| 00:00.05 | flenders | k-man_: you have to set it up with them |
| 00:00.18 | flenders | call forward if unavailable |
| 00:00.19 | k-man_ | flenders, oh... |
| 00:00.21 | flenders | I know engin does that over here |
| 00:00.47 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 00:00.49 | k-man_ | flenders, so if the voip provider went down, ie, they had a major hardware crash or something, would that call forward if unavailable still work? |
| 00:01.38 | flenders | k-man_: I don't believe they would have a major crash like that, or at least, chances are pretty small |
| 00:02.10 | flenders | k-man_: i've been with engin for about 2 years now, and they never went down... |
| 00:03.20 | diclophis-work | has anyone seen a flash (swf) application that talks VOIP ? |
| 00:03.20 | flenders | k-man_: I think any service provider (especially telcos) must have a decent failsafe system in place |
| 00:03.21 | k-man_ | flenders, anything can happen, i am just wondering what happens in the worst case scenario.. .say there was an earthquake and the whole building collapsed for example |
| 00:03.32 | JT | engin does go down flenders |
| 00:03.36 | JT | just not very often |
| 00:03.41 | flenders | k-man_: dunno mate... ring them and ask. :o) |
| 00:03.47 | k-man_ | ok |
| 00:03.49 | k-man_ | thanks :) |
| 00:03.50 | flenders | JT: I guess I'm lucky then |
| 00:04.10 | JT | much more than the pstn though, which generally has never ever gone down except when they fuck up our line phsyically due to changing service |
| 00:04.21 | JT | flenders: no, you just didn't notice them |
| 00:04.24 | flenders | JT: I also have my sip channels on my nagios, and they haven't been down since I put them in there |
| 00:04.31 | JT | they had a few major outages in the last couple of months |
| 00:04.57 | flenders | JT: that might have been just before I added them to the monitoring system |
| 00:05.12 | JT | once they were down for a few hours |
| 00:05.20 | JT | lots of people complaining on voip boards |
| 00:05.53 | JT | engin is good as long as you understand what you are getting and what service levels to expect |
| 00:06.16 | JT | don't expect a good personally relationship with their staff or the staff to be that cluey |
| 00:06.21 | JT | it's like a big behemoth |
| 00:06.35 | JT | s/personally/personal/ |
| 00:07.04 | flenders | JT: yeah, I know, but so is telstra |
| 00:07.10 | flenders | telstra is a lot less likely to go down, though |
| 00:07.10 | JT | yeah |
| 00:07.12 | flenders | :o) |
| 00:07.28 | JT | although the way their cust svc reps speak, you'd think they only have 1 tech |
| 00:07.36 | JT | that operates the entire network |
| 00:07.46 | flenders | hahaha |
| 00:08.01 | JT | "yeah the tech guy is out at the moment, i'll find out with him and call you back later" |
| 00:08.16 | JT | i mean props to him if it's pretty much just him, as it's a massive network :P |
| 00:08.57 | flenders | I was shocked with telstra cust service the other day |
| 00:09.21 | JT | telstra can be useful if you speak to the right person |
| 00:09.27 | JT | speaking to the right person is the hard bit |
| 00:09.29 | k-man_ | how is nodephone's DID plans going? |
| 00:09.39 | flenders | true |
| 00:09.47 | JT | nodephone isn't that cheap |
| 00:10.00 | *** join/#asterisk coppice (n=chatzill@55.157.17.210.dyn.pacific.net.hk) |
| 00:10.07 | k-man_ | no |
| 00:10.09 | k-man_ | its not cheap |
| 00:10.13 | k-man_ | but as i use internode |
| 00:10.18 | k-man_ | and i am in a testing phace |
| 00:10.19 | flenders | we changed our carrier for PSTN here to newtel, which is owned by commander, and at least, their tech team seems to be alright. |
| 00:10.20 | k-man_ | phase |
| 00:10.27 | JT | k-man_: hey your question about if the provider is down, was that inbound or outbound calls? |
| 00:10.37 | flenders | you can always talk to the tech people |
| 00:10.47 | k-man_ | JT, i was talking about inbound |
| 00:10.52 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:10.52 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:10.54 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 00:10.56 | JT | flenders: do newtel just rebill? |
| 00:11.02 | JT | k-man_: ok, provider problem then |
| 00:11.04 | flenders | JT: yeah |
| 00:11.18 | flenders | JT: their rates are great for landlines |
| 00:11.31 | JT | optus have some exchanges with their own gear now |
| 00:11.43 | JT | i doubt they'd beat optus's line rental rates :P |
| 00:11.45 | k-man_ | jt, i guess my real question is "what should one do to ensure you are always contactable on your main phone number if you plan to implement a mostly voip based system?" |
| 00:11.54 | flenders | JT: how mych? |
| 00:11.56 | flenders | much? |
| 00:11.59 | JT | k-man_: pray? |
| 00:12.08 | k-man_ | hmm |
| 00:12.25 | k-man_ | jt, any other suggestions about building a bullet proof system? |
| 00:12.26 | JT | k-man_: imho, main numbers should come in over digital BRI or PRI circuits, but that's just me |
| 00:12.35 | k-man_ | jt, ok |
| 00:12.43 | k-man_ | jt, i think that answers my question |
| 00:12.58 | JT | the Internet is unreliable, but if your provder can reliably implement call forwarding, they might be ok |
| 00:13.05 | JT | your voip provider, that is |
| 00:13.10 | JT | flenders: $20/mo |
| 00:13.23 | JT | flenders: analogue or PRI. |
| 00:13.26 | flenders | wow! |
| 00:13.29 | JT | min 10ch pri |
| 00:13.36 | JT | no line hunt fee |
| 00:13.37 | flenders | that's cheap |
| 00:13.50 | JT | not sure if it's just a temporary offer |
| 00:14.02 | JT | but they've been phone marketing it like crazy |
| 00:14.06 | flenders | are they in north syd? |
| 00:14.15 | JT | yeah i almost want to get a personal 10ch pri just for the fun of it, but not quite |
| 00:14.19 | JT | absolutely |
| 00:14.51 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 00:15.49 | flenders | can you set your outbound caller id to anything you want on a PRI? |
| 00:16.02 | JT | not with telstra |
| 00:16.07 | JT | they sanity check |
| 00:16.13 | JT | to make sure it's a number you own |
| 00:16.28 | flenders | what if the number is a landline owned by us? |
| 00:16.52 | JT | not sure, you might have to enquire if they can add it to an allowed list |
| 00:16.56 | JT | dunno if they would |
| 00:17.19 | mogorman | flenders, if your provider lets you you can |
| 00:17.24 | mogorman | most will |
| 00:17.38 | JT | most, most .au providers, ie. telstra? |
| 00:17.57 | mogorman | sorry |
| 00:18.13 | JT | ah, i was giving him the specific answer for over here |
| 00:18.17 | JT | i've tried it |
| 00:18.28 | Cheetah | thanks for your support earlier. I somehow managed to compile the driver ont he newer kernel by using a suse .rpm ;) |
| 00:18.32 | Cheetah | err wrong channel :D |
| 00:18.37 | Cheetah | supposed to go to #debian |
| 00:19.10 | Cheetah | thanks for YOUR help, too, of course :D |
| 00:21.21 | JT | flenders: the optus office is really good |
| 00:21.29 | JT | if i'm reading my notes right |
| 00:21.37 | JT | if they "changeover" your lines from telstra to their pri |
| 00:21.42 | JT | it's $1200 install |
| 00:21.54 | JT | but they put that cost as a credit on your account |
| 00:22.33 | JT | way to make them lose money, use the lines for incoming calls only ;) |
| 00:22.41 | *** join/#asterisk station49 (n=station4@host-24-225-204-251.patmedia.net) |
| 00:22.53 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 00:22.59 | JT | s/office/offer/ |
| 00:24.02 | station49 | can someone tell me if I am able to register with a SIP server via something like X-LIte, shoudl i be able to via Asterisk in sip.conf using the register => ..... |
| 00:25.24 | flenders | JT: fuck! that's a pretty decent offer |
| 00:26.49 | JT | flenders: yeah they're crazy |
| 00:27.09 | *** join/#asterisk ttuttle (n=tom@gentoo/contributor/ttuttle) |
| 00:28.05 | ttuttle | I added the client in sip.conf, but it (Ekiga) says "Registration failed". |
| 00:28.10 | JT | he should use the dial command to call the sip entry for the soft phone |
| 00:28.14 | JT | hmm |
| 00:28.38 | ttuttle | JT: exten => 617xxxxxxx,1,Answer then exten => 6178808012,n,Dial(SIP/hostname)? |
| 00:29.00 | JT | if 'hostname' is what is in square brackets in sip.conf |
| 00:29.09 | ttuttle | JT: It is. |
| 00:29.16 | JT | is it type=friend |
| 00:29.24 | ttuttle | Yes. |
| 00:29.35 | ttuttle | type=friend, host=ip username=user, secret=pass, context=from-sip. |
| 00:29.42 | *** join/#asterisk arctic_import (n=jasonj@mail.uui-alaska.com) |
| 00:30.00 | arctic_import | Can someone help me get a PRI (TE110P) working? |
| 00:30.08 | arctic_import | I have the T1/PRI wired up to the local telco but they are telling me they get nothing on the D channel or something? How can I debug this? |
| 00:30.31 | JT | arctic_import: have you configured zaptel.conf and zapata.conf? |
| 00:30.43 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 00:30.53 | arctic_import | my /etc/zapata.conf shows span=1,1,0,esf,b8zs |
| 00:31.09 | JT | that bit is right |
| 00:31.15 | JT | there is more to it than that |
| 00:31.18 | JT | ~pb |
| 00:31.28 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 00:31.28 | ttuttle | JT: /me wonders if there is a better (less user-friendly, more clear about the terms) softphone to use than Ekiga? |
| 00:31.46 | JT | ttuttle: not sure |
| 00:32.10 | arctic_import | JT, also using bchan=1-8 and dchan=24 |
| 00:33.15 | *** join/#asterisk adker (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net) |
| 00:33.33 | JT | arctic_import: sorry, please pastebin.ca the whole of zaptel.conf AND zapata.conf or we are unable to assist |
| 00:33.43 | arctic_import | JT, in the /etc/asterisk/zaptel.conf I have : switchtype=national, signalling=pri_cpe, group=1, channel => 1-8 |
| 00:33.48 | arctic_import | JT, okay. |
| 00:34.06 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 00:35.01 | arctic_import | JT, http://pastebin.ca/344780 |
| 00:35.39 | ttuttle | JT: Do I need a separate daemon to do SIP registration? |
| 00:35.49 | JT | arctic_import: 8ch fractional t1? |
| 00:36.18 | arctic_import | JT, yes its a PRI with 8 B channels, and 1 D channel |
| 00:36.49 | JT | argg, AMP |
| 00:36.59 | JT | is there anything in zapata_additional? |
| 00:37.11 | ttuttle | Anyone? I'm basically trying to get Asterisk to act as a SIP server. |
| 00:37.23 | arctic_import | JT, no its empty |
| 00:38.02 | JT | arctic_import: seems ok, as long as your telco is has a national switchtype |
| 00:38.23 | JT | arctic_import: next thing to try is a pri intense debug in the console |
| 00:38.47 | arctic_import | JT, yes the telco is using national as well. We even tried 5ess but we still couldn't get it working. |
| 00:39.32 | arctic_import | JT I've done this but I don't know what it means. |
| 00:40.42 | JT | you may have to paste some into pastebin then |
| 00:41.01 | *** join/#asterisk doolph (i=doolph@200.46.148.43) |
| 00:41.03 | arctic_import | JT, http://pastebin.ca/344787 |
| 00:41.17 | doolph | anyone know how to determine the callprogress / indications.conf for my country? |
| 00:41.47 | JT | arctic_import: yep so it's clearly receiving nothing back from the line |
| 00:42.04 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 00:42.05 | JT | which may mean your cabling is bad or any number of things |
| 00:42.22 | JT | doolph: it's not in there already? |
| 00:42.30 | doolph | nop :( |
| 00:42.39 | JT | what country? |
| 00:42.43 | doolph | panama |
| 00:43.00 | JT | hmm, check if it's in zonedata.c by any weird chance |
| 00:43.14 | JT | in the zaptel sources |
| 00:43.38 | JT | if not, you may have to try and see if your country has any standards documents published on it |
| 00:44.06 | doolph | they are Cable&Wireless |
| 00:44.21 | JT | also, you could record them, and use audio software with spectrum analyser capability to see the frequencies of the tone |
| 00:44.36 | arctic_import | JT, so when I use zttool it tells me its up. but maybe that doesn't mean anything? The telco can see the span go down when I unload the modules. and it comes back when I reload the modules. Would this still happen with a bad cable? |
| 00:44.57 | doolph | umm |
| 00:45.18 | JT | maybe they are getting sync from you, but you aren't receiving theirs? |
| 00:45.24 | JT | did you make the cable |
| 00:46.33 | arctic_import | JT, no its a pre-made cat5 cable. Straight thru. |
| 00:46.36 | doolph | JT have you done this before? |
| 00:46.40 | ttuttle | How do I configure Asterisk to act as a SIP server for softphones on my LAN? I tried it but my softphone is getting "Registration failed". |
| 00:46.46 | anonymouz666 | exten => 123,1,Set(COUNT=1) |
| 00:46.46 | anonymouz666 | exten => 123,2,While($[ ${COUNT} < 5 ]) |
| 00:46.54 | doolph | i mean record, audio software... etc |
| 00:46.55 | anonymouz666 | set count 1 will increase the count? |
| 00:46.58 | JT | doolph: analysed tones? not really, the ones i needed already exist |
| 00:47.07 | JT | doolph: but i've done similar stuff in audio software |
| 00:47.22 | doolph | JT so you can use callprogress=yes without any problem |
| 00:47.24 | JT | anonymouz666: err, that will just set it to 1 |
| 00:47.34 | ttuttle | Anyone, please? |
| 00:47.38 | JT | doolph: i guess so |
| 00:47.53 | doolph | ttuttle put nat=yes |
| 00:48.02 | ttuttle | doolph: In the softphone entry in sip.conf? |
| 00:48.15 | doolph | in any sip user context |
| 00:48.23 | JT | arctic_import: hrm ok |
| 00:48.39 | JT | arctic_import: is the jumper set to T1? |
| 00:48.51 | arctic_import | JT, whats the default |
| 00:48.57 | ttuttle | doolph: Hmm. |
| 00:49.02 | JT | probably t1, but i'd never assume |
| 00:49.06 | ttuttle | doolph: Then what? |
| 00:49.27 | ttuttle | doolph: But there's no NAT between Asterisk and the phone. |
| 00:49.33 | arctic_import | JT, I didnt' change the jumper from default so I can't say for sure that its set to T1 |
| 00:49.46 | doolph | ttuttle no? |
| 00:49.55 | doolph | what softphone are you using |
| 00:49.57 | JT | something you should've checked before putting the card in ;) |
| 00:50.24 | JT | arctic_import: so the card never goes up in asterisk? |
| 00:50.36 | ttuttle | doolph: It's Ekiga. |
| 00:50.41 | ttuttle | doolph: They're on the same LAN. |
| 00:51.12 | doolph | ummm try another softphone first |
| 00:51.30 | doolph | like SJphone |
| 00:51.33 | ttuttle | doolph: ok |
| 00:52.12 | ttuttle | doolph: installing... |
| 00:52.25 | doolph | cool |
| 00:53.03 | ttuttle | doolph: Where do I enter the server name? |
| 00:53.24 | doolph | options, profiles |
| 00:53.43 | ttuttle | doolph: Hmm... |
| 00:53.47 | *** join/#asterisk lba (n=lba@user-12lml5g.cable.mindspring.com) |
| 00:53.49 | *** part/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:54.01 | ttuttle | doolph: General doesn't have it, Advanced doesn't have it, DTMF doesn't have it, is it in STUN? |
| 00:54.22 | JT | no |
| 00:54.36 | ttuttle | Oh. |
| 00:54.37 | doolph | where are you |
| 00:54.41 | ttuttle | doolph: Where? |
| 00:54.48 | ttuttle | doolph: On my bed, looking at my laptop ;-) |
| 00:54.54 | thx2000 | obaby |
| 00:54.57 | doolph | i said options, then profiles, then new |
| 00:55.01 | ttuttle | doolph: yeah |
| 00:55.13 | doolph | ok, dont ask any other stupid questions |
| 00:55.25 | ttuttle | doolph: /me gets it. |
| 00:55.29 | lba | Please help parse this to show result of ChanIsAvail: exten => _4XX,n,VERBOSE("Status: is ${ChanIsAvail(EXTEN)}" 1) |
| 00:55.31 | doolph | cool lol |
| 00:55.39 | arctic_import | JT, well I don't really know how to check the status in asterisk. |
| 00:55.42 | ttuttle | doolph: So it says "squirrel: Service unavailable" (squirrel is the serveR). |
| 00:56.07 | doolph | then your server is not available ? |
| 00:56.14 | ttuttle | doolph: But it's running. |
| 00:56.23 | doolph | can you connect to it? |
| 00:56.28 | ttuttle | doolph: Okay, I'm getting "Username/auth name mismatch" from Asterisk. Just a sec. |
| 00:56.38 | ttuttle | doolph: Yes, but I get an error when I try to log in. |
| 00:56.48 | arctic_import | JT if I do a pri show span 1, it always reports Status: Provisioned, Down, Active |
| 00:56.51 | doolph | maybe you havent reload the sip config |
| 00:57.36 | ttuttle | just a sec. |
| 00:58.01 | JT | lba: ${EXTEN} not (EXTEN) |
| 00:58.20 | lba | JT: Thanks. I'll try that. |
| 00:58.39 | *** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) |
| 00:58.47 | JT | arctic_import: hmm so it's down |
| 00:59.09 | ttuttle | doolph: Nah, I reloaded the SIP config. |
| 00:59.36 | doolph | give me user/pass/host |
| 00:59.38 | doolph | to test it here |
| 01:00.04 | ttuttle | doolph: It's not on a public IP. |
| 01:00.10 | lba | JT: Better coz it dosn't report an error but the only result is: "Status: is " 1 |
| 01:00.13 | doolph | umm |
| 01:00.14 | JT | doolph: what makes you think it's even Internet accessible? |
| 01:00.15 | JT | :) |
| 01:00.22 | doolph | lol |
| 01:00.47 | lba | JT: I need to know the status == return code of ChanIsAvail |
| 01:01.02 | JT | not familiar with it |
| 01:01.20 | doolph | JT dude, any idea to dont make TDM answer all my calls even when its ringing? i cannot have a correct billing in this way |
| 01:01.43 | doolph | this is sad |
| 01:02.01 | JT | lba: did you do ${ChanIsAvail(${EXTEN})} ? |
| 01:02.24 | lba | JT: Was "not familiar with it" addressed to me or doolph? |
| 01:02.28 | JT | doolph: huh? |
| 01:02.31 | JT | the variable |
| 01:02.36 | JT | sounds like a function though |
| 01:02.37 | joe | Is anyone here doing central provisioning w/ polycom phones who has there own local-settings-sip.cfg file so as to not touch the distributed sip.cfg? |
| 01:02.41 | lba | JT: Do spaces matter? |
| 01:02.44 | *** join/#asterisk ttuttle (n=tom@gentoo/contributor/ttuttle) |
| 01:02.58 | JT | so if it is, you will need brackets around the EXTEN var |
| 01:03.05 | ttuttle | Sorry about that, IPv6 went down. |
| 01:03.14 | lba | JT: I'll try again. brb |
| 01:04.13 | ttuttle | doolph: What now? |
| 01:04.31 | doolph | ttuttle is it still available or its password problem |
| 01:04.46 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 01:05.00 | ttuttle | doolph: I'm just getting that error message -- username/auth name mismatch. |
| 01:05.25 | mosty | why does $DIALSTATUS rely on the peer being qualified? |
| 01:06.21 | lba | JT: Your line give WARNING No application '${ChanIsAvail' for extension (default, 430, 3) |
| 01:06.30 | doolph | ttuttle where |
| 01:06.33 | JT | ttuttle: to be honest, it's very pointless us trying to debug your problem until you paste sip.conf into pastebin.ca after removing the actual password(s) |
| 01:06.36 | ttuttle | doolph: From Asterisk. |
| 01:06.40 | ttuttle | JT: Ok. |
| 01:06.59 | doolph | ttuttle i think its your sip config |
| 01:07.04 | doolph | JT |
| 01:07.05 | doolph | <PROTECTED> |
| 01:07.05 | doolph | <PROTECTED> |
| 01:07.27 | doolph | the zap line was still ringing, not really answered |
| 01:08.33 | JT | analogue? |
| 01:08.42 | doolph | tdm400 yes |
| 01:08.50 | JT | that is normal operation |
| 01:08.55 | JT | analogue has crap signalling |
| 01:08.56 | doolph | arghh |
| 01:09.03 | doolph | really? |
| 01:09.04 | JT | hard for a computer to detect events |
| 01:09.28 | doolph | if I use T1 then it can detects it correctly |
| 01:09.45 | JT | yeah you need answer supervision if zaptel even supports it, and maybe it can try and detect it but it may not be reliable |
| 01:10.05 | JT | of course, each stage of call progress is a Q.931 data signalling message sent over the D channel |
| 01:10.11 | JT | very easy for a computer to understand |
| 01:10.24 | ttuttle | Here: http://pastebin.ca/344818 |
| 01:10.29 | doolph | u mean the T1 |
| 01:10.33 | JT | yes |
| 01:10.39 | CJLinst | Is there a quick way to run commands like ChanIsAvail(SIP/112) from the CLI and look at the result? |
| 01:10.52 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:10.52 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:10.53 | JT | lba: you are doing things totally wron |
| 01:10.55 | JT | wrong |
| 01:10.59 | JT | lba: i checked the docs |
| 01:11.01 | Qwell | Bhaal: must you keep doing that? ;/ |
| 01:11.18 | ttuttle | doolph: Can you look at my sip.conf? |
| 01:11.19 | JT | chanisavail is an app, not a variable or function |
| 01:11.27 | JT | Qwell: he does that to check for onjoin spammers |
| 01:11.49 | Qwell | yeah, an onjoin spammer isn't gonna grep -v "freenode/staff" ... |
| 01:12.22 | Qwell | him leaving and joining every couple hours is FAR more spammy |
| 01:12.43 | ttuttle | Qwell: Every couple hours? That's not spammy. |
| 01:12.50 | doolph | analog line is very stupid |
| 01:12.51 | ttuttle | JT: Can you take a look at my sip.conf? |
| 01:12.58 | Qwell | ttuttle: it is when you have 1 spammer every month or so |
| 01:13.05 | ttuttle | Qwell: ah |
| 01:13.12 | Qwell | 120 times/mo > 1 time/mo |
| 01:13.23 | doolph | omg |
| 01:13.31 | doolph | i invested more than $400 |
| 01:13.37 | doolph | and this is not working like I want |
| 01:13.46 | ttuttle | doolph: What are you doing? |
| 01:13.58 | arctic_import | JT: okay here some more info. When I load the modules. and cat /proc/zaptel/* I show the channels and they are all "CLEAR" after I start asterisk this changes to "In use" |
| 01:14.04 | lba | JT: I'm also looking at the docs at voip-info and can't figure how to parse ChanIsAvail. |
| 01:14.06 | arctic_import | JT: Is that normal? |
| 01:14.06 | doolph | just trying to have log every call |
| 01:14.09 | JT | doolph: you could always use an itsp |
| 01:14.27 | doolph | an what |
| 01:14.35 | lba | JT: What I want to do is dial an internal exten and get a Congested if the exten doesn't exist, else call the exten |
| 01:14.42 | JT | lba: show application chanisavail |
| 01:15.04 | JT | chanisavail is an app, that returns vars |
| 01:15.21 | JT | arctic_import: i believe so |
| 01:15.32 | JT | doolph: internet telephony service provider |
| 01:15.38 | lba | JT: I'll examine and play with that info. |
| 01:15.40 | JT | sip has sufficient signalling |
| 01:15.41 | lba | JT: Thanks |
| 01:17.37 | doolph | JT ah, yes but I have a telular |
| 01:18.01 | JT | doolph: what does that mean? |
| 01:18.33 | lba | JT: I tried: exten => _4XX,n,ChanIsAvail(${EXTEN}) exten => _4XX,n,VERBOSE("${AVAILSTATUS}" 1) |
| 01:18.35 | doolph | telular=a gsm to pstn |
| 01:18.43 | doolph | its to make mobile calls |
| 01:18.54 | JT | you mean a gsm to pots gateway? |
| 01:19.04 | lba | JT: Result: ChanIsAvail argument takes format ([technology]/[device]) |
| 01:19.23 | JT | lba: you are missing sip/zap/etc |
| 01:19.43 | lba | JT: device is just 'SIP' No number? |
| 01:19.49 | doolph | its a equipment that convert mobile network to any standard line fxs |
| 01:20.07 | JT | doolph: yes, a gsm to pots gateway |
| 01:20.15 | doolph | yes that thing |
| 01:20.30 | doolph | i need to do what you said |
| 01:20.36 | JT | lba: err, exactly *What* are you try to check, what channel? |
| 01:20.37 | doolph | record the line |
| 01:20.50 | doolph | then do the indications.conf strings |
| 01:21.04 | JT | ok |
| 01:21.51 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 01:22.27 | lba | JT: I am trying to check if an sip extension is not an actual channel in sip.conf and Congested if so. |
| 01:22.44 | lba | JT: This is to prevent mis-dials which cause * to go wild. |
| 01:23.13 | lba | JT: I check 430 which does not exist, then 413 which is right next to me. |
| 01:23.18 | JT | lba: still have no idea what you're trying to do |
| 01:23.26 | JT | the concept of channels doesn't quite align to sip |
| 01:23.33 | ttuttle | Where do I put the username and password that I want to have to enter on a SIP softphone that will connect to my Asterisk server? |
| 01:23.43 | JT | as you can establish sip connections at will |
| 01:24.16 | lba | JT: OK. _4XX numbers dial sip phones thruout the house. But sometimes wife dials a non-existing number. Asterisk goes crazy and I want to detect this. |
| 01:24.38 | JT | ttuttle: you must set host=dynamic |
| 01:24.46 | JT | otherwise registrations are pointless |
| 01:25.03 | lba | JT: And run Congestion. host _is_ dynamic in sip.conf |
| 01:25.05 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 01:25.16 | ttuttle | JT: Ok. |
| 01:25.22 | JT | i sait ttuttle not lba |
| 01:25.30 | ttuttle | JT: Can I just not do registration? |
| 01:25.31 | lba | JT: Sorry |
| 01:25.37 | ttuttle | JT: IT's a static IP on my LAN. |
| 01:25.54 | JT | ttuttle: you either set the host statically, like you are now, and stop the client from registering, or you register |
| 01:26.07 | ttuttle | JT: Okay, I've set it to host=dynamic, and it's still getting an error logging in. |
| 01:26.25 | ttuttle | JT: Asterisk says "Feb 7 20:26:10 NOTICE[7508]: chan_sip.c:11131 handle_request_register: Registration from '<sip:tom@10.42.0.3:5060>' failed for '10.42.0.8' - Username/auth name mismatch". |
| 01:26.34 | JT | lba: umm, it sounds like a very esoteric setup, why don't you have each extension in your house setup with their own entries in extensions.conf and sip.conf? |
| 01:26.49 | JT | ttuttle: did you do a sip reload? |
| 01:26.58 | ttuttle | JT: Yes, I've restarted Asterisk itself several times. |
| 01:27.34 | JT | ttuttle: i'd set the sip entry name to conf just to test |
| 01:27.41 | JT | sometimes it can be anal about it |
| 01:27.42 | lba | JT: Each extension _is_ setup in sip.conf. I use _4XX because there are 15 sip phones and it's cumbersome to have so many separate extens |
| 01:27.45 | ttuttle | JT: Instead of lion? |
| 01:27.55 | JT | ttuttle: yes |
| 01:28.00 | ttuttle | JT: "conf"? |
| 01:28.12 | JT | ttuttle: if that fails, try adding insecure=very |
| 01:28.13 | doolph | ttuttle why you are setting it as tom |
| 01:28.16 | JT | sip.conf, where else? |
| 01:28.21 | doolph | put a number |
| 01:28.24 | ttuttle | doolph: Wait, the username? |
| 01:28.26 | doolph | yes |
| 01:28.31 | JT | err, names work fine, doolph |
| 01:28.36 | ttuttle | doolph: Because I figured I would be logging in with a username. |
| 01:28.41 | doolph | umm |
| 01:28.43 | ttuttle | doolph: My upstream provider, for example, uses a username. |
| 01:28.46 | mosty | JT: why cant you have registrations AND host=<some fixed address> ? i would still like to be able to see if the peer is contactable even if it is static |
| 01:28.58 | doolph | just try it before |
| 01:29.31 | JT | mosty: arrgh, it specifically states they're mutally exclusive in the default documentation like the sample sip.conf, registration is ONLY to know what dynamic ip a host is on |
| 01:29.44 | wunderkin | mosty, that is what qualify is for |
| 01:29.46 | ttuttle | doolph: Still doesn't work. |
| 01:29.51 | JT | mosty: qualify finds if hosts are up |
| 01:29.54 | wunderkin | lba, if each valid exten has a voicemail box, use mailboxexists instead |
| 01:30.18 | ttuttle | JT: Okay, it's still not working, it says "Username/auth name mismatch". Should I set authuser in the [lion] setting too? |
| 01:30.29 | lba | wunderkin: They don't. We use a single voicemail box for the whole house (residence) |
| 01:30.34 | JT | yes, give it a go |
| 01:30.58 | JT | lba: how does your system dial? |
| 01:31.25 | lba | JT: Not sure I understand. Dialing is done with Dial() Is that what you asked? |
| 01:31.26 | mosty | jt/wunderkin: perhaps i am confused, i thought i tried that, and qualify wasn't happy if it was a static host |
| 01:31.47 | wunderkin | i thought you could, dunno |
| 01:31.53 | JT | lba: yeah i'd like to see the dial line if possible, i'm not sure i understand your setup |
| 01:31.58 | JT | mosty: that's not so |
| 01:32.03 | ttuttle | JT: Still same problem. |
| 01:32.19 | JT | ttuttle: look at the actuall error using sip debug |
| 01:32.44 | wunderkin | lba, chanisavail is yucky for this, what do you mean that asterisk goes weird if they dial an invalid exten? |
| 01:33.21 | lba | JT: exten => _4XX,n,Macro(stdexten,${EXTEN},SIP/${EXTEN},{20,rt}) |
| 01:33.35 | JT | i think he is using some sort of crackpot method of dialling his extensions |
| 01:33.37 | JT | arrgh |
| 01:33.39 | JT | yep |
| 01:33.41 | lba | wunderkin: It continually issues messages in the CLI. Each 2 lines each. |
| 01:33.59 | wunderkin | use the real macro-stdexten |
| 01:34.25 | ttuttle | JT: I'll post the errors to pastebin.ca. |
| 01:34.27 | lba | wunderkin: This is not the real macro-stdexten? |
| 01:34.30 | JT | lba: ok, if you want to continue using that crackpot method, there are easier methods :P |
| 01:34.40 | JT | lba: are all the extensions in a continuous block |
| 01:34.46 | JT | with no gaps in numbering? |
| 01:34.58 | *** join/#asterisk orkid (n=orkid@dataq2.utias.utoronto.ca) |
| 01:35.11 | lba | JT: Not in a continuous block. I'm still installing phones. |
| 01:35.16 | JT | hrm |
| 01:35.23 | JT | because you could use a better pattern |
| 01:35.27 | JT | instead of stuffing around |
| 01:35.28 | ttuttle | JT: http://pastebin.ca/344841 |
| 01:35.57 | JT | _4[0-2][0-9],n......... etc |
| 01:36.05 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 01:36.07 | JT | or you could specify them all |
| 01:36.39 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
| 01:37.09 | lba | JT: I'll think about it. I really like the present numbering system. How would your method prevent misdials? |
| 01:37.26 | JT | well |
| 01:37.38 | wunderkin | your problem is just with the macro or extensions.conf |
| 01:37.42 | wunderkin | looping |
| 01:38.14 | JT | say the pattern is _4[0-2][0-3], that'd only match an extension between 400 and 423 |
| 01:38.17 | wunderkin | show the complete macro, all of 4xx, and the cli output |
| 01:38.58 | ttuttle | JT: Can you take a look at that pastebin? |
| 01:38.58 | JT | wunderkin: the problem is also that his pattern is broad and catches extensions that do not exist |
| 01:38.58 | doolph | omg |
| 01:38.59 | wunderkin | no... the 2nd number could only be 0,1,2 and the 3rd number 0,1,2,3 |
| 01:39.09 | lba | wunderkin: As far as I can tell it's the stock * stdexten macro |
| 01:39.25 | wunderkin | that doesn't matter, he could just play a congestion or something, it should not be looping |
| 01:39.41 | wunderkin | lba, nope |
| 01:40.17 | JT | wunderkin: umm, yes, which is the same as saying "the range between 400 and 423" |
| 01:40.17 | wunderkin | nope |
| 01:40.20 | JT | why not? |
| 01:40.32 | JT | what are you on about? |
| 01:40.51 | ttuttle | JT: Me? |
| 01:40.52 | wunderkin | [0-2] means 0,1,2 on the 2nd digit, [0-3] means 0,1,2,3 on the 3rd digit, not a range as a whole |
| 01:41.07 | wunderkin | the whole range would be 400-423 |
| 01:41.15 | JT | i know how the patten works wunderkin |
| 01:41.16 | JT | err |
| 01:41.20 | JT | what's your point |
| 01:41.28 | JT | so it's between 400 and 423 effectively |
| 01:42.06 | wunderkin | oh, i mean that would not match 419 or 418, etc |
| 01:42.18 | wunderkin | the 3rd number is not 0,1,2,3 |
| 01:42.28 | JT | ah true |
| 01:42.30 | JT | my mistake |
| 01:43.57 | lba | wunderkin: I don't understand why you say 'nope' about my stdexten macro being the standard * one. |
| 01:44.44 | wunderkin | [macro-stdexten]; ; Standard extension macro: ; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well ; ${ARG2} - Device(s) to ring |
| 01:45.16 | lba | wunderkin: That's exactly what I have in extensions.conf |
| 01:45.27 | wunderkin | you're not calling it like that |
| 01:45.33 | wunderkin | oh |
| 01:46.05 | wunderkin | i guess, with extra cruft |
| 01:46.13 | wunderkin | but anyways, pastebin |
| 01:47.17 | ttuttle | Hmm. |
| 01:47.38 | ttuttle | How do I specify the authentication that Asterisk should require of a softphone? |
| 01:49.27 | doolph | ttuttle why don't you try freepbx or something |
| 01:49.43 | doolph | or just install asterisk 1.4 web |
| 01:51.02 | arctic_import | Well I officially have no Idea what to try next on this stupid PRI. I've verified the Jumper is set to T1, I've changed cables. I cannot get the darn thing to work. |
| 01:51.27 | lba | JT: wunderkin general pastebin - Miscellany - post number 344861 |
| 01:51.53 | wunderkin | arctic_import, well if it says provisioned, down, active, um its turned down on the other side, i dont know if there are any other causes of that, from your side |
| 01:51.58 | ttuttle | doolph: Never mind, I fixed it. |
| 01:52.02 | wunderkin | lba, whole url |
| 01:52.34 | lba | http://pastebin.ca/344861 |
| 01:52.45 | arctic_import | wunderkin, yeah except the Telco is blaming my equipment. |
| 01:52.53 | wunderkin | everything else? |
| 01:52.59 | wunderkin | ^ lba |
| 01:53.22 | Lurchtoke | shit |
| 01:53.22 | wunderkin | arctic_import, you probably need to talk to someone that is not lazy and high |
| 01:53.37 | Lurchtoke | 8080 is the default port for remote gui access? |
| 01:53.41 | arctic_import | wunderkin, haha ya maybe. |
| 01:54.36 | lba | wunderkin: JT This is the original DP to dial any house extension. I want to modify it to detect and Congestion on bad exten numbers. |
| 01:54.58 | wunderkin | show the complete macro, all of 4xx, and the cli output <-- |
| 01:55.12 | lba | wunderkin: JT My extensions for in-house dialing are between 400 and 499 |
| 01:55.20 | lba | wunderkin: The std-exten macro? |
| 01:55.22 | Bobthehunter | is there such a thing as strstr for dialplans ? |
| 01:55.31 | wunderkin | lba, yes, [macro-stdexten] |
| 01:55.35 | Bobthehunter | i need oh! |
| 01:55.38 | Bobthehunter | chantype =zxp |
| 01:55.40 | Bobthehunter | zap |
| 01:56.06 | wunderkin | what bob |
| 01:56.39 | Bobthehunter | GotoIf($[${ChannelType} = Zap]?10) |
| 01:56.44 | Bobthehunter | that wat im looking for lol |
| 01:56.59 | Bobthehunter | i assume its not case sensitive |
| 01:57.26 | Bobthehunter | but can i goto a macro ? |
| 01:57.35 | Bobthehunter | GotoIf($[${ChannelType} = Zap]?nameofmacro1) |
| 01:58.07 | lba | wunderkin: http://pastebin.ca/344868 |
| 01:58.30 | wunderkin | Bobthehunter, well, i guess you can use ${CHANNEL}, check the doc directory for the variable stuff |
| 01:58.39 | Bobthehunter | kk |
| 01:58.49 | lba | wunderkin: I got this from someone elses DP but thought it was the standard * std-exten |
| 01:58.58 | wunderkin | ... |
| 01:59.08 | wunderkin | packet corruption |
| 02:06.58 | *** part/#asterisk hkdaylxb (n=chatzill@144.214.37.27) |
| 02:07.07 | *** join/#asterisk hkdaylxb (n=chatzill@144.214.37.27) |
| 02:07.33 | lba | wunderkin: I paste binned my macro-stdexten at http://pastebin.ca/344868 |
| 02:07.43 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 02:07.48 | wunderkin | you did not |
| 02:08.38 | lba | wunderkin: Maybe I did something wrong. Will pastebin again ... |
| 02:09.07 | wunderkin | that or there is interference in one of your internet tubes |
| 02:09.52 | lba | wunderkin: http://pastebin.ca/344879 |
| 02:09.54 | Bobthehunter | actually i dont know channel type yet.. since call not trough.. |
| 02:10.05 | Bobthehunter | im trying to see if zap avail.. already got a checkpri macro |
| 02:10.37 | Bobthehunter | so i need a REGEX in the strgin ill ddial.. and i got that in ${ARG2} |
| 02:10.52 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:10.52 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:12.48 | wunderkin | set to every hour now? |
| 02:13.19 | wunderkin | well im out of energy and patience at the moment |
| 02:13.36 | wunderkin | lba, use the stock macro-stdexten and go from there, get rid of the call forward crap |
| 02:14.34 | wunderkin | looks to me like it will keep looping |
| 02:15.31 | wunderkin | Bobthehunter, you can try one more time in english, otherwise im out for awhile |
| 02:15.37 | Bobthehunter | ouch |
| 02:15.58 | Bobthehunter | im trying to regex the channel to see if ZAP in the name so i can call my macro checkpriavail |
| 02:16.28 | Bobthehunter | sorry.. im trying to REGEX my newdialstring.. to see if ZAP in it.. |
| 02:16.48 | wunderkin | uh yeah thats what i said |
| 02:16.53 | Bobthehunter | my dial strign is FOO and containx either SIP/BLAH/${EXTEN} or ZAP/g1/${EXTEN} |
| 02:17.10 | wunderkin | well you just said zap before |
| 02:17.21 | Bobthehunter | yeah sorry lol long day here too |
| 02:17.44 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 02:17.50 | lba | wunderkin: I didn't realize my macro-stdexten was not stock. I'll look for the stock one. However, my problem has not been with the std-exten except for the looping was a bad exten is dialed. That's why I'd like to check the status |
| 02:17.56 | wunderkin | well if ${CHANNEL} does not work, then i don't know what you're doing so i don't really feel like it now, sorry |
| 02:18.49 | wunderkin | lba, well.. if you are using stock asterisk, it is in /usr/src/asterisk/configs/extensions.conf.sample |
| 02:19.17 | *** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net) |
| 02:21.29 | k-man_ | how can i make my phone and asterisk make sounds that are more like the australian signals? |
| 02:21.31 | teknoprep | hey anyone here use voicepulse ? |
| 02:22.11 | JT | k-man_: in asterisk you set the tonezone to au |
| 02:22.17 | teknoprep | been having choppyness problems until i moved to connect03.voicepulse.com iax2 trunk |
| 02:22.18 | JT | in sip phones, you'll have to set it there too |
| 02:22.29 | k-man_ | jt, oh.... |
| 02:22.32 | k-man_ | ok |
| 02:22.46 | k-man_ | jt, where is it in asterisk? |
| 02:23.16 | JT | if there's any zap channels, you set it in zapata.conf |
| 02:23.32 | JT | indications.conf for default asterisk indications |
| 02:23.41 | JT | you can also set it with a var per call iirc |
| 02:23.52 | k-man_ | oh |
| 02:24.08 | k-man_ | but i only have a sip phone so its a setting in the phone? |
| 02:24.24 | JT | there are two tone sources in asterisk, zaptel ones, for zap channels, and asterisk ones, for Congestion() and playtones, etc |
| 02:24.36 | JT | well the sip phone makes the noise, so yes |
| 02:24.40 | JT | at least the dialtone |
| 02:25.34 | k-man_ | in regional settings, there is a bunch of tones that can be set |
| 02:25.40 | k-man_ | cryptic strings |
| 02:27.10 | lba | wunderkin: OK I've copied macro-stdexten into my extensions.conf but I'll have to carefully examine it. |
| 02:27.35 | lba | wunderkin: It doesn't seem to allow options in the dial command like rtT |
| 02:28.16 | lba | wunderkin: I appreciate your help. Unfortunately, I have been called for dinner right now and this is last night before my wife flys off early tomorrow. |
| 02:28.26 | lba | wunderkin: Please excuse me. |
| 02:30.17 | lba | wunderkin: And thank you very much for showing me that my macro-stdexten wasn't stock like I thought. |
| 02:31.12 | lba | wunderkin: and JT Thanks to both of you. Bye |
| 02:37.45 | flenders | how do I get a dialtone after dialing an extension? |
| 02:40.34 | [TK]D-Fender | flenders: To do what? What happens between your initial dial and your 2nd dialtone? Are you expecting this to be an * provided tone? |
| 02:41.05 | flenders | I can dial out |
| 02:41.08 | JT | DISA |
| 02:42.01 | flenders | JT: http://www.voip-info.org/wiki-Asterisk+cmd+DISA ? |
| 02:43.29 | doolph | anyone got a good qos solution for linux router? |
| 02:43.43 | doolph | i got a network that is mixing p2p & voip |
| 02:44.23 | Qwell | qos solution: Don't do that |
| 02:44.32 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 02:45.07 | doolph | what |
| 02:47.16 | JT | flenders: i'd say that'd be it |
| 02:47.24 | JT | what other disa would there be in asterisk |
| 02:47.33 | flenders | JT: just tested, works great! thanks mate |
| 02:47.44 | JT | yeah it's a useful command |
| 02:50.18 | *** join/#asterisk angler_ (i=angler@nat/digium/x-5495ada1a5dab369) |
| 02:56.25 | *** join/#asterisk soylent (n=soylent@38.99.80.85) |
| 02:58.23 | k-man_ | hey, can you use asterisk to do voice directed ivr? |
| 02:58.27 | k-man_ | like telstra has here? |
| 02:58.40 | soylent | uhg |
| 02:58.51 | [TK]D-Fender | k-man_: lookup Sphinx on the WIKI |
| 02:59.07 | soylent | anyone familiar with setting up asterisk to connect to a cisco router holding the T1 line? |
| 03:00.42 | soylent | guess not? |
| 03:00.47 | soylent | :-) |
| 03:01.19 | *** join/#asterisk xpot (n=xpot@dsl093-228-250.slc1.dsl.speakeasy.net) |
| 03:01.34 | k-man_ | thanks |
| 03:01.35 | soylent | for some reason, I'm not allowed to get a digium or equivalent card and connect the T1 to asterisk directly. I'm only allowed (business decision, not mine) to use the Cisco. |
| 03:02.07 | soylent | so just curious if people use that setup much or if that is "offensive" in this realm. |
| 03:02.10 | soylent | hehe |
| 03:02.20 | xpot | anyone know the proper use of calleridnum in 1.4 using agi? Here is what I currently have> my $callerid = $input{'calleridnum'}; |
| 03:05.18 | [TK]D-Fender | soylent: I've heard of a number of people in here that have done this, and it is not offensive |
| 03:05.30 | soylent | whew |
| 03:05.33 | soylent | :-) |
| 03:05.37 | k-man_ | anyone know of nodephone have a callback service for testing purposes? |
| 03:05.47 | k-man_ | so i can test DID |
| 03:06.30 | soylent | so Fender, maybe this question has nothing to do with Cisco though. My problem, through cisco, is how do I handle early media, if that's a term used here. |
| 03:08.02 | soylent | right now the Cisco is sending me a 100 message and then nothing until the call gets dropped. |
| 03:10.43 | jpablo | soylent, i once almost lost all my hair with a cisco passing me sip calls. finally i found that the stupid cisco expects that you send them a ringing message before anything |
| 03:10.52 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:10.52 | jpablo | so start with a exten bla,1,ringing |
| 03:10.52 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:11.08 | [TK]D-Fender | soylent: You can't really. SIP has no means or representing it IIRC |
| 03:11.50 | soylent | so busy signals and such? won't get passed back? or fast busies or just "sorry, number not in use any more" messages? |
| 03:12.27 | soylent | does that work ok with the T1 attached to the asterisk box directly? |
| 03:12.41 | soylent | cause so far, none of that seems to be getting through. |
| 03:12.56 | soylent | or maybe it's a config issue on the cisco? |
| 03:13.20 | soylent | sorry to be so noisy. just been pulling my hair out as you say... |
| 03:14.02 | jpablo | soylent, thoses messages dont work for me in a ISDN E1 connected to a digium e1 card. |
| 03:14.13 | soylent | :-( |
| 03:14.13 | jpablo | soylent, other people say the messages work for them |
| 03:14.28 | soylent | anyone know for sure?? |
| 03:14.48 | [TK]D-Fender | soylent: * will have no way to TELL the PRI to send early media |
| 03:14.52 | jpablo | soylent, they work when i connect 24 fxo channels to a channel back and connect the channel bank to the digium card. |
| 03:16.05 | jpablo | [TK]D-Fender, why not ? I'll like to get that working ... |
| 03:16.31 | [TK]D-Fender | jpablo: there is no such functionality in SIP |
| 03:17.25 | jpablo | [TK]D-Fender, you can answer the sip half and pass the audio |
| 03:19.11 | [TK]D-Fender | jpablo: You are missing the point. Early media is a digital telecom signaling method. There is not translation from SIp to that. You can RECEIVE early media just fine, but thats THEIR side doing the work. don't expect to dial OUT and send early media... |
| 03:19.33 | soylent | oh wait Fender |
| 03:19.38 | soylent | I'm having a problem receiving it. |
| 03:20.03 | soylent | the cisco should be passing it from the tdm to my asterisk but is not, or seems not to be. |
| 03:20.15 | soylent | the only sip message I get is 100 trying |
| 03:20.16 | jpablo | early media makes no sense when you dialout |
| 03:20.34 | jpablo | you could talk for free |
| 03:20.43 | [TK]D-Fender | jpablo: Hrm |
| 03:21.29 | [TK]D-Fender | No matter.. I'm still overjoyed at Polycom SIP 2.1.0 today :) |
| 03:21.37 | jpablo | jeje |
| 03:21.40 | xpot | anyone know of a way to count the number of digits entered? |
| 03:22.14 | [TK]D-Fender | xpot: Entered on what? when? How? |
| 03:22.30 | soylent | damn, I'm still on 2.0.1.b |
| 03:22.34 | xpot | sip chan, when requested |
| 03:22.47 | xpot | for ex: Read(blah) |
| 03:22.54 | xpot | count(blah) |
| 03:22.54 | jpablo | soylent, i remember cisco sip implementation sucking badly, you can try to buy a sangoma or digium card. |
| 03:23.00 | jpablo | soylent, are you using e1 or t1? |
| 03:23.09 | soylent | t1 in us |
| 03:23.15 | soylent | can't remember for the other office |
| 03:23.21 | soylent | I think it's a t1 too though. |
| 03:23.25 | [TK]D-Fender | xpot: "show function LEN" |
| 03:23.31 | xpot | thanks |
| 03:24.16 | xpot | Fender: would you be able to answer my previous question as well? |
| 03:24.35 | xpot | here it is again: anyone know the proper use of calleridnum in 1.4 using agi? Here is what I currently have> my $callerid = $input{'calleridnum'}; |
| 03:24.39 | *** join/#asterisk ttuttle (n=tom@gentoo/contributor/ttuttle) |
| 03:24.58 | ttuttle | sjphone is the only softphone I've tried so far that works right with Asterisk. Neither Kphone nor Ekiga manage to register properly. |
| 03:25.25 | soylent | could this have anything to do with the progressinband setting in asterisk? |
| 03:25.27 | [TK]D-Fender | xpot: Never did AGI |
| 03:25.32 | ttuttle | Are there any other open source softphones that might work? |
| 03:25.34 | ttuttle | soylent: me? |
| 03:25.35 | jpablo | soylent, i guess i should work. |
| 03:25.36 | xpot | ok, thank you |
| 03:25.55 | [TK]D-Fender | ttuttle: ekiga works fine for me... |
| 03:26.36 | ttuttle | [TK]D-Fender: Hmm. |
| 03:30.02 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 03:30.29 | *** join/#asterisk wubba (n=kmurrey@cable-76-215.sssnet.com) |
| 03:30.37 | Carp1 | Is it possible to get a T1 with ONLY 5 or 6 voice channels? and no data. |
| 03:30.56 | ShadowHntr | Carp1: you can get a fractional t1 perhaps... |
| 03:31.10 | ShadowHntr | check with your local telco and data providers |
| 03:31.10 | Carp1 | Whoops, I meant that. |
| 03:31.26 | ShadowHntr | what country are you in |
| 03:31.28 | Carp1 | Where I live, there really isnt local anything |
| 03:31.30 | Carp1 | USA |
| 03:31.30 | ShadowHntr | oh |
| 03:31.32 | ShadowHntr | new york |
| 03:31.36 | Carp1 | Yes |
| 03:31.42 | ShadowHntr | check with like XO Communications (www.xo.com) or local large isps |
| 03:31.56 | Carp1 | Thanks. |
| 03:31.59 | ShadowHntr | you can get a fractional t1. it doesn't matter to the ISP or telco what you use the channels for |
| 03:32.08 | ShadowHntr | perhaps Verizon Business will be able to help you out =) |
| 03:32.38 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 03:33.19 | *** join/#asterisk Avochelm (n=damien@gw-morphett.koalatelecom.com.au) [NETSPLIT VICTIM] |
| 03:33.20 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) [NETSPLIT VICTIM] |
| 03:33.34 | Carp1 | With a fractional T1, you dont get 24 channels tho right? |
| 03:33.42 | Carp1 | Im reallt not sure how a T1 works lol |
| 03:34.11 | ShadowHntr | a T1 consists of 24 8-bit channels |
| 03:34.14 | JT | it's a fraction |
| 03:34.16 | ShadowHntr | that can be used for either data or voice |
| 03:34.23 | ShadowHntr | fractional means you don't get the whole T1 circuit |
| 03:34.27 | Carp1 | Not a fractional though |
| 03:34.30 | *** join/#asterisk Milk_ (n=None@74-134-97-185.dhcp.insightbb.com) |
| 03:34.31 | ShadowHntr | you tell the ISP how many of the 24 channels you want. |
| 03:34.33 | Milk_ | good evening! |
| 03:34.36 | JT | basically they allow you to only use a certain amount of channels |
| 03:34.38 | Carp1 | Ok. |
| 03:34.43 | Carp1 | Thanks. |
| 03:34.43 | JT | telcos provide T1s, not isps, really |
| 03:35.07 | JT | 8bit channels, that's a bit misleading |
| 03:35.15 | [TK]D-Fender | I want a 100% fraction of a T1. THERE! 24 channels! |
| 03:35.18 | JT | it has 24 * 64kbit/s |
| 03:35.25 | ShadowHntr | http://en.wikipedia.org/wiki/Digital_Signal_1 |
| 03:35.37 | Milk_ | I was here last night, but blew a fuse and lost connection, so.. I'm back |
| 03:35.57 | Milk_ | I'm trying to use an fxo card in my trixbox to allow me to dial out from a sip phone on my pots line |
| 03:36.04 | JT | it's just a tdm interface, and you only use what timeslots you need |
| 03:36.05 | Milk_ | I'm getting a "all circuits are busy" message |
| 03:36.10 | Milk_ | but not sure whats wrong |
| 03:36.14 | Carp1 | Milk_: there is a channel fro trixbox |
| 03:36.21 | Carp1 | try #trixbox or #freepbx |
| 03:36.36 | Milk_ | Carp1, no one ever answers in there |
| 03:36.41 | [TK]D-Fender | Milk_: .... |
| 03:36.42 | Milk_ | :) |
| 03:36.44 | [TK]D-Fender | ~trixbox |
| 03:36.46 | jbot | rumour has it, trixbox is unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
| 03:37.05 | Milk_ | ok... lets ignore that its trixbox |
| 03:37.06 | JT | ShadowHntr: 8bits is irrelevant at the interface level, as it's just a serial bidirectional data stream |
| 03:37.14 | [TK]D-Fender | Milk_: That unfortunately falls under the typically category of "TFB" |
| 03:37.15 | Milk_ | its a problem with the asterisk install on the trixbox machine... |
| 03:37.26 | JT | it's not asterisk |
| 03:37.30 | JT | it's freepbx on trixbox |
| 03:37.35 | [TK]D-Fender | Milk_: No, we won't. |
| 03:37.41 | ShadowHntr | JT: you're right. just trying to relay the info to Carp1. |
| 03:37.41 | ShadowHntr | :) |
| 03:37.48 | JT | freepbx is asterisk with very complicated included dialplans |
| 03:38.01 | ShadowHntr | :] |
| 03:38.07 | JT | ShadowHntr: :) |
| 03:38.28 | Milk_ | no one is even willing to give it a go huh |
| 03:39.34 | JT | Milk_: if you do lots of hunting to narrow down the problem beforehand, people might find it possible to give a little advice |
| 03:39.49 | Milk_ | I've done quite a bit |
| 03:40.31 | Milk_ | I've opened the CLI, set verbosity and debug to 10, and was in the process of pastebining the output when I lost power last night.... |
| 03:40.34 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 03:40.53 | Milk_ | I can recreate the error and start with that, but I assume I will be told TFB and RTFM instead of getting any help |
| 03:41.09 | [TK]D-Fender | Milk_: Trying to find out what you put in wrong to cause your failure just isn't worth the time for most of us. Check the forums if their IRC channels aren't actiice enough |
| 03:41.17 | [TK]D-Fender | Milk_: there are plenty of those |
| 03:42.02 | [TK]D-Fender | Milk_: the odds of you're conveying enough useful info for us to help find out what you did wrong are somewhat bleak |
| 03:42.23 | Milk_ | [TK]D-Fender, I find you offensive and a terrible representative of OSS, I'm trying to get help, and all your giving me is attitude |
| 03:42.31 | JT | err |
| 03:42.31 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 03:42.32 | JT | truth |
| 03:42.36 | JT | is not attitude |
| 03:42.40 | Milk_ | I'm not a newbie, scared of the CLI, scared of criticism, or scared of config files |
| 03:42.57 | JT | how can we help with something we aren't familiar with? |
| 03:42.59 | [TK]D-Fender | Milk_: Tell you what : Show us where you think the problem is. |
| 03:43.15 | Milk_ | I've been running a non-windows environment for many years and I am quite capable of problem solving, with some basic guidance |
| 03:43.17 | JT | if you ask for a very specific advice about something, with actual config data, we may be able to assist |
| 03:43.17 | Milk_ | now.... |
| 03:43.19 | wubba | All circuits are busy - usually a registration problme. |
| 03:43.25 | [TK]D-Fender | Milk_: the fact you get an error recording is of no assistance' |
| 03:43.35 | Milk_ | I appear to be registered just fine |
| 03:43.49 | [TK]D-Fender | Milk_: CLI output, config files for the channels affected, etc... |
| 03:43.51 | putzz | how would I go on to block only a certain area code, instead of listing all the allowed ones? |
| 03:44.01 | Milk_ | [TK]D-Fender, if you would read and get off your high horse, you would see that I said I had more info last night |
| 03:44.11 | Milk_ | maybe.. . just maybe I'm running debug now |
| 03:44.12 | Milk_ | just maybe |
| 03:44.17 | Milk_ | oh.. but wait.. I'm a newbie and TFB |
| 03:44.24 | wubba | Screw that |
| 03:44.26 | [TK]D-Fender | Milk_: If you hear a recording then your phone isn't the problem, its the other end you're trying to bridge in |
| 03:44.31 | JT | putzz: s/_123XXXXXXXX, or similar |
| 03:44.36 | wubba | I was going to jump on and help - but this guy is a goof. |
| 03:44.44 | JT | Milk_: last night is not now |
| 03:44.52 | JT | we can't see last night's conversation |
| 03:44.58 | JT | it's a long time ago in #asterisk land |
| 03:45.00 | Milk_ | it appears..... |
| 03:45.07 | [TK]D-Fender | Milk_: So you had more info tlast night. That to say you're going to provide even less now? |
| 03:45.07 | putzz | lol |
| 03:45.18 | Milk_ | I'm working on it |
| 03:45.19 | [TK]D-Fender | conterproductive :) |
| 03:45.20 | Milk_ | geez |
| 03:45.20 | JT | if we scrollback 10million lines, no thanks? |
| 03:45.30 | wubba | CLI would probably show exactly what the problem is... |
| 03:45.35 | Milk_ | do you guys really treat everyone this way? |
| 03:45.41 | Milk_ | <PROTECTED> |
| 03:45.43 | [TK]D-Fender | JT : "load chan_fluxcapacitor.so" |
| 03:45.53 | wubba | Nope - only people that act like we owe them something |
| 03:45.54 | [TK]D-Fender | Milk_: Again, not an * generated error. |
| 03:46.08 | Milk_ | further up |
| 03:46.10 | Milk_ | <PROTECTED> |
| 03:46.12 | JT | only people who complain about us telling you that it's hard to fix vague problems in unsupported software without lots of details |
| 03:46.14 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:46.14 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:46.21 | JT | that is not debug output |
| 03:46.33 | [TK]D-Fender | Milk_: Thats just part of the spam of message put out by FreePBX as a marker, but not truely indicitive of what channel caused the error |
| 03:46.41 | Milk_ | ok.. how can I differeitiate between debug and verbosity |
| 03:46.52 | JT | there's sip debig |
| 03:46.56 | Milk_ | ok |
| 03:46.57 | [TK]D-Fender | Milk_: www.pastebin.ca |
| 03:47.01 | Milk_ | let me pop that up |
| 03:47.04 | JT | which may or may not be necessary |
| 03:47.12 | [TK]D-Fender | Milk_: Dump the CLI output of an ENTIRE failed call |
| 03:47.23 | [TK]D-Fender | Milk_: and do "set verbose 10' prior to the call |
| 03:47.29 | JT | probably worth seeing what the hell your configs are actually trying to do though |
| 03:47.47 | [TK]D-Fender | JT : we dont even know WHICh oe to look foor yet :) |
| 03:47.56 | [TK]D-Fender | JT : No channel types mentioned! |
| 03:48.05 | [TK]D-Fender | JT : lets work from the ground up... |
| 03:48.17 | [TK]D-Fender | JT : or at least our head start at 6' under :) |
| 03:48.22 | JT | < Milk_> I'm trying to use an fxo card in my trixbox to allow me to dial |
| 03:48.22 | JT | <PROTECTED> |
| 03:48.32 | JT | true |
| 03:48.50 | [TK]D-Fender | JT : Still I wanna see the DIAL command and the lines immediately following it |
| 03:48.56 | JT | yeah |
| 03:49.00 | JT | would be useful |
| 03:49.04 | [TK]D-Fender | ( Milk_ this should be a HINT for you ) |
| 03:49.11 | [TK]D-Fender | PSSSSSSST |
| 03:49.13 | Milk_ | I'm pasting it now |
| 03:49.13 | *** join/#asterisk acecase (n=fu@h175.65.40.69.ip.alltel.net) |
| 03:49.14 | [TK]D-Fender | ^^^^^^^^^^^ |
| 03:49.17 | Milk_ | geez |
| 03:49.23 | acecase | hello |
| 03:49.33 | Milk_ | http://www.pastebin.ca/344958 |
| 03:49.52 | *** join/#asterisk CrashSys (n=kumba@158-211.187-72.tampabay.res.rr.com) |
| 03:49.58 | *** join/#asterisk shodan (n=shodan@ip047.96-113-216.pppoe1.joliette.intermonde.net) |
| 03:50.23 | JT | people are annoyed because you're making all sorts of remarks like "geez", Milk_, might be why you're getting a little bit of resistance |
| 03:50.26 | [TK]D-Fender | -- Executing Dial("SIP/3101-09c10640", "ZAP/g0/13092872137|300|") in new stack |
| 03:50.29 | [TK]D-Fender | PROGRESS |
| 03:50.35 | Milk_ | I can pare that down to remove the basic debug and basic verbosity, if its helpfull |
| 03:50.42 | CrashSys | Corydon: Ping ...? |
| 03:50.43 | Milk_ | geez.... haha |
| 03:50.48 | [TK]D-Fender | at least it looks like you're dialing a legit NA LD # |
| 03:51.28 | [TK]D-Fender | Milk_: No, phase 1 complete. now pastebin your zapata.conf , zaptel.con, and ALL files they link to (#include) |
| 03:52.10 | JT | Milk_: well you're doing an awful lot of complaining, keep in mind people here aren't paid to help |
| 03:52.24 | Milk_ | I wasn't complaining about the help |
| 03:52.27 | Milk_ | just the attitude |
| 03:52.29 | [TK]D-Fender | JT : Ok, I think he's recovering a bit... |
| 03:52.32 | Milk_ | ... I appreciate the help! |
| 03:52.47 | *** join/#asterisk marc7 (n=marc@S0106000f66461bdb.gv.shawcable.net) |
| 03:52.49 | [TK]D-Fender | ok, everyone cool it a bit. We'll see how this progresses |
| 03:53.00 | [TK]D-Fender | Milk_: get to that 2nd set of PB's you |
| 03:53.05 | [TK]D-Fender | ve been requested to provide |
| 03:53.19 | Milk_ | working on it now |
| 03:53.20 | marc7 | does anybody have any ideas how I can configure asterisk to beep every 60 seconds during a Record() statement? |
| 03:53.29 | Milk_ | heres zapata.conf |
| 03:53.30 | Milk_ | http://www.pastebin.ca/344961 |
| 03:54.16 | Milk_ | I don't see a zaptel.con |
| 03:54.36 | putzz | *conf |
| 03:54.40 | Milk_ | ok |
| 03:55.01 | acecase | can anyone point me to a real good set of asterisk configuration and language documentation? |
| 03:55.08 | Qwell | ~wikis |
| 03:55.21 | jbot | i guess wikis is http://www.voip-info.org |
| 03:55.21 | Milk_ | http://www.pastebin.ca/344964 |
| 03:55.22 | Milk_ | zaptel.conf |
| 03:55.24 | Qwell | Why does jbot hate me so much? |
| 03:55.35 | JT | mm, wouldn't it be useful for channels to be in zapata.conf? |
| 03:56.02 | acecase | thanks Qwell. I have been looking through that and its very fast but kinda unorganized for a noob |
| 03:56.12 | acecase | very Vast* |
| 03:56.13 | Qwell | ~book |
| 03:56.15 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:56.15 | putzz | ~seen mygod_run |
| 03:56.45 | jbot | putzz: i haven't seen 'mygod_run' |
| 03:56.45 | JT | and zaptel.conf |
| 03:56.45 | JT | looks like both files do not have channel numbers in them |
| 03:56.45 | sumasuma | hi, Is there is any asterisk GUI, so that i can just install it on my webserver and provide services to my customers and manage my asterisk ? |
| 03:56.45 | JT | slight issue |
| 03:56.48 | Milk_ | hrm... |
| 03:56.51 | acecase | thanks |
| 03:56.58 | [TK]D-Fender | Milk_: Please provide the 2 #include-ed files from zapata.conf |
| 03:57.35 | acecase | and its a free pdf even :) I apriciate it |
| 03:59.02 | Milk_ | http://www.pastebin.ca/344970 |
| 04:00.22 | sumasuma | any help for asterisk GUI please ? commercial ones is also ok |
| 04:00.23 | marc7 | our SIP carrier is prematurely terminating phone calls because someone is in the middle of a "record" statement... and because asterisk is just "receiving" sound, not "transmitting" or echoing anything back... our carrier thinks the call has ended. is there any way we can have asterisk echo a "beep" every minute, without interrupting a Record() command? |
| 04:00.45 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
| 04:01.04 | [TK]D-Fender | sumasuma: Wrong place for that. Please read the channel topic to see which other channel to go to for support |
| 04:01.11 | marc7 | we need to be able to record ~10 minute long messages, and we can only get to 3-4 minutes of our asterisk server not echoing anything before our carrier drops the call |
| 04:01.33 | [TK]D-Fender | Milk_: Ok, that all looks fine. pastebin the * CLI output of "zap show channels" |
| 04:01.36 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:01.36 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:01.41 | file | marc7: set transmit_silence_during_record to yes in asterisk.conf under the options context |
| 04:01.51 | file | it will probably not be there, so just add it |
| 04:01.55 | file | record will then transmit silence |
| 04:01.58 | marc7 | file: you rock steady |
| 04:01.59 | [TK]D-Fender | Milk_: FYI, the X100P has 2 ports. i already suspect you plugged the line in the wrong jack... |
| 04:02.04 | marc7 | thanks |
| 04:02.08 | Milk_ | http://www.pastebin.ca/344973 |
| 04:02.19 | Qwell | [TK]D-Fender: "But I plugged a phone into the phone port" |
| 04:02.21 | Milk_ | my card has only 1 port |
| 04:02.22 | Qwell | $20 |
| 04:02.26 | Qwell | damn |
| 04:02.32 | [TK]D-Fender | Milk_: And its not really 2 ports so much as a "pass-through" |
| 04:03.18 | *** join/#asterisk canadiancow (n=canadian@CPE00134640ff2f-CM00159a40c0be.cpe.net.cable.rogers.com) |
| 04:03.23 | JT | file: what's the alternate to transmitting silence? |
| 04:03.23 | [TK]D-Fender | Milk_: Well... everything else looks fine.. |
| 04:03.33 | file | JT: not transmitting silence? |
| 04:03.44 | JT | file: i though ast didn't support rtp silenve supression |
| 04:03.45 | [TK]D-Fender | Qwell : duh....duhh..duh...DUMB :) |
| 04:03.53 | Milk_ | hrm... |
| 04:03.59 | *** join/#asterisk ManxPower (n=manxpowe@203.sub-70-216-232.myvzw.com) |
| 04:04.13 | file | JT: it doesn't. |
| 04:04.43 | JT | well it seems to transmit rtp packets for me even when there is no activity |
| 04:05.05 | file | Record does not transmit packets while recording unless you turn on that option |
| 04:05.30 | JT | so whilst Record is being used, asterisk stops transmitting rtp packets? |
| 04:05.42 | [TK]D-Fender | ManxPower: SIP 2.1.0 iss out and add MicroBrowser support for IP 501 & 430 :) |
| 04:05.54 | file | yes |
| 04:06.11 | ManxPower | [TK]D-Fender: I'll play with it when a client wants to pay me to. |
| 04:06.19 | JT | is there any other situation when asterisk stops transmitting rtp packets during an in media path call? |
| 04:06.27 | Milk_ | the fxo card is deffinatly 1 port, and the line is active as I just tested on an analogue phone |
| 04:06.31 | ManxPower | [TK]D-Fender: What do you use the microbrowser for anyway? |
| 04:06.41 | file | I do not know off the top of my head at this time of night |
| 04:06.43 | Qwell | ManxPower: ascii pr0n |
| 04:07.05 | [TK]D-Fender | ManxPower: Live Queue status, VM count in multiple boxes. Line concurrency checks, etc |
| 04:07.05 | JT | file: ok, isn't that a bug, if asterisk doesn't support rtp silence supression? |
| 04:07.49 | ManxPower | [TK]D-Fender: I knew there was something missing from my life. |
| 04:08.00 | [TK]D-Fender | Milk_: :hrm. put that card on a splitter with another phone and try while offhook on the parallel phone |
| 04:08.17 | file | JT: if you want to add support, feel free |
| 04:08.18 | [TK]D-Fender | ManxPower: Sorry, I don't work for Pfizer... |
| 04:08.19 | [TK]D-Fender | ;) |
| 04:08.25 | file | but not supporting something does not a bug make |
| 04:08.31 | ManxPower | [TK]D-Fender: Anyway, 1.4 is an example of why I don't install .0 releases. |
| 04:08.51 | Milk_ | that line is connected to the whole house.. so I can just pick up another line |
| 04:08.53 | Milk_ | let me try |
| 04:09.01 | Qwell | what? |
| 04:09.07 | Qwell | ugh |
| 04:09.11 | [TK]D-Fender | ManxPower: Yeah I suppose, but it does solve a number of existing issues. I'll simply keep you abreast of my progress then. |
| 04:09.12 | JT | file: i understand, but shouldn't asterisk always transmit rtp packets during an in-media sip session, if it is the case that rtp silence supression is not supported? just curious |
| 04:09.43 | Milk_ | [TK]D-Fender, I hear nothing on the house like when trying to dial |
| 04:10.02 | [TK]D-Fender | ManxPower: seriously ups its functionality sa well. Tables support (about time!) and a few things |
| 04:10.21 | [TK]D-Fender | Milk_: Hrm. now its moved on to being a mystery... |
| 04:10.41 | Qwell | unplug all phones |
| 04:10.50 | file | JT: should? probably |
| 04:10.53 | ManxPower | [TK]D-Fender: Yes, but other than queues and trivial things like stoke quotes and weather, what USE is the microbrowser. |
| 04:10.53 | [TK]D-Fender | Milk_: Try PB-ing "show channels" |
| 04:11.08 | Qwell | ManxPower: I'm telling you...ascii pr0n |
| 04:11.10 | JT | file: okay |
| 04:11.13 | Qwell | You could sell that |
| 04:11.24 | russellb | Qwell: that would rock |
| 04:11.25 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 04:11.28 | [TK]D-Fender | ManxPower: Anything beyond nearly incidental info / company directory (good reason), begins asking why you're not using a PC. |
| 04:11.28 | Milk_ | http://www.pastebin.ca/344982 |
| 04:11.33 | Qwell | russellb: totally |
| 04:11.38 | Qwell | right bkruse_home ? |
| 04:11.55 | ManxPower | [TK]D-Fender: um, many of my users are not technical enough to use a computer. |
| 04:12.06 | [TK]D-Fender | Milk_: Ok, we've covered just about everything.... |
| 04:12.26 | Milk_ | [TK]D-Fender, and no thoughts on why? |
| 04:12.32 | [TK]D-Fender | ManxPower: Ok, get them a channel bank and a bunch of analo gphones and ship me their overstock :) |
| 04:12.47 | [TK]D-Fender | Milk_: Not offhand, still looking at to see if there's something I missed. |
| 04:13.09 | Milk_ | [TK]D-Fender, thanks! |
| 04:13.13 | canadiancow | are there any mirrors i can download asterisk from? |
| 04:13.18 | [TK]D-Fender | Milk_: maybe try replugging the line. maybe loose/ |
| 04:13.33 | [TK]D-Fender | canadiancow: www.asterisk.org has links |
| 04:13.55 | ManxPower | [TK]D-Fender: Maybe I could tap into the account database and post the most recent check written to the top 10 paid people in the company. |
| 04:14.20 | bkruse_home | Qwell: yes! |
| 04:14.21 | canadiancow | [TK]D-Fender, it has links to the companies that provide bandwidth, but no other download links |
| 04:14.36 | russellb | bkruse_home: have you done your homework young man? |
| 04:14.46 | Milk_ | [TK]D-Fender, I had just done that a minute ago |
| 04:15.28 | Qwell | pear flavored vodka? wtf |
| 04:15.53 | bkruse_home | russellb: no ;[ |
| 04:15.59 | bkruse_home | russellb: i got some pre-cal :[ |
| 04:16.07 | russellb | lame |
| 04:16.10 | bkruse_home | but ill do it tomorrow in physics, its resource writing time |
| 04:16.11 | bkruse_home | :] |
| 04:16.22 | russellb | didn't your mother tell you to do your homework before getting on IRC? |
| 04:17.15 | russellb | :-p |
| 04:17.28 | russellb | rewritten the CLI commands? |
| 04:17.57 | Qwell | bkruse_home: You should do well in physics. We need somebody to write res_physics |
| 04:18.11 | russellb | why do we need a physics engine in asterisk ? |
| 04:18.13 | Qwell | oh wait, that's res_psychic :P |
| 04:18.14 | bkruse_home | Qwell: what do we need res_physics to do? |
| 04:18.16 | bkruse_home | LOL! |
| 04:18.17 | *** join/#asterisk ManxPower (n=manxpowe@150.sub-70-216-156.myvzw.com) |
| 04:18.33 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:18.34 | russellb | that would be silly |
| 04:18.34 | Qwell | russellb: and the obvious answer, is...Wii bowling |
| 04:18.49 | ManxPower | As you can see the infrastructire in New Orleans is not very reliable yet |
| 04:18.50 | bkruse_home | Qwell: i wanted to through down some wii tenis today |
| 04:18.52 | [TK]D-Fender | Milk_: Ok, zaptel.conf one last time please... |
| 04:18.59 | bkruse_home | mitchel got op'd but bkruse didnt?? :[ no irc op's under 20? |
| 04:19.02 | Qwell | see, what you do, is swing an analog handset around... |
| 04:19.12 | Qwell | and it can "hear" it's direction |
| 04:19.23 | *** mode/#asterisk [+v bkruse_home] by Qwell |
| 04:19.28 | bkruse_home | Qwell: and it tells you the area of the room and providers echo can? |
| 04:19.30 | *** join/#asterisk andres_pag (n=andres_p@70.56.223.76) |
| 04:19.59 | CrashSys | res_psychic?!?! |
| 04:20.24 | *** join/#asterisk ez` (n=ez@c66.203.210-59.clta.globetrotter.net) |
| 04:20.32 | Milk_ | http://www.pastebin.ca/344986 |
| 04:20.57 | CrashSys | bowling for TDM |
| 04:20.59 | CrashSys | I like it |
| 04:21.12 | *** join/#asterisk fwp (n=FWP@unaffiliated/fwp) |
| 04:22.09 | [TK]D-Fender | Milk_: Ok, I officially can't see why.... |
| 04:22.26 | Milk_ | [TK]D-Fender, well.. I appreciate your time! |
| 04:22.33 | *** join/#asterisk bpiper (n=bpiper@user-142gior.cable.mindspring.com) |
| 04:22.36 | CrashSys | well that's interesting... chan_oss shoots the CPU usage to 160% when I call it... |
| 04:22.38 | CrashSys | I like it... |
| 04:22.56 | [TK]D-Fender | Milk_: Well you started producing materials. that'll increase your odds. |
| 04:23.25 | Milk_ | [TK]D-Fender, I always produce... just no point in spilling your guts off the bat when you don't know if anyone is willing |
| 04:24.17 | russellb | ooh, code that uses 160% of a processor? |
| 04:24.21 | russellb | that must be pretty l33t |
| 04:24.31 | CrashSys | yeah |
| 04:24.37 | CrashSys | it's so l33t that it goes oops :D |
| 04:24.54 | bkruse_home | russellb: its pretty crazy,i use res_fakesmellingdwaynecandleforrussell |
| 04:25.01 | bkruse_home | its a thread hog, and smells like bubble gum. |
| 04:25.02 | Milk_ | well.. I'm off to do some reading |
| 04:25.05 | Milk_ | thanks again guys |
| 04:25.15 | russellb | bkruse_home: o.O |
| 04:25.30 | bkruse_home | russellb: I want HALF of your candle, its delighful |
| 04:25.45 | russellb | why do you want my candle? |
| 04:25.55 | russellb | it's ... a cheap candle from walmart |
| 04:26.06 | CrashSys | chan_alsa fires the CPU up to about 2%... |
| 04:26.15 | CrashSys | but the dsp/answer and dsp/noanswer patch aint workin'... |
| 04:26.26 | bkruse_home | russellb: its an awesome color, and an awesome scent |
| 04:26.32 | bkruse_home | and does NOT smell like bubble gum. |
| 04:26.40 | *** join/#asterisk pirulo (n=andres_p@70.56.223.76) |
| 04:26.49 | russellb | <3 |
| 04:27.06 | russellb | mmmm |
| 04:27.14 | Qwell | PQ would be...mad :p |
| 04:27.18 | russellb | i think burning incense would be over the line ... |
| 04:27.32 | russellb | i guess if you closed your door :) |
| 04:27.32 | Qwell | "Are you... ... ...baking something?" |
| 04:27.38 | bkruse_home | Qwell: omg. matt would not shutup about the bubble gum |
| 04:28.03 | bkruse_home | Qwell: get the cinnamon candle and fan the door |
| 04:28.04 | *** part/#asterisk pirulo (n=andres_p@70.56.223.76) |
| 04:28.07 | *** part/#asterisk andres_pag (n=andres_p@70.56.223.76) |
| 04:28.31 | russellb | i wonder if they have chocolate chip cookie incense |
| 04:29.26 | bkruse_home | russellb: I found it! |
| 04:29.35 | Qwell | russellb: http://www.naturesgardencandles.com/candlemaking-soap-supplies/item/rf-32 |
| 04:29.45 | ManxPower | russellb: They have leather scented candles, chocolate chip cookie incense doesn't seem like a stretch. |
| 04:30.12 | rudholm | what about "New Car" candles? |
| 04:30.20 | Qwell | rudholm: heh |
| 04:31.03 | russellb | http://cgi.ebay.com/Chocolate-Chip-Cookie-Premium-Incense-Sticks-20-pk_W0QQitemZ280034334575QQihZ018QQcategoryZ43405QQssPageNameZWDVWQQrdZ1QQcmdZViewItem?hash=item280034334575 |
| 04:31.03 | rudholm | where's Strom? don't they have teh intarweb in Canada by now? |
| 04:31.32 | bkruse_home | russellb: HA, i like it |
| 04:31.32 | ManxPower | rudholm: It's Feb. The internet tubes are frozen up there |
| 04:31.37 | rudholm | oh right |
| 04:31.44 | rudholm | he must be freezing |
| 04:31.53 | canadiancow | its not that cold ;) |
| 04:32.13 | rudholm | he was one big shiver in Seattle a couple weeks ago. |
| 04:32.21 | [TK]D-Fender | canadiancow: yes it is. Spit BOUNCES |
| 04:32.27 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:32.31 | putzz | heh |
| 04:32.32 | rudholm | spit shatters |
| 04:34.17 | bkruse_home | msg russellb russellb: some of these NEW_CLI declarations in the ast_cli_entry struct dont have cli commands to get to them?? |
| 04:34.20 | bkruse_home | whoops. |
| 04:34.33 | bkruse_home | there :] |
| 04:34.43 | Corydon76-home | Oops |
| 04:34.51 | bkruse_home | oh noes! |
| 04:35.08 | Corydon76-home | Evening |
| 04:36.06 | *** join/#asterisk alindeman (i=adml@freenode/staff/alindeman) |
| 04:37.30 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 04:38.06 | bpiper | interesting question for you guys... I've been running asterisk for 1.2.7.1 since it first came out. Had no problems, then all of the sudden the other day it started crashing. I checked ps aux and it shows the asterisk process at 99.9% cpu. Since then I upgraded to 1.2.14 and same problem. It runs for about 15 min & crashes again... Anyone know where to look to figure this one out? |
| 04:38.31 | Qwell | bpiper: using mpg123 for MoH? |
| 04:38.39 | bpiper | nope |
| 04:38.58 | CrashSys | Using Chan_OSS? :D |
| 04:39.12 | bpiper | huh? Chan_OSS? |
| 04:39.26 | CrashSys | well mine uses all the CPU |
| 04:39.34 | CrashSys | but i'm special so far |
| 04:40.16 | bpiper | it used to use less than 1% on a dual xeon 3.2ghz |
| 04:40.55 | bpiper | if I do "restart now" it will kick me out & never let me back in until I reboot |
| 04:41.16 | bpiper | it says that it can't connect to asterisk.ctl |
| 04:41.51 | ManxPower | bpiper: perhaps you need to do some debugging |
| 04:42.07 | ManxPower | like run asterisk -cvvv then do a restart now and see what the error message is |
| 04:42.07 | bpiper | that's why I'm here, I'm not sure where to start... |
| 04:42.23 | bpiper | it says segmentation fault if I do that after it crashes |
| 04:42.50 | ManxPower | bpiper: that is not typical What version of the linux kernel, what verison of Asterisk, any custom patches? |
| 04:43.20 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-177-0-178.vic.bigpond.net.au) |
| 04:43.36 | bpiper | ManxPower... asterisk-1.2.14, CentOS, no custom patches |
| 04:44.20 | [TK]D-Fender | CrashSys: Thats why you get to ride in the little bus :) |
| 04:45.59 | bpiper | ManxPower, everything worked perfectly for months... nothing was changed and then it started crashing every day... the only way to route traffic again is to reboot |
| 04:46.01 | CrashSys | d-fendeR: the twinkie bus |
| 04:46.14 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:46.14 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:46.21 | CrashSys | bpiper: HW Failure? |
| 04:46.25 | CrashSys | bad PS? |
| 04:46.42 | CunningPike | bpiper: Do you yum automatically? |
| 04:46.46 | bpiper | how would a bad power supply affect asterisk? |
| 04:47.02 | bpiper | no, but I just ran it today to see if it would help |
| 04:47.07 | CrashSys | a bad power supply with low voltage on the supply rails will effect everything |
| 04:47.21 | CrashSys | you say it causes a kernel panic right? |
| 04:47.42 | bpiper | not really, asterisk seems to be the only thing that is affected |
| 04:47.48 | CrashSys | ahhh |
| 04:47.57 | CrashSys | nevermind then... I came in half baked... |
| 04:48.13 | bpiper | smoke one for me ;-) |
| 04:48.15 | CrashSys | I may ride this thing, but fender does the driving... |
| 04:48.34 | CunningPike | bpiper: I would try gdb - see if a stack trace shows anything |
| 04:49.08 | bpiper | not familiar with gdb, how do I use it? |
| 04:49.22 | CrashSys | Maybe if I try using chan_oss with OSS turned on in kernel (instead of the alsa wrapper) it'll work better... |
| 04:49.30 | CrashSys | hope my sound card loads... |
| 04:49.50 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:49.50 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:53.16 | bpiper | CrashSys: is it odd that ps aux shows both asterisk -vvv -c and safe_asterisk? |
| 04:53.24 | CrashSys | no |
| 04:53.28 | bpiper | oh |
| 04:53.34 | CrashSys | safe_asterisk makes sure asterisk -vvv -c is running |
| 04:53.39 | CrashSys | like a watchdog |
| 04:53.47 | bpiper | I'm one stop away from formatting this damn thing & starting over |
| 04:53.59 | CrashSys | what distro? |
| 04:54.00 | ManxPower | bpiper: it won't help. |
| 04:54.01 | bpiper | I really don't want to do that though since it is in a colo about 2 hours away |
| 04:54.05 | bpiper | no? |
| 04:54.15 | CrashSys | it didn't just break... |
| 04:54.20 | CrashSys | something happened... |
| 04:54.28 | ManxPower | bpiper: not unless you updated something betwen the time it worked and the time it stopped working. |
| 04:54.34 | CrashSys | if you dont figure out what happened, it will just happen again |
| 04:54.39 | CunningPike | bpiper: Rebuild with DONT_OPTIMIZE selected in menuselect |
| 04:55.06 | bpiper | CunningPike: rebuld asterisk? with DONT_OPTIMIZE? |
| 04:55.19 | CunningPike | bpiper: Then, when asterisk is in its 99% state: sudo gdb /usr/sbin/asterisk `cat /var/run/asterisk.pid` |
| 04:55.43 | CunningPike | bpiper: Then: sudo gdb /usr/sbin/asterisk `cat /var/run/asterisk.pid` |
| 04:56.01 | CunningPike | bpiper: That will yield stack traces that you can pastebin for the smarties in here to look at |
| 04:56.09 | CrashSys | will it his 99% at idle? |
| 04:56.13 | CrashSys | his = hit |
| 04:56.35 | bpiper | yea, not a single call going through right now & it is at 103% |
| 04:56.39 | eald | hi, anyone had has any problem with Hyperthreading enabled computer when running asterisk? It is recommended to turn off? |
| 04:56.52 | CrashSys | hmmm |
| 04:56.57 | CunningPike | bpiper: Yes - but previous activity may have done that |
| 04:57.08 | bpiper | i rebooted about 3 minutes ago |
| 04:57.10 | CrashSys | using realtime or anything fancy like that? |
| 04:57.19 | CunningPike | eald: Yes, it is. Turn it off in the BIOS and boot with noht |
| 04:57.20 | bpiper | no realtime, static only |
| 04:57.38 | CrashSys | Hyperthreading = a windows thing :) |
| 04:58.15 | bpiper | perhaps but it's been running some form of asterisk for almost 2 years now |
| 04:58.23 | bpiper | no problems until 2 days ago |
| 04:58.33 | CrashSys | what distro? |
| 04:58.48 | eald | I had a 10 segmentation fault in one hour in a 60 people configuration, after two weeks of testing without problems |
| 04:59.12 | bpiper | started back in 1.2.1 and presently on 1.2.14 |
| 05:00.16 | eald | I really don't *know* what was the problem, I ran the hardware diagnostic utility that comes in the BIOS and everything is fine there, but hyperthreading is enable and now is the primary suspect of the multiple crime |
| 05:02.38 | bpiper | freaking yum was on in a cron.daily job... I wasn't aware of that... |
| 05:02.55 | bpiper | anyone know if I can "uninstall" a yum update? |
| 05:03.25 | ManxPower | bpiper: I would recommend rebuilding asterisk first |
| 05:03.41 | bpiper | already done... |
| 05:03.46 | ManxPower | any any asterisk related software. If the issue is a library update compat issue, that would fix it. |
| 05:03.54 | ManxPower | Ah, OK. |
| 05:04.03 | CunningPike | bpiper: kernel updates too? |
| 05:04.55 | bpiper | I'm not sure... |
| 05:05.01 | k-man_ | how can i make a dial plan so, on my local phone I can hear what an inbound call would hear? |
| 05:05.37 | bpiper | CunningPike, here is the cron... perhaps that can tell you if it was a kernel update too |
| 05:05.38 | bpiper | #!/bin/sh |
| 05:05.38 | bpiper | <PROTECTED> |
| 05:05.38 | bpiper | if [ -f /var/lock/subsys/yum ]; then |
| 05:05.38 | bpiper | <PROTECTED> |
| 05:05.38 | bpiper | <PROTECTED> |
| 05:05.40 | bpiper | fi |
| 05:05.41 | ManxPower | k-man_: your question makes no sense |
| 05:05.53 | k-man_ | doesn;'t it? |
| 05:05.55 | k-man_ | made sense to me |
| 05:05.57 | k-man_ | ;) |
| 05:06.12 | ManxPower | nope. |
| 05:06.20 | k-man_ | imaging i am setting up an ivr for inbound calls |
| 05:06.38 | ManxPower | k-man_: dial the extension of the IVR |
| 05:06.40 | k-man_ | i want to access that ivr from my local sip phone |
| 05:06.44 | k-man_ | o |
| 05:06.44 | k-man_ | h |
| 05:06.46 | k-man_ | i see |
| 05:07.08 | ManxPower | Well designed systems would not have the IVR on exten => s. |
| 05:07.19 | *** join/#asterisk [TK]D-Fender (n=joe_blow@64.235.216.2) |
| 05:07.29 | ManxPower | Well designed systems would use a Goto in exten => s to go to the real IVR extension |
| 05:08.09 | CunningPike | bpiper: On the phone - sorry |
| 05:08.17 | ManxPower | since exten s is ONLY executed when Asterisk does not receive a destination number for the call and that usually only happens on analog fxo ports and (arguably) no well designed system would use analog fxo ports |
| 05:08.33 | bpiper | No prob, CunnincPike, I appreciate the help |
| 05:09.04 | ManxPower | We use XX01 as the IVR extension where XX is the 2 digits that are assigned to all the extensions at that office. |
| 05:09.37 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 05:09.37 | *** mode/#asterisk [+o mitcheloc] by ChanServ |
| 05:11.08 | ManxPower | XX09 for voicemail, XX15 for parking, XX16 - XX19 for picking up a parked call, etc |
| 05:11.32 | mitcheloc | <PROTECTED> |
| 05:11.42 | Qwell | mitcheloc: next to the V key |
| 05:12.27 | ManxPower | you could of course use exten s for the ivr and set up an extension with a goto, of course. |
| 05:12.54 | mitcheloc | Qwell: can't seem to find it |
| 05:12.56 | CunningPike | bpiper: Does your yum.conf have exclude=kernel in it? |
| 05:13.38 | bpiper | CunningPike, no |
| 05:14.28 | CunningPike | bpiper: Rebooted recently (i.e. between when it worked and when it started this behavior)? |
| 05:14.51 | CrashSys | Whatever this yum thing is it doesn't sound very delicious... |
| 05:14.59 | bpiper | many times, and upgraded from 1.2.7.1 to 1.2.14... nothing made any difference |
| 05:15.24 | ManxPower | crash it is a URPMI wannabe, which is a replacement for RPM |
| 05:15.25 | CunningPike | bpiper: What might have happened is that a kernel updated borked zaptel |
| 05:15.42 | CunningPike | s/updated/update/ |
| 05:15.44 | ManxPower | I doubt that bpiper has zaptel |
| 05:15.48 | bpiper | I don't |
| 05:15.56 | CunningPike | Oh |
| 05:15.57 | CunningPike | :) |
| 05:16.02 | ManxPower | since if he did then ASTERISK WOULD NOT START AT ALL |
| 05:16.13 | CunningPike | No need to shout, ManxPower |
| 05:16.17 | bpiper | hehe |
| 05:16.41 | ManxPower | s/ASTERISK WOULD NOT START AT ALL/<em>ASTERISK WOULD NOT START AT ALL</em>/ |
| 05:16.52 | ManxPower | stupid jbot |
| 05:16.54 | bpiper | <bitchslap> |
| 05:17.36 | CunningPike | bpiper: Well, I would try rebuilding everything in any case..... |
| 05:18.00 | ManxPower | bpiper: As I said, your experience is not typical. |
| 05:18.08 | *** join/#asterisk remowylliams (n=Mare@207.65.58.203) |
| 05:18.12 | bpiper | tried that too, if by everything you mean asterisk. I completely deleted all instances of asterisk & rebuilt |
| 05:18.26 | bpiper | my next thought was to reinstall the OS |
| 05:18.57 | bpiper | something sure is f&^ked up and I need this server back up and running |
| 05:19.02 | remowylliams | I'm sorry bringing this up if it's in the wrong room. I just updated my trixbox to 2.0 and it seems like I have to register on the trixbox site first. Is this for real? |
| 05:19.20 | *** part/#asterisk mogorman (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 05:19.21 | bpiper | eek, trixbox... I believe they have their own forum |
| 05:19.28 | CrashSys | I dunno |
| 05:20.26 | *** join/#asterisk tim0123 (n=cash247@adsl-75-39-213-70.dsl.rcsntx.sbcglobal.net) |
| 05:20.52 | bpiper | CunningPike: Thanks for all the help, I'm just going to have to make the 2 hr drive & format the server. |
| 05:21.03 | CunningPike | bpiper: OK - have fun :) |
| 05:21.05 | bpiper | I'm betting it was the yum after all |
| 05:21.07 | CunningPike | ~wglwat |
| 05:21.09 | jbot | rumour has it, wglwat is well, good luck with all that |
| 05:21.12 | CunningPike | ;) |
| 05:23.06 | tim0123 | Are there any Third party verification add on's out there for asterisk |
| 05:23.21 | *** join/#asterisk angler_ (i=angler@nat/digium/x-fda8e0c326320b1a) |
| 05:27.41 | *** part/#asterisk misc-- (n=misc@203.87.183.134) |
| 05:30.39 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.253.70) |
| 05:30.51 | yidiyuehan | hello good afternoon everybody. |
| 05:31.08 | yidiyuehan | could anyone tell me how i can allow h.263+ in asterisk? |
| 05:31.16 | Qwell | allow=h263 |
| 05:31.27 | yidiyuehan | just put allow=h263+ in /etc/asterisk/sip.conf? |
| 05:31.29 | Qwell | plus you need to enable video support |
| 05:31.30 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 05:31.45 | Qwell | hmm, I'm not sure if there's a separate one for h263+ |
| 05:31.49 | yidiyuehan | but allow =h263 is just H263 instead of H.263+ right? |
| 05:31.50 | Qwell | I guess there would have to be though |
| 05:32.10 | yidiyuehan | yes i can use H263 for video call and i have tested it successfully. |
| 05:32.25 | Qwell | then yeah, it's probably allow=h263+ |
| 05:32.26 | yidiyuehan | however H.263+ is better right? but i cannot use it with X-lite 3. |
| 05:32.53 | Qwell | x-lite does video now? |
| 05:32.58 | yidiyuehan | yes,;-) |
| 05:33.01 | Qwell | neat |
| 05:33.11 | yidiyuehan | x-lite 3.0 is a lower versio of eyebeam |
| 05:33.32 | yidiyuehan | but qwell, do you know how i can patch H264 for asterisk 1.2.14? |
| 05:33.39 | Qwell | patch it how? |
| 05:33.47 | yidiyuehan | as i am using freepbx, i cannot install asterisk 1.4.0 with it |
| 05:34.04 | yidiyuehan | well, ok, i know there is a way, but forgot the link:), thanks although |
| 05:35.53 | *** join/#asterisk litage (n=nick@203.220.55.70) |
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| 05:38.10 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 05:38.10 | *** mode/#asterisk [+o russellb] by ChanServ |
| 05:46.14 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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| 05:46.50 | Qwell | alindeman: ^^^ |
| 05:47.12 | brookshire | QWELL!!!!!!!!!!! |
| 05:47.22 | Qwell | brookshire: !! when you gonna stop by? :p |
| 05:47.29 | brookshire | i tried to today |
| 05:47.35 | brookshire | but my id wouldn't let me inside |
| 05:47.39 | Qwell | heh |
| 05:47.45 | Qwell | You could've...called somebody ;/ |
| 05:47.50 | brookshire | i did |
| 05:47.59 | brookshire | no one let me in |
| 05:48.03 | Qwell | lame |
| 05:48.12 | SwK | brookshire: maybe it was a hint |
| 05:48.15 | SwK | heh |
| 05:48.20 | brookshire | probably |
| 05:48.21 | brookshire | :) |
| 05:57.44 | [TK]D-Fender | Well i'm talking the dive into SIP 2.1.0. Will know how it all turns out momentarily |
| 05:57.57 | CrashSys | 2.1.0? |
| 05:57.59 | CrashSys | who's 2.1.0? |
| 05:58.08 | danp | the polycom firmware |
| 05:58.19 | wunderkin | we're still having reboot problems with 2.1.0 :P |
| 05:59.10 | danp | what's new in it? |
| 06:00.26 | CrashSys | 2.1.0 - 1.6.7 = approximately 0.4.3 things worth :) |
| 06:01.01 | wunderkin | close to half |
| 06:01.03 | danp | i use 2.0.1 currently |
| 06:03.55 | [TK]D-Fender | danp: MicroBrowser support for IP 501 & 430,a pile of other fixes, not redundent server support, dialplan prefix/suffis ability, etc. |
| 06:06.16 | *** part/#asterisk bpiper (n=bpiper@user-142gior.cable.mindspring.com) |
| 06:08.26 | danp | interesting |
| 06:08.36 | danp | where did you obtain it? |
| 06:09.11 | [TK]D-Fender | danp: My reseller |
| 06:10.56 | k-man_ | i still don't understand the difference between type=friend and type=peer in sip.conf |
| 06:11.16 | k-man_ | which one should I have to allow DID? or is that not connected to DID? |
| 06:12.50 | CrashSys | Anyone ever used the #8 (call pickup on ringing phone) with Polycom IP430's? |
| 06:15.05 | JT | k-man_: it's in the book |
| 06:15.11 | JT | friend is an alias for peer and user |
| 06:18.42 | [TK]D-Fender | 501 MB is nifty.... |
| 06:18.48 | [TK]D-Fender | seems to be the same |
| 06:19.00 | [TK]D-Fender | new tags supported including tables |
| 06:19.02 | [TK]D-Fender | thanks god |
| 06:19.14 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 06:22.25 | remowylliams | can rockwell hfc modems be used with asterisk? I"m not sure if this card has voice on it or not. |
| 06:25.30 | *** join/#asterisk loca|host (n=fourat@196.203.53.221) |
| 06:25.53 | JT | remowylliams: i don't think so |
| 06:27.06 | remowylliams | Well darn |
| 06:27.39 | JT | i assume you're talking about isdn |
| 06:28.10 | hads | Check if it's supported by mISDN |
| 06:28.11 | remowylliams | jt: No I'm talking a plain old modem |
| 06:28.23 | hads | Oh, then don't :) |
| 06:28.24 | JT | remowylliams: oh, definately not then |
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| 06:35.18 | CunningPike | Is there an openSER channel? |
| 06:35.32 | CunningPike | nm - found it |
| 06:35.33 | CunningPike | :) |
| 06:35.37 | sevard | heh |
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| 06:41.03 | CrashSys | ok... paging and ringing notification taken care of... |
| 06:41.13 | CrashSys | i'll just record the page, then issue a command to the OS to play the file |
| 06:41.19 | CrashSys | and alsa can just ring |
| 06:41.21 | CrashSys | all solved |
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| 07:11.27 | joe | anyone know what a config error 0x4020 is? |
| 07:11.41 | joe | not many answers via google... |
| 07:19.52 | CunningPike | joe: Is that from a Polycom phone? |
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| 07:20.28 | joe | CunningPike: yes, polycom 301 |
| 07:21.45 | CunningPike | joe: Usually a syntax error in your config file |
| 07:22.06 | joe | CunningPike: heh, k. thanks |
| 07:22.28 | joe | CunningPike: any config file parsers available from polycom that you know of? |
| 07:22.40 | CunningPike | joe: I wish :D |
| 07:22.51 | joe | you'd think they would have one! |
| 07:23.25 | joe | figured it was a silly error of some sort, I just can't see it atm and I'm too tired so off to bed ... |
| 07:23.36 | CunningPike | joe: Yup - fresh eyes are better |
| 07:24.52 | J4k3 | "Remedial Mail... Mailbox" |
| 07:24.54 | J4k3 | hehe |
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| 07:48.22 | adeeln | asterisk is core dumping on me, and i can't seem to figure out what's causing it...the last thing it loads is app_chanisavail.so |
| 07:49.55 | adeeln | hmmm...there are some unresolved symbols |
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| 08:12.18 | Mavvie | is it just me, or is it not possible to disable CallerID on zap channels. |
| 08:12.36 | Mavvie | I see #defines like AST_PRES_RESTRICTED, but it is nowhere used. |
| 08:12.55 | Corydon76-home | Use SetCallerPres |
| 08:13.27 | Mavvie | aaah. thanks Corydon76-home |
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| 08:50.01 | parag_ast | hi can anybody tell me that if i want to accept any call from ip address 72.36.131.23 without authenticating then what context do i need to write.... |
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| 09:06.39 | asteriskdude9 | Hi, it got disconnected before. |
| 09:06.40 | asteriskdude9 | I have a fairly complicated setup. Extensions (1,2 and 3). In 3 - I execute AGI in java which play few wav files depending on external parameters. How do I accomodate user who needs to reach extension 2 from my agi. Can I have a dial plan inside my AGI? |
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| 09:51.11 | hkdaylxb | I have receive an undefined indication from zap , how can I figure out the tonelist ? |
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| 10:17.20 | tzafrir | hkdaylxb, what do you mean by "undefined indication"? |
| 10:19.20 | mkl1525 | Hi, I've setup hints for my snom phones so that I can see who's calling an other phone and pick it up (using debian etch asterisk-bristuff version). Problem is that I see on my display "441 > 441" where 441 is the called phone. Can anybody please give a hint which config might be wrong? |
| 10:21.01 | hkdaylxb | tzafrir, I mean a tone that asterisk cannot resolve within the dialplan |
| 10:21.43 | hkdaylxb | such as Goto(s-${DIALSTATUS},1) |
| 10:22.52 | hkdaylxb | if the indication tone is defined , asterisk will be able to jump to s-something |
| 10:23.18 | tzafrir | indications are defined in indications.conf |
| 10:24.24 | hkdaylxb | yes, but I have received an undefined tone . It may be generated by a special switch that does not follow standards |
| 10:25.52 | corruptor | has any1 got problems with ver 1.2.14 when asterisk drops some calls with "No response to our critical packet" after 20 secs? I've watched sip debug and everythings seems ok with signalling but it seems that * just ignores ACKs. I've rolled back to 1.2.13 and that's solved the problem. |
| 10:26.23 | hkdaylxb | I would like to record the tone list, so that I can append a special extension for it. |
| 10:26.25 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 10:26.38 | corruptor | it looks like a bug |
| 10:36.26 | Mavvie | hmm... do you think that the features described in features.conf works via a SIP phone towards a PRI channel? |
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| 10:51.45 | lyGP | hi |
| 10:51.59 | lyGP | any guide to run 2 asterisk together? |
| 11:09.52 | dlynes_laptop | Mavvie, which features, specifically? afaik, all the features in features.conf are pbx features; they're not channel features |
| 11:10.39 | Mavvie | dlynes_laptop: I just want to see if they work. I can press * as often as I want, but it still doesn't disconnect the call. |
| 11:10.54 | *** join/#asterisk af_ (n=getsmart@ip-179-53.sn1.eutelia.it) |
| 11:10.56 | Mavvie | which makes me go "You're doing it wrong again dutchman!" |
| 11:11.04 | dlynes_laptop | hrm |
| 11:11.18 | dlynes_laptop | I think you're doing it wrong again, trekkie |
| 11:12.59 | dlynes_laptop | You mean like '*0'? |
| 11:13.09 | Mavvie | yeah, same result. |
| 11:13.17 | dlynes_laptop | What's your Dial command? |
| 11:14.05 | dlynes_laptop | And also do you have canreinvite=no set on your sip channel? |
| 11:14.29 | Mavvie | Dial(Zap/g3/0409227633,,wW) |
| 11:14.45 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 11:15.11 | Mavvie | there is no canreinvite=no |
| 11:15.57 | dlynes_laptop | Mavvie, do a show peer peername on the sip peer in question |
| 11:16.24 | dlynes_laptop | Mavvie, do you see 'CanReinvite : No' in the properties? |
| 11:16.51 | Mavvie | <PROTECTED> |
| 11:17.06 | Mavvie | does it matter, it's the SIP phone on my desk. |
| 11:17.09 | Mavvie | ? |
| 11:17.28 | Mavvie | aha! |
| 11:17.46 | Mavvie | first a #, then it works with the tT option in the dial command |
| 11:18.18 | dlynes_laptop | Mavvie, now is this phone connected to asterisk from behind a nat? |
| 11:18.25 | dlynes_laptop | Mavvie, or is it and asterisk on the same network? |
| 11:18.44 | Mavvie | dlynes_laptop: it works with a hH option. nice nice. |
| 11:18.49 | Mavvie | dlynes_laptop: all on the same subnet. |
| 11:19.06 | dlynes_laptop | ah...yeah...you forgot hH :) |
| 11:19.08 | dlynes_laptop | hehehe |
| 11:19.45 | dlynes_laptop | Anyways |
| 11:20.23 | dlynes_laptop | When you add t, T, h, H, w, or W it's the same as specifying canreinvite=no in your sip.conf file |
| 11:22.03 | dlynes_laptop | If you want *0 to work for hangup, though |
| 11:22.09 | dlynes_laptop | You need w and/or W |
| 11:22.13 | Mavvie | yups. |
| 11:22.22 | dlynes_laptop | And you also need to do a Set(DYNAMIC_FEATURES=automon) |
| 11:22.33 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 11:22.47 | dlynes_laptop | That should be placed before the Dial command |
| 11:22.47 | Mavvie | strangely enough it's not shown up as "show features" |
| 11:23.09 | dlynes_laptop | What isn't? |
| 11:23.12 | dlynes_laptop | *0? |
| 11:23.23 | Mavvie | *0 is "disconnect call" |
| 11:23.24 | dlynes_laptop | Or '*'? |
| 11:23.31 | Mavvie | One Touch Monitor |
| 11:23.31 | Mavvie | Disconnect Call * *0 |
| 11:24.01 | dlynes_laptop | Yeah, cause you have it remapped to '*0' |
| 11:24.07 | dlynes_laptop | In your features.conf file |
| 11:24.39 | Mavvie | yeah, but I'm more worried about the absence of things with regarding to automon |
| 11:24.55 | Mavvie | one touch monitor |
| 11:26.22 | Mavvie | aaaah |
| 11:26.29 | Mavvie | reload chan_features.so doesn't reload the configuration file. |
| 11:30.03 | ThoMe | kann hier wer deutsch? :-) |
| 11:30.31 | ThoMe | if i have incomming call, and the caller is forward to my mailbox, how i can get the call back? |
| 11:30.44 | ThoMe | caller back from the mailbox to my phone? |
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| 11:36.12 | Nobbie | heya =) |
| 11:36.58 | Nobbie | has anyone tried setting up a BLF to indicate on each dynamic queue agent wether they're logged in or not ? |
| 11:37.34 | Mavvie | I wonder if I can set that DYNAMIC_FEATURES in my [globals] section. |
| 11:37.45 | Mavvie | aha |
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| 12:28.46 | Nobbie | has anyone tried setting up a BLF to indicate on each dynamic queue agent, on their ip phone, wether they're logged in to a queue or not ? |
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| 12:45.21 | backblue | hi, anyone using call-limit? |
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| 13:06.06 | zeeesh | hi |
| 13:06.24 | HarryR | Hi |
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| 13:21.02 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 13:25.36 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 13:27.27 | *** join/#asterisk Kernel_Core (i=Kernel_C@217.218.80.135) |
| 13:28.35 | *** join/#asterisk UVSoft (n=UVSoft@84.204.154.31) |
| 13:33.02 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 13:33.35 | UVSoft | Hi. I've got a problem. Asterisk became absolutely silent, it doesn't play the dial/busy/etc tones, however it works properly, so I can phone someone and talk to him... why can it behave so strange? |
| 13:36.03 | UVSoft | Asterisk doesn't need any gsm files, does it? It generates such tones himself, right? |
| 13:36.21 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:36.30 | *** join/#asterisk hyphen (n=hyphen@c-69-136-84-149.hsd1.pa.comcast.net) |
| 13:37.27 | tzafrir | UVSoft, do you see any warnings regarding indications? Do you have indications.conf? |
| 13:39.05 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:39.31 | UVSoft | tzafrir: there's no any warning or errors.... and i have this config |
| 13:39.40 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 13:44.05 | *** join/#asterisk coppice (n=chatzill@118.202.17.210.dyn.pacific.net.hk) |
| 13:46.14 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:46.14 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:46.29 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 13:48.20 | anonymouz666 | when using .call files... how to check the DIALSTATUS? |
| 13:48.23 | anonymouz666 | on busy |
| 13:48.44 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
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| 13:51.43 | *** part/#asterisk ConstyXIV (n=andrew@216.69.48.69) |
| 13:51.52 | *** join/#asterisk Ebola (n=Ebola@host81-151-91-139.range81-151.btcentralplus.com) |
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| 14:04.22 | jojo^ | Isn't the 's' extension suppose to be catch all? How do I make it work with IAX2? (diax windows client to be specific).. Trying to get pattern matching working |
| 14:04.50 | clive- | jojo use _X. |
| 14:06.22 | *** join/#asterisk Gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 14:06.48 | *** join/#asterisk Ryushin (n=Ryushin@windwalker.openinnovations.com) |
| 14:07.22 | orlock | whats the diff between X. and _X.? |
| 14:08.06 | Nivex | X. won't work, _X. will |
| 14:08.17 | Nivex | the _ means pattern match |
| 14:13.30 | Corydon76-home | X. means match a LITERAL "X." |
| 14:14.13 | Ahrimanes | _ turns on pattern matching |
| 14:14.20 | Ahrimanes | d'oh |
| 14:14.22 | Ahrimanes | sorry |
| 14:14.24 | *** join/#asterisk lowlevel (n=Stuart@CPE00145176d140-CM000f9f1e356e.cpe.net.cable.rogers.com) |
| 14:14.34 | anonymouz666 | what happens if I put a dial() inside a while() on dialplan? while() dial() endwhile() |
| 14:14.53 | anonymouz666 | on success |
| 14:17.22 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 14:20.11 | *** join/#asterisk Gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 14:21.04 | codefreeze | anonymouz666: It will dial multiple times? Until a hangup. |
| 14:21.58 | acecase | if i may get by with another unrelated question here. Does anyone know of any activex or other .net control for iax2? google is giving me too much junk to weed through |
| 14:22.57 | anonymouz666 | codefreeze: so I am thinking to put a condition ANSWERED after this Dial() |
| 14:23.20 | anonymouz666 | because this loop will try 5 five times, on sucess, I don't need to try again |
| 14:24.30 | codefreeze | anonymouz666: sounds like a plan. |
| 14:24.45 | jojo^ | clive-, Thanks |
| 14:25.19 | *** join/#asterisk leejohn (n=jsharryp@58.69.36.87) |
| 14:27.04 | leejohn | hi! good day guys, what's the prefered rtptimeout setting i could set any idea? TIA |
| 14:30.13 | evgeni | maybe somebody know how to make that asterisk will send userpassword to as5300 ? |
| 14:33.36 | leejohn | evgeni: register => myusername:mypassword@your_ip_of_as5300 ? is this want you are saying? |
| 14:34.04 | *** join/#asterisk Tili (n=tili@87.219.93.154) |
| 14:36.36 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:36.36 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:45.08 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:46.14 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:46.14 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:47.23 | giasai68 | hello |
| 14:47.45 | giasai68 | can I to have complete dialplan example with ZAP channel? |
| 14:47.47 | giasai68 | thanks |
| 14:54.07 | anonymouz666 | exten => s-BUSY,2,While($[ ${COUNT} < 5 ]) |
| 14:54.08 | anonymouz666 | exten => s-BUSY,3,SET(COUNT=$[${COUNT} + 1]) |
| 14:54.28 | anonymouz666 | do I need to Set(COUNT=1) to work? |
| 14:55.33 | *** join/#asterisk ToyMan (n=Stuart@12.23.30.130) |
| 15:00.16 | giasai68 | is it possible to have complete dialplan example with ZAP channel? |
| 15:02.52 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:05.02 | *** join/#asterisk gerhard7 (n=gerhard@82-171-117-191.dsl.ip.tiscali.nl) |
| 15:13.49 | *** join/#asterisk s1gny (n=s1gny@p54916BD5.dip.t-dialin.net) |
| 15:14.00 | *** part/#asterisk s1gny (n=s1gny@p54916BD5.dip.t-dialin.net) |
| 15:16.29 | *** join/#asterisk Yanik (i=yanik@freeswitch.rocks.yanik.com) |
| 15:16.53 | jontow | any chance anyone has had 1.4 working on freebsd with a sangoma A101? |
| 15:20.21 | brookshire | i've gotten a te110p to work with freebsd :) |
| 15:21.02 | *** join/#asterisk hyphen (n=hyphen@c-69-136-84-149.hsd1.pa.comcast.net) |
| 15:22.03 | nfi|ermes | hi all |
| 15:22.21 | nfi|ermes | where can i download and find documentation about asterisk-gui ? |
| 15:23.15 | in-pt | nfi|ermes: asteriskguru.com for documentation and svn for download |
| 15:23.46 | *** join/#asterisk orkid_ (n=orkid@dataq2.utias.utoronto.ca) |
| 15:23.58 | jontow | brookshire; i have t100p's and they don't :/ |
| 15:24.02 | jontow | <-- sick of gentoo |
| 15:24.03 | *** join/#asterisk angler_ (i=angler@nat/digium/x-bfb1a8b821d23eb3) |
| 15:24.59 | jontow | nothing but HDLC FCS errors/aborts and clicking audio and echo |
| 15:25.31 | jontow | i can get 1.2 to work with the sangoma, but the lack of support from digium regarding their cards and freebsd is discouraging at absolute best |
| 15:25.50 | jontow | and res_smdi would be damned nice, so i don't have to keep using my weak-ass shell script hack to signal MWI to the softswitch ;) |
| 15:26.15 | jontow | i don't really want to backport all of 1.4 to 1.2 to get a compatible zaptel interface |
| 15:26.45 | jontow | (if i do; i get a very distinct "we don't care if you have bugs, you're using a wildly unsupported codebase and configuration" (which indeed is the case) |
| 15:26.56 | *** join/#asterisk chat_jokey (n=chat_jok@202-63-175-78.static.exatt.net) |
| 15:26.58 | chat_jokey | helloo |
| 15:27.10 | jontow | i wouldn't expect support on such a creepy setup :))) |
| 15:27.10 | chat_jokey | anyone knows if there is any way to connect Asterisk using SS7 to transfer SMSs ? .. i have tried libss7 and chan_ss7 .. they both dont work |
| 15:27.44 | jontow | just wish i could deal with linux.. this would all be much less painful.. (i guess) |
| 15:28.11 | evgeni_71 | I'm having issues with my Viatalk sip trunks. outgoing works flawlessly... incoming calls make it in about 1/4 of the time.. Any ideas? |
| 15:29.47 | penguinFunk | snprintf(fileNoExt, 19, "%c%c%c%c%c%", c[0], c[1], c[2], c[3], c[4]); |
| 15:29.53 | penguinFunk | theres got to be an easier way of doing this |
| 15:30.24 | penguinFunk | in python you can do c[0:4] |
| 15:30.43 | penguinFunk | to reference multiple elements in an array |
| 15:31.34 | penguinFunk | wrong channel |
| 15:32.14 | *** part/#asterisk Rhizome (n=Rhizome@81.191.147.145) |
| 15:37.57 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 15:38.00 | *** join/#asterisk andrew` (i=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 15:39.19 | endre | thanks anyway |
| 15:39.32 | endre | i didn't know this multiple reference thing in python |
| 15:39.41 | *** join/#asterisk dasenjo (n=dasenjo@190.24.176.244) |
| 15:44.36 | *** join/#asterisk yansolo90 (n=yann@par69-3-82-224-162-203.fbx.proxad.net) |
| 15:45.50 | *** join/#asterisk chiang_sg (i=kodok@121.7.15.196) |
| 15:45.52 | yansolo90 | hello, anybody knows what default login/password are to ssh Cisco 79XX ? |
| 15:46.09 | in-pt | cisco |
| 15:46.14 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:46.15 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:46.29 | chiang_sg | hi anyone can suggest free sip softphone that can dial 123932@192.168.1.1 directly ? |
| 15:47.02 | in-pt | chiang_sg: sjphone |
| 15:48.09 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 15:48.39 | yansolo90 | i can log me with log/log and debug/debug but i want to be able to reboot the phone... |
| 15:49.23 | in-pt | yansolo90: which cisco phone u have? |
| 15:50.13 | yansolo90 | 7941 |
| 15:50.18 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:50.45 | yansolo90 | and 7961 |
| 15:50.57 | in-pt | do telnet |
| 15:50.57 | Nugget | telnet is eeeeeeevil! |
| 15:51.06 | in-pt | and use cisco password |
| 15:52.22 | yansolo90 | u have 2 login. first is login/password u put in SEP<MAC>.cnf.xml and u have a second login to acces the CLI |
| 15:53.20 | giasai68 | hello |
| 15:54.30 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 15:54.54 | giasai68 | I have configured asterisk with TE205P and making a dialplan, if I try to do a call dont have answer and there is 'status is 'CONGESTION' any help? |
| 15:55.01 | yansolo90 | and you cant telnet the phone with a SIP firmware |
| 15:55.08 | giasai68 | please, give me a feedback, thank you :) |
| 16:02.18 | *** join/#asterisk xpato (n=pato@bart.it-linux.cl) |
| 16:02.18 | xpato | hi anyone using asterisknow? |
| 16:02.47 | *** join/#asterisk CrashSys (n=kumba@208.177.233.66.ptr.us.xo.net) |
| 16:05.23 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 16:05.32 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 16:05.45 | giasai68 | I have configured asterisk with TE205P and making a dialplan, if I try to do a call dont have answer and there is 'status is 'CONGESTION' any help? |
| 16:05.47 | giasai68 | please, give me a feedback, thank you :) |
| 16:07.30 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 16:09.16 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 16:09.32 | vooduhal | Were voicemail distribution lists added to app_voicemail in 1.4? |
| 16:13.16 | Qwell[] | distribution lists? |
| 16:13.27 | Qwell[] | Why not just send the voicemail to multiple mailboxes? |
| 16:16.35 | *** join/#asterisk pirulo (n=andres_p@65.19.28.123) |
| 16:17.24 | lude | man what a weird issue |
| 16:17.47 | jm|work | No manual entry for what |
| 16:17.48 | jm|work | No manual entry for a |
| 16:17.48 | jm|work | No manual entry for weird |
| 16:17.48 | jm|work | Reformatting issue(5), please wait... |
| 16:18.00 | endre | lol |
| 16:18.02 | lude | hahah |
| 16:18.04 | mercestes | ROFL |
| 16:18.08 | mercestes | Supersweet! |
| 16:18.18 | lude | asterisk isn't passing callerid to *just* sip phones from a certain zap group |
| 16:18.37 | mercestes | man jm|work you are a retarded bot. |
| 16:18.40 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:18.40 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:18.44 | jm|work | :( |
| 16:18.52 | vooduhal | Qwell, That works but the user wants to be able to save these groups. We wrote a patch for a very old version of asterisk that we've been running but we would like to upgrade our voicemail server. |
| 16:18.57 | mercestes | aww..I'm sorry. |
| 16:19.00 | mercestes | i take it back |
| 16:19.06 | jm|work | (: |
| 16:19.13 | dasenjo | [OT] Hi! I want to change my asterisk-users mailling lists email address to a gmail one, mailman says me that a confirmation was sent to the address .. but it never arrives |
| 16:19.22 | dasenjo | can someone here help me? |
| 16:20.32 | mercestes | daenjo: Help you what? |
| 16:20.32 | pirulo | with what? |
| 16:21.34 | mut | whats a good place to lookup clec's for an areacode? |
| 16:22.43 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:22.43 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:22.49 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 16:22.54 | Corydon-w | Well, you could purchase a copy of the LERG |
| 16:22.57 | mercestes | mut: lerg tables. |
| 16:23.08 | MrTelephone | hi, can someone direct me to an example dialplan using DID from a pri to route calls to the proper SIP endpoint? |
| 16:23.10 | CrashSys | corydon: Get my message? |
| 16:23.10 | Corydon-w | jinx |
| 16:23.16 | Corydon-w | CrashSys: no |
| 16:23.23 | *** join/#asterisk RoyK (n=roy@213.160.242.90) |
| 16:23.35 | CrashSys | Corydon: the alsa patch doesn't have an effect... |
| 16:23.41 | Corydon-w | Ah |
| 16:23.43 | CrashSys | it still reads from alsa.conf |
| 16:24.13 | Corydon-w | I'll look at it at some point if I have time |
| 16:24.17 | CrashSys | so I made chan_alsa not answer (to get ringing), and made a [page] context that records the page, then issues a system command to play it |
| 16:24.55 | CrashSys | the page isn't "live"... but that will work out fine... |
| 16:25.02 | *** join/#asterisk af_ (n=getsmart@ip-179-53.sn1.eutelia.it) |
| 16:25.02 | CrashSys | specially for the 2-3 phones that are near the horn... |
| 16:25.49 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 16:26.56 | *** join/#asterisk AF-Slash (n=AF-Slash@209-181-28-69.hlna.qwest.net) |
| 16:30.00 | MrTelephone | Is there a variable for DID in asterisk? |
| 16:30.17 | CrashSys | yes, somewhere... |
| 16:30.21 | *** join/#asterisk lba (n=lba@user-12lml5g.cable.mindspring.com) |
| 16:30.39 | mut | um |
| 16:30.46 | mut | does asterisk peer with vonage right? |
| 16:30.58 | Qwell[] | it's just SIP |
| 16:31.07 | mut | i didn't know if they did anything weird to disallow it |
| 16:31.21 | CrashSys | Other then being vonage? |
| 16:31.23 | Qwell[] | Do you have a biz account? |
| 16:31.31 | mut | i dun have an acct at all |
| 16:31.32 | mut | was just asking |
| 16:31.43 | Qwell[] | well, unless you do, it's against their TOS or whatever |
| 16:31.52 | Qwell[] | or, used to be... maybe they've gotten undumb |
| 16:32.10 | yansolo90 | hello, anybody knows what default login/password are to ssh Cisco 79XX ? |
| 16:33.09 | MrTelephone | cisco cisco or something |
| 16:33.22 | CrashSys | cisco/payus |
| 16:33.23 | CrashSys | ? |
| 16:33.31 | *** join/#asterisk Jynger (n=rip@tti.tt.ee) |
| 16:33.47 | cpm | default? default is no password, and ssh isn't enabled. If ssh is enabled, the unit has been configured |
| 16:34.01 | MrTelephone | I can't find many examples of dialplans that include a 24 channel pri |
| 16:34.18 | Jynger | hi |
| 16:34.56 | yansolo90 | no cisco/cisco does'nt work but i'm sure cisco/payus will |
| 16:35.07 | CrashSys | MrTelephone: Dialplan examples? |
| 16:35.24 | wunderkin | MrTelephone, the number is sent as the exten in the context you specify |
| 16:35.41 | CrashSys | not sure a dial-plan example would do you much good... since they're all specific to their usage |
| 16:36.19 | lba | How to insert context B in the middle of a context A and have it execute in-line? |
| 16:36.47 | *** join/#asterisk riddlebox (n=riddlebo@24-207-167-95.dhcp.stls.mo.charter.com) |
| 16:36.52 | lba | lba Other than typing the exten => lines? |
| 16:36.59 | MrTelephone | do phone companies usually send 888 777 3939 10 numbers? |
| 16:37.07 | Corydon-w | lba: include => othercontext |
| 16:37.25 | Jynger | i got a problem with asterisk1.4.0svn trunk, i cannot get no output to rawman query but asterisk/mxml?action=l and manager?action= querys produce correct output |
| 16:37.34 | wunderkin | lba, the order it is typed in is not the order followed |
| 16:37.36 | Corydon-w | MrTelephone: it depends upon what you want them to send |
| 16:37.38 | yansolo90 | <PROTECTED> |
| 16:38.08 | Jynger | does that suppose to be like that? but then gui doesnt work if rawman doesnt produce output reply |
| 16:38.11 | CrashSys | if you cascade the include statements from context to context you can get the context's to go in order specified |
| 16:38.19 | lba | wunderkin: Hello. Is there anyway to have it execute in-line except typing the lines in the first context? |
| 16:38.29 | anonymouz666 | exten => s-BUSY,3,SET(COUNT=$[${COUNT} + 1]) - can I use that without SET(COUNT=1) first ?? |
| 16:38.33 | wunderkin | huh |
| 16:38.36 | lba | wunderkin: These contexts all use n type priorities |
| 16:38.37 | MrTelephone | so if I have 15 zap channels in [from-pstn], I just start putting the extensions in there? I don't have to GotoIf(${DID}=???:Extension) <-- cheap example |
| 16:39.12 | wunderkin | MrTelephone, if they send 4 digits for did, then just exten => 1234,1,blah |
| 16:39.42 | lude | jeez this is weird |
| 16:39.54 | lude | <PROTECTED> |
| 16:39.54 | MrTelephone | ok |
| 16:39.59 | lude | <PROTECTED> |
| 16:40.05 | lude | asterisk knows the caller id |
| 16:40.11 | lude | but check out what it puts in the sip headers |
| 16:40.20 | lude | From: "Unknown" <sip:Unknown@10.1.61.2>;tag=as4cf3d02e |
| 16:40.20 | lude | To: <sip:6096@10.1.61.253:5061;user=phone;transport=udp> |
| 16:40.49 | Corydon-w | ~pb |
| 16:40.51 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 16:40.54 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 16:40.54 | CrashSys | I'm confused |
| 16:40.58 | CrashSys | too much chat |
| 16:41.01 | CrashSys | someone bring me a beer |
| 16:41.05 | lude | i can paste the whole thing in there if someone cares to look |
| 16:41.11 | CrashSys | I think i'm an alchy |
| 16:41.33 | *** join/#asterisk dasenjo_ (n=dasenjo@190.24.179.95) |
| 16:41.45 | lude | http://channels.debian.net/paste/5311 |
| 16:41.59 | CrashSys | beer-cannon... |
| 16:42.04 | CrashSys | I like it |
| 16:42.42 | HarryR | beer balloons :) |
| 16:42.59 | CrashSys | too messy if one pops |
| 16:43.16 | lude | so asterisk does know tha caller id |
| 16:43.18 | HarryR | nah, just the right size to fit in your mouth |
| 16:43.25 | lude | it'll even pass it off to a zap channel |
| 16:43.26 | CrashSys | Hmmm |
| 16:43.32 | lude | it just doesn't include it in the sip headers |
| 16:45.22 | MrTelephone | well i got the motorole sbv520 cable modems working good with asterisk |
| 16:45.33 | *** join/#asterisk hal2385614 (n=chatzill@86.149.54.0) |
| 16:45.41 | MrTelephone | but i need to use embedded transaction requests in the chan_mgcp.c to cut down on traffic |
| 16:45.48 | hal2385614 | hi all - how are you all today?! ;-) |
| 16:45.55 | MrTelephone | right now everytime you press a digit it sends a packet |
| 16:46.14 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:46.14 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:46.43 | hal2385614 | Can someone tell me how to install the fxotune utility from the zap source, please? |
| 16:47.08 | MrTelephone | did you compile it? |
| 16:47.13 | MrTelephone | just move it into /usr/sbin or something |
| 16:48.29 | joe | on polycom phones how does one downgrade sip? |
| 16:48.52 | hal2385614 | It did seem to compile, but when I run it from the tmp directory it complains that Cannot open /etc/fxotune.conf, but there isn't one |
| 16:49.05 | MrTelephone | the sip version? I remember reading once you go to a certain version you can't go back.. better check out polycoms website |
| 16:49.24 | MrTelephone | fxotune is supposed to create that file |
| 16:49.38 | hal2385614 | yes, mrT that's what I thought |
| 16:49.38 | MrTelephone | oh because you have to run it like fxotune -i 4 |
| 16:50.06 | MrTelephone | I'm using polycom 501's and I didn't do the "secure" firmware install because of the warning |
| 16:50.22 | nfi|ermes | if i use chan_mISDN with asterisk 1.4, do i need to install zaptel and libpri too ? |
| 16:50.29 | MrTelephone | and I'm having a bitch of a time getting ntp working with the 501's so I had to install sntp |
| 16:51.15 | MrTelephone | you can tell asterisk is putting a dent in commercial pbx systems.. a lot of bad articles about how hard it is to configure and setup |
| 16:51.40 | MrTelephone | If I get my echo issues ironed out (when my pri is installed) I beleive its the best thing ever |
| 16:52.26 | *** join/#asterisk critch (n=critch@steven.basesys.com) |
| 16:52.30 | anonymouz666 | answering my own question if you don't initialize a variable the value will be 0. |
| 16:52.43 | hal2385614 | I expected a bit more output from fxotune (I thought there was last time I ran it a couple of years ago), It just says fxotune: successfully set echo coeffecients on FXO modules |
| 16:52.50 | hal2385614 | I hope that has worked |
| 16:53.05 | anonymouz666 | do that with an int using C language... |
| 16:53.43 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 16:54.23 | hal2385614 | has anyone tried attaching a fax machine to an FXO port? I am finding the sound quality of the sender's fax through it is not very good. Does anyone have any ideas about how I could improve it? |
| 16:55.22 | jm|work | hal2385614: which codec? |
| 16:56.07 | hal2385614 | jm - there is no codec, it's just straight from the FXS port to the FXO port |
| 16:56.23 | hal2385614 | I just want it to pass it straight through |
| 16:57.29 | *** join/#asterisk angler_ (i=angler@nat/digium/x-cf05be3f5e5007df) |
| 16:57.43 | CrashSys | MrTelephone: What problems you having with NTP? |
| 17:02.11 | *** join/#asterisk Lann (i=Lanniste@adsl-63-200-88-82.dsl.scrm01.pacbell.net) |
| 17:05.29 | *** join/#asterisk dasenjo_ (n=dasenjo@190.24.177.243) |
| 17:09.15 | hal2385614 | am I correct that when a call comes into asterisk via an FXS interface, and is routed to dial a traditional phone on an FXO interface, there is no codec that can be changed or configured to improve the sound quality? |
| 17:09.55 | critch | corect, no codec involved |
| 17:10.17 | critch | you probably need to consider wiring problems |
| 17:11.19 | brad_mssw | or get better quality FXO/FXS ports ... try messing with the gain levels, echo cancellation, etc |
| 17:11.42 | critch | echo cancel should be thwarted by the fax call |
| 17:12.37 | anonymouz666 | http://www.pastebin.ca/345584 - where in this macro I can set COUNT, I think the way is configured today won't work COUNT will always have value 1 before while. |
| 17:13.44 | anonymouz666 | I am stuck on that |
| 17:13.55 | *** part/#asterisk pythos (i=lanebob@unaffiliated/pythos) |
| 17:15.09 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:15.09 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:15.37 | *** join/#asterisk UlbabraB (n=salama@81.72.43.241) |
| 17:17.17 | hal2385614 | tank you critch - I have just called the fax port using a phone, and the sound is very clear, however, when a fax calls it, it sounds "gurgly" |
| 17:19.02 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 17:19.03 | *** join/#asterisk Gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 17:19.36 | critch | unfortunately, most of my asterisk knowledge is old-skool compared to what is current. |
| 17:20.13 | HarryR | I'm the same with windows :\ |
| 17:20.38 | HarryR | which is why I don't do desktop support anymore ;) "Win.ini, wtf is that!" |
| 17:21.00 | CrashSys | Yeah, my aeroglass interface isn't coming up... |
| 17:21.03 | CrashSys | wtf is that? |
| 17:21.14 | HarryR | aeroglass? |
| 17:21.19 | HarryR | oh... vista |
| 17:21.43 | CrashSys | I dont even know what it is |
| 17:21.46 | CrashSys | other then the name |
| 17:22.11 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 17:22.30 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 17:22.51 | CrashSys | Although I do enjoy the Mac vs. PC commercials... |
| 17:22.55 | CrashSys | they're entertaining... |
| 17:23.12 | HarryR | have you been to the mac stores? |
| 17:23.43 | HarryR | I went to one on regent street or something, very very nice, but all the staff were airheads |
| 17:24.12 | CrashSys | They can play iTunes and surf the web! |
| 17:24.15 | *** part/#asterisk jp_away (n=jpablo@linuxuanl.org) |
| 17:24.34 | CrashSys | I've been to a few "geeksquad" places to pick up some people's computer and they aren't much better... |
| 17:24.48 | lba | Is there any way to force a macro to "return" to the line after it was called in an exten? |
| 17:25.41 | giasai68 | HELLO |
| 17:25.44 | CrashSys | lba: set the priority number when calling the macro (pass it as a variable), and script yer macro with an exit to return to the context,priority+1 |
| 17:26.13 | lba | CrashHD: Is 'return' an Asterisk command? |
| 17:26.24 | CrashSys | lba: no, it's how you script things |
| 17:26.46 | *** join/#asterisk w0ls0n (n=Me@43-141-135-64.dsl.sacoriver.net) |
| 17:26.58 | lba | CrashHD: The 'exit' command? I just want the name so I can look up how the command works in a macro |
| 17:27.01 | w0ls0n | hi all |
| 17:27.11 | CrashSys | lba: there is no return or exit command |
| 17:27.24 | Qwell[] | CrashHD: there is in ael2 I believe |
| 17:27.38 | w0ls0n | I have a asterisk box setup and working. I need a card that will take 2 analog phone jacks. |
| 17:27.51 | Qwell[] | w0ls0n: tdm400p |
| 17:27.52 | w0ls0n | I can't seem to find anything on http://www.digium.com |
| 17:27.53 | lba | CrashHD: OK. How do I specify a macro should return to a certain line in the calling context? |
| 17:28.18 | w0ls0n | I was looking at that one |
| 17:28.27 | w0ls0n | but those look like ethernet jacks |
| 17:28.34 | Qwell[] | w0ls0n: it's analog :) |
| 17:28.42 | w0ls0n | oh ok |
| 17:28.43 | lba | CrashSys: OK. How to I specify that the macro should return to the calling context at a certain priority? |
| 17:28.45 | Qwell[] | the older cards did use RJ45, but RJ11 works in an RJ45 jack just fine |
| 17:28.53 | w0ls0n | anyone know if that comes with freebsd drivers |
| 17:29.20 | Qwell[] | zaptel support on freebsd is fairly minimal |
| 17:29.23 | Qwell[] | and unsupported |
| 17:29.29 | CrashSys | lba: You program it to do so |
| 17:29.32 | critch | anyone know how to read documentation for w0ls0n? |
| 17:29.50 | Qwell[] | critch: his questions are fair |
| 17:30.06 | Qwell[] | he did his research - it's just that the conclusion he came to was incorrect |
| 17:30.20 | lba | CrashSys: Do you think it's possible to specify the returning priority as n+1? I use most n as priorities in dialplan |
| 17:30.37 | anonymouz666 | can use macro dial inside macro dial ? |
| 17:30.39 | anonymouz666 | lool |
| 17:30.45 | Qwell[] | anonymouz666: "macro dial"? |
| 17:30.46 | w0ls0n | :-( |
| 17:30.47 | evgeni | Maybe someone know ? Is it possible to authorize cisco with radius by sip userid and password ? |
| 17:30.54 | Qwell[] | w0ls0n: you're fine :) |
| 17:30.56 | giasai68 | I have a dial problem: exten => s,1,Dial(Zap/G1) how I can dial to my equipement the same number that I have calling? E.G. if I call with SJphone my asterisk machine with number XXXXX if I write: exten => s,1,Dial(Zap/G1/XXXX) all work fine, if I dont specified XXXX number in dialplan dont work. Any help? |
| 17:31.03 | w0ls0n | well had I found documentation, I wouldn't have asked :-) |
| 17:31.09 | anonymouz666 | Qwell recursive |
| 17:31.14 | CrashSys | lba: You would have a line like this Macro(macro,arg1,arg2,arg3) |
| 17:31.16 | Qwell[] | anonymouz666: You don't "dial" a macro |
| 17:31.28 | critch | w0ls0n: some of that was just teasing you |
| 17:31.30 | anonymouz666 | [macro-dial] |
| 17:31.35 | anonymouz666 | :D |
| 17:31.37 | Qwell[] | w0ls0n: hmm, you're right - they do look like ethernet jacks |
| 17:31.38 | w0ls0n | die |
| 17:31.56 | Qwell[] | anonymouz666: yes, you can call a macro from within a macro - up to like 7 levels deep |
| 17:31.57 | w0ls0n | is it the leds throwing me off? |
| 17:32.18 | Qwell[] | w0ls0n: no, we need to update the image, since afaik, it doesn't use RJ45 jacks anymore ;) |
| 17:32.33 | lba | CrashSys: Thanks for your help. I'll work on this 'returning macros |
| 17:32.38 | w0ls0n | ahhhh |
| 17:33.01 | w0ls0n | well what is the recommended distro to run asterisk on? |
| 17:33.08 | Qwell[] | w0ls0n: whatever you know best |
| 17:33.15 | w0ls0n | I know BSD best :-) |
| 17:33.26 | Juggie | CentOS! :P |
| 17:33.27 | mut | uh |
| 17:33.28 | HarryR | portage is your friend :) |
| 17:33.33 | critch | BSD isn't a distro |
| 17:33.37 | mut | they just look like it cause its a tall slot |
| 17:33.40 | Qwell[] | ~kill Juggie |
| 17:33.41 | jbot | ACTION shoots a inverse pseudotachyon gun at Juggie |
| 17:33.41 | w0ls0n | FreeBSD that is |
| 17:33.48 | mut | its a 4 pin thing so it looks bigger |
| 17:33.52 | mut | wider |
| 17:33.58 | mut | mine look the exact same |
| 17:33.59 | HarryR | You could try asterisk on freebsd |
| 17:34.09 | HarryR | just don't expect support for tdm cards & stuff |
| 17:34.10 | w0ls0n | it works |
| 17:34.18 | w0ls0n | yea thats what I figured |
| 17:34.46 | anonymouz666 | Qwell http://www.pastebin.ca/345584 my problem is with set(count=1) so the while loop never will reach endwhile() |
| 17:35.39 | anonymouz666 | I think the Set(COUNT=1) must be elsewhere |
| 17:36.42 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:38.10 | giasai68 | I have a dial problem: exten => s,1,Dial(Zap/G1) how I can dial to my equipement the same number that I have calling? E.G. if I call with SJphone my asterisk machine with number XXXXX if I write: exten => s,1,Dial(Zap/G1/XXXX) all work fine, if I dont specified XXXX number in dialplan dont work. Any help? |
| 17:38.17 | giasai68 | please, give me a feedback |
| 17:38.18 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 17:38.58 | critch | giasai68: exten => _X.,1,Dial(Zap/G2/${EXTEN}) |
| 17:39.12 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:39.46 | giasai68 | thanks, I'll try |
| 17:39.47 | giasai68 | :) |
| 17:40.44 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:40.44 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:42.43 | critch | Qwell[]: I think they can get chanserv to remove the +b |
| 17:43.04 | Qwell[] | sure, that's easy, but I wonder if they're completely immune to it |
| 17:44.21 | critch | ircops, as in freenode operators, or members in the channel with +o? |
| 17:44.25 | Qwell[] | freenode opers |
| 17:44.49 | critch | I don't know |
| 17:45.37 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:45.37 | *** mode/#asterisk [+b Bhaal!bhaal@freenode/staff/bhaal] by Qwell[] |
| 17:45.43 | Qwell[] | let's find out |
| 17:46.03 | Juggie | hats +b? |
| 17:46.08 | Juggie | *whats |
| 17:46.09 | critch | ban |
| 17:46.14 | Qwell[] | Juggie: ircnub |
| 17:46.22 | Juggie | oh jeeze... |
| 17:46.23 | Juggie | nm,. |
| 17:46.26 | w0ls0n | is the Asterisk: The Future of Telephony by O'Reilly really a good book to get? I got a SpamAssassin book by O'Reilley and it sux0red |
| 17:46.37 | Juggie | temporary brain fart. |
| 17:46.37 | Corydon-w | Qwell[]: not exactly a way to make friends |
| 17:46.40 | Qwell[] | w0ls0n: yes, it's very good |
| 17:46.47 | Qwell[] | Corydon-w: he's spammy, and it's annoying |
| 17:47.03 | Qwell[] | I've asked him to stop several times. I've asked another oper to remove him/ask him to stop. |
| 17:47.08 | Corydon-w | He is? |
| 17:47.09 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 17:47.19 | Qwell[] | he does a part/join every hour or so |
| 17:47.27 | Qwell[] | a /cycle probably |
| 17:47.38 | Juggie | so? :) |
| 17:47.41 | *** join/#asterisk RoyK (n=roy@ti211310a080-5551.bb.online.no) |
| 17:47.50 | Qwell[] | it's more spammy than what he's trying to stop |
| 17:48.00 | Qwell[] | it completely defeats the purpose |
| 17:48.09 | [TK]D-Fender | w0ls0n: Best book out there to date. |
| 17:48.18 | w0ls0n | ok |
| 17:48.59 | Corydon-w | Qwell[]: it defeats the purpose of getting rid of bots that spam on join? |
| 17:49.05 | Qwell[] | yes |
| 17:49.08 | CrashSys | I dont own any software books |
| 17:49.13 | CrashSys | prolly explains why I suck as an admin |
| 17:49.14 | Qwell[] | one bot per month sends one line of text |
| 17:49.19 | [TK]D-Fender | w0ls0n: If you don't like it you can return the freely downloadable PDF for a full refund... |
| 17:49.27 | Corydon-w | Qwell[]: sure, NOW |
| 17:49.31 | Qwell[] | an oper who /cycle's every HOUR, is like...3600 times more annoying |
| 17:49.46 | Qwell[] | off by a factor of 10 there... |
| 17:49.53 | Qwell[] | or 1.. I give up |
| 17:49.54 | cpatry | <PROTECTED> |
| 17:50.06 | Qwell[] | Juggie: :p |
| 17:50.11 | Corydon-w | Qwell[]: 720 |
| 17:50.14 | Juggie | Qwell, relax :) |
| 17:50.20 | Qwell[] | Corydon-w: true, two lines |
| 17:50.28 | Qwell[] | so yeah |
| 17:50.39 | w0ls0n | thanks all |
| 17:50.42 | [TK]D-Fender | cpatry: Your nick looks a little off there :) |
| 17:50.42 | CrashSys | or wrote c-code |
| 17:51.14 | cpatry | [TK]D-Fender: i know, JunK-Y is @home |
| 17:51.56 | [TK]D-Fender | cpatry: Ever consider Reg-ing JunK-Y[notathome] ? :) |
| 17:52.00 | Juggie | by day he is 'claude'! :) |
| 17:52.36 | cpatry | im like neo and mr.anderson. :P |
| 17:52.58 | *** part/#asterisk Navman (n=Navman@62.108.206.82) |
| 17:53.13 | CrashSys | The matrix needed a fat-guy doing wire-fu |
| 17:53.18 | CrashSys | like me |
| 17:53.38 | Corydon-w | On IRC, everyone is thin |
| 17:54.00 | CrashSys | i FORGOT |
| 17:54.13 | CrashSys | damn caps |
| 17:54.14 | cpatry | Corydon-w: you scare too much ppl, they are all running away! |
| 17:54.33 | CrashSys | on the internet everyone's a porn-star/model |
| 17:55.31 | Corydon-w | cpatry: people aren't scared of me; they're scared of the TRUTH |
| 17:55.37 | critch | just because some one is a pornstar doesn't mean you want to see it |
| 17:56.48 | Juggie | eugh. |
| 17:56.59 | Juggie | this conversation is going in the wrong direction |
| 17:57.08 | HarryR | critch, a perfect example would be "My mum used to be a porn star" |
| 17:57.12 | [TK]D-Fender | Corydon : Does it bear a remarkable likeness? ;) |
| 17:57.15 | w0ls0n | pix please? |
| 17:57.16 | w0ls0n | LOL |
| 17:58.32 | Corydon-w | [TK]D-Fender: dunno, do you like gay porn? |
| 17:58.41 | anonymouz666 | haha |
| 17:59.18 | CrashSys | ... |
| 17:59.27 | anonymouz666 | my dialplan in a forever loop and you are talking about pron stuff :D |
| 17:59.39 | [TK]D-Fender | Corydon-w: Don't know, have never seen. Being that I'm most assuredly heterosexual means I'm likely to never know :) |
| 17:59.49 | Corydon-w | angler_: usually that's a way to end the topic |
| 17:59.56 | [TK]D-Fender | Corydon-w: (Not falling for the obvious baited question) :D |
| 17:59.59 | Corydon-w | err, anonymouz666 |
| 18:00.10 | cpatry | with MOH, the time diff between each packet is huge, is there any way to stabilize it? |
| 18:00.51 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 18:00.58 | [TK]D-Fender | Corydon-w: Just joking around and all, and my comment was in your resemblance to the "truth" you are the carrier of... |
| 18:01.16 | danp | [TK]D-Fender: how did 2.1.0 go? |
| 18:01.24 | Corydon-w | [TK]D-Fender: oh, that. That was a reference to a movie |
| 18:01.27 | [TK]D-Fender | cpatry: 1-800-SHIP-UPS ! |
| 18:01.43 | [TK]D-Fender | Corydon-w: Doesn't ring a bell. Which one? |
| 18:01.53 | *** part/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 18:01.55 | Corydon-w | [TK]D-Fender: A Few Good Men |
| 18:01.59 | [TK]D-Fender | danp: Works great, as does the IP 501 MicroBrowser |
| 18:02.06 | cpatry | [TK]D-Fender: and seriously? |
| 18:02.11 | danp | cool |
| 18:02.16 | danp | what are you using the MB for? |
| 18:02.18 | [TK]D-Fender | Corydon-w: Only saw it once, shortly after it hit video. I should revisit it in my collection. |
| 18:02.24 | danp | that's something i'd like to explore |
| 18:02.41 | wunderkin | gay text porn |
| 18:03.07 | Corydon-w | [TK]D-Fender: Jack Nicholson, screaming "You can't handle the TRUTH" |
| 18:03.24 | critch | when it is text, it is called literotica, unless it is ascii pron |
| 18:03.27 | [TK]D-Fender | cpatry: Check your MP3's (assuming thats what you're using) forID3 tags, etc.... then barring that try another tech (Native vs MPG123), and failing even that, re-encode them as a more native codec and go Native MoH |
| 18:03.46 | [TK]D-Fender | Corydon-w: Ah, paraphrased, not verbatim. I do remember that line..... |
| 18:03.53 | cpatry | its native and ulaw |
| 18:04.04 | [TK]D-Fender | cpatry: And only MoH sucks? |
| 18:04.16 | [TK]D-Fender | cpatry: and the overall call is ULAW? |
| 18:04.24 | cpatry | everything is alaw ya |
| 18:04.28 | cpatry | ulaw sorry |
| 18:04.33 | [TK]D-Fender | danp: At home, nothing yet. Will have to come up with some stuff. |
| 18:04.37 | *** join/#asterisk lorinc (n=ang@caracas-0648.adsl.interware.hu) |
| 18:05.01 | giasai68 | hello |
| 18:05.02 | [TK]D-Fender | danp: Were I to reinstall all of my home-automation gear I'd assuredly to an interface for that. |
| 18:05.25 | [TK]D-Fender | cpatry: Is it shitty on all phones? |
| 18:05.31 | cpatry | ya |
| 18:05.35 | giasai68 | when I receive a call and hang up appear this worning: WARNING[30055]: src/chan_h323.c:977 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_86 and call is terminate |
| 18:05.48 | giasai68 | please, give me any feedback... thanks |
| 18:06.09 | cpatry | i ran ethereal, for a playback its almost standard at 20ms, but MOH has a lot of variation. |
| 18:06.30 | [TK]D-Fender | cpatry: Give another MoH a try. |
| 18:06.33 | Qwell[] | cpatry: you don't have VAD on the other end? |
| 18:06.56 | cpatry | another, like? |
| 18:06.58 | cpatry | nope |
| 18:07.00 | *** join/#asterisk pirulo (n=andres_p@65.19.28.123) |
| 18:07.14 | Qwell[] | cpatry: Do you see RTP coming back from the other end? |
| 18:07.25 | cpatry | wait |
| 18:07.43 | cpatry | i close my window, i will reproduce it in 10 minutes. |
| 18:08.53 | [TK]D-Fender | cpatry: MPG123, Madplay, etc.... |
| 18:09.11 | giasai68 | when I receive a call and hang up appear this worning: WARNING[30055]: src/chan_h323.c:977 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_86 and call is terminate |
| 18:09.13 | giasai68 | please, give me any feedback... thanks |
| 18:10.22 | [TK]D-Fender | giasai68: We heard you the first time, and if somebody knew, they'd answer you |
| 18:11.05 | cpatry | [TK]D-Fender: im using native and ur telling me to use mpg123? stop smokin' dude :) |
| 18:11.45 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 18:13.48 | cpatry | tk: which polycom firmware has the microbrowser again? |
| 18:13.51 | anonymouz666 | why ${DIALSTATUS} is always 0 using call files to dial out ???? |
| 18:13.57 | dlynes_laptop | cpatry, I would still try [TK]D-Fender's suggestion, and also try different files |
| 18:14.06 | cpatry | ok |
| 18:14.13 | dlynes_laptop | cpatry, especially try the default moh files for asterisk...make sure those ones work |
| 18:14.28 | dlynes_laptop | cpatry, it might be your custom moh files |
| 18:14.51 | cpatry | dlynes_laptop: its the original files, nothing customized so far. |
| 18:14.58 | dlynes_laptop | cpatry, one of my customers is using custom files and their autoattendant files sound like crap |
| 18:15.13 | dlynes_laptop | cpatry, they had a professional record them, and the volume is way too high on them |
| 18:15.21 | dlynes_laptop | cpatry, so of course they're distorted like crap |
| 18:15.23 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 18:15.50 | dlynes_laptop | cpatry, yeah...try a different moh method then |
| 18:15.56 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 18:16.00 | dlynes_laptop | cpatry, what cpu are you using? |
| 18:16.29 | dlynes_laptop | cpatry, and are you using the 1000Hz timer in the kernel? |
| 18:16.32 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 18:17.05 | Qwell[] | cpatry: make sure that you're getting rtp back from the other side. |
| 18:18.05 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:18.06 | dlynes_laptop | Qwell[], he said he was having variations in packets, not one way audio for moh |
| 18:18.07 | giasai68 | when I receive a call and hang up appear this worning: WARNING[30055]: src/chan_h323.c:977 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_86 and call is terminate |
| 18:18.08 | giasai68 | please, give me any feedback... thanks |
| 18:18.16 | Qwell[] | dlynes_laptop: I'm thinking vad |
| 18:18.19 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 18:18.27 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 18:18.28 | Qwell[] | which would possibly explain the "choppiness" |
| 18:18.33 | cpatry | Qwell[]: from packet 426 to packet 470, no RTP is comin back. |
| 18:18.36 | dlynes_laptop | Qwell[], ah...but asterisk doesn't support vad |
| 18:18.41 | Qwell[] | dlynes_laptop: EXACTLY |
| 18:18.50 | Qwell[] | which is why it needs to be turned off on the remote end |
| 18:18.53 | dlynes_laptop | Qwell[], ah...didn't think that would affect anything, then |
| 18:19.04 | Qwell[] | dlynes_laptop: we time outgoing packets based on incoming packets |
| 18:19.11 | Qwell[] | if there are no incoming packets... |
| 18:19.26 | dlynes_laptop | Qwell[], asterisk usually just silently ignores it for me, but i always disable it on all my clients, because i hate the spam in the cli window when it happens |
| 18:19.41 | Qwell[] | cpatry: If you blow into the phone on the other end, you may start getting rtp flowing "properly" again |
| 18:19.47 | *** join/#asterisk hal2385614 (n=chatzill@86.149.48.4) |
| 18:19.47 | [[blah]asfd | i am having complaints of users who are taking calls over sip. the calls work just fine. they can hear the person they are talking to and the person can hear them. but if they transfer to another station, the person who picks up the transfer cannot hear the caller, but the caller can hear them. what could be causing this? |
| 18:19.50 | Qwell[] | like, try blowing into it for 5-10 seconds |
| 18:20.09 | dlynes_laptop | Qwell[], sounds kinky....giving your phone a blow |
| 18:20.12 | [TK]D-Fender | cpatry: SIP 2.1.0 (brand new) |
| 18:20.20 | dlynes_laptop | things that make you go hmmmmmm |
| 18:20.21 | *** join/#asterisk remiss (i=bofh@151.80-203-38.nextgentel.com) |
| 18:24.18 | Jynger | can somebody help me with a asterisk manager interface problem ? |
| 18:24.30 | [[blah]asfd | any idea on why that would happen with my sip calls? |
| 18:24.46 | cpatry | Qwell[]: it works great now, it has something related to VAD, thanks bro. |
| 18:24.53 | Qwell[] | cpatry: np |
| 18:25.15 | cpatry | Qwell[]: but what about if they have VAD turn on? |
| 18:25.26 | Qwell[] | then you'll get what you just described - or worse |
| 18:25.51 | cpatry | is there any tweak we can do to get it work with VAD? |
| 18:25.58 | [[blah]asfd | anyone ever experienced that? where after a tranfer, only half of the call is there? |
| 18:26.45 | Qwell[] | cpatry: I think you can use the zap timing thing |
| 18:27.00 | Qwell[] | I don't recall the option, but you can time packets with zap or something |
| 18:28.56 | *** part/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 18:29.45 | giasai68 | hello, when I receive a call with asterisk 1.4 (trougth a gateway) and hungup phone call is terminate |
| 18:29.58 | giasai68 | please, give me a help |
| 18:30.00 | giasai68 | thank ypou |
| 18:30.20 | remiss | hehe |
| 18:30.35 | Qwell[] | Chapter 12, Page 346, Paragraph 3: "When you hangup a call - the call gets hung up." |
| 18:31.13 | Qwell[] | giasai68: Unless you've described the problem incorrectly...I see no issue |
| 18:31.21 | w0ls0n | heh |
| 18:31.36 | critch | you must be on an old version, should be "When you hangup a 2 party call - the call gets hung up" |
| 18:31.53 | Qwell[] | critch: ahh, didn't realize they released a new book with the errata |
| 18:32.21 | Qwell[] | critch: 3 party call, with the hanguper in the middle |
| 18:32.30 | giasai68 | Qwell[]: when I hangup call after 1 or 2 second call is autoterminate, why? |
| 18:32.32 | Qwell[] | that would also be a valid call termination |
| 18:32.53 | critch | Well, the editor of the original book didn't think about anything more complex than a simple call |
| 18:33.04 | Qwell[] | critch: silly editors |
| 18:33.16 | Qwell[] | I bet they used emacs in rev 1 |
| 18:33.24 | [TK]D-Fender | Qwell : tautology :) |
| 18:33.47 | giasai68 | Qwell[]: I can paste log call, if u read it |
| 18:34.43 | remiss | wow |
| 18:34.54 | Qwell[] | giasai68: I really still don't see an issue |
| 18:35.10 | clyrrad | remiss: its your dial plan you should be worried about :p |
| 18:35.33 | giasai68 | Qwell[]: call is autoterminate when I hangup phone |
| 18:35.40 | remiss | asterisk answered when I called it.. so good so far.. *try incoming call* |
| 18:35.44 | Qwell[] | giasai68: yes, that is supposed to happen |
| 18:36.19 | critch | giasai68: what are you expecting to happen that you don't seem to be observing? |
| 18:36.38 | giasai68 | Qwell[]: I need talk when hangup phone |
| 18:36.44 | Qwell[] | talk to who? |
| 18:37.08 | *** join/#asterisk Chris-NB (n=chris@argos.campus-sbg.at) |
| 18:37.12 | clyrrad | to Record() ? |
| 18:37.46 | remiss | clyrrad: should I replace the dialplan? |
| 18:37.53 | remiss | seems to work ok |
| 18:37.54 | Qwell[] | remiss: read UPGRADE.txt |
| 18:38.02 | *** join/#asterisk jpe-nyc (n=jpe@p77-37.acedsl.com) |
| 18:38.04 | Jynger | please help, raw asterisk manager interface does not reply to querys... |
| 18:38.09 | clyrrad | remiss: alot has changed, lots has been added, depreciated etc... |
| 18:38.39 | remiss | Qwell[]: I was going to -- but then I saw how long it was :p |
| 18:38.43 | clyrrad | remiss: if you see the change log - its huge! |
| 18:38.47 | giasai68 | Qwell[]: I'm calling with IPPhone my asterisk machine trougth PRI card call is forward to gsmgateway. The gateway dial number, when otherside hungup phone call is aoutoterminate and there is any conversation |
| 18:39.07 | Qwell[] | giasai68: if the other end hangs up, who do you expect to be talking to? |
| 18:39.30 | giasai68 | Qwell[]: yes |
| 18:39.40 | Qwell[] | That...wasn't a yes/no question |
| 18:39.51 | clyrrad | lol |
| 18:40.04 | giasai68 | Qwell[]: whats do u mean? |
| 18:40.14 | Qwell[] | [TK]D-Fender: cue |
| 18:40.16 | mafkees | If I pickup my phone I hear a dialtone, can someone help with that ? |
| 18:40.37 | clyrrad | mafkees: what is the issue with that? |
| 18:40.46 | Qwell[] | clyrrad: He's trolling. ;P |
| 18:40.50 | mafkees | ;) |
| 18:40.52 | *** join/#asterisk Opperior (n=chatzill@24.61.165.73) |
| 18:40.55 | clyrrad | ah thank gawd! |
| 18:41.01 | mafkees | sorry |
| 18:41.06 | mafkees | couldn't resist |
| 18:41.07 | J4k3 | mafkees: call my local telco named "Windstream", they're the experts in removing dialtone from active phone lines. |
| 18:41.09 | clyrrad | hahaha |
| 18:41.11 | J4k3 | ... and the ability to ring, too. |
| 18:41.21 | giasai68 | Qwell[]: u still there? |
| 18:41.29 | Qwell[] | giasai68: no, I left, sorry |
| 18:41.29 | mafkees | J4k3: hahahaha |
| 18:41.40 | clyrrad | LOL |
| 18:41.50 | mafkees | actually, that's not funny if you want to place a call |
| 18:41.58 | remiss | will extensions reload tell me if I'm using something deprecated/removed? |
| 18:42.11 | remiss | ugh.. it's deprecated :p |
| 18:42.13 | mafkees | remiss: some of it will be reported |
| 18:42.14 | clyrrad | remiss: usually you will see some error or warning |
| 18:42.16 | Qwell[] | remiss: :P |
| 18:42.22 | Qwell[] | remiss: the irony there is...nice |
| 18:42.30 | mafkees | some of it will be notified when you go through the dialplan |
| 18:42.44 | remiss | first i did reload extensions -- that's deprecated.. ok I go extensions reload.. that's deprecated :p |
| 18:42.50 | clyrrad | remiss: you really need to look at the change log |
| 18:42.53 | Qwell[] | dialplan reload |
| 18:43.12 | remiss | clyrrad: ok.. will do |
| 18:43.24 | mafkees | yeah, I was puzzled with the new cli as well |
| 18:43.36 | clyrrad | I am waiting to upgrade |
| 18:43.40 | mafkees | first thing I always do is: set verbose 255 |
| 18:43.49 | mafkees | naw that did work great in 1.4 ;) |
| 18:44.12 | clyrrad | I will upgrade when its 1.4.2 |
| 18:44.40 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 18:44.54 | remiss | this asterisk-gui thing.. is that some sort of web-interface to asterisk or what? |
| 18:45.43 | mafkees | yeah |
| 18:45.56 | clyrrad | GUI's are evil |
| 18:45.57 | *** join/#asterisk shodan (n=shodan@ip047.96-113-216.pppoe1.joliette.intermonde.net) |
| 18:46.07 | mafkees | yeah |
| 18:46.13 | remiss | clyrrad: yes, yes.. but it's great to show off |
| 18:46.17 | mafkees | use echo and sed to edit configfiles |
| 18:46.25 | remiss | hehe |
| 18:46.39 | clyrrad | remiss: nah that has zero coolness factor |
| 18:46.51 | remiss | :-/ |
| 18:46.54 | clyrrad | show me a nice script with sed etc - thats great to show off |
| 18:47.24 | shmaltz | anyone in central New Jersy looking for a job msg me |
| 18:48.37 | w0ls0n | what kind of job |
| 18:48.42 | w0ls0n | I know someone looking for one |
| 18:49.04 | shmaltz | w0ls0n, managin and installing configuring PBXs |
| 18:49.08 | w0ls0n | and he has a good solid asterisk background but I'm not sure if hes willing to move yet |
| 18:49.23 | shmaltz | w0ls0n, don't want him to move |
| 18:49.26 | mafkees | w0ls0n: I did not know you knew me |
| 18:49.30 | shmaltz | did you see my request? |
| 18:49.39 | shmaltz | anyone in central New Jersy |
| 18:49.45 | w0ls0n | oh sorry |
| 18:49.51 | w0ls0n | hehe |
| 18:50.25 | Opperior | bah, remote office |
| 18:52.02 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 18:52.26 | *** join/#asterisk trixman (n=andy@rrcs-67-53-168-147.west.biz.rr.com) |
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| 18:58.57 | *** part/#asterisk cristiandavidgm (n=CDGM@201.236.251.105) |
| 19:05.10 | CrashSys | Dont suppose anyone is aware of a way to get timestamps on the console? :) |
| 19:05.19 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 19:05.29 | CrashSys | Guess I coult put no-op's in the dialplan |
| 19:05.57 | CrashSys | or something |
| 19:06.15 | joelsolanki | how many simulteneous calls does P4 with 1 GB SIP g729 calls run on asterisk ? |
| 19:06.44 | CrashSys | 250 channels (ballpark) |
| 19:07.13 | joelsolanki | at a time 250 channels g729 SIP ? |
| 19:07.27 | CrashSys | dunno about G729... |
| 19:07.38 | CrashSys | but an asterisk server will handle 250-ish channels |
| 19:07.51 | CrashSys | you get around the 300-mark and funny things happen |
| 19:08.05 | CrashSys | So it depends on what your doing... |
| 19:08.06 | joelsolanki | hmm ok. |
| 19:08.35 | CrashSys | Launch like 10 calls on the platform, see how much CPU it's burning for your G729, and use that as a metric... |
| 19:08.49 | critch | joelsolanki: if you are talking SIP to SIP, asterisk would try and get out of the loop, and can handle many more calls |
| 19:09.13 | CrashSys | Yeah, i'm talking zap to zip |
| 19:09.17 | CrashSys | err zap to sip |
| 19:09.33 | critch | I want chan_zip, care to share? |
| 19:09.47 | CrashSys | sure |
| 19:09.53 | CrashSys | call PKWare |
| 19:09.59 | CrashSys | tell them you want chan_zip |
| 19:10.16 | critch | ahh but that is chan_pkzip |
| 19:10.27 | CrashSys | :( |
| 19:10.34 | critch | what about bzip2? |
| 19:12.07 | joelsolanki | yes SIP to SIP ? how many calls ? |
| 19:12.25 | *** join/#asterisk Jared_Leto (n=Lostprop@80-89-104-241.DSL.ycn.com) |
| 19:12.28 | critch | joelsolanki: depends on many variables outside of what you have given us |
| 19:12.38 | joelsolanki | hmm ok |
| 19:12.56 | critch | joelsolanki: if calls can be sent from endpoint to endpoint, then asterisk isn't involved shortly after call setup |
| 19:13.05 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 19:13.52 | joelsolanki | it connects to our parent service provider |
| 19:15.06 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-59-250.dsl.tul2ok.sbcglobal.net) |
| 19:15.12 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
| 19:16.27 | critch | joelsolanki: do you have enough bandwidth to transport 250 channels to your parent service provider? |
| 19:16.37 | *** join/#asterisk karmatronic (n=karmatro@84.77.153.132) |
| 19:18.01 | flying_Luck | I'm trying to link asterisk(1.2.14/fbsd6.2/cronyx tau32pci) via E1 with Nec NEAX ips 2000 station. 'pri show span 1' shows it as 'Provisioned, Up, Active'. switchtype is set to q.sig because as i undertand station want q.sig. When i'm trying to call to station i'm getting some 'T200 counter expired, What to do' and than - ''== Primary D-Channel on span 1 down. Where should i dig ? |
| 19:19.04 | CrashSys | 250 channels is not 250 calls either |
| 19:20.16 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 19:20.47 | giasai68 | hello |
| 19:21.32 | giasai68 | when I do a call using ooh323 protocol, whenI hangup call is destroit immediatly... any help please??? |
| 19:21.47 | joelsolanki | yes i have bandwidth |
| 19:21.52 | critch | hangups SHOULD destroy the call |
| 19:21.56 | joelsolanki | we have about 20 Mb bw. |
| 19:22.10 | joelsolanki | so thats not an issue. |
| 19:22.23 | [TK]D-Fender | giasai68: What do you THINK it should be doing after you hangup? |
| 19:22.25 | giasai68 | critch: how I can fix? |
| 19:22.36 | critch | hangups SHOULD destroy the call _need_ to be in the loop when calls are placed? |
| 19:22.45 | critch | opps |
| 19:23.05 | critch | joelsolanki: Does asterisk _need_ to be in the loop for all calls? |
| 19:23.07 | [TK]D-Fender | giasai68: Why should a call continue AFTER you've hung up? That makes little sense unless you can tell us what actions you need it to perform... |
| 19:23.41 | giasai68 | yes, I need continue calls after I have hung up the call |
| 19:23.56 | *** join/#asterisk Telamon (i=telamon@blk-137-96-217.eastlink.ca) |
| 19:24.06 | *** part/#asterisk Telamon (i=telamon@blk-137-96-217.eastlink.ca) |
| 19:24.22 | giasai68 | is it possible fix this issue? |
| 19:24.35 | critch | giasai68: take about 4 steps back from the problem and give us the bigger picture of what you want to do. |
| 19:24.49 | Bobthehunter | can i do || ? in a check as in GotoIf($[${ACCOUNTCODE} = ("123" || "234") |
| 19:24.50 | CrashSys | you want to make more calls after you hangup |
| 19:24.55 | CrashSys | so make more calls? |
| 19:24.58 | *** join/#asterisk angler_ (i=angler@nat/digium/x-cfdfcde2a0a086d1) |
| 19:25.14 | Qwell[] | Bobthehunter: not like that, no |
| 19:25.18 | Bobthehunter | hm |
| 19:25.25 | Bobthehunter | im searching for format on vinfo |
| 19:25.37 | Qwell[] | $[$[${ACCOUNTCODE} = 123] || $[${ACCOUNTCODE} = 234]] |
| 19:25.42 | Qwell[] | something like that should work |
| 19:25.49 | giasai68 | sjphone ---> astersik ---> PRI ---> gsmgateway ----> call in gsm network |
| 19:26.08 | critch | giasai68: take another step back, bigger picture |
| 19:27.03 | giasai68 | after I have generate call and I have hangup with gsm phone call is immediatly destroit and can do continue |
| 19:27.19 | Bobthehunter | bx.c:1814 pbx_substitute_variables_helper_full: Error in extension logic (missing ']') |
| 19:27.32 | critch | giasai68: at this level of detail, NO you can not continue without hanging up and calling back. |
| 19:27.46 | giasai68 | why? |
| 19:27.52 | critch | giasai68: are you trying to implement call center where agent is always on the phone? |
| 19:27.56 | remiss | giasai68: why would you want to continue the call when it is hung up? |
| 19:28.06 | Bobthehunter | me dumb |
| 19:28.11 | [TK]D-Fender | giasai68: "show application dial" |
| 19:28.20 | giasai68 | remiss: yes, I want to continue the call |
| 19:28.30 | remiss | giasai68: but why and to who? |
| 19:28.45 | remiss | you want the call to return to asterisk? |
| 19:28.47 | [TK]D-Fender | giasai68: Go type what I just gave you in * CLI and READ THE INSTRUCTIONS. |
| 19:29.03 | [TK]D-Fender | remiss: I've just figered where he needs to go. Let him run with it... |
| 19:29.25 | remiss | [TK]D-Fender: okay.. I don't get it :p |
| 19:29.56 | giasai68 | ok |
| 19:32.39 | giasai68 | [TK]D-Fender: how I can apply all that? |
| 19:32.54 | critch | by reading and understainding documenation |
| 19:33.09 | giasai68 | [TK]D-Fender: in my dialplan? |
| 19:34.24 | critch | J4k3: elaborate? |
| 19:34.42 | [TK]D-Fender | giasai68: Quick answer : you are loking for the "g" parameter for your dial. That handles when the REMOTE side hangs up. You need to read bout the "h" standard extension for the LOCAL side hangs up the call to handle that case. |
| 19:35.05 | CrashSys | Trix? |
| 19:35.22 | [TK]D-Fender | J4k3: Silly moron, Trixbox is for kids! |
| 19:36.04 | CrashSys | The best thing I ever did with Trixbox was installing Slack :D |
| 19:36.08 | *** join/#asterisk ChicagoBud (n=Bud@adsl-70-228-35-78.dsl.chcgil.ameritech.net) |
| 19:36.47 | CrashSys | maybe fonality will improve that |
| 19:37.49 | CrashSys | or whoever bought 'em |
| 19:37.59 | ChicagoBud | is there an "application" that will write to the log/console? For debugging, I'd like to write the caller id of a call to the console |
| 19:38.08 | J4k3 | [TK]D-Fender: exactly! kids! |
| 19:38.13 | J4k3 | ;) |
| 19:38.45 | jpe-nyc | hello all... does anyone have any leads for a hook-up line app (chat-line)? |
| 19:38.55 | giasai68 | [TK]D-Fender: I want call isnt termiante befoar the calling party hangs |
| 19:39.23 | [TK]D-Fender | giasai68: I just told you with method accounts for each condition. Get reading. |
| 19:40.52 | ChicagoBud | I guess NoOp() |
| 19:41.15 | [TK]D-Fender | ChicagoBud: and "show application verbose" |
| 19:41.28 | CrashSys | D-Fender: Read? Learn? What's that? |
| 19:41.37 | ChicagoBud | [TK]D-Fender, thanks |
| 19:41.38 | giasai68 | [TK]D-Fender: thanks |
| 19:41.50 | acecase | hows it goin Moobius? |
| 19:42.17 | *** join/#asterisk Ebola (n=Ebola@host81-151-91-139.range81-151.btcentralplus.com) |
| 19:42.28 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:44.09 | acecase | CrashSys you just getting into asterisk? |
| 19:44.18 | *** join/#asterisk BruXo (n=celio@c91192a6.static.bhz.virtua.com.br) |
| 19:44.48 | *** join/#asterisk [Mr_X] (i=1000@88.118.89.212) |
| 19:44.51 | w0ls0n | I am too. I'm just starting doing research on asterisk |
| 19:45.22 | acecase | I was just going to point to http://www.asteriskdocs.org . Someone from here pointed me to it and its full |
| 19:46.02 | acecase | I had no idea just how huge asterisk was when I decided to set it up. Universities should offer cirtificate courses |
| 19:47.17 | [TK]D-Fender | acecase: Not enough deployments, or a stable enough product to truely evaluate some on at that level. |
| 19:47.42 | [TK]D-Fender | acecase: I trust you've downloaded THE BOOK, by this point? |
| 19:47.49 | acecase | its the only full software solution though rite? |
| 19:47.57 | acecase | yes I have the book. thanks for that |
| 19:48.18 | [TK]D-Fender | acecase: Solution to WHAT :) |
| 19:48.31 | acecase | PBX? rite? |
| 19:48.32 | acecase | lol |
| 19:48.37 | *** join/#asterisk Wubba (n=kmurrey@adsl-76-211-157-158.dsl.akrnoh.sbcglobal.net) |
| 19:49.22 | acecase | yeah PBX. i was hoping I got that rite |
| 19:49.40 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 19:50.24 | acecase | what I'm getting at is, it is bound to be the future. its the human condition to want control and hardware PBX systems don't give you that |
| 19:50.28 | CrashSys | ace: I dunno what I am to asterisk but i'm not quite new |
| 19:50.31 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 19:50.31 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:50.34 | CrashSys | i'm more noobish |
| 19:51.02 | acecase | Im completely new so I don't meen any disrespect. |
| 19:51.28 | CrashSys | None taken :) |
| 19:51.32 | [TK]D-Fender | acecase: Well of course * is only software :) |
| 19:51.54 | acecase | i actualy know what you meen by * now though. :) |
| 19:52.10 | [TK]D-Fender | acecase: By which point, so are SIPX, OpenPBX, FreeSWITCH, and a pile of other softwares.... |
| 19:52.25 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 19:52.25 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:52.29 | acecase | ic. I thought asterisk was the only one |
| 19:52.31 | CrashSys | Some of them are piles for sure |
| 19:52.39 | acecase | :) |
| 19:52.46 | [TK]D-Fender | acecase: Stop thinking and start reading.... |
| 19:52.58 | acecase | alright!! :) |
| 19:53.19 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 19:53.45 | CrashSys | www.voip-info.org = da bomb |
| 19:53.51 | CrashSys | sometimes the diggity too |
| 19:54.09 | CrashSys | other times it's crap |
| 19:54.10 | [TK]D-Fender | CrashSys: A good place to go once you know the basics, and need some specifics. |
| 19:54.24 | CrashSys | voip-info.org is how I learned :( |
| 19:54.33 | mafkees | same here |
| 19:54.39 | [TK]D-Fender | CrashSys: It isn't a place you should go without having gone through th BOOK first. |
| 19:55.05 | CrashSys | I've never read the future of telephony |
| 19:55.20 | acecase | rite. its a lot of info but its scattered. need to know a little to get much out of it |
| 19:55.24 | CrashSys | I just starting changing things and seeing what happened... |
| 19:55.31 | [TK]D-Fender | CrashSys: Technically the same here, but I come from a old school programming background when they were still printing BASIC raw function & syntax books... |
| 19:56.00 | [TK]D-Fender | CrashSys: : I am the kind that can imagine the pieces fitting together just by looking at the raw pieces. many/most aren't built that way... |
| 19:56.18 | CrashSys | d-fender: Yup... |
| 19:56.41 | CrashSys | d-fender: I just compiled it... then started poking around conf's, changing things and seeing what happened... |
| 19:56.46 | CrashSys | same way i've learned everything |
| 19:56.58 | CrashSys | which means I threw away my first 2-3 server's configs :D |
| 19:57.03 | Wubba | TKDfender - Did you get that Milk guy taken care of last night? |
| 19:57.57 | acecase | CrashSys I envy people like you. I'm not natural at all. I have to learn from the ground up with everything and then I need a reference laying around still |
| 19:58.02 | *** join/#asterisk Opperior (n=chatzill@24.61.165.73) |
| 19:59.52 | CrashSys | ace: Everyone finds what works for them... if that works for you then go with it... |
| 19:59.53 | acecase | I still have a copy of complete idiots guide to C# laying here and have been using C# since 2003 |
| 20:00.16 | acecase | I just messed up and said C sharp in a channel full of linux gurus. now everyone hates me don't they :) |
| 20:00.24 | CrashSys | I have yet to learn C or C# |
| 20:00.30 | CrashSys | one of these days I will |
| 20:00.41 | CrashSys | C# = C++ aint it? |
| 20:01.06 | hads | No |
| 20:01.18 | mafkees | no |
| 20:01.19 | acecase | C# is a Microsoft variant of C thats mainly used for programming in .NET |
| 20:01.27 | CrashSys | Ahhh |
| 20:01.30 | mafkees | it will run with mono |
| 20:01.34 | acecase | the only thing C about it is the syntax realy |
| 20:01.44 | mafkees | c++ syntax yeah |
| 20:01.54 | acecase | mafkees is mono getting good yet? havn't looked at it in a couple years |
| 20:02.20 | mafkees | acecase: it's ok. but it's not really fast |
| 20:02.26 | mafkees | we only use it for one thing |
| 20:02.29 | mafkees | beagle indexing |
| 20:02.58 | mafkees | I dont know a lot of it, I'm not a C# guru |
| 20:03.43 | acecase | I'm gonna break down some day and learn the STL and standard C libraries and use C++ everywhere. rite now I use C# for windows and perl or just bash for what little I do in linux |
| 20:04.25 | *** join/#asterisk ElKronos (n=elkronos@adsl188-238-centenario.neunet.com.ar) |
| 20:04.31 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 20:04.34 | *** join/#asterisk bkruse_home (i=kruz@nat/digium/x-f35f4b1235a3fcad) |
| 20:04.35 | mafkees | brb |
| 20:06.01 | acecase | [TK]D-Fender do you know of any good IAX2 controls or libraries? I have looked at a few but most seem incomplete |
| 20:06.57 | remiss | ugh.. is there nothing I can do about the client noise thing with sip? I sendt an email to the sip-provider but they didn't even bother replying :S |
| 20:09.18 | *** join/#asterisk heison (n=heison@dns.somanetworks.com) |
| 20:09.52 | w0ls0n | remiss what voip provider do you have? |
| 20:10.20 | acecase | gotta go. thanks for letting me loiter |
| 20:10.26 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 20:11.08 | remiss | I mean silence support thingy not noise :S |
| 20:11.17 | remiss | w0ls0n: a norwegian one... televoip |
| 20:11.21 | w0ls0n | ah ok |
| 20:11.42 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 20:12.07 | remiss | bah.. comfort noise support* |
| 20:12.29 | Bobthehunter | anyway to delay zap .. i mean when i send to an ivr from zap it missing the start of playback.. tried playing a silence but same.. |
| 20:13.06 | Bobthehunter | ? |
| 20:14.03 | *** part/#asterisk hyphen (n=hyphen@c-69-136-84-149.hsd1.pa.comcast.net) |
| 20:14.17 | *** join/#asterisk RoyK (n=roy@ti211310a080-5551.bb.online.no) |
| 20:14.20 | CrashSys | bob: like it's breaking up? |
| 20:14.24 | J4k3 | wow, this is an interesting couple of bugs |
| 20:14.24 | J4k3 | Feb 8 14:01:53 WARNING[3571] channel.c: Unable to find a codec translation path from g729 to unknown |
| 20:14.24 | J4k3 | Feb 8 14:01:53 WARNING[3571] file.c: Unable to open vm-password (format g729): No such file or directory |
| 20:14.54 | *** join/#asterisk [Mr_X] (i=1000@88.118.89.212) |
| 20:15.03 | [TK]D-Fender | J4k3: Looking like you don't have G.729 licenses or native encoded recordings. |
| 20:15.45 | J4k3 | yeah... which is odd because I installed g729 last night. |
| 20:15.58 | J4k3 | oh well, this is what I get for mixing proprietary crap up in the mix. |
| 20:16.10 | heison | i have been relying on app_pgsql, but it's no longer available in 1.4... does anyone know of a way to access a postgres database directly from the dialplan? |
| 20:16.41 | critch | heison: func_odbc |
| 20:16.52 | J4k3 | blah... I need to reinstall this box, then play the kiss-digium's-ass routine because this will be my second reinstall with the g729 codec license |
| 20:16.59 | J4k3 | and if they give me any lip I'll just stop payment on my credit card :P |
| 20:17.04 | heison | critch: thought so... thanks!! |
| 20:17.30 | CrashSys | heison: ODBC? |
| 20:19.15 | heison | CrashSys: thx |
| 20:19.22 | *** join/#asterisk flying_Luck (n=melifaro@ppp85-140-137-33.pppoe.mtu-net.ru) |
| 20:22.16 | *** join/#asterisk CrashHD (n=crashhd@c-76-20-22-240.hsd1.ca.comcast.net) |
| 20:23.18 | [TK]D-Fender | J4k3: You should be able to save your license file if you're using the sme hardware |
| 20:25.07 | Bobthehunter | ok so i need a zap inbound predelay... |
| 20:25.13 | Bobthehunter | anyway to do this ? |
| 20:26.07 | CrashSys | inbound pre-delay |
| 20:26.46 | [TK]D-Fender | Bobthehunter: Clarify plesae... |
| 20:26.48 | CrashSys | like your trying to dial and it's not reading all the digits (dialing out) |
| 20:27.04 | [TK]D-Fender | CrashSys: that'd be OUTBOUND... |
| 20:27.05 | shmaltz | anybody here interested please follow up: |
| 20:27.07 | shmaltz | http://lists.digium.com/pipermail/asterisk-biz/2007-February/019943.html |
| 20:27.13 | CrashSys | d-fender: could be :) |
| 20:28.40 | CrashSys | Too bad i'm not in jersey... i've got experience with asterisk, and a toshiba strata DK40i... |
| 20:28.48 | CrashSys | and all the toshiba CRAP |
| 20:29.22 | Bobthehunter | i mean when i send to an ivr from zap it missing the start of playback.. tried playing a silence but same.. |
| 20:29.30 | shmaltz | CrashSys, where do you live? |
| 20:29.46 | CrashSys | Florida |
| 20:29.49 | CrashSys | it's pretty cold today |
| 20:29.53 | CrashSys | like 70 |
| 20:29.59 | shmaltz | lol |
| 20:30.11 | shmaltz | 30 here in the UK where I am at the momen |
| 20:30.13 | shmaltz | t |
| 20:31.06 | *** join/#asterisk msupino (n=marco@192.114.87.134) |
| 20:31.42 | Bobthehunter | onyly thing i found is $agi->answer() then sleep(1); then do the shit |
| 20:34.14 | CJLinst | 1.4 isd giving me a segmentation fault trying to transfer to park orbit (700). |
| 20:34.29 | [TK]D-Fender | Bobthehunter: Thats pretty normal to wait 1-2s after answering to start audio on analog. |
| 20:34.57 | CrashSys | I've gotta use w's in my dial commands for the line to come up |
| 20:35.29 | Bobthehunter | hmm |
| 20:35.34 | Bobthehunter | its PRI TDM |
| 20:35.39 | Bobthehunter | no analgo here |
| 20:36.29 | CrashSys | interesting... |
| 20:37.14 | *** join/#asterisk [Mr_X] (i=1000@88.118.57.67) |
| 20:37.52 | *** join/#asterisk Gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 20:37.54 | CrashSys | The the set-up of the channel is taking a long time... Hummm... |
| 20:43.49 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 20:46.15 | elriah | Hi all. I'm about to buy a bunch of Cisco 7941G's. I found one for $299 called "Cisco CP-7941G IP Phone Global Spare" and another one for $399.00 called "Cisco CP-7941G-CH1 IP Phone w / License", I obviously want to use SIP, so which of these phones should I get? |
| 20:46.38 | Qwell[] | elriah: You should call your Cisco sales rep and ask him that question. |
| 20:46.52 | Qwell[] | elriah: The answer _I_ was given, however, is that the license is to use the software on the phone |
| 20:47.20 | *** join/#asterisk Skarmeth (n=Skarmeth@201009052174.user.veloxzone.com.br) |
| 20:47.25 | Qwell[] | I basically had to beat that answer out of them though |
| 20:47.30 | CunningPike | elriah: Or, you could get Polycoms instead and pay less money and no license |
| 20:47.32 | Skarmeth | hi all |
| 20:47.37 | Qwell[] | CunningPike: there is that ;) |
| 20:48.02 | Juggie | i would avoid cisco for sure :) |
| 20:48.16 | J4k3 | isn't that the master plan to modern internet existance? |
| 20:48.19 | J4k3 | "avoid cisco" |
| 20:48.42 | w0ls0n | I thought Cisco was good |
| 20:48.49 | Skarmeth | any news about blocking charged calls in ISDN/PRI channels?? |
| 20:48.50 | Qwell[] | w0ls0n: the hardware is |
| 20:49.10 | bcnl | is there a reason why ${TIMESTAMP} would work for mixmonitor just ifne in 1.2.x but not in 1.4.0? |
| 20:49.31 | elriah | Hey thanks. |
| 20:49.35 | bcnl | I have it set exactly the same exten => s,n,MixMonitor(/audio/uk_calls/uk-sales-${TIMESTAMP}.wav) |
| 20:49.46 | bcnl | but I'm not getting the timestamp in 1.4 |
| 20:50.51 | Opperior | I use Snom myself. Cost less the Polycom, same features |
| 20:50.51 | *** join/#asterisk [Mr_X] (i=1000@88.118.57.67) |
| 20:51.20 | mafkees | I really like the quality of the cisco phones |
| 20:51.34 | Juggie | hah |
| 20:51.47 | [TK]D-Fender | elriah: You should forget Cisco alotogether and go Polycom... |
| 20:53.25 | [TK]D-Fender | Opperior: No, Polycom doesn't have the "flakey firmware", "only passable audio quality", and "crappy display" features of Snom :) |
| 20:53.26 | Opperior | or Snom. never found a good comparison between Snom and Polycom, though. Always thought you were paying extra for name with Polycom |
| 20:53.35 | Opperior | hmm |
| 20:53.40 | [TK]D-Fender | Opperior: Surprisingly, no. |
| 20:54.10 | [TK]D-Fender | Opperior: they are the "gold standard" SIP phone of #asterisk currently. |
| 20:54.42 | Opperior | wonder if I could get a demo unit to compare... |
| 20:54.49 | syzygyBSD | what is the platinum standard? |
| 20:54.53 | trixman | i need help with the local prefix error |
| 20:55.06 | russellb | the platinum standard is the polycom 650 |
| 20:55.09 | russellb | with HDvoice :-p |
| 20:55.16 | syzygyBSD | :( never got one that high |
| 20:55.21 | syzygyBSD | just 501's |
| 20:55.25 | mafkees | does it come with bluray ? |
| 20:55.27 | syzygyBSD | and a 301 somewhere... |
| 20:55.49 | mafkees | hhmm, a phone that can play divx |
| 20:55.54 | mafkees | that would be awesome |
| 20:56.13 | syzygyBSD | they have them don't they? |
| 20:56.17 | Bobthehunter | anyway to use RGN on pri's ? like a 302 on the pri to say hey hes not here hes at NXX XXX XXXX |
| 20:56.33 | mafkees | or like: nethack on the display |
| 20:56.33 | Opperior | play a movie across the phone line |
| 20:56.48 | syzygyBSD | Bobthehunter: what does RGN stand for? |
| 20:57.09 | *** part/#asterisk BruXo (n=celio@c91192a6.static.bhz.virtua.com.br) |
| 20:57.23 | syzygyBSD | Opperior: look at "video" phones |
| 20:58.06 | Opperior | I was referring to playing DivX ^ |
| 20:58.09 | Bobthehunter | Redirecting Number (RGN) |
| 20:58.21 | Bobthehunter | http://www.voip-info.org/wiki/view/RDNIS |
| 20:58.49 | Bobthehunter | meaning can asterisk say on the pri zap line ..hey its not here its there and push back to hairpin |
| 20:59.16 | syzygyBSD | ya.. you need to be on the ss7 network for RGN I believe |
| 20:59.27 | syzygyBSD | or for it to take up 2 zap lines |
| 21:01.25 | w0ls0n | holy shit |
| 21:01.26 | w0ls0n | Anna Nicole Smith is dead |
| 21:01.57 | J4k3 | WHAT?! |
| 21:02.00 | mafkees | and your point is ? |
| 21:02.13 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 21:02.13 | w0ls0n | um, shes like dead |
| 21:02.29 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 21:02.45 | Opperior | she was dead inside long before |
| 21:02.57 | w0ls0n | cocaine overdose prob |
| 21:03.04 | mafkees | yeah |
| 21:03.10 | J4k3 | nah |
| 21:03.11 | J4k3 | downers |
| 21:03.20 | J4k3 | or both |
| 21:03.27 | J4k3 | rollercoaster. |
| 21:05.07 | J4k3 | or... she hit her limit on trashyness. |
| 21:05.29 | syzygyBSD | start the consipricy theories about how it was the RIAA that killed her |
| 21:05.45 | Opperior | no, it was the pirates! |
| 21:05.53 | J4k3 | no, that'd be Brooke Hogan... because if anyone ever hears her godawful "music" they'll never want to hear any other music-like thing again. |
| 21:05.57 | mafkees | HARRRRRRRRRRRR |
| 21:06.07 | syzygyBSD | pirates and ninja's agree, cowboys suck |
| 21:06.21 | *** join/#asterisk Growly (n=growly@is.hibs.school.nz) |
| 21:06.53 | *** join/#asterisk duel (i=duel@mail.asaiatm.com) |
| 21:07.19 | *** join/#asterisk kg12gk (n=none@wsip-66-210-250-2.ph.ph.cox.net) |
| 21:07.21 | kg12gk | hello |
| 21:07.32 | syzygyBSD | hi kg12gk how are you today |
| 21:07.39 | kg12gk | good how r u |
| 21:07.43 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 21:08.09 | kg12gk | guys I need to setup a phone system. should I go with plain asterisk, or asteriskNOW or Trixbox ? |
| 21:08.09 | syzygyBSD | I'm good, kinda, well to be honest life sucks, what a pointless existance with no hope for.... |
| 21:08.22 | syzygyBSD | lol, j/k I'm doing well |
| 21:08.22 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 21:08.57 | syzygyBSD | kg12gk: well, depends on what you want to do, how much time you want to devote, your skill level.... |
| 21:09.38 | kg12gk | well ... intermediate level, very quick setup we have no time to mess around so we need a phone system quick |
| 21:10.00 | kg12gk | is asterisk now reliable ? |
| 21:10.05 | kg12gk | it's in beta4 now |
| 21:10.29 | syzygyBSD | kg12gk: any special features you want on your phone system? |
| 21:10.35 | J4k3 | haha |
| 21:10.52 | kg12gk | qeues, and multiple auto atendents |
| 21:11.12 | syzygyBSD | kg12gk: ya, they all can do that |
| 21:11.15 | kg12gk | version numbers no but version NAMES hehe |
| 21:11.19 | kg12gk | BEAT is a bad name |
| 21:11.28 | J4k3 | actually |
| 21:11.32 | J4k3 | its not really a "Beta" or anything else |
| 21:11.35 | J4k3 | its a source snapshot |
| 21:11.36 | J4k3 | tahts it. |
| 21:11.44 | kg12gk | hehe there u go |
| 21:11.49 | syzygyBSD | kg12gk: so gmail beta isn't stable? |
| 21:11.58 | J4k3 | syzygyBSD: no, its not. |
| 21:12.10 | kg12gk | it is but it's features are limited :P |
| 21:12.11 | mafkees | indeed |
| 21:12.17 | syzygyBSD | hmm, well for the last 2 years I have used it it has been |
| 21:12.27 | mafkees | J4k3: you cannot use it with telnet ? |
| 21:12.30 | syzygyBSD | kg12gk: how much more can you ask for from a mail client? |
| 21:12.41 | J4k3 | mafkees: well that, and the fact that if it deems something spam I CAN NOT FORCE IT TO STOP. |
| 21:12.41 | syzygyBSD | J4k3: what problem? |
| 21:12.47 | kg12gk | hehe it doesn't have imap :P |
| 21:12.47 | kg12gk | hehe |
| 21:12.52 | J4k3 | and once it deems it spam, it blocks all the images off the email |
| 21:13.06 | J4k3 | be it attachments or URLs... so therefore the mail is now useless, spam or not. |
| 21:13.14 | syzygyBSD | hmm... |
| 21:13.15 | J4k3 | and I can't convience gmail to stop that crap. so I quit using gmail |
| 21:13.19 | kg12gk | guys now back to asterisk |
| 21:13.29 | J4k3 | I only used it for silly crap anyways. Its the new hotmail - too useless to be useful but convienent to give spammers. |
| 21:13.35 | syzygyBSD | J4k3: I just checked, I can view images in spam just fine |
| 21:13.37 | kg12gk | I tried trixbox and it keeps crashing with EXT3-fs errors every now and then |
| 21:13.44 | kg12gk | so I just gave up on it |
| 21:13.52 | J4k3 | syzygyBSD: well, I certainly can't on a few mailing lists I was on. |
| 21:13.58 | mafkees | kg12gk: that's not trixbox fault |
| 21:14.02 | J4k3 | which work fine when they're delivered to a real mail account and a real mail application |
| 21:14.03 | mafkees | it means your disk is rotten |
| 21:14.03 | kg12gk | then I saw that we have asteriskNOW |
| 21:14.10 | syzygyBSD | J4k3: might have been a past error, they have gotten a lot better |
| 21:14.16 | J4k3 | too little/too late. |
| 21:14.24 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
| 21:14.28 | J4k3 | I prefer my real mail account. I prefer ownership of my own content. |
| 21:14.29 | CrashSys | LOL, version numbers... reminds me of this "IT" guy that came in to set-up servers after I wired an office... told the owner that I had wired everything incorrectly... |
| 21:14.29 | syzygyBSD | meh, I like the fact that it is a POP3 client now |
| 21:14.36 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 21:14.49 | *** join/#asterisk PhilKC (i=greece@freenode/staff/about.linux.philkc) |
| 21:14.50 | fiber0pti | Does anyone know where I can download the SIP firmware for a Cisco 7912 without the need for a cisco login? |
| 21:14.50 | mafkees | I dont use gmail because I cannot read it with mutt |
| 21:14.53 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 21:14.58 | mafkees | fiber0pti: you cant |
| 21:15.11 | syzygyBSD | mafkees: you haven't really tried have you |
| 21:15.18 | syzygyBSD | you can read it with mutt just fine |
| 21:15.19 | *** join/#asterisk dasenjo (n=dasenjo@190.24.176.206) |
| 21:15.24 | CrashSys | so after driving up there, and getting grilled, I ask the guy what was incorrect, and he goes "It's not wired to T568b!"... to which I reply "Yeah, it's T568a"... and he goes "Well B is the newer one!" |
| 21:15.40 | J4k3 | well, the bigger problem is... gmail offers nothing and delivers nothing |
| 21:15.47 | J4k3 | nobody on earth wanted another fucking hotmail, and thats what was delivered. |
| 21:15.49 | LeddyHM | yay for vendors who no longer support us making changes w/o our knowledge or approval |
| 21:15.55 | fiber0pti | mafkees: i have for a 7940.. someone had a link too it a long time ago |
| 21:15.57 | LeddyHM | they borked up DTMF |
| 21:16.10 | Juggie | J4k3? |
| 21:16.18 | LeddyHM | you guys know of anywhere I can look to see why the autoattendant doesn't recognize dtmf tones? |
| 21:16.39 | CrashSys | I check my e-mail with rm |
| 21:16.42 | syzygyBSD | J4k3: if you think gmail is another hotmail.... I'm sorry. |
| 21:17.01 | J4k3 | syzygyBSD: thats exactly what it is. Nobody else wants any other gmail features unless they're idiots. |
| 21:17.02 | Juggie | gmail is miles better then hotmail, speed, threading, built in IM, etc... |
| 21:17.17 | J4k3 | wow, like I want an IM. |
| 21:17.22 | J4k3 | I don't want to store files on gmail's servers |
| 21:17.22 | mafkees | fiber0pti: that's not legal |
| 21:17.34 | J4k3 | (come on, $4GB usb flash devices are under $40 now...) |
| 21:17.35 | mafkees | syzygyBSD: does gmail offer imap ? |
| 21:17.46 | syzygyBSD | mafkees: no, pop3 though |
| 21:17.46 | Juggie | then dont, no one's asking you do, furthermore this isnt #gmail, so i dont see how any of this matters. |
| 21:17.50 | fiber0pti | mafkees: slap the cuffs on and take me to jail ;) |
| 21:17.53 | CrashSys | I hate my thumb drive |
| 21:17.57 | CrashSys | I always loose it |
| 21:18.05 | CrashSys | I just ftp to my server anyways |
| 21:18.14 | CrashSys | or http if I know I need it |
| 21:18.16 | J4k3 | Juggie: ok congrats, content whiner. I didn't see you adding anything to the discussion before the OT so fuck off. |
| 21:18.48 | *** join/#asterisk Modcuts (n=Moducts@88-109-60-190.dynamic.dsl.as9105.com) |
| 21:19.10 | CrashSys | that's 3l33t |
| 21:19.19 | CrashSys | or something |
| 21:19.57 | Corydon-w | J4k3: calm down |
| 21:20.15 | syzygyBSD | J4k3: just for me, the ability to have all of my email ever from all of my accounts available everywhere with an internet connection is very very convinent |
| 21:20.24 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:20.39 | J4k3 | syzygyBSD: except you lost the ownership of that mail. I personally have no desire for my words to be (c) Google. |
| 21:20.50 | J4k3 | if google wants to pay me for content, they can. I don't think they'll like my price. |
| 21:20.53 | mafkees | ah well |
| 21:20.56 | Corydon-w | J4k3: OT, please take it elsewhere |
| 21:21.12 | J4k3 | Corydon-w: you realize this conversation has more than me in it. |
| 21:21.15 | syzygyBSD | J4k3: lol, you lose "ownership" of your email if it ever touches a router on the internet |
| 21:21.17 | CrashSys | I'd pay to have someone delete my e-mail for me |
| 21:21.21 | LeddyHM | hmmm. no ideas? |
| 21:21.28 | mafkees | lol CrashSys |
| 21:21.32 | J4k3 | I'm going to be quiet before Corydon76 has a fit. |
| 21:21.47 | CrashSys | He's crazy like that |
| 21:21.49 | mafkees | J4k3: you dont want to mess with Corydon ;) |
| 21:21.57 | CrashSys | I once saw him fling trixbox CD's at people |
| 21:22.07 | CrashSys | ninja-style.. |
| 21:22.14 | J4k3 | mafkees: yeah, I'm sure he'll ban me or something... oooh. |
| 21:22.16 | mafkees | lol, that's one of the 2 things they are made for |
| 21:22.22 | mafkees | fling at ppl |
| 21:22.22 | syzygyBSD | i KNEW those cds were good for something, smart guy... |
| 21:22.25 | mafkees | put your mug on it |
| 21:22.33 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-9e3c03818032d015) |
| 21:22.37 | CrashSys | CD's make bad coasters for cold drinks tho :( |
| 21:23.21 | syzygyBSD | I never thought they were good coasters, I like something that is a little absorbant to soak up the sweat from the glass |
| 21:23.32 | CrashSys | Exactly... |
| 21:23.44 | CrashSys | floppies work better, but only marginally |
| 21:24.41 | syzygyBSD | CrashSys: have any 11" floppies? |
| 21:24.49 | CrashSys | bsd: you win :( |
| 21:24.53 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 21:24.57 | syzygyBSD | oh, I don't, I want some though |
| 21:24.59 | CrashSys | I think I have one 5 1/4" somewhere |
| 21:25.03 | J4k3 | hey, is this #coasters? |
| 21:25.08 | syzygyBSD | ya, 5.25 was the most I ever had |
| 21:25.10 | J4k3 | NO, STOP OR CORYDON WILL GET PISSY WITH YOU |
| 21:25.52 | Corydon-w | J4k3: if you're campaigning to piss me off, you're doing a good job |
| 21:26.10 | Corydon-w | Calm down |
| 21:26.12 | J4k3 | Corydon-w: I'm not responsible for your emotions. |
| 21:26.28 | J4k3 | I was just trying to get the channel back on topic |
| 21:26.38 | J4k3 | since its apparently a high priority for you |
| 21:26.41 | Corydon-w | An admirable goal |
| 21:26.45 | syzygyBSD | "it's not my fault you got angry when I pissed on your car" |
| 21:27.52 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 21:27.59 | mafkees | lol syzygyBSD |
| 21:29.03 | blitzrage | I have an 8" floppy |
| 21:29.16 | blitzrage | (and I mean the disk) |
| 21:29.40 | Corydon-w | Oooooo. Hi, blitzrage. (oh, darn) |
| 21:29.55 | blitzrage | Corydon-w: lol -- you are the reason I qualified :) |
| 21:30.24 | blitzrage | ok, off to the Tragically Hip concert! |
| 21:30.26 | mafkees | asterisknow == linux2.6 right ? |
| 21:37.56 | JT | is it wrong to have 8" disks as well as mainframe reel to reel? |
| 21:38.47 | *** join/#asterisk ttuttle (n=tom@gentoo/contributor/ttuttle) |
| 21:39.46 | *** join/#asterisk anthony] (n=anthony@175.21.188.72.cfl.res.rr.com) |
| 21:39.51 | ttuttle | Is it possible (using AGI or something else) to get Asterisk to spontaneously initiate a call to a number, and then do whatever it would do as if the user had initiated the call? Would I be right in assuming that using Dial would not allow me to execute other applications until the user hangs up? |
| 21:42.14 | Corydon-w | ttuttle: call files |
| 21:42.31 | ttuttle | Corydon-w: What do you mean? |
| 21:42.45 | Corydon-w | ttuttle: see sample.call in the root directory |
| 21:42.53 | ttuttle | Corydon-w: Ah. |
| 21:43.23 | ttuttle | Corydon-w: Ooh, this is cool! |
| 21:43.25 | JT | also search the wiki |
| 21:43.31 | JT | they're briefly mentioned in the book too |
| 21:44.01 | JT | the manager interface is the other way to spawn calls from a process |
| 21:45.32 | ttuttle | JT: Mmm, automated phone calls. |
| 21:45.41 | ttuttle | Is it normal that Vitelity allows me to completely spoof my CID? |
| 21:46.18 | JT | apparently it's normal for a lot of us providers |
| 21:46.18 | ttuttle | Nice. =D |
| 21:46.18 | Opperior | Most PRI lines let you specify your Caller-ID |
| 21:46.31 | Opperior | useful for extension-specific caller-IDs |
| 21:46.42 | Opperior | just thought I'd mention |
| 21:46.46 | Opperior | :P |
| 21:46.53 | ttuttle | Opperior: Ah. |
| 21:46.53 | JT | sip terminates at pri |
| 21:47.56 | ttuttle | PRI = ? |
| 21:48.00 | ttuttle | ISDN? |
| 21:48.06 | Opperior | sort-of |
| 21:48.07 | mafkees | isdn30 or T1 |
| 21:48.12 | ttuttle | ah |
| 21:48.27 | mafkees | oh, or J1 |
| 21:48.35 | *** join/#asterisk dj-fu (n=deejay@203-167-190-166.dsl.clear.net.nz) |
| 21:48.55 | Opperior | I think the closest description is a particular use of a T1. It's a T1 where one of the chanells is used for management info |
| 21:49.06 | JT | Opperior: not sort of, yes |
| 21:49.11 | JT | PRI is ISDN |
| 21:49.17 | ttuttle | Are there any open-source voice recognition programs that can be used with Asterisk? |
| 21:49.28 | JT | mafkees: isdn30 = E1, not T1 |
| 21:49.37 | mafkees | JT: uhhuh |
| 21:49.42 | JT | all voice recognition sucks major arse |
| 21:49.46 | mafkees | JT: PRI == E1, T1 or J1 |
| 21:49.55 | ttuttle | JT: I know. But is there any open source? |
| 21:49.58 | mafkees | that's what I answered |
| 21:50.06 | JT | mafkees: well a T1 has 23 traffic channels in PRI mode |
| 21:50.07 | JT | not 20 |
| 21:50.11 | JT | 30 |
| 21:50.16 | mafkees | I know |
| 21:50.29 | mafkees | hence the: isdn30 _or_ T1 |
| 21:50.33 | JT | your answer sounded like it equated isdn30 to t1 |
| 21:50.41 | mafkees | ah |
| 21:50.42 | mafkees | sorry |
| 21:50.45 | mafkees | not meant to be |
| 21:50.56 | Corydon-w | ttuttle: practically, there are no open source programs. There's Sphinx, but that doesn't do an acceptable job. |
| 21:51.10 | JT | sometimes hard to express stuff easily over irc :) |
| 21:51.14 | Corydon-w | ttuttle: You may be interested in a Lumenvox license, however |
| 21:51.20 | ttuttle | Corydon-w: How much? |
| 21:51.20 | mafkees | yeah |
| 21:51.32 | ttuttle | Corydon-w: Mind you, I'm just tinkering with this. |
| 21:51.46 | Corydon-w | ttuttle: don't quote me, but I think the unlimited license is around $150, while the developer license is around $50 |
| 21:51.57 | ttuttle | Corydon-w: Hmm. |
| 21:52.04 | ttuttle | Corydon-w: Too much for playing, but reasonable for an app. |
| 21:52.25 | Corydon-w | ttuttle: I dunno, the text-to-speech stuff is $30/voice |
| 21:52.37 | Corydon-w | ttuttle: cepstral kicks ass |
| 21:52.53 | mafkees | cepstral is paid as well right ? |
| 21:53.01 | Corydon-w | Correct |
| 21:53.33 | mafkees | must be better then the robotlike stuff festival creates |
| 21:53.43 | ttuttle | Mmm, festival isn't that great, but it's free. |
| 21:53.54 | mafkees | yeah |
| 21:53.56 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:53.58 | Corydon-w | It's noticeable on some words, but over all, Cepstral is very natural sounding |
| 21:54.05 | mafkees | ever tried to let festival read a mail message to you |
| 21:54.29 | mafkees | it's horrible |
| 21:54.44 | ttuttle | mafkees: No, I haven't. |
| 21:54.50 | ttuttle | mafkees: But I imagine it sucks. |
| 21:54.55 | mafkees | Corydon-w: does cepstral come in non-uk languages ? |
| 21:54.56 | Corydon-w | You can download Cepstral and listen to the samples, if you don't mind it tacking on an unlicensed message to the beginning |
| 21:54.58 | mafkees | like Dutch |
| 21:55.11 | anthony] | anyone looking for a dedicated to run your pbx on? |
| 21:55.19 | Corydon-w | mafkees: it does, but I don't know about dutch |
| 21:55.32 | mafkees | cool, I think I'll need to check it out |
| 21:55.43 | mafkees | during testing I can live with some extra message |
| 21:56.33 | Corydon-w | US English, UK English, Italian, Canadian French, German, Americas Spanish |
| 21:56.56 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 21:58.07 | *** join/#asterisk nachophone (i=boster@ivan.dreamhost.com) |
| 21:58.31 | Corydon-w | mafkees: http://www.cepstral.com/downloads/ |
| 21:59.25 | *** part/#asterisk Opperior (n=chatzill@24.61.165.73) |
| 21:59.33 | *** join/#asterisk duel (i=duel@mail.asaiatm.com) |
| 22:00.07 | robl^ | Corydon-w: can it do Kilingon? |
| 22:00.25 | Corydon-w | Not yet |
| 22:01.09 | joe | anyone know how to fix a config file error 0x4020? |
| 22:01.27 | mafkees | thnx |
| 22:01.32 | nachophone | in my logs, i'm getting app_queue.c: No one is answering queue followed by ast_expr2.y: non-numeric argument followed by app_dial.c: Unable to create channel of type 'SIP' (cause 3- No route to destination) |
| 22:01.46 | nachophone | is it possible I'm hitting some sort of upper limit on SIP sessions? |
| 22:03.15 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 22:03.17 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 22:03.33 | CrashHD | I'm getting unauthorized messages |
| 22:03.36 | CrashHD | can someone tell me what |
| 22:03.38 | CrashHD | why |
| 22:03.39 | CrashHD | http://www.pastebin.ca/345860 |
| 22:03.40 | CrashHD | ? |
| 22:04.18 | *** join/#asterisk max_______ (i=max__@ts.bestserversllc.net) |
| 22:04.26 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242381245.dsl.bell.ca) |
| 22:05.17 | *** join/#asterisk orkid_ (n=orkid@bas1-barrie18-1242381245.dsl.bell.ca) |
| 22:05.52 | *** join/#asterisk marv0997 (n=marv@205.211.247.190) |
| 22:06.41 | Modcuts | Does anybody know of a sip soft phone that allows multiple extensions to be logged in? |
| 22:06.55 | mafkees | thnx all. I'm downloading cepstral now and I'll start to rip festival from all my proof-of-concepts |
| 22:07.12 | mafkees | finally something good enough to put this stuff into production soon |
| 22:07.23 | CrashHD | I'm getting unauthorized messages |
| 22:07.25 | CrashHD | http://www.pastebin.ca/345860 |
| 22:07.26 | mafkees | integrate it into my monitoring |
| 22:07.27 | CrashHD | anyone know why? |
| 22:07.46 | mafkees | my nice menu with coffeemachine and vcr |
| 22:08.02 | mafkees | no more listening to my own voice there :) |
| 22:08.11 | mafkees | no more prerecording every message I can have |
| 22:10.18 | mafkees | hhmm |
| 22:10.35 | mafkees | I need 2 licenses before I can use it in asterisk ? |
| 22:10.48 | mafkees | 1 for the voice and 1 for the ADL ? |
| 22:11.39 | jpe-nyc | i think it is a single user and multi user license schema |
| 22:11.50 | *** join/#asterisk digilink (n=digilink@66-191-246-176.dhcp.kgpt.tn.charter.com) |
| 22:13.46 | mafkees | so for my home pbx I can use the 29,99 license |
| 22:13.53 | mafkees | but in my hosted pbx I have to get the ADL |
| 22:15.09 | mafkees | I'll mail cepstral about it :) |
| 22:15.14 | mafkees | they will know best |
| 22:18.32 | *** join/#asterisk ruied (n=ruied@bl7-219-82.dsl.telepac.pt) |
| 22:19.00 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 22:21.01 | elriah | So I have 25 Cisco 7941's on the way. Any suggestions for use with asterisk beyond what's on voip-info.org? |
| 22:22.33 | perd | guys |
| 22:22.40 | perd | i think you should all know, anna nicole smith is dead |
| 22:22.55 | [TK]D-Fender | elriah : Sad overexpenditure, but hey, whatever. Hope you got your SmartNET support with it for firmware, etc. |
| 22:23.22 | *** join/#asterisk J4k3 (i=jsuter@78.sub-70-216-60.myvzw.com) |
| 22:23.29 | elriah | Yep, sure did. Not my decision, really. |
| 22:23.32 | mafkees | perd: yeah. overdose |
| 22:24.01 | mafkees | NOT |
| 22:24.37 | *** join/#asterisk Opperior (n=chatzill@c-75-69-247-108.hsd1.nh.comcast.net) |
| 22:25.09 | mafkees | n=stkn@gentoo/developer/pdpc.active.stkn |
| 22:25.15 | mafkees | what kindda hostnames is that ? |
| 22:27.58 | elriah | Which tftp server are you guys using on debian/ubuntu? |
| 22:28.55 | mafkees | pure-ftpd or proftpd |
| 22:29.01 | mafkees | depends on what it has to do |
| 22:29.05 | elriah | They do tftpd? |
| 22:29.12 | mafkees | no |
| 22:29.29 | elriah | For cisco phones, wil the new 8.6 firware do ftp? |
| 22:29.33 | elriah | *will |
| 22:29.39 | mafkees | not that I know |
| 22:29.43 | mafkees | it uses tftp |
| 22:31.02 | elriah | I have a barely used TDM400P w/4fxs (red) cards installed if anyone wants it, first reasonable PM offer gets it. |
| 22:31.03 | mafkees | for debian: apt-get install tftpd |
| 22:31.14 | elriah | mafkees: thanks |
| 22:35.07 | *** join/#asterisk olinux (n=olinux@starbucks.wellspublishing.net) |
| 22:35.23 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com) |
| 22:35.24 | Corydon-w | red is fxo, not fxs |
| 22:35.42 | elriah | fxs is station, right? o is telco? |
| 22:35.55 | Corydon-w | Correct |
| 22:36.06 | elriah | Anyway, they are red. Designed to plug 1fb's into. |
| 22:36.27 | elriah | Fromt he phone company |
| 22:36.32 | elriah | *from the |
| 22:37.56 | *** join/#asterisk Ironhand (i=arjen@meek.xs4all.nl) |
| 22:40.33 | mafkees | I'm off |
| 22:40.35 | mafkees | latero all |
| 22:41.45 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242379851.dsl.bell.ca) |
| 22:42.09 | nachophone | is there a maximum amount of calls that can be in a a queue? |
| 22:45.14 | *** join/#asterisk megasquid (n=asdf@ip3d.campustech.net) |
| 22:49.07 | perd | damnit, i cant figure out this sound quality issue over sip |
| 22:49.49 | perd | anyone have issues with prompts garbled every now and then? if i go into voicemail i can reproduce it pretty regularly, one or two syllables is garbled and the rest is fine... |
| 22:50.11 | perd | connection is cisco 7912 -> POE switch -> asterisk box |
| 22:52.28 | Opperior | I had something similar once. At random times, sound in voicemail would be garbled for a second or so. Turned out to be an interrupt issue with my TDM card. |
| 22:52.48 | perd | the TDM card affected your SIP audio? |
| 22:52.55 | perd | or were you using an analog phone |
| 22:53.26 | Opperior | I had two TDm cards and they were trying to share the interrupt. It went away when I replaced my analog lines with PRI, so I only needed one card |
| 22:53.36 | perd | my digium pri card and both my ethernet cards all have their own irqs |
| 22:53.38 | perd | :/ |
| 22:53.43 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 22:53.55 | Opperior | hrmm |
| 22:54.11 | perd | i have like 0% cpu usage |
| 22:54.13 | perd | and memory usage |
| 22:54.23 | perd | no transcoding, my sound files are in ulaw format |
| 22:54.29 | perd | shrug. |
| 22:54.56 | *** join/#asterisk jm|laptop (n=jm@zen.jamiem.com) |
| 22:55.03 | perd | oh wth they're gsm now hmm |
| 22:55.08 | perd | maybe that's my problem |
| 22:55.09 | *** join/#asterisk [shodan] (n=shodan@ip044.96-113-216.pppoe1.joliette.intermonde.net) |
| 22:55.52 | CrashHD | hey guys what would cause retransmits from my server? |
| 22:56.02 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 22:56.16 | *** part/#asterisk karmatronic (n=karmatro@84.77.153.132) |
| 23:02.31 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:02.31 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:04.45 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 23:04.45 | *** mode/#asterisk [+o angler] by ChanServ |
| 23:05.17 | critch | what is this, the watched * install never goes to zero channels in use? Just logged in to our switch and was finally down to 2 calls, and then it mushroomed to 8 |
| 23:05.29 | elriah | I have a barely used TDM400P w/4fxo (red) cards installed if anyone wants it, first reasonable PM offer gets it. |
| 23:05.58 | putzz | $10 |
| 23:05.59 | putzz | hehe |
| 23:06.10 | elriah | *reasonable* ;P |
| 23:06.20 | mog | 10.50 |
| 23:06.26 | putzz | more reasonable then 0 |
| 23:06.27 | putzz | ;-) |
| 23:06.30 | putzz | 11 |
| 23:06.36 | putzz | ok 40 |
| 23:06.43 | elriah | oh geez lol |
| 23:06.45 | mog | 40.50 |
| 23:06.49 | *** join/#asterisk dasenjo_ (n=dasenjo@190.24.177.245) |
| 23:06.52 | putzz | 50 |
| 23:06.58 | mog | 50.50 |
| 23:07.12 | putzz | heh |
| 23:07.15 | putzz | I give u 120 |
| 23:07.18 | critch | elriah: does $10 sound reasonable for one of the jumpers, or is that a jumperless card? |
| 23:07.23 | putzz | and thats all u get with all 4 modules |
| 23:07.52 | elriah | I'll do $150+shipping |
| 23:08.11 | putzz | 150 including shipping with the 4 modules with DOA garantee |
| 23:08.12 | putzz | ;-) |
| 23:08.21 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 23:08.25 | mog | i got all day ^_^ |
| 23:08.45 | mog | i actually have 30 or 40 cards at home |
| 23:09.16 | JT | critch: indeed |
| 23:09.20 | JT | analogue is poop |
| 23:09.23 | denon | mog: and you've never sent me any freebies? |
| 23:09.31 | mog | lol |
| 23:09.33 | mog | they are dead |
| 23:09.38 | mog | i was making a sculpture |
| 23:09.39 | denon | ah well .. |
| 23:09.43 | denon | I'll just RMA em <G> |
| 23:09.51 | mog | heh |
| 23:10.19 | denon | a pci card sculpture? |
| 23:10.22 | denon | that'd be ... interesting |
| 23:10.24 | mog | yeah |
| 23:10.32 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 23:10.33 | denon | kinda hard to mold, no? |
| 23:10.37 | *** part/#asterisk bkruse_home (i=kruz@nat/digium/x-f35f4b1235a3fcad) |
| 23:10.39 | mog | my wife was making for her art class |
| 23:10.47 | denon | huh |
| 23:10.56 | mog | it was really her sculpture |
| 23:11.07 | mog | for her art class |
| 23:11.10 | denon | so .. what's it a sculpture of? |
| 23:11.13 | anthony] | anyone looking for a dedicated server to run your pbxs on? |
| 23:11.47 | denon | big red phone |
| 23:11.49 | putzz | anthony], where, how much, specs? |
| 23:12.06 | denon | anthony]: sure, free? I'll take 10 |
| 23:12.33 | Opperior | Title it " Hello -ello -lo -o" |
| 23:12.37 | perd | is there a CLI command to show me what codec a registered sip client is using |
| 23:12.38 | elriah | Later, all. |
| 23:13.29 | critch | perd: is there a codec used if the client is only registered? |
| 23:13.44 | perd | not sure |
| 23:14.03 | critch | perd: answer is no. codecs are part of a channel |
| 23:14.12 | critch | a channel is part of a call |
| 23:14.33 | perd | ah ok |
| 23:14.35 | perd | sip show channels did it |
| 23:15.08 | critch | cool, I was a bit of a smart ass, and still answered a question right |
| 23:15.09 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242380975.dsl.bell.ca) |
| 23:15.19 | perd | you asshole! |
| 23:15.20 | perd | thank you! |
| 23:15.41 | critch | your welcome |
| 23:16.21 | critch | ahh, but I see what that did to my karma, the server I need to clear of calls ended up getting to more calls |
| 23:16.25 | JT | anthony]: hmm, details? |
| 23:16.28 | wunderkin | the great cornholio! |
| 23:16.48 | critch | s/to/two/ |
| 23:17.23 | critch | oops, guess I should have better specified the replacement |
| 23:18.57 | anthony] | A few people in this channel rent dedicated servers from our company, the reason I am mentioning it is because of our sale. |
| 23:19.02 | critch | hmm, just looked over at one of the big air handlers in our colo facility and noticed it is reading 80F inside, why should they have cooling problems when it is 37F outside? |
| 23:19.04 | anthony] | I'm trying to hook the community up :) |
| 23:19.43 | CunningPike | critch: Maybe it's on fire |
| 23:19.59 | critch | anthony]: purchase computers, or hosted? |
| 23:20.04 | anthony] | Heh. |
| 23:20.14 | anthony] | Hosted, in a datacenter.. not my home office. |
| 23:20.27 | anthony] | Level3, TWTC multihomed. |
| 23:20.48 | anthony] | Can be setup within a few hours of ordering. |
| 23:20.49 | critch | ahh, then not so interested. We are about to purchase another machine to have as a warm spare |
| 23:21.01 | anthony] | Okay |
| 23:21.16 | critch | kind of limited to physical access to our PRI circuit |
| 23:22.39 | anthony] | Ah. |
| 23:23.04 | anthony] | I do have a personal Dual opteron 246, 2GB memory, 80GB HDD server (1U) that I might be interested in letting go of at a good price. |
| 23:23.21 | mercestes | anthony]: can I msg you? :D |
| 23:23.21 | critch | and didn't level3 get bought recently, or was it they bought someone else? |
| 23:23.36 | mercestes | level3 generically sucks, don't see why anyone would buy them. |
| 23:23.45 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 23:24.03 | anthony] | mercestes, you've been in one of my servers right? |
| 23:24.03 | critch | ahh yeah, they bought Telcove, formerly Adelphia |
| 23:24.10 | anthony] | Level3, doesn't get bought. |
| 23:24.12 | anthony] | They buy :) |
| 23:24.24 | mercestes | anthony]: no... |
| 23:24.44 | mercestes | you'd know if I was on yoru server. |
| 23:25.03 | anthony] | mercestes |
| 23:25.06 | anthony] | <- potential1 |
| 23:25.08 | anthony] | Heh |
| 23:25.10 | mercestes | but I wanna be...amybe |
| 23:25.16 | anthony] | you're my friend. |
| 23:25.20 | anthony] | (i hope) |
| 23:25.21 | mercestes | ...oh |
| 23:25.24 | anthony] | <3 |
| 23:25.26 | mercestes | Yea! |
| 23:25.30 | mercestes | I was on one of your servers..:D |
| 23:25.30 | anthony] | weeeeeehhh |
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| 23:39.40 | Modcuts | I'm currently working on a java based setup app for asterisk , and was wondering what you thought was better trying to parse the configs directly or using a db to store the information which is then written to the configs? |
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| 23:40.01 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- Asterisk, Asterisk-addons, Zaptel, and Libpri 1.4.0 released!!! (December 23, 2006) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. -=- |
| 23:45.18 | dlynes_laptop | Modcuts, I think it would all depend on which is easier for you to program |
| 23:45.45 | dlynes_laptop | Modcuts, from a user interface perspective, it shouldn't make much difference as long as you don't force them to install a sql server |
| 23:46.23 | dlynes_laptop | Modcuts, i.e. just use berkeley db, or one of the many embedded db's |
| 23:47.00 | dlynes_laptop | Modcuts, or even just a binary file with a binary index, using your own predefined data structure |
| 23:47.03 | ThoMe | has anyone the phone SI-7800 ? |
| 23:47.12 | ThoMe | senao |
| 23:47.34 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:47.39 | J4k3 | ThoMe: just don't get an F1000G :) |
| 23:47.44 | J4k3 | (the utstarcom one) |
| 23:47.49 | ThoMe | aha |
| 23:47.50 | ThoMe | ;) |
| 23:48.05 | J4k3 | I'm filling out return forms for mine right now |
| 23:48.15 | J4k3 | I'm going to get a Linksys WIP300 |
| 23:48.40 | *** join/#asterisk Vec (n=Vector@dsl-243-103-241.telkomadsl.co.za) |
| 23:48.40 | J4k3 | see if its any better |
| 23:49.18 | Vec | Does anyone know if asterisk or the zaptel drivers will have any issues compiling on a x86-64 processor and distribution ? |
| 23:49.55 | CrashSys | Vec: I know of people who compile it without issues |
| 23:50.25 | Vec | CrashSys : I mean no more issues then normal x86 |
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| 23:50.32 | CrashSys | nope... |
| 23:50.42 | CrashSys | They were gentoo users too! |
| 23:50.49 | CrashSys | dunno what USE flags they had |
| 23:51.16 | CrashSys | prolly fOMG and l33t :D |
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| 23:52.27 | Vec | CrashSys : what u mean USE flasgs ? |
| 23:52.32 | CrashSys | nevermind |
| 23:52.40 | CrashSys | it's a gentoo thing |
| 23:52.57 | critch | Vec: what gentoo users do to pass info in like what would go to a configure script |
| 23:53.16 | Vec | CrashSys : ok, sounds odd |
| 23:53.18 | mercestes | like -fomg-optimized. |
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| 23:53.56 | Vec | oh, they sound clever, so u don't always have to set configure options over and over again |
| 23:54.38 | critch | Vec: it is more because they compile most everything from source,and therefore need to specify things like use GTK, or use Gnome, or don't use KDE |
| 23:55.02 | CrashSys | or fOMG |
| 23:55.38 | Modcuts | <dlynes_laptop> : Cheers |
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| 23:57.26 | Opperior | I've seen the WIP300. It feels like it's break if you dropped it on a pillow |
| 23:57.40 | Opperior | er, it would |
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