| 00:02.27 | *** join/#asterisk Ebola (n=Ebola@host86-136-134-202.range86-136.btcentralplus.com) |
| 00:02.37 | *** join/#asterisk VOiCi (n=o@132-199.sh.cgocable.ca) |
| 00:02.52 | *** join/#asterisk UnixDog (n=UnixDog@adsl-69-234-211-33.dsl.irvnca.pacbell.net) |
| 00:03.03 | UnixDog | anyone here using nufone ? |
| 00:05.04 | *** join/#asterisk Greenbox (n=Brett@user-24-214-124-177.knology.net) |
| 00:05.06 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 00:05.20 | polerin | meh. Is it a bad idea to use the record() application to record IVR prompts on a temp basis? |
| 00:05.42 | JT | no... |
| 00:05.44 | polerin | or is it easier to just record them on another machine, convert, and move them over |
| 00:06.04 | JT | some would argue that a pro microphone/etc is better, but meh |
| 00:06.20 | JT | unless you're going to postproduce all the sound files, Record() is fine |
| 00:06.43 | polerin | well it'll be with a mic rather than a standard phone anyway |
| 00:06.52 | polerin | softphone's yehknow :/ |
| 00:06.57 | JT | oh |
| 00:07.10 | JT | because audio from a good hardphone sounds quite good :) |
| 00:07.30 | polerin | the key phrase there is "good hardphone" |
| 00:07.31 | polerin | heh |
| 00:07.57 | Strom_M | polerin: a decent headset should work fine too |
| 00:07.59 | polerin | well, we'll see. thanks though |
| 00:08.34 | JT | softphones still pick up a lot of background noise, headset or not |
| 00:10.38 | polerin | JT: yeah, i'm doing what I can to nix the bg noise while recording |
| 00:10.57 | Innatech | Certianly for temporary use, recording from an extension is fine. Once you know what you want, it is usually worth the time to sit down in front of a real mic and then do some cleanup. |
| 00:11.16 | polerin | it's more of a test anyway, I'll probably end up redoing all the prompts or having someone else do them. As good as I am at the smooth voice thing, I also have a... distinctive voice :P |
| 00:11.46 | dc3aes | UnixDog, i use nufone sometimes in my setup |
| 00:12.38 | Innatech | Lately I've been tempted by a perverse urge to record IVR prompts in kitteh.... |
| 00:12.50 | De_Mon | lastlog ThoMe |
| 00:13.06 | polerin | Innatech: heh |
| 00:13.20 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:13.45 | Innatech | " U CAN PRES 0 FOR INVISIBLE RECEPTIONIST!" [0] --> Hangup. |
| 00:14.33 | dc3aes | ha |
| 00:14.38 | Strom_M | "I CAN HAS VOICEMAIL?" |
| 00:15.01 | polerin | Actually that's really really really tempting |
| 00:15.04 | Strom_M | "IT CAN BE QUEUE TIEM NOW PL?" |
| 00:15.07 | UnixDog | dc3aes, have you been able to reach them |
| 00:15.10 | Innatech | Yeah. Best not to think about it. |
| 00:15.13 | Strom_M | s/PL/PLZ/ |
| 00:15.25 | UnixDog | been trying since friday afternoon to setup a account |
| 00:15.35 | Innatech | Please stop before I can no longer control myself. My clients WON'T understand. |
| 00:15.54 | polerin | "YOU Haz Calle teh Buisness! we are closed, but I can haz recording?" |
| 00:16.19 | Mercestes | can someone give me some handholding on the aastra config files? |
| 00:16.22 | Strom_M | IM IN UR FONE PLAYIN' MAH HARBLZ |
| 00:16.27 | polerin | LOL |
| 00:16.33 | Mercestes | actually, I'd be happy with just a copy of template config files for the aastra 480i CT |
| 00:16.36 | JT | i actually once recorded some choptel prompts |
| 00:16.45 | JT | had to put on a very redneck australian voice ;) |
| 00:17.35 | Mercestes | polerin! |
| 00:17.41 | Strom_M | i dont know what a redneck australian voice sounds like |
| 00:17.54 | Mercestes | STrom_M: Like any other australian. |
| 00:17.59 | Mercestes | m8. |
| 00:18.01 | Innatech | "That's not an endpoint. THIS is an ENDPOINT." |
| 00:18.16 | Mercestes | lol |
| 00:18.19 | JT | Innatech: pretty much, a stronger version of that |
| 00:18.21 | Mercestes | POLERIN!!!! |
| 00:18.23 | Mercestes | How's it hangin? |
| 00:18.45 | polerin | Arrrrr! |
| 00:18.54 | dc3aes | UnixDog, never tried to contact them formally.. if i understand correctly there may be an employee or two milling in here but i could have misinterpreted |
| 00:18.58 | polerin | Haveatcheeee |
| 00:18.58 | Strom_M | ....still not able to figure it out |
| 00:19.05 | fujin | IN UR *, ROUTIN UR CALLZ |
| 00:19.11 | Mercestes | Arr! |
| 00:19.14 | UnixDog | hmm |
| 00:19.24 | UnixDog | well it tells me to contact them |
| 00:19.33 | dc3aes | UnixDog, my CID function with NuFone nolonger works but I have yet to prove that it is not our cities telephone system |
| 00:19.33 | Mercestes | I will poke ye til you beg me to stop...then poke ye some more! Arr. |
| 00:19.36 | polerin | Noo |
| 00:19.38 | UnixDog | but no one answers the phone or calles you back |
| 00:19.46 | Mercestes | Surrender yer booty! |
| 00:19.50 | Mercestes | ..... |
| 00:19.55 | Mercestes | I never realized how gay pirates sound... |
| 00:20.03 | dc3aes | i feel for ya, I went through that with LES.NET for months.. |
| 00:20.09 | polerin | .. you know you're not my type Mercestes, sorry |
| 00:20.14 | Innatech | UnixDog: try and find one of their authorized resellers, and bug them for contact info that works. |
| 00:20.19 | Mercestes | Because I'm white meat and i wiggle? |
| 00:20.36 | dc3aes | i called over and over and over, and left messages, and emailed over and over.. nothing. I finally started typing in all capitol letters and they responded angrily, but hey they responded!! |
| 00:20.43 | Innatech | Spare me references to your wiggling meat, I implore you! |
| 00:20.51 | polerin | yeah uh.. basicly |
| 00:20.52 | polerin | lol |
| 00:20.59 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 00:21.08 | polerin | and I agree. I get enough references to wiggling meat in other channels |
| 00:21.11 | polerin | :/ |
| 00:21.32 | Barmal | Does anybody have a firmware release 8.2(2)SR3 for SIP for me to download? |
| 00:21.36 | JT | Strom_M: http://youtube.com/watch?v=lZc0JxdCsCg |
| 00:21.42 | fujin | lol, SIP firmwarez! |
| 00:21.51 | fujin | Barmal: for what product |
| 00:22.00 | JT | ^ Mark "Chopper" Read, inspiration for "ChopTel" ;) |
| 00:22.38 | JT | he was a hitman in the 90s, and is not a minor celebrity |
| 00:22.43 | Barmal | fujin 7970 |
| 00:22.47 | JT | but has a very redneck sounding accent |
| 00:22.50 | fujin | cisco.com |
| 00:23.01 | JT | s/not/now/ |
| 00:23.02 | Barmal | fujin: you right..... no service agreement |
| 00:23.05 | Strom_M | JT: hahahah |
| 00:23.12 | fujin | pay for a service agreement |
| 00:23.14 | fujin | stop being a cheapass |
| 00:23.45 | Barmal | fujin: nobody wants to sell for 1 phone. tryed to call over 10 cisco partners today |
| 00:24.07 | Mercestes | So how ya been? you never come to visit me anymore. :( I was hurt. |
| 00:24.18 | fujin | buy a linksys |
| 00:24.18 | *** part/#asterisk Here_And_There (n=Here_And@pool-68-238-252-162.phlapa.fios.verizon.net) |
| 00:24.20 | Mercestes | b11d even more so tho...sorry biotch....never even sent flowers. |
| 00:24.21 | JT | Strom_M: he did a whole bunch of skits for the rohnnie johns half hour, if you search youtube for "chopper" you'll find them, but i think that's the funniest :) |
| 00:24.29 | flenders | Barmal: http://www.xs4all.nl/~graver1/cisco/SIP-7960/ |
| 00:24.50 | flenders | oh, 7970 |
| 00:24.51 | Strom_M | i'm just amused you can broadcast that :) |
| 00:25.28 | JT | Strom_M: yeah, you can broadcast anything after 9:30pm except for an errect penis |
| 00:25.34 | fujin | lol |
| 00:25.43 | JT | and maybe a vagina, i'm not sure on that one |
| 00:25.50 | Barmal | fujin: too late bought couple 7970's on ebay... :( |
| 00:25.57 | JT | i hear the US is prudish about the word "shit" on radio too |
| 00:25.59 | fujin | buy a service agreement then |
| 00:26.05 | JT | Strom_M: this is free to air, not cable, btw |
| 00:26.27 | polerin | JT: yeah we're fucking lame about language |
| 00:26.34 | *** join/#asterisk CpuID2 (n=nathan@gentoo/contributor/cpuid) |
| 00:26.38 | CpuID2 | hey ppls |
| 00:26.41 | Barmal | fujin: would pay you $50 if you could find somebody who would sell service agreement for me |
| 00:26.50 | JT | polerin: here, breakfast show announces use the word "shit" whenever they feel like it |
| 00:27.10 | Barmal | fujin: nobody wants to sell s c for 2 phones |
| 00:27.10 | JT | only fuck is not permitted on radio, but it is permitted on government run radio stations |
| 00:27.14 | fujin | Barmal: now why would I do that? |
| 00:27.29 | fujin | your lack of planning does not cause an emergency for me |
| 00:28.04 | CpuID2 | anyone here know if its possible to have queues in asterisk, where members are progressively added in? |
| 00:28.08 | Barmal | fujin: cause I am tired of everybody saying buy a service agreemtn but nobody wants to sell it... |
| 00:28.15 | CpuID2 | eg. some staff here are dedicated to handling inbound calls |
| 00:28.24 | fujin | cisco.com! |
| 00:28.27 | fujin | that is where you buy them, lol |
| 00:28.28 | CpuID2 | yet other staff have other jobs, and are more just there for overflow in terms of inbound calls |
| 00:28.32 | Strom_M | CpuID2: you mean....dynamic members? |
| 00:28.32 | JT | Barmal: why buy cisco? |
| 00:28.38 | fujin | reckon |
| 00:28.44 | CpuID2 | id love to not interrupt the overflow staff until the dedicated staff havent answered for lets say 30-60 seconds |
| 00:28.46 | fujin | why buy cisco if you don't have a sc, that's just stupiud |
| 00:28.54 | CpuID2 | an dpreferably without using 2 queues, but it might be the only way :) |
| 00:29.00 | Strom_M | CpuID2: yes, look at "penalties" when you add queue members |
| 00:29.00 | CpuID2 | with a 30-60sec timeout on the first |
| 00:29.00 | Barmal | jt: cI do not know why just thought this is a good phone |
| 00:29.05 | CpuID2 | ah, thx :) |
| 00:29.20 | JT | Barmal: polycom is better, and the company has a little less contempt towards customers |
| 00:29.28 | fujin | I find the linksys ones to be great |
| 00:29.31 | fujin | if not on par with polycom |
| 00:29.35 | Mercestes | CpuID2: Be aware, penalties do not work with the "ringall" strategy. |
| 00:29.37 | Mercestes | ...I wish they did. |
| 00:29.44 | Mercestes | Maybe I should put in a feature request. |
| 00:29.47 | JT | fujin: the speakerphone in the linksys doesn't come close to polycom |
| 00:30.14 | fujin | ah |
| 00:30.15 | fujin | I see |
| 00:30.23 | fujin | haven't had any reason to test the speakerphone yet ;) |
| 00:30.26 | Barmal | too late have those highly prices ciscos here.... If I knew they need that stupid license.... |
| 00:30.45 | flenders | speakerphones on cisco 7940s are good too |
| 00:30.51 | fujin | although I'm sure polycom are great. The distributor we get all our linksys and cisco stuff off here in NZ wasn't reselling Polycom, I don't think there is a distributor for em |
| 00:30.51 | flenders | linksys is shit |
| 00:30.57 | flenders | speakerphones, I meant |
| 00:30.59 | fujin | would have to get em overseas |
| 00:31.03 | JT | flenders: they're probably designed by polycom ;) |
| 00:31.25 | JT | fujin: i thought there was one in nz |
| 00:31.37 | JT | but sometimes it is cheaper to get them from the us |
| 00:31.43 | flenders | fujin: buy it in australia |
| 00:31.44 | fujin | mm |
| 00:31.45 | JT | they only have a 1yr warranty anyway |
| 00:32.02 | fujin | not like I'm going to box up all my Linksys' and send em back for Polycoms though |
| 00:32.10 | fujin | but when I looked I couldn't find a distributor for em |
| 00:32.15 | JT | heh |
| 00:32.22 | JT | at least you didn't get grandstream |
| 00:32.40 | fujin | haha |
| 00:32.40 | Barmal | no honestly is it really so hard to get this one file or am I missing something.... I feel like I am stupid ???? |
| 00:32.40 | fujin | fuck that |
| 00:32.42 | fujin | I'm not an idiot |
| 00:32.47 | flenders | I have a bunch of linksys here too... would swap them for polycoms if I could |
| 00:33.01 | fujin | haven't had any issue with these 942's they've been great, provisioning was a snap |
| 00:33.11 | fujin | 7 lines per-phone of XML, a generic file of about 10 lines |
| 00:33.26 | Barmal | does polycom come in color screen? |
| 00:33.33 | fujin | a far shot better than the Mitel 5224's we were running *shudder* |
| 00:33.59 | JT | Barmal: no, not at the moment |
| 00:34.03 | JT | not for their phones anyway |
| 00:34.38 | flenders | Barmal: why do you need colour screen? |
| 00:35.04 | Barmal | mmy customer needs. in my house I have a sipura 841 and I am ok |
| 00:35.27 | JT | they need it why |
| 00:35.29 | Barmal | would probably change it to wip330 but not sure how good are they |
| 00:35.55 | Barmal | realestate and financing office I mean upscale shit |
| 00:36.18 | JT | i would think they need a phone that works |
| 00:36.22 | JT | colour is just a wank |
| 00:36.49 | Strom_M | step 1 in consulting: separate "needs" from "wants" |
| 00:37.07 | Barmal | :) |
| 00:38.55 | Barmal | everything would be alright if not a stupid cisco policy... You got to buy a service agreement for $8 so nobody wants to sell it or you have to buy a license for sip witch is $250 but they do not know if it will support not call manager... |
| 00:39.08 | Mercestes | needs can be easily seperated from wants the moment you put a rediculous price tag on it. |
| 00:42.08 | flenders | so, they wanted colour phones, but wanted it cheap from ebay... |
| 00:42.28 | fujin | sounds like someone is a dodgy contractor |
| 00:44.02 | flenders | I bought a couple of cisco phones for demo, but if clients ever want them, I'll buy them from a proper reseller |
| 00:47.17 | fujin | yeah |
| 00:47.20 | fujin | at least then you can buy licenses |
| 00:47.21 | Barmal | no they asked me to buy 8 phones. I was trying to save money and got on ebay... :) |
| 00:47.31 | *** join/#asterisk Here_And_There (n=Here_And@pool-68-238-252-162.phlapa.fios.verizon.net) |
| 00:47.44 | Strom_M | and now you know what kind of mess being a cheapskate gets you into |
| 00:47.47 | Strom_M | ~cheap |
| 00:47.47 | jbot | i guess cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 00:47.56 | fujin | cheapskate=fail |
| 00:48.09 | fujin | buy linksys 962's if you want colour |
| 00:48.16 | fujin | they're only about $400nzd/handset |
| 00:48.32 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.91) |
| 00:48.54 | Barmal | but you guys gave me an idea. Let me you: if i buy new phones they will come with service agreement and those files? |
| 00:48.58 | *** join/#asterisk wunderkin (n=wunderki@dslstat-ppp-95.fastq.com) |
| 00:49.07 | [[blah]asfd | i have a call in queue: Callers: |
| 00:49.07 | [[blah]asfd | <PROTECTED> |
| 00:49.11 | [[blah]asfd | it is not a real call |
| 00:49.15 | [[blah]asfd | how can I kill it |
| 00:49.32 | [[blah]asfd | I tried doing a soft hangup SIP/3712-00b4d9e0 |
| 00:49.35 | fujin | Barmal: no, you have to purchase a license at the same time |
| 00:50.07 | *** part/#asterisk Here_And_There (n=Here_And@pool-68-238-252-162.phlapa.fios.verizon.net) |
| 00:50.08 | [[blah]asfd | i get SIP/3712-00b4d9e0 is not a known channel |
| 00:50.53 | Barmal | fujin: yes but if I buy a license I will have a this .cop file too yes or no? |
| 00:51.15 | Barmal | then I can upload to those couple phones? |
| 00:51.15 | *** join/#asterisk Here_And_There (n=Here_And@pool-68-238-252-162.phlapa.fios.verizon.net) |
| 00:51.27 | fujin | no idea |
| 00:51.30 | fujin | I don't roll ciscoz ;] |
| 00:51.41 | fujin | 1license per phone though, I think you generally enroll a license to a phone |
| 00:51.52 | JT | Barmal: their stuff is licensed per phone |
| 00:52.10 | JT | Barmal: if you misuse firmware on unlicensed phones, they can sue you |
| 00:52.33 | Barmal | ok but technically I mean it can be done... |
| 00:52.51 | fujin | probably |
| 00:53.06 | fujin | you'd be better off selling them on eBay and buying alternative more friendly phones |
| 00:53.07 | JT | just buy a polycom phone, much easier |
| 00:53.09 | fujin | polycoms, linksys |
| 00:53.10 | fujin | ;][ |
| 00:53.14 | JT | and they work better behind nat |
| 00:53.21 | JT | ciscos have almost no hope behind nat |
| 00:54.03 | Barmal | does polycom have intercom / autoanswer? |
| 00:54.15 | JT | you can set that using a sip header |
| 00:56.56 | Strom_M | JT: FWIW, my ciscos work fine behind nat |
| 00:57.12 | JT | interesting |
| 00:57.15 | JT | what models and setup |
| 00:57.21 | Strom_M | 7960 |
| 00:57.37 | Strom_M | cisco 7960 -> linksys wrt54g -> asterisk |
| 00:57.44 | JT | no Internet? |
| 00:57.58 | Strom_M | no internet |
| 00:58.11 | JT | is it actually being NATed? |
| 00:58.15 | Strom_M | though if i do add internet, it works fine |
| 00:58.16 | Strom_M | yes |
| 00:58.22 | Mercestes | I've got ciscos workign behind nat too. |
| 00:58.32 | JT | why do you use nat for lan to lan comms? |
| 00:58.41 | Mercestes | There is some really cryptic "media interface gateway" or some crap like that in the .cnf files, but...yea, it works. |
| 01:02.09 | fujin | yuck |
| 01:02.10 | fujin | nat for lan to lan |
| 01:02.28 | fujin | arrrh, why do people try to nat between phones and * |
| 01:02.46 | CpuID2 | Mercestes, If the strategy is defined as 'ringall', then only those available members with the lowest priorities will ring. In the example above, if 200 is not busy, then only 200 will ring. If 200 is busy, then only 201 and 203 will ring. If 200, 201 and 203 are busy, then 202 will ring. |
| 01:02.53 | CpuID2 | sound about right? :) |
| 01:03.06 | CpuID2 | looks perfect to me, means i can have the dedicated staff as priority 1, and everyone else as priority 2 :) |
| 01:03.15 | Mercestes | CpuID2: Wrong. |
| 01:03.20 | CpuID2 | oh?, whack |
| 01:03.22 | CpuID2 | http://www.voip-info.org/wiki/view/Asterisk+call+queues |
| 01:03.25 | fujin | bow-wow. |
| 01:03.26 | CpuID2 | somethings outta date :P |
| 01:03.32 | CpuID2 | meh |
| 01:03.43 | *** join/#asterisk obnauticus (n=obnautic@c-71-59-162-60.hsd1.wa.comcast.net) |
| 01:03.57 | Mercestes | oh, yea right. |
| 01:04.04 | Qwell | CpuID: See doc/queue-callback-members.txt, or whatever it's called |
| 01:04.06 | Mercestes | but by busy they mean "on the phone" not "did not answer." |
| 01:04.09 | CpuID2 | ok coo |
| 01:04.15 | CpuID2 | ya...thats the only downside :) |
| 01:04.16 | CpuID2 | timeouts |
| 01:04.17 | Qwell | That shows a good way of doing priorities |
| 01:04.24 | CpuID2 | Qwell, thx |
| 01:04.29 | JT | fujin: umm, nat works fine when setup right, and the reason is probably because NAT is very widely implemented for a variety of reasons |
| 01:07.41 | *** join/#asterisk Eminence (n=achin@ip68-227-184-126.ri.ri.cox.net) |
| 01:08.35 | Eminence | with sip, do any ports need to be opened, if you only make outgoing calls? |
| 01:09.03 | Strom_M | yes |
| 01:09.13 | Strom_M | because you have to receive audio too, you know |
| 01:10.01 | Eminence | incoming audio isn't sent along the same [outgoing] connection that was just opened? |
| 01:10.10 | *** join/#asterisk Zilasb (n=litcom@c-24-99-8-41.hsd1.ga.comcast.net) |
| 01:10.11 | JT | indeed |
| 01:10.14 | JT | different ports |
| 01:10.25 | Strom_M | different protocol, actuallu |
| 01:10.29 | Eminence | curious |
| 01:10.33 | Strom_M | SIP handles signaling, RTP handles media |
| 01:10.39 | Eminence | ah |
| 01:10.47 | JT | yes, and rtp uses 2 ports |
| 01:10.49 | JT | one each way |
| 01:10.55 | Eminence | what are the ports? |
| 01:11.15 | JT | it coses ports between 10000-20000 with the default asterisk rtp.conf |
| 01:11.20 | JT | choses |
| 01:11.26 | Strom_M | chooses |
| 01:11.43 | JT | right |
| 01:11.58 | Eminence | if my asterisk server is behind a nat, can i tighten that range down and port forward? |
| 01:12.04 | Eminence | or just put it on a dmz? |
| 01:12.34 | JT | Eminence: what needs to access asterisk? |
| 01:12.41 | Greenbox | you can make it smaller, there's two options iirc rtpstart and rtpend |
| 01:13.07 | JT | there's generally no great benefit to making it smaller |
| 01:13.10 | Eminence | just some sip clients that sit on the inside of the nat (both clients are * are 192.168.1.x) |
| 01:13.17 | Eminence | and i just make outgoing calls with iconnecthere |
| 01:13.35 | JT | ok, so no clients accessing asterisk from the outside world? |
| 01:13.41 | Eminence | correct |
| 01:13.42 | JT | no port forwarding needed. |
| 01:13.51 | shido6 | dmz bad |
| 01:13.53 | Eminence | ok |
| 01:13.58 | JT | just need stuff to not be firewalled |
| 01:14.53 | Eminence | well, then next question: when trying to dial out with ICH, i hear it ringing, but it eventually times out. asterisk reports Auto-congesting, and circuit is busy |
| 01:15.25 | JT | the number was busy/not answering maybe? |
| 01:15.43 | Eminence | i tried dialing my cellphone, so i know it's not busy |
| 01:16.10 | Eminence | so even if my cell didn't pick up, i should at least gotten my voice mail |
| 01:16.24 | JT | maybe you passed the number incorrectly? |
| 01:16.45 | Eminence | meaning i typed in my cell number incorrectly? no, that's not the case |
| 01:17.56 | JT | meaning maybe asterisk sent it not in the format your provider would like |
| 01:18.22 | Eminence | ah. well, that's a possibility, however the configuration i'm using now worked several months ago |
| 01:18.37 | JT | i see |
| 01:18.46 | JT | surely there's some info on your console |
| 01:19.01 | Eminence | sure. i can pastebin it, if you'd like |
| 01:19.24 | JT | if you wish |
| 01:20.51 | *** join/#asterisk Ebola (n=Ebola@host86-136-134-202.range86-136.btcentralplus.com) |
| 01:21.08 | *** join/#asterisk guillote_GNU (n=guillote@190.7.23.224) |
| 01:22.28 | *** join/#asterisk wunderkin- (n=wunderki@dslstat-ppp-95.fastq.com) |
| 01:23.11 | [[blah]asfd | is there any other way to kill a "phantom" call without rebooting the server besides soft hangup <channel> ? |
| 01:23.49 | Eminence | JT, http://pastebin.ca/600691 |
| 01:23.59 | [[blah]asfd | have a call in queue that does not really exist and it is tying up the queue. Callers:1. SIP/3712-00b4d9e0 (wait: 148:41, prio: 0) |
| 01:24.09 | Eminence | (amazingly, pastebin.ca has syntax hilighting for asterisk!) |
| 01:24.18 | [[blah]asfd | i have tried to soft hang it up, reload app_queue.so and reload chan_sip also... nothing works. |
| 01:24.25 | [[blah]asfd | cannot reboot right now... this is killing me. |
| 01:24.27 | *** part/#asterisk UnixDog (n=UnixDog@adsl-69-234-211-33.dsl.irvnca.pacbell.net) |
| 01:24.43 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 01:25.06 | JT | Eminence: ok, now try it again with sip debug turned on |
| 01:25.20 | JT | Eminence: btw, SetCallerID is deprecated |
| 01:26.08 | Eminence | so i noticed. i'll remove it from the dialplan |
| 01:27.32 | Eminence | this is quite a lot of debugging data. anything i should be looking for in particular? |
| 01:28.40 | JT | you should be looking to pastebin an entire call from start to end :) |
| 01:28.58 | Eminence | okay :) |
| 01:31.50 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 01:32.40 | Eminence | JT, http://pastebin.ca/600711 |
| 01:35.17 | JT | err BT110 :P |
| 01:35.19 | JT | anyway |
| 01:35.22 | JT | in sip.conf |
| 01:35.26 | JT | set externip= |
| 01:35.32 | JT | and localnet= |
| 01:36.15 | Eminence | heh, the BT100 was the cheapest one out there :) |
| 01:36.22 | Sedorox | bt100 == cheap.. but works |
| 01:36.26 | Sedorox | in my experience |
| 01:36.27 | JT | shittest |
| 01:36.33 | JT | Sedorox: not in most others' |
| 01:36.36 | Sedorox | bt101/110 is ugly (whatever it was) |
| 01:36.50 | Sedorox | dunno.. mine's in a box somewhere |
| 01:36.53 | Sedorox | since I got a 7960 |
| 01:36.54 | *** join/#asterisk obnauticus (n=obnautic@c-71-59-162-60.hsd1.wa.comcast.net) |
| 01:36.56 | *** join/#asterisk LakeSolon (n=blake@64-83-205-22.dhcp.stcd.mn.charter.com) |
| 01:37.03 | Eminence | so far, i'm happy with it. granted, i don't use it everyday |
| 01:37.17 | Sedorox | if you use it for hwat it is.. a phone.. its not really that bad |
| 01:37.25 | JT | it's easy to be happy with something if you've never used better ;) |
| 01:37.34 | Eminence | touche |
| 01:37.44 | JT | the audio on grandstreams is terrible |
| 01:37.53 | JT | Eminence: tried those config changes? |
| 01:38.44 | Sedorox | JT: well likw I said.. I not have a 7960.. and debating on a IP600... yes.. they are better phones.. but if you want something simple... but as all things.. you get what you pay for |
| 01:39.07 | JT | heh |
| 01:39.23 | Eminence | trying them now. so far, no luck. i'm just double checking them now |
| 01:40.26 | Nephfl | this polycom config crap is taking forever |
| 01:40.57 | [[blah]asfd | could anyone help me out with this error: |
| 01:40.57 | Nephfl | anybody got "buddy watching" set up? |
| 01:40.58 | [[blah]asfd | [Jul 2 16:22:04] ERROR[5477]: chan_zap.c:8158 zt_pri_error: !! Got reject for frame 118, retransmitting frame 118 now, updating n_r! |
| 01:41.13 | JT | uhoh |
| 01:41.20 | JT | running a real pri? |
| 01:41.25 | [[blah]asfd | yes |
| 01:41.35 | JT | how often do you get the error? |
| 01:41.40 | [[blah]asfd | i get that error on two different pris in two different locations |
| 01:41.44 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:41.46 | [[blah]asfd | nearly every time i place a call |
| 01:41.51 | JT | err |
| 01:41.53 | [[blah]asfd | let me get my configs up on pastbin. |
| 01:41.56 | JT | check zap timing |
| 01:41.58 | JT | zttest |
| 01:42.13 | [[blah]asfd | ok... zttest? from CLI>? |
| 01:42.18 | JT | yes |
| 01:42.25 | JT | run it for a little while |
| 01:42.30 | [[blah]asfd | must not have it installed. |
| 01:42.31 | [[blah]asfd | hang on. |
| 01:42.33 | JT | keep a look out for the lowest scores |
| 01:42.41 | [[blah]asfd | version 1.4.4 |
| 01:42.47 | [[blah]asfd | is that in the make menuconfig? |
| 01:42.50 | Nephfl | anybody know if i can get this polycom to monitor any channel via hint? or am i pissin' up a rope? |
| 01:42.52 | JT | and what scores you generally see |
| 01:42.57 | JT | it'd be in zaptel |
| 01:43.20 | [[blah]asfd | hmmm. it was installed. hang on..... |
| 01:43.32 | Eminence | JT, no luck. |
| 01:44.02 | Eminence | should nat= be set to something in sip.conf ? |
| 01:44.13 | shido6 | yes |
| 01:44.19 | [[blah]asfd | ./zttest in /usr/src/zaptel-1.4.2.1 |
| 01:44.21 | [[blah]asfd | that did the trick |
| 01:44.41 | [[blah]asfd | 99.9whatever and up on most of the scores... I am looking for something odd? |
| 01:45.11 | Eminence | same behavior with nat undefinied, and nat=yes |
| 01:46.00 | *** part/#asterisk Here_And_There (n=Here_And@pool-68-238-252-162.phlapa.fios.verizon.net) |
| 01:47.22 | JT | [[blah]asfd: it's not "whatever" |
| 01:47.26 | JT | you need to be VERY SPECIFIC |
| 01:47.29 | JT | for zttest |
| 01:48.15 | JT | Eminence: i think the problem is that your ITSP is down. |
| 01:48.24 | JT | or at least that server is |
| 01:49.13 | *** join/#asterisk lgc (n=lgc@189.146.42.145) |
| 01:49.26 | [[blah]asfd | JT:... just letting it run for a bit, then I will pastbin it all up. |
| 01:49.27 | Eminence | okay |
| 01:50.18 | *** join/#asterisk Here_And_There (n=Here_And@pool-68-238-252-162.phlapa.fios.verizon.net) |
| 01:50.18 | [[blah]asfd | JT: ok, here are the results: http://pastebin.ca/600735 |
| 01:50.18 | JT | [[blah]asfd: are any of the errors coming up while zttest is running? |
| 01:50.27 | [[blah]asfd | I did not see any. |
| 01:50.29 | Eminence | is there some public asterisk server i can connect to, maybe do an echo test? (just to make sure that my end is working correctly) |
| 01:51.02 | JT | [[blah]asfd: your zap timing results aren't completely abysmal, but they also are not acceptable for digital circuits |
| 01:51.11 | JT | [[blah]asfd: you have a zap timing issue |
| 01:53.16 | [[blah]asfd | what would good timing look like? |
| 01:53.38 | JT | never ever falling below 99.975586% |
| 01:54.39 | [[blah]asfd | ok... so take a look and see if this is better: http://pastebin.ca/600741 |
| 01:55.04 | blitzrage | yo yo |
| 01:55.30 | JT | [[blah]asfd: it looks better, but it still falls below a couple of times |
| 01:56.24 | CpuID2 | JT, Best: 100.000000 -- Worst: 99.731445 -- Average: 99.938286 lol looks like i fall below your standards :P |
| 01:56.49 | JT | CpuID2: i usually ignore the totals, as zttest is buggy and can't count half of the time |
| 01:56.50 | [[blah]asfd | ok... so how can I adjust... Here is my zapata.conf: http://pastebin.ca/600743 |
| 01:57.09 | JT | i just look on the screen to see what the general pattern is, and what the worst numbers are |
| 01:57.28 | JT | CpuID2: 99.73%, if you actually got that, is awful |
| 01:57.36 | JT | and would cause a frame slip on digital |
| 01:57.51 | Eminence | i'm curious: what is zttest actually measuring? |
| 01:58.04 | JT | accuracy of your zaptel timing |
| 01:58.07 | CpuID2 | hehe |
| 01:58.13 | JT | Eminence: any interrupt sharing? |
| 01:58.20 | CpuID2 | when you say digital, as in T1/E1? |
| 01:58.31 | CpuID2 | or to put that better, PRI/BRI? :) |
| 01:58.35 | JT | CpuID2: yes, bri/pri/channelised t1 |
| 01:58.45 | Eminence | on my system? i suspect there might be, but i'm not using zap |
| 01:58.45 | lgc | Hi! I just installed Asterisk on my machine and I've never used it. How can I send a fax? Thanks. |
| 01:58.49 | CpuID2 | were only using tdm400p cards, analog only here |
| 01:58.54 | CpuID2 | seems ok :) |
| 01:58.59 | JT | Eminence: whoops :) |
| 01:59.14 | CpuID2 | are active calls likely to affect zttest's timing output? |
| 01:59.17 | JT | [[blah]asfd: any interrupt sharing? |
| 01:59.21 | JT | CpuID2: they can |
| 01:59.27 | Eminence | JT: :) |
| 01:59.30 | CpuID2 | theres at least 2-3 active calls on there atm so :) |
| 01:59.51 | JT | [[blah]asfd: you have NFAS? |
| 02:00.08 | JT | CpuID2: on a good system, calls will make no difference |
| 02:00.15 | [[blah]asfd | JT: yes |
| 02:00.20 | [[blah]asfd | :-) |
| 02:00.27 | [[blah]asfd | everyone had a smiley, so i did one too |
| 02:00.40 | JT | [[blah]asfd: your NFAS only uses 1 D channels? |
| 02:00.49 | JT | i always thought 2 was the minimum |
| 02:00.57 | CpuID2 | np |
| 02:01.08 | [[blah]asfd | seems to work |
| 02:01.20 | JT | [[blah]asfd: if that one pri dies, you will lose all 4 PRIs |
| 02:01.24 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 02:01.29 | [[blah]asfd | yeah... that I know |
| 02:01.31 | JT | you should check with the telco to see what the backup D channel is |
| 02:01.35 | JT | and set it up |
| 02:01.50 | [[blah]asfd | the do not provide me one... have to pay extra |
| 02:01.58 | JT | what tightarses |
| 02:02.05 | [[blah]asfd | yeah. |
| 02:02.05 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:02.10 | JT | what telco is this? |
| 02:02.15 | [[blah]asfd | Global Crossing |
| 02:02.20 | JT | hrm |
| 02:02.39 | JT | zap timing is not usually related to your config files, but i'd like to see zaptel.conf anyway |
| 02:06.25 | JT | [[blah]asfd: ? |
| 02:07.02 | lgc | Hi! I just installed Asterisk on my machine and I've never used it. How can I send a fax? Thanks. |
| 02:07.20 | JT | with great difficulty |
| 02:07.49 | [[blah]asfd | zaptel.conf: http://pastebin.ca/600752 |
| 02:09.56 | JT | that looks fine |
| 02:10.01 | JT | [[blah]asfd: so any irq sharing? |
| 02:10.20 | [[blah]asfd | no |
| 02:10.46 | JT | what sort of machine is it? |
| 02:10.52 | [[blah]asfd | http://pastebin.ca/600755 |
| 02:10.56 | JT | and what card? |
| 02:12.19 | [[blah]asfd | 1500MHZ ITX with a TE405P 1GB ram |
| 02:12.34 | JT | omg |
| 02:12.36 | JT | VIA? |
| 02:12.38 | [[blah]asfd | yeah |
| 02:12.41 | JT | no wonder |
| 02:12.41 | [[blah]asfd | :-D |
| 02:12.45 | JT | get a real machine |
| 02:12.55 | JT | that thing can't handle 4 PRIs |
| 02:13.12 | [[blah]asfd | well, it sits along side all of my HP DL380s |
| 02:13.40 | JT | via is evil, especially with interrupt fussy stuff like asterisk |
| 02:13.52 | JT | well digium/zaptel, more specifically |
| 02:14.02 | [[blah]asfd | yeah... so lost cause on that one? |
| 02:14.16 | [[blah]asfd | anyone want a TE405P? |
| 02:14.18 | JT | i think so, you can try swapping the pci slots if there are any others |
| 02:14.24 | [[blah]asfd | nope just one |
| 02:14.39 | JT | selling the card instead of the via? ;) |
| 02:14.49 | [[blah]asfd | its extra |
| 02:14.51 | [[blah]asfd | side project |
| 02:14.59 | [[blah]asfd | can shoot the via ;-) |
| 02:15.04 | [[blah]asfd | cards worth money |
| 02:15.19 | JT | heh, so you have something else running the pris? |
| 02:15.32 | [[blah]asfd | well... these are extra PRIs |
| 02:15.35 | JT | well i could be interested, it all depends ;) |
| 02:15.38 | JT | heh ok |
| 02:15.39 | [[blah]asfd | was trying to set up a SIP gateway |
| 02:15.49 | [[blah]asfd | make me an offer |
| 02:16.04 | [[blah]asfd | PM me |
| 02:17.07 | lgc | Hi! I just installed Asterisk on my machine and I've never used it. How can I send a fax? Thanks. |
| 02:17.40 | [[blah]asfd | lgc: move from asterisk to hylafax |
| 02:17.54 | [[blah]asfd | you interested in someone doing fax for you? |
| 02:19.14 | lgc | [[blah]asfd, I'm interested in sending a fax from my computer. |
| 02:19.29 | lgc | [[blah]asfd, what is hylafax? |
| 02:19.29 | [[blah]asfd | lgc: check out hylafax and iax modem |
| 02:19.40 | [[blah]asfd | theres an awesome site that has info on that... |
| 02:19.45 | [[blah]asfd | let me see if i can find it... |
| 02:19.48 | [[blah]asfd | ummm....... |
| 02:19.50 | [[blah]asfd | google.com |
| 02:19.52 | [[blah]asfd | ;-) |
| 02:19.55 | [[blah]asfd | just kidding. |
| 02:20.14 | [[blah]asfd | enter it into google and it will take you to the main site. |
| 02:21.41 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:22.08 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 02:23.51 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:24.32 | lgc | [[blah]asfd, but can you tell me if Asterisk is not good for sending faxes? |
| 02:28.15 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-63-246.lns2.syd6.internode.on.net) |
| 02:28.36 | [[blah]asfd | lgc: most people have moved away from asterisk for fax. |
| 02:28.48 | [[blah]asfd | there is asterfax, but I have not heard many success stories with that. |
| 02:28.58 | *** join/#asterisk johann8384 (n=johann83@gateway.myogre.com) |
| 02:30.59 | JT | lgc: asterisk can send faxes with spandsp if you have a pstn connection |
| 02:31.39 | lgc | [[blah]asfd, JT, I see, thanks. |
| 02:33.24 | *** join/#asterisk saftsack (n=saftsack@pD9E07F8F.dip.t-dialin.net) |
| 02:35.23 | *** join/#asterisk Marshall- (n=Marshall@cpe-76-181-113-136.columbus.res.rr.com) |
| 02:51.33 | [[blah]asfd | I have a server (server_02) which has 4 t1s in it. all calls from phones on this server to the t1s are high quality and sound great. very stable. I connected phones from server_01 to server_02 to call out over the server_02 t1s but the call quality for those phones is scratchy and sometimes drops. I have set disallow=all allow=gsm on both sides of the connection as well as other combos, but it makes no effect. what else can I look into to correct this |
| 02:52.16 | alrs | [[blah]asfd Does server_01 have a timing source other than ztdummy? |
| 02:52.30 | JT | gsm isn't the best |
| 02:52.36 | lgc | [[blah]asfd, JT, does Hylafax need Asterisk to run? |
| 02:52.48 | JT | no |
| 02:53.03 | lgc | JT does it have a GUI? |
| 02:53.14 | JT | probably not, no idea |
| 02:53.32 | [[blah]asfd | alrs: yes... it has a failed T1 card in it. |
| 02:53.42 | [[blah]asfd | alrs: that was for you |
| 02:54.24 | [[blah]asfd | hylafax... no on the gui and no on asterisk. |
| 02:54.35 | lgc | JT, do I need a modem to use it or my Ethernet link will do? |
| 02:54.36 | [[blah]asfd | it will use asterisk if you want to use iaxmodem instead of a hardware modem |
| 02:54.50 | lgc | [[blah]asfd, Oh. |
| 02:54.53 | [[blah]asfd | you will need a true hardware modem, not a winmodem |
| 02:54.55 | JT | lgc: you need a modem or spandsp, how else will it connect to the pstn? |
| 02:55.07 | Qwell | JT: magic |
| 02:55.11 | [[blah]asfd | spandsp also is an option. |
| 02:55.16 | [[blah]asfd | we used hylafax |
| 02:55.20 | JT | [[blah]asfd: spandsp works with iaxmodem |
| 02:55.24 | [[blah]asfd | iaxmodem |
| 02:55.28 | JT | iaxmodem is useless without spandsp |
| 02:55.30 | [[blah]asfd | is what I meant |
| 02:55.49 | JT | you must use iaxmodem to use hylafax with asterisk |
| 02:56.02 | alrs | [[blah]asfd Any insight as to why the T1 card failed? |
| 02:56.03 | JT | and iaxmodem must use spandsp :) |
| 02:56.14 | lgc | JT, sorry, I don't know what 'spandsp' is. I figure I can get to my VoIP to actually make the call. |
| 02:56.27 | [[blah]asfd | alrs: dunno the details... digium just said to send it back |
| 02:56.29 | JT | it's a library |
| 02:56.37 | JT | lgc: you figure wrong |
| 02:56.42 | JT | you cannot fax over VoIP |
| 02:56.53 | alrs | [[blah]asfd I'd switch out to ztdummy to see if things got worse or better |
| 02:57.04 | JT | you need to use FoIP for faxing, ie. T.38, asterisk does not do T.38 endpoint |
| 02:57.21 | [[blah]asfd | alrs: what is the way to do that? |
| 02:57.46 | alrs | [[blah]asfd load the ztdummy module instead of the wc-whatever module |
| 02:58.05 | lgc | JT, what do I need for FoIP? |
| 02:58.17 | JT | something that supports T.38 |
| 02:58.27 | lgc | JT, such as? |
| 02:58.41 | JT | lgc: what are you trying to do anyway? |
| 02:58.54 | lgc | JT, just send out a fax from my machine. |
| 02:59.01 | alrs | [[blah]asfd Maybe try using SIP instead of IAX |
| 02:59.06 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 02:59.08 | JT | lgc: do you have a landline? |
| 02:59.42 | lgc | JT, (sorry, I'm a newbie!) by a landline you mean a regular phone line? |
| 03:00.07 | JT | a phone line of some description, yes |
| 03:00.15 | JT | that travels over land |
| 03:00.15 | lgc | JT, yes. |
| 03:00.25 | JT | as opposed to one that travels over air |
| 03:00.51 | JT | lgc: it's easier to hook up a fax machine to that line |
| 03:01.11 | *** join/#asterisk wulfy814 (n=askme@pool-71-240-198-202.alt.east.verizon.net) |
| 03:01.39 | lgc | JT, I agree. And it'd be easier to just go to Office Max and have it sent. |
| 03:01.43 | wulfy814 | ok this is killing me, I can't get asterisk 1.2.17 to recognize pressing "0" for operator during voicemail |
| 03:01.53 | lgc | But it's no fun that way. |
| 03:01.55 | JT | lgc: there are also email to fax services out there |
| 03:01.56 | wulfy814 | I have a basic std-exten Macro that's called |
| 03:03.02 | snuff-work | lgc u can just send ulaw version of fax signal.. but it rarely worked (before T38) but i did attempt it |
| 03:03.07 | lgc | JT, I figured I could have some fun and learn something by trying to fax from my computer, if the learning curve isn't too steep (that's why I ask about the GUI). |
| 03:03.37 | JT | lgc: yeah we generally advise against GUIs |
| 03:03.38 | lgc | snuff-work, what is ulaw? |
| 03:03.44 | JT | i take it you're not using asterisk already |
| 03:04.37 | lgc | JT, never used it. In fact, I installed it and removed it a while a go when you guys told me it wasn't necessary to use Hylafax. |
| 03:04.42 | snuff-work | urmm.. a codec.. |
| 03:04.57 | wulfy814 | http://www.pastebin.ca/600794 is the macro called |
| 03:04.58 | JT | lgc: i think you should do some reading |
| 03:05.01 | JT | ~thebook |
| 03:05.02 | jbot | thebook is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:06.18 | [[blah]asfd | alrs: of these modules, what all needs to be unloaded: zttranscode,wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2 |
| 03:06.25 | [[blah]asfd | anything wc...? |
| 03:06.48 | JT | [[blah]asfd: err, they pretty much should all NOT be loaded |
| 03:07.01 | JT | [[blah]asfd: why do you have so many loaded? |
| 03:07.03 | [[blah]asfd | so can i just rmmod zaptel? |
| 03:07.11 | [[blah]asfd | dunno... |
| 03:07.20 | JT | no, you will need to get rid of them first |
| 03:07.27 | [[blah]asfd | ok |
| 03:08.44 | wulfy814 | no ideas on my wanting to press "0" to get out of voicemail? |
| 03:08.57 | wulfy814 | I have it working on an older system |
| 03:09.29 | wulfy814 | 1.2.10 or something like that |
| 03:12.43 | *** join/#asterisk demiv (n=demiv@134.42.128.66.PPPoECali.dynamic.telesat.net.co) |
| 03:13.39 | [[blah]asfd | i think i have a locked up zap channel and cannot kill it. tried zap destroy channel Zap/11-1. it did not die. any other way to kill it? |
| 03:13.41 | snuff-work | wulfy814, i could have sworn its a flag u pass to vm command.. i think the extension should be 'o' the letter |
| 03:13.58 | [[blah]asfd | soft hangup also failed. |
| 03:14.42 | wulfy814 | snuff-work: hmm I have extension 'o' the letter defined in my macro |
| 03:14.51 | wulfy814 | but not sure about a flah |
| 03:14.53 | wulfy814 | flag that is |
| 03:14.55 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 03:16.25 | wulfy814 | you were right about the flag - it's in voicemail.conf |
| 03:16.28 | wulfy814 | thank you |
| 03:16.57 | [[blah]asfd | I cannot kill wct4xxp: Could anyone help me out here. http://pastebin.ca/600800 |
| 03:18.12 | *** join/#asterisk johann8384 (n=johann83@gateway.myogre.com) |
| 03:30.24 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 03:34.42 | [[blah]asfd | what other possibilities could there be causing poor sound quality between my two servers. One sounds fine on its own t1s, but when the other one tries to use its t1s those calls sound terrible. |
| 03:35.09 | JT | what protocol |
| 03:35.14 | [[blah]asfd | IAX2 |
| 03:35.17 | [[blah]asfd | gsm |
| 03:35.25 | JT | is trunk=yes or trunk=no? |
| 03:35.33 | [[blah]asfd | yes |
| 03:35.34 | JT | why on earth would you use gsm codec on a lan? |
| 03:35.39 | JT | set it to no |
| 03:35.41 | [[blah]asfd | ok |
| 03:35.56 | JT | use alaw or ulaw, depending upon your country |
| 03:37.24 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:39.36 | [[blah]asfd | set it to use ulaw only and it gives me these errors: Call rejected by 10.0.1.182: Unable to negotiate codec |
| 03:39.50 | [[blah]asfd | Rejected connect attempt from 10.0.1.187, requested/capability 0x2/0xff03 incompatible with our capability 0xf804. |
| 03:40.02 | JT | set BOTH sides to ulaw :) |
| 03:40.11 | [[blah]asfd | I thought i did. |
| 03:40.18 | [[blah]asfd | reloaded and everything... trying one more time. |
| 03:42.03 | [[blah]asfd | yeah, same thing... double checked and reloaded each one twice. |
| 03:42.29 | JT | pastebin.ca the iax.conf files minus passwords |
| 03:43.02 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-17-47-67.pskn.east.verizon.net) |
| 03:43.30 | [[blah]asfd | ok... hang on |
| 03:45.24 | [[blah]asfd | http://pastebin.ca/600836 |
| 03:46.11 | [[blah]asfd | error from 10.0.1.182: http://pastebin.ca/600837 |
| 03:46.42 | JT | they aren't the whole files are they? |
| 03:46.49 | JT | and you only have user entries on each server? |
| 03:47.16 | [[blah]asfd | those are just the entries for those machines.. not using any globals or anything. |
| 03:47.36 | [[blah]asfd | i guess they are user... should be friend? |
| 03:48.03 | [[blah]asfd | i dont have a user and a peer for each connection... just the user |
| 03:48.15 | JT | yes, if you want to send and receive calls |
| 03:48.24 | JT | well the globals are important |
| 03:49.09 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 03:49.10 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:49.12 | [[blah]asfd | is there a ctrl+a for nano ;-) |
| 03:50.29 | JT | a what? |
| 03:50.37 | [[blah]asfd | select all? |
| 03:50.49 | *** join/#asterisk linex (n=Krooks@202.184.116.210) |
| 03:52.19 | JT | ah |
| 03:54.19 | [[blah]asfd | http://pastebin.ca/600850 |
| 03:54.23 | *** join/#asterisk bmg505 (n=leon@196.209.180.242) |
| 03:54.33 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 03:54.34 | [[blah]asfd | globals are the same on both servers |
| 03:56.07 | JT | [[blah]asfd: bandwidth=high |
| 03:57.21 | *** join/#asterisk DanielHolth (n=dholth@2002:18aa:c06f:1:211:2fff:fed5:f7c7) |
| 03:58.08 | DanielHolth | Hi, can anyone recommend the best local number portability VOIP provider? |
| 03:58.09 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 03:58.29 | JT | where on earth for? |
| 03:59.27 | *** join/#asterisk |R (i=bob@modemcable241.28-203-24.mc.videotron.ca) |
| 04:00.52 | |R | Is anyone familiar with ekiga |
| 04:01.30 | |R | i can get it to register with asterisk, call the demo, but i can't hear the demo back... and both end point should be on the same subnet... :/ |
| 04:01.54 | |R | and it works from an SPA-2102 with a hardphone... |
| 04:03.59 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 04:04.49 | *** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal) |
| 04:05.16 | Aces1Up | just a quick question for you professionals, to use meetme in asterisk 1.4 do i need a timing device? |
| 04:06.43 | *** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 04:08.08 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 04:13.23 | [[blah]asfd | Aces1Up: yes |
| 04:13.30 | [[blah]asfd | if you do not have a card.. use ztdummy |
| 04:13.39 | Aces1Up | thank buuuuudy, i'm the weasel! |
| 04:13.52 | |R | hehe |
| 04:14.41 | JT | Aces1Up: sometimes ztdummy is not accurate enough |
| 04:15.03 | JT | [[blah]asfd: tried the change i suggested? |
| 04:15.37 | [[blah]asfd | yeah... asking for feedback from users now. |
| 04:15.43 | JT | so it works? |
| 04:15.51 | [[blah]asfd | dunno yet |
| 04:16.00 | [[blah]asfd | still waiting for feedback from people on the phones. |
| 04:16.01 | JT | well you aren't getting codec errors |
| 04:16.09 | [[blah]asfd | oh... havent tried it like that. |
| 04:16.13 | [[blah]asfd | i see what you are saying. |
| 04:16.19 | JT | well what did you try then? |
| 04:16.31 | [[blah]asfd | just changing the bandwidth to high |
| 04:16.46 | JT | and codecs are now ulaw not gsm? |
| 04:16.59 | [[blah]asfd | changing it back... |
| 04:16.59 | [[blah]asfd | sorry |
| 04:17.02 | [[blah]asfd | misunderstood |
| 04:17.24 | JT | you need to set bandwidth=high and allow=ulaw |
| 04:17.31 | JT | gsm is pointless on a lan |
| 04:17.33 | [[blah]asfd | yeah... |
| 04:17.36 | JT | lower sound quality |
| 04:17.38 | JT | higher cpu load |
| 04:18.13 | Aces1Up | hrmm i'm reading the future of telephony book, and the say to uncomment the ztdummy in the makefile to compile it, but i can't seem to find the line from the book, is it different in 1.4? |
| 04:18.29 | JT | it might be |
| 04:18.36 | JT | it will be part of zaptel |
| 04:19.12 | RyanW | is it possible to send an SMS out an E1 ? |
| 04:19.25 | JT | RyanW: not that i know of |
| 04:19.29 | Aces1Up | so i have to compile the zaptel drivers and that will install the ztdummy driveR? |
| 04:19.37 | JT | you'd need telco support, and asterisk support |
| 04:19.52 | JT | you could establish a modem connection, but that's different again |
| 04:20.00 | RyanW | so the best way to send SMS's from * is to use a perl script and web based sms gateway? |
| 04:20.18 | JT | or email |
| 04:20.27 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 04:20.28 | JT | and the perl bit is optional :) |
| 04:21.21 | rue_mohr | IDLERS NEEDED, PLEASE JOIN #garfield, thanks! |
| 04:21.33 | JT | ... |
| 04:21.50 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:23.11 | [hC] | why would you want idlers? |
| 04:23.48 | rue_mohr | cause I have to get some people out :) its a twisted social experiment gone wrong |
| 04:24.10 | linex | I'm trying linphone |
| 04:28.42 | CpuID2 | lol |
| 04:29.17 | CpuID2 | er ok... :) |
| 04:41.32 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 04:43.32 | JT | rue_mohr: winning friends and influencing people i see |
| 04:43.56 | rue_mohr | nono |
| 04:44.14 | rue_mohr | I need 30 bodies in the channel for a few hours |
| 04:44.24 | rue_mohr | seeif it makes some people leave |
| 04:44.28 | JT | that doesn't really make spamming ok |
| 04:50.21 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 04:50.48 | kingsob | im reading the book asterisk: the future of telephone, and its refering to a package called "asterisk-sounds", but i cant seem to find it, has it been removed? |
| 04:51.43 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 04:53.48 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 04:54.30 | rue_mohr | appologies |
| 04:54.33 | *** part/#asterisk rue_mohr (n=rue@h24-207-90-24.cst.dccnet.com) |
| 04:54.56 | [TK]D-Fender | kingsob, http://www.asterisk.org/downloads |
| 04:55.22 | [TK]D-Fender | kingsob, First step : get off ass and check for self : check! |
| 04:55.52 | [TK]D-Fender | kingsob, In 1.4 the included more in the base. |
| 04:56.01 | *** join/#asterisk smultron (n=lauren@cpe-72-179-47-78.austin.res.rr.com) |
| 04:56.18 | kingsob | dood, calm down.. i gaurantee you i have looked |
| 04:56.25 | kingsob | and its not where they claim it should be |
| 04:57.25 | kingsob | or even called the same thing |
| 04:58.46 | [TK]D-Fender | kingsob, always check asterisk.org, and be ready for things to change and accept that the layout describes what is being distributed, and with a little IQ you can answer your own questions. Life goes on and you book is not rewriting itself in front of your eyes |
| 05:01.15 | kingsob | ur my hero |
| 05:01.28 | fujin | anyone have any experience with heartbeat and asterisk? |
| 05:02.01 | *** join/#asterisk remmo (n=junk@203.62.147.3) |
| 05:03.22 | fujin | not the answer I was hoping for :( |
| 05:05.24 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 05:07.04 | OloBola | I am unable to dial out to my voip provider since updating asterisk |
| 05:13.53 | tzafrir_laptop | OloBola, provide more information here, and/or on bugs.digium.com |
| 05:14.31 | tzafrir_laptop | e.g: from what version to what version? what error do you get? |
| 05:16.30 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:18.24 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 05:18.45 | OloBola | hmm.. not sure what is going on. Does voipjet have a network status page? |
| 05:20.49 | rob0 | First I'd go back and check release notes to see if you did the upgrade right. |
| 05:21.37 | OloBola | incoming is fine, the only thing I can't do is dial out to my provider |
| 05:22.11 | rob0 | "sip set debug" can help |
| 05:22.30 | rob0 | "sip set debug off" can help restore your sanity later |
| 05:23.00 | OloBola | thank you |
| 05:26.14 | OloBola | IAX2/voipjet-3 is circuit-busy |
| 05:26.19 | OloBola | whoops |
| 05:26.51 | OloBola | sorry anonymous voip provider |
| 05:29.49 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 05:30.48 | kiscokid | voipjet never works on the free server |
| 05:31.29 | OloBola | I tried west. and east. It might be my account balance. |
| 05:32.33 | OloBola | 19.60, 40 cents short! |
| 05:32.47 | OloBola | let's see... |
| 05:34.10 | rob0 | Oh, um, "sip set debug" won't help with an IAX2 provider. :) |
| 05:34.48 | OloBola | yep |
| 05:35.04 | OloBola | so that was it, my account balance was too low. |
| 05:37.14 | linex | Setting up asterisk is relatively easy. Now how do I make money out of it ? |
| 05:41.56 | *** join/#asterisk Krooks (n=Blahme@124.82.39.240) |
| 05:49.48 | OloBola | I can paypal you to install lumenvox on my machine (one way to make money from *) |
| 05:50.10 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
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| 06:03.35 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 06:14.03 | kingsob | has anyone had any experince with http://vonage-asterisk.com/ ? |
| 06:21.01 | Strom_M | kingsob: it looks questionable at best, and a scam at worst |
| 06:21.12 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.91) |
| 06:21.14 | Strom_M | and anyway, why would you want to use vonage? their policies are draconian |
| 06:22.10 | Strom_M | and those prices are ridiculous |
| 06:22.42 | Strom_M | their cheapest plan is, at minimum, 3c per minute; you can get per-minute rates in the 1.5-2c range with no minimum commitment |
| 06:27.24 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 06:32.04 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 06:34.31 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 06:35.07 | Aces1Up | when compiling the zaptel drivers, do i compile and install after the normal asterisk installation? |
| 06:35.17 | Strom_M | before |
| 06:35.29 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 06:35.38 | Strom_M | always compile zaptel, then libpri, and finally asterisk |
| 06:36.14 | Aces1Up | ok doke. |
| 06:36.41 | *** join/#asterisk Krooks (n=Blahme@124.82.39.48) |
| 06:36.44 | Aces1Up | strom one last questions, do i have to modify the current zaptel drivers 1.4 makefile to include the ztdummy drivers on compiling? |
| 06:38.47 | Strom_M | .....just make sure it's selected when you do make menuselect |
| 06:40.29 | linex | Anyone setup hylafax before ? |
| 06:41.13 | Aces1Up | strom to build zaptel drivers i need to creat symbolic link to my linux sources? |
| 06:42.15 | kingsob | im trying to get and incoming call from FWD to work, but i dont seem to get receiving the call.. I have forwarded port 4569 UDP and TCP to my asterisk server, are there any other ports i need to forward? |
| 06:42.51 | Strom_M | kingsob: did you see my response to your earlier question? |
| 06:43.08 | kingsob | no i didnt |
| 06:43.18 | Strom_M | Aces1Up: the kernel headers for your currently running kernel must be available |
| 06:43.23 | Strom_M | kingsob: scroll up and read |
| 06:43.28 | kingsob | im in canada, and there arent too many options.. |
| 06:43.32 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 06:43.50 | Strom_M | kingsob: there are still better options than that |
| 06:44.04 | kingsob | i guess ill just got with Unlimitel then :) |
| 06:44.13 | kingsob | i've heard lost of good things about them |
| 06:45.12 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 06:45.58 | kingsob | i think u might be right about a scam lol -> The security certificate presented by this website was issued for a different website's address. |
| 06:46.01 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58.8.3.120.revip2.asianet.co.th) |
| 06:46.03 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:46.12 | kingsob | from ie lol |
| 06:48.24 | Aces1Up | strom when i extract zaptel the directory should be in a seperate directory of the root of the asterisk source files correct? |
| 06:49.54 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:49.57 | Strom_M | it's a completely separate tarball |
| 06:50.06 | Strom_M | have you never compiled software on linux before? |
| 06:50.53 | Aces1Up | not too often, but if its completely seperate that i just throw it in its own directory right/ |
| 06:52.33 | *** join/#asterisk sergee (i=kvirc@195.94.224.197) |
| 06:55.07 | *** join/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at) |
| 06:55.26 | nicox | hello, |
| 06:56.06 | nicox | do anybody know what the error "!! Got reject for frame 17, retransmitting frame 17 now, updating n_r!" mean? |
| 06:56.15 | kingsob | can someone tell me how to fix this error? http://rafb.net/p/FPYXM035.html |
| 06:56.52 | Strom_M | kingsob: codec issue |
| 06:57.00 | Strom_M | make sure you and the other endpoint support the same codecs |
| 06:57.31 | kingsob | oh ok, im just using the FWD "Call Me" test thing |
| 06:59.46 | kingsob | woohoot, she works nowq |
| 07:00.03 | kingsob | looks like someone had the same problem as me --> http://paste.lisp.org/display/13396 |
| 07:01.51 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:05.40 | nicox | do anybody know what the error "!! Got reject for frame 17, retransmitting frame 17 now, updating n_r!" mean? |
| 07:06.00 | JT | it means you should check your zap timing |
| 07:14.08 | *** join/#asterisk redax (n=redax@mail.caracom.hu) |
| 07:14.10 | redax | hi, |
| 07:15.01 | *** join/#asterisk Krooks (n=Blahme@60.52.11.214) |
| 07:15.29 | nicox | its a timing-problem? |
| 07:15.57 | *** join/#asterisk phez72 (n=redcap@xs3.xs4all.nl) |
| 07:16.23 | redax | if trustrpid=yes and a isdn->sip gateway sends Remote-Party-Id: <sip:03@192.168.1.1>;party=called |
| 07:16.27 | nicox | i have 3 E1 to 3 different peers, and the 2 of this E1 are in NT mode, the third gives the timing |
| 07:16.51 | redax | do I have to do exten => 03,1,Dial(...) in my dialplan? |
| 07:17.09 | redax | ie. is remote-party-id is replaces the to: header? |
| 07:17.28 | HaMYaI | hi JT, I am still strugling with the test where I made a cross over E1 to test btw the two ports on my quad card |
| 07:18.31 | JT | nicox: PRIs do not have NT mode, they have Network modes though |
| 07:18.34 | HaMYaI | I tried to pull out the junk in zapata.conf and now this config here http://en.pastebin.ca/601008 |
| 07:18.41 | JT | nicox: how often does the error occur? |
| 07:19.35 | JT | HaMYaI: we told you about 5 times you can only have the "1" timing priority present ONCE, please fix it already |
| 07:21.14 | HaMYaI | JT: sorry I thought I had to change only for span 1 & 2 |
| 07:21.23 | JT | no |
| 07:22.41 | HaMYaI | JT: can I put 3 or 4 as timimg in span 3 & 4? |
| 07:22.54 | HaMYaI | or they can be 0 |
| 07:23.00 | JT | can you count? it would be 2 and 3 |
| 07:23.05 | JT | 1 2 3 |
| 07:23.10 | HaMYaI | err ok |
| 07:23.22 | JT | 0 means give timin to the line |
| 07:23.37 | JT | i pointed you to an article on the wiki already about this |
| 07:23.42 | JT | it really isn't that difficult |
| 07:25.03 | HaMYaI | I thought I needed 0 when signalling is pri_net and 1 when signalling is pri_cpe |
| 07:25.23 | JT | you need 1+ for pri_cpe |
| 07:25.28 | HaMYaI | so I was wrong |
| 07:25.28 | JT | it is a priority |
| 07:25.41 | JT | they cannot all have the same priority for sync |
| 07:26.32 | HaMYaI | I never use more than 1 E1 before so my previous zaptel.conf was very simple |
| 07:26.50 | JT | ok |
| 07:28.37 | HaMYaI | now I have http://en.pastebin.ca/601021 |
| 07:29.14 | HaMYaI | I haven't changed anything for spans 1 & 2 |
| 07:30.08 | JT | ok, so what's happening now? |
| 07:32.06 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:32.22 | HaMYaI | I have this is /proc/zaptel/1 and /proc/zaptel/2 -> http://en.pastebin.ca/601024 |
| 07:32.57 | HaMYaI | but "zap show channels" provides some fine looking channel info |
| 07:34.16 | JT | so is the pri coming up or what? |
| 07:35.35 | HaMYaI | how do I acknowledge that? |
| 07:36.00 | JT | is it appearing in the asterisk cli? |
| 07:37.18 | HaMYaI | yes it doesfrom "zap show status", the alarm is OK for span 1 & 2 |
| 07:37.32 | HaMYaI | and BLU/RED for 3 & 4 |
| 07:38.35 | HaMYaI | I will use the dial() cmd on zap channels to test then |
| 07:40.21 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 07:41.45 | JT | gar |
| 07:41.51 | JT | it's OK, then what's the issue |
| 07:41.54 | JT | yes, try dialling |
| 07:44.04 | HaMYaI | I added a sip user and modify extensions.conf http://en.pastebin.ca/601033 |
| 07:44.36 | HaMYaI | and put the results under TEST RESULTS |
| 07:48.27 | *** join/#asterisk BugKhaM (n=LAMER@ppp-58.8.1.189.revip2.asianet.co.th) |
| 07:48.47 | BugKhaM | sorry JT, I was HaMYaI and got disconnected |
| 07:49.07 | BugKhaM | I may miss some of your messages |
| 07:50.01 | BugKhaM | I placed my test results and modifications to sip.conf and extensions.conf here -> http://en.pastebin.ca/601033 |
| 07:52.02 | JT | BugKhaM: do not enable g.723 |
| 07:52.03 | BugKhaM | anything suspecious? |
| 07:52.09 | JT | and do not enable ilbc unless you installed it |
| 07:52.22 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 07:53.02 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 07:53.21 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 07:53.46 | BugKhaM | ok removed that now, I just saw that show codecs had those |
| 07:55.35 | *** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 07:57.51 | RyanW | Hi, i just upgraded to a new version of asterisk but the astvarlib var in asterisk.conf is no longer honoured. |
| 07:58.00 | BugKhaM | RT: and it works now! |
| 07:58.23 | *** join/#asterisk Swat2 (n=bler@218-215-197-210.people.net.au) |
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| 07:58.49 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:59.27 | RyanW | after upgrading my IVR's no longer work either, it falls through straight away, how do i fix |
| 07:59.54 | Strom_M | RyanW: what version did you upgrade from |
| 08:00.02 | Strom_M | and what version are you using now? |
| 08:00.15 | Swat2 | Are there any how-to's on SLA/BLA ? |
| 08:00.24 | BugKhaM | RT: so what's the problem with having those codecs allowed? I thought it will just select other codecs when one isn't available |
| 08:00.51 | RyanW | i'm now running 1.4.5 and previously i was running 1.4.something |
| 08:01.05 | Strom_M | RyanW: if it's within 1.4 release there shouldn't be much difference |
| 08:01.30 | JT | RT? |
| 08:01.33 | JT | who is RT? |
| 08:01.39 | BugKhaM | ohh sorry |
| 08:01.53 | BugKhaM | I was exciting |
| 08:02.00 | BugKhaM | thanks JT |
| 08:02.04 | JT | no probs |
| 08:02.12 | BugKhaM | for your great help |
| 08:02.19 | JT | seems there is a use for old tormenda 2 cards ;) |
| 08:02.27 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 08:02.28 | BugKhaM | what about the errors in /proc/zaptel/* |
| 08:03.08 | BugKhaM | JT: yeah I bought it three months ago without knowing it would work |
| 08:03.31 | vlt | Hello. I'm using "overlapdial=yes" in zapata.conf on some channels. Can I modify the timeout value when dialling begins after no more digit is pressed? Can I set the "#" key as a delimiter (for the impatient callers)? |
| 08:03.43 | BugKhaM | JT: I am using E100P amd TE110P too |
| 08:05.16 | JT | BugKhaM: almost like a distory museum ;) |
| 08:07.30 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58.8.3.120.revip2.asianet.co.th) |
| 08:07.44 | RyanW | Its now executing hangup instead of exten => t in my ivr context, any ideas? |
| 08:08.49 | HaMYaI | err disconnected again |
| 08:09.38 | *** join/#asterisk dharrigan (n=dharriga@dsl-217-155-228-129.zen.co.uk) |
| 08:11.11 | *** join/#asterisk alomelo (n=wyginwys@85.105.204.40) |
| 08:12.12 | *** join/#asterisk Bu5hm4nn (n=Miranda@pD9E0CFB6.dip0.t-ipconnect.de) |
| 08:13.45 | Aces1Up | jt, ok i it looks like everything compiled ok on the zaptel drivers, is there a way to check to see if the ztdummy driver is correctly loaded? |
| 08:13.57 | RyanW | ls /dev/zap ? |
| 08:14.49 | Aces1Up | no such file or directory :( |
| 08:15.32 | *** join/#asterisk vgster (n=vgster@host81-149-46-66.in-addr.btopenworld.com) |
| 08:16.23 | Aces1Up | do i have to load the module manually? |
| 08:17.23 | Aces1Up | ok loaded the zap drivers via modprobe zaptel |
| 08:17.34 | Aces1Up | now i have /dev/zap |
| 08:17.37 | HaMYaI | JT: should the span 2 (pri_net) provide standard ISDN protocol? as I am planing to use it to connect to dialogic D/300SC E1 card |
| 08:17.53 | Aces1Up | is there anything else i should check to make sure ztdummy is correctly working? |
| 08:18.12 | JT | HaMYaI: yes, etsi if you have it set to euroisdn |
| 08:18.19 | *** part/#asterisk Andri[DK] (n=andri@hydrogen.calidris.com) |
| 08:18.19 | JT | as long as the dialogic isn't on asterisk |
| 08:19.54 | HaMYaI | JT: nope it's using another solution for it, here's the list of supported protocol http://resource.intel.com/telecom/support/releases/unix51/linux51fp2/Onldoc/htmlfiles/install/r4conf23.htm |
| 08:20.30 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 08:21.45 | HaMYaI | JT: there are a few there namely "CTR4" , "NE1" that I might try |
| 08:22.57 | JT | give NE1 a go i guess |
| 08:23.04 | HaMYaI | JT: both were described as "EURO-ISDN ETSI300-102" |
| 08:23.28 | HaMYaI | JT: ok, will tell you the results =) |
| 08:23.30 | JT | correct |
| 08:27.03 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:28.46 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 08:29.57 | *** join/#asterisk xermesx (n=ermsewrk@217.220.121.62) |
| 08:40.14 | *** join/#asterisk voipman (i=distorti@junipero.3sheep.com) |
| 08:41.05 | voipman | 2092 active channels |
| 08:41.06 | voipman | 1103 active calls |
| 08:41.09 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:41.13 | voipman | not bad asterisk ;) |
| 08:41.36 | zoa | how did you do that ? |
| 08:41.50 | voipman | dont ask me, i just route em |
| 08:42.11 | voipman | mostly simple routing tho, all passthrough rtp |
| 08:42.19 | voipman | dual xeon 3gig |
| 08:42.46 | zoa | 1103 shouldnt be possible with quality audio if you ask me |
| 08:43.04 | zoa | how much packets per second is it handling ? |
| 08:43.25 | voipman | ur right, i see quality problems on this box at about 12-1300 calls |
| 08:43.43 | *** join/#asterisk af_ (n=getsmart@81-174-8-1.dynamic.ngi.it) |
| 08:44.15 | voipman | hmm pps, havent checked- i usually keep close eye on cpu tho- which > ~18% usually means quality ichues |
| 08:46.08 | *** join/#asterisk pogo123 (n=hh@buero-gw.dortmund.loca.net) |
| 08:46.09 | voipman | what do you use to check pps? |
| 08:46.17 | zoa | there is a tool called rated or so |
| 08:47.17 | zoa | but i cant find the link |
| 08:48.06 | zoa | its a console tool |
| 08:48.11 | zoa | its called rate or rated |
| 08:48.44 | voipman | <PROTECTED> |
| 08:48.51 | voipman | err * console |
| 08:49.13 | zoa | linux |
| 08:50.32 | voipman | meh, i guess the smart thing to do would be to check the router and not load extra ish on the busy machine, but for testing i'd love to find a nice pps tool |
| 08:53.24 | voipman | no findy w/centos yum, I would have expected ethtool to have an option to check- wonder if tcpdump has an option.. |
| 08:53.42 | *** join/#asterisk purplet (n=purplet@010.041.dsl.concepts.nl) |
| 08:55.34 | voipman | err found rate on voip-info, thanks zoa |
| 08:59.57 | OloBola | so is lumenvox a pain to install? |
| 09:02.02 | Aces1Up | i'm running my asterisk box on a vmware image of fedora 7, and getting the following error, is this a bad compile type error or connection / NAT issue? Sip_Xmit of 0xaddress f8 (len 492) to ip:5060 returned -2 Bad file descriptor |
| 09:04.16 | *** join/#asterisk Ebola (n=Ebola@host86-136-134-202.range86-136.btcentralplus.com) |
| 09:05.21 | Aces1Up | my setup is as follows... NAT Router --> Computer (192.168.1.200) --> Vmware Box (192.168.206.128) |
| 09:08.06 | voipman | Aces1Up: u might have better luck googling on that one bud: setup is quite unique. or at least try back in the am (US) when more digium peeps are on this chan |
| 09:10.06 | Zeeek | ~seen wasim |
| 09:10.25 | jbot | wasim <n=wasim@pdpc/supporter/active/wasim> was last seen on IRC in channel #asterisk, 189d 2h 34m 28s ago, saying: 'hmm .. poor england'. |
| 09:15.53 | Aces1Up | ~seen voipmasta |
| 09:15.55 | jbot | voipmasta <n=dopeshow@201.139.156.12.cable.dyn.cableonline.com.mx> was last seen on IRC in channel #asterisk, 15d 23h 42m 6s ago, saying: 'Well, gotta go, see you later'. |
| 09:16.36 | Zeeek | ~seen YerMutha |
| 09:16.37 | jbot | i haven't seen 'yermutha', Zeeek |
| 09:18.14 | krdian_ | ~seen krdian_ |
| 09:18.15 | jbot | krdian_ is currently on #asterisk (23h 32m 25s). Has said a total of 2 messages. Is idling for 1s, last said: '~seen krdian_'. |
| 09:18.17 | Zeeek | OloBola what do you want to install? |
| 09:18.36 | krdian_ | hi |
| 09:18.55 | *** join/#asterisk xermesx (n=ermsewrk@217.220.121.62) |
| 09:21.12 | *** join/#asterisk Andri[DK] (n=andri@hydrogen.calidris.com) |
| 09:22.07 | Andri[DK] | I've got a question; When dialing in through a SIP extension Playback(tt-monkeys) gives me nothing while the echo test works perfectly, as well as directing to other SIP extensions. Any ideas ? |
| 09:22.25 | creativx | does tt-monkeys exist |
| 09:22.27 | Zeeek | did you answer the call? |
| 09:23.08 | creativx | " Otherwise, unless 'noanswer' is specified, the channel will |
| 09:23.08 | creativx | be answered before the sound is played. Not all channels support playing |
| 09:23.08 | creativx | messages while still on hook. |
| 09:23.08 | creativx | " |
| 09:23.36 | JT | Andri[DK]: you must Answer first. |
| 09:23.36 | Uatec | Hi, i have loads of different extensions |
| 09:23.54 | Uatec | how can i make it so that after doing all of the instructions for any particular extension, it will call my final extension (hangup) |
| 09:23.55 | Uatec | ? |
| 09:24.13 | Andri[DK] | I answer the call before playing the monkeys, and yes the file exists. I'm running full verbose with SIP debug as well |
| 09:24.50 | Andri[DK] | Running v1.4.6 of Asterisk |
| 09:25.05 | JT | Andri[DK]: then what's the error? |
| 09:26.28 | Andri[DK] | no error, it reports the channel as up |
| 09:26.38 | Zeeek | but you don't hear anything |
| 09:26.39 | Andri[DK] | I just can't hear anything on the other end |
| 09:26.48 | JT | so it all seems to execute ok on console |
| 09:26.50 | Andri[DK] | yup |
| 09:26.54 | JT | Andri[DK]: add Wait(3) after Answer. |
| 09:26.58 | Zeeek | and the echo test works |
| 09:27.01 | JT | extensions reload |
| 09:27.03 | JT | and try again |
| 09:27.44 | OloBola | Zeeek: lumenvox speech starter kit |
| 09:27.55 | Zeeek | it takes a few minutes to install |
| 09:28.15 | Zeeek | sign up for asterisk speech rec mailing list |
| 09:28.30 | Zeeek | it's so qiet all questions are answered right away |
| 09:28.52 | OloBola | yum? |
| 09:28.52 | Andri[DK] | JT: Executing: Answer, Wait, Playback... Still can't hear anything |
| 09:29.04 | JT | Wait(3)? |
| 09:29.09 | Andri[DK] | yes |
| 09:29.24 | JT | ok, pastebin it |
| 09:29.27 | Zeeek | try background and see if you hear that |
| 09:29.28 | Andri[DK] | roger |
| 09:30.36 | Andri[DK] | http://pastebin.co.uk/18429 |
| 09:30.54 | *** join/#asterisk proti (i=seb@calypso.frankengul.org) |
| 09:31.08 | proti | hi |
| 09:32.01 | JT | brb |
| 09:33.59 | proti | A sip show registry shows always nothing despite having 2 client registered. |
| 09:34.02 | proti | Any hint why ? |
| 09:34.38 | Zeeek | because they're not SIP? |
| 09:34.48 | proti | Yes they are. |
| 09:34.58 | proti | But I'm sure I'm missing something. |
| 09:35.07 | Zeeek | do you see them in show peers? |
| 09:35.31 | Andri[DK] | sip show users ? |
| 09:35.32 | proti | yes |
| 09:35.38 | proti | user1/user1 195.115.23.39 D 5070 Unmonitored |
| 09:36.12 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 09:37.07 | proti | The port has changed. And match the one of the client. |
| 09:37.42 | proti | The show users is like that: |
| 09:37.44 | proti | server1 server1 incoming No RFC3581 |
| 09:38.09 | Zeeek | exit |
| 09:38.15 | Andri[DK] | ps ax |
| 09:38.20 | proti | ? |
| 09:38.39 | af_ | there are iax2 videphone software or hardware? |
| 09:38.42 | Uatec | Zeeek, wrong window? |
| 09:38.48 | Zeeek | ye |
| 09:38.48 | Andri[DK] | I'm sorry proti, but I'm not that familiar with the new version. |
| 09:38.58 | Zeeek | I have nothing in registry |
| 09:38.58 | proti | Me neither. |
| 09:39.24 | Andri[DK] | my registry is empty as well, I just don't see it as a problem |
| 09:40.03 | proti | I got a server1 which receives incoming calls. And user1 that sends invite. |
| 09:40.27 | proti | But When I send an invite to server1, asterisk replies with 404. |
| 09:40.36 | Zeeek | yeah it's funny but I have nothing in registry on the 1.4 box |
| 09:40.49 | proti | Work in progress ? |
| 09:40.58 | Andri[DK] | proti: you sure that the incoming extension isn't just missing from extensions.conf ? |
| 09:41.09 | proti | I'll check |
| 09:41.40 | Zeeek | on 1.2 box, the registry shows what the pbx is registered to, not users who are registered |
| 09:44.55 | proti | Ok, it's the reverse I was thinking of. |
| 09:46.16 | stoffell | any idea how I can figure out wich password is being transmitted during a sip register? I'm using a Kirk phone and always get sip unauthorized.. ( http://paste.debian.net/31887 ) |
| 09:47.03 | Andri[DK] | ethereal? |
| 09:47.23 | stoffell | Andri[DK], yeah, tried that, but I can't find the password in real text... |
| 09:47.43 | Andri[DK] | stoffell: actually your Call-ID looks like an md5sum |
| 09:47.59 | proti | normal, it's a digest. |
| 09:48.11 | Andri[DK] | then I've got nothing |
| 09:48.16 | Zeeek | there's an option to send clear passwords IIRC |
| 09:48.53 | stoffell | hm, the problem is the phone that tries to register is using a Kirk 600v3 gateway, I configured it and all, but can't get the authorization okay.. weird.. |
| 09:49.08 | stoffell | so my first guess was it was sending a wrong password (clearly.. unauthorized :-) ) |
| 09:49.12 | Andri[DK] | I'm still stuck on why Playback doesn't work on my extensions in 1.4 while Echo works perfectly |
| 09:50.06 | Zeeek | auth=plaintext |
| 09:50.10 | proti | stoffell: You can try to build the digest by hand. |
| 09:50.26 | JT | Andri[DK]: what are you connecting with? |
| 09:50.30 | Zeeek | that may be iax only |
| 09:50.46 | stoffell | Zeeek, will look into it |
| 09:50.52 | stoffell | proti, oh, how could i do that? |
| 09:51.06 | proti | md5sum and the rdc http-digest. |
| 09:51.10 | Uatec | hey, i just saw: Asterisk Business Edition is currently sold with a 40-call limit and can support up to 240 calls in increments of 40 calls |
| 09:51.12 | proti | rfc* |
| 09:51.23 | Uatec | is there a call limit on open source asterisk? |
| 09:51.30 | stoffell | okay, i see |
| 09:51.30 | snuff-work | nope |
| 09:51.31 | JT | no |
| 09:51.42 | JT | except it often doesn't work very well after a few hundred |
| 09:51.48 | Uatec | none at all? just a performance limit? |
| 09:51.49 | Uatec | ah.. |
| 09:51.57 | J4k3 | you're paying for support (the bigger the system, the more hassles there are) |
| 09:51.58 | Andri[DK] | JT: Tried both SIP directly to the PBX and with a cellphone through my provider>sip>pbx |
| 09:51.59 | snuff-work | look at good old voip-info |
| 09:52.04 | Uatec | so after 200 calls what would you do? get multiple asterisk servers? |
| 09:52.11 | snuff-work | that's it uatec |
| 09:52.14 | Uatec | ok |
| 09:52.23 | JT | Andri[DK]: pastebin.ca your sip.conf minus passwords |
| 09:52.27 | Uatec | that's actually completely irrelevant to my question |
| 09:52.31 | snuff-work | guess if u wanted to play it safe.. 150:) |
| 09:52.41 | JT | Uatec: or use a more robust program |
| 09:52.47 | J4k3 | it all depends on the hardware really |
| 09:53.17 | Uatec | i just want to know what spec i would need on a asterisk server which has 20 users, and 8 isdn channels (on a b410p) |
| 09:53.21 | Uatec | jt, such as what? |
| 09:54.17 | Andri[DK] | JT: http://pastebin.ca/601131 |
| 09:54.20 | snuff-work | well i'd be thinking if u just wanted to pass sip traffic dont do all the complex dialplan stuff.. go like openser/ser for few 1000's calls |
| 09:55.14 | Uatec | heh, i'm not looking at that many |
| 09:55.15 | Andri[DK] | JT: Does Playback use Alsa in anyway ? |
| 09:55.17 | Uatec | i wa sjust interested |
| 09:55.21 | JT | nup |
| 09:55.22 | Uatec | i only want 20 users |
| 09:55.48 | Uatec | and at most 8 concurrent external (isdn) calls and maybe some internal (sip) calls too |
| 09:56.07 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 09:56.41 | dharrigan | Hey there. Question, has asterisk the ability to recognise callerid coming in and simple save the incoming call into a mailbox of that number's number? |
| 09:57.21 | creativx | dharrigan: yes. |
| 09:57.22 | Zeeek | it's available in voicemail |
| 09:57.36 | dharrigan | so, if a call came in for 343134123 then it could create a mailbox called 343134123 |
| 09:57.39 | *** join/#asterisk fyrestrtr (n=burhan@pdpc/supporter/student/fyrestrtr) |
| 09:57.48 | dharrigan | s/for/from/ |
| 09:58.12 | fyrestrtr | is there a *hardware* appliance that has asterisk + supported hardware cards available? |
| 09:59.11 | Zeeek | Digium has one |
| 10:00.11 | Zeeek | trixbox has one too |
| 10:00.59 | Andri[DK] | JT: The only thing I can think of is that it's failing to convert the gsm files to alaw |
| 10:02.08 | *** join/#asterisk Fl1p (n=david@port-83-236-208-174.static.qsc.de) |
| 10:02.27 | dharrigan | Does anyone know of anyone offering virtual private numbers -> voip? |
| 10:02.51 | J4k3 | doesn't FWD do that? |
| 10:03.01 | Zeeek | ipkall |
| 10:03.20 | Zeeek | ideasip.com |
| 10:05.05 | dharrigan | thanks al |
| 10:05.09 | dharrigan | thanks all |
| 10:05.24 | JT | do you mean DIDs? |
| 10:06.18 | JT | Andri[DK]: dunno |
| 10:06.25 | JT | Andri[DK]: try some other format then |
| 10:06.56 | fyrestrtr | what is this > "One 4- port FXS module & two 4-port FXO modules" ?? |
| 10:09.12 | _E-bola | Do anybody know a loadbalancing solution for linux that has a nice usercontrolable webinterface? |
| 10:09.42 | Zeeek | fyrestrtr what are you looking for? |
| 10:10.05 | JT | _E-bola: dreaming? |
| 10:10.32 | _E-bola | JT? |
| 10:10.42 | Zeeek | _E-bola ? |
| 10:10.46 | J4k3 | _E-bola: if you find one, plz, by all means, let everyone know :D |
| 10:10.54 | _E-bola | hmm k |
| 10:11.33 | JT | Ebola: real administrators do not use web interfaces |
| 10:11.46 | _E-bola | JT: its not for me, its for my linux hating windows co-workers |
| 10:11.48 | Ebola | Sue don't |
| 10:11.53 | MrWup | what cool things can i do with ISDN? |
| 10:11.54 | _E-bola | to be able to change data flow if necesary |
| 10:12.08 | MrWup | that will give a really good impression to clients |
| 10:12.08 | fyrestrtr | Zeeek: I have an application that can connect to asterisk (or other SIP-compatible voip solutions) and I am trying to test it out by purchasing a hardware appliance that already comes preconfigured with asterisk. I just want to have internal voip sets, and connection to an external pstn for dialing out. |
| 10:12.33 | JT | MrWup: accurate billing? |
| 10:12.40 | voipman | _E-bola: check out PEN. No native web inteface but simple enough to construct and powerful enough for whateva |
| 10:12.50 | MrWup | no need to bill them |
| 10:13.05 | voipman | if sip only ser/openser is a great solution |
| 10:13.09 | _E-bola | voipman: yep ive looked at pen |
| 10:13.10 | Zeeek | fyrestrtr you won't need FXS or FXO then. They are for connecting local analog phones and analog lines respectively |
| 10:13.12 | *** join/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy) |
| 10:13.14 | _E-bola | if only it had a web interface |
| 10:13.20 | _E-bola | http://www.vmware.com/vmtn/appliances/directory/117 looks somewhat useable |
| 10:13.32 | Zeeek | fyrestrtr I'd look at trixbox applicane |
| 10:13.33 | JT | MrWup: DIDs |
| 10:13.36 | fyrestrtr | Zeeek: what do you mean 'analog lines' ?? |
| 10:13.46 | Zeeek | prolly the fastest way to go |
| 10:13.47 | fyrestrtr | Zeeek: is that the same as the line that I get from my telco? |
| 10:13.54 | Zeeek | analog lines as in PSTN |
| 10:13.56 | voipman | _E-bola: odesk.com $10/hr for capable web programers |
| 10:13.58 | MrWup | didnt get that installed with my lines, no need for it |
| 10:13.59 | JT | if it's analogue |
| 10:14.00 | Zeeek | regular phone line |
| 10:14.05 | JT | no, analogue as in POTS |
| 10:14.09 | JT | pstn can be either |
| 10:14.18 | Zeeek | correct, you anal sob :) |
| 10:14.28 | fyrestrtr | yes, right. So how do I dial out without having the appliance connected to the POTS? |
| 10:14.36 | Zeeek | via SIP? |
| 10:14.55 | fyrestrtr | so then I need another provider to give me an outbound SIP number? |
| 10:15.12 | Zeeek | fyrestrtr please read this: http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 10:15.21 | Zeeek | it's a good article that will answer ALL those questions |
| 10:16.09 | Zeeek | Title: "VoIP and POTS Integration with Asterisk" |
| 10:17.03 | *** part/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy) |
| 10:17.44 | fyrestrtr | hrmm. |
| 10:18.14 | fyrestrtr | Okay I get one part of the puzzle, how to connect to the POTS and any 'normal' phones to the system, so that they will rung. |
| 10:18.18 | fyrestrtr | err, ring. |
| 10:19.03 | Zeeek | Why not read the article? |
| 10:19.27 | OloBola | public key not available for //var/cache/yum/LumenVox/packages/LumenVoxLicenseServer-7.5-303.fc5.i386.rpm |
| 10:20.18 | OloBola | trying to install lumenvox using yum |
| 10:20.49 | OloBola | the license server |
| 10:21.04 | kingsob | has exten => s,1,Answer( ) changed in version 1.4? |
| 10:21.09 | kingsob | the s part |
| 10:21.59 | OloBola | any suggestions? |
| 10:22.18 | fyrestrtr | Zeeek: because have of it doesn't apply to me, since there is no such thing as a IPCSP in my location. |
| 10:22.24 | fyrestrtr | s/have/half/ |
| 10:22.44 | fyrestrtr | oh hey .. that's neat :) |
| 10:23.32 | Zeeek | fyrestrtr there's no such thing in mine either. Doesn't stop me from having a lot of services |
| 10:23.51 | Zeeek | in fact, that article is old and I still never heard of that |
| 10:23.54 | *** join/#asterisk Corydon76-home (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 10:23.54 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 10:24.52 | Zeeek | but it does give all the basic concepts |
| 10:25.21 | fyrestrtr | yes, that it does. |
| 10:26.41 | OloBola | hey! exciting stuff here >>> public key not available for //var/cache/yum/LumenVox/packages/LumenVoxLicenseServer-7.5-303.fc5.i386.rpm |
| 10:26.58 | fyrestrtr | I don't need the device to make long distance calls, or even accept calls from other SIP providers (externals). But what I do want is to have ... local extensions (this part I understood) and ability to dial out on the POTS (this part I understood). What is confusing me is -- do I need another provider if I *just* want these two things to work? I don't think I do. |
| 10:28.37 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:29.02 | OloBola | all you'll need is a phone line and a card, as far as I can tell. |
| 10:29.10 | fyrestrtr | yeah exactly. |
| 10:29.40 | fyrestrtr | I need two or three voip phones that hook into the asterisk appliance, and then the regular lines that I have, hook into the asterisk appliance. |
| 10:30.12 | fyrestrtr | once I get this going *then* I can test the custom application that will hook into asterisk. |
| 10:30.52 | fyrestrtr | asterisk has STUN? |
| 10:31.36 | nicox | what does that mean? "XXX Missing handling for mandatory IE 8 (cs0, Cause) XXX" |
| 10:31.37 | nicox | ? |
| 10:32.12 | fyrestrtr | http://www.digium.com/en/wheretobuy/digiumdirect/productview.php?product_code=AADKADMIN -- this page doesn't exist :( |
| 10:35.03 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 10:38.17 | kingsob | can someone help me out with this?? -> http://rafb.net/p/iepDKe49.html |
| 10:39.47 | ManxPower | kingsob: look at the goto. |
| 10:40.00 | ManxPower | Goto(context,extension,priority) |
| 10:40.23 | ManxPower | exten => s is NOT a wildcard |
| 10:40.30 | kingsob | shouldnt s pick up everyhting? |
| 10:40.32 | creativx | no |
| 10:41.27 | kingsob | oh, i guess i have misunderstood its meaning then |
| 10:41.41 | kingsob | i thought it was for when you didnt know the extension |
| 10:42.11 | ManxPower | kingsob: No, it is for FXO ports where the destination number is not known. It is also used in macros |
| 10:43.03 | ManxPower | kingsob: you didn't read the book. |
| 10:44.00 | Uatec | WTF? I have two asterisk processes running that look like "asterisk -vvvg -c" |
| 10:44.07 | Uatec | why would i have two? |
| 10:44.12 | Uatec | surely i don't need more than one? |
| 10:45.04 | ManxPower | Uatec: How do you know they are not two THREADS |
| 10:46.01 | Uatec | well, they have different PIDs, which leads me to believe they are processes |
| 10:46.09 | ManxPower | As I'm sure you know from reading this question in the mailing list archives -- the "ps" program can show all threads of a process as just one process or it can show each thread as it's own thread |
| 10:46.41 | kingsob | im on page 103 lol |
| 10:46.47 | *** join/#asterisk henryv2 (n=henry_vo@82-39-113-10.cable.ubr03.newy.blueyonder.co.uk) |
| 10:46.49 | kingsob | where it explains s |
| 10:46.49 | Uatec | also one is using 99% cpu and one is using 0.5% |
| 10:46.53 | Uatec | that can't be right |
| 10:47.00 | Uatec | at the moment, asterisk isn't even doing anything |
| 10:47.02 | kingsob | and none of these examples work |
| 10:47.05 | ManxPower | Uatec: Are you on *BSD? |
| 10:47.11 | Uatec | no |
| 10:47.13 | Uatec | pound key linux |
| 10:47.17 | Uatec | (rpath) |
| 10:47.28 | ManxPower | kingsob: What verison of Asterisk are you using? |
| 10:47.38 | kingsob | 1.4 |
| 10:47.50 | ManxPower | And what verison was The Book written for? |
| 10:47.53 | kingsob | 1.2 |
| 10:48.08 | ManxPower | And you are suprized that none of the examples work? |
| 10:48.12 | kingsob | yes |
| 10:48.14 | kingsob | lol |
| 10:48.28 | ManxPower | I assume you read UPGRADE.txt to see what things changed in 1.4? |
| 10:49.32 | kingsob | not really read |
| 10:49.38 | kingsob | looked at for 10 sevs |
| 10:49.50 | henryv2 | does anyone know how using Asterisk and which VoIP phones could be used to show which park extension a caller has been parked on on the VoIP phone screen? I know it is announced when the call is parked but as a reminder for a busy office! |
| 10:50.03 | ManxPower | It is the key to translating the 1.2 examples into 1.4 |
| 10:50.18 | fyrestrtr | does the asterisk appliance provide PoE? |
| 10:50.22 | kingsob | i shoudla gone with 1.2 to start out with |
| 10:50.31 | kingsob | its hard to try translate soemthing you dont understand yet |
| 10:50.40 | ManxPower | Exactly |
| 10:50.55 | creativx | the recommended way to upgrade is to go from 1.2 to 1.4 on all production servers, and then wait and see what breaks |
| 10:51.02 | Uatec | fyrestrtr, probably not. you would need a switch that provides PoE, otherwise the power would not transmit through your normal switch |
| 10:51.15 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:51.20 | fyrestrtr | ah, thanks :) |
| 10:51.30 | ManxPower | henryv2: That is a feature of KEY SYSTEMS, not PBXs. |
| 10:51.42 | ManxPower | fyrestrtr: Call Digium to be sure. |
| 10:52.01 | ManxPower | Also remember that Asterisk Appliance does not support T-1/E-1. |
| 10:52.05 | ManxPower | It only supports analog |
| 10:52.19 | Zeeek | or read the digium site |
| 10:52.59 | fyrestrtr | its not mentioned there if it does or not, and when I tried to get pricing on it, the page doesn't display. The trixbox appliance won't be out till "summer '07" and in the Middle East, it was summer two months ago. |
| 10:53.01 | ManxPower | Zeeek: That would be too logical |
| 10:53.15 | creativx | henryv2: do what real men does.. write a fancy GUI and wrap it around the asterisk manager interface |
| 10:53.16 | Zeeek | gee, how did i find this? http://www.digium.com/en/products/hardware/asteriskappliance.php |
| 10:55.01 | kingsob | hmm i think i figured out hwy its not working!!! enter-ext-of-person doesnt exist anymore! |
| 10:55.41 | henryv2 | creativx: there are already GUIs out there which can achieve a similar goal but I was hoping to keep it simple rather than getting each member of staff to keep a GUI on their desktop |
| 10:56.05 | creativx | henryv2: we took the different approach.. we are hiding the phones from the staff, and letting them get all the juicy details onscreen instead |
| 10:56.23 | henryv2 | ManxPower: I'm not really talking about the key system style of assinging a physical line to a caller, but a virtual extension where the call is now parked similar to the way the Avaya Index works |
| 10:57.09 | henryv2 | just i find that people forget which extension a caller is on when transferring them and there is no way to reference the extension easily once the call is parked |
| 10:57.33 | Zeeek | what ever happened to "JIM! LINE 3!" |
| 10:57.33 | Andri[DK] | creativx: is there any fancy gui for asterisk manager interface? |
| 10:57.42 | Andri[DK] | Can't say I was impressed with the AsteriskGUI |
| 10:58.11 | *** join/#asterisk Bu5hm4nn (n=Miranda@pD9E0CFB6.dip0.t-ipconnect.de) |
| 10:58.31 | ManxPower | henryv2: I believe that Polycoms can do something like what you want. There have been discussions on the mailing lists about this in the past. |
| 10:58.45 | ManxPower | I'm sure some other phones like the SNOM can as well. |
| 10:59.33 | Bu5hm4nn | Hello. does anyone know how to solve this problem: NOTICE[4100]: cryptostub.c:42 stub_ast_key_get: Crypto support not loaded! |
| 10:59.48 | henryv2 | I've got a SNOM300, not sure how I'd achieve it with that phone - I'll check the Polycoms |
| 10:59.53 | Zeeek | you may need to load crypto support |
| 11:00.10 | Bu5hm4nn | Zeeek: how, sir? |
| 11:00.41 | Zeeek | what version? |
| 11:00.45 | ManxPower | Bu5hm4nn: I suspect you do not have the openssl dev package installed for your distro when you built Asterisk |
| 11:00.53 | Zeeek | 1.4 make menuselect |
| 11:01.04 | Zeeek | 1.2 what ManxPower said |
| 11:01.09 | henryv2 | Andri[DK]: Fonality offer a good GUI - http://www.fonality.com/pbxtra_features.html as do Soft Telecom: http://www.soft-telecom.com/vox-office.shtml |
| 11:01.40 | Bu5hm4nn | ok, i have 1.4.6 (last night) ... so i'll try rebuilding with openssl-dev |
| 11:01.52 | Bu5hm4nn | thx |
| 11:02.02 | Zeeek | kerchunk |
| 11:02.50 | Andri[DK] | henryv2: commercial stuff? |
| 11:02.50 | *** join/#asterisk vlt (n=dm@port-87-234-95-227.dynamic.qsc.de) |
| 11:02.58 | henryv2 | Andri[DK]: yep, but if you're providing telecoms equipment for a business the Vox Office is still good value |
| 11:03.02 | henryv2 | really very cheap for what it does |
| 11:03.59 | Andri[DK] | yeah, I'm thinking a frontend would be nice for the company since I'm just here for the summer |
| 11:04.06 | Zeeek | Everyone be sure to watch the season premiere of Asterisk TV Friday at 12:30PM EDT |
| 11:04.08 | creativx | Andri[DK]: i dunno of any particular.. ive made my own and integrated it to our cmr |
| 11:04.10 | creativx | crm |
| 11:04.24 | Andri[DK] | creativx: cool, php or ? |
| 11:04.26 | Zeeek | In fact, I think Kerry from FOnality will be in the conference room |
| 11:04.37 | creativx | Andri[DK]: VB |
| 11:04.47 | Andri[DK] | *shivers* |
| 11:05.13 | creativx | asterisk and vb is a sweet combo |
| 11:05.36 | Andri[DK] | perhaps, vb just always gives me the shivers :P |
| 11:05.42 | Bu5hm4nn | uh, mixing redmond with the gnu world ... |
| 11:05.49 | creativx | it gives me sweet dineros |
| 11:06.04 | creativx | which in turn gives me beer |
| 11:06.10 | creativx | ergo, its a recipe for success. |
| 11:06.16 | Zeeek | and a desire for salty chips? |
| 11:07.21 | creativx | not really, im not much of a chips person |
| 11:07.33 | Andri[DK] | I prefer my dineros to be created with software, as free as beer :P |
| 11:07.37 | Zeeek | you prefer pringle? |
| 11:07.53 | Zeeek | green olives? |
| 11:08.02 | creativx | i prefer beer |
| 11:08.04 | Zeeek | beer nuts? |
| 11:08.06 | creativx | with more beer |
| 11:08.08 | *** join/#asterisk maduxi (n=skdbjf@89.129.170.96) |
| 11:08.16 | Zeeek | you must accompany beer with a solid food |
| 11:08.22 | creativx | yes, beer is a solid food |
| 11:08.24 | creativx | so it solves itself |
| 11:09.01 | Zeeek | beer is good |
| 11:10.05 | creativx | it sure is |
| 11:12.42 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 11:13.02 | *** join/#asterisk saftsack (n=saftsack@pD9E0605E.dip.t-dialin.net) |
| 11:17.33 | *** part/#asterisk maduxi (n=skdbjf@89.129.170.96) |
| 11:20.34 | kingsob | can anyone tell me why my timeout extension is never getting executed? |
| 11:20.34 | kingsob | http://rafb.net/p/GqTEEU52.html |
| 11:20.41 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 11:21.21 | *** join/#asterisk emily_25 (n=skdbjf@89.129.170.96) |
| 11:21.33 | Zeeek | does i work? |
| 11:21.42 | kingsob | yeah |
| 11:21.59 | emily_25 | hi |
| 11:22.02 | Zeeek | then there may be another 't' precluding this one in another context |
| 11:22.09 | Zeeek | Hi emily_25 |
| 11:22.19 | kingsob | thats my entire extensions.conf file |
| 11:22.30 | creativx | autofallthrough? |
| 11:22.41 | Zeeek | not a very interesting one in that respect |
| 11:23.06 | emily_25 | where can i have help with asterisknow? |
| 11:23.21 | kingsob | creativx, yeah |
| 11:23.27 | kaldemar | /topic |
| 11:23.32 | Zeeek | Emily isn't there an IRC channel of that name? |
| 11:23.39 | kaldemar | emily_25: ^ |
| 11:23.44 | emily_25 | yes sory |
| 11:23.44 | creativx | [tk]-fender said something interesting about that yesterday kingsob |
| 11:23.47 | emily_25 | ibe just found it |
| 11:23.59 | Zeeek | you may have to wait for someone to be there |
| 11:24.20 | emily_25 | ok thx |
| 11:24.23 | Zeeek | kingsob you might try setting the digit timeout |
| 11:24.47 | kingsob | how would i do that? |
| 11:24.57 | Zeeek | by googling for asterisk digit timeout |
| 11:24.59 | *** join/#asterisk obnauticus (n=obnautic@c-71-59-162-60.hsd1.wa.comcast.net) |
| 11:25.18 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 11:25.26 | Zeeek | the first ten entries are of interest |
| 11:26.12 | Zeeek | Interestingly, the first site referenced has this line in it: "NOTE:Information about the Asterisk functions could be obtained by typing the show functions command." |
| 11:26.35 | Zeeek | then folows a full tutorial on timeout |
| 11:27.21 | kingsob | ok i added exten => s,2,Set(TIMEOUT(digit)=10) |
| 11:27.25 | kingsob | and same thing :( |
| 11:27.36 | Zeeek | set it to like 1 or 2 just to not waste so much time :) |
| 11:27.59 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 11:28.03 | kingsob | well it hangs up the second its done playing the file anyways |
| 11:28.13 | Zeeek | take a look at show dialplan to see if it shows what you would expect |
| 11:32.20 | kingsob | looks good :( |
| 11:33.02 | Zeeek | what is this, 1.2 ? |
| 11:34.32 | Zeeek | exit |
| 11:34.35 | Zeeek | again? |
| 11:35.18 | kingsob | 1.4 |
| 11:36.43 | kaldemar | you should set response timeout, not digit timeout. |
| 11:36.55 | Zeeek | whoops, that's right |
| 11:37.27 | kingsob | http://rafb.net/p/kCMg9g31.html |
| 11:37.28 | kingsob | o ok |
| 11:37.54 | kingsob | same thing tho :( |
| 11:37.58 | kaldemar | on the other hand someone has written to wiki that you must disable autofallthrough or use WaitExten, otherwise it will fall through immediately. |
| 11:39.57 | kingsob | ohhh i think i just found that --> http://freepbx.org/trac/ticket/1820 |
| 11:40.17 | kaldemar | i'd keep autofallthrough and use WaitExten to have better control over the dialplan. |
| 11:40.32 | emily_25 | i have this queue, but this does only put the call in hold with music, but does not ring the agent http://pastebin.ca/601215 why? |
| 11:41.51 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 11:42.08 | Uatec | what spec hardware would i need for an asterisk server with 20 sip clients and 8 isdn channels? |
| 11:43.00 | kingsob | woot, she works now, thx alot kaldemar, you just saved me from bashing my head against the wall |
| 11:43.07 | creativx | get softer walls |
| 11:43.15 | creativx | after a while it gets pleasant |
| 11:43.16 | creativx | ;) |
| 11:43.27 | Zeeek | so it was autofallthrough? |
| 11:43.36 | kingsob | i just had to add |
| 11:43.42 | kingsob | exten => s,3,WaitExten( ) |
| 11:43.48 | Zeeek | oh |
| 11:44.15 | kingsob | --> http://rafb.net/p/jdmMOV76.html |
| 11:44.28 | kaldemar | emily_25: are your agents configured correctly? |
| 11:45.05 | kingsob | onto the next page, im sure i will be back soon lol |
| 11:45.05 | Zeeek | I'll have to check it on 1.4, on 1.2 it works as expected |
| 11:45.17 | *** join/#asterisk shinao1 (n=shinao1@196.1.179.225) |
| 11:46.02 | creativx | emily_25: show queue 6004 ? |
| 11:47.37 | emily_25 | is asterisknow gui making it, the extensions is all mixed up |
| 11:47.45 | emily_25 | i show u the 7000 |
| 11:47.47 | emily_25 | http://pastebin.ca/601224 |
| 11:47.55 | emily_25 | in users.conf |
| 11:57.40 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 12:01.50 | *** join/#asterisk matsk (n=mk@194.68.102.175) |
| 12:10.09 | *** part/#asterisk DanielHolth (n=dholth@2002:18aa:c06f:1:211:2fff:fed5:f7c7) |
| 12:11.51 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 12:12.00 | jeremy_g | hello girls |
| 12:14.36 | Uatec | don't be silly jeremy_g |
| 12:14.41 | Uatec | there are no girls on the internet |
| 12:14.56 | henryv2 | what about her? |
| 12:15.07 | Uatec | obviously a delusion |
| 12:15.11 | negativeduck | yes there are... but they are all supposedly half naken and wanting me |
| 12:15.17 | negativeduck | atleast according the websites I visit |
| 12:15.26 | negativeduck | s/naken/naked |
| 12:15.46 | kaldemar | http://tolitx.blogs.friendster.com/tatak_tolitz/images/teenager.jpg |
| 12:16.01 | Uatec | ? |
| 12:16.08 | Uatec | sfw? |
| 12:16.14 | creativx | sfw |
| 12:16.28 | creativx | if you work at a fat guy factory |
| 12:16.40 | negativeduck | ofcourse on those same sites I'm also some buff dude with Chiseled abs, instead of my normal fat ass keyboard riding self. |
| 12:16.41 | Uatec | lol |
| 12:16.42 | Uatec | ew |
| 12:18.38 | Bu5hm4nn | hello. i've compiled crypto so that errors gone. now i've got the next one. anyone an idea?[Jul 3 14:14:13] WARNING[479]: channel.c:718 ast_best_codec: Don't know any of 0xe000 formats |
| 12:18.38 | Bu5hm4nn | [Jul 3 14:14:13] ERROR[479]: chan_iax2.c:7467 socket_process: No best format in 0xe000??? |
| 12:18.38 | Bu5hm4nn | [Jul 3 14:14:13] NOTICE[479]: chan_iax2.c:7472 socket_process: Rejected connect attempt from 192.246.69.186, requested/capability 0x4/0xf804 incompatible with our capability 0xe703. |
| 12:21.08 | jeremy_g | Bu5hm4nn:dude,you could me of great help to me. I need libssl 0.9.8 binaries for my debian box. |
| 12:21.24 | jeremy_g | Bu5hm4nn:compiling libssl by hand is a pita |
| 12:21.32 | Uatec | what's wrong with apt? |
| 12:21.37 | Uatec | apt has super cow powers, so i hear tell |
| 12:22.33 | Bu5hm4nn | jeremy_g: i have ubuntu LTS; and there's a package called libssl0.9.8 |
| 12:22.39 | jeremy_g | Uatec:the system already has a little older version of libssl and apt get would just say if u upgrade then you might have to upgrade some other packages to which i dont give a %#$@. But even -f option wont work. |
| 12:22.52 | jeremy_g | Bu5hm4nn:send |
| 12:22.57 | Uatec | why not just let it upload those other packages? |
| 12:23.07 | jeremy_g | Uatec:it wont do that |
| 12:23.17 | Uatec | why not? |
| 12:24.20 | jeremy_g | Uatec:its a complicated situation. i am infact burning asterisk into knoppix which uses debian packaging system |
| 12:24.34 | *** join/#asterisk Greek-Boy (n=grb@196.46.105.176) |
| 12:24.45 | Uatec | :o |
| 12:24.54 | negativeduck | is there a huge reason to not let it update the dist? |
| 12:25.01 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 12:25.04 | Bu5hm4nn | what is capability 0xe703 , or where can i find out? |
| 12:25.11 | jeremy_g | Uatec:so i have this rw cloop image chrooted into and doing apt-get and dpkg stuff which is working except for this ssl thingie and another libc6-dev for which it complained for /usr/include/sys/types.h or sth |
| 12:26.59 | Uatec | umm |
| 12:27.01 | Uatec | i have a penny |
| 12:27.12 | kaldemar | Bu5hm4nn: i found the answer with google in a few seconds. |
| 12:28.04 | Bu5hm4nn | sry, i didn't even try cause i had the irc right here |
| 12:28.34 | kaldemar | search first and if you can't find the answer by yourself, then ask. |
| 12:29.16 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 12:29.25 | Bu5hm4nn | kaldemar: i know. I usualy do so. *shame* |
| 12:33.51 | Bu5hm4nn | jeremy_g: i can send u the ubuntu libssl, but u could also get it directly from the package servers |
| 12:38.43 | *** join/#asterisk guillote_GNU (n=guillote@190.7.23.224) |
| 12:39.44 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:43.22 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 12:43.22 | *** mode/#asterisk [+o anthm] by ChanServ |
| 12:43.42 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 12:48.06 | *** join/#asterisk Phuntom (n=Phuntom@80.233.159.254) |
| 12:48.10 | Phuntom | hi ya |
| 12:49.18 | Phuntom | any alive? |
| 12:49.53 | Phuntom | join #asterisknow |
| 12:50.09 | Phuntom | hm |
| 12:50.15 | *** join/#asterisk xermesx (n=ermsewrk@217.220.121.62) |
| 12:51.48 | JT | eh |
| 13:00.11 | *** join/#asterisk wunderkin (n=wunderki@dslstat-ppp-95.fastq.com) |
| 13:06.02 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 13:11.59 | *** join/#asterisk irule (n=irule@189.164.43.194) |
| 13:12.36 | Andri[DK] | has anyone worked with AsteriskGUI ? |
| 13:12.56 | Andri[DK] | I'm having problems connecting it with my custom extensions, doesn't see them |
| 13:12.58 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:13.54 | *** join/#asterisk el_4_jinete (n=el_4_jin@mail.pulxar.com.co) |
| 13:14.19 | el_4_jinete | Hi all... :) |
| 13:15.38 | Andri[DK] | hi |
| 13:15.42 | Fl1p | what's the easiest way to check if a channel is still available while using the ManagerInterface ? |
| 13:17.04 | Andri[DK] | show channels ? |
| 13:17.19 | creativx | i was wondering if sox was any good for lowering the volume ? |
| 13:17.25 | creativx | my moh files are nazi loud |
| 13:18.10 | rob0 | You play Deutschland Uber Alles for MOH? |
| 13:18.41 | polerin | Dood... that'd be aweseome |
| 13:18.44 | Uatec | hey, in my cdr_custom.conf, how can i make it display the time of a call with the seconds? |
| 13:18.51 | Andri[DK] | creativx: http://www.voip-info.org/wiki/index.php?page=Asterisk+sound+files |
| 13:18.54 | rob0 | it would be pretty funny, yes |
| 13:18.59 | polerin | Can I play Too Drunk to Fuck as MOH? |
| 13:19.02 | Fl1p | Andri[DK] : the show channels doesn't show the full channel ID it's not shown completet. I need the available channels and grep it to redirect it for example.... |
| 13:19.28 | *** join/#asterisk af_ (n=getsmart@81-174-8-1.dynamic.ngi.it) |
| 13:19.37 | Andri[DK] | Fl1p: not sure if there's any good way of showing "available" channels. What kind of channels are you referring to |
| 13:20.09 | polerin | actually you know what would make good MOH? |
| 13:20.14 | polerin | Portishead |
| 13:20.40 | Uatec | GO PORTISHEAD |
| 13:21.10 | Andri[DK] | Coldplay does it for me |
| 13:21.22 | Andri[DK] | Clocks on repeat until they hang up |
| 13:21.25 | Uatec | coldplay teh boring |
| 13:21.41 | *** join/#asterisk dickyjoe (n=richardl@dsl-220-235-101-16.nsw.westnet.com.au) |
| 13:21.58 | Fl1p | www.freeplaymusic.com got a lot license free stuff |
| 13:22.05 | Andri[DK] | Disney soundtracks from the movies might also make people go insane |
| 13:22.26 | Andri[DK] | depends ofcourse what you're going for |
| 13:22.29 | dickyjoe | Hello all |
| 13:22.39 | Uatec | also, all pop music will make people get stupid |
| 13:22.50 | Fl1p | Andri : i want to check if user on SIP haven't hung up the channel, if not i would transfer him to another channel etc.... |
| 13:23.33 | dickyjoe | I'd like to introduce myself if I may... |
| 13:24.17 | dickyjoe | My name is Richard aka dickyjoe and I'm new to asterisk |
| 13:24.52 | *** part/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 13:24.59 | Andri[DK] | welcome dickyjoe |
| 13:25.07 | dickyjoe | I bought myself the book "Asterisk - The future of telephony" and have read it a couple of times |
| 13:25.18 | Andri[DK] | Fl1p: not sure, sorry |
| 13:25.27 | dickyjoe | thanks andri |
| 13:26.03 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:26.03 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:26.07 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 13:27.01 | dickyjoe | I want to ask a techo question about linking an old system to asterisk |
| 13:27.08 | dickyjoe | pabx that is |
| 13:27.56 | Qwell | dickyjoe: fire away |
| 13:28.33 | Andri[DK] | Just describe the old system as well as you can |
| 13:28.48 | ai-a | Where do i place AGI scripts for using within the extention conf file ? |
| 13:29.06 | ai-a | is AGI scripting the best way to write scripts for * ? |
| 13:29.19 | dickyjoe | Our company has a fujitsu pabx with an E1 (ISDN 30) connected to our australian telco (Telstra) |
| 13:29.49 | JT | heh, telstra, ditch them ;) |
| 13:30.19 | dickyjoe | I am not sure if it supports q-sig but wish to interconnect asterisk between to interpret calls |
| 13:30.42 | JT | why do you need q.sig? |
| 13:31.10 | dickyjoe | some calls can then go via sip and others via analouge |
| 13:31.26 | JT | there is no need for Q.SIG |
| 13:31.28 | dickyjoe | JT: I don't actually know what I need |
| 13:31.28 | *** join/#asterisk currach (n=currach@89.16.90.180) |
| 13:31.40 | JT | dickyjoe: i'm in sydney and have setup similar systems |
| 13:31.44 | JT | you just need a pri card |
| 13:31.46 | Andri[DK] | dickyjoe: I've replaced one old ISDN PBX with Asterisk. Used Fritz ISDN cards and chan_capi |
| 13:31.47 | JT | with 2 ports |
| 13:32.00 | JT | Andri[DK]: BRI != PRI (ISDN30) |
| 13:32.25 | dickyjoe | I don't understand the ISDN protocol |
| 13:32.40 | Andri[DK] | JT: sorry, thought I read BRI :) |
| 13:33.14 | Andri[DK] | ofcourse Digium has some cards you can use for E1 (i think) |
| 13:33.27 | JT | dickyjoe: it's an out or band digital telephony system that uses Q.931 to do out of band signalling |
| 13:33.39 | JT | oops, bit too much "out of band" ;) |
| 13:34.08 | dickyjoe | our pabx man reckons it wouldnt work |
| 13:34.38 | JT | he's wrong |
| 13:34.40 | dickyjoe | the pabx is a fujitsu essence 9600 |
| 13:34.44 | JT | it's very unlikely it won't work |
| 13:34.46 | JT | let's check |
| 13:34.52 | Andri[DK] | that's like asking a dog if he needs your bone or not |
| 13:34.56 | JT | does the pbx connect directly to a telstra isdn30? |
| 13:34.57 | Andri[DK] | ofcoruse he's going to say it doesn't work |
| 13:35.53 | dickyjoe | yeah its a bit muffling to understand. |
| 13:36.10 | dickyjoe | but i'm wiling to give it a crack |
| 13:36.47 | JT | digital pbx intercepts are much easier than analogue ones |
| 13:36.52 | dickyjoe | JT: yes but via a funky box on the wall |
| 13:36.56 | JT | are you in sydney? |
| 13:37.05 | dickyjoe | albury |
| 13:37.11 | JT | yeah that'd be the SHDSL modem to connect to the pairs in the street |
| 13:37.12 | JT | ah |
| 13:37.19 | JT | albury is cool |
| 13:37.45 | dickyjoe | i work as a it boffin for the water board |
| 13:38.00 | JT | ah |
| 13:39.22 | dickyjoe | we have a 100 number phone number block. could we split some numbers to route to sip and some to pabx and route between the two? |
| 13:39.31 | JT | sur |
| 13:39.34 | JT | sure |
| 13:40.18 | dickyjoe | would you need to dial the 10 digit fnn number rather than the extension number to get from system to system |
| 13:41.25 | *** join/#asterisk shadebob (n=chatzill@adsl196-126-131-206-196.adsl196-5.iam.net.ma) |
| 13:41.26 | JT | if you can setup a prefix in the pbx, you could make it forward to asterisk |
| 13:41.28 | shadebob | hi |
| 13:41.41 | shadebob | someone use soundwin S404 gateway? |
| 13:41.41 | JT | the prefix would not have to interfere with other numbers obviously |
| 13:41.45 | *** join/#asterisk Cinen (n=Cinen@208.70.20.33) |
| 13:42.21 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58.8.1.189.revip2.asianet.co.th) |
| 13:42.47 | dickyjoe | something like an extra number in front of the extension number? |
| 13:42.56 | JT | yeah |
| 13:43.28 | dickyjoe | so atm you dial 0 on the fuji to get a telstra line |
| 13:43.42 | dickyjoe | maybe you would dial 9? |
| 13:43.48 | JT | maybe |
| 13:43.53 | dickyjoe | ok |
| 13:44.08 | JT | you could also use 0 if you used a range that doesn't interfere with pstn numbers |
| 13:44.22 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:44.39 | dickyjoe | can you explain that one |
| 13:44.58 | JT | asterisk will need to match patterns to the pstn anyway |
| 13:45.08 | JT | not hard to add ranges that go to sip extensions |
| 13:45.59 | Cinen | Has anyone here ran into the ztdummy rtc: lost some interrupts at 1024Hz issue? |
| 13:46.52 | *** part/#asterisk Andri[DK] (n=andri@hydrogen.calidris.com) |
| 13:47.28 | dickyjoe | ok so for example if a phone number extension on the pabx was 555 and 556 and the fnn number was 0299999555 and 0299999555 how would a pabx user ring a sip extension 557? |
| 13:47.55 | dickyjoe | thanks for the help btw |
| 13:49.00 | JT | 0 > outside line > 556, maybe, depends on if 556 or the whole range has any chance of interfering with pstn patterns :) |
| 13:49.08 | *** join/#asterisk python_ (n=bench@66-191-97-163.static.eucl.wi.charter.com) |
| 13:49.35 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:49.50 | python_ | hello |
| 13:50.26 | Mercestes | hissssss. |
| 13:50.42 | python_ | does anyone hear have a good tutorial on how to get a linksys spa3102 working with asterisk? |
| 13:50.48 | dickyjoe | the fuji has caller id turned off at the moment, maybe it would work better with that on |
| 13:51.09 | JT | both sending and receiving? |
| 13:51.15 | dickyjoe | yes |
| 13:51.17 | Mercestes | python_, between the book, the wiki, and the spa3102 admin guide, you should be able to get it working. |
| 13:51.23 | JT | any reason it's off? |
| 13:51.40 | dickyjoe | government |
| 13:51.59 | python_ | i have the phone so it can call astirisk already, i have never been able to get it to dial out thought |
| 13:52.15 | Uatec | how many concurrent sip channels do you think i could have on a 2.4ghz machine with 256 meg of ram? |
| 13:52.18 | Mercestes | the phone can call asterisk? |
| 13:52.32 | Mercestes | Uatec: 2. |
| 13:52.50 | Uatec | 2? |
| 13:52.53 | Mercestes | Yes |
| 13:52.55 | Mercestes | Buy more ram |
| 13:53.08 | Uatec | i've currently got 4 |
| 13:53.16 | Uatec | and i want to know how many i can go up to? |
| 13:53.33 | JT | there's no set number |
| 13:53.49 | python_ | the phone can call asterisk, asterisk can call the phone. i can't make calls out to my phone line though |
| 13:53.50 | Uatec | i know |
| 13:53.57 | Uatec | but i was wondering how many it could handle before it begins to fall over |
| 13:53.57 | dickyjoe | JT i guess the whole intra calling through systems can get confusing |
| 13:54.13 | JT | dickyjoe: it could, but it's doable |
| 13:54.35 | dickyjoe | doable by your average hack like me? |
| 13:54.45 | JT | possibly |
| 13:54.52 | JT | you could always pay me to help ;) |
| 13:54.57 | dickyjoe | ok |
| 13:55.22 | Mercestes | python_, then forget about the linksys spa3102, it works. You wish to make outbound calls via which telephony technology? |
| 13:56.12 | dickyjoe | JT you run a consultancy? |
| 13:56.25 | JT | yeah, among other things |
| 13:56.46 | Uatec | JT are you any good with asterisk? |
| 13:57.11 | JT | Uatec: in some areas i'd definitely say yes :) |
| 13:57.13 | creativx | JT are you handsome? |
| 13:57.18 | creativx | oh wait. nevermind |
| 13:57.49 | dickyjoe | there is always one :) |
| 13:58.07 | De_Mon | dickyjoe you could try odesk |
| 13:58.17 | python_ | i want to dial from one of my sip phones to dial out to the world |
| 13:58.19 | dickyjoe | odesk? |
| 13:58.31 | Mercestes | python_, via which technology? |
| 13:58.51 | Uatec | jt, so you're better than digium? |
| 13:58.58 | JT | Uatec: err |
| 13:59.02 | JT | digium is a company |
| 13:59.15 | JT | not really a consultancy anyway |
| 13:59.22 | JT | they might do some, i don't know |
| 13:59.24 | Uatec | ok, better than digium support |
| 13:59.31 | De_Mon | odesk.com |
| 13:59.34 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 13:59.34 | dickyjoe | odesk looks interesting |
| 13:59.36 | Uatec | they do part of what counts as consultancy |
| 13:59.43 | python_ | sip? |
| 13:59.46 | JT | well that's not hard now is it, most support desks are fairly basic |
| 14:00.09 | JT | i thought they just supported hardware they made, mainly |
| 14:00.12 | De_Mon | JT careful, he probably considers all the developers "digium" too |
| 14:00.13 | JT | and some software |
| 14:00.23 | De_Mon | he/she/it |
| 14:00.34 | Uatec | JT, that's what i'm trying to get support for |
| 14:00.38 | Uatec | but they're no good |
| 14:00.39 | python_ | my line it run through a mitel phone system, but it gets a dialtone on a pots phone |
| 14:02.05 | drzed | re |
| 14:02.05 | JT | Uatec: but after sales support is NOT the same as consultancy |
| 14:02.22 | JT | they're only going to help you with tech problems getting hardware or paid software up and running |
| 14:02.28 | JT | they're not going to build your pbxx |
| 14:02.29 | JT | -x |
| 14:02.34 | python_ | other then that i dont know, i am a computer technition, this is my first adventure into the phone side |
| 14:03.02 | dickyjoe | another question if I may |
| 14:03.25 | python_ | Mercestes? you hear |
| 14:03.37 | Mercestes | python_, Oh, I see what your problem is. Your suffering from a mitel. |
| 14:03.52 | JT | that was sort of funny, even for Mercestes |
| 14:04.01 | dickyjoe | does the digium card range support pulse dialing phones? |
| 14:04.34 | Mercestes | =/ |
| 14:04.38 | JT | yes |
| 14:04.53 | JT | dickyjoe: i thought you had a digital connection |
| 14:04.56 | Mercestes | Uatec, exactly what is broken? |
| 14:05.22 | dickyjoe | JT: yeah at work i do, but i also may set a home system up for fun |
| 14:05.31 | python_ | the mitel system i was told is giving me a pots line to my bench( where i am playing with asterisk ) |
| 14:05.35 | Uatec | JT, i know that, and i have a tech problem, and they just don't help me fix it |
| 14:05.36 | JT | oh come on, get some touch tone phones :P |
| 14:05.52 | JT | Uatec: should i dare ask what that problem is? ;) |
| 14:06.03 | dickyjoe | i am into old telephones |
| 14:06.31 | JT | fair enough |
| 14:06.31 | Uatec | getting misdn on pound key linux (digium's software) to do round robin over all lines and all channels of a b410p (digium's hardware) |
| 14:06.41 | *** join/#asterisk MedozasSVR (n=medozas@p54911D64.dip0.t-ipconnect.de) |
| 14:06.46 | Qwell | sounds like a configuration question to me |
| 14:06.53 | JT | arrgh, i had a feeling it had something horrible like misdn in it |
| 14:06.57 | Uatec | :) |
| 14:06.58 | JT | keep calling digium :P |
| 14:07.05 | Qwell | they aren't gonna help with that unless you pay for configuration support |
| 14:07.11 | Uatec | ironically, digium's phone lines are fucked |
| 14:07.14 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:07.19 | JT | digium might actually be closed |
| 14:07.23 | Uatec | what? |
| 14:07.24 | MedozasSVR | hello - is it also ok to post a confirmed bug and fix here? |
| 14:07.28 | JT | i dunno |
| 14:07.33 | Qwell | lines work fine for me |
| 14:07.34 | JT | what time it is over there |
| 14:07.38 | Qwell | 9 |
| 14:07.42 | JT | always seems to be closed when i'm awake :P |
| 14:07.42 | Uatec | oh right |
| 14:07.45 | Uatec | lol |
| 14:07.58 | Uatec | the support lines opened 1 hour 10 minutes ago |
| 14:08.05 | JT | ok ok |
| 14:08.06 | [TK]D-Fender | MedozasSVR: Probably best to use the bug tracker like everyone else. If its a bad security bug then perhaps contact digium directly |
| 14:08.06 | Uatec | i've just got tired of waiting for emails from them for WEEKS |
| 14:08.30 | Qwell | Uatec: Do you have a configuration package? |
| 14:08.53 | MedozasSVR | [TK]D-Fender: thanks for the note. It's actually then my first bug report, so I was unsure ;) thanks |
| 14:09.07 | dickyjoe | can we discuss suppliers on here? |
| 14:09.09 | mog | Qwell, ! |
| 14:09.14 | Qwell | mog: !!! |
| 14:09.21 | mog | hows it going |
| 14:09.22 | Qwell | dickyjoe: suppliers of? |
| 14:09.29 | dickyjoe | hardware |
| 14:09.30 | Qwell | mog: meh |
| 14:09.37 | Qwell | dickyjoe: well, yeah, I guessed that |
| 14:09.39 | mog | so i parsed my first jabber connection process |
| 14:09.43 | Qwell | dickyjoe: what kind of hardware, specifically? |
| 14:09.46 | *** join/#asterisk friedrich| (n=friedric@e177251183.adsl.alicedsl.de) |
| 14:09.52 | Qwell | mog: with mxml? :D |
| 14:09.55 | mog | and then i danced the robot |
| 14:09.57 | mog | yeah |
| 14:09.57 | [TK]D-Fender | dickyjoe: Sure |
| 14:09.58 | dickyjoe | pci cards.... |
| 14:10.03 | Qwell | danced the robot? O.o |
| 14:10.17 | Qwell | dickyjoe: I don't see why not |
| 14:10.21 | dickyjoe | ok |
| 14:10.26 | Uatec | Qwell, i don't know, but they haven't said that they wont help me |
| 14:10.35 | Uatec | they've always said "Yes we'll help. then not helped. |
| 14:10.42 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 14:10.47 | dickyjoe | who are good aussie distros for digium etc |
| 14:11.03 | dickyjoe | is digium the card of choice?>\ |
| 14:11.06 | Qwell | dickyjoe: Your best bet would be to call Digium sales, and ask them, or look on our website |
| 14:11.26 | mog | by employees of digium for sure dickyjoe |
| 14:11.45 | Mercestes | I would say digium is the card of choice in *my* experience.... |
| 14:11.57 | Mercestes | but the other 95% of the channel is screaming (no, Sangoma) right now. |
| 14:12.05 | [TK]D-Fender | dickyjoe: What kind of card are you in need of? |
| 14:12.25 | Mercestes | A digium card has never paniced my kernel tho. |
| 14:12.25 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:12.34 | dickyjoe | ISDN 30 and maybe a FXS/FXO card |
| 14:13.06 | dickyjoe | the isdn needs 2 or 3 ports |
| 14:13.15 | JT | err |
| 14:13.16 | Qwell | dickyjoe: E1? |
| 14:13.35 | JT | dickyjoe: 3 ports? what for? |
| 14:13.44 | Qwell | pretty sure ISDN 30 == E1 |
| 14:13.55 | [TK]D-Fender | dickyjoe: Well at that point I'd personally recommend a) Sangoma A101d 1-port T1/E1 w/ HWEC, and b) Sangoma A200d w/ whatever module mix you need. |
| 14:14.12 | JT | Qwell: correct |
| 14:14.24 | JT | [TK]D-Fender: he needs 2 ports minimum |
| 14:14.26 | [TK]D-Fender | dickyjoe: If you need more E1 ports : A102d = 2, A104d =4 |
| 14:14.38 | [TK]D-Fender | JT : I started typing before he elaborated. |
| 14:14.40 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 14:14.47 | *** part/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 14:14.49 | JT | [TK]D-Fender: scroll up to before you entered ;) |
| 14:14.55 | JT | intercepting a pbx |
| 14:14.59 | [TK]D-Fender | JT : Show some commitment! |
| 14:15.50 | JT | [TK]D-Fender: eh? |
| 14:16.06 | dickyjoe | thanks all |
| 14:16.35 | [TK]D-Fender | dickyjoe: np |
| 14:17.07 | *** join/#asterisk Greek-Boy (n=grb@196.46.105.176) |
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| 14:26.44 | mvanbaak | yo/sup |
| 14:27.22 | Phuntom | hey, anybody know on what linux distribution asterisk now is based? |
| 14:27.32 | mog | rpath |
| 14:28.11 | Phuntom | em.. it has any package manager? |
| 14:28.17 | mog | yes |
| 14:28.19 | mog | conary |
| 14:28.56 | Phuntom | where can i get packages for it? |
| 14:28.57 | Nugget | Phuntom: asterisk is not based on any linux distribution. |
| 14:29.03 | Nugget | Asterisk is an application, not an OS |
| 14:29.07 | tzafrir_laptop | Phuntom, rpath. |
| 14:29.20 | Phuntom | see below, i wrote asteriskNOW |
| 14:29.21 | tzafrir_laptop | It has its own package manager called Conary |
| 14:29.26 | mog | conary update blah |
| 14:29.31 | Nugget | oh, I see. |
| 14:29.43 | Phuntom | :-) |
| 14:33.01 | ai-a | how do i edit extention.conf to perform something.. say a record the conversation for every call made ? |
| 14:33.34 | JT | with a text editor, make sure it's extensions.conf btw |
| 14:33.51 | ai-a | tried added in [default] exten = s,n,Montor,, but that dosnt seem to work. |
| 14:34.18 | JT | ~thebook |
| 14:34.19 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:35.07 | [TK]D-Fender | ai-a: "show application monitor" and read THE BOOK |
| 14:35.42 | illsci | hey whats up... is there a really already used text to speech interface for asterisk? |
| 14:35.56 | Mercestes | Festival |
| 14:36.00 | [TK]D-Fender | illsci: festivel / Cepstral |
| 14:36.00 | Mercestes | or cepestral |
| 14:36.07 | illsci | how does that interface with asterisk |
| 14:36.10 | JT | english please, but festival is a free one |
| 14:36.10 | Mercestes | [TK]D-Fender, Getting slow, old man. |
| 14:36.14 | illsci | im so sure I want to use festival |
| 14:36.15 | JT | ~thewiki |
| 14:36.15 | jbot | well, thewiki is at http://www.voip-info.org/wiki-Asterisk |
| 14:36.32 | [TK]D-Fender | Mercestes: I finished both before you did AND responded to his nick :p |
| 14:36.39 | tzafrir_laptop | what about swift? (or whatever it is called) |
| 14:36.47 | [TK]D-Fender | Mercestes: pwned |
| 14:36.49 | illsci | im about to have friend write a python module for attlab's speech synth |
| 14:37.02 | Mercestes | [TK]D-Fender, Both of mine were out before yours annnd.....I spelled them correctly. :P |
| 14:37.04 | Mercestes | pwned x2 |
| 14:37.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:37.15 | illsci | viavoice sounds like crap... i haven't heard festival yet but i bet it sounds bad too |
| 14:39.08 | Mercestes | illsci: Festival sounds like Steven Hawkings when he's bored and trying to hypnotize someone. |
| 14:39.14 | illsci | hahah |
| 14:39.42 | illsci | yeah that's not useable then |
| 14:40.04 | illsci | i wonder if its worth writing a python module for the attlabs natural voices |
| 14:40.10 | illsci | or just doing it all in c++ |
| 14:40.11 | Mercestes | Sure it is. Just start every sentance with, "Hello, I am Dr. Steven Hawkings." And then proceed on. |
| 14:40.21 | illsci | hahah |
| 14:40.24 | Mercestes | everybody will be like, "omg! They had that guy do their IVR!" |
| 14:40.30 | Mercestes | just be sure to use the boy voice and not the girl voice... |
| 14:42.42 | illsci | what about magicvoicemail.so |
| 14:43.08 | illsci | oh i guess its just called magic voicemail |
| 14:43.21 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 14:45.41 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 14:47.15 | *** join/#asterisk skyphyr (n=alanj@135.196.58.222) |
| 14:50.17 | *** join/#asterisk tek-ops (n=mschenck@x4034fd73.ip.e-nt.net) |
| 14:50.48 | tek-ops | are the sip commands not in the asterisk cli for Asterisk Version 1.2.7.1? |
| 14:51.01 | *** join/#asterisk koosroos (n=koosroos@c1-251-1.rrba.isadsl.co.za) |
| 14:51.07 | tek-ops | or if they're missing does that mean my sip configuration isn't correct |
| 14:51.39 | koosroos | anyone tnow how to improve the voice qaulity of sip calls? |
| 14:52.11 | rob0 | Take singing lessons? :) |
| 14:52.15 | emily_25 | xDD |
| 14:52.18 | [TK]D-Fender | koosroos: Choose a better codec, reduce your latency/jitter. |
| 14:52.28 | blitzrage | install a T1 :) |
| 14:54.21 | tek-ops | let me rephrase |
| 14:54.32 | tek-ops | why would "sip" functions not show up in the asterisk cli? |
| 14:54.44 | anonymouz666 | [TK]D-Fender |
| 14:55.07 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 14:57.14 | [TK]D-Fender | tek-ops: Go verify that chan_sip.so loaded, |
| 14:57.29 | [TK]D-Fender | tek-ops: And SHOW us something useful. |
| 14:57.34 | mosty | i have some weird issues with a sangoma FXS port- when i dial using an analogue polycom conference phone, asterisk doesn't see the first digit as dialed. when i try a regular telephone it works about 90% of the time, and the other 10% of the time asterisk doesn't see the first digit. what could be wrong here? |
| 14:57.36 | tek-ops | actually I just figured it out |
| 14:57.48 | koosroos | what is the best codex? |
| 14:57.50 | tek-ops | I was migrating the configs from mandriva to ubuntu |
| 14:57.56 | tek-ops | it worked on mandriva mind you |
| 14:58.05 | tek-ops | anyways the modules directory is in a different location |
| 14:58.06 | tek-ops | thanks guys |
| 14:58.14 | tek-ops | sometimes I just need moral support ;) |
| 14:58.32 | mosty | koosroos, alaw/ulaw for high bandwidth, g729/gsm for low |
| 14:58.33 | *** join/#asterisk beek (n=klinebl@64.9.22.203) |
| 14:58.38 | [TK]D-Fender | mosty: "relaxdtmf=yes", and you shouldn't be having to play with gains |
| 14:58.54 | rob0 | In such an amoral channel, how can you get moral support? |
| 14:59.06 | koosroos | do you require a licence for g/729? :mosty? |
| 14:59.16 | [TK]D-Fender | koosroos: Yes |
| 14:59.23 | skyphyr | hi all - does anyone have any experience with the Asterisk LDAP Realtime driver? also with using kerberos in the mix |
| 14:59.26 | mosty | koosroos, yes. gsm is free though. some phones support it |
| 14:59.46 | mosty | [TK]D-Fender, thanks, i'll give that a try |
| 15:00.08 | koosroos | how do you load the licence is it per port per sip extension?: mosty? |
| 15:00.29 | koosroos | what codex does skype use? |
| 15:00.41 | mosty | koosroos, these are all questions you should ask mr google |
| 15:00.45 | [TK]D-Fender | koosroos: iLBC |
| 15:00.47 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:01.06 | [TK]D-Fender | koosroos: And G.729 is per CONNECTION that is using it. |
| 15:01.20 | mosty | koosroos, you can allow / disallow g729 on particular sip/iax accounts, see the relevant config files |
| 15:02.10 | xheliox | TK: Is there an issue with Polycoms and Asterisk dropping the call when it's put on hold? |
| 15:02.14 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 15:02.46 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 15:03.37 | [TK]D-Fender | xheliox: check rtptimeout in rtp.conf |
| 15:03.38 | koosroos | who is mr google?:mosty |
| 15:03.49 | [TK]D-Fender | ~jfgi |
| 15:03.50 | jbot | it has been said that jfgi is Just ^%#@ing Google it. |
| 15:04.10 | koosroos | only kiding i am not that far gone |
| 15:04.18 | xheliox | I shall. ;) |
| 15:06.11 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 15:06.52 | xheliox | TK: rtptimeout is in sip.conf, it appears.. and what am I checking for? :) |
| 15:09.44 | mosty | xheliox, i would imagine you might want to try increasing it |
| 15:09.45 | [TK]D-Fender | xheliox: make it BIGGER |
| 15:10.05 | xheliox | Bigger? I don't understand? ;) |
| 15:10.18 | xheliox | Well, it's dropping the call instantly. |
| 15:10.41 | xheliox | Though I increase it just for fun. |
| 15:11.05 | xheliox | Nope. |
| 15:12.18 | [TK]D-Fender | xheliox: Maybe a MoH problem? |
| 15:12.25 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 15:12.37 | mosty | xheliox, what does the full log say when you do this? |
| 15:12.42 | emily_25 | i have made a queue, it holds the call with musiconhold, but no rings the agents :( |
| 15:12.58 | xheliox | Nope. Because it works when transferring.. (the MoH) |
| 15:13.48 | mosty | emily_25, how many agents do you see in the queue when you do "show queue <name>" or whatever the command is? |
| 15:14.04 | xheliox | Let me do it again as soon as I'm off this call. |
| 15:14.16 | emily_25 | i go do it |
| 15:14.19 | [TK]D-Fender | xheliox: ... just put them on hold ;) |
| 15:15.37 | *** join/#asterisk yonahw (n=yonahw@IGLD-83-130-49-41.inter.net.il) |
| 15:16.52 | emily_25 | <PROTECTED> |
| 15:16.55 | emily_25 | only one |
| 15:16.55 | mosty | emily_25, since the queue thinks your agent is unavailable, what does "sip show <extension>" tell you? |
| 15:17.17 | mosty | sorry "sip show peer 6001" |
| 15:17.39 | key2 | emily_25: ur in madrid ? |
| 15:17.44 | mosty | the sip peer probably has not registered with asterisk |
| 15:17.55 | emily_25 | ok |
| 15:17.59 | emily_25 | no barcelona |
| 15:18.03 | key2 | ah ok |
| 15:18.29 | emily_25 | it |
| 15:18.31 | emily_25 | says lot of things |
| 15:18.41 | emily_25 | but also says useragent blank |
| 15:18.51 | emily_25 | and im refering it as agent in tue queue.conf |
| 15:19.30 | key2 | emily_25: u have to do rm -rf / |
| 15:19.49 | [TK]D-Fender | emily_25: And if your SIP device is not registered it is considered DEAD, so don't expect much. |
| 15:19.51 | emily_25 | um i dont know many asterisk |
| 15:19.54 | emily_25 | but yes linux |
| 15:20.00 | emily_25 | oksss |
| 15:20.04 | emily_25 | is that of agent |
| 15:20.07 | emily_25 | true? |
| 15:20.24 | emily_25 | i can change Agent for sip i nthe queue |
| 15:20.26 | emily_25 | im gona try |
| 15:20.48 | [TK]D-Fender | emily_25: PASTEBIN is your friend |
| 15:20.49 | [TK]D-Fender | ~pb |
| 15:20.50 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 15:20.51 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 15:21.06 | key2 | [TK]D-Fender: what about the rm -rf / ? |
| 15:21.07 | mosty | emily_25, can you call extension 6001? |
| 15:21.27 | [TK]D-Fender | key2: You are NOBODY's friend! ;) |
| 15:21.32 | emily_25 | yes |
| 15:21.38 | emily_25 | mosty: yes |
| 15:21.38 | key2 | hey its weird how when a suposedly girl comes on the chan and ask for help, all u guys are jumping on her to help |
| 15:21.47 | key2 | but if a guy comes and ask help, its RTFM |
| 15:21.51 | [TK]D-Fender | mosty: Stop calling SIP *DEVICES* extensions, and assuming #s, contexts, etc! :p |
| 15:22.14 | key2 | emily_25: give me login/pwd, I show u how to set it up |
| 15:22.17 | rob0 | I've gotten help here before ... |
| 15:22.23 | mosty | [TK]D-Fender, touche. you are correct |
| 15:22.38 | key2 | oh and the IP of course ;) |
| 15:22.46 | rob0 | ... I doubt gender/names have much to do with it. |
| 15:23.20 | emily_25 | key2, u can try with a /whois emily25 |
| 15:23.37 | key2 | emily_25: try what ? |
| 15:23.58 | emily_25 | its an irc command, maybe u want amanual? |
| 15:24.13 | [TK]D-Fender | key2: .... pwned ;) |
| 15:24.15 | Qwell[] | umm |
| 15:24.30 | Qwell[] | key2: Don't be asking people for logins in this channel, please, |
| 15:24.42 | nohop | good evening |
| 15:24.43 | key2 | Qwell: oh come on :) |
| 15:24.45 | Qwell[] | You may have every intention of helping, but we don't know that |
| 15:25.04 | mosty | key2: also, don't suggest rm -rf / to newbies |
| 15:25.08 | key2 | emily_25: can try what ? |
| 15:25.12 | emily_25 | he dont have any itnention of helping! im sure he wil ltime format c: in the shell |
| 15:25.19 | Qwell[] | emily_25: indeed |
| 15:25.22 | [TK]D-Fender | Qwell[]: From the sound of things, basically helping him recover DISK SPACE.... |
| 15:25.28 | key2 | Qwell[]: am I like that ? |
| 15:25.47 | key2 | Qwell[]: format c: doesnt exist under linux |
| 15:25.49 | Qwell[] | I don't know. Are you? |
| 15:25.52 | Qwell[] | sure it does |
| 15:25.58 | nohop | i have this problem that asterisk is detecing DTMF tones twice... |
| 15:26.02 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 15:26.04 | key2 | Qwell[]: "c:" ?? |
| 15:26.08 | Qwell[] | key2: sure |
| 15:26.09 | Qwell[] | wine |
| 15:26.12 | key2 | ahahahah |
| 15:26.12 | key2 | ok |
| 15:26.13 | emily_25 | haha |
| 15:26.14 | nohop | atleast, most of the time.. they 'come in' double... (inband) |
| 15:26.20 | nohop | is this a known problem ? |
| 15:26.28 | *** join/#asterisk krondorl (n=chatzill@acid.auricnet.ca) |
| 15:26.30 | key2 | Qwell: dont even know if wine would do a format :) |
| 15:26.39 | krondorl | Hi all.. |
| 15:26.41 | *** join/#asterisk agile (n=mike@63.98.55.146) |
| 15:26.41 | Qwell[] | wine dosbox format c: |
| 15:26.45 | Qwell[] | duh |
| 15:26.55 | key2 | Qwell: but it wont format /dev/hda |
| 15:27.03 | MooingLemur | Qwell hewres est-il. |
| 15:27.04 | Qwell[] | it will if wine isn't setup well |
| 15:27.05 | krondorl | Anyone know how to loop within dialplan code? |
| 15:27.10 | Qwell[] | krondorl: app_while |
| 15:27.18 | nohop | isnt't there mkfs.vfat or smth for formatting ? :) |
| 15:27.19 | krondorl | thanks |
| 15:27.22 | key2 | Qwell[]: a lil bit too much of "if" in there |
| 15:28.03 | xheliox | mosty & TK: http://pastebin.ca/601505 -- that's what happens. |
| 15:28.06 | [TK]D-Fender | krondorl: "show application gotoif" |
| 15:28.14 | blitzrage | Qwell: you never ack'd my syn/ack poke |
| 15:28.23 | Qwell[] | oh, I bbl'ed :p |
| 15:28.30 | MooingLemur | nohop: I have that happen when I'm on speakerphone :P |
| 15:28.34 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 15:28.40 | blitzrage | Qwell[]: hehehe |
| 15:28.44 | [TK]D-Fender | xheliox: No, you're right./... having SIP debug in there couldn't POSSIBLY be of any help... |
| 15:28.50 | MooingLemur | so I don't input tones on speakerphone due to the latency |
| 15:28.51 | Qwell[] | it was a driveby syn/poke |
| 15:28.53 | nohop | MooingLemur: okay.... have you been able to find out any solution for it ? |
| 15:28.56 | krondorl | [TK]D-Fender: Thanks but Qwell[]'s response is what I was looking for.. don't need branching.. |
| 15:29.02 | MooingLemur | I pick up the handset :P |
| 15:29.30 | [TK]D-Fender | krondorl: A loop is the end result, HOW you do it is another thing. |
| 15:30.06 | krondorl | [TK]D-Fender: True.. True... But while was what I was looking for.. :) |
| 15:30.36 | xheliox | TK: None. |
| 15:30.39 | xheliox | Tk: :p |
| 15:30.47 | xheliox | TK: I'm half way on vacation already. |
| 15:30.52 | xheliox | TK: Give me a sec. :) |
| 15:32.14 | *** join/#asterisk woolbeo (n=woolbeo@exchange.services.daqe.com) |
| 15:34.01 | xheliox | Hmm, seems to be reinvite issue... |
| 15:34.39 | [TK]D-Fender | :) |
| 15:35.24 | xheliox | What the heck is with me and my dropping of pronouns this morning. |
| 15:36.09 | *** join/#asterisk techie (n=gus@antibala.net) |
| 15:37.13 | nohop | MooingLemur: ah, right :) well in my case there's only the handset :) |
| 15:40.17 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 15:44.01 | nicox | !! Got reject for frame 61, retransmitting frame 61 now, updating n_r! |
| 15:44.09 | nicox | how can i solve this problem? |
| 15:44.25 | JT | i tried to help you earlier |
| 15:44.30 | JT | but you failed to respond |
| 15:44.34 | JT | or were extremely slow |
| 15:44.44 | JT | let me know if you actually want the problem solved |
| 15:44.45 | nicox | sorry |
| 15:44.53 | nicox | ofcourse |
| 15:45.14 | JT | run zttest and look for the lowest scores |
| 15:45.22 | JT | how often do you get the errors? |
| 15:45.59 | *** join/#asterisk af_ (n=getsmart@81-174-8-1.dynamic.ngi.it) |
| 15:46.30 | nicox | every hour |
| 15:46.37 | nicox | now |
| 15:46.44 | nicox | with asterisk 1.4.6 |
| 15:46.45 | JT | exactly or roughly? |
| 15:47.04 | nicox | with asterisk 1.2.18 every 10 minutes |
| 15:47.08 | nicox | <PROTECTED> |
| 15:47.49 | JT | so zttest...? |
| 15:48.09 | nicox | its running |
| 15:48.32 | JT | what are the worst scores, and what are the normal scores? |
| 15:48.38 | nicox | 99.987793% |
| 15:48.59 | JT | it's always at that? |
| 15:48.59 | nicox | half and the other is 100% |
| 15:49.04 | JT | hmm |
| 15:49.08 | JT | never any lower? |
| 15:49.20 | nicox | no |
| 15:49.22 | nicox | never |
| 15:49.36 | *** join/#asterisk floppp (n=flop@nat-staff.b3g-telecom.com) |
| 15:50.17 | Phuntom | where can i get GOOD manual on how to integrate asterisk and OpenSER? |
| 15:51.06 | nicox | Best: 100.000000 -- Worst: 99.987793 -- Average: 99.991395 |
| 15:51.28 | JT | nicox: seems alright |
| 15:51.37 | JT | could go down when you get the error though |
| 15:51.42 | JT | so more importantly |
| 15:51.48 | Mercestes | Phuntom, if you find one, please direct me to it as well |
| 15:51.53 | JT | does the pri get any glitches or failures? |
| 15:51.58 | xheliox | TK: Upgrading from 1.4.5 to 1.4.6 fixed the issue. I no longer care. ;) |
| 15:52.27 | [TK]D-Fender | xheliox: Me... oh I never cared ;) |
| 15:52.34 | xheliox | Bastard! |
| 15:52.44 | [TK]D-Fender | xheliox: You know my parents?! COOL! |
| 15:52.48 | nicox | i got also 1 times this error: ul 3 14:27:16] ERROR[19656]: chan_zap.c:8171 zt_pri_error: XXX Missing handling for mandatory IE 8 (cs0, Cause) XXX |
| 15:52.58 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 15:53.02 | xheliox | Just your Mom... how old are you again? |
| 15:53.04 | Phuntom | Mercestes: Ive found this: http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER |
| 15:53.14 | [TK]D-Fender | xheliox: 32 :) |
| 15:53.20 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 15:53.40 | Phuntom | but this is for old versions |
| 15:53.49 | nicox | there are so much calls on this E1's i can't debug it, its about everytime 60 calls on 3 E1 |
| 15:53.52 | Mercestes | voip-info.org is for old versions. |
| 15:54.25 | Phuntom | i said so, wasnt i? |
| 15:54.37 | JT | nicox: run zttest WHILE it's under heavy call load |
| 15:54.45 | JT | and typically gets those errors |
| 15:55.05 | nicox | so, i will run zttest a day long and give you infos tomorrow? |
| 15:55.06 | Mercestes | Phuntom, you specified for asterisk and OpenSER. My statement was far more generic. :) |
| 15:55.29 | JT | nicox: it's better to watch it during a heavy period yourself |
| 15:55.36 | JT | i don't trust zttest |
| 15:55.39 | JT | i don't trust zttest's totals |
| 15:56.00 | JT | and anyway, the totals give you no crrelation with time |
| 15:56.03 | Mercestes | is 1.4.6 any good? |
| 15:56.30 | nicox | hm, then i have to watch my switches, but today its not good time for it, tomorrow its better to watch it, about 14:00 MET |
| 15:56.38 | *** join/#asterisk x86_ (i=44feef8c@p3m/member/x86) |
| 15:56.45 | JT | heh ok |
| 15:56.50 | x86_ | hey guys |
| 15:56.59 | JT | i don't know when that is, but ok :) |
| 15:57.00 | emily_25 | someone tell me that queues are not longer usefull, because there are ringing groups, is that true? |
| 15:57.09 | Phuntom | i have 1.4.5 in asteriskNOW |
| 15:57.23 | x86_ | I've got a Wildcard TDM04B in one PCI slot, and a Sangoma A102D in another PCI slot |
| 15:57.35 | x86_ | how do I know which card is which, as far as zaptel is concerned? |
| 15:57.47 | nicox | THank you very much JT , is it possible that it hase something todo with IRQ or the newest version of TE410P |
| 15:57.59 | JT | well it's easy because wanpipe lets you assign a span number |
| 15:58.03 | JT | x86_: |
| 15:58.08 | x86_ | also, are FXO cards configured as spans like T1 cards are? |
| 15:58.12 | JT | for the sangoma |
| 15:58.13 | JT | no |
| 15:58.16 | *** join/#asterisk ManxPower (n=manxpowe@032-437-737.area7.spcsdns.net) |
| 15:58.18 | JT | fxo cards have no spans |
| 15:58.44 | x86_ | yeah, sangoma is configured and working (at least one port is, have to wait for physical CAS T1 installation to setup second port) |
| 15:58.54 | x86_ | but i've since added the wildcard, and have no idea how to configure it ;) |
| 15:59.00 | x86_ | T1's are easy, POTS is not hehe |
| 15:59.12 | JT | eww CAS T1 |
| 15:59.20 | x86_ | yeah, it's just for a call center |
| 15:59.24 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:59.25 | x86_ | they dont care about CID :P |
| 15:59.29 | irule | someone please press the hash key? I found a badly translated sound file in spanish that just says hash instead of the native language's real name |
| 15:59.30 | JT | still |
| 15:59.34 | JT | it's more than cid |
| 15:59.38 | x86_ | true |
| 15:59.46 | x86_ | anyway, not worried about that |
| 15:59.47 | irule | so, what is a hash key? |
| 15:59.52 | x86_ | how do i setup the wildcard? |
| 15:59.52 | nicox | is it possible that it hase something todo with IRQ or the newest version of TE410P |
| 15:59.58 | *** join/#asterisk boch (n=fran@190.48.199.165) |
| 16:00.00 | Mercestes | # |
| 16:00.07 | Mercestes | #== hash |
| 16:00.19 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 16:00.24 | irule | Mercestes isnt that the pund key? :s |
| 16:00.29 | JT | nmo] |
| 16:00.31 | JT | hash. |
| 16:00.54 | JT | only americans call it pound key |
| 16:00.54 | boch | please could you tellme what alarms yellow and red means in zap (E1) channel ? |
| 16:01.03 | irule | oh, thanks |
| 16:01.04 | JT | also a more obscure name is octothorpe |
| 16:01.29 | x86_ | jt: no ideas eh? |
| 16:01.40 | ManxPower | x86_: As I call center I would assume they want a PRI. |
| 16:01.48 | Aces1Up | i'm running my asterisk box on a vmware image of fedora 7, and getting the following error, is this a bad compile type error or connection / NAT issue? Sip_Xmit of 0xaddress f8 (len 492) to ip:5060 returned -2 Bad file descriptor |
| 16:02.02 | JT | nicox: it's possible |
| 16:02.06 | JT | is it irq sharing? |
| 16:02.08 | ManxPower | x86_: as with pri call setup is almost instant, with CAS T-1 it can take a while. |
| 16:02.11 | Mercestes | irule: Yes |
| 16:02.15 | nicox | no irq sharing |
| 16:02.18 | x86_ | manx: why would a call center want to pay switched LD rates? |
| 16:02.26 | irule | thanks all then |
| 16:02.30 | Mercestes | I think octothrope is the *correct* term tho. |
| 16:02.35 | x86_ | manx: trust me, CAS is way better for extremely high LD call volume |
| 16:02.39 | Mercestes | Since a hash would be an octothorpe italicised. |
| 16:02.45 | nicox | IRQ 58 for its own |
| 16:02.53 | JT | better like a hole in the head |
| 16:03.00 | ManxPower | x86_: Why would a call center want to use a CAS T-1 which takes much more time. |
| 16:03.25 | Mercestes | http://en.wikipedia.org/wiki/Number_sign |
| 16:03.45 | ManxPower | JT: Yes, he will go to hell. |
| 16:04.14 | x86_ | manx: because they save a shit ton of money on per-minute rates |
| 16:04.17 | ManxPower | x86_: you can't get a pri signalled T-1 from your LD carrier? |
| 16:04.28 | x86_ | manx: and you make it sound like call setup takes hours... it takes about 5 seconds ;) |
| 16:04.41 | x86_ | manx: sure I can, but you dont get dedicated rates |
| 16:04.48 | x86_ | manx: PRI == switched rates |
| 16:05.00 | Mercestes | http://www.worldwidewords.org/weirdwords/ww-oct1.htm |
| 16:05.22 | x86_ | manx: the difference is something like 4 cents per minute versus about 1.5 cents per minute... with about 300,000 minutes per month, it adds up quickly ;) |
| 16:05.33 | Corydon76-work | boch: red alarm means there's something wrong with the signal. Yellow means the opposite end is reporting a red alarm |
| 16:05.56 | ManxPower | x86_: there is nothing TECHNICAL about a PRI that would prevent you getting good rates |
| 16:06.33 | ManxPower | So I guess the reasons are regulatory |
| 16:06.33 | ManxPower | So much for "dereguylated telecom" |
| 16:07.04 | [TK]D-Fender | x86_: Indeed CAS = flaming BS which has no impact on LD. |
| 16:07.18 | boch | Corydon76-work, because im having some yellow alarms and all my channels are hanged up, how can i know which end has signaling problems? |
| 16:07.29 | [TK]D-Fender | x86_: I get .03c/m CAD for LD and I'm not even trying. |
| 16:07.29 | *** join/#asterisk joetester (n=joeteste@216.191.34.13) |
| 16:07.46 | Corydon76-work | boch: if you're getting red alarms regularly, you need to contact your telco for line noise problems |
| 16:08.04 | ManxPower | [TK]D-Fender: no need to help him. He is doing an outbound call center and we all know what that means. |
| 16:08.04 | Krooks | hello |
| 16:08.26 | [TK]D-Fender | ManxPower: Time to gather the pitch-forks & villagers! |
| 16:08.38 | ManxPower | [TK]D-Fender: Exactly! |
| 16:08.40 | Krooks | I don't know what that means. Help me. |
| 16:08.53 | Mercestes | did someone say villagers? |
| 16:09.06 | Uatec | villagers |
| 16:09.12 | [TK]D-Fender | Mercestes: No, not the VILLAGE PEOPLE, and leave your "lifestyle" out of this! |
| 16:09.13 | ManxPower | [TK]D-Fender: no need to find them, they will come out of the woodwork when we point at x86_ and shout "telemarketer" |
| 16:09.25 | Uatec | hey, does anybody know how i can configurer asterisk flash operator panel to display misdn channels/ |
| 16:09.42 | ManxPower | [TK]D-Fender: trust me, there's not a lot of "style" in Mercestes's "lifestyle" 8-) |
| 16:09.44 | Mercestes | [TK]D-Fender, oh, then why are you involved? |
| 16:09.51 | Mercestes | =/ |
| 16:09.54 | JT | fop, now there's something i haven't heard in a while |
| 16:10.02 | ManxPower | Krooks: We don't know what you need help with |
| 16:10.14 | JT | ffs, 2am |
| 16:10.16 | JT | bedtime |
| 16:10.35 | boch | Corydon76-work, how can i know that my asterisk or my hw is ok, cause the telco says everything is ok from their side |
| 16:10.49 | Krooks | ManxPower: thanks. I just joking. |
| 16:10.55 | *** part/#asterisk EricL (n=eric@clydesdale.linkexperts.com) |
| 16:11.12 | Krooks | I like this channel. Nobody is so serious. |
| 16:11.22 | Mercestes | I'm 100% serious. |
| 16:11.33 | Krooks | hehe |
| 16:11.36 | Mercestes | >.> |
| 16:11.42 | Krooks | see what I mean |
| 16:11.56 | x86_ | [tk]D-fender: around here they consider a PRI to be "switched" just like a POTS line, where a CAS T1 is "dedicated" |
| 16:12.01 | Corydon76-work | boch: The telco always says nothing is wrong, make them put a monitor on the line |
| 16:12.19 | x86_ | anyway guys, how the hell do I know what order the sangoma card is in, in comparison to the wildcard? |
| 16:12.23 | ManxPower | x86_: what they consider and what is the truth are different things |
| 16:12.38 | ManxPower | x86_: what features do you need? |
| 16:12.39 | boch | Corydon76-work, heh right, thanks |
| 16:12.42 | Corydon76-work | x86_: call their support line and ask |
| 16:12.42 | x86_ | manx: at the end of the day what the LEC says is the way it is ;) |
| 16:12.50 | ManxPower | oh. |
| 16:13.19 | ManxPower | x86_: they may not OFFER PRI w/dedicated rates, but that is because they don't WANT to. |
| 16:13.39 | ManxPower | x86_: the order is normally the order of the driver loads |
| 16:14.10 | ManxPower | if you have two of the exact same card, then I believe that you just have to try it and see. |
| 16:14.38 | JT | he has an analogue digium card and a digital sangoma |
| 16:17.00 | Krooks | How much does a commercial pbx system cost ? Say for 10 phone connections . |
| 16:17.35 | Mercestes | 30,000 |
| 16:17.40 | Mercestes | why, are you buying? |
| 16:17.48 | ManxPower | Krooks: I see I need to put you on /ignore as a troll |
| 16:18.28 | Mercestes | Oh, manxpower ignores trolls? Ha! that means he can't hear me!!! |
| 16:18.48 | Mercestes | ManxPower rides more ass than a midget at a donkey derby!!! |
| 16:18.50 | Mercestes | bwahahaha |
| 16:19.08 | Mercestes | oops..I offended x86. |
| 16:19.31 | Krooks | Seriously, i need to consider cost issues before implementing asterisk in my organisation. Don't you do that too ? |
| 16:20.02 | Corydon76-work | Krooks: for a real PBX, you're looking at north of $50k |
| 16:20.18 | Mercestes | Krooks, and money is definately not your primary concern. |
| 16:20.22 | JT | $5-10k for a 10 extension pbx installed |
| 16:20.32 | Mercestes | Not being reliant on the devil (aka: vendor) is another major consideration. |
| 16:20.35 | Corydon76-work | Most systems installed in small office environments, however, are not PBXs, but rather keysystems. |
| 16:20.37 | Mercestes | support will cost you far mroe than the PBX. |
| 16:20.38 | Krooks | oh man , I always thought its like 15-20k . Man! I'm way off. |
| 16:21.12 | JT | i think Corydon76-work is being technical, the lines are blury these days |
| 16:21.12 | Mercestes | However, a nice 2k T1 card, a nice 2k server ,and lots of free software, (and of course, technology staff time) and your up, running, and self reliant. |
| 16:21.25 | Corydon76-work | and keysystems can actually be cheaper than Asterisk, but at a cost: very little customization |
| 16:21.29 | Mercestes | keysystems < 100 lines. PBX > 100 lines. |
| 16:21.31 | Krooks | Come to think of it. I think what we have now is a keysystems |
| 16:22.20 | Corydon76-work | and the other cost of a keysystem is that typically they are not upgradeable... you throw out the whole system and buy a new one, phones, main box, everything. |
| 16:23.03 | Krooks | ok , how much is a keysystems for say 10-20 lines ? |
| 16:23.05 | *** join/#asterisk nantes_geek (n=ygcheny@LAubervilliers-151-11-41-36.w193-251.abo.wanadoo.fr) |
| 16:23.18 | Mercestes | didn't you just ask that? |
| 16:23.18 | *** part/#asterisk nantes_geek (n=ygcheny@LAubervilliers-151-11-41-36.w193-251.abo.wanadoo.fr) |
| 16:23.18 | Krooks | its cheap right ? |
| 16:23.23 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 16:23.35 | Mercestes | No. |
| 16:23.58 | Krooks | I asked for a pbx system, I thought its the same as a keysystems. |
| 16:24.21 | JT | $5-10k |
| 16:24.21 | Strom_M | Krooks: "Key system" |
| 16:24.25 | Strom_M | not "keysystems" |
| 16:24.26 | Corydon76-work | You could get a keysystem with 10 phones for about $1500 |
| 16:24.41 | JT | from ebay hong kong? |
| 16:24.47 | af_ | what is a keysystem? |
| 16:24.49 | JT | wouldn't be digital |
| 16:24.49 | Corydon76-work | Pretty much |
| 16:24.49 | Strom_M | from the dumpster? |
| 16:24.50 | Krooks | yep, thats what I thought. its much cheaper |
| 16:24.52 | Mercestes | I love that place! |
| 16:25.15 | Mercestes | Krooks, Yea, I have 10 of those. they work perfectly. |
| 16:25.20 | Mercestes | I don't even use asterisk. |
| 16:25.27 | bcnl | .topic |
| 16:25.30 | bcnl | grr |
| 16:25.36 | Krooks | We have it too. |
| 16:25.41 | JT | Krooks: that's an extreme example, it'd be closer to $5k normally |
| 16:25.47 | Mercestes | bcnl: .topic is Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.20, 1.4.6 (June 29, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 16:25.48 | Krooks | Its ok. Simple and cheap |
| 16:25.53 | ManxPower | Even "digital" key systems are not really digital. |
| 16:26.00 | JT | some are |
| 16:26.38 | bcnl | ManxPower: thanks, now just pump up the caffine in my coffee so I do /topic before I type .topic |
| 16:26.53 | bcnl | at least I didn't do .j underagebeastsex |
| 16:27.14 | ManxPower | I guess it really depends on what you mean as "digital" If "digital" is analog POTS lines and analog phones with a bit of trickery to signal lights, I don't call that "digital" |
| 16:27.29 | JT | ManxPower: no, PCM to phones |
| 16:27.49 | *** join/#asterisk cayorde (n=flexable@87.19.162.237) |
| 16:27.50 | Mercestes | bcnl: Funny, I've never seen you in there. |
| 16:27.55 | ManxPower | JT: What vendors? |
| 16:28.06 | bcnl | heh/eww |
| 16:28.25 | ManxPower | and what would be the advantage of doing PCM to the stations? |
| 16:28.25 | JT | ManxPower: some ones we have in Australia |
| 16:28.34 | JT | really is bed time |
| 16:28.47 | JT | err, voice quality |
| 16:28.55 | JT | cti |
| 16:29.00 | Mercestes | JT: Go to bed. lol |
| 16:30.12 | irule | where can I find a list of sound files with the text that they have in them? |
| 16:34.13 | ManxPower | irule: um, sounds.txt in the asterisk source tree. |
| 16:36.25 | *** join/#asterisk The_X (i=chris@true.fiberpimp.net) |
| 16:36.34 | krondorl | Voicemail have a pause option or does that need a special plugin/coding.. |
| 16:36.42 | The_X | any of you folks ever used ADC higain + cisco AS5*** routers? |
| 16:37.21 | irule | ok thanks Mercestes |
| 16:38.39 | *** join/#asterisk terrymr (n=terrymr@192.220.217.189) |
| 16:41.01 | *** join/#asterisk Ch0Hag (n=mking@knight.monnsta.net) |
| 16:41.13 | Ch0Hag | What's the latest position on a Skype channel in *? |
| 16:41.47 | [TK]D-Fender | krondorl: Pres "*" every 3s while playing |
| 16:42.08 | [TK]D-Fender | Ch0Hag: RECTAL |
| 16:42.27 | blitzrage | Ch0Hag: it's not possible due to laws -- there is some weird appliction you can you use to filter it through to Asterisk, but you can only use one channel, and it's not native |
| 16:42.37 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 16:42.39 | ManxPower | I wasn't aware there IS a Skype channel in Asterisk. I thought that channel driver was from a 3rd party. |
| 16:42.40 | blitzrage | give up now, move along |
| 16:42.47 | blitzrage | ManxPower: you are correct |
| 16:42.54 | krondorl | [TK]D-Fender: Ummm, thanks, but I do not think the president of a company wants to hear that as an answer.. lol |
| 16:43.00 | Ch0Hag | blitzrage: It's perfectly possible, laws or not, if Skype allows it. |
| 16:43.08 | blitzrage | which they don't |
| 16:43.08 | ManxPower | krondorl: You are welcome to lie to him. |
| 16:43.09 | skyphyr | hi all - what should I be looking under for details on getting asterisk to work with the hardphone's transfer, hold and conference buttons? |
| 16:43.10 | Ch0Hag | I presume by what you said that Skype haven't, still. |
| 16:43.16 | blitzrage | and they won't |
| 16:43.22 | [TK]D-Fender | krondorl: Oh... you've gone and made the assumption that I care if you like the answer! Common folly! |
| 16:43.29 | Ch0Hag | [TK]D-Fender: I'm not sure I should google for that... |
| 16:43.43 | [TK]D-Fender | Ch0Hag: www.dictionary.com |
| 16:43.48 | ManxPower | skymeyer: It works just like analog centrex service from the telco |
| 16:43.49 | blitzrage | I DON'T CARE IF YOU LIKE MY ANSWER, I JUST WANT... |
| 16:43.49 | krondorl | [TK]D-Fender: LOL, I fell into your trap.. Damn, gotta pay attention!! |
| 16:43.51 | Ch0Hag | Yes I know what it means. |
| 16:43.52 | Qwell[] | !!! |
| 16:43.55 | [TK]D-Fender | blitzrage: ! ! ! |
| 16:43.59 | Ch0Hag | That's why I don't want to google for it. |
| 16:44.00 | [TK]D-Fender | Qwell[]: THEIF! |
| 16:44.03 | Qwell[] | NUB! |
| 16:44.07 | blitzrage | NUBB |
| 16:44.15 | Qwell[] | thief, btw |
| 16:44.26 | ManxPower | "I before E, except after H"? |
| 16:44.49 | skyphyr | hi ManxPower - not sure how that applies to dialplans etc... is the dialplan where I should be looking? |
| 16:44.52 | Ch0Hag | Or were you just being facetious? |
| 16:45.16 | [TK]D-Fender | Ch0Hag: **MOI*?!?! never |
| 16:45.18 | ManxPower | skymeyer: no. It is all handled in the zaptel configuration |
| 16:45.20 | blitzrage | Qwell[]: registertimeout in sip.conf is in milliseconds? |
| 16:45.24 | Qwell[] | dunno |
| 16:45.31 | blitzrage | ... |
| 16:45.32 | skyphyr | oh there is a skymeyer :$ |
| 16:45.48 | ManxPower | sorty. |
| 16:45.48 | Ch0Hag | blitzrage: "Give up" is not wise. A client wants me to setup Skype. If I can say "... and we can connect our VoIP service to Skype also" then I have more selling power. |
| 16:45.50 | skyphyr | hehe sorry ManxPower - thought it was a typo hehe |
| 16:46.03 | [TK]D-Fender | Ch0Hag: Skype the the bastard child of telephony und used only by little punks who think theya re cool and "pioneers" in VoIP |
| 16:46.13 | Ch0Hag | Otherwise another company is lost to the mire that is Skype. |
| 16:46.16 | *** part/#asterisk MedozasSVR (n=medozas@p54911D64.dip0.t-ipconnect.de) |
| 16:46.33 | emily_25 | ive just buyet $10 of skype! |
| 16:46.39 | emily_25 | bought? |
| 16:46.40 | emily_25 | xD |
| 16:46.49 | emily_25 | bye! |
| 16:47.16 | Ch0Hag | Rather interestingly their website says the download contains no spyware, adware or malware. |
| 16:47.17 | ManxPower | Ch0Hag: Well go do it. But nobody here knows how to do it. |
| 16:47.17 | ManxPower | skyphyr: since you said analog hardphones, I assumed you have zap cards. |
| 16:47.29 | *** part/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net) |
| 16:47.37 | Ch0Hag | To say that implies there's a suspicion otherwise, which surely is not good (for them). |
| 16:48.28 | skyphyr | sorry - voip hardphones - though I have an analogue hardphone with a zap card at home |
| 16:49.25 | ManxPower | skyphyr: you press read the manual for the phone to learn how to do transfers. |
| 16:49.34 | krondorl | Ok gonna ask another question which I expect to get a smart answer from [TK]D-Fender, Is there any plans of there being a pause/resume option for voicemail?? |
| 16:49.45 | skyphyr | the phone didn't come with a manual |
| 16:49.48 | skyphyr | honest |
| 16:50.04 | skyphyr | plus I was more concerned what I need to do on my asterisk server :-) |
| 16:50.08 | ManxPower | krondorl: not that I'm aware of, but of course, all that sort of stuff is in the bugrracker if it exists. |
| 16:50.15 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 16:50.22 | *** join/#asterisk yxa (n=lonari@bb116-14-3-116.singnet.com.sg) |
| 16:50.23 | [TK]D-Fender | krondorl: Check the bug-tracker. I doubt it, but I'm sure it'd be relatively easy to create |
| 16:50.27 | ManxPower | skyphyr: you don't have to do ANYTHING on your asterisk server |
| 16:51.04 | skyphyr | ok as long as I have dialplans in place it'll just allow it to work and conference/transfer etc as though they'd dialed the number directly |
| 16:51.19 | ManxPower | skyphyr: correct. |
| 16:51.36 | skyphyr | wow - very cool :-) thanks for the help |
| 16:51.51 | ManxPower | the transfer stuff in features.conf is, in my opinion, a terrible hack for phones that are too stupid to have a transfer feature. |
| 16:52.08 | ManxPower | it uses DTMF for transfers |
| 16:52.51 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 16:58.29 | blitzrage | Ch0Hag: well, you can't use Skype with Asterisk (at least not with more than 1 channel at a time) -- so saying, "give up", seems the best course of action (with regards to utilzing a real skype channel in Asterisk) |
| 16:59.34 | Ch0Hag | I'm not going to lose the money I could make converting a client to VoIP just because of minor difficulties. |
| 17:00.12 | Ch0Hag | The systems I've seen all seem to want a Skype client running in X, so a solution (if more than one channel is needed) is lots of virtual machines running lots of Skype clients. |
| 17:00.25 | Ch0Hag | It is a not insurmountable problem, and giving up is for losers. |
| 17:00.28 | Strom_M | Ch0Hag: you have that little professional integrity that you would rather sell a client a piece of unusable, unstable hackery? |
| 17:00.29 | Ch0Hag | Poor losers, usually. |
| 17:00.50 | rob0 | Hmmm, doing it right seems more important to me. |
| 17:01.03 | Juggie | doing it right would be not using skype |
| 17:01.11 | rob0 | zackly. |
| 17:01.20 | Ch0Hag | I have enough professional integrity to do everything I can to give the client what they need instead of what they want whilst still taking what they want into consideration. |
| 17:01.23 | blitzrage | Ch0Hag: sounds like a support nightmare which won't scale -- good luck |
| 17:01.42 | blitzrage | and I honestly mean that -- if you can get it to to work, awesome, but doesn't sound like a good usage of time and resources |
| 17:01.59 | Ch0Hag | Which is why what I would prefer is a full-on Skype channel in Asterisk proper. |
| 17:02.19 | Ch0Hag | If that is not possible, I will do what is necessary to support the client and, in turn, make money to support myself. |
| 17:02.19 | blitzrage | which you can't have |
| 17:02.32 | blitzrage | chan_skype does not exist, and probably never will |
| 17:03.00 | rob0 | If a client dictates the low-level details of a project, then they're not trusting you as a professional. |
| 17:05.03 | Juggie | there is only one that works decentaly at all |
| 17:05.06 | Juggie | and it doesnt pass dtmf |
| 17:06.18 | Ch0Hag | If it lets my client call their clients/suppliers/friends who have Skype accounts for free, then they will buy my VoIP system (or, rather, my time to install and support a VoIP system). |
| 17:06.26 | Ch0Hag | This is a Good Thing. |
| 17:07.19 | Juggie | oh yeah, and are you going to manage the mappings in the phone system |
| 17:07.23 | Juggie | for all the skype accounts |
| 17:07.37 | Juggie | so, superfruity1234@skype is extension 2001 |
| 17:08.58 | Juggie | that would be a management nightmare |
| 17:09.50 | Ch0Hag | It's a flat file with 2 or 3 fields. |
| 17:09.52 | Juggie | maybe you let them do it, ok that would be fine, except, would you rather your client goes through the trouble of setting up an extension, then picking up their phone, calling that extension, OR, just use skype when they need to, and then get the benefit of skype presense, and instant messaging, etc. |
| 17:09.58 | Ch0Hag | You wouldn't even need a full-blows rdbms. |
| 17:10.06 | Juggie | Ch0Hag, that you are going to let EVERY user edit |
| 17:10.20 | Juggie | or do you want an email every time soneone has a new skype friend to call |
| 17:10.22 | Ch0Hag | I am? |
| 17:10.30 | Ch0Hag | Oh that way? |
| 17:10.42 | Juggie | well its one way or the other |
| 17:10.47 | Juggie | someone has to input it |
| 17:10.49 | Ch0Hag | Nono, they'll just call <skype-userid>@skype.domain.com |
| 17:10.50 | Ch0Hag | Or something. |
| 17:11.05 | ManxPower | Ch0Hag: How do I dial that on my phone? |
| 17:11.15 | Juggie | yeah, that will be fun |
| 17:11.22 | Ch0Hag | I don't know about you, but I just type it in. |
| 17:11.48 | rob0 | My phone has a 12-key dialpad. |
| 17:11.57 | Ch0Hag | So does mine. |
| 17:12.04 | Ch0Hag | Have you never sent a text message? |
| 17:12.16 | rob0 | not from a desktop phone, no |
| 17:12.21 | Juggie | so, you REALLY think users will want to type in a sip user name via a alpha numeric keypad |
| 17:12.34 | rob0 | s/sip/skype/ |
| 17:12.44 | Juggie | yes. |
| 17:12.44 | Juggie | thanks rob. |
| 17:12.44 | ManxPower | so robert dobbs would be 7776662233777366622227777 |
| 17:13.05 | Ch0Hag | No. I think they'll type it into their computer-to-deskphone software applet. |
| 17:13.06 | rob0 | J.R. "Bob" Dobbs, to be precise ;) |
| 17:13.09 | ManxPower | Anyway, good luck with that. |
| 17:13.17 | Juggie | the sad thing about sip names is that no one actually uses them. |
| 17:13.34 | Ch0Hag | But they could type it into their phone if they want to. |
| 17:13.36 | Juggie | no one wants to type names in on their phone |
| 17:13.41 | [TK]D-Fender | Non VoIP question I could use a quick hand with. Anyone know of a free Windows Web filter so I can block off access to all but a few sites/IP's on a single client (local install). |
| 17:13.51 | Ch0Hag | They can also set up a quick-dial keys and the like if they want to. |
| 17:14.03 | Ch0Hag | [TK]D-Fender: Squid? |
| 17:14.14 | rob0 | haha I was going to say that too |
| 17:14.18 | Juggie | [TK], win xp firewall? |
| 17:14.24 | [TK]D-Fender | Ch0Hag: They have a windows build suitable for stand-alone installation? |
| 17:14.41 | rob0 | probably not |
| 17:14.42 | [TK]D-Fender | Juggie: Can I PW it and URL limit in/out? |
| 17:14.55 | Juggie | [TK]D-Fender, no. |
| 17:14.57 | Ch0Hag | I don't see why they wouldn't. |
| 17:15.14 | rob0 | But, I've set up a few squid transparent proxies. |
| 17:15.46 | Ch0Hag | http://www.squid-cache.org/Download/ -- ".. install a binary package of Squid, available for many different platforms including Windows." |
| 17:15.56 | rob0 | [TK]D-Fender: another idea which MIGHT work on windows: run a local nameserver |
| 17:16.16 | rob0 | just redirect the names you don't want them going to |
| 17:16.28 | Juggie | [tk], http://www.personalfirewall.comodo.com/overview.html?currency=USD®ion=North%20America&country=US |
| 17:16.42 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 17:16.59 | [TK]D-Fender | rob0: Nifty idea, but too random and I'd be going back all the time. |
| 17:18.48 | Juggie | [TK]D-Fender, comodo or zonealarm |
| 17:18.49 | Juggie | whichever |
| 17:19.37 | Juggie | comodo is probally better then zonealarm free |
| 17:21.10 | *** join/#asterisk inv_arp[work] (n=junya@c-71-229-122-61.hsd1.fl.comcast.net) |
| 17:23.05 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 17:26.05 | *** mode/#asterisk [-b sunsmasher!*@*] by Qwell[] |
| 17:26.17 | Qwell[] | somebody poke me if his client screws up again |
| 17:34.08 | blitzrage | sweet! |
| 17:34.18 | Qwell[] | :p |
| 17:34.52 | Strom_M | haha |
| 17:36.07 | *** join/#asterisk dennisharrison (n=dennisha@71-81-37-101.dhcp.slid.la.charter.com) |
| 17:38.43 | *** join/#asterisk perf3ktion (n=perf3kt@adsl-68-73-150-67.dsl.ipltin.ameritech.net) |
| 17:38.47 | perf3ktion | anyone know of good autodialers that interface with * |
| 17:39.01 | dennisharrison | hardware or software? |
| 17:39.05 | perf3ktion | not even good, just one at all |
| 17:39.06 | perf3ktion | software |
| 17:39.19 | dennisharrison | I have read about some |
| 17:39.23 | dennisharrison | haven't used any personally |
| 17:39.31 | dennisharrison | did you check voip-info ? |
| 17:39.35 | Qwell[] | They have some at telemarketer-spam-tools.com |
| 17:39.36 | perf3ktion | just point, I dont' need hand holding |
| 17:39.57 | perf3ktion | is that a joke qwell? |
| 17:40.02 | Qwell[] | Yes. |
| 17:40.07 | Strom_M | Qwell[]: I thought it was go-die-in-a-fire.com |
| 17:40.14 | Qwell[] | Strom_M: CNAM |
| 17:40.36 | perf3ktion | hahaha |
| 17:40.48 | perf3ktion | why did I still put it in the browser, lol |
| 17:41.02 | Mercestes | In the famous words of the Protoss Dragoons. "I have returned." |
| 17:41.42 | perf3ktion | dennisharrison: yeah looked at the voip-info |
| 17:42.15 | *** join/#asterisk stmaher_ (n=stmaher@mail.voxpilot.com) |
| 17:42.34 | stmaher_ | Hello everyone |
| 17:42.47 | dennisharrison | hello stmaher_ :) |
| 17:43.18 | stmaher_ | Im new to asterisk.. Just wondering you could answer a few questions? |
| 17:43.38 | stmaher_ | I have the reilly book downloaded.. but would really like to get up and running to play with it.. |
| 17:43.51 | stmaher_ | I have an IVR and a sip phone.. |
| 17:43.59 | stmaher_ | I need to use asterisk in the middle.. |
| 17:44.10 | dennisharrison | stmaher_, you want to do sip to sip? |
| 17:44.16 | stmaher_ | Yes please :-) |
| 17:44.26 | dennisharrison | what is your sticking point right now? |
| 17:44.38 | dennisharrison | like, how much do you know about your current problem? :) |
| 17:44.49 | stmaher_ | Incoming calls work.. |
| 17:45.04 | stmaher_ | I have the hello world example played when a sip call is made into the asterisk box |
| 17:45.10 | dennisharrison | k |
| 17:45.11 | perf3ktion | are you behind nat |
| 17:45.21 | stmaher_ | but I dont know how to handle it to move it to the other sip phone |
| 17:45.24 | stmaher_ | no.. |
| 17:45.26 | stmaher_ | Office LAN |
| 17:45.43 | perf3ktion | you just mean sip to sip |
| 17:45.49 | stmaher_ | All the examples on the net seem to be for pots and pstn |
| 17:45.54 | perf3ktion | both extension are setup and registered? |
| 17:46.05 | stmaher_ | Im trying to do it without registering |
| 17:46.22 | Strom_M | why? |
| 17:46.30 | perf3ktion | well register in the meaning of the sip device knowing what extension to connect to |
| 17:46.46 | stmaher_ | Oh right.. i played around with the sip.conf file.. |
| 17:46.46 | Strom_M | uh, no |
| 17:46.48 | Ch0Hag | Can you get TT to do the equivalent of a print STDERR? |
| 17:46.52 | Strom_M | that's not what a registration is |
| 17:46.59 | Ch0Hag | Oops wrong # |
| 17:47.00 | stmaher_ | and tried adding it in there.. |
| 17:50.45 | *** join/#asterisk koosroos (n=koosroos@c1-251-1.rrba.isadsl.co.za) |
| 17:51.38 | koosroos | to make your extension.cong file can one set spaces between diffrent secments of ones dialplan? |
| 17:51.40 | stmaher_ | do you have to register it in order to move calls to it |
| 17:52.08 | Strom_M | stmaher_: "place calls", not "move calls" |
| 17:52.21 | Strom_M | and your desk sets generally should be registering with your PBX, yes |
| 17:52.31 | Ch0Hag | koosroos: Whitespace is largely ignored except in the definition lines themselves. |
| 17:52.58 | koosroos | what do you mean with definition lines |
| 17:52.59 | koosroos | ? |
| 17:53.12 | stmaher_ | I dont understand the need to register... If I make a call it .. can i route it directly to another ip address? |
| 17:53.14 | Ch0Hag | ie. 'exten => 100,1,NoOp' is not the same as 'exten => 100, 1, NoOp' but is identical to ' exten => 100,1,NoOp' |
| 17:53.27 | *** join/#asterisk rstob911 (n=russ@d4-187.rb2.clm.centurytel.net) |
| 17:53.36 | Ch0Hag | Blank lines are irrelevent to the best of my knowledge. |
| 17:53.45 | Ch0Hag | I certainly use them liberally. |
| 17:53.53 | krondorl | Qwell: What svn version is the pause/resume in?? |
| 17:54.09 | koosroos | k i see, do you ming=d looking at my dialplan to ee if there might be any problems |
| 17:54.11 | Qwell[] | krondorl: don't know when it got put in, but it's in trunk though |
| 17:54.25 | Qwell[] | You could try doing an `svn blame` on app_voicemail.c, and see if you can find out what rev |
| 17:55.20 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-97-164.ph.ph.cox.net) |
| 17:58.17 | koosroos | just mentioned to past my dialplan for a lookover and everybody went silent |
| 17:58.40 | *** part/#asterisk rstob911 (n=russ@d4-187.rb2.clm.centurytel.net) |
| 17:59.09 | *** join/#asterisk x86_ (n=bryce@p3m/member/x86) |
| 18:00.10 | x86_ | how do i do math in the dialplan? for example, I have 4 digit "account codes" that people dial before a number, but if they dial 1-NPA-NXX-XXXX, I need to strip the 1 before dialing the call out to the PSTN |
| 18:00.20 | x86_ | I have the account code length in a variable |
| 18:00.54 | Strom_M | koosroos: are you having a specific issue with your dialplan:? |
| 18:01.06 | Strom_M | x86_: you want substrings, not math |
| 18:01.07 | x86_ | I want something like Dial(Zap/g3|${EXTEN:${ACCOUNTCODE}+1}}) |
| 18:01.17 | Strom_M | no |
| 18:01.18 | koosroos | no just need to know if i will have hassels in the feature |
| 18:01.26 | x86_ | Strom_M: still need math, as I have to add 1 to $ACCOUNTCODE |
| 18:01.32 | *** join/#asterisk rstob911 (n=russ@d4-187.rb2.clm.centurytel.net) |
| 18:01.36 | x86_ | substring or not :) |
| 18:01.52 | *** join/#asterisk casimir (n=casimir@rrcs-71-43-154-55.se.biz.rr.com) |
| 18:02.10 | Strom_M | so you if they dial 5555-311-555-2368, you want the circuit to be sent 1-311-555-2368? |
| 18:02.12 | x86_ | wait, I could start at the end, and take the next 10 digits to the left |
| 18:02.12 | *** part/#asterisk rstob911 (n=russ@d4-187.rb2.clm.centurytel.net) |
| 18:02.21 | *** join/#asterisk rstob911 (n=russ@d4-187.rb2.clm.centurytel.net) |
| 18:02.24 | polerin | meh. small phone switch, is it worth it to learn AstDB, or just skip to the *SQL's? |
| 18:02.35 | polerin | (given that i'm already familiar with SQL) |
| 18:02.38 | koosroos | ${EXTEN:1:10} x86 |
| 18:02.41 | x86_ | Strom_M: no, they dial 5555-1-222-333-4444, and I need to send the circuit 222-333-4444 |
| 18:02.47 | [TK]D-Fender | polerin: To do WHAT exactly? |
| 18:03.03 | Strom_M | koosroos: ugh, well i suppose so |
| 18:03.04 | polerin | mostly user storage, possible autodialing stuff |
| 18:03.09 | x86_ | koosroos: wont that strip the first digit off only? |
| 18:03.15 | *** join/#asterisk devnet (n=ddevine@rdu-nat.rpath.com) |
| 18:03.18 | *** part/#asterisk devnet (n=ddevine@rdu-nat.rpath.com) |
| 18:03.22 | koosroos | yes |
| 18:03.23 | Juggie | x86, check out exten |
| 18:03.24 | polerin | though the autodialing wouldn't be soon, I'd need to set that up seperately :P |
| 18:03.25 | *** join/#asterisk devnet (n=ddevine@rdu-nat.rpath.com) |
| 18:03.31 | x86_ | koosroos: not what i want then ;) |
| 18:03.32 | *** part/#asterisk devnet (n=ddevine@rdu-nat.rpath.com) |
| 18:03.36 | Juggie | to strip digits |
| 18:03.41 | [TK]D-Fender | x86_: ${EXTEN:4} <- whoopie shit :) |
| 18:03.47 | x86_ | koosroos: I want a variable length $ACCOUNTCODE, plus an additional 1 |
| 18:03.47 | koosroos | what do you need then? x86 |
| 18:03.59 | x86_ | [TK]D-Fender: that 4 is variable |
| 18:04.02 | polerin | actually I think I answered my own question, the info for the autodial would likely be in a sql db anyway so.. |
| 18:04.05 | polerin | yeah |
| 18:04.06 | koosroos | explain |
| 18:04.12 | x86_ | [TK]D-Fender: it may change to 5 some day, and I dont want the dialplan to break |
| 18:04.12 | koosroos | explain with sample |
| 18:04.14 | *** join/#asterisk devnet (n=ddevine@rdu-nat.rpath.com) |
| 18:04.20 | [TK]D-Fender | x86_: "show function CUT" <------------- |
| 18:04.39 | Juggie | x86, if it may change to 5 then plan for 5 now. |
| 18:04.40 | *** part/#asterisk devnet (n=ddevine@rdu-nat.rpath.com) |
| 18:04.45 | x86_ | there's no way to do some simple math? |
| 18:04.49 | Juggie | do, 05555-1-111-1111-1111 |
| 18:04.54 | [TK]D-Fender | x86_: SEE ABOVE |
| 18:04.57 | x86_ | Juggie: it may change to 6 some time |
| 18:05.15 | Juggie | x86, what are you doing |
| 18:05.17 | polerin | jeebus, what are you needing that long of a dial for? |
| 18:05.18 | x86_ | <PROTECTED> |
| 18:05.18 | x86_ | No function by that name registered. |
| 18:05.22 | krondorl | qwell[] Since I'm running 1.2.13 I can't find it there, so I was wondering how far up the line it was from where I am now. But as you don't know off the top of your head I see if I can find it. |
| 18:05.22 | Juggie | because what you are explaning to me makes no sense |
| 18:05.23 | Ch0Hag | You can do ${EXTEN:${VAR1}:${VAR2}} |
| 18:05.29 | skyphyr | anyone here used the c460 gigaset? |
| 18:05.39 | x86_ | Ch0Hag: that doesn't solve the problem :) |
| 18:05.56 | Juggie | man, google bought grandcentral for 50million :( |
| 18:06.00 | Juggie | that breaks my heart. |
| 18:06.07 | *** part/#asterisk rstob911 (n=russ@d4-187.rb2.clm.centurytel.net) |
| 18:06.08 | [TK]D-Fender | x86_: Pay attention to UPPERCASE. |
| 18:06.15 | [TK]D-Fender | x86_: "show function CUT" <------------- |
| 18:06.27 | Juggie | x86_, explain your system to me |
| 18:06.28 | polerin | functions are all uppercase :) |
| 18:06.29 | x86_ | show function CUT |
| 18:06.29 | x86_ | No function by that name registered. |
| 18:06.38 | Juggie | [TK]D-Fender, isnt CUT removed in 1.4 |
| 18:06.46 | x86_ | Juggie: i'm running 1.2 still |
| 18:06.47 | [TK]D-Fender | Juggie: is it? |
| 18:06.47 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 18:06.48 | Qwell[] | krondorl: It won't be in 1.2 at all |
| 18:06.57 | dc3aes | Juggie.. after reading some conspiracy theory stuff about Google/CIA and owning all services where they can funnel personal communications, archive desktops, email, etc.. i actually start "worrying" about all the stuff google gobbles up lol |
| 18:06.59 | x86_ | Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-d |
| 18:07.08 | krondorl | qwell[] ok, thanks.. |
| 18:07.15 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 18:07.16 | Juggie | x86_, can you explain WHAT you want to do |
| 18:07.24 | sahafeez | dumb questions - what file is it that the name of the smtp server is in. i need to change it and set it up so long ago i forget where |
| 18:07.27 | Juggie | WHY you need 4,5,6 digits to preempt a number |
| 18:07.57 | x86_ | Juggie: call center account codes |
| 18:08.01 | [TK]D-Fender | x86_: http://www.pastebin.ca/601744 |
| 18:08.10 | [TK]D-Fender | x86_: from my work 1.2 server |
| 18:08.13 | Juggie | x86_, call center account codes? |
| 18:08.24 | Juggie | you mean for the agents to make outbound calls? |
| 18:08.40 | x86_ | [TK]D-Fender: that's nice, but there is no standard delimiter |
| 18:08.44 | koosroos | can anyone help lpease need to set up a call pickup group i have 8 sip phones with 4 fxo ports i am routing calls to a common queue where the it will ring cyclic. but i need to pick up the call if i want to |
| 18:08.46 | x86_ | Juggie: right |
| 18:08.46 | [TK]D-Fender | x86_: Appears in my home 1.4.2 system as well |
| 18:08.56 | x86_ | [TK]D-Fender: wont solve the problem anyway |
| 18:08.59 | Juggie | x86_, why dont you just use the agent thats logged into the phone. |
| 18:09.05 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 18:09.06 | Juggie | or aer you not using asterisk agents |
| 18:09.09 | x86_ | Juggie: they dont log in |
| 18:09.12 | *** join/#asterisk guillote_GNU (n=guillote@190.7.23.224) |
| 18:09.28 | x86_ | Juggie: plus, they can make 10 different calls, and use 10 different codes, depending on what they are working on |
| 18:09.38 | x86_ | Juggie: and 5 people may make a call with the same code at once |
| 18:09.39 | [TK]D-Fender | x86_: {EXTEN:${LEN(ACCOUNTCODE)}) ? |
| 18:09.47 | Ch0Hag | x86_: There *is* a MATH function, but it sounds like you want something else. |
| 18:09.53 | x86_ | [TK]D-Fender: that looks interesting |
| 18:10.19 | x86_ | [TK]D-Fender: now, how do i add 1 to it? :) |
| 18:10.22 | koosroos | does this apply to pickupgrp or callgroup in my sip.conf? |
| 18:10.26 | Juggie | x86_, ic, well... my suggestion is to make it big enough so you dont have to keep changing |
| 18:10.55 | [TK]D-Fender | x86_: {EXTEN:$[${LEN(ACCOUNTCODE)} + 1]} ? |
| 18:11.02 | x86_ | [TK]D-Fender: THANKS |
| 18:11.11 | x86_ | that's all I've been wanting the whole time ;) |
| 18:11.15 | *** join/#asterisk fiber0pti (i=fiber0pt@216.31.101.41) |
| 18:11.30 | [TK]D-Fender | x86_: READ! |
| 18:11.31 | Juggie | i still dont get how that is going to work |
| 18:11.34 | blitzrage | yay! Copy edit review done! |
| 18:11.47 | Strom_M | yay! |
| 18:11.54 | Ch0Hag | In fact one of my testing extensions reads exten => 503,4,Set(ODBC_GETC('b')='$[${ODBC_GETC('b')} + 1]'), so I don't think the function is even required. |
| 18:12.00 | Juggie | if each agent has multiple account codes based on what they are working on how is ACCOUNTCODE going to get filled in |
| 18:12.02 | Strom_M | blitzrage: gonna celebrate with fireworks? |
| 18:12.02 | blitzrage | FYI: TFoT2 should be released sometime in August |
| 18:12.04 | Ch0Hag | (I *knew* I had some maths in it somewhere) |
| 18:12.06 | koosroos | can anyone help lpease need to set up a call pickup group i have 8 sip phones with 4 fxo ports i am routing calls to a common queue where the it will ring cyclic. but i need to pick up the call if i want to |
| 18:12.08 | blitzrage | Strom_M: unfortunately not |
| 18:12.16 | koosroos | does this apply to pickupgrp or callgroup in my sip.conf? |
| 18:12.21 | Strom_M | blitzrage: it was a joke, silly :) |
| 18:12.28 | x86_ | [TK]D-Fender: read what? that was mentioned before? |
| 18:12.31 | blitzrage | Strom_M: I have no sense of humour |
| 18:12.36 | x86_ | [TK]D-Fender: i must have missed it because no one said it ;) |
| 18:12.50 | [TK]D-Fender | x86_: This is EXPRESSIONS 101 + VARIABLES 101 |
| 18:13.06 | [TK]D-Fender | x86_: 2 things which the WIKI pages coive quite WELL |
| 18:13.10 | [TK]D-Fender | cover* |
| 18:13.17 | x86_ | [TK]D-Fender: notice how my original question was alluding to "where can i find EXPRESSIONS 101?" |
| 18:13.21 | x86_ | ;) |
| 18:13.25 | Strom_M | google.com |
| 18:13.45 | x86_ | 13:01 < Strom_M> x86_: you want substrings, not math |
| 18:13.49 | [TK]D-Fender | x86_: I only started paying attention as your whining turned to braying ;) |
| 18:14.04 | Strom_M | x86_: because your original question was unclear and I had to guess |
| 18:14.15 | x86_ | [TK]D-Fender: k, next time scroll up before <troll_mode> please ;) |
| 18:14.17 | [TK]D-Fender | x86_: http://www.voip-info.org/wiki/view/Asterisk+Expressions <- yeah... you looked REALLY ahrd ;) |
| 18:14.30 | x86_ | yes, i did look ahrd ;) |
| 18:14.34 | [TK]D-Fender | ;) |
| 18:14.47 | x86_ | i was googling asterisk math |
| 18:14.48 | [TK]D-Fender | whadt adis askjkjd a? |
| 18:14.51 | x86_ | and didn't find much :P |
| 18:14.58 | x86_ | heh |
| 18:15.01 | x86_ | thanks man |
| 18:15.13 | *** join/#asterisk kesiev[ita] (n=kes@host234-117-dynamic.5-87-r.retail.telecomitalia.it) |
| 18:15.14 | [TK]D-Fender | x86_: Nice & functional? |
| 18:15.34 | x86_ | can't test it yet, took my channel bank to the other site |
| 18:15.44 | x86_ | my IP phone is teh broke too :P |
| 18:16.06 | [TK]D-Fender | x86_: Soft-phone <- |
| 18:16.21 | kesiev[ita] | Hi all. I've a problem with the dial command and a Sangoma card. |
| 18:16.23 | x86_ | yeah, true |
| 18:16.44 | x86_ | [TK]D-Fender: for testing stuff like dialplan, no biggie, but for stuff that requires speakers and/or microphone, I'm kinda screwed ;) |
| 18:18.08 | [TK]D-Fender | x86_: Better stock up on KY, because you're looking to get burned... |
| 18:18.29 | [TK]D-Fender | kesiev[ita]: Details + Pastebin <- |
| 18:18.31 | [TK]D-Fender | ~pb |
| 18:18.31 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 18:18.33 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
| 18:19.57 | kesiev[ita] | Everything is working. No PB needed, i think. I need to dial a number and pass the dialing/calling tones to the caller. Tried the "t" command but seems to generate a "fake ringing tone". |
| 18:19.58 | x86_ | [TK]D-Fender: rofl |
| 18:20.01 | x86_ | [TK]D-Fender: sicko |
| 18:20.09 | *** join/#asterisk saftsack (n=saftsack@pD9E0605E.dip.t-dialin.net) |
| 18:20.50 | Strom_M | kesiev[ita]: don't use the "r" flag when you execute Dial() |
| 18:21.09 | [TK]D-Fender | kesiev[ita]: Thats not how you pass tones to an established call. Go read Dials instructions : "show application dial" |
| 18:21.10 | Juggie | sometimes trying to help people in here is like talking to a wall |
| 18:21.25 | Juggie | even when i answer, or ask follow up questions |
| 18:21.27 | Juggie | no one listens |
| 18:21.31 | [TK]D-Fender | Strom_M: And that's not what he's asking about |
| 18:21.40 | Strom_M | jizzmop |
| 18:21.51 | [TK]D-Fender | Juggie: I'm sorry... were you saying something? ;) |
| 18:22.18 | Strom_M | kesiev[ita]: pastebin your dialplan |
| 18:22.53 | [TK]D-Fender | Strom_M: He jsut wants to pass DTMF on an outbound call. thats a simple DIAL option he just needs to go READ. |
| 18:22.54 | Jingles | I've got a great question. |
| 18:23.05 | [TK]D-Fender | Strom_M: Nothing to bother pastebinning. |
| 18:23.25 | Jingles | So far, I haven't had a lot of luck getting my SIP providers to pass ANI so that I can manage the callerID that is going out to folks I'm calling. |
| 18:23.51 | [TK]D-Fender | Jingles: Find a better one. |
| 18:24.01 | Strom_M | [TK]D-Fender: he didn't ask about DTMF. he asled about call progress tones |
| 18:24.14 | Jingles | Over the PRI, I can control the clid, since I'm signaling the TDMA equipment at the phone company - but the SIP providers don't seem to want to pass callerID. |
| 18:24.18 | *** join/#asterisk dillydally (n=email@58.69.243.203) |
| 18:24.27 | Strom_M | Jingles: caller ID is not ANI |
| 18:24.28 | kesiev[ita] | if I toggle the "r" flag, I hear the "free line" tone for two seconds, then it starts with the "dialing" tone. |
| 18:24.30 | [TK]D-Fender | Strom_M: Ok, could be interpreted that way. |
| 18:24.47 | kesiev[ita] | Storm_M: I'm using Trixbox 1.2.3 |
| 18:24.53 | [TK]D-Fender | ~trixbox |
| 18:24.54 | jbot | well, trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 18:24.56 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 18:24.59 | Jingles | Strom_M : I realize. But when Asterisk sets 'callerID', it signals (via the PRI) the TDMA equipment to provide ANI |
| 18:25.07 | Strom_M | uh no |
| 18:25.09 | *** join/#asterisk bobdraken (n=bobdrakk@75-138-105-182.dhcp.hckr.nc.charter.com) |
| 18:25.10 | bobdraken | yes finaly |
| 18:25.14 | Jingles | poop. |
| 18:25.15 | bobdraken | i have some questions |
| 18:25.16 | Strom_M | TDMA is a mobile phone technology |
| 18:25.24 | bobdraken | i have a spare old old box |
| 18:25.29 | Strom_M | you're confusing it with TDM |
| 18:25.41 | bobdraken | im wondering if it will run asterisk.. i have ubuntu on it.. it has 256 mem and a 6 gig hdd |
| 18:25.45 | Jingles | ok. let me look up my terms real quick. |
| 18:25.50 | Strom_M | which is merely what ISDN PRI sits on top of :) |
| 18:25.54 | Jingles | boo. gotcha. |
| 18:26.02 | kesiev[ita] | as jbot_ is saying, I have to ask to their forum. Well, thank you guys. :) |
| 18:26.12 | Jingles | anyhow. the PRI can control caller ID |
| 18:26.24 | Strom_M | and caller ID on the PSTN is /not/ the same as ANI |
| 18:26.35 | *** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 18:26.44 | Strom_M | caller ID is handled in the 'calling party number' field of the SS7 IAM |
| 18:26.56 | Strom_M | ANI is the 'charge number' field of the SS7 IAM |
| 18:26.58 | bobdraken | anyone? do you think asterisk will run on a box that small |
| 18:27.01 | [TK]D-Fender | bobdraken: Probably fine depending on how much load you'll ask of *. |
| 18:27.11 | [TK]D-Fender | bobdraken: what CPU? |
| 18:27.12 | bobdraken | im wodering |
| 18:27.21 | bobdraken | can i set it up and hook it up to my home line |
| 18:27.32 | bobdraken | to call my home phone with. .im must wanting to play with it before i set up a voip line |
| 18:27.34 | Jingles | Strom_M : ok - I'm reading as fast as I can. |
| 18:27.51 | [TK]D-Fender | bobdraken: Sure, for basic home use you're pretty much garaunteed a 100% yes for your setup |
| 18:28.07 | bobdraken | how do i hook it up... ethernet? |
| 18:28.19 | bobdraken | do i need a ethernet 2 reg phone line adapter |
| 18:28.25 | [TK]D-Fender | bobdraken: If you're thinkiing about going VoIP soon after, then forget your landline altogether. |
| 18:28.34 | bobdraken | well i mean |
| 18:28.42 | bobdraken | what voip service u use... |
| 18:28.46 | [TK]D-Fender | bobdraken: Need, no, but its a good way to have a hardware phone for your calling. |
| 18:28.47 | bobdraken | nufone |
| 18:29.02 | [TK]D-Fender | bobdraken: there are MANY options out there |
| 18:29.10 | Jingles | Strom_M : Ok - since I can control the PRI, I'll ask my questions there first. |
| 18:29.11 | [TK]D-Fender | bobdraken: My circumstances are special. |
| 18:29.17 | bobdraken | special how so |
| 18:29.28 | bobdraken | [TK]D-Fender: how so |
| 18:29.49 | [TK]D-Fender | bobdraken: I connect through my work's PRI. |
| 18:29.58 | [TK]D-Fender | bobdraken: They are my ITSP effectively. |
| 18:30.04 | bobdraken | cool... |
| 18:30.21 | bobdraken | so i can run asterisk from ubuntu |
| 18:30.25 | [TK]D-Fender | bobdraken: And as I'm an * consultant I'm also linked to numerous other systems world-wide |
| 18:30.29 | [TK]D-Fender | bobdraken: Sure |
| 18:30.34 | *** join/#asterisk twitchnln (n=Administ@70.43.112.117.nw.nuvox.net) |
| 18:30.38 | bobdraken | ok ill d/l it and install |
| 18:30.39 | [TK]D-Fender | bobdraken: Plenty of guides on setting it up under Ubuntu |
| 18:30.41 | twitchnln | greetings everyone |
| 18:30.49 | Hmmhesays | Aren't we all [TK]D-Fender? |
| 18:30.51 | bobdraken | can u link me to one that u would recomend using |
| 18:31.06 | [TK]D-Fender | bobdraken: Before you even worry about that, jsut DL *, and THE BOOK |
| 18:31.08 | [TK]D-Fender | ~book |
| 18:31.08 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:31.25 | bobdraken | [TK]D-Fender: what? |
| 18:31.33 | [TK]D-Fender | bobdraken: And get your hands dirty with * and learn how it works for free and see if it suits your needs & wants |
| 18:31.34 | bobdraken | just d/l the asterisk book |
| 18:31.35 | *** join/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 18:31.45 | bobdraken | how can i do that... set it up on my home line? |
| 18:31.47 | bobdraken | or something |
| 18:31.54 | bobdraken | or hook it up to a router and diff phones |
| 18:32.04 | Jingles | crud. I don't have my T1 book here. |
| 18:32.10 | MooingLemur | is there a way to bump up the number of possible saved voicemail messages from 99? |
| 18:32.12 | polerin | everything from FWD and softphones to several other things |
| 18:32.14 | twitchnln | i have a couple of mitel 5224 dual mode phones that i have setup as extensions, does anyone know if there is any way to set these phones not to require me to hit the dial button to send calls? |
| 18:32.15 | [TK]D-Fender | bobdraken: That requires a hardware purchase of a compatable TDM card or gateway. |
| 18:32.20 | Qwell[] | MooingLemur: voicemail.conf, maxmsgs |
| 18:32.28 | bobdraken | walmart sell those.. and how much |
| 18:32.30 | [TK]D-Fender | bobdraken: And if you are considering dumping your analog line I'd say save your $ |
| 18:32.34 | bobdraken | yeah |
| 18:32.38 | bobdraken | so how can i play with it |
| 18:32.39 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 18:32.44 | MooingLemur | ahh, k. It wasn't in there (commented) |
| 18:32.45 | MooingLemur | thanks |
| 18:32.48 | [TK]D-Fender | bobdraken: SIP soft-phone is the typical way |
| 18:32.50 | Jingles | but ok, I can see that the 11 B and 1 D channels that the PRI uses - and the D channel is how the phone company gets the signal |
| 18:32.54 | polerin | voip ftw :) |
| 18:32.58 | bobdraken | i have a xact voip phone |
| 18:33.01 | [TK]D-Fender | bobdraken: www.xten.com <- go DL X-Lite. |
| 18:33.01 | Jingles | (signal for call info) |
| 18:33.08 | bobdraken | can i hook that up with no voip service |
| 18:33.15 | bobdraken | and call internally inside my house and play that way |
| 18:33.18 | [TK]D-Fender | bobdraken: or Idefisk from www.asteriskguru.com |
| 18:33.30 | polerin | yeah |
| 18:33.33 | [TK]D-Fender | bobdraken: You will hook it up to * . |
| 18:33.33 | polerin | you can |
| 18:33.38 | bobdraken | yeah \ |
| 18:33.41 | bobdraken | kk ill do that |
| 18:33.47 | bobdraken | call the kitchen or something |
| 18:33.47 | bobdraken | lol |
| 18:33.50 | [TK]D-Fender | bobdraken: * will be the one handling all of your calls and features, etc. |
| 18:33.53 | bobdraken | kitchen line 3 |
| 18:33.59 | bobdraken | lol |
| 18:34.00 | [TK]D-Fender | bobdraken: something like that |
| 18:34.04 | Jingles | Strom_M : So, you're saying that when Asterisk makes a call, and 'sets callerID' during the dialout over the PRI, that it's not signalling the hardware at the far end to set the ANI? |
| 18:34.22 | bobdraken | ok so ill install it now |
| 18:34.24 | bobdraken | brb |
| 18:35.40 | [TK]D-Fender | bobdraken: Go google up some Ubuntu specific guides as well |
| 18:35.40 | koosroos | is it nesesary to end off all commands with the hangup() command? |
| 18:35.50 | bobdraken | would you recomend i install asterisk |
| 18:35.51 | bobdraken | or trixbox |
| 18:35.56 | bobdraken | is that the same thing? |
| 18:35.59 | bobdraken | just diff package? |
| 18:36.02 | [TK]D-Fender | bobdraken: Asterisk. Trixbox will not teach you anything. |
| 18:36.09 | bobdraken | kk |
| 18:36.10 | Hmmhesays | trixbox is kind of like asterisknow |
| 18:36.11 | *** join/#asterisk wyoming (n=steve_mu@216.166.159.235) |
| 18:36.11 | bobdraken | thanks in advance |
| 18:36.13 | [TK]D-Fender | ~trixbox |
| 18:36.14 | jbot | i heard trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 18:36.15 | Hmmhesays | got a webgui |
| 18:36.16 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 18:36.39 | [TK]D-Fender | bobdraken: Its lets you setup an * config in a terribly cookie-cutter way. |
| 18:36.58 | bobdraken | can i burn the asterisk to cd and install it like that on the other box |
| 18:37.00 | bobdraken | will it fit |
| 18:37.26 | krondorl | YAY!!!! I did it!!!! Finally!!! |
| 18:37.32 | [TK]D-Fender | bobdraken: * is just a bunch of Tarballs. Install a normal distro and install * after |
| 18:37.52 | bobdraken | i have ubuntu already installed on the box |
| 18:38.27 | [TK]D-Fender | bobdraken: Fine, so go install *'s prerequisite packages and get to it! |
| 18:38.54 | bobdraken | lol ok i have to turn on the other box and stuff |
| 18:38.56 | bobdraken | give me a few |
| 18:39.11 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 18:40.54 | krondorl | And all with about 20 lines of dialplan coding. |
| 18:41.01 | Hmmhesays | file. |
| 18:44.21 | Hmmhesays | I need a list of north american area codes |
| 18:44.31 | Hmmhesays | all the that all use 1 as the country code |
| 18:45.48 | Mercestes | Hmmhesays, http://www.thedirectory.org/pref/ |
| 18:46.29 | Mercestes | >.> Sorry, wrong nick. |
| 18:48.40 | koosroos | is it nesesary to end off all commands with the hangup() command? |
| 18:49.07 | Mercestes | koosroos, No. |
| 18:49.55 | Jingles | boo. |
| 18:50.16 | Jingles | I sorted out that SetCallerID() has an ANI flag. Setting it doesn't seem to help my situation. |
| 18:50.33 | Jingles | I still get the SIP provider's callerID. |
| 18:52.30 | agile | anyone here ever used supernettel.com ? |
| 18:54.09 | *** join/#asterisk zeeesh (n=aadilism@202.125.143.67) |
| 18:54.19 | Hmmhesays | Mercestes: thanks, now I just have to format the data |
| 18:54.42 | Mercestes | that's what secretaries are for. |
| 18:54.46 | Mercestes | that reminds me. |
| 18:54.49 | Mercestes | ~seen b11d |
| 18:54.52 | jbot | b11d <n=no@234-200-29-134.hcc.mnscu.edu> was last seen on IRC in channel #asterisk, 26d 3h 44m 46s ago, saying: 'if it's possible to harvest uptime info from polycom phones via snmp or something like that?'. |
| 18:55.38 | *** join/#asterisk torch (n=logan@200.99.7.98) |
| 18:55.42 | torch | hi guys ... |
| 18:55.54 | krondorl | Hi torch |
| 18:55.55 | Mercestes | Greetings, earthling |
| 18:55.56 | *** join/#asterisk mindCrime_ (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 18:55.59 | torch | .I've learned how to add a prefix to a trunk ... but now I need to match a number ..for example 50 ..but now I need to add a suffix ..let's say 90 ... If I do 50+90 ... the asterisk will try to match 90 and add 50 as prefix ...any ideas ? |
| 18:56.52 | Mercestes | torch: ... |
| 18:57.01 | torch | hey mercestes ... :-) |
| 18:57.10 | Mercestes | If you do 50 + 90 why would you need to add 50 as a prefix? |
| 18:57.39 | Mercestes | and how would you tell 50 + 90 from 70 + 70? |
| 18:57.41 | [TK]D-Fender | Set(imavarwithstuffaddedtobothendsofanothervar=123${someothervar}456) |
| 18:57.44 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 18:57.45 | torch | nope ... if I dail 50 .. I need to add 90 meaning ...the trunk will dial 5090 |
| 18:58.03 | [TK]D-Fender | torch: show how you're calling it now. |
| 18:58.20 | torch | k .. I have a outbound route matching 50 |
| 18:58.27 | Mercestes | torch: Dial(zap/g1/${EXTEN}90) |
| 18:59.04 | torch | right Mercestes ...right now I'm using freepbx to configure everything ... do you know how to do that in there? |
| 18:59.10 | Mercestes | oh! |
| 18:59.13 | Mercestes | I know the answer to that!!! |
| 18:59.15 | Mercestes | ~freepbx |
| 18:59.16 | jbot | methinks freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:59.32 | torch | well ..I'm there .. but no luck so far :-) |
| 18:59.40 | Mercestes | I could have guessed that. ;) |
| 18:59.45 | torch | ;-) |
| 18:59.47 | [TK]D-Fender | torch: Get rid of that flaming pile of shit and you should know better than to ask that here. |
| 18:59.51 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:59.51 | Mercestes | freepbx== "asterisk? what's asterisk?" |
| 19:00.09 | Strom_M | Jingles: yes, that's exactly what I'm saying |
| 19:00.09 | russellb | that makes me sad |
| 19:00.36 | torch | TK .. can't do that right now ... to newbie for that .. |
| 19:00.37 | Mercestes | freepbx == "Voip? oh..you mean vonage, right? Man, what a typo." |
| 19:00.41 | *** join/#asterisk ManxPower (n=manxpowe@015-817-840.area5.spcsdns.net) |
| 19:01.38 | Strom_M | Jingles: as an end user, there is absolutely no way for you to manipulate the ANI of the call |
| 19:01.49 | *** join/#asterisk bobdraken (n=haxxor@75-138-105-182.dhcp.hckr.nc.charter.com) |
| 19:01.59 | bobdraken | im back sorry it took so long ok i have the thing d/l but |
| 19:02.02 | bobdraken | how do i install it |
| 19:02.13 | Mercestes | run setup.exe |
| 19:02.35 | torch | ok Mercestes .. let's say I do forget freepbx ... where should I put the Dial(zap/g1/${EXTEN}90) ? |
| 19:02.44 | bobdraken | still here? |
| 19:02.46 | Mercestes | torch: In your outbound context. |
| 19:02.54 | martin[ug] | Mercestes: don't forget to attach a /silent :) |
| 19:03.00 | Mercestes | martin[ug], ;) |
| 19:03.28 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-b7e73d00a3159bae) |
| 19:03.38 | torch | thx Mercestes .. let me do that ;-) |
| 19:03.39 | torch | brb |
| 19:03.44 | Mercestes | Good luck. |
| 19:03.59 | Mercestes | bobdraken, google wiki asterisk install. |
| 19:04.07 | Mercestes | bobdraken: btw...you *are* referring to asterisk, right? |
| 19:04.14 | Mercestes | that may be a silly assumption on my part. |
| 19:04.34 | ManxPower | torch: that is not a valid Dial line |
| 19:04.44 | ManxPower | you are missing a , before the 90 |
| 19:04.49 | bobdraken | yes |
| 19:04.54 | bobdraken | asterisk |
| 19:04.58 | bobdraken | im lost though i have no idea |
| 19:04.58 | torch | k .... got that ManxPower ..thx |
| 19:05.03 | ManxPower | also, g1 refers to a zaptel group. I assume you have one defined. |
| 19:05.06 | bobdraken | i d/l the thing from teh site |
| 19:05.11 | Mercestes | ManxPower, 90 isn't a timeout.... |
| 19:05.11 | bobdraken | but what do i do to install it |
| 19:05.21 | Mercestes | He wants to postfix a 90 to whatever he's dialing. |
| 19:05.29 | ManxPower | Ah. Then it is correct |
| 19:05.33 | Mercestes | So he dials 50 and the system dials 5090. |
| 19:05.37 | x86_ | hmm, making a 911 context in my dialplan, now if someone wants to dial 911 the office is small enough that sending the call out via one of the regular POTS lines is fine, but what if all POTS lines are busy (the get inbound faxing, send outbound faxes, etc) |
| 19:05.56 | ManxPower | x86_: then you are screwed. |
| 19:06.02 | x86_ | does anyone have a good 911 macro that checks to see if any POTS lines are available, otherwise boot one off? |
| 19:06.03 | torch | ok guys ... back to the dial line without comma ..right? |
| 19:06.15 | x86_ | ManxPower: nah, i seen a macro for it once, can't remember where |
| 19:06.26 | x86_ | ManxPower: asterisk is more powerful than you give credit ;) |
| 19:06.29 | ManxPower | What is torch's specific question? |
| 19:06.46 | bobdraken | anyone? |
| 19:06.48 | x86_ | that's a good question |
| 19:06.53 | Mercestes | ManxPower, blah blah postpend a 90 if I dial 50, blah blah freepbx. |
| 19:06.55 | ManxPower | x86_: Well if you want to spend a month writing a manager app to kill off all calls..... |
| 19:06.56 | Mercestes | I missed the rest. |
| 19:07.03 | torch | added a outbound route to number 50 ... and now I need to add a suffix 90 to it |
| 19:07.19 | ManxPower | torch: go to #trixbox |
| 19:07.28 | Mercestes | Yea, that's what I said :) |
| 19:07.42 | torch | Mercestes .. you said #freepbx ;-) |
| 19:07.44 | Mercestes | I tossed him a bone in that direction to start him running tho. |
| 19:07.50 | Mercestes | torch: yes, yes I did. |
| 19:07.50 | ManxPower | exten => 50,1,Dial(Zap/g1/${EXTEN}90) |
| 19:08.07 | ManxPower | whatever the channel name is. |
| 19:08.11 | bobdraken | can anyone help me install asterisk |
| 19:08.17 | Mercestes | freepbx is Mary Kate and trixbox is Ashley. |
| 19:08.17 | ManxPower | bobdraken: no. |
| 19:08.25 | torch | k guys ...thank you all. |
| 19:08.28 | x86_ | ManxPower: why spend a month re-writing something that's already written? |
| 19:08.30 | x86_ | :) |
| 19:08.30 | [TK]D-Fender | x86_: asterisk -rx "soft hangup zap/1-1" |
| 19:08.31 | Mercestes | technically they're different but.....I defy you to tell them apart. |
| 19:08.37 | ManxPower | bobdraken: There are many consultants that will install it for a fee. |
| 19:08.39 | [TK]D-Fender | x86_: THERE |
| 19:08.41 | Mercestes | wouldn't mind having a threesome with them tho... |
| 19:08.45 | Mercestes | freepbx and trixbox I mean. |
| 19:08.53 | x86_ | [TK]D-Fender: what about 2 people dialing 911 at once? |
| 19:08.55 | bobdraken | ManxPower: so why are you in this room |
| 19:09.13 | ManxPower | bobdraken: To help with issues with Asterisk. |
| 19:09.17 | [TK]D-Fender | bobdraken: Go read THE BOOK, and google up some Ubuntu specific guides as I told you to, and check out www.asterisk.org |
| 19:09.19 | Mercestes | bobdraken, I can help you. |
| 19:09.20 | Mercestes | :) |
| 19:09.22 | Mercestes | ~book |
| 19:09.22 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:09.23 | Mercestes | ~wiki |
| 19:09.24 | bobdraken | ManxPower: isnt installing an issue |
| 19:09.28 | ManxPower | If you have a specific issue then I'm pretty sure we can help. |
| 19:09.30 | Mercestes | ~wikis |
| 19:09.31 | jbot | from memory, wikis is http://www.voip-info.org |
| 19:09.32 | Mercestes | ~docs |
| 19:09.33 | jbot | methinks docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 19:09.33 | bobdraken | Mercestes: thanks |
| 19:09.40 | Mercestes | bobdraken: NP! Good luck. |
| 19:09.50 | bobdraken | Mercestes: fresh instasll of ubuntu on a very small box.. wanting to install asterisk |
| 19:10.03 | Mercestes | oh, sorry.. |
| 19:10.07 | Mercestes | all that jbot stuff was your help. |
| 19:10.10 | bobdraken | Mercestes: i have the d/l .tar.gz from the asterisk site |
| 19:10.10 | Mercestes | Off you go then. |
| 19:10.17 | x86_ | [TK]D-Fender: hmm interesting... POTS channels are designated differently than channels on a T1? |
| 19:10.27 | ManxPower | x86_: not at all |
| 19:10.29 | [TK]D-Fender | x86_: No. |
| 19:10.48 | x86_ | [TK]D-Fender: for example, Zap/1 is the first DS0 on the DS1 going from my asterisk box to the channel bank |
| 19:10.49 | Jingles | I must just be forming this incorrectly. |
| 19:11.00 | x86_ | [TK]D-Fender: Zap/1-1 looks different than Zap/1 to me |
| 19:11.04 | Jingles | Set(CALLERID(ANI)=##########) |
| 19:11.11 | Jingles | the wiki isn't super clear. |
| 19:11.12 | bobdraken | Mercestes: well thanks agin.. so much sarcasam in this room |
| 19:11.13 | Damin | Hmm.. how to unbrick a polycom 501 stuck at the Welcome Screen! |
| 19:11.14 | [TK]D-Fender | x86_: Zap/1 is a DEVICE. Zap/1-1 is a CHANNEl |
| 19:11.20 | Mercestes | Jingles, I am pretty certain ANI is read only. |
| 19:11.25 | Jingles | ? |
| 19:11.25 | ManxPower | x86_: each channel has a -x appended to is for an active call. That is the call instance. in a three-way call or call waiting there would be a -2 as well |
| 19:11.26 | [TK]D-Fender | Damin: Fix your provisioning. |
| 19:11.46 | x86_ | ah right |
| 19:11.51 | Damin | stkn_: But.. but.. but.. it works for every OTHER phone? :) |
| 19:11.54 | x86_ | active channel vs. physical channel |
| 19:11.58 | x86_ | i was cornfused ;) |
| 19:12.32 | ManxPower | Damin: if you bricked your phone and it is not a provisioning issue then you need to send it back to Polycom |
| 19:12.35 | Jingles | this page: http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCIDNum says to use Set(CALLERID... for asterisk 1.2.* and SetCIDNum() for older versions. |
| 19:12.45 | Mercestes | bobdraken, not really sarcasm. If you want help beyond that I suggest a consultant. |
| 19:12.45 | ManxPower | Jingles: ANI is not Callerid |
| 19:12.47 | Jingles | and I need to control the callerID that's going out. |
| 19:13.03 | Jingles | well, SetCallerID() has an ANI flag, but that's not being respected. |
| 19:13.06 | ManxPower | Jingles: Do you want to control the CallerID or the ANI? |
| 19:13.15 | Mercestes | Jingles: And again, I believe that ANI flag is readonly. |
| 19:13.24 | ManxPower | Jingles: how are you determining that ANI is not being set. |
| 19:13.40 | Jingles | at the end of the day - all I care about is when my phone rings, going through a SIP trunk, that I get the callerID *I* want - not 'unavailable' or the SIP provider's #. |
| 19:13.43 | *** join/#asterisk jm|laptop (n=jm|home@cpc1-papw3-0-0-cust844.cmbg.cable.ntl.com) |
| 19:13.51 | Damin | ManxPower: A client sent this one back and said "It acts funny". ;) |
| 19:13.54 | ManxPower | Jingles: then set Callerid NAME,not ANI |
| 19:14.04 | ManxPower | ..e.r.. number, actually |
| 19:14.17 | *** join/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 19:14.19 | Jingles | ok. do I wrap it in < >? |
| 19:14.27 | ManxPower | ANI Automatic Number Identification Service. Typically only available on PRI and CT1 |
| 19:14.35 | Jingles | Set(CALLERID(NUMBER)=<######>) |
| 19:14.35 | bobdraken | there isnt a tutorial for asterisk and ubuntu on asterisk guru |
| 19:14.46 | ManxPower | Oh, and typically only available on toll free inbound numbers |
| 19:14.49 | Jingles | or do I leave the < > out? |
| 19:14.55 | ManxPower | Jingles: Are you stubborn or just stupid? |
| 19:15.02 | Jingles | probably both. |
| 19:15.15 | ManxPower | Try Set(CALLERID(number)=5554441212) |
| 19:15.17 | Damin | Yay!! |
| 19:15.23 | ManxPower | since that SETS THE CALLERID NUMBER |
| 19:15.23 | Damin | I unbricked it! :) |
| 19:15.42 | Jingles | right. However, the SIP provider doesn't seem to respect that I'm setting that value. |
| 19:15.51 | Jingles | as I don't get what I've set on my phone, when I call it. |
| 19:15.58 | *** join/#asterisk Gh0sty (n=ghosty@ip-81-11-190-106.dsl.scarlet.be) |
| 19:16.05 | ManxPower | You have an orange and are asking how to make apple sauce with it. |
| 19:16.14 | Mercestes | Then you call up your SIP provider, scream "retard" into the phone, and then tell them that you wish to pass them the callerID number you specify. |
| 19:16.23 | Mercestes | and ask them to not reset it |
| 19:16.34 | ManxPower | Jingles: Mercestes it won't do him any good as he is not setting outbound callerid |
| 19:16.34 | Jingles | *nods* ok. |
| 19:16.45 | Mercestes | Of course, now I have to ask, are you passing them a *REAL* callerID number or some random fake caller ID number? |
| 19:16.58 | Jingles | That makes sense. Now that I'm setting the outbound callerID correctly, I'm getting *something* - instead of 'unavailable'. |
| 19:17.18 | Jingles | I'm sending them the main callerID for our office, so that our clients/customers know we're trying to call them. |
| 19:17.21 | Gh0sty | Anyone have asterisk 1.2 running on debian etch? (i installed this one since the 1.4 does not support freepbx yet) but i get a lot of times: asterisk died, restarting. It seems with the startscript it is possible to double-start it and then i get the same effect ... :/ |
| 19:17.28 | ManxPower | BTW, aren't variables case sensitive? i.e. CALLERID(NUMBER) is not the same as CALLERID(number)? |
| 19:17.55 | ManxPower | Jingles: what is the ACTUAL line you have in your dialplan? |
| 19:18.20 | Mercestes | They are. Set(CALLERID(numanumadance)=3115551212) |
| 19:18.21 | ManxPower | Gh0sty: start asterisk as "asterisk -cvvv" and you will see what the real error messages are |
| 19:18.23 | Jingles | exten => _31X.,3,Set(CALLERID(NUMBER)=9075551212) |
| 19:18.29 | Jingles | gotcha. |
| 19:18.37 | Mercestes | Jingles: NUMBER is lowercase. |
| 19:18.57 | ManxPower | Good. You don't have extra quotes, spaces, dashes, etc. |
| 19:19.01 | *** join/#asterisk nitomartinez (n=nito@8.Red-80-33-122.staticIP.rima-tde.net) |
| 19:19.18 | Jingles | *nods* just tested - and I'm getting 'unavailable' again. |
| 19:19.45 | ManxPower | Jingles: does it work when you use the correct case? |
| 19:19.46 | Jingles | all I can say is I wish the SIP provider would respond to my requests for support - I want to see what their logs say. |
| 19:20.16 | Jingles | nope. Just get 'unavailable'. when I make number all caps, I get the caller ID # of the SIP trunk I'm dialing through. |
| 19:20.22 | Jingles | which is... odd? to say the least. |
| 19:20.22 | bobdraken | : ( |
| 19:20.28 | Mercestes | lmao |
| 19:20.32 | ManxPower | Jingles: use all lowercase |
| 19:20.36 | ManxPower | not really unusual at all. |
| 19:21.01 | Jingles | well, 8 months into deploying and supporting asterisk, and I'm still 'new' - learning crazy new things every day. |
| 19:21.14 | nitomartinez | Anyone there who could help out with a tdm2400 (wctdm24xxp) module in debian? |
| 19:21.21 | ManxPower | Jingles: Make the priority after the Set(CALLERID(number)=332323) be Noop(CALLERID(number) is ${CALLERID(number)}) |
| 19:21.33 | Jingles | good idea. one sec, whilst I test. |
| 19:21.40 | twitchnln | does anyone have any experience with mitel 5224 dual mode phones? |
| 19:21.41 | bobdraken | i need to remove my isdn card? |
| 19:21.45 | bobdraken | before installation? |
| 19:21.48 | ManxPower | Jingles: remember asterisk is case sensitive |
| 19:21.53 | Mercestes | Jingles: I suggest a nice copy paste of his correct syntax. |
| 19:22.05 | Mercestes | bobdraken, no. |
| 19:22.29 | bobdraken | do i need to install any extra packages |
| 19:22.31 | bobdraken | devel stuff or anything |
| 19:22.37 | blitzrage | I like Verbose() more than NoOp() |
| 19:22.41 | ManxPower | bobdraken: that information is in the asterisk readme or install doc |
| 19:22.54 | [TK]D-Fender | bobdraken: www.asterisk.org <----------- |
| 19:23.03 | ManxPower | blitzrage: I was not aware that 1.2 had a Verbose() application |
| 19:23.16 | blitzrage | hrmmm.... I think it does |
| 19:23.18 | koosroos | hi there the bug is back |
| 19:23.23 | blitzrage | but I haven't used 1.2 in almost a year though |
| 19:23.47 | rob0 | koosroos: Yes, but I like the old classic VW's better. |
| 19:24.06 | nitomartinez | Anyone there who could help out with a tdm2400 (wctdm24xxp) module in debian. The module loads but it doesn't seem to find any card (HW problem perhaps?) |
| 19:24.11 | koosroos | i have installed asterisk on a suse 10.2 version does anyone know where the zaptel source files are? |
| 19:24.18 | rob0 | I did have an '85 Golf which wasn't a bad little car, tho. |
| 19:24.24 | Mercestes | koosroos, wherever you install them to |
| 19:24.26 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 19:24.35 | Jingles | ManxPower : ok. the NoOp line is in, working like you'd expect, and spits out CALLERID(number) is ... |
| 19:24.38 | Jingles | and the # I put in. |
| 19:24.40 | koosroos | do a ztcfg -vvv nitomartinez |
| 19:24.54 | Jingles | what I get on my phone? is the # of the trunk I'm dialing out through. |
| 19:24.55 | nitomartinez | I have done that |
| 19:25.05 | nitomartinez | koosroos: asterisk:~# ztcfg -vvv |
| 19:25.05 | nitomartinez | Zaptel Configuration |
| 19:25.05 | nitomartinez | ====================== |
| 19:25.05 | nitomartinez | Channel map: |
| 19:25.05 | nitomartinez | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 19:25.06 | nitomartinez | Channel 02: FXS Kewlstart (Default) (Slaves: 02) |
| 19:25.08 | nitomartinez | Channel 03: FXS Kewlstart (Default) (Slaves: 03) |
| 19:25.10 | nitomartinez | Channel 04: FXS Kewlstart (Default) (Slaves: 04) |
| 19:25.10 | Mercestes | ... |
| 19:25.12 | nitomartinez | Channel 05: FXO Kewlstart (Default) (Slaves: 05) |
| 19:25.13 | Mercestes | PASTEBIN |
| 19:25.14 | nitomartinez | Channel 06: FXO Kewlstart (Default) (Slaves: 06) |
| 19:25.14 | koosroos | what do you get if you do a zttool? |
| 19:25.16 | nitomartinez | Channel 07: FXO Kewlstart (Default) (Slaves: 07) |
| 19:25.17 | Jingles | waaaaait.... |
| 19:25.18 | nitomartinez | Channel 08: FXO Kewlstart (Default) (Slaves: 08) |
| 19:25.20 | nitomartinez | Channel 09: FXO Kewlstart (Default) (Slaves: 09) |
| 19:25.21 | blitzrage | oh great... we're going to get 24 channels |
| 19:25.22 | nitomartinez | Channel 10: FXO Kewlstart (Default) (Slaves: 10) |
| 19:25.24 | nitomartinez | Channel 11: FXO Kewlstart (Default) (Slaves: 11) |
| 19:25.24 | Mercestes | LMAO |
| 19:25.26 | nitomartinez | Channel 12: FXO Kewlstart (Default) (Slaves: 12) |
| 19:25.28 | joetester | lmao |
| 19:25.28 | nitomartinez | Channel 13: FXO Kewlstart (Default) (Slaves: 13) |
| 19:25.30 | koosroos | nito is in trouble |
| 19:25.30 | nitomartinez | Channel 14: FXO Kewlstart (Default) (Slaves: 14) |
| 19:25.30 | blitzrage | ~pb |
| 19:25.31 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 19:25.31 | Jingles | AH HA! |
| 19:25.31 | Mercestes | Damnit, you ruined the ending for me. |
| 19:25.31 | [TK]D-Fender | gsl;'dkfgl;'skfdl;''l;klgsd |
| 19:25.32 | nitomartinez | Channel 15: FXO Kewlstart (Default) (Slaves: 15) |
| 19:25.34 | nitomartinez | Channel 16: FXO Kewlstart (Default) (Slaves: 16) |
| 19:25.36 | nitomartinez | Channel 17: FXO Kewlstart (Default) (Slaves: 17) |
| 19:25.38 | nitomartinez | Channel 18: FXO Kewlstart (Default) (Slaves: 18) |
| 19:25.40 | nitomartinez | Channel 19: FXO Kewlstart (Default) (Slaves: 19) |
| 19:25.42 | nitomartinez | Channel 20: FXO Kewlstart (Default) (Slaves: 20) |
| 19:25.42 | koosroos | 24 to go |
| 19:25.44 | nitomartinez | 20 channels configured. |
| 19:25.49 | Mercestes | ... |
| 19:25.50 | Mercestes | Aww. |
| 19:25.53 | Mercestes | he ended prematurely. |
| 19:25.59 | Mercestes | he must've gotten excited. |
| 19:25.59 | koosroos | k i was wrong only 20 |
| 19:26.05 | [TK]D-Fender | Mercestes: Now go clean yourself up! ;) |
| 19:26.12 | koosroos | lol |
| 19:26.13 | Mercestes | nitomartinez, it's ok, it happens to everyone |
| 19:26.17 | De_Mon | ???? |
| 19:26.24 | nitomartinez | koosroos: OK Wildcard TDM2400P Board 1 |
| 19:26.27 | Jingles | there's a bit in the notes that say 'hey! n00b! if you provide 'fromuser' in sip.conf, it's going to override your callerid every day of the week! even on wednesday!' |
| 19:26.27 | *** join/#asterisk antongch (n=antongch@cm198234.red83-165.mundo-r.com) |
| 19:26.42 | koosroos | do you have fxo mods on the card? |
| 19:26.48 | bobdraken | so i dont need any of this kernel sources kernal headers bison openssl |
| 19:26.53 | nitomartinez | Sorry |
| 19:26.58 | bobdraken | time sources |
| 19:27.12 | nitomartinez | koosroos: 1 fxo, 5 fxs |
| 19:27.47 | [TK]D-Fender | bobdraken: ..... go to the site and READ. open up THE BOOK and read. |
| 19:27.48 | koosroos | past you zaptel.conf and zapata.conf in pastbin |
| 19:28.09 | nitomartinez | koosroos: I have tried zaptel 1.4.2 and zaptel 1.2.11 |
| 19:29.24 | koosroos | paste you zaptel.conf and zapata.conf in pastebin:nito |
| 19:29.33 | [TK]D-Fender | PASTEBIN <------------------ |
| 19:29.37 | [TK]D-Fender | not PASTE |
| 19:29.38 | javar | :D |
| 19:30.24 | koosroos | kkk |
| 19:30.33 | koosroos | back to my problem |
| 19:30.40 | Mercestes | koosroos: that's racist. |
| 19:30.47 | koosroos | suse 10.2 |
| 19:30.58 | *** join/#asterisk gardo (n=gardo@121.97.211.162) |
| 19:31.24 | *** join/#asterisk zeeesh (n=aadilism@202.125.143.66) |
| 19:31.38 | bobdraken | asterisk now will erase everything else and just leave asterisk |
| 19:31.41 | bobdraken | that stinks |
| 19:31.52 | bobdraken | *asterisknow .. the iso cd install |
| 19:32.14 | [TK]D-Fender | bobdraken: This is NOT what you were advised to use. |
| 19:32.21 | koosroos | need to know where to locate the F&(*&(& zatel source files looking to get rid of echo on zap channels |
| 19:32.42 | *** join/#asterisk joe-f (n=joef@c-71-201-188-239.hsd1.il.comcast.net) |
| 19:33.11 | Mercestes | koosroos: Wherever you *PUT* them. Most likely in /usr/src or C:/Documents and Settings/Windows N00b/My Documents/Downloads |
| 19:33.14 | joe-f | have you guys heard of dialing a conf # too quickly and having it not recognize it? because its seeming to only get it when typed in slowly..? |
| 19:34.25 | koosroos | suse loaded asterisk for me i just checked the little box |
| 19:35.30 | koosroos | if no one can help me i willlllll read in the suse for dummy book |
| 19:35.49 | nitomartinez | kooroos: http://paste.debian.net/31930 |
| 19:36.23 | *** part/#asterisk terrymr (n=terrymr@192.220.217.189) |
| 19:37.45 | nitomartinez | Hi koosroos |
| 19:37.48 | x86_ | http://pastebin.ca/601876 |
| 19:38.14 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 19:38.20 | Mercestes | koosroos: then you need to install asterisk the hard way. There is no guarantee you even *have* source files. |
| 19:38.20 | x86_ | i'm trying to dial 868618006668686 with this dialplan, and I'm getting WARNING[7393]: pbx.c:1721 pbx_extension_helper: No application 'Macro' for extension (from-internal, 868618006668686, 3) |
| 19:38.53 | Mercestes | koosroos: They could be shacked up on your CD where you'd never see them or SUSE could have done a binary install with no source involved. |
| 19:38.55 | x86_ | Mercestes: the "hard way" really isn't so hard ;) |
| 19:39.15 | x86_ | [TK]D-Fender: any ideas? |
| 19:39.29 | Mercestes | x86_: Yea, it's a relative term. Honestly, all those "auto-magic install" processes are far mroe difficult than the source install. |
| 19:39.33 | [TK]D-Fender | x86_: on? |
| 19:39.43 | x86_ | [TK]D-Fender: WARNING[7393]: pbx.c:1721 pbx_extension_helper: No application 'Macro' for extension (from-internal, 868618006668686, 3) |
| 19:39.57 | x86_ | [TK]D-Fender: here's my dialplan: http://pastebin.ca/601876 |
| 19:40.07 | x86_ | Mercestes: most of the time yeah |
| 19:40.08 | *** part/#asterisk twitchnln (n=Administ@70.43.112.117.nw.nuvox.net) |
| 19:41.15 | x86_ | show application Macro shows nothing! (no matter the capitalization) |
| 19:41.21 | Qwell[] | module load app_macro.so |
| 19:41.44 | [TK]D-Fender | x86_: With a dialplan pattern match like this how the hell do you expect your account code to be VARIABLE?! exten => _XXXX866NXXXXXX,1,Set(accountcode=${EXTEN:0:${PGCODE}}) |
| 19:42.22 | [TK]D-Fender | x86_: And missing "Macro" tells me you are missing a primary module... |
| 19:43.41 | x86_ | [TK]D-Fender: it starts at position 0, and captures however man PGCODE tells it to |
| 19:43.47 | x86_ | [TK]D-Fender: pretty basic concept :P |
| 19:43.53 | *** join/#asterisk ManxPower (n=manxpowe@015-843-170.area5.spcsdns.net) |
| 19:44.18 | x86_ | [TK]D-Fender: (PGCODE is a variable of the length of the "Program code", what I was calling account code earlier) |
| 19:44.19 | [TK]D-Fender | x86_: "exten => _XXXX866NXXXXXX" <- looko closer and SLAP YOURSELF. |
| 19:45.13 | koosroos | nitomartinez:http://paste.debian.net/31933 lood this and only this for your zaptel and zapata also check in you indication.conf if you have context [es] |
| 19:45.37 | x86_ | XXXX is the account code, 866 starts the toll-free number, N matches [2-9] once , and X matches [0-9] six times |
| 19:45.46 | x86_ | [TK]D-Fender: what are you blabbering about? |
| 19:46.03 | x86_ | ah |
| 19:46.08 | x86_ | variable length-ness ;) |
| 19:46.21 | x86_ | hmm, interesting |
| 19:46.29 | x86_ | how can i get around that? |
| 19:46.53 | ManxPower | x86_: use all fixed length items |
| 19:46.58 | ManxPower | really, that is the way to get around it. |
| 19:47.17 | [TK]D-Fender | ManxPower *A* way anyways ;) |
| 19:47.18 | ManxPower | all other options would require a timeout |
| 19:47.27 | [TK]D-Fender | ManxPower: Nope! ;) |
| 19:47.50 | ManxPower | [TK]D-Fender: my way is always the right way. I thought you knew that. 8-) |
| 19:47.58 | x86_ | oh snap |
| 19:48.02 | x86_ | you gonna take that? |
| 19:48.07 | x86_ | </instigator> |
| 19:48.09 | x86_ | :P |
| 19:48.11 | [TK]D-Fender | ManxPower: To the wrong end ;) |
| 19:48.32 | x86_ | [TK]D-Fender: can i do a substring from the other side? |
| 19:48.32 | ManxPower | x86_: I assume the account codes are the variable length items |
| 19:48.40 | x86_ | ManxPower: correct |
| 19:48.57 | ManxPower | x86_: and there is no way to make the a fixed length? |
| 19:49.33 | ManxPower | x86_: the issue is that you need a timeout if you want a variable length extension pattern match. |
| 19:49.34 | x86_ | i think i'll have to substring from the left 13 digits to get the dialed number, substring from the right ${PGCODE} digits to get the account code, the trash the rest (which may or may not be a single 1) |
| 19:49.39 | x86_ | ManxPower: no, i dont |
| 19:49.40 | x86_ | ;) |
| 19:49.43 | Jingles | ManxPower : thanks muchly for the pointers. I've resolved the callerID issues for one provider, determined that another isn't honoring or reading the value in right, and that a third is currently down. ;) |
| 19:50.09 | ManxPower | Jingles: Welcome to the world of VoIPoInternet |
| 19:50.12 | ManxPower | I'm glad I don't do that. |
| 19:50.14 | Jingles | *grins* |
| 19:50.34 | Jingles | I guess Set(CALLERID(number)=###) is better than SetCallerID()? |
| 19:50.37 | ManxPower | I do VoIPoCorporateWANorLANthenOverPSTN |
| 19:50.38 | Jingles | or more usable. |
| 19:50.53 | ManxPower | Jingles: SetCallerID will be removed in 1.6 |
| 19:51.06 | Jingles | *nods* |
| 19:51.19 | Jingles | so, for the sake of later migration/upgrades - I should use Set() |
| 19:51.22 | Jingles | which makes sense. |
| 19:52.40 | Hmmhesays | hmm where can I download func_odbc for asterisk 1.2? |
| 19:54.21 | De_Mon | Hmmhesays :) |
| 19:54.44 | De_Mon | !func_odbc |
| 19:54.53 | *** join/#asterisk Meaty (n=meaty3@office.abi.ca) |
| 19:55.28 | De_Mon | Hmmhesays try googling for "func_odbc asterisk 1.2" |
| 19:56.58 | Corydon76-work | http://svncommunity.digium.com/view/func_odbc/ |
| 19:57.20 | De_Mon | func_odbc is the best thing since sliced bread |
| 19:57.54 | Corydon76-work | Heh |
| 19:58.15 | *** join/#asterisk bancus (n=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net) |
| 19:58.36 | Hmmhesays | hmm at can't svn checkout at that address |
| 19:58.43 | bancus | Hi, I'm having trouble getting incoming callerid to work on PSTN with a TDM card. Anyone around that might be able to help? |
| 19:58.50 | Corydon76-work | Hmmhesays: change "view" to "svn" |
| 19:59.04 | nitomartinez | koosroos: Thanks koosroos, just trying |
| 20:01.35 | nitomartinez | koosroos. I get "chan_zap.c: Unable to register channel '1-4'" |
| 20:01.43 | nitomartinez | I think that the module doesn't load |
| 20:02.07 | koosroos | nitomartinez:what do you get if you zttool? |
| 20:02.41 | *** join/#asterisk wyoming (n=steve_mu@216.166.159.235) |
| 20:03.04 | nitomartinez | koosroos: UNCONFIGURED Wildcard TDM2400P Board 1 |
| 20:03.35 | nitomartinez | koosroos: Just run ztcfg -vvvv |
| 20:03.47 | x86_ | Do I want "4E Custom", "5E NI2" ? |
| 20:03.50 | x86_ | which one? |
| 20:03.51 | nitomartinez | koosroos: and now I get OK Wildcard TDM2400P Board 1 |
| 20:03.57 | x86_ | for a CAS T1 |
| 20:04.02 | x86_ | Sangoma T1 card |
| 20:04.46 | Corydon76-work | Hmmhesays: get it? |
| 20:04.48 | *** join/#asterisk ReD-MaN (i=daemon@CPE0002b38bce8b-CM0018c0b357cc.cpe.net.cable.rogers.com) |
| 20:04.51 | koosroos | do a ztcfg -vv |
| 20:05.08 | koosroos | nitomartinez:do a ztcfg -vvvv |
| 20:05.31 | nitomartinez | koosroos: I have already done it, now I get OK Wildcard TDM2400P Board 1 |
| 20:05.55 | nitomartinez | koosroos: And after that I restarted asterisk, but it doesn't see the channels |
| 20:06.11 | nitomartinez | koosroos: See http://paste.debian.net/31934 |
| 20:06.17 | koosroos | nitomartinez:do a lsmod and see if you see zaptel and wctdm24xxp |
| 20:06.58 | nitomartinez | koosroos: both loaded |
| 20:07.07 | nitomartinez | See in the same post |
| 20:07.51 | Hmmhesays | what does multirow do? |
| 20:07.56 | nitomartinez | But the reference count for wctdm is 0 |
| 20:08.23 | Corydon76-work | Hmmhesays: it's new |
| 20:08.38 | Corydon76-work | Hmmhesays: allows you to retrieve multiple row sets |
| 20:08.52 | Corydon76-work | Hmmhesays: but it's only available in 1.4's backport |
| 20:09.08 | koosroos | are you sure you have a quad fxo mod in the slot closest to the cable adapter? |
| 20:09.15 | Corydon76-work | func_odbc normally only lets you retrieve the first row of a resultset |
| 20:09.59 | Strom_M | x86_: none of those options are for CAS T1; those are ISDN signaling flavors |
| 20:10.39 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:10.57 | koosroos | let me get my settings of my working server will be right back :nitomartinez |
| 20:11.04 | nitomartinez | Thanks |
| 20:12.06 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 20:12.08 | x86_ | Strom_M: that's what I thought |
| 20:12.50 | x86_ | Strom_M: I was on a "TAC" call with AT&T ordering my CAS T1, and they explicitly said "non-ISDN service", then asked if I wanted 4E Custom or 5E NI2 |
| 20:12.57 | *** part/#asterisk bobdraken (n=haxxor@75-138-105-182.dhcp.hckr.nc.charter.com) |
| 20:13.21 | x86_ | Strom_M: I told them 5E NI2, since on-the-spot googling seemed to make me believe it worked with the sangoma A102D card |
| 20:13.36 | Strom_M | 5ESS and NI2 are different |
| 20:13.51 | x86_ | yeah I told them 5ESS |
| 20:14.00 | x86_ | not NI2 |
| 20:14.07 | Strom_M | should have picked NI2 |
| 20:14.14 | x86_ | for CAS? |
| 20:14.32 | x86_ | i've never had to answer that question before :P |
| 20:14.56 | *** join/#asterisk OloBola (n=not@74.95.13.57) |
| 20:15.02 | x86_ | i've always just said give it to me b8zs, esf, e&m dial start, and they say OK! |
| 20:15.24 | x86_ | LD only, no local, no inbound |
| 20:15.39 | Corydon76-work | Yeah, strange. Switchtype makes no difference for CAS |
| 20:16.17 | x86_ | and they repeatedly told me "24 channels", so it can't be PRI hehe |
| 20:16.24 | Corydon76-work | I wonder if the phone tech is new and thought she had to fill in ALL of the fields... |
| 20:16.34 | x86_ | perhaps |
| 20:16.39 | Strom_M | Corydon76-home: thats possible |
| 20:16.50 | Strom_M | but still, wouldnt they have separate forms for ISDN and CAS? |
| 20:16.51 | Corydon76-work | Wouldn't be the first time ILEC phone techs were clueless |
| 20:16.54 | x86_ | i was all like "uhhhh... hmm.... lemme look online" lol |
| 20:17.12 | koosroos | nitomartinez:http://paste.debian.net/31935 just remeber my system is a 24 port fxo server |
| 20:17.19 | Strom_M | but...on the flip side, what the hell are you doing ordering a CAS T1? it's the future now; you should be ordering ISDN :) |
| 20:17.27 | nitomartinez | koosroos: thanks |
| 20:17.34 | x86_ | Strom_M: they sent me a single form in an email, which was a fill-in-the-blank word document with check boxes and stuff, and it was for both CAS and PRI |
| 20:18.00 | OloBola | trying to install lumenvox using yum: warning: rpmts_HdrFromFdno: V3 DSA signature: NOKEY, key ID de376963 |
| 20:18.14 | x86_ | Strom_M: that's an old conversation from earlier today ;) |
| 20:18.27 | *** join/#asterisk mindCrime_ (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 20:18.32 | OloBola | public key not available for //var/cache/yum/L.... |
| 20:18.51 | Corydon76-work | OloBola: ignore the error |
| 20:19.16 | Corydon76-work | in this case only |
| 20:19.35 | *** join/#asterisk joxny (n=lukes@gprs.t-mobile.sk) |
| 20:19.50 | Corydon76-work | theoretically, lack of a key might mean that you had an unauthenticated package |
| 20:20.02 | Corydon76-work | which could be a security risk |
| 20:20.08 | OloBola | it looks like it's quiting after the warning, or maybe it's finished? |
| 20:20.09 | Corydon76-work | blitzrage: yes, luv? |
| 20:20.17 | Corydon76-work | OloBola: it's finished |
| 20:20.29 | blitzrage | Corydon76-work: I'm just on a conference call and wasting some unused cycles |
| 20:20.33 | OloBola | oh great, thanks. |
| 20:20.38 | Corydon76-work | blitzrage: ah |
| 20:21.04 | Corydon76-work | blitzrage: I find conference calls in general to be wasted cycles... ;-) |
| 20:21.04 | x86_ | hmm, ok, so chan_zap.so is not wanting to load... zaptel kernel module is running fine, /proc/zaptel/1 shows my TDM04B, /proc/zaptel/2 shows my sangoma A102D |
| 20:21.05 | blitzrage | waiting for someone to build me some RPMs with a patch for OpenSER so I can test another patch which should make transfers work when I have multiple Asterisk boxes in a dispatcher.list file |
| 20:21.14 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) |
| 20:21.15 | blitzrage | Corydon76-work: yep... but I get paid to be on them usually... so that's ok |
| 20:21.16 | x86_ | Jul 3 15:19:45 ERROR[5556]: chan_zap.c:12297 setup_zap: Unable to register channel '1-4' |
| 20:21.30 | x86_ | why would it not be able to register channels? |
| 20:21.40 | x86_ | shouldn't it just be registering devices on load, not channels? |
| 20:21.51 | Corydon76-work | x86_: incorrect signalling most likely |
| 20:21.55 | [TK]D-Fender | x86_: Stop talking about the errors your config is creating and pasbin the damn CONFIGS! :) |
| 20:22.08 | x86_ | [TK]D-Fender: mmk :) |
| 20:22.26 | Corydon76-work | x86_: most common error is when your zaptel.conf and zapata.conf don't agree |
| 20:22.37 | x86_ | Corydon76-work: does it matter that nothing is hooked up to the two T1 spans, and no POTS lines are in the TDM04B? |
| 20:22.37 | Corydon76-work | x86_: so pastebin both, please |
| 20:22.43 | x86_ | ok cool |
| 20:23.16 | Corydon76-work | or if you fail to run 'ztcfg' after loading modules |
| 20:23.29 | *** part/#asterisk joxny (n=lukes@gprs.t-mobile.sk) |
| 20:26.04 | x86_ | ah |
| 20:26.05 | x86_ | no |
| 20:26.07 | x86_ | user error ;) |
| 20:26.34 | x86_ | i put all my trunk definitions under [channels] in zapata.conf, instead of [trunkgroups] |
| 20:26.37 | *** join/#asterisk galeras (n=root@200.31.204.42) |
| 20:26.57 | IOscanner | When starting asterisk from inittab It seems to be spawning more then one asterisk process. It seems it forks a new process off the parent every so often. |
| 20:27.30 | x86_ | is that a bad thing? |
| 20:27.45 | IOscanner | I have seen this a few time before and I think it was on a box that was having a hardware problem. All seem to be related to ZAP. |
| 20:27.58 | Corydon76-work | IOscanner: a new process or a new thread? |
| 20:28.06 | IOscanner | It is when in a week it has forked 100's and crashs asterisk |
| 20:29.14 | IOscanner | http://papernapkin.org/pastebin/view/865 |
| 20:29.21 | agile | I've got a weird thing happening and I can't figure out why.. my asterisk seems to have stopped listening to dtmf codes from incoming calls. I've tried using a sip trunk and an iax trunk from different providers so I'm thinking it must be something on my end. |
| 20:29.54 | agile | eg: if I set up an extension that just has WaitExten(60) and dial it, I get nothing |
| 20:30.13 | agile | I mean the WaitExten executes but it's like it doesn't get my key presses |
| 20:30.17 | IOscanner | even is I stop asterisk I have to keep the subs with -9 before I can start asterisk again |
| 20:30.55 | koosroos | how does one load the timers for your coutry ???? indications can someone please light up this subject! |
| 20:32.14 | *** join/#asterisk x86_ (n=bryce@p3m/member/x86) |
| 20:32.53 | Corydon76-work | IOscanner: ps auxwww |grep asterisk |
| 20:33.04 | Corydon76-work | IOscanner: that'll show process state |
| 20:34.32 | x86_ | Access Remarks: NI-2 5ESS Protocol |
| 20:34.41 | x86_ | 1st T1 Channel Layout: 24 non isdn channels |
| 20:35.08 | x86_ | Service Type: Non-ISDN |
| 20:35.23 | x86_ | obviously the protocol is in some standard template on their side |
| 20:35.36 | agile | I have 2 young virgin goats I will give to the first man you returns my stolen dtmf codes |
| 20:35.37 | x86_ | just got that email regarding the "TAC" call with AT&T on this CAS T1 order |
| 20:37.22 | Corydon76-work | agile: try it with IAX only |
| 20:37.37 | Corydon76-work | agile: that will eliminate SIP-specific issues |
| 20:37.44 | IOscanner | http://papernapkin.org/pastebin/view/866 |
| 20:38.20 | Corydon76-work | IOscanner: yep, those are all threads |
| 20:38.35 | agile | Corydon76-home, I've got a teliax account and have tried using only it but get the same issue :( |
| 20:38.36 | Corydon76-work | The first one is the process you started |
| 20:38.52 | IOscanner | yep |
| 20:38.56 | IOscanner | starting from inittab |
| 20:38.59 | Corydon76-work | agile: Ah, I think I know what the problem is |
| 20:39.01 | agile | I see nothing on the console when I try to enter anything |
| 20:39.17 | Corydon76-work | agile: did you forget to Answer the channel prior to running WaitExten() ? |
| 20:39.20 | *** join/#asterisk Abe_Froman (i=holcomb@oc9.org) |
| 20:39.37 | x86_ | so does my numbering in /etc/zaptel.conf for span= have to match what zaptel shows in /proc/zaptel/1, /proc/zaptel/2, /proc/zaptel/3 ? |
| 20:39.40 | agile | Corydon76-home, hmm... *blush* |
| 20:39.43 | Corydon76-work | Unanswered channels cannot receive DTMF |
| 20:39.48 | x86_ | because in zaptel.conf I number 0,1,2 |
| 20:41.14 | agile | Corydon76-home, you were correct, but even after adding the answer I get the same thing :( |
| 20:41.47 | Corydon76-work | agile: add a Playback(tt-weasels). do you get any audio at all? |
| 20:42.36 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 20:42.37 | agile | Corydon76-home, I get the audio prompts, (was actually trying VoicemailMain now) |
| 20:43.05 | Corydon76-work | agile: can you try a different provider? |
| 20:43.24 | Corydon76-work | agile: or use a local SIP phone? |
| 20:44.01 | agile | Corydon76-home, we tried using a sipphone account as the trunk but get the same thing which is why I'm thinking it's something on my end |
| 20:44.59 | Corydon76-work | If you type 'iax2 debug', you should see IAX2 DTMF generated as separate events |
| 20:45.18 | Corydon76-work | If you don't see that, the provider isn't sending them correctly |
| 20:45.52 | IOscanner | any idea why it is threading so many? |
| 20:46.11 | Corydon76-work | IOscanner: you running any 3rd party modules? |
| 20:47.42 | Voicemeup | agile |
| 20:47.48 | Voicemeup | restart the box |
| 20:47.54 | Voicemeup | you have zap on it ? |
| 20:48.08 | Voicemeup | i happened to me on 1 box.. zap got a few dealocks.. and killed dtmf recognition |
| 20:48.16 | Voicemeup | had to actually reboot it .. then all came back |
| 20:48.16 | *** join/#asterisk javb (n=javb@190.80.224.21) |
| 20:48.32 | *** join/#asterisk bobdraken (n=haxxor@75-138-105-182.dhcp.hckr.nc.charter.com) |
| 20:48.47 | bobdraken | i tried to restart asterisk i just installed it and i get this |
| 20:48.49 | agile | Voicemeup, no zap but what the heck :) restarting |
| 20:48.49 | bobdraken | trying to start it |
| 20:48.51 | bobdraken | Asterisk not yet configured. Edit /etc/default/asterisk first. |
| 20:48.53 | Voicemeup | no |
| 20:48.54 | javb | people, is there a chance that i can set an account to be part of more that one group (not callgroup neither pickupgroup) talking about 'group', what u set on zapata.conf |
| 20:48.56 | Voicemeup | hmm |
| 20:49.02 | Voicemeup | reboot |
| 20:49.10 | Corydon76-work | Voicemeup: IAX2 doesn't rely on recognition sequences |
| 20:49.12 | Voicemeup | unless it never worked |
| 20:49.13 | bobdraken | Voicemeup: me? |
| 20:49.24 | Voicemeup | agile did dtmf ever work ? |
| 20:49.31 | bobdraken | Voicemeup: me reboot? |
| 20:49.33 | krondorl | Night all. Working day over for me... |
| 20:49.38 | Voicemeup | agile :could be a codec issue to.. |
| 20:49.39 | *** join/#asterisk tako-san (n=Tako-san@24.68.129.29) |
| 20:49.40 | Corydon76-work | DTMF isn't transmitted as audio on IAX2 |
| 20:49.43 | Voicemeup | bobdraken no |
| 20:49.43 | Corydon76-work | ever |
| 20:49.44 | agile | Voicemeup, yes.. it mysteriously stopped a couple weeks ago |
| 20:49.53 | bobdraken | Asterisk not yet configured. Edit /etc/default/asterisk first. |
| 20:49.56 | Voicemeup | agile ok : try rebooting the hwole server to see |
| 20:50.04 | bobdraken | what should i do about that im tryint to start asterisk |
| 20:50.13 | Voicemeup | agile: start fresh |
| 20:50.29 | Corydon76-work | bobdraken: just a guess, but maybe you ought to edit the file it's telling you about |
| 20:50.34 | agile | Voicemeup, yea doing it now |
| 20:50.39 | bobdraken | im in it but edit what |
| 20:50.43 | IOscanner | Nope just asterisk |
| 20:50.56 | Corydon76-work | bobdraken: no idea, I'm on Mandriva here |
| 20:51.04 | javb | people, is there a chance that i can set an account to be part of more that one group (not callgroup neither pickupgroup) talking about 'group', what u set on zapata.conf |
| 20:51.14 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 20:51.16 | bobdraken | im on ubuntu |
| 20:51.17 | IOscanner | I only see then when I am using digium cards (TDM400P) with FXO ports |
| 20:51.23 | Voicemeup | ok lets see the results.. like i said only times DMTF stopped on my side .12.X branches.. is zap crying and messing something up .. then dtmf stops for soem obscure reasons.. happened only 3 times in 1 year but never know |
| 20:51.34 | Corydon76-work | bobdraken: email the maintainer |
| 20:51.40 | bobdraken | what? |
| 20:51.44 | bobdraken | Corydon76-do whawt? |
| 20:51.54 | Corydon76-work | bobdraken: or put the error message into google |
| 20:52.22 | *** join/#asterisk jm|laptop (n=jm|home@cpc1-papw3-0-0-cust844.cmbg.cable.ntl.com) |
| 20:52.46 | Corydon76-work | bobdraken: google suggests that "RUNASTERISK" is set to "no" |
| 20:53.01 | Corydon76-work | or not "yes" anyway |
| 20:53.24 | *** join/#asterisk troy- (n=tabmeist@CPE00907f17e478-CM00186845db94.cpe.net.cable.rogers.com) |
| 20:53.25 | Corydon76-work | https://wiki.ubuntu.com/AsteriskOnUbuntu |
| 20:53.31 | Corydon76-work | ~ubuntu |
| 20:53.32 | jbot | from memory, ubuntu is a Linux distribution for average computer users |
| 20:53.47 | Corydon76-work | jbot: ubuntu is also https://wiki.ubuntu.com/AsteriskOnUbuntu |
| 20:53.48 | jbot | ACTION lovingly explains to is also https://wiki.ubuntu.com/AsteriskOnUbuntu in a way that causes is also https://wiki.ubuntu.com/AsteriskOnUbuntu to weep with gratitude that is also https://wiki.ubuntu.com/AsteriskOnUbuntu must read the fine, friendly manual |
| 20:54.28 | *** join/#asterisk currach (n=currach@213-202-141-138.bas502.dsl.esat.net) |
| 20:54.28 | Corydon76-work | ~ubuntu |
| 20:54.29 | jbot | methinks ubuntu is a Linux distribution for average computer users |
| 20:54.36 | troy- | does linux come with a tftp server? |
| 20:54.37 | Corydon76-work | jbot: ubuntu is also https://wiki.ubuntu.com/AsteriskOnUbuntu |
| 20:54.38 | jbot | ACTION lovingly explains to is also https://wiki.ubuntu.com/AsteriskOnUbuntu in a way that causes is also https://wiki.ubuntu.com/AsteriskOnUbuntu to weep with gratitude that is also https://wiki.ubuntu.com/AsteriskOnUbuntu must read the fine, friendly manual |
| 20:54.56 | Corydon76-work | wtf? |
| 20:54.57 | bobdraken | it wont let me look in the ect/asterisk folder |
| 20:55.01 | bobdraken | it says i dont have permsion |
| 20:55.07 | Corydon76-work | bobdraken: you're not root, then |
| 20:55.08 | agile | Voicemeup, after a fresh boot still have the problem :( |
| 20:55.09 | Strom_M | bobdraken: try /etc/ not /ect/ |
| 20:55.15 | bobdraken | thats what i ment |
| 20:55.23 | bobdraken | i dont know roots pwd |
| 20:55.24 | Corydon76-work | bobdraken: sudo su - |
| 20:55.28 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 20:55.35 | Corydon76-work | It's your own password |
| 20:55.46 | Corydon76-work | assuming you installed the box |
| 20:55.47 | tako-san | or sudo -s |
| 20:56.29 | bobdraken | k term says im root now |
| 20:56.34 | bobdraken | do i need to restart my session or |
| 20:56.40 | bobdraken | the folder says i still dont have permsion |
| 20:57.00 | Corydon76-work | Eh? |
| 20:57.23 | Corydon76-work | Does the directory exist? |
| 20:58.09 | bobdraken | what do u mean |
| 20:58.30 | bobdraken | i see etc, asterisk and it has a red x above the folder |
| 20:58.41 | bobdraken | i click it and it says that i dont have permisoin |
| 20:58.45 | bobdraken | but im root |
| 20:58.51 | tako-san | you should access it via the command line |
| 20:58.53 | Corydon76-work | Use the terminal, not your GUI |
| 20:58.56 | tako-san | not the gui |
| 20:58.59 | tako-san | exactly |
| 20:59.02 | bobdraken | hum... |
| 20:59.07 | bobdraken | ill try |
| 20:59.13 | bobdraken | im trying to start asterisk |
| 20:59.44 | Corydon76-work | bobdraken: tell me you're doing this as a hobbyist and not as a businessman |
| 20:59.46 | bobdraken | Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk -r' to connect. |
| 20:59.54 | bobdraken | yes just to learn cory |
| 20:59.57 | bobdraken | im new to linux also |
| 20:59.57 | bobdraken | so |
| 20:59.59 | bobdraken | hehe |
| 21:00.02 | Corydon76-work | Thank god |
| 21:00.05 | fujin | might help to learn linux first |
| 21:00.17 | bobdraken | well i was hoping it would have gui but its ok ... now i got to set up my usb phone |
| 21:00.22 | bobdraken | and try to get that to work |
| 21:00.24 | bobdraken | im soooo lost lol |
| 21:00.48 | fujin | #ubuntu |
| 21:00.53 | *** part/#asterisk javb (n=javb@190.80.224.21) |
| 21:01.19 | bobdraken | what do i need to open to mess with my phone.. i need to install it i guess |
| 21:01.22 | bobdraken | or set it up |
| 21:01.33 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:01.42 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 21:02.39 | bobdraken | 6363 |
| 21:03.00 | bobdraken | opps lol sorry |
| 21:03.11 | bobdraken | i was dialing on the phone and it typed in here |
| 21:03.28 | bobdraken | do i need to install some voip phone software cause i dont think the software that came with the phone will work on linux |
| 21:03.35 | bobdraken | or can i do it through asterisk' |
| 21:04.07 | troy- | does tftp come preloaded on linux? |
| 21:04.10 | polerin | bobdraken: your hardphone right? |
| 21:04.15 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 21:04.19 | bobdraken | its a voip xact phone |
| 21:04.29 | bobdraken | i pluged it into my usb |
| 21:04.46 | polerin | troy-: do dual zone climate controls come preloaded on cars? |
| 21:04.58 | polerin | bob: ew. no ethernet? |
| 21:05.00 | [TK]D-Fender | troy-, No, TFTF is a protocol. Now there are programs that speak that protocol. But they don't come with Linux. |
| 21:05.17 | bobdraken | polerin: what do you mean |
| 21:05.25 | troy- | [TK]D-Fender, i have one installed now in.tftpd but i cant seem to find another copy of it on the net? |
| 21:05.27 | bobdraken | no ethernet.. the phone is usb |
| 21:05.32 | [TK]D-Fender | troy-, They come with DISTROs (or downlaoded sperately. Now DISTROs tend to come with Linux ;) |
| 21:05.39 | troy- | hah |
| 21:05.44 | polerin | bobdraken: yeah.. well hook it up to your windows system :P |
| 21:05.52 | polerin | does it support sip? |
| 21:05.58 | [TK]D-Fender | troy-, because you see, Linux is a KERNEL |
| 21:06.02 | bobdraken | i dont have a windows systtem... are you saying that this phone wont work |
| 21:06.12 | polerin | [TK]D-Fender: heh yeah I was about to get on you because I know YOU know better ;) |
| 21:06.31 | polerin | bobdraken: nope, I'm saying you should hit google and find out :) |
| 21:06.36 | [TK]D-Fender | RMS would be proud of me right now :) |
| 21:06.47 | bobdraken | im confused lol |
| 21:07.05 | bobdraken | is there something i can look at to see what asterisk is doing |
| 21:07.08 | polerin | meh, I care less about what RMS thinks of the syntax then the clarity of the answers :D |
| 21:07.14 | fujin | troy-: generally not, I usually run atftpd - found it's the best tftpd so far |
| 21:07.15 | bobdraken | options or a command line for asterisk |
| 21:07.24 | polerin | bobdraken: find out if you can get the phone drivers working |
| 21:07.30 | polerin | you don't have to have it directly on the * box. |
| 21:07.30 | troy- | fujin, i have one installed called in.tftpd but i cant find it on the net? |
| 21:07.38 | bobdraken | so google... xact asterisk |
| 21:07.40 | [TK]D-Fender | troy-, tftpd is FINE. Whats your bloody problem? :) |
| 21:07.42 | bobdraken | or xact linux drivers |
| 21:07.44 | fujin | in.tftpd is the default one, it's terrible |
| 21:07.45 | fujin | I forget who makes it |
| 21:08.04 | troy- | fujin, its worked fine, i just need another copy of it |
| 21:08.12 | fujin | what distro? |
| 21:08.15 | fujin | apt-get install tftpd ;] |
| 21:08.16 | troy- | centos |
| 21:08.22 | fujin | yum? |
| 21:08.34 | fujin | yum search tftpd |
| 21:08.41 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 21:08.41 | troy- | yum is right :D |
| 21:08.42 | irule | what is the meaning of zed.gsm? I see it in sounds.txt but it did is not included in my latest asteris/asterisk-addons installation |
| 21:08.44 | [TK]D-Fender | troy-, What do you mean "another copy of it"?! |
| 21:08.47 | polerin | **headdesk** |
| 21:08.59 | troy- | [TK]D-Fender, i want to reinstall the OS but i want to make sure i have a copy off it first |
| 21:09.02 | [TK]D-Fender | irule, its a pronunciation of "z". |
| 21:09.13 | polerin | irule: I suspect it's the non american pronounciation of z... aw foo |
| 21:09.14 | [TK]D-Fender | irule, Hukt on fonix werkt 4 u! |
| 21:09.16 | polerin | dangit |
| 21:09.36 | irule | oh, thanks |
| 21:10.03 | [TK]D-Fender | troy-, its on the install ISO's if thats what you mean :) |
| 21:10.23 | polerin | or you can get it remotely |
| 21:10.27 | polerin | doesn't yum do fetches |
| 21:10.41 | [TK]D-Fender | yum install tftpd <0 |
| 21:10.48 | fujin | ;] |
| 21:10.58 | [TK]D-Fender | Yippy-kia-yay |
| 21:11.10 | troy- | [TK]D-Fender, ah okay so i dont need to download it obviously? |
| 21:11.34 | [TK]D-Fender | troy-, Depends if you INSTALLED it when you had the chance. Did *I* pick your packages?! NO~! |
| 21:11.42 | troy- | hah |
| 21:11.50 | [TK]D-Fender | troy-, "man tftpd" <------------- |
| 21:11.55 | [TK]D-Fender | troy-, good way to find out. |
| 21:11.56 | troy- | but worst case scenario i can just do yum install tftpd? |
| 21:12.02 | bobdraken | no luck so far im still googling |
| 21:12.16 | [TK]D-Fender | troy-, Now did you CONFIGURE it? Is the service RUNNING? Who shot J.R?!?! |
| 21:12.27 | polerin | yeeeeeee |
| 21:12.41 | [TK]D-Fender | bobdraken, extremely high odds that USB phone is worthless to you. Stopp looking and go try something productive. |
| 21:12.56 | troy- | [TK]D-Fender, via the /etc/xinetd.d/tftp file |
| 21:13.22 | [TK]D-Fender | troy-, check netstat. |
| 21:13.22 | polerin | troy-: ok thats ONE part of it, but that's far from the only place you configure things |
| 21:13.34 | tako-san | troy-: did you restart xinetd.d? |
| 21:13.41 | polerin | troy-: google "in.tftd configuration tutorial" |
| 21:13.52 | troy- | polerin, its already configured |
| 21:14.01 | troy- | i just cant remember how i installed it |
| 21:14.26 | fujin | you use inetd |
| 21:14.27 | fujin | to listen |
| 21:14.28 | fujin | and then spawn it |
| 21:14.37 | bobdraken | [TK]D-Fender: like what |
| 21:14.44 | bobdraken | [TK]D-Fender: buy a diff phone? |
| 21:14.56 | polerin | bobdraken: like use a softphone |
| 21:15.19 | polerin | or two, or whatever |
| 21:15.23 | bobdraken | whats a "softphone"? |
| 21:15.29 | troy- | fujin, or xinetd? |
| 21:15.32 | bobdraken | a regular landline phone? |
| 21:15.37 | fujin | like hardware and software |
| 21:15.37 | fujin | use brain |
| 21:15.38 | polerin | no |
| 21:15.41 | bobdraken | oOo |
| 21:15.56 | bobdraken | ok i have a mic so yeah i can do that... which should i d/l and use |
| 21:15.57 | fujin | learn2google |
| 21:16.05 | fujin | wikipedia has a nice list |
| 21:16.24 | [TK]D-Fender | bobdraken, Ever use Skype? Thats a soft-phone. MSN Messgenger w/ voice? DITTO. NetMeeting? DUH |
| 21:16.35 | [TK]D-Fender | bobdraken, now NONE of THOSE are recommended. |
| 21:17.00 | [TK]D-Fender | bobdraken, for Windows : X-Lite, Ifdefisk. For Linux : Ekiga, Idefisk |
| 21:17.13 | polerin | [TK]D-Fender: first VoIP anything I ever used was I-chat. rember that? |
| 21:17.16 | OloBola | aftering trying to install Lumenvox license server, the directory where it's supposed to be install is empty (/opt/ is empty) |
| 21:17.17 | OloBola | http://www.pastebin.ca/602034 |
| 21:17.27 | [TK]D-Fender | polerin, No, I avoid Mac :) |
| 21:17.43 | polerin | it |
| 21:17.47 | polerin | it's not a mac product |
| 21:17.55 | polerin | it was an intel proof of concept thing |
| 21:17.56 | [TK]D-Fender | polerin, I was using Netmeeting a decade ago for video conferencing :) |
| 21:18.06 | polerin | this was back in `95 |
| 21:18.15 | polerin | or around it. |
| 21:18.59 | polerin | it was bad... you pushed to talk, it sent it up and buffered on the other side. when the whole send was buffered, it waited till it's chance to play, then played one clip at a time |
| 21:19.45 | OloBola | does it look like it's quitting ? http://www.pastebin.ca/602034 |
| 21:20.37 | [TK]D-Fender | *b00m* |
| 21:20.38 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) [NETSPLIT VICTIM] |
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| 21:20.39 | *** mode/#asterisk [+oo blitzrage Corydon76-home] by irc.freenode.net |
| 21:20.41 | polerin | nice. split |
| 21:21.05 | *** join/#asterisk bobdraken (n=haxxor@75-138-105-182.dhcp.hckr.nc.charter.com) |
| 21:21.05 | bobdraken | sorry froze up |
| 21:21.16 | [TK]D-Fender | *b00m* |
| 21:21.16 | polerin | bobdraken: you forgot lube didn't you |
| 21:21.25 | bobdraken | i started eikga |
| 21:21.37 | bancus | Hi, I'm having trouble getting incoming callerid to work on PSTN with a TDM card. Anyone around that might be able to help? |
| 21:21.39 | bobdraken | and it said that something else was using sip something i dono ... probably asterisk |
| 21:21.41 | javar | :O |
| 21:21.47 | bobdraken | but i have eikga |
| 21:21.55 | [TK]D-Fender | bobdraken, you have a LOT of reading to do. Stop worrying about these fine bits and start learning the BASICS |
| 21:22.03 | polerin | bobdraken: go read the book |
| 21:22.04 | polerin | :P |
| 21:22.24 | bobdraken | i guess i cant play with these softphones unless i set up other soft phones in the house |
| 21:22.25 | polerin | bobdraken: how many computers do you have there? |
| 21:22.31 | [TK]D-Fender | bobdraken, and indeed * & ekiga are fighing to reserve SIP' port (5060) for inbound. |
| 21:22.35 | bobdraken | unless... i use that usb phone and set it up on another computer |
| 21:22.38 | troy- | polerin, btw i googled as you said and no ersults L:P |
| 21:22.38 | bobdraken | in this house i have... |
| 21:22.41 | bobdraken | 5 computers |
| 21:22.49 | polerin | troy-: lol, well it was just a guess :) |
| 21:22.53 | bobdraken | [TK]D-Fender: what does that mean |
| 21:22.54 | troy- | :P |
| 21:22.54 | [TK]D-Fender | bobdraken, I would suggest you start with a soft-phone on ANOTHER computer for now. |
| 21:23.01 | OloBola | does it look like it's quitting ? http://www.pastebin.ca/602034 (lumenvox license server installation) |
| 21:23.03 | polerin | troy-: intended to make you google for information and look it up. |
| 21:23.04 | bobdraken | this is my play box |
| 21:23.05 | troy- | [TK]D-Fender, btw tftp wont show up in netstat if there are no requests |
| 21:23.19 | [TK]D-Fender | bobdraken, this is networking 101. you can't have 2 apps fighting to LISTEN on the same port. |
| 21:23.19 | troy- | xinetd starts the process when it sees requests for the daemon |
| 21:23.24 | polerin | troy-: well you can always just nmap yourself |
| 21:23.25 | polerin | :P |
| 21:23.25 | bobdraken | oOo |
| 21:23.29 | Qwell[] | it doesn't always get started by inetd |
| 21:23.32 | bobdraken | so have this as an asterisk box |
| 21:23.38 | bobdraken | and... my phone on another |
| 21:23.45 | troy- | Qwell[], in my case xinetd = inetd |
| 21:23.51 | Qwell[] | it doesn't always get started by an inetd |
| 21:23.54 | [TK]D-Fender | troy-, Comment out from xinetd and start it as a daemon yourself then! ;) |
| 21:23.58 | Qwell[] | ^^ |
| 21:24.07 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 21:24.14 | Defraz | Here is a weird issue I am having, When person calls into our Asterisk server it gets routed to a ring group, Someone in the ring group answers the phone, then they transefer the call to another extention, the caller gets music when the person gets placed on hold but then the minute call starts to ring threw they get dead silence until the caller picks up the extention. |
| 21:24.24 | *** join/#asterisk fujin (n=fujin@unaffiliated/fujin) |
| 21:24.30 | bobdraken | i can install my usb xact phone on my my brothers window box.. but can i configure it to connect to my asterisk box |
| 21:24.31 | troy- | Qwell[], in a brand spanking new centos installation will in.tftpd come with? |
| 21:24.33 | bobdraken | were on a router |
| 21:24.38 | polerin | woh |
| 21:24.45 | polerin | run on sentance |
| 21:24.48 | Qwell[] | troy-: If you tell it to, sure |
| 21:24.49 | [TK]D-Fender | Defraz, pastebint he full CLI output of the call at verbose 10 w/ channel debug. |
| 21:24.50 | polerin | bobdraken: you are both on the same router? |
| 21:24.54 | bobdraken | yes |
| 21:24.55 | fujin | troy-: why does it matter? yum install it! |
| 21:25.02 | OloBola | does it look like it's quitting ? http://www.pastebin.ca/602034 (lumenvox license server installation) |
| 21:25.04 | bobdraken | eth connection not wifi |
| 21:25.07 | troy- | fujin, will yum install tftpd install in.tftpd or another tftp daemon? |
| 21:25.13 | [TK]D-Fender | bobdraken, that is NOT a phone. It is the equivalent to a stupid speaker & mic |
| 21:25.16 | fujin | no idea |
| 21:25.18 | fujin | why? |
| 21:25.23 | troy- | because i want in.tftpd |
| 21:25.25 | bobdraken | ok i undestand... but |
| 21:25.26 | [TK]D-Fender | bobdraken, You will need to tell a soft-phone to try and USE it. |
| 21:25.35 | bobdraken | so tell my brother to set up a softphone |
| 21:25.38 | [TK]D-Fender | bobdraken, www.xten.com <- go download X-Lite |
| 21:25.45 | *** join/#asterisk Ein (i=nassarmu@gateway/web/cgi-irc/beitsahour.net/x-2402d58794f27b86) |
| 21:25.47 | bobdraken | does that work on M$ |
| 21:25.50 | bobdraken | he has M$ |
| 21:25.52 | [TK]D-Fender | YES |
| 21:25.57 | bobdraken | ok one sec |
| 21:26.05 | [TK]D-Fender | bobdraken, get off your but. Go to the site. DOWNLOAD & INSTALL |
| 21:26.18 | Ein | i have a voip question but it is not Asterisk related, anyone here familiar with Avaya equipment? |
| 21:26.41 | polerin | Ein: I'm familiar with the gawdam queue entry beep in my ear |
| 21:26.41 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 21:26.50 | polerin | Ein: fucking queue. arg. |
| 21:26.51 | Ein | polerin: heh |
| 21:27.06 | bobdraken | hes insatlling it now.. what do oi need to do |
| 21:27.12 | bobdraken | on the asterisk box |
| 21:28.06 | bobdraken | it says that cost 60 bucks |
| 21:28.51 | Ein | basically i have a 4610SW IP phone that is connecting to an Avaya voip box over a vpn, it seems to authenticate fine but no dialtone |
| 21:29.02 | bobdraken | the free version will work? |
| 21:29.38 | polerin | bobdraken: TRY IT |
| 21:29.41 | Ein | i ran a tcpdump and the voip box keeps trying to connect to the phone on port 49300-49309, and the phone keeps responding, destination port unreachable |
| 21:29.44 | polerin | **headdesks** |
| 21:30.13 | polerin | Ein: are the ports open on that phone in that range? |
| 21:30.15 | [TK]D-Fender | Ein, this isn't #avaya |
| 21:30.32 | *** part/#asterisk putnopvut (i=putnopvu@nat/digium/x-a728a28792c373e1) |
| 21:30.36 | [TK]D-Fender | bobdraken, I said X-LITE! |
| 21:30.57 | Ein | [TK]D-Fender: i know, i stated that it was not asterisk related and people responded that they are familiar with avaya, i took that to mean that they would help |
| 21:31.12 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-97-164.ph.ph.cox.net) |
| 21:31.12 | Ein | polerin: nmap of the phone shows nothing open |
| 21:32.03 | OloBola | does it look like my installation is quitting ? http://www.pastebin.ca/602034 (lumenvox license server installation) The /opt/ dir is empty |
| 21:33.15 | polerin | Ein: I'm familiar as a user, not admin. Sounds like you need to find what ports the phone is listening on or get into the phone config and set it up right **shrugs** that's the end of my support |
| 21:33.41 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 21:33.45 | FuriousGeorge | hey all. |
| 21:33.45 | Ein | heh, oh well |
| 21:33.47 | Ein | thanks |
| 21:34.12 | polerin | :) |
| 21:34.32 | polerin | Ein: To be honest it was nice to at least hear a sanely phrased question ;P |
| 21:34.39 | *** join/#asterisk sunsmasher (n=Beamer@66.251.47.154) |
| 21:34.51 | Defraz | TK doing it now. Just taking a bit to stage the call, and there are tons of other calls going on too. So let me get that done thanks. |
| 21:35.01 | x86_ | what do yall recommend for call accounting? I've tried Areski, but it's not really that good in my opinion |
| 21:35.47 | FuriousGeorge | been using asterisk for a few years now. its not often im at a total loss like this. for some reason, when the internet (vi which i route outgoing calls) goes down it causes some sort of deadlock in asterisk 1.4.5 users cannot complete outbound calls, the cli stops responding, stop now doesnt work, all commands that start with sip (like sip show peers) are gone. users can call eachother via sip, but routing calls over POTS by dia |
| 21:35.51 | FuriousGeorge | what sort of madness is this |
| 21:36.03 | FuriousGeorge | oh yeah, and all the calls i attempt to make result in the channel hanging |
| 21:36.18 | [TK]D-Fender | FuriousGeorge, ..... 1.4.6 <- |
| 21:37.03 | FuriousGeorge | [TK]D-Fender: k |
| 21:37.27 | polerin | FuriousGeorge: btw, the last words that came over were "...POTS by dia" |
| 21:37.33 | polerin | yah char limit :) |
| 21:37.52 | FuriousGeorge | polerin: , but routing calls over POTS by dialing 9 doesnt work |
| 21:38.06 | FuriousGeorge | (and it should as per my setup) |
| 21:38.55 | FuriousGeorge | ironic that i upgrade to 1.4.5 on thurday and my ISP goes down for the first time in 12 months |
| 21:39.00 | FuriousGeorge | on Sat |
| 21:39.35 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9ui.cable.mindspring.com) |
| 21:40.07 | VJFROMGT | my entire outbound route fails when a single trunk fails http://www.asteriskguru.com/board/here-vp6105.html#6105 |
| 21:43.23 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 21:44.05 | [TK]D-Fender | VJFROMGT, ...... |
| 21:44.09 | [TK]D-Fender | ~freepbx |
| 21:44.10 | jbot | methinks freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 21:44.12 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 21:44.15 | *** part/#asterisk Ein (i=nassarmu@gateway/web/cgi-irc/beitsahour.net/x-2402d58794f27b86) |
| 21:44.19 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 21:44.47 | EmleyMoor | I have installed asterisk-web-vmail from Debian, got round a permissions problem, but it doesn't find messages |
| 21:45.13 | tzafrir_laptop | EmleyMoor, I'm not sure it ever worked out of the box |
| 21:45.28 | EmleyMoor | tzafrir: Hmmm... |
| 21:45.43 | tzafrir_laptop | You probably need to mess a bit with apache or whatever your httpd is |
| 21:45.56 | tzafrir_laptop | I consider that script buggy and don't use it myself |
| 21:46.21 | EmleyMoor | tzafrir_laptop: I have implemented a recommended fix and it now lets me log in, but finds no messages |
| 21:46.51 | tzafrir_laptop | EmleyMoor, hmmm... what recommended fix? from where? |
| 21:47.05 | irule | are ALL vm-* files used by voicemail application? these are the only ones that I asume that * uses, and are hard-coded, so, they are the only ones that dont need to be in extensions.conf to be needed? I am just making a list of the files I am using. |
| 21:47.34 | EmleyMoor | tzafrir-laptop: Make www-data a member of asterisk - from the Debian BTS |
| 21:49.17 | x86_ | what do yall recommend for call accounting? I've tried Areski, but it's not really that good in my opinion |
| 21:49.34 | [TK]D-Fender | irule, FFS just grep that shit out of the * source! |
| 21:50.00 | [TK]D-Fender | irule, and you assume WRONG. |
| 21:50.05 | irule | [TK]D-Fender good FFFS point, thanks lol |
| 21:50.10 | VJFROMGT | tk,, i am back |
| 21:50.41 | VJFROMGT | my entire outbound route fails when a single trunk fails http://www.asteriskguru.com/board/here-vp6105.html#6105 |
| 21:50.59 | OloBola | does it look like my installation is quitting ? http://www.pastebin.ca/602034 (lumenvox license server installation) The /opt/ dir is empty |
| 21:51.40 | [TK]D-Fender | VJFROMGT, this is NOT a FreePBX support channel. You already know not to bring that crap here. This is like the 100th time you've been told. |
| 21:52.10 | VJFROMGT | tk,, its not a freepbx question, just a general asterisk question |
| 21:53.16 | [TK]D-Fender | VJFROMGT, No, it is YOU having a problem witht he DIALPLAN that FreePBX creates. The call does not DIE, yuo just don't like HOW it processes. And that all FreePBX. |
| 21:53.54 | [TK]D-Fender | VJFROMGT, "How can I set my outbound route to try all 6 trunks before giving up? (at least top down) " <- BS this a BLATANTLY a FreePBX dialplan issue. |
| 21:55.14 | obnauticus | ROFL |
| 21:55.15 | obnauticus | [TK]D-Fender |
| 21:55.18 | obnauticus | Sprint changed my umm |
| 21:55.18 | obnauticus | CID |
| 21:55.24 | obnauticus | to 911 emergency <911> |
| 21:56.23 | *** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 21:58.04 | OloBola | does it look like my installation is quitting ? http://www.pastebin.ca/602034 (lumenvox license server installation) The /opt/ dir is empty, where the server should have been installed |
| 21:58.31 | dillydally | can anybody help me or point me to a link dealing with directed call pickup |
| 21:58.41 | Hmmhesays | someone wouldn't happen to have a comma delimited north american areacode list would they? |
| 21:59.05 | *** join/#asterisk Eminence (n=achin@ip68-227-184-126.ri.ri.cox.net) |
| 22:00.22 | dillydally | i tried this but it still wouldn't work "exten => _77xxxx,1,pickup(${EXTEN:2}@my-extensions) |
| 22:01.06 | *** join/#asterisk MedozasSVR (n=medozas@p54911D64.dip0.t-ipconnect.de) |
| 22:01.17 | bobdraken | how do i tell the ip address of asterisk |
| 22:01.20 | bobdraken | so i can set up x lite |
| 22:01.28 | MedozasSVR | is qwell somewhere here? |
| 22:01.45 | bobdraken | i dont have voip service yet i just want to do inside calls to another phone in my house |
| 22:02.36 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 22:03.13 | bobdraken | anyone? how can i find out the ip address to my asterisk box |
| 22:04.03 | Corydon76-work | bobdraken: /sbin/ifconfig eth0 |
| 22:05.14 | bobdraken | Corydon76-what do you mean |
| 22:05.19 | bobdraken | type that in term? |
| 22:05.33 | Corydon76-work | Correct |
| 22:05.36 | Eminence | pbx_dundi.so and chan_skinny.so seem to be causing * to segfault on exit. i don't use these (so i'm just not loading them), but does anyone want a gdb backtrace? |
| 22:05.46 | bobdraken | so i need to be in the sbin and ifconfig folers |
| 22:05.54 | bobdraken | or just type /sbin/ifconfig eth0 |
| 22:05.59 | Eminence | just type /sbin/ifconfig eth0 |
| 22:06.01 | Corydon76-work | Just type it |
| 22:06.03 | bobdraken | k one sec |
| 22:06.10 | bobdraken | i add that to xlites doman |
| 22:06.12 | bobdraken | for the sip set up |
| 22:06.13 | bobdraken | correct |
| 22:06.48 | [TK]D-Fender | bobdraken, Correct. |
| 22:06.59 | [TK]D-Fender | bobdraken, But like I told you before... STOP. NOW. |
| 22:07.12 | Eminence | (and this is a clean 1.4.6 installation) |
| 22:07.20 | [TK]D-Fender | bobdraken, You are flying blind and we are not here to hand hold you through learning every little thing about * from scratch. |
| 22:07.31 | [TK]D-Fender | bobdraken, Stop and rad THE BOOK. |
| 22:07.33 | x86_ | how do i reset a polycom IP501 again? |
| 22:07.44 | [TK]D-Fender | bobdraken, Don't stop until you've finished through chapter 5. |
| 22:08.41 | Corydon76-work | x86_: Menu, Settings, Advanced, 456, Admin, Reset to Default, Format File System |
| 22:10.37 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 22:11.23 | Defraz | TK http://www.pastebin.ca/602132 here is the info you requested. |
| 22:12.39 | Kwakwa | Does anyone know anything about decibels? Apparently in the UK you can't subject an employee to anything over 85dB over the working day. Some sales person from sennheiser came in and saw our cheap headsets n said "ur breaking the law, I dont think the speaker in the headset adheres to the legislation". Surely the dB of a headset is just the volume so if its too loud we just turn it down? :) |
| 22:12.59 | [TK]D-Fender | ~freepbx |
| 22:12.59 | jbot | i guess freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 22:13.01 | EmleyMoor | Decibels are merelt a ratio |
| 22:13.04 | EmleyMoor | merely |
| 22:14.11 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 22:14.16 | Kwakwa | Indeed, so if the dB is > 85dB they just turn it down? :/ |
| 22:14.49 | EmleyMoor | Kwakwa: You can have a signal-to-noise ratio of 85dB on something extremely quiet |
| 22:15.02 | EmleyMoor | It all depends on what exactly is being measured |
| 22:15.10 | bobdraken | stop now why |
| 22:15.16 | andyd | I asked in here during the day once but think it's probably worth mentioning again -- if someone wishes to give some expansion to the asterisk instructions on www.tollfreeuk.com - or add intructions for other telephony apps, then please let me know. |
| 22:15.16 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:15.19 | EmleyMoor | And as for the guy from Sennheiser, he was looking for business |
| 22:15.22 | andyd | its for free calls to some uk destinations |
| 22:15.48 | Eminence | often when they say something like 85dB it's usually db spl |
| 22:16.08 | Eminence | which is relative to the quietest sound a human can hear |
| 22:16.43 | Kwakwa | I read something about that, the sound pressure level.. how would I work out if the headsets are ok? |
| 22:16.45 | Corydon76-work | 85dB is typically the volume of street traffic |
| 22:17.13 | EmleyMoor | andyd: I see no problem with that... though I use FWD and have an account with VoIPtalk also |
| 22:17.34 | Kwakwa | Surely the headsets should be fine then, there's no cars that ring us up :) |
| 22:17.37 | Eminence | a loud concert, or maybe near a race track is about 120 (if i'm remembering correctly) |
| 22:17.55 | Kwakwa | Apparently Niagra Falls is 90dB |
| 22:18.02 | andyd | You're down the road from me :-) so you can get toll free access to 0800 in a payphone or probably even on your domestic landline - this is for people who don't live in the UK really. But thank you for looking, it is appreciated. |
| 22:18.11 | Corydon76-work | 120dB is supposedly the loudness of a gunshot |
| 22:18.33 | EmleyMoor | andyd: Indeed - I find it interesting to know about such facilities |
| 22:19.44 | Eminence | Corydon76-work, from what distance, abouts? |
| 22:20.02 | Corydon76-work | Eminence: good question, the sources don't say |
| 22:20.22 | Corydon76-work | I assume if you're holding the gun |
| 22:20.41 | Corydon76-work | Average voice is in the range of 50-60dB |
| 22:20.54 | Kwakwa | I'll just tell them to bog off, shame I can't figure out exactly what dB they're at. Cheers guys. |
| 22:20.55 | Eminence | i wonder what the fall off is |
| 22:21.16 | bobdraken | ok how do i stop asterisk from running while i read up |
| 22:21.20 | Corydon76-work | Kwakwa: you can get a device to tell you |
| 22:21.25 | Eminence | i.e. double the distance from the source, what happens to spl? |
| 22:21.33 | dennisharrison | bobdraken, asterisk stop |
| 22:21.35 | Kwakwa | Corydon76-work: Whats it called? |
| 22:21.42 | *** part/#asterisk MedozasSVR (n=medozas@p54911D64.dip0.t-ipconnect.de) |
| 22:21.45 | Kwakwa | bobdraken: Use screen |
| 22:21.47 | bobdraken | dennisharrison: does it auto start when i start this box |
| 22:21.55 | bobdraken | Kwakwa: use what screen |
| 22:21.59 | Corydon76-work | Kwakwa: oddly enough, a decibel meter |
| 22:22.01 | dennisharrison | bobdraken, I didn't build or configure your box :) |
| 22:22.02 | Kwakwa | `screen` |
| 22:22.12 | dennisharrison | screen = linux nerds katana |
| 22:22.13 | bobdraken | dennisharrison: how do i check the config for it lol |
| 22:22.22 | bobdraken | screen as in terminal |
| 22:22.26 | Eminence | everybody should be using screen :) |
| 22:22.27 | dennisharrison | bobdraken, usually you could check /etc/rc.local |
| 22:22.38 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 22:22.42 | dennisharrison | screen = new hottness |
| 22:22.46 | dennisharrison | only ... its old :) |
| 22:22.51 | Kwakwa | screen is the bees knees, heh |
| 22:22.56 | dennisharrison | bobdraken, what distro? |
| 22:22.56 | Kwakwa | Corydon76-work: haha, ta :) |
| 22:22.59 | bobdraken | ubuntu |
| 22:23.06 | Kwakwa | apt-get install screen |
| 22:23.07 | bobdraken | im very very new to asterisk |
| 22:23.14 | bobdraken | Kwakwa: what is screen? |
| 22:23.21 | bobdraken | and new to linux hehe |
| 22:23.27 | Eminence | do a google search for "gnu screen" |
| 22:23.28 | dennisharrison | bobdraken, so is this server edition or you have gnome in front of you? |
| 22:23.34 | dennisharrison | screen is like 'tabs' for terminal |
| 22:23.45 | Kwakwa | "Screen is a full-screen window manager that multiplexes a physical ter- minal between several processes (typically interactive shells)" |
| 22:23.46 | bobdraken | gnome |
| 22:23.51 | Corydon76-work | bobdraken: you need to read some books on Linux and Asterisk before you go any further |
| 22:24.02 | bobdraken | yeah whats the link to the asterisk box |
| 22:24.03 | bobdraken | and... |
| 22:24.07 | Corydon76-work | ~thebook |
| 22:24.08 | jbot | hmm... thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:24.12 | dennisharrison | ok so just go into system / admin / services (I think) |
| 22:24.13 | bobdraken | should i go ahead and set up an account with skype |
| 22:24.16 | dennisharrison | and deselect asterisk |
| 22:24.18 | bobdraken | so i can use my skype phone |
| 22:24.25 | dennisharrison | although depending how you installed it... I doubt it starts at boot |
| 22:24.32 | Corydon76-work | Don't buy the Asterisk book now... there's a new edition due out in August |
| 22:24.46 | dennisharrison | bobdraken, uhh .. yeah well thats skype |
| 22:24.48 | bobdraken | ooo its a buyable book.. i thought it was something i could read online |
| 22:24.59 | Corydon76-work | It's both |
| 22:25.14 | bobdraken | i guess im asking should i use skype for a voip service.. or would yall sugest something else |
| 22:25.16 | Corydon76-work | Kwakwa: http://scientificsonline.com/product.asp?pn=3038732&sid=shopmore&eid=shopmore&mr:trackingCode=6182C82C-B728-DC11-9D0E-000423C27502&mr:referralID=NA&bhcd2=1183501285 |
| 22:25.30 | EmleyMoor | bobdraken: Something else... anything else (!) |
| 22:25.45 | bobdraken | lol see thats why i asked.. EmleyMoor what service would you sugest |
| 22:26.00 | Corydon76-work | You're welcome to use Skype, but it's not going to interoperate with Asterisk |
| 22:26.09 | bobdraken | ahh good thing i didnt |
| 22:26.11 | bobdraken | what should i use |
| 22:26.43 | dennisharrison | bobdraken, |
| 22:26.50 | dennisharrison | what do you want to do? |
| 22:26.50 | bobdraken | dennisharrison: yes |
| 22:26.52 | Corydon76-work | http://www.voip-info.org/wiki/view/VOIP+Service+Providers+Residential |
| 22:26.57 | Kwakwa | thanks Corydon76-work |
| 22:26.58 | Corydon76-work | There's a whole list of them |
| 22:27.09 | bobdraken | well... asterisk box, voip service and play around.. learn pretty much |
| 22:27.11 | bobdraken | set up a pbx |
| 22:27.16 | EmleyMoor | Something based on either SIP or IAX2 |
| 22:27.20 | dennisharrison | bobdraken, for incoming calls? |
| 22:27.22 | EmleyMoor | I use VoIPtalk |
| 22:27.30 | dennisharrison | I have used a few |
| 22:27.35 | dennisharrison | voicepulse isn't bad for me |
| 22:27.37 | bobdraken | yes |
| 22:27.37 | EmleyMoor | FreeWorldDialup is good for starting out |
| 22:27.37 | dennisharrison | its cheap |
| 22:27.47 | bobdraken | how about |
| 22:27.49 | bobdraken | nufone |
| 22:28.13 | dennisharrison | you should use something iax2 based for correctness, and less headaches :) |
| 22:29.15 | *** join/#asterisk demiv (n=demiv@134.42.128.66.PPPoECali.dynamic.telesat.net.co) |
| 22:29.21 | bobdraken | hum... |
| 22:29.30 | bobdraken | do those have a monthly charge |
| 22:30.05 | dennisharrison | usually a number will net you monthly |
| 22:30.08 | EmleyMoor | bobdraken: VoIPtalk make monthly charges for some kinds of number, one-off for others, and you have to pay for calls |
| 22:30.16 | dennisharrison | along with a channel |
| 22:30.24 | dennisharrison | and then your minutes are billed at whatever your rate is |
| 22:30.36 | bobdraken | dang that stinks |
| 22:30.43 | bobdraken | skype is only like what 40 bucks for a year |
| 22:30.51 | dennisharrison | skype is dog shit |
| 22:31.02 | EmleyMoor | It all depends on what you want to achieve |
| 22:31.16 | dennisharrison | true EmleyMoor |
| 22:31.29 | dennisharrison | you might not need a pbx bobdraken |
| 22:31.41 | dennisharrison | but if you want to learn asterisk, then skype is out of the question |
| 22:31.45 | dennisharrison | well .. afaik |
| 22:31.48 | EmleyMoor | VoIPtalk is free, with a free 0871 number, if you don't call chargeable numbers with it |
| 22:31.53 | bobdraken | kk well what provider do yall ulse |
| 22:31.55 | Corydon76-work | You probably shouldn't be signing up to any service at all until you've successfully configured Asterisk and your SIP phone to talk |
| 22:31.56 | bobdraken | *use |
| 22:31.58 | Voicemeup | lol skype |
| 22:32.07 | bobdraken | EmleyMoor: voiptalk is free? |
| 22:32.30 | Voicemeup | basic math you get waht you pay for |
| 22:32.40 | Voicemeup | pay for 0 get shitty service and no support |
| 22:32.41 | EmleyMoor | Free for VoIP-to-VoIP and PSTN-to-VoIP (0871 number) |
| 22:32.48 | Strom_M | i wish skype to die a horrible painful pointy death |
| 22:32.55 | Voicemeup | pay 0.004 blended on all calls get no service and down time |
| 22:32.58 | Voicemeup | you get the point |
| 22:33.04 | dennisharrison | well either that or stop trashing what should have been a decent protocol Strom_M |
| 22:33.24 | Voicemeup | pay 0.02 and get pstn |
| 22:33.32 | Voicemeup | and or 10 cents and get pots term |
| 22:33.39 | dennisharrison | bobdraken, heres the question, what do you want to do in the long run? |
| 22:33.49 | bobdraken | ahh i dono... well |
| 22:33.51 | bobdraken | i do know |
| 22:33.53 | Corydon76-work | I pay .02/minute for my 800 termination |
| 22:33.57 | dennisharrison | you want to be able to deploy asterisk boxes in a call center environ? ;p |
| 22:34.04 | dennisharrison | or you want to play with it? |
| 22:34.10 | dennisharrison | or you want to use it for small business? |
| 22:34.15 | *** join/#asterisk smultron (n=lauren@cpe-72-179-47-78.austin.res.rr.com) |
| 22:34.38 | Defraz | Okay I think I have it figured out. |
| 22:34.39 | bobdraken | well, small business what do you mean |
| 22:34.44 | bobdraken | what guess defraz |
| 22:34.57 | Defraz | haha |
| 22:34.59 | Defraz | sorry |
| 22:35.00 | dennisharrison | bobdraken, like ... why do you want to learn asterisk? |
| 22:35.04 | dennisharrison | what is the goal ? |
| 22:35.09 | Defraz | Okay, my issue with the transfering |
| 22:35.19 | bobdraken | learn about the phone system... cid ani pbx |
| 22:35.31 | bobdraken | set up a voicemail system... pbx |
| 22:35.33 | bobdraken | just learn |
| 22:35.59 | Strom_M | for the difference between CID and ANI, may I suggest a good book on SS7? |
| 22:36.04 | dennisharrison | bobdraken, what type of internet connection do you have ? |
| 22:36.09 | bobdraken | cable |
| 22:36.18 | dennisharrison | do you have a land line in your house now? |
| 22:36.22 | bobdraken | Strom_M: what book |
| 22:36.25 | syzygyBSD | can't wait till I get on the SS7 network.. shouldn't be too long |
| 22:36.26 | Voicemeup | we charge for 800 term lol 1 penny per call |
| 22:36.26 | bobdraken | dennisharrison: yes |
| 22:36.34 | dennisharrison | bobdraken, just get an fxo card then |
| 22:36.36 | dennisharrison | but NOT a cheap one |
| 22:36.38 | Voicemeup | so no call center decides to dropp 100,000 calls a day |
| 22:36.41 | bobdraken | whats fxo |
| 22:36.44 | dennisharrison | and learn all those features first |
| 22:36.50 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 22:36.55 | bobdraken | and what kind of business |
| 22:36.57 | dennisharrison | you plug a land line into your box using fxo |
| 22:37.00 | bobdraken | are people offering with asterisk |
| 22:37.15 | bobdraken | i can get a fxo box from walmart correct? |
| 22:37.21 | JT | no |
| 22:37.21 | Strom_M | lol |
| 22:37.25 | dennisharrison | are you a bot? |
| 22:37.37 | syzygyBSD | well, I know some major hosting companies use asterisk for their phone system |
| 22:37.43 | bobdraken | hi im sally visit my site for pics |
| 22:37.46 | bobdraken | lol no im not a bot |
| 22:37.59 | dennisharrison | that was a bot'ish reply :) |
| 22:38.06 | bobdraken | lol |
| 22:38.10 | syzygyBSD | ya, bots use lol a lot |
| 22:38.18 | bobdraken | jesh i was kidding |
| 22:38.24 | dennisharrison | we are switching our legacy clients who are coming out of term on their lucent and avaya equipment to asterisk |
| 22:38.32 | syzygyBSD | I don't think he could pass the turning test. |
| 22:38.35 | dennisharrison | ... and asterisk really works, for the most part :) |
| 22:38.45 | dennisharrison | syzygyBSD, lol |
| 22:39.13 | bobdraken | ill set up my M$ box with my xact phone and a voip service |
| 22:39.26 | bobdraken | read up on asterisk and connect my other computers voip to the asterisk |
| 22:39.53 | syzygyBSD | ha, bots need more typo's! |
| 22:40.12 | bobdraken | sound like a plan? |
| 22:41.00 | syzygyBSD | typo's are only part of a plan, need a good vocabulary too |
| 22:41.01 | dennisharrison | bobdraken, just buy a single port fxo card from somewhere (suggestions anyone? no clue on this one) |
| 22:41.05 | dennisharrison | and cram it in your box |
| 22:41.21 | dennisharrison | then get asterisk working with it to where when you call you hear the test messages, etc |
| 22:41.24 | EmleyMoor | Really you want a TDM01B |
| 22:41.25 | syzygyBSD | I'd just suggest a out of box one for testing |
| 22:41.39 | smultron | if i need an asterisk box to take 6 POTS lines in, i'd need 2 cards, right? one card with 4, another with 2...? |
| 22:41.40 | bobdraken | how do you call to ur own asterisk box |
| 22:41.44 | bobdraken | whats the number set to it |
| 22:41.46 | dennisharrison | then learn about dialplans, context, etc until you can get outgoing calls to use your local line for local calls |
| 22:41.53 | EmleyMoor | smultron: No - you can get an 8-port capable card |
| 22:41.58 | syzygyBSD | bobdraken: all of them? |
| 22:42.00 | smultron | :o |
| 22:42.03 | smultron | whoah |
| 22:42.16 | bobdraken | ok last question promise |
| 22:42.17 | smultron | EmleyMoor: where are those ones? |
| 22:42.20 | Corydon76-work | TDM800P |
| 22:42.25 | Voicemeup | smultron or get the new linksys spa8000 |
| 22:42.26 | bobdraken | what service should i sign up with |
| 22:42.35 | Voicemeup | in september with 8 analog ports and tie to asterisk |
| 22:42.48 | Voicemeup | bobdraken really depends on your needs |
| 22:42.50 | dennisharrison | EmleyMoor, syzygyBSD ... yeah I don't use those things, although we did just replace a fried r4fxo-ec |
| 22:43.20 | EmleyMoor | I have a TDM31B - one line on it currently stands "spare" |
| 22:43.25 | syzygyBSD | all I really use is pri cards |
| 22:43.26 | bobdraken | whats a good cheap provider that works with asterisk |
| 22:43.37 | Corydon76-work | smultron: http://www.digium.com/en/wheretobuy/digiumdirect/productview.php?product_code=1TDM808BF-01 |
| 22:43.41 | EmleyMoor | bobdraken: Providing what exactly? |
| 22:43.49 | bobdraken | voip service |
| 22:43.53 | bobdraken | i guess i dono im a newb lol |
| 22:44.03 | Voicemeup | he could be talking about hosting/ voip or maybe therapy |
| 22:44.07 | Voicemeup | ;) |
| 22:44.11 | dennisharrison | lol |
| 22:44.12 | EmleyMoor | Yes - but providing calling to PSTN? Incoming PSTN numbers? Just plain VoIP? |
| 22:44.23 | bobdraken | pstn? |
| 22:44.26 | Corydon76-work | I really like the 800P... finally, an analog card that works with fax machines |
| 22:44.31 | dennisharrison | public switch telephone network |
| 22:44.34 | dennisharrison | voip-info.org |
| 22:44.37 | smultron | Corydon76-work: thanks, looking now |
| 22:44.52 | bobdraken | land line |
| 22:44.52 | dennisharrison | Corydon76-home, really eh? |
| 22:44.57 | bobdraken | yeah i wana call land lines |
| 22:45.07 | dennisharrison | just get a damn fxo card bobdraken :) |
| 22:45.17 | dennisharrison | put a voip provider in there later |
| 22:45.19 | Corydon76-work | dennisharrison: yeah, we've been testing it in that configuration for a while now and haven't had any troubles |
| 22:45.21 | Voicemeup | heeheh , that yeah prefix in your phrase would suggest you write down on a piece of paper what you require before looking for it |
| 22:45.21 | EmleyMoor | bobdraken: In which country are you based? |
| 22:45.22 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 22:45.29 | dennisharrison | Corydon76-home, what fax machines? |
| 22:45.44 | bobdraken | usa |
| 22:45.49 | Corydon76-work | dennisharrison: el cheapo one we rescued from a dumpster, actually |
| 22:46.01 | dennisharrison | Corydon76-home, haha, get out |
| 22:46.10 | Corydon76-work | Well, almost. |
| 22:46.21 | Corydon76-work | It would have ended up there if we hadn't taken it |
| 22:46.23 | dennisharrison | Corydon76-home, we do some pretty stupid crap to get most of the more expensive faxes to work |
| 22:46.29 | Corydon76-work | It may still end up in the dumpster |
| 22:46.29 | dennisharrison | wind up using them as a printer :) |
| 22:46.34 | dennisharrison | lol |
| 22:47.01 | Corydon76-work | but both my boss and I are habitual packrats, so... |
| 22:47.05 | dennisharrison | hey you guys been looking at the openmoko project? |
| 22:47.24 | dennisharrison | Corydon76-home, hell, thats us too... I think it comes with the telecom industry |
| 22:47.29 | dennisharrison | I still have some old key systems |
| 22:47.55 | smultron | Corydon76-work: TDM808B can support 6 FXO lines? and I can get another for FXS? |
| 22:48.01 | Strom_M | what about pac-man? |
| 22:48.13 | dennisharrison | Strom_M, I was thinking that ... or some sort of midi device :) |
| 22:48.26 | Strom_M | i'm a pack rat too |
| 22:48.35 | Strom_M | trying to break that habit tho |
| 22:48.44 | dennisharrison | the woman keeps throwing stuff out |
| 22:48.53 | dennisharrison | I will sneak out behind her and go re'acquire :) |
| 22:49.14 | dennisharrison | my 'shop' looks like a ministorage :) |
| 22:49.42 | bobdraken | asterisk stop in the terminal didnt work |
| 22:50.06 | Corydon76-work | smultron: no, you'd need another card for that |
| 22:50.23 | Corydon76-work | 808 supports 8 FXO lines |
| 22:50.29 | dennisharrison | asterisk -rvvvv then and tell me what you get bobdraken |
| 22:51.08 | bobdraken | rvvvv |
| 22:51.11 | bobdraken | 4 v's? |
| 22:51.29 | dennisharrison | ... |
| 22:51.37 | *** join/#asterisk saftsack (n=saftsack@pD9E0605E.dip.t-dialin.net) |
| 22:51.37 | dennisharrison | use as many v's as you want :) |
| 22:51.52 | dennisharrison | thats the verbosity level |
| 22:52.00 | bobdraken | says its running |
| 22:52.12 | *** join/#asterisk obnauticus (n=obnautic@c-71-59-162-60.hsd1.wa.comcast.net) |
| 22:52.13 | dennisharrison | so you are now in the asterisk cli? |
| 22:52.29 | bobdraken | ooo sweet |
| 22:52.38 | *** join/#asterisk stack_ (n=stack@198.30.100.203) |
| 22:53.05 | dennisharrison | ? |
| 22:53.08 | dennisharrison | tell it stop now |
| 22:53.26 | stack_ | I upgraded my * box from ubuntu breezy to dapper. When I run 'ztcfg', I get 'ZT_SPANCONFIG failed on span 1: Invalid argument (22)' |
| 22:53.29 | stack_ | any ideas? |
| 22:53.50 | dennisharrison | stack_, |
| 22:53.53 | dennisharrison | what version of zaptel? |
| 22:54.00 | *** join/#asterisk tako-san (n=Tako-san@24.68.129.29) |
| 22:54.03 | stack_ | I just compiles the 1.4.3 version |
| 22:54.12 | dennisharrison | what version of asterisk ? |
| 22:54.18 | stack_ | 1.2.7.1 |
| 22:54.27 | dennisharrison | compile zap 1.2 |
| 22:54.40 | stack_ | are the versions supposed to match for the two? |
| 22:54.50 | dennisharrison | yep |
| 22:54.58 | *** part/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 22:55.00 | stack_ | I was afraid of that, but I didn't see it metioned anywhere |
| 22:55.06 | bobdraken | ok how do i stop asterisk now |
| 22:55.22 | bobdraken | stop now |
| 22:55.23 | bobdraken | i see it |
| 22:58.23 | *** join/#asterisk cnet2 (n=nada@196.40.38.130) |
| 22:58.51 | cnet2 | hi guys, what is the best international voip carrier? I'm having issues landing in argentina with callcentric |
| 22:59.36 | cnet2 | no one ?.. any suggestions? |
| 22:59.49 | Eminence | anybody in here use chan_gtalk? |
| 23:00.45 | cnet2 | ..another interesting subject.. I tried to use it, and it didn't create one of the 2 channels.. (don't remember which).. but registration on asterisk was a success |
| 23:01.42 | *** join/#asterisk neon (n=davor@189.148.26.145) |
| 23:01.43 | tzafrir_laptop | EmleyMoor, here? |
| 23:05.08 | JT | arrgh, just been browsing the scrollback, why did i do that |
| 23:05.23 | obnauticus | Is the SMS function automatically compiled during ./configure and make with Asterisk 1.4 |
| 23:05.44 | *** part/#asterisk woolbeo (n=woolbeo@exchange.services.daqe.com) |
| 23:05.50 | neon | where can i get the default config for asterisk? |
| 23:05.58 | *** join/#asterisk guillote_GNU (n=guillote@host252.200-82-63.telecom.net.ar) |
| 23:06.12 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 23:06.23 | JT | the sms function... i don't think it can do much, obnauticus |
| 23:06.28 | obnauticus | K |
| 23:06.39 | obnauticus | It's free to SEND SMS's through the internet right? |
| 23:06.43 | obnauticus | or through your own thing |
| 23:06.45 | obnauticus | not using a telco |
| 23:06.51 | JT | ... |
| 23:06.57 | obnauticus | i don't know what im talking about |
| 23:06.57 | obnauticus | is it? |
| 23:07.04 | JT | i assume you want the sms to get so someone with a phone |
| 23:07.10 | obnauticus | ya |
| 23:07.15 | JT | s/get so/get to/ |
| 23:07.24 | JT | no, it's usually not free |
| 23:07.36 | obnauticus | Sofa king stupid |
| 23:07.37 | JT | anything that involves connecting to a phone network is usually not free |
| 23:07.47 | obnauticus | sending* |
| 23:07.49 | obnauticus | to a phone network |
| 23:07.50 | obnauticus | ahh |
| 23:08.00 | JT | apparently it costs money to maintain phone networks ;) |
| 23:08.06 | obnauticus | Not if you are me. |
| 23:08.08 | obnauticus | LOL |
| 23:08.11 | Jingles | much to my boss's chagrin. |
| 23:08.13 | obnauticus | In a perfect world |
| 23:08.16 | obnauticus | peopel will do it for fun. |
| 23:08.26 | obnauticus | And it will be 100x better. |
| 23:08.29 | obnauticus | and free |
| 23:08.29 | obnauticus | lol. |
| 23:08.51 | Strom_M | i do it for fun for myself |
| 23:08.58 | Strom_M | then i charge my clients a lot to do it for them |
| 23:13.09 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 23:14.07 | *** join/#asterisk saftsack (n=saftsack@pD9E0605E.dip.t-dialin.net) |
| 23:15.32 | aptura | This industrial phone has penatrated a commmercial complex, a condo complex a house and a townohouse complex before the audio quality was starting to become difficult to hear. Not bad :) up to 36 of these phones can communicate between each other. |
| 23:16.09 | JT | using only 5gigawattas of HF |
| 23:16.22 | aptura | hehe |
| 23:25.33 | OloBola | how do I get the version number for * |
| 23:25.43 | [hC] | show version |
| 23:25.44 | OloBola | thank you. |
| 23:25.48 | [hC] | np |
| 23:28.14 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:30.03 | Hmmhesays | hmm is it possible for func_odbc to return more than one value from a query? |
| 23:32.23 | OloBola | so what version am I running: SVN-trunk-r7230 |
| 23:32.37 | JT | trunk |
| 23:33.16 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 23:33.55 | OloBola | I think I'm looking for 1.2 or 1.4 or whatever. Am I prior to 1 ? |
| 23:34.34 | JT | no, trunk is the beta release, will be the next 1.6 |
| 23:34.46 | JT | then again i don't know how old a trunk version you are using :) |
| 23:35.15 | OloBola | I see. |
| 23:39.16 | *** join/#asterisk cynjut (n=burgess@ip68-225-142-95.om.om.cox.net) |
| 23:40.01 | cynjut | Hello All |
| 23:40.30 | cynjut | Hello Anyone? |
| 23:42.37 | cynjut | Anyone interested in a helping me solve a conundrum? |
| 23:46.21 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 23:47.53 | *** part/#asterisk bobdraken (n=haxxor@75-138-105-182.dhcp.hckr.nc.charter.com) |
| 23:49.34 | OloBola | yum LumenVoxSRE stops download one file or another every single time |
| 23:49.48 | OloBola | ing |
| 23:50.09 | OloBola | primary.xml.gz 33% |======== | 464 kB 00:06 ETA |
| 23:52.46 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 23:56.53 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
