| 00:04.35 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 00:04.39 | glaz | Will a cross over cable work on a T1 to PRI card? |
| 00:05.02 | JT | you're connecting what with what? |
| 00:05.15 | glaz | a T1 to a Digium T122 Card |
| 00:09.09 | JT | you mean TE122? |
| 00:15.03 | gandhijee | anyone know what toning tiawan uses? |
| 00:15.05 | *** join/#asterisk anthm (n=anthm@70-9-230-238.area4.spcsdns.net) |
| 00:15.05 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:16.29 | gandhijee | <PROTECTED> |
| 00:16.57 | Greek-Boy | the volume on my pri channels are a lil low. what levels are recommended for txgain and rxgain? |
| 00:19.02 | *** join/#asterisk robeph (n=robf@router.asteriasgi.com) |
| 00:19.17 | robeph | is it possible to pass an outbound call over dundi? |
| 00:19.50 | drmessano | uh yes |
| 00:19.56 | [hC] | Kinda strange. Anyone seen an issue where calling a busy number (over IAX or PRI) plays busy tone for only a second or two, before the call hangs up on you automatically? |
| 00:19.58 | robeph | i mean, I got it working for extensions, dialing say 5000 from this box goes across dudni to the other, but when i set it up to dial with _9NXXNXXXXXX it says no |
| 00:20.02 | jblack | omg. callwithus is so the sux0rs |
| 00:20.20 | drmessano | They are? |
| 00:20.29 | robeph | and I fig it should just treat this as another extension, so not sure why it's ignorin it when I try n pass it |
| 00:20.35 | jblack | THey have 8 or 9 different routes. The higher the route you pick, the more it costs. |
| 00:20.58 | drmessano | I thought they were the best thing since sliced bread? |
| 00:21.12 | jblack | Some of the routes accept CID. Some don't. Some have horrid jitter. One had latency of 500ms. Some ring while connecting, some don't. |
| 00:21.29 | *** join/#asterisk boster (n=boster@chi-pat.cashnetusa.com) |
| 00:21.30 | jblack | And the very best one on the list? It does enum lookups, and fails the call if the enum fails. |
| 00:22.16 | drmessano | Nice |
| 00:22.31 | jblack | It's more random than a 14 year old girl's feelings |
| 00:23.19 | jjshoe | mm 14 year old girls... |
| 00:23.26 | jjshoe | http://www.jailbait-gallery.com |
| 00:23.44 | drmessano | You have to be kidding me |
| 00:23.49 | drmessano | GTFO with that crap |
| 00:24.20 | boster | I'm trying to use the MEMBERINTERFACE variable with little to no luck in asterisk 1.4.17. The setinterfacevar=yes option is in queues.conf and I can't seem to access it either by passing it as an argument to and AGI or by having the AGI attempt to pull it directly. |
| 00:24.25 | jblack | Heh. It's a dropped domain ad page, is all |
| 00:24.26 | boster | any ideas? |
| 00:25.10 | jblack | drmessano: Who do you use, how much do they charge, and do they support CID modification? |
| 00:25.30 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:25.39 | Qwell | jblack: you checked? |
| 00:25.51 | jblack | qwell: Every single one, at least thrice. |
| 00:25.58 | jblack | qwell: Oh, the page? Yeah. |
| 00:26.02 | drmessano | Les.net... the one you said sucked balls and that callwithus was so much better than. |
| 00:26.30 | jblack | I still stand by that. |
| 00:26.43 | drmessano | Sounds like callwithus is doing great |
| 00:26.48 | jblack | Anyone that charges my credit card money, then claims they never took it, gets graded a 0 to me. |
| 00:27.08 | jblack | Heh. I'd hold both of them back a grade at this moment. |
| 00:27.17 | jblack | Perhaps send them to the trade-school fair. |
| 00:27.48 | jjshoe | you guys are no good at jokes, lol |
| 00:28.19 | jblack | I'm not funny after 50 serial phone calls to myself |
| 00:28.46 | jblack | Clearly, I need a new upstream. |
| 00:28.59 | Jumpie | hey guys |
| 00:29.06 | JT | he's a Serial Caller |
| 00:29.08 | Jumpie | make clean for asterisk-addons i get this error |
| 00:29.09 | Jumpie | Package gtk+-2.0 was not found in the pkg-config search path. |
| 00:29.09 | Jumpie | Perhaps you should add the directory containing `gtk+-2.0.pc' |
| 00:29.09 | Jumpie | to the PKG_CONFIG_PATH environment variable |
| 00:29.15 | Jumpie | majjor problerm? |
| 00:29.19 | Jumpie | sorry didnt mean to spam so many |
| 00:29.47 | robeph | http://pastebin.ca/979002 there, someone take a look and perhaps gimme a reason why this isn't working |
| 00:32.28 | *** join/#asterisk Simzilla (n=Sim@bas9-quebec14-1168091844.dsl.bell.ca) |
| 00:32.58 | Simzilla | hello, can I ask a question about asterisk? |
| 00:33.01 | robeph | all pertanent info should be there |
| 00:33.38 | drmessano | Q&A time is between 11:01 and 11:02PM on June 31st |
| 00:33.41 | drmessano | Sorry :( |
| 00:33.49 | robeph | drmessano: :( |
| 00:33.53 | robeph | drmessano: thats so long away |
| 00:34.02 | drmessano | Longer than you think |
| 00:34.10 | robeph | is there even a 31-6? |
| 00:34.28 | drmessano | plays the windows 98 tada.wav |
| 00:34.32 | robeph | hah |
| 00:34.39 | robeph | so anyhow |
| 00:34.41 | Simzilla | lol june 31st.... |
| 00:34.49 | robeph | about http://pastebin.ca/979002 what is the failure here... |
| 00:34.58 | drmessano | robeph: Please don't pastespam |
| 00:35.09 | robeph | >:O |
| 00:36.00 | drmessano | Asking for help is one thing, shoving your problem down everyone's throats with repeated links to a pastebin is just bad karma, dude |
| 00:36.11 | drmessano | baaad... bad karma |
| 00:36.16 | jjshoe | robeph let me look, I won't be a grumpy asshole, just a sec. |
| 00:36.23 | robeph | drmessano: :p |
| 00:36.27 | robeph | it worked see? |
| 00:36.36 | robeph | that was the master plan, a) paste it until someone is angered |
| 00:36.39 | drmessano | jjshoe works for Fonality.. he can help |
| 00:36.45 | robeph | b) someone feels bad that angry person assaults me, |
| 00:36.51 | robeph | c) person helps me =) |
| 00:36.52 | Simzilla | we did setup at work our asterisk pretty much without a problem, we used Polycom 330 phones, and we have people working from home that we want them to be able to use it. The phone at home doesn't seem to be able to register even with the phone ip in DMZ, is there any hints we should know? |
| 00:36.53 | drmessano | Who is angry? |
| 00:36.57 | robeph | I'm not serious :p |
| 00:36.59 | robeph | so no one |
| 00:37.16 | drmessano | No, you're just sounding like a douche, that's all |
| 00:37.19 | jjshoe | robeph what exactly are you attempting/hoping to do? |
| 00:37.25 | drmessano | Karma.. see.. Karma.. |
| 00:37.35 | *** join/#asterisk cyberjackal (n=jackal@pool-70-109-57-67.clppva.fios.verizon.net) |
| 00:37.47 | jjshoe | Simzilla forward 5060. |
| 00:38.01 | robeph | jjshoe: well at the moment there is no provider for the box I'm dialing out, I just want this to pass the call to numbers off across dundi and out their provider on the remote pbx |
| 00:38.16 | Simzilla | we did that, we also tried to forward 15060 in case ISP block 5060 and doesn't work either |
| 00:38.21 | robeph | since dundi is already in place, figured I'd just pass ito ut that way rather than making trunks and dicking with the dialplan |
| 00:38.36 | robeph | until they get a provider for that box that I'm trying to set up |
| 00:39.17 | robeph | i just want it to dial across and out the foreign boxes trunk... |
| 00:39.29 | jjshoe | robeph I've never used dundi. |
| 00:39.45 | robeph | heh no one has it seems ;p |
| 00:40.07 | jjshoe | robeph maybe that's a sign? |
| 00:40.32 | jjshoe | Simzilla what happens when they take the routers out of the picture? |
| 00:40.56 | robeph | yeah, could be heh, except its just odd that it works for the 40XX extensions |
| 00:41.07 | robeph | but not the outward call |
| 00:41.09 | Simzilla | it's ppoe, can't be taking out of the picture :S |
| 00:41.52 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 00:42.01 | drmessano | Can other users register to the * box externally? |
| 00:42.29 | Qwell | ~sipnat |
| 00:42.30 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 00:42.41 | Simzilla | is there anyway to test the connection directly like telnet or something I know telnet won't work because of udp but something to see if the problem is from the asterisk internet connection or home |
| 00:42.41 | Nugget | telnet is eeeeeeevil! |
| 00:42.55 | drmessano | Can other users register to the * box externally? |
| 00:44.01 | Simzilla | not at all |
| 00:44.14 | drmessano | Follow the SIP NAT guides |
| 00:44.19 | drmessano | Thats your issue |
| 00:46.45 | Jumpie | drmessano, |
| 00:46.54 | *** join/#asterisk ectospasm (n=ectospas@c-71-207-229-248.hsd1.al.comcast.net) |
| 00:46.54 | Jumpie | im vanilla :) just getting ready to start config'in |
| 00:47.29 | drmessano | About time |
| 00:48.12 | Jumpie | yeah |
| 00:48.17 | Jumpie | it was like having a tumor removed |
| 00:49.48 | *** join/#asterisk Jumpie (n=jumpie@pool-96-231-155-171.washdc.fios.verizon.net) |
| 00:49.50 | Jumpie | oops |
| 00:50.22 | drmessano | ~n00bkfletcher |
| 00:50.23 | jbot | Everyone was a n00b once, but if you're a non-listening n00b, expect a kvelching from the n00bkfletcher! |
| 00:50.29 | boster | I'm trying to use the MEMBERINTERFACE variable with little to no luck in asterisk 1.4.17. The setinterfacevar=yes option is in queues.conf and I can't seem to access it either by passing it as an argument to and AGI or by having the AGI attempt to pull it directly. Any idea where I went wrong? |
| 00:50.45 | Jumpie | lol |
| 00:51.06 | C4colo | is there a SayDigits() function that can work like Background() where you can listen for DTMF while it is playing the digits? |
| 00:51.31 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-fa5d4e4eb2a04f9d) |
| 00:52.50 | glaz | Anyone ever seen this error ? wcte12x[p]: Unable to ping the DSP (2)! |
| 00:53.17 | Jumpie | drehlecom, |
| 00:53.19 | Jumpie | er drmessano |
| 00:53.31 | Jumpie | am i good with gsm only core sound packages? |
| 00:54.10 | drmessano | There will be some overhead for transcoding, but not terrible bad.. if you have the drive space, get em all |
| 00:54.29 | C4colo | the format doesn't really matter until you have a lot of call volume where the transcode time is a problem |
| 00:54.34 | Jumpie | i dont have a ton of hd space for this test server |
| 00:54.41 | drmessano | How much space? |
| 00:54.45 | Jumpie | ah well my target audience is 25 or less |
| 00:54.47 | Jumpie | 80gb |
| 00:54.51 | drmessano | Oh jesus |
| 00:54.52 | Jumpie | i was thinkin maybe wav and gsm? |
| 00:55.02 | Jumpie | :P this is MY test server, not what ill give others |
| 00:55.10 | C4colo | although some sound files are different for example the Meetme Menu file, whatever it is called, is the full menu in GSM and only the first half in WAV/ulaw/alaw/etc |
| 00:55.22 | drmessano | You do realize you could get by with 10GB, right? |
| 00:55.23 | C4colo | I delete the other versions of the meetme menu and just leave the gsm one |
| 00:55.25 | drmessano | Download them all |
| 00:55.29 | Jumpie | drmessano, ah...k |
| 00:56.04 | C4colo | I would suggest ulaw and gsm instead of wav, if you will be allowing ulaw and gsm codecs primarily |
| 00:56.16 | Jumpie | right |
| 00:56.22 | Jumpie | ok im going whole 9 yards :D |
| 00:56.24 | Jumpie | thanks |
| 00:56.33 | C4colo | and ulaw files are slightly smaller than the wav ones too |
| 00:56.53 | Jumpie | i was also looking at, ultimately once im more familiar, to do a move or copy/purge of cdr records every X days |
| 00:56.57 | Jumpie | to another server |
| 00:57.14 | *** join/#asterisk Kumbang (n=dsp@167.205.24.69) |
| 00:57.25 | Jumpie | i know you can do rotating but i think i wanna do it offl to another server, but i think ill just get the box going as is for now :P |
| 00:57.44 | C4colo | you can set up the odbc cdr function to just save all cdr to another sql server |
| 00:58.24 | drmessano | Howdy Kumbang from Bangkok, how are you? |
| 00:58.41 | C4colo | by the way Jumpie, did you get a new chair? |
| 00:58.47 | Jumpie | haha |
| 00:58.49 | Jumpie | not till payday |
| 00:58.54 | Jumpie | im sitting on the floor at the other end of my office |
| 00:58.57 | C4colo | lol |
| 00:58.59 | Jumpie | heh |
| 00:58.59 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 00:59.08 | C4colo | I need a new chair too, but same thing, not until payday |
| 00:59.22 | Jumpie | im apprehensive to get a small rack cabinet because im gonna move shortly, and im currently on the 4th floor of a condo community and dont wanna drag it up :P |
| 00:59.52 | C4colo | lol |
| 01:00.41 | C4colo | yea, I just got a cheap set of metal shelves from target or somewhere and put my servers on there, I only have one 1u and I'm shipping it off to a datacenter in a week anyway, so no need for a rack (even though I would like one) |
| 01:00.55 | C4colo | but, honestly, rack-mount servers, especially 1u are way too loud |
| 01:01.08 | C4colo | even locked away in the basement the thing is like a hoover running 24/7 |
| 01:01.17 | Jumpie | really? |
| 01:01.26 | Jumpie | hmm im considering gettin 1ru for my servers |
| 01:01.31 | *** part/#asterisk boster (n=boster@chi-pat.cashnetusa.com) |
| 01:01.31 | Jumpie | gotta hookup on dell power edge |
| 01:02.18 | C4colo | well, imagine this, there are at least 10 40mm fans in my server, all are high-powered high-volume fans each putting out about 50 dba |
| 01:02.34 | jjshoe | I have a 1u server with two mobo's in it that is super duper quiet. |
| 01:03.06 | Jumpie | heh |
| 01:04.36 | C4colo | are they mini-itx? |
| 01:04.39 | C4colo | or micro atx? |
| 01:04.48 | C4colo | and what are the processors? |
| 01:05.23 | C4colo | mine is a dual-opteron, it needs those 10 fans |
| 01:05.52 | jjshoe | mini-itx |
| 01:09.13 | Jumpie | C4colo, |
| 01:09.21 | Jumpie | at this time im not 100% sure on exactly which one im going to get |
| 01:09.37 | Jumpie | there was a good quad core one with 2gb and raid 1 160gb drives |
| 01:09.38 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 01:09.40 | Jumpie | for about $1200 |
| 01:09.44 | *** join/#asterisk nyarkni (i=barry@pool-72-76-182-105.nwrknj.east.verizon.net) |
| 01:09.46 | Jumpie | redundant nic |
| 01:12.16 | JT | jjshoe: diy, or prebuilt? |
| 01:12.51 | C4colo | I got a barebones from newegg |
| 01:12.54 | C4colo | tyan makes great servers |
| 01:13.37 | C4colo | mine is no longer available, spent $1300 for it 3 years ago, the one they replaced it with is only $800ish |
| 01:13.53 | C4colo | 4 SATA hot-swap on the front and dual opteron with max 16gb memory |
| 01:14.48 | C4colo | has the SiI 3114 SATA controller with 4 independant channels for the drives, skip the hardware RAID and just used mdadm you will get equal or better performance with better managability |
| 01:15.16 | *** part/#asterisk nyarkni (i=barry@pool-72-76-182-105.nwrknj.east.verizon.net) |
| 01:15.23 | C4colo | oh, and if you don't need the super-best-fasted processors, there is a site that is selling opteron 242s for $35 each |
| 01:16.07 | jjshoe | jt diy. but the 1u case was designed to hold two mini-itx's |
| 01:16.09 | ectospasm | glaz: usually that means a problem with the hardware echo canceling module... call Digium... |
| 01:16.22 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:16.22 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 01:16.25 | robeph | heh I could walk to digium =s |
| 01:16.31 | robeph | its like... not far from me at all |
| 01:16.32 | C4colo | the opteron 242 beats a P4 2.9ghz by about 50% even though it is at 1.6ghz |
| 01:16.44 | ectospasm | it'd take me a good 30min to an hour to walk there |
| 01:16.53 | ectospasm | I drive instead (-; |
| 01:16.54 | robeph | ectospasm: you in south hsv? |
| 01:16.55 | robeph | hahah |
| 01:17.27 | robeph | actually it'd take me that long less I was at my friends apt... though I think they moved so now I'm not sure wtf they're at |
| 01:17.32 | Jumpie | C4colo, whoa |
| 01:17.43 | Jumpie | i'd kinda want a single resource though |
| 01:17.44 | JT | jjshoe: mm, more info on this case? |
| 01:17.53 | jjshoe | jt mini-itx.com carries it |
| 01:17.55 | Jumpie | they still have the tyan servers on newegg/ |
| 01:18.15 | JT | ah ok |
| 01:18.20 | C4colo | yea, they disapeard for a bit but came back with a bunch of them |
| 01:18.20 | JT | jjshoe: model number? |
| 01:18.35 | C4colo | Jumpie, what do you mean a single resource? |
| 01:19.00 | C4colo | mdadm will support raid 10 or 5 or whatever using the four drives presented by the sata controller |
| 01:19.17 | jjshoe | jt no clue, look for the only 1u that supports more then one mobo |
| 01:19.52 | ectospasm | robeph: I walk slow |
| 01:19.59 | C4colo | I"m about to reinstall my server with raid 10 through mdadm, I'll post my results on the voipcoop.org forums when I have completed it |
| 01:20.00 | robeph | ah |
| 01:20.11 | *** part/#asterisk LemensTS (i=CustomGT@adsl-70-238-131-138.dsl.stlsmo.sbcglobal.net) |
| 01:20.22 | jjshoe | jt http://mini-itx.com/store/?c=33#c147 |
| 01:20.31 | JT | cool |
| 01:21.39 | *** join/#asterisk jeev (i=jeev@naptime.net) |
| 01:21.51 | C4colo | they have a dual opteron on newegg with 2 hot-swap bays for about $400, not sure of the specs just saw it the other day |
| 01:22.22 | robeph | I dislike the tersity of iax2 debug :'( |
| 01:22.31 | robeph | with sip I can make some semblence of wtf I've done wrong |
| 01:22.42 | robeph | iax2 is just like "cause code...0... holla" |
| 01:23.08 | jeev | hi guys, i'm pretty new at this.. started with asterisknow for a few days.. really happy.. but it seems like i will be extremely out of date with asterisk now if i continue it.. do you guys recommend i switch to Asterisk and just build it ? |
| 01:24.04 | ectospasm | jeev: depends on how dirty you want to get your hands... |
| 01:24.25 | Jumpie | C4colo, sorry, semo afk |
| 01:24.30 | Jumpie | C4colo, i meant like one place for everything |
| 01:24.34 | Jumpie | not this site for this, that site for that |
| 01:24.53 | Jumpie | i did see on voipsupply somethin called a phone bochs that had everything including a sangoma t1 card |
| 01:25.07 | Jumpie | but i think i could get a better dell 1ru and just add a sangoma / digium |
| 01:26.00 | jeev | ectospasm, i'm going to link 3 of my offices together.. and put the box in a local datacenter.. i'd PREFER the gui but.... i'm very picky with updates.. |
| 01:26.35 | *** join/#asterisk BeeBuu (n=beebuu@218.13.87.215) |
| 01:26.48 | ectospasm | I dunno if IAX trunking is easy to setup for AsteriskNOW... |
| 01:26.53 | Jumpie | jeev i've noticed the consensus here is dont go gui anything :D |
| 01:27.00 | ectospasm | I wouldn't think so |
| 01:27.10 | jeev | hmm, i dont know what IAX is.. i guess it's like SIP.. but i'm using SIP |
| 01:27.22 | jeev | i just dont understand why the gui is pitting my service provider in users.conf when it should be in sip.conf, no ? |
| 01:27.27 | Jumpie | like sip |
| 01:27.28 | Jumpie | no |
| 01:27.48 | ectospasm | jeev: Right... but if you're linking three asterisk systems, IAX is the natural protocol to use... |
| 01:28.07 | jeev | i think that i should keep asteriskNOW in a devel environment if i can't figure out how to do something in the text editor.. for regular asterisk, but product wise, should go asterisk. non gui |
| 01:28.23 | jeev | ectospasm, i actually am just setting them up with softphones.. and pointing them to the ip.. is that a stupid idea ? |
| 01:28.42 | jeev | it's just going to be 1 asterisk box. |
| 01:28.51 | ectospasm | SIP will have NAT issues that way. Not insurmountable, but can be difficult to fix |
| 01:29.04 | jeev | so "IAX" is preferred in this ENV ? |
| 01:29.17 | ectospasm | jeev: IMO, yes |
| 01:29.30 | jeev | i noticed a few tests.. at a certain point, i had one-way audio.. but that's fixed. |
| 01:29.38 | ectospasm | ...but I thought you would have multiple asterisk servers... |
| 01:29.44 | Jumpie | i had the same until i edited sip_nat.conf |
| 01:29.46 | jeev | so if i have all my clients running NATD.. then the smart way to go would be IAX ? |
| 01:29.59 | *** join/#asterisk Simzilla (n=Sim@bas9-quebec14-1168087145.dsl.bell.ca) |
| 01:30.14 | ectospasm | jeev: if you've gotten SIP to work, then no need to change |
| 01:30.39 | Jumpie | ok so like..im all done |
| 01:30.43 | Jumpie | how can i be sure its all 'there' lol |
| 01:30.44 | C4colo | Jumpie, ah, yes they have opterons on newegg, and actually a very good deal on DDR 400 ECC REG, kingston brand for $50/1gb so you can get the whole mess from them, just saying that if you don't need the best and biggest processor you can get them for $35 from the other site, and to be honest I have never had an AMD cpu die on me |
| 01:30.49 | jeev | i'm very excited, i'm currently paying $1600/month.. i could put this in my cage in the datacenter.. probably cost me 100 in bandwidth.. or less... and $120/month in calls. |
| 01:31.01 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 01:31.05 | Jumpie | c4 but can they handle 20-25 concurrent call servers? |
| 01:31.17 | C4colo | 242s yea |
| 01:31.20 | jeev | i've got to step out but this is cool, i will be back soon :) |
| 01:31.25 | Jumpie | jeev, :) |
| 01:31.39 | Jumpie | C4colo, so you recommend buliding them? the problem is i can get stuff a bit cheaper probably |
| 01:31.41 | Jumpie | but im strapped on time |
| 01:31.44 | C4colo | I'm going to be running a virtualization server on them with 5 virt servers running on it, one windows 2003, and four asterisk |
| 01:31.49 | Jumpie | was hoping for a decent prebuild and just install the digium/sangoma cards |
| 01:31.57 | C4colo | I'll let you know how the performance is if you want |
| 01:32.00 | Jumpie | ok |
| 01:32.11 | Jumpie | so how do you feel about supermicro |
| 01:32.21 | C4colo | but you can run two dual-core opterons if you are ok paying $200 each for the processors mnimum |
| 01:32.34 | C4colo | supermicro is a very good company but only really makes intel boards so I don't use them |
| 01:32.43 | C4colo | I have one supermicro that is about 10 years old and still running strong |
| 01:32.47 | C4colo | it is a dual p3 800 |
| 01:33.03 | C4colo | they may have started making amds recently, if so I would look at them |
| 01:33.05 | robeph | ok I got these boxes trunked via iax, I'm trying to send across a call but it's just sitting for a while, no output on remote end, and local side just hangs up after a while |
| 01:33.14 | C4colo | their barebones are good servers I know that |
| 01:33.24 | Jumpie | hmm ok..i was thinking intel was the way to go for cpus on this |
| 01:33.31 | Jumpie | if amd can handle, definately i'd do that |
| 01:33.36 | Jumpie | i see the dell power edges have 2 choiceds |
| 01:33.46 | Jumpie | i just cant get involved in buliding all these |
| 01:33.50 | C4colo | I haven't and wouldn't use intel, I may be biased but the performance:$ ratio is way out of wack |
| 01:34.44 | Jumpie | i hear you |
| 01:34.50 | Jumpie | when i built copmputers for people a few years back |
| 01:34.55 | Jumpie | it was always amd unless they specifically wanted intel |
| 01:35.13 | Jumpie | does newegg have any prebuild 1ru w/ amd? or did you hand build em all |
| 01:36.17 | robeph | oh nm, someone killed the process and it nuked my implicit -vvvvv wondered why the output was so damned terse >>< |
| 01:39.19 | robeph | how annoying... I dislike iax so much, |
| 01:39.23 | robeph | does weird stuff |
| 01:39.57 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 01:40.31 | C4colo | Jumpie, from my experience intel gets better performance by running the processors faster and faster, better for marketing anyway 3ghz sounds better than 2.1, however AMD has optimized the way the processors work so that the 2.1 ghz processor has the same performance and uses 2/3rds the power and heat |
| 01:41.04 | C4colo | my 1.6 ghz opteron 242 compresses files faster and with less overhead than a P4 3ghz |
| 01:41.12 | *** join/#asterisk JerJer[mobile] (n=jj@dhcp-11-163.hstf.interop.net) |
| 01:41.19 | *** join/#asterisk DirtyD (n=DirtyD@ool-4571b24d.dyn.optonline.net) |
| 01:41.34 | DirtyD | Wow. iaxmodem/hylafax works great! |
| 01:42.05 | DirtyD | wonder why rxfax was such a piece of poop. |
| 01:42.12 | *** join/#asterisk Katty (n=The@adsl-68-92-251-223.dsl.stlsmo.swbell.net) |
| 01:43.33 | jblack | Yeah. I only had one problem with it; I had to make sure it starts _after_ asterisk, and not before. |
| 01:44.03 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:44.13 | robeph | ok, anyone know why I'd be able to call via a sip provider from this one pbx, yet if I pass a call across an iax trunk and it passes it on to the sip provider I get a "604 this number does not exist anywhere" |
| 01:44.27 | robeph | it's trying to dial the exact same number though heh... |
| 01:44.44 | DirtyD | jblack: is that because iaxmodem needs to register with asterisk? |
| 01:44.48 | *** join/#asterisk tobias (n=tobias@user-0ce2hpk.cable.mindspring.com) |
| 01:44.49 | Greek-Boy | is there anyway to ensure that the music class starts from the beginning of the mp3 file? I am calling it through a Dial command. the idea is to use it as customized ringback tone |
| 01:45.37 | susinths | <C4colo> : u r right, but Intel seems to have taken AMD on performance nowadays |
| 01:45.52 | robeph | susinths: I <3 q6600 |
| 01:46.03 | susinths | ?? |
| 01:46.07 | Greek-Boy | intel are better at multiple cores |
| 01:46.08 | Greek-Boy | :) |
| 01:46.11 | robeph | intels quad core |
| 01:46.18 | robeph | for desktops |
| 01:46.18 | Greek-Boy | yeah |
| 01:46.19 | susinths | yeah |
| 01:46.28 | Greek-Boy | i have a quad core xeon for a server. wow! |
| 01:46.29 | robeph | I've yet to max it out doing anythign realworld |
| 01:46.34 | Greek-Boy | that thing is crazy |
| 01:46.37 | robeph | of course... |
| 01:46.47 | robeph | I got this q6600 OC'd to like 3.29Ghz |
| 01:46.49 | susinths | the quad cores both desktop & server cpu seems to perform better than amd right? |
| 01:46.50 | robeph | from 2.4 |
| 01:46.54 | robeph | yeah |
| 01:47.02 | DirtyD | if spandsp can emulate a fax machine, can it also be used to emulate a dialup modem? |
| 01:47.03 | robeph | susinths: plus its very very stable > AMD for OC'ing |
| 01:47.30 | Greek-Boy | i wouldn't wanna overclock a multiple core intel CPU |
| 01:47.36 | robeph | Greek-Boy: and why not =) |
| 01:47.38 | Greek-Boy | AMD is the way to go if u into overclocking |
| 01:47.39 | susinths | i c |
| 01:47.49 | robeph | Greek-Boy: you've not played with the q6600 yet then I presume |
| 01:47.58 | Greek-Boy | i haven't |
| 01:48.01 | susinths | intel r making in 45nm |
| 01:48.01 | robeph | I've seen it water cooled to like 3.85ghz |
| 01:48.08 | robeph | from 2.4 stock |
| 01:48.10 | susinths | thats better for oc i guess |
| 01:48.10 | asdx | ~book |
| 01:48.11 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 01:48.12 | Greek-Boy | will attempt too maybe in 2 years :P |
| 01:48.14 | robeph | I got it air cooled at 3.25ghz |
| 01:48.37 | Jumpie | ok boy i dunno where to start |
| 01:48.43 | Jumpie | i edtied my local net and externip stuff |
| 01:48.45 | DirtyD | You're water cooling your asterisk box? |
| 01:48.47 | susinths | u got a intel 2.4 quad cpu to over 3Ghz?? |
| 01:48.50 | Jumpie | now trying to setup my local extension and sip registryation |
| 01:48.51 | robeph | DirtyD: hahah no :p |
| 01:48.55 | robeph | susinths: eh yeh |
| 01:49.00 | susinths | shitt |
| 01:49.02 | robeph | susinths: well over |
| 01:49.05 | susinths | which one |
| 01:49.10 | robeph | susinths: q6600 g0 |
| 01:49.15 | susinths | i c |
| 01:49.16 | DirtyD | robeph: oh lol. I was gonna say, hope it's not going in a data center.. :) |
| 01:49.20 | robeph | DirtyD: :p |
| 01:49.24 | susinths | how much costs the cpu? |
| 01:49.32 | robeph | susinths: hrmm 260? |
| 01:49.43 | susinths | $ |
| 01:50.01 | robeph | http://www.newegg.com/Product/Product.aspx?Item=N82E16819115018 |
| 01:50.03 | susinths | ohhh, man |
| 01:50.05 | robeph | USD 235 |
| 01:50.14 | robeph | check specs on it :p |
| 01:50.14 | susinths | it like over 3Ghz each core?? |
| 01:50.18 | robeph | susinths: yes |
| 01:50.19 | susinths | damn!!! |
| 01:50.22 | robeph | susinths: 3.25ghz |
| 01:50.32 | robeph | at full 100%x4 I hit sabout 53F |
| 01:50.34 | robeph | air cooled |
| 01:50.34 | susinths | ohhh shit |
| 01:50.49 | robeph | but I never touch 100%x4 @ real world application |
| 01:50.56 | susinths | in Celsius? |
| 01:51.07 | robeph | 11.5 |
| 01:51.12 | robeph | er |
| 01:51.13 | robeph | no |
| 01:51.14 | robeph | shit |
| 01:51.16 | susinths | really? |
| 01:51.17 | robeph | lol it is C |
| 01:51.19 | robeph | I'm retarded |
| 01:51.20 | robeph | no |
| 01:51.22 | Jumpie | wow amazing how there is much more info on the config files than there was on the trixbox versions :D |
| 01:51.28 | susinths | cant be tight |
| 01:51.33 | robeph | I view my temps in C, I just was lookign at the weather and talking bout temps on my cpu |
| 01:51.36 | susinths | its lowe than idle temp |
| 01:51.42 | robeph | my weather = F... |
| 01:51.42 | susinths | hehe |
| 01:51.45 | robeph | it's 53C |
| 01:51.49 | susinths | oh i v |
| 01:51.52 | susinths | c* |
| 01:52.10 | susinths | so it will handle MANY * users right? |
| 01:52.12 | robeph | I new that looked wrong.. |
| 01:52.21 | robeph | I dunno it's not runnign a* :p |
| 01:52.21 | susinths | hehe |
| 01:52.33 | susinths | ok |
| 01:52.33 | robeph | I'd not try to use anything oc'd in any production environ, |
| 01:52.37 | robeph | it's asking for asplosion |
| 01:52.50 | susinths | i'm sure it can handle at least 50users |
| 01:53.22 | susinths | which motherboard? |
| 01:54.08 | susinths | <robeph> : is the cpu 45nm based? |
| 01:54.17 | robeph | http://www.newegg.com/Product/Product.aspx?Item=N82E16813131235 |
| 01:54.19 | robeph | this mobo I got |
| 01:54.28 | robeph | no 65nm |
| 01:54.36 | susinths | really |
| 01:54.45 | susinths | and it oc's so good |
| 01:54.53 | robeph | yep |
| 01:55.07 | robeph | of course I have a case with like 6 fans |
| 01:55.10 | susinths | i like asus mb's |
| 01:55.10 | *** join/#asterisk Telamon (n=telamon@blk-89-240-175.eastlink.ca) |
| 01:55.12 | robeph | and a huge huge heat sink |
| 01:55.16 | susinths | very good quality |
| 01:55.19 | robeph | indeed |
| 01:55.24 | susinths | Zalman? |
| 01:55.37 | robeph | hahah no |
| 01:55.43 | robeph | sounds like a small airport |
| 01:55.53 | susinths | which heatsink |
| 01:55.55 | susinths | hehe |
| 01:55.56 | Telamon | Anyone know why a Polycom 550 wouldn't be dialing extensions that start with an *? IE, if I dial *99 I just get a busy tone without it trying to send it to Asterisk. |
| 01:56.08 | susinths | its a heatsink called Zalman 9500 |
| 01:56.11 | susinths | its very good |
| 01:56.17 | robeph | its in an antec nine case chassis, with a zerotherm http://www.newegg.com/Product/Product.aspx?Item=N82E16835887011&Tpk=nirvana%2bheatsink |
| 01:56.20 | Jumpie | hah im lost |
| 01:56.21 | *** join/#asterisk ectospasm (n=ectospas@c-71-207-229-248.hsd1.al.comcast.net) |
| 01:56.35 | robeph | ok, anyone know why I'd be able to call via a sip provider from this one pbx, yet if I pass a call across an iax trunk and it passes it on to the sip provider I get a "604 this number does not exist anywhere" |
| 01:56.36 | susinths | i c |
| 01:58.49 | robeph | hmmm |
| 02:00.10 | [hC] | anyone upgrade to recent polycom firmware (3.0, 2.2, etc) and have an issue where after a short period of time, the ringer stops working on a phone? |
| 02:01.14 | tengulre | how to connect asterisk to skype? |
| 02:01.50 | ectospasm | tengulre: I'm sure there's a channel driver for it, but officially it doesn't support Skype... |
| 02:02.31 | robeph | is there some caveat about sending outward calls via iax trunk -> a* pbx -> sip |
| 02:02.50 | robeph | cos something is mighty odd here =s |
| 02:05.23 | *** join/#asterisk cyberjackal (n=jackal@pool-70-109-57-67.clppva.fios.verizon.net) |
| 02:17.35 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 02:20.21 | robeph | so does anyone have any idea? |
| 02:25.56 | Corydon76-dig | robeph: Nope |
| 02:25.58 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.133.55) |
| 02:26.05 | robeph | weird |
| 02:26.06 | Corydon76-dig | robeph: you're probably doing something wrong |
| 02:26.12 | robeph | well yeah, I'm guessing that |
| 02:26.23 | robeph | I was looking more for a "what" could I be doing wrong cos nothing is apparent to me |
| 02:26.31 | Corydon76-dig | http://pastebin.ca |
| 02:26.45 | Corydon76-dig | Pastebin your configs |
| 02:26.50 | robeph | ah ok |
| 02:27.09 | robeph | I did earlier but that was for dundi... not this hold on |
| 02:27.11 | *** join/#asterisk ldsjohn (n=darksage@c-24-21-115-243.hsd1.mn.comcast.net) |
| 02:27.24 | *** join/#asterisk ahbritto (n=guest@adsl-69-104-245-232.dsl.pltn13.pacbell.net) |
| 02:27.24 | ldsjohn | anyone ever see this error ? Progress Description: Call is not end-to-end ISDN; further call progress information may be available inband |
| 02:27.46 | ldsjohn | its a pri debug error that happens when I call a certain few 800 numbers |
| 02:28.00 | *** join/#asterisk asteriskj (n=infin1te@72.243.59.33) |
| 02:28.43 | *** join/#asterisk JayTee52 (n=jforde05@c-69-243-161-112.hsd1.in.comcast.net) |
| 02:29.34 | ldsjohn | my phone company tells me I need to make my PRI connect the audio imeadiatly but I can't find any documentation on this |
| 02:30.45 | C4colo | does the call fail? |
| 02:32.19 | C4colo | it sounds like it is just an informative message, if the call progresses normally then you should be fine, if it fails after that message you won't know why because the call progress is in-band |
| 02:32.30 | ldsjohn | yes it fails |
| 02:32.36 | ldsjohn | it rings for 11 rings or so |
| 02:32.51 | ldsjohn | then sometimes I see a CONNECT ACKNOWLEDGE and bam it works |
| 02:33.14 | C4colo | do the numbers you are dialing work from a normal telephone or cellphone line? |
| 02:34.03 | C4colo | the ones that ring and then fail? |
| 02:34.34 | *** join/#asterisk ta^3 (n=tacvbo@189.146.179.149) |
| 02:36.21 | robeph | Corydon76-dig: http://pastebin.ca/979137 |
| 02:38.05 | ldsjohn | yup |
| 02:38.09 | ldsjohn | im calling lenovo |
| 02:38.15 | ldsjohn | any number for lenovo wont work |
| 02:38.19 | ldsjohn | but on my cell it works |
| 02:38.23 | ldsjohn | on my house phone it works |
| 02:39.07 | Corydon76-dig | robeph: check the outgoing callerid when it succeeds and compare to when it fails |
| 02:39.08 | JT | "doesn't work"? |
| 02:39.22 | ldsjohn | it rings 10-15 times |
| 02:39.30 | Jumpie | m |
| 02:39.31 | ldsjohn | then either disconnects me or connects me to them |
| 02:39.37 | Jumpie | uh i tried to load up asterisk and it says command not found |
| 02:39.38 | Jumpie | lol |
| 02:39.41 | Jumpie | wasnt that put in my path? |
| 02:39.43 | Corydon76-dig | ldsjohn: that's a question for your provider |
| 02:39.51 | ldsjohn | they tell me its my fault |
| 02:40.05 | ldsjohn | they brought in a cisco system and plugged it into the pri |
| 02:40.05 | Corydon76-dig | Can they tell you why? |
| 02:40.06 | ldsjohn | it worked |
| 02:40.15 | robeph | Corydon76-dig: can I just noop that? |
| 02:40.24 | ldsjohn | they tell me asterisk needs to on some calls through the pri do progress montiering |
| 02:40.32 | JayTee52 | Jumpie, what linux distro did you install * on? |
| 02:40.35 | ldsjohn | connect the audio instantly and then do call progres monitering |
| 02:40.40 | Corydon76-dig | ldsjohn: ask for a PRI trace and have them tell you what you need to set that isn't getting set |
| 02:40.42 | ldsjohn | I dont see that as being possible though |
| 02:40.46 | lmadsen | hilo |
| 02:40.50 | ldsjohn | they did a trace / trap and sent it to me |
| 02:41.02 | Corydon76-dig | ldsjohn: and? |
| 02:41.09 | ldsjohn | and thats what I see the only difference between working and not working is |
| 02:41.35 | ldsjohn | <PROTECTED> |
| 02:41.48 | ldsjohn | I get that message only on the ones that dont work |
| 02:41.59 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-28.dllstx.fios.verizon.net) |
| 02:42.06 | jblack | Oh YUCK! I'm looking for a new provider, so I did a search for "Termination". after the 3 top links, google suggests that what I really meant was abortion. http://www.google.com/search?hl=en&safe=off&q=termination&btnG=Search |
| 02:42.22 | ldsjohn | lol |
| 02:42.23 | Corydon76-dig | ldsjohn: ask them to drill down a bit further. That's just an error message from their switch, not a diagnosis of what you need to be sending |
| 02:42.33 | ldsjohn | no thats the error from asterisk |
| 02:42.37 | ldsjohn | if i do pri debug span 1 |
| 02:42.39 | ldsjohn | i get that message |
| 02:42.47 | ldsjohn | they gave me a dump of their messages |
| 02:42.52 | ldsjohn | do you wan tto see it |
| 02:43.01 | Corydon76-dig | Sure, why not |
| 02:43.03 | ldsjohn | they say it proves its my system |
| 02:43.10 | ldsjohn | because asterisk is sending HEX data |
| 02:43.16 | Corydon76-dig | ROFL |
| 02:43.25 | Corydon76-dig | It's all hex data |
| 02:43.30 | ldsjohn | yup |
| 02:43.48 | Corydon76-dig | That's a "we don't know what's wrong, either" |
| 02:43.58 | Corydon76-dig | Ask for a senior switch tech |
| 02:44.54 | ldsjohn | http://pastebin.ca/979147 |
| 02:45.04 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 02:45.43 | Jumpie | JayTee52, sorry, missed your response |
| 02:45.45 | Jumpie | centos 5 |
| 02:46.09 | Jumpie | i opened terminal and su |
| 02:46.13 | JayTee52 | did you use the voip-info.org guide I sent you the link to? |
| 02:46.14 | Jumpie | then did asterisk -vvvr |
| 02:46.24 | Jumpie | JayTee52, yea thast what i've been using |
| 02:46.41 | Jumpie | and actually has been pretty much spot on |
| 02:46.55 | JayTee52 | did you do the checkconfig part and the service asterisk start? |
| 02:47.25 | Jumpie | yes |
| 02:47.29 | Jumpie | that was one part that was different |
| 02:47.31 | JayTee52 | open a terminal and cd to /etc/asterisk and then type asterisk -vvvr |
| 02:47.35 | Jumpie | i had to do chkconfig --add asterisk |
| 02:47.41 | JayTee52 | ? |
| 02:47.41 | Jumpie | ok |
| 02:47.56 | Jumpie | argh still says cmd not found |
| 02:48.04 | Corydon76-dig | ldsjohn: that trace says the call went through |
| 02:48.05 | Jumpie | JayTee52, i didnt need to restart my system did i? |
| 02:48.10 | ldsjohn | yup |
| 02:48.14 | Corydon76-dig | ldsjohn: 2 calls, in fact |
| 02:48.20 | JayTee52 | Jumpie, yes |
| 02:48.21 | ldsjohn | but there is a 10-12 second delay on one of them |
| 02:48.31 | Corydon76-dig | That's their switch |
| 02:48.42 | Jumpie | JayTee52, well crap |
| 02:48.43 | Jumpie | lol |
| 02:48.44 | ldsjohn | yup |
| 02:48.44 | JayTee52 | or you can type service asterisk start at a termnial prompt |
| 02:48.46 | Corydon76-dig | A routing delay of some sort |
| 02:48.49 | ldsjohn | they are finally believing me |
| 02:48.54 | Jumpie | JayTee52, i ahd done that already |
| 02:49.32 | ldsjohn | those 2 calls are the same |
| 02:49.35 | ldsjohn | but one is a good number |
| 02:49.36 | JayTee52 | Jumpie, from that point on after a boot you can type ps aux | grep asterisk and you should see the startup script. You installed 1.4.19? |
| 02:49.38 | ldsjohn | and one was a bad number |
| 02:49.53 | Jumpie | JayTee52, correct |
| 02:50.05 | Jumpie | ok so first, lemme go reboot |
| 02:50.08 | Jumpie | brb :P |
| 02:50.16 | ldsjohn | the only difference was the bad number delays 12 seconds or so before they respond, their tech is now pulling another dump |
| 02:50.18 | Corydon76-dig | ldsjohn: I'm not sure why you're asking for help here. It's clearly a carrier issue |
| 02:50.31 | ldsjohn | I needed some backup, sorry |
| 02:50.42 | ldsjohn | they have argued all day for the past 2 days that its my fault |
| 02:51.02 | Corydon76-dig | ldsjohn: That's normal for the telco |
| 02:51.24 | Corydon76-dig | "We're the phone company. We don't care. We don't have to." |
| 02:51.48 | ldsjohn | I dont know how to argue against them im the only guy at my company that works on linux and everyones pointing a finger at me... |
| 02:51.54 | ldsjohn | their cisco system worked |
| 02:51.55 | Corydon76-dig | Only when you hammer the point that it's their fault, will they acknowledge |
| 02:52.06 | ldsjohn | but i think its because the cisco system has some type of workaround for the problem |
| 02:52.12 | Corydon76-dig | ldsjohn: always escalate |
| 02:52.14 | ldsjohn | not because the problem is with asterisk |
| 02:52.55 | JT | ldsjohn: are you using the correct switchtype? |
| 02:53.01 | ldsjohn | yup |
| 02:53.19 | ldsjohn | esf,b8zs |
| 02:53.27 | ldsjohn | thats what they told me it was |
| 02:53.28 | JT | that's not switchtype |
| 02:53.32 | JT | that's line encoding |
| 02:53.32 | ldsjohn | national |
| 02:53.33 | outtolunc | that isn't the switchtype |
| 02:53.36 | ldsjohn | is switchtype |
| 02:53.51 | Corydon76-dig | national-1 or national-2 ? |
| 02:53.51 | JT | are you sure they're using NI1? |
| 02:53.52 | outtolunc | 1 or 2 |
| 02:53.55 | ldsjohn | hrm |
| 02:54.01 | ldsjohn | switchtype=national |
| 02:54.08 | JT | that is NI1 |
| 02:54.08 | ldsjohn | would switchtype affect 10 numbers |
| 02:54.08 | Corydon76-dig | That's national-2 |
| 02:54.12 | JT | what are *they* using? |
| 02:54.15 | ldsjohn | hrm |
| 02:54.15 | JT | is it? |
| 02:54.18 | Corydon76-dig | JT: yes |
| 02:54.26 | JT | what's ni1? |
| 02:54.33 | Corydon76-dig | switchtype=ni1 |
| 02:54.43 | JT | ah |
| 02:55.03 | JT | too many switch types in the us :D |
| 02:55.21 | ldsjohn | and wrong switchtype |
| 02:55.26 | ldsjohn | would it affect all calls? |
| 02:55.28 | jblack | Yeah. |
| 02:55.34 | ldsjohn | cause we have thousands of numbers that we can call |
| 02:55.36 | ldsjohn | 10 that we can't |
| 02:55.43 | x86 | any arabic speakers about? |
| 02:55.53 | Corydon76-dig | ni1 and ni2 are the two main ones. Most of the others are simply variants of ni2 |
| 02:56.18 | Corydon76-dig | Extra sets of IEs, basically |
| 02:56.38 | Corydon76-dig | The only exceptions to that are euroisdn and qsig |
| 02:56.39 | jblack | I got hit with the switchtype question for the first time yesterday. |
| 02:56.40 | ldsjohn | so wrong switchtype could plausibly work with most numbers? just be missing one set? |
| 02:57.21 | Corydon76-dig | ldsjohn: it's more likely they have a route screwed up in their switch |
| 02:57.33 | ldsjohn | yeah that was the solution a year ago |
| 02:57.35 | *** join/#asterisk Jumpie (n=jumpie@pool-96-231-155-171.washdc.fios.verizon.net) |
| 02:57.40 | Jumpie | ok JayTee52 ... |
| 02:57.41 | ldsjohn | we had 3 numbers not working they fixed their route and wala |
| 02:57.52 | Corydon76-dig | "voila" |
| 02:57.56 | drmessano | "viola" |
| 02:58.00 | drmessano | err yeah |
| 02:58.03 | ldsjohn | but they are argueing against it because they plugged a cisco system into my pri |
| 02:58.05 | ldsjohn | and it worked fine |
| 02:58.06 | Jumpie | works now hehe |
| 02:58.28 | drmessano | "voila", not the instrument |
| 02:58.28 | Corydon76-dig | ldsjohn: did they try ALL of the numbers? |
| 02:58.38 | ldsjohn | all the ones I complained about |
| 02:58.42 | Corydon76-dig | ldsjohn: or did they just call the local telco and said "works for us" |
| 02:58.49 | JayTee52 | it's weird that you did checkconfig -add asterisk because I only did checkconfig asterisk on on all my installs. |
| 02:58.55 | ldsjohn | I heard them |
| 02:58.59 | Jumpie | JayTee52, ya, it didnt work |
| 02:58.59 | ldsjohn | they called and talked to lenovo |
| 02:59.04 | ldsjohn | through their cisco box |
| 02:59.06 | Jumpie | guys in #centos said to try that and it worked |
| 02:59.08 | Corydon76-dig | JayTee52: "chkconfig" |
| 02:59.27 | ldsjohn | if I do the same thing through my asterisk box I get 12 second delay by the time i get connected the lenovo system has hung up most of the time |
| 02:59.27 | JayTee52 | Corydon76-dig, yeah. sorry, it's a bit late and I'm getting crispy |
| 02:59.28 | Jumpie | ok im at asterisk cli |
| 02:59.37 | JayTee52 | yay!!! type help |
| 02:59.39 | Jumpie | and my sip registrion is succesful woohoo |
| 03:00.11 | JayTee52 | Jumpie, you're off and running then! don't stay up past your bedtime! :-) |
| 03:00.11 | Jumpie | JayTee52, im curious though, the sip.conf is humongeos with tons of comments, how do i know where i can setupa new context? i am concerned where [general] really ends? hehe |
| 03:00.16 | Jumpie | lol :) |
| 03:00.17 | *** part/#asterisk cyberjackal (n=jackal@pool-70-109-57-67.clppva.fios.verizon.net) |
| 03:00.51 | robeph | Corydon76-dig: I should be able to simply NoOp("${CALLERID(all)}") right? |
| 03:00.52 | Jumpie | my goal tonight is just to get internal extensions established and calling |
| 03:01.09 | JayTee52 | read it carefully, you'll see where general ends and the examples for phones start. You can make your own without all the clutter by copying and pasting just what you need but save the original for reference. |
| 03:01.23 | robeph | Corydon76-dig: for that extension I'm dialing to display it? cos when I do this though, I see no diff except one says IAX2/pbx1-iax other is saying OSS/dsp for console |
| 03:01.42 | Jumpie | ahh like sip_bak.conf or something |
| 03:01.58 | Corydon76-dig | robeph: You could also do a sip debug |
| 03:02.12 | robeph | ahh didn't thnk of that for the remote box |
| 03:02.19 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 03:02.46 | Jumpie | JayTee52, as far as i can see, [general] exstends all the way until [authentication] |
| 03:03.39 | Jumpie | ah..i think i see it what your talking about now |
| 03:04.13 | JayTee52 | just start at adding stuff at the end of the file. Most everything that could cause issues is commented out by default. But there is one thing you really need to do first and foremost. |
| 03:04.35 | Jumpie | heh |
| 03:04.42 | Jumpie | wazzat |
| 03:04.45 | JayTee52 | read the book! |
| 03:04.58 | JayTee52 | ~book |
| 03:04.58 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 03:05.07 | Jumpie | yea ..i have it up |
| 03:05.12 | Jumpie | plus i also have asterisk for dummies |
| 03:05.12 | Jumpie | :D |
| 03:05.19 | JayTee52 | I'd toss the latter |
| 03:05.26 | Jumpie | k |
| 03:05.29 | drmessano | Up.. or READING them.. |
| 03:05.30 | drmessano | ? |
| 03:05.43 | JayTee52 | :-) |
| 03:06.14 | drmessano | You just totally asked "I just installed Asterisk, now what?".. in a round about way. |
| 03:06.24 | drmessano | That question is LOVED here |
| 03:07.11 | Jumpie | drmessano, actually i was trying to follow a bit and start |
| 03:07.15 | drmessano | ~nowwhat |
| 03:07.15 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 03:07.16 | Jumpie | i just wanted to be sure i was in right part ot file |
| 03:07.21 | Jumpie | rofl |
| 03:07.26 | JayTee52 | <PROTECTED> |
| 03:07.43 | drmessano | JayTee52: you're losing more than that |
| 03:07.59 | JayTee52 | just call me turnip brain |
| 03:08.16 | drmessano | JayTee52: I'd recommend "Orange Crush" up loud when this is over |
| 03:08.43 | JayTee52 | do you know what that song's about? |
| 03:09.02 | [TK]D-Fender | JayTee52, I wouldn't insult a perfectly nutient-rich vegetable by comparing it to you ;) |
| 03:09.36 | [TK]D-Fender | Warning to all you vegans out there : You are what you eat. |
| 03:09.50 | JayTee52 | rumor is that it's about dioxin poisoning because Agent Orange would cause your the lumbar vertebrae and discs to deteriorate. |
| 03:10.06 | [TK]D-Fender | knaws on some steaks ands revels at being suitably on top of the food chain... |
| 03:10.07 | drmessano | Yep |
| 03:10.33 | *** join/#asterisk apollonx (i=kit@193.19.189.38.STATIC.ISP.KZ) |
| 03:10.40 | *** join/#asterisk jsoftw (n=Administ@60.234.135.124) |
| 03:10.44 | drmessano | and Blister in the Sun is not about a bad suntanning accident |
| 03:10.51 | jsoftw | How do I stop MusicOnHold from another channel? |
| 03:10.53 | plik | I considered becoming a predatarian,,,, |
| 03:11.10 | plik | one who only eats thier predators |
| 03:11.20 | C4colo | heh |
| 03:11.21 | drmessano | plik: the leniancy for eating people is not what it used to be |
| 03:11.21 | [TK]D-Fender | jsoftw, What do you mean FROM another channel? |
| 03:11.42 | jsoftw | [TK]D-Fender: doing a conference, and I want to turn off the music when another user joins. |
| 03:11.58 | C4colo | there is a meetme option for that |
| 03:12.06 | [TK]D-Fender | jsoftw, "core show application meetme" <-- go read the instructions. |
| 03:12.11 | C4colo | it is not m, that's join muted |
| 03:12.13 | jsoftw | Im not using meetme |
| 03:12.25 | C4colo | what are you using for the conference? |
| 03:12.43 | ldsjohn | okay so the switch tech and they guy on my site are still convinced |
| 03:13.14 | ldsjohn | they want to know why their system would work perfectly they now have a sundial system a cisco system and a "meter" all calling lenovo perfectly when plugged into my pri but my system doesn't |
| 03:13.28 | ldsjohn | they are using switchtype ni2 and I ahve switchtype=national |
| 03:13.44 | jsoftw | [TK]D-Fender: I forget what its called, but it does not use the ztdummy |
| 03:13.47 | jsoftw | thing |
| 03:14.11 | JT | ldsjohn: that sounds ok |
| 03:14.20 | jsoftw | But basically I need a way to stop MusicOnHold, so it carrys on |
| 03:14.25 | jsoftw | to the next instruction |
| 03:14.39 | glaz | I have unplugued a T1 from an asterisk server to an other one, and now when I try to dial my number it says this number is not in service |
| 03:14.56 | glaz | is this normal? |
| 03:15.16 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:16.15 | jsoftw | AppConference |
| 03:16.18 | jsoftw | C4colo: ^^ |
| 03:16.36 | jsoftw | Mostly done via dial plans |
| 03:17.51 | jsoftw | It appears MusicOnHold has no way of being told to stop :/ |
| 03:20.45 | jblack | as in someone wants peace and quiet while on hold? |
| 03:21.20 | *** join/#asterisk bintut (n=bintut@bb121-6-167-197.singnet.com.sg) |
| 03:21.43 | jblack | You could add a feature that calls StopMusicOnHold if the user presses # |
| 03:22.47 | C4colo | does anyone here understand why the hell digium decided that DBget(bar/${CALLERIDNUM}) should be replaced with something like Set(foo=${DB(bar/${CALLERID(num)})}) ??? |
| 03:23.37 | C4colo | my god, look at the trailing grouping symbols! I guess too many people were figuring out how to write useful dialplans and they had to keep it an elite group of people who could understand and use their software |
| 03:24.02 | jsoftw | StopMusicOnHold eh |
| 03:24.37 | jblack | it's a thought |
| 03:25.01 | [TK]D-Fender | C4colo, Because this way you don't have to have 20 vaiables for callerID, and you can simply get the value from Db for comparison without having to save it in a var first, etc |
| 03:25.26 | jsoftw | I cant seem to find docs for that |
| 03:25.33 | [TK]D-Fender | C4colo, And we've got bigger shmucks to dispose of. Ones who can't understnat PRIORITY numbers. |
| 03:25.48 | JayTee52 | moi? |
| 03:26.05 | [TK]D-Fender | jsoftw, "core show application conference" <- read its instructions then |
| 03:26.15 | [TK]D-Fender | understand* |
| 03:26.20 | [TK]D-Fender | damn I can't type tonight... |
| 03:26.34 | C4colo | I didn't even notice the typo, apparently I read fluent typo |
| 03:26.35 | jsoftw | :\ |
| 03:26.44 | jsoftw | Conference? It has little to do with it I think. |
| 03:26.52 | jsoftw | The conference its self is working fine. |
| 03:26.56 | jsoftw | The only issue is the music. |
| 03:27.08 | jsoftw | I could just not play music and it would be sweet. |
| 03:27.12 | jsoftw | But that would bore people :) |
| 03:27.17 | C4colo | I just have a dialplan that I'm rewriting for 1.4 from 1.2 and they have about a million DBput / DBget calls in the thing |
| 03:27.18 | [TK]D-Fender | jsoftw, and what do you think is INITIATING the MoH? App_conference. |
| 03:27.26 | [TK]D-Fender | jsoftw, Go read its instructions. |
| 03:27.32 | C4colo | I'm tired of writing that big long string for every call to DB |
| 03:27.44 | jsoftw | [TK]D-Fender: no, _I_ am. |
| 03:27.46 | JayTee52 | cut and paste? |
| 03:27.57 | C4colo | you should use appconference to generate the moh |
| 03:27.58 | [TK]D-Fender | jsoftw, pastebini your dialplan |
| 03:27.59 | [TK]D-Fender | ~pb |
| 03:28.00 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 03:28.01 | [TK]D-Fender | ^^^^^^^ |
| 03:28.32 | [TK]D-Fender | jsoftw, I'd like to see where you're doing this.... sounds ... "interesting". |
| 03:28.49 | *** join/#asterisk blq (n=Bl@dslb-088-064-141-136.pools.arcor-ip.net) |
| 03:29.46 | plik | hmmm "pastebini" sounds like something you might order at an italian :) |
| 03:29.46 | C4colo | and what is wrong with meetme again? just curios, not criticizing your choice |
| 03:29.58 | C4colo | lol |
| 03:30.02 | JayTee52 | mmmm, pastebini and a cappucino! |
| 03:30.06 | C4colo | I'll have a pastebini on the rocks |
| 03:30.18 | jsoftw | [TK]D-Fender: http://paste.debian.net/436 |
| 03:30.19 | C4colo | or would that be pastebini and some breadsticks |
| 03:30.26 | jsoftw | heh |
| 03:31.16 | *** join/#asterisk jmesquita (n=jmesquit@200.170.114.149) |
| 03:31.17 | JayTee52 | doesn't strike me as a cocktail beverage, more like a pastry. Maybe a member of the canoli family. |
| 03:31.27 | jsoftw | mmm food. |
| 03:31.30 | robeph | Corydon76-dig: yeh indeed seems to be the problem I tried setting the CID to what it ought be, but to no avail, it doesn't seem to care if I tell it what it should be or not... |
| 03:32.13 | [TK]D-Fender | jsoftw, Looks like your broken dialplan trying to "queue" people up before sending them to the conference room. This is not * or app_conference's fault then. |
| 03:32.25 | C4colo | hmm, a chocolate covered pastebini, that actually sounds good |
| 03:32.44 | [TK]D-Fender | jsoftw, Nifty hack attempt though. |
| 03:32.54 | [TK]D-Fender | jsoftw, Meetme not working for you? |
| 03:32.59 | jsoftw | [TK]D-Fender: erm, it works? |
| 03:33.04 | robeph | pastebini? I prefer Macadamian nut covered no pastes myself |
| 03:33.14 | jsoftw | The _only_ thing im having issues with here is the music on hold. |
| 03:33.34 | jsoftw | [TK]D-Fender: meetme needs ztdummy drivers. |
| 03:33.43 | jsoftw | [TK]D-Fender: I dont want to have to use that. |
| 03:34.19 | JayTee52 | I try to avoid pastry all together but I have my weak moments (all the time) when I end up caving and buying a box of raspberry filled glazed Krispy Kremes. |
| 03:34.29 | jsoftw | :\ |
| 03:34.35 | JayTee52 | nothing quite as satisfying as a lard induced coma. |
| 03:34.38 | jsoftw | sounds horribly sweet |
| 03:34.41 | robeph | heh |
| 03:34.57 | robeph | sadly, I order rib eyes instead of newyork strip steak...because I like the excess fat :'( |
| 03:35.14 | robeph | JayTee52: .... |
| 03:35.18 | robeph | JayTee52: I love doughnuts, but |
| 03:35.19 | C4colo | there is no music option for the Conference application? |
| 03:35.28 | C4colo | I haven't yet used it so I don't know |
| 03:35.35 | robeph | raspberry are the only ones I hate, they remind me of an offbrand generic toothpaste flavor |
| 03:35.39 | JT | a fatty steak is way healthier than krispy kreme :P |
| 03:35.52 | jsoftw | Theres nothing wrong with a bit of natural fat. |
| 03:35.58 | *** part/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 03:36.01 | jsoftw | Steak ftw. |
| 03:36.02 | C4colo | mmmm... fat |
| 03:36.10 | robeph | I just ate steak out |
| 03:36.11 | jsoftw | I mean the fat that comes out of steaks when you cook them |
| 03:36.18 | robeph | yes |
| 03:36.21 | robeph | good stuff |
| 03:36.30 | robeph | unlike pork fat which is like...horrible stuff |
| 03:36.31 | C4colo | any time someone says FTW it reminds me of my highschool days when FTW had an entirely different meaning |
| 03:36.41 | robeph | Fight the Whites? |
| 03:36.41 | C4colo | and apparently still does for me |
| 03:36.46 | C4colo | F!@#$ the world |
| 03:36.47 | jsoftw | :/ |
| 03:37.18 | C4colo | pork fat is "natural" too |
| 03:37.27 | C4colo | but I agree, not as tasty |
| 03:37.29 | robeph | tastes like craptula |
| 03:37.44 | AndyGraybeal_ | i love lard, i always have some in the friedge |
| 03:37.56 | C4colo | spread it on your toast in the morning? |
| 03:37.59 | robeph | hahaha |
| 03:38.03 | robeph | vegemite and crisco plz |
| 03:38.04 | C4colo | I use it for making gravy |
| 03:38.06 | C4colo | LOL |
| 03:38.07 | C4colo | EEEW |
| 03:38.08 | [TK]D-Fender | jsoftw, #18 puts them on MoH indefinitely. How are they expected to get out of that? |
| 03:38.16 | jsoftw | [TK]D-Fender: no kidding. |
| 03:38.34 | robeph | does anyone know why setting CID would fail and not set caller id :-\ |
| 03:38.44 | JT | crisco is pretty tasteless |
| 03:38.46 | jsoftw | [TK]D-Fender: isnt that what I was asking about? |
| 03:39.05 | robeph | I'm passing a call off over an iax trunk to anoterh box that has to pass it along to a sip provider for outgoing calls, but it clobs my CID so I can't auth with the sip provider when I go over the iax trunk >< |
| 03:39.16 | [TK]D-Fender | jsoftw, Well this is your dialplan. It won't work that way at all. You can't "background" MoH like that. |
| 03:39.27 | robeph | I can't set it locally on the outbound box cos it ignores me when I say to, it says "setting caller id blah blah" but never actually happens |
| 03:39.33 | C4colo | have you looked at http://sourceforge.net/projects/appconference/ ?? |
| 03:39.34 | jsoftw | [TK]D-Fender: *sigh* |
| 03:39.54 | [TK]D-Fender | jsoftw, The only non "C" way I can think of is using AGI to queue them up and streaming the audio. |
| 03:40.01 | jsoftw | Hmm. |
| 03:40.06 | C4colo | I don't know if that third party conference app needs ztdummy though |
| 03:40.17 | jsoftw | C4colo: thats what I am using. |
| 03:40.18 | [TK]D-Fender | C4colo, He IS using App_conference and trying to hack MoH into it via the fialplan. |
| 03:40.26 | robeph | wait |
| 03:40.29 | C4colo | then it supports music when only one |
| 03:40.32 | robeph | can't you just background madplay or something? |
| 03:40.40 | robeph | or mpg123 |
| 03:40.43 | robeph | and ringout |
| 03:40.46 | [TK]D-Fender | C4colo, It doesn't and thats why he's using it. Welcome to the conversation.... |
| 03:40.49 | C4colo | here is a list of the features http://www.freeswitch.org/node/100 |
| 03:41.10 | robeph | freeswitch >< |
| 03:41.10 | C4colo | read that page before insulting me kthk |
| 03:41.14 | C4colo | kthx rather |
| 03:41.24 | C4colo | there is meetme, conference, and confcall |
| 03:41.30 | C4colo | confcall is meetme+ |
| 03:41.38 | C4colo | but I don't know if they removed the need for the zaptel rtc |
| 03:41.48 | jsoftw | Nya ha! |
| 03:41.59 | robeph | every time I see kthx, I think of this trite lil slag who was so readily annoying.... she'd be yap yap yap yap and then when she realized you werent' listening she'd be like "okthxbuhbye" and walk off =\ |
| 03:42.02 | C4colo | the feature matrix shows the features and confcall allows moh when only one |
| 03:42.16 | C4colo | lol |
| 03:42.19 | jsoftw | I think the way to do it would be to get a channel to join the conference, play music, and kick it out when a second real person joins the conference. |
| 03:43.03 | robeph | so why not run ztdummy though *ducks".... |
| 03:43.05 | C4colo | I was just responding in like with your "welcome to the conversation" ... I was hoping the trite response would illicit a similar emotional response to that which your comment caused myself |
| 03:43.06 | jsoftw | operation DJ is underway |
| 03:43.31 | jsoftw | robeph: I dont have any VOIP hardware, and I its never accurate enough with the ztdummy. |
| 03:43.46 | C4colo | are you using the Conference() application (as it seems) or the third party confcall application? |
| 03:43.50 | robeph | jsoftw: I set up some moh type junk via an agi that would provide random low bitrate shoutcast channels via mpg123.... |
| 03:44.02 | C4colo | I'd look at the requirements of the confcall, looks like I might use it to replace meetme on my server |
| 03:44.03 | jsoftw | Trues. |
| 03:44.14 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 03:44.24 | C4colo | oh, I pasted the wrong link earlier |
| 03:44.25 | robeph | sounded like pure buttered arse. |
| 03:44.27 | C4colo | hince the confusion |
| 03:44.34 | robeph | C4colo: indeed |
| 03:44.36 | robeph | kthx |
| 03:44.49 | C4colo | here is the confcall thingie download place stuff http://www.freeswitch.org/node/50 |
| 03:44.57 | JayTee52 | <PROTECTED> |
| 03:45.17 | C4colo | oooh, they have an app_changrab on that page too |
| 03:45.27 | C4colo | I could use that |
| 03:45.31 | jsoftw | Anyone running asterisk in a fbsd jail? |
| 03:45.32 | robeph | is listening to his girlfriend play everquest 2 in the background while I fuddle with this stupid box somewhere in chicago... |
| 03:45.46 | jsoftw | You have a geek gf? Lucky bastard |
| 03:45.53 | robeph | hahah no just a gamer gf |
| 03:45.56 | jsoftw | Everquest though :/ |
| 03:46.09 | robeph | jsoftw: heheh well that box she's on is the "tv" box low end gaming hardware |
| 03:46.14 | robeph | so COD4 won't run on it =p |
| 03:46.17 | JayTee52 | I bought and played that like twice |
| 03:46.17 | jsoftw | :) |
| 03:46.25 | robeph | she'll have to go play on her box for that, and I gave her that one and its at her house |
| 03:46.33 | C4colo | yea, playing eq does not mean geek |
| 03:46.42 | jsoftw | yeah well meh |
| 03:46.42 | robeph | C4colo: not at all |
| 03:46.53 | robeph | I was like yeh, come ver here lemme show you me failing to setup an iaxtrunk |
| 03:47.02 | robeph | and she's like, meh, no , order some steak out |
| 03:47.06 | C4colo | I have a lot of friends who play eq and/or WoW 24/7 and they are not geeks, just addicted to MMORPGs |
| 03:47.24 | C4colo | lol, now I want steak |
| 03:47.27 | jsoftw | Ok so lets talk about how gamers != geeks some more then |
| 03:47.29 | robeph | she'll play any game really... FPS/MMO/RTS |
| 03:47.38 | jsoftw | My favourarte game is dark crusade |
| 03:47.42 | robeph | hey I'm a geek and a gamer :-s |
| 03:47.44 | C4colo | it's almost 10pm on a week night though, I don't know if there is anywhere still open with good stek |
| 03:47.45 | jsoftw | likewise |
| 03:47.46 | robeph | jlsoft |
| 03:48.07 | C4colo | I'm a geek that never has time for games =( |
| 03:48.09 | robeph | er, I still play ADoM like... at least a few hours a week |
| 03:48.10 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 03:48.16 | jsoftw | ADoM ? |
| 03:48.20 | robeph | actually its open in a console here |
| 03:48.25 | robeph | yeah, a rogue like game |
| 03:48.29 | jsoftw | rouge? |
| 03:48.32 | C4colo | my newest game is called 1.2 to 1.4 dialplan conversion |
| 03:48.32 | robeph | :o |
| 03:48.39 | jsoftw | C4colo: awesome |
| 03:48.46 | C4colo | the best part is that for every level I complete I am rewarded with real money instead of game gold |
| 03:48.48 | jsoftw | C4colo: single-player only, no co-op |
| 03:49.04 | C4colo | well you can use a hack called 'screen -x' but that can get confusing |
| 03:49.13 | [TK]D-Fender | My favourite game is banned in 17 states ;) |
| 03:49.23 | jsoftw | What is it? |
| 03:49.31 | [TK]D-Fender | jsoftw, Not that kind of channel ;) |
| 03:49.36 | robeph | [TK]D-Fender: sleeping with minors? |
| 03:49.38 | jsoftw | :/ |
| 03:49.38 | robeph | :o |
| 03:49.41 | C4colo | LOL |
| 03:49.46 | jsoftw | Dodgy. |
| 03:49.49 | jsoftw | Whatever it is. |
| 03:50.00 | robeph | but only 17? |
| 03:50.01 | robeph | O.o |
| 03:50.06 | C4colo | haxxoring using CID spoofing ? |
| 03:50.08 | robeph | *shrug* |
| 03:50.11 | robeph | heh |
| 03:50.32 | [TK]D-Fender | C4colo> haxxoring using CID spoofing ? <- no that IS this kind of channel |
| 03:50.37 | C4colo | LOL |
| 03:51.19 | robeph | I wrote a script a long long time ago that would scour online rev directories, for a given npa-npx and then dial all the unlisted numbers that weren't found on a dialback and spoof the CID to itself so it'd log the CNAM and infos for the unlisted numbers |
| 03:51.29 | robeph | broadvoice called and griped loudly about me doing that |
| 03:51.34 | JT | [TK]D-Fender: evasive enough? :P |
| 03:52.13 | drmessano | Say that again... in my good ear? |
| 03:52.22 | robeph | anyhow about CID, why does asterisk ignore my command to set the CID name and number in my dialplan? |
| 03:52.28 | C4colo | wow, that's a neat trick |
| 03:52.31 | robeph | console says it's doing it, but then when it goes out. |
| 03:52.33 | C4colo | can I have a copy of the results? |
| 03:52.39 | robeph | C4colo: it's ancient |
| 03:52.43 | C4colo | hmm |
| 03:52.46 | robeph | check on bellsmind.net they had the script on there |
| 03:53.04 | robeph | it was just a perl script and partner agi |
| 03:53.05 | drmessano | Is this 1337 night? |
| 03:53.26 | robeph | C4colo: I'm sure some of the results are up on some of the forum archives |
| 03:53.37 | robeph | check back bout 3-4 yrs on binrev forums |
| 03:53.48 | robeph | if they keep attachments n stuff you can find em |
| 03:54.03 | C4colo | I'll have to do that sometime, I should get back to my dialplan though =( |
| 03:54.11 | *** join/#asterisk Micc (n=Micc@207.66.239.101) |
| 03:54.24 | C4colo | but does app_confcall need ztdummy? that's what I want to know and there is no documentation just a .c file |
| 03:54.25 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 03:54.53 | robeph | C4colo: http://www.binrev.com/forums/lofiversion/index.php?t7870.html Thats some early stuff, don't think any of the spoof back info is in that, that looks like when I first started on it |
| 03:55.52 | [TK]D-Fender | C4colo, that page linked said it needed a timer like meetme does. Doesn't say WHICH, but first guess would be a Zaptel timer. |
| 03:56.30 | drmessano | I haxored my asterisk box so I don't have to put any coins into my grandstream phone to make calls now |
| 03:56.34 | drmessano | I am so 1337 |
| 03:56.46 | JayTee52 | hahahaha |
| 03:56.48 | robeph | drmessano: heh |
| 03:57.04 | robeph | drmessano: lol |
| 03:57.24 | robeph | man this is pissing me off :'''''( |
| 03:57.45 | drmessano | I've seen WarGames 2163 times, and I have an ASCII txt file of all the errors in the movie. Is that bad? |
| 03:57.50 | JayTee52 | "put the mouse down and step away from the console!" |
| 03:58.35 | drmessano | SSH does not mean "Shuddup Silly Haxor" |
| 03:58.45 | *** part/#asterisk killfill (n=killfill@200.73.13.54) |
| 03:58.51 | JayTee52 | I know people who've watched every episode of every season of Stargate SG-1 just to count how many times Teal'c has said, "Indeed". |
| 03:58.51 | jeev | asterisk FOR LIFE |
| 03:59.18 | drmessano | HA |
| 03:59.20 | JayTee52 | but I avoid anyone that speaks fluent Klingon |
| 03:59.25 | plik | app_redbox |
| 03:59.27 | *** part/#asterisk Micc (n=Micc@207.66.239.101) |
| 03:59.33 | drmessano | Star Trek is for losers.. STAR WARS FOR THE WIN!!!! |
| 03:59.40 | jeev | hahah |
| 03:59.45 | jeev | anything that starts with star is for losers! |
| 03:59.52 | drmessano | BOOO HISSSS |
| 04:00.00 | JayTee52 | I liked the first 3 Star Wars but Jar Jar made me want my money and my time back. |
| 04:00.21 | drmessano | I'm a 4th level Jedi.. Don't try to go to work tomorrow, jeev.. Trust me. |
| 04:00.57 | [TK]D-Fender | These aren't the 4th quarter reports you're looking for.... |
| 04:01.04 | drmessano | app_jedi .. I can make free calls.. on your phone. |
| 04:02.13 | jsoftw | sorted! :D :D |
| 04:02.39 | jeev | i'm gonna run for asterisk party and win the presidency. |
| 04:02.53 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:03.10 | drmessano | app_haxor ... I can hang up mom's phone calls by typing "core ihatemom" on my console. I R SO 1337 |
| 04:04.05 | JT | drmessano: i read that as "ISO1337" and was thinking "I'm not sure that's a legitimate standard" |
| 04:04.15 | C4colo | why compile an app for redbox? you can just generate the tones from the dialplan |
| 04:04.18 | drmessano | LOL |
| 04:04.28 | C4colo | haha |
| 04:05.00 | drmessano | ISO1337 is the CD-ROM file system that actually works on a Compaq branded CD-ROM drive. |
| 04:05.23 | jsoftw | music on hold stuff working mint in the conference for the first person only! :D |
| 04:05.25 | C4colo | compaq != 1337 |
| 04:05.27 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25) |
| 04:05.29 | jsoftw | then it goes off once another joins |
| 04:05.39 | jsoftw | and its all happy :) |
| 04:06.15 | drmessano | when one person is left, does it play The Doors - The End? |
| 04:06.22 | drmessano | That.. would... rock |
| 04:06.30 | jsoftw | it could do |
| 04:06.31 | JayTee52 | none of my computer have a label on them. If they did it would be one I designed because I build my own. |
| 04:06.58 | C4colo | I had Pyrotechnic Systems case badges made |
| 04:07.03 | jsoftw | That was easy. |
| 04:07.15 | C4colo | of course I build custom systems as a business, so yea |
| 04:07.16 | JT | it would be better if it played a recording saying "Get the f*ck out of my house motherf*cker!" :D |
| 04:07.18 | drmessano | I've never bought a store brand PC.. I would rather sell out and use Skype |
| 04:07.18 | jsoftw | So everyone should remember the StartMusicOnHold / StopMusicOnHold is for the win. |
| 04:07.50 | C4colo | I paid a company to build the first computer I ever bought, it was a custom build though ... after that I just built my own |
| 04:07.51 | JayTee52 | Fraken Toaster Model 20 would be a good name for a home built box. |
| 04:08.15 | JayTee52 | I'm an Intel snob. |
| 04:08.40 | C4colo | ah, one of "those" |
| 04:09.01 | C4colo | that comment can be applied to the preceeding two comments at your discretios |
| 04:09.05 | C4colo | discretion rather |
| 04:09.12 | JayTee52 | although I have to admit the tri-core Phenom AMD is coming out with looks interesting. |
| 04:09.34 | C4colo | I've never paid money for an intel |
| 04:09.37 | jsoftw | Intel seems to be winning atm |
| 04:09.42 | jsoftw | Quite fast chip=s |
| 04:09.46 | C4colo | wait, I paid $40 for a dual proc P3 server once about 5 years ago |
| 04:10.14 | JT | the quad core phenom looks more interesting than the tri core |
| 04:10.15 | C4colo | although the price for performance is way too much for the new intel |
| 04:10.21 | drmessano | Did someone say Intel? |
| 04:10.27 | C4colo | I'd rather pay the same amount for one intel chip for two dual-core opterons |
| 04:10.34 | jsoftw | Sun has some mean chips though |
| 04:10.42 | drmessano | That word shows up as a big red box on my screen |
| 04:10.42 | jsoftw | Fairly whoop ass servers |
| 04:10.46 | JT | jsoftw: intel are losing against amd's hypertransport.... for now |
| 04:10.58 | jsoftw | JT: yeah but that only matters in big ass servers |
| 04:11.00 | jsoftw | ? |
| 04:11.02 | *** join/#asterisk andresmujica (n=andresmu@190.25.66.158) |
| 04:11.08 | JT | jsoftw: yeah, pity those T2000s are useless for transcoding |
| 04:11.09 | jsoftw | When there are lots of CPU/Cores |
| 04:11.09 | andresmujica | hi all! |
| 04:11.14 | jsoftw | JT: yeah :/ |
| 04:11.16 | C4colo | yep, especially in dual processors systems, the hypertransport makes handoffs for processes much faster |
| 04:11.50 | C4colo | well you tell me is an 800mhz unidirectional FSB faster than a 1000mhz bi-directional HT port? |
| 04:12.12 | C4colo | memory controller is on-die for AMD chips too, so memory access is faster |
| 04:12.12 | andresmujica | i wonder i ¡f it's possible to get some meetme room at a remote pbx location, and access those from a different pbx, both are iax2 trunked, and i can access remote extensions from both of them |
| 04:12.12 | jsoftw | erm no |
| 04:12.17 | JT | also all intel's qc offerings so far have been dual dies in a package |
| 04:12.20 | JT | not single die |
| 04:12.33 | C4colo | yea, dual dies in a package, not a dual-core die |
| 04:12.43 | JT | quad core you mean? |
| 04:12.46 | C4colo | same as two socketed cpus on one chip |
| 04:12.49 | JayTee52 | wondering what impact the new architecture Intel is bringing out with Nehalem with QuickPath instead of a FSB is going to have. |
| 04:12.53 | C4colo | AMD actually has dual-core chips |
| 04:13.30 | jsoftw | Im going to start pushing for sun gears at work :) |
| 04:13.38 | jsoftw | And probably start using solaris too |
| 04:13.46 | JT | sun makes good shit |
| 04:13.51 | JT | sun and ibm ftw |
| 04:13.51 | jsoftw | Yeap |
| 04:13.52 | JayTee52 | oooh, imagine a Sunfire running Asterisk |
| 04:13.55 | JT | hp next |
| 04:14.10 | jsoftw | Ive some new HP's in the room, they are rather nice |
| 04:14.13 | JT | JayTee52: can't do transcoding though |
| 04:14.28 | jsoftw | Cant or cant do it well? |
| 04:14.28 | JT | to g.729 anyway |
| 04:14.34 | JT | can't to g.729 |
| 04:14.38 | JayTee52 | the Ultrasparc can't? |
| 04:14.38 | JT | no code |
| 04:14.39 | jsoftw | why cant it? |
| 04:14.42 | jsoftw | Ahh right |
| 04:14.42 | JT | x86 |
| 04:14.50 | jsoftw | So just a port then |
| 04:14.52 | JayTee52 | ah, damn |
| 04:15.05 | JT | and the T2000 hasn't got the floating point you need for transcoding |
| 04:15.06 | jsoftw | Ahh right, proprietry? |
| 04:15.07 | *** join/#asterisk solar_ant (n=John@122.164.241.120) |
| 04:15.19 | JT | unless you rewrite all the transcoders to use fixed point |
| 04:15.21 | JayTee52 | some clever bastard will write an x86 emulator to handle the transcoding |
| 04:15.22 | C4colo | bbl I have to get some work done |
| 04:15.24 | jsoftw | Some of the AMD boxen from sun would whoop ass it though |
| 04:15.31 | JT | right |
| 04:15.36 | jsoftw | They have some seriously scary hardware :\ |
| 04:16.52 | jsoftw | has the task of making asterisk redundant :/ |
| 04:17.07 | JayTee52 | ah, crap in a can. it's after midnite. I should try this thing called sleep out, I hear it's pretty good. |
| 04:17.08 | jsoftw | so other asterisk machines take over |
| 04:17.15 | JT | jsoftw: fun! |
| 04:17.17 | JayTee52 | Skynet! |
| 04:17.32 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:17.34 | jsoftw | I might use carp and some other trickery |
| 04:17.41 | jsoftw | carp on a jail ip |
| 04:17.45 | jsoftw | That should do it |
| 04:18.17 | JT | carp works ok on linux? |
| 04:18.19 | JayTee52 | jsoftw, what about Linux HA |
| 04:18.24 | jsoftw | I dont use linux |
| 04:18.28 | JayTee52 | ah |
| 04:18.29 | JT | ah ok |
| 04:18.29 | jsoftw | Not for servers anyway |
| 04:18.37 | JayTee52 | what do you use? |
| 04:18.40 | JayTee52 | BSD? |
| 04:18.43 | jsoftw | free/open bsd |
| 04:18.58 | JT | wonders what niche netbsd has, if any |
| 04:18.58 | jsoftw | yeah. |
| 04:19.02 | JayTee52 | how big of a pain is it getting asterisk running on that? |
| 04:19.15 | jsoftw | JayTee52: asterisk its self was easy. |
| 04:19.47 | plik | JayTee52: not bad : cd /usr/ports/net/asteris && make install :) |
| 04:19.48 | jsoftw | JayTee52: but I put in freepbx and AGI stuff, and that was a pain in the ass, because all that shit assumed different path locations. |
| 04:19.54 | JayTee52 | what hardware you running it on? |
| 04:19.58 | jsoftw | opeteron |
| 04:20.04 | jsoftw | opteron even |
| 04:20.12 | JayTee52 | good chip |
| 04:20.16 | jsoftw | or something like that, I forget |
| 04:20.32 | jsoftw | either way, its 100% idle all day |
| 04:21.12 | JayTee52 | Opterons are feisty and solid but they definitely like a good heatsink/fan combo |
| 04:21.25 | JayTee52 | Semprons are just crap |
| 04:21.29 | jsoftw | yah |
| 04:21.47 | JT | why are semprons so bad? |
| 04:21.56 | *** join/#asterisk crbrocket (n=bob@fl-207-30-6-36.sta.embarqhsd.net) |
| 04:21.59 | jsoftw | they dont seem that fast. |
| 04:22.20 | JT | as in ridiculously slower than an opteron/athlon? |
| 04:22.23 | jsoftw | Ive got 4 small servers running them though, no problems |
| 04:22.28 | jsoftw | nah they are fast enough |
| 04:22.28 | JayTee52 | they run really hot, they've got a real small cache compared to the Opterons. They're the AMD equivalent of Intel's Celerons |
| 04:22.39 | JT | yes |
| 04:22.52 | JT | but how they perform in practice is better than comparing specs |
| 04:22.57 | JayTee52 | yep |
| 04:23.13 | jsoftw | In practace, my benchmarks show that they are fail. |
| 04:23.20 | JT | lol |
| 04:23.32 | JayTee52 | I wouldn't run a production server on one |
| 04:23.37 | jsoftw | *shrug* |
| 04:23.43 | JT | i probably wouldn't either |
| 04:23.45 | jsoftw | They have given me zero shit |
| 04:23.51 | JayTee52 | but then nothing I buy branded comes with anything other than Opterons or Xeons |
| 04:24.02 | jsoftw | about a year of uptime before I had to reboot for kernel updates |
| 04:24.44 | drmessano | I wouldn't run a server with a Sempron |
| 04:24.51 | jsoftw | *shrug* |
| 04:25.09 | JayTee52 | I've got * running on a Quad Core Xeon, 4GB RAM and 2 SAS drives with RHEL5 64bit and nothing slows it down. |
| 04:25.20 | drmessano | F Minus Minus for having the word PRON in it and not being good for PRON |
| 04:25.48 | *** join/#asterisk ahbritto (n=guest@adsl-69-104-245-232.dsl.pltn13.pacbell.net) |
| 04:26.38 | jeev | the on hold music on my asterisk.. i wanna change it to live streaming porn music |
| 04:27.11 | plik | sounds like a job for app_conference again |
| 04:27.17 | jsoftw | Give it a bit of IO and it will slow down. |
| 04:27.22 | jsoftw | Linux has sucky IO multitasking :\ |
| 04:28.18 | andresmujica | anyone knows how can i access a meetme room held at a remote pbx via an iax2 trunk? i can access remote extensions, but no conferences... |
| 04:28.38 | jsoftw | *shrug* |
| 04:28.44 | JayTee52 | well, I'll find out soon enough when I go live with it handling 4 PRI circuits on 2 Digium TE212P cards. |
| 04:29.08 | jsoftw | Trues. |
| 04:29.14 | JT | 2 digium cards at once, adventureous |
| 04:29.17 | jsoftw | Worked with some sangoma gear the other day |
| 04:29.22 | jeev | ok guys, i'm really curious.. is it possible to connect an existing nortel system to asterisk and f the telephone company lines ? |
| 04:29.29 | jsoftw | That stuff is nice, though we had to replace the echo cancelation chip, it had gone bung |
| 04:29.31 | [TK]D-Fender | andresmujica, If you can reach extensions. make sure your conference extens are accessable where you pass your call off to. |
| 04:29.36 | JayTee52 | JT, think it'll choke? |
| 04:29.49 | JT | JayTee52: maybe, depends on the computer |
| 04:29.51 | *** join/#asterisk |omni| (n=rob@c-67-185-180-203.hsd1.wa.comcast.net) |
| 04:30.06 | JT | might have mysterious interrupt/bad zap timing issues, might not |
| 04:30.15 | JayTee52 | JT, Dell PowerEdge 2950, Quad Core Xeon 4GB ram, 2 SAS drives. |
| 04:30.31 | JT | yeah i have no idea specifically if you'll run in to trouble or not with that |
| 04:30.33 | JayTee52 | zttest runs 99.98 ALL the time |
| 04:30.41 | JT | digium cards seem to like some hw |
| 04:30.45 | JT | not others |
| 04:30.54 | JT | JayTee52: it may go down during actual load |
| 04:30.57 | JayTee52 | jeev, I'm trying to do what you want to do |
| 04:30.59 | JT | the score |
| 04:31.05 | jeev | oh really |
| 04:31.17 | JayTee52 | JT, yeah, I'm setting up a load test before going into a live test scenario. |
| 04:31.18 | *** join/#asterisk sione (i=sione@ocs.net) |
| 04:31.19 | jeev | man i gave up on that shit, i've been using asterisk for 3-4 days now.. i'll just convert everyone to a headset/softphone. |
| 04:31.27 | jeev | who be your provider |
| 04:31.28 | [TK]D-Fender | JayTee52, What are you planning on doing with those calls? |
| 04:31.54 | JayTee52 | [TK]D-Fender, live calls or test calls? |
| 04:32.15 | [TK]D-Fender | JayTee52, Your 4 PRI's. What actual load is your server going to be doing? |
| 04:32.34 | [TK]D-Fender | JayTee52, load means call recording, conferencing, transcoding, etc. |
| 04:33.01 | JayTee52 | it's going to handling inbound outbound calls from/to our local telco and also routing calls that aren't in the dialplan to a Nortel Option 11c PBX. |
| 04:33.16 | sione | has a Zaptel issue with the TDM410 |
| 04:33.26 | [TK]D-Fender | JayTee52, So far that doesn't sound like any kind of load. Your box is serious overkill |
| 04:33.39 | JayTee52 | better more than less |
| 04:33.58 | [TK]D-Fender | JayTee52, Yup, so theres my vote of confidence in its (over)suitability. |
| 04:35.20 | JayTee52 | eventually when all of the Nortel extensions have been migrated to Asterisk and it's all working smooth we'll pull the plug on the Option 11 and the two PRI's going to it and sell the old girl on Ebay. |
| 04:36.30 | jeev | you gotta give up like me, fark the nortel system. |
| 04:36.39 | JayTee52 | this scenario will allow me to migrate in phases rather than a complete cutover all at once which is what I'd prefer but when you're a non-profit org with penny pinchers driving the bus you do what you can. |
| 04:37.04 | [TK]D-Fender | JayTee52, Why are they switching? |
| 04:37.41 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:37.42 | JayTee52 | lower cost of maintenance, equipment etc. Integration with Exchange UM and video conferencing. |
| 04:37.47 | C4away | robeph you still there? |
| 04:38.07 | JayTee52 | I can do it for about 20K in equipment costs versus 100K from AT&T for Nortel upgrades. |
| 04:38.17 | drmessano | Two more words I block out |
| 04:38.59 | JayTee52 | and that also eliminates about 30K a year in support contracts and makes my position more vital (but no one's indispensable) |
| 04:39.11 | jeev | put a backdoor in the system! |
| 04:39.33 | jeev | well, if you're running redhat, you could pretty much root yourself 1 week after you're fired. |
| 04:39.33 | JayTee52 | jeev, you mean in case they can me aftewards? |
| 04:39.47 | jeev | :> |
| 04:39.51 | jeev | ya |
| 04:39.56 | jeev | start routing their calls to porn 900 numbers |
| 04:40.11 | JayTee52 | that wouldn't be ethical! :P |
| 04:40.20 | jeev | :) |
| 04:40.38 | jeev | have cheated on your spouse ? |
| 04:40.48 | JayTee52 | nope |
| 04:40.58 | JayTee52 | or on my taxes |
| 04:41.04 | jeev | yea, taxes |
| 04:41.13 | JayTee52 | and I don't lie to any of my true friends, ever. |
| 04:41.16 | jeev | i['d never cheat on my girl.. but taxes, yes. |
| 04:41.22 | *** join/#asterisk Kyoshi (i=whoa@pool-71-167-117-15.nycmny.fios.verizon.net) |
| 04:41.23 | jeev | i dont lie.. except on taxes |
| 04:42.01 | JayTee52 | people that know me well trust me and know they can count on me when the chips are down and their backs to the wall. |
| 04:42.02 | Kyoshi | can i have 2 inbound routes for the same zap channel? 1 for a blank CID and 1 to capture 8NNXXXXXXX ? |
| 04:42.38 | JayTee52 | don't think so |
| 04:43.03 | JayTee52 | umm, wait |
| 04:43.44 | JayTee52 | if I understand your question you can route based on the pattern match and if there is no CID. for a second I thought you were saying 2 incoming contexts. |
| 04:44.56 | *** join/#asterisk loompek (n=NoName@noname.rula.net) |
| 04:44.58 | loompek | morning |
| 04:45.16 | JayTee52 | where you are! morning :-) |
| 04:45.58 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 04:46.00 | jeev | i dont rip off clients either |
| 04:46.01 | jeev | so |
| 04:46.05 | jeev | i was kidding about the backdoor |
| 04:46.13 | JayTee52 | I figured as much |
| 04:46.28 | *** join/#asterisk mecredis (n=fcb@cpe-74-73-135-105.nyc.res.rr.com) |
| 04:46.40 | jeev | i'm running asteriskNOW now but am considering asterisk... source manually crap all that wonderful stuff |
| 04:46.45 | mecredis | just got a weird "Crypto Support not loaded!" error |
| 04:46.51 | mecredis | what are my options? |
| 04:46.59 | mecredis | do I have to recompile asterisk? |
| 04:47.10 | JayTee52 | what version? |
| 04:47.41 | sione | anyone know how to prevent the FXO port on the TDM410 from picking up the line when another phone on the same analog line making a call? |
| 04:47.48 | mecredis | JayTee52.4.19 |
| 04:47.55 | mecredis | err, 1.4.19 |
| 04:48.16 | [TK]D-Fender | Kyoshi, Just do a catch-all, and GotoIf your way out to somewhere else where appropriate. |
| 04:48.17 | JT | sione: no, asterisk works like a pbx |
| 04:48.28 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:48.42 | sione | so you cant have alarm system on the same line? |
| 04:49.03 | mecredis | gah, looks liek I have to do a recompile |
| 04:49.05 | Kyoshi | tk, actually.. its more like.. i got a zap channel and i wanna catch all 800, 866 and 877 numbers that call in and send them to an ivr menu. |
| 04:49.13 | JT | not if you want to guarantee there's no interference, sione |
| 04:49.22 | sione | hmmm |
| 04:49.37 | JayTee52 | mecridis, what OS is it on? |
| 04:49.47 | mecredis | Linux |
| 04:49.53 | *** join/#asterisk bintut (n=bintut@bb121-6-167-197.singnet.com.sg) |
| 04:49.55 | JayTee52 | yeah, but what distro |
| 04:50.20 | mecredis | think its fedora |
| 04:50.23 | mecredis | its amazon ec2 |
| 04:50.25 | mecredis | which may be stupid |
| 04:51.20 | mecredis | ahh |
| 04:51.27 | [TK]D-Fender | Kyoshi, Just take the call in on "s" or a generic pattern match, and GOTOIF your way out for these cases. |
| 04:51.30 | mecredis | it seems that its commented out in asterisk.conf |
| 04:51.36 | JayTee52 | I'd recompile and make sure you check all your options when you do make menuselect but you might need to edit your make file. I saw a post about that error on one of the forums but it was for Debian I think. |
| 04:51.53 | JayTee52 | uncomment it and restart |
| 04:52.02 | mecredis | trying that now |
| 04:52.07 | JayTee52 | and check your modules.conf too |
| 04:52.09 | Kyoshi | thanks tk |
| 04:52.20 | JayTee52 | I've gotta go. Nite all |
| 04:52.44 | JayTee52 | don't stay up too late! |
| 04:52.53 | mecredis | gah |
| 04:52.58 | mecredis | no luck uncommenting it |
| 04:53.09 | sione | I have another problem with my zaptel driver, caller ID would not work unless I do "#define POLARITY_DEBOUNCE 6000" in wctdm24xxp.h |
| 04:53.11 | mecredis | will do make menuselect this time |
| 04:53.13 | mecredis | see what happens |
| 04:53.21 | mecredis | if you could point me to that forum it'd be great, JayTee52 |
| 04:53.59 | sione | if I dont do that, asterisk takes the caller ID polairty change as if the line was picked up |
| 04:54.33 | JayTee52 | mecredis, do you have OpenSSL loaded? Crypto is dependent on that |
| 04:54.41 | mecredis | ah just found that in the menuselect |
| 04:54.42 | mecredis | hahah |
| 04:54.46 | mecredis | christ! |
| 04:54.54 | mecredis | OK i'll figure it out |
| 04:54.59 | mecredis | thanks for the help |
| 04:55.04 | JayTee52 | there's a section on it in the book |
| 04:55.12 | mecredis | ah, lost my copy stupidy |
| 04:55.12 | *** join/#asterisk steliosk (n=Stelios@athedsl-282332.home.otenet.gr) |
| 04:55.20 | JayTee52 | ~book |
| 04:55.20 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 04:55.27 | JayTee52 | free download of the pdf |
| 04:55.32 | mecredis | ah, forgot that |
| 04:55.41 | mecredis | awesome thanks |
| 04:55.53 | JayTee52 | np, have a good one |
| 04:55.58 | mecredis | 'night |
| 04:56.24 | JayTee52 | [TK]D-Fender, drmessano good night to you both! |
| 04:56.42 | sione | anyone know a correct way to fix asterisk for not picking up the call when the CallerID tone is coming in? |
| 04:57.59 | sione | it seems asterisk cuts off the CallerID data when it picks up the call when revers polarity happens just before the CID tone |
| 05:00.00 | sione | my FXO line when a call comes it it does the follwoing "RING (Revers Polairty) CID DATA (Nomarl Polarity) RING" |
| 05:00.20 | robeph | anyone know why passing a call across an IAX trunk losing its CID info before going out the next hop |
| 05:01.46 | steliosk | sione : Are you using reverse polarity for signalling ? |
| 05:01.53 | sione | ya |
| 05:02.19 | steliosk | sione : then its "normal" |
| 05:02.28 | sione | so if I delay the pickup on reverspolaity in the header file the CID works fine |
| 05:02.51 | sione | normal for asterisk to cut it off? |
| 05:02.58 | steliosk | sione : theoretically it should |
| 05:03.20 | sione | cuz asterisk picks up right when the RP happens so the CID data did not make it |
| 05:04.35 | sione | the weird part about this is I have 2 lines coming in ffrom PacBell and one line does the RP on CID data and my other line does not which works fine. |
| 05:05.22 | steliosk | what happens if you disable Rp signalling ? |
| 05:05.23 | sione | tempted to call pacbell to ask why I have 2 diffrent signaling |
| 05:05.35 | sione | how I go about that? |
| 05:05.41 | sione | would like to test that |
| 05:08.43 | *** part/#asterisk jsoftw (n=Administ@60.234.135.124) |
| 05:08.45 | steliosk | answeronpolarityswitch=yes/no |
| 05:09.06 | sione | thought that was for FXS port |
| 05:09.09 | steliosk | sione : also check if your tip/ring is reversed |
| 05:09.49 | sione | I played with the tip/ring as well it was correct but I reverse it anyways as a test and did the same thing so I put it back to normal |
| 05:13.02 | sione | i made the change to my zapata.conf had no affect |
| 05:13.36 | steliosk | sione : usually the CID starts 200-300 ms after the reverse polarity signal |
| 05:13.57 | sione | ya thats the problem |
| 05:14.05 | sione | asterisk is picking up right away |
| 05:14.13 | sione | so the CID gets cut off |
| 05:14.54 | sione | log says "Got event 2 (Ring/Answered).." before CID data righ when RP happens |
| 05:15.47 | robeph | lol how annoying |
| 05:15.57 | robeph | I figured out my problem with CID being clobbered.... |
| 05:16.01 | steliosk | Does that happens if you set answeronpolarityswitch=no ? |
| 05:16.13 | sione | ya I just tested it with it set to no |
| 05:16.26 | sione | had no affect |
| 05:16.41 | sione | I thought that option was for FXS port only |
| 05:17.17 | steliosk | no its for the zaptel channel in general |
| 05:17.24 | sione | robeph: did you figure out how to fix it? |
| 05:17.34 | sione | ah ok |
| 05:18.10 | steliosk | what happens if you swap the lines i.e switch the ports the lines get connected to the card |
| 05:18.22 | sione | the problem follows the line |
| 05:18.37 | sione | that was the 1st thing I did when I installed the TDM410 thinking I had a bad card |
| 05:19.58 | sione | so my only fix was to add a 6sec delay when RP happens in the kernel module |
| 05:20.24 | steliosk | well that means that you do not have a warped config for the line in question and probably there is an issue with your provider |
| 05:20.33 | sione | the thing about that, I might forget about that hack and when I upgrade the zaptel module i will be back to the problem |
| 05:21.10 | steliosk | .me goes to get some coffee |
| 05:22.14 | sione | I was thinking opening a problen ticket with my telco but I feel they will say I am hooking up an unsupported device to the phone line |
| 05:23.33 | *** join/#asterisk MicW (n=michael@dslb-088-074-162-136.pools.arcor-ip.net) |
| 05:23.35 | MicW | hi |
| 05:24.51 | MicW | we are using chan_zap with qozap and a multiport card. when all lines are busy, an incomming caller hears a "currently not available" text. how can i change it to a bussy tone? (i think it has to do with isdn reject codes) |
| 05:27.00 | MicW | and where do i get "all-circuits-busy-now" as german sound file? |
| 05:35.21 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 05:36.17 | robeph | you can't set CID var remotely for a call that is passing through another |
| 05:36.22 | robeph | sione: yes |
| 05:43.21 | MicW | robeph: was this an answer to my question above? |
| 05:43.34 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 05:48.58 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 05:56.03 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 06:04.37 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:05.39 | *** join/#asterisk yang (i=yang@static-ip-62-75-255-125.inaddr.intergenia.de) |
| 06:07.10 | robeph | MicW: no it was to sione before you entered |
| 06:07.23 | robeph | sorry I'm slow, I been on the phone for like 7 hours with this custmer working the kinks outta these machiens |
| 06:08.37 | MicW | ;) |
| 06:15.54 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:19.02 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:20.32 | *** join/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 06:20.58 | arekm | looking for a solution of some asterisk problem. I see that someone already reported the problem.... but I don't know what's the bug number 8-) |
| 06:21.10 | arekm | is anyone able to tell which bug has this attachment? http://bugs.digium.com/file_download.php?file_id=18240&type=bug |
| 06:25.02 | *** join/#asterisk LiuYan (n=LiuYan_f@211.154.128.135) |
| 06:26.25 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 06:28.24 | *** join/#asterisk gmaruzz (n=maruzz@213-140-6-122.ip.fastwebnet.it) |
| 06:29.50 | arekm | found but the bug is about something else :\ |
| 06:32.33 | arekm | I have tons of: [Apr 10 08:32:24] DEBUG[20118] audiohook.c: Read factory 0x826da70 was pretty quick last time, waiting for them. |
| 06:33.03 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 06:35.51 | joelsolanki | HI room |
| 06:36.11 | joelsolanki | sip show channels show me 245 channels |
| 06:36.28 | joelsolanki | show channels shows me 5 active calls and 10 active channels. |
| 06:36.37 | joelsolanki | i see sip show channels show me stale entereies |
| 06:36.41 | joelsolanki | enteries |
| 06:37.00 | joelsolanki | which is creating problems when 60 calls simulteneously |
| 06:37.07 | joelsolanki | unable to allocate sip channel |
| 06:37.18 | Aces1up | man allocate deez |
| 06:37.18 | *** join/#asterisk Sir0x (n=cmayer@p5B0842A7.dip.t-dialin.net) |
| 06:37.25 | joelsolanki | 202.160.161.42 sarangpate GW5d286e241 00101/00001 unkn No (d) Rx: CANCEL |
| 06:37.34 | Sir0x | morning |
| 06:37.38 | joelsolanki | stale enteries are like above. |
| 06:38.00 | joelsolanki | CANCEL BYE INVITE |
| 06:38.10 | joelsolanki | so anybody know how to remove it. |
| 06:38.16 | joelsolanki | reload doesnt work |
| 06:38.22 | joelsolanki | i have to stop the asterisk and start it again |
| 06:39.25 | C4away | is there a way to set the context WaitExten executes in during a macro? the documenation doesn't say anything about setting a context like you can for Background |
| 06:39.50 | C4away | joelsolanki, Those calls are supposed to be destroyed by asterisk automatically |
| 06:40.11 | C4away | after 15 seconds or something |
| 06:40.28 | C4away | do you have that many showing up during the 15 seconds asterisk waits? |
| 06:40.42 | C4away | how many users do you have? |
| 06:40.54 | C4away | or sip registrations for phones/devices rather |
| 06:41.00 | joelsolanki | no it shows me still now. |
| 06:41.05 | joelsolanki | it doesnt go :( |
| 06:41.22 | C4away | well, if you have 60 calls active, you must have 100 or 200 devices registering right? |
| 06:41.28 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.134) |
| 06:41.30 | joelsolanki | NO. |
| 06:41.47 | joelsolanki | i have softswitch which registers to asterisk and makes call. |
| 06:42.08 | joelsolanki | yes that softswitch have lot of users. but simulteneously calls are always 20 to 25 |
| 06:42.13 | C4away | so you have one sip registration? |
| 06:42.18 | C4away | the softswitch? |
| 06:42.27 | joelsolanki | yes softswitch uses sip registeration |
| 06:42.37 | C4away | but it is the only sip user? |
| 06:43.07 | C4away | so any NAT keepalive packets or other misc packets the phones are sending out are going to softswitch and not the asterisk box right? |
| 06:43.17 | joelsolanki | yes only sip users |
| 06:43.44 | C4away | and those entries are staying the same and just increasing in number? |
| 06:43.45 | joelsolanki | correct. everything is handled at softswitch. |
| 06:43.59 | joelsolanki | correct it is staying there and always increasing. |
| 06:45.03 | joelsolanki | is there a process that i can remove it manually / |
| 06:45.04 | joelsolanki | ? |
| 06:45.21 | C4away | you can soft hangup each channel |
| 06:45.33 | joelsolanki | soft hungup doesnt show those stale enteries :( |
| 06:45.34 | C4away | but you have to know which ones are actual calls and which are bogus |
| 06:45.37 | C4away | oh |
| 06:45.47 | C4away | one sec |
| 06:45.49 | joelsolanki | ok |
| 06:47.49 | jeev | what is the best possible codec to use ? |
| 06:47.54 | C4away | do you have qualify=yes set for the softswich? |
| 06:47.59 | C4away | jeev: depends on your needs |
| 06:48.09 | jeev | hmm |
| 06:48.12 | jeev | high quality voice ? |
| 06:48.19 | C4away | high quality would be ulaw |
| 06:48.32 | C4away | or alaw if you are anywhere but north america |
| 06:48.42 | jeev | ahh |
| 06:48.54 | C4away | gsm is free and low bandwidth, that's the one I use |
| 06:48.57 | jeev | ok so i wanna connect 3 of my offices together |
| 06:49.14 | jeev | right now, i'm testing it on a dsl.. 768 up |
| 06:49.21 | jeev | but it'd obviously be smart to move it to my datacenter, right ? |
| 06:49.30 | C4away | will there be more than 5 or 10 calls on that line at any time? |
| 06:49.38 | jeev | quite possibly, yes. |
| 06:49.47 | C4away | gsm using IAX2 trunking is your best bet then |
| 06:49.49 | jeev | now, i do have a 1mbit up also at the office |
| 06:49.58 | jeev | i simply do not know what IAX2 is, my provider uses SIP |
| 06:50.00 | C4away | and putting the hub server in the datacenter would be a good idea |
| 06:50.06 | jeev | i want to softphone all the computers, have them x-lite to the datacenter |
| 06:50.09 | C4away | then connect each office pbx to that hub server |
| 06:50.17 | jeev | there will just be 1 pbx |
| 06:50.23 | C4away | oh, how many users will be in each office? |
| 06:50.23 | jeev | at the datacenter, is that dumb ? |
| 06:50.38 | jeev | about 14 in 1, 2 in another, 6 in the other |
| 06:50.43 | C4away | there are many ways to do it |
| 06:50.55 | C4away | in the smaller offices you could just use softphones or sip phones |
| 06:51.09 | jeev | all 3 offices have a nortel system |
| 06:51.09 | C4away | in the larger one(s) I would put a pbx and trunk to reduce overhead |
| 06:51.09 | *** part/#asterisk Aces1up (n=Now@ip70-173-52-152.lv.lv.cox.net) |
| 06:51.12 | jeev | well, the 2 user office doesn't |
| 06:51.34 | C4away | replacing the nortel or running the softphones for internal? |
| 06:51.45 | joelsolanki | C4away: yes i have qualify=yes |
| 06:51.47 | jeev | yea, i'm gonna leave the nortel, too much headache until i consolidate all 3 offices into a building |
| 06:51.49 | joelsolanki | for the softswitch config |
| 06:51.59 | jeev | so i will do softphones everywhere. |
| 06:52.36 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 06:52.48 | C4away | joelsolanki: the (d) in the channel means it is marked for destruction |
| 06:53.47 | C4away | joelsolanki: can you do a 'sip show history ' on one of the channes that is hung |
| 06:53.51 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 06:53.58 | C4away | then paste it to http://www.pastebin.ca |
| 06:54.06 | C4away | and give me the link |
| 06:54.19 | joelsolanki | ok |
| 06:54.57 | C4away | joelsolanki: the qualify=yes only applies to asterisk installations with many sip peers, it seems that wouldn't be causing this from what I can tell |
| 06:55.21 | joelsolanki | spare*CLI> sip show history |
| 06:55.21 | joelsolanki | Usage: sip show history <channel> |
| 06:55.21 | joelsolanki | <PROTECTED> |
| 06:55.32 | joelsolanki | i have to provide sip channel |
| 06:55.39 | joelsolanki | should i provide u any sip channels ? |
| 06:55.55 | C4away | any of them that are stale |
| 06:56.03 | C4away | a CANCEL or INVITE |
| 06:56.27 | joelsolanki | ok hold on |
| 06:56.47 | *** join/#asterisk Nuonec (n=nuonec@124.43.42.125) |
| 06:57.23 | joelsolanki | oh sip history recording is currently disabled. use sip history to enable :( |
| 06:57.24 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 06:57.29 | C4away | oh |
| 06:57.35 | C4away | yea |
| 06:57.37 | C4away | do that |
| 06:57.46 | joelsolanki | k |
| 06:57.52 | C4away | can you restart the box or are there active users? |
| 06:58.29 | joelsolanki | let me restart the asterisk ? |
| 06:58.31 | C4away | jeev: so you are using the astersik as a voice chat program for internal calls? |
| 06:58.43 | C4away | if you do it will clear up the old ones that history is not logged for |
| 06:58.52 | C4away | then you will just see the new ones as they show up |
| 06:59.03 | C4away | I don't know if you can, if it is production and you have users making calls don't worry |
| 06:59.12 | C4away | it will just be harder to find the ones that have history recorded |
| 07:00.47 | joelsolanki | yes i restarted it. |
| 07:00.56 | C4away | ok, what version of asterisk are you running? |
| 07:01.02 | joelsolanki | lets see once i have stale entery |
| 07:01.02 | C4away | CLI> version |
| 07:01.18 | C4away | sorry CLI> show version |
| 07:01.26 | joelsolanki | <PROTECTED> |
| 07:02.21 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 07:02.50 | C4away | is the ip address in the channel always the one for the softswitch |
| 07:02.53 | C4away | ? |
| 07:03.16 | Nuonec | I am trying to configure Asterisk with Festival. When I place a call using xlite the CLI says "Unable to read from cache/festival". Whats need to be done here? |
| 07:03.58 | joelsolanki | yes |
| 07:04.24 | C4away | is it possible that the two could be having problems with the network? |
| 07:04.29 | C4away | packets being lost/re-routed? |
| 07:04.37 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 07:05.28 | C4away | it seems that up until asterisk 1.4.18 there were problems with channels marked for destruction not being cleared properly if the other host did not respond |
| 07:05.32 | C4away | http://bugs.digium.com/view.php?id=1604 |
| 07:05.36 | joelsolanki | oh |
| 07:05.56 | C4away | update to 1.4.19 (make sure your 1.2 dialplan is ready to upgrade, test it on another server first) |
| 07:06.03 | C4away | that should solve the problem |
| 07:06.27 | joelsolanki | ok so upgrading to 1.4.18 should solve the issue |
| 07:06.47 | C4away | 18 had the problem 19 is the current version |
| 07:07.24 | joelsolanki | hmm |
| 07:07.43 | joelsolanki | i will give it a try |
| 07:07.48 | C4away | make sure to test your dialplan on a 1.4 server and set verbose to 6 on the CLI |
| 07:07.55 | C4away | watch the log files and make sure there aren't any warnings |
| 07:08.00 | C4away | many things have changed between 1.2 and 1.4 |
| 07:08.10 | joelsolanki | hmm |
| 07:08.23 | C4away | when you download the new source there will be a file in there called UPGRADE.TXT |
| 07:08.39 | joelsolanki | ok.. |
| 07:08.42 | C4away | read that, it has everything that has been deprecated or removed |
| 07:08.48 | C4away | and things that have ben added |
| 07:08.55 | joelsolanki | got it |
| 07:09.02 | C4away | are there calls on it 24/7? |
| 07:09.15 | joelsolanki | yes |
| 07:09.24 | joelsolanki | it is wholesale softswitch. |
| 07:09.31 | C4away | hmm |
| 07:09.35 | joelsolanki | i use for to terminationg calls to different voip provider |
| 07:09.51 | C4away | could you run asterisk -rx "restart when convenient" as a cron job? |
| 07:10.03 | C4away | every night at midnight or sometihng |
| 07:10.15 | C4away | then if there is one or two calls on it, it would wait until 0 active calls and restart |
| 07:10.16 | joelsolanki | do u think it will remove those stale enteries ? |
| 07:10.18 | C4away | that would be a workaround |
| 07:10.27 | C4away | doing a restart now clears them right? |
| 07:10.32 | C4away | "restart now" |
| 07:10.56 | joelsolanki | yes i think i can setup a cron job. |
| 07:10.58 | joelsolanki | :) |
| 07:11.36 | C4away | "restart when convenient" does the same thing as "restart now" just waits for 0 calls |
| 07:12.38 | C4away | make sure the "" is around the command that you want to run |
| 07:12.51 | C4away | asterisk -rx "restart when convenient" |
| 07:13.09 | C4away | if you have to escape them in the cron file (depending on how you set it up) make sure you do |
| 07:13.46 | joelsolanki | correct |
| 07:13.48 | joelsolanki | i will do that |
| 07:14.01 | C4away | will once every 24 hours work to clear them? |
| 07:14.13 | joelsolanki | yes 24 hours |
| 07:14.15 | C4away | you could do it at any time really, it will just sit there as long as there is still a call |
| 07:14.41 | C4away | although a new call might fail if it tries to connect while asterisk is restarting |
| 07:14.42 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:15.10 | joelsolanki | correct |
| 07:15.30 | joelsolanki | have u worked on codec translation with asterisk ? |
| 07:15.37 | joelsolanki | h323 to sip ? |
| 07:15.47 | C4away | uh, yes and no |
| 07:15.58 | C4away | ooh323 is installed through asterisk-addons |
| 07:16.04 | C4away | but I never got it to work properly |
| 07:16.14 | C4away | and that would be a format not a codec |
| 07:17.06 | C4away | codecs are just for the audio, like GSM, uLaw, etc |
| 07:17.38 | joelsolanki | hmm |
| 07:18.03 | joelsolanki | i had tested h323 but didnt worked properly but i have to give it a try again |
| 07:18.10 | C4away | yea, I never got h323 to work |
| 07:18.55 | C4away | did you ever get a sip show history <channel> output for me to look at? |
| 07:19.06 | C4away | just curious if it matches with the information I was reading in that bug report |
| 07:21.10 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:21.19 | kamanashisroy | wget cannot follow the redirection in asterisk addon download page .. :( |
| 07:21.34 | C4away | no, just grab the last half |
| 07:21.43 | C4away | ref=http://etc |
| 07:21.48 | C4away | from the http:// to the end |
| 07:21.54 | C4away | that is the actual file |
| 07:30.39 | joelsolanki | sorry i was away |
| 07:30.42 | joelsolanki | let me check |
| 07:31.59 | joelsolanki | yes i have 1 entry |
| 07:33.24 | joelsolanki | http://www.pastebin.ca/979303 |
| 07:33.33 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 07:35.19 | joelsolanki | give me 10 mins. brb. |
| 07:37.07 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 07:37.51 | *** join/#asterisk matt_ (n=matt@2001:770:168:1:20b:cdff:fe04:843a) |
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| 07:44.05 | bougie | hello :) |
| 07:45.06 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 07:57.43 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:57.47 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:02.09 | joelsolanki | C4away: u there ? |
| 08:02.18 | *** join/#asterisk bps (n=none@host.250.19.23.62.rev.coltfrance.com) |
| 08:04.54 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 08:21.05 | *** join/#asterisk qdk_ (n=qdk@85.235.253.139) |
| 08:22.37 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 08:30.15 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 08:38.36 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:40.16 | *** join/#asterisk chigital (n=chigital@p4FE5DEFC.dip.t-dialin.net) |
| 08:41.04 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 08:42.02 | *** join/#asterisk xenonex (n=xenonex@89.218.233.251) |
| 08:45.16 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 08:47.29 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 08:47.33 | ice_croft | hi all |
| 08:50.14 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:53.34 | *** join/#asterisk thesyko (n=theSyko@135.216.48.60.klj04-home.tm.net.my) |
| 08:57.43 | thesyko | hi there guys |
| 08:57.51 | thesyko | wanna ask |
| 08:58.11 | thesyko | whats the minimum hardware requirement to get asterisk working ? |
| 08:58.48 | jfg | few. it runs on my soekris :) |
| 08:59.08 | jfg | it depends what you want to do with your asterisk |
| 08:59.52 | jfg | for personal use, you don't need core quad server :) |
| 09:01.02 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 09:01.21 | thesyko | i mean can it run on a p 1 srv with 256 ram ? |
| 09:02.01 | jql | well, asterisk will suffer a bit if it has less than 100mhz of cpu time |
| 09:02.15 | mort_gib | How can I force a termination of a call from the cli |
| 09:02.27 | thesyko | hmmm |
| 09:02.34 | tzafrir | not force, but ask nicely: |
| 09:02.49 | tzafrir | soft hangup <tab><tab> and pick the channel name |
| 09:03.00 | tzafrir | Normally it works |
| 09:03.02 | thesyko | so its not advisable to run it from my old p1 machine rite |
| 09:03.11 | tzafrir | No |
| 09:03.23 | tzafrir | Maybe it will work for a single channel or so |
| 09:03.42 | tzafrir | But try to get a PII at the near-by junk yard |
| 09:03.45 | jql | thesyko: it would perhaps manage to keep up with packet-pushing, but I doubt you could setup an IVR |
| 09:04.03 | thesyko | maybe then its possible to run it from my hosted VPS server? |
| 09:04.15 | thesyko | it has 512mb ram |
| 09:04.22 | thesyko | and gentoo |
| 09:04.41 | jql | if it can start emacs in less than 3 seconds, you're gold |
| 09:05.25 | jql | tries it on his computer... ~3 seconds indeed |
| 09:09.08 | thesyko | jfg |
| 09:09.23 | thesyko | u said you gt asterisk working on your soekris board |
| 09:09.27 | thesyko | what model is it? |
| 09:09.35 | jfg | net5501 |
| 09:09.55 | *** join/#asterisk tiav (n=tiav@ivr94-4-82-227-121-53.fbx.proxad.net) |
| 09:09.56 | jfg | 512Mb of ram |
| 09:10.00 | thesyko | Ok |
| 09:10.08 | thesyko | how many users? |
| 09:10.09 | jfg | and yes you can with your p1 running asterisk ;) |
| 09:10.17 | jfg | just me and my girlfriend |
| 09:11.02 | *** join/#asterisk iunixan (n=iunixan@41.234.141.122) |
| 09:11.06 | yang | jfg: a P2 should not cost you more than 10 EUR |
| 09:11.17 | *** join/#asterisk scurb (n=scurb@194.218.238.2) |
| 09:11.32 | thesyko | hmmm |
| 09:11.44 | thesyko | i may just get a soekris board |
| 09:11.57 | thesyko | or some embeded system |
| 09:12.27 | jfg | on our providers boxes (in france) there is asterisk.. |
| 09:14.30 | jfg | i mean, they use an optimised version of asterisk with few features on a ~100mhz passive proc |
| 09:15.30 | *** join/#asterisk thesyko (n=theSyko@135.216.48.60.klj04-home.tm.net.my) |
| 09:15.53 | thesyko | sorry guys got dc just now |
| 09:17.55 | yang | thesyko: you might get a better performance on a VPS those cost ~5 EUR per month |
| 09:18.15 | thesyko | i have a VPS hosted with vr.org |
| 09:18.44 | thesyko | http://vr.org/virtual-server-plans/plans-and-pricing/ |
| 09:18.56 | thesyko | plan XV2 |
| 09:20.06 | yang | check www.netdirekt.de for better pricing |
| 09:21.14 | thesyko | the reason i chose vr.org because of their low latency and their vps performnce |
| 09:22.05 | jfg | ovh have better config at same price |
| 09:22.15 | jfg | http://www.kimsufi.com/ |
| 09:22.54 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 09:23.20 | thesyko | but honestly guys |
| 09:23.35 | thesyko | peering from europe to asia (malaysia) is very bad |
| 09:23.49 | thesyko | latency's are very high |
| 09:24.03 | thesyko | i tried many providers |
| 09:24.07 | yang | yep, you will get 300-500ms what isn't really something you can go with |
| 09:24.17 | thesyko | close to 10 already then i stoped at vr.org |
| 09:24.25 | yang | what kind of latency do you have to thailand? |
| 09:24.26 | thesyko | vr.org gives me max of 250ms |
| 09:24.52 | thesyko | you gotta gimme a server and i'll do a ping test |
| 09:29.04 | dandre | Hello, |
| 09:30.02 | dandre | Is there any possibility to include a context in another one base on a variable test condition? |
| 09:31.50 | mvanbaak | no |
| 09:32.03 | mvanbaak | but you can use a macro for that |
| 09:32.09 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 09:32.47 | mvanbaak | include => can only take a context name, and optionally timebased settings so you can include the context based no time |
| 09:33.02 | mvanbaak | but you cannot use it to include based on a variable |
| 09:36.03 | dandre | ok and how can I use a macro for that? |
| 09:41.25 | *** join/#asterisk PTuser1 (n=59b474a7@67.159.55.26) |
| 09:41.47 | mvanbaak | using Goto and IF |
| 09:42.08 | PTuser1 | hello |
| 09:42.49 | PTuser1 | I need some help, is anyone available to give me a few hints? |
| 09:43.31 | justdave | notices the room suddenly get quiet |
| 09:43.54 | justdave | probably would help if you said what you needed help with |
| 09:44.07 | justdave | (er, no pun intended there) |
| 09:44.20 | PTuser1 | ok |
| 09:45.06 | PTuser1 | i've a asterisk server 1.4.16 and wanted to upgrade do 1.4.19, But I can't find any documentation on upgrade procedures. |
| 09:45.43 | PTuser1 | Is the a specific procedure in order to maintain configurations and accounts and so on? |
| 09:46.14 | justdave | just don't "make samples" and you'll be fine :) |
| 09:46.37 | justdave | make install will only overwrite the program files and not your config |
| 09:48.10 | justdave | you basically just act like you're doing a fresh install, except you can just restart after installing instead of having to set up your config first |
| 09:49.40 | PTuser1 | justdave, any ideas? |
| 09:50.05 | PTuser1 | this web intrerface sucks |
| 09:54.37 | *** join/#asterisk PTuser1 (n=59b474a7@67.159.55.26) |
| 09:54.51 | PTuser1 | sorry, web interface sucks |
| 09:55.02 | PTuser1 | any help on my question? |
| 09:57.37 | mvanbaak | PTuser1: justdave answered already |
| 09:57.52 | mvanbaak | just do a new install, but DONT run 'make samples' |
| 09:58.01 | mvanbaak | after that, restart asterisk |
| 09:58.20 | mvanbaak | to be complete: |
| 09:58.33 | PTuser1 | ok thanks, and that will maintain the present configurations, right? |
| 09:58.34 | mvanbaak | tar zxvf asterisk-1.4.19.tar.gz |
| 09:58.38 | mvanbaak | cd asterisk-1.4.19 |
| 09:58.46 | mvanbaak | ./configure && make && make install |
| 09:58.53 | mvanbaak | asterisk -rx 'restart' |
| 10:00.27 | justdave | although you might want to throw a "make menuselect" in there and doublecheck that your build options are how you want them |
| 10:00.45 | *** join/#asterisk amaache (n=maache76@41.221.19.156) |
| 10:01.16 | justdave | the most likely thing you would probably have different than the default is the voice prompt or music on hold options (which packs to install) |
| 10:01.37 | justdave | in case you have different formats than the defaults or different languages |
| 10:07.48 | *** join/#asterisk jarod14 (n=jarod14@viatel-i3f-55-186.cnt.nerim.net) |
| 10:09.42 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 10:13.09 | mort_gib | How big a difference in real life is G729 over ulaw/alaw ?? |
| 10:15.12 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 10:16.05 | jfg | mort_gib: in real life ? |
| 10:17.21 | sysreq | mort_gib: bandwidth wise, ulaw takes approximately 10-15kb/s, G729 around 2-5kb/s.. as far as voice quality is concerned, i found them to be pretty much the same, but then i'm not an expert and didn't test g729 thorougly. |
| 10:17.33 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 10:17.48 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 10:17.51 | ice_croft | hi |
| 10:17.51 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 10:18.06 | ice_croft | how can i bind several address to listen sip? |
| 10:18.34 | ice_croft | bindaddr attr can be only 1? :) |
| 10:19.31 | bps | first link in google: http://www.voip-info.org/wiki/view/Asterisk+multiple+bindaddr+bounty |
| 10:19.40 | ice_croft | thanx man |
| 10:20.44 | *** join/#asterisk floppp (n=flopp@nat-staff.b3g-telecom.com) |
| 10:22.56 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.140) |
| 10:23.58 | *** join/#asterisk xenonex (n=xenonex@89.218.233.251) |
| 10:24.21 | *** join/#asterisk xenonex (n=xenonex@89.218.233.251) |
| 10:25.45 | mort_gib | sysreq: I'm finishing off an installation, and it would make sense to make all my clients go for G.729, like supporting Digium andf all |
| 10:25.48 | *** join/#asterisk xenonex (n=xenonex@89.218.233.251) |
| 10:26.10 | mort_gib | It would be even better if it made a difference... |
| 10:26.45 | *** join/#asterisk xenonex (n=xenonex@89.218.233.251) |
| 10:27.53 | *** join/#asterisk PTuser1 (i=59b474a7@gateway/web/ajax/mibbit.com/x-f6757d68cd749129) |
| 10:28.01 | *** join/#asterisk xenonex (n=xenonex@89.218.233.251) |
| 10:28.05 | PTuser1 | finally |
| 10:28.18 | PTuser1 | I think i found a decent web irc client |
| 10:28.43 | tzafrir | cgiirc is nice |
| 10:28.45 | *** join/#asterisk xenonex (n=xenonex@89.218.233.251) |
| 10:28.57 | PTuser1 | i'm using mibbit |
| 10:29.15 | PTuser1 | i can't remember who I was talking to |
| 10:29.35 | *** join/#asterisk a-s (n=user@89.38.174.194) |
| 10:29.35 | PTuser1 | i asked for guidance on upgrade asterisk 1.4.16 to 1.4.19 |
| 10:29.53 | a-s | hello |
| 10:29.58 | PTuser1 | hi |
| 10:30.34 | sysreq | mort_gib: well, the big advantage is that it takes a lot less bandwidth to get similar voice quality. |
| 10:31.01 | PTuser1 | really? |
| 10:31.07 | mort_gib | I was just wondering if that was always the case in real life... |
| 10:31.08 | a-s | I have a phone that sends me RTP packets of 10 ms and 20 ms , combinated. The timestamps supersposes in the moment that a packet of 10ms comes |
| 10:31.33 | PTuser1 | I want to upgrade, but dont wanna lose configurations |
| 10:31.35 | mort_gib | I mean, my clients are small |
| 10:31.41 | PTuser1 | how can I do this? |
| 10:32.06 | tzafrir | PTuser1, get shell hosting and use irssi |
| 10:32.31 | tzafrir | Run it in a screen |
| 10:32.52 | PTuser1 | thx tzafrir |
| 10:32.58 | a-s | for example: |
| 10:32.58 | a-s | RTP 20ms timestamp 1000 |
| 10:32.58 | a-s | RTP 20ms timestamp 1160 |
| 10:32.58 | a-s | RTP 10ms timestamp 1080 |
| 10:33.01 | a-s | RTP 10ms timestamp 1160, and so on |
| 10:33.36 | a-s | what shell I do ? to write the 1st packet of 10ms over the last of 20ms, or to append it to the buffer? |
| 10:35.49 | *** part/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 10:36.09 | sysreq | mort_gib: you get the final say, but it's probably not worth it if your clients and asterisk servers are confined to a lan.. if a lot of calls have to go through a smaller pipe (internet) though, then you should consider g729.. |
| 10:36.11 | hi365 | anyone ever have this? SPEECH_SCORE(0) isnt getting set (using Speechabackground w/ Lumenvox) |
| 10:37.39 | mort_gib | I don't, maybe 2 concurrent calls at any given time, -Supporting Digium?? I don't use Digium hardware |
| 10:37.48 | a-s | can somebody explain me, please, what the timestamp field means? |
| 10:38.10 | PTuser1 | is there a specifi procedure to upgrade asterisk, but not losing present configurations? |
| 10:38.39 | tzafrir | 'make install' does not overwrite configuration |
| 10:38.41 | sysreq | PTuser1: just don't run 'make samples' at the end of the installation, and you'll keep your current .conf files. |
| 10:38.58 | PTuser1 | ok many thanks |
| 10:39.01 | tzafrir | Hopefully the old configuration will still work with the new server... |
| 10:39.18 | PTuser1 | i'm upgrading 1.4.16 to 1.4.19 |
| 10:39.26 | tzafrir | That should generally work |
| 10:39.47 | tzafrir | I have a little script to check most things without really installing |
| 10:39.58 | PTuser1 | thsi version improves performance or just bugs and fixes? |
| 10:40.24 | PTuser1 | tzafrir, what does script do? |
| 10:40.35 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 10:40.59 | tzafrir | http://bugs.digium.com/11680 |
| 10:42.15 | tzafrir | basically make install to a subdirectory, and (almost) run asteirks as usual from there |
| 10:42.17 | PTuser1 | thanks i'll try it |
| 10:42.28 | tzafrir | Slightly edit the generated config, and you'll use /etc/asterisk |
| 10:43.08 | sysreq | mort_gib: i wouldn't really worry about that, digium is a business after all and in my opinion will still have a bright future as open source voip spread, even if you don't go for that g729 license ;p if you really want to support the asterisk project, i'd suggest testing new releases and filing bugs. |
| 10:43.45 | *** join/#asterisk apollonx (i=kit@193.19.189.38.STATIC.ISP.KZ) |
| 10:44.17 | PTuser1 | from your experience, do you think it's worthwhile to upgrade from 1.4.16 to 19? |
| 10:44.39 | mort_gib | Ah, but the bugs only become apparent when you have 20+ users screaming :-) |
| 10:45.01 | mort_gib | But I understand what you mean |
| 10:45.06 | sysreq | ;) |
| 10:45.57 | PTuser1 | i have about 10 users |
| 10:46.31 | *** join/#asterisk chigital (n=chigital@91.90.144.102) |
| 10:46.33 | PTuser1 | i thought version 19 would bring better performance, but is it only bug fixes? |
| 10:48.32 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 10:48.33 | sysreq | PTuser1: just look at the ChangeLog and see for yourself.. http://svn.digium.com/view/asterisk/tags/1.4.19/ChangeLog?view=markup |
| 10:49.21 | *** join/#asterisk Kalamansi (n=k@unaffiliated/davao) |
| 10:49.22 | PTuser1 | thanks |
| 10:49.37 | Kalamansi | hello asterisk |
| 10:49.56 | Kalamansi | what is asterisk? is it operating system? |
| 10:50.09 | Kalamansi | i know freebsd,openbsd,netbsd but this one is different |
| 10:50.13 | sysreq | Kalamansi: www.asterisk.org ..? |
| 10:51.00 | Kalamansi | sysreq yes but i dont see any word that says its an operating system.... |
| 10:51.10 | mort_gib | It's not |
| 10:51.16 | sysreq | maybe because it's not? |
| 10:51.17 | PTuser1 | lol |
| 10:51.18 | sysreq | heh |
| 10:51.51 | Kalamansi | ok thanks |
| 10:52.35 | tzafrir | Asterisk is a character |
| 10:52.37 | sysreq | Kalamansi: it's a toolkit, a piece of software.. some linux distributions come pre-packaged with it though. |
| 10:52.55 | tzafrir | ~wiki asterisk |
| 10:53.31 | tzafrir | duh. We're not important enough to be the first hit on wikipedia |
| 10:57.14 | PTuser1 | have to go |
| 10:57.19 | PTuser1 | many thanks |
| 10:57.25 | PTuser1 | see ya |
| 10:57.58 | Kalamansi | thank you |
| 10:58.09 | Kalamansi | ill try to install asterisk in gentoo |
| 11:02.19 | Kalamansi | sysreq theres a job opening . they pay 600usd per month is it cool nice salary? |
| 11:03.15 | a-s | I have a phone that sends packets on 2 different ports... an ArtDio |
| 11:03.29 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 11:04.35 | sysreq | Kalamansi: it depends on where you're located, what's the job and how many hours you'll do a week, i guess.. |
| 11:05.41 | sysreq | but 600usd/mo is really not a lot |
| 11:06.02 | Kalamansi | monday to friday 8 hours. i am from philippines. my expertise are based in server. freebsd,openbsd,netbsd,gentoo and windows. and the work or company is USA base |
| 11:06.37 | mort_gib | Kalamansi: Are you looking for work?? |
| 11:06.40 | Kalamansi | never tried handling asterisk. i thought it was an os. i know how to secure server and web/file server but not that good |
| 11:06.54 | Kalamansi | mort_gib : yes. someone offerd me that asterisk |
| 11:07.18 | hi365 | anyone ever have this? SPEECH_SCORE(0) isnt getting set (using Speechabackground w/ Lumenvox) |
| 11:07.27 | sysreq | getting paid 3.75$ an hour, here (north america), would be called exploitation ;p |
| 11:07.28 | mort_gib | So they want you to do Asterisk for them 8 hours a day, 5 days a week for 600 USD?? |
| 11:07.38 | Kalamansi | they give me these Asterisk, and VoIP |
| 11:07.55 | mort_gib | Yep, even in Europe that would be on the petty side of salaries |
| 11:07.59 | Kalamansi | mort_gib : yes |
| 11:08.26 | mort_gib | Dude if you are good with those things you can get a damn sight more |
| 11:08.48 | Kalamansi | mort_gib : they offer it to me : Asterisk PBX including SIP and Asterisk's AGI and C API (SIP Express Router and FreeSwitch) |
| 11:09.16 | Kalamansi | i thought asterisk is an os hehe. my mistake. |
| 11:09.24 | mort_gib | ?? Sorry, they offered you a position taking care of that stuff?? |
| 11:09.33 | Kalamansi | yes |
| 11:09.41 | Kalamansi | for 600USD |
| 11:09.42 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 11:09.45 | Kalamansi | fair enough mate? |
| 11:10.21 | mort_gib | Nope, I would pay more for a skilled worker, seems you need to read up a bit though |
| 11:10.23 | mort_gib | :-) |
| 11:15.10 | C4away | are you moving to the USA to do this or will it all be remote? |
| 11:18.35 | *** join/#asterisk Rico29 (n=rico@free.lycee-ledantec.ac-rennes.fr) |
| 11:18.39 | Rico29 | hi |
| 11:19.21 | Rico29 | how can I do an action (i.e. modify a variable) when a call ends ? |
| 11:21.29 | *** join/#asterisk scurb (n=scurb@194.218.238.2) |
| 11:21.34 | *** part/#asterisk scurb (n=scurb@194.218.238.2) |
| 11:21.46 | Rico29 | i've seen the 'g' option in Dial cmd, but it's just when the called party hangs up |
| 11:22.08 | Rico29 | (sorry for english-writing errors) |
| 11:23.11 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:26.49 | Rico29 | anybody has an idea ? |
| 11:28.39 | Rico29 | ok... |
| 11:31.41 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 11:33.25 | Kalamansi | Rico29 mate its late some americans sleep |
| 11:33.34 | Kalamansi | Rico29 what time is it in your country? |
| 11:33.38 | kamanashisroy | somehow the asterisk realtime extension loading is not working .. :( res_mysql_realtime is connected to mysql .. extconfig is specifying the exact table ..that table is configured correctly .. but alas when I do dial s@myrealtimecontext it fails :( |
| 11:33.47 | Kalamansi | time in america i guess its early in the morning |
| 11:33.55 | Rico29 | 2 PM :) |
| 11:34.15 | Rico29 | kamanashisroy, > firewall ? |
| 11:34.43 | kamanashisroy | Rico29: not sure what firewall can do !! .. |
| 11:34.43 | Kalamansi | Rico29 : its probably 3am in america i guess. for some helper i guess from america |
| 11:35.04 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 11:35.05 | Rico29 | when you do a "realtime load <name of table in extconfig> <field in database><name in database>" in the CLI, what's the answer ? |
| 11:35.09 | kamanashisroy | I have checked my laptop db (show processlist) .. it is showing that asterisk is connected .. |
| 11:35.25 | Kalamansi | kamanashisroy where you from? |
| 11:35.30 | kamanashisroy | Rico29: thanks let me try .. |
| 11:35.30 | Kalamansi | kamanashisroy are you from philippines? |
| 11:35.31 | Rico29 | i.e. "realtime load sippeers name like 205 |
| 11:35.47 | kamanashisroy | Kalamansi: no .. from Bangladesh .. why ? |
| 11:36.35 | kamanashisroy | Rico29: no such command realtime load |
| 11:36.58 | *** join/#asterisk the_5th_wheel (n=edd@41.223.244.9) |
| 11:37.01 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 11:37.33 | Kalamansi | kamanashisroy just asking. |
| 11:37.42 | Kalamansi | kamanashisroy can you teach me asterisk? hehehe |
| 11:37.54 | kamanashisroy | Rico29: realtime mysql status .. it is saying that I asterisk is connected to mysql fine .. |
| 11:38.10 | kamanashisroy | Kalamansi: I do not know asterisk actually ;) |
| 11:38.12 | Rico29 | kamanashisroy, >PV |
| 11:38.14 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:38.39 | Kalamansi | kamanashisroy then why are you here? hehehe c`mon teach me |
| 11:38.48 | Kalamansi | hey Rico29 teach me asterisk |
| 11:38.53 | Kalamansi | wb sysreq |
| 11:39.22 | sysreq` | thanks. |
| 11:39.23 | Rico29 | i'm at work |
| 11:40.41 | kamanashisroy | Rico29: once I wrote realtime resource myself ;) |
| 11:41.09 | Kalamansi | sysreq` you have two nicks whats wrong? lag at the second nick? |
| 11:42.16 | sysreq` | Kalamansi: i have an irc client opened both at home and now at work. |
| 11:43.32 | *** join/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 11:44.00 | Kalamansi | sysreq nice and cool :) |
| 11:45.00 | *** join/#asterisk Great_Randew (n=Andrew@mail.claired.com) |
| 11:47.38 | kamanashisroy | I hate trixbox .. they ruin my life .. |
| 11:47.47 | kamanashisroy | :) |
| 11:53.00 | Kalamansi | kamanashisroy hehehe why not try gentoo? |
| 11:53.53 | kamanashisroy | I always compile asterisk in my computer .. but my clients buy them from some unknown places .. (where I do not know mysql root password) |
| 11:59.21 | kamanashisroy | Here( http://pastebin.com/m95afdae ) is my debuging commands, it seems that in asterisk console the inbound_main context is not available wheras it is there in mysql web_context_view :( . |
| 11:59.54 | kamanashisroy | I enabled the full logging in logger.conf too .. |
| 12:00.19 | kamanashisroy | but there is no res_mysql_config logs searching the dialplan .. |
| 12:00.32 | kamanashisroy | is there anything that I am missing ? |
| 12:01.42 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 12:03.46 | *** join/#asterisk quigon (n=matias@nat/redhat/x-2310d0a26c9d0409) |
| 12:08.31 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 12:11.29 | C4away | how do you look up the callerid= info on a peer? |
| 12:11.40 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:13.12 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:13.31 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:14.16 | *** join/#asterisk Tili (n=tili@58.27.158.19) |
| 12:15.05 | Rico29 | doeas anybody knows how to do an action when the call ends (called OR callee hangs up) |
| 12:15.08 | Rico29 | ?? |
| 12:15.23 | tzafrir | the h extension? |
| 12:15.52 | Rico29 | ok, i'l try |
| 12:15.57 | Rico29 | thanks a lot |
| 12:20.00 | kamanashisroy | ^^^ I answer my question , you need to switch realtime in extensions.conf to load a context from realtime .. |
| 12:20.36 | *** part/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 12:24.01 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 12:24.39 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 12:28.15 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 12:30.32 | Rico29 | tzafrir, > thanks, it works ;) |
| 12:31.01 | *** join/#asterisk tobias (n=tobias@user-0ce2hpk.cable.mindspring.com) |
| 12:37.26 | *** join/#asterisk _Krieger_ (n=krieger@193.39.118.158) |
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| 12:46.32 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 12:46.59 | *** join/#asterisk guillote_GNU (n=guillote@host100.201-253-185.telecom.net.ar) |
| 12:47.35 | *** join/#asterisk Telemac (n=telemac@213.223.113.74) |
| 12:47.37 | Telemac | Hello |
| 12:48.58 | kamanashisroy | hi, can I use 'sip/123@mygateway.com' as queue_member interface ? |
| 12:49.13 | *** join/#asterisk jmesquita (n=jmesquit@200.170.114.149) |
| 12:49.13 | Telemac | Is there any misdn guru there ? |
| 12:49.58 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 12:50.54 | *** join/#asterisk bougie (n=bougie@APoitiers-256-1-10-5.w90-11.abo.wanadoo.fr) |
| 12:54.08 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:54.08 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:54.22 | x86 | morning |
| 12:54.25 | x86 | heya lmadsen |
| 12:55.43 | *** join/#asterisk tobias (n=tobias@user-0ce2hpk.cable.mindspring.com) |
| 12:59.37 | lmadsen | howdy |
| 13:00.31 | yang | heya lmadsen |
| 13:00.46 | lmadsen | thinks he hears an echo |
| 13:01.14 | yang | heya heya hey he h .... :) |
| 13:01.41 | yang | bbl |
| 13:02.31 | [TK]D-Fender | Echo Cancelled! |
| 13:03.42 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 13:04.49 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:06.04 | *** join/#asterisk Darthclue (n=chatzill@fw149.nisd.net) |
| 13:07.30 | *** join/#asterisk jelkimantis (n=jelkiman@dynamic-acs-24-154-50-205.zoominternet.net) |
| 13:07.49 | jelkimantis | Hi all! |
| 13:07.59 | *** join/#asterisk loompek (n=NoName@noname.rula.net) |
| 13:08.05 | loompek | morning.. smee again :D |
| 13:08.34 | loompek | anyone of you guys ever played with videophones? |
| 13:08.41 | jelkimantis | Here's a question: why does echo sometimes occur on a teleconference? That is, what is the most common cause of this? |
| 13:09.19 | *** join/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 13:09.39 | loompek | i'd like to test the capabilities... using h263.. is there any way to transcode a test movie in h263 and play it using playback or something? |
| 13:10.14 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-48-245.pskn.east.verizon.net) |
| 13:10.32 | [TK]D-Fender | loompek: Hi.. goan f.... ;) |
| 13:10.56 | loompek | anyways.. the phone is grandstream gvx-3000 |
| 13:11.07 | loompek | s/gvx/gxv/ |
| 13:12.07 | jelkimantis | is trying to figure out asterisk in some meaningful way. |
| 13:12.41 | jelkimantis | sorry, that was meant for another room. |
| 13:12.44 | dandre | how can I restrict all my iax accounts to a network range addresses? I have tried to pu deny = ... in global section of iax2.conf with no success |
| 13:13.11 | *** join/#asterisk M1s3ry (n=M1s3ry@216.207.245.1) |
| 13:14.01 | *** join/#asterisk drehlecom (i=ircbnc@2001:6f8:1153:2:208:c7ff:feac:d1fb) |
| 13:15.20 | dandre | sorry in general section of iax.conf |
| 13:18.20 | [TK]D-Fender | dandre: read the big print : http://www.voip-info.org/wiki/index.php?page=Asterisk+config+iax.conf |
| 13:19.37 | dandre | ok thanks, |
| 13:21.24 | kamanashisroy | where is my console commands ? console dial s@inbound_main -> no such command 'console dial' !! |
| 13:22.10 | kamanashisroy | what can I do to get back the console commands ? |
| 13:23.33 | [TK]D-Fender | kamanashisroy: What ver of *? |
| 13:23.46 | kamanashisroy | 1.4 |
| 13:23.52 | [TK]D-Fender | kamanashisroy: Go verify exactly. |
| 13:24.15 | kamanashisroy | [TK]D-Fender: Asterisk 1.4.16.2 |
| 13:24.26 | [TK]D-Fender | kamanashisroy: go check what modules are loaded. |
| 13:24.45 | RoyK | remembers ast 0.6 |
| 13:25.49 | kamanashisroy | [TK]D-Fender: what is the module responsible for console dial commands ? |
| 13:25.51 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:26.05 | kamanashisroy | [TK]D-Fender: please let me know, I shall menuselect that .. |
| 13:26.07 | [TK]D-Fender | kamanashisroy: probably a PBX one. |
| 13:26.22 | kamanashisroy | [TK]D-Fender: so it must be loaded .. |
| 13:26.29 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 13:26.39 | [TK]D-Fender | kamanashisroy: Go look in detail yourself. You've been cutting corners blindly... |
| 13:27.29 | kamanashisroy | [TK]D-Fender: you are right .. I feel blind (earlier I found the same issue, now I am lost ..) |
| 13:28.09 | [TK]D-Fender | kamanashisroy: Stop trying to cherry-pick your modules and auto-load them all and "noload" specific ones you know you don't want |
| 13:28.30 | kamanashisroy | [TK]D-Fender: :) thanks for clue |
| 13:28.32 | [TK]D-Fender | <PROTECTED> |
| 13:29.07 | coppice | mod_sakura |
| 13:30.43 | *** join/#asterisk LemensTS (i=CustomGT@adsl-70-238-131-138.dsl.stlsmo.sbcglobal.net) |
| 13:30.46 | *** join/#asterisk Shotygun (n=thorn@213.31.43.3) |
| 13:31.27 | Shotygun | Hello. How can I determine if a SIP destination has DND enabled so I can provide a better result in Dial() than CONGESTION ? |
| 13:31.37 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.235.113) |
| 13:31.47 | LemensTS | WHen I call the ${EPOCH} variable, it is giving me a time that is 5 hours off what my asterisk box is set at |
| 13:32.55 | kamanashisroy | [TK]D-Fender: I remembered .. |
| 13:32.59 | *** join/#asterisk macros73 (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 13:33.11 | kamanashisroy | [TK]D-Fender: the server does not have a sound device .. etc .. |
| 13:33.19 | kamanashisroy | [TK]D-Fender: no OSS module .. |
| 13:33.28 | kamanashisroy | [TK]D-Fender: > finally no dial .. |
| 13:33.35 | kamanashisroy | [TK]D-Fender: thank you for help |
| 13:34.13 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
| 13:34.29 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 13:38.30 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:38.30 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:38.32 | *** join/#asterisk yang (i=yang@static-ip-62-75-255-125.inaddr.intergenia.de) |
| 13:39.47 | yang | sip show peers (status) doesn't seem to show the correct values - everything seems to be about 60ms over the actual value...? |
| 13:41.00 | *** join/#asterisk jcims (n=jcims4@cpe-71-72-93-210.columbus.res.rr.com) |
| 13:41.20 | agx | is there a way to set 40ms as packet size when using G.711 ? |
| 13:42.30 | jcims | is anyone working on an app to create and make notification calls from Asterisk? e.g. i'm a baseball coach and want to call all my players to tell them practice is cancelled. pull up app, record message, pick distribution list, go |
| 13:43.44 | *** join/#asterisk duckz (n=duckz@81-180-102-217.etth.opensys.ro) |
| 13:43.49 | *** join/#asterisk ]technophreak[ (n=]technop@modemcable048.23-81-70.mc.videotron.ca) |
| 13:44.22 | ]technophreak[ | would anyone know if realtime music on hold is actually part of 1.4.18 ? |
| 13:44.31 | ]technophreak[ | or was it just a patch I have to apply ? |
| 13:45.31 | keith4 | LemensTS: what time zone are you in? |
| 13:45.36 | LemensTS | Central |
| 13:45.52 | LemensTS | VoIP timeclock_1_04 # date |
| 13:45.52 | LemensTS | Thu Apr 10 08:45:43 CDT 2008 |
| 13:46.57 | keith4 | isn't central GMT-5 currently? |
| 13:49.19 | [TK]D-Fender | GMT-6 |
| 13:49.37 | ]technophreak[ | anyone has an answer ? |
| 13:50.13 | keith4 | sure, it's GMT-6 normally, but it's daylight savings time |
| 13:50.36 | [TK]D-Fender | jcims: Go make a bunch of call-files and drop them into the spool folder. Or make a web-app or something to make it pretty and maybe use AMI instead |
| 13:58.49 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 14:03.55 | anonymouz666 | call-files sucky. |
| 14:04.19 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 14:04.23 | anonymouz666 | unless you use chan_local does not store CDRs. |
| 14:04.48 | ManxPower | yang: "sip show peers" does NOT show the network latency |
| 14:05.18 | ManxPower | yang: it shows how long it takes to get a response to an SIP OPTIONS packet from the remote device. |
| 14:07.50 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) |
| 14:08.04 | [TK]D-Fender | ]technophreak[: Yes, it'd be an add-on patch as it wouldn't be added to mainline. It may be accepted for 1.6 possibly |
| 14:08.17 | ]technophreak[ | hmm shit |
| 14:08.42 | ManxPower | ]technophreak[: not all that many people use realtime |
| 14:08.54 | ManxPower | even many Asterisk service providers do not use Realtime |
| 14:08.55 | jcims | [TK]D-Fender: yeah, just wondering if someone already did part of it...lol. i'm thinking about extending poMMo (email distribution) as it has a great base |
| 14:08.58 | [TK]D-Fender | ]technophreak[: Sorta the "big print" about how patches are accepted in. More or less a feature lock unless its part of a bug-fix |
| 14:09.17 | file | but those who do... want everything including the kitchen sink to use it |
| 14:09.24 | ManxPower | If the feature was not in 1.4.0 then it won't be in 1.4.19 |
| 14:09.27 | jcims | there's no over the wire protocol to feed audio to asterisk is there? |
| 14:09.39 | jcims | aside from standard file transfer stuff |
| 14:09.46 | [TK]D-Fender | ManxPower: really sad. means no new ideas for years... |
| 14:09.47 | ManxPower | jcims: you are not making any sense |
| 14:09.54 | ]technophreak[ | would this patch be stable ? |
| 14:09.59 | ManxPower | [TK]D-Fender: it means fewer bugs |
| 14:10.25 | file | the 1.6 release cycle changes things... |
| 14:10.30 | [TK]D-Fender | jcims: CC a mailer daemon that will see the topic and trigger your callout using the list of other CC's as a look-up |
| 14:10.35 | file | 1.6.0->1.6.1 will add new features for example |
| 14:10.36 | ManxPower | Well in theory fewer bugs -- it does not seem to help with the recent 1.4s |
| 14:10.43 | jcims | ManxPower: sorry. for example, if i have an app that i want to make calls (aforementioned web app), i need to integrate with asterisk using file transfer to push audio content, then something else to write call files or do agi |
| 14:10.50 | [TK]D-Fender | file: THANK JEEBUS |
| 14:10.50 | jcims | n00b here...obviously :) |
| 14:10.57 | ]technophreak[ | what do you mean does not seem to help .. ? |
| 14:11.10 | ManxPower | [TK]D-Fender: you'll stop thanking when your shiny new asterisk update segfaults on you. |
| 14:11.27 | ManxPower | ]technophreak[: I mean the new release cycle does not seem to reduce bugs |
| 14:11.28 | [TK]D-Fender | ManxPower: unload chan_curmudgeon.so |
| 14:11.28 | file | [TK]D-Fender: but then we've got people complaining that we are releasing too often and shouldn't add new features... can't please everyone! |
| 14:11.53 | [TK]D-Fender | file: Then learn to prioritize, I'm right here! :p |
| 14:11.56 | ManxPower | file: Digium customers that SPEND MONEY want a stable non-crashing, non-segfaulting Asterisk. |
| 14:12.22 | file | ManxPower: ...ABE? |
| 14:12.32 | ManxPower | [TK]D-Fender: Shall I forget the segault problem with 1.4.19? |
| 14:12.34 | [TK]D-Fender | ManxPower: Not to say that the flood-gates should be opened wide, but right now they're shut and we wait years for important stuff. |
| 14:12.59 | ManxPower | [TK]D-Fender: maybe they should do releases more often? |
| 14:13.01 | [TK]D-Fender | ManxPower: And 1.2 never had issues, yadda yadda... |
| 14:13.14 | [TK]D-Fender | ManxPower: All of life is a tradeoff.... |
| 14:13.20 | ManxPower | [TK]D-Fender: The new release cycle is supposed to reduce bugs -- it does not seem to be. |
| 14:13.37 | [TK]D-Fender | ManxPower: Well time will tell as always |
| 14:13.40 | ]technophreak[ | Of course I dont want it segfaulting ... |
| 14:13.47 | ManxPower | file: really I don't have a lot of confidence in ABE. |
| 14:13.56 | file | so be it. |
| 14:14.01 | anonymouz666 | ManxPower is very very pessimist. :) |
| 14:14.13 | ]technophreak[ | I just dont want to have to update th config file and reload everytime .. that's the whole purpose of realtime |
| 14:14.35 | ManxPower | anonymouz666: I've been screwed by Asterisk bugs many times -- so many times that I basically don't update Asterisk anymore. |
| 14:15.52 | anonymouz666 | yes, If it works, I don't update. That's why I still run 1.2 in many customers. |
| 14:15.59 | M1s3ry | ManxPower, that just means you've found bugs that can be fixed now. :p |
| 14:16.02 | mocker | 1.2 for life. :) |
| 14:16.38 | ManxPower | M1s3ry: I prefer dealing with bugs I know about rather than new bugs. |
| 14:16.39 | florz | M1s3ry: Well, finding bugs in asterisk really is simple. Fixing them is not, and so nobody does fix them. |
| 14:16.58 | file | nobody fixes them? |
| 14:17.06 | file | I'm insulted by that. |
| 14:17.31 | florz | file: Given the huge number of bugs still in there, I am at least not getting the impression from the source =:-) |
| 14:17.32 | M1s3ry | There is a team that fixes them. It may be the priority in which they are fixed that you mean. |
| 14:17.35 | ManxPower | Most bugs get fixed eventually |
| 14:17.54 | file | florz: bugs aren't just bugs... |
| 14:18.10 | florz | file: how ya mean? |
| 14:18.16 | file | the issue tracker also includes additional features |
| 14:18.32 | file | plus not just Asterisk issues. |
| 14:18.42 | florz | well, I'm not talking about the issue tracker at all, just about my impression from the source |
| 14:19.33 | ManxPower | Asterisk does lack some pretty standard features like dialtone detection, but that only really is an issue for FXO signaled ports. |
| 14:19.40 | file | patches always welcome. |
| 14:20.36 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:21.16 | *** join/#asterisk M1s3ry (n=M1s3ry@216.207.245.1) |
| 14:21.59 | ManxPower | file: many of these features are "needed" only be people that are not qualified to write a patch. |
| 14:22.06 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 14:22.30 | *** join/#asterisk Acxty (n=FAclon@201.220.132.138) |
| 14:22.36 | ZPertee | can someone tell me what is wrong with this http://pastebin.ca/979637 |
| 14:22.40 | ManxPower | i.e. "winmodem as fxo", "dialtone detection", "working call progress", etc. |
| 14:23.01 | ZPertee | FYI: It is voicemail.conf |
| 14:23.17 | ManxPower | ZPertee: no you tell US what is happening that is not supposed to be happening. |
| 14:23.34 | [TK]D-Fender | ZPertee: Why don't you tell us whats NOT WORKING so we don't have to waste our time playing the guessing game? |
| 14:23.52 | coppice | I thik anyone using analogue lines needs dialtone detection |
| 14:23.56 | mocker | ManxPower won that one. |
| 14:24.05 | ManxPower | coppice: nobody should be using analog 8-) |
| 14:24.08 | [TK]D-Fender | mocker: Mine was longer :) |
| 14:24.15 | ManxPower | coppice: but I agree with you if they must use analog. |
| 14:24.21 | Acxty | Hi guys, I want to the install asterisk on a single computer. I had been searching but I only find the package to install it on any linux distribution. My question is this one, do you recommend a single computer only for asterisk or can it be handle by another system, running several other applications?Web,mail, ftp server etc.. The problem will be a memory leak of something like that |
| 14:24.42 | ZPertee | ManxPower and [TK]D-Fender: [Apr 10 10:23:56] WARNING[4378]: app_voicemail.c:7843 load_config: Failed to load configuration file. |
| 14:24.56 | [TK]D-Fender | Acxty: Depends on your overall system load including what * will place. |
| 14:25.10 | [TK]D-Fender | ZPertee: check your permissions, etc. |
| 14:25.27 | ManxPower | ZPertee: what is the file size reported by "ls -l /etc/asterisk/voicemail.conf" |
| 14:25.30 | Acxty | [TK]D-Fender, how much memory and space does asterisk need? |
| 14:25.58 | ManxPower | Acxty: if you don't use voicemail, asterisk requires about 5MB of disk space and 128MB of RAM. |
| 14:26.07 | Acxty | what I will configure is an ivr + db application |
| 14:26.26 | [TK]D-Fender | Acxty: I run *, apache, Mysql, Gnome, ftp, and a ton of other shit on my home XP2000+ w/ 256 meg ram. I have nearly non-existant call volume, but its all G.711 anyways |
| 14:26.46 | ManxPower | Acxty: VoIP requires a fast CPU, but does not use much disk space or memory |
| 14:27.10 | *** join/#asterisk blaylock (n=blaylock@snap.helixsystems.com) |
| 14:27.14 | [TK]D-Fender | ManxPower: um, transcoding would, but not "VoIP" as a general term. |
| 14:27.33 | ManxPower | [TK]D-Fender: I don't think he knows the difference. |
| 14:27.44 | Acxty | ManxPower, I have a 3.0 Dual Core with 2 GB of Ram. Don't know if I install it on that computer or use a similar computer to make a single install of it |
| 14:27.48 | [TK]D-Fender | ManxPower: But you should state it properly as not to mislead the rest... |
| 14:27.54 | ManxPower | ZPertee: are you going to ask questions and then go away away for 30 mins again? |
| 14:28.16 | ManxPower | [TK]D-Fender: I sit corrected. |
| 14:28.20 | ZPertee | [TK]D-Fender and ManxPower: Man if only I could spell, or see, or at least know how to spell VOICEMAIL NOT VOICMAIL |
| 14:28.22 | [TK]D-Fender | Acxty: What kind of load are you looking to place on it? How many calls? Coming in how? |
| 14:28.33 | ZPertee | ManxPower, why do you hate me |
| 14:28.34 | [TK]D-Fender | ZPertee: load chan_clue.so |
| 14:28.38 | ManxPower | ZPertee: THAT is why I asked you for the "ls" |
| 14:29.02 | ManxPower | ZPertee: Don't take it personally -- I hate anyone that wants free support and then does not stick around to get the support. |
| 14:29.18 | ZPertee | ManxPower, understandable |
| 14:29.24 | ZPertee | ManxPower, and sorry |
| 14:29.26 | [TK]D-Fender | ZPertee: Don't ask my why your car is broken and only show me the glove compartment which is working fine. |
| 14:29.33 | Acxty | [TK]D-Fender, will be like 2 calls per minute |
| 14:29.41 | ManxPower | And make no mistake -- almost every single person here is a volunteer with NO obligation to help anyone. |
| 14:29.55 | [TK]D-Fender | Acxty: how many simultaneous, coming in how? |
| 14:30.04 | Acxty | and each call will ask for password, and menu options 1,2.3 for do something? |
| 14:30.05 | florz | file: Well, actually they are not. Without this disclaimer stuff, I might even feel inclined to submit patches at times. |
| 14:30.25 | file | that is your option. |
| 14:30.31 | ZPertee | [TK]D-Fender, yeah I know. the problem is that I'm never sure how much info or what info is needed to solve the problem |
| 14:30.49 | file | but the license agreement lets Digium sell ABE, which provides some income which can be used to pay people like me to work on issues and core API development |
| 14:31.06 | [TK]D-Fender | ZPertee: if it says it can't load the file, forget the contents, the FILE is bad. |
| 14:31.22 | ManxPower | I personally have NO problem with either of Digium's two disclaimers |
| 14:31.46 | file | ManxPower: we have a new one, lawyers like it more - and it can be signed electronically |
| 14:32.01 | Acxty | [TK]D-Fender, don't know that exact amout, but there will be like 10, 12 |
| 14:32.07 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 14:32.24 | florz | file: Yeah - however, given the really bad state of the code base, I doubt whether the project wouldn't be in a much better shape with less participation by digium but more code from others instead. |
| 14:32.57 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 14:33.16 | file | florz: perhaps you might be interested in Callweaver then |
| 14:34.06 | [TK]D-Fender | Acxty: coming in how? |
| 14:35.00 | florz | file: I kindof am, but they seem to be replacing broken stuff with stuff that's broken even worse, from what I've seen, even though that wasn't that much, really. |
| 14:35.21 | Acxty | don't get what you are trying to ask |
| 14:36.39 | [TK]D-Fender | Acxty: Exactly what tech/codec/etc are the calls coming in under? |
| 14:37.00 | file | florz: interesting. |
| 14:37.47 | thesyko | hi guys, |
| 14:37.50 | thesyko | wanna ask |
| 14:37.52 | file | hasn't kept up with Callweaver much |
| 14:38.06 | thesyko | i'm intrested to build an asterisk for my home use |
| 14:38.10 | Acxty | I don't have that answer right now |
| 14:38.19 | thesyko | do you think this peice of hardware is suitable ? |
| 14:38.20 | thesyko | http://www.mini-itx.com/store/?c=2#j7f2 |
| 14:39.11 | M1s3ry | don't think there's been any testing on that. Wanna test it? :D |
| 14:39.31 | thesyko | or is there any recomended hardware type? |
| 14:41.29 | mocker | Has anyone played w/ FreeSWITCH? |
| 14:41.54 | LemensTS | Its actually GMT-5. But I figured out the problem. This php timeclock program im using, has 2 options. One to use browsers time when it displays the epoch, one to use servers time. I turned them both off, then it displayed correctly on the asterisk timestamps, but not when you did it from the php portal. I just subtracted 18000seconds from the timestamp when it posts the form...looks like it all works thanks. |
| 14:42.02 | M1s3ry | thesyko, try this... http://www.voip-info.org/wiki/view/Asterisk+hardware+recommendations Although there is alot of hardware that you could use |
| 14:43.44 | LemensTS | btw people can call in from anywhere and clock on, or they can clock on from the php page. Can review reports, total hours worked etc. I used an open source php time clock, and just modified it to work with asterisk...ill post a zip file up when im done |
| 14:43.44 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 14:46.24 | Acxty | [TK]D-Fender, don't know if this is what you are asking. I have been searching between SIP and IP telephone |
| 14:48.28 | [TK]D-Fender | Acxty: SIP phones are IP phones. SIP is a protocol. Either way for the amount of concurrency you're mentioning I see no problem for the most-part. |
| 14:48.37 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:49.20 | Acxty | [TK]D-Fender, hehe will read a little more to make the correct answers. I am reading Asterisk: The Future of Telephony |
| 14:49.46 | *** join/#asterisk sPOiDar (n=spoidar@59.167.192.109) |
| 14:50.14 | thesyko | hmm k |
| 14:50.47 | thesyko | i'm looking for generally itx / mini-itx hardware |
| 14:51.03 | thesyko | lowpowered embeded type hardware |
| 14:51.36 | coppice | the itx stuff looks nice until you see the price |
| 14:51.37 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
| 14:51.51 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 14:52.23 | [TK]D-Fender | thesyko: What kind of load/purpose? |
| 14:56.07 | *** join/#asterisk Telemac (n=telemac@213.223.113.74) |
| 14:56.09 | Telemac | Hello |
| 14:56.17 | *** join/#asterisk UnixDog (n=UnixDog@ppp-69-238-167-52.dsl.irvnca.pacbell.net) |
| 14:58.04 | sPOiDar | anyone know mISDN inside-out, particularly hfcmulti (b410p)? I can't seem to make it not broken, been at it for about 7hr non-stop |
| 14:58.28 | Telemac | I've a b410 card and I'd like to know what's the module for proper zaptel channel ? |
| 14:59.00 | sPOiDar | you don't get proper zap channels, you either need chan_misdn or chan_capi |
| 14:59.26 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 14:59.32 | thesyko | for home use |
| 15:00.10 | thesyko | maybe 4 ~ 6 extensions |
| 15:00.20 | Telemac | sPOiDar: I have chan_misdn but docs on web say that I need to configure zapata. I do so and when I start asterisk I get warning about missing zap channel. |
| 15:01.22 | sPOiDar | Telemac: nope, no zap configuration with misdn, you will need mISDN and mISDNuser for it though |
| 15:01.59 | Uatec | sPOiDar, i struggled with it for month |
| 15:02.02 | Uatec | eventually got it working |
| 15:02.17 | sPOiDar | Telemac: it's painful though - I'm hoping someone can offer some advice as to why my (previously working) mISDN setup now won't make or receive calls |
| 15:02.25 | Uatec | but now we're fixing it by puttin a sangoma a500 in (i got given one by the very nice sangoma lady i met yesterday) |
| 15:02.51 | Telemac | sPOiDar: Do you have a good doc about it ? I'm struggling to have some complete and good doc about that setup |
| 15:03.16 | [TK]D-Fender | thesyko: Perhaps http://www.soekris.com/net5501.htm |
| 15:03.24 | *** join/#asterisk nephfl (n=none@wsip-70-168-186-225.ga.at.cox.net) |
| 15:04.02 | sPOiDar | Telemac: There's not much to it, problem is the drivers/channel don't output anything in the way of useful debug data that I can see... check out http://misdn.org |
| 15:04.04 | [TK]D-Fender | I'm considering getting on to build a router with. |
| 15:05.33 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 15:05.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:05.33 | Telemac | sPOiDar: Thanx |
| 15:05.43 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 15:05.44 | nephfl | hello, im having trouble with this whole voip thing... maybe you guys can help... i can make local calls, but the system appears to connect toll free and long distance calls fine, but i get no audio...anyone know why that might be? I'm using VTWhite and polycom 320s |
| 15:05.59 | spjuden | I'm noticing quite a few lines in my log of the "callerid.c: Caller*ID failed checksum" variety from my fxo lines.. what could be causing this? |
| 15:06.16 | spjuden | nephfl, sounds like a nat problem |
| 15:06.38 | nephfl | why would local calls through vtwhite (same area code) work but not ld |
| 15:06.39 | nephfl | ? |
| 15:06.42 | *** join/#asterisk spoidar (n=spoidar@59.167.192.109) |
| 15:07.06 | spjuden | nephfl, are your local calls going thru the phone and ld through voip? |
| 15:07.24 | nephfl | guess i wasnt clear..i didnt mean local (internal calls) were working... imeant calls to other area codes other than the did i have on the system |
| 15:08.03 | nephfl | i can call through vtwhite to my cell phone, but not to toll free numbers or to other area codes |
| 15:09.06 | nephfl | they appear to be passed in the same way that the working calls do, i just get no sound |
| 15:09.07 | spjuden | nephfl, what are you using to call outbound, internet or pstn? |
| 15:09.12 | nephfl | internet |
| 15:09.39 | spjuden | and you're sure that the calls are being connected? The called party reports the phone ringing? |
| 15:09.47 | nephfl | no pstn at all on this system |
| 15:09.56 | nephfl | no..i dont know |
| 15:10.28 | nephfl | i just see that it passes the calls in cli the same way as when im calling same area code numbers |
| 15:10.51 | mandd | On astersik -r, [Apr 10 07:08:40] WARNING[4957]: manager.c:3031 init_manager: Unable to bind socket: Address already in use |
| 15:10.57 | spjuden | hmm..post your extensions.conf on pastebin |
| 15:11.05 | mandd | alwats happends |
| 15:11.10 | mandd | what is it? |
| 15:11.33 | Telemac | sPOiDar: I've installed files from http://downloads.digium.com/pub/zaptel/b410p/, Do you know if it is ok ? |
| 15:12.07 | nephfl | i also didnt mention, that i didnt write the extensions.conf ...it's trixbox |
| 15:12.49 | bitzero | Thats a really great way to make friends around here in a hurry (form what I've seen). |
| 15:12.58 | M1s3ry | lol |
| 15:13.06 | spjuden | nephfl, hah...well ive never seen what the conf files it generates looks like, but i guess you could try posting it |
| 15:13.40 | M1s3ry | ha, I wouldn't dream of trying to read extensions.conf from a trixbox generated file |
| 15:13.50 | spjuden | that bad? |
| 15:14.01 | spjuden | like FrontPage makes html bad? |
| 15:14.03 | nephfl | ok, you want all extensions.conf type files...or just the section on my extension? |
| 15:14.05 | M1s3ry | well, and it's just a personal opinion, but it's ugly |
| 15:14.38 | spjuden | nephfl, if you can find just the relevant portions that connect your local and ld calls, that would be best it seems :) |
| 15:14.43 | M1s3ry | exactly... there are mulitple includes to OTHER extension.conf file, that make it even more confusing |
| 15:14.49 | nephfl | yeah, it breaks everything out into a bunch of files and makes a mess... |
| 15:15.30 | M1s3ry | nephfl, I know you may have worked on it for some time, but I HIGHLY recommend to just going over to regular open source asterisk |
| 15:15.41 | spjuden | i tried elastix for about a day and then got rid of it.. too many features where i didn't need them and not enough where i did |
| 15:16.34 | M1s3ry | Trixbox => Macros, Includes, and Argument's Oh MY! |
| 15:17.18 | nephfl | i have no problem setting it up with asterisk, but for regular use where things need to change quickly... it doesnt make sense to always code your own stuff.. |
| 15:18.02 | bitzero | nephfl: what kind of crazy world do you live in? |
| 15:19.01 | M1s3ry | well, if your proficient at it then sure it does, but if you want a GUI, which is fine to, and support for digium cards then I suggest Asterisknow or switchvox |
| 15:19.26 | nephfl | Last i looked at asterisknow, it was a buggy piece of crap |
| 15:19.43 | bitzero | wants an asterisknow VMWare image... |
| 15:19.51 | bitzero | The only one's I've found are broken torrents. |
| 15:20.02 | bitzero | just wants to get eyes on that GUI - my employers might like it. |
| 15:20.26 | UnixDog | the current iso has issues also |
| 15:20.31 | C4away | what about SARK? |
| 15:20.34 | M1s3ry | it's out of beta now, but look at switchvox also if you get a chance. Otherwise open source asterisk would be your best bet |
| 15:20.34 | *** join/#asterisk bougie (n=bougie@APoitiers-256-1-10-5.w90-11.abo.wanadoo.fr) |
| 15:21.14 | spjuden | switchvox is proprietary isnt it? |
| 15:21.39 | M1s3ry | the gui is, but there is a free version that's good for home/ small offices installations. |
| 15:21.50 | *** join/#asterisk sPOiDar (n=spoidar@59.167.192.109) |
| 15:22.11 | M1s3ry | otherwise you can go to the SOHO and SMB versions that do cost |
| 15:22.38 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 15:23.56 | Telemac | What modes/type settings for misdn/b410 so it can be used with asterisk as main PBX ? nt_ptmp ? nt_ptp ? TE ? |
| 15:24.15 | spjuden | hmm.. |
| 15:25.13 | *** join/#asterisk AsteriskRo (n=Asterisk@190.36.185.138) |
| 15:25.45 | sPOiDar | Telemac: depends on the equipment you're connecting it to |
| 15:25.55 | AsteriskRo | hi all... please, can anybody help me on registering my pbx throught a outbound proxy ? |
| 15:26.10 | sPOiDar | Telemac: you should read up on isdn in general to understand that stuff |
| 15:26.29 | Telemac | sPOiDar: I'd like to connect this card directly to the isdn jack |
| 15:27.17 | AsteriskRo | to my SIP Provider but passing throught an outbound proxy ? |
| 15:27.36 | Telemac | sPOiDar: I have read docs but they do not tell the same thing and I'm expert to understand which doc is the right one |
| 15:27.47 | sPOiDar | Telemac: it will need to be TE, but whether it's PTP or PTMP depends on whether you want to bridge it to other devices, and how your ISDN line is configured |
| 15:27.49 | *** join/#asterisk blq (n=Bl@dslb-088-066-226-059.pools.arcor-ip.net) |
| 15:28.03 | *** join/#asterisk blq (n=Bl@dslb-088-066-226-059.pools.arcor-ip.net) |
| 15:28.27 | Telemac | sPOiDar: Digium (http://www.asteriskguru.com/tutorials/digium_b410p_installation_guide.html) tell that TE should be used when direcly connected to the ISDN line ... |
| 15:28.49 | sPOiDar | Telemac: basically ask your telco, but it's probably PTP, so TE_PTP |
| 15:29.49 | Telemac | sPOiDar: Ok, thanx. I will try with te_ptp |
| 15:31.42 | *** part/#asterisk ]technophreak[ (n=]technop@modemcable048.23-81-70.mc.videotron.ca) |
| 15:33.13 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 15:33.33 | bsdwarrior | im trying to change the musicon hold and it simply wont play the new music. I have done a reload |
| 15:33.38 | ZPertee | is there any app that will cause asterisk to play ringing for x seconds |
| 15:33.56 | UnixDog | GIve me BSD or give me Death |
| 15:34.07 | blaylock | Death it is |
| 15:34.16 | blaylock | unless you're on Mac |
| 15:34.19 | blaylock | :-D |
| 15:34.26 | tzafrir | hands UnixDog Blue Screen of Death |
| 15:34.38 | tzafrir | how's that for both? |
| 15:34.42 | UnixDog | mac = wannabe bsd user |
| 15:34.49 | blaylock | UnixDog, haha |
| 15:34.58 | ZPertee | UnixDog, now that's funny |
| 15:35.03 | coppice | either will fit a tight budget |
| 15:35.07 | blaylock | no they got BSD to work right |
| 15:35.12 | loompek | anyone of you guys played with video? |
| 15:35.17 | flujan | hello guys, does sip phones works with AMD? |
| 15:35.25 | blaylock | flujan, ? |
| 15:35.33 | UnixDog | AMD CPU yes |
| 15:35.41 | bsdwarrior | http://pastebin.com/db712a9d |
| 15:35.44 | blaylock | yeah i was gonna say |
| 15:35.55 | bsdwarrior | the agentmusic defined is playing the old music. |
| 15:36.27 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.140) |
| 15:37.25 | flujan | blaylock: http://pastebin.com/d6204ae40 |
| 15:37.34 | flujan | I am trying to use amd, without success.. |
| 15:38.06 | *** join/#asterisk b11d` (n=no@234-200-29-134.hcc.mnscu.edu) |
| 15:38.14 | Telemac | sPOiDar: I think I've bad mISDN as I get module error (kernel: Oops: 0000). Do you get mISDN, mISDNuser from digium or beronet ? Which version of asterisk do you use ? (I'm trying latest, asterisk 1.6.x) |
| 15:38.33 | blaylock | flujan, so what are you trying to do and what value does HUMAN have in it? |
| 15:38.44 | b11d` | hey all |
| 15:39.19 | sPOiDar | Telemac: I don't know how stable mISDN is with 1.6, in fact I thought they were including native ISDN in it? |
| 15:40.00 | ZPertee | blaylock, were you able to get asterisk on a mac? |
| 15:40.12 | blaylock | ZPertee, never tried |
| 15:40.18 | Telemac | sPOiDar: Yes, I don't know if it could be ok in place of beronet stuff. |
| 15:40.23 | blaylock | easier on linux anyway |
| 15:40.35 | AsteriskRo | please, how do i configure an outbound proxy for registering a PBX ? |
| 15:41.04 | ZPertee | blaylock, ok yeah I never did either but I carry my mac around everywhere so it would be sweet to have Asterisk on it. guess I could just run linux virtually and save myself some headaches |
| 15:41.12 | blaylock | AsteriskRo, IAX or SIP trunk provider shoul have that info for you |
| 15:41.17 | loompek | is there a h263 loopback test for asterisk? |
| 15:41.18 | sPOiDar | Telemac: I finally just got mine working using beronet on 1.2 |
| 15:41.40 | blaylock | ZPertee, get vmware fusion and install a debian box, compile asterisk |
| 15:41.47 | sPOiDar | Telemac: I seriously tried about 8 other versions and combinations, this one finally worked... |
| 15:41.57 | ZPertee | blaylock, is vmware fusion worth the price? |
| 15:42.03 | AsteriskRo | blaylock: thanks for answering... |
| 15:42.38 | Telemac | sPOiDar: I was also try to find the good combination so I'm very interessted in known which one you use. |
| 15:42.54 | AsteriskRo | blaylock: i'm using my own sip trunk , but i don't know how to configure it to register to my outbound proxy |
| 15:43.05 | blaylock | ZPertee, yeah man...its only like $80 |
| 15:43.08 | flujan | blaylock: http://pastebin.com/d5ed63936 |
| 15:43.16 | flujan | blaylock: looks like AMD is not working |
| 15:43.17 | *** join/#asterisk ManxPower (n=manxpowe@213.sub-75-203-228.myvzw.com) |
| 15:43.31 | sPOiDar | Telemac: I do a lot of Asterisk work, but usually with E1 - I tend to avoid BRI like the plague because it's always such a painful experience |
| 15:43.47 | Telemac | sPOiDar: I think so |
| 15:43.53 | ZPertee | blaylock, I know I know. I need to quit being the broke college kid that I am and fork out the dough |
| 15:44.14 | AsteriskRo | blaylock: in any softphone or grandstream hardphone i configure it using the parameters for that, but i can't find them in the asterisk configuration |
| 15:44.22 | blaylock | AsteriskRo, well an outbound proxy is normally used to send outgoing calls to so Im not sure I understand what you are trying to do |
| 15:44.31 | Telemac | sPOiDar: asterisk 1.2 with mISDN and mISDNUser from beronet if I understand ? |
| 15:44.49 | blaylock | ZPertee, there is also VirtualBox but I can never get networking to work right |
| 15:44.50 | [TK]D-Fender | flujan: I'd suggest NoOp-ing your variable first. Curious that I don't se an evaluation return in : GotoIf("SIP/40005-b721fb00", "?15:16") in new stack |
| 15:45.22 | bsdwarrior | anyone know ? |
| 15:45.29 | AsteriskRo | blaylock: I want to register my pbx on the outbound proxy to be able to receive incoming calls from other SIP users |
| 15:45.32 | blaylock | AsteriskRo, you have to configure a SIP user in sip.conf |
| 15:45.38 | sPOiDar | Telemac: The machine I've been working on was an old 1.2 install anyway, so I just grabbed the last 1.2 releases of libpri, zaptel, asterisk and then used the install-misdn-mqueue.tar.gz package from beronet to install mISDN, mISDNuser and chan_misdn |
| 15:46.02 | AsteriskRo | blaylock: that's what i'm trying but the pbx is not registering |
| 15:46.14 | blaylock | AsteriskRo, im not sure 'register' is the proper word are you talking about a port forward maybe? |
| 15:46.42 | ManxPower | blaylock: how far have you read The Book? |
| 15:46.52 | flujan | [TK]D-Fender: hello TK |
| 15:46.54 | blaylock | ManxPower, ? |
| 15:46.56 | AsteriskRo | blaylock: i want to receive calls from other SIP users of the SIP service |
| 15:46.57 | flujan | [TK]D-Fender: http://pastebin.com/d347b54d7 |
| 15:47.05 | ManxPower | blaylock: the book everyone should read before using Asterisk. |
| 15:47.08 | ManxPower | ~book |
| 15:47.09 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 15:47.10 | flujan | it looks like amd is not setting the amdstatus variable |
| 15:47.33 | blaylock | ManxPower, dude I work with asterisk every day so I dont know what you are talking bout |
| 15:47.45 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:48.44 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 15:48.46 | flujan | [TK]D-Fender: looks like that AMD is not executing... |
| 15:48.48 | Telemac | sPOiDar: Thanx, I will try that and then see if I can upgrade asterisk at least to 1.4 (for some MGCP issue) |
| 15:48.53 | blaylock | AsteriskRo, if you are trying to register a SIP phone outside of your network and you have your asterisk box behind a firewall, then you need to set up NAT translations |
| 15:49.16 | ZPertee | blaylock, ok thanks. I'll give VirtualBox a try anyhow. I worked with creating virtual machines pretty much all last summer so I'll see if I can't get VirtualBox to work. |
| 15:49.19 | sPOiDar | Telemac: np, best of luck - BRI is the sux ;) |
| 15:49.21 | [TK]D-Fender | flujan: -- AMD: initialSilence [2500] greeting [1500] afterGreetingSilence [800] totalAnalysisTime [5000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256] |
| 15:49.22 | UnixDog | mgcp --- grr cisco strikes again |
| 15:49.34 | [TK]D-Fender | flujan: No, it clearly is, its jsut not doing what you expect.. |
| 15:49.39 | AsteriskRo | <PROTECTED> |
| 15:49.44 | Qwell | since when is MGCP a Cisco tech? |
| 15:49.56 | flujan | [TK]D-Fender: yeap, I know. But it is not setting the AMDSTATUS VARIABLE. |
| 15:50.00 | flujan | ops |
| 15:50.01 | flujan | variable |
| 15:50.10 | Qwell | since they helped write the RFC, apparently |
| 15:51.05 | blaylock | AsteriskRo, yes you probably need to set up a NAT traversal then to make it work |
| 15:51.16 | davevg-btwtech | flujan, what version of asterisk? |
| 15:51.26 | flujan | davevg-btwtech: 1.4.19 |
| 15:51.27 | blaylock | AsteriskRo, btw SIP is UDP 5060 if you didnt already know |
| 15:51.32 | AsteriskRo | <PROTECTED> |
| 15:52.27 | blaylock | AsteriskRo, http://www.voip-info.org/wiki/view/SIP+outbound+proxy |
| 15:52.36 | AsteriskRo | <PROTECTED> |
| 15:52.45 | blaylock | AsteriskRo, I think you might be using it wrong or misunderstanding something |
| 15:52.48 | [TK]D-Fender | AsteriskRo: READ ---> |
| 15:52.50 | [TK]D-Fender | ~sipnat |
| 15:52.51 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:52.52 | [TK]D-Fender | ^^^^^^^^^^^ |
| 15:52.52 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.140) |
| 15:53.01 | AsteriskRo | <PROTECTED> |
| 15:53.18 | flujan | davevg-btwtech: 1.4.18 same problem. |
| 15:53.24 | flujan | [TK]D-Fender: any ideas? |
| 15:53.30 | kamanashisroy | hi, I am using asterisk as iax client ;) |
| 15:53.39 | blaylock | AsteriskRo, i never use those variables so i dunno |
| 15:53.47 | davevg-btwtech | flujan, do you happen to have silence suppression turned on the phones you are testing with? |
| 15:53.51 | kamanashisroy | but, how to send dtmf using this client ? |
| 15:53.51 | [TK]D-Fender | flujan: Nope, its 3rd party, I've never used it, its executing and not doing what it says it should. I smell "bug" |
| 15:54.15 | AsteriskRo | blaylock: and what variables should i use? |
| 15:54.18 | ManxPower | [TK]D-Fender: um AMD is part of Asterisk |
| 15:54.37 | [TK]D-Fender | ManxPower: It is? I sit mistaken. The rest stands for now :) |
| 15:55.04 | davevg-btwtech | 1.4+, 1.2 was 3rd party and had different variable names, it was why i asked |
| 15:55.14 | blaylock | AsteriskRo, i dont use any. I just set up a sip account and tell my phone to use that account either by boot files or direct configuration of the phone via its web interface |
| 15:55.19 | blaylock | AsteriskRo, thats all you need |
| 15:55.25 | *** join/#asterisk sacitec (n=tobi@201.144.211.83) |
| 15:55.40 | ManxPower | damn it, I was SURE it was part of at least 1.6, but I can't find it there. |
| 15:56.02 | blaylock | AsteriskRo, and if you have a dialplan in place in the context the sip account is in plus an outgoing TRUNK (IAX or SIP) then you should be good |
| 15:56.37 | ManxPower | Yes, AMD is at least part of 1.6 |
| 15:56.40 | blaylock | AsteriskRo, also have to tell the phone what IP address your asterisk server has |
| 15:57.02 | AsteriskRo | blaylock: ok, i setup the accound info in sip.conf and have the context in the dialplan, but can receive any call |
| 15:57.11 | ManxPower | blaylock: we don't normally use the word "trunk" here. |
| 15:57.17 | ManxPower | ~trunk |
| 15:57.17 | jbot | somebody said trunk was is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 15:57.43 | AsteriskRo | blaylock: i don't understand your last part: also have to tell the phone what IP address your asterisk server has |
| 15:58.11 | AsteriskRo | blaylock: i'm making test with two phones registered in my pbx |
| 15:58.15 | blaylock | ManxPower, then reprogram Asterisk to use something different. otherwise thats how I refer to it |
| 15:58.19 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-177.static.twtelecom.net) |
| 15:58.34 | JT | blaylock: asterisk makes no mention of sip trunks |
| 15:58.41 | ManxPower | blaylock: Asterisk ALSO does not use the word trunk, except as it applies to IAX2 |
| 15:58.56 | [TK]D-Fender | AsteriskRo: Read the guide that I linked to you. You have clearly not set up the basics for * to work behind NAT |
| 15:58.59 | blaylock | AsteriskRo, the SIP phone needs to know the IP address of the Asterisk server dude. So it can register to it |
| 15:59.02 | ManxPower | JT: I'm pretty sure extensions.conf.sample stupidly uses the word "trunk". |
| 15:59.17 | blaylock | fine ManxPower a SIP PEER excuse me |
| 15:59.56 | AsteriskRo | <PROTECTED> |
| 15:59.57 | blaylock | err i meant JT |
| 16:00.18 | blaylock | AsteriskRo, what does 'sip show peers' give you |
| 16:00.36 | ManxPower | I wish I knew why some marketing terms like "sip trunk" make people think it's a technical term. |
| 16:00.38 | blaylock | AsteriskRo, can you make outgoing calls? |
| 16:00.39 | JT | blaylock: a trunk means specific thing in telephony, sip does not qualify |
| 16:00.48 | *** join/#asterisk dagobart (n=dagobart@ip-62-143-215-156.1411O-CUD12K-01.ish.de) |
| 16:01.04 | eric2 | how do I do a realtime database reload |
| 16:01.11 | ManxPower | calling it a sip trunk makes about as much as calling a Volvo a tree |
| 16:01.15 | blaylock | JT, thats great thanks for clearing that up. Also thanks for giving me input when I didnt ask for it |
| 16:01.17 | AsteriskRo | blaylock: this is my "sip show peers": 6001/6001 192.168.1.106 D 5060 Unmonitored 6000/6000 192.168.1.117 D N 5060 Unmonitored |
| 16:01.41 | ManxPower | AsteriskRo: Go READ the damn BOOK |
| 16:01.45 | JT | blaylock: thanks for being receptive to corrections, and for not being agressive |
| 16:01.48 | blaylock | AsteriskRo, can you dial between the extensions? |
| 16:01.49 | dagobart | can anyone of you point me to a good short intro to VoIP using Asterisk? |
| 16:02.07 | AsteriskRo | blaylock: yes i can dial between extensions |
| 16:02.53 | blaylock | AsteriskRo, then your problem is that you do not have your incoming and outbound connections set up properly |
| 16:03.11 | flujan | kicks flujan in the head |
| 16:03.12 | ManxPower | dagobart: Telecom is complicated, VoIP even more so. You need to understand linux, networking, nat, Asterisk, and telecom -- there is no such thing as a short document that covers what you need. However, some people have written a book about Asterisk -- to bad so many people on this channel are either stupid, illiterate, or just plain lazy and don't read the book. |
| 16:03.13 | blaylock | AsteriskRo, you can do this 3 ways (IAX, SIP, or Zap) |
| 16:03.20 | [TK]D-Fender | dagobart: What kind of information are you loking for? |
| 16:03.24 | flujan | [TK]D-Fender: got it... My mistake, using codec g729 on the phone |
| 16:03.25 | flujan | :D |
| 16:03.38 | [TK]D-Fender | flujan: I was wondering about that... |
| 16:03.40 | Daviey | ~thebook |
| 16:03.40 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 16:03.57 | dagobart | ManxPower: Thanks for telling that to me. I know that I need to understand. Poor boss forbid me to do so. hence I need something short, sadly. |
| 16:04.10 | ManxPower | dagobart: you will fail. |
| 16:04.14 | blaylock | haha |
| 16:04.19 | flujan | [TK]D-Fender: lol |
| 16:04.24 | AsteriskRo | blaylock: ok, i'm trying to configure SIP first, then i want to configure with ZAP, i have a digium card (not installed yet) |
| 16:04.29 | dagobart | ManxPower: I've even got the book here, the one of O'Reilly. |
| 16:04.33 | blaylock | dagobart, and so will your boss |
| 16:05.05 | dagobart | blaylock: yes, I know, but still I'm with them. |
| 16:05.28 | dagobart | ManxPower: I'm sure, I will. Poor thing, it'll be my fault, in their eyes. |
| 16:05.32 | blaylock | AsteriskRo, you will need a SIP account with some provider then |
| 16:05.50 | blaylock | AsteriskRo, which means you need to call them up and get an account, etc etc |
| 16:06.23 | blaylock | dagobart, I hear you man we all have those bosses |
| 16:06.24 | AsteriskRo | blaylock: yes, i already have an account |
| 16:06.52 | blaylock | AsteriskRo, and did they give you registration strings and instructions? |
| 16:06.56 | dagobart | blaylock: Thanks, so how do you deal with them to get things straight? |
| 16:07.09 | outtolunc | dagobart: any short *review* of asterisk (which there are many) should be enough to make your boss realize it will be worth the investment in time for you to read ~thebook |
| 16:07.22 | AsteriskRo | blaylock: yeap, i have them |
| 16:07.24 | ManxPower | dagobart: it sucks to be you. |
| 16:07.54 | ManxPower | dagobart: people either fail or they convince their boss to remove the requirements that will ensure failure. It's quite simple, really. |
| 16:07.58 | blaylock | AsteriskRo, then I think you need to call up your provider Initiate the conversation with a quick WTF and ask them what you need to do |
| 16:08.02 | dagobart | outtolunc: we'll he tells me it were quicker to just try & error, instead of rtfm |
| 16:08.19 | dagobart | ManxPower: thanks |
| 16:08.25 | blaylock | dagobart, RTFM while trying |
| 16:08.38 | dagobart | blaylock:am trying to ;-) |
| 16:09.08 | dagobart | *well |
| 16:09.09 | blaylock | dagobart, so i forgot your question already |
| 16:09.16 | ManxPower | dagobart: I suggest you find another job fast. |
| 16:09.21 | spjuden | Even when reading the book, setting it up properly is going to be an iterative process when new at the game |
| 16:09.21 | *** join/#asterisk CallCtr4Sale (n=hellowor@203.177.237.26) |
| 16:09.25 | CallCtr4Sale | hello everyone |
| 16:09.26 | dagobart | ManxPower: I realized that already |
| 16:09.37 | [TK]D-Fender | dagobart: Here's something quick that might help. |
| 16:09.41 | [TK]D-Fender | ~jerjerguide |
| 16:09.42 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 16:09.43 | ManxPower | blaylock: he wants a complete guide to setting up Asterisk all in 4 pages or less. |
| 16:09.54 | [TK]D-Fender | ManxPower: this'll do it :) |
| 16:09.56 | CallCtr4Sale | has anyone worked with asterisk as a trunk for a toshiba strata |
| 16:09.57 | AsteriskRo | blaylock: it's that difficult? it should be easy.... |
| 16:10.00 | dagobart | spjuden: as far as I understood him, he wants it to be done until mid-next week |
| 16:10.15 | dagobart | Thanks [TK]D-Fender |
| 16:10.27 | blaylock | ManxPower, haha well i can give him 4 words or less. extensions.conf, sip.conf, iax.conf and voicemail.conf |
| 16:10.28 | [TK]D-Fender | dagobart: Does he often ask you to do things you don't know how to do and forbid you to learn how as well? |
| 16:10.32 | blaylock | there you go |
| 16:10.32 | spjuden | dagobart, in that case I hope you get paid overtime :) |
| 16:10.59 | dagobart | [TK]D-Fender: other thing is, I don't have any clue of VoIP, don't know what the client wants to know from em |
| 16:11.25 | blaylock | dagobart, you need to tell your boss forget it then |
| 16:11.29 | dagobart | spjuden: that's a nice dream, yes. Unfortunately, I'm ... well ... paid poorly, even that. |
| 16:11.36 | dagobart | no overtime |
| 16:11.49 | [TK]D-Fender | dagobart: If you are loking to provide services to clients as well you are walking into a war-zone. Good luck, you need LOTS of it,a nd forget about "next week". |
| 16:12.05 | blaylock | exactly |
| 16:12.14 | dagobart | [TK]D-Fender: I meant client software, linphone, kphon etc |
| 16:12.34 | blaylock | dagobart, ask your boss for some boot camp courses or something |
| 16:12.34 | spjuden | lol |
| 16:12.37 | [TK]D-Fender | dagobart: Instll *, read the quick guide, read the book, start testing now. |
| 16:12.39 | ManxPower | dagobart: those are called "softphones" |
| 16:13.01 | dagobart | ManxPower: Oh, read that term before. Thanks. |
| 16:13.14 | dagobart | There are too many new terms at once, just forgot it. |
| 16:13.38 | [TK]D-Fender | dagobart: Go get your hands dirty and come back when you've actually started working with *. |
| 16:13.48 | blaylock | dagobart, what kind of time frame did your boss give you? |
| 16:14.02 | dagobart | [TK]D-Fender: thanks for the invitation. I'll come back then |
| 16:14.10 | ManxPower | dagobart: as you can see using the wrong term causes confusion |
| 16:14.18 | [TK]D-Fender | blaylock: 0-to-hero by mid next week ;) |
| 16:14.29 | blaylock | [TK]D-Fender, bwahahaha |
| 16:14.42 | dagobart | blaylock: no real one. Last week the superior told me "some months", today boss came in told me, they're ready, would not like to wait for me |
| 16:15.05 | dagobart | ManxPower: of yourse, hence I've ben forbidden to learn terms but instead shall try & error |
| 16:15.11 | dagobart | since that were quicker |
| 16:15.35 | blaylock | dagobart, well then they are retarted |
| 16:16.05 | blaylock | dagobart, asterisk will work but you need to make sure they know you need to learn it first |
| 16:16.10 | flujan | guys, there is some application that allows me to jump between the contexts in asterisk? |
| 16:16.27 | spjuden | flujan, like goto? |
| 16:16.30 | blaylock | dagobart, if they cant except that then move on to something commercial with lots of paid tech support |
| 16:16.38 | dagobart | blaylock: well, they laughed at me, when I told them I were interested in Ruby, but they were proud of doing php. ... anyways, that's another story. |
| 16:16.45 | blaylock | dagobart, Avaya VoIP might work for them |
| 16:17.16 | flujan | just like it spjuden |
| 16:17.18 | flujan | thanks |
| 16:17.26 | blaylock | please excuse if I have offended anyone by mentioning Avaya in here :-D |
| 16:17.38 | spjuden | flujan, np |
| 16:17.57 | blaylock | damned H.323 |
| 16:18.07 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:18.17 | ManxPower | dagobart: You realize that Asterisk is not actually a PBX, right? |
| 16:18.39 | blaylock | umm |
| 16:18.42 | dagobart | ManxPower: if I got it correctly, it's some kind of can-everything tool |
| 16:19.02 | ManxPower | dagobart: Asterisk is a TOOLKIT that allows you to CREATE a PBX |
| 16:19.15 | dagobart | ManxPower: oops. missed. |
| 16:19.21 | ManxPower | Much like Perl is a language that lets you CREATE programs |
| 16:19.22 | alrs | ARG BLARG ARG |
| 16:19.29 | dagobart | ManxPower: oic |
| 16:19.30 | M1s3ry | dagobart, or a tool that can assist in watering your plants |
| 16:19.48 | dagobart | yes, that's a good hint. I'll pass that. |
| 16:19.59 | *** part/#asterisk jarod14 (n=jarod14@viatel-i3f-55-186.cnt.nerim.net) |
| 16:20.25 | blaylock | dagobart, now that you have been flamed relentlessly try visiting http://www.voip-info.org |
| 16:20.39 | dagobart | They just want an can do everything, because they want it to put it into web app. To check back ... err, with customers. But outgoing traffic will be by landline |
| 16:20.53 | M1s3ry | blaylock, ahh... there's no flaming here XD |
| 16:21.08 | blaylock | heh |
| 16:21.10 | dagobart | has to leave, everyone's leaving... sorry, I'll check back with you tonight (GMT+2) or tomorrow |
| 16:21.36 | blaylock | we shall hold our breath |
| 16:21.54 | dagobart | blaylock: is okay, I don't agree with their approach neither. But they refuse to listen. |
| 16:22.27 | ManxPower | dagobart: At least they can't fire you (I assume you are in the EU) |
| 16:22.50 | spjuden | ahh..socialism at its best |
| 16:23.24 | dagobart | ManxPower: well, there are exceptions to that rule, and actually, I really don't love this job |
| 16:23.42 | coppice | if people refuse to listen you only need half as much bandwidth |
| 16:23.43 | dagobart | ManxPower: So, only issue is to get an alternative soon |
| 16:24.24 | dagobart | coppice: the other one is then to IRC where I bother #asterisk with my job issues. ;-) |
| 16:24.28 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 16:24.53 | bsdwarrior | Im trying to setup new songs for music on hold. for some reason if I rename the files, they dont play at all. I get dead air |
| 16:25.19 | bsdwarrior | im simply playing them from a directory no mpg123 |
| 16:25.24 | AsteriskRo | blaylock: thanks for your help blaylock... |
| 16:25.28 | dagobart | okay, thanks for all and to everyone. Have a nice day/evening/night & bye. |
| 16:25.33 | AsteriskRo | blaylock bye |
| 16:26.45 | [TK]D-Fender | bsdwarrior: pastebin your CLI output at verbose 10, your musiconhold.conf and a detailed and explicit folder dump via "ls -l". |
| 16:26.47 | [TK]D-Fender | ~pb |
| 16:26.47 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:26.48 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 16:27.15 | bsdwarrior | http://pastebin.com/db712a9d |
| 16:27.25 | spjuden | wow, you guys can be assholes :) |
| 16:27.26 | M1s3ry | bsdwarrior, it sounds like asterisk just doesn't know what to play if you change the name, be sure that if you do change the name, that asterisk knows what soundfile to look for, where to look for it, and that the soundfile has the correct name, and is in the right place |
| 16:28.11 | *** part/#asterisk AsteriskRo (n=Asterisk@190.36.185.138) |
| 16:28.30 | bsdwarrior | M1s3ry I was trying to add new music but it simply plays the old songs. pretty retarted. |
| 16:28.40 | *** join/#asterisk TheSov (n=a@204-16-231-218-risingmedical.pkh.ord.sparkplugbb.net) |
| 16:28.44 | bsdwarrior | Ive moved the directory of old songs did a reload, and same thing |
| 16:29.23 | M1s3ry | make sure the moh directory is /var/lib/asterisk/moh |
| 16:29.42 | bsdwarrior | tkd- fender here |
| 16:29.44 | bsdwarrior | http://pastebin.com/d3bdf8097 |
| 16:29.49 | [TK]D-Fender | bsdwarrior: You have noe "mode=" directive. hence no class at all |
| 16:30.04 | M1s3ry | or that musiconhold.conf is set to look at the directory you placed the songs. |
| 16:30.11 | *** join/#asterisk riddlebox (n=JamesMid@75-128-170-26.static.stls.mo.charter.com) |
| 16:30.29 | M1s3ry | "mode=files" should be right... right tk? |
| 16:30.37 | [TK]D-Fender | bsdwarrior: in fact doesn't look like musiconhold.conf at all. |
| 16:30.43 | bsdwarrior | m1s3ry im using the agentmusic for gnudialer. |
| 16:30.48 | [TK]D-Fender | bsdwarrior: What is all of that? |
| 16:31.10 | bsdwarrior | tkdd-fender cli output and the ls |
| 16:31.23 | [TK]D-Fender | bsdwarrior: Oh, non * original stuff... WGLWAT |
| 16:31.27 | bsdwarrior | hereis moh http://pastebin.com/de23b75a |
| 16:32.26 | CallCtr4Sale | hi everyone how do i do call recording using asterisk with my toshiba Strata CTX670 KSU? any ideas |
| 16:32.32 | [TK]D-Fender | bsdwarrior: Those aren't normal directives for musiconhold.conf period. |
| 16:32.51 | bsdwarrior | tkd-fender I dont even know how its working |
| 16:33.13 | bsdwarrior | the normal hold music works, the agentmusic is goofy |
| 16:34.44 | M1s3ry | ls /var/lib/asterisk/moh, what does it look like? |
| 16:34.56 | [TK]D-Fender | bsdwarrior: You're running non-standard apps/modules for MOH it seems. We can't help you there... |
| 16:35.11 | *** join/#asterisk RoyK (n=roy@gprs-ggsn6-nat.mobil.telenor.no) |
| 16:35.44 | bsdwarrior | whats non-standard ? |
| 16:35.51 | M1s3ry | not what asterisk comes with |
| 16:36.08 | M1s3ry | ^^^^ => non-standard |
| 16:36.22 | bsdwarrior | m1s3ry - I know what the word means, thanks. |
| 16:36.27 | [TK]D-Fender | bsdwarrior: what you showed is not standard musiconhold.conf stuff. |
| 16:36.50 | bsdwarrior | can anyone pastebin a working musiconhold.conf with at least one context |
| 16:37.42 | [TK]D-Fender | bsdwarrior: look at the asterisk samples. |
| 16:37.43 | *** join/#asterisk wonderworld (n=voici@ip-62-143-163-248.1211G-CUD12K-01.ish.de) |
| 16:37.55 | [TK]D-Fender | bsdwarrior: Where the heck did you get those keywords from? |
| 16:37.57 | [TK]D-Fender | bsdwarrior: http://www.google.ca/search?hl=en&q=agentmusic+hold&btnG=Search&meta= |
| 16:38.04 | wonderworld | hey, is there an easy way to convert a asterisk recorded gsm-file to mp3 ? |
| 16:38.05 | [TK]D-Fender | bsdwarrior: complete 0 on Google. |
| 16:38.19 | bsdwarrior | kk |
| 16:38.57 | bsdwarrior | tkd-fender btw I got that callback system working nicely. been for a while. |
| 16:39.41 | M1s3ry | wonderworld, yes, try sox |
| 16:39.55 | wonderworld | i did on debian stable |
| 16:40.05 | M1s3ry | work? |
| 16:40.09 | wonderworld | it saqys: sorry, no mp3 files supported |
| 16:40.39 | M1s3ry | ahhh... hmm. i'm sure there's something out there, but I don't know of it off of the top of my head... |
| 16:40.42 | alrs | wonderworld: the sox in debian won't convert to .mp3 |
| 16:41.04 | wonderworld | hmm, i can't switch servers. is there another way? |
| 16:41.06 | CallCtr4Sale | has anyone integrated Toshiba Strata CTX670 and Asterisk? |
| 16:41.08 | alrs | wonderworld: you might check to see if a no-holds-barred version of sox is available from debian-multimedia.org, or compile it yourself. |
| 16:41.18 | wonderworld | ok thanks |
| 16:42.51 | [TK]D-Fender | CallCtr4Sale: only way would be to have it pipe out the call for recording via T1 or something, or pass the entire call out via T1, and back in to bridge to the target phone (taking up 2 channels) |
| 16:43.03 | bsdwarrior | tkd-fender - previously a retarted donkey configured our system. (wasnt me) |
| 16:43.07 | CallCtr4Sale | thanks TK |
| 16:43.14 | [TK]D-Fender | CallCtr4Sale: This would be PRI signalling clearly so you could target DID's |
| 16:43.24 | [TK]D-Fender | CallCtr4Sale: Sounds hackish at best. |
| 16:43.28 | UnixDog | bsdwarrior: some linux wannabe guru right |
| 16:43.57 | bsdwarrior | unixdog - no. |
| 16:44.02 | CallCtr4Sale | TK, could it be possible that I will setup asterisk as a gateway to catch and forward calls to the strata via a sip trunk? |
| 16:44.20 | [TK]D-Fender | bsdwarrior: One who invents syntax out of thin air and things it means something? |
| 16:44.25 | CallCtr4Sale | in that way i can have asterisk record all the incoming and outgoing calls |
| 16:44.26 | ManxPower | CallCtr4Sale: you mean "sip peer" or "sip connection" There is no such thing as a "sip trunk" |
| 16:44.30 | [TK]D-Fender | bsdwarrior: Clearly a total retard. |
| 16:44.43 | CallCtr4Sale | yeah sip peer |
| 16:44.44 | CallCtr4Sale | sorry |
| 16:44.50 | bsdwarrior | tkd-fender - yeah exactly. no longer here. |
| 16:44.57 | LemensTS | exten => _1231,n,MYSQL(Query resultid ${connid} SELECT \`empfullname\` FROM \`timeclock\`.\`employees\` WHERE \`employid\` = \'${EMPID}\') |
| 16:45.04 | [TK]D-Fender | CallCtr4Sale: Everything depends on where you are expecting * to sit in this equation and how the CTX is capable of talking to it. |
| 16:45.07 | LemensTS | does that not look right? |
| 16:45.20 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 16:45.58 | CallCtr4Sale | what basically asterisk would do is that it will be configured with the DID's and just forwards each call received to the strata |
| 16:46.18 | [TK]D-Fender | LemensTS: Wy all of the --'-- s? |
| 16:46.20 | [TK]D-Fender | LemensTS: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MYSQL |
| 16:47.51 | LemensTS | TKD-Fender: Is thier any kind of mysql debugging from the cli in 1.6? THat sure would be helpful |
| 16:48.20 | *** part/#asterisk gmaruzz (n=maruzz@213-140-6-122.ip.fastwebnet.it) |
| 16:48.21 | [TK]D-Fender | LemensTS: 1.6? You should be asking in the -dev channel |
| 16:48.37 | [TK]D-Fender | LemensTS: We can't comment on stuff that could be really screwed up. |
| 16:48.47 | *** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 16:52.20 | jeev | ok guys, i try to call a number, try to put in the extension during the message.. and it doesn't go through.. is there anything i could put in the config |
| 16:55.14 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 16:55.55 | JayTee52 | anyone using jbenable=yes with Asterisk 1.4? |
| 16:56.05 | JayTee52 | in SIP |
| 16:56.11 | jeev | JayTee52, help me out during my bug testing ;/ |
| 16:56.21 | JayTee52 | jeev, not today |
| 16:56.30 | jeev | :O |
| 16:56.30 | JayTee52 | too many things on my desk right now |
| 16:56.32 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 16:56.37 | JayTee52 | maybe later |
| 16:56.40 | jeev | damn |
| 16:56.44 | jeev | can i just tell you what the problem is |
| 16:56.48 | JayTee52 | sure |
| 16:57.00 | jeev | i'm calling the court.. |
| 16:57.05 | jeev | we regulary press, 4,3,2 |
| 16:57.12 | jeev | on the nortel phones, it bypasses the message |
| 16:57.21 | jeev | when i dial it in the phone, it just keeps going.. it's not like going through |
| 16:58.09 | JayTee52 | bypasses what message? |
| 16:58.20 | jeev | the automated attendant/IVR of the court |
| 16:58.29 | jeev | on the nortel phone, they'd press 4.. then 3 then 2 |
| 16:58.45 | jeev | but now with this.. it doesn't go through, you gotta wait till message is done |
| 16:58.46 | JayTee52 | so you're passing DTMF tones from the Nortel switch AFTER you connect? |
| 16:59.03 | *** join/#asterisk shuri (n=shuri@64.235.209.226) |
| 16:59.04 | jeev | no, bypass nortel, i'm using softphone |
| 17:00.17 | Jumpie | hi JayTee52 :D |
| 17:00.27 | JayTee52 | jeev, what I'm asking is that when you "regularly pres 4,3,2" from a Nortel phone are you already connected to the court? |
| 17:01.26 | jeev | yea |
| 17:02.32 | JayTee52 | so when you connect to the court's IVR with a softphone through * it plays the whole message and won't bypass. If you let it play the whole message and then dial 4,3,2 what happens? |
| 17:02.59 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584618.dsl.bell.ca) |
| 17:03.16 | shuri | hi there, is there a way to blacklist an entire area code for exemple database put blacklist 418XXXXXXX 1 |
| 17:03.17 | shuri | HI there, i need to find a way to blacklist a wildcard range callerid |
| 17:04.09 | shuri | humm sorry for double post |
| 17:04.37 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:04.57 | JayTee52 | hi Jumpie |
| 17:06.28 | UnixDog | why would you block a whole area code |
| 17:06.32 | jeev | so sorry jayrod422. |
| 17:06.41 | jeev | i THINK it works, i'll ask the employee |
| 17:06.48 | jeev | it works when you wait the whole message.. JayTee52 |
| 17:07.25 | JayTee52 | jeev, what softphone are you using? |
| 17:07.29 | Jumpie | when looking at motherboard linux distro compatability charts, i never see centos listed |
| 17:07.41 | Jumpie | what distro can i use to most closely parallel? a redhat release, fedora, what? |
| 17:07.42 | JayTee52 | Jumpie, just look for RHEL |
| 17:07.48 | Jumpie | thanks |
| 17:07.59 | jeev | x-lite |
| 17:08.01 | shuri | Unixdog we got a Free tool number and a local number we want to check if call is local or not |
| 17:08.08 | shuri | if local play msg to call thelocal number |
| 17:08.08 | JayTee52 | CentOS 5 is RHEL 5 with the RedHat branding removed. |
| 17:08.32 | JayTee52 | jeev, turn off Silence suppression and try again |
| 17:09.21 | tzafrir | JayTee52, hey, RedHat don't even call it RHEL anymore :-) |
| 17:10.21 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 17:11.01 | bsdwarrior | tkd-fender I got rid of the "urban-engineering" in the config did it the right way and it works now. |
| 17:11.01 | bsdwarrior | thanks |
| 17:11.07 | jeev | it doesn't work.. at ALL |
| 17:11.12 | jeev | even after the message.. |
| 17:12.20 | JayTee52 | jeev, after turning it off it doesn't work at all? |
| 17:12.56 | JayTee52 | X-Lite calls it Acoustic Echo Cancellation in the devices config under Options. |
| 17:14.22 | JayTee52 | tzafrir, what does RedHat call it now? |
| 17:14.43 | tzafrir | They don't like the shorthand |
| 17:14.50 | shuri | UnixDog any idea? |
| 17:15.02 | JayTee52 | oh, I guess I should sell my Red fedora on Ebay then. |
| 17:15.12 | Jumpie | eheh im about to attempt my first extension :D |
| 17:15.37 | spjuden | tzafrir, does the tco on the astribank use a standard connector that i might find a prewired connection block for? |
| 17:15.38 | tzafrir | shuri, what technology do you get the calls? SIP? Zaptel? |
| 17:15.40 | jeev | i'm gonna try turning it on right now |
| 17:15.44 | JayTee52 | but I'm reluctant. When I first ordered it I had to wait, it was on backorder for my size because I've got such a fat head. |
| 17:15.55 | shuri | tzafrir sip |
| 17:16.33 | JayTee52 | jeev, also look in your sip.conf and see if you have a dtmfmode setting for the phone and whether its rfc2833, inband or info and let me know. |
| 17:17.29 | jeev | JayTee52, i love you. |
| 17:17.41 | jeev | it's rfc2833, i altered acoustic on x-lite and it works |
| 17:18.00 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25) |
| 17:18.19 | JayTee52 | Attention, for all you trixbox and other voip software people that use "sip trunks" I have just started production of the new "SIP valise". It's lighter than a trunk and comes with optional shoulder strap and is available in a wide range of colors. |
| 17:18.33 | *** join/#asterisk marc7 (n=marc@64.46.14.64.novuscom.net) |
| 17:18.46 | JayTee52 | jeev, I figured it was "silence suppression" or AEC as X-Lite calls it. |
| 17:19.37 | tzafrir | spjuden, it's in the manual. I don't remember off-hand if it's a standard wiring |
| 17:20.12 | jeev | gimme 5! |
| 17:20.28 | UnixDog | gives jeev gizmo 5 |
| 17:20.36 | jeev | no way! not you. |
| 17:21.41 | spjuden | tzafrir, the only mention i can find in the manual is that the "option includes a special connector on the rear panel".. im taking that as a no :) |
| 17:22.08 | tzafrir | looking... |
| 17:22.34 | JayTee52 | ^5's jeev |
| 17:23.03 | JayTee52 | so I'll ask again, anyone using jbenable in sip.conf with 1.4.18? |
| 17:23.03 | jeev | :> |
| 17:23.14 | jeev | not sure what jbenable is :> |
| 17:23.20 | JayTee52 | jbenable=yes |
| 17:24.20 | JayTee52 | I know it's a Grandstream issue, I just know it. I can't detect any jitter on the two Polycoms I have, just the Grungepuddles GPX-2000s. |
| 17:24.34 | JayTee52 | jeev, jitterbuffer |
| 17:26.35 | jeev | i'm having live one way audio |
| 17:26.36 | jeev | fuck |
| 17:26.44 | jeev | i hav an employee using it and she can't speak.. they wont hear it |
| 17:26.48 | JayTee52 | potty mouth! |
| 17:26.48 | jeev | the other person can't hear it |
| 17:26.54 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 17:27.33 | JayTee52 | the same softphone dialing the court IVR? |
| 17:28.14 | jeev | ya |
| 17:28.19 | jeev | i was having this issue earlier.. |
| 17:28.23 | jeev | but it started to work |
| 17:28.26 | Jumpie | woohoo!!! |
| 17:28.29 | Jumpie | made my first calls! |
| 17:28.33 | Jumpie | NO GUI |
| 17:28.45 | JayTee52 | congratulations |
| 17:28.51 | Jumpie | this is a crowning moment |
| 17:29.15 | JayTee52 | I remember my first call......it was to Sarah.......she said no. :-( |
| 17:29.19 | Jumpie | lo |
| 17:29.24 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.134) |
| 17:29.42 | jeev | [Apr 10 10:28:16] WARNING[12769]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 17:29.46 | jeev | i dont know what that has to do with |
| 17:29.51 | jeev | i called her extensions and she can hear me but i can't hear her. |
| 17:30.01 | Jumpie | nat issues? |
| 17:30.15 | Jumpie | JayTee52, i keep gettin these received sip subscribe notice, like...spammage |
| 17:30.16 | Jumpie | lol |
| 17:30.29 | jeev | i'm calling her locally ;/ |
| 17:30.40 | [TK]D-Fender | jeev: Firewall issues then |
| 17:30.58 | [TK]D-Fender | jeev: Betting you've got the inbound port blocked |
| 17:31.08 | JayTee52 | jeev, X-Lite doesn't support IAX2, it only supports SIP |
| 17:31.25 | jeev | local firewall? let me check |
| 17:31.48 | jeev | let me wqlk to her temrinal |
| 17:31.50 | jeev | terminal |
| 17:32.12 | JayTee52 | terminal? is he in an airport or sumthin? |
| 17:33.17 | [TK]D-Fender | jeev: Server & local |
| 17:33.58 | jeev | :) |
| 17:34.04 | jeev | server is local.. it's 192.168.0.72 |
| 17:34.16 | jeev | it will be at the datacenter once i've perfected it i guess. |
| 17:34.21 | jeev | it doesn't have to be for now though. |
| 17:34.37 | jeev | i have nat checked for the user |
| 17:35.02 | [TK]D-Fender | jeev: "checked"? |
| 17:35.09 | jeev | ahh, i'm using gui :D |
| 17:35.13 | jeev | but yea, nat=yes |
| 17:35.15 | jeev | for the user |
| 17:35.31 | [TK]D-Fender | jeev: for a local user that doesn't mean much, and doesn't say anything about firewalling. |
| 17:35.49 | jeev | yea.. |
| 17:36.24 | jeev | what do you think it could be |
| 17:36.53 | [TK]D-Fender | jeev: I jsut told you. So until you show us something of value don't expect much. |
| 17:37.02 | jeev | what would you like to see? |
| 17:37.23 | [TK]D-Fender | jeev: dump your firewall setup on both sides, pastebin your sip.conf and phone entries and shove them all in a pastebin. |
| 17:37.26 | [TK]D-Fender | ~pb |
| 17:37.26 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:37.50 | jeev | ok great.. i disabled firewall on her computer and my computer, would you like the router natd file ? |
| 17:39.41 | Jumpie | hey i dont tottaly understand the need for "hint" |
| 17:39.52 | Jumpie | why you'd preface the actual dial line with it? |
| 17:41.32 | CallCtr4Sale | how do i interconnect asterisk to another ip pbx via sip? |
| 17:41.33 | jeev | http://pastebin.com/m66d1262d |
| 17:41.42 | [TK]D-Fender | CallCtr4Sale: ...~book |
| 17:41.48 | [TK]D-Fender | ~book |
| 17:41.48 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:41.50 | [TK]D-Fender | ^^^^^^^^^^ |
| 17:42.41 | Jumpie | lol |
| 17:43.03 | [TK]D-Fender | jeev: /sbin/natd -n fxp0 -l -redirect_port udp 192.168.0.72:5060 5060 -redirect_port udp 192.168.0.72:10000-20000 10000-20000 -redirect_port tcp 192.168.0.72:8088 8088 |
| 17:43.06 | [TK]D-Fender | jeev: ? |
| 17:43.29 | jeev | yes sir |
| 17:43.32 | jeev | that is my freebsd router. |
| 17:43.37 | [TK]D-Fender | jeev: pastebina call with1 -way audio from beginning to end, SIP debug enabled./ |
| 17:43.48 | [TK]D-Fender | jeev: and on your * server itself? |
| 17:44.11 | jeev | ok cool |
| 17:44.20 | jeev | no, asterisk is behind the freebsd router. |
| 17:44.43 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 17:45.13 | jeev | i dunno where to start so i'll copy the whole thing bro |
| 17:46.52 | [TK]D-Fender | jeev: You don't seem to get it. You * server has its OWN FIREWALL. Go see what its doing. |
| 17:46.57 | CallCtr4Sale | thanks... |
| 17:47.13 | jeev | http://pastebin.com/m502d4e1 |
| 17:47.14 | CallCtr4Sale | anyway what digium card is specifically to be used for analog trunkline |
| 17:47.40 | [TK]D-Fender | CallCtr4Sale: How many lines? |
| 17:48.12 | CallCtr4Sale | just 1 analog trunk line |
| 17:48.16 | jeev | iptables is blank, nothing in it |
| 17:48.35 | Qwell | CallCtr4Sale: TDM410P |
| 17:48.51 | M1s3ry | be sure to get it with one FXO module |
| 17:48.58 | Qwell | (not to be confused with 410B) |
| 17:49.05 | M1s3ry | or, at least one fxo module |
| 17:49.36 | CallCtr4Sale | ok.. can it handle simultaneous calls? |
| 17:49.49 | jeev | no firewall on the server unless if it's not iptables. |
| 17:50.01 | Qwell | CallCtr4Sale: analog can't, so...no? |
| 17:50.05 | CallCtr4Sale | like a pstn trunkline can handle simultaneous calls |
| 17:50.06 | M1s3ry | one analog line = one simultaneous call. |
| 17:50.37 | Qwell | what is a trunkline? seems a bit non-standard, or your naming is wrong |
| 17:50.54 | CallCtr4Sale | trunkline sorry |
| 17:50.55 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 17:51.00 | *** join/#asterisk mindCrime_ (n=chatzill@216.27.62.2) |
| 17:51.17 | CallCtr4Sale | is there a digium card specifically made to handle a trunkline |
| 17:51.26 | Qwell | again - what is a "trunkline"? |
| 17:51.44 | M1s3ry | what do you mean? T1? analog |
| 17:51.47 | M1s3ry | etc... |
| 17:51.59 | CallCtr4Sale | its a single pstn line that can handle simultaneous calls... |
| 17:52.04 | *** part/#asterisk LemensTS (i=CustomGT@adsl-70-238-131-138.dsl.stlsmo.sbcglobal.net) |
| 17:52.29 | Jumpie | maybe thats a telecom marketing blunder |
| 17:52.49 | Qwell | CallCtr4Sale: no such thing |
| 17:53.25 | M1s3ry | CallCtr4Sale, it sounds like your talking of something along the lines of a T1 |
| 17:53.41 | Qwell | or multiple lines in a single cable |
| 17:53.46 | [TK]D-Fender | CallCtr4Sale: Your term "trunk line" is meaningless. |
| 17:53.49 | Qwell | (ie; 2 line phones) |
| 17:54.02 | M1s3ry | Qwell, true true |
| 17:54.12 | CallCtr4Sale | because we have this 1 pstn line in the office that can be called more than one at a time |
| 17:54.26 | jeev | [TK]D-Fender, i ran iptables -L and didn't detect any rules. |
| 17:54.37 | Qwell | CallCtr4Sale: you mean multiple lines with the same telephony number? |
| 17:54.41 | Qwell | telephone* |
| 17:54.53 | M1s3ry | CallCtr4Sale, ok here's another question for ya. RJ-11, or RJ-45? |
| 17:54.59 | CallCtr4Sale | rj11 |
| 17:55.08 | Qwell | what does it currently connect to? |
| 17:55.14 | CallCtr4Sale | an analog pbx |
| 17:55.19 | Qwell | clearly |
| 17:55.21 | Qwell | which analog pbx? |
| 17:55.39 | CallCtr4Sale | forgot the specific model, its a NEC pbx |
| 17:55.40 | outtolunc | waits for him to say 'the one in the other room' <G> |
| 17:55.50 | C4away | you mean 8p8c. RJ-45 is an old standard that only had two active data pins and had a shunt resistor on two of the other pins ... it is not used anymore |
| 17:55.51 | [TK]D-Fender | jeev: Everything looks OK, since you're dealing with softphones I'd seriously check out your sound cards & recording sources. |
| 17:55.53 | Qwell | and how many calls, specifically, can one "trunkline" handle simultaneously? |
| 17:56.00 | jeev | i am using USB headsets.. |
| 17:56.17 | jeev | what i'm curious about is.. is it possible to disable NAT everywhere but have the server authenticate with my SIP provider? cause if nat is disabled, it wont. |
| 17:56.24 | [TK]D-Fender | CallCtr4Sale: Well a single RJ11 typically means 1 analog line, means handles 1 call. |
| 17:56.32 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 17:56.40 | Qwell | chances are, this "problem" can be solved with a $2 splitter. |
| 17:56.43 | M1s3ry | CallCtr4Sale, look here... http://www.digium.com/en/products/analog/ this lists a various number of cards that can help you. It depends on the number of incoming lines that you have, and the number of analog phones you have for the specific version that you will need |
| 17:56.57 | CallCtr4Sale | ok thanks alot guys... |
| 17:57.01 | Qwell | CallCtr4Sale: see my above question |
| 17:57.53 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584618.dsl.bell.ca) |
| 17:58.26 | Jumpie | hmm, im showing an active registry with my provider, but its not showing as an unmonitored peer like normal, so when i try to make an outbound call, which i have specified in my outbound context, it rejects because it has no path and tries to resolve as an internal extension |
| 17:58.29 | Jumpie | am i missing something? |
| 18:01.31 | jeev | well, i disabled nat on both mine and her extension and i'm still unable :/ |
| 18:02.20 | [TK]D-Fender | jeev: check your SOUND CARDS. |
| 18:03.04 | [TK]D-Fender | Jumpie: And you still haven't learned to pastebin your problems. |
| 18:04.00 | *** join/#asterisk bougie (n=bougie@APoitiers-256-1-10-5.w90-11.abo.wanadoo.fr) |
| 18:04.42 | jeev | hm |
| 18:04.52 | jeev | i will try right now tk. |
| 18:06.19 | jeev | wow, tk... her shit was on mute.. OMG |
| 18:06.44 | Jumpie | fender..sorry |
| 18:06.54 | Jumpie | sec |
| 18:07.07 | mvanbaak | pastebins [TK]D-Fender |
| 18:07.28 | mvanbaak | there, I pastebinned my problem ;) |
| 18:09.32 | Jumpie | fender http://jumpie.pastebin.org/28798 |
| 18:09.40 | Jumpie | my contexts, asterisk output, etc |
| 18:09.50 | Jumpie | basically its trying to dial it locally because it cant get outbound, yet i do have a registry |
| 18:09.53 | Jumpie | not sure if i've missed something |
| 18:10.03 | Jumpie | i know the sip show peers should have it listed but doesnt |
| 18:12.14 | C4away | is there a way to access the raw ANI info on an inbound call? |
| 18:12.27 | *** join/#asterisk angom_w (n=angom@201.170.65.143) |
| 18:12.35 | [TK]D-Fender | Jumpie: And where is [2000] 's sip.conf entry? |
| 18:12.53 | C4away | ${CALLERID(ani)} can be set, does that include the ani info if I read it on the way in? |
| 18:13.21 | Jumpie | fender, its there, i didnt paste it |
| 18:13.25 | [TK]D-Fender | C4away: Go try |
| 18:13.27 | Jumpie | that 2000 was another nonr elated entry |
| 18:13.31 | Jumpie | i can call between 1000 and 2000 |
| 18:13.40 | Jumpie | did you stillw ant me to pastebin those? |
| 18:13.42 | [TK]D-Fender | Jumpie: 2000 is trying to dial out. |
| 18:13.53 | [TK]D-Fender | Jumpie: he's the one with the PROBLEM. |
| 18:13.59 | C4away | ok I'll try putting that as a Noop on the way in |
| 18:14.01 | *** join/#asterisk mindCrime_ (n=chatzill@216.27.62.2) |
| 18:14.05 | Jumpie | yea,s ame prob on 1000 |
| 18:14.10 | Jumpie | i thought iw ent off my outbound context? |
| 18:14.10 | [TK]D-Fender | Jumpie: because his calls don't have access to the extensions you are dialing. |
| 18:14.15 | Jumpie | hmm |
| 18:14.38 | [TK]D-Fender | Jumpie: stop talking about "outbound" like that word has some sort of magical meaning. |
| 18:14.49 | [TK]D-Fender | Jumpie: You once again fail to show the whole picture. |
| 18:15.23 | [TK]D-Fender | Jumpie: * says that IT doesn't like whyt you dialed, and you then fail to show me that phone's config, and the context in your dialplan it points to |
| 18:15.23 | Jumpie | i meant outbound under my [outbound] context, sorry, i know thats user defined |
| 18:15.30 | Jumpie | ok..my bad just a sec |
| 18:15.39 | [TK]D-Fender | Jumpie: EVERYTHING is user defined. |
| 18:16.13 | Jumpie | ah....well i think i know the problem, but ill paste it anyway hehe |
| 18:16.30 | Jumpie | http://jumpie.pastebin.org/28801 |
| 18:16.34 | Jumpie | i have context set to internal |
| 18:16.36 | [TK]D-Fender | Jumpie: Yeah, they are using another context. yes, I'm already quite sure of that |
| 18:16.45 | Jumpie | because there where i defined the extensions in extensions.conf |
| 18:16.52 | Jumpie | can you set 2 contexts? |
| 18:17.03 | [TK]D-Fender | Jumpie: Well guess what, as it they can't dial those extens you now made |
| 18:17.08 | [TK]D-Fender | Jumpie: No, you can't |
| 18:17.33 | [TK]D-Fender | Jumpie: So you intead make a 3rd which will "include =>" the multiple other contexts you want them to have access to. |
| 18:18.07 | krdian | hi |
| 18:18.09 | Jumpie | aah |
| 18:18.09 | krdian | is back. you can continue now. |
| 18:18.41 | Jumpie | its not prefaced by a # is it? thats just for files? |
| 18:18.47 | outtolunc | hands krdian ticket number 234123984924 and sends him to the back of the line <G> |
| 18:18.53 | Jumpie | lol |
| 18:19.19 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 18:20.01 | Jumpie | fender, i guess the reason i was thinking it wasnt related to that, is when my outbound was working, i had an additional sip show peers entry to callwithus, which i dont |
| 18:20.05 | Jumpie | i thought it was my config to them :P |
| 18:20.26 | *** join/#asterisk mindCrime__ (n=chatzill@216.27.62.2) |
| 18:20.44 | krdian | is there possible to configure mindspeed card cn8478 with asterisk ? |
| 18:20.47 | x86 | whould echo cancellation be prohibiting feature code DTMF from working properly? |
| 18:21.23 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:21.48 | mvanbaak | n/16 |
| 18:21.51 | mvanbaak | oops |
| 18:21.51 | UnixDog | yes brb |
| 18:22.31 | Jumpie | ok fender, something different is happening now lol |
| 18:24.30 | jeev | ok, i'm pressing buttons into x-lite.. trying to do key presses and it's probably working 1 out of 3 or 4.. |
| 18:24.43 | [TK]D-Fender | Jumpie: * is telling you to your face that the # you want doesn't match. |
| 18:25.15 | *** join/#asterisk LemensTS (i=CustomGT@adsl-70-238-131-138.dsl.stlsmo.sbcglobal.net) |
| 18:25.34 | Jumpie | yeah |
| 18:25.38 | Jumpie | im working on it now... |
| 18:25.50 | Jumpie | the includes trick worked, its actually trying to go out the right path now |
| 18:26.00 | Jumpie | but i've fudged somethin in my reg string |
| 18:26.03 | C4away | can you call a macro from within a macro? |
| 18:26.26 | x86 | when I try the sequence defined in features.conf for blind transfer, it's like asterisk is not seeing the DTMF because it just sends the key preses to the other party on the bridged call and does nothing |
| 18:26.30 | [TK]D-Fender | Jumpie: sip.callwithus.com:5060 545345523 105 Registered Thu, 10 Apr 2008 02:08:45 |
| 18:26.36 | [TK]D-Fender | Jumpie: looks registered to me. |
| 18:26.47 | x86 | would echo cancellation be causing the DTMF tones to get skewed and not allow asterisk to interpret them? |
| 18:27.02 | [TK]D-Fender | x86: Speed counts, as do Dial options. |
| 18:27.04 | LemensTS | is thier a SayNumber cmd that will say 1111 as one one one one, and not one thousand one hundred eleven? |
| 18:27.05 | Jumpie | yes, but i had this problem last time |
| 18:27.12 | Jumpie | it shows up in registry, but not in peer |
| 18:27.28 | [TK]D-Fender | LemensTS: "core show application SayDigits" |
| 18:28.05 | x86 | [TK]D-Fender: what dial options do i want? |
| 18:28.17 | [TK]D-Fender | x86: "t" or "T" |
| 18:28.27 | [TK]D-Fender | x86: Depending on who you're giving the rights to. |
| 18:28.57 | Jumpie | this is pissing me off |
| 18:29.31 | x86 | [TK]D-Fender: inbound call --> asterisk --> analog user |
| 18:29.41 | x86 | [TK]D-Fender: i want analog user to be able to transfer the call |
| 18:29.48 | Jumpie | ah..i think there is a part i'm missing |
| 18:29.50 | x86 | [TK]D-Fender: 't' then? |
| 18:30.37 | x86 | [TK]D-Fender: T would allow the caller to transfer... I want the callee to transfer |
| 18:30.41 | x86 | right? |
| 18:31.08 | [TK]D-Fender | x86: "core show application dial" |
| 18:31.41 | x86 | [TK]D-Fender: yep, that was exactly my problem... was not using 't' |
| 18:31.45 | x86 | works great now, thanks man :) |
| 18:33.12 | Jumpie | woohoo it worked |
| 18:33.30 | Jumpie | fender, i had the registry string, but i did not have the actual context in my sip.conf for them |
| 18:35.18 | [TK]D-Fender | Jumpie: in sip.conf those are not "contexts". You can call them either "sections", or "users/peers" |
| 18:35.22 | [TK]D-Fender | x86: You're welcome. |
| 18:35.30 | Jumpie | sorry |
| 18:35.49 | Jumpie | i wish things would reference things the same across the board lol |
| 18:36.03 | Jumpie | i had to create a [section] withmy callwithus username |
| 18:36.42 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 18:36.42 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:41.02 | jeev | should "full => notice,warning,error,debug,verbose,dtmf" in logger.conf be logging keypresses? |
| 18:43.01 | *** join/#asterisk Defraz (i=t0tal@72.24.26.7) |
| 18:46.02 | *** join/#asterisk angom_w (n=angom@201.170.65.143) |
| 18:47.23 | *** join/#asterisk mknerd2008 (i=3f951603@gateway/web/ajax/mibbit.com/x-5535517876637c2c) |
| 18:48.34 | mknerd2008 | hello world |
| 18:48.36 | mknerd2008 | :-) |
| 18:50.23 | LemensTS | TKD-Fender: Im just curious, do you work for Digium? |
| 18:51.41 | Jumpie | hm |
| 18:51.49 | Jumpie | i think my outbound dialing method is incorrect |
| 18:52.46 | jeev | i'm trying to log key presses but full => notice,warning,error,debug,verbose,dtmf isn't helping. |
| 18:55.06 | [TK]D-Fender | LemensTS: Nope, independent consultant. |
| 18:59.28 | jeev | [TK]D-Fender .. do you know how i could log key presses ? |
| 19:00.36 | [TK]D-Fender | jeev: What "keypresses"? When? How? Why? |
| 19:01.59 | jeev | i'm calling an extension.. she answers, i press random keys.. i want it to log.. cause she's claling the courts and it's working randomly.. |
| 19:02.05 | krdian | jeev: you should see dtmfs in full.log |
| 19:02.15 | jeev | full => notice,warning,error,debug,verbose,dtmf |
| 19:02.34 | jeev | i have that in logger.conf, reload.. it's logging full.log but i dont see any key preses, but when i or her press it while we're on the extension, we hear the noise |
| 19:03.46 | [TK]D-Fender | jeev: Test it out your PSTN link to another phone directly. |
| 19:04.51 | *** join/#asterisk Defraz (i=t0tal@72.24.26.7) |
| 19:04.53 | jeev | i did, works on the original nortel system |
| 19:07.17 | LemensTS | is thier anyway to do 1 for yes, 2 for no, without being in an 's' extension? |
| 19:08.22 | LemensTS | i usually create multiple contexts with 's' extensions so i can keep asking more questions, but im sure thier is a better way? |
| 19:09.28 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 19:11.13 | Jumpie | fender |
| 19:11.14 | Jumpie | http://jumpie.pastebin.org/28818 |
| 19:11.17 | Jumpie | can you look at that please? |
| 19:11.41 | Jumpie | im getting failed to authenticate...i know this is somethin simple im missing but i cant figure it out, also im following the settings per callwithus.com's config page |
| 19:11.54 | *** join/#asterisk Defraz (i=t0tal@72.24.26.7) |
| 19:13.51 | [TK]D-Fender | Jumpie: -- Executing [17037070075@fromhome:1] Dial("SIP/2000-089eba40", "SIP/sip.callwithus.com/17037070075|120") in new stack |
| 19:14.30 | [TK]D-Fender | Jumpie: you have a peer called [545345523]to call out your provider and here you are trying to dial by IP with NO auth or anything. |
| 19:14.53 | jeev | :< |
| 19:14.53 | outtolunc | change your this CALLWITHUS=SIP/sip.callwithus.com to this CALLWITHUS=SIP/545345523 |
| 19:15.01 | riddlebox | I hate trying to walk salesmen through opening ports on their firewalls |
| 19:15.09 | Jumpie | that sip peer does have a user/pass on it fender |
| 19:15.17 | Jumpie | outtolunc, i thikn i tried that earlier, same problem |
| 19:15.21 | Jumpie | but ill try again |
| 19:15.48 | [TK]D-Fender | Jumpie: the peer does... and you're not SUING IT |
| 19:15.51 | [TK]D-Fender | USING* |
| 19:16.06 | Jumpie | ok ..hmm sec |
| 19:16.30 | x86 | [TK]D-Fender: Dyslexia means never having to say that you're ysror. |
| 19:16.48 | [TK]D-Fender | x86: ind'd |
| 19:16.51 | outtolunc | also, try not to use [xxxxxx] and username=xxxxxx with the same strings |
| 19:17.18 | Jumpie | outtolunc, sorry, i pasted from their site |
| 19:17.20 | jeev | what could cause some key presses to work and some not to work ? |
| 19:17.22 | Jumpie | so which one should i do? |
| 19:17.28 | *** join/#asterisk talntid (n=t@66.208.251.170) |
| 19:17.28 | outtolunc | using [theprovidername] makes it easier to read |
| 19:17.47 | outtolunc | just a suggestion |
| 19:18.14 | Jumpie | ah..ok |
| 19:18.27 | Jumpie | i guess that was just their 'recommendation' i was guessing it was required..duh..thanks :D |
| 19:18.55 | *** join/#asterisk MikeJ (n=MikeJ@33.203.64.208.static.accentrainc.com) |
| 19:20.30 | jeev | [TK]D-Fender, i think it has to do with toneduration, it says it's quite short, i notice that when i tap it, the IVR for the court figures it out |
| 19:20.42 | Jumpie | outtolunc, is my dial out syntax right? |
| 19:20.53 | outtolunc | it should work.. |
| 19:21.04 | Jumpie | i changed it to what you said and now i get this: [Apr 10 03:19:56] WARNING[5830]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 19:21.05 | outtolunc | but also, do not make it 'provider specific' |
| 19:22.31 | outtolunc | exten => _1NXXNXXXXXX,1,Dial(${CALLWITHUS}/${EXTEN},120) should remain as ${TRUNK} and set TRUNK= yadda at top.. that way (once again, only a suggestion) because if you change providers.. and someone reads dial(CALLWITHUS... ) they start thinking oh its provider specific |
| 19:22.33 | Jumpie | [Apr 10 03:19:56] WARNING[5830]: chan_sip.c:2925 create_addr: No such host: 545345523 |
| 19:22.52 | Jumpie | outtolunc, ah didnt know that was intended for me...ok |
| 19:23.06 | outtolunc | repaste what you have now |
| 19:23.45 | Jumpie | K....lol |
| 19:24.01 | outtolunc | note: if you can't get your register to work with a proper friend/peer setup.. this is not going to work |
| 19:25.14 | Jumpie | right |
| 19:25.17 | Jumpie | i think its somethin so simple |
| 19:25.21 | Jumpie | hold on compiling stuff forf pastebin |
| 19:25.22 | outtolunc | do this |
| 19:25.27 | outtolunc | going priv |
| 19:26.19 | [TK]D-Fender | Jumpie: I'm guessing you renamed your sip peer name.... |
| 19:26.46 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 19:28.58 | *** part/#asterisk MikeJ (n=MikeJ@33.203.64.208.static.accentrainc.com) |
| 19:29.35 | *** join/#asterisk Maqiavelo (n=dfinol@64.116.134.90) |
| 19:29.41 | Maqiavelo | Good evening guys |
| 19:31.10 | Maqiavelo | I have a question, we have an Avaya PBX at work. The guys here wanted to buy something really expensive for IVR but I sort of shot em down told them Asterisk could be used for IVR. I've done it before using SIP, however Avaya only supports H323 |
| 19:31.10 | Maqiavelo | The only thing I need the Asterisk is to receive the "call" and start the IVR through H323 |
| 19:31.11 | Maqiavelo | So it's one way only Avaya -> Asterisk |
| 19:31.17 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 19:31.31 | Maqiavelo | the Asterisk server won't be making any calls, just IVR |
| 19:31.51 | Maqiavelo | I've tried googling a lot and I even have the Asterisk book from O'reilly, but it's 95% SIP stuff |
| 19:32.33 | Maqiavelo | How can I tell Asterisk that any call from H323 start the IVR? |
| 19:33.43 | keith4 | H323 is just a channel type |
| 19:33.53 | keith4 | just like SIP, Zap, etc. |
| 19:34.21 | Maqiavelo | I assume I have to edit h323.conf because I saw the file |
| 19:34.49 | Maqiavelo | Whoever, I see something like [Billy Bob] and where it falls |
| 19:34.51 | *** join/#asterisk asdx (n=diego@adsl-154-206.click.com.py) |
| 19:34.59 | Maqiavelo | But I want _all_ h323 "calls" be routed to the IVR |
| 19:35.03 | RobH | Maqiavelo: what keith4 said, the codec doesnt normally get used to set where in the dialplan call goes |
| 19:35.19 | riddlebox | wohoo just picked up 2 S110M cards for $31 a piece |
| 19:35.22 | RobH | Maqiavelo: you direct where it goes based on channel, not codec. |
| 19:35.27 | RobH | zap, iax2, sip |
| 19:35.31 | asdx | hi, can i set a variable in the dialplan for matching a string, like _X and that _X will get any quantity of numbers like 03459348534583453 or whatever |
| 19:35.54 | Jumpie | fender |
| 19:35.57 | RobH | asdx: Yep, _XX will match any number 0 to 9, two digits |
| 19:35.58 | Jumpie | im dialing outbound now :) |
| 19:36.02 | Maqiavelo | RobH, mhmh |
| 19:36.15 | Maqiavelo | RobH, There's this in the h323.conf sample conf |
| 19:36.16 | Maqiavelo | ;[BillyBob] |
| 19:36.16 | Maqiavelo | ;type=user |
| 19:36.16 | Maqiavelo | ;host=192.168.1.1 |
| 19:36.17 | Jumpie | i was right around it, but i was a bit off, outtolunc helped me point to my [calwithus] in sip.conf |
| 19:36.27 | Maqiavelo | I just change the host there to the Avaya IP? |
| 19:36.33 | keith4 | not quite |
| 19:36.33 | Maqiavelo | and then edit the incoming context? |
| 19:36.48 | RobH | Maqiavelo: There may be a setting where you dump the channel into a context, test for what kind of codec its using, then use if statement to send it elsewhere, but thats beyond me |
| 19:36.57 | asdx | RobH: what about if people wants to call to other places where there must be more than two digits? i have to set another extension for that? |
| 19:37.04 | Maqiavelo | Ok I'm lost. |
| 19:37.22 | asdx | RobH: it means that i have to ask my customers, "how much digits does the phone number have?" |
| 19:37.22 | keith4 | how are you going to tell the avaya box to pass the call to asterisk? |
| 19:37.22 | plik | Jumpie: well done, now make notes and do it again ;) |
| 19:37.45 | RobH | asdx: chapter 6 in the book |
| 19:37.48 | RobH | ~book |
| 19:37.48 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 19:37.52 | Jumpie | wash rinse repeat :D |
| 19:37.56 | Maqiavelo | keith4, Probably when a call comes to the Avaya redirect it to certain IP address (the asterisk box) |
| 19:38.10 | plik | Jumpie: it's a good way to learn |
| 19:38.11 | Maqiavelo | using h323 and regarding codec, I think g711 |
| 19:38.34 | RobH | stuff like _9NX. means match anything starting 9(2to9)(0to9)plus anyhting after |
| 19:38.50 | RobH | the book explains it much better than I do. |
| 19:39.26 | UnixDog | no |
| 19:39.36 | UnixDog | [2-9] = 2 to 9 |
| 19:39.41 | UnixDog | 9= 9 only |
| 19:39.51 | UnixDog | sorry miss red |
| 19:39.53 | UnixDog | lol |
| 19:39.59 | Maqiavelo | keith4, Does that make sense? |
| 19:40.10 | keith4 | Maqiavelo: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+h323.conf |
| 19:41.54 | keith4 | i don't know enough about h323 to offer specific tips |
| 19:42.26 | keith4 | but from what I understand, you need to do some configuration on both ends, here |
| 19:42.52 | Jumpie | jis there a way to do a quick stream/play of audio files in /var/lib/asterisk/sounds so i can see what i like? :D |
| 19:42.56 | *** join/#asterisk BCS-Satori (n=BCS-Sato@75.148.21.113) |
| 19:43.02 | Jumpie | without file/opening in a gui sound app |
| 19:43.13 | keith4 | Jumpie: yes. playback() or background() |
| 19:43.28 | BCS-Satori | Has anyone messed around with Asterisk running on an citrix xenserver and rtp perforamnce. Not talking high loads just for devel and demo |
| 19:43.44 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:43.50 | Jumpie | keith4, i meant like right from the command line |
| 19:43.55 | Jumpie | so i can do a lot in a row |
| 19:44.25 | Maqiavelo | keith4, Thanks I'll look into that |
| 19:44.52 | Jumpie | like literally, i wanna listen to each and every sound file in this dir |
| 19:44.57 | keith4 | Jumpie: they're just GSM audio. mplayer should be able to play them, no? |
| 19:45.24 | Jumpie | ya |
| 19:45.26 | Jumpie | play worked too |
| 19:45.27 | Jumpie | :P |
| 19:45.30 | Jumpie | thanks |
| 19:45.35 | Jumpie | right from bash |
| 19:45.43 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 19:46.12 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 19:48.21 | Jumpie | blue-eyed-polar-bear.gsm |
| 19:48.22 | Jumpie | wtf |
| 19:48.58 | keith4 | there are lots of "wtf" sound files in there |
| 19:49.11 | Jumpie | ya, a naughtyboy and girl too haha |
| 19:49.14 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 19:50.08 | Jumpie | oh there..im gonna play deadbeat.gsm and bad.gsm when you try to dial intl |
| 19:50.10 | Jumpie | good practice |
| 19:50.40 | asdx | RobH: chapter 6 didn't answer my question... sadly |
| 19:50.53 | RobH | hrmm, maybe I misreferened |
| 19:50.54 | RobH | lemme check |
| 19:51.19 | asdx | RobH: maybe the pattern matching section? |
| 19:51.37 | RobH | 137 |
| 19:51.38 | RobH | pg |
| 19:51.39 | RobH | sorry |
| 19:51.41 | RobH | heh. |
| 19:51.41 | jblack | I got IPKall working with IAX. ;) |
| 19:51.49 | Jumpie | hmm i was trying to do a core show application play |
| 19:51.52 | Jumpie | and it says im not registered.... |
| 19:51.54 | Jumpie | to what? |
| 19:52.07 | outtolunc | ...play<tab> |
| 19:52.18 | asdx | RobH: "configuring the dialplan for testing" ? |
| 19:52.18 | RobH | I am a horrid person, I just glanced at the table of contents and did not verify the actual text. |
| 19:52.22 | asdx | RobH: that one? |
| 19:52.27 | RobH | dialplan basics chapter |
| 19:52.45 | RobH | in 'building an interactive dialplan' |
| 19:52.51 | RobH | page 137 of the second edition book |
| 19:52.55 | Jumpie | oh.. |
| 19:53.10 | RobH | asdx: see it? |
| 19:53.30 | asdx | yeah |
| 19:53.40 | Jumpie | imidiot |
| 19:54.05 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:54.06 | RobH | asdx: hope that answers it =] |
| 19:54.38 | Jumpie | outtolunc, im trying to play 2 files, with a delay, i guesst he delay isnt in the playback |
| 19:55.06 | Jumpie | exten => _011.,1,Playback(deadbeat),1,Playback(bad) kinda like that |
| 19:55.09 | Jumpie | but i think the 1 is wrong |
| 19:55.20 | outtolunc | no and it is &file1&file2 |
| 19:55.26 | outtolunc | just do |
| 19:55.45 | outtolunc | playback(file1); waitexten(1); playback(file2); |
| 19:55.52 | outtolunc | or .5 |
| 19:56.13 | Jumpie | rofl...that works |
| 19:56.23 | outtolunc | er file1&file2 |
| 19:56.28 | Jumpie | it broke cleanly enough to just do deadbead&bad |
| 19:57.02 | Jumpie | so now my next project is incoming |
| 19:57.07 | Jumpie | ill do that later :) thanks for all your help |
| 19:57.32 | outtolunc | thank fender, he is the one that invested a lot of time |
| 19:58.26 | Jumpie | thank you fender :P |
| 19:58.50 | M1s3ry | outtolunc, any reason to use waitexten instead of wait? |
| 19:58.52 | *** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 19:58.57 | plik | wonders how many call minutes worldwide can be attributed to [TK]D-Fender in any given month |
| 19:59.10 | outtolunc | because wait doesn't work in ael's so i just just waitexten everywhere <G> |
| 19:59.35 | M1s3ry | ahhh... I knew I was missing a bit of information... k, just wanted to be sure |
| 20:00.22 | Jumpie | outtolunc, |
| 20:00.27 | Jumpie | one last question |
| 20:00.27 | outtolunc | jumpie, |
| 20:00.57 | Jumpie | my caller id is going through my provider as my 2000, vs matching what is my DID, if they dont get a valid CID, they use a bogus one |
| 20:01.05 | Jumpie | ist hat setup in my extensions.conf portion? |
| 20:01.19 | Jumpie | like anything going out that provider i want 7034364555 |
| 20:01.23 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 20:01.30 | Jumpie | just wasnt sure if its on the first line or make a second line |
| 20:01.34 | Jumpie | under [outbound] |
| 20:01.42 | Jumpie | or in sip.conf |
| 20:01.50 | outtolunc | then set the callerid for that [2000] (and all other users) to whatever you want |
| 20:01.57 | outtolunc | the point being.. |
| 20:02.33 | outtolunc | if you set it as a global, it will get overwritten by the individual one and if it isn't set, that callerid is set to the [username] |
| 20:02.33 | Jumpie | from inside i want it to show the internal extension |
| 20:02.42 | Jumpie | yea... |
| 20:02.48 | outtolunc | so.. you have to set it for each one |
| 20:03.02 | Jumpie | but isnt there a way to say anything dialed out on the SIP connection to have one caller id? |
| 20:03.02 | outtolunc | if you want to 'make sure' |
| 20:03.08 | Jumpie | if its an internal dial, it;'ll never hit that |
| 20:03.34 | outtolunc | not that i'm aware of because of the code that says.. if it isn't set, use the [xxxxx[ |
| 20:03.40 | outtolunc | you can try |
| 20:03.47 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 20:03.50 | outtolunc | just watch the asterisk manager |
| 20:03.53 | Jumpie | ok |
| 20:03.54 | Jumpie | sec |
| 20:04.03 | outtolunc | it will show you the callerid being set from unknown to something |
| 20:04.21 | outtolunc | not the CLI, the asterisk manager |
| 20:04.53 | Jumpie | oh....maybe im not used to checking that :) |
| 20:04.56 | outtolunc | http://dynx.net/ASTERISK/DOCS/RTF/MANAGER.RTF |
| 20:05.03 | Jumpie | well when i check my provider call history, it shows 2000 |
| 20:05.31 | outtolunc | and you are registered as user 2000 for your extension right? |
| 20:05.42 | Jumpie | yes |
| 20:05.43 | outtolunc | hello <G> |
| 20:05.50 | Jumpie | but i know theres a way to do this somehow |
| 20:06.05 | Jumpie | lol |
| 20:06.05 | nny_1 | so i am trying to transfer a call from Zap to a SIP outbound like: exten => s-Zap,12,Transfer(ZAP/g1/${ARG1}) exten => s-Zap,13,Transfer(SIP/${AREACODE}${ARG1}@${SIPOUTBOUND}) but it fails and dies. If a call comes in on the SIP channel and goes out it is ok, any advice? |
| 20:06.25 | outtolunc | you can set it in your phone, softphone, the [sip user], the globals.. |
| 20:06.34 | outtolunc | each will have less and less priority |
| 20:06.42 | nny_1 | oops er, the first exten => s-Zap,12,Transfer(ZAP/g1/${ARG1}) trys the zap channel, and if it doesn't work, than tries Zap,13,Transfer(SIP/${AREACODE}${ARG1}@${SIPOUTBOUND}) |
| 20:06.47 | Jumpie | outtolunc, basicaly if an internal calls and internal, preserve that caller id (laptop, 1000 whatever) but if i make an outbound call from ANY internal extension, carry the 17034364555 cid |
| 20:06.54 | Jumpie | argh |
| 20:06.54 | Jumpie | ok |
| 20:07.04 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 20:07.20 | outtolunc | then just before the dial(trunkout).. do a setcallerid or SET(callerid(num)) |
| 20:07.24 | outtolunc | er = |
| 20:07.40 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 20:08.17 | Jumpie | thats what i was lookin for :) |
| 20:08.19 | Jumpie | thanks |
| 20:09.08 | outtolunc | nny, why not just use dial (instead of transfer) |
| 20:09.16 | *** join/#asterisk angom_w (n=angom@201.170.65.143) |
| 20:11.01 | ManxPower | nny_1: Transfer is mostly for IVRs and their ilk. It is not a good thing to use in general |
| 20:11.16 | Jumpie | wait...is it setcallerid(num) or setcallerid = "num" |
| 20:11.43 | outtolunc | core show application setcallerid |
| 20:11.44 | nny_1 | outtolunc: ManxPower noobie mistake thanks |
| 20:12.17 | *** join/#asterisk fransena (n=fransena@207.229.0.38) |
| 20:12.18 | Jumpie | ah it worked |
| 20:12.24 | Jumpie | setcallerid(num) worked :) thanks! |
| 20:12.29 | *** join/#asterisk Evilkiksass (n=do@75.35.230.5) |
| 20:12.32 | mknerd2008 | whats the command to accept DMTF tones into a variable? |
| 20:12.38 | ManxPower | Jumpie: It is WHATEVER channelvariables.txt tells you it is. |
| 20:12.43 | outtolunc | read |
| 20:12.45 | ManxPower | mknerd2008: Read |
| 20:13.00 | ManxPower | or "core show applications like DTMF" |
| 20:13.14 | mknerd2008 | thx |
| 20:14.36 | fransena | I have a new asterisk appliance, and I've got it going, but the park function doesn't appear to be operating properly. When a call is parked, the "parkee" goes to 701, but the "parker" does not get the announcement and goes to MOH. No config in extensions.conf, only using the std lines in featurs.conf |
| 20:14.57 | Evilkiksass | Does anyone know if Asterisk officialy supports lldp? |
| 20:15.52 | [TK]D-Fender | mknerd2008: "core show application read" |
| 20:16.34 | [TK]D-Fender | fransena: If the transfer is underway properly and you hear nothing, this is typically a sign that you have a BLIND transfer to the parking lot, and not an ATTENDED one like you need to be doing. |
| 20:17.30 | *** part/#asterisk riddlebox (n=JamesMid@75-128-170-26.static.stls.mo.charter.com) |
| 20:19.11 | Jumpie | i have actualy dialed a phone with my voice before :) |
| 20:20.44 | nny_1 | whats the most efficient way to play moh while a extension is being rung? Do I have to park the call, ring the extension, and then bridge the parked call to the extension, or can asterisk just play moh instead of ringing noises? |
| 20:21.16 | nny_1 | (i suspect that i answered my own question, as the ringy noises come from the phone, and that asterisk has already bridged the call to the sip channel) |
| 20:21.42 | file | nny_1: the 'm' option to Dial allows it to provide music on hold, instead of ringing |
| 20:21.48 | nny_1 | pardon my 12 year old speech, someone left a bowl of crack and i can haz cheezeburger pitcures at the office |
| 20:21.54 | nny_1 | well hell that is efficient |
| 20:22.00 | outtolunc | <PROTECTED> |
| 20:22.00 | outtolunc | <PROTECTED> |
| 20:22.00 | outtolunc | <PROTECTED> |
| 20:22.11 | outtolunc | core show application dial |
| 20:22.17 | nny_1 | outtolunc: ty |
| 20:22.36 | outtolunc | sorry for the big paste |
| 20:23.09 | nny_1 | outtolunc: does that work with Dial and Transfer? |
| 20:23.28 | outtolunc | core show application dial does NOT equal tranfer |
| 20:24.07 | nny_1 | yeah |
| 20:24.22 | outtolunc | dial is like a a huge dictionary, where transfer is a 2 page fold out <G> |
| 20:24.43 | nny_1 | hehe yeah nm i don't use transfer atm anyways in it, <see crack statement> |
| 20:26.35 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 20:26.50 | teknoprep | for polycom phones where do i set which file is my speed dial list ? |
| 20:28.04 | UnixDog | mac-directory.cfg |
| 20:28.30 | teknoprep | yeah is there a way to make a global ? |
| 20:28.44 | UnixDog | for all phones |
| 20:28.48 | teknoprep | yeah |
| 20:28.59 | UnixDog | just he 00000000000.directory.cfg |
| 20:29.10 | teknoprep | 000000000000-directory~.xml |
| 20:29.13 | UnixDog | and not making a personal on |
| 20:29.15 | teknoprep | ahh i had a tilda lol |
| 20:29.18 | UnixDog | xml sorry |
| 20:29.31 | *** join/#asterisk Kyoshi (i=whoa@pool-71-167-117-15.nycmny.fios.verizon.net) |
| 20:29.33 | teknoprep | it was default with my updates i downloaded from polycom |
| 20:29.37 | UnixDog | been coding now I cant type |
| 20:29.42 | teknoprep | lol thats fine |
| 20:30.09 | UnixDog | 1.6 is almost fixed on fbsd 7.0 |
| 20:31.05 | outtolunc | it is *almost* the weekend also <G> |
| 20:31.07 | *** join/#asterisk plik (i=gorph@phalse.2600.COM) |
| 20:31.39 | fransena | ok, after reading up on blind vs attended transfers, I think I'm just as confused as when I started. I'm using the Aastra 57i phones, and they have a built in "Park" function which I have dialing 700. If I read the docs right, I need to instead dial *2 (or whatever I have in features.conf) THEN 700 to park the call? |
| 20:31.54 | UnixDog | I have 2 more files to patch |
| 20:32.26 | UnixDog | blind means you just transfer the call and it hangs up on you |
| 20:32.52 | UnixDog | attends means you stay inline during the transfer and then after they connect you talk then hang up |
| 20:32.53 | *** join/#asterisk Bourrelle (n=Bourrell@132.207.156.100) |
| 20:32.59 | UnixDog | whats so confusing about that |
| 20:33.09 | UnixDog | they stay connected |
| 20:33.21 | Bourrelle | Hello, i just read the whole asterisk documentation but I have 1 more little question remaining |
| 20:33.24 | fransena | Sorry, I should have been clearer...I understand the difference between the transfer types, I just don't "get" how to implement them. |
| 20:33.52 | UnixDog | features.conf |
| 20:34.01 | Bourrelle | Im making a SIP phone that can handle conference |
| 20:34.15 | Bourrelle | in a conference, theres 3 persons right |
| 20:34.29 | Bourrelle | so one of then will mixer the audio/video to the 2 others |
| 20:34.35 | ManxPower | fransena: HOW exactly you do a specific type of transfer depends on the device you are using to do the transfer. If you are using an IP phone, CHECK THE DOCS for that phone |
| 20:34.39 | UnixDog | depends on the confrence |
| 20:34.54 | UnixDog | some confrence calls can be nway and or 3 way |
| 20:34.58 | Bourrelle | Im wondering if asterisk can handle 2 connection from the same |
| 20:35.04 | Bourrelle | computer |
| 20:35.08 | Bourrelle | at the same time |
| 20:35.10 | fransena | gotcha. thanks ManxPower. |
| 20:35.13 | Bourrelle | without putting it on hold |
| 20:35.22 | ManxPower | fransena: For parking you want to do a supervised/consultative transfer. If you do a blind transfer the caller will hear the parking lot number read back, not you. |
| 20:35.53 | ManxPower | Bourrelle: Asterisk has handled hundreds of connections from the same computer. |
| 20:36.58 | jeev | i'm still having trouble with DTMF, i'm pressing a button softly, like you'd normally press 1 if it said press 1... but it's not working, only way it'll recognize it is if i tap it REALLY fast |
| 20:37.36 | UnixDog | jeev inband/outofband/rfc2833/auto ? |
| 20:37.59 | jeev | rfc2833 |
| 20:38.06 | jeev | is what my provider set me up on |
| 20:38.32 | Bourrelle | so I could call 2 persons, receives there audio , mix it, and hear the composition of both audio ? |
| 20:38.32 | UnixDog | you have to check with your provider |
| 20:38.41 | UnixDog | and thier trunking configuration |
| 20:38.50 | jeev | they told me 2833. |
| 20:39.05 | UnixDog | ok |
| 20:39.06 | jeev | i read up on dtmfduration and toneduration.. but unable to find anything in docs anymore, was it depricated? |
| 20:39.21 | Bourrelle | How could Exosip handle this localy... |
| 20:39.28 | UnixDog | and on the exten you set dtmf=rfc2833 dtmfmode=rfc2833 |
| 20:39.34 | UnixDog | and on the trunk ? |
| 20:39.37 | Bourrelle | I just dont get it |
| 20:39.38 | jeev | corect |
| 20:39.42 | jeev | correct |
| 20:39.42 | Bourrelle | When i try to make another call |
| 20:39.50 | Bourrelle | asterisk put 1 call on hold |
| 20:40.00 | ManxPower | jeev: no, Asterisk never supported durations for RFC2833 DTMF. Perhaps someone was talking about the BETA 1.6? |
| 20:40.00 | Bourrelle | without asking me |
| 20:40.08 | UnixDog | your phone puts the call on hold not asterisk |
| 20:40.36 | ManxPower | Bourrelle: I hope you are looking at SIP packet traces before you start accusing Asterisk of putting a call on hold automatically |
| 20:40.42 | jeev | ManxPower: http://www.voip-info.org/wiki/view/Asterisk+DTMF |
| 20:40.54 | Jumpie | hmm im going nuts here |
| 20:41.20 | Jumpie | trying to set a pattern so that if i dont dial 1 it can add it to calls |
| 20:41.25 | Jumpie | but only when prefaced by 703 |
| 20:41.27 | file | Asterisk 1.4 and higher preserves the duration of DTMF |
| 20:41.36 | Jumpie | exten => 1+703NXXXXXX,2,Dial(${TRUNKOUT}/${EXTEN},120) |
| 20:41.43 | Jumpie | am i jacking that up? |
| 20:41.55 | outtolunc | yes |
| 20:42.02 | outtolunc | just do |
| 20:42.05 | jeev | does anyone want the number im' calling? ( san diego superior court ) |
| 20:42.08 | jeev | and try it |
| 20:42.38 | outtolunc | exten => _NXXNXXXXXX,1,Dial(${TRUNKOUT}/${EXTEN},120) |
| 20:42.39 | UnixDog | 120 |
| 20:42.39 | jeev | if i dial 4 normally, it wont go through, if i lightly tap it.. it'll recognize. |
| 20:42.39 | ManxPower | jeev: It clearly states on that page that those two options are for ZAP!!!!! |
| 20:42.43 | outtolunc | exten => _1NXXNXXXXXX,1,Dial(${TRUNKOUT}/${EXTEN},120) |
| 20:42.43 | UnixDog | wow thats a long ring time |
| 20:42.45 | jeev | 0_o |
| 20:42.50 | jeev | i didn't notice |
| 20:42.53 | Bourrelle | thanks, il take a look, maybe I got it wrong |
| 20:43.02 | Jumpie | heh that meant to be 20 sory |
| 20:43.09 | jeev | ManxPower, any recommendations? |
| 20:43.10 | Jumpie | outtolunc, i have that 2nd one already |
| 20:43.26 | Jumpie | i just want to be able to 10 digit dial a number starting with 703 |
| 20:43.29 | Jumpie | all others fail without 1 |
| 20:43.31 | mknerd2008 | I am trying to use read and then say the digits |
| 20:43.31 | ManxPower | jeev: I recommend that you stop thinking zap options apply to SIP. |
| 20:43.41 | jeev | i understood that. |
| 20:43.44 | outtolunc | jumpie you are not understanding or have them wrong |
| 20:44.00 | Jumpie | exten => _1NXXNXXXXXX,1,Dial(${TRUNKOUT}/${EXTEN},120) i have already |
| 20:44.02 | ManxPower | jeev: perhaps you might look at sip.conf.sample |
| 20:44.09 | Jumpie | im trying to allow myself to dial 703 numbers without a 1 |
| 20:44.09 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584618.dsl.bell.ca) |
| 20:44.15 | jeev | ok let me check it out |
| 20:44.17 | ManxPower | exten => _703NXXXXXX,1,Dial(${TRUNKOUT}/${EXTEN},120) |
| 20:44.17 | mknerd2008 | its trying to pass the digits as an extension |
| 20:44.26 | Jumpie | ManxPower, i tried that too |
| 20:44.30 | ManxPower | Jumpie: are you really that inexperienced with numbers. |
| 20:44.33 | Jumpie | its still trying to dial an extension called 703XXXXXXX |
| 20:44.41 | Jumpie | hah no..im only asking because i tried and failed :( |
| 20:44.48 | Jumpie | maybe i goofed let me try again |
| 20:44.54 | ManxPower | Jumpie: You just SAID you wanted to match 703 |
| 20:45.00 | Jumpie | yes |
| 20:45.05 | outtolunc | _1NXX and _NXX are 2 different extension patterns |
| 20:45.09 | ManxPower | I just told you how to match 803 |
| 20:45.11 | Jumpie | i realize this |
| 20:45.16 | outtolunc | they are both ...,1's |
| 20:45.17 | Jumpie | ManxPower, ok ...let me try that again |
| 20:45.20 | ManxPower | now what is the problem. |
| 20:45.30 | ManxPower | Jumpie: Are you going to ask a coherent question this time? |
| 20:45.45 | Jumpie | how was i not being coherent? |
| 20:46.02 | Jumpie | im pretty sure i did what you had asked before and failed, but i'll try again, may have been mistraken |
| 20:46.16 | ManxPower | "Jumpie: i just want to be able to 10 digit dial a number starting with 703" "ManxPower: exten => _703NXXXXXX,1,Dial(${TRUNKOUT}/${EXTEN},120) " |
| 20:46.36 | outtolunc | on the _NXX one you make it Dial(${TRUNKOUT}/1{$EXTEN},120) |
| 20:46.56 | jeev | ManxPower, anything special to look for in the example? |
| 20:47.10 | ManxPower | jeev: I never have DTMF problems. |
| 20:47.19 | Evilkiksass | Would anyone happen to know if Asterisk supports LLDP? |
| 20:47.44 | ManxPower | Evilkiksass: I can say with a reasonable level of confidence that Asterisk does not support LLDP. |
| 20:48.01 | Evilkiksass | Thank you ManxPower. |
| 20:48.01 | ManxPower | But perhaps Asterisk supports something similar. |
| 20:48.18 | Evilkiksass | Well I need to know because I need to test my product with LLDP. |
| 20:48.29 | ManxPower | Exactly what is LLDP? |
| 20:48.39 | Evilkiksass | http://en.wikipedia.org/wiki/Link_Layer_Discovery_Protocol |
| 20:49.14 | Jumpie | ManxPower, still no cigar |
| 20:49.31 | ManxPower | Jumpie: Can you be a little more vague please? |
| 20:49.39 | Jumpie | http://jumpie.pastebin.org/28845 |
| 20:49.44 | Jumpie | hold on thats why i paste bined :) |
| 20:50.13 | Jumpie | when i try to dial just 703 707 0075 i still get this: [Apr 10 04:48:25] NOTICE[5785]: chan_sip.c:13952 handle_request_invite: Call from '2000' to extension '7037070075' rejected because extension not found. |
| 20:50.32 | ManxPower | Jumpie: you don't have a prioirty 1 |
| 20:50.47 | Jumpie | oh...doh |
| 20:50.54 | Jumpie | each new one starts a priority 1 doesnt it |
| 20:51.00 | Jumpie | i have a priority one for the very first line and thats it |
| 20:51.22 | ManxPower | Jumpie: 703 is a different extension, so you need a priority 1 |
| 20:51.32 | ManxPower | actual order and spacing do not matter |
| 20:51.34 | Jumpie | yes my mistake, too swift on copy paste |
| 20:52.15 | M1s3ry | ManxPower, hehe... so I got a support call this morning regarding nearly the same thing that jumpie had. so I looked in the cli saw the same thing, asked the guy if he was dialing a "9" as per what was in his extensions.conf.... He started cussing at himself, thanked me, then hung up... |
| 20:52.29 | outtolunc | order only matters when you have a wildcard and a specific, the specific needs to be first |
| 20:52.54 | ManxPower | M1s3ry: I hope they pay you well 8-) |
| 20:52.54 | ManxPower | outtolunc: as I understand it, that only applies to include => stuff |
| 20:52.59 | outtolunc | (specific being a subset of the wildcard) |
| 20:53.18 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 20:53.25 | M1s3ry | ManxPower, it's digium... |
| 20:53.31 | ManxPower | M1s3ry: Ah. |
| 20:53.34 | outtolunc | manx, a _X, and 1 in that order the wildcard wins |
| 20:53.52 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 20:53.52 | outtolunc | if you do 1, then the _X your 1 will win |
| 20:54.13 | ManxPower | outtolunc: I'm too lazy to prove you wrong 8-)_ |
| 20:54.21 | Jumpie | hmm |
| 20:54.43 | Jumpie | if you have 2 wildcards, but one is more specific, the more specific wins right |
| 20:55.01 | M1s3ry | yes |
| 20:55.03 | Jumpie | ok now.im not getting the error that its trying to dial an internal extension |
| 20:55.06 | Jumpie | but my provider is rejecting it |
| 20:55.11 | ManxPower | Jumpie: as long as they are in the same context, the most specific extension should always win. |
| 20:55.12 | Jumpie | i wonder if it wants a 1 |
| 20:55.40 | Jumpie | http://jumpie.pastebin.org/28848 |
| 20:55.44 | ManxPower | Jumpie: Do you have any extensions or patterns that could match 702 |
| 20:55.46 | ManxPower | 703 |
| 20:55.49 | jeev | so there is no way to inrease the minimum for a dtmf duration |
| 20:55.56 | jeev | so there is no way to increase the minimum for a dtmf duration ? |
| 20:55.57 | Jumpie | nope |
| 20:56.15 | *** join/#asterisk mattva01 (n=matthew@c-69-143-130-97.hsd1.va.comcast.net) |
| 20:56.16 | M1s3ry | try it, but as ManxPower suggested, be sure that you have set the first priority for the _703 exten to 1 |
| 20:56.28 | Jumpie | i did |
| 20:56.30 | ManxPower | jeev: remember to specify your technology |
| 20:56.32 | M1s3ry | kk, making sure |
| 20:56.55 | Jumpie | http://jumpie.pastebin.org/28849 is what i have now |
| 20:56.57 | jeev | i'm just using the x-lite softphone, USB headset.. asterisNOW |
| 20:57.15 | ManxPower | Jumpie: your provider is rejecting the call. |
| 20:57.16 | mattva01 | quick question, is there anyway to get an asterisk dialplan to ring the pstn phones on the line its connected to (with an x100p) |
| 20:57.29 | Jumpie | ManxPower, yea...i thin its cause it doesnt like 1 not coming from it? |
| 20:57.34 | ManxPower | mattva01: no, that would be called FXS |
| 20:57.36 | Jumpie | cant i do a 1+ and then ther est? |
| 20:57.43 | ManxPower | Jumpie: Am I your provider? |
| 20:57.46 | Jumpie | that way if i dont dial 1, append 1 to the front |
| 20:57.57 | Jumpie | ManxPower, im not sure why its rejecting, but i know if i can send a 1 itll accept |
| 20:58.10 | ManxPower | then you know what to do. |
| 20:58.10 | Jumpie | i want to add a 1 to 10 digit calls , but only if the first 3 is 703 |
| 20:58.12 | Jumpie | k |
| 20:58.22 | mattva01 | ManxPower : thought so , now I can have the last laugh at my place of work :) |
| 20:58.34 | ManxPower | ~fxofxs |
| 20:58.35 | jbot | hmm... fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 20:58.35 | M1s3ry | Jumpie, is the callerid set correctly? sometimes providers reject if you don't have it right |
| 20:58.48 | ManxPower | mattthe 2nd port on the X100P is just wired directly to the pins on the first port. |
| 20:58.56 | ManxPower | think of it as a built-in splitter |
| 20:58.56 | Jumpie | M1s3ry, that exact same one works with the other plan |
| 20:59.11 | M1s3ry | what error is it giving you/ |
| 20:59.32 | Jumpie | exten => _1NXXNXXXXXX,1,Set(CallerID=17034364555) |
| 20:59.32 | Jumpie | exten => _1NXXNXXXXXX,n,Dial(${TRUNKOUT}/${EXTEN},20) |
| 20:59.37 | Jumpie | is what i have..and that works |
| 20:59.40 | Jumpie | im gettin a reject |
| 20:59.44 | Jumpie | but not why |
| 20:59.48 | Jumpie | on the _703 one |
| 20:59.57 | M1s3ry | in the CLI, what error does it give when rejecting? |
| 21:00.08 | Jumpie | http://jumpie.pastebin.org/28848 |
| 21:00.21 | Jumpie | er my bad |
| 21:00.23 | outtolunc | Set(CallerID=17034364555) is not valid |
| 21:00.45 | Jumpie | when i did setcallerid(number) it worked, but i got a deprecated error and it said to do set(callerid) |
| 21:00.56 | ManxPower | Jumpie: It did not. |
| 21:01.15 | outtolunc | yes with is Set(CALLERID(num)=7034364555) |
| 21:01.20 | outtolunc | er which |
| 21:01.38 | mknerd2008 | read() is passing to a extention |
| 21:01.49 | Jumpie | hmm i guess i interpreted it incorrectly |
| 21:02.11 | ManxPower | I can't handle all this newbieishness |
| 21:02.15 | *** part/#asterisk ManxPower (n=manxpowe@213.sub-75-203-228.myvzw.com) |
| 21:02.19 | Jumpie | hmm i wouldnt have known that because i had no errors in cli |
| 21:02.33 | outtolunc | jumpie just do as i said before |
| 21:02.55 | Jumpie | so Set(CallerID(num)=17034364555) |
| 21:03.14 | outtolunc | not the 1 |
| 21:03.30 | Jumpie | if i dont pass the 1, callwithus will put its own caller id |
| 21:03.33 | outtolunc | the callerid does not need nor want the 1 |
| 21:03.46 | outtolunc | the PHONE you dial needs the 1 |
| 21:03.49 | outtolunc | not the callerid |
| 21:03.56 | Jumpie | my bad |
| 21:03.59 | outtolunc | read your priv |
| 21:04.03 | outtolunc | and do it |
| 21:04.19 | outtolunc | remove that 703 crap you have |
| 21:05.14 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:06.27 | jeev | fark, wish you'd send me a priv with a fix to my issue. |
| 21:06.29 | jeev | :> |
| 21:11.10 | *** join/#asterisk talntid (n=t@66.208.251.170) |
| 21:15.06 | Jumpie | jeev still same prob man? |
| 21:15.32 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 21:16.02 | jeev | i'm having key press problems.. i have to lightly and quickly tap the number for remote IVR's to pick up, i can't press the buttons regularly.. |
| 21:16.25 | Jumpie | man one of my credit card companies is like that |
| 21:16.38 | Jumpie | form my home phone you have to press and hold forever (ok kinda opposite problem) |
| 21:16.40 | Jumpie | for it to register |
| 21:16.42 | Jumpie | but on cell phone no prob |
| 21:17.14 | jeev | :< |
| 21:17.17 | jeev | i dont know what's causing this |
| 21:17.36 | Jumpie | do you have some kinda global setting thats causing a very low tolerance? |
| 21:18.48 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 21:18.51 | [hC] | bludgeons polycom |
| 21:18.59 | [hC] | How do you still crash with hints on. how. |
| 21:19.10 | Jumpie | grenkubs |
| 21:19.12 | Jumpie | gremlins |
| 21:20.49 | *** join/#asterisk saftsack (n=saftsack@p4FC77E2B.dip.t-dialin.net) |
| 21:20.54 | Jumpie | ~book |
| 21:20.55 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 21:20.59 | Jumpie | i needed that url...thx |
| 21:21.52 | jeev | no i dont even know what to look for Jumpie. |
| 21:22.43 | Jumpie | holy crap i just got acrobat reader from adobe.com , 47mb file at 4700k/s |
| 21:22.45 | Jumpie | :) |
| 21:23.32 | jeev | heh |
| 21:23.36 | jeev | helppppppppppppp |
| 21:28.18 | Jumpie | ok gonna setup ring group |
| 21:28.44 | patrick-- | hey everyone. im trying to install some fax support to my asterisk machine. the perl script from http://www.voip-info.org/wiki/view/app_rxfax+mISDN+Asterisk+1.4 only gives errors |
| 21:29.55 | nny_1 | would a global variable in extensions.conf work in sip.conf or voicemail.conf? |
| 21:30.43 | Jumpie | doesnt that mean global...within that file? |
| 21:31.11 | nny_1 | Jumpie: i think so, was asking lol |
| 21:32.15 | nny_1 | i can define globals in sip.conf and voicemail.conf though can't i? |
| 21:38.39 | *** join/#asterisk error404 (n=error404@c174287.static.uniserve.ca) |
| 21:39.14 | *** join/#asterisk Darthclue (n=chatzill@fw149.northside.isd.tenet.edu) |
| 21:39.43 | Jumpie | hmm wtf is this? |
| 21:39.44 | Jumpie | exten => s-TORTURE,1,Goto(${ARG4},s,1) ; Callee chose to send this call to a telemarketer torture script. |
| 21:39.48 | Jumpie | that sounds enticing |
| 21:40.10 | Darthclue | it's exactly what the comment says it is |
| 21:40.28 | error404 | bashes head on desk |
| 21:40.43 | Jumpie | i know that |
| 21:40.44 | Jumpie | lol |
| 21:40.52 | Jumpie | im just curious as to whats entailed in a 'torture script' |
| 21:41.01 | Jumpie | like annoying audio files? or screeching/beeps, etc |
| 21:41.08 | *** join/#asterisk tobias (n=tobias@user-0ce2hpk.cable.mindspring.com) |
| 21:41.13 | error404 | what i want to do is FWD local dialin -> asterisk -> sip provider for cheap cellphone long distance, can someone provide some advice? |
| 21:42.08 | Jumpie | jeev im reading in the book |
| 21:42.15 | Jumpie | it says the [globals] is specifically for extensions.conf |
| 21:42.30 | Jumpie | however there are other things you can declare probably like in general for other files |
| 21:42.51 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:43.22 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 21:44.03 | jeev | hmm ? |
| 21:44.08 | jeev | what you speaking of? |
| 21:44.27 | Jumpie | sorry i meant to address nny_1 |
| 21:44.32 | jeev | fark! |
| 21:44.33 | jeev | i got excited. |
| 21:45.24 | Jumpie | sowwy |
| 21:46.18 | jeev | hey |
| 21:46.21 | jeev | i wonder if i could make a dialplan |
| 21:46.25 | jeev | to send the extension itself |
| 21:47.12 | jeev | can i make it call a number.. and send the buttons? sure i can |
| 21:47.41 | *** join/#asterisk ymonsalv1z (n=ymonsalv@200.27.102.130) |
| 21:48.31 | Jumpie | wait back up |
| 21:48.35 | Jumpie | what exactly are you trying to do |
| 21:49.09 | outtolunc | send buttons eh <G> |
| 21:49.19 | Jumpie | lol |
| 21:49.43 | outtolunc | scottie prepare app_teleporter |
| 21:49.57 | jeev | scottie doooooooooooo |
| 21:50.48 | jeev | outtolunc, what you grinning at |
| 21:52.01 | Jumpie | im tryin gto help understand jeev's problem outtolunc |
| 21:52.05 | outtolunc | i'm just waiting for the video of you yanking the button off the phone, and swallowing it.. then waiting to see of you fart the right freqs to cause the dtmf you want |
| 21:52.07 | ymonsalv1z | hi |
| 21:52.34 | Jumpie | he wants to dial an internal ext, to call to a number, pause 5 seconds, then send the dtmf tones, slightly spaced out, of the original int ext he dialed |
| 21:52.46 | jeev | i love farting |
| 21:52.55 | Jumpie | i just thought, during the dial application, you are at a standstill for the next line until that call is terminated |
| 21:53.00 | outtolunc | dial(tech/dev/xxxxxx w xxx) |
| 21:53.26 | outtolunc | core show application dial |
| 21:53.43 | ymonsalv1z | I installed asterisk, but I was unable to configure the PSTN is with a card tdm400 |
| 21:54.03 | ymonsalv1z | i from chile |
| 21:54.09 | jeev | k, outtolunc, reason i wanna do that is cause my key presses aren't working properly unless if i quickly tap the number, you know what i mean ? |
| 21:54.15 | jeev | so i just wanna automate it and have the system send it. |
| 21:55.35 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
| 21:55.38 | outtolunc | first i would try a diff softphone .. weren't you the one with xlite duration issue |
| 21:55.49 | Jumpie | yeah |
| 21:55.54 | jeev | yes i was |
| 21:56.07 | ymonsalv1z | The deviation from the PSTN to an annex (extension.conf) |
| 21:56.07 | outtolunc | try another softphone before you yank your hair out |
| 21:56.30 | jeev | let me try firefly |
| 21:57.49 | ymonsalv1z | i need help please |
| 21:58.07 | jeev | we all need help! |
| 21:58.21 | jeev | so should i expect this minor static to be gone once i move my box to the datacenter? |
| 21:58.27 | *** join/#asterisk trippss (n=ss@66.60.235.100) |
| 21:58.53 | ymonsalv1z | ok |
| 21:58.56 | ymonsalv1z | sorry |
| 21:59.24 | trippss | Q: if I'm using an external SIP gateway that is in turn connected to PRI trunk, and I want to configure the national NPI dialing plan from *, how would I do this? |
| 21:59.30 | [TK]D-Fender | jeev, static on what? |
| 21:59.39 | jeev | the calls outbound or inbound |
| 21:59.49 | [TK]D-Fender | jeev, on what interface? |
| 21:59.50 | jeev | right now the box is sitting in a 768k up network where there are quite a few users |
| 21:59.57 | error404 | man this application has the worst docs and steepest learning curve of anything i've seen |
| 22:00.15 | jeev | error404, it's rather complex. |
| 22:00.23 | jeev | that's why i'm greatful for people like fender. |
| 22:00.29 | [TK]D-Fender | tripps : You don't set NPI over SIP, thats a core setting on your gateway |
| 22:00.30 | jeev | jumpie and the others |
| 22:00.31 | jeev | ;) |
| 22:00.33 | error404 | nods |
| 22:00.39 | error404 | is there something simpler that can do what i want? |
| 22:00.45 | jeev | i'm gonna drop it in the cage, 100mbit up.. wondering if that should help |
| 22:00.45 | [TK]D-Fender | error404, ... |
| 22:00.46 | [TK]D-Fender | ~book |
| 22:00.48 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 22:00.48 | [TK]D-Fender | ~wikis |
| 22:00.49 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 22:00.56 | carrar | hahah |
| 22:00.58 | carrar | nice spam |
| 22:01.04 | carrar | (mailing list) |
| 22:01.20 | [TK]D-Fender | jeev, You've missed the question again... what INTERFACE are you calls on that have static on them? Exact harware and connections of BOTH ends please. |
| 22:01.42 | trippss | [TK]D-Fender: that's what I was afraid of . . . the mediant docs aren't clear about what each NPI manipulation table and context table actually does. I presume if I want national then E.164 NPI and national TON is what I want? |
| 22:01.56 | *** join/#asterisk b1shop (n=b1shop@c-76-16-229-8.hsd1.il.comcast.net) |
| 22:02.12 | [TK]D-Fender | tripps : that sounds familiar, but I'm not an expert. Give it a shot. |
| 22:02.32 | error404 | [TK]D-Fender i've read a fair bit of the available docs, but i'm still having trouble. this whole dialplan/extensions etc. architecture is very strange and counter-intuitive. |
| 22:02.52 | [TK]D-Fender | error404, Funny thing is its a whole lot simpler than you think.... |
| 22:03.04 | Jumpie | i think its just because it was written with developers in mind |
| 22:03.16 | [TK]D-Fender | error404, but I agree I could do a much better job documenting much of this myself the much of whats out there. |
| 22:03.18 | Jumpie | this isnt something i'd bundle up and give my mom for her birthday |
| 22:03.20 | error404 | i *am* a developer, but i know next to nothing about pbx, which is probably the problem |
| 22:03.24 | [TK]D-Fender | Something I intend to do soon. |
| 22:03.27 | jeev | hardware? you want the actual ethernet interface? it's just softphone > ethernet > SIP in datacenter belonging to my provider |
| 22:03.46 | jeev | what is another free softphone i can test with.. firefly wasn't it heh |
| 22:03.47 | trippss | [TK]D-Fender: i have tried a few combos - but it doesn't seem to "take". the SIP messages on the forums show npi=1;ton=0 within the Remote-Party-ID portion but mine never shows up anywhere . . . I'll keep picking away at it. . . |
| 22:04.04 | [TK]D-Fender | jeev, Static can only be due to your mic/soundcard, or your ITSP. Them you must replace, or your headset for you. |
| 22:04.16 | *** join/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 22:04.17 | outtolunc | noone, including your mom <G> should be editing extensions and conf files unless they know what they are doing <G> |
| 22:04.36 | Jumpie | hah |
| 22:04.46 | Jumpie | outtolunc, im a bit lost as wher to start for my incoming |
| 22:04.53 | outtolunc | you don't drop her in front of a callmanager setup and say fix it <G> |
| 22:04.55 | [TK]D-Fender | error404, Ask some specific questions here and there are plenty of people well versed in dialplan around to help |
| 22:05.01 | Jumpie | like, i just want my sip/1000 and sip/2000 to ring when people call my ID |
| 22:05.01 | Jumpie | DID |
| 22:05.07 | [TK]D-Fender | Myself, I'm about to head out though. |
| 22:05.21 | lanning | SPAMMED! from Asterisk-announce! |
| 22:05.21 | Jumpie | i called my did and got my asterisk demo thing :P bad juju |
| 22:05.42 | error404 | i guess what i'm trying to do is just not well covered by the wiki stuff out there. i'm trying to use asterisk to answer a FWD extension and provide a dialtone to another SIP provider. |
| 22:05.58 | outtolunc | for the did you set an exten for it.. and just dial(SIP/1000&SIP/2000) |
| 22:06.50 | jeev | so it must not be a bandwidth issue Fender? |
| 22:07.44 | [TK]D-Fender | jeev, No. Static is not bandwidth. If it was just broken I'd say yes, but actual static sound means someones analog gear is crap |
| 22:07.56 | [TK]D-Fender | ok, I'm off. back in a few hours. |
| 22:08.37 | outtolunc | static can from from any wiring also, example would be a headset for your softphone |
| 22:09.07 | outtolunc | which has nothing to do with the actual voip call |
| 22:09.16 | Jumpie | outtolunc, oh, i just create an extension that matches my DID? |
| 22:09.17 | Jumpie | duh |
| 22:09.33 | outtolunc | nods, in the context for which you inbound comes in on |
| 22:09.56 | outtolunc | exten yourdid,1,Dial(sip/1000&sip/2000) |
| 22:09.57 | Jumpie | but, isnt the idea of contexts, that somethin else references it |
| 22:10.00 | outtolunc | er => |
| 22:10.02 | Jumpie | right |
| 22:10.04 | Jumpie | hmm ok |
| 22:10.23 | outtolunc | contexts are just containers |
| 22:10.24 | Jumpie | thanks lol |
| 22:10.37 | Jumpie | so in reality its just my local specific preference to organize |
| 22:10.42 | husimon | Is there a way to set static ip and network settings in a cisco 7940? |
| 22:10.44 | Jumpie | books/sites may give suggestions, but in the end it doesnt matter |
| 22:10.54 | husimon | I wanted to use one at a remote network but I don't want to have to setup a dhcp and tftp server |
| 22:11.07 | outtolunc | you *could* have every last thing in a [default] context |
| 22:11.17 | outtolunc | you would be INSANE to do so |
| 22:11.20 | Jumpie | outtolunc, when specifying, i leave off the 1 again right :) |
| 22:11.22 | outtolunc | but you could <G> |
| 22:11.35 | Jumpie | ya..no thx heh |
| 22:11.49 | outtolunc | it is whatever your provider sends you |
| 22:12.05 | outtolunc | if that is 10 digits, then 10 digits.. if it is 4, then 4 |
| 22:12.19 | Jumpie | no..im talking about my did |
| 22:12.26 | outtolunc | so am i <G> |
| 22:12.30 | Jumpie | when im doing the exten => do i do 1703.... or 703xxxxxxx |
| 22:12.36 | husimon | Juggie, 1703 |
| 22:12.38 | outtolunc | that is outbound |
| 22:13.15 | husimon | Juggie, oh i see what you're saying |
| 22:13.22 | outtolunc | just dial your did from a cell phone and see how asterisk (watch the cli) sees it as |
| 22:13.31 | Jumpie | hmm...i bet it'd be easier to just setup a ring group and dial that ? |
| 22:13.32 | husimon | Juggie, for incoming I just put the extension, the digits that change in my did group |
| 22:13.33 | outtolunc | (you do have it pointed at your asterisk box right) |
| 22:13.36 | Jumpie | ok |
| 22:13.38 | husimon | ie 301, 302, 304 |
| 22:13.41 | Jumpie | yes |
| 22:14.16 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:14.23 | outtolunc | Jumpie: you can make it more complicated later, just get it working first <G> |
| 22:14.37 | Jumpie | outtolunc, all it shows is my incoming SIP/ doesnt show how the caller id comes across |
| 22:14.43 | Jumpie | ah..i have to check my provider logs |
| 22:14.52 | jeev | ok, i tried another softphone and i'm still having the same problem with the DTMF thing |
| 22:15.23 | Jumpie | lol dang jeev |
| 22:15.46 | Jumpie | yep...it prepends it with 1 |
| 22:15.58 | outtolunc | jeev, what codec |
| 22:16.58 | Jumpie | lol outtolunc argh its still going to demo haha |
| 22:17.25 | outtolunc | Jumpie: find and ; out that demo stuff |
| 22:17.29 | Jumpie | yeah |
| 22:17.39 | outtolunc | then you will see your error and not a flood of crap |
| 22:17.40 | Jumpie | i shoulda just copied the conf file toa bakup and started scratch after all |
| 22:18.08 | outtolunc | your backup is in asterisksource/configs |
| 22:18.22 | outtolunc | where it 'appeared from' in the first palce |
| 22:18.26 | Jumpie | if i just ; out [demo] will itt ake care of everything? |
| 22:18.30 | Jumpie | or do i have to ; each line :P |
| 22:18.41 | outtolunc | find include => demo |
| 22:18.45 | outtolunc | and comment it out |
| 22:18.46 | Jumpie | ah... |
| 22:18.50 | Jumpie | yep i know exactly where that is |
| 22:19.15 | Jumpie | was under [dfault] |
| 22:19.45 | outtolunc | remember that asterisk will 'fall back' to context default when you aren't configured correctly |
| 22:19.51 | Jumpie | [Apr 10 06:19:35] NOTICE[5785]: chan_sip.c:13952 handle_request_invite: Call from '545345523' to extension 's' rejected because extension not found. |
| 22:19.58 | outtolunc | so do a 'set verbose 3' from the CLI |
| 22:19.59 | Jumpie | i dont have that as an extension |
| 22:20.15 | Jumpie | but thats how my provider sends it to me i guess, lookin for that extension |
| 22:20.33 | outtolunc | because you didn't have the proper extension, it fell back to 's' |
| 22:20.40 | outtolunc | so, set verbose 3 |
| 22:20.46 | outtolunc | and do it again |
| 22:20.58 | Jumpie | at my provider, the termination is SIP/545345523 (im just a sip peer) |
| 22:21.01 | Jumpie | ok lemme do that |
| 22:21.26 | Jumpie | same msg..i think thats what i had it on first time |
| 22:21.28 | outtolunc | most times |
| 22:21.32 | Jumpie | [Apr 10 06:21:21] NOTICE[5785]: chan_sip.c:13952 handle_request_invite: Call from '545345523' to extension 's' rejected because extension not found. |
| 22:21.33 | outtolunc | er.. what i do |
| 22:21.47 | Jumpie | do i have to make a bogus extension, and forward that to sip/1000&sip/2000? |
| 22:21.51 | outtolunc | in sip.conf make 2 entries a user and peer for that provider |
| 22:22.04 | outtolunc | in each of those a different context |
| 22:22.18 | outtolunc | so you now control both directions |
| 22:22.34 | outtolunc | (and stay the hell out of default) <G> |
| 22:22.48 | susinths | have someone here heard about Soekris Net4801? |
| 22:23.01 | Jumpie | outtolunc, ya,, so far in sip.conf i just have [callwithus] for outbound |
| 22:23.04 | susinths | i need to know how much it costs aprox? |
| 22:23.06 | Jumpie | i mean wouldnt i just be creating a duplicate? |
| 22:23.21 | Jumpie | or are you referring to one [545345523] blah blah |
| 22:23.30 | outtolunc | jumpie.. the example would be |
| 22:24.08 | outtolunc | [CWU] type=user context=cwu-in |
| 22:24.26 | outtolunc | [CWU] type=peer context=cwu-out |
| 22:24.38 | outtolunc | understand? |
| 22:24.47 | Jumpie | yea |
| 22:24.54 | Jumpie | callwithus says to do type=friend though |
| 22:24.58 | Jumpie | for outbound |
| 22:24.59 | outtolunc | so |
| 22:25.06 | Jumpie | k |
| 22:25.14 | outtolunc | that is because most people can't figure out user/peer correctly |
| 22:25.19 | Jumpie | lol |
| 22:25.21 | Jumpie | ah ok |
| 22:25.58 | Jumpie | but those are both within [callwithus] or i create two? |
| 22:26.12 | outtolunc | 2 |
| 22:26.16 | Jumpie | k |
| 22:26.22 | outtolunc | read the example i just did again |
| 22:26.59 | Jumpie | yes, so i need to change up my extensions.conf as well to reflect |
| 22:28.05 | outtolunc | you would ADD to extensions conf a [cwu-in] and [cwu-out] |
| 22:28.23 | Jumpie | exactly thats what i meant :) |
| 22:28.36 | outtolunc | and copy/move your previous ..NXX and 1NXX stuff to it |
| 22:28.41 | Jumpie | right |
| 22:29.01 | outtolunc | or put those in longdistance and just include them to the cwu-out |
| 22:29.13 | outtolunc | now |
| 22:29.44 | outtolunc | ANYTHING that hits [cwu-in] you know came inbound from that provider |
| 22:30.03 | outtolunc | and you can create dialplan logic reflecting that |
| 22:30.26 | Jumpie | aside from the context and type, all the other stuff like host/secret, etc all stays the same |
| 22:30.32 | outtolunc | yes |
| 22:30.36 | Jumpie | k |
| 22:30.38 | Jumpie | almost done :) |
| 22:32.05 | outtolunc | brb, nature |
| 22:34.39 | *** join/#asterisk korihor (n=humberto@190.78.209.202) |
| 22:40.07 | *** join/#asterisk mindCrime (n=chatzill@nat/redhat/x-5681a27fd256023e) |
| 22:40.40 | korihor | hi to everyone |
| 22:41.14 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:41.15 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:41.34 | korihor | the new zaptel 1.2.25 break the option "answeronpolarityswitch" in zapata.conf? |
| 22:42.22 | korihor | i'm using the driver wctdm |
| 22:43.41 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:45.35 | *** join/#asterisk tobias (n=tobias@user-0ce2hpk.cable.mindspring.com) |
| 22:52.02 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 22:53.43 | tzafrir | korihor, hmm... how exactly do you see that? |
| 23:00.24 | jeev | outtolunc, im' using ulaw i think |
| 23:02.08 | korihor | on 1.2.24 the polarity reverse answer worked, but on 1.2.25 it stopped working. i believe that's because the changes on the wctdm.c file related to the new structure fxo |
| 23:02.23 | *** part/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 23:04.52 | korihor | maybe on the line 942 the wctdm.c file? |
| 23:04.55 | *** join/#asterisk JayTee52 (n=jforde05@c-69-243-161-112.hsd1.in.comcast.net) |
| 23:06.12 | *** part/#asterisk LemensTS (i=CustomGT@adsl-70-238-131-138.dsl.stlsmo.sbcglobal.net) |
| 23:08.02 | *** join/#asterisk sack (n=sack@247.Red-81-32-161.dynamicIP.rima-tde.net) |
| 23:11.34 | *** join/#asterisk jsmith (n=jsmith@72.21.36.138) |
| 23:11.34 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 23:12.10 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 23:13.41 | *** join/#asterisk RoyK (n=roy@ip-210-6-149-91.dialup.ice.no) |
| 23:21.50 | *** join/#asterisk MDK2MDK (n=NanoTec@41.249.112.174) |
| 23:23.04 | MDK2MDK | hello |
| 23:23.16 | *** join/#asterisk s0lid (n=s0lid@210.213.198.61) |
| 23:23.31 | MDK2MDK | can some one tel me if we can use asterisk as a solution for call centers slz ?? |
| 23:23.33 | MDK2MDK | pls |
| 23:26.05 | denon | yes |
| 23:26.14 | denon | but if you have to ask .. odds are you're not quite there :) |
| 23:27.14 | MDK2MDK | and were can i ask ?? because i'm new to asterisk |
| 23:29.33 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 23:29.33 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:29.47 | _ShrikE | MDK2MDK: there is a really good book that you can read that will at the least point you in the right direction. |
| 23:29.54 | _ShrikE | ~book |
| 23:29.54 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 23:31.15 | MDK2MDK | ok thx so much :) |
| 23:32.19 | korihor | tzafrir: did you saw what i wrote? do you know anything about that? |
| 23:36.14 | *** part/#asterisk RoyK (n=roy@ip-210-6-149-91.dialup.ice.no) |
| 23:37.04 | jsmith | korihor: Apparently he didn't... have you opened a bug report? |
| 23:38.30 | *** join/#asterisk WindBack (n=jorge@host43.200-117-119.telecom.net.ar) |
| 23:39.13 | *** join/#asterisk WindBack (n=jorge@host43.200-117-119.telecom.net.ar) |
| 23:40.47 | korihor | jsmith: i haven't. i wanted to see if anybody knew something about it |
| 23:41.11 | jsmith | korihor: No clue... your best bet is to open a bug report |
| 23:41.41 | korihor | jsmith: i know you are lds like me :) |
| 23:42.28 | jsmith | korihor: I guessed that by your nick :-) |
| 23:43.25 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 23:44.12 | bitzero | are we being attacked by mormons? |
| 23:45.10 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 23:45.21 | drmessano | huh? |
| 23:45.27 | korihor | jsmith: i will open the bug report after i do some more test |
| 23:46.07 | jsmith | bitzero: Well, I'm not sure if "attacked" is the right word... some might say "invaded" or "infiltrated" :-p |
| 23:46.23 | bitzero | feels assaulted. |
| 23:46.39 | jsmith | bitzero: But yes, there are Mormons all over the place :-) |
| 23:47.09 | *** part/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 23:48.30 | Jumpie | ahah!! |
| 23:48.34 | Jumpie | figured out something interesting |
| 23:48.36 | korihor | jsmith: i'm from venezuela, i know that you worked here for some time. is that right? |
| 23:48.53 | Jumpie | is lds too |
| 23:49.08 | jsmith | korihor: I spent about three weeks in Venezuela about ten years ago |
| 23:49.35 | Jumpie | i cant get my box ti look outside of [default] for inbound calls |
| 23:49.41 | Jumpie | oh..nm |
| 23:49.56 | jsmith | Jumpie: What type of calls? Analog? SIP? IAX? |
| 23:50.49 | Jumpie | well my stupid callwithus example expects you to defer everything to [default] |
| 23:50.56 | Jumpie | i ha changed it and had to set context= |
| 23:51.18 | Jumpie | im actually learning by looking at the logs |
| 23:51.24 | Jumpie | glances at drmessano |
| 23:51.27 | Jumpie | :P |
| 23:52.34 | jeev | ahhhhhhhhh! |
| 23:53.02 | jeev | i'm still farked Jumpie |
| 23:53.11 | Jumpie | jeev :( |
| 23:53.18 | Jumpie | did you manage to pass the dtmf?" |
| 23:53.21 | jeev | nope |
| 23:53.22 | Jumpie | with delay? |
| 23:53.24 | Jumpie | hmm |
| 23:53.54 | jeev | yea, really sucks man |
| 23:53.59 | jeev | it's a nusance |
| 23:54.03 | jeev | or however you spell it |
| 23:54.07 | MDK2MDK | can some one tel me , to build a call center with asterisk what kind on hardware do i need ?? |
| 23:54.11 | jsmith | jeev: What are you trying to do? |
| 23:54.26 | jsmith | MDK2MDK: Some Asterisk boxes, some phones, some T1/E1 cards, and a whole lot of learning |
| 23:54.34 | jeev | jsmith, i call a number, their IVR answers.. i have to press the number (extension) really fast for it to pick it up.. if i press it like a regular tone, their IVR wont pick it up |
| 23:55.02 | jsmith | jeev: Version of Asterisk? Is your call going out over SIP or IAX or analog or T1/E1? |
| 23:55.12 | jeev | SIP, asteriskNOW latest. |
| 23:55.23 | jeev | 1.4.18.1 |
| 23:55.24 | jsmith | MDK2MDK: Learn Asterisk first, then start small, and keep learning until you have the power to build a call center |
| 23:55.54 | jsmith | jeev: And your SIP provider... any guess if they're still using Asterisk 1.2? If so, you may have to set rfc2833compensate=yes in sip.conf |
| 23:55.55 | MDK2MDK | thx jsmith :), but asterisk box mean a normal server or computer on a special things ? |
| 23:56.03 | jsmith | jeev: (Assuming you're using rfc2833 for your dtmfmode) |
| 23:56.18 | jeev | yes, i'm using rfc2833, i could ask my provider, i think he's up to date though, is tehre a method to check remotely ? |
| 23:56.22 | jsmith | MDK2MDK: A normal everyday server, running Linux and Asterisk |
| 23:56.27 | jsmith | jeev: Nope. |
| 23:56.45 | MDK2MDK | ok thx soooooo much i'll start right now :) |
| 23:56.57 | jeev | that did not help after a reload jsmith. |
| 23:57.06 | drmessano | glances up in the air |
| 23:57.12 | jsmith | jeev: Which codec are you using? |
| 23:57.42 | jeev | i think im' using ulaw |
| 23:58.37 | jsmith | Then you could try using inband DTMF... I usually don't recommend it, but in your case, it may just solve your issues |
| 23:58.45 | jeev | i have already tried that.. |
| 23:59.11 | Jumpie | jeev or did you? |
| 23:59.17 | jeev | i already did |
| 23:59.51 | jsmith | runs out of ideas |