| 00:00.29 | jamicque | no the network is ok. I've tested it in numerous fdifferent networks (different machines) etc. |
| 00:00.35 | jamicque | The result is always the same |
| 00:01.13 | jamicque | also in local network (the statiscis of connectio n between 2 sip phones) |
| 00:02.41 | jamicque | I've also google'ed it and everyone has similar statistics but no one has taken it under consideration |
| 00:05.18 | hardwire | :( |
| 00:06.07 | citywok | hardwire the same bug exists in 1.6.1.9 lol |
| 00:06.15 | hardwire | :( |
| 00:08.11 | citywok | bug report submitted |
| 00:09.53 | *** join/#asterisk josh (n=josh@c-67-177-6-66.hsd1.ut.comcast.net) |
| 00:09.56 | *** part/#asterisk josh (n=josh@c-67-177-6-66.hsd1.ut.comcast.net) |
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| 00:11.03 | josh | I have an Aastra phone that won't download a new firmware, I've tried two different tftp servers, HTTP, and FTP. I've tried when it automatically boots, from the Web UI and from the Web-Recovery Mode |
| 00:11.39 | josh | The phone just aborts the transfer after reading a little of the file... |
| 00:12.33 | *** join/#asterisk Elvanor (n=elvanor@gentoo/developer/elvanor) |
| 00:12.49 | josh | It's currently running 2.1.1 firmware which is from June 2007 |
| 00:13.25 | Elvanor | I know next to nothing about VoIP services, but I wonder if someone can help me. Is it possible to get a permanent normal phone number that will ring on my computer via SIP ? |
| 00:14.46 | *** join/#asterisk kamikami (n=alisajja@brn29-1-82-245-188-240.fbx.proxad.net) |
| 00:16.55 | Katty | hi |
| 00:21.29 | Katty | http://www.youtube.com/watch?v=4Y7TV0gEri8 <- omnomnomnomnomnom |
| 00:24.31 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:24.45 | Elvanor | Can someone point me to a channel where I can get advice about what I want? |
| 00:24.57 | Elvanor | Since I think it's not really asterisk related |
| 00:25.23 | Qwell | ~itsplist-us |
| 00:25.24 | infobot | extra, extra, read all about it, itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 00:25.25 | Qwell | Elvanor: ^ |
| 00:27.59 | Elvanor | Are there some companies that propose several phone numbers in different countries? all linked to a SIP account? |
| 00:28.35 | Katty | Elvanor: call them and ask. |
| 00:29.59 | Katty | http://www.youtube.com/watch?v=DPf8GcNNybg <- :> |
| 00:32.22 | jaytee | watching Star Trek the reboot |
| 00:33.36 | *** join/#asterisk Tech_Travis (n=tech_tra@mail.techglia.com) |
| 00:33.47 | elliot98 | wonders which version of * did the USS Enterprise us |
| 00:36.39 | jaytee | they don't run * |
| 00:36.58 | jaytee | they run Switchvox-LCARS |
| 00:46.26 | *** join/#asterisk Tako-san (n=Tako-san@p1216-ipad308osakakita.osaka.ocn.ne.jp) |
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| 00:49.47 | s34n | is there a way to connect a meetme with an ices command? |
| 00:50.26 | Katty | pokes jblack |
| 00:51.51 | *** join/#asterisk ming_zym (n=ming_zym@121.0.29.237) |
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| 01:19.32 | *** join/#asterisk WiCoder (n=bobbo@c-75-74-232-212.hsd1.fl.comcast.net) |
| 01:19.32 | Katty | pokes people |
| 01:21.02 | flyback | won eight 256 meg registered ecc sticks for 99 cents on ebay, micron ones made for dell :) w00h000 |
| 01:22.16 | Katty | :> |
| 01:22.46 | WiCoder | slaps WiCoder around a bit with a large trout |
| 01:23.56 | akyaki_newbie | Hi dlynes, I've tried 'ring splash', but it's for BLF ... |
| 01:24.31 | akyaki_newbie | It doesn't do anything 'On hold', or do you know somehting special ? |
| 01:27.11 | *** join/#asterisk Julian|Work (n=jostrow@74.85.23.194) |
| 01:27.13 | *** part/#asterisk Julian|Work (n=jostrow@74.85.23.194) |
| 01:28.09 | *** join/#asterisk ariel_ (n=ariel_@c-71-196-99-26.hsd1.fl.comcast.net) |
| 01:28.20 | Katty | hi ariel_ |
| 01:29.49 | ariel_ | hi Katty |
| 01:31.52 | *** join/#asterisk akyaki_newbie (n=aki@S0106004f4e164e77.gv.shawcable.net) |
| 01:33.49 | flyback | akyaki_canuck! |
| 01:33.56 | flyback | bites akyaki_newbie |
| 01:35.33 | jblack | wow, a judge came awfully close to saying 'everyone's a dumbass' |
| 01:35.42 | jblack | The judge ruled Wednesday that while the ads may be "sneaky," they are not misleading, and do not justify any action to take them off the air. The judge did state however, that most TV watchers are "semi-catatonic" and therefore can easily misinterpret information presented. |
| 01:35.44 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:38.26 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 01:39.24 | akyaki_newbie | whois flyback |
| 01:40.08 | flyback | a canuck hater |
| 01:40.19 | flyback | canada must pay for it's many acts of terrorism |
| 01:40.25 | flyback | such as the canadian rapper "snow" |
| 01:40.26 | flyback | ducks |
| 01:42.03 | Tako-san | ignore the buzzing fly. |
| 01:43.12 | WiCoder | a unrolls the fly tape eh |
| 01:43.13 | Katty | jblack: i agree with the judge |
| 01:43.37 | Katty | jblack: also, what adverts were these? |
| 01:44.24 | flyback | i'm kidding around akyaki_newbie |
| 01:44.29 | flyback | have a good day bud :) |
| 01:44.46 | jblack | katty: the 3g ads. |
| 01:44.55 | Katty | link? |
| 01:45.26 | jblack | Katty: http://www.engadget.com/2009/11/03/atandt-sues-verizon-over-theres-a-map-for-that-ads/ |
| 01:46.52 | Katty | nods |
| 01:46.58 | Katty | well it says right there 5x more 3g coverage |
| 01:47.11 | Katty | but you're right, people will just think /coverage/ in general |
| 01:47.19 | Katty | no more misleading that Happy Meals |
| 01:47.46 | jblack | I didn't express an opinion. I was reporting on a judge's statement. :) |
| 01:47.50 | flyback | what verzion should do is |
| 01:47.59 | Katty | jblack: (= |
| 01:48.01 | flyback | say "ok we will talk about your wiretapping skills instead" |
| 01:48.01 | Katty | jblack: <3 |
| 01:48.18 | Katty | http://i.imgur.com/tpELY.jpg <- hahaha |
| 01:48.37 | flyback | needs to get 2 512mb unbuffered pc133 dimms next for his mini-itx :P |
| 01:48.43 | jblack | Though I don't necessarily disagree with the judge. well over 80% of the US populate is christian. About half of the US is _republican_ |
| 01:48.56 | flyback | what's your point |
| 01:49.14 | jblack | That there's no shortage of stupid people |
| 01:49.19 | flyback | yawns |
| 01:49.35 | Katty | that's rather rude, flyback |
| 01:49.44 | jblack | Maybe he's tired. ;) |
| 01:49.47 | flyback | actually he's the one being rude |
| 01:49.51 | flyback | i'm just blowing him off |
| 01:50.01 | Katty | hugs jblack |
| 01:50.06 | Katty | jblack: yes perhaps he is. |
| 01:50.19 | jblack | flyback: You did ask me what my point was..... |
| 01:50.24 | flyback | I know I did :) |
| 01:50.45 | jblack | I thought it was obvious, so I'm guessing you're either republican or christian, or both. |
| 01:50.59 | flyback | not so much republican anymore |
| 01:51.08 | flyback | I say just drop a nuke on dc and start fresh |
| 01:51.09 | jblack | does a touchdown dance |
| 01:51.13 | [TK]D-Fender | "Christian". Yeah... that's nice and vague |
| 01:51.15 | flyback | like the big red reset button on my 286 |
| 01:51.18 | Katty | hi fender. |
| 01:51.37 | [TK]D-Fender | Katty: You have a few interesting ones at the office and in your family as I recall :) |
| 01:51.40 | jblack | time to go use the trudgemill. bbl |
| 01:51.57 | Katty | well...interesting is putting it nicely. |
| 01:51.58 | Katty | but yes. |
| 01:52.20 | flyback | I won't be around in a few more yrs anyways |
| 01:52.29 | flyback | so the world can burn to a crisp for all I care |
| 01:53.03 | Katty | [TK]D-Fender: http://imgur.com/vnpKL.jpg <- i found your drawing. it's cute. |
| 01:53.36 | [TK]D-Fender | Katty: LOL... |
| 01:53.54 | *** join/#asterisk xa0z (n=Interex@75-129-244-178.dhcp.mtvr.il.charter.com) |
| 01:54.28 | xa0z | Anyone have any reccommendations for an inexpensive CNAM lookup? Maybe 100 lookups a month, or less? |
| 01:54.36 | Katty | well hello there |
| 01:55.56 | xa0z | I have Flowroute but not getting any CNAM on incoming calls, so I would assume they don't supply it. |
| 01:55.59 | flyback | I gave up on life yrs ago |
| 01:56.02 | flyback | but anyways ... |
| 01:56.46 | *** join/#asterisk KingDavidNYC (n=Chris123@pool-96-224-162-219.nycmny.east.verizon.net) |
| 01:57.58 | KingDavidNYC | Hello everybody!!! |
| 01:58.36 | xa0z | jblack, :) |
| 02:01.49 | *** join/#asterisk kamikami (n=kamikami@brn29-1-82-245-188-240.fbx.proxad.net) |
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| 02:07.26 | *** join/#asterisk manxpower (n=ewieling@107.sub-70-214-135.myvzw.com) |
| 02:07.54 | *** join/#asterisk Deeewayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) |
| 02:07.54 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 02:08.30 | ariel_ | ~nat |
| 02:08.31 | infobot | [nat] Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 02:09.16 | *** join/#asterisk ent (n=ent@unaffiliated/ent) |
| 02:09.30 | phix | good idea |
| 02:09.35 | phix | reading docs is how you learn stuff |
| 02:10.20 | Katty | hugs Deeewayne |
| 02:10.40 | Deeewayne | hello Katty! |
| 02:10.48 | xa0z | No one has an idea? |
| 02:15.35 | Deeewayne | I have an idea |
| 02:15.49 | xa0z | But probably not pertaining to my question above. |
| 02:16.03 | Deeewayne | unfortunately, I missed your question. :-/ |
| 02:16.41 | xa0z | Anyone have any reccommendations for an inexpensive CNAM lookup? Maybe 100 lookups a month, or less? |
| 02:16.43 | xa0z | ^ |
| 02:17.22 | Deeewayne | no recommendation, sorry |
| 02:18.37 | xa0z | I'm with Flowroute, and I'm starting to think they don't have CNAM on inbound... but someone else told me they did. |
| 02:19.52 | jblack | europe is in trouble again |
| 02:20.03 | xa0z | Hello James. |
| 02:20.11 | Katty | what happened this time |
| 02:20.17 | Katty | did louis walsh get a photo with angelina jolie? |
| 02:21.12 | p3nguin | With VoIP.ms, you have to explicitly enable CNAM lookups so they can charge you per call. |
| 02:21.47 | xa0z | p3nguin, not an option with Flowroute, that I've seen. Someone else has said they do offer CNAM on inbound, but that it's a hit/miss situation. |
| 02:21.55 | xa0z | :) |
| 02:22.24 | xa0z | pokes jblack in the eye |
| 02:22.52 | jblack | Gas supplies again. |
| 02:23.05 | xa0z | bets jblack has no clue who he is. |
| 02:23.10 | xa0z | yet... |
| 02:23.21 | Katty | http://tinypic.com/view.php?pic=accxp4&s=5 <- wow, nice. |
| 02:23.39 | Katty | that's that french girl |
| 02:23.49 | Katty | can't think of her name. |
| 02:23.50 | xa0z | Katty, that file is so old. I remember seeing that in my young IRC days. |
| 02:24.02 | jblack | xa0z: I don't play "who am I" games. Sorry. |
| 02:24.02 | Katty | xa0z: good for you. |
| 02:24.16 | jblack | Ahh. Hi |
| 02:24.24 | Interex | ;) |
| 02:24.42 | jblack | friend me on facebook. |
| 02:24.50 | Interex | I don't use Facebook. |
| 02:24.52 | Interex | It's evil. |
| 02:25.00 | jblack | The face part, or the book part? |
| 02:25.16 | Interex | hah, the site in general |
| 02:25.44 | p3nguin | drmessano probably knows if Flowroute has the feature or not. I'm pretty sure he uses their services and is probably familiar with what they have to offer. |
| 02:25.47 | jblack | You sound like my grandfather and computers. :) |
| 02:26.01 | jblack | before you know it, you'll be screaming at kids to get off yer lawn |
| 02:26.07 | Interex | jblack, Yeah, but you know I know my stuff :P |
| 02:26.38 | Interex | I didn't even know you were on here until I saw an IRC log from a google search. |
| 02:26.46 | jblack | so, how are you doin |
| 02:27.01 | Interex | Life is okay. |
| 02:27.14 | Interex | Have I talked with you since March ? |
| 02:27.32 | jblack | That's back when I popped over on dal for an afternoon? |
| 02:27.56 | Interex | Probably. I don't get on DALnet much. Usually to bug someone in the ole channel. |
| 02:28.04 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 02:28.12 | jblack | Yeah. march would probably be that last time. |
| 02:28.30 | Interex | I'm forgetting if I told you I got custody of my daughter. |
| 02:28.44 | jblack | No, that's excellent news! |
| 02:28.57 | Interex | Yea, good times. |
| 02:29.25 | flyback | is still trying to find people he lost contact info for 10 yrs ago on this network :/ |
| 02:29.29 | jblack | you see, if you had a facebok page, I could be subscribed to it, and I'd know important shit like that! |
| 02:29.38 | flyback | about to start putting my recovery box together |
| 02:29.43 | Interex | jblack, haha, I wouldn't ever update it |
| 02:29.43 | flyback | to pull data from some old damaged hd's |
| 02:30.17 | jblack | What else is goin on with you? |
| 02:30.37 | Interex | Not too much really. Been workin, playing around with voip stuff, etc. |
| 02:31.00 | jblack | You'll have fun with that. = |
| 02:31.29 | jblack | This is a good place. An open place, though. If you play the "I want an answer without explaining things", you won't get very far here... just a heads up. |
| 02:31.52 | Interex | I've had voip for a long while now, just within the past several months have I started to mess with my own PBX rather than just using a DTA/ATA. |
| 02:31.58 | Interex | I don't use Asterisk, :( |
| 02:31.58 | jblack | People here will plonk you for a week if you hand over just 1/2 a config file. |
| 02:32.12 | jblack | Ahh, ok. |
| 02:32.31 | jblack | Yeah, you used to um... packet8, right? |
| 02:32.37 | Interex | I was with 8x8 for a long time. |
| 02:32.41 | Interex | Then moved to CallCentric |
| 02:32.45 | Interex | Now with Flowroute. |
| 02:33.29 | Interex | CallCentric is one fucked up ITSP. |
| 02:34.01 | *** join/#asterisk RypPn (i=TuMbL@rosscom.co.uk) |
| 02:34.13 | jblack | What sort of problems did they give you? |
| 02:34.58 | Interex | ptime was a big issue for a while, then DNS SRV. They had their weights ALL wrong, finally myself and a developer friend contacted them and got that fixed. |
| 02:35.26 | Interex | I know you are, or were with them, but for me... they sucked big time. |
| 02:35.27 | jblack | huh. I see 2ms between them and my linode. |
| 02:35.44 | jblack | Them and others. I think I have a balance at 5 or 6 itsps. |
| 02:36.07 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 02:36.47 | Interex | It took a lot of communicating with CallCentric to fix their DNS Weights. |
| 02:37.06 | Interex | They claimed they with in compliance with RFC standards |
| 02:37.30 | Interex | Actually, I think right now it's all set to 0 |
| 02:37.38 | jblack | as far as I'm aware, there are no rules with weighting. |
| 02:37.55 | jblack | Though I haven't looked at the standard in quite some time... |
| 02:37.56 | Interex | I dunno that there are or not, just a response I got from them. |
| 02:38.11 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 02:38.34 | jblack | I don't think there's many dns servers that really care about 'em. It's a nearly-forgotten part of the protocol. |
| 02:38.44 | jblack | Other than for mx records, of course. ;) |
| 02:39.13 | jblack | Even then, I've seen a mailserver or three that won't even bother contacting a secondary mx. |
| 02:39.25 | Interex | The issue was registration. |
| 02:39.41 | Interex | Would register to 1 IP, then auth fail on another, then so on, then calls would be hit or miss. |
| 02:40.05 | jblack | so your client was making frequent lookups. |
| 02:40.19 | jblack | what did their zone timeouts look like? |
| 02:40.56 | Interex | I don't remember, it's been a while since I had to dick around with them. I'm looking to see if I have any logs left from my conversations with them |
| 02:41.35 | jblack | nah. don't bother. |
| 02:42.09 | Interex | So far Flowroute has been great quality... just no incoming CNAM as of yet, even if they do supply it. |
| 02:43.51 | Interex | Not to mention I've been fucking around with LCDd, and the MTXORB Driver on FreeBSD. |
| 02:43.56 | Interex | GAH, been driving me nuts |
| 02:45.23 | Katty | http://melayukini.net/wp-content/uploads/2009/11/funny-baby.JPG <- eppigy as a kid |
| 02:45.35 | jblack | I haven't heard anyone mention freebsd in awhile. =) |
| 02:45.55 | Interex | heh |
| 02:46.31 | Interex | I run all my stuff on my little custom FreeBSD box. |
| 02:46.54 | Interex | I took an old Motorola Digital cable tuner box and built my mini-itx setup inside it. |
| 02:46.58 | jblack | did they ever get smp into fbsd? |
| 02:47.38 | jblack | Never mind. I see they did about 10 years ago |
| 02:47.39 | Interex | Heh, like back in 2003 or something |
| 02:48.05 | Interex | brb |
| 02:48.27 | jblack | Yeah, I was thinking of openbsd, and it went in in '04 |
| 02:49.30 | p3nguin | I've been running SMP on FreeBSD for several years, but only run a single core OpenBSD box. |
| 02:49.59 | Katty | Maliuta: http://i.imgur.com/WU5vZ.jpg <- |
| 02:50.40 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 02:53.37 | *** join/#asterisk levity (n=levity@unaffiliated/canuck) |
| 02:54.16 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 02:56.55 | TJNII | Cute pic |
| 02:58.11 | jblack | That's so me! |
| 02:58.55 | Interex | And me, http://msp50.photobucket.com/albums/f317/gatorflyer/big_nutts.jpg |
| 02:58.59 | Interex | ;) |
| 02:59.33 | jblack | . |
| 02:59.52 | jblack | I'm off to do chores. have fun all |
| 03:00.25 | Interex | heh, later man |
| 03:06.01 | *** join/#asterisk levity (n=levity@unaffiliated/canuck) |
| 03:10.23 | Katty | hmmmmmmmmmmm |
| 03:16.02 | Katty | http://imgur.com/o23i5.jpg |
| 03:16.43 | manxpower | Aren't Squirrels the incarnation of Evil? |
| 03:16.58 | Katty | no |
| 03:18.14 | manxpower | Katty: http://www.anti-squirrel.com/ |
| 03:19.27 | manxpower | I KNEW IT!!!!! "Sure they look cute and harmless, seeming to live their lives eating nuts, playing in the trees, and chasing eachother across your front lawn. The fact is they have far bigger plans: Global domination and human enslavement. " |
| 03:21.25 | Katty | mhmm |
| 03:21.35 | *** part/#asterisk Interex (n=Interex@75-129-244-178.dhcp.mtvr.il.charter.com) |
| 03:26.21 | *** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru) |
| 03:27.38 | KingDavidNYC | manxpower, do you remember a coupe of days ago I was asking anybody to help me create an outgoing call through the pstn port of a linksys? |
| 03:29.35 | flyback | manxpower, no that's.....................STUPID CANUCKS |
| 03:29.39 | flyback | not squireels |
| 03:32.45 | *** join/#asterisk yosi1234 (i=Bmw_Toro@CPE001217239daa-CM000e5c6f225a.cpe.net.cable.rogers.com) |
| 03:38.26 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 03:40.36 | jblack | Katty: ping |
| 03:43.02 | *** join/#asterisk Meaw (n=dino@188.161.191.27) |
| 03:55.45 | Katty | jblack: pong! |
| 04:02.56 | Katty | jblack: going to bed here shortly. find me aim if i'm not around when you get back |
| 04:03.47 | jaytee | nitey nitey |
| 04:04.07 | jaytee | hugs katty |
| 04:05.17 | jblack | Katty: Ahh. |
| 04:05.49 | jblack | Tomorrow then. It's just a cooking thing |
| 04:12.45 | *** join/#asterisk DarkRift (n=dark@modemcable015.68-200-24.mc.videotron.ca) |
| 04:13.58 | flyback | DarkCANUCK |
| 04:13.59 | flyback | DarkCANUCK |
| 04:14.02 | flyback | bites DarkRift |
| 04:22.19 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 04:25.15 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
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| 04:37.33 | flyback | bbl |
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| 04:48.21 | p3nguin | ~sipnat |
| 04:48.27 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:58.54 | *** join/#asterisk errotan (n=errotan@81.0.115.48) |
| 05:08.44 | ChannelZ | can anyone resolve www.ubercart.org ? |
| 05:12.05 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
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| 05:21.50 | p3nguin | Anyone having any trouble with DTMF when using Google Voice and Gizmo together with Asterisk? First I wasn't able to verify my Gizmo number when GV called it (and it was SIP forwarded to *). I finally got that taken care of by forwarding Gizmo to my Skype account, where I was able to enter DTMF successfully. But now when I call my GV number, which forwards to Gizmo, which SIP forwards to *, I cannot enter my PIN once * picks up. |
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| 05:25.33 | *** join/#asterisk brunner (n=chris@24-158-231-185.dhcp.leds.al.charter.com) |
| 05:25.44 | brunner | Can I hire someone here to backport ChanSpy from 1.6 to 1.4? |
| 05:26.22 | brunner | I mean the version of ChanSpy that is in 1.6, of course. I'm well aware that there is already a ChanSpy function in 1.4. |
| 05:26.52 | *** join/#asterisk DND (n=arabia@80.227.221.34) |
| 05:27.08 | DND | guys, anyone know a ulaw player for windows? |
| 05:28.32 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 05:29.12 | ChannelZ | uses Sound Forge.. |
| 05:31.06 | [TK]D-Fender | brunner: The channel model changed somewhat significanlt from 1.4 to 1.6 so I'm betting that isn't going to be feasible. What was your reason for it? |
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| 05:33.04 | [TK]D-Fender | brunner: The channel model changed somewhat significanlt from 1.4 to 1.6 so I'm betting that isn't going to be feasible. What was your reason for it? |
| 05:33.09 | brunner | [TK]D-Fender: there's a feature that my developer needs, and we've already put tons of development into 1.4 |
| 05:33.35 | [TK]D-Fender | brunner: Congratulations on painting, and continuing to paint yourself into a corner |
| 05:34.14 | brunner | [TK]D-Fender: your incredible social skills must be why you're constantly getting laid |
| 05:35.50 | [TK]D-Fender | brunner: Oh this was CONSIDERABLY more gentle than many would give this scenario |
| 05:37.12 | [TK]D-Fender | brunner: As long as you're realizing how big the stream you are swimming upwards is |
| 05:38.02 | [TK]D-Fender | brunner: This particular one is really tough |
| 05:40.59 | ChannelZ | especially if he can't even keep is IRC running |
| 05:47.32 | *** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1096762451.dsl.bell.ca) |
| 05:49.29 | DND | guys, anyone know a ulaw player for windows? |
| 05:50.49 | dlynes | DND: audacity? |
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| 05:57.45 | DND | dlynes, tried that but its giving out a chipmunk voice |
| 05:57.54 | DND | its very fast |
| 05:58.04 | p3nguin | You've got the sampling rate set wrong. |
| 05:58.32 | DND | hmm is that in the options? i never tried audacity before |
| 05:59.14 | DND | lemme check |
| 05:59.21 | DND | but what should be the sampling rate? |
| 05:59.28 | DND | for ulaw/alaw? |
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| 06:01.50 | DND | p3nguin, i just changed the sample rate to 8khz and sample format to 16-bit |
| 06:01.53 | [TK]D-Fender | checkout time, later all |
| 06:01.55 | DND | but its still fast |
| 06:04.18 | p3nguin | hmm |
| 06:07.33 | ChannelZ | * is playing fast or Audacity is? |
| 06:07.44 | DND | audacity |
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| 06:08.14 | DND | i have the recorded ulaw file in my win desktop and trying audacity to play it |
| 06:10.28 | *** join/#asterisk jermudgeon (n=jhaustin@216-67-61-242.static.acsalaska.net) |
| 06:11.57 | ChannelZ | you did Import Raw? |
| 06:13.39 | DND | sorry i think this is not the right forum :) im using *now with freepbx |
| 06:13.47 | DND | and i set the recording to save as ulaw |
| 06:14.25 | ChannelZ | sighs |
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| 06:59.37 | jkroon | hi guys, can anybody please explain to me CONGESTION vs CHANUNAVAIL ? |
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| 08:16.44 | Nest0r | agentlogin application not loaded, it worked before, how can I do |
| 08:16.45 | Nest0r | what charge? |
| 08:16.54 | bala | hi, could someone help me get an idea of what is going on here: Dropping incompatible voice frame on Mobile/zphone-617f of format slin since our native format has changed to 0x0 (nothing) |
| 08:26.06 | *** join/#asterisk [netman] (n=netman@126.Red-88-19-164.staticIP.rima-tde.net) |
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| 08:39.26 | bala | hi, i am trying to debug asterisk, and am unable to see the symbols belonging to a channel from the addons package. any hints on what i could be doing wrong? |
| 08:39.54 | bala | i can see the symbols with nm |
| 08:45.44 | dlynes | DND: doesn't matter...you're not asking a gui question, so you're safe |
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| 08:53.02 | Get_The_Fish | does anyone know if there are plans to designate a 1.6.x version as LTS? |
| 08:58.32 | kaldemar | Get_The_Fish: see the table in this post: http://blogs.asterisk.org/2009/11/10/asterisk-project-update-astricon-2009/ |
| 08:59.00 | tzafrir__laptop | bala, what package exactly? What distribution? What repository? |
| 09:00.14 | bala | tzafrir__laptop: ubuntu sources. i just saw that only 2 symbols are defined as 't' from chan_mobile, rest are 'U' |
| 09:00.46 | Get_The_Fish | oh hey, I didnt see that, thanks Kaldemar, much appreciated... |
| 09:00.55 | tzafrir__laptop | bala, what do you mean by "ubuntu sources"? Rebuilt the ubuntu package? installed from source on Ubuntu? Using Ubuntu package? |
| 09:01.03 | Get_The_Fish | (that pretty much sums it all up doesnt it :)) |
| 09:01.14 | bala | tzafrir__laptop: apt-source asterisk-addons, from ubuntu, karmic |
| 09:01.29 | Get_The_Fish | I like the "new" versioning schema, I think that it makes more sense. I really like the LTS concept though. |
| 09:01.41 | Get_The_Fish | Do you think that we will see ABE on 1.8? |
| 09:01.46 | bala | tzafrir__laptop: how can i force the inclusion of all symbols? |
| 09:02.15 | tzafrir__laptop | bala, it should create a separate package , asterisk-dbg , IIRC |
| 09:02.44 | bala | tzafrir__laptop: oh, asterisk itself is fine, i can see the symbols from there, |
| 09:03.12 | bala | tzafrir__laptop: i can't see the symbols in the chan_mobile.so itself, |
| 09:03.56 | kaldemar | Get_The_Fish: i don't know about the plans on ABE, but it would sound logical to have it use a LTS release |
| 09:04.19 | Get_The_Fish | I would think so too, yeah. It appears that is what is done now, as it is only 1.4 |
| 09:04.48 | *** join/#asterisk knobo (n=user@90.149.4.182) |
| 09:05.10 | Get_The_Fish | I'm excited about 1.8 actually... unlike some pundits, I have been impressed with the changes from 1.4 to 1.6, so 1.8 is really exciting |
| 09:05.15 | kaldemar | i wouldn't be too surprised if the whole idea of LTS came from ABE |
| 09:05.35 | Get_The_Fish | well, it makes sense. There is kind of an evolution there. |
| 09:06.05 | Get_The_Fish | You go from hobbyist to testing with an eye to production, and you are good with the latest greatest and no support... |
| 09:06.35 | Get_The_Fish | when you have it in production for a while, you start to want that support cushion and the QA vetting that goes with it. |
| 09:07.02 | knobo | what is the reason for not installing all header files necessary for compiling asterisk, and then compiling asterisk on the production server? |
| 09:07.02 | kaldemar | even 1.6.2 is exciting, it has some nice new apps :) |
| 09:07.03 | Get_The_Fish | with ABE, that is |
| 09:07.28 | Get_The_Fish | I have seen that, REALTIME_FIELD being at the top of my list, solves a pretty big issue for me |
| 09:07.33 | knobo | and instead compile asterisk on a different computer, then installing the binaries on the server. |
| 09:07.42 | Get_The_Fish | at least I hope it will :) |
| 09:08.02 | Get_The_Fish | kaldemar, what apps are you looking at that are new and shiny? |
| 09:08.48 | knobo | or is there no reason for it? |
| 09:09.28 | *** join/#asterisk yosi1234 (i=Bmw_Toro@CPE001217239daa-CM000e5c6f225a.cpe.net.cable.rogers.com) |
| 09:10.18 | yosi1234 | anyone here use lylex.net for VOIP hosting? or any voip pbx hosting I need to host 10 incomming SIP trunk 1-800 toll free channels, have the in a queue and 5 remote agents working from home... i guess effectivly 15 channels.. any comments? |
| 09:11.11 | yosi1234 | i was going to buy bandwith for my home server but the most they offer is 50/2 mbs and i'm not sure if that is fast enough over cable, so i'm assuming hosted ina datacenter will eb faster |
| 09:12.05 | Get_The_Fish | yosi, did I misread or did you just say that 50/2 mbps is not enough for 15 channels? |
| 09:12.13 | Get_The_Fish | or "may not be enough" |
| 09:12.39 | yosi1234 | well my stupid cable provider claims 2 mbs per sec upstream, but i have a feeling that it is really 1.5mbs variable over cable... |
| 09:12.41 | kaldemar | Get_The_Fish: app_confbridge and app_originate |
| 09:13.06 | yosi1234 | is that enough for 15 channels? is it really the upsteam or downstream that i should care about? |
| 09:13.44 | Get_The_Fish | uh, yeah, that should be enough :) are you using compression? |
| 09:13.51 | yosi1234 | when they say g711 is 80kilobytes is that up and down combines or 80kbytes each way?so really 160? |
| 09:14.11 | yosi1234 | no compression i want g711, not too happy with g729 when i tried it |
| 09:16.10 | Get_The_Fish | yoshi, here: http://www.bandcalc.com/ and here: http://www.asteriskguru.com/tools/bandwidth_calculator.php... these will help you make an informed decision. |
| 09:16.18 | kaldemar | yosi1234: 80 kiloBITS per second, not bytes. try this: http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 09:16.24 | Get_The_Fish | and I use g729 in a call center. |
| 09:16.25 | kaldemar | oops. duplicating. |
| 09:16.27 | Get_The_Fish | lol lol |
| 09:16.38 | yosi1234 | see i don't understand this, I ran PBX in a Flash, on my server at home and it took up, 400 megs of ram vanilla install.. but when i spoke to the ppl at Lylex.net hosting they said on their hosted platform a 256 mb ram machine should be enough for 15 channels... they said cause thie virtualized setyup it takes less memory, does that make any sense? |
| 09:17.16 | *** join/#asterisk tuxx- (n=tuxx@nightshade.nl) |
| 09:17.30 | yosi1234 | i will check out those calculators... |
| 09:17.47 | Get_The_Fish | uh, not bevuase of virtualization specifically, at least not that I know of anyway |
| 09:17.56 | kaldemar | yosi1234: pbx in a flash is a whole bunch of stuff besides asterisk. you can run asterisk on a SOHO router with 16 megs of RAM. |
| 09:18.15 | yosi1234 | wow... |
| 09:18.29 | yosi1234 | i guess freepbx takes up a bunc, apache etc.. |
| 09:18.39 | kaldemar | it won't handle 15 channels, but just as an example. |
| 09:19.17 | yosi1234 | does the more simultaneous channels i connect take up more memory? 15 channels take up a lot of ram or not really more ram? |
| 09:19.42 | yosi1234 | wonder why the hosting provider, suggested 256 is enough for 15 channels using pbiaf? |
| 09:20.18 | yosi1234 | 256mb that is |
| 09:22.16 | yosi1234 | wow, just ran the bandwidth calc, and it say g711 at 15 channel is approx 960kilobits... but does that mean i need that on the upstream, downstream or both?? |
| 09:22.23 | Get_The_Fish | kaldemar, app_originate is pretty interesting indeed... is that replacing dial()? It probably could, yeah? Interesting, send a call into an application |
| 09:23.27 | kaldemar | Get_The_Fish: no, it can't replace Dial. |
| 09:24.16 | kaldemar | but no more scripts that generate call files, CLI commands or send manager commands if you want to originate calls from the dialplan. that's nice. |
| 09:24.30 | *** part/#asterisk mumtazah (n=anees@180.96.48.60.wmu01-home.tm.net.my) |
| 09:25.03 | Get_The_Fish | oh shit, yeah, I dunno what I was thinking there.... |
| 09:25.14 | Get_The_Fish | yeah, thats extremely nice indeed |
| 09:27.02 | bala | tzafrir__laptop: i tried building the addons package again, with DEB_BUILD_OPTIONS="nostrip noopt debug", but still no luck. the package is getting built with "-g" option, but the functions are not visible |
| 09:27.16 | bala | tzafrir__laptop: any idea about what i could be doing wrong? |
| 09:27.47 | tzafrir__laptop | bala, dh_strip can strip debug symbols to a separate package |
| 09:28.25 | tzafrir__laptop | I'm not sure about the Ubuntu packaging (IIRC they do this with a separate repo for debug symbols) |
| 09:28.25 | Get_The_Fish | and app_confbridge is going to be cool too, I saw that one the other day. The new apps and functions are already incorporated into the docs site... |
| 09:29.10 | bala | tzafrir__laptop: thanks, i will try that. |
| 09:29.22 | Get_The_Fish | I didnt notice the app_originate though...there is something specific that I can do with that now that you mention it |
| 09:29.22 | tzafrir__laptop | But even if they removed the package asterisk-dbg, you can re-add it, and use: dh_strip --dbg-package-asterisk-dbg |
| 09:29.27 | tzafrir__laptop | in debian/rules |
| 09:30.59 | Get_The_Fish | say, does anyone have any other documentation on the MXML manager interface other than whats on asterisk.org/docs? |
| 09:34.29 | *** join/#asterisk frogzoo (n=frogzoo@203.171.192.201) |
| 09:37.04 | *** join/#asterisk gaetronik (n=gaetan@LPuteaux-156-15-5-233.w80-14.abo.wanadoo.fr) |
| 09:37.14 | gaetronik | hi! |
| 09:37.42 | tzafrir__laptop | Long time no see! |
| 09:38.23 | *** join/#asterisk CrashSys (i=Kumba@azrael.crashsys.com) |
| 09:39.57 | CrashSys | Don't suppose anyone has any insight on how close Asterisk 1.4.27 is to general release? |
| 09:41.58 | *** part/#asterisk Nest0r (n=nestor@200.121.194.14) |
| 09:42.58 | gaetronik | i've a strange one way sound issue |
| 09:43.24 | gaetronik | sometimes when i receive a call i can't hear the caller |
| 09:43.41 | gaetronik | but i can hear the sound of the gangup |
| 09:43.48 | gaetronik | like the plastic chock sound |
| 09:44.26 | gaetronik | can i blame my *** ISP modem routing for doing garbage with nat |
| 09:47.49 | *** join/#asterisk contrabanda (n=c0ntr@79.99.248.8) |
| 09:48.09 | contrabanda | hello everyody |
| 09:50.29 | tzafrir | CrashSys, it has been tagged already |
| 09:50.52 | tzafrir | So I guess it will be released today or tommorow |
| 09:51.49 | yosi1234 | anyone here have any experience with hosted pbx's? |
| 09:52.24 | frogzoo | a hosted pbx is an offsite pbx, with only IP to site? |
| 09:53.19 | yosi1234 | well lylex.net the company i'm looking zat is claiming i don't need more than 256 mbs of RAM, as some of the memory too boot up the virtual hosted pbx, kernal etc.. is already taken care of |
| 09:53.51 | yosi1234 | does this make sense hence the virtual environment... on my pbx in a flash it takes 484mb just to load up |
| 09:55.20 | *** part/#asterisk contrabanda (n=c0ntr@79.99.248.8) |
| 09:55.50 | xrmx__ | yosi1234, i have some virtual pbxs with 256mb each. used memory +/- buffers is 57MB |
| 09:57.14 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 09:59.15 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 10:00.51 | yosi1234 | xrmx, i don't understand why is takes so much less than the distro on a virtual pbx? the base distro for pbx in a flash takes up 484mb ram on my computer, but the hosting provider claims it takes up on 70-120 megs to tun? |
| 10:01.28 | tzafrir | yosi1234, give it a shot yourself: install it under e.g. kvm/virtualbox at home |
| 10:02.15 | tzafrir | This does not emulate the bad network, but is good enough to test the memory requirements and such |
| 10:02.28 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:7811:c820:f908:cc6a) |
| 10:03.56 | yosi1234 | cool, thanks |
| 10:04.21 | dpryo | Depends on the virtual env |
| 10:04.30 | yosi1234 | well i have it running in vmware, and its taking up the 484 in it |
| 10:04.37 | dpryo | some doesn't emulate the whole x86, bios and anything |
| 10:04.45 | dpryo | like kvm, vmware and virtualbox does |
| 10:05.14 | yosi1234 | i wonder what lylex is using |
| 10:10.02 | dpryo | Virtuozzo/OpenVZ shares memory, caching and filesystems with all VMs. That means you don't need to use your memory on kernel, drivers etc. |
| 10:10.11 | dpryo | You get a jail for your distribution |
| 10:13.52 | *** join/#asterisk mechbangirc (n=mech@static-host119-73-2-167.link.net.pk) |
| 10:15.04 | mechbangirc | i am looking for a web application that shows call recordings and its corrosponding CDR in one place, any idea??? |
| 10:27.05 | *** join/#asterisk tarik (n=chatzill@41.251.250.56) |
| 10:43.39 | yosi1234 | thanks guyz... |
| 10:43.54 | *** join/#asterisk voxter (n=voxter@76.77.91.250) |
| 11:01.18 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
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| 11:03.22 | shamelessn00b | Hi all |
| 11:03.49 | shamelessn00b | I'm having sound quality issues on playback |
| 11:03.55 | shamelessn00b | can anyone help |
| 11:03.56 | shamelessn00b | ? |
| 11:06.11 | *** join/#asterisk c0rnoTa (n=c0rnoTa@188.162.32.198) |
| 11:12.10 | *** join/#asterisk Dabian (n=M0RTEN@fsf/member/dabian) |
| 11:12.23 | Dabian | Hi |
| 11:12.48 | Dabian | I come with a question. |
| 11:14.32 | *** join/#asterisk nicola_pav (n=chatzill@83.244.78.241) |
| 11:14.34 | Dabian | I am reading "Aterisk The Future of Telephony", and I've reached page 12. Here they talk about "channels". How many "channels" you can have in a hobby-system, a SOHO-system, and so on. What is "channel" in this context? |
| 11:14.53 | nicola_pav | kaldemar: there? |
| 11:17.25 | Dabian | Hmm .. on what time of the day is this channel most active? |
| 11:20.58 | *** join/#asterisk mohawk (n=ross@host217-40-110-153.in-addr.btopenworld.com) |
| 11:22.52 | UQlev | Dabian: channel is communication session |
| 11:23.26 | Dabian | UQlev: So for the hobby setup, no more than 5 conversations should be simultanious? |
| 11:23.31 | UQlev | Dabian: usually 2 channels per session |
| 11:24.12 | Dabian | (It says no more than 5 channels .. oh .. I guess that means 2-3 conversations at a time?) |
| 11:24.19 | UQlev | Dabian: no problem with multiple conversations once your badnwidth can fit it |
| 11:24.45 | Dabian | What is a "communication session" then? |
| 11:24.56 | UQlev | Dabian: phone call |
| 11:25.52 | tzafrir | ~channel |
| 11:25.53 | infobot | rumour has it, channel is This refers to the group of resellers that supply most companies with software, hardware, and support. The channel is a force to be reckoned with, and it competes directly against companies like Gateway 2000 and Dell. A channel can also be a content container, like a television channel. You may be watching TV channels on the Internet someday. |
| 11:26.17 | tzafrir | grrr... not very useful |
| 11:26.34 | UQlev | Dabian: it depends on whether you use local zap cards or do simple SIP/IAX calls |
| 11:26.57 | Dabian | UQlev: I only do SIP |
| 11:27.06 | Dabian | At least for now. |
| 11:27.14 | tzafrir | A channel is mostly a single "leg" of a call (if you think of a call as connecting - bridging - two "legs") |
| 11:27.33 | *** join/#asterisk uluatu (n=uluatu@189.4.53.18) |
| 11:27.39 | UQlev | Dabian: problem with multiple sip/iax calls only if your VoIP provider support simultations multiple termination |
| 11:27.53 | *** join/#asterisk RussellAlan (n=RussellA@cpe-76-185-155-8.tx.res.rr.com) |
| 11:28.28 | tzafrir | Dabian, "number of channels" is too general a question to ask, |
| 11:28.47 | RussellAlan | Any pbxinaflash'ers? |
| 11:28.51 | Dabian | tzafrir: Well, I didn't have a clue what they were refering to. :) |
| 11:28.52 | nicola_pav | hello. i use an IVR script to check pin and take destination number. after that i ForkCDR(a) and after that i use Dial() |
| 11:29.14 | nicola_pav | when i started to use ForkCDR(a), the billing does not deduct anything from PIN account |
| 11:29.16 | nicola_pav | y? |
| 11:29.16 | tzafrir | for instance, a call that requires transcoding to some compressed codec requires many more CPU cycles that a simple G.711 call |
| 11:29.48 | Dabian | Basicly, I am setting Asterisk up for my personal (private) use, aka. hobby - for now. I plan later on doing some kinda provider setup, so I can at least service my friends. |
| 11:30.26 | UQlev | Dabian: you can easily prvide SIP service to 20-30 of your friends |
| 11:30.35 | Dabian | Maybe even more than that .. so that is why I am reading Chapter 2 with great interest. |
| 11:30.48 | Dabian | UQlev: Cool. |
| 11:31.24 | UQlev | Dabian: for your friends iax will be more convenient to use |
| 11:31.42 | UQlev | Dabian: and iax clients respectively |
| 11:32.10 | Dabian | UQlev: Because? |
| 11:32.33 | UQlev | because of nasty nutted firewalls |
| 11:32.45 | Dabian | Ahhh |
| 11:33.15 | Dabian | I have a friend, who I gave a (somewhat) cheap IP-phone (vlines) |
| 11:34.05 | UQlev | Dabian: I believe harware ip phones support sip only |
| 11:34.15 | Dabian | Quite often, I cannot call him .. the line will just go silent, not even the repeating "doood"-sound when you call him. I will then make him restaart it. |
| 11:34.27 | Dabian | And it will work. |
| 11:34.37 | Dabian | I think maybe this phone supports IAX. |
| 11:34.47 | UQlev | very doubtfully |
| 11:35.04 | Dabian | I guess when my asterisk works, it would be worth a try to connect him with IAX, if it indeed does support IAX? |
| 11:35.29 | Dabian | (His ISP had forced a lousy ZyXEL-router upon him) |
| 11:35.36 | UQlev | Dabian: if possible run your asteris on a public interface |
| 11:35.46 | Dabian | I will |
| 11:36.03 | Dabian | A virtual server with public IP. |
| 11:36.06 | UQlev | not behind soho routers |
| 11:36.52 | Dabian | I should hope not. Its a hosting location. I dont own the hardware, I just lease it. |
| 11:37.08 | UQlev | i see |
| 11:37.21 | Dabian | Well, not the hardware, but the "virtual machine" |
| 11:37.33 | *** join/#asterisk shamelessn00b (n=chatzill@58-65-172-114.nayatel.pk) |
| 11:39.52 | Dabian | I am concerned about the "real time" thing though. When I start up Asterisk, it sometimes complains about it .. (right now I cannot get the error) |
| 11:40.22 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 11:40.38 | Dabian | UQlev: IAX is good for softphones? |
| 11:40.46 | UQlev | Dabian: what OS and what asterisk version? |
| 11:41.12 | UQlev | Dabian: not all softphones support iax |
| 11:41.28 | Dabian | The OS is an older Ubuntu. |
| 11:41.28 | UQlev | Dabian: I like zoiper for it |
| 11:41.58 | nicola_pav | hello. i have a problem with ForkCDR() |
| 11:42.03 | Dabian | asterisk version 1.2.7.1 |
| 11:42.27 | nicola_pav | i use option "a". when i use it, billing stops to bill |
| 11:43.05 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
| 11:43.55 | shamelessn00b | using asterisk 1.4 on debian 5 |
| 11:44.05 | shamelessn00b | is there a tool to analyze sound quality |
| 11:44.06 | *** join/#asterisk arossouw (n=arossouw@dsl-146-48-160.telkomadsl.co.za) |
| 11:44.12 | UQlev | Dabian: not very up-to-date |
| 11:44.13 | shamelessn00b | or to improve it |
| 11:44.36 | Dabian | True |
| 11:44.39 | arossouw | can anyone give me advice on call pickup see http://slexy.org/view/s20tt417E9, thats what im getting from asterisk |
| 11:44.49 | kaldemar | nicola_pav: how does your billing know which entry to look for in the CDR? |
| 11:44.58 | Dabian | UQlev: Its just the version stuck with the Ubuntu-release. |
| 11:45.38 | UQlev | Dabian: it will be enough to try it |
| 11:45.46 | Dabian | UQlev: Cool. |
| 11:46.02 | Dabian | I am very excited about Asterisk. |
| 11:46.06 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
| 11:46.26 | *** join/#asterisk Stese (n=Someone@adsl.ntsols.com) |
| 11:46.36 | Stese | Morning alll |
| 11:46.39 | Stese | -l |
| 11:46.53 | UQlev | Dabian: for most users big problem is to make microphone working properly |
| 11:47.27 | Dabian | From soundcard? |
| 11:47.29 | nicola_pav | kaldemar: it looks for the master.csv |
| 11:47.35 | UQlev | Dabian: yes |
| 11:47.57 | nicola_pav | when i used ForkCDR(a), for each call there are two entries |
| 11:48.01 | kaldemar | nicola_pav: re-read my question. what does it look for in the file? |
| 11:48.05 | UQlev | Dabian: that is why sip is not so popular yet as skype |
| 11:48.28 | Dabian | I guess I will have no console Microphone, since asterisk is running remotely. |
| 11:48.31 | kaldemar | nicola_pav: how does it identify a line for billing? |
| 11:48.53 | Stese | Running Asterisk 1.6.1.1 and am getting loads of "> Doing dnsmgr_lookup for 'sip.provider.com'" messages in CLI. is the reason for this going to be shown in a sip debug or do I need to be looking elsewhere? |
| 11:49.02 | UQlev | Dabian: I mean microphones on client softphones |
| 11:49.05 | Dabian | UQlev: OK. I always wondered why people were so excited about Skype, when you got SIP. :) |
| 11:49.27 | *** join/#asterisk lordmortis (n=lordmort@203-206-121-244.dyn.iinet.net.au) |
| 11:49.33 | Dabian | OK |
| 11:50.03 | shamelessn00b | anyone halp this poor soul :( |
| 11:50.40 | nicola_pav | kaldemar: i am looking into it now |
| 11:51.10 | UQlev | shamelessn00b: sound quality mostly depends on client |
| 11:51.16 | Dabian | My setup for now, doesn't need to involve a phone. I just want to connect asterisk to a SIP-provider, and then .. (lack english knowledge) "move" the calls to another phone number. |
| 11:51.53 | UQlev | Dabian: it is called "termination service" or sip trunk |
| 11:51.53 | Stese | Dabian > You want it to redirect all calls to a particular number? |
| 11:52.14 | Dabian | Stese: Excactly! |
| 11:52.35 | Dabian | Laster I want Asterisk to handle both numbers .. but for now, that would be great. |
| 11:53.31 | Stese | Dabian > Where is that number you want to direct calls to? |
| 11:53.51 | Dabian | Stese: Its another SIP-provider. |
| 11:54.55 | Stese | So you have Number on Sip-Provider 1 -> Asterisk ->NUmber on Sip Provider 2? |
| 11:55.00 | UQlev | Dabian: for this situation it seems you don't need your asterisk server |
| 11:55.15 | Dabian | stese: Yes |
| 11:55.56 | Dabian | UQlev: My SIP-hardware can only handle one provider on each line. Its a SPA2K. |
| 11:56.22 | Stese | Dabian > I agree with UQlev. You shuold be able to do that with your provider |
| 11:57.18 | Dabian | Stese: They want money to redirect the number. I suspect they will also charge me for each call going to the new number. |
| 11:57.36 | Dabian | So I thought it was a way to get started with Asterisk. |
| 11:57.54 | Dabian | s/a way/a good excuse/ |
| 11:58.07 | Dabian | Exactly :) |
| 11:58.09 | UQlev | Dabian: then your friends should have accounts on your * server as well |
| 11:58.42 | Dabian | Indeed. If they want to. |
| 11:59.44 | Stese | Dabian > Fair enough. You would need to have * 'register' with your providers, and then a very small dialplan redirect between them |
| 11:59.47 | Stese | the book would be the best way to learn that |
| 11:59.50 | Stese | ~book |
| 11:59.51 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 11:59.58 | Dabian | The thing is .. I had a great provider .. but they were bought by another company with no visions. Now they're rising the price, so we figure they're squeezing the orange before they close it down. |
| 12:00.11 | Dabian | Stese: Cool! I am reading that one. I bought it from Amazone. |
| 12:00.23 | Stese | He he... you'll have all you need in there :) |
| 12:00.40 | Stese | Anyone help me with this one? |
| 12:00.44 | Stese | Running Asterisk 1.6.1.1 and am getting loads of "> Doing dnsmgr_lookup for 'sip.provider.com'" messages in CLI. is the reason for this going to be shown in a sip debug or do I need to be looking elsewhere? |
| 12:03.28 | *** join/#asterisk coppice (n=chatzill@210.17.168.171) |
| 12:04.28 | *** join/#asterisk grharry (n=root@ppp-94-65-202-227.home.otenet.gr) |
| 12:05.56 | grharry | hi, what is the tag name <from-port - to port> in order to comply with cisco start_media_port: , end_media_port: in sip.conf ?? |
| 12:08.22 | kaldemar | arossouw: has some channel set PICKUPMARK when you try to pick it up? |
| 12:09.02 | kaldemar | arossouw: core show application Pickup gives a rather nice explanation on how it works. |
| 12:13.27 | *** part/#asterisk grharry (n=root@ppp-94-65-202-227.home.otenet.gr) |
| 12:19.38 | garymc | Anyone know how i can stop all my polycoms displaying missed calls even though another extension in the ring group answered it, so it wasnt actually a missed call? |
| 12:23.53 | Stese | garymc > in systems ai've seen the call is sent to each phone specifically, so the phone records as a missed call since it wasn't answered on that handset |
| 12:24.57 | garymc | Stese : So is there no way to change this? |
| 12:25.17 | garymc | I suppose it doesnt matter really but it is starting to annoy me |
| 12:26.18 | Stese | unless you can do it on the polycom, no |
| 12:26.28 | kaldemar | the missed call indications can be disabled entirely, but that may not be what you want. |
| 12:26.56 | garymc | no, i just want it to not display if the call is answered by another extension |
| 12:26.57 | kaldemar | ask polycom if they've implemented some way to disable it for a single call. |
| 12:27.23 | dpryo | Anyone using text-to-speech from Acapela Group? |
| 12:28.11 | garymc | See when ive been out of the office and someone else has been at their desk i caom back and there is 15 missed calls on my phone but they where all answered. so if the guy whos answered the calls has gone out when i come in i dont know if they where answerd without looking at his phone to see if he has any missed calls |
| 12:29.03 | Stese | what happens to calls that are actually missed by all? |
| 12:29.36 | garymc | they get directed to an extension voicemail. But not everyone leaves voicemail |
| 12:29.53 | Stese | true... |
| 12:30.44 | Stese | but if they don't leave voicemail, they can't honestly expect a call back... |
| 12:31.00 | garymc | nope but its more out of curiosty at times |
| 12:31.19 | kaldemar | i once set an installation to send an SMS if a call wasn't answered at all. |
| 12:31.43 | kaldemar | you could do something in the next priority to know which calls were not answered by anyone |
| 12:32.10 | *** join/#asterisk voipdeveloper (n=chatzill@122.170.34.164) |
| 12:32.16 | voipdeveloper | hi |
| 12:33.46 | Stese | if you are a programmer, i'm sure you can have something written to look at the call logs or some such to see if the call has been answered |
| 12:35.18 | voipdeveloper | I am facing crash in asterisk |
| 12:35.35 | voipdeveloper | anybody there...who help me out? |
| 12:36.02 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 12:36.20 | Stese | voipdeveloper > You'll need to give more information for assistance |
| 12:36.44 | kaldemar | ~ask |
| 12:36.45 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:38.12 | voipdeveloper | I will keep it in mind |
| 12:38.17 | voipdeveloper | new to use irc |
| 12:38.20 | voipdeveloper | sorry for that |
| 12:39.13 | *** join/#asterisk levity (n=levity@unaffiliated/canuck) |
| 12:40.44 | elliot98 | what does pernicious mean? |
| 12:40.54 | Dabian | Cantide: What kinda name would that be? DollarSucker? |
| 12:41.08 | Dabian | <wrong window> |
| 12:41.28 | kaldemar | elliot98: http://lmgtfy.com/?q=define%3Apernicious |
| 12:42.13 | elliot98 | what is lmgtfy? |
| 12:44.35 | coppice | let me google that for you |
| 12:45.02 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
| 12:45.18 | Stese | Basically it means, next time, ask that type of question on google |
| 12:45.37 | kaldemar | if i see someone asking what google is, it's cluebat time |
| 12:46.42 | coppice | http://www.fuckinggoogleit.com/?q=define%3Apernicious works, too |
| 12:46.54 | elliot98 | http://lmgtfy.com/?q=define%3cluebat |
| 12:47.14 | Stese | don't think he's got the idea |
| 12:47.45 | *** join/#asterisk Buklov (n=buklov@80.68.15.79) |
| 12:48.18 | shamelessn00b | If anyone could help me on sound quality issues in asterisk |
| 12:48.48 | *** join/#asterisk uluatu (n=uluatu@189.4.53.18) |
| 12:50.59 | kaldemar | ~cluebat elliot98 |
| 12:50.59 | infobot | ACTION pulls out a ClueBat (tm) and thwaps elliot98. |
| 12:52.04 | elliot98 | all shaken up, he begins to see g's and o' floating around his head |
| 12:55.50 | voipdeveloper | I am using asterisk 1.4.24.1 , 8Gb of RAM, centos 5.3 64 bit os, it's in production |
| 12:56.06 | elliot98 | go on |
| 12:56.06 | voipdeveloper | asterisk crashes after each 12 hours approx |
| 12:56.21 | elliot98 | using any queues? AGIs? |
| 12:56.37 | Stese | voipdeveloper > Any error messages? prior faults? |
| 12:56.42 | elliot98 | Dahdi? |
| 12:57.21 | voipdeveloper | Stece > Yes.... |
| 12:57.29 | voipdeveloper | WARNING[8730] pbx.c: Failed to create new channel thread |
| 12:57.31 | voipdeveloper | [Nov 18 06:47:23] WARNING[8730] chan_sip.c: Failed to start PBX :( |
| 12:58.34 | elliot98 | does "show channels" show any zombie calls? |
| 12:58.53 | Stese | ok, pastebin your configs and link them |
| 12:58.57 | Stese | ~pastebin |
| 12:58.58 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 12:59.24 | elliot98 | how come infobot does show pastie.org? |
| 13:00.21 | *** join/#asterisk areay (n=areay@188.220.19.179) |
| 13:00.56 | voipdeveloper | asterisk gets crash.... |
| 13:01.20 | voipdeveloper | it generates corefile |
| 13:01.32 | Stese | Paste that to Pastebin too then |
| 13:01.39 | voipdeveloper | but GDB does not show any debug symbols |
| 13:02.09 | areay | hi everyone... would there be any reason why asterisk thinks a call that's no longer connected should still be in a queue? i hung up like 5mins ago and my extensions are still ringing :/ |
| 13:02.09 | kaldemar | voipdeveloper: see doc/backtrace.txt in the source package |
| 13:02.16 | Stese | voipdeveloper > we need INFOMATION, before we can help you |
| 13:03.09 | Stese | So, error codes from CLI and your main config files from the device |
| 13:03.38 | voipdeveloper | we are doing dial only |
| 13:04.14 | *** join/#asterisk bbt (n=sam@180.189.139.92) |
| 13:04.39 | Stese | voipdeveloper > Seriously, if you want help, paste all the info you can find from your system into paste bin.... we can't guess your problem |
| 13:05.41 | voipdeveloper | ok fine |
| 13:12.55 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:12.55 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:15.47 | *** part/#asterisk manxpower (n=ewieling@107.sub-70-214-135.myvzw.com) |
| 13:18.14 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 13:19.12 | kisom | just a quick question, im new to all this voip and asterisk stuff, but.. having a server on a public IP address, is it possible to have it relay the speech for clients that are NATed on networks i have no control over? I've got asterist running fine on a home server, but i cannot place calls when outside my house because most public networks are using NAT... |
| 13:20.18 | kisom | i only want to call other people on the server, we've all got cellphones which supports SIP |
| 13:20.58 | kaldemar | ~sipnat |
| 13:20.59 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:21.07 | kaldemar | kisom: no problem there |
| 13:21.37 | *** join/#asterisk Jacke (i=jacke@valhalla.bofh.org) |
| 13:21.59 | Jacke | argh! anyone proficient with dahdi? |
| 13:22.53 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:22.53 | Zhad | depends on what you cann proficient :-p |
| 13:23.01 | Zhad | s/cann/call/; |
| 13:23.26 | *** join/#asterisk chendy (n=chatzill@116.30.194.119) |
| 13:23.31 | Chainsaw | Jacke: Please skip the introduction and move on the problem at hand. What hardware are you trying to drive with it? |
| 13:23.33 | Jacke | Zhad: how do i disable the EC on a single channel ? |
| 13:24.02 | Jacke | Chainsaw: TE412p, but that's irrelevant - my problem is generic to any hw-ec enabled card by digium ;-) |
| 13:24.34 | Jacke | Chainsaw: I want to disable ec on a given channel, leaving it enabled on all others. Is that even doable? |
| 13:25.14 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:26.04 | *** join/#asterisk lordmortis (n=lordmort@203-206-121-244.dyn.iinet.net.au) |
| 13:27.30 | kaldemar | Jacke: there's always the echocancel parameter... |
| 13:28.36 | Jacke | kaldemar: no use for me, i need it to be disabled permanently before i even run asterisk. |
| 13:30.15 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 13:30.54 | *** join/#asterisk ManxPower-work (n=EWieling@107.sub-70-214-135.myvzw.com) |
| 13:31.10 | ManxPower-work | ~answers |
| 13:31.11 | infobot | from memory, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 13:33.01 | *** join/#asterisk styelz (n=yoohoo@2001:5c0:1500:800:0:0:0:1) |
| 13:36.28 | dlynes | I've got an issue with T.38 refusing to work; the errors and the SIP invites from both the end point and the upstream (but not the asterisk box in the middle) are as follows: http://pastebin.ca/1677070 |
| 13:36.40 | dlynes | I'm just wondering if anyone might be able to shed some light on how I can solve the problem with the error....the error ultimately causes the communication to end in failure |
| 13:36.52 | dlynes | I've tried using the parameters to t38udptl=yes mentioned in M15586, to no avail |
| 13:37.06 | dlynes | I'm using Asterisk 1.6.1.8 |
| 13:38.47 | areay | why are my extensions still ringing for an incoming call that has hung up? |
| 13:40.01 | *** join/#asterisk oej (n=olle@gate.ipvision.dk) |
| 13:41.48 | ManxPower-work | areay: Are you using analog? |
| 13:44.13 | *** join/#asterisk wry (n=wry@catv-89-133-123-4.catv.broadband.hu) |
| 13:44.47 | wry | hey |
| 13:45.19 | wry | anyone knows what 'zt_pri_error: Warning: unknown/inappropriate protocol discriminator received (xx/xx)' is meant to mean? |
| 13:46.32 | Chainsaw | wry: The ISDN version of "Help I can't understand what they're saying!" |
| 13:46.45 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 13:48.56 | Tim_Toady | any suggestions for cards with bri interface? |
| 13:49.01 | Faustov | Chainsaw: what do you think about getting wanpipe to the main tree? |
| 13:49.23 | Chainsaw | Faustov: Which main tree? |
| 13:49.59 | Faustov | Chainsaw: well right now there is an obsolete ebuild in the voip overlay, would be nice to have it in portage |
| 13:50.01 | wry | Chainsaw: thanks. |
| 13:50.31 | Pan3D | morning |
| 13:50.43 | Chainsaw | Faustov: You'd have to sponsor me an appropriate piece of hardware so I can test it. |
| 13:51.02 | Chainsaw | Faustov: I have (very) bad experiences with ebuilds for hardware that I don't actually have. |
| 13:51.06 | Faustov | Chainsaw: that's 500 bucks unfortunately :( |
| 13:51.41 | Faustov | I barely got the funds from the CFO for one :> |
| 13:51.41 | Chainsaw | Faustov: Feel free to talk to Sangoma. If they're open to it, I will use it and make sure you get your ebuild. |
| 13:52.04 | Faustov | Chainsaw: you mean, ask them for a free sample? |
| 13:52.35 | Chainsaw | Faustov: Or whatever other strategy you'd wish to apply. I have a working phone system already. |
| 13:52.51 | Faustov | I'll see what I can do |
| 13:53.06 | Chainsaw | Faustov: Triple-BRI would fit my situation best. |
| 13:53.09 | flyback | suspects Chainsaw has "canuck tendencies" |
| 13:53.36 | Chainsaw | You're going to have to explain that one. |
| 13:53.37 | Faustov | might be hard if yo ask me, because last time I worked with their support, they weren't too happy to face a number of gentoo-only bugs |
| 13:54.10 | Chainsaw | looks at flyback |
| 13:54.29 | *** join/#asterisk Corydon76-dig (i=indigo@c-69-137-80-31.hsd1.tn.comcast.net) |
| 13:54.29 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 13:54.35 | *** join/#asterisk superbeef (n=superbee@74.84.194.4) |
| 13:54.37 | [TK]D-Fender | flyback: This schtick of yours is getting old, fast... |
| 13:54.39 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 13:54.42 | flyback | ok ok |
| 13:54.49 | flyback | but I been doing this 12+ yrs |
| 13:54.53 | flyback | IT NEVER GETS OLD |
| 13:54.54 | flyback | muhahaha |
| 13:54.56 | flyback | :P |
| 13:55.01 | flyback | i'll keep it out of the channel though |
| 13:55.13 | [TK]D-Fender | flyback: good idea |
| 13:55.40 | flyback | bah |
| 13:55.54 | Chainsaw | Faustov: I can help them out. It's a two-way thing. |
| 13:57.00 | Faustov | Chainsaw: I'll write them an email and see what they say |
| 13:57.12 | *** join/#asterisk Ad-Hoc (n=nimbus@62.169.216.185) |
| 13:57.47 | Chainsaw | Faustov: Alright. You can reach me on tony@linx.net |
| 13:59.24 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 13:59.56 | shamelessn00b | is there a way to benchmark audio quality |
| 14:00.05 | shamelessn00b | asterisk is generating |
| 14:00.08 | shamelessn00b | ?? |
| 14:00.10 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 14:00.45 | Chainsaw | shamelessn00b: You could use a handset that allows recording. |
| 14:00.48 | Chainsaw | shamelessn00b: And then plot it. |
| 14:00.55 | ManxPower-work | shamelessn00b: 1.6 (maybe 1.4) keeps track of audio quality. See the channelvariables.tex in the doc/ directory of Asterisk |
| 14:01.22 | ManxPower-work | See the QOS info in that directory. It is a newer feature and not well known |
| 14:01.35 | shamelessn00b | were using 1.4 |
| 14:01.41 | shamelessn00b | (deployed) |
| 14:06.00 | shamelessn00b | ManxPower-work: is there some documentation I could read regarding the file you mentioned? |
| 14:06.18 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:2857:564b:b2c9:8871) |
| 14:06.23 | cusco | hi |
| 14:06.24 | ManxPower-work | shamelessn00b: all official documentation is in /path/to/src/asterisk/doc |
| 14:06.37 | ManxPower-work | You can check voip-info.org but that is not official documentation |
| 14:06.47 | levity | sorry for sounding lazy just need the proper google search term, i would like my home SIP phone and cellphone to ring at the same time? What would be official Asterisk way of achieving this? |
| 14:06.48 | ManxPower-work | It's a Wiki so it's frequently out of date |
| 14:07.08 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-204-20-90.acanac.net) |
| 14:07.18 | ManxPower-work | levity: Dial(SIP/phone1&SIP/cellnumber@provider) |
| 14:07.27 | cusco | We have an asterisk getting calls from pstn, and dialing them on to another asterisk. that 2nd asterisk queues calls and passes them on to extensions. |
| 14:07.41 | ManxPower-work | levity: read the Asterisk book or heck, even read the extensions.conf.sample that comes with Asterisk |
| 14:07.43 | levity | wow that simple thanks ManxPower-work |
| 14:08.02 | ManxPower-work | levity: I am assuming there are no analog lines involved |
| 14:08.30 | levity | ok I will, no analog |
| 14:09.05 | shamelessn00b | ManxPower-work: what should I be seaching for channelvariables.tex? |
| 14:09.21 | ManxPower-work | shamelessn00b: That's just one of several files you should look at. |
| 14:09.31 | ManxPower-work | The names should be pretty obvious |
| 14:10.50 | *** join/#asterisk cusco_ (n=trilili@142.119.54.77.rev.vodafone.pt) |
| 14:10.52 | cusco_ | sorry |
| 14:10.56 | cusco_ | sorry, by sip packets I mean audio only, udp traffic |
| 14:11.12 | cusco_ | (I haven't read anything else, connection fell) |
| 14:11.20 | ManxPower-work | SIP does not carrt audio |
| 14:11.28 | ManxPower-work | RTP is the audio protocol |
| 14:11.33 | cusco_ | yes that is right |
| 14:11.35 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:11.52 | cusco_ | I meant for RTP to travel directly between extension and 1st asterisk |
| 14:11.58 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 14:12.05 | cusco_ | how could I accomplish that? |
| 14:12.33 | ManxPower-work | cusco_: If there is NAT involved don't expect to be able to do that |
| 14:12.53 | cusco_ | there is no nat. we are all on the same subnet |
| 14:13.06 | *** join/#asterisk abatista (n=chatzill@63.214.236.169) |
| 14:13.43 | kaldemar | use sip between the asterisk boxes and allow re-invites. canreinvite or directmedia parameters depending on your version |
| 14:14.06 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:14.07 | cusco_ | 1.6 |
| 14:14.13 | cusco_ | ok let me read about it |
| 14:14.18 | ManxPower-work | then as long as there is no codec conversion (transcoding), no NAT, no Dial options that would prevent reinvites (like T/t/W/w), and as long as you do not have canreinvite=no then RTP should flow directly between the two endpoints |
| 14:15.02 | kaldemar | 1.6 is not a version, it's a branch. there are differences between 1.6 versions. |
| 14:15.43 | cusco_ | 1.6.1.1 |
| 14:15.49 | cusco_ | ok thanks |
| 14:16.11 | kaldemar | then it's canreinvite. directmedia comes with 1.6.2 i see.. |
| 14:16.31 | ManxPower-work | always read the UPGRADE*.txt |
| 14:17.13 | Faustov | Chainsaw: mail sent, I've requested for any hardware they could give for non profit testing and non commercial use |
| 14:17.25 | Katty | yawns |
| 14:17.26 | *** join/#asterisk feature (n=wea@82.103.141.216) |
| 14:17.34 | elliot98 | when I issue a "show queues" what does does Invalid next to the agent mean? |
| 14:17.40 | ManxPower-work | I doubt Sangoma wants to give away their products to poor people. |
| 14:17.51 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 14:18.17 | [TK]D-Fender | elliot98: It means "this isn't possibly contactable, I give up" |
| 14:18.47 | feature | hello. anyone had success story with asterisk 1.6 & res_config_ldap? |
| 14:19.07 | Chainsaw | Katty: Morning :) |
| 14:19.15 | Katty | hi Chainsaw |
| 14:19.29 | Chainsaw | ManxPower-work: I doubt my employer wants to sponsor Sangoma drivers if we have no use for them in our phone system. |
| 14:19.55 | ManxPower-work | "sponsor sangoma drivers"? |
| 14:19.57 | Chainsaw | So, just another approach to the problem. I can't package drivers that I don't have the hardware for. |
| 14:20.08 | Faustov | well |
| 14:20.10 | Faustov | worth asking |
| 14:20.18 | Faustov | if it fails, too bad, no wanpipe ebuild |
| 14:21.13 | shamelessn00b | ManxPower-work: I cant locate the file |
| 14:21.14 | shamelessn00b | :( |
| 14:23.05 | Faustov | ManxPower-work: btw, poor people? Just because we don't want to spend $$$ on every possible hardware bit to bring support for it? |
| 14:24.09 | voipmonk | wakes up |
| 14:24.16 | voipmonk | YaAaawWWWWWWWWn |
| 14:24.18 | Katty | hugs voipmonk |
| 14:24.36 | voipmonk | does a triple take.. uhh hi |
| 14:24.41 | voipmonk | whats cookin' ? |
| 14:24.47 | voipmonk | :) |
| 14:24.52 | Katty | it's not even close to lunch time |
| 14:25.25 | Chainsaw | Faustov: You're expecting reason out of ManxPower. That's a bit much :) |
| 14:25.47 | Faustov | :) |
| 14:27.16 | Faustov | Chainsaw: btw, support for wanpipe could be bit hard, the whole software set depends quite a lot on having access to dahdi-linux sources that are currently installed and depends even more on an interactive ./Setup script |
| 14:27.29 | Faustov | might be a road through pain to get it working |
| 14:27.39 | Chainsaw | Faustov: Yes, it would have to be a USE-flag on the dahdi ebuild. |
| 14:27.49 | Chainsaw | Faustov: You can't really have it as a separate ebuild, it's just asking for trouble. |
| 14:28.01 | Faustov | makes sense |
| 14:28.04 | *** join/#asterisk dmz (n=dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 14:28.33 | Faustov | what about the interactive setup? Should it remain as it is in the ebuild? |
| 14:28.52 | Chainsaw | Faustov: Ebuilds can't be interactive, so that chattiness will have to be patched out. |
| 14:29.03 | ManxPower-work | Oddly, I never had to have all the different cards installed in order to build zaptel or dandi. |
| 14:29.26 | Chainsaw | ManxPower-work: You're rebranding again? Dandi? |
| 14:29.38 | Faustov | ManxPower-work: dahdi and zaptel are supported, we're just talking about installing the wanpipe drivers for Sangoma cards and the appropriate firmware |
| 14:29.43 | ManxPower-work | build the drivers, package them, done. Don't even need Digium hardware installed. I was building zaptel packages for internal use |
| 14:30.32 | ManxPower-work | Faustov: I thought you were talking about creating an "ebuild". |
| 14:31.12 | Faustov | ManxPower-work: that was my idea but i was wrong. |
| 14:31.31 | Faustov | USE-flag for an existing dahdi ebuild is better |
| 14:31.31 | Chainsaw | ManxPower-work: That is "correct". |
| 14:32.08 | Chainsaw | Faustov: But yes, that's still packaging a driver/firmware package. Where I come from, we test things before we ship them. |
| 14:32.12 | ManxPower-work | So why would Sangoma require the hardware to be installed when Zaptel/Dahdi doesn't? |
| 14:32.25 | Chainsaw | If you don't believe in that, that's fine. You can't all work from the same principles. |
| 14:32.41 | Faustov | heh |
| 14:33.55 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 14:36.57 | dlynes | ManxPower-work: they don't |
| 14:38.34 | leifmadsen | Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10 are now available! For more information please see the release announcement: http://www.asterisk.org/node/49870 |
| 14:38.57 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.10 (2009/11/18), 1.6.0.18 (2009/11/18), 1.4.27 (2009/11/18), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #switchvox #asterisk-bugs |
| 14:39.25 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 14:39.31 | Katty | hi leif |
| 14:39.45 | leifmadsen | Katty: yo! |
| 14:40.14 | Katty | tell me you don't actually say yo |
| 14:40.34 | leifmadsen | oh. I do. |
| 14:40.48 | dlynes | leifmadsen: Great work! |
| 14:40.59 | leifmadsen | dlynes: thanks! I did it all myself :) |
| 14:41.05 | dlynes | leifmadsen: so what exactly does the new 'a' parameter to SendFax() do? |
| 14:41.08 | leifmadsen | </utter_lie> |
| 14:41.16 | Katty | perks up |
| 14:41.17 | leifmadsen | I have absolutely not idea |
| 14:41.25 | dlynes | leifmadsen: ah, ok |
| 14:41.27 | ManxPower-work | waves to leifmadsen |
| 14:41.28 | Katty | :< |
| 14:41.40 | Katty | maybe it has something to do with the audio buffer |
| 14:41.49 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 14:41.52 | leifmadsen | waves back |
| 14:41.53 | Katty | hi Naikrovek |
| 14:42.14 | Naikrovek | howdy |
| 14:43.02 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:44.22 | dlynes | Katty: that's just a wag, right? |
| 14:44.34 | Katty | define wag. |
| 14:44.42 | dlynes | wild ass guess |
| 14:44.49 | ManxPower-work | wag = "you read the docs"? |
| 14:46.05 | Katty | ^_- |
| 14:46.12 | Katty | that is my wishful thinking |
| 14:47.51 | *** join/#asterisk jtodd (i=xm9k84ou@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 14:47.51 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 14:47.59 | saxa | is it better to stick with 1.4.x version or the last one 1.6.1.x ? |
| 14:48.19 | Faustov | I've migrated recently to 1.6 |
| 14:48.21 | saxa | i'm running now 1.4.26.1 |
| 14:48.31 | Faustov | some changes to configs are required |
| 14:48.36 | Chainsaw | is on 1.6.1.9 (and made the jump from 1.2, skipping 1.4 entirely) |
| 14:48.37 | Faustov | but other than that no problems |
| 14:48.41 | leifmadsen | saxa: that isn't really a properly phrased question as it depends on your requirements |
| 14:48.44 | saxa | i'm running now 1.4.24.1 |
| 14:49.08 | leifmadsen | there is no point in going to 1.6.x unless you need a) new features b) additional capacity to handle more channels |
| 14:49.23 | saxa | leifmadsen: ok, i have troubles now to connect 2 boxes with iax |
| 14:49.26 | leifmadsen | c) installing a new server anyways and not using an existing version |
| 14:49.29 | ManxPower-work | 1.6 is totally awesome. |
| 14:49.34 | dlynes | saxa: I find 1.6.1 is considerably more stable than 1.4 |
| 14:49.37 | saxa | and still cant get out of it, why it wont work anymore |
| 14:49.54 | ManxPower-work | saxa: try SIP between the two boxes |
| 14:50.02 | dlynes | saxa: however, that being said, if you're running a 1.4.25 or earlier box, good luck connecting to a 1.6 box on iax2 |
| 14:50.07 | saxa | ManxPower-work: yup, will try that too |
| 14:50.26 | saxa | dlynes: i have on both boxes 1.4.24.1 |
| 14:50.28 | dlynes | saxa: 1.4.26 fixed that, and 1.4.26.2 fixed a few other things |
| 14:50.45 | saxa | probably i should try to upgrade |
| 14:50.51 | ManxPower-work | saxa: Several years ago I was having major issues with IAX2, when I switched to SIP I no longer had problems. |
| 14:50.59 | dlynes | saxa: all of my end points are on 1.4.26.2 and my main server is on 1.6.1.8....zero problems |
| 14:51.21 | saxa | ManxPower-work: the thing is that iax was working ok, but then it stopped for some reason |
| 14:51.29 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 14:51.58 | dlynes | ManxPower-work: yeah...I had the same problems....however, iax2 seems to be a lot more stable now than it used to be...before it would affect call quality in a big way...not so, now |
| 14:52.04 | ManxPower-work | saxa: I never figured out WHY I was having IAX2 problems, but since users stopped screaming at me I didn't care about the WHY |
| 14:52.10 | saxa | ok thx, so i will try to do an upgrade, my reason using 1.4.x was problems loading the driver for tdm400 on my x86-64 |
| 14:52.20 | Faustov | saxa: this reminds me... I had a similar issue - the problem was I had a rubbish character out of any context and as a result the config file was getting ignored without any notification |
| 14:52.25 | leifmadsen | please don't upgrade a production box without testing |
| 14:52.36 | leifmadsen | 1.4 -> 1.6.x is a major upgrade |
| 14:52.55 | *** join/#asterisk mark_p (n=mpe@gate.ipvision.dk) |
| 14:52.56 | dlynes | saxa: one thing you might want to note |
| 14:52.58 | saxa | Faustov: that can be, since i have just touched things in extensions.conf |
| 14:53.07 | dlynes | saxa: if you do upgrade to 1.6, and you want faxing to work |
| 14:53.28 | saxa | dlynes: thats interestin |
| 14:53.29 | dlynes | saxa: i.e. using Fax for Asterisk, make sure you're running a 32-bit OS |
| 14:53.42 | saxa | ok, than i wont :) |
| 14:53.43 | dlynes | saxa: Fax for Asterisk isn't available for 64-bit binaries yet |
| 14:53.52 | Chainsaw | leifmadsen: Thanks, 1.6.1.10 is in the Portage tree now. |
| 14:53.53 | saxa | since my machine and os on the server is 64bit |
| 14:54.00 | leifmadsen | Chainsaw: w00t! |
| 14:54.05 | leifmadsen | that didn't take long :) |
| 14:54.14 | elliot98 | from a business point of view, is it ok to have the same person maintain and also develop asterisk systems? |
| 14:54.27 | Chainsaw | leifmadsen: It's had build-system love and it shows :) |
| 14:54.47 | Chainsaw | leifmadsen: Shame about the two bugs that are still open (BT line test & FXO state), but at least my patch queue is getting smaller now. |
| 14:54.50 | dlynes | saxa: well, you could upgrade and just wait for the 64-bit version of the faxing modules |
| 14:54.55 | Katty | elliot98: i do |
| 14:55.03 | Faustov | Chainsaw: Damn, I was about to dcc you the ebuild :D |
| 14:55.03 | Katty | elliot98: but it's not a very good idea. |
| 14:55.05 | leifmadsen | Chainsaw: ya, that's good :) what bug numbers are those? |
| 14:55.09 | dlynes | saxa: after all...doesn't matter whether you're on 1.4 or 1.6...you won't have a 64-bit fax module |
| 14:55.30 | Katty | elliot98: for example, i went into the hospital awhile back with my appendix. during that time...there was no one around if the phone server decided to get hung up in its underwear. |
| 14:55.34 | Chainsaw | leifmadsen: https://issues.asterisk.org/view.php?id=14163 & https://issues.asterisk.org/view.php?id=14577 |
| 14:55.35 | dlynes | Chainsaw: you maintain an asterisk build binary package for a given distribution? |
| 14:56.06 | Chainsaw | leifmadsen: 14577 is waiting for feedback so I'll give you that one. 14163 seems very ready and has been for a long time though. |
| 14:56.08 | saxa | dlynes: that probably will be my thing, just to wait for faxing |
| 14:56.19 | Chainsaw | dlynes: For Gentoo, yes. |
| 14:56.28 | dlynes | Chainsaw: ah...cool |
| 14:56.30 | coppice | you could, of course, use an alternative to fax for asterisk, and run 64 bits today :-) |
| 14:56.36 | Chainsaw | dlynes: It's not a binary package; it's a script that compiles it automatically. |
| 14:56.53 | dlynes | Chainsaw: yeah...I know how Gentoo works...much like portstree on BSD |
| 14:56.59 | Chainsaw | *nod* |
| 14:57.10 | saxa | Chainsaw: where can i see your build scripts ? If possible of course :) |
| 14:57.30 | saxa | maybe i can use a part of it for my packages |
| 14:57.30 | Chainsaw | saxa: Let me find you the CVS tree. It's not pure bash, it integrates with a package manager. |
| 14:57.39 | saxa | of course |
| 14:57.50 | saxa | no problem at all, will get out the right bits |
| 14:57.51 | dlynes | coppice: Yeah, but the problem I had with that library is that I often couldn't get it to succeed with more than 1 page at a time, and it wouldn't do t.38 |
| 14:58.38 | ManxPower-work | dlynes: spandsp has had amazing improvements in the last few years |
| 14:58.40 | dlynes | coppice: well, and it wouldn't work reliably with newer versions of asterisk, wouldn't work reliably with certain fax modems/fax machines, ... |
| 14:58.47 | Chainsaw | saxa: They're here: http://sources.gentoo.org/viewcvs.py/gentoo-x86/net-misc/asterisk/ |
| 14:58.50 | coppice | its in use in a lot of big systems |
| 14:58.54 | dlynes | ManxPower-work: so it "just works" like the digium version now? |
| 14:58.59 | Chainsaw | saxa: 1.6.1.10 hasn't made it in there yet but it should show up soon. |
| 14:59.02 | *** join/#asterisk clart001 (n=clart@host219-94-dynamic.20-87-r.retail.telecomitalia.it) |
| 14:59.11 | ManxPower-work | dlynes: It did the last time I used it about a year ago |
| 14:59.20 | dlynes | ManxPower-work: ah |
| 14:59.44 | ManxPower-work | dlynes: when I first started using it I had similar problems |
| 14:59.56 | dlynes | coppice: does it build with asterisk in the newer versions, or do I need to use callweaver? |
| 15:00.09 | coppice | dlynes: digium had to do something about their screwed up timing to get their own fax module to work. they benefits any other fax module, too :-) |
| 15:00.28 | *** join/#asterisk drepan (n=pandre@apcdns2.autopage.co.za) |
| 15:00.46 | dlynes | coppice: ah...is that why faxes kept dying after 1 or less pages? |
| 15:00.51 | clart001 | Could anybody tell me how can i update asterisk using tarball? |
| 15:00.55 | drepan | Hi, just installed Asterisk GUi |
| 15:00.55 | saxa | Chainsaw: thx |
| 15:01.13 | Katty | drepan: i'm sorry to hear that. |
| 15:01.20 | drepan | sip.conf users do not reflect on the GUI |
| 15:01.31 | drepan | Katty, is it that bad |
| 15:01.38 | ManxPower-work | drepan: Try asking on the AsteriskGUI channel |
| 15:01.40 | coppice | dlynes: that depends on your system. lots of big systems have been running multiple E1s of FAXing all day for years |
| 15:01.53 | dlynes | clart001: tar zxvf file.tar.gz && cd file && ./configure && make menuconfig && make && make install && make config && make samples |
| 15:02.07 | Katty | http://likebit.com/wallpapers/images/phocagallery/Snow/1%20%2810%29.jpg <- pretty! |
| 15:02.33 | *** join/#asterisk Godfather_ (n=Godfathe@187.Red-83-38-229.dynamicIP.rima-tde.net) |
| 15:02.43 | dlynes | coppice: Yeah...we were trying to use it on a T1 using Sangoma hardware previously, and the only way we could get it work was recently switching over to Elastix...and as you probably know, Elastix is a huge POS |
| 15:02.51 | ManxPower-work | Katty: looks like one of the levels f Dante's Inferno |
| 15:02.57 | drepan | My problem, is that I have moved the user to the users.conf file the sip user can make outgoing calls but not receive incoming |
| 15:03.32 | ManxPower-work | drepan: I doubt anyone here can help you. but feel free to keep asking |
| 15:03.36 | dlynes | coppice: so spandsp and the various asterisk modules work with asterisk 1.6.1 now? |
| 15:03.43 | drepan | Is Asterisk GUI a waste of time |
| 15:03.49 | dlynes | drepan: no |
| 15:03.52 | ManxPower-work | dlynes: Sendfax Receive Fax uses spandsp |
| 15:04.17 | dlynes | ManxPower-work: The digium version uses coppice's library? |
| 15:04.22 | drepan | ManxPower-work: the problem would seem that some routing is not correct, not necessarily a GUI issue |
| 15:04.24 | coppice | dlynes: they should do. use spandsp-0.0.6pre12 and results should be very good |
| 15:04.42 | *** join/#asterisk Godfather_ (n=Godfathe@187.Red-83-38-229.dynamicIP.rima-tde.net) |
| 15:04.59 | ManxPower-work | dlynes: I don't know about their COMMERCIAL fax stuff, but the fax stuff that is included in Asterisk 1.6.x uses spandsp |
| 15:05.09 | dlynes | ManxPower-work: Digium = SendFax/ReceiveFax; Coppice = TxFax/ReceiveFax |
| 15:05.18 | ManxPower-work | ~users.conf |
| 15:05.18 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 15:05.19 | clart001 | @dlynes thank you |
| 15:05.22 | leifmadsen | Katty: lol, my new dual screen wallpaper --> http://www.dualscreenwallpaper.com/photos/image_302.jpg |
| 15:05.28 | Chainsaw | uses the Digium fax modules and is quite happy with them |
| 15:05.46 | coppice | dlynes: in asterisk 1.6 you use SendFax and ReceiveFax to access spandsp |
| 15:05.47 | Chainsaw | (I need the aforementioned BT linetest fix so I don't have to restart my analog interface board on a daily basis though) |
| 15:05.49 | dlynes | ManxPower-work: eh? asterisk 1.6.x doesn't include faxing support (other than t.38 passthrough) |
| 15:05.58 | Chainsaw | dlynes: Mine does. |
| 15:06.03 | ManxPower-work | dlynes: Yes. It. does. |
| 15:06.04 | dlynes | coppice: oh....they're included, but you have to install spandsp? |
| 15:06.05 | Chainsaw | dlynes: It helps if you enable spandsp support which is optional. |
| 15:06.07 | coppice | Chainsaw: how many channels do you use? |
| 15:06.21 | dlynes | coppice: and if you install digium's fax for asterisk, it'll unload the spandsp support? |
| 15:06.22 | Chainsaw | coppice: For the fax reception? One analog port. |
| 15:06.50 | dlynes | coppice: or at least override it? |
| 15:06.51 | muiro | can I set IAX to use TCP instead of UDP? |
| 15:06.56 | coppice | I've yet to hear from anyone doing any serious FAXing with digium module. maybe people only ever use it when its free :-) |
| 15:06.57 | ManxPower-work | muiro: no |
| 15:07.03 | muiro | poo :( |
| 15:07.20 | voipmonk | wow... can I make my wheels square? |
| 15:07.30 | Katty | leifmadsen: mahhhh |
| 15:07.36 | [TK]D-Fender | voipmonk: Yes, but highly unrecommended |
| 15:07.40 | coppice | dlynes: I'm not sure what they do about selecting a fax module. app_fax in the asterisk distribution uses spandsp |
| 15:07.48 | leifmadsen | Katty: I found it too funny |
| 15:08.17 | *** join/#asterisk brunner (n=chris@m4a5736d0.tmodns.net) |
| 15:08.52 | brunner | Can anyone give me some idea as to how much more or less stable 1.6 is verses 1.4? |
| 15:09.02 | coppice | I wonder why digium didn't make their FAX module do V.34, since commetrex says they've had that available for a long time |
| 15:09.39 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:10.10 | dlynes | coppice: thanks for the info about spandsp and app_fax, then |
| 15:11.22 | Katty | http://www.youtube.com/watch?v=VXSQ9fsVWxc :> |
| 15:14.26 | coppice | dlynes: If you think spandsp is a problem, consider how those guys doing hundreds of thousands of FAXes a day get by :-) |
| 15:14.57 | *** join/#asterisk TommyBotten (n=noosjent@services.nbi.freecode.no) |
| 15:15.18 | TommyBotten | Is the 1.6.1 series ready for production? |
| 15:15.33 | drepan | Although I prefer using the vi , management want a GUI to see call logs and stats, which is the best solution for this |
| 15:15.53 | *** part/#asterisk clart001 (n=clart@host219-94-dynamic.20-87-r.retail.telecomitalia.it) |
| 15:16.54 | *** join/#asterisk dmz (n=dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 15:17.37 | *** join/#asterisk vally (n=vally@62.48.76.175) |
| 15:17.42 | dlynes | coppice: yeah...I just remember trying to use it in asterisk 1.2 and up to about asterisk 1.4.6, and how many problems i had with it...I just gave up on getting faxing to work, altogether |
| 15:17.53 | dlynes | TommyBotten: in my mind, yes |
| 15:18.04 | Chainsaw | TommyBotten: I run it in production, but my call volume is rather low. |
| 15:18.11 | dlynes | TommyBotten: Personally, I find 1.6.1 way more stable than 1.4 |
| 15:18.17 | *** join/#asterisk ruchir (n=ruchir@122.170.34.164) |
| 15:18.34 | ruchir | hi all |
| 15:18.35 | Chainsaw | TommyBotten: I've made/received ~1800 calls in 2 weeks. |
| 15:18.45 | *** join/#asterisk saftsack (n=oliver@p579DD294.dip.t-dialin.net) |
| 15:18.52 | dlynes | TommyBotten: at our busiest time of the day, we have about 12 simultaneous calls, we have two SIP upstreams, and over a thousand customers |
| 15:19.03 | Katty | Chainsaw: between which dates? |
| 15:19.08 | saftsack | hi, can i connect multiple times to asterisk manager over 5038 at the same time? |
| 15:19.08 | Chainsaw | So indeed, dlynes is a better example. |
| 15:19.26 | Chainsaw | Katty: Between today (just now) and a week and 6 days back when I last restarted. |
| 15:19.34 | voipmonk | saftsack, use astmanproxy |
| 15:19.43 | Katty | Chainsaw: what was that date? |
| 15:19.53 | Katty | the 5th? |
| 15:20.03 | Chainsaw | Katty: The 6th. |
| 15:20.04 | TommyBotten | dlynes: Thanks for the info ... It' |
| 15:20.10 | dlynes | TommyBotten: all of our endpoints are currently running 1.4.26.2 (7 end points, serving between 20 and 200 customers each) over iax2, and then we have a small smattering of SIP end points for the smaller installs (1 or 2 stations each) |
| 15:20.24 | TommyBotten | dlynes: Sorry... eh. It's a low traffic site, but much of the features would be very nice |
| 15:20.24 | Katty | Chainsaw: 4270 calls |
| 15:20.32 | dlynes | TommyBotten: but our main asterisk swtich is running 1.6.1.8 |
| 15:20.42 | Katty | Chainsaw: incoming/outgoing/internal |
| 15:20.58 | Katty | Chainsaw: what do you use for your call logs? |
| 15:20.59 | TommyBotten | dlynes: This site will be having max 10 simultaenous calls, SIP only. |
| 15:21.01 | Chainsaw | Katty: *nod* My total is of the same numbers. Not extremely high load :) |
| 15:21.21 | Chainsaw | Katty: We have no need for them as we don't bill our staff. |
| 15:21.25 | dlynes | TommyBotten: the reason the simultaneous call volume is so low, is because most of our customers are care facilities with a high percentage of dementia patients |
| 15:21.34 | Katty | Chainsaw: i was just curious how you looked at it. |
| 15:21.36 | Katty | Chainsaw: csv then? |
| 15:21.41 | ruchir | can someone point out why asterisk may not be generating proper debug sysmbols in core dump |
| 15:21.46 | Chainsaw | Katty: Everything chargeable goes out over BT (ISDN BRI) lines for which we get itemised bills. |
| 15:21.53 | Chainsaw | Katty: The total calls you mean? |
| 15:22.04 | Chainsaw | Katty: I just use core show calls for that :) |
| 15:22.13 | Katty | ah, right. k |
| 15:23.11 | dlynes | TommyBotten: to put it in perspective, over the last two days, we've had 2155 calls |
| 15:23.13 | Chainsaw | I'll get asked for that feature eventually, I'm sure of it. But for now it's not used :) |
| 15:23.26 | TommyBotten | dlynes: Ah... just about the same as our estimates. |
| 15:23.33 | TommyBotten | dlynes: Thank you very much for your insight |
| 15:23.42 | *** join/#asterisk yosi1234 (i=Bmw_Toro@CPE001217239daa-CM000e5c6f225a.cpe.net.cable.rogers.com) |
| 15:25.25 | yosi1234 | anyone here used either LyLex or Aretta for hosting? |
| 15:31.45 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 15:32.37 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:32.38 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:32.52 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
| 15:33.40 | Dr-Linux|home | I've still two unsolved issue, i thought let's discuss with experts. |
| 15:35.04 | Dr-Linux|home | 1. when i call from PSTN to my * box and dial the meetme extensions .. and after activity when hangup the call .. this call still stuck in asterisk. I tried alot .. but no luck |
| 15:35.07 | Dr-Linux|home | any suggestion |
| 15:36.21 | *** join/#asterisk ecrist (n=ecrist@pdpc/supporter/professional/ecrist) |
| 15:36.26 | ecrist | I have a remote user who has no solid broadband connection. We're considering getting her an ISDN for voip purposes. What do I need to know to get that to work? Can I have the other end terminate at my rack and provide voip over that so it's virtually a direct connection? What hardware do I need? |
| 15:38.48 | kaldemar | Dr-Linux|home: are you using an analog line? |
| 15:39.15 | Dr-Linux|home | kaldemar: Yes analog/Dahdi |
| 15:39.16 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 15:39.51 | kaldemar | ask your telco if they support CDS (=call disconnect supervision) |
| 15:39.58 | ruchir | does asterisk has limit on how much ram it can utlilize? |
| 15:40.36 | dlynes | Dr-Linux|home: woah...dood....you're still here? |
| 15:40.47 | Dr-Linux|home | ruchir: i'm using 16 GM RAM each server |
| 15:41.01 | ruchir | does asterisk utilize more than 8G ram?/ |
| 15:41.07 | dlynes | Dr-Linux|home: still in Karachi I guess, too? |
| 15:41.07 | Dr-Linux|home | dlynes: oo yeah .. you CA guy? |
| 15:41.08 | kaldemar | Dr-Linux|home: you could try busydetect=yes and busycount and busypattern in chan_dahdi.conf |
| 15:41.08 | *** join/#asterisk TimToady_ (n=moi@188.4.43.103.dsl.dyn.forthnet.gr) |
| 15:41.22 | dlynes | Dr-Linux|home: yeah...if CA means Canada |
| 15:41.26 | Dr-Linux|home | dlynes: no, i live in Lahore |
| 15:41.32 | Dr-Linux|home | yeah Canada |
| 15:41.56 | ruchir | the version i have seems to have some memory leaks |
| 15:42.06 | ruchir | unfortunately core dump doesn't contain debug info |
| 15:42.11 | dlynes | Dr-Linux|home: so how's life? Are you understanding asterisk better than you were? |
| 15:42.12 | ruchir | so hard to fix |
| 15:42.28 | Dr-Linux|home | kaldemar: yes, I already had these option in chan_dahdi.conf but no luck. After that i changed to an option to Polarity Reversall |
| 15:42.44 | dlynes | ruchir: doesn't use anywhere close to 8GB's |
| 15:43.12 | Dr-Linux|home | dlynes: lol yeah life is good .. but you know here in Pakistan .. and As i belog to Tribal area of PK ... .. |
| 15:43.30 | Dr-Linux|home | dlynes: I'm okay with Asterisk ;) |
| 15:43.44 | ruchir | dlynes: for me, it keeps increasing with number of calls processed |
| 15:43.52 | ruchir | and crashes in 12 to 15 hours |
| 15:43.57 | kaldemar | ruchir: what version are you using? |
| 15:44.03 | ruchir | 1.4.24.1 |
| 15:44.13 | ruchir | there is some custom code done |
| 15:44.29 | ruchir | but strange thing is debug symbols |
| 15:44.34 | ruchir | why it may not be generating? |
| 15:44.48 | dlynes | ruchir: on my main server, with only asterisk, a web server, mysql (and I don't think anything else), i'm using 245,792KB's (246MB's) |
| 15:44.50 | Dr-Linux|home | ruchir: how many simultenious calls? |
| 15:45.10 | ruchir | it ranges from 60 to 180 |
| 15:45.24 | ruchir | i have 8G ram |
| 15:45.31 | dlynes | ruchir: ah...I'm only doing max 10 simultaneous calls, probably |
| 15:45.38 | kaldemar | ruchir: several memory leaks have been fixed since that version |
| 15:45.46 | dlynes | ruchir: but even then...you shouldn't be that much higher |
| 15:46.00 | dlynes | ruchir: are you using linux? |
| 15:46.02 | Dr-Linux|home | I had no issue on 380/cun calls on 3 GB ram |
| 15:46.07 | ruchir | dlynes: yes |
| 15:46.11 | dlynes | ruchir: pastebin a dump of the command 'free' |
| 15:46.39 | dlynes | ~pb |
| 15:46.39 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 15:46.53 | ruchir | http://pastebin.com/m7408f1c6 |
| 15:47.22 | dlynes | ruchir: you're not actually using as much memory as you think you are |
| 15:47.23 | ruchir | this is affter crash though |
| 15:47.33 | ruchir | let me tell before crash |
| 15:47.44 | dlynes | ruchir: you're only using about 42MB's |
| 15:48.16 | dlynes | ruchir: much of that memory you show there is cached by the kernel...it's not actually used by your application |
| 15:49.04 | ruchir | should i give pmap o/p? |
| 15:49.07 | dlynes | erm...more than 42MB's actually |
| 15:49.17 | ruchir | the o/p was in mb |
| 15:49.44 | dlynes | ruchir: you're using about 464MB's actually |
| 15:49.53 | dlynes | ruchir: o/p? |
| 15:50.02 | ruchir | output |
| 15:50.12 | dlynes | ruchir: i know that...you used 'free -m', not 'free' |
| 15:50.13 | ruchir | dlynes: i see |
| 15:50.28 | ruchir | dlynes: sorry :) |
| 15:50.42 | dlynes | ruchir: so 464MB's is not out of line for the number of simultaneous calls you're doing |
| 15:50.56 | ruchir | dlynes: then why it might keep crashing every 12-15 hours and that also without proper core dump |
| 15:50.56 | dlynes | ruchir: it's probably a little bit on the lean side |
| 15:51.19 | ruchir | when i checked in pmap |
| 15:51.21 | ruchir | it appears that |
| 15:51.21 | dlynes | ruchir: because you probably have a memory leak...it's just not apparent from looking at your current snapshot |
| 15:51.30 | ruchir | asterisk process used ~8G before it crashed |
| 15:51.31 | elliot98 | meetings over |
| 15:52.06 | *** part/#asterisk jermudgeon (n=jhaustin@216-67-61-242.static.acsalaska.net) |
| 15:52.13 | dlynes | ruchir: with your current snapshot, that you pastebin'd, your system looks normal |
| 15:52.35 | ruchir | dlynes: thats current state after crash :) |
| 15:52.49 | dlynes | ruchir: give it several hours and then check it again (after it's had a chance to set up and tear down several thousand calls) |
| 15:52.50 | ruchir | dlynes: i have pmap output before crash |
| 15:52.52 | ruchir | but its too big |
| 15:53.05 | dlynes | ruchir: just try using 'free' for now |
| 15:53.47 | dlynes | ruchir: check your swap space used, and subtract 'buffers' and 'cached' from 'used' to arrive at a more true value |
| 15:53.57 | ruchir | i've been monitoring since 2 days |
| 15:53.58 | dlynes | ruchir: buffers and cached are used internally by the kernel |
| 15:54.09 | ruchir | yeah |
| 15:54.22 | ruchir | dlynes: can u guide why it doesnt generate proper core dump? |
| 15:54.44 | dlynes | ruchir: because your resources got starved too fast for it to be able to react in time |
| 15:55.06 | ruchir | dlynes: umm, what to do then? |
| 15:55.25 | dlynes | ruchir: keep actively monitoring; don't depend on the core dump for now |
| 15:55.50 | dlynes | ruchir: introduce more logging into asterisk, if you suspect a memory leak there |
| 15:55.54 | ruchir | dlynes: the processor is quite powerful |
| 15:56.19 | dlynes | ruchir: there's a number of memory debugging options you can enable during make menuconfig |
| 15:56.29 | ruchir | which ones |
| 15:56.38 | ruchir | i have dont optimize and thread debug enabled |
| 15:56.44 | Dr-Linux|home | does asterisk support P-Asserted? i mean the 1.6.1.x? |
| 15:56.59 | Dr-Linux|home | RPID is working fine though |
| 15:57.40 | dlynes | ruchir: under 10. compiler flags, try DEBUG_FD_LEAKS, MALLOC_DEBUG, LOW_MEMORY |
| 15:57.44 | ruchir | Dr-Linux|home: u can add manually if required |
| 15:58.09 | leifmadsen | Dr-Linux|home: I don't find anything that says "P-Asserted" when doing a grep for it in 1.6.1 |
| 15:58.09 | dlynes | ruchir: but even just MALLOC_DEBUG, you have access to some more features from the asterisk cli to monitor resource usage |
| 15:58.11 | elliot98 | Katty: did anything happen while you were out? |
| 15:58.13 | Dr-Linux|home | ruchir: it is required but the provider |
| 15:58.35 | *** join/#asterisk RypPn (i=TuMbL@rosscom.co.uk) |
| 15:58.40 | elliot98 | ~svn |
| 15:58.41 | infobot | Subversion (aka SVN) is version control software that aims to be a better CVS than CVS. See http://subversion.tigris.org/. |
| 15:58.43 | Dr-Linux|home | yeah, so Asterisk doesn't support it |
| 15:58.54 | *** join/#asterisk ManxPower-work (n=EWieling@107.sub-70-214-135.myvzw.com) |
| 15:59.05 | leifmadsen | I only find "P-Asserted-Identity" |
| 15:59.21 | *** part/#asterisk feature (n=wea@82.103.141.216) |
| 16:00.17 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 16:00.26 | Dr-Linux|home | leifmadsen: yeah but how to use that properly .. not manually in dialplan |
| 16:00.35 | leifmadsen | not my issue to resolve :) |
| 16:00.46 | Dr-Linux|home | :) |
| 16:01.02 | Dr-Linux|home | I often see on WIKI: |
| 16:01.03 | Dr-Linux|home | exten => _1NXXNXXXXXX,1,SipAddHeader(P-Asserted-Identity: <sip:${CALLERIDNUM}>) |
| 16:01.10 | Dr-Linux|home | but that is quite old |
| 16:02.00 | leifmadsen | still seems to be the method to use that I imagine |
| 16:02.49 | dlynes | Dr-Linux|home: any of that SipAddHeader() stuff is usually very vendor-dependent, too |
| 16:04.24 | Dr-Linux|home | dlynes: I replied verizon that we are sending RPID .. will that be enough for callerid showing at for end or we need P-Asserted.... in any case |
| 16:04.42 | ruchir | http://pastebin.com/m6ddeb0dc |
| 16:04.48 | ruchir | this is the top output before crash |
| 16:05.45 | vally | is there a list around of all options possible when using Dial in a dialplan? the list on voip-info is not complete. |
| 16:06.23 | dlynes | Dr-Linux|home: i have no idea |
| 16:06.24 | kaldemar | vally: core show application Dial in CLI |
| 16:06.40 | Dr-Linux|home | no prob |
| 16:06.58 | vally | ty |
| 16:07.01 | kaldemar | vally: asterisk itself is the best place to look for application documentation |
| 16:07.52 | dlynes | ruchir: so at the time of crash, you were using up 2GB's of RAM, which is in my mind too high for your call volume, but that's not the reason it crashed |
| 16:08.05 | dlynes | ruchir: i suspect there's another reason it crashed |
| 16:09.01 | dlynes | ruchir: like maybe a deadlock in asterisk |
| 16:09.50 | dlynes | ruchir: so you might be better off to use a flag like 'DEBUG_CHANNEL_LOCKS', instead |
| 16:10.07 | dlynes | ruchir: that being said, someone was saying there was a number of bugs in your particular version of asterisk |
| 16:10.25 | dlynes | ruchir: i can't see upgrading to a new version being worse than crashing every 14 to 15 hours |
| 16:10.45 | dlynes | ruchir: maybe you'll start crashing every 25 to 30 hours now, instead |
| 16:11.02 | dlynes | ruchir: at the worst, probably :) |
| 16:11.16 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 16:11.30 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 16:17.27 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:20.22 | *** join/#asterisk ryduh (n=ryduh@204.16.143.186) |
| 16:21.53 | *** join/#asterisk ManxPower-work (n=EWieling@107.sub-70-214-135.myvzw.com) |
| 16:22.55 | *** part/#asterisk Jacke (i=jacke@valhalla.bofh.org) |
| 16:24.59 | ManxPower-work | Ugh! I really need a laptop that supports more then 3GB or RAM |
| 16:27.15 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 16:27.40 | *** join/#asterisk Dabian (n=M0RTEN@fsf/member/dabian) |
| 16:27.56 | Katty | elliot98: what do you mean whiel i was out |
| 16:28.13 | Dabian | To make outgoing calls that will cost me .. I apparently need something callled a "TRUNK" ?!?? |
| 16:28.34 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 16:28.36 | Katty | that question hits an alltime low. |
| 16:28.45 | Katty | infobot: the book |
| 16:28.55 | Katty | infobot: thebook? |
| 16:28.56 | infobot | thebook is, like, Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 16:28.57 | Dabian | katty: I got it form the book .. if you mean me. |
| 16:29.05 | Katty | ^- please read more. |
| 16:29.15 | Dabian | katty I read to the end of the chapter. |
| 16:29.20 | [TK]D-Fender | Dabian: Phone service doesn't tend to come for free |
| 16:29.23 | Dabian | I am not english .. |
| 16:29.32 | Katty | or, if you are unwilling to read or don't have time, you might try findind a consulstant to help you |
| 16:29.39 | [TK]D-Fender | Dabian: "trunk" is a often misused term |
| 16:29.49 | Dabian | OK |
| 16:30.08 | Dabian | Well, the book says something like OUTBOUNDTRUNK=ZAP/4 |
| 16:30.13 | [TK]D-Fender | damnIf you want to use an ITSP with * you can. If you want to use physical lines you already have, you can do that too, provided you have a supported hardware interface for it. |
| 16:30.15 | [TK]D-Fender | ~itsp |
| 16:30.16 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 16:30.24 | Dabian | which I guess means that it uses an FXO connected to the asterisk box. |
| 16:30.52 | Katty | in other mews, i'm considering lunch. |
| 16:30.57 | [TK]D-Fender | Dabian: Yes, that sample refers to using Zaptel (DAHDI) hardware interface to interact with a physical line. |
| 16:31.12 | Dabian | I don't have FXO or FXS ... so I want to use my ITSP as trunk. |
| 16:31.18 | Dabian | I guess? |
| 16:31.29 | elliot98 | is it harmful to smoke a unlit cigarette? |
| 16:31.30 | Katty | the considerations so far are, mexican, indian, or that leftover broccoli chicken and rice casserole i brought. |
| 16:31.45 | Katty | elliot98: an unlit cigarette does not smoke. |
| 16:32.01 | Katty | elliot98: however, that's like asking if it's harmful to stand next to a bucket of /dangerious substance/ |
| 16:32.03 | Dabian | Katty: The latter .. if you want my vote. :) |
| 16:32.09 | [TK]D-Fender | Dabian: Sure |
| 16:32.27 | Katty | elliot98: why are you standing next to the bucket/unlit cigarette in the first place? :P |
| 16:32.40 | *** join/#asterisk cesar_CR (n=cesar@201.196.51.10) |
| 16:32.53 | elliot98 | hmmms |
| 16:32.55 | [TK]D-Fender | [11:31]<Katty>elliot98: however, that's like asking if it's harmful to stand next to a bucket of /dangerious substance/ <- Ok, I'll get you a bucket of U-235 and you can just sit next to it for a few hours, k? ;) |
| 16:32.58 | Dabian | Now, I defined my ITSP as peer .. (and I registered them to receive calls) |
| 16:34.01 | Katty | [TK]D-Fender: :< |
| 16:34.07 | Katty | elliot98: why so hmmy |
| 16:34.11 | Dabian | D-Fender: Isn't it hard to get hold of U235? |
| 16:34.24 | elliot98 | Fender's in Iran, no? |
| 16:34.32 | Katty | ^_- |
| 16:34.35 | Katty | no |
| 16:35.05 | Katty | pretty much all military weapons use depleted uranium i believe |
| 16:35.25 | [TK]D-Fender | AL AQUEBA, JIHAD JIHAD!! |
| 16:35.26 | elliot98 | more like titanium |
| 16:35.26 | [TK]D-Fender | ERRR.. I MEAN .. PASS THE PEANUTS PLEASE! |
| 16:35.31 | Katty | of course that's not exactly U-235 either |
| 16:35.53 | elliot98 | cause I got no chicken casserole |
| 16:36.05 | elliot98 | wipes the sweat of his brow |
| 16:36.18 | Katty | it's not sounding very appetizing |
| 16:36.25 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 16:36.26 | Katty | casserole or uranium |
| 16:36.48 | elliot98 | uranium casserole - at you'll know when it's done |
| 16:36.50 | Dabian | Never mind .. I guess I need u238 to build a newclear power plant anyhow. |
| 16:36.57 | [TK]D-Fender | elliot98: bunker buster bombs use depleted uranium as a shell casing for penetration |
| 16:37.27 | elliot98 | can't get much denser then depleted uranium |
| 16:37.43 | Katty | it's used for armor piercing thingies |
| 16:37.53 | elliot98 | cluster bombs |
| 16:38.04 | levity | ManxPower-work: thanks for the help with Dial -it works perfectly- that's why this project is so brilliant people don't mind answering questions |
| 16:38.06 | Katty | anti tank mines |
| 16:38.14 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 16:38.16 | elliot98 | I'm just a simple developer, don't know things like that. |
| 16:38.26 | Dabian | But how do I tell asterisk, thats its ok make a paycall? Is the difference just that I define a variable in the global section? |
| 16:38.27 | Katty | you ever seen a grenade? |
| 16:38.37 | Katty | it's absolutely retarded how LITTLE explosive there is in it |
| 16:38.55 | elliot98 | it looks like a little pack of coolaid |
| 16:38.56 | Katty | warefare is just awful |
| 16:39.01 | Katty | warfare, too |
| 16:39.15 | Katty | yes, let's pack explosives with diamond cut metal! |
| 16:39.37 | Katty | and it will ASPLODE and go everywhere! |
| 16:39.43 | [TK]D-Fender | [11:37]<elliot98>can't get much denser then depleted uranium <- you are clearly new around here..... the crazies come out late at night ;) |
| 16:39.48 | Katty | hopefully near trees, so trees will asplode and go everywhere! |
| 16:40.04 | Dabian | lol fender :) |
| 16:40.16 | elliot98 | katty...how's your co-worker doing? |
| 16:40.22 | Katty | which one |
| 16:40.27 | Katty | there's like 30 of those here |
| 16:40.36 | elliot98 | *the* one |
| 16:40.44 | Katty | ^_- |
| 16:40.50 | Katty | i don't understand your question. |
| 16:40.53 | elliot98 | with that clomping footstep |
| 16:41.02 | Katty | that statement does not parse. |
| 16:41.31 | elliot98 | the one who wants a fax from the internet of something like that |
| 16:41.31 | Katty | that or i'm missing an index statement somewhere |
| 16:41.36 | Katty | oh /that/ one |
| 16:41.46 | Katty | still alive, unfortunately. |
| 16:42.06 | ruchir | is it safe to bypass ast_hangup call ? |
| 16:42.20 | ruchir | when new thread can't be created |
| 16:42.28 | ruchir | my asterisk seems to crash in that case |
| 16:42.42 | ruchir | i think it may introduce some memory leaks but it may run longer |
| 16:42.42 | Katty | dude, i should just go home for the rest of the day. |
| 16:42.48 | Katty | sit around in my jammies |
| 16:42.52 | ruchir | what everyone suggest? |
| 16:42.52 | elliot98 | after all that ruckess...she's a superwoman |
| 16:43.12 | Katty | yeah she made herself look like a real moron. |
| 16:43.16 | Katty | everyone got a good laugh. |
| 16:43.39 | Dabian | Am I right to assume that the book only covers ZAP trunks? |
| 16:43.53 | Dabian | (as in, it doesn't cover other trunks) |
| 16:44.51 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 16:45.02 | *** join/#asterisk Malkor (n=marco@hlle-d9ba03e9.pool.mediaWays.net) |
| 16:46.08 | Katty | hmm. yes. jammies and star trek all afternoon sounds pretty darn good |
| 16:46.39 | [TK]D-Fender | Dabian: read the book... and then realize that there are a lot of other guides out there, including guides for individual TSP's on specifically how to set them up with * |
| 16:46.52 | Dabian | oh darn |
| 16:46.58 | Dabian | "TRUNKING = YES" |
| 16:47.02 | Dabian | missed that |
| 16:47.13 | ManxPower-work | ~siptrunk |
| 16:47.14 | infobot | extra, extra, read all about it, siptrunk is To set a SIP peer/friend/user as a trunk add either trunk=yes or wombat=yes (they both do the same thing) in the peer/friend/user definition in sip.conf |
| 16:47.14 | Katty | i think he wants someone to read to him. |
| 16:47.19 | *** join/#asterisk cesar_CR (n=cesar@201.196.51.10) |
| 16:47.47 | Dabian | Katty: I am just stupid as coitus. |
| 16:48.22 | [TK]D-Fender | suspects a translator app is in use here... |
| 16:48.25 | Katty | firstly, coitus is not stupid. |
| 16:48.34 | Katty | it's /very/ normal. |
| 16:48.39 | leifmadsen | siptrunk? |
| 16:48.44 | leifmadsen | never heard of that option |
| 16:48.46 | Katty | secondly, i didn't say you were stupid..i just think you want someone to read to you (= |
| 16:48.50 | Dabian | Katty: I am neither though. :) |
| 16:49.00 | leifmadsen | oh, I get what it is saying now... how... terribly unuseful |
| 16:49.08 | ManxPower-work | leifmadsen: they both do the same thing "nothing"./ |
| 16:49.21 | leifmadsen | ya... what a stupid runabout way to say that though |
| 16:49.22 | Katty | where's eppigy? |
| 16:49.30 | ManxPower-work | But since SO many people seem desperate to set up SIP trunks and think there's a special option for it.... |
| 16:50.22 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 16:50.25 | leifmadsen | it would be much more productive not to belittle people so much and just say straight out the answer, and not write some whacky runabout way of saying the same thing |
| 16:50.42 | thehar | lol leif |
| 16:50.47 | ManxPower-work | And yet, that has not worked for the several years I tried to do it the nice way. |
| 16:51.05 | *** join/#asterisk eppigy (n=Dave@snugglenets.com) |
| 16:51.11 | Dabian | Well .. I thought I followed the instructions .. but when I /try/ to make outbound calls .. only the free ones go through .. I got to have done something stupid or misread something. |
| 16:51.13 | [TK]D-Fender | leifmadsen: Actually one thing it doesn't mention is that while those 2 "options" have the same worthyness... it fails to say that they are both actually worthless |
| 16:51.13 | leifmadsen | infobot: siptrunk is something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 16:51.14 | infobot | ...but siptrunk is already something else... |
| 16:51.25 | leifmadsen | infobot: no, siptrunk is something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 16:51.26 | infobot | okay, leifmadsen |
| 16:51.30 | thehar | sip trunk is marketing for magic |
| 16:51.33 | [TK]D-Fender | leifmadsen: Before you know it people will start asking what the wombat= parameter is for... |
| 16:51.34 | leifmadsen | ManxPower-work: then just stop helping, because you're not |
| 16:51.43 | eppigy | HELLO |
| 16:51.43 | thehar | boys. boys. simmer! |
| 16:52.09 | Katty | there you are |
| 16:52.10 | Dabian | ~iaxtrunk |
| 16:52.11 | Katty | hugs eppigy |
| 16:52.38 | [TK]D-Fender | thehar: simmer is a low boil... I think you're looking for "chil" :) |
| 16:52.41 | [TK]D-Fender | chill* |
| 16:52.43 | thehar | hahaha |
| 16:52.45 | Katty | eppigy: Dabian was talking about coitus, and you didn't make an appearance. that's when i realized you were missing. |
| 16:52.48 | thehar | indeed [TK]D-Fender |
| 16:52.57 | *** part/#asterisk ryduh (n=ryduh@204.16.143.186) |
| 16:53.49 | Dabian | Btw., D-Fender .. I did find a guide for my TSP .. but it didn't seem to help me. I know I should probably be patient and read the whole book before I expect stuff to work .. but it feels like I am so close. |
| 16:53.55 | leifmadsen | infobot: iaxtrunk is IAX2 trunking enables Asterisk to send media (as mini-frames) from multiple channels using a single header. The reduction in overhead makes the IAX2 protocol more efficient when sending multiple streams to the same endpoint (usually another Asterisk server). The option is 'trunk=<yes|no>' in iax.conf, and is peer configurable. |
| 16:53.56 | infobot | okay, leifmadsen |
| 16:54.31 | eppigy | Katty: lol |
| 16:54.47 | eppigy | yeah I'm getting a catalytic converter replaced |
| 16:54.58 | Dabian | leifmadsen: In |
| 16:55.00 | thehar | Katty: can i have some of your special alcoholic cider? |
| 16:55.09 | eppigy | well now I am going to leave and come back tommorow because the part is in charlotte |
| 16:55.10 | leifmadsen | Dabian: Out? |
| 16:55.18 | Dabian | leifmadsen: On page 110, it is called "trunking=yes" .. unless that is something different. |
| 16:55.19 | Katty | eppigy: that means your vehicle was made after 1975 |
| 16:55.32 | eppigy | yes 2002 |
| 16:55.37 | Katty | :> |
| 16:55.45 | Katty | thehar: that statement does not parse. |
| 16:55.47 | eppigy | TRANS AM WS6 |
| 16:55.53 | eppigy | VROOM |
| 16:55.54 | Katty | oh? |
| 16:55.56 | Katty | trans am? |
| 16:56.01 | thehar | Katty: you spoke of some delicious cider that included liquor |
| 16:56.04 | Katty | grabs racing scarf and mittens |
| 16:56.05 | Katty | LET"S GO |
| 16:56.18 | Katty | thehar: oh. ummm. hmm. |
| 16:56.40 | Katty | thehar: http://42ndrecipestreet.blogspot.com/2009/11/steves-apple-pie-liquor.html <- That? |
| 16:56.52 | leifmadsen | Dabian: please consider submitted errata then: http://oreilly.com/catalog/9780596510480 |
| 16:56.56 | thehar | yes! |
| 16:56.58 | thehar | i require it! |
| 16:57.07 | Katty | it will cost you about 30 bucks to make it |
| 16:57.14 | Katty | and you'll need a large heavy stockpot |
| 16:57.23 | eppigy | Katty: lol |
| 16:57.23 | thehar | i have all of the ingredients |
| 16:57.24 | Katty | and about 3 hours of time |
| 16:57.26 | thehar | but i don't want to make it |
| 16:57.34 | Katty | it simply requires babysitting |
| 16:57.34 | eppigy | yeah it is really fun with the t-tops off |
| 16:57.59 | Katty | i have a friend who drives a trans am. |
| 16:58.08 | Katty | he scares the snickerdoodles out of me when he drives |
| 16:58.16 | Katty | but it's fun! |
| 16:58.33 | eppigy | yesh |
| 16:58.39 | Katty | what color is yours? |
| 16:58.55 | Katty | if you tell me red, i'm going to throw up |
| 16:58.56 | thehar | mmm snickerdoodles |
| 16:59.47 | Katty | the only proper color for a trans am is black. |
| 16:59.59 | vally | is there any way to originate using Dial in the dialplan and having an Agi to an external interface open all the time and simultaneously with that call? |
| 17:03.14 | Katty | hmmmmmmmmmmmmmm. burritos or biryani |
| 17:03.54 | leifmadsen | both! |
| 17:04.02 | leifmadsen | biryani burrito |
| 17:04.02 | *** join/#asterisk uluatu (n=uluatu@189.4.53.18) |
| 17:04.04 | Katty | ^_- |
| 17:04.06 | Katty | eww. no. |
| 17:06.36 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 17:07.57 | afink | hello everyone. How can I have asterisk give a user's caller id over an IAX trunk? Previously I was using SIP for trunks between asterisk servers and using the sendrpid and trustrpid settings to do this. |
| 17:08.09 | *** join/#asterisk Dabian (n=M0RTEN@fsf/member/dabian) |
| 17:08.42 | ManxPower-work | afink: by default whatever callerid is set for a sip user / zap port will be sent. |
| 17:09.02 | ManxPower-work | if you do an IAX2 debug you should see the number being sent. |
| 17:09.25 | afink | hmm, ok everything comes up default user...so that must be set somewhere |
| 17:10.58 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:7811:c820:f908:cc6a) |
| 17:11.14 | ManxPower-work | afink: then the incoming call is not being authenticated against the correct iax.conf user. Historically if an incoming call did not match something in iax.conf then the last entry in the file will be used. I don't know why. I think it's a terrible thing to do. I know this applies at least to 1.2, I don't know about later versions but I've not seen any indication that this has changed in later releases. |
| 17:11.41 | ManxPower-work | Are you setting a callerid= line in the destination iax.conf? |
| 17:11.52 | afink | sure is, in iax.conf my guest context has "Guest IAX user" what you said above is dean on |
| 17:12.07 | ManxPower-work | also how are the users making the calls? SIP phone? |
| 17:12.15 | ManxPower-work | afink: remove the callerid |
| 17:12.17 | afink | ManxPower-work: no not intentionally. Yes they are |
| 17:12.39 | ManxPower-work | You need to fix your iax settings. |
| 17:12.48 | afink | yep. thank you |
| 17:13.17 | ManxPower-work | afink: what version of Asterisk are you using? |
| 17:13.30 | afink | 1.4.23 |
| 17:13.41 | ManxPower-work | afink: I spent several days trying to figure out a similar issue in 2002. |
| 17:14.06 | ManxPower-work | (Asterisk 0.65 I think) |
| 17:18.27 | afink | Thanks ManxPower-work worked like a charm |
| 17:19.02 | ManxPower-work | afink: you should still figure out your issue. Random calls could hit that account and have all the access that account has. |
| 17:19.13 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:19.32 | *** join/#asterisk Cuz (n=plastik@mail.gradeatechs.com) |
| 17:19.48 | Cuz | what would you guys recommend for queue stats? |
| 17:20.22 | kisom | Does anyone know where i can get a high-resolution image of the asterisk logo? |
| 17:20.22 | Cuz | i run a small call centre off the box, with 2 queues, and i'd like to get statistics generated / displayed for the queues |
| 17:20.36 | Cuz | images.google.com |
| 17:20.40 | kisom | tried that. |
| 17:20.54 | kisom | with high resolution i mean > 2048 pixels |
| 17:21.05 | Cuz | http://www.asteriskguru.com/tools/queue_stats.php |
| 17:21.10 | Cuz | http://www.orderlyq.com/orderlystatsse/install.html |
| 17:23.52 | [TK]D-Fender | kisom: ... |
| 17:23.57 | [TK]D-Fender | ~asterisk-trademark |
| 17:24.04 | [TK]D-Fender | ~asterisktrademark |
| 17:24.04 | infobot | [~asterisktrademark] See http://www.digium.com/en/company/view-policy/5 for Digium's trademark policy. |
| 17:24.17 | [TK]D-Fender | kisom: Check with their marketing dept |
| 17:24.21 | kisom | Will do. |
| 17:24.48 | kisom | It's really for promotion only tho. |
| 17:25.08 | mort_gib | [TK]D-Fender: They don't reply to email tho |
| 17:25.31 | [TK]D-Fender | kisom: doesn't matter... tis a legal question |
| 17:25.43 | kisom | Oh well, I'll just go ahead and vector down the logo myself. |
| 17:29.10 | ecrane | "shift-8"? (sorry, I couldn't resist...) |
| 17:29.17 | yosi1234 | anyone here ever used Ring Central?? I was going to get a asterisk hosting provider + a sip provider, 10 channels for my 5 inbound agents, and deall with all the tech issue, but ringcentral, says they can do all this for $50 a month?!?!?they even claim to have 50 channels? any feedback on this? |
| 17:29.46 | yosi1234 | including unlimited long distance? |
| 17:30.01 | ecrane | yosi1234: there might be a per-minute cost associated with phone calls as well. |
| 17:30.28 | yosi1234 | they rold me it includes upto 5000 minutes of toll free, which is a lot |
| 17:30.44 | ecrane | yosi1234: (just a guess.. I haven't actually used them.) |
| 17:30.55 | yosi1234 | fair enough... |
| 17:31.44 | yosi1234 | i was thinking with a hosted solution i need a soft phone for my agents, this way it includes theirs... only downfall is that there is no FOP or FOP to monitor or manage the call queue |
| 17:32.27 | Dabian | OK .. .since there is no "trunk=yes" for sip .. how do I make a paycall since Dial(SIP/provider/ext) doesn't work? |
| 17:32.48 | ecrane | 5000 minutes (per month?) is about 17 hours per line per month. But other then a WAG and simple math I can't help you make a decision. Others in this channel have more experience then I do in dealing with VoIP Providers.. |
| 17:34.12 | yosi1234 | fair enough... i will download their user guide and see what features this cheap setup has.. |
| 17:35.09 | Dabian | No matter what I read, I cannot spot the error in my config. |
| 17:35.24 | p3nguin | pastebin! |
| 17:35.44 | p3nguin | dabian: Put your configs in a pastebin and someone will look over them to help you. |
| 17:35.56 | kaldemar | Dabian: yes it does if you configure it correctly. and iax.conf's trunk=yes has nothing to do with this kind of behavior. |
| 17:36.34 | elliot98 | where does asterisk stand with regard to encrypted calls? |
| 17:36.43 | Dabian | kaldemar: I can make free calls, but not charged calls. I assume somehow the authorisation with my VSP is failed. |
| 17:36.51 | p3nguin | yosi1234: Do you know how to configure Asterisk? |
| 17:37.11 | Dabian | kaldemar: However, I suspect the password is correct, since I can receive calls. |
| 17:38.27 | kaldemar | Dabian: incoming and outgoing calls are two separate animals. show a trace of a failed call so you don't have to guess. |
| 17:39.05 | Dabian | kaldemar: You mean goto console, and do a "sip debug peer <provider>" ? |
| 17:40.07 | kaldemar | Dabian: yes. and also core set verbose 10. |
| 17:40.52 | yosi1234 | p3nguin: i do |
| 17:40.55 | kaldemar | Dabian: the syntax is sip set debug peer peername |
| 17:41.07 | elliot98 | ~asteriskuptospeed |
| 17:41.22 | kaldemar | never before have i heard someone complain about free calls.. :P |
| 17:41.34 | yosi1234 | but asterisk, hosted (enough bandwidth for 10 call) can't run on my home account will cost me $35/month +sip + long distance |
| 17:41.42 | yosi1234 | ringcentral has everything for $49 |
| 17:41.50 | Dabian | Ahh .. apparently Asterisk uses SIP2.0! |
| 17:43.22 | p3nguin | yosi1234: Why will it cost you this much at home? |
| 17:43.34 | Dabian | circuit busy .. whats that? It means the line is busy? |
| 17:44.12 | kaldemar | Dabian: show a cli output |
| 17:44.48 | Dabian | I'm on the console .. everything looks good .. it calls <sip:PSTN_NUMBER@provider.tld> |
| 17:44.59 | Dabian | but gets a "circuit busy" or something. |
| 17:45.42 | Dabian | No .. "Not found - no such extension" ? |
| 17:46.02 | dlynes | Dabian: that's why you're getting a 'circuit-busy'...you're trying to send a call to a non-existent extension |
| 17:46.23 | yosi1234 | home there is not enough bandwith for 10 channels + 5 remote agents... thats about 1.5mbs upstream assuming the connection is good |
| 17:46.28 | kaldemar | Dabian: don't tell what happens, pastebin the whole output |
| 17:46.30 | Dabian | dlynes: But the number exits |
| 17:46.33 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.136.2.dsl.dyn.forthnet.gr) |
| 17:46.48 | dlynes | Dabian: pastebin your console output and your extensions.conf file |
| 17:47.03 | *** join/#asterisk hobodave (n=hobodave@63.149.16.13) |
| 17:47.04 | dlynes | ~pb |
| 17:47.05 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 17:47.08 | Dabian | impossible .. I'm on the console . |
| 17:47.22 | dlynes | Dabian: and? how does that make it impossible? |
| 17:47.59 | Dabian | dlynes: How can I copy out that scrolled out the screen in the console? |
| 17:48.08 | Dabian | s/out /output / |
| 17:48.09 | dlynes | Dabian: you mean you're at a normal vc, and you're not in a gui, right? |
| 17:48.11 | hobodave | besides the license, is ABE significantly different than Asterisk Community Edition? |
| 17:48.26 | Dabian | dlynes: Yes, I am on VC |
| 17:48.29 | hobodave | infobot and Dabian: use screen |
| 17:48.29 | Dabian | No GUI. |
| 17:48.35 | ManxPower-work | hobodave: you have no access to modify the system outisde of Asterisk |
| 17:48.43 | dlynes | hobodave: ABE gives you paid support....asterisk community edition doesn't |
| 17:48.43 | ManxPower-work | Dabian: use a terminal that allows copy and paste |
| 17:48.47 | Dabian | hobo: I am using screen. |
| 17:49.02 | dlynes | Dabian: load up gpm |
| 17:49.04 | ManxPower-work | Dabian: then learn how you can copy and paste in screen |
| 17:49.09 | hobodave | dlynes: is that the extent of the differences? |
| 17:49.26 | kaldemar | Dabian: man script |
| 17:49.39 | Dabian | dlynes: The problem is that there is so much output, that it scrolls off the screen .. so its "very hard" (at least) to copy paste. |
| 17:49.49 | Dabian | kalde: ahh .. nice idea! |
| 17:49.54 | dlynes | Dabian: then scroll back using shift-pageup, then select the text to paste, go to another terminal, load up lynx to pastebin.ca, go to the edit field, then click your middle button (or left and right buttons together) to paste |
| 17:50.02 | Dabian | kalde: IIRC that will put output in a file. |
| 17:50.10 | hobodave | my company paid for ABE last October, and our server just fell over, we don't have the original binaries anymore. Digium is giving me the runaround and not letting me download the *same* versions that we purchased |
| 17:50.36 | kaldemar | Dabian: yes it will, while still showing you stdout and stderr |
| 17:50.40 | Dabian | dlynes: OK .. I'll need to disable my other provider, because that idiot requires me to register every 30 seconds. |
| 17:51.01 | dlynes | Dabian: you also have your /var/log/asterisk/full file, too |
| 17:53.20 | dlynes | Dabian: just grep -v your sip registrations out of there |
| 17:53.50 | hobodave | dlynes: could you clarify please? Is there no difference between ABE and community besides the support subscription? |
| 17:54.24 | dlynes | hobodave: when did I say that was the only difference? |
| 17:54.33 | dlynes | hobodave: I know that's one of the differences |
| 17:54.43 | ManxPower-work | hobodave: Nobody knows since you don't have access to the ABE source. Digium says that ABI and Asterisk are built from the same source tree. I don't know if ABE has any GUI stuff, but if so that's not part of the standard Asterisk. |
| 17:54.45 | dlynes | hobodave: I don't know what all the differences are, because I don't use ABE |
| 17:54.57 | Dabian | I think it doesn't authenticate before it makes the call. |
| 17:54.58 | hobodave | I didn't say you did, that's why I asked for clarification |
| 17:55.21 | dlynes | hobodave: but the paid support is the MAIN difference, and that's where the value is |
| 17:56.08 | *** join/#asterisk Tim_Toady (n=moi@77.49.31.217.dsl.dyn.forthnet.gr) |
| 17:56.20 | Dabian | dlynes: There is no full. |
| 17:56.32 | dlynes | Dabian: how about messages? |
| 17:56.41 | Dabian | dlynes: Bingo! |
| 17:56.45 | hobodave | I think our server intentionally chose to die 2 weeks after our support subscription expired |
| 17:56.51 | hobodave | #$%^& |
| 17:57.30 | Dabian | It complains about using "_." as extension. |
| 17:57.44 | dlynes | Dabian: that's an invalid extension |
| 17:57.45 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:57.51 | dlynes | Dabian: I think you really meant _X. |
| 17:58.05 | kaldemar | it's not invalid, just poor practice |
| 17:58.16 | Dabian | Right |
| 17:58.23 | dlynes | kaldemar: so what on earth does repeat the previous non-existent atom mean? |
| 17:58.34 | Dabian | dlynes: Ahh. |
| 17:58.45 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:58.47 | dlynes | kaldemar: because that's effectively what _. means, right? |
| 17:59.57 | kaldemar | dlynes: it just matches 1 or more of everything, including the special extensions. syntax wise it is valid. |
| 18:00.04 | ManxPower-work | _. means "match anything including extensions "h", "a", "o", and any others. |
| 18:00.17 | ManxPower-work | You do NOT want that. |
| 18:00.54 | Dabian | The only clue I get is "SIP/2.0 404 Not found -- no such extension" |
| 18:01.00 | dlynes | kaldemar: ah, ok...good way to screw up your system, in other words |
| 18:01.20 | kaldemar | dlynes: definitely |
| 18:01.25 | dlynes | Dabian: we've asked you for a log, and your dialplan several times now |
| 18:01.35 | dlynes | Dabian: without that, it's like shooting in the dark |
| 18:01.38 | Dabian | Right |
| 18:01.42 | ManxPower-work | Dabian: we won't know what is happening until we see a sip debug |
| 18:01.56 | p3nguin | dabian: If you aren't willing to provide the information, you may stop asking for help at any time. |
| 18:02.03 | Dabian | But last time I showed a log .. I was just told "Oh then the extension doesn't exist." |
| 18:02.05 | dlynes | Dabian: are you logged in on the local console, or are you logged in using ssh? |
| 18:02.18 | Dabian | I got the output in a file now. |
| 18:02.24 | Dabian | I used kaldemars idea |
| 18:02.25 | dlynes | Dabian: ok...pastebin it |
| 18:02.31 | kaldemar | Dabian: so what? it was last time and an extension may not exist. |
| 18:02.48 | Dabian | Its a valid PSTN number. |
| 18:03.06 | Dabian | I can call free PSTN numbers, but no pay numbers. |
| 18:03.18 | ManxPower-work | Dabian: less talk, more pastebin |
| 18:03.19 | dlynes | Dabian: pastebin!!!! :) |
| 18:03.58 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 18:04.26 | *** part/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 18:09.01 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:10.55 | *** join/#asterisk akyaki_ (n=aki@S0106004f4e164e77.gv.shawcable.net) |
| 18:12.36 | *** join/#asterisk astrutt (i=astrutt@pinky.ratman.org) |
| 18:13.04 | hobodave | hey Dabian |
| 18:13.12 | hobodave | http://wgetpaste.zlin.dk/wgetpaste-1 |
| 18:13.41 | Katty | peeks in |
| 18:14.36 | *** join/#asterisk joako (n=ston3d@opensuse/member/joak0) |
| 18:14.43 | hobodave | heh the siteis shutdown |
| 18:14.58 | joako | Is there a simple way to enforce a do not call list via Asterisk? Maybe to store it in a MySQL database? |
| 18:15.17 | ManxPower-work | Dabian is never going to get this problem solved. |
| 18:16.00 | dlynes | joako: you mean a blacklist? |
| 18:16.15 | dlynes | joako: or you mean for outbound calls? |
| 18:16.26 | Katty | HELLO THAR |
| 18:16.46 | joako | dlynes: For outbound calls. |
| 18:16.47 | Nugget | either way, the answer to joako's question is the same. "It depends on how you define 'simple'" |
| 18:16.57 | Katty | hi nugget |
| 18:17.01 | Katty | you nuggety nuggateer |
| 18:17.04 | Nugget | huggles Katty |
| 18:17.07 | dlynes | joako: for outbound calls or inbound calls, you can enforce a blacklist, by either using sql functions or asterisk database functions |
| 18:17.09 | Katty | hugs Nugget |
| 18:17.23 | hobodave | ok |
| 18:17.25 | hobodave | Dabian: http://gist.github.com/238948 |
| 18:17.33 | joako | dlynes: Any perfomance issues with the asterisk DB in 1.4? |
| 18:17.34 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:17.43 | hobodave | that supports dpaste and pastebin |
| 18:17.56 | dlynes | joako: not afaik |
| 18:18.18 | dlynes | joako: asterisk database is pretty time tested and proven...it's been in wide use since before v1.0 |
| 18:18.44 | *** join/#asterisk brunner (n=chris@m3a5736d0.tmodns.net) |
| 18:18.47 | Katty | hmmm. |
| 18:18.49 | Katty | infobot: winter |
| 18:18.50 | infobot | i guess winter is the coldest season of the year!, or the hottest season of the year for the other hemisphere which is in summer at the same time |
| 18:18.53 | *** join/#asterisk areay (n=areay@ns0.thebosscorporation.com) |
| 18:18.57 | brunner | Are there any stats about the reliability of 1.6 vs. 14? |
| 18:19.00 | brunner | 1.4* |
| 18:19.05 | Cuz | anyone know of a good free queue statistics tool? |
| 18:19.21 | eppigy | NEIN |
| 18:19.21 | Cuz | i'm looking at QrderlyQ, but it's not free |
| 18:19.22 | brunner | I want to switch to 1.6, but I've got a developer who swears it's significantly less reliable |
| 18:19.23 | areay | hi everyone... is it me or is ChanSpy/ExtenSpy extremely buggy, to the point of being unusable? |
| 18:19.34 | Katty | hi dave |
| 18:19.36 | eppigy | Katty: it is the really nice silver color |
| 18:19.36 | Katty | HELLO DAVE |
| 18:19.42 | Katty | :> |
| 18:19.47 | Katty | silver is also an acceptabuhls color |
| 18:19.47 | joako | dlynes: Right well if the list were 500 numbers I don't doubt the perfomance would be fine, but what if the DB had 100,000 entries or even more could Asterisk DB handle that OK? |
| 18:19.49 | Cuz | and Asterisk Queuestats 0.3 everyone seems to have problems setting up since it uses postgres instead of mysql |
| 18:19.49 | Katty | tho i prefer black |
| 18:19.54 | Cuz | so are there any others? |
| 18:20.04 | Katty | what kind of queue data |
| 18:20.38 | Cuz | small call center queue stats, so time people were in queues, who's answering the most calls out of my agents, etc |
| 18:20.45 | Cuz | i have 2 queues with 4 agents in each |
| 18:20.54 | jblack | joako: It's sqlite3, I believe. why don't you do some testing |
| 18:21.00 | ManxPower-work | joako: AstDB is BerkelyDB |
| 18:21.08 | Katty | hmmmmmmmmm. |
| 18:21.14 | Katty | well Asterisk-stat might do something |
| 18:21.22 | Katty | but it's meant for CDR stuff |
| 18:21.31 | jblack | berkely? Wow |
| 18:22.05 | dlynes | joako: berkeleydb's performance is quite good....it's been around for years and years |
| 18:22.07 | areay | has anyone else had any experience using ChanSpy/ExtenSpy? |
| 18:22.07 | Cuz | Katty: found a few commercially available packages, but this has got to be a common request, no? |
| 18:22.18 | ManxPower-work | jblack: I don't know which version |
| 18:22.19 | Cuz | like there should be a FOSS project for this |
| 18:22.19 | Faustov | arossouw: logging fixed, restart of asterisk solved the issue |
| 18:22.20 | dlynes | joako: it's not as good as btrieve, but it's the next best thing |
| 18:22.29 | Katty | Cuz: dunno. i don't commonly work with queues and agents. |
| 18:23.01 | Katty | Cuz: asterisk-stat queries the cdr database |
| 18:23.09 | ManxPower-work | Every time I work with queues I make sure to have garlic, holy water, a cross, and someone that runs slower than me |
| 18:23.26 | dlynes | Dabian: dood....have you figured out how to pastebin yet? |
| 18:23.26 | Katty | ha |
| 18:23.28 | joako | areay: I use chanspy. It works. |
| 18:23.35 | Katty | dlynes: ;P |
| 18:23.37 | Katty | dlynes: that is hilarious |
| 18:23.53 | Cuz | Katty: yea, thats a far too high level overview of what's going on |
| 18:24.03 | Katty | Cuz: bummer. |
| 18:24.03 | kaldemar | brunner: 1.6.1 maybe, but 1.6.0 should be pretty close. you probably won't find any exact data on this. |
| 18:24.17 | areay | joako, i'm having problems with it... sending a dtmf * does nothing but cut the sound out... and it cycles through calls for about 30 secs, and then it cuts out (still connected, but no sound) |
| 18:24.33 | areay | really weird |
| 18:24.35 | Katty | http://www.youtube.com/watch?v=VXSQ9fsVWxc <- put on mittens before viewing. |
| 18:24.40 | brunner | kaldemar: do you have an opinion about the stability of 1.6 as it compares to 1.4? |
| 18:24.47 | elliot98 | does using addqueuemember require me to list the agent in agent.conf? |
| 18:24.50 | brunner | maybe I just need commercial support |
| 18:24.54 | dlynes | kaldemar: 1.6.1 is unstable? I've been using it for about 3 months now, and haven't had any instability issues |
| 18:25.01 | areay | i would expect it to run fine for hours without needing to redial the ChanSpy extension |
| 18:25.23 | areay | but my monitors have to redial every 5mins or so |
| 18:25.41 | Nugget | 1.6 broke my SIP/tcp phones, and I had core dumps until I re-worked my dialplan to eliminate macros. Other than that it's been ok. |
| 18:25.41 | brunner | dlynes: how many concurrent channels? |
| 18:25.48 | dlynes | brunner: up to 10 |
| 18:25.55 | areay | and whisper doesn't work at all |
| 18:26.00 | dlynes | brunner: well...up to 10 concurrent calls (20 concurrent channels) |
| 18:26.01 | joako | areay: When you press * it switchs between channels. What I have notied because I use SIP and progressinband=no is that progress tones (ringing, etc) are not played when using ChanSpy() you also might want to try ChanSpy(|b) |
| 18:26.06 | brunner | dlynes: okay, well, I'm always running at least 400 |
| 18:26.06 | eppigy | Katty: http://i58.photobucket.com/albums/g245/eppigy/585219951_l.jpg |
| 18:26.24 | areay | joako, thanks, i'll give that a go |
| 18:26.25 | brunner | sorry, that's 400 calls. I have no idea how many channels it's typically using. |
| 18:26.32 | dlynes | brunner: so what kind of instability have you seen with 1.6.1? |
| 18:27.01 | joako | NOW THAT YOU MENTION IT when I am running > 100 channels I seem to have some issues, but I always thought it was the call dropping... |
| 18:27.04 | kaldemar | dlynes: i didn't say it was unstable. all don't consider it as stable as 1.4 though. |
| 18:27.22 | dlynes | kaldemar: really? I found it's just the opposite |
| 18:27.26 | Katty | eppigy: hmmmmmm. keeps timing out |
| 18:27.26 | brunner | dlynes: I haven't yet. I'm still running 1.4 because someone keeps telling me that 1.6 isn't stable enough yet, and I'm seeking more opinions |
| 18:27.31 | dlynes | kaldemar: I've found 1.6.1 to be more stable than 1.4 |
| 18:27.32 | Katty | eppigy: put it somewhere else temporarily |
| 18:27.33 | kaldemar | brunner: i'd go for a 1.6 branch. |
| 18:28.10 | citywok | has anybody ever fought calls dropping at 29.5 minutes before? |
| 18:28.14 | kaldemar | dlynes: it's also a question of what you do with it of course. |
| 18:28.37 | *** join/#asterisk albertoandrade (n=albertoa@189.4.53.18) |
| 18:28.40 | dlynes | citywok: yes...I narrowed my issue down to some audiocodes boxes |
| 18:28.48 | Katty | http://www.youtube.com/watch?v=-kpEOQWN39s <- DUCK! |
| 18:29.02 | dlynes | citywok: in my case it was pretty much at 30 minutes right on the nose, every time |
| 18:29.14 | citywok | dlynes: i'm not using anything but SIP -> SIP |
| 18:29.20 | dlynes | citywok: same here |
| 18:29.28 | citywok | what was the issue that you found? |
| 18:29.54 | dlynes | citywok: after 30 minutes, the call would just drop...couldn't get any conversations longer than 30 minutes on the audiocodes 24 port fxs gateway boxes |
| 18:29.58 | citywok | it seems to not happen when i'm trying to make it happen in testing, and always happens i dont want it to. |
| 18:30.08 | dlynes | citywok: the solution we found was to use mediatrix 4124's instead |
| 18:30.16 | citywok | ah, darn i'm using all asterisk |
| 18:30.24 | dlynes | citywok: mostly because audiocodes' support was horrible |
| 18:31.05 | dlynes | citywok: which version of asterisk? |
| 18:31.17 | dlynes | citywok: or version(s)? |
| 18:31.37 | areay | joako, ok i have a room full of people on the phone, and extension 777 mapped to ChanSpy(SIP|b).. when i dial 777 i get a beep and nothing happens... if i dial * nothing happens... what am i doing wrong? |
| 18:32.26 | brunner | holy crap! $8k/yr for support? |
| 18:32.27 | brunner | that's nuts |
| 18:32.39 | [TK]D-Fender | [13:24]<elliot98>does using addqueuemember require me to list the agent in agent.conf? <- only if you are referring to an Agent channel |
| 18:33.00 | ManxPower-work | areay: did you try specifying a SIP channel? |
| 18:33.05 | p3nguin | areay: Is "SIP" a valid channel on which to spy? I think not. |
| 18:33.25 | Cuz | updating 156 packages during production hours *crosses fingers* |
| 18:33.27 | Cuz | lmao |
| 18:33.30 | Cuz | i'm such a bofh |
| 18:33.34 | Cuz | i hope no one gets cut off |
| 18:33.36 | areay | ManxPower-work, p3nguin: i thought you just specified the prefix for that and it cycled thru available channels? |
| 18:33.36 | ManxPower-work | p3nguin: "core show application chanspy" will tell you. |
| 18:33.57 | p3nguin | manxpower-work: You've misdirected your statement again. |
| 18:34.05 | kaldemar | p3nguin: it takes a prefix, not a channel |
| 18:34.09 | *** join/#asterisk errotan (n=errotan@a1678.adsl.pool.eol.hu) |
| 18:34.18 | p3nguin | kaldemar: You've misdirected your statement, as well. |
| 18:34.27 | ManxPower-work | p3nguin: I was suggesting you read the documentation for him. |
| 18:34.38 | kaldemar | p3nguin: no, i haven't and neither did ManxPower-work. |
| 18:34.46 | p3nguin | I really don't care to read hte documentation for him. |
| 18:35.00 | ManxPower-work | p3nguin: Me neither |
| 18:35.05 | kaldemar | p3nguin: you suggested that ChanSpy needs a channel as an argument, which isn't the case. |
| 18:35.09 | p3nguin | My ChanSpy works just fine, by the way. |
| 18:35.19 | saxa | <PROTECTED> |
| 18:35.23 | elliot98 | Fender: So I can dynamically add sip devices even if they are not in the agents.conf. Does this act as a callback or must the agents stay online? |
| 18:35.27 | saxa | any ideas what this means ? |
| 18:36.23 | joako | saxa: What brand/model device are you using? I would not worry about it... do you have any "real" issues? |
| 18:36.29 | areay | ok i've read the documentation, googled, etc... i wouldn't come here asking for help if i hadn't done the above |
| 18:36.44 | [TK]D-Fender | saxa: Means * doesn't seem to like what the other side is saying |
| 18:36.46 | joako | elliot98: Do you add them e.g. as Agent/something or SIP/something? |
| 18:36.48 | ManxPower-work | areay: You read "core show application chanspy"? That's the only official documentation |
| 18:37.04 | areay | ManxPower-work, yeah... the same info is posted all over google too |
| 18:37.09 | [TK]D-Fender | elliot98: Means the Queue will directly generate a call to that channel |
| 18:37.16 | [TK]D-Fender | (device actually) |
| 18:37.24 | ManxPower-work | areay: The info on "google" could be from ANY version of Asterisk |
| 18:37.27 | kaldemar | areay: what version are you using? |
| 18:37.36 | saxa | joako: its * to * connection |
| 18:37.42 | saxa | [TK]D-Fender: thx |
| 18:37.43 | areay | ManxPower-work, kaldemar: i'm using 1.6 |
| 18:37.52 | kaldemar | areay: change SIP|b to SIP,b and try again |
| 18:38.00 | areay | kaldemar, thx i'll try that |
| 18:38.09 | saxa | seems that by sip protocol my 2 boxes get registered one to another |
| 18:38.17 | ManxPower-work | or SIP/ or SIP/123 or whatever. |
| 18:38.39 | kaldemar | ManxPower-work: just SIP should be fine according to the docs |
| 18:39.07 | elliot98 | well, I was using agentcallcaklogin |
| 18:39.12 | areay | kaldemar, seems to be working fine... thanks :) |
| 18:39.31 | ManxPower-work | "Dialing a series of digits followed by # builds a channel name to append to 'chanprefix'. For example, executing ChanSpy(Agent) and then dialing the digits '1234#' while spying will begin spying on the channel 'Agent/1234'. |
| 18:39.31 | ManxPower-work | " |
| 18:39.40 | citywok | dlynes: 1.6.anything and the SVN tree |
| 18:40.24 | areay | kaldemar, ok it worked as it should for about minute and a half... now pressing * to cycle thru does nothing and it's just quiet :/ |
| 18:40.39 | ManxPower-work | areay: then file a bug |
| 18:41.32 | areay | ManxPower-work, i don't know enough about my configuration to provide information to reproduce this "bug" (if it is one) |
| 18:41.43 | areay | ManxPower-work, would i be better using 1.4? |
| 18:41.43 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 18:41.50 | areay | better off* |
| 18:41.55 | Dabian | Whats digest authorisation? |
| 18:42.12 | ManxPower-work | areay: I have no idea. |
| 18:42.27 | eppigy | Katty: try now :> |
| 18:42.30 | [TK]D-Fender | Dabian: Permission to take a snack-break |
| 18:42.31 | ManxPower-work | Dabian: A lot less talk, a lot more pastebin action. |
| 18:43.00 | *** join/#asterisk Godfather_ (n=Godfathe@187.Red-83-38-229.dynamicIP.rima-tde.net) |
| 18:43.17 | ManxPower-work | Dabian: At this point you are just pissing people off because you are not follwing directions. |
| 18:43.32 | [TK]D-Fender | areay: [13:37]<kaldemar>areay: change SIP|b to SIP,b and try again <-- reason itsn't working after this is because you didn't read the "|" is deprecated as a parameter delimiter |
| 18:43.35 | p3nguin | Is he still here? |
| 18:44.01 | areay | [TK]D-Fender, yeah i changed that and it worked... but after a minute and a half or so it goes quiet |
| 18:44.10 | ManxPower-work | [TK]D-Fender: it still shows | in the show application into in 1.6.1 |
| 18:44.16 | kaldemar | [TK]D-Fender: or before that change.. |
| 18:45.30 | areay | seems strange... i would have thought ChanSpy/ExtenSpy would be in use by the majority of asterisk's userbase |
| 18:45.56 | p3nguin | ChanSpy works fine for me, but I have never been successful using ExtenSpy. |
| 18:46.05 | Godfather_ | Im searching a VoIP phone 802.11g (and also not cordless) like the SPA525G but more cheaper |
| 18:46.25 | areay | ChanSpy/ExtenSpy are just means to an end... i need to be able to monitor active outbound calls |
| 18:46.58 | *** join/#asterisk akyaki_ (n=aki@S0106004f4e164e77.gv.shawcable.net) |
| 18:47.48 | *** join/#asterisk dirtyd (n=DirtyD@41330563.cst.lightpath.net) |
| 18:47.50 | dirtyd | Hi |
| 18:48.16 | dirtyd | Question, can I use an asterisk box to "provide" T1 PRI service to another PBX? |
| 18:48.54 | kaldemar | dirtyd: yes |
| 18:48.54 | ManxPower-work | dirtyd: yes |
| 18:50.47 | areay | i think it must be a bug... DTMF presses during ChanSpy are really, well, buggy |
| 18:50.54 | areay | surely this isn't the same system as everyone's using |
| 18:51.49 | areay | is 1.6 my problem here? |
| 18:52.02 | areay | is confused..... still. |
| 18:52.07 | joako | saxa: Interesting. Honestly I don't do much SIP-to-SIP between asterisk machines. If you are not having any noticeable issues I would not bother with it. If you really want to resolve the warnings I would first double check your SIP configuration is the same on both ends and if you are using dissimilar Asterisk versions switch them to the same version |
| 18:52.23 | p3nguin | If I wanted to use ExtenSpy instead of ChanSpy, I would probably try to use ExtenSpy(202@internal). Would that be correct syntax, or would ExtenSpy(SIP/202@internal) be required? or are both wrong? |
| 18:52.46 | areay | p3nguin, don't bother. neither work as advertised |
| 18:52.58 | dirtyd | kaldemar: what's the technology I should be focusing on? Is there a how to on it? |
| 18:53.03 | ManxPower-work | p3nguin: Extensions don't normally have a TECH specified (TECH=SIP/ZAP/etc) |
| 18:53.06 | p3nguin | areay: That's been my problem in the past, and, hence, my reason to use ChanSpy. |
| 18:54.03 | areay | p3nguin, it's a shame... i think if the developers spent a little more time working with it it would be one of asterisk's most useful features |
| 18:55.13 | areay | maybe we should start a petition |
| 18:55.27 | kaldemar | dirtyd: what kind of service are you going to provide? |
| 18:55.42 | kaldemar | ~book |
| 18:55.43 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:55.48 | ManxPower-work | dirtyd: First step is to learn how the PSTN works and how T-1s work. |
| 18:55.58 | kaldemar | dirtyd: ^^ that's a good read on asterisk and telephony in general |
| 18:58.05 | areay | it would be nice, at least, to know whether or not i'm making a mistake, or if this is a limitation of asterisk |
| 18:58.13 | dirtyd | kaldemar, I have a digium card with two ports, 1 port is connected to a T1 PRI, I have another PBX that I'd like to share the T1 PRI with through the asterisk box. |
| 18:58.37 | dirtyd | The second PBX is not an asterisk box. |
| 18:58.45 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 18:59.00 | ManxPower-work | dirtyd: You can't technically SHARE a T-1 PRI, but Asterisk can accept the call from the telco and then send the call to your PBX. |
| 18:59.15 | ManxPower-work | You will need a T-1 crossover cable to go between Asterisk and the PBX. Ethernet Crossover won't wokr |
| 18:59.18 | ayeso | how can i specify what Ethernet interface asterisk should listen on for sip requests? |
| 18:59.47 | ManxPower-work | ayeso: that is almost always a bad idea. |
| 19:00.08 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 19:00.18 | ayeso | ManxPower-work: so If i have multiple interfaces, will it automatically listen on all interfaces? |
| 19:00.39 | ManxPower-work | ayeso: yes, and it will rely on the operating system to send the packet out the correct interface. |
| 19:00.51 | p3nguin | areay: So ExtenSpy simply does not work, and so all my attempts to use it have been in vain? |
| 19:00.57 | *** join/#asterisk xpot-mobile (n=xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
| 19:01.01 | ayeso | ManxPower-work: This is just for testing something out, thanks for your info. |
| 19:03.40 | [TK]D-Fender | p3nguin: ExtenSpy() target an exten, not a device |
| 19:03.45 | areay | p3nguin, i'm sure, deep inside a cave somewhere in the world, there's one fat developer with pizza stains on his t-shirt and a grin on his face that knows how to use ExtenSpy... you have to ask yourself what's going to be quicker... finding this man, or guessing his syntax |
| 19:04.05 | [TK]D-Fender | areay: this is not Raw-Cat Sigh-Hence |
| 19:04.16 | p3nguin | [tk]d-fender: I used ExtenSpy(211@internal), but that failed to work. Is that incorrect? |
| 19:04.39 | areay | [TK]D-Fender, huh? |
| 19:04.45 | areay | p3nguin, i think you've found your man ;) |
| 19:04.48 | [TK]D-Fender | p3nguin: How do I know it should be legal? I don't see your calls, or your dialplan, or your attempt |
| 19:05.04 | [TK]D-Fender | p3nguin: It looks like it COULD be legal. |
| 19:05.43 | ManxPower-work | assuming you have an exten => 211,1,whatever in the [internal] context in asterisk. |
| 19:05.57 | p3nguin | Dialing SIP/211@internal is how I reach the phone registered to 211. |
| 19:06.03 | ManxPower-work | if that extension is in a context like [from-internal] then of course it won't work |
| 19:06.17 | ManxPower-work | p3nguin: you do NOt dial SIP/211@internal |
| 19:06.55 | ManxPower-work | since that would specify calling the sip.conf device named [internal] in sip.conf and asking that device to send the all to extension 211 |
| 19:07.03 | p3nguin | Seems like it to me. In the internal context, Dial(SIP/211,50) rings that phone. |
| 19:07.15 | ManxPower-work | p3nguin: then you are dialing SIP/211 |
| 19:07.19 | areay | [TK]D-Fender, can you shed any light on why ChanSpy is cutting out randomly? |
| 19:07.23 | p3nguin | That's what I said. |
| 19:07.29 | [TK]D-Fender | p3nguin: areay I don't see any debug. |
| 19:07.51 | [TK]D-Fender | [14:05]<p3nguin>Dialing SIP/211@internal is how I reach the phone registered to 211. <-- you did not say "SIP/211" here |
| 19:08.39 | p3nguin | Sure seems like that's what the text says. |
| 19:08.41 | areay | [TK]D-Fender, i'm rounding up some debug for you... gonna be about 15mins... |
| 19:08.50 | areay | brb |
| 19:08.58 | p3nguin | I'll be pasting sip debug in a minute, as well. |
| 19:09.07 | ManxPower-work | p3nguin: if you Dial(SIP/211@internal) it will not work |
| 19:09.37 | [TK]D-Fender | p3nguin: Note the "@internal" on the end. NOT the same |
| 19:11.05 | [TK]D-Fender | ManxPower-work: reminds me of internet newbs who try to e-mail things to places like "www.joe.blow@http://hotmail.com" |
| 19:11.27 | [TK]D-Fender | ManxPower-work: and just tack on prefix's and suffixes wherever they like |
| 19:11.51 | *** join/#asterisk magic_hat (n=geoffdou@adsl-69-211-62-116.dsl.chcgil.ameritech.net) |
| 19:11.52 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 19:12.23 | p3nguin | Interpret what I said however you wish; you know what I said and you know what it meant, and I additionally provided the exact syntax for clarification. |
| 19:12.27 | ManxPower-work | *nod* The lack of being specific and correct is going to eventually open the PBX up to being used to make unauthorized calls. |
| 19:13.02 | Cuz | after adding a name to the "accountcode" section in sip_additional.conf/sip.conf should create new CDR records with whatever name i give it in /var/log/asterisk/cdr-csv/"name".csv, no? |
| 19:13.33 | [TK]D-Fender | Cuz: for devices using those entries, yes |
| 19:13.35 | Cuz | i've done this, and added "support" as the account code for all my queue agents, but i still see no file, so i touched the file, fixed the permissions, still no data |
| 19:13.45 | Cuz | do i need to restart asterisk before it will take affect? |
| 19:14.12 | dirtyd | kaldemar: I'd like to the second PBX to see my asterisk BOX as an ordinary T1 PRI connection. |
| 19:14.40 | kaldemar | dirtyd: it will, nothing special there |
| 19:14.50 | ManxPower-work | dirtyd: People do this ALL the time. |
| 19:16.27 | [TK]D-Fender | Cuz: A reload of SIP should do it for those devices |
| 19:17.15 | Dabian | Not sure what I did .. but somehow I managed to call out now. |
| 19:17.30 | Cuz | [TK]D-Fender: yea, i thought i did already through freepbx (don't beat me please) |
| 19:17.34 | *** join/#asterisk lozarythmic (n=lpraties@e1-1.ns500-1.ts.milt.as9105.net) |
| 19:17.48 | magic_hat | hey all. anyone got a good VOIP provider recommendation for me? |
| 19:17.53 | Cuz | because i grepped, and support is in the agent's "accountcode" section in sip_additional.conf |
| 19:18.06 | Cuz | but that csv file contains no data |
| 19:21.35 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 19:23.20 | *** join/#asterisk Tech_Travis (n=tech_tra@208.179.137.131) |
| 19:23.44 | Katty | :> |
| 19:24.22 | SuPrSLuG | polycom question. is it possible to disable the mystats and buddies softkeys and still have presence enabled on the phone? |
| 19:24.38 | ManxPower-work | SuPrSLuG: yes |
| 19:27.47 | SuPrSLuG | ManxPower-work: http://pastebin.com/m12dfc368 what am i missing? the buddy and mystat softkeys still show on the phone |
| 19:28.03 | ManxPower-work | SuPrSLuG: Read the admin buide. |
| 19:28.06 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 19:28.07 | ManxPower-work | And the Admin Guide too |
| 19:28.11 | Dabian | Neat .. .I can use both my VSP's to dial out .. only one of them is exceptionally slow. |
| 19:28.14 | p3nguin | ExtenSpy() SIP debug -> http://pastebin.com/d2f51e4a4 |
| 19:28.20 | ManxPower-work | That's what I did when I disabled those keys |
| 19:28.33 | *** join/#asterisk Cuz (n=plastik@mail.gradeatechs.com) |
| 19:28.38 | Cuz | weird |
| 19:28.48 | Dabian | yea |
| 19:28.53 | ManxPower-work | you are also welcome to look at http://www.fnords.org/~eric/polycom-config-examples/ |
| 19:28.55 | SuPrSLuG | ManxPower-work: i did it said set these values to "0" , but they're still there |
| 19:29.16 | Cuz | so i restarted asterisk, but i still see 0 data in /var/log/asterisk/cdr-csv/support.csv |
| 19:29.28 | Cuz | though i've confirmed that accountcode=support is set for all my static agents in the queues |
| 19:30.56 | Katty | so you are going to use the cdr |
| 19:31.24 | Katty | very enteresting |
| 19:35.01 | Dabian | 10XXXX ... Shouldn't that match any 6-digit number starting with 10? |
| 19:35.14 | p3nguin | _10XXXX might |
| 19:35.38 | *** join/#asterisk _maxiwll_AR (n=_maxiwll@201-75-21-11-ma.cpe.vivax.com.br) |
| 19:35.52 | Dabian | p3nguin: Thanks. |
| 19:35.56 | Dabian | Why the _? |
| 19:36.13 | leifmadsen | because it means "pattern match" |
| 19:36.23 | leifmadsen | this is one of the most basic syntax forms in the dialplan you should know |
| 19:36.26 | Dabian | Ahh .. of course. |
| 19:36.27 | leifmadsen | please read the documentation |
| 19:37.25 | Dabian | I have read .. and I remember that X matched digit .. but I didn't get what the "_"'s was for. Thank you for stressing that. |
| 19:37.46 | Cuz | oh man |
| 19:37.59 | Cuz | something in the latest wanpipe updates borked my box |
| 19:38.12 | Cuz | asterisk is working fine internally, my trunk is down for my PRI |
| 19:38.51 | Katty | Dabian: wow, you're still here? |
| 19:39.16 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 19:39.17 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 19:39.25 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 19:39.29 | Cuz | master of queues?!! |
| 19:39.34 | Cuz | zomg help me pl0x :) |
| 19:39.46 | Cuz | i need good queue statistic generation tool :P |
| 19:39.48 | Dabian | Katty: Yeah. Its working now. Somehow I fixed it myself. |
| 19:40.26 | Katty | Cuz: well if you're using cdr now, i'd highly recommend asterisk-stat |
| 19:40.27 | SuPrSLuG | ManxPower-work: I had to put a 0 in the entry instead of Null, pres.idleSoftKeys="0" /> work now :-D |
| 19:40.51 | SuPrSLuG | ManxPower-work: Thanks |
| 19:41.12 | elliot98 | every few days for some reason, queues seems to stop working and I need to restart * |
| 19:41.40 | Cuz | Katty: but will this give me the breakdown that I'm looking for, like how long agents stay on a call, etc... |
| 19:42.04 | Katty | Cuz: asterisk-stat queries your cdr |
| 19:42.06 | *** join/#asterisk torrancew (n=torrance@btr.macconsultinggroup.com) |
| 19:42.25 | Katty | Cuz: it's a nice pretty php wraparound for the database |
| 19:43.24 | magic_hat | hey all. our current voip provider is having some call quality probs... looking around for a US based sip alternative. Suggestions? |
| 19:43.25 | _maxiwll_AR | in my accounts IAX2 ta giving this message Unmonitored oq can this be? |
| 19:45.45 | Cuz | Katty: alright, i'll check it out |
| 19:45.54 | Cuz | i got my wanpipe problem fixed |
| 19:45.56 | [TK]D-Fender | _maxiwll_AR: Because you didn't tell * to monitor them |
| 19:46.09 | Cuz | fucking yum updated my wanpipe modules and core |
| 19:46.09 | Katty | Cuz: http://www.areski.net/asterisk-stat-v1_3/ |
| 19:46.21 | Cuz | and i guess decided to blow away my wanpipe config files for my t1/PRI |
| 19:46.24 | Cuz | lol |
| 19:46.35 | Cuz | i had to set up my PRI from scratch |
| 19:46.58 | Cuz | but its working again, fully updated :) |
| 19:47.14 | areay | ok people my ChanSpy works perfectly now... if anyone asks in the future tell them they need to setup SPYGROUPs for it to function correctly. |
| 19:47.18 | Cuz | guys in the queues thanked me for the 15 min break from the phones |
| 19:47.31 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:47.38 | areay | does a dance |
| 19:48.23 | Cuz | oh asterisk-stat? |
| 19:48.27 | Cuz | i already have that |
| 19:48.40 | Cuz | uses trixbox |
| 19:48.48 | Cuz | dodges flying rotten vegetables |
| 19:49.18 | Katty | hmmmmmmmmmmmmmmmmm |
| 19:49.22 | Cuz | i need something fancier than asterisk-stat, more queue-centric reporting is what i'm looking for |
| 19:49.36 | Katty | i have a feeling you're going to pay for it too |
| 19:49.57 | _maxiwll_AR | [TK]D-Fender: how do I resolve this? |
| 19:50.38 | ManxPower-work | _maxiwll_AR: notice the quality= option in iax.conf.sample |
| 19:50.51 | Cuz | Katty: yea, i've only found one free package that has potential |
| 19:51.01 | Cuz | http://www.asteriskguru.com/tools/queue_stats.php |
| 19:51.22 | p3nguin | Hmm. I don't have any SPYGROUPs and my ChanSpy works just fine. ExtenSpy is the one that doesn't work. |
| 19:51.45 | p3nguin | manxpower-work: quality or qualify? |
| 19:51.57 | [TK]D-Fender | SHH! |
| 19:52.08 | ManxPower-work | Sorry, qualify |
| 19:52.42 | _maxiwll_AR | ManxPower-work: /etc/asterisk/iax.conf ? |
| 19:53.48 | *** join/#asterisk Alagar (n=Administ@122.164.41.133) |
| 19:53.55 | ManxPower-work | _maxiwll_AR: I don't know what your /etc/asterisk/iax.conf contains. I do know that the iax.conf.sample that comes with the Asterisk source has that option and should have info about that option. |
| 19:56.10 | _maxiwll_AR | ManxPower-work: here in my = on the quality has to be yes right? |
| 19:56.49 | Cuz | so i still can't figure out why after specifying an "accountcode" of support, why I'm still not seeing any data in /var/log/asterisk/cdr-csv/support.csv |
| 19:56.54 | Cuz | anyone have any ideas? |
| 19:57.17 | ManxPower-work | _maxiwll_AR: qualify not quality. I mistyped the first time |
| 19:58.24 | *** join/#asterisk [netman] (n=netman@112.Red-81-36-134.dynamicIP.rima-tde.net) |
| 19:58.38 | Katty | DANCING QUEEEN |
| 20:00.20 | Katty | i would not be surprised if it was snowing outside right now. |
| 20:00.24 | Katty | p3nguin: what's the temp up north |
| 20:00.33 | p3nguin | katty: 48 over here. |
| 20:00.58 | Katty | that's practically balmy |
| 20:01.02 | Katty | oh snap |
| 20:01.04 | Katty | 57F here. |
| 20:01.06 | p3nguin | It's chilly. |
| 20:01.10 | Katty | guess i should get outside more often |
| 20:01.21 | Katty | it was cold and overcast this morning |
| 20:03.38 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 20:04.19 | p3nguin | The sun is shining, now, so it might warm up just in time to get cold again when the sun sets. |
| 20:05.47 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 20:06.02 | *** join/#asterisk levity (n=levity@unaffiliated/canuck) |
| 20:08.08 | elliot98 | pretty cold by you guys |
| 20:08.54 | [TK]D-Fender | and that's just our "welcome"! :p |
| 20:10.54 | *** join/#asterisk nix8n82 (n=nate@63.162.27.14) |
| 20:12.13 | *** part/#asterisk superbeef (n=superbee@74.84.194.4) |
| 20:13.44 | *** join/#asterisk Alagar (n=Administ@122.164.41.133) |
| 20:14.51 | p3nguin | So... why doesn't ExtenSpy work for me? I'm using the syntax as prescribed in The Book. |
| 20:15.04 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 20:16.18 | *** join/#asterisk Alagar (n=Administ@122.164.41.133) |
| 20:17.54 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:21.16 | Katty | p3nguin: it doesn't love you. |
| 20:21.27 | Katty | p3nguin: i blogged it, awhile back, but i believe my dydns is down |
| 20:21.35 | Katty | checks |
| 20:21.38 | Katty | yep. |
| 20:22.09 | p3nguin | If I remember right, your blogging matched up with my configuration, and it still didn't work out for me. |
| 20:22.40 | nix8n82 | Have you ever wondered if you were retarded? if you have done anything like the following it is highly likely you are. At 11:20pm Est someone went to test an asterisk box setup to record a conference call, and then take that call and call other numbers and play that recording to them, for a sample audio tt-monkeys |
| 20:23.34 | p3nguin | Yeah? Is there more to it? |
| 20:23.34 | Katty | p3nguin: ExtenSpy(110@foo)? |
| 20:23.49 | Dabian | nix8n82: I think your message was cut. Please paste from "for a sample audio". |
| 20:24.02 | p3nguin | katty: Pretty much. http://pastebin.com/d2f51e4a4 |
| 20:24.05 | Katty | p3nguin: ExtenSpy(115@default) |
| 20:24.13 | nix8n82 | and forgot to disable the rining to a fire station and an emergency call center(basically 911) |
| 20:24.34 | Dabian | Hehe :) |
| 20:24.58 | Katty | p3nguin: turn debug off and show me the cli when you do it |
| 20:25.05 | Katty | p3nguin: verbose 10 please |
| 20:25.35 | p3nguin | katty: Verybose was on 100 in that paste. |
| 20:25.38 | nix8n82 | is that retarted |
| 20:25.40 | p3nguin | verbose, even |
| 20:26.08 | p3nguin | Oh, nevermind... you wanted what was BEFORE the debug. One second. |
| 20:27.21 | nix8n82 | I also had tt-monty-knights set to play but thankfully no picked up when that one went through |
| 20:27.23 | Dabian | p3nguin: Why are you using private IP numbers? |
| 20:27.46 | Katty | facepalms |
| 20:27.51 | p3nguin | dabian: Obviously because the devices are on a private network. |
| 20:28.09 | Katty | no, that deserves the double facepalm |
| 20:28.19 | Dabian | p3nguin: That sounds reasonable .. I guess I need to rephrase my question. :) |
| 20:28.23 | p3nguin | katty: http://pastebin.com/d36e3484a |
| 20:28.26 | *** join/#asterisk Cuz (n=plastik@mail.gradeatechs.com) |
| 20:28.29 | p3nguin | katty: Nothing really useful here. |
| 20:28.32 | Katty | http://www.youtube.com/watch?v=13eDfrMgFQM <- see 0:30 |
| 20:28.44 | Dabian | Whats faceplam? |
| 20:28.55 | Katty | ^- Dabian see above video. about :30 |
| 20:29.23 | Dabian | katty: Hmm .. maybe I can get the liba-plugin for mplayer ... |
| 20:29.29 | Katty | p3nguin: hmmm. |
| 20:29.38 | Katty | p3nguin: try something more direct. |
| 20:29.44 | Dabian | libaa, even. |
| 20:29.52 | Katty | p3nguin: skip your reads and what not, make an extension dial extenspy(211@internal) |
| 20:30.11 | p3nguin | katty: Okay. However, it shows that the varibale was read in directly, and it shows that it tried to spy on 211@internal. |
| 20:30.17 | Katty | i see that |
| 20:30.17 | Dabian | Oh .. that wont work .. Youtube is adobe flash. |
| 20:30.23 | [TK]D-Fender | p3nguin: I don't see a call for it to spy on in there anywhere |
| 20:30.26 | Katty | do it anyway |
| 20:31.12 | Katty | i just assumed 201 was bridged to 211 |
| 20:31.16 | p3nguin | 211 is on a call, but since that isn't the problem, I didn't enable debug until I was ready to check ExtenSpy. |
| 20:31.19 | Katty | p3nguin: the call was bridged, right? |
| 20:31.22 | p3nguin | 211 is on a call with someone else. |
| 20:31.31 | Katty | k |
| 20:31.46 | [TK]D-Fender | p3nguin: We don't see this anywhere |
| 20:32.13 | p3nguin | Then you didn't read the paste I left. |
| 20:32.29 | [TK]D-Fender | p3nguin: [15:28]<p3nguin>katty: http://pastebin.com/d36e3484a <-- this one? |
| 20:32.34 | [TK]D-Fender | p3nguin: Indeed I did. |
| 20:32.44 | Katty | he had a previous one with debugg info |
| 20:32.44 | p3nguin | (1328.13) <p3nguin> ExtenSpy() SIP debug -> http://pastebin.com/d2f51e4a4 |
| 20:32.46 | p3nguin | this one. |
| 20:33.33 | p3nguin | katty: Same thing when using it without the variable. |
| 20:33.51 | Dabian | katty: LOL .. very nice clip. :) |
| 20:34.00 | Katty | Dabian: yes i thought so. |
| 20:34.22 | [TK]D-Fender | p3nguin: I still don't see the call you are spying on.... |
| 20:34.28 | Dabian | Katty: So double facepalming, is what Picard does near the end, where he realises his error? |
| 20:34.43 | [TK]D-Fender | p3nguin: Show the the call being PLACED that you are trying to spy on. |
| 20:35.00 | p3nguin | It's hard to do that if the call is already in effect. |
| 20:35.15 | [TK]D-Fender | p3nguin: "core show channels concise" |
| 20:35.26 | Katty | Dabian: yes. |
| 20:35.30 | Dabian | Katty: Decending his face into his palms. |
| 20:35.33 | Katty | Dabian: that's the Double Facepalm |
| 20:35.33 | Dabian | :) |
| 20:35.40 | Katty | Dabian: of which you are worthy |
| 20:35.43 | Katty | applauds |
| 20:35.45 | Katty | it is a rare honor. |
| 20:36.04 | Dabian | Katty: Thank you? But how have I earned this great honor? |
| 20:36.12 | Katty | giggles |
| 20:36.19 | thehar | giggles |
| 20:36.22 | thehar | i had beers with lunch |
| 20:36.24 | thehar | it was fantastic |
| 20:36.28 | Katty | hugs on thehar |
| 20:36.35 | Katty | i think i shall drink tonight |
| 20:36.39 | Katty | it will be guuuud |
| 20:36.42 | Dabian | thehar: Sounds like a revolutionary concept! I have to try that! :D |
| 20:36.51 | thehar | you've never had a beer lunch? |
| 20:36.54 | thehar | oh Dabian .. |
| 20:36.55 | thehar | tisk tisk |
| 20:37.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:37.03 | Katty | i usually opt for the vodka tho |
| 20:37.13 | thehar | i have beers so i don't smell like a bar |
| 20:37.19 | thehar | that. and. beer > * |
| 20:37.25 | Katty | vodka > beer |
| 20:37.28 | Dabian | Vodka is very good .. but I often get dizzy when drinking it. |
| 20:37.28 | thehar | beer. wine. spirits.. in that order |
| 20:37.52 | Dabian | Esp. in larger amounts. |
| 20:38.12 | Katty | thehar: i'm going to have to call snickerdoodles on that statement |
| 20:38.46 | thehar | you will not |
| 20:38.51 | thehar | i have an open palm right now. ready to be used |
| 20:39.22 | Katty | SNICKERDOODLES! |
| 20:39.37 | Katty | *hee* |
| 20:39.42 | Dabian | is not going to ask this time.. |
| 20:40.00 | [TK]D-Fender | p3nguin: and? |
| 20:40.02 | Katty | probably for the best. |
| 20:40.07 | thehar | mmmm i LOVE snickerdoodles |
| 20:40.11 | thehar | delicious and sugary |
| 20:40.25 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 20:40.33 | elliot98 | snickerdoodles |
| 20:40.37 | elliot98 | ? |
| 20:40.47 | thehar | i'm going to put this mini bottle of baileys in my coffee |
| 20:40.54 | Dabian | Its incredible cool .. My Asterisk is working. :) |
| 20:41.12 | Dabian | thehar: You can mix baileys with beer also! |
| 20:41.14 | elliot98 | claps in applause |
| 20:41.20 | Dabian | thehard: (Not a pretty sight, though). |
| 20:41.28 | thehar | Dabian: i refer it over ice or with coffee |
| 20:41.41 | Dabian | *nod* |
| 20:42.08 | Dabian | With beer, I prefer something like Nordsoe Olie. |
| 20:42.16 | Dabian | (Nord sea oil) |
| 20:42.29 | thehar | or shit. |
| 20:42.43 | Dabian | A spirit you can get in Scandinavia and some places in Germany |
| 20:43.25 | elliot98 | new website got a "looks good to me" stamp of approval Yippy! |
| 20:43.58 | Dabian | elliot98: Gratties! :) |
| 20:45.15 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 20:45.34 | Dabian | elliot98: Can we see it? |
| 20:48.25 | *** join/#asterisk mr_ian (n=mr_ian@S0106001b63f49383.du.shawcable.net) |
| 20:49.47 | Katty | i prefer baileys with milk and ice. |
| 20:50.19 | Katty | too strong otherwise (= |
| 20:52.57 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
| 20:53.10 | p3nguin | new call -> http://pastebin.com/d769db925 |
| 20:54.19 | [TK]D-Fender | p3nguin: there is no call to 211@admin2 there |
| 20:54.40 | p3nguin | I agree. 211 made and outgoing call. |
| 20:54.47 | *** join/#asterisk thepopolinux (n=cyril@AToulouse-552-1-73-215.w92-136.abo.wanadoo.fr) |
| 20:54.50 | [TK]D-Fender | p3nguin: SIP/211 made a call |
| 20:54.55 | thepopolinux | Hello all |
| 20:55.20 | _maxiwll_AR | I'm having a problem when I call a branch to the other I hear the voice of the person in it does not listen to me, how can I solve? |
| 20:55.38 | [TK]D-Fender | p3nguin: You have the really large conceptual issue I suspected from the moment this question came up. |
| 20:55.40 | p3nguin | That's right. "core show application ExtenSpy" says Only channels created by outgoing calls for the specified extension will be selected for spying. |
| 20:56.05 | ManxPower-work | p3nguin: that means the DIALED extension, not the calling device |
| 20:56.21 | [TK]D-Fender | ManxPower-work: *sigh* will they ever learn? |
| 20:56.42 | ManxPower-work | [TK]D-Fender: no |
| 20:56.45 | *** part/#asterisk dD0T (n=dD0T@unaffiliated/dd0t) |
| 20:57.39 | muiro | I have such a hard time explaining that to my coworkers |
| 20:57.58 | [TK]D-Fender | muiro: Consider blunt force trauma |
| 20:57.58 | [TK]D-Fender | muiro: High impat training! |
| 20:57.58 | [TK]D-Fender | impact* |
| 20:58.21 | thepopolinux | I have a problem : I want to forward SIP incoming call trought IAX trunk, to another server. It work, but when the destination serveur Playback a sound, the sound cut after few seconds. The call is not hung up, there is simply no more sound |
| 20:58.33 | thepopolinux | Have you an idea please? |
| 20:59.08 | muiro | [TK]D-Fender: that might work, though I usually default to highliting my dialplan on printouts |
| 20:59.09 | p3nguin | Well, that's just ridiculous. How would I know every "extension" that one of my devices would call? |
| 20:59.26 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) [NETSPLIT VICTIM] |
| 20:59.26 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) [NETSPLIT VICTIM] |
| 20:59.26 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) [NETSPLIT VICTIM] |
| 20:59.26 | *** join/#asterisk Woody2143 (n=Woody214@machine76.Level3.com) [NETSPLIT VICTIM] |
| 20:59.26 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 20:59.26 | *** join/#asterisk rpm (n=russell@S0106000c29898b7e.cg.shawcable.net) [NETSPLIT VICTIM] |
| 20:59.26 | *** join/#asterisk brian (n=brian@unaffiliated/brian) [NETSPLIT VICTIM] |
| 20:59.26 | *** join/#asterisk evil_gordita (n=evilgord@ip70-188-50-186.rn.hr.cox.net) [NETSPLIT VICTIM] |
| 20:59.26 | *** join/#asterisk iCEBrkr (n=icebrkr@72.251.206.106) [NETSPLIT VICTIM] |
| 20:59.26 | *** join/#asterisk Knightfal (n=Knightfa@75.142.144.171) [NETSPLIT VICTIM] |
| 20:59.26 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) [NETSPLIT VICTIM] |
| 20:59.28 | [TK]D-Fender | p3nguin: You missed the POINT of that application. |
| 20:59.34 | p3nguin | Obviously! |
| 20:59.49 | [TK]D-Fender | p3nguin: If you want to spy on a call placed by a DEViCE, that's what ChanSpy is for <- |
| 21:00.01 | p3nguin | Yeah, ChanSpy works GREAT! |
| 21:00.05 | _maxiwll_AR | <PROTECTED> |
| 21:00.12 | [TK]D-Fender | p3nguin: If you want to spy on WHOEVER is calling an extension in your dialplan, that's what ExtenSpy is for |
| 21:00.20 | p3nguin | ChanSpy(SIP/211) ;bingo |
| 21:00.54 | [TK]D-Fender | p3nguin: "Someone called John's extension and is int he middle of leaving a voicemail and I want to SPY on the EXtensION the caller is executing" <- |
| 21:01.04 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) [NETSPLIT VICTIM] |
| 21:01.04 | *** join/#asterisk iCEBrkr (n=icebrkr@72.251.206.106) |
| 21:01.32 | ManxPower-work | This is important: An extension cannot make calls. Extensions can only receive calls. Devices make calls. |
| 21:02.15 | p3nguin | They need to stop changing the definition of "extension." |
| 21:02.31 | [TK]D-Fender | p3nguin: No, you need to stop |
| 21:02.44 | p3nguin | I didn't change anything. |
| 21:02.47 | Katty | both of you need to simmer |
| 21:02.53 | Katty | and then ice. |
| 21:02.54 | [TK]D-Fender | p3nguin: extensions.conf <- this has EXTENSIONS. sip.conf <- this has DEViCES |
| 21:03.06 | thepopolinux | someone would have an idea for the problem please? |
| 21:03.07 | p3nguin | ya think? |
| 21:03.19 | Katty | SIMMER |
| 21:03.27 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:03.27 | p3nguin | bubbles slightly |
| 21:03.32 | Katty | sends [TK]D-Fender and p3nguin to different corners. |
| 21:03.35 | *** join/#asterisk Zambezi (i=Zulu@80.67.9.2) |
| 21:03.54 | [TK]D-Fender | looks around in his round room in confusion |
| 21:04.06 | [TK]D-Fender | Katty: FOILED AGAIN! |
| 21:04.13 | p3nguin | Why, because circular rooms have no corners? |
| 21:04.14 | Katty | hands [TK]D-Fender the cluebat for comfort |
| 21:05.09 | p3nguin | I often hear people talking about corners of circular objects, and I never quite understood. |
| 21:06.48 | Katty | ponders dinner. |
| 21:06.51 | [TK]D-Fender | p3nguin: another sample "Which moron keeps f-ing dialing 911?" : Extenspy(911@outgoingcontext) |
| 21:07.04 | Katty | oh for goodness sake. |
| 21:07.16 | Katty | i think this conversation has gone on quite long enough |
| 21:07.21 | [TK]D-Fender | Katty: Sorry, all out of goodness... come back again tomorrow! |
| 21:07.24 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 21:07.24 | Katty | let it go :P |
| 21:07.26 | citywok | i'm testing with the SVN tree, and it does not work at all with one of my 2 sip providers... |
| 21:07.47 | citywok | it works fine with voicepulse, but calls fail after 5 seconds or so with bandwidth.com |
| 21:07.48 | *** join/#asterisk chuckf_ (n=chuckf@ubuntu/member/chuckf) |
| 21:07.52 | Pan3D | "does not work" |
| 21:07.58 | [TK]D-Fender | citywok: What "tree"? SVN is a problem, not a specific branch of * |
| 21:08.11 | Katty | Pan3D: nono, "does not work at all" |
| 21:08.16 | citywok | the SVN tree of asterisk 1.6.1 |
| 21:08.17 | Pan3D | lol |
| 21:08.25 | elliot98 | For anyone who wants to see the new site or has seen it already, here we go: http://asteriskspeed.linuxinnovations.com |
| 21:08.32 | citywok | if i go back to 1.6..1.9 i have no problems |
| 21:08.32 | [TK]D-Fender | sitAt what point in time? |
| 21:08.36 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 21:08.39 | Pan3D | Katty: ok, thanks. for a second I thought he hadn't provided enough information. |
| 21:08.42 | elliot98 | any suggestions *greatly* appreciated! |
| 21:08.43 | kaldemar | oak or birch? |
| 21:08.45 | Pan3D | that clears it up |
| 21:08.50 | _maxiwll_AR | [TK]D-Fender: I'm having a problem when I call a branch to the other I hear the voice of the person in it does not listen to me, how can I solve? |
| 21:08.54 | [TK]D-Fender | citywok: 1.6.1.10 <- |
| 21:08.58 | Katty | Pan3D: yes i thought so too. |
| 21:09.11 | [TK]D-Fender | kaldemar: Birch... for fastest ignition |
| 21:09.15 | citywok | [TK]D-Fender: the call links up sometimes for about a second, before audio goes dead, and then 5 seconds later or so the entire call goes dead |
| 21:09.33 | Pan3D | citywok: ports open? |
| 21:09.41 | Pan3D | did you look at logs? |
| 21:09.42 | citywok | lol, i downloaded and compiled 1.6.1.9 yesterday as it was the current one on digium, doh. |
| 21:09.43 | Pan3D | etc. |
| 21:09.48 | *** join/#asterisk Amorsen (n=Amorsen@94.127.50.7) |
| 21:09.58 | citywok | Pan3D: no firewalling, directly connected to the internet, iptables -F, iptables -P INPUT ACCPET |
| 21:10.06 | Pan3D | logs? |
| 21:10.11 | citywok | that was one of the first things i tested actually |
| 21:10.15 | kaldemar | [TK]D-Fender: and better heating power! screw the stability of oak. |
| 21:10.17 | p3nguin | For ExtenSpy, what determines if a channel is "outgoing" or not? |
| 21:10.25 | citywok | i did verbose 10, sip debug on, and used putty to log the console |
| 21:10.34 | [TK]D-Fender | kaldemar: Birch doesn't burn as long though... depends on your goal |
| 21:10.37 | citywok | just in the few seconds the log is massive |
| 21:10.41 | thepopolinux | guys, could you give me ways to respond to my problem please? it would be very nice because I really don't know how to do |
| 21:10.57 | citywok | with 35 phones registered there's a lot of chatter which makes it difficult for me to read |
| 21:10.59 | elliot98 | popo...wazzup? |
| 21:11.08 | kaldemar | p3nguin: extenspy doesn't care about channels, only extensions |
| 21:11.11 | Pan3D | citywok: grep is your friend |
| 21:11.16 | *** join/#asterisk Godfather_ (n=Godfathe@187.Red-83-38-229.dynamicIP.rima-tde.net) |
| 21:11.19 | citywok | looking for what? |
| 21:11.29 | thepopolinux | elliot98 > I have a problem : I want to forward SIP incoming call trought IAX trunk, to another server. It work, but when the destination serveur Playback a sound, the sound cut after few seconds. The call is not hung up, there is simply no more sound |
| 21:11.35 | p3nguin | "Only channels created by outgoing calls for the specified extension will be selected for spying." |
| 21:11.37 | Pan3D | well, for starters, traffic between your server and the endpoint. |
| 21:11.42 | p3nguin | "Only channels created by outgoing calls for the specified extension will be selected for spying." |
| 21:11.51 | Katty | glares at p3nguin |
| 21:12.00 | Katty | you are not 4 anymore |
| 21:12.10 | kaldemar | p3nguin: you give the app an extension, not a channel. |
| 21:12.12 | p3nguin | What determines if a channel is created by an OUTGOING CALL? |
| 21:12.22 | Pan3D | heh |
| 21:12.23 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 21:12.24 | citywok | Pan3D: because it's all on new lines, you end up removing pretty much all of the detail though |
| 21:12.26 | elliot98 | the destination server is also a *? |
| 21:12.28 | [TK]D-Fender | p3nguin: A call TO an extension in your dialplan. |
| 21:12.31 | Pan3D | citywok: learn2grep |
| 21:12.36 | Pan3D | -A -B |
| 21:12.37 | [TK]D-Fender | p3nguin: doesn't amtter what the exten bridges to if anything |
| 21:12.41 | [TK]D-Fender | matter* |
| 21:12.54 | elliot98 | popO; did you check the logs of both servers? |
| 21:13.34 | p3nguin | So a call from SIP/201 to exten 211 (which subsequently dials sip/211) would be an outgoing call TO 211, yes? |
| 21:13.38 | thepopolinux | elliot98 > yes, no error message, both servers are Asterisk 1.6 |
| 21:13.58 | [TK]D-Fender | p3nguin: better |
| 21:14.32 | citywok | Pan3D: do you use something like 10 +/-? |
| 21:14.37 | [TK]D-Fender | p3nguin: the fact that it calls SIP/211 is irrelevant. its the fact the dialplan extension is 211 |
| 21:14.46 | thepopolinux | elliot98 > when I register incoming sip trunks on the second server directly, it work well |
| 21:14.47 | citywok | (i was just reading the man page and saw those lines and figured thats what you were tlaking about) |
| 21:15.01 | p3nguin | I was just adding it to let you know what exten 211 did. I realize it's not a critical bit of info. |
| 21:15.01 | *** join/#asterisk Woody2143 (n=Woody214@machine76.Level3.com) |
| 21:15.41 | p3nguin | katty: And what the fuck was your remark supposed to me? You don't need to talk to me like a child. |
| 21:15.47 | p3nguin | mean |
| 21:16.00 | elliot98 | so the setup is: Sip Phone -> Sip Server -> IAX Trunk |
| 21:16.24 | elliot98 | wipes his screen, but the expletive is still there |
| 21:16.37 | Dabian | If I define a friend called "carrot" in sip.conf correctly, and connect a sip-phone to it. I guess I could reach that phone in extensions.conf by typing "Dial(SIP/carrot)" in the right place. |
| 21:17.08 | [TK]D-Fender | Katty: Of course he's not 4 any more. p3nguin > p4nguin |
| 21:17.11 | Pan3D | citywok: improvise. You (should) know what the general conversation between a server and client are. So, grep for the IP of the server/client, then use -A -B to narrow it down. |
| 21:17.36 | [TK]D-Fender | Dabian: Yes |
| 21:17.49 | *** join/#asterisk rpm (n=russell@S0106000c29898b7e.cg.shawcable.net) |
| 21:17.56 | elliot98 | citywok: also, do a quick logger reload |
| 21:17.56 | thehar | russellb: i just found your card you gave me at con! |
| 21:17.58 | thehar | tender |
| 21:17.58 | elliot98 | so the messages file will be shorter |
| 21:17.58 | thehar | awwwls |
| 21:18.06 | citywok | Pan3D: yea i'm finding better luck grepping for the phone number i dialed actually |
| 21:18.08 | thepopolinux | elliot98 > yes, i get incoming calls from a number with sip trunk. the server forward call on sip trunk to another server. the destination server play sound with Playback or MOH and the sound cut after few seconds |
| 21:18.10 | russellb | thehar: w00t! |
| 21:18.14 | Pan3D | citywok: whatever works! |
| 21:18.15 | russellb | thehar: I have your business card at my desk! |
| 21:18.32 | elliot98 | popo: do IVR's work? |
| 21:18.35 | thehar | russellb: you too! i have you. mark. aaron. and allison all taped to my 21" lcd |
| 21:18.39 | Dabian | Asterisk is sweet. I guess maybe my problem was wrong file permissions. |
| 21:18.47 | elliot98 | popo: and live phone calls? |
| 21:18.59 | citywok | hmmmm SIP/2.0 500 Server Internal ErrorRecord-Route |
| 21:19.12 | Dabian | I am thankful for your help, I was getting real stressed. |
| 21:19.13 | rpm | anyone know which codec i should be using for g729 (codec_g729a.so) on freebsd 7.2 (a vps of mine)? i downloaded the "codec_g729a-1.4_3.1.3-core2_64.tar.gz" but it doesn't seem to load.. |
| 21:19.33 | rpm | i don't really want to register my licenses until i know it will work on here. |
| 21:19.38 | thepopolinux | elliot98 > yes, the extension work because when i register number directly on destination server, is ok |
| 21:19.39 | citywok | [TK]D-Fender: same problem exists in 1.6.1.10, appears to be a new bug |
| 21:19.46 | citywok | problem does not exist in 1.6.1.9 |
| 21:19.58 | elliot98 | popo: I mean if there is an IVR on the IAX trunk |
| 21:20.00 | [TK]D-Fender | ok, checkout time... back later |
| 21:20.07 | Dabian | Weee .. a new bug! :) |
| 21:20.07 | Qwell | rpm: You'll have to get one of the (old, unsupported) FreeBSD versions |
| 21:20.08 | Pan3D | run! |
| 21:20.15 | elliot98 | do the DTMFs give any response |
| 21:20.33 | elliot98 | even if the sounds stops, can you still press a button |
| 21:20.33 | Pan3D | that guy is amazingly cool for all the questions he has to field. |
| 21:20.35 | elliot98 | ? |
| 21:21.09 | thepopolinux | elliot98 > ok, I've not tested, I test now, please wait (sorry for my very bad english !) |
| 21:21.49 | thehar | russellb: i never got a card from corydon :( |
| 21:21.50 | elliot98 | ok |
| 21:21.55 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 21:21.57 | muiro | Pan3D: agreed |
| 21:21.58 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) [NETSPLIT VICTIM] |
| 21:22.58 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 21:24.18 | *** join/#asterisk fnordus (n=dnall@70.70.0.215) |
| 21:25.04 | thepopolinux | elliot98 > when I execute my IVR with multi-choise, it block on first Playback and it not go to next instruction |
| 21:25.18 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 21:28.53 | *** join/#asterisk knobo (n=user@ti100710a080-1351.bb.online.no) |
| 21:29.19 | *** part/#asterisk knobo (n=user@ti100710a080-1351.bb.online.no) |
| 21:29.30 | *** join/#asterisk evil_gordita (n=evilgord@ip70-188-50-186.rn.hr.cox.net) |
| 21:32.19 | thepopolinux | elliot98 > During my tests, a call is made correctly but the sound sacadas, it was possible to navigate through the IVR with DTMF. Without changing anything, now it's cut after a few seconds and the running of dialplan be stopped. |
| 21:33.16 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:34.28 | hardwire | I have a client with a channel bank connected via CAS T1 to a 4 port digium card. |
| 21:34.50 | hardwire | They have implemented a physical toggle switch thinger that moves POTS from one location off the channel bank, onto the channel bank. |
| 21:34.54 | hardwire | between two systems basically |
| 21:35.02 | hardwire | I think they toggle it often |
| 21:35.09 | rpm | Does the "codec_g729a_v33_nocona" supply the CLI command "g729 show licenses" ? I have it loaded, my 10 licenses are registered but I am not able to verify they've been read. |
| 21:35.19 | hardwire | and asterisk freaks out often. |
| 21:36.04 | hardwire | I'm wondering how exactly lots of toggling which phones are connected to a channel bank would error the T1 link between Asterisk and the bank enough to cause issues. |
| 21:36.40 | hardwire | doesn't matter what version of asterisk/zap/dahdi I use. |
| 21:36.40 | hardwire | :( |
| 21:39.43 | thepopolinux | elliot98 > no idea ? |
| 21:39.53 | Qwell | rpm: no |
| 21:41.41 | *** join/#asterisk DerkKo (n=afernand@75-149-178-131-Miami.hfc.comcastbusiness.net) |
| 21:42.07 | DerkKo | Hey guys... Im wondering if i could modify the from headers on the invites asterisk sends out |
| 21:43.09 | muiro | thepopolinux: can we see any code samples/logs? use pastebin please |
| 21:43.22 | DerkKo | IE From: <sip:15551212@10.20.2.129:5060>;Information-I-Added;bla234234=12341 |
| 21:43.53 | thepopolinux | muiro > thank you for your help! I do that now |
| 21:46.37 | muiro | I have a soft spot in my heart for people newer to Asterisk coding IVR's. â« Memories â« |
| 21:48.02 | thepopolinux | muiro > http://pastebin.com/d3e6cd18c |
| 21:48.41 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:49.12 | thepopolinux | muiro > incoming calls sip trunk is register on Zurix. Zurix forward calls trought IAX trunk to Idfix. |
| 21:51.39 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
| 21:52.43 | muiro | thepopolinux: I'm opening an msg to you |
| 21:58.49 | elliot98 | did you try changing the timeout? |
| 22:02.10 | elliot98 | also, why is there a ringing command? |
| 22:03.16 | thepopolinux | elliot98 > you talk to me ? |
| 22:03.39 | elliot98 | yes |
| 22:03.47 | elliot98 | did you solve it yet? |
| 22:05.06 | thepopolinux | elliot98 > muiro speak to me in private, I have a new pastebin : http://pastebin.com/d5a8eff8b |
| 22:06.24 | *** part/#asterisk DerkKo (n=afernand@75-149-178-131-Miami.hfc.comcastbusiness.net) |
| 22:07.39 | *** join/#asterisk evil_gordita (n=evilgord@ip70-188-50-186.rn.hr.cox.net) |
| 22:12.49 | Katty | allo |
| 22:14.03 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 22:16.13 | beek | 'allo Katty |
| 22:16.43 | Katty | hugs beek |
| 22:16.50 | beek | smiles |
| 22:31.52 | *** join/#asterisk nny (n=scott@64.203.239.83) |
| 22:32.27 | nny | anyone here know if the cisco 50X (504g for example) SPA phones can be setup to auto answer (for office paging)? |
| 22:34.10 | nny | hmm |
| 22:34.16 | nny | Auto Answer Page:Â |
| 22:34.29 | nny | in config UI, so I assume I just need to craft the right bit for it to AA |
| 22:34.30 | nny | cool |
| 22:37.03 | dlynes | rpm: are you running in linux compatibility mode? |
| 22:38.44 | p3nguin | nny: http://www.voip-info.org/wiki/view/Asterisk+cmd+Page There's a section on SPA- phones. |
| 22:38.47 | dlynes | Dabian: so did you ever figure out how to pastebin? |
| 22:38.51 | saxa | joako: the * version are same on both ends, now probably my sip.conf has still some problems since I did it just quickly and dirty way |
| 22:39.08 | nny | p3nguin: my hero, thanks |
| 22:40.39 | hobodave | hey guys |
| 22:40.55 | hobodave | how many concurrent connections does Asterisk community version support? |
| 22:41.45 | Qwell | hobodave: there is no limit |
| 22:41.51 | hobodave | weird |
| 22:41.57 | Qwell | need more channels? throw more hardware at it |
| 22:42.01 | hobodave | why is ABE sold with limits then? |
| 22:42.05 | lesouvage | hobodave: it is not asterisk that wil restrict your concurrent calls (within reasonable range) but the hardware used, the codecs used and the bandwidth available and the features used. |
| 22:42.21 | Qwell | hobodave: because that's how it's sold |
| 22:42.25 | hobodave | gotcha |
| 22:42.54 | elliot98 | popo: was the issue resolved? |
| 22:42.59 | Qwell | Asterisk is Open Source.. even if we *did* add a limit to the code... |
| 22:43.02 | hobodave | so we have a 20 connection copy of ABE that has expired, is it reasonable to think that the community version of Asterisk will support at least the same 20 connections, given teh same hardware? |
| 22:43.05 | lesouvage | hobodave: that has to do with businesmodels. |
| 22:43.08 | hobodave | Qwell: good point |
| 22:43.13 | Qwell | hobodave: yes |
| 22:44.14 | [TK]D-Fender | hobodave: they were hardly ever really different |
| 22:45.31 | lesouvage | hobodave: there is a good change that you cn do more concurrent calls bacause lots of AGI scripts used might not be that cpu and resource efficient. |
| 22:45.34 | *** join/#asterisk Corydon76-dig (i=three@c-69-137-80-31.hsd1.tn.comcast.net) |
| 22:45.37 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 22:45.44 | Qwell | Corydon76-dig: YOU |
| 22:46.33 | Qwell | Corydon76-dig: I do believe you left a jacket downstairs. I guessed it was yours, and put it in your office. |
| 22:47.51 | hobodave | thanks for the help guys |
| 22:48.49 | Qwell | hobodave: also, if you're not familiar - http://www.digium.com/en/supportcenter/asterisk.php |
| 22:49.09 | hobodave | yea I saw that :) |
| 22:49.20 | hobodave | we're actually going to run 1.2 I think though, since that's the latest provided by Gentoo |
| 22:49.26 | Qwell | umm |
| 22:49.26 | Qwell | no |
| 22:49.28 | hobodave | and yes I know it's wtf crazy to run Gentoo in production |
| 22:49.31 | Qwell | Switch distros. |
| 22:49.34 | [TK]D-Fender | hobodave: 1.2 = ANCIENT |
| 22:49.40 | hobodave | not an option :( |
| 22:49.50 | hobodave | I didn't make this decision |
| 22:49.53 | Qwell | Gentoo is supposed to be a bleeding edge distro, yet has a 3 year old version of Asterisk |
| 22:49.55 | *** join/#asterisk KingDavidNYC (n=Chris123@66.7.86.2) |
| 22:50.02 | hobodave | lol |
| 22:50.12 | [TK]D-Fender | Qwell: YES, THERE WILL BE BLOOD |
| 22:50.12 | KingDavidNYC | Hello Everybody!!!! |
| 22:51.15 | KingDavidNYC | I am very well, thanks |
| 22:51.29 | Qwell | hobodave: no, but seriously |
| 22:51.31 | Qwell | don't use 1.2 |
| 22:51.42 | hobodave | Qwell: can you provide specifics? |
| 22:52.11 | hobodave | I'm willing to argue it to my people, but we're reluctant to not use Portage for our package maintenance |
| 22:52.14 | Qwell | not much to be specific about. just don't use 1.2 |
| 22:52.24 | Qwell | portage = crap. 1.2 = old and *unsupported* |
| 22:52.49 | Qwell | You'd be going from a tested/supported build to...that. |
| 22:52.58 | tzafrir | hobodave, what stops Gentoo from using a newer version of Asterisk? |
| 22:53.08 | hobodave | well, actually I'm looking in portage now |
| 22:53.09 | Qwell | tzafrir: skill |
| 22:53.15 | hobodave | they have 1.6.1.10 |
| 22:53.18 | hobodave | but it's masked |
| 22:53.24 | hobodave | meaning, you have to explicitly unmask it to install it |
| 22:53.43 | Qwell | or just install it yourself |
| 22:53.44 | hobodave | which means either 1. it's not stable , or 2. everyone is too lazy to test it |
| 22:53.53 | Qwell | it's 2. |
| 22:54.02 | Qwell | well, 1 too, heh |
| 22:54.03 | *** join/#asterisk stupidnic (n=foo@cpe-70-94-229-122.sw.res.rr.com) |
| 22:54.20 | tzafrir | Qwell, they have dahdi. They had 4 patches, 3 were related to a newer kernel (fixed in SVN) , last one was also quite justified |
| 22:54.48 | Qwell | hobodave: There are a lot of open source projects that will yell at you (very loudly) if you ask for help using a gentoo build. Pidgin is a good example of this. They'll /kick you from the channel if you say the word. |
| 22:55.06 | hobodave | Qwell: I'm very aware of this |
| 22:55.19 | hobodave | I like gentoo for messing around |
| 22:55.25 | Qwell | hobodave: On what distro were you running BE? |
| 22:55.27 | hobodave | but I would never have made the decision to put it on our production servers |
| 22:55.33 | hobodave | on Gentoo |
| 22:55.34 | Qwell | because it sure as heck wasn't supported on Gentoo... |
| 22:55.47 | hobodave | it is |
| 22:55.51 | hobodave | the "install" isn't |
| 22:56.05 | hobodave | but as ABE is precompiled, they'll support it post-install |
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| 22:56.45 | flyback | bids on a licenced copy of office 2003 pro for his mom, (don't even mention star office, it's not going to happen) |
| 22:56.46 | Qwell | rolls his eyes in the direction of the third floor |
| 22:56.52 | tzafrir | Qwell, actually Gentoo provides a nice smoke-testing lab to see which build tweaks break Asterisk |
| 22:58.52 | Qwell | glares at flyback |
| 22:59.01 | flyback | yes? |
| 22:59.03 | hobodave | is there a way to know what version of community ABE was built with? |
| 22:59.14 | flyback | that was a global message so just ignore it here |
| 22:59.26 | Qwell | flyback: Turn them off. |
| 22:59.35 | Qwell | hobodave: I don't know what you mean |
| 22:59.46 | hobodave | Qwell: ABE versions things differently |
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| 22:59.53 | Qwell | very much so, yes |
| 22:59.56 | hobodave | so I dont know if our version was built from 1.4, 1.6 or anything |
| 23:00.02 | Qwell | What version of BE? |
| 23:00.17 | hobodave | ABE-C.3.2.2-rpms-x86_64 |
| 23:00.20 | flyback | or what? you will harass me in #linuxhelp? |
| 23:00.20 | flyback | :P |
| 23:00.23 | Qwell | Based on 1.4. |
| 23:00.28 | *** join/#asterisk cnu (i=cnu@113.80-203-44.nextgentel.com) |
| 23:00.31 | *** kick/#asterisk [flyback!i=north@pdpc/sponsor/digium/Qwell] by Qwell (or this) |
| 23:00.38 | hobodave | should our scripts that work on 1.4 work on 1.6? |
| 23:00.48 | Qwell | hobodave: UPGRADE.txt in the 1.6 source |
| 23:00.55 | hobodave | k |
| 23:01.27 | Qwell | why do I keep running into that kid? *mutter* |
| 23:01.47 | hobodave | thanks for the help Qwell |
| 23:01.53 | Qwell | hobodave: yep |
| 23:02.09 | hobodave | my company is so cheap |
| 23:02.14 | thepopolinux | Good bye all, thank you for your help |
| 23:02.19 | *** part/#asterisk thepopolinux (n=cyril@AToulouse-552-1-73-215.w92-136.abo.wanadoo.fr) |
| 23:02.23 | hobodave | I wish they'd just buy the stupid copies of ABE and be done with it |
| 23:02.48 | hobodave | I told my boss that if something goes wrong with Asterisk, they're going to get a big NFC from me |
| 23:02.50 | Qwell | hobodave: I think everything is moving in the open source subscriptions direction |
| 23:02.52 | hobodave | and our systems guys said the same |
| 23:03.13 | tzafrir | hmmm... they have a stupid patch to fix the local copy of libgsm . Instead of just using a system one |
| 23:03.43 | Qwell | tzafrir: I'm not really surprised |
| 23:04.02 | tzafrir | There's a patch "add missing LIBS for uclibc" |
| 23:04.20 | tzafrir | And then there are a number of bug-fixes for some open bugs |
| 23:04.40 | Qwell | with no comments from them saying they tested well? |
| 23:04.53 | tzafrir | With references to the bug reports |
| 23:04.59 | Qwell | I mean on the bug reports |
| 23:05.23 | tzafrir | https://issues.asterisk.org/view.php?id=14163 is "ready for review" |
| 23:05.54 | Qwell | Chainsaw |
| 23:06.07 | Qwell | that's good |
| 23:06.49 | tzafrir | All in all not many patches |
| 23:07.01 | Qwell | more than Debian though, right? |
| 23:07.23 | tzafrir | nah. The debian one now has more |
| 23:07.53 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:544e:fd77:dc84:5e4a) |
| 23:07.58 | Qwell | tzafrir: somebody should have a talk with the maintainer :p |
| 23:08.05 | Qwell | </troll> |
| 23:08.46 | Qwell | I wish Gentoo would hurry up and unmask it though.. it's been there forever. |
| 23:09.47 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:09.49 | russellb | i wish people would stop using gentoo |
| 23:09.55 | russellb | doesn't duck. |
| 23:10.39 | [TK]D-Fender | gooses |
| 23:12.50 | hobodave | I'm talking to our systems guys |
| 23:12.56 | hobodave | apparently 90% of their systems are gentoo |
| 23:13.02 | hobodave | sooo bizarre |
| 23:13.08 | hobodave | the other 10% CentOS |
| 23:13.10 | toresbe | wow, that's fucked up |
| 23:13.14 | hobodave | why the two extremes? |
| 23:13.17 | hobodave | sigh |
| 23:13.29 | toresbe | Hm, perhaps incompetence? |
| 23:13.35 | toresbe | Stop hiring 14-year-olds. |
| 23:13.42 | hobodave | heh |
| 23:14.43 | hobodave | sigh Centos 5.4 just came out, and it's still on PHP 5.1 |
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| 23:16.45 | hobodave | I suggested Ubuntu |
| 23:16.58 | hobodave | but we'd be the only Ubuntu installation they would support |
| 23:17.04 | toresbe | not a bad choice... I prefer Debian, though |
| 23:17.20 | hobodave | I have no experience with Debian, I just know that Ubuntu is debian based |
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| 23:20.06 | Dabian | russelllb: Whats wrong with Gentoo, in your opinion? |
| 23:21.03 | Qwell | Dabian: portage |
| 23:21.08 | Qwell | all of it |
| 23:21.13 | hobodave | meh |
| 23:21.17 | hobodave | I love portage and gentoo |
| 23:21.25 | hobodave | but I just think it's a bad choice for our production servers |
| 23:21.30 | Dabian | What is portage? Another dist? |
| 23:21.32 | hobodave | all 3 of our servers are WIDELY out of sync |
| 23:21.37 | hobodave | portage is a "package manager" |
| 23:21.46 | hobodave | where packages are source tarballs |
| 23:22.01 | Dabian | Replacement for "emerge" or is emerge a tool of the manager? |
| 23:22.08 | hobodave | manages dependencies, and has a clever system called USE flags that control what gets compiled into your stuff |
| 23:22.11 | hobodave | emerge is a tool of portage |
| 23:22.20 | Dabian | ok |
| 23:22.44 | Dabian | I considered trying Gentoo more than once, because of the supposed cleverness of "overlays" |
| 23:23.01 | hobodave | so we're undertaking this huge project to synchronize the systems of all our production servers |
| 23:23.07 | hobodave | kernel versions, software versions, etc. |
| 23:23.22 | hobodave | and then have regular upgrade reviews every quarter, and mandatory every year |
| 23:23.41 | hobodave | and we're talking about compiling binaries on only one server, and then distributing the binaries to the remainder of the systems |
| 23:23.41 | Dabian | Sometimes I make a stupid change in some program, and either nobody else would every like it, or maybe I am just too lazy to post a patch, and then I upgrade, and have to do it all again. |
| 23:23.43 | beek | I considered using Gentoo, as well. Then I realized that statistically I only have twenty-five years remaining life span and I don't want to waste it compiling my whole #$%&ing OS and apps. |
| 23:23.53 | hobodave | it's like we're implementing our own $%^ing CentOS |
| 23:24.09 | Dabian | beek: You hit 50? |
| 23:24.22 | beek | just about, one year to go. |
| 23:24.33 | Dabian | beek: Enjoy it! :) |
| 23:24.51 | Dabian | beek: You're not old yet! |
| 23:24.57 | Dabian | :) |
| 23:25.10 | toresbe | Dabian: portage is wget with duct tape :) |
| 23:25.23 | beek | The warranty on my body expired when I turned 40. Anything I get now is just gravy. |
| 23:25.27 | Dabian | hobo: Sure sounds like trouble. |
| 23:25.44 | Dabian | beek: I *love* gravy. :) |
| 23:25.50 | toresbe | hobodave: Debian does all of that really cleverly, though - distributed installations are a snap with FAM |
| 23:25.54 | toresbe | FAI* |
| 23:26.26 | Dabian | toresbe: A combination of debian and gentoo would be great. apt-get source doesn't quite do the trick, even though its close. |
| 23:26.39 | toresbe | you don't need to use apt-get source |
| 23:26.58 | Dabian | toresbe: No, but you can, which is nice. |
| 23:27.02 | toresbe | sure |
| 23:27.23 | toresbe | just get the source - from SVN if you want - make the Debian package information, then invoke the package builder and voila deb package |
| 23:27.43 | Dabian | Yeah .. but thats when you want the new version. |
| 23:28.14 | Dabian | And then you have to get (some) knowledge of the debian package system. |
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| 23:29.35 | tzafrir | toresbe, Debian has similar tools |
| 23:29.47 | lesouvage | Dabian: i have a debian asterisk howto that I use for myself when I have to install an asterisk system. It might not be perfect but it is wrking if you follow all the steps. |
| 23:29.48 | tzafrir | svn-buildpackage and such |
| 23:30.12 | toresbe | I'm talking about Deb now :) |
| 23:30.33 | lesouvage | Dabian: if you are interested I can pastebin it somewhere. |
| 23:31.28 | Dabian | Please do! My system works fine now, but later I might wanna setup a system for production, and possibly with a newer version of Asterisk, than I have now. |
| 23:34.03 | lesouvage | Dabian: there are some lines in Dutch, but it is basically just execute the command lines you see in the text. |
| 23:39.17 | Dabian | lesouvage: I don't have a working knowledge of Dutch, but ignoring them would be feasable. |
| 23:42.08 | lesouvage | Dabian: I'm cleaning the instruction. Just wait a moment. |
| 23:45.37 | lesouvage | Dabian: see http://www.pastebin.be/22047 I started it once based on the info on Asterisk guru ad it has been very helpful for me. |
| 23:47.14 | Dabian | lesouvage: Thanks! |
| 23:47.23 | Godfather_ | i'm compiling * 1.6.1 and i'm getting this error, http://pastebin.com/m57278c8f |
| 23:50.47 | Dabian | lesouvage: Looks very useful! What license, if any, are you distributing this text with? |
| 23:53.33 | lesouvage | Dabian: I haven't thought about it. If we ever met a beer would be perfect :-) |
| 23:54.27 | Dabian | :-) |
| 23:55.21 | lesouvage | Dabian: it is more or less what is on the asterisk guru site with some adjustments based on my own experience during the install process. |
| 23:55.46 | Dabian | lesouvage: O |
| 23:55.48 | Dabian | lesouvage: OK |
| 23:57.06 | Dabian | lesouvage: I am just thinking that it could help a lot of people running Debian getting started. |
| 23:59.44 | Dabian | Step-by-step quicksters like this, is really great, and I think many people look for stuff like this. |
