| 00:00.35 | smackus | sorry... never sent pastebin :-D |
| 00:00.42 | smackus | http://pastebin.ca/80959 |
| 00:00.45 | smackus | im a jackass |
| 00:01.46 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 00:02.10 | rene- | why make config says that asterisk cannot stat init.asterisk because it cant find it |
| 00:02.19 | rene- | where should init.asterisk be? |
| 00:02.32 | rene- | and why 1.2.9.1 makefile isnt finding it |
| 00:02.41 | r_evolution | make config is for redhat |
| 00:02.50 | Qwell[] | r_evolution: only on 1.2 |
| 00:03.35 | russellb | Qwell[]: which is what everyone uses :) |
| 00:03.37 | r_evolution | ^ |
| 00:03.40 | r_evolution | he's got you there |
| 00:03.47 | Qwell[] | russellb: oh, sure :p |
| 00:04.00 | rene- | no make config for debian? mmm |
| 00:04.03 | Qwell[] | but, as a general statement, that is false |
| 00:04.09 | r_evolution | jerk. |
| 00:04.09 | russellb | that's true. |
| 00:04.17 | [Airwolf] | Qwell, do you know when 1.4 will be avalible ? |
| 00:04.26 | russellb | [Airwolf]: some day ... |
| 00:04.26 | Qwell[] | "soon" |
| 00:04.31 | [Airwolf] | hehe |
| 00:04.31 | r_evolution | heh russell. |
| 00:04.34 | russellb | [Airwolf]: you can blame file |
| 00:04.35 | r_evolution | 'some day soon' |
| 00:04.36 | r_evolution | i love it |
| 00:04.37 | [Airwolf] | Like duke nukem forever |
| 00:04.38 | [Airwolf] | :P |
| 00:04.41 | r_evolution | yesss |
| 00:04.52 | r_evolution | the duke nukem movie will be out... someday |
| 00:04.57 | file | blame me |
| 00:05.00 | r_evolution | sorta like the neuromancer movie will be out... some day |
| 00:05.03 | Qwell[] | file: we always do |
| 00:05.14 | r_evolution | not true... i prefer to blame you qwell |
| 00:05.14 | file | Magical eight ball says... next week |
| 00:05.16 | rene- | well now thats some creativity |
| 00:05.31 | r_evolution | just because you're a heretic |
| 00:05.34 | russellb | file: that's a lie |
| 00:05.34 | r_evolution | even when i wake up for work |
| 00:05.36 | r_evolution | i blame you |
| 00:05.45 | r_evolution | magical lie ball? |
| 00:05.47 | file | russellb: nope! |
| 00:05.52 | russellb | file: w00t |
| 00:05.56 | r_evolution | O_o |
| 00:05.59 | smackus | so can anyone see anything wrong with the way I am calling voicemail? |
| 00:06.01 | smackus | http://pastebin.ca/80959 |
| 00:06.13 | r_evolution | people tend to be a little ADD smackus |
| 00:06.23 | smackus | it does not play the unavail or busy messages, just /tmp |
| 00:06.36 | smackus | r_evolution: this I know |
| 00:06.48 | [Airwolf] | I really hate it that I can't find any voip clients voor windows mobile 5 |
| 00:07.01 | [Airwolf] | Or that they don't exist |
| 00:07.02 | r_evolution | well... is not the busy to be called by s+101? |
| 00:07.12 | file | russellb: think positive |
| 00:07.19 | smackus | is it? |
| 00:07.27 | smackus | i just went off of the sample in the extensions.conf |
| 00:08.07 | r_evolution | erm |
| 00:08.09 | r_evolution | n+101 |
| 00:08.11 | r_evolution | my bad |
| 00:08.26 | russellb | file: i suppose while we're there we might be able to gitrdone |
| 00:08.32 | rene- | r_evolution: http://bugs.digium.com/view.php?id=6922 has the definite answer |
| 00:08.35 | file | russellb: yes! |
| 00:08.51 | smackus | so instead of s-NOANSWER, you are saying n+101? |
| 00:08.57 | smackus | am i following you on that? |
| 00:09.45 | smackus | dial+101,Voicemail() |
| 00:09.48 | russellb | n+101 is going away, you should not be using it |
| 00:09.52 | Qwell[] | russellb: how done will you gitr, exactly? |
| 00:09.53 | XARiUS | anyone know why if I just this past weekend pulled from svn asterisk-1.2 branch, why I wouldn't have the hint application for presence? |
| 00:10.03 | smackus | ok... so which way is correct? |
| 00:10.04 | Qwell[] | XARiUS: It isn't an application |
| 00:10.07 | smackus | n+ or what? |
| 00:10.08 | r_evolution | cool deal rene- :) |
| 00:10.08 | russellb | XARiUS: there is no hint application |
| 00:10.21 | Qwell[] | It's a prirority |
| 00:10.22 | r_evolution | ok nevermind listen to russellb then |
| 00:10.28 | XARiUS | eh, okie, when I try and use hint as described in dialplan, * just bitches about no application.. |
| 00:10.38 | XARiUS | ahhhh |
| 00:10.44 | XARiUS | okie, brb, lemme hack this up then |
| 00:10.46 | Qwell[] | XARiUS: 5551211,hint,5551212@somecontext |
| 00:10.51 | XARiUS | gotchya |
| 00:10.53 | smackus | so have I dont it correctly then? |
| 00:10.57 | smackus | russellb: |
| 00:11.03 | smackus | http://pastebin.ca/80959 |
| 00:11.08 | russellb | smackus: if you have gone by the example, then yes |
| 00:11.40 | smackus | ok. then can you explain to me why then when i get a noanswer or busy on an extension it still calls the tmp message? |
| 00:13.02 | russellb | tmp message? |
| 00:15.47 | smackus | russellb: http://pastebin.ca/80976 |
| 00:16.07 | smackus | i delete tmp, and it recreates it |
| 00:16.39 | smackus | tmp is empty |
| 00:16.41 | russellb | well if it's playing it, it didn't get deleted ... |
| 00:16.47 | burnproof | good day guys, busydetect seem's like doesn't work for me, can anyone suggest some tweaks? |
| 00:16.48 | burnproof | thanks |
| 00:17.12 | smackus | there is nothing in /tmp |
| 00:17.19 | smackus | so there is nothing to play. |
| 00:17.21 | burnproof | since my telco line doesn't have disconnection supervision it's really give me an headache |
| 00:17.27 | smackus | even if i deleted it. |
| 00:17.28 | russellb | ls /var/spool/asterisk/voicemail/progrexion/156/ |
| 00:18.25 | smackus | http://pastebin.ca/80980 |
| 00:18.26 | *** join/#asterisk mrdigital (n=mr@pool-72-81-28-149.phlapa.east.verizon.net) |
| 00:18.37 | infinity1 | is it posible to play the gsm files in windows media player? |
| 00:18.41 | russellb | smackus: do you not see the temp message there? |
| 00:19.02 | russellb | temp.gsm temp.wav temp.WAV |
| 00:19.08 | infinity1 | ? |
| 00:19.22 | infinity1 | oh. yer not talking to me :) |
| 00:19.28 | mrdigital | hey infinity1? |
| 00:19.28 | korihor | jajja |
| 00:19.40 | smackus | so when I create a voicemail account, do I have to go in and delete the temp files? |
| 00:19.41 | infinity1 | ? |
| 00:19.45 | smackus | cuz they all have them. |
| 00:19.46 | infinity1 | mrdigital: yea |
| 00:19.47 | russellb | smackus: no, those are created by the user |
| 00:19.52 | mrdigital | Can i pm you infinity1 |
| 00:19.56 | infinity1 | sure |
| 00:20.03 | r_evolution | dude. |
| 00:20.05 | r_evolution | mrdigital |
| 00:20.08 | russellb | smackus: or they should only be present if created by the user |
| 00:20.19 | *** join/#asterisk Krush (n=Krush@p5092342B.dip0.t-ipconnect.de) |
| 00:20.25 | Krush | good evening |
| 00:20.27 | r_evolution | you need to learn basic dialplan stuff before you can hope to implement mysql stuff in the dialplan :-\ |
| 00:20.35 | russellb | infinity1: i think you use the wav49 format if you want to play in win media player ... |
| 00:20.51 | smackus | But why would my dial plan not call busy or unavail rather than tmp? i will delete those and see what happens. |
| 00:20.58 | mrdigital | russellb: you know dialplans and agis? |
| 00:21.04 | r_evolution | heh ;x |
| 00:21.09 | russellb | mrdigital: no, i know nothing |
| 00:21.11 | mrdigital | hey r_evolution you got my pm? |
| 00:21.13 | infinity1 | russellb: yes. i don't want to convert a bunch of gsms to wav. thoguth there might be an easy way |
| 00:21.14 | Krush | is there a way to redefine the voicemail storage directory of asterisk at runtime? |
| 00:21.24 | burnproof | :) |
| 00:21.30 | russellb | infinity1: well, in the trunk there is a "convert" CLI command ..... |
| 00:21.45 | r_evolution | russellb doesnt do anything for Asterisk... nothing at all ;) |
| 00:21.57 | burnproof | r_evolution: :) |
| 00:22.01 | r_evolution | yes mrdigital... I typed all that out earlier for you... im not doing it again :) |
| 00:22.02 | russellb | r_evolution: nope, i don't even know how i got here ... |
| 00:22.06 | infinity1 | hm |
| 00:22.08 | file | russellb poses for Digium/Asterisk calendar pictures |
| 00:22.13 | XARiUS | oh hell, freenode's been hacked again! |
| 00:22.14 | XARiUS | ;) |
| 00:22.17 | r_evolution | me neither!! i thought you were here for the purposes of juggling |
| 00:22.51 | r_evolution | speaking of useless where is that damn madsen kid... |
| 00:23.02 | file | haha... kid... |
| 00:23.17 | r_evolution | ;) |
| 00:23.50 | r_evolution | why is it that im speaking |
| 00:23.55 | r_evolution | and i feel like this woman isnt listening |
| 00:24.07 | r_evolution | i say... open the web browser and go to <internal IP> |
| 00:24.23 | r_evolution | she says she cant open web pages... i say it doesnt matter... just do ^ |
| 00:24.28 | r_evolution | she repeats that she cant open up web pages |
| 00:24.30 | rene- | is being ADD is yet another thing that is nerdy if nerds have it but becomes fashionable and something people will pretend having? |
| 00:24.35 | mrdigital | you did? where was i |
| 00:24.46 | r_evolution | and the scary thing is that she's one of the techs :( |
| 00:24.52 | r_evolution | you were on the other end of the PM i presume |
| 00:25.24 | r_evolution | i wasnt at your house... but im pretty sure you were on the other end of the PM :) |
| 00:25.32 | mrdigital | r_evolution i left when you typed it |
| 00:27.24 | r_evolution | im sorry :( |
| 00:30.02 | r_evolution | now i really gotta figure out why the ping-time is such the suck |
| 00:31.46 | r_evolution | 1005ms O_O |
| 00:32.36 | XARiUS | ouch. |
| 00:34.58 | r_evolution | yeah |
| 00:35.01 | r_evolution | no kidding! |
| 00:35.10 | r_evolution | im trying to figure out if the ATA is just sucking... or whaaaaaat |
| 00:35.44 | rene- | my ata is stupid too |
| 00:35.45 | phigwork | i have this linksys rt31p2 sitting here |
| 00:35.48 | rene- | it looses its registration |
| 00:35.53 | phigwork | it'd be so nice if i could install linux on it |
| 00:36.37 | rene- | i want a linksys to install kismet on it and get wifi |
| 00:36.49 | phigwork | the wrt's used to be $30-50 |
| 00:37.05 | rene- | they are like 100 |
| 00:37.08 | rene- | here |
| 00:37.19 | phigwork | why can't we get a retail fully configurable voip router for that much :/ |
| 00:37.46 | phigwork | stupid locked hardware |
| 00:37.58 | drray | I wonder why digium won't make a fxs/router box based on the iaxy |
| 00:38.18 | phigwork | does the handytone box do iax? |
| 00:38.20 | phigwork | or just sip? |
| 00:38.48 | phigwork | i was wonderin what the big deal was aboot the iaxy over the handytone |
| 00:41.02 | XARiUS | iax requires no proxy or stun servers to traverse nat. |
| 00:41.12 | XARiUS | SIP.. well.. behind most sane routers, sip works. |
| 00:41.18 | XARiUS | then again, I've seen it totally not work :) |
| 00:41.26 | phigwork | sip clients work find behind nat |
| 00:41.31 | phigwork | err fine |
| 00:41.44 | phigwork | getting someone to register to you with nat doesn't work so well |
| 00:41.51 | XARiUS | most do now days, but there's lots of, shall we call them legacy routers out there that totally muck up sip. |
| 00:42.17 | phigwork | but i thought the handytone did iax. i guess i was wrong |
| 00:42.24 | XARiUS | I dunno, maybe it does? |
| 00:43.06 | r_evolution | actually XARiUS... we've got a lady on the phone with a WRT54G that is refusing to let the ATA connect to * |
| 00:43.29 | phigwork | why? |
| 00:43.30 | r_evolution | and given that it just took me 45minutes to walk her and the tech through putting the line directly to the ATA to check and make sure an IP was being assigned |
| 00:43.31 | XARiUS | is it a NEW wrt54g? |
| 00:43.37 | r_evolution | i doubt it :) |
| 00:43.40 | XARiUS | I picked up a brand new one last week for a client (the ones running that other linux version) |
| 00:43.49 | XARiUS | it refused ANY sip until I upgraded the firmware, flat out blocked it |
| 00:44.04 | phigwork | i've got that sveasoft one on mine |
| 00:44.14 | phigwork | not the latest version tho, cause my membership expired |
| 00:44.19 | r_evolution | given that it took me 45 mins to do the simplest thing |
| 00:44.21 | r_evolution | :) |
| 00:44.34 | r_evolution | i dont think ill be helping them setup port forwarding |
| 00:44.37 | XARiUS | yeah same here, but the newer wrt54g's run a different sort of chip or linux, I forget which, but sveasoft won't work on them. |
| 00:44.37 | r_evolution | when i leave in 15 minutes |
| 00:44.39 | r_evolution | :) |
| 00:44.47 | XARiUS | lol evo, why should they even need it? |
| 00:44.59 | r_evolution | welllllllll |
| 00:45.04 | r_evolution | you might ask that question |
| 00:45.23 | XARiUS | or is the ata dirt old? |
| 00:45.34 | r_evolution | nah |
| 00:45.51 | r_evolution | anyways... the ATA seems to drop the internet connection sporadically when hooked directly to cable or dsl |
| 00:45.53 | r_evolution | whicheva it is |
| 00:46.02 | r_evolution | only does this with a couple of crappy ISP |
| 00:46.04 | r_evolution | ISPs* |
| 00:46.14 | r_evolution | in that case we usually put it behind a router and everything works fine |
| 00:46.41 | r_evolution | and no one is claiming the ATA is close to high-quality (UT Starcom) |
| 00:46.44 | r_evolution | but it works decently :) |
| 00:46.56 | XARiUS | yeah well good luck with that dude! |
| 00:46.59 | XARiUS | happy hour time! |
| 00:47.01 | XARiUS | haha |
| 00:47.09 | r_evolution | haha |
| 00:47.10 | r_evolution | yeah |
| 00:47.15 | r_evolution | unhappy hour :( |
| 00:47.19 | r_evolution | actually 10 mins |
| 00:47.21 | r_evolution | and im out |
| 00:47.29 | XARiUS | you must be pacific. |
| 00:47.31 | r_evolution | *counts down* |
| 00:47.37 | r_evolution | nah EST |
| 00:47.40 | XARiUS | ew. |
| 00:47.44 | r_evolution | mmmhmm |
| 00:47.45 | XARiUS | slave labor, there till 9? |
| 00:47.54 | r_evolution | slave labor at a good rate :) |
| 00:48.01 | XARiUS | fair enuff :) |
| 00:48.15 | r_evolution | there's 3 of us here... who are capable of more than the basic crap |
| 00:48.21 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 00:48.23 | r_evolution | and one of us has to be here at all times |
| 00:48.25 | r_evolution | per management |
| 00:48.32 | r_evolution | so some days i get to work 10 - 9 |
| 00:48.34 | r_evolution | which means |
| 00:48.41 | r_evolution | one day a week... i hate life |
| 00:48.42 | XARiUS | where do ya work? |
| 00:48.52 | r_evolution | i would be an agent of hell |
| 00:49.02 | XARiUS | if ya don't mind me askin |
| 00:49.03 | r_evolution | ;) |
| 00:49.08 | r_evolution | little ITSP in the making |
| 00:49.12 | r_evolution | everything based off * |
| 00:49.14 | *** join/#asterisk Katty (n=Administ@dialup-4.244.123.121.Dial1.StLouis1.Level3.net) |
| 00:49.17 | XARiUS | ohhh I've heard of that place!! nice! |
| 00:49.18 | XARiUS | :) |
| 00:49.19 | r_evolution | which is just FUN FUN for me some days |
| 00:49.20 | r_evolution | haha |
| 00:49.33 | XARiUS | northeast? southeast? |
| 00:49.40 | Katty | allo. |
| 00:49.49 | r_evolution | basically it's a small relatively unknown place that relies on a much larger pyramid marketing group to bring in new customers |
| 00:49.50 | r_evolution | SE |
| 00:49.53 | r_evolution | VA to be exact |
| 00:50.03 | r_evolution | hey it's one of like |
| 00:50.05 | r_evolution | 3? |
| 00:50.06 | XARiUS | phew.. you lucked out, almost landed in WV :) |
| 00:50.08 | r_evolution | females in here |
| 00:50.11 | r_evolution | no kidding! |
| 00:50.21 | r_evolution | tho... i DID think i was in WV the other night |
| 00:50.33 | XARiUS | plenty of females there! tho usually related :x |
| 00:50.36 | XARiUS | JUST KIDDING ANYONE FROM WV |
| 00:50.37 | r_evolution | at a field party with my girl and saw some other girl dancing/grinding on her younger brother... |
| 00:50.42 | XARiUS | hahahaha |
| 00:50.42 | r_evolution | i was like... |
| 00:50.43 | r_evolution | wtfux? |
| 00:50.53 | Katty | this is riddiculus. |
| 00:50.57 | rene- | i saw an IAX phone with builtin modem |
| 00:50.57 | r_evolution | im serious... i was damn near physically revolted... |
| 00:51.00 | XARiUS | whats up Katty? |
| 00:51.05 | r_evolution | b/c the girl is cute... but im WTFUX |
| 00:51.07 | Katty | clearly not your iq. |
| 00:51.13 | r_evolution | what? this place is ridiculous |
| 00:51.27 | XARiUS | Oh, I see, it's time for personal insults because it's happy hour and someone decides to poke a little fun? |
| 00:51.39 | r_evolution | pretty much XARiUS :) |
| 00:51.41 | r_evolution | that's ok |
| 00:51.42 | Katty | way to take a joke. |
| 00:51.45 | XARiUS | I was born and raised in Alabama, Katty. I'm allowed to make comments such as that, since I've had them directed toward me. |
| 00:51.46 | r_evolution | i'll laugh with you :) |
| 00:51.58 | r_evolution | AL scares me... |
| 00:52.12 | r_evolution | mosquitos so big... they dont just bite you... those fuckers EAT YOU WHOLE |
| 00:52.52 | XARiUS | AL is awesome. Well, the northern part, anyway. |
| 00:52.54 | XARiUS | Digium is in Huntsville :) |
| 00:53.08 | r_evolution | as are really freaking big mosquitos |
| 00:53.24 | r_evolution | frighteningly big actually |
| 00:53.25 | r_evolution | like horses. |
| 00:53.44 | Katty | i have nothing against AL |
| 00:53.48 | Katty | twisted's from there. |
| 00:53.54 | XARiUS | this I know :) |
| 00:53.58 | Katty | he's mostly normal. |
| 00:54.06 | Katty | except when there's bad weather |
| 00:54.21 | r_evolution | something just doesnt jive right about that... |
| 00:54.41 | XARiUS | I'm in San Diego now. Thats what I miss most about the southeast, the weather. |
| 00:54.54 | XARiUS | When you see rain 2.. maybe 3 times a year, you start to really miss a lil thunderstorm every now and again. |
| 00:55.33 | *** join/#asterisk eDIsonxl (n=eDison@60-248-182-162.HINET-IP.hinet.net) |
| 00:55.41 | Qwell[] | Katty: ...twisted...normal...? |
| 00:55.49 | Qwell[] | Katty: are we talking about the same person? |
| 00:55.53 | Katty | Qwell: yup. |
| 00:55.58 | Katty | Qwell: he's normal around me. |
| 00:56.07 | XARiUS | I do remember him mentioning freaking out about a thunderstorm a long time ago. |
| 00:56.08 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:56.14 | XARiUS | It's been ages since I've been in here on a regular basis. |
| 00:56.24 | Katty | he doesn't like tornados. |
| 00:56.26 | Katty | at /all/ |
| 00:56.29 | Katty | hi file! |
| 00:56.31 | r_evolution | file seems to be rather hyper tonight |
| 00:56.33 | XARiUS | ahh maybe thats what it was :) |
| 00:56.34 | Katty | have you brought me a muffin? |
| 00:56.37 | r_evolution | wait... who DOES like tornados? |
| 00:56.38 | file | I have! |
| 00:56.42 | Katty | oh boy! |
| 00:57.02 | hads | Damn, now I feel like a muffin. |
| 00:57.08 | r_evolution | i've only seen two or three... i still don't like them |
| 00:57.12 | Katty | r_evolution: that's normal file behavior. |
| 00:57.22 | Katty | cept he doesn't do it /for real/ |
| 00:57.23 | Katty | much |
| 00:57.25 | XARiUS | I was about a mile from the one that ripped thru Huntsville back in the 80's. |
| 00:57.28 | XARiUS | it went thru my back yard |
| 00:57.40 | Katty | file: i'd say you hug people more than run around them in circles |
| 00:57.47 | file | this is true |
| 00:58.36 | r_evolution | O_o |
| 00:58.39 | X-Rob_ | ~book |
| 00:58.40 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 00:59.02 | r_evolution | not quite sure why that happened |
| 00:59.16 | r_evolution | Rob forget where the book was located? |
| 00:59.19 | r_evolution | got nostalgic? |
| 00:59.31 | DrkShdw | ? thebook |
| 00:59.33 | Katty | file: you should call me tomorrow |
| 00:59.37 | r_evolution | XARiUS... im curious... |
| 00:59.40 | Katty | file: sometime...but not from 11-12 cst. |
| 00:59.47 | file | Katty: eeeeeeep! perhaps |
| 00:59.53 | r_evolution | how is one a mile from something... but it goes through one's backyard? |
| 00:59.59 | r_evolution | one must've had a BIG FUCKING BACKYARD |
| 01:00.00 | Katty | file: cause if you call then, you'll have to listen to my voicemail. |
| 01:00.07 | XARiUS | I grew up on quite a bit of land. |
| 01:00.09 | Katty | file: and that'd be Just Awful(tm) |
| 01:00.15 | XARiUS | which included a 15 acre lake. |
| 01:00.18 | file | Katty: is your voicemail darn good? |
| 01:00.20 | r_evolution | ahhh |
| 01:00.23 | XARiUS | :) |
| 01:00.27 | r_evolution | i grew up in the woods :) |
| 01:00.32 | r_evolution | literally... feral child. |
| 01:00.46 | Katty | file: call me now and find out! |
| 01:00.49 | rene- | does make config build startup scripts for zaptel drivers?? |
| 01:00.52 | r_evolution | my first arrest was at the ripe old age of 14 ;x |
| 01:00.57 | r_evolution | seems to rene- |
| 01:00.57 | rene- | hah |
| 01:01.02 | file | Katty: nah |
| 01:01.06 | file | I'll let it be a surprise |
| 01:01.07 | rene- | r_evolution: what were you doing |
| 01:01.08 | *** join/#asterisk EvilDeshi (n=EvilDesh@pool-71-98-15-189.mdsnwi.dsl-w.verizon.net) |
| 01:01.20 | Katty | file: kayo. |
| 01:01.22 | r_evolution | being extremely intoxicated at a 21+ club |
| 01:01.23 | XARiUS | rene: last time I ran it, it did for ztdummy back in the day |
| 01:01.27 | XARiUS | rene: I think it still might |
| 01:01.50 | r_evolution | anyhow. |
| 01:01.50 | EvilDeshi | im having a sound issue with my asterisk where when i call the box from a softphone the audio is really choppy and hard to understand we are running debian linux 2.6 kernel on vmware |
| 01:01.52 | r_evolution | im going home clowns |
| 01:02.02 | r_evolution | good night to all... and to all a good night |
| 01:02.03 | XARiUS | cya evo. |
| 01:02.18 | r_evolution|afk | slave labor ends for the day |
| 01:02.22 | rene- | hah |
| 01:02.44 | rene- | time to get busted, jk |
| 01:05.20 | rene- | the bot is cool i want to take it home |
| 01:05.52 | XARiUS | so how bout those elections rene? :) |
| 01:06.02 | rene- | jeezus christ: i dont know |
| 01:06.09 | rene- | <PROTECTED> |
| 01:06.41 | XARiUS | hopefully he has a better run than the recently elected (meaning plurarl) chief's of police in TJ. |
| 01:06.42 | XARiUS | :) |
| 01:06.44 | rene- | but the right wing guy one for like 200k votes out of 41Million |
| 01:06.53 | XARiUS | errr *plural |
| 01:07.19 | XARiUS | yeah it was really close |
| 01:07.27 | *** join/#asterisk adker (n=adker@74-33-220-165.br1.glv.ny.frontiernet.net) |
| 01:08.08 | rene- | both parties are really corrupt, |
| 01:08.17 | rene- | pan has just access to greater power |
| 01:08.29 | rene- | and stole the most |
| 01:08.44 | drray | I'm wondering why so many elections in north america have been so close |
| 01:09.13 | rene- | what kindof sucks is that IFE or the election governing body IT systems are being run by PAN brother in law |
| 01:09.31 | XARiUS | oh yeah, that could really suck. |
| 01:11.52 | rene- | the leftist guy here is more like brazil's lula than venezuela chavez or cubas castro |
| 01:13.09 | *** join/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com) |
| 01:13.43 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 01:14.18 | PMantis | If I have two iaxtel numbers, anyone know how to route each one differently? I tried specifying a context on the register => line... |
| 01:14.42 | PMantis | But, they still both come in as 's@default' |
| 01:15.48 | orlock | PMantis: i know this one! |
| 01:15.56 | orlock | PMantis: i just figured it out last night |
| 01:16.05 | PMantis | haha, awesome! :) |
| 01:16.15 | orlock | PMantis: in your sip registration, what are you sending as the Contact: string? |
| 01:16.27 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
| 01:16.44 | orlock | PMantis: i noticed that mine was sending <sip: s@my.ip.address> |
| 01:16.45 | PMantis | orlock, You mean the iax registration. :) |
| 01:16.55 | orlock | PMantis: Hmm.. ahh. may be different then |
| 01:17.05 | file | IAXtel doesn't have the capability to do that PMantis |
| 01:17.22 | orlock | PMantis: but if its the issue i had, it defaults to "s" for the registrant username |
| 01:17.26 | orlock | but i was using sip trunks |
| 01:17.43 | PMantis | Yeah, that makes a difference.. |
| 01:17.45 | orlock | PMantis: my fix was to add the phone number at the end of the registration string |
| 01:18.17 | orlock | but i dont know about iax |
| 01:18.22 | PMantis | file, That stinks. I have a DID for business calls, 3 for home use, wanted 1 IAXtel # for business, and 1 for home. |
| 01:18.28 | Ritalin- | you can't do it with whatevernumber/s@default ? |
| 01:18.39 | file | I don't know if I can change it either |
| 01:19.42 | PMantis | I tried: register => username:password@iaxtel.com/17001234567 |
| 01:19.50 | PMantis | and... register => username:password@iaxtel.com/in-iaxtel-business |
| 01:19.52 | PMantis | etc |
| 01:20.17 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 01:20.20 | PMantis | oh and register => username:password@iaxtel.com/17001234567@in-iaxtel |
| 01:20.25 | file | long story but server side it won't |
| 01:20.30 | file | it will always send to s |
| 01:21.08 | PMantis | file, You mean 'technical' story? |
| 01:21.08 | Ritalin- | pmantis: who is your itsp? |
| 01:21.21 | PMantis | Ritalin-, I'm trying to do this with IAXtel |
| 01:21.40 | PMantis | file, Or.. was there abuse? |
| 01:21.54 | file | when IAXtel was setup I don't believe it was made to send it to your number... just s |
| 01:22.47 | Ritalin- | I've got a SPA3000 that I'm having bad echo problems with. I turn down the gain and then there's a volume problem *sigh* |
| 01:23.03 | *** join/#asterisk linlin (n=linlin@c-67-184-159-30.hsd1.il.comcast.net) |
| 01:23.09 | PMantis | file, I guess if I want it bad enough I'll instal another copy of * running on the same box |
| 01:23.30 | file | let me look in here... |
| 01:23.53 | Ritalin- | pm: what hardware are you using? just out of curiosity |
| 01:24.33 | PMantis | But.. how could this work in a hosted environment then? If one asterisk server served multiple clients and more than one had an IAXtel account ? |
| 01:24.40 | PMantis | Ritalin-, What do you mean? Phones? |
| 01:24.50 | Ritalin- | tdm400? |
| 01:24.57 | file | then it wouldn't work :) |
| 01:25.09 | X-Rob_ | heh |
| 01:25.41 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 01:26.08 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 01:26.09 | PMantis | Ritalin-, My server has 2 tdm400's for 6 extensions, plus IAXy's, a Polycom IP501, and a Linksys PAP2 |
| 01:26.46 | file | PMantis: I can... try to change this... |
| 01:27.08 | PMantis | file, you can... what? :-) |
| 01:27.14 | file | I have IAXtel access |
| 01:27.14 | Ritalin- | pm: have a lot of people working for you? |
| 01:27.29 | X-Rob_ | heh |
| 01:27.31 | PMantis | file, Ahhhh, ok, cool. :) |
| 01:27.53 | *** join/#asterisk pdtmobile (n=ptinsley@209.12.249.243) |
| 01:27.57 | Strom_C | oh, the joy of coffeeshop wireles |
| 01:28.01 | PMantis | Ritalin-, Well, there's me, my wife, 3 kids... :-) |
| 01:28.01 | Strom_C | s |
| 01:28.10 | X-Rob_ | Ritalin-, on my desk I have a SPA941, GXP2000, PAP2 with an analog phone, SPA3000 with the input from port2 on the pap2 and the output to another analog phone, a BT101 and a snom360. |
| 01:28.19 | PMantis | Ritalin-, i'm a small business |
| 01:28.19 | X-Rob_ | that's not an unusually large number of phones for a dev 8) |
| 01:28.28 | Strom_C | it's frankenpbx |
| 01:28.33 | PMantis | lol |
| 01:28.33 | Strom_C | that's what it is :) |
| 01:29.07 | Strom_C | fine for a devleoper, but man, is it painful to go find that at a client's location |
| 01:29.16 | PMantis | Man, my desk was obnoxious when i was setting up a call center phone system for a Montreal company. |
| 01:29.18 | Ritalin- | what do you think of the gxp2k vs the spa941? |
| 01:29.38 | pdtmobile | X-Rob_: one of my guys is growing a lab slowly in his office. I walked in there yesterday and he has grown a cisco switch and a 601 with two sidecars since the last time I was in there |
| 01:29.46 | pdtmobile | they are multiplying |
| 01:29.57 | PMantis | I had 14 phones spread out all over my 12x10 room. heh |
| 01:29.58 | ManxPower | Grandstream has had several years to make decent firmware. They continue to fail. Enough said. |
| 01:30.00 | Strom_C | the phones are like rabbits in that regard |
| 01:30.18 | Strom_C | I have tons of phones in my apartment...lemme think |
| 01:30.19 | X-Rob_ | Ritalin-, handset qualtiy, the GXP sounds better and has better features. However, the SPA941 has a surprisingly good speakerphone |
| 01:30.21 | pdtmobile | ManxPower: isn't that the truth, but I guess you get what you pay for |
| 01:30.27 | *** join/#asterisk yanz7 (n=tysovka@ool-182f74a0.dyn.optonline.net) |
| 01:30.27 | Ritalin- | x-rob: have you had any kind of echo problems with the spa3k ? |
| 01:30.48 | *** join/#asterisk Samoied (n=Samoied@201.47.218.152) |
| 01:30.56 | pdtmobile | hows the seen thing work I can't remember |
| 01:31.05 | PMantis | file, are you attempting to make a change right now? |
| 01:31.19 | Strom_C | two cisco 7960s, two spa-2002s, an iaxy, a pap2, two tdm400s, three payphones, three nortel analog desk sets, and about a dozen Western Electric desk and wall sets |
| 01:31.27 | file | PMantis: I will soon, reading the SQL |
| 01:31.34 | pdtmobile | lol payphone |
| 01:31.38 | file | it's not... easy... |
| 01:31.44 | PMantis | file, Cool! thanks |
| 01:31.49 | hads | X-Rob_: Do you think the GXP2000 handset sounds better than the SPA941? |
| 01:31.51 | Strom_C | pdtmobile: THREE of them! |
| 01:31.53 | pdtmobile | select * from everything |
| 01:32.06 | PMantis | pdthome, no... |
| 01:32.08 | pdtmobile | hehhe |
| 01:32.15 | pdtmobile | kidding |
| 01:32.24 | PMantis | DELETE FROM IAXTEL_USERS |
| 01:32.31 | mtgh | Is there a C++ room on freenode, I need to figure out what a UCHAR is |
| 01:32.33 | PMantis | Oh, crap, forgot my WERE clause |
| 01:32.36 | PMantis | :) |
| 01:32.47 | pdtmobile | ya, everybody has to experience that at least once |
| 01:33.07 | PMantis | Too much here! |
| 01:33.18 | file | PMantis: ha |
| 01:33.19 | pdtmobile | backups |
| 01:33.27 | pdtmobile | snapshots are your friend |
| 01:33.33 | Qwell | mtgh: an unsigned char |
| 01:33.33 | PMantis | Yup |
| 01:33.49 | mtgh | qwell: can I msg you? |
| 01:33.53 | Qwell | no |
| 01:34.00 | pdtmobile | or just use transactions when you are doing manual deletes |
| 01:34.02 | PMantis | That's also why I rsync'd the whole server before I packaged it up for shipment to my client. |
| 01:34.05 | pdtmobile | on crap roll back roll back! |
| 01:34.11 | PMantis | if something breaks, i have a backup. |
| 01:34.27 | mtgh | I need to know how to create that datatype in C++ as a pointer |
| 01:34.28 | Qwell | mtgh: It's actually unicode char |
| 01:34.36 | X-Rob_ | ~book |
| 01:34.40 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:34.41 | yanz7 | hey everyone..whats a good service that works with asterisk ... |
| 01:34.46 | mtgh | I am looking for a place to get help on that as I don't know C++ and I have someone else's mess dropped in my lap |
| 01:35.00 | Qwell | yanz7: googletalk |
| 01:35.04 | pdtmobile | ~seen dlynes_home |
| 01:35.28 | jbot | dlynes_home is currently on #asterisk. Has said a total of 136 messages. Is idling for 22h 22m 7s, last said: 'heh'. |
| 01:35.28 | pdtmobile | how doe sthat work? |
| 01:35.28 | yanz7 | googletallk? |
| 01:35.28 | pdtmobile | seen dlynes_home |
| 01:35.28 | file | darn, problem |
| 01:35.28 | PMantis | yanz7, That's almost a religious answer... but we also need one other questions answered... Where do you need the DID to be from? |
| 01:35.28 | pdtmobile | crap |
| 01:35.28 | Qwell | file: there always is |
| 01:35.29 | pdtmobile | i suck |
| 01:35.29 | yanz7 | new york |
| 01:35.32 | pdtmobile | oh! it's just slow |
| 01:35.39 | yanz7 | i am just new at this |
| 01:35.43 | pdtmobile | now i duno which one worked :/ |
| 01:35.49 | yanz7 | wanted to try to set it but cant get it to work with vonage |
| 01:35.51 | Qwell | mtgh: http://www.ph.unimelb.edu.au/~ssk/kde/srcdoc/kjs/KJS__UChar.html |
| 01:36.02 | X-Rob_ | ~seen dlynes_home |
| 01:36.06 | jbot | dlynes_home is currently on #asterisk. Has said a total of 136 messages. Is idling for 22h 22m 45s, last said: 'heh'. |
| 01:36.07 | yanz7 | now i found out it doesnt really support it anyway |
| 01:36.07 | PMantis | yanz7, I have great luck ith ViaTalk, others have good luck with Broadvoice, voicepulse, etc. |
| 01:36.15 | *** part/#asterisk eDIsonxl (n=eDison@60-248-182-162.HINET-IP.hinet.net) |
| 01:36.19 | X-Rob_ | ~seen X-Rob |
| 01:36.20 | jbot | x-rob <n=rob@CPE-60-231-85-52.qld.bigpond.net.au> was last seen on IRC in channel #asterisk, 4d 3h 17m 38s ago, saying: '~centosbug'. |
| 01:36.21 | X-Rob_ | ~seen X-Rob_ |
| 01:36.22 | jbot | x-rob_ is currently on #asterisk (2d 1h 17m 27s). Has said a total of 20 messages. Is idling for 1s, last said: '~seen X-Rob_'. |
| 01:36.31 | PMantis | yanz7, I use multiple providers |
| 01:36.33 | Qwell | ~seen yourmom |
| 01:36.35 | jbot | yourmom <~RyanF@ip70-179-112-231.dc.dc.cox.net> was last seen on IRC in channel #tfp, 539d 23h 30m 51s ago, saying: ':)'. |
| 01:36.40 | X-Rob_ | hahahaha |
| 01:36.40 | Qwell | heh, oh |
| 01:36.46 | Strom_C | hahahahaha |
| 01:36.47 | X-Rob_ | *bahahaha* |
| 01:36.52 | X-Rob_ | OK |
| 01:36.53 | PMantis | Qwell, wow, didn't expect that |
| 01:36.55 | X-Rob_ | now that _is_ funny. |
| 01:37.00 | Qwell | PMantis: yeah, no kidding :p |
| 01:37.13 | PMantis | ~seen QwellsMom |
| 01:37.15 | jbot | PMantis: i haven't seen 'qwellsmom' |
| 01:37.15 | Strom_C | youmom is at cox.net |
| 01:37.25 | PMantis | ROFL |
| 01:37.33 | yanz7 | thanks pmantis |
| 01:37.34 | PMantis | Strom_C, That's *funny*! |
| 01:37.34 | X-Rob_ | ~seen QwellsMom, naked - this week |
| 01:37.36 | jbot | X-Rob_: i haven't seen 'qwellsmom, naked - this week' |
| 01:38.24 | Qwell | pwned |
| 01:38.33 | pdtmobile | lol |
| 01:38.46 | pdtmobile | look what I started |
| 01:38.48 | X-Rob_ | bah |
| 01:38.56 | X-Rob_ | ~lart pdt |
| 01:38.57 | Katty | this is completely off topic... |
| 01:38.59 | X-Rob_ | ~lart pdtmobile |
| 01:39.08 | Katty | but what's the difference between a mudslide and a white russian? |
| 01:39.24 | Katty | and don't tell me that one is fxo and one is fxs |
| 01:39.28 | file | ugh |
| 01:39.29 | file | Query OK, 42030 rows affected (2.06 sec) |
| 01:39.32 | PMantis | hah |
| 01:39.39 | X-Rob_ | A mudslide is usually caused by torrential rains |
| 01:39.42 | PMantis | file, Uh, oh |
| 01:39.48 | yanz7 | pmantis with viatalk do i have to get softphone to setup up with asterisk |
| 01:39.48 | Ritalin- | x-rob: have you had any kind of echo problems with the spa3000 ? |
| 01:39.49 | X-Rob_ | whilst an unusually white russian would proabbly be an albino |
| 01:39.51 | Katty | X-Rob_: freak. |
| 01:40.04 | orlock | and deforestation |
| 01:40.07 | X-Rob_ | Ritalin-, google spa3000 firmware echo |
| 01:40.08 | *** join/#asterisk bkervaski (n=bkervask@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 01:40.09 | Katty | where is hmmhesays when i need him )= |
| 01:40.12 | pdtmobile | mudslide has chocolate in it, white russian doesn't if I remember correclty |
| 01:40.17 | X-Rob_ | there's some firmware that sucks for echo |
| 01:40.22 | Katty | pdtmobile: kthx. |
| 01:40.23 | orlock | deforestation is an important part, as theres nothing holding the soil together anymore |
| 01:40.31 | X-Rob_ | except for the kittens |
| 01:40.42 | file | PMantis: almost done |
| 01:40.43 | bkervaski | Hi all. In my extensions.conf, If I have 4 IAX channels setup for FAX and I have them as priority 1,2,3,4, if one is busy, it should automatically goto the next one right? |
| 01:40.59 | PMantis | yanz7, ViaTalk does BYOD, but I opted to order a Linksys PAP2 anyhow,. Then they gave me the passwrd for my account, and the PAP2. So, Asterisk answers al calls directly, and I can route that to a zap channel, IAX phone, SIP phone, the PAP2, etc. |
| 01:41.04 | X-Rob_ | Man, I love how people come in here and are _so_ confused. |
| 01:41.07 | *** join/#asterisk eDIsonxl (n=eDIsonxl@60-248-182-162.HINET-IP.hinet.net) |
| 01:41.21 | PMantis | file, Awesome |
| 01:41.26 | Ritalin- | x-rob: ah, okay cool good tip thanks. I was having a bitch with getting it setup because the PSTN ring-thru wasn't documented. and changed firmware. may have to change it again. good to know |
| 01:41.34 | Strom_C | white russian is kahlua, milk, vodka |
| 01:41.56 | yanz7 | thanks pmatis ..now i have to decode this :) |
| 01:41.57 | file | actually, meh |
| 01:41.58 | Strom_C | holy god, i'm lagged to hell |
| 01:41.59 | X-Rob_ | I personally before a 'robsmix' |
| 01:42.07 | file | PMantis: try it now |
| 01:42.13 | X-Rob_ | which is 5 parts scotch, 2 icecubes, and 1 part pepsi max. |
| 01:42.17 | pdtmobile | Katty: i waswrong |
| 01:42.19 | PMantis | bkervaski, Priority is actually a line number... just like a line number in BASIC programming (anyone remember that??) |
| 01:42.20 | bkervaski | Hi all. In my extensions.conf, If I have 4 IAX channels setup for FAX and I have them as priority 1,2,3,4, if one is busy, it should automatically goto the next one right? |
| 01:42.30 | Katty | pdtmobile: enlighten me |
| 01:42.33 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:42.37 | pdtmobile | Katty: the difference is a mudslide has Bailey's in it the other doesn't |
| 01:42.50 | Katty | pdtmobile: do they both have vodka, tho? |
| 01:42.52 | pdtmobile | both have cream, vodka and kahlua |
| 01:42.56 | Katty | k |
| 01:43.03 | file | your peer isn't registered it looks like |
| 01:43.04 | Ritalin- | I love a good scotch on the rocks |
| 01:43.04 | pdtmobile | or milk instead of cream |
| 01:43.08 | pdtmobile | depends on who makes it |
| 01:43.25 | Katty | i'll have them hold the vodka and kahlua ;) |
| 01:43.30 | bkervaski | Hi all. In my extensions.conf, If I have 4 IAX channels setup for FAX and I have them as priority 1,2,3,4, if one is busy, it should automatically goto the next one right? |
| 01:43.33 | Katty | mew. |
| 01:43.43 | pdtmobile | damnit Strom_C you have me looking at payphones on ebay |
| 01:43.44 | Strom_C | what?! |
| 01:43.44 | file | there it is... |
| 01:43.55 | pdtmobile | i have always wanted one but i was like, hmm... ebay |
| 01:44.02 | PMantis | bkervaski, No. See my previous comment. |
| 01:44.08 | Strom_C | bkveraski: will you pleaste stop asking the same damned question every single minute? |
| 01:44.32 | pdtmobile | Strom_C: In my extensions.conf, If I have 4 IAX channels setup for FAX and I have them as priority 1,2,3,4, if one is busy, it should automatically goto the next one right? |
| 01:44.39 | pdtmobile | :) sorry couldn't help myself |
| 01:44.45 | Nugget | heh |
| 01:44.59 | Strom_C | pdtmobile: In my extensions.conf, If I have 4 IAX channels setup for FAX and I have them as priority 1,2,3,4, if one is busy, it should automatically goto the next one right? |
| 01:44.59 | file | Nugget'n'fries! |
| 01:45.02 | pdtmobile | bkervaski: thats what it should do |
| 01:45.04 | Katty | Nugget, deary, how're you? |
| 01:45.13 | Katty | Nugget: and will you come with me to get a mudslide? |
| 01:45.32 | file | PMantis: better? |
| 01:46.29 | Nugget | I'm grumpy tonight, but otherwise doing ok. |
| 01:46.31 | PMantis | file, It looks like it tries 's', then if that doesn't work, try the number... |
| 01:46.49 | Strom_C | ok, to hell with coffeeshop wifi...imgonna switch to my mobile phone |
| 01:46.49 | Strom_C | brb |
| 01:46.49 | file | PMantis: right now it dials them both, but I like your idea better |
| 01:46.56 | bkervaski | @pdtmobile: Thanks. |
| 01:46.59 | Katty | Nugget: grumpy? )= |
| 01:47.16 | PMantis | file, heh, cool |
| 01:47.17 | yanz7 | let me ask you one more questoin if you dont mind .....if i have one line with a number and 3 ppl call i get busy signal for the third person on that number and if i get an extra number how do i switch that 3rd person to the second number if they call the same number.... |
| 01:47.27 | PMantis | file, But, yea... it seemed to work. |
| 01:47.29 | yanz7 | sorry for a long question |
| 01:47.38 | PMantis | I was simply trying to test both numbers before I commented. |
| 01:47.54 | pdtmobile | yanz7: i assume you are talking about analog lines? |
| 01:47.58 | file | PMantis: well lemme change it again... then you can test once more |
| 01:48.01 | *** join/#asterisk calvinhp (n=calvin@cpe-65-29-88-222.indy.res.rr.com) |
| 01:48.19 | yanz7 | i guess so |
| 01:48.20 | PMantis | file, Currently works, but go ahead. |
| 01:48.40 | pdtmobile | yanz7: you need to talk to the phone company about getting a rollover |
| 01:48.50 | Strom_C | wow |
| 01:48.52 | yanz7 | ok so thats done on the phone companies side |
| 01:48.59 | Strom_C | I have never had GPRS actually be faster than coffeeeshop wifi |
| 01:49.06 | pdtmobile | yanz7: yeppers |
| 01:49.10 | yanz7 | but what about voip |
| 01:49.11 | pdtmobile | Strom_C: where you located? |
| 01:49.13 | yanz7 | and asterisk |
| 01:49.14 | PMantis | Strom_C, LOL wow |
| 01:49.42 | Strom_C | I'm in Northridge right now |
| 01:49.51 | Snake-Eyes | Does Asterisk take advantage of 64 bit archtecture? |
| 01:50.02 | file | PMantis: done |
| 01:50.09 | pdtmobile | I run across that from time to time here (Nashville, TN) my 3G with Sprint is better than wifi cause a bunch of idiots bring their laptops and play WoW eating up every single bit of bandwidth |
| 01:50.31 | Strom_C | well there is some guy on a laptop in the corner, but i dont know what hes doing |
| 01:50.33 | PMantis | lol |
| 01:50.37 | PMantis | file, testing |
| 01:50.40 | Strom_C | he looks too professional to be playing WoW |
| 01:50.41 | pdtmobile | yanz7: so you have a voip provider? |
| 01:50.43 | pdtmobile | lol |
| 01:50.48 | Strom_C | maybe he's bittorrenting |
| 01:50.48 | yanz7 | not yet |
| 01:50.52 | yanz7 | just trying to figure this out |
| 01:50.53 | pdtmobile | you never know, it's the sneaky ones |
| 01:50.56 | yanz7 | i have vonage |
| 01:50.57 | Strom_C | or maybe this coffeeshop just blows |
| 01:50.57 | PMantis | file, I'm using DISA out to another PBX, and dialing back to mine |
| 01:51.05 | yanz7 | but i guess it doesnt work well with asterisk |
| 01:51.07 | file | the excitement never ends |
| 01:51.14 | pdtmobile | yanz7: not really |
| 01:51.35 | yanz7 | so i am gonna check out another provider for this |
| 01:51.48 | Ritalin- | are there better fxs->voip gateways over the spa3k? for <$100 ? |
| 01:51.52 | pdtmobile | yanz7: just check out how many concurrent calls the provider allows when you switch to a new one |
| 01:52.26 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 01:52.32 | yanz7 | thanks for the info |
| 01:52.35 | *** join/#asterisk s0lid (n=s0lid@61.28.161.132) |
| 01:52.44 | PMantis | file, works great! |
| 01:52.59 | PMantis | file, thank you! |
| 01:53.01 | yanz7 | pmantis do u know how many concurrent connection viatalk support |
| 01:53.04 | file | not bad for a box with 822 active registrations |
| 01:53.07 | pdtmobile | holy crap faxgetty is spamming me |
| 01:53.12 | yanz7 | vt_unlimited plan |
| 01:53.14 | calvinhp | anyone ever experience a growing lag time during a meetme conference? We had 2 calls using g729a and 2 channels using ulaw |
| 01:53.14 | PMantis | yanz7, Yes... 2 in, 2 out |
| 01:53.27 | yanz7 | is that at the same time |
| 01:53.30 | calvinhp | after 30 minutes the lag was at least 2 seconds |
| 01:53.35 | PMantis | yanz7, So, a total of 4, but not 4 in ONE direction |
| 01:53.43 | calvinhp | all parties hung up and called back and it was fine |
| 01:53.44 | yanz7 | gotcha thanks |
| 01:53.53 | calvinhp | until about 30 minutes when it had grown back again |
| 01:53.55 | X-Rob_ | calvinhp, yes, that's an issue with timing |
| 01:54.02 | X-Rob_ | use ztdummy with use-rtc enabled |
| 01:54.12 | calvinhp | X-Rob_: I have a TDM04B card in the machine |
| 01:54.16 | X-Rob_ | and make sure you're running 1.2.9.1 |
| 01:54.24 | X-Rob_ | calvinhp, bugger. |
| 01:54.26 | PMantis | yanz7, No, if your needs are mostly inbound, then voicepulse allows for 4 inbound, and unlimited? outbound. but you pay by the minute for all outbound. |
| 01:54.34 | X-Rob_ | in that case, you're screwed. |
| 01:54.39 | PMantis | yanz7, Sorry.. No=Now |
| 01:54.44 | X-Rob_ | use anotehr machine without hardware |
| 01:54.50 | calvinhp | X-Rob_: the only thing that is not normal about my setup is that I'm using FreeBSD |
| 01:55.06 | X-Rob_ | calvinhp, *shrug* no idea |
| 01:55.07 | calvinhp | could it be an issue with the zaptel drivers? |
| 01:55.08 | pdtmobile | so i have been having ALOT of problems with my PRI freaking out with 1.2.9.1 where all my inbound numbers no longer make it in and you get a fast busy from the telco |
| 01:55.09 | X-Rob_ | It's a timing issue |
| 01:55.09 | yanz7 | i see..also do i have to get the pap2 device |
| 01:55.11 | PMantis | yanz7, It's for reasons like this that I use multiple VoIP providers... |
| 01:55.19 | pdtmobile | outbound still typically works for awhile, then it goes dead as well |
| 01:55.27 | PMantis | yanz7, No, you don't have to. The offer BYOD |
| 01:55.29 | yanz7 | that makes sence |
| 01:55.33 | pdtmobile | any ideas? |
| 01:55.51 | yanz7 | but then i would need regular acount and a soft phone |
| 01:55.55 | pdtmobile | an asterisk restart makes everything "happy" again |
| 01:55.55 | yanz7 | to be able setup asterisk |
| 01:56.14 | *** join/#asterisk Strom_C (n=strom@gateway.digium.com) |
| 01:56.29 | PMantis | yanz7, But, you can order the PAP2, then call them for the password, and use asterisk instead. Then will also give you the password to the PAP2, so you can "hack" it, and make it register to your asterisk server, instead of their service. |
| 01:56.30 | Strom_C | if its not crappy coffeeshop wifi, it's screen going bonkers |
| 01:56.44 | Strom_C | PMantis: which company? |
| 01:56.44 | pdtmobile | Strom_C: maybe you should give up |
| 01:56.51 | pdtmobile | it's a sign |
| 01:56.52 | Strom_C | pdtmobile: yes perhaos I should |
| 01:56.53 | PMantis | Strom_C, ViaTalk |
| 01:56.56 | Strom_C | er, perhaps |
| 01:56.59 | Strom_C | ah, interesting |
| 01:57.09 | Strom_C | i thought you were going to say "vonage" |
| 01:57.12 | file | Strom_C: you are silly for IRCing from Digeum |
| 01:57.17 | Strom_C | to which I would have responded "in which universe" |
| 01:57.22 | Strom_C | file: yeah true |
| 01:57.32 | Qwell | file: I give it...4 days |
| 01:57.36 | PMantis | Strom_C, They'd be perfect for me if the would just allow for breaking the mold... more than 2 in 2 out. |
| 01:57.56 | Strom_C | heh right? |
| 01:58.05 | pdtmobile | so no ideas on the PRI freak out? |
| 01:58.13 | Qwell | <file> Qwell: for?! |
| 01:58.18 | Qwell | file: for you to be doing the same :D |
| 01:58.22 | Strom_C | im disappointed that the PAP2T has no vertical stand thing and has green lights instead of blue |
| 01:58.22 | PMantis | Strom_C, But, I'm using them anyhow, cause it's still the best I found... *and* I purchased a whole year of unlimited when, they gave me an additional year for free. |
| 01:58.26 | file | lol |
| 01:58.31 | *** join/#asterisk gandhijee (n=gandhije@c-24-147-213-182.hsd1.ma.comcast.net) |
| 01:58.43 | Strom_C | PMantis: how much did a whole year of unlimited run you |
| 01:58.58 | PMantis | Strom_C, So, 2 years of unlimited US and Canada calling for $200?? I jumped on it. |
| 01:59.09 | pdtmobile | PMantis: run that by me again |
| 01:59.14 | PMantis | lol |
| 01:59.18 | PMantis | pdthome, Which part? |
| 01:59.23 | pdtmobile | the 200 dollar part |
| 01:59.25 | pdtmobile | :) |
| 01:59.28 | yanz7 | so then i can just order the do it yourself option get the sip password and instead of buying pap2 device i can set that up in asterisk |
| 01:59.34 | Strom_C | thats actually a good deal, assuming you talk for more than $8 worth of minutes per month |
| 02:00.08 | pdtmobile | i haven't heard of those guys before, pretty happy with it? |
| 02:00.12 | PMantis | pdtmobile, It was a $199 for an unlimited 1 year plan... but they had a promotion when i signed up... for another year free. |
| 02:00.31 | PMantis | Strom_C, Yeah! and we regularly hit 2500 mins a month of outbound. |
| 02:00.32 | *** join/#asterisk file2 (n=jcolp@gateway.digium.com) |
| 02:00.35 | file2 | omg hi |
| 02:00.40 | Strom_C | was it free, or FREEEEEEEEEEEEEEEEEEEEEEE? |
| 02:00.49 | PMantis | ha |
| 02:01.02 | pdtmobile | looks like they are doing 3 months free now |
| 02:01.11 | pdtmobile | 3 months for 12 month contract, 6 for 24 |
| 02:01.12 | PMantis | still NB |
| 02:01.13 | Qwell | file2: They said no |
| 02:01.24 | file | Qwell: that sux0rz |
| 02:01.26 | Qwell | telecomjoshvoxmart FAILS |
| 02:01.31 | Strom_C | theres a mattress store chain here in southern california that clains "we'll beat anyone's advertised price or your mattress is FREEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEE!!!!" |
| 02:01.36 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 02:01.42 | Qwell | 800-freeeee |
| 02:01.44 | Qwell | :D |
| 02:01.46 | Strom_C | yes |
| 02:01.51 | Strom_C | 1-800-FREEEEE |
| 02:02.08 | PMantis | Strom_C, Haha, well, doens't "free' beat the other persons' price? |
| 02:02.10 | Strom_C | its a damned good adcampaign |
| 02:02.23 | Qwell | Strom_C: Now...name the company |
| 02:02.24 | Qwell | :P |
| 02:02.27 | Qwell | because I can't |
| 02:02.28 | Strom_C | well, in reality, they never need to give away a free mattress |
| 02:02.30 | Strom_C | sit n sleep |
| 02:02.33 | pdtmobile | 2 in 2 out isn't bad |
| 02:02.35 | yanz7 | pmantis:so then i can just order the do it yourself option get the sip password and instead of buying pap2 device i can set that up in asterisk |
| 02:02.36 | Qwell | ahh |
| 02:02.49 | Strom_C | they just need to undercut the other guy by like $30 |
| 02:02.51 | PMantis | yanz7, Exactly. |
| 02:03.00 | Qwell | the last matress I bought... |
| 02:03.06 | Qwell | was from a lady, out of a storage unit |
| 02:03.14 | Strom_C | I bought my mattress at IKEA |
| 02:03.20 | PMantis | yanz7, But like I said, I opted for the PAP2, then "cleaned up" after the preconfig, and stuck my asterisk box in the middle. |
| 02:03.21 | Qwell | oej would be proud |
| 02:03.26 | file | Strom_C: are you an IKEA whore? |
| 02:03.37 | PMantis | yanz7, Now I use the PAP2 when I travel. 2 lines.. one for me, one for my wife. |
| 02:03.38 | pdtmobile | lol |
| 02:03.41 | Strom_C | file: I took Mark there when I was in the bay area with him once |
| 02:03.52 | file | dangerous |
| 02:04.12 | Strom_C | he bought a lamp with a little gel filter thing that made it oroject asterisks on the wall |
| 02:04.27 | Qwell | Strom_C: nice |
| 02:04.28 | file | ha |
| 02:04.28 | yanz7 | pmantis:meaning that u can take that box anywhere connect to regular phone to it and asterisk and be online anywhere ..like you are your own provider |
| 02:04.31 | Strom_C | this was a year ago. The thing is still sitting half-assembled in his study |
| 02:04.43 | PMantis | yanz7, Yes |
| 02:04.51 | yanz7 | thanks for the explanation!! |
| 02:04.54 | file | half assembled? ooh, better then usual |
| 02:05.16 | Qwell | the man can build a robot, and write a pbx... |
| 02:05.18 | Strom_C | hah |
| 02:05.26 | Qwell | typical computer geek.. :p |
| 02:05.35 | Strom_C | I vote "easily distracted" |
| 02:05.48 | pdtmobile | man... I hate that |
| 02:05.49 | yanz7 | pmantis: just one more question i think you said that you have different providers to be able to have more incoming/outgoing then how do u i guess its called callhunting if you have different numbers from different providers |
| 02:05.53 | file | poor robot needs to be upgraded |
| 02:05.54 | yanz7 | or thats not possible thne |
| 02:05.56 | pdtmobile | I am actually going to finish at least one of my projects |
| 02:05.57 | PMantis | yanz7, You're welcome. Just beware that this takes some knowledge of the SIP protocol to get that working. It's easier to order w/o a device, use asterisk.. then travel with an "IAXy" |
| 02:05.59 | pdtmobile | one of these days |
| 02:06.01 | Qwell | file: with a lamp perhaps? |
| 02:06.09 | file | yessss |
| 02:06.18 | file | and festival! |
| 02:06.28 | Qwell | That would be very very bad |
| 02:06.31 | Strom_C | oh christ, not festival |
| 02:06.45 | Strom_C | although...it is a robot.... |
| 02:06.56 | Strom_C | HELLO I AM FESTIVAL THE ROBOT |
| 02:07.00 | pdtmobile | lol |
| 02:07.06 | Qwell | I could just imagine there being a chat form on markocam... |
| 02:07.15 | Strom_C | "make the robot talk" |
| 02:07.30 | Qwell | yes...very very bad |
| 02:07.31 | Strom_C | so mark's on a very important call and the robot starts going "fuck shit piss" |
| 02:07.36 | PMantis | yanz7, Well, I use viatalk for our "main" number (outbound)... then I bought a few DID from other providers... one for business, one for drirect ring to my extension, for my wife, etc. I can't get one provider to pass to another when I've used up all my allocated connections. |
| 02:07.37 | Qwell | Strom_C: exactly |
| 02:07.52 | file | Strom_C: 'tsk 'tsk |
| 02:07.56 | rene- | thats why i have a cursing parrot, it keeps going until it dies |
| 02:08.03 | file | oh oh oh with lumenvox so he could say shut up... |
| 02:08.21 | Qwell | heh |
| 02:08.21 | file | yessss |
| 02:08.28 | Qwell | "STFU robot!" |
| 02:08.32 | yanz7 | pmantis: i see...but its posible to get more numbers from lets say viatalk and they will be able to call hunt |
| 02:08.40 | yanz7 | pmantis again thanks a lot |
| 02:08.41 | file | I'd have to run the SRE on another box, the robot is no where near powerful enough |
| 02:08.48 | rene- | file: have you used lumenbox? does it works well |
| 02:08.52 | rene- | lumenbox |
| 02:08.55 | PMantis | yanz7, NP... |
| 02:08.56 | rene- | vox shit |
| 02:09.01 | file | rene-: yes, it does |
| 02:10.03 | rene- | file: what do you use for tts? cepstral? |
| 02:10.27 | file | TTS is completely separate, but I think Cepstral is available |
| 02:10.31 | rene- | i know |
| 02:10.35 | rene- | ah |
| 02:10.39 | rene- | do you mean lumen sells a tts also? |
| 02:10.43 | file | no |
| 02:10.46 | rene- | ok |
| 02:11.22 | Strom_C | what the crap? lumenvox has no "buy" button. how much does their product cost? |
| 02:11.29 | rene- | it was like 400/channel, |
| 02:11.31 | Qwell | Strom_C: free |
| 02:11.39 | file | lalalala I am in engineering lalalala |
| 02:11.45 | Strom_C | Qwell: ORLY? |
| 02:11.47 | Qwell | no |
| 02:11.50 | Qwell | but ask file :p |
| 02:11.57 | Strom_C | pfft |
| 02:12.15 | file | they're selling something for regular developers that isn't expensive... |
| 02:12.23 | Qwell | "regular"? |
| 02:12.30 | file | regular ol' joes |
| 02:12.42 | rene- | Qwell: meaning those that dont sing lalala i am in engineering |
| 02:12.42 | Strom_C | what about regular ol' stroms |
| 02:12.52 | file | Strom_C: you could if you wanted! |
| 02:12.55 | pdtmobile | I would assume it's per concurrent channel? |
| 02:13.13 | rene- | that is what the lumen guy told me |
| 02:13.28 | file | per concurrent speech port... but you don't have to have a speech port open for the entire call |
| 02:13.43 | file | if you only need to use it at a main menu, then fine... |
| 02:13.46 | pdtmobile | so just like where you would have waitexten |
| 02:14.42 | Strom_C | I had hoped to get some work done, but instead im just using irc |
| 02:14.53 | pdtmobile | i am about 50/50 so far on that |
| 02:14.56 | Strom_C | ah, irc |
| 02:14.56 | file | Strom_C: awww we love you |
| 02:15.00 | pdtmobile | was hoping for 90/10 |
| 02:15.07 | Strom_C | <3 |
| 02:15.20 | file2 | mooooooooo |
| 02:15.41 | file | Strom_C: logged in enough? |
| 02:15.49 | Strom_C | ?? |
| 02:16.01 | Strom_C | do i have like five thousand sessions open? |
| 02:16.11 | file | nah :P |
| 02:16.15 | file | 4999 |
| 02:16.21 | Strom_C | yow |
| 02:16.27 | Katty | meesha. |
| 02:16.39 | Katty | Strom_C: you should be chased with a bat. |
| 02:17.00 | Strom_C | it only claims i have one instance of bash open when i run ps |
| 02:17.35 | Qwell | Strom_C: it forked...repeatedly |
| 02:17.55 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 02:18.02 | Strom_C | oh well, what do i know |
| 02:18.05 | Strom_C | im a linux nub |
| 02:18.07 | Strom_C | i know phones |
| 02:18.13 | Strom_C | not linux :) |
| 02:19.01 | PMantis | Anyony know if a Polycom IP501 can use 1 line appearance as internal, then another line appearance to register with an external server using NAT? |
| 02:19.15 | pdtmobile | polycom and nat are not the best pair |
| 02:19.20 | Strom_C | PMantis: I dont see why not |
| 02:19.39 | pdtmobile | but yes, you can specify a different server for every line on the phone |
| 02:20.13 | PMantis | pdtmobile, I've done that, but the SIP register request comes from my internal IP... so the other * server can't send a response. |
| 02:20.43 | PMantis | Then there's the RTP ports, but I'm not there yet... |
| 02:21.01 | pdtmobile | back to the nat makes polycom cry comment :) |
| 02:21.11 | PMantis | heh\ |
| 02:21.28 | pdtmobile | leme dig out a config real quick |
| 02:21.53 | *** join/#asterisk segfault_rb (n=segfault@201-34-84-211.paemt705.dsl.brasiltelecom.net.br) |
| 02:22.12 | PMantis | pdtmobile, k |
| 02:22.30 | PMantis | pdtmobile, I'm not using XML via FTP fr this phone yet, BTW |
| 02:22.44 | rob0 | Bats don't chase people. They chase bugs. |
| 02:22.44 | pdtmobile | oh, thats the only way to fly |
| 02:22.47 | Katty | guns don't kill people |
| 02:22.49 | Katty | people kill people |
| 02:22.53 | Katty | but i think the gun helps. |
| 02:22.55 | Katty | just a bit |
| 02:23.00 | pdtmobile | i honestly haven't ever used the web interface, never seen it :) |
| 02:23.03 | segfault_rb | hi people |
| 02:23.04 | PMantis | pdtmobile, I did that for the call center I setup... but mine is manual config ATM |
| 02:23.13 | pdtmobile | cant help ya on that |
| 02:23.50 | PMantis | pdtmobile, Ok, but I'll swap to FTP/XML in a couple minutes if you can give me an viable solution. :-) |
| 02:23.55 | pdtmobile | lol |
| 02:23.55 | segfault_rb | i haven't found anything about it on the net, so i've come here to know something about it |
| 02:24.03 | pdtmobile | leme dig around then I might have something here |
| 02:24.34 | pdtmobile | am i the only one that does aix2 show peers all the time? |
| 02:24.34 | PMantis | pdtmobile, okie. I've just held back, 'cause my * box is currently Gentoo... and I didn't feel like compiling vsftpd |
| 02:24.50 | pdtmobile | ya gentoo is damn good at the compiling |
| 02:24.52 | rob0 | Oh, chased WITH a bat, not BY a bat ... I missed that |
| 02:25.13 | segfault_rb | has anyone used asterisk compiled with hoard [http://www.hoard.org] memory allocator ? any production server made this way ?? |
| 02:25.21 | PMantis | pdtmobile, True, I'm just over with my Gentoo phase.. wanna replace them all with Ubuntu. |
| 02:25.51 | segfault_rb | PMantis: please don't do it :-) |
| 02:26.02 | PMantis | segfault_rb, why? |
| 02:26.28 | pdtmobile | CentOS is what I use for all mine |
| 02:26.28 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 02:26.29 | segfault_rb | PMantis: as a desktop [K|U]buntu is great, but not as a server |
| 02:26.36 | pdtmobile | got too used to RHES at my last job |
| 02:26.54 | PMantis | segfault_rb, Ubuntu 6.06 LTS is nice for a server! :) |
| 02:26.58 | wwalker | Question about complex call transfer scenario - http://rafb.net/paste/results/H9yvDk32.html |
| 02:27.21 | *** join/#asterisk CodyC (n=cody@ppp-70-251-189-238.dsl.austtx.swbell.net) |
| 02:27.38 | *** join/#asterisk calvinhp (n=calvin@cpe-65-29-88-222.indy.res.rr.com) |
| 02:28.52 | segfault_rb | PMantis: say what you want, but Ubuntu is not "tried&tortured" as Debian packages |
| 02:28.58 | pdtmobile | PMantis: i can't find all my config crap :( |
| 02:30.05 | *** join/#asterisk phalacee (n=Sunforge@202.3.110.33) |
| 02:30.16 | PMantis | wwalker, I'm not specificly sure about the phone, but I know that's easily doable in asterisk. The phone only needs a transfer button. |
| 02:30.58 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:31.03 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:31.35 | wwalker | PMantis: what search terms? |
| 02:31.42 | segfault_rb | PMantis: changing the subject, have you ever done something like what i'm talking about ? |
| 02:32.13 | PMantis | segfault_rb, Something like what, exactly? |
| 02:32.46 | segfault_rb | compiling and running asterisk with hoard [http://www.hoard.org] memory allocator ? any production server made this way ?? |
| 02:32.49 | PMantis | wwalker, I'd just lookup the specs on the phone itself. If it's SIp based, and has a transfer button, you should be good. |
| 02:33.07 | PMantis | segfault_rb, oh.. no. Never tried hoard |
| 02:33.15 | wwalker | right, what do I look up on voipinfo to find how to do it in asterisk? |
| 02:33.29 | wwalker | ti's SIP and has a transfer button |
| 02:33.44 | wwalker | Does Asterisk just do that by default? |
| 02:33.53 | pdtmobile | wwalker: it should "just work" |
| 02:33.57 | segfault_rb | digium lists have nothing about it, so i came here :-) |
| 02:34.09 | PMantis | wwalker, Then as long as each phone has the abilit to directly dial every other phone, then Asterisk should allow that "out-of-the-box". |
| 02:34.27 | wwalker | Thanks guys |
| 02:34.34 | PMantis | wwalker, NP! |
| 02:35.04 | PMantis | pdtmobile, "just work" as long as the contexts of each phone allows them to dial each otehr. :) |
| 02:35.13 | pdtmobile | granted |
| 02:35.26 | PMantis | But I know.. thats assumed. :) |
| 02:35.30 | pdtmobile | was assuming he had all of his agents in the same contxt |
| 02:35.31 | segfault_rb | PMantis: it's on Solaris, but you'll get the idea: http://developers.sun.com/solaris/articles/multiproc/multiproc.html |
| 02:36.12 | Qwell | umm |
| 02:36.17 | Qwell | segfault_rb: Why not just use mtmalloc? |
| 02:36.52 | PMantis | Qwell, http://www.hoard.org says that mtmalloc isnt' as good. |
| 02:37.22 | file | ha, more kitkat bars made of... pure chocolate! |
| 02:37.44 | Qwell | file: eww |
| 02:37.57 | Qwell | That completely defeats the purpose |
| 02:38.15 | segfault_rb | Qwell: i came here for enlightment :-) |
| 02:38.58 | pdtmobile | segfault_rb: my assumption would be that it hasn't become such an issue that a new allocator is needed |
| 02:39.13 | file | enlightenment is available for those who are pure and free |
| 02:39.23 | segfault_rb | Qwell: i'm pretty new to asterisk, i'll thank you for any tip |
| 02:39.34 | PMantis | file, so is XFCU, Gnome, KDE. ;-) |
| 02:39.55 | pdtmobile | PMantis: wow, REALLY bad joke :) |
| 02:40.01 | PMantis | file, ok, I'm cnfused. XFCU is my credit union.. |
| 02:40.12 | segfault_rb | PMantis: XFCE ? |
| 02:40.24 | PMantis | pdtmobile, I'm the king of corny jokes. |
| 02:40.26 | PMantis | :) |
| 02:41.31 | file | king... queen... |
| 02:41.41 | PMantis | hey, hey |
| 02:42.15 | file | keep it gay, keep it gay, keep it gay |
| 02:42.55 | segfault_rb | file: ? |
| 02:43.13 | *** part/#asterisk korihor (n=humberto@200.35.213.46) |
| 02:44.55 | Strom_C | gay gay gay! |
| 02:45.03 | hohum | straight straight straight! |
| 02:45.10 | hohum | bi! bi! bi! |
| 02:45.13 | pdtmobile | purple |
| 02:46.51 | rene- | are those who are advocating for more gay, merely looking for a bigger pool for choosing a mate? |
| 02:47.14 | Strom_C | well, sure |
| 02:47.31 | Strom_C | anything for more potential mates :) |
| 02:47.49 | coppice | rene- did you meet your mate in a pool? I met mine at work. |
| 02:47.50 | rene- | fair enough |
| 02:48.16 | segfault_rb | rene-: lesser competition may result in a bigger chance of success :-) |
| 02:49.12 | rene- | fair answers |
| 02:49.25 | coppice | segfault_rb: do you really want to win by default? |
| 02:49.33 | rene- | coppice: no not at a pool |
| 02:49.41 | rene- | coppice: and heh |
| 02:49.57 | pdtmobile | he is shooting for the desert island mentality |
| 02:50.16 | *** join/#asterisk Coeus (n=Coeus@ip24-255-125-43.dc.dc.cox.net) |
| 02:50.18 | *** part/#asterisk phalacee (n=Sunforge@202.3.110.33) |
| 02:50.54 | segfault_rb | rene-: not every single time, obviously |
| 02:51.26 | coppice | you mean like "if you were the last man on earth, you would definitely be the last man on earth"? :-) |
| 02:51.49 | pdtmobile | :) |
| 02:52.04 | rob0 | I met mine through a dialup BBS network |
| 02:52.11 | pdtmobile | oh look at that the last man, guess i'm asexual |
| 02:52.14 | rob0 | (mate) |
| 02:53.05 | rene- | my plants change sex according to the environment |
| 02:53.29 | pdtmobile | frat party, pool party, technology company |
| 02:54.09 | pdtmobile | i feel sorry for most women that work at tech companies, at least around here |
| 02:54.14 | pdtmobile | there is like 1 or two per company |
| 02:54.26 | pdtmobile | i am sure they have to beat dorks off with a stick |
| 02:54.53 | rene- | pdtmobile: when they are surrounded by so many males and getting all the attention they get because of that i think they do pretty well |
| 02:55.08 | pdtmobile | well thats if any got the courage to talk to her :) |
| 02:56.03 | rene- | hah thats true |
| 02:58.15 | segfault_rb | pdtmobile: it remembers me my first semester at university, where the only woman in my class wasn't a student :-( |
| 02:58.24 | pdtmobile | hehe |
| 02:58.43 | pdtmobile | ya, i think there were only one or two girls that spoke my language in my CS department |
| 02:58.52 | pdtmobile | the rest were oriental and didn't speak but like three words of english |
| 02:58.53 | file | was your language C? |
| 02:59.15 | pdtmobile | c++ |
| 02:59.24 | Ritalin- | I'm going to write a language called filthy |
| 02:59.34 | pdtmobile | Ritalin-: there probably is one already |
| 02:59.41 | PMantis | lol |
| 02:59.43 | pdtmobile | there are a bazillion of them |
| 03:00.12 | pdtmobile | anybody in here mess with building rpm packages? |
| 03:01.11 | pdtmobile | for some reason building spec files is my least favorite thing in the world |
| 03:01.54 | russellb | jcollie in #asterisk-dev builds some for asterisk, i think |
| 03:02.46 | segfault_rb | Qwell: back to hoard, i'm using linux, not solaris, that's another reason i'm asking about it |
| 03:02.48 | pdtmobile | i don't know what my problem is with them, it's not really hard... just sucks, it's like cleaning the bathroom, it has to be done sometimes but why do I have to do it ;) |
| 03:03.26 | russellb | in theory, for asterisk, "make rpm" should do it for you |
| 03:03.27 | pdtmobile | that reminds me, i'm not in asterisk-dev |
| 03:03.48 | pdtmobile | i love the topic in there :) |
| 03:04.03 | russellb | yes, file does a nice job updating the topic :) |
| 03:04.16 | file | I try |
| 03:04.19 | russellb | we get rather silly in that channel. |
| 03:04.23 | Qwell | file: I'm about to update you |
| 03:04.29 | file | eep |
| 03:04.33 | pdtmobile | that sounds dirty |
| 03:04.41 | Qwell | pdtmobile: for good reason |
| 03:05.22 | Qwell | http://uuoc.com/1506 |
| 03:05.23 | Qwell | see? |
| 03:05.43 | pdtmobile | lol |
| 03:06.00 | pdtmobile | that reminds me of the biggest geek joke in college |
| 03:06.09 | pdtmobile | finger <insert girls name here on edu server> |
| 03:06.12 | rene- | i dont want to break the mood but i have a technical question, the manager interface, does it sends events as they happen or does it queues them some how |
| 03:06.35 | russellb | rene-: queues them i believe |
| 03:06.40 | russellb | but you can assume that it's pretty instant |
| 03:06.43 | file | jbot: sex? |
| 03:06.46 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/ |
| 03:06.46 | pdtmobile | people wouldn't give that a rest, it's when they started fingering the professors that I started to worry |
| 03:06.56 | coppice | pdtmobile: you shouldn't study in these backwater universities in China, just to save money. you'll find at their prestige universities costs are still low, but the english standards are better |
| 03:07.33 | pdtmobile | damn i knew I got swindled |
| 03:08.08 | rene- | coppice: you live in HK right? |
| 03:08.16 | rene- | are you origiinally from there also? |
| 03:08.20 | coppice | the UK's university of nottingham now has a campus in china, too. cheaper than studying at the campus in the UK |
| 03:08.33 | orlock | coppiceL i thought chinese prestige universities were all australian |
| 03:08.34 | coppice | me no chinky. me engrishman |
| 03:08.38 | rene- | heh |
| 03:10.47 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
| 03:11.30 | *** join/#asterisk alexns (n=ibtek04@static-acs-24-154-114-15.zoominternet.net) |
| 03:12.20 | coppice | the nottingham campus in china actually looks like the one in china - looks like a typical case of chinese copying :-) |
| 03:12.37 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 03:12.39 | coppice | s/one in china/one in the UK |
| 03:13.12 | pdtmobile | i love vmware server, makes life much better |
| 03:13.17 | segfault_rb | well, seems i have to get my hands dirty on this :-) |
| 03:13.19 | *** join/#asterisk mosty (n=mostynm@60-241-198-194.static.tpgi.com.au) |
| 03:13.27 | tengulre11 | hi,all |
| 03:13.45 | segfault_rb | anyway, thanks for the good laughs, creatures |
| 03:13.47 | pdtmobile | segfault_rb: i think you are on your own, you could ask in asterisk-dev |
| 03:13.57 | pdtmobile | somebody may have toyed with the idea |
| 03:13.58 | tengulre11 | anybody know where have IAX or SIP stack library? (WINDOWS PLATFORM) |
| 03:14.05 | segfault_rb | i'll do it, thx |
| 03:14.45 | Strom_C | i can sing that whole song |
| 03:15.10 | pdtmobile | i never can remember the one part about 7 o'clock |
| 03:15.25 | Strom_C | six o'clock tv hour dont get caught in foreign towers |
| 03:15.31 | Strom_C | slash and burn return listen to yourself churn |
| 03:15.31 | Strom_C | etc |
| 03:15.33 | pdtmobile | oh ya, that too |
| 03:15.42 | pdtmobile | hehe, see how well i remember it |
| 03:16.09 | tengulre11 | who can help me?? |
| 03:16.28 | pdtmobile | tengulre11: what do you mean |
| 03:17.36 | tengulre11 | pdtmobile: do u know where have IAX or SIP stack library (windows platform)? |
| 03:18.02 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-116-157.dsl.chcgil.sbcglobal.net) |
| 03:18.04 | pdtmobile | are you trying to run asterisk on windows? |
| 03:18.17 | mosty | is there a sound player for linux that supports g.711 ulaw? |
| 03:18.39 | rob0 | There are IAX and SIP *clients* for Windows ... |
| 03:18.42 | tengulre11 | pdtmobile: I want running the client on windows. |
| 03:19.01 | Flauto | anyone here is using trixtel.com for toll free termination? |
| 03:19.23 | tengulre11 | rob0: where? |
| 03:19.38 | pdtmobile | http://www.voip-info.org/wiki/view/VOIP+Phones#SoftPhones |
| 03:20.09 | pdtmobile | it may take you a week to find one you like, but at least you have some choices |
| 03:21.36 | tengulre11 | pdtmobile: yes! thank you! |
| 03:21.43 | pdtmobile | tengulre11: no problem |
| 03:22.28 | pdtmobile | i sure hope this pbx replacement fixes my PRI problem |
| 03:22.31 | Flauto | is trix around today? |
| 03:22.35 | pdtmobile | otherwise I quit life |
| 03:23.44 | pdtmobile | between this and another pbx core dumping it's not been a good week |
| 03:24.54 | pdtmobile | wow, thats a niche for ya |
| 03:24.57 | rob0 | (considering installing * on /dev/laptop for that) |
| 03:26.04 | pdtmobile | so you can use /dev/dsp on /dev/laptop ;) |
| 03:26.19 | rob0 | yeah |
| 03:27.08 | coppice | symlinks are your fiend |
| 03:27.25 | PMantis | Does * have a little gui for softphone use? |
| 03:27.35 | Flauto | is anyone here using trxtel for tollfree termination |
| 03:27.45 | mosty | can anyone recommend a util for linux that can convert wav files to g.711 ulaw? |
| 03:27.51 | mosty | and back? |
| 03:27.53 | pdtmobile | sox |
| 03:27.53 | drray | sox |
| 03:28.00 | alexns | hmm bad day.... got a lady who wants to sue cause of a bad day with teliax :) |
| 03:28.10 | pdtmobile | nice |
| 03:28.12 | *** join/#asterisk dacleric (n=dacleric@p54823C60.dip0.t-ipconnect.de) |
| 03:28.27 | pdtmobile | alexns: I had a customer end up with a line hung open with them for like 36 hours in a conference room |
| 03:28.38 | alexns | hehe |
| 03:28.39 | mosty | what file extension does sox expect for ulaw? it tells me "Do not understand format type: ulaw" |
| 03:28.52 | alexns | i got dcap but asterisk scares me |
| 03:29.01 | PMantis | use gsm |
| 03:29.20 | alexns | pdtmobile: analog line ? |
| 03:29.34 | mosty | pmantis: i need to convert from ulaw to wav first, just to hear what these sounds are exactly |
| 03:29.42 | pdtmobile | it's a pbx that is teliax only |
| 03:29.46 | pdtmobile | no idea how it happened |
| 03:29.55 | pdtmobile | i kicked whatever it was out of the conference room and everything was ok |
| 03:30.09 | alexns | pdtmobile: ive seen hangs like that but they usually clear, thats crazy |
| 03:30.11 | pdtmobile | they use it for conferencing and calling overseas |
| 03:30.45 | rene- | moziax is cool, and runs on windows |
| 03:30.46 | alexns | pdtmobile: shit just gets crazy |
| 03:30.54 | pdtmobile | yep |
| 03:31.14 | alexns | pdtmobile: not to mention i used snom 360 instead of polycoms and they suck balls when it comes to sound quality |
| 03:31.31 | pdtmobile | really? that sucks i was looking at them for a project |
| 03:31.37 | pdtmobile | i am polycom only mainly |
| 03:31.50 | pdtmobile | have some linksys wireless at a couple customers |
| 03:32.00 | alexns | pdtmobile: polycom are much better quality, ive had a few dead snoms right out of the box |
| 03:32.07 | pdtmobile | that sucks |
| 03:32.18 | *** join/#asterisk los415 (n=los415@sfca-office.corp.race.com) |
| 03:32.42 | alexns | pdtmobile: even better... same office junk netgear poe switch gets surge and blows poe circuts on 8 phones and takes out pos BTX dell server |
| 03:32.56 | pdtmobile | I have two gripes about the polycom, everything else i like. #1 i wish they booted faster #2 i wish the directories worked a little better |
| 03:33.01 | alexns | pdtmobile: pbx is still up.. homemade quality |
| 03:33.02 | *** join/#asterisk dan__t (n=dant@ip70-162-170-55.ph.ph.cox.net) |
| 03:33.05 | dan__t | damn |
| 03:33.10 | dan__t | the Intertel guy just left. |
| 03:33.23 | orlock | alexns: i have netgear switches |
| 03:33.25 | orlock | hate, even |
| 03:33.35 | orlock | especially the "smart" swiches |
| 03:33.37 | pdtmobile | i use cisco and dlink, that dlink 24 port PoE works pretty well |
| 03:33.47 | orlock | which are like a managed switches retarded nephew |
| 03:33.51 | alexns | pdtmobile: Hmm i used 601 with expansions bug in firmware on all polycoms only can monitor 7 hints |
| 03:34.06 | pdtmobile | alexns: it's fixed now |
| 03:34.15 | alexns | pdtmobile: i leave turds that are better quality than netgear |
| 03:34.27 | pdtmobile | the 501 can only monitor 7 still |
| 03:34.30 | alexns | i havn't been watching the channel |
| 03:34.34 | pdtmobile | maybe it's 8 |
| 03:34.36 | alexns | good to know |
| 03:34.45 | pdtmobile | but the latest sip gets you 48 |
| 03:34.49 | alexns | ahh |
| 03:34.50 | pdtmobile | want a copy? |
| 03:34.57 | alexns | i have the account |
| 03:35.01 | pdtmobile | k, nevermind |
| 03:35.02 | alexns | thanks anyway :) |
| 03:35.18 | los415 | i know this is a shot in the dark since digium said this is the first cable shipped. i got a timing ribbon cable to be able to share timing across multi t1 cards has anyone worked on this or know what you got to do to get it to work |
| 03:35.29 | alexns | what do you think of the linksys 942 ? |
| 03:35.43 | alexns | pdtmobile |
| 03:35.55 | mosty | ahh, sox -t .ul |
| 03:35.57 | file | los415: if you call support tomorrow they should be able to help you |
| 03:36.11 | pdtmobile | i honestly haven't touched the linksys phones yet, I have been considering grabbing a few for the lab |
| 03:36.42 | alexns | i have one its pretty decent although it has an annoying double ring when calling someone |
| 03:36.47 | alexns | just a bitchpoint |
| 03:36.54 | pdtmobile | asterisk and the phone? |
| 03:36.58 | los415 | file i know but whats the fun in that lol |
| 03:37.13 | file | los415: nobody here probably has any idea about it... or even knew it existed |
| 03:37.20 | alexns | yep no matter how you dial |
| 03:37.24 | los415 | yea i know |
| 03:37.32 | file | alexns: there's a setting |
| 03:37.35 | los415 | they shipped it to me and said good luck |
| 03:37.36 | los415 | lol |
| 03:37.41 | los415 | let us know how it works |
| 03:37.47 | pdtmobile | hehe |
| 03:37.50 | file | alexns: it's because the phone is getting a 180 Ringing and 183 Session Progress, and mixing a generated ring and ring from the audio stream |
| 03:37.52 | alexns | file: setting on phone.. like call progress |
| 03:37.54 | pdtmobile | los415: guess so far it's not? |
| 03:38.09 | file | los415: I can get to the info... if I get out of bed |
| 03:38.26 | los415 | hrmmm |
| 03:38.34 | file | which I may do in a moment |
| 03:38.34 | pdtmobile | you could also just turn off the ring in the dial string as well if the phone is being super brain dead |
| 03:38.48 | alexns | it is off |
| 03:38.51 | los415 | well it would be apreciated |
| 03:38.51 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 03:38.57 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 03:39.19 | pdtmobile | ya I guess you can't really kill the external telco ring |
| 03:39.21 | pdtmobile | hehe |
| 03:39.28 | file | okay, please be holding! |
| 03:41.52 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 03:42.38 | pdtmobile | otay, time to go home, been fun |
| 03:44.05 | Flauto | is trx here today? |
| 03:44.32 | Flauto | i have dtmf problem with trxtel.com's toll free termination |
| 03:44.38 | Flauto | anyone would help |
| 03:45.49 | rene- | if only polycoms were easy to set up, in that respect they suck balls |
| 03:47.28 | dan__t | i'm totally seeing how this Intertel PBX relates to Asterisk. |
| 03:51.53 | *** join/#asterisk bmg505 (n=leon@c1-181-8.rndf.isadsl.co.za) |
| 03:59.44 | CunningPike | rene-: Polycoms are easy, imho |
| 04:00.11 | *** join/#asterisk m_a_g_o (i=maxgluck@201.243.102.189) |
| 04:02.52 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 04:03.46 | m_a_g_o | good evening, could anyone give me a hint on how to use "originate" from a perl script using * manager? I have used the command method, but can't find originate... |
| 04:03.53 | rene- | i cant get them right, i always have to go tru the web, tftp and manual(onscren) configuration to get them working, snoms and grandstreams are a breeze, |
| 04:04.32 | m_a_g_o | is it also with command, like ->command('originate' '...')? |
| 04:04.38 | CunningPike | rene-: We just use FTP, and began with the default config files that come with the firmware, making tweaks as we need |
| 04:05.12 | CunningPike | rene-: Tonight, all my Polycoms will automatically reboot themselves and download the latest SIP application to themselves |
| 04:05.26 | CunningPike | rene-: How much easier could that be? |
| 04:07.38 | PMantis | CunningPike, Script sending a SIP notify? |
| 04:07.59 | CunningPike | PMantis: No, they have an autopoll option |
| 04:08.26 | CunningPike | PMantis: They wake up at 0200, see if _any_ of their config files have been updated and, if so, reboot themselves |
| 04:08.54 | PMantis | CunningPike, Oh? I just setup a call center with Polycoms.. I didn't notice that option. |
| 04:09.19 | CunningPike | PMantis: I'm at home right now, but if you're around tomorrow I can dig it out for you |
| 04:09.50 | PMantis | CunningPike, Awesome... or leave me a voicemail. w/ iaxtel |
| 04:10.04 | m_a_g_o | !? |
| 04:10.22 | CunningPike | PMantis: You'd better ping me - I'll forget |
| 04:10.26 | *** join/#asterisk tlow (i=tlowe@omfg.wtf.no) |
| 04:10.28 | *** join/#asterisk marv0997 (i=marv0997@190.4.2.86) |
| 04:10.45 | PMantis | CunningPike, Me too, that's the problem. :) |
| 04:10.54 | CunningPike | PMantis: What is? |
| 04:11.04 | PMantis | CunningPike, I'll forget, too. |
| 04:11.10 | CunningPike | PMantis: Forget what? |
| 04:11.27 | PMantis | CunningPike> PMantis: You'd better ping me - I'll forget |
| 04:11.44 | CunningPike | PMantis: I have no recollection of that |
| 04:11.48 | CunningPike | ;) |
| 04:12.04 | PMantis | Doh! |
| 04:13.40 | PMantis | CunningPike, BTW, do you use a web interface to cnfig your * servers? |
| 04:14.09 | CunningPike | PMantis: No, our phones are on their own VLAN, so I couldn't even if I wanted to |
| 04:14.30 | PMantis | CunningPike, :-) I mean Asterisk itself. |
| 04:14.35 | CunningPike | PMantis: SOrry - misread the question - but the answer's still the same - I use BBEdit |
| 04:15.12 | CunningPike | PMantis: It's a text editor....... |
| 04:15.14 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 04:15.18 | PMantis | Ahhh |
| 04:15.22 | PMantis | for the Mac |
| 04:15.30 | CunningPike | PMantis: What else? |
| 04:15.32 | CunningPike | :) |
| 04:15.59 | PMantis | CunningPike, Well, I usually use vi, but when deploying to a client, I used FreePBX... |
| 04:16.18 | PMantis | Was hoping to have a little module that would write the XML for the polycoms from the web interface. |
| 04:16.38 | PMantis | Right now it's manual |
| 04:17.02 | CunningPike | PMantis: Ah. I am working on a bunch of shell scripts that will manage our configs, so our help desk can use a curses style CLI to make changes |
| 04:17.20 | CunningPike | Add a new phone, change a name, reset a voicemail password etc |
| 04:17.23 | PMantis | That'd still be slick |
| 04:17.42 | CunningPike | I plan to release them when they're ready - should be over the summer |
| 04:17.45 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 04:18.13 | PMantis | Nice... make it feel like a Nortel switch to those used to it. lol |
| 04:18.35 | *** join/#asterisk phalacee (n=Sunforge@202.3.110.33) |
| 04:20.46 | Strom_C | aghhh |
| 04:21.01 | Strom_C | my mobile phone lost track of the network twenty minutes ago |
| 04:21.41 | Strom_C | I've been a happy t-mobile customer for two years now but this is ridiculous |
| 04:24.50 | St1ckm4n | Strom_C: what phone do you have? |
| 04:25.13 | Strom_C | HTC Wizard |
| 04:25.18 | Strom_C | aka "Tmobile MDA" |
| 04:25.22 | St1ckm4n | do you like it |
| 04:25.27 | Strom_C | yes |
| 04:25.32 | Strom_C | I'm quite satisfied with it |
| 04:25.37 | St1ckm4n | I've been wanting to get it, but worried it might be buggy |
| 04:25.51 | St1ckm4n | the sidekick worked good for email but I need full windows capability |
| 04:25.53 | Strom_C | woot, it found the network |
| 04:26.11 | St1ckm4n | do you have alot of lockup issues with it? |
| 04:26.14 | Strom_C | no |
| 04:26.16 | file | Strom_C: I hid the network |
| 04:26.17 | Strom_C | never |
| 04:26.22 | St1ckm4n | good to know |
| 04:26.23 | Strom_C | file: oh ok |
| 04:26.24 | St1ckm4n | thanks |
| 04:26.35 | Strom_C | alright, ive gotta run |
| 04:26.50 | file | walk, don't run |
| 04:27.18 | St1ckm4n | nah, go ahead and run this ain't no kiddy pool |
| 04:42.05 | *** part/#asterisk tlow (i=tlowe@omfg.wtf.no) |
| 04:42.50 | *** join/#asterisk HawkmanAZ (n=JamesHaw@ip70-176-233-229.ph.ph.cox.net) |
| 04:47.07 | HawkmanAZ | What is the main reason for dropped call in Asterisk? I am running Asterisk with no IRQ misses and all phones on a local LAN but we have 20 to 30 dropped calls a day. I have tried every configuration but to no avail. |
| 04:50.52 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 04:52.27 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 05:10.45 | file | moo |
| 05:14.24 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 05:19.55 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:25.08 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 05:27.08 | file | I go nini, bye |
| 05:27.11 | russellb | nini. |
| 05:31.45 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 05:31.47 | TheCops | Hi |
| 05:33.36 | TheCops | Someone know when a hold event is generated, this is not working here.. |
| 05:33.39 | CunningPike | Run! It's the cops |
| 05:36.12 | TheCops | :) |
| 05:48.06 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 05:55.58 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 05:58.50 | Strom_C | oh hell, here's a weird problem |
| 05:59.10 | Strom_C | two fax machines on two PAP2Ts |
| 05:59.21 | Strom_C | suddenly, they both stop responding to inbound calls |
| 05:59.31 | Strom_C | the PAP2s are ringing the lines just fine |
| 05:59.58 | Strom_C | I can plug my buttset into the FXS port of the PAP2 and place and answer calls all day long |
| 06:02.24 | Strom_C | I want to blame the equipment, but two pieces of equipment failing simultaneously? |
| 06:03.17 | *** join/#asterisk twisted[asteria] (n=twisted@pdpc/supporter/active/twisted) |
| 06:03.17 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 06:03.26 | Qwell | twisted[asteria]: y0 |
| 06:03.29 | Strom_C | hey twisted[asteria]! |
| 06:03.34 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 06:03.40 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 06:06.34 | *** join/#asterisk SunMoon_ (n=Ravi@ip68-13-60-184.om.om.cox.net) |
| 06:07.15 | SunMoon_ | help help help |
| 06:07.27 | Strom_C | SunMoon_: pour salsa on it |
| 06:07.37 | SunMoon_ | I need some help with asterisk running on Trixbox version 1.1 |
| 06:07.45 | Strom_C | sigh |
| 06:07.49 | Strom_C | #freepbx |
| 06:07.58 | tzafrir_laptop | this is a place for asterisk questions |
| 06:08.08 | SunMoon_ | I thought it was the place |
| 06:08.16 | Strom_C | i mean, uh, what is your question |
| 06:08.16 | SunMoon_ | correct me if I am wrong ? |
| 06:08.25 | tzafrir_laptop | if you can phrase that as an asterisk question, as here. Else, try #freepbx |
| 06:08.44 | SunMoon_ | should I got to that channel |
| 06:08.50 | SunMoon_ | #freepbx ? |
| 06:08.54 | Strom_C | oh just bloody ask the question already |
| 06:09.00 | Strom_C | :) |
| 06:09.25 | SunMoon_ | Seems like Strom is pretty angry |
| 06:09.27 | SunMoon_ | :)) |
| 06:09.40 | Strom_C | yes, i'm dealing with a pair of completely bonkers fax machines |
| 06:09.48 | Strom_C | so I apologize if I seem....edgy. |
| 06:10.08 | SunMoon_ | thats ok my friend for all you know I don't even know u from adam |
| 06:10.42 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:10.50 | PakiPenguin_ | anyone has some example of asterisk interaction with a mysql db from dial plan? |
| 06:10.53 | Strom_C | so come on...ask us your question |
| 06:11.10 | *** part/#asterisk SunMoon_ (n=Ravi@ip68-13-60-184.om.om.cox.net) |
| 06:11.26 | tzafrir_laptop | PakiPenguin_, what do you mean? real time? static/dynamic? |
| 06:11.26 | Strom_C | or not |
| 06:11.56 | PakiPenguin_ | nope , tzafrir_laptop , like getting data from a mysql db ( like order number / status ) and playing it back to the caller |
| 06:12.18 | dlynes_home | PakiPenguin_: that sounds like an agi script, not a dialplan |
| 06:12.33 | PakiPenguin_ | ah dlynes_home okay .. /me looks into php-agi |
| 06:12.35 | tzafrir_laptop | there is something in addons just for that |
| 06:12.49 | russellb | there is also func_odbc |
| 06:12.53 | tzafrir_laptop | The syntax has changed between 1.0 and 1.2, I believe |
| 06:12.53 | PakiPenguin_ | tzafrir_laptop, name it namee it :) |
| 06:13.36 | tzafrir_laptop | show application mysql<tab><tab> helps? |
| 06:14.04 | tzafrir_laptop | PakiPenguin_, is that this specific 1.0 system? |
| 06:14.16 | PakiPenguin_ | yup |
| 06:15.11 | pdtmobile | dlynes_home: the faxing is working like a champ so far |
| 06:15.18 | dlynes_home | pdthome: cool |
| 06:15.21 | dlynes_home | pdthome: good to hear |
| 06:15.37 | pdtmobile | about 10 to 20% success rate with rxfax |
| 06:15.42 | pdtmobile | so far 100% with iaxmodem |
| 06:15.55 | dlynes_home | pdthome: exactly why i suggested it to you :) |
| 06:16.09 | PakiPenguin_ | iaxmodem? |
| 06:16.12 | dlynes_home | pdthome: frankly, i wasn't even getting 10-20% with rxfax |
| 06:16.21 | pdtmobile | well with PRI i get around 80% |
| 06:16.22 | dlynes_home | pdthome: i was getting 0% |
| 06:16.28 | pdtmobile | but analog for some reason i get HOSED |
| 06:16.32 | dlynes_home | pdthome: and that was on a pri |
| 06:16.32 | EvilDeshi | im having a problem getting my asterisk to register with fwd iax2 Jul 6 22:42:17 NOTICE[2518]: chan_iax2.c:7438 socket_read: Registration of '766860' rejected: 'Registration Refused' from: '192.246.69.186' can someone help me out with what could be causing this |
| 06:16.35 | PakiPenguin_ | we've a very good rate with rxfax |
| 06:16.45 | Strom_C | hello, EvilDeshi :) |
| 06:16.53 | EvilDeshi | hey St1ckm4n |
| 06:16.55 | EvilDeshi | Strom_C |
| 06:17.05 | EvilDeshi | funny finding you here i am on the phone with lamerjoe |
| 06:17.28 | pdtmobile | dlynes_home: neway, thanks for the suggestion |
| 06:17.36 | Strom_C | EvilDeshi: heh |
| 06:17.41 | dlynes_home | EvilDeshi: ummmm...you need to go through a stun server, or a proxy more than likely |
| 06:17.43 | tzafrir_laptop | PakiPenguin_, show application mysql should help |
| 06:17.54 | dlynes_home | EvilDeshi: it's grabbing your private subnet address by the looks of it |
| 06:18.00 | PakiPenguin_ | thanks tzafrir_laptop |
| 06:18.04 | EvilDeshi | thats not my subnet |
| 06:18.09 | EvilDeshi | that ip belongs to fwd |
| 06:18.28 | dlynes_home | ah...couldn't recall if 192.246 was class c private or not |
| 06:18.54 | dlynes_home | EvilDeshi: you're either in their list of denied ips, or you got your username or password incorrect for the registration |
| 06:19.01 | pdtmobile | PakiPenguin_: you using it with pri or analog? |
| 06:19.06 | PakiPenguin_ | analog |
| 06:19.06 | dlynes_home | EvilDeshi: or you're just not set up yet |
| 06:19.11 | pdtmobile | hmm |
| 06:19.12 | EvilDeshi | sip works |
| 06:19.20 | EvilDeshi | i have it set in fwd to use iax |
| 06:19.34 | pdtmobile | it definitely seems to be a YMMV type of beast, guess your mileage is better than mine :) |
| 06:20.13 | PakiPenguin_ | :) |
| 06:20.21 | PakiPenguin_ | pdtmobile, which country ? us? |
| 06:20.26 | pdtmobile | ya |
| 06:21.15 | dlynes_home | PakiPenguin_: you're in lahore? |
| 06:21.26 | PakiPenguin_ | dlynes_home, nope :) Islamabad |
| 06:21.32 | EvilDeshi | i just cant figure this out |
| 06:21.38 | PakiPenguin_ | been to lahore dlynes_home ? |
| 06:21.41 | dlynes_home | PakiPenguin_: ah |
| 06:21.47 | dlynes_home | PakiPenguin_: nope...Dr-Linux is from there |
| 06:21.53 | PakiPenguin_ | ah i see |
| 06:22.08 | EvilDeshi | i mean if i change the line iax2.fwdnet.net to sip.fwdnet.net it will register |
| 06:22.14 | dlynes_home | PakiPenguin_: my only pakistani friend in Canada is from a small town outside of Karachi |
| 06:22.19 | EvilDeshi | i have my fwd account setup with iax enabled |
| 06:22.22 | PakiPenguin_ | dlynes_home, i think dr-linux works in a call center |
| 06:22.27 | PakiPenguin_ | i see :) |
| 06:22.27 | dlynes_home | PakiPenguin_: yeah, he does |
| 06:22.49 | PakiPenguin_ | i work in a cellular company |
| 06:22.56 | dlynes_home | lots of punjabis here...not many pakistanis though |
| 06:23.09 | PakiPenguin_ | i see which city? |
| 06:23.16 | PakiPenguin_ | canada :) |
| 06:23.17 | dlynes_home | Vancouver |
| 06:23.39 | dlynes_home | almost all of the punjabis here are jutt's, too |
| 06:24.20 | PakiPenguin_ | jutt? |
| 06:24.24 | dlynes_home | yeah |
| 06:24.28 | PakiPenguin_ | whats that? |
| 06:24.33 | dlynes_home | It's a caste in India |
| 06:25.00 | dlynes_home | If it's not the top caste, it's one of the top castes |
| 06:26.08 | PakiPenguin_ | i see |
| 06:26.22 | PakiPenguin_ | we have something called "jaat" here , i think they are the same |
| 06:26.28 | dlynes_home | probably |
| 06:26.35 | dlynes_home | jutt is a punjabi word |
| 06:26.46 | dlynes_home | erdu is very similar to punjabi |
| 06:26.55 | dlynes_home | or urdu or however you spell it |
| 06:27.03 | dlynes_home | urdu's also pretty similar to farsi |
| 06:27.58 | PakiPenguin_ | yup it was made from farsi, urdu in general is not a language , its a mixture of languages |
| 06:28.05 | dlynes_home | ah |
| 06:28.14 | dlynes_home | well, farsi isn't really a language, either |
| 06:28.19 | dlynes_home | it's just a mixture as well |
| 06:28.37 | dlynes_home | I think it might have 10-20% original words |
| 06:28.42 | dlynes_home | everything else is some other language |
| 06:29.00 | dlynes_home | It's got a lot of french, english and arabic mixed in |
| 06:29.11 | PakiPenguin_ | haha yeah |
| 06:29.33 | dlynes_home | even their word for police is french :) |
| 06:29.50 | dlynes_home | Les gendremeris |
| 06:29.53 | *** join/#asterisk CMike (i=daemon@c-874071d5.116-1-64736c10.cust.bredbandsbolaget.se) |
| 06:30.20 | dlynes_home | it means rural constabulary |
| 06:30.22 | CMike | morning |
| 06:30.50 | dlynes_home | sup, esse? |
| 06:32.26 | PakiPenguin_ | i see |
| 06:35.05 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 06:40.32 | *** join/#asterisk DaPriv (i=Privatee@crimson.66fruit.com) |
| 06:53.24 | *** join/#asterisk Eecplat (n=ouarf@AStDenis-105-1-63-231.w80-8.abo.wanadoo.fr) |
| 06:54.50 | *** join/#asterisk P-NuT (n=nut@fw.office.unitedip.net.au) |
| 06:54.59 | P-NuT | Hi everyone |
| 06:55.18 | P-NuT | I've used the record command like this..... |
| 06:55.30 | P-NuT | exten => *77,3,Record(/var/lib/asterisk/custom-sounds/myivr.gsm) |
| 06:55.40 | P-NuT | it works and records the file, |
| 06:55.50 | P-NuT | but when I try to play it back with this.... |
| 06:55.59 | P-NuT | exten => *99,3,Playback(/var/lib/asterisk/custom-sounds/myivr.gsm) |
| 06:56.09 | P-NuT | it says that the file doesn't exist.. |
| 06:56.18 | CMike | loose the .gsm |
| 06:56.18 | Strom_C | does the file show up when you ls? |
| 06:56.28 | Strom_C | kill the .gsm and use .wav :) |
| 06:56.31 | P-NuT | Now, I AM trying to record and playback this with an IAX softphone external |
| 06:56.50 | P-NuT | Strom_C: yes, file is there... |
| 06:56.52 | Strom_C | does asterisk have write permissions to the directory? |
| 06:56.59 | *** join/#asterisk inv_Arp (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 06:57.02 | *** join/#asterisk junbug (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 06:57.11 | P-NuT | well, the file is there, so I can only assume so |
| 06:57.14 | CMike | have you read "show application playback" ? |
| 06:57.17 | CMike | Playback(filename[|option]): Plays back a given filename (do not put |
| 06:57.17 | CMike | extension). |
| 06:57.19 | P-NuT | yes, |
| 06:57.20 | CMike | sort of.. |
| 06:57.25 | P-NuT | there's not many options to it |
| 06:57.25 | CMike | so loose the exten |
| 06:57.32 | P-NuT | exten? |
| 06:57.36 | CMike | eg: exten => *99,3,Playback(/var/lib/asterisk/custom-sounds/myivr) |
| 06:57.39 | Strom_C | CMike: it's "lose" not "loose" |
| 06:57.43 | CMike | oops |
| 06:57.54 | Strom_C | loose == opposite of tight |
| 06:57.57 | Strom_C | lose == to misplace |
| 06:58.01 | CMike | <-- still not awake :) |
| 06:58.06 | P-NuT | ok, so what am I doing? |
| 06:58.19 | CMike | exten => *99,3,Playback(/var/lib/asterisk/custom-sounds/myivr) |
| 06:58.29 | CMike | without the ".gsm" |
| 06:59.17 | CMike | anyone in here using the sangoma a104 ? |
| 06:59.44 | CMike | I've got really bad sound on the outgoing E1 |
| 07:00.28 | P-NuT | CMike: Will that really help?? |
| 07:00.50 | P-NuT | what about this? exten => *99,3,Playback(/var/lib/asterisk/custom-sounds/myivr:gsm) |
| 07:00.54 | CMike | you should not use extensions on filename in the playback command |
| 07:01.09 | P-NuT | ok then.... |
| 07:01.12 | CMike | as the helpfile says |
| 07:01.16 | P-NuT | I'll try that |
| 07:01.18 | P-NuT | thanks |
| 07:01.26 | CMike | try that.. |
| 07:01.52 | CMike | brb need morningcoffee |
| 07:03.32 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 07:05.08 | P-NuT | fixed it... |
| 07:05.23 | P-NuT | you need to press # after you record or else it records nothing. |
| 07:05.28 | P-NuT | Thanks guys! |
| 07:07.07 | Strom_C | wow, it only took an hour to fix that fax problem |
| 07:07.27 | Strom_C | apparently the fax machines suddenly got very picky about the ring voltage they were receiving |
| 07:08.37 | Strom_C | voodoo |
| 07:08.40 | Strom_C | it's all voodoo |
| 07:08.54 | Strom_C | waving a dead chicken isnt enough |
| 07:09.06 | Strom_C | you actually have to bludgeon the machine to smithereens with the dead chicken |
| 07:09.07 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 07:13.37 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 07:16.11 | E-bola | Do anybody have examples or a howto on how to connect asterisk to a SIP gateway/provideR? |
| 07:16.22 | E-bola | so we can dial out on the PSTN net |
| 07:22.43 | *** join/#asterisk SwK (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 07:22.47 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:37.49 | SwK | damn its quiet in here tonight |
| 07:40.14 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 07:57.34 | *** join/#asterisk CoffeeKid (n=kirkalle@dsl093-224-026.slc1.dsl.speakeasy.net) |
| 07:59.04 | *** join/#asterisk RoyKa (n=roy@chello080109196173.3.graz.surfer.at) |
| 08:00.29 | tzafrir_laptop | SwK, good morning to you too |
| 08:02.24 | _4d4m_ | just got some polycom 601's through. Everything working great, except occasional calls from internal extensions only, dont cause the unit to ring (or register an incoming call at all). * thinks all is well however, and the call rings for specified duration before returning an unavailable status. anyone got any advice for me? |
| 08:02.48 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 08:04.11 | fenlander | _4d4m_: might be worth taking an ethereal trace to see if the polycom is responging to the INVITE at all |
| 08:04.26 | fenlander | _4d4m_: then we can start pointing fingers |
| 08:04.48 | _4d4m_ | fenlander: am right in the middle of that as we speak ;) |
| 08:06.23 | *** join/#asterisk RoyK[at] (n=roy@80.109.196.173) |
| 08:06.42 | fenlander | _4d4m_: let us know what happens :) |
| 08:11.24 | RoyK[at] | <PROTECTED> |
| 08:12.23 | *** join/#asterisk TeePOG (n=1234@196.207.45.253) |
| 08:12.36 | TeePOG | good morning |
| 08:14.32 | *** join/#asterisk Laerte (n=bho@217.221.36.10) |
| 08:14.37 | Laerte | hy |
| 08:14.53 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 08:15.13 | e-ddie | hi folks |
| 08:15.40 | *** part/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 08:15.47 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 08:16.39 | e-ddie | what's a good way to implement whether the call goes to voicemail, or not by dialing a specified nr? |
| 08:17.11 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 08:17.12 | *** join/#asterisk zetp (i=ZeT@politechnika.lublin.pl) |
| 08:17.47 | zetp | hi! |
| 08:18.02 | *** join/#asterisk nfi|ermes (n=nfi_erme@217.220.121.62) |
| 08:18.12 | zetp | i have to develop a IVR system on asterisk |
| 08:18.59 | zetp | i have got to connnect only 2 pstn networks |
| 08:19.57 | zetp | when call is from pstn there should be a voice response |
| 08:20.11 | zetp | like "welcom ... please press 1 for ..." |
| 08:20.49 | zetp | and call shold be pass to second pstn network |
| 08:21.01 | zetp | have someone try it? |
| 08:31.12 | tzafrir_laptop | zetp, it's a standard Asterisk config |
| 08:31.40 | tzafrir_laptop | Use Background in the s extension on the IVR's context to play the message |
| 08:31.53 | tzafrir_laptop | ~docs |
| 08:31.55 | jbot | docs is probably probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 08:32.06 | zetp | i'm new in asterisk |
| 08:32.11 | eaperezh | zept: check here http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu |
| 08:32.27 | zetp | i thought there will be problem with passing a signalisation from isdn to isdn |
| 08:32.59 | zetp | thanks for answer |
| 08:35.49 | eaperezh | zept: The thing is the dialing command. check that link... |
| 08:36.36 | eaperezh | I need to connet 8 FXO ports from a Panasonic PBX to asterisk. Is there a device to do that? |
| 08:40.18 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 08:40.22 | rene- | hey |
| 08:40.32 | rene- | i am having an issue with PRI |
| 08:40.41 | rene- | anyone want to take alook at it? |
| 08:40.42 | rene- | http://pastebin.ca/81260 |
| 08:41.46 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 08:47.33 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 08:55.24 | eaperezh | I guess my question is confusing. A customer has a Panasonic PBX with 8 CO and 24 analog extensions. |
| 08:55.24 | eaperezh | Currently is setup like this: |
| 08:55.24 | eaperezh | analog_phones----panasonic_pbx----PSTN |
| 08:55.24 | eaperezh | And now he wants this: |
| 08:55.24 | eaperezh | analog_phones----panasonic_pbx---ASTERISK---PSTN |
| 08:55.59 | eaperezh | what should I buy to connect the asterisk to the panasonic? FXS or FXO? |
| 08:56.50 | tzafrir_laptop | how do you want to connect it? |
| 08:57.18 | tzafrir_laptop | do you want Asterisk for the voicemail? for voip? |
| 08:57.34 | tzafrir_laptop | Should phones be still connected to the panasonic pbx? |
| 08:57.57 | tzafrir_laptop | should pstn be still connected to the panasonic? |
| 08:58.14 | *** join/#asterisk Corydon76-home (i=eight@pdpc/supporter/sustaining/Corydon76-home) |
| 09:01.11 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 09:03.08 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:05.17 | eaperezh | tzafrir: Asterisk should provide voicemail to the users and pstn dialing capabilities. |
| 09:06.15 | eaperezh | tzafrir: phones still connected to panasonic. pstn only on asterisk and removed from the panasonic |
| 09:08.20 | *** join/#asterisk rkr245 (n=ravi@81.21.33.35) |
| 09:09.03 | rkr245 | hi all |
| 09:09.11 | rkr245 | goodmorning |
| 09:10.42 | *** join/#asterisk cbkm (n=kenny@mail.3ait.co.uk) |
| 09:11.33 | cbkm | Hi, We recently installed Asterisk 1.2.9.1 on a client of ours and noticed that qview.pl (contrib/scripts/) didn't appear to work, I've got it working, created a svn diff... where's the best place to put the patch file? :-) |
| 09:13.32 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:14.11 | *** join/#asterisk azeteg (n=azeteg@c115.brewhouse.se) |
| 09:16.25 | rkr245 | hi guys |
| 09:17.42 | rkr245 | when i am using x-lite i can make and recieve calls but i got a new phone its " ip-300 " made in china its a sip phone but i can only recieve calls and cannot make calls |
| 09:17.56 | rkr245 | does any body know what is wrong here? |
| 09:18.17 | rkr245 | when i used x-lite my sip messages are like this |
| 09:18.21 | *** join/#asterisk RoyK[at] (n=roy@chello080109196173.3.graz.surfer.at) |
| 09:18.34 | rkr245 | v=0 |
| 09:18.47 | rkr245 | o=32331003 1361195779 1361196364 IN IP4 192.168.0.79 |
| 09:19.00 | rkr245 | s=X-Lite |
| 09:19.08 | rkr245 | but with ip-300 |
| 09:19.12 | rkr245 | v=0 |
| 09:19.18 | *** join/#asterisk tparcina (n=tparcina@lns02-0003.dsl.iskon.hr) |
| 09:19.26 | tparcina | hi channel!¨ |
| 09:19.35 | rkr245 | o=- 93075978 03769140 IN IP4 192.168.0.74 |
| 09:20.01 | rkr245 | here for x-lite o=32331003 its the x-lite number |
| 09:20.15 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
| 09:20.24 | rkr245 | and for ip-300 o=-93075978 |
| 09:20.38 | rkr245 | its not the number i gave it to that |
| 09:20.48 | rkr245 | its sending some thing else |
| 09:20.56 | rkr245 | any guys any clue |
| 09:21.11 | rkr245 | please help me |
| 09:21.11 | rkr245 | when you see this message |
| 09:21.19 | rkr245 | ;-( |
| 09:22.09 | tparcina | rkr, what's the problem? - i have just join |
| 09:23.17 | *** join/#asterisk kay2 (n=ashdown@sd-420.dedibox.fr) |
| 09:27.03 | e-ddie | can anyone point me to a document about BLA settings in asterisk for snom phones? |
| 09:27.05 | rkr245 | tparcina,igot a new phone |
| 09:27.28 | rkr245 | its "ip-300" sip phone |
| 09:28.01 | rkr245 | when i used to make calls from x-lite its every thing o.k |
| 09:28.35 | rkr245 | but when iam using ip-300 it can only recieve calls but cannot make calls |
| 09:29.16 | rkr245 | tparcina,and its showing 401 gone message to me |
| 09:30.09 | rkr245 | so whats the problem in behalf of this phone settings ;-( |
| 09:30.25 | tparcina | rkr, sorry, can't help you out |
| 09:30.39 | rkr245 | o.k np |
| 09:33.18 | *** join/#asterisk nXOR (n=drade@pdpc/supporter/sustaining/nXOR) |
| 09:33.29 | tzafrir_laptop | eaperezh, what interfaces other than plain telephony does this panasonic pbx support? Any voip? ISDN? |
| 09:33.37 | nXOR | hello dear people, i have a query that u might answer or point me towards some resource |
| 09:33.55 | nXOR | i have finally managed to setup asterisk with visdn and its working properly |
| 09:34.01 | tzafrir_laptop | nXOR, ask, but wait for answers... |
| 09:34.20 | nXOR | however the outgoing sound is great no jitters or anythign, but incoming soud is echoish .... |
| 09:34.53 | nXOR | for instance i call srom a soft phone to my gsm, i recieve an answer however i can hear sound bouncing back in the gsm |
| 09:34.56 | nXOR | hello -- hello |
| 09:35.07 | nXOR | anythign i can do about it |
| 09:35.20 | nXOR | qos ? |
| 09:35.23 | nXOR | or whatnot |
| 09:36.13 | tzafrir_laptop | eaperezh, if not, you could use a sort of "pass-through": on Asterisk: 8 FXS ports for the "pstn" ports of the panasonic, and 8 FXO ports to the real PSTN |
| 09:37.57 | tzafrir_laptop | nXOR, what are both ends? |
| 09:38.34 | nXOR | well i have asteisk box connected with rj-45 cable to NT switch |
| 09:38.42 | nXOR | the other part of asterisk box is LAN |
| 09:38.48 | tzafrir_laptop | echo can't be generated in ISDN and in voip. I gather it is generated on the soft-phone. Can you try to cancel it there? (or maybe use a better headset?) |
| 09:38.55 | nXOR | hm |
| 09:39.12 | tzafrir_laptop | s/a better/a/ |
| 09:39.13 | nXOR | yes but i can hear gsm to soft phone clearly |
| 09:39.24 | nXOR | however the gsm echos back everythign i say |
| 09:39.43 | nXOR | let me see the soft phone options |
| 09:39.48 | nXOR | ill try from ip phone aswell |
| 09:39.51 | nXOR | moment |
| 09:40.03 | tzafrir_laptop | this is probably because sound in one of the ends "leaks" back |
| 09:40.45 | tzafrir_laptop | What are both ends of the call? |
| 09:41.00 | nXOR | yes hard phone is clear, and gsm is clear aswell |
| 09:41.14 | nXOR | there is some echoing but its calmed down immidiately |
| 09:41.22 | tzafrir_laptop | On which side do you hear the echo? |
| 09:41.23 | nXOR | ip phone that is* |
| 09:41.26 | nXOR | gsm side |
| 09:41.31 | nXOR | when talkign to a soft phone |
| 09:41.45 | nXOR | ip phone to gsm works fine, so i guess that narrows it down to soft phone |
| 09:42.51 | nXOR | yep to be exact its sjphone |
| 09:42.54 | nXOR | x-lite works fine |
| 09:42.56 | lilalinux | does anyone know a sipclient for nokia 9500 (OS: S80)> |
| 09:43.07 | nXOR | tzafrir_laptop: uve been an inspiration :D |
| 09:43.09 | tzafrir_laptop | For starters, get a simple headset. |
| 09:43.20 | nXOR | yeah, im in the testing process atm |
| 09:43.27 | nXOR | we will buy ip phones |
| 09:44.49 | *** join/#asterisk daysmen3 (n=primus@host86-139-212-252.range86-139.btcentralplus.com) |
| 09:47.34 | daysmen3 | test |
| 09:48.17 | *** join/#asterisk linuX|Reneger (n=linuxren@Q7866.q.strato-dslnet.de) |
| 09:50.35 | nXOR | great, it works, the sound is astill a lil bit metally, but thats nothign to worry about |
| 09:55.35 | nXOR | ty boys and girls |
| 09:55.37 | *** part/#asterisk nXOR (n=drade@pdpc/supporter/sustaining/nXOR) |
| 09:58.31 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 10:04.58 | *** join/#asterisk loopt (n=pt@gw1.sanyo.hu) |
| 10:09.10 | *** part/#asterisk tparcina (n=tparcina@lns02-0003.dsl.iskon.hr) |
| 10:19.50 | *** part/#asterisk eDIsonxl (n=eDIsonxl@60-248-182-162.HINET-IP.hinet.net) |
| 10:30.45 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 10:35.18 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 10:38.51 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 10:43.45 | *** part/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 10:51.58 | *** part/#asterisk Samoied (n=Samoied@201.47.218.152) |
| 10:52.17 | *** join/#asterisk eDIsonxl (n=xian-lia@60-248-182-162.HINET-IP.hinet.net) |
| 11:08.26 | *** join/#asterisk groogs (n=greg@d38-54-164.commercial1.cgocable.net) |
| 11:13.17 | _problem_ | Does any body working with chan_sccp here ? |
| 11:18.52 | RoyK[at] | http://asterisk-backports.org/wiki/index.php/Max_open_files-1.2 <-- can someone help me port that to trunk? I don't know my way around the code there too well |
| 11:21.54 | *** join/#asterisk Zerthimon (n=Zerthimo@80.74.110.153) |
| 11:22.16 | Zerthimon | Hi, is there any way to trap an application error in dialplan ? |
| 11:27.56 | *** join/#asterisk burnproof (n=burnproo@210.213.244.61) |
| 11:28.04 | FlatFoot | got a codec question , what would you reccommend to use for 1 phone on the end of an ADSL ( 1 meg ) registering to a * on the end of a wireless ( 8 meg SDSL ) |
| 11:28.15 | burnproof | good day ! guys |
| 11:28.40 | tzafrir_laptop | Zerthimon, what application? |
| 11:29.11 | tzafrir_laptop | Start with 'show application Thatpplication' in the cli |
| 11:31.58 | *** join/#asterisk evisu (n=hIRC@bzq-88-152-238-38.red.bezeqint.net) |
| 11:33.07 | Zerthimon | tzafrir: well, I'm using a Read ( ) and when noone make an option the system acts weirdly |
| 11:33.15 | burnproof | how can i initiate a transfer without loosing the caller ? for example the called party is not on his/her desk? |
| 11:33.46 | Zerthimon | tzafrir: perhaps I need to figureout what to do if the variable isn't set |
| 11:34.50 | tzafrir_laptop | "Read should disconnect if the function fails or errors out" |
| 11:35.27 | RoyK[at] | FlatFoot: SDSL is <= 2.3Mbps by stanard... |
| 11:35.35 | RoyK[at] | s/stanard/standard/ |
| 11:35.40 | tzafrir_laptop | The variable may be empty if the user just dialed # |
| 11:35.45 | Zerthimon | it doesn't disconnect, it says "User entered Nothing" and dailplan continues with variable contains "" |
| 11:37.08 | Zerthimon | tzafrir: and then GotoIf fails cos the variable is unset |
| 11:37.19 | Zerthimon | any idea ? |
| 11:38.20 | tzafrir_laptop | What do you want to do in case of an error? As opposed to an empty input? |
| 11:38.55 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 11:39.01 | Zerthimon | tzafrir: |
| 11:39.32 | Zerthimon | tzafrir: I wanna handle the situation when caller doesn't make any choice |
| 11:40.52 | FlatFoot | sorry a meant SDSL as in smae speed both directions |
| 11:41.29 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 11:42.05 | RoyK[at] | FlatFoot: SDSL is symmetric, but leased lines are not SDSL, but usually just as symmetric...... |
| 11:42.34 | FlatFoot | thats the word i was looking for , sorry having a bit of a blonde day |
| 11:42.48 | RoyK[at] | :) |
| 11:43.07 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 11:43.28 | FlatFoot | anyway need to find the best codec you would recommend for the one phone on the ADSL |
| 11:43.53 | FlatFoot | got a bit of delay / crackling on the calls |
| 11:44.19 | FlatFoot | need to get it right cos it's a firm of solicitors ( don't wanna go to court ) |
| 11:46.18 | RoyK[at] | FlatFoot: which version of asterisk? |
| 11:46.38 | FlatFoot | hang on i'll look it's remote and i cant remeber |
| 11:46.40 | RoyK[at] | the most important is getting a good jitterbuffer and plc even if you're on a good link |
| 11:47.12 | FlatFoot | 1.0.9 |
| 11:47.47 | FlatFoot | built on debian from the quick install disc |
| 11:48.27 | RoyK[at] | then you need to either upgrade to trunk, or to 1.2.9.1 and hand-patch it with the code from http://asterisk-backports.org/wiki/index.php/Rtp-jb-1.2 |
| 11:48.42 | RoyK[at] | if this is sip/pstn, that tis |
| 11:48.44 | RoyK[at] | that is |
| 11:49.00 | FlatFoot | it's sip to sip |
| 11:49.19 | FlatFoot | basically goes ADSL 1 to WIRELESS to ADSL 2 |
| 11:49.30 | RoyK[at] | then use the patch anyway to keep RTP timestamps so the dejittering can be done on the endpoints |
| 11:49.40 | RoyK[at] | with the current code, RTP timestamps are lost when bridging a call |
| 11:49.41 | FlatFoot | there are 3 offices main one is on the big wireless that remotes are on ADSL |
| 11:50.11 | RoyK[at] | also wireless == packet loss |
| 11:50.18 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 11:50.21 | RoyK[at] | meaning you really need dejittering+plc |
| 11:50.47 | FlatFoot | the wireless has been running without packet loss for a year now |
| 11:51.25 | RoyK[at] | anyway, patch up and try again, and you will hear the difference |
| 11:51.35 | FlatFoot | wireless from one ADSL is round trip of no more than 20ms |
| 11:51.52 | RoyK[at] | since asterisk destroys timestamps, it fucks up dejittering |
| 11:51.56 | RoyK[at] | making it all worse |
| 11:51.59 | RoyK[at] | beleive me. |
| 11:52.03 | burnproof | RoyK[at]: can i ask a few question please |
| 11:52.07 | RoyK[at] | or keep on trying := |
| 11:52.14 | RoyK[at] | burnproof: sure |
| 11:52.18 | FlatFoot | so upgrade the whole thing to 1.2.9.1 then patch yea ? |
| 11:52.33 | FlatFoot | ask away Roy |
| 11:52.34 | RoyK[at] | download 1.2.9.1, apply the patch, make install, restart |
| 11:52.49 | FlatFoot | ta |
| 11:53.03 | FlatFoot | help is much appreciated |
| 11:53.13 | burnproof | RoyK[at]: how can i initiate a transfer without losing the caller |
| 11:53.37 | burnproof | RoyK[at]: for e.g the user doesn't pick up the phone? |
| 11:54.05 | RoyK[at] | burnproof: i guess attended transfer should work |
| 11:54.21 | burnproof | RoyK[at]: attended transfer is not the solution since there's no real operator that will be permanently assign |
| 11:54.36 | RoyK[at] | dunno, then. sorryh |
| 11:54.38 | RoyK[at] | sorry |
| 11:54.44 | burnproof | RoyK[at]: heheh thanks a lot |
| 11:55.00 | burnproof | RoyK[at]: it really gives me a lot of headache :) |
| 11:55.20 | RoyK[at] | try asking on the ml |
| 11:55.32 | burnproof | RoyK[at]: ok i will thanks |
| 11:56.59 | *** join/#asterisk clive- (n=pirch@dsl-146-103-184.telkomadsl.co.za) |
| 11:57.59 | *** join/#asterisk tamp4x (n=tampon@64.201.13.51) |
| 11:58.03 | tamp4x | helllp |
| 11:58.20 | burnproof | RoyK[at]: this is the most frequently ask from the client perspective, and i think this should be fix ^_^ there's a similar question being asked on the bug tracker :) |
| 11:58.27 | tamp4x | when i try a zap command it does not work but asterisk compiled |
| 11:58.46 | r_evolution|afk | are you referring to like a call fwd'ing? |
| 11:58.50 | burnproof | tamp4x: did you compile libpri + zaptel |
| 11:58.53 | r_evolution|afk | or a roll-over line perhaps? |
| 11:58.59 | tamp4x | yes burn |
| 11:59.24 | burnproof | tamp4x: are you using trunk or stable release? |
| 11:59.38 | tamp4x | stable |
| 11:59.46 | r_evolution | burnproof... are you trying to roll-over a call? |
| 12:00.03 | r_evolution | i.e. when extension A is not answered after X rings, then you want to transfer to exten B? |
| 12:00.27 | burnproof | r_evolution: nope the call should be back to caller A |
| 12:00.53 | burnproof | r_evolution: User A answer the phone then Transfer to B if B is not answering it should be back to A |
| 12:00.56 | tamp4x | any ideas |
| 12:01.16 | burnproof | tamp4x: show modules like zap what is the output? |
| 12:01.18 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:01.18 | r_evolution | ooooo |
| 12:01.28 | tamp4x | hmm chan zap.so compiled |
| 12:01.46 | r_evolution | other than perhaps an attended transfer like RoyK suggested... unless you want to perhaps park the call? then no |
| 12:02.05 | tamp4x | only app_zap stuff |
| 12:02.15 | tamp4x | chan_zap isnt loading |
| 12:02.16 | burnproof | r_evolution: heheh yeah i know it's hard they don't buy my idea on parking a call |
| 12:02.33 | RoyK[at] | burnproof: which bug? |
| 12:02.39 | stoffell | I have an international call to italy that should connect to a cell phone, but the dialstatus in the log says "cancel", anyone else experienced this? (no other problems) |
| 12:02.41 | burnproof | r_evolution: since there will be no permanent person will be assign as _REAL_ operator |
| 12:03.04 | r_evolution | but i dont understand why you need an operator? |
| 12:03.25 | r_evolution | you've already GOT User A trying to transfer |
| 12:03.26 | burnproof | RoyK[at]: it's not a bug i think it's the design by asterisk |
| 12:03.31 | r_evolution | ^ |
| 12:03.36 | burnproof | r_evolution: ok here's the situation |
| 12:03.55 | burnproof | r_evolution: the company want's to assign a specific extension acting like an operator |
| 12:04.04 | [TK]D-Fender | r_evolution : Political question. My head office insists on always having a person answer the phone. In an age of VoIP / IVR / VR, you'd think someone would WAKE UP. |
| 12:04.26 | RoyK[at] | burnproof: not a bug it's a feature? i've heard that before... :) |
| 12:04.29 | Zerthimon | Anyone know how to handle the situation where a caller doesn't enter anything on Read ( ) ?? |
| 12:04.32 | tamp4x | nothing in the channels directory got copied to the modules directory wtf |
| 12:04.33 | burnproof | hehehehe |
| 12:04.37 | r_evolution | hehe TK ;x |
| 12:04.46 | r_evolution | oic burn |
| 12:04.50 | burnproof | :) |
| 12:04.57 | [TK]D-Fender | RoyK[at] : And Fatal Bugs are from this day forward to be called "SPECIAL Features" |
| 12:05.20 | burnproof | [TK]D-Fender: heheh yeah great |
| 12:05.22 | RoyK[at] | [TK]D-Fender: oxymoron of the day :) |
| 12:05.45 | [TK]D-Fender | I personally LOVE IVRs if they are properly designed. If I know exactly where I want to go the system should allow me to get there with as little intervention as possible. |
| 12:05.46 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 12:05.51 | burnproof | tamp4x: can you copy chan_zap.so then reload asterisk |
| 12:06.20 | RoyK[at] | in trunk, do you still have to do a _restart_ if changing zaptel configs? |
| 12:06.29 | burnproof | tamp4x: i mean copy it to /usr/lib/asterisk/modules |
| 12:06.42 | RoyK[at] | burnproof: just 'load chan_zap.so' |
| 12:06.48 | RoyK[at] | no need to restart/reload |
| 12:06.59 | [TK]D-Fender | RoyK[at] : I've long lived with the approach (corporately) that hardware failure = upgrade opportunity, and I tend to make all of my purchases a "cascading" upgrade for numerous people. |
| 12:07.00 | burnproof | RoyK[at]: hhehe yes |
| 12:07.11 | Zerthimon | [TK]D-Fender: how do I handle the situation where a caller doesn't enter anything on Read ( ) ?? |
| 12:08.49 | [TK]D-Fender | Zerthimon : How do you WANT to handle it? |
| 12:09.00 | tamp4x | it was my fault, because i ahd old modules so i did rm -f chan* |
| 12:09.21 | burnproof | tamp4x: ok |
| 12:09.25 | Zerthimon | [TK]D-Fender: I wan't to GotoIf ( ) if caller doesn't press anything |
| 12:09.41 | [TK]D-Fender | Zerthimon : pastebin what you've got for it. |
| 12:09.55 | Zerthimon | [TK]D-Fender: aight |
| 12:11.34 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:11.38 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 12:14.03 | Zerthimon | [TK]D-Fender: http://pastebin.ca/81347 |
| 12:18.14 | [TK]D-Fender | Zerthimon : After you have finished all your GotoIf's for valid options, do one to test for a null entry. hat will split you off so you can choose how you want to loop back |
| 12:19.14 | Zerthimon | [TK]D-Fender: GotoIf's don't work when the variable is empty |
| 12:19.36 | [TK]D-Fender | Zerthimon : you mean YOURS don't. :) |
| 12:19.37 | Zerthimon | how do I test a null entry ? |
| 12:19.47 | Zerthimon | yeah mine |
| 12:20.09 | [TK]D-Fender | Zerthimon : GotoIf($["${MYVAR}"=""]?10,20) |
| 12:20.38 | [TK]D-Fender | Zerthimon : Your problem was that when it evaled your variable it left NOTHING on that side of the = sign. |
| 12:21.01 | [TK]D-Fender | Zerthimon : Add quotes (or ANYTHING) to both ends and you'll know when its null. |
| 12:21.10 | Zerthimon | [TK]D-Fender: aye |
| 12:21.12 | RoyK[at] | [TK]D-Fender: s/(\d+)/label$1/ |
| 12:21.28 | [TK]D-Fender | RoyK[at] : I != RegExp. |
| 12:22.54 | RoyK[at] | [TK]D-Fender: I just meant 'use labels instead of bloody numbers' :) |
| 12:24.36 | [TK]D-Fender | RoyK[at] : Screw labels.... how often do I see dialplans complicated enough to warrant it? Easier to use alpha extens (like the STDEXTEN sample macro does), or just hard code (I don't use "n", leads to sloppy stupid things). If you can't do it right the first time, move over :) |
| 12:25.01 | mut | o_O |
| 12:25.26 | mut | i label almost every single line of my dialplan |
| 12:25.27 | [TK]D-Fender | "If at first you don't succeed, skydiving may not be for you" |
| 12:26.02 | mut | it's like 300 lines |
| 12:26.14 | burnproof | [TK]D-Fender: of course you will found yourself dead on the water :) joke |
| 12:26.36 | X-Rob_ | " (I don't use "n", leads to sloppy stupid things)" |
| 12:26.50 | rene- | my pri take inconming calls but no outgoing calls, why? |
| 12:26.54 | X-Rob_ | Like slitting your wrists after renumbering a 50 line macro 4 times? |
| 12:27.09 | mut | 50 line macro?! |
| 12:27.13 | [TK]D-Fender | mut : 300 lines? not so bad. One of my STDEXTEN's and its supporting setup macros was that big! |
| 12:27.35 | *** join/#asterisk Bert- (n=bert@LAubervilliers-151-12-81-84.w193-252.abo.wanadoo.fr) |
| 12:27.40 | Bert- | hello there :) |
| 12:27.43 | mut | not it's not too bad as i don't hardcode any extensions in mine |
| 12:27.47 | drray | HEY! I have dialplans that predate the N feature |
| 12:27.48 | mut | it's all NXXNXXXXX |
| 12:27.55 | [TK]D-Fender | X-Rob_ : Please refer to the "get it right the first time" part ;) |
| 12:28.08 | mut | whats the N feature anyway? |
| 12:28.16 | rene- | what s Z used for? |
| 12:28.26 | Bert- | what is the difference between the two extensions when I type sip show peers plz ? I mean 2010/2010 |
| 12:28.27 | X-Rob_ | [TK]D-Fender, and how does that avoid the 'customer requests additional things' issue? |
| 12:28.27 | [TK]D-Fender | mut : exten => s,n,NoOp() |
| 12:28.43 | burnproof | rene-> please check extensions.conf you will find what you are looking for |
| 12:28.46 | mut | like.. n = next? |
| 12:28.48 | rene- | X-Rob, punch? |
| 12:29.08 | rene- | burnproof thanks |
| 12:29.46 | r_evolution | n = nuts |
| 12:29.52 | r_evolution | :( |
| 12:29.53 | [TK]D-Fender | mut : Yup.. and welcome to * 1.2 ;) |
| 12:29.58 | r_evolution | heh... |
| 12:30.04 | r_evolution | ugh my eyes are fuzzy... :-\ |
| 12:30.05 | drray | heh |
| 12:30.10 | mut | *shrug* |
| 12:30.21 | mut | i don't keep up with things that aren't really that useful |
| 12:30.24 | rene- | This bad behaved PRI takes calls but wont place them, it runs ISDN and well there are some ISDN error messages but i kist cant figure them out, i mangled a little bit with zapata localdialplan, etc |
| 12:30.41 | drray | the problem is asterisk is a moving target |
| 12:30.53 | rene- | and manageg to get the telco tell me that the number you dialed has been disconnected |
| 12:30.59 | r_evolution | thats why we all just need to utilize the tin can w/ string method |
| 12:31.27 | rene- | s/kist/just/ |
| 12:31.33 | drray | I remember when they changed the kernel modules name |
| 12:31.33 | mut | can't make long distance calls to europe with that method tho |
| 12:31.36 | mut | the string gets too wet |
| 12:31.37 | r_evolution | seriously... the tin cans i have running around the office have 0 latency... 0 jitter... 0 echo |
| 12:31.44 | drray | 0 sound |
| 12:31.46 | r_evolution | who needs europe? |
| 12:31.51 | rene- | heh |
| 12:31.56 | r_evolution | who needs canada? |
| 12:32.00 | r_evolution | ;) |
| 12:32.05 | r_evolution | who needs anything father than like |
| 12:32.08 | r_evolution | 5 feet from you |
| 12:32.08 | drray | I do, they put codeine in the aspirin |
| 12:32.10 | mut | who needs anyone outside a 50 foot circle eh |
| 12:32.15 | r_evolution | YES! |
| 12:32.17 | r_evolution | exactly. |
| 12:32.26 | r_evolution | wait. where am I going to buy asprin? |
| 12:32.47 | r_evolution | man i've got a headache... |
| 12:32.57 | r_evolution | mmmm |
| 12:33.02 | rene- | who needs aspirin |
| 12:33.04 | drray | lightwieght |
| 12:33.14 | r_evolution | heh |
| 12:33.19 | rene- | real man hammers head against the wall |
| 12:33.27 | r_evolution | O_o |
| 12:33.34 | r_evolution | yeah fuck man... who NEEDS that dumb frontal lobe |
| 12:33.36 | drray | I also remember when they merged the fxo and fxs drivers into one module |
| 12:34.07 | rene- | real man grow another one when the old one is done |
| 12:34.08 | [TK]D-Fender | r_evolution : I'd rather have a bottle in front of me than a frontal lobotomy ;) |
| 12:34.09 | drray | back in the day |
| 12:34.52 | drray | I halfway expect them to disable the 4 key on the phone with a note in the source code that it has been deprecieated |
| 12:35.14 | r_evolution | kinda what im thinking TK |
| 12:35.52 | drray | buh bye 4 key |
| 12:36.36 | r_evolution | hah |
| 12:36.46 | r_evolution | fuck... i need that to dial my house :( |
| 12:36.51 | drray | move |
| 12:37.03 | r_evolution | would have to movefar |
| 12:37.05 | r_evolution | area code is 434 |
| 12:37.07 | r_evolution | ;) |
| 12:38.26 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 12:38.52 | rene- | since my ISDN line (e1) is not placing calls i gathered some debug output, does anyone care to look at them (http://pastebin.ca/81260) |
| 12:39.02 | Ahrimanes | r_evolution: lol |
| 12:39.09 | rene- | insert real man joke there ^ |
| 12:39.46 | r_evolution | real men don't place calls... they run jog over and speak with the person? |
| 12:40.53 | RoyK[at] | r_evolution: then it's good I'm not a 'real man' when I want to talk to my friend in .au |
| 12:40.56 | trelane_ | anyone recommend a good stable motherboard that will handle IRQ steering per slot for digium's hardware? |
| 12:41.11 | RoyK[at] | trelane_: anything ACPI |
| 12:41.16 | RoyK[at] | APIC, i mean |
| 12:41.18 | r_evolution | no kidding! it's good for me too! b/c I'd hate like hell to run the distance from FL to VA |
| 12:41.30 | trelane_ | RoyK[at], apic tends to cause timing issues for me |
| 12:41.38 | RoyK[at] | trelane_: it does? |
| 12:41.42 | RoyK[at] | works for me (tm) |
| 12:42.14 | RoyK[at] | MrChimpy: for doing what? |
| 12:43.06 | r_evolution | for bringing RoyK in... alive or dead O_O |
| 12:43.24 | MrChimpy | some addition to the dial app |
| 12:43.43 | [TK]D-Fender | rene- : " -- Nobody picked up in 30000 ms" |
| 12:43.54 | *** join/#asterisk esculapio__ (n=ESCulapi@200.88.44.66) |
| 12:44.02 | MrChimpy | oh. being able to specify a dialplan location to jump to when the L() limit expires |
| 12:44.17 | RoyK[at] | ah |
| 12:44.24 | [TK]D-Fender | rene- : "Message type: CALL PROCEEDING (2)" |
| 12:44.34 | [TK]D-Fender | rene- : Looks like it dialed and nobody answered... |
| 12:44.53 | [TK]D-Fender | MrChimpy : Once you gain tenure you get the big $ ;) |
| 12:45.31 | Ahrimanes | hm |
| 12:46.02 | Ahrimanes | "Jul 7 14:35:57 WARNING[82702] pbx.c: No application 'mcc2' for extension (from-sip, 00441534284726, 15)" <- i have an application mcc2 installed and loaded.. could this be due to a call arriving exactly during a reload |
| 12:46.06 | rene- | [TK]D-Fender: just a no answer? did something that simple keep three ppl awake all night? i |
| 12:46.31 | rene- | i was thinking it was something regarding pridialplan and such things |
| 12:47.37 | rene- | and we were dialing to another pbx we had in the same room via the PSTN and it never worked |
| 12:48.43 | [TK]D-Fender | rene- : LOOKS fine to me.... |
| 12:49.02 | Bert- | There is something I really don't understand |
| 12:49.05 | Bert- | I read doc |
| 12:49.10 | rene- | [TK]D-Fender: thanks |
| 12:49.16 | Bert- | but I still not understand |
| 12:50.33 | Bert- | I have 10 people, each with a softphone I have extensions for them (like user1 : 333). but I would like to see their name instead of their extensions with the sip show peers cmd. I tried to declare them as agents. but now they are unable to login. |
| 12:50.57 | Bert- | I think I REALLY don't understand agents at all :( |
| 12:51.06 | RoyK[at] | MrChimpy: hm... that kind of sounds a good idea :) |
| 12:52.11 | burnproof | Bert-: what part of agent you don't understand? |
| 12:52.24 | *** join/#asterisk nortex (n=nortex@64.136.65.142) |
| 12:52.44 | burnproof | Bert-: what it has something to do with sip show peers? could you please elaborate more? |
| 12:52.53 | Bert- | yep |
| 12:53.16 | [TK]D-Fender | Bert- : Agents are for QUEUES. |
| 12:54.09 | Bert- | My boss have choosen extensions for employees. Like that : Alexandre RELOU 2328 AREL |
| 12:54.39 | Bert- | If I declare a user in my sip.conf like [2328], all is fine |
| 12:54.54 | Bert- | ubt unable to associate [AREL] with an extensions |
| 12:55.05 | Bert- | so I understand [xxx] MUST be extension to dial |
| 12:55.26 | Bert- | [TK]D-Fender I gave to use queue, because of IVR |
| 12:55.36 | Bert- | when an options rings all phones |
| 12:55.52 | Bert- | In fact |
| 12:56.11 | Bert- | I would like to use agents and be able to call them without user any [222] declaration in my sip.conf |
| 12:56.25 | Bert- | user -> use |
| 12:57.37 | Bert- | You agree agent is a "real" human |
| 12:57.38 | Bert- | ? |
| 12:57.48 | [TK]D-Fender | Bert- : why can't you make an exten for [AREL]? Normally it'd be like "exten => 222,1,Dial(SIP/AREL,30)" |
| 12:58.16 | [TK]D-Fender | Bert- : I still don't see how queues are requiren here... |
| 12:58.49 | Bert- | [TK]D-Fender because I use queue for IVR like that : press 0 to reach an operator. Then when 0 is pressed, several phones will ring |
| 12:59.04 | *** join/#asterisk jetway2008 (n=asd@218.111.11.189) |
| 12:59.11 | Bert- | I sue queue to easyly add ore remove agent from this menu |
| 12:59.22 | jetway2008 | hi |
| 12:59.23 | [TK]D-Fender | Bert- : Dial(SIP/100&SIP/200&SIP/300,30) <------------- |
| 12:59.39 | jetway2008 | what does Forward Disconnect Supervision mean? |
| 12:59.48 | [TK]D-Fender | Bert- : You can use dialplan logic to change who you dail rather easily. |
| 12:59.53 | Bert- | [Tk] and when a new employee comes, I have to add him in alls my dialplans ?? |
| 12:59.54 | Bert- | boring |
| 13:00.01 | Bert- | with queue, I only add or remove it |
| 13:00.31 | Bert- | in fact my IVR works like that : you type direct extensions to directly call one people |
| 13:00.42 | Bert- | if you use menu options, then you enter in a queue |
| 13:00.46 | Bert- | and concerned phones rings |
| 13:00.47 | [TK]D-Fender | Bert- : No, there are other ways.... |
| 13:01.05 | Bert- | <[TK]D-Fender> Bert- : Dial(SIP/100&SIP/200&SIP/300,30) <------------- what about my 22 users ? |
| 13:01.39 | [TK]D-Fender | Bert- : You want to call 22 people? So you are going to make 50 logins so that when you grow you just tell the next person "you are agent 44 now"? |
| 13:02.27 | [TK]D-Fender | Bert- : if its a legit queue ok... |
| 13:02.55 | [TK]D-Fender | Bert- : For mapping the agent logins to "sip show peers" you're going to have to make your own app to do that. |
| 13:03.20 | hwt | i get these after a while: Jul 7 15:01:56 ERROR[24436]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql: Unknown connection error: (2006) MySQL server has gone away |
| 13:03.27 | hwt | any way to "keepalive" mysql? |
| 13:03.50 | QuAtRo[NL] | mysql_pconnect ? |
| 13:03.51 | Bert- | [TK]D-Fender okay |
| 13:04.03 | Bert- | yep |
| 13:04.10 | Bert- | <[TK]D-Fender> Bert- : You want to call 22 people? So you are going to make 50 logins so that when you grow you just tell the next person "you are agent 44 now"? |
| 13:04.15 | jetway2008 | what does Forward Disconnect Supervision mean? |
| 13:04.56 | Bert- | and her extensions will depends of his name (eg marie gonzales will be MGON ->6466 |
| 13:07.15 | Ahrimanes | e-ddie: hey |
| 13:07.19 | e-ddie | hej |
| 13:07.27 | [TK]D-Fender | Bert- : Wait a sec.... you want to do "dial by name"? |
| 13:08.42 | burnproof | [TK]D-Fender: by the way that's one of my TODO list :) how can i accomplished that dial by name stuff? |
| 13:09.29 | *** join/#asterisk Eecplat (n=ouarf@AStDenis-105-1-25-146.w81-248.abo.wanadoo.fr) |
| 13:09.53 | *** join/#asterisk dynia80 (n=chris@62.232.50.3) |
| 13:10.21 | *** join/#asterisk paolob-parroquia (n=paolob-p@pri-214-b7.codetel.net.do) |
| 13:10.47 | *** join/#asterisk Katty (n=aisaacs@64.82.232.54) |
| 13:11.04 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 13:11.24 | dynia80 | hi, just finished setting up TE110P <--ISDN PRI trunk--> AVAYA G3 and everything seems to be working ok, except AVAYA switch is not sending CallerID to asterisk box. Any ideas? |
| 13:11.32 | *** join/#asterisk NeonLevel (i=NeonLeve@201.145.221.21) |
| 13:11.50 | paolob-parroquia | Hi guys! I have a statemente language=es in sip.conf, but it isn't honored when calling the stations. Sound files are looked for in the english sounds directory. Anyone could help me? |
| 13:12.10 | *** join/#asterisk ariel_ (n=Ariel@70.46.87.158) |
| 13:12.12 | NeonLevel | necesitas crear el directorio es dentro de sounds |
| 13:12.12 | Katty | morning |
| 13:12.14 | NeonLevel | con eso |
| 13:12.17 | Katty | herro, ariel_ |
| 13:12.28 | *** join/#asterisk postel_ (n=jp@unaffiliated/postel) |
| 13:12.30 | Bert- | no I don't want dial by name |
| 13:12.32 | paolob-parroquia | NeonLevel, it's there |
| 13:12.45 | Bert- | it is just a tip if you don't remember extensions for a personn |
| 13:12.51 | ariel_ | morning everyone |
| 13:13.23 | *** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net) |
| 13:13.40 | Qb3rt | here is my problem ----> http://pastebin.ca/81370 |
| 13:13.48 | Bert- | but I think I will declare people as agents, and then manually add dialplans for internal calls ... |
| 13:13.51 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 13:13.56 | Bert- | hope w'll never grow up to thousands :d |
| 13:13.58 | [TK]D-Fender | paolob-parroquia : pastebin your dialplan |
| 13:13.58 | hwt | QuAtRo[NL]: well, it's asterisk that fails to use a persistent connection. |
| 13:13.59 | [TK]D-Fender | ~pb |
| 13:14.04 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 13:14.04 | hwt | QuAtRo[NL]: it times out. |
| 13:14.27 | dynia80 | Hi everyone! I have a problem with AVAYA G3 switch not passing CallerID over ISDN PRI (E1) trunk to asterisk box with TE110P |
| 13:14.58 | [TK]D-Fender | dynia80 : We heard you the first time.... |
| 13:15.15 | dynia80 | oops, sorry about that |
| 13:15.22 | QuAtRo[NL] | hwt: Wherefore do you need it? |
| 13:18.18 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 13:18.33 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:18.33 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:18.36 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 13:18.52 | knarfly | Hey I just got my remote X-Lite to authenticate with my * server and make some calls. Trouble is I can't hear audio on this end. Any suggetstions how to solve one-way audio issue? |
| 13:19.27 | hwt | QuAtRo[NL]: because of this: Jul 7 15:01:56 ERROR[24436]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql: Unknown connection error: (2006) MySQL server has gone away |
| 13:19.34 | hwt | QuAtRo[NL]: the connection to the mysql server times out. |
| 13:19.44 | MrChimpy | hmm. odd. I have my inbound E1 calls going to a default dialplan context where I have a list of DDIs defined as extensions. Thing is exten => i, doesn't seem to do anything - unrecognised DDIs are just dropped with "Extension '4642607' in context 'default' from '8707886753' does not exist. Rejecting call on channel 0/1, span 1" - doesn't even touch the dialplan. |
| 13:19.55 | hwt | knarfly: behind NAT? |
| 13:20.03 | hwt | knarfly: in that case, set nat=yes in you sip.conf |
| 13:20.42 | *** join/#asterisk tafazzi (n=Dario@85-18-114-176.ip.fastwebnet.it) |
| 13:21.19 | [TK]D-Fender | MrChimpy : you can bypass that by using _X. in your inbound context, then attmpting a GOTO into another. if the Goto fails it will execute the next priority allowing you to capture the "unknown" DID. |
| 13:21.29 | paolob-parroquia | [TK]D-Fender, apparently pastebin.com doesn't perform the upload... |
| 13:21.42 | knarfly | Yes, both the * server and the phone are behind NAT's but on separate networks. I have set nat=yes in sip.conf |
| 13:21.54 | [TK]D-Fender | paolob-parroquia : us pastebin.ca . the .com one doesn't work so well lately. |
| 13:22.02 | hwt | knarfly: qualify=yes? could be nat table timeout |
| 13:22.23 | MrChimpy | TKD: thanks |
| 13:22.25 | [TK]D-Fender | knarfly : pastebin your [general] section of SIP.CONF. |
| 13:22.38 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.236.230.Dial1.SanJose1.Level3.net) |
| 13:22.49 | knarfly | hwt: Should I set nat=yes in the [general] section or under the individual phone settings? |
| 13:23.32 | Katty | morning fender (= |
| 13:23.39 | paolob-parroquia | [TK]D-Fender, http://pastebin.ca/81377 |
| 13:23.45 | knarfly | [TK]D-Fender: Can you help me with pastebin. I've never used it before. |
| 13:24.07 | hwt | knarfly: well, that depends on how you want it. general=default settings, individual phone settings overrides settings in general. |
| 13:24.15 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.236.230.Dial1.SanJose1.Level3.net) |
| 13:24.23 | [TK]D-Fender | Katty : Mew. |
| 13:25.31 | [TK]D-Fender | knarfly : just go to www.pastebin.ca and copy/paste your sip.conf's [general] section there. |
| 13:25.50 | paolob-parroquia | [TK]D-Fender, the problem is when I park a call: it gives me the parked extension in english |
| 13:26.22 | paolob-parroquia | [TK]D-Fender, and the moh file: it looks for it in the english dir |
| 13:26.33 | paolob-parroquia | other messages come out in spanish |
| 13:27.41 | [TK]D-Fender | paolob-parroquia : So SOME files work, others don't? |
| 13:28.25 | paolob-parroquia | [TK]D-Fender, when I transfer a call, it say it in spanish, but when I park a call not, and the sound file isn't looked for in the spanish dir |
| 13:29.17 | [TK]D-Fender | paolob-parroquia : pastebin the CLI output of a call that shows this in action. |
| 13:29.21 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 13:29.32 | Dr-Linux | [TK]D-Fender: can i use a "name" instead of "number" as agentid in agents.conf? |
| 13:31.15 | FlatFoot | anyone recommend a good door entry system to add to * ( UK ) |
| 13:31.41 | *** join/#asterisk evilmnky (n=evilmnky@69-163-156-13.atlsfl.adelphia.net) |
| 13:31.56 | paolob-parroquia | [TK]D-Fender, http://pastebin.ca/81400 |
| 13:32.09 | rene- | D-Fender, i m still getting no body pickup in 30000ms but i hear telco messages ' the number you dialed is incorrect, please try again' |
| 13:32.58 | Qb3rt | in my asterisk server the sound stop once per month and i have to reboot the server to make the sound work again ----> http://pastebin.ca/81370 |
| 13:33.22 | knarfly | [TK]D-Fender: Okay, I think I did that right! |
| 13:35.06 | [TK]D-Fender | Dr-Linux : How would you type a name inside of AgentLogin, or AgentCallbackLogine? |
| 13:35.27 | knarfly | [TK]D-Fender: Sorry...Now I think I did it right ... http://pastebin.ca/81411 |
| 13:35.45 | *** join/#asterisk darkfiber1 (n=bwarner@65.113.208.21) |
| 13:35.55 | [TK]D-Fender | paolob-parroquia : And which line shows a non-es file being played? |
| 13:36.23 | paolob-parroquia | [TK]D-Fender, the digits 7,0,1 after parking the call |
| 13:36.32 | Dr-Linux | [TK]D-Fender: that's why i'm doing, i don't want my agents to dial, they will login to the queue via web interface, so in that case they will enter their name and password and they will be logged in. |
| 13:36.41 | paolob-parroquia | [TK]D-Fender, Playing 'digits/7' (language 'en') |
| 13:36.48 | [TK]D-Fender | knarfly : You need NAT=YES in [general], also I'm not sure localnet accepts CIDR. |
| 13:37.04 | darkfiber1 | getting warning chan_sip.c:695 retrans_pkt on call.....anyone know what might be causing this (incoming and outgoing calls work fine) |
| 13:37.15 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 13:37.22 | [TK]D-Fender | paolob-parroquia : You have a folder called "es" inside the DIGITS folder? |
| 13:37.54 | knarfly | [TK]D-Fender: Sorry...should I change it to 10.0.0.0/255.0.0.0? |
| 13:38.00 | paolob-parroquia | [TK]D-Fender, /usr/share/asterisk/sounds/digits/es/ |
| 13:38.55 | [TK]D-Fender | paolob-parroquia : So this happens when you PARK? |
| 13:39.17 | [TK]D-Fender | paolob-parroquia :I think you should be setting the language for each user in SIP.CONF (or whatever) as well... |
| 13:39.30 | [TK]D-Fender | knarfly : I'd try that... What else isn't working? |
| 13:39.33 | paolob-parroquia | [TK]D-Fender, yes, and when a moh in invoked |
| 13:39.56 | paolob-parroquia | [TK]D-Fender, "for each user" means for each extension? |
| 13:41.05 | paolob-parroquia | [TK]D-Fender, in extensions.conf I have a "include => parkedcalls". Where is the parkedcalls context defined? |
| 13:41.14 | nortex | I am getting ready to build 8 phone systems for office that have 2 to 4 employees and are all part of a larger company. I planned on using Dell Optiplex machines that have new warranties and what not. My question is would Digium TDM400 or a pair of them in some cases, or use the Sangoma A200 cards? |
| 13:41.31 | [TK]D-Fender | paolob-parroquia : I don't think language applies to MoH, and you do "language=es" in [general] I think to apply it globally. |
| 13:42.50 | nortex | That sounds odd, I mean which cards would be recommended. |
| 13:44.13 | [TK]D-Fender | nortex : you mean several seperate PBX's located at customer premesis? |
| 13:44.27 | knarfly | [TK]D-Fender: It's still a no go. I hear the ringing and I can see my * cli console at home answering the call and sending to VM. But nothing is heard on this end. |
| 13:44.52 | [TK]D-Fender | knarfly : Ah, no audio. What ports are you forwarding? |
| 13:45.34 | nortex | [TK]D-Fender, No, we have 8 remote offices, sales office, and each one has 2-4 employees and will get their own * PBX |
| 13:46.25 | [TK]D-Fender | nortex : Physically located at their site then? |
| 13:46.33 | knarfly | [TK]D-Fender: All the calls I make work. Even the call through my PSTN provider got through to my office phone. I can hear audio through my desktop phone but when I speak into it I do not hear it back on my PC speakers. |
| 13:46.40 | nortex | [TK]D-Fender, yes. |
| 13:47.11 | [TK]D-Fender | nortex : Well if you need basic voice, Sangoma's are a lot more forgiving about your choice of system. |
| 13:47.27 | *** join/#asterisk festr_ (n=festr@ns.regnet.cz) |
| 13:47.31 | festr_ | hello |
| 13:47.57 | [TK]D-Fender | nortex : And scale better as well. Pricing scales cheaper than TDM400 without increasing IRQ load. |
| 13:48.09 | paolob-parroquia | [TK]D-Fender, in extensions.conf I have a "include => parkedcalls". Where is the parkedcalls context defined? |
| 13:48.48 | nortex | [TK]D-Fender, Are there any issues with Sangoma A200 cards? |
| 13:49.03 | knarfly | [TK]D-Fender: 2727, 4569, 5060 thru 5082, 8000 thru 20000, |
| 13:49.25 | dynia80 | EXIT |
| 13:49.26 | dynia80 | QUIT |
| 13:49.28 | dynia80 | HELP |
| 13:49.29 | [TK]D-Fender | nortex : Don't work so great for faxing right now, but that most of it. |
| 13:49.29 | festr_ | i've locked two channels (asterisk 1.2 revision about 35xxx) in ring state for 220hour. Data between thouse sip clients and pbx doesnt show any RTP data nor sip signalization. whats wrong? |
| 13:49.38 | festr_ | how to debug this? is it debuggable? |
| 13:49.39 | festr_ | :) |
| 13:49.59 | [TK]D-Fender | knarfly : Could be your router. Several D-LInk's have been reported to be screwy in their NAT implementation. |
| 13:50.32 | knarfly | [TK]D-Fender: My router is a FreeBSD-6.0 server running IPFW and NATD |
| 13:50.46 | kay2 | [TK]D-Fender: do you know a stressing tool for asterisk? so i can see how many concurrent calls it could handle ? |
| 13:51.06 | nortex | [TK]D-Fender, It llokms like the Digium cards give you the flexibility of a module per port, where the Sangoma has 2 ports per module. What kind of faxing problem, can it bridge a incoming fax to a fxs port? |
| 13:52.55 | [TK]D-Fender | knarfly : well if you added nat=yes, and you IP's are right I'm not sure what to say... turn on SIP debug, make a call and pastebin it. |
| 13:53.26 | [TK]D-Fender | nortex :If you would simply bridge it, you are better off leaving faxes out of the picture 100%. |
| 13:54.19 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 13:54.38 | [TK]D-Fender | nortex : Yes TDM400 scales per port to 4 but uncurs a heavy interrupt load, and is really picky about IRQ's, etc. You are strongly recommended not to pass 2 cards in a server whereas with the A200 you can scale up at no penalty. |
| 13:55.45 | nortex | [TK]D-Fender, Well, I know in one office we would like to try and do fax detection on a third line and that way it could work as a voice line if the others are busy. The telco does the hunt group on all 3 lines already. |
| 13:56.48 | nortex | [TK]D-Fender, Would the Sangoma have problems if I used rxfax to accept the fax? |
| 13:56.51 | [TK]D-Fender | nortex : well TDM400P look cheaper for that solution. |
| 13:57.29 | [TK]D-Fender | nortex : I hear faxing is a little hit/miss right now. I don't have any real complaints on my A104d, but others have noted it. |
| 13:57.41 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 13:57.51 | *** join/#asterisk kcio (n=cassio@c92509a4.rjo.virtua.com.br) |
| 13:57.53 | [TK]D-Fender | nortex : Analog faxing sucked on it, but I'm dealing with PRI. RxFAX seems pretty decent |
| 13:57.59 | kcio | guys what is this chan_zap.c:11062: error: dereferencing pointer to incomplete type |
| 13:58.20 | knarfly | [TK]D-Fender: Got something here. My firewall rules open up 5062 but I suddenly saw a packet being dropped on that port. I don't undertsand this well enough but I added a new rule to open the firewall wide open and now the audio works. I just need to figure out a new rule to allow this traffice. The existing rule I have isn't cutting it. |
| 13:58.57 | nortex | [TK]D-Fender, I know our 104D here has destroyed faxes, and so we split most of the fax lines out of our pri. |
| 13:59.15 | nortex | [TK]D-Fender, I hoped the analog was more stable |
| 13:59.20 | [TK]D-Fender | knarfly : Every time I hear people going "Security is super important" I hear them screw themselves with firewalls and *. |
| 13:59.43 | [TK]D-Fender | nortex : I ran into the same scenario. RxFax DOES seem ok though. |
| 14:00.00 | [TK]D-Fender | nortex : But shifted off our primary analog fax machines. |
| 14:00.39 | knarfly | [TK]D-Fender: Security is like a hot woman...can't live with 'em....can't live without 'em... |
| 14:01.04 | *** join/#asterisk odrm (n=ckarg@83.148.181.218) |
| 14:01.17 | kcio | anyone can help me with that? |
| 14:01.18 | [TK]D-Fender | knarfly : More like "liable to explode for little reason and less notice and some its always your fault. |
| 14:01.21 | rob0 | Ralf the dog: "There's somethin' irresista-bullish about 'em!" |
| 14:01.27 | [TK]D-Fender | somehow* |
| 14:01.33 | darkfiber1 | anyone familiar with the error: WARNING[15689]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on ... |
| 14:01.46 | rob0 | (original _The Muppet Movie_ ) |
| 14:01.48 | nortex | [TK]D-Fender, So if I have to keep the fax machine then I should look at a dedicated line and then Sangoma cards for the ease of use and scalibility. |
| 14:02.39 | [TK]D-Fender | nortex : yup. |
| 14:02.45 | nortex | [TK]D-Fender, Have you tried an Astribank? |
| 14:03.00 | odrm | could someone please offer some basic (i.e. n00b-digestible) advice on getting a Voicetronix OpenLine4 card recognised by asterisk? driver & asterisk compiled fine & running. How do I tell asterisk that the vpb card exists? |
| 14:03.10 | knarfly | rob0: I saw that one...his line about my problem is women...kermit says "Why don't you leave them alone?" Ralphs reply..."Well....that's my problem...." |
| 14:03.24 | [TK]D-Fender | nortex : I mean you could get an FXS for it, and I presume bridging would be as reliable, but in the case of a 3 line co like you sampled it wouldn't be cost effective. |
| 14:03.47 | [TK]D-Fender | nortex : that'd be 2 remora frames, 2 FXO, 1 FXS, with 2 ports effectively unused. |
| 14:04.06 | rob0 | That is a really great movie. One of my top ten favorites. |
| 14:04.23 | pdtmobile | nortex: I started using IAXModem + Hylafax instead of rxfax the other day and haven't had any of the problems I was having with rxfax |
| 14:04.42 | [TK]D-Fender | nortex : EW.... wouldn't touch. |
| 14:04.55 | [TK]D-Fender | nortex : USB + limited to *? Ick. |
| 14:05.31 | nortex | [TK]D-Fender, True, but an extra fxs and fxo could be used fo things like paging or analog cordless phones. |
| 14:05.37 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 14:05.42 | jbalcomb | [TK]D-Fender: YIPPY!! phone server down! two hours to bring up the back up! BLAH!! |
| 14:06.07 | [TK]D-Fender | nortex : yes, but I am trying to think based on what you SAID you want. Cost matters to many. |
| 14:06.18 | [TK]D-Fender | jbalcomb : 2 hours? WTF?! |
| 14:06.34 | *** part/#asterisk kcio (n=cassio@c92509a4.rjo.virtua.com.br) |
| 14:06.40 | nortex | [TK]D-Fender, It does, thanks for the advice and help. |
| 14:06.53 | [TK]D-Fender | nortex : np |
| 14:07.13 | Sonderblade | it would be cool if there was some free software hardphone |
| 14:08.25 | [TK]D-Fender | Sonderblade : ......huh? |
| 14:08.28 | rene- | pa1688 |
| 14:08.33 | rene- | based devices |
| 14:08.35 | ManxPower | Sonderblade, search for PA168 |
| 14:09.01 | rene- | Manx Have you seen the IAX+modem hardphone? |
| 14:09.03 | rene- | it is cool |
| 14:09.06 | rob0 | Sonderblade probably means the firmware would be Free Software |
| 14:09.27 | tzafrir_laptop | [TK]D-Fender, limited to Asterisk? last time I heard some other programs supported zaptel drivers |
| 14:09.36 | Sonderblade | yes, then you wouldn't have to deal with manufacturers crappy bug-ridden firmware |
| 14:09.37 | rene- | rob0 isnt that the case with Pa1688? |
| 14:09.46 | rob0 | I don't know |
| 14:09.52 | rene- | as in thats the reason it runs IAX |
| 14:10.03 | tzafrir_laptop | As for USB: it happens to actually work. And does not require opening any case to install... |
| 14:11.34 | [TK]D-Fender | tzafrir : Astribank is a USB only device no? |
| 14:11.41 | tzafrir_laptop | Sonderblade, go andbuild one. The standard uclinux distro now include linphone |
| 14:11.50 | tzafrir_laptop | [TK]D-Fender, right |
| 14:11.52 | [TK]D-Fender | PA168(x) = POS |
| 14:13.06 | [TK]D-Fender | tzafrir : I like standards compliant stuff... Sangoma works on pretty much everything. SIP is portable. In my opinion the best thing about * my telephony solutions is... T REPLACABILITY. |
| 14:13.21 | Sonderblade | tzafrir: what kind of hw do i need for that? |
| 14:13.52 | [TK]D-Fender | tzafrir : No piece of my solutions owns me and I can take all the hardware to any platform I want and change the PBX core as I please. Sure I love *... for NOW. who knows down the road? |
| 14:13.54 | tzafrir_laptop | Sonderblade, frankly: never tried it with any embedded hardware |
| 14:13.56 | jbalcomb | [TK]D-Fender: yeah, fun stuff. |
| 14:14.14 | [TK]D-Fender | jbalcomb : So how the hell does it take 2 hours to bring up a friggen server? |
| 14:14.18 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 14:14.49 | *** join/#asterisk coppice (n=chatzill@61.197.17.210.dyn.pacific.net.hk) |
| 14:14.50 | Sonderblade | if you could use a cheap grandstream phone and upload your own firmware that would have been nice |
| 14:14.57 | Sedorox | http://nymph.daforest.org:6937/uncached/21.5E7913000000/temperature |
| 14:15.01 | Sedorox | errrr sorry |
| 14:15.36 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:16.20 | *** part/#asterisk Ritalin- (i=dave@c-68-47-199-178.hsd1.tn.comcast.net) |
| 14:16.30 | jbalcomb | [TK]D-Fender: switches IPS, lines, copying VM, voice promts, IVRs, rebooting phones, re-adding routes that weren't persistent for some reason.. |
| 14:16.56 | [TK]D-Fender | yOU MEAN REBUILDING A SERVER FROM SCRATCH? |
| 14:21.13 | kay2 | [TK]D-Fender: do you know a stressing tool for asterisk? so i can see how many concurrent calls it could handle ? |
| 14:21.27 | jbalcomb | [TK]D-Fender not quite but it certainly wasn't ready to be switched over. |
| 14:21.32 | [TK]D-Fender | kay2 : no idea |
| 14:21.54 | [TK]D-Fender | jbalcomb : Diet-Fun! All the pain, none of the fun! Get yours NOW!~ |
| 14:22.14 | *** join/#asterisk _problem_ (n=lokesh_k@estrela.nortenet.pt) |
| 14:22.18 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 14:22.20 | *** part/#asterisk tafazzi (n=Dario@85-18-114-176.ip.fastwebnet.it) |
| 14:22.54 | *** join/#asterisk Soulbane (n=Sunforge@202.3.110.33) |
| 14:28.04 | knarfly | Bingo...found it. My firewall rules did not include opening up for port 5062. It's working great now. Just called my office phone from the * server at home and moh cane through clear and crisp. Thanks everyone! |
| 14:30.58 | smackus | I know the issue keeps coming up, but what is the deal with Avoided initial deadlock for...? |
| 14:31.12 | smackus | what really causes that? I am getting upwards of 5 of those per minute. |
| 14:31.40 | smackus | I have read some bugs that say that is fixed in version 1.2 and such, but I am getting them in 1.2.9.1 |
| 14:31.56 | smackus | http://bugs.digium.com/view.php?id=7226 |
| 14:34.12 | *** join/#asterisk Pepse (n=pepse@ip68-109-169-37.ph.ph.cox.net) |
| 14:34.25 | ManxPower | smackus, "latest code branch" does NOT mean 1.2.9.1 |
| 14:34.43 | ManxPower | it means SVN checkout of the latest dev branch for 1.2 |
| 14:35.00 | smackus | true... but do I want to be running a production system on a dev branch? |
| 14:35.25 | ManxPower | smackus, the branch I am referring to will become 1.2.10.0 |
| 14:35.39 | ManxPower | It's usually fairly stable. |
| 14:35.55 | ManxPower | and I imagine it would be better than your current situation. |
| 14:36.49 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.146) |
| 14:36.55 | ManxPower | the download page or #asterisk-dev should be able to tell you how to get the correct branch |
| 14:37.01 | smackus | ok, i got bitch slapped for using anything besides the released version, so please forgive my hesitation. anyone else want to chime in and give advice on this? [TK]D-Fender: any advice? you are my favorite bitch slapper! :-D |
| 14:37.59 | smackus | I would have called you a flamer for flaming me when i am dumb, but that didnt sound quite right |
| 14:38.49 | smackus | http://bugs.digium.com/view.php?id=6997 This seems to describe my issue to the "T" |
| 14:38.56 | ManxPower | smackus, only run a SVN branch if you MUST. |
| 14:39.17 | smackus | ok, thats what I wanted to hear. Now I have to figure out if I must |
| 14:40.21 | [TK]D-Fender | smackus : Sorry, nothing to contribute on this one... |
| 14:52.22 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:52.27 | *** join/#asterisk hohum (n=dcorbe@mail.interceltelecoms.com) |
| 14:56.25 | knarfly | Okay...next stupid question...how do I properly exit from * when using ssh and I started the * session from the ssh session. If I type exit at the server prompt it will just dump me back into *. |
| 14:57.53 | ManxPower | knarfly, What command did you use to start Asterisk? |
| 14:58.04 | tzafrir_laptop | when you exit the terminal, ssh notifies its dauter processes that their terminal is gone by sending SIGHUP . Asterisk will probably take that as a signal to reload... |
| 14:58.55 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 14:59.06 | PakiPenguin | evening |
| 15:03.17 | ManxPower | I guess knarfly doesn't really want help. |
| 15:03.53 | SplasPood | Question for all... if you were going to be handling some pure SIP routing, no PBX style features.. just SIP in and SIP out... with some LCR in the mix, what software would you use? OpenSER? |
| 15:04.55 | ManxPower | SplasPood, *I* would use Asterisk, but OpenSER might be a better choice. |
| 15:05.39 | SplasPood | ManxPower: *I* would want to do the same, cause OpenSER confuses the hell out of me... but I get the feeling there's a better choice like you suggested |
| 15:06.36 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 15:13.23 | *** join/#asterisk redondos (n=redondos@190.48.1.154) |
| 15:13.40 | redondos | Good morning. I need some insights about manipulating CDR logs to create monthly reports. |
| 15:13.51 | redondos | What do you guys use for this task? |
| 15:14.00 | drray | perl/sed/awk |
| 15:14.18 | redondos | drray: Can I see your scripts, please? |
| 15:14.57 | drray | got an email? |
| 15:15.41 | redondos | redondos [at] gmail [dot] com |
| 15:15.48 | redondos | Thanks in advance, man. |
| 15:15.57 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 15:16.19 | drray | I'll need to sanitize it |
| 15:16.26 | drray | but I'll send it to you |
| 15:16.33 | redondos | drray: Sounds great. :) |
| 15:16.37 | *** part/#asterisk odrm (n=ckarg@83.148.181.218) |
| 15:17.01 | redondos | Continuing my analysis, what I need is to create different reports. One per customer. |
| 15:17.28 | redondos | Do your scripts cover this, drray? (Send them anyway if they don't, I'll see if I can modify them accordingly) |
| 15:17.54 | *** join/#asterisk EinsteinTaylor (n=dtaylor@216.243.100.29) |
| 15:18.01 | EinsteinTaylor | morning all |
| 15:20.03 | redondos | morn |
| 15:20.11 | *** join/#asterisk pdtmobile (n=ptinsley@209.12.249.243) |
| 15:21.02 | EinsteinTaylor | quiet this morning... |
| 15:21.17 | EinsteinTaylor | figured by now we'd all have a nice caffeine buzz goin |
| 15:21.25 | redondos | Perhaps you can help me creating per-customer reports? :) |
| 15:21.56 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 15:22.18 | Toerkeium | redondos: stop bein lazy.. what are you? from Argentine? :9= |
| 15:22.26 | redondos | Toerkeium: Errr... no! |
| 15:22.35 | Toerkeium | ;)) |
| 15:22.37 | redondos | :) |
| 15:22.38 | St1ckm4n | drray: i have a little experience with dumping the queue logs to a db and pulling the info, is this what you're looking at? |
| 15:22.54 | EinsteinTaylor | i'm probably the wrong guy to ask...i'm pretty good with systems and code but just getting my feet wet with asterisk |
| 15:23.18 | redondos | Well can you at least tell me what format you present your reports in? Plaintext? Stylesheets? HTML? |
| 15:23.48 | redondos | EinsteinTaylor: Hope you don't electrocute yourself int he attempt. |
| 15:24.12 | EinsteinTaylor | indeed...although i have found red bull and sunflower seeds usefull to the endeavour |
| 15:24.54 | redondos | You drop some seeds inside a red bull can and drink it? Is it supposed to give you even more energy? |
| 15:25.01 | EinsteinTaylor | lol :) |
| 15:25.07 | EinsteinTaylor | i'm willing to try anything |
| 15:25.13 | redondos | Heh. |
| 15:25.26 | EinsteinTaylor | right now the only production asterisk we have is voicemail |
| 15:25.31 | EinsteinTaylor | have about 1000 clients on it |
| 15:25.45 | EinsteinTaylor | i have a few sip phones going but it's just testing right now |
| 15:25.55 | Toerkeium | there is a web site with addons for * ? |
| 15:26.07 | Toerkeium | aparts from homepage |
| 15:26.12 | Toerkeium | -s |
| 15:26.16 | EinsteinTaylor | most distro's have a package with asterisk extras |
| 15:26.45 | Toerkeium | I am searching something like.. hmm not sure how to call it in english.. |
| 15:27.09 | Toerkeium | the machine homer simpson found that calls everyone automatically ? |
| 15:27.11 | Toerkeium | ;) |
| 15:27.40 | redondos | Hrm. |
| 15:27.46 | redondos | That sounds annoying. |
| 15:27.50 | redondos | :) |
| 15:28.05 | St1ckm4n | drray: have you looked into using asterisk guru queue statistics, it creates some decent reports and lets you pick the queue, it includes pretty graphs and crap |
| 15:28.05 | Toerkeium | probably.. :) |
| 15:29.03 | Toerkeium | but usefull in some scenarios |
| 15:29.06 | stephane_ | re |
| 15:29.11 | EinsteinTaylor | anyone played with stun a all? |
| 15:29.20 | Toerkeium | I have stunt |
| 15:29.29 | Toerkeium | works quite well |
| 15:29.48 | redondos | St1ckm4n: Thanks! |
| 15:30.17 | EinsteinTaylor | does the stun server run inside the network in question? or is it out on the net? |
| 15:30.32 | St1ckm4n | redondos: Thank You? (what did I do) |
| 15:30.33 | EinsteinTaylor | like if i deploy sip phones all over town can i have them connect to one central stun server? |
| 15:30.40 | EinsteinTaylor | or does each location need a stun server? |
| 15:30.50 | redondos | St1ckm4n: You showed me * Guru queue stats. |
| 15:31.04 | smackus | so i am interested to know if anyone else is using version 1.2.9.1 and not getting these avoided initial deadlock channels. if so I want to compare systems a bit |
| 15:31.14 | Toerkeium | not sure what your asking EinsteinTaylor |
| 15:31.30 | St1ckm4n | yeah it's not bad, but you have to work with it to dump your queue logs into a database, we're not doing realtime with it currently but we dump the logs every hour |
| 15:31.40 | EinsteinTaylor | maybe i need to read more so i know what i'm asking :) |
| 15:31.53 | redondos | EinsteinTaylor: You only need one. |
| 15:31.58 | redondos | EinsteinTaylor: http://www.voip-info.org/wiki-STUN |
| 15:32.19 | EinsteinTaylor | we're a full blown ISP/Telco |
| 15:32.22 | EinsteinTaylor | and i'm just wondering |
| 15:32.34 | Toerkeium | oh |
| 15:32.38 | EinsteinTaylor | if we deploy this all overtown can they all talk back to the stun server at my data center |
| 15:32.43 | knarfly | Manx: Sorry my friend I was off in the other room. I use asterick -v v v v c |
| 15:32.45 | Toerkeium | I was talking about voipstunt :P |
| 15:33.19 | EinsteinTaylor | i'm talking about stun to help with the NAT issues |
| 15:33.30 | knarfly | ManxPower: Sorry my friend I was off in the other room. I use asterick -v v v v c |
| 15:33.31 | *** join/#asterisk Pegasus_Epsilon (i=pegasus@198.174.233.133) |
| 15:33.44 | smackus | anyone using 1.2.9.1 and not getting avoided initial deadlock errors? |
| 15:34.01 | St1ckm4n | I get those errors but I'm still on 1.2.5 |
| 15:34.42 | smackus | is your asterisk crashing? |
| 15:34.42 | ManxPower | use safe_asterisk instead, then use asterisk -rvvv to reconnect |
| 15:34.42 | Pegasus_Epsilon | is splash ring for MWI possible on ZAP lines? |
| 15:35.00 | St1ckm4n | every couple of days |
| 15:35.25 | St1ckm4n | but I looked back in my logs and we were getting those errors ever since we first got the phone switch and it went for 2 weeks without an issue |
| 15:35.56 | St1ckm4n | when yours crashes, do all calls in process disconnect and you get all circuits are busy when you try to call in? |
| 15:36.09 | smackus | loose everything |
| 15:36.28 | smackus | It sounds to me that there is no one on 1.2.x that is running stable is this inaccurate? |
| 15:36.43 | Pegasus_Epsilon | i know it apparently does FSK by default, and i have the stutter dialtone working, but i kinda need splash rings |
| 15:36.53 | St1ckm4n | when ours crashes, I can connect to * CLI and run some commands but phone calls wont work unless I kill-9 the asterisk process and let it restart |
| 15:37.13 | [TK]D-Fender | smackus : mY SET SEEMS PRETTY STABLE. uP FOR WEEKS ON END |
| 15:37.35 | smackus | [TK]D-Fender: what version are you? |
| 15:39.20 | [TK]D-Fender | smackus : 1.2.9.1. I've run all sorts of releases all over the place throughout my client base as well with little or no trouble |
| 15:39.51 | smackus | yeah, check out my logs for just the last few minutes: |
| 15:39.52 | smackus | http://pastebin.ca/81504 |
| 15:40.06 | Pegasus_Epsilon | [TK]D-Fender: don't suppose you've got an answer to my question |
| 15:40.33 | [TK]D-Fender | Pegasus_Epsilon : Missed it. |
| 15:40.41 | [TK]D-Fender | smackus : HDLC errors! Yay! |
| 15:40.44 | Pegasus_Epsilon | splash rings on zap channels for MWI. possible? |
| 15:40.46 | SwK[Work] | anyone have a lead on a good SMS Shortcode wholesaler? |
| 15:41.02 | smackus | <PROTECTED> |
| 15:41.04 | [TK]D-Fender | smackus : Jul 7 09:36:00 WARNING[28474] chan_zap.c: Unable to request echo training on channel 51 |
| 15:41.11 | [TK]D-Fender | smackus : Get an A104d ;) |
| 15:41.17 | smackus | what is that? |
| 15:41.35 | [TK]D-Fender | Pegasus_Epsilon : * does it already in Zaptel. |
| 15:41.47 | [TK]D-Fender | smackus : Sangoma 4-Port HWEC card. |
| 15:41.48 | smackus | echo canceling card? |
| 15:41.52 | Pegasus_Epsilon | doesn't seem to be working for me, anything i might be missing about that? |
| 15:42.10 | [TK]D-Fender | Pegasus_Epsilon : what kind of interface are you running? |
| 15:42.17 | Pegasus_Epsilon | TDM2400P |
| 15:42.18 | smackus | [TK]D-Fender: are you saying that this is an issue with my card? |
| 15:42.35 | Pegasus_Epsilon | 1 FXS module |
| 15:42.48 | [TK]D-Fender | smackus : Well you are being flooded with Echotraining issues, and are getting HDLC aborts.. not a good sign. |
| 15:42.48 | Pegasus_Epsilon | (4 lines) |
| 15:43.09 | Pegasus_Epsilon | presume that's what you meant |
| 15:43.16 | smackus | so my card then...? |
| 15:43.22 | [TK]D-Fender | Pegasus_Epsilon : If you have "mailbox=[boxnumber]@[context]" in your zapata channel definition you will get the stutter. |
| 15:43.41 | Pegasus_Epsilon | i have the stutter dialtone, i need splash rings |
| 15:43.41 | [TK]D-Fender | smackus : Your card may very well be a flakey POS. |
| 15:44.10 | smackus | i would believe you based on results. |
| 15:44.13 | [TK]D-Fender | smackus : Echo training... *shudder* |
| 15:44.34 | [TK]D-Fender | <- master of the blatantly obvious |
| 15:45.51 | Pegasus_Epsilon | so uhhh |
| 15:45.59 | coppice | they train the echos to speak loud and clear |
| 15:46.35 | [TK]D-Fender | coppice : What pitch! What sustain! ARE YOU NOT ENTERTAINED?! |
| 15:47.07 | Pegasus_Epsilon | [TK]D-Fender: i have the stutter dialtone, i need splash rings |
| 15:48.53 | *** join/#asterisk inv_Arp (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 15:50.11 | *** join/#asterisk inv_Arp (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 15:51.57 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 15:52.25 | Pegasus_Epsilon | [TK]D-Fender: is that an "i don't know" or a "not possible" or what |
| 15:52.52 | dlynes_home | Pegasus_Epsilon: you might try giving him a chance to answer :) |
| 15:53.12 | Pegasus_Epsilon | didn't think he was writing a novella.... |
| 15:54.00 | dlynes_home | Pegasus_Epsilon: anyways...splash rings are usually a phone-based configuration setting |
| 15:54.15 | Pegasus_Epsilon | well i know sipuras generate them |
| 15:54.31 | dlynes_home | Pegasus_Epsilon: So try looking through your sipura manual then |
| 15:54.37 | dlynes_home | Pegasus_Epsilon: it has nothing to do with asterisk |
| 15:54.42 | Pegasus_Epsilon | i'm using a TDM with an FXS module |
| 15:54.49 | dlynes_home | Pegasus_Epsilon: asterisk only lets the sipura know that it has voicemail |
| 15:54.54 | Pegasus_Epsilon | ^^^ |
| 15:55.11 | dlynes_home | Pegasus_Epsilon well i know sipuras generate them |
| 15:55.16 | pdtmobile | Pegasus_Epsilon: are you paying him to help you? |
| 15:55.16 | Pegasus_Epsilon | exactly |
| 15:55.56 | dlynes_home | Pegasus_Epsilon: so are you using a tdm400, or both? |
| 15:56.14 | Pegasus_Epsilon | TDM2400P+1xFXS(4 lines) |
| 15:56.34 | Pegasus_Epsilon | but i guess i'm trying to get it to mimic a sipura |
| 15:56.48 | dlynes_home | Pegasus_Epsilon: take a look at your sample zapata.conf file then |
| 15:57.01 | dlynes_home | Pegasus_Epsilon: if it supports ring splashing at all, there'll be a setting for it in there |
| 15:57.22 | Pegasus_Epsilon | i didn't see it in there, but i'll look again. |
| 15:57.29 | dlynes_home | Pegasus_Epsilon: you do have it configured for voicemail, right? |
| 15:58.03 | dlynes_home | Pegasus_Epsilon: and if it doesn't have such a feature, there's nothing stopping you from writing such a feature |
| 15:58.09 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 16:00.05 | Pegasus_Epsilon | i do have the stutter ring for voicemail waiting working fine |
| 16:01.35 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 16:02.10 | dlynes_home | you mean stutter tone, not stutter ring |
| 16:03.19 | [TK]D-Fender | Pegasus_Epsilon : oH SPLASH rings... HMMM.. NEVER HEARD OF ON ANALOG BEFORE... |
| 16:03.35 | [TK]D-Fender | Pegasus_Epsilon : And keep the heck away from my Caps-Lock key! ;) |
| 16:04.17 | [TK]D-Fender | Pegasus_Epsilon : That is a worthy addition to chan_zap ..... |
| 16:06.34 | *** join/#asterisk nettie (i=[U2FsdGV@85-18-54-38.ip.fastwebnet.it) |
| 16:10.04 | *** join/#asterisk Bacta (n=silvermo@222.155.135.57) |
| 16:10.07 | Bacta | can i was my wanky in ur sinky? |
| 16:13.31 | *** join/#asterisk Toerkeium (n=oo@201.216.206.221) |
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| 16:27.20 | Toerkeium | is it possible to make * work against a mysql database instead of configuation text files? |
| 16:28.16 | PakiPenguin | yup |
| 16:28.25 | PakiPenguin | asteirsk realtime |
| 16:29.33 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 16:31.30 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 16:31.56 | Toerkeium | Thanks PakiPenguin |
| 16:32.19 | PakiPenguin | yw Toerkeium |
| 16:32.27 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:36.13 | vader-- | hola |
| 16:36.15 | vader-- | hey qwell |
| 16:37.37 | *** join/#asterisk marv0997 (n=marv@65.165.87.198) |
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| 16:43.45 | vader-- | hmmm |
| 16:43.52 | vader-- | im still pondering going to sccp over sip |
| 16:44.04 | Qwell[] | as you should be |
| 16:44.10 | vader-- | im trying to find something that shows me a feature set of sip vs sccp on the 7940gs |
| 16:44.22 | file | SCCP is for silly people |
| 16:44.26 | Qwell[] | sip: lame stuff |
| 16:44.29 | Qwell[] | sccp: everything |
| 16:44.43 | vader-- | file why do you say sccp is for silly people? |
| 16:44.58 | Qwell[] | vader--: the maintainer of chan_skinny is a total nub |
| 16:45.24 | file | that being said chan_skinny does need more work |
| 16:45.43 | Qwell[] | file: 2 days, and it'll get all the love in the world. ;) |
| 16:45.47 | vader-- | ? |
| 16:45.50 | vader-- | 2 days |
| 16:45.56 | Qwell[] | nothing |
| 16:46.07 | vader-- | they releasing something new? |
| 16:46.29 | Qwell[] | vader--: releasing...no |
| 16:47.25 | vader-- | so whats happening in two days? |
| 16:47.43 | coppice | i'm going to europe :-) |
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| 16:48.16 | PakiPenguin | :p |
| 16:49.20 | vader-- | qwell is something happening with sccp in two days? |
| 16:49.30 | Qwell[] | vader--: it's gonna get some lovin' |
| 16:50.03 | vader-- | oh ya? |
| 16:50.07 | vader-- | by who] |
| 16:50.13 | Qwell[] | by its maintainer |
| 16:50.14 | coppice | what a coincidence. i'm going to europe to get some lovin. wifey is there |
| 16:50.21 | Qwell[] | namely...me... |
| 16:50.25 | vader-- | gotcha |
| 16:50.51 | vader-- | who maintains sccp2? |
| 16:50.55 | Strom_C | silly client control protocol? |
| 16:51.09 | Qwell[] | vader--: some dumbass named Sergio |
| 16:51.16 | Qwell[] | Strom_C: ha, I like it |
| 16:51.17 | coppice | Some Completely Cockedup Protocol |
| 16:51.26 | Qwell[] | file: What? :( |
| 16:51.30 | Qwell[] | He is a dumbass ;/ |
| 16:51.56 | Qwell[] | alright, fine...I appologize...he wasn't always that way |
| 16:52.09 | Qwell[] | -p |
| 16:55.03 | Qwell[] | so, moving on :p |
| 16:55.10 | file | moving swiftly on |
| 16:55.29 | Qwell[] | /clr anybody? |
| 16:55.44 | vader-- | qwell do you use a large number of cisco phones? |
| 16:55.58 | Qwell[] | vader--: like...a few |
| 16:56.02 | mut | phisco cones?! |
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| 17:06.36 | jbalcomb | is there a canned audio prompt for "you are next" or "there are"? |
| 17:08.07 | KranZ | i think there's a "you are number" |
| 17:08.19 | KranZ | used in the queues |
| 17:08.36 | KranZ | bbl |
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| 17:13.53 | *** join/#asterisk jamincollins (n=jcollins@ptech7-44.acdmis.com) |
| 17:15.31 | jamincollins | I've read through the information on voip-info.org and seem to be missing something necessary to override the CallerID set when a SIP phone dials out... |
| 17:15.52 | file | jamincollins: either set it in sip.conf, or in the dialplan |
| 17:16.46 | jamincollins | I'm hoping to be able to set it on a per station basis... and have been trying to set it in sip.conf, but every change I make seems to be ignored (yes I am issuing a "sip reload") |
| 17:17.03 | file | pastebin an entry |
| 17:17.25 | jbalcomb | ok, so i guess they were just copies of queue-youaremext and queue-thereare |
| 17:17.34 | jamincollins | I've tried specifing "callerid=1234567" and "fromuser=1234567" |
| 17:18.04 | jamincollins | however I'm still getting what the phone sends which would be 4567 in these cases |
| 17:18.20 | file | so a station calls into Asterisk in order to place a call and you want to control the callerid, okay - you need to show me an example entry so I can tell you what's potentially wrong |
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| 17:21.20 | *** part/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:22.19 | syzygybsd | jamincollins: pastebin an entry from your sip.conf |
| 17:23.16 | jamincollins | attempting to... just waiting on pastebin at this point |
| 17:24.03 | jamincollins | it's not rocket science I know... but it's not accepting the submission... =( |
| 17:24.08 | CunningPike | jamincollins: Use pastebin.ca - pastebin.com is fscked |
| 17:24.20 | jamincollins | ahhh |
| 17:24.49 | Toerkeium | ARA docs are very poor |
| 17:24.59 | jamincollins | http://pastebin.ca/81573 |
| 17:25.05 | Toerkeium | any place to find a more extensive docs? |
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| 17:25.51 | file | jamincollins: you have no password in that... but anyway, fromuser isn't what you want... callerid=My Polycom <1000> |
| 17:25.57 | file | that will set name to My Polycom and number to 1000 |
| 17:26.06 | jamincollins | tried that too |
| 17:26.09 | file | provided the phone is matching that entry |
| 17:26.15 | syzygybsd | jamincollins: in there add the line callerid=<number> |
| 17:26.27 | jamincollins | sip debug peer polycom-jwc <- catches the dial request |
| 17:26.46 | file | pastebin that |
| 17:26.55 | jamincollins | but callerid=1234 has no effect on the pushed ANI |
| 17:27.07 | syzygybsd | surrond it in <> |
| 17:27.44 | jamincollins | no change |
| 17:27.52 | jamincollins | still getting the lineid of the phone instead |
| 17:28.02 | Strom_C | "lineid"? |
| 17:28.20 | file | do a sip debug so we can make sure it's catching the right entry, and also a sip show user polycom-jwc |
| 17:28.25 | file | to make sure your callerid is being read in |
| 17:29.30 | *** join/#asterisk Rick_Hunter (n=rhunter@01-118.008.popsite.net) |
| 17:29.30 | jamincollins | No such command 'sip show user' (type 'help' for help) |
| 17:29.46 | file | what version of Asterisk? |
| 17:29.51 | Strom_C | 0.4 |
| 17:29.52 | Qwell[] | 1.2.27 |
| 17:30.01 | jamincollins | 1.0.7 (Debian Stable) |
| 17:30.25 | Strom_C | holy ass, thats ooooooooooooooold |
| 17:32.40 | syzygybsd | how can I tell if asterisk is running chrooted and if so what the root is (it is failing on my System(/bin/echo 1)) |
| 17:33.09 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
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| 17:33.44 | jamincollins | Strom_C: thanks for the insightful observation |
| 17:33.52 | jsaunders | What is the term that describes having a pbx for your company that resides at your voip providers location? |
| 17:34.03 | jsaunders | I can't remember it for the life of me. |
| 17:34.04 | Strom_C | jsaunders: hosted pbx |
| 17:34.06 | jamincollins | co-located? |
| 17:34.09 | jsaunders | Nope.... |
| 17:34.11 | Strom_C | colocated or hosted |
| 17:34.21 | jsaunders | Yes, those are valid, but there's a voip term related to it. |
| 17:34.28 | jsaunders | Ugh |
| 17:34.36 | Strom_C | dead hookers |
| 17:34.40 | jsaunders | That too but..... |
| 17:34.44 | Strom_C | :) |
| 17:34.46 | Corydon-w | cat sex |
| 17:34.55 | jsaunders | Ok, now we're getting off topic here. |
| 17:34.55 | jsaunders | Heh |
| 17:35.01 | jsaunders | Man, what is it. |
| 17:35.12 | file | dogballs |
| 17:35.12 | Corydon-w | Lame |
| 17:35.31 | jamincollins | so, was it simply not possible in v1.0.7 to control the SIP CallerID? |
| 17:35.43 | Strom_C | tandem? gateway? |
| 17:35.44 | jsaunders | You know when you hear a song on the radio but have no clue what it is... and then they don't annouce it at the end and go straight to commercial instead... dunno about you but I'm cursed w/ having to find out... that sorta thing. |
| 17:35.44 | file | jamincollins: all 1.0 knowledge has been wiped from my brain |
| 17:35.54 | jsaunders | tandom is voip to voip I believe. |
| 17:35.58 | jsaunders | tandem |
| 17:36.02 | jamincollins | I've even tried user the "setvar" command |
| 17:36.14 | file | 1.0 doesn't have that |
| 17:36.14 | Strom_C | jsaunders: depends. |
| 17:36.18 | Corydon-w | jamincollins: SetCallerID ? |
| 17:36.24 | jamincollins | but regardless of setting CALLERNUM or CALLERID nothing changes |
| 17:36.39 | syzygybsd | Virtual PBX |
| 17:36.44 | Strom_C | jsaunders: google for a phrase in the song |
| 17:36.52 | Corydon-w | jamincollins: you can't set builtin variables. That's why we hadSetCallerID |
| 17:37.17 | jsaunders | syzygybsd: Nope. |
| 17:37.49 | syzygybsd | jamincollins: I still haven't seen the pastbin of the sip debug peer polycom-jwc they asked for... |
| 17:37.55 | Corydon-w | jamincollins: besides, you should be using 1.0.11.1, due to the denial-of-service attack possible in 1.0.7 |
| 17:38.39 | jamincollins | pretty sure Debian back ported the security fix to their stable version |
| 17:38.41 | syzygybsd | Centrex? |
| 17:39.06 | Strom_C | jamincollins: hahahahahahaha, i doubt that |
| 17:39.13 | jamincollins | syzygybsd, I'm not going to go through and santize out the IP information for the whole session |
| 17:39.34 | syzygybsd | your on linux right? it is one easy sed command but anyway |
| 17:40.11 | syzygybsd | jsaunders: it is centrex isn't it |
| 17:40.15 | jamincollins | syzygybsd, yes, one for the IP, one for the phone numbers, etc |
| 17:40.16 | syzygybsd | http://en.wikipedia.org/wiki/Centrex |
| 17:40.16 | Strom_C | no |
| 17:40.22 | jsaunders | ip centrex. Yes. |
| 17:40.32 | jsaunders | Bing bing bing |
| 17:40.40 | Corydon-w | jamincollins: well, we could ask the Debian maintainer about that. He hangs out in here, and I doubt he has done anything about it |
| 17:40.51 | Strom_C | that wins the "biggest abuse of the word Centrex" award of the year |
| 17:40.54 | jsaunders | http://www.ip-centrex.org/whatis/index.html |
| 17:40.55 | syzygybsd | http://www.google.com/search?num=50&hs=77Y&hl=en&lr=&safe=off&client=firefox-a&rls=org.mozilla%3Aen-US%3Aofficial&q=*+is+a+hosted+pbx&btnG=Search around result 40 I think, I have 50 show up on a page |
| 17:41.11 | Corydon-w | jamincollins: the standard answer is to upgrade to the latest release version |
| 17:41.28 | jsaunders | heheh |
| 17:41.37 | file | Strom_C: Digital Centrex Supreme ftw |
| 17:41.42 | syzygybsd | or 5 on the advertizements |
| 17:41.46 | Strom_C | Centrex is PBX-like services served off of the telephone company's central office switch |
| 17:42.16 | Strom_C | how the hell you transmogrify that into "PBX we own colocated at the ISP" is beyond me |
| 17:42.16 | jamincollins | Corydon-w, you talking about the buffer overflow? |
| 17:42.54 | jamincollins | There was a security update made on 4/26 to correct that by the Debian security team |
| 17:42.59 | jsaunders | Read this strom_c, that will explain to you how made the transformation. http://www.ip-centrex.org/whatis/index.html |
| 17:43.12 | *** join/#asterisk Mattwj2005 (n=Matt@user-12l3n0n.cable.mindspring.com) |
| 17:43.18 | jsaunders | And dictionary.com has centrex listed as... (CENTRal EXchange) a kind of telephone exchange. |
| 17:43.32 | Corydon-w | jamincollins: what about the one at the beginning of June? |
| 17:43.37 | jsaunders | So, if you slap 'ip' in front, it comes accross as fairly straight forward to me. |
| 17:43.46 | Strom_C | jsaunders: what a bunch of morons |
| 17:43.47 | syzygybsd | so any idea how to tell if asterisk is chrooted? |
| 17:43.48 | jsaunders | Heheh |
| 17:44.12 | jsaunders | So you suggest I don't get in the habbit of using this term. |
| 17:44.13 | jsaunders | Heh |
| 17:44.21 | Strom_C | exactly |
| 17:44.24 | jsaunders | :D |
| 17:44.25 | jsaunders | word |
| 17:44.30 | Corydon-w | syzygybsd: can you access /usr/sbin/traceroute ? If not, you're chrooted |
| 17:44.45 | syzygybsd | heh.. can't access /bin/echo |
| 17:44.56 | *** join/#asterisk alexis101 (n=alexis@70.54.204.92) |
| 17:45.21 | jamincollins | Corydon-w, don't see an update since June for it, no |
| 17:45.23 | syzygybsd | well, now that that issue is cleared.. how to solve this problem... |
| 17:46.39 | *** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41) |
| 17:46.46 | alexis101 | hi everyone I have a problem with my asterisk server at my job ... Everything was ok since 2 days ago when my server reboot with no reason (dint see anything in system log) now sometime my asterisk do strange thing |
| 17:46.52 | GerbilWrk | Anyone else just have Teliax go down on them? |
| 17:47.46 | alexis101 | like it can run any application (queue voicemail ...) and when i restart it nothing work ... i have to reboot the computer |
| 17:48.26 | alexis101 | and i dint change anything in my configuration files |
| 17:48.34 | alexis101 | anyone ever had this problem ??? |
| 17:48.36 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.187.115) |
| 17:49.02 | *** join/#asterisk Bullseye_Network (n=info@216.143.192.69) |
| 17:52.05 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001217014b92.vc.shawcable.net) |
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| 17:52.45 | jamincollins | I can override the CallerID if I set the fromuser on the outgoing SIP trunk but then I have a singular Caller ID for all stations rather than pushing each stations DID |
| 17:55.37 | dlynes_laptop | ~q931 |
| 17:55.51 | dlynes_laptop | ~wiki q.931 |
| 17:56.31 | *** join/#asterisk andrebarbosa (n=andrebar@62.48.215.102) |
| 17:56.38 | dlynes_laptop | Anyone happen to know where the list of q.931 error codes is located? |
| 17:56.54 | jamincollins | search for ISDN CAUSE CODE |
| 17:56.59 | coppice | try the source code for libpri |
| 17:57.23 | andrebarbosa | anyone as a figure for how many invites can asterisk handle per second, on a high-end machine like dual-xeon or something like that? |
| 17:58.18 | dlynes_laptop | thanks |
| 17:58.24 | *** join/#asterisk ariel_ (n=FreePBX6@70.46.87.158) |
| 17:59.03 | syzygybsd | anyone have any idea why this line would fail with the error "Spawn extension (test, 18317671212, 1) exited non-zero on 'SIP/QCC1-82e8'"? Line in extensions.conf "exten => _1.,1,System(echo 1)" |
| 17:59.06 | *** join/#asterisk kiong (n=alamek@bb58-185-164-145.singnet.com.sg) |
| 17:59.25 | syzygybsd | it works if I go straight to dial, and from the console "!echo 1" works |
| 17:59.26 | dlynes_laptop | syzygybsd: that means there wasn't any errors |
| 17:59.40 | dlynes_laptop | syzygybsd: if there was an error, it would have exitted zero |
| 18:00.05 | PakiPenguin | hi dlynes_laptop |
| 18:01.06 | syzygybsd | other way around dlynes_laptop, if there were errors it would exit 1 or -1... 0 if it succeeded. The command isn't run as I have tried piping it to a file, etc and the next command (a dial, never occurs) |
| 18:01.51 | dlynes_laptop | syzygybsd: if you're looking for the specific reason why it disconnected, check ${HANGUPCAUSE} |
| 18:02.07 | dlynes_laptop | heya pp |
| 18:02.29 | syzygybsd | it never gets to that line in the extensions.conf. But I will Noop that and hope it gives me some clue |
| 18:02.43 | *** join/#asterisk Krush (n=easys@p50922BCC.dip0.t-ipconnect.de) |
| 18:02.47 | Krush | good evening |
| 18:04.39 | syzygybsd | hmm, not defined it is 0, Also dlynes_laptop it never gets to the PRI, it fails at the system command so that variable would never get set anyway |
| 18:05.15 | Krush | i am trying to sed group permissions on my voicemails, as they need to be accessed via web frontent. i have modified apps_voicemail to VOICEMAIL_DIR_MODE and VOICEMAIL_FILE_MODE |
| 18:05.35 | Krush | DIR_MODE works correctly but FILE permissions are simply ignored |
| 18:07.27 | *** join/#asterisk s0lid (n=s0lid@124.107.26.34) |
| 18:07.34 | dlynes_laptop | syzygybsd: ummm...you're dialing out on sip |
| 18:07.42 | dlynes_laptop | syzygybsd: what does your pri have to do with sip? |
| 18:08.34 | dlynes_laptop | Krush: the permissions issue is a known issue |
| 18:08.51 | jbalcomb | [TK]D-Fender ok, everything is back to normal, sort of. On the bright side, we are now on Asterisk 1.2.5 instead of 1.2.1 |
| 18:08.51 | dlynes_laptop | Krush: I think I remember seeing somewhere that a lot of that will be fixed in 1.4 |
| 18:08.52 | syzygybsd | lol.. just how the call is comming into the system, sorry, totally missed that, that is how I am generating the call (ie, a sip client). I am dialing out on PRI but that has nothing to do with this error as it is failing on the System command. |
| 18:09.05 | syzygybsd | It shouldn't have any affect where the call is comming from should it? |
| 18:09.19 | dlynes_laptop | Krush: regardless, there's a patch available on bugs.digium.com that will fix that problem too |
| 18:09.31 | dlynes_laptop | syzygybsd: nope |
| 18:09.59 | dlynes_laptop | syzygybsd: oops...didn't see it's failing on echo 1 |
| 18:10.17 | syzygybsd | is there another way to call a system command? maybe use the AGi but that seems overkill for echo |
| 18:10.33 | ToyMan | ok, hopefully without starting any flamewars.. .what is the prefered/easiest/most supported OS to get asterisk up and running on at the moment? |
| 18:10.39 | dlynes_laptop | syzygybsd: maybe you have to enclose echo 1 in quotes? i.e. "echo 1"? |
| 18:10.56 | dlynes_laptop | ToyMan: linux |
| 18:11.06 | dlynes_laptop | ToyMan: it's the only officially supported OS |
| 18:11.16 | ToyMan | any specific distro? |
| 18:11.31 | dlynes_laptop | ToyMan: whichever one you're most familiar with is the best one to run it on |
| 18:12.05 | dlynes_laptop | ToyMan: there's peeps running it on slackware, gentoo, debian, redhat, centos, fedora, ... |
| 18:12.13 | ToyMan | cool... i like the debian-based distros myself.. are there .debs for everything? |
| 18:12.24 | Krush | dlynes_home, damn. i cant find that patch |
| 18:12.26 | jamincollins | anyway to see which (if any) directives in a conf file are being ignored? |
| 18:12.30 | dlynes_laptop | ToyMan: there's even the odd brave soul running it on FreeBSD and solaris |
| 18:12.39 | Krush | could you please tell me where you have seen it? |
| 18:12.49 | dlynes_laptop | Krush: i can't recall offhand...gimme a sec |
| 18:12.53 | jamincollins | ToyMan, there are debs for it, but they are a tad old |
| 18:13.09 | ToyMan | FreeBSD is my OS of choice, but not always worth the struggle for specific cases like this |
| 18:13.13 | syzygybsd | wish it were that simple :( tried it, tried having the 1 in quotes, giving the absolute path to echo, making sure it works in the console. huh, just tried the actuall command "/bin/sh -c echo 1" and it didn't return what it should, at least that is somewhere to go.... |
| 18:13.25 | ToyMan | jamincollins: so compiling is the way to go? |
| 18:13.31 | dlynes_laptop | ToyMan: FreeBSD is fine, providing you don't need to do TDM hardware |
| 18:13.46 | dlynes_laptop | ToyMan: FreeBSD's zaptel driver support is not up to spec |
| 18:14.05 | dlynes_laptop | ToyMan: a telephony board |
| 18:14.11 | ToyMan | ah... |
| 18:14.13 | coppice | its short for tedium |
| 18:14.15 | ToyMan | heh |
| 18:14.21 | dlynes_laptop | ToyMan: if you're running pure ATA's and gateways, FreeBSD won't be an issue |
| 18:14.24 | *** join/#asterisk pdtmobile (n=ptinsley@209.12.249.243) |
| 18:14.24 | mog | lol CoaxD |
| 18:14.27 | mog | er coppice |
| 18:14.44 | ToyMan | hmmm. |
| 18:14.57 | ToyMan | ok, thanks |
| 18:15.19 | ToyMan | I might just try a few test installs on dif. OS's and see what works... |
| 18:16.02 | dlynes_laptop | Krush: http://bugs.digium.com/view.php?id=6334 |
| 18:16.23 | dlynes_laptop | Krush: i just did a filter search of 'umask' |
| 18:16.28 | dlynes_laptop | Krush: it was the only hit |
| 18:16.34 | Krush | ok. thanks |
| 18:17.02 | dlynes_laptop | It's listed as severity, 'tweak' |
| 18:17.09 | dlynes_laptop | but I think it's much more than that |
| 18:17.25 | dlynes_laptop | I'd think it should be considered severe, as it's a critical security flaw |
| 18:17.51 | Krush | this fix sets makes use of general umask settings which is not what i need. |
| 18:18.37 | dlynes_laptop | ToyMan: if you decide to try redhat/centos/fedora, you might want to check out the following known issue, too: |
| 18:18.42 | dlynes_laptop | ~redhatbug |
| 18:18.43 | jbot | rumour has it, redhatbug is is a problem with the latest RedHat Enterprise Linux and CentOS kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 18:19.29 | ToyMan | dlynes_laptop: thanks, but I'll stick with a debian base or bsd.. |
| 18:20.07 | *** join/#asterisk JonR800 (i=jon@p1mp.org) |
| 18:20.21 | dlynes_laptop | Krush: if you look further down the comments, you'll see references to the voicemail patches |
| 18:20.55 | dlynes_laptop | Krush: the original poster didn't change voicemail, but other people did |
| 18:21.21 | dlynes_laptop | anyways...gotta run... |
| 18:25.59 | Bullseye_Network | anybody think of a reason asterisk would ignore a callout file that was dropped in via a nfs partition? The permissions are the same as ones dropped in locally. |
| 18:26.00 | Krush | hmm. how could i access this patch? |
| 18:26.56 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 18:29.07 | syzygybsd | is it possible that my version of asterisk, 1.2.7, doesn't have the system or trysystem or AGi commands? |
| 18:29.08 | jamincollins | ToyMan: I haven't had problems with the v1.0.7 build from debian stable until this recent SIP Caller ID scenario |
| 18:29.39 | jamincollins | that said, there may be some security concerns not yet backported to stable |
| 18:29.40 | ToyMan | SIP scenario? a recent bug? |
| 18:30.06 | jamincollins | ToyMan, not a bug so much as an inability to override the outgoing Caller ID |
| 18:30.28 | jamincollins | could just be I don't have the right syntax... or could be that it just isn't possible in this version |
| 18:30.42 | ToyMan | ah, ok. I'm still at the basics level |
| 18:30.49 | ToyMan | up and running first.. ;-) |
| 18:30.57 | *** join/#asterisk Waverly360 (n=mirc@209.12.249.243) |
| 18:31.24 | ToyMan | someone mentioned to me that they found a flash-based bsd OS device that was running asterisk.. anyone heard of this? |
| 18:31.54 | ToyMan | something along the lines of a sokeris box... |
| 18:32.31 | jamincollins | there are also backports of the 1.2.7.1 build if you so desire |
| 18:32.47 | Waverly360 | Anyone here played around much with polycom phones and the "Do Not Disturb" feature? |
| 18:33.16 | jamincollins | AFAIK, the DND feature simply causes the phone to reject the invite |
| 18:33.16 | Waverly360 | Or even the different statuses you can set on the phones... Particularly the 501s. |
| 18:34.03 | Waverly360 | There's no way to make them notify asterisk so that the proper presence can be set? |
| 18:35.47 | jamincollins | that I don't know |
| 18:36.00 | syzygybsd | AHHHHHHHHHHHHHH, k, somehow when this version of asterisk was compiled the System and AGi applications wern't included |
| 18:36.22 | Waverly360 | Hrm. I have a 601 with a sidecar attached. I have the sidecar monitoring my 501 extension just fine. |
| 18:36.38 | Waverly360 | So whenever I dial something with the 501, the light lights up on the sidecar. |
| 18:37.04 | Waverly360 | But I know the sidecar can show a lot more than just if a line is in use...I'm wanting to see if it's possible to make it recognize the statuses that can be set on the 501 |
| 18:37.06 | syzygybsd | I didn't know that was an option |
| 18:40.05 | syzygybsd | I should never trust my clients builds me thinks |
| 18:43.27 | syzygybsd | wow, the source on this computer for asterisk won't compile the system or agi commands, maybe I should grab fresh code... |
| 18:46.34 | jamincollins | it was a version problem.... upgraded to the 1.2.7.1 backport and it's working fine now |
| 18:54.57 | *** join/#asterisk s0lid (n=s0lid@203.177.12.98) |
| 18:58.09 | syzygybsd | ok, before I go beat this server into the ground.. what libraries does asterisk need to compile system and AGI functionality? |
| 18:59.37 | syzygybsd | hmm.. curl maybe... |
| 19:03.54 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 19:04.54 | ToyMan | ok, more flame-war material ... ;-) |
| 19:05.00 | SwK[Work] | anyone know of a good place to get a sack of Argentine DIDs? |
| 19:05.14 | ToyMan | do you guys have a favorite VoiP provider? |
| 19:05.30 | SwK[Work] | yeah me |
| 19:05.37 | SwK[Work] | I am my favorite voip provider |
| 19:05.50 | file | SwK[Work]: no, I'm my favorite voip provider |
| 19:06.20 | GerbilWrk | I highly recommend against Teliax, i'll just say that |
| 19:06.50 | *** join/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
| 19:06.52 | pdtmobile | has anybody tried to port a number from one? |
| 19:06.55 | pdtmobile | to another |
| 19:06.58 | trixter | The first 100 paid ClueCon attendees will recieve a FREE Asterisk Compatible T1 Card (value $450), an Asterlink account with 1000 outbound minutes, and a FREE inbound US DID. Register now to secure yours today! |
| 19:07.10 | trixter | wow cluecon basically pays for itself just by registering |
| 19:07.35 | file | that was not advertising at all |
| 19:07.52 | SwK[Work] | file: I'm a better voip supplier then you |
| 19:07.54 | trixter | nope it wasnt |
| 19:08.14 | file | SwK[Work]: I supply like the best, after you pass my 100 page interoperability test! |
| 19:08.17 | trixter | mostly becuase I am not affiliated with cluecon |
| 19:08.25 | SwK[Work] | file: who are you? Level3? |
| 19:08.34 | file | SwK[Work]: Level3.141592653 |
| 19:08.37 | SwK[Work] | hah |
| 19:08.39 | *** join/#asterisk Drew99 (n=top@ppp83-237-244-47.pppoe.mtu-net.ru) |
| 19:09.20 | trixter | now if I actually worked for cluecon.com then it might be advertising, but since I dont work for cluecon.com its hard to say that I am advertising.. I get nothing out of talking about it, hwoever I think that since its a good open source telephony conference covering most of the major open source telephony projects I like to share with others |
| 19:09.59 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 19:10.00 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:10.00 | TrixV0x | ToyMan: I also highly recommend *against* Teliax... Go with VoicePulse Connect, they've been great to me so far |
| 19:10.46 | SwK[Work] | ToyMan: MixNetworks, CXMUSA.com, Asterlink are all 3 good |
| 19:10.50 | Drew99 | hi! anybody can help me to split sip-operator into 2 contexts - incoming and outgoing? i need to route incoming calls into context=sipin, and outgoing calls - using context=sipout |
| 19:11.08 | SwK[Work] | VoicePulseConnect keep raising their prices and if you decide you dont like them they REFUSE to let your port out |
| 19:11.13 | TrixV0x | Hmm, never even heard of those... |
| 19:11.18 | SwK[Work] | Teliax is the suck |
| 19:11.41 | ToyMan | thanks... and I'll stay away from Teliax... |
| 19:11.58 | SwK[Work] | last time I had to work with a customer using them, they could send outbound calls but no inbound calls would work at all sip or iax |
| 19:12.52 | TrixV0x | I don't know, VoicePulse just dropped their prices so most of my calls are < 0.007... Can't complain too much about that, everyone else seems to charge for incoming per-minute also |
| 19:13.12 | trixter | I pay people for inbound minutes when they get one of my DIDs |
| 19:13.20 | SwK[Work] | Mix and CXM have flat rate plans for US/.CA calls |
| 19:13.26 | syzygybsd | k, i can't figure out why the system application won't compile |
| 19:13.28 | trixter | but then trxtel.com is "the VoIP provider that pays you" |
| 19:13.29 | znoG | if I want to set a hard limit on how long a call can go active, should I use AbsoluteTimeout? I find the L option in dial USUALLY hangs up the line after X ms, but a couple of times it hasn't. |
| 19:13.39 | trixter | we never charge anything and only pay our customers |
| 19:13.42 | *** part/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 19:13.47 | TrixV0x | Yeah, that's a good business model :) |
| 19:13.59 | SwK[Work] | sounds like a 419 telephonecompany |
| 19:14.02 | TrixV0x | And you're going to pay salary how? |
| 19:14.15 | JonR800 | second job |
| 19:14.15 | trixter | trixter: quite easily infact |
| 19:14.15 | SwK[Work] | hah |
| 19:14.35 | trixter | er trixv0x ... if people wouldnt steal variations of my nick tab would work :P |
| 19:14.42 | znoG | ......?¿? |
| 19:14.55 | TrixV0x | Ok, so, how? |
| 19:15.19 | trixter | as a clec we get the ability to make a little money per minute depending on the direction the call goes. this is how ipkall gives away numbers.. the only difference is we share some of that revenue with our customers |
| 19:15.37 | *** join/#asterisk firejon (n=jellis@206-169-49-105.static.twtelecom.net) |
| 19:15.44 | trixter | but the reality is does it matter so long as you do infact get paid as the webpage says? :) |
| 19:16.15 | TrixV0x | Yeah, because you'll shut down before the first check ever comes :) |
| 19:16.49 | trixter | and if you want to test it skype is free til the end of the year so you can make calls without it costing you anything ... although we dont officially support the concept of calling yourself for profit, it is legal according to http://hraunfoss.fcc.gov/edocs_public/attachmatch/FCC-04-32A1.pdf |
| 19:17.11 | Drew99 | hi! anybody can help me to split sip-operator into 2 contexts - incoming and outgoing? i need to route incoming calls into context=sipin, and outgoing calls - using context=sipout |
| 19:17.23 | trixter | TrixV0x: well we have been operating for over a year.. and as a clec we do get paid per minute to receive calls, so ... |
| 19:18.08 | SwK[Work] | Drew: any context can Dial(SIP/WhateverPeer) reguardless of what context that peer is in |
| 19:18.23 | SwK[Work] | the context on a sip peer definition is just to define what context incoming calls go into |
| 19:19.39 | Drew99 | SwK[Work]: ii define a context for incoming calls using context=sipin, but they go to context=default =(( |
| 19:19.47 | *** join/#asterisk trbldwine (i=trbldwin@adam.ur.northwestern.edu) |
| 19:19.51 | syzygybsd | can anyone help me figure out why system and agi arn't compiled? my make results http://pastebin.ca/81681 |
| 19:20.04 | SwK[Work] | drew99 did you restart asterisk or reload sip after you defined that? |
| 19:20.44 | Drew99 | SwK[Work]: certainly. I need to define type=user, context=sipin and type=peer, context=sipout, right? |
| 19:22.16 | jarrod | anyway to check why i have 5 asterisk instances running each using 169M memory |
| 19:22.29 | jarrod | well, its one instance, but ps reports 5 separates |
| 19:22.52 | SwK[Work] | drew99: why create both just create a type=friend and context=sipin.... |
| 19:22.58 | SwK[Work] | context=sipout doesnt do shit |
| 19:23.24 | *** join/#asterisk userdefined (n=jross@cpe-24-169-142-23.rochester.res.rr.com) |
| 19:23.41 | *** join/#asterisk linlin (n=linlin@c-67-184-159-30.hsd1.il.comcast.net) |
| 19:25.27 | Drew99 | SwK[Work]: but in sip.conf: Tip 2: Use separate type=peer and type=user sections for SIP providers (instead of type=friend) if you have calls in both directions |
| 19:27.44 | jarrod | how can i see why asterisk is using so much memory? |
| 19:28.19 | trixter | gdb |
| 19:28.50 | Drew99 | SwK[Work]: when someone calls me (context=sipin), * trasmit this call to mymobile using context=sipout throught the same sip-operator. I need to transfer, etc. this call from my mobile using context=sipout, but user that calls me must have it's own opotunities using context=sipin |
| 19:30.47 | Drew99 | SwK[Work]: I know that I can give 't' into Dial-params and I can transfer, caller don't, but if i need other dtmf-options, unique for me and for caller. |
| 19:31.12 | Drew99 | SwK[Work]: I think only splitting contexts can help, right? |
| 19:31.33 | *** join/#asterisk backblue (n=moo@87-196-42-182.net.novis.pt) |
| 19:36.45 | Drew99 | nobody know about splitting context of sip-operator? |
| 19:41.54 | *** join/#asterisk paryl (n=chatzill@216-201-177-82.res.logixcom.net) |
| 19:50.46 | smackus | apparently digium does not like hyperthreading. |
| 19:51.01 | trixter | they dont appear to like normal threading either :P |
| 19:51.41 | CunningPike | smackus: Apparently it kacks up how the card get interrupts |
| 19:52.36 | smackus | yeah... i never had deadlock errors until I went to a threaded system. |
| 19:52.41 | smackus | never would have thought.... |
| 19:52.47 | smackus | thought it would do much better. |
| 19:52.55 | trixter | you can do threads without deadlocks if you do em right |
| 19:53.02 | paryl | when you do 'show queues', what do these mean?: W:0, C:67, A:9 |
| 19:53.24 | smackus | isnt it wait, calls and something else? |
| 19:53.38 | smackus | average maybe? |
| 19:58.15 | paryl | hmmm |
| 19:58.50 | *** part/#asterisk Drew99 (n=top@ppp83-237-244-47.pppoe.mtu-net.ru) |
| 19:59.59 | *** join/#asterisk NewSole (n=dave@d226-105-226.home.cgocable.net) |
| 20:02.55 | dpryo | Somebody know if it's possible to remote control Avaya 4620 phones? (sip) Like the Avaya IPAgent does.. |
| 20:03.52 | coppice | a remote control phone? I can just imagine the handset mysteriously rising fro the desk on its own :-) |
| 20:04.01 | dpryo | :) |
| 20:04.15 | dpryo | I mean, answering the headset etc. |
| 20:04.54 | PakiPenguin | which code is used to do pickup from a zaptel group? |
| 20:04.57 | Bullseye_Network | I am TRYING to use NFS to share the outgoing directory for call files. Which direction is gonna work the best for asterisk? The share on the * server or the share on the remote server? |
| 20:04.57 | dpryo | Perhaps it only works with the h.323 firmware. |
| 20:06.10 | coppice | NFS == not fast system that should give a clue where it belongs :-) |
| 20:06.29 | Bullseye_Network | That crossed my mind too. |
| 20:09.22 | kRutOn | Bullseye_Network: I'd look at it from the standpoint of whether it's more critical for the asterisk server to read/write the files or for the remote server to read/write the files |
| 20:09.39 | kRutOn | I'd say put it on the * server since it's probably the most critical piece |
| 20:10.06 | kRutOn | having asterisk wait when the NFS server is unavailable is probably not desireable. |
| 20:12.02 | *** part/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
| 20:13.31 | PakiPenguin | i want to allow people to pickup other people's extensions when they are ringing and they are not around ( zap channels ) , i know *8# is the code to use , but where do i define the pickup groups and how? |
| 20:16.35 | Bullseye_Network | kRutOn:I currently have the nfs share on the * server and it kinda ignored the callout file that are created there from the remote server. |
| 20:17.39 | *** join/#asterisk Galeras (n=Galeras@litigaractivos1.att.net.co) |
| 20:18.20 | Galeras | hi! |
| 20:18.56 | *** join/#asterisk swytch (n=ezcall@d80-170-25-137.cust.tele2.fr) |
| 20:19.21 | *** part/#asterisk firejon (n=jellis@206-169-49-105.static.twtelecom.net) |
| 20:19.29 | swytch | i have a question about the g729 codec and asterisk .. |
| 20:19.39 | paryl | ok, my agents log in, but show queues shows them all unavailable, so they aren't getting calls. any idea why that would be? |
| 20:20.28 | swytch | does it matter if i have the licensed g729 package or not when i pass rtp thru * in codec g729. |
| 20:20.41 | Qwell[] | swytch: not if it's passthrough |
| 20:20.56 | Waverly360 | Is it possible to customize the softkey menus on a polycom phone? |
| 20:21.10 | Qwell[] | Waverly360: don't think so, no |
| 20:21.10 | swytch | Qwell[]: passthrough just means that g729 is coming in, and is whats going out too? |
| 20:21.22 | Qwell[] | swytch: and that nothing is trying to transcode it in between |
| 20:21.34 | Qwell[] | like, say, Playback, using a gsm file |
| 20:21.51 | swytch | Qwell[]: * will not ever try to transcode g729 if there is no licence, right? |
| 20:21.54 | Waverly360 | Qwell[]: So I can't even disable a softkey button that I don't want a user to have access to? Primarily the MyStat softkey on my 501. |
| 20:22.04 | Qwell[] | swytch: it might still try...but it'll fail |
| 20:22.14 | Qwell[] | Waverly360: ask [TK]D-Fender .. |
| 20:22.24 | Qwell[] | he's the resident polycom fanatic |
| 20:22.32 | swytch | Qwell[]: ..and is it shure that * does not choose some lossier variant of the codec? |
| 20:22.32 | Waverly360 | [TK]D-Fender: Are you present? |
| 20:22.32 | Qwell[] | I'm just the cisco guy :p |
| 20:22.42 | Qwell[] | swytch: lossier? |
| 20:22.43 | Waverly360 | Qwell[]: Thanks :) |
| 20:22.51 | swytch | Qwell[]: g729a |
| 20:23.12 | swytch | Qwell[]: in the negotiation for on the legs |
| 20:24.37 | swytch | Qwell[]: well, if you say that asterisk will use plain g729 as usual and only have the possibility of pass it right tru, then thats the asnwer. thanx |
| 20:28.14 | Strom_C | http://www.ip-centrex.org/whatis/index.html |
| 20:28.14 | Strom_C | http://www.ip-centrex.org/whatis/index.html[#asterisk] |
| 20:29.10 | swytch | Strom_C: was that for me? |
| 20:40.11 | Galeras | Please, does anybody knows how to detect a hangup on a Grandstream HT488, which FXO port is connected to an old panasonic KX-TEM824? |
| 20:42.56 | znoG | if I want to set a hard limit on how long a call can go last, should I use AbsoluteTimeout? I find the L option in dial USUALLY hangs up the line after X ms, but a couple of times it hasn't. |
| 20:45.13 | PakiPenguin | how does iaxmodem work? do i need another card apart from my zaptel card to use iaxmodem? |
| 20:50.26 | *** join/#asterisk jeremib (n=netnameu@c-71-203-209-162.hsd1.tn.comcast.net) |
| 20:51.18 | jeremib | what can I check to see why my text messages are not being sent when an voicemail arrives? |
| 20:51.46 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 20:53.13 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 20:53.50 | *** part/#asterisk jeremib (n=netnameu@c-71-203-209-162.hsd1.tn.comcast.net) |
| 20:55.54 | *** join/#asterisk Coeus (n=Coeus@ip24-255-125-43.dc.dc.cox.net) |
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| 21:00.41 | NewSole | hellno folks |
| 21:02.50 | *** part/#asterisk smackus (n=smackus@63.149.122.93) |
| 21:03.11 | Galeras | Please give me some advice about how to detect a hangup on a Grandstream HT488 which FXO port is connected to an old panasonic KX-TEM824. This Panasonic doesn't provide polarity reversal. It just plays a busy tone.. |
| 21:04.54 | syzygybsd | say i did some load testing on a T1... my provider wouldn't disconnect/give me congestion the next day would they? |
| 21:06.30 | *** join/#asterisk gorauskas (n=gorauska@66-224-20-131.atgi.net) |
| 21:06.41 | *** part/#asterisk gorauskas (n=gorauska@66-224-20-131.atgi.net) |
| 21:10.24 | ariel_ | have you gone out to American Lake? Oh cold is not the word. I can't belive people go swiming there. |
| 21:11.12 | ariel_ | Galeras, hummm I have no Idea. |
| 21:12.01 | *** join/#asterisk swytch (n=ezcall@d80-170-25-137.cust.tele2.fr) |
| 21:32.25 | *** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 21:33.51 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 21:36.52 | CoffeeKid | Anyone ever install astGUIclient? If so, do you know of a tutorial that's less obnoxious than the on one thier web site (the "From Scratch" one). |
| 21:37.05 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 21:37.09 | MikeJ[Laptop] | hello |
| 21:37.17 | CoffeeKid | hi |
| 21:37.29 | MikeJ[Laptop] | anyone know of a java based softphone that could be used off a webpage |
| 21:37.51 | MikeJ[Laptop] | doesn't need any fancy dial stuff.. just need to hit a button to call a hard coded voip address |
| 21:38.06 | MikeJ[Laptop] | but should handle nat and such |
| 21:38.20 | Katty | MikeJ[Laptop]: hey hun |
| 21:38.28 | MikeJ[Laptop] | so either iax, or sip with stun should work cool |
| 21:38.29 | Toerkeium | MikeJ[Laptop]: in which color? |
| 21:38.31 | MikeJ[Laptop] | hey Katty |
| 21:38.42 | MikeJ[Laptop] | Toerkeium, heh |
| 21:38.45 | Toerkeium | :P |
| 21:38.49 | *** join/#asterisk tamp4x (n=tampon@64.201.13.51) |
| 21:39.21 | tamp4x | i jsut ordered the g729 codecs, did the read me for the install, now when i try to turn asterisk i get this.... [codec_g729a.so]Ouch ... error while writing audio data: : Broken pipe |
| 21:39.25 | tamp4x | any ideas why? |
| 21:39.53 | MikeJ[Laptop] | tamp4x, that's a digium commercial product.. they should support it... have you tried support? |
| 21:40.25 | NewSole | Dam people Mike.... (Hi Mike) |
| 21:40.25 | tamp4x | it takes a while to get sto someoen irc is faster =] |
| 21:41.22 | NewSole | Hey Mike We got our Softphone almost wrapped up |
| 21:41.24 | MikeJ[Laptop] | NewSole? |
| 21:41.37 | MikeJ[Laptop] | don't even need a gui |
| 21:41.43 | Dr-Linux | tamp4x: kill mpg123 and then start asterisk |
| 21:41.48 | MikeJ[Laptop] | looking for somthing that can go embedded in a webpage |
| 21:41.50 | NewSole | o just people in news groups |
| 21:42.01 | MikeJ[Laptop] | just with a call and hangup |
| 21:42.11 | MikeJ[Laptop] | and cross platform. |
| 21:42.30 | NewSole | You use .Net for Web |
| 21:42.42 | Dr-Linux | MikeJ[Laptop]: let me know, if you got one |
| 21:43.03 | MikeJ[Laptop] | NewSole, do the users need to install that? |
| 21:43.05 | NewSole | if you do you can use Core of Softphone for Web |
| 21:43.17 | NewSole | nope |
| 21:43.21 | MikeJ[Laptop] | hmmm |
| 21:43.36 | MikeJ[Laptop] | even on linux? |
| 21:43.41 | NewSole | the whole softphone is 100% .Net |
| 21:43.58 | MikeJ[Laptop] | yeah.. so you need mono runtime and such on linux or it won't run? |
| 21:44.03 | tamp4x | now that i killed it it doenst even give the reason why g729 wont laod |
| 21:44.05 | NewSole | the APS.Net on web Server does it |
| 21:44.20 | MikeJ[Laptop] | how does that access local sound device? |
| 21:44.29 | NewSole | java |
| 21:44.33 | MikeJ[Laptop] | hmmm |
| 21:44.34 | MikeJ[Laptop] | nice |
| 21:45.07 | NewSole | a java Socket IO |
| 21:46.04 | NewSole | its used to pump audio data to sound card |
| 21:47.21 | pdtmobile | that sounds very rube goldbergish to me |
| 21:47.38 | Bullseye_Network | Ok here is an odd problem. Server A runs a perl script to create a callout file, NFS to Server B(Asterisk) file moved to outgoing directory. * ignores file. BUT if I open the file on EITHER server with VIM then resave it and put it in the correct directory it works fine. |
| 21:48.13 | Bullseye_Network | If I run the perl script on the Asteriskserver it created callout file and reads it with no problem |
| 21:49.03 | CunningPike | What are the good ACD statistics packages out there? I'm trying to get the AsteriskGuru one working, but it's proving to be a bit of a PITA |
| 21:49.09 | syzygybsd | oops.. I killed asterisk with too high a load :( |
| 21:50.41 | swytch | Bullseye_Network: may be a CR and LF thing. your perl script partially uses CR and CRLF, and vim fixes it. |
| 21:51.31 | Bullseye_Network | swytch: will check that, but it works fine if run on the actual asterisk server. |
| 21:52.19 | pdtmobile | why do you need .net webserver to serve a java applet? |
| 21:52.40 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 21:52.44 | swytch | Bullseye_Network: cmp and hexdump will tell you the difference after and before you edit it. |
| 21:53.10 | Bullseye_Network | swytch: will try that out now. |
| 21:53.44 | swytch | Bullseye_Network: what is the thing that doesnt manage to read it? |
| 21:54.03 | Bullseye_Network | swytch: Asterisk, its a callout file |
| 21:54.27 | syzygybsd | huh.. 1.2.9.1 seems less stable then 1.2.7 |
| 21:55.39 | *** join/#asterisk Waverly360 (n=mirc@209.12.249.243) |
| 21:55.46 | TheCops | There's a way to get Hold event with asterisk (When a call is on hold in the API manager event) ? |
| 21:57.22 | swytch | syzygybsd: killed * ? what happened? what was the load? |
| 21:57.35 | Bullseye_Network | swytch: it is the EOF |
| 21:58.03 | syzygybsd | lol.. well I spawned 70 some calls from sip to zap in about 10 seconds, on a pentium4 3.2.... |
| 21:58.21 | syzygybsd | it at least came back on 1.2.7.. 1.2.9 it just dies |
| 21:58.30 | swytch | Bullseye_Network: is it an explisit EOF file in it _before_ or _after_ your vi session? |
| 21:59.08 | Bullseye_Network | swytch: cmp: EOF on file1 file1 is BEFORE the VIM edit |
| 21:59.54 | swytch | Bullseye_Network: so vim actually removes the explisit EOF at the end. and * likes that. haha. what a heck. |
| 22:00.09 | Bullseye_Network | swytch: so that means the VIM toof the EOF off??? |
| 22:00.39 | swytch | Bullseye_Network: why is the EOF there in the first place? |
| 22:01.06 | Bullseye_Network | Its a file written by a perl script |
| 22:01.37 | Bullseye_Network | I dont know why one server is putting it and another not |
| 22:02.04 | swytch | Bullseye_Network: different perl version? |
| 22:03.32 | Bullseye_Network | swytch: same |
| 22:03.59 | swytch | Bullseye_Network: same OS on both? |
| 22:04.09 | Bullseye_Network | yes |
| 22:04.45 | swytch | Bullseye_Network: the original source of the data in that perl script is the same? over netweok? |
| 22:05.19 | swytch | s/netw.../network/i |
| 22:05.47 | Bullseye_Network | yes copied directly |
| 22:06.20 | swytch | Bullseye_Network: copied from what? |
| 22:09.09 | Bullseye_Network | ok switched it back to echo the callout file from within perl with same problem. BUT when I not do a cmp or hexdump they are exactally the same |
| 22:10.05 | Bullseye_Network | So I too the 2 compaired files and dropped them in the outgoing dir and it read the exited file and not the other |
| 22:10.09 | Bullseye_Network | AHHHH! |
| 22:10.37 | Bullseye_Network | So its not the eof |
| 22:13.28 | Bullseye_Network | the 2 files are identical in cmp and hexdump YET one asterisk ignores |
| 22:14.30 | *** join/#asterisk XARiUS (n=bdarcy@66-146-191-242.skyriver.net) |
| 22:14.37 | swytch | Bullseye_Network: so its the path or the permission or owner then. |
| 22:14.39 | Bullseye_Network | after 5 mins the callout file asterisk ignores it deletes it |
| 22:15.05 | Bullseye_Network | asterisk deletes the file to it has permission |
| 22:15.13 | Bullseye_Network | but not until 5 mins |
| 22:15.16 | swytch | s/to/so/ |
| 22:15.32 | Bullseye_Network | s/to/so/ |
| 22:15.33 | Bullseye_Network | lol |
| 22:15.36 | Bullseye_Network | yea |
| 22:16.02 | swytch | in this case asterisk is evil, and i dont know whats happening. |
| 22:16.30 | Bullseye_Network | It REFUESES to read it but has no problem deleteing it. :( |
| 22:17.15 | swytch | Bullseye_Network: yes. deleting a file depends on the permission of the dir containg it. |
| 22:17.37 | XARiUS | so, can anyone recommend a small nat router that will work with IP501's?? |
| 22:17.57 | swytch | Bullseye_Network: reading it depends on the permission flags of the file itself and the owner. |
| 22:18.35 | swytch | Bullseye_Network: what the owner and permissions of the file before and after you vim it? |
| 22:18.40 | *** join/#asterisk luke-jr_ (n=luke-jr@2002:1891:f657:0:20e:a6ff:fec4:4e5d) |
| 22:18.43 | Bullseye_Network | -rw-r--r-- 1 root root <---- on both files.... The one it will read and the one it wont. |
| 22:19.04 | swytch | ..and when you move it to that evil directory? |
| 22:19.04 | Bullseye_Network | and yes. Asterisk is running as root on this one. |
| 22:19.09 | Bullseye_Network | Same |
| 22:19.36 | swytch | whats the file name? |
| 22:20.03 | Bullseye_Network | callout_####### |
| 22:20.13 | Bullseye_Network | the # = a customer number |
| 22:21.27 | swytch | and * deletes for a specific customer, or does it delete only your non-vim versions? |
| 22:21.56 | Bullseye_Network | non-vim |
| 22:22.09 | Bullseye_Network | Or when a call hangs up for the ones it does read |
| 22:23.42 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 22:23.42 | swytch | summing up: file content, name, permission, owner and path is same, but when you "vim" it then * reads it. when * read it then * deletes it. |
| 22:25.23 | Bullseye_Network | <PROTECTED> |
| 22:25.53 | swytch | i must say "/&|@#!#&/(.*&%?+$}#" |
| 22:26.09 | Bullseye_Network | If I run the SAME script ON the * server it works fine. |
| 22:26.24 | Bullseye_Network | I can VIM the callout file on EITHER server and it will work. |
| 22:26.29 | Bullseye_Network | which make no sence |
| 22:26.54 | Bullseye_Network | BUT hexdump says both are the same, when I use ECHO in perl script |
| 22:27.07 | swytch | conclusion: vim makes the bytes go brown in the file. |
| 22:27.12 | Bullseye_Network | lol |
| 22:27.49 | swytch | and the brown color makes * shit on it in 5 minutes |
| 22:27.52 | Bullseye_Network | http://pastebin.ca/81831 |
| 22:28.02 | Bullseye_Network | this is the part of the script that writes the file |
| 22:28.24 | swytch | i must say that the brown color makes * shit on it in 5 minutes |
| 22:28.53 | swytch | i dont need the pastebin. two identical files ARE two identical files. this means the same bytes. |
| 22:29.14 | swytch | but my conclusion with the shitting octets remainss |
| 22:29.51 | nextime | swytch : are your absolutly sure that the two files are identical? have you tried an md5sum? |
| 22:30.10 | swytch | he did cmp on them |
| 22:30.15 | Bullseye_Network | and hexdump |
| 22:30.15 | swytch | no diffs |
| 22:30.24 | nextime | swytch : try md5sum. |
| 22:30.38 | swytch | Bullseye_Network: try md5sum |
| 22:30.52 | CunningPike | ~seen waverly |
| 22:31.06 | jbot | i haven't seen 'waverly', CunningPike |
| 22:31.06 | Bullseye_Network | will do |
| 22:31.11 | CunningPike | ~seen waverley |
| 22:31.15 | jbot | i haven't seen 'waverley', CunningPike |
| 22:31.30 | swytch | ~seen shitting_octets |
| 22:31.32 | jbot | i haven't seen 'shitting_octets', swytch |
| 22:31.40 | swytch | i have |
| 22:31.55 | Bullseye_Network | Same |
| 22:31.56 | CunningPike | ~seen waverly360 |
| 22:31.58 | jbot | waverly360 is currently on #asterisk (36m 19s), last said: 'Qwell[]: Thanks :)'. |
| 22:32.13 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 22:32.14 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 22:32.18 | CunningPike | Waverly360: You were asking about turning off the 'MyStat' softkey? |
| 22:32.22 | swytch | nextime: md5sum says their identical. |
| 22:32.31 | CunningPike | ~they're |
| 22:32.37 | nextime | Bullseye_Network : ok, NOW you can say that are identical for sure. |
| 22:32.53 | Bullseye_Network | GREAT it still doesnt work.... :( |
| 22:33.30 | swytch | nextime: theire not. the owner,permission,path,content are unchanged. the problem is that vim turns the octets magic. |
| 22:33.34 | nextime | Bullseye_Network : BUT you now know that the problem isn't in the file that you have compared, so, it's anyway a little step behind |
| 22:34.12 | swytch | nextime: so what attribute of a file are we unaware of here? |
| 22:34.39 | nextime | swytch : i don't know the problem, i'm saying about md5sum only cause i've see that Bullseye_Network say something about "vim hexdump and echo show that files are identical" |
| 22:34.50 | CunningPike | jbot, their is the possessive of them, as in "She cut off their tails with a carving knife". They're is a contraction of 'they are', as in "They're tail-less mice, now". |
| 22:34.51 | jbot | ...but their is already something else... |
| 22:34.53 | swytch | if you know, Bullseye_Network would be glad. |
| 22:34.57 | CunningPike | ~their |
| 22:34.58 | jbot | extra, extra, read all about it, their is a possessive pronoun, whereas they're is a contraction for "they are" |
| 22:35.13 | CunningPike | Good enough, jbot |
| 22:36.01 | nextime | maybe a different version or .vimrc of vim is the magic trick? |
| 22:36.22 | nextime | ( i'm saying something random ) |
| 22:37.15 | *** join/#asterisk tenlet (n=tenlet@pool-141-153-164-186.mad.east.verizon.net) |
| 22:39.01 | tenlet | hi, new to asterisk... anyone around to help a newb? |
| 22:39.48 | inv_Arp | tenlet: not if you dont ask a specific question... |
| 22:40.04 | tenlet | alright |
| 22:40.45 | tenlet | inv_Arp: i cant seem to get a text definition of asterisk, it is a software program that allows you to make phone calls and receive? basically your own phone provider? |
| 22:40.58 | [TK]D-Fender | tenlet : No. * is a PBX. |
| 22:41.16 | [TK]D-Fender | tenlet : In a way I guess you COULD do taht, but thats not the poitn |
| 22:41.51 | tenlet | trying to understand the function of the program, the about and the books i read seem vague on that part |
| 22:46.53 | Bullseye_Network | ls |
| 22:46.56 | Bullseye_Network | ops sorry |
| 22:47.08 | Strom_C | tenlet: essentially, asterisk is an extremely versatile and flexible telephone switching system |
| 22:47.41 | rob0 | /bin/ls: Permission denied |
| 22:48.20 | Bullseye_Network | del /* |
| 22:48.39 | Bullseye_Network | -r -f |
| 22:49.38 | CunningPike | ~lart AsteriskGuru queue_stats |
| 22:49.50 | CunningPike | Ah, that's better |
| 22:49.57 | Waverly360 | You rank Cunning? :) |
| 22:50.01 | Waverly360 | rang* even |
| 22:50.19 | CunningPike | I did - you had a question about turning off softkeys on the Polycom |
| 22:50.30 | Waverly360 | I do |
| 22:50.35 | Waverly360 | OR |
| 22:50.37 | Waverly360 | making them work |
| 22:50.43 | Waverly360 | I'll take either answer at this time ;) |
| 22:51.16 | *** join/#asterisk RoyK[at] (n=roy@chello080109196173.3.graz.surfer.at) |
| 22:51.22 | CunningPike | Waverly360: In your sip.cfg, look for <features . You will see the features listed there, with an .enabled attribute for each. Guess what that does? :) |
| 22:52.02 | Bullseye_Network | I think I figured out my callout file problem..... |
| 22:52.05 | Waverly360 | CunningPike: Hmm...it makes Julianne fries? |
| 22:52.07 | CunningPike | Waverly360: You can also specify that in a phone.cfg file, to enable/disable features by individual phone |
| 22:52.18 | Waverly360 | CunningPike: Well..I have presence enabled |
| 22:52.34 | CunningPike | Waverly360: Disabling it gets rid of the softkeys |
| 22:53.02 | Waverly360 | CunningPike: Well..here's the thing..I want presence enabled. Because I have a 601 with a sidecar that's actually functioning properly. |
| 22:53.04 | Bullseye_Network | swytch: remember the 5 min wait then delete the file? WELL the clocks are off by approx 5mins. |
| 22:53.29 | Waverly360 | CunningPike: What I'm also wanting though, is for the MyStat button and the Do Not Disturb button to affect the sidecar as well. |
| 22:53.30 | CunningPike | Waverly360: But you want to disable the softkeys on that phone? Can't be done |
| 22:53.44 | Waverly360 | CunningPike: I don't want them disabled if I can make them work correctly. |
| 22:54.03 | CunningPike | Waverly360: When you say affect the sidecar...... |
| 22:54.03 | Waverly360 | CunningPike: The problem is..the Statuses don't actually do anything on those phones that I can tell. |
| 22:54.06 | Bullseye_Network | swytch: So asterisk will not process a callout file until the creation time is current |
| 22:54.14 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 22:54.16 | [TK]D-Fender | Waverly360 : That requires support in * which it does not have ATM. |
| 22:54.28 | Waverly360 | CunningPike: If someone sets themself to away or busy..I want it to show up on the sidecar |
| 22:54.34 | CunningPike | Waverly360: No, they don't - you only get Online, Offline and On the Phone |
| 22:54.36 | Waverly360 | [TK]D-Fender: crap... |
| 22:55.00 | *** join/#asterisk shashu (n=shashwat@61.246.43.91) |
| 22:55.14 | shashu | hi |
| 22:55.27 | shashu | i have a query with asterisk |
| 22:55.47 | shashu | i am a new to it and can someone help me out in this |
| 22:55.48 | TheCops | netxusa own |
| 22:56.30 | Waverly360 | [TK]D-Fender: So it's an asterisk problem then. Are the phones supposed to send out packets when you change the status on them? |
| 22:57.18 | TheCops | There's a way to get Hold event with asterisk (When a call is on hold in the API manager event) ? |
| 22:57.24 | shashu | i need to install asterisk and i have a redhet 9 CD server. Can i install Asterisk 1.2.9.1 on redhat9 ? |
| 22:57.27 | shashu | please help me |
| 22:57.44 | Strom_C | yes |
| 22:57.51 | Waverly360 | I believe you can Shashu. Shouldn't be any real problems. |
| 22:57.58 | shashu | oh thanks |
| 22:58.11 | shashu | thanks waverly |
| 22:58.39 | anonymouz666 | led zeppelin owns |
| 22:58.40 | anonymouz666 | lol |
| 22:58.42 | anonymouz666 | :D |
| 22:59.18 | *** join/#asterisk terrapen (n=cjs@166.70.183.108) |
| 22:59.35 | Waverly360 | CunningPike: So the polycoms (501, 601) have those features for..what? |
| 23:00.03 | CunningPike | Waverly360: Come again? |
| 23:00.18 | Waverly360 | CunningPike: Heh..I'm just confused. I have status features on my phone..that do nothing. |
| 23:00.26 | Waverly360 | CunningPike: If they don't work..I wanna hide them. |
| 23:00.48 | CunningPike | Waverly360: But if you turn off presence, your SIP hints will stop working |
| 23:01.01 | CunningPike | Waverly360: Put some tape over them ;) |
| 23:01.01 | Waverly360 | CunningPike: Catch 22....damnit |
| 23:01.10 | CunningPike | ~catch22 |
| 23:01.23 | Waverly360 | CunningPike: Hah...not sure that'll be ok'd by my manager ;) but good idea |
| 23:01.28 | CunningPike | :D |
| 23:01.48 | CunningPike | Now, is anyone else using AsteriskGuru's queue_stats? |
| 23:02.45 | Waverly360 | CunningPike: I put in a support call with Polycom, just to ask them what they were supposed to do. The guy told me that the phones would determine what the pbx was capable of doing, and if the status feature wasn't handled by the pbx, that they simply wouldn't do anything on the phone. |
| 23:03.09 | Waverly360 | CunningPike: He claims that it does work on other pbx's...though he may have been BSing me. |
| 23:03.51 | CunningPike | Waverly360: It probably works on the commercial ones that Polycom has certified |
| 23:04.05 | CunningPike | RoyK[at]: Why? Why would you do that? That's just mean! |
| 23:04.11 | [TK]D-Fender | Waverly360 : they have it for OTHER PBX's |
| 23:04.24 | CunningPike | RoyK[at]: Anyway, it doesn't run on an iMac |
| 23:04.37 | RoyK[at] | CunningPike: exactly |
| 23:05.07 | Waverly360 | CunningPike: Hrm..... |
| 23:05.22 | CunningPike | I think RoyK[at] Is off his meds again |
| 23:05.58 | Waverly360 | CunningPike: Well..if that's the case...if the statuses do work with other phones..there must be a way to trick the phones into enabling those buttons.. |
| 23:06.15 | Waverly360 | CunningPike: I mean with other pbx's.. |
| 23:06.40 | Waverly360 | CunningPike: Then I would think the asterisk source code could be modified a bit to recognize when those statuses are set. |
| 23:07.05 | CunningPike | Waverly360: When can you have it ready? 1.4. is almost ready...... |
| 23:07.20 | Waverly360 | CunningPike: I need to know how to enable the features on the phone first. |
| 23:07.46 | Waverly360 | CunningPike: I'll have to brush up on my C++ though ;) it's been a few years. |
| 23:08.17 | CunningPike | Waverly360: I think, although I'm not sure, that the hone sets it's status OK - it's that Asterisk either doesn't ask, or doesn't do anything with the response |
| 23:08.24 | CunningPike | s/hone/phone/ |
| 23:08.58 | Waverly360 | CunningPike: Well..I know for a fact that the phone doesn't do ANYTHING on the network when I change my status. |
| 23:09.22 | CunningPike | Now, who's up for a little command line php support? I'm trying to run a php file that uses pgconnect() to connect to a database, except I keep getting 'Connection refused'. Wtf am I doing wrong? |
| 23:09.41 | CunningPike | Waverly360: Interesting...... |
| 23:10.03 | Waverly360 | CunningPike: I hooked my phone and computer up to a hub, and sniffed the packets. The phones don't send ANY packets when you hit "Do Not Disturb" or change the status via the "MyStat" softkey |
| 23:10.19 | Waverly360 | CunningPike: Commandline php eh? been a couple of years since I did that..I stick to perl mostly now |
| 23:10.22 | CunningPike | The same connection string works fine in another php file that's loaded in a browser |
| 23:10.36 | CunningPike | It's part of this queue_stats thing I' |
| 23:10.41 | CunningPike | I'm trying to get working |
| 23:10.44 | CunningPike | :( |
| 23:10.57 | Waverly360 | CunningPike: Oh...I'll bet it has to do with environment variables..I remember having a similar problem. |
| 23:11.11 | CunningPike | Ah - that may be so........ |
| 23:12.01 | Waverly360 | CunningPike: You can probably modify your code to print out a list of all environment variables in both scripts. |
| 23:12.01 | *** part/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 23:12.01 | Waverly360 | CunningPike: See what's set...might help you out. |
| 23:12.01 | CunningPike | OK- thanks |
| 23:12.21 | Waverly360 | It's been awhile... $_ENV? |
| 23:12.21 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:12.29 | Waverly360 | my php foo is gone |
| 23:13.28 | Waverly360 | [TK]D-Fender: Do you know how to trick the status buttons into working on the Polycoms? |
| 23:14.01 | [TK]D-Fender | Waverly360 : .... huh? |
| 23:14.13 | ariel_ | ~centos |
| 23:14.14 | jbot | from memory, centos is better than Fedora Core except for that silly bug, see ~centosbug for details |
| 23:14.14 | *** join/#asterisk Bullseye_Network (n=info@216.143.192.69) |
| 23:14.27 | ariel_ | ~centosbug |
| 23:14.28 | jbot | centosbug is, like, a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 23:14.40 | Waverly360 | [TK]D-Fender: Heh...well, you said that other PBXs support those statuses...meaning there has to be a way to enable the buttons on the phones. Right now they do nothing. |
| 23:16.44 | ariel_ | evening everyone |
| 23:20.30 | *** join/#asterisk knarfly (n=billtill@c-69-180-98-189.hsd1.fl.comcast.net) |
| 23:22.16 | *** join/#asterisk jaike (i=jaike@124.106.143.106) |
| 23:23.01 | jaike | anyone familiar with this CLI message? == Forcing Marker bit, because SSRC has changed |
| 23:24.06 | jaike | happens after SIP tries native bridge between two * servers |
| 23:24.29 | jaike | i get flooded with like 40 lines of it |
| 23:25.35 | [TK]D-Fender | Waverly360 : Doable = yes, done = no. |
| 23:25.45 | [TK]D-Fender | Waverly360 : get coding little monkey! |
| 23:26.32 | CunningPike | What's the command to see what services are on what ports - I'm having a brain frt |
| 23:26.38 | CunningPike | s/frt/fart/ |
| 23:27.14 | Waverly360 | netstat -amv |
| 23:27.17 | Waverly360 | ? |
| 23:27.26 | Waverly360 | netstat -anv |
| 23:27.31 | Waverly360 | my bad...I think that's right |
| 23:27.37 | CunningPike | THanks :D |
| 23:27.57 | Waverly360 | netstat -anp |
| 23:27.59 | Waverly360 | damnit |
| 23:28.05 | Waverly360 | I keep getting the options wrong :P |
| 23:28.40 | Waverly360 | [TK]D-Fender: Well..any idea on where I can get info on how to do that? |
| 23:28.56 | Waverly360 | [TK]D-Fender: I'm serious about actually implementing that in Asterisk..but I need a good kickstart. |
| 23:30.28 | *** part/#asterisk knarfly (n=billtill@c-69-180-98-189.hsd1.fl.comcast.net) |
| 23:31.22 | CunningPike | OK - there's something borked with my postgresql - I can connect locally (from console) but not to port 5432 |
| 23:31.40 | Waverly360 | ummm |
| 23:31.53 | justinu | you have to enable the tcpip sockets option in the pgsql.conf |
| 23:32.11 | justinu | otherwise it only works on unix sockets |
| 23:32.27 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 23:33.02 | CunningPike | justinu: Would that affect the ability of pgconnect to, like, pgconnect? |
| 23:33.28 | CunningPike | justinu: Even from a script run on the console? I guess it would, come to think of it..... |
| 23:33.52 | justinu | depends on whether pgconnect is trying to use unix or tcpip sockets |
| 23:37.24 | *** part/#asterisk jaike (i=jaike@124.106.143.106) |
| 23:38.05 | CunningPike | justinu: OK - getting a hba error now, so that fixed it - thanks! |
| 23:38.10 | [TK]D-Fender | Waverly360 : * already shows ringing, on call, idle, and at least one other useful one... |
| 23:38.31 | Waverly360 | [TK]D-Fender: But I need to know how to enable the statuses on the phone. |
| 23:38.41 | Waverly360 | [TK]D-Fender: That's the issue I'm having right now. |
| 23:38.42 | [TK]D-Fender | Waverly360 : as to where to start, i wouldn't know... I'm not a C programmer and don't know * that well/ |
| 23:39.34 | Waverly360 | [TK]D-Fender: Well..if I can figure out how to make the Polycoms actually NOTIFY asterisk when I change my status...then I can do the rest. |
| 23:40.02 | Waverly360 | [TK]D-Fender: Right now the buttons are worthless. |
| 23:42.50 | Bullseye_Network | YEAH!!! Taking apache off the server fixed my deadlock issues. |
| 23:43.01 | *** part/#asterisk Bullseye_Network (n=info@216.143.192.69) |
| 23:43.12 | Waverly360 | but apparently not his network connectivity... |
| 23:48.16 | file | there is no network... |
| 23:48.27 | mishehu | there is no spoon |
| 23:48.35 | file | mishehu, ran out |
| 23:48.42 | Waverly360 | hrm |
| 23:50.36 | RoyK[at] | coppice: ping |
| 23:51.23 | *** join/#asterisk Bullseye_Network (n=info@216.143.192.69) |
| 23:51.47 | Bullseye_Network | im running 1.2.9.1 and just stared getting Forcing Marker bit, because SSRC has changed why? |
| 23:52.11 | RoyK[at] | because of users |
| 23:52.37 | file | Bullseye_Network: is it causing a problem? |
| 23:53.04 | Bullseye_Network | yes. It making my screen so I cant read itits throwing like 10-20 a second |
| 23:53.22 | Bullseye_Network | and causes load to go up when I turn up verbosity |
| 23:53.45 | Waverly360 | ah screw it today. I can't find much more info. Hopefully someone from polycom will have emailed me by next week. I'm taking off for the weekend. Have a good one everyone. |
| 23:54.21 | Bullseye_Network | I thought they fixed that after 1.2.7 |
| 23:56.44 | XARiUS | Can anyone recommend a small nat router that will work with IP501's? |
| 23:56.52 | XARiUS | I'm ready to kick this wrt in the nuts. |
| 23:57.00 | XARiUS | (if it had any) |
| 23:57.02 | ariel_ | network router the linksys wrt54g work fine |
| 23:57.11 | XARiUS | well this is the *new* v6 one.. |
| 23:57.16 | ariel_ | ahh |
| 23:57.21 | XARiUS | I couldn't find any v2-4's |
| 23:57.26 | ariel_ | most of mine are still in the 4.0 version |
| 23:57.26 | XARiUS | I *know* those work. |
| 23:57.47 | ariel_ | XARiUS, what is the issue your having? |
| 23:57.48 | XARiUS | so before I goto frys and drop 300 in random routers to test, I figured I'd ask if anyone's had success with some. |
| 23:58.07 | XARiUS | from here, I have a wrt54g running sveasoft talisman.. it handled the phones just the way it should to my remote * box.. |
| 23:58.17 | ariel_ | I use for most of my biz accounts one setup on monowall |
| 23:58.23 | XARiUS | my client's office, just got a new wrt54g v6.. the phones requests never even MAKE it to *. |
| 23:58.25 | XARiUS | nothing, nodda. |
| 23:58.43 | ariel_ | http://www.logicsupply.com/default.php/cPath/73?gclid=CIDa2aDqgIYCFUw2OAodckPIEg |
| 23:58.45 | XARiUS | no failed reg, nothin. |
| 23:58.59 | XARiUS | yikes |
| 23:59.03 | XARiUS | nice box, too pricey. |
| 23:59.06 | znoG | if I want to set a hard limit on how long a call can go last, should I use AbsoluteTimeout? I find the L option in dial USUALLY hangs up the line after X ms, but a couple of times it hasn't. |
| 23:59.15 | XARiUS | this office has 3 ip501's in it.. nothing that nice required |
| 23:59.17 | XARiUS | lol |
| 23:59.33 | ariel_ | I understand |
| 23:59.49 | XARiUS | I searched all thru frys at the serial numbers.. all they have are v5 and v6's.. |