00:01.03 | *** join/#asterisk luke-jr (n=luke-jr@2002:1891:f663:0:20e:a6ff:fec4:4e5d) |
00:05.05 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
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00:12.34 | [hC] | Is it possible for queue agents to continue to be run while an announcement is being played? |
00:12.44 | [hC] | i get the impression that while an announcement is being played, no agents are being rang. |
00:13.56 | *** join/#asterisk tonyb2006 (n=tonyb@2002:4571:29c2:0:0:0:0:1) |
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00:27.34 | LennonNZ | Hello. has anyone got asterisk to peer with a Siemens Hiq8000 properly? |
00:27.37 | LennonNZ | I am having troubles |
00:28.03 | LennonNZ | it registers ok, but I can only call from hiq->asterisk. but asterisk->hiq gives wrong password error |
00:35.25 | LennonNZ | ah. worked it out.. |
00:37.10 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
00:37.17 | saint_ | hi all |
00:37.35 | zoa | hi |
00:38.01 | saint_ | so i am following the doc, and I am trying the simple SIP trunk.. with no success... I just configured the sip.conf , and should I expect asterisk to work ? or should i do something else ? |
00:42.06 | *** join/#asterisk wubba (n=kmurrey@cable-76-215.sssnet.com) |
00:42.11 | zoa | you need to alter extensions.conf too |
00:42.49 | wubba | Is this discussion here just about develpment of * or do you all talk about real world installations? |
00:43.05 | zoa | this channel is more like a support channel |
00:43.11 | zoa | there is different channel for development |
00:43.25 | wubba | ok |
00:43.43 | saint_ | zoa, i m doing a real world install |
00:43.48 | saint_ | asterisk <--> Alcatel |
00:43.51 | _DAW | Anyone interested in helping test some SIP DID's? I need some regular, reliable, intelligent feedback on voice quality and am willing to give away 10 for 6 months. Can provide number to most of the US. |
00:44.11 | wubba | _daw - I could help you |
00:44.23 | _DAW | message me off channel if your interested. |
00:44.26 | wubba | Saint - tell me about your install. |
00:44.27 | _DAW | please |
00:44.53 | wubba | sent |
00:45.51 | saint_ | wubba, I have a company with a Alcatel PBX in the US |
00:45.58 | saint_ | they have clients all over the world who use sip |
00:46.02 | saint_ | inside it works |
00:46.08 | saint_ | but the alcatel sip can't NAT |
00:46.12 | wubba | So your using * |
00:46.30 | saint_ | so I am setting up a Asterisk box which is able to NAT sip clients, and I'll try to use a SIP trunk between it and alcatel |
00:46.50 | wubba | interesting |
00:47.04 | saint_ | _DAW, are you saying that you are providing a SIP trunk ? |
00:47.14 | saint_ | wubba, yeah, if it works, I'll push asterisk more to my clients, lol |
00:47.19 | saint_ | I have a site with 14,000 extensions |
00:47.23 | wubba | We have been messing around with the Trixbox/Freepbx. We recently installed the Asterisk Now and it looks very good. |
00:47.45 | saint_ | Sweet.. |
00:47.46 | _DAW | saint_ inbound.. yes. |
00:47.58 | saint_ | Alcatel came up with 7.0 and SIP trunks, but it looks like none in the US tried it yet |
00:48.00 | wubba | We sell key systems now - but have started to run into some great opportunities for some * boxes. |
00:48.11 | saint_ | I did a 7.1 install at AT&T and they were trying sip trunking and gateway in their lab |
00:48.23 | saint_ | wubba, really ? |
00:48.28 | saint_ | _DAW, what do you need ? |
00:48.35 | saint_ | wubba, like what ? |
00:49.29 | wubba | We have a school system that has about 30 sites throughout our area. They are looking to tie two of them together, and them move the rest later on. |
00:49.45 | saint_ | cool... is it in the US ? |
00:49.49 | wubba | yes |
00:50.04 | saint_ | what school district ? |
00:50.21 | wubba | It's not really a 'district' - its a company that does Charter Schools. |
00:50.28 | saint_ | oh, ok .. |
00:50.52 | wubba | But they have 30 sites now and are expanding |
00:50.55 | saint_ | I'm just starting with Asterisk (a couple of hours ago.) It looks pretty neat |
00:51.08 | saint_ | how are they connected together ? |
00:51.32 | wubba | They are not now - they are all on key systems. |
00:51.37 | saint_ | oh, ok |
00:51.43 | saint_ | have them switch to Alcatel - Lucent :-D |
00:52.03 | wubba | We are going to put a data T-1 in and do sip trunks with DID to each location. |
00:52.38 | saint_ | then how do you connect the key system to asterisk ? |
00:53.09 | wubba | We are replacing the key systems with * |
00:53.15 | saint_ | oh ! |
00:53.32 | saint_ | I thought key system <-> * <---- SIP ---> * <-> Key system |
00:53.41 | saint_ | so what kind of phones are you going to give to the clients ? |
00:53.45 | wubba | Polycom |
00:53.48 | saint_ | SIP ? IP ? Digitals ? |
00:53.54 | saint_ | which model ? |
00:53.55 | wubba | All SIP |
00:53.59 | wubba | 330 |
00:54.50 | saint_ | let me check on line |
00:54.57 | *** join/#asterisk Fieldy (i=Yi5dPltt@gentoo/contributor/Fieldy) |
00:55.00 | saint_ | soundpoint ip 330 ? |
00:55.54 | wubba | yes |
00:56.23 | bsd_tech | wubba wubba |
00:56.49 | saint_ | wubba, interesting... |
00:56.55 | wubba | hey bsd |
00:57.02 | saint_ | the good feature with alcatel, is that they have a keyboard on the screen, and you can do dial by name |
00:57.20 | saint_ | the latest ip touch phone has a color screen too , where you can feed video |
00:57.26 | saint_ | like a doorcam .. |
00:57.37 | wubba | interesting |
00:57.49 | saint_ | yeah.. you can also program neat XML applications for it |
00:57.50 | bsd_tech | ? |
00:58.17 | saint_ | send/receive SMS, check the price of a stock, weather, other shit like that |
00:58.33 | saint_ | whatever -almost- you can do in xml, you can do it on this phone |
00:58.35 | saint_ | it s pretty neat |
00:58.49 | bsd_tech | url? |
00:58.55 | saint_ | hold on |
00:59.20 | bsd_tech | no now |
00:59.24 | bsd_tech | now now now |
00:59.27 | bsd_tech | now |
00:59.31 | bsd_tech | lol |
00:59.38 | saint_ | http://www1.alcatel-lucent.com/enterprise/en/solutions/mobility/on_site_mobility/iptouch_phones.html;jsessionid=HDX0W4OXDQLVVLAWFRUE1DNMCYWGI3GC |
01:00.16 | saint_ | i have one on my desk right now :-D |
01:00.31 | saint_ | + they have bluetooth.. sswwwweeeeeeeettt |
01:01.10 | bsd_tech | how much |
01:01.17 | saint_ | for you ? |
01:01.23 | saint_ | or as if you were my customer ? |
01:01.33 | wubba | cost/retail |
01:01.35 | bsd_tech | wich is cheaper |
01:01.36 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
01:02.09 | bsd_tech | and the page does not display correctly |
01:02.18 | saint_ | about 280 for my to buy, about 580 me to sell to you ... anywhere in between |
01:02.38 | saint_ | but this is the top .. you have other IP phones with black & white screen |
01:02.41 | saint_ | instead of the color |
01:02.47 | saint_ | iptouch 4038 |
01:02.48 | Innatech | keerist. That's spendy. |
01:03.09 | saint_ | yeah.... Alcatel-Lucent dude :-) |
01:03.21 | wubba | The phones are quite hideous looking. |
01:03.58 | saint_ | nnnnaaaaa |
01:04.02 | saint_ | better than ciscoc |
01:04.04 | saint_ | cisco |
01:04.04 | bsd_tech | that 200% markup on price |
01:04.07 | bsd_tech | ouch |
01:04.10 | saint_ | yup |
01:04.19 | saint_ | but we give 35% to most of customers |
01:04.31 | wubba | so your work for them? |
01:04.44 | saint_ | used to. i work for a business partner now |
01:05.06 | bsd_tech | donate one for review. so I can write a article on it |
01:05.13 | saint_ | hehehe |
01:05.16 | bsd_tech | but I need 30 days of testing |
01:05.23 | wubba | I'd buy for $580 - then I have to put my markup on it - man that's expensive. |
01:05.25 | Qwell | donate one for review. so I can burn it in effigy |
01:05.27 | saint_ | you won't be able to test it without a PBX ! |
01:05.35 | Innatech | None of my clients would want all that, anyway. They just want the same functions that they're used to from various Nortel PBXes. |
01:05.39 | wubba | bsd has about 20 of them |
01:05.56 | bsd_tech | I have a asterisk pbx |
01:06.19 | saint_ | Innatech, because they never played with it |
01:06.30 | saint_ | I went into a hospital where they changed from nortel to cisco (ip) |
01:06.46 | Innatech | saint: very true, at least for some of them. But I service a lot of technophobes too. |
01:06.47 | saint_ | I came and left a demo AFTER the cisco was installed... 6 months after, the removed the cisco to install Alcatel .. |
01:06.54 | bsd_tech | they should have saved thier money and gone polycom |
01:07.04 | saint_ | I'm telling you.. gotta play with it, and check all the functionalities of the system .. it's a killer system .. running on linux ! |
01:07.16 | Qwell | They should have saved their money and gone with asterisk, and reused the ciscos |
01:07.29 | saint_ | yeah, could of.. but then... do you have dial by name ? |
01:07.38 | Qwell | If I write it, sure |
01:07.40 | bsd_tech | asterisk is not ready for hospitals |
01:07.49 | saint_ | it sounds stupid, but in a hospital connected to other hospital, where you have over 4,000 people ... dial by name suddently makes sense to have .. |
01:07.51 | bsd_tech | still work to be done |
01:08.28 | Innatech | Yeah....again, my clients needs may not be the most complicated (definitely not hospitals) and for them Asterisk is just perfect. |
01:08.45 | wubba | Innatech - what is your average install in # of phones? |
01:08.53 | *** join/#asterisk piper69 (n=piper@unaffiliated/piper69) |
01:09.07 | piper69 | hi all |
01:09.11 | bsd_tech | well I would like to get one of the phones to write a review |
01:09.12 | Innatech | hmm----probably 8 to 20. |
01:09.19 | bsd_tech | get them to donate one |
01:09.38 | bsd_tech | wont happen thou |
01:09.39 | piper69 | i got offered a 5ESS DCS switch engineer today |
01:09.52 | piper69 | am very excited |
01:09.58 | Innatech | Then I show up and add a few every so often. But they tend be in that range when I go out to install or convert. |
01:10.30 | wubba | [innatech] We are getting ready to really start pushing some of these on our current installs. Are you using SIP? |
01:10.39 | saint_ | bsd_tech, an alcatel phone ? |
01:10.51 | Innatech | wubba - yes, primarily SIP. |
01:11.11 | wubba | [innatech] So are you replacing Key Systems? |
01:11.20 | wubba | [innatech] - What phones are you using? |
01:11.36 | Innatech | wubba - I'm either replacing key systems or setting up new small businesses. |
01:11.46 | wubba | [innatech] Excellent. |
01:12.00 | Innatech | wubba - I've had good luck with the Snoms, but I know some find them buggy or lacking in features. |
01:12.22 | wubba | [innatech] So what are you using as far as the * box - you just building something? |
01:12.28 | piper69 | hopefully one day i will catch up with you guys on Asterisk |
01:12.36 | Innatech | wubba - A lot of small shops without much startup money can also be serviced by using ATAs with good old analog desktop speakerphones. This often provides better sound than a dirt cheap IP phone. |
01:12.53 | Innatech | wubba - I build a box for it when I can get it approved. |
01:13.02 | wubba | k |
01:13.10 | wubba | [innatech] Hope you don't mind the questions... |
01:13.17 | Innatech | wubba - otherwise we repurpose a WinXP workstation by adding a RAID card and installing CentOS or Ubuntu LTS> |
01:13.28 | Innatech | wubba - not at all. |
01:13.48 | wubba | [innatech] We have been installing mostly key systems - but I have had * running in our office for about 3 months now runnign with 5 SIP trunks. |
01:14.38 | wubba | [innatech] So I have had our sales people call on a bunch of folks that we quoted lasted year that didn't do anything. Hoping that we can move them over to a full * install. |
01:14.43 | Innatech | It's truly excellent for new/small businesses. People get very excited when they find out what they can get for next to nothing in captial investment. |
01:14.51 | Innatech | wubba - that's a very smart tactic. |
01:15.20 | wubba | [innatech] Especially since it was close to $300k in Quotes. |
01:15.41 | Innatech | wubba - very nice! |
01:15.43 | wubba | [innatech] So I have close to 100 or so quotes. |
01:16.02 | wubba | [innatech] How are you marketing? |
01:16.22 | Innatech | wubba - Heh. I wish I had leads like that. I don't do much marketing yet, mostly word of mouth in the professional communities I work with. |
01:16.35 | Innatech | wubba - marketing to attys and docs gets expensive fast. |
01:16.52 | wubba | [innatech] I paid for them... we belong to a service where we pay for leads - $29 a peice. |
01:17.13 | wubba | But when you sell one and make $2500 - it's well work ith |
01:17.18 | Innatech | wubba - What is your conversion rate like? If I may pry a little? |
01:17.32 | wubba | Terrible - |
01:17.36 | Innatech | hehe. |
01:17.53 | wubba | Like I said I have 100 that we are recalling - and I sold maybe 4. |
01:18.02 | Innatech | Yeah, I end up doing work for almost everyone who calls. But it does get sloooow at times. |
01:18.25 | wubba | Alot of people just kicking the tires |
01:18.32 | Innatech | I should get the marketing religion, but I'm hesitant to throw money at the problem. |
01:18.50 | wubba | It's actually just an investement. |
01:19.15 | wubba | You invest in it to make you more money. It's hard at times to spend the money - but it usually helps. |
01:19.23 | Innatech | It's true, I've heard that from everyone. I just have to make myself pick a venue and do it. |
01:19.39 | Innatech | AdWords seems like a waste, for instance. |
01:19.44 | Innatech | afk 1 min, brb. |
01:19.45 | wubba | Yes |
01:20.42 | cpm | who's selling what? |
01:22.30 | wubba | We are just talking about small * installs 8-20 endpoints |
01:22.30 | Qwell | "corruped"? |
01:25.22 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
01:25.44 | TripleFFFF | anyone know why if i dont use fromuser= i get non authoriezed.. but if i use it then i loose callerid |
01:30.56 | Innatech | (BAK.) Yes, * for small or new businesses. And how to market it effectively. |
01:33.12 | wubba | [innatech] - In the US? |
01:33.48 | Innatech | wubba - Yes. Los Angeles. |
01:34.18 | wubba | [innatech] OK |
01:34.47 | wubba | [innatech] How many installs have you done? |
01:35.26 | Innatech | wubba - counting everything, including tiny installs for friends, probably 15. |
01:35.43 | Innatech | wubba - installs worth talking about-- probably 5. |
01:35.57 | wubba | ok |
01:36.27 | Innatech | wubba - essentially one every couple months for the last year. |
01:36.31 | Innatech | brb |
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01:57.30 | Innatech | bak |
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02:10.41 | JT | Innatech: so most customers can only cough up for a second hand or retasked machine? |
02:11.17 | TripleFFFF | wow i do 15 a day |
02:11.18 | TripleFFFF | lol |
02:11.55 | JT | 15 what? |
02:12.03 | TripleFFFF | installs |
02:12.12 | JT | of what? |
02:12.15 | TripleFFFF | asterisk |
02:12.15 | JT | specs? |
02:12.16 | TripleFFFF | lol |
02:12.20 | JT | LOLLOlollllk |
02:12.20 | JT | k |
02:12.32 | TripleFFFF | was refering to mesage form innatech and wubba |
02:12.38 | JT | lol! |
02:13.18 | JT | and i asked for the specs, not more lols |
02:13.27 | JT | like how many stations |
02:13.32 | JT | what level of features |
02:14.40 | Innatech | Triple: When they're first getting started and have no capital? Yes, that's right. Some of the others can pay for more, and I sell them dual proc dual core rack servers. :) |
02:14.40 | Innatech | err, that should have been to JT. |
02:14.40 | JT | ah ok |
02:14.44 | Innatech | Triple: I'd love to do 15 a day. If that's possible. |
02:14.58 | JT | how cheap can you make an asterisk install though, Innatech ? |
02:15.00 | *** join/#asterisk techie (n=gus@voip.routedsystems.com) |
02:15.11 | Innatech | Practically free, save my time and cost to set up the trunks, etc. |
02:15.11 | JT | Innatech: must be 15 pretty simple installs |
02:15.19 | Innatech | JT: yeah, no kidding. |
02:15.53 | JT | Innatech: got to be some hardware cost unless you just give them VoIPoInternet and softphones |
02:16.30 | Innatech | Yeah--practically free. And it is VOIP over the WAN and usually cheap ATAs with existing analog deskphones. For the cheapest installs. |
02:17.09 | Innatech | So, the cost of the ATA plus (trunk costs / # of users) is the per seat cost, before they pay me. |
02:17.42 | Innatech | Of course, I advise them that they'll need to upgrade once cash flow happens. But its good for getting up off the ground with a new business. |
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02:22.54 | JT | hrm i would never give a business VoIPoI only |
02:23.14 | JT | doesn't matter how small they are, they're better off even with 1 analogue phone and phone line |
02:23.20 | *** join/#asterisk MrTelephone (n=DeaLER25@bas13-toronto63-1177850949.dsl.bell.ca) |
02:23.28 | JT | unless they don't actually need their phone to work |
02:23.38 | MrTelephone | does anyone ever have a pri just stop working and you have to reload the drivers and restart asterisk? |
02:23.57 | JT | MrTelephone: what card? |
02:24.04 | MrTelephone | sangoma dual port pri |
02:24.42 | MrTelephone | how often should asterisk restart the b channels? |
02:25.09 | MrTelephone | there was no alarms showing |
02:26.13 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
02:26.17 | Innatech | JT: Yeah, I give them the spiel about reliability, etc. Some of them care enough to spring for an actual phoneline and a TDM card, but most of them don't care. Ultimately its their decision. And most of them are good about upgrading once they are established and aren't worried about month-to-month survival. |
02:26.48 | JT | hmm ok |
02:27.08 | Innatech | *digium card that is/ |
02:27.25 | JT | sure |
02:28.10 | MrTelephone | hmmmm |
02:28.21 | MrTelephone | funny stuff happening |
02:28.23 | Innatech | When you think about it, Asterisk gives you amazing capabilities compared to what most businesses at that level are working with. |
02:28.35 | JT | Innatech: but how much can you possibly make off a project like that? |
02:28.42 | JT | Innatech: definitely |
02:29.14 | Innatech | JT: More than you'd think. Especially since I usually set up the rest of their computer network at the same time. |
02:29.29 | JT | hmm |
02:29.34 | JT | numbers? :) |
02:29.51 | Innatech | JT: mmm.....I'd prefer to be vague about my actual rates.... |
02:29.56 | MrTelephone | as long as you can keep your pri up |
02:30.02 | Innatech | JT: but, a few hundred, usually more. |
02:30.36 | Innatech | JT: If I'm putting everything in for them (server, workstations, phones etc) its usually over 1K. |
02:30.49 | JT | hmm not bad |
02:31.01 | Innatech | JT: Yeah. And my own overhead is *very* low. |
02:31.09 | MrTelephone | 1 thousand dollars |
02:31.12 | *** join/#asterisk Maghteridon (n=Maghteri@88-149-168-208.f5.ngi.it) |
02:31.14 | MrTelephone | shit im wiring a hotl for 9 thousand |
02:31.15 | Maghteridon | hi all |
02:31.31 | JT | MrTelephone: how many extensions? |
02:31.38 | MrTelephone | just runnign network cable |
02:31.40 | MrTelephone | no phones or anything |
02:31.45 | Innatech | MrT: If I had a CAT5 license, I could clean up on that stuff. I have to farm it out at the moment. |
02:31.46 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
02:31.52 | JT | how many extensions |
02:31.54 | MrTelephone | 13 extensions pc, polycom 501 phones was like 7 grand |
02:32.03 | Innatech | MrT: I'm not into dealing with insurance and building contractors, tho. |
02:32.04 | JT | ah only 13 |
02:32.16 | MrTelephone | you need a cat5 license in the states? |
02:32.20 | MrTelephone | ahhh |
02:32.23 | bsd_tech | where do you buy your phones MRT |
02:32.39 | MrTelephone | deny it all if something happens |
02:32.39 | MrTelephone | j/k |
02:32.46 | MrTelephone | ingrammicro |
02:32.52 | MrTelephone | wholesaler |
02:33.02 | TripleFFFF | ? |
02:33.03 | MrTelephone | the digium card was the most expensive thing |
02:33.05 | TripleFFFF | simple vanilla install |
02:33.06 | bsd_tech | http://ipphone-warehouse.com/ |
02:33.06 | TripleFFFF | yes |
02:33.09 | MrTelephone | 24 port linksys poe switch was 600 |
02:33.11 | bsd_tech | they rock |
02:33.12 | TripleFFFF | sorry was doing another ;) |
02:33.16 | Maghteridon | Let's say that you have at least 3 numbers on asterisk :x,y,z. Let's suppose X is calling Y; Y answers and then gives the communication to Z, so now X is talking with Z. How is it called this process that transfers the call once Y answered? |
02:33.25 | bsd_tech | I love polycom |
02:33.27 | MrTelephone | 200 for each polycome phone, 950.00 for a polycom voicestation 4000 |
02:33.28 | TripleFFFF | takes me around 15 minutes per get all libs/core+ fax up. |
02:33.42 | TripleFFFF | then 30 min to 1 hour for all tdm stuff |
02:33.44 | MrTelephone | so it adds up pretty quick |
02:34.19 | bsd_tech | you overpayed for the 501 |
02:34.19 | MrTelephone | 200 canadian |
02:34.19 | bsd_tech | they are down to amost 150 a piece now |
02:34.19 | bsd_tech | ok |
02:34.20 | bsd_tech | Canuck |
02:34.21 | JT | TripleFFFF: what sort of faxing, fax to email, or fax over pri out to an analogue port? |
02:34.22 | MrTelephone | they went down? |
02:34.32 | TripleFFFF | fax2pdf |
02:34.34 | MrTelephone | im using spandsp works great |
02:34.43 | TripleFFFF | and fax to analog etc |
02:34.48 | bsd_tech | nvfax is also good |
02:34.49 | JT | TripleFFFF: what about sending faxes... email to fax? |
02:34.54 | TripleFFFF | hehehe |
02:34.56 | TripleFFFF | not included |
02:34.57 | bsd_tech | but needs updating for 1.4.2 |
02:34.59 | TripleFFFF | but i can do |
02:35.00 | TripleFFFF | ;) |
02:35.17 | MrTelephone | nvfax is another asterisk addin? |
02:35.26 | MrTelephone | who here uses sangoma cards? |
02:35.29 | anthonyl | are you using spandsl on asterisk 1.4.x? |
02:35.32 | anthonyl | spandsp* |
02:35.34 | TripleFFFF | lol |
02:35.38 | TripleFFFF | never.. not using 1.4 |
02:35.46 | MrTelephone | im using asterisk.1.4.11 |
02:35.49 | MrTelephone | i mean 1.2.11 |
02:35.53 | TripleFFFF | just done a client that he pressed save on gui and it erased all the configs from extensions |
02:35.59 | bsd_tech | 1.4.2 is very nice and we have spandsp faxing now |
02:36.02 | MrTelephone | but im trying to develop NCS in MGCP |
02:36.06 | anthonyl | ah word, just checking a friend of mine was having issues with it on 1.4 yesterday |
02:36.10 | TripleFFFF | also had intermittent registrations /authorisation issues |
02:36.18 | bsd_tech | 1.4.2? |
02:36.20 | TripleFFFF | so we rm -rf /asterisk |
02:36.23 | bsd_tech | or 1.4.0 |
02:36.31 | TripleFFFF | and hmm cough.. installed opb |
02:36.31 | MrTelephone | where can i find out how my pri lost sync with asterisk? |
02:36.42 | TripleFFFF | check the util with it |
02:37.06 | TripleFFFF | the apps that comes with the libs for the card ;) |
02:37.11 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
02:37.18 | TripleFFFF | k friday night beer |
02:37.20 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
02:39.10 | MrTelephone | Mar 30 19:36:42 prdc-asterisk-1 kernel: wanpipe1: OOF alarm is ON |
02:39.10 | MrTelephone | Mar 30 19:36:42 prdc-asterisk-1 kernel: wanpipe1: RED alarm is OFF |
02:39.10 | MrTelephone | Mar 30 19:36:42 prdc-asterisk-1 kernel: wanpipe1: T1 disconnected! |
02:39.10 | MrTelephone | Mar 30 19:36:42 prdc-asterisk-1 kernel: wanpipe1: AFT communications disabled! |
02:39.10 | MrTelephone | Mar 30 19:36:42 prdc-asterisk-1 kernel: wanpipe1: Starting TDMV 1ms Timer |
02:39.11 | MrTelephone | Mar 30 19:36:53 prdc-asterisk-1 kernel: wanpipe1: T1 connected! |
02:39.22 | MrTelephone | Mar 30 19:36:53 prdc-asterisk-1 kernel: wanpipe1: AFT communications enabled! |
02:39.22 | MrTelephone | Mar 30 19:36:53 prdc-asterisk-1 kernel: wanpipe1: AFT Global TDM Intr |
02:39.27 | MrTelephone | it was connected but never restarted the bchannels |
02:39.28 | MrTelephone | wtf |
02:39.37 | MrTelephone | is that the phone company screwing around? |
02:40.26 | MrTelephone | i'll have to install more stable zaptel drivers |
02:40.32 | bsd_tech | rum and coke |
02:40.56 | techie | and pirates of the caribbean |
02:41.08 | [TK]D-Fender | MrTelephone, and stop SPAMMING THE CHANNEL |
02:41.25 | MrTelephone | sorry skippy |
02:41.37 | MrTelephone | 7 teeny lines |
02:41.42 | JunK-Y | isnt spam. |
02:41.45 | MrTelephone | 8 :-/ |
02:41.54 | [TK]D-Fender | JunK-Y, Sure it is. |
02:42.09 | [TK]D-Fender | Anything over 3 is highly suspect and a block like that, hell yeah |
02:42.13 | JunK-Y | spam isnt unsolited publicity? |
02:42.31 | [TK]D-Fender | *I* didn't ask for it, now did I? :) |
02:42.44 | JT | it's very simple |
02:43.03 | JT | IF lines.numbers => 3 THEN do not send to channel |
02:43.28 | *** join/#asterisk Defraz (n=t0tal@209.137.240.88) |
02:43.39 | MrTelephone | my information could help others |
02:43.47 | MrTelephone | not like i pasted a bunch of C, eek |
02:44.08 | JT | stop defending yourself, and just admit you made a newbie mistake... |
02:44.10 | MrTelephone | OOF is a loss of signal? |
02:44.11 | JT | ~pb |
02:44.15 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
02:44.47 | JT | there is 268 nicks in here, it's not 8 teeny lines, it's 268 * 8 lines |
02:44.50 | *** join/#asterisk f1r3w0rm (n=warirc@adsl-070-145-060-050.sip.bix.bellsouth.net) |
02:45.42 | MrTelephone | i've been using irc for 20 years |
02:45.59 | JT | great |
02:46.00 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
02:46.02 | JT | :) |
02:46.06 | MrTelephone | a flood is like maxchars * maxlines |
02:46.19 | MrTelephone | usually ends up with a disconnect saying queue exceeded |
02:46.20 | MrTelephone | lol |
02:46.34 | MrTelephone | it happened to me before many moons ago |
02:46.35 | JT | no, that's an IRCD enforced excess flood kill |
02:46.49 | MrTelephone | i understand anyways |
02:46.49 | JT | i'm talking about what humans determine to be a flood :) |
02:46.58 | bkruse_home | russellb: you there/ |
02:47.01 | bkruse_home | ? |
02:47.18 | MrTelephone | at lease give me a break and subtract 260 people as they are drones :) |
02:47.32 | f1r3w0rm | hello all |
02:50.46 | MrTelephone | why is 802.11b/g so flakey these days |
02:50.50 | MrTelephone | is it the interference from others? |
02:51.07 | bkruse_home | MrTelephone: actually, pine trees will own your wireless, seriously |
02:51.07 | bkruse_home | lol |
02:51.30 | f1r3w0rm | i know |
02:51.39 | f1r3w0rm | i live around a bunch of em |
02:51.57 | bkruse_home | true isnt it? |
02:52.10 | f1r3w0rm | i can go into city and i have perfect sig i go home im 10 ft from router and i got 20 % sig |
02:52.12 | [TK]D-Fender | bkruse_home = Arborially challenged :D |
02:52.18 | bkruse_home | :P |
02:52.22 | bkruse_home | i just hate trees :P |
02:52.39 | Strom_M | let's send bkruse on a vacation to the pacific northwest |
02:52.59 | bkruse_home | im down |
02:53.08 | bkruse_home | nah, i dont need a vacation, just a full 8 hours of sleep |
02:53.09 | Strom_M | all trees |
02:53.13 | Strom_M | nothing but trees |
02:53.13 | bkruse_home | NO! |
02:53.14 | Innatech | PacNW has AWESOME trees. Heh. |
02:53.23 | bkruse_home | </3 trees |
02:53.27 | bkruse_home | ive had big tests this week :[ |
02:53.54 | f1r3w0rm | lol u think Innatech try livin in south ms |
02:53.59 | MrTelephone | thats right i have a 2km 900mhz link that the pine trees are devastating |
02:54.13 | MrTelephone | i ordered a nice yagi to put up next week to see if it makes a differnece |
02:54.45 | fx0 | what difference will it make with the pine trees ? |
02:55.07 | MrTelephone | its at -80. 6 more db will bring me to an acceptable -75 |
02:55.26 | MrTelephone | the old antenna is 9db gain and the yagi is 14db |
02:55.36 | MrTelephone | so i guess thats 5 db difference |
02:55.39 | fx0 | i'd rather buy a chainsaw :) |
02:55.43 | MrTelephone | me too |
02:55.58 | MrTelephone | whats funny is that he radios are pretty high |
02:56.12 | MrTelephone | 60 ft at one side and 30 ft on the other |
02:56.17 | MrTelephone | the trees are roughly 25-30ft |
02:56.33 | bkruse_home | i think im going to play counterstrike, or work on clientside graphing for asterisk......hmm |
02:56.35 | MrTelephone | elevation difference is nil |
02:56.50 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
02:57.15 | f1r3w0rm | i use linksys wrt 54 g |
02:57.24 | f1r3w0rm | i got a custom firmware on it |
02:57.43 | f1r3w0rm | cranked the antennae power to 126 |
02:57.55 | f1r3w0rm | and i actually maintain connection now |
02:58.45 | MrTelephone | lol |
02:58.51 | MrTelephone | by knocking out the neighborhood |
02:59.05 | MrTelephone | i installed some nice proxim access points.. 300 each or something.. and they are flawless |
02:59.14 | MrTelephone | but who can afford that at home? |
02:59.19 | Innatech | Cranking up TX isn't always a good idea. Then the access point just steps all over the client transmissions. |
02:59.37 | MrTelephone | your raising your noise floor |
03:00.30 | MrTelephone | i installed some firmware too and it didn't make a difference when i jacked up the xmit power.. I was watching it on netstumbler.. |
03:00.32 | MrTelephone | but who knows |
03:00.36 | *** join/#asterisk dj-fu (n=ajc@unaffiliated/dj-fu) |
03:00.49 | MrTelephone | qdon't give yourself cancer |
03:01.13 | Innatech | yah. I've never found anything but a modest increase in TX power on WRTs makes a difference. After that, you're just cooking nearby soft tissues. ;) |
03:01.40 | Innatech | Hey, look down this waveguide! >sizzle< |
03:01.53 | MrTelephone | yeah there is a guaard on the chip or something |
03:02.03 | MrTelephone | FCC wouldn't allow it |
03:02.23 | Innatech | Well....in theory. |
03:02.25 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
03:02.32 | MrTelephone | im having a lot of issues with the new wrt54g's, have to power cycle them all the time? |
03:02.42 | Innatech | Yeah, me too, actually. |
03:02.48 | Innatech | WTF is up with that? |
03:02.57 | MrTelephone | the only theory i know is plug in |
03:03.04 | MrTelephone | not sure |
03:03.11 | MrTelephone | the last 2 i sold were shit |
03:03.28 | MrTelephone | dlink used to give me problems like that |
03:03.46 | Innatech | Yeah. I've had to warranty return a bunch lately. Serious pain and a waste of time, and makes me look like a chump w/the clients. |
03:03.51 | MrTelephone | i think the wireless part is crashing the unit |
03:04.02 | MrTelephone | yeah i hear you |
03:04.11 | MrTelephone | go with dlink for now |
03:04.26 | Innatech | I had one recently where basic switching was failing. QC issues @ the factory, maybe. |
03:04.35 | MrTelephone | maybe |
03:04.35 | Innatech | Yeah, dlink is OK. |
03:04.45 | MrTelephone | i know they changed their software platform |
03:05.09 | MrTelephone | now if you want one that will take linux you have to buy the wRT54G-l or something |
03:05.09 | Innatech | yeah, and the board design changes too. My older WRT54s are my best ones. |
03:05.25 | MrTelephone | and i bought a rangebooster model and it sucks too |
03:05.36 | MrTelephone | dropping connections all the time, doesn't have the qos option in the firmware |
03:06.06 | MrTelephone | i just hope their switches arn't as crappy because I'm using a lot of them |
03:06.33 | Innatech | Most of their *managed* switches are rebadged low-end Cisco gear, which is OK. |
03:06.43 | MrTelephone | i feel your pain though.. I don't sell much stuff but when I do the last thing I want to do is deal with returns |
03:06.47 | Innatech | Their dumb switches.....not so hot. |
03:07.17 | Innatech | Yeah. It's awkward. That's why I try not to resell equipment as much as possible. |
03:07.18 | MrTelephone | well i went with some 24 port cheapos that were 110 brand new |
03:07.30 | *** join/#asterisk pkempgen (n=pkempgen@AC9E8CD7.ipt.aol.com) |
03:07.36 | MrTelephone | trial and error I guess |
03:07.45 | Innatech | Well, they should be OK. Honestly, tho, the linksys stuff on that level is no better than, say, TrendNet. |
03:07.48 | MrTelephone | hey do u have the same problems with the access points? |
03:08.04 | Innatech | I dunno, I use WRT54s *as* access points. :) |
03:08.15 | MrTelephone | ohhh |
03:08.15 | MrTelephone | ok |
03:08.27 | *** part/#asterisk pkempgen (n=pkempgen@AC9E8CD7.ipt.aol.com) |
03:08.44 | MrTelephone | i installed some firmware once and couldn't get wap working so i took it off |
03:08.50 | MrTelephone | WPA rather? |
03:09.22 | f1r3w0rm | i use the dd-wrt firmware |
03:09.26 | f1r3w0rm | micro |
03:09.31 | f1r3w0rm | its a nice one |
03:09.58 | f1r3w0rm | wrt54G cant handle the standard >.< |
03:10.07 | f1r3w0rm | not the older models ne wyas |
03:10.40 | f1r3w0rm | if u get a newer wrt54 u install the dd-wrt standard it had ipv6 support |
03:11.29 | MrTelephone | sweet |
03:11.36 | MrTelephone | i never even looked at ipv6 yet |
03:11.45 | MrTelephone | i don't think its supported in canada yet |
03:11.46 | MrTelephone | lol |
03:11.56 | f1r3w0rm | i use it mainly for lan |
03:12.02 | f1r3w0rm | specially @ work |
03:12.16 | f1r3w0rm | we jus installed 45k computers |
03:12.57 | MrTelephone | zaptel-core is zaptel-base in 1.2.16? |
03:13.18 | MrTelephone | 45K extended memory? |
03:13.30 | MrTelephone | j/k |
03:13.32 | dj-fu | 45,000 computers? |
03:13.38 | dj-fu | That's a huge company |
03:13.40 | dj-fu | in one building? |
03:13.45 | MrTelephone | theres no way |
03:13.50 | dj-fu | dude :\ |
03:13.50 | coppice | was that before or after lunch? |
03:14.04 | MrTelephone | im not understanding the concept of vlans |
03:14.05 | f1r3w0rm | that was 3 weeks work |
03:14.11 | MrTelephone | if you have one router routing the vlans |
03:14.14 | JT | MrTelephone: very simple |
03:14.19 | MrTelephone | then isn't that physical link easily saturated? |
03:14.33 | JT | depnds on the links |
03:14.50 | dj-fu | you use a layer4 switch and loadbalance across a few routers. |
03:14.54 | dj-fu | don't be daft :0 |
03:14.59 | f1r3w0rm | dj-fu |
03:15.00 | MrTelephone | i guess |
03:15.04 | MrTelephone | seems like a pain |
03:15.08 | dj-fu | not really |
03:15.14 | dj-fu | less of a pain that non vlanning |
03:15.15 | f1r3w0rm | its a call center for ATT |
03:15.37 | fx0 | fully voip ? |
03:15.40 | JT | f1r3w0rm: 45k PCs at one site? |
03:15.44 | f1r3w0rm | ya |
03:15.50 | dj-fu | that's a big ass building. |
03:15.52 | f1r3w0rm | ya |
03:15.53 | f1r3w0rm | it is |
03:15.54 | JT | how big was the place? |
03:15.55 | dj-fu | link |
03:16.10 | f1r3w0rm | its in downtown Dallas |
03:16.31 | dj-fu | TRUE BRO? |
03:16.35 | f1r3w0rm | true |
03:16.44 | dj-fu | I'm from texan. |
03:16.45 | fx0 | how long did you take |
03:16.53 | JT | how many employees do they have their at one time? |
03:17.07 | JT | there, even |
03:17.07 | f1r3w0rm | iuno i was jus a tech installing the shit |
03:17.21 | dj-fu | how big of a team to install 45k pc's? |
03:17.24 | dj-fu | what deployment tools? |
03:17.24 | MrTelephone | its more like downtown world trade center building #3 |
03:17.26 | dj-fu | what OS? |
03:17.38 | coppice | let's see. a 45k seat call centre would emply >100k people. that's about the same as the total call centre business in Bangalore :-\ |
03:17.55 | f1r3w0rm | all we did was roll out hardware |
03:18.06 | f1r3w0rm | another team did setup |
03:18.17 | JT | i'm calling bullshit |
03:18.21 | JT | maybe 4500 stations |
03:18.24 | JT | not 45000 |
03:18.29 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
03:18.49 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
03:19.09 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
03:19.26 | fx0 | what type of hardware did you use for asterisk, i cant even imagine |
03:20.06 | [TK]D-Fender | fx0, Who said this had anything to do with*? :) |
03:20.16 | blitzrage | incase anyone missed it earlier: http://digg.com/software/Asterisk_Users_Weekly_Friday_Conference_Podcast_at_12_30_PM_EDT |
03:20.19 | fx0 | lol dont know man, that black container from sun comes in mind, blackbox. |
03:20.46 | JT | eh, you can handle 45000 users in 1 rack cabinet |
03:20.56 | MrTelephone | im calling that im upgradeing to zaptel-1.2.16 |
03:21.14 | MrTelephone | please don't judge me becaused I have a sangoma card.. I bought 3K worth the digium already |
03:21.18 | JT | well, depends what they're using though |
03:21.25 | JT | 45000 telephony users... |
03:22.20 | blitzrage | MrTelephone: it's fine -- whatever works |
03:22.24 | MrTelephone | i feel like bcause i had a problem with my pri card noone cares because I didn't buy digium |
03:22.54 | [TK]D-Fender | MrTelephone, So whats the issue. Your earlier over-paste just looked like someone pulled the plug and jacked it in 11 sec later |
03:23.02 | fx0 | true, we will call jan dowse @ sales to stab you with an icepick while you are sleeping |
03:23.31 | Strom_M | hahaha |
03:23.38 | JT | MrTelephone: no, people use both cards, just the traffic flow is high in the channel |
03:23.46 | Strom_M | imagining jan doing that is amusing :) |
03:23.57 | Innatech | While everyone's talking, my oft repeated legacy question: any one have experience interfacing * with Meridian/Nortel? |
03:24.36 | [TK]D-Fender | Innatech, Over FXO, I do |
03:25.12 | Innatech | TK: Is it as a terrible pain in the rear as I'm imagining it to be? |
03:25.36 | [TK]D-Fender | Innatech, What are you trying to acheive? |
03:27.48 | Innatech | I'd like to use * to control the extensions on the meridian, and to allow it to access VOIP trunks through Asterisk. Honestly, I'm a little vague on what *is* possible. In short, one of my close family members has a big legacy meridian PBX and wants to salvage as much of it as possible while moving forward into VOIP. |
03:27.55 | MrTelephone | ok i'm just checking |
03:27.55 | MrTelephone | some drunk guy is probably tripping over my cords |
03:28.21 | MrTelephone | asterisk with fxs cards i guess |
03:28.30 | *** join/#asterisk thoughtpolice (n=austin@c75-111-145-138.plaicmtc01.tx.dh.suddenlink.net) |
03:28.41 | fx0 | a t1 card will do the job. |
03:28.51 | Innatech | yeah...that's more or less what I figured. |
03:28.58 | JT | MrTelephone: your server as in a place that a drunk guy can trip over it? |
03:29.13 | fx0 | meridian cards are way more expensive tho. |
03:29.14 | [TK]D-Fender | Innatech, how many phones/lines? |
03:29.15 | Innatech | They originally wanted to focus on resuing the phones, but I gently suggested that was the wrong focus. ;) |
03:29.24 | Innatech | Gah. A lot of phones..... |
03:29.31 | MrTelephone | i work in a native reserve |
03:29.40 | MrTelephone | a native has keys to my office |
03:29.41 | [TK]D-Fender | Innatech, NUMBERS please. What kind of lines, and how many. How man ext? |
03:29.48 | MrTelephone | anything is possible |
03:29.54 | MrTelephone | friday night :-/ |
03:29.55 | Innatech | Heh, I'm thinking. |
03:29.58 | JT | MrTelephone: i see... |
03:30.16 | fx0 | 45k lines! |
03:30.18 | fx0 | omg |
03:30.25 | MrTelephone | i don't think sangoma likes the 1.2.16 zaptel :( |
03:31.01 | Innatech | I think its around 25 lines or so. I'm not sure how they're coming into the switch. Roughly 35 extensions....at any given time some of them are unplugged. |
03:31.54 | Innatech | Some of the lines are parcelled out to subtenants, so it gets a little confusing when they try and explain it to me, especially since they don't really know either. |
03:32.42 | Innatech | But speaking generally: is the meridian hardware nice enough to bother with holding onto? I have no basis for comparison for older systems. |
03:33.17 | Strom_M | Innatech, only 35 extensions? you sure it's not a Norstar? |
03:33.53 | Innatech | Norstar sounds familiar, but the phones and cabinet of the PBX say Meridian. I'll go take a look at it, its around the corner. |
03:34.39 | Strom_M | Innatech, is it a fairly large standalone cabinet, or is it fairly flat modules on the wall? |
03:35.30 | MrTelephone | nortstar nortel |
03:35.32 | MrTelephone | same crap |
03:35.36 | Innatech | Storm--you called it. Flat modules, open the cabinet and it says Norstar. |
03:35.38 | MrTelephone | reliable but overprices |
03:36.04 | Innatech | Worth holding onto? Better to abandon? |
03:36.06 | MrTelephone | i ordere an adit 600 |
03:36.20 | Innatech | If I did a conversion, I'd be doing it for free....so.....be painfully honest. :) |
03:36.23 | JT | MrTelephone: new or second hand? |
03:36.50 | [TK]D-Fender | Innatech, Norstar is annoying pile of crap. Ditch it... it'll cost you less. |
03:36.58 | MrTelephone | new |
03:37.06 | JT | MrTelephone: how much are they new? |
03:37.24 | MrTelephone | 1600 canadian for 16fxs and dual t1 |
03:37.25 | [TK]D-Fender | Innatech, about 5K in gear + PC and its over with |
03:37.39 | Innatech | Yeah, OK. That was my instinct, thanks. |
03:37.50 | MrTelephone | supposed to be good equipment |
03:38.18 | JT | MrTelephone: dual t1? only 16 ports.... |
03:38.22 | Strom_M | Innatech, yeah, ditch the Norstar |
03:38.27 | Innatech | cool. |
03:38.28 | MrTelephone | yeah because it supports up to 48 |
03:38.32 | MrTelephone | or something |
03:38.37 | JT | oh right |
03:38.45 | JT | how much does the adit cost with 48 fxs ports? |
03:38.59 | MrTelephone | not sure i thinjk the blades are like 3 or 4 hundred each |
03:39.05 | JT | hmm |
03:39.07 | MrTelephone | ? |
03:39.13 | JT | how many ports a blade? |
03:39.16 | MrTelephone | 8 |
03:39.20 | JT | oh |
03:39.23 | MrTelephone | i bought it from texas actually |
03:39.29 | f1r3w0rm | Night All |
03:39.34 | MrTelephone | vox technologys or soemthing |
03:39.48 | JT | i'm in australia, so getting channel banks is harder |
03:39.51 | JT | no-one uses them |
03:40.02 | cpm | sell ya mine |
03:40.04 | cpm | :) |
03:40.11 | JT | heh |
03:40.11 | MrTelephone | the office nextdoor just bought a brand new meridian system |
03:40.12 | cpm | I gotta pile of 'em here that suck |
03:40.19 | JT | what model? |
03:40.24 | cpm | CAC |
03:40.27 | MrTelephone | i think so |
03:40.31 | JT | why do they suck? |
03:40.33 | cpm | CAC |
03:40.34 | MrTelephone | with the call pilot |
03:40.37 | cpm | the suck model |
03:40.47 | JT | i thought CAC wasn't that bad |
03:40.53 | MrTelephone | they are good, but rather have installed a crashing asterisk box |
03:40.54 | MrTelephone | lol |
03:40.55 | cpm | Carrier Access Group, the sucky ones |
03:41.20 | MrTelephone | i need something so i can power cycle a computer |
03:41.23 | cpm | seriously, I have a closet full of these damned things |
03:41.37 | JT | MrTelephone: ip powerboard |
03:41.38 | MrTelephone | over the internet |
03:41.48 | JT | cpm: what bit sucks? |
03:42.00 | cpm | these are the CB1s |
03:42.04 | MrTelephone | cuz i just crashed my computer at work now i have to drive 20km to reboot it |
03:42.04 | Innatech | My relative's company bought the Norstar back in the 90s. It was a *ton* of money back then, and it has done alright over the years. But it's good to get some endorsement on getting rid of it. |
03:42.14 | Innatech | Anyway, thanks everyone. TTYL. |
03:42.22 | cpm | I think they were custom burned for the telco I got them from, |
03:42.38 | JT | MrTelephone: you can get ip powerboards |
03:42.42 | cpm | I have a CB2 that sucks less |
03:42.47 | cpm | thats the one I use |
03:42.50 | MrTelephone | im gonna buy one |
03:43.08 | JT | MrTelephone: they cost about the price of a new desktop pc though |
03:43.19 | MrTelephone | yeah i know |
03:43.23 | MrTelephone | same with those nice ip kvms |
03:43.27 | MrTelephone | 2k |
03:43.29 | cpm | I've seen'em for as little as $250 or so |
03:43.30 | JT | yeah |
03:43.42 | MrTelephone | expensive crap |
03:43.46 | cpm | just for an IEC power interrupter |
03:43.50 | MrTelephone | anyways gotta go for a drive |
03:43.54 | cpm | n'joy |
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03:50.49 | coppice | data centre staff will interrupt your power for free. |
03:51.02 | coppice | restoring it can cost, though :-) |
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03:59.38 | cpm | my local power company does it for me. |
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04:35.20 | SECGOD | anyone have success with Cisco 7960 phones with Asterisk ? |
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04:43.31 | fx0 | thousands of times |
04:44.31 | fx0 | whats up |
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05:17.07 | [[blah]asfd | i added accountcode = dialer to my extensions in sip.conf. but when I dial those extensions there is nothing added to the cdr with that account code. is that not the right way to do that? |
05:22.22 | dc3aes | im having a really weird sporatic bug here guys.. got an IAX trunk to NuFone from our PBX and completely randomly (about 30% on average) of the time it the CPN/CID is being dropped and it shows up as "unavailable" to the dialed party.. but in the nufone logs it in fact shows that we set the CID properly.. anyone heard of this? |
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05:22.51 | [[blah]asfd | I just left nufone.. i was very unhappy with their service. |
05:23.17 | [[blah]asfd | VERY unhappy!!! |
05:24.27 | [[blah]asfd | I did not think that nufone supported outbound caller id |
05:26.38 | dc3aes | they do, and let you set it at will... however if its this sporadic whats the point? |
05:27.37 | [[blah]asfd | yeah, I had them for 4 months and had nothing but "sporadic issues" and could never get them to answer trouble tickets or calls. |
05:44.27 | [[blah]asfd | so i am setting accountcode = dialer in sip.conf, but when I do a NoOp(${ACCOUNTCODE}) it displays nothing. what am i doing wrong? |
05:53.07 | SwK | try getting rid of the spaces aroudn the = sign |
05:54.33 | [[blah]asfd | no difference. |
05:54.41 | [[blah]asfd | still does not display properly. |
05:55.20 | [[blah]asfd | is account code applied to inbound calls to a device? or only outbound? |
06:04.50 | dlynes_laptop | [[blah]asfd: only outbound for sip |
06:05.01 | dlynes_laptop | [[blah]asfd: when you set it in sip.conf that is |
06:05.49 | dlynes_laptop | [[blah]asfd: its purpose is for billing, which would have limited use for inbound (only on systems that charge both for outgoing and incoming calls) |
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06:11.10 | DoDaT69 | asdf |
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06:22.11 | rollergrrl | Anyone sober? |
06:22.27 | dlynes_laptop | nope |
06:22.38 | Tene | yus |
06:23.02 | rollergrrl | I'm barely able to type |
06:23.13 | rollergrrl | and I'm supposed to fix a serve outage |
06:23.48 | Tene | ack. that's trouble. |
06:24.10 | rollergrrl | Drinking coffee like mad |
06:24.16 | rollergrrl | damn birthday parties |
06:24.47 | rollergrrl | I got a 1080p LCD 47" today though |
06:24.52 | rollergrrl | it's happy! |
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06:29.16 | dlynes_laptop | rollergrrl: great for administering asterisk, i would imagine :0 |
06:34.54 | fx0 | it is a tv. |
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07:06.56 | tzafrir_laptop | anybody here with some understanding of ISDN? |
07:07.22 | tzafrir_laptop | what does 'pridialplan=unknown' do? |
07:08.01 | tzafrir_laptop | I know it technically sets the Type Of Number . But what does this do in practice? |
07:08.32 | tzafrir_laptop | Should I use "unknown" or national/international/local ? |
07:14.00 | JerJer | tzafrir_laptop: really depends on what the provider of the ISDN expects |
07:15.23 | tzafrir_laptop | JerJer, it is actually ISDN BRI providers, in two different countries (Italy and Australia). I tried messing with local, national, etc, but no luck: I could dial national numbers, but not to mobile phones. |
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07:18.18 | JerJer | strange |
07:19.24 | tzafrir_laptop | I figure that that telco is more tolerant to ISDN phones, and thus tolerates "unknown" |
07:19.59 | tzafrir_laptop | But this is just a hunch |
07:21.04 | tzafrir_laptop | But I don't really understand why I should have this knowledge. Isn't it the job of my provider? |
07:21.21 | tzafrir_laptop | I guess I'm not familiar enough with how telcos work |
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07:52.22 | DarKnesS_WolF | what is group paging means ? |
07:55.02 | piper69 | DarKnesS_WolF: i would think paging a group of users at once |
07:57.11 | DarKnesS_WolF | piper69: yes what is paging i'm trying to dig it in voip-infoorg |
07:58.39 | DarKnesS_WolF | ahh got it |
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08:05.36 | SoftIce | hi qualify=yes what does this do exactly? |
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08:20.16 | mvanbaak | it will send a SIP OPTIONS packet every now and then |
08:20.26 | SoftIce | to see latency, etc ? |
08:20.30 | mvanbaak | that waay asterisk knows the latency between itself and the phone |
08:20.43 | SoftIce | or the latency between itself and the trunk ? |
08:20.52 | mvanbaak | if the latency is more then 2 seconds asterisk will mark the phone unreachable |
08:20.56 | SoftIce | is it advised to use when using nat and using a trunk |
08:21.10 | SoftIce | mvanbaak: but what about trunks to carriers |
08:21.13 | mvanbaak | SoftIce: yeah s/phone/trunk/ |
08:21.15 | SoftIce | with latency of 50/60ms ? |
08:21.26 | SoftIce | should i disable qualify then ? |
08:21.34 | mvanbaak | no |
08:21.44 | SoftIce | but i dont want it to say its unreachable |
08:21.54 | SoftIce | id prefure it to try everything first |
08:22.05 | mvanbaak | it only goes to unreachable if the latency is above 2 seconds |
08:22.24 | SoftIce | ahh, 200ms |
08:22.30 | mvanbaak | 2000 ms |
08:22.35 | SoftIce | ahh, I see |
08:22.45 | SoftIce | so allways best to use qualify then ? |
08:22.55 | SoftIce | any other tricks to help with carriers using voip |
08:22.56 | mvanbaak | if nat is involved, yeah |
08:23.07 | SoftIce | becasue i sometimes get all curcuits are busy |
08:23.19 | mvanbaak | the option packets will keep the nat state open in you nat device |
08:23.52 | SoftIce | and the options packet is built into qualify ? |
08:23.56 | SoftIce | well what qualify sends out? |
08:23.58 | mvanbaak | yeah |
08:24.02 | SoftIce | great |
08:24.03 | piper69 | DarKnesS_WolF: to be honest with you i really don't know shit about asterisk. i am dying to learn |
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08:24.12 | SoftIce | so thats about the best effor I can give to sip with nat and using a carrier ? |
08:24.35 | mvanbaak | if you get 'all circuits busy' you can check to see if the trunk is marked 'unreachable' |
08:24.35 | SoftIce | im using a dynamic ip so I have externip=hostname and have externrefresh=10 |
08:24.41 | piper69 | i want to go as cheap as possiable |
08:24.45 | mvanbaak | if not, the remote end might be on the phone indeed |
08:24.55 | SoftIce | mvanbaak or else does it mean that the far end doesnt have any trunks available? |
08:25.01 | mvanbaak | or...... let's hope not.....the carrier has no more free outbound lines |
08:25.02 | SoftIce | mvanbaak: ahh |
08:25.07 | SoftIce | mvanbaak :) |
08:25.08 | SoftIce | could be |
08:25.40 | SoftIce | mvanbaak: any idea what might cause this then, im on a call and I get a break for a second I an hear the person or visa versa then it kicks in again? |
08:25.45 | piper69 | mvanbaak: what do you mean by carrier, is it the T1 |
08:25.55 | SoftIce | piper69: im using a carrier for voip |
08:25.59 | SoftIce | to other countrys |
08:26.26 | mvanbaak | SoftIce: that sounds like a bandwidth issue |
08:26.44 | SoftIce | mvanbaak: even though the call is crystal clear? |
08:26.47 | piper69 | SoftIce: oh, is it possable i can know the name or wesite for them |
08:26.50 | SoftIce | and im using g729 |
08:26.59 | SoftIce | piper69: hey ? |
08:27.03 | SoftIce | www.govoip.co.za |
08:27.51 | mvanbaak | SoftIce: I never used g729 |
08:27.57 | mvanbaak | I always use g711 |
08:28.48 | mvanbaak | but we had the same problem as you describe |
08:29.01 | mvanbaak | and it turned out to be an issue with our dsl line |
08:29.11 | mvanbaak | we enabled QOS on the ATM level |
08:29.17 | mvanbaak | and everything is fine since then |
08:29.47 | mvanbaak | we let the ISP enable QOS, sorry |
08:30.22 | piper69 | SoftIce: i don't think they provide calls to Africa, sudan |
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08:34.31 | piper69 | do i have to have all the hardware to run asterisk, i am running debian and i want to install asterisk to test and see how it looks like |
08:35.25 | SoftIce | mvanbaak: ahh, luck you :) |
08:35.33 | SoftIce | to get our ISP's to even bother with that is a no go |
08:37.08 | SoftIce | what sound is 'busy' ? |
08:37.12 | SoftIce | I cant find busy.gsm |
08:37.20 | SoftIce | just to do a normall engaged sound ? |
08:37.33 | mvanbaak | busy() |
08:37.38 | SoftIce | not playing that gastly message |
08:37.54 | SoftIce | all cuircuts, etc :P |
08:38.08 | SoftIce | mvanbaak: ye, do you know what busy() actually plays, what gsm file? |
08:38.09 | mvanbaak | you can try: Busy() |
08:38.20 | mvanbaak | I dont think it plays a file |
08:38.31 | mvanbaak | I think it uses indications.conf to create the tones |
08:38.51 | mvanbaak | but I'm not 100% sure |
08:38.56 | mvanbaak | you could check |
08:39.02 | SoftIce | thanks |
08:39.14 | mvanbaak | exten => 13,1,Busy(30) |
08:39.23 | mvanbaak | that will play the busy tones for 30 seconds |
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08:43.11 | SoftIce | mvanbaak you ever had this |
08:43.17 | SoftIce | you dial from you box to a phone |
08:43.19 | SoftIce | you hangup |
08:43.24 | SoftIce | but it is not allways hanging up the call |
08:43.29 | SoftIce | so it keeps ringing but you have put down the phon? |
08:43.50 | JT | tzafrir_laptop: just a quick tip: NO-ONE uses ISDN hardphones in Australia |
08:43.58 | JT | or for values very near to none |
08:44.05 | JT | people use isdn |
08:44.08 | JT | not isdn phones |
08:44.19 | SoftIce | who has here an engaged.gsm file ? |
08:44.45 | JT | SoftIce: what do you need all that crap for? just use generated tones |
08:44.58 | SoftIce | JT: playing with stuff |
08:45.04 | SoftIce | JT: any idea on my last query ? |
08:45.20 | SoftIce | i phone from my phone through my asterisk to my cell I hang up my phone |
08:45.23 | SoftIce | the cell keeps ringing? |
08:45.29 | JT | look up playtones, congestion, and busy in the wiki |
08:45.36 | JT | they use indications.conf |
08:45.40 | SoftIce | thanks |
08:46.03 | mvanbaak | hhmm |
08:46.12 | mvanbaak | SoftIce: I have no idea. I never had that |
08:49.12 | SoftIce | ye, its weir |
08:49.17 | SoftIce | it doesnt happend all the time |
08:49.19 | SoftIce | but it happens |
08:49.24 | SoftIce | maybe a sip bridge between each other |
08:49.29 | SoftIce | or somethig to that affect |
08:49.48 | SoftIce | thats the only reason I prefure iax, is all these issues with sip/nat |
08:50.15 | piper69 | what is it called when you can someone and instade of you hear the ringing tone you hear music.? |
08:50.25 | SoftIce | MOH |
08:50.28 | SoftIce | music on hold |
08:50.44 | SoftIce | oh, i miss understood |
08:50.49 | piper69 | no MOH is when you put someone on hold |
08:50.59 | SoftIce | well it can be an IVR |
08:51.04 | SoftIce | that will take the call and play music |
08:51.09 | SoftIce | untill some line becomes avaiable |
08:51.12 | piper69 | nope |
08:51.18 | piper69 | thats not it |
08:51.26 | JT | it's called a musical ringing indication |
08:51.29 | JT | or similar :) |
08:51.32 | piper69 | i think its PVI or something like that |
08:51.39 | JT | also known as pain in the arse |
08:51.55 | SoftIce | oh, you talking about pre-answer |
08:52.06 | SoftIce | ye, just like you can generate a ringing tone with r |
08:52.16 | SoftIce | you can do the same to do nothing orplay music |
08:52.17 | SoftIce | or what ever |
08:52.41 | piper69 | SoftIce: either you type really too fast, or you need a new keyboard |
08:53.14 | piper69 | lol |
08:53.29 | SoftIce | what do you mean ? |
08:53.48 | JT | possiblysomethingtodowiththelackofspaces |
08:54.09 | SoftIce | spaces, you mean many lines to structure a sentence |
08:54.19 | JT | <PROTECTED> |
08:54.22 | JT | ^ |
08:54.24 | JT | ;) |
08:54.39 | piper69 | SoftIce: you miss alot or s a r's :) |
08:54.48 | SoftIce | ahh, well i've been a unix admin for over 12 years :) you get used to tab completion :P |
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08:55.23 | piper69 | SoftIce: hahahaha yeah |
08:55.23 | SoftIce | and swopping keyboards also make a difference, laptop UK, desktop US, etc :P |
08:55.30 | mvanbaak | SoftIce: learn to touchtype |
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08:55.43 | SoftIce | I bang type :) |
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08:55.47 | mvanbaak | that way you can set your keyboard layout in the os ok |
08:55.58 | mvanbaak | no matter what the keys are on your hardware |
08:56.12 | piper69 | you know what the funny thing, i was working in a windows machine the other day and for some reason i hit the tab com after a command and it did complete it . i was stun |
08:56.20 | SoftIce | well, let me quote what dragon natrual speaking once said, real men dont have to type, use a head set and dictiate what you want :P |
08:56.34 | SoftIce | piper69 mwahah , had the same awhile back :) |
08:57.28 | piper69 | SoftIce: what distro are you using |
08:57.30 | AndyCap | SoftIce: so what did it type on your screen if you said that. :P |
08:57.56 | SoftIce | :P |
08:58.16 | SoftIce | piper69: well mainly fbsd, untill i got a large contract with 2 companys that use ubuntu :P |
08:58.27 | SoftIce | im south africa, i should use it anyway : |
08:58.28 | SoftIce | :) |
08:58.52 | piper69 | where in south africa? we are brother then |
08:58.58 | piper69 | *s |
08:59.03 | SoftIce | i live close to durban |
08:59.10 | SoftIce | lived all over though, cape town, jhb, :P |
08:59.20 | SoftIce | didnt like jhb, im a surfer! :) need the ocean |
08:59.42 | piper69 | SoftIce: i am originaly from sudan |
08:59.50 | piper69 | but i live in USA now |
09:00.28 | SoftIce | nice man |
09:00.33 | SoftIce | id love to live in the us too :) |
09:01.49 | piper69 | yep you will brother, one day you will. that was my dream |
09:02.53 | piper69 | what do you do for living |
09:05.12 | piper69 | i am running debian etch and in the process of downgrading to sarge ao that i can insall asterisk |
09:05.57 | SoftIce | haha |
09:06.16 | SoftIce | why downgrade? why not use ubuntu at least it has support up till 2011 or something for 6.06 |
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09:12.13 | piper69 | isn't ubuntu from debian thu |
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09:12.47 | SoftIce | well its very much the same, except you cant use debian in an enterprise solution because they dont have support |
09:13.23 | SoftIce | also, ubuntu has ubuntu dapper what is version 6.06 that has a support till for another 4 or so years |
09:13.33 | SoftIce | untill their next long term release is issued |
09:14.15 | SoftIce | also, if you compare packages, you will see that debian packages are slightly higher(newer) than ubuntu, as ubuntu tests their stuff thoroughly |
09:16.45 | mvanbaak | SoftIce: that last one is the other way around |
09:16.52 | mvanbaak | ubuntu has newer packages |
09:17.02 | mvanbaak | unless you are running debian testing |
09:17.12 | mvanbaak | that one gets new packages quickly |
09:17.15 | mvanbaak | so ppl can test it |
09:17.22 | mvanbaak | before it becomes debian stable |
09:17.42 | mvanbaak | debian stable is very picky about new packages |
09:17.46 | piper69 | mvanbaak: it's called etch |
09:18.11 | piper69 | will astersik work on ubuntu |
09:18.13 | mvanbaak | that's testing |
09:18.18 | mvanbaak | piper69: yeah it will |
09:19.27 | SoftIce | mvanbaak ye |
09:19.34 | SoftIce | well im talking about long term relase ubuntu |
09:19.38 | SoftIce | their packages are not newer |
09:19.41 | SoftIce | but edgy, etc are |
09:20.39 | SoftIce | hmm, i have this issue, i phone, right it goes through the carrier rings on the caller phone |
09:20.41 | SoftIce | they answer |
09:20.47 | SoftIce | but it keeps ringing on my side? |
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09:25.49 | piper69 | SoftIce: i was pormoted today to be an 5ESS DCS Switch Engineer |
09:25.58 | SoftIce | nice |
09:26.02 | SoftIce | you working saturdays? |
09:26.49 | piper69 | SoftIce: nope , no sat or sunday |
09:28.12 | piper69 | i haven't work sat and sunday for almost 2 yrs now , thats when i was an IT manager for the wireless company i work for, but now since i am going to be a switch eng i think i will work sat or sun as it required |
09:30.56 | SoftIce | :) |
09:30.57 | SoftIce | nce bro |
09:31.11 | piper69 | mvanbaak: what version of ubuntu do you think , |
09:31.43 | SoftIce | piper69 6.06 |
09:31.46 | mvanbaak | piper69: I dont use ubuntu |
09:31.47 | SoftIce | dapper |
09:31.50 | mvanbaak | I use debian |
09:32.14 | SoftIce | debian is great you just cant use it in an enterprise env |
09:32.26 | piper69 | mvanbaak: so what do you think , i am currently runnin etch |
09:32.40 | mvanbaak | etch is fine |
09:32.44 | mvanbaak | I'm running it too |
09:32.53 | mvanbaak | SoftIce: I'm running debian in enteprise env |
09:33.07 | SoftIce | mvanbaak: well you not understanding what im saying, you dont have support |
09:33.35 | piper69 | is there is a pakage for Asterisk |
09:33.45 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
09:33.47 | SoftIce | piper69 apt-get install asterisk zaptel |
09:34.14 | DarKnesS_WolF | SoftIce: what version of astersiks debian has ? |
09:34.24 | piper69 | but why when i did apt-cache search asterisk nothing came |
09:34.56 | SoftIce | piper69 you need to add some repos |
09:35.06 | SoftIce | to the sources.list file |
09:35.06 | piper69 | DarKnesS_WolF: i will assume a lower version of the current onw |
09:35.19 | SoftIce | piper69 wrong, as the lower version has sip vulns |
09:35.22 | SoftIce | so they are up to date |
09:35.55 | SoftIce | grrrrr, please can somebody sugest something to me, I phone out, pass call to carrier via sip they pass the call to my cell phone I answer but on my side it keeps ringing |
09:36.53 | DarKnesS_WolF | SoftIce: hmm do u have -r option in ur dial line ? |
09:37.30 | SoftIce | DarKnesS_WolF: yes |
09:37.31 | piper69 | all what i need is VOIP phone and i can run asterisk? |
09:37.40 | DarKnesS_WolF | SoftIce: remove it and try |
09:37.48 | DarKnesS_WolF | this r i think to foce asterisk to give ring tone. |
09:38.09 | piper69 | and sign up with voip carrier |
09:38.20 | SoftIce | DarKnesS_WolF: yes but Ihave a delay with the carrier |
09:38.23 | SoftIce | thats why I have it in there |
09:38.30 | SoftIce | so people can least here its doing something |
09:38.36 | DarKnesS_WolF | SoftIce: try without it 1st |
09:38.51 | SoftIce | great |
09:40.11 | piper69 | so can someone answear my question |
09:41.27 | piper69 | please |
09:42.01 | SoftIce | what? |
09:42.39 | piper69 | for hardware, all what i need is an VOIP phone |
09:43.01 | piper69 | or ATA |
09:43.24 | AndyCap | piper69: you don't really need hardware either, but it depends on what you want to do.. |
09:44.16 | piper69 | all what i want is to have a phone that i can use to call my parents in africa |
09:44.28 | piper69 | and maybe use it here localy |
09:46.14 | SoftIce | DarKnesS_WolF it still happens :P |
09:46.42 | DarKnesS_WolF | SoftIce: hmm no idea ;-) |
09:48.04 | piper69 | <PROTECTED> |
09:49.23 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
09:49.53 | EmleyMoor | How do I specify the timezone that covers Shanghai in the SayUnixTime command? |
09:52.09 | piper69 | Setting up asterisk-sounds-main (1.2.13~dfsg-2) ... |
09:52.09 | piper69 | Setting up fxload (0.0.20020411-1) ... |
09:52.09 | piper69 | Setting up zaptel (1.2.11.dfsg-1) ... |
09:52.09 | piper69 | Zaptel telephony kernel driver: FATAL: Module ztdummy not found. |
09:52.09 | piper69 | Notice: Configuration file is /etc/zaptel.conf |
09:52.11 | piper69 | line 0: Unable to open master device '/dev/zap/ctl' |
09:52.14 | piper69 | 1 error(s) detected |
09:52.16 | piper69 | /sbin/ztcfg failed. Check /etc/zaptel.confzaptel. |
09:52.19 | piper69 | Setting up asterisk-classic (1.2.13~dfsg-2) ... |
09:52.21 | piper69 | Setting up asterisk (1.2.13~dfsg-2) ... |
09:52.24 | piper69 | Asterisk not yet configured. Edit /etc/default/asterisk first. |
09:52.32 | piper69 | !Topic |
09:52.42 | piper69 | !topic |
09:53.52 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
09:54.12 | *** part/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
10:05.28 | *** join/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy) |
10:08.48 | dlynes_laptop | EmleyMoor: export TZ=UTC+8 doesn't work? |
10:09.16 | *** part/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy) |
10:36.11 | tzafrir_laptop | piper69, what version of Debian is it? |
10:36.50 | tzafrir_laptop | apt-get install zaptel-source build-essenials |
10:37.00 | tzafrir_laptop | m-a a-i zaptel |
10:37.14 | tzafrir_laptop | /etc/init.d zaptel start |
10:38.38 | *** join/#asterisk lorinc (n=ang@pool-1270.adsl.interware.hu) |
10:43.43 | *** join/#asterisk ming_zym (n=ming_zym@124.254.52.3) |
11:06.00 | tzafrir_laptop | EmleyMoor, Asia/Shanghai ? |
11:06.22 | tzafrir_laptop | EmleyMoor, generally look under /usr/share/zoneinfo |
11:11.07 | *** join/#asterisk stony (n=steinche@p5B15196E.dip0.t-ipconnect.de) |
11:11.10 | stony | hi |
11:11.20 | stony | can i add more than one device to a trunk ? |
11:11.35 | stony | i need 4 misdn channels in a trunk so that asterisk can get the next free one and use it |
11:12.39 | *** join/#asterisk Ebola (n=Ebola@host86-136-190-11.range86-136.btcentralplus.com) |
11:13.33 | *** join/#asterisk AndyMcGee001 (n=root@88.134.202.82) |
11:14.26 | tzafrir_laptop | stony, in zap (bri/pri) that's pretty simple: a group. I imagine a similar concept exists with misdn |
11:14.54 | AndyMcGee001 | Hi |
11:16.29 | stony | tzafrir_laptop: i read it in this moment |
11:16.35 | stony | thx anyway |
11:18.26 | AndyMcGee001 | Can i do more than one "hint" before i dial more than one sip-phone and will it pickup this dial if someone picksup`s one the hinted extensions? |
11:25.45 | EmleyMoor | AndyMcGee001: What are you trying to achieve |
11:25.47 | EmleyMoor | ? |
11:25.55 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
11:27.44 | AndyMcGee001 | I want to dispactch a incomming call to a group of people. When i do this all the line buttons of the people not in this group start to flash... |
11:28.26 | AndyMcGee001 | When someone not in the group now hits one of the flashing keys a pickup is issued to the numbered extension programmed for that key... |
11:28.56 | EmleyMoor | Hmmm... I would think it possible - but I don't know how |
11:29.04 | AndyMcGee001 | but the pickup goes to nirvana because the dial is not assotioated with the extension, only with the SIP username(s) |
11:31.58 | EmleyMoor | Sounds like you need a dialable thing to flash the line buttons... |
11:35.56 | AndyMcGee001 | Yes, i thought about that to. But i think i need to assosiate all the extensions (101, 105, 103 etc.) with the call because thats whats programmed to the buttons inside the phones. |
11:36.42 | AndyMcGee001 | and "hint" seems applicable. I just dont know if i can attach more than one hint to a call. |
11:37.31 | EmleyMoor | Am I right in thinking PoE (the standard kind) will work over 2-pair wiring, btw? |
11:37.33 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
11:40.34 | *** join/#asterisk l1nux (n=moi@jof81-1-82-245-67-40.fbx.proxad.net) |
11:40.44 | l1nux | hi |
11:40.54 | AndyMcGee001 | Yes i think so, uses some kind of phantom powering then (dc overlay) |
11:41.12 | l1nux | asterisk 1.4.2 not support sqlite3 ? |
11:41.36 | tzafrir_laptop | l1nux, there are a number of patches for sqlite support |
11:41.58 | tzafrir_laptop | There is also something in asterisk_addons , though it is not really maintained |
11:42.44 | l1nux | tzafrir_laptop, please where get patch to fix it ? |
11:43.04 | l1nux | i try patch from http://bugs.digium.com/view.php?id=6754 |
11:43.15 | tzafrir_laptop | that i one of them |
11:43.28 | tzafrir_laptop | It should reference the other ones, I believe |
11:43.41 | l1nux | i get "configure: *** The SQLite installation on this system appears to be broken" |
11:43.44 | l1nux | :/ |
11:44.41 | l1nux | using Debian testing (sqlite3) |
11:46.14 | tzafrir_laptop | at least one of them is ued by the developers of destar, who mainly use debian Etch |
11:47.50 | AndyMcGee001 | Bye |
11:48.30 | tzafrir_laptop | anyway, http://bugs.digium.com/view.php?id=7149 is the one that was merged into trunk |
11:48.40 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
11:49.30 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
11:49.52 | tzafrir_laptop | actually a working sqlite3 logging was added to the Debian asterisk package at some stage. I believe t is included in Etch. checking... |
11:50.58 | tzafrir_laptop | yup. It's in there. http://packages.debian.org/changelogs/pool/main/a/asterisk/asterisk_1.2.13~dfsg-2/changelog . apt-get install asterisk |
11:51.46 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
11:54.11 | l1nux | tzafrir_laptop i am using astersik 1.4.2 from asterisk.org |
11:56.23 | l1nux | http://lists.digium.com/pipermail/asterisk-dev/2006-December/025266.html :/ |
11:56.26 | tzafrir_laptop | right. the asterisk 1.4.2 in experimental does not have that patch applied |
11:59.37 | tzafrir_laptop | I do see cdr_sqlite.so which is sqlite2 |
12:01.30 | l1nux | yes |
12:17.31 | *** join/#asterisk MrTelephone (n=DeaLER25@bas13-toronto63-1177850949.dsl.bell.ca) |
12:17.43 | MrTelephone | hey is there a program to freemem for linux |
12:17.58 | MrTelephone | my ram usage slowly creeps up |
12:18.49 | *** join/#asterisk theBong (i=jvckso@84.94.98.200.cable.012.net.il) |
12:20.36 | theBong | newb q: agi-test.agi does not play back the command SEND TEXT.. Any suggestions/Alternatives? The log reads 200 as the return code |
12:23.00 | theBong | anyone? |
12:24.28 | MrTelephone | are those external scripts your running/ |
12:24.30 | MrTelephone | ? |
12:25.29 | theBong | i copied the contents of agi-test.agi to swift.agi ( I use RAGI) and the script runs. SAY NUMBER works but not SAY TEXT |
12:26.20 | theBong | also , where is STDERR by default (Linux)? Thanks. |
12:26.45 | *** join/#asterisk bjohnson (n=bjohnson@i209-195-93-204.cia.com) |
12:27.53 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
12:28.00 | florz | theBong: Usually on the left. |
12:30.14 | theBong | the log read Mar 31 04:27:34 VERBOSE[8666] logger.c: AGI Rx << SEND TEXT "hello world" |
12:30.14 | theBong | Mar 31 04:27:34 VERBOSE[8666] logger.c: AGI Tx >> 200 result=0 |
12:30.18 | MrTelephone | my dhcp server is whacked out |
12:30.32 | theBong | and nothing gets played |
12:31.57 | florz | theBong: Why do you think anything should get played? |
12:32.23 | theBong | SEND TEXT should play the text to the user. SAY NUMBER works fine |
12:32.42 | florz | theBong: Why do you think anything should get played? |
12:33.31 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
12:33.33 | theBong | see my answer above |
12:33.56 | florz | theBong: Well, then: Why do you think SEND TEXT should play the text to the user? |
12:34.16 | florz | how ever one would "play a text" |
12:35.15 | theBong | how else would you recommend to say text in agi? I need a simple AGI that receives 1 param and reads it to the user |
12:35.19 | theBong | thanks again |
12:35.24 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
12:35.57 | florz | theBong: Well, I guess you would need a speech synthesis program first. |
12:36.26 | theBong | i have flite |
12:36.51 | florz | theBong: Well, and then use it like you would use it from the dialplan |
12:39.24 | theBong | that's where I am struggling. RAGI (Ruby AGI) has a say method which in turn calls swift.agi. I wanted to create this swift.agi to simply read the text sent by RAGI |
12:39.24 | *** join/#asterisk DrCron (n=rszasz@c-67-174-231-152.hsd1.ca.comcast.net) |
12:39.37 | theBong | perhaps I should change the ragi code... |
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13:03.31 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
13:04.02 | MrTelephone | is there a pri failover device that switches to computer 2 if something fails? |
13:04.33 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
13:05.27 | *** join/#asterisk ming_zym (n=ming_zym@124.254.56.158) |
13:08.21 | l1nux | tzafrir_laptop, fixed in trunk (svn) :D |
13:13.20 | *** join/#asterisk _DAW (n=chatzill@adsl-156-109-78.msy.bellsouth.net) |
13:18.13 | *** join/#asterisk ApEtc (i=apetc@ip70-162-197-214.ph.ph.cox.net) |
13:21.53 | _DAW | Anyone interested in testing some SIP DID's? I have 9 to give away for the next 6 months in exchange for regular feedback on the voice quality. I can do most of the US. |
13:33.08 | *** join/#asterisk shinux__ (n=shinux@86.62.8.178) |
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13:36.19 | anthonyl | im up for freevoip |
13:36.24 | anthonyl | er pstn stuffs |
13:36.33 | *** join/#asterisk AbsTradELic (n=vldmr@201.79.156.219) |
13:36.37 | anthonyl | wait,no nevermind |
13:36.41 | *** part/#asterisk anthonyl (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
13:36.45 | _DAW | ok |
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13:57.50 | ligoban | Addons 1.4.0 installation question |
14:00.16 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
14:00.36 | *** join/#asterisk zoa (n=zoa@pirus.securax.be) |
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14:07.10 | f1r3w0rm | hmm |
14:07.10 | f1r3w0rm | anyone tried asterisk for windows ? |
14:07.26 | sbingner | blasphemy |
14:07.38 | JunK-Y | f1r3w0rm: where did u see * for windows? |
14:07.41 | Gido-E | f1r3w0rm are you seriuos? |
14:07.48 | stony | anyone ever tried to get a siemens hipath running together with asterisk ? |
14:07.50 | f1r3w0rm | http://www.asteriskwin32.com/ |
14:08.08 | f1r3w0rm | i guess im pretty damn serious |
14:09.29 | Gido-E | f1r3w0rm: heueu dont run asterisk on a windows machine. |
14:10.02 | f1r3w0rm | do what |
14:10.08 | drfreeze | Morning. Can someone point me to some docs on updating the firmware for Polycom 501 phones? |
14:10.34 | JunK-Y | drfreeze: just put ur new in the directory |
14:10.39 | zeedo | f1r3w0rm: I'd say you'd be better of just installing it and trying it, most people here will think it's silly - personally I'd say try it and if it's good for you then great |
14:10.46 | [TK]D-Fender | AsteriskWin32 0.56 build from Asterisk 1.0.10 |
14:10.49 | [TK]D-Fender | ANCIENT |
14:11.01 | [TK]D-Fender | AsteriskWin32 0.60 build from Asterisk 1.2.14 |
14:11.05 | [TK]D-Fender | Slightly better... |
14:11.05 | zeedo | f1r3w0rm: one thing I would point out is that they dont seem to keep up with asterisk updates, so you could be left open to security issues |
14:11.19 | JunK-Y | 1.2.14 is so old too. |
14:11.24 | f1r3w0rm | this is for a non inet connect system |
14:11.25 | [TK]D-Fender | All * for Win is is a virtual linux env running *. |
14:11.31 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1088929080.dsl.bell.ca) |
14:11.31 | f1r3w0rm | so security aint much issue |
14:11.48 | [TK]D-Fender | You have to config it EVERY BIT the same |
14:11.49 | zeedo | f1r3w0rm: security was an example, all sorts of bugs will be missed |
14:11.58 | [TK]D-Fender | and invite trouble for the virtual aspect of it. |
14:12.01 | drfreeze | JunK-Y: I'm talking about the firmware, not the config. Are they updated in the same way? |
14:12.15 | [TK]D-Fender | zeedo: No... they won't be missing any bugs... they're all still in there ;) |
14:12.25 | JunK-Y | drfreeze: yes |
14:12.29 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
14:12.31 | drfreeze | Any trixbox experts here? |
14:12.35 | drfreeze | JunK-Y: thx |
14:12.37 | [TK]D-Fender | ~trixbox |
14:12.44 | jbot | [trixbox] junk - avoid. It is also unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/, or known as 'sh1tbox', STAY AWAY! |
14:12.44 | JunK-Y | drfreeze: see #trixbox |
14:12.57 | blitzrage | morning all! |
14:13.01 | zoa | yo yo |
14:13.03 | JunK-Y | hiya blitzrage ! |
14:13.14 | drfreeze | I have a client who is using trixbox, and we could not get a phone to authorize. |
14:13.27 | blitzrage | ummmm... that description of trixbox is a bit harsh |
14:13.29 | drfreeze | The passwords on the phone and the sip_additional.conf matched however |
14:13.31 | zeedo | drfreeze: try the #trixbox channel |
14:13.35 | JunK-Y | like we just said, goto #trixbox! |
14:13.37 | zeedo | err sorry #freepbx |
14:14.08 | drfreeze | are there other user/passwords in asterisk that I should know about to authenticate with * |
14:14.15 | blitzrage | anyone know if whisper paging can whisper page all currently bridged channels on a system? |
14:14.31 | blitzrage | I want to use it to announce system outages to current callers |
14:14.45 | blitzrage | i.e. "the system will be going down in 5 minutes" |
14:15.16 | [TK]D-Fender | blitzrage: I suspect if the source end of the whisper is the "Page" app... |
14:15.54 | drfreeze | ok guys, forget I said trixbox. :) |
14:16.03 | Iamnach0 | lol |
14:16.17 | blitzrage | actually, more than likely the ChanSpy() app |
14:16.34 | JunK-Y | a process for that will be excellent! |
14:16.46 | [TK]D-Fender | blitzrage: Use Page to do a Local channel where you call the whisperpage instead of Dial. |
14:17.00 | blitzrage | this would be for active channels -- I don't think that would work |
14:17.07 | drfreeze | Other than username and secret inside sip.conf, is there another user/pass combination used by the phone to authenticate with * |
14:17.09 | f1r3w0rm | zeedo |
14:17.12 | f1r3w0rm | u there |
14:17.13 | blitzrage | i.e. already bridged |
14:17.17 | zeedo | f1r3w0rm: yeh |
14:17.17 | blitzrage | not inactive channels |
14:17.20 | f1r3w0rm | pm plz |
14:17.24 | zeedo | sure |
14:17.33 | [TK]D-Fender | blitzrage: You might want to use an AGI to poll for active channels |
14:17.35 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
14:17.44 | [TK]D-Fender | blitzrage: And then dumpt them. |
14:17.59 | JunK-Y | [TK]D-Fender: theres only one problem to that |
14:18.00 | blitzrage | I think this could be done from a separate dialplan app |
14:18.06 | blitzrage | and would be the most efficient |
14:18.10 | JunK-Y | ya loop to stream to all channels right? |
14:18.33 | blitzrage | an Outage() app, or an 'o' flag to the ChanSpy() application ('o' for outage) |
14:18.34 | JunK-Y | so ya must wrote ur agi in C to start x threads. |
14:18.38 | [TK]D-Fender | JunK-Y: I presume he may only want to whisper to the INESIDE channels, which would require some intelligence to the selection method... |
14:19.14 | [TK]D-Fender | JunK-Y: As in "don't tell customers why we have to hang up on them" |
14:19.39 | JunK-Y | to the inside channels??? |
14:19.59 | blitzrage | I'd just be happy with keeping it more simple, and just broadcast to all active bridges |
14:20.26 | blitzrage | once that worked, then more complex methods of determining who is "inside" and who is "outside" could occur |
14:20.52 | blitzrage | you'd basically have to parse the sip.conf file, and say, "ok, if any of these peers are active, whisper page them" |
14:21.17 | JunK-Y | what do ya mean by inside and outside? |
14:21.18 | blitzrage | JunK-Y: yes, I'd say all channels, not just bridged would be the ideal |
14:21.29 | blitzrage | JunK-Y: he means local extensions vs. outside callers |
14:21.50 | JunK-Y | that could be a later option to the outage app, right? |
14:21.54 | blitzrage | yes |
14:22.00 | blitzrage | for now you'd just say "all channels" |
14:22.09 | blitzrage | just to prove it can work |
14:22.14 | blitzrage | then you can fine tune its settings |
14:22.32 | JunK-Y | but the stream have to be on all the channels, at the same time, right? |
14:22.37 | blitzrage | this might even be a CLI option and not an application.... |
14:22.41 | blitzrage | actually, both would be ideal |
14:22.44 | blitzrage | I can think of uses for box |
14:22.46 | blitzrage | both* |
14:23.00 | blitzrage | JunK-Y: yes at the same time |
14:23.03 | blitzrage | (or within reason) |
14:23.41 | JunK-Y | Outage(filetostream) right? |
14:23.50 | blitzrage | stop gracefully with warning at 5,1 mins |
14:23.55 | blitzrage | JunK-Y: yes |
14:24.20 | blitzrage | stop gracefully notice 5 warn 1 |
14:24.36 | blitzrage | that'd send a notice at the 5 minute mark, then warn with 1 minute left |
14:24.48 | blitzrage | and of course not accept any new calls |
14:25.14 | JunK-Y | it has to avoid Zap/pseudo, any special channels to avoid? |
14:25.37 | blitzrage | you could even get fancy and have a Dial() option or Queue() option that could listen for DTMF so agents could extend / request a longer time out if they are in the middle of a sale |
14:26.07 | blitzrage | JunK-Y: ideally it'd work for Local and application channels (i.e. MeetMe, Queue, etc...) |
14:26.27 | blitzrage | but I'm no channel expert |
14:26.55 | JunK-Y | i will work on that this evening and see what can I do exactly. |
14:27.01 | blitzrage | sounds good! |
14:27.09 | JT | MrTelephone: yes, you can get pri failover devices |
14:27.14 | JunK-Y | that will not be kpflemming's code, but that will work :) |
14:27.17 | blitzrage | I've suggested a couple of good ones to russellb lately |
14:27.33 | zoa | is chanspy stable yet ? |
14:27.44 | JunK-Y | i had in mind, for conferences servers, a way, to do that |
14:27.55 | blitzrage | zoa: I'm using it in testing with no crashes thus far |
14:28.00 | JunK-Y | but only to pseudo/zap, so all meetme users could hear that. |
14:28.11 | zoa | we have some whisper thing we use here |
14:28.13 | blitzrage | zoa: although I think file is building a new version in his audiohooks branch |
14:28.17 | zoa | dunno if its stable though |
14:28.35 | zoa | we dont really use it in production |
14:29.22 | blitzrage | anyone know if you can "redirect" an IAX2 channel via the dialplan like the SIP channel via the Transfer() application? |
14:29.33 | JunK-Y | zoa: it just grab all actives channels and stream a specific message? |
14:29.43 | zoa | we dont have it for all channels :) |
14:29.48 | zoa | but that shouldnt be too hard to make |
14:29.52 | blitzrage | because if that's possible, then I can build an IAX2 registration server, and redirect calls to my Asterisk cluster.... |
14:29.52 | zoa | i guess |
14:30.02 | JunK-Y | zoa: do u think its a good idea? |
14:30.03 | zoa | blitzrage |
14:30.03 | zoa | : i think thats possible |
14:30.08 | zoa | yes it would rock in some cases |
14:30.11 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
14:30.20 | zoa | blitzrage: i was thinking to do something else |
14:30.24 | blitzrage | zoa: yah? As long as its the same channel type right? (i.e. I can't redirect SIP to IAX2) |
14:30.31 | blitzrage | zoa: what's your idea sir? |
14:30.32 | zoa | to actually build a registration server outside of asterisk |
14:30.34 | zoa | just doing that |
14:30.36 | zoa | like SER |
14:30.38 | blitzrage | yes! |
14:30.42 | zoa | as it would be faster |
14:30.53 | blitzrage | it needs to support IAX2 and DUNDi |
14:31.01 | zoa | and it would shield the iax2 port on the asterisk |
14:31.04 | blitzrage | yes |
14:31.06 | blitzrage | good idea! |
14:31.17 | zoa | similar to what we do now for SER and RTP on our setup |
14:31.19 | JunK-Y | blitzrage: i will change it a bit |
14:31.28 | zoa | asterisk is not visible to the outside world |
14:31.31 | blitzrage | JunK-Y: np, let me know when you have something and I will test |
14:31.41 | JunK-Y | Outage(), will go at a particular cep, so we could use anything ya want. |
14:31.45 | blitzrage | zoa: same with us -- we distribute calls to Asterisk via SER |
14:31.51 | blitzrage | cep? |
14:32.00 | JunK-Y | context extension priority |
14:32.03 | blitzrage | ahhh |
14:32.07 | JunK-Y | so you could stream 2 streams |
14:32.09 | blitzrage | yes, that'd be handy |
14:32.15 | zoa | yeah but i dont have audio going straight either |
14:32.18 | JunK-Y | or dial, or email, or whatever ya want. |
14:32.21 | blitzrage | zoa: ahhhhh I see |
14:32.26 | blitzrage | via the mediaproxy? |
14:32.35 | zoa | via our own mediaproxy |
14:32.35 | blitzrage | or whatever SER calls it |
14:32.39 | blitzrage | gotcha |
14:32.51 | JunK-Y | cause if ur on iax2, u could dial ur remote server, and the transfer will do the rest |
14:32.56 | zoa | we can do 3000 bi directional calls through one proxy |
14:32.59 | JunK-Y | so ur server is no longer in the path. |
14:33.03 | zoa | and those proxys are clustered |
14:33.08 | *** join/#asterisk uwe (n=uwe1@dogbert.palnet.com) |
14:33.13 | blitzrage | I have a business edition license here I should register so I can get priority to bug reports :) |
14:33.20 | zoa | lol |
14:33.38 | JunK-Y | blitzrage: i need one too! |
14:33.38 | JunK-Y | :) |
14:33.57 | blitzrage | I had Digium send me one :) |
14:34.25 | JunK-Y | the latest ABE is based on 1.4 or still 1.2? |
14:34.32 | blitzrage | 1.2 I think |
14:34.36 | JunK-Y | kk |
14:34.50 | JunK-Y | ive to jet, breakfast with brother, we'll be in touch soon. |
14:35.15 | blitzrage | sounds good buddy |
14:35.40 | JunK-Y | dont say that to Corydon! |
14:35.43 | JunK-Y | hehehe |
14:36.08 | blitzrage | :) |
14:36.13 | blitzrage | hey, where is that .version file? |
14:37.52 | blitzrage | oh yes, make update :) |
14:37.54 | blitzrage | lets see if it works |
14:38.01 | blitzrage | sometimes its the simple things that you forget |
14:39.04 | JunK-Y | root@troy:/usr/src/asterisk-1.4# build_tools/make_version . |
14:39.05 | JunK-Y | SVN-branch-1.4-r59289M |
14:42.58 | *** join/#asterisk zoa (n=zoa@pirus.securax.be) |
14:46.16 | f1r3w0rm | anyone know where i can get a cheap telephony card |
14:49.11 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
14:49.33 | mmartinn | Please everyone take a moment to rejoice that our PRI problem may be solved :) |
14:49.51 | blitzrage | hallelujah!!! |
14:50.00 | blitzrage | CAN I GET AN "AMEN"!? |
14:50.16 | blitzrage | *may* be solved?! |
14:50.17 | Strom_M | ACK |
14:50.25 | blitzrage | Strom_M: heh :) |
14:50.35 | mmartinn | Well, we're pretty sure. No problems for over 24 hours |
14:50.47 | Strom_M | i'm not terribly good at coming up with snappy ISDN jokes |
14:51.02 | mmartinn | Strom_M: LOL... I didn't get it first, but now I do |
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14:51.36 | mmartinn | I'm now looking forward to figuring out why an identical box isn't working with the PRI |
14:52.32 | mmartinn | My bets are (a) the interface card (b) the network/cable/switch or (c) bad hardware like the mobo/memory/cpu |
14:53.24 | stony | can i do something like: msns=1234* in misdn.conf ? |
14:55.08 | f1r3w0rm | damn Telephony cards arent cheap |
14:55.38 | blitzrage | telephony cards for Asterisk are relatively cheap compared to standard commercial telephony hardware |
14:55.58 | mmartinn | You can always ebay them too if you want cheaper |
14:58.31 | f1r3w0rm | i jus wanted to make a home setup to page my staff |
14:58.54 | stony | if you use sip internaly and a isdn-card for the extern stuff it's really cheap |
14:59.26 | f1r3w0rm | my whole setup is internal |
15:00.12 | f1r3w0rm | im using my house to demo the setup if i like the way everything comes together im gonna do it @ my buddies bussiness |
15:00.20 | stony | go, use voip ... |
15:01.51 | riddlebox | can you record conference calls in asterisk? |
15:02.01 | Qwell | riddlebox: sure, show application meetme |
15:02.10 | riddlebox | ok cool thanks |
15:05.42 | *** join/#asterisk SoftIce (n=phil@vc-196-207-45-253.3g.vodacom.co.za) |
15:06.30 | SoftIce | hmf, anyone know if/when bristiff will release a bristuff using 1.4 asterisk ? |
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15:29.30 | tzafrir_laptop | SoftIce, no idea. There's a letter on their site where they promise to do so |
15:35.00 | SoftIce | tzafrir_laptop: ahh, thank you |
15:51.24 | JT | tzafrir_laptop: catch my note about isdn phone here? :) |
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15:51.35 | tzafrir_laptop | yes |
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15:52.17 | JT | tzafrir_laptop: what telco are you having trouble with? |
15:52.19 | tzafrir_laptop | I was going to answer: well: maybe they were hoping they will have phones when they planned the ISDN BRI service |
15:52.47 | tzafrir_laptop | JT, I was helping someone, so I'm not really sure which telco. Is there more than one? |
15:53.15 | JT | i don't know if any other than Telstra offer BRI |
15:53.20 | JT | others offer POTS and PRI |
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16:00.00 | Maghteridon | Heelloo |
16:00.05 | Maghteridon | ph [TK]D-Fender my hero |
16:00.26 | Maghteridon | *oh =) Can I bother you with a couple of questions? These are fast tbh |
16:00.29 | Maghteridon | mmmm? =) |
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16:01.42 | Maghteridon | I have some problems with music on hold... I'm not able to get it work fine, even tough I followed examples... |
16:01.56 | Maghteridon | all I can hear is a "heavy noise" |
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16:04.03 | Maghteridon | supposing I'd like to play "example.mp3", is musiconhold.conf all I need to modify, if the mp3 file is in /var/lib/asterisk/mohmp3/example.mp3 ? |
16:07.56 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
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16:10.59 | riddlebox | I have setup exten =>698,1,Answer(), exten =>698,2,MeetMe(699,DprM,1234) |
16:11.52 | riddlebox | and also setup in meetme.conf conf => 699,1234,123456 |
16:12.06 | riddlebox | but when I call 698 it says it is an invalid conference |
16:18.48 | *** join/#asterisk eliXier (n=GTI16V@gti.twice-irc.de) |
16:19.59 | _DAW | Anyone interested in some free SIP DID's to help test? I have 9 left I can give away for the next 6 months in exchange for regular feedback on the voice quality, message me off channel if your interested. |
16:24.24 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
16:30.14 | [TK]D-Fender | Maghteridon: Yes? |
16:30.43 | Maghteridon | How do you call the function that let you pass a call from one number to another (internal numbers) ? |
16:30.56 | Maghteridon | I don't know what to look for because I don't know the translation |
16:31.00 | Maghteridon | in english ^^ |
16:31.12 | [TK]D-Fender | Maghteridon: You mean send a call on 1 phone to another? |
16:31.37 | [TK]D-Fender | Maghteridon: That would be "Transfer". There are 2 kinds of transfers, ATTENDED, and BLIND. |
16:31.46 | Maghteridon | I mean X calls Y, Y answers, and then he passes the call to Z |
16:32.36 | [TK]D-Fender | Maghteridon: Attended transfers means YOU cann person #2. They See YOUR CallerID. they pick up. You ask if they want to talk to this other person. They say "yes", tyou then COMPLETE the transfer |
16:33.02 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
16:33.09 | [TK]D-Fender | Maghteridon: Blind transfer is where you jsut tell it where you want to transfer them to and the call just GOES AWAY on your phone and starts ringing at the destination. |
16:33.56 | Maghteridon | Yes, then I need the first one... is it already implemented or should I configure anything? |
16:34.16 | [TK]D-Fender | Maghteridon: It depends on what kind of phones you are using. Most SIP phones do this INTERNALLY. |
16:34.21 | Maghteridon | On normal phone here in italy it works like pressing the "R" key and the internal number where to transfer to |
16:34.27 | [TK]D-Fender | Maghteridon: Which takes NO configuration to do. |
16:34.40 | [TK]D-Fender | Maghteridon: So what kind of phones are you using? |
16:35.10 | Maghteridon | X-ten (not really me actually... a friend) , and a phone adapter connected to a normal cordless phone |
16:35.13 | *** join/#asterisk gustavo (n=G@87-196-142-229.net.novis.pt) |
16:35.26 | Maghteridon | a lynksys pap2 port adapter |
16:35.40 | ManxPower | I thought X-Lite did not have an attended transfer feature. I thought that feature was only availble in the commervial "Pro" version |
16:36.11 | Maghteridon | It might be actually... I'm not using it now |
16:36.14 | ManxPower | on the Linksys, "R" aka "RECALL" aka "FLASH" key should initate a transfer |
16:36.31 | Maghteridon | So R+number? |
16:36.34 | Maghteridon | or number+R? |
16:36.35 | [TK]D-Fender | x-ten (if you mean X-Lite) does not have a transfer feature. You will need to learn how to use FEATURES.CONF and let * do the work for this one. |
16:36.36 | ManxPower | the Linksys will then handle all the magic to transfer a call. |
16:36.51 | [TK]D-Fender | Maghteridon: For your ATA, it should have its own way of doing this. Check its manual |
16:37.01 | Maghteridon | allright then |
16:37.05 | Maghteridon | and last question... |
16:37.17 | Maghteridon | the music on hold... all I hear is n heavy noise :| |
16:37.27 | [TK]D-Fender | Maghteridon: The PAP2 does have its own way, just read the manual for it |
16:37.43 | ManxPower | Maghteridon: the Music On Hold files may have a sample rate asterisk can't handle. |
16:37.55 | Maghteridon | actuallt look... : |
16:38.11 | ManxPower | "Asterisk: The Perl of PBXs" i.e. there are always several ways to do something. |
16:38.15 | [TK]D-Fender | Maghteridon: And what MODE are they using in "musiconhold.conf"? |
16:38.16 | *** join/#asterisk mihinomenest (i=QWD2@cerebus.clandestineresearch.com) |
16:38.22 | Maghteridon | [default] |
16:38.22 | Maghteridon | mode=files |
16:38.22 | Maghteridon | directory=/var/lib/asterisk/mohmp3/ |
16:38.34 | Maghteridon | it says that cannot find the directory |
16:38.42 | [TK]D-Fender | ManxPower: http://www.sofaswitch.org/d/ |
16:39.00 | [TK]D-Fender | Maghteridon: Does * have UNIX rights to that folder? |
16:39.06 | Maghteridon | Ofcourse |
16:39.43 | [TK]D-Fender | Maghteridon: Also remov the trailing slash |
16:39.53 | ManxPower | Maghteridon: does the user asterisk is running as have rights to that directory and does the directoy actually conain any files? |
16:40.00 | ManxPower | I can't seem to type today |
16:40.00 | riddlebox | [TK]D-Fender, can you help me with my meetme conference problem? |
16:40.12 | [TK]D-Fender | riddlebox: Namely? |
16:40.23 | Maghteridon | ok my bad, I didn't copy fine: here is the actualt string |
16:40.26 | Maghteridon | [default] |
16:40.26 | Maghteridon | mode=files |
16:40.26 | Maghteridon | directory=/var/lib/asterisk/mohmp3/muse |
16:40.34 | gustavo | can someone recomend one or two reliable voip providers in europe? |
16:40.39 | Maghteridon | and the answer is: yes, good permissions and real file existing |
16:40.48 | gustavo | there are so many, it's hard to test them all |
16:41.04 | Maghteridon | gustavo, where are you from btw? |
16:41.14 | [TK]D-Fender | Maghteridon: If it finds your files and they sound like crap, check the encording rate, and make sure they aren't VBR. |
16:41.14 | gustavo | Maghteridon: Portugal |
16:41.19 | drfreeze | While trying to install asterisk-1.4.2 I get the following: configure: error: *** termcap support not found |
16:41.24 | Maghteridon | ah portugal then nothing, tought ou were italian |
16:41.39 | Maghteridon | ... VBR...? =) |
16:41.41 | drfreeze | One source says that libncurses5 should fix that |
16:41.49 | gustavo | it's not clear I should go with a local provider |
16:41.52 | riddlebox | [TK]D-Fender, let me pastebin it |
16:42.10 | drfreeze | /usr/lib/libncurses.so.5 /usr/lib/libncurses.so.5.4 /usr/lib/libncursesw.so.5 /usr/lib/libncursesw.so.5.4 |
16:42.27 | Maghteridon | btw ManxPower is that entry in musiconhold.conf right? |
16:43.37 | riddlebox | [TK]D-Fender, http://pastebin.ca/418400 |
16:45.27 | Maghteridon | ManxPower, it says actually: Unable to open file '/var/lib/asterisk/mohmp3/muse/massive' |
16:45.32 | Maghteridon | no such file or directory... but it exists... |
16:48.55 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
16:52.10 | stony | hmm how can i print variables with NoOp() ? |
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16:52.30 | stony | NoOp(${CALLERID(num)} only shoes "CALLERID(num)" in the cli |
16:52.44 | stony | s/shoes/shows |
16:52.59 | blitzrage | missing a closing brace |
16:53.06 | blitzrage | and you should use Verbose() |
16:53.12 | *** join/#asterisk boojit (n=boojit@gw.carter.to) |
16:53.23 | blitzrage | Verbose(1|${CALLERID(num)}) is better |
16:53.27 | stony | blitzrage: yeah typo here... |
16:53.37 | blitzrage | yah... like I said :) |
16:53.56 | stony | blitzrage: in the config it's written correctly and still not working |
16:55.05 | blitzrage | something else is wrong... because that works |
16:55.08 | boojit | hi folks |
16:55.21 | stony | blitzrage: the verbose() function works fine ... using that... |
16:55.34 | blitzrage | NoOp() is for no operation, not really for debugging |
16:55.39 | *** join/#asterisk Innatech (n=daf@netblock-72-25-97-119.dslextreme.com) |
16:56.17 | boojit | I'm doing some discovery for a client and I'm wondering if anyone here has experience doing systems for voter response. |
16:56.49 | boojit | not using SMS, but rather just regular phone access. |
16:57.09 | *** join/#asterisk chrisknight (n=explodin@cpe-71-79-81-174.columbus.res.rr.com) |
16:57.12 | riddlebox | is this the correct way to do meetme http://pastebin.ca/418400 ? |
16:57.16 | drfreeze | Anyone know how to solve this problem? configure: error: *** termcap support not found |
16:57.46 | russellb | drfreeze: install libncurses-dev |
16:58.02 | chrisknight | yum install libtermcap libtermcap-devel newt newt-devel ncurses ncurses-devel |
16:58.53 | boojit | If anyone has any experience with such systems in a high-availability large-scale implementation, please ping me. |
16:59.03 | drfreeze | chrisknight: ahh, it wants the devel packages. Trying now |
16:59.15 | chrisknight | When all the "default" responses dont work, how can I factory reset a cisco 7960 phone? |
16:59.24 | drfreeze | chrisknight: thx. that worked |
16:59.32 | chrisknight | no prob |
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17:00.54 | chrisknight | **# does not unlock the network config... Holding down # during reboot wont work either. Someone has set a password on the network config as well as telnet... |
17:05.48 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
17:09.30 | ManxPower | chrisknight: Cisco has documents on their web site about how to unlock their phones |
17:11.31 | chrisknight | I know, I believe I have tried all of them. This thing must have some "special" load... |
17:11.52 | drfreeze | Is zaptel required for an VoIP only asterisk install |
17:12.27 | ManxPower | drfreeze: no. neither is a sound card |
17:14.24 | russellb | neither are glow in the dark IDE cables |
17:14.49 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
17:18.04 | coppice | but glow in the dark SATA cables are this year's must have |
17:21.20 | Qwell | crap, I better get some then |
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17:21.25 | *** mode/#asterisk [+o anthm] by ChanServ |
17:22.06 | uwe | um, what value does astdb use for enabling/diabling dnd ? is it YES or ON and Off or NO or by deleting the record ? i mean which does asterisk use to decide if calls should be sent to that user or not ? |
17:22.58 | blitzrage | uwe: it depends how you want to implement it in your dialplan |
17:23.11 | Qwell | (or chan_skinny) |
17:23.23 | blitzrage | GotoIf($[${DB(user/dnd)} = 1]?dnd,1) |
17:23.23 | Qwell | actually, maybe not |
17:23.26 | blitzrage | as an example |
17:23.27 | *** part/#asterisk bsd_tech (n=bsdtech@ppp-69-238-48-150.dsl.irvnca.pacbell.net) |
17:24.08 | blitzrage | GotoIf($[${DB(user/dnd)} = OFF]?dnd,1) is just has valid |
17:24.16 | blitzrage | s/has/as/ |
17:25.11 | uwe | hmm |
17:25.47 | uwe | thank you blitzrage |
17:26.43 | Qwell | blitzrage: why not like this? GotoIf($[${DB(${ODBC_GETUSER(${EXTEN})}/dnd)} = ${ODBC_GETOFFVALUE()}]?dnd,1) |
17:26.46 | Qwell | :p |
17:27.43 | Corydon76-home | Heh |
17:27.57 | Qwell | russellb: that's nothing |
17:28.09 | Qwell | ODBC_GETUSER is a 40 line query that uses 20 tables |
17:28.11 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
17:28.31 | Qwell | ooo |
17:28.34 | Qwell | we need 'in' |
17:28.52 | Qwell | GotoIf($[${somevar} in ${MYHASH}]) |
17:29.17 | Qwell | GotoIf($[${DB(${ODBC_GETUSER(${EXTEN})}/dnd)} in ${ODBC_GETOFFVALUEs()}]?dnd,1) |
17:29.40 | Qwell | where's murf? heh |
17:30.50 | blitzrage | Qwell: because it seems kinda dumb to use func_odbc to lookup a value in AstDB? :) |
17:30.58 | Qwell | You must be new here. |
17:31.28 | blitzrage | :) |
17:31.28 | Qwell | blitzrage: it's funny - laugh :) |
17:32.05 | Qwell | GotoIf($[${ODBC_IsDND(${EXTEN})}]?dnd,1) |
17:32.06 | Qwell | :) |
17:32.22 | blitzrage | :) |
17:32.36 | Corydon76-home | Hey, that's lowercase... |
17:32.42 | blitzrage | FIRED! |
17:32.46 | Qwell | Corydon76-home: yeah... |
17:32.58 | Corydon76-home | Bad convention! |
17:33.10 | *** part/#asterisk chrisknight (n=explodin@cpe-71-79-81-174.columbus.res.rr.com) |
17:33.18 | Qwell | sure, and I'm really strict when doing SQL related stuff usually.. just for readabilities sake in this case |
17:33.37 | Qwell | I can't stand it when people use lowercase "select" in queries |
17:33.48 | russellb | select *; |
17:33.52 | Corydon76-home | heh |
17:34.10 | Qwell | russellb: no FROM clause :D |
17:34.15 | russellb | pwned! |
17:34.16 | Corydon76-home | Actually, I use lowercase keywords when morons capitalize all their table names and column names |
17:34.32 | Qwell | Corydon76-home: I use uppercase table names ALL the time :p |
17:34.43 | file | SELECT russellb FROM world; |
17:34.50 | russellb | i got really irritated the other day when I realized I couldn't do the equivalent of "SELECT * FROM whatever;" through the realtime API |
17:35.20 | russellb | I modified res_config_odbc to let you do it, but haven't finished changing all of the others ... |
17:35.30 | *** join/#asterisk mrbnet (n=sureal@corpmail1.mrbnetworks.com) |
17:35.50 | russellb | if it worked, then all of the "sip show peers", "voicemail show users", etc. CLI commands could work with realtime |
17:36.08 | russellb | </small rant> |
17:36.42 | blitzrage | russellb: :-O |
17:37.04 | Qwell | russellb: You can't just set no WHERE clause? |
17:37.14 | *** join/#asterisk stony (n=steinche@p5B15196E.dip0.t-ipconnect.de) |
17:37.14 | russellb | correct, you have to have at least one |
17:37.17 | Qwell | lame! |
17:37.38 | Qwell | well |
17:38.04 | Qwell | int one=1;\nsomefunction(table, one, 1); |
17:38.24 | Qwell | that would probably fail miserably on some non-DB things |
17:40.39 | russellb | yes, kevin said that, too |
17:40.48 | russellb | I said, what about LDAP? :) |
17:40.54 | Qwell | yeah |
17:41.00 | Qwell | stupid ldap :p |
17:41.09 | russellb | it's not even in the tree ... |
17:41.24 | russellb | someone should probably take that on to get it done by 1.6 ... |
17:41.40 | ManxPower | russellb: it is nice of your to volunteer |
17:42.02 | russellb | heh |
17:42.21 | russellb | just tack it on my list |
17:42.52 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-db3b326591e9148d) |
17:42.52 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
17:42.58 | Qwell | Cresl1n: !!! |
17:43.08 | Cresl1n | Qwell !!!!! |
17:43.11 | Cresl1n | are you in the office today? |
17:43.14 | Qwell | nah |
17:43.19 | Cresl1n | lame..... |
17:43.23 | Cresl1n | everybody cool is here :-P |
17:43.38 | *** join/#asterisk ToyMan (n=Stuart@74-32-55-210.dsl1.mdl.ny.frontiernet.net) |
17:43.38 | Qwell | You're there by yourself? :) |
17:43.43 | Cresl1n | heh :-) |
17:43.46 | Qwell | (kidding) :) |
17:43.55 | blitzrage | Everyone cool and Cresl1n too |
17:44.00 | Qwell | blitzrage: pwned |
17:44.00 | ManxPower | Cresl1n: Does Zaptel support DACS? |
17:44.02 | Cresl1n | nah, spiceland and rick loveman are here too |
17:44.04 | russellb | bkruse and I were thinking about going there and having a jam session, heh |
17:44.08 | Qwell | Cresl1n: ahh |
17:44.10 | Cresl1n | ManxPower: oh no you didn't! |
17:44.18 | Cresl1n | how many times do you have to explain it to someone |
17:44.20 | Qwell | russellb: brt, I play a MEAN cowbell |
17:44.28 | russellb | sweet! |
17:44.34 | russellb | I have a couple cowbells I could bring. |
17:44.38 | Qwell | haha, seriously? |
17:44.39 | russellb | except ... they are in storage ... |
17:44.42 | Cresl1n | well, I explained I think well enough so that next time I can just tell the person to read the archives |
17:44.44 | Qwell | I was totally joking |
17:44.59 | russellb | lame |
17:46.01 | blitzrage | Zaptel supports DACS? |
17:46.12 | ManxPower | Cresl1n: We used to use DACS extensively. |
17:46.19 | Qwell | I need DACS! Does Zaptel support it? What is DACS?! |
17:46.32 | russellb | I totally need DACS for my office |
17:46.36 | russellb | else I can't work |
17:46.40 | russellb | and I blame Cresl1n |
17:46.44 | ManxPower | Qwell: Sangoma did not support DACS RBS until I complained to them and they fixed it in a driver update. |
17:46.58 | Qwell | ManxPower: well, what'd you go and do that for? |
17:47.16 | russellb | (I don't know what DACS is ...) |
17:47.32 | ManxPower | That was the only significant issue we have had with Sangoma. And yes, we have had HDLC abort errors with a Sangoma card, so we know it DOES happen. |
17:48.16 | ManxPower | russellb: Basically you digitally cross connect 1 T-1 channel to another T-1 channel. Sort of a digital patch panel at the 64k/56k channel level |
17:48.41 | russellb | ah, neat ... |
17:48.48 | *** join/#asterisk Insane00 (n=aamirwah@74-128-211-33.dhcp.insightbb.com) |
17:48.50 | Qwell | used for what? |
17:48.54 | russellb | I heard Sangoma eats babies |
17:49.01 | ManxPower | handy for example if you have a mixed voice and data T-1. You can DACS the data channels out another T-1 port into whatever data device you have. |
17:49.04 | Qwell | russellb: Well, they are Canadian |
17:49.09 | Qwell | ...fast |
17:49.24 | blitzrage | Qwell: I can run faster than you |
17:49.31 | Qwell | probably |
17:49.33 | Insane00 | Hello |
17:49.33 | russellb | ManxPower: oh, alright, makes sense |
17:49.34 | ManxPower | It is VERY handy when putting Asterisk between the Telco and other T-1 equipment like channel banks, routers, and PBXs with T-1 ports |
17:49.35 | blitzrage | I'm sure of it :) |
17:49.42 | Qwell | but, luckily, I've got a 3k mile headstart :P |
17:50.03 | ManxPower | DACS is WHY we were able to transition to Asterisk for a significant number of our phones |
17:51.13 | ManxPower | Qwell: We tried Sangoma because of the long standing issue Digium cards have had with a large enough number of motherboards as to cause us issues. |
17:51.36 | Qwell | ManxPower: You shouldn't have those problems anymore |
17:52.04 | ManxPower | As most people know, many of those issues are supposed to be resolved in recent Zaptel releases, but we were committed to Sangoma well before those Zaptel releases |
17:52.24 | Insane00 | I need help to undedrstand asterisk ? any body |
17:52.36 | ManxPower | Insane00: you're insane. |
17:52.45 | Insane00 | thats why i am here |
17:52.54 | russellb | lol |
17:53.07 | russellb | I can tell you Asterisk starts with an A |
17:53.13 | russellb | ~thebook |
17:53.25 | jbot | it has been said that thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
17:53.25 | Qwell | ~book |
17:53.27 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
17:53.28 | Qwell | jbot: you suck |
17:53.29 | jbot | no, *you* suck! |
17:53.34 | ManxPower | Insane00: I assume you also want to start your own VoIP company next week. |
17:53.34 | Qwell | wtf |
17:53.35 | Insane00 | did i asked you about spelling |
17:53.48 | Insane00 | we are actually |
17:54.06 | Insane00 | but i want to jump in to know some basics |
17:54.07 | russellb | Qwell: Can you be helping me be the next Vonage? |
17:54.46 | Qwell | Insane00: such as? |
17:54.53 | Qwell | We can't answer such a broad question |
17:55.16 | ManxPower | Insane00: The Book is a good start. PBXs are complex devices. Linux is a complex system. Networking is a complex thing. Telco interconnects are complex. Asterisk requires knowledge of all of these things |
17:55.50 | Insane00 | hmm wow |
17:55.51 | Qwell | correction: Telco interconnects are incredibly complex |
17:55.52 | Insane00 | i know linux |
17:56.38 | Insane00 | but i think if you guys are hel me to understand the how asterisk route the calls |
17:56.44 | ManxPower | BTW, www.sandman.com has lots of really cool telco stuff for very good prices. Including things that take CPC tones and turn them into a battery drops, etc |
17:57.24 | Insane00 | or in other words when a user dial a local number how its terminated locally |
17:57.34 | ManxPower | http://www.sandman.com/loop.html#CPCGenerator |
17:58.07 | Cresl1n | Yeah, I got some cool new firmware for the TE410P/TE405P as well |
17:58.12 | Cresl1n | hot rod firmware |
17:58.19 | Cresl1n | we're beta testing it right now |
17:58.27 | Insane00 | i will really apreciate if some one explain me |
17:59.02 | *** join/#asterisk boch (n=fran@190.48.225.254) |
17:59.04 | ManxPower | I'll be using them for things like emergency phones at clients with only 1 local line. |
18:00.22 | ManxPower | The site also has things like impedance matching devices. |
18:00.23 | drfreeze | To use tftp to provision phones, the DHCP server doesn't need to be run from the Asterisk box, right? |
18:00.53 | *** join/#asterisk thoughtpolice (n=austin@c75-111-145-138.plaicmtc01.tx.dh.suddenlink.net) |
18:00.59 | ManxPower | drfreeze: no. |
18:01.11 | boch | is it possible to make Record() stop recording with any key instead # ? |
18:01.14 | drfreeze | ManxPower: thx |
18:01.43 | ManxPower | you just have to have the DHCP server add the correct option in it's response. drfreeze: Without the DHCP option "next server" is one of them, many phones will just default to loading their configs from the DHCP server. |
18:01.50 | ManxPower | Ciscos do this AFIK |
18:03.15 | Cresl1n | it should help with motherboard compatibilty issues as well |
18:03.18 | ManxPower | Qwell you might find this interesting. I got a line tester out of storage. Calling a nearby 105 Test number (104Hz tone at 1db), my test set is showing a -10 to -13 db level at my 66 block |
18:03.42 | *** join/#asterisk ming_zym (n=ming_zym@124.254.55.207) |
18:03.53 | ManxPower | I suspect I have an impedance issue as I get echo even on TDM bridged Zap calls. |
18:03.57 | Qwell | Insane00: That isn't really a question that can be answered |
18:04.09 | Cresl1n | ugh |
18:04.13 | Cresl1n | ManxPower: run fxotune |
18:04.38 | ManxPower | Cresl1n: on a T-1 card. |
18:04.40 | ManxPower | the analog lines are coming into an Adtran |
18:04.58 | Cresl1n | ManxPower: oh, that's a whole different colored horse then |
18:05.32 | ManxPower | Cresl1n: I am 11,000 ft from the CO and going thru a SLC96 |
18:05.43 | un_j | :-) |
18:05.57 | Insane00 | Qwell: any book you recommnad |
18:06.02 | ManxPower | so basically the worst possible case for PBX |
18:06.06 | Qwell | Insane00: The one jbot said..twice |
18:06.26 | un_j | hey how to set up timeouts for iax (drops after 60 seconds) and if I call from sip ext to sip ext it drops also after 60 seconds :-) |
18:06.34 | blitzrage | someone wrote a book? |
18:06.40 | blitzrage | those guys must be super cool |
18:06.49 | Qwell | blitzrage: sooo cool |
18:06.52 | ManxPower | un_j: we can't help you with FreePBX and/or AMP |
18:06.55 | russellb | omg I wish I could meet them |
18:06.58 | Qwell | somebody should give them a lot of money |
18:07.02 | blitzrage | totally |
18:07.27 | blitzrage | I heard they are building their own space station |
18:07.49 | ManxPower | blitzrage: for total world domination, I assume? |
18:07.55 | un_j | I don't user free pbx its 1.4.2. pure no gui |
18:07.57 | blitzrage | most likely |
18:08.28 | ManxPower | un_j: Good. then we can help you 8-) |
18:08.59 | un_j | please ;-) |
18:10.21 | ManxPower | un_j: IAX and SIP should not timeout unless they are not getting any response from the far end or you have a timeout on the Dial line |
18:12.10 | un_j | hm, its gets: Auto fallthrough, channel 'IAX2/chicago-3' status is 'CHANUNAVAIL' |
18:15.24 | un_j | how about rtptimeout? |
18:22.01 | zoa | yoyo |
18:28.34 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
18:33.37 | JacksLivr | how do i make a channel hangup from the console? |
18:33.54 | blitzrage | soft hangup |
18:34.25 | JacksLivr | blitzrage: THANK YOU!!!!!!!!!!!! <3 |
18:34.39 | JacksLivr | so easy, just not where i was looking |
18:40.52 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
18:41.06 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
18:42.06 | un_j | hey sip is ok but iax gets: [Mar 31 13:38:38] DEBUG[11742] chan_iax2.c: Received iseqno 12 not within window 13->15 |
18:42.06 | un_j | <PROTECTED> |
18:42.18 | un_j | 60 sorry |
18:42.30 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
18:42.33 | *** join/#asterisk eald_home (n=eald@189.157.104.40) |
18:52.40 | *** join/#asterisk techie (n=gus@antibala.com) |
18:54.14 | *** join/#asterisk techie (n=gus@voip.routedsystems.com) |
19:04.58 | robin_sz | sigh ... echo is beginning to bug me bigtime |
19:05.17 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
19:05.22 | robin_sz | chan_mISDN is the culprit I suspect |
19:06.19 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
19:11.46 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
19:12.23 | *** join/#asterisk haagenti (n=user@212.91.233.233) |
19:15.17 | haagenti | can asterisk do ss7/camel yet? |
19:16.11 | ManxPower | un_j: That might indicate a serious networking issue. |
19:17.14 | ManxPower | haagenti: Short Answer: No. Long Answer: Yes, but it is from some 3rd party and (I think) is commercial. |
19:17.33 | *** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net) |
19:18.34 | haagenti | hmmm... just curious |
19:19.02 | d00gster | guys are dial plans universal? I mean a dial plan for asterisk works on sipura? |
19:19.37 | Qwell | d00gster: no, ATAs have their own dialplans |
19:19.41 | *** join/#asterisk piper69 (n=piper@unaffiliated/piper69) |
19:19.48 | piper69 | hi all |
19:19.56 | d00gster | I see |
19:20.03 | Qwell | d00gster: there are some good examples on the wiki though |
19:21.13 | piper69 | i use debian etch i am trying to install asterisk i get Setting up asterisk-sounds-main (1.2.13~dfsg-2) ... |
19:21.14 | piper69 | Setting up fxload (0.0.20020411-1) ... |
19:21.16 | piper69 | Setting up zaptel (1.2.11.dfsg-1) ... |
19:21.19 | piper69 | Zaptel telephony kernel driver: FATAL: Module ztdummy not found. |
19:21.21 | piper69 | Notice: Configuration file is /etc/zaptel.conf |
19:21.24 | piper69 | line 0: Unable to open master device '/dev/zap/ctl' |
19:21.26 | piper69 | 1 error(s) detected |
19:21.29 | piper69 | /sbin/ztcfg failed. Check /etc/zaptel.confzaptel. |
19:21.31 | piper69 | Setting up asterisk-classic (1.2.13~dfsg-2) ... |
19:21.34 | piper69 | Setting up asterisk (1.2.13~dfsg-2) ... |
19:21.34 | Qwell | ~paste |
19:21.43 | jbot | somebody said paste was http://rafb.net/paste/ |
19:21.50 | piper69 | sorry |
19:22.22 | piper69 | i do apologize |
19:23.04 | *** join/#asterisk Zaw (i=zaw@unaffiliated/zaw) |
19:24.57 | JunK-Y | Qwell: so how's the newest ubuntu? |
19:25.10 | Qwell | JunK-Y: incomplete |
19:25.22 | Qwell | there are quite a few packages missing |
19:25.26 | JunK-Y | like? |
19:25.28 | Qwell | otherwise, it's the same as edgy |
19:25.37 | JunK-Y | is it ubuntu or kubuntu? |
19:25.40 | Qwell | like, vmware-server, some amd64 things |
19:25.43 | Qwell | ubuntu |
19:25.45 | piper69 | Qwell: if you have time can you help me please i pasted my issue http://rafb.net/p/x95dil59.html |
19:25.49 | Qwell | some kernel modules |
19:25.52 | Qwell | audio is b0rked for my card |
19:26.11 | *** join/#asterisk Ebola (n=Ebola@host86-136-190-11.range86-136.btcentralplus.com) |
19:26.13 | JunK-Y | kk |
19:26.32 | Qwell | oh, and compiz/beryl is hosed |
19:26.51 | JunK-Y | ive heard hood things about beryl |
19:32.35 | *** join/#asterisk kannan (n=kannan@210.211.178.104) |
19:32.36 | Qwell | are you saying it's ghetto? |
19:32.49 | Qwell | I *love* typos that make the sentence mean something completely different, heh |
19:33.11 | JunK-Y | s/hood/good/ |
19:33.12 | JunK-Y | :) |
19:33.25 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
19:33.29 | JunK-Y | there ya go jbot , you rock! |
19:36.29 | blitzrage | Qwell: good stuff |
19:37.20 | *** join/#asterisk Innatech (n=daf@netblock-72-25-97-119.dslextreme.com) |
19:37.42 | drfreeze | Does anyone know what version of version of BootROM and SIP SW I should be using for Polycom? |
19:37.50 | drfreeze | is 1.6.7 ok for SIP SW? |
19:40.57 | Qwell | russellb: ping? |
19:41.14 | Dovid | 1.6.7 should be good |
19:41.17 | Dovid | i use it on my phones |
19:42.23 | drfreeze | Dovid: thx. and bootrom 2.6.1 ok? |
19:42.39 | Dovid | dont have a phone wit me now. dont know what boot rom i am using. |
19:42.57 | Dovid | the SIP 2.x.x is what made and issue with asterisk 1.2.x |
19:43.03 | Maghteridon | Is there a way from the CLI to record a conversation that a user has with an external line? |
19:43.42 | Dovid | Maghteridon: i dont think so. but u can spy in to a conversation and record it from there |
19:43.42 | drfreeze | Dovid: what about asterisk 1.4.2? is it ok with SIP 2 |
19:43.48 | Dovid | or u should be able |
19:44.08 | Dovid | dr: i heard with 1.4.2 it's good but i never touched 1.4.x yet |
19:44.14 | Maghteridon | Dovid, do you have to be on the asterisk box to do it? |
19:44.28 | Dovid | to spy |
19:44.29 | Dovid | ? |
19:44.30 | Maghteridon | yes |
19:44.34 | Dovid | idk if u can record from spying |
19:44.48 | Maghteridon | how do you spy? |
19:44.49 | Dovid | no. it's a feature on asteirsk. u can do it from any phone |
19:44.51 | *** join/#asterisk lung (n=lung@24-148-96-186.ip.mhcable.com) |
19:44.56 | Dovid | look on the wiki |
19:45.02 | Dovid | i think the cmd is ChanSpy |
19:45.06 | Maghteridon | ah okay thanks |
19:46.08 | lung | ive got asterisk 1.2.17 core dumping on three seperate machines.. is this a proper outlet to dsicuss? |
19:47.20 | JunK-Y | lung: can ya reproduce the problem at any time? |
19:47.28 | JunK-Y | <PROTECTED> |
19:47.28 | JunK-Y | <PROTECTED> |
19:47.30 | JunK-Y | oups |
19:48.08 | lung | JunK-Y: only in the sense that it happens so often, but no, im not seeing what exactly is triggering it.. its happening very often across all three servers though.. at least hourly |
19:48.29 | JunK-Y | lung: read the file: README.backtrace |
19:48.34 | JunK-Y | in ur doc/ dir |
19:49.27 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
19:49.33 | lung | i have many cores :p and i compiled valgrind |
19:50.15 | JunK-Y | try to see if theres no duplcate already existing bugs first at bugs.digium.com |
19:50.21 | lung | yeap |
19:50.31 | zoa | http://www.autoblog.com/2007/03/30/bulgaria-gets-a-loan-to-buy-fleet-of-porsche-cayenne-ambulances/ |
19:50.33 | zoa | loool |
19:50.34 | zoa | stupid fucks |
19:51.19 | Qwell | zoa: Where are you from? I always get those B's mixed up... |
19:51.30 | zoa | im from belgium |
19:51.32 | Qwell | Bulgaria, Bolivia, Belgium.. |
19:51.33 | zoa | but living in bulgaria |
19:51.34 | Qwell | ahh, okay |
19:51.38 | zoa | o ho ho |
19:51.44 | zoa | im coming to the devcon!!! |
19:51.45 | zoa | whiii |
19:51.58 | Qwell | I told redbull they need to sponsor us just now :P |
19:52.01 | JunK-Y | zoa: you live in bulgaria? didnt know taht. |
19:52.08 | Qwell | how freaking awesome would free redbull be? |
19:52.17 | JunK-Y | zoa: i wont be able to afford it sadly :( |
19:52.18 | zoa | i dont drink redbul |
19:52.22 | zoa | its killing me |
19:52.25 | Qwell | zoa: bah! |
19:52.29 | Qwell | quit for a month |
19:52.30 | mvanbaak | Qwell: any change there will be Jolt ? |
19:52.35 | Qwell | mvanbaak: none |
19:52.38 | zoa | i sometimes drink it |
19:52.40 | Qwell | Jolt is illegal in Georgia |
19:52.44 | zoa | but it makes my heart go crazy |
19:52.45 | mvanbaak | really ? |
19:52.47 | Qwell | no |
19:52.47 | mvanbaak | why ? |
19:52.57 | zoa | JunK-Y: where do you live ? |
19:53.04 | mvanbaak | it's hard to get here in .nl |
19:53.12 | mvanbaak | I always go to Germany to get it |
19:53.17 | Qwell | mvanbaak: isn't redbull too? |
19:53.19 | zoa | mvanbaak: je kan het online vinden denk ik :) |
19:53.23 | Qwell | or was that...norway? |
19:53.24 | zoa | redbull is very easy to find |
19:53.25 | Qwell | stupid N's! |
19:53.25 | mvanbaak | zoa: klopt |
19:53.27 | JunK-Y | zoa: im in montreal, quebec in canada. |
19:53.28 | zoa | they have it everywhere |
19:53.31 | zoa | aha |
19:53.37 | mvanbaak | Qwell: redbull is legal here |
19:53.49 | Qwell | somebody said it was illegal in some country that started with an N, I thought |
19:53.52 | Qwell | mog would know |
19:54.00 | zoa | Qwell: everything is legal in the netherlands |
19:54.05 | zoa | they sell magic mushrooms there |
19:54.06 | mvanbaak | lol |
19:54.08 | Dovid | damn |
19:54.10 | mvanbaak | yeah! |
19:54.12 | mvanbaak | shrooms ! |
19:54.13 | Dovid | zoa: when can i move ? |
19:54.20 | zoa | you dont want to |
19:54.21 | zoa | :) |
19:54.32 | LennonNZ | does anyone have a working configure to set up sipdiscount.com to asterisk |
19:54.38 | LennonNZ | all I'm getting at the moment from them is |
19:54.42 | LennonNZ | Got SIP response 500 "Internal server error" back from 80.239.235.200 |
19:54.54 | mvanbaak | I get that a lot from sipdiscount as well |
19:54.56 | mvanbaak | it's them |
19:55.05 | mvanbaak | because sometimes it works out of the blue |
19:55.23 | LennonNZ | I had it working.. and then suddenly I get this all the time |
19:55.30 | mvanbaak | yeah |
19:55.33 | mvanbaak | get used to it |
19:55.42 | mvanbaak | happens at least twice a week with them |
19:56.02 | LennonNZ | I am looking at getting some free callin/out numbers |
19:56.08 | Cresl1n | redbulll....... |
19:56.15 | LennonNZ | there is a list os free providers anyplace I can get numbers/connetions from? |
19:56.27 | Qwell | Cresl1n: :D |
19:56.33 | mvanbaak | LennonNZ: remember that you get what you pay for |
19:56.35 | JunK-Y | Cresl1n: ya just heard the magic word huh?! |
19:56.36 | Qwell | somebody tried that last year, didn't they? |
19:56.49 | LennonNZ | mvanbeek: true |
19:56.57 | Cresl1n | oh yeah |
19:57.22 | zoa | hey look its Cresl1n |
19:57.25 | zoa | its a live |
19:57.25 | zoa | alive |
19:57.30 | Cresl1n | zoa!~!!! |
19:57.35 | zoa | Cresl1n: are you taking your wife ? |
19:57.44 | zoa | i showed you mine now show me yours |
19:57.46 | zoa | :P |
19:57.54 | mvanbaak | ehm...... |
19:57.54 | LennonNZ | is there anyway I can play music and a "ring ring".. "ring ring" sound at the same time whilst calling someone bfore they answer the phone? |
19:57.55 | mvanbaak | lol |
19:58.23 | zoa | SHOW ME THE GOODIES! |
19:58.24 | zoa | :) |
19:58.24 | LennonNZ | at the moment I can do either, but not both |
19:58.32 | mvanbaak | the only way to do that is to put that ringring in your music |
19:58.35 | mvanbaak | mix it together |
19:59.12 | LennonNZ | oh well |
19:59.31 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
20:02.02 | dan42 | JunK-Y: most of the lines in my backtrace are "in ?? ()" .. am i doing something wrong? |
20:02.21 | blitzrage | dan42: did you compile with DONT_OPTIMIZE? |
20:02.30 | JunK-Y | dan42: you need make dont-optimize |
20:02.34 | dan42 | blitzrage: 1.2.17 compiled valgrind |
20:02.43 | blitzrage | 1.2: make dont-optimize ; 1.4: Set DONT_OPTIMIZE in menuselect |
20:02.43 | dan42 | which does the same thing |
20:03.05 | dan42 | valgrind: dont-optimize |
20:03.23 | JunK-Y | have ya started with -g too? |
20:03.27 | dan42 | seems like its lines that would be in the libs outside of asterisk |
20:03.29 | dan42 | yes |
20:03.54 | dan42 | i dont think i would have a core otherwise, right? |
20:04.01 | dan42 | but yes all thew same |
20:04.22 | dan42 | i have my output from gdb, but its not leading ME anywhere |
20:04.36 | dan42 | not that that means anything |
20:05.50 | *** join/#asterisk nighty^^ (n=nighty@sushi.rural-networks.com) |
20:11.14 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1088929080.dsl.bell.ca) |
20:11.54 | *** join/#asterisk harleya (n=harleya@c-67-161-253-232.hsd1.ut.comcast.net) |
20:12.15 | robin_sz | ok, so to improve the crap echo performance of chan_mISDN I should? |
20:12.30 | drfreeze | Is there a way to have an auto provisioning file for all 301 phones and one for all 501 phones? |
20:12.36 | *** join/#asterisk rad07 (i=raca@64-126-95-37.static.everestkc.net) |
20:12.48 | drfreeze | robin_sz: call digium? |
20:12.58 | robin_sz | drfreeze, do they retreive their configs from a web swerever? |
20:13.09 | drfreeze | don't know |
20:13.12 | d00gster | guys, would can I force international calls to dual a callingcard DiD before the actual number? |
20:13.35 | robin_sz | d00gster, yes |
20:14.04 | zoa | the HPEC might work on misdn |
20:14.08 | zoa | check with creslin |
20:14.09 | d00gster | how do I do that for oubound calls? |
20:14.13 | zoa | he should know |
20:14.28 | Qwell | zoa: do any software echo cans work with misdn? |
20:14.31 | Qwell | If so, hpec should work just fine |
20:14.35 | Cresl1n | nah, HPEC doesn't work with misdn |
20:14.41 | Cresl1n | there'd be licensing issues anyways |
20:14.47 | Cresl1n | misdn kernel code is only GPL |
20:14.52 | Cresl1n | it'd be ugly |
20:15.13 | robin_sz | d00gster, inyour dialplan, just test for the first digits of an international number, and then put the calling card numbers before the EXTEN |
20:15.34 | robin_sz | use includes to include that pattern before the local patterns |
20:16.09 | robin_sz | there are several examples of this in the default config |
20:16.29 | dan42 | JunK-Y: thanks for the help, just wanted to make sure i did that right.. i just posted a bug.. didnt see another one like it |
20:16.39 | robin_sz | zoa, HPEC? |
20:16.43 | d00gster | robin_sz I need that for outbound calls |
20:16.54 | robin_sz | there are several examples of this in the default config |
20:17.12 | d00gster | ok |
20:18.01 | zoa | Cresl1n: doesnt have to be a problem the octasic people seem to have gotten around that |
20:18.14 | JunK-Y | dan42: <value optimized out> so isnt dont-optimize |
20:18.18 | *** join/#asterisk Corydon76-home (i=brown@pdpc/supporter/sustaining/Corydon76-home) |
20:18.18 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
20:18.29 | robin_sz | so, basically ... unless chan_mISDN suddenyl has a new burst of code ... I need to look for a real card, rather than this cologne card? |
20:18.47 | JunK-Y | dan42: and we need a bt full too. |
20:18.49 | rad07 | Everybody: I installed Asterisk 1.4.2. Any Web interface that works with it? I didn't install samples and I need to connect Linksys ATA SPA3102. I know where is web based config page for my ATA, but I am not sure of the steps I need to do to setup Asterisk to accept/receive calls via ATA. Any step by step guides? If I have to do it manually which files (I assume after installing Samples) should I configure? Please help me. I asked this q |
20:19.08 | dan42 | JunK-Y: its all there.. "bt", "bt full", and "thread apply all bt" |
20:19.52 | Qwell | dan42: from a non-optimized build |
20:19.52 | robin_sz | rad07, configure you sip.conf ... as show in the book |
20:19.52 | *** join/#asterisk Assid (n=assid@203.212.204.107) |
20:19.53 | dan42 | Qwell: its compiled non-optimized |
20:19.53 | JunK-Y | oops, sorry, miss the first part. |
20:20.46 | *** join/#asterisk boch (n=fran@190.48.209.242) |
20:21.14 | rad07 | robin_sz: Which book? Is it the first step to install Samples? After that what I configured actually? Is it just sip.conf that I need to touch? |
20:21.24 | Qwell | ~book |
20:21.30 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
20:21.30 | Qwell | that one |
20:21.31 | Qwell | ... |
20:21.32 | Qwell | that one |
20:21.46 | dan42 | Qwell: i did "make valgrind" actually since thats what was in voip-info .. but that seems to just run dont-optimize from what i can tell |
20:21.53 | Qwell | dan42: yeah |
20:22.01 | rad07 | Is it still relevant (accurate) for latest Asterisk version? |
20:22.11 | Qwell | rad07: mostly |
20:22.31 | Qwell | a new book will be published eventually |
20:22.53 | rad07 | 1. Install samples. 2. Configure sip.conf, Next??? |
20:23.06 | Qwell | rad07: Profit!!! |
20:23.13 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-179-77.ny325.east.verizon.net) |
20:23.20 | dan42 | heh |
20:23.23 | robin_sz | connfigure your ata to connect to your * server, using the name/pw you set in sip .conf |
20:23.26 | rad07 | It cannot be that simple |
20:23.31 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
20:23.34 | dan42 | is that res_profit or app_profit? |
20:23.49 | robin_sz | step 1) steal underpants |
20:23.53 | robin_sz | step 2) .... |
20:23.58 | robin_sz | step 3) profit! |
20:24.43 | Qwell | dan42: chan_prophet |
20:25.28 | robin_sz | so ... where was I? a single hfc/cologne card // what are my options other than chan_mISDN |
20:25.51 | rad07 | Guys. Does anybody teach Asterisk? I am willing to pay small amount of money to learn basics. I need to integrate Web based conferencing with Asterisk and I need to be able to compile some modules/packages from third party? Anybody with programming knowledge? |
20:25.56 | Qwell | robin_sz: no hardware echo can for those cards? |
20:26.37 | Cresl1n | robin_sz, is it not a b410p? |
20:26.42 | Cresl1n | they have HW echo can on them |
20:26.42 | robin_sz | absolutely not |
20:26.53 | robin_sz | b410p? |
20:26.56 | Cresl1n | yeah |
20:26.59 | Cresl1n | the digium bri card |
20:27.07 | wwq222 | is there a way to do outgoing call queueing in asterisk w/o manually controlling the number of files in the outgoing folder? |
20:27.11 | robin_sz | its just a plain old HFC/cologne data card |
20:27.15 | Cresl1n | oh |
20:27.26 | Qwell | wwq222: you can `touch` the files into the future |
20:27.34 | robin_sz | echo cancel works ok say, 80% of the time |
20:27.34 | Qwell | wwq222: I don't know the syntax offhand, but `man touch` |
20:27.47 | *** join/#asterisk cthorner (n=cthorner@209-234-185-130.static.twtelecom.net) |
20:29.21 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
20:29.21 | robin_sz | I don't think I can afford a digium card, out of my league |
20:29.21 | wwq222 | qwell: thanks - i'm trying to implement some kind of queuing scheme because i can only make so many outgoing calls at once - so i want to try to get the next call to go once the current one finishes |
20:29.21 | *** join/#asterisk duki (n=duki@host-85-27-49-12.brutele.be) |
20:29.22 | Qwell | ahh |
20:29.22 | *** part/#asterisk cthorner (n=cthorner@209-234-185-130.static.twtelecom.net) |
20:29.22 | Qwell | You probably want to write something that connects via manager |
20:29.22 | wwq222 | qwell: I thought about just checking the number of files in the outgoing directory, andthen moving more in there when it drops below the limit |
20:29.45 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
20:29.48 | d00gster | robin_sz, I can't find the pattern in the configs. In my sipura, If I put <999011:011> is will append 999 before outbound number that starts with 011. what's the equivilance in asterisk? |
20:30.25 | *** join/#asterisk alejandro (n=asanchez@kde/developer/alejandro) |
20:30.37 | Qwell | d00gster: _011.,1,Dial(999${EXTEN}) |
20:30.42 | Qwell | something like that would do |
20:30.47 | [TK]D-Fender | d00gster, There is none, because the * dialplan does whatever you TELL it to. |
20:30.59 | [TK]D-Fender | qwell : yes.. something LIKE that ;) |
20:31.12 | Qwell | except I completely hosed the tech |
20:31.17 | [TK]D-Fender | shhh! |
20:31.57 | [TK]D-Fender | qwell : Tell not the see-rat! |
20:32.07 | [TK]D-Fender | seek* |
20:32.14 | *** join/#asterisk Fieldy (i=suQbustO@gentoo/contributor/Fieldy) |
20:32.16 | alejandro | hi, Im trying to configure a SPA 3102 with Asterisk. I configured in the SPA as SIP client and it's connected to the PSTN. How i can configure a basic config so softphones can call the PBX and later redirect to the PSTN ? |
20:32.29 | d00gster | Qwell, I lost you there, what I am trying to do is to prepend a 2121111111 <Pause 2 sec> before a 011. number .... can you help me with that? |
20:32.29 | alejandro | It's the first time that I configure Asterisk as a PBX, and it's a little hard. :-) |
20:32.43 | *** join/#asterisk Corydon76-home (i=green@pdpc/supporter/sustaining/Corydon76-home) |
20:32.43 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
20:32.51 | Qwell | haha, I just got my t-mobile bill |
20:32.59 | Qwell | and they charged me for sending a txt msg to myself |
20:33.04 | Qwell | that is awesome |
20:33.13 | robin_sz | d00gster, did you find the "international" bit of the diaplan? |
20:33.24 | d00gster | no robin_sz |
20:33.33 | robin_sz | well, you need to find that first |
20:33.38 | Qwell | I'm surprised they didn't double-charge me for it, since incoming isn't free |
20:33.55 | zoa | lol |
20:34.07 | zoa | its abnormal if you ask me |
20:34.08 | robin_sz | d00gster, do you know which file to look in to find the dialplan? |
20:34.16 | Qwell | ooo, but I did get charged double for sending one to my wife |
20:34.17 | zoa | they should have charged you double |
20:34.26 | d00gster | nope. |
20:34.40 | Qwell | and I got charge for sending messages to an invalid number |
20:35.20 | Qwell | ooo, I'm mad now :D |
20:35.37 | robin_sz | d00gster, its in extensions.conf ... now, you need to read the docs. really. what you will get on here is help if you have read the docs and tried to learn but we will not just spponfeed you .. we will tell you were to look for the sppon though :) |
20:35.47 | [TK]D-Fender | ~book |
20:35.50 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
20:35.52 | robin_sz | and that |
20:35.53 | [TK]D-Fender | ~wikis |
20:35.54 | jbot | somebody said wikis was http://www.voip-info.org |
20:35.59 | Mahmoud | Qwell, lol @ double charge |
20:40.32 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
20:41.47 | *** join/#asterisk MrTelephone (n=DeaLER25@bas13-toronto63-1177850949.dsl.bell.ca) |
20:42.14 | MrTelephone | can someone help me with the Cellular gateways? Is there something that acts like a mini cellular base station? |
20:42.31 | zoa | yes there are |
20:42.36 | zoa | but i dont have any links |
20:42.38 | zoa | they are affordable |
20:45.26 | MrTelephone | are you responding to me? |
20:45.29 | zoa | yes |
20:45.52 | MrTelephone | so they are base station then? when I read one description it sounded like it uses the preexisting cellular network |
20:46.02 | MrTelephone | i want to put up a cellular network in the bush |
20:46.06 | zoa | yes there are base stations |
20:46.10 | zoa | it exists |
20:46.20 | zoa | look for micro cells or so |
20:46.22 | MrTelephone | http://www.hyperms.com/index.asp?mainpage=prod_enlarge&prodcat=2603&prodtbl=260300&prodid=106 |
20:46.27 | blitzrage | how much is a base station? That'd be fun to have a base station in downtown Toronto :) |
20:46.47 | drfreeze | Can someone explain to me what a LineKey is (reg.1.lineKeys on Polycom cfg) |
20:46.51 | MrTelephone | it's a licensed spectrum but im putting it on native reserve land |
20:47.08 | blitzrage | that makes sense |
20:47.08 | MrTelephone | linekey is the lcd butotns on the display |
20:48.26 | MrTelephone | u can have up to 3 line keys on a polycom 501 |
20:48.26 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
20:48.26 | MrTelephone | is the link I pasted a base station? |
20:48.26 | zoa | i thought that was the name |
20:48.26 | MrTelephone | so if I plug that crap in I should be able to use my cell phone? |
20:48.26 | zoa | blitzrage |
20:48.26 | drfreeze | MrTelephone: how does that differ from callsperlinekey? |
20:48.27 | zoa | i dont remember |
20:48.27 | zoa | but its cheap |
20:48.27 | [TK]D-Fender | You DO have exactly 3 LineKeys on an IP 501. What you DO with them is up to you. |
20:48.27 | MrTelephone | you can have more than one conversation with each line key |
20:48.27 | drfreeze | [TK]D-Fender: what is a LineKeys? A 'line'? |
20:48.27 | MrTelephone | its like call-limit in sip.conf |
20:48.37 | [TK]D-Fender | drfreeze, No, not quite. |
20:48.55 | MrTelephone | usually u want one line key.. 1 call per line key or something |
20:49.14 | MrTelephone | or 3 line keys, 1 call per line, then when a second person calls it shows up on line key 2 |
20:49.36 | zoa | or try pico cell |
20:49.42 | MrTelephone | pico cell? |
20:49.53 | [TK]D-Fender | drfreeze, An IP 501 can support up to 3 completely distict regitrations. to do so you must allocate 1 linkey to each. From there though you can support from 1-8 calls on each line key. |
20:50.35 | drfreeze | [TK]D-Fender: by registration - do you mean their own phone number? |
20:51.12 | [TK]D-Fender | drfreeze, Most people only use a SINGLE registration with a phone however. With this in mind you can sake "linekeys" = 3, and callperlinekey=1, and that way when youa re on a call using the first linekey, a subsequent incoming call will ring on the SECOND. And the tird on the thrid |
20:51.32 | [TK]D-Fender | drfreeze, Yes. registration = seperate account entirely. |
20:52.16 | MrTelephone | i need a cdma base station |
20:52.16 | [TK]D-Fender | drfreeze, 1 same would be to use reg1 to have 2 linekeys @ 1 call max each, and reg2 having 1 key supporting 5 calls. |
20:52.32 | MACscr | how can i resolve/verify a sip address? |
20:52.47 | drfreeze | [TK]D-Fender: so, if I want a 'traditional feel' for a phone and want it to have 3 lines, I set linekeys=3 and callsperlinekey=1 or linekeys=1 and callsperlinekey=3? |
20:53.00 | drfreeze | MACscr: sip show peers |
20:53.21 | [TK]D-Fender | drfreeze, The former |
20:53.28 | drfreeze | [TK]D-Fender: thx |
20:53.34 | [TK]D-Fender | drfreeze, And that will be a "traditional" feel |
20:53.46 | MACscr | drfreeze : that would only work if it was a peer, what if wanted to verify and address |
20:53.52 | MACscr | whoops, any |
20:54.15 | [TK]D-Fender | drfreeze, since I use my IP 501 with multiple clients, I give 1 line key to each reg w/ 5 calls each max (you use the cursor keys to scroll through calls. Its vrey nice actually). |
20:54.25 | MACscr | basically i can receive phone calls just fine with my number, but not by my sip address |
20:54.39 | [TK]D-Fender | drfreeze, Polycom's call handling capabilities are superior to every phone I have ever used. |
20:54.53 | MACscr | and i want to figure out if its my address (received from provider) or my * box |
20:55.02 | drfreeze | [TK]D-Fender: what's the advantage to having multiple lines on a linekey? |
20:55.33 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
20:56.02 | drfreeze | [TK]D-Fender: is that like call waiting, but with 5 lines? |
20:56.40 | drfreeze | MACscr: I don't think I'm the one to ask. :( |
20:57.11 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
20:57.21 | [TK]D-Fender | drfreeze, Well if you want multiple regs, this way you can support multiple calls per and conference any which way you want. Also you can use a single line-key even for just 1 reg supporting multiple call so as to leave 2 available for use as speed-dials. Very useful when associated for presence. |
20:57.30 | zoa | 3600 downloads of idefisk 2.0 so far |
20:57.32 | zoa | not bad |
20:57.41 | zoa | for 60 hours or so |
20:58.29 | MACscr | drfreeze : i only asked after you responded to me =P |
20:58.39 | drfreeze | [TK]D-Fender: oh, so the conf button on the phones is to connect lines on a linekey? |
20:59.05 | drfreeze | MACscr: I know. sorry about that |
20:59.19 | [TK]D-Fender | <PROTECTED> |
20:59.49 | [TK]D-Fender | drfreeze, If you simply have 2 distinct calls in progrees and you want to merge into a conference, that is the purpose of the "join" soft-key |
21:00.06 | LennonNZ | zoa: your dev of it? |
21:00.18 | Qwell | LennonNZ: he pays the bills ;) |
21:00.33 | LennonNZ | how much is the OEM vers of it? |
21:00.34 | *** part/#asterisk MrTelephone (n=DeaLER25@bas13-toronto63-1177850949.dsl.bell.ca) |
21:02.05 | zoa | qwell see my privmsg |
21:02.47 | Vec | Is it possible to set an ip address range of allowed peers in sip.conf (for each peer) I remember seeing it somewhere but can't find it anywhere i.e a ACL? |
21:02.57 | Dovid | zoa: Qwell dont like PM's |
21:03.10 | zoa | sure he does |
21:03.13 | MACscr | so no one can tell me how to verify a sip address? |
21:03.17 | drfreeze | [TK]D-Fender: In a polycom.cfg file, is the reg.1 a linekey? |
21:03.25 | Qwell | zoa: I saw it :p |
21:03.58 | zoa | so reply you little bastard :p |
21:04.02 | drfreeze | MACscr: ok, let's give is a try |
21:04.09 | drfreeze | what exactly are you trying to do? |
21:04.13 | Assid | zoa: new one out ? |
21:04.37 | zoa | yes |
21:04.39 | zoa | doing sip now |
21:04.41 | MACscr | drbreeze: im trying to verify that my sip address is correct as i cant seem to register it with enum |
21:04.59 | boch | is it possible to make Record() stop recording with any key instead # ? |
21:05.02 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
21:05.06 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
21:05.10 | Assid | zoa: yeah i know.. you sent me the beta remember.. |
21:05.13 | Dovid | yes boch: |
21:05.14 | Assid | did you fix those bugs? |
21:05.16 | Dovid | simple hang up |
21:05.23 | drfreeze | MACscr: lets get some context. your phone registering with *? |
21:05.26 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
21:05.28 | markit | hi, the "voice" I get from a ISDN BRI card, what "codec" is it using? |
21:05.39 | Qwell | markit: g711 |
21:05.40 | drfreeze | does 'sip show peers' show unkown? |
21:05.50 | markit | Qwell oh, thanks a lot |
21:05.51 | boch | Dovid, what do you mean with hangup ? |
21:05.55 | russellb | Qwell: pong |
21:05.58 | zoa | assid: so it got a little more stable in the mean time |
21:06.15 | zoa | russel did you see my message about the 1 port on asterisk-dev ? |
21:06.21 | Dovid | boch: my bad. didnt see what u wrote. idk of switching it rom # to another key |
21:06.22 | zoa | 1 port 1 thread ? |
21:06.30 | zoa | maybe we need 2 ports :) |
21:06.32 | russellb | zoa: yeah, I sure did |
21:06.33 | russellb | heh |
21:07.12 | zoa | we could do a double register for it |
21:07.16 | Qwell | russellb: nevermind :p |
21:07.18 | zoa | wouldnt break the nat stuff |
21:07.18 | russellb | zoa: I'm still pondering it. You have definitely made me wonder if putting in IAX2 is actually the best idea |
21:07.25 | zoa | hehe :) |
21:07.26 | zoa | great |
21:07.52 | *** join/#asterisk Fieldy (i=TlY8ZsWD@gentoo/contributor/Fieldy) |
21:08.13 | Qwell | man, gentoo gives out cloaks like candy |
21:08.14 | zoa | i think the kernel proposal might be ok |
21:08.15 | russellb | It make more sense to go with a new protocol using TCP ... |
21:08.20 | drfreeze | MACscr: solve your problem? |
21:08.21 | zoa | i also think so |
21:08.40 | Qwell | zoa: eh, there are other ways to do that.. no need for a kernel modules (that would be linux specific...) |
21:09.08 | zoa | maybe its something more for dundi ? |
21:09.12 | zoa | cant we put that into dundi ? |
21:09.51 | MACscr | drfreeze : my context works fine as far as incoming and outgoing when it comes to regular PTSN numbers. |
21:09.54 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
21:09.57 | zoa | and move dundi into more like an asterisk signalling / clustering protocol |
21:10.23 | Qwell | well, it WOULD be for talking to asterisk |
21:10.31 | Qwell | and, if you'll all recall what IAX stands for... :p |
21:10.33 | russellb | zoa: That's an interesting thought, I have been wondering that myself |
21:10.42 | Qwell | I mean, perhaps a tcp thingie for iax |
21:10.48 | Qwell | (but NOT for audio) |
21:10.50 | russellb | but then I was thinking use DUNDi for the discovery of event peers ... and IAX2 as the event peering protocol |
21:11.15 | zoa | for messaging stuff, iax would be fine |
21:11.16 | Qwell | there really is no reason iax2 couldn't listen on udp and tcp |
21:11.26 | zoa | and why not we could put CDR's in there too |
21:11.27 | Qwell | ...for signalling and stuff like that |
21:11.31 | zoa | but im not sure it would be ok for all cases |
21:12.07 | Qwell | but the thing is, even with tcp, you're still gonna need to be 100% certain that the record got to the other machine |
21:12.16 | zoa | yes |
21:12.22 | [TK]D-Fender | drfreeze, No, reg1 is not a linekey. You have to tell it HOW MANY it will receive, and how many calls will be supported on each. If you get reg1 2 keys with 2 calls each, it will then support up to 4 calls, with the third & forth (if received consecutively) will fall on the 2nd line-key. |
21:12.24 | zoa | my collegue will post some questions about failover with it |
21:12.25 | Qwell | server could be flat out down/off |
21:12.30 | zoa | what if the receiving server is down |
21:12.36 | Qwell | So you'd need to keep queueing them up |
21:12.39 | zoa | even that should be handled |
21:12.44 | zoa | with the possibility of multiple ones |
21:12.47 | zoa | with resynching |
21:12.51 | russellb | Dundi could also be the transport of events, i guess. You'd only send events to a few servers and they would make it through the network ... |
21:12.54 | Qwell | so, IMO, I don't think we really need to even go the tcp route |
21:14.00 | Qwell | this is probably a somewhat inappropriate place for this discussion too.. |
21:14.09 | drfreeze | [TK]D-Fender: thx for that explanation. What is the purpose of reg.2 - reg.6? |
21:14.11 | russellb | we could switch to #asterisk-dev :) |
21:14.24 | [TK]D-Fender | drfreeze, And there is no polycom.cfg, and you should never put registration stuff in sip.cfg either, only in the phoneXXX.cfg. |
21:14.38 | [TK]D-Fender | drfreeze, other completely differnt identities. |
21:14.47 | drfreeze | [TK]D-Fender: yeah, that is what I meant. phonexxx.cfg |
21:15.25 | drfreeze | [TK]D-Fender: ok, so I think I'm getting it. So they can connect to different providers. |
21:15.25 | [TK]D-Fender | drfreeze, I am registered to MY server at home with 1 reg (using 1 linekey @ 5 calls), and I register to REMOTE PBX's with my other 2. |
21:15.29 | *** join/#asterisk Fieldy (i=1PCoslZL@gentoo/contributor/Fieldy) |
21:15.35 | [TK]D-Fender | drfreeze, NOW you're getting it... |
21:15.50 | [TK]D-Fender | drfreeze, This is not typical in your average home/office |
21:15.57 | drfreeze | [TK]D-Fender: is there a way for the user to tell where the calls are coming from? other than ring type? |
21:16.06 | [TK]D-Fender | drfreeze, But jsut another reason for Polycom's superior call handling :) |
21:16.23 | [TK]D-Fender | drfreeze, can you clarify that a bit... |
21:16.48 | Assid | you really should take up polycom support man.. |
21:17.39 | drfreeze | [TK]D-Fender: when a comes to your phone thru your server and when one comes thru your remote pbx. Can you tell from the LCD display on your 501? |
21:17.56 | drfreeze | *when a call |
21:18.16 | [TK]D-Fender | drfreeze, You mean as opposed to from those 2 other regs? |
21:18.27 | drfreeze | yes |
21:18.28 | Assid | drfreeze: the caller id shows up as what is received |
21:18.46 | [TK]D-Fender | drfreeze, as I gave each reg its own line key, the one thats ringing is animated. |
21:18.53 | drfreeze | Assid: that won't tell you the src |
21:19.18 | [TK]D-Fender | drfreeze, So on 1 client, linkey 2 will be flashing for that incoming call. |
21:19.19 | Assid | welll shows you the line key thats calling in |
21:20.28 | drfreeze | maybe it doesn't matter. Was just wondering if you could tell when a call came thru a provider A vs provider B |
21:21.16 | Assid | drfreeze if its over a different line key (whichever is registered with whichever pbx), that linekey will show ringing |
21:21.43 | [TK]D-Fender | <PROTECTED> |
21:21.52 | [TK]D-Fender | drfreeze, Instantly |
21:22.48 | [TK]D-Fender | drfreeze, You have no idea what other mucking I do to accomodate the fact that MY * is reg'd to a lot of other PBX for which I use a prefix code to dial out of. :) |
21:23.06 | drfreeze | :) |
21:23.38 | drfreeze | I have an office that may need to start doing just such trickery. :) |
21:23.49 | [TK]D-Fender | drfreeze, I then prefix callerid coming in from each to make sure it goes back out the right system :) |
21:24.09 | Assid | [TK]D-Fender: sipbroker ? |
21:24.11 | drfreeze | [TK]D-Fender: woo, smart |
21:24.26 | [TK]D-Fender | Assid, huh? |
21:24.39 | *** join/#asterisk `p4r14h (n=j0sh@69.92.145.178) |
21:24.52 | Assid | have you tried sipbroker? |
21:25.19 | [TK]D-Fender | Assid, Nope. |
21:25.29 | [TK]D-Fender | Assid, What do i need ITSP's or... I have CLIENTS ;) |
21:25.44 | [TK]D-Fender | s/or/for/ |
21:29.53 | l1nux | good night all :) |
21:29.56 | *** part/#asterisk l1nux (n=moi@jof81-1-82-245-67-40.fbx.proxad.net) |
21:31.44 | *** join/#asterisk ruied (n=ruied@bl7-217-252.dsl.telepac.pt) |
21:33.29 | [TK]D-Fender | ~book |
21:33.38 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
21:40.05 | *** join/#asterisk FieldySnuts (i=0TnvqF3f@gentoo/contributor/Fieldy) |
21:40.13 | MACscr | anyone know of a service i can use to make a test sip call to me? |
21:41.28 | Dovid | MACscr: u wana test incoming sip ? |
21:41.38 | zoa | its called asterisk and you can download it for free |
21:41.39 | Dovid | u can use a sogphone w |
21:41.40 | MACscr | David: yes |
21:41.41 | Dovid | hehe |
21:41.42 | zoa | :) |
21:42.20 | MACscr | my DIDs work fine, but i need to get my sip address to work so i can register with enum |
21:42.49 | [TK]D-Fender | MACscr, Gimme a URI and I'll call you |
21:44.05 | MACscr | PM sent |
21:44.53 | MACscr | my sip provider provides URI's as welll that i should be able to use too, right? |
21:47.43 | zoa | im off to bed |
21:47.44 | zoa | bbye |
21:48.19 | [TK]D-Fender | zoa, later |
21:49.47 | Assid | alrite im off to bed as well |
21:49.53 | Assid | gnight |
21:59.07 | *** join/#asterisk AMUG (n=junky@modemcable140.185-70-69.mc.videotron.ca) |
21:59.36 | JunK-Y | %ast core show application record |
22:02.51 | russellb | %ast originate Local/12562486000@default extension Local/12564286098 |
22:03.11 | russellb | :) |
22:03.26 | JunK-Y | %ast core show channels |
22:03.41 | [TK]D-Fender | <PROTECTED> |
22:03.50 | JunK-Y | russel: 0 active channels |
22:03.51 | [TK]D-Fender | :( pwnd |
22:03.52 | Qwell | %ast !shutdown -h now |
22:03.58 | JunK-Y | %ast stop now |
22:04.01 | *** mode/#asterisk [+b *!*n=junky@*.185-70-69.mc.videotron.ca] by Qwell |
22:04.01 | *** kick/#asterisk [AMUG!n=north@pdpc/sponsor/digium/Qwell] by Qwell (bot) |
22:04.08 | russellb | %ast !cat /dev/urandom /tmp/fillmydisk |
22:04.13 | [TK]D-Fender | z0mg! |
22:04.18 | Qwell | oh, heh, JunK-Y |
22:04.25 | *** mode/#asterisk [-b *!*n=junky@*.185-70-69.mc.videotron.ca] by Qwell |
22:04.33 | Qwell | sorry :D |
22:04.36 | JunK-Y | hehe |
22:04.59 | JunK-Y | %ast dialplan add foo |
22:05.03 | Qwell | %ast \!shutdown -h now |
22:05.07 | *** join/#asterisk AMUG (n=junky@modemcable140.185-70-69.mc.videotron.ca) |
22:05.08 | Qwell | %ast \!shutdown -h now |
22:05.20 | Qwell | boo |
22:05.23 | Qwell | %ast !shutdown -h now |
22:05.36 | JunK-Y | %ast \!ls |
22:05.46 | Qwell | %ast !reboot |
22:06.04 | JunK-Y | Qwell: continue, he just told me that he likes it. |
22:06.07 | russellb | % !cat /etc/shadow |
22:06.22 | russellb | %ast !cat /etc/shadow |
22:06.27 | JunK-Y | %ast !cat /etc/passwd |
22:06.46 | Qwell | %ast dialplan add extension die,1,System,reboot |
22:06.56 | russellb | lol |
22:06.58 | JunK-Y | dialplan add extension die,1,System,reboot |
22:07.02 | JunK-Y | %ast dialplan add extension die,1,System,reboot |
22:07.19 | russellb | %ast sip show users |
22:07.30 | russellb | I got a password! |
22:07.31 | russellb | score! |
22:07.34 | Qwell | %ast dialplan add extension die,1,System,reboot into blah replace |
22:07.43 | JunK-Y | now exploit it :) |
22:07.52 | russellb | %ast manager show users |
22:08.00 | JunK-Y | russellb: thats why i submit a damn patch! |
22:08.12 | russellb | :-p |
22:08.20 | russellb | %ast iax2 show users |
22:08.21 | JunK-Y | http://bugs.digium.com/view.php?id=9273 |
22:08.22 | Qwell | %ast unload chan_sip |
22:08.24 | JunK-Y | could ya commit it? |
22:08.25 | Qwell | %ast unload chan_sip.so |
22:08.40 | russellb | %ast restart now |
22:08.42 | JunK-Y | %ast restart now |
22:08.43 | JunK-Y | hjeheje |
22:08.49 | JunK-Y | i allow a restart now so far |
22:08.52 | JunK-Y | %ast core show uptime |
22:09.00 | JunK-Y | since, this is only a dev machine. |
22:09.02 | russellb | %ast core show channels verbose |
22:09.07 | JunK-Y | %ast core show uptime |
22:09.21 | russellb | %ast convert tt-weasels.gsm tt-weasels.wav |
22:09.23 | russellb | %ast convert tt-weasels.gsm tt-weasels1.wav |
22:09.25 | russellb | %ast convert tt-weasels.gsm tt-weasels2.wav |
22:09.27 | russellb | %ast convert tt-weasels.gsm tt-weasels3.wav |
22:09.29 | russellb | %ast convert tt-weasels.gsm tt-weasels4.wav |
22:09.36 | Qwell | no, no, no! |
22:09.36 | JunK-Y | mouahah ah |
22:09.48 | Qwell | %ast convert tt-weasels.gsm tt-weasels.ilbc |
22:09.51 | Qwell | %ast convert tt-weasels.gsm tt-weasels.lpc10 |
22:11.38 | Qwell | harder codecs :P |
22:11.38 | russellb | %ast convert /var/lib/asterisk/moh/fpm-calm-river.wav /var/lib/asterisk/moh/fpm-calm-river.ilbc |
22:11.38 | russellb | %ast convert /var/lib/asterisk/moh/fpm-calm-river.wav /var/lib/asterisk/moh/fpm-calm-river1.ilbc |
22:11.39 | russellb | %ast convert /var/lib/asterisk/moh/fpm-calm-river.wav /var/lib/asterisk/moh/fpm-calm-river2.ilbc |
22:11.39 | russellb | %ast convert /var/lib/asterisk/moh/fpm-calm-river.wav /var/lib/asterisk/moh/fpm-calm-river3.ilbc |
22:11.39 | JunK-Y | No such command 'convert tt-weasels.gsm' (type 'help' for help) |
22:11.39 | russellb | :( |
22:11.39 | Qwell | trunk, file convert |
22:11.39 | russellb | bah |
22:11.39 | JunK-Y | %ast core show uptime |
22:11.39 | russellb | %ast core show version |
22:11.44 | russellb | it should have that command! :) |
22:12.12 | JunK-Y | %ast core show version |
22:12.16 | JunK-Y | %ast core show channels |
22:12.16 | Qwell | %ast dialplan show |
22:12.24 | JunK-Y | Qwell: i autorize it too |
22:12.28 | Qwell | lol, gui |
22:12.30 | Qwell | nice |
22:13.08 | Qwell | %ast include context default into DID_trunk_1 |
22:13.08 | Qwell | :D |
22:13.19 | JunK-Y | %ast include context |
22:13.27 | JunK-Y | could ya commit: http://bugs.digium.com/view.php?id=9273 ? |
22:14.00 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
22:15.55 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-14.ph.ph.cox.net) |
22:16.15 | *** join/#asterisk garreel (n=garreel@host48-2-dynamic.16-87-r.retail.telecomitalia.it) |
22:16.45 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
22:17.47 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
22:18.34 | Qwell | %ast moh show files |
22:18.45 | Qwell | boo |
22:19.21 | JunK-Y | theres no crucial information on that machine. |
22:21.26 | russellb | why not |
22:21.43 | Qwell | %ast indication en add 247/500,494/328,0/78,370/94,370+247/31,247/94,0/375,185/500,220/500,220+247/16,247/484,0/16,294/484,0/16 |
22:21.49 | Qwell | erm |
22:21.51 | russellb | %ast console dial demo |
22:22.01 | russellb | ooh, that one is allowed |
22:22.11 | Qwell | %ast indication remove en ring |
22:22.14 | russellb | %ast dialplan show |
22:22.18 | Qwell | %ast indication add en ring 247/500,494/328,0/78,370/94,370+247/31,247/94,0/375,185/500,220/500,220+247/16,247/484,0/16,294/484,0/16 |
22:22.30 | Qwell | %ast indication add en ring "247/500,494/328,0/78,370/94,370+247/31,247/94,0/375,185/500,220/500,220+247/16,247/484,0/16,294/484,0/16" |
22:22.36 | Qwell | hot |
22:22.58 | russellb | %ast console dial vmenu_record |
22:24.09 | cr4z3d | yay for ssh <3 |
22:24.14 | russellb | AMUG: stfu |
22:24.37 | Qwell | %ast indication add us ring 247/500,494/328,0/78,370/94,370+247/31,247/94,0/375,185/500,220/500,220+247/16,247/484,0/16,294/484,0/16 |
22:24.38 | Qwell | better :p |
22:24.46 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
22:25.02 | Qwell | %ast indication show us |
22:25.11 | Qwell | silly bot |
22:25.13 | russellb | %ast console dial 1000 |
22:25.44 | russellb | don't ask for the dialplan, it'll go for 15 minutes |
22:25.47 | JunK-Y | hes lagged, wait |
22:25.49 | russellb | one line at a time, slowly |
22:26.06 | Qwell | russellb: yeah, I already did, heh |
22:27.57 | garreel | is there a way to show caller id even if it's hidden by default on a E1 PRI line? |
22:28.12 | Qwell | garreel: hidden by who? |
22:28.19 | garreel | hidden by the telco |
22:28.23 | Qwell | then no.. |
22:28.36 | garreel | :-( |
22:29.12 | Qwell | why would they hide cid on a PRI? |
22:29.29 | garreel | becouse we are a call center |
22:29.53 | garreel | who work for the telco |
22:30.04 | garreel | we make outbound |
22:30.15 | garreel | so it's hidden |
22:30.40 | garreel | I was thinking to try something such the *82 code |
22:31.26 | garreel | but if I put a *XX code before the number I get an error |
22:32.27 | Qwell | JunK-Y: Playtones(ring) \n Wait(30) |
22:32.36 | Qwell | Do that, you'll like it :p |
22:32.56 | Qwell | Answer() first |
22:38.56 | *** join/#asterisk `p4r14h (n=j0sh@69.92.145.178) |
22:39.33 | mmartinn | Will [general] / disallow = all in skinny.conf disable all? |
22:39.45 | Qwell | it should, in 1.4 |
22:39.51 | mmartinn | in 1.2? |
22:39.55 | Qwell | no |
22:40.02 | Qwell | don't use chan_skinny in 1.2 |
22:40.14 | mmartinn | We don't use skinny at all but there's a log message about it |
22:40.25 | Qwell | just noload the module |
22:41.03 | mmartinn | okay :) |
22:41.59 | *** join/#asterisk Jubei (n=Stormtro@147.27.46.26) |
22:42.35 | Jubei | could somebody either explain to me or point me to a document on the web about the diferrences between TE and NT mode in ISDN BRI interfaces |
22:43.54 | mmartinn | Qwell: There's no weird side effects of noload=chan_iax2, right? |
22:43.55 | Qwell | mmartinn: not if you don't use it, and it'd be chan_iax2.so |
22:43.55 | mmartinn | Qwell: Right! Thanks :) |
22:43.58 | garreel | Qwell[]: but if it's not possible to override the default hidden caller id state... what is *82 code meant of? |
22:46.06 | gambolputty | jubei - look at cisco ccna documents |
22:46.11 | gambolputty | they talk about isdn |
22:47.19 | gambolputty | and then after that, watch Ninja Scroll |
22:50.22 | *** join/#asterisk crudi (n=crudi@ppp-70-247-203-146.dsl.snantx.swbell.net) |
22:52.29 | Jubei | guys mISDN speaks of a command line called "misdnportinfo". I don't have that in my system, why?:) |
22:52.44 | Jubei | is it part of the mISDN package? |
22:53.16 | crudi | I'm having issues with AsteriskNow. I'm willing to paypal $$ if someone can help me get it working. I have a Triple span FXO digium card, GXP-2000 phones, and 2 POTS lines. |
22:54.21 | Qwell | crudi: what issues? |
22:55.02 | crudi | well lets start with if dial an internal extension, we can't hear each other. |
22:55.23 | Qwell | are the grandstreams on the same LAN? |
22:55.28 | crudi | yes |
22:55.28 | Qwell | same lan as asterisk, I should say |
22:55.32 | crudi | yup |
22:56.02 | *** join/#asterisk jdiskywlkr (n=kvirc@adsl-70-234-164-77.dsl.tul2ok.sbcglobal.net) |
22:56.07 | Qwell | Do you get audio if you call something like, say, voicemail? |
22:56.59 | crudi | let me check |
22:57.13 | jdiskywlkr | If Asterisk is used to proxy a mgcp to sip phone call, will Asterisk stay in between the two endpoints, or will the conversation be handed off to the endpoints using rtp? |
22:58.36 | *** join/#asterisk saftsack (n=saftsack@pD9E07CF9.dip.t-dialin.net) |
22:58.56 | saftsack | is the via c3 corefusion board suitable for asterisk? |
22:59.10 | saftsack | pbx system w/o transcoding with < 20 users |
22:59.17 | crudi | yes i get audio when i call VM |
22:59.30 | Qwell | crudi: So, it's only a problem between the grandstreams? |
22:59.36 | crudi | yup |
22:59.44 | Qwell | and did you verify that they both get audio from voicemail? |
23:01.37 | crudi | yes they both do |
23:02.18 | crudi | now i get that the other # is unavailable |
23:02.36 | Qwell | it probably isn't registered |
23:02.48 | crudi | I have registration turned off in the gui on the phone |
23:02.58 | crudi | thats what i read to do |
23:03.14 | Qwell | yeah, that's so asterisk doesn't register to the phone.. the phone isn't registering to asterisk, and it needs to |
23:03.48 | JunK-Y | any way to register and set an outbound-proxy btw? |
23:04.51 | Jubei | anybody know anything about mISDN ? i've done everything as the documentation describes but then "misdnportinfo" says : "Found no card. Please be sure to load card drivers." isn't that what the mISDN start script is supposed to do? |
23:04.52 | crudi | yeah, i get |
23:04.56 | crudi | Name/username Host Dyn Nat ACL Port Status |
23:04.56 | crudi | 02 (Unspecified) D 0 Unmonitored |
23:04.56 | crudi | 01 (Unspecified) D 0 Unmonitored |
23:04.56 | crudi | 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] |
23:05.37 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
23:17.42 | crudi | any ideas on that qwell? |
23:17.50 | tzafrir_laptop | I'm trying to sign up for an account in iaxtel . Funny enough they have a large choices of countries to choose from |
23:17.55 | tzafrir_laptop | (USA alone) |
23:18.29 | Qwell | crudi: You'll need to fix up your phones to register |
23:18.36 | tzafrir_laptop | Jubei, misdn-init might help |
23:18.42 | tzafrir_laptop | it has some useful options |
23:18.45 | crudi | ive turned them on to register. still not registering. |
23:19.18 | Qwell | crudi: Did you setup the phones in the GUI? |
23:19.25 | crudi | yes |
23:19.39 | tzafrir_laptop | what phones? |
23:19.42 | Qwell | using the Users tab? |
23:19.45 | Qwell | tzafrir_laptop: gxp's |
23:19.46 | crudi | gxp-2000 |
23:19.52 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
23:20.02 | crudi | Users tab in the asterisknow gui, and the account 1 tab in the phone gui |
23:20.13 | Qwell | what account? |
23:20.26 | crudi | the extensions |
23:20.27 | crudi | 01 |
23:20.28 | crudi | 02 |
23:20.36 | Qwell | those are added in the Users tab... |
23:21.02 | crudi | yes |
23:21.05 | crudi | now i have |
23:21.06 | crudi | Name/username Host Dyn Nat ACL Port Status |
23:21.06 | crudi | 02/02 192.168.1.202 D 5060 Unmonitored |
23:21.06 | crudi | 01/01 192.168.1.201 D 5060 Unmonitored |
23:21.06 | crudi | 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline] |
23:21.14 | Qwell | much better |
23:21.28 | crudi | but still no audio |
23:22.14 | crudi | it rings, we pick up, cant hear |
23:22.20 | tzafrir_laptop | hmm... they also have a typo in the screen after the registration: "minuites" instead of "minutes" |
23:22.32 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
23:22.35 | Qwell | Does the asterisk console give any useful output, if you do a `core set verbose 50`? |
23:22.44 | Qwell | ..and make the call again afterwards, of course |
23:23.33 | garreel | crudi: try canreinvite=no |
23:23.39 | crudi | Verbosity was 3 and is now 50 |
23:23.39 | crudi | <PROTECTED> |
23:23.39 | crudi | <PROTECTED> |
23:23.39 | crudi | <PROTECTED> |
23:23.39 | crudi | <PROTECTED> |
23:23.39 | crudi | <PROTECTED> |
23:23.41 | crudi | <PROTECTED> |
23:23.43 | crudi | <PROTECTED> |
23:23.46 | crudi | <PROTECTED> |
23:23.46 | Qwell | garreel: they're on the same NAT |
23:23.56 | Qwell | crudi: please use a pastebin next time |
23:23.58 | Qwell | ~pb |
23:24.00 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
23:24.00 | crudi | sorry |
23:24.01 | crudi | k |
23:24.39 | crudi | i know i know, i will |
23:25.09 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-66-89.dsl.peoril.sbcglobal.net) |
23:25.58 | MACscr | whats the minimum about of ports i need open for rtp? |
23:26.09 | Qwell | MACscr: 0, if you aren't using it |
23:26.49 | *** join/#asterisk thevoke (n=michiel@missioncontrol.eng.gbxs.net) |
23:27.18 | MACscr | lol, true |
23:27.33 | Jubei | anybody running mISDN? |
23:27.56 | crudi | http://www.pastebin.ca/418975 |
23:28.39 | Qwell | are your phones set to use a stun or anything silly? |
23:28.51 | Qwell | and no firewall of any kind between them? |
23:29.01 | crudi | nope to both |
23:29.11 | crudi | straight out of the box |
23:29.15 | crudi | both on the same switch |
23:29.23 | Qwell | switch or router? |
23:29.27 | crudi | switch |
23:29.34 | Qwell | ok |
23:29.39 | crudi | router connected to switch of course |
23:30.05 | Qwell | go into users.conf, and add a canreinvite=no for each of those phones.. you can't do it from the GUI afaik |
23:31.47 | crudi | didnt work |
23:31.53 | Qwell | Did you reload? |
23:31.56 | crudi | yes |
23:32.10 | Qwell | Does it still show the "Native bridging" line? |
23:32.24 | crudi | no |
23:32.26 | crudi | packet2packet |
23:32.59 | Qwell | any chance I can get ssh access to the box? |
23:33.38 | crudi | its on a private lan |
23:34.43 | Qwell | You didn't edit sip.conf at all? |
23:34.48 | crudi | no |
23:35.00 | Qwell | there's gotta be a setting in the phones causing that |
23:35.15 | Qwell | You're gonna have to look through the menus and find something about stun, or nat, or something, and disable it |
23:35.26 | crudi | stun is blank |
23:35.33 | garreel | crudi: try iptraf to se if data is flowing trough the phones |
23:35.34 | crudi | so is nat ip |
23:35.46 | crudi | where at? |
23:35.48 | Qwell | crudi: anything on rtp debug? |
23:35.59 | crudi | rtp debug? |
23:36.11 | crudi | just turned that on |
23:36.28 | Qwell | make a call, and see if you see the rtp packets |
23:36.39 | crudi | i see a bunch |
23:36.45 | Qwell | pastebin about 20-30 lines of that |
23:37.01 | Qwell | start from the beginning if possible |
23:37.13 | crudi | http://www.pastebin.ca/418981 |
23:37.34 | Qwell | no "Got RTP packet from "? |
23:37.38 | mmartinn | Hey folks; I notice that when I use the UserEvent manager action, I see the event, but I get an "Action: UserEvent" header in the response -- I don't see it explicitly in the source; does this mean the UserEvent treated Action: as a custom header? |
23:37.53 | crudi | nope |
23:38.06 | Qwell | the len is weird |
23:38.10 | Qwell | 160 vs 33 |
23:38.19 | Qwell | 160 would be ulaw I think.. not sure what 33 is |
23:38.27 | Qwell | that's tiny.. gsm/g729 maybe |
23:38.46 | crudi | ya we keep seeing GSM and PCMU |
23:38.50 | crudi | on our phones |
23:38.54 | Qwell | for both phones, try adding disallow=all and allow=ulaw |
23:39.25 | Qwell | not sure why they would be negotiating different codecs... |
23:39.46 | Qwell | russellb: re-ping |
23:40.25 | crudi | that worked |
23:40.42 | Qwell | cool |
23:40.43 | crudi | but i still dont see got rtp |
23:40.58 | Qwell | yeah, it may not show that in P2P bridging mode |
23:41.23 | Qwell | try removing those canreinvite lines now - you shouldn't need it |
23:42.24 | crudi | ya still works once i remove |
23:42.32 | Qwell | cool :) |
23:42.36 | garreel | :-) |
23:42.37 | crudi | How do I get my two pots lines to work |
23:42.45 | crudi | ive got the 4000p |
23:42.46 | Qwell | crudi: tdm400p? |
23:42.50 | crudi | 400 |
23:42.51 | crudi | ya |
23:42.53 | crudi | 3 fxo |
23:42.53 | Qwell | then on the setup wizard, it should've detected them |
23:42.56 | crudi | it did |
23:43.25 | crudi | i get a 404 when i call out |
23:43.35 | Qwell | the Incoming Calls and Calling Rules tabs |
23:44.03 | MACscr | anyone have experience with e164.org? I cant get ti to register my hostname. It says thats its unable to verify the route. But I am able to receive sip calls on that box, so im not sure what its complaining about |
23:44.18 | crudi | http://www.pastebin.ca/419003 |
23:44.42 | Qwell | and are the phones using those dialing rules? |
23:44.50 | crudi | yes |
23:44.51 | Qwell | "Dial Plan" |
23:45.14 | crudi | DialPlan1 |
23:45.14 | crudi | yeah |
23:45.29 | *** join/#asterisk brussel (n=brussel@cpe-24-165-7-252.san.res.rr.com) |
23:45.32 | Qwell | try unchecking the "Or More" checkbox for all of those |
23:45.37 | Qwell | it was buggy previously |
23:46.41 | Qwell | and you of course need to strip off 1 digit from those |
23:46.51 | Qwell | well, except the 911 rule |
23:48.51 | crudi | that seemed to work. removing the or more. but only on the first line. on the 2nd one i get a "603" |
23:49.04 | Qwell | the 911? |
23:49.20 | crudi | no - Line2-Local |
23:49.25 | crudi | its my 2nd pots |
23:49.31 | crudi | I dial 8 then the number |
23:50.12 | Qwell | is the phones dialplan set to use 8? |
23:50.20 | Qwell | phones have their own dialplans too |
23:50.24 | crudi | its the same dialplan |
23:50.34 | crudi | it just says if you get an 8, use this port |
23:50.44 | Qwell | in the phone itself |
23:51.36 | crudi | i have a dial plan prefix |
23:51.37 | crudi | blank |
23:52.55 | crudi | and it wont let me do a 9*73 to disable call forwarding |
23:53.06 | Qwell | because X doesn't match * |
23:53.23 | Qwell | You'll need another pattern for that |
23:54.02 | *** join/#asterisk chrisknight (n=explodin@cpe-71-79-81-174.columbus.res.rr.com) |
23:54.43 | crudi | i think i got that one |
23:55.19 | chrisknight | Iv'e been pulling my hair out for about a month now. Searched the net all over... Can anyone reset a cisco 7960 phone? I have tried everything. |
23:55.53 | crudi | qwell: whenever i call, i get the ivr. but i didnt set one up. incoming rule is set to go straight to an extension |
23:59.51 | garreel | trying MP3Player application... but it seems to play slooooooooooooooooooooooooowly the file... any ideas? |