| 00:00.35 | Un1x | alright i'll just get the i386 version |
| 00:00.36 | syzygyBSD | most of the security issues on a box come from having too much shit installed, just go with the basics, ssh, and asterisk |
| 00:00.42 | syzygyBSD | then keep those 2 patched |
| 00:01.12 | Un1x | I See, btw can i use the x86 version |
| 00:01.24 | Un1x | or must i use the i386 |
| 00:01.29 | Un1x | since my processor is 64 bit |
| 00:01.41 | syzygyBSD | um... what is your question? |
| 00:01.51 | Un1x | well FC has those x86_64 releases |
| 00:01.58 | Un1x | my processor is a AMD X2 its 64 bit |
| 00:02.27 | syzygyBSD | I haven't done anything with 64 bit, so I wouldn't be the best person to answer you |
| 00:02.52 | rtcg-tx | Un1x: I got * to detect my hardware on Slack 11.. PHEW! :) |
| 00:03.20 | syzygyBSD | rtcg-tx: did that include a sata drive? |
| 00:03.32 | rtcg-tx | Ah.. not on Slack11 yet. |
| 00:03.33 | Un1x | rtcg * to detect hardware... |
| 00:03.37 | rtcg-tx | but sata in general...yes.. |
| 00:03.48 | rtcg-tx | don't have slack 11 sata setup |
| 00:03.50 | Un1x | mine isn't that if i can get *nix installed |
| 00:03.54 | Un1x | my problems solved for me |
| 00:04.02 | Un1x | im trying sata and its fialing |
| 00:04.07 | Un1x | so accoding to nvidia |
| 00:04.09 | Un1x | fc5 works |
| 00:04.12 | Un1x | so i'm trying Fc5 now |
| 00:04.13 | Un1x | :( |
| 00:04.24 | Un1x | if it works i'll be happy |
| 00:04.28 | Un1x | because i'll be able to use my phone |
| 00:04.35 | Un1x | althought im downloaing the i386 version |
| 00:04.38 | syzygyBSD | i had some ugly issues with FC5 and asterisk, but it was a while ago |
| 00:04.38 | Un1x | instead of the 64 |
| 00:04.49 | Un1x | where you using Zaptel? |
| 00:04.52 | rtcg-tx | I had a system board that didn't export the SATA drives properly. I had to flash the BIOS to get the sata drive to export to those 0x80 0x81 addresses.. |
| 00:04.58 | syzygyBSD | always |
| 00:05.05 | Un1x | hmm |
| 00:05.10 | Un1x | maybei should flash the bios |
| 00:05.15 | rtcg-tx | can you get something like FREEDOS and install IT on those sata drives? |
| 00:05.18 | Un1x | i'll flash it after i check whats up if fc5 dont work |
| 00:05.24 | rtcg-tx | in other words..can the sata drives be system drives??? |
| 00:05.33 | Un1x | i beleive so |
| 00:05.43 | Un1x | i know there used on dell servers afaik |
| 00:05.47 | Un1x | same with windows computers |
| 00:05.51 | Un1x | withouit a problem |
| 00:05.52 | syzygyBSD | Un1x: do you have a knoppix cd lying around? |
| 00:06.03 | Un1x | but linux is behin coz fucking vendors dont give, devs documentation |
| 00:06.11 | Un1x | so somone has to go around the backdoor hacking the code to get at it |
| 00:06.16 | Un1x | syzygyBSD yes i do |
| 00:06.19 | Un1x | but i beleive its old |
| 00:06.22 | Un1x | about a 1yr+ |
| 00:06.29 | Un1x | i can always download it |
| 00:06.34 | Un1x | be on the pc in about 10 min |
| 00:06.39 | syzygyBSD | should still have sata support, try that while FC5 is downloading |
| 00:07.09 | Un1x | Alright |
| 00:07.13 | syzygyBSD | it is really sad I can download stuff faster than I can burn it |
| 00:07.45 | Un1x | i can burn pretty quick too |
| 00:07.46 | Un1x | lol |
| 00:08.11 | syzygyBSD | well gl |
| 00:08.55 | JT | if knoppix, FC5, GRML, or ubuntu cds can't get the sata up, i don't like your chances :P |
| 00:09.19 | Un1x | i feel like getting FreeBSD on this |
| 00:09.28 | Un1x | i bet you it would work there pretty well connected and organised |
| 00:09.45 | JT | then most zap hardware doesn't work, yay |
| 00:10.05 | Un1x | yea that sucks |
| 00:10.11 | Un1x | but hey devs donate there time they dont get paid :S |
| 00:10.11 | qdk | how do i use ForkCDR()? i tried answering the call, ForkCDR() and then Dial out again, but that doesnt work. :-( |
| 00:10.33 | JT | what are you talking about? heaps of devs get paid |
| 00:10.54 | CunningPike | Does anyone know if the 'New Call' timeout for Polycom phones can be changed? In other words, if you press 'New Call' and then do nothing, you get a reorder tone after 15 seconds - can this 15 seconds be changed? |
| 00:11.16 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 00:11.30 | Un1x | if i get asterisk working tonite |
| 00:11.33 | Un1x | imma be happy if not |
| 00:11.37 | Un1x | im heading straight for a bar |
| 00:11.37 | Un1x | lol |
| 00:11.39 | Un1x | fuck the night |
| 00:11.42 | ucfMethod | ha |
| 00:11.43 | qdk | Un1x: I have * running on FreeBSD. |
| 00:11.45 | Un1x | pissing me the hell off :( |
| 00:11.55 | Un1x | qdk i know it works but zaptel doesn't well it does but stably... |
| 00:12.02 | Un1x | or have you tried zaptel |
| 00:12.16 | JT | not all hardware cards even have drivers in fbsd |
| 00:12.21 | ucfMethod | those freebsd guys love bragging about their ports, why can't they get asterisk wroking? |
| 00:12.28 | qdk | Un1x: well i guess zaptel and stability never really worked out. .-) |
| 00:12.32 | Un1x | because there not doing zaptel |
| 00:12.39 | Un1x | its a seperate team porting it |
| 00:12.44 | JT | ucfMethod: because kernel drivers need to be redeveloped, it's no small task |
| 00:12.46 | Un1x | not in relation to freebsd devs |
| 00:12.51 | Un1x | <3 freebsd i will never let it go |
| 00:12.54 | Un1x | its where i started off :D |
| 00:13.25 | qdk | Un1x: but i think you should try out OpenPBX, which i also have working in BSD, but they dont do that much support for zaptel, but you might be able to get it to work anyway. |
| 00:13.51 | Un1x | well qdk if i wanted to waste time i could have a box for transcoding |
| 00:14.02 | Un1x | like so |
| 00:14.32 | Un1x | Box a (ulaw) --->>> Box b (transcodes to gsm aka into G.729) .--->>> iax provider |
| 00:14.37 | qdk | Un1x: huh? |
| 00:14.39 | *** join/#asterisk MoutaPT (n=root@a213-22-40-63.cpe.netcabo.pt) |
| 00:14.49 | Un1x | nevermind different concept... |
| 00:15.02 | Un1x | man fuck cabinet is soo louad damn fans lol |
| 00:15.13 | qdk | Un1x: why would you use GSM in the middle? |
| 00:15.25 | Un1x | because bandwidth cap = 120KB/s |
| 00:15.35 | Un1x | and i cant have more then 1 call per time |
| 00:15.44 | Un1x | but with compression i could squeeze in few more :) |
| 00:15.48 | Un1x | just for bandwidth wise |
| 00:15.55 | Un1x | im not that bored to waste that much time tho |
| 00:15.58 | Un1x | pondering over it |
| 00:16.04 | cloud9 | ok, I've rid myself of nast echos with fxotune.. I'm using ulaw(g.711), I think it sounds good, but other people are complaining that it still doesn't sound quite good enough. Is there anything you guys could suggest I do to tune my audio qulaity? |
| 00:16.19 | cloud9 | I'm refering specificaly to jitters |
| 00:16.50 | qdk | Un1x: and why are you bringing the "waste of time" subject up? |
| 00:17.03 | Un1x | bored atm |
| 00:17.06 | Un1x | waiting for fc5 to download. |
| 00:18.43 | ucfMethod | Un1x: 120KB/s up speed? |
| 00:18.45 | ucfMethod | thats a T1 |
| 00:18.50 | ucfMethod | give or take a few |
| 00:18.57 | Un1x | Yes |
| 00:19.19 | Un1x | my download is 2300KB/s |
| 00:19.24 | ucfMethod | only 1 call |
| 00:19.26 | Un1x | maybe few KB less or Higher |
| 00:19.29 | Un1x | yes |
| 00:19.30 | ucfMethod | ??? i am confused... |
| 00:19.39 | ucfMethod | how are you getting these calculations |
| 00:19.49 | Un1x | ucfMethod 90KB = Ulaw + ovehead... |
| 00:20.00 | ucfMethod | 90kb not KB |
| 00:20.10 | ucfMethod | i have an office of 40+ |
| 00:20.13 | ucfMethod | on 1 T1 |
| 00:20.17 | ucfMethod | with no probs |
| 00:20.26 | ucfMethod | routinely have 10 simul calls going |
| 00:20.32 | ucfMethod | chatterboxes i tell ya |
| 00:20.38 | Un1x | heh |
| 00:20.40 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 00:21.03 | JT | ucfMethod: i think you mean kb/s |
| 00:21.03 | ucfMethod | and funny things, is internet for office is running on that T1 as well |
| 00:21.13 | Un1x | T1 is sloooow :( |
| 00:21.19 | ucfMethod | i know |
| 00:21.26 | ucfMethod | buts its got great SLA and latency |
| 00:21.28 | JT | faster than your upload, Un1x |
| 00:21.43 | Un1x | JT? |
| 00:21.56 | JT | a T1 |
| 00:22.04 | Un1x | how fast is T1 exactly |
| 00:22.07 | JT | if you only have 128kbit/s |
| 00:22.11 | JT | 1.5Mbit/s |
| 00:22.23 | Un1x | no dude i get 120KB not kbits |
| 00:22.33 | JT | i can be more exact, but i'd have to look it up |
| 00:22.47 | JT | oh bull crap, then why would you have issues with more than one g.711 call? |
| 00:22.57 | Un1x | hmmm i never tried 2 calls :P |
| 00:23.07 | Un1x | lol i always assumed that one call = 90KB not kb |
| 00:23.09 | ucfMethod | its the age old confusion |
| 00:23.13 | ucfMethod | kb is not KB |
| 00:23.16 | *** join/#asterisk luisgrin (n=chatzill@201-213-24-146.net.prima.net.ar) |
| 00:23.24 | JT | what is the speed of your connection in kilobits per second? |
| 00:23.26 | Un1x | yea i know that, its just i thought it was KB and not kbit |
| 00:23.37 | JT | no-one measures connection speed in kilobytes per second |
| 00:23.40 | Un1x | i dont know let me get a speedtest on this connection |
| 00:23.45 | JT | it's kB/s, not KB/s, btw |
| 00:23.47 | JT | argh |
| 00:24.01 | JT | no speedtests, you should know the speed you are signed up to with your isp |
| 00:24.05 | JT | hate web speed tests |
| 00:24.23 | Un1x | not really if youre not signed up :p |
| 00:24.28 | JT | err ok |
| 00:24.40 | JT | it's 85kbit/s per G.711 session in SIP |
| 00:24.45 | Un1x | 9893 kbps |
| 00:24.48 | Un1x | is download |
| 00:24.58 | Un1x | 1024kbps is upload |
| 00:24.59 | JT | what is this, cable, dsl? |
| 00:25.02 | Un1x | cable |
| 00:25.28 | JT | T1 upload is still faster |
| 00:25.39 | JT | it's also much more business grade than cable will ever be |
| 00:25.40 | Un1x | yes upload... |
| 00:25.45 | Un1x | i know that |
| 00:25.47 | ucfMethod | man |
| 00:25.51 | Un1x | this is my home connection tho |
| 00:25.53 | Un1x | not business... |
| 00:25.57 | ucfMethod | this is still going on, i went and got a coke and came back |
| 00:26.13 | JT | sure, then it's a silly comparison to make |
| 00:26.22 | JT | when you say things like "T1 is slooow" |
| 00:26.32 | ucfMethod | thats why, home connections are shite, even if you got a 15mbit/2mbit connection |
| 00:26.47 | ucfMethod | you use a portion of that up speed, and voila, your latency goes to crapola |
| 00:26.48 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 00:26.49 | Un1x | yea hey man optononline uploads are nice |
| 00:26.54 | Un1x | 600kB + |
| 00:27.20 | markit | after a svn up of asterisk 1.2svn, now seems that asterisk don't recongnize anymoer dtfm in the conversation (i.e. for transfers with combination in features.conf), or if I call a IVR, the dtfm are not get... any clue? |
| 00:27.22 | JT | anyway, you should be able to do more than 1 call at once |
| 00:27.26 | ucfMethod | i have a 15mbit/2mbit cox connection at home, its great for downloading 'stuff' from usenet, but serving is crap |
| 00:27.28 | Un1x | is 1.4 stabl now |
| 00:27.31 | Un1x | or is still beta |
| 00:27.44 | Un1x | ucf i know hah all cable isps suck |
| 00:27.50 | Un1x | they should fix there upload ratios |
| 00:27.52 | JT | and if you get 600kB/s uploads, not sure what made you say your upload was only 120kB/s before |
| 00:28.01 | JT | 1.4 is beta |
| 00:28.05 | Un1x | im not saying its my connection |
| 00:28.10 | Un1x | im saying its what optonline gives there customers |
| 00:28.14 | JT | right |
| 00:28.20 | Un1x | im not optonline tho... |
| 00:28.31 | JT | ah ok |
| 00:28.55 | Un1x | so on my 1024kbps how many calls can i handle somoen wanna do the math plz? |
| 00:28.57 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 00:28.59 | Un1x | since i apperently suck at it |
| 00:29.06 | markit | no one with and updated 1.2 from svn? |
| 00:29.34 | JT | 1024 / 85 |
| 00:29.37 | JT | minus a bit |
| 00:29.44 | JT | for overhead, and other usage |
| 00:29.52 | JT | and perhaps you won't get full upload, as cable is shared |
| 00:30.58 | infinity1 | what does TRUNKMSD (MSD) stand for? |
| 00:31.05 | Un1x | i see |
| 00:31.22 | Un1x | so g729 encoding must be... 10kb? |
| 00:31.31 | JT | most significant digit |
| 00:31.34 | qdk | noone using ForkCDR()? |
| 00:31.36 | Strom_C | 8kbps per channel plus overhead |
| 00:31.49 | Strom_C | Un1x: what protocol are you planning on using? |
| 00:31.52 | JT | Un1x: 16 or 20kbit/s with overhead iirc |
| 00:31.55 | markit | infinity1: MSD probably your isdn number, could it be? (not expert here) |
| 00:32.03 | JT | no, that's MSN |
| 00:32.09 | infinity1 | i think most significatn digit |
| 00:32.09 | JT | multiple subscriber number |
| 00:32.11 | markit | ok, |
| 00:32.22 | JT | msd is computers/digit related |
| 00:32.53 | *** join/#asterisk jaike (n=jaike@210.213.168.135) |
| 00:34.49 | Un1x | fuck man |
| 00:34.52 | Un1x | fc5 is hanging |
| 00:34.53 | Un1x | lol |
| 00:35.36 | ucfMethod | im out... later folks |
| 00:36.54 | *** join/#asterisk Luke-Jr (n=luke-jr@CPE-24-31-246-32.kc.res.rr.com) |
| 00:38.08 | *** part/#asterisk jaike (n=jaike@210.213.168.135) |
| 00:38.13 | JT | sounds not good |
| 00:40.26 | *** join/#asterisk shimi (n=shimi@bzq-88-153-244-238.red.bezeqint.net) |
| 00:40.28 | cloud9 | JT : here's a dumb question, the codecs are used for compression, right? and I've read that the more compression, the worse the sound quality.. is there a way to use voip without compression? |
| 00:41.02 | qdk | Un1x: why not getting FC6 now you are at it? |
| 00:41.36 | JT | cloud9: g.711 |
| 00:41.37 | *** part/#asterisk dasenjo (n=dasenjo@63.245.86.186) |
| 00:41.46 | Un1x | because |
| 00:41.48 | JT | is pcm with companding, no compression |
| 00:41.50 | Un1x | i was told by nvida to get 5 |
| 00:42.10 | JT | err |
| 00:42.16 | JT | is this an asterisk box |
| 00:42.19 | JT | or a desktop box |
| 00:42.20 | cloud9 | JT : thanks bud |
| 00:42.20 | cloud9 | yea |
| 00:42.38 | JT | i don't see why you would need any nvida drivers on an asterisk box, Un1x |
| 00:42.49 | Un1x | Umm the chipset man |
| 00:42.52 | Un1x | sata controller |
| 00:42.56 | JT | hrm |
| 00:43.05 | JT | sounds like a pain in the arse |
| 00:43.16 | JT | this is not for business purposes i assume? |
| 00:43.24 | Un1x | nopes |
| 00:44.05 | JT | cloud9: companding expands the dynamic range, so technically it could be seen as compression, but it increases the quality of audio over standard pcm |
| 00:44.46 | JT | Un1x: why not get something simpler to use with asterisk? |
| 00:45.40 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:45.58 | Shaun2222 | i need to validate user input, i basically want to make sure they enter somthing in the 300's |
| 00:46.09 | Shaun2222 | is there a regex type compare? |
| 00:46.38 | *** part/#asterisk MoutaPT (n=root@a213-22-40-63.cpe.netcabo.pt) |
| 00:46.41 | Shaun2222 | so i could say if ${OPT} = 3** then... |
| 00:52.56 | *** join/#asterisk fr0zen (n=fr0zen@bas2-toronto12-1168022550.dsl.bell.ca) |
| 00:53.43 | fr0zen | guys what does anyone know wat the following error means: "Starting Zap/1-1 at from-pstn,s,1 failed so falling back to exten 's'" |
| 00:54.57 | *** join/#asterisk MoutaPT (n=kiitos@a213-22-40-63.cpe.netcabo.pt) |
| 00:58.15 | Shaun2222 | is this right? exten => _X.,103,GotoIf($["${${callee_opt}:0:1}" != "3"]?100); |
| 00:58.17 | JT | looks like it's trying to start at named extension "from-pstn" if i'm reading that right |
| 00:58.38 | *** join/#asterisk insomnia41 (n=dasver96@69.254.230.163) |
| 00:59.11 | Shaun2222 | tring to do if ${callee_opt} does not start with 3 then go to 100 |
| 00:59.34 | *** join/#asterisk drm (n=DRM@ool-45786908.dyn.optonline.net) |
| 01:00.27 | *** join/#asterisk sloth_ (n=josh@cpe-69-203-212-182.nyc.res.rr.com) |
| 01:00.40 | JT | fr0zen: what sort of zap channel is it? |
| 01:01.14 | cloud9 | LOL, I just tried a VOIP bandwidth test on bandwidth.com.. it recommends only 1 concurrent call at a time. |
| 01:01.27 | JT | what's your connection? |
| 01:01.41 | cloud9 | well, for that test lab it's a DSL line. |
| 01:01.57 | cloud9 | but supposedly, according to my boss, it's supposed to compare to a T1 line. |
| 01:02.02 | cloud9 | I guess not so much.. LOL |
| 01:02.36 | JT | probalby not for upload |
| 01:02.45 | JT | you don't even know the specs of the dsl line? |
| 01:03.09 | cloud9 | nope, I just started with this company about 2 months ago. |
| 01:03.19 | JT | right |
| 01:03.23 | *** part/#asterisk lullabud (n=lullabud@12.24.42.67) |
| 01:03.32 | fr0zen | JT: what do u mean ? its running on a PSTN line on a Sangoma card.. |
| 01:03.46 | JT | fwiw, i wouldn't trust web speed tests |
| 01:03.50 | cloud9 | and I've got a ton of other things that have been a higher priority than looking of the specs for a DSL line that we used solely for failover on our router. |
| 01:04.09 | JT | fr0zen: well your sangoma card would be the zap hardware then |
| 01:04.27 | JT | fr0zen: sounds like zapata.conf and/or extensions.conf is configured wrong |
| 01:04.43 | fr0zen | i am sure the extensions.conf is the one not configured properly |
| 01:05.39 | JT | cloud9: it's important if voip will be going over it |
| 01:05.39 | JT | fr0zen: pastebin the relevant sections of both |
| 01:05.39 | JT | ~pb |
| 01:05.45 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 01:05.45 | fr0zen | but i cant find a tutorial which can help me setup a PSTN line comin into Asterisk and then using a ATA to dial out to the internal phones |
| 01:05.54 | cloud9 | JT : agreed |
| 01:06.56 | fr0zen | www.pastebin.ca/261228 |
| 01:08.14 | nosbig | fr0zen, what card are you using to receive the PSTN call? |
| 01:08.15 | JT | fr0zen: what does "context="? in zapata.conf? |
| 01:08.35 | fr0zen | context = from-pstn |
| 01:08.43 | fr0zen | and its a Sangoma A200 |
| 01:09.04 | *** join/#asterisk hoobastooba (n=ckwall@65.205.41.35) |
| 01:09.59 | JT | well the problem would be that you have no [from-pstn] context in your extensions.conf |
| 01:10.13 | hoobastooba | there is an option to add to the Makefile when compiling asterisk which will allow me to show memory consumption by module within the asterisk cli... I am searching for the info, but dont know the term to look for. can anyone tell me what I should be searching for? |
| 01:10.56 | fr0zen | umm but the call comes in fine, it gives that error, but automatically points to [default] which points to [demo] and eventually i hear the demo msg... |
| 01:11.03 | fr0zen | so where would i add this from-pstn? |
| 01:11.17 | fr0zen | wait so instead of [demo] should it say [from-pstn] ? |
| 01:11.18 | nosbig | I was trying to find that context, also. |
| 01:11.34 | JT | another block in the dialplan, or you could change the context in zapata.conf |
| 01:11.41 | nosbig | fr0zen, you can create your own dialplan from scratch... |
| 01:11.51 | JT | nosbig: i did a quick look, then did a ctrl + f :) |
| 01:12.07 | nosbig | Keeping that name, something that makes sense, would be a good idea. |
| 01:12.09 | JT | fr0zen: what created that dialplan? |
| 01:12.23 | nosbig | JT: That looks like the included one in Debian. |
| 01:12.29 | fr0zen | umm what created it? its the default sample fine |
| 01:12.43 | JT | sample from what install? |
| 01:12.46 | fr0zen | its too bloated, i am gonna try one from scratch but couldnt figure it out |
| 01:12.47 | JT | was it source or packaged |
| 01:12.55 | fr0zen | source |
| 01:13.04 | JT | fr0zen: i suggest you take a look at the asterisk book |
| 01:13.08 | JT | it makes it a lot simple |
| 01:13.12 | nosbig | JT: Right on! |
| 01:13.19 | JT | ~thebook |
| 01:13.21 | jbot | somebody said thebook was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:13.22 | nosbig | I bought that book because it was so good..... |
| 01:13.34 | JT | i am going to buy it |
| 01:13.46 | JT | it's a pain to scroll through such a huge pdf with no hyperlinks |
| 01:13.59 | nosbig | Just make sure to use Zap groups if trying to dial multiple lines as a trunk on outbound calls... ;-) |
| 01:14.12 | nosbig | I have so many bookmarks in it, it isn't funny. |
| 01:14.13 | fr0zen | well k i am gonna try from scratch...i am gonna paste it again, lets see what it does... |
| 01:14.32 | *** part/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 01:14.34 | fr0zen | wait what do i need in the new dialplan? any basic tutorual which covers the pstn lines? |
| 01:14.38 | nosbig | We just got done deploying it at work on Cisco IP phones. |
| 01:14.50 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:15.36 | nosbig | fr0zen, the [trunkld], [trunklocal], and [trunktollfree] context all look good... |
| 01:16.05 | fr0zen | why does this dialplan not work: www.pastebin.ca/261234 |
| 01:16.28 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 01:16.31 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 01:16.43 | nosbig | Hang on... We'll check it out... |
| 01:16.51 | fr0zen | i am startin from scratch now..... i don thave any of those [trunkid] etc |
| 01:17.12 | JT | fr0zen: you must start with priority 1 |
| 01:17.22 | nosbig | fr0zen, make sure that your first priority in an extension is 1. |
| 01:17.26 | JT | and priority 2 is in the wrong place |
| 01:17.29 | JT | change it to 3 |
| 01:17.30 | JT | or n |
| 01:17.35 | nosbig | Dang it... You are speedy, JT... ;-) |
| 01:17.39 | JT | n means next |
| 01:17.48 | JT | but you must always start with a number |
| 01:17.49 | JT | ;) |
| 01:18.06 | fr0zen | sweet it worked!:) |
| 01:18.14 | fr0zen | i love u all!!! |
| 01:18.15 | JT | awesome |
| 01:18.32 | fr0zen | now final question...how do i make a phone dial out using the pstn line? |
| 01:18.52 | nosbig | Use the Dial Application... |
| 01:19.09 | nosbig | Dial Zap/g2/${EXTEN} |
| 01:19.35 | hoobastooba | is there somewhere that lists the flag options in the asterisk Makefile? |
| 01:19.52 | nosbig | hoobastooba, Try ./configure --help |
| 01:20.06 | fr0zen | so does Dial Zap/g2/${EXTEN} go under [from-pstn] as well? |
| 01:20.56 | fr0zen | shouldnt the phones dial 9 or something before being able to dial out? currently the phones are able to call each other |
| 01:20.59 | JT | no you should use a different context |
| 01:21.16 | JT | you can use a prefix such as 9 |
| 01:21.39 | JT | or no prefix, if all your internal lines have numbers that do not interfere with pstn numbers in your location |
| 01:21.46 | nosbig | fr0zen, duplicate those trunk contexts from the example file, and include them in the internal context... |
| 01:21.55 | hoobastooba | nosbig: that will list out the flag possibilities specific to that make file? |
| 01:22.06 | fr0zen | trunk contexts? like trunklocal etc? |
| 01:22.12 | nosbig | hoobastooba, Yes... |
| 01:22.15 | hoobastooba | cool |
| 01:22.16 | nosbig | fr0zen, Yes. |
| 01:22.48 | nosbig | hoobastooba, That is the case with basically any auto-toolsd enable source tarball (nearly any modern mature package). |
| 01:23.01 | JT | fr0zen: if you use a prefix, you will need to use ${EXTEN:1} instead of ${EXTEN} for a single digit prefix |
| 01:23.03 | hoobastooba | i love learning new things. |
| 01:24.16 | cloud9 | what's the diff between Mbps, and MBps |
| 01:24.35 | Nugget | Mbps is megabits per second, MBps is megabytes per second. |
| 01:24.43 | fr0zen | k well i am gonna post the config again..feedback for dialing out pls |
| 01:24.50 | cloud9 | thanks Nugget |
| 01:25.05 | cloud9 | so 0.78 MBps, is almost a t1 line right? |
| 01:25.07 | JT | usually a factor of 8 difference |
| 01:25.11 | fr0zen | www.pastebin.com/261242 |
| 01:25.44 | JT | nothing there |
| 01:25.53 | hoobastooba | yep, blank |
| 01:26.08 | cloud9 | so, according to bandwidth calculator, to have 40 concurrent SIP calls using G711, I'd need .78MBps (incoming and outgoing) .. does that sound correct? |
| 01:26.09 | fr0zen | hmm k let me check |
| 01:26.21 | JT | you're abviously hand typing your urls instead of copy and past |
| 01:26.24 | JT | it's pastebin.ca |
| 01:26.29 | fr0zen | www.pastebin.ca/261244 |
| 01:26.36 | fr0zen | ahhh |
| 01:26.42 | fr0zen | sorry yes, cuz two different computers |
| 01:26.58 | JT | ever heard of remote terminals? :P |
| 01:27.31 | hoobastooba | fr0zen: [dialout] you need to use exten => here |
| 01:27.39 | hoobastooba | is that where you are trying to dial from? |
| 01:27.54 | JT | i don't think you've quite got the hang of extensions |
| 01:28.07 | JT | fr0zen: i suggest you read up on extensions in the book |
| 01:28.09 | fr0zen | not at all....this is the first time |
| 01:28.24 | JT | well look at the difference between that line and the others |
| 01:28.36 | JT | exten => extension,priority,command |
| 01:28.41 | fr0zen | welll i would, i just need to get this running for a school demo, and the demo is tomorrow. Unfortunately i dont have the time to read the full book right now. |
| 01:28.41 | hoobastooba | fr0zen: look up the dial command in the wiki |
| 01:28.47 | fr0zen | got this far using voip-info |
| 01:28.48 | *** join/#asterisk sloth_ (n=josh@cpe-69-203-212-182.nyc.res.rr.com) |
| 01:29.06 | hoobastooba | fr0zen... look up the dial command |
| 01:29.10 | fr0zen | k thnx |
| 01:29.44 | JT | fr0zen: then just go to specific sections for reference |
| 01:30.29 | nosbig | http://pastebin.ca/261246 |
| 01:30.42 | hoobastooba | nosbig: asterisk does not use ./configure. so ./configure --help wont work |
| 01:30.53 | hoobastooba | am i wrong? |
| 01:31.02 | nosbig | Hmmm..... Sorry. I haven't built Asterisk... |
| 01:31.22 | nosbig | I typically use the Debian etch packages. |
| 01:31.40 | fr0zen | nosbig: what is that? |
| 01:31.45 | JT | fr0zen: chapter 5, book page 77, pdf page 95 |
| 01:31.48 | *** join/#asterisk tengulre (n=tengulre@221.11.5.182) |
| 01:32.25 | nosbig | fr0zen, your dialplan, revised. |
| 01:32.47 | fr0zen | k well...ah i am confused |
| 01:33.09 | nosbig | What confused you? |
| 01:34.06 | nosbig | You create your dialplan in logical sections.... |
| 01:34.30 | nosbig | Then, you can include only the parts you need in the contexts where calls get dumped. |
| 01:34.35 | fr0zen | well i see that...but [dialout] is that correct? |
| 01:34.45 | nosbig | Sorry, I forgot to remove dialout... |
| 01:34.51 | nosbig | Get rid of it entirely... |
| 01:34.54 | fr0zen | k |
| 01:35.14 | *** join/#asterisk osiris (n=osiris@71.205.27.131) |
| 01:35.58 | fr0zen | k i did...when trying to dial out it gives a busy tone |
| 01:36.01 | nosbig | Your SIP phones all fall into [internal], I assumed. This is based on your sip.conf configuration for each sip client. |
| 01:36.06 | JT | you'll need to add international dialling to that too |
| 01:36.22 | nosbig | I keep forgetting... Where are you from, fr0zen? |
| 01:36.33 | fr0zen | toronto,canada |
| 01:36.37 | nosbig | oic |
| 01:36.52 | nosbig | Me and my arrogant assumption that you were in the USA... Sorry... ;-) |
| 01:36.58 | fr0zen | lol |
| 01:37.03 | hoobastooba | he was speaking american |
| 01:37.15 | fr0zen | eh? |
| 01:37.17 | fr0zen | :p |
| 01:37.20 | nosbig | rotfl! |
| 01:37.44 | fr0zen | JT: well i need to get local working before getting the international |
| 01:39.06 | fr0zen | nosbig: any idea why it isnt dialing out? |
| 01:39.14 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:39.45 | nosbig | And you get a busy signal? |
| 01:40.08 | nosbig | Have you started asterisk on a second console or xterm as "asterisk -vvvvr"? |
| 01:40.10 | fr0zen | yeap...internal calls work fine... |
| 01:40.21 | *** join/#asterisk stuq (n=stuq@user-12lcqia.cable.mindspring.com) |
| 01:40.24 | fr0zen | its as asterisk -vvvvc |
| 01:40.24 | nosbig | You can watch on the console exactly what is happening... |
| 01:40.56 | nosbig | OK. Then, what does zapata.conf look like? Was that in one of your previous pastes? |
| 01:41.10 | JT | it wasnt |
| 01:41.21 | fr0zen | i'll paste the zapata one moment |
| 01:42.03 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 01:42.23 | nosbig | I have fought so many battles with out asterisk implementation... Zap trunk issues, Cisco firmware upgrade issues, fax tone detection, and SIP termination of a fax machine, among other things.... |
| 01:42.32 | hoobastooba | still not finding the flag for my make file... does this sound familiar to anyone? that I can add a flag to the make file so that asterisk cli will show me memory consumption of modules? |
| 01:43.00 | hoobastooba | *Makefile |
| 01:43.11 | fr0zen | u guys want the zaptel or zapata ? |
| 01:43.12 | nosbig | hoobastooba, how much memory are you trying to run Asterisk with? |
| 01:43.15 | nosbig | zapata |
| 01:43.20 | hoobastooba | 4 gb |
| 01:43.33 | hoobastooba | and asterisk chews it up in about 2 days and crashes |
| 01:43.34 | nosbig | How many concurrent calls are you looking to havee at once? |
| 01:43.43 | nosbig | What version are you using? |
| 01:43.49 | hoobastooba | 1.2.13 |
| 01:43.56 | fr0zen | www.pastebin.ca/261253 |
| 01:44.31 | hoobastooba | not an issue of number of calls... issue of something using up all of the ram and not ever giving it back. |
| 01:44.40 | hoobastooba | only happens on this one server. |
| 01:44.49 | hoobastooba | have 5 production servers... 4 working rock solid |
| 01:44.56 | hoobastooba | this one has something unique that i cannot find |
| 01:44.57 | fr0zen | fyi: i am using a Cisco ATA as well |
| 01:45.35 | JT | the cisco ata needs its dialplan setup correctly too |
| 01:46.03 | hoobastooba | jared refered me to the make file flag, but he could not remember what it was or where to find it... |
| 01:46.12 | hoobastooba | trying to do my research and am coming up blank |
| 01:46.14 | fr0zen | umm the internal calls work fine? does that maybe work that the ata setup is correct? |
| 01:46.23 | nosbig | fr0zen, change the g2 to g1.... |
| 01:46.31 | fr0zen | and incomin calls can go to the pstn phones connected via the ata as well |
| 01:46.48 | nosbig | The FXO signalling is used on FXS sevices and vice versa.... |
| 01:47.04 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 01:47.17 | JT | maybe the ata does timeout dialling, shrug |
| 01:47.30 | nosbig | Right now, the outbound calls are being directed to the ATA devices, which are not going to ring that way... ;-) |
| 01:47.32 | JT | is there a big pause between dialling the last digit and it ringing? |
| 01:47.59 | fr0zen | JT: yes...about 5 seconds |
| 01:48.07 | JT | yeah, timeout :) |
| 01:48.17 | fr0zen | nos: k changing that in the extensions.conf |
| 01:48.36 | fr0zen | JT: i am guessing thats not good.... |
| 01:48.48 | JT | oh, it depends if you like to wait 5 seconds or not |
| 01:48.55 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 01:48.59 | fr0zen | really dont care as long as i can dial out |
| 01:49.04 | JT | if you don't, you'll have to setup a dialplan in the ATA |
| 01:49.06 | fr0zen | this is not for home use, its a demo for school project |
| 01:49.10 | JT | as sip has to dial group at a time |
| 01:49.25 | JT | then what will you do with it after the project? |
| 01:49.43 | fr0zen | the ATA and Sangoma card is not mine...its schools property |
| 01:49.45 | fr0zen | they want it back |
| 01:49.54 | JT | ah lol |
| 01:50.15 | JT | do they often give out asterisk projects? |
| 01:50.17 | fr0zen | howcome when i dial out e.g. 9416422xxxx nothing shows up in the asterisk console? all i get is a busy dialtone |
| 01:50.35 | fr0zen | JT: this is the first time, maybe from next year yes |
| 01:50.46 | JT | make sure the verbosity is at least 5 |
| 01:50.57 | fr0zen | JT: everything else shows up..except this..it is set to 5 |
| 01:51.13 | JT | did you reload extensions or asterisk after making changeS? |
| 01:51.34 | *** join/#asterisk MeRwEciq`asi (i=Girl__15@62.162.14.80) |
| 01:51.35 | fr0zen | yeap |
| 01:51.54 | JT | perhaps the busy dialtone is from the ATA |
| 01:52.02 | JT | and there is some sort of dialplan in it |
| 01:52.10 | JT | you will need to look into the ata configuration |
| 01:55.12 | *** join/#asterisk Newbie___ (n=me@211.24.146.11) |
| 01:55.31 | *** join/#asterisk Mad|Cow (n=thirt@74.92.109.201) |
| 01:57.02 | Mad|Cow | Has anyone got zaprtc running on kernel 2.6? I'm pulling my hair out trying to figure this one out ;-) |
| 01:58.02 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-145-177.dyn.embarqhsd.net) |
| 01:58.39 | fr0zen | hmmm not cisco is a pain in the ass...! aregh |
| 01:59.06 | fr0zen | looks like u are right, nothing shows up in asterisk console, so i guess its not going pass the ata...but it does DIAL internally and that shows up in the console |
| 01:59.12 | fr0zen | anything with 9XXXXXXXXx doesnt show up |
| 02:00.40 | *** join/#asterisk sloth_ (n=josh@cpe-69-203-212-182.nyc.res.rr.com) |
| 02:03.20 | *** join/#asterisk lakapaka (n=lakapaka@69-174-196-11.chvlva.adelphia.net) |
| 02:04.04 | fr0zen | does anyone here have experience with Cisco ATA 186? |
| 02:07.47 | *** part/#asterisk MoutaPT (n=kiitos@a213-22-40-63.cpe.netcabo.pt) |
| 02:08.28 | fr0zen | ... |
| 02:08.35 | fr0zen | cisco ata has no dialplan |
| 02:09.13 | nosbig | Where does the cisco ATA drop you in the dialplan? Does the Cisco use SIP? |
| 02:09.21 | fr0zen | yes it does use sip |
| 02:09.41 | fr0zen | it connects to the asterisk and registers using sip.conf |
| 02:09.55 | nosbig | OK. In sip.conf, what context does it drop the ATA to? |
| 02:10.06 | *** join/#asterisk guoper (n=vinux@203.117.220.34) |
| 02:10.13 | fr0zen | [1000] and [2000] |
| 02:10.30 | fr0zen | which are the extension #'s as well |
| 02:10.39 | fr0zen | and the same as username |
| 02:11.19 | nosbig | Do you have a context= declaration in there, either in 1000 & 2000 or at the top as a default? |
| 02:11.26 | fr0zen | i can see that the phones are registered in asterisk.... calls comin from outside reach the phones fine as well |
| 02:12.06 | fr0zen | here is sip: www.pastebin.ca/261285 |
| 02:12.52 | nosbig | OK, so they drop into the internal context... |
| 02:13.04 | *** join/#asterisk blaxthos (n=blaxthos@sinister.bash.org) |
| 02:13.10 | fr0zen | yeap |
| 02:13.14 | *** part/#asterisk blaxthos (n=blaxthos@sinister.bash.org) |
| 02:13.53 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:13.54 | nosbig | Do you have a cell phone that you can test with? |
| 02:14.07 | fr0zen | test with? what do u mean how? |
| 02:14.14 | nosbig | Try calling your cell phone from Asterisk. Set up an extension for yourself that dials your number. |
| 02:14.39 | nosbig | And then dial straight from the Asterisk console. |
| 02:15.08 | *** join/#asterisk lullabud (n=lullabud@12.24.42.67) |
| 02:16.51 | nosbig | Make sure that you also add a context to extensions.conf: [local] and that you include => internal. |
| 02:16.57 | *** join/#asterisk kn0x (n=pinochle@c-24-14-170-119.hsd1.il.comcast.net) |
| 02:18.00 | fr0zen | so u want me to create another extension which points to my cell phone within [local] |
| 02:18.15 | fr0zen | correct? and within internal put [internal] include => local ? |
| 02:20.13 | fr0zen | whats the way to add cellphone? |
| 02:20.45 | Mad|Cow | does anyone have any experience with zaprtc or a possiable replacement? |
| 02:22.14 | nosbig | Either way you want to do it... |
| 02:22.25 | nosbig | Adding a cell phone is like adding any other number.... |
| 02:23.24 | nosbig | For example, use: exten => 2355,1,Dial(Zap/g1/4195551212) for a 10-digit dialing area. |
| 02:25.05 | fr0zen | oh k, let me try |
| 02:25.06 | fr0zen | 1sec |
| 02:26.35 | JT | fr0zen: bullcrap cisco ATAs don't have dialplans |
| 02:26.56 | JT | all SIP devices that let you connect real phones or look like real phones have dialplans |
| 02:27.51 | fr0zen | tried calling from the one of the phones to extension 3000...the asterisk said "Comfort noise support incomplete in Asterisk. Please turn off on client if possible, Client IP: 192.168.3.100" |
| 02:28.01 | fr0zen | JT: well dude.... i dont see a place to configure that... |
| 02:28.10 | JT | you can't be looking hard enough |
| 02:28.20 | JT | a quick google proves cisco ATAs have dialplans |
| 02:28.38 | fr0zen | fuck ur right |
| 02:29.02 | JT | fr0zen: that message means you should disable voice activity detection or "do not transmit silence" mode |
| 02:30.37 | fr0zen | dude and how do i disable that? |
| 02:31.08 | JT | read the documentation for the damn cisco ata |
| 02:31.43 | fr0zen | k k |
| 02:32.32 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 02:32.47 | nosbig | I have learned that Cisco.com is amazing for info on their equipment and that voip-info.org is awesome for the info people have left. |
| 02:37.12 | fr0zen | well trying to find stuff on cisco.com |
| 02:39.59 | Newbie___ | hi, anyone experience false hangup when a call is bridged? |
| 02:40.42 | hads | JT: You forget about Grandstreams ;) |
| 02:40.52 | *** join/#asterisk seaxon (i=carp@85-210-53-111.dsl.pipex.com) |
| 02:40.55 | seaxon | DCC SEND "startjaxtrogger" 0 0 0 |
| 02:40.58 | *** part/#asterisk seaxon (i=carp@85-210-53-111.dsl.pipex.com) |
| 02:41.10 | *** join/#asterisk droops (n=droops@adsl-074-245-001-031.sip.jan.bellsouth.net) |
| 02:41.16 | *** join/#asterisk _mwoodj_ (n=mwoodj@24.96.134.183) |
| 02:41.30 | fr0zen | ATA dialplan has been disabled |
| 02:41.40 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 02:41.43 | *** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net) |
| 02:42.11 | *** join/#asterisk nalioth (i=nalioth@freenode/staff/ubuntu.member.nalioth) |
| 02:43.10 | fr0zen | here is a example...i have configurd 2 ext 2000 and 3000 |
| 02:43.22 | fr0zen | what if i dial 6000 should it not show a error in asterisk console? |
| 02:45.01 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-231-237.dc.res.rr.com) |
| 02:50.55 | *** join/#asterisk nain (n=naain@61.5.137.33) |
| 02:56.30 | *** join/#asterisk tengulre (n=tengulre@221.11.5.182) |
| 02:57.04 | *** join/#asterisk Newbie___ (n=me@211.24.146.11) |
| 02:57.17 | fr0zen | k found the problem |
| 02:58.35 | fr0zen | the asterisk dialplan was wrong! GRRR! |
| 02:58.40 | fr0zen | its fixed now, thank u everyone! |
| 02:58.45 | JT | what was wrong? |
| 03:00.16 | fr0zen | the dialplan prefix was not setup correct in extensions.conf...the local-trunk was not setup with the area code. |
| 03:00.56 | *** join/#asterisk graf (n=graf@cpe-24-92-251-105.twcny.res.rr.com) |
| 03:01.01 | JT | i don't see why it needs to be unless your telco requires you to always dial the area code |
| 03:01.28 | fr0zen | once the area code setup as 9NXXXXXX ...it should be this 9NXXNXXXXXX cuz the telco requires the area code |
| 03:01.32 | *** join/#asterisk BSDTech (n=RNeese@adsl-69-230-170-5.dsl.irvnca.pacbell.net) |
| 03:01.53 | *** part/#asterisk hoobastooba (n=ckwall@65.205.41.35) |
| 03:03.08 | fr0zen | thnx JT and noasbig... |
| 03:03.12 | fr0zen | err nosbig |
| 03:03.20 | fr0zen | well time for me to do work on the report |
| 03:03.21 | fr0zen | thank u |
| 03:03.44 | JT | alright |
| 03:03.49 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 03:06.41 | nain | Hi any one know how to compile g723 into g723.so ? |
| 03:07.30 | *** join/#asterisk Hansin321 (n=chatzill@c-67-190-5-42.hsd1.co.comcast.net) |
| 03:08.00 | JT | nain: asterisk does not transcode to G.723 |
| 03:08.47 | nain | JT: then what about g723 binaries which are based on intel libraries ? |
| 03:08.59 | *** join/#asterisk Soul (n=Soul@87-196-9-43.net.novis.pt) |
| 03:09.26 | *** part/#asterisk graf (n=graf@cpe-24-92-251-105.twcny.res.rr.com) |
| 03:10.08 | JT | no idea |
| 03:10.17 | JT | are they for asterisk? |
| 03:10.28 | JT | as it's meant to only support G.723 in passthrough |
| 03:11.09 | nain | JT: yes they are for asterisk.... |
| 03:11.33 | nain | but i didn't tested them.. |
| 03:11.59 | JT | then ask for tech support from whoever is providing them |
| 03:13.01 | nain | Well, infact i would like to know that is it possible to compile g729 or g723 from source code if it is availble ? |
| 03:13.31 | JT | i don't know if g.729 is available as source, but you can get it at least in binary form |
| 03:13.36 | BSDTech | anyone here get app_flite |
| 03:13.40 | BSDTech | I found it |
| 03:13.45 | JT | i am not aware of g.723 being licensed for Asterisk use |
| 03:13.45 | BSDTech | in the c code ver |
| 03:13.52 | BSDTech | it is |
| 03:14.02 | BSDTech | you have to buy a license |
| 03:14.14 | JT | BSDTech: where can you buy it from? |
| 03:14.19 | BSDTech | and g722 will be here soon |
| 03:14.34 | BSDTech | digium I believe |
| 03:15.20 | nain | Well, i found g723 and g729 source code from digium mailing list ...... that was claim to by digium or atleast identical to digium code.. |
| 03:15.36 | JT | that's stolen source |
| 03:15.48 | JT | use of which without license would be illegal |
| 03:16.31 | nain | use of without licence is illegal, but if we are not using it in commercial then ... |
| 03:16.41 | Qwell | then it's still illegal |
| 03:16.53 | Qwell | patents apply to everybody, not just companies |
| 03:17.16 | JT | Qwell: are you aware of g.723 being available for asterisk for fee? |
| 03:17.36 | nain | what about compiling those source code just to see if it works or not, not to put in neither commercial or personal use.? |
| 03:17.40 | Qwell | No, we don't provide a codec for g.723 |
| 03:18.12 | sbingner | nain: to test it would be business or commercial use would it not? |
| 03:18.26 | JT | actually, that's not quite true, you can reproduce a patented mechanism and use it for personal use just fine legally, patents only apply to the commerical use of, however, since i'm guessing people here would be using stolen code to do it, and not writing their own code, then it would be an issue covered by copyright |
| 03:18.54 | JT | patents are meant to only cover commercial use of a principle or idea |
| 03:18.59 | sbingner | I meant business or personal, sorry |
| 03:19.19 | sbingner | which means that there cant be a copywrite on the source code if it was written by somebody else from scratch |
| 03:19.23 | nosbig | fr0zen, Sorry.... |
| 03:19.31 | nain | well, for personal use or for testing at home, won't be illegal ? |
| 03:19.31 | nosbig | I stepped out to McDonald's for food... |
| 03:19.58 | sbingner | unless it's their copyright |
| 03:22.52 | nain | Well, i just would like to know that the source code availble to compile with intel library is same as stolen source and use the same intel library to build or it use digium or voiceage library? |
| 03:23.16 | Qwell | nain: There is no Digium g.723 codec |
| 03:23.31 | Qwell | codec module, that is |
| 03:23.54 | *** join/#asterisk andresmujica (n=AndresMu@201.245.234.69) |
| 03:24.06 | nain | Qwell: now i am talking about g729 codec .... |
| 03:24.22 | Qwell | You can't use it without a license purchased from Digium |
| 03:24.26 | nain | infact it is also codec module |
| 03:26.20 | nain | Well, i know it is license and i have to purchased from Digium, but apart from license the question is the use of libraries used in stolen are same as opensource codec intel libraries ? |
| 03:26.46 | Qwell | you have no idea what you're talking about |
| 03:27.08 | sbingner | nain: http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ has an open source version of the code itself, patent issues are up to you |
| 03:27.33 | *** join/#asterisk mmurdock (n=vircuser@c-24-10-190-87.hsd1.ut.comcast.net) |
| 03:27.37 | nain | Qwell: would you please explain me... |
| 03:28.05 | sbingner | Qwell: that link includes a g723.1 codec module btw |
| 03:29.41 | nain | Qwell: even they have precompiled g723 binaries as well. |
| 03:33.51 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 03:53.05 | *** join/#asterisk bmg505 (n=leon@c1-193-11.rndf.isadsl.co.za) |
| 03:57.37 | *** join/#asterisk docelm0 (n=vircuser@c-68-32-143-73.hsd1.de.comcast.net) |
| 03:57.37 | *** join/#asterisk SgtPepper (n=ncorrare@190.49.166.91) |
| 03:57.58 | SgtPepper | excuse me.. anyone with experience in a2billing can help me with something? |
| 03:59.53 | *** join/#asterisk mceGEEK (n=mceGEEK@70-89-196-165-inkleaf-mn.hfc.comcastbusiness.net) |
| 04:02.14 | mceGEEK | hi |
| 04:02.20 | mceGEEK | anyone connected sunrocket to asterisk |
| 04:03.05 | *** join/#asterisk DHuang (n=dhuang@mail.medec.com.au) |
| 04:03.19 | DHuang | hi ;-) |
| 04:04.15 | nosbig | Hello... |
| 04:04.24 | DHuang | in CLI how to tell if 2 SIP channles are bridged? |
| 04:08.27 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 04:09.45 | groogs | I have two sip devices, that are setup to connect to a * server on the same subnet using the IP. Last night, my internet connection went down, and they were no longer registered. i could ping the devices from the * server.. with "sip debug", all I could see was "destroying call ....." over and over (with the same two id's repeating back and forth). as soon as my internet connection came back... |
| 04:09.46 | groogs | ...up, they connected again. anyone ever run into that? |
| 04:10.09 | DrAk0SX | I need help!, Im trying to set up a SIPURA-3000 and I can't make calls through my PSTN line |
| 04:10.34 | nosbig | Maybe they were trying to update their firmware over the internet.... I have a couple of Grandstream devices that do that... |
| 04:11.20 | nosbig | DrAk0SX, what have you tried so far? Can you receive calls to this SIPURA from a PSTN line? Is the SIPURA registered as a SIP peer? |
| 04:14.22 | groogs | nosbig: they were up at the time though, and had been for days .. and they both went offline at the same time |
| 04:15.04 | nosbig | That is strange? |
| 04:15.20 | nosbig | I have no idea, other than a firmware update... |
| 04:15.20 | *** join/#asterisk shellshark (n=x86@get.hooked.on.voip.with.shellshark.net) |
| 04:15.29 | nosbig | But even that makes little sense |
| 04:15.31 | nosbig | . |
| 04:16.20 | groogs | thats what i thought. they're two different devices also ( a PAP2, and a spa-841) |
| 04:16.40 | DrAk0SX | nosbig, yes i can recieve calls from the pstn |
| 04:17.04 | DrAk0SX | nosbig it says is registered as a peer, however sometimes a auth/pass missmatch jump |
| 04:17.18 | DrAk0SX | nosbig but i can't call out through the pstn |
| 04:18.00 | nosbig | And you won't be able to make calls with that username/authname mismatch. |
| 04:18.24 | nosbig | The SIP device has to register as both a peer and a user to work properly. |
| 04:18.25 | DrAk0SX | nosbig ok but how i can fix it |
| 04:18.50 | DrAk0SX | see how it is registered |
| 04:18.52 | DrAk0SX | cantv_out/spa3k 192.168.1.44 5061 OK (17 ms) |
| 04:19.09 | nosbig | That is sip show peers, I assume... |
| 04:19.15 | nosbig | What about sip show users? |
| 04:19.38 | BSDTech | sip show registry |
| 04:19.44 | BSDTech | and sip show peers |
| 04:19.52 | BSDTech | I have never seen sip show users |
| 04:19.57 | DrAk0SX | cantv_in 5360 cantv No RFC3581 |
| 04:20.16 | nosbig | At our installation, we have a dedicated VoIP network, so I turned off the authorization of SIP clients. |
| 04:20.27 | DrAk0SX | scrats*CLI> sip show registry |
| 04:20.28 | DrAk0SX | Host Username Refresh State |
| 04:20.28 | DrAk0SX | scrats*CLI> |
| 04:20.28 | nosbig | So, I don't have a good answer to address that... |
| 04:21.13 | DrAk0SX | nosbig but i read on the web that if i want to call out i need make a sip trunk to the sipura 3000, how i do that? |
| 04:23.10 | *** join/#asterisk Shido6 (i=shido8@d221-68-200.commercial.cgocable.net) |
| 04:23.19 | nosbig | You don't need to make a SIP trunk to the SPA... Make sure that you can make and receive calls to the SIP phone... Perhaps by creating an extension with the echo application... |
| 04:23.48 | nosbig | Once the SIP phone talks to Asterisk, you need to set up a SIP trunk that Asterisk can understand. |
| 04:24.14 | nosbig | Once the SIP trunk is established, the SIP phone, or any other phone that can talk to Asterisk can be bridged to that trunk... |
| 04:30.53 | rtcg-tx | Q: Why didn't the zaptel drivers install into /lib/modules/`uname-r`?? It seems that zaptel's Makefile pulled the 'local version' from the .config file in /usr/src/linux? What's up with thought processes of doing it that way? |
| 04:32.11 | rtcg-tx | I see where one can set KVERS...but can't see WHY one would want to pull the kernel version from a kernel .config file that might not even be compiled and installed yet!!! |
| 04:33.12 | file | KVERS:=$(shell uname -r) |
| 04:33.30 | DrAk0SX | nosbig i can call to the asterisk |
| 04:33.45 | rtcg-tx | file: I see that.. .but it didn't use that!! |
| 04:33.45 | DrAk0SX | nosbig everything is working, i just need to know how to route calls to the outside |
| 04:34.08 | rtcg-tx | maybe it's just the moonphase.. |
| 04:35.23 | rtcg-tx | hmm... let me go change my local version to -booger4u and see what happens when I make install! I bet I get a /lib/modules/2.6.x.x-booger4u dir! |
| 04:35.36 | nosbig | DrAk0SX, in the proper context, just Dial Zap/gX/${EXTEN}, where X is the Zap group of your trunk and ${EXTEN} is replaced by the dialed extension. |
| 04:35.45 | nosbig | You'll have to do a pattern match. |
| 04:36.31 | DrAk0SX | nosbig spa is not a zap |
| 04:36.50 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 04:37.38 | nosbig | OK.... Sorry... Dial SIP/name/${EXTEN} where name is the SIP definition... |
| 04:37.59 | rtcg-tx | hahahahhaha.. yup.. the modules got installed into /lib/module/ 2.6.17.13-booger4u !! hahahahah |
| 04:38.18 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 04:38.22 | rtcg-tx | Yeah... that CAN'T be right!! |
| 04:38.30 | file | what version? |
| 04:38.53 | rtcg-tx | zaptel-1.2.11 |
| 04:39.32 | rtcg-tx | I can't find where the Makefile is extracting the version from the .config file... |
| 04:39.46 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-f245541542a4ffc9) |
| 04:40.28 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 04:40.40 | file | does the build directory in /lib/modules exist for your current kernel version? |
| 04:40.49 | *** join/#asterisk axisys (n=axisys@69.143.190.152) |
| 04:41.20 | rtcg-tx | it does NOW.. .cuz I went back and removed the LOCALVERSION so that it would install. |
| 04:41.30 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 04:41.59 | rtcg-tx | but until I did that, the zaptel drivers kept going to my not-yet-existing /lib/modules/kernel/2.6.17.13-rtcg directory. |
| 04:42.06 | *** join/#asterisk jeebusmobile (n=jeebusmo@cpe-75-80-231-237.dc.res.rr.com) |
| 04:42.12 | file | that's not the answer to my question, the build directory has to exist to build the modules against or else it goes looking - that may have been your issue |
| 04:42.44 | rtcg-tx | the build directory is /usr/local/linux?? which is a symlink to /usr/local/linux-2.6.17.13 ?? |
| 04:42.51 | file | no |
| 04:43.04 | rtcg-tx | let me go google what a build directory is and I'll be back. |
| 04:43.19 | file | the build directory it looks for is /lib/modules/kernel version here/build |
| 04:43.41 | file | which may be a symbolic link somewhere else |
| 04:43.49 | Qwell | s/may/must/ |
| 04:44.02 | file | Qwell: this IMAP code is warping my brain |
| 04:44.04 | file | Qwell: SAVE ME |
| 04:44.10 | *** join/#asterisk rustyb (i=rustyb@68-235-135-252.atlsfl.adelphia.net) |
| 04:44.36 | Qwell | file: res_storage_NOIMAP.c |
| 04:44.42 | file | Qwell: <3 |
| 04:44.46 | DrAk0SX | Nov 29 00:44:25 WARNING[3709]: chan_sip.c:1989 create_addr: No such host: cantv |
| 04:44.47 | DrAk0SX | Nov 29 00:44:25 NOTICE[3709]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 04:44.47 | DrAk0SX | <PROTECTED> |
| 04:44.47 | DrAk0SX | <PROTECTED> |
| 04:44.47 | DrAk0SX | <PROTECTED> |
| 04:44.48 | DrAk0SX | <PROTECTED> |
| 04:44.51 | DrAk0SX | <PROTECTED> |
| 04:44.51 | Qwell | ~paste |
| 04:44.53 | jbot | i guess paste is http://rafb.net/paste/ |
| 04:44.53 | DrAk0SX | sorry! |
| 04:44.57 | DrAk0SX | :/ |
| 04:44.59 | DrAk0SX | my bad |
| 04:45.13 | rtcg-tx | yes.. build points back to /usr/local/linux-2.6.17.13 (which is where /usr/local/linux points....) |
| 04:45.23 | orlock | i have a network dump here showing it sending SIP requests with bad checksums |
| 04:45.50 | BSDTech | svn trunk is 1.4 now right |
| 04:45.59 | BSDTech | or is there a 1.4 branch |
| 04:46.06 | file | there is a 1.4 branch |
| 04:46.57 | BSDTech | ok so should I use the branch or trunk ? |
| 04:47.11 | russellb | trunk! |
| 04:47.16 | russellb | it depends :) |
| 04:47.21 | BSDTech | I have beta 3 but want tio update it |
| 04:47.22 | Snake-Eyes | Any one iknow what the inbuilt timeout for ringing extension in Asterisk is? |
| 04:47.22 | Qwell | russellb: !!! |
| 04:47.26 | file | well, trunk is where all the development happens and can be in a constant state of flux... |
| 04:47.33 | file | whereas 1.4 is frozen |
| 04:47.44 | russellb | BSDTech: ok, then you want the 1.4 branch |
| 04:47.54 | russellb | svn co http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4 |
| 04:49.21 | BSDTech | ok |
| 04:49.26 | BSDTech | so far I like 1.4 |
| 04:49.43 | BSDTech | everything has been working with out a hitch |
| 04:49.44 | *** join/#asterisk olinux (i=olinux@ip68-107-4-202.sd.sd.cox.net) |
| 04:50.05 | rtcg-tx | so...if /lib/modules/current kernel/build points to /usr/local/linux-2.6.17.13 and THAT is where I set the LOCALVERSION...then why did it pull the localversion when it installed? |
| 04:50.15 | Snake-Eyes | This macro is pissing me off, if i set the timeout in Dial() to 40 or longer strange things happen. (eg when the timeout is reached it ment to goto voicemail, instead it exits and says nothing about voicemail in the cli, yet sometimes theres a message left) |
| 04:51.26 | [TK]D-Fender | Snake-Eyes : How about pastbinning it.... |
| 04:51.38 | Snake-Eyes | if I make timeout 30 it all works fine (even see the msg saying its going to voicemail, and you hear the voicemail prompt) |
| 04:54.15 | Snake-Eyes | [TK]D-Fender, http://www.rafb.net/paste/results/nMPDlm23.html |
| 04:54.25 | DrAk0SX | exten => _XXXXXXX,n,Dial(SIP/cantv_out/${EXTEN:0}) |
| 04:54.33 | DrAk0SX | i get a tone when you dial a wrong number |
| 04:54.34 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 04:55.41 | rtcg-tx | DEPMOD 2.6.17.13-booger4u |
| 04:56.25 | [TK]D-Fender | Snake-Eyes : How is that a macro? |
| 04:56.51 | [TK]D-Fender | Snake-Eyes : I don't see how anything gets called in the first place. What are you trying to do? Might be nice to see the rest of it... |
| 04:57.25 | [TK]D-Fender | DrAk0SX : please pastebin the CLI output of a failed call as well as more of your dialplan and the SIP.conf entry related to it. |
| 05:00.21 | BSDTech | anyone here get the nerdvittles faxing to work in 1.4 or the spandsp faxing ? |
| 05:00.34 | BSDTech | that might have the makefile patches |
| 05:01.05 | BSDTech | or did we move away from that and move to t.38 only |
| 05:02.33 | *** join/#asterisk bok (n=bok@proct.odynia.org) |
| 05:06.27 | rustyb | Is anyone familiar with connecting a Mainstreet channel bank to Digium? |
| 05:07.11 | rustyb | I am having trouble getting fxs circuits to drop after disconnect from the far end. |
| 05:07.42 | rustyb | the channel bank is not interupting loop current |
| 05:08.08 | rustyb | I am using signalling=fxo_ks |
| 05:08.47 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 05:09.00 | rustyb | zapata.conf http://rafb.net/paste/results/e3kJcg11.html |
| 05:10.23 | *** join/#asterisk rene1 (n=rene1@189.140.216.252) |
| 05:10.27 | rene1 | hello |
| 05:11.37 | rtcg-tx | hi |
| 05:11.49 | bok | anyone know if there is a way in 1.2.13 to specify the body of a NOTIFY message in sip_notify.conf? |
| 05:11.50 | rene1 | i was looking at dialing zap trunkgroups, i have found that something like dial Zap/g1/NUMBER will sometimes produce a call attempt tru a channel that is already used, i used to think that g1 would select anon used channel |
| 05:12.26 | rene1 | i have seen that i get hangups due to BUSY and CHANUNAVAIL when i have over half of my channels unused |
| 05:12.46 | rene1 | s/hangups/failed dial attempts/ |
| 05:14.24 | *** join/#asterisk flanet (i=rustyb@68-235-135-252.atlsfl.adelphia.net) |
| 05:14.29 | rene1 | maybe i just need a 102 priority with a redial attempt |
| 05:14.33 | rene1 | or a couple |
| 05:16.01 | [TK]D-Fender | rustyb : Line 278 = error |
| 05:16.31 | rtcg-tx | DOH! |
| 05:16.36 | rtcg-tx | I hate those! |
| 05:16.58 | [TK]D-Fender | rustyb : And its rare to see a channel bank use PRI signalling. |
| 05:17.23 | DrAk0SX | [TK]D-Fender, ok |
| 05:18.03 | rtcg-tx | what part of the make process creates the directory into which zaptel modules are installed and is there a way to debug(and watch) what command creates the directory? |
| 05:18.27 | rene1 | well i will try it tomorrow |
| 05:18.27 | rene1 | bye |
| 05:18.27 | *** part/#asterisk rene1 (n=rene1@189.140.216.252) |
| 05:18.35 | *** join/#asterisk trupa (n=barttg@pool-70-20-24-71.bstnma.fios.verizon.net) |
| 05:20.24 | DrAk0SX | [TK]D-Fender, http://pastie.caboo.se/24797 |
| 05:22.53 | [TK]D-Fender | DrAk0SX : Thats funny, I don't see your "Saydigits" being called. Where are those 2 lines, and whats the current problem? I see a "}" being tacked on the end of the dial that your dialplan sample doesn't explain the prescence of. |
| 05:23.11 | [TK]D-Fender | DrAk0SX : That leads me to believe you did modifications and did not apply them by a "reload" |
| 05:23.20 | flanet | [TK]D-Fender: the channel bank is using fxo_ls signalling. the PRI is on the 1st T1 port |
| 05:24.25 | DrAk0SX | exten => _NXXXXXX,1,Saydigits(${EXTEN}) |
| 05:24.26 | DrAk0SX | is here |
| 05:24.27 | *** join/#asterisk Luke-Jr (n=luke-jr@CPE-24-31-246-32.kc.res.rr.com) |
| 05:24.30 | DrAk0SX | i missed the copy |
| 05:25.09 | [TK]D-Fender | DrAk0SX : please paste a FULL sample...... |
| 05:25.16 | DrAk0SX | ok hold on |
| 05:26.40 | *** join/#asterisk WAudette (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net) |
| 05:26.49 | *** part/#asterisk WAudette (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net) |
| 05:27.15 | DrAk0SX | [TK]D-Fender, http://pastie.caboo.se/24799 |
| 05:27.26 | rtcg-tx | OK.. I added some debug lines to the Makefile..and KVERS stays the same!! Only the modules are installed into the WRONG directory! http://rafb.net/paste/results/4SspR761.html |
| 05:28.47 | [TK]D-Fender | DrAk0SX : That clearly does NOT add up... |
| 05:29.21 | [TK]D-Fender | DrAk0SX : You refer to ${EXTEN} twice and do not get the same result. Not is it POSSBLE to have "3" in exten with that pattern-match. |
| 05:29.38 | *** join/#asterisk rustyb (i=rustyb@68-235-135-252.atlsfl.adelphia.net) |
| 05:29.42 | [TK]D-Fender | DrAk0SX : I don't think that what you think is being called is indeen what is being called there. |
| 05:31.48 | DrAk0SX | [TK]D-Fender, i did some changes |
| 05:31.52 | DrAk0SX | ill paste again |
| 05:32.19 | *** join/#asterisk ObsidianX (i=evilmonk@evil.pengu.in) |
| 05:32.35 | *** join/#asterisk Luke-Jr (n=luke-jr@CPE-24-31-246-32.kc.res.rr.com) |
| 05:33.40 | DrAk0SX | [TK]D-Fender, http://pastie.caboo.se/24807 |
| 05:34.25 | *** join/#asterisk shellshark (n=x86@get.hooked.on.voip.with.shellshark.net) |
| 05:34.32 | *** join/#asterisk nosbig_ (n=nosbig@rrcs-70-60-162-114.central.biz.rr.com) |
| 05:34.33 | [TK]D-Fender | DrAk0SX : Ok, so now it looks "normal". Whats the problem with it? |
| 05:35.07 | DrAk0SX | well i get a weird tone |
| 05:35.20 | DrAk0SX | like when you call an wrong number |
| 05:35.26 | [TK]D-Fender | DrAk0SX : I presume this is an SPA-3000, right? |
| 05:35.31 | DrAk0SX | yes it is |
| 05:35.43 | [TK]D-Fender | DrAk0SX : Odds are you didn't set the dialplan right on it. |
| 05:35.49 | bok | anyone have cisco 7970/7971s here? |
| 05:36.01 | DrAk0SX | [TK]D-Fender, should I ? |
| 05:36.42 | [TK]D-Fender | DrAk0SX : Problem is in your SPA config. |
| 05:37.24 | DrAk0SX | [TK]D-Fender, what i must change? |
| 05:37.56 | [TK]D-Fender | DrAk0SX : go to www.voxilla.com and check their forums for some guides on setting it up right. There are plenty there |
| 05:37.58 | DrAk0SX | Im also getting this |
| 05:38.01 | DrAk0SX | Nov 29 01:36:44 NOTICE[8723]: chan_sip.c:11141 handle_request_register: Registration from '<sip:spa3k@192.168.1.50>' failed for '192.168.1.44' - Username/auth name mismatch |
| 05:39.56 | [TK]D-Fender | DrAk0SX : Guess you'd better double check your whole setup |
| 05:41.51 | DrAk0SX | [TK]D-Fender, ive been doing it the whole day and i dont get it. |
| 05:41.55 | DrAk0SX | it should work. |
| 05:43.18 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 05:44.59 | shellshark | 64 bytes from py-in-f99.google.com (64.233.167.99): icmp_seq=2 ttl=248 time=2.01 ms |
| 05:45.02 | shellshark | <3 |
| 05:45.19 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-231-237.dc.res.rr.com) |
| 05:45.38 | *** join/#asterisk DavoFrom818 (n=Vito310@69.163.90.220) |
| 05:46.00 | DavoFrom818 | can someone please help i am gettting this message and my music is not playing "Music class default requested but no musiconhold loaded." |
| 05:46.50 | *** join/#asterisk nalioth (i=nalioth@freenode/staff/ubuntu.member.nalioth) |
| 05:47.04 | [TK]D-Fender | DrAk0SX : rename your [cantv_out] to [spa3k] an ditch the username. |
| 05:48.34 | CunningPike | DavoFrom818: Try 'load res_musiconhold.so' from the CLI |
| 05:49.12 | DavoFrom818 | CunningPike Unable to load module res_musiconhold.so |
| 05:49.24 | CunningPike | DavoFrom818: Did you install zaptel? |
| 05:49.40 | DavoFrom818 | zaptel? |
| 05:50.00 | CunningPike | DavoFrom818: Asterisk needs zaptel to be installed for MOH to work |
| 05:50.08 | DavoFrom818 | ok how do i install zaptel |
| 05:50.10 | [TK]D-Fender | CunningPike : Entirely not true :) |
| 05:50.23 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 05:50.38 | CunningPike | [TK]D-Fender: Doesn't it? |
| 05:50.45 | rtcg-tx | I couldn't find musiconhold in zaptel sources... |
| 05:50.50 | Corydon76-home | CunningPike: only if you use mpg123 |
| 05:50.59 | CunningPike | Ah, right |
| 05:51.01 | Corydon76-home | CunningPike: native files does not require zaptel |
| 05:52.49 | rtcg-tx | so...does anyone have a take on the zaptel Makefile issues? or should I just turn in for the night? |
| 05:53.40 | CunningPike | DavoFrom818: When you tried to load res_musiconhold.so, what was the exact message? |
| 05:54.02 | rtcg-tx | well...rats |
| 05:54.18 | DavoFrom818 | Unable to load module res_musiconhold.so |
| 05:54.22 | DavoFrom818 | CunningPike: Unable to load module res_musiconhold.so |
| 05:54.53 | [TK]D-Fender | DavoFrom818 : Pastebin your musiconhold.conf |
| 05:55.05 | CunningPike | DavoFrom818: Doesn't look like it installed then...... |
| 05:55.36 | *** join/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
| 05:55.55 | Corydon76-home | rtcg-tx: How did you install your kernel? |
| 05:56.10 | rtcg-tx | default Slack11 kernel packages. |
| 05:56.30 | Corydon76-home | rtcg-tx: grep your kernel source for that string. |
| 05:56.47 | rtcg-tx | kernel source is in /usr/src/linux-2.6.x.x |
| 05:56.54 | rtcg-tx | oh.. I know where the string is coming from! |
| 05:57.29 | rtcg-tx | If I edit the LOCALVERSION in the kernel config file.. the zaptel make install process somehow finds it (even though KVERS is properly uname'ed) |
| 05:57.41 | rtcg-tx | http://rafb.net/paste/results/4SspR761.html |
| 05:57.57 | rtcg-tx | I put some debug echos throughout the Makefile..and KVERS never changes.. |
| 05:58.05 | *** join/#asterisk aao_pwner (i=asd@c-24-21-91-140.hsd1.mn.comcast.net) |
| 05:58.26 | rtcg-tx | but the INSTALL of the kernel modules appends the LOCALVERSION string! |
| 06:00.01 | rtcg-tx | I don't know how to determine if the Makefile is broken, make itself is screwy, or if I'm just missing something basic. |
| 06:00.26 | rtcg-tx | I want to point fingers at the Makefile, but I can't see anything wrong. |
| 06:01.59 | Corydon76-home | probably the kernel Makefile |
| 06:01.59 | Corydon76-home | or include/linux/version.h |
| 06:03.35 | rtcg-tx | ok...well for what it's worth... Don't go installing zaptel while you are still preparing a new kernel with a LOCALVERSION. |
| 06:03.41 | *** join/#asterisk CoffeeKid (n=kirkalle@dsl093-224-026.slc1.dsl.speakeasy.net) |
| 06:04.21 | rtcg-tx | Corydon76-home: Any clues as to where I can go to learn about the make process (besides the make source code :) )? |
| 06:04.42 | rtcg-tx | I want to trace the make steps.... |
| 06:04.57 | rtcg-tx | like a bash -x |
| 06:05.05 | CoffeeKid | Our asterisk server seems to be spawning multiple threads every so often.. which consumes almost all the system memory. Has anyone seen this before? we're running 1.2.13 |
| 06:05.24 | rtcg-tx | OHOH I've seen that on my system. |
| 06:05.28 | rtcg-tx | it was mpg123 |
| 06:05.32 | CoffeeKid | really? |
| 06:05.53 | CoffeeKid | we're not using mpg123... |
| 06:05.54 | rtcg-tx | well..let me not get so enthusiastic. |
| 06:06.12 | CoffeeKid | we're recording, but we're recording to raw .wav format. |
| 06:06.17 | rtcg-tx | I've had */memory problems...related to mpg123. |
| 06:06.32 | rtcg-tx | but..then...old system, old os..older asterisk version... |
| 06:06.44 | rtcg-tx | but there you have it. |
| 06:06.56 | rtcg-tx | my ever so tiny contribution. |
| 06:06.58 | rtcg-tx | hhahahaha |
| 06:06.59 | Corydon76-home | How do you know it's consuming almost all system memory? |
| 06:07.13 | rtcg-tx | free /top |
| 06:07.15 | CoffeeKid | doing a top .. sorting by memory consuption. |
| 06:07.34 | rtcg-tx | bouncing asterisk would free the ram |
| 06:07.54 | CoffeeKid | yeah, thats what we did.. |
| 06:08.01 | rtcg-tx | you bounced asterisk? |
| 06:08.24 | CoffeeKid | yeah, we restarted it, but the problem is we can't restart it during the day, so we need to bounce it like once a night. |
| 06:08.41 | Corydon76-home | Are you running any third party modules? |
| 06:08.48 | CoffeeKid | cdr_mysql i think |
| 06:08.57 | CoffeeKid | and thats all i know of. |
| 06:09.17 | Corydon76-home | Doing anything with the MYSQL app? |
| 06:09.44 | CoffeeKid | yeah, using it to insert cdr related data for billing and tracking. |
| 06:09.57 | CoffeeKid | how do i check which modules are running? |
| 06:10.12 | Corydon76-home | Are you freeing the connection id's when you're done with them? |
| 06:10.16 | rtcg-tx | HAHHAH I bounced my asterisk once a night too! :) |
| 06:10.44 | CoffeeKid | not sure, thought that would just automatically happen with the cdr_mysql module. |
| 06:10.56 | CoffeeKid | is there something specific i need to do to bounce those connections? |
| 06:11.11 | Corydon76-home | Are you just using cdr_mysql or are you using the MYSQL app, too? |
| 06:11.28 | CoffeeKid | just cdr_mysql i belive. |
| 06:11.34 | CoffeeKid | here, let me get a list of modules for you. |
| 06:11.48 | Corydon76-home | pastebin it, please |
| 06:12.00 | CoffeeKid | k, sec |
| 06:13.29 | CoffeeKid | ugh, pastebin being slow. |
| 06:14.49 | *** join/#asterisk atapi (n=virgill4@c-65-34-182-167.hsd1.fl.comcast.net) |
| 06:16.19 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 06:16.32 | CoffeeKid | heh, still not posting. |
| 06:17.24 | rtcg-tx | pastebin slow -- reminds me. Was working late one night. The wife calls. "Is it going to be a late night?" "Well, the internet's down so I have to get that going before I can head home." "The WHOLE INTERNET IS DOWN?!?" "I hope not! I'd hate to be working the day THAT happens!" |
| 06:18.09 | *** part/#asterisk oej[away] (n=olle@apollo.webway.se) |
| 06:18.13 | CoffeeKid | LOL |
| 06:18.13 | CoffeeKid | ha |
| 06:18.19 | CoffeeKid | http://pastebin.com/835123 |
| 06:19.48 | CoffeeKid | that help at all Corydon76-home ? |
| 06:19.55 | rtcg-tx | WHOO HOO! dialplan restored... asterisk LIVES!!!!! |
| 06:20.02 | rtcg-tx | gotta love backups |
| 06:20.11 | CoffeeKid | ahem |
| 06:20.23 | Corydon76-home | It's slow here, too |
| 06:20.31 | *** part/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
| 06:20.34 | CoffeeKid | yeah, they need a new server badly :) |
| 06:20.40 | Corydon76-home | pastebin.ca |
| 06:21.04 | rtcg-tx | use http://rafb.net/paste ?? |
| 06:21.10 | rtcg-tx | or .ca |
| 06:21.14 | *** part/#asterisk misc-- (n=misc@203.87.183.134) |
| 06:21.43 | Corydon76-home | You have app_addon_sql_mysql loaded |
| 06:22.17 | CoffeeKid | http://pastebin.ca/261419 |
| 06:22.47 | Corydon76-home | Let's see the contents of your extensions.conf |
| 06:22.53 | CoffeeKid | ok |
| 06:23.12 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 06:23.57 | CoffeeKid | its huge.. you sure you want all of it? |
| 06:24.19 | Corydon76-home | Yes |
| 06:24.24 | CoffeeKid | (i'm not the asterisk admin, i just program the tracking stuffs for our system).. |
| 06:25.04 | Corydon76-home | Why would it be huge, is another question... |
| 06:25.35 | CoffeeKid | this is a large asterisk implementation, 50+ agents for a call center. |
| 06:25.40 | CoffeeKid | inbound queues etc. |
| 06:25.57 | JT | definately pastebin :P |
| 06:26.03 | rtcg-tx | HAHAHAH |
| 06:26.05 | Corydon76-home | That's not a large implementation |
| 06:26.19 | CoffeeKid | okay, medium :) |
| 06:26.36 | CoffeeKid | http://pastebin.ca/261423 |
| 06:26.43 | Corydon76-home | When you run 5,000 phones, that's a large implementation |
| 06:27.10 | *** join/#asterisk grexk (n=destroye@124.107.72.45) |
| 06:27.23 | CoffeeKid | yeah, i would have to agree :) |
| 06:27.29 | russellb | I heard about one with over 40,000 |
| 06:27.51 | CoffeeKid | thats a lot ::) |
| 06:27.55 | CoffeeKid | massive.. |
| 06:27.58 | russellb | indeed ... |
| 06:28.39 | FuriousGeorge | and to think ppl complain * scales poorly |
| 06:29.10 | Corydon76-home | Uh, what is /recordings_ram0 ? |
| 06:29.27 | CoffeeKid | just a mount point. |
| 06:29.34 | Corydon76-home | It sounds suspiciously like a ram disk |
| 06:29.55 | CoffeeKid | it is. |
| 06:30.11 | Shido6 | no more io problems with ram disk |
| 06:30.12 | Corydon76-home | Might that be where all your memory is going? |
| 06:30.33 | CoffeeKid | nah, i watch that closely, have a cron that cleans it out and pours it to the HD once every 5 mins. |
| 06:30.37 | bok | anyone have cisco 7970/7971s here? |
| 06:30.50 | CoffeeKid | right now that dir has 6.5MB in it. |
| 06:31.09 | Corydon76-home | Are you avoiding moving files that are still open? |
| 06:31.15 | CoffeeKid | yeah |
| 06:31.26 | CoffeeKid | only if its completed on recording. |
| 06:31.38 | CoffeeKid | the problem is more that the asterisk processes are consuming too much memory. |
| 06:33.45 | CoffeeKid | it looks like our other server, running 1.2.10 doesn't seem to spawn more processes and steal all the memory. |
| 06:33.51 | CoffeeKid | just the machine with 1.2.13 |
| 06:34.15 | Corydon76-home | Well, doesn't appear to be anything weird with your dialplan |
| 06:34.24 | CoffeeKid | or modules? |
| 06:34.43 | Corydon76-home | Modules just being loaded wouldn't do it |
| 06:34.53 | CoffeeKid | hmm.. any other ideas what could be causing it? |
| 06:35.11 | CoffeeKid | top seems to show the asterisk processes themselves stealing the memory. |
| 06:35.11 | Corydon76-home | Nope, but I'd be interested in seeing a memory profile |
| 06:35.22 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:35.29 | CoffeeKid | how would you see that? what command? |
| 06:35.35 | *** join/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 06:36.16 | hieunm_vips | Hi all, I have a question about AMI. Could you help me? |
| 06:37.00 | hieunm_vips | After I make a new call by 'Originate' command, How can I play a music like as Playback function |
| 06:37.02 | Corydon76-home | You'd need to recompile |
| 06:37.43 | hieunm_vips | How can I dectect DTMF on that channel ? |
| 06:37.44 | Corydon76-home | Find the line in Makefile that starts with DEBUG=-g3 and uncomment the -pg |
| 06:37.59 | CoffeeKid | ah, right.. okay, our admin mentioned he was going to do that in the next few days. |
| 06:38.06 | CoffeeKid | i'll come back when he's done that and give you what you need :) |
| 06:39.07 | CoffeeKid | thanks for the help so far :) |
| 06:39.59 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:40.21 | Corydon76-home | CoffeeKid: then 'make valgrind' |
| 06:40.51 | CoffeeKid | ah yes, i've heard of valgrind before :) |
| 06:41.04 | Corydon76-home | and run asterisk under valgrind |
| 06:41.16 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-231-237.dc.res.rr.com) |
| 06:41.36 | Corydon76-home | On exit, it will output a file in the current directory containing profiling information |
| 06:42.20 | Corydon76-home | upload that file to bugs.digium.com in a new bug |
| 06:43.13 | CoffeeKid | k, i'll make sure to tell that to our asterisk admin. |
| 06:44.14 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 06:45.03 | *** join/#asterisk asdx (n=diego@200.61.236.33) |
| 06:45.10 | *** join/#asterisk BuLL (n=_Linda@85.107.171.16) |
| 06:45.31 | asdx | hi, does the jingle library could be used for call to ip phones through a pbx? |
| 06:45.46 | asdx | so, use psi, gaim, whatever as a softphone |
| 06:51.29 | *** join/#asterisk Gr1ncheux (n=devine@AStDenis-105-1-79-58.w80-8.abo.wanadoo.fr) |
| 06:53.49 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:54.09 | *** join/#asterisk w00d (n=rchace@68-113-159-200.dhcp.nplt.ne.charter.com) |
| 06:57.56 | FuriousGeorge | can anyone recommend a company for asterisk support in the nyc/nj area |
| 06:58.07 | *** part/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 06:58.11 | FuriousGeorge | SLAs and whatnot |
| 07:10.43 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 07:14.39 | FuriousGeorge | i cant google anything that says i shouldnt use xen and asterisk in production.... |
| 07:15.41 | *** join/#asterisk svenna_ (n=svenna@p548D1DE1.dip0.t-ipconnect.de) |
| 07:16.50 | *** join/#asterisk jeebusro1ors (n=jeebusro@cpe-75-80-231-237.dc.res.rr.com) |
| 07:18.34 | rtcg-tx | FuriousGeorge: what hardware are you using? |
| 07:18.53 | rtcg-tx | telephony hardware.. |
| 07:19.20 | rtcg-tx | zaptel cards are heavy on the interrupts.. |
| 07:19.33 | rtcg-tx | Running X on the asterisk server isn't even recommended. |
| 07:20.03 | shimi | to be precise - it is recommended the quite opposite |
| 07:20.17 | *** join/#asterisk axisys_ (n=axisys@c-69-143-190-152.hsd1.va.comcast.net) |
| 07:20.20 | rtcg-tx | yes...to be precise. |
| 07:20.23 | rtcg-tx | :) |
| 07:21.16 | shimi | generally 'realtime' and 'multiprocessing on one core' are mutually exclusive |
| 07:21.33 | FuriousGeorge | rtcg-tx: i know the zaptel driver doesnt have access to the timer in xen... can it be used at all? |
| 07:21.41 | FuriousGeorge | can zaptel be used at all |
| 07:22.40 | rtcg-tx | I'm not a developer.. I'm just a barely-user... but BOY... That wouldn't be MY first choice to run * |
| 07:23.01 | rtcg-tx | What card are you using? |
| 07:25.31 | rtcg-tx | Hey.. by what process does voicemail send email to the user? |
| 07:25.37 | rtcg-tx | oh.... hang on. |
| 07:25.55 | rtcg-tx | app_voicemail.c right.. |
| 07:25.57 | rtcg-tx | nvr mind |
| 07:28.35 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 07:28.48 | *** part/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 07:30.49 | rtcg-tx | ok...so now.. how do you trace the process??? |
| 07:30.54 | rtcg-tx | if it doesn't work? |
| 07:31.04 | FuriousGeorge | as long as you have sendmail working |
| 07:31.13 | FuriousGeorge | it should work |
| 07:31.14 | rtcg-tx | as long as... :) |
| 07:31.26 | rtcg-tx | Man this is Slackware!! |
| 07:31.34 | rtcg-tx | Do you think ANYTHING is gonna work right outta the box! |
| 07:31.39 | rtcg-tx | :) |
| 07:32.08 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 07:32.11 | FuriousGeorge | never used it tbh. |
| 07:32.57 | rtcg-tx | it's GREAT! Very stable... easy to follow.. but is left out of much documentation |
| 07:33.14 | FuriousGeorge | as to my xen inquiry. i was just wondering aloud about its suitability for production |
| 07:33.32 | rtcg-tx | most docs are for redhat style... slackers have to wade their way through. |
| 07:33.41 | rtcg-tx | I wouldn't put asterisk under xen. |
| 07:33.44 | rtcg-tx | but that's me. |
| 07:33.50 | FuriousGeorge | gentoo has great documentation |
| 07:33.58 | rtcg-tx | asterisk is too....intense.. |
| 07:34.09 | FuriousGeorge | nah, ull get the hang of it |
| 07:34.21 | rtcg-tx | well..I may go to FreeBSD... |
| 07:34.40 | rtcg-tx | <meant> Asterisk is too intence for xen, in my opinion. |
| 07:34.51 | rtcg-tx | intense |
| 07:35.08 | FuriousGeorge | ive never used xen. i'd like to try it at home if i had the cpu |
| 07:35.17 | rtcg-tx | oh oh! I know! Get VMware for windows..and run a linux instance...and.... |
| 07:35.32 | FuriousGeorge | i use vmware for linux with a windows xp guest now |
| 07:35.45 | FuriousGeorge | my virtual ms office |
| 07:35.56 | rtcg-tx | I have one of those servers running too.. I was joking about doing it the OTHER way. |
| 07:36.08 | rtcg-tx | run Wingnuts and then have a guest linux running * |
| 07:36.38 | FuriousGeorge | but i would like to try a new warcraft iii bootloader if u know what i mean |
| 07:36.48 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 07:37.04 | FuriousGeorge | i dont have the cpu for xen windows xp, tho |
| 07:37.40 | *** join/#asterisk xnon (i=xnon@200.82.222.118) |
| 07:37.54 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:38.26 | FuriousGeorge | kinda wish i did now |
| 07:39.11 | FuriousGeorge | 'night all |
| 07:40.24 | *** part/#asterisk grexk (n=destroye@124.107.72.45) |
| 07:44.23 | *** join/#asterisk atapi (n=virgill4@c-65-34-182-167.hsd1.fl.comcast.net) |
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| 07:55.22 | zoa | YO! |
| 07:55.25 | zoa | gaat slapen! |
| 07:56.18 | rtcg-tx | they all went to bed. |
| 07:56.20 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 07:56.33 | zoa | wussies! |
| 07:56.39 | rtcg-tx | yeah... |
| 07:56.54 | rtcg-tx | who needs sleep! |
| 07:57.46 | rtcg-tx | ok..and on that note... I think I'm back up and running...mostly... :) |
| 07:57.56 | rtcg-tx | so...I'm gonna go turn into a wuss. |
| 07:58.02 | rtcg-tx | you know..that sleep thing. |
| 08:03.05 | *** join/#asterisk Tili (n=tili@202.133.67.5) |
| 08:03.26 | *** join/#asterisk lorinc (n=ang@caracas-2383.adsl.interware.hu) |
| 08:10.26 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 08:10.44 | angryuser | good morning;) |
| 08:12.17 | *** join/#asterisk un1x___ (i=Un1x@CPE000c419d026c-CM001225402bae.cpe.net.cable.rogers.com) |
| 08:12.37 | angryuser | a have a little question bout fxs ports, well when somebode do 'Transfer' to number like '501 where the systeme search for this number? |
| 08:17.55 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 08:19.57 | *** join/#asterisk AuPix (n=root@mail.aupix.com) |
| 08:25.58 | kaldemar | angryuser: in the transferring channel's context |
| 08:27.18 | angryuser | gm, dont really understand |
| 08:28.33 | angryuser | kaldemar: i neet te define in each context transfers possible? |
| 08:29.59 | kaldemar | i'm not quite sure i understand what you mean. |
| 08:33.22 | angryuser | kaldemar:well i dont really understood about transfers;) |
| 08:33.58 | *** join/#asterisk speekac (n=alwin@60.51.217.61) |
| 08:36.44 | Newbie___ | hi all, would asterisk caused false hangup if kewlstart is used instead of actual loopstart ? |
| 08:36.50 | kaldemar | angryuser: so if A calls B, and B transfers the call to C, you want to know where asterisk tries to find C's number. is this the case? |
| 08:37.48 | kaldemar | Newbie___: kewlstart is actually loopstart but with disconnect supervision. you could be able to produce false hangups with it with busydetect and a small busycount. |
| 08:39.22 | Newbie___ | kaldemar: can i use loopstart and disable all busycount and busydetect? i am using ks and busy detect and still experience random hangup |
| 08:40.56 | kaldemar | if you set busydetect=no, it won't be used. |
| 08:41.39 | kaldemar | if the random hangups still occur after that change, something else migth cause them. |
| 08:42.27 | brian | hey |
| 08:42.30 | brian | i have a extensions.conf section |
| 08:42.34 | Newbie___ | meaning to say, i can use ks but use busydetect=no ? |
| 08:43.03 | brian | when I call in asterisk complains about 404 |
| 08:43.10 | brian | http://rafb.net/paste/results/tDLALZ50.html |
| 08:43.26 | kaldemar | Newbie___: yes, you can use ks with busydetect=no |
| 08:43.58 | Newbie___ | kaldemar: callprogress=no as well ? |
| 08:45.17 | *** join/#asterisk stephane (n=stephane@gw.sortilege.net) |
| 08:46.10 | kaldemar | Newbie___: it seems to be observing the line for a busy tone also, you better set it to no too. |
| 08:46.16 | brian | hey |
| 08:46.32 | Newbie___ | kaldemar: thanks a lot, will try that |
| 08:46.38 | brian | what is wrong with my extensions.conf pasted here: http://rafb.net/paste/results/tDLALZ50.html |
| 08:46.46 | kaldemar | Newbie___: np |
| 08:47.34 | brian | heeelp |
| 08:47.41 | kaldemar | brian: how's the rest of your dialplan? how are you trying to pass the call to that context? |
| 08:48.04 | brian | in sip.conf |
| 08:48.17 | brian | with context=recordvoiceprompt |
| 08:48.48 | kaldemar | are you trying to give the prompt for all calls? |
| 08:49.14 | brian | yes |
| 08:49.37 | kaldemar | what number are you dialing? |
| 08:49.48 | brian | let's say it's 5555555555 |
| 08:49.59 | kaldemar | replace the s with _X. and give it a try |
| 08:50.07 | brian | yeah i know _X. works |
| 08:50.13 | brian | but why doesn't s work? |
| 08:50.34 | kaldemar | because s doesn't match to 5555555555 |
| 08:50.42 | brian | what does it match to |
| 08:51.21 | angryuser | dont put the point at the end, replate it with _XXXXXXXXX, its better i think |
| 08:51.50 | angryuser | if the lenght of numero is fixed of course |
| 08:52.05 | Newbie___ | kaldemar: random hangup seems to happen when incoming/outgoing. no problem when FXS-FXS |
| 08:52.24 | Newbie___ | kaldemar: so safe to assume is the busydetct issue |
| 08:52.40 | angryuser | can somebody point me at free musiononhold mp3's? |
| 08:53.33 | brian | With _XXXXXXXXXX I Get SIP/2.0 484 Address Incomplete |
| 08:53.39 | *** join/#asterisk KermitTheFragger (n=ktf@118-197.bbned.dsl.internl.net) |
| 08:54.19 | angryuser | X=one number |
| 08:54.43 | angryuser | so if u have 555 => _XXX |
| 08:54.50 | angryuser | not more or less |
| 08:55.12 | *** join/#asterisk djPepse (n=pepse@ip68-109-169-37.ph.ph.cox.net) |
| 08:55.29 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:57.08 | kaldemar | brian: if you dial less numbers than you have X's in your exten, asterisk keeps sending 484 until you have sent enough numbers. |
| 08:57.40 | kaldemar | overlap dialing defined by sip rfc. |
| 09:03.31 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:03.47 | *** join/#asterisk alex_112 (n=admin@fw.packetfront.com) |
| 09:05.33 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-205-211.tx.res.rr.com) |
| 09:08.35 | *** join/#asterisk knarfly (n=knarfly@c-65-34-177-3.hsd1.fl.comcast.net) |
| 09:08.43 | knarfly | howdy 8-) |
| 09:10.21 | djPepse | hey, any of you guys use gizmoproject? |
| 09:17.44 | *** join/#asterisk spaghetty (n=user@lugbari/people/spaghetty) |
| 09:20.32 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 09:23.36 | *** join/#asterisk fantasio (i=fantasio@banane.zedat.fu-berlin.de) |
| 09:23.39 | fantasio | hi there |
| 09:24.04 | spaghetty | hi |
| 09:24.24 | fantasio | someone around who owns a Siemens Hicom 300E and tried to connect asterisk through a digium t100p ? |
| 09:24.50 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 09:27.19 | *** join/#asterisk rtcg-tx (n=chatzill@mail.richardthecomputerguy.com) |
| 09:31.22 | *** join/#asterisk adlerweb (i=nodeadle@unaffiliated/adlerweb) |
| 09:32.20 | *** join/#asterisk CleanerX (n=nix@p54A392E4.dip0.t-ipconnect.de) |
| 09:32.55 | *** join/#asterisk key2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 09:33.25 | *** join/#asterisk parag_ast (n=parag_as@dxb-b112451.alshamil.net.ae) |
| 09:33.56 | parag_ast | Can anybody tell me asterisknow Username and Password ?? |
| 09:34.03 | parag_ast | default username and password |
| 09:34.13 | parag_ast | WebAdmin |
| 09:34.40 | fantasio | hum |
| 09:34.45 | fantasio | webinterface? |
| 09:34.48 | fantasio | is it any good? |
| 09:34.55 | parag_ast | Yes very good |
| 09:35.01 | parag_ast | but not able to get into it |
| 09:35.11 | parag_ast | as QuickStart Guide of ASterisknow |
| 09:35.17 | parag_ast | is having only 2 pages |
| 09:35.19 | fantasio | well, is here someone arround who can confirm that? |
| 09:35.32 | parag_ast | Russellb, ;) |
| 09:36.21 | zoa | im writing a tutorial on it |
| 09:36.29 | parag_ast | hehehe |
| 09:36.41 | parag_ast | Quickstart Guide is having only |
| 09:36.42 | parag_ast | 2 pages |
| 09:36.53 | zoa | asteriskguru is writing a tutorial for it |
| 09:36.56 | *** part/#asterisk adlerweb (i=nodeadle@unaffiliated/adlerweb) |
| 09:37.10 | zoa | http://www.asteriskguru.com/tutorials/asterisk_gui.html |
| 09:37.15 | zoa | there is already a part of it there |
| 09:37.15 | parag_ast | good but kindly let us know the default Web Access username and password |
| 09:37.16 | zoa | :) |
| 09:37.35 | zoa | check in manager.conf |
| 09:37.41 | zoa | it sprobably going to be the same |
| 09:37.53 | *** join/#asterisk [hC] (n=hardcore@S01060018394c6be3.vc.shawcable.net) |
| 09:38.27 | [hC] | Yo |
| 09:38.38 | pif | I'm failing to init chan_misdn with a junghans card, do they work out-of-the box ? |
| 09:38.50 | pif | or should I modify a setting? |
| 09:40.33 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 09:41.25 | parag_ast | Zoa, it seems its having lots of Bugs right now |
| 09:43.10 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 09:43.57 | djPepse | yeah, i tried asterisk gui and it didn't work :) |
| 09:44.19 | djPepse | btw, idefisk rox |
| 09:51.20 | *** part/#asterisk Newbie___ (n=me@211.24.146.11) |
| 09:52.03 | Kigh | does somebody know if the asterisk gui will be open source? |
| 09:52.22 | zoa | Kigh: afaik it is |
| 09:52.24 | zoa | download it and see |
| 09:52.40 | Kigh | i will :) |
| 09:52.45 | Kigh | thx |
| 09:54.04 | E-bola | Do anybody know how i can accept incoming calls without using register lines? |
| 09:54.10 | djPepse | hehe |
| 09:54.17 | E-bola | both my sip->pstn providers are changing their setup, so im required to change mine as well |
| 09:54.33 | E-bola | they just forward calls to my static wan ip |
| 09:54.48 | E-bola | but i dont know how i can explain asterisk to accept incoming calls without using register lines? |
| 09:54.58 | *** join/#asterisk Shaun2222 (n=ndci@ip68-5-62-187.oc.oc.cox.net) |
| 09:54.58 | parag_ast | asterisk GUI is having so many bugs right now |
| 09:55.16 | hads | E-bola: Look at the insecure option in sip.conf |
| 09:56.03 | E-bola | hads: do you have anywher ei can read or some examples? |
| 09:56.27 | hads | ~/src/asterisk/configs/sip.conf.sample |
| 09:57.02 | brian | what is wrong with my extensions.conf pasted here: http://rafb.net/paste/results/tDLALZ50.html |
| 09:57.39 | brian | http://rafb.net/paste/results/g3XoSm60.html <--- When I press 1 or 2 asterisk hangs up on me |
| 09:58.00 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 09:58.54 | Shaun2222 | anybody know whats going on with this... http://rafb.net/paste/results/8bJ3RK14.html |
| 09:58.58 | Shaun2222 | the problem is with the last line |
| 09:59.09 | Shaun2222 | i'm basically comparing the var to see if it starts with 2 |
| 09:59.12 | Shaun2222 | it's 3 digits long |
| 09:59.21 | brian | ? |
| 10:00.12 | *** join/#asterisk uwe (n=uwe@213.6.11.22) |
| 10:00.24 | Shaun2222 | it's not working, for some reason it's dumping me back to the start of the context |
| 10:00.44 | brian | http://rafb.net/paste/results/g3XoSm60.html <--- When I press 1 or 2 asterisk hangs up on me |
| 10:00.56 | *** join/#asterisk __AK__ (n=ak@28.228.210.62.te-dns.org) |
| 10:01.08 | *** join/#asterisk Scherrie (n=alwin@60.51.217.61) |
| 10:01.22 | Shaun2222 | brian: i think this is wrong... Goto(_XXXXXXXXXX|7) |
| 10:01.35 | Shaun2222 | i think it should be Goto([recordvoiceprompt] |
| 10:01.36 | brian | why is it wrong |
| 10:01.50 | Shaun2222 | bah... i think it should be Goto(recordvoiceprompt|7) |
| 10:02.02 | Kigh | right |
| 10:02.20 | Kigh | Goto(context|priority) |
| 10:02.23 | E-bola | can anybody confirm that http://www.asterisknow.org/ is down? |
| 10:02.31 | Shaun2222 | goto(recordvoiceprompt,,7) actually i think |
| 10:02.38 | Kigh | or Goto(priority) in same context (eg Goto(4) ) |
| 10:02.48 | Shaun2222 | it's somthing like context,exten,priority |
| 10:02.55 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 10:03.05 | Kigh | err right |
| 10:03.19 | brian | It still hangs up |
| 10:03.55 | brian | Goto(context|priority) works for everything else |
| 10:04.28 | Shaun2222 | what context do you start in? |
| 10:04.41 | brian | recordvoiceprompt |
| 10:04.59 | Shaun2222 | so when you dial into it, it doesnt do anything just hangs up? |
| 10:05.03 | brian | no |
| 10:05.12 | brian | when i press a digit 1 or 2 it hangs up |
| 10:05.15 | brian | and it shouldn't |
| 10:05.32 | Shaun2222 | uhhh |
| 10:05.43 | Shaun2222 | dont you have to do read() |
| 10:05.54 | brian | what the hell are you talking about |
| 10:06.23 | Shaun2222 | well you start in recordvoiceprompt... |
| 10:06.23 | brian | what is read? |
| 10:06.31 | Shaun2222 | so it's matching _XXXXXXXXX or whatever.. |
| 10:06.50 | Shaun2222 | well as far as i know you cant just dial in and press 1, it doesnt work like that |
| 10:07.16 | brian | yes it does |
| 10:07.35 | brian | you set 1 an an extension |
| 10:07.39 | Shaun2222 | oh, your phone is also set to use that context? |
| 10:08.01 | brian | no, you press 1 and 1 is an extension so asterisk listens to it |
| 10:08.38 | Shaun2222 | right, i think i see what your doing... |
| 10:08.56 | brian | but rather than listen to it |
| 10:09.00 | brian | when you press 1 it hangs up immediately |
| 10:09.02 | Shaun2222 | i'm just used to seeing 2 seperate contexts for outside-incomming and inside-incomming.. |
| 10:09.33 | brian | i just want to record some voice prompts |
| 10:09.55 | brian | so if you can like make this work that would be nice because i'm totally a asterisk noob and i just copied and pasted this from voip-info.org |
| 10:11.17 | Shaun2222 | i do somthing simular but mine looks like this... |
| 10:11.18 | Shaun2222 | http://rafb.net/paste/results/wmsNxC32.html |
| 10:11.28 | Shaun2222 | just for recording temp shit for testing |
| 10:11.37 | Shaun2222 | then i just move it from /tmp to whatever i want it named. |
| 10:11.37 | brian | i don't want to hang up |
| 10:11.47 | brian | i want to keep recording |
| 10:11.52 | brian | i have lots of stuff to record |
| 10:11.59 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 10:12.47 | Shaun2222 | you need to look at your logs |
| 10:12.49 | Shaun2222 | log into the console |
| 10:12.51 | Shaun2222 | whats it say |
| 10:13.09 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 10:13.19 | brian | it says |
| 10:13.34 | brian | Normal clearing |
| 10:14.47 | Shaun2222 | are you running asterisk with -vvvvvvvvvvvv |
| 10:15.50 | Shaun2222 | E-bola: works for me btw |
| 10:16.01 | E-bola | Shaun2222: the homepage? |
| 10:16.17 | Shaun2222 | E-bola: ya brings up the normal page.. |
| 10:16.43 | E-bola | hmm weird |
| 10:19.28 | djPepse | Anyone using Gizmoproject? I'm having a weird problem where when I answer a phone, the caller goes directly to Gizmo's voicemail |
| 10:19.36 | djPepse | even tho I shut off voicemail from their site |
| 10:19.53 | djPepse | i can only assume it's a configuration error of some kind. |
| 10:21.59 | *** join/#asterisk daysmen3 (n=primus@host86-139-113-76.range86-139.btcentralplus.com) |
| 10:22.02 | brian | the problem is that you're using gizmoproject |
| 10:22.03 | brian | :P |
| 10:22.24 | brian | Shaun2222: verbose didn't help |
| 10:22.36 | brian | Shaun2222: it just told me that a command was deprecated so i changed it and now back to square one |
| 10:23.12 | djPepse | can you recommend a provider with good, steady call clarity for $3/mo unlimited incoming? |
| 10:23.38 | brian | no |
| 10:24.00 | brian | there is just lots of horror stories to be heard about gizmo (or sipphone) |
| 10:24.22 | brian | they "lose" your DID |
| 10:25.23 | brian | i really want this stupid asterisk dial plan to work |
| 10:27.35 | djPepse | i hear nothin but good so far, but i haven't talked to too many people |
| 10:27.36 | brian | can someone take a look at: http://rafb.net/paste/results/Qqng4G22.html |
| 10:27.36 | djPepse | jnctn.net's site sucks ass, and i guess my 800did has been reporting disconnected |
| 10:27.43 | brian | djPepse: well, you have to realize that you get what you pay for |
| 10:28.08 | djPepse | i wouldn't say that |
| 10:28.17 | djPepse | cough vonage cough |
| 10:28.17 | brian | djPepse: and if you want to pay for "all circuits busy" and all sorts of other problems then be my guest |
| 10:28.36 | djPepse | several people recommended jnctn |
| 10:28.50 | brian | never heard of it |
| 10:29.07 | brian | there is a good company i found, www.les.net |
| 10:29.20 | brian | unlimited incoming with 2 channels |
| 10:29.59 | brian | it's a little more expensive, and I have not really been with them for too long... |
| 10:30.12 | brian | so if you have a problem (I haven't experienced one yet) |
| 10:30.55 | djPepse | why not use Record()? |
| 10:31.32 | djPepse | more expensive? I just signed up, can't do anything else until I verify, but it looks to be cheaper than junction |
| 10:31.54 | djPepse | and it's the same price as gizmo i think for the did.. maybe $4 instead of 3 |
| 10:32.49 | brian | djPepse: i do use record() |
| 10:33.34 | brian | the problem is not the recording that works great |
| 10:33.38 | djPepse | oh right |
| 10:33.40 | brian | the problem is that pressing 1 or 2 doesn't do anything |
| 10:33.50 | brian | it just hangs up |
| 10:34.43 | djPepse | do the other exten =>s work? the t, and i, |
| 10:34.54 | brian | i have no idea |
| 10:34.57 | brian | i copied and pasted this |
| 10:35.02 | brian | the t and i are probably deprecated |
| 10:35.17 | brian | they are supposed to be the extensions for "invalid extension" |
| 10:35.18 | djPepse | try putting the rest of the stuff in its own context? |
| 10:35.33 | brian | man i just want this to work so i can record the files |
| 10:35.37 | djPepse | everything below the 1-9 lines |
| 10:35.47 | brian | i don't know how to do that |
| 10:35.50 | djPepse | [blah] |
| 10:35.57 | brian | not yet anyways |
| 10:36.07 | djPepse | [nameofcontext] |
| 10:36.10 | djPepse | kinda like |
| 10:36.14 | djPepse | [recordvoiceprompt] |
| 10:36.23 | brian | i know that part |
| 10:36.27 | brian | but i don't know how to make it work |
| 10:36.40 | djPepse | put [blah] before the exten=> t line |
| 10:36.43 | E-bola | hads i looked at the sip.conf.sample |
| 10:36.54 | E-bola | but i didnt find any info on how i can stop using register lines |
| 10:37.10 | djPepse | what do you mean stop? |
| 10:37.24 | E-bola | my isp gave me a bunch of lines in this format somerandomnumber@thier_ip |
| 10:37.31 | E-bola | and told me to stop registering |
| 10:37.42 | E-bola | and that they will send all my calls to my server based on my static WAN ip |
| 10:37.53 | E-bola | and im not sure what to change in my asterisk server to do that? |
| 10:38.40 | E-bola | currently i register each telephone number on their server, and then have extensions that match the number and handle incoming calls like that |
| 10:38.49 | E-bola | I got no clue how to handle incoming calls without registering |
| 10:39.09 | brian | just take out the register line |
| 10:39.19 | djPepse | i guess just leave out the register line, and in your Dial() command do SIP/number@their_ip/{$EXTEN} |
| 10:39.27 | djPepse | err ${EXTEN} |
| 10:39.31 | E-bola | dial command? |
| 10:39.32 | brian | but keep the peer definition |
| 10:39.36 | E-bola | this is for incoing calls not outgoing? |
| 10:39.57 | brian | djPepse: hi i still need help :( |
| 10:40.08 | E-bola | and i asume i need to use the info they send me? |
| 10:40.16 | cy3o3 | hmmm |
| 10:40.17 | djPepse | brian: try my last suggestion |
| 10:40.36 | brian | what was that again |
| 10:40.41 | cy3o3 | Anyone around with some RealTime experieince? |
| 10:40.56 | brian | cy3o3: What is RealTime? |
| 10:41.01 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 10:41.09 | djPepse | brian: put [blah] before the exten=> t line |
| 10:41.13 | cy3o3 | trying to get asterisk going with mysql realtime / odbc |
| 10:41.20 | Shaun2222 | cy3o3: ya some |
| 10:41.25 | brian | djPepse: That's going ot make it not work even more |
| 10:41.31 | cy3o3 | not sure if I'm good to go or not there... but peep this: |
| 10:41.39 | cy3o3 | voip*CLI> show modules like odbc |
| 10:41.39 | cy3o3 | Module Description Use Count |
| 10:41.40 | *** join/#asterisk __AK__ (n=ak@28.228.210.62.te-dns.org) |
| 10:41.40 | Shaun2222 | it's been a while. |
| 10:41.42 | cy3o3 | res_config_odbc.so ODBC Configuration 1 |
| 10:41.45 | cy3o3 | res_odbc.so ODBC Resource 0 |
| 10:41.47 | cy3o3 | cdr_odbc.so ODBC CDR Backend 0 |
| 10:41.50 | cy3o3 | func_odbc.so ODBC lookups 0 |
| 10:41.53 | cy3o3 | shouldn't I have some sort of res_mysql module in there? |
| 10:42.17 | djPepse | brian: well, you just have to do include => blah |
| 10:42.20 | Shaun2222 | cy3o3: if i remember correctly there's 2 diffrent ways of connecting to mysql.. |
| 10:42.20 | cy3o3 | my name is also shaun by the way.. interesting that you spell your name the same |
| 10:42.31 | brian | djPepse: but that's not going to fix anything |
| 10:42.32 | djPepse | brian: along with your other includes, wherever recordvoiceprompt is |
| 10:42.36 | cy3o3 | Shaun2222: yeah for sure... |
| 10:42.38 | Shaun2222 | cy3o3: you can either use res_odbc or res_mysql |
| 10:42.42 | djPepse | brian: ok, then i guess don't try it |
| 10:42.53 | brian | djPepse: it will work exactly the same but it will be in different contexts |
| 10:43.03 | cy3o3 | well shit I'd rather just use res_odbc since it's already there and appears to be working, heh |
| 10:43.07 | Shaun2222 | if you use odbc you have to install unixodbc on the system and configure it.. |
| 10:43.11 | djPepse | that's cool, mang |
| 10:43.12 | brian | odbc isn't a database |
| 10:43.19 | Shaun2222 | i think when i did it i ditched the odbc idea and went to using res_mysql |
| 10:43.39 | *** join/#asterisk daysmen3 (n=primus@host86-139-113-76.range86-139.btcentralplus.com) |
| 10:43.44 | brian | odbc is an interface to databases |
| 10:43.48 | Shaun2222 | one sec |
| 10:43.59 | Shaun2222 | let me look i think i still have the devel box i was testin git on |
| 10:44.04 | cy3o3 | yeah, when I did this before I was using res_mysql |
| 10:44.15 | cy3o3 | but that module is not installed currently |
| 10:44.19 | cy3o3 | is that add-ons? |
| 10:44.20 | djPepse | e-bola: did you figure it out? |
| 10:44.32 | Shaun2222 | load => res_config_mysql.so |
| 10:44.34 | Shaun2222 | thats what i'm using |
| 10:44.42 | cy3o3 | yeah |
| 10:44.43 | Shaun2222 | i didnt use odbc because it was annoying. |
| 10:44.48 | cy3o3 | I have no res_config_mysql.so |
| 10:44.55 | Shaun2222 | ya, i think it comes from asterisk-addons |
| 10:45.01 | cy3o3 | hmmm |
| 10:45.04 | cy3o3 | okay |
| 10:45.14 | cy3o3 | I've installed asterisk from apt-get under ubuntu |
| 10:45.15 | brian | guys make my extensions work for me |
| 10:45.16 | brian | :( |
| 10:45.31 | cy3o3 | there is unfortunately no asterisk-addons package for ubuntu (in apt at least) |
| 10:45.54 | djPepse | brian: is there even an include => recordvoiceprompt? |
| 10:46.05 | Shaun2222 | cy3o3: here's my modules.conf incase your interested... |
| 10:46.05 | Shaun2222 | http://paste.uni.cc/11724 |
| 10:46.16 | Shaun2222 | cy3o3: so go download it from asterisk and build it. |
| 10:46.33 | Shaun2222 | cy3o3: http://ftp.digium.com/pub/asterisk/releases/asterisk-addons-1.2.5.tar.gz |
| 10:46.38 | Shaun2222 | cd /usr/src |
| 10:46.42 | Shaun2222 | wget http://ftp.digium.com/pub/asterisk/releases/asterisk-addons-1.2.5.tar.gz |
| 10:47.07 | Shaun2222 | tar zxvf asterisk-addons-1.2.5.tar.gz;asterisk-adons-1.2.5;make;make install |
| 10:47.14 | Shaun2222 | make sure you have mysql-devel installed. |
| 10:47.14 | brian | djPepse: no |
| 10:47.21 | brian | djPepse: recordvoiceprompt is the default context |
| 10:47.29 | brian | djPepse: i dont' think you understand at all |
| 10:47.31 | E-bola | djPepse: yes |
| 10:47.36 | E-bola | i simply removed the register lines |
| 10:47.39 | brian | i wish an asterisk expert was here :( |
| 10:47.41 | E-bola | and everything continued to work |
| 10:47.47 | E-bola | that just seems a bit too easy.... |
| 10:47.56 | djPepse | brian: guess not, mr. knowitallcan'tsolvemyownproblem |
| 10:48.16 | E-bola | doesnt this mean anything can send me sip packets claiming to be calling my number, but in fact could be from anywhere? |
| 10:48.26 | E-bola | since i dont validate incoming calls at all? |
| 10:48.38 | Shaun2222 | brian: like i said you need to look at the console and see where it's stoping |
| 10:48.47 | djPepse | e-bola: well, for your incoming type=peer extension, you should use host=theirip |
| 10:48.49 | cy3o3 | Shaun2222: no mysql-devel in apt |
| 10:48.49 | brian | i know where it's stopping shaun |
| 10:49.09 | brian | it's stopping when i press a digit |
| 10:49.09 | Shaun2222 | cy3o3: well i dont know what it's called in debian... |
| 10:49.13 | brian | it's kind of obvious |
| 10:49.14 | djPepse | haha |
| 10:49.20 | djPepse | brian: what does it say on the console? |
| 10:49.28 | brian | one second |
| 10:49.29 | Shaun2222 | brian: that doesnt mean anything.. |
| 10:49.30 | E-bola | djpepse: but i have nothing at all related to incoming in my sip.conf now |
| 10:49.37 | cy3o3 | there's libmysql++-dev - mysql C++ library bindings (development) |
| 10:49.54 | Shaun2222 | it could be dieing on 1 because you dont have system support, or it could be dieing on 2 because that file doesnt exist... |
| 10:50.02 | Shaun2222 | you would never know by hitting 1 which one it died on |
| 10:50.03 | E-bola | the only peers i have in sip.conf are the ones for outgoing calls |
| 10:50.06 | Shaun2222 | look at the console |
| 10:50.07 | djPepse | e-bola: does your incoming # work now that you've removed the register line? |
| 10:50.15 | E-bola | djPepse: yes |
| 10:50.24 | E-bola | djPepse: even restarted |
| 10:50.26 | djPepse | shaun2222: he doesn't have to look at anything, he already knows what the problem is |
| 10:50.28 | djPepse | :) |
| 10:50.31 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 10:50.32 | Shaun2222 | cy3o3: ya i think thats it. |
| 10:50.39 | djPepse | it's hanging up after he presses 1! |
| 10:50.43 | cy3o3 | okeee mang, I'll give it a whirl.. thanks |
| 10:51.03 | Shaun2222 | cy3o3: have phun, it took me a while to get it working... all the docs on this are stupid.. |
| 10:51.16 | Shaun2222 | cy3o3: i must have switched between mysql and odbc 10 times |
| 10:51.24 | cy3o3 | seriously dude.. jesus |
| 10:51.30 | cy3o3 | yeah, I had it working GREAT on debian |
| 10:51.31 | djPepse | e-bola: i think there's an option that's on by default for guest calls |
| 10:51.34 | Shaun2222 | cy3o3: i got res_mysql to work... |
| 10:51.38 | djPepse | e-bola: i could be wrong tho |
| 10:51.40 | Shaun2222 | so i would stick with that route. |
| 10:51.42 | brian | something else is worng now |
| 10:51.49 | cy3o3 | now our company has decided to standardize everything on ubuntu, so I'm stuck with that |
| 10:51.59 | cy3o3 | (even though ubuntu is basically debian or whatever) |
| 10:51.59 | cy3o3 | blah |
| 10:52.00 | Shaun2222 | cy3o3: well ubuntu and debian are basically the same thing. |
| 10:52.06 | cy3o3 | hopefully I can get this going |
| 10:52.15 | djPepse | cy3o3: ubuntu is pretty swanky |
| 10:52.18 | brian | my provider wont' accept my registration |
| 10:52.28 | cy3o3 | I wanted to go gentoo |
| 10:52.29 | Shaun2222 | i use CentOS for most stuff... |
| 10:52.36 | cy3o3 | kust for like... asterisk |
| 10:52.41 | E-bola | djPepse: i dont understand.... what do i need to put in sip.conf to restric incoming calls to only be allowed from a certain ip? |
| 10:52.52 | djPepse | cy3o3: hehe, there's a funny site comparing gentoo users to ricers |
| 10:52.53 | cy3o3 | cause that seems to be what everyone uses... everythings all packaged and shit with the latest and greatest |
| 10:52.55 | Shaun2222 | used debian back in the day... for hosting most people want CentOS/RHEL or Fedora(blah) these days |
| 10:53.04 | cy3o3 | ricers? lolz |
| 10:53.32 | E-bola | debian forever :) |
| 10:53.43 | Shaun2222 | sorry, i've never understood the hype of gentoo... i absolutly hate working with that OS and even more the customers who use it... |
| 10:53.45 | Shaun2222 | :) |
| 10:54.00 | backblue | i love it! :P |
| 10:54.09 | Shaun2222 | they always know everything, but really know nothing... it's rare i see somebody using it that actually can manage it. |
| 10:54.39 | backblue | Shaun2222: well, dont speak about what you dont know |
| 10:54.42 | djPepse | i installed it on my ibook :) |
| 10:54.46 | backblue | i work with gentoo for 2 years |
| 10:54.50 | backblue | and no problems at all |
| 10:54.58 | E-bola | djPepse: hmm now my numbers stopped working |
| 10:55.01 | backblue | and i have passed along by mostly all distros |
| 10:55.05 | E-bola | after removing my register lines |
| 10:55.10 | cy3o3 | yeah, asterisk addons straight up just fails miserably on make |
| 10:55.10 | Shaun2222 | backblue: i'm not saying that gentoo doesnt work... i'm saying that 99% of the people i see using it... shouldnt be... |
| 10:55.11 | djPepse | other than waiting a year for things to compile |
| 10:55.13 | djPepse | like mozilla |
| 10:55.14 | djPepse | haha |
| 10:55.16 | backblue | i dont use gentoo for production, only for desktop. |
| 10:55.23 | djPepse | e-bola: yeah that's what should have happened |
| 10:55.31 | cy3o3 | fuck |
| 10:55.35 | Shaun2222 | it's like those people who use freebsd because they think it's secure... but can bearly somthing like mandrake ;) |
| 10:55.45 | djPepse | e-bola: so what all did your isp give you? did they tell you to configure an extension or anything? |
| 10:56.11 | backblue | Shaun2222: that's stupid, they know freebsd its better than mandrake, so why use mandrake? |
| 10:56.13 | Shaun2222 | cy3o3: whats the errors |
| 10:56.22 | backblue | they dont have to know how kernel stacks works |
| 10:56.28 | backblue | for having the right to use it |
| 10:56.37 | cy3o3 | app_saycountpl.c:12:29: error: asterisk/logger.h: No such file or directory |
| 10:56.38 | cy3o3 | app_saycountpl.c:13:30: error: asterisk/channel.h: No such file or directory |
| 10:56.38 | cy3o3 | app_saycountpl.c:14:26: error: asterisk/pbx.h: No such file or directory |
| 10:56.38 | cy3o3 | app_saycountpl.c:15:29: error: asterisk/module.h: No such file or directory |
| 10:56.41 | cy3o3 | etc etc etc etc |
| 10:56.56 | Shaun2222 | cy3o3: is there a asterisk-lib package? |
| 10:57.01 | Shaun2222 | or asterisk-devel |
| 10:57.07 | djPepse | dude just apt-cache search asterisk |
| 10:57.31 | Shaun2222 | backblue: what i was saying is that these people can bearly use mandrake let alone freebsd... bsd is far more complex to manage. |
| 10:57.58 | backblue | gentoo it's very simple |
| 10:57.59 | Shaun2222 | djPepse: would if i was on the box :) |
| 10:58.05 | djPepse | i hate bsd as a desktop |
| 10:58.08 | backblue | and very powerfull all the way |
| 10:58.27 | djPepse | blackblue: you can accomplish the same thing by just getting source from the source :) |
| 10:58.46 | djPepse | screw emerging |
| 10:58.49 | backblue | it's like having a car that if you dont forces it, it only drinks about 5l/100km, but if you want to run, it has 200 horses on the engine |
| 10:58.53 | backblue | what can be better? |
| 10:59.06 | backblue | it's the perfect solution for most of the common |
| 10:59.30 | E-bola | guys in my dialplan |
| 10:59.32 | dlynes_laptop | Shaun2222: all that crap gets installed with the main asterisk install package |
| 10:59.36 | E-bola | when i dial a user and he is busy |
| 10:59.43 | djPepse | if you're going to wait for it to build, you may as well go to the source, and save the room better used for other things by removing the emerge stuff |
| 10:59.44 | E-bola | asterisk hangsup, saying everyone is congested/busy atm |
| 10:59.51 | E-bola | instaid of moving on to the next line |
| 11:00.08 | Shaun2222 | cy3o3: can you paste the full dump of make to a pastebin or somthing |
| 11:00.13 | djPepse | e-bola: next line? |
| 11:00.24 | E-bola | djPepse: next priority |
| 11:00.25 | dlynes_laptop | E-bola: Are you using trixbox, or vanilla asterisk? |
| 11:00.31 | brian | ok |
| 11:00.34 | E-bola | dlynes_laptop: no |
| 11:00.36 | brian | the sip proxy was down for a few minutes heh |
| 11:00.54 | E-bola | i simply want it to move to the next priority if the user is busy |
| 11:01.13 | E-bola | i thought that was the default behavior? |
| 11:01.13 | djPepse | e-bola: it's whatever you put in the dialplan |
| 11:01.13 | brian | ok well |
| 11:01.13 | dlynes_laptop | E-bola: Just curious because asterisk's default behaviour is not to tell the caller that everyone is congested/busy atm |
| 11:01.59 | cy3o3 | Shaun2222: asterisk-dev - development files for asterisk |
| 11:02.12 | cy3o3 | Shaun2222: not sure what you mean by a pastebin... |
| 11:02.13 | brian | ok |
| 11:02.18 | dlynes_laptop | E-bola: asterisk's default behaviour is continue on to the next dialplan priority; if there isn't a next dialplan priority, depending on your settings in the dialplan general section, it'll either hangup, or it's undefined, I believe |
| 11:02.20 | brian | It actually doesn't hang up when you press a button |
| 11:02.22 | TimothyP | FWD IAX still not working ? |
| 11:02.33 | cy3o3 | I can dump the output to a textfile and host it somewhere so you can see the entire output |
| 11:02.34 | E-bola | dlynes_laptop: what could stop it from moving on to the next priority |
| 11:02.35 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 11:02.43 | djPepse | TimothyP: what's not working? |
| 11:02.45 | brian | exten => _XXXXXXXXXX,8,Set(TIMEOUT(response)=10) |
| 11:02.51 | brian | That's where it hangs up djPepse |
| 11:02.52 | dlynes_laptop | E-bola: maybe you have priority jumping enabled? |
| 11:02.56 | TimothyP | djPepse since yesterday I can't register anymore |
| 11:02.59 | TimothyP | it always says |
| 11:03.04 | TimothyP | Request Sent and then Timeout |
| 11:03.09 | dlynes_laptop | E-bola: priority jumping is not enabled by default in asterisk 1.2, but it was the default in 1.0 |
| 11:03.16 | TimothyP | 192.246.69.186:4569 814179 <Unregistered> 60 Request Sent |
| 11:03.16 | TimothyP | 192.246.69.186:4569 805208 <Unregistered> 60 Request Sent |
| 11:03.33 | TimothyP | it worked before, since yesterday it stopped working |
| 11:03.42 | cy3o3 | Shaun2222: weeee.. after installing asterisk-dev: |
| 11:03.43 | cy3o3 | root@voip:/usr/src/asterisk-addons-1.2.5# make install |
| 11:03.44 | cy3o3 | for x in format_mp3/format_mp3.so app_saycountpl.so cdr_addon_mysql.so app_addon_sql_mysql.so res_config_mysql.so; do install -m 755 $x /usr/lib/asterisk/modules ; done |
| 11:03.44 | djPepse | TimothyP: I'm registered |
| 11:03.53 | brian | X-Asterisk-HangupCause: Normal Clearing |
| 11:03.55 | TimothyP | yep some people are, other say they still can't |
| 11:04.03 | TimothyP | did they change something which requires a change? |
| 11:04.10 | brian | TimothyP: it is a problem with your provider? |
| 11:04.16 | Shaun2222 | cy3o3: ya, i figured they didnt include those in the standard asterisk package... |
| 11:04.23 | brian | TimothyP: see if that IP responds to ping |
| 11:04.24 | TimothyP | connectoing to FWD using a SIP phone works fine |
| 11:04.25 | Shaun2222 | they also have seperate dev packages.... |
| 11:04.26 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:04.28 | djPepse | TimothyP: i don't believe so, I haven't changed anything. I'm using * 1.4 btw |
| 11:04.32 | Shaun2222 | s/also/always/ |
| 11:04.40 | brian | djPepse: help me i'm a miserable failure |
| 11:04.42 | puzzled | hi |
| 11:04.56 | cy3o3 | yayyyyyyyyyy |
| 11:04.57 | cy3o3 | res_config_mysql.so MySQL RealTime Configuration Driver 0 |
| 11:04.57 | cy3o3 | cdr_addon_mysql.so MySQL CDR Backend 0 |
| 11:04.57 | cy3o3 | app_addon_sql_mysql.so Simple Mysql Interface 0 |
| 11:05.00 | TimothyP | and there are no blocked ports as we are on a business account and have to manage our own firewall, we also have no upload/download limitations |
| 11:05.22 | Shaun2222 | alright, i'm outta here, time for bed... |
| 11:05.32 | brian | i need help please please please |
| 11:05.33 | brian | :( |
| 11:05.39 | TimothyP | yes I responds to ping |
| 11:05.41 | djPepse | brian: pastebin the whole stuff at verbosity 6 |
| 11:05.51 | cy3o3 | Shaun2222: later man, thanks for your help |
| 11:06.07 | Shaun2222 | djPepse: your determined guy ;) |
| 11:06.17 | Shaun2222 | cy3o3: np. |
| 11:06.20 | TimothyP | iax.conf contains lines like register => myfwdnumber:mypassword7@iax2.fwdnet.net |
| 11:06.56 | djPepse | that's what i have |
| 11:07.05 | TimothyP | figured that much :) |
| 11:07.23 | TimothyP | some other people on the forum are complaining about it as well, for some it has been solved, for others it has not |
| 11:08.07 | brian | djPepse: ok |
| 11:08.58 | djPepse | timothyp: can you connect with another client? Try idefisk |
| 11:09.17 | TimothyP | is that a software iax client? |
| 11:09.21 | djPepse | yeh |
| 11:09.21 | zoa | yes |
| 11:09.44 | TimothyP | I'll have a look at it in a minute |
| 11:10.29 | TimothyP | zoa, any idea about this one: I can place outgoing calls using misdn now, but incoming calls don't seem to work, it always says the extention can never match and people get a message from belgacom saying the number is not connected, asterisk does see the incoming call however |
| 11:11.00 | zoa | paste me the exact number in private |
| 11:11.06 | zoa | euh |
| 11:11.11 | zoa | exact warning / error message on asterisk |
| 11:11.24 | TimothyP | hang on I'll call myself first :p |
| 11:12.07 | djPepse | i have the problem of calls ringing in, asterisk acknowledges the pickup, but the caller gets provider's voicemail |
| 11:12.19 | brian | djPepse: http://rafb.net/paste/results/dJLVyn69.txt |
| 11:13.15 | brian | djPepse: http://rafb.net/paste/results/dJLVyn69.txt |
| 11:13.23 | brian | djPepse: hi i pasted it |
| 11:13.25 | djPepse | brian: so like, it looks like it just goes right through the recording |
| 11:13.50 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 11:14.05 | djPepse | maybe you need to add a timeout, like exten => etc,4,Record(mymessage:ulaw,##) |
| 11:14.17 | brian | nope |
| 11:14.31 | brian | don't you see that it doesn't hang up at record |
| 11:14.38 | Modcuts | morning when writing a app, is there an operation for reading back the digits pressed. |
| 11:14.39 | djPepse | does it record at record? |
| 11:14.46 | brian | <PROTECTED> |
| 11:14.47 | brian | yes it records |
| 11:14.52 | brian | and it plays it back |
| 11:14.53 | brian | that works fine |
| 11:15.00 | brian | That is the problem line, right there |
| 11:15.00 | djPepse | oh, that's not what you said before |
| 11:15.05 | brian | It's falling through |
| 11:15.11 | brian | yes i did, i guess you didn't catch it it's ok |
| 11:15.15 | brian | now you know |
| 11:15.42 | brian | anyways, why is this auto fallthrough occurring |
| 11:15.53 | TimothyP | OH Boy |
| 11:15.54 | TimothyP | :p |
| 11:16.10 | TimothyP | I've been sending private messages, never noticed the fact that I wasnt registered anymore so they didn't arrive |
| 11:16.27 | djPepse | hm |
| 11:16.42 | TimothyP | :p |
| 11:17.02 | TimothyP | btw |
| 11:17.04 | brian | djPepse: can you help meeeee |
| 11:17.07 | TimothyP | using idefisk I get the same timeout |
| 11:17.17 | brian | djPepse: play my song....the solution to my problem song |
| 11:17.33 | brian | :-D |
| 11:19.02 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:19.27 | brian | :( |
| 11:19.35 | brian | cpm: hello! |
| 11:19.39 | E-bola | dlynes_laptop: http://pastebin.com/835238 |
| 11:19.45 | E-bola | thats the part thats not working properly |
| 11:19.55 | cpm | lo |
| 11:19.56 | E-bola | if the SIP/Philip phone is busy it just dies |
| 11:20.58 | brian | cpm: you shall help me with my extensions.conf problems! |
| 11:21.29 | E-bola | can anybody tell me why thats happening |
| 11:22.28 | djPepse | brian: 1.2? or 1.4? |
| 11:22.42 | brian | 1.2.13 |
| 11:23.22 | cy3o3 | guh.. anyone know wtf this means? heh |
| 11:23.24 | cy3o3 | chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory |
| 11:23.42 | cy3o3 | is that just audio on the server basically? |
| 11:23.51 | E-bola | i just dont get why asterisk says Everyone is busy/congested at this time (1:1/0/0) instaid of simply moving on to next priority |
| 11:24.09 | puzzled | cy3o3: just noload => chan_oss.so if you want to get rid if it |
| 11:24.51 | brian | djPepse: you shall assist! |
| 11:24.58 | djPepse | brian: in 1.4, i get the goto recordvoiceprompt|2 in new stack, then "Sent into invalid extension 'recordvoiceprompt'" and it gives me the Playback invalid over and over again |
| 11:24.58 | dlynes_laptop | E-bola: can you pastebin your entire extensions.conf file? |
| 11:25.07 | dlynes_laptop | E-bola: I'm missing part of the picture |
| 11:25.16 | E-bola | dlynes_laptop: sure thing |
| 11:25.42 | brian | djPepse: well i want it to work in 1.2.18 :P |
| 11:26.21 | brian | i mean 13 |
| 11:26.34 | djPepse | brian: yeah but you actually get it to work, recording and playing back, whereas it doesn't do that with that syntax in 1.4 |
| 11:27.06 | djPepse | i don't see the problem with it falling through if it's gotten to the end of your context |
| 11:27.37 | E-bola | dlynes_laptop: i dont have any options in general though so i doubt showing the whole thing makes anything more clear... |
| 11:27.49 | brian | djPepse: that's because 1.4 is different... |
| 11:28.25 | E-bola | dlynes_laptop: http://pastebin.com/835253 |
| 11:28.36 | E-bola | please let me know asap, im going crazy with this :) |
| 11:28.38 | E-bola | it used to work fine |
| 11:28.45 | Modcuts | So there is no function for relaying the numbers dialed on the channel? |
| 11:28.49 | E-bola | only thing i have done is remove some register lines for other numbers |
| 11:30.08 | E-bola | ok |
| 11:30.15 | E-bola | i knwo whats wrong! |
| 11:30.25 | E-bola | its jumping to n + 101 with n being the current priority |
| 11:30.25 | cy3o3 | puzzled: sweet thanks |
| 11:30.33 | dlynes_laptop | E-bola: that's what I said |
| 11:30.37 | E-bola | question: why on earth does it jump +101 prioty instaid of +1?` |
| 11:30.45 | cy3o3 | puzzled: but that is indeed what's up right... like dialing from the console would produce audio output on the server? |
| 11:30.47 | E-bola | dlnyes_laptop: yep i know thats how i found out :) |
| 11:30.51 | E-bola | But WHY is it doing that? |
| 11:30.52 | cy3o3 | cause uhh, my server just chills in a colo |
| 11:30.54 | dlynes_laptop | E-bola: Because you've got priority jumping enabled |
| 11:31.03 | E-bola | dlynes: that must be by default then? |
| 11:31.06 | dlynes_laptop | I'm just trying to figure out why |
| 11:31.08 | E-bola | i dont enable it anywhere specifically |
| 11:31.08 | dlynes_laptop | erm where |
| 11:31.12 | cy3o3 | so I could care less if audio is coming from the server.. in fact, it really should not, heh |
| 11:31.19 | dlynes_laptop | E-bola: Which version of asterisk are you using? |
| 11:31.23 | E-bola | im using version Asterisk 1.2.9.1 |
| 11:31.43 | puzzled | cy3o3: yes. if it's supported in the version of asterisk that you have than you have the command "Dial" from the CLI |
| 11:31.50 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-ea18298ad5a2c7e8) |
| 11:31.57 | brian | E-bola: you should upgrade there are lots of security updates |
| 11:32.21 | brian | djPepse: hi there |
| 11:32.21 | E-bola | brian: ya i just found out |
| 11:32.27 | dlynes_laptop | brian: yeah, but if you're using faxing, anything later than 1.2.9.1 won't work with spandsp, either |
| 11:32.39 | brian | dlynes_laptop: eh |
| 11:32.53 | brian | people still use fax? |
| 11:32.57 | dlynes_laptop | brian: asterisk kept breaking compatibility with spandsp; I think coppice finally gave up on trying to stay compatible with it |
| 11:33.02 | E-bola | dlynes_laptop: maybe its easier if u just tell me how to turn off priority jumping |
| 11:33.07 | E-bola | i cant find any documentation on it |
| 11:33.15 | cy3o3 | puzzled: I still want to "dial" from the CLI for testing purposes, but no need to hear any audio output on the server.. any way to do that if I disable chan_oss.so |
| 11:33.19 | brian | i am very sad |
| 11:33.20 | dlynes_laptop | E-bola: If you use the 'j' option to dial, it enables priority jumping, too |
| 11:33.22 | cy3o3 | or just ignore that output from the cli |
| 11:33.30 | dlynes_laptop | brian: yes, people still use fax |
| 11:33.33 | dlynes_laptop | brian: i still use fax |
| 11:33.33 | brian | i know |
| 11:33.35 | brian | it was a joke |
| 11:33.39 | brian | you dinosaur |
| 11:33.40 | brian | :P |
| 11:34.02 | puzzled | cy3o3: disable it and you loose at least sound. not sure if you will still have the dial command. why not disable it and just use a softphone? |
| 11:34.10 | E-bola | dlynes_laptop: i only use m option when dialing |
| 11:34.40 | cy3o3 | puzzled: ugh, well I'm in a totally wacked out environment, heh |
| 11:34.40 | dlynes_laptop | yeah, i seen that |
| 11:34.40 | cy3o3 | puzzled: guess I'll just leave it enabled for now |
| 11:35.19 | cy3o3 | also, yet another question.. heh, I rock |
| 11:35.20 | brian | Hello guys! |
| 11:35.26 | cy3o3 | how come I don't get the 0day l33t color CLI |
| 11:35.30 | brian | Please to help me with extensions.conf problem! |
| 11:35.45 | cy3o3 | I'm running asterisk -vcr in a screen session.. TERM is set to vt100 |
| 11:35.53 | E-bola | dlynes_laptop: how would i turn on/off priority jumping globally? |
| 11:36.05 | E-bola | since im using 1.2 it must be turned on somewhere... |
| 11:36.12 | Nugget | with TERM=vt100 you should never get color in anything. |
| 11:36.19 | Nugget | vt100 isn't a color-capable terminal |
| 11:36.20 | brian | http://rafb.net/paste/results/1YlOdV83.html |
| 11:36.32 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:36.33 | cy3o3 | well I tried xterm-color |
| 11:36.35 | puzzled | cy3o3: no idea. don't use the colors but isn't vt100 non-color? |
| 11:36.38 | Nugget | try TERM=xterm or TERM=xterm-color (depending on your flavor of unix) |
| 11:36.39 | cy3o3 | TERM=screen is what the default is |
| 11:36.48 | cy3o3 | since I'm in a screen session |
| 11:36.52 | Nugget | *nod* |
| 11:36.56 | Nugget | dunno then |
| 11:37.13 | dlynes_laptop | E-bola: what recording file are you using if the caller is busy? |
| 11:37.30 | dlynes_laptop | E-bola: i.e. is the recording you're hearing if the caller is busy one of your recordings? |
| 11:37.30 | cy3o3 | it's not a setting in one of the bazillion .conf files or anything is it? |
| 11:37.48 | E-bola | dlynes_laptop: if the caller is busy, he should just be moved to another priority |
| 11:37.55 | Nugget | not that I've ever seen |
| 11:37.58 | E-bola | he shouldnt hear a busy tone |
| 11:38.03 | E-bola | and that used to work |
| 11:38.18 | E-bola | if my priority 101 was dial(user) then my 102 was a goto |
| 11:38.22 | dlynes_laptop | E-bola: but is the busy tone and the busy recording being generated by you, or the dialed party's telco? |
| 11:38.32 | E-bola | and that stopped working cuz asterisk suddenly started jumping to 202 instaid of 102 |
| 11:38.46 | E-bola | dlynes_laptop: there is no busy tone, nor should their be |
| 11:38.49 | cy3o3 | Heh, and ANOTHER question for ya'll! heh. I have a TDM400P card in that box.. |
| 11:39.05 | cy3o3 | my only intention of using it is for timing in meetme conferences.. |
| 11:39.13 | cy3o3 | any idea how I can tell if that's in place? |
| 11:39.19 | E-bola | dlynes_laptop: understand? |
| 11:39.33 | cy3o3 | I see a bunch of zap modules in show modules like zap |
| 11:39.36 | brian | cy3o3: zttest |
| 11:39.51 | dlynes_laptop | yeah, so it's 'echbiz_kontoret_lukket' that's getting played? |
| 11:39.56 | dlynes_laptop | erm |
| 11:40.02 | dlynes_laptop | ' techbiz_kontoret_lukket I mean? |
| 11:40.04 | brian | dlynes_laptop: hi you know lots about asterick |
| 11:40.12 | E-bola | dlynes_laptop: well thats if we are closed |
| 11:40.12 | dlynes_laptop | a bit |
| 11:40.15 | E-bola | thats handled by the goto |
| 11:40.21 | E-bola | this is different |
| 11:40.23 | brian | dlynes_laptop: help me you asterisk guy you |
| 11:40.26 | E-bola | it shoudl simply be like this |
| 11:40.26 | dlynes_laptop | E-bola: so which file is it playing when i'ts busy then? |
| 11:40.33 | E-bola | dlynes: no files are played |
| 11:40.39 | E-bola | the call is simply passed on to another person |
| 11:40.46 | dlynes_laptop | E-bola: Which person? |
| 11:41.02 | brian | dlynes_laptop: hey... |
| 11:41.03 | E-bola | dlynes: a list of persons |
| 11:41.04 | E-bola | exten => 43332730,103,Goto(Internal,41,1) |
| 11:41.05 | dlynes_laptop | brian: ? |
| 11:41.15 | brian | dlynes_laptop: can you look at this and tell me why it's not working right: http://rafb.net/paste/results/wmsNxC32.html |
| 11:41.28 | E-bola | and that looks like: exten => 41,1,Dial(SIP/Jonas&SIP/Philip&SIP/Gudme&SIP/Joakim&SIP/Nikolaj&SIP/Christian,99,tm) |
| 11:41.29 | brian | dlynes_laptop: it hangs up after 8 |
| 11:41.32 | dlynes_laptop | E-bola: Ok, but it's not getting to there, right? |
| 11:41.39 | brian | dlynes_laptop: due to fall through |
| 11:42.01 | E-bola | dlynes_laptop: correct, instaid of going to the goto priority which is the next one, it goes to a priority +101 |
| 11:42.23 | E-bola | it basically increments the current priority 101 "priorities" instaid of just 1 |
| 11:42.32 | E-bola | and i dont know why :( |
| 11:42.55 | dlynes_laptop | So, after exten => 43332730,101,Dial(SIP/Philip,15,m), it's going to exten => 43332730,202,Hangup(), right? |
| 11:42.57 | E-bola | so if the user is busy instaid of going to priority 102 from 101 it goes to prirority 202 |
| 11:42.58 | cy3o3 | brian: yeah dude.. no zztest even exists.. |
| 11:43.06 | E-bola | dlynes_laptop: precisely |
| 11:43.12 | cy3o3 | brian: |
| 11:43.13 | cy3o3 | voip*CLI> zap show status |
| 11:43.13 | cy3o3 | No Zaptel interface found. |
| 11:43.13 | cy3o3 | Nov 29 04:42:26 WARNING[5279]: chan_zap.c:9937 zap_show_status: Unable to open /dev/zap/ctl: No such file or directory |
| 11:43.16 | cy3o3 | :( |
| 11:43.20 | E-bola | dlynes_laptop: and i want it to go to 102 instaid |
| 11:43.22 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:44.03 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:44.21 | dlynes_laptop | E-bola: it's not going to priority 200 instead? |
| 11:44.32 | E-bola | dlynes_laptop: no its doing what u said |
| 11:44.33 | brian | dlynes_laptop: when is it my turn??? |
| 11:44.39 | brian | :-D |
| 11:44.41 | dlynes_laptop | brian: calm the hell down |
| 11:44.48 | dlynes_laptop | brian: i'm helping someone else already |
| 11:44.49 | brian | never! |
| 11:44.59 | brian | asterick excites me |
| 11:45.26 | dlynes_laptop | E-bola: Well, if you'll notice priority 103 and 104 are commented out |
| 11:45.27 | E-bola | dlynes_laptop: as far as i can see the question is why is my asterisk 1.2 using priority jumping when it was supposed to have been disabled in 1.2 |
| 11:46.12 | dlynes_laptop | E-bola: in your original pastebin, 104 was hangup, and 103 was commented out |
| 11:46.21 | dlynes_laptop | E-bola: but in your full pastebin, they're both commented out |
| 11:46.47 | cy3o3 | most of you must not be in the US |
| 11:46.56 | cy3o3 | considering it's going on 5am here... |
| 11:47.01 | dlynes_laptop | cy3o3: I'm in Canada |
| 11:47.04 | cy3o3 | coo |
| 11:47.08 | dlynes_laptop | cy3o3: It's 3:45am here |
| 11:47.15 | cy3o3 | dlynes_laptop: watch Trailer Park Boys? |
| 11:47.16 | cy3o3 | heh |
| 11:47.20 | E-bola | dlynes_laptop: but 102 isnt? and thats what sould be the next priority, after the call in 101? |
| 11:47.22 | dlynes_laptop | cy3o3: that show sucks |
| 11:47.29 | cy3o3 | awwww, I find it hilarious |
| 11:47.46 | dlynes_laptop | cy3o3: it's a bunch of red-neck trailer trash losers :) |
| 11:47.46 | brian | dlynes_laptop: my extensions.conf! it is riddled with deprecations and such |
| 11:48.01 | cy3o3 | dlynes_laptop: well... that's the idea of the show, yes. |
| 11:48.07 | dlynes_laptop | brian: so fix it, then |
| 11:48.08 | cy3o3 | and that's why it's funny as hell |
| 11:48.08 | cy3o3 | heh |
| 11:48.33 | brian | dlynes_laptop: i'm not smart enough |
| 11:48.39 | dlynes_laptop | brian: there's nothing wrong with the dialplan you showed me |
| 11:48.48 | dlynes_laptop | brian: it should do exactly what it says |
| 11:48.48 | brian | yes there is! |
| 11:48.55 | brian | it hangs up on me! |
| 11:48.59 | dlynes_laptop | brian: but you're using _666, and _ is a pattern matcher |
| 11:49.03 | dlynes_laptop | brian: 666 isn't a pattern |
| 11:49.06 | brian | after priority 8 it hangs up on me |
| 11:49.15 | brian | dlynes_laptop: 666? huh? |
| 11:49.18 | dlynes_laptop | brian: and there's no priority 8, either |
| 11:49.22 | brian | yes there is |
| 11:49.32 | dlynes_laptop | brian: then you gave me the wrong pastebin |
| 11:49.37 | dlynes_laptop | brian: smarten up |
| 11:49.38 | brian | exten => _XXXXXXXXXX,8,Set(TIMEOUT(response)=10) |
| 11:49.48 | brian | dlynes_laptop: http://rafb.net/paste/results/p2nmRY22.html |
| 11:49.56 | dlynes_laptop | brian: that's not the one you gave me |
| 11:50.01 | brian | weird |
| 11:50.06 | brian | what one did i give you |
| 11:50.11 | dlynes_laptop | brian: look up the documentation for TIMEOUT(response) |
| 11:50.22 | dlynes_laptop | brian: you obviously don't fully understand how it works |
| 11:51.10 | dlynes_laptop | brian: you gave me http://rafb.net/paste/results/wmsNxC32.html |
| 11:51.46 | brian | dlynes_laptop: i don't fully understand how it works? |
| 11:51.51 | dlynes_laptop | brian: no |
| 11:51.58 | brian | dlynes_laptop: it cuts you off after 10 seconds of no input... |
| 11:51.58 | dlynes_laptop | brian: read the documentation on it |
| 11:52.05 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 11:52.07 | dlynes_laptop | brian: yes, but only when you use it properly |
| 11:52.19 | brian | dlynes_laptop: and if i don't use it properly? |
| 11:52.28 | dlynes_laptop | brian: then it returns immediately |
| 11:53.00 | dlynes_laptop | brian: well, regardless, it returns immediately, but it causes the dialplan to wait when it's appropriate to do so |
| 11:53.15 | dlynes_laptop | brian: when it's expecting a response from the user |
| 11:53.34 | dlynes_laptop | brian: the end of the effective dialplan is not a place in the dialplan that waits for input from the user |
| 11:53.46 | dlynes_laptop | brian: dialplan terminations do not wait for input |
| 11:54.16 | dlynes_laptop | brian: Wait() waits for input, Background() waits for input, ... |
| 11:54.35 | dlynes_laptop | brian: Set(TIMEOUT(response)=...) does not wait for input |
| 11:54.58 | brian | dlynes_laptop: now it hangs up at Playing '/home/asterisk/confirm-voice-prompt' (language 'en') |
| 11:55.08 | brian | dlynes_laptop: i know it doesn't wait for input |
| 11:55.24 | dlynes_laptop | TIMEOUT(response)...keyword being 'response' |
| 11:55.41 | *** join/#asterisk JT (n=jon@141.168.62.0) |
| 11:55.49 | dlynes_laptop | E-bola: not sure |
| 11:55.57 | dlynes_laptop | E-bola: try putting that call into its own dialplan context |
| 11:56.34 | brian | I added Wait(10) dlynes_laptop |
| 11:56.37 | brian | It doesn't make any difference |
| 11:56.40 | brian | Wait I forgot to reload |
| 11:56.51 | dlynes_laptop | E-bola: i.e. exten => 43332730,100,Goto(velkomst) |
| 11:57.06 | dlynes_laptop | E-bola: i.e. exten => 43332730,100,Goto(velkomst,s,1) I mean |
| 11:57.24 | dlynes_laptop | E-bola: and then under [velkomst] put the rest of that dialplan for that number |
| 11:57.28 | brian | wait() doesn't wait for input |
| 11:57.33 | E-bola | hmmm its strange though.... |
| 11:57.35 | brian | it just makes asterick sleep for x seconds |
| 11:57.41 | brian | asterisk |
| 11:58.36 | dlynes_laptop | brian: sorry...meant waitexten, not wait |
| 11:58.41 | dlynes_laptop | brian: wait ignores all input |
| 11:59.02 | dlynes_laptop | brian: make sure the set(timeout(response)=...) is before the waitexten(), also |
| 11:59.07 | dlynes_laptop | brian: not after it |
| 11:59.53 | dlynes_laptop | brian: all timeout(response) does is set a variable in the dialplan that certain applications check |
| 12:00.02 | dlynes_laptop | brian: it doesn't actually do a delay by itself |
| 12:00.07 | E-bola | how hard is it to upgrade my 1.2.9 asterisk to the newest 1.2? |
| 12:00.19 | dlynes_laptop | E-bola: pretty easy and straight-forward |
| 12:00.46 | dlynes_laptop | E-bola: Unless you're using faxing, upgrading to a new version should not be fraught with problems |
| 12:00.47 | E-bola | dlynes_laptop: so easy there is no docs on it anywhere? |
| 12:00.47 | E-bola | hehe |
| 12:00.58 | dlynes_laptop | E-bola: man make |
| 12:01.10 | E-bola | can i just make install |
| 12:01.17 | E-bola | or do i need to do some manualy binary copying? |
| 12:01.27 | dlynes_laptop | E-bola: yeah...make sure you don't do a make samples though |
| 12:01.35 | dlynes_laptop | E-bola: or you'll overwrite all your config files |
| 12:01.54 | dlynes_laptop | E-bola: also keep in mind that make install will overwrite the sound files |
| 12:02.13 | dlynes_laptop | E-bola: so if you've got custom sound files that have replaced stock files, you might want to back them up before installing |
| 12:02.33 | E-bola | thanks for the advice |
| 12:02.37 | E-bola | ile do that and then try a make install |
| 12:03.00 | brian | dlynes_laptop: now when i press 1 or 2 i get "invalid extension" infinitely looped |
| 12:03.27 | dlynes_laptop | brian: becuase you don't have an extension 1 or 2 defined |
| 12:03.33 | brian | yes i do |
| 12:03.49 | brian | exten => 1,1,System(/bin/mv /usr/local/share/asterisk/sounds/mymessage.ulaw /home/asterisk/${TIMESTAMP}.ulaw) |
| 12:03.49 | brian | exten => 1,2,Playback(/home/asterisk/voice-prompt-saved) |
| 12:03.49 | brian | exten => 1,3,Playback(/home/asterisk/ask-record-again) |
| 12:04.09 | dlynes_laptop | brian: repost your new extensions.conf file |
| 12:04.20 | dlynes_laptop | brian: there's probably something messed up |
| 12:04.37 | brian | http://rafb.net/paste/results/2q4DGn10.html |
| 12:06.05 | brian | dlynes_laptop: k |
| 12:06.30 | E-bola | is it normal to get a few warnings when compiling? |
| 12:06.34 | E-bola | got some signed warnings in enum.c |
| 12:07.00 | dlynes_laptop | E-bola: yeah, that's normal |
| 12:07.06 | dlynes_laptop | E-bola: as long as there's no errors |
| 12:07.06 | brian | dlynes_laptop: hi ? |
| 12:07.17 | E-bola | dlynes_laptop: thanks |
| 12:07.56 | dlynes_laptop | E-bola: did you try adjusting your dialplan yet, to use a goto? |
| 12:08.15 | brian | dlynes_laptop: ! |
| 12:08.20 | E-bola | dlynes_laptop: nope not yet, wanted to try and update first |
| 12:08.32 | dlynes_laptop | brian: pastebin your log |
| 12:08.34 | E-bola | as brian pointed out i should ahve done it a long time ago anyway hehe |
| 12:13.27 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:14.34 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 12:15.06 | brian | dlynes_laptop: i fixed it |
| 12:15.28 | dlynes_laptop | brian: so what was your problem, then? |
| 12:16.26 | E-bola | dlynes_laptop: what about my asterisk addons? |
| 12:16.36 | E-bola | app_addon_sql_mysql.so cdr_addon_mysql.so etc |
| 12:16.42 | E-bola | will they need to be recompiled as well? |
| 12:16.49 | dlynes_laptop | E-bola: make ; make install for asterisk-addons, yes |
| 12:17.20 | dlynes_laptop | E-bola: and then cd format_mp3 ; make ; make install to make sure mp3 support gets installed from asterisk-addons |
| 12:18.10 | E-bola | thanks |
| 12:18.14 | E-bola | your an invalueable help :) |
| 12:18.32 | dlynes_laptop | no applause needed |
| 12:18.34 | dlynes_laptop | Just send money |
| 12:19.25 | *** join/#asterisk _Garry__ (n=gkg@epf-1.de-cix.net) |
| 12:19.59 | E-bola | :) |
| 12:20.31 | stephane | re |
| 12:22.01 | *** join/#asterisk beyond (n=beyond@c9346fb2.virtua.com.br) |
| 12:22.27 | E-bola | dlynes_laptop: asterisk's make install keep claiming that the addon modules arent made by that version |
| 12:22.39 | E-bola | even if i remove them and re-run a make install from the addon directory |
| 12:22.43 | E-bola | so i guess its alright |
| 12:23.00 | dlynes_laptop | E-bola: yes...that's why you have to run make ; make install from asterisk-addons __AFTER__ installing asterisk |
| 12:23.08 | E-bola | i did |
| 12:23.10 | dlynes_laptop | E-bola: i.e. don't try reinstalling asterisk after addons |
| 12:23.19 | E-bola | well i didnt remove the addons |
| 12:23.23 | E-bola | before installing asterisk |
| 12:23.23 | TimothyP | anybody using misdn? |
| 12:23.29 | E-bola | i installed newest asterisk, then enwest addons |
| 12:23.39 | dlynes_laptop | E-bola: it'll complain about those modules every time then, because the timestamp on asterisk is going to be newer than the modules froma ddons |
| 12:23.55 | E-bola | dlynes_laptop: ahh ok in that way, but everything should work fine regardless? |
| 12:23.59 | dlynes_laptop | E-bola: exactly...don't do asterisk, asterisk-addons, and then asterisk again...just do asterisk and then asterisk-addons |
| 12:24.20 | E-bola | hmm im on newest 1.2.x version |
| 12:24.26 | E-bola | but its still jumping +101 :( |
| 12:24.40 | dlynes_laptop | E-bola: like i said...fix your dial plan the way I asked you to :) |
| 12:24.49 | dlynes_laptop | E-bola: that'll help you nail down your problem |
| 12:25.01 | dlynes_laptop | E-bola: you might have part of the rest of your dial plan conflicting with it |
| 12:25.10 | E-bola | true |
| 12:26.50 | dlynes_laptop | E-bola: i don't see it happening, but then again, not everything is always obvious, either |
| 12:27.01 | dlynes_laptop | E-bola: so doing it the way i suggested is almost like having a second set of eyes |
| 12:27.55 | *** join/#asterisk ruied (n=ruied@mail.es-juliodantas.net) |
| 12:30.58 | *** join/#asterisk zapp-branigan (n=zapp-bra@81-202-140-115.user.ono.com) |
| 12:31.51 | E-bola | dlynes_laptop: again ur right |
| 12:32.01 | E-bola | if i make priority 101 a goto in another context |
| 12:32.10 | E-bola | and then have a simple dial and another dial as the next priority |
| 12:32.18 | E-bola | it correctly jumps to the second dial command if the first is busy |
| 12:32.25 | E-bola | now why on earth doesnt it work in my normal context? |
| 12:33.59 | dlynes_laptop | E-bola: because you've got something fuxored there |
| 12:34.10 | dlynes_laptop | E-bola: maybe an include is screwing everything up |
| 12:34.55 | zapp-branigan | hi, i'm new in this, how to know if a zap is busy ? |
| 12:36.09 | dlynes_laptop | zapp-branigan: ChanIsAvail(Zap/1-1) |
| 12:36.15 | brian | exten => 1,3,Goto(recordvoiceprompt_step2|s) |
| 12:36.19 | dlynes_laptop | zapp-branigan: or whatever zap channel you're looking at |
| 12:36.26 | brian | How do I properly Goto another extension dlynes_laptop |
| 12:36.43 | zapp-branigan | ok i have a digium card pci |
| 12:36.49 | brian | I mean not an extension dlynes_laptop, a context |
| 12:37.01 | zapp-branigan | and i want send a call for the avaliable zap |
| 12:37.52 | *** join/#asterisk lorinc (n=ang@caracas-2383.adsl.interware.hu) |
| 12:38.15 | zapp-branigan | the extension.conf can be programed as c or something else ? |
| 12:38.25 | zapp-branigan | i see more easy the c :P |
| 12:38.27 | E-bola | dlynes_laptop: i have no includes... |
| 12:38.31 | E-bola | and uve seen my entire dialplan :( |
| 12:38.51 | brian | dlynes_laptop: heeey how do i goto another context |
| 12:39.26 | E-bola | hmm i tracked it closer! |
| 12:39.30 | E-bola | its my gotoifdate |
| 12:39.43 | dlynes_laptop | brian: Goto(contextname,extension,priority) |
| 12:39.48 | RoyK | I wonder how asterisk compares to this http://www.voip-info.org/wiki/view/OpenPBX.org#TargetedOSplatformsandarchitectures |
| 12:39.57 | dlynes_laptop | E-bola: so it's not priority jumping then |
| 12:40.11 | E-bola | exten => 43332730,1,GotoIfTime(9:00-17:00|mon-thu|*?Incoming,43332730,100) |
| 12:40.11 | E-bola | exten => 43332730,2,GotoIfTime(9:00-16:30|fri|*?43332730,100) |
| 12:40.11 | E-bola | exten => 43332730,3,Goto(Incoming,43332730,200) |
| 12:40.15 | dlynes_laptop | E-bola: it's not going to the original line number you thought it was |
| 12:40.16 | E-bola | these 3 lines somehow are wrong |
| 12:40.36 | E-bola | they should simply jump to priority 100 if we are open, and 200 if we are not |
| 12:40.44 | E-bola | atm i cant figure out what they are doing heh |
| 12:40.54 | dlynes_laptop | E-bola: You can do time-based includes, too |
| 12:41.04 | E-bola | dlynes_laptop: i know, but this seemed easier |
| 12:41.09 | dlynes_laptop | E-bola: that might help you narrow down your problem as well |
| 12:41.20 | dlynes_laptop | E-bola: then you can do a noop as soon as you enter the included context |
| 12:43.36 | E-bola | dlynes_laptop: what does the ? do in gotoiftime? |
| 12:43.39 | E-bola | its not explained anywhere |
| 12:45.54 | E-bola | hmm |
| 12:45.59 | E-bola | it continoues to jump +101 |
| 12:46.03 | E-bola | in the same extension |
| 12:46.07 | E-bola | and i simpyl cant see why.... |
| 12:46.21 | dlynes_laptop | E-bola: ? is a conditional operator |
| 12:46.41 | E-bola | i think this might be a bug... |
| 12:47.00 | E-bola | it seems using gotoiftime enables priority jumping... |
| 12:47.07 | dlynes_laptop | E-bola: it means if the previous expression is true, it goes to Incoming context, extension 43332730, priority 100 |
| 12:47.19 | E-bola | dlynes_laptop: ahh ok |
| 12:47.35 | E-bola | can anybody test? |
| 12:47.37 | dlynes_laptop | E-bola: it's the same as the C operator |
| 12:47.50 | E-bola | my claim is that if u gotoiftime priority jumping is enabled |
| 12:47.58 | dlynes_laptop | E-bola: a==b ? 1 : 2 |
| 12:48.10 | dlynes_laptop | E-bola: if( a==b ) { then 1 ; } else { 2 ; } |
| 12:48.27 | *** join/#asterisk DrAk0SX (n=luisjose@190.38.151.37) |
| 12:48.35 | TimothyP | oh that's great, fwd's email isn't working, 55555 isn't working ... what are they up to ? |
| 12:48.53 | E-bola | dlynes_laptop: ye i get it now, just think its a bit weird syntax to use in this context but fair enough |
| 12:49.23 | *** join/#asterisk gaspiz (n=gaspiz@86.35.34.63) |
| 12:49.47 | dlynes_laptop | TimothyP: it's called quality server |
| 12:49.49 | dlynes_laptop | TimothyP: it's called quality service |
| 12:49.58 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 12:50.40 | gaspiz | hi, any agi interface gurs? I have a hard question/ possible bug |
| 12:51.02 | E-bola | heh |
| 12:51.04 | E-bola | this is strange |
| 12:51.06 | E-bola | i finally fixed it |
| 12:51.08 | E-bola | priorityjumping=no |
| 12:51.14 | E-bola | in the general context of extensions.conf |
| 12:51.25 | E-bola | i dont get it at all though, i thought it was supposed to be default=no? |
| 12:51.34 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 12:51.35 | E-bola | but maybe you meant that default in the sample conf it was set as no? |
| 12:52.16 | gaspiz | using deadagi I dial a person. After the called person hangs up I make a sleep for n seconds. The channel for the caller is hung up only AFTER those n seconds |
| 12:52.52 | TimothyP | dlynes_laptop that was a joke right? |
| 12:53.17 | dlynes_laptop | TimothyP: of course :) |
| 12:54.11 | TimothyP | hehe |
| 12:55.07 | dlynes_laptop | E-bola: and which version of asterisk are you using? |
| 12:55.13 | dlynes_laptop | E-bola: i thought you were using 1.2.9.1? |
| 12:58.04 | cy3o3 | ugh |
| 12:58.11 | cy3o3 | so what the hell is up with this? : |
| 12:58.12 | cy3o3 | Nov 29 05:55:27 WARNING[5961]: res_musiconhold.c:421 spawn_mp3: Found no files in '/usr/share/asterisk/mohmp3' |
| 12:58.15 | cy3o3 | Nov 29 05:55:27 WARNING[5961]: res_musiconhold.c:493 monmp3thread: Unable to spawn mp3player |
| 12:59.11 | cy3o3 | oh.. well I guess the files are in /var/lib/asterisk/mohmp3 heh |
| 13:02.21 | *** join/#asterisk stuq (n=stuq@user-12lcqia.cable.mindspring.com) |
| 13:06.40 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 13:07.36 | angryuser | good day, i have routed port 5060 to server, but from external my sip client still have no sound, which ports do i need to route more? |
| 13:08.42 | gaspiz | angryuser: did you open the rtp ports? |
| 13:08.47 | brian | how do I detect hangup so I can do some clean up? |
| 13:09.02 | brian | is there a special extension like s for hangup |
| 13:09.08 | gaspiz | brian: h |
| 13:09.09 | dlynes_laptop | brian: h |
| 13:09.13 | brian | so obvious |
| 13:09.14 | brian | :P |
| 13:09.16 | angryuser | gm not sure |
| 13:09.46 | gaspiz | angryuser: the sound is sent on a higher port 10000 to 20000 I guess |
| 13:10.03 | angryuser | udp? |
| 13:10.12 | gaspiz | angryuser: for sound you need to enable them in your firewall |
| 13:10.17 | gaspiz | yes udp |
| 13:10.23 | angryuser | ok il try |
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| 13:10.57 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 13:11.14 | dlynes_laptop | good morning, drumkilla |
| 13:11.27 | cy3o3 | any way to test music on hold using mpg123 from the console with no client connected to the server? heh.. just to make sure it's working somehow? |
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| 14:15.34 | froguz | hi |
| 14:16.20 | froguz | do i need to install Libpri even if i'm not going to configure zaptel hardware? |
| 14:17.19 | froguz | my guess is that i don't need it, but i preffer to be really sure of that |
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| 14:26.59 | Katty | morning. |
| 14:28.22 | Juggie | froguz, no. |
| 14:28.48 | froguz | Juggie, thanks |
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| 14:40.58 | Dr-Linux | what's wrong with this: |
| 14:40.58 | Dr-Linux | Nov 29 06:38:05 NOTICE[22990]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 14:44.42 | __AK__ | you have to disable silence suppression on your phone |
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| 14:45.14 | *** join/#asterisk wiseguy_ (n=chivilis@infospalvos.lt) |
| 14:45.16 | wiseguy_ | hello |
| 14:45.29 | wiseguy_ | anybody using AVM Fritz cards having echo problems? |
| 14:45.38 | Dr-Linux | hhm.. ok |
| 14:45.54 | wiseguy_ | http://groups.google.lt/group/Asterisk-users/browse_thread/thread/25f7a6baaa06cf48/1e292eb10eca1fca?lnk=st&q=&rnum=1&hl=lt#1e292eb10eca1fca - i have the same problem |
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| 14:46.46 | __AK__ | you're welcome Dr-Linux ...... |
| 14:48.06 | puzzled | wiseguy_: did you try chan_capi-0.7.1? |
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| 14:54.01 | angryuser5883 | ;) |
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| 15:02.03 | sergee | is there Asterisk logo in high resolution anywhere? i need it to preapre some papers but can't find it with google :( maybe someone will share image with me? :) |
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| 15:03.51 | heh_v_water | Has anyone got experience compiling the jiax client |
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| 15:09.47 | heroine | hi ppl |
| 15:09.56 | Winkie | hi there |
| 15:11.30 | wiseguy_ | puzzled: yes |
| 15:11.39 | wiseguy_ | puzzled: and you? |
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| 15:12.25 | santibiotico | hi |
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| 15:29.03 | Katty | so, if i had a polycom phone who's light didn't flash when they got a voicemail...but it flashes on boot and works and such.......where do i go hunting for answers? |
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| 15:31.16 | hmmhesays | ~seen JunK-Y |
| 15:31.31 | jbot | junk-y is currently on #asterisk (2d 20h 28m 11s), last said: 'going to bed, peace hmmhesays !'. |
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| 15:32.32 | [420]riOn | hello all, can anyone help me with a simple question about a TDM400P |
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| 15:41.34 | russellb | ~lart file |
| 15:41.40 | file | eeeeep |
| 15:42.12 | TimothyP | Is is possible to somehow (perhaps using AGI) make Asterisk force a SIP phone to make a call to a number, so that i could have a desktop appliction with all my contacts in it and when I press a contact my phone calls that number? |
| 15:42.58 | russellb | you can do that pretty easily using the manager interface |
| 15:43.02 | russellb | using the Originate action |
| 15:43.15 | russellb | you would originate a call between your phone and the number you want to connect it to |
| 15:43.46 | heroine | damn |
| 15:44.08 | heroine | has somebody managed to use the pickup command with ser in front of asterisk ? |
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| 15:46.25 | equinox0r | hi.. |
| 15:46.47 | equinox0r | i have some problems with my mysql/sip authentication .. |
| 15:46.59 | Katty | file: do you know why my phones have no flashy when they have a voicemail? )= |
| 15:47.06 | Katty | file: they elude me. |
| 15:47.09 | file | Katty: no :( |
| 15:47.10 | equinox0r | the sql-statement is sent correctly to the mysql server but asterisk tells me that the username or auth name mismatches .. |
| 15:47.21 | Katty | file: i think it's cause people don't hug them enough )= |
| 15:47.25 | Katty | file: my phone works fine )= |
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| 15:48.01 | equinox0r | anyone got an idea why i cant authenticate? |
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| 15:48.42 | sloth_ | When compiling zaptel on a FC4 box what do i need for kernel sources? |
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| 16:02.14 | TimothyP | russellb the manager interface? the idea is that I create small desktop app which gets all contacts from our crm :) |
| 16:02.28 | Ifaistos | sloth_: The sources of the kernel you use |
| 16:02.41 | sloth_ | thnx |
| 16:03.51 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:04.01 | blitzrage | anyone know if there is a variable that keeps track of the filename for One-Touch Recording? |
| 16:05.11 | blitzrage | file: see above! :D |
| 16:05.56 | heroine | Katty: do you use asterisk or ser/asterisk ? |
| 16:06.04 | ACiDV | I have Pickup Groups working perfectly. I use GROUPCOUNT to limit calls on an extensions, but the channel that pick the call isnt 'controlled' by dialplan. Does it's possible to set a variable or whatever to increment GroupCount on pickup call ? |
| 16:06.51 | ACiDV | I think that this can be possible with some hack on ast_pickup_call function (res_features.c) but need to know if others way exist |
| 16:07.14 | *** join/#asterisk denny (n=denny@about/linux/staff/pdpc.professional.denny) |
| 16:09.16 | Katty | heroine: asterisk. |
| 16:09.30 | *** part/#asterisk denny (n=denny@about/linux/staff/pdpc.professional.denny) |
| 16:09.31 | Katty | file: i think i found something! (dundundun) |
| 16:10.24 | BSDTech | I believe its in thsql and its recordings |
| 16:10.35 | BSDTech | th my |
| 16:10.43 | BSDTech | cant tpye |
| 16:10.54 | file | Katty: a good something, or a bad something? |
| 16:10.56 | heroine | i'm really fed up with ser breaking any asterisk implementation of features |
| 16:11.22 | heroine | pickup fucked up ... blf fucked up .. nice \o/ |
| 16:12.32 | *** join/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at) |
| 16:14.18 | nicox | Hello |
| 16:15.01 | hmmhesays | since when does ser break asterisk? |
| 16:15.18 | blitzrage | not knowing how to do something doesn't mean the software broke it :) |
| 16:17.15 | Katty | file: will you look at a tcpdump for me? |
| 16:17.37 | windygeek | A few months I was talking to a SIP provider based the the US's east coast. They offered incoming channels for $20 each per month, but I can't find them now. Any idea who I'm talking about? |
| 16:17.48 | heroine | hmmhesays: let's that some features implemented in asterisk seems to badly work with ser in front of asterisk |
| 16:19.12 | hmmhesays | hey katty |
| 16:19.53 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 16:20.05 | Katty | hmmhesays: how ya doin hun? |
| 16:23.26 | hmmhesays | scratching my head |
| 16:23.37 | hmmhesays | anyone know anything about direct tv boxes and when they use the phone line? |
| 16:23.58 | hmmhesays | I got a two ports on my gateway that go dead when you plug a dtv box into them |
| 16:24.48 | *** join/#asterisk hfb (n=hfb@pool-71-118-252-158.lsanca.dsl-w.verizon.net) |
| 16:26.35 | BSDTech | anyone know who did the actual port of flite to bsd |
| 16:26.52 | BSDTech | I need input on updating it to 1.3 from 1.2 |
| 16:27.17 | BSDTech | seems it has not been updated in awhile |
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| 16:29.42 | sloth_ | what module do i load for TE210P? |
| 16:30.55 | *** join/#asterisk ramtha (n=ramtha@195.14.234.165) |
| 16:30.57 | ramtha | hi |
| 16:31.11 | ramtha | is there any chance to get asterisk 1.4 and zaptel 1.2 working together? |
| 16:31.19 | SplasPood | why? |
| 16:31.36 | ramtha | because my server freezes wenn i install zaptel 1.4 |
| 16:31.39 | MrChimpy | Background appears to return an extension... |
| 16:31.39 | ramtha | i dont need it |
| 16:31.47 | ramtha | but i need t.38 passthrough |
| 16:31.58 | SplasPood | well you need it insofar as i believe asterisk 1.4 "requires" it |
| 16:32.07 | ramtha | ok |
| 16:32.17 | ramtha | i thought about something like that |
| 16:32.30 | SplasPood | It works well for me.. it actually freezes your system? |
| 16:32.36 | SplasPood | capture any output before it does? |
| 16:33.23 | ACiDV | Anyone have example of Feature Proxy Channel use ? Dial(Feature/Tech/Dest) ? |
| 16:33.31 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
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| 16:34.33 | ramtha | i don´t now......the server is not here, it is in the serverroom and it doesn´t come up again |
| 16:34.34 | MrChimpy | Read has a "filename" field for "sample to play before reading the digits" but that doesn't imply it can be interrupted |
| 16:35.48 | zapp-branigan | hi, i'm compiling the new zaptel 2.4 but there are some errors in my computer |
| 16:35.51 | zapp-branigan | /home/asterisk/zaptel-1.4.0-beta2/zconfig.h:9:26: error: linux/config.h: No existe el fichero o el directorio |
| 16:35.51 | zapp-branigan | make[2]: *** [/home/asterisk/zaptel-1.4.0-beta2/pciradio.o] Error 1 |
| 16:35.52 | zapp-branigan | make[1]: *** [_module_/home/asterisk/zaptel-1.4.0-beta2] Error 2 |
| 16:35.58 | *** join/#asterisk spunz (n=spunz@h081217096236.dyn.cm.kabsi.at) |
| 16:36.05 | ramtha | install linux source |
| 16:36.09 | zapp-branigan | can't find the linux/config.h |
| 16:36.20 | zapp-branigan | yum install yum-utils kernel-devel |
| 16:36.31 | ramtha | you need kernel-headres |
| 16:36.51 | ramtha | headers |
| 16:36.55 | zapp-branigan | yumdownloader kernel-2.6.18-1.2798.fc6.i686 |
| 16:37.21 | zapp-branigan | su -c "rpm -Uvh --replacefiles --replacepkgs kernel-2.6.18-1.2798.fc6.i686.rpm" |
| 16:37.22 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 16:37.32 | djflux | config.h doesn't exist in the fc6 kernel-devel package |
| 16:37.40 | zapp-branigan | where is ? |
| 16:37.55 | ramtha | in the header package!!! |
| 16:37.56 | djflux | you can safely comment that line out in the code or symlink to autoconf.h |
| 16:38.03 | ramtha | or that way |
| 16:38.04 | ramtha | : |
| 16:38.08 | djflux | ramtha: it doesn't exist in the package anymore on FC6 |
| 16:38.33 | zapp-branigan | yum install kernel-headers |
| 16:38.36 | zapp-branigan | and too |
| 16:39.41 | zapp-branigan | but the zaptel work if i remove this line? |
| 16:39.47 | djflux | yes |
| 16:39.50 | zapp-branigan | := |
| 16:40.04 | zapp-branigan | ok |
| 16:40.06 | zapp-branigan | :) |
| 16:40.25 | zapp-branigan | and this line what is the use to do =? |
| 16:40.30 | Katty | i'm so confuzzled. |
| 16:40.33 | hmmhesays | anyone use direct tv with an ATA? |
| 16:40.37 | Katty | my phone claims 0 gmt )= |
| 16:40.48 | Katty | but the phone settings are -6, and the ftp is -6. |
| 16:41.56 | zapp-branigan | in zaptel i not found autoconf.h |
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| 16:43.54 | djflux | autoconf.h is in the /lib/modules/`uname -r`/build/include/linux |
| 16:44.02 | djflux | directory |
| 16:44.40 | djflux | not sure what it used to do, but there wasn't anything really interesting in the FC5 kernel-devel package |
| 16:45.07 | zapp-branigan | ok |
| 16:45.09 | djflux | zaptel compiles fine without that line and it's actually removed of ifdef'd out in svn |
| 16:45.15 | djflux | s/of/or |
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| 16:46.45 | zapp-branigan | in this directory there is olly 2 files.fc6 |
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| 16:47.32 | zapp-branigan | directories |
| 16:47.33 | zapp-branigan | ok |
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| 16:50.34 | santibiotico | i have a problem with musiconhold....it was working ok, but right now, when it is suposed to be playing music, music does not play...and i get the following message: |
| 16:50.44 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 16:50.50 | santibiotico | <PROTECTED> |
| 16:50.51 | santibiotico | <PROTECTED> |
| 16:50.51 | santibiotico | <PROTECTED> |
| 16:50.54 | *** join/#asterisk Katty_ (n=angela@hera.copi-rite.com) |
| 16:50.59 | santibiotico | any idea?? |
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| 16:55.53 | *** join/#asterisk slappin (n=slappin@c-69-247-217-56.hsd1.mo.comcast.net) |
| 16:56.00 | *** join/#asterisk dacleric (n=dacleric@p54821723.dip0.t-ipconnect.de) |
| 16:57.50 | blitzrage | anyone happen to know if you can use MixMonitor() to do the recording in one-touch recording (wW flag in Dial()) instead of standard Monitor() ? |
| 16:58.10 | Qwell[] | blitzrage: hack the code? |
| 16:58.23 | blitzrage | Qwell[]: was afraid you might say that :) |
| 16:58.30 | Qwell[] | I don't know, otherwise |
| 16:58.38 | blitzrage | I think I'll be better off just mixing the channels myself in my DeadAGI then |
| 16:58.49 | blitzrage | problem is my DeadAGI is moving those files around before Asterisk mixes them itself |
| 16:59.25 | blitzrage | lets hope asterisk doesn't freak out when it can't find the files |
| 16:59.41 | SplasPood | asterisk, freak out? never.... ;) |
| 16:59.46 | blitzrage | :) |
| 16:59.50 | Qwell[] | It's 100% error free, of course. |
| 16:59.54 | blitzrage | obviously |
| 16:59.57 | blitzrage | everything is user-error |
| 17:00.55 | blitzrage | goddamn layer 8 |
| 17:00.56 | Qwell[] | heh |
| 17:00.56 | SplasPood | he |
| 17:01.18 | file | blitzrage: you dare strike me? |
| 17:01.32 | blitzrage | I date and I do |
| 17:01.35 | blitzrage | and I'd do it again |
| 17:01.42 | file | you... date? |
| 17:01.49 | blitzrage | s/date/dare |
| 17:02.01 | blitzrage | I obviously don't date; I use Asterisk |
| 17:02.39 | blitzrage | Qwell[]: so once I get some of this stuff done, I can start doc'ing IMAP+voicemail :) |
| 17:02.40 | *** join/#asterisk Del_Mon (n=de_mon@fl-69-69-155-23.dyn.embarqhsd.net) |
| 17:02.47 | Qwell[] | blitzrage: sweet |
| 17:02.49 | blitzrage | I'm getting ever so close |
| 17:02.54 | Qwell[] | blitzrage: You do know I was just joking though, right? :) |
| 17:02.54 | file | blitzrage: yay |
| 17:03.05 | blitzrage | Qwell[]: yah well, I wanna play with it anyways |
| 17:03.08 | Qwell[] | heh |
| 17:03.09 | blitzrage | and it'd be a good thing to doc I think |
| 17:03.12 | Qwell[] | I was hoping you might say that |
| 17:03.14 | Qwell[] | ...and that |
| 17:03.16 | santibiotico | anyone can help me with musiconhold?? |
| 17:03.19 | file | I was also hoping that... |
| 17:03.20 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:03.26 | blitzrage | I was hoping that you'd hope |
| 17:03.30 | rob0 | IMAP+voicemail, does that mean * will have an IMAP daemon? |
| 17:03.33 | Qwell[] | I was h...ahh hell |
| 17:03.40 | blitzrage | lol |
| 17:03.48 | Qwell[] | rob0: No, it connects to an IMAP client to save/read voicemail |
| 17:03.53 | Qwell[] | ie; your email server |
| 17:04.15 | DrAk0SX | anyone here from spain? |
| 17:04.30 | blitzrage | no, but I've been to Spain once :) |
| 17:04.46 | Qwell[] | blitzrage: That clearly makes you a representative of the country |
| 17:04.49 | rob0 | I've heard about the rain there, staying mainly in the plain. |
| 17:05.03 | file | c'est la vie! |
| 17:07.27 | blitzrage | Qwell[]: clearly |
| 17:09.29 | file | blitzrage: would you have all the money in the world if you were a wealthy girl? |
| 17:09.44 | *** join/#asterisk knathraak (n=zach@151.196.142.242) |
| 17:09.58 | *** part/#asterisk Zothar_Work (n=chatzill@ppp-70-242-106-135.dsl.rcsntx.swbell.net) |
| 17:09.58 | blitzrage | file: sounds like a trick question |
| 17:10.05 | blitzrage | lets see what Qwell[] would say |
| 17:10.18 | knathraak | hi, all. need some pointers with music-on-hold. |
| 17:10.20 | Qwell[] | Nope, he wouldn't. |
| 17:10.31 | sloth_ | If I have an ata behind nat and and asterisk 1.4 b3 with public ip, can i do t.38 passthru? |
| 17:11.04 | knathraak | i need to dial a series of numbers (fxo channels) sequentially until someone answers, and I need seamless music on hold the entire time |
| 17:11.27 | knathraak | MusicOnHold() doesn't return control like Background(), so that won't work. |
| 17:11.58 | knathraak | and using the m option for Dial() results in a break in the music, between dialed extensions. |
| 17:12.08 | knathraak | any ideas? |
| 17:13.12 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 17:13.14 | CunningPike | knathraak: Would some dialplan trick using Transfer() work? |
| 17:13.43 | *** join/#asterisk stephane (n=stephane@nostromo.cabale.net) |
| 17:13.55 | knathraak | CunningPike: hmm... not sure how transfer works(i'll go look it up)...what are you thinking |
| 17:14.35 | stephane | re |
| 17:14.37 | CunningPike | knathraak: Not sure if it will work either, but what I'm thinking is that, all the time you're doing an attended transfer, the party being transfered gets to hear MOH |
| 17:14.40 | *** join/#asterisk topping_ (n=topping@207.47.6.182.static.nextweb.net) |
| 17:16.20 | knathraak | CunningPike: sounds interesing....looking it up now |
| 17:16.36 | CunningPike | knathraak: OK - just a possibility...... |
| 17:16.57 | dennisharrison | alright |
| 17:17.12 | dennisharrison | so, I compile and configure zaptel or asterisk first ? |
| 17:17.14 | dennisharrison | zap huh> |
| 17:18.05 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 17:18.29 | Corydon-w | zaptel first |
| 17:23.42 | dennisharrison | thanks |
| 17:29.03 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 17:29.55 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 17:31.42 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-235-141.red.bezeqint.net) |
| 17:34.45 | Katty | Corydon-w: if you put settings into sip.cfg on ftp, and then settings into a polycom phone, who wins? |
| 17:36.02 | Corydon-w | The settings on the phone override |
| 17:37.07 | Katty | Corydon-w: if i told it -6 gmt via the soft buttons, and at the IP too.. |
| 17:37.18 | Katty | Corydon-w: any idea why status says 0{0.0}gmt |
| 17:37.26 | [TK]D-Fender | Katty: sip.cfg - phoneXXX.cfg - <mac>-phone.cfg (local). In order of application. |
| 17:38.19 | Corydon-w | Katty: what does your DHCP server say? |
| 17:38.32 | [TK]D-Fender | Katty: I'd suggest wiping out any <mac>-phone.cfg for it, and make sure that your DHCP server is handing out the GMT if at all possible. If not, you may want to upgrade your firmware as there is better control in the newer release. |
| 17:38.37 | Corydon-w | I typically set the timezone offset at the dhcp server |
| 17:38.42 | Katty | [TK]D-Fender: yeah, i've wiped out all of them. |
| 17:38.48 | Katty | Corydon-w: well... |
| 17:38.52 | Corydon-w | Katty: how did you wipe them? |
| 17:38.58 | Katty | Corydon-w: i thought so too, but our dhcp server is a firewall. |
| 17:39.04 | Katty | Corydon-w: hardware firewall. |
| 17:39.16 | Katty | Corydon-w: so i changed the time server at the hardware firewall and it didn't affect the phones at all. |
| 17:39.19 | Katty | Corydon-w: i deleted them. |
| 17:39.28 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.89) |
| 17:39.30 | Corydon-w | Katty: not time server. Time offset. |
| 17:39.37 | Katty | [TK]D-Fender: phones don't pull anything from sip.cfg. |
| 17:39.38 | Corydon-w | Time server always sends as GMT |
| 17:39.54 | Katty | [TK]D-Fender: and now there are no phone cfg files on the ftp, because i deleted them. |
| 17:39.58 | Corydon-w | Katty: time offset is usually specified in seconds |
| 17:40.05 | Katty | [TK]D-Fender: but the phone still think that it's 0 gmt |
| 17:40.21 | Katty | Corydon-w: i've doublechecked it's in seconds in sip.cfg, doesn't matter. phones always think they're 0 gmt |
| 17:40.30 | Corydon-w | Katty: have you tried a reformat of the filesystem on the Polycom yet? |
| 17:40.33 | Katty | they're right when they first come up. |
| 17:40.35 | [TK]D-Fender | Katty: Also key idea : change your refresh timeout in sip.cfg to like 2 minutes for a test |
| 17:40.42 | Katty | but after their sync period, they switch to 0, rather than 6 |
| 17:40.49 | [TK]D-Fender | Katty: z0mg |
| 17:40.51 | *** join/#asterisk dhill (i=dhill@fog.mindcry.org) |
| 17:40.53 | Katty | [TK]D-Fender: yeah, i kicked it way down. |
| 17:40.57 | Katty | [TK]D-Fender: to 100 |
| 17:41.03 | Katty | [TK]D-Fender: didn't make a difference at all. |
| 17:41.12 | [TK]D-Fender | Katty: Then its being overriden on either the phone or your sip.cfg |
| 17:41.21 | [TK]D-Fender | Katty: And coming right from DHCP |
| 17:41.46 | sloth_ | does anyone know why i get "process_sdp: Unsupported SDP media type in offer: image 2672 udptl t38"? |
| 17:42.04 | Katty | [TK]D-Fender: well it can't come from dhcp. |
| 17:42.27 | Corydon-w | Katty: Menu / Settings / Advanced / Admin / Reset / Format |
| 17:42.28 | Katty | [TK]D-Fender: i went and looked at the firewall, and there's no option for setting a time server in there |
| 17:42.39 | Katty | Corydon-w: this isn't a single phone issue. it's company wide. |
| 17:42.58 | Katty | Corydon-w: i somehow doubt formatting all of our phones is going to do the trick here ;) |
| 17:43.02 | Corydon-w | Katty: try one phone and get that one working |
| 17:43.07 | *** join/#asterisk [420]riOn (n=ryan@c-69-242-252-152.hsd1.fl.comcast.net) |
| 17:43.10 | dhill | i am using asterisk 1.2.13 - if i have call-limit=1 set for a friend. if we are on the phone.. it shows 1 in use (sip show inuse). if we remain on the phone and i do a "sip reload", it goes to 0 in use.. but we are still on the phone... is this a known bug? |
| 17:43.13 | [TK]D-Fender | Katty: Ok, take a troubled phone and upgrade it to 2.0.X, rebuild a set of configs for it |
| 17:43.32 | Corydon-w | Katty: I'm constantly surprised at what all formatting the filesystem on the Polycoms will fix. ;-) |
| 17:43.47 | [420]riOn | can someone answer me a simple question about the zaptel.conf file? |
| 17:43.51 | Katty | Corydon-w: that's not a step i'm willing to take at the moment. :P |
| 17:44.04 | *** join/#asterisk Shido6 (i=shido8@d221-68-200.commercial.cgocable.net) |
| 17:44.11 | Corydon-w | Katty: all it does it redownload the settings from the FTP server |
| 17:44.23 | Corydon-w | Katty: it's fairly harmless |
| 17:44.46 | Corydon-w | Katty: but you then absolutely KNOW there's no errant settings left on the phone |
| 17:45.28 | Katty | Corydon-w: i'm not going to format a couple hundred phones :P |
| 17:45.34 | Katty | i'll format mine tho. |
| 17:45.42 | Corydon-w | Katty: I'm not suggesting all for the time being |
| 17:45.56 | Katty | [TK]D-Fender: i think you're going to be right. |
| 17:46.12 | Corydon-w | I just know that the Polycoms tend to accumulate cruft in the local config |
| 17:46.39 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 17:47.04 | dennisharrison | ~zaptel |
| 17:47.30 | jbot | [zaptel] zapata telephony interface. A low level interface designed to abstract hardware access to a variety of devices for BRI, PRI or analogue access. |
| 17:52.28 | [420]riOn | can someone please help me with the zaptel.conf file for a TDM400P card? i have a very simple question |
| 17:54.09 | Katty | well i formatted my phone. |
| 17:54.23 | CunningPike | [420]riOn: Try asking it |
| 17:54.39 | Katty | and it's still 0 gmt |
| 17:55.42 | *** join/#asterisk slappin (n=slappin@c-69-247-217-56.hsd1.mo.comcast.net) |
| 17:55.49 | *** join/#asterisk sloth_ (n=josh@64.3.170.41.ptr.us.xo.net) |
| 17:56.28 | Katty | the phone's ip pulled it from sip.cfg tho |
| 17:56.45 | [420]riOn | ok, my card has 2 FXS ( port 1 & 2) and 2 FXO ports (port 3 & 4), in the zaptel.conf file do i use fxoks=3,4 or fxoks=1,2? |
| 17:56.52 | Katty | and if you get to the settings via the softphones, it's right too |
| 17:57.09 | *** join/#asterisk cloud9 (n=root@smtp.a-cti.com) |
| 17:57.11 | sloth_ | anyone using t.38 passthru with asterisk 1.4? |
| 17:57.16 | cloud9 | what up |
| 17:57.43 | *** part/#asterisk slappin (n=slappin@c-69-247-217-56.hsd1.mo.comcast.net) |
| 17:58.00 | SplasPood | Is setting a Dial() timeout of 0 the same as not passing a timeout value at all? |
| 17:58.24 | rob0 | [420]riOn: genzaptelconf is your friend |
| 17:59.26 | [420]riOn | ok, i was reading the digium manual that was sent with the card, i will try that thank you |
| 17:59.33 | *** join/#asterisk LordBacon (n=frb@75.160.2.201) |
| 17:59.34 | rob0 | But my guess, included at No Extra Charge, is that you use fxoks=1,2\nfxsks=3,4 |
| 17:59.42 | LordBacon | oink, oink |
| 18:00.46 | LordBacon | I just got asterisk 1.0.9 on suse 10.0, is there a good document for getting this running as quickly as possible? |
| 18:01.09 | rob0 | ~wikis |
| 18:01.20 | jbot | extra, extra, read all about it, wikis is http://www.voip-info.org |
| 18:02.18 | *** join/#asterisk ESCulapio__ (n=ESCulapi@200.88.44.66) |
| 18:02.52 | LordBacon | that's what I was looking for, I didn't find a link on asterisk.org |
| 18:04.20 | ESCulapio__ | <PROTECTED> |
| 18:04.20 | ESCulapio__ | Traducción de Babel Fish Ayuda |
| 18:04.20 | ESCulapio__ | In English: |
| 18:04.20 | ESCulapio__ | as I say to him to call to manager that envie the call to asterisk and vice turns working with oh323 |
| 18:04.43 | ESCulapio__ | as I say to him to call to manager that envie the call to asterisk and vice turns working with oh323 |
| 18:06.29 | *** join/#asterisk Fibersrv (n=IceChat7@66-189-233-17.dhcp.cpgr.mo.charter.com) |
| 18:06.50 | blitzrage | LordBacon: 1.0.9? that's pretty old! |
| 18:07.38 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:08.24 | heh_v_water | Has anyone ever compiled jiaxclient before? I've tried with 1.4 and 1.5 versions of Sun Java, tried various versions of gcc amongst other thigns with no luck.. right now the compile is spitting out a pthread error but i have pthread installed.. |
| 18:08.27 | cloud9 | hey, anybody here having success with faxes using asterisk? |
| 18:08.57 | Fibersrv | How does Asterisk handle a rollover line? For example, a school has 1 phone line incoming, but it rolls over to 4 other phone numbers if the main line is busy. |
| 18:09.31 | dhill | it handles it just fine |
| 18:09.41 | *** join/#asterisk mw46_1 (i=marcow@nat/sun/x-c1b575cb252699dd) |
| 18:09.54 | Fibersrv | What type of PC, also would be recommended on handling the phone server? |
| 18:10.07 | SplasPood | PS/2 |
| 18:10.26 | Fibersrv | what specifications I meant to say, sorry. |
| 18:10.35 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-183.mtco.com) |
| 18:10.55 | SplasPood | Fibersrv: depends on what you're doing.. there were some reasonable dimensioning docs on www.voip-info.org |
| 18:11.30 | heh_v_water | Fibersrv, I believe i seen the specs o voip-info.org.. IIRC it all depends on how many lines and how much phone traffic you have |
| 18:12.00 | *** join/#asterisk b11d (n=noway@234-200-29-134.hcc.mnscu.edu) |
| 18:12.01 | b11d | allo |
| 18:13.02 | Fibersrv | Basically a school wants to switch over to VOIP. They have 1 phone line incoming, that rolls over to 4 additional phone lines when the main one is busy. They have no extensions going to the classroom, as of yet. So, what I am trying to do is recommend a system (PC) to them so that it will handle the traffic. There will not be a lot of traffic, it is a small school. |
| 18:13.28 | dhill | a small system will work fine.. |
| 18:13.32 | dhill | pII :) |
| 18:13.54 | Fibersrv | Also, it has been told to me that Asterisk will only work with a T-1. Will it not work with a cable connection? |
| 18:13.55 | Fibersrv | ok thanks |
| 18:14.08 | dhill | well |
| 18:14.19 | heh_v_water | just wondering about the 4 lines thing.. that would require 4 FXO ports? |
| 18:14.22 | dhill | you are looking at about 90Kbps per channel |
| 18:14.23 | *** part/#asterisk jpk (i=M40hgItf@mail.seceidos.de) |
| 18:14.35 | dhill | so if you have 5 lines.. 5 * 90.. = 450 |
| 18:14.40 | dhill | you need prolly 512Kbps upload |
| 18:14.48 | *** join/#asterisk xnon (i=xnon@200.82.222.118) |
| 18:14.54 | dhill | just for voip |
| 18:15.32 | *** join/#asterisk lukketto (n=lukketto@host68-190-dynamic.7-87-r.retail.telecomitalia.it) |
| 18:15.43 | Fibersrv | now the FXO ports, are they cards that are needed? If so, what is recommended? Sorry to ask these so many questions, I am new to Linux and Asterisk. |
| 18:15.54 | mw46_1 | Fibersrv: does the school want to switch the outside connection to VOIP or the inside? |
| 18:16.07 | Fibersrv | inside |
| 18:16.17 | Fibersrv | they have a T-1 |
| 18:16.41 | mw46_1 | I'm looking at a similar setup;-) |
| 18:17.31 | mw46_1 | but I'm new to all that as well;-) |
| 18:17.58 | heh_v_water | Fibersrv, Now I'm no expert by any means with this stuff so take my info with a grain of salt and be sure to verify.. there's two ways to get the FXO ports.. 1 is with a card, the other is with an FXO box |
| 18:17.58 | b11d | im converting a community college right now.. |
| 18:18.14 | LordBacon | 1.0.9 are the rpms I found for suse |
| 18:18.18 | b11d | I dont recommend the "box" |
| 18:18.19 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 18:18.22 | LordBacon | I'd like to figure out how to make it work, then upgrade |
| 18:18.23 | b11d | get a digium FXO/FXS card |
| 18:18.59 | Fibersrv | The number of cards will depend on the number of lines incoming into the building correct? |
| 18:19.01 | b11d | LordBacon.. thats not a good idea.. download & learn to use the newest versions from the source code |
| 18:19.13 | b11d | packages are shortcuts & shortcuts mean future failures |
| 18:19.32 | b11d | Fibersrv.. yes.. depending on the card & how many fxo & fxs ports it has |
| 18:19.49 | LordBacon | packages mean I don't have to watch the security lists |
| 18:19.53 | b11d | HAHAHAHAHA |
| 18:19.56 | mw46_1 | Fibersrv: digium has multiport cards but in the end yes. |
| 18:19.58 | b11d | yeah.. keep believing that |
| 18:20.06 | LordBacon | packages mean I don't have to spend all my time digging up dependencies |
| 18:20.11 | b11d | uhh.. yeah .. right |
| 18:20.11 | Fibersrv | what about extensions, they don't need those type of cards correct. It isd all handled internally right |
| 18:20.20 | b11d | whatever works for you Bacon |
| 18:20.31 | b11d | that is correct Fibersrv.. assuming the inside will be pure voip |
| 18:20.50 | b11d | if one of your 'extensions' is an analog POTS phone, then you need a card with an FXS port |
| 18:20.53 | LordBacon | if I wanted to spend all my time building stuff, I'd run gentoo |
| 18:21.05 | b11d | sure.. sure.. :P |
| 18:21.13 | Fibersrv | b11d it will be VOIP inside |
| 18:21.16 | heh_v_water | Fibersrvit depends, if you want to connect analog phones to the asterisk server then you need FXS ports, if not then all is good |
| 18:21.28 | b11d | cool.. then you should only need a card with FXO ports (unless you're going to do voice over a T1) |
| 18:22.20 | b11d | dont forget about non-voip analog devices that might need to use your system.. like elevator phones, and fire alarms. |
| 18:23.22 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:23.51 | dennisharrison | is /usr/src/zaptel/zaptel.c:226: warning: ‘fcstab’ defined but not used during make linux26 ok? :) |
| 18:23.52 | b11d | so..... |
| 18:24.01 | Qwell[] | dennisharrison: ignore it |
| 18:24.13 | dennisharrison | Qwell[] thats what I thought, thanks mate |
| 18:24.22 | b11d | whats new Qwell? |
| 18:24.26 | b11d | make it to the gangbang last week? |
| 18:25.24 | heh_v_water | not to repeat but.. I'm still looking for jiaxclient info if anyone has any experience with it |
| 18:26.24 | b11d | no experience with that here.. what is it? |
| 18:26.47 | heh_v_water | it is an open source web based IAX client ;) |
| 18:27.26 | heh_v_water | the binary works I just want to tweak it |
| 18:27.40 | heh_v_water | compiling has not been friendly at all |
| 18:28.02 | b11d | ahhh.. yeah that'll happen |
| 18:28.12 | *** join/#asterisk xnon_ (i=xnon@200.82.222.118) |
| 18:28.21 | Fibersrv | <b11d> what is the difference in changing it from VOIP on the outside or inside. They have a T-1 connection, and the last working system I seen running Asterisk had it changed from the outside. The connection changed at a channel bank provided by the phone company. |
| 18:28.41 | Fibersrv | which was a box that hung on the wall |
| 18:28.54 | b11d | well if you go voip on the outside, then you need a voip provider.. |
| 18:29.01 | b11d | and the T1 should be all data.. |
| 18:29.11 | *** join/#asterisk xnon_ (i=xnon@200.8.87.11) |
| 18:29.12 | mw46_1 | and your old phone numbers might get lost? |
| 18:29.14 | b11d | if you want to just do voip on the inside, then the T1 needs channels assigned for voice.. like a PRI |
| 18:29.16 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 18:29.25 | b11d | no, you can keep your old numbers.. at least here in the USA you can |
| 18:29.39 | mw46_1 | that's good to know;-) |
| 18:29.56 | b11d | thanks to LNP |
| 18:30.42 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 18:31.24 | mw46_1 | Does somebody know of a good list of Asterisk (++) consultants in tha central CA area? |
| 18:31.53 | blitzrage | mw46_1: there are consultant lists on the voip-info wiki |
| 18:32.09 | *** join/#asterisk lullabud (n=lullabud@12.24.42.67) |
| 18:32.17 | mw46_1 | yes, but nobody really close:-( |
| 18:32.43 | b11d | give enough cash, people will travel :) |
| 18:33.11 | mw46_1 | from a little charter school? |
| 18:33.11 | syzygyBSD | smart man.. |
| 18:33.58 | Mad|Cow | I'm using the ztdummy module (I have verifed its loaded in the kernel), can someone tell me how to use it as a |
| 18:34.02 | Mad|Cow | time channel? |
| 18:34.03 | *** join/#asterisk xnon__ (i=xnon@200.8.87.11) |
| 18:34.09 | syzygyBSD | mw46_1: why do you need a local consultant? if the hardware is connected correctly, which most tech people can do, then all you need is configuration, which can be done remotely |
| 18:34.13 | *** join/#asterisk xnon__ (i=xnon@200.8.87.11) |
| 18:34.16 | BSDTech | modeprobe zaptel |
| 18:34.26 | b11d | mw46_1.. just sit in here and talk for a week.. you'll learn it |
| 18:34.27 | BSDTech | if you have a card intsalled it willindit |
| 18:34.29 | b11d | thats what i've been doing |
| 18:34.55 | BSDTech | did you configure your card |
| 18:34.57 | mw46_1 | syzygyBSD: it's more a spec thing, right now. |
| 18:34.59 | Mad|Cow | bsdtech: the module is already loaded. Dont I have to do anything though to make asterisk recogizine it? |
| 18:35.04 | mw46_1 | How big, how much etc. |
| 18:35.05 | BSDTech | or you just trying to use ztdummy |
| 18:35.10 | syzygyBSD | I come and go depending on how much work I have.. speaking of that... I have to go... |
| 18:35.19 | Mad|Cow | I'm trying to use ztdummy |
| 18:35.32 | syzygyBSD | mw46_1: well specs can all be done remotely as well |
| 18:35.39 | BSDTech | ok did you setmodprobe to load ztdummy |
| 18:35.50 | Fibersrv | How does Asterisk know what channels have been assigned for voice by the T-1? If this in the configuration of Asterisk or the phone provider? |
| 18:35.52 | BSDTech | and did you edit the zaptel.conf |
| 18:36.03 | Mad|Cow | ztdummy is also loaded as a module |
| 18:36.15 | BSDTech | its how you configure the t1 on the system |
| 18:36.26 | Mad|Cow | I'm not sure what needs to be in the /etc/zaptel.conf though for the ztdummy driver |
| 18:36.27 | BSDTech | they if its loaded its runnign |
| 18:36.58 | dennisharrison | any tips for adding CRC-CCITT to debian 2.6 kernel ? |
| 18:37.15 | BSDTech | take lots of asprin |
| 18:37.30 | Mad|Cow | I'm getting warrnings from asterisk that I dont have a "timing interface" |
| 18:37.34 | *** join/#asterisk Agrizzi (n=Brandon@65.97.153.236) |
| 18:37.34 | BSDTech | and put a pillow on the wall where you will bang your head |
| 18:37.56 | BSDTech | brb fone |
| 18:38.02 | Mad|Cow | oh... I have done that too ;-) |
| 18:38.20 | Agrizzi | hello everyone |
| 18:38.24 | Agrizzi | good afternoon |
| 18:38.37 | mw46_1 | syzygyBSD: yes, in principle. But as far as I've seen, that school has an inherited net infrastructure which nobody knows any more:-( |
| 18:38.57 | mw46_1 | That's probably the first thing to be rediscovered. |
| 18:38.58 | Agrizzi | ? dose anyone know were the system recordings go after i set the up what folder i want to make a backup of the files |
| 18:39.02 | mw46_1 | :-( |
| 18:39.21 | syzygyBSD | :) as do most schools, I remember setting up 10baseT at my old high school |
| 18:40.15 | syzygyBSD | Agrizzi: are you looking for ones that are recorded, say with monitor or ones that can be played ? |
| 18:40.35 | jmesquita | Hello, yall! |
| 18:40.49 | syzygyBSD | <PROTECTED> |
| 18:40.51 | mw46_1 | OK, I'll listen in here from time to time and in the meantime will try to find out more about what's there so far. |
| 18:40.54 | jmesquita | I am running asterisk 1.2.13 on a linux 2.4.27-3-686 |
| 18:40.54 | Agrizzi | no i had my office girl make the auto attendent and now it is stored in the server as a wav file, but were at |
| 18:40.57 | Agrizzi | i want to copy |
| 18:41.00 | mw46_1 | Thanks for all your help;-) |
| 18:41.01 | Agrizzi | it to my harddrive |
| 18:41.06 | jmesquita | But I have about 400 threads running at the same time |
| 18:41.15 | jmesquita | Is that supposed to be correct behaviour? |
| 18:41.20 | syzygyBSD | Agrizzi: /var/lib/asterisk/sounds/ |
| 18:41.27 | Agrizzi | i checked there |
| 18:41.29 | Agrizzi | didn't see it |
| 18:41.40 | syzygyBSD | ls -lt|head |
| 18:41.57 | *** join/#asterisk MAxiMuS (n=manken__@88.241.106.252) |
| 18:42.33 | jmesquita | I am running asterisk 1.2.13 on a linux 2.4.27-3-686 and I have about 400 asterisk threads runnning at the same time. Is that supposed to be normal behaviour or do I have something weird on my system? |
| 18:45.30 | *** join/#asterisk henrique (n=henrique@201-1-133-64.dsl.telesp.net.br) |
| 18:51.20 | *** join/#asterisk sloth_ (n=josh@64.3.170.41.ptr.us.xo.net) |
| 18:52.49 | b11d | ouch |
| 18:52.52 | b11d | 400 threads? wow |
| 18:54.19 | *** join/#asterisk tomhoover (n=thoover@valgate.valcom.com) |
| 18:54.41 | tomhoover | register cofebrk |
| 18:55.42 | *** join/#asterisk tessier (n=treed@kernel-panic/sex-machines) |
| 18:59.22 | tomhoover | identify cofebrk |
| 18:59.58 | *** join/#asterisk ambriento (n=melcon@200.192.160.100) |
| 19:01.42 | *** part/#asterisk tomhoover (n=thoover@valgate.valcom.com) |
| 19:04.42 | wunderkin | ... |
| 19:13.14 | *** join/#asterisk alerios (n=alerios@69.79.145.100) |
| 19:13.35 | *** join/#asterisk tomhoover (n=kvirc@valgate.valcom.com) |
| 19:15.12 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 19:18.33 | *** join/#asterisk |tomhoover| (n=kvirc@valgate.valcom.com) |
| 19:19.28 | *** part/#asterisk |tomhoover| (n=kvirc@valgate.valcom.com) |
| 19:21.44 | dennisharrison | I get FATAL: Module zaptel not found. when I modprobe zaptel |
| 19:21.48 | *** join/#asterisk Robyn (n=Ebola@host86-136-86-71.range86-136.btcentralplus.com) |
| 19:21.59 | dennisharrison | followed the tut at http://www.voip-info.org/wiki/view/Asterisk+Zaptel+Installation |
| 19:22.09 | dennisharrison | any hints of where to go next ? :p |
| 19:23.00 | *** join/#asterisk _ebola (n=freenode@veryniceshop.com) |
| 19:23.24 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 19:24.35 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 19:25.18 | CunningPike | I'm fighting to get multiple registrations working on a Polycom with type=peer - does anyone have any suggestions? |
| 19:25.38 | sloth_ | Anybody working with t.38 and 1.4? |
| 19:25.59 | *** part/#asterisk knathraak (n=zach@151.196.142.242) |
| 19:26.01 | *** join/#asterisk phil_z (n=jiriki@74.132.237.142) |
| 19:26.02 | SplasPood | CunningPike: different peers, I assume? |
| 19:26.33 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 19:27.09 | CunningPike | SplasPood: Yes - the last one in sip.conf works, but the others use that peer for authentication instead of their own |
| 19:27.28 | CunningPike | username= has no effect...... |
| 19:27.36 | *** part/#asterisk phil_z (n=jiriki@74.132.237.142) |
| 19:27.36 | SplasPood | CunningPike: they're all named diff? |
| 19:27.43 | SplasPood | ~pb |
| 19:27.53 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 19:27.58 | CunningPike | SplasPood: Yes - I'll pastebin the peers and the polycom config |
| 19:28.01 | SplasPood | CunningPike: throw yer sip.conf t.. |
| 19:28.02 | SplasPood | :) |
| 19:28.14 | SplasPood | CunningPike: also, what firmware ya running on the phone |
| 19:29.11 | CunningPike | 1.6.7 |
| 19:29.51 | *** join/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
| 19:31.19 | *** join/#asterisk adorah (n=admin@87.68.172.77.cable.012.net.il) |
| 19:31.26 | dennisharrison | with modprobe zaptel I get FATAL: Module zaptel not found. I got no errors during install |
| 19:31.41 | *** join/#asterisk santiago (n=santiago@208.195.214.146) |
| 19:31.42 | dennisharrison | any ideas ? |
| 19:32.40 | CunningPike | SplasPood: http://pastebin.ca/261666 |
| 19:32.59 | CunningPike | dennisharrison: modprobe zaptel |
| 19:33.47 | *** join/#asterisk Agrizzi (n=Brandon@65.97.153.236) |
| 19:33.57 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 19:35.02 | CunningPike | SplasPood: It's something simple, I'm sure - we used to use type=friend and it worked fine, but I'm doing some work with hints in 1.4, and they need type=peer to work |
| 19:35.09 | [TK]D-Fender | CunningPike: You cannot do them in seperate tags. <reg> is meant to be done all in 1 tag |
| 19:35.50 | CunningPike | [TK]D-Fender: For peers only? It works for type=fried |
| 19:35.56 | CunningPike | s/fried/friend/ |
| 19:37.18 | CunningPike | [TK]D-Fender: I'll try it as one tag..... |
| 19:38.30 | [TK]D-Fender | CunningPike: You should always be using them as friend.... |
| 19:38.56 | CunningPike | [TK]D-Fender: Well, Polycom users will have trouble using hints in 1.4, then....... |
| 19:39.02 | dennisharrison | CunningPike FATAL: Module zaptel not found. |
| 19:39.14 | *** join/#asterisk hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 19:39.57 | SplasPood | CunningPike: sorry, I got distracted... what TK said is correct |
| 19:40.10 | SplasPood | CunningPike: how are hints affected? |
| 19:40.21 | *** join/#asterisk apardo (n=apardo@87.217.145.139) |
| 19:40.34 | CunningPike | dennisharrison: Do a 'find / -name zaptel.ko and see where it is located |
| 19:40.50 | SplasPood | CunningPike: with 1.2, at least, and type=friend my hints work as expected |
| 19:40.54 | SplasPood | CunningPike: haven't tried with 1.4 yet |
| 19:40.57 | [TK]D-Fender | CunningPike: Why is that? |
| 19:41.03 | CunningPike | SplasPood: They don't work unless type=peer and call-limit>0 |
| 19:41.14 | CunningPike | [TK]D-Fender: Don't know :) Ask oej |
| 19:41.28 | SplasPood | CunningPike: has this been your experience, or are you citing some source |
| 19:41.32 | CunningPike | SplasPood: Our hints work great on 1.2 as well |
| 19:41.56 | CunningPike | SplasPood: Experience - one of the first things we noticed on our test server was that hints stopped working |
| 19:42.04 | CunningPike | http://bugs.digium.com/view.php?id=8405 |
| 19:42.19 | dennisharrison | CunningPike /lib/modules/2.6.17/misc/zaptel.ko other then the src dir |
| 19:42.44 | SplasPood | CunningPike: hrm.. maybe it's a bug.. have you found mention of this issue anywhere? |
| 19:42.59 | CunningPike | dennisharrison: And 'uname -r' says? |
| 19:43.07 | CunningPike | SplasPood: Check out the Mantis link I posted |
| 19:43.20 | dennisharrison | 2.6.17-2-686 |
| 19:43.54 | dennisharrison | should I symlink 2.6.17 to 2.6.17-2-686 ? |
| 19:44.30 | SplasPood | dennisharrison: just mv em over |
| 19:44.37 | SplasPood | and do a depmod -a |
| 19:44.45 | dennisharrison | roger that |
| 19:46.46 | [TK]D-Fender | CunningPike: another reason I haven't touched 1.4 yet :) |
| 19:47.04 | CunningPike | [TK]D-Fender: Hmmm - this is going to pose a real problem........ |
| 19:47.15 | dennisharrison | that did it |
| 19:47.16 | dennisharrison | thanks |
| 19:49.32 | *** join/#asterisk AuPix (n=root@adsl-04-85.abel.net.uk) |
| 19:52.08 | CunningPike | [TK]D-Fender: OK - insecure=peer gets around that one...... |
| 19:52.16 | SplasPood | CunningPike: hrm, good to know... |
| 19:52.53 | *** join/#asterisk luca (n=luca@host68-190-dynamic.7-87-r.retail.telecomitalia.it) |
| 19:54.10 | *** join/#asterisk lukketto (n=lukketto@host68-190-dynamic.7-87-r.retail.telecomitalia.it) |
| 19:54.28 | *** part/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 20:03.30 | dennisharrison | ok zaptel is compiled configured installed and loaded |
| 20:03.49 | dennisharrison | now how do I check to see that my cards are working before I do the same with asterisk ? |
| 20:03.49 | *** join/#asterisk sloth (n=josh@64.3.170.41.ptr.us.xo.net) |
| 20:05.32 | *** part/#asterisk lukketto (n=lukketto@host68-190-dynamic.7-87-r.retail.telecomitalia.it) |
| 20:06.25 | *** join/#asterisk xnon (n=xnon@200.8.87.11) |
| 20:07.31 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 20:07.31 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:07.57 | dennisharrison | hmm |
| 20:07.58 | dennisharrison | awesome |
| 20:07.59 | docelm0 | Anyone in here good w/ SIP realtime? I have a couple questions that just odnt make any fricken sense.. |
| 20:08.03 | dennisharrison | looks like it picked up the first card |
| 20:08.04 | blitzrage | dennisharrison: zttool |
| 20:08.05 | dennisharrison | but not the second |
| 20:08.37 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 20:09.33 | dennisharrison | wooo ... zttool looks pretty messed up |
| 20:09.33 | dennisharrison | heh |
| 20:09.38 | Agrizzi | hello everyone |
| 20:09.44 | dennisharrison | hopefully its just my ssh client |
| 20:09.51 | Agrizzi | have a question dose anyone know of a web sip client |
| 20:09.53 | *** join/#asterisk saftsack (n=oliver@p54A7E20E.dip.t-dialin.net) |
| 20:09.56 | Agrizzi | so users can go to my website |
| 20:10.02 | *** join/#asterisk xlogik (n=Miranda@pawn.twbg.com) |
| 20:10.09 | Agrizzi | and login to a web sip phone |
| 20:10.18 | docelm0 | Agrizzi not opensource |
| 20:10.22 | blitzrage | dennisharrison: probably don't have ncurses-devel installed then |
| 20:10.42 | dennisharrison | thought I did |
| 20:10.47 | dennisharrison | let me check |
| 20:10.56 | Agrizzi | ok which ones are not open source |
| 20:10.58 | dennisharrison | but do I need to use zttool to config the cards ? |
| 20:11.10 | docelm0 | no |
| 20:11.12 | blitzrage | dennisharrison: no, its just a status monitor |
| 20:11.38 | docelm0 | There was an exhibitor @ astricon that has what your looking for.. its like 2000 USD |
| 20:11.38 | dennisharrison | yeah I have ncurses5-dev |
| 20:11.43 | dennisharrison | blitzrage thanks |
| 20:11.48 | Agrizzi | shit |
| 20:11.52 | Agrizzi | man |
| 20:12.02 | docelm0 | Agrizzi I can code you one if you want. |
| 20:12.12 | dennisharrison | well when I ztcfg -vv I get |
| 20:12.29 | docelm0 | Do you want an actual webphone or click2call application? |
| 20:12.38 | dennisharrison | ZT_CHANCONFIG failed on channel 2: No such device or address (6) |
| 20:12.38 | dennisharrison | at the end |
| 20:12.47 | docelm0 | hmm.. |
| 20:12.53 | docelm0 | realtime is being a pain in my ass |
| 20:13.22 | Agrizzi | hmmm i was thinking i want to maybe setup some type of web conference room system with a websoft phone |
| 20:13.26 | Agrizzi | what do you think |
| 20:13.37 | CunningPike | SplasPood: insecure=peer allows them to place calls, but the callerid is that of the last peer in sip.conf, not of the peer that is placing the call |
| 20:14.50 | heh_v_water | Agrizzi, I am currently trying to make an open source IAX client work take a look here http://www.hem.za.org/jiaxclient/ |
| 20:15.59 | *** join/#asterisk ESCulapio__ (n=ESCulapi@200.88.44.66) |
| 20:16.02 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 20:16.05 | ESCulapio__ | quien puede ayudarme con h323 |
| 20:16.06 | Agrizzi | wow looks great |
| 20:16.11 | ESCulapio__ | quien habla espanol |
| 20:16.12 | *** part/#asterisk alerios (n=alerios@69.79.145.100) |
| 20:16.12 | ESCulapio__ | ?? |
| 20:16.20 | Agrizzi | i am currently working for a comapny who has alot of offices |
| 20:16.22 | markit | anyone with dtfm recognition problem with recent 1.2svn? |
| 20:16.26 | Agrizzi | but i think they might close this office |
| 20:16.36 | Agrizzi | so i am trying to build a solutions business |
| 20:16.44 | ESCulapio__ | i have problem with oh323 |
| 20:16.54 | Agrizzi | which will allow the other offices to feed off of |
| 20:17.06 | heh_v_water | Agrizzi, if you make it work you should be proud.. i've been trying to compil emy own for days now... |
| 20:18.06 | Un1x | anyone around here using g729 encoding? |
| 20:19.03 | *** join/#asterisk mfroes (n=froes@200-162-218-81.corp.ajato.com.br) |
| 20:19.08 | mfroes | cant load zaptel module |
| 20:19.27 | *** join/#asterisk zefk (n=zefk@81.181.249.106) |
| 20:19.34 | robin_z | meep? |
| 20:19.39 | mfroes | Nov 29 17:18:06 WARNING[19967]: loader.c:554 load_modules: Loading module chan_zap.so failed |
| 20:19.47 | robin_z | so .. mISDN ... |
| 20:19.49 | *** join/#asterisk NormanASD (n=norman@206.135.58.98) |
| 20:20.05 | robin_z | why does it go tits up on me once a day? |
| 20:21.19 | Agrizzi | very ncie |
| 20:21.24 | Agrizzi | hmmm i will look at it |
| 20:21.28 | Agrizzi | some more |
| 20:21.32 | Agrizzi | and see what i can come up with |
| 20:22.50 | ESCulapio__ | estoy integrando ccm 3.3 h323 with asterisk |
| 20:22.52 | NormanASD | I've been looking, but cannot find, documentation on how Asterisk handles SIP NOTIFY messages. For example, Client 1 and Client 2 are connected via Asterisk (noreinvite.) If Client 1 sends a SIP NOTIFY, will Client 2 receive it or will Asterisk assume it's for it and drop it? |
| 20:23.11 | ESCulapio__ | quien puede ayudarme |
| 20:23.12 | ESCulapio__ | ? |
| 20:23.28 | ESCulapio__ | necesito orientacion |
| 20:23.33 | Agrizzi | ? on nat ports what are the correct ports needed to run asterisk behined my firewall |
| 20:26.49 | *** join/#asterisk raetin (n=thomas@rrcs-71-41-120-142.se.biz.rr.com) |
| 20:27.28 | *** join/#asterisk MattH (n=MattH@cloud2.chilitech.net) |
| 20:27.45 | blitzrage | Agrizzi: look at rtp.conf |
| 20:27.52 | *** join/#asterisk pjz (n=pj@zachs.place.org) |
| 20:27.54 | blitzrage | plus SIP signalling port |
| 20:28.10 | pjz | if I'm building a new asterisk box, should I go with a 2.4 or 2.6 kernel? |
| 20:28.14 | blitzrage | 2.6 |
| 20:28.20 | pjz | 2.6.? |
| 20:28.25 | pjz | er, 2.6.what ? |
| 20:28.25 | blitzrage | 2.6.newest |
| 20:28.29 | pjz | okay |
| 20:28.32 | pjz | thanks |
| 20:28.43 | blitzrage | and asterisk doesn't care really all that much |
| 20:28.50 | blitzrage | but support for 2.4 will go away at some point |
| 20:29.13 | pjz | there's 2.6 support for the latest zap modules? |
| 20:29.13 | blitzrage | and I say newest because 1) asterisk doesn't care and 2) it's probably got less security flaws |
| 20:29.17 | blitzrage | of course |
| 20:29.37 | pjz | cool |
| 20:30.39 | MattH | Hi... does anyone know.. if I want to use an MMS:// stream for musiconhold will mpg123 play it? |
| 20:30.48 | MattH | I'm trying .. but it just does silence |
| 20:30.52 | *** join/#asterisk dasuberd1vid (i=david@nat/digium/x-bf1b55433e0a0963) |
| 20:31.00 | *** join/#asterisk RoyK (n=roy@ti211310a080-6121.bb.online.no) |
| 20:31.13 | *** join/#asterisk dasuberdavid (n=edwardfa@199.227.185.35) |
| 20:31.48 | pjz | is there a debian repository for the latest asterisk? |
| 20:32.05 | hmmhesays | doubtful |
| 20:32.28 | docelm0 | Anyone know why realtime would not let someone to register when realtime is setup correctly and the client is right .. But when I check debug it shows that its pulling from the db then doesnt do anything |
| 20:32.59 | dennisharrison | ok guys, after some reading it looks like I am having irq issues or something with my two x100p's |
| 20:33.28 | Shido6 | yep |
| 20:34.27 | docelm0 | x100 eww.. Go pickup a TDM400P w/ 2 FXO |
| 20:34.35 | docelm0 | I have one I love it. |
| 20:35.22 | dennisharrison | well this is from x100p.com I know it sucks |
| 20:35.25 | dennisharrison | but its supposed to work |
| 20:35.37 | dennisharrison | this has two fxo's on it also |
| 20:36.34 | *** join/#asterisk AuPix (n=root@adsl-04-85.abel.net.uk) |
| 20:37.03 | *** join/#asterisk s1gny|wrk (n=s1gny@p5491703D.dip.t-dialin.net) |
| 20:37.50 | dennisharrison | oh ... |
| 20:37.51 | dennisharrison | nm |
| 20:37.56 | dennisharrison | I think its working |
| 20:38.04 | dennisharrison | each card is 1 channel and I have two channels working ? |
| 20:38.08 | *** part/#asterisk s1gny (n=s1gny@p5491703D.dip.t-dialin.net) |
| 20:38.09 | dennisharrison | shit, im confused |
| 20:39.45 | *** join/#asterisk topping_ (n=topping@207.47.6.182.static.nextweb.net) |
| 20:40.24 | dennisharrison | anyone here use x100p cards ? |
| 20:41.39 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 20:43.14 | BSDTech | its it a clone or areal x100p |
| 20:43.39 | dennisharrison | from x100p.com |
| 20:43.39 | BSDTech | I find 90% of them only work 20% of the time |
| 20:43.39 | dennisharrison | according to http://www.voip-info.org/tiki-index.php?page=X100P+clone |
| 20:43.42 | *** join/#asterisk mpreslic (n=FreePBX6@infraservtech.com) |
| 20:43.44 | dennisharrison | it should work ok |
| 20:43.54 | dennisharrison | hehe heya BSDTech was talking to you yesterday |
| 20:43.59 | BSDTech | should and do are 2 diff things |
| 20:44.05 | dennisharrison | thats true |
| 20:44.09 | dennisharrison | however I see the channels |
| 20:44.14 | dennisharrison | im just looking for some clarification |
| 20:44.38 | *** join/#asterisk GaVak (n=denniso@adsl-074-228-124-003.sip.sav.bellsouth.net) |
| 20:45.18 | GaVak | Any idea on what could cause a SIP to ZAP call to get the telco providors 'You must dial a 1 or 0 before this call...' message 2 out of 5 dial attempts? |
| 20:45.19 | *** join/#asterisk zefk (n=zefk@81.181.249.106) |
| 20:45.29 | BSDTech | ok |
| 20:45.38 | dennisharrison | http://pastebin.ca/261733 |
| 20:45.50 | dennisharrison | does that mean I am good to go on installing asterisk now? |
| 20:46.11 | dennisharrison | or do I need to clear up that ZT_CHANCONFIG failed on channel 2: No such device or address (6) error message with some sort of magic wand ? |
| 20:46.21 | *** join/#asterisk zapp-branigan (n=zapp-bra@81-202-140-115.user.ono.com) |
| 20:46.53 | dennisharrison | or rather since this is the same thing I get no matter how verbose I set ztcfg too is there somewhere else to go to get more information on the error? |
| 20:47.16 | sloth | T.38 anyone? |
| 20:47.35 | Shido6 | ZZzZz |
| 20:47.42 | Shido6 | dennis, whats up? |
| 20:48.24 | dennisharrison | two x100p's using latest zaptel from svn on debian, everything is configured and compiled according to the various tuts I found on voip-info.org |
| 20:48.43 | PupenoR | Any ideas how to get rid of this notices and warnings: http://paste.lisp.org/display/30955 This is an my (development) box and I want to only see notices and warnings of the module I am developing. |
| 20:48.51 | dennisharrison | ZT_CHANCONFIG failed on channel 2: No such device or address (6) is what I get when I ztcfg |
| 20:49.05 | dennisharrison | http://pastebin.ca/261733 is the output of dmesg and ztcfg |
| 20:49.07 | Shido6 | heh |
| 20:49.10 | Shido6 | ok |
| 20:49.32 | Shido6 | configure 1 |
| 20:49.42 | Shido6 | then rerun ztcfg |
| 20:49.52 | Shido6 | then try the other. |
| 20:49.57 | dennisharrison | you mean use only one card ? |
| 20:49.58 | dennisharrison | ahhh gotcha :P |
| 20:50.54 | *** join/#asterisk MIXX941 (n=mark@unaffiliated/mixx941) |
| 20:51.43 | dennisharrison | brb |
| 20:52.08 | *** join/#asterisk Katty (n=angela@hera.copi-rite.com) |
| 20:55.43 | ESCulapio__ | quien me puede ayudar |
| 20:55.53 | ESCulapio__ | quien habla espanol |
| 20:55.54 | pjz | nadie |
| 20:56.19 | synthetiq | anyone here use meetme? when i use the P value you and specifiy the password, it doesnt not prompt for the password...wtf? |
| 20:56.35 | ESCulapio__ | pjz, tengo un problemilla con gnugk oh323 |
| 20:57.00 | ESCulapio__ | como le digo al asterisk que me envie las llama por oh323 al call manager |
| 20:57.08 | ESCulapio__ | utilizando gnugk |
| 20:58.22 | *** join/#asterisk afrosheen (n=test@cpe-76-183-174-247.tx.res.rr.com) |
| 20:58.43 | blitzrage | synthetiq: are you creating a dynamic conference? |
| 20:59.13 | *** part/#asterisk zefk (n=zefk@81.181.249.106) |
| 20:59.28 | *** join/#asterisk poller_ (n=poller@poller.se) |
| 21:01.47 | synthetiq | no static |
| 21:02.49 | synthetiq | meetme(1234|P|2222) |
| 21:02.49 | *** part/#asterisk poller_ (n=poller@poller.se) |
| 21:02.49 | synthetiq | that should prompt the person even tho i specified the pass... |
| 21:02.58 | *** join/#asterisk [edwardfaulk] (n=edwardfa@199.227.185.35) |
| 21:06.01 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 21:06.38 | *** join/#asterisk insomnia41 (n=dasver96@c-69-254-230-163.hsd1.ks.comcast.net) |
| 21:12.08 | *** join/#asterisk jeffik (n=Jeff@208-41-192-106.client.dsl.net) |
| 21:12.34 | [edwardfaulk] | <PROTECTED> |
| 21:13.56 | *** join/#asterisk jaffar (n=jaffar@ppp85-140-58-226.pppoe.mtu-net.ru) |
| 21:16.46 | adorah | Hi has anyone here experienced with Astlinux on net4801? |
| 21:18.46 | *** part/#asterisk BSDTech (n=RNeese@adsl-69-230-170-5.dsl.irvnca.pacbell.net) |
| 21:22.28 | SplasPood | synthetiq: hrm yea.. according to 'show application' at least |
| 21:22.52 | *** join/#asterisk towel (n=phph@67.58.10.46) |
| 21:24.13 | *** part/#asterisk towel (n=phph@67.58.10.46) |
| 21:27.47 | *** join/#asterisk af_ (n=af@ip-156-221.sn2.eutelia.it) |
| 21:29.09 | *** join/#asterisk saftsack (n=oliver@pD9E06858.dip.t-dialin.net) |
| 21:29.30 | cloud9 | hey, anybody here having success with faxes using asterisk? |
| 21:29.48 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 21:32.39 | dennisharrison | anyone have a further troubleshooting step for the ZT_CHANCONFIG failed on channel 1: No such device or address (6) message ? |
| 21:32.42 | *** join/#asterisk uwe (n=uwe@213.6.14.163) |
| 21:32.51 | dennisharrison | I am now trying to only use one card |
| 21:36.31 | *** join/#asterisk bkw_ (n=brian@24.249.5.154) |
| 21:36.50 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 21:41.13 | *** join/#asterisk lorinc (n=ang@caracas-1213.adsl.interware.hu) |
| 21:42.06 | zoa | hey you little fagot |
| 21:42.17 | zoa | :p |
| 21:43.02 | *** join/#asterisk alamantia (i=alamanti@nat/digium/x-8f8980c0462a43c6) |
| 21:43.06 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:43.35 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 21:45.49 | afrosheen | dennisharrison: is the proper module loaded? |
| 21:47.55 | afrosheen | !seen [tkd]fender |
| 21:48.06 | afrosheen | uhmmm |
| 21:48.38 | file | he'll probably be home in a bit |
| 21:48.48 | *** part/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 21:48.57 | afrosheen | I just wanted to give him props for mentioning that Sangoma PRI card...it is perfect |
| 21:49.09 | afrosheen | we got our PRI rolled over yesterday and it rocks |
| 21:50.40 | afrosheen | previously we were using a sip trunk to a provider over the internet and nobody was happy..not a day went by that one of our employees wasn't bitching about it |
| 21:51.25 | file | the internet is a dangerous place for VoIP packets... |
| 21:53.37 | *** join/#asterisk oomph-work (n=jimmy@65.216.185.17) |
| 21:53.43 | pjz | iss 1.2.13 okay to use or is 1.4 so much better than I should go with it even though I'll have to build it since there's no debian packages for it? |
| 21:54.02 | afrosheen | file: it was the quick and dirty solution until we got a wad of cash to invest in this |
| 21:54.35 | afrosheen | btw is it always this dead in here these days? last year it was chaos |
| 21:54.43 | zoa | dunno what happened here |
| 21:54.47 | dennisharrison | afrosheen wcfxo is loaded |
| 21:54.54 | zoa | when this place had only 10 people |
| 21:54.55 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 21:55.00 | zoa | there was a lot of talking all the time |
| 21:55.07 | afrosheen | dennisharrison: check /dev and see if udev or whatever created the device when the module loaded |
| 21:55.30 | dennisharrison | zap right ? |
| 21:56.34 | *** join/#asterisk RoyK (n=roy@ti211310a080-14619.bb.online.no) |
| 21:57.03 | afrosheen | dennisharrison: what card is it |
| 21:58.19 | dennisharrison | x100p |
| 21:58.52 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 21:58.56 | dennisharrison | hrmm .. do I have to redo something with the module now that I took one card out ? |
| 21:59.23 | afrosheen | dude you're gonna have to outline the situation for me, I'm confused already |
| 22:00.27 | dennisharrison | ahh alright cool |
| 22:00.42 | dennisharrison | I have two x100p's in here with the latest zaptel from svn |
| 22:01.05 | Agrizzi | hey guys |
| 22:01.06 | Agrizzi | ? |
| 22:01.09 | dennisharrison | followed the tuts on voip-info.org about configuring compiling and loading zaptel |
| 22:01.16 | Agrizzi | dose anyone know how i can get the mac address of |
| 22:01.20 | Agrizzi | my system in putty |
| 22:01.20 | Agrizzi | ? |
| 22:01.24 | dennisharrison | http://pastebin.ca/261733 |
| 22:01.38 | dennisharrison | is the dmesg output and the ztcfg -vv from the system after that proccess |
| 22:01.51 | dennisharrison | I asked about the ZT_CHANCONFIG failed on channel 2: No such device or address (6) message |
| 22:01.59 | *** join/#asterisk sloth (n=josh@64.3.170.41.ptr.us.xo.net) |
| 22:02.16 | dennisharrison | and someone told me to take one of the cards out and set the zaptel.conf to have only one card instead of the two |
| 22:02.18 | dennisharrison | so I did that |
| 22:02.29 | dennisharrison | now when I check dmesg I get this wcfxo: probe of 0000:02:0c.0 failed with error -5 |
| 22:02.49 | dennisharrison | no matter with either card or whatever slot I put it in |
| 22:03.19 | afrosheen | ouch |
| 22:03.19 | dennisharrison | however with both cards in I get the successful fxo card found stuff that is shown at pastebin link |
| 22:03.23 | afrosheen | what does zttool tell you |
| 22:03.31 | dennisharrison | let me check now |
| 22:03.46 | dennisharrison | I only see ztdummy |
| 22:04.32 | dennisharrison | I just verified /etc/zaptel.conf and I have fxsks=1 in there |
| 22:04.38 | dennisharrison | instead of the fxsks=1-2 for both cards |
| 22:04.40 | *** join/#asterisk CunningPike_ (n=CunningP@204.239.12.189) |
| 22:04.48 | dennisharrison | I feel like I am lost in the woods :) |
| 22:05.20 | CunningPike_ | New name for Asterisk - Blair Witch Project |
| 22:05.31 | dennisharrison | lol |
| 22:05.32 | CunningPike_ | ;) |
| 22:05.43 | [TK]D-Fender | dennisharrison : That would be a problem.... |
| 22:05.48 | Agrizzi | how do i get the mac address for my system in putty |
| 22:05.58 | dennisharrison | [TK]D-Fender: haha I hear ya man |
| 22:06.07 | CunningPike_ | Agrizzi: ifconfig, usually |
| 22:06.08 | dennisharrison | [TK]D-Fender: any suggestions ? |
| 22:06.24 | *** join/#asterisk linuxtuxie (n=kku@168.50-241-81.adsl-dyn.isp.belgacom.be) |
| 22:06.34 | [TK]D-Fender | dennisharrison : "fxsks=1-2" is looking pretty good right now |
| 22:06.37 | Agrizzi | in putty |
| 22:06.40 | Agrizzi | not windows |
| 22:06.49 | dennisharrison | yeah isn't that funny? |
| 22:06.56 | [TK]D-Fender | Agrizzi : ipconfig |
| 22:06.56 | dennisharrison | I mean how does that make sense ? |
| 22:07.15 | dennisharrison | unless when I compiled the modules they probed the cards and somewhere there is a config to tell it where to look for what card |
| 22:07.25 | dennisharrison | is that possible with the zaptel driver ? |
| 22:07.28 | dennisharrison | im just guessing |
| 22:07.57 | Agrizzi | command not found |
| 22:08.27 | CunningPike_ | Agrizzi: ifconfig |
| 22:08.45 | dennisharrison | should I just put both cards back in or should I reconfigure something ? |
| 22:08.46 | infinity1 | how can i make a zap channel for dial out only? no matter what, the phones keep ringing when i dial into the channel |
| 22:09.00 | linuxtuxie | Hi all, can somebody help me out with setting up Asterisk as SIP client behind NAT? I browsed around a lot...but I still can not create a working setup. My current sip.conf can be read at http://pastebin.be/3986/ and my fw rules can be read at http://pastebin.be/3987/...any help would be greatly appreciated!! |
| 22:09.02 | CunningPike_ | [TK]D-Fender: Have you ever gotten a Polycom to register on a port other than 5060? |
| 22:09.58 | [TK]D-Fender | CunningPike : Not me personally, but once for a guy who had more than 1 behind NAT (remote to *) |
| 22:10.22 | [TK]D-Fender | dennisharrison : Pastebin your zaptel.conf, zapata.conf, and output of "cat /proc/interrupts" |
| 22:10.41 | CunningPike_ | [TK]D-Fender: OK - what I am trying to do is register more than one peer from the same phone. For Asterisk to 'see' them properly, they need to be on different ports.... |
| 22:10.44 | LordBacon | does asterisk come with the webui or is that a separate package? |
| 22:10.46 | *** part/#asterisk jeffik (n=Jeff@208-41-192-106.client.dsl.net) |
| 22:11.11 | afrosheen | there's the man |
| 22:11.17 | afrosheen | what's up [TK]D-Fender |
| 22:11.41 | [TK]D-Fender | afrosheen : not too much... |
| 22:12.00 | afrosheen | [TK]D-Fender: good call on the Sangoma card for our new PRI, it's perfect |
| 22:12.09 | [TK]D-Fender | afrosheen : Glad to hear. |
| 22:12.10 | afrosheen | everyone here is happier than pigs in shit |
| 22:12.13 | *** join/#asterisk bkw_ (n=brian@24.249.5.154) |
| 22:12.29 | afrosheen | and we have 200 DIDs at a dime each so everyone gets a direct number :) |
| 22:12.57 | afrosheen | haven't heard any echo yet either |
| 22:12.58 | [TK]D-Fender | linuxtuxie : More your "register" in ther below "nat=yes", and add "externrefresh" in there.... |
| 22:12.58 | *** join/#asterisk System_Hoax (n=System_H@pD9E6ECC7.dip.t-dialin.net) |
| 22:13.00 | *** part/#asterisk System_Hoax (n=System_H@pD9E6ECC7.dip.t-dialin.net) |
| 22:13.14 | LordBacon | ok, I finally got asterisk 1.2.13 installed |
| 22:13.40 | afrosheen | mmmm..bacon |
| 22:15.31 | brian | is there anyway to create a menu using AGI for a MeetMe conference |
| 22:15.42 | *** join/#asterisk saftsack (n=saftsack@pD9E06858.dip.t-dialin.net) |
| 22:16.43 | *** part/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 22:17.18 | zapp-branigan | hi, there is another way to route the iax2 chanels to the free chanels without ChanIsAvail ? |
| 22:17.19 | linuxtuxie | [TK]D-Fender: done that, ran sip reload...and If I place a call, I see with sip debug enabled "Retransmitting #1->6 (NAT) to 81.241.50.168:5060:" where 81.241.50.168 is my correct extenal ip |
| 22:18.08 | *** join/#asterisk remmo (n=chatzill@smack.isp.net.au) |
| 22:18.14 | [TK]D-Fender | CunningPikenot in my experience. My * box is my NAT as well, and I have an IP 301 here that reg's to the local IP of my server, and the other reg out to a remote client. |
| 22:18.21 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 22:19.24 | LordBacon | I guess now I want freepbx |
| 22:19.45 | *** join/#asterisk aao_pwner (i=asd@c-24-21-91-140.hsd1.mn.comcast.net) |
| 22:19.52 | dennisharrison | [TK]D-Fender: here it is http://pastebin.ca/261844 |
| 22:20.26 | infinity1 | my zap channel 2 is set to a context with nothing in it an when someone dials in, the phone rings. is there a ghost in my box? |
| 22:20.32 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 22:20.45 | nextime | hi, anyone use jiaxclient here? |
| 22:21.05 | [TK]D-Fender | dennisharrison : Do yourself and those helping you a big favour and trash everything thats commented out in those 2 files.... |
| 22:21.42 | [TK]D-Fender | dennisharrison : Second, I don't see the modules loaded for your cards (of which your config only accounted for 1. |
| 22:21.46 | dennisharrison | [TK]D-Fender : sure thing at the top of the pastebin I have the only options that are uncmmmented |
| 22:22.10 | [TK]D-Fender | dennisharrison : Do "modprobe wcfxo" and repastebin "cat /proc/interrupts" |
| 22:22.30 | brian | how do I make my fastagi application be able to broadcast a menu to the meetme conference |
| 22:22.45 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 22:22.57 | dennisharrison | yeah my current config only has one in it |
| 22:23.00 | [TK]D-Fender | dennisharrison : Please redo them all without any comments.... |
| 22:23.08 | dennisharrison | alright |
| 22:23.13 | dennisharrison | what should be in the conf file ? |
| 22:23.17 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 22:23.40 | [TK]D-Fender | dennisharrison : Remove what shouldn't and we'll correct the rest after. |
| 22:23.47 | dennisharrison | just fxsks=1 and loadzone right ? |
| 22:23.49 | dennisharrison | ok sure thing |
| 22:24.29 | *** join/#asterisk brut- (n=brutatl@1022host168.starwoodbroadband.com) |
| 22:26.03 | afrosheen | dennisharrison: btw which OS are you running |
| 22:26.03 | Un1x | where is the documentation |
| 22:26.06 | Un1x | on how to compile asterisk |
| 22:26.09 | Un1x | with g729 support |
| 22:26.25 | *** part/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 22:26.52 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 22:28.15 | infinity1 | how can i create a channel for only dialing out and not for receiving calls when its part of a group? |
| 22:28.26 | linuxtuxie | In http://pastebin.be/3988/ you can find back the attempt of making a call with the suggested change in the sip.conf. The retransmit part is repeated 6 times |
| 22:30.10 | linuxtuxie | sip show registry nicely shows: sip.voipstunt.com:5060 linuxtuxie 105 Registered |
| 22:30.10 | dennisharrison | afrosheen : debian |
| 22:30.24 | [TK]D-Fender | linuxtuxie : please pastebin your peer entry, and the dialplan calling it |
| 22:31.19 | DrAk0SX | what's this? http://pastie.caboo.se/24939 |
| 22:32.29 | dennisharrison | [TK]D-Fender : Hopefully this is easier to read :) http://pastebin.ca/261863 |
| 22:32.45 | [TK]D-Fender | DrAk0SX : Running TDM400 for FXS? |
| 22:32.49 | dennisharrison | just look for -- to tell when a new file starts |
| 22:33.05 | *** join/#asterisk entersecurity (n=cblonski@h-67-100-250-106.lsanca54.covad.net) |
| 22:33.45 | [TK]D-Fender | dennisharrison : Did you compile Zaptel yourself or are you using a package? |
| 22:33.52 | Un1x | !g729 |
| 22:33.56 | Un1x | ~g729 |
| 22:33.58 | jbot | from memory, g729 is It was in November 1995 that the G.729 standard, also referred to as CS-ACELP was adopted by the ITU, a United Nations organization. Similar, quality-wise, to 32 kbps ADPCM, G.729 offers toll quality speech. Furthermore, being only an 8 kbps codec, G.729 offers opportunities for significant increases in bandwidth utilization to existing ... |
| 22:34.05 | dennisharrison | [TK]D-Fender : I compiled it myself |
| 22:34.11 | dennisharrison | from svn today |
| 22:34.21 | Un1x | well |
| 22:34.26 | Un1x | where is the god damn documentation :P |
| 22:34.28 | linuxtuxie | [TK]D-Fender: My dialplan can be found at http://pastebin.be/3989/ |
| 22:34.37 | [TK]D-Fender | Un1x : http://www.digium.com/en/supportcenter/documentation/viewdocs/G729 |
| 22:34.39 | brian | how do I make my fastagi application be able to broadcast a menu to the meetme conference |
| 22:34.47 | dennisharrison | followed the tut at http://www.voip-info.org/wiki/index.php?page=Asterisk+Linux+Debian and also |
| 22:34.48 | dennisharrison | http://www.voip-info.org/wiki/view/Asterisk+Zaptel+Installation |
| 22:34.52 | brian | I want to listen to the meetme conference DTMF |
| 22:35.07 | [TK]D-Fender | dennisharrison : Ok, well the lack of modules listed in /proc/interrupts is very wrong.... |
| 22:35.21 | brian | On my FastAGI script |
| 22:35.21 | dennisharrison | [TK]D-Fender I hear ya |
| 22:35.23 | brian | Is it possible? |
| 22:35.45 | linuxtuxie | [TK]D-Fender: the peer entry (in fact it is friend...but that should not be a problem) can be read at http://pastebin.be/3986/ |
| 22:35.48 | afrosheen | dennisharrison: what kernel series is it, 2.4 or 2.6 and is udev enabled |
| 22:35.59 | dennisharrison | 2.6.17-2-686 |
| 22:36.17 | dennisharrison | I make linux26'ed and make install-udev |
| 22:36.37 | dennisharrison | modules load on system start via /etc/modules |
| 22:36.43 | afrosheen | ok so you did read the install docs :) |
| 22:36.57 | dennisharrison | yes |
| 22:37.04 | afrosheen | did you include the wait post-zaptel module install? |
| 22:37.04 | [TK]D-Fender | linuxtuxie : Looks largely legit, though I'd change your dials to look like : # |
| 22:37.04 | [TK]D-Fender | xten => _013XXXXXX,1,Dial(SIP/0032${EXTEN:1}@voipstunt,60) |
| 22:37.04 | [TK]D-Fender | # |
| 22:37.08 | dennisharrison | for README.linux26 ? |
| 22:37.11 | afrosheen | yeah |
| 22:37.14 | DrAk0SX | [TK]D-Fender, no, sipuras |
| 22:37.19 | dennisharrison | afrosheen yep |
| 22:37.23 | [TK]D-Fender | exten => _013XXXXXX,1,Dial(SIP/voipstunt/0032${EXTEN:1},60) |
| 22:37.32 | [TK]D-Fender | dang pase caught 2 EOL's |
| 22:37.46 | tm | wer da der deutsch spricht? ;) |
| 22:37.51 | [TK]D-Fender | DrAk0SX : Ok, thats wierd then.... |
| 22:38.12 | afrosheen | du lieb micht night ich lieb dicht nicht, ah hah |
| 22:38.23 | mercestes | Any clue if Asterisk can interface with Microsoft Exchange Server 2007. Particularly the Voicemail servicing capability? |
| 22:38.23 | afrosheen | oops |
| 22:38.28 | dennisharrison | and the only thing I didn't do was enable CRC-CCITT because #debian guys said its already there |
| 22:38.30 | tm | afrosheen: *lach* |
| 22:38.32 | [TK]D-Fender | linuxtuxie : Though DO move the register to jsut above your first context entry |
| 22:38.36 | afrosheen | sinus medicine makes me type like a moron |
| 22:39.00 | mercestes | afrosheen: urgen durgen vergen bergen nurgen durgen? |
| 22:39.23 | entersecurity | Hello, I am looking to set up an Asterisk system in an internal "help desk" type functionality such that users on the lan call into a queue that is answered at one of 8 PC workstations. I have general questions about the interface to such a system. |
| 22:40.31 | mercestes | entersecurity: That seems to be a pretty straight forward setup well covered in the wiki. What are your general questions? |
| 22:40.31 | Un1x | w00t |
| 22:40.32 | Un1x | Asterisk supoports 64 bit |
| 22:40.38 | Un1x | or well atleast the encoding does |
| 22:40.51 | tm | Spricht hier wer deutsch? |
| 22:41.32 | linuxtuxie | <PROTECTED> |
| 22:41.40 | mercestes | tm: nein. Spricht hier wer leiderhausen. |
| 22:41.58 | tm | mercestes: Ja ist ja oK! Sorry. |
| 22:42.00 | awannabe | is there any reason why some voicemails would get cut off, like a user leaves a voicemail and when the * user goes to check it it just says the callerid and no message |
| 22:42.21 | [TK]D-Fender | linuxtuxie : hrm |
| 22:42.30 | entersecurity | I am a little unsure of the system architecture at the call center as I there are probably many ways to do it, I am looking for the most common or most functional, i.e. do most people use a physical phone with headset at each PC workstaion and answer calls via some sort of web interface on the PC or are softphones more appropriate |
| 22:43.04 | Katty | mercestes: who are you? |
| 22:43.04 | [TK]D-Fender | entersecurity : Depends on the admins style. Personally mine use hardphones with headsets. |
| 22:43.11 | afrosheen | tm: english only buddy :( |
| 22:43.25 | tm | afrosheen: ok :) |
| 22:43.46 | *** join/#asterisk sloth_ (n=josh@64.3.170.41.ptr.us.xo.net) |
| 22:43.52 | afrosheen | awannabe: have you updated the server lately |
| 22:43.54 | entersecurity | but the calls in the queue can be answered/transfered via a web interface on the PC, or are you stuck pressing buttons on the hardphone? |
| 22:44.12 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 22:44.18 | mercestes | Katty: some random internet man |
| 22:44.30 | Katty | mercestes: i see. |
| 22:44.41 | awannabe | afrosheen: nope, default install of 1.2.13 |
| 22:44.54 | afrosheen | awannabe: check permissions on the voicemail directory |
| 22:45.04 | awannabe | when I send a call to voicemail I do, VoiceMail(exten) is that wrong? |
| 22:45.11 | afrosheen | ]/var/spool/asterisk/whatever |
| 22:45.14 | mercestes | Katty: I talk to you all the time..:/ |
| 22:45.22 | [TK]D-Fender | linuxtuxie : What router are you behind, and what do you have forwarded? |
| 22:45.29 | Katty | mercestes: :< |
| 22:45.34 | afrosheen | awannabe: I think that's correct |
| 22:45.35 | Katty | mercestes: whooooooooo are you :P |
| 22:45.47 | [TK]D-Fender | entersecurity : We just use the phone like a phone. As you should. |
| 22:46.08 | linuxtuxie | [TK]D-Fender: my router is a debian box running on soekris hardware...making use of iptables...which can be viewed at http://pastebin.be/3987/ |
| 22:46.25 | afrosheen | [TK]D-Fender: polycom is about to release the 650 with a backlit screen..*drool* |
| 22:46.27 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 22:46.27 | mercestes | entersecurity: Via Operator Flash Panel it is possible to make transfers via a web interface, and it may work very well for the implementation you suggest, but, I can't advocate it openly. |
| 22:46.38 | mercestes | entersecurity: However, D-Fender is right as usual, hardphones are the way to go. |
| 22:46.43 | linuxtuxie | the forwared rtp ports are also defined in rtp.conf on my asterisk box (8000-2000) |
| 22:46.56 | afrosheen | softphones are fun to test with but you shouldn't run a business on them |
| 22:46.59 | mercestes | Katty: I'm a * guy. |
| 22:47.00 | linuxtuxie | 20000 I mean :) |
| 22:47.04 | entersecurity | mercestes: it's a bit more complicated as we are integrating "intercom" type fucntionality to the call center with an IP video system |
| 22:47.09 | mercestes | Katty: I'm always Mercestes.... |
| 22:47.23 | Katty | mercestes: mew?! |
| 22:48.13 | [TK]D-Fender | afrosheen : Overpriced. not on my "recommend" list |
| 22:48.37 | afrosheen | [TK]D-Fender: well it hasn't even been released yet, all of our phones are 501's or 601's (a handful) |
| 22:48.42 | [TK]D-Fender | entersecurity : Complicated how? |
| 22:48.48 | mercestes | entersecurity: Hrm. Might I suggest the Polycom suite of video products then? |
| 22:49.15 | entersecurity | mercestes: basically we'd use some IP phone and do all call answering from there then, and they wouldn't use any web interface aside from maybe a view of calls in the queue |
| 22:49.21 | pjz | why am I getting ZT_SPANCONFIG failed on span 0: Invalid argument (22) when I do ztcfg? |
| 22:49.35 | pjz | dmesg says it found the wildcart TE110P |
| 22:49.47 | afrosheen | pjz: what does ztcfg -vvv and zttool tell you |
| 22:50.04 | [TK]D-Fender | entersecurity : Downside with softphones is all your eggs are in one basket. Their PC crashes and they lose everything. Also you are wired closer to the box, and have to deal witht he softphones kludgy interface and the hassle of task-switching to deal with it. |
| 22:50.29 | entersecurity | the cameras are behind ADA complient 3"x3" call stations in parking garages with a single call button |
| 22:50.35 | mercestes | entersecurity: Do a web search for operator flash panel. Contact polycom for VoIP video conferencing. |
| 22:50.53 | pjz | afrosheen: ztcfg -vvv shows that it's configuring 13 channels and even says '13 channels configured.' but then it says 'ZT_SPANCONFIG failed on span 0: Invalid argument (22)' |
| 22:51.24 | entersecurity | no video conference, think more of one way streaming video and image capture to make sure the person pressing the button isn't a vandal / robbing someone |
| 22:51.59 | [TK]D-Fender | entersecurity : So you are doing 1-to-1 videoconfereced calls into a processing queue? |
| 22:52.00 | mercestes | entersecurity: Then it needn't be SIP driven video and you can use any of the other network video systems available. |
| 22:52.12 | entersecurity | the phone portion is really basic i think, just a queue that can be picked up by one of 8 people and forwarded out of the system on a time out |
| 22:52.32 | mercestes | entersecurity: Yea, pretty simple. I already have the code for that in my head. |
| 22:52.35 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 22:52.48 | entersecurity | correct, we have a good IP video soulution in place and would like to integrate the intercom functionality, but it sounds like just using the phones as a hard phone alone would be the most appropriate |
| 22:53.08 | entersecurity | no desktop integration required |
| 22:53.12 | Katty | mercestes: you do not parse. |
| 22:53.20 | [TK]D-Fender | entersecurity : Then a SIP video-softphone like eyebeam might just be the thing for you. |
| 22:53.28 | [TK]D-Fender | entersecurity : Go buy a copy for testing. |
| 22:53.40 | [TK]D-Fender | entersecurity : www.xten.com |
| 22:54.00 | mercestes | entersecurity: Depending on the intercom it may plug into an fxs port. |
| 22:55.12 | mercestes | Katty: I do not parse? :( aww... |
| 22:55.12 | entersecurity | yeah, vikingelectronics looks like they have a good ADA complient package we can plug into an fxs port |
| 22:55.30 | dennisharrison | [TK]D-Fender : Looks like when I put both cards in there it only sees one |
| 22:55.40 | entersecurity | D-Fender: thanks for the link I will check them out |
| 22:55.56 | dennisharrison | [TK]D-Fender : and with one card in there it sees none but it doesnt matter what card I use in what pci slot ... weird ? |
| 22:56.05 | awannabe | I think customer is on crack, when I leave voicemails they work great! |
| 22:56.31 | pjz | afrosheen: ztcfg -vvv shows that it's configuring 13 channels and even says '13 channels configured.' but then it says 'ZT_SPANCONFIG failed on span 0: Invalid argument (22)' Any ideas? |
| 22:56.41 | syzygyBSD | ahh, the problem of tech support, well it didn't work fine 5 minutes ago |
| 22:56.50 | afrosheen | pjz: is that wildcard TE110P the only card in that system? |
| 22:56.53 | [TK]D-Fender | dennisharrison : Yeah, not sure what to make of it.... |
| 22:57.07 | entersecurity | there are other cameras in the system and they will need to be streaming up to 12 cameras on the desktop at one time. Our video IP solution is very robust to handle all of those requirements, we are more concerned with the intercom (PBX) as we are unfamiliar with Asterisk and others |
| 22:57.09 | CunningPike_ | [TK]D-Fender: Polycoms can't specify a local port for each peer registration like an SPA-3000 can, so Polycom users are going to have problems with multi-peer phones when type=friend/user goes away |
| 22:57.10 | dennisharrison | [TK]D-Fender : sweet! I always get the best problems :) |
| 22:57.11 | [TK]D-Fender | dennisharrison : What version are you trying to run? |
| 22:57.22 | pjz | afrosheen: no, I think ther's one of the 4-pots cards in there, but it's not configured |
| 22:57.24 | dennisharrison | [TK]D-Fender : of zaptel ? svn from today |
| 22:57.33 | pjz | afrosheen: and it's not used so it doesn't need to be configured I think ? |
| 22:57.38 | dennisharrison | [TK]D-Fender : asterisk isn't on the system yet |
| 22:57.49 | afrosheen | pjz: does zttool show it as being configured or no |
| 22:57.59 | [TK]D-Fender | CunningPike : Depends on what * does to compensate. Keeping in mind Digium & Polycom are getting "tight" I wouldn't bet on these problems lasting |
| 22:58.16 | [TK]D-Fender | dennisharrison : Skip SVN, and get the 1.2 series FTP release |
| 22:58.21 | Shaun2222 | is there a default timout for dial()? |
| 22:58.22 | pjz | afrosheen: no zttool on this box for some reasn |
| 22:58.27 | CunningPike_ | [TK]D-Fender: Sure hope not....... :) |
| 22:58.29 | pjz | afrosheen: ztcfg is there tho |
| 22:58.36 | dennisharrison | pjz : you have to make it |
| 22:58.37 | afrosheen | pjz: you probably need to build it, it's part of the zaptel stuff |
| 22:58.54 | CunningPike_ | [TK]D-Fender: Right now, you can't have multi-line phones with BLF |
| 22:59.08 | pjz | afrosheen: ah, okay, found it |
| 22:59.10 | dennisharrison | [TK]D-Fender : ok ... will I be able to just overwrite everything here with that or should I go through and remove everything zaptel ? |
| 22:59.17 | CunningPike_ | [TK]D-Fender: Without having separate peer and user entries for each line |
| 22:59.25 | mercestes | Can anyone give me a clue/hint on how voicemail integration will work with Microsoft Exchange 2007? :/ |
| 22:59.26 | pjz | afrosheen: it doesn't list the wctdm card, just the wcte110p card |
| 22:59.30 | linuxtuxie | on a regular basis I see the following output in my asterisk console with sip debug enabled -> http://pastebin.be/3990/ ...I think it is normal output..but I am not 100% sure |
| 22:59.52 | afrosheen | pjz: that's good then but if the tdm card isn't being used, yank it so it's not using an IRQ |
| 22:59.59 | [TK]D-Fender | CunningPike : You mean right now with * 1.4 BETA.... |
| 23:00.16 | dennisharrison | OHH! no errors :) |
| 23:00.21 | [TK]D-Fender | CunningPike : Run BETA in production at your own risk :) |
| 23:00.35 | *** join/#asterisk sloth (n=josh@64.3.170.41.ptr.us.xo.net) |
| 23:00.37 | CunningPike_ | [TK]D-Fender: Correct - we're not in production - we're testing in our lab |
| 23:00.55 | afrosheen | well I'm off to play Resistance on my ps3..bbl |
| 23:01.00 | dennisharrison | ok ... so now I have a channel working with no errors because ztcfg -vv returns and says evertyhing is fine |
| 23:01.01 | blitzrage | ps3? ugh |
| 23:01.14 | dennisharrison | is it time to install asterisk ? |
| 23:01.20 | dennisharrison | or is there more testing to be done here first? |
| 23:01.44 | afrosheen | dennisharrison: go for it |
| 23:01.48 | dennisharrison | lol ;p |
| 23:01.53 | dennisharrison | what version ? |
| 23:01.56 | afrosheen | reboot once to make sure everything is kosher |
| 23:02.02 | afrosheen | we use 1.2.11 in production |
| 23:02.04 | dennisharrison | sure thing |
| 23:02.10 | dennisharrison | ok cool |
| 23:02.14 | afrosheen | bbl |
| 23:02.16 | dennisharrison | thanks |
| 23:02.19 | [TK]D-Fender | dennisharrison :frankly you could have it all installed already as it is. Didn't have to do zaptel 100% first you know... |
| 23:02.36 | *** join/#asterisk _DAW (n=_DAW@adsl-156-78-145.msy.bellsouth.net) |
| 23:03.00 | entersecurity | thanks all for your help, I think I am pointed in a better direction now to use hardphones and not worry about the visual integration of the phone interface |
| 23:03.14 | dennisharrison | [TK]D-Fender : I know, i just move slower then others ;p |
| 23:04.00 | *** part/#asterisk entersecurity (n=cblonski@h-67-100-250-106.lsanca54.covad.net) |
| 23:04.41 | *** join/#asterisk nalioth (i=nalioth@freenode/staff/ubuntu.member.nalioth) |
| 23:05.09 | _DAW | Hey everyone. I all of a sudden have started receiving warnings from asterisk saying "Ring/Off-hook in strange state 6 on channel 5". Nothing has changed. The channels are T1 kewlstarts through a channelbank (FXO) to the pstn. Any suggestions? |
| 23:05.47 | linuxtuxie | Does anybody still have some pointers for my "Asterisk as SIP client behind NAT" problem ? |
| 23:06.17 | awannabe | does anyone have the snoms setup so when you dial a 7/10 digit number it just auto dial the numbers? |
| 23:06.35 | *** join/#asterisk oomph-work (n=jimmy@65.216.185.17) |
| 23:06.59 | [TK]D-Fender | linuxtuxie : I might suspect your firewall rules... double check them. |
| 23:07.55 | pjz | afrosheen: okay, shtudown the machine, yanked the card, turned machine back on, still no go - same thing |
| 23:08.08 | linuxtuxie | [TK]D-Fender: Am I correct if I say that udp port 5060 & rtp port range 8000->20000 needs to be forwarded to my asterisk box ? |
| 23:08.09 | pjz | afrosheen: ZT_SPANCONFIG failed on span 0: Invalid argument (22) |
| 23:08.36 | pjz | span=0,1,0,esf,b8zs bchan=1-12 dchan=24 |
| 23:09.19 | [TK]D-Fender | linuxtuxie : Yup. Just make sure your syntax is right. |
| 23:09.40 | linuxtuxie | ok, let me doublecheck |
| 23:09.50 | [TK]D-Fender | pjz : Than should be span=1,1,0,esf,b8zs |
| 23:11.05 | [TK]D-Fender | rm -q hads |
| 23:11.14 | [TK]D-Fender | :O |
| 23:11.24 | hads | :S |
| 23:11.41 | *** join/#asterisk sloth_ (n=josh@64.3.170.41.ptr.us.xo.net) |
| 23:12.14 | Shaun2222 | is there a default timeout for dial()? |
| 23:12.35 | [TK]D-Fender | Shaun2222 : Yeah "never" |
| 23:12.50 | Shaun2222 | ok good.. |
| 23:13.03 | hads | show application Dial will tell you what you want to know |
| 23:13.46 | Shaun2222 | so if i wanted to leave it unchanged i would have my dial look like dial(local/ext||mgM(macro)) |
| 23:14.18 | [TK]D-Fender | Shaun2222 : Correct. That will dial until its answered, or Hell Freezes Over TM |
| 23:14.45 | linuxtuxie | <PROTECTED> |
| 23:15.05 | cloud9 | anybody here setup the real time architecture with mysql? |
| 23:15.21 | mercestes | cloud9: I have...ask away |
| 23:15.21 | Shaun2222 | cloud9: yes. |
| 23:15.25 | cloud9 | cool |
| 23:15.28 | cloud9 | so, .. |
| 23:15.58 | cloud9 | when I make the addons, does it add the res_mysql.conf to /etc/asterisk? |
| 23:15.58 | cloud9 | I don't seem to have the file |
| 23:16.13 | linuxtuxie | [TK]D-Fender: the output from iptables -L can be viewed at http://pastebin.be/3991/ |
| 23:16.46 | mercestes | CLoud9: In the src directory for asterisk-addons somewhere there are "examples" and res_mysql.conf is among them. |
| 23:17.00 | cloud9 | locate says I have it in /usr/src/asterisk/asterisk-addons-1.2.5/configs/res_mysql.conf.sample. |
| 23:17.06 | mercestes | cloud9: could do an updatedb && locate res_mysql.conf but it should be in the source somewhere. |
| 23:17.10 | mercestes | cloud9: That's it then. |
| 23:17.12 | Shaun2222 | cloud9: i think you need to do make install samples |
| 23:17.21 | Shaun2222 | or make samples cant remember.. |
| 23:17.24 | cloud9 | I just did, but I'll do it again. |
| 23:17.28 | mercestes | Shuan2222: not with mysql real time. |
| 23:17.29 | mercestes | it won't work. |
| 23:17.38 | mercestes | cloud9: Copy that file to /etc/asterisk. |
| 23:17.42 | [TK]D-Fender | linuxtuxie : Ok, I'm all out of ideas at this point... |
| 23:17.53 | cloud9 | mercestes : thanks bud |
| 23:17.55 | Shaun2222 | cloud9: my res_mysql.conf has a sample header on it still |
| 23:17.59 | linuxtuxie | <PROTECTED> |
| 23:18.15 | syzygyBSD | linuxtuxie: whats the problem, I'm lazy and don't want to scroll up |
| 23:18.42 | linuxtuxie | syzygyBSD: Asterisk as SIP client behind NAT |
| 23:18.48 | syzygyBSD | I just finished my 60 line routing table the other week.. so I am a bit fresh |
| 23:19.06 | syzygyBSD | nat is done on linux right? |
| 23:19.15 | mercestes | cloud9: anytime |
| 23:19.25 | cloud9 | mercestes : have you found anything you didn't like about RTA? |
| 23:19.27 | cloud9 | with mysql? |
| 23:19.32 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 23:19.39 | cloud9 | any features not present that you expected? |
| 23:19.41 | linuxtuxie | syzygyBSD: yip, I have debian asterisk box behind a debian firewall running on soekris hardware |
| 23:19.44 | Shaun2222 | cloud9: the documentation sucks.. |
| 23:20.00 | syzygyBSD | you have a pastebin of all your iptables? |
| 23:20.02 | mercestes | Shaun2222: I found the documentation to be pretty ok. |
| 23:20.04 | Shaun2222 | cloud9: i found my self changing the database schemes like crazy trying to find the ones that where right. |
| 23:20.17 | linuxtuxie | syzygyBSD: the fw is created with iptables -> http://pastebin.be/3987/ |
| 23:20.25 | mercestes | cloud9: there are nuances...however, there is far more available in realtime than I think is beneficial. |
| 23:20.36 | mercestes | extensions.conf in mysql basically turns mysql into an overglorified text editor. |
| 23:20.42 | linuxtuxie | syzygyBSD: http://pastebin.be/3991/ show the iptables -L output |
| 23:20.58 | pjz | when asterisk complains 'unknown directive 'exten' at line 523 of extensions.conf', how accurate is that line number? |
| 23:21.06 | linuxtuxie | syzygyBSD: o this is a minimal fw rule set |
| 23:21.07 | mercestes | Sip.conf in mysql is very handy for scripting and allowing your users to change their own dialplans, however, a sip reload becomes rather disastrous in the middle of hte day as it seeds Sip peeers. |
| 23:21.39 | mercestes | However, having cdr's in mysql is *extremely* handy and works really well. |
| 23:22.10 | NormanASD | Does anyone know how to get Asterisk to pass SUBSCRIBE request from a client to another client? |
| 23:22.19 | syzygyBSD | linuxtuxie: that isn't all of your nat rules is it? |
| 23:22.29 | [TK]D-Fender | pjz : Pastebin the whole thing and scan it |
| 23:22.58 | pjz | got it |
| 23:23.15 | pjz | I tried commenting a line out with a leading # instead of ; |
| 23:23.53 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 23:23.58 | linuxtuxie | syzygyBSD: no, in front of the ones you saw in pastebin all the chains get the accept policy |
| 23:24.15 | syzygyBSD | linuxtuxie: would you mind pastebining all of them? |
| 23:24.34 | syzygyBSD | I need to see the masq rules as well |
| 23:25.22 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 23:25.53 | linuxtuxie | syzygyBSD: http://pastebin.be/3992/ |
| 23:28.12 | awannabe | for priority "t" can you adjust the timeout at all? |
| 23:28.15 | linuxtuxie | syzygyBSD: the vars equal to -> ASTERISK_IP="192.168.200.1" | EXT_ETH="ppp0" | IPTABLES="/sbin/iptables" |
| 23:28.57 | syzygyBSD | k, thanks, I am writing one that should work |
| 23:30.27 | syzygyBSD | this firewall... is it your nat router as well? |
| 23:30.35 | NormanASD | Does anyone know how to get Asterisk to pass SUBSCRIBE request from a client to another client? I hope this is possible... |
| 23:30.38 | [TK]D-Fender | awannabe : "show function TIMEOUT" |
| 23:31.17 | tzafrir_laptop | I remember that someone posted a patch to remember echotraining parameters |
| 23:31.18 | *** join/#asterisk g0tw00d (n=rchace@68-113-159-200.dhcp.nplt.ne.charter.com) |
| 23:31.25 | tzafrir_laptop | zaptel echo training) |
| 23:31.28 | shido6 | http://pastebin.ca/259667 |
| 23:31.49 | tzafrir_laptop | Has it been accepted? If not: any idea where I can find it? |
| 23:32.07 | awannabe | shido6: i want basically instant, like right now this is setup on a auto attendant, and if no one presses any keys i need to go to the t priority, but immaidelly |
| 23:32.16 | awannabe | so just make those timeouts 0 then? |
| 23:32.39 | shido6 | weird.. |
| 23:32.46 | shido6 | if im slow in pressing a key |
| 23:32.54 | shido6 | its going to go to t if you set it to 0 |
| 23:32.58 | shido6 | that might be too quick |
| 23:33.03 | awannabe | i agree, this customer is a total iddiot, heh |
| 23:33.08 | [TK]D-Fender | awannabe : "t" is an exten, and there has to be a time, it can't be instant How you be instant and WAIT for keys? |
| 23:33.29 | awannabe | so set the digit and timeout to say 1 then? |
| 23:33.32 | syzygyBSD | linuxtuxie: http://pastebin.be/3993/ |
| 23:33.34 | syzygyBSD | try that |
| 23:33.50 | [TK]D-Fender | awannabe : What are their options, and where are you sending it on timeout? |
| 23:33.56 | *** join/#asterisk Alric (n=nbowyer@wireless-062.1stel.com) |
| 23:34.40 | awannabe | [TK]D-Fender: press 1 for celluar, 2 for voicemail, 0 for operator, if nothing is pressed we want it to go to voicemail |
| 23:34.49 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-183.mtco.com) |
| 23:36.18 | Shaun2222 | is there anyway to make a macroexit if it's passed onto voicemial.. |
| 23:38.36 | awannabe | [TK]D-Fender: http://pastebin.ca/261916 thats what i have, and now i broke it lol |
| 23:39.05 | linuxtuxie | syzygyBSD: The problem still remains |
| 23:39.19 | litage | what are the most probable causes of audio cutting in and out and stuttering for outgoing sip calls and incoming zap calls? |
| 23:39.30 | syzygyBSD | linuxtuxie: what is the problem? |
| 23:39.40 | linuxtuxie | registration works...but when placing a call, it just retransmits 6 times |
| 23:39.56 | *** part/#asterisk linuxtuxie (n=kku@168.50-241-81.adsl-dyn.isp.belgacom.be) |
| 23:40.25 | *** join/#asterisk linuxtuxie (n=kku@168.50-241-81.adsl-dyn.isp.belgacom.be) |
| 23:40.35 | linuxtuxie | hmmm, got disconnected |
| 23:40.48 | syzygyBSD | hmm, i hoped the -A Forward rule woulda fixed that |
| 23:44.35 | [TK]D-Fender | awannabe : Ditch s,5 |
| 23:44.55 | *** join/#asterisk jeffik (n=Jeff@208-41-192-106.client.dsl.net) |
| 23:45.52 | awannabe | [TK]D-Fender: ill try that, right now it playing the greeting then disconnecting the call, heh |
| 23:46.05 | *** join/#asterisk jm|work (n=jamiem@zen.jamiem.com) |
| 23:46.25 | *** join/#asterisk linuxtuxi1 (n=kku@220.50-200-80.adsl-dyn.isp.belgacom.be) |
| 23:46.37 | [TK]D-Fender | awannabe Do other IVR's work? |
| 23:46.58 | awannabe | yeah, im trying with the s,5 being removed |
| 23:47.30 | awannabe | "auto fallthough, channel zap/xxx is status unknonw" |
| 23:48.01 | [TK]D-Fender | awannabe : Autofallthrough = BAD. disable that in [globals] |
| 23:49.46 | linuxtuxi1 | syzygyBSD: do you have such a working nat setup ? |
| 23:49.50 | awannabe | [TK]D-Fender: thanks, that fixed it! |
| 23:49.58 | *** join/#asterisk [Outcast] (n=bill@219-89-206-239.adsl.xtra.co.nz) |
| 23:50.34 | [Outcast] | does 1.4 support SMP ? |
| 23:50.58 | syzygyBSD | linuxtuxil mine is more complicated, I have a multihomed setup |
| 23:51.04 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 23:51.48 | CunningPike | [Outcast]: As in SMP kernels? |
| 23:52.31 | brian | hey, just wondering if there is anyway at all to listen for DTMF on a meetme conference using fastagi? |
| 23:52.47 | syzygyBSD | linuxtuxil couple questions, your firewall box, how many nics does it have? |
| 23:52.56 | [Outcast] | CunningPike: If i am running a SMP kernel will asterisk threads run different processors or will still run on CPU 0 |
| 23:53.30 | CunningPike | [Outcast]: Don't know about that aspect of 1.4 - sorry |
| 23:53.33 | *** join/#asterisk Tako-san (n=jmkiffia@24.108.162.254) |
| 23:54.38 | linuxtuxi1 | This nat problem is a hard one, I will give it another go tomorrow, nevertheless I would like to thank you all for the great help & insights! |
| 23:55.30 | cloud9 | Shaun2222 : I'm seeing this message when I try to use an extension from the real time architecture: "Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available" |
| 23:55.42 | syzygyBSD | I have nat working on a ppp0 setup so I know how it is needed to work, but I have custom routes as well because of my multi homed setup, |
| 23:55.42 | cloud9 | did you happen to run into that? |
| 23:56.03 | syzygyBSD | send me a message tomorrow and I will help you get that up and working |
| 23:56.59 | NormanASD | On SUBSCRIBE support in Asterisk, I always seem to get "403 Forbidden (within dialog)" from asterisk when my client sends a SUBSCRIBE request (that I want to go to the other peer.) Any way to make this work? |
| 23:57.07 | linuxtuxi1 | syzygyBSD: Already thanks in advance! I can reach you again on the #asterisk forum tomorrow ? |
| 23:57.24 | syzygyBSD | um.. in this channel, I am never on the forum |
| 23:57.47 | linuxtuxi1 | syzygyBSD: indeed, my mistake :) |
| 23:58.08 | syzygyBSD | does a wget work from the asterisk box? |
| 23:58.42 | Shaun2222 | cloud9: i actually never finished my realtime project... i had extensions, queues, queue memebers, voicemail and iax users and peers in realtime |
| 23:58.45 | linuxtuxi1 | sure |
| 23:58.51 | Shaun2222 | cloud9: i never got to adding sip yet. |
| 23:59.40 | Shaun2222 | cloud9: i would imaging setting it up is basically the same as iax though... in the extconfig.conf sipusers => mysql,asterisk,sip_users |
| 23:59.47 | Shaun2222 | and then creating the table.. |
| 23:59.55 | Shaun2222 | you can combine the user/peers tables if you want |
| 23:59.58 | Shaun2222 | i kept them seperate |