNamu | well, im audi5000 |
| 00:18.37 | Qwell | What they need to do... |
| 00:18.48 | Qwell | is have something where a customer can explicitly opt the fuck out. :p |
| 00:18.58 | opus__ | but what about grandma |
| 00:19.09 | Qwell | screw grandma, she can use my cellphone too |
| 00:19.28 | Qwell | Thats why it should be opt out, and not opt in |
| 00:19.30 | opus__ | you know, when she's like debugging the lastest asterisk CVS head on her linux 1 U server and like falls |
| 00:19.48 | Qwell | well then, she can use her own cellphone :p |
| 00:20.02 | opus__ | what if she has a rotary cell phone |
| 00:20.21 | Qwell | then she wouldn't opt out, now would she? |
| 00:20.46 | Qwell | I, for one, trust Sprint PCS (haha, damn I suck) more then I trust Adelphia |
| 00:21.42 | opus__ | when you need delta force, trust sprint pcs |
| 00:21.52 | *** join/#asterisk netvulture (0@63.174.172.245) |
| 00:23.08 | netvulture | anybody having tos issues with IAX or SIP when running HEAD? |
| 00:24.24 | terrapen | OHHHHH |
| 00:24.29 | terrapen | i think i figured out this problem |
| 00:24.55 | terrapen | so it looks like maybe Asterisk is using the file extenssion to determine the format of the recorded voicemail |
| 00:25.12 | terrapen | whatever.wav goes to format_wav.so |
| 00:25.29 | terrapen | and whatever.WAV goes to format_wav_gsm.so |
| 00:25.43 | terrapen | but the mac is case insensitive or something |
| 00:26.12 | terrapen | so somehow, its getting whatever.wav as the name of the file when it actually is whatever.WAV |
| 00:26.18 | terrapen | and sending it to the wrong processor |
| 00:26.56 | terrapen | maybe i can write a patch for this |
| 00:32.57 | jesster | is FreeBSD + Asterisk considered stable? I want to run SIP with MeetMe conferences |
| 00:33.06 | jesster | over G729 |
| 00:33.39 | *** join/#asterisk bjohnson (~bjohnson@66.11.188.215) |
| 00:36.39 | *** join/#asterisk SkySky (~Miranda@toronto-HSE-ppp4257121.sympatico.ca) |
| 00:37.33 | CoderCR | I belive Asterisk has a problem handling zaptel -> sip and vice versa |
| 00:38.57 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
| 00:39.29 | terrapen | i need to figure out where app_voicemail.c determines what format a recorded VM is |
| 00:41.13 | opus__ | how do I make festival sound like a chick |
| 00:43.00 | CoderCR | opus: take festival from behind |
| 00:44.55 | Qwell | bug 4403 could use some testers, if anybody is interested. It's for adding context and mailbox to the database for voicemails in odbc. Makes it nice and easy to parse, and can be used wonderfully across multiple * installations |
| 00:45.28 | xheliox | Bug testing? That's blasphemy! |
| 00:45.48 | xheliox | It's much easy to expect someone else to do it and then whine when it doesn't work right. :) |
| 00:46.03 | Qwell | :p |
| 00:46.14 | Qwell | well, if it isn't tested, I doubt it'll go into cvs |
| 00:47.14 | xheliox | Whew, that's a relief. ;) |
| 00:48.46 | terrapen | qwell |
| 00:49.09 | terrapen | any idea where app_voicemail.c determines the file type of a voicemail file? |
| 00:49.11 | Qwell | ? |
| 00:49.25 | Qwell | You'd think I'd know...but, I don't. I could try to look really quick |
| 00:49.40 | bewest | is it set in the conf file? |
| 00:50.03 | terrapen | i think it *may* be in make_file() |
| 00:50.07 | terrapen | im not sure |
| 00:50.21 | *** join/#asterisk zonezero (~zonezero@59.b167.bendtel.net) |
| 00:50.47 | Qwell | if (!strcasecmp(fmt, "wav49")) |
| 00:50.52 | Qwell | There are alot of those |
| 00:51.14 | terrapen | qwell, i think i've found a bug related to Mac OS X's case insensitivity |
| 00:51.16 | *** join/#asterisk lpires (~lpires@200.243.188.2) |
| 00:51.27 | *** part/#asterisk lpires (~lpires@200.243.188.2) |
| 00:51.33 | terrapen | i think Asterisk is using the wrong format to play my recorded VMs |
| 00:51.34 | Qwell | its case insensitive? |
| 00:51.37 | terrapen | yes |
| 00:51.45 | terrapen | Mac OS X is, i believe case insensitive |
| 00:52.03 | Druken | Qwell: any way to add the "press * when voicemail begins to play to login??" to voicemail? |
| 00:52.05 | terrapen | cjs@gunga-din [502] 19:51:52 [~]-> echo "dog" > Foo |
| 00:52.05 | terrapen | cjs@gunga-din [503] 19:51:57 [~]-> cat foo |
| 00:52.05 | terrapen | dog |
| 00:52.19 | Qwell | Druken: eh? |
| 00:52.26 | Qwell | terrapen: fun |
| 00:52.45 | terrapen | Asterisk, as far as i can tell, is using the file extension to determine the format |
| 00:53.07 | terrapen | and whatever C function it uses to get the filename probably returns the filename in all lower-case, no matter what it really is |
| 00:53.54 | Druken | Qwell: say i call myself, and i get the voicemail... can i press * to login and check my messages? |
| 00:53.55 | terrapen | which results in format_wav trying to play the file, when it really should be handled by format_wav_gsm |
| 00:53.55 | netvulture | ? |
| 00:54.21 | Qwell | Druken: dunno |
| 00:54.54 | terrapen | i really want to patch this so that i can say that i've contributed source code to Asterisk :) |
| 00:55.17 | Druken | Qwell: any idea who would know? |
| 00:55.25 | Druken | or if it could be added |
| 00:57.41 | terrapen | drunken, you can probably set that up in extensions.conf |
| 00:57.44 | terrapen | do some reading on that |
| 00:57.58 | *** join/#asterisk visik7 (~ciao@visik7.user) |
| 00:58.21 | terrapen | check to see if calling extension == called extension and if true, call the voicemail app |
| 00:58.48 | terrapen | or set up another extension |
| 00:58.53 | *** join/#asterisk bjohnson (~bjohnson@66.11.188.215) |
| 00:59.12 | terrapen | for example, if your extension is 108, set up extension 10800 to go to voicemail for mailbox 108 |
| 00:59.26 | terrapen | tons of ways to accomplish this |
| 00:59.31 | Druken | no... i men say i'm at your house and i call home, i want to be able to press * and be asked for my password |
| 01:00.48 | Qwell | Druken: have it run a macro on dial |
| 01:00.52 | Qwell | or something |
| 01:03.00 | Druken | it would have to be an option in the voicemail app |
| 01:03.49 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 01:04.12 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 01:04.33 | AgiNamu | Whoa.. Xwin from a Windows RDP session via VPN is a bit slow.. |
| 01:12.36 | opus__ | whats a good way to have a dial plan loop a few times, |
| 01:13.20 | AgiNamu | use a chanvar |
| 01:13.26 | AgiNamu | and a gotoif |
| 01:14.14 | cluecon[file] | chan_local |
| 01:14.16 | cluecon[file] | *G* |
| 01:14.54 | AgiNamu | chan_local to loop? |
| 01:15.02 | AgiNamu | that sounds evil |
| 01:15.03 | terrapen | hey qwell |
| 01:15.10 | cluecon[file] | it is, it literally loops |
| 01:15.12 | terrapen | did you write this ODBC stuff in app_voicemail? |
| 01:15.14 | cluecon[file] | [loop] |
| 01:15.18 | terrapen | <PROTECTED> |
| 01:15.20 | cluecon[file] | exten => 42,1,Dial(Local/42@loop) |
| 01:15.22 | cluecon[file] | ;) |
| 01:15.38 | Qwell | ? |
| 01:15.59 | Qwell | terrapen: not all of it, no. just the stuff that deals with context/mailbox in the DB |
| 01:15.59 | terrapen | qwell, was it you who was talking about ODBC & the voicemail app? |
| 01:16.04 | *** join/#asterisk lpires (~lpires@200.243.188.2) |
| 01:16.05 | Qwell | terrapen: yeah |
| 01:16.10 | Qwell | bug 4403, to see what I did |
| 01:16.11 | terrapen | how about the stuff in store_file()? |
| 01:16.14 | terrapen | is that you? |
| 01:16.17 | terrapen | ou/me looks |
| 01:16.21 | Qwell | some of it... |
| 01:16.51 | terrapen | would you mind answering a C question for me? i'm trying to sort out this case sensitivity problem |
| 01:17.20 | *** join/#asterisk map71 (~map@pfo-06-095.pfo.ras.cantv.net) |
| 01:17.47 | *** join/#asterisk apardo (~apardo@80.26.160.180) |
| 01:17.48 | terrapen | line 1072 |
| 01:17.52 | terrapen | <PROTECTED> |
| 01:17.53 | terrapen | <PROTECTED> |
| 01:17.59 | Qwell | lemme look |
| 01:18.20 | Qwell | ok, what about it? |
| 01:18.37 | terrapen | oh wait |
| 01:18.38 | terrapen | hrmm |
| 01:18.44 | *** join/#asterisk Eduardo_Henrique (~ehst@201008101247.user.veloxzone.com.br) |
| 01:18.51 | terrapen | store_file() is only used for the ODBC stuff? |
| 01:18.52 | Qwell | that gets pulled from the config, or something |
| 01:18.54 | Qwell | yes |
| 01:19.11 | Qwell | it gets called by the STORE() macro |
| 01:19.32 | terrapen | im trying to figure out where exactly app_voicemail determines the format of a voicemail audio file |
| 01:19.35 | terrapen | any ideas? |
| 01:19.58 | Qwell | looking still |
| 01:20.59 | Qwell | ret |= f->format; perhaps? |
| 01:21.01 | Qwell | file.c |
| 01:21.15 | Qwell | fn = build_filename(filename, ext); |
| 01:21.17 | Qwell | if (fn) { |
| 01:21.29 | Qwell | I assume build_filename returns fn, which is a bool whether it exists |
| 01:21.52 | terrapen | im not sure that app_voicemail uses build_filename |
| 01:21.58 | Qwell | it doesn't |
| 01:22.06 | Qwell | it uses a bunch of other crap, which eventually calls it |
| 01:22.12 | terrapen | ahhh |
| 01:22.14 | terrapen | ok |
| 01:22.16 | Qwell | ast_fileexists, which calls ast_filehelper |
| 01:22.32 | Qwell | figure out where the "formats" var comes from, and I'm pretty sure you've got it |
| 01:22.36 | Qwell | gotta head out though |
| 01:22.48 | terrapen | k |
| 01:22.49 | terrapen | thanks |
| 01:24.25 | harryvv | i cannot stant woman at times and there emotional hangups :) |
| 01:25.02 | *** join/#asterisk metacove (~metacove@mail.gettcomm.com) |
| 01:26.25 | Qwell | yeah...ast_filehelper and build_filename are your likely culprits. I'd start there |
| 01:26.40 | zonezero | anyone here know how the new 911 rules for voip will change things? esp with voicepulse and the like? |
| 01:26.45 | Pete_Largo | why is the CLI saying yay! ? |
| 01:26.57 | bewest | it's happy |
| 01:27.07 | Qwell | if (fn) |
| 01:27.09 | Qwell | snprintf(fn, fnsize, "%s.%s", filename, type); <--- bingo |
| 01:27.16 | Pete_Largo | what's it happy about? |
| 01:27.24 | Qwell | off I go |
| 01:27.25 | bewest | you had a choice and you chose * |
| 01:27.33 | Pete_Largo | lol |
| 01:27.53 | Qwell | Pete_Largo: apps/app_voicemail.c |
| 01:27.56 | Qwell | I was just looking at that... |
| 01:28.05 | Pete_Largo | hmm? |
| 01:28.25 | meppl | gute nacht |
| 01:28.33 | Pete_Largo | I guess the voicemail is happy? my 3 year old daughter will love that, she's into the are you happy phase... |
| 01:29.14 | bewest | your 3 year old daughter has voicemail? |
| 01:29.30 | opus__ | maybe she can do my voice over |
| 01:29.32 | opus__ | haha |
| 01:29.45 | Pete_Largo | how in the hell did you add that sentence together and end up with that result? |
| 01:29.51 | metacove | I have a quick ast question, can I bother you all with it ? |
| 01:29.57 | opus__ | sure |
| 01:29.59 | metacove | I installed asterisk on gentoo using the ebuild. I had previously installed it on CentOS 3 and 4 with no problems. For some reason I cannot make the modules autoload. |
| 01:29.59 | metacove | I have autoload=yes in /etc/asterisk/modules.conf but modules still don't dynamically load. I still have to explicitly load each module or I get unresolved dependancies in the /var/log/asterisk/messages file. |
| 01:29.59 | metacove | Does anyone have a suggestion ? I asked in gentoo-voip which seems rather dead =p |
| 01:30.08 | opus__ | will answer 4 food |
| 01:30.16 | metacove | ./dcc opus__ cheeseburger |
| 01:30.21 | metacove | answers plz =p |
| 01:31.33 | metacove | [res_features.so]May 31 20:12:32 WARNING[23270]: /usr/lib/asterisk/modules/res_features.so: undefined symbol: adsi_available |
| 01:31.39 | metacove | thats what i get in my error log |
| 01:31.54 | metacove | shouldnt autoload=yes in the modules.conf make it autoload the res_adsi.so or whatever the module is ? |
| 01:32.30 | bewest | is there a translater from g726 to ulaw? |
| 01:33.42 | Pete_Largo | what does this line mean? May 31 20:18:55 NOTICE[2633]: indications.c:402 ast_unregister_indication_country: Removed default indication country 'us' |
| 01:33.57 | Pete_Largo | why is it removing the country? |
| 01:34.11 | bewest | niZon, do you have an awesome providers list? |
| 01:34.12 | metacove | shhh. my questions first =p |
| 01:34.17 | niZon | nope |
| 01:34.23 | metacove | voipjet is pretty good provider |
| 01:34.27 | metacove | i think... |
| 01:34.33 | lpires | Which of the bolts of connector RJ45 I must use to connect FXO interface? |
| 01:34.37 | cluecon[file] | Asterlink is a pretty good provider |
| 01:34.41 | cluecon[file] | so I've heard |
| 01:34.51 | bewest | you work for them? |
| 01:34.59 | cluecon[file] | that's beside the point |
| 01:35.02 | bewest | hehe |
| 01:35.05 | bewest | I'm with them now |
| 01:35.07 | niZon | I have yet to find a good provider that can do 204 or cheap toll free over iax |
| 01:35.10 | bewest | err... you |
| 01:35.29 | cluecon[file] | our toll-free has some nifty stuff that nobody else has... |
| 01:35.55 | cluecon[file] | like we can send you the ANI2, meaning you'll know if the call comes from say like a payphone - so you can reject the call and not pay the extra fee |
| 01:36.14 | Druken | Qwell: do you know the voicemail source at all? |
| 01:37.07 | metacove | whats asterlinks pricing like ? |
| 01:37.50 | *** join/#asterisk obiyoda (~chatzilla@24-119-124-104.cpe.cableone.net) |
| 01:37.56 | metacove | anypne have a clue on the gentoo - asterisk question i posted ? |
| 01:38.00 | metacove | Im desperate :-) |
| 01:38.01 | cluecon[file] | metacove: 2 cents, just like everyone else |
| 01:38.17 | cluecon[file] | no international at the moment, but we're working on that |
| 01:38.29 | cluecon[file] | well, I worked on it and got it working... waiting on someone else *nudge nudge* to get rating and junk done |
| 01:38.30 | Eduardo_Henrique | Which of the bolts of connector RJ45 I must use to connect FXO interface? |
| 01:38.55 | niZon | cluecon[file]: more info on asertlink? |
| 01:39.03 | cluecon[file] | asterlink? what'cha wanna know |
| 01:39.18 | bewest | can you put more technical info on the website? |
| 01:39.23 | cluecon[file] | technical info such as? |
| 01:39.24 | *** part/#asterisk CoderCR (~creyna@209.242.148.51) |
| 01:39.25 | zonezero | cluecon[file], what company was that? |
| 01:39.34 | cluecon[file] | you get all the technical info when you signup, it comes in your welcome e-mail |
| 01:39.44 | cluecon[file] | zonezero: http://www.asterlink.com/ |
| 01:39.52 | cluecon[file] | we're also the ones putting on Cluecon: http://www.cluecon.com/ |
| 01:39.53 | bewest | actually, it'd be nice to switch from sip to iax o nthe website |
| 01:40.05 | bewest | cluecon[file], the person that signed up gets the email |
| 01:40.06 | zonezero | cool...voicepulse is pissing me off |
| 01:40.12 | bewest | sometimes that person is not the one setting it up :-) |
| 01:40.31 | niZon | cluecon[file]: can you accept paypal? |
| 01:40.33 | cluecon[file] | bewest: pfft |
| 01:40.49 | cluecon[file] | niZon: no |
| 01:41.13 | niZon | damn |
| 01:41.27 | cluecon[file] | I had a thought and one of you made me lose it |
| 01:41.42 | bewest | *inno* |
| 01:41.57 | cluecon[file] | meh whatever |
| 01:45.11 | metacove | ill repost my question just in case it disappeared =p |
| 01:45.11 | metacove | I installed asterisk on gentoo using the ebuild. I had previously installed it on CentOS 3 and 4 with no problems. For some reason I cannot make the modules autoload. |
| 01:45.11 | metacove | I have autoload=yes in /etc/asterisk/modules.conf but modules still don't dynamically load. I still have to explicitly load each module or I get unresolved dependancies in the /var/log/asterisk/messages file. |
| 01:45.11 | metacove | Does anyone have a suggestion ? I asked in gentoo-voip which seems rather dead =p |
| 01:45.41 | bewest | I've heard a lot of comments that the ebuild for * is no good |
| 01:45.51 | bewest | typical of packaging systems, if you ask me |
| 01:46.04 | opus__ | metacove |
| 01:46.08 | metacove | gentoo packager is better than most |
| 01:46.14 | metacove | its rather flexible |
| 01:46.23 | metacove | yes opus ? |
| 01:46.28 | opus__ | the problem is gentoo, |
| 01:46.34 | metacove | i got to visit digium a few weeks ago |
| 01:46.36 | bewest | that may be true, however I have specifically heard that the ebuild for * is very bad |
| 01:46.41 | opus__ | you did depmod -a of course |
| 01:46.46 | metacove | yes |
| 01:46.48 | metacove | but... |
| 01:46.48 | metacove | opus |
| 01:46.54 | metacove | im talking asterisk modules |
| 01:47.02 | metacove | ./usr/lib/asterisk/modules |
| 01:47.03 | opus__ | whats the unresolved dependancies |
| 01:47.11 | opus__ | oh |
| 01:47.14 | metacove | well, doesnt asterisk autoload modules on demand ? |
| 01:47.16 | opus__ | ok |
| 01:47.18 | metacove | yea. now u get me =p |
| 01:47.21 | metacove | i have autoload=yes |
| 01:47.27 | metacove | and my modules dir defined in asterisk.conf |
| 01:47.30 | metacove | but they dont autoload |
| 01:47.30 | opus__ | then you intsalled an old version, and then you installed a newer version on top of that |
| 01:47.33 | metacove | what more is there ? |
| 01:47.37 | metacove | not true |
| 01:47.39 | opus__ | rm -rf /usr/lib/asterisk/modules and reinstall |
| 01:47.44 | metacove | i installed 1.0.7 on a fresh box |
| 01:47.46 | metacove | i tried that opus |
| 01:47.50 | metacove | i saw a post that mentioned that |
| 01:47.52 | *** join/#asterisk Moonwick (~moonwick@core.dump.net) |
| 01:48.05 | opus__ | ok, whats the first unresolve depend |
| 01:48.23 | metacove | well asterisk spawns fine if i define each module with a load=> ... line in modules.conf |
| 01:48.32 | opus__ | ok |
| 01:48.34 | metacove | but the first one it tries to load is a dependancy of features.so |
| 01:48.39 | metacove | which happens to be res_adsi.so |
| 01:48.53 | metacove | <PROTECTED> |
| 01:49.09 | opus__ | well, i used gentoo exclusively for two years and then just gave it up for this exact reason |
| 01:49.24 | opus__ | and also having it replace /etc/fstab like every fuckign day |
| 01:49.31 | CoaxD | opus: gentoo sucks donkey balls |
| 01:49.32 | metacove | lol how come fstab ? |
| 01:49.42 | metacove | I've used gentoo since early days and its great |
| 01:49.44 | opus__ | they updated it or some crap |
| 01:49.47 | metacove | its not for point clickers or redhat lovers |
| 01:49.50 | metacove | but its quite nice |
| 01:49.51 | CoaxD | opus: the 5% performance increase (overall) it gives you just is not worth it |
| 01:49.58 | opus__ | coaxd yup |
| 01:50.00 | CoaxD | metacove: if you like compiling all your shit from SOURCE, yea! |
| 01:50.02 | metacove | I dont use it for the increase in optimizations |
| 01:50.09 | opus__ | why do you use it? |
| 01:50.32 | metacove | it does an excellent job of keeping configuration data in etc |
| 01:50.37 | CoaxD | opus: I think the whole reason gentoo users use gentoo is because they want to say they're not a point and click user |
| 01:50.42 | metacove | emerge / portage is so powerful |
| 01:50.51 | CoaxD | Do you realize that still, to this day, the package manager of gentoo has a tmpfile vulnerability? |
| 01:50.51 | metacove | you never get into dependancy hell like some other binary distros |
| 01:51.09 | CoaxD | well, emerge |
| 01:51.15 | metacove | im not trying to fight with you coaxd |
| 01:51.15 | CoaxD | Dunno if you'd call it a 'package manager' |
| 01:51.18 | metacove | dont turn it into that |
| 01:51.24 | metacove | why not ? |
| 01:51.29 | metacove | it manages the packages on the box |
| 01:51.32 | CoaxD | metacove: Dude, its not a fight. its truth. its a stupid fucking distro that makes no sense |
| 01:51.46 | opus__ | Emerging gcc... |
| 01:51.46 | metacove | wow |
| 01:51.47 | metacove | k |
| 01:51.47 | opus__ | faild |
| 01:51.48 | CoaxD | metacove: I will not drive my point home. I dont care if you agree with me or not. It is what it is.. |
| 01:51.50 | opus__ | hehe |
| 01:51.56 | metacove | i went to the wrong channel for help |
| 01:52.04 | CoaxD | metacove: Hey, dont get me wrong, asterisk works |
| 01:52.04 | metacove | thanks a whole lot for valuable contributions coaxd |
| 01:52.10 | opus__ | metacove - why not use centos?? |
| 01:52.23 | metacove | 90% of our boxes are gentoo |
| 01:52.25 | CoaxD | metacove: And it even works on gentoo |
| 01:52.30 | metacove | have a portage / distfiles mirror in house |
| 01:52.33 | CoaxD | metacove: Oh god. *groan* I'm sorry. |
| 01:52.43 | CoaxD | metacove: Must be a maintenence nightmare |
| 01:52.43 | metacove | ignored |
| 01:52.48 | metacove | =p |
| 01:52.55 | CoaxD | metacove: I mean, seriously. how do you handle mass upgrades? |
| 01:53.04 | CoaxD | metacove: (Aside from rsyncing a brand new 'world'>) |
| 01:53.07 | opus__ | you can't have nightmares if you don't go to sleep w/ on all those god damn gentoo boxes :) |
| 01:53.25 | metacove | heh |
| 01:53.35 | CoaxD | metacove: debian, i 'apt-get update;apt-get dist-upgrade' |
| 01:53.43 | CoaxD | metacove: fc2, 'yum update' |
| 01:53.45 | metacove | Ill keep working at it myself |
| 01:53.51 | CoaxD | metacove: what do gentoo users do? |
| 01:53.54 | metacove | thanks for your feedback opus |
| 01:54.04 | CoaxD | metacove: I mean, with FreeBSD users, they gotta run some nifty thingie to update the ports tree |
| 01:54.14 | metacove | and coaxd, Ive never experienced such an unhelpful group of responses from opensource ppl |
| 01:54.16 | CoaxD | metacove: And update the core fs with stuff that needs updating, and figures out which is what |
| 01:54.17 | metacove | thanks for a first :-) |
| 01:54.18 | opus__ | metacove - hope you find your problem. |
| 01:54.30 | CoaxD | metacove: um, i'm asking you a real question. |
| 01:54.45 | metacove | im sure i will |
| 01:54.58 | metacove | im gonna modify the ebuild and take out any gentoo-esque patches that seem iffy |
| 01:54.58 | CoaxD | metacove: And should you actually tell me what the fucking problem is, maybe I can help with your asterisk shit |
| 01:55.12 | metacove | ... |
| 01:55.16 | metacove | more helpful response =p |
| 01:55.32 | CoaxD | metacove: How am I supposed to help if I don't know what the problem is? |
| 01:55.38 | opus__ | unresolved shared library dependencies w/ asterisk modules loading, however works fine when loading one by one |
| 01:55.45 | CoaxD | Hmmmmmmm |
| 01:55.52 | CoaxD | whacky. shit. |
| 01:55.54 | opus__ | autoload is broken |
| 01:56.03 | metacove | its not dependancies / resolve issues in gentoo, but asterisk isnt dynamically loading its internal modules |
| 01:56.05 | cluecon[file] | autoload works fine in my world |
| 01:56.06 | opus__ | metacove -- try, strace -f asterisk |
| 01:56.09 | metacove | so assumed asterisk experts could help =p |
| 01:56.18 | CoaxD | metacove: Well, sorta |
| 01:56.22 | metacove | alright opus =p |
| 01:56.26 | CoaxD | metacove: there are a gazillion gentoo/asterisk users |
| 01:56.34 | CoaxD | metacove: None just happen to be here at the moment :/ |
| 01:56.44 | opus__ | d00d just 3mr3g3 $ync !!! :) |
| 01:56.45 | metacove | heh. i understand coax :-) |
| 01:56.52 | metacove | hehe opus |
| 01:57.01 | CoaxD | metacove: You know, maybe just a post to asterisk-users would give you a response back in 10 minutes? |
| 01:57.07 | CoaxD | metacove: Could be something stupid simple |
| 01:57.25 | CoaxD | metacove: Then again, most gentoo users i know say to screw the gentoo package and build their own |
| 01:57.35 | metacove|away | yea |
| 01:57.41 | metacove|away | i think i may have to come to that fairly soon |
| 01:57.53 | opus__ | what??? you didn't srat that way??? |
| 01:57.55 | CoaxD | metacove: It really isnt that hard to do... |
| 01:58.07 | CoaxD | metacove: Plus, you get the latest and greatest CVS tree, which is, sadly enough, probably the least broken tree |
| 01:58.11 | metacove|away | im very familiar with compiling coax =p |
| 01:58.22 | CoaxD | metacove: Well then, why aren't you doing it? :) |
| 01:58.27 | opus__ | dude never use gentoo ebuilds for anything production worthy. thats like hiring a hippy to smoke weed all day |
| 01:58.27 | metacove|away | but if i can keep it managed by the package manager it would be nice =p |
| 01:58.42 | CoaxD | opus: Can I send my resume, please? |
| 01:58.43 | metacove|away | lol opus |
| 01:59.07 | CoaxD | opus: Oh. and do you supply the weed? |
| 01:59.10 | opus__ | only if you got the hook up |
| 01:59.12 | opus__ | :) |
| 01:59.17 | CoaxD | god damnit |
| 01:59.18 | CoaxD | *spit* |
| 01:59.21 | opus__ | yeah, this shit is generation two |
| 01:59.36 | CoaxD | which reminds me... where *is* my frickin' bowl, anyway? |
| 01:59.42 | cluecon[file] | I melted it |
| 01:59.53 | CoaxD | doh :( |
| 02:00.02 | CoaxD | you musta had an acetylene torch, then. *g* |
| 02:00.05 | opus__ | you smoked it all when you rewrote the gentoo ebuild last night |
| 02:00.09 | cluecon[file] | yup |
| 02:00.15 | CoaxD | opus: Bahahahaha |
| 02:00.24 | *** join/#asterisk baka03 (~mike@60-240-47-106.tpgi.com.au) |
| 02:00.28 | CoaxD | opus: I think you got it there, sir |
| 02:00.29 | opus__ | hehe:) |
| 02:00.48 | CoaxD | you know, i remember back when I worked for SourceForge... The lead admin on the team really liked gentoo |
| 02:00.52 | SplasPood | coaxd: fuckin stoner |
| 02:00.58 | CoaxD | Hey Splas.. :) |
| 02:01.14 | CoaxD | Splas: How goes? |
| 02:01.22 | baka03 | anyone willing to help me out with setting up my iax.conf and extensions.conf so i can dial out |
| 02:01.42 | opus__ | sure |
| 02:01.45 | CoaxD | baka03: Sure, but the thing is, there are really good examples out there already |
| 02:01.57 | opus__ | yeah what coaxd said |
| 02:02.04 | baka03 | i have been looking at the examples at voip-info.org |
| 02:02.12 | CoaxD | baka03: And unfortunately, asterisk's config *is* a bit of a bitch |
| 02:02.29 | baka03 | even had a section for my provider but i just cant seam to get it to work |
| 02:02.34 | CoaxD | baka03: But the idea is to wrap your head around it. Took me a couple weeks of fucking with it, but i get it now |
| 02:02.54 | CoaxD | baka03: But once you DO get your head around it, you'll be able to do ANYTHING with asterisk |
| 02:03.04 | baka03 | i hope so |
| 02:03.08 | CoaxD | baka03: (because you will understand simple extension logic, etc) |
| 02:03.10 | opus__ | baka i think you can just get away with Dial(IAX2/user:pass@host/theextenionnumber) |
| 02:03.26 | CoaxD | baka03: And BTW, voip-info.org woulda been the site i would send you to |
| 02:03.37 | CoaxD | baka03: So at least you were in the right place! *lol* |
| 02:03.55 | cluecon[file] | my young mind took so little time to grasp asterisk |
| 02:04.10 | opus__ | i hate writing invoices |
| 02:04.13 | baka03 | ok in my extensions.conf i have a [default] section im assuming when asterisk is working out how to dial a number it starts there? |
| 02:04.16 | CoaxD | cluecon: Yeah, it seems younger minds are more pliable.. |
| 02:04.27 | cluecon[file] | indeed |
| 02:04.42 | CoaxD | baka03: It does, but only because all things are currently pointed to look at default |
| 02:04.57 | CoaxD | baka03: In iax2.conf, in any stanza that defines a connection, you tell it what context to default to |
| 02:04.58 | cluecon[file] | who wants to see a short video of me?!? |
| 02:05.08 | CoaxD | cluecon: Nobody, sir. |
| 02:05.12 | cluecon[file] | too bad |
| 02:05.49 | baka03 | ok i have run iax2 show registry in asterisk cli and i dont get anything in the list |
| 02:06.07 | baka03 | so im assuming im doing something wrong as far as the iax is concerned |
| 02:06.09 | cluecon[file] | http://file-radio.com/movies/file_upclose.mov |
| 02:06.22 | cluecon[file] | omg Becky! it's meeeeeee |
| 02:07.01 | cluecon[file] | me claiming at that |
| 02:07.08 | *** join/#asterisk omarc55 (~omarc55@adsl-146-66-29.mia.bellsouth.net) |
| 02:07.11 | baka03 | pbx*CLI> iax2 show registry |
| 02:07.11 | baka03 | Host Username Perceived Refresh State |
| 02:07.11 | baka03 | pbx*CLI> |
| 02:07.29 | cluecon[file] | baka03: iax2 show peers |
| 02:07.42 | baka03 | same |
| 02:07.51 | cluecon[file] | then you have no peers, welcome to the world |
| 02:07.53 | opus__ | reload |
| 02:08.34 | opus__ | i have the biggest problem with CLI output truncating columns, esp the Channel field |
| 02:08.51 | opus__ | i'm thinking about opening up a bug, but then it'all go on on about how it needs to be all xml |
| 02:09.18 | *** join/#asterisk lattice (~lattice@dsl092-015-074.sfo1.dsl.speakeasy.net) |
| 02:10.08 | baka03 | my iax.conf is http://www.wollongongastronomy.com/asterisk/iax.conf |
| 02:10.10 | omarc55 | hi all, I am trying to setup spandsp with a TE110P on asterisk but I can't get it to receive faxes, all I get is Executing RxFAX(...) in debug mode and thats it, what could I be doing wrong? |
| 02:10.50 | baka03 | i see now im getting this error when i reload: config.c:323 cfg_process: parse error: No category context for line 4 of /etc/asterisk/iax.conf |
| 02:11.11 | opus__ | you have context=category ? |
| 02:11.20 | opus__ | but no context actually called 'category' in extensions.conf |
| 02:11.33 | opus__ | change it to default |
| 02:11.36 | baka03 | ahh ok |
| 02:11.47 | opus__ | that will be one taco sir |
| 02:12.25 | CoaxD | opus: Be sure to specify hardshell or soft |
| 02:13.05 | opus__ | ok |
| 02:13.18 | twisted | yee haw |
| 02:13.24 | opus__ | one soft taco with a hot dog in the middle please |
| 02:13.25 | twisted | think i can git me a phone on this here thing? http://www.bumperdumper.com/ |
| 02:14.05 | baka03 | what is all this stuff in extenstions.conf |
| 02:14.07 | baka03 | dundi-e164-local |
| 02:14.50 | sigterm | twisted: lol interesting site |
| 02:14.54 | baka03 | opus__: im still getting that error cfg_process: parse error: No category context for line 4 of /etc/asterisk/iax.conf |
| 02:15.11 | twisted | sigterm, i tend to find the most bizzare stuff on the net |
| 02:15.17 | sigterm | indeed |
| 02:15.45 | *** join/#asterisk Redb3ard (~oylerj@c-24-125-77-17.hsd1.va.comcast.net) |
| 02:15.53 | opus__ | jhehe |
| 02:15.56 | sigterm | bumperdumper-RC1 = electronic flush mechanism with handle |
| 02:15.59 | opus__ | that guy thought he was going to be rich |
| 02:16.10 | opus__ | i'm going to crank call him and say I want to license it |
| 02:16.15 | opus__ | for airplanes or something |
| 02:16.17 | sigterm | any bets hunters buy that thing? |
| 02:16.30 | twisted | lol |
| 02:16.32 | Redb3ard | anyone in here ever messed with a cidco ip210 voip phone? |
| 02:17.00 | sigterm | lol i like the hitch extension with the toilet paper holder |
| 02:17.07 | opus__ | hehe |
| 02:17.28 | twisted | i guess the only thing you gotta worry about is pooping on your feet |
| 02:17.29 | sigterm | thats just funny stuff |
| 02:17.37 | twisted | maybe he should attach a mud flap to it |
| 02:17.43 | opus__ | haha |
| 02:17.47 | sigterm | hah |
| 02:18.05 | *** join/#asterisk santiago (~santiago@63.245.86.198) |
| 02:19.14 | twisted | what a sappy quit message |
| 02:19.36 | Redb3ard | You can't put love in a box - but you can wrap a corpse in an old rug |
| 02:19.38 | drumkilla | i'll be sure to make fun of him later :) |
| 02:20.15 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 02:21.55 | *** join/#asterisk orbi (~dantate@pcp08696782pcs.500ash01.tn.comcast.net) |
| 02:25.05 | *** join/#asterisk jo3sm1th (~Fullname@ppp-69-233-0-194.dsl.irvnca.pacbell.net) |
| 02:25.47 | jo3sm1th | How hard is it to make your DID play a msg in mp3 and then offer like 4 extensions all of which go to different landline phone#s |
| 02:26.11 | jo3sm1th | I have a DID nad i have asterisk set up to use with voicepulse but I always just make any incoming call go straight to my softphone |
| 02:26.30 | jo3sm1th | IS greg from nufone here?! |
| 02:26.38 | xeet2 | jo3sm1th: quite easy |
| 02:26.46 | jo3sm1th | I know |
| 02:26.53 | jo3sm1th | and I can make a simple mp3 file now |
| 02:27.10 | xeet2 | so why do you ask? =) |
| 02:27.21 | jo3sm1th | Because I dont have time to do it |
| 02:27.29 | Nugget | then pay someone to do it for you |
| 02:27.32 | jo3sm1th | Deal |
| 02:27.34 | jo3sm1th | Anyone here? |
| 02:27.49 | jo3sm1th | cut me some slack this is liek 30mins max. of work ONE mp3 file |
| 02:28.01 | xeet2 | how much are you paying? |
| 02:28.20 | baka03 | is anyone here using freshtel? |
| 02:28.22 | jo3sm1th | Wheres a website that can make me a quick mp3 file i saw some crude thing at ATT once that would work |
| 02:28.26 | jo3sm1th | xeet2 how much do you want |
| 02:28.54 | xeet2 | jo3sm1th: what do you have in mind? |
| 02:29.23 | jo3sm1th | Greg from nufone is what i have in mind |
| 02:29.28 | jo3sm1th | is he here? |
| 02:29.52 | xeet2 | you think you'd be able to get greg to do some mp3 work for you? hahaha |
| 02:29.58 | cluecon[file] | jo3sm1th: nope |
| 02:30.04 | xeet2 | he's got way too many other things making him alot more money =) |
| 02:30.52 | jo3sm1th | Anyone else besies xeet here? |
| 02:31.35 | xeet2 | so, I'm excluded because I'm asking you how much you'd pay? |
| 02:35.27 | jo3sm1th | I'm paying 75 per hour... billed in half our increments |
| 02:37.05 | cluecon[file] | jo3sm1th: what do you want? |
| 02:37.33 | xeet2 | jo3sm1th: http://thevoice.digium.com |
| 02:37.46 | cluecon[file] | Allison rocks |
| 02:38.10 | xeet2 | jo3sm1th: you can get what you need there, and it will actually sound good, for alot less than 75$, if you can wait 24 hours for it |
| 02:38.45 | jo3sm1th | Allison the voice? |
| 02:38.48 | jo3sm1th | I already got a short mp3 file |
| 02:39.23 | cluecon[file] | jo3sm1th: but you want the dialplan logic and stuff to make it happen like you said above? |
| 02:39.34 | xeet2 | so, I'm confused as to what you actually want then, you want the dialplan stuff? |
| 02:39.47 | *** join/#asterisk ZX81 (matt@222-153-114-70.jetstream.xtra.co.nz) |
| 02:40.02 | cluecon[file] | ZX81!!!!!!!! |
| 02:40.06 | *** part/#asterisk ZX81 (matt@222-153-114-70.jetstream.xtra.co.nz) |
| 02:40.09 | cluecon[file] | gah |
| 02:40.15 | JunK-Y | u scared him! |
| 02:40.20 | xeet2 | hehe |
| 02:40.21 | cluecon[file] | yup |
| 02:40.36 | *** join/#asterisk ZX81 (matt@222-153-114-70.jetstream.xtra.co.nz) |
| 02:40.36 | JunK-Y | ~cluecon |
| 02:40.37 | jbot | it has been said that cluecon is http://www.cluecon.com |
| 02:40.45 | ZX81 | hi people |
| 02:40.48 | cluecon[file] | hey |
| 02:40.50 | ZX81 | only here for a sec |
| 02:40.56 | ZX81 | :) |
| 02:41.01 | ZX81 | how are you file? |
| 02:41.07 | cluecon[file] | how are you my dear? |
| 02:41.11 | ZX81 | kewl |
| 02:41.16 | ZX81 | yah good |
| 02:41.21 | ZX81 | a little stressed |
| 02:41.26 | ZX81 | getting reading for astricon |
| 02:41.28 | ZX81 | :D |
| 02:41.36 | cluecon[file] | ah |
| 02:41.45 | ZX81 | you know what getvar is in new cvs for agi? |
| 02:41.52 | ZX81 | has it changed? |
| 02:41.57 | ZX81 | or is still get variable? |
| 02:42.11 | ZX81 | just wondering cos of the change to dialplan app |
| 02:42.16 | ZX81 | :) |
| 02:42.17 | ZX81 | ok |
| 02:42.20 | drumkilla | cluecon[file] uses chickens |
| 02:42.28 | ZX81 | hehe |
| 02:42.45 | ZX81 | are the agi commands defined in each individual src file? |
| 02:42.53 | ZX81 | agi_register? |
| 02:43.11 | JunK-Y | res_agi.c |
| 02:43.18 | ZX81 | ok |
| 02:43.19 | ZX81 | ty |
| 02:43.23 | cluecon[file] | DUH |
| 02:43.24 | JunK-Y | its 5$. |
| 02:44.06 | ZX81 | :) |
| 02:44.22 | AgiNamu | Anyone interested in an E911 solution/ |
| 02:44.32 | JunK-Y | ZX81: what are u presenting at astricon? |
| 02:44.37 | ZX81 | C# vs * |
| 02:44.39 | ZX81 | :) |
| 02:44.45 | ZX81 | and predictive algorythims |
| 02:44.46 | ZX81 | etc |
| 02:44.52 | ZX81 | fuzzy logic too |
| 02:44.53 | ZX81 | :) |
| 02:44.55 | ZX81 | tutorial |
| 02:44.57 | ZX81 | 2 hours |
| 02:44.57 | AgiNamu | C# and *? :) |
| 02:44.57 | ZX81 | :) |
| 02:45.12 | AgiNamu | explain how to fix threading issues between Mono and asteirsk, and you earn a lot of friends |
| 02:45.24 | JunK-Y | after the presentation, msg me the link for the pdf, i'll take a look. |
| 02:45.35 | xeet2 | AgiNamu: whats this about a e911 solution |
| 02:45.36 | ZX81 | JunK-Y: has video too |
| 02:45.37 | ZX81 | :) |
| 02:45.52 | ZX81 | AgiNamu: yah, you providing e911 |
| 02:45.56 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 02:45.59 | ZX81 | WilliamK: I think it sux |
| 02:46.06 | cluecon[file] | ZX81: you're very fuzzy |
| 02:46.14 | heison | ~seen jerjer |
| 02:46.28 | jbot | jerjer is currently on #asterisk (8h 53m 22s). Has said a total of 6 messages. Is idling for 5h 14m 56s |
| 02:46.28 | ZX81 | WilliamK: I'm trying to get people to send prior art for him |
| 02:46.28 | ZX81 | :) |
| 02:46.32 | AgiNamu | www.dash911.com |
| 02:46.36 | AgiNamu | I'm welcoming feedback. |
| 02:46.39 | WilliamK | somehow, I think Nortel could probably beat them on their voicemail claim |
| 02:46.46 | xeet2 | ah yes, I believe I got an e-mail about this |
| 02:46.47 | AgiNamu | I want to announce to Asterisk-biz, just wanted to see what ppl think first. |
| 02:46.59 | ZX81 | WilliamK: as could others |
| 02:47.44 | PyroSteve | the callerid option in my sip.conf wont override the sip clients callerid info |
| 02:48.07 | PyroSteve | what else do I have to ignore the sip clients caller id string ? |
| 02:48.14 | ZX81 | OT: anyone know how to send keepalive on irssi for irc? |
| 02:48.28 | ZX81 | WilliamK: I have posted a news article |
| 02:48.39 | WilliamK | dang merdian originals are pre-1995 equipment |
| 02:49.46 | ZX81 | yep |
| 02:50.00 | ZX81 | and email filters?! |
| 02:50.05 | PyroSteve | hey gyes |
| 02:50.08 | PyroSteve | guys |
| 02:50.08 | ZX81 | hi |
| 02:50.14 | ZX81 | :) |
| 02:50.27 | PyroSteve | what options cause asterisk to ignore a sip clients caller id info |
| 02:50.45 | Juggie | zapata.conf |
| 02:50.50 | Juggie | callerid=??? |
| 02:50.52 | Juggie | look it up |
| 02:50.56 | PyroSteve | well I did |
| 02:51.01 | PyroSteve | in my sip.conf |
| 02:51.04 | ZX81 | or sip.conf |
| 02:51.06 | Juggie | no |
| 02:51.06 | ZX81 | :) |
| 02:51.14 | Juggie | look there :) |
| 02:51.20 | PyroSteve | I have the callerid="name" <###-###-####> |
| 02:51.28 | Juggie | right, that should override anything a user might send |
| 02:51.40 | PyroSteve | but the sip client is still able to provide his own number |
| 02:51.46 | ZX81 | :) |
| 02:51.55 | ZX81 | but it takes the name? |
| 02:52.04 | PyroSteve | umm dont know |
| 02:52.23 | ZX81 | NoOp(${CALLERID}) |
| 02:52.25 | ZX81 | :) |
| 02:52.29 | Juggie | callerid= if set properly in sip.conf |
| 02:52.37 | Juggie | will override whatever a sip phone might send |
| 02:55.42 | *** join/#asterisk dial_911_now (~niggerplz@66-55-197-254.gwi.net) |
| 02:56.29 | *** join/#asterisk remmo (~rem@smack.isp.net.au) |
| 02:59.03 | ZX81 | ~ping |
| 02:59.04 | jbot | pong |
| 02:59.10 | ZX81 | ~pang |
| 02:59.14 | AgiNamu | blitzrage, i hear ya! |
| 02:59.31 | AgiNamu | I hate getting messages saying "What's an smime.p7m file??" |
| 03:00.35 | ZX81 | I hate getting messages saying "What's a hardcorePr0n.mpg file??" |
| 03:00.39 | ZX81 | :) |
| 03:01.06 | AgiNamu | isn't it pr0n.mpg.exe? |
| 03:01.13 | ZX81 | :) |
| 03:01.14 | ZX81 | heh |
| 03:01.23 | *** join/#asterisk REdOG (~REdOG@REdOG.user.gentoo) |
| 03:01.50 | REdOG | anyone here familliar with iaxcomm have some time to try and help me? |
| 03:05.00 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 03:09.53 | JerJer | does anyone have a good apt server for Rh9? |
| 03:10.06 | *** join/#asterisk cool2nigh (~cjs@adsl-68-75-180-59.dsl.emhril.ameritech.net) |
| 03:10.11 | cool2nigh | hello |
| 03:10.35 | cool2nigh | i was wondering if it was possible to run asterisk on a dedicated server? |
| 03:10.50 | JerJer | why wouldn't it be possible? |
| 03:11.30 | ZX81 | Why would a get variable return a result of 1, a code of 200, but no data? (phpagi) |
| 03:12.01 | *** join/#asterisk doolph (doolph@201.226.146.178) |
| 03:12.02 | ZX81 | (using exten=>889,1,Set(test=12345) |
| 03:12.31 | ZX81 | HEAD |
| 03:12.32 | cool2nigh | jerjer- im not sure. would i need a ata. like i used for vonage or broadvoice. or would it interface directly with the voip provider through the internet |
| 03:12.42 | PyroSteve | why would one not want to use asterisk@home ? |
| 03:13.08 | ZX81 | PyroSteve: I think people should learn asterisk configs 1st then use a@h |
| 03:13.13 | ZX81 | but that's just me |
| 03:13.19 | dr123 | here is a reason has anyone gotten asterisk@home to working with SIP call Parking!!! |
| 03:13.29 | ZX81 | I have to deal with customers using a@h who know nothing about * :D |
| 03:13.55 | PyroSteve | ZX81: and how does that work out ? |
| 03:14.02 | ZX81 | PyroSteve: not so well |
| 03:14.03 | ZX81 | :) |
| 03:14.15 | ZX81 | I pass them on to the resident a&h guru |
| 03:14.19 | ZX81 | well offsite |
| 03:14.33 | ZX81 | I really should learn it's configs and setup |
| 03:14.34 | ZX81 | :) |
| 03:15.08 | dr123 | Hey Does anyone know how you would make asterisk detect and ring an extension when it detects Call Waiting from an FXO (PSTN) line / card... such that on whatever phone you are on currently with the FXO you would hear the standard call waiting beep |
| 03:15.24 | dr123 | Does that question make sense |
| 03:15.38 | cool2nigh | jerjer= ??? |
| 03:16.12 | CoaxD | Bwahahahaha |
| 03:16.22 | CoaxD | err. not irc safe. forget it |
| 03:17.04 | ZX81 | could somone have a look at: |
| 03:17.06 | ZX81 | http://pastebin.ca/13073 |
| 03:17.08 | JerJer | cool2nigh: ok and the problem is? |
| 03:17.11 | ZX81 | simple phpagi |
| 03:18.01 | JerJer | ZX81: where does get_var get defined? |
| 03:18.21 | ZX81 | in phpagi |
| 03:18.39 | ZX81 | $res=$this->agi_exec("GET VARIABLE $var"); |
| 03:18.45 | ZX81 | then returns res |
| 03:18.53 | cool2nigh | jerjer- weather it's possible to install asterisk on a dedicated server. Eg. off the shelf rack running redhat, and installing asterisk on it. Is it possible? |
| 03:19.30 | ZX81 | cool2nigh: yes I have asterisk running on multiple 1U rack units |
| 03:19.38 | ZX81 | some using Fedora Core |
| 03:19.40 | ZX81 | bleh |
| 03:20.02 | REdOG | Im overwhelmed, we have 3 * servers and I really have no clue how to use them. Where might I begin without trashing stuff and pissing my coworkers off? |
| 03:20.12 | xeet2 | dr123: you shouldn't use call waiting on pots lines that go into a pbx |
| 03:20.31 | ZX81 | and then in res_agi.c: Usage: GET VARIABLE <variablename> |
| 03:20.35 | dr123 | well... it cant turn it off |
| 03:20.39 | dr123 | so i guess i just wont use it |
| 03:20.40 | cool2nigh | zx- thanks.. may i ask what voip provider you use? i was looking at iax.org they seem pretty good |
| 03:20.55 | dr123 | I use Packet8 for my PSTN |
| 03:21.00 | dr123 | and it comes with Call waiting |
| 03:21.05 | ZX81 | cool2nigh: nufone.net |
| 03:21.09 | ZX81 | + about 20 others |
| 03:21.14 | ZX81 | :) |
| 03:21.30 | xeet2 | dr123: you should be able to turn it off with a *xx number on your local telco |
| 03:21.47 | xeet2 | cool2nigh: there are almost 100 startup * based itsps |
| 03:21.49 | dr123 | this is true... i did not think of that |
| 03:22.16 | xeet2 | and only about 1/4th of them will make it past the first 6 months |
| 03:22.48 | cool2nigh | yeah |
| 03:22.59 | ZX81 | xeet2: 1/4th is pretty generous :) |
| 03:23.03 | dr123 | hey does anyone have the Cisco 6.x (6.3 SIP) firware |
| 03:23.05 | dr123 | firmware |
| 03:23.06 | xeet2 | zx: maybe so |
| 03:23.10 | ZX81 | :) |
| 03:23.16 | *** join/#asterisk Newbie___ (me@211.24.146.11) |
| 03:23.44 | xeet2 | zx81: there's no howto yet |
| 03:23.47 | dr123 | or a place other than ebay to get some Cisco 7960's |
| 03:23.54 | dr123 | I got one from ebay for 220 |
| 03:23.57 | dr123 | not bad |
| 03:24.22 | xeet2 | dr123: you have to buy the license for the code from cisco |
| 03:24.33 | dr123 | i understand that |
| 03:24.56 | xeet2 | I can sell you some but not for 220 =) |
| 03:25.04 | xeet2 | with 7.3 |
| 03:25.31 | dr123 | I heard that 7 was a little buggy and that 6 was preferred is that incorrect |
| 03:25.43 | xeet2 | depends on what bugs you want =) |
| 03:26.01 | dr123 | Do you use SIP w/ 7.3 and it works fine? |
| 03:26.19 | dr123 | also can you upload custom ringers to the Cisco 7960's or no? |
| 03:26.25 | xeet2 | I've had 6.3 crash daily for a few people and 7.3 fixed it, whatever they were doing that made it happen |
| 03:26.49 | xeet2 | but I've also had 7.3 crash a few times |
| 03:26.58 | dr123 | awsome... |
| 03:27.28 | Newbie___ | hi anyone know what is this ?? codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes at the end) |
| 03:27.33 | xeet2 | but in general 7 has been more stable than 6, for me |
| 03:27.43 | *** join/#asterisk CdtDelta_PM (~CdtDelta_@dsl081-225-161.chi1.dsl.speakeasy.net) |
| 03:29.23 | dr123 | Here is what I want to do with the Cisco since it has 6 lines (I understand this would be mostly Configured in the extension.conf) but I want the last time to display the Caller ID of any incoming Calls from the PSTN (packet8 telco) that is not hard but I want it to display them without ringing... like if I want to know who is calling but they are not calling me just calling from off network... is there a way to turn off the ringer on the c |
| 03:30.15 | AlexCeli | I have the firmware 6.3 Cisco 7960 and work fine for me.. |
| 03:30.33 | xeet2 | dr123: you can set a different ringer in your sip messages |
| 03:30.55 | dr123 | and just set it to null or something |
| 03:31.13 | xeet2 | you might have to go as far as making a mute ring tone and use that |
| 03:31.25 | AlexCeli | I can't install the firmware 7.4, the phone ask for a missing file |
| 03:31.59 | Redb3ard | i have some cheap knockoff voip phone that doesnt even do sip |
| 03:32.04 | xeet2 | AlexCeli: 7 runs things a bit differently, you have a boot loader and then an image, as opposed to just an image |
| 03:32.07 | Redb3ard | not without a firmware upgrade that i cant find |
| 03:32.20 | wankel | alex: i had to ugprade to the latest skinny image, then to the latest sip image. |
| 03:32.30 | wankel | the old skinny image won't upgrade direct to 7.x sip |
| 03:32.51 | AlexCeli | xeet2: and where i can find the boot loader? |
| 03:33.19 | xeet2 | AlexCeli: its included in the file, you just have to change your config |
| 03:33.21 | xeet2 | hang on |
| 03:33.29 | AlexCeli | xeet2: i got a lot of error messages... |
| 03:34.10 | dr123 | by the way i couldnt find much on this... what is the difference between the 7960 and 7960G |
| 03:34.14 | dr123 | are their any really? |
| 03:34.33 | xeet2 | alexceli: OS79XX.TXT has to be "P003-07-X-00" |
| 03:34.46 | xeet2 | and you need a .bin, .sbn, .loads, and a .sb2 |
| 03:35.00 | baka03 | is it possible to use an ATA as an outbound PSTN line for asterisk? |
| 03:35.14 | xeet2 | baka: yes |
| 03:35.28 | baka03 | do you know where there is some good info on this? |
| 03:35.29 | AlexCeli | xeet2: yeah the file is with this info.. |
| 03:35.35 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 03:35.53 | AlexCeli | let me find the error... |
| 03:35.59 | baka03 | or even if it has a specific name |
| 03:36.00 | xeet2 | baka: not really, it just works |
| 03:36.40 | baka03 | so i plug an ATA into the asterisk server and have a phone line in the ATA |
| 03:36.43 | loud | or P0S3-07-3-00 |
| 03:36.50 | loud | thats what i have on my tftp |
| 03:37.06 | xeet2 | really? cisco docs say it has to be P00 for the boot image |
| 03:37.50 | loud | he will find out |
| 03:37.52 | loud | :> |
| 03:38.14 | loud | seems that is bad packaged, the .bin |
| 03:38.35 | AlexCeli | xeet2: the phone ask me for "P0S3-07-4-00.sbn" |
| 03:38.38 | Redb3ard | guys, if this phone claims it needs an h.323 gateway, what are the chances asterisk can fit that bill? |
| 03:39.01 | xeet2 | alex: and what does your OS79XX.txt say? |
| 03:39.08 | loud | that file exists |
| 03:39.13 | loud | when you decompress the .zip |
| 03:39.21 | AlexCeli | xeet2: in the cisco files, i can only find P0O3-07-4-00.sbn |
| 03:39.21 | dr123 | mine says p003-07-4-00 |
| 03:39.22 | xeet2 | loud: does it? not for me |
| 03:39.33 | dr123 | err P003-07-4-00 |
| 03:39.50 | xeet2 | alexceli: what is the exact output of os79xx.txt? |
| 03:39.51 | AlexCeli | xeet2: i rename the P0O3-07-4-00.sbn to P0S3-07-4-00.sbn and don't work... |
| 03:40.01 | xeet2 | yeah, 7 works differently, you can't do that |
| 03:40.10 | xeet2 | and its 0 not o |
| 03:40.41 | AlexCeli | xeet2: [root@alex][/home/alex/central/P0S3-07-4-00> cat OS79XX.TXT |
| 03:40.41 | AlexCeli | P003-07-4-00 |
| 03:41.08 | xeet2 | and P003-07-4-00.sbn and .bin don't exist? |
| 03:41.10 | AlexCeli | xeet2: this is in the default distro |
| 03:41.28 | AlexCeli | i change to P0S3-07-4-00 |
| 03:41.40 | xeet2 | hang on, going to paste you something |
| 03:41.54 | AlexCeli | xeet2: and even the phone want the P0S3-07-4-00.sbn |
| 03:44.41 | dr123 | hey do you have any idea why parking would not work in sip it works with IAX but with SIP it puts the caller on hold but never reads the digits of the parking space so you dont know where it is parked |
| 03:45.34 | *** join/#asterisk Ahewes (~rsb@adsl-69-107-50-127.dsl.pltn13.pacbell.net) |
| 03:45.43 | xeet2 | dr123: which parking app are you using? |
| 03:45.56 | xeet2 | dr123: the new one in res_features fixes that problem |
| 03:47.18 | dr123 | res_features... i mean i edited the parking info in features.conf |
| 03:47.30 | dr123 | what is this other one? |
| 03:50.00 | xeet2 | ok, when you do a show dialplan parkedcalls, do you see anything in there? |
| 03:50.54 | dr123 | yes |
| 03:51.01 | baka03 | can someone explain how you use an ATA to dial out to the standard telephone line in asterisk |
| 03:51.08 | dr123 | Include => Parkedcalls [pbx_config |
| 03:51.09 | dr123 | and |
| 03:51.32 | dr123 | 70 => 1. Park() [res_features] |
| 03:51.52 | xeet2 | dr123: so, when you want to park a call, are you including parkedcalls into that context and using 70? or do you have a separate extension configured to park a call? |
| 03:52.16 | Corydon76-home | baka03: you put a Dial line which dials the external line in the starting context for which the ATA is connected to Asterisk |
| 03:52.32 | dr123 | in my dial plan i have a context called [parkedcalls] and under that it says include => parkedcalls |
| 03:52.35 | dr123 | that is it |
| 03:52.47 | xeet2 | hmmm |
| 03:52.50 | xeet2 | it should work then |
| 03:53.15 | Corydon76-home | baka03: that explanation is technically correct, confusing as hell, and illustrates that Asterisk is fairly complex |
| 03:53.32 | baka03 | Corydon76-home: :) |
| 03:53.41 | *** join/#asterisk mud (~mud@CPE0050fc54ad50-CM0011ae8fd786.cpe.net.cable.rogers.com) |
| 03:53.57 | SwK | Corydon76-home if you didnt say asterisk there I would have sworn you were talking microsoft |
| 03:54.13 | Corydon76-home | baka03: if you understood what I was talking about, you wouldn't have asked the question, would you? |
| 03:54.29 | baka03 | Corydon76-home: i think i understand where your going |
| 03:54.46 | baka03 | Much reading to be done |
| 03:55.02 | Corydon76-home | so make sure you have a type=user configured in sip.conf corresponding to your ATA. Note the context in that config. |
| 03:55.12 | xeet2 | dr123: are you testing this with another call holding or just calling 70? |
| 03:55.16 | SwK | Asterisk: King of the Steep learning Curve |
| 03:55.30 | xeet2 | swk: how true |
| 03:55.30 | Corydon76-home | Then set up a dialplan in extensions.conf for that same context |
| 03:55.49 | baka03 | ok |
| 03:56.05 | dr123 | yeah another call holding |
| 03:56.19 | xeet2 | dr123: are you doing a blind transfer or an attended transfer? |
| 03:56.25 | dr123 | like i call exten 1101 (my computer - SJPhone, from 1100 - Grandstream 100) |
| 03:56.28 | Corydon76-home | Assuming you're in North America (NANP), you might do exten => _9NXXXXXX,1,Dial(Zap/1/${EXTEN:1}) |
| 03:56.46 | dr123 | I am assuming it is attended... i press Transfer then 70 then send |
| 03:56.51 | Corydon76-home | Assuming your phone line is channel 1. |
| 03:57.00 | dr123 | and it puts the call into parking lot but does not playback the parking space |
| 03:57.02 | SwK | or you just might do exten => 666,1,Dial(Zap/1/911) |
| 03:57.20 | xeet2 | dr123: when you press send, does it immediatly disconnect you and put the call into the space? |
| 03:58.04 | Corydon76-home | SwK: that should be 616, right? |
| 03:58.20 | SwK | 616? |
| 03:58.20 | Juggie | http://code.google.com/summerofcode.html |
| 03:58.28 | Corydon76-home | SwK: the real mark of the beast |
| 03:58.32 | SwK | heh |
| 03:58.33 | baka03 | Corydon76-home: and i can get a few ATA's connected to seperate lines and just add them in the same fashion? |
| 03:58.41 | Corydon76-home | SwK: you haven't paid attention to the news, have you? |
| 03:58.51 | SwK | the media bored me |
| 03:58.58 | xeet2 | baka: if you're doing alot of voip, you really should use a service that you can use * with directly |
| 03:59.21 | Corydon76-home | SwK: someone found an old script that found that Revelations was copied incorrectly... the mark of the beast is not 666, but 616. |
| 03:59.30 | xeet2 | dr123: when you press send, does it immediatly disconnect you and put the call into the space? |
| 03:59.30 | SwK | ha |
| 03:59.31 | SwK | nice |
| 03:59.43 | Corydon76-home | baka03: yep, different configs for each |
| 03:59.52 | dr123 | i dont know it does something weird like it is disconnected in the sense that audio is not passing either way but the call is still kind of connected becasue you have to hang up |
| 04:00.03 | Corydon76-home | SwK: ironically, 616 is Ann Arbor, MI |
| 04:00.22 | SwK | hah |
| 04:00.25 | xeet2 | dr123: that doesn't sound like an attended transfer |
| 04:00.25 | dr123 | but after is press 70 then send the other side of the call is fully parked and can be pickup no problem assuming you are looking at the CLI and know wher eit put it based on that beucase it doesnt say it back on the phone that did the parking |
| 04:00.31 | dr123 | does that make snese |
| 04:00.41 | baka03 | Corydon76-home: So then i wont need to plug a modem(s) into the asterisk box as it can use the ATAs |
| 04:00.44 | SwK | Corydon76-home it was also one helluva club in memphis |
| 04:00.55 | xeet2 | dr123: not really but it doesn't sound like an attended transfer. you have to do an attended transfer to use call parking and sip |
| 04:01.23 | jo3sm1th | How do you use A(x) with Dial |
| 04:01.34 | jo3sm1th | play an announcement to the called party, using x as file |
| 04:01.35 | Corydon76-home | baka03: I wouldn't plug a modem into the Asterisk box, anyway... unless you have a channel bank |
| 04:02.30 | Corydon76-home | TDM400P doesn't support modems/faxes |
| 04:02.31 | xeet2 | dr123: try just calling 70, see if you get a parking space spoken to you |
| 04:03.00 | xeet2 | Corydon76-home: really? I've used them with fax machines before |
| 04:03.17 | Corydon76-home | xeet2: according to Digium, it's an unsupported configuration |
| 04:03.21 | xeet2 | hmmm |
| 04:03.30 | xeet2 | that might explain some issues |
| 04:03.31 | Corydon76-home | xeet2: and it doesn't work with latest CVS |
| 04:03.47 | baka03 | Corydon76-home: my plan is to have 2-3 normal phone lines available to asterisk |
| 04:04.03 | baka03 | + the VoIP |
| 04:04.10 | xeet2 | whats the limitation with that? just that the cards can't handle the amount of traffic fax/modem calls would pass? |
| 04:04.26 | Corydon76-home | baka03: then you'll probably want to group them in zapata.conf and use something like Zap/g1 to select the first available analog line |
| 04:04.38 | Corydon76-home | assuming group=1 in zapata.conf |
| 04:04.59 | baka03 | Corydon76-home: is it possible to group the ATA's like that? |
| 04:05.36 | Corydon76-home | Uh, well, you don't do it that way |
| 04:05.52 | Corydon76-home | You'd do Dial(SIP/11&SIP/12&SIP/13) |
| 04:06.01 | Corydon76-home | and whichever one picked up first gets the call |
| 04:06.09 | xeet2 | uhm |
| 04:06.28 | xeet2 | hmm |
| 04:06.32 | dr123 | ok try that now |
| 04:06.58 | dr123 | if I dial 70 .. it says 71...so it does do that correctly |
| 04:07.03 | baka03 | ok, so its possible to do what im asking, have multiple ATA's plugged into phone lines and have asterisk use the ATA's for dialing out. |
| 04:07.32 | Corydon76-home | Uh, well, maybe |
| 04:07.37 | xeet2 | ata's plugged into phone lines? |
| 04:07.42 | xeet2 | thats more of a voip gateway |
| 04:07.45 | xeet2 | not an ata |
| 04:08.07 | Corydon76-home | Yeah, ATAs are FXS devices, I think |
| 04:08.11 | Corydon76-home | not FXO devices |
| 04:08.12 | xeet2 | normally |
| 04:08.16 | dr123 | so how do i make it do the attended transfer |
| 04:08.35 | xeet2 | an fxo "ata" is a gateway, and is normally about 2-300$ |
| 04:08.44 | xeet2 | and most of them suck. |
| 04:09.00 | xeet2 | multitech isn't bad if they would fix their caller id problems |
| 04:09.09 | xeet2 | but at 3k for 8 ports, its no bargain |
| 04:09.27 | xeet2 | dr123: what kind of phone is it? |
| 04:09.37 | baka03 | so the only normal method of connecting std lines into asterisk is with a FXO pci card? |
| 04:10.02 | xeet2 | baka: thats the cheapest if you don't mind some echo issues |
| 04:10.26 | xeet2 | which won't always be the case, just most of the time |
| 04:10.43 | dr123 | Grandstream BudgetTone 100 but the Cisco as mentioned before arrieves tomorrow |
| 04:11.21 | xeet2 | dr123: mmm, dunno... did you try a <flash>70<wait for parking space number><flash> ? |
| 04:11.35 | Corydon76-home | baka03: or a channel bank and a TE110P |
| 04:11.46 | doolph | anyone got fixed the duration/billsec error using zaptel device (X100P) ?? |
| 04:11.49 | Corydon76-home | Or T100P if you can still find one |
| 04:12.01 | baka03 | hrm, how do people normally use asterisk to make calls over the PSTN |
| 04:12.04 | dr123 | trying now |
| 04:12.16 | xeet2 | does the te110p do anything better than the t100p, other than handle an e1? |
| 04:12.16 | Corydon76-home | baka03: I use a PRI |
| 04:12.31 | *** join/#asterisk matobago (~matobago@einstein.transtelco.com.mx) |
| 04:12.43 | Corydon76-home | xeet2: I think there's an improvement in the firmware, also |
| 04:12.46 | doolph | anyone got fixed the duration/billsec error in mysql using zaptel device (X100P) ?? |
| 04:12.48 | xeet2 | hmmm |
| 04:12.49 | dr123 | no that didnt work |
| 04:12.58 | Corydon76-home | doolph: what error? |
| 04:13.08 | xeet2 | is digium offering any kind of trade-up? |
| 04:13.18 | matobago | hi everyone |
| 04:13.27 | dr123 | i was just reading about these blind / attend transfer I know that I need to do an attended transfer is there a way that you say like the transfer using the transfer button is ALWAYS attended |
| 04:13.37 | baka03 | using ISDN lines |
| 04:13.46 | Corydon76-home | xeet2: don't think so, it's $100 more anyway |
| 04:13.50 | xeet2 | dr123: you'd have to look at grandstream for that |
| 04:13.59 | matobago | does anyone know why i get this message when i'm tring to transfer a recived call with # |
| 04:14.01 | matobago | Got SIP response 400 "Bad Request" back from 200.52.114.47 |
| 04:14.11 | xeet2 | well I mean, 100$ plus a t100p card getting a te110p card |
| 04:14.26 | doolph | <Corydon76-home> doolph: what error? <-- the calllog says that the call is answered even when it is not, it start to log when ring |
| 04:14.45 | Corydon76-home | doolph: that's not an error... that's due to the fact that you're using an analog line |
| 04:14.57 | Corydon76-home | and there's no true callprogress on an analog line |
| 04:15.45 | xeet2 | ah well, been running fine on the t100p's for 2 years now |
| 04:16.10 | Corydon76-home | doolph: it's not an error, and it's never going to change |
| 04:16.45 | SwK | anyone use poly's? |
| 04:17.02 | matobago | anyone? |
| 04:17.26 | SwK | damned things are sending 603's after 60 secs and I cant seem to find the swich/knob to adjust that |
| 04:18.00 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
| 04:24.48 | elric | does anyone know of a good guide to setup voicemail with asterisk? |
| 04:26.32 | baka03 | hrm, i have 2 telephone lines and a VoIP service. I want to have asterisk control the use of these lines so i can use any phone and use any service. Whats the best method of connecting both these standard POTS lines to asterisk, from what i read is i only have the the X100P as an option? |
| 04:27.53 | Corydon76-home | That's not completely true, you also have the FXO boards on the TDM400P |
| 04:27.57 | *** join/#asterisk dariob (~tom@c6476113-171.impsat.com.co) |
| 04:28.07 | Corydon76-home | Just use it only for voice |
| 04:28.49 | *** join/#asterisk loud (~ariel@datamarkets.com.mx) |
| 04:29.04 | baka03 | TDM04B - 4-port FXO bundle |
| 04:29.29 | Corydon76-home | Yeah, but the FXO port is still not recommended in port 1 |
| 04:29.59 | baka03 | so the x100p has echo problems? |
| 04:30.14 | Corydon76-home | Anything analog is going to have echo problems |
| 04:31.54 | *** join/#asterisk Mavvie (edwin@dialin-42.barnet.com.au) |
| 04:32.24 | baka03 | will the x100p work in australia? |
| 04:32.56 | SwK | the x100p should work in .au and .nz |
| 04:33.13 | Corydon76-home | Should work, yes. Approved to work? Dunno |
| 04:33.14 | SwK | now if its approved or not is another question |
| 04:33.16 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
| 04:34.03 | SwK | now why do i have to fight the urge to make nasty comments about down under? |
| 04:35.13 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 04:36.01 | baka03 | this any good? http://cgi.ebay.com.au/ws/eBayISAPI.dll?ViewItem&category=70811&item=5777703353&rd=1 |
| 04:37.05 | SwK | baka03 thats just another close x100p |
| 04:37.07 | SwK | err clone |
| 04:37.47 | SwK | now will it work, probably, will you have far end hangup detection issues, most likely |
| 04:38.01 | SwK | i havent seen a clone that did that properly |
| 04:38.26 | baka03 | what is the diff between x100p and x101p |
| 04:38.55 | Corydon76-home | Later chip revision |
| 04:39.06 | Corydon76-home | plus the x101p has a heatsink on the chip |
| 04:39.12 | loud | and you are contributing to the asterisk project. |
| 04:40.01 | loud | hey, mark likes bitches and alcohol too. |
| 04:40.32 | orbi | Corydon. |
| 04:41.06 | Corydon76-home | Mark doesn't drink that much |
| 04:41.18 | Corydon76-home | I've never seen him drunk |
| 04:41.20 | Juggie | send him red bull |
| 04:41.23 | Corydon76-home | Hey, orbi |
| 04:41.27 | Corydon76-home | LTNS |
| 04:43.19 | SwK | does the heatsink on the 101 actually do anything other then cover the chip numbers? |
| 04:43.47 | Sedorox | Umm... help spread the heat? |
| 04:43.53 | SwK | hah |
| 04:44.09 | SwK | yeah thats it |
| 04:44.22 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:45.12 | baka03 | http://www.voip-info.org/wiki-Asterisk+hardware+home+analog Option 2 |
| 04:45.35 | baka03 | doesnt that allow me to connect an ATA to the POTS line and use it in asterisk? |
| 04:46.04 | Sedorox | yes |
| 04:47.25 | *** join/#asterisk techie (gus@antibala.com) |
| 04:49.21 | baka03 | does it mean asterisk can use the POTS line or just the phone plugged into the ATA can use it? as in the phone on the ATA can use the POTS line and the Asterisk connection. |
| 04:49.46 | baka03 | or can any SIP phone connected to asterisk use that POTS line |
| 04:51.11 | *** join/#asterisk trig_hm (~jb@home.monkeypr0n.org) |
| 04:52.30 | lichen | HA, i'm wondering the same thing |
| 04:52.36 | lichen | er oops |
| 05:01.19 | dudes | anyone ever got this error when compiling zap /usr/src/zaptel/zaptel.c:168: warning: `fcstab' defined but not used |
| 05:01.35 | bewest | it's not an error |
| 05:01.45 | bewest | that would not stop compilation |
| 05:01.57 | bewest | it's harmless |
| 05:02.24 | dudes | after I make install it says FATAL: Module zaptel not found. |
| 05:02.39 | bewest | ok, so it didn't compile |
| 05:02.44 | bewest | but not because of that warning |
| 05:02.44 | dudes | it did |
| 05:03.02 | bewest | it compiled and installed? |
| 05:03.06 | dudes | it finishes compiling with no more erros |
| 05:03.23 | dudes | [ -f /etc/zaptel.conf ] || install -D -m 644 zaptel.conf.sample /etc/zaptel.conf |
| 05:08.21 | bewest | that's just the conf file though |
| 05:08.27 | bewest | not the module |
| 05:08.46 | bewest | you see zaptel.o in the directory? |
| 05:09.16 | dudes | I'm just in awe that I can't modprobe, but the module is installed in /lib/modules/(kernel ver) |
| 05:09.45 | bewest | I think I had a similar problem once |
| 05:09.52 | bewest | I don't remember what I did |
| 05:12.17 | dudes | it went in the wrong kernel dir |
| 05:12.17 | dudes | opps ... damn I need sleep |
| 05:12.37 | bewest | hehehe |
| 05:14.02 | Corydon76-home | Hmmm, maybe you need to do 'depmod -a' ? |
| 05:18.40 | dudes | for some reason there was a 2.6.8 in /lib/modules ... it was suppose to use 2.6.8-smp |
| 05:18.59 | dudes | probally the base debian kernel |
| 05:20.21 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 05:33.16 | dudes | TE410 cards are a pain in the .... |
| 05:33.20 | *** join/#asterisk mrkyr (bviitanen@h24-207-81-82.cst.dccnet.com) |
| 05:37.35 | dudes | anyone have any idea's why one out of two TE410P cards will a inturupt |
| 05:37.46 | dudes | interrupt* |
| 05:39.29 | drumkilla | contact support@digium.com |
| 05:39.52 | dudes | I did that today ... and lets say their suggestion was to disable smp. |
| 05:39.54 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
| 05:40.03 | dudes | So I did, and well, it didn't work. |
| 05:40.09 | drumkilla | so reply |
| 05:40.25 | tengulre | hi, all |
| 05:40.36 | drumkilla | they deal with this stuff every day |
| 05:41.02 | *** join/#asterisk jo3sm1th (~Fullname@ppp-69-233-0-194.dsl.irvnca.pacbell.net) |
| 05:41.12 | drumkilla | you can call in during business hours as well |
| 05:41.46 | dudes | I did speak with them over the phone. |
| 05:42.18 | dudes | I was just looking for any suggestions. I think upgrading the firmwire will fix the issue. |
| 05:42.32 | dudes | So I suppose I'll have to have them send 4 new cards with the new firmwire. |
| 05:46.56 | JerJer | so like did someone break ast_request_and_dial ? |
| 05:47.00 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
| 05:50.45 | rvhi | hi |
| 05:51.01 | rvhi | try to use * manager, telnet to port 5038 |
| 05:51.07 | rvhi | got Response: Error |
| 05:51.07 | rvhi | Message: Authentication failed |
| 05:51.15 | rvhi | any suggestion? |
| 05:54.04 | *** join/#asterisk Lun1028 (~mrprotoco@h-66-167-201-50.snvacaid.covad.net) |
| 06:00.09 | JerJer | rvhi: authenticate properly |
| 06:03.05 | *** join/#asterisk Death_INC (~thanotos@adsl-699.flex.com) |
| 06:08.58 | *** join/#asterisk Nukemizer (~Nuke@160.7.249.15) |
| 06:13.28 | Nukemizer | is it possible to have Zap/1-1 play MOH while picking up Zap/3-1 to dial a phone number ? I can not find a combination that works |
| 06:15.05 | Nukemizer | I want to revrse the Dial plan for incoming calls so the caller hears MOH while waiting for someone to answer so I imagine the Dialplan is near the same |
| 06:15.34 | tzafrir | dudes, problems with debian? |
| 06:15.56 | twisted | Nukemizer, show application dial. |
| 06:16.04 | twisted | in particular, the m flag. |
| 06:16.07 | twisted | http://fuckallyall.com/modules/gallery/albums/album13/pk_Cat1.sized.jpg |
| 06:16.09 | twisted | errrr |
| 06:16.15 | twisted | http://www.albinoblacksheep.com/flash/llama.php |
| 06:17.13 | Nukemizer | when I pick on Zap/1-1 and dial a 7 digit number ... |
| 06:17.15 | Nukemizer | exten => _XXXXXXX,1,SetMusicOnHold(home) |
| 06:17.15 | Nukemizer | exten => _XXXXXXX,2,MusicOnHold() |
| 06:17.15 | Nukemizer | exten => _XXXXXXX,3,Dial(Zap/3/${EXTEN}) |
| 06:17.24 | twisted | Nukemizer, show application dial. |
| 06:17.25 | Nukemizer | twisted, is that what you mean ? |
| 06:17.27 | twisted | in particular, the m flag. |
| 06:17.45 | baka03 | reading more can some one confirm that this sentence means i can use this ATA as a PSTN trunk to asterisk (therefor adding a PSTN line to asterisk without using a PCI card) http://geekgazette.com/index2.php?option=com_content&task=view&id=28&Itemid=26&pop=1&page=0 |
| 06:17.52 | Nukemizer | ambarrassed, I don't know what the "m" flag is . I am sorry |
| 06:18.03 | twisted | Nukemizer, for the 3rd time. SHOW APPLICATION DIAL |
| 06:18.03 | baka03 | under VoIP PBX |
| 06:18.13 | Nukemizer | ahhh. |
| 06:19.28 | Nukemizer | twisted... thank you ! I am sorry i did not get that :) must be late |
| 06:19.35 | Nukemizer | i see now |
| 06:19.38 | twisted | baka03, if you read the article, it plainly states your answer. |
| 06:20.28 | baka03 | yes but im wanting some one who hasnt spend all day reading asterisk docs to confirm im not gone crazy. |
| 06:21.02 | twisted | Nukemizer, :) |
| 06:21.13 | twisted | baka03, reading is good. reading will find you answers. |
| 06:21.40 | twisted | and to confirm, the article shows you in great detail how to perform the task |
| 06:21.56 | baka03 | twisted: also can make brain squishier than normal |
| 06:22.04 | Nukemizer | in the end Asterisk Always seems simpler than I make it ! must not try so hard |
| 06:22.12 | twisted | seeing as I haven't used one of the devices, and how this article is written by someone who has, I would say that the article has the answer |
| 06:22.23 | twisted | Nukemizer, hehe |
| 06:31.00 | *** join/#asterisk syslinux (syslinux@203-173-150-1.bliink.ihug.co.nz) |
| 06:31.06 | syslinux | Hi, |
| 06:31.20 | syslinux | Has anyone tried the prerelease chan_bluetooth? |
| 06:34.54 | syslinux | ive installed chan_bluetooth, my phone successfully pairs when i dial the cellphone from the landline i get "May 31 23:32:02 WARNING[2523]: Device T68i Philip: Unhandled Unsolicited: +CLIP: "078550260",129" how can i get asterisk to forward this to an extension |
| 06:43.46 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 06:44.13 | *** join/#asterisk kusznir (~kusznir@c-24-22-196-105.hsd1.wa.comcast.net) |
| 06:48.53 | tzafrir | what's the mp3 file RoyK always refers people to? |
| 06:50.28 | tzafrir | BTW: It probably won't be relevant for folks here, but anyway: http://asterisk-configuration.xorcom.com/ |
| 06:51.04 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 06:52.48 | kusznir | Anyone here with experinence running asterisk as a provider of many POTS lines? |
| 06:53.01 | kusznir | (and willing to talk a bit about it?) |
| 06:53.32 | inspired | kusznir: you want to set up a CLEC with asterisk? :D |
| 06:54.18 | kusznir | :) (I've actually been kind of curious as to exactly what a CLEC is) |
| 06:54.38 | inspired | competitive local exchange carrier |
| 06:54.50 | kusznir | actually my intention is to use Asterisk as a PBX for ~1,000 phone lines (so I guess many people wouldn't consider that "large", upon reflextion....) |
| 06:54.52 | inspired | a local exchange which isn't Bell, AT&T etc |
| 06:55.30 | inspired | sure, if you set up a cluster it's doable, but you'll need a bunch of channel banks and PRI cards |
| 06:55.35 | kusznir | But a CLEC would still have to use the local telco's copper to provide services to thers (well, in the pre-VoIP days, anyway..) |
| 06:55.53 | inspired | isn't that what you want? |
| 06:56.00 | kusznir | Ok, that's pretty much what I thought I'd have to do from my research so far. |
| 06:56.24 | inspired | you can have 2x Sangoma a104 in one pc |
| 06:56.32 | inspired | that's 248 channels |
| 06:56.49 | inspired | buy 8 channel banks and you have a 248 user setup |
| 06:57.06 | twisted | since when are 2 a104's 248 channels? |
| 06:57.20 | inspired | uhm, 124+124? |
| 06:57.28 | twisted | 124? |
| 06:57.30 | kusznir | Sorry, two seperate threads (to me, anyway). I'm actually researching the option for a local college to provide phone services to their dorms. Previous estimates have put this at ~$500,000 in hardware and barely profitable in monthy expenses; I think asterisk can do it better. |
| 06:57.34 | inspired | twisted: E1 |
| 06:57.42 | twisted | uhm |
| 06:57.50 | inspired | I'm sorry, I don't automatically THINK that everyone is a ulaw |
| 06:57.51 | twisted | e1 EUROPE |
| 06:57.58 | twisted | t1 USA |
| 06:58.12 | twisted | no, but you could look at his hostmask. |
| 06:58.22 | inspired | why would I care? |
| 06:58.33 | twisted | if you're going to provide info, provide it in a logical fashion. |
| 06:58.47 | twisted | WA - US, is not quite europe, now is it? |
| 06:58.51 | inspired | ok, from now on you want me to look at the hostmask of everyone I'm talking to. that sounds a bit ridiculous |
| 06:59.10 | Nukemizer | twisted, is ther a variable that lets MOH play after the call has been bridged, and stop when the called party answers ? I get MOH but only for 2 sec until call is bridged |
| 06:59.38 | twisted | Nukemizer, if you're dialing via POTS, the call will be bridged when the line is seized. |
| 07:00.06 | *** join/#asterisk DeFi (~DeFi@ip68-6-40-245.sb.sd.cox.net) |
| 07:00.17 | twisted | inspired, quit being an ass, I was trying to keep you from sounding like an idiot, but apparently have failed in doing so. |
| 07:00.39 | inspired | sounding like an idiot? you're the one being an ass right now |
| 07:00.47 | *** mode/#asterisk [+q *!*@inspired.] by twisted |
| 07:00.48 | inspired | but sure, I don't want to argue anymore |
| 07:00.51 | *** mode/#asterisk [+q inspired!*@*] by twisted |
| 07:01.00 | *** mode/#asterisk [-q *!*@inspired.] by twisted |
| 07:01.20 | twisted | actually, rather than jumping down my throat accusing me of "THINKING" that everyone is ulaw, stop assuming they are alaw. |
| 07:01.21 | Nukemizer | twisted, so because I am using Zap1-1 to start my call and Asterisk is placing the call on Zap3-1 I will not be able to do create my MOH while ringing ? |
| 07:01.34 | *** mode/#asterisk [-q inspired!*@*] by twisted |
| 07:02.04 | twisted | Nukemizer, are these fxo ports, or a digital circuit? |
| 07:02.10 | kusznir | So I'd be using T1's then, which get me 24 lines per t1. Are there any better/cheaper options than the Digium 4-port T1 card? |
| 07:02.13 | Nukemizer | one is FXS and the other is FXO |
| 07:02.28 | twisted | Nukemizer, channel 1 i'm assuming is the fxs |
| 07:02.36 | twisted | based on your previous statement |
| 07:02.37 | kusznir | And how many can I run on one server? I've read some people saying 4 max, others saying 10 max. |
| 07:02.43 | Nukemizer | I pickup on fxs to dial and the PTOS Trunk is connected to the FXO |
| 07:02.49 | twisted | yep |
| 07:03.01 | twisted | fxo automatically bridges the call on line seizure, therefore it will stop as soon as it goes to dial |
| 07:03.34 | inspired | kusznir: uhm, I wonder.. why can't you use E1 even though you are in USA? |
| 07:03.42 | inspired | between the channel banks and PRIs |
| 07:03.57 | Nukemizer | twisted, so I could only benefit if I use SIP phones behind the Asterisk ? |
| 07:04.09 | twisted | inspired, he could, although there will be a limitation mismatch between the PRI's and channel banks |
| 07:04.19 | twisted | Nukemizer, or digital circuits |
| 07:04.26 | inspired | aren't there E1 channel banks? |
| 07:04.33 | twisted | that's what i'm talking about |
| 07:04.39 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 07:04.45 | twisted | e1 circuits (last I checked) aren't used in the usa from ANY LEC |
| 07:04.55 | inspired | ok, I see |
| 07:04.57 | kusznir | true, can e1 channel banks provide standard POTS lines compatable with US standars. |
| 07:04.57 | Nukemizer | twisted, ok that helps me to understand. Thank you very much for your help |
| 07:05.13 | twisted | so the PRI will be 23 channels of useful voice, and the channel bank would be (what, 32?) |
| 07:05.18 | twisted | Nukemizer, np |
| 07:05.29 | inspired | kusznir: if you do this in the university it could probably work |
| 07:05.44 | inspired | I don't think anyone will stop you from using e1 circuits in it |
| 07:06.00 | twisted | not internally, but on the connectivity side, it would be hindered. |
| 07:06.14 | inspired | how? |
| 07:06.44 | twisted | agian, PRI from telco in US = 23 channels of audio, E1 circuit to channel bank = ~32 |
| 07:06.48 | kusznir | I was planning on trunking through SIP. |
| 07:06.56 | inspired | ehm |
| 07:06.58 | twisted | ah that works |
| 07:07.07 | twisted | although I hope ya got a nice pipe :) |
| 07:07.27 | inspired | twisted: and what's the probability of all his lines being in use at the same time? |
| 07:07.51 | inspired | switchtype=euroisdn internally and whatever you use in the US on the other side |
| 07:07.53 | twisted | probability vs. availibility |
| 07:08.19 | inspired | so you never have more than 96 users on your asterisk boxes? |
| 07:08.27 | kusznir | I was planning on using comcast enterprise services, where they say you can get any size you wnat pretty much. I was wondering how many trunks I'd need, though, as that would deturmine my connectivity need (i was thinking on the order of 5-10 Mbps) |
| 07:08.29 | twisted | if i build a system, I want all channels to be able to be in use, and all features I provision on each channel to be able to be in use in any aspect available by the system |
| 07:09.33 | twisted | i figure on a 24 port (US T1) that if i sell 3 way calling on all channels (or call waiting) i would need 2 t1's available to use. Even if they're not both at 100% utilization, that also puts me in a good position if i have a circuit failure. |
| 07:12.19 | *** join/#asterisk jbot_ (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 07:12.19 | *** topic/#asterisk is Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm | The FCC ruled for forced E911 by pstn-terminated voip carriers - http://www.fcc.gov |
| 07:21.52 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 07:22.31 | kusznir | how many boxes would be needed to service the 1000'ish lines (assuming the PTSN connection would be via SIP)? |
| 07:24.52 | tzafrir | anybody using mozphone? the link on voip-info seems to lead to a page with missing download links |
| 07:25.01 | *** part/#asterisk visik7 (~ciao@visik7.user) |
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| 07:30.27 | *** join/#asterisk wasim (~wasim@203.81.201.188) |
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| 08:34.28 | *** join/#asterisk Jas_Williams (~Jason@host217-43-100-176.range217-43.btcentralplus.com) |
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| 08:39.17 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
| 08:42.51 | robin_sz | meep? |
| 08:44.24 | robin_sz | assume for a moment I am stupid ... now ... voice mail and my Grandstream ting ... I never seem to be able to log in what am I missing? |
| 08:44.53 | robin_sz | its 1234<wait and wait>4242<wait and wait> .. |
| 08:48.18 | gordonjcp | robin_sz: what *exactly* happens when you try |
| 08:48.20 | *** join/#asterisk kajtzu (~kajtzu@shell1.fi.basen.net) |
| 08:48.49 | kajtzu | hmm. is it just me or has the public cvs not been updated for a few days? |
| 08:48.53 | potboy | hello room, I have a intel HT 2.8 box that is just missing a motherboard is there any motherboards you can recommend or a compatability list of sorts? |
| 08:49.24 | robin_sz | I get the "mailbox" prompt .. I type 1234 ... I wait .. I get the "password" prompt .. I type 4242 . I wait ... "login incorrect .. M<ailbox" |
| 08:49.52 | robin_sz | wait =~ 5+ seconds |
| 08:50.11 | *** join/#asterisk udppacket (~acydlynk@73.155.39-62.rev.gaoland.net) |
| 08:53.32 | Jas_Williams | robin_sz: Post your voicemail.conf for us to check |
| 08:56.37 | *** join/#asterisk vandien (~stephan@p509199F5.dip0.t-ipconnect.de) |
| 09:00.39 | *** join/#asterisk bublbobl (~chatzilla@nat-80-64-224-17.e-qual.fr) |
| 09:00.51 | gordonjcp | robin_sz: hm, are those set as your login and password? |
| 09:01.04 | gordonjcp | robin_sz: can you post the relevant line of your voicemail.conf? |
| 09:01.08 | gordonjcp | ~pastebin |
| 09:01.10 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 09:02.49 | bublbobl | hi all ! |
| 09:06.23 | bublbobl | where is the most up-to-date documentation about asterisk ? Is it "Asterisk Handbook Project" or the "hitchhiker's guide to *" ? |
| 09:07.28 | RoyK | ~docs |
| 09:07.29 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 09:10.06 | bublbobl | RoyK> thx |
| 09:10.38 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:15.01 | *** join/#asterisk dalfz (~dman@mojope.com) |
| 09:15.36 | dalfz | does the digium g729 codec support inband dtmf? |
| 09:16.09 | wasim | dalfz: no g729 codec supports inband, afaik |
| 09:16.35 | wasim | dalfz: no codec other than g.711 would either |
| 09:16.46 | dalfz | wasim, i need inband dtmf for i.e. banking telephone service. how would i fix that? |
| 09:16.59 | wasim | dalfz: use g711 |
| 09:17.19 | wasim | dalfz: or do out of band |
| 09:17.47 | dalfz | if i select rfc2833 or info dtmfmode, nothing happens when i press keys |
| 09:18.02 | wasim | nothing happens on which end? |
| 09:18.04 | dalfz | at the bank |
| 09:18.24 | dalfz | but i can access asterisk voicemail |
| 09:18.59 | *** join/#asterisk darkskiez (~mbryars@194.247.78.146) |
| 09:19.28 | dalfz | if i select inband dtmf, i can access my bank but not asterisk voicemail |
| 09:19.29 | darkskiez | my pri line isnt accepting calls with the error Channel Unsuitable, what does that mean ? |
| 09:21.21 | Jas_Williams | darkskiez: are these voice or data calls, is the NOA correct ? |
| 09:21.21 | darkskiez | Jas_Williams: voice, and excuse my ignorance, what's an NOA? |
| 09:21.34 | Jas_Williams | Nature of Address (national, International or unknown) |
| 09:21.49 | darkskiez | its set to unknown as recommended by voip-info for a BT line. |
| 09:22.13 | Jas_Williams | darkskiez: is this a new installation ? |
| 09:22.49 | darkskiez | Jas_Williams: yes. |
| 09:22.53 | Jas_Williams | k |
| 09:23.20 | Jas_Williams | can you post a pri debug span 1 to pastebin.ca for a call attempt |
| 09:23.43 | darkskiez | i'm rebuilding asterisk at the moment |
| 09:23.48 | darkskiez | will do in a few minutes |
| 09:23.57 | Jas_Williams | k |
| 09:24.28 | darkskiez | i was testing with cvs head, (i know bad idea dont tell me off) but i'm trying with stable now. |
| 09:26.31 | Jas_Williams | darkskiez: what is the build of the BT PRI euroISDN |
| 09:27.49 | darkskiez | Jas_Williams: also head, rebuilding all with stable at the mo. |
| 09:27.55 | dalfz | if i select rfc2833 dtmfmode, will the sip->pstn gateway be responsible for mixing in inband dtmfs? |
| 09:28.57 | Jas_Williams | darkskiez: Head should not be the issue at this level, but go for stable and then we will know :) |
| 09:30.11 | *** join/#asterisk Newbie___ (me@211.24.146.11) |
| 09:30.25 | Newbie___ | hi , how do i destroy a sip channel |
| 09:30.50 | Newbie___ | tired use soft hangup |
| 09:31.36 | Jas_Williams | Newbie___: is the phantom calls in progress ? |
| 09:32.01 | Newbie___ | no, call has ended but * still hang on to it |
| 09:32.21 | darkskiez | i rmmoded the old driver and modprobed the new one and the light hasnt come back on |
| 09:32.44 | Newbie___ | 62.xxx.151.245 6674260125 6f06bf400e9 00102/00002 UNKN (d) |
| 09:33.15 | Newbie___ | i try soft hangup .... dont know which is the channel number |
| 09:33.32 | Jas_Williams | darkskiez: reboot required |
| 09:33.39 | darkskiez | k |
| 09:33.40 | darkskiez | back later |
| 09:33.41 | Jas_Williams | Newbie___: show sip channels |
| 09:33.41 | RoyK | ~seen zoa |
| 09:33.42 | jbot | zoa <~zoa@ip-212-239-162-34.dsl.scarlet.be> was last seen on IRC in channel #asterisk, 3d 19h 7m 50s ago, saying: 'away from what ? having all wrong ?'. |
| 09:34.32 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 09:34.35 | Jas_Williams | Newbie___: sip show channels |
| 09:35.05 | Newbie___ | Peer User/ANR Call ID Seq (Tx/Rx) Format |
| 09:35.10 | Newbie___ | 219.95.155.242 (None) 21c00c23-68 00101/00022 UNKN |
| 09:35.22 | Newbie___ | Jas_Williams: which is the channel ? |
| 09:36.09 | Newbie___ | wont work with soft hangup 21c00c23-68 |
| 09:36.10 | *** join/#asterisk dwmw2 (dwmw2@baythorne.infradead.org) |
| 09:36.47 | Jas_Williams | Newbie___: do a sip show channel 21c00c23-68 |
| 09:37.52 | Newbie___ | Jas_Williams: did that, any place i should be looking at ? |
| 09:38.00 | Jas_Williams | hmm |
| 09:38.35 | Newbie___ | nothing there that say 'channel' |
| 09:38.46 | Jas_Williams | Newbie___ show channels |
| 09:39.56 | Newbie___ | Jas_Williams: the sip calls i try to destroy does not show up on show channels |
| 09:40.53 | Newbie___ | zap has terminated the calls, but sip still hang on to it |
| 09:41.03 | Jas_Williams | it should be there |
| 09:42.25 | Newbie___ | show channels tell me there are 8 active channels while sip show users say 10 active channels |
| 09:42.57 | Newbie___ | i mean sip show channels |
| 09:43.17 | Jas_Williams | Newbie___: but that chould be sip to sip calls on line for each part of the call |
| 09:44.16 | *** join/#asterisk iheartcanada (~iheartcan@lfc.tor.istop.com) |
| 09:44.33 | Newbie___ | Jas_Williams: under normal condition, sip to sip call will also be shown as active channels in show channels |
| 09:46.56 | *** join/#asterisk darkskiez (~mbryars@194.247.78.146) |
| 09:47.59 | Jas_Williams | darkskiez: any progress |
| 09:48.45 | darkskiez | Jas_Williams: its different, Ring requested on unconfigured channel 0/1 span 1 |
| 09:49.07 | Jas_Williams | k |
| 09:49.45 | Jas_Williams | darkskiez: can you post your /etc/zaptel.conf and /etc/asterisk/zapata.conf to pastebin.ca |
| 09:49.59 | darkskiez | i'll try, no mouse :) |
| 09:51.47 | Jas_Williams | darkskiez: or give me ssh access to help you get this working :) |
| 09:52.29 | darkskiez | Jas_Williams: would, but i'm behind a firewall |
| 09:52.36 | Jas_Williams | k |
| 09:53.20 | kajtzu | hmm. does srvlookup=yes only apply to configured sip peers? |
| 09:53.43 | kajtzu | or is it possible to dial someone@foo.com assuming foo.com has a SRV record? it doesn't seem to work out of the box |
| 09:57.45 | tzafrir | darkskiez, use script(1) |
| 09:57.52 | tzafrir | script logfile |
| 09:57.53 | *** join/#asterisk Chotaire (chotaire@chotaire.net) |
| 09:57.56 | tzafrir | command1 |
| 09:57.59 | tzafrir | command2 |
| 09:58.01 | tzafrir | command3 |
| 09:58.03 | tzafrir | exit |
| 09:58.18 | tzafrir | and you have all of the commands and their output in logfile |
| 09:59.18 | darkskiez | Jas_Williams: 13091 |
| 09:59.26 | Jas_Williams | off to look |
| 09:59.54 | darkskiez | Jas_Williams: you're a star |
| 10:00.32 | tzafrir | anybody wants to try to write a script that uses wget to post to pastebin? |
| 10:00.47 | tzafrir | I'd be very helpful for minimal distros |
| 10:01.28 | iheartcanada | heh |
| 10:01.28 | tzafrir | There's already a nice minimal irc client written in perl (sirc), if you really-really want |
| 10:01.51 | tzafrir | and a very minimal one written in C that takes about 20kb |
| 10:03.29 | Jas_Williams | darkskiez: comment out these lines |
| 10:03.30 | Jas_Williams | <PROTECTED> |
| 10:03.30 | Jas_Williams | trunkgroup=>1,16 |
| 10:03.30 | Jas_Williams | spanmap=>1,1,1 |
| 10:03.43 | darkskiez | I was suspicious of those |
| 10:03.49 | darkskiez | for other types of switches |
| 10:03.59 | Jas_Williams | They are not valid for euro isdn |
| 10:04.33 | Jas_Williams | what is the output of ztcfg -vvv is good with no errors |
| 10:05.57 | darkskiez | Amazing |
| 10:05.59 | darkskiez | ringing |
| 10:06.19 | darkskiez | yes |
| 10:06.26 | *** join/#asterisk kapejod (~kapejod@e178053088.adsl.alicedsl.de) |
| 10:06.33 | Jas_Williams | working |
| 10:07.33 | darkskiez | If i was a woman Jas :) |
| 10:07.40 | darkskiez | Cheers loads |
| 10:07.41 | Jas_Williams | :-P |
| 10:08.00 | darkskiez | I think if those options are incompatible with the others, it should b0rk :) |
| 10:08.33 | Jas_Williams | now try head :) |
| 10:08.58 | darkskiez | Jas_Williams probably wont :) |
| 10:09.09 | Jas_Williams | least you got it working |
| 10:09.11 | darkskiez | Maybe when its stable |
| 10:09.24 | darkskiez | I would like alot of the features though |
| 10:09.28 | darkskiez | but hey. |
| 10:10.08 | darkskiez | One touch recording, better regionalisation, database integration are all tasty |
| 10:11.25 | darkskiez | Is there any way to find out how many channels we have on the line without contacting the provider? |
| 10:11.59 | RoyK | hm |
| 10:12.09 | RoyK | sip debug peer xxx doesn't work with realtime :( |
| 10:12.21 | darkskiez | RoyK with the cache peers option on ? |
| 10:12.32 | kajtzu | RoyK: sure does |
| 10:12.34 | RoyK | what's that? |
| 10:12.38 | RoyK | darkskiez: ? |
| 10:12.42 | darkskiez | RoyK I think it makes that work. |
| 10:12.45 | dalfz | if i do an outband dtmf mode (g729), is the sip server providing pstn access responsible for mixing them inband audio? |
| 10:12.50 | *** join/#asterisk iheartcanada (~iheartcan@lfc.tor.istop.com) |
| 10:12.54 | kajtzu | rtcachefriends=yes |
| 10:13.00 | RoyK | kajtzu: where? |
| 10:13.00 | darkskiez | that one |
| 10:13.01 | kajtzu | and rtautoclear=yes in sip.conf |
| 10:13.12 | kajtzu | and rtnoupdate=no |
| 10:14.07 | kajtzu | you guys know if asterisk just uses srvlookup=yes for configured peers or is it possible to dial anyone? the dial anyone part doesn't seem to work for me |
| 10:14.44 | RoyK | nice |
| 10:14.46 | RoyK | thanks |
| 10:17.26 | RoyK | hm |
| 10:17.33 | wasim | dalfz: theoretically |
| 10:17.41 | RoyK | <PROTECTED> |
| 10:17.41 | RoyK | Jun 1 12:16:57 NOTICE[4045]: app_dial.c:972 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) |
| 10:17.44 | RoyK | <PROTECTED> |
| 10:17.52 | RoyK | that's with realtime |
| 10:17.52 | RoyK | but the peer is indeed available |
| 10:21.41 | *** join/#asterisk kletter-matze (~kletter-m@212.126.219.82) |
| 10:28.48 | *** join/#asterisk ManxPower (~eric@32.199-78-194.adsl-fix.skynet.be) |
| 10:32.00 | *** join/#asterisk pietro (~pietro@host164-68.pool8248.interbusiness.it) |
| 10:32.09 | pietro | hello |
| 10:32.36 | pietro | is possible confiuring a language in oh323 channel ? |
| 10:34.50 | *** join/#asterisk apardo (~apardo@96.Red-83-43-209.pooles.rima-tde.net) |
| 10:37.48 | *** join/#asterisk Big_moth (taCo@220.187.90.174) |
| 10:39.17 | *** part/#asterisk Big_moth (taCo@220.187.90.174) |
| 10:44.03 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
| 10:47.16 | RoyK | pietro: I don§t think the channel is language related :P |
| 10:49.40 | *** join/#asterisk newl (~newlook@203-59-15-197.dyn.iinet.net.au) |
| 10:51.07 | *** join/#asterisk oden (~oden@194-237-146-22.customer.telia.com) |
| 10:51.51 | scanna | hi all, it's possible to obtain an evaluation licence for g729? |
| 10:52.35 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
| 10:53.13 | RoyK | what is "screening" |
| 10:53.15 | Newbie___ | scanna: you do not need to evaluate g729, g729 is good |
| 10:53.31 | RoyK | scanna: there's a "free" g.729 somwhere |
| 10:53.37 | RoyK | ~wiki g.729 |
| 10:54.23 | *** join/#asterisk darkskiez (~mbryars@194.247.78.146) |
| 10:55.35 | darkskiez | i've got the zapata prefixes defined for the different call types, but the callerid is not being pre-pended with them despite the pri debug showing it correctly recognises the type. hmm. |
| 10:57.06 | tzafrir | scanna, OTOH, it is probably a different implementation than the one of digium, and thus yo can't ssume it will perform the same |
| 11:04.11 | *** join/#asterisk orlon_ (~apathy@c220-237-74-194.kelvn1.qld.optusnet.com.au) |
| 11:05.00 | pietro | ! 8139810 1 1 |
| 11:05.07 | pietro | continua a sparami riche come questa |
| 11:05.11 | pietro | che vul dire ? |
| 11:05.17 | pietro | ops |
| 11:05.19 | pietro | sorry |
| 11:11.55 | pietro | when i use a mISDN channell in CLI i have too "! 8139810 1 1" |
| 11:14.24 | *** join/#asterisk postel (~canonical@postel.user) |
| 11:15.49 | dalfz | wasim, is that a commonly neglected feature of sip proxies? |
| 11:16.24 | orlon_ | anyone using res_config_odbc with #include's?? |
| 11:18.51 | dalfz | wasim, my external sip->pstn proxy (ABS GW v5.1.0) does nothing about INFO or RFC2833 dtmfs.. shrug |
| 11:20.04 | RoyK | hm |
| 11:20.09 | RoyK | i need call diverts in zap |
| 11:22.51 | *** join/#asterisk P-NuT (~P-NuT@CPE-60-225-214-14.nsw.bigpond.net.au) |
| 11:23.00 | P-NuT | Hi all. |
| 11:23.58 | P-NuT | Say is anyone here experienced with setting up asterisk on Ubuntu or Debian? |
| 11:24.07 | orlon_ | yup |
| 11:24.10 | orlon_ | Debian |
| 11:24.16 | P-NuT | Ubuntu specifically.. |
| 11:24.29 | orlon_ | shouldn't be that much difference, wassup? |
| 11:24.37 | P-NuT | Well, |
| 11:24.48 | P-NuT | I apt-get install asterisk |
| 11:25.23 | orlon_ | first off, I'd recommend using aptitude rather than apt-get. but that won't break things |
| 11:25.32 | P-NuT | and all's well with the sip phones talking to each other, however setting up the card I have no idea about. |
| 11:25.48 | P-NuT | Why's aptitude better? |
| 11:25.52 | orlon_ | what sort of card? |
| 11:26.04 | orlon_ | aptitude supports reverse dependencies |
| 11:26.05 | P-NuT | oh.. umm Tiger... something |
| 11:26.19 | P-NuT | ok then.... |
| 11:26.35 | orlon_ | so if you install something and it drags in other libs etc they're automagically marked as 'Auto' and uninstalled when no longer in use |
| 11:26.45 | orlon_ | and it has a builtin in Minesweeper game =] |
| 11:26.52 | P-NuT | oh cool! |
| 11:27.29 | orlon_ | aptitude does pretty much everything apt-get _and_ apt-cache do on the CLI |
| 11:27.46 | P-NuT | when you uninstall asterisk it leaves all the config files there it sux. |
| 11:27.50 | orlon_ | so this Tiger card, is it ISDN, PSTN? |
| 11:27.58 | P-NuT | I think ISDN |
| 11:28.02 | orlon_ | P-NuT: that's a remove as opposed to a purge |
| 11:28.03 | P-NuT | it said ISDN. |
| 11:28.25 | P-NuT | ok then.... |
| 11:28.37 | P-NuT | I'm new to ubuntu. I was a redhat man. |
| 11:28.43 | orlon_ | ok, you need to work out whether it's support by chan_capi, chan_misdn or the i4l stuff |
| 11:28.49 | orlon_ | you'll never look back ;) |
| 11:28.56 | P-NuT | can I just install them all? |
| 11:29.04 | RoyK | nonononono |
| 11:29.09 | P-NuT | oh.. |
| 11:29.10 | RoyK | what sort of hardware do you have? |
| 11:29.16 | P-NuT | Sorry thats the redhat in me talking |
| 11:29.20 | P-NuT | well... |
| 11:29.31 | P-NuT | I'm not sure.. Kudzu detected it.. |
| 11:29.40 | P-NuT | ...in centOS |
| 11:29.46 | P-NuT | but I didnt write it down. |
| 11:29.49 | orlon_ | i'm using Cisco gateways rather than cards |
| 11:29.53 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 11:29.57 | RoyK | P-NuT: what does lspci say? |
| 11:29.59 | P-NuT | It's a digium clone that uses the zaptel driver |
| 11:30.07 | RoyK | ok |
| 11:30.10 | P-NuT | yeah.... |
| 11:30.17 | P-NuT | lspci.... |
| 11:30.23 | P-NuT | ....yeah.... |
| 11:30.25 | RoyK | P-NuT: that's not isdn, right? |
| 11:30.32 | RoyK | iirc digium has no bri cards |
| 11:30.55 | orlon_ | P-NuT: msg me the output of lspci |
| 11:31.02 | RoyK | no |
| 11:31.04 | RoyK | pastebin it |
| 11:31.05 | P-NuT | let me google tiger isdn asterisk.. |
| 11:31.06 | P-NuT | one sec... |
| 11:31.13 | RoyK | ~pastebin |
| 11:31.28 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 11:31.28 | P-NuT | in var/log? |
| 11:31.32 | orlon_ | RoyK: damn new fangled web thingies |
| 11:31.55 | orlon_ | P-NuT: no, just run lspci |
| 11:32.39 | P-NuT | ok |
| 11:33.01 | Aze` | Is it possibile empty cache of system utilized by *? |
| 11:33.09 | P-NuT | ok |
| 11:33.10 | orlon_ | 14.4! luxury! |
| 11:33.12 | P-NuT | pasting now |
| 11:33.16 | P-NuT | root@netbox:/home/dave # lspci |
| 11:33.16 | P-NuT | 0000:00:00.0 Host bridge: Silicon Integrated Systems [SiS] 661FX/M661FX/M661MX Host (rev 11) |
| 11:33.16 | P-NuT | 0000:00:01.0 PCI bridge: Silicon Integrated Systems [SiS]: Unknown device 0003 |
| 11:33.17 | P-NuT | 0000:00:02.0 ISA bridge: Silicon Integrated Systems [SiS] SiS963 [MuTIOL Media IO] (rev 25) |
| 11:33.17 | P-NuT | 0000:00:02.1 SMBus: Silicon Integrated Systems [SiS]: Unknown device 0016 |
| 11:33.17 | P-NuT | 0000:00:02.5 IDE interface: Silicon Integrated Systems [SiS] 5513 [IDE] |
| 11:33.21 | P-NuT | 0000:00:02.7 Multimedia audio controller: Silicon Integrated Systems [SiS] Sound Controller (rev a0) |
| 11:33.23 | P-NuT | 0000:00:03.0 USB Controller: Silicon Integrated Systems [SiS] USB 1.0 Controller (rev 0f) |
| 11:33.25 | P-NuT | 0000:00:03.1 USB Controller: Silicon Integrated Systems [SiS] USB 1.0 Controller (rev 0f) |
| 11:33.27 | P-NuT | 0000:00:03.3 USB Controller: Silicon Integrated Systems [SiS] USB 2.0 Controller |
| 11:33.29 | RoyK | ~lart P-NuT |
| 11:33.29 | P-NuT | 0000:00:04.0 Ethernet controller: Silicon Integrated Systems [SiS] SiS900 PCI Fast Ethernet (rev 91) |
| 11:33.31 | P-NuT | 0000:00:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 11:33.32 | orlon_ | P-NuT: don't paste to the channel |
| 11:33.33 | P-NuT | 0000:01:00.0 VGA compatible controller: Silicon Integrated Systems [SiS] 661FX/M661FX/M661MX/741/M741/760/M760 PCI/AGP |
| 11:33.36 | P-NuT | how's that? |
| 11:33.37 | RoyK | ~lart P-NuT for not using the pastebin |
| 11:33.38 | P-NuT | oh |
| 11:33.39 | RoyK | ~pastebin |
| 11:33.56 | jbot | somebody said pastebin was a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 11:33.56 | P-NuT | sorry |
| 11:33.56 | sigterm | whoa there fella |
| 11:34.00 | P-NuT | it's the 0000:00:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 11:34.07 | orlon_ | yup |
| 11:34.25 | P-NuT | So can anyone help a brother out? |
| 11:34.34 | P-NuT | ;-) |
| 11:35.00 | P-NuT | I've looked all over the net for a ubuntu how to but I cant find one.. |
| 11:35.07 | orlon_ | never used one...let's see what google says |
| 11:35.08 | P-NuT | :'( |
| 11:35.38 | P-NuT | it works right out of the box with asterisk at home. |
| 11:36.25 | P-NuT | Any ideas people? |
| 11:36.37 | orlon_ | P-NuT: try this -- dmesg | grep -iE 'tiger|isdn' |
| 11:36.45 | orlon_ | and use pastebin this time ;) |
| 11:36.49 | P-NuT | What I thought was the go is: |
| 11:36.49 | P-NuT | apt-get install asterisk |
| 11:36.52 | P-NuT | oh ok.. |
| 11:36.54 | P-NuT | one sec. |
| 11:37.20 | P-NuT | root@netbox:/home/dave # dmesg | grep -iE 'tiger|isdn' |
| 11:37.20 | P-NuT | ISDN subsystem Rev: 1.1.2.3/1.1.2.3/1.1.2.2/1.1.2.3/1.1.2.2/1.1.2.2 loaded |
| 11:37.20 | P-NuT | HiSax: Linux Driver for passive ISDN cards |
| 11:37.48 | P-NuT | this makes no sense to me though.. |
| 11:37.53 | P-NuT | :'( |
| 11:38.28 | orlon_ | ok, so the driver has loaded |
| 11:38.54 | orlon_ | looks like that thing uses the isdn4linux drivers (ie HiSax) |
| 11:39.35 | P-NuT | oh.. |
| 11:39.40 | P-NuT | I've heard of that |
| 11:40.10 | orlon_ | http://www.voip-info.org/tiki-index.php?page=Asterisk%20ISDN4Linux |
| 11:40.11 | orlon_ | try that |
| 11:40.21 | RoyK | i4l is not good |
| 11:40.34 | RoyK | better spend $30 on an HFC PCI card |
| 11:40.45 | cochi | i got no probs with i4l ;) |
| 11:41.23 | jalsot | hello |
| 11:41.27 | cochi | well i need it anyway (driver issue) |
| 11:41.28 | orlon_ | P-NuT: might be worth searching the asterisk lists to see if someone has got chan_capi working with that card |
| 11:42.27 | jalsot | does anybody know if it is possible to mix in and out channel with monitor application into one stream on-the-fly? |
| 11:42.32 | orlon_ | anybody using res_config_odbc and #include's ?? |
| 11:43.13 | ManxPower | jalsot, no. You can do it after. See "show application monitor" |
| 11:43.30 | P-NuT | The thing is guys that with asterisk@home it configures EVERYTHING out of the box. |
| 11:43.37 | P-NuT | I dont have to do anything. |
| 11:43.44 | jalsot | ManxPower: yes, I know that it is possible after, but would it be possible to make continuously? |
| 11:43.58 | jalsot | I would like to save some I/O and space... |
| 11:44.00 | ManxPower | jalsot, I just said no. |
| 11:44.05 | orlon_ | P-NuT: so use that then or copy their config |
| 11:44.07 | P-NuT | But with Ubuntu I have no idea.... |
| 11:44.15 | P-NuT | will that work do you think?> |
| 11:45.03 | orlon_ | P-NuT: well....yes |
| 11:45.09 | P-NuT | really? |
| 11:45.35 | jalsot | ManxPower: ok, thx. Theoretically by hacking res_monitor should be the right way to make it, right? |
| 11:45.37 | ManxPower | jalsot, You don't want to force Asterisk to mix the stream and do any format conversion in realtime. |
| 11:45.40 | *** join/#asterisk cpm (~Chip@cpm.sustaining.supporter.pdpc) |
| 11:45.44 | orlon_ | P-NuT: <sarcasm>no, i'm just messing with you</sarcasm> |
| 11:46.05 | ManxPower | you want to write it to disk as fast as you can and then let the system convert it post-call as slowly as it needs to. |
| 11:46.22 | orlon_ | P-NuT: the driver's already loaded for the card, you've just got to configure Asterisk. that'll be the same no matter what linux distro you use |
| 11:47.25 | jalsot | ManxPower: is mixing a CPU consuming task? |
| 11:47.46 | P-NuT | ok, so in theory I could copy all of the /etc/asterisk directory from asterisk@home install and use it in ubuntu? |
| 11:47.57 | ManxPower | jalsot, I doubt it, but any format conversion could be. but Asterisk needs every bit of CPU it can get. |
| 11:49.34 | jalsot | ManxPower: my problem: 30 concurrent channel recording, probably it makes problems with voice quality; so I would like to record onto tmpfs but my RAM/budget is limited |
| 11:49.42 | ManxPower | jalsot, I'm sure what you want to do will work....for small numbers of calls |
| 11:50.05 | ManxPower | jalsot, what makes you think you are disk I/O bound? |
| 11:50.14 | jalsot | ManxPower: right I'm recording into sln, than soxmix and after that lame |
| 11:50.20 | ManxPower | And have you fiddled with IDE DMA, Unmask IRQ, etc |
| 11:51.08 | jalsot | ManxPower: I just read one presentation which said I should omit any unnesessary disk activity |
| 11:51.19 | ManxPower | egads! well convery to GSM for one thing |
| 11:51.22 | orlon_ | P-NuT: possibly. you might need to adjust some things like asterisk.conf as the debian/ubuntu packages move some things to obey the FHS |
| 11:51.47 | ManxPower | encoding mp3s is VERY CPU intensive. |
| 11:52.16 | jalsot | ManxPower: what to do if customer wants mp3 and I cannot put another box there? |
| 11:52.39 | *** part/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 11:52.56 | ManxPower | jalsot, You have 1 liter of needs and a .5 liter container. |
| 11:53.16 | P-NuT | ok, maybe that will be my best bet.. |
| 11:53.16 | P-NuT | thanks for your help guys.. |
| 11:53.36 | ManxPower | jalsot, try using whatever options lame recommends for speeding up encoding time (usually at the expense of larger file sizes) |
| 11:53.39 | orlon_ | P-NuT: make a backup first |
| 11:53.47 | orlon_ | gotta go |
| 11:53.54 | P-NuT | ok will do |
| 11:53.58 | P-NuT | thanks again... |
| 11:54.07 | jalsot | ManxPower: so the tmpfs idea won't help, what is your opinion? |
| 11:54.27 | ManxPower | And have you fiddled with IDE DMA, Unmask IRQ, etc? |
| 11:54.55 | jalsot | yes |
| 11:55.04 | jalsot | no IRQ sharing, DMAs are ok |
| 11:55.39 | ManxPower | are you calling hdparam before asterisk, I assume? |
| 11:55.49 | jalsot | yes |
| 11:55.57 | ManxPower | what value is your -X? |
| 11:56.02 | ManxPower | 66 or 69? |
| 11:56.29 | tzafrir | I'm intersted to to know what you folks think of http://asterisk-configuration.xorcom.com/ . If you think it is OT here, then PM me, or use #xorcom-rapid |
| 11:57.16 | jalsot | my current hdparm.conf looks as: |
| 11:57.17 | tzafrir | ManxPower, usually simply hdparm -d1 gets it right, and with -X you may just get it wrong |
| 11:57.20 | jalsot | /dev/hda { |
| 11:57.20 | jalsot | <PROTECTED> |
| 11:57.23 | jalsot | <PROTECTED> |
| 11:57.23 | jalsot | <PROTECTED> |
| 11:57.26 | jalsot | <PROTECTED> |
| 11:57.26 | jalsot | } |
| 11:57.34 | cochi | grmbl tzafrir, why not some days earlier ;) |
| 11:57.46 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
| 11:57.52 | tzafrir | jalsot, hdparm -v /dev/hda |
| 11:58.06 | tzafrir | jalsot, pastebin |
| 11:58.09 | tzafrir | ~pastebin |
| 11:58.13 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 11:58.39 | ManxPower | tzanger, just smack him. He's been here long enough to know better. |
| 11:58.49 | jalsot | http://pastebin.ca/13095 |
| 11:59.03 | jalsot | :) |
| 11:59.17 | jalsot | yes, some times I got a note ;) |
| 11:59.34 | tzafrir | jalsot, hdparm -c1 |
| 11:59.54 | tzafrir | though it shouldn't have too great an impact on your performance |
| 12:00.06 | jalsot | tzafrir: ok |
| 12:00.26 | jalsot | tzafrir: what about SATA disks? |
| 12:00.41 | jalsot | I'm planning to upgrade to SATA... |
| 12:00.50 | tzafrir | jalsot, no eye deer |
| 12:00.52 | jalsot | there are no such settings in hdparm afaik |
| 12:01.00 | blitzrage | ManxPower: tzafrir != tzanger afaik :) |
| 12:02.01 | tzafrir | 'tzafrir' normally gets confused for tzanger, when someone uses the tab completion of, e.g. xchat |
| 12:02.23 | tzafrir | But the other way around? strange indeed |
| 12:03.48 | gordonjcp | tzafrir: in irssi, if you keep pressing tab it alternates between the two |
| 12:03.57 | gordonjcp | and the first choice is the last one you used |
| 12:04.13 | gordonjcp | so I can tab complete again - tz<tab> tzafrir |
| 12:05.00 | ManxPower | I'm off to the diamond museum. leave me alone. 8-) |
| 12:06.09 | Druken | Jun 1 08:06:42 WARNING[3308]: channel.c:511 ast_channel_walk_locked: Avoided deadlock for 'Local/1@pgdel-0126,2', 10 retries |
| 12:06.12 | Druken | any ideas? |
| 12:06.46 | *** join/#asterisk _omer (dfsdf@202.147.167.213) |
| 12:07.33 | Jas_Williams | Druken: are you using the manager interface ? |
| 12:07.46 | Druken | what manager interface? |
| 12:07.55 | wasim | heh |
| 12:08.04 | Druken | i use FOP |
| 12:09.28 | _omer | "SIP show inuse" doesnt show my anything but only the headings... |
| 12:09.47 | _omer | ?? |
| 12:09.48 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:10.21 | Druken | _omer: are you using realtime for your sip.conf ? |
| 12:10.26 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
| 12:10.34 | _omer | realtime? |
| 12:10.43 | wasim | heh |
| 12:11.19 | Druken | that answers my question :) |
| 12:11.21 | Druken | carry on |
| 12:11.29 | _omer | :-D |
| 12:11.42 | _omer | oh come on ..... |
| 12:12.22 | _omer | I'm a newbie....yesterday I configured Asterisk with FWD and today, I dont see anything when I do "sip show inuse" |
| 12:12.23 | Druken | was just going to say if your using realtime none of that stuff will work |
| 12:13.05 | *** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 12:14.16 | *** join/#asterisk festr_ (~festr@ns.regnet.cz) |
| 12:14.30 | _omer | I think I saw this stuff working before putting FWD in the box ... :S and it happened yesterday. |
| 12:14.55 | festr_ | hello, i've problem with digium E1 card. After reboot PC, card is not detected byt the driver. After shutdown (unplug power) and boot, everything is ok, any idea? |
| 12:15.27 | wasim | festr_: wierd, try moving slots |
| 12:15.39 | Druken | festr_: you must power down, they will not fix with a soft reboot |
| 12:15.58 | _omer | how could he move slots.....it may damages his board ;) |
| 12:16.14 | festr_ | hmm.. this is normal "feature" ? |
| 12:16.22 | festr_ | that you have to power down? |
| 12:16.26 | Druken | not much of a feature, just a reality |
| 12:16.55 | Druken | i used to pull out my hair trying to figure that one out |
| 12:16.57 | festr_ | on another machine no power down needed |
| 12:17.07 | Druken | i'd sit there for hours, rebooting the damn server |
| 12:17.10 | festr_ | so where is problem? |
| 12:17.16 | wasim | i've never had that happen |
| 12:17.46 | festr_ | wasim: you have the same problem? |
| 12:17.58 | wasim | festr_: nope, Druken does |
| 12:18.42 | Druken | yeah, i have 3 cards in my system, and when they go poof, i have to hard reboot the server or none of them come bacl |
| 12:18.49 | Druken | s/bacl/back |
| 12:19.11 | festr_ | what about another motherboard |
| 12:19.31 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 12:19.37 | *** part/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 12:19.38 | wasim | i've got about 100 cards in various servers, and thank god haven't faced this problem |
| 12:19.39 | Druken | that might fix the problem.. i don't know... |
| 12:19.43 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 12:19.46 | Druken | mine running on an intel board |
| 12:20.36 | Druken | wasim: luckily my boards don't go poof very often, not anymore.. thank christ |
| 12:24.10 | jalsot | does anybody know how CPU demanding is GSM codec? Is it worth to record files into gsm in place of mp3? |
| 12:25.21 | _omer | hey I dont even have "Sip debug" command specifically. |
| 12:25.29 | _omer | whats going on with my box. |
| 12:25.36 | wasim | jalsot: "show translation" |
| 12:25.55 | *** join/#asterisk P-NuT (~P-NuT@CPE-60-225-214-14.nsw.bigpond.net.au) |
| 12:26.01 | P-NuT | Hi all. |
| 12:26.27 | P-NuT | got a few debian asterisk questions. |
| 12:26.45 | _omer | shoot! |
| 12:26.55 | Jas_Williams | _omer: what version of * are you running |
| 12:26.56 | jalsot | wasim: slin2gsm is 1ms |
| 12:27.02 | gordonjcp | P-NuT: apt-get install zaptel && apt-get install asterisk |
| 12:27.12 | _omer | Jas_Williams: how to check the version? |
| 12:27.15 | P-NuT | thats it? |
| 12:27.18 | jalsot | wasim: I don't know details for mp3 |
| 12:27.41 | bublbobl | anyone receiving FAX in a SIP context ? I have rxfax(${FAXFILE}) exiting non zero. |
| 12:27.42 | Jas_Williams | from the cli type show version |
| 12:28.33 | _omer | Jas_Williams: Asterisk CVS-HEAD-05/17/05-08:00:05 built by root@MYLINUX on a i686 running Linux |
| 12:28.50 | P-NuT | gordon: will that give me fax and musiconhold |
| 12:28.51 | gordonjcp | P-NuT: well, then you need to actually set asterisk up to do stuff |
| 12:29.03 | gordonjcp | you can live without the zaptel stuff if you're not using meetme and iax trunking |
| 12:29.03 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
| 12:29.16 | gordonjcp | P-NuT: no idea about fax because I don't use it |
| 12:29.24 | gordonjcp | you will need mpg123 for music on hold |
| 12:29.27 | Jas_Williams | _omer: you could try updating to the latest version of HEAD do a make update in the asterisk src directory |
| 12:29.29 | P-NuT | I just have 3 sip phones and a card. |
| 12:29.34 | P-NuT | thats all I want |
| 12:30.00 | P-NuT | I tried to get music on hold to work before, but it didnt work. |
| 12:31.26 | Jas_Williams | P-NuT: try a make mpg123 in the asterisk src directory as a starting point for music on hold |
| 12:31.28 | _omer | Jas_Williams: ok....I check it out... |
| 12:31.59 | P-NuT | really? |
| 12:31.59 | bublbobl | P-NuT> Didin't you install mpg321 instead of mpg123 (many distros replace it for security reasons, but I read asterisk needs the original mpg123) |
| 12:32.18 | P-NuT | no I have the real mpg123 |
| 12:32.36 | bublbobl | P-NuT> :-s |
| 12:32.49 | Druken | what version of mpg123 ? |
| 12:33.13 | P-NuT | no idea. |
| 12:33.21 | bublbobl | (mine is Version 0.59r and it works) |
| 12:33.24 | tzafrir | mpg123 from where? |
| 12:35.08 | tzafrir | if you use the debian package and use it for streaming, be sure to update to the latest version (0.59r-20) as they fixed some holes in the streaming code |
| 12:37.47 | *** join/#asterisk Broesmeli (~broesme@195.65.2.68) |
| 12:38.56 | jalsot | btw, you can use mpg123 from the digium cvs ;) [in asterisk dir. make 'make mpg123'] |
| 12:40.35 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
| 12:43.00 | _omer | <PROTECTED> |
| 12:44.44 | *** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 12:44.49 | Ariel_ | Morning everyone. |
| 12:45.19 | _omer | good morning boy... |
| 12:45.19 | tzafrir | jalsot, and I bet they don't bother patching half the holes distros like Debian and Gentoo patch in mpg123 |
| 12:45.56 | tzafrir | which means: don't use it for streaming |
| 12:46.34 | jalsot | tzafrir: yes, probably you're right! |
| 12:49.45 | *** join/#asterisk coppice (~chatzilla@14.198.17.210.dyn.pacific.net.hk) |
| 12:49.58 | *** join/#asterisk Meaty (~cp_simbul@office.abi.ca) |
| 12:50.26 | *** join/#asterisk Lee__ (~lee@ool-44c26fa3.dyn.optonline.net) |
| 12:50.55 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 12:53.21 | darkskiez | why wouldnt asterisk prefix incoming calls callerid with the prefixes defined in the zapata.conf ? |
| 12:59.03 | _omer | I have two asterisk machines.....one have "SIP DEBUG" command but the second one doesnt have "sip debug" command specifically.... |
| 13:00.59 | tzafrir | _omer, is the second system an asterisk one? are you in the asterisk CLI? |
| 13:01.02 | tzafrir | asterisk -r |
| 13:01.47 | _omer | yes |
| 13:01.57 | _omer | both have same scripts.. |
| 13:02.01 | tzafrir | _omer, and chan_sip is loaded? |
| 13:02.11 | _omer | how to check that? |
| 13:02.21 | _omer | both are working fine with SIP |
| 13:03.27 | Jas_Williams | darkskiez: depends on how the carrier presents the NOA on the numbers |
| 13:04.43 | Jas_Williams | _omer: after make update do a make install then an restart when convenient on * |
| 13:05.22 | _omer | Jas_Williams : ok I do it |
| 13:05.34 | darkskiez | Jas_Williams: it says TON: National Number (2) NPI: ISDN E.164/163 |
| 13:05.51 | darkskiez | and i have nationalprefix = 0 in my zapata |
| 13:09.26 | darkskiez | Oh, thats a CVS head setting. |
| 13:09.43 | darkskiez | thats why it doesnt work. |
| 13:10.57 | kapejod | or a bristuff setting |
| 13:11.29 | _omer | Jas_Williams: after make update and restart my machine....SIP DEBUG is still not there... |
| 13:11.51 | Druken | did you make install ? |
| 13:11.55 | _omer | yes |
| 13:12.08 | _omer | make update, make install, and restart.. |
| 13:12.27 | _omer | I have problem with SIP SHOW INUSE as well.. |
| 13:13.05 | _omer | here is what I get when I do SIP SHOW INUSE |
| 13:13.06 | _omer | * User name In use Limit |
| 13:13.06 | _omer | * Peer name In use Limit |
| 13:13.10 | _omer | that's it |
| 13:16.45 | _omer | any help?? plz |
| 13:16.55 | *** join/#asterisk Katty (~katrina@68.112.15.110) |
| 13:17.14 | Katty | miow. |
| 13:17.23 | Wonka | *mew* :x) |
| 13:19.21 | Katty | izzoquiet |
| 13:19.36 | Chad-wl | On Fedora Core3 where would one get Timezone.h from? Chan_zap needs it and it's missing? |
| 13:19.53 | tzafrir | it comes from zaptel's installation |
| 13:20.18 | tzafrir | and it's *t*imezone.h . case matters. |
| 13:21.15 | tzafrir | hmmm.... |
| 13:21.27 | tzafrir | timezone or tonezone? |
| 13:21.35 | Chad-wl | tzafrir: ya, I realize that, no idea why I typed it in upper case. |
| 13:21.44 | Chad-wl | Tonezone.. dooh.. |
| 13:21.51 | Chad-wl | er.. tonezone.h |
| 13:22.26 | _omer | I have pasted all of my old scripts.....but problems are stil there :S |
| 13:22.56 | tzafrir | _omer, so you better start giving details. |
| 13:23.14 | _omer | I have given all the details... |
| 13:23.21 | tzafrir | Any way of getting the system to a "working" state? |
| 13:23.35 | tzafrir | can you define extly what's the problem is? |
| 13:23.42 | _omer | yeah sure... |
| 13:24.21 | _omer | 1- I all the commands but not "SIP DEBUG" command . |
| 13:24.58 | tzafrir | what version do you use? |
| 13:25.08 | _omer | 2- "SIP SHOW INUSE" ...was working fine....but now it's not working... |
| 13:25.12 | Katty | how about a sip smack command. |
| 13:25.15 | _omer | "show version"? |
| 13:25.30 | tzafrir | what exactly should happen when you run sip debug? |
| 13:25.39 | tzafrir | and what does happen? |
| 13:25.54 | _omer | No such command 'sip debug' (type 'help' for help) |
| 13:26.10 | _omer | show version: Asterisk CVS-HEAD-05/17/05-08:00:05 built by root@MYLINUX on a i686 running Linux |
| 13:26.52 | tzafrir | but there are other sip commands, right? |
| 13:27.02 | _omer | right |
| 13:27.13 | Druken | like sip show peers works? |
| 13:27.18 | _omer | yes.... |
| 13:27.54 | SwK | are you sure yor build isnt busted? |
| 13:28.37 | _omer | my box is able to get the calls thru sip...how could it be busted. |
| 13:28.55 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 13:29.10 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 13:29.31 | SwK | well head has has sip debug in it for quite a while |
| 13:29.39 | SwK | sip ? |
| 13:29.42 | SwK | should show it |
| 13:30.07 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 13:30.11 | SwK | sip debug peer <tab> should give you a list of peers if not something aint right |
| 13:30.13 | tzafrir | _omer , could you try: |
| 13:30.23 | tzafrir | set debug 99 |
| 13:30.26 | tzafrir | set verbose 99 |
| 13:30.34 | tzafrir | unload chan_sip.so |
| 13:30.43 | tzafrir | load chan_sip.so |
| 13:31.03 | _omer | ok |
| 13:32.50 | cochi | [OT] mh anybody else having probs with eyeBeam? Slow connect, failed auth despite correct credentials? |
| 13:35.18 | Hmmhesays | heh what the heck, no loaded rtc kernel module on this install |
| 13:35.51 | Hmmhesays | odd |
| 13:35.54 | Hmmhesays | quite |
| 13:38.04 | Katty | ooooh, look who i found! |
| 13:39.38 | jalsot | how can I mix 2 gsm files together into one gsm file? |
| 13:39.56 | tzafrir | jalsot, sox? |
| 13:40.37 | jalsot | can it mix 2 gsm files without decoding them? |
| 13:40.37 | Hmmhesays | haha i'm not compiling a new kernel on this machine |
| 13:40.49 | tzafrir | A universal sound sample translator |
| 13:40.52 | tzafrir | ~sox |
| 13:40.53 | jbot | hmm... sox is Sound Processing Tool. URL: http://sox.sourceforge.net/ |
| 13:41.06 | Hmmhesays | jalsot: lol |
| 13:41.14 | jalsot | Hmmhesays: why? :) |
| 13:41.24 | MikeJ[Laptop] | bkw_, COME BACK! |
| 13:41.33 | bkw_ | jalsot, you have to convert to sln before you mux the files |
| 13:41.34 | MikeJ[Laptop] | your too young too die. |
| 13:41.40 | MikeJ[Laptop] | heh |
| 13:41.44 | Katty | Hmmhesays: that would be kneading action |
| 13:41.59 | jalsot | bkw_: ah, so it's not effective |
| 13:42.10 | bkw_ | effective? |
| 13:42.14 | Hmmhesays | cat 1.gsm 2.gsm 3.gsm > new.gsm |
| 13:42.31 | Hmmhesays | Katty: learn something new every day :D |
| 13:42.48 | tzafrir | Hmmhesays, don't those files have headers? |
| 13:42.49 | Katty | Hmmhesays: you would |
| 13:42.54 | bkw_ | no |
| 13:42.54 | jalsot | what I would like to overcome is big sln files [what I have now] and much disk activity |
| 13:42.56 | bkw_ | gsm doesn't have headers |
| 13:43.02 | *** join/#asterisk jhava (~icechat5@200.58.26.21) |
| 13:43.16 | bkw_ | jalsot, you're thinking too much |
| 13:43.17 | Hmmhesays | <jalsot> can it mix 2 gsm files without decoding them? <-- that's what i was answering |
| 13:43.20 | bkw_ | if the file is gsm |
| 13:43.23 | bkw_ | and your caller is ulaw |
| 13:43.33 | bkw_ | the file goes to the caller via sln |
| 13:43.39 | Katty | i always feel dumb when i show up in here. |
| 13:43.49 | bkw_ | robin_sz, not that kind of header |
| 13:43.51 | bkw_ | gsm is headerless |
| 13:43.52 | *** join/#asterisk Eduardo_Henrique (~ehst@201.19.5.145) |
| 13:43.55 | robin_sz | dang ;) |
| 13:44.00 | Katty | the headless gsman |
| 13:44.07 | tzafrir | robin_sz, that's why compilation is so slow |
| 13:44.08 | jalsot | bkw_: my case: 30 channels recording, I would like to find the best way to make it |
| 13:44.08 | robin_sz | bkw_: like a headless horseman? |
| 13:44.10 | Katty | Hmmhesays: thx |
| 13:44.13 | Hmmhesays | shh... it's actually just a wire strainer |
| 13:44.18 | Katty | hot |
| 13:44.23 | Katty | now i can make pasta! |
| 13:44.27 | bkw_ | jalsot, slinear is the least cost format |
| 13:44.44 | Hmmhesays | better share |
| 13:44.50 | robin_sz | so .. these single span FXO cards are basically just cheap modems? |
| 13:44.50 | Katty | fxo in a tangy bbq sauce |
| 13:45.08 | Hmmhesays | indeed they are |
| 13:45.10 | bkw_ | hahaha |
| 13:45.12 | [TK]D-Fender | I was told there is a parameter for ChanIsAvail that will indicate if the target Technology is in use even though it may allow extra connections. Can anyone confirm if this is so and how to use it? |
| 13:45.15 | bkw_ | Katty, you funny |
| 13:45.16 | jalsot | bkw_: yes, but... I have 2 streams, than I make a 3rd, mixed stream and after that gsm or mp3 - which means around 90MB/min |
| 13:45.34 | bkw_ | you wanna mux em |
| 13:45.40 | bkw_ | the best way is to record slinear |
| 13:45.46 | bkw_ | then mux down to gsm |
| 13:45.55 | Katty | bkw_: and you make my head hurt. |
| 13:46.04 | *** join/#asterisk NTJOCK (~brian@txshirts.com) |
| 13:46.07 | bkw_ | you can't natively mux gsm |
| 13:46.11 | Zeeek | Katty long time no see |
| 13:46.12 | NTJOCK | hello |
| 13:46.16 | jalsot | :\ |
| 13:46.19 | Katty | Zeeek! |
| 13:46.19 | bkw_ | OMG another brian |
| 13:46.41 | bkw_ | NO DADDY NO!!! |
| 13:47.07 | jalsot | bkw_: would i be possible to store muxed stream from res_monitor? |
| 13:47.08 | NTJOCK | I'm getting this annoying "timeout, but no rule 't' in context blah. |
| 13:47.21 | NTJOCK | It's occurring during a macro that handles some phones in a common area that don't have vm |
| 13:47.24 | bkw_ | jalsot, it would be more resource intensive than you think |
| 13:47.24 | jalsot | I know, it's not possible right now... |
| 13:47.37 | jalsot | bkw_: either with sln? |
| 13:47.49 | jalsot | is mixing CPU intensive? |
| 13:47.51 | [TK]D-Fender | NTJOCK : can you put your extensions.conf in www.pastebin.ca for us to look at. |
| 13:48.02 | cochi | any way in * to "fake" ringing (catching a code) before real ringing? |
| 13:48.05 | jalsot | unfortunately I don't have audio coding background :( |
| 13:48.07 | robin_sz | so to configure this system to see my fxo card .. is there some handy tool? |
| 13:48.12 | NTJOCK | I can, but it's very complicated and makes extensive use of macros. |
| 13:48.13 | newl | NTJOCK: add a 't' rule with Nop then if it's that annoying. :) |
| 13:48.16 | bkw_ | jalsot, with slinear the very basic of muxing slinear is adding the two arrays of values together... |
| 13:48.19 | bkw_ | then dividing by two |
| 13:48.36 | bkw_ | you can also raise the volume or lower it by multiplying by 2's or dividing by 2's |
| 13:48.45 | bkw_ | slinear is NICE |
| 13:48.49 | bkw_ | me luv it long time |
| 13:49.08 | NTJOCK | newl: I tried that, but it doesn't like it in the context, and it doesn't like it in the macro. |
| 13:49.14 | jalsot | bkw_: but it doesn't sound very CPU hungry |
| 13:49.20 | NTJOCK | newl: the right spot for it is in the macro. |
| 13:49.47 | NTJOCK | essentially what's happening is there are two macros used for a sipura 841 |
| 13:49.56 | NTJOCK | one access the primary line |
| 13:50.03 | NTJOCK | and forwards to the second if the primary is busy |
| 13:50.19 | bkw_ | jalsot, its more disk hungry than CPU |
| 13:50.24 | NTJOCK | the second macro tries the second line, and if it gets confused it dumps. |
| 13:50.27 | shido | anyone remember how to change every file in a directory from buzz45454 to buzz54545.tar ? (2000 fils) |
| 13:50.33 | bkw_ | jalsot, i'm sure if you sweet talk anthm |
| 13:50.38 | bkw_ | he could whip you up a patch to do what you want |
| 13:50.39 | NTJOCK | I'm in the process of modifying it to go to a customer service voicemail, but I keep getting this t error. |
| 13:50.41 | jalsot | bkw_: my idea: store temporary sound files onto tmpfs |
| 13:50.48 | [TK]D-Fender | NTJOCK : Try using "ChanAvail" for that with both registrations. |
| 13:50.51 | Katty | kinky. |
| 13:50.53 | bkw_ | you don't gain much really |
| 13:50.58 | jalsot | but tmpfs while uses RAM has limited capacity |
| 13:51.01 | NTJOCK | chanavail? |
| 13:51.04 | jalsot | bkw_: why? |
| 13:51.08 | NTJOCK | hmm, i'll have to go see if that's in the wiki. |
| 13:51.10 | bkw_ | because of ram limits |
| 13:51.13 | NTJOCK | hadn't heard of that. |
| 13:51.14 | NTJOCK | :) |
| 13:51.22 | jalsot | bkw_: you mean, capacity limits? |
| 13:51.28 | bkw_ | yep |
| 13:51.32 | bkw_ | unless you have 10 gigs of rram |
| 13:51.34 | bkw_ | I woudln't do that |
| 13:51.44 | NTJOCK | if I dial an unavailable channel will that cause a timeout issue? |
| 13:51.53 | jalsot | what I calculated, 1.5GB would be enough for 15minutes/30channel |
| 13:51.56 | [TK]D-Fender | exten => 1234,1,ChanIsAvail(SIP/2500&SIP/2501) |
| 13:52.06 | jalsot | which can be enough for this deployment |
| 13:52.14 | [TK]D-Fender | if you dial an unavail channel it will show busy. |
| 13:52.19 | [TK]D-Fender | just like normal. |
| 13:52.22 | bkw_ | jalsot, multiply that by two |
| 13:52.26 | NTJOCK | right.... and should progress as if busy |
| 13:52.31 | bkw_ | remember you have two channels per call going to disk |
| 13:52.36 | jalsot | in other way - not from the capacity point of view - will itt help on performance? |
| 13:52.42 | [TK]D-Fender | If ChanisAvail doesn't find one you can just right to VM |
| 13:52.44 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 13:52.47 | bkw_ | still won't help much on performance |
| 13:52.48 | jalsot | yes, everything calculated into |
| 13:53.01 | NTJOCK | ah, so you go chanavail, and if it finds one then it goes there, and if not it goes to the next item? |
| 13:53.20 | jalsot | bkw_: what I read is that asterisk doesn't like disk activity, tmpfs would help on it, or not? |
| 13:53.33 | bkw_ | no asterisk will do fine |
| 13:53.39 | bkw_ | it all depends on how you set it up |
| 13:53.55 | jalsot | :) |
| 13:54.02 | NTJOCK | [TK]D-Fender: so in your exten above, if 2500 and 2501 are unavailable then it goes to 1234,2, but if 2500 is busy and 2501 is available then it goes to 2501? |
| 13:54.06 | bkw_ | like I said sweet talk anthm |
| 13:54.07 | jalsot | bkw_: how would you set up? ;) |
| 13:54.22 | bkw_ | i'm sure he could whip up something to mux them and output in realtime to disk in gsm or something already muxed |
| 13:54.36 | bkw_ | lowering your disk activity considerably |
| 13:54.43 | jalsot | bkw_: when is anthm used to be on irc? |
| 13:54.56 | bkw_ | expect him to pop online any moment |
| 13:55.03 | jalsot | thx :) |
| 13:55.10 | bkw_ | when I say sweet talk... that means PAY HIM |
| 13:55.10 | bkw_ | haha |
| 13:55.14 | bkw_ | just in case you questioned that |
| 13:55.41 | jalsot | if you had to do a box with 30channel call recording, how would you make it? |
| 13:56.02 | bkw_ | how big a box? |
| 13:56.30 | bkw_ | I would dedicate a drive on its own controller for recording for one |
| 13:56.55 | bkw_ | use SCSI U320 if you can |
| 13:57.32 | jalsot | bkw_: budget is for SATA and P4 :( |
| 13:58.42 | Hmmhesays | how much |
| 13:59.19 | bkw_ | SCSI isn't that expensive anymore |
| 13:59.20 | *** join/#asterisk docelm0 (~docelm0@67.106.194.90.ptr.us.xo.net) |
| 13:59.25 | Hmmhesays | because you can get an expensive p4 with sata |
| 13:59.31 | wasim | bah, 30 channels of gsm only compute to 48 KB/s, any drive can sustain that |
| 13:59.32 | bkw_ | its like this.. you're in a race.. you have the budget for a pinto.. but to win you need the porsche |
| 13:59.35 | docelm0 | Is something wrong with the lists? |
| 13:59.41 | bkw_ | wasim, no slinear |
| 14:00.01 | wasim | again, 240 KB ... thats no big sweat |
| 14:00.03 | Makenshi | scsi is allright, fiberchannel is the best option :> |
| 14:00.08 | bkw_ | 30 channels of slinear? |
| 14:00.10 | bkw_ | times two |
| 14:00.22 | wasim | 480 KB ... keep in coming |
| 14:00.28 | bkw_ | thats more than that |
| 14:00.35 | jalsot | we had 1st this idea: to have SCSI disc for online storing and to store final mp3 files to SATA, because 200GB SCSI is still not cheap [RAID1] |
| 14:00.35 | bkw_ | slinear is close to 128k |
| 14:00.52 | Nuxi | Acutally, hire a java developer. He/She/It will tell you that you can always justify spending more on hardware because it is cheaper than man hours. |
| 14:01.04 | bkw_ | hahahahahah |
| 14:01.05 | bkw_ | so true |
| 14:01.06 | jalsot | slinear is 16bit/8kHz, right? |
| 14:01.33 | bkw_ | slinear isn't 64k if thats where you're going.. its a bit more |
| 14:01.40 | Hmmhesays | heh It, ever seen a wookie java dev? |
| 14:01.57 | Hmmhesays | i'm so tired that's all that came to mind |
| 14:02.12 | *** join/#asterisk [Jedi] (~fdsaasdf@ns1.flozano.com) |
| 14:02.14 | Nuxi | Of course the response to the java programmer is always. 1 and a half young programmers is cheaper than you. |
| 14:02.22 | bkw_ | 875 kilobyte is what you're looking at for 30x2 channels of slinear or there abouts. |
| 14:02.29 | jalsot | bkw_: as far as I know, slinear is 16bit/8kHz, so 128kbit/s for one call leg. 256kbit/s for 2 legs |
| 14:02.50 | bkw_ | well if you can get a patch to mux them in realtime and barf out gsm in realtime |
| 14:02.54 | bkw_ | you'll need more CPU |
| 14:03.09 | bkw_ | get something with hyper threading |
| 14:03.30 | Nuxi | hyper threading is not safe... |
| 14:03.34 | bkw_ | yes it is |
| 14:03.43 | bkw_ | I don't know what you have been smokin but it sure as hell is |
| 14:03.45 | Nuxi | could lead to crypto leak. |
| 14:04.01 | jalsot | I still have a big question: 2xslinear than sox than lame or online gsm+sox |
| 14:04.04 | bkw_ | ok but if you have someone on your box you got bigger issues |
| 14:04.21 | *** join/#asterisk jcims (~jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 14:04.29 | newl | Expert troll. |
| 14:04.33 | *** part/#asterisk jcims (~jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 14:04.37 | bkw_ | its like you have a lock on your door to keep crazy people out.. but if you have a crazy person that has a key.. you're fucked. mmmkay |
| 14:04.37 | [Jedi] | I need to know with AGI the exact time in which a call is answered, but in real-time, not by reading the ANSWEREDTIME |
| 14:04.40 | [Jedi] | is that possible? |
| 14:04.47 | bkw_ | [Jedi], not with AGI |
| 14:04.55 | [Jedi] | great, absolutely great |
| 14:05.02 | [Jedi] | how can I do it then? |
| 14:05.06 | bkw_ | AGI - > Anogher Great Idea |
| 14:05.10 | bkw_ | C baby. |
| 14:05.22 | bkw_ | AGI -> Another Goofy Idean? |
| 14:05.24 | bkw_ | er Idea |
| 14:05.30 | Hmmhesays | you might be able to get it from the cdr's on hangup |
| 14:05.31 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) |
| 14:05.35 | [Jedi] | don't mind coding in C, but don't know where to start |
| 14:05.36 | bkw_ | he said REALTIME |
| 14:05.38 | bkw_ | oh oh oh |
| 14:05.40 | bkw_ | I know how you can get it |
| 14:05.42 | bkw_ | muhahahahahah |
| 14:05.45 | bkw_ | you running CVS head? |
| 14:05.48 | [Jedi] | no |
| 14:05.51 | Hmmhesays | heh, yeah he did... i'm going to stop answering questions now |
| 14:05.52 | [Jedi] | stable + bristuff |
| 14:05.52 | bkw_ | damn |
| 14:05.54 | bkw_ | shot that down |
| 14:05.58 | bkw_ | NEXT!!! |
| 14:06.03 | [Jedi] | hehehehe |
| 14:06.14 | [Jedi] | no chances of survival for me then? |
| 14:06.20 | bkw_ | use head |
| 14:06.25 | Katty | sure there is jedi |
| 14:06.28 | Katty | use The Force |
| 14:06.30 | [Jedi] | head + bristuff works? |
| 14:06.37 | bkw_ | [Jedi], if you hand patch it in |
| 14:06.41 | bkw_ | its not trivial |
| 14:06.43 | bkw_ | but you can do it |
| 14:06.45 | coppice | bkw_: as it says on huge signs an Manilla airport "death to patent lawyers" |
| 14:06.46 | [Jedi] | brrr |
| 14:06.47 | Nuxi | Actually, the unique id of the call is based on the time stamp. Couldn't we reverse hack the time it started from that? |
| 14:06.57 | [Jedi] | things go worse :D |
| 14:06.57 | bkw_ | coppice, HAHAHAHAHAHAHAHAHAHA |
| 14:07.00 | coppice | bkw_: or was that drug trafficers |
| 14:07.15 | bkw_ | either way.. J2 is evil |
| 14:07.24 | Katty | bkw_: also, when you get done there, i'd like some documentation about cluecon |
| 14:07.25 | bkw_ | one of their patents is getting re-examined |
| 14:07.39 | Katty | bkw_: something i can give to teh owner and say, Send Me Here |
| 14:07.40 | bkw_ | may 18th that was granted |
| 14:07.55 | bkw_ | Katty, http://www.cluecon.com ;) |
| 14:08.03 | bkw_ | Katty, I'll also sell your boss for you |
| 14:08.10 | coppice | bkw_: I was surprised they are still in business. i scanned around today. looks like UMS providers are making a small comeback |
| 14:08.18 | Katty | bkw_: sell? |
| 14:08.32 | bkw_ | Katty, bug him to death till he sends you |
| 14:08.33 | bkw_ | ;) |
| 14:08.36 | bkw_ | thats what I can do |
| 14:08.38 | Katty | oh. |
| 14:08.42 | Katty | k'then |
| 14:08.47 | bkw_ | coppice, ya |
| 14:08.53 | coppice | bkw_: you'll onyl get scrap value for most bosses |
| 14:08.57 | Katty | nah, i'll go |
| 14:09.00 | Katty | i just have to scowl a little |
| 14:09.14 | bkw_ | Cluecon is not 100% asterisk oriented |
| 14:09.33 | Nuxi | But * is 100% cluecon oriented. |
| 14:09.34 | Katty | i don't suppose you'd let me attend for free |
| 14:09.44 | coppice | bkw_:you mean there are babes too? |
| 14:09.55 | bkw_ | hahaha |
| 14:09.58 | bkw_ | shhhh |
| 14:10.01 | newl | Will you submit to being hit by Cluebat for entry? |
| 14:10.02 | bkw_ | i'm trying to get Katty there |
| 14:10.11 | coppice | oh, sorry, hunky guys |
| 14:10.12 | bkw_ | we'll have cluebats there |
| 14:10.14 | Hmmhesays | cluecon is in IL huh |
| 14:10.21 | bkw_ | yes |
| 14:10.25 | Hmmhesays | hrm, not to far from here |
| 14:10.35 | bkw_ | Katty, hrm.. i'll talk to da boss |
| 14:10.42 | bkw_ | da boss = anthnm |
| 14:10.44 | Katty | i don't think the boss is going to liek that 650 attendance fee |
| 14:10.44 | bkw_ | er anthm |
| 14:10.45 | *** join/#asterisk RoyK (~roy@213.160.242.93) |
| 14:10.58 | Hmmhesays | most bosses wouldn't |
| 14:11.05 | Hmmhesays | but will suck it up and pay anyway |
| 14:11.21 | Katty | maybe if i was in st. louis |
| 14:11.24 | Katty | but this isn't st. louis |
| 14:11.26 | coppice | i'd like the 650 attendance fee. give it to me |
| 14:11.28 | bkw_ | Katty, its cheaper than VON |
| 14:11.28 | Katty | there's a cow right down the road |
| 14:11.30 | Nuxi | $650 is less than cost of man hours ... |
| 14:11.36 | bkw_ | and you might actually learn something |
| 14:11.46 | newl | Most PHB's will send you anywhere if you tell them there's marketing hype and buzzwords. 8) |
| 14:11.56 | bkw_ | OH that reminds me |
| 14:11.59 | bkw_ | when I was at VON |
| 14:12.01 | Katty | yeah, they're sending me to the vonexus thing too |
| 14:12.03 | bkw_ | I could hear people on the phone |
| 14:12.05 | Katty | i do /not/ want to go to that |
| 14:12.09 | bkw_ | man they were buzzword whores |
| 14:12.11 | bkw_ | it was quite funny |
| 14:12.24 | bkw_ | what this one guy was saying made NO sense |
| 14:12.29 | bkw_ | but it was a long string of buzwords |
| 14:12.33 | Katty | i don't wanna go! |
| 14:12.48 | bkw_ | hehe |
| 14:13.08 | Hmmhesays | yeah, I remember when i was starting out talking to people like that |
| 14:13.18 | Hmmhesays | it was like "wtf are you talking about" |
| 14:13.22 | bkw_ | haha |
| 14:13.37 | bkw_ | my bullshit alarm goes off when I hear buzword whores |
| 14:13.50 | Katty | i'm still confused :P |
| 14:13.54 | Hmmhesays | one of my favorites is "softswitch" |
| 14:14.00 | Katty | strange geeks, the lot of you |
| 14:14.12 | newl | "Speak Engrish prease!" |
| 14:14.31 | *** part/#asterisk dalfz (~dman@mojope.com) |
| 14:14.40 | *** join/#asterisk pat_lehem (~lehem@212.150.210.2) |
| 14:14.43 | newl | heh soft switch..as if there's no hardware what so ever involved..I too am always amazed at that one. |
| 14:14.44 | pat_lehem | Hello All. |
| 14:14.45 | *** join/#asterisk mogorman (~mogorman@207.111.174.1) |
| 14:14.45 | Juggie | my pet peave is SNMP... it was supposed to get information from devices... then the buzzword whores came along, and make SNMP a protocol for telling a server to do something. |
| 14:14.59 | Juggie | what happened, SNMP got ruined. |
| 14:15.02 | Hmmhesays | a year or so ago, seemed like every sales person under the sun was looking for a softswitch and they would all be talking about a different piece of equipment |
| 14:15.06 | pat_lehem | did anyone try out the NVLineDetect application mentioned in the Wiki? |
| 14:15.14 | coppice | Hmmhesays: a spandsp fan, then, eh? :-) |
| 14:15.16 | pat_lehem | for progress detection on analog lines? |
| 14:15.26 | Katty | bkw_: i have put teh page on boss's desk |
| 14:15.34 | newl | Juggie: Well, the "Management" abbreviation wasn't for any one particular reason ya think? |
| 14:15.35 | bkw_ | goodluck young jedi |
| 14:15.44 | Katty | bkw_: and if i show up, there better be vegan munchables |
| 14:15.53 | Katty | bkw_: or i will not be a happy little geek |
| 14:15.54 | Juggie | newl, what do u mean? |
| 14:16.14 | bkw_ | we'll find something mmkay |
| 14:16.17 | Hmmhesays | free admission and special meals? demanding |
| 14:16.28 | jontow | im kinda interested in the coppercom switches |
| 14:16.39 | Hmmhesays | lol j/k |
| 14:16.44 | jontow | essentially channel banks connected to a sun netra of some sort; from what i have seen |
| 14:16.46 | newl | Juggie: well, it means simple network management protocol. Management implies just that. |
| 14:16.53 | Katty | Hmmhesays: i'll demand /you/ in a minute |
| 14:17.00 | Katty | k, maybe nto |
| 14:17.01 | Katty | i mean not |
| 14:17.05 | Juggie | newl, SNMP was never intended to remote control devices. |
| 14:17.07 | jontow | .. which is interesting; but i've also looked a little behind the scenes at some of the scripting on the console.. its f'knugly |
| 14:17.10 | Hmmhesays | Katty: is that a threat or a promise? |
| 14:17.12 | Juggie | it was intended for retreiving information |
| 14:17.16 | Katty | Hmmhesays: why not both? |
| 14:17.22 | Hmmhesays | :D |
| 14:17.27 | bkw_ | Hmmhesays, hrm.. ok i'm not going there... NEXT!!! |
| 14:17.44 | Juggie | and as for vegan? :) vegtables are for suckers |
| 14:17.47 | Juggie | bring on the beef |
| 14:17.50 | jontow | they don't follow ANY of the UNIX ideals/conventions.. no manpages for anything custom, instead poorly formatted txt files that you have to run a command to, assumes 'more' as $PAGER and doesn't accept a change there, etc.. just not polished |
| 14:17.56 | Katty | mmm, suckers. |
| 14:18.10 | Hmmhesays | i prefer chicken, grilled, bbq sauce |
| 14:18.14 | jontow | 'tis a big hack, though a working one with lots of options and complicated scripts :) |
| 14:18.23 | Katty | Juggie: i bet you don't even know what vegan food is if it smacked you upside the back of the head |
| 14:18.28 | jontow | asterisk is way more clean, which, given some background there too.. is scary :D |
| 14:18.30 | Hmmhesays | over a fire, wood or charcoal |
| 14:18.40 | Juggie | Katty, i know its not beef, chicken, or pork |
| 14:18.47 | Katty | Juggie: k |
| 14:19.00 | Hmmhesays | or anything having to do with animals at all |
| 14:19.00 | Juggie | and i for one cannot live without beef chicken and pork :) |
| 14:19.09 | coppice | Juggie: SNMP was always for remote control. Simple Network Management Procotol. The M wouldn't make much sense without control |
| 14:19.48 | *** join/#asterisk brettnem (~brettnem@user-0ccsr4m.cable.mindspring.com) |
| 14:20.00 | Hmmhesays | speaking of spandsp, I wonder if I can get that working on this p233 i got sitting here |
| 14:21.22 | newl | Juggie: I beg to differ with your thinking on SNMP. Read the first paragraph of RFC1067 dated Aug 1988. :) |
| 14:22.46 | brettnem | hello all! |
| 14:22.47 | cochi | any ideas on how to listen for dtmf during some "fake ring" period? i'm kinda stuck :| |
| 14:22.47 | cyberknet | heh heh heh =) <-- Got the cable box he wanted for no additional charge |
| 14:23.24 | brettnem | cochi: use the "d" option for the dial parameter |
| 14:23.38 | brettnem | cochi: 'd' -- allow the calling user to dial a 1 digit extension while waiting for a call to |
| 14:23.39 | Hmmhesays | it's a conference.... i'm pretty sure they'll let you ask questions |
| 14:23.41 | brettnem | <PROTECTED> |
| 14:23.41 | gordonjcp | coppice: was it not "Simple Network Monitoring Protocol"? |
| 14:23.45 | brettnem | <PROTECTED> |
| 14:24.07 | jhava | Hello, has anyone experienced a segmetation fault in asterisk for using G729 against an AS5300 in SIP ? I'm having this problem. Also, any other codec I use gives me incompatible media. AS5300 configured for multiple codecs, so is Asterisk. Tried G711u/A G723 and G.729. Any clues ? |
| 14:24.10 | Hmmhesays | unless bkw_ starts slapping people around |
| 14:24.21 | cochi | oh? that's not on the wiki. thanks brett |
| 14:24.39 | jontow | cochi; from the CLI: "show application dial" :) |
| 14:24.41 | brettnem | cochi: remember --> show applications <application_name> |
| 14:24.47 | gordonjcp | ah, no it wasn't |
| 14:24.49 | jontow | always gonna give you more up-to-date information over the wiki :) |
| 14:24.51 | cochi | i'm very web-focussed. sry ;)) |
| 14:24.55 | *** join/#asterisk file[class] (~root@66.199.241.90) |
| 14:24.57 | Juggie | gordonjcp, thats what *I* thought. |
| 14:25.01 | file[class] | yay |
| 14:25.05 | jontow | 'tis ok :) just another avenue you can take to double-check your work |
| 14:25.08 | cochi | and i'm a noob. learning each second ;))) |
| 14:25.12 | cochi | k *happy* |
| 14:25.34 | coppice | Like I said: RFC1067 begins |
| 14:25.36 | coppice | This memo defines a simple protocol by which management information |
| 14:25.37 | coppice | for a network element may be inspected or altered by logically remote |
| 14:25.38 | bewest | and it's a lot faster than searching the wiki |
| 14:25.39 | coppice | users. |
| 14:26.00 | newl | What they're thinking about is SGMP which is RFC1028. |
| 14:28.31 | cochi | yuk. lemme guess. d option just on HEAD mh |
| 14:29.22 | bkw_ | someone wants to be slapped? |
| 14:29.30 | bkw_ | file[class], do you have the list? |
| 14:29.50 | Katty | bkw_: you won't loose me at cluecon, will you? |
| 14:29.51 | *** join/#asterisk rephorm (~brian@ip67-95-13-60.z13-95-67.customer.algx.net) |
| 14:30.26 | file[class] | bkw_: I'm in class, that's on my laptop in my backpack |
| 14:30.38 | file[class] | I can probably swing online in, like, 50 minutes |
| 14:31.13 | rephorm | has anyone ever had problems with Playback()/Background() audio not working? all other audio is going fine, but things like VoicemailMain() just play dead air. i've checked permissions on everything, and can't seem to locate the problem. was working fine up until a power outage last night |
| 14:32.19 | *** join/#asterisk hatton (~hatton@24-54-72-88.kntnny.adelphia.net) |
| 14:32.40 | hatton | morning/afternoon/evening folks |
| 14:32.56 | Godsey | anyone using Radius for call accounting with Asterisk? |
| 14:33.23 | gordonjcp | no but I'm about to start look at it |
| 14:33.49 | *** join/#asterisk drumkilla (~russell@user-24-214-77-225.knology.net) |
| 14:33.49 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 14:34.13 | file[class] | drumkilla! |
| 14:34.39 | Godsey | when following the dialplan |
| 14:34.47 | Godsey | if I have 2,Dial... |
| 14:34.58 | Godsey | 3,AGI(this) .... 4,Hangup |
| 14:35.02 | drumkilla | hi file[class] |
| 14:35.13 | Godsey | if both people hang up on the dial, will it follow through to the agi or simply abort? |
| 14:35.33 | file[class] | waddup? |
| 14:35.35 | hatton | need a hand getting an internal SIP phone to connect to * |
| 14:35.44 | hatton | name=Sip1 |
| 14:36.03 | drumkilla | file[class]: sitting at my desk, eating a biscuit |
| 14:36.08 | *** join/#asterisk cluecon (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net) |
| 14:36.16 | cluecon | bkw_: you here? |
| 14:36.28 | cochi | god i'm silly. now i read "sitting at my desk, drinking a biscuit"... |
| 14:36.31 | cochi | :) |
| 14:36.36 | cochi | probably too warm or so ;) |
| 14:39.31 | drumkilla | is there a mysql app? |
| 14:40.00 | Juggie | drum, for what? |
| 14:40.08 | file[class] | this system is so so... not working good |
| 14:40.13 | file[class] | it dislikes Win2k3 Small Business Server |
| 14:40.19 | drumkilla | well .. executing sql statements, i don't know |
| 14:40.24 | drumkilla | someone is asking me |
| 14:40.28 | drumkilla | i've never used one ... |
| 14:40.43 | drumkilla | looks like he wants to set vars from sql statements |
| 14:41.00 | drumkilla | <-- obviously a db expert. |
| 14:41.01 | Juggie | drumkilla, mysql it should be |
| 14:41.09 | Godsey | I set vars from mysql using agi :) |
| 14:41.21 | drumkilla | Godsey: yeah, that's what I would do if I were writing it ... |
| 14:41.30 | *** join/#asterisk Essobi (kstone@75.137.26.216.host.teledvance.com) |
| 14:42.02 | Essobi | Hmm. Anyone think of a way off hand to tell if a call is offhook or not in an AGI? I can't think of a way to do it... |
| 14:42.20 | Juggie | if a call is active or not you mean? |
| 14:42.30 | Essobi | yea yea |
| 14:42.47 | Juggie | here's what i do |
| 14:42.47 | Juggie | if ($result['result']==-1) |
| 14:42.47 | Juggie | { |
| 14:42.47 | Juggie | $result=$agi->agi_verbose("Detected Hangup"); |
| 14:42.47 | Juggie | $state=CONFCLEANUP; |
| 14:42.48 | Juggie | } |
| 14:43.24 | Essobi | $result['result'] is populated by what? |
| 14:43.48 | Essobi | CHANNEL STATUS? |
| 14:44.09 | *** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 14:44.15 | Seyr | greets |
| 14:44.28 | *** part/#asterisk pat_lehem (~lehem@212.150.210.2) |
| 14:44.29 | *** join/#asterisk oej (~oej@213.204.186.40) |
| 14:44.44 | Juggie | no by every agi call you make |
| 14:44.45 | *** join/#asterisk shanky (~shanky@238.Red-80-33-29.pooles.rima-tde.net) |
| 14:44.49 | Juggie | assuming your using phpagi |
| 14:44.50 | shanky | hi, good afternoon |
| 14:44.54 | Seyr | I just got latest CVS and am having problems compiling zaptel. If anyone is bored, feel free to look at my error :-) http://pastebin.ca/13106 |
| 14:44.57 | Juggie | every call returns $result which is an array |
| 14:45.39 | Seyr | this is on a new system though, so I might have just forgot a lib or something |
| 14:45.57 | shanky | I'm trying to install asterisk 1.0.7 in a debian woody with kernel 2.4.18 and when I try to do "make" I get: configure: error: termcap support not found |
| 14:46.02 | shanky | make: *** [editline/libedit.a] Error 1 |
| 14:46.09 | Juggie | shanky |
| 14:46.11 | shanky | and I have done "apt-get install term-cap" |
| 14:46.16 | Juggie | this coudnt be easier :) |
| 14:46.19 | Godsey | shanky: does /lib/modules/*/build exist? |
| 14:46.20 | Juggie | oh yah |
| 14:46.24 | Juggie | how about termcap-devel :) |
| 14:46.43 | shanky | Juggie: E: Couldn't find package termcap-devel |
| 14:46.53 | drumkilla | libtermcap |
| 14:46.53 | Juggie | apt-cache search termcap |
| 14:46.59 | drumkilla | i think |
| 14:47.03 | Juggie | then install it |
| 14:47.16 | shanky | Godsey: no, there is no build, thanks |
| 14:47.32 | *** join/#asterisk Poincare (~jefffnode@dD5779BD2.access.telenet.be) |
| 14:47.43 | Godsey | I think you simply link that to /usr/src/linux |
| 14:47.48 | Godsey | but I'm not positive :) |
| 14:48.08 | Juggie | libtermcap - A basic system library for accessing the termcap database. |
| 14:48.08 | Juggie | libtermcap-devel - Development tools for programs which will access the termcap database. |
| 14:48.14 | *** join/#asterisk jeffik (~Jeff@69.158.0.210) |
| 14:48.26 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 14:48.37 | file[class] | yada yada yada |
| 14:48.41 | cjk | hi, does anyone know a softphone compatile to the siemens m34 |
| 14:48.50 | shanky | Godsey: the problem is that "apt-get install kernel-source" I only have a .tar.bz2 in /usr/src |
| 14:49.06 | shanky | I'm rigth now uncompressing |
| 14:49.15 | Godsey | shanky: I don't do debian anymore sorry :) |
| 14:49.37 | Godsey | not since potato |
| 14:50.17 | Seyr | I just got CVS on a new system and tried to compile zaptel, but im getting an error. http://pastebin.ca/13106 |
| 14:50.34 | Seyr | make: *** SUBDIRS=/usr/src/zaptel: No such file or directory. Stop. |
| 14:50.34 | shanky | Juggie: I think I have a problem with my sources.list |
| 14:52.28 | *** join/#asterisk stkn (~stkn@stkn.developer.gentoo) |
| 14:54.29 | tzafrir | shanky, is it on debian? |
| 14:54.44 | tzafrir | if so: you don't need kernel-source |
| 14:54.57 | Juggie | it is debian |
| 14:55.03 | Juggie | i forget the command to make the proper link |
| 14:55.25 | tzafrir | you need kernel-headers-`uname -r` |
| 14:56.07 | tzafrir | It has all the files needed to build extra modules |
| 14:56.51 | tzafrir | shanky, but do you try to build zaptel-modules from the standard debian package? |
| 14:57.11 | tzafrir | If so: |
| 14:57.24 | tzafrir | http://tzafrir.org.il/rapid/APT.html |
| 14:59.46 | makhtar | Has anyone had problems with the lcd display on a Polycom IP500? One of ours has gone blank and nothing seems to fix it. Otherwise, the phone is working correctly. |
| 15:00.08 | dudes | http://www.voip-info.org/wiki-Asterisk+Linux+Debian |
| 15:00.18 | Godsey | makhtar: even if you factory reset it? |
| 15:00.41 | makhtar | my co-worker said he tried that |
| 15:00.44 | JerJer | large hammer |
| 15:00.52 | makhtar | hammer is being considered |
| 15:00.56 | tzafrir | Seyr, strange. It should run there make SUBDIRS=$(PWD) modules, right? |
| 15:01.07 | Godsey | makhtar: when you call polycom to get waranty replacment |
| 15:01.16 | Godsey | tell them it's brand new out of box, never used |
| 15:01.17 | makhtar | godsey. what is the procedure for the factory reset? |
| 15:01.18 | shanky | tzafrir: rigth now I don't need zaptel |
| 15:01.24 | Godsey | jumpers |
| 15:01.37 | Godsey | I don't remember the right jumpers to cross |
| 15:01.44 | makhtar | godsey, then i doubt he did that |
| 15:01.57 | *** join/#asterisk Assid (~assid@203.115.64.62) |
| 15:01.58 | Godsey | if you don't tell them it's fresh out of the box never used they won't replace it |
| 15:01.59 | Assid | heya |
| 15:02.00 | makhtar | anywhere i might look to find that? |
| 15:02.11 | Godsey | and they'll make you return it to your place of purchase (pain in ass normally) |
| 15:02.24 | Assid | i am having a small issue configuring sip.. if i call my own extension.. it yet goes to another one |
| 15:02.30 | makhtar | ive gotten the impression they are difficult to deal with. thanks for the advice. |
| 15:03.18 | hatton | dumb question - do I have to have a working incoming configuration before I can get a working internal config? |
| 15:03.25 | makhtar | any idea about which jumpers need to be crossed or where i can find that info? we looked pretty carefully in the polycom documentation and didnt see anything like that? |
| 15:03.43 | Godsey | ours have a brick of pins under the phone |
| 15:03.54 | Assid | i have exten => 2530,1,Dial(SIP/user/${EXTEN}) and similar for 2531 |
| 15:03.55 | makhtar | yep, 6 pins i believe |
| 15:04.02 | hatton | in other words I've got the SIP phone that I want to use as an extension and a FWD account I'm using to test... which comes first with the setup? |
| 15:04.34 | Assid | from 2531 if i dial 2531 or 2530 .. it always goes to 2530 |
| 15:04.34 | Godsey | off to work |
| 15:04.40 | makhtar | is it a particular pair or just jump them all? |
| 15:05.01 | makhtar | thanks godsey |
| 15:06.19 | heison | ~seen oej |
| 15:06.20 | jbot | oej is currently on #asterisk (21m 51s) |
| 15:10.45 | Seyr | tzafrir: yeh |
| 15:11.46 | *** part/#asterisk shanky (~shanky@238.Red-80-33-29.pooles.rima-tde.net) |
| 15:11.49 | Katty | tzanger: were you supposed to get me a recipe last night? |
| 15:12.01 | Katty | Ariel_: or was it you who was supposed to get me a recipe? |
| 15:12.01 | cluecon | ~cluecon |
| 15:12.03 | jbot | you are, like, http://www.cluecon.com |
| 15:15.47 | Seyr | tzafrir: i even copied over the zaptel dir from a working server and i get the same error. i must be missing a lib or something |
| 15:17.24 | *** join/#asterisk blop (blop@2001:6f8:204:33:bbbb:bbbb:bbbb:bbbb) |
| 15:17.25 | Ariel_ | Katty, not me. Hello hope all is well today? |
| 15:17.32 | Katty | Ariel_: so far |
| 15:18.29 | Katty | Ariel_: do you have a breakfast recipe of any kind? |
| 15:18.53 | Katty | bye |
| 15:18.59 | Ariel_ | Katty, unless you like Cuban or phillipino food yes. |
| 15:19.14 | Katty | oh. |
| 15:19.28 | coppice | surely nobody likes fillipino food |
| 15:19.39 | Katty | and surely nobody likes coppice either |
| 15:20.10 | Seyr | Anyone have any clue why I would get this error when compiling Zaptel? http://pastebin.ca/13106 |
| 15:20.14 | tzafrir | Seyr, try remming-out the line in the makefile that says 'PWD=$(shell pwd) |
| 15:20.37 | tzafrir | It is totally unnecessary |
| 15:20.52 | coppice | Mangos in .ph are the most wonderful things in the world. those grill blue something fish are wonderful too. most other stuff in .ph seems to be seriously abused chicken :-) |
| 15:21.14 | tzafrir | Seyr, is it HEAD or stable? |
| 15:21.17 | Seyr | tzafrir: I get the same error |
| 15:21.19 | Seyr | head |
| 15:21.19 | coppice | the only place in earth where macdonalds serve rice |
| 15:21.29 | Seyr | it compiles fine on another system though :-( |
| 15:21.34 | Seyr | same head |
| 15:21.43 | Ariel_ | coppice, lumpia |
| 15:21.54 | Seyr | i think im missing a library or something, just no clue what |
| 15:21.57 | robin_sz | never eat at the golden arches |
| 15:22.02 | Seyr | libpri and asterisk compile just fine |
| 15:22.14 | Katty | mcdonalds isn't vegan :< |
| 15:22.31 | Katty | they even have a disclaimer on their website saying nothing is guaranteed vegetarian |
| 15:22.47 | bublbobl | Hard to make faxs work... Has anyone already experienced "TIFF directory is missing required "StripOffsets" field" after rxfax got the fax ? |
| 15:22.51 | *** join/#asterisk riksta (~rick@host81-155-216-40.range81-155.btcentralplus.com) |
| 15:22.56 | RoyK | ~seen zoa |
| 15:22.57 | jbot | zoa <~zoa@ip-212-239-162-34.dsl.scarlet.be> was last seen on IRC in channel #asterisk, 4d 57m 5s ago, saying: 'away from what ? having all wrong ?'. |
| 15:23.11 | robin_sz | bublbobl: just use a real faxmodem and hylafax |
| 15:24.11 | coppice | bublbobl: one of the buggy libtiffs does that. it will also do it with Hylafax |
| 15:24.28 | bublbobl | robin_sz> thanx, next step is to have hylafax working w/ hylafax, but 4 now I have incoming sip fax I have to email, and i don't think hylafax can manage sip fax :-( |
| 15:25.05 | coppice | nothing can manage SIP fax unless its T.38.... which * can't yet handle |
| 15:25.08 | bublbobl | coppice> I've read this but don't remember wheere :-$ Do you remember which versio is the best ? |
| 15:25.20 | robin_sz | well, it can, but fax over sip/iax seems to be not worth trying |
| 15:25.37 | *** join/#asterisk km- (~pgrace@brdgw1.rttx.com) |
| 15:25.50 | km- | howdy! |
| 15:26.07 | *** join/#asterisk Jas_Williams (~Jason@host217-43-100-176.range217-43.btcentralplus.com) |
| 15:26.43 | coppice | bublbobl: it depends how they have been patched. there are fixes for the various versions of libtiff, but thety keep breaking things. watch out for having more than one version on your machine. that causes lots of people problems |
| 15:26.57 | bjohnson | damn. my sixtel DIDs aren't working |
| 15:27.12 | Katty | km-: mew |
| 15:28.12 | robin_sz | bublbobl: remember, a fax is basically a 14.4K modem ... trying to squeeze ti down a 3Kbits compressed-to-hell GSM stream leads to "issues" |
| 15:28.31 | bublbobl | coppice> Ok, i'll check. I removed my sarge pakages (3.7.2) to install tarball version 3.5.7 |
| 15:28.53 | bublbobl | bublbobl> no, i don't have issue bandwidth so I will use pcma or pcmu (80kbps per call) |
| 15:29.03 | coppice | 3.5.7 is usually a safe bet |
| 15:29.24 | coppice | pcma and pcmu guarantee nothing with FAX |
| 15:29.45 | tzanger | Katty: you didn't email me |
| 15:29.53 | tzanger | I'm off to pittsburgh right now though |
| 15:29.59 | bewest | she doesn't want your chile |
| 15:30.09 | tzanger | bewest: then what's she asking for |
| 15:30.12 | coppice | even most of the ATAs which have a FAX detect, and lock down the jitter buffer mode have other things which screw up. |
| 15:30.26 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
| 15:30.35 | bublbobl | coppice&robin_sz> thank you |
| 15:30.36 | bjohnson | tzanger: have you checked your sixtel DIDs lately? |
| 15:30.38 | Katty | tzanger: i don't have your email address |
| 15:30.45 | Katty | tzanger: have fun (= |
| 15:30.56 | tzanger | Katty: akohlsmith@mixdown.ca; I told you what it was yesterday :-=) |
| 15:31.01 | tzanger | bjohnson: worked a couple days ago, why? |
| 15:31.08 | bjohnson | mine just ring |
| 15:31.14 | bjohnson | (on incoming calls) |
| 15:31.15 | Katty | tzanger: i obviously missed that line. thx |
| 15:31.28 | tzanger | ok I'm off to pittsburgh; latera ll |
| 15:31.31 | tzanger | er later all |
| 15:31.33 | bewest | later |
| 15:31.34 | Katty | bi. |
| 15:31.39 | *** join/#asterisk Assid (assid@203.115.64.58) |
| 15:31.42 | Assid | heya |
| 15:31.48 | bjohnson | beye |
| 15:32.09 | Assid | can someone help me set this up? |
| 15:32.13 | *** join/#asterisk kFuQ (~somedude@c-24-17-224-78.hsd1.wa.comcast.net) |
| 15:32.28 | Assid | i managed to make 2 extensions |
| 15:32.34 | *** join/#asterisk YaP (~YaP@host-84-222-179-24.cust-adsl.tiscali.it) |
| 15:32.37 | YaP | hi |
| 15:32.42 | Assid | however .. all my calls always end up going to the first extension |
| 15:33.37 | YaP | what's the right syntax to use capi channels for outgoing calls? Dial(CAPI/msn:${EXTEN}) ? |
| 15:34.35 | Jas_Williams | YaP: That is correct or Dial(CAPI/msn:b${EXTEN}) |
| 15:34.52 | Nuxi | grrrrrrr. my bank now REQUIRES internet explorer 4 or better for security reasons for online banking. That's retarded. Please use the least secure browser for security purposes. |
| 15:35.19 | bewest | Nuxi, that's awful |
| 15:35.26 | bewest | time to switch banks? |
| 15:35.26 | YaP | so what's wrong with exten => _050.,1,Dial(CAPI/mynumber:${EXTEN}) ? |
| 15:35.31 | heison | ~seen oej |
| 15:35.33 | jbot | oej is currently on #asterisk (51m 4s) |
| 15:35.33 | YaP | when i try to make a call |
| 15:35.43 | YaP | Jun 1 17:23:28 ERROR[24443]: chan_capi.c:1181 capi_request: Syntax error in dialstring. read the docs! |
| 15:35.43 | YaP | Jun 1 17:23:28 NOTICE[24443]: chan_capi.c:1238 capi_request: didn't find capi device with controller = 0 or group = 0. |
| 15:35.43 | YaP | Jun 1 17:23:28 NOTICE[24443]: app_dial.c:746 dial_exec: Unable to create channel of type 'CAPI' |
| 15:35.45 | Nuxi | They also have stupid ad banners, which could do some nasties. |
| 15:35.47 | YaP | i get sorry for the paste... |
| 15:36.05 | bewest | your bank has ad banners? |
| 15:36.14 | km- | howdy |
| 15:36.14 | blitzrage | I'm curious if anyone else has implemented a FindMe FollowMe service, and if so, how they went about it? |
| 15:36.28 | bewest | what is that service? |
| 15:36.44 | bewest | it chases a prioritized list of numbers based on availability? |
| 15:36.53 | Nuxi | It's some local ad service. |
| 15:36.59 | blitzrage | basically |
| 15:38.45 | km- | I <3 eclipse |
| 15:40.13 | Jas_Williams | YaP: post your capi.conf to pastebin.ca |
| 15:40.40 | *** join/#asterisk RoyK (~roy@217-157-34.0101.adsl.tele2.no) |
| 15:42.14 | Qwell | Nuxi: which bank? |
| 15:42.16 | YaP | http://pastebin.ca/13110 |
| 15:43.09 | Nuxi | tierone |
| 15:44.01 | Jas_Williams | YaP: under [interfaces] are you missing a line msn=yourmsn ? |
| 15:44.07 | *** join/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 15:44.16 | YaP | there isn't that line |
| 15:44.22 | YaP | i try to add it |
| 15:44.25 | Jas_Williams | you need ome |
| 15:44.40 | _omer | can I clean the database ? |
| 15:44.51 | cpatry | with the manager, is there any way to know a specific is associated with which request? |
| 15:45.00 | Jas_Williams | YaP: now restart asterisk |
| 15:45.01 | *** join/#asterisk redeemed (~minded@65.211.26.66) |
| 15:45.04 | redeemed | helllllo |
| 15:45.39 | redeemed | im trying to set asterisk up on my works network |
| 15:45.49 | redeemed | im going to be running gentoo linux |
| 15:46.03 | redeemed | i need help with what card(s) to get, as well as what phones |
| 15:46.06 | YaP | Jas_Williams: i still get the same error |
| 15:46.14 | bjohnson | bewest: just use the superdial macro |
| 15:46.24 | *** join/#asterisk MikeJ[Jayden] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 15:46.39 | bjohnson | blitzrage: ^^ that was for you |
| 15:46.44 | mrtwister | i need help. i trying to make callback application with realtime billing. there is all ok, working. but i have to dial fires to caller. i using DIAL command in agi. when calling AGI (from call file, channel is not answered (1,deadagi(...) 2,hangup). agi script does authentication and statying dialing. and stopping. how i can "imitate" dial command? |
| 15:47.11 | [Jedi] | mrtwister: you can exec dial from AGI |
| 15:47.27 | Jas_Williams | YaP: does capi info from the cli identify your card |
| 15:47.33 | bjohnson | redeemed: get the SUX2000 |
| 15:47.45 | Qwell | bjohnson: I'll bet it sucks. :D |
| 15:47.52 | redeemed | bjohnson :( |
| 15:47.58 | bjohnson | I'd buy that for a buck |
| 15:47.59 | blitzrage | bjohnson: yah, thats what I figure is just some fancy macro |
| 15:49.16 | YaP | Jas_Williams: i can receive incoming call on sip phone (xlite) |
| 15:50.34 | JerJer | mrtwister: or write a C app using the Asterisk C API |
| 15:50.46 | JerJer | and re-implement the functionality you need |
| 15:51.10 | JerJer | i do this quite a lot, because i hate 5,000 line extensions.conf files |
| 15:51.28 | Juggie | agi works quit well too |
| 15:51.33 | Jas_Williams | YaP: are you running in ptp mode ? |
| 15:51.35 | JerJer | agi doesn't scale |
| 15:51.43 | [Jedi] | i love java+agi :) |
| 15:51.45 | YaP | Jas_Williams: hmm what does it mean? |
| 15:51.55 | Juggie | JerJer, theres also res_js |
| 15:52.01 | JerJer | Juggie: fuck that |
| 15:52.03 | JerJer | hell no |
| 15:52.05 | *** join/#asterisk RoyKa (~roy@217-157-34.0101.adsl.tele2.no) |
| 15:52.20 | JerJer | not in your wettest dream |
| 15:52.26 | Juggie | then whats your solution that doesnt require recompiling * |
| 15:52.30 | Jas_Williams | Yap what type of BRI are you connected to ? PTP or PTMP (who is the provider) |
| 15:52.30 | Qwell | JerJer: morning |
| 15:52.43 | *** join/#asterisk phantam (~cchance@72.252.15.246) |
| 15:52.49 | YaP | Jas_Williams: telecomitalia |
| 15:52.55 | phantam | hi guys |
| 15:53.00 | YaP | i don't know what type of bri is it |
| 15:53.09 | *** join/#asterisk jaike (~a@203.131.137.77) |
| 15:53.11 | JerJer | Juggie: i don't recompile asterisk |
| 15:53.18 | Qwell | JerJer: Do you know if there is any type of "opt out" for a user with the new E911 crap? |
| 15:53.32 | JerJer | see asterisk-addons for how to compile asterisk modules outside of the asterisk tree |
| 15:53.34 | Qwell | I just know its gonna end up costing me money, and I simply don't want it... |
| 15:53.38 | terrapen | any Mac users here? |
| 15:53.46 | Qwell | terrapen: get that working last night? |
| 15:53.49 | terrapen | qwell: I'm still trying to debug this code :( |
| 15:53.53 | JerJer | remember asterisk is modular ... just plugin more or less .so's and run with it |
| 15:53.55 | mrtwister | JerJer, and [Jedi] i doing DIAL from AGI - but channel is unavailable before dial so - it going to hangup after :( |
| 15:53.56 | Jas_Williams | YaP: remove this line from your capi.conf isdnmode=ptmp |
| 15:53.59 | terrapen | dude, i've stared at file.c for hours |
| 15:54.02 | Qwell | heh |
| 15:54.07 | jaike | anyone know what may be causing one-way audio? called party cant hear caller |
| 15:54.16 | Qwell | jaike: nat/firewall |
| 15:54.19 | terrapen | i should just comment code as i read it |
| 15:54.28 | terrapen | that way, Asterisk will have some comments :) |
| 15:54.36 | redeemed | =] |
| 15:54.36 | YaP | Jas_Williams: still the same error |
| 15:54.36 | Qwell | jaike: nat=yes, canreinvite=no, qualify=yes |
| 15:54.37 | phantam | i have a question not sure if anyone can help ... im trying to reduce my office costs on phone bills... the local telco charges alot to call cells but local cell carrier has free mobile to mobile... so i wanted to route all calls 55xxxxxx via asterisk to a cellphone to make the call... like using a cellsocket is this possible what would it intale |
| 15:54.45 | redeemed | can anyone help me setup asterisk at my work?!? |
| 15:54.57 | terrapen | redeemed: yes |
| 15:55.02 | Qwell | redeemed: I'm sure you could find a consultant |
| 15:55.03 | jaike | qwell: where do add those? sip.conf? |
| 15:55.05 | [Jedi] | phantam: free mobile to mobile? where do you live? |
| 15:55.07 | terrapen | redemeed: My rate is $70/hr |
| 15:55.12 | Qwell | jaike: yeah |
| 15:55.13 | phantam | caribbean |
| 15:55.15 | Qwell | terrapen: cheap |
| 15:55.17 | redeemed | i dont need to actually set it up |
| 15:55.20 | jaike | ok..thanks..will try it |
| 15:55.20 | [Jedi] | wow |
| 15:55.21 | Qwell | redeemed: don't msg me...thanks |
| 15:55.26 | redeemed | i need help with what to get |
| 15:55.45 | terrapen | phantom, i can help you, too |
| 15:56.01 | Jas_Williams | YaP: This is my capi.conf http://pastebin.ca/13112 |
| 15:56.01 | phantam | ? |
| 15:56.12 | *** join/#asterisk postel (~canonical@postel.user) |
| 15:57.24 | YaP | Jas_Williams: hmm |
| 15:58.09 | phantam | terrapen u there? |
| 15:58.10 | RoyKa | anyone using SER? |
| 15:58.12 | terrapen | yeah |
| 15:58.16 | RoyKa | ~ser |
| 15:58.18 | jbot | rumour has it, ser is Sip Express Router - see http://www.iptel.org/ser/ |
| 15:58.24 | phantam | u know how to do it /whats needed |
| 15:58.30 | terrapen | phatam, i was saying that if you needed consulting, that i'm available |
| 15:58.47 | *** join/#asterisk rephorm (~brian@ip67-95-13-60.z13-95-67.customer.algx.net) |
| 15:59.04 | phantam | i was planning on just getting a handytone sip->fxo and plug the cellsocket into that |
| 15:59.27 | terrapen | that's kind of a crappy way to do it |
| 15:59.34 | phantam | it is/ |
| 15:59.43 | terrapen | why go digital <-> analog <-> digital |
| 15:59.47 | terrapen | keep it all digital |
| 15:59.51 | YaP | Jas_Williams: is pretty the same |
| 15:59.57 | phantam | they dont have a cheap tdma that is sip |
| 16:00.00 | YaP | btw are you sure the problem is in capi.conf? |
| 16:00.01 | phantam | or not that i could find |
| 16:00.08 | YaP | and not in extension.conf sytax? |
| 16:00.11 | phantam | i found the gsm one but not the tdma one |
| 16:00.12 | terrapen | tdma? heh |
| 16:00.22 | phantam | ya i found gsm but not tdma |
| 16:00.26 | terrapen | well, yeah, maybe your cellsocket then |
| 16:00.30 | terrapen | tdma is old skool |
| 16:00.31 | phantam | lol |
| 16:00.32 | Hmmhesays | mamma says cell sockets are the devil |
| 16:00.32 | Jas_Williams | YaP: Here is my sample dial line exten = _3XXXXX,1,Dial(CAPI/330417:b${EXTEN}) |
| 16:00.37 | phantam | Hmmhesays why? |
| 16:00.53 | terrapen | any asterisk devs around? |
| 16:01.04 | Hmmhesays | I always end up having to do installs where I swear people are terminating traffic out of their grass huts |
| 16:01.16 | MikeJ[Jayden] | what's an asterisk devs |
| 16:01.25 | terrapen | heh :) |
| 16:01.29 | phantam | terrapen can it work like that tho? to have asterisk send 55xxxxx# to the handytone? cause the cellsocket requires the # at the end to "press send" |
| 16:01.39 | YaP | Jas_Williams: i don't know... |
| 16:01.47 | MikeJ[Jayden] | phantam, sure.. |
| 16:01.56 | phantam | i dont wanna have to dial a # from my handsets that would be gay |
| 16:02.04 | Hmmhesays | and of course they have purchased the cheapest pile of /shi@t they can find |
| 16:02.24 | terrapen | phantam, sure |
| 16:02.24 | _omer | how to clean or make the database empty that I see,,,,,,with "database show" |
| 16:02.40 | MikeJ[Jayden] | _omer, dbdelltree |
| 16:03.28 | *** join/#asterisk map71 (~map@plmras2-194.ras.plm.cantv.net) |
| 16:04.22 | *** join/#asterisk meppl (mephisto@p54AAC0E7.dip.t-dialin.net) |
| 16:04.34 | _omer | Mike: at CLI? |
| 16:06.33 | map71 | hi |
| 16:06.33 | phantam | Hmmhesays so u saying the cellsocket + handytone is bad choice ? i mean its only 150$ total lol |
| 16:06.55 | Hmmhesays | phantam: i'm not saying anything, more or less complaining because i'm tired and irritated |
| 16:06.59 | Hmmhesays | :D |
| 16:07.02 | phantam | lol |
| 16:07.39 | phantam | ok so im gonna go ahead and order the darn handytone and cellsocket... and if i get it to work ill order another set for the other mobile carriers that we make calls to |
| 16:10.00 | *** part/#asterisk redeemed (~minded@65.211.26.66) |
| 16:13.07 | _omer | //Queue/PersistentMembers/english-support : SIP/1100@1100;0;0| |
| 16:13.17 | _omer | which one is key and which one is family? |
| 16:13.29 | _omer | how to delete this entry from database? |
| 16:14.07 | phantam | hmmm |
| 16:14.12 | phantam | whats a cheap fxo besides the handy tone |
| 16:14.43 | jaike | anyone using sipura spa 841? is it a good phone? |
| 16:14.49 | *** join/#asterisk Nukemizer (~Nuke@160.7.249.15) |
| 16:19.16 | *** join/#asterisk claude2005 (~claude@kcassidy.plus.com) |
| 16:19.20 | robin_sz | ok, so .. what do I need to put in etc/zaptel.conf to make my FXO card work? |
| 16:19.29 | robin_sz | it doesnt seem very clear to me .. |
| 16:19.35 | *** join/#asterisk dr123 (~temp@12-202-51-38.client.insightBB.com) |
| 16:19.59 | *** join/#asterisk _omer (dfsdf@202.147.167.213) |
| 16:21.06 | phantam | is microsoft LCS compatible with asterisk? |
| 16:21.24 | JerJer | robin_sz: you haven't studied the documenation |
| 16:21.40 | JerJer | find zaptel.conf.sample |
| 16:21.59 | *** join/#asterisk darkskiez (~mbryars@194.247.78.146) |
| 16:22.03 | darkskiez | when an app says it returns 0 or -1, what does that mean to a dialplan? |
| 16:22.31 | cpatry | darkskiez: -1 stops, 0 continue. |
| 16:22.57 | Nuxi | I think it means nothing. You can't read them from the dial plan, sooo.... ???? |
| 16:23.24 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 16:27.27 | Katty | mmm, lunch. |
| 16:27.46 | map71 | robin_sz: Something like fxssks=1 (for one channel) |
| 16:28.22 | map71 | robin:sz: Typo: 'fxsks=1' for one channel |
| 16:28.22 | Katty | i would recommend fxsls unless you've been told otherwise |
| 16:28.42 | Katty | loopstart is more widely used that koolstart |
| 16:28.53 | Katty | s/that/than/ |
| 16:29.04 | robin_sz | JerJer: looking |
| 16:29.11 | robin_sz | map71: trying |
| 16:29.27 | Nukemizer | Can mp3's created in Winders be used in asterisk ? Do I need to run a converter like sox ? or is my problem more simple like permissions ? I do have mpg123 on the box but get nothing but immediate hangup when testing |
| 16:29.34 | Katty | hmm, i don't think he should be looking for zaptel.conf.sample |
| 16:29.41 | Katty | making changes to the sample is not going to affect asterisk |
| 16:29.54 | robin_sz | Nukemizer: why not use asterisk to record them? |
| 16:30.01 | Katty | but i'll stay out of this one |
| 16:30.06 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 16:30.37 | robin_sz | Katty: even I woudnt edit the sample ... |
| 16:30.48 | Nuxi | darkskiez: IF might be able to help with return codes. |
| 16:30.57 | cochi | sry guys. reconnecting once more. dsl speed dropped to 50 kbit :| |
| 16:31.14 | Nukemizer | can I rip from my CD whilist using Asterisk ? I just had the files from my media server and thought I could use them, that was what I had hopped |
| 16:32.33 | robin_sz | JerJer: OK, the sample is the same as the one in /etc ... and it still means nothing to me. |
| 16:33.42 | Seyr | Anyone use iax.cc ?? |
| 16:34.57 | blop | whats the diff btw "s" and _X. ? |
| 16:35.23 | map71 | robin_sz: Perhaps you should visit www.voip-info.org and search for the documention of 'zapata.conf'. Did help me |
| 16:35.31 | blop | i mean, why somepeople are using _X. in order to receive sms ? |
| 16:35.39 | blop | i arrives as s here |
| 16:36.00 | gordonjcp | Nuxi: that's crazy |
| 16:36.03 | robin_sz | map71: zapata? oh, ok, not heard of that |
| 16:36.05 | gordonjcp | Nuxi: you could use base 46 |
| 16:36.12 | gordonjcp | bah, 36 |
| 16:36.46 | Nuxi | Nope, the 0-9 are used for SIP signaling. (base 10 encoded) |
| 16:37.04 | map71 | robin_sz: I still am sleeping. Look for zaptel.conf. This is the configuration needed for the drivers |
| 16:37.24 | *** join/#asterisk diablopico (~russ@ip68-8-227-7.sd.sd.cox.net) |
| 16:37.27 | robin_sz | right, thats what I have here ... |
| 16:37.39 | map71 | robin_sz: There is another one called zapata.conf that must be modified to (is the configuration for asterisk to talk with the driver) |
| 16:37.52 | robin_sz | 'k |
| 16:37.55 | JerJer | robin_sz: then its time you pay someone (not me) to hold your hand |
| 16:38.16 | diablopico | Does anyone know af a Media gateway software for windows that will work with h323 to dialogic e1 cards ? |
| 16:38.38 | robin_sz | JerJer: thanks *so* much ... |
| 16:38.45 | JerJer | diablopico: how does that even remotely relate to asterisk? |
| 16:39.08 | JerJer | robin_sz: no problem |
| 16:39.14 | Nuxi | Wonderful thing about flashlight is there shouldn't be any jitter. |
| 16:39.28 | gordonjcp | Nuxi: ahaa |
| 16:39.54 | diablopico | it doesnt , and asterisk may be the solution that i go with, but i need something to compare it to for analysis |
| 16:40.20 | phigan | how can I check my * voicemail remotely? |
| 16:40.26 | Nuxi | diablopico, asterisk runs on windows with cygwin. I even got some agi to work. |
| 16:40.32 | gordonjcp | diablopico: does windows have any support for voip stuff? |
| 16:40.51 | Nuxi | The flashlight setup is faster tho. |
| 16:41.03 | diablopico | thanks |
| 16:41.11 | phigan | flashlight setup? |
| 16:41.13 | gordonjcp | Nuxi: yeah, but I've had some good results with ppp over ssh over teletype |
| 16:41.24 | robin_sz | gordonjcp: sort of .. if you include softphones, and the Java RTP stack .. |
| 16:41.43 | Nuxi | phigan: base26 encoding of rtp to transport over morse code flashlight system. |
| 16:41.46 | diablopico | and i have not been successful in finding windows support for VoIP at an enterprise level, but there is some for residential gateway stuff |
| 16:41.53 | gordonjcp | I suppose it must do, then |
| 16:41.54 | phigan | Nuxi: ahhhh :) |
| 16:41.58 | jontow | yes, there is definitely enterprise level VoIP for windows |
| 16:42.12 | jontow | its ugly as shit, and quirky too.. but it works and i know it to be deployed in many a callcenter |
| 16:42.23 | jontow | (google: CosmoCall, CosmoCom Universe) |
| 16:42.24 | Nuxi | The enterprise stuff for windows looks like the original series. |
| 16:42.34 | jontow | and yes, it transports with H.323 |
| 16:42.37 | phigan | I am going out of town tomorrow, and I would really like to set up my * so that I can receive my voicemails |
| 16:42.43 | jontow | it is INSANELY expensive |
| 16:42.53 | *** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 16:43.01 | phigan | Unfortunately logging into it is not going to work because my * is behind nat |
| 16:43.14 | gordonjcp | jontow: wow |
| 16:43.15 | Nuxi | It's enterspensive |
| 16:43.21 | phigan | and my nat router won't forward the 10000-20000 udp range with autofw for some reason |
| 16:43.26 | jontow | phigan; do you have an incoming line into your * box? |
| 16:43.36 | phigan | jontow: Incoming voip lines, yeah |
| 16:43.46 | gordonjcp | jontow: all the power and flexibility of a wizard, all the visual appeal of the console... |
| 16:43.46 | jontow | no PSTN connection at all? |
| 16:43.51 | phigan | jontow: No |
| 16:43.56 | jontow | ouch |
| 16:43.58 | phigan | why? |
| 16:44.12 | phigan | I can call into it ok |
| 16:44.24 | jontow | i was gonna say; if you have a direct line in; and your extension hangs off it.. just program another extension or call-forward-all from your phone to the voicemailmain extension :) |
| 16:44.52 | phigan | Wouldn't it be the same for non-pstn? |
| 16:44.53 | _omer | I want to remove Asterisk from my system?? |
| 16:44.56 | gordonjcp | register a freebie dial-in SIP line and use that to forward to your voicemailmain |
| 16:45.08 | gordonjcp | _omer: os? |
| 16:45.13 | _omer | RH9 |
| 16:45.14 | jontow | should be; as long as you can call inward to a phone :) |
| 16:45.19 | phigan | ok so I would need a seperate number to connect to the voicemail? |
| 16:45.28 | jontow | not if you forward your phone's extension to it |
| 16:45.29 | _omer | Red Hat 9 |
| 16:45.32 | gordonjcp | _omer: how do you normally remove packages in RH9? |
| 16:45.35 | phantam | guys |
| 16:45.38 | jontow | this is all dialplan magic :) |
| 16:45.46 | phantam | does the handytone 485 really do the sip->fxo translation |
| 16:45.47 | _omer | I have never worked in RH9 :) |
| 16:45.47 | phigan | jontow: when my phone rings, it waits for 4 rings then sends to voicemail |
| 16:45.50 | jontow | can you call and leave yourself a voicemail? |
| 16:45.54 | phigan | yeah |
| 16:46.02 | jontow | ok |
| 16:46.04 | phantam | or is it actually to turn a regular phone into ip phone |
| 16:46.46 | jontow | when you get to the "please leave a message after the tone" prompt.. hit * and it should ask fro a password if your pbx is setup right :) |
| 16:48.06 | *** join/#asterisk Moc_ (~mochouina@64.235.210.66) |
| 16:48.10 | phantam | ? |
| 16:48.46 | Nukemizer | I am stuck on my moh not playing mp3's. I CAN play my mp3 files on my home asterisk just fine. My office asterisk will not play my mp3's. I get no error message so I do not know where to look for trouble shooting. Can anyone suggest a path for me to trouble shoot ? |
| 16:49.25 | jontow | nukemizer; what platform you running * on? mpg123 installed? or is it mpg321? what version? are you using ztdummy or a real zaptel device? |
| 16:50.16 | Essobi | Is there any standard operator intercept messages available for *? |
| 16:50.30 | jontow | checkout asterisk-sounds from cvs :) |
| 16:50.35 | jontow | there are a bunch of random ones |
| 16:51.27 | robin_sz | JerJer: so, do you happen to know whether a single-span FXO card counts as an E1 or a T1, the only two options. |
| 16:51.34 | *** part/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 16:51.46 | jontow | single span FXO? |
| 16:51.52 | jontow | (uh) |
| 16:51.56 | phigan | jontow: What's weird is, I can't get it to ask me for a password when I dial the voicemail extension |
| 16:51.57 | robin_sz | Digium X100P |
| 16:52.00 | Nukemizer | jontow, Office = Digium single T1 card - home Zaptel T400 2fx0 and 2 fxs modules |
| 16:52.01 | phigan | jontow: Is it supposed to? |
| 16:52.05 | Nuxi | I want to play with distorting wav files. Can anyone recommend me a good www reference or a book? |
| 16:52.11 | jontow | yeah, that ain't a T1 or E1 :) |
| 16:52.15 | jontow | its a single line, hehe |
| 16:52.45 | bewest | I wonder if anyone's ever connected csound to asterisk |
| 16:52.45 | phantam | so the handy tone will do sip->fxo |
| 16:52.45 | Nukemizer | jontow, office Version 0.59r of mpg123 and at home system |
| 16:52.46 | robin_sz | jontow: well, thats where my confusion arises .. zaptel.conf only deals with E1 and T1 in the documentation |
| 16:52.57 | phantam | cause i dont have a pci slot for a fxo card |
| 16:53.14 | *** join/#asterisk RoyKa (~roy@179.80-203-29.nextgentel.com) |
| 16:53.30 | Nukemizer | ** Note , I can play the included moh files. they work just fine |
| 16:53.40 | robin_sz | jontow: # The coding is one of "ami" or "b8zs" for T1 or "ami" or "hdb3" for E1 ... |
| 16:53.49 | Essobi | jontow I don't mean random ones.. I mean the standards.. "I'm sorry, the number you've called is no longer in service, if you feel this is an error, please hang up and try your call again", and etc, etc, etc. |
| 16:53.55 | robin_sz | so what should I put for an X100P which is netiher? |
| 16:54.02 | jontow | if you're hooking up an FXO .. that isn't applicable, iirc |
| 16:54.18 | robin_sz | right. |
| 16:54.23 | bewest | Essobi, yeah |
| 16:54.32 | bewest | Essobi, there are a few in the standard distribution |
| 16:54.42 | bewest | Essobi, if you check out asterisk-sounds there are many many more |
| 16:54.48 | Essobi | Oh? Pfft, guess I'll have to go listen to them all. :| |
| 16:54.54 | jontow | check the wiki |
| 16:54.56 | bewest | Essobi, check the wiki for descriptions |
| 16:55.01 | Essobi | WIKI WIKI WOW WOW |
| 16:55.03 | jontow | it gives a basic description of what each actually says :) |
| 16:55.03 | *** join/#asterisk AlexCeli (~Alex@200.37.85.99) |
| 16:55.25 | bewest | Essobi, there are random funny ones too |
| 16:55.35 | phantam | is there a better place to ask questions like that |
| 16:55.35 | phantam | ? |
| 16:56.25 | jontow | back in the 60s my great aunt was the operator and the person who recorded all those prompts for the local telco :) |
| 16:56.37 | Essobi | neeet |
| 16:56.41 | Essobi | heh |
| 16:56.45 | Essobi | did she have a blue box? |
| 16:56.50 | jontow | not a clue ;) |
| 16:56.54 | jontow | i met her twice |
| 16:57.00 | jontow | and i haven't since seen her (that was years ago) |
| 16:57.21 | jontow | and since then the telco has consolidated a bunch of crap into a remote office, including a lot of their switching equipment.. so its all generic now i think |
| 16:57.28 | robin_sz | oh and if anyone cares, I am still not liking my Grandstream GXP2000 |
| 16:57.33 | jontow | :( |
| 16:57.51 | jontow | rating on a scale of 1-10? |
| 16:58.03 | robin_sz | 3 |
| 16:58.06 | phantam | why ? |
| 16:58.32 | robin_sz | 1) bad hardware desing (wont go on hook) |
| 16:58.51 | *** join/#asterisk ender (~me@ender.fedora) |
| 16:58.52 | robin_sz | 2) poor physical feel (light and tacky) |
| 16:58.53 | jontow | :( |
| 16:59.02 | jontow | sounds much like the BT100s |
| 16:59.06 | ender | has anybody dealt with Sipura SPA-841 2-Line IP Phone and has opinions on it? |
| 16:59.20 | robin_sz | 3) it doesnt wat to updtae tis firmaware |
| 16:59.22 | jontow | get yourself an 80s or early 90s bell phone |
| 16:59.32 | jontow | crack open the handset and steal the weighted bar from it |
| 16:59.36 | robin_sz | heh |
| 16:59.38 | jontow | then retrofit into a grandstream phone :) |
| 16:59.38 | *** join/#asterisk bofh42 (~bofh42@p548222F7.dip0.t-ipconnect.de) |
| 16:59.43 | jontow | that'd make them 20x better |
| 16:59.45 | robin_sz | I figure on putting it back on ebay |
| 16:59.57 | robin_sz | and getting a Cisco |
| 17:00.02 | jontow | indeed |
| 17:00.11 | jontow | i think if i can convince my employer to give it to me |
| 17:00.16 | robin_sz | the on-hook problem they MUST have known about |
| 17:00.27 | jontow | im gonna take this budgetone and try to fit decent audio components into it someday |
| 17:00.30 | jontow | i think that'd really help |
| 17:00.41 | jontow | would be nice to get reasonable volume controls too.. :) |
| 17:01.08 | robin_sz | the GXP has a an on hook switch that meets the handset, except the handset ear bit is recessed .. so it never could have operated it .. |
| 17:01.19 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 17:01.38 | jontow | heh.. that sucks.. |
| 17:01.47 | robin_sz | perhaps I can glue a lump of plastic to the switch ... |
| 17:01.53 | *** join/#asterisk cmk (~cmk_@p54A3F981.dip.t-dialin.net) |
| 17:02.33 | jontow | thats annoying.. :) |
| 17:02.55 | robin_sz | given that it costs say, twice what an POTS phone of good quality costs |
| 17:03.16 | robin_sz | and there is like an additional $2 of components ... |
| 17:03.23 | robin_sz | its not really acceptable |
| 17:04.42 | jontow | yeah, makes lots of sense :( |
| 17:05.13 | ender | has anybody dealt with Sipura SPA-841 2-Line IP Phone and has opinions on it? |
| 17:05.14 | robin_sz | ah well ... eBay will save me :) |
| 17:06.11 | robin_sz | oh, and will someone PLEASE fix the broken HTML on the wiki |
| 17:06.31 | [TK]D-Fender | ender : http://voip-info.org/wiki-SPA-841 http://www.geekgazette.com/index.php?option=com_content&task=view&id=24&Itemid=26 |
| 17:06.42 | JerJer | robin_sz: its a wiki - you can't fix it ? |
| 17:07.00 | robin_sz | JerJer: content yes, broken layout, no |
| 17:07.55 | robin_sz | JerJer: main content block ends up off to the tright hand side in a colum about 150 pixels wide, way oof the right of where it should be. presumably a wrongle closed <tr> or something |
| 17:08.29 | Hmmhesays | get out an html debugger |
| 17:08.35 | Hmmhesays | and have it tell you were there are possible problems |
| 17:08.47 | ender | [TK]D-Fender: thanks |
| 17:08.53 | robin_sz | Hmmhesays: I just did .. I got bored at error 128 |
| 17:09.03 | Hmmhesays | or get out the ol' wysiwyg editor and click on the area that is giving you issue |
| 17:09.09 | Hmmhesays | to narrow it down |
| 17:09.55 | robin_sz | I'll gladly sort it if someone sends me the Wiki source ... |
| 17:10.44 | Hmmhesays | what main content block are you talking about? |
| 17:11.18 | robin_sz | the main content .. the MAIN CONTENT .. like .. the whataver the wiki page is about content content. |
| 17:11.33 | Hmmhesays | haha i'm lost, link me again |
| 17:11.57 | robin_sz | http://www.voip-info.org/tiki-index.php?page=voip-info.org#comments |
| 17:12.13 | robin_sz | for me .. I get a big blue exapanse smack in the middle of the page |
| 17:12.29 | robin_sz | the real info is in a white comun off the right of the screen |
| 17:12.45 | Hmmhesays | hrm, looks fine here |
| 17:13.11 | robin_sz | well, validator.w3.org says it has man, many html errors |
| 17:13.30 | Hmmhesays | are you trying to use lynx on the wiki again? |
| 17:13.34 | Hmmhesays | xD |
| 17:13.38 | robin_sz | no, netscape |
| 17:13.45 | *** part/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 17:13.46 | Hmmhesays | it renders right in firefox and IE |
| 17:14.04 | robin_sz | neither of which i can run on my machine .. |
| 17:14.05 | jontow | netscape has rendered things correctly 2 out of 10 times that i'vfe used it |
| 17:14.12 | jontow | what is your machine? |
| 17:14.21 | robin_sz | Redhat 7.3 laptop |
| 17:14.29 | jontow | why won't mozilla run on it? |
| 17:14.42 | Hmmhesays | have you been drinking? |
| 17:15.07 | jontow | i've been using mozilla 1.x (now 1.8b1) on my netbsd 2.0.2 laptop for weeks and its doing great |
| 17:15.11 | robin_sz | well, it renders code that aitn brok |
| 17:15.16 | robin_sz | http://validator.w3.org/check?uri=http%3A%2F%2Fwww.voip-info.org%2Ftiki-index.php%3Fpage%3Dvoip-info.org%23comments |
| 17:15.25 | Hmmhesays | no one is going to fix it you realize |
| 17:15.30 | robin_sz | I would .. |
| 17:15.35 | robin_sz | if I had the code .. |
| 17:15.40 | jontow | yeah you gotta remember though.. maybe half of the people on the web write decent HTML, if that :( |
| 17:15.44 | jontow | im not one of them. |
| 17:15.46 | Hmmhesays | you'd be better off rpm -i mozilla |
| 17:16.09 | jontow | no, not the rpm; it never is up to date ;) grab the installer or running-copy directory-tree-tarball from mozilla.org |
| 17:16.22 | Hmmhesays | haha i have used redhat in ages |
| 17:16.31 | Hmmhesays | just trying to get a point across |
| 17:16.32 | robin_sz | well, yeah |
| 17:16.42 | jontow | last time i installed an RPM on RH9.0 of mozilla it was still 1.6.x |
| 17:16.51 | robin_sz | its old .. I really REALLY should go debian |
| 17:16.53 | jontow | granted that was when 1.7.x was "new" or whatever :) |
| 17:17.08 | Hmmhesays | debian sarge is a good os |
| 17:17.26 | Hmmhesays | woody is alright too if you dont' need to be up to date |
| 17:17.28 | *** join/#asterisk apropos (tom@wsip-68-15-205-134.ok.ok.cox.net) |
| 17:17.32 | robin_sz | hey, Sarge is "stable" this weekend |
| 17:17.37 | robin_sz | perhaps, this is upgrade time |
| 17:17.44 | Hmmhesays | did sid move into testing? |
| 17:17.51 | robin_sz | not yet |
| 17:17.58 | robin_sz | it doesnt all go over in one lump |
| 17:18.08 | Hmmhesays | which is why i asked |
| 17:18.11 | robin_sz | :) |
| 17:18.12 | Hmmhesays | ;) |
| 17:18.15 | *** join/#asterisk RoyKa (~roy@179.80-203-29.nextgentel.com) |
| 17:18.16 | *** join/#asterisk joe (~jsauer@ip66-107-33-195.z33-107-66.customer.algx.net) |
| 17:18.26 | robin_sz | so who is the Wiki maintainer then? |
| 17:19.04 | Hmmhesays | you realize that this will be a losing battle for you |
| 17:19.15 | Katty | hi lads. |
| 17:19.39 | Hmmhesays | speaking of hello's... I gotta run |
| 17:19.43 | robin_sz | shrug ... |
| 17:20.12 | Hmmhesays | but i'll be lurking in the background again soon enough |
| 17:20.38 | Katty | Hmmhesays: :< |
| 17:22.51 | RoyKa | wtf? is there a FEMALE person in #asterisk??? |
| 17:23.51 | blitzrage | RoyKa: I've seen two recently actually :) |
| 17:23.59 | phigan | heh |
| 17:24.03 | robin_sz | who cares. male, female, martian .. does it matter in any way? |
| 17:24.04 | blitzrage | ~seen sarahemm |
| 17:24.06 | jbot | sarahemm <~sarahemm_@Toronto-HSE-ppp3704598.sympatico.ca> was last seen on IRC in channel #asterisk, 6d 14h 40s ago, saying: 'twisted: i'll try to post a patch when i get to sjc (i have inet access there, and will be working on * inflight)'. |
| 17:24.07 | RoyKa | blitzrage: apart from l-fy??? |
| 17:24.22 | blitzrage | that'd make 3 |
| 17:24.41 | niZon | ~seen boris |
| 17:24.42 | jbot | boris <boris@S01060040ca1e5b54.wp.shawcable.net> was last seen on IRC in channel #asterisk, 1d 10h 56m 14s ago, saying: 'Who's the stud? :-p'. |
| 17:24.50 | robin_sz | oh, and then theres the girl that does the voiceovers ... |
| 17:24.54 | RoyKa | ~seen l-fy |
| 17:24.56 | jbot | l-fy <~diana@l-fy.developer.yate> was last seen on IRC in channel #asterisk, 18d 2h 37m 41s ago, saying: 'hello'. |
| 17:26.13 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 17:26.13 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:30.02 | jontow | im likin' blackbox |
| 17:30.16 | jontow | though its a severely ugly process to configure bbkeys unless bbconf matches perfectly.. *cough* |
| 17:30.16 | jontow | :) |
| 17:30.18 | niZon | bah |
| 17:30.19 | jontow | stupid API changes |
| 17:30.29 | niZon | link2voip said 24-48 hours for a toll free |
| 17:30.33 | niZon | it's been almost a week |
| 17:30.36 | Nuxi | Hmmm. maybe it does matter (male/female). If you are female everyone wants to answer your questions, but they all think you are stupid. |
| 17:30.55 | RoyK | fsck |
| 17:32.35 | docelm0 | ~seen doce |
| 17:32.36 | jbot | doce <~docelmo@116-39.202-68.tampabay.res.rr.com> was last seen on IRC in channel #asterisk, 6d 17h 5m 52s ago, saying: 'Your kidding?'. |
| 17:32.50 | *** join/#asterisk _omer (dfsdf@202.147.167.213) |
| 17:32.51 | RoyK | bkw_: ping |
| 17:33.04 | Katty | <bkw_> pong |
| 17:33.15 | _omer | want to change #ztdummy to ztdummy in Makefile....may I know the line number plz ? |
| 17:33.26 | robin_sz | Nuxi: but this is the net, like how do you know who is male or female? |
| 17:34.23 | robin_sz | _omer: search for it .. if using vi, just do /ztdummy<return> |
| 17:35.42 | RoyK | _omer: search for it? |
| 17:36.07 | robin_sz | IIRC its in the first page or so |
| 17:36.15 | _omer | yes...found at line 100 |
| 17:36.22 | _omer | ztd-loc # ztdummy |
| 17:36.24 | robin_sz | well, page 4 then :) |
| 17:36.25 | _omer | correct??? |
| 17:36.39 | robin_sz | sounds about right ... |
| 17:36.40 | _omer | is this the correct line? |
| 17:36.43 | _omer | good... |
| 17:36.44 | robin_sz | delete the # |
| 17:37.28 | phigan | I can't for the life of me get * to let me enter a password for voicemail |
| 17:37.52 | _omer | thanks guys |
| 17:38.05 | phigan | so when I call myself I can't hit * to get my voicemail |
| 17:38.47 | phigan | I'm assuming that it doesn't work because the password doesn't work |
| 17:39.19 | phigan | My voicemail works just fine |
| 17:39.24 | phigan | I just can't check it remotely :/ |
| 17:39.29 | Katty | _omer: don't forget the girls. |
| 17:39.34 | Nuxi | robin_sz, hmmmm. So it's more a matter of appearing female. |
| 17:39.34 | phigan | which I could really really use, leaving town tomorrow |
| 17:40.44 | robin_sz | Nuxi: couldbe, after all I am an 18yr old, blonde female, high-school student, Im just pretending to be a ageing male. |
| 17:40.48 | _omer | Katty: I am a gay... |
| 17:40.51 | _omer | :D |
| 17:40.55 | Katty | k'then |
| 17:41.26 | phigan | Thingies are fun to touch. |
| 17:41.44 | Katty | and by thingies you mean fxos |
| 17:41.50 | phigan | sure |
| 17:42.05 | phigan | if by fxos you mean thingies |
| 17:42.12 | Nuxi | Sure..., and I'm a 31 year old programmer that works from home. |
| 17:42.15 | Katty | i see |
| 17:42.32 | Katty | Nuxi: i'm a windows admin :< |
| 17:42.43 | Katty | let's trade jobs! |
| 17:42.55 | Katty | Essobi: thx. |
| 17:42.58 | phigan | so like, how do I get Asterisk to have a password to my voicemail? |
| 17:42.58 | Nuxi | I'm happy to say I don't know anything about windows except that * will run on it. |
| 17:43.07 | Essobi | Hehehe. |
| 17:43.15 | robin_sz | Nuxi: no you aint .. |
| 17:43.26 | robin_sz | Nuxi: theres no such thing as a windows admin |
| 17:43.35 | Katty | oh? |
| 17:43.39 | Katty | my job is suddenly obsolete? |
| 17:43.42 | Katty | and all my clients too? |
| 17:43.45 | Katty | YEAH! |
| 17:43.52 | phigan | not obsolete |
| 17:43.54 | phigan | doesn't exist |
| 17:43.58 | phigan | :) |
| 17:43.58 | Essobi | PFFT. I got kung foo skills for days. Windows, Unix, it doesn't matter, just thrown down on it. :) |
| 17:44.02 | Katty | hot dang, i'm excited |
| 17:44.10 | Essobi | Katty Me too. :) |
| 17:44.18 | robin_sz | Katty: yep, as soon as they produce a automated way of hitting the reset button every hour, you get your pink slip |
| 17:44.24 | Katty | Essobi: gosh. |
| 17:44.34 | Katty | robin_sz: well, possibly. |
| 17:44.34 | Essobi | robin_sz I think that's called a watchdog. |
| 17:44.42 | Katty | robin_sz: they'd have to keep me around for the asterisk box |
| 17:44.42 | phigan | except, I can't use my computer to get asterisk to use a password for my voicemail for some reason :) |
| 17:44.45 | Katty | robin_sz: and networking |
| 17:44.50 | Katty | robin_sz: and probably the website |
| 17:44.52 | gordonjcp | Essobi: see, I can rustle up spicy stir-fried chicken with cashew nuts with such blinding speed and skill |
| 17:44.55 | gordonjcp | robin_sz: 'cos... |
| 17:44.57 | robin_sz | networking on windows? |
| 17:44.59 | gordonjcp | Essobi: 'coss |
| 17:45.01 | Essobi | Katty Ugh.. one of those jobs.. Ewww. |
| 17:45.06 | Katty | robin_sz: amongst other things |
| 17:45.09 | gordonjcp | Essobi: I *know* Kung Po |
| 17:45.12 | Nuxi | I saw an ad in the paper looking for someone to keep their office computers from crashing. |
| 17:45.13 | Essobi | MEhehe |
| 17:45.15 | Katty | Essobi: yeah i seem to be the catch all closet |
| 17:45.15 | robin_sz | you can coonect windows to a network? |
| 17:45.25 | Katty | robin_sz: yes, with duct tape |
| 17:45.26 | Nuxi | So there are windows admin jobs. |
| 17:45.29 | Essobi | Katty I had one of those jobs once. Sucked. |
| 17:45.30 | robin_sz | wow :) |
| 17:45.41 | *** join/#asterisk jief- (~jief@modemcable196.182-80-70.mc.videotron.ca) |
| 17:45.42 | Katty | Essobi: i kinda like it. |
| 17:45.43 | robin_sz | heh .. heres a good one, |
| 17:45.48 | Katty | Essobi: i get to work with a lot of things... |
| 17:45.49 | Essobi | Katty It'll get old. |
| 17:45.56 | Katty | Essobi: i even got a kitten out of the warehouse once :> |
| 17:46.04 | Essobi | I did it for... |
| 17:46.05 | Essobi | ... |
| 17:46.08 | Essobi | Damnit, I'm old. |
| 17:46.14 | Katty | you so are |
| 17:46.15 | robin_sz | "I hate creases in duct-tape, it just smacks of low professional standards" ;) |
| 17:46.18 | jief- | hi, i was wondering if anyone was using Aastra 480i generic sip phones with asterisk? Mine isn't working properly right now |
| 17:46.35 | Essobi | Katty Pffft, you're practically a twinkle in your dad's eye still. :) |
| 17:46.37 | Katty | nope, i'm using polycom500s |
| 17:46.46 | Essobi | Poly's ain't bad. |
| 17:46.53 | Katty | Essobi: my dad can die for all i care |
| 17:46.58 | Essobi | Wrote Ciscos DSPs after all for a few years.. |
| 17:47.15 | Essobi | Katty Gee.. you don't have "male" issues, do you? |
| 17:47.31 | Katty | well let's see |
| 17:47.38 | Katty | i had androphobia until i moved away from home |
| 17:47.42 | Katty | and i prefer females over males |
| 17:47.47 | jief- | hi, i was wondering if anyone was using Aastra 480i generic sip phones with asterisk? Mines are able to place calls. But can't receive calls, after 2 seconds, calls are being disconnected |
| 17:47.48 | Katty | yeah, i think i might have a 'few' |
| 17:47.57 | Essobi | Katty That was sarcasm. |
| 17:48.09 | Sn0p | can any one recommend an cheap and good sip phone? and a url to where i could buy it? |
| 17:48.19 | Katty | Sn0p: software phone? |
| 17:48.21 | Katty | Sn0p: those are free |
| 17:48.24 | Essobi | wtf do you get androphobia from? |
| 17:48.35 | Essobi | nm |
| 17:48.37 | Katty | Essobi: if you don't know what the word means, google it. |
| 17:48.38 | Essobi | I don't want to know |
| 17:48.42 | Sn0p | Katty:hardware |
| 17:48.47 | Katty | Essobi: if you don't get how...then you have a very crappy imagination |
| 17:48.51 | Katty | Sn0p: what do youw ant it to do? |
| 17:48.58 | Katty | Sn0p: multiple lines? |
| 17:48.58 | Sn0p | call?:p |
| 17:48.58 | Essobi | what/where/why are all interchangable in wtf |
| 17:49.00 | cluecon | ~cluecon |
| 17:49.01 | jbot | it has been said that cluecon is http://www.cluecon.com |
| 17:49.21 | jief- | Sn0p: if you are not looking for the prettiest phones, the Gnet phones are cheap and work quite well |
| 17:49.21 | loud | ~androphobia |
| 17:49.24 | Sn0p | Katty: no, just one line |
| 17:49.34 | robin_sz | ooh, look, this X100P came with a an asterisk Live CD .. hey, I should try it. |
| 17:49.42 | terrapen | anyone seen a problem with -HEAD where a SIP phone calls another SIP phone but nobody answers so the caller hangs up and can no longer make any outbound calls at this point? |
| 17:49.44 | Sn0p | i was watching Budgetone phones are they any good? |
| 17:49.52 | loud | till they explode |
| 17:50.00 | loud | or just die. |
| 17:50.02 | robin_sz | cheap |
| 17:50.04 | Katty | Sn0p: digital or analog? |
| 17:50.05 | Essobi | Katty Call me niave, but I have a belief that all people are inherently good. *SHRUG* |
| 17:50.09 | terrapen | also, does anybody have the latest IP500 firmware? |
| 17:50.09 | Sn0p | digital |
| 17:50.15 | terrapen | and SIP sw |
| 17:50.22 | Sn0p | jief-: do you have a url ? |
| 17:50.24 | Essobi | Sn0p Does "BUDGETtone" sound like a quality phone to you? |
| 17:50.29 | Sn0p | no |
| 17:50.36 | robin_sz | Essobi: no, most people are inherently selfish, most are good, a few are plain evil |
| 17:50.41 | Essobi | Follow your gut instinct then. |
| 17:50.59 | Essobi | robin_sz Jeez, remind me not to visit your hometown. |
| 17:51.01 | *** join/#asterisk clive- (~pirch@rrba-146-101-112.telkomadsl.co.za) |
| 17:51.08 | Katty | Sn0p: http://www.voip-info.org/wiki-VoIP+Phones#id997987 |
| 17:51.08 | bewest | the budgetones have no side tone |
| 17:51.12 | Sn0p | my instinct told me to get in here and ask :) |
| 17:51.16 | gordonjcp | Essobi: ayup, with you on that |
| 17:51.17 | bewest | so it's kind of weird |
| 17:51.25 | Katty | Sn0p: take your pick, prices listed |
| 17:51.28 | jief- | Sn0p: hold on |
| 17:51.31 | *** join/#asterisk smash- (~smash@198.107.16.189) |
| 17:51.33 | smash- | hello |
| 17:51.33 | Essobi | bewest wtf is a side tone? |
| 17:51.46 | bewest | you can hear yourself in the ear piece |
| 17:51.47 | Katty | Essobi: it's the tone you make when i smack your side with a chair |
| 17:51.49 | gordonjcp | Essobi: it's the amount of your own voice you hear |
| 17:51.58 | Essobi | Katty rofl |
| 17:51.59 | jief- | Sn0p: http://www.gentek.com/products.cgi?cat=100 |
| 17:52.02 | Sn0p | kajtzu: i have seen the list but, what to pick ? i have no idea whats good |
| 17:52.06 | Sn0p | jief-: thank you |
| 17:52.10 | Sn0p | err |
| 17:52.16 | Essobi | jjez |
| 17:52.18 | robin_sz | Essobi: just remember, you probably meet the better half of the planet onthe net, and you probably only interact with members of the better 10% ... |
| 17:52.19 | Sn0p | katty : i have seen the list but, what to pick ? i have no idea whats good |
| 17:52.35 | Essobi | no side tone? That's a shitty way to deal with full duplex speakerphones. |
| 17:52.48 | Essobi | robin_sz ;) |
| 17:52.50 | PyroSteve | <PROTECTED> |
| 17:52.55 | Katty | Sn0p: that's why you have to compare your options |
| 17:53.02 | PyroSteve | my budgetone has plenty of "side tone" |
| 17:53.02 | Katty | Sn0p: you don't just walk up to a car deal and say hi! i want a car. |
| 17:53.18 | Essobi | PyroSteve AH HA! |
| 17:53.22 | Essobi | I'm not wrong! |
| 17:53.24 | Katty | http://www.voip-info.org/wiki-Budgetone <- this is a budgetone for about 75-80 bucks |
| 17:53.40 | Essobi | pft |
| 17:53.53 | PyroSteve | then why can I hear my self through the speaker when yall say I cant |
| 17:54.05 | phigan | wish i had a ht486 for my trip |
| 17:54.06 | Katty | PyroSteve: cause you've insaned. |
| 17:54.08 | bewest | PyroSteve, what model do you have? |
| 17:54.15 | Essobi | PyroSteve I didn't say that. :) |
| 17:54.16 | Sn0p | Katty: if my friend would recommand it would take a closer look, better then go to all the car delears and ask them huh?;) |
| 17:54.16 | bewest | I have the bt-100 |
| 17:54.21 | cpm | yeah, my "new" LG cell phone the replaced my dutiful motorola has no side tone, I HATE it. |
| 17:54.22 | Essobi | SO IM RIGHT! HAH! |
| 17:54.27 | gordonjcp | Katty: I was going to buy a budgetone 102, but I got an Avaya 4602 instead for £40-odd |
| 17:54.45 | Katty | gordonjcp: that price is what's listed on voip-info website. |
| 17:55.01 | jief- | i was wondering if anyone was using Aastra 480i generic sip phones with asterisk? Mines are able to place calls. But can't receive calls, after 2 seconds, calls are being disconnected |
| 17:55.08 | gordonjcp | Katty: yup, so divide by 2-ish for GBP |
| 17:55.10 | cluecon | ~cluecon |
| 17:55.13 | jbot | hmm... cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. |
| 17:55.13 | Katty | i think jief- is an automated recording. |
| 17:55.17 | bkw_ | someone called my name? |
| 17:55.18 | jief- | Katty: you wish |
| 17:55.18 | *** join/#asterisk jgaviria (~c81fcd05@208.195.214.137) |
| 17:55.26 | gordonjcp | the only thing is, if I register the voicemail notifications, it craps itself after about 5 minutes |
| 17:55.29 | Katty | bkw_: your wife. |
| 17:55.31 | gordonjcp | some weird bug somewhere |
| 17:55.32 | bkw_ | haha |
| 17:55.33 | PyroSteve | bewest: same here |
| 17:55.33 | bkw_ | wife |
| 17:55.35 | bkw_ | thats funny |
| 17:55.37 | bkw_ | Royk did |
| 17:55.38 | Katty | isn't it |
| 17:55.39 | PyroSteve | or i think I have the 101 |
| 17:55.40 | cluecon | bkw_: your just hearing things. |
| 17:55.40 | Assid | umm.. |
| 17:55.46 | jief- | Katty: im a frustrated admin wasting time with what looks like a very crappy phone |
| 17:55.50 | bkw_ | no the tab changes color when someone says bkw_ |
| 17:55.51 | Assid | i need some help setting up a sip extension |
| 17:55.53 | Assid | can someone please help me |
| 17:55.56 | bkw_ | someone said my name boi |
| 17:55.57 | Katty | jief-: i hate when that happens :< |
| 17:56.03 | Katty | bkw_: HEY |
| 17:56.07 | Katty | bkw_: why don't you help jief- |
| 17:56.09 | jief- | Katty: even the manufacturer can't help |
| 17:56.15 | Katty | bkw_: he seems desperate |
| 17:56.18 | jgaviria | hi, im connecing an E100P to a PRI that have three blocks of 10 channels configured, sharing the D channel, and i cant do outgoing calls, somebody could helpme? |
| 17:56.30 | blitzrage | bkw_: |
| 17:56.39 | PyroSteve | does AMP keep the config files in a neat manner |
| 17:57.12 | PyroSteve | should one be able to modify the config files in conjuction with using AMP ? |
| 17:57.27 | cluecon | bkw_: will there be wifi at cluecon? |
| 17:57.42 | blitzrage | anyone know the Polycom three finger salute to reboot the phone before I google for it? |
| 17:58.03 | *** join/#asterisk Falle (falstaf@voip-forum.se) |
| 17:58.03 | Essobi | umm |
| 17:58.05 | twisted[work] | blitzrage, + - msg hold |
| 17:58.14 | twisted[work] | it's a four finger salute ;) |
| 17:58.18 | *** join/#asterisk docelm0 (~docelm0@67.106.194.90.ptr.us.xo.net) |
| 17:58.19 | Essobi | nice |
| 17:58.20 | PyroSteve | thats four fingers ! |
| 17:58.20 | Nuxi | ~google reboot polycom |
| 17:58.34 | Essobi | It's the vulcan death grip |
| 17:58.35 | twisted[work] | PyroSteve, yes, thus "it's a four finger salute ;)" |
| 17:58.41 | blitzrage | twisted[work]: thanks! |
| 17:58.42 | docelm0 | Ya.. That was fun and interesting.. sigh.. Damn Cisco's.. |
| 17:58.44 | twisted[work] | blitzrage, np |
| 17:59.21 | blitzrage | heh, worked too :) |
| 17:59.42 | Essobi | I wish I could access the XML engine in my 7960 |
| 17:59.49 | Essobi | That'd be super sweet. |
| 18:01.23 | terrapen | any Mac people here? |
| 18:01.24 | Katty | i wonder if he'd be possible to make a website that writes to a tftp server, ..and the website writing to a text file that would change the settings on a polycom phone |
| 18:01.28 | terrapen | Astmasters folks? |
| 18:02.27 | PyroSteve | an off topic topic but I dont understand Secret Window |
| 18:02.36 | Katty | he killed her |
| 18:02.39 | Katty | and burried her with the corn |
| 18:02.41 | Katty | cause he was insane |
| 18:02.56 | Katty | and the ending of his book didn't turn out the way it was supposed to |
| 18:03.27 | [TK]D-Fender | cluecon : Hey, I just implemented your method of using CVS' ChanIsAvail to circumvent placing a queue call for a phone in use and it work well. |
| 18:03.36 | Nuxi | Katty, I have a tftp server that writes config files on the fly. |
| 18:03.44 | PyroSteve | which what is he did at the end .. was change the ending by acting it out ? |
| 18:03.47 | cluecon | TKD: Glad to hear it. |
| 18:03.58 | [TK]D-Fender | cluecon : actually it still dials the agent, but I bomb out manually and immediately. |
| 18:04.06 | Katty | Nuxi: but i don't want them to be written on the fly. i want a user to login into the website and configure their own phone using dropdown menus, checkboxes, etc. |
| 18:04.09 | [TK]D-Fender | So, so far so good. |
| 18:04.35 | Nuxi | Katty, exactly. All the configs are in the db. The tftp server reads them and makes a config file. |
| 18:04.44 | cluecon | TKD: Come to cluecon and we can blame bkw_ for all of * failings especially the queues. |
| 18:05.20 | Katty | Nuxi: database? |
| 18:05.33 | Nuxi | Having thousands of config files for customers configurations is messy IMO. |
| 18:05.43 | Katty | Nuxi: why are you storing settings in a database if they're going to ............................................oh. |
| 18:05.45 | Hmmhesays | indeed |
| 18:05.53 | Hmmhesays | db storage just makes sense |
| 18:06.05 | Katty | i'm not talking about the asterisk |
| 18:06.12 | Katty | i'm talking about the polycom phone |
| 18:06.23 | Katty | or are we both talking about the polycom phones? |
| 18:06.28 | Hmmhesays | hah I gotta quit jumping into of the middle of conversations |
| 18:06.33 | Nuxi | Sure. you just need to figure out the config files and write a handler for them. |
| 18:06.39 | [TK]D-Fender | cluecon : I don't want to place blame, I want SOLUTIONS. And wish I could but its a little far for me and costly for a project I'm not too certain to be able to implement |
| 18:06.48 | Nuxi | Then the handler reads the settings from a db and makes the file. |
| 18:07.02 | Katty | why not write directly to a text file? |
| 18:07.09 | Katty | there's only 15 phones in this office |
| 18:07.16 | Katty | somethings will be the same over and over again |
| 18:07.36 | Nuxi | For your situation, this might not be ideal. For thousands, it works well. |
| 18:07.40 | cluecon | TKD: Just keep me posted on what solutions you need. I'm making progress on my end and thus far, * is the only solution for us that makes sense. |
| 18:07.49 | Katty | i see. |
| 18:07.51 | Hmmhesays | don't polycom's support centralized provisioning? |
| 18:07.56 | Katty | Hmmhesays: yes |
| 18:08.07 | Hmmhesays | what's yoru issue then? |
| 18:08.07 | Katty | Hmmhesays: but i don't really want that. i'd like to have an office with 5 rooms in it |
| 18:08.26 | Katty | Hmmhesays: everyone gets told which office their into today...then they go, login into the domain, login to the website and configure the phone sitting next to them |
| 18:08.32 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 18:08.38 | terrapen | does anybody have the latest polycom firmware? |
| 18:08.38 | Katty | Hmmhesays: the next day, someone else is in the office. repeat |
| 18:08.42 | [TK]D-Fender | cluecon : lucky you. basically * does everything *I* care about, its other people's concerns that are the challenges to meet. |
| 18:08.46 | Katty | terrapen: i bet polycomd oes |
| 18:08.52 | Hmmhesays | end users configuring phones is bad |
| 18:08.56 | [TK]D-Fender | terrapen : Wouldn't suggest the "latest" |
| 18:09.05 | terrapen | they only let authorized dealers have them now, katty :( |
| 18:09.11 | [TK]D-Fender | terrapen : BR 1.4.2 and SIP 1.5 though |
| 18:09.12 | terrapen | fender: what's wrong with it? |
| 18:09.22 | [TK]D-Fender | Just that you can't go back |
| 18:09.26 | terrapen | ah |
| 18:09.38 | Katty | Hmmhesays: well how else is phone xyz going to be logged in as 'dan' |
| 18:09.46 | terrapen | fender, i upgraded to -HEAD and now if my polycom dials an extension which does not answer, it can no longer dial anything |
| 18:09.49 | anthm | [TK]D-Fender, why do you think his nick is cluecon!, he's telling you to come to the conference and work toward solving your problems =D |
| 18:09.49 | Katty | i don't really want them poking about on the phone |
| 18:09.55 | cluecon | TKD: same here. It does everything I need out of the box. What everyone else wants, I'll have to build. |
| 18:09.55 | terrapen | my cisco does not seem to be affected |
| 18:10.02 | Katty | Hmmhesays: with a form on a website, i can only give them the options i want them to have |
| 18:10.03 | bewest | Hmmhesays, Katty, they aren't manipulating the phone directly necessarily, just a web interface of your design |
| 18:10.21 | Hmmhesays | yeah, hmmm.... a good question |
| 18:10.31 | Hmmhesays | you can do it |
| 18:10.38 | Katty | sure i can |
| 18:10.42 | Katty | i just need a few hours |
| 18:10.50 | Hmmhesays | ok so you really don't have an issue? |
| 18:10.52 | Hmmhesays | <chuckle> |
| 18:10.58 | Katty | pffft. |
| 18:11.02 | Katty | don't make me come over there |
| 18:11.11 | Hmmhesays | :D |
| 18:11.23 | Hmmhesays | mmm pizza |
| 18:11.31 | Katty | cause if i have to go over there, i'm stealing that bike |
| 18:12.21 | Hmmhesays | ruh roh, if by "stealing" you mean hoping on the back and going for a ride |
| 18:13.35 | [TK]D-Fender | cluecon : One of the next things to implement is a Queue_log entry for when agents DON'T answer their calls. |
| 18:14.05 | anthm | how bout a shiny new queue system go with it |
| 18:14.20 | dudes | [TK]D-Fender - that's not a bad idea |
| 18:14.56 | docelm0 | cluecon likes to speak alot in the 3rd person.. |
| 18:15.08 | robin_sz | [TK]D-Fender: and you could even auto-print their dismissal notice on their printer ... |
| 18:15.38 | dudes | robin_sz - Agent_send_message == f'n fired |
| 18:15.53 | cluecon | TKD: the queue system itself needs to be re-fashioned to allow more flexibility. I think that the problem is that * isn't heavily used for call center environments. I also think that you and I have unique environments that have special needs which haven't really come up before. |
| 18:16.38 | dudes | cluecon - asterisk does work well enought. Though, patching is required and patching one head to the next get's rough. |
| 18:16.42 | robin_sz | sigh .. adding an FXO should not be this hard ... or I might be stupid |
| 18:16.44 | anthm | hmm maybe you should go test ICD |
| 18:16.52 | [TK]D-Fender | True, but one which is required for "respecability" for corporate use. * needs it to escape home/smb status. |
| 18:16.52 | dudes | cluecon - enough* |
| 18:17.05 | cluecon | docelm0: I've been told that paranoid schizophrenic usually do have multiple personalities but I prefer to simply refer to them as The Voices since they won't ever shutup. |
| 18:17.29 | jontow | my boss is threatening the equipment with a big foam hammer.. this is a bad sign.. |
| 18:17.35 | jontow | i think he should take up smoking; it'd mellow him out a bit |
| 18:17.52 | Katty | smoking :< |
| 18:17.52 | robin_sz | jontow: he has a headache ... |
| 18:17.57 | dudes | jobtow - smoking ... something green? He'd forget why he was mad in the first place. |
| 18:18.03 | Katty | jontow: he needs a nap |
| 18:18.06 | jontow | i didn't say anything about anything green |
| 18:18.11 | Katty | let's all nap! |
| 18:18.12 | jontow | but a nap would probably help quite nicely.. |
| 18:18.13 | dudes | jontow - I did. |
| 18:18.15 | robin_sz | jontow: its easy to cure his headache ... without drugs too. |
| 18:18.26 | jontow | and it isn't a headache so much as retarded equipment :) |
| 18:18.40 | cluecon | dudes: i'm not saying that * doesn't work well enough, it's getting it beyond enough to get it into the mainstream. I live in an area that is primarily call centers. If * had more features to deal with call center scenarios then it could easily replace existing dialers and call center setups to the point of exponential growth. |
| 18:18.41 | robin_sz | jontow: you know what a TENS machine is? |
| 18:18.49 | jontow | offhand; i do not. |
| 18:19.29 | robin_sz | jontow: used to cure muscle pain by applying fairly heavy electrical pulses via sticky pads. used in childbirth sometimes |
| 18:19.37 | jontow | aha |
| 18:20.05 | robin_sz | jontow: I reckon you could strap one across his temples for a couple of hours and he'll be real freindly and mild afterwards |
| 18:20.20 | robin_sz | perhaps drooling a little too. |
| 18:20.28 | jontow | :D |
| 18:20.36 | dudes | cluecon - I know what you're saying. |
| 18:20.43 | robin_sz | worked fine on my kids ;) |
| 18:20.44 | Katty | gosh |
| 18:21.07 | docelm0 | So clue Im guessing your going to cluecon with the rest of us? :) |
| 18:21.31 | cluecon | i'm trying to convice the boss. it all hinges on money. if the * setup works, then it's pretty much a sure thing. |
| 18:21.38 | dudes | cluecon - We've been working on asterisk to make it more acceptable for call centers in the main stream. Though we're a short time off, we have made some good progress. |
| 18:21.42 | Katty | cluecon: same for me.. |
| 18:21.53 | Lee__ | me too :) |
| 18:22.04 | Katty | cluecon: except it does work....but they're also thinking about vonexus :<<<<<<<<<<<<<<<<<<<<<< |
| 18:22.12 | dudes | cluecon - What exactly are you trying to do? |
| 18:22.59 | anthm | email sellmyboss@cluecon.com (not a joke) with details and bkw will call and get your boss to take you |
| 18:23.32 | *** join/#asterisk AntiPool (~bostjan@BSN-77-143-148.dsl.siol.net) |
| 18:23.41 | Lee__ | wow, the only Microsoft based VoIP service. thank god. |
| 18:23.51 | AntiPool | hi all, |
| 18:24.02 | blop | anyone has configured receiving SMS from Belgacom ? :) |
| 18:24.18 | blop | it works .. at half :p |
| 18:24.36 | AntiPool | i configured my X100P card and asterisk is running, how do i convice it to answer on ring ? |
| 18:25.02 | bkw_ | exten => s,1,Answer |
| 18:25.19 | cluecon | dudes: we are a collections company. we call an average of 25,000 people every day because they owe somebody else money. our present dialer limits us to 12 users max and 2 pri connections. we are preparing to roll out an * solution capable of at least 40 users and 8 outgoing pri connections. It all hinges on financing at this point and me sitting down and creating a little predictive dialer program to run the system. |
| 18:25.45 | cluecon | anthm: the boss is sold on cluecon, as long as * works out since it will be the moneymaker. |
| 18:26.00 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 18:26.17 | anthm | if you want it to work for callcenter |
| 18:26.36 | dudes | cluecon - have you looked at gnudialer |
| 18:26.37 | [TK]D-Fender | cluecon : www.aheeva.com <------ |
| 18:26.39 | anthm | look at icd, i've been working on it for > a year |
| 18:26.49 | anthm | it needs a few finishing touches |
| 18:27.07 | anthm | makes app_queue look like hello world |
| 18:27.18 | dudes | aheeva ... *sigh* |
| 18:27.23 | cluecon | anthm: I'm looking at icd. Just the first glance at it makes me want to try it. |
| 18:27.29 | *** join/#asterisk Corydon-w (beige@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 18:27.49 | bewest | icd? |
| 18:27.59 | JerJer | a few ? |
| 18:28.06 | JerJer | how about blown up and done with less bloat? |
| 18:28.36 | cluecon | dudes: i wasn't too thrilled with gnudialer. It did give me a few ideas to start with. Basically, we have the theory worked out on our predicitive stuff, we just have to write the code. It's very simple logic for us and will be fairly easy to do. |
| 18:29.04 | jontow | it looks good but isn't what i'd aim for |
| 18:29.15 | jontow | lots of goo^Wgui |
| 18:29.19 | jontow | :D |
| 18:29.47 | jontow | i looked at ICD too; but never could get it to compile/install cleanly (many months ago) |
| 18:29.56 | jontow | think i'll give it another shot here pretty soon ;) |
| 18:30.02 | dudes | cluecon - 1.5 kind of sucked, but that was a hackable workable dialer. 2.0 when it's out is probally more in your direction. |
| 18:30.56 | anthm | ICD needs people , it's on sf now so you have 3 developers waiting for feedback |
| 18:31.04 | cluecon | cost is another factor for us. * is free. digium hardware is cheap. our present dialer will cost us nearly $35,000 to add 12 more users and that is unreasonable for us. we can put in a whole new * system for about $18,000 and then it only cost us roughly $300 to add a person and that is only because we can't get the hardware for free. |
| 18:32.04 | JerJer | rm -rf icd* |
| 18:32.07 | dudes | cluecon - asterisk is the ticket because it is cheaper than conventional dialer |
| 18:32.48 | DeeJayTwo | In order to connect a T1 (PRI) with a PBX T1 interface, what kind of RJ-45 is needed? (pins order).. |
| 18:32.55 | JerJer | 12 45 |
| 18:33.00 | DeeJayTwo | IS it the same kind as for T1->Channel bank? |
| 18:33.22 | JerJer | just use a straight thru cat-5 patch cable if you are lazy |
| 18:33.24 | cluecon | DeeJayTwo, a standard RJ45 Cable should do it. |
| 18:33.30 | robin_sz | heh: http://voip-forum.tmcnet.com/voip-forum/forum/forum_posts.asp?TID=4300&PN=2 |
| 18:33.34 | dudes | T1 to T1 should be corssover |
| 18:34.01 | cluecon | yes, but a T1 to PBX would be straight-thru |
| 18:34.10 | JerJer | dudes: not from the smart jack to a TXXXP |
| 18:34.18 | DeeJayTwo | erm.. |
| 18:34.27 | robin_sz | so, basically, a cheap winmodem will work as an FXO, if you can fugre out how to configure * |
| 18:34.32 | Assid | umm.. can someone help me configure sip .. for softphone |
| 18:34.36 | JerJer | DeeJayTwo: you have a PRI from the telco, correct? |
| 18:34.40 | DeeJayTwo | yes |
| 18:34.47 | DeeJayTwo | I want to connect it to a TE410P |
| 18:34.56 | cluecon | DeeJayTwo: to connect the PRI to your * box, use a standard Network Patch Cable |
| 18:34.56 | JerJer | then jack in a standard RJ-45 patch cable into the plug on the smartjack |
| 18:35.01 | DeeJayTwo | ok... |
| 18:35.04 | JerJer | then jack the other in into the TE410P |
| 18:35.06 | DeeJayTwo | same pins alignment at both side.. |
| 18:35.11 | cluecon | yes |
| 18:35.13 | DeeJayTwo | nice |
| 18:35.15 | DeeJayTwo | thank you very much |
| 18:35.23 | cluecon | it is a standard network (rj45) cable. |
| 18:35.39 | dudes | robin_sz - it is possible, though you'd be better off buying a cheap intel537 modem (aka x100P) |
| 18:35.52 | JerJer | you can make one cat-5 cable run Two PRI if you custom crimp the ends |
| 18:35.54 | JerJer | 12 45 |
| 18:35.59 | Assid | A> it always shows me as no cmpatible codec |
| 18:36.11 | blop | dam http://www.voip-info.org/wiki-Asterisk+priorities :( |
| 18:36.18 | JerJer | robin_sz: no you are not better off to buy a cheap intel537... you will not enjoy it |
| 18:36.43 | JerJer | buy a TDM01B or nothing at all |
| 18:37.00 | dudes | JerJer - I was saying that'd be a better alternative to a cheap winmodem |
| 18:37.05 | JerJer | no |
| 18:37.22 | dudes | JerJer - they suck though, the intel537 X100P's, but they do work. |
| 18:37.32 | blop | I'm currently using Koolstart signalling protocol for my zap channels here in belgium, anyone using Loopstart / Groundstart ? should i ? |
| 18:37.37 | JerJer | if you do not have the PRECISE motorolla chipset that mark originally chose, it will cause problems |
| 18:37.56 | JerJer | echo, no caller*id, etc |
| 18:38.31 | JerJer | all these driver compatable clones are technically inferior and will not operate 100% correctly |
| 18:38.40 | JerJer | why do you think digium dropped them like a bad date? |
| 18:39.09 | dudes | JerJer - you don't need to explain to me. I've used intel537's and they work good for answering... nothing more. |
| 18:39.19 | JerJer | see nothing more |
| 18:39.44 | JerJer | Caveat Emptor |
| 18:39.52 | dudes | If you want cheap ... expect nothing more than the basics man |
| 18:39.52 | JerJer | you get what you pay for |
| 18:40.14 | [TK]D-Fender | Best POTS option IMO : Sipura SPA-3000 |
| 18:40.44 | JerJer | the spa-3000 is decent - has trouble with DSL on the lines though - even with multiple filters |
| 18:41.02 | [TK]D-Fender | Haven't heard about that yet... will look into. |
| 18:41.26 | JerJer | sipura acknowledged it was an issue |
| 18:41.33 | blop | spa3000 looks nice :) thx for the link |
| 18:42.57 | blop | no belgian zap channels users arround ? :D |
| 18:44.59 | *** join/#asterisk rephorm (~brian@ip67-95-13-60.z13-95-67.customer.algx.net) |
| 18:45.08 | blop | robin_sz: u know if someone has already suceeded ? |
| 18:45.45 | jalsot | hello |
| 18:46.16 | jalsot | does anybody know what is the difference between app_record and res_monitor? which use in which case? |
| 18:48.46 | Juggie | bah |
| 18:48.52 | Juggie | res_monitor allows you to snoop on an yline |
| 18:49.04 | Juggie | where as app_record will record what it hears on the line its running on |
| 18:49.43 | jalsot | but the channel is bi-directional so it will make the same, won't? |
| 18:51.37 | Lee__ | I'm having trouble loading zaptel as a module in a xen machine with a 2.6.11 kernel. it's missing a symbol called crc_ccitt_table. I recompiled the kernel with this .config option set to "y" yet zaptel is still complaining. How can I verify this symbol exists in my running kernel? |
| 18:52.31 | jalsot | anybody else in app_record vs. res_monitor question? |
| 18:52.36 | tzafrir_laptop | Lee__, modprobe it. don't insmod |
| 18:52.53 | tzafrir_laptop | alternatively: insmod crc32 |
| 18:52.57 | Lee__ | tzafrir_laptop: no difference. I haven't used insmod. |
| 18:53.52 | tzafrir_laptop | Lee__, and if there are still problems: and did 'depmod -a' run OK? |
| 18:53.56 | Assid | i still cant get this codecs to sync |
| 18:54.33 | Lee__ | tzafrir_laptop: yes, depmod -a is fine and crc32 is built into the kernel |
| 18:55.15 | Lee__ | I'll try rebuilding the entire kernel again... |
| 18:55.21 | *** join/#asterisk truescot (~na@c117243.upc-c.chello.nl) |
| 18:55.49 | truescot | can anyone tell me if there is any good documentation in AMP(asterisk management portal) |
| 18:55.59 | *** join/#asterisk stkn (nobody@stkn.developer.gentoo) |
| 18:56.03 | Lee__ | truescot: amportal.sf.net |
| 18:57.03 | truescot | yea been there, not much documentation |
| 18:57.25 | Lee__ | there's enough. I think you need to read a little closer. |
| 18:57.37 | truescot | i already have it installed etc its more the deatures and things i am looking for |
| 18:57.42 | Lee__ | if you are looking for something specific, try #amportal |
| 18:58.20 | truescot | am there but no-one else around so thought i would try here |
| 18:58.22 | Assid | hrmm |
| 18:58.37 | truescot | but i am sure i will work it out eventually :) |
| 18:58.42 | Assid | i am almost done getthing this crap to work.. if only i can get it to call back to this number |
| 18:59.32 | Assid | i have 2 extensions.. 2530 and 2530 .. when i login with the username of 2531 and call 2530 OR 2531 .. it keeps going to the 2530 user |
| 18:59.44 | Assid | errr.. 2 extension -> 2530/2531 |
| 18:59.51 | *** join/#asterisk outofjungle (~outofjung@61.17.134.123) |
| 19:00.54 | *** join/#asterisk {sean} (~sean@adsl-69-214-130-169.dsl.lgtpmi.ameritech.net) |
| 19:01.00 | {sean} | w00t |
| 19:01.48 | *** join/#asterisk doughecka_ (~Tad@doughecka.user) |
| 19:02.21 | outofjungle | i'm trying to emerge zaptel on gentoo 2.6 w/ udev. and it is failing |
| 19:02.31 | Assid | can someone help me with this part? |
| 19:02.32 | oej | l |
| 19:02.43 | {Sean} | does anybody have any resources for a good call queue management application for *? |
| 19:03.54 | phigan | jontow: Hey when I hit * during my outgoing message, it doesn't ask for password or do anything |
| 19:04.02 | Lee__ | outofjungle: how is it failing? |
| 19:04.40 | outofjungle | Lee__: http://rafb.net/paste/results/NoQnYW75.html |
| 19:05.12 | doughecka_ | I have a question about cisco phones... my cisco phones take FOREVER to boot |
| 19:05.30 | doughecka_ | we have cisco switches here, but when we boot on a non-cisco network, it boots fast |
| 19:05.54 | Lee__ | outofjungle: 1) that version is mad old. Don't use it. 2) I think that's an issue for bugs.gentoo.org or #gentoo-voip |
| 19:07.21 | robin_sz | w00t! ... FXO configured |
| 19:07.28 | outofjungle | Lee__: thanks. I'll look into that. actaully, i may not need the zaptel module as i dont have the hardware. i just want the SIP side of astereisk |
| 19:07.45 | Lee__ | outofjungle: you need it if you want meetme conferencing |
| 19:07.54 | Lee__ | ztdummy |
| 19:08.07 | *** join/#asterisk Juxt (~Juxt@64.135.20.202) |
| 19:08.16 | robin_sz | or a digium card ... |
| 19:08.29 | robin_sz | or a modem card ;) |
| 19:08.33 | outofjungle | Lee__: oh okay. i'll compile it anyways, as i'm playing with it as of now. |
| 19:09.08 | Lee__ | 1.0 stable from CVS is pretty consistent and should compile on gentoo. |
| 19:09.18 | Juxt | good afternoon |
| 19:09.27 | Juxt | i have a transcoding related question |
| 19:09.38 | robin_sz | I might even be able to test this interface if my wife gets off the phone ;) |
| 19:10.00 | robin_sz | stop now |
| 19:10.19 | rephorm | i have a simple dialplan: http://pastebin.ca/13122 any ideas why none of the NoOp's trigger after "main-menu" is played? (i just get an "Auto fallthrough") |
| 19:11.04 | rephorm | the fallthrough is immediately after the Background() is finished, with no time to enter an extension |
| 19:11.48 | rephorm | (also, i _do_ get the no-op's on ext. 1 & 2, provided i hit 1 or 2 while main-menu is playing) |
| 19:14.29 | rephorm | ahh. it goes to hangup, not timeout... |
| 19:14.58 | *** join/#asterisk manipura (~phpwizard@S01060011954c9d46.cg.shawcable.net) |
| 19:15.41 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
| 19:16.56 | bkw_ | ok guys you can register for cluecon now.. and pay later .... I have it in the registration option now. |
| 19:16.58 | bkw_ | ;) |
| 19:17.59 | {Sean} | how about go and not pay ever? |
| 19:18.17 | bkw_ | haha |
| 19:18.19 | Assid | can someone hear me? |
| 19:18.28 | robin_sz | bkw_: can I pay after the event? |
| 19:18.30 | *** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3771041.sympatico.ca) |
| 19:18.35 | DaLion | hi all |
| 19:18.38 | bkw_ | robin_sz, haha no |
| 19:18.41 | bkw_ | doughecka_, yo yo |
| 19:18.41 | rephorm | does ResponseTimeout work with recent cvs head? |
| 19:18.47 | bkw_ | yes |
| 19:18.49 | DaLion | can one tel me if you can push a dial to a specific channel ? |
| 19:18.50 | bkw_ | but use the function |
| 19:18.52 | DaLion | like xten.. |
| 19:18.59 | doughecka_ | bkw_: wheres echo coming from on a T1? |
| 19:19.01 | DaLion | Dial (SIP/username/channel) |
| 19:19.07 | DaLion | so would ring on channgel channel |
| 19:19.08 | bkw_ | doughecka_, you hear echo? |
| 19:19.09 | {Sean} | i will pay in access to my resold vonage account |
| 19:19.10 | Elshar | Is anyone aware of any OSS stun servers other than what's listed on the voip-info wiki? ;) |
| 19:19.11 | shido | has anyone ever burned a CD image to a DVD , how do you make an iso from the dvd the cd is 700mb but the dvd is 4.7Gb how do you make the 700mb iso off the dvd? |
| 19:19.12 | bkw_ | of the person you're calling? |
| 19:19.13 | doughecka_ | I only get echo on a couple calls, and I can figure out how to tweak it.. |
| 19:19.36 | bkw_ | you can't |
| 19:19.36 | doughecka_ | I hear it, the other person cant |
| 19:19.37 | doughecka_ | crap |
| 19:19.37 | *** part/#asterisk jaike (~a@203.131.137.77) |
| 19:20.19 | bkw_ | if you hear it.. its the other end thats causing it |
| 19:20.20 | DaLion | anyone got an idea ? |
| 19:20.20 | doughecka_ | hmm |
| 19:20.20 | bkw_ | DaLion, be more specific |
| 19:20.20 | {Sean} | what's the best call queue management application available for *? |
| 19:20.20 | doughecka_ | but its annoying and calling from a regular analog line, I dont get it |
| 19:20.22 | DaLion | ok i Dial(SIP/myusername/3) would ring channel 3 of my eyebeam or xten or whatever |
| 19:20.27 | DaLion | not channel 1 |
| 19:20.28 | bkw_ | you can't |
| 19:20.31 | DaLion | oh |
| 19:20.31 | bkw_ | NEXT!!! |
| 19:20.33 | DaLion | darn |
| 19:20.36 | DaLion | !next it is |
| 19:20.42 | DaLion | thanks Brian |
| 19:20.51 | bkw_ | ;) |
| 19:21.18 | DaLion | well.. but a cisco or polycom .. you cant push neither to a specific line ? |
| 19:21.18 | doughecka_ | bkw_: ok, next question, my cisco phone takes FOREVER to boot |
| 19:21.22 | doughecka_ | close to 2-3 min to completly boot |
| 19:21.23 | DaLion | i guess you would need to tie the lines to a seperate account then |
| 19:21.27 | bkw_ | doughecka_, and? |
| 19:21.32 | bkw_ | DaLion, ya |
| 19:21.32 | Juxt | is G729 that much better than iLBC? |
| 19:21.45 | bkw_ | ilbc uses more CPU in some cases |
| 19:21.47 | doughecka_ | well, I took it to a non-cisco network, and it booted in 30 seconds |
| 19:21.48 | DaLion | ya to .. #1 or #2 |
| 19:21.50 | bkw_ | but they are about the same |
| 19:21.56 | bkw_ | doughecka_, turn of CDP |
| 19:21.59 | bkw_ | er off |
| 19:22.02 | DaLion | like ". but a cisco or polycom .. you cant push neither to a specific line ?" |
| 19:22.06 | doughecka_ | CDP? |
| 19:22.09 | doughecka_ | on the phone? |
| 19:22.11 | DaLion | or ya to "guess you would need to tie the lines to a seperate account then" |
| 19:22.21 | bkw_ | try that |
| 19:22.29 | Juxt | i am using iLBC for trunking once the call goes out of my office, all intra-office calls are ULAW |
| 19:22.31 | DaLion | that being.. #2 |
| 19:22.33 | DaLion | ok thanks |
| 19:22.37 | doughecka_ | bkw_: whats cdp standfor? |
| 19:22.42 | DaLion | hmm.. |
| 19:22.46 | Juxt | i was wondering how many calls i can do on a P4 box |
| 19:22.57 | doughecka_ | oh, google |
| 19:22.58 | doughecka_ | yay |
| 19:23.09 | DaLion | http://nltk.sourceforge.net/ |
| 19:23.22 | DaLion | ok thank Brian again ill leave you to your stuff now |
| 19:23.45 | Elshar | Does anyone know what stern is? ;) |
| 19:24.49 | wasim | last name of a howard? |
| 19:25.18 | jalsot | does anybody know what is the difference between app_record and res_monitor? which use in which case? |
| 19:25.18 | Essobi | ummm |
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| 19:25.31 | wasim | business school at nyu? |
| 19:25.39 | Elshar | wasim: Funny. :) |
| 19:25.41 | wasim | rear of a boat? |
| 19:25.48 | Assid | can someone help me with something please |
| 19:25.57 | Assid | i cant seem to make a call route correctly |
| 19:26.14 | Assid | all my calls always end up on the same extension |
| 19:26.27 | blop | who's playing with SMS ? |
| 19:26.31 | Elshar | I'm trying to find a stun server, and one of them has an option for a "STERN server starting port" whatever that is. :P |
| 19:26.35 | blop | i got some questions :) |
| 19:26.48 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
| 19:27.03 | gordonjcp | Elshar: it's like FRIENDLY server starting port, but not so friendly |
| 19:27.27 | rephorm | can anyone help me with my simple IVR? |
| 19:27.35 | Elshar | Hrm.. Maybe I should just read the code :) |
| 19:27.43 | Elshar | Hopefully he'd have documented it somewhere? :P |
| 19:28.15 | rephorm | i'm not getting any sort of response time after the last priority is complete. instead it drops directly to 'h'... |
| 19:28.47 | *** part/#asterisk apropos (tom@wsip-68-15-205-134.ok.ok.cox.net) |
| 19:28.55 | lesouvage | callerid isn't working with incoming calls. Should I enable it in one of the .conf files? |
| 19:29.18 | Elshar | I guess this all started because the stupid linksys rt31p2-na's have some odd settings I didn't see in the pap2-na's.. :P |
| 19:30.51 | Juxt | does anyone know if it's possible to do "voice coloring" with asterisk? |
| 19:31.05 | bewest | voice coloring? |
| 19:31.10 | Elshar | I like my voices blue :) |
| 19:31.13 | bewest | like a disguise? |
| 19:31.13 | Elshar | Makes me more.. relaxed |
| 19:31.25 | Juxt | well it's very useful in a conference call |
| 19:31.31 | *** join/#asterisk darkskiez (~mhb@host-84-9-78-76.bulldogdsl.com) |
| 19:31.34 | *** join/#asterisk santiago (~santiago@63.245.86.184) |
| 19:31.35 | Juxt | this way when people speak everyone knows who they are |
| 19:31.44 | bewest | ? |
| 19:31.52 | Juxt | it adds some "color" to the voice, e.g. 1 person sounds like a baritone |
| 19:32.09 | Juxt | and one sounds like darth vader |
| 19:32.16 | rephorm | am i mistaken in assuming that asterisk should wait 15 seconds for a response after Menu|s|1 in http://pastebin.ca/13122? |
| 19:32.19 | Elshar | Heh, do any of them ever play with it? |
| 19:32.25 | robin_sz | so .. callerid on an FXO .. POTS line incoming .. FXO .. * ... will I get callerid? |
| 19:32.34 | robin_sz | this is UK CID by the wya |
| 19:32.35 | bewest | Juxt, I'm not aware of anything |
| 19:32.39 | Elshar | "looke, I am your faathor" |
| 19:32.41 | robin_sz | not the USian one |
| 19:32.51 | Juxt | it's a pretty nifty concept |
| 19:32.57 | bewest | Elshar, noooOOOOOOOOO |
| 19:32.57 | Elshar | Sounds cool |
| 19:32.57 | DaLion | how would asteirsk handle carp ? |
| 19:33.07 | Juxt | if it was possible to apply some soft of an eq to the channel it'd make it possible |
| 19:33.10 | DaLion | any evident problems with arp and mac cloning ? |
| 19:33.30 | Juxt | could it be possible to pass the entire channel thru something like sox? |
| 19:33.57 | lesouvage | robin_sz: I have the same problem. No callerid on an incoming line. I live in Holland |
| 19:34.12 | darkskiez | does call parking actually work in asterisk stable? How can i convince asterisk where to send the call if it times out ? |
| 19:34.13 | doughecka_ | bkw_: any suggestions on how to turn it off? I cant find anything on my switches about cdp |
| 19:34.52 | robin_sz | lesouvage: I have the CID on the line .. but I dont think it is decoded .. UK CID is different to everyone elses anyway ... |
| 19:35.25 | robin_sz | lesouvage: we start by reversing the line voltage ... AFAIK no other country does that |
| 19:35.49 | bkw_ | doughecka_, no cdp run |
| 19:36.04 | doughecka_ | do I have to do something to save it over a reboot? |
| 19:36.07 | lesouvage | robin_sz: something like driving on the left site of the street |
| 19:36.28 | robin_sz | lesouvage: no, thats a sensible thing ;) |
| 19:37.14 | doughecka_ | schweet, thats faster |
| 19:37.29 | [TK]D-Fender | rephorm : You shouldn't assume the timeout, you should DEFINE it. exten =>s,2,digittimeout(15) |
| 19:38.05 | gambolputty | hi |
| 19:38.22 | *** join/#asterisk marten_ (~sm7syx@212-162-184-20.skbbip.com) |
| 19:38.31 | gambolputty | Has anyone gotten an update to work with the mysql command? |
| 19:39.12 | *** join/#asterisk kant (~bernd@metrored-gw.tropicohn.com) |
| 19:39.29 | rephorm | [TK]D-Fender: yeah, i tried that. forgot to update the pastebin. i have this now: http://pastebin.ca/13127 and still get dropped to Menu|h|1 as soon as the Background() is finished |
| 19:40.45 | [TK]D-Fender | Switch the order of the timeout and background commands. I believe Background expects something to follow it. |
| 19:41.48 | rephorm | [TK]D-Fender: nope. |
| 19:42.16 | marten_ | Newbie question ! I have a 'nice' error when staring up. (Registered translator 'gsmtolin' from format GSM to SLINR, cost 9) Illegal instruction !!!! Any idea anyone ? |
| 19:42.42 | rephorm | [TK]D-Fender: all of the examples i've seen look like what i did. i've been hacking around it by addind a 10 sec silent file. but, i'd rather be able to have true timeouts. |
| 19:43.22 | [TK]D-Fender | rephorm : can you paste your current version. |
| 19:44.32 | *** join/#asterisk goldenolden (~goldenold@65.171.196.23) |
| 19:44.33 | goldenolden | hi |
| 19:44.39 | marten_ | Newbie question 2 ! I would like to set my asterisk up as a sip-sip gateway no phonline just sip ! Any good starting point / howto / examples ? |
| 19:44.50 | goldenolden | anyone have any issues with no audio over a LAN based sip connection to asterisk? |
| 19:44.57 | goldenolden | like if calling voicemail, no audio |
| 19:45.01 | goldenolden | or doing a .gsm file playback |
| 19:45.04 | rephorm | http://pastebin.ca/13129 |
| 19:45.13 | goldenolden | essentially anything that's not bridged between two sip clients produces no audio |
| 19:47.18 | [TK]D-Fender | rephorm : do you get your step 4 NoOp? |
| 19:47.32 | rephorm | [TK]D-Fender: http://pastebin.ca/13130 has the console output |
| 19:47.35 | Assid | damn |
| 19:47.38 | rephorm | [TK]D-Fender: yes. |
| 19:47.38 | Assid | i cant make a new account |
| 19:48.19 | tzafrir_laptop | marten_, if this is to another asterisk, use iax instead of sip |
| 19:48.21 | *** join/#asterisk muxxum (~kvirc@ppp-69-217-121-102.dsl.chcgil.ameritech.net) |
| 19:49.07 | muxxum | Hey guys |
| 19:49.12 | DaLion | he |
| 19:50.00 | *** part/#asterisk Juxt (~Juxt@64.135.20.202) |
| 19:50.04 | marten_ | tzafrir_laptop, Sorry you lost me there ! IAX ??? And yes I like to connect my asterisk to my providers asterisk. |
| 19:50.34 | tzafrir_laptop | marten_, searches keyword: trunk |
| 19:50.35 | marten_ | tzafrir_laptop, But why does it crash ? |
| 19:50.38 | rephorm | [TK]D-Fender: do you have a working IVR snippet i could look at? (or could you test that one on your setup?) i'm starting to think its something outside of extensions.conf |
| 19:51.16 | marten_ | tzafrir_laptop, I look into trunk. Need to switch computer... |
| 19:51.35 | muxxum | If I dial into Asterisk can I make a menu selection and have that selection automatically kickoff I program I wrote?ogram |
| 19:51.52 | DaLion | yeha |
| 19:51.56 | DaLion | read about AGI |
| 19:52.20 | tzafrir_laptop | marten_, as for crashing: it shouldn't. my doesn't. If your does and you don't understand why and want help, give more details |
| 19:52.20 | muxxum | AGI? googling now |
| 19:52.22 | rephorm | muxxum: or see the System application |
| 19:52.30 | rephorm | show application System |
| 19:52.33 | marten_ | tzafrir_laptop, I |
| 19:52.47 | Nuxi | muxxum http://www.voip-info.org/tiki-index.php?page=Asterisk%20AGI |
| 19:52.57 | muxxum | wow thanks everyone |
| 19:53.01 | tzafrir_laptop | muxxum, what program? For very simple cases you can use 'System' from the dialplan |
| 19:53.11 | goldenolden | god this is annoying |
| 19:53.12 | Nuxi | hehe muxxum spelt wow backwards. |
| 19:53.17 | goldenolden | no audio on basic voicemail playback... any ideas? |
| 19:53.32 | rephorm | Nuxi: and you spelled muxxum backwards... |
| 19:54.08 | tzafrir_laptop | even worse, with the proper font, that nick can be mirrored |
| 19:54.32 | rephorm | ok, can ANYONE here verify that ResponseTimeout() is in fact working in CVS Head? |
| 19:54.46 | muxxum | tzafrir_laptop a network/system configuration management prog I wrote |
| 19:55.10 | rephorm | goldenolden: do you get audio from Playback()? |
| 19:55.36 | marten_ | tzafrir_laptop, As i stated I'm new at asterisk, I downloaded and installed and tried to start it up whitout editing the config files ... (asterisk -vvvgc) and got the 'illigal instruction' error. So trying to edit some of the sample config files I'm still stuck with that error. And as beeing new to asterisk i don't really know what all the files are doing :-( |
| 19:55.38 | rephorm | goldenolden: and, do you get two way phone audio? |
| 19:55.44 | muxxum | So it seems one could use Asterisk as gateway to initiate other programs.... |
| 19:55.55 | [TK]D-Fender | rephorm : Yeah, thats wierd.... |
| 19:56.13 | Nuxi | muxxum, yup and you can communicate with * too. See also the manager api |
| 19:56.24 | [TK]D-Fender | rephorm : Strange idea - Put a redundant "answer" at the start of your menu. |
| 19:56.36 | tzafrir_laptop | muxxum, you could initial through System, and probably through curl (to post to a URL) as well |
| 19:56.42 | Nuxi | http://www.voip-info.org/wiki-Asterisk+manager+API |
| 19:57.14 | muxxum | Hmmm, I think system becuase my program would reside on the same box but that is good to know |
| 19:57.20 | rephorm | [TK]D-Fender: nope. i've tried that before also. adding/removing Answer's in any resonable location :) |
| 19:57.38 | [TK]D-Fender | BLARG! |
| 19:57.44 | goldenolden | rephorm: no audio |
| 19:57.45 | tzafrir_laptop | marten_, if you want a simpler start, I can also recommend you Rapid |
| 19:57.48 | tzafrir_laptop | ~rapid |
| 19:57.49 | jbot | methinks rapid is [xorcom rapid] at http://www.xorcom.com/rapid . A Debian based distro that features an auto-installer which installs and configures both Debian, and Asterisk. Maintainer: tzafrir |
| 19:57.59 | lesouvage | I'm runing gastman on windows xp home edition to monitor a Xorcom box. When running a cli command gastman brakes. Is this a known error with the Gastman Windows installer? On linux it runs fine. |
| 19:58.04 | goldenolden | rephorm: I do get two way audio only on bridged SIP calls that do not have the media path going through asterisk |
| 19:58.24 | rephorm | goldenolden: hmm. firewall issues? |
| 19:58.32 | tzafrir_laptop | lesouvage, could you give more details? |
| 19:58.34 | rephorm | goldenolden: the asterisk box blocking rtp ports? |
| 19:59.17 | rephorm | [TK]D-Fender: i guess its time to post to the list... :) |
| 19:59.23 | marten_ | tzafrir_laptop, Now I'm lost "rapid" how where what ? |
| 19:59.34 | muxxum | I guess one could use Asterisk to make phone calls as well based on certain menu selections or events fed into an Asterisk API... |
| 19:59.37 | lesouvage | Marten: I'm not the maintainer but I have good experiences with Rapid. It's ready out of the box. |
| 20:00.22 | tzafrir_laptop | marten_, it is a linux distro that installs Asterisk |
| 20:00.33 | tzafrir_laptop | http://xorcom.com/rapid |
| 20:00.40 | [TK]D-Fender | rephorm : I've seen other posts on the subject, but no solutions. |
| 20:00.42 | robin_sz | OK, so .. I can do the "press 1 for sales, press 2 for support" thing ... how do I do the "or please hold for the operator"?? |
| 20:00.49 | tzafrir_laptop | a 200MB ISO image |
| 20:01.25 | tzafrir_laptop | robin_sz, goto(context,s,1) |
| 20:01.28 | cluecon | robin_sz: have your timeout send to the operator, like the following: |
| 20:01.49 | tzafrir_laptop | oops, robin_sz , ignore |
| 20:02.01 | marten_ | tzafrir_laptop, Ok got it, I tried to use my server and a new image on that is not recomended ! So I have to stay with my config files.. (reading doc) |
| 20:02.09 | robin_sz | yeak, got I have timeout .. thats the key. |
| 20:02.18 | cluecon | exten => t,1,Goto(operator,s,1) ; If they take too long, send to the operator |
| 20:02.44 | *** join/#asterisk Corydon76-home (four@pcp08665860pcs.500ash01.tn.comcast.net) |
| 20:02.53 | robin_sz | right |
| 20:02.56 | robin_sz | thanks |
| 20:03.00 | cluecon | yvw |
| 20:03.10 | cluecon | ~cluecon |
| 20:03.12 | jbot | from memory, cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. |
| 20:03.39 | robin_sz | and finally .. to ring a group of phoens and let any one pick it up? |
| 20:03.39 | tzafrir_laptop | where's jbot? |
| 20:03.44 | tzafrir_laptop | hmmm? |
| 20:03.51 | tzafrir_laptop | ~seen jbot |
| 20:04.05 | jbot | jbot <ibot@apt.bot.TimRiker.active.supporter.pdpc> was last seen on IRC in channel #dub, 83d 4h 59m 26s ago, saying: 'somebody said leave was what lazarus should do now'. |
| 20:04.05 | robin_sz | (the operator in this case is anyone isn sales ;) |
| 20:04.21 | jalsot | does anybody know how to mix slinear channels? I mean, from the mathematical side... |
| 20:04.35 | cluecon | robin: exten => 2,1,Dial(SIP/234&SIP/250,10,t) where SIP/234 and SIP/250 are your sales extensions. |
| 20:04.46 | robin_sz | cluecon: ahh, & |
| 20:04.49 | robin_sz | thanks |
| 20:05.21 | cluecon | tzafrir: jbot is now known as jbot_ |
| 20:05.31 | cluecon | ~seen jbot_ |
| 20:05.33 | jbot | i haven't seen 'jbot_', cluecon |
| 20:05.34 | tzafrir_laptop | robin_sz, don't you get a small "tick" in the end when recording with Record? |
| 20:05.57 | rephorm | cluecon: i'm having trouble actually getting to a timeout extension. i have a really simple IVR set up, and i'm getting dropped to 'h' instead of 't' (immediately after the last priority -- i.e. without a timeout). the extensions.conf is at: http://pastebin.ca/13130 |
| 20:05.58 | robin_sz | tzafrir_laptop: not that I noticed ... |
| 20:06.06 | robin_sz | I'll try again later .. |
| 20:06.09 | rephorm | cluecon: any ideas? |
| 20:06.34 | lesouvage | tzafrir_laptop: Gastman breaks when I use the cli option of Gastman. That's sad because the panel en the call-list look nice and easy. |
| 20:06.36 | cluecon | rephorm: lots of ideas, but none that relate to your problem. Let me take a minute to ponder it and I'll let ya know. |
| 20:06.42 | rephorm | cluecon: thanks ;) |
| 20:07.00 | tzafrir_laptop | lesouvage, can you pastebin the CLI of what happens? |
| 20:07.16 | tzafrir_laptop | set verbose 10 before that, please |
| 20:07.50 | lesouvage | tzafrir_laptop: Yes, wait a moment. |
| 20:08.14 | bkw_ | ok quick |
| 20:08.25 | bkw_ | everyone raise your hands if you know how to make something 110% reliable |
| 20:08.46 | cluecon | i do. but i would have to kill you if i told you how. |
| 20:08.56 | claude2005 | Hello there |
| 20:08.56 | claude2005 | Can any one help, I am having a problem, when an external caller calls, and then drops the call, the internal phone keeps ringing even though the external call is been dropped. |
| 20:09.08 | DaLion | hmm.. congestion mayne |
| 20:09.18 | DaLion | claude2005 try a congestion error handling |
| 20:09.21 | DaLion | or something |
| 20:09.31 | claude2005 | am new to this |
| 20:09.38 | DaLion | look at logs also could give you a clue |
| 20:09.43 | claude2005 | so you have to give me abit more direction please |
| 20:09.59 | DaLion | and casued by.. maybe codec negociation |
| 20:10.02 | tzafrir_laptop | claude2005, first off, soft hangup <tab><tab> |
| 20:10.03 | PBXtech | go left |
| 20:10.46 | claude2005 | i am using a BT uk land line for incomming |
| 20:10.56 | claude2005 | tdm22p |
| 20:11.55 | *** join/#asterisk jayk_ (jayk@vapid.reprehensible.net) |
| 20:12.16 | cluecon | rephorm: try changing your s extension to this and see if it works: |
| 20:12.17 | cluecon | exten => s,1,Answer() |
| 20:12.17 | cluecon | exten => s,2,Background(main-menu) |
| 20:12.17 | cluecon | exten => s,3,WaitExten(15) |
| 20:12.17 | cluecon | exten => s,4,NoOp(After Background) |
| 20:12.22 | jayk_ | i have asterisk running with Cisco 7960 phones..whenever anybody leaves a voicemail, the light on the handset does not come on. |
| 20:12.33 | PBXtech | that sucks |
| 20:12.36 | jayk_ | does anybody know why? i also enabled stutter but i don't hear a stutter when i pick up the handset. |
| 20:12.50 | Juggie | because you dont have it setup right |
| 20:12.52 | Juggie | it works for me |
| 20:12.57 | jayk_ | :) |
| 20:13.06 | Juggie | did you set mailbox=exten@context |
| 20:13.09 | Juggie | in sip.conf |
| 20:13.30 | jayk_ | i have mailbox=extension_number |
| 20:14.00 | Juggie | add the VM context as well |
| 20:14.06 | Juggie | tho i'm not sure if its required |
| 20:14.38 | jayk_ | i don't have the context defined |
| 20:14.43 | *** join/#asterisk santiago (~santiago@63.245.86.184) |
| 20:14.44 | jayk_ | in voicemail.conf, my mailboxes are all under other |
| 20:15.00 | Juggie | then say mailbox=exten@other |
| 20:15.01 | jayk_ | should i put mailbox=exten#@other? |
| 20:15.05 | jayk_ | ok |
| 20:15.16 | jayk_ | man if it is that easy im gonna kick myself |
| 20:15.34 | Juggie | your also gonna wanna make sure when you call voicemail from your dialplan you specify other as well. |
| 20:15.56 | rephorm | cluecon: ok. that works. however, after the WaitExten is over, i still get sent to 'h' instead of 't'. (in other words, practically its a good work around, but something is definitely not right with ResponseTimeout()) |
| 20:16.15 | cluecon | what * version are you running? |
| 20:16.22 | *** join/#asterisk jsolares (~jsolares@200.30.141.86) |
| 20:16.33 | Assid | whats a good sip softphone? |
| 20:16.38 | *** part/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 20:16.44 | Assid | besides xlite |
| 20:16.49 | rephorm | cluecon: cvs head as of today. however, i've had the same problem since i got into asterisk in february. |
| 20:17.35 | cluecon | try taking out the s,4 entry. |
| 20:17.43 | muxxum | There is even a tcl "amclient.tcl" to access the manager. Awesome. |
| 20:18.23 | jayk_ | oh that's sweet |
| 20:18.25 | jayk_ | the light came on |
| 20:18.32 | rephorm | cluecon: i originally just had the responsetimeout and the background lines |
| 20:18.34 | rephorm | cluecon: no luck. |
| 20:18.38 | *** join/#asterisk claude2005 (~claude@kcassidy.plus.com) |
| 20:18.48 | rephorm | cluecon: (well, and the h,i,t NoOp's to see what was going on) |
| 20:19.05 | rephorm | wait. |
| 20:19.30 | anthm | asterisk motto get your shit together, extens s,h,i and t |
| 20:19.38 | rephorm | gah. sorry. figured it out. there was an "autofallthrough=yes" at the top |
| 20:19.59 | rephorm | never noticed that. (modified a coworkers ext.conf to get started...) |
| 20:20.13 | rephorm | <PROTECTED> |
| 20:21.21 | cluecon | yvw rephorm. |
| 20:22.57 | *** part/#asterisk km- (~pgrace@brdgw1.rttx.com) |
| 20:23.25 | rephorm | hmm. the default extensions.conf has autofallthrough on. the wiki needs some updating |
| 20:23.27 | darkskiez | how is call parking supposed to correctly send the call back to the caller in v1-0 ? |
| 20:25.46 | *** join/#asterisk file[mac] (~jcolp@mctn1-7469.nb.aliant.net) |
| 20:25.49 | jalsot | does anybody know what is the difference between app_record and res_monitor? which use in which case? is it possible to record mixed channels rather than in/out channels separately? |
| 20:26.28 | PBXtech | yea like one records a file, the other monitors/records a channel |
| 20:26.45 | lesouvage | tzafrir_laptop: The window application Gastman breaks without a trace in the cli. Maybe it's because it's compiled for window 2k and I run it on Windows XP home edition. |
| 20:27.05 | jalsot | PBXtech: both records channels into file, don't they? |
| 20:27.35 | PBXtech | record records to a file |
| 20:27.56 | jalsot | I use monitor [res_monitor] but found app_record and I don't know what is the difference |
| 20:28.07 | jalsot | PBXtech: into 1 file? |
| 20:28.17 | jayk_ | Juggie: thanks, that context did the trick |
| 20:28.25 | PBXtech | if your recording a channel use monitor |
| 20:28.34 | PBXtech | if your recording a menu prompt use record |
| 20:28.50 | jalsot | records recor only one direction? |
| 20:28.57 | PBXtech | no |
| 20:28.58 | jalsot | on a bi-directional channel? |
| 20:29.05 | PBXtech | no |
| 20:29.15 | PBXtech | maybe |
| 20:29.24 | kajtzu | SIP/2.0 488 Not Acceptable Here (codec error) |
| 20:29.29 | kajtzu | hmm. what does that really mean? |
| 20:29.30 | PBXtech | it records you to a file for a menu prompt |
| 20:29.45 | PBXtech | bad codec eh |
| 20:29.51 | jalsot | sorry, still don'tknow what is the difference |
| 20:29.59 | jalsot | I have a channel, e.g. IAX2 |
| 20:30.01 | kajtzu | the only difference is that the user is using GPRS instead of fixed ethernet or Wireless lan |
| 20:30.11 | jalsot | if I use app_record, what is recorded? |
| 20:30.21 | jalsot | [I know what does res_monitor] |
| 20:30.30 | PBXtech | how are you going to use app_record to record a channel |
| 20:30.49 | jalsot | PBXtech: didn't try, I'm just curious |
| 20:31.03 | *** join/#asterisk cluecon[file] (~file@mctn1-7469.nb.aliant.net) |
| 20:31.12 | PBXtech | try it then :) |
| 20:31.39 | jalsot | from the description of app_record: "Records from the channel into a given filename." |
| 20:31.58 | PBXtech | yup try it |
| 20:32.11 | PBXtech | http://www.voip-info.org/wiki-Asterisk+cmd+Record |
| 20:32.39 | PBXtech | http://www.voip-info.org/wiki-Asterisk+cmd+Monitor |
| 20:33.13 | jalsot | so record records only voice from asterisk |
| 20:33.24 | jalsot | not both directions |
| 20:33.45 | muxxum | Thanks for all the help! |
| 20:33.48 | *** part/#asterisk muxxum (~kvirc@ppp-69-217-121-102.dsl.chcgil.ameritech.net) |
| 20:34.26 | rephorm | jalsot: app_record records what's coming to asterisk from the phone. its used to do things like record voicemail prompts, etc. |
| 20:34.51 | jalsot | rephorm: oh, now I underestood, thanks! |
| 20:34.58 | rephorm | jalsot: np :) |
| 20:35.22 | jalsot | so recording bi-directional calls is what res_monitor is used for |
| 20:35.41 | jontow | anthm around? |
| 20:36.36 | jontow | ;) |
| 20:36.45 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 20:36.59 | jontow | more seriously; what version of asterisk is ICD tested working with? |
| 20:37.03 | anthm | oh you're here to register for cluecon yay! |
| 20:37.05 | jontow | rather; what is the *recommended* version? |
| 20:37.09 | jontow | i wish :( |
| 20:37.14 | jalsot | is it possible [hack?] to store recorded in/out streams mixed? [e.g. slinear format for saving disk space and I/O] |
| 20:37.18 | cluecon | anthm is more square than round even though he does a pretty good ball on occassion. |
| 20:37.21 | cluecon | ~cluecon |
| 20:37.24 | jbot | i heard cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. |
| 20:37.39 | anthm | icd tried to keep up with head |
| 20:37.52 | jontow | did it work? :D |
| 20:37.55 | anthm | so it probably works best there but it might |
| 20:38.11 | anthm | i meant it tries |
| 20:38.30 | rephorm | jalsot: app_monitor mixes them together (wav format) |
| 20:38.49 | jalsot | rephorm: I mean, on-the-fly, not after call hangup |
| 20:39.09 | jontow | :) will give it a shot, then |
| 20:39.17 | jalsot | the current way is not an optimized way, IMO |
| 20:39.39 | jalsot | higher disk usage and capacity requirements |
| 20:40.02 | darkskiez | I like it |
| 20:41.22 | goldenolden | nope, no firewall issues on the one way audio |
| 20:41.24 | *** join/#asterisk alegh (~ag11@OL12-112.fibertel.com.ar) |
| 20:41.31 | goldenolden | this is killing me |
| 20:42.19 | alegh | Hi anyone with experience with Grandstream GXP-2000 compared with sipura 841? |
| 20:42.59 | jontow | golden; in my experience, things that kill you are bad. and often very simply solved. |
| 20:44.30 | goldenolden | :) |
| 20:44.49 | rephorm | jalsot: ahh. i see what you mean. hmm. you'd have to modify monitor to do it. |
| 20:45.15 | jalsot | rephorm: do you know how to mix 2 slinear channels? |
| 20:45.27 | rephorm | jalsot: not sure what slinear is... |
| 20:45.36 | jalsot | I mean, mathematical side... |
| 20:45.45 | rephorm | jalsot: i'm guessing its not a raw waveform? |
| 20:45.53 | jalsot | slinear is 16bit samples at 8kHz rate, AFAIK |
| 20:46.04 | jalsot | rephorm: it's a raw voice |
| 20:46.38 | Assid | hrmm |
| 20:46.41 | Assid | i keep getting No entry in voicemail config file for '' |
| 20:46.51 | Assid | but i already put in a entry for this extension |
| 20:47.20 | claude2005 | Any one can help, I am still having a problem when an external caller calls and drops the call the internal phone keeps ringing regardless the external caller has hung-up |
| 20:47.21 | rephorm | jalsot: can't you just add them then? |
| 20:47.51 | jalsot | rephorm: unfortunately I don't have signal processing background, I don't know :( ...anybody? |
| 20:48.44 | jalsot | I found a thread on asterisk-users from 2003 where there was a talk about recording into 1 file, but it looks it has never applied |
| 20:49.01 | jalsot | http://lists.digium.com/pipermail/asterisk-users/2003-May/011923.html |
| 20:49.03 | Nuxi | I think that averaging the samples would be fine. You coud dft the wave and then average and the dft again, but it should turn out the same. |
| 20:49.48 | rephorm | Nuxi: yeah. averaging makes sense |
| 20:49.59 | jalsot | Nuxi: do you mean: (sampleIn+sampleOut)/2? |
| 20:50.18 | jalsot | I read something like that, but don't know if it gives good quality... |
| 20:50.20 | Nuxi | You'll have a lot of hard times trying to do a dft as you get data, but an average can be done realtime. |
| 20:50.28 | jalsot | and what about rounding... |
| 20:50.48 | Nuxi | You'll have far worse rounding with normal signal processing techniques. |
| 20:50.50 | jalsot | dft means discrete Fourier transformation? |
| 20:50.54 | Nuxi | yup. |
| 20:51.03 | jalsot | ah |
| 20:51.19 | jalsot | maybe taking a look into sox's source will tell something |
| 20:51.33 | goldenolden | okay so no audio when I dial into the PBX for voicemail... same LAN, no difference... |
| 20:52.00 | jontow | golden; what phone? |
| 20:52.10 | Nuxi | (wave1 + wave2) /2 == dft^-1((dft(wave1) + dft(wave2)) / 2) |
| 20:52.16 | goldenolden | using both a SIPURA 841 and Cisco 7940 |
| 20:52.19 | goldenolden | same problem |
| 20:52.24 | goldenolden | I've played with CODECS too |
| 20:52.25 | *** join/#asterisk irv999 (~irv999@ool-44c241c7.dyn.optonline.net) |
| 20:52.27 | jontow | what is 'dtmfmode' set to in sip.conf ? |
| 20:52.28 | goldenolden | using just straight ULAW right now |
| 20:52.30 | irv999 | LO ALL! :-D |
| 20:52.38 | Nuxi | but (wave1 + wave2) /2 is faster and less float error. |
| 20:52.51 | irv999 | Any polycom experts herE? |
| 20:52.54 | goldenolden | it's not... and DTMF is fine, just no audio |
| 20:53.10 | jontow | set: dtmfmode=inband |
| 20:53.18 | jontow | then 'sip reload' and tell me what happens.. im just curious |
| 20:53.28 | jontow | odd coincidence is that it fixed my problems many many months ago |
| 20:53.31 | jontow | (same problem..) |
| 20:53.34 | goldenolden | k |
| 20:53.36 | goldenolden | one sec thanks |
| 20:53.46 | jalsot | Nuxi: hmm, so is it worth to use dft or not? |
| 20:53.52 | jontow | btw, thats ona per-phone basis |
| 20:53.57 | jontow | not global |
| 20:54.09 | Nuxi | Not unless you are doing something more than averaging. |
| 20:54.32 | *** join/#asterisk DougWare (~admin@198.67.41.187) |
| 20:54.45 | goldenolden | notta :( |
| 20:54.47 | jalsot | Nuxi: do you know how sox does it's job? |
| 20:54.52 | jontow | same stuff huh.. |
| 20:54.55 | Nuxi | nope. |
| 20:55.17 | goldenolden | yep |
| 20:55.26 | jontow | i have: canreinvite=yes, type=friend, host=dynamic, nat=no, dtmf=inband |
| 20:55.34 | jontow | shit, dtmf=inband, not dtmfmode=inband (my bad) |
| 20:55.38 | jalsot | Nuxi: thx for suggestions! |
| 20:55.40 | jontow | thats the config for my 7960 |
| 20:56.01 | Nuxi | Oh, wait. you can't average them... |
| 20:56.23 | jontow | waaait a minute; i have no idea.. i musta been on the crack |
| 20:56.26 | jontow | i see both in my sip.conf |
| 20:56.41 | Nuxi | In the case where person A is silent, person B will be quiter than he should be. |
| 20:56.51 | Nuxi | Hmmmm.... |
| 20:57.01 | DougWare | Sorry to but in, but I need to know where to find the latest copy of the jitterbuf-rtp-sip patch for the HEAD version. The one I have found is dated for the 4th of May, and it doesn't apply correctly to the current head version, does anyone have or know where I can get the newer one from? |
| 20:57.35 | Nuxi | you get that jalsot? |
| 20:57.45 | jalsot | Nuxi: yep :( |
| 20:58.02 | jalsot | Nuxi: I don't know how to step further :( |
| 20:58.07 | cluecon | have fun guys. I'm gonna go see what the stupid idiot did to my car! |
| 20:59.00 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 20:59.06 | jontow | hmm.. anthm, having the same problems i had with HEAD long ago -- channel_pvt.h isn't there and it whines about a lack of a 'callerid' member in a structure |
| 20:59.10 | *** join/#asterisk jterrero (~jterrero@mcse-irc.isys-networks.com) |
| 20:59.30 | anthm | this is the one from sf.net? |
| 20:59.47 | jontow | yep, 1.0.0 |
| 20:59.48 | jontow | and CVS HEAd |
| 20:59.50 | jontow | (D) |
| 21:00.20 | *** join/#asterisk jellytot (~terry@196.38.236.210) |
| 21:00.33 | jellytot | hello guys and girls |
| 21:00.33 | Nuxi | jaslot. In the real world, if you have two people talking you have two people talking. You don't average you just add. Boy I'm silly some times. |
| 21:00.55 | jellytot | i am wondering if i could get some help here |
| 21:00.55 | bewest | anyone have an idea of how many asterisk servers are running/serving phones? |
| 21:01.12 | jalsot | Nuxi: yep, underestood, half duplex ;) |
| 21:01.21 | irv999 | when I change an extension, do I need to reboot the system? |
| 21:01.31 | jontow | chan->callerid is mentioned .. 4 times in the file from what i can see; 2 are commented out with the word "BAD" above them ;) |
| 21:01.40 | jontow | i have a feeling it fails because i lack channel_pvt.h :) |
| 21:02.21 | terrapen | i wish file.c wasn't so bloody confusing |
| 21:03.03 | jontow | http://www.voip-info.org/wiki-Asterisk+fax <-- toward the bottom, if you search for channel_pvt.h gives an explanation (brief) |
| 21:03.10 | jontow | bug #3573 |
| 21:03.25 | *** join/#asterisk openfly (~openfly@howard.the.duck.can.eviscerate.us) |
| 21:03.30 | openfly | so uhm... |
| 21:03.30 | jalsot | hmm, why is res_monitor not made [optionaly?] to make 1 mixed stream if mostly everybody is mixing those 2 call legs and than making some encoding? |
| 21:03.36 | openfly | anyone got any spare cisco power cubes? |
| 21:03.43 | openfly | like uhm... four of em? |
| 21:04.06 | Assid | umm.. |
| 21:04.13 | jellytot | i have been running BRIstuff RC8 for a while now, using 2 HFC-s cards. today i tried to install a third card and seem to get a error |
| 21:04.19 | jalsot | the way we have now consumes more disk usage and also more capacity - is there a reason for that? |
| 21:04.25 | Assid | how many consecutive lines can asterisk hold? |
| 21:04.27 | anthm | you are getting the code from sf cvs server ? |
| 21:04.48 | jontow | no; the tarball.. i'll grab the cvs copy now |
| 21:04.53 | Assid | like suppose i want everyone to authenticate.. and then maybe have 300 people having a live feed ( conference) is it possible? |
| 21:04.55 | jontow | i see that the tarball is a little dated :) |
| 21:04.59 | jellytot | once asterisk starts i get a zaphfc lost sync, pci performance too low |
| 21:05.04 | Assid | basically they should only be able to hear.. |
| 21:05.06 | jellytot | has anybody see this before |
| 21:05.07 | Assid | not speak |
| 21:05.40 | DougWare | Does anyone know where I can find the CVS tar for 3-31-05? |
| 21:06.06 | wasim | DougWare: i think gnuinter.net has nightly tars |
| 21:06.13 | DougWare | Thanks |
| 21:06.28 | wasim | not sure though, he might have stopped |
| 21:06.39 | Juggie | you can request a date from the cvs server |
| 21:06.45 | DougWare | I will check, right now it is my bet |
| 21:06.47 | Juggie | any snapshot in time that you want |
| 21:06.53 | wasim | or what juggie said |
| 21:07.00 | Juggie | now, how you do that, i forget but i know it can be done :) |
| 21:07.03 | alegh | exit |
| 21:07.09 | DougWare | Ok, that might be the better way, I will try that, I will check the man page for cvs |
| 21:07.18 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
| 21:07.36 | wasim | -d cvs checkout -D 2005-05-11 -r v1-0 zaptel libpri asterisk |
| 21:07.45 | DougWare | Thank you |
| 21:08.19 | wasim | just an example, skip -r if you want HEAD, and change the date |
| 21:08.49 | *** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 21:09.26 | jontow | score, CVS HEAD for ICD compiled :) |
| 21:09.32 | *** join/#asterisk logicalonline (~logicalon@border.logicalonline.com) |
| 21:09.48 | *** join/#asterisk [hC] (~hardcore@8.10.2.4) |
| 21:10.10 | [hC] | What is the highest-density card available for terminating PRI into asterisk? (like, number of ports per card) |
| 21:10.24 | Juggie | T1/E1 * 4 |
| 21:10.31 | [hC] | Digium make one? |
| 21:10.32 | Juggie | DS3 comming soon |
| 21:10.33 | Juggie | yes |
| 21:10.38 | Juggie | www.digium.com |
| 21:10.42 | Juggie | go look |
| 21:10.48 | [hC] | Cool, I'll go look. I have the 1 port model right now. |
| 21:11.03 | Juggie | ohhh... digium has a new website |
| 21:11.05 | *** part/#asterisk marten_ (~sm7syx@212-162-184-20.skbbip.com) |
| 21:11.24 | Juggie | http://www.digium.com/index.php?menu=news/ds3000p |
| 21:11.51 | Juggie | 644 lines should be plenty :) |
| 21:12.06 | RoyK | it won't work |
| 21:12.22 | Juggie | what wont work |
| 21:12.52 | RoyK | with the interrupt rate of those cards, you'll need an octo opteron or something to run them |
| 21:13.19 | *** join/#asterisk stkn (~stkn@stkn.developer.gentoo) |
| 21:13.34 | RoyK | better stick with small 1U boxes, lots of them, with sangoma A104 cards |
| 21:13.40 | Juggie | well i'm sure if the board exists that it works tho it may not be perfect |
| 21:13.47 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 21:13.50 | Juggie | is the A104 a 1/2 height card? |
| 21:14.16 | claude2005 | Any one can help, I am still having a problem when an external caller calls and drops the call the internal phone keeps ringing regardless the external caller has hung-up |
| 21:14.16 | RoyK | I think it can be fit in low profile |
| 21:14.24 | [hC] | sangoma can fit in 1u? |
| 21:14.45 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 21:14.46 | RoyK | [hC]: why not? |
| 21:14.55 | [hC] | Im not disagreeing :) |
| 21:14.55 | RoyK | [hC]: any pci card can fit into 1u |
| 21:14.59 | drumkilla | RoyK: you don't knwo what you are talking about |
| 21:15.05 | Juggie | not so |
| 21:15.07 | drumkilla | the interrupt rate will be no different than any other card. |
| 21:15.09 | RoyK | drumkilla: no? |
| 21:15.10 | Juggie | only 1/2 height cads |
| 21:15.31 | [hC] | yeah, why would someone buy the sangoma over the digium? does it require less cpu to run it or something? |
| 21:15.33 | blitzrage | Juggie: correct |
| 21:15.45 | blitzrage | [hC]: not that I'm aware of |
| 21:15.46 | jsharp | Just a metric buttload of data transferred per interrupt. |
| 21:15.55 | Juggie | blitzrage, thanks i know i'm right :) |
| 21:15.59 | blitzrage | Juggie: :D |
| 21:16.05 | blitzrage | Juggie: yah, you seem like you know stuff |
| 21:16.08 | blitzrage | :D |
| 21:16.23 | RoyK | drumkilla: I'm just disturbed by the digium hardware quality. they suck. I've had to cold restart, as in unplug, wait a minute, re-pulg, my servers several times just because of driver failure |
| 21:16.33 | *** part/#asterisk logicalonline (~logicalon@border.logicalonline.com) |
| 21:17.05 | RoyK | drumkilla: if the driver isn't correctly unloaded, the te410p may hang until it loses power |
| 21:17.12 | RoyK | which is bad |
| 21:17.49 | Ahewes | Anyone using a TA750? |
| 21:18.09 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 21:18.17 | RoyK | wtf is ta750? |
| 21:18.23 | robin_sz | oh great. this grnadstream is now taking 5 minutes to display its confgiuation pages. wonderful. |
| 21:18.24 | drumkilla | adtran channel bank |
| 21:18.55 | *** join/#asterisk Nukemizer (~Nuke@67.137.28.165) |
| 21:19.04 | blitzrage | RoyK: you clearly know nothing if you don't know what a TA750 is! |
| 21:19.12 | blitzrage | Ahewes: they rock |
| 21:19.13 | *** join/#asterisk doolph (doolph@201.226.146.178) |
| 21:19.33 | RoyK | blitzrage: please, I don't do anything but PRI/E1 |
| 21:19.48 | blitzrage | RoyK: even more reason to know what a TA750 is |
| 21:19.49 | RoyK | blitzrage: but I do know how that works |
| 21:20.04 | robin_sz | arse arse arse. how can a single page of html take so long on a 100mbs network? |
| 21:20.24 | bewest | robin_sz, take the unit and throw it out the window |
| 21:20.28 | bewest | robin_sz, problem solved |
| 21:20.39 | RoyK | blitzrage: do you feal a good reason to pick on people that don't know the same things that you do, sir? |
| 21:20.41 | robin_sz | bewest: nah, target practice |
| 21:20.48 | bewest | :-> |
| 21:20.54 | bkw_ | RoyK did you have a question? |
| 21:20.59 | bkw_ | or comment |
| 21:21.01 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
| 21:21.13 | robin_sz | bewest: the sad thing is I was really happy to get a real hardware phone .. now, I'm just disappointed. |
| 21:21.13 | RoyK | bkw_: when? earlier today? |
| 21:21.20 | bkw_ | ya |
| 21:21.21 | bkw_ | I missed it |
| 21:21.23 | bkw_ | and you were gone |
| 21:21.37 | RoyK | bkw_: I was just amused by that patent email you sent |
| 21:21.43 | RoyK | bkw_: fucking crazy |
| 21:21.46 | bewest | robin_sz, I have a budgetone 100... it's been ok for me so far |
| 21:21.50 | bkw_ | did you read the patents |
| 21:21.54 | bkw_ | its entertainment to the max |
| 21:22.00 | RoyK | hehe :) |
| 21:22.04 | bewest | robin_sz, my needs are limited |
| 21:22.10 | bewest | bkw_, which thread was that? |
| 21:22.18 | bkw_ | mime multipart/mixed |
| 21:22.21 | bkw_ | violates that stupid patent |
| 21:22.23 | RoyK | bewest: patent lawyer |
| 21:22.35 | robin_sz | bewest: my basic need is "works" .. so far, not acheived |
| 21:22.45 | RoyK | bkw_: did you see my reply? |
| 21:22.48 | bkw_ | I was reading them and going "Ok you have got to be fucking kidding me" the whole time I was reading them |
| 21:22.51 | bkw_ | ya |
| 21:23.09 | bkw_ | I was like lets be a bit more general here why don't we |
| 21:23.31 | robin_sz | was this patents on concepts like "pressing buttons to dial" ?? |
| 21:23.35 | bkw_ | ok |
| 21:23.40 | bkw_ | sex causes O2 consumption to go up |
| 21:23.45 | bkw_ | go for it |
| 21:23.54 | irv999 | Has anyone had any experience with brands of DID's? and how to do DID's over the internet with NAT on both ends? |
| 21:23.59 | bkw_ | Can't forget you must license O2 from me |
| 21:24.05 | bkw_ | so everyone stop breating |
| 21:24.07 | RoyK | ~sex? |
| 21:24.08 | 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/ |
| 21:24.08 | bkw_ | er breathing |
| 21:24.09 | Katty | what's this about sex? |
| 21:24.20 | bkw_ | "man sex" |
| 21:24.29 | bkw_ | haha |
| 21:24.33 | Katty | what's this about man sex? |
| 21:24.43 | bkw_ | the sex man page |
| 21:24.45 | cluecon[file] | bkw_: get back to work, slacker! |
| 21:24.49 | irv999 | man sex.. I hear some women like that |
| 21:24.51 | Katty | oh, that thing. |
| 21:24.59 | bkw_ | I like just man sex |
| 21:25.02 | bkw_ | but thats me |
| 21:25.13 | Katty | you'd be that way |
| 21:25.29 | bkw_ | ain't nuttin wrong with a little gay porn! |
| 21:25.32 | bkw_ | haha |
| 21:26.07 | Nivex | BOOBIES! |
| 21:26.10 | Nivex | sorry, had to be done. |
| 21:26.12 | irv999 | bkw: as long as I am not in it.. I don't mind |
| 21:26.15 | irv999 | hahaha |
| 21:26.25 | irv999 | hand sized or huge boobs? |
| 21:26.50 | MikeJ[Laptop] | bkw, who was the company? |
| 21:26.50 | bkw_ | hahaha |
| 21:26.55 | bkw_ | for? |
| 21:27.06 | MikeJ[Laptop] | who has the patents |
| 21:27.14 | bkw_ | go read them.. you'll notice the theme |
| 21:27.16 | bkw_ | its J2 |
| 21:27.35 | Katty | mmm, gay. |
| 21:27.38 | bkw_ | so far reading them has almost put me in the hospital from laffing so hard |
| 21:27.40 | Katty | i hope bi counts. |
| 21:27.45 | MikeJ[Laptop] | why don't we get a group of people and get a bunch of general patents that cover their entire business |
| 21:27.45 | Katty | or else i'd be all left out. sniffle. |
| 21:27.45 | Nethab | is it just a cease and desist letter or are they before a judge |
| 21:27.48 | bkw_ | Katty, it does |
| 21:27.58 | MikeJ[Laptop] | then we can do the you drop your suit first game |
| 21:28.00 | Katty | i see. (:>>>) |
| 21:28.29 | devel | i hesitate to ask a question somewhat VOIP related, but.... |
| 21:28.38 | bkw_ | "Get your suit on" by Missy Elliot |
| 21:28.56 | Nethab | like a schoolgirl |
| 21:28.57 | MikeJ[Laptop] | devel, good, then don't :) |
| 21:29.04 | devel | ha ha ha |
| 21:29.07 | devel | fair enough. |
| 21:29.16 | bkw_ | remember CLUECON is where its at |
| 21:29.47 | robin_sz | bkw_: while they are no doubt amusing, I suspect you wont laugh so hard when they are granted, and or they start suing people ... the patents systems, especially in the US is dumb. |
| 21:29.48 | bkw_ | i'll be there.. i'll dress like Cher did on her farewell tour.. |
| 21:30.11 | bkw_ | robin_sz, they are already granted |
| 21:30.23 | Nethab | there not still pending? |
| 21:30.28 | bkw_ | nope |
| 21:30.32 | robin_sz | bkw_: well, you are stuff then. even with prior art. |
| 21:30.38 | Nethab | we were sued by a company with patents still pending |
| 21:30.39 | robin_sz | stuffed |
| 21:30.52 | Nethab | for interlacing images |
| 21:30.57 | bkw_ | robin_sz, not really |
| 21:31.02 | bkw_ | if we have to show up with prior art |
| 21:31.07 | bkw_ | its quite easy at that point to get it thrown out |
| 21:31.10 | bkw_ | moving on |
| 21:31.23 | darkskiez | is native sip transfer to parking working in stable or just head or neither ? This issue is confusing me reading old mailing list posts. |
| 21:31.34 | robin_sz | ummm, the system in the US and the rest of the world for prior arts is different |
| 21:31.35 | bkw_ | try it |
| 21:31.46 | bkw_ | robin_sz, yep |
| 21:31.59 | bkw_ | :) <-- this violates a patent |
| 21:32.04 | bkw_ | that little smiley face |
| 21:32.17 | bkw_ | but that was just filed as an example of how lame our patent system is |
| 21:32.30 | robin_sz | yeah |
| 21:32.46 | Nethab | ok here's a question, why does an outgoing call to a [peer] use the peername but an incoming call from same peer use the username = name |
| 21:32.48 | robin_sz | its *slightly* saner in .eu .. but only slightly |
| 21:32.51 | darkskiez | bkw_: It seemed to semi-work but try to return the call to the old context with the extension s pri 1 and not the originating extension or anything. How is this meant to find the original parker ? |
| 21:33.10 | bkw_ | darkskiez, well the parking that is in asterisk right now I do NOT like |
| 21:33.16 | bkw_ | it doesn't work how parking SHOULD work |
| 21:33.33 | bkw_ | it works how lame ass old phone systems worked back in the day |
| 21:33.41 | bkw_ | this is why we have valetparking |
| 21:33.42 | bkw_ | it works |
| 21:33.43 | bkw_ | it works great |
| 21:33.47 | bkw_ | and its like a shelf |
| 21:33.51 | bkw_ | you pick where you park it |
| 21:33.54 | Nethab | we should patent the method of suing someone over a patent |
| 21:33.56 | bkw_ | it doesn't return |
| 21:34.03 | bkw_ | till they hangup or you go get it |
| 21:34.12 | bkw_ | what what what? |
| 21:34.13 | bkw_ | my dna? |
| 21:34.19 | Nethab | no no no TnA |
| 21:34.21 | darkskiez | I saw valet parking, what happens if you park over an existing parked thing ? |
| 21:34.28 | bkw_ | ya know what is scary.. you're not the first to say something like that |
| 21:34.43 | bkw_ | darkskiez, it will fail and you take the call back |
| 21:34.46 | bkw_ | on my 7960's it works great |
| 21:34.50 | bkw_ | you can even stack them |
| 21:35.01 | bkw_ | fifo or lifo them out of the stack |
| 21:35.08 | darkskiez | bkw_: do you have a sample call plan using that ? |
| 21:35.14 | bkw_ | if I recall you can even stack them |
| 21:35.19 | bkw_ | then dump them all into the same exten |
| 21:35.25 | bkw_ | like stack callers for a meetme |
| 21:35.33 | bkw_ | then dump them all at once |
| 21:35.36 | darkskiez | bkw_: Oooo nice. |
| 21:35.41 | bkw_ | darkskiez, I think the wiki has an example |
| 21:35.55 | darkskiez | bkw_: cheers, i'll re-investigate it |
| 21:36.53 | *** join/#asterisk mkrufky (~mike@68.160.103.76) |
| 21:36.58 | darkskiez | bkw_: voip-info ? |
| 21:37.03 | bkw_ | file[mac], WAKE UP |
| 21:37.07 | bkw_ | darkskiez, yes |
| 21:37.10 | bkw_ | or google fer it |
| 21:39.17 | Nethab | stacker 2 the only way to bulk up |
| 21:39.24 | MikeJ[Laptop] | j2 is the result of a marriage of a revolutionary patented suite of services and a financially strong and disciplined organization. |
| 21:39.32 | MikeJ[Laptop] | hehe |
| 21:39.34 | bkw_ | yep |
| 21:39.36 | MikeJ[Laptop] | wow... |
| 21:39.44 | Nethab | financially strong enough for a patent lawsuit |
| 21:39.50 | MikeJ[Laptop] | they are just pissed you sell their services cheaper |
| 21:39.51 | darkskiez | bkw_: lots of references to it on your site which seem to be dead. I fould the loligo one with the source, but every reference to it seems dead |
| 21:39.54 | MikeJ[Laptop] | what jerks |
| 21:40.06 | bkw_ | darkskiez, ya loligo has it |
| 21:40.21 | jsharp | Nethab: Probably not strong at all...looking to make money through lawsuits rather than selling a product. |
| 21:40.38 | [hC] | I am sending calls out over my PRI, but "zap show channels" doesnt seem to indicate that there is an outbound call in progress. neither does zttool. Is there another command i should be using to look at current active channels? |
| 21:40.56 | jsharp | show channels |
| 21:40.56 | bkw_ | the problem is they can go after anyone that sells services that could be unified messaging |
| 21:41.02 | bkw_ | voice, fax and more |
| 21:41.13 | *** part/#asterisk mkrufky (~mike@68.160.103.76) |
| 21:41.19 | jsharp | Lets see em go after vonage, then. |
| 21:41.21 | Nethab | patents can only be for certain methods |
| 21:41.37 | Nethab | make one small change in your method and your different |
| 21:41.59 | [hC] | jsharp: ok i see it there, but is there something that will show me used channels, like a count? of all 23 channels on my pri? |
| 21:42.14 | bkw_ | Nethab, you have to just violate one claim in the patent |
| 21:42.16 | bkw_ | not the whole thing |
| 21:42.22 | bkw_ | hell google violates one of them |
| 21:42.58 | bkw_ | document archival and searching stuffs |
| 21:42.59 | jsharp | show channels will show all active calls on your system, voip or TDM. You'll just have to count them there. |
| 21:43.01 | Nethab | Silicon Graphics was sued for violating an 'image compression' patent because we used interlaced images on our website |
| 21:43.27 | [hC] | jsharp: huh. so is zttool broken, or what? It looks like it has a place where it would show me which channels are active, but when i place a call, it doesnt show up. |
| 21:43.33 | robin_sz | patnets, in general, suck |
| 21:43.34 | Nethab | so we deinterlaced all our images and said STFU |
| 21:43.35 | bkw_ | Nethab, SGI also was the key person in making Hylafax too weren't they? |
| 21:43.53 | Nethab | hydrafax? |
| 21:43.59 | bkw_ | hylafax |
| 21:44.08 | bkw_ | sam leifler I think it was |
| 21:44.21 | irv999 | Has anyone had any experience with brands of DID's? and how to do DID's over the internet with NAT on both ends? |
| 21:44.23 | Nethab | we had an internal fax relay thing that interfaced with IRIX but we didn't call it hylafax |
| 21:44.29 | Nethab | i forget what it was called |
| 21:44.41 | Nethab | of course we made up names all the time |
| 21:44.53 | bkw_ | haha |
| 21:45.05 | Nethab | like what's an Altix |
| 21:45.09 | bkw_ | irv999, the key is to force the rtp port ranges |
| 21:45.14 | bkw_ | if you can control both ends good |
| 21:45.17 | bkw_ | if not you're SOL |
| 21:45.20 | irv999 | bkw: in the firewall? |
| 21:45.33 | bkw_ | double nat is easy if you understand the concepts |
| 21:45.40 | irv999 | bkw: I can't control the remote ports but I can do it on my end.. |
| 21:45.42 | *** join/#asterisk outofjungle (~outofjung@61.17.134.123) |
| 21:45.49 | bkw_ | too bad |
| 21:45.50 | bkw_ | so sad |
| 21:45.51 | bkw_ | you're SOL |
| 21:46.05 | Nethab | one way audio here you come |
| 21:46.08 | bkw_ | you "might" get it working |
| 21:46.12 | bkw_ | if you're lucky |
| 21:46.12 | irv999 | hrmm I do have a VPN device on my end.. |
| 21:46.18 | bkw_ | VPN it |
| 21:46.28 | irv999 | bkw: sonicwall.. but I need a SMALL vpn device to connect to the DID |
| 21:46.33 | irv999 | portable.. |
| 21:46.43 | irv999 | bkw: so it can be put in a pocket or bag |
| 21:46.52 | bkw_ | I love how the term "DID" is misused these days |
| 21:47.18 | bkw_ | its what it does.. not what it is. |
| 21:47.30 | irv999 | bkw: I did not mean that |
| 21:47.57 | robin_sz | bewest: hold it boy, I patented subtraction, and addition is justy a special case |
| 21:47.59 | Nethab | just call it your rate center |
| 21:48.18 | jhava | hello all, one question.... anybody knows how to transfer a SIP line if it is not connected (eg, by configuration file, or web app, but not from SIP phone) I'm using AMP also. |
| 21:48.31 | [hC] | is there a way to shut down a sip channel in use? |
| 21:48.33 | robin_sz | oh fsck. this GXP 2000 has shat itself. |
| 21:48.37 | [hC] | like destroy a call forcefully.. |
| 21:48.38 | Nethab | soft hangup |
| 21:48.40 | robin_sz | wont even boot :( |
| 21:48.50 | Nethab | cli> soft hangup <channel> |
| 21:48.55 | opus__ | jhava you can originate a call |
| 21:48.58 | bewest | looks like target practice is sooner than you thought |
| 21:49.13 | jhava | by transfer I meant forwarding.... |
| 21:49.36 | opus__ | explain |
| 21:49.37 | irv999 | bkw: I meant ATA |
| 21:49.57 | [hC] | I had two calls that for some reason still seem to be established |
| 21:50.02 | [hC] | yet they quite obviously are not. |
| 21:50.12 | Nethab | [hC]: soft hangup |
| 21:50.17 | [hC] | I did. |
| 21:50.19 | [hC] | Is that common? |
| 21:50.26 | irv999 | bkw: any suggestions on pocket VPn device to make the ATA portable |
| 21:51.13 | Nethab | not too common but it happens |
| 21:51.39 | jhava | opus__: I have AMP that lets a SIP phone user forwards calls anywhere with *79, but not all phones are connected, and need to configure transfer for those phones by other means |
| 21:51.40 | *** join/#asterisk mrkyr (~bviitanen@h24-207-80-55.cst.dccnet.com) |
| 21:51.46 | Nethab | don't they make an ATA with PPPoE built in |
| 21:52.12 | opus__ | not all other phones have a DTMF pad? I don't understand by what you mean 'not all phones are connected' |
| 21:53.04 | jhava | Other phones are in the internet, users are not connected, but I need to forward their calls to land lines anyway |
| 21:53.47 | jhava | without reconfiguring the Asterisk box, so I need to emulate what it is done by *79 by software, is this configuration saved anywhere ? |
| 21:54.08 | *** part/#asterisk clive- (~pirch@rrba-146-101-112.telkomadsl.co.za) |
| 21:54.10 | opus__ | you need an AGI script |
| 21:54.59 | opus__ | go into the extension.conf in your /etc/asterisk directory and look for '*79', see which AGI script it calls |
| 21:54.59 | irv999 | bkw: what do most other pepple do with their ATA's when they NAT |
| 21:55.01 | jhava | Ok, I need to check then AGI scripts that does that function, maybe in phpagi from AMPORTAL, that was helpful, thanks. |
| 21:55.06 | opus__ | should be a line AGI('yadayada') |
| 21:55.12 | opus__ | look for that script |
| 21:55.34 | jhava | thank you opus__ |
| 21:55.35 | opus__ | copy it, just modify it, then, where you extension is for outside users place that script at a higher prority |
| 21:56.14 | blitzrage | me too |
| 21:56.18 | blitzrage | I'm starving |
| 21:56.29 | Nethab | i am too dang it, thanks for reminding me |
| 21:56.30 | opus__ | taco bell |
| 21:56.41 | blitzrage | gross |
| 21:56.48 | blitzrage | I want real food, not liquid meat |
| 21:56.56 | *** join/#asterisk jeffik (~Jeff@69.158.21.177) |
| 21:57.04 | blitzrage | taco bell has lower grade meat than dog food |
| 21:57.07 | opus__ | word |
| 21:57.20 | Nethab | that's only because i buy premuim grade dog food |
| 21:57.25 | blitzrage | lol |
| 21:57.25 | opus__ | hahaaha |
| 21:58.07 | opus__ | i just get the bean burrito ssupreme and the cheese queadea |
| 21:58.08 | irv999 | opus: other than the voicemail.conf where else do I specify a voicemail setup? do I need to reboot asterisk for it to take effect? |
| 21:58.23 | Nethab | that's like serving your dog filet minion while you're eating refried beans |
| 21:58.36 | Nethab | voicemail reload |
| 21:58.41 | irv999 | ty |
| 21:58.50 | blitzrage | just voicemail and dialplan logic to get into the voicemail - should just be able to do reload app_voicemail.so |
| 21:58.57 | opus__ | irv999 - there is a Voicemail(u<EXTENNUM>) function you need as the last priority as an extension, but yeah, thats it. 'reload' yup |
| 21:59.11 | blitzrage | lol |
| 21:59.31 | opus__ | that will be one pound of dog food |
| 21:59.51 | cluecon[file] | blitzrage! |
| 22:00.49 | irv999 | opus: what is the voicemauil(u191)? where is that run? in asterisk? |
| 22:01.19 | irv999 | or a command in the normal unix prompt |
| 22:01.58 | *** join/#asterisk manox (~boanthrap@pool-141-157-189-99.bos.east.verizon.net) |
| 22:02.51 | irv999 | isnt voicemail(u191) in the dialplan? I just want to check the voicemail by dialing in |
| 22:03.00 | blitzrage | irv999: yes, dialplan |
| 22:03.31 | irv999 | blitz: is that necessary to have to check login information?. |
| 22:03.45 | blitzrage | irv999: huh? |
| 22:03.58 | blitzrage | irv999: you need VoicemailMain() |
| 22:04.04 | blitzrage | if you want to login to voicemail |
| 22:04.23 | irv999 | blitz: Before I assign it to an extension on timout.. can I just enter the information into voicemail.conf and login? or do I need to do anything else |
| 22:04.38 | irv999 | sorrt I am such a dumbass.. I am trying to plug through this on my own.. my programmer is strining my a long |
| 22:05.52 | Nethab | what are you trying to do |
| 22:05.58 | blitzrage | irv999: you setup the voicemail user in the default voicemail context (voicemail.conf), then use the VoicemailMain() application in the dialplan (extensions.conf) to login. If you supply no information to the application, it will ask you for the mailbox, then the pin |
| 22:06.08 | Nethab | login to voicemail to check messages, or leave a voicemail for a person |
| 22:07.06 | blitzrage | irv999: be sure to do a 'show application voicemail' and 'show application voicemailmain' from the Asterisk CLI to learn the difference |
| 22:07.48 | irv999 | ok checking |
| 22:08.10 | irv999 | blitz: thanks |
| 22:08.17 | *** part/#asterisk manox (~boanthrap@pool-141-157-189-99.bos.east.verizon.net) |
| 22:08.24 | JerJer | you know it is a very good day when the telco asks you how much they should bill you for a circuit |
| 22:08.49 | blitzrage | :D |
| 22:09.11 | mogorman | anyone use realtime sip here? |
| 22:10.52 | JerJer | i hope not |
| 22:10.58 | Nethab | what the heck, they asked you? |
| 22:11.08 | JerJer | yep |
| 22:11.25 | Nethab | is because of pro-rating? |
| 22:11.32 | JerJer | nope |
| 22:11.32 | Nethab | or they forgot what there prices were |
| 22:11.44 | JerJer | we do most of our own transport |
| 22:11.55 | opus__ | mogmorman -- i am about to use it, what are you useing it for? |
| 22:12.10 | JerJer | opus__: don't safe yourself some hair |
| 22:12.24 | JerJer | dont...save yourself some hair |
| 22:12.25 | darkskiez | bkw_: sorry to bug you about valetparking, but it doesnt seem to build on stable, missing parking.h for starters..? that sound right? |
| 22:13.12 | JerJer | don't run stable |
| 22:14.00 | darkskiez | oh, parking.h isnt on head either ? weird. |
| 22:14.06 | darkskiez | JerJer: why not ? |
| 22:14.35 | JerJer | parking.h has been killed |
| 22:14.38 | darkskiez | JerJer: i got slated for days here when I was talking about running head on a production server. |
| 22:14.39 | JerJer | its now features |
| 22:14.59 | bkw_ | I don't think it will build on stable |
| 22:14.59 | JerJer | darkskiez: cvs head is far superior code than -stable |
| 22:15.08 | bkw_ | we target for cvs-head |
| 22:15.15 | bkw_ | as JerJer said it is far superior |
| 22:15.15 | *** join/#asterisk robbins (~robbins@samson1.500.megarouting.com) |
| 22:15.17 | JerJer | just because there hasn't been a release in quite a while |
| 22:15.40 | bkw_ | it will be a "release" next time right? |
| 22:15.45 | bkw_ | not "stable" |
| 22:15.45 | bkw_ | haha |
| 22:15.50 | JerJer | fuck i hope so |
| 22:16.01 | JerJer | cuz -stable is not 'stable' |
| 22:16.15 | robbins | hey folks, anyone know of recent issues of using IAX in queues.conf... it was working, now the member IAX2 statement is coming up as invalid? |
| 22:16.20 | darkskiez | hmmm |
| 22:16.22 | bkw_ | haha |
| 22:16.24 | bkw_ | it sure isn't |
| 22:16.37 | bkw_ | cvs-head is closer to "stable" than -stable |
| 22:16.45 | JerJer | yeppers |
| 22:16.57 | bkw_ | I just updated a few boxes here with it |
| 22:17.00 | bkw_ | got 3 more to update |
| 22:18.05 | bkw_ | get ur freak on |
| 22:18.09 | robbins | bkw, any code changes that you are aware of that disabled IAX connects in Queues? |
| 22:18.16 | bkw_ | no clue |
| 22:18.30 | bkw_ | I know we have to remove a few things from chan_agent and add a few things to make it NOT deadlock |
| 22:18.38 | bkw_ | otherwise its chan_deadlock |
| 22:18.48 | JerJer | app_noqueue |
| 22:18.54 | robbins | yeah.... |
| 22:18.56 | bkw_ | what we remove would never get added to cvs either |
| 22:19.04 | bkw_ | and what we add would never fly either |
| 22:19.26 | JerJer | so run a patch repository |
| 22:19.27 | cluecon[file] | but it darn well works |
| 22:19.30 | bkw_ | ok its like this.. you can deadlock once ever 5 reloads.. or deadlock every 100+ |
| 22:19.34 | bkw_ | JerJer, soon ;) |
| 22:19.40 | opus__ | jerjer whats broken in realtime sip? |
| 22:19.55 | bkw_ | other than when zaptel kernel panics for no good reason |
| 22:20.24 | JerJer | realtime is just broken |
| 22:20.27 | JerJer | period |
| 22:20.32 | bkw_ | yes flawed |
| 22:20.33 | opus__ | why? |
| 22:20.36 | bkw_ | realtime perl is nice |
| 22:20.40 | bkw_ | you can work around those silly flaws |
| 22:20.44 | bkw_ | ok I gotta jet |
| 22:20.45 | bkw_ | bbl |
| 22:20.49 | opus__ | thanks |
| 22:20.59 | Nethab | because the underlying assumption that asterisk is thread safe, and deadlock free was flawed |
| 22:21.54 | Nethab | the whole of asterisk needs to be abstracted to be completely asynchronous, because other than zap, it's all message based, which can come and go at any time |
| 22:22.17 | *** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 22:22.24 | Ariel_ | hello everyone. |
| 22:23.23 | JerJer | moo |
| 22:23.27 | Nethab | hello |
| 22:23.56 | Nukemizer | I am searching for a way to determine why I can play mp3's in /var/lib/asterisk/mohmp3 but I can not ply mp3's that I have created. the proccess does work on one box for me, but not my second one. I can not find any error messags |
| 22:24.13 | JerJer | Nethab: code up a small example and post it on the bug tracker - voice your opinion |
| 22:24.22 | *** join/#asterisk bjohnson (~bjohnson@66.11.188.215) |
| 22:25.14 | robbins | <PROTECTED> |
| 22:25.27 | Nethab | the fact that the manager functions are a part of the channel driver is completely flawed and can not scale |
| 22:25.33 | robbins | is that not a proper format for a member entry in queues.conf |
| 22:25.40 | robbins | ? |
| 22:25.41 | Nethab | JerJer: cause it would be a rewrite of * not a patch |
| 22:25.58 | JerJer | and the problem is? |
| 22:25.58 | opus__ | is there a log of whats new in cvs head? |
| 22:26.05 | opus__ | duh, changelog |
| 22:26.25 | Nethab | don't go by changelog check the cvs lists |
| 22:28.47 | JerJer | htt://lists.digium.com |
| 22:28.51 | JerJer | +p |
| 22:30.22 | *** join/#asterisk Marlow (~marlow@159-134-144-183.as1.mvw.galway.eircom.net) |
| 22:30.35 | *** join/#asterisk goatmilk (~goatmilk@user-24-214-24-45.knology.net) |
| 22:31.16 | *** join/#asterisk SimonR (~SimonR@HSE-Toronto-ppp3488582.sympatico.ca) |
| 22:31.36 | Nethab | i prefer my trout beating live, because they're flopping will only increase the beating |
| 22:32.19 | goatmilk | Using a dead trout allows me to vent more frustration since I am hitting harder |
| 22:32.38 | goatmilk | i assure you the beating is not diminished |
| 22:32.54 | goatmilk | i'll send that punk ass to his momma cryin' like a school sissy |
| 22:33.16 | Nethab | why do you have to be in school to be a sissy |
| 22:33.25 | Nethab | mommas boys usually stay at home |
| 22:34.31 | goatmilk | Nethab: stop asking questions or you'll get the trout next! |
| 22:34.38 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 22:34.39 | opus__ | wc -l `find . '.c'` |
| 22:34.42 | opus__ | <PROTECTED> |
| 22:34.52 | Nethab | *.c? |
| 22:34.54 | goatmilk | shit yeah!! |
| 22:35.05 | Nethab | find . -name '*.c' |
| 22:35.08 | goatmilk | *.c* (non-regex) |
| 22:35.25 | drumkilla | what about .h! |
| 22:35.40 | Nethab | c and h pure cane sugar |
| 22:35.46 | MikeJ[Laptop] | drumkilla, why you always thinking of the .h? |
| 22:35.50 | MikeJ[Laptop] | wassup with that? |
| 22:36.48 | opus__ | <PROTECTED> |
| 22:36.50 | darkskiez | arse:# rmmod wcte11xp |
| 22:36.50 | darkskiez | Connection to 194.xxx.xxx.xxx closed. |
| 22:37.00 | opus__ | whats that about |
| 22:38.09 | citats | opus__: final message from the most intelligent creatures on the planet |
| 22:38.20 | citats | well not the most, i think 2nd most |
| 22:38.27 | darkskiez | the asterisk devs ? |
| 22:38.38 | citats | the dolphins |
| 22:38.39 | MikeJ[Laptop] | what are the asterisk devs? |
| 22:38.49 | darkskiez | the most intelligent creatures on the planet :) |
| 22:39.05 | MikeJ[Laptop] | heh... who's butt are you trying to kiss |
| 22:39.16 | darkskiez | timecops |
| 22:39.56 | darkskiez | Oh, he's lurking. |
| 22:40.12 | darkskiez | anyone else miss the fun the other day? |
| 22:40.37 | MikeJ[Laptop] | what fun? |
| 22:40.53 | darkskiez | he threw a wobbler when a bug got introduced to cvs head. |
| 22:41.19 | MikeJ[Laptop] | which bug? |
| 22:42.37 | darkskiez | http://bugs.digium.com/view.php?id=4386 |
| 22:43.12 | *** join/#asterisk jdg (~jdg@CA03F3D6.adsl.mana.pf) |
| 22:44.03 | MikeJ[Laptop] | well.. that is why we need more people reviewing stuff before it gets commited... |
| 22:45.22 | Nethab | has Kris Fleming started yet? |
| 22:45.47 | cypromis | mr. F |
| 22:45.49 | darkskiez | Nethab: he checked in that bug, so i guess so ;) |
| 22:46.03 | blitzrage | isn't it Kevin Fleming? :) |
| 22:46.06 | darkskiez | oh |
| 22:46.13 | darkskiez | i've not a scoob. |
| 22:49.23 | *** join/#asterisk guugmember (~guugmembe@200.6.219.35) |
| 22:49.35 | darkskiez | rmmod wcte11xp - that shouldnt crash a machine, should it ? |
| 22:49.45 | *** join/#asterisk CoolAcid (~jk@216.99.98.39) |
| 22:51.17 | Nethab | What are your opinions about seperating out everything but the protocol specific stuff from the channel drivers, ie. the management stuff, channel tracking, user/peer definitions, etc |
| 22:51.44 | guugmember | who knows any 12fxo and 12fxs channel bank that works good with asterisk? |
| 22:52.00 | Nethab | guugmember: sip gateways? |
| 22:52.21 | Nethab | or like PCI cards |
| 22:53.14 | guugmember | Nethab, channel banks like the rhino ones |
| 22:53.39 | jsolares | a pci card with that many fxs or fxo ports would work as well i guess |
| 22:53.52 | darkskiez | 3xtdm400 cards ? |
| 22:54.05 | guugmember | what about the irqs in the machine? |
| 22:54.05 | jsolares | the same pc would then need 6 cards |
| 22:54.23 | darkskiez | be cheaper than a te4xxp and a channel bank tho |
| 22:54.51 | guugmember | darkskiez, that is true |
| 22:54.52 | darkskiez | guugmember: not so sure about the irq demands from a tdm card versus a te card |
| 22:54.56 | jsolares | i dont know of any board that can accept 6 cards living happily |
| 22:55.34 | guugmember | jsolares, that is why I want to go for the channel bank, but cant find any 12fxo and 12fxs on |
| 22:55.35 | guugmember | one |
| 22:55.43 | guugmember | s/on/one |
| 22:55.51 | *** join/#asterisk ldav15 (~ldavis@208.2.145.2) |
| 22:57.01 | guugmember | http://www.channelbanks.com/ |
| 22:57.18 | ldav15 | anyone have time to help diagnose a realtime setup problem? |
| 23:00.02 | blitzrage | ldav15: just ask - someone will help if they can |
| 23:00.12 | darkskiez | guugmember: that rhino one says its modular and can be changed on a 4port basis |
| 23:03.10 | ldav15 | I've setup odbc,postgres and am trying to get asterisk to allow a sip phone to register. I can't get anymore info out of asterisk other than, "Registration from <SIP:404@10.1.7.102> failed for 10.1.3.29 |
| 23:04.24 | ldav15 | does anyone know of a good step by step setup other than voip-info.org or is that my best bet? |
| 23:10.20 | *** join/#asterisk robin_sz (~robin@adsl.redpoint.org.uk) |
| 23:10.45 | *** join/#asterisk outsidefactor (~blah@203-206-241-250.dyn.iinet.net.au) |
| 23:10.46 | robin_sz | ok, one more dumb ass question ... howcome I have no dialtone on this extension? |
| 23:11.20 | *** part/#asterisk guugmember (~guugmembe@200.6.219.35) |
| 23:12.39 | *** join/#asterisk ZOP (NachoGod@ZOP.sustaining.supporter.pdpc) |
| 23:12.47 | ZOP | hola |
| 23:13.04 | *** part/#asterisk outsidefactor (~blah@203-206-241-250.dyn.iinet.net.au) |
| 23:13.21 | ZOP | drumkilla: i wont get a chance to upgrade asterisk here until tonight but i'll be sure to give you any feedback after i do. |
| 23:13.43 | ZOP | drumkilla: it might have to be pushed off to friday though if things don't go well with my rebuilding of my personal firewall stuff heh. |
| 23:14.01 | ZOP | (re 1.0.8 rc) |
| 23:14.56 | claude2005 | Any one knows why when external caller hangup the internal phone keeps ringing |
| 23:17.05 | *** join/#asterisk stkn (nobody@stkn.developer.gentoo) |
| 23:19.14 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 23:25.29 | JunK-Y | moooo |
| 23:29.41 | robin_sz | claude2005: yes, on an FXO card? |
| 23:30.54 | robin_sz | claude2005: US and some other systems revers the line voltage or some such on hangup, which is detected by the FXO, most euro systems do not do this and the card does not see the hangup |
| 23:33.53 | *** part/#asterisk ldav15 (~ldavis@208.2.145.2) |
| 23:34.36 | Katty | Ariel_: way to go away to dinner right when i turn the webcam on |
| 23:34.45 | *** join/#asterisk SimonR (~SimonR@67.71.52.223) |
| 23:38.14 | Marlow | Katty : that's how it goes sometimes |
| 23:39.07 | Katty | indeed |
| 23:39.30 | Marlow | robin_sz : some EU switches even do weirder stuff with the FXO .. |
| 23:40.11 | Marlow | robin_sz : like I moved to a new place and i would get busy when i hook the X101P in .. allways .. |
| 23:40.26 | Marlow | robin_sz : if i took a cable with crossed wires, I would get a line fault .. |
| 23:40.30 | robin_sz | Marlow: yeah ... and UK CID is another mess as well |
| 23:40.48 | Marlow | robin_sz : this behavior stayed even if the pabx was off ... not powered .. |
| 23:40.55 | robin_sz | heh |
| 23:40.56 | Marlow | robin_sz : UK CID is one ... |
| 23:41.13 | *** join/#asterisk dial_911_now (~niggerplz@66-55-197-254.gwi.net) |
| 23:41.29 | Marlow | robin_sz : and you know swedish, danish and dutch CID is completely different to anyone else |
| 23:41.33 | robin_sz | I presume there is no way for me to get UK CID into * via my FXO? |
| 23:42.01 | Marlow | robin_sz : wasn't there a patch for that ? or is it only for the nordic CID until now ? |
| 23:42.09 | robin_sz | not knowing |
| 23:42.22 | Marlow | robin_sz : search the bugs |
| 23:42.29 | Marlow | robin_sz : on CID or UK CID |
| 23:42.45 | robin_sz | shall I tell you about UK ISDN and cid, especially Home Highway ...and business highway? |
| 23:43.01 | Marlow | robin_sz : eh ? how can they fuck that up .. |
| 23:43.07 | robin_sz | youd be suprised |
| 23:43.18 | Marlow | robin_sz : DSS1 is one standard .. all over europe .. it's digital .. |
| 23:43.23 | robin_sz | on hoem highway, they send CID on the D channel ... |
| 23:43.30 | Marlow | robin_sz : even though Eircom does really weird stuff .. |
| 23:43.34 | *** join/#asterisk mikewho2 (Schnappi@ip68-105-227-82.lu.dl.cox.net) |
| 23:43.36 | mikewho2 | anyone arounmd? |
| 23:43.38 | Marlow | robin_sz : that's how it should be .. |
| 23:43.46 | robin_sz | then at the rx terminal .. strip it off and covert it to CID |
| 23:43.48 | Marlow | robin_sz : d-channel is for signalling |
| 23:43.58 | robin_sz | and removbe it from the customer side of the D channel |
| 23:44.08 | mikewho2 | anyone know why when i try to update my 7960 it gets the 7.4 sip goes the verifying load, then says 'protocol appliucation invalid' |
| 23:44.14 | mikewho2 | the 'settings' button doesnt even work |
| 23:44.16 | robin_sz | if you pay another 50 quid a month, they will not filter it |
| 23:44.16 | Marlow | robin_sz : why remove it ? |
| 23:44.31 | robin_sz | because its BT and they are bastards |
| 23:44.42 | Marlow | robin_sz : the usual is, that you can set any CID you want, of the MSNs assigned to you |
| 23:44.58 | Marlow | robin_sz : if you set one, that isn't assigned to you, CID defaults to your main number |
| 23:45.04 | robin_sz | right |
| 23:45.18 | Marlow | robin_sz : and if you want to be able to set anything, you'll have to pay |
| 23:45.27 | Marlow | robin_sz : that is european std. for isdn |
| 23:45.40 | *** join/#asterisk P-NuT (~dave@CPE-60-225-214-14.nsw.bigpond.net.au) |
| 23:45.44 | P-NuT | Hi all |
| 23:45.54 | robin_sz | this is on incoming calls, they wont let 'home' customers rx incoming cid ... except by the 'analogue' way on the line, not on the isdn ports. |
| 23:45.55 | Marlow | robin_sz : the one where you can set anything is "unscreened" |
| 23:46.14 | Marlow | robin_sz : WHAT ? that's pure bullcrap |
| 23:46.24 | robin_sz | yeah .. I know |
| 23:46.43 | robin_sz | but if you pay the extra 50 quid, they let it through |
| 23:46.50 | Marlow | robin_sz : now .. here is how eircom does it .. and it's not consitent throughout the country .. |
| 23:47.30 | Marlow | robin_sz : anywhere in the country, you will have to send your number without regional prefix, to show the right CID |
| 23:48.08 | Marlow | robin_sz : in Dublin you will however only receive the last 4 digits of your MSN when the call comes in (not CID, but MSN) |
| 23:48.34 | Marlow | robin_sz : that is so far ok, because most phones can figure that out .. they go from right to left and match |
| 23:49.15 | Marlow | robin_sz : in Galway, you will get a MSN send, that includes the "0"+regional prefix and full number |
| 23:49.21 | Marlow | robin_sz : now go figure on that .. |
| 23:49.47 | Marlow | robin_sz : hat to use a d-channel analyzer to figure out, what the f****** f*** they were up to |
| 23:49.49 | robin_sz | shrug .. its Eire .. what did you expect ;) |
| 23:50.03 | Marlow | robin_sz : not inconsistency within the same company .. |
| 23:50.10 | P-NuT | Hi all, I wonder if anyone can point me in the direction of an example to setup and extension that can access the server through 2 nat firewalls. |
| 23:50.17 | Marlow | robin_sz : but then again .. nothing can shock me anymore in this country |
| 23:50.31 | Marlow | P-NuT : use IAX :) |
| 23:50.32 | robin_sz | Marlow: my wife is from Dublin ... nothing suprises me anymore |
| 23:50.50 | Marlow | robin_sz : ehehe ..... the irish are odd .. but friendly .. |
| 23:51.22 | Marlow | robin_sz : and they know to brew good stout ... for the lager, that's us danes, that are unbeatable :) |
| 23:51.29 | robin_sz | Marlow: and from the wrong side of the Liffey too ;) |
| 23:51.51 | Marlow | robin_sz : what ? north side ? .. where did you grab her ? |
| 23:52.02 | robin_sz | from behind ;) |
| 23:52.08 | Marlow | robin_sz : ehehehe .. |
| 23:52.42 | Marlow | robin_sz : tjaeh ... once it's done, it's marriage .. no way around it with the irish girls .. |
| 23:54.01 | *** join/#asterisk cluecon[file] (~file@mctn1-4363.nb.aliant.net) |
| 23:54.01 | robin_sz | Marlow: sigh ... i fear you are right ... 20 years now and no sign of early release for good behaviour |
| 23:55.31 | Marlow | robin_sz : eh ... it can't be that bad ... |
| 23:56.25 | robin_sz | Marlow: well, shrug, maybe not *that* bad ;) |
| 23:56.33 | Marlow | robin_sz : anyhow ... i'll be heading of to bed ... |
| 23:56.40 | robin_sz | night ... |
| 23:56.42 | Marlow | robin_sz : it's getting late .. |
| 23:59.19 | Katty | Ariel_: webcam is off now, but last shot is still up there if you want to see it |
