00:03.32 | _570RM_ | yo, what is the Dial command to initiate a call out of a isdn4linux card.i can already receive calls, but cant make them |
00:06.08 | *** part/#asterisk santiago (~santiago@63.245.86.95) |
00:07.38 | *** join/#asterisk MavvieSJH (~root@barnetworks-link.syd.comindico.com.au) |
00:08.55 | MavvieSJH | only, after analyzing the HTTP packet, I wonder why they actually bother: |
00:09.12 | MavvieSJH | GET / HTTP/1.1..Host: 202.83.176.33:80 |
00:09.32 | MavvieSJH | whose idea was it to put an IP address for the hostname there? |
00:09.57 | srt | and including the port is funny, too ;) |
00:10.07 | firestrm | what is wrong with this dial command? exten =>1010,dial(IAX2/10@200.106.63.85,30) |
00:10.19 | MavvieSJH | but it made me understand why they needed four computers instead of one to roll out the callmanager system. |
00:10.34 | MavvieSJH | srt: hadn't even spotted that one yet. |
00:10.34 | firestrm | thas should work souldnt it? |
00:10.48 | MavvieSJH | firestrm: no priority |
00:11.18 | Juggie | exten => _401X,1,Dial(IAX2/asterisk2:asterisk@192.168.0.52/${EXTEN}@iax,${DIALTIME},r) |
00:11.38 | Juggie | compare :) |
00:11.56 | puppet | juggie: hia |
00:12.09 | puppet | juggie: i got basic webinterface up now ;D |
00:12.30 | puppet | juggie: all incoming call canbe redirected to queue,and two upphones |
00:12.32 | Juggie | patch any of the c stuff? |
00:12.34 | puppet | ipphones |
00:12.37 | *** join/#asterisk obelisque (~samifruit@70.48.19.123) |
00:12.46 | obelisque | hello! |
00:12.52 | obelisque | Hello guys! |
00:12.56 | obelisque | and GIRLS! |
00:12.57 | puppet | juggie: nah havent had to use that yet the apifunctions that are done are |
00:13.18 | YoYo | ~seen bkw |
00:13.20 | jbot | bkw <~bkw@u201.udal.afb.lu.se> was last seen on IRC in channel #debian, 20d 12m 23s ago, saying: 'uhm, the latter one or both of them?'. |
00:13.20 | YoYo | ~seen bkw_ |
00:13.21 | jbot | bkw_ <~brian@bkw.developer.and.friend.of.asterisk> was last seen on IRC in channel #asterisk, 15h 12m 2s ago, saying: 'Delmar,where?'. |
00:13.21 | puppet | get_channels,get_channel,redirect,start_monitor,stop_monitor |
00:13.25 | Juggie | puppet, spent some time talking to someone today, asterisk needs a whole new cli system |
00:13.42 | puppet | juggie: iiik, does that mean that i have to recode? |
00:13.55 | Juggie | one where the command which determines the output isnt responsible for formatting it |
00:14.06 | obelisque | Any good FREE iax2 client VIDEO for windows? |
00:14.55 | Juggie | puppet, write a list of commands that would be good to have for the web gui, and i'll look at patching them to support concise |
00:14.58 | srt | MavvieSJH: what firmware are you running on your cisco? |
00:15.00 | Juggie | i need the same stuff... |
00:15.27 | puppet | juggie: its really all commands :s |
00:15.45 | puppet | juggie: we want to get dialplan in better format we want sip peers we want.. yeah :/ |
00:15.48 | Juggie | puppet, you better start learning C then because i aint patching every command ;) |
00:15.58 | puppet | juggie: yeah i have to ;p |
00:16.10 | Juggie | i'll do what i consider important |
00:16.26 | MavvieSJH | srt: I think it's TERM70.6-0-2SR1-0s or 7970_64060118 |
00:16.44 | puppet | dialplan aint that important relative easy to just put the lines with Context at start in array on that way we get all arrays we can redirect people to |
00:16.48 | puppet | contexts* |
00:17.05 | puppet | juggie: wanne see my progress? |
00:17.06 | MavvieSJH | srt: at this moment I suspect a bad configuration in the call manager, but I can't login to it since I don't know which username for the webinterface (I know the password :-) |
00:17.19 | srt | *gg* |
00:17.38 | *** join/#asterisk Damin_Mobile (~pocketirc@79.sub-166-155-81.myvzw.com) |
00:17.42 | srt | my 7960 with at least doesn't put the port there |
00:17.57 | Damin_Mobile | Back in Cleveland... |
00:18.07 | Damin_Mobile | Just landed |
00:18.20 | Damin_Mobile | It is CcOLD! |
00:18.45 | *** join/#asterisk n3tar (~geno@201.254.93.202) |
00:18.47 | n3tar | hi |
00:19.41 | Damin_Mobile | #asterisk-bugs |
00:20.04 | brc_ | hi Damin_Mobile |
00:20.19 | srt | MavvieSJH: hmm at least not if i dont explicitly include it in the url ;) |
00:20.20 | firestrm | dial(IAX2/10@200.106.63.85,30) still doesnt work.. i have iax.conf sent up on the remote machine, any idea where its going wrong? |
00:20.42 | Damin_Mobile | brc: You up for doing some more bug hunting tonight? |
00:20.43 | firestrm | comes back busy, and nothing appears on the remote machine |
00:20.50 | brc_ | Damin_Mobile, sure |
00:21.08 | MavvieSJH | srj: I suspect it's a configuration issue on the callmanager, because the guys who installed it refused to use DNS ("it only slows everything down") |
00:21.09 | brc_ | Damin_Mobile, be a minute...on hold with united to reserve my cluecon ticket |
00:21.20 | obelisque | Do you guys know any good FREE iax2 VIDEO client for windows? |
00:21.20 | Damin_Mobile | brc: You have to go to Von in Boston |
00:21.28 | brc_ | I do? |
00:21.29 | brc_ | when |
00:21.30 | brc_ | where |
00:21.34 | brc_ | who's going |
00:21.40 | Damin_Mobile | von.com... |
00:21.45 | brc_ | why do people never tell me these things |
00:21.49 | brc_ | who's going? |
00:22.30 | Damin_Mobile | brc; I am still on the plane, so it will be much later tonight. byee for now... |
00:22.37 | brc_ | Damin_Mobile, hangon a sec |
00:22.49 | brc_ | Damin_Mobile, I've gotta book my ticket by 6pm |
00:23.20 | Juggie | internet on the plane? :) |
00:23.21 | Juggie | geek |
00:23.33 | Juggie | tho i bought a train ticket to toronto in april in first class |
00:23.41 | Juggie | so i could have internet on the train |
00:24.12 | pauldy | man this is a mess does anyone have any tips for broadvoice to asterisk to softphone |
00:24.21 | pauldy | I can dial extensions form the softphone no problem |
00:24.30 | Juggie | can you call out? |
00:24.34 | pauldy | full audio everything |
00:24.40 | pauldy | I can dial to the softphone form my cell |
00:24.42 | pauldy | full audio |
00:24.52 | pauldy | but when I dial from the softphone to my cell I can't hear anything |
00:24.58 | Juggie | does it connect? |
00:24.59 | pauldy | and the call timer on the softphone never starts up |
00:25.22 | Juggie | does your cell ring when u dial? |
00:25.22 | pauldy | so when I dial out from the softphone it rings my cell |
00:25.32 | pauldy | but I cannot hear anything when I pick up |
00:25.56 | Juggie | does the machine your sip phone is on have more then one ip? |
00:26.17 | pauldy | localhost and the dhcp address |
00:26.24 | pauldy | thats it |
00:26.45 | Juggie | what does the cli say? |
00:26.48 | *** join/#asterisk stustu (~stustu@fluffy.fatburen.org) |
00:27.08 | pauldy | crom the console |
00:27.16 | pauldy | it says pretty much what I would expect it to |
00:27.17 | Juggie | ? |
00:27.21 | pauldy | crom = from |
00:27.28 | Juggie | did you try enabeling sip debug |
00:27.30 | wildcard0 | hey. WAY offtopic. if anyone knows stuff about stacked LNB satellite stuff, please msg me :) |
00:27.30 | wildcard0 | thanx |
00:27.31 | stustu | What's the differnece between "Transmitting" and "Transmitting reliably" in the SIP debug log? |
00:27.36 | Juggie | to see if there was anything odd |
00:27.38 | pauldy | yea the call looks fine |
00:27.58 | Juggie | have u tried calling anything besides your cell? |
00:28.06 | Juggie | not that it should matter |
00:29.38 | pauldy | one thing wierd when I hang up i get Got SIP response 481 "Call/Transaction Does Not Exist" back |
00:30.35 | Juggie | check the wiki for broadvoice sounds like something is misconfigured |
00:30.37 | Juggie | but i am not sure what |
00:30.53 | opus___ | make sure your context is correct and your extensions.conf |
00:30.56 | pauldy | me either and the configs are all over the place most of them won't even allow incoming calls |
00:31.03 | doughecka | when passing caller id out |
00:31.07 | doughecka | do I include a 1? |
00:31.28 | doughecka | eh, guess so |
00:31.29 | doughecka | :) |
00:31.38 | MavvieSJH | doughecka : ,1,SetCallerId(1${CALLERIDNUM}) |
00:31.54 | doughecka | well, I think the proper way is without the 1 |
00:32.06 | pauldy | yea right now the dial context is default for the softphone |
00:32.07 | MavvieSJH | oh, read it wrong. |
00:32.10 | doughecka | cause calling 18005558355 rejected it while removing the 1 took it :) |
00:32.17 | doughecka | haha |
00:32.21 | Matt-E- | why, when i do an outside dial it rings once then nothing...? |
00:33.06 | stustu | Any experts on SIP authentication here? |
00:34.07 | pauldy | should I have to worry with anything like TRUNK= |
00:34.34 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
00:34.34 | *** mode/#asterisk [+o anthm] by ChanServ |
00:35.40 | pauldy | found the problem |
00:35.56 | firestrm | AgiNamu, still here? |
00:35.56 | pauldy | one o fthe tuts said to set pedantic=yes |
00:36.06 | pauldy | I disabled that and now everything is working |
00:36.06 | ManxPower | ~docs |
00:36.07 | 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 |
00:38.33 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
00:42.23 | *** join/#asterisk jeffik (~jeffik@m2b7c36d0.tmodns.net) |
00:45.04 | *** join/#asterisk miguellinux (~miguellin@200.47.223.190) |
00:45.11 | *** join/#asterisk lindi- (~lindi@kulho150.adsl.netsonic.fi) |
00:45.24 | Matt-E- | why is the console answering my outgoing calls ? |
00:45.34 | miguellinux | Hi, I have problems with IAXtel |
00:45.36 | YoYo | because you mucked up extensions.conf |
00:45.50 | YoYo | then talk to iaxtel ppl =D |
00:45.51 | Matt-E- | for example? |
00:46.11 | miguellinux | it doesnt suppor ulaw? |
00:46.49 | jeffik | Anybody have any exprience wiht livevoip? |
00:46.49 | miguellinux | it forces to other very crancky codecs |
00:53.59 | *** join/#asterisk lqdengr (~fweston@69.172.205.68.cfl.res.rr.com) |
00:56.32 | Darwin35 | Freeze Drop the mouse |
00:56.47 | Darwin35 | hands off the keyboards |
00:57.07 | Darwin35 | this is a mandatory 10 min break from typing. |
00:57.24 | Darwin35 | step away from the pc and stretch and forage for food and fluids |
00:58.20 | Eight | the #include command doesn't accept wildcards does it? |
01:00.21 | Darwin35 | #include or include => |
01:00.26 | Eight | #include |
01:00.38 | Eight | if it did, that'd be handy =) |
01:02.49 | Darwin35 | <=== brain just went fart |
01:03.04 | Darwin35 | man all this work in extensions.conf |
01:03.12 | Darwin35 | everything should work now |
01:04.21 | *** join/#asterisk DaLion (DaLion@Toronto-HSE-ppp3881624.sympatico.ca) |
01:04.25 | DaLion | anyone got radio working on with .. rawplayer ???.. loooking for a decent broadcast? |
01:05.03 | *** join/#asterisk ACiDV (~acidvicio@122-68-181.dr.cgocable.ca) |
01:06.48 | *** join/#asterisk tuxinator_linuxM (~tuxinator@m410e36d0.tmodns.net) |
01:07.02 | ACiDV | Hi =) I try to dial multiple channel like: Dial(SIP/1000&SIP/1001&Local/819477XXXX@outgoing). The problem is that the Local/ call a remote IAX server and when it call, it reply with Call accepted and other devices dont ring. Does exist a way to bypass this ? |
01:07.26 | KalD|Work | ok this is a long shot - but anyone here have an old nightowl subscription? |
01:12.18 | *** join/#asterisk jeffik (~jeffik@m6d9836d0.tmodns.net) |
01:15.31 | *** join/#asterisk Newbie___ (some@60.48.48.175) |
01:15.46 | Newbie___ | hi |
01:15.57 | Newbie___ | anyone heard of telextreme.com ? |
01:19.05 | ariel_ | Newbie___, have you signed up with them? |
01:19.11 | *** join/#asterisk Gator (~krp@ip24-250-198-83.ga.at.cox.net) |
01:21.38 | *** join/#asterisk zapa (~zant@201.135.137.236) |
01:22.05 | *** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net) |
01:23.07 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
01:23.42 | Newbie___ | ariel_: no, i tried calling my experience with them is scary |
01:23.43 | zapa | hi all, anybody has expirence with echo cancel options in MARK2 MARK3 STEVE or STEVE2 , i canīt resolve my echo troubles with my e1 any clue |
01:23.58 | Newbie___ | i dont think that comapny even exists |
01:24.08 | ariel_ | The rates at Broadvoice are better. |
01:24.19 | Newbie___ | i am going for unlimted cell |
01:24.25 | Newbie___ | ariel_: u signed up ? |
01:25.13 | Newbie___ | ariel_: brodvoice do not accept non USA credit card so i have problem there |
01:25.24 | denon | zapa: its worth searching the wiki on echo problems .. tons of articles and good info |
01:25.32 | ariel_ | no I have been thinking of BV but keep reading all the problems people keep having. So I am staying for now with my did from VoicePulse and ld from voipjet,nufone and livevoip. |
01:26.31 | Newbie___ | tried ping to broadvoice server, i am getting like over 200-300 ms |
01:26.48 | zapa | thanks denon, really i try a lot of configuration and only one side have echo troubles, thank again |
01:27.14 | Qwell | Newbie___: I believe nufone accepts paypal |
01:27.54 | Qwell | not sure if they have an unlimited plan though... |
01:28.04 | Newbie___ | cheking out nufone now |
01:28.52 | Newbie___ | it doesnt say anything about unlimted calls, so i assume there is none |
01:29.16 | Qwell | well, how many minutes do you use a month? |
01:29.32 | Newbie___ | about 5000+ world wide |
01:29.42 | Qwell | Thats a bit, yeah |
01:29.49 | Qwell | not sure an unlimited plan would really cover you there, heh |
01:30.23 | Newbie___ | broadvoice does, but initial ping result does not look good and the crdit card |
01:30.27 | Newbie___ | problem |
01:30.43 | Qwell | "unlimited" doesn't mean quite what it sounds like it means |
01:31.20 | Eight | Qwell: at what point does BV start shouting "unreasonable usage"? |
01:31.21 | *** join/#asterisk santiago (~santiago@63.245.86.95) |
01:31.26 | Qwell | got me |
01:31.31 | *** part/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
01:31.32 | Qwell | surely before 5000 minutes though |
01:31.34 | |Vulture| | ~5k |
01:31.45 | Newbie___ | Qwell: someone told me that, and i wrote to BV and they affirm me that unlimited is unlimted. no hidden charge |
01:31.56 | Qwell | Newbie___: Get it in writing. |
01:32.00 | |Vulture| | Ive heard if your consistantly getting around 5k for 2 months your gone |
01:32.18 | |Vulture| | Newbie___: no hidden charge.. but they will dump the account |
01:32.38 | Qwell | maybe dan2 will speak up |
01:32.49 | Newbie___ | |Vulture|: yeah, BV said they have the right to terminate the account if they get 'suspicious' |
01:33.16 | Qwell | honestly though, if you need 5000 minutes, an unlimited plan isn't right for you |
01:33.34 | Newbie___ | Qwell: any recommendation ? |
01:33.43 | Qwell | I'd still say nufone |
01:34.09 | moonwick | I'd say that if he needs 5000 minutes that an unlimited plan is perfect. :P |
01:34.22 | Qwell | nah, any unlim plan will boot him for 5k |
01:34.51 | Newbie___ | finding provider is one thing and voice clarity is another |
01:34.57 | Newbie___ | wish i am in the USA |
01:35.08 | Qwell | Newbie___: another great thing about non unlim providers, is that you can test them out first |
01:35.17 | Qwell | drop $2-5, test it for a bit...see how it works out |
01:35.42 | Newbie___ | Qwell: i agrre |
01:35.45 | Newbie___ | agree |
01:35.50 | Qwell | and for the record, if you're going over an ocean(I assume you are), anything will be a little higher |
01:36.22 | Newbie___ | how will the voice sound like if is 200-300 ms ? |
01:36.31 | Qwell | probably as expected |
01:36.54 | *** join/#asterisk godsmoke (~godsmoke@66-108-159-216.nyc.rr.com) |
01:37.36 | Newbie___ | life is a bitch |
01:37.47 | Qwell | where are you hailing from? |
01:38.09 | Newbie___ | Malaysia |
01:39.25 | Eight | And you're trying to get cheap calls in the US? |
01:39.29 | ACiDV | Does it's possible to dial (ring) multiple extensions that are connected on different servers ? like : Dial(IAX2/server1/ext1&IAX2/server1/ext2&IAX2/server2/ext1&...) ? |
01:39.56 | Newbie___ | Eight: no, USA + other part of the world |
01:40.03 | *** join/#asterisk puppet (puppet@1-1-3-3b.ox.mlm.bostream.se) |
01:40.12 | Eight | Newbie___: I wouldn't worry about the BV ping, for calls to the US (and other part of the world) |
01:40.13 | puppet | there :) |
01:40.32 | Eight | Newbie___: it's just going to take a LONG time for the info to make the trip, wether it spends most of it on the 'net or most of it on some long haul carrier's phone network. |
01:40.38 | puppet | Anyone that live in scandianvia and wanne get rid of rackboxes? ;p |
01:41.43 | Newbie___ | i was given from some provider * forgot the name* the clarity is good provided the voice reach me. speak from 1-10, 20% are missing |
01:42.11 | Newbie___ | i mean free 0.25 credit to try out |
01:43.14 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
01:46.29 | *** join/#asterisk xachen (justin@toto.citelnetworks.com) |
01:46.56 | xachen | I'm trying to develop a Dial Command for a extension so I can dial out onto the phone system |
01:47.06 | xachen | how would I make it send a call if its prefixed with 9 |
01:47.16 | xachen | I just need it to strip out the 9 and send the ${exten} |
01:47.17 | Qwell | ${EXTEN:1} |
01:47.21 | xachen | thanks :) |
01:49.35 | Eight | alright, i'm going to settle once and for all how ast' needs to be configured for BV. |
01:50.07 | Eight | fromuser and authname aren't even valid values, and they're in the only config that worked for me =p |
01:50.52 | WhiteWlf | I' |
01:51.14 | WhiteWlf | In my dialplan, how could I prompt for an extension then direct them to the approiate voicemail box? |
01:51.46 | Eight | WhiteWlf: check out the macro in the default extensions.conf sample file. |
01:51.54 | Eight | WhiteWlf: I don't know how it works, but I think that does what you want. |
01:52.00 | Eight | or similar. |
01:52.03 | *** join/#asterisk SPoon_TSX (~SPoon_TSX@24.83.96.211) |
01:52.16 | WhiteWlf | Eight: the uhh... default dial thingy one that I can't remember the name of? |
01:52.19 | Newbie___ | hmmmm net2hone is using SIP now |
01:52.31 | SPoon_TSX | Hi out there, Just wondering how is your experience on buying VoIP Equitment off voipsupply.com? |
01:52.56 | Newbie___ | i bought mine from voxilla |
01:52.59 | WhiteWlf | Eight: macro-stdexten? |
01:53.16 | SPoon_TSX | voxilla? Is it a Canadian or US company? |
01:53.18 | Qwell | SPoon_TSX: The only time I tried, I had an issue...turned me off to them. |
01:53.35 | Eight | WhiteWlf: that's what I had in mind. I've never looked at it though, so you're on your own from here =) |
01:53.55 | SPoon_TSX | Where I can get the VoIP equitment in Canada? |
01:54.16 | Newbie___ | SPoon_TSX: is a US company |
01:54.23 | Qwell | SPoon_TSX: I'm convinced that many of these will ship to canada |
01:54.34 | WhiteWlf | Eight: It's called with ${EXTEN}, and sends them there using that variable... but I thought that contained the exten you were calling from... no? |
01:54.46 | SPoon_TSX | Qwell: Do have their website? |
01:54.51 | Eight | SPoon_TSX: gah, y'know... there was one store that was listing prices in CAD and I almost didn't delete it from my list thinking "Maybe someone will ask for the link... naaah" |
01:54.51 | Newbie___ | SPoon_TSX: they sent to me half way around the world |
01:55.02 | Qwell | SPoon_TSX: dunno, voxilla.com, I'd imagine |
01:55.02 | Eight | SPoon_TSX: that was last night... There *is* one though. |
01:55.23 | SPoon_TSX | Eight: May i have the web address? |
01:55.31 | SPoon_TSX | Qwell: Thanks. |
01:55.33 | Eight | SPoon_TSX: I was just saying, I deleted it last night. |
01:55.34 | Newbie___ | http://www.voxilla.com/ |
01:55.46 | SPoon_TSX | Eight: O, is okay. |
01:59.01 | Eight | It's amazing to me there are more, and better, hardware SIP phones than software. |
01:59.12 | WhiteWlf | Eight: Same here |
01:59.13 | Eight | Seems so backwards from the norm. |
01:59.23 | Qwell | I'd think there would be more hardware ones |
01:59.45 | WhiteWlf | It's easier to make a piece of software than hardware. |
02:00.03 | WhiteWlf | More so, a piece of hardware that runs software. |
02:00.03 | Qwell | sure, but who's gonna buy a softphone? |
02:00.15 | Eight | Who said anything about buy? |
02:00.17 | Nugget | I own two eyebeam licenses. |
02:00.17 | WhiteWlf | who's going to buy Asterisk? |
02:00.27 | Qwell | Nugget: I didn't quite mean it like that. :P |
02:00.43 | Qwell | I mean...is the general public more likely to buy a softphone, or hardphone? |
02:00.44 | WhiteWlf | Good software doesn't mean it costs... |
02:01.11 | WhiteWlf | mmm... hardware |
02:01.13 | Qwell | I'm thinking more from a "my company only exists to make money" point of view |
02:03.07 | Eight | anyway to kick Ast' into re-registering? |
02:03.41 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
02:04.38 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
02:06.22 | WhiteWlf | I think reloading re-registers |
02:07.17 | Nukemizer | I want to link 2 Asterisk boxes for calling between systems. To help me seearch for my answers can anyone tell me what the acronym to describe * networking ? in my PBX world it is called QSIG |
02:07.57 | Nukemizer | or is it just that simple "asterisk networking" ? |
02:09.16 | *** join/#asterisk DaLion (DaLion@Toronto-HSE-ppp3884408.sympatico.ca) |
02:09.25 | DaLion | <PROTECTED> |
02:09.25 | DaLion | <PROTECTED> |
02:10.21 | shido6 | show lag? |
02:10.33 | shido6 | sip show peers if you have a qualify statement set in sip.conf |
02:10.33 | DaLion | well i want to see channel quiality |
02:10.51 | DaLion | <PROTECTED> |
02:11.14 | DaLion | hmm |
02:11.20 | DaLion | that not the call quality |
02:11.31 | DaLion | tring to see if QOS working |
02:12.02 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.res.rr.com) |
02:13.52 | Eight | er, non-root. |
02:14.25 | Eight | oooh, I kinda like the new 2.6 device system... /dev/ is actually readable! |
02:15.54 | godsmoke | I'm getting this strange behavior when trying upgrade my cisco 7960 -- since it's not asterisk related, if someone could help, pm me |
02:16.09 | *** join/#asterisk jeffik (~jeffik@m919f36d0.tmodns.net) |
02:17.08 | Eight | and I broke BV again. |
02:22.25 | Essobi | Somewhat readable. |
02:23.11 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
02:23.42 | Newbie___ | hey Essobi |
02:26.43 | *** join/#asterisk bjohnson (~bjohnson@ip137-172.dsl.istop.com) |
02:32.42 | Newbie___ | hi bjohnson |
02:33.18 | Newbie___ | bjohnson: u once told me to get SPA 2000, and is been working great |
02:33.50 | *** join/#asterisk blaisen1 (~blaisen1@tightcode.ofpower.net) |
02:34.02 | blaisen1 | anyone know where to get toronto DIDs? |
02:34.38 | Essobi | Umm. |
02:34.51 | Essobi | Bell Canada? |
02:35.29 | blaisen1 | well i meant someone doing SIP or IAX origination |
02:35.35 | blaisen1 | but yeah i guess they'd work |
02:35.39 | *** join/#asterisk DaLion (DaLion@Quebec-HSE-ppp224769.qc.sympatico.ca) |
02:35.50 | DaLion | hey |
02:35.50 | DaLion | QOS on sveasoft works great ! |
02:36.18 | nine76 | x100p's here,rebooting/installing. bye |
02:36.28 | blaisen1 | hey, anyone know why i have two mpg123 processes for my music on hold (coming from a radio station's mp3 stream)? |
02:36.37 | DaLion | yes |
02:36.43 | DaLion | mpg123 has a multi thread bug |
02:36.47 | DaLion | use something else |
02:36.53 | blaisen1 | like? |
02:36.57 | DaLion | let me check |
02:37.03 | DaLion | what station u using btw ? |
02:37.12 | blaisen1 | LIFE 100.3 FM Barrie, Ont |
02:37.29 | DaLion | default => custom:/var/lib/asterisk/mohmp3_raw,/usr/bin/rawplayer |
02:37.35 | DaLion | blaisne whats the link ? |
02:37.51 | DaLion | check wiki for compiling.. thinks its... |
02:37.58 | DaLion | ~rawplayer |
02:38.06 | blaisen1 | www.fm100.net |
02:38.09 | blaisen1 | rawplayer? |
02:38.12 | DaLion | yeah |
02:38.38 | *** join/#asterisk Damin_Mobile (~pocketirc@242.sub-166-155-102.myvzw.com) |
02:38.39 | godsmoke | for some reason -- my cisco 7960 tries to load a file called P0S3-07-.bin |
02:38.43 | godsmoke | anyone seen this before? |
02:38.53 | DaLion | hy.. no.. no cisco here ;( |
02:38.55 | PatrickDK | ya, that is normal |
02:39.10 | DaLion | it tftpinh |
02:39.10 | godsmoke | PatrickDK: it can't be -- that file shouldn't exist, and doesn't |
02:39.14 | dan2 | Qwell: if your burning 5000 minutes a month you'll be terminated if you are running a residential plan |
02:39.16 | DaLion | tftping the bin.. |
02:39.22 | *** join/#asterisk firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
02:39.28 | godsmoke | DaLion: heh -- forget it -- you don't understand |
02:39.48 | DaLion | ah.. lol |
02:39.52 | Nugget | yeah, that filename doesn't look right |
02:39.56 | PatrickDK | hmm, maybe it forgot it's mac? |
02:39.58 | blaisen1 | dalion: rawplayer will play shoutcast/mp3 streams? |
02:39.58 | godsmoke | of course not |
02:40.05 | godsmoke | no, this has nothing to do with its mac address |
02:40.11 | godsmoke | it does this before trying to load the .cnf |
02:40.14 | Damin_Mobile | godsmoke: Look at your tftp config files. |
02:40.24 | PatrickDK | heh, it's been awhile since I loaded the 7960 |
02:40.37 | DaLion | ~rtfm |
02:40.38 | jbot | rtfm is, like, read the f*cking manual... try asking me about "FAQ" |
02:40.47 | DaLion | ~faq |
02:40.48 | firestrm | anyone know what mightMar 12 10:51:28 NOTICE[5023]: chan_iax2.c:5444 socket_read: Rejected connect attempt from 24.68.44.53 |
02:40.48 | firestrm | Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT |
02:40.48 | firestrm | <PROTECTED> |
02:40.48 | firestrm | <PROTECTED> |
02:40.58 | DaLion | ~tiki |
02:41.00 | blaisen1 | dalion: I can't find where to download rawpalyer..? |
02:41.08 | DaLion | im trying to find tikiwiki addy |
02:41.10 | DaLion | lol |
02:41.14 | DaLion | ~wiki |
02:41.36 | DaLion | ~asterisk |
02:41.37 | jbot | hmm... asterisk is a PBX (Private Branch eXchange) and telephony toolkit. http://www.asterisk.org |
02:41.52 | Damin_Mobile | firestrm; rejected connect attempt due to wrong credentials |
02:42.13 | *** join/#asterisk bjohnson_ (~bjohnson@ip137-172.dsl.istop.com) |
02:42.23 | DaLion | ok hold on |
02:42.29 | blaisen1 | hmmm... looks like rawplayer won't decode mp3 |
02:42.41 | DaLion | http://www.voip-info.org/tiki-index.php?page=Asterisk%20mpg123%20faking%20it |
02:42.42 | blaisen1 | so it will probably not work in my application (receiving an mp3 stream from a radio station) |
02:42.54 | Qwell | dan2: Thanks. |
02:43.11 | Qwell | dan2: How much lower is the "limit"? |
02:43.17 | firestrm | im having problems connecting 2 * boxes together.. anyone know what might be happining here? http://pastebin.ca/7282 |
02:43.35 | firestrm | somthing is breaking in iax, but i dont know what |
02:44.38 | DaLion | ah true |
02:45.51 | blaisen1 | well it looks like it is only downloading 1 stream, so the second instance may be harmless |
02:46.12 | blaisen1 | i was worried it would be using twice the bandwidth maintaining two connections to the radio station |
02:46.19 | bjohnson_ | ManxPower: earlier you asked my DID provider for SW Ontario .. sixtel |
02:46.32 | blaisen1 | sixtel is sloooooooooooooooow at doing anything |
02:46.42 | Damin_Mobile | firestrm: http://lists.digium.com/pipermail/asterisk-users/2004-August/057681.html |
02:46.45 | shido6 | wow |
02:46.51 | blaisen1 | like adding new ontario dids |
02:46.53 | shido6 | SW ontario, eh? |
02:46.58 | blaisen1 | or cancelling stuff they say they've already ancelled |
02:47.13 | shido6 | how many dids and where do u need them? |
02:47.41 | blaisen1 | toronto/london/kitchener/ottawa and if possible windsor/hamilton/st catherines |
02:47.59 | blaisen1 | it took me over 3 weeks to get a london ont did with sixtel, and even then it didn't work |
02:48.10 | blaisen1 | it looks like its going to some cell phone calling card reseller.. wtf...? |
02:48.27 | Essobi | nice |
02:48.34 | blaisen1 | i would ixnay on the sixtelay |
02:48.47 | Essobi | ixstelay |
02:48.50 | blaisen1 | it is such a pita to find a good canadian DID/800 provider |
02:49.03 | Essobi | Hmm. |
02:49.06 | blaisen1 | i have service with livevoip which is good but the fact that they change their website and offerings on an hourly basis makes me nervous |
02:49.11 | Essobi | I'll ask at work monday. |
02:49.20 | Essobi | How many DIDs you interested in? |
02:49.31 | Eight | grrr, anyone know what file permissions I might be overlooking that would prevent a remote connection to asterisk from the shell? |
02:49.33 | blaisen1 | when i signed up i was under the impression canadian 800 originatoin was 1.29 cents/min now i'm told its 5 cents/min USD! i can get allstream 800 orgin to pstn for 4.25 cents |
02:49.36 | blaisen1 | and sprint for 3.5 |
02:49.38 | bjohnson_ | shido6: you have yours yet? |
02:49.43 | Eight | I have already looked through this: http://www.voip-info.org/wiki-Asterisk+non-root |
02:49.46 | Essobi | 5 cents USD? |
02:49.50 | Essobi | You're getting raped. |
02:49.53 | blaisen1 | essobi: no doubt |
02:50.09 | *** join/#asterisk BoboTWF (~joshuas-a@rrcs-66-27-57-228.west.biz.rr.com) |
02:50.28 | Essobi | I guess it might depends on how many DIDs you want too thou. |
02:50.32 | blaisen1 | ok so where in the heck do i get ontario DIDs for a reasonable price and canadian 800 origination for sane prices that SHOULD be below what the telcos will offer |
02:50.46 | Essobi | We got US50 at work. |
02:50.46 | bjohnson_ | Essobi: that is cheap for CDN 800 voip number |
02:50.55 | blaisen1 | well we want to resell service to our ADSL customers but right now I'm concerned about all this crtc reg bs |
02:50.56 | Essobi | I dunno if there is anything in the works for canada. |
02:50.59 | Eight | Grrrr, it's still putting its PID file in /var/run/ not /var/run/asterisk |
02:51.02 | Essobi | I know we cover some of mexico too. |
02:51.05 | bjohnson_ | Essobi: exactly |
02:51.24 | Essobi | 5 cents USD for a canadian 800? |
02:51.28 | bjohnson_ | Essobi: everything that covers canada is more expensive for voip |
02:51.30 | bjohnson_ | yes |
02:51.37 | Essobi | Rock. I'm moving to canada. |
02:51.39 | bjohnson_ | shido6: London and Kitchener |
02:51.52 | blaisen1 | so how can sprint canada/callnet do 3.5 cents/min canadian orig? |
02:52.02 | blaisen1 | and thats canadian $$ not US |
02:52.03 | bjohnson_ | gotta go .. be back in 15 minutes |
02:52.18 | shido6 | how many? shoot requirements over to shido6@gmail.com |
02:52.28 | shido6 | cdn dids and 8xx |
02:52.55 | firestrm | ok here is my iax config on the two machines http://pastebin.ca/7284 any one know why its not accepting credentials? |
02:53.00 | Essobi | So what company do you exactly hustle for Shido? |
02:53.33 | *** join/#asterisk xyharley (~daecon@xyharley.dsl.xmission.com) |
02:54.43 | blaisen1 | i've seen shido6@gmail.com on calltermination.com |
02:55.18 | blaisen1 | i have no volume right now because i have no reliable carrier to work some packages around for my potential customers |
02:55.36 | blaisen1 | so i'm just looking for reasonable rates with potential for discounts based on volume |
02:56.05 | blaisen1 | i found unlimitel which seemed like a good deal.. until you realize they bill by the minute |
02:56.06 | _570RM_ | hmmm |
02:56.18 | _570RM_ | does anyone see why this line is not working: |
02:56.21 | _570RM_ | exten => _0.,1,Dial(Modem/g1:${EXTEN:1}) |
02:56.40 | _570RM_ | group 1 is a BRI passive i4l card |
02:57.20 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
02:57.41 | _570RM_ | it can receive calls alright, but i cant get it to dial out |
02:58.33 | firestrm | do i have to set a username /password or will static ip do? |
02:58.38 | firestrm | in IAX |
03:00.32 | modulus_ | yeah i got your IAX right here baby |
03:00.58 | firestrm | modulus_, thanks :~ |
03:01.06 | Eight | @#$# @#$# @##$ |
03:01.20 | firestrm | Eight, i know how you feel |
03:01.41 | Eight | it keeps dropping the pid file in /var/run/, not /var/run/asterisk/! |
03:01.46 | Eight | or, rather, failing to. |
03:01.49 | firestrm | Eight, i think swearing is allowed here, it part of asterisk experince :) |
03:02.46 | firestrm | modulus_, any ideas where my config is going wrong? |
03:03.12 | firestrm | im allmost understanding IAX config, but im missing somthing.. |
03:05.29 | bjohnson_ | shido6: I signed up for 1 London and 1 Kitchener at sixtel on Feb 17. The London one I got about a week later. I'm still waiting for Kitchener with them |
03:05.44 | Eight | asterisk.h:#define AST_PID ASTVARRUNDIR "/asterisk.pid" |
03:06.03 | Eight | Makefile:ASTVARRUNDIR=$(INSTALL_PREFIX)/var/run/asterisk |
03:06.15 | bjohnson_ | shido6: both would be low usage .. likely about 30 minutes per month to start .. maybe up to 200 each per month after a year |
03:06.19 | Eight | [blake@star run]$ ls /var/run/asterisk.pid |
03:06.27 | Eight | <PROTECTED> |
03:06.40 | Eight | what the hell? |
03:06.59 | jeffik | bjohnson_ may i ask what DID provider you are using? |
03:07.21 | bjohnson_ | jeffik: sixtel |
03:07.31 | bjohnson_ | jeffik: sorry .. iax.cc |
03:07.44 | bjohnson_ | jeffik: sorry .. iax.cc = sixtel |
03:08.14 | jeffik | I'm looking for a provide for Toronoto DID s |
03:08.28 | bjohnson_ | there's a bunch that do Toronto |
03:08.35 | bjohnson_ | even some that ONLY do Toronto |
03:08.59 | jeffik | i saw but could only get a response from unlimitel |
03:09.13 | bjohnson_ | I don't remember there names because I was looking for other areas |
03:09.19 | jeffik | i's for my own asterisk@home |
03:09.24 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net) |
03:10.26 | bjohnson_ | one guy at comcast told me they could hook up a SIP connection for me but another guy said no .. I decided I wasn't interested since they only had major cities |
03:11.04 | firestrm | arrrgh IAX blows! |
03:11.05 | DaLion | No one do Quebec Province.. |
03:11.52 | jeffik | ok, well i gotta get someone other than livevoip my number has not worked rihght in weeks and they blame it on their canadian provider |
03:12.09 | bjohnson_ | I think someone in here about 2 months ago found 405 DIDs |
03:12.24 | Syncros | DaLion ? |
03:12.27 | firestrm | why cant they give a )($#*)# proper example of how to use iax between two machines anywhere. F*&K this pisses me off |
03:12.39 | bjohnson_ | firestrm: it's no big deal |
03:12.52 | bjohnson_ | firestrm: when you get it working make sure you document it then |
03:13.07 | firestrm | bjohnson_, is is for me.. ive been trying to get it working for 3 hours |
03:13.36 | bjohnson_ | firestrm: set up a friend on each side .. use the same username, secret, and section name on both machines |
03:13.47 | jontow | what about the 'asterisk dual servers' entry on the wiki? does that have one? |
03:14.38 | bjohnson_ | firestrm: if one has a dynamic IP .. it has to have a register command. If one has a static IP .. it doesn't need a register but the oppisite machine needs a host= line in iax.conf |
03:14.39 | *** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg) |
03:15.01 | firestrm | bjohnson_ http://pastebin.ca/7284, im using static ip's so no pass/user |
03:15.29 | jontow | that doesn't make much sense |
03:16.05 | firestrm | bjohnson_, i have them set up as peer though |
03:16.18 | jeffik | bjohnson_: sorry, are you using sixtel now? |
03:18.14 | bjohnson_ | firestrm: start with friend .. peer is for sending only |
03:18.25 | firestrm | bjohnson_, i keep getting no athority found.. |
03:18.39 | bjohnson_ | jeffik: yes .. for one DID .. been waiting since Feb 17 for the second one |
03:18.54 | firestrm | bjohnson_, changed to friend, still same result |
03:19.08 | jeffik | ok, cause i see they offer 416 numbers |
03:19.53 | jontow | yeah.. set them up as friend, put a user:pass in there |
03:20.37 | Eight | This is insane... There is something very fragile about BV. |
03:20.39 | jontow | i have many machines working IAX to IAX |
03:20.48 | Eight | I've been fiddling with this config file for hours now and I still haven't identified it. |
03:21.01 | Eight | I went back to my 'known working', and now it doesn't work either. |
03:21.30 | bjohnson_ | firestrm: did you read this? http://www.voip-info.org/tiki-index.php?page=Asterisk+IAX+authentication |
03:21.54 | bjohnson_ | firestrm: I have it working but am using the same username, secret, and section names on both servers |
03:22.42 | opus___ | eight - yeah |
03:22.55 | bjohnson_ | firestrm: adding username and secret info will not add or detract from security .. I just know it works |
03:23.11 | opus___ | eight - curious, what do you notice right now thats strage? |
03:23.24 | opus___ | strange even |
03:23.31 | Eight | opus___: outgoing calls seem to work when expected... |
03:23.37 | Eight | opus___: but incoming are flakey as all hell. |
03:23.45 | firestrm | bjohnson_, yes but im barely grasping the concepts as it is, usr/pass confuses me even more.. |
03:24.00 | bjohnson_ | firestrm: then just do it |
03:24.35 | opus___ | eight - shit, maybe we should all go in on a T1 |
03:24.42 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
03:24.46 | bjohnson_ | firestrm: you can do what works and figure it out later .. or try to figure out why the one you have doesn't work (and doesn;t match the config of 2 people with working systems) |
03:24.55 | Eight | opus___: I thought we had, and it was called Broadvoice =p |
03:25.20 | jontow | i have a freebsd machine with 4-7 IAX connections to various other machines (with differing OS'/distributions of linux) |
03:25.32 | opus___ | So basically they just went around to each state and colocated a T1 box.. hmmm |
03:25.46 | Eight | opus___: yu'up. |
03:25.47 | jontow | i can put the config for a pair of them online if you want. |
03:26.08 | heison | <PROTECTED> |
03:26.31 | opus___ | at $600 per t1 card, $1600 1U ibm machine, $200 per T1 = $2400 costs plus $400 /month for rackspace and T1 ... |
03:26.37 | MikeJ[Jayden] | heison, you just copied and pasted that from 5 hrs ago. |
03:26.40 | heison | turns out cisco3 does not register with asterisk upon reboots |
03:26.54 | jontow | heison; you need to check hardware settings.. |
03:26.55 | Eight | opus___: you apparently know the numbers better than I do. |
03:26.56 | MikeJ[Jayden] | there you go |
03:26.57 | heison | MikeJ[Jayden]: yes, almost |
03:27.03 | MikeJ[Jayden] | hehe |
03:27.05 | Eight | opus___: but ya, if you want to go ahead and do that, start in Minneapolis =) |
03:27.12 | opus___ | hehe |
03:27.17 | Eight | opus___: I've been thinking about it... |
03:27.17 | heison | how do i force the 7960 to register? |
03:27.19 | *** join/#asterisk warmfeet (~c@213.78.240.109) |
03:27.31 | heison | jontow: can u be more specific? |
03:27.34 | opus___ | Eight - it sucks that we'd have to colocate in each city with low capital. |
03:27.41 | jontow | heison; SIPmacaddr.cnf is where you'd have to set all that |
03:27.44 | firestrm | bjohnson_, what does the dial format look like for that. i want to dial remote extension 10 username vince pass blah |
03:28.00 | opus___ | i wish you could like VPN to other state's T1 lines.. |
03:28.04 | jontow | make user and [name] the same.. |
03:28.07 | jontow | ie. |
03:28.07 | Eight | opus___: well, you don't HAVE to collocate in each city... but you pay connection fees otherwise. |
03:28.09 | jontow | [iaxlink] |
03:28.13 | jontow | user=iaxlink |
03:28.15 | jontow | secret=blah |
03:28.18 | jontow | type=friend |
03:28.23 | warmfeet | Is it possible for asterisk to create the voicemail INBOX and paths when first voicemail comes, rather than precreating them |
03:28.27 | opus___ | i don't fully understand the telcom industry |
03:28.27 | jontow | host=XXX.YYY.ZZZ.QQQ |
03:28.35 | jontow | ... |
03:28.47 | jontow | then in the dialplan, set IAXLINK=IAX2/iaxlink@iaxlink |
03:29.07 | Eight | opus___: it's old, bureaucratic, heavily regulated, and just kinda funky. |
03:29.11 | heison | jontow: and what param forces the phone to register if it doesn't do it by itself |
03:29.13 | firestrm | jontow on which end.. |
03:29.15 | jontow | exten => n,1,Dial(${IAXLINK}/${EXTEN}) |
03:29.22 | jontow | both ends |
03:29.25 | warmfeet | and does neone has exp with MWI,it seems that u need to hv individual users in sip.conf...which I dont since I have them in SER |
03:29.27 | *** part/#asterisk DaLion (DaLion@Quebec-HSE-ppp224769.qc.sympatico.ca) |
03:29.40 | jontow | if you want to make calls from either end, and receive them on the other, that is.. |
03:30.17 | bjohnson_ | firestrm: I just use dial(IAX2/remoteserver/${exten}) |
03:30.31 | jontow | and heison, check the firmware revisions.. what are the phones running? |
03:30.32 | bjohnson_ | where remoteserver matches the section name in iax.conf |
03:30.45 | heison | jontow: they are all SIP 7.3 |
03:30.49 | jontow | hmm |
03:31.06 | jontow | heison; if they can't contact the tftp server they don't register |
03:31.16 | jontow | but they WILL be able to make calls, as is the nature of SIP (doesn't need a proxy, per se.) |
03:31.22 | Eight | And now BV is working again, but I didn't change anything... |
03:31.34 | jontow | thats why i said.. check the hardware (network) settings. |
03:31.35 | firestrm | jontow, dont you need a host= somwhere in there? otherwhise how can it find the other server? |
03:31.39 | heison | they do contact the tftp server, but doesn't register with * |
03:31.53 | *** join/#asterisk alexns (~alex@acs-24-154-114-15.zoominternet.net) |
03:31.57 | jontow | firestrm; re-read my 'paste' .. :) |
03:32.05 | heison | anyone here using SIP 7.x? |
03:32.12 | jontow | those aren't complete entries.. let me just put mine up, it'll make this hours less of a discussion :D |
03:32.14 | heison | on 7960s? |
03:32.26 | jontow | heison; i am.. i had 13 of them running 7.3 with no problems except slow bootup time. |
03:32.34 | alexns | i would be if i could get sip 7.x |
03:32.45 | heison | jontow: do you see a time stamp on your phone? |
03:32.49 | jontow | yes |
03:32.52 | jontow | thats another network feature |
03:32.58 | heison | i don't have those anymore... |
03:32.59 | firestrm | jontow, oops there it was a few lines down :) |
03:33.07 | jontow | make sure you've got the netmask and gateway setup correctly.. your problems are ALL in your network settings |
03:33.19 | heison | jontow: during the swapping of hardware, i lost my SIPDefault.cnf |
03:33.21 | jontow | it needs proper DNS and gateway information |
03:33.33 | jontow | want a sample, while i'm pulling config files? |
03:33.37 | heison | jonton: can you show me your SIPDefault.cnf? |
03:33.45 | heison | jontow: yes please |
03:33.51 | jontow | ;) yep, gimme a bit |
03:33.55 | Bacon | Howdy. |
03:33.55 | heison | thanks |
03:34.08 | Bacon | I'm trying to setup Asterisk with BroadVoice. |
03:34.20 | Bacon | I have inbound working, but not outbound... |
03:34.39 | Bacon | Mar 11 23:28:25 VERBOSE[1487]: -- Got SIP response 404 "Not Found" back from 147.135.8.128 |
03:35.00 | Bacon | I don't know enough about asterisk to start debugging... |
03:35.04 | Bacon | Any suggestions? |
03:36.08 | opus___ | bacon - what is your outgoing configuration |
03:36.13 | opus___ | brb reboot |
03:36.57 | firestrm | jontow, bjohnson_, that worked!! once again im in your debt.. i sure hope we meet some time so i can repay you guys in the canadian beverage of choice (beer) |
03:37.34 | firestrm | where i was going wring was i was trying to send the extension where the username is supposed to go.. |
03:37.37 | bjohnson_ | firestrm: I suspect you could do it your way and the answer lies somwhere in http://www.voip-info.org/tiki-index.php?page=Asterisk+IAX+authentication |
03:37.42 | *** join/#asterisk Goshen (~Goshen@c-67-172-238-57.client.comcast.net) |
03:37.47 | bjohnson_ | but that's not how I did it |
03:38.03 | bjohnson_ | (ie the way you were unsuccessfully trying) |
03:38.05 | Dandan | heh, anyone had any success with MPG321? |
03:38.10 | firestrm | bjohnson_, at this point im just glad it worked.. |
03:38.12 | bjohnson_ | firestrm: now edit the wiki page |
03:38.19 | Dandan | instead of security crippled mpg123 ? |
03:38.19 | firestrm | bjohnson_, i WILL!! |
03:38.27 | bjohnson_ | Dandan: no |
03:38.49 | jontow | firestrm; http://mno.bsd.st/~jontow/2005-03-11/iax.conf.txt |
03:39.02 | jontow | sorry its a little late |
03:39.09 | jontow | but thats my example.. :) |
03:39.29 | Dandan | bjohnson_: maybe asterisk community should look into incorporating mpg321 |
03:39.37 | Dandan | which is still under development... |
03:39.38 | Bacon | opus___: You rebooted yet? |
03:39.40 | Dandan | and maintained |
03:39.45 | opus___ | yup |
03:40.06 | jeffik | bjohnson_: using asterisk@home and wonder how difficul to get DID from sixel running |
03:40.07 | opus___ | Bacon - just to let you know I haven't got mine to work either, but I'm working on it now |
03:40.22 | firestrm | jontow, where i was going wrong was i was passing the extension where the username was .. ie dial(iax/10@iaxconnect,30) rather than dial(iax/iaxconnect/10). |
03:40.26 | Bacon | opus___: What do you have working? |
03:40.35 | Bacon | opus___: Like I said, I have inbound working... |
03:40.46 | firestrm | it was a dumbass mistake that was invisible die to it being right infront of me.. |
03:41.43 | firestrm | but im going to go throw up a wiki page especially on the subject now.. mind if i use your example as one of the ways to do it? |
03:41.57 | jontow | and heison; http://mno.bsd.st/~jontow/2005-03-11/SIPDefault.cnf |
03:42.09 | jontow | for the place to specify the server, look for "192.168.2.1" as a string |
03:42.40 | jontow | keep in mind that config file sets 'telnet_level' to 2 (privileged mode) |
03:42.55 | jontow | you need to get into the settings on the phone itself before you may get it all working |
03:43.19 | bjohnson_ | jeffik: easy if you edit the config files .. I don;t know the gui |
03:43.31 | jontow | go for it, don't even care if you mention me :) |
03:43.41 | firestrm | jontow, mind if i use you example for the wiki? |
03:44.03 | bjohnson_ | firestrm: add to the dual server page .. don't start a new one |
03:44.27 | opus___ | bacon - i just have inbound working |
03:44.31 | firestrm | bjohnson_ i'll try to find that one.. |
03:44.31 | opus___ | lemme post my config |
03:44.37 | opus___ | for outgoing, ... |
03:44.55 | heison | jontow: i found a couple problem already... |
03:44.58 | channan | hi, anyone's familiar with dialing to Toluca, Mexico? I've tried to call an old friend but automated operator announced busy tone all the time |
03:45.08 | heison | proxy_register, sntp_server |
03:45.11 | channan | the number is: 011-52-72 xxxxx |
03:45.42 | channan | I looked at from the web and it seemd to change to 011-52-722, but still did not work |
03:45.46 | jontow | heison ;) |
03:45.51 | jeffik | bjohnosn_: well you can edit the files direcly or use the gui. all i need is the settngs, i'm asking as i had a less than possitive experience wiht livevoip.com |
03:46.20 | heison | it's taking forever to reboot... but will see that's my only problem, thanks. |
03:46.34 | bjohnson_ | they give you the setting when you get a DID |
03:46.55 | firestrm | bjohnson_, i think i found it, do you mean this page ? http://www.voip-info.org/tiki-index.php?page=Asterisk%20-%20dual%20servers#comments |
03:46.57 | heison | jontow: the time is back, and i no longer have a "x" besides the extension |
03:47.04 | jontow | ;) |
03:47.04 | opus___ | bacon http://pastebin.ca/7287 |
03:47.10 | bjohnson_ | they even substitute your username, password, and DID into the config examples (they're dynamically created) |
03:47.23 | bjohnson_ | firestrm: yes |
03:47.31 | heison | and Asterisk now sees the phone as registered! |
03:47.34 | heison | thanks man |
03:47.36 | firestrm | ok.. i will add my bit there.. |
03:47.37 | jontow | np |
03:48.02 | heison | i guess i took it for granted without looking thru the .cnf file before. |
03:48.05 | jeffik | bjohnson_: thanks, seems like they are worth a try |
03:48.22 | opus___ | bacon - when you run asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvgc you can see the output when it tries to call, I updated that url with my error |
03:48.40 | opus___ | <PROTECTED> |
03:48.43 | modulus_ | how about asterisk -vvvvvvvvvvvvvvvvvvvvvvvvgc? |
03:48.43 | jontow | heison; btw, those are hacked up examples based on the ones that are available at cisco.com (if you dig REAAAALLLLY deep and have an account) |
03:48.43 | opus___ | was what it said |
03:48.44 | mikegrb | ... |
03:48.59 | *** join/#asterisk zagaya972 (~d2s-compa@APointe-a-Pitre-102-1-18-150.w81-248.abo.wanadoo.fr) |
03:49.05 | mikegrb | I run asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvgc so I can be cool like opus___ |
03:49.05 | opus___ | modulus is there a better way to do it |
03:49.13 | Bacon | opus___: You should have host=proxy.lax.broadvoice.com |
03:49.21 | heison | jontow: i got my account about 3 hrs ago, and yes i have seen something similiar on their site |
03:49.35 | jontow | :) |
03:49.45 | opus___ | bacon - that one seems to go down, but let me try it again |
03:50.33 | Bacon | opus___: Are you behind a nat? |
03:50.44 | opus___ | no, but my phone is |
03:51.01 | Bacon | opus___: Is your phone working with your asterisk box ok? |
03:51.03 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
03:51.06 | opus___ | yes |
03:51.31 | opus___ | proxy.lax won't let me register, again... |
03:52.15 | Bacon | opus___: Have you been here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20settings%20Broadvoice |
03:52.28 | opus___ | yes |
03:53.17 | jeffik | opus___: i am using x-lite begind a wifi router, can i access my asteisk without opening ports on the router using nat? |
03:53.33 | opus___ | I am able to call in while dialing out, but dialing out gives me the 'nobody picked up in 30000 ms' |
03:54.05 | opus___ | jeffik - I think so |
03:54.55 | jeffik | opus___: what do i need to set? and do i need cooresponding settings on asterisk? |
03:55.43 | *** join/#asterisk JmanA9 (~josh@pa-murraysville2b-141.pit.adelphia.net) |
03:55.51 | JmanA9 | hello |
03:56.04 | JmanA9 | when i'm running asterisk for the first time, i get this error message in the console and it doesn't start up: |
03:56.16 | JmanA9 | [pbx_loopback.so]Mar 11 22:55:20 WARNING[21616]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_loopback.so: undefined symbol: pbx_substitute_variables_varshead |
03:56.16 | JmanA9 | Mar 11 22:55:20 WARNING[21616]: loader.c:440 load_modules: Loading module pbx_loopback.so failed! |
03:56.24 | JmanA9 | google's turning up nothing, anyone know what to do? |
03:56.45 | jontow | jmana9; how did you install? compile from source? binary? where'd you get the tarball? cvs? |
03:57.07 | JmanA9 | i got the tarball from the cvs |
03:57.24 | Eight | uh, what? |
03:57.26 | JmanA9 | just did make clean and make samples |
03:57.44 | Bacon | Mar 11 23:51:03 DEBUG[1487]: Stopping retransmission on '20596ebe59e1e1a8538812d3222448f0@sip.broadvoice.com' of Request 102: Found |
03:57.46 | jontow | what about 'make install' ? |
03:57.56 | jontow | and 'make' for that matter? |
03:58.03 | Bacon | Mar 11 23:51:03 VERBOSE[1487]: -- Got SIP response 480 "Temporarily Not Available" back from 147.135.12.128 |
03:58.16 | opus___ | jeffik - here is an example config http://pastebin.ca/7290 |
03:58.27 | JmanA9 | i think i may have forgotten to do a make install.... |
03:58.30 | JmanA9 | i'm such an idiot |
03:58.35 | jontow | ;) |
03:58.36 | JmanA9 | lol |
03:58.56 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.we.client2.attbi.com) |
03:58.57 | opus___ | jeffik - the key is for sip.conf/[general] to have nat=yes externip=yadayadayad and for your sip device [xlite]/nat=yes/qualify=yes |
03:59.54 | *** join/#asterisk jets (~jetsn@xyharley.dsl.xmission.com) |
03:59.55 | opus___ | jeffik - i've never used xlite, there might be additional special lines needed in its sip entry. I posted how i got sjphone to work |
04:01.04 | opus___ | bacon, any luck? |
04:01.41 | *** join/#asterisk Himeko (~himeko@S01060040ca128fc3.ed.shawcable.net) |
04:02.37 | *** part/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
04:03.16 | jeffik | opus___: thnanks, this is a good start i will try it |
04:03.59 | *** join/#asterisk jsolares (~jsolares@200.12.33.64) |
04:04.38 | opus___ | hmmm, eyebeam |
04:06.13 | jeffik | opus___: you like sjphone? |
04:07.05 | opus___ | jeffik - yes |
04:07.35 | jeffik | opus___: are you in Canada? |
04:07.37 | opus___ | simple, works, easy to setup. not much debugging info/or I haven't really looked |
04:07.44 | opus___ | jeffik - usa oregon |
04:07.55 | warmfeet | anyone good with perl regex here....I need to match sip.domain.com in vmail.cgi as opposed to having a context with no '.' |
04:08.17 | warmfeet | wS I guess |
04:08.34 | warmfeet | i mean ~\wS$ |
04:08.40 | jeffik | opus___: Toronto/Chicago |
04:09.28 | opus___ | jeff cool. |
04:09.39 | *** part/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
04:10.00 | JmanA9 | well, i recomplied everything twice, i'm still getting that error |
04:10.14 | opus___ | jeffik - i'm looking at https://sip-communicator.dev.java.net/download.html right now, apparently it does video with asterisk |
04:10.33 | JmanA9 | i obtained everything from the cvs, did a make clean ; make install for everything, also did a make samples for asterisk :/ |
04:11.17 | jeffik | opus___: would be nice but got to get my asteisk@home running with Toroto/Chicago DIDs and outgoing |
04:11.32 | opus___ | jman - try removing /var/lib/asterisk |
04:11.35 | opus___ | or whatever |
04:12.12 | *** join/#asterisk Jabreity (~jfkdlsjk@12-222-3-81.client.insightBB.com) |
04:12.20 | JmanA9 | ok, i'll try that |
04:12.36 | JmanA9 | still no good :( |
04:16.31 | Eight | http://www.voip-info.org/tiki-index.php?page=Asterisk+settings+Broadvoice |
04:16.39 | Eight | I've added the 'second example' on that page. |
04:16.49 | Eight | I think it clarifies some things somewhat. |
04:19.22 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
04:21.10 | Jabreity | howdy all |
04:21.38 | *** join/#asterisk afrosheen (~afro@c-67-166-172-141.client.comcast.net) |
04:21.41 | afrosheen | hey gang |
04:24.05 | afrosheen | lively bunch tonight eh |
04:24.12 | Grooby | aye |
04:24.13 | afrosheen | that must mean everything is working right |
04:24.13 | jsolares | very |
04:24.25 | afrosheen | anyone tackle VOIP payphones before? |
04:24.58 | jsolares | nope |
04:25.07 | afrosheen | I may have a huge project on my hands if I can figure out a nice way of implementing it |
04:25.13 | JmanA9 | any known issues with fc3 and this error i'm getting? |
04:25.54 | afrosheen | yes |
04:26.03 | jsolares | afrosheen: have you looked into analogue payphones but connected to an iaxy? |
04:26.04 | JmanA9 | have anything to do with udev? |
04:26.11 | Bacon | opus___: I was afk for a bit. |
04:26.15 | Bacon | opus___: Any luck? |
04:26.16 | Jabreity | hey, i have what i think is a valid question |
04:26.27 | afrosheen | jsolares: yeah and I did some thinking on it, it seems like a good idea for a number of reasons |
04:26.33 | Bacon | There are no invalid questions, only invalid people. |
04:26.40 | Jabreity | :) |
04:26.43 | jsolares | hehe |
04:26.45 | _6Flamez_ | jman: v1 or HEAD? |
04:27.00 | JmanA9 | i have no clue, whatever comes with fc3 :/ |
04:27.02 | afrosheen | jsolares: reason one: keep the analog lines from the telco closet to the phones = no rewiring |
04:27.16 | Jabreity | ok, its gotta be a simple config err somewhere, but i have a handfull of extensions setup |
04:27.21 | _6Flamez_ | what * version? |
04:27.29 | afrosheen | jsolares: reason two: with the iaxy's in the telco closet, they can easily plug into the UPS for failsafe-ness |
04:27.32 | jsolares | afrosheen: i have no experience whatsoever with payphones be it analog or ip. but how does the analog pay phone determine the time it has |
04:27.43 | JmanA9 | 1.0.3 |
04:27.48 | afrosheen | jsolares: that's handled on the * server |
04:28.07 | opus___ | bacon -- no :( |
04:28.11 | afrosheen | jsolares: the phone must pass some kind of tones via the iaxy to * I guess :~ |
04:28.18 | jsolares | ah |
04:28.19 | opus___ | I think I have mine setup exactly like everyone elses as well |
04:28.23 | Bacon | opus___: What errors are you getting? |
04:28.32 | jsolares | good thing the iaxy uses ulaw then :) |
04:28.43 | Bacon | opus___: Mar 12 00:23:38 VERBOSE[1487]: -- Got SIP response 404 "Not Found" back from 147.135.0.128 |
04:28.47 | Bacon | Thats my pain. |
04:28.49 | afrosheen | oh |
04:28.53 | afrosheen | registration issues |
04:29.59 | afrosheen | jsolares: yeah but it'll end up g729 before it hits the t3 |
04:30.28 | Jabreity | ok, i have voicemail configed proper, i can send vm between extensions by dialing into the vm system, however i dont get rollover after x rings, or when busy when i dial an extension |
04:30.36 | jsolares | but the * is between that? eg. iaxy to asterisk in "local" net and then onto a voip provider? |
04:30.52 | jsolares | i havent seen many voip payphones |
04:30.54 | Bacon | afrosheen: I have registration issues, or someone else does? |
04:31.07 | afrosheen | jsolares: yeah iaxy is physically 2 feet from the switch, 2 feet from the * box |
04:31.14 | Essobi | How much does a 5300 with 4 PRI cards cost? |
04:31.18 | _6Flamez_ | Bacon: you do |
04:31.22 | jsolares | then it *should* be doable hehehe |
04:31.23 | afrosheen | Bacon: you do |
04:31.41 | Bacon | Odd, I'm getting inbould calls. |
04:31.44 | afrosheen | jsolares: you got me thinking about how * will handle the time from each phone |
04:32.17 | afrosheen | jsolares: with a normal analog payphone, each coin generates a tone that's picked up by a switch somewhere I believe |
04:32.25 | Jabreity | yup |
04:32.38 | jsolares | hmmm |
04:32.43 | afrosheen | if * can listen for those tones it can calculate how long the call should last |
04:32.47 | Bacon | Any hints as to where to start looking for my problems? I'm pretty new to Asterisk. |
04:32.52 | Essobi | jsolares It's a 4 wire POTS with a special switch on the other end.. atleast.. that's how bell does it. |
04:33.02 | Jabreity | in some fortresses |
04:33.04 | Essobi | All the rest COCOTS are not like that. |
04:33.24 | Essobi | 2 wire, and everything is completely physically driven inside the phone. |
04:33.31 | Jabreity | correct |
04:33.34 | Essobi | No change, no bring bring. |
04:33.34 | afrosheen | the iaxy *should* pass those coin tones to * right? |
04:33.39 | jsolares | get 2 wire payphones afrosheen! |
04:33.40 | jsolares | hehe |
04:33.49 | Essobi | 2 wires don't drop remote tones. |
04:33.56 | Essobi | only the 4 wires. |
04:34.02 | afrosheen | I imagine they'll be 4 wire phones |
04:34.03 | Essobi | 2 wires work like a normal phone |
04:34.13 | afrosheen | it'll be a retrofit in a big place, like an airport |
04:34.20 | Essobi | and the other 2 are odd pass or no pass signaling for the $$ |
04:34.25 | Essobi | Umm. |
04:34.26 | Jabreity | in theroy you could lay cc number on the line, and after approval permit carrier on line |
04:34.35 | Essobi | I doubt you'd get a BIG airport. |
04:34.42 | afrosheen | Essobi: why's that |
04:34.45 | Essobi | As those want dataports on their payphones. |
04:34.48 | Essobi | for the laptops. |
04:34.51 | Jabreity | BIG airports = BIG money |
04:34.55 | Essobi | As those want dataports on their payphones. |
04:34.59 | afrosheen | there is big money at stake here |
04:35.15 | Essobi | Umm.. Yea. |
04:35.16 | afrosheen | I'm in with a guy that owns phones in 70-something airports nationwide |
04:35.29 | Essobi | So? |
04:35.35 | afrosheen | I suggested voip termination and a light went on above his head |
04:35.51 | afrosheen | so now it's my baby to see if it can be done |
04:35.54 | jsolares | hehe |
04:35.56 | Essobi | Hire me to engineer them, and I'll drop, wire and write an LCD for him. |
04:36.04 | Essobi | :) |
04:36.16 | Essobi | Um... anything can be done. |
04:36.18 | opus___ | dude, i want to red box a iaxy |
04:36.20 | Essobi | Anything. |
04:36.21 | Jabreity | ... just give him lsd. wont know the diff |
04:36.22 | Essobi | Haha. |
04:36.23 | opus___ | tell me which airport :) |
04:36.31 | afrosheen | 'all of them' |
04:36.37 | jsolares | hehehe |
04:36.44 | Essobi | umm.. he "Owns" them? |
04:36.46 | Jabreity | All your base... |
04:36.59 | Essobi | RBELONGINZORSTOUZES |
04:36.59 | afrosheen | yeah he owns the phones, has the right to rip them all out and replace them |
04:37.03 | Essobi | Ahh. |
04:37.11 | afrosheen | like I said, big money |
04:37.14 | Essobi | I thought you meant... he owned the airports.. |
04:37.15 | Essobi | hah |
04:37.23 | Jabreity | mmmmm, smells like someon stepped in some fresh capitalism |
04:37.27 | afrosheen | he said bell earns $25 per phone on average |
04:37.40 | afrosheen | and if he could cut them out of the picture = more moeny |
04:37.41 | afrosheen | :) |
04:37.44 | Jabreity | per day/month/yr? |
04:37.45 | Essobi | What's he going to do about faxing and data? |
04:37.49 | afrosheen | daily I believe |
04:37.56 | Jabreity | nice |
04:37.59 | afrosheen | dude these are just payphones |
04:38.04 | Essobi | He could put some 1FBs in. |
04:38.06 | Essobi | Dude. |
04:38.08 | afrosheen | we're not starting a kinko's at each damn phone |
04:38.17 | Essobi | My local airport.. has data jacks on the payphones. |
04:38.19 | Jabreity | ooooh, add a wap |
04:38.25 | Essobi | :) |
04:38.36 | opus___ | yeah, they got data jacks. but there's also free wireless too.. |
04:38.39 | afrosheen | I considered some kinda funky mesh network..but maybe too much hassle and fcc/faa issues |
04:38.41 | Dandan | anyone using BV can show me his register string? |
04:38.47 | Dandan | <PROTECTED> |
04:38.48 | *** join/#asterisk Weezey (~Weezey@206.210.109.226) |
04:38.51 | Dandan | that's what i get |
04:38.54 | jeffik | opus___/bjohonson_: thanks for the help |
04:38.59 | afrosheen | Dandan: did you patch asterisk yet |
04:39.08 | Weezey | is there a device out there that acts as a SIP peer and a MGCP client? |
04:39.12 | Dandan | asterisk from CVS... |
04:39.14 | afrosheen | I thought I saw a posting from broadvoice recently mentioning some patch |
04:39.17 | Dandan | read that you need no patch |
04:39.33 | opus___ | dandan http://www.voip-info.org/tiki-index.php?page=Asterisk+settings+Broadvoice |
04:39.41 | Dandan | opus___: that's what i did |
04:39.49 | Dandan | and I am getting 404 |
04:39.53 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
04:39.59 | Dandan | so i wanted a real life example |
04:40.12 | *** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net) |
04:40.17 | Bacon | Dandan: Are you behind a nat? |
04:40.22 | Dandan | no |
04:40.24 | opus___ | register => 555121212@sip.broadvoice.com:mypassword:mynumberagain@sip.broadvoice.com |
04:40.25 | Dandan | external ip |
04:40.45 | opus___ | thats my real life fake example |
04:40.56 | opus___ | i don't live in 555 yet:) |
04:40.59 | Bacon | Yup. That is what I have. |
04:41.05 | Dandan | shhh |
04:41.07 | Weezey | I call 1212121 locally. |
04:41.09 | Dandan | lemmie look :) |
04:41.31 | *** join/#asterisk jjg (~clh@adsl-69-107-18-183.dsl.pltn13.pacbell.net) |
04:41.38 | Dandan | hmmm |
04:41.43 | Dandan | i built asterisk today |
04:41.47 | Weezey | yay! |
04:41.48 | Dandan | it says i need no patch |
04:41.53 | Dandan | anyone can verify? |
04:42.00 | opus___ | yes |
04:42.10 | opus___ | with cvs head from friday |
04:42.28 | Dandan | yes: do i need it? |
04:42.34 | Dandan | or yes: i am correct? |
04:42.42 | Bacon | My config: http://pastebin.ca/7299 |
04:42.56 | jjg | were there any enhancements available via CVS due to VON? |
04:43.19 | Dandan | Bacon: tx |
04:43.28 | Dandan | will work with what you pasted |
04:43.36 | Bacon | Dandan: I can get inbould calls, but I can't dial out. |
04:43.44 | _6Flamez_ | host=proxy.lax.broadvoice.com is wrong i believe should be host=sip.broadvoice.com |
04:43.46 | Dandan | Bacon: y? |
04:43.52 | Bacon | Some of the folks here say I have registration problems. |
04:44.04 | Bacon | _6Flamez_: I'll try that. |
04:44.17 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
04:44.42 | Sedorox | Does anyone know if mpg123 has a problem with SMP FreeBSD systems? |
04:45.11 | Jabreity | ok can anyone tell me why my voicemail isint picking up after 6 rings? |
04:45.24 | Bacon | _6Flamez_: kickass. |
04:45.27 | Bacon | It works. |
04:45.39 | _6Flamez_ | cool |
04:45.45 | Bacon | So much for broadvoice's instructions. |
04:45.51 | Dandan | Bacon: the password is whatever you use to get to their website |
04:45.52 | opus___ | bacon - you can call out? |
04:45.55 | Sedorox | Jabreity: do you have it where the extention will timeout? e.g. exten => 1000,Dial(SIP,user,25,rt), where 25 is about 4 rings... |
04:46.02 | Sedorox | 25 secs... |
04:46.09 | Jabreity | oh |
04:46.14 | Jabreity | craptastic |
04:46.15 | Sedorox | Bacon: BV's instructions are messed |
04:46.15 | Bacon | opus___: Yup. |
04:46.25 | opus___ | bacon - hmmm, is it any different from my config? |
04:46.25 | Jabreity | im a crackhead |
04:46.45 | Bacon | opus___: Dunno, click the link and check it out. |
04:46.48 | Sedorox | Jabreity: gotta have it will it'll time out and goto the next priority :-p |
04:46.59 | Sedorox | will=where |
04:47.01 | opus___ | http://pastebin.ca/7288 |
04:48.48 | Jabreity | sedrox - |
04:48.49 | Jabreity | 101=> 101, 101,jasonbreitwieser@hotmail.com |
04:48.49 | *** join/#asterisk JMcA (~jmcadams@67.141.1.51) |
04:49.14 | Dandan | Mar 11 23:48:31 NOTICE[13759]: chan_sip.c:4309 sip_reg_timeout: -- Registration for '860XXXXXXX@sip.broadvoice.com@sip.broadvoice.com' timed out, trying again |
04:49.18 | Jabreity | im missing something for sure |
04:49.20 | Dandan | <PROTECTED> |
04:49.26 | Dandan | anyone can help me with that? |
04:49.42 | Sedorox | Jabreity: thats in voicemail.conf, right? |
04:49.47 | Jabreity | yes |
04:50.02 | Sedorox | Jabreity: no.. the example I gave is extentions.conf |
04:50.08 | Jabreity | ? |
04:50.11 | Jabreity | duh |
04:50.20 | Jabreity | sorry, long evening |
04:50.29 | Sedorox | 'tis fine |
04:50.29 | Sedorox | :-p |
04:50.31 | Sedorox | I've been there |
04:50.51 | Sedorox | yea.. you want to have a timeout on the extention so after 25 seconds.. if no pickup.. it goes to the next priority.. which should be voicemail |
04:50.56 | Sedorox | or dialing another phone.. etc.. |
04:51.11 | Bacon | opus___: Here is mine: http://pastebin.ca/7299 |
04:51.18 | Jabreity | ok, lemmie post my config online, i think i got it right |
04:51.23 | Sedorox | ok |
04:51.31 | Bacon | Dandan: http://pastebin.ca/7299 |
04:51.32 | tuxinator_linuxM | Anybody still in San Jose, or is it just me? |
04:51.53 | Sedorox | well the extention should just look like exten => 1000,1,Dial(SIP,user,25,rt) exten => 1000,2,Voicemail(u1000) or something like that |
04:51.57 | Sedorox | depending on your setup... |
04:52.15 | Sedorox | I use 25.. beause its about 4-5 rings.. the average for answering machines.. altho some people's are higher |
04:53.18 | _6Flamez_ | after 25 secs if will jump to 101 not 2 |
04:53.37 | Sedorox | mine always jumps to 2... hmmm |
04:53.46 | Sedorox | if busy.. goes to 102, right? |
04:53.48 | Dandan | Bacon: that's what i have |
04:53.59 | tuxinator_linuxM | just 101, not to 101 |
04:54.03 | tuxinator_linuxM | jump |
04:54.06 | Jabreity | Sedorox: http://www.theextremeoutfitters.com/extensions.txt |
04:54.19 | Dandan | Bacon: how did you build your *? |
04:54.28 | Dandan | did you cvs? ftp? patched? |
04:54.55 | Sedorox | Jabreity: ok.. the extentions will timeout after 60 seconds.. or 30.. depending on which one your using... |
04:55.50 | Sedorox | except somewhere in there you have to have exten => 100,101,Voicemail(u100) (where 101 apparently is where it goes after 60 secs.. and the u100 is user 100 in voicemail.conf) |
04:56.06 | Bacon | Dandan: I cheated, I install asterisk@home |
04:56.08 | jjg | has anyone built an iax client? |
04:56.09 | Sedorox | if you want.. I can put up some examples of mine... |
04:56.09 | Jabreity | Sedorox: bearing in mind im emotionally fragile at this point, this is my first time... trying to learn |
04:56.27 | Sedorox | Jabreity: thats perfectly fine... should I start to curse you out? :-p j/k!!! |
04:56.38 | Sedorox | I was there a few weeks ago learning the dialplan |
04:56.56 | Jabreity | :) go for it. i prolly deserve it from the blasting i gave my boxen |
04:57.05 | Sedorox | ahah |
04:57.07 | _6Flamez_ | heh |
04:57.16 | Sedorox | here.. let me paste a few examples outta my config |
04:57.21 | Jabreity | okies |
04:57.46 | opus___ | bacon - it just doesn't ring.. hmm |
04:58.34 | Dandan | Bacon: LOL :) |
05:00.13 | Sedorox | Jabreity: http://pastebin.ca/7300 |
05:00.17 | *** join/#asterisk nine76 (~t00r@cpe-69-135-184-24.woh.rr.com) |
05:00.20 | Jabreity | Sedorox: did i make a noob mistake or what :) |
05:00.26 | Sedorox | dunno |
05:00.26 | Sedorox | hehe |
05:00.32 | Sedorox | probably.. don't think your including everything |
05:01.00 | lqdengr | hi guys, i'm new to *, i've got my inbound VoIP service working with an IVR menu where I can dial my extension and ring my x-lite softphone, but I can't seem to figure out how i would go about configuring * to let me dial out of the system from a x-lite, it says call not permitted |
05:01.39 | _6Flamez_ | huh |
05:01.57 | Sedorox | lqdengr: you just need to configure a extention to dial out on... or a matching plan I think.. like if you dial 928005551212.. where 9 is used to say "hey.. I'm dialing out" |
05:02.04 | Sedorox | there's some good stuff in the wiki about it... |
05:02.26 | lqdengr | sedorox: so basically i need to set up some extension or "code" to tell * that i need an outside line? |
05:02.31 | Sedorox | also.... you can look at the example configs that are posted there.. they help if you learn that way (like me.. bu seeing and implamenting) |
05:02.37 | Dandan | ok, quick hack, how to make an extensions which would say date/time |
05:02.38 | Dandan | ? |
05:02.45 | Dandan | as 612 in *@home |
05:02.55 | *** join/#asterisk ikey (~kirankuma@202.54.37.186) |
05:03.08 | lqdengr | sedorox: is http://www.voip-info.org/wiki-Outbound+call+handling what i need to be looking at? |
05:03.15 | Sedorox | well now.. you just need to have it match.. you can have it where you can just dial the number... or you can have it where you need to press 9 first.. or any other number (most poeple do standard and pick 9 to dial out of) |
05:03.22 | *** join/#asterisk JmanA9 (~josh@pa-murraysville2b-141.pit.adelphia.net) |
05:03.26 | Juggie | 8 is cooler :P |
05:03.29 | lqdengr | lol |
05:03.30 | ikey | hi can any one help me in configuring two sip channels in asterisk |
05:03.32 | _6Flamez_ | Dandan: exten => 999,2,DateTime |
05:03.41 | _6Flamez_ | be sure to Answer first! |
05:03.44 | Dandan | _6Flamez_ thx a lot |
05:03.44 | Dandan | :) |
05:03.46 | Sedorox | lqdengr: looks a little complicated.. but yes |
05:04.02 | lqdengr | sedorox: ok ill putz around with that for a while, thx for your help! |
05:04.39 | Sedorox | yup... like I said.. the best way would probably be under the example configs.. people have it where they can dial out a FXO.. or dial out a SIP.. its normally the same.. if you can't get it.. let me know.. I'll help ya |
05:05.23 | *** join/#asterisk sudhir492 (~sudhir@4.7.58.171) |
05:05.34 | afrosheen | ikey: just start asking, don't ask to ask |
05:05.52 | sudhir492 | anyone using PAP2-NA or Sipura SPA-2000 here? |
05:06.02 | *** join/#asterisk locovox (~locovox@218-153-89-200.fibertel.com.ar) |
05:06.27 | sudhir492 | I cannot use both phones at the same time with PAP2-NA |
05:06.34 | locovox | hi just need a little help configuring my first asterisk with a TDM21B |
05:07.06 | afrosheen | locovox: what's in a tdm21b |
05:07.41 | Sedorox | afrosheen: I think its a FXO and FXS.... |
05:07.50 | afrosheen | let the man answer :) |
05:07.57 | Sedorox | hehe |
05:07.59 | locovox | i'm trying to make it work, however i think this time i got dialton!!! |
05:08.29 | Sedorox | oh wow.. I was wrong... |
05:08.43 | locovox | gimme five mins |
05:08.46 | opus___ | bacon -- -- Executing Dial("SIP/175-faee", "SIP/18005551212@sip.broadvoice.com") in new stack |
05:08.47 | opus___ | <PROTECTED> |
05:08.50 | opus___ | does your line say that? |
05:09.07 | opus___ | when dialing? |
05:09.57 | Dandan | opus___: u using BV? |
05:11.06 | opus___ | yes |
05:11.08 | opus___ | when dialing out |
05:11.36 | Goshen | howdy...funny you two are talking about that...I just switched over here to get help with my outbound broadvoice |
05:11.46 | Goshen | I can receive calls, but not dial out |
05:11.57 | opus___ | goshen - well, I can give you my config. For some reason it works fine with other ppl:) |
05:12.06 | Goshen | ok, lets have it |
05:12.14 | Goshen | patebin.ca or query window |
05:12.15 | _6Flamez_ | sure a lot of BV related probs... |
05:12.19 | Dandan | opus___: can i have it also? |
05:12.21 | opus___ | goshen - http://pastebin.ca/7288 |
05:12.27 | Goshen | http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup didn't work for me |
05:12.29 | opus___ | Hey it works for everyone but me! :) |
05:12.33 | Dandan | _6Flamez_: their website is confusing at best |
05:12.40 | Goshen | 6Flamez: thats because its crappy SIP! I hate sip |
05:12.43 | *** join/#asterisk _santiago_ (~santiago@63.245.86.95) |
05:12.58 | _6Flamez_ | what link... one is corrent and one isnt |
05:12.59 | Goshen | but they have LMP..so they get my ring |
05:13.03 | _6Flamez_ | correct* |
05:13.17 | opus___ | LMP? |
05:13.22 | _6Flamez_ | ya i dropped them due to sip issues |
05:13.35 | afrosheen | sip is good people |
05:13.54 | afrosheen | until polycom and others start making iax phones |
05:13.59 | Goshen | sip and its huge range of natdefying ports stinks |
05:14.04 | Dandan | opus___: and your register string? |
05:14.12 | Goshen | my register string works |
05:14.14 | opus___ | polycom probably won't make iax phones |
05:14.20 | afrosheen | yeah I know |
05:14.25 | Goshen | polycom has IAX phones... |
05:14.27 | afrosheen | we're lucky they're making good sip phones |
05:14.32 | opus___ | register => 555121212@sip.broadvoice.com:mypassword:mynumberagain@sip.broadvoice.com |
05:14.40 | opus___ | goshen oh? |
05:14.42 | Dandan | hm :/ |
05:14.50 | Goshen | make sure you use the sip registration password |
05:14.56 | Goshen | NOT the website password |
05:15.12 | Bacon | Yeah, that tricked me. |
05:15.13 | afrosheen | since when does polycom make iax hardware? |
05:15.16 | opus___ | goshen -- incoming works fine, outgoing has problems.. that wouldn't be it would it? |
05:15.19 | Goshen | you get your sip password from the account page |
05:15.38 | Goshen | if incoming works fine it isn't your registration string |
05:15.44 | _6Flamez_ | Goshen: what do u have as your host= line? |
05:18.09 | Goshen | in my nonworking outbound sip.conf? |
05:18.18 | _6Flamez_ | ya |
05:18.25 | Goshen | host=sip.broadvoice.com |
05:18.43 | _6Flamez_ | ok.. usually that's the problem, but yours is correct |
05:18.50 | opus___ | exten=_9NXXNXXXXXX, 1, Dial(SIP/${EXTEN:1}@sip.broadvoice.com) still won't dial out, I just get dead air and a timeout |
05:18.54 | Goshen | the other nonworking one that I tried is |
05:18.56 | Goshen | host=proxy.broadvoice.com |
05:19.01 | opus___ | audio works full duplex on incoming just fine |
05:19.10 | _6Flamez_ | has to be sip.broadvoice.com |
05:19.30 | opus___ | goshen - proxy=sip.broadvoce.com works too |
05:19.32 | _6Flamez_ | if you want to use a different proxy, u'd need to add a /etc/hosts entry etc.. |
05:19.54 | opus___ | that will work but is not required |
05:20.09 | opus___ | proxy=proxy.***.broadvoice.com works |
05:20.14 | opus___ | on some servers... |
05:20.23 | _6Flamez_ | that changed? |
05:20.24 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
05:20.30 | opus___ | It really all depeneds on random numbers it seems:) |
05:20.40 | Eight | _6Flamez_: I tried the /etc/hosts trick, no joy. |
05:20.46 | Goshen | bah! same crap... |
05:20.47 | Eight | It seems some accounts just aren't on some proxies. |
05:20.47 | Goshen | Executing Dial("SIP/21-b012", "SIP/8017123381@sip.broadvoice.com|30") in new stack |
05:20.47 | Goshen | <PROTECTED> |
05:20.47 | Goshen | <PROTECTED> |
05:21.09 | Eight | http://www.voip-info.org/wiki-Asterisk+settings+Broadvoice |
05:21.09 | opus___ | gosh hmm |
05:21.14 | Eight | Have a look at the second example. |
05:21.15 | Goshen | I am using the cvs stable from today |
05:21.17 | Eight | That one works for me. |
05:21.41 | Goshen | I am using the 1.0.7RC |
05:21.55 | opus___ | thats todays cvs right? |
05:22.06 | Goshen | yup |
05:22.13 | Goshen | from this afternoon MST |
05:22.20 | Goshen | so it should include the patch |
05:22.29 | Eight | 1.0.6 works without patching |
05:23.01 | Dandan | *CLI> Mar 12 00:22:42 NOTICE[13979]: chan_sip.c:8776 sip_poke_noanswer: Peer 'sip.broadvoice.com' is now UNREACHABLE! Last qualify: 0 |
05:23.07 | Dandan | huh? |
05:23.08 | Dandan | which means? |
05:23.34 | Nugget | it means pretty much exactly what it says. |
05:23.36 | Eight | Dandan: your 'net connection is flaking, and you can't ping sip.broadvoice.com |
05:23.39 | Nugget | which part is confusing? |
05:23.45 | Eight | Dandan: did you try /etc/hosts |
05:23.48 | opus___ | Uhh, is this correct: |
05:23.53 | Dandan | Eight: no, should i? |
05:23.59 | opus___ | To get the current stable release, issue the following command: |
05:23.59 | Eight | actually... |
05:24.02 | Dandan | Mar 12 00:23:44 NOTICE[13979]: chan_sip.c:4309 sip_reg_timeout: -- Registration for '8607772005@sip.broadvoice.com@sip.broadvoice.com' timed out, trying again |
05:24.03 | opus___ | # cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds |
05:24.09 | opus___ | won't that download 1.0.7RC? |
05:24.21 | Eight | Actually, I *just* had BV go unreachable for a moment. |
05:24.24 | Nugget | that will download cvs stable. |
05:24.31 | Eight | I think their network flaked out for a moment. |
05:24.31 | opus___ | oh i'm dum |
05:24.38 | Dandan | Eight: oh :) |
05:24.41 | Dandan | cool |
05:24.54 | Dandan | 8) |
05:25.02 | Goshen | ITS A MIRACLE!!! |
05:25.10 | Goshen | The second example on the wiki works!!! |
05:25.12 | Dandan | doesn't make it any less confusing for a newbie... |
05:25.16 | Goshen | THANKS! |
05:25.16 | Eight | Goshen: Your welcome. |
05:25.22 | Eight | That's my config. |
05:25.27 | Sedorox | lol |
05:25.32 | Eight | err, "you're", rather. |
05:25.32 | Goshen | now we just need to delete all of the other crap... |
05:25.42 | opus___ | nugget -- wait, is stable = 1.0.7rc in cvs? |
05:25.43 | Goshen | and send broadvoice your config |
05:25.53 | Nugget | stable is stable. |
05:25.58 | Goshen | and tell them to send that out as the example |
05:26.00 | opus___ | http://bugs.digium.com/bug_view_page.php?bug_id=0003746 saids We are ready to release 1.0.7 but need some people to test and verify that the code is without any major bugs. |
05:26.08 | lqdengr | sedorox: i looked around at some sample configs for outbound calls, and i still can't seem to get it working |
05:26.10 | Goshen | Asterisk CVS-v1-0-03/11/05-12:59:27 built by root@localhost on a i686 running Linux |
05:26.24 | opus___ | "Please grab the latest code from stable CVS (cvs co -r v1-0 asterisk)." |
05:26.30 | Sedorox | lqdengr: how do you want to dial out? PSTN, SIP, IAX, CAPI? |
05:26.44 | lqdengr | IAX |
05:26.48 | Goshen | ok, everyone have their broadvoice working now? |
05:26.50 | Sedorox | with who? |
05:26.55 | Goshen | I have incoming and outgoing working now |
05:27.07 | opus___ | goshen -- glad I could help. DOesn't work for me though:) hehe |
05:27.07 | lqdengr | i have a line in my extensions.conf that looks like exten => _1NXXNXXXXXX,1,Dial(IAX2/foo:bar@gwiaxt01.voicepulse.com/${EXTEN}) |
05:27.16 | Goshen | oh..and if you have pulver communicator and upnp on your nat...you are going to have problems |
05:27.26 | Goshen | because pulver communicator takes sip ports |
05:27.38 | Sedorox | lqdengr: then you should be able to dial 1 and the numbers |
05:27.40 | Goshen | opus: what problems are you having? |
05:27.41 | Sedorox | number* |
05:27.52 | Sedorox | so like... |
05:27.58 | opus___ | goshen -- can't dial out, I'm going to try today's cvs "stable" first |
05:27.59 | lqdengr | sedorox: i think i must have it under the wrong contect or something, because it still isn't working when i try that |
05:27.59 | Sedorox | 1-123-555-1212 |
05:28.04 | Dandan | Eight: pls pastebin it :) |
05:28.06 | Goshen | ahh ok |
05:28.09 | Sedorox | what context do you have it under? |
05:28.16 | Goshen | you may not have the latest patches for BV |
05:28.26 | lqdengr | [outbound] |
05:28.35 | lqdengr | thats what the sample file from voicepulse had it listed under |
05:28.46 | Sedorox | do you have include => outbound, under either local or default? |
05:28.53 | lqdengr | no, let me try that |
05:28.56 | Sedorox | do that |
05:29.03 | Sedorox | then you should have access to that extnetion |
05:29.07 | Sedorox | which will allow you to dial out |
05:29.07 | Sedorox | :) |
05:29.27 | Eight | Dandan: do the who what? |
05:29.38 | Dandan | http://pastebin.ca - your configs |
05:30.11 | *** join/#asterisk spackle (~spackle@209.234.83.19) |
05:30.14 | Eight | Dandan: I already put mine into the wiki. |
05:30.18 | lqdengr | hmm weird, i think i have my exensions.conf file all screwed up or something. i put it under default, but still no go |
05:30.28 | Eight | Dandan: you don't want the rest of my configs, doesn't have anything to do with BV. |
05:30.35 | Dandan | Eight: ok |
05:30.45 | Sedorox | pastebin you extentions.conf.. just change your passwords |
05:31.02 | Dandan | or don't change them :D |
05:31.10 | Eight | I have only two lines in my extensions.conf that are relevant to bv |
05:31.17 | Eight | [from-broadvoice] |
05:31.19 | Eight | and |
05:31.27 | Eight | exten => s,1,goto(default,s,1) |
05:31.34 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
05:31.50 | Eight | Or are you talking about dialing out? |
05:33.08 | Eight | exten => _NXXNXXXXXX,1,Dial(SIP/${EXTEN}@sip.broadvoice.com) |
05:33.14 | Eight | That's a 10 digit US number. |
05:34.09 | opus___ | later.. |
05:34.27 | Eight | So... does everyone here have BV functioning? |
05:34.47 | Goshen | Hey Eight: there is only one thing you are missing in your register line...that is the extension to put the incoming call in to |
05:34.50 | Sedorox | then again.. I don't have BV :-p |
05:35.03 | Eight | Goshen: if you leave off the /extension, it just goes to s |
05:35.07 | Goshen | @sip.broadvoice.com/s on the end will put them to s |
05:35.12 | Sedorox | I actually just got a toll free number with link2voip... so gonna set that up now... |
05:35.18 | Eight | Goshen: you don't even need the /s |
05:35.33 | Goshen | so you really don't need from-broadvoice if you put them to to your default incoming context |
05:35.48 | Goshen | unless of course you just want to handle them differently |
05:36.15 | Goshen | something that I feel needs to be made clear to new users is the fact that you must 10 digit dial without the 1 always now |
05:36.18 | Goshen | like a cellphone |
05:36.27 | Dandan | Eight: i don't |
05:37.17 | Dandan | heh would like sbdy to help me step by step |
05:37.18 | Dandan | :/ |
05:37.25 | Dandan | it is waaaay too confusing for me... |
05:37.26 | Dandan | :/ |
05:37.35 | Eight | Goshen: really? 1 works fine for me. |
05:37.39 | Eight | Goshen: or did last night. |
05:37.40 | Goshen | Dandan: download Asterisk@Home |
05:37.55 | Dandan | Goshen: i have it on my desktop |
05:38.05 | Dandan | problem is i like to know it from the roots |
05:38.09 | Eight | Asterisk@home will DELETE YOUR HARD DRIVE> |
05:38.11 | Dandan | and that's how I learn |
05:38.15 | Goshen | Eight: that dial string you posted is without a 1 |
05:38.25 | Eight | Only install it on a machine you want to dedicate to asterisk |
05:38.28 | Sedorox | Eight: eh? |
05:38.29 | lqdengr | sedorox: sorry, took me a while to get it copied: http://pastebin.ca/7302 |
05:38.29 | Eight | Goshen: I have another with the 1 =) |
05:38.36 | Sedorox | lqdengr: thats ok |
05:39.05 | Eight | I actually have it setup to give an 'outside dialtone' for Broadvoice, in a different context. |
05:39.11 | Sedorox | what context do you have your phones connected to? |
05:39.17 | Dandan | so that's i would like to help me with BV priv if possible |
05:39.35 | Goshen | ~docs |
05:39.36 | 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 |
05:39.56 | Goshen | Dandan: spend a week or so reading :) |
05:39.57 | lqdengr | see, i'm not sure, in sip.conf i set it to outbound |
05:40.00 | Eight | Sedorox: well, SIP phones come into [sippers] which includes [default], and calls from broadvoice come into [from-broadvoice] which right now just kicks right to [default] |
05:40.05 | Dandan | i bought the yellow book too |
05:40.43 | Sedorox | Eight: hmm.. even tho I was referring to lqdengr :-p not sure what your thing was |
05:41.00 | Sedorox | lqdengr: can you dial the other extentions listed there? |
05:41.09 | Goshen | Dandan: this page helped me quite a bit when I was getting started.... http://iheavy.com/modules.php?op=modload&name=News&file=article&sid=35&mode=thread&order=0&thold=0 |
05:41.10 | Eight | Sedorox: oh, silly me... I was thinking you had 'eight' on the beginning of your comment =p |
05:41.15 | Goshen | it has a nice simple working config |
05:41.24 | jjg | can anyone tell me how to terminate a call through nufone from a laptop? |
05:41.26 | Sedorox | Eight: tis fine :-p what are you trying to do? lol |
05:41.30 | lqdengr | sedorox: nope, i sure cant |
05:41.31 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
05:41.34 | Goshen | and was the exact configuration I have...one FXO card, and a grandstream sip phone |
05:41.40 | Sedorox | ok... |
05:41.44 | Eight | Sedorox: I think I have everything I've tried so far working fine, actually =) |
05:41.45 | Sedorox | is that your entire extentions.conf? |
05:41.48 | lqdengr | yes |
05:41.53 | Sedorox | Eight: cool |
05:41.56 | Sedorox | lqdengr: weird.... |
05:42.17 | Sedorox | ok |
05:42.18 | Sedorox | do this |
05:42.35 | Sedorox | let me see if I can update it on the pastebin to make it easier |
05:42.35 | lqdengr | sedorox: this is my first * experiment, and i've been messing with that file for the past day or two getting inbound working correctly, so i must have screwed something up |
05:42.40 | lqdengr | ok |
05:43.04 | *** join/#asterisk santiago (~santiago@63.245.86.95) |
05:43.08 | Goshen | opus____: you still compiling? |
05:43.19 | Sedorox | lqdengr: everyone's gotta learn somehow.. like I said.. was in your posistion a few weeks ago... |
05:43.50 | lqdengr | sedorox: heh. well i appreciate it '-) |
05:43.55 | lqdengr | err. ;-) too |
05:44.43 | Dandan | ok till tomorrow :) |
05:44.44 | Dandan | [d] |
05:45.30 | bonez39 | Goshen: I have a linksys rt31p2 2 port phone adapter/router...I have flashed it once to get the newest firmware...wouldn't the fact that I can flash it suggest that the hardware is open..i.e., could be used with asterisk? |
05:46.39 | Goshen | bonez39: hey :) its possible...can you get in and change the server it connects to? |
05:46.44 | Sedorox | lqdengr: I see a few things... so you might just wanna copy this over your exsisting one.. |
05:46.48 | Sedorox | but save a compy |
05:46.50 | lqdengr | roger |
05:46.50 | Sedorox | copy* |
05:46.52 | Sedorox | to compare |
05:47.01 | Sedorox | and you have it as outgoing.. not outbound.. hence why it didn't work :-p |
05:47.04 | Sedorox | but anyway |
05:47.08 | lqdengr | lol! |
05:48.28 | lqdengr | so, did you update the copy on pb? |
05:48.37 | Sedorox | not yet... still checking it over |
05:48.43 | Sedorox | trying to do several things here.. so bear with me... |
05:48.56 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
05:49.08 | lqdengr | n/p |
05:49.38 | Sedorox | lets see if pastebin will ne nice |
05:49.39 | Sedorox | be* |
05:51.10 | Goshen | bonez39: do this...take screenshots of all of your current config, and print them |
05:51.18 | Goshen | so you can switch back easy |
05:51.46 | Sedorox | lqdengr: |
05:51.48 | Sedorox | http://pastebin.com/253643 |
05:52.04 | Goshen | bonez39: see this page http://www.voip-info.org/tiki-index.php?page=Linksys |
05:52.07 | Sedorox | and either remove context=, from your sip.conf |
05:52.11 | Sedorox | or set it to context=default |
05:53.00 | lqdengr | i'm guessing there's not supposed to be a php tag on the first line there |
05:53.05 | Sedorox | nooo |
05:53.10 | Sedorox | pastebin adds that |
05:53.13 | lqdengr | oh |
05:53.16 | Sedorox | you'll se the end tag at the end too |
05:53.24 | Sedorox | the ?> |
05:53.25 | Sedorox | is php ending |
05:53.26 | lqdengr | actually, i dont see one |
05:53.28 | lqdengr | ohh |
05:54.40 | jjg | can someone please help me make a call through nufone with a laptop? |
05:55.19 | lqdengr | hmm, nope i'm still not having any luck with it |
05:55.29 | Sedorox | did you do a reload on the console? |
05:55.35 | Essobi | Hmm. |
05:55.39 | lqdengr | i wonder if i could have some sort of config problem with x-lite. do you have to tweak any of the default settings? |
05:55.41 | rvhi | anyone knows how to run two instances of * on one server? |
05:55.42 | lqdengr | yes, i reloaded |
05:55.57 | Sedorox | rvhi: just have it bind to different addresses/ports? |
05:56.02 | Essobi | What kind of neet shit would you guys want from the management interface on a web control interface? |
05:56.06 | Sedorox | ummm.. for xlite.... |
05:56.52 | Sedorox | Essobi: adding/removing extentions, possibly uploading files for moh or background/playback, voicemail control (add/remove users, reset passes, etc) |
05:57.40 | lqdengr | when i try to place a call, should i see some activity on the console? |
05:57.54 | Sedorox | lqdengr: yes |
05:57.56 | Sedorox | if you have verbose on |
05:58.00 | lqdengr | strange |
05:58.03 | Sedorox | like I always start mine with -vvvc |
05:58.08 | Essobi | Ehh, MOH is easy. |
05:58.08 | Sedorox | in a screen session.. but anyway |
05:58.09 | lqdengr | ok, thats what i did |
05:58.15 | *** join/#asterisk ikey (~ikey@202.54.37.183) |
05:58.29 | lqdengr | and i turned on sip debug to see if that showed anything useful |
05:58.41 | Sedorox | and does it show anything? |
05:58.49 | lqdengr | but i don't see anything at all when i try to place a call, however * can call my softphone with no problems |
05:58.56 | lqdengr | so the sip connection is good... |
05:59.11 | Sedorox | ummmm |
05:59.12 | Sedorox | not all the time |
05:59.15 | ikey | hi can any one help me in configuring sip channels and to use it with grandstream ip phones |
05:59.15 | *** join/#asterisk Tarox (someone@pD9E7B0C0.dip.t-dialin.net) |
05:59.25 | Sedorox | does 'sip show peers' show your phone? |
05:59.30 | Sedorox | and IP and all that |
05:59.36 | lqdengr | sip show users does |
05:59.39 | Sedorox | ikey: its the same as anything else |
05:59.50 | lqdengr | yeah peers does as well |
05:59.58 | Sedorox | there is actually a example in there for a grandstream |
06:00.05 | Sedorox | lqdengr: hmmmm |
06:00.26 | *** join/#asterisk sleepy_one (~chatzilla@dhcp16632045.neo.rr.com) |
06:00.38 | Sedorox | double check your setting in xlite.. I think something is messed up where it isn't going to the server for outgoing calls... because your console should be floodded with stuff when you do anything on the sip channel. woith debug on |
06:00.42 | lqdengr | ikey, look in sip.conf, there's a good example you can build from |
06:00.53 | lqdengr | sedorox: k |
06:01.33 | sleepy_one | hello everyone |
06:01.33 | lqdengr | sedorox: k, i dont have any idea WHY, but it magically started working |
06:02.12 | Sedorox | ahah |
06:02.14 | Sedorox | dunno |
06:02.23 | Sedorox | computers are weird like that |
06:02.34 | sleepy_one | does anyone know how to get mpg123 working on FC3x86_64 ? |
06:02.36 | lqdengr | i clicked on the menu button in x-lite, then the console started exploding, so maybe it was just "stuck" |
06:02.49 | Sedorox | lol |
06:04.29 | Essobi | sleepy_one Umm. compile it? |
06:04.30 | Essobi | :) |
06:04.42 | sudhir492 | Does asterisk run on FC3, kernel 2.6.9? |
06:04.58 | lqdengr | it should |
06:05.01 | sleepy_one | Absolutely! runs very well thank you! |
06:05.26 | Goshen | asterisk works well on my 2.6.10 kernel |
06:05.44 | sudhir492 | Ok. when I try to compile asterisk, I get the following error: /usr/bin/ld: cannot find -lidn |
06:05.44 | sudhir492 | what is libidn? |
06:06.08 | sudhir492 | thanks |
06:07.00 | sudhir492 | Goshen: what distro are you running? |
06:07.09 | Goshen | Mandrake 10.0 |
06:07.22 | Goshen | with custom compiled kernel 2.6.10 |
06:07.39 | Nugget | linux is poo. |
06:07.43 | tuxinator_linuxM | make sure Makefile is looking for the right source/build files |
06:07.47 | Essobi | PEE! |
06:08.00 | sudhir492 | I just installed FC3. trying to compile asterisk but got stumped by lidn |
06:09.05 | sudhir492 | tuxinator_linuxM: Anything specific do you have in mind? |
06:09.18 | tuxinator_linuxM | Nugget: I don't think this is the right channel to bash on linux |
06:09.29 | Essobi | Why not? |
06:09.31 | Essobi | :) |
06:09.44 | tuxinator_linuxM | Have you read wiki pages on 2.6 |
06:09.56 | Nugget | doesn't change the fact that linux is poo. :) |
06:10.10 | Goshen | sudhir492: do you have everything listed on this page? http://www.asterisk.org/index.php?menu=download |
06:11.03 | nine76 | Hey all,I would appreciate anyone looking over my configs and giving me some clues as to why * says "No channel type registered for 'Zap'" . Thanks:-/ http://pastebin.ca/7303 |
06:12.02 | sudhir492 | I do, except bison devel, which I thought may not be necessary. |
06:13.06 | lqdengr | Is there any way to set outbound caller id info with voicepulse? My calls show up as some New York phone # instead of my actual #. |
06:16.51 | Sedorox | how many calls can a voice T1 support? |
06:16.58 | lqdengr | 23 for pri |
06:17.00 | lqdengr | 24 for t1 |
06:17.05 | Sedorox | ah thats right... |
06:17.21 | Sedorox | what do you think it is if it was d data T1 with VoIP |
06:17.24 | Sedorox | around 10? |
06:17.34 | lqdengr | no, a lot more than that i'd say |
06:17.40 | lqdengr | i think they say about 50kbps/call |
06:17.47 | lqdengr | so 1544 / 50 |
06:17.53 | Sedorox | thats 25 in and out... |
06:18.00 | Sedorox | so yea... |
06:18.03 | lqdengr | about 31, minus tcp overhead, etc. |
06:18.09 | *** join/#asterisk ikey (~ikey@202.54.37.183) |
06:18.10 | Goshen | nine76 did you modprobe your modules? |
06:18.24 | nine76 | Goshen: Yes,no errors. x100p card |
06:18.26 | ikey | hi can any one help in configuring sip channels in asterisk |
06:18.35 | Sedorox | ikey: what do you need? |
06:18.50 | Goshen | nine76: wcfxo? |
06:19.01 | nine76 | yes,I am doing it again to be completely sure:) |
06:19.13 | nine76 | I did zaptel,wcfxs and wcxfo |
06:19.25 | Goshen | why wcfxs? |
06:19.32 | Goshen | that isn't right for a x100p |
06:19.43 | nine76 | Was following directions on "getting started with asterisk" guide |
06:19.57 | Goshen | where? |
06:20.10 | nine76 | http://www.automated.it/guidetoasterisk.htm#_Toc49248763 |
06:20.28 | Goshen | wrong card...that is for a TDM400P Installation |
06:20.31 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
06:20.35 | nine76 | I also tried others configs, as I noticed they varied. Gettign started guide had "channel=1" while an onlamp article had channel => 1 |
06:20.58 | Goshen | get rid of the wcfxs module |
06:21.16 | Goshen | x100p only has fxo |
06:21.44 | Goshen | nine76: http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf |
06:22.20 | Goshen | also, did you run ztcfg? |
06:22.47 | lqdengr | anyone know if you can do ivr from within your voicemail greeting? i'd like to give people the option of pressing 9 to have * connect them to my cell |
06:23.06 | sudhir492 | There was a symbolic link for libidn.so in /usr/lib directory. After creating the link, asterisk builds fine |
06:23.21 | sudhir492 | I mean sym link was missing |
06:23.39 | Goshen | lqdengr: I think there is an option to dial 0 for the operator |
06:23.44 | nine76 | I removed wcfxs, restarted *, still no chan type.. ztcfg --v output is at bottom of pastebin. exited ok saying 1 chan configured succesfully |
06:24.10 | nine76 | dmesg also reports wilcard there |
06:24.20 | lqdengr | goshen: ok. thanks |
06:24.29 | sudhir492 | lqdengr: do the ivr first before running voicemail |
06:24.47 | xkev | zaptel pri gurus in here tonight? |
06:25.05 | xkev | problem with facility message failing to handle caller-name |
06:26.57 | lqdengr | sudhir493: duh, thanks for pointing out the obvious to me. :-) /me rethinks getting another beer |
06:27.19 | Essobi | Mmm. |
06:27.34 | Essobi | How is * going to interact with a T1 card with DSPs on it? |
06:27.50 | Goshen | any suggestions for a gsm player for windows? |
06:28.10 | Essobi | That'd be a massive code change ehh? DTMF detection, silence detection, tone playback, IVR playback.. |
06:28.14 | Eight | Goshen: quicktime |
06:28.16 | lqdengr | Quicktime |
06:28.23 | *** join/#asterisk jeffik (~jeffik@m9f7236d0.tmodns.net) |
06:29.30 | xkev | "Do not handle argument of type 0x80" |
06:30.32 | Eight | xkev: http://www.voip-info.org/wiki-PRI |
06:31.09 | xkev | this is 0x80 |
06:31.15 | xkev | not 0x84 |
06:31.20 | Eight | ah, I see that. |
06:31.21 | Eight | my bad. |
06:31.24 | xkev | heh |
06:31.49 | xkev | but I just restowed some older cvs, and I bet it's my fault for something I did in zapata.conf, since it didn't fix it (twas working) |
06:33.46 | xkev | oh wait, I just rolled asterisk back not zaptel |
06:34.23 | shido6 | boink |
06:34.35 | JmanA9 | well, i got asterisk working :) |
06:34.43 | JmanA9 | i just deleted every folder that had the word asterisk in it and reinstalled :) |
06:34.57 | shido6 | hehehe |
06:35.39 | Sedorox | find / -name *asterisk* | rm -rf |
06:35.40 | Sedorox | :-p |
06:35.53 | JmanA9 | it worked :) |
06:36.10 | Sedorox | find / -name * | rm -rf |
06:36.13 | Sedorox | O:-) |
06:36.32 | Eight | I knew *someone* was going to have to give the command to accomplish that =p |
06:37.02 | Sedorox | ahahah |
06:37.15 | Sedorox | eh.. I prefer dd if=/dev/null of=/dev/hda |
06:37.15 | ikey | hi can any one help in configuring sip channels with grandstream phones |
06:37.56 | Eight | Sedorox: /dev/random =) |
06:38.02 | Sedorox | hehe |
06:38.19 | Goshen | ikey: http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone |
06:38.56 | Eight | We need jbot to send a welcome /msg to people as they join. |
06:39.01 | Eight | listing stuff like ~docs |
06:39.17 | Eight | jbot welcome? |
06:39.20 | Sedorox | Goshen: I think he has the channel on mute or something.. he's asked three times.. and each gotten a answer |
06:39.42 | Goshen | hmm |
06:40.59 | Eight | Anyone have an SRV record for their server? |
06:41.17 | Nugget | at least one person does, yes. :) |
06:41.42 | Eight | Nugget: =p |
06:41.45 | Eight | lol... whoops. |
06:41.55 | xkev | eight I do yeah |
06:41.58 | Eight | this song on Philadelphonic starts with a phone ring. |
06:42.08 | Eight | I was like "WTF?! Why is my computer ringing?! I didn't do anything!" |
06:42.22 | Goshen | lol |
06:42.35 | Nugget | there's a song by the cure that has a noise in it that sounds exactly like my alarm clock. it makes me grumpy whenever I hear it |
06:42.35 | xkev | you've been sucked into the void for too long |
06:42.58 | Eight | xkev: Mind if I verify my Asterisk install is properly doing SRV lookups for SIP addresses? |
06:43.28 | xkev | sure, try kevin@pbx.xmission.com, you'll just get my voicemail though |
06:44.02 | Eight | uh. |
06:44.13 | Eight | you only have A and NS records... |
06:44.24 | Nugget | slacker.com has a srv record for sip. |
06:44.33 | Nugget | try nugget@slacker.com if you want |
06:44.40 | xkev | _sip._udp.pbx.xmission.com. 3600 IN SRV 1 0 5060 pbx.xmission.com. |
06:44.42 | xkev | I do? |
06:44.53 | Eight | hmmm, been a while since I've used dig =) |
06:44.56 | xkev | pbx.xmission.com. 3600 IN NAPTR 2 0 "s" "SIP+D2U" "" _sip._udp.pbx.xmission.com. |
06:44.57 | Nugget | heh |
06:45.04 | Goshen | a fellow utahn :) |
06:45.20 | xkev | hola! |
06:45.34 | xkev | you in utopia territory? |
06:45.41 | Goshen | xkev: unfortunately NO! |
06:45.45 | Goshen | I want fiber to my home! |
06:45.48 | xkev | me either |
06:46.01 | xkev | we're connecting up in a few months (XM is) |
06:46.14 | Goshen | XM? |
06:46.23 | xkev | xmission |
06:46.43 | xkev | dig SRV _sip._udp.foo.bar @ns |
06:47.26 | xkev | goshen, just IP data at first, but we'll do voip and video on demand eventually |
06:47.38 | Goshen | very nice |
06:47.45 | Eight | xkev: thanks... I thought I could get it without being so specific =) |
06:48.01 | xkev | that's what naptr does |
06:48.18 | xkev | such a kludge :) |
06:48.51 | Eight | I've got a friend who's into HAM radio and stuff... |
06:49.03 | Eight | Pondering how we might assemble some neat toys. |
06:49.13 | Nugget | I don't understand how people can be interested in ham radio now that the internet exists. |
06:49.25 | Eight | Nugget: the internet isn't everywhere =) |
06:49.50 | Nugget | It's everywhere I want to be. :) |
06:49.57 | xkev | the internet exists on ham too |
06:49.58 | Eight | Nugget: It's not everywhere I want to be. |
06:50.00 | xkev | packet radio |
06:51.11 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
06:53.47 | jeffik | packet radio like GPRS |
06:54.18 | xkev | slow and shitty, exactly. :) |
06:55.06 | Eight | xkev: kevin@pbx.xmission.com is supposed to go to voicemail? |
06:55.15 | xkev | it'll ring a few times first |
06:55.18 | jeffik | yes slow and shitty but available where no wifi or dsl/cable is availablre |
06:55.34 | Eight | jeffik: packet radio can work where GPRS isn't available. |
06:55.39 | Eight | you can MAKE it available where YOU want it. |
06:55.45 | Eight | as opposed to where the telco feels like it. |
06:56.01 | xkev | eight, but maybe there's some proxy auth getting in the way |
06:57.49 | bonez39 | Goshen: have any idea how soon you could order fiber? or the cost? |
06:58.08 | xkev | erm, seems to work for me unauth |
06:58.39 | *** join/#asterisk criptos (~criptos@201.135.97.238) |
06:58.55 | xkev | res_search OK (len=242) |
06:58.55 | xkev | NAPTR: _sip._udp.pbx.xmission.com |
06:58.55 | xkev | res_search OK (len=197) |
06:58.55 | xkev | SRV: 1,0,5060 |
06:58.55 | xkev | SRV: pbx.xmission.com |
06:58.55 | xkev | SipClient: Sending to 'pbx.xmission.com:5060' |
06:59.09 | criptos | Can I use a fxo port and a sintax like Zap/1/1234 as an agent for a queue? |
06:59.14 | Eight | ya, I'm not sure I've got my end configured very well atm. |
06:59.25 | xkev | (that's kphone output, btw) |
07:00.10 | lqdengr | anyone know how to make x-lite tell * to play hold music when you put someone on hold? |
07:00.25 | slePP | has anyone done load testing on asterisk/PRI to see how many calls/second you can drive through it? |
07:00.29 | xkev | * plays moh when it gets a sendonly |
07:00.30 | Eight | <PROTECTED> |
07:00.30 | Eight | <PROTECTED> |
07:00.30 | Eight | <PROTECTED> |
07:00.30 | Eight | <PROTECTED> |
07:00.30 | Eight | <PROTECTED> |
07:00.31 | Eight | <PROTECTED> |
07:00.33 | Eight | There's mine =p |
07:00.41 | Eight | But I don't hear a voicemail announcement. |
07:00.47 | xkev | do you hear ringing? |
07:01.01 | Eight | ya, I got the ringing, and the ringing stopped when it picked up. |
07:01.05 | Eight | then nothing. |
07:01.08 | Newbie___ | slePP: i dont think x-lite is capable of doing music on hold |
07:01.09 | xkev | odd |
07:01.20 | xkev | try music@pbx.xmission.com |
07:01.20 | Eight | I think it's a reinvite problem. |
07:01.26 | Eight | I get a click. |
07:01.35 | slePP | Newbie___: i didn't ask :> |
07:01.56 | Eight | ah, i'm an idiot. |
07:02.01 | Newbie___ | slePP: sorry wrong person |
07:02.04 | lqdengr | lol |
07:02.04 | Newbie___ | hehe |
07:02.04 | Eight | at some point I commented out my disabling of reinvite. |
07:02.09 | Eight | forgot to put it back =p |
07:02.17 | slePP | okay, who here hasn't entered the pastebin draw yet? |
07:02.18 | xkev | I had to reinvite=no for the polycoms to behave |
07:02.22 | slePP | http://pastebin.ca/draw.php |
07:02.23 | slePP | go do so :> |
07:02.25 | xkev | ..only had a problem on manager redirect though |
07:02.43 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
07:02.49 | Newbie___ | hmmm let me see who asked that question earlier |
07:03.04 | slePP | and no, X-Lite doesn't provide MoH, the server would do that for you instead |
07:03.31 | Eight | xkev: heh, where is this music feed from? |
07:03.31 | Newbie___ | i use firefly, mp3 music on hold |
07:03.47 | xkev | eight, we have like 50 hours of hold music |
07:04.13 | xkev | from our collections of downtempo and ambient. there's a dash of sesame street here and ther etoo |
07:04.36 | xkev | I haven't counted :) |
07:04.40 | slePP | xkev: i use Elmo singing the alphabet for the support queue |
07:04.51 | Eight | well, asterisk is using SRV properly, and I didn't even have to screw with it for 37 hours =) |
07:05.05 | xkev | you need the drunken russian sailors singing 'my heart will go on' |
07:05.20 | slePP | xkev: that's in my personal collection, i bet ;> |
07:05.25 | xkev | eight, now I need someone with e164org to test with |
07:05.40 | slePP | e164.org! |
07:05.42 | slePP | thassme |
07:05.47 | slePP | well, i'm part of it :> |
07:05.48 | xkev | word2u |
07:05.54 | Eight | xkev: what did you need to configure to get *incoming* SRV to work like it just did with me? other than the DNS record of course. |
07:06.01 | xkev | dns record |
07:06.06 | slePP | though i haven't seen evilbunny in a while |
07:06.31 | Eight | xkev: doesn't there need to be a 'guest' user for random schmucks off the 'net? |
07:06.37 | xkev | the dns just tells the calling UA where to send its packets |
07:06.38 | slePP | xkev: 8829900003305 |
07:06.41 | xkev | oh yeah, that |
07:07.04 | slePP | of course, there's a chance those numbers are pointing at the wrong spot atm |
07:07.25 | xkev | I have context=anonymous in sip.conf [general] |
07:07.27 | slePP | yup, they are... |
07:07.31 | slePP | they're pointed at the old server |
07:07.51 | xkev | and I created an [anonymous] with stuff I let people call, including some realtime derived from staff email addresses, etc |
07:08.24 | nine76 | whome was it that came in and mentioned "I got it to work by deleting every folder found for a search on "asterisk" and recompiled,then it worked". Thanks to that guy:) |
07:08.37 | Eight | nine76: hah. |
07:08.44 | nine76 | didnt wanna have to redo so many configs,but it works,so hapiness |
07:09.12 | slePP | okay, i'm done. time for beer'n'TV |
07:09.13 | xkev | I've got like 12 CVS versions under /usr/local/stow, and not a single problem with version skew |
07:09.13 | Eight | xkev: so there's nothing special to set, I'm already accepting anonymous connections? |
07:09.15 | lqdengr | anybody have an opinion on switchvox? |
07:09.33 | xkev | eight, the default context in sip.conf handles that yeah |
07:09.41 | Eight | xkev: ah. |
07:09.57 | xkev | ..I think :) |
07:10.03 | xkev | mine's all over the map |
07:10.07 | Eight | hah. |
07:10.37 | xkev | yeah, I bet that's what does it |
07:11.00 | xkev | since the type=user entry for my SER proxy (where you really send port 5060 on this box) is typoed :) |
07:12.13 | xkev | (btw, all ser does is send presence subscribes to the right place, pretty much) |
07:14.54 | jeffik | anybody using asterisk@home? |
07:15.08 | xkev | I have a box warming the basement, yeah |
07:15.34 | xkev | case ROSE_NAME_PRESENTATION_ALLOWED_SIMPLE: |
07:15.35 | xkev | memcpy(call->callername, comp->data, comp->len); |
07:15.45 | xkev | default: |
07:15.51 | xkev | pri_message("Do not handle argument of type 0x%X\n", comp->type); |
07:16.08 | xkev | libpri/pri_facility.h:#define ROSE_NAME_PRESENTATION_ALLOWED_SIMPLE 0x80 |
07:16.16 | xkev | "Do not handle argument of type 0x80 |
07:16.18 | Eight | Someone feel like testing something for me? |
07:16.24 | xkev | WTF?!?!? |
07:16.58 | Eight | xkev: hah. |
07:17.34 | Eight | xkev: I had a similar paradox earlier with /var/run/asterisk.pid |
07:17.44 | Eight | I'm damned sure I know where I'm telling it to put it. |
07:17.46 | Eight | and it damned sure isn't. |
07:19.07 | xkev | I can test somehin for ya |
07:19.18 | Eight | call weasels@3.141592.net |
07:19.29 | xkev | that's a hell of a domain name |
07:19.32 | Eight | =) |
07:19.38 | ikey | Goshen: saw the documents but these are for the advanced sip phone configurations |
07:19.50 | Eight | There's no SRV, that should use the A rec. |
07:19.50 | xkev | res_search: NO result ! |
07:20.24 | sleepy_one | gnite y'all |
07:20.36 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
07:20.39 | ikey | Goshen: i need to check with basic configurations |
07:20.53 | PTG123 | Does ANYONE know how rto reset the admin password on a polycom? |
07:21.03 | xkev | eight, it figured it out, but it was slower not having the srv |
07:21.04 | Eight | xkev: you heard the weasels message? |
07:21.06 | ikey | can any one help me in configuring sip channels? |
07:21.30 | xkev | but kphone being a total piece of crap is now deadlocked :) |
07:21.40 | Sedorox | ikey: there is a TON of examples between the sip.conf and the wiki |
07:21.41 | Eight | xkev: cool, I just wanted to make sure I knew I was 'catching' the calls properly. |
07:21.54 | Eight | xkev: I'll poke the guy who runs my DNS records about it. |
07:22.13 | PTG123 | Someone mus t know how to use the polycom phones :) |
07:22.50 | xkev | ptg I haven't even changed my admin pass |
07:23.09 | PTG123 | xkev: damn :( |
07:23.24 | xkev | I figure, if they want to get in and screw up their phone, then they can have no phone :) |
07:23.32 | *** join/#asterisk TauReX (~james@colossus.trustmatta.com) |
07:23.41 | PTG123 | xkev: can you set the admin password from the tftp server? |
07:23.44 | PTG123 | when it does its update? |
07:24.00 | xkev | not that I've found |
07:24.02 | xkev | I wish I could |
07:24.20 | PTG123 | does the tftp stuff specify all the sip settings etc though? |
07:24.27 | xkev | yeah |
07:24.39 | xkev | but I don't tftp, I use ftp (for the polycoms anyway) |
07:24.39 | PTG123 | can you give me a copy of your setting files? Maybe i can just set it all up that way |
07:24.55 | PTG123 | any way to know which its set up to use |
07:25.00 | PTG123 | what does the polycom specify for the login and password |
07:25.01 | PTG123 | when you ftp? |
07:25.28 | xkev | PlcmSpIp for both, is default |
07:25.41 | xkev | and it'll get the server from dhcp op 66 (tftp server name) |
07:25.46 | PTG123 | whats the default for the web iface? |
07:25.55 | xkev | Polycom and 456 |
07:26.02 | PTG123 | yah i think i tried that |
07:26.08 | PTG123 | well so if i make a dummy ftp server |
07:26.11 | PTG123 | and let it get the files |
07:26.15 | PTG123 | it should fix my problem? |
07:26.15 | xkev | Polycom and your admin pass :) |
07:26.21 | PTG123 | can you specify the tftp server to use in the feature? |
07:26.29 | xkev | 'the feature'? |
07:26.54 | rvhi | anyone knows how to have 2 instances of * on one server? |
07:27.15 | xkev | rvhi, if you ever figure it out, let me know :) |
07:27.17 | Sedorox | rvhi: just have another bind to different IPs/ports? |
07:27.36 | xkev | you'll have to keep all sorts of things from colliding |
07:27.53 | rvhi | quite a few things use sock |
07:27.59 | *** join/#asterisk jmhunter (~jmhunter@64.77.200.148) |
07:27.59 | *** mode/#asterisk [+o jmhunter] by ChanServ |
07:28.09 | PTG123 | http://eknowledge.polycom.com/SRVS/CGI-BIN/WEBCGI.EXE/,/?St=14,E=0000000000001066275,K=1150,Sxi=3,Case=obj(34787) |
07:28.15 | PTG123 | do those instructions look like they would work xkev? |
07:28.15 | rvhi | they won't work well with 2 instances |
07:28.23 | xkev | make an /etc/asterisk2 and copy everything, change things to suit, and use asterisk -C /etc/asterisk2/asterisk.conf |
07:28.54 | rvhi | does voicemail support vacation greating? something similar to the email vacation setting. |
07:29.07 | rvhi | greeting... |
07:29.38 | xkev | main nasty collides would be /var/run/asterisk/asterisk.ctl and /var/lib/asterisk/astdb |
07:30.14 | xkev | but if you do some fancy symlinking, you should be able to juust specify alt dirs in asterisk.conf |
07:30.33 | rvhi | can you specify the name in two different *.conf? |
07:30.36 | xkev | rvhi, CVS has a temporary greeting option |
07:30.55 | rvhi | then load * with different .conf? |
07:30.55 | ikey | hi can anyone help us out in configuring sip channels |
07:31.12 | xkev | rvhi, asterisk -C /etc/asterisk2/asterisk.conf |
07:31.25 | xkev | I haven't done this yet, but I will |
07:31.32 | rvhi | cvs? i tried to avoid it as much as i can |
07:31.55 | rvhi | can i just copy the app_voicemail.c to 1.0.7 |
07:31.58 | rvhi | would that work? |
07:32.16 | xkev | I need some public voip<->pstn and I don't want to wedge that simple setup into my 1500 line office pbx dialplan |
07:32.26 | jjg | is there no way to terminate through nufone via laptop????????? |
07:32.28 | xkev | rvhi, no |
07:32.42 | xkev | you can look for the patch on mantis or asterisk-cvs list and try to backport it |
07:33.16 | xkev | stable has many many many major differences (such as flags and the channel struct) |
07:33.17 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
07:33.50 | rvhi | why wouldn't anyone backport these new features? realtime and voicemail? |
07:33.56 | xkev | e.g. chan->callerid in stable == chan->cid.num and chan->cid.name in cvs |
07:34.06 | PTG123 | anyone know polycom phones |
07:34.19 | xkev | because cvs doesn't suck, and it has lots of things that make you hate stable :) |
07:34.29 | xkev | or rather, things you can't do in stable |
07:34.38 | xkev | ptg123 lookin at your url |
07:35.04 | jjg | PTG123 : lots of luck...i thoght termination through nufone via laptop wold be an easy question |
07:35.18 | PTG123 | jjg: whats you prob with that? :) |
07:35.30 | jjg | prob is , how to do it |
07:35.54 | jjg | does nufone support SIP? not that i'm aware of |
07:35.57 | PTG123 | thats why it sbetter to use companies with good support who would help with those things :) |
07:35.57 | xkev | ptg123, on your link: try it |
07:36.08 | PTG123 | xkev: i don't see those menus |
07:36.11 | PTG123 | xkev: are there? |
07:36.17 | rvhi | when is cvs becoming the new stable? is it on the horizon? or i have no hope now? |
07:36.37 | xkev | you're looking at a doc for VSX, not SoundPoint IP |
07:36.44 | xkev | I don't have a phone at home today |
07:37.08 | jjg | PTG123 : what is your point in asking the problem without offering someing like a solution? |
07:37.12 | xkev | but look for a menu item that is reset factory defaults, I think I recall one being there |
07:38.00 | xkev | rvhi, I wish I knew. I'm going live with my CVS after exhaustive testing and minor bug fixes (which incidentally, this latest bug exists in stable too, heh) |
07:38.01 | PTG123 | jjg: b/c i am not gonna take you step by step how to do something the only reason you can't get it doen is b/c you were too cheap to spend an extra .5c :) |
07:38.15 | jjg | doesn't it kinda mean that nufone sucks if there isnt a softphone? |
07:38.17 | PTG123 | xkev: can't find it, sucks.. |
07:38.31 | PTG123 | jjg: well for a softphone use xlit |
07:38.31 | jjg | PTF123 : don't follow, sorry |
07:38.38 | jjg | through nufone? |
07:38.39 | PTG123 | i use xpro + bluetooth headset |
07:38.40 | xkev | ..but I must have cvs for some features |
07:38.44 | PTG123 | but nufone doesn't support sip directly |
07:38.54 | jjg | well like i said |
07:38.54 | PTG123 | use firefly |
07:39.01 | jjg | i didn't say nufone + asterisk |
07:39.02 | PTG123 | but like i said |
07:39.03 | xkev | ..for a simple pstn gateway with some voicemail and a feature or two, I'd use stable |
07:39.11 | PTG123 | i don't think nufone is the best thing to use for that |
07:39.23 | PTG123 | find a good cvs build, and stick with it |
07:39.26 | PTG123 | thats what i did |
07:39.28 | jjg | ...s/nufone/laptop |
07:39.57 | xkev | ptg, I keep patching so I have to keep updating to get my features committed :) |
07:40.01 | jjg | so there is no fucking way to terminate through nufone without an asterisk box? jesus |
07:40.27 | PTG123 | xkev: their doesnt seem much interested in submitting my features.. so i don't wanna waste the time even committing them :) |
07:40.39 | PTG123 | jjg: i gave you the answer |
07:40.46 | PTG123 | jjg: but this is #asterisk not #nufone :) |
07:40.49 | jjg | uh, and the answer was? |
07:40.52 | PTG123 | so why would anyone want to help you |
07:40.53 | jjg | i'm on #nufone |
07:40.58 | jjg | they have shit to say too |
07:41.06 | PTG123 | PTG123: but nufone doesn't support sip directly |
07:41.06 | PTG123 | PTG123: use firefly |
07:41.17 | PTG123 | use teliax, i know that guy would help you out :) |
07:41.36 | jjg | and firefly is a protocol?...how does firefly relate to sip? |
07:41.43 | PTG123 | ~firefly |
07:41.44 | jbot | somebody said firefly was http://virbiage.com/firefly/download/firefly-thirdparty.exe |
07:41.46 | Eight | <PTG123> find a good cvs build, and stick with it <-- That's what releases are supposed to be! =) |
07:41.47 | PTG123 | since nufone doesn't support sip |
07:41.51 | PTG123 | has nothing to do with it |
07:41.57 | xkev | eight, we need a 1.1 yes |
07:42.04 | PTG123 | ~firefly |
07:42.06 | jbot | from memory, firefly is http://virbiage.com/firefly/download/firefly-thirdparty.exe |
07:42.23 | xkev | I'd like to see a freeze in the timeline |
07:42.29 | Eight | jbot jbot? |
07:42.30 | jbot | jbot is probably the shipboard computer, but you may call me eddie if it helps you relax |
07:42.41 | Eight | eddie eddie? |
07:42.42 | Eight | =p |
07:42.45 | shido6 | whats up? |
07:42.51 | `Sauron | Eight, do you never sleep? |
07:42.54 | shido6 | watching the Detroit / Boston Game |
07:43.03 | Eight | `Sauron: =) |
07:43.26 | ikey | jjg: can u help me out in configuring sip channels for outbound |
07:43.59 | jjg | wuss the issus? |
07:45.08 | *** join/#asterisk file (~file@251.134.218.209.transedge.com) |
07:45.13 | jjg | besides, ikey |
07:45.35 | jjg | you told me earlier today that you had 120 simultaneous calls going on a single CPU |
07:45.41 | jjg | so what's the problem? |
07:46.07 | jjg | forgot how you got all those calls rolling? |
07:46.55 | shido6 | ok back , timeout |
07:48.11 | rvhi | is there a way to define the extension to reach after user presses 0 in voicemail? |
07:48.29 | rvhi | e.g. ceo's voicemail, press 0 goes to secretary |
07:48.41 | rvhi | other people, press 0 goes to receptionist |
07:49.23 | rvhi | the special extension 'o' is reached after 0 is pressed |
07:49.41 | rvhi | how to i know if the original caller is in which email? |
07:50.30 | *** join/#asterisk porkchop (~porkchop@porkchop.nat.cccp.porkchop.net) |
07:51.12 | porkchop | I'm having a problem with callerid. No matter what I do, inbound calls appear on my screen as "porkchop", the username for the phone. Any ideas? I'm not using fromuser= (anymore) in sip.conf .... |
07:51.16 | JmanA9 | can anyone recommend a good asterisk gui? |
07:51.39 | PTG123 | can you buy 48v adapters in stores? |
07:51.41 | JmanA9 | one for changing config files and stuff, not a softphone ;) |
07:52.22 | ikey | jjg: yeah i think i told u that on PSTN e1 Lines |
07:52.33 | shido6 | double overtime |
07:52.44 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
07:52.56 | ikey | jjg: so was trying to work on SIP too for outbound calls |
07:53.08 | jjg | ok |
07:53.15 | PTG123 | JmanA9: there really isn't a good one publically available |
07:53.17 | jjg | it's easier than outboudn pstn |
07:53.20 | jjg | what's the prob? |
07:53.33 | JmanA9 | ok, editing config files it is :) |
07:53.58 | Eight | JmanA9: it's just as well... it means you can SSH in and do everything as normal =) |
07:54.01 | ikey | jjg: yeah i tried with grandstream and configured sip.conf and extensions.conf |
07:54.04 | *** join/#asterisk jeffik (~jeffik@me97b36d0.tmodns.net) |
07:54.48 | ikey | jjg: asterisk can be configured as sip client as well as sip server ...i think there was some confusion when i configured |
07:55.00 | jjg | ikey : i'm sorry for the sarcasm, i'm totally exhausted ... pastebin yrou stuff and i'll chekc it in a coupla hours...private message me |
07:55.20 | ikey | jjg : ok great |
07:55.25 | ikey | will do that now it self |
07:58.06 | shido6 | hehe |
08:01.04 | PTG123 | a sip client is a sip server, really no difference :) to be one you need to be the other |
08:01.09 | rvhi | xdev, i looked the cvs app_voicemail.c code |
08:01.20 | rvhi | can't find anything about vacation greeting |
08:03.42 | *** join/#asterisk eric_ (Ap0ll0@modemcable081.176-201-24.mc.videotron.ca) |
08:03.57 | eric_ | anyone awake on this friday night? |
08:04.18 | Eight | eric_: nope. |
08:04.33 | jeffik | eric_: only those who need help |
08:05.13 | tuxinator_linuxM | I am |
08:05.20 | eric_ | haha |
08:05.25 | eric_ | nice to see some other insomniacs out there |
08:05.44 | porkchop | insomnia? Its only 314am |
08:05.50 | porkchop | :) |
08:05.52 | jeffik | anyone work with *@home? |
08:05.53 | eric_ | :P |
08:05.53 | tuxinator_linuxM | What does the sun look like? |
08:05.54 | Eight | mmm... pi. |
08:05.58 | eric_ | yeah, so basically, I compiled asterisk, and am using default config, but when i start it up, it segfaults |
08:06.10 | Eight | eric_: lol. |
08:06.10 | tuxinator_linuxM | jeffik: I hear it works well |
08:06.11 | Eight | eric_: nice =) |
08:06.13 | porkchop | tuxinator_linuxM: you mean...the day star? |
08:06.14 | eric_ | does anyone have any pointers where to start? google hasnt found me anything useful :/ |
08:06.17 | eric_ | thanks eight :P |
08:06.32 | tuxinator_linuxM | eric_: hardware? |
08:06.36 | tuxinator_linuxM | What is your |
08:06.47 | tuxinator_linuxM | OS? |
08:06.49 | Eight | eric_: It sounds like you have problems that aren't terribly related to asterisk... is my first (uneducated) guess. |
08:07.03 | eric_ | well, its all emulated hardware, its RHEL running under UML |
08:07.06 | drumkilla | tuxinator_linuxM: how was training |
08:07.12 | Eight | eric_: well there ya go =) |
08:07.16 | eric_ | someone else is doing the same thing and its working for him though |
08:07.24 | Eight | ok, I'll just shut upnow. |
08:07.28 | tuxinator_linuxM | drumkilla: well, good and bad... |
08:07.30 | Zeeek | eric_ you have zaptel hardware? |
08:07.30 | eric_ | haha dont shut up |
08:07.49 | Eight | Alright, to heck with extensions.conf |
08:07.53 | eric_ | no hardware being used -- its at a remote location... im just going to use it to route |
08:07.55 | tuxinator_linuxM | drumkilla: Good cuz I reallize I know * pretty well... |
08:08.00 | drumkilla | :) |
08:08.10 | Eight | exten => _.,1,agi,everything.py ; =p |
08:08.11 | Zeeek | eric_ when you start do yiou get anything before the fault? any messages? |
08:08.15 | tuxinator_linuxM | drumkilla: Badd cuz I wasted $200 to learn that I already know |
08:08.17 | Newbie___ | can anyone please give me a 1800 number to try out the sound quality, i am not from the US |
08:08.17 | eric_ | no messages at all |
08:08.32 | drumkilla | tuxinator_linuxM: well it sounded like it was intended for people that didn't know anything about it |
08:08.33 | jeffik | *@home uses amp gui |
08:08.35 | eric_ | but i can do asterisk ---nothing, and it tells me "error, no option" |
08:08.39 | Zeeek | sounds like it may be mobo related - fcertain chipsets exhibit this behavior |
08:08.56 | eric_ | Zeeek: me? all my hardware is basically emulated |
08:09.02 | tuxinator_linuxM | eric_: use -vvv |
08:09.12 | eric_ | thats what im doin :P |
08:09.17 | tuxinator_linuxM | k |
08:09.21 | eric_ | hehe |
08:09.23 | tuxinator_linuxM | noting in log fiel? |
08:09.28 | eric_ | nothing at all |
08:09.31 | eric_ | core dump actually :P |
08:09.42 | Newbie___ | any 800, 888 number i can test dial to the US ? |
08:09.56 | eric_ | 8009993355 -- its dell ;) |
08:10.09 | Newbie___ | tks |
08:10.26 | eric_ | i'm stumped... i have no idea where to look first |
08:10.36 | jeffik | any canadians on tonight |
08:11.01 | eric_ | anyone else have any delightful insight? |
08:11.16 | PTG123 | anyone awake now know anything about polycoms? :) |
08:11.24 | PTG123 | drumkilla: you don't by chance? |
08:11.29 | tuxinator_linuxM | PTG123: their good |
08:11.47 | Newbie___ | trying out simpleconnect.com with firefly IAX, couldnt hear a thing calling 800 999 3355 |
08:11.51 | PTG123 | tuxinator_linuxM: little more info then that :) |
08:11.55 | jeffik | no canadians? |
08:12.00 | shido6 | whats wrong with the polycom? |
08:12.09 | shido6 | [02:52] <shido6> big project |
08:12.09 | shido6 | [02:52] <shido6> heading to bed |
08:12.09 | shido6 | [02:52] <shido6> greg@nufone.net |
08:12.09 | shido6 | [02:52] <shido6> IM: shido6@msn.com |
08:12.11 | PTG123 | shido6: i can't find out how to reset admin password |
08:12.12 | shido6 | crap |
08:12.18 | shido6 | then |
08:12.19 | shido6 | err |
08:12.24 | shido6 | put next server in the dhcp server |
08:12.24 | PTG123 | shido6: any idea? |
08:12.27 | shido6 | as your tftp box |
08:12.29 | tuxinator_linuxM | PTG123: I havevn't played with them yet |
08:12.41 | PTG123 | shido6: what do you mean? |
08:12.43 | shido6 | and reboot that summu mumma snitch |
08:13.02 | shido6 | err |
08:13.04 | shido6 | http://lists.digium.com/pipermail/asterisk-users/2004-October/069585.html |
08:13.06 | shido6 | try that |
08:14.08 | PTG123 | in the cfg files can you specify admin pwrd? |
08:16.08 | shido6 | stdby |
08:17.45 | porkchop | I'm having a problem with callerid. No matter what I do, inbound calls appear on my screen as "porkchop", the username for the phone. Any ideas? I'm not using fromuser= (anymore) in sip.conf .... |
08:18.10 | tuxinator_linuxM | porkchop: I think it is a bug |
08:18.26 | tuxinator_linuxM | check bugs.digium.com |
08:23.08 | shido6 | ok |
08:23.09 | shido6 | To reset a forgotten Admin Password: |
08:23.09 | shido6 | 1. Get the systems serial number from the system or from the System |
08:23.09 | shido6 | Information screen. |
08:23.09 | shido6 | 2. Go to System >Diagnostics > Reset System. |
08:23.09 | shido6 | 3. Enter the systems serial number and select Delete System Settings. |
08:23.10 | shido6 | 4. Select Reset System. |
08:24.16 | PTG123 | shido6: i don't have a system diagnostics screen |
08:24.55 | porkchop | tuxinator_linuxM: can't seem to find a specific report that applies to me. I'll keep working on it I suppose. |
08:25.11 | shido6 | sorry |
08:25.12 | shido6 | im an idiot |
08:25.14 | eric_ | hrmm, it seems to sefgault after parsing an almost-empty extconfig.conf |
08:25.14 | shido6 | wrong box |
08:25.52 | shido6 | thats for the viewstation |
08:25.53 | shido6 | sorry |
08:26.05 | PTG123 | damn :( |
08:26.08 | PTG123 | any other ideas? |
08:27.31 | *** join/#asterisk santiago (~santiago@63.245.86.95) |
08:27.43 | shido6 | still lookin |
08:29.12 | *** join/#asterisk srt (~nobody@gw0-cgn.reucon.net) |
08:29.25 | *** join/#asterisk Zgarbi (~my@212.58.125.68) |
08:30.32 | *** join/#asterisk tessier (~treed@210.245.96.123) |
08:31.34 | *** join/#asterisk herag (herag@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net) |
08:32.28 | herag | what does it mean when I get an SIP 400, bad request when trying to take an incoming call and dial forward it out to another number? |
08:33.46 | shido6 | I dont see anything PT |
08:33.51 | shido6 | u have physicall access to the phone? |
08:33.54 | herag | I hadn't changed any configs, but suddenly I'm getting these weird sip 400 errors...but it only happens when a call tries to be forwarded out, I can dial straight out just fine, and calls can go into my device, but if I try to take a call coming in from broadvoice, and then forward it out to another number, it won't work anymore |
08:33.56 | PTG123 | shido6: yah me either, not sure what to do |
08:34.04 | shido6 | you could do the "short the damn thing" trick |
08:34.17 | shido6 | unplug , spark , plug , spark, plug |
08:34.26 | shido6 | but you risk turning the phone into a paperweight |
08:34.36 | shido6 | hopefully it will kickstart to factory defaults |
08:34.45 | shido6 | u know what I mean by spark? |
08:34.53 | shido6 | plug the power in just enough to spark it |
08:34.56 | shido6 | and unplug |
08:34.59 | shido6 | then plug it back in |
08:35.03 | shido6 | then unplug it and spark again |
08:35.12 | PTG123 | that works? |
08:35.14 | PTG123 | its poe though |
08:35.19 | shido6 | oh lord |
08:35.43 | shido6 | Im trying my damndest to not say, "fou're yucked." |
08:36.03 | PTG123 | haha |
08:36.07 | PTG123 | well they have shit on their site |
08:36.12 | PTG123 | on how to reset everything but this phone |
08:36.54 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
08:37.26 | modulus_ | jbot stable? |
08:37.27 | jbot | i guess stable is the status of a Debian release when no packages will be added or changed unless a security fix is needed, or sta-ble adj; uptime in excess of 365days, or where the horses live The current stable version of Debian is woody (3.0). |
08:37.31 | modulus_ | wtf |
08:37.35 | modulus_ | jbot stable asterisk? |
08:37.54 | modulus_ | jbot you whore of a bot, tell me the latest stable version you bitch machine |
08:38.13 | drumkilla | 1.0.6 - 1.0.7 will be very soon |
08:38.40 | modulus_ | jbot should divulge that info |
08:38.49 | drumkilla | it's in the topic ... |
08:39.03 | modulus_ | this irc session is screen'd |
08:39.13 | drumkilla | feel free to go test it for me :) |
08:40.27 | footnote | Where the Horses Live Linux(TM) |
08:43.10 | rvhi | try to compile voicemail with mysql support |
08:43.14 | rvhi | get this app_voicemail.c:371:31: mysql-vm-routines.h: No such file or directory |
08:43.24 | rvhi | any suggestion? |
08:43.51 | xkev | ast-- switch (comp->type & PRI_DEBUG_APDU) { |
08:43.51 | xkev | + switch (comp->type) { |
08:43.58 | xkev | s/ast--/-/ |
08:44.12 | xkev | that typo was breaking caller name facility message. |
08:44.31 | xkev | Eight, ^^ |
08:46.23 | godsmoke | wohoo |
08:46.25 | godsmoke | FWD on my cisco |
08:46.27 | godsmoke | it all works |
08:47.09 | ta[i]nted | what's up with the crazy |
08:48.05 | drumkilla | rvhi: it's in asterisk-addons |
08:50.55 | rvhi | i found it in asterisk-addons 1.0.6 |
08:51.00 | rvhi | i was using 1.0.0 |
08:51.12 | rvhi | guess was added after 1.0.0 |
08:51.19 | drumkilla | nah |
08:51.24 | drumkilla | it was always in addons ... |
08:51.39 | rvhi | maybe i didn't get the right addons for 1.0.0 |
08:51.46 | rvhi | hard to find it now... :) |
08:51.50 | drumkilla | no big deal |
08:51.57 | rvhi | anyone i copy the file from 1.0.6 |
08:51.58 | drumkilla | you want to be running the new stuff anyway |
08:52.05 | rvhi | praying it is going to work |
08:52.12 | xkev | lousy mysql licensing |
08:52.12 | drumkilla | rvhi: you should run the code from stable cvs at the moment, actually |
08:52.28 | drumkilla | then, make a note on bug 3746 that you're using it without problems |
08:53.11 | rvhi | i made some code change to 1.0.0 |
08:53.32 | rvhi | going to take some time to merge my change to 1.0.6/stable |
08:53.45 | drumkilla | ok ... |
08:53.54 | *** join/#asterisk Tommmo (~tps@203.62.181.52) |
08:54.36 | Tommmo | anyone know if it's possible to configure the call progress tones on cisco 7940/60 ? |
08:55.55 | godsmoke | what are call progress tones? |
08:56.33 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
08:57.13 | Tommmo | e.g. the tone you hear when you call someone, and you are waiting for them to pickup |
08:57.18 | Tommmo | busy tones, etc |
08:57.22 | godsmoke | ah |
09:02.02 | *** join/#asterisk r0d3nt|m (~RatMan@4.19.77.194) |
09:12.00 | modulus_ | i think someone needs to port asterisk-addons to the freebsd ports tree |
09:15.24 | ikey | can anyone explain how to configure asterisk as sip client and server |
09:16.28 | Zeeek | is it possible to specify callerid in a .call file? |
09:17.09 | Zeeek | never mind |
09:17.11 | Zeeek | got it |
09:17.21 | srt | Zeeek: sure look at sample.call |
09:17.25 | Zeeek | got it |
09:19.03 | PTG123 | anyone know how to make the polycom webserver answer? :) |
09:19.07 | PTG123 | i got into the phone to config it |
09:19.11 | PTG123 | but now webserver won't respond |
09:19.45 | Zeeek | how did this problem come about? I'm considering buying a Polycom |
09:19.59 | Zeeek | did you just forget the password? |
09:21.10 | PTG123 | i got it from ebay :) |
09:21.14 | PTG123 | buy the cisco |
09:21.17 | PTG123 | i bought both |
09:21.19 | PTG123 | cisco is much better |
09:21.20 | PTG123 | :) |
09:21.40 | Zeeek | the poly I'm considering can be had new for $220 or so |
09:21.46 | Sedorox | hmm |
09:21.48 | Zeeek | the cisco is more I think |
09:21.52 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
09:22.00 | PTG123 | umj |
09:22.05 | PTG123 | i got one new for $137 :) |
09:22.06 | Sedorox | the channel I think, over all likes polycom more |
09:22.08 | PTG123 | for a 7960 |
09:22.16 | PTG123 | and my cisco is a 6 line |
09:22.25 | Zeeek | I don't do ebay though |
09:22.36 | Zeeek | so I'm talking new from a distributor |
09:23.21 | *** join/#asterisk brimston3 (me@146.229.178.20) |
09:24.09 | brimston3 | i'm new to asterisk, is there a quickstart guide to getting a TDM11B running on a debian box with 2.6.8 ? |
09:24.23 | Zeeek | Starter tutorial: |
09:24.23 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
09:24.23 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
09:24.23 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
09:24.23 | Zeeek | THE reference of the moment: |
09:24.24 | Zeeek | http://www.asteriskdocs.org |
09:24.31 | Zeeek | try number three above |
09:24.42 | brimston3 | thanks Zeeek |
09:24.49 | Zeeek | then search for 2.6 kernel woes |
09:25.05 | Zeeek | on the wiki |
09:26.55 | brimston3 | number three as in automated.it right? |
09:26.59 | brimston3 | or asteriskdocs ? |
09:27.15 | Zeeek | automated has a complete guide |
09:27.20 | Zeeek | but not 2.6 |
09:27.24 | brimston3 | ah |
09:27.34 | Zeeek | also the latest asteriskdocs.org has a complete setup guide |
09:27.47 | Zeeek | also the wiki has all the stuff but not easy to find |
09:28.04 | Zeeek | http://www.voip-info.org/wiki-Asterisk |
09:28.26 | Zeeek | http://www.voip-info.org/wiki-Asterisk+installation+tips |
09:28.46 | Zeeek | Look for 2.6 HERE : http://www.voip-info.org/wiki-Asterisk+OS+Platforms |
09:28.57 | brimston3 | thanks |
09:32.43 | *** join/#asterisk nine76 (~t00r@cpe-69-135-184-24.woh.rr.com) |
09:32.50 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
09:34.47 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
09:35.33 | PTG123 | anyone know why i can dial numbers with my polycom but it shows unregistered? |
09:37.14 | Zeeek | beacuse there's no password in the peer/user/friend entry? |
09:37.27 | PTG123 | call wouldnt work if their wasn't |
09:37.31 | Zeeek | because default has all possible extensions? |
09:37.32 | GMsoft | is there a command to send MWI messages for SIP ? I've got many asterisk and a centralized voicemail here and I can't find such command on voip-info |
09:37.58 | PTG123 | Zeek: no its working right with my sip account |
09:38.02 | PTG123 | Zeek: just not registering |
09:38.06 | Zeeek | PTG is it unregistered or unreachable? |
09:38.13 | PTG123 | unregistered |
09:38.18 | PTG123 | think it just never sent the register package |
09:38.21 | PTG123 | its not registering |
09:38.25 | Zeeek | as in not in the show peers list? |
09:38.59 | PTG123 | don';t believe so |
09:39.14 | Zeeek | belief has nothing to do with voIP |
09:39.25 | Zeeek | it is or is not in the list |
09:40.57 | tuxinator_linuxM | Man, I'm tired |
09:41.02 | Zeeek | sleep |
09:41.08 | Zeeek | your eyes are closing |
09:41.12 | Zeeek | sowly... |
09:41.17 | tuxinator_linuxM | yah they are |
09:41.19 | Zeeek | slowly even... |
09:41.33 | Zeeek | think of J'Lo |
09:41.40 | tuxinator_linuxM | you should of seen file fall asleep with his hands on the keyboard |
09:41.55 | Zeeek | did it look like this? |
09:42.03 | Zeeek | fffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffff |
09:42.20 | modulus_ | beep! |
09:42.26 | Zeeek | meep |
09:42.31 | tuxinator_linuxM | we got a picture, I think it is on drumkillers camera |
09:42.38 | modulus_ | zeeek |
09:42.47 | Zeeek | meeep |
09:43.01 | Eight | ergh... wtf. |
09:43.09 | Eight | So... I'm playing with AGI. |
09:43.15 | Zeeek | Eight wft are you doing up at this hour? |
09:43.23 | Eight | I SAY NUMBER twice, and then "HANGUP". |
09:43.30 | Eight | and my SIP client hears both numbers... |
09:43.34 | Zeeek | no one does good AGI at 4AM |
09:43.44 | Eight | Zeeek: depends on when they woke up =) |
09:43.54 | Zeeek | no, it's an absolute |
09:44.08 | Zeeek | 4AM is the hour where most people die at hospitals |
09:44.13 | Eight | Anyways... HANGUP seems to send 403 forbidden. |
09:44.21 | Eight | *even though* the call was already in progress. |
09:44.24 | Eight | it's fucked up. |
09:44.34 | Zeeek | write a module |
09:44.41 | Zeeek | you'll have more control |
09:44.44 | Eight | Zeeek: god no. I want to use python not c. |
09:44.47 | *** join/#asterisk brimstone (~brimstone@146.229.178.20) |
09:44.50 | Zeeek | muhahaha |
09:45.08 | Eight | I was thinking last night "hey, maybe I'll actually try using C for something other than a CompSci course..." |
09:45.20 | Eight | after 2 hours bouncing around the asterisk source... to hell with that. |
09:45.27 | Zeeek | http://wiki.e164.org/moin.cgi/WordIndex |
09:46.13 | Zeeek | c is like the molecule that all unix is made from |
09:46.27 | Eight | Zeeek: I know. |
09:46.27 | Zeeek | it's the one thing they have in common |
09:46.38 | Zeeek | and it's so easy and powerful |
09:46.50 | Eight | Zeeek: hahaha. |
09:47.01 | Zeeek | someone might make up some handy macros for use with asterisk though |
09:47.06 | Eight | Zeeek: spend some time with a decent language some time, you'll NEVER go back to C. |
09:47.08 | Zeeek | that would help a lot |
09:47.16 | Zeeek | name a decent language? |
09:47.25 | Zeeek | besides py |
09:47.28 | Eight | Python, Java, Ruby... |
09:47.37 | Eight | Lisp. |
09:47.44 | Eight | Smalltalk |
09:47.50 | brimstone | perl? |
09:47.51 | Zeeek | ah you want objects... use c++ |
09:48.02 | Eight | brimstone: perl's ok, if you can code responsibly enough. |
09:48.15 | Eight | brimstone: but it's really conducive to writing difficult to read code. |
09:48.24 | Eight | Zeeek: C++ is an abomination. |
09:48.29 | Eight | Objective C is much, MUCH better. |
09:48.35 | Zeeek | anyway, great thngs have been done with all languages, so it's infantile to argue for one or the other. Only your own limitations are being argued. (someone write that down) |
09:48.48 | Eight | esp' in the NeXTSTEP/GNUStep/Mac OSX environment |
09:49.18 | Eight | Zeeek: Are you arguing that Asterisk might as well have been written in COBOL? or Assembler? |
09:49.32 | Eight | Zeeek: There's a right tool for the job, and there are new developments in tools. |
09:49.33 | Zeeek | again, useless conjecture |
09:49.44 | Zeeek | Eight yes, but you are arguing your opinion |
09:49.52 | Eight | Zeeek: I have nothing else to argue =) |
09:50.03 | Zeeek | I didn't mention COBOL which is made to do certain things as you note |
09:50.04 | Eight | Zeeek: Seriously, do you have much experience with languages other than C? |
09:50.09 | Zeeek | are you over 20 ? |
09:50.12 | Eight | yes. |
09:50.19 | Zeeek | how much over |
09:50.41 | Zeeek | or, if you prefer what is the year you made your first program? |
09:50.49 | Eight | oh geezus. |
09:50.50 | Eight | I have no idea. |
09:50.52 | Zeeek | (and in what lmanbgua) |
09:50.56 | Zeeek | language? |
09:50.57 | Eight | It was in C. |
09:51.04 | Zeeek | what decade? |
09:51.27 | Eight | I'd have to say... early 90s. |
09:51.44 | Zeeek | that's a decent exposure |
09:51.50 | Eight | Zeeek: oh gee, thanks. |
09:51.53 | Zeeek | so I'm surprised you want to make these statements |
09:52.14 | Eight | Zeeek: What experiences make your opinions better than mine? =p |
09:52.17 | Zeeek | but I do think you should use whatever language you want |
09:52.19 | Zeeek | none |
09:52.29 | Zeeek | that('s myy whole point - you are full of sheisse |
09:52.43 | Zeeek | no one is capable of maintaing these kinds of args |
09:52.56 | Zeeek | so you're just talking at 4AM |
09:53.08 | Zeeek | and not getting your AGI to work |
09:53.13 | Zeeek | in py |
09:53.21 | Eight | Dude, the python code is like 6 lines long. |
09:53.27 | Eight | It's the HANGUP command that's being weird. |
09:53.31 | Zeeek | easy to debug then |
09:54.06 | Zeeek | talk to people on dev about this, maybe they can "fix" it |
09:54.09 | Eight | Zeeek: let me know when you fall off your high horse and I'll start speaking to you again =) |
09:54.17 | Zeeek | you misunderstand |
09:54.32 | Zeeek | I don't care, I just think these blanket statements are silly |
09:54.46 | Zeeek | I do not defend or recommend any distro or language |
09:55.20 | Eight | Ah, now you've said something concrete. |
09:55.26 | Zeeek | I said it before |
09:55.36 | Zeeek | I was just curious about your experience - and now I know |
09:55.47 | Zeeek | have you slept much ? |
09:56.08 | Eight | Zeeek: yes, I've only been awake 8 hours maybe (don't recall exactly). |
09:56.21 | Eight | ok, my turn to quiz. |
09:56.27 | Eight | What languages have you had serious exposure to? |
09:56.28 | Zeeek | ok, I think you read some kind of animosity in to what I was trying to say |
09:56.40 | Zeeek | Eight "serious" I don't know |
09:56.53 | Eight | Zeeek: you were wholly discounting my opinions prima fascia... that's generally a hostile thing to do =) |
09:56.53 | Zeeek | I'm a lousy programmer - let's get that out of the way |
09:57.11 | Zeeek | no I discount ALL opinions that say "sucks" etc |
09:57.14 | Zeeek | but let me answer |
09:57.39 | Zeeek | I began with Z80, 6800 and 6502 machine, then assebly for same |
09:58.09 | Eight | Zeeek: You may be a lousy programmer, but perhaps it's just because you've not found a language that lets you spend enough time expressing yourself, and less time controlling bits (or in C's case, pointers). |
09:58.11 | Zeeek | then did Fortran (ans basic obviously) then used PDP11 under two OS, then VAX11 assembly |
09:58.22 | Zeeek | no I am a lousy programmer |
09:58.40 | Zeeek | then worked ona system that had terminals that spoke Forth |
09:58.55 | Zeeek | and installed forth on my little Radio Shack with 32k |
09:58.57 | Eight | alright, it's not AGI... for some reason my client is getting 403 all the time when I use hangup right now. |
09:59.24 | Zeeek | again, concretely, talk to the guys on dev (or wait until way later int he day) |
09:59.34 | Zeeek | there's always the mailing list too |
09:59.47 | Eight | exten => t,1,hangup |
09:59.48 | Eight | no error |
09:59.52 | Eight | exten => 2,1,agi,default.py |
09:59.52 | Eight | exten => 2,2,hangup |
09:59.53 | Eight | error |
10:00.01 | Eight | It's not really an error, exactly... |
10:00.04 | Zeeek | stack problems? |
10:00.13 | Eight | it's just the SIP client gets a 403 instead of exiting cleanly when the connection drops. |
10:00.47 | Zeeek | what's the mast line of the AGI? |
10:00.50 | Zeeek | last line |
10:01.13 | Eight | exten => 2,1,agi,default.py |
10:01.13 | Eight | exten => 2,2,saynumber(9) |
10:01.13 | Eight | exten => 2,3,hangup |
10:01.17 | Eight | still 403. |
10:01.27 | Eight | and I hear '9' before it drops me. |
10:01.32 | Zeeek | what's the last line of the python AGI? |
10:01.39 | Zeeek | or last few |
10:01.52 | Eight | actually, I lied, the script is 3 lines long =p |
10:02.13 | Eight | sys.exit(0) |
10:02.25 | Eight | I put it in explicitly, jus tin case. |
10:02.32 | Zeeek | ok then the first two |
10:03.11 | Eight | dude, all it does is send "SAY NUMBER 1 """ and "SAY NUMBER 2 """ to stdout and flush |
10:03.25 | Eight | I really, really don't think it's my code =p |
10:04.36 | RaYmAn-Bx | have you checked it actually runs if you run it manually? |
10:04.37 | *** join/#asterisk coppice (~chatzilla@111.196.17.210.dyn.pacific.net.hk) |
10:04.40 | Eight | exten => 2,1,saynumber(8);agi,default.py |
10:04.40 | Eight | exten => 2,2,saynumber(9) |
10:04.40 | Eight | exten => 2,3,hangup |
10:04.47 | Eight | I still get 403 =p |
10:04.55 | Eight | and my code doesn't even run. |
10:05.01 | Eight | it's an extensions.conf issue. |
10:05.33 | Eight | stop talking about my code =) |
10:05.33 | Zeeek | significant debug element there |
10:05.33 | djorange | hi hi |
10:06.14 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
10:06.32 | RaYmAn-Bx | Eight: not sure if it's needed, but have you considering doing an Answer first? |
10:06.38 | RaYmAn-Bx | considered* |
10:07.12 | djorange | okay i got two trunks coming in and i need to get one trunk to hit auto attendent1 and then transfer to group 200. |
10:07.21 | Eight | RaYmAn-Bx: aaah, good call. |
10:07.43 | Eight | RaYmAn-Bx: damn =/ |
10:07.49 | Eight | I really thought that might have been it = |
10:08.57 | Eight | oh, it was. |
10:09.01 | Eight | Forgot to reload that time =p |
10:09.15 | Eight | I guess the client just thinks the call fails unless it gets 'answered' properly. |
10:09.29 | Eight | and 'saynumber' doesn't "answer" properly. |
10:10.08 | Eight | if I use playback, I don't get 403. |
10:11.10 | Eight | RaYmAn-Bx: Thanks man =) |
10:11.21 | Eight | Zeeek: next time when i say it's not my code... it ain't my friggin' code =) |
10:11.29 | Zeeek | "New versions of Asterisk have added "Answer" capabilities to several functions like Playback(), which means that those functions will answer themselves if necessary. " |
10:11.42 | Eight | Ya... I think that's actually a BAD idea, to be honest. |
10:11.51 | Eight | it promotes problems like I just had =p |
10:11.54 | Zeeek | I never said it was - again - why the hostility? - I was curious to see what your py looked like |
10:11.56 | Eight | (and I thought it was a bad idea before this!) |
10:12.26 | RaYmAn-Bx | Eight: regardless of how correct your code is it's always a good idea to run it manually and make sure it doesn't have any stupid errors :P |
10:12.33 | Eight | Zeeek: I wasn't being hostile, just trying to make a joke =) |
10:12.51 | Eight | RaYmAn-Bx: Ya, I'm actually watching from the Asterisk console. if Python errors it displays there. |
10:13.02 | *** join/#asterisk Inv_arp (junya@adsl-8-232-168.mia.bellsouth.net) |
10:13.13 | Zeeek | Eight I have no doubt that anyone here is stonger in programming that I am and wouldn't question their stuff |
10:13.15 | Eight | RaYmAn-Bx: and I *know* it was all working, because the only code in the file results in an audible playback on the channel, which I was hearing. |
10:13.46 | Zeeek | Eight however I have lived lionger than anyone here and sometimes questiion the wisdom of some statements |
10:14.32 | Zeeek | s/wisdom/"Absolute Veracity"/ |
10:14.37 | Eight | =) |
10:14.45 | *** join/#asterisk ikey (~ikey@202.54.37.183) |
10:15.16 | Eight | I was going to say, "I do mistake age for wisdom, nor experience for expertise." |
10:15.20 | Eight | but you defused it =p |
10:15.26 | Eight | err, 'do not'. |
10:15.38 | Zeeek | Eight if I may drag this on one more time, a lot of people speak of things they know little about, for example I have never used H323 and wouldn't say it sucks |
10:16.01 | Zeeek | yet I see lots of people talking about countries they have never visited |
10:16.11 | Zeeek | or people they've never met |
10:16.16 | Eight | Zeeek: I have used C, and in my opinion it does suck for all but very specific tasks (building operating systems is a specific task). |
10:16.24 | Zeeek | I know and understand |
10:16.45 | Eight | actually, it quite excels at building operating systems. |
10:17.04 | Zeeek | My own prejudice wrt to asterisk devel is that I don't like talking to stdin/out |
10:17.14 | Eight | you mean w/ AGI? |
10:17.14 | Zeeek | it seems flaky to me |
10:17.18 | Zeeek | yeah |
10:17.25 | Zeeek | doesn't feel right |
10:17.32 | ManxPower | ~docs |
10:17.33 | 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 |
10:17.37 | Zeeek | but that's just my own experience with API |
10:17.39 | Eight | It's *really* common to do things that way in Unix. |
10:17.48 | Zeeek | Go back to bed Manx ! |
10:18.44 | ManxPower | Zeeek, I just slept 8 hours. |
10:18.51 | Zeeek | ok then you're cool |
10:19.04 | Zeeek | someone was having big Polycom problems a while ago |
10:19.10 | RaYmAn-Bx | 8 hours just isn't enough :/ |
10:19.25 | ManxPower | But as a cat I know there is no such thing as "too much sleep" |
10:19.27 | Zeeek | ~seen PTG123 |
10:19.29 | jbot | ptg123 is currently on #asterisk (2h 58m 53s). Has said a total of 83 messages. Is idling for 40m 30s |
10:19.39 | Zeeek | hehe |
10:19.59 | Zeeek | Manx I'm watching the Polycom stuff carefully as I want to buy one |
10:20.18 | Eight | ~seen |
10:20.22 | Eight | ~seen eight |
10:20.23 | jbot | eight <~blake@12-205-155-39.client.mchsi.com> was last seen on IRC in channel #asterisk, 1s ago, saying: '~seen eight'. |
10:20.39 | Eight | what... no messages? |
10:20.54 | Zeeek | ~seen Mel Gibson's latest movie |
10:20.55 | jbot | i haven't seen 'mel gibson's latest movie', Zeeek |
10:21.12 | Eight | how do you add info to jbot? |
10:21.13 | Zeeek | ~got milk |
10:21.14 | jbot | ACTION chugs down a carton |
10:21.22 | Zeeek | lazy bot |
10:21.34 | Eight | jbot jbot? |
10:21.35 | jbot | methinks jbot is the shipboard computer, but you may call me eddie if it helps you relax |
10:21.53 | Zeeek | ~eddie |
10:21.54 | jbot | Robust, clustering, load balancing, high availability, web server tool.. URL: http://www.eddieware.org/ |
10:22.07 | Eight | ~agi |
10:22.08 | jbot | [agi] the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
10:22.16 | *** join/#asterisk wasim (~wasim@203.81.216.2) |
10:22.24 | Eight | Zeeek: there, go get pyst. It does the API for you =p |
10:22.33 | Zeeek | Rappers are even better |
10:22.52 | Zeeek | "I don't need no AGI... I got it all in mah CLI!" |
10:23.23 | Zeeek | ok it must be late for humor |
10:24.10 | Zeeek | go for it! |
10:24.24 | Zeeek | (sorry, forgot the duuuuude) |
10:24.45 | ManxPower | Command Line Interface Terminanal? |
10:25.23 | Zeeek | Timing With Asychronous Tones |
10:26.04 | Zeeek | Push Under Symmetrical Signalling Yeti |
10:26.32 | Eight | ManxPower: was 'terminAnal' intentional? =/ |
10:26.58 | Eight | I didn't even see it for a while =p |
10:27.15 | Zeeek | you are supposed to be implementing |
10:27.21 | Eight | I can type and type at the same time. |
10:27.38 | Zeeek | multiple windows? When did that get invented? |
10:27.51 | Zeeek | What editor do you use, Eight? |
10:27.57 | Eight | xemacs |
10:28.04 | Eight | or, vi, depending on what I'm doing. |
10:28.09 | Eight | or TextEdit.app. |
10:28.10 | Zeeek | ok, you're not a vi nazi |
10:28.25 | Eight | vi is for config files, xemacs is for code, textEdit is for everything else. |
10:28.41 | Zeeek | "the right tool..." |
10:29.20 | Zeeek | I must be a wimp for using pico/nano - though I have used and liked emacs |
10:29.43 | Eight | I used to use pico for a while, until I got around to learning vi. |
10:29.46 | Zeeek | In fact I did use "mouseemacs" years ago |
10:29.48 | Eight | I still use pico in pine. |
10:29.58 | Zeeek | pine! whoa |
10:31.07 | ta[i]nted | pine was made at my old university |
10:31.33 | modulus_ | how do i ignore exten => i,1,?? |
10:31.37 | coppice | you went to university in a lumber forest? :-\ |
10:31.51 | Inv_arp | bah paco...nano :) |
10:31.55 | Eight | modulus_: what do you mean, ignore? |
10:32.06 | Eight | exten => i,1,noop |
10:32.09 | modulus_ | ignore invalidly pressed extensions |
10:32.16 | modulus_ | NoOp does not ignore |
10:32.23 | modulus_ | it takes over the context and extension |
10:33.07 | Zeeek | coppice SawMill U. |
10:33.34 | modulus_ | zeeek, help me out here? |
10:33.47 | Zeeek | why do you want to ignore ignore? |
10:34.00 | ta[i]nted | pretty close |
10:34.12 | ta[i]nted | there are a lot of trees in my state |
10:34.14 | Inv_arp | modulus_: exten => i,1,Goto(s,1) ; go back to first step? |
10:34.18 | modulus_ | i want my menu to continue even if invalid exten |
10:34.21 | ta[i]nted | coppice university of washington |
10:34.26 | modulus_ | inv_arp, i don't want to restart the menu |
10:34.32 | Zeeek | modulus_ put a goto to the entry of the extension |
10:34.52 | modulus_ | i want the menu to keep going and ignore if extension pressed is invalid |
10:34.54 | Zeeek | either before any prompts or after |
10:35.12 | Inv_arp | modulus_: use Playback () instead of Background() |
10:35.13 | modulus_ | b/c ppl press extensions before i can read(VAR) |
10:35.19 | Zeeek | i,1,goto(3) |
10:35.23 | modulus_ | inv_arp, good idea i should've thought of that |
10:36.00 | [ro]nic3try | HOW do i stop asterisk to ask for autentification on incoming calls ? |
10:36.10 | modulus_ | Authenticate() |
10:36.54 | [ro]nic3try | no i have an sip_proxy_out in sip.conf |
10:37.04 | [ro]nic3try | i have an sip_proxy_out in sip.conf |
10:37.48 | Eight | wait for digit returns the decimal ascii code of the digit dialed. |
10:37.55 | [ro]nic3try | if sip_proxy_out is set as user it doesnt ask for sipProxy autentification |
10:37.59 | Eight | A string, or a number... nah, can't have that =p |
10:38.33 | Eight | ~agi |
10:38.34 | jbot | agi is probably the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
10:38.34 | [ro]nic3try | if sip_proxy_out is set as peer or friend it asks for sipProxy autentification |
10:38.36 | *** join/#asterisk Dibbler (~Dibbler@snaddy.plus.com) |
10:39.37 | Zeeek | Eight isn't everything strings in asterisk ? |
10:40.56 | *** join/#asterisk mbranca (~matteo@80.152.73.227) |
10:41.27 | Eight | '200 result=51\n' |
10:41.34 | Eight | that's the string I get back for 3. |
10:41.42 | Eight | I think pyst will do the conversion for me, though. |
10:42.13 | ManxPower | Does anyone else think that Kevin Flemming is very smart and very crazy? |
10:42.28 | *** join/#asterisk brimstone (~brimstone@146.229.178.20) |
10:43.06 | Zeeek | him? http://www.integritypersonnel.com/kflemming.shtml |
10:43.08 | brimstone | thanks again Zeeek, i got wctdm to compile under debian 2.6.8 |
10:43.15 | Zeeek | great! |
10:43.22 | Zeeek | where do you find the info? |
10:43.46 | brimstone | required me to rm all of 2.6.* and then load the kernel again from a knoppix chroot, but it's working |
10:43.49 | ManxPower | nomad_, the one on the mailing lists. |
10:43.56 | brimstone | no place special, just kind of kicked stuff around |
10:44.09 | ManxPower | At first I thought he was just plain crazy, but now I think he's very smart too. |
10:44.10 | Zeeek | Manx users list or other? |
10:44.12 | modulus_ | if ($bal < 1.05){ |
10:44.12 | modulus_ | <PROTECTED> |
10:44.12 | modulus_ | <PROTECTED> |
10:44.12 | modulus_ | <PROTECTED> |
10:44.12 | modulus_ | } |
10:44.17 | ManxPower | This is either a very bad thing or a very good thing. |
10:44.20 | modulus_ | for some reason that $bal comparison tickles me |
10:44.46 | modulus_ | guess what i'm doing wheeee! |
10:44.51 | ManxPower | kpfleming@starnetworks.us |
10:46.37 | *** join/#asterisk lidl (~little@213-140-6-96.fastres.net) |
10:47.26 | Zeeek | Yes, sometimes what happens is that you are accidentally using |
10:47.26 | Zeeek | undocumented behavior, and when that gets "fixed" your system breaks. |
10:47.35 | Zeeek | That's a good one! |
10:47.36 | modulus_ | $AGI->debug() is a real function right? |
10:47.40 | modulus_ | i read it once somewhere |
10:48.02 | *** join/#asterisk ikey (~ikey@202.54.37.183) |
10:49.01 | ikey | can any one help me on configuring sip channel with iptel.org |
10:49.10 | ta[i]nted | modulus_ your code sucks |
10:50.07 | ta[i]nted | what exactly do u want to debug |
10:50.39 | *** join/#asterisk afe ([KFVNEtAOw@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
10:50.51 | afe | Morning, folks! |
10:51.13 | Eight | oh my god. |
10:51.20 | Eight | I've been spending waaaay too much time in extensions.conf |
10:51.25 | Eight | I just blanked on how to do a goto in python =p |
10:51.29 | Eight | Oooh ya =p |
10:52.04 | afe | You know, it's just too much fun pokin around in extensions.conf all the time :D |
10:54.32 | felipex | any info about bluetooth ? |
10:54.58 | ikey | can any one help me on configuring sip channel with iptel.org |
10:57.08 | [ro]nic3try | HELP |
10:57.29 | [ro]nic3try | i canot pe called from a ser server |
10:57.41 | [ro]nic3try | even i have been registered |
10:57.54 | [ro]nic3try | and i can call on tha server |
10:58.46 | ManxPower | Does anyone know of a deaktop pager for Win32 like the ones in KDE, GNOME, FVWM, etc? |
10:58.51 | ManxPower | desktop too. |
10:59.34 | Zeeek | Manx you mean a pop window? |
10:59.50 | Zeeek | for callerid and such? |
11:00.29 | Zeeek | We use Yac |
11:01.48 | *** join/#asterisk afe ([p7Gv1IiBf@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
11:02.02 | ManxPower | Zeeek, No, like the multiple virtual desktop application that basically every linux GUI has. |
11:02.31 | Zeeek | ya xp has it |
11:02.51 | Ash | ManxPower: XP PowerToys has what you want |
11:03.19 | ManxPower | Ash, Does it work in Win3k? |
11:03.22 | ManxPower | ..er.... |
11:03.24 | ManxPower | Win2k |
11:03.30 | Ash | ManxPower: No, it's only for XP. |
11:03.36 | Ash | There are other things for 2k, though. |
11:03.39 | Zeeek | but there is free ware for 2k |
11:03.40 | Ash | (I don't know what they are) |
11:03.46 | Ash | http://download.microsoft.com/download/whistler/Install/2/WXP/EN-US/DeskmanPowertoySetup.exe |
11:03.50 | ManxPower | I don't even know the correct terms to search for. |
11:03.52 | Ash | is the XP one |
11:04.01 | Zeeek | virtual desktop |
11:04.04 | Ash | ManxPower: "virtual desktop win32" |
11:04.09 | Ash | or win2k or what have you |
11:04.27 | ManxPower | Yeah. I get all sorts of stuff that's pretty useless with that search. |
11:04.45 | afe | ~google nstall/2/WXP/EN-US/DeskmanPowertoySetup.exe |
11:04.45 | ManxPower | It will be less of an issue once the new motherboard for my linux box arrives. |
11:05.00 | Zeeek | ok, manx I see I need to stop what I'm doing to help you |
11:05.08 | Zeeek | wait a second adn I 'll find it for you |
11:05.13 | afe | ~google windows 2000 virtual desktop |
11:05.25 | ManxPower | Zeeek, I'm looking for sometthing that works very much like the linux version. |
11:05.48 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
11:05.59 | afe | I think the first link points you to what you want |
11:06.07 | Zeeek | These are little icos you hit to change the display to a diffenent destop |
11:06.26 | ManxPower | That's much closer to what I'm looking for. |
11:06.27 | Zeeek | I find this kind of thing on freeware Usenet groups |
11:06.47 | Zeeek | but the message "virtual desktop" isn't there, only the header :( |
11:07.00 | Zeeek | try googling the freeware goup |
11:07.07 | Zeeek | group |
11:07.20 | Zeeek | this is a question that comes up often there |
11:07.23 | afe | try http://virt-dimension.sourceforge.net/ |
11:07.47 | ManxPower | afe, Yes, that one looks closer to what I'm looking for. |
11:08.06 | ManxPower | It still is a seperate window, rather than living on the taskbar, of course. |
11:08.30 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
11:09.01 | Zeeek | look here: http://www.ams.as.ro/index.htm |
11:09.39 | *** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
11:09.51 | Zeeek | no, don't |
11:10.07 | Zeeek | I had one of these for Win2K but I can't find it - sorry |
11:11.14 | ManxPower | Zeeek, "looking there" doesn't seem to work for me. |
11:11.37 | Zeeek | you want freeware, right? |
11:11.49 | ManxPower | Zeeek, If it's good enough I'll pay a small amount. |
11:12.04 | Zeeek | http://www.tucows.com/virtualdesk95_default.html |
11:12.15 | Zeeek | tucows is a good source for this kind of stuff |
11:12.26 | *** join/#asterisk djin (~djin@62.58.40.196) |
11:12.28 | Zeeek | oops all 4 are shareware |
11:12.40 | Zeeek | I often buy good programs for up to $30 though |
11:13.36 | Zeeek | I know there is at least opne free one but I can't find it. They aren't maintained because most people have moved to XP which has a free one in XP Toys |
11:14.43 | Zeeek | Manxpower, here ya go, this looks promising: http://www.free-soft.ro/desktop/desktop.html |
11:15.05 | ManxPower | I'm running Windows. That means I should be utterly terrified of installing software. |
11:15.16 | Zeeek | yes be afraid. Be very afraid |
11:15.21 | *** join/#asterisk afe ([oZrbWPFNw@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
11:15.50 | Zeeek | software "firewall" like Kerio is an absolute must (to signal callouts) |
11:15.57 | coppice | anything that looks like work gets me pretty scared |
11:16.05 | ManxPower | i.e. I don't want to randomly install these apps and hope one of them doesn't fuck up my system. |
11:16.09 | brimstone | in a call file, how do i make it wait till someone picks up the line on the other end? |
11:16.22 | ManxPower | brimstone, you stop using analog ports. |
11:16.26 | Zeeek | brimstone put a value like 9999999 |
11:16.45 | Zeeek | or none at all in dial |
11:17.06 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
11:17.18 | brimstone | in the Channel: line? |
11:17.23 | Zeeek | Manx, still sourceforge |
11:17.24 | Zeeek | http://sourceforge.net/projects/virtual-desktop/ |
11:17.28 | brimstone | no value? |
11:17.55 | Zeeek | Dial with no timeout value will wait IIRC |
11:17.57 | afe | I have a X100P clone that I'd like to put in a computer that only accepts half height pci cards - think it would be possible to just cut the metal plate to make it fit? |
11:18.44 | *** join/#asterisk zoa (~zoa@dD577788C.access.telenet.be) |
11:18.46 | zoa | hello htere |
11:19.28 | ManxPower | If you are using analog ports Asterisk will consider the call answered when it finishes dialing. |
11:19.39 | brimstone | :/ |
11:20.00 | brimstone | i only got one analog channel to play with |
11:20.04 | Zeeek | brimstone what are you trying to do? |
11:20.14 | brimstone | just playing with asterisk |
11:20.37 | brimstone | i'd like for the one fxo to just ring the one fxs like a simple pass through |
11:20.40 | brimstone | that's the current goal |
11:21.09 | Zeeek | and you can't call the fxo? |
11:21.21 | brimstone | i can now, just got that working |
11:21.33 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
11:22.00 | ManxPower | I can't even figure out how to use these damn things. |
11:22.07 | Zeeek | which one Manx? |
11:22.25 | ManxPower | Zeeek, The two that I downloaded. |
11:22.28 | ManxPower | 8-) |
11:22.43 | ManxPower | k http://sourceforge.net/projects/virtual-desktop/ is the one I'm trying to look at now. |
11:22.44 | Zeeek | you mean make clean; make ; make install didn't work? |
11:22.57 | ManxPower | ran the installer. |
11:23.00 | Zeeek | and? |
11:23.03 | Zeeek | nothing? |
11:23.07 | ManxPower | no virtual desktops, no icon, no nothing. |
11:23.17 | Zeeek | the trojan is installed and talking on port n? |
11:23.24 | ManxPower | It's prolly uploading my harddrive to a server in russia. 8-) |
11:23.26 | Zeeek | this is Windows - you have to reboot |
11:23.57 | Zeeek | the sourceforge one has the sources obviuosly |
11:24.10 | ManxPower | Installing Linux would be easier. |
11:24.16 | Zeeek | that's an option |
11:24.26 | Zeeek | then use wine :) |
11:24.34 | ManxPower | Zeeek, not really. I need windows for pcANYWHERE at least. |
11:24.51 | Zeeek | that's some strong medecine |
11:24.55 | Zeeek | pcA |
11:25.11 | Zeeek | seriously the sourceforge thing looks good |
11:25.20 | ManxPower | On my desktop I can at least run vmware. |
11:25.37 | brimstone | wow, that wasn't hard to do at all... |
11:25.54 | ManxPower | Zeeek, Install it, let me know how to use it 8-) |
11:26.32 | Zeeek | Don't need, went XP everywhere |
11:26.38 | Zeeek | it's worth it |
11:26.46 | ManxPower | Will XP run well on a 600Mhz machine? |
11:27.02 | Zeeek | I'v only seen a single blue screen once in 400+ days of 4 PC |
11:27.11 | afe | how much memory do you have in it? |
11:27.18 | Zeeek | I'd say if Win2k runs, XP might |
11:27.28 | ManxPower | I've not seen any BSODs since I got the laptop 6 months ago. |
11:27.36 | Zeeek | you have at least 256M ? |
11:27.47 | ManxPower | Can you get rid of that sissy GUI on XP? |
11:27.56 | Zeeek | more or less |
11:27.56 | RaYmAn-Bx | yeah |
11:28.06 | afe | I wouldn't run XP with less than 512 MB and yes, you can turn off the sissy stuff :) |
11:28.12 | ManxPower | So it looks like a computer, rather than a hello kitty playland? |
11:28.16 | coppice | you *must* get rid of that GUI unless you have a fast machine |
11:28.41 | coppice | if its was hello kitty like it would be pink |
11:28.46 | coppice | very pink |
11:28.49 | Zeeek | in defense of XP, it handles stuff like USB a lot better than Win2k |
11:28.57 | *** join/#asterisk cero64 (ruiner@fantab.ulo.us) |
11:28.57 | Zeeek | they're all pink INSIDE |
11:29.00 | ManxPower | "Oh, look, such cute little icons, *pet* Ouch! That icon bit me! Oh god they are swarming! Help! Help!" |
11:29.04 | afe | I went the other way and installed lots of extra eyecandy :) |
11:29.20 | coppice | but any third party USB driver with the windows logo certification will blue screen |
11:29.23 | Zeeek | Manx it will popup and lecture you about having too many UNUSED icons |
11:29.42 | ManxPower | zeedo, I really hope you are joking. |
11:29.44 | Zeeek | coppice that'w why the blue screen driver was invented |
11:29.55 | Zeeek | Marxo, no not at all |
11:29.57 | RaYmAn-Bx | you can just switch off desktop icons :> |
11:30.01 | ManxPower | Someone kick zeedo. He's messing up my auto complete. |
11:30.15 | Zeeek | It's Zeppo actually |
11:30.32 | coppice | i dunno what testing the MS labs do on these third party drivers, but its a joke. |
11:30.43 | afe | ManxPower: unless you tell xp not too, it will eventually pup exactly those things :) |
11:30.51 | ManxPower | Anyway, I didn't find anything suitable so I'll just live with it. |
11:30.55 | afe | pup=pop |
11:31.07 | ManxPower | I ran Win98 until 6 months ago. |
11:31.09 | Zeeek | you tied the sourceforge dealie ManxPower |
11:31.11 | Zeeek | ? |
11:31.16 | RaYmAn-Bx | ManxPower: considered a shell replacement? :> Like blackbox for windows or whatever? |
11:32.01 | RaYmAn-Bx | I think they can do multidesktop thingy..and no startmenu (or desktop icons..) |
11:32.04 | ManxPower | Zeeek, Yes. It happily installed, installed the "NT Service" then did nothing. No icon, no taskbar, when I run the menu option for it it happily accessed the HD for a min or 3 and then...did nothing. |
11:32.41 | ManxPower | RaYmAn-Bx, I basically want the GNOME windows manager and desktop pager applette for Win2k |
11:33.08 | RaYmAn-Bx | cygwin-X? |
11:33.22 | ManxPower | RaYmAn-Bx, I was looking for something simple. |
11:33.51 | ManxPower | It's not like I'm going to spend a lot of time getting it to work, I only use the win2k machine when my linuxbox has hardware problems or when I'm on the road. |
11:34.52 | ManxPower | The Win32 "window manager" is pretty similar to my GNOME desktop. A "start" menu, a task bar, an applette ssection. It's just missing a desktop pager. |
11:36.02 | afe | anyone knows if it would be possible to connect a digium TE100P to a PRI interface in an old LG PBX? |
11:39.12 | ManxPower | afe, We don't generally care what the far end hardware is, as long as it supports a protocol Asteriskk supports (like PRI) |
11:39.29 | *** join/#asterisk dreamcode (~je@81.181.199.39) |
11:39.59 | afe | ManxPower: ok, so I guess then it would be possible to use it to get 32 channels between asterisk and the pbx? |
11:40.15 | ManxPower | afe, Well 30 channels, since 2 channels an an E-1 are used for signaling. |
11:40.29 | ManxPower | sorry, 2 channels on E-1 PRI are used for signaling |
11:40.49 | afe | oh, well that's enough :) |
11:40.56 | coppice | 1 channel for signalling, and one for sync |
11:41.12 | ManxPower | afe, The PBX could be doing something stipud, but if it's PRI, it SHOULD work. |
11:41.16 | *** join/#asterisk jeffik (~jeffik@m807a36d0.tmodns.net) |
11:41.30 | afe | unfortunately, the expensive part to make it work is probably hiring someone to configure the old pbx ... |
11:46.39 | Mavvie | 15:43:25.727288 10.192.15.229.1177 > 10.192.0.2.53: 48+ Type1907 (Class 29802)?. (33) [tos 0x60] |
11:46.52 | Mavvie | that's what my cisco phone (7970) uses for a DNS packet. |
11:46.57 | dreamcode | why is asterisk asking to autentificate a user from an SER server which is calling me ? |
11:49.12 | dreamcode | or.. how do i set asterisk to match first the to then the from ip in sip.conf |
11:49.27 | *** join/#asterisk gst (~gst@wireless.sysfrog.org) |
11:49.54 | ManxPower | dreamcode, You really can't. You want to either match on username/secret or match on ip address, but you don't want to do both. |
11:50.37 | ManxPower | dreamcode, I seem to recall that SER does not authenticate at all. You need to use allow/deny and match on IP address. |
11:51.04 | ManxPower | dreamcode, How many thousand users will you have? |
11:51.13 | dreamcode | ManxPower: my problem is :that i want to foward call on ser server, and also to be able to receive calls from the same ser server |
11:51.31 | Zeeek | Entropy infers that nothing in the universe can ever be "unlimited" |
11:51.32 | dreamcode | i just use one user on the ser server |
11:51.44 | ManxPower | dreamcode, don't use SER. |
11:52.04 | Zeeek | ManxPower did you look at the control panel after installing any of those desktop apps? |
11:52.07 | dreamcode | but that's my VOIP provider |
11:52.27 | ManxPower | dreamcode, then use permit/deny to allow SER to connect. |
11:52.57 | dreamcode | ok.. thx.. at that i didn't thought |
11:53.51 | lidl | if i buy a cisco 7940g, will i be able to use it with * ? |
11:56.00 | gst | does with IAX the voice traffic between 2 clients always travel through the server or is IAX able to transfer the voice stream directly between 2 clients (like with SIP)? |
11:56.33 | ManxPower | gst, Yes. |
11:56.47 | ManxPower | lidl, Yes, if yo have the SIP firmware (extra cost) |
11:57.13 | gst | ManxPower: tnx |
11:57.13 | Zeeek | ManxPower I just installed the VDM on SourceForge - it works beautifully |
11:57.19 | ManxPower | lidl, Cisco phones only come with the phone, no power and no SIP firmware. Both are extra cose ($45 for power, $125? for SIP firmware) |
11:57.32 | ManxPower | Zeeek, I've already moved on. |
11:57.35 | Zeeek | The switching is donbe with Alt-1 2 and 3 |
11:57.43 | ManxPower | lidl, Consider Polycom. |
11:57.59 | Zeeek | yeah but it works - seems to do what you wanted |
11:58.35 | ManxPower | Zeeek, it puts a little pager between the windows on hte task bar and the clock in the lower right? |
11:59.03 | Zeeek | NO NONE OF THAT |
11:59.06 | Zeeek | ooops |
11:59.15 | Zeeek | it doesn't put anythiong anywhere |
11:59.23 | ManxPower | Zeeek, Um, that's what the Linux pagers do and that's what I was looking for. |
11:59.31 | Zeeek | it just uses Alt1 2 or 3 to switch between three desks |
11:59.49 | Zeeek | well, I installed it just for you :) |
11:59.55 | ManxPower | Zeeek, Thank you. |
12:00.55 | *** join/#asterisk LorenzoMarouani (~LorenzoMa@AVelizy-112-1-27-252.w80-13.abo.wanadoo.fr) |
12:00.56 | Zeeek | fyi if you even need to install windows programs, there is usually docs in the directory |
12:01.04 | lidl | ManxPower, thx for the hints |
12:01.16 | Zeeek | they could have called it README but they chose desktop.html |
12:01.18 | Mavvie | http://weblog.barnet.com.au/edwin/000094.html <- yippee cisco |
12:01.28 | ManxPower | lidl, polycom comes with power and SIP firmware (if you get the right model). |
12:01.29 | Zeeek | ManxPower a little ironic |
12:01.32 | LorenzoMarouani | Hi |
12:01.40 | ManxPower | lidl, Neither Polycom nor Cisco support their phones with Asterisk |
12:02.23 | LorenzoMarouani | Someone can tell me if there is an handler on incomming channel event in astersik ? |
12:03.24 | LorenzoMarouani | I need to dev a module, and pass out extensions |
12:05.42 | afe | you guys know the typical street price of a polycom 600 in the US? |
12:05.56 | afe | or a cisco 7940 with SIP |
12:06.10 | Zeeek | afe I'd gueszs around $340 |
12:06.43 | lidl | ManxPower, so you wouldn't recommend policom either, would you? |
12:06.48 | afe | hmm... that's about 50% of what it costs here (Sweden) |
12:06.48 | ManxPower | lidl, If you want a cheap, but good phone, consider the SIPura SPA-841 |
12:07.04 | ManxPower | lidl, Polycom is the Official VoIP Phone for my customers. |
12:07.30 | Zeeek | I keep thin king this is a Polycom dealer : http://aticom.com/ |
12:07.49 | ManxPower | The SIPura is not a "pretty" as Polycom or Cisco, but it works well. It doesn't have a second ethernet port and it doesn't support PoE, however. |
12:08.11 | Zeeek | BT102 has a second port |
12:08.23 | afe | Zeeek: Umm... that page was... well, interesting |
12:08.39 | Zeeek | I know there is a store that sells them with a URL near that |
12:09.03 | ManxPower | You can frequently get pretty good discounts on polycom stuff if you find the right dealer. |
12:09.13 | afe | I have a collegue going to NYC in a couple of weeks, but he'll only be there for 4 days, so no time to order it I'm afraid |
12:09.27 | lidl | ManxPower, let's say I could pay 200Ī (or dollars) for a phone. is it policom a good choice? |
12:10.00 | ManxPower | lidl, The polycom IP 300 should be available for under $200 |
12:10.04 | ManxPower | The SIPura is only $100 |
12:10.28 | Zeeek | here's one afe: http://www.voipsupply.com/product_info.php?cPath=95_107&products_id=251 |
12:10.52 | Zeeek | $199.95 no supply or shipping |
12:11.44 | Zeeek | ManxPower the Sipura has had a few bad reports on the way the phone is built, sticking keys, etc |
12:12.50 | ManxPower | Zeeek, I know. I've not had significant problems with my two SPA-841 phones |
12:12.59 | ManxPower | As long as you use the latest firmware |
12:13.27 | Zeeek | the URL I was looking for is http://www.atacomm.com |
12:13.42 | ManxPower | It will prolly become the Recommended Official Manx Power Personal Phone |
12:14.02 | ManxPower | Neither ATAcom nor VoIP supply seem to have great prices. |
12:14.23 | Zeeek | Cisco 7960G with SIP $379 |
12:15.02 | Zeeek | ip500 $209 + ship |
12:15.17 | Zeeek | not good? |
12:15.22 | Zeeek | where should I look? |
12:15.47 | ManxPower | Zeeek, Maybe they reduced their pricing. |
12:16.03 | ManxPower | IP 500 for $210 is not a terrible price. |
12:16.21 | ManxPower | Keep in mind that if you want PoE on the 300 or 500 it's $30 for the special cable. |
12:16.21 | Zeeek | includes the 110-250 PS |
12:16.28 | Zeeek | I do not want POE |
12:17.07 | Zeeek | I'm told I'd pay $220 including PS and shipping - I though that was decent |
12:17.40 | Zeeek | the ip600 is about $100 more for the person asking about that model earlier |
12:18.08 | afe | Zeeek: the first page had 7960:s for $299 |
12:18.33 | ManxPower | afe, I'm sure that's with the Cisco SCCP firmware. |
12:18.45 | ManxPower | and no power supply, of course. |
12:18.55 | Zeeek | <PROTECTED> |
12:18.59 | afe | ManxPower: actually, that was with the power supply |
12:19.11 | ManxPower | they must be getting them cheaper then. |
12:19.24 | ManxPower | I should look at ATAcomm again if they have lowered their prices. |
12:19.50 | Zeeek | no wait $319.95 includes SIP |
12:19.53 | coppice | speaking of atacomm, what happened to ipvolution? |
12:20.11 | afe | however, if I would order it the shipping would cost a lot, and I might have to pay customs and VAT as well :/ |
12:20.24 | Zeeek | afe no have your friend bring it back |
12:20.48 | ManxPower | coppice, I don't know, but I suspect the usual problem of "Wow, this is a hell of a lot more complicated and expensive than I thought!" problem. Which I refer to as "The Wasim Problem". |
12:20.53 | afe | Zeeek: yeah, but I'm not sure he'd be able to get it his hotel fast enough |
12:21.03 | Zeeek | <PROTECTED> |
12:21.17 | Zeeek | usually they won't deliver to hotel anyway |
12:21.41 | Zeeek | he's have to go pick it up +$40 taxi (assuling he takes the subway there) |
12:21.59 | afe | they're in NYC? |
12:22.27 | ManxPower | Fly to NYC for vacation, then write it off as a business expense since you were there to pick up the phone. |
12:22.32 | Zeeek | heh |
12:22.54 | Zeeek | I thought they were in Joizy actaulluy but I suddenly can't find that info |
12:23.08 | Zeeek | Buffalo, NY 14225 USA |
12:23.17 | Zeeek | close |
12:23.19 | ManxPower | That's not even close to NYC |
12:23.27 | Zeeek | close phonetically |
12:23.37 | ManxPower | I think Buffalo is closer to Toronto than to NYC. |
12:23.43 | afe | on a 4 day holiday, I guess he might be reluctant to go to Buffalo :) |
12:23.46 | Zeeek | ok, vacation in Toronto |
12:23.53 | lidl | how about a clipcomm phone? has anyone tried http://www.voipsupply.com/product_info.php?cPath=95_106&products_id=307 ? |
12:24.06 | Zeeek | there is no such thing a 4-day holiday for Europeans! |
12:24.20 | afe | unless I can convince him he needs to see the Niagara Falls :) |
12:24.40 | Zeeek | what about a cheaper Cisco? |
12:24.57 | *** join/#asterisk feral_kid (~not@209.205.207.130) |
12:25.03 | Zeeek | ah no, doesn't look very good |
12:25.33 | afe | I can get the low end ciscos for about $100 here, but they look like crap |
12:25.43 | ManxPower | I was at Niagra Falls (Canada) last summer for a couple of days. Pretty cool, but very touristy. |
12:25.44 | Zeeek | no SIP, no speakerphone |
12:26.19 | Zeeek | watch out, those prices are sometimes for refurbished phones on voipsupply |
12:26.48 | afe | and to get a 7940 with SIP here, I'd have to get it from a certified Cisco dealer = $$$ |
12:27.04 | Zeeek | or order from Russia :) |
12:27.35 | feral_kid | I just bought a Tiger Jet Network card of of E-Bay. Although ztcfg shows that one a channel is on, I can't get dialtone of the the card... Is there something about that card that I don't know about? |
12:27.57 | Zeeek | you mean an X100P clone? |
12:28.06 | feral_kid | Zeeek: Yes |
12:28.10 | afe | I can travel to russia for about $150 to get one, but I hate that place :) |
12:28.10 | Zeeek | that's a FXO card to be connected tot he phone line |
12:28.28 | Zeeek | you mean you can't make the card talk to PSNT? |
12:29.07 | feral_kid | Isn't that both a FXO/FXS card? |
12:29.08 | Zeeek | feral_kid it's connected to PSTN and you aren't able to get it to go offhook? |
12:29.15 | Zeeek | errrrr no, not at all |
12:29.24 | lidl | on voipsupply a polycom ip500 is sold at 200$ |
12:29.26 | Zeeek | I've never heard of such an animal |
12:29.40 | Zeeek | so we were saying lidl |
12:29.50 | afe | feral_kid: you can plug a phone into the phone jack on it, but it won't have anything to do with asterisk |
12:29.51 | Zeeek | $220 with power supply |
12:30.04 | lidl | i would use POE |
12:30.16 | ManxPower | Which is really weird because Polycom always ships with a power supply, as far as I know. |
12:30.23 | Zeeek | feral_kid actually there is a phone jack but it's for when the card is not powered (PC off) |
12:30.40 | lidl | ManxPower, so i'm compelled to buy the powersupply? |
12:30.46 | Zeeek | Manx ya they do it to make it look cheaper - in fairness it's a good option |
12:30.55 | feral_kid | Zeeek: Ah... That was what I was missing... :) |
12:31.03 | Zeeek | lidl not if you already have a good 12v |
12:31.06 | ManxPower | lidl, It's not like the power spupply adds more than a few dollars to the phone. |
12:31.10 | feral_kid | Zeeek: NOw it becomes clear... |
12:31.28 | Zeeek | feral_kid you need an FXS or an ATA (cheaper) |
12:31.42 | ManxPower | lidl, I suppose they may have a version without power, but I've never heard of anyone getting the wrong product and screaming on the mailing lists about lack of power supply. |
12:31.46 | afe | feral_kid: actually, it works when it's powered as well, but it will "steal" the line from asterisk |
12:32.05 | feral_kid | Zeeek: I have two Sipuras, but I was was just testing out the card... |
12:32.06 | ManxPower | The IP 300 and IP 500 both require special PoE cables. |
12:32.12 | Zeeek | afe does it ring when the line rings? I guess so |
12:32.19 | afe | Zeeek: yes |
12:32.38 | Zeeek | I ran for a few weeks with phone in parallel |
12:32.41 | Zeeek | it sucks |
12:32.43 | afe | it's really not that bad to use as a backup |
12:32.59 | Zeeek | the asterisk phones will keep ringing for one or two rings |
12:33.02 | ManxPower | The two ports on the X100P are hardwires togather. |
12:33.07 | lidl | ManxPower, are the 'special' cables shipped with it, or si it an extracost? |
12:33.11 | Zeeek | $30 |
12:33.18 | Zeeek | so you see |
12:33.20 | afe | the tdm11b is a much better card |
12:33.26 | Zeeek | no free lucnh anywhere |
12:33.28 | ManxPower | lidl, PoE cables are $30 from what I understand and are a seperate product. |
12:33.36 | Zeeek | or lunch even |
12:33.39 | lidl | :/ |
12:33.50 | Zeeek | except maybe in prison |
12:34.01 | Zeeek | speaking of lunch.... |
12:34.05 | afe | if now only digium support could get my dialback problem fixed :) |
12:34.07 | feral_kid | Zeeek: The only reason I picked up the X100 clone was so I could play around with FWD-IN |
12:34.20 | Zeeek | FWD-IN ? |
12:34.29 | Zeeek | I know about FWD OUT |
12:34.55 | Zeeek | get a free software phone and use that for now |
12:35.02 | *** join/#asterisk nassy (~mark@24-193-228-118.nyc.rr.com) |
12:35.04 | feral_kid | Zeeek: It is 4:30A, so by this time I don't know IN from OUT... :) |
12:35.14 | Zeeek | haha |
12:35.21 | Zeeek | watch behind you then ;) |
12:35.25 | marlowe | hahah |
12:38.29 | afe | anyone knows if SJPhone or X-Pro for pocketpc works with a bluetooth headset? |
12:40.12 | lidl | on the specs i read the ip500 supports multilanguage, but it doesn't show what languages are available. does anyone know? |
12:40.17 | marlowe | yesitdid on mine |
12:40.52 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
12:41.36 | *** join/#asterisk [Paul] (~paul@80.100.33.108) |
12:41.53 | [Paul] | hi |
12:42.26 | [Paul] | i've just upgraded eyebeam to the newest version (3000W) |
12:42.30 | [Paul] | 3004W |
12:42.48 | *** join/#asterisk NosDe (~joernhall@c224016.adsl.hansenet.de) |
12:42.52 | [Paul] | when i try to make a call i always get a 407 error |
12:43.20 | [Paul] | when i reinstall 3002S (i think) it works fine |
12:43.25 | NosDe | hi. anyone here with some skills in chan_capi ?? |
12:43.42 | [Paul] | and when i'm in my lan it also works |
12:43.49 | [Paul] | but via NAT it doesn't |
12:44.22 | Zgarbi | I just run Asteriks for first time and dial to number 1234 but answer quality is very bad, like zigzaged voice. may it's because cpu=600MHz and RAM=128k on fedora core 3? |
12:44.53 | [Paul] | i'm running asterisk on 500mhz and 128mb and that works fine |
12:45.32 | NosDe | i'm running * an an pentium 233 (down to 90 MHz) and 128MB |
12:45.53 | Zgarbi | so strange... :( |
12:46.59 | Zgarbi | I tryed to change codecs on my x-lite, but same, ping between mycomp and asterixhost less then 5ms |
12:47.13 | afe | I first ran * on a PII 333 and it was ok |
12:47.43 | Zgarbi | so what can be a problem? |
12:47.45 | Zeeek | Zgarbi do you use ulaw? |
12:47.48 | afe | 128 k (k???) might be a bit low ;) |
12:48.12 | Zgarbi | I don't konw what is ulaw, i'm newbie |
12:48.19 | Zgarbi | 128M |
12:48.36 | Zgarbi | sorry |
12:48.37 | afe | I use fedora 2 and it eats quite a lot of memory |
12:48.50 | afe | Zgarbi: I guessed that, just teasing you ;) |
12:49.15 | Zgarbi | Cpu(s): 1.0% us, 12.1% sy, 2.0% ni, 83.3% id, 1.6% wa, 0.0% hi, 0.0% si |
12:49.15 | Zgarbi | Mem: 125704k total, 119472k used, 6232k free, 1084k buffers |
12:49.15 | Zgarbi | Swap: 1020116k total, 45864k used, 974252k free, 16520k cached |
12:49.38 | Zeeek | Zgarbi the codec being used on X-Lite is shown during the call - what codec do you see? |
12:49.49 | Zgarbi | GSM |
12:49.54 | afe | ulaw is a sound codec, and if asterisk does codec translation it might get CPU heavy |
12:49.59 | Zgarbi | I has changed it on 711 |
12:50.01 | Zgarbi | but same |
12:50.36 | afe | in the asterisk CLI, type show translation |
12:50.39 | Zeeek | Is that the onbe that is selected suring the call? |
12:50.41 | NosDe | does anyone have some skills in compiling chan_capi (junghans) with capi20 (avm) for * |
12:50.46 | Zgarbi | I using x-lite to connect other host (not mine) and it's works perfect |
12:51.48 | *** join/#asterisk izo (~izo@izo.warpl.ipxxi.pl) |
12:52.37 | *** join/#asterisk D1ng0 (~dingo@202.57.43.4) |
12:54.44 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
12:56.07 | D1ng0 | hello from the warm tropical phillipines |
12:57.19 | coppice | hello from a cold wet miserable place just a short flight from you :-) |
12:57.31 | marlowe | hello from nj. |
12:57.33 | marlowe | hah |
12:57.41 | Zeeek | whadya mean? |
12:58.07 | afe | ~weather ESSA |
12:58.30 | afe | so don't complain about weather ;) |
12:58.36 | Zeeek | Stockholm is a beautiful city |
12:58.47 | coppice | its OK |
12:59.32 | Zeeek | coppice your favorite city? |
12:59.33 | marlowe | ~weather 07746 |
12:59.38 | afe | I'm actually in Uppsala (70 km from Stockholm), but no weather code for that ;) |
12:59.39 | marlowe | darn |
12:59.40 | marlowe | lol |
12:59.58 | marlowe | ill load weather.com |
13:05.39 | marlowe | ~weather kttn |
13:05.44 | marlowe | I knew that too. |
13:06.23 | marlowe | It should say snowing |
13:06.29 | marlowe | 'cause it is out my window |
13:06.45 | Zgarbi | all right, I didnt understand what problem was but as I see there was 2 answer machine was executed together and sound was mixed. wher first finished second was fine |
13:06.53 | *** join/#asterisk Chuji (Chuji@pcp09930052pcs.tulipgrove.tn.nash.comcast.net) |
13:08.18 | Eight | aaaand BV is broken again. |
13:08.22 | Eight | surprise surprise. |
13:10.31 | Eight | heh, funny you guys should be playing with jbot's weather feature. |
13:10.48 | Eight | I'm in the middle of writing up a weather thing in python for Asterisk =p |
13:10.58 | Eight | it works with zip codes though =) |
13:11.53 | Grooby | ~weather taiwan |
13:12.02 | Grooby | doh! |
13:12.19 | coppice | taiwan is cool and damp right now |
13:13.16 | Eight | http://www.voip-info.org/tiki-index.php?page=Asterisk+settings+Broadvoice |
13:13.29 | Eight | You guys have any ideas as to why the second example might STOP working? |
13:14.11 | Grooby | BV can go out |
13:14.15 | Grooby | let me dial my self |
13:14.21 | Eight | I can dial out, but not in. |
13:14.29 | Zgarbi | bye all |
13:14.40 | Eight | and I haven't changed anything in the register => section in ages. |
13:14.45 | afe | http://hem.bredband.net/b282251/images/IMAGE_00003.jpg = view from my window right now :) |
13:14.49 | Eight | sometimes it works, sometimes it doesn't. |
13:14.56 | Grooby | works for me |
13:15.03 | Grooby | i never had any problems.... |
13:15.23 | Eight | Grooby: could you post your sip.conf as a third example on the wiki? |
13:15.50 | Eight | or atleast just pastebin for now. |
13:16.06 | Eight | and I'll work out the differences and make the changes to the second example. |
13:17.45 | Grooby | just remember I am using AMP |
13:17.52 | Eight | AMP? |
13:17.56 | Eight | jbot AMP? |
13:17.57 | jbot | [amp] an Audio MPEG Player. [non-free] |
13:20.21 | Grooby | try that |
13:20.52 | *** join/#asterisk carlosh (~carlosh@203-96-159-89.paradise.net.nz) |
13:20.58 | afe | umm... wouldn't that be asterisk management portal (or something like that) ... silly jbot :) |
13:21.28 | Grooby | nah |
13:21.31 | Grooby | that's win AMP |
13:21.38 | Grooby | :P |
13:21.50 | afe | ~kick jbot |
13:21.52 | jbot | bugger off sod! |
13:22.16 | afe | ~hugs jbot |
13:22.46 | carlosh | howdy all: here trying to get CallingCard Applications to work.. with postgreSQL.. having difficulties compiling I think because I haven't defied or configured postgresql client libary and header files, Anyone please care to help me doing this? thanks. |
13:23.17 | carlosh | defined |
13:27.13 | Grooby | you there eight? |
13:27.18 | Grooby | did you see my pastebin? |
13:29.36 | Grooby | ok..i am off |
13:29.39 | Grooby | good luck Eight |
13:30.25 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
13:34.22 | feral_kid | Anyone using Asterisk@Home? |
13:34.35 | *** join/#asterisk threat|BX (threat@dsl-41.16.240.220.rns02-kent-syd.dsl.comindico.com.au) |
13:34.38 | threat|BX | G'Day |
13:34.44 | afe | howdy |
13:35.46 | Zeeek | Broadvoice questions, RFC: does BV encourage, discourage use of their service witrh asterisk (or are they indifferent)? |
13:35.59 | carlosh | howdy all: here trying to get CallingCard Applications to work.. with postgreSQL.. having difficulties compiling I think because I haven't defined or configured postgresql client libary and header files, could anyone please help me with this? thanks. |
13:36.25 | Zeeek | Is there a compelling reason why BV or any provider should encourage (or not) the asterisk community |
13:36.57 | afe | Zeeek: maybe they don't wan't their customers to be able to provide services for other people? |
13:37.30 | feral_kid | I tried to set up a trunk for FWD using iax2.fwdnet.net... I have yet to make a call through FWD (using iax2 or for that matter using just fwd)... Anyone know how to get this working behind a double NATted machine? |
13:37.37 | Eight | Zeeek: There was an article where a BV rep' discussed people using Asterisk with BV. Shouldn't be too hard to find. He welcome the market, but said 'you're on your own' for support. |
13:37.57 | Zeeek | what d'you think of his comment, Eight? (in a few words!) |
13:38.37 | Eight | That more service providers in more industries should have that stance. |
13:38.52 | Zeeek | no support but c'mon in? |
13:38.58 | Eight | I dislike having a cookie cutter applied to me and getting parts I like lopped off. |
13:39.12 | Zeeek | is BV broken for everyone or just asterisk (several weeks now, all the postings) |
13:40.34 | afe | I'm using a Swedish provider (Rix Telecom) - they don't officially support asterisk, but has an asterisk forum |
13:40.46 | afe | and they're using asterisk for their service |
13:41.32 | Eight | Zeeek: I got it working again... |
13:41.39 | Eight | I had to put 'insecure' in for them to log into me. |
13:41.57 | Eight | I kept removing it because when I did so it KEPT working. |
13:42.08 | afe | Eight: I had to put insecure=very for my provider as well - otherwise it didn't work |
13:42.19 | Zeeek | but there has been so much talk about them recently, I wondered a lot about the compelling reasons on both sides |
13:42.33 | Zeeek | FWD requires that too IIRC |
13:42.40 | Zeeek | but FWD is free as we all know |
13:42.51 | afe | does bv limit the number of simultaneous calls (spelling? :)) |
13:44.39 | Eight | afe: afaik they didn't used to, but do now. |
13:44.57 | Eight | I can not, for example, call in and out at the same time. |
13:45.03 | Eight | atleast, however I tried it didn't work. |
13:45.10 | afe | Eight: that kinda sux |
13:45.40 | Eight | oh, hey... I guess I can. |
13:46.20 | afe | at work we use asterisk for incoming support calls, and there's no limit in the number of calls for just one account :) |
13:47.36 | afe | our provider, however, refuses to sell their service to call centers and call marketing companies (for obvious reasons) |
13:48.28 | threat|BX | afe hows it going? |
13:49.09 | afe | threat|BX: just fine, thanks :) But, did you mean with anything in particular? |
13:49.13 | Zeeek | I was kind of worndering whether when possible we should strive to support service providers that try to lean twoards, not against asterisk |
13:49.15 | Eight | aha... |
13:49.26 | Eight | I just did a PSTN->BV->Asterisk->BV->PSTN call. |
13:50.00 | *** join/#asterisk Tili (~Tili@202-133-67-12-dialup.sat.net.pk) |
13:50.09 | Eight | Zeeek: I think that'll happen naturally. |
13:50.19 | Eight | Zeeek: if anyone gets a good rep it'll spread pretty quick. |
13:50.59 | Eight | BV seems to be the leader now because they have unlimited minutes plans, and a fairly broad selection of DIDs. |
13:51.11 | threat|BX | afe I want to know more about asterisk :) in particular what I need to know for it to be a standalone PBX replacement |
13:51.21 | threat|BX | afe or if its feasable :) |
13:51.26 | Eight | threat|BX: it is very feasible. |
13:51.32 | threat|BX | afe what hardware I need, etc.. |
13:51.35 | Eight | threat|BX: describe your current setup? |
13:52.17 | *** join/#asterisk nix000 (~nix000@66.11.191.103) |
13:52.18 | afe | threat|BX: I suggest you take a look at the wiki - that's the best place to start |
13:52.22 | afe | ~wiki |
13:52.30 | Eight | jbot wiki? |
13:52.31 | jbot | hmm... wiki is http://www.voip-info.org |
13:52.43 | nix000 | anyone can recomend a good ss7 gateway that works with asterisk ? |
13:52.48 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
13:52.50 | threat|BX | Eight small business, four lines, probably need ~10 - 20 phones connected to the system |
13:53.00 | *** join/#asterisk Goldenear (~Nicolas@d193.dhcp212-198-200.noos.fr) |
13:53.10 | Zeeek | Eight yes that is the case (good/bad rep being known in the "community") |
13:53.13 | Eight | threat|BX: keep your existing phones, or switch to new ones? |
13:53.30 | *** part/#asterisk Goldenear (~Nicolas@d193.dhcp212-198-200.noos.fr) |
13:53.48 | Zeeek | threat|BX here a few decent links with some into stuff |
13:53.49 | Zeeek | Starter tutorial: |
13:53.49 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
13:53.49 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
13:53.49 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
13:53.49 | Zeeek | THE reference of the moment: |
13:53.50 | Zeeek | http://www.asteriskdocs.org |
13:53.50 | threat|BX | Eight This would be for a new branch office for this particular company, so all new equipment |
13:53.58 | threat|BX | Zeeek thanx :) |
13:54.03 | Eight | Just a fairly simple computer (anything modern), with atleast one TDM400P card in it to talk to the analog lines (4 ports per card). |
13:54.11 | Eight | And then put SIP phones on the network. |
13:54.23 | Eight | the card is ~320 bucks with 4x FXO. |
13:54.40 | Eight | the phones are anywhere from 60->350 depending on how fancy the model is. |
13:54.53 | threat|BX | I see |
13:55.03 | Eight | Some have pass-through ability so you can put them between the computer and the rest of the network on each desk, no extra wiring. |
13:55.38 | Eight | An' that's it. |
13:55.40 | Zeeek | Eight good summing up! |
13:56.04 | Zeeek | Someone needs to write a paper on the pbx user experience though :) |
13:56.13 | Zeeek | believe me, it's not protty |
13:56.16 | Zeeek | or pretty |
13:56.18 | threat|BX | ok, I would need to get a price on a PBX system with phones for a comparison, but the astericks solution sounds pretty good |
13:56.23 | Eight | threat|BX: be warned, however. The actually configuration is extremely arcane. |
13:56.27 | *** part/#asterisk srt (~nobody@gw0-cgn.reucon.net) |
13:56.52 | Eight | the software is free, the hardware is inexpensive, but the man-hours put into learning the configuration can be intensive. |
13:56.52 | Zeeek | threat|BX you can set up an asterisk pbx for way less than a classic pbx, but there are disadvantages |
13:57.17 | Eight | If you find a local company that can do asterisk install/support, go for it. |
13:57.28 | Eight | If you have the time to dedicate to it yourself, go for it. |
13:57.36 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
13:57.42 | Zeeek | only if you have no job and aren't married though |
13:57.43 | Eight | If you want to just plug it in and go... use something else. |
13:58.08 | Zeeek | service providers sell plug and play including a day of setup and config |
13:58.18 | threat|BX | Eight what are some common problems that I may encounter? phone calls drop out? asterisk starts randomly playing voice messages at inapropiate times?? :) |
13:58.30 | afe | Eight: I agree man-hours cost, but if you're true geek (like most of us) it's just fun :S |
13:58.37 | Zeeek | yup |
13:58.43 | Eight | ya, I'm just playing with it right now. |
13:58.52 | Eight | It could turn into something more soon, but for now it's just neat =) |
13:59.00 | Eight | I have all my friends setup with x-lite, and so on. |
13:59.04 | threat|BX | afe hehe i would find enjoyment out of setting it up :P |
13:59.22 | afe | threat|BX: when you get it working, you won't have any problems - its (in my experience) very stable |
13:59.33 | Eight | threat|BX: since you're asking this on a saturday, I assume you have some personal interest in it. |
13:59.40 | afe | threat|BX: it's the first time configuration that takes time |
13:59.51 | threat|BX | afe can asterisk be comfigured for one line using a cheap ass modem? or is it more envolved then that? :) (basically just want voice messages if no answer etc.. :)) |
14:00.08 | afe | threat|BX: yes |
14:00.11 | Eight | threat|BX: now that you know it *can* server your purpose... try tinkering with it. setup a computer with it and tinker. See if it's something you want to learn yourself, or would rather pay someone else to do. |
14:00.19 | threat|BX | Eight hehe :) well actually its sunday morning 1:17AM :P so yes, your guess is correct :) |
14:00.21 | afe | threat|BX: if it's a X100P modem (or clone) |
14:00.31 | Zeeek | threat|BX take a look at those pages I showed. They have a plain english explanation of all the basic concepts |
14:00.44 | Zeeek | a lot of answers to your questions |
14:01.29 | threat|BX | Zeeek ok I will :) I am still asking quick questions before I dedicate my unmarried and unemployed life to reading the docs :P |
14:01.29 | Eight | the X100P works, but as far as I know 'you get what you pay for', and it can be a little flakey. Not sure what the issues are though. |
14:01.29 | afe | kinda odd actually, this community is not always very newbie friendly ;) |
14:01.30 | threat|BX | afe X100P modem aye, so you need a specific type? |
14:01.42 | Zeeek | afe I would'nt say that - except in a very few cases I've seen |
14:01.57 | afe | threat|BX: yes, that's the cheapest option available, but it's not very good though |
14:02.17 | afe | Zeeek: hmm... true, the mailing list is much more unforgiving |
14:02.18 | threat|BX | lol yes I learnt linux 5yrs back, I know how unfriendly people can get :) |
14:02.46 | threat|BX | afe well I would want cheap to test it out at home before implementing it else where |
14:03.06 | afe | threat|BX: if you have the dough, I'd suggest getting a tdm11b from digium (devkit) - works very good and allows you to connect an analog phone as well |
14:03.07 | Zeeek | threat|BX you can do a lot of playing with no hardware at all |
14:03.32 | threat|BX | Zeeek really? like what? (I would like to test it out also) |
14:03.54 | Zeeek | if you'd read the four page article you'd already know! :) |
14:04.12 | threat|BX | Zeeek LOL ok already :) I will read it, grrr :P |
14:04.22 | Zeeek | it's worth the few minutes, believe me |
14:04.33 | Zeeek | you'll have more complex questions after |
14:04.39 | afe | what really made me like asterisk is the possibility to answer my home POTS from anywhere, and for that you need hardware ;) |
14:04.49 | Zeeek | no |
14:04.58 | Zeeek | you can have a voip number at home |
14:05.05 | Zeeek | a lot of people are doing that |
14:05.13 | Zeeek | and have no POTS service even at home |
14:05.20 | afe | unfortunately, I can't here - I need a POTS for my DSL :/ |
14:05.25 | Zeeek | me too |
14:05.37 | Zeeek | but now we have offers of no-voice POTS lines |
14:05.42 | Zeeek | (for DSL) |
14:06.05 | afe | We're still waiting for that to arrive here - but the main Swedish Telco owns all the lines |
14:07.10 | Zeeek | FT owns them all here too but they are foprced to capitulate |
14:07.47 | Zeeek | most DSL offers now include voIP "unlimited" national calling |
14:07.48 | threat|BX | damn monopoly phone companies :( theres one here at AU too :( |
14:08.08 | afe | threat|BX: yeah, but it will probably change |
14:08.23 | nix000 | anyone interfaced cisco gatways with asterisk ? |
14:08.36 | Zeeek | for example there is Free who offers $40/mo 20M down DSL + unlimited phone |
14:09.28 | afe | I have an offer from my voip provider for about the same, but no traffic limit and free national calls |
14:09.46 | Zeeek | the down was speed |
14:09.51 | afe | however, atm I can only get 512KB from them |
14:09.54 | Zeeek | 20Mbs |
14:10.02 | afe | ah, ok |
14:10.05 | Zeeek | actually 16 |
14:10.09 | Zeeek | ATM |
14:10.22 | Zeeek | still very good - I don't know the upload speed though |
14:10.23 | afe | I have about 8 MBs now |
14:10.31 | afe | and 800Kbps up |
14:10.38 | Zeeek | We have 2 at the office and 1 at home |
14:10.44 | Zeeek | 256 up |
14:10.49 | Zeeek | works ok for my use |
14:10.58 | Zeeek | naturally I'd like more |
14:11.29 | Zeeek | but in many parts of the USA, 256K/64K is like $40/month |
14:11.42 | afe | right now we have some really aggressive anti-piracy stuff going on, which kinda limit the use of a high band-width ;) |
14:11.58 | Zeeek | you mean like file-sharing? |
14:12.08 | Zeeek | there si a big crackdown on that |
14:12.10 | afe | that, and private ftp archives |
14:12.16 | Zeeek | same idea |
14:12.49 | Zeeek | ~seen bacondoublechz |
14:12.55 | jbot | bacondoublechz <~bacon@69-162-37-142.stcgpa.adelphia.net> was last seen on IRC in channel #asterisk, 16h 52m 2s ago, saying: 'xantus, I was taking about the option in the call preferences menu'. |
14:12.55 | afe | two days ago they busted in at one of the ISP:s and found one of the biggest archives in Europe ... |
14:13.46 | afe | those responsible will face a lawsuit of several hundred millions |
14:13.48 | Zeeek | afe yeah they're working on EU laws to make it mandatory for ISP to provide user info |
14:14.25 | Zeeek | how large is your asterisk installation at the office? |
14:14.54 | afe | lots of people have gotten raided at home, where they take the computers and you can get both jail time and have to pay millions |
14:15.03 | Zeeek | ours is just 2FXO 3FXS plus three SIP phones outside |
14:15.19 | Zeeek | and 1 IAXy for now |
14:15.31 | afe | at office we use SIP only (for about 5 people) |
14:15.44 | Zeeek | I recently bought some music at Virgin store - 1eu per song |
14:15.48 | carlosh | guys, what's definitely the best pre-paid app. for * from your personal experience? |
14:16.12 | Zeeek | afe and what in from POTS? anything? |
14:16.31 | afe | Zeeek: atm, we just forward the pots line to a voip number :) |
14:16.55 | Zeeek | waiting to see if voIP is just a passing fad? |
14:17.33 | afe | nah, but it's a bit difficult to move the number to voip atm, so I put a recording there telling people to use the voip number instead |
14:17.54 | afe | we'll probably keep it since it's a lot cheaper, and we can record calls etc |
14:17.55 | Zeeek | this stuff will change radically when every single household is connected by high speed and you TV screen (hi-res, huge, flat on wall) will show live scenes of a beach on another planet |
14:18.18 | Zeeek | I'll gibve that about 10 years - except for the planet part |
14:18.21 | lidl | is it possible to connect iaxy to a fax? |
14:18.40 | afe | eventually, we'll try to connect our old pbx to asterisk, so everyone can call out for free |
14:18.41 | Zeeek | lidl in theory absolutely |
14:18.58 | lidl | Zeeek, thx |
14:19.08 | afe | lidl: fax and modems over IP are a bit shaky though |
14:19.10 | Zeeek | but results may vary depending on the lag etc |
14:19.16 | *** join/#asterisk sysdef (~sysdef@sysdef.admin.debiancenter) |
14:19.27 | Zeeek | well it is ulaw, but still, it better be a good connection |
14:20.07 | Zeeek | hey I wonder if I could connect my laptop via modem to the IAXy and get faxes ? |
14:20.11 | afe | if the server gets busy with some codec conversion for some other call, that could possibly affect a fax transmission |
14:20.38 | Zeeek | but then why would I want to do that... |
14:20.45 | lidl | afe, i'm going to use a p4 to serve 9 telephones and a fax, with two incoming lines |
14:21.00 | afe | lidl: that should be enough :) |
14:21.08 | lidl | :) |
14:21.21 | afe | why would anyone want to fax anything anyways? :D |
14:21.27 | Zeeek | lidl with 2 TDM400 ? |
14:22.01 | *** join/#asterisk jeffik (~jeffik@m506e36d0.tmodns.net) |
14:22.02 | lidl | Zeeek, yes, i was thinking about having a tdm400 with 2 fxo and 1 fxs dedicated to the fax |
14:22.19 | Zeeek | and the other phones would be SIP? |
14:22.26 | lidl | Zeeek, yes |
14:22.38 | Zeeek | sounds like a good plan |
14:22.51 | carlosh | anyone using the chinese IPPhone: HOP8T ? have some issues with hold music and also 178ms delay on my LAN ! (compared to 1ms using soft client) |
14:22.57 | lidl | well 7 sip voip phones and 2 iaxy/atas for two pots phones |
14:23.13 | lidl | .. and 1 fax |
14:23.50 | Zeeek | fux sax |
14:24.09 | Zeeek | fax sucks |
14:24.24 | afe | in 2004 I think we might have received a maximun of 10 faxes that actually meant anything |
14:24.24 | Zeeek | but yes, some people need them |
14:24.38 | Zeeek | my last two were both spams |
14:25.01 | Zeeek | I was so pissed I called one company and began an order for 10,000 of their machines to send to china :) |
14:25.10 | afe | lol |
14:25.32 | Zeeek | then at athe end I said and if I ever get a fax from you again, better lock your doors! |
14:25.37 | lidl | sadly the office i'm going to serve receives a lot of faxes :/ |
14:25.56 | lidl | with a mean of 1 x day |
14:26.10 | afe | if faxing is important, I'd probably just keep a separate pots for that |
14:26.23 | Zeeek | lidl the surest way wouldbe to have an FXS module and connect a faw to that |
14:26.27 | lidl | afe, so an fxs |
14:26.28 | Zeeek | fax |
14:26.28 | lidl | yes |
14:27.01 | afe | or skip the fxs and keep one analog line for the fax |
14:27.01 | Zeeek | he meant separate line and just connect the fax to it, no FXS? |
14:27.05 | Zeeek | ya |
14:27.41 | Zeeek | or use a service like j2 or efax and receive the fax by email |
14:28.07 | carlosh | for fax, I'd use an fxs direct to dedicated modem/hylafax => gfax (linux app) to get them in tiff format files... or convert to tiff using something else, and get it in email.. |
14:28.27 | Zeeek | carlosh good plan also |
14:28.42 | afe | that would work aswell :) |
14:29.20 | lidl | Zeeek, i would do so, but my customers are not that computer literate.. i'm sure they do prefer getting paper other than an email |
14:29.39 | carlosh | then, send tiff file to local printer.. lol |
14:29.45 | lidl | carlosh, :) |
14:29.49 | Zeeek | heh |
14:30.06 | afe | lidl: ouch... then I'd go the separate line route |
14:30.14 | Zeeek | compuserve used to have a service to convert email to a printed page that they then mailed! |
14:30.25 | *** join/#asterisk smurfix (~smurf@smurfix.developer.debian) |
14:30.51 | Zeeek | the previous version was delivered by a lone horseman |
14:30.53 | afe | no need to involve the asterisk for faxes if you always want them on paper |
14:31.58 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
14:32.02 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
14:32.12 | Faithful | Hey, can we peer with skype? |
14:32.18 | Zeeek | no |
14:32.21 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
14:32.23 | Faithful | boo |
14:32.33 | Zeeek | shame, in'it |
14:33.07 | lidl | afe, but using a dedicated pstn would not be feasible in their situation imho, since they just have 2 pstn lines :/ |
14:33.42 | carlosh | shame indeed.. guys, out of all the logged on the irc, there must be one using any prepaid application.. I have some problems setting any from the wiki.. any reccomendations ? |
14:33.42 | lidl | afe so connecting the fax to the fxs would let me to use both incoming lines for the two fxo |
14:33.42 | afe | lidl: then I'd make one of the modules on the two TDM400:s an FXS port |
14:34.00 | afe | carlosh: sorry mate - never tried it |
14:34.16 | Faithful | is G729 better than iLBC? |
14:34.31 | carlosh | ilbc works beautyfully on dial ups.. |
14:34.50 | Faithful | hence skype uses it |
14:34.57 | carlosh | better handling of packet losses |
14:35.10 | Zeeek | skype will prolly never peer with anyone though - except Motorola (done) |
14:35.10 | Faithful | better than G729 |
14:35.42 | Faithful | can we connect to skype with * |
14:35.46 | Faithful | at all? |
14:35.50 | afe | Faithful: no |
14:35.56 | Zeeek | two handsets taped toghter might work? |
14:36.02 | zoa | omg |
14:36.04 | Zeeek | head to tail |
14:36.08 | zoa | this jetlag is killing me |
14:36.35 | afe | Faithful: Skype might eventually get a pots number option, and that would of course work |
14:36.36 | Faithful | well you could get a Skype FXO and plug it into * |
14:37.05 | carlosh | Faithful, true |
14:37.06 | Zeeek | you can dial in ti DISA with skype paid PSTN service |
14:37.21 | afe | I don't really like Skype, since it's only purpose is to make its creator (a fellow Swede) more money |
14:37.34 | Faithful | good on him |
14:37.41 | afe | indeed |
14:37.41 | Zeeek | You need more rich people in Sweden now that Abba left |
14:37.44 | lidl | afe, so 2 fxo and 1 fxs on the TDM400, right? |
14:37.58 | Faithful | I like people with iniative |
14:38.07 | afe | lidl: yep |
14:38.13 | lidl | ok :) |
14:38.17 | Zeeek | why not build a skype pbx ? |
14:38.33 | carlosh | zoa: are you the zoa i am thinking you are who just arrived from Italy? |
14:38.40 | afe | Faithful: yeah, I guess I'm just envious :) |
14:39.05 | zoa | nah i just came from von |
14:39.17 | Faithful | I don't know why we cant plugin |
14:39.19 | carlosh | oh, sorry |
14:39.23 | lidl | what's the average uptime for asterisk? |
14:39.42 | Faithful | about 10min |
14:39.45 | afe | lidl: mine is exactly 24 hours since I restart it nighly :D |
14:39.46 | lidl | i mean, in your experience |
14:39.50 | carlosh | a few weeks.. I got it installed at a customer's .. they have only had to reset it once... |
14:40.01 | Zeeek | lidl it's very hard to tell since most of us experiemnt so much |
14:40.02 | carlosh | in 4 months.. |
14:40.02 | lidl | afe, is it a good practise to restart it? |
14:40.14 | Zeeek | common wisdom is once a week |
14:40.24 | Chuji | lidl : what kind of load are you putting on it? |
14:40.30 | afe | lidl: I don't think it's really necessary, but the mpg123 might act up (if you use music on hold) |
14:40.32 | lidl | for a production system, would you recommend to cron its restart? |
14:40.45 | Zeeek | many people do that lidl |
14:41.01 | Faithful | what about Alcatel PBX they run a form of linux I believe ... do they self reboot once a week or so (or just work) |
14:41.01 | Zeeek | as long as you can safely stop all service at some predefined time |
14:41.04 | lidl | Chuji, not much load, 2 incoming lines, and 10 voip phones |
14:41.10 | eKo1 | I don't restart, I just let it run until it dies. |
14:41.18 | Zeeek | so how ioften is that? |
14:41.28 | eKo1 | Every week. |
14:41.32 | Zeeek | really ? |
14:41.35 | Zeeek | of what? |
14:41.46 | carlosh | so, no one experience with pre-paid apps... no one with HOP8T IP Phone (Chinese), anyone experience with SER? |
14:41.48 | afe | lidl: the restart takes like 2 seconds on a small system, so if you do it night time it won't hurt anyone :) |
14:41.48 | eKo1 | All sorts of misc. problems. |
14:41.50 | Chuji | lidl : I wouldn't worry about it then |
14:42.00 | Zeeek | eKo1 tell me about this |
14:42.01 | Faithful | one down side with OSS is that we patch so many disparate bits of software together the outcome is sometimes a little hard to stabelize |
14:42.42 | lidl | does asterisk offer a something like a watchdog to check it is actually running? |
14:42.50 | afe | yikes... heavy snowing going on outside now |
14:42.58 | *** join/#asterisk [Paul] (~paul@80.100.33.108) |
14:42.58 | eKo1 | Zeeek: I've had so many problems, I can't remember any specific one anymore. |
14:43.00 | Zeeek | lidl the safe_asterisk script restarts it when it dies IIRC |
14:43.12 | eKo1 | So far: System uptime: 5 days, 10 hours, 39 seconds |
14:43.13 | Zeeek | or can anway |
14:43.26 | lidl | Zeeek, ok |
14:43.40 | Zeeek | eKo1 ok, I was wondering - mine usually is up until I recboot |
14:43.54 | Zeeek | but then I do that at least once a week lately |
14:43.56 | Faithful | for instance, someone leaving voicemail on my system creates a 25% chance that the ISDN module will drop out of the kernel and the box doesn't answer any more calls |
14:44.34 | afe | ouch |
14:44.37 | lidl | Faithful, that's quite unstable then |
14:44.46 | Chuji | afe : Where are you? |
14:44.53 | afe | Chuji: Sweden |
14:44.57 | eKo1 | Well, I think * should be stable enough to not warrant a restart at all. |
14:45.05 | *** join/#asterisk dimmik (~kold@ip-92-216.first.gr) |
14:45.07 | eKo1 | Given the right environment. |
14:45.22 | carlosh | anyone experience with status management, so you know other parties are available or not (IAX or SIP) .. ? thanks. |
14:45.27 | lidl | since the system i'm going to install is at about 1000km from where i live, i'd like a stable system |
14:46.03 | afe | lidl: such a small system, with only tdm cards would probably be perfectly stable |
14:46.09 | Chuji | carlosh : How do you want to use this *status*? |
14:46.21 | lidl | afe, thanks that's reassuring :) |
14:46.47 | *** part/#asterisk dimmik (~kold@ip-92-216.first.gr) |
14:46.47 | *** join/#asterisk d00gster (~in_ter@70.48.207.77) |
14:46.52 | Faithful | lidl: stick to digium equipment |
14:47.05 | lidl | Faithful, for sure |
14:47.17 | afe | digium + fedora works perfectly for me |
14:47.26 | carlosh | Chuji: I want to be able to tell if a SIP or IAX user is available.. or not.. similar to firefly, when you change your status, the others will see you in a different color.. |
14:48.13 | carlosh | I use fedora 3, latest kernel, latest cvs.. always very stable.. |
14:48.55 | Chuji | carlosh : what do you want to display this in? |
14:48.56 | afe | calisto: I use fedora 2, but always latest kernel and asterisk CVS head - works like a charm |
14:49.21 | eKo1 | using CVS head is very risky for production use. |
14:49.22 | afe | that calisto should have been carlosh :) |
14:49.32 | Faithful | except we talk about rebooting once a week... which has never been linux practice |
14:49.37 | carlosh | Chuji: Gnomemeeting (the last ver) also has some support for statuses.. but Asterisk would not do anything when you change your status... |
14:50.05 | afe | Faithful: I don't reboot (unless I upgraded the kernel) - I just restart asterisk |
14:50.42 | Faithful | That |
14:50.52 | Faithful | 's not so bad I guess. |
14:51.06 | carlosh | Chuji: the soft clients should support this ... only firefly does, and if you use their private service to send these availability messages (add on or patched asterisk).. |
14:51.24 | afe | Faithful: restarting asterisk also restarts mpg123, which is what I believe is causing most problems for me :) |
14:54.09 | eKo1 | afe: Yes, that causes problems for me too. |
14:54.32 | lidl | i'm going to use a debian as a platform |
14:54.51 | afe | bleh... my mouse just refused to work on my icemat |
14:55.03 | eKo1 | I kill those mpg123 procs. after I stop * and start * thereafter. |
14:55.22 | carlosh | you should only get one hold-music process per hold music file on your system.. |
14:55.37 | Grooby | ok back |
14:55.39 | PatrickDK | no, it's not per file |
14:55.51 | PatrickDK | it's per musiconhold definition |
14:55.59 | carlosh | ok.. |
14:56.23 | carlosh | i have only two processes.. no problems at all.. |
14:58.03 | afe | emergency over - mouse working again :) |
14:58.19 | carlosh | the only issue wih hold music I have at present.. is this IP phone.. it would not play it... zaptel extensions and incoming callers no problem.. bt not this chinese IP phone.. |
14:58.45 | afe | carlosh: that's probably a codec issue |
14:58.47 | threat|BX | hmm so how do these X100P PCI thingies differ from regular modems? |
14:58.58 | carlosh | afe: ta, will check.. |
14:59.21 | *** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com) |
14:59.54 | TheBear | In order to use IAXTEL, can one use a SIP phone connected to an * server, or MUST it be a IAX phone ? |
15:00.07 | Eight | threat|BX: They're basically sound cards. |
15:00.28 | Eight | threat|BX: all the processing is done in the CPU, instead of on card. That's why they're so cheap (and require special drivers in windows to work as modems) |
15:00.44 | Grooby | eight |
15:00.48 | Grooby | did you fix your BV problem? |
15:00.57 | Eight | Grooby: ya, 'insecure=very'. |
15:01.01 | Grooby | okie |
15:01.04 | Eight | when you remove it, it will continue to work for a while. |
15:01.11 | Eight | but to kick it into working again you have to replace it. |
15:01.14 | Eight | that seems to be the issue. |
15:01.20 | Grooby | ok |
15:01.28 | threat|BX | Eight oh I see, so the hardware versions of them are expensive? (how much extra are the hardware ones?) |
15:01.36 | Grooby | hehe..I remember having that issue but i figure everyone knew about it |
15:01.45 | Grooby | oh well..back to my Mythtv research |
15:01.47 | threat|BX | Eight heh they wont be used in windows "{ |
15:01.48 | afe | TheBear: you can use any phone that can talk to asterisk |
15:01.50 | threat|BX | :P |
15:02.08 | TheBear | afe: thanks, |
15:02.18 | afe | TheBear: asterisk will translate the call for you |
15:02.36 | TheBear | cool ! |
15:02.56 | Eight | threat|BX: there are a bunch of places that sell the TDM400P online, check froogle |
15:03.11 | threat|BX | heh ok |
15:05.55 | TheBear | Am I right that you can have an extension (44) that if I dial (44pstn_number) It will then be able to repeat dial the pstn_number until it is answered (assuming the first attempt) the number is busy ? |
15:07.22 | *** join/#asterisk \Grooby\ (~sl9z@ip24-250-126-171.dc.dc.cox.net) |
15:08.26 | afe | TheBear: well, you would probably want to wait a few seconds until dialing it again, but yes |
15:08.54 | *** join/#asterisk kodomo (~memyself@emu.net.informatik.tu-muenchen.de) |
15:09.00 | kodomo | hi folks :) |
15:09.24 | kodomo | Can anybody tell me if early B3 is possible with chan_misdn? |
15:09.33 | carlosh | re this unavailable status, GM reports back (on the * console): Got SIP response 486 "Busy Here" back from <IP ADDRESS> .. but .. I'd like to know somehow, beforehand like with firefly.. that the user is available or not... |
15:10.21 | TheBear | afe: right, if exten => _44.,3,Dial(${EXTEN:2}), then _44.,103,Wait(3) _44.,104,goto(3) right to loop back on a busy signal ? |
15:10.44 | threat|BX | hmmmm ok so I need a TDM400P card for incomming lines and a TDM400P card with other modules to connect my phones to? |
15:10.48 | threat|BX | am I on track? |
15:11.29 | kodomo | anybody using misdn for the external device? |
15:11.54 | afe | TheBear: not sure... the 104 will kick in when busy, and you should probably keep the call somewhere else and then retry |
15:12.27 | afe | threat|BX: all tdm400 cards can be equipped with any module |
15:12.57 | MikeJ[Jayden] | ~seen xkev |
15:12.59 | jbot | xkev <kevin@orbit.xmission.com> was last seen on IRC in channel #asterisk, 6h 20m 47s ago, saying: 'lousy mysql licensing'. |
15:12.59 | TheBear | afe: isn't it when a Dial() get a busy signal the next seq in the dialplan is 100+Dial seq number, thus 3 become 103 ? |
15:13.24 | afe | TheBear: its n+101 |
15:13.27 | threat|BX | afe ok but basically I need a tdm400, and aport for every phone I want to connect to it? or are ther eother ways? ":) |
15:14.09 | TheBear | afe: ok n+101, then 104 should be wait and 105 goto(3)? |
15:14.28 | afe | threat|BX: you could use a Sipura (or other) that makes an analog phone able to talk SIP |
15:14.37 | afe | TheBear: I think so, yes |
15:15.03 | afe | TheBear: never tried it though, so you better do :) |
15:15.32 | afe | threat|BX: but I think a tdm400 with the number of ports needed works best |
15:15.33 | TheBear | afe: thanks yeah I will just wanted to check it was possible, seems logical, but logical is not always possible |
15:15.43 | lidl | threat|BX, http://www.voipsupply.com/product_info.php?cPath=96_118&products_id=33 |
15:15.59 | afe | threat|BX: unless you go with SIP phones |
15:16.54 | threat|BX | afe ok, but for each SIP device I need to connect it to a port on a special type of PCI card? (E.g. TDM400?) |
15:17.05 | threat|BX | oh, SIP is good? |
15:17.43 | *** join/#asterisk CosmicRay (~jgoerzen@2002:4545:7206:1:20e:a6ff:fe5c:55e1) |
15:18.16 | *** join/#asterisk Nohair (~Jez@srscomp.demon.co.uk) |
15:18.20 | afe | threat|BX: the sip devices connects over ethernet, and your server won't need anything but a ethernet connection for that |
15:18.51 | afe | the TDM400 is only needed for analog stuff |
15:18.55 | threat|BX | afe nice |
15:18.58 | Nohair | Hi can any one help I have asterisk running on Macos when I am left a voicemail its just white noise |
15:19.46 | PatrickDK | tdm400 is needed for a zaptel timing device in 3.3v pci systems |
15:19.55 | PatrickDK | unless you feel liking paying for the t1 card |
15:20.09 | threat|BX | hmmmmm |
15:20.16 | threat|BX | I cant get T1 here :( |
15:20.28 | threat|BX | E1 maybe, but I believe its uber expensive |
15:20.32 | afe | PatrickDK: you don't need the timing device if you're not doing meetme, and you can use the ztdummy |
15:20.34 | PatrickDK | heh t1/e1 |
15:20.35 | Nohair | any one help with the voicemail noise problems???? |
15:20.42 | PatrickDK | ztdummy sucks |
15:20.42 | threat|BX | ÖP |
15:20.46 | threat|BX | ÖP even |
15:20.51 | threat|BX | gaz kezboard is fscking up |
15:20.57 | PatrickDK | and meetme/iax trunking is required for me |
15:21.25 | TheBear | anyone know why? (1) I have subscribed to asterisk-user@digium yet I have not received any acceptance email to approve my subscription (2) I have registered with IAXTEL and also not received any email with my password ? |
15:21.48 | CosmicRay | sounds like your e-mail is hosed |
15:21.59 | Nohair | Any one running asterisk on a mac???? |
15:22.26 | afe | PatrickDK: if only for timing I suppose any X100P clone would work |
15:22.35 | TheBear | CosmicRay: I can get junk and other mail fine :( in fact I've been checking every 5 sec. and keep receiveing mail I don't want, and not the mail I'm waiting for |
15:22.38 | PatrickDK | afe, I said 3.3v pci |
15:22.39 | threat|BX | afe, thanx for zour help Ö= same with the others Ö= I will havea read and think about setting up one of these szstems Ö= |
15:22.50 | PatrickDK | servers don't normally have 32bit 5v pci slots |
15:22.55 | threat|BX | PatrickDK 3.3v PCI isnt normal_ |
15:23.02 | afe | PatrickDK: my mistake |
15:23.08 | PatrickDK | 3.3v pci is standard for servers |
15:23.13 | threat|BX | nice |
15:23.18 | threat|BX | well BBL |
15:23.36 | Nohair | and one running asterisk on a mac |
15:24.05 | afe | Nohair: running asterisk on OSX is not very common :) |
15:24.39 | Nohair | aft :-( |
15:25.12 | afe | PatrickDK: do the X100P really require 5V= |
15:25.22 | PatrickDK | afe, the 3 I have do |
15:25.37 | PatrickDK | and so said digium |
15:25.47 | PatrickDK | though the card has the 3.3v cut in it |
15:27.35 | afe | PatrickDK: hmm... I know the TDM400 needs extra power, but I didn't think the X100p (or clones) did |
15:27.46 | PatrickDK | they don't need extra power |
15:27.50 | PatrickDK | they need a 5v pci slot |
15:28.01 | PatrickDK | most 32bit pci slots are 5v |
15:28.11 | PatrickDK | and ALL 64bit slots are 3.3v |
15:28.42 | tzanger | werd 'em up |
15:28.48 | afe | PatrickDK: ah... sorry - now I see your problem (I'm a bit tired atm) |
15:29.02 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
15:29.04 | coppice | all 66MHz slots are 3.3V. Not all 64bit slots |
15:29.25 | PatrickDK | heh, ya, someone always makes it different |
15:29.37 | PatrickDK | pci was made to do almost anything, makes it more confusin |
15:30.38 | afe | PatrickDK: however, if you run asterisk on a Soho system you probably don't have those slots anyway ;) |
15:31.06 | PatrickDK | hmm, my soho systems do |
15:31.59 | PatrickDK | heh, 7 people in the office, 3 servers |
15:32.04 | afe | if that's a problem I would run asterisk on a desktop computer rather than a server :) |
15:32.14 | PatrickDK | not reliable enough |
15:32.24 | PatrickDK | I perfer redundant cpu/memory/bios |
15:32.46 | afe | PatrickDK: of course, most soho don't do that :) |
15:32.48 | WhiteWlf | standalone old-school PBX'es didn't have much redundancy... but they never went down either |
15:33.11 | PatrickDK | whitewlf, I have seen enough break |
15:33.16 | WhiteWlf | me too |
15:33.17 | WhiteWlf | heh |
15:33.48 | afe | if you need the redundancy, you can probably pay for a tdm400 I guess |
15:33.48 | *** join/#asterisk Ron-Na (~ronald@203.70.36.126) |
15:33.48 | tzanger | wtf do you need 3 servers in a 7-person office for? |
15:34.07 | Ron-Na | Does anybody has experience with Realtime? |
15:34.13 | PatrickDK | one for telephone |
15:34.15 | tzanger | WhiteWlf: actually they went down often enough... usually the trunk cards (makes sense) |
15:34.23 | PatrickDK | one for harddrive shared storage |
15:34.33 | PatrickDK | one doing backups of all systems |
15:34.45 | WhiteWlf | tzanger: the one at meh office is rebooted once a week and has 450 users and one of those oldsckewl tape backup drives in it |
15:34.47 | tzanger | I think we have 6 or 7 for a 50-person manufacturing org, could consolidate it down to probably 3 |
15:34.50 | PatrickDK | have had too many harddrive failures already in he last 4 years |
15:34.58 | PatrickDK | about 1 drive every year |
15:35.03 | tzanger | PatrickDK: *nods* |
15:35.04 | PatrickDK | for ide drives |
15:35.12 | tzanger | that's what RAID'xs for :-) |
15:35.13 | PatrickDK | servers running scsi |
15:35.18 | tzanger | and hotswap, it just works too well |
15:35.22 | PatrickDK | hmm, you don't normally raid desktop systems |
15:35.22 | tzanger | painfully well |
15:35.27 | tzanger | PatrickDK: no not desktops |
15:35.29 | tzanger | you're correct |
15:35.44 | Moc | the problem is hotswap is that it the cause of most problem |
15:35.51 | tzanger | we've got a company who loves those diskless systems |
15:35.56 | tzanger | I can't think of the anme of htem offhand |
15:35.57 | Moc | because of the connection of the hotswap bay |
15:35.58 | PatrickDK | I have never had a problem with hotswap |
15:35.59 | Ron-Na | Does anybody use realtime? |
15:36.03 | tzanger | Ron-Na: not me |
15:36.04 | WhiteWlf | WinTerms.... |
15:36.11 | PatrickDK | moc, that is just compaq hoswap, many roblems with them |
15:36.12 | tzanger | WhiteWlf: yeah something like that |
15:36.16 | file | ooh twisted is unhappy at being woken up |
15:36.19 | Moc | PatrickDK, Dell also |
15:36.20 | WhiteWlf | oh man do i hate winterms |
15:36.20 | PatrickDK | I haven't had a problem with my intel hotswap |
15:36.24 | WhiteWlf | That leads me into a question about creating a dialplan... what if I want to prompt a user for an extension... then route them to voicemail... like directly to voicemail... how do I capture the user input at a menu to use it to send it to voicemail? |
15:36.30 | PatrickDK | hell, I would never trust dell |
15:36.32 | tzanger | file: you shouldn't get out of bed so noisily then |
15:36.43 | file | well it was bkw who slammed the door |
15:36.46 | WhiteWlf | dell is just a company that puts parts together... how the hell can you not trust it? |
15:36.49 | tzanger | file: heh |
15:36.57 | PatrickDK | whitewlf, theydon't put parts together |
15:37.05 | PatrickDK | have you seen thier motherboard designs? |
15:37.12 | lidl | how much it costs in your country a t1 or e1 line? |
15:37.17 | PatrickDK | they have them made to their specs |
15:37.23 | PatrickDK | and isn't always reliable |
15:37.26 | WhiteWlf | yeah, and i can order exact replacements from the company they get them from |
15:37.37 | Moc | Canada/Montreal, about 700$/month+ for 1 year contract |
15:37.39 | Moc | CND |
15:37.40 | WhiteWlf | viva la asus, abit etc |
15:37.42 | afe | dell is working perfectly fine for me (and for lots of our customers) - however, don't expect to be able to upgrade them with standard parts |
15:37.49 | Ron-Na | I want to charge incoming calls in ASTCC, does anybody know how to set this up? |
15:38.20 | PatrickDK | I'll stick to intel and supermicro systems |
15:38.20 | file | Ron-Na: astcc is a _calling card_ application |
15:38.30 | file | lol, twisted just freaked out at the TV because it was loud |
15:38.34 | Moc | Ron-Na adjust your outgoing price in consequence |
15:38.37 | lidl | Moc, canada, so t1, so 24 channels + signalling, isn't it? |
15:38.45 | PatrickDK | 23+signal |
15:38.48 | PatrickDK | 24 channels total |
15:38.52 | Moc | it 23B channel + 1 dchannel |
15:39.01 | coppice | If you are in a big company Dells are great. for a small business or home user results vary |
15:39.05 | lidl | yes, my fault |
15:39.06 | Ron-Na | Moc, I don't understand that |
15:39.31 | WhiteWlf | It has 23 channels, and you have to have 1 signal for every set.... 23+1 = 24 total |
15:39.52 | afe | coppice: I wouldn't get a dell form home use (since I like to build my own computers, and I have 5 at home) |
15:40.12 | WhiteWlf | did anyone understand my question or did I just like... do a morning? |
15:40.13 | Moc | I got a Dell 1850 for my personal use, it aint bad |
15:40.42 | afe | coppice: but at work we just got 4 simple dells for $250 each - that really good if you only need them for surfing and office use |
15:41.38 | coppice | Moc: its not so much he machine. its the support. if a big company has problems Dell's response is excellent. if you are small potatoes to them, they couldn't care less |
15:41.50 | afe | dells High End computers are not bad either, but they can't be upgraded (easily) |
15:42.20 | coppice | afe: in asia we never get any of those cheap deals. Dell machines are quite expensive |
15:42.34 | lidl | Moc, just a curiosity, for each voice channel do you get a different phone number? |
15:42.37 | WhiteWlf | I'd really prefer not to spout my question again... it's rude... but I'd like to know if anyone saw it? |
15:42.44 | afe | Big companies and government agencies don't upgrade computers, so for that purpose they're excellent |
15:43.03 | Nugget | clearly the only rational solution is to buy a powermac. |
15:43.16 | Moc | lidl, you can get as many phone number you wish |
15:43.35 | afe | coppice: here they vary from extremely cheap to very expensive :) |
15:44.09 | afe | you can buy a machine one day, and it can be bought at half that price one week after ... |
15:44.16 | coppice | lidl: phone numbers are not normally related to channels on a PRI, unless its being used in a channel bankstyle to connect to 23 phones |
15:44.20 | Moc | lidl, the B Channel is the channel that allow voice or data communication. the D Channel have the 'signaling'. For example, a incoming call with DID 555 555 5555 is calling, so my system tell them ok use this B channel |
15:44.35 | MikeJ[Jayden] | hey Moc.. |
15:45.18 | coppice | afe: they endlessly advertise cheap deals in the newspapers. go to their site the day the paper is published and try the code - the system says it has expired. even those deals are not that cheap, though |
15:45.19 | lidl | Moc, coppice, thanks for the clarification :) |
15:45.30 | Moc | hi MikeJ |
15:46.55 | Faithful | I just experienced skype |
15:47.20 | *** part/#asterisk sysdef (~sysdef@sysdef.admin.debiancenter) |
15:47.32 | Faithful | It's got me worried |
15:47.34 | MikeJ[Jayden] | well.. I'm off to go have some exciting reading of Advanced UNIX Programming, second edition :) |
15:47.41 | Ron-Na | I use to say Skype is a toy, but it worked,.... on my Athlon it stopped to work |
15:48.01 | lidl | so, having a t1 let's you have a phone number which can handle 23 total calls (incoming/outgoing) concurrently |
15:48.08 | Ron-Na | The service of Skype is zero, ... |
15:48.22 | Faithful | the issue for me is it works better than G729 does |
15:48.45 | coppice | Faithful: of course it works better than G.729 |
15:48.46 | Faithful | Maybe I should change codecs |
15:48.51 | Moc | lidl, yes, |
15:48.56 | lidl | ok, thx |
15:49.03 | Faithful | coppice: why do you say that? |
15:49.09 | *** join/#asterisk gonzo- (~gonzo@icc-nat.univ.kiev.ua) |
15:49.20 | coppice | it uses a wideband codec at a much higher bit rate |
15:49.33 | tzanger | morning coppice |
15:49.59 | tzanger | I should soon be able to start the discovery phase of when the wctdm driver started sucking |
15:50.00 | Faithful | I thought G729 was the VoIP codec of the gods |
15:50.01 | coppice | if you don't have jitter and loss Skype should be considerably better than a land-line |
15:50.18 | coppice | G.729 is the codec of the bandwidth starved |
15:50.44 | Faithful | Oh... so I bought it for nothing |
15:50.45 | Moc | Faithful, the best codec in term of bandwidth and qualify, I think it g726 |
15:50.50 | coppice | its not great, its just pretty good for 8kbps |
15:50.54 | *** join/#asterisk Mike (~mike@201.135.48.217) |
15:51.10 | Moc | qualify=quality |
15:51.13 | Faithful | it suffers from jitter pretty bad |
15:51.21 | coppice | G.729 is mostly something you need because the other end is using it |
15:51.25 | Faithful | I gather that iLBC is better |
15:51.51 | Moc | iLBC is DEAD FOR ME ... it the worst codec !!! (mostly because the code is buggy) |
15:52.06 | Mike | Moc, buggy? |
15:52.13 | Faithful | My VoIP provider told me G729 is so much better than GSM |
15:52.16 | Moc | I had terrible jitter with ilbc, and had issue that Asterisk go take 100% CPU also using that codec .. |
15:52.23 | coppice | Moc: iLBC is fine. It must be how you are using it |
15:52.37 | Faithful | Skype is iLBC correct |
15:52.52 | coppice | Skype is iLBC wideband |
15:52.53 | Moc | Im talking about Asterisk implementation of iLBC |
15:53.07 | mikegrb | skype is proprietary crap |
15:53.17 | Moc | Every time I get to use ilbc, problem appear ... |
15:53.26 | WhiteWlf | In an IVR menu... how can I have it prompt an extension and then dial that extension? |
15:53.39 | mikegrb | WhiteWlf: by reading some damn docs |
15:53.54 | mikegrb | WhiteWlf: and not expecting everyone to hand you everything without work on your part |
15:53.55 | WhiteWlf | mikegrb: I'm not framiliar with that specific doc ;) |
15:54.02 | tzanger | coppice: I spoke to someone in France (I'm in Canada) over Skype (he called my POTS line) ... there was signficant delay but other than that it wasn't bad |
15:54.04 | Moc | ~docs |
15:54.05 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
15:54.12 | Mike | hi coppice !! |
15:54.22 | tzanger | it was kind of funny, he was going on about how amazing voip is but he had no idea that ever single call we've been taking and placing over the past year has been VOIP :-) |
15:54.32 | WhiteWlf | Thank you moc... mikegrb: this is a hobby... not a requirement of myself... so I enjoy the help I get when I get it. :D |
15:54.33 | coppice | tzanger: skype is designed to work really great when conditions are good. |
15:54.42 | Faithful | coppice: wideband? |
15:54.45 | tzanger | coppice: I find iLBC's voice quality subpar, at least compared to GSM |
15:54.52 | mikegrb | WhiteWlf: read some docs and you will get a lot more help |
15:55.18 | Moc | WhiteWlf, if it a hobby, you should be searching by yourself even more ;) |
15:55.27 | coppice | Faithful: 16k samples/second == 8kHz bandwidth. you can tell "s" from "f" :-) |
15:55.29 | tzanger | coppice: I'm not sure why though -- but every time I change my default codec from GSM to iLBC I get complaints... and the hardware's more than enough to be able ot handle it (Xeon systems) |
15:55.51 | Moc | tzanger, yes, ilbc have weird problems |
15:55.54 | coppice | Mike: hi. how's things? |
15:56.02 | Moc | in asterisk atless |
15:56.19 | Mike | coppice, after e1 worked they didnt call me no more:( |
15:56.53 | Mike | Moc, i use alot iLBC i find it really good |
15:56.58 | Mike | Moc, even better than gsm |
15:57.31 | Faithful | so should I be using gsm over G729? |
15:57.34 | Faithful | man |
15:57.40 | coppice | Mike: R2 seems pretty solid now. There are still a couple of odd things which happen infrequently, though |
15:57.45 | Mike | Faithful, anything is better than g729 |
15:58.03 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
15:58.04 | Faithful | Oh, no... and I paid money for it |
15:58.17 | Faithful | I was told the opposite |
15:58.26 | Moc | Faithful, g729 is good with you got very low bandwidth |
15:58.27 | coppice | comparing codecs is a complex thing. saying one is better is meaningless on its own. better at what? |
15:58.47 | Faithful | I got jitter... what's good for gitter? iLBC? |
15:58.53 | Moc | but g729 is CPU hungry, and qualify aint that great, but it aint that bad |
15:59.29 | *** join/#asterisk MicH323 (~micosat@host-84-9-63-27.bulldogdsl.com) |
15:59.31 | Faithful | when I switched everyone I spoke to commented |
15:59.40 | Faithful | without my saying |
15:59.43 | coppice | iLBC is good for packet loss tolerance. jitter is a jitter buffering issue. not a codec one |
15:59.53 | Faithful | quality went down from GSM |
16:00.46 | Faithful | but I gess the difference between jitter & packet loss can be undescernable at times |
16:00.47 | Moc | I always use ulaw, except for 2office that I had to make 20 voice channel go throught a DSL, so I used g726 (sound nearly as ulaw) |
16:02.13 | Faithful | Man I have so much to learn about all this stuff ... codecs etc |
16:02.20 | lidl | on voipsupply i see no gsm gateways. has anyone experienced them? |
16:02.37 | Moc | gsm gateway ??? |
16:02.58 | puppet | this is evil |
16:04.32 | coppice | the nastiest things with codecs is mixing them. avoid transcoding at all costs |
16:04.42 | lidl | like this one: http://shop.voismart.it/proddetail.php?prod=GSM-1ANA |
16:05.55 | lidl | basically, you put a sim inside, connect it to an fxs and apply least call routing |
16:09.52 | file | lalala |
16:10.09 | lidl | http://www.2n.cz/products/gsm_gateways/analog/analog_gsm_gateway.html |
16:10.10 | puppet | Have you all seen German Darth Vader? |
16:10.18 | puppet | http://www.efterbliven.de/pics/ohhgawd23.jpg < German Darth Vader |
16:10.25 | coppice | those GSM units require GSM<->analogue<->something digital with a quality loss. there are some units which allow digital audio to go straight into the GSM unit now. |
16:10.41 | *** join/#asterisk RoyK (~roy@ti211210a080-2089.bb.online.no) |
16:11.13 | lidl | coppice, any vendor? |
16:12.06 | puppet | coppice: oh what do they cost? |
16:12.15 | coppice | dunno. try google. it used to be all the GSM modems only allowed audio to be fed in as analogue, but i was told recently some of them support digital audio now |
16:12.38 | RoyK | hmmmm |
16:12.38 | carlosh | guys: what is aec ? |
16:13.09 | RoyK | automatic echo cancellation? |
16:13.14 | carlosh | ta. |
16:13.23 | coppice | We used to use Falcom GSM modems for mass SMS. those were about $250. if units like that now support digital audio they should still be the same price |
16:13.31 | coppice | acoustic echo cancellation |
16:14.08 | carlosh | thanks.. been trying to tune up this IP phone, so I can hear the hold music... grrr.. |
16:14.29 | coppice | by which they usually mean voice on a line, rather than modem signals on a line. AEC sounds like it means cancelling rooms echos, but it usually doesn't |
16:14.37 | RoyK | if I have a queue with 10 SIP queue members, can I, when someone calls in, check wheather there are any SIP clients connected, and if not, play some message saying "fuck off" or something? |
16:15.17 | `Sauron | How polite. |
16:15.36 | RoyK | whatever. |
16:15.40 | RoyK | is it possible? |
16:15.59 | *** join/#asterisk _RaYmAn_ (user@213.237.12.147.adsl.vby.tiscali.dk) |
16:16.09 | `Sauron | You could read the documentation and find out. |
16:16.10 | coppice | RoyK: I suggest you just tell them to go away. leave it up to them whether the use the time for sexual activities |
16:16.23 | RoyK | ~lart coppice |
16:16.57 | puppet | coppice: http://www.falcom.dk/fala2d-3.pdf ? |
16:17.51 | Nugget | http://lnk.nu/slacker.com/lt <-- lart |
16:18.32 | *** join/#asterisk Frantic (~ab@24-193-46-85.nyc.rr.com) |
16:19.16 | `Sauron | Nugget: It's nice outside. What are you doing not driving? |
16:19.26 | Nugget | gearing up for sxsw. |
16:19.36 | `Sauron | Going today? |
16:19.39 | Nugget | yeah |
16:20.09 | `Sauron | Hum. |
16:20.23 | `Sauron | With all the sxsw traffic, wonder how hard it's gunna be to get to Music Makers |
16:20.27 | carlosh | summary of my testing.. gee... this ip phone only works kind of ok with ulaw and alaw.. g729 not fully licensed here.. and it won't play the holf music.. I had to allow only either ala or ulaw at a time in order to get good quality audio at the other end.. a software client... |
16:20.58 | carlosh | 5:20 am here... off to bed.. :/ |
16:21.24 | file | be back tomorrow, going to... well... fly home |
16:21.28 | carlosh | if anyone can recommend a good/affordable iphone, with ilbc support, much appreciaed beforehand.. |
16:21.47 | Zeeek | bt100 |
16:21.58 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
16:23.07 | carlosh | Zeeek: thanks. |
16:23.23 | Zeeek | meets the two criteria you gave |
16:23.39 | lidl | Zeeek, budgetone100? |
16:23.53 | Zeeek | to be precise BT102 is the one I rccomand |
16:24.08 | Zeeek | has a hub for easier connection |
16:24.14 | *** join/#asterisk jeffik (~jeffik@m5f7436d0.tmodns.net) |
16:24.28 | lidl | a friend of mine said they look like toys, but are ok for residential use |
16:24.42 | Zeeek | many, many of these phones look like toys |
16:24.49 | `Sauron | why ilbc? |
16:24.50 | carlosh | sure, it has the two ethernet ifaces as well |
16:24.59 | Zeeek | the cisco looks like it should be in the war room of the pentagon |
16:25.11 | Zeeek | and it probably is |
16:25.16 | carlosh | Zeeek: ilbc support ? |
16:25.32 | Zeeek | again: meets the two criteria you gave |
16:26.02 | zoa | hey royk |
16:26.08 | Zeeek | file your photo has now been shown worldwide |
16:26.11 | carlosh | :o) |
16:26.24 | `Sauron | which photo? |
16:26.30 | Zeeek | not the nude one |
16:26.35 | Zeeek | that requires a password |
16:26.39 | zoa | where is file ? |
16:26.42 | Zeeek | or an .htaccess hack |
16:27.04 | Zeeek | you'll have to search |
16:27.27 | Zeeek | exten => hint,1,mailingList(recent) |
16:27.42 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
16:27.42 | Nugget | heh |
16:28.43 | puppet | http://www.efterbliven.de/pics/ohhgawd23.jpg < German Darth Vader |
16:28.46 | puppet | that one |
16:28.51 | puppet | thats nugget |
16:28.53 | puppet | oh wait |
16:28.55 | puppet | wrong pic |
16:28.56 | carlosh | Zeeek: do you know what is the standard number for ilbc ? gxxx ? |
16:29.00 | puppet | http://slacker.com/photos/2002mroadster/IMG_1381 |
16:29.02 | puppet | there ;p |
16:29.08 | Zeeek | carlosh no |
16:29.34 | carlosh | the pdf of the budgetone does not mention ilbc.. |
16:29.49 | Zeeek | carlosh it does ilbc |
16:30.08 | carlosh | Zeeek: thanks mate, off now, thanks all |
16:30.11 | Zeeek | introduced around 5.11 firmware or something like that - a while ago now |
16:30.57 | zoa | i have such a terrible jetlag now |
16:31.06 | zoa | struggling to stay awake |
16:31.16 | Zeeek | afraid of nightmares? |
16:32.08 | Zeeek | search for "pictures" in the asterisk users mailing list reveals a few interesting voIP personalities |
16:32.36 | tzanger | Zeeek: eh? |
16:33.22 | Zeeek | AKA "Spring VON Asterisk Pavilion" |
16:34.58 | `Sauron | alright |
16:35.05 | `Sauron | Off to do a brake job. How fun. |
16:35.16 | tzanger | `Sauron: doing it yourself or getting it done |
16:35.18 | afe | mamma mu!!! |
16:35.21 | `Sauron | myself |
16:35.27 | `Sauron | c'mon, I'm a real MAN. :) |
16:35.27 | tzanger | `Sauron: excellent |
16:35.38 | tzanger | I have to do brakes on my vehcile too |
16:35.40 | tzanger | I miss my jeep so much |
16:36.52 | zoa | hehe |
16:36.54 | zoa | i was there too |
16:37.47 | Zeeek | someone was talking about how linux uses RAM on the ML... I noticed free RAM went from like 400M to 100M between 4:30 and 4:45 - that's when the daily and weekly cronjobs are run |
16:38.04 | Zeeek | I've been monitoring RAM because ti appeared there was a leak |
16:38.45 | Inv_arp | if context A includes => B and B includes => C can A access C? |
16:38.46 | Zeeek | but the cronjob that ran is just a stock script to archive logs |
16:38.55 | Nugget | Inv_arp: yes |
16:38.57 | Zeeek | it would seem |
16:39.41 | Inv_arp | hmm not good... when anyonecalls in they can access my outside lines by pressing 91NXXNXXX |
16:39.49 | agave-txlink | heh |
16:39.52 | agave-txlink | asterisk has memory leaks |
16:40.03 | agave-txlink | everytime I do a reload, mem usage goes up 2M |
16:40.24 | zoa | hmm |
16:40.26 | zoa | report it |
16:40.36 | zoa | its not certain the actual mem usage goes up |
16:40.43 | zoa | as linux doesnt reclaim memory until it needs it |
16:40.43 | agave-txlink | yeah, it's certain |
16:40.48 | agave-txlink | once I hit total mem used |
16:40.50 | agave-txlink | asterisk crashes |
16:40.54 | zoa | aha |
16:40.55 | zoa | yeah |
16:40.58 | agave-txlink | i'm running 1.0.6 though |
16:41.00 | zoa | thats a memleak |
16:41.01 | agave-txlink | i'll try 1.0.7 before I bitch about it |
16:41.17 | Zeeek | by golly, he's right! |
16:41.44 | Zeeek | no, in fact it went down - 1.0.6 |
16:41.47 | Eight | I don't suppose there's magically a changelog for .0.6->.0.7? |
16:43.33 | Inv_arp | ok context A includes => B and B includes =>C anyway to prevent A knowing about C? |
16:44.07 | agave-txlink | what are you trying to accomplish invarp? |
16:44.12 | Zeeek | Inv_arp I think you need to build your contexts differently |
16:44.47 | Zeeek | with the sooper power stuff included only in the priviliged users context |
16:44.55 | Inv_arp | agave-txlink: when anyone dials in my menus "A" if they press 91 they get outside line "C" |
16:45.03 | agave-txlink | okay |
16:45.09 | agave-txlink | your menus should have their own context |
16:45.22 | agave-txlink | if you're using a DID, trap the DID and then do a Goto to your menu context |
16:45.46 | agave-txlink | speaking of asterisk bugs, if you do too many include => then asterisk will crash |
16:45.57 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
16:46.01 | agave-txlink | hey kevin |
16:46.40 | Zeeek | how many includes? |
16:47.02 | agave-txlink | zeeek: i didn't count, it was a production system so I had to fix quickly, but I'd say around 100 or so |
16:47.13 | Zeeek | ya well that is a lot of includes |
16:47.33 | agave-txlink | true |
16:48.18 | Zeeek | not nested? |
16:48.24 | agave-txlink | no, not nested |
16:48.33 | Zeeek | excuse me I need to go open a bottle of Château La Mission Haut-Brion 1997 |
16:48.36 | agave-txlink | which is why I was able to switch to #include |
16:48.55 | Zeeek | #include rocks for making the dialpan hipper |
16:49.18 | Inv_arp | i seperate internal/external using #include |
16:54.21 | zoa | hey mark |
16:54.24 | zoa | im home already |
16:54.28 | kram | yay! |
16:54.33 | kram | glad you made it safe |
16:54.44 | zoa | it was horrible |
16:54.49 | zoa | couldnt sleep all night |
16:54.50 | kram | how horrible? |
16:54.53 | kram | i sowwy! |
16:54.57 | zoa | took ages to get home |
16:54.58 | zoa | :) |
16:54.59 | kram | you must have been thinking about asterisk |
16:55.03 | zoa | hehe lol |
16:55.51 | *** join/#asterisk bah (048830696@AC90B4BA.ipt.aol.com) |
16:58.15 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
16:58.30 | *** join/#asterisk Xander77 (~Alex@exten-halls-243.soton.ac.uk) |
16:59.26 | *** join/#asterisk Frantic (~ab@w020.z066088084.nyc-ny.dsl.cnc.net) |
16:59.32 | *** join/#asterisk ws9455 (~ws9455@adsl-68-94-10-246.dsl.rcsntx.swbell.net) |
17:11.32 | *** part/#asterisk jeffik (~jeffik@m5f7436d0.tmodns.net) |
17:14.01 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfjhe.dialup.mindspring.com) |
17:20.38 | *** join/#asterisk n3tar (~geno@201.254.93.202) |
17:20.41 | n3tar | hi |
17:20.47 | *** join/#asterisk spackle (~spackle@209.234.83.19) |
17:22.09 | *** join/#asterisk Necko (~roy@IGLD-83-130-105-236.inter.net.il) |
17:22.16 | Necko | hello |
17:22.43 | PTG123 | hey anyone awake know the polycom well? |
17:23.00 | Zeeek | here we go again :) |
17:23.07 | PTG123 | hehe |
17:23.10 | PTG123 | well i unlocked it now |
17:23.14 | Zeeek | you missed ManxPower |
17:23.18 | PTG123 | now i just need to know how to make it register |
17:23.21 | PTG123 | i can make a call |
17:23.26 | PTG123 | its just not registering |
17:23.32 | Zeeek | you were there this morning |
17:23.45 | PTG123 | you mean he was here? |
17:23.53 | Zeeek | right after yuou left |
17:23.55 | spackle | Anybody know specifically what nat=yes or no does? I did a search of the wiki and just find samples of it. |
17:24.04 | PTG123 | yes |
17:24.14 | Necko | Is there a version of asterisk that i can install on an exisitng linux distro? |
17:24.25 | PTG123 | nat doesnt' listen to the headers for the ip address of the host, and uses the actualyip it comes from.. nat=yes should ALWAYS be on |
17:24.28 | Zeeek | Necko sure |
17:24.41 | Zeeek | use cvs or download a tarball |
17:24.47 | Necko | thanks.:) |
17:24.55 | spackle | PTG123 does it send a keepalive or anything? |
17:24.58 | Zeeek | see digium.com -> download |
17:25.04 | Zeeek | or |
17:25.05 | Zeeek | Starter tutorial: |
17:25.05 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
17:25.05 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
17:25.05 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
17:25.05 | Zeeek | THE reference of the moment: |
17:25.06 | Zeeek | http://www.asteriskdocs.org |
17:25.21 | Zeeek | the automated stuff shows a whole procedure |
17:25.35 | Zeeek | so does asteriskdocs.org |
17:25.43 | PTG123 | spackle: no |
17:26.11 | *** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com) |
17:26.12 | Necko | thanks! |
17:26.20 | Zeeek | cheers |
17:26.31 | spackle | PTG123: thanks |
17:26.35 | Necko | another little question ...can asterisk behave as an h323 gw? |
17:26.47 | Zeeek | it would appear that it can |
17:26.56 | Zeeek | ~wiki |
17:26.59 | TheBear | I have my TDM10B (with FXS module) installed, and a std plugged in. What config files to I need to change to have * dial the std phone on an incoming call |
17:27.08 | Zeeek | I can never remember the URL |
17:27.14 | PTG123 | ok i helped someone, so should karma make someone help me |
17:27.19 | TheBear | I know my extensions.conf must have Zap/2 probm but how do I define zap/2 |
17:27.34 | Zeeek | PTG123 wait for Manxpower - wait do you use 1.0.6 STABLE? |
17:27.39 | Essobi | Morning peeps |
17:27.57 | Zeeek | PTG123 if so, go chase him down in #asterisk-stable |
17:28.10 | PTG123 | hehe |
17:28.16 | PTG123 | stable is for chickens |
17:28.22 | Zeeek | TheBear ZAP channels are defined in zapata.conf |
17:28.34 | Zeeek | preted you use STABLE to find ManxPower |
17:28.36 | TheBear | Zeeek: ok what about zaptel.con ? |
17:28.38 | Necko | i mean,will i be able to create a dial peer of h323 which will get a request from an ip and if the destination pattern starts with ,03 forward it to the PSTN?(through a specified modem) |
17:28.48 | PTG123 | Zeek: he only likes stable? :) |
17:28.52 | Zeeek | yes zaptel, not zapata |
17:29.24 | Zeeek | fxsks = 1,2 |
17:29.24 | Zeeek | fxoks = 3,4,5 |
17:29.38 | Zeeek | you have statements like those ytwo in yours? |
17:29.58 | TheBear | fxsks=1 only yes, which is my X100p |
17:30.06 | Zeeek | so it's ZAP/1 |
17:30.10 | Zeeek | and that's an end to it |
17:30.40 | TheBear | yes, now I have installed my TDM10B, along with the x100p |
17:30.41 | Zeeek | you ain't got no ZAP/2 bro |
17:30.47 | Necko | if i were to connect 4 phones to the asterisk machine i would have need 4 fxs(modem-line or phone?) right? |
17:30.51 | Zeeek | ah |
17:30.55 | *** join/#asterisk Gh0sty (~Ghosty@81.11.192.116) |
17:30.59 | Zeeek | then you will need to install the driver |
17:31.04 | TheBear | I want to use the FXS module in the TDM10B to dial a std analog phone |
17:31.07 | Zeeek | the name of which escapes me |
17:31.26 | Zeeek | you have an FXS module. You need another statement in zaptel |
17:31.35 | TheBear | wctdm, I found that on wiki, but I'm not clear on what other conf files need to be changed |
17:31.36 | Zeeek | like fxoks=2 ? |
17:31.49 | Zeeek | add that |
17:32.03 | TheBear | ok I just added that, what about zapata.conf it only have channel => 1 |
17:32.20 | Zeeek | first you'll want to see if ztcfg says anything |
17:32.35 | Zeeek | I can't remember if you need to unload and reload the drivers first though |
17:32.52 | Zeeek | type this ztcfg -vvv and see what iot says |
17:33.05 | TheBear | 2 channels configured. |
17:33.10 | Zeeek | you got it! |
17:33.12 | TheBear | ZT_CHANCONFIG failed on channel 2: No such device or address (6) |
17:33.19 | Zeeek | now go read about zapata in the samples |
17:33.53 | Zeeek | i think you need to stop astrisk and unload/reload the zaptel drivers |
17:35.10 | Zeeek | TheBear check this out : http://www.voip-info.org/wiki-Asterisk+config+zapata.conf |
17:36.09 | TheBear | ok I'll try that |
17:39.25 | TheBear | reload the machine, and the modules, still the same messages from ztcfg -vvvv |
17:39.39 | Zeeek | anyone discoverdid you read the wiki page? |
17:40.06 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net) |
17:41.47 | *** join/#asterisk rumba (~ropawa@cpe-68-201-148-205.sw.res.rr.com) |
17:41.51 | TheBear | Zeeek: also * fails to start "WARNING[6710]: chan_zap.c:848 zt_open: Unable to specify channel 2: No such device or address" |
17:42.09 | Zeeek | not surprising that asterisk wouldn't like it |
17:42.14 | Zeeek | if ztcfg doesn't |
17:47.14 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
17:47.23 | TheBear | How can I get zrtcfg to recognize the phone connected to the FXS module ? |
17:47.26 | ariel_ | good afternoon everyone. |
17:47.47 | ariel_ | TheBear, ztcfg -vv |
17:47.49 | Zeeek | it doesn't care about phones |
17:47.57 | Zeeek | only the hardware itself |
17:48.14 | Zeeek | do this: |
17:48.18 | Zeeek | cat /proc/interrupts |
17:49.13 | PTG123 | Mar 12 10:45:41 NOTICE[6672]: chan_sip.c:8152 handle_request: Registration from '<sip:66.235.234.131@66.235.234.131>' failed for '68.106.24.139' |
17:49.15 | Zeeek | do you see wctdm ? |
17:49.21 | PTG123 | does that mean its specifying the ip as username? |
17:49.23 | TheBear | no |
17:49.35 | Zeeek | TheBear did you modprobe it? |
17:49.51 | TheBear | yes modprobe wctdm and it is listed in lsmod |
17:50.03 | Zeeek | and ztcfg gives an error? |
17:50.09 | TheBear | Module Size Used by |
17:50.09 | TheBear | wctdm 121728 - |
17:50.09 | TheBear | wcfxo 10688 - |
17:50.09 | TheBear | zaptel 220516 - |
17:50.21 | TheBear | 2 channels configured. |
17:50.21 | TheBear | ZT_CHANCONFIG failed on channel 2: No such device or address (6) |
17:50.40 | Zeeek | did you remove and then reload the drivers a few minutes ago like I asked? |
17:51.09 | TheBear | yes I even rebooted the machine and then did the modprobe from scratch |
17:51.39 | Zeeek | what distro and kernel? |
17:51.49 | TheBear | gentoo kernel 2.6.10 |
17:51.57 | Zeeek | sounds like it's time to do some serious mailing list searching |
17:52.15 | Zeeek | 2.6 kernel has issues that should be found somewhere |
17:52.50 | Zeeek | something about that here: |
17:52.51 | Zeeek | http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation |
17:53.02 | TheBear | I've tried to subscribe to the mailing list since thursday I'm still waiting for my subscription confirmation |
17:53.08 | Zeeek | Support for Kernel 2.6 |
17:53.08 | Zeeek | <PROTECTED> |
17:53.18 | Zeeek | did you read that? ^^^^^^^^^^^^^^^^^ |
17:53.56 | Zeeek | or this: |
17:53.58 | Zeeek | "Sep 2004: When compiling zaptel for ztdummy, be sure to link /usr/src/linux-2.6 to /lib/modules/2.6.x.x.etc/build so modprobe will succeed. |
17:54.28 | TheBear | yes I did the make linux26 I found that when I compiled and installed * |
17:54.36 | Zeeek | TheBear you can search the list without subscribing using google |
17:54.42 | TheBear | perhaps the acpi=off then ? |
17:54.47 | TheBear | let me try that |
17:56.28 | Zeeek | If you do a google search on "ZT_CHANCONFIG failed on channel 2: No such device or address" 45 pages come up |
17:57.29 | TheBear | <PROTECTED> |
17:58.37 | TheBear | most that I have seen so far are talking about the USB version |
18:01.25 | ManxPower | ~docs |
18:01.26 | 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 |
18:01.42 | TheBear | Hi Eric |
18:01.54 | *** join/#asterisk _tommyg_ (~tom@vsat-148-64-73-166.c119.t7.mrt.starband.net) |
18:02.18 | ManxPower | TheBear, Does ztcfg -vvv show any errors? |
18:02.34 | PTG123 | i got my phone working, woo hoo ;) |
18:02.37 | *** join/#asterisk LorenzoMarouani (~LorenzoMa@AVelizy-112-1-9-62.w81-49.abo.wanadoo.fr) |
18:02.41 | _tommyg_ | Any one out there tried a ShoreTel 530 with asterisk? Having a prob with MGCP |
18:06.00 | spackle | ~NAT |
18:06.03 | jbot | nat is probably Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
18:06.20 | TheBear | ManxPower: no just the two line about the channels and thos two give me a sec to bring the machine up and I'll display it all |
18:06.52 | TheBear | Module Size Used by |
18:06.52 | TheBear | wcfxo 10688 - |
18:06.52 | TheBear | wctdm 121728 - |
18:06.52 | TheBear | zaptel 220516 - |
18:07.05 | TheBear | Zaptel Configuration |
18:07.05 | TheBear | ====================== |
18:07.09 | TheBear | Channel map: |
18:07.14 | TheBear | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
18:07.14 | TheBear | Channel 05: FXO Kewlstart (Default) (Slaves: 05) |
18:07.18 | TheBear | 2 channels configured. |
18:07.23 | TheBear | ZT_CHANCONFIG failed on channel 5: No such device or address (6) |
18:07.57 | TheBear | I tried changing fxoks=5 from =2 no joy, also can't seem to get to the acpi=off |
18:08.05 | Essobi | I'm coming up with a new acronym for a project.. Someone give me a word that begins with F that fits... fields, forums, forms... F.. something that means website.. MMM |
18:08.48 | Essobi | _tommyg_ Why would you want to use MGCP? No sip images for a 530? |
18:09.18 | nestAr | isn't the shoretel just a Polycom IPX00? |
18:09.29 | _tommyg_ | I do not believe so, if so I have no way of getting them |
18:09.34 | _tommyg_ | Not sure |
18:09.38 | Essobi | Heh. |
18:09.41 | _tommyg_ | docs only speak of mgcp |
18:09.45 | Essobi | Nothing is impossible. |
18:09.47 | puppet | thebear: dont spam channel PLEASE |
18:09.53 | puppet | ~pastebin |
18:09.54 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
18:09.54 | Essobi | MGCP is so durty |
18:09.58 | puppet | thebear: use that please |
18:10.06 | tessier | Essobi: Division by zero is impossible |
18:10.25 | TheBear | sorry |
18:10.31 | TheBear | :=( |
18:10.39 | CosmicRay | does anyone here have rules to support the 1010 carrier selection when dialing out? |
18:10.51 | Essobi | Tessier Unless you redefine 0. Or Division. |
18:11.28 | tessier | zero is zero. You can write it with a different character but the concept is still the same. |
18:11.29 | _tommyg_ | OK I am gonna look for some SIP images, not sure if they are out there or not |
18:11.34 | nestAr | the shoreline IP 100 is a polycom, i bet the 530 is too |
18:12.17 | *** part/#asterisk Ash (aaron@fudgecom.net) |
18:12.19 | nestAr | http://www.voip-info.org/wiki-Polycom+Phones#comments <--- there's mention of the Shoretel 100 here |
18:12.20 | TheBear | Let me go read some more thanks |
18:12.48 | _tommyg_ | Thanks, gonna check it out |
18:15.57 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
18:20.56 | *** join/#asterisk ckruetze (~ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
18:23.02 | *** join/#asterisk darby_t (~tom@dnk250.neoplus.adsl.tpnet.pl) |
18:23.18 | *** join/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net) |
18:25.03 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
18:25.03 | *** mode/#asterisk [+o twisted] by ChanServ |
18:25.20 | twisted | *sigh* |
18:25.24 | ariel_ | ~seen drumkilla |
18:25.27 | jbot | drumkilla <~russell@12.21.241.80> was last seen on IRC in channel #asterisk, 9h 31m 42s ago, saying: 'ok ...'. |
18:25.37 | mrgoby | I'm having some issues with the Redirect command in the manager api .... I'm trying to redirect a zap channel to a conference room.... it transfers to the context i specify and to the extension, but to the priority-1.... also, it doesnt actually connect to the extension .. but it hangs until either input or timeout... any ideas ? |
18:25.40 | ariel_ | hello drumkilla are you around today? |
18:25.55 | ariel_ | hello twisted how are you doing today? |
18:26.09 | twisted | ariel_, would be better if I wasn't sitting in an airport :P |
18:26.22 | ariel_ | your heading back from Von? |
18:26.24 | twisted | yea |
18:26.38 | shido6 | mrgoby |
18:26.39 | twisted | drumkilla is in terminal C with bkw, and file is somewhere in terminal A here in san jose |
18:26.41 | ariel_ | Did you like it? Was it worth it? |
18:26.45 | twisted | it was fun |
18:26.47 | shido6 | mrgoby show me your dialplan |
18:27.01 | mrgoby | ok, pastebin-ing it now |
18:27.08 | ariel_ | I guess I will have to wait. I have a bug question. |
18:27.19 | modulus_ | i'll show you mine if you show me yours |
18:27.49 | twisted | ariel_, why is that? |
18:28.42 | ariel_ | I have a question on bug 3577 which he put that it was not added to stable. But as far as I can see stable still has incominglimit active. |
18:29.24 | twisted | right.... |
18:29.43 | twisted | the patch wasn't added to stable to handle limit on peers |
18:29.43 | ariel_ | so the bug 3577 is a fix and it should be added to stable. |
18:29.59 | *** join/#asterisk adorah (~jack@80.179.34.21.forward.012.net.il) |
18:30.34 | twisted | actually, it adds support for other items |
18:30.38 | twisted | not fixes any other bugs |
18:30.47 | mrgoby | http://pastebin.ca/7326 |
18:31.19 | mrgoby | that is after the login and not including the logout on the manger api |
18:31.22 | ariel_ | at least it looks like then this type of setting will stay with the product. I am having major problems with setgroup and getgroupcounts |
18:31.48 | modulus_ | "Sorry, the requested ID value is way too large or too small." |
18:31.55 | modulus_ | how brain-dead. |
18:32.01 | modulus_ | pastebin.ca/0 |
18:32.10 | twisted | ariel_, what difficutlies? I haven't had a problem using them |
18:32.27 | mrgoby | i was digging in manager.c and i saw that it uses ast_asnyc_goto() ... would this be why ? |
18:32.51 | mrgoby | why it seems to hang, that is ? |
18:33.14 | ariel_ | I am getting wrong counts from them and also it's not taking account when the users call outbound to exten only to pots or pri lines. It's just a very funny thing. |
18:34.02 | adorah | Have a query regarding setting a remote IAX extension: Once I added an extension in iax.conf do I set the extension in the extension.conf under context=default or open a new context say [iax]? |
18:34.47 | mrgoby | i also saw that on the other params, it is taking string length -1 for the value, which i assume is to get rid of null space and new line chars.... but with priority it just takes the val - 1... which i think may be a bug |
18:34.52 | *** join/#asterisk santiago (~santiago@63.245.86.111) |
18:35.06 | ariel_ | twisted main problem is when used in a rollover macro type it looses the callerIDNUM. or it just keeps the orginal one. Which I have been trying to reset when it goes to then next device. |
18:35.26 | adorah | Have a query regarding setting a remote IAX extension: Once I added an extension in iax.conf do I set the extension in the extension.conf under context=default or open a new context say [iax]? |
18:36.48 | *** join/#asterisk Spooch (~rath@p549A1CF8.dip0.t-ipconnect.de) |
18:38.14 | PTG123 | anyone have any idea why when i dial with my polycom it doesn't seem to look up a sip user? |
18:38.19 | PTG123 | but yet it dials fine |
18:39.56 | *** join/#asterisk file[airport] (file@dhcp64-134-126-76.sjca.sjc.wayport.net) |
18:41.37 | twisted | heh |
18:42.02 | file[airport] | don't you dare |
18:42.05 | twisted | nah |
18:42.11 | file[airport] | I've got a mac mini and I'm not afraid to knock you unconcious with it |
18:42.21 | Sedorox | ahah |
18:42.27 | Sedorox | don't abuse it!!! sned it to me instred |
18:42.30 | Sedorox | stead* |
18:42.32 | file[airport] | nah |
18:43.20 | shido6 | make a fish tank |
18:43.34 | Darwin35 | macmini wow |
18:44.00 | ariel_ | mac oh boy. So did you change the os to a real linux build?? |
18:44.13 | Darwin35 | Darwin |
18:45.34 | Sedorox | the mini's are nice |
18:45.52 | Sedorox | even tho I like OSX.. I'll probably load gentoo on it |
18:46.20 | ariel_ | is OS X not made from a version of BSD? |
18:46.24 | file[airport] | yes |
18:46.36 | PTG123 | osx is freebsd basically |
18:46.43 | mrgoby | darwin is anyway |
18:46.45 | Sedorox | well the kernel |
18:46.45 | Sedorox | yes |
18:50.19 | *** join/#asterisk bah (048830696@AC892FC3.ipt.aol.com) |
18:51.45 | PTG123 | you know i think i found a way to make a call on an asterisk server without registering, or authenticating |
18:52.00 | PTG123 | no server is secure :) |
18:52.01 | PTG123 | heh |
18:52.28 | mikegrb | sure |
18:52.29 | *** join/#asterisk Dark40rce (~vince@dsl-17-106.cofs.net) |
18:52.56 | ariel_ | PTG123, how are you doing this? |
18:53.08 | PTG123 | put the username as the ip of the server |
18:53.12 | Essobi | Mmm. |
18:53.24 | PTG123 | seems to be letting my phone make calls no problem |
18:53.27 | ariel_ | sample of the dial string? |
18:53.38 | *** join/#asterisk lidl (~little@213-140-6-96.fastres.net) |
18:53.46 | Essobi | Twisted and File are going to throw down. |
18:53.47 | PTG123 | actually one sec |
18:53.51 | Essobi | :) |
18:54.07 | PTG123 | i think its setting the address as the server in the sip packet |
18:54.08 | file[airport] | haha |
18:54.09 | PTG123 | not the username |
18:55.02 | ariel_ | PTG123, so you have as username in lets say xlite with no password or any other settings? |
18:55.11 | PTG123 | one sec checking |
18:55.15 | PTG123 | put in some debugging in the code |
18:55.16 | Essobi | I got $5 on twisted |
18:55.24 | Essobi | What's the spread? :) |
18:55.31 | PTG123 | yes its specifying usename as server ip address |
18:55.35 | PTG123 | and its letting me make calls no problem |
18:55.39 | PTG123 | i think i have a real password |
18:55.51 | PTG123 | my polycom for some reason is doing it that way and its letting it make calls |
18:56.01 | PTG123 | In find peer: 66.235.234.131 |
18:56.01 | PTG123 | TABLE: sipfriends |
18:56.01 | PTG123 | ==> name=66.235.234.131 |
18:56.01 | PTG123 | <PROTECTED> |
18:56.01 | PTG123 | <PROTECTED> |
18:56.02 | PTG123 | <PROTECTED> |
18:56.04 | PTG123 | <PROTECTED> |
18:56.05 | PTG123 | <PROTECTED> |
18:56.08 | PTG123 | <PROTECTED> |
18:56.10 | PTG123 | <PROTECTED> |
18:56.12 | PTG123 | <PROTECTED> |
18:56.14 | PTG123 | see its specifying my server ip as the name |
18:56.16 | PTG123 | and it lets i go right in and dial |
18:56.18 | file[airport] | argh you silly person |
18:56.50 | PTG123 | its whacky too |
18:56.53 | PTG123 | it sets caller id to my ip |
18:56.57 | PTG123 | er to the server ip |
18:56.59 | PTG123 | when i call out |
18:57.40 | puppet | ptg123: please dont spam channel.. |
18:57.45 | puppet | ~pastebin |
18:57.47 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
18:59.21 | mrgoby | that should be the topic |
19:00.01 | mrgoby | pastebin .... noahs of the irc-world ... floods can go there |
19:00.25 | Essobi | Umm. |
19:00.57 | Essobi | I was just wondering.. Why is there no way so see the duration of a channels current status? |
19:00.58 | ariel_ | PTG123, so if you try your settings I can use your server to make a call.. hummmm should I try? |
19:01.13 | PTG123 | it should work on any asterisk server |
19:01.23 | Essobi | That's kind of... smart to be able to get that from the AGI, Manager, or CLI |
19:01.27 | PTG123 | i am more frustrated that the polycom isn't sending the proper username |
19:02.23 | ariel_ | in most cases you can dial an extension@IP address and you have a good chance of getting it. |
19:02.56 | PTG123 | but you don't understand, i can dial ANYWHERE in the world from my box |
19:03.02 | PTG123 | without authenticating |
19:03.06 | PTG123 | any phone number you want |
19:04.31 | mrgoby | can you dial santa clause at the north pole ? |
19:05.06 | PTG123 | if i knew his number |
19:07.25 | ariel_ | PTG123, yes your right. I just did it via your server. |
19:08.27 | PTG123 | ariel: major problem isn't it |
19:08.48 | ariel_ | yes it is in my view. The dial string was so easy you would not belive it. |
19:09.03 | PTG123 | ariel_: you want to submit it to mantis? |
19:09.09 | *** join/#asterisk NYBLAZE10 (alex@ACA0FDF4.ipt.aol.com) |
19:09.16 | *** join/#asterisk Dark40rce (~vince@dsl-17-106.cofs.net) |
19:09.41 | ariel_ | I just did on my asterisk box. exten => _1760.,1,Dial/sip/${EXTEN:4}@yourIPaddress. |
19:10.29 | PTG123 | well i didn't have to do anything on my box to make it work |
19:10.32 | PTG123 | just set username to ip |
19:10.34 | PTG123 | then dial anything iw ant |
19:10.39 | PTG123 | it goes through normal dial strings |
19:11.10 | ariel_ | hummm I wonder if I can do this via someone else without logging in...... |
19:11.29 | *** join/#asterisk Tili (~Tili@202-133-65-63-dialup.sat.net.pk) |
19:11.52 | PTG123 | yah you can i think |
19:11.55 | PTG123 | try it on mine |
19:12.00 | PTG123 | 66.235.234.131 |
19:12.13 | PTG123 | but even if you had to be logged in |
19:12.15 | PTG123 | still a major problem |
19:12.20 | PTG123 | you could just get free calls |
19:12.22 | PTG123 | and buy a $2 account |
19:12.26 | PTG123 | on someones servers :) |
19:12.41 | PTG123 | damn i should keep this bug secret, gives me ideas :) |
19:13.54 | ariel_ | PTG123, make a guest account on your sip. and put it to a context=block and in the extensions.conf put [block] exten => X.,1,Congestion |
19:14.15 | PTG123 | whats that gonna do? |
19:14.46 | PTG123 | ok well i fixed my polycom so it sends the proper stuff |
19:14.57 | ariel_ | It just might send these calls to that context instead of wide open server. |
19:15.41 | PTG123 | well |
19:15.47 | PTG123 | see if you can make a call through my server right now |
19:16.05 | ariel_ | whats your exten? |
19:16.37 | PTG123 | um |
19:16.39 | PTG123 | call a phone number |
19:16.43 | PTG123 | you don't need to know an extension |
19:16.53 | *** join/#asterisk anthm (~anthmct@69.76.83.52) |
19:16.53 | *** mode/#asterisk [+o anthm] by ChanServ |
19:16.59 | ariel_ | I just made another ld call... |
19:17.13 | PTG123 | yep |
19:17.14 | PTG123 | it went through |
19:17.19 | PTG123 | 305 number? |
19:17.23 | ariel_ | yes |
19:17.26 | PTG123 | man bad bug |
19:17.30 | PTG123 | ok trying your changes now |
19:17.34 | PTG123 | now try |
19:17.59 | PTG123 | bad bug |
19:17.59 | PTG123 | :) |
19:18.08 | jontow | why do you think the makefile says to read SECURITY ;) |
19:18.11 | ariel_ | PTG123, still worked. |
19:18.20 | PTG123 | jontow: you got a fix for this? |
19:18.31 | jontow | make sure your contexts are safe |
19:18.38 | PTG123 | jontow: how would you do that? |
19:18.41 | jontow | and yeah.. put SIP traffic behind the firewall |
19:18.55 | PTG123 | jontow: i need to allow people to use sip to my server |
19:19.15 | jontow | if nothing is accessible to [default] and your sip.conf says throw all traffic that isn't registered to specific phones in default or from-sip-restricted, or whatever the hell.. |
19:20.10 | Darwin35 | anyone done overhead paging with dsp |
19:20.18 | PTG123 | ariel_: now try? |
19:20.42 | ariel_ | ok it's blocked |
19:20.53 | PTG123 | ok i bet this works on 99% of peoples servers |
19:20.56 | PTG123 | heh |
19:21.22 | PTG123 | free phone calls for everyone |
19:21.27 | Darwin35 | ptg what you working on |
19:21.56 | PTG123 | Darwin35: found a way to make free phone calls on everyones asterisk servers |
19:22.00 | *** join/#asterisk Smythe (~Smythe@spock.cbcag.edu) |
19:22.09 | Darwin35 | ? |
19:22.10 | ariel_ | PTG123, well at least there is a way to stop this. |
19:22.21 | PTG123 | ariel_: true |
19:22.23 | Darwin35 | I want the fix to block it |
19:22.33 | Darwin35 | hehhe |
19:22.34 | PTG123 | just set your default to context=block |
19:22.38 | Darwin35 | just for safty |
19:22.44 | PTG123 | and put a block context in extensions that dumps to congestion |
19:22.57 | ariel_ | Darwin35, it's easy just have your default have only one line that says exten => X.,1,Congestion |
19:23.02 | Smythe | does anyone have any expertise in tying an Asterisk box to a Mitel over a T1? (not PRI) |
19:23.04 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
19:23.18 | Darwin35 | hh |
19:23.21 | PTG123 | ok ariel wanna try it on another ip? |
19:23.30 | Darwin35 | well back to overhead paging with dsp |
19:23.54 | PTG123 | 208.139.204.228 |
19:23.56 | PTG123 | try that one |
19:24.30 | Darwin35 | need to be able to tie * into a pa system |
19:25.16 | *** join/#asterisk nicholas_ (~nicholas@pD953A7A7.dip.t-dialin.net) |
19:25.20 | nicholas_ | hi |
19:26.28 | Smythe | my Mitel SX2000 doesn't like calls initiated from the Asterisk box. Any ideas? |
19:27.16 | Ron-Na | Can anybody help me with regex in ASTCC? |
19:28.01 | modulus_ | how about just plain ol' regex? |
19:28.01 | Ron-Na | How to setup a country and exclude some cities with different rates? |
19:28.04 | modulus_ | i prefer POR |
19:30.36 | *** join/#asterisk Syncros (~sysop@noc.routermonkey.net) |
19:30.36 | Darwin35 | ok where is the console ansewr setup .. grr |
19:33.22 | Darwin35 | grrrr |
19:33.27 | Darwin35 | this should be easy |
19:34.29 | _RaYmAn_ | PTG123, what was the actual problem? default context not being setup correctly? Or can it be done regardless of that? |
19:36.44 | ariel_ | sorry PTG123 I was busy with my baby. did you try that ip address you just posted yet? |
19:38.34 | *** join/#asterisk Gh0sty (~Ghosty@81.11.192.116) |
19:44.41 | *** join/#asterisk mikegrb (~michael@thegrebs.com) |
19:46.57 | *** part/#asterisk Smythe (~Smythe@spock.cbcag.edu) |
19:50.09 | nicholas_ | are there any problems with authentication from kphone with asterisk? it seems generates the wrong response during digest authentication |
19:51.22 | modulus_ | i prefer POR |
19:52.50 | nicholas_ | linphone works |
19:53.48 | jontow | :) |
19:54.26 | jontow | i've yet to test a soft-phone under a platform other than windows |
19:54.39 | jontow | and i was told i need to have recommendations ;) sooo.. saturday at work compiling X.org |
19:54.39 | jontow | hehe |
19:58.04 | nicholas_ | can i enable output why asterisk reject a SIP register? "sip debug" only shows what pakets are transmitted |
19:58.59 | Darwin35 | why compile when you can install via pkgs |
20:01.31 | jontow | because i'm paid hourly |
20:01.33 | riksta | jontow: x-lite under wine |
20:01.40 | jontow | :P |
20:01.45 | riksta | i'm not joking |
20:01.50 | riksta | thats about the best |
20:01.50 | jontow | works alright? |
20:01.53 | riksta | yep |
20:01.53 | jontow | nice |
20:02.03 | riksta | there's iaxcomm and a few others |
20:02.05 | riksta | they are crap |
20:02.11 | jontow | :/ |
20:05.25 | *** join/#asterisk Gh0sty (~Ghosty@81.11.192.116) |
20:06.08 | Darwin35 | hmm make a way for them to record and them play it back on th e pa hmm that might work |
20:06.40 | *** join/#asterisk AhmedFouad (~xor@82.201.208.1) |
20:06.46 | AhmedFouad | hi all |
20:06.52 | AhmedFouad | i need an advice in something |
20:07.17 | riksta | don't eat yellow snow |
20:07.19 | AhmedFouad | need to connect 2 connect two seperate offoices overseas with asterrisk |
20:07.38 | file[airport] | lol I'm just sitting here and wireless APs keep appearing |
20:07.54 | file[airport] | originally at 12, now at 17... lol |
20:07.57 | riksta | file[airport]: you wanna try living in student halls like me :) |
20:08.17 | file[airport] | oh well, I like my paid wifi |
20:08.22 | file[airport] | not NATTed |
20:08.33 | riksta | ya |
20:08.52 | riksta | crazy how many unsecured APs there are tho |
20:09.26 | file[airport] | woot, fileserver at the airport! max out the bandwidth! |
20:10.37 | *** join/#asterisk ionix (ionix@MTL-HSE-ppp184758.qc.sympatico.ca) |
20:10.50 | ionix | sup sup |
20:11.12 | ionix | Anyone has an idea on how to do prepaid for an asterisk<->asterisk solution ? |
20:11.22 | *** join/#asterisk claint (~claint@195.174.25.120) |
20:11.24 | ionix | so that there might be multiple SIP/IAX2 channels at the same time |
20:11.25 | file[airport] | yes, it's called using astcc or writing code :) |
20:11.28 | riksta | wiki |
20:11.35 | ionix | astcc == calling card |
20:11.40 | ionix | I want to do SIP-SIP |
20:11.43 | file[airport] | astcc can be modified |
20:11.44 | ionix | or IAX2-IAX2 |
20:12.01 | ionix | and this will allow multiple concurrent connections ? |
20:12.06 | file[airport] | sure, why not? |
20:12.13 | ionix | last I check, it has multiple use prevention |
20:12.21 | file[airport] | you can modify it... |
20:12.27 | file[airport] | there is no package out there to do what you want, it's not that easy |
20:12.30 | riksta | it's silly questions night |
20:12.33 | file[airport] | you have to create or modify |
20:12.40 | ionix | yeh, I planned to create |
20:12.40 | Qwell | -- if (conn.count > 1) return; |
20:12.41 | Qwell | ++ |
20:12.51 | riksta | ill write it for you, for a fee :) |
20:12.55 | ionix | however, I was thinking about the best approach |
20:13.10 | ionix | like, use asterisk manager, AGIs or wrote a custom module |
20:14.08 | Darwin35 | http://pastebin.ca/7329 |
20:14.45 | wildcard0 | ionix, are you planning on gpl'ing it? |
20:14.50 | ionix | yeh |
20:15.10 | ionix | I hate private source code, this is why I won't use the NACT we have to do it |
20:15.16 | wildcard0 | then i'll tell you how to modify asterisk/astcc to do it without changing huge amounts :) |
20:15.24 | ionix | hehe nice |
20:15.33 | *** join/#asterisk Lee__ (~lee@69-203-206-248.nyc.rr.com) |
20:15.46 | wildcard0 | lemme just pull up my notes then we can take this private so we don't annoy the rest of the channel |
20:15.55 | ionix | ok |
20:16.00 | riksta | why not leave it public so we can chuck in our comments |
20:16.16 | wildcard0 | sure. i just didn't want to get too off topic with coding stuffs |
20:16.33 | riksta | i fail to see how that's off topic |
20:16.35 | riksta | :P |
20:16.38 | wildcard0 | ok. this is how i was planning on doing it: |
20:16.40 | wildcard0 | riksta, :) |
20:17.06 | wildcard0 | first, modify app_dial so that it stores the time for it's 'L' variable in shared memory |
20:17.15 | wildcard0 | http://www.voip-info.org/wiki-Asterisk+cmd+Dial for reference |
20:17.47 | wildcard0 | and give it an optional key also. |
20:18.08 | wildcard0 | so that all the cc times can be stored in the same place |
20:18.15 | file[airport] | it's so very dead at this terminal |
20:18.17 | file[airport] | there's, like, nobody |
20:18.36 | Qwell | file[airport]: San Jose? |
20:18.41 | Darwin35 | http://pastebin.ca/7331 |
20:18.42 | file[airport] | yes |
20:18.46 | Qwell | odd, for a Saturday |
20:18.57 | Qwell | I'd bet LAX is extremely packed right now. |
20:19.04 | zoa | heyf ile |
20:19.06 | zoa | hey file |
20:19.06 | wildcard0 | then it's just a matter of modifying astcc to write to that shared memory location to compute it's times. |
20:19.15 | file[airport] | zoa: twisted and bkw hate you, they got what you had |
20:19.19 | Qwell | file[airport]: See if you can't get a flight down to LAX or ONT. :p |
20:19.24 | file[airport] | <PROTECTED> |
20:19.26 | zoa | file how do you mean ? |
20:19.31 | file[airport] | zoa: they're sick |
20:19.33 | zoa | the bulgarian virus ? |
20:19.36 | zoa | Haha cool |
20:19.37 | zoa | :) |
20:19.43 | Qwell | They got sick from food zoa suggested? |
20:19.49 | Qwell | erm, nm |
20:19.50 | zoa | i dont think it was infectuus though |
20:19.51 | zoa | :) |
20:19.57 | file[airport] | apparently same symptoms you had |
20:20.06 | Qwell | file's next |
20:20.07 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
20:20.09 | wildcard0 | it's not really that difficult. i keep meaning to do it, but i haven't had the time yet |
20:20.12 | zoa | that will teach em from touching me all the time :p |
20:20.16 | file[airport] | :p |
20:20.20 | ionix | hmm astcc... |
20:20.21 | ionix | <PROTECTED> |
20:20.21 | ionix | <PROTECTED> |
20:20.21 | ionix | <PROTECTED> |
20:20.21 | ionix | <PROTECTED> |
20:20.21 | ionix | <PROTECTED> |
20:20.23 | ionix | <PROTECTED> |
20:20.23 | file[airport] | oh, c'mon... lemme on my fileserver |
20:20.24 | file[airport] | PALEEZ |
20:20.25 | zoa | do they also have the horrible jetlag ? |
20:20.28 | ionix | so it means that it doesn't allow many calls |
20:20.37 | file[airport] | zoa: ha |
20:20.43 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
20:20.51 | riksta | ionix: dude, how hard can it be to remove that functionailty.....i take it you are obviously not much of a coder? |
20:20.55 | wildcard0 | ionix, it's the 'inuse' variable |
20:21.06 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
20:21.07 | wildcard0 | you can just set that to return 0 all the time |
20:21.24 | wildcard0 | but if you do that, then calls can run over the max amount if there are concurrent calls at the end of the card's limit |
20:21.41 | wildcard0 | the shared memory thingy fixes that correctly |
20:22.01 | ionix | k |
20:22.02 | WhiteWlf | what's the bitrate of ulaw? |
20:22.06 | ionix | 64kbits |
20:22.11 | WhiteWlf | thanks :) |
20:22.27 | ionix | riksta: Problem is that I want to manage simultanious connections |
20:22.34 | WhiteWlf | GSM is ~13, correct? |
20:22.35 | wildcard0 | ionix, otherwise it's just a matter of adding 'return;' as the first line in the checkinuse() function |
20:22.40 | file[airport] | frell frell frell |
20:22.43 | ionix | wildcard0: How it fixes that ?? |
20:22.56 | file[airport] | this thing has such a crappy route to Canada |
20:23.15 | ionix | i.e: Will it have to adjust all timeouts on all connections for the account on each connect/disconnect |
20:23.21 | file[airport] | oh wait, someone is eating all the bandwidth |
20:23.37 | wildcard0 | ionix, the shared memory thing will take care of that cause they'll all be reading from the same memory location |
20:23.45 | _Vile | rm -rf file |
20:23.57 | file[airport] | brb |
20:23.57 | lesouvage | I read on the digium site that the X100P is discontinued. What is the (productcode) of the new alternative? |
20:24.25 | *** join/#asterisk Jer13261 (~Jer@rdu57-251-152.nc.rr.com) |
20:24.46 | Jer13261 | can someone help me in getting a channel var set in a dialplan that isnt working? |
20:24.57 | ionix | wildcard0: Hmm, let say I make a call to Canada for 1.1Ē/min. The app calculates 200 minutes timeout. Then an other call with the same account is made to Morocco at 10Ē/min. How will the shared memory update the Canadian call timeout ? |
20:25.11 | jontow | lesouvage; the alternative is a TDM400P card, iirc |
20:25.24 | jontow | or you can buy a cheap X100P clone on ebay, i hear.. |
20:25.39 | Jer13261 | yea like $7 each... |
20:26.11 | wildcard0 | ionix, astcc will make that change in the db. that's not difficult. the hard part is changing the limits for the calls in progress |
20:26.20 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
20:26.27 | ionix | yeh excatly |
20:26.30 | *** part/#asterisk Jer13261 (~Jer@rdu57-251-152.nc.rr.com) |
20:26.40 | wildcard0 | which is what the shared memory thing does |
20:26.49 | *** join/#asterisk Jer13261 (~Jer@rdu57-251-152.nc.rr.com) |
20:26.52 | ionix | I see |
20:26.56 | wildcard0 | you need to modify app dial to do it cause it's passed in as a parameter |
20:27.51 | zoa | do they also have the horrible jetlag ? |
20:28.15 | *** join/#asterisk file[airport] (file@dhcp64-134-126-76.sjca.sjc.wayport.net) |
20:28.40 | wildcard0 | ionix, basically you need to modify chan->whentohangup to be point to a shared memory location |
20:28.46 | zoa | i gained 4 kg when i was at von |
20:28.47 | wildcard0 | er -be |
20:28.50 | zoa | damn american food |
20:29.12 | wildcard0 | heh funny. i go down to the states and i lose weight |
20:29.44 | ionix | k, I'll try that |
20:30.24 | wildcard0 | ionix, it'll take some mucking, but once it's set up there, you'll be able to modify that number directly from astcc |
20:31.04 | wildcard0 | just make sure each card gets its own shared memory location for the time variable |
20:31.33 | *** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com) |
20:31.51 | TheBear | ok solved my zap/2 problem the power cable had come out of the TDM card ??? |
20:32.18 | TheBear | When I pick up the phone connected to the TDM card I get a dial tone, when I dial number it is not recognised, why ? |
20:32.53 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
20:36.09 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
20:37.44 | lesouvage | jontow: a telephone line in and a network connection is all I'm looking for. What should be the benefits of spending $ 125 for a TDM400P, what are the extra's? |
20:38.41 | zoa | did anyone in here use app_icd lately ? |
20:42.58 | eKo1 | lesouvage: Get an SPA-3000 |
20:43.18 | Jer13261 | is there any way to set a channel var without using an exten line?...eg becuase the user hasnt dialed anything yet :) |
20:44.00 | anthm | setvar => var=val in the friend def |
20:45.03 | Jer13261 | setvar foo=bar |
20:47.08 | Jer13261 | setvar isnt listed as a opt in sip.conf |
20:47.58 | wildcard0 | globals? kinda messy... |
20:48.51 | Jer13261 | well i need to define a default area code on a per phone basis....its not even accepting me trying to set a global |
20:49.35 | lesouvage | eKo1: I 'm working out a plan to built a dedicated asterisk box based on the epia 5000 (mini) motherboard. I will need a kind of X100P card to connect the box to the world. |
20:51.43 | jontow | lesouvage; maybe you should buy a few of the ebay X100P clones then.. |
20:51.50 | jontow | i bought 3 and have been experimenting slowly with 1 of them |
20:51.51 | Darwin35 | ok it seems to work where is my speaker |
20:51.58 | jontow | mostly to test the netbsd drivers |
20:52.33 | anthm | line 197 of the cvs sip.conf.sample |
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20:53.08 | *** join/#asterisk rowter (~Drake@201.135.26.29) |
20:53.30 | rowter | anyone has manage to work soyo G668 with *? |
20:53.50 | file[airport] | anthm: so, like, bkw is in the air |
20:53.57 | rowter | I have firmware version 1.42 |
20:53.58 | *** join/#asterisk Himeko (~himeko@S01060040ca128fc3.ed.shawcable.net) |
20:54.10 | zoa | anthm, when do we finally get to meet ya ? |
20:54.23 | anthm | aug 3 when you come to cluecon |
20:54.55 | file[airport] | so, 1PM and this airport is not packed at all |
20:55.28 | zoa | anthm, what is the most recent version for app_icd and is it considered stable ? |
20:56.00 | anthm | probably the on in it's cvs |
20:56.12 | zoa | thats 10 months old ? |
20:56.26 | zoa | could that be ? |
20:56.33 | zoa | or am i looking at the wrong stuff |
20:56.49 | anthm | probably |
20:57.02 | anthm | the one on orson.callenish.com |
20:57.39 | zoa | got a full link ? |
21:00.08 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
21:00.52 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
21:02.12 | anthm | http://voip-info.org/wiki-ICD |
21:02.24 | anthm | that'll be $50 google fee |
21:02.27 | zoa | thnx i should have knowmn L) |
21:02.30 | zoa | :) |
21:02.51 | *** join/#asterisk dev-null (~real@meitner.wh.Uni-Dortmund.DE) |
21:03.28 | *** join/#asterisk darby_t (~tom@dnk250.neoplus.adsl.tpnet.pl) |
21:03.29 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
21:06.14 | lesouvage | jontow: I just did. There are plenty of x100p and clowns on ebay. It's my first ordered part of the asterisk-mini. |
21:06.38 | *** part/#asterisk eKo1 (~bernd@63.245.57.70) |
21:10.36 | *** join/#asterisk zippp (~zip@c66.190.109.98.ts46v-01.rckprt.tx.charter.com) |
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21:16.42 | Darwin35 | is there a list of all the functions in * |
21:16.49 | Darwin35 | that can be mapped |
21:17.20 | Juggie | functions in the code? |
21:17.24 | Juggie | or dialplan functions |
21:17.42 | Darwin35 | dialplan functions |
21:18.00 | Juggie | on the wiki |
21:18.12 | Juggie | use google and search for dialplan applications |
21:18.20 | Juggie | the google search on the page i mean |
21:18.46 | *** join/#asterisk buskila (~buskila@CBL217-132-75-104.bb.netvision.net.il) |
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21:21.17 | *** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net) |
21:21.34 | *** join/#asterisk [spi] (~nothing@dD5E021D2.access.telenet.be) |
21:21.39 | [spi] | hi ppl |
21:22.00 | *** join/#asterisk _Bradders (~Bradders@c220-237-64-249.fitzg1.qld.optusnet.com.au) |
21:22.04 | _Bradders | Heh all! |
21:22.08 | Moc | hi |
21:22.50 | [spi] | I have the following error when trying to call a netmeeting client with sjphone in sip mode to test sip <-> h323: Spawn extension (default, #, 2) exited non-zero on 'SIP/mysjphone-e386' |
21:22.50 | [spi] | <PROTECTED> |
21:22.50 | [spi] | <PROTECTED> |
21:22.50 | [spi] | <PROTECTED> |
21:22.50 | [spi] | <PROTECTED> |
21:22.52 | [spi] | <PROTECTED> |
21:23.34 | Moc | h323 is evil |
21:23.46 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
21:24.23 | [spi] | what are these security checks and how can I fix this? |
21:24.40 | Darwin35 | ok I need to add *7-100 dialing |
21:24.54 | Darwin35 | for each extension |
21:25.01 | _Bradders | Just doing a bit of a hunt around for a suitable solution to add SIP capabilities to GAIM for Linux |
21:25.16 | _Bradders | Anyone have any suggestions? |
21:25.28 | Moc | that be cool |
21:25.37 | Moc | but IAX in gaim might be better |
21:25.47 | _Bradders | There are a couple of ports of linphone |
21:25.48 | _Bradders | phonegaim |
21:25.49 | Moc | especially that GAIM was first made by Mark Spencer ;) |
21:25.50 | _Bradders | gaim-vv |
21:25.54 | _Bradders | IAX? |
21:25.59 | Darwin35 | I wish yahoo would get voip in thier *nix client |
21:26.11 | _Bradders | MSN is _almost_ VoIP/SIP |
21:26.27 | Moc | whinning again about BSD ? ;) |
21:26.46 | Darwin35 | no |
21:27.04 | Darwin35 | I have asterisk kicking ass on fbsd |
21:27.37 | Lee__ | anyone have positive useage with asterisk on OpenBSD? |
21:27.43 | Darwin35 | just adding feature/dialplan functions |
21:27.45 | _Bradders | OOh... IAX! |
21:27.53 | Moc | ;) |
21:27.58 | _Bradders | Works through NAT, no need for STUN! |
21:28.03 | Moc | oh yea |
21:28.21 | Qwell | could probably borrow chan_iax, heh |
21:28.30 | Moc | well there is a libiax I think |
21:28.32 | Qwell | as long as its GPL...which it probably will be |
21:28.39 | Qwell | dunno, that would work too :p |
21:28.40 | _Bradders | Perhaps I should be looking at AIX then! |
21:28.45 | _Bradders | IAX that is |
21:28.47 | Qwell | eww, not aix |
21:28.47 | _Bradders | faux paus! |
21:29.08 | Qwell | yeah...if you release an iax plugin for gaim, I'll use it. |
21:29.19 | Darwin35 | ok everyone over to Solaris |
21:29.20 | Moc | I got samba + LDAP + Domain controler working like I want today |
21:29.37 | Moc | except for a minor Local Administrator issues |
21:29.52 | _Bradders | Is IAX supported on anything but Asterisk? |
21:30.02 | file[airport] | ya know what, I have yet to have been asked to turn on my laptop when going through security |
21:30.10 | Moc | yes, yat or something support it I think |
21:30.15 | Darwin35 | use speex |
21:30.20 | Moc | file, they dont anymore |
21:30.20 | Darwin35 | or ilbc |
21:30.41 | file[airport] | I even snuck my Mac Mini past security |
21:30.44 | Moc | file[airport], they use somekind of smell detector |
21:30.44 | file[airport] | they didn't ask me to take it out or anything |
21:31.08 | _Bradders | So its really an Asterisk initiative? |
21:31.16 | Moc | because they know that doesnt mean the laptop power, it aint have nice C4 in the DVD Bay |
21:31.23 | Qwell | _Bradders: It stands for "inter asterisk exchange", so... |
21:31.26 | _Bradders | Can IAX go directly from client to client? |
21:31.29 | _Bradders | Ahh...! |
21:31.32 | _Bradders | That explains it |
21:31.40 | Darwin35 | iax is best for trunking asterisk to asterisk |
21:31.55 | _Bradders | Ok.. so I'm probably stuck with SIP |
21:31.55 | Qwell | but it works great for clients too...NAT and all |
21:31.58 | Moc | yes trunking help alot, could double the number of channel with the same bandwidth |
21:32.16 | Moc | _Bradders, good luck ;) |
21:32.20 | _Bradders | I've gone to the trouble of writing STUN support for linphone, so I think I might stick with that :) |
21:32.54 | _Bradders | I am writing a wrapper for linphone, if It gets good I might write a IAX wrapper ! |
21:32.59 | [spi] | I have a (Call ended due to security checks) error which config files do I have to check for a sip to h323 client? |
21:33.15 | Darwin35 | most *nix voip clients suck |
21:33.19 | _Bradders | Wraps SIP in HTTP/TLS or STUN |
21:33.45 | Darwin35 | asterisk is the best thing to come along |
21:34.01 | _Bradders | Linphone is kind of cool http://www.linphone.org/?lang=fr&rubrique=3 |
21:34.15 | *** join/#asterisk Stbjr[PuterShow] (~stboch@pcp0010759468pcs.howard01.md.comcast.net) |
21:34.17 | _Bradders | Asterisk is a full blown PABX, all I want is p2p messaging using voice :) |
21:34.28 | _Bradders | We already have SIP comliant PABX's |
21:36.49 | _Bradders | Just need to the p2p client |
21:38.07 | Lee__ | skype, but it isn't free software. some buddies of mines are raving about it. |
21:39.07 | lindi- | somebody should put together a free alternative indeed |
21:40.32 | Qwell | skype kinda sucks ;/ |
21:40.40 | Qwell | proprietary protocols and all |
21:40.56 | Lee__ | I didn't say my friends were above sucking :) |
21:41.04 | Qwell | most aren't |
21:41.45 | Lee__ | I don't know anything about it except that they are using it to make international phone calls on the cheap and record the session with nothing more than a laptop. |
21:42.08 | [spi] | I have a (Call ended due to security checks) error which config files do I have to check for a sip to h323 client? |
21:44.16 | lindi- | Qwell: point is that i have a feeling there already are free alternatives but nobody has put them together into something that an average windows user can install |
21:44.55 | Lee__ | that assumes one cares about the average Windows user as a top priority |
21:46.38 | Qwell | Lee__++ |
21:46.52 | lindi- | Lee__: i see your point but when it comes to communicating with people then it's quite important |
21:47.00 | spackle | ~karma |
21:47.00 | jbot | spackle has neutral karma |
21:47.30 | Lee__ | when it comes to making a commercial product it's important. communicating can be done for free for those who wish. |
21:47.34 | *** join/#asterisk Mike (~mike@201.129.122.206) |
21:48.00 | lindi- | Lee__: er, those who have the technical skills :) |
21:48.13 | Beirdo | anyone here from the UK who can give my UK DID a ding so I can see how the callerID is formatted? |
21:48.54 | Lee__ | wow. Icecast + asterisk! |
21:49.09 | Lee__ | anyone here actually using that feature? |
21:49.17 | Qwell | Lee__: That exists? |
21:49.21 | Qwell | oh, for MoH? |
21:49.30 | Lee__ | it's in the Debian package documentation |
21:49.53 | Lee__ | <PROTECTED> |
21:52.49 | rvhi | anyone use proftpd? |
21:52.58 | rvhi | i created the account, but can't ftp into it |
21:53.05 | rvhi | i can login |
21:53.14 | rvhi | so my phone can't get the config |
21:53.15 | Lee__ | man proftpd |
21:54.09 | rvhi | proftpd config should be fine. it was running ok |
21:54.14 | rvhi | i just added this new user |
21:54.20 | rvhi | my linux admin is on vacation |
21:54.23 | rvhi | :( |
21:54.25 | *** join/#asterisk robbins (~robbins@adsl-068-209-107-007.sip.mia.bellsouth.net) |
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21:56.31 | robbins | hey folks, i want to load only the modules i'll need for asterisk like an embedded system, is there a doc describing the modules and their associated functions? |
21:57.12 | kimc | hello asterisk |
21:58.34 | robbins | i've read asterisk slimming on voip-info, but it's pretty sparse |
21:58.58 | kimc | I'd be happy just get dialtone out of a digium port |
22:00.00 | kimc | I've got battery on the port but no asterisk dialtone.. |
22:04.37 | *** join/#asterisk l-fy (~diana@l-fy.developer.yate) |
22:04.41 | l-fy | ihaaaaaaaaaaaa |
22:05.23 | l-fy | bwahahahahaha |
22:05.24 | *** part/#asterisk l-fy (~diana@l-fy.developer.yate) |
22:12.52 | *** join/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net) |
22:12.54 | Lee__ | so I think I have this right. I can set up an Asterisk server in my lab and have two computers with softphones connect to it just to test it's basic VoIP->VoIP capabilities, right? |
22:13.13 | mrgoby | i'm having dtmf detection problems using iax using nufone |
22:13.18 | mrgoby | doesnt work using ulaw or gsm |
22:13.28 | file[airport] | mrgoby: outbound or inbound? |
22:13.34 | mrgoby | out |
22:13.42 | file[airport] | it's not NuFone, it's your SIP phone |
22:13.45 | mrgoby | to my cell |
22:14.02 | mrgoby | it is iax and asterisk |
22:14.05 | mrgoby | so, i doubt it |
22:14.20 | file[airport] | outline how you're placing the call |
22:14.31 | mrgoby | call file |
22:14.40 | mrgoby | answer |
22:14.44 | mrgoby | wait |
22:14.57 | mrgoby | no extens can be dialed |
22:15.09 | mrgoby | audio is fine |
22:15.12 | file[airport] | okay, well... uh... answer and wait won't accept DTMF... |
22:15.20 | file[airport] | you have to use background or waitexten... |
22:15.20 | mrgoby | echo works |
22:15.23 | *** join/#asterisk tecnico (~tecnico@user-24-236-123-31.knology.net) |
22:15.39 | mrgoby | background is what i'm using |
22:15.44 | file[airport] | see, you didn't say that |
22:16.05 | file[airport] | and the context is included in the one you're in so background can match against the extension? |
22:16.14 | mrgoby | i know,srry typing one handed :-) |
22:16.31 | mrgoby | same context,yes |
22:16.43 | file[airport] | then do an iax2 debug and see if you get packets for digits |
22:16.50 | mrgoby | ok |
22:19.56 | mrgoby | is rfc2833 sip only ? |
22:20.05 | file[airport] | yes |
22:20.22 | file[airport] | IAX2 uses out of band dtmf all the time |
22:20.28 | mrgoby | ok |
22:20.37 | mrgoby | debugging now |
22:20.39 | mrgoby | one sec |
22:21.01 | file[airport] | now how nice am I, debugging your problem while at the airport |
22:22.08 | kimc | file: can you help me with a basic config? |
22:23.25 | kimc | Trying to get a port to deliver dialtone to a pots phone |
22:25.12 | mrgoby | http://pastebin.ca/7333 |
22:25.16 | mrgoby | i appreciate |
22:25.22 | mrgoby | i know i should have a t exten |
22:25.29 | mrgoby | this is just for debugging though |
22:26.08 | mrgoby | when i pressed the dialpad, no output from iax2debug |
22:26.28 | mrgoby | so, it appears it is not accepting it at all |
22:26.35 | mrgoby | my provider on this cell is sprint |
22:27.03 | mrgoby | it works fine if i dial the fxo line |
22:27.24 | mrgoby | from my cell |
22:29.36 | mrgoby | any ideas ? i'm using switch-1.nufone.net.... just fyi |
22:33.01 | mrgoby | file: http://pastebin.ca/7333 |
22:34.36 | *** part/#asterisk kimc (~kimc@pcp04039944pcs.wbrmfd01.mi.comcast.net) |
22:34.39 | mrgoby | brb |
22:38.53 | *** join/#asterisk nwhit (nwhit@wsip-24-234-120-72.lv.lv.cox.net) |
22:39.34 | nwhit | hello all... i have an easy (i hope) question |
22:40.14 | nwhit | i want to allow the incoming lines to dial an extension... there are about 60 sip phones connected... but i don't want to have to put them all in the dial plan |
22:40.54 | nwhit | something like: exten => _NXX,2,Dial(Sip/${EXTEN},20,m) |
22:41.04 | *** join/#asterisk ckruetze (~ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
22:41.08 | nwhit | but it needs to make sure that the extension that they enter is valid |
22:41.13 | nwhit | any way to check first? |
22:42.06 | *** join/#asterisk jeffik (~jeffik@m8b7936d0.tmodns.net) |
22:49.52 | *** join/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net) |
22:51.16 | mrgoby | i've got to get a new phone... crappy 2.4ghz phone wipes out my wireless everytime it rings... even though i have it set to like channel 32 |
22:51.48 | mrgoby | so, anyone have any ideas about this nufone/iax/dtmf dealio ? |
22:53.01 | mrgoby | driving me nuts |
22:53.19 | modulus_ | what's a voip? |
22:53.29 | mrgoby | ~voip |
22:53.30 | jbot | extra, extra, read all about it, voip is Voice over IP |
22:53.30 | *** join/#asterisk lters (~lters@mrtc-mm-600046.mis.net) |
22:55.53 | modulus_ | what's a IP? |
22:56.01 | mrgoby | ~IP |
22:56.02 | jbot | mrgoby IP is a connectionless, best-effort packet switching protocol. It provides packet routing, fragmentation and re-assembly through the data link layer. [internet protocol] |
22:56.28 | mrgoby | ~newbie |
22:56.30 | jbot | rumour has it, newbie is someone who is new to linux or debian, and should read the docs (/usr/share/doc/) |
22:56.49 | nwhit | ~funnyguy |
22:56.58 | mrgoby | ~justkidding |
22:57.04 | nwhit | ~haha |
22:57.05 | jbot | heh |
22:57.08 | *** join/#asterisk Nebukadneza (~daddel9@i3ED6E179.versanet.de) |
22:57.08 | *** join/#asterisk Delmar (~Delmar@222-152-57-78.adsl.inspire.net.nz) |
22:57.10 | *** part/#asterisk Nebukadneza (~daddel9@i3ED6E179.versanet.de) |
22:57.33 | *** join/#asterisk Nebukadneza (~daddel9@i3ED6E179.versanet.de) |
22:57.36 | *** part/#asterisk Nebukadneza (~daddel9@i3ED6E179.versanet.de) |
22:57.52 | modulus_ | who wants to see my calling card agi script? |
22:57.58 | modulus_ | it's in perl |
22:58.12 | mrgoby | pastebin dat mug |
22:58.21 | Delmar | sure modulus_ |
22:58.38 | modulus_ | http://pastebin.ca/7335 |
22:58.41 | modulus_ | tell me what you guys think |
22:59.10 | Delmar | before i look... |
22:59.10 | nwhit | hey can someone help with a dial plan problem i am having |
22:59.13 | Delmar | what does it do :P ? |
22:59.19 | modulus_ | calling card |
22:59.29 | modulus_ | read it |
22:59.57 | modulus_ | line 047 is my favorite line: if ($bal < 1.05){ |
23:00.24 | modulus_ | perl automagically knows a null value is less than "1.05" |
23:01.12 | nwhit | i want to allow the incoming lines to dial an extension... there are about 60 sip phones connected... but i don't want to have to put them all in the dial plan |
23:01.17 | nwhit | something like: exten => _NXX,2,Dial(Sip/${EXTEN},20,m) |
23:01.34 | nwhit | but it needs to make sure that the extension that they enter is valid... what should i do to check first |
23:01.56 | modulus_ | use exten => i,1,Playback(invalid) |
23:02.04 | Essobi | WASABI |
23:02.05 | modulus_ | <PROTECTED> |
23:02.43 | nwhit | it just hangs up the line if, for example they enter 555 and Sip/555 doesn't exist |
23:02.50 | mrgoby | i dont know much about perl::AGI |
23:03.01 | Essobi | AGI ain't hard |
23:03.10 | modulus_ | there's some fun regex in my code too: |
23:03.11 | modulus_ | 041 $bal = $dollars = $cents = $row[0]; |
23:03.11 | modulus_ | 042 $dollars =~ s/(\d*)\.\d*/$1/; |
23:03.12 | modulus_ | 043 $cents =~ s/\d*\.(\d{2}).*/$1/; |
23:03.12 | mrgoby | but it would be helpful to make the provider a variable |
23:03.21 | Essobi | It's actually pretty lame. :) |
23:03.24 | mrgoby | perhaps |
23:03.31 | modulus_ | yeah asterisk::AGI is dumb |
23:03.38 | mrgoby | why lame? |
23:03.47 | modulus_ | it's just a module that does STDIN and STDOUT |
23:03.51 | Essobi | I wrote a CallerID/CMS lookup push screen in 20 minutes. |
23:04.59 | Essobi | Hell.. it took me longer to get the XMLhttpget refresh code right then that AGI. |
23:05.18 | modulus_ | that's b/c XML is brain-dead |
23:05.27 | modulus_ | and so is html |
23:05.31 | Essobi | But I tell you it's better then Cisco TCL.. Statefull/Callback TCL.. What a fricking joke. |
23:05.54 | modulus_ | jbot tcl? |
23:05.55 | jbot | it has been said that tcl is at http://www.scriptics.com/ or in feed "http://handhelds.org/feeds/tcl" |
23:06.18 | Essobi | I mean really.. who the HELL thought it was a good idea to put a state engine in a halfass implemented TCL. |
23:06.52 | *** join/#asterisk Frantic (~ab@24-193-46-85.nyc.rr.com) |
23:06.52 | modulus_ | yeah hardware sucks in general |
23:06.58 | Essobi | The felling blow.. I dled a sample script from the Cisco Engineering FTP site.. and the 2nd line in the program was a comment, stating.. "Does this work?" |
23:07.41 | modulus_ | no perlers here want to comment on my code? |
23:08.07 | Essobi | What? That regexp? It makes me go blind looking at pattern matching positionals |
23:08.40 | modulus_ | you should see my regex with pointers to arrays of associative array pointers |
23:09.00 | modulus_ | only in the perl world |
23:09.05 | Essobi | FFS.. Just build multi-demenstional arrays. |
23:09.12 | Essobi | you know what I meant. |
23:09.21 | modulus_ | that's boring |
23:09.28 | Essobi | Yea, and readable. |
23:09.35 | modulus_ | and slower |
23:09.50 | Essobi | WTF are you running on? A 386? who cares. |
23:10.01 | nwhit | any other suggestions? |
23:10.02 | modulus_ | high volume load |
23:10.03 | Essobi | WOOOO 0.001seconds FASTER! |
23:10.08 | nestAr | your mom goes to college |
23:10.16 | *** join/#asterisk jets (~jetsn@guardian.pmt.org) |
23:10.22 | Essobi | hi vol is saved for C. |
23:10.31 | modulus_ | not necessarily |
23:11.27 | *** join/#asterisk cjk (~cjk@80.92.75.100) |
23:11.41 | cjk | hi, is there a limit of callgroups i can have? |
23:11.51 | modulus_ | cjk, rtfs |
23:11.56 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
23:14.14 | cjk | modulus_, ok you see i know the word callgroup, so where to you guess did i found that. now think again. why might i have asked that question. and now you pretty cool rtfX-answering guy if you do not want to help me, than just do not answer. english is not everyone's native language. |
23:14.16 | Essobi | he Callgroup= setting defines call group for calls to this device. |
23:14.18 | Essobi | In v1.0 and previous versions of Asterisk, call groups are numbered 0-31. |
23:14.20 | Essobi | In v1.1dev and v1.2, call groups are numbered 0-63 |
23:14.35 | cjk | Essobi, yeah thata what i read |
23:15.03 | cjk | so i can have 64 callgroups |
23:15.56 | Essobi | Wow.. Call groups are kinda.... Retarded. |
23:15.58 | Essobi | Heh. |
23:16.33 | modulus_ | cjk, thanks for your rather long response to my short, terse, and to-the-point response. i've always understood that it's just a suggestion. you're also correct about english, it's not my first nor native language. |
23:16.39 | Essobi | So a call group is the inbound trunk of a pickup group? |
23:17.42 | Essobi | Hey cjk .. WTF does engrish have to do with reading code? |
23:17.52 | Essobi | :) |
23:18.21 | cjk | Essobi, well i try to get the answer without reading the whole * sources, but thanks for the suggestion |
23:19.20 | Essobi | who said you had to read the whole thing? |
23:19.35 | Essobi | I use grep everyday to find what I'm looking for and pull it apart. |
23:19.38 | Essobi | You have to. |
23:19.39 | modulus_ | jbot modular? |
23:19.50 | Essobi | A lot of code just isn't documented. |
23:19.58 | Essobi | like show channels concise |
23:20.08 | modulus_ | essobi, it's not? *shock* |
23:20.12 | nwhit | figured it out... using chanlisavail |
23:20.13 | Essobi | Just ain't documented.. ain't in the help CLI. |
23:20.17 | modulus_ | essobi, like ALL of it isn't documented |
23:20.25 | modulus_ | except stupid dial commands and functions |
23:22.20 | modulus_ | damnit someone tell me how to write my perl code better |
23:22.23 | *** join/#asterisk marc324 (~marc32344@65-39-197-107.dsl.teksavvy.com) |
23:23.49 | Chuji | modulus_ : Heh, don't ask me, I'm usually getting help from you |
23:24.02 | Nugget | `Sauron: http://nugget.livejournal.com/88629.html |
23:24.04 | Chuji | /j #perl |
23:24.27 | modulus_ | yeah asterisk help |
23:24.30 | modulus_ | not perl |
23:25.15 | Chuji | Nuh uh, I don't need asterisk help |
23:25.27 | Chuji | You've helped me with my lousy perl skillz |
23:25.41 | modulus_ | i have? |
23:25.42 | Chuji | Ask anthm, he's a good perler |
23:25.45 | modulus_ | *scratches head* |
23:26.00 | modulus_ | you must be an ultra-ultra-noob then |
23:26.07 | Chuji | haha... yup |
23:26.19 | modulus_ | 'cause i'm pretty noobie at this perl stuff |
23:26.45 | Chuji | Well I'm getting better |
23:27.06 | modulus_ | well i'm getting more obfiscated with my perl |
23:27.09 | Chuji | Thanks to my safari subscription |
23:27.43 | modulus_ | chuji, did you look at my calling card agi script? |
23:27.57 | Chuji | I don't think so, where's it posted? |
23:28.29 | Chuji | I made a calling card app too. Needed to have one that interfaced with our Intranet data in MSSQL |
23:28.59 | *** join/#asterisk modulus_ (modulus@rm-f.net) |
23:29.02 | modulus_ | damnit |
23:29.04 | modulus_ | sorry |
23:29.38 | modulus_ | http://pastebin.ca/7335 |
23:29.54 | Chuji | Did you paste it to irc or something? |
23:29.59 | Chuji | booted cuz of flood |
23:30.07 | modulus_ | pasted the wrong buffer |
23:30.12 | ta[i]nted | modulus_ |
23:30.18 | modulus_ | taintedulus_ |
23:30.19 | ta[i]nted | no money in calling card |
23:30.24 | ta[i]nted | use astcc |
23:30.35 | ta[i]nted | if u are that determined |
23:30.43 | modulus_ | what's astcc? |
23:30.51 | ta[i]nted | RTFS |
23:30.53 | ta[i]nted | jk |
23:30.58 | Chuji | ~astcc |
23:30.59 | jbot | i heard astcc is the asterisk calling card platform. There have been patches so that now you can use it in either a pre-pay or post-pay model. You can find more information about it on the wiki (www.voip-info.org) |
23:30.59 | modulus_ | jbot astcc? |
23:31.00 | jbot | i guess astcc is the asterisk calling card platform. There have been patches so that now you can use it in either a pre-pay or post-pay model. You can find more information about it on the wiki (www.voip-info.org) |
23:31.04 | ta[i]nted | just giving u taste of own medicine |
23:31.20 | modulus_ | astcc is too complex i've looked at this before |
23:31.23 | ta[i]nted | astcc is all hooked up .. u just plug in values and go |
23:31.29 | ta[i]nted | astcc is too complex? |
23:31.37 | ta[i]nted | it's perl for christ's sake |
23:32.15 | file[airport] | this airport is interesting |
23:32.28 | ta[i]nted | rtfc - it'll put hair on your chest |
23:32.33 | ta[i]nted | file[airport] which one are u at |
23:32.42 | file[airport] | San Jose |
23:32.45 | file[airport] | their wireless has a WINS server |
23:32.52 | file[airport] | so you can see all the other Windows boxes connected |
23:32.56 | ta[i]nted | yea |
23:33.00 | Essobi | scarey |
23:33.06 | ta[i]nted | i used to be able to get free net access at sjc |
23:33.15 | modulus_ | tainted, compared to my script it's complex |
23:33.26 | file[airport] | it only cost me $6.95 for an unlimited day pass with Wayport |
23:33.29 | file[airport] | and that's full terminal coverage |
23:33.31 | ta[i]nted | just ssh'd into my own box |
23:33.31 | Chuji | how was von file? |
23:33.38 | file[airport] | loved it |
23:33.40 | file[airport] | miss everyone |
23:33.50 | ta[i]nted | any highlights? |
23:34.00 | file[airport] | the Grandstream business phone is sweet |
23:34.17 | Chuji | I'm good for Von Boston, and Astricon, but I can't do anymore |
23:34.21 | ta[i]nted | what about their other ATAs |
23:34.48 | file[airport] | I can do Astricon atlanta |
23:35.00 | file[airport] | ta[i]nted: it's just combos... but really sweet |
23:35.04 | file[airport] | like FXO/FXS combos, routers, etc |
23:35.07 | Chuji | Atlanta is pretty close for me. That is easy |
23:35.16 | ta[i]nted | nice |
23:35.36 | file[airport] | I have their flyer in my bag |
23:36.04 | modulus_ | tainted, perl can get very complicated |
23:36.22 | file[airport] | it's fun to see all these people go through |
23:36.34 | file[airport] | hot guys and hot girls... |
23:36.57 | ta[i]nted | hot girls at a voip convention? |
23:37.04 | file[airport] | yes there was |
23:37.13 | file[airport] | Tristan was nice, she was at the Switchvox booth |
23:37.31 | hermie | booth babes or actual human beings? |
23:37.39 | file[airport] | actual human beings |
23:37.45 | ta[i]nted | i hope a pic gallery pops up sometime |
23:37.46 | file[airport] | she designed the web interface/look of Switchvox |
23:37.53 | file[airport] | they went out with us a few times for dinner |
23:38.04 | file[airport] | Joshua and Tristan from Switchvox |
23:38.11 | ta[i]nted | switchvox has a nice website |
23:38.13 | ta[i]nted | very clean |
23:38.33 | file[airport] | http://www.desktopsummit.com/photos.php?category_id=19 |
23:38.34 | file[airport] | Tristan is there |
23:38.41 | modulus_ | astcc is borderline bloatware |
23:38.46 | file[airport] | Joshua is the guy with the glasses |
23:39.22 | modulus_ | tainted, there are niches in credit cards still |
23:40.01 | ta[i]nted | modulus_ what do u mean credit cards? |
23:40.08 | modulus_ | oops |
23:40.11 | modulus_ | s/credit/calling/ |
23:40.17 | modulus_ | hahaha |
23:41.02 | Essobi | I need to get my boss to take me to VON next year. |
23:41.03 | Essobi | :| |
23:41.37 | file[airport] | it was a nice experience |
23:41.39 | file[airport] | lemme find pics |
23:42.20 | file[airport] | http://host-a.starnetworks.us/Members/kpfleming/spring_von/photoalbum_view |
23:42.29 | file[airport] | that was the Asterisk Pavilion! |
23:43.10 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
23:43.21 | *** join/#asterisk onixx (1000@CPE0040f47145d1-CM000f9f7f2290.cpe.net.cable.rogers.com) |
23:43.46 | onixx | hi all, anyone else is getting "smsq.c:422: `POPT_ARGFLAG_SHOW_DEFAULT' undeclared " when compiling today's cvs ? |
23:44.37 | Essobi | My boss said the pavilion was pretty tight |
23:44.48 | file[airport] | yeah, full of people |
23:44.54 | Essobi | So what's the news on these new cards with built in DSPs? |
23:45.00 | Essobi | Tight as in.. Nice. |
23:45.19 | file[airport] | what new cards with built in DSPs? |
23:45.33 | file[airport] | the only things we had was an echo canceller card for the TE410P and an IAXy version 2 |
23:45.49 | file[airport] | which were both VERY cute |
23:46.39 | file[airport] | Essobi: I espically liked the carpet myself |
23:47.30 | nwhit | has anyone worked with cisco 7905g phones? |
23:47.48 | Essobi | It was someone else's cards |
23:47.55 | Essobi | $3K a piece |
23:48.50 | Essobi | T1 with a hardware DSP |
23:49.07 | Essobi | 410 with echo cancel ehh? |
23:49.09 | Essobi | MM. |
23:49.24 | nwhit | i am having trouble with the mwi on this cisco phone |
23:49.27 | Essobi | Maybe I can talk the boss into selling those 5400's after all. :) |
23:50.21 | Essobi | I just replaced two 5300's this week with an * box. |
23:50.36 | file[airport] | people in the US are odd |
23:50.49 | Essobi | LOL |
23:50.57 | *** join/#asterisk tuxinator_linuxM (~tuxinator@m010e36d0.tmodns.net) |
23:50.58 | Essobi | No shit.. We're all crazy. |
23:51.12 | file[airport] | tuxinator_linuxM: are you at the San Jose airport? |
23:51.35 | tuxinator_linuxM | file[airport]: Ya |
23:51.41 | tuxinator_linuxM | whre you at? |
23:51.44 | file[airport] | which terminal? |
23:51.51 | file[airport] | I'm in Terminal A |
23:51.55 | tuxinator_linuxM | C |
23:51.57 | *** join/#asterisk bjohnson (~bjohnson@66.11.188.184) |
23:51.58 | file[airport] | between gates A8 and A9 |
23:52.02 | tuxinator_linuxM | I have two hours to kill |
23:52.15 | Essobi | You guys are sad.. Logoff and go have a beer. |
23:52.16 | file[airport] | I have, oh, 7 |
23:52.29 | Essobi | Or seven. :) |
23:52.37 | file[airport] | haha |
23:52.40 | tuxinator_linuxM | man, I am at the expediat cafe, had laptop hookups |
23:52.52 | file[airport] | I just paid $6.95 for wifi access via Wayport |
23:52.58 | file[airport] | full terminal coverage ;) |
23:53.07 | tuxinator_linuxM | Wayport is over here too |
23:53.36 | file[airport] | T-Mobile only covers a portion of here |
23:53.48 | file[airport] | the roaming is nice too, I've been through this entire place and haven't lost my connection |
23:53.51 | Essobi | PSssh... Log off and go have a beer together. Jeees. High-ball, martini.. beer. |
23:54.00 | file[airport] | I can't drink here, or at home |
23:54.02 | file[airport] | I'm only 18 :p |
23:54.06 | Essobi | Baah. |
23:54.07 | tuxinator_linuxM | he he |
23:54.15 | file[airport] | legal age here is 21 |
23:54.55 | tuxinator_linuxM | Do I need to come over there file? I just got comforatable. |
23:55.07 | file[airport] | you don't _need_ to |
23:55.18 | tuxinator_linuxM | you have more time to kill, you come here |
23:55.27 | file[airport] | but I've already been through security once |
23:55.40 | file[airport] | once is enough for anyone |
23:55.49 | tuxinator_linuxM | did they strip search you? |
23:55.55 | file[airport] | no |
23:55.56 | file[airport] | :p |
23:56.14 | Essobi | file[airport] WTF you from? |
23:56.25 | file[airport] | I really wonder sometimes if those things damage any equipment |
23:56.27 | file[airport] | Essobi: Canada |
23:56.29 | file[airport] | Atlantic Canada |
23:56.55 | Essobi | Hah.. "Personal" equipment, or hardware? |
23:56.57 | Essobi | Hehe. |
23:57.02 | file[airport] | both |
23:57.11 | file[airport] | :p |
23:57.12 | Essobi | they answer is both then |
23:57.13 | Essobi | Hehe |
23:57.36 | Essobi | So why was there no gear for the show? |
23:57.43 | file[airport] | it never arrived |
23:57.46 | Essobi | Ahh. |
23:58.00 | Essobi | Hell.. Shoulda hollered. I'd sent 10 phones with my boss. |
23:58.07 | file[airport] | oh we had phones and stuff |
23:58.26 | file[airport] | ...yeah |
23:58.30 | file[airport] | and channel banks! |
23:58.30 | tuxinator_linuxM | file[airport]: they won't let me trough security over there |
23:58.42 | Essobi | haha |
23:58.43 | file[airport] | tuxinator_linuxM: so we shall converse on IRC |
23:58.48 | Essobi | SHUTDOWN! |
23:58.56 | file[airport] | NO! |
23:59.11 | Essobi | "I don't know bob... This guy looks kinda shifty, with a loptop and all..." |
23:59.17 | tuxinator_linuxM | file[airport]: So will be both be losers with our laptops |
23:59.31 | file[airport] | I need streaming TV on my laptop |
23:59.46 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
23:59.49 | tuxinator_linuxM | I'm sure you could file a good porn movie |