| 00:00.15 | terrapen | me too. i just found out about it today... the voipsupply guy recommended it to me |
| 00:00.21 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 00:00.25 | justinu | tarrapen: i think it's a kick ass solution, and I'm gonna set it up as soon as I find a customer who needs it. |
| 00:00.54 | terrapen | justinu, cool, add your findings to the wiki when you do |
| 00:01.58 | [av]bani | terrapen: redfone has no echo canceller afaik |
| 00:02.10 | glm2k | aahh, that's gonna hurt |
| 00:02.38 | terrapen | that's true |
| 00:02.41 | Abydos313 | [av]bani so is it designed to be used without echo cancellation or another device is needed? |
| 00:02.45 | terrapen | i wonder how much that will matter |
| 00:02.50 | glm2k | terrapen: a lot |
| 00:03.32 | terrapen | how else would you share one PRI line between two redundant asterisk servers? |
| 00:04.09 | glm2k | people will tolerate pops and clicks to a certain extent, but from experience, they will not tolerate echo. |
| 00:04.10 | redax | is there anybody using chan_capi-cm ? |
| 00:04.32 | De_Mon | I bought a X101p clone (tiger jet) fxo card, but its not loading with the wcfxo module.. suggestions? |
| 00:04.51 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:05.10 | terrapen | where is the echo coming from? |
| 00:05.13 | FuriousGeorge | i know this is gonna sound stupid but can anyone recommend me a good one line analog phone |
| 00:05.17 | terrapen | and can the asterisk server not take care of it? |
| 00:05.18 | [av]bani | Abydos313: i guess you're supposed to use external ECs, eg tellabs |
| 00:05.21 | justinu | echo comes from the PSTN |
| 00:05.26 | FuriousGeorge | dont say slimline |
| 00:05.39 | justinu | or poor acoustic coupling on shitty phones |
| 00:05.56 | FuriousGeorge | or untwisting your wires too much |
| 00:06.00 | [av]bani | justinu: speakerphones have to deal with EC too |
| 00:06.03 | terrapen | why don't we have echo with our current crappy ass 24-port Adtran FXS? |
| 00:06.10 | FuriousGeorge | what justinu said |
| 00:06.42 | terrapen | rather, we don't have any echo problems with that... |
| 00:06.43 | justinu | there's two kinds of echo, hybrid echo (2-4 wire conversion at the far end) |
| 00:06.49 | [av]bani | terrapen: you do, you just dont hear it because the delay is low |
| 00:06.50 | Gennaro | i done sip-context for 2 phone and works |
| 00:06.53 | justinu | and acoustic echo (the phone can hear it's own speaker) |
| 00:07.10 | Gennaro | so i ask how can i do to let ring 2 or more phone?!? |
| 00:07.16 | terrapen | ok, i just e-mailed the Redfone folks asking about echo cancellation |
| 00:07.25 | [av]bani | :) |
| 00:07.33 | Gennaro | how can i do to do a queue?!? |
| 00:07.40 | Abydos313 | [av]bani thx |
| 00:07.41 | [av]bani | they dont mention EC anywhere on their pages, in their product announcements, press releases, or datasheets |
| 00:08.04 | justinu | someone here theorized that redphones were just running asterisk inside them |
| 00:08.10 | justinu | i dunno about that |
| 00:08.20 | justinu | be interesting to know more about how they work tho |
| 00:09.04 | asterisk99 | russellb: Are you using res_perl or Asterisk::AGI ? |
| 00:11.00 | russellb | asterisk99: neither, heh |
| 00:11.11 | [av]bani | justinu: they are just PCs with quad t1 pci card, in a red box |
| 00:11.28 | [av]bani | justinu: and theyd be silly not to use asterisk |
| 00:11.36 | russellb | asterisk99: I just work on Asterisk code. I don't actually "use" it all that much. :) |
| 00:11.58 | russellb | I pay attention to what you guys are saying and doing to get a feel for what needs to be done ... |
| 00:12.18 | terrapen | avbani, how do you know this? |
| 00:12.26 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 00:13.45 | *** join/#asterisk propagandhi (n=opera@CPE-144-131-132-107.nsw.bigpond.net.au) |
| 00:13.59 | propagandhi | can you send SMS messages to cell phones with asterisk |
| 00:14.13 | *** join/#asterisk fugitivo (n=ajf@201.255.176.38) |
| 00:14.16 | terrapen | prop: voip-info.org |
| 00:14.45 | [TK]D-Fender | propagandhi : no but I'm sure there are a dozen or so other tools yuo can trigger from it just as easily.... |
| 00:15.44 | Gennaro | i tried with & and works |
| 00:16.13 | Gennaro | how can i do to use an n number without write a mass of codes? |
| 00:16.23 | propagandhi | [TK]D-Fender: yeah I can trigger it to send through an SMS gateway, but I was wondering if there was an inbuilt way of doing it direct from asterisk |
| 00:16.42 | terrapen | go to that site |
| 00:17.26 | Gennaro | exten => 110,1,Dial(SIP/100&SIP/101,20) can i write it in 2 line!? |
| 00:21.02 | [TK]D-Fender | propagandhi : nope... this isn't a "standard" thing..... |
| 00:21.34 | *** join/#asterisk forao (n=fasdfasd@pool-138-89-178-72.mad.east.verizon.net) |
| 00:21.51 | propagandhi | [TK]D-Fender: thanks at least I know not to waste any time on it then |
| 00:22.42 | *** join/#asterisk backblue (n=moo@87-196-32-165.net.novis.pt) |
| 00:22.44 | SkramX | is there a special dial-option for it to go to vm if it times out. |
| 00:23.09 | [av]bani | [TK]D-Fender: i talked to atacomm. its still $199 |
| 00:23.12 | SkramX | if i put the voicemail command after the dial, then after I hangup, and the user is left on the line by him/herself, it goes to VM. |
| 00:23.33 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 00:23.53 | [av]bani | [TK]D-Fender: they said directly, the front page price is correct |
| 00:24.46 | SkramX | ? |
| 00:25.31 | *** join/#asterisk rene- (n=rene-@201.127.24.218) |
| 00:27.50 | [TK]D-Fender | [av]bani : Any explanation why it changed on the item detail level then? It WAS 199$ there before... |
| 00:31.24 | [av]bani | [TK]D-Fender: they are 'out of stock', apparently the price is raised when that happens |
| 00:31.32 | [av]bani | [TK]D-Fender: but i asked them if it was $199 and they said yes |
| 00:34.28 | Goral | if i'm running an asterisk box at home and i have two people outside my local network calling each other do i need the bandwidth to support the conversation or does asterisk point it to each other like a peer2peer network? |
| 00:35.24 | [TK]D-Fender | Goral : If they are behind NAT, that would be painful to attempt. Just use a lighter codec |
| 00:35.41 | docelmo | I know this is the wrong channel but any knowegable SER people in here? |
| 00:35.56 | docelmo | Need a little help. If you know C and Asterisk Dev that would be killer also |
| 00:37.16 | Goral | [TK]D-Fender : i'm guessing i will learn that in Asterisk book |
| 00:37.41 | [TK]D-Fender | Goral : possibly :) What would each side be using as a phone? |
| 00:37.43 | r_evolution | i. |
| 00:37.44 | r_evolution | swear. |
| 00:37.45 | r_evolution | im. |
| 00:37.46 | r_evolution | going. |
| 00:37.46 | r_evolution | to |
| 00:37.48 | r_evolution | kill. |
| 00:37.48 | r_evolution | this. |
| 00:37.50 | r_evolution | guy. |
| 00:38.00 | r_evolution | sorry had to get some frustration out all better now... how goes it TK? |
| 00:38.18 | [TK]D-Fender | NTB.... go bury that body now ;) |
| 00:38.34 | Hmmhesays | asterisk book |
| 00:38.36 | Hmmhesays | good luck |
| 00:38.39 | r_evolution | no kidding man. |
| 00:38.48 | r_evolution | So here's the issue... justin and i were talking about it earlier... |
| 00:39.00 | r_evolution | the * switch i've built does NOT recognize DTMF tones... from most phones |
| 00:39.11 | r_evolution | but only when coming through one of the providers... |
| 00:39.18 | r_evolution | when i use the old provider? works fine. |
| 00:39.28 | r_evolution | When I use the * as teh origination so i control the dtmf method |
| 00:39.30 | r_evolution | works fine. |
| 00:39.39 | r_evolution | going from PSTN to the new provider to the * box |
| 00:39.41 | r_evolution | NO JOY |
| 00:40.17 | r_evolution | and the guy keeps getting me to do this and that on this end... right now he's wanting me to set dtmf to info |
| 00:40.23 | r_evolution | this is after having me set it to inband |
| 00:40.31 | [TK]D-Fender | What code are you using? |
| 00:40.40 | *** join/#asterisk kratzers (n=kratzers@65.119.216.4) |
| 00:40.40 | [TK]D-Fender | codec? |
| 00:40.43 | r_evolution | (justin suggested earleir that they want to use inband anyway based on eval of the debug) |
| 00:40.46 | kratzers | lear |
| 00:40.46 | r_evolution | right now? just ulaw |
| 00:40.49 | Goral | r_evolution : i hope that wasn't directed @ me... |
| 00:41.00 | r_evolution | when it goes live we're switching to 729 |
| 00:41.03 | [TK]D-Fender | r_evolution : inband is believable. |
| 00:41.25 | r_evolution | yeah but inband isn't going to work when we go to 729 |
| 00:41.29 | r_evolution | or am i mistaken in that? |
| 00:41.33 | [TK]D-Fender | 729 warps things too much... |
| 00:41.37 | r_evolution | (p.s. i already tried both info and inband... no joy) |
| 00:41.52 | rene- | rfcXXXX? |
| 00:41.55 | r_evolution | yeah... so basically it *MUST* be rfc2833... |
| 00:42.03 | r_evolution | and im like just FIX IT ON YOUR END. |
| 00:42.04 | r_evolution | PLZ! |
| 00:42.15 | [TK]D-Fender | r_evolution : pastebin your sip.conf.... |
| 00:42.34 | r_evolution | no sip.conf TK |
| 00:42.37 | r_evolution | realtime |
| 00:42.43 | [TK]D-Fender | :/ |
| 00:42.47 | r_evolution | i can replicate it without a problem |
| 00:42.51 | r_evolution | but trust the problem isn't in this box |
| 00:42.59 | r_evolution | like i said... when i use the old switch |
| 00:43.01 | r_evolution | DTMF works fine |
| 00:43.11 | r_evolution | just with this new provider |
| 00:43.13 | r_evolution | no joy. |
| 00:43.32 | r_evolution | when I start a call from the * box (i.e. I call to the DID from the SIP and it routes back into *) |
| 00:43.35 | r_evolution | it works fine |
| 00:43.39 | r_evolution | if I use rfc2833... |
| 00:43.50 | r_evolution | I was speaking with justin earlier... and he pointed out that they are apparently trying to use inband... |
| 00:44.04 | r_evolution | hence my belief that the problem is lying on their end |
| 00:44.09 | [TK]D-Fender | r_evolution : pastebin a dump of your SIP data |
| 00:45.06 | r_evolution | here i'll show you the one earlier... let me find it |
| 00:46.33 | *** join/#asterisk Eitch (i=[U2FsdGV@unaffiliated/eitch) |
| 00:46.58 | *** join/#asterisk delmar (n=Delmar@203-114-178-231.inspire.net.nz) |
| 00:47.07 | *** part/#asterisk kratzers (n=kratzers@65.119.216.4) |
| 00:47.33 | r_evolution | TK get that? |
| 00:49.40 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 00:49.53 | [TK]D-Fender | eek |
| 00:49.53 | [av]bani | haha now that i have a 7970, Qwell will never be seen in this channel again |
| 00:50.25 | [av]bani | =) |
| 00:51.05 | r_evolution | ? |
| 00:52.05 | [TK]D-Fender | r_evolution : A bit much for me tonight..... |
| 00:52.27 | *** join/#asterisk AAH_USER (n=none@c-67-186-207-234.hsd1.ut.comcast.net) |
| 00:52.42 | r_evolution | trust me homie... the problem isn't on my end :) |
| 00:52.48 | r_evolution | but thanks for the offer |
| 00:52.54 | r_evolution | this is one of the 3 i've been on all day |
| 00:54.34 | *** join/#asterisk nanogeek (n=nanogeek@207.244.10.65) |
| 00:55.18 | r_evolution | the other two have been solved |
| 00:55.19 | r_evolution | this one... |
| 00:55.20 | r_evolution | :( |
| 01:02.51 | nanogeek | Does anyone know how to forward inbound CID/CPN info to the CID of an off-net forwarded number? |
| 01:03.33 | *** join/#asterisk bkw_ (n=bkw_@ppp-70-128-118-15.dsl.tulsok.swbell.net) |
| 01:04.49 | syzygybsd | does anyone know of a web based sip phone? |
| 01:05.08 | r_evolution | im not sure nano... but perhaps trust rpid = yes? |
| 01:05.14 | asterisk99 | russellb: Asterisk::AGI is cool - I like this |
| 01:05.43 | asterisk99 | syzygybsd: XTen Lite |
| 01:06.01 | asterisk99 | syzygybsd: I know what a syzygy is |
| 01:06.03 | syzygybsd | that isn't web based and reqires an install |
| 01:06.16 | syzygybsd | it is what i use now |
| 01:06.33 | asterisk99 | syzygybsd: You are correct --- I should have read your question better |
| 01:07.03 | syzygybsd | though i am suprised you know what a syzygy is... |
| 01:07.50 | asterisk99 | syzygybsd: An alignment of planets; sexual union of two gametes; plus about 15 other definitions |
| 01:08.13 | nanogeek | I haven't seen trustrpid before. What does it do? |
| 01:09.44 | r_evolution | eh from my guessing (i havent used it... just had it mentioned) |
| 01:09.49 | r_evolution | it would trust the remote peer id |
| 01:10.12 | russellb | asterisk99: I bet. The guy that wrote that is a great programmer. He used to contribute to Asterisk development long before I was around. |
| 01:10.23 | r_evolution | 'or' |
| 01:10.37 | r_evolution | you could just use callerid=""<NUMBERGOESHERE> |
| 01:10.42 | r_evolution | and the * box will provide whatever ID |
| 01:11.07 | r_evolution | the Trust RPID comes if you look in your * box |
| 01:11.20 | r_evolution | and enter sip show peer <SIPID> at the CLI |
| 01:15.42 | nanogeek | I don't think that will quite do it. I am sitting on a 10 channel PRI. I am using * to forward incoming calls to off-net phones (like a cell phone or home PSTN phone.) I am not sure that a SIP configuration option will do it. |
| 01:16.52 | nanogeek | In fact in some cases I am forwarding calls to multiple destinations (for first come first serve call pick up). |
| 01:17.33 | nanogeek | I've tried the ANI Spoof AGL without success. |
| 01:18.09 | r_evolution | then you should be able to just fwd the caller id via zaptel yes? |
| 01:18.35 | nanogeek | You would think. I cant seem to find any docs on it though. |
| 01:18.43 | r_evolution | no no i mean |
| 01:19.04 | r_evolution | in the zapata I thought there was a callerid option |
| 01:19.54 | r_evolution | callerid=asreceived |
| 01:20.19 | nanogeek | its configured |
| 01:20.36 | r_evolution | When you call into * are you receiving the caller properly? |
| 01:20.53 | nanogeek | It appears so. |
| 01:21.15 | r_evolution | (P.s. i don't actually use any PRI lines... all my calls are going out over SIP to a provider ... so i'm throwing knives in the dark) |
| 01:22.06 | nanogeek | <PROTECTED> |
| 01:22.06 | nanogeek | <PROTECTED> |
| 01:22.06 | nanogeek | <PROTECTED> |
| 01:22.06 | nanogeek | <PROTECTED> |
| 01:22.06 | nanogeek | <PROTECTED> |
| 01:22.08 | r_evolution | I'm considering putting in a 24line card just for backup... but that's down the road a bit |
| 01:22.41 | nanogeek | So caller id is detected in * |
| 01:23.16 | *** join/#asterisk sofh (i=realvir@27-150-254-84.skylogicnet.it) |
| 01:23.18 | r_evolution | you sure? that looks like part of your dial-plan setting the caller ID |
| 01:23.27 | nanogeek | All the LEC's will want your left nut to install a PRI. |
| 01:23.49 | r_evolution | exactly why I'm not exactly pressing the MUST HAVE ONE FOR BACKUP issue |
| 01:24.04 | sofh | hi all! |
| 01:24.11 | r_evolution | someone's excited! |
| 01:24.13 | r_evolution | ;) |
| 01:24.24 | sofh | i've some issue regading codecs in asterisk |
| 01:25.11 | r_evolution | nano... that really looks like your callerID is being set in your dial plan... but perhaps I'm wrong? |
| 01:25.20 | r_evolution | I only used a TDM for perhaps two weeks and changed the ID a bit... |
| 01:25.30 | sofh | g729 is consuming arround 31kbps whereas ive been informed tht it shoud not go more then 10kbit |
| 01:25.45 | nanogeek | they lie! |
| 01:25.48 | r_evolution | hey sofh is that counting the up and down? |
| 01:26.01 | r_evolution | i.e. |
| 01:26.03 | r_evolution | incoming and outgoing |
| 01:26.17 | r_evolution | sorta like ulaw is only supposed to be like 60?70? |
| 01:26.25 | sofh | 31 up and 31 down on sip with g729 :$ |
| 01:26.27 | mover | any codec guru here? |
| 01:26.29 | r_evolution | but with overhad and all that shite it ends up noticeably more |
| 01:26.30 | nanogeek | bits or bytes? |
| 01:26.41 | r_evolution | true. the kb implies bits... |
| 01:26.45 | sofh | kilo bits..not bytes |
| 01:26.50 | mover | my g726-32 sounds like an alien |
| 01:26.58 | nanogeek | unless you have fat fingers... |
| 01:27.08 | r_evolution | and isn't it supposed to be 8 - 10 kB? |
| 01:27.11 | nanogeek | digitizing |
| 01:27.14 | r_evolution | 8 bits per byte etc |
| 01:27.27 | r_evolution | or isit just 8 - 10 kb? |
| 01:27.29 | r_evolution | O_o |
| 01:27.33 | r_evolution | damn you conversions! |
| 01:27.36 | sofh | so i want to either asterisk consumes so much bw or it is normal what i am facing ? |
| 01:28.06 | nanogeek | depends on whether your on broadband or a dial up... |
| 01:28.09 | sofh | not according to quintum and cisco gw..they are consuming 10 up and 10 down with g729 |
| 01:28.10 | r_evolution | hey nano... how are you passing the calls back out? are they coming in pri and going out pri? nothing in between |
| 01:28.34 | nanogeek | yes, in and out pri. exten => 1,6,Dial(${JGG}&${JGGHM}&${JGGCELL}&${JGGWIFI},30,Ttro) |
| 01:29.04 | sofh | whereas my box is eating 31upk and 31k down :< |
| 01:29.08 | r_evolution | What all is coming before the dial in teh dialplan? |
| 01:29.34 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 01:29.55 | r_evolution | Executing SetCallerID("Zap/1-1 (which appears to be part of the dial-plan ) would be setting the caller ID |
| 01:30.00 | nanogeek | exten => 1,1,NoOp(${CALLERID}) |
| 01:30.00 | nanogeek | exten => 1,2,SetCallerID(${CALLERID}) |
| 01:30.00 | nanogeek | exten => 1,3,Playback(one-moment-please) |
| 01:30.00 | nanogeek | exten => 1,4,GotoIfTime(07:00-21:00|mon-fri|*|*?6) |
| 01:30.00 | nanogeek | exten => 1,5,Goto(7) |
| 01:30.02 | r_evolution | do you get any caller ID when you fwd to a number? |
| 01:30.02 | nanogeek | exten => 1,6,Dial(${JGG}&${JGGHM}&${JGGCELL}&${JGGWIFI},30,Ttro) |
| 01:30.05 | nanogeek | exten => 1,7,Macro(vmessage,${JGGVM}) |
| 01:30.07 | r_evolution | ahh |
| 01:30.08 | nanogeek | exten => 1,8,Hangup |
| 01:30.36 | r_evolution | Is any caller ID being passed? |
| 01:30.41 | r_evolution | or is it just coming through as unknown? |
| 01:31.02 | sofh | r_evaolution! any idea about bw usage by asterisk ? |
| 01:31.04 | r_evolution | that's odd sofh... :( |
| 01:31.18 | nanogeek | that assumes all outgoing on the same port as opposed to any available ports? |
| 01:31.28 | *** join/#asterisk Andre3w_ (n=andrew@stjhnf0122w-142162051236.pppoe-dynamic.nl.aliant.net) |
| 01:31.32 | sofh | or is there any QoS or some thing lke that to get the max packets compression ? |
| 01:31.35 | r_evolution | no sir. I am not a codec hacker... |
| 01:31.46 | sofh | :) |
| 01:31.59 | r_evolution | yeah... just try straight up passing it directly to the number... and see if ANY caller ID is passed |
| 01:32.05 | r_evolution | without worrying about moving through a dialplan |
| 01:32.41 | sofh | but you can atleast let me know how much its consuming at your side , so that i can ensure my usage is normal and everybody have the same |
| 01:33.07 | sofh | but if your bw usage is less then mine..then offcourse i am doing some mistake or missing something |
| 01:33.24 | r_evolution | I've not installed the g729 licenses yet homie... I'm working out all the kinks before i move into production with this one |
| 01:33.41 | sofh | ok then may be u will be doing gsm ? |
| 01:33.56 | r_evolution | nah using ulaw right now |
| 01:34.00 | nanogeek | I'm not exactly sure how to approach that. |
| 01:34.20 | sofh | ok...ulaw shud consume 64kbit/s for a channel..as far as i know |
| 01:34.48 | sofh | but its consuming around 350kbit/s in my case :$ |
| 01:35.01 | r_evolution | HOLY SHIT |
| 01:35.08 | techie | amen. |
| 01:35.28 | r_evolution | nano... try commenting everything else out... and change to 1,1,Dial etc |
| 01:35.29 | sofh | now you get that i am realy in a pain :( |
| 01:35.37 | r_evolution | dude... |
| 01:35.39 | r_evolution | what the HELL?! |
| 01:35.49 | r_evolution | sorry you just absolutely floored me on that one sofh |
| 01:36.17 | sofh | :< |
| 01:36.20 | r_evolution | and why are you NoOp'ing? |
| 01:36.21 | nanogeek | give me a sec to config and test. |
| 01:36.27 | r_evolution | NoOp = No Operation |
| 01:37.16 | r_evolution | wow sofh... maybe you should look at what's really going on there man... |
| 01:37.31 | r_evolution | Are you using a softphone? a hardphone? an ATA? |
| 01:37.40 | sofh | tried both |
| 01:37.49 | r_evolution | both? there's three options... |
| 01:37.53 | r_evolution | what are you using right now? |
| 01:38.20 | sofh | soft switch , ATA , and soft client even with relavent supported codecs |
| 01:38.32 | sofh | i changed the OS, kernel everything... |
| 01:39.09 | r_evolution | damn man... what's your build? |
| 01:39.14 | sofh | actualy i am just connecting a carrier to my pri gsm gw via * box |
| 01:39.16 | r_evolution | something HAS to be wrong in the box but I have no idea what it could be |
| 01:39.30 | sofh | asterisk 1.2.4 on gentoo linux |
| 01:39.49 | r_evolution | oh... wow man... |
| 01:40.03 | sofh | i think i can work more..but i need an idea..what is the normal bw usage (both up and down) in asterisk others users have |
| 01:40.12 | r_evolution | have you tried gathering an ethereal log? |
| 01:40.26 | r_evolution | http://www.ethereal.com/ |
| 01:40.44 | sofh | nah, but i monitor the packets via iptraf |
| 01:40.58 | r_evolution | and you don't see any other incoming/outgoing packets? |
| 01:41.15 | sofh | offcourse not ! here is the confusion |
| 01:41.28 | sofh | only packets upd and ip as i am using sIP |
| 01:42.00 | sofh | i want to use h323 but i didnt find it stable so left and switch to SIP..but now this bandwidth issue is in front of me |
| 01:43.34 | r_evolution | well... with our old switch... we usually advised our customers to have 256 up/down |
| 01:43.44 | r_evolution | and that was for 3way as well as ease of browsing |
| 01:45.08 | nanogeek | sofh, how much bandwidth do you have on your switch now? |
| 01:45.19 | r_evolution | he dipped. |
| 01:45.39 | r_evolution | no joy on your end nano? |
| 01:45.56 | nanogeek | <PROTECTED> |
| 01:45.56 | nanogeek | <PROTECTED> |
| 01:45.56 | nanogeek | <PROTECTED> |
| 01:45.56 | nanogeek | <PROTECTED> |
| 01:45.56 | nanogeek | <PROTECTED> |
| 01:45.59 | nanogeek | <PROTECTED> |
| 01:46.21 | nanogeek | only problem is that I got the default CID for the trunk |
| 01:46.32 | nanogeek | :- |
| 01:46.34 | *** join/#asterisk forao (n=fasdfasd@pool-138-89-178-72.mad.east.verizon.net) |
| 01:46.51 | r_evolution | So why are you NoOp at the beginning? |
| 01:47.59 | nanogeek | for debug purposes. otherwise too lazy to remove it. |
| 01:48.05 | *** join/#asterisk enimihil (n=enimihil@70-98-228-219.dsl1.hol.ny.frontiernet.net) |
| 01:48.46 | r_evolution | ah |
| 01:48.49 | r_evolution | ok in the zapata.conf |
| 01:48.51 | r_evolution | you ahve callerid=asreceived |
| 01:48.53 | r_evolution | and |
| 01:48.55 | r_evolution | usecallerid=yes |
| 01:49.02 | nanogeek | si |
| 01:49.28 | nanogeek | priindication=outofband |
| 01:49.28 | nanogeek | callerid=asreceived |
| 01:49.28 | nanogeek | usecallerid=yes |
| 01:49.28 | nanogeek | hidecallerid=no |
| 01:49.28 | nanogeek | callwaiting=yes |
| 01:49.30 | nanogeek | callwaitingcalerid=yes |
| 01:49.33 | nanogeek | usecallingpres=yes |
| 01:49.35 | nanogeek | threewaycalling=yes |
| 01:49.38 | nanogeek | transfer=yes |
| 01:49.40 | nanogeek | cancellforward=yes |
| 01:49.43 | nanogeek | stripmsd => 1 |
| 01:49.45 | nanogeek | busydetect=no |
| 01:50.26 | r_evolution | I dunno then man... what i've read and what little i've done says that should be working |
| 01:50.33 | r_evolution | im assuming you just typoed on cancallforward |
| 01:50.41 | r_evolution | try usecallingpres=no? |
| 01:51.04 | r_evolution | restrictcid: (PRI channels only) This option has effect only when hidecallerid=no. If hidecallerid=no and restrictcid=yes, Asterisk will prevent the sending of the Caller ID data as a presentation number when making outgoing calls (ANI data is still sent). |
| 01:51.10 | nanogeek | hmmm |
| 01:51.16 | r_evolution | so what im sayin |
| 01:51.23 | r_evolution | is maybe it's having an issue sending the text |
| 01:51.34 | r_evolution | like... if you're forwarding to a cell... |
| 01:51.40 | r_evolution | does your cell support textual cid? |
| 01:51.56 | r_evolution | but if you restrictcid=yes |
| 01:52.06 | r_evolution | then ^ says the ani is still sent... |
| 01:52.06 | r_evolution | so? |
| 01:53.25 | nanogeek | my cell supports text however I'll try usecallingpres=no next to see what happens. |
| 01:54.22 | r_evolution | well mine supports text if i already have the number in... but not textual CID |
| 01:54.29 | r_evolution | i didnt want the thing to begin with... now i'm stuck with it :( |
| 01:54.38 | r_evolution | ex-gf : GET A CELL PHONE IN CASE I NEED YOU?!?!? |
| 01:54.47 | r_evolution | me : digital leash! *revolt* NO!! |
| 01:54.52 | r_evolution | we all know how that goes :-\ |
| 01:55.23 | nanogeek | tell me about it! |
| 01:56.36 | r_evolution | that's ok |
| 01:56.39 | r_evolution | i got the V-card ;) |
| 01:56.41 | r_evolution | so i won in the end |
| 01:56.43 | nanogeek | do i need to restart or reload after mod'ing zapata? |
| 01:56.46 | r_evolution | that's what i do... i win. |
| 01:56.49 | r_evolution | restart I thought? |
| 01:56.57 | r_evolution | not like it takes much |
| 01:56.58 | r_evolution | stop now |
| 01:56.59 | r_evolution | asterisk |
| 01:57.01 | r_evolution | asterisk -r -vvvvv |
| 01:57.09 | r_evolution | are you taking full logs? |
| 01:57.24 | nanogeek | good question. i'm not sure... |
| 01:57.31 | r_evolution | check out logger.conf |
| 01:57.35 | r_evolution | if the full is commented out... |
| 01:57.36 | r_evolution | then yer not |
| 01:57.44 | r_evolution | i always take full logs when im tryin to work through something |
| 01:57.47 | r_evolution | partially for myself |
| 01:57.50 | r_evolution | and partially for pastebins |
| 01:58.36 | r_evolution | hey here's an interesting line |
| 01:58.37 | r_evolution | http://www.voip-info.org/wiki/view/CallerID |
| 01:58.54 | r_evolution | On ISDN PRI lines (US NI-2 type) Callerid name information CNAM is transmitted in a separate FACILITY IE. some time AFTER the initial SETUP message. If you are using CNAM in your dialplan make sure to insert a wait statement before using the calleridname variable. Otherwise CALLERIDNAME will not be populated initially. |
| 01:59.40 | r_evolution | seriously man... check out http://www.voip-info.org/wiki/view/CallerID |
| 01:59.43 | r_evolution | wiki holds ALL ANSWERS |
| 01:59.46 | r_evolution | hahah |
| 01:59.57 | r_evolution | im leaving tho... time to run off into the wilderness until the next day of slave labor |
| 01:59.57 | r_evolution | i mean |
| 02:00.01 | r_evolution | code work |
| 02:00.14 | r_evolution | uhhmm try to catch up with justinu... he's usually the guy who i get my answers from |
| 02:00.37 | r_evolution | hate to break out on ya nano |
| 02:00.38 | r_evolution | but |
| 02:00.40 | nanogeek | thanks! I'm going to work long into the night to see if I can finger this thing out. |
| 02:00.46 | r_evolution | seriously bro |
| 02:00.47 | r_evolution | check the wiki |
| 02:00.50 | r_evolution | and ask justin |
| 02:00.57 | r_evolution | he usually helps me more than a little bit |
| 02:00.59 | nanogeek | I'll do both! |
| 02:03.01 | *** part/#asterisk enimihil (n=enimihil@70-98-228-219.dsl1.hol.ny.frontiernet.net) |
| 02:07.22 | *** join/#asterisk WAudette (n=WAudette@c-67-170-156-3.hsd1.or.comcast.net) |
| 02:12.00 | Homer99 | Greetings good people! Problem -- Just installed asterisk server with sangoma 4 port fxo card. Can dial out, but when dialing in, it sounds like it rings once and something picks up to the caller (silence), but phone does not ring and nothing happens -- help? |
| 02:16.39 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-190-62.red.bezeqint.net) |
| 02:21.37 | *** join/#asterisk stealthmethod (n=123@216.49.220.243) |
| 02:23.10 | *** join/#asterisk Young (i=Dunno@218.14.136.44) |
| 02:24.45 | *** part/#asterisk stealthmethod (n=123@216.49.220.243) |
| 02:26.48 | tuxinator_linux | Homer99: I assume you configured your dial plan? |
| 02:28.47 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 02:32.15 | *** join/#asterisk justinu (n=justin@eowyn.blacksun.net) |
| 02:35.25 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 02:35.40 | alephcom | Greeting Everyone |
| 02:36.11 | AAH_USER | Greetings Canadian |
| 02:37.03 | heison | ~seen coppie |
| 02:37.13 | jbot | heison: i haven't seen 'coppie' |
| 02:37.13 | Abydos313 | Hello |
| 02:37.13 | heison | ~seen coppice |
| 02:37.16 | jbot | coppice <n=chatzill@212.197.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 2d 17h 39m 3s ago, saying: 'you mean packets are delivered by a cron job? :-\'. |
| 02:40.21 | fifer | Anyone know about any issues wiht Kapu and naving to links? |
| 02:41.38 | fifer | We migrated content from a 2.0.x site over to 2.1.2 and Kapu does not recognize any of the old data |
| 02:41.42 | fifer | Only new objects |
| 02:41.47 | fifer | Very odd |
| 02:43.18 | *** join/#asterisk fafnir (i=hahaha@unaffiliated/fafnir) |
| 02:44.17 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 02:45.20 | justinu | I've got a system with a single x100p card in it. no matter what slot I put the card in, it always wants to share an IRQ w/ the ethernet card. any advices? |
| 02:46.05 | greendisease | configure the irq's manually in the bios, or get another ethernet card, or change the mobo |
| 02:46.19 | justinu | tried 2 of the 3 |
| 02:46.29 | greendisease | get a new mobo |
| 02:46.35 | justinu | is that really the bottom line? |
| 02:47.01 | greendisease | basically. if you tried doing the irqs manually and it didnt work, something there is funky |
| 02:47.09 | justinu | definitely... |
| 02:57.55 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 03:04.13 | justinu | greendisease: figured it out. two PCI slots are sharing IRQ 15 and so is the onboard ethernet |
| 03:04.26 | justinu | if I put it in the other slots, it's fine |
| 03:04.29 | justinu | thx |
| 03:04.35 | justinu | time to buy a new mobo |
| 03:05.05 | justinu | this bios doesn't allow me to set IRQs per slot, so I'm SOL |
| 03:07.10 | niZon | 18 hours to decide if I really want to buy a 7970G.... hmm |
| 03:07.44 | *** join/#asterisk john867530 (n=chatzill@207.177.124.87) |
| 03:08.26 | implicit | :) |
| 03:08.37 | implicit | ~seem implicit |
| 03:08.42 | implicit | ~seen implicit |
| 03:08.46 | jbot | implicit <n=implicit@ip68-4-84-39.oc.oc.cox.net> was last seen on IRC in channel #asterisk, 4s ago, saying: '~seen implicit'. |
| 03:08.49 | implicit | ~seen implicit recently |
| 03:08.51 | jbot | i haven't seen 'implicit recently', implicit |
| 03:09.16 | implicit | then you are a shitty bot. Learn to process natural languages |
| 03:09.23 | niZon | lol |
| 03:10.17 | implicit | niZon, u do know that 7970 has no SIP image right? |
| 03:11.27 | *** join/#asterisk grandy (n=mmmurf@c-68-42-64-205.hsd1.mi.comcast.net) |
| 03:11.58 | grandy | hello... what is the preferred method of scaling beyond one asterisk server? |
| 03:12.13 | implicit | what are you trying to do |
| 03:12.17 | implicit | what sort of scenario? |
| 03:12.25 | implicit | what protocols |
| 03:12.37 | implicit | what type of service |
| 03:12.58 | grandy | implicit: iax only ... |
| 03:13.15 | grandy | implicit: some 2 way calls but most 1 way (voicemail, automated outbound) |
| 03:13.53 | implicit | what is it exactly, PBX type setup? |
| 03:13.59 | grandy | implicit: i would rather have the system run on two+ instances of asterisk in parallel, for example, in case of a hardware failure... |
| 03:13.59 | implicit | what are the UAs |
| 03:14.13 | grandy | implicit: UAs? |
| 03:14.29 | implicit | how are your phones connected to *? |
| 03:14.35 | implicit | IAX? |
| 03:14.38 | grandy | implicit: yes, iax |
| 03:14.43 | implicit | what phones are you using |
| 03:14.46 | *** join/#asterisk glm2k_ (n=GLM@rrcs-24-199-11-46.west.biz.rr.com) |
| 03:15.17 | grandy | implicit: not using iax phones, connecting asterisk to an asterisk gateway that provides dial tone, inbound numbers, etc. |
| 03:15.50 | implicit | so you are using digium cards or something? |
| 03:16.01 | implicit | a bit confused about your setup |
| 03:16.05 | [av]bani | 7970... sipless indeed :( |
| 03:16.08 | grandy | implicit: nope, just connecting to the provider of dialtone via iax |
| 03:16.27 | implicit | so, explain to me again where your calls originate |
| 03:16.30 | implicit | and where they terminate |
| 03:16.55 | implicit | PSTN->IAX->*->IAX->PSTN?? |
| 03:17.07 | grandy | implicit: pretty much, for two way calls |
| 03:17.14 | implicit | ok |
| 03:17.16 | implicit | gotcha |
| 03:17.30 | grandy | so i'm just dealing with the middle part, the * |
| 03:17.34 | implicit | ok |
| 03:17.57 | implicit | well if you are just looking to have a hot failover |
| 03:18.01 | implicit | that should be extremely easy then |
| 03:18.09 | implicit | in case you have a hot failover |
| 03:18.19 | grandy | ok... just copy the dialplan? |
| 03:18.39 | implicit | yeah, and have heartbeat between the two |
| 03:18.41 | implicit | when the first dies |
| 03:18.48 | implicit | the other will steal its IP |
| 03:18.51 | implicit | and take its place |
| 03:18.58 | implicit | of course your in progress calls will die |
| 03:19.07 | grandy | is there a tool already in existence that would handle that IP reassignment? |
| 03:19.16 | implicit | www.linux-ha.org |
| 03:19.23 | implicit | should do exactly what you need |
| 03:19.32 | grandy | oh cool |
| 03:19.40 | implicit | for '2 way' calls, if at all possible use something like SER |
| 03:19.42 | implicit | with SIP |
| 03:19.46 | implicit | so that you don't lose in progress calls |
| 03:19.50 | implicit | and when you have it failover |
| 03:19.56 | implicit | you will still be able to keep all your accounting records |
| 03:20.07 | implicit | and end the calls properly |
| 03:20.12 | implicit | not sure if it will work in your scenario |
| 03:20.18 | grandy | ser with sip... hmm... |
| 03:20.19 | implicit | but worth looking into |
| 03:20.55 | grandy | ahh... so now what about just scaling for capacity? is there a way to do that and distribute the load over multiple * boxes in the setup I described? |
| 03:20.58 | implicit | yeah, cause SER is only transaction stateful, it doesn't keep call state, and your media will go point to point (unless you force it to go otherwise) |
| 03:20.59 | implicit | etc |
| 03:21.05 | jontow | :) |
| 03:21.07 | JunK-Y | implicit: ! |
| 03:21.10 | implicit | hey JunK-Y |
| 03:21.11 | implicit | how are you! |
| 03:21.12 | implicit | !!! |
| 03:21.17 | JunK-Y | im fine. |
| 03:21.17 | implicit | when the fuck are you coming down to california |
| 03:21.24 | implicit | ;) |
| 03:21.27 | JunK-Y | ive been there twice! |
| 03:21.29 | implicit | i just got back from colombia |
| 03:21.32 | implicit | this friday |
| 03:21.33 | JunK-Y | when are u comin' up? |
| 03:21.38 | implicit | to montreal? |
| 03:21.41 | JunK-Y | yes |
| 03:21.50 | implicit | when it gets warmer |
| 03:21.51 | implicit | hahaha |
| 03:21.51 | implicit | :) |
| 03:21.57 | JunK-Y | then, this summer! |
| 03:21.59 | implicit | i'm going to brazil again though pretty soon |
| 03:22.00 | implicit | come down |
| 03:22.08 | implicit | amazing girls in rio de janeiro |
| 03:22.15 | implicit | you wouldn't believe |
| 03:22.29 | JunK-Y | ive no doubt on that. |
| 03:24.15 | implicit | sorry grandy |
| 03:24.21 | grandy | implicit: np |
| 03:24.26 | implicit | why don't you send me a message, and i'll talk to you about this stuff in more detail |
| 03:24.58 | grandy | what kind of message? |
| 03:25.10 | implicit | aim or something, i'm going on my other computer |
| 03:25.21 | grandy | oh ok... that's cool if you've got a few minutes... |
| 03:25.34 | implicit | pm it to me |
| 03:26.09 | grandy | just need to install gaim real quick |
| 03:26.13 | implicit | k |
| 03:26.29 | implicit | which one do you use? |
| 03:26.39 | implicit | icq? |
| 03:27.03 | grandy | implicit: skype or gaim usually but i recently reinstalled this laptop and hadn't gotten around to installing everything on it just yet |
| 03:27.28 | tengulre | who is a SOHO user? |
| 03:28.05 | grandy | oh do you have google talk? |
| 03:28.15 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 03:28.24 | implicit | nop :( |
| 03:28.39 | grandy | implicit: ok, gaim is installing now... |
| 03:31.44 | niZon | implicit: I'm aware the 7970 has no SIP image :P |
| 03:31.57 | niZon | I have chan_sccp installed :) |
| 03:32.15 | niZon | it's said to work quite well with 7970s |
| 03:32.33 | [av]bani | needs more cowbell |
| 03:32.53 | [av]bani | niZon: chan_sccp works fine with 7970... with the usual warts |
| 03:33.09 | niZon | tell me more |
| 03:33.42 | FuriousGeorge | i got an echo... |
| 03:33.46 | [av]bani | well, i dont like sccp in general |
| 03:33.56 | FuriousGeorge | ...and the only presciption is more cowbell |
| 03:34.09 | [av]bani | and cisco phones give you barely any control from the UI... its all done from static XML config files |
| 03:34.45 | niZon | hmm |
| 03:34.55 | niZon | how's the sound quality? |
| 03:35.09 | [av]bani | fine, but it had damn well better be for what you pay |
| 03:35.14 | [av]bani | i like the polycom audio better |
| 03:35.16 | niZon | yeah |
| 03:35.18 | niZon | hmm |
| 03:35.33 | [av]bani | but the polycom has no backlighted phones |
| 03:36.02 | mogorman | you work in dark av |
| 03:36.09 | niZon | I want an uber professional looking phone, I'll be making use of the XML features too |
| 03:36.12 | MstlyHrmls | how big of a demand is a backlight, really? |
| 03:37.05 | *** join/#asterisk mrkyr (n=bviitane@h24-207-83-214.cst.dccnet.com) |
| 03:37.38 | [av]bani | mogorman: i dont sleep with the lights on, no |
| 03:37.49 | mogorman | lol |
| 03:37.54 | MstlyHrmls | I keep hearing about it as a deficiency, but is it really, truly, honestly going to be worth the $10 or whatever extra they'll tack onto the price? |
| 03:37.57 | [av]bani | niZon: ip601 fits the bill |
| 03:38.00 | mogorman | i never sleep without my night light |
| 03:38.20 | MstlyHrmls | are there more people using this as a residential phone than I realize? :-) |
| 03:38.41 | [av]bani | MstlyHrmls: if you want a phone for your nightstand, its nice to have. also MUCH easier to see the display even in well lighted areas |
| 03:38.42 | niZon | small display |
| 03:39.00 | [av]bani | MstlyHrmls: eg even in a well lit office its 3-4x easier to see a backlighted display |
| 03:39.26 | niZon | hmm.. I can get a 7970 for $525CDN shipped |
| 03:39.50 | [av]bani | its a HUGE phone |
| 03:40.52 | niZon | yeah |
| 03:41.00 | [av]bani | biggest phone i have |
| 03:41.54 | mogorman | anyone know where i can find a good used server for my house |
| 03:42.07 | MstlyHrmls | [av]bani: sure, it's easier, but is it enough easier that justifies the cost (plus the added drain on PoE)? |
| 03:42.38 | MstlyHrmls | [av]bani: I dunno, I've never really had a problem with the lack of backlight; yet it's a common complaint here... |
| 03:43.31 | mogorman | im looking on spending 300 bucks i dont need a lot |
| 03:43.58 | niZon | mogorman: ebay |
| 03:44.16 | mogorman | yeah im looking |
| 03:44.24 | mogorman | but nothing really jumping out at me |
| 03:45.44 | mogorman | i need to find something closer to huntsville |
| 03:45.49 | mogorman | as shipping will kill me |
| 03:45.57 | [av]bani | mogorman: i have phones with and without. i strongly prefer with |
| 03:45.58 | mogorman | might just build it |
| 03:46.10 | mogorman | yeah [av]bani i was just trolling |
| 03:46.18 | [av]bani | :) |
| 03:46.18 | mogorman | its not a big deal to me |
| 03:46.24 | mogorman | i know it can be useful |
| 03:47.48 | heison | ~seen coppice |
| 03:47.50 | jbot | coppice <n=chatzill@212.197.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 2d 18h 49m 37s ago, saying: 'you mean packets are delivered by a cron job? :-\'. |
| 03:48.57 | litage | what do the "h" and "i" in these 2 dialplan entries mean?: exten => h,1,Hangup exten => i,2,Hangup |
| 03:49.03 | *** join/#asterisk bmg505 (n=leon@dsl-165-149-69.telkomadsl.co.za) |
| 03:49.14 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 03:59.08 | *** join/#asterisk maayani (n=maayani@cpe-24-90-246-25.nyc.res.rr.com) |
| 04:02.00 | *** part/#asterisk maayani (n=maayani@cpe-24-90-246-25.nyc.res.rr.com) |
| 04:03.34 | *** join/#asterisk jdiskywlkr (n=kvirc@ip68-0-83-251.tu.ok.cox.net) |
| 04:03.46 | *** join/#asterisk maayani (n=maayani@cpe-24-90-246-25.nyc.res.rr.com) |
| 04:05.44 | *** join/#asterisk bewest (n=bewest@ool-435012e6.dyn.optonline.net) |
| 04:05.46 | harryvv | quiet in here |
| 04:06.01 | docelmo | yep yep |
| 04:06.30 | *** join/#asterisk webman (n=adamg@52.87.233.220.exetel.com.au) |
| 04:07.29 | webman | does anyone know of a VOIP service which will allow calling a US based 1-888 phone number for free ?? I need to call AOL Postmaster helpdesk to fix my mail server :( |
| 04:07.52 | FuriousGeorge | sipphone (with no minutes purchaces) |
| 04:07.54 | trixter | www.trxtel.com -> tollfree termination |
| 04:07.58 | trixter | its free, dont even have to register |
| 04:07.58 | FuriousGeorge | nufone asterlink |
| 04:08.03 | FuriousGeorge | i assume most of them |
| 04:08.12 | FuriousGeorge | webman: !skype |
| 04:08.24 | trixter | trxtel supports both sip and iax2 as well |
| 04:09.27 | *** part/#asterisk maayani (n=maayani@cpe-24-90-246-25.nyc.res.rr.com) |
| 04:10.23 | *** join/#asterisk maayani (n=maayani@cpe-24-90-246-25.nyc.res.rr.com) |
| 04:11.04 | webman | thanks people, will try that... |
| 04:11.48 | *** join/#asterisk iaxy (n=iaxy@modemcable236.55-131-66.mc.videotron.ca) |
| 04:11.53 | trixter | FuriousGeorge: most voip providers require you to register and typically pay before they give you free US tollfree, so if its just 1 or 2 calls then it may not be worth it :/ |
| 04:12.11 | harryvv | how can these 1800 services survive if thay dont charge? |
| 04:12.27 | mogorman | you just have to be in my apt |
| 04:12.32 | mogorman | or on my network |
| 04:12.33 | trixter | they expect to make money on other calls and/or monthly subscriptions |
| 04:12.53 | FuriousGeorge | trixter: sippohone, but its not iax |
| 04:13.19 | FuriousGeorge | will give you free calls to 800#s with an account |
| 04:13.29 | FuriousGeorge | i got a feeling gtalk will too |
| 04:13.47 | AAH_USER | mogorman, please send info so I can terminate all long distance and international with you |
| 04:14.09 | AAH_USER | Thise $15 / minute calls hurt |
| 04:15.18 | AAH_USER | Ha-ha |
| 04:16.18 | FuriousGeorge | i terminate calls only from my front door to the kitchen for free, and thats only cuz the landlord wont fix the doorbelll |
| 04:16.44 | bewest | how can I set the voicemail greeting for a context? |
| 04:17.04 | bewest | I tried directoryintro=custom/mygreet but it didn't work |
| 04:17.09 | bewest | vm-greet is always played |
| 04:17.16 | AAH_USER | Better doorbell anyhow, you can tell salesmen to buzz off from your recliner |
| 04:17.32 | FuriousGeorge | and religious zealots |
| 04:17.33 | jontow | i think, perhaps, i'll try to get SQL-based sip config workin tonight :) |
| 04:18.11 | FuriousGeorge | jontow: i was thinking about that. how would that work? would the internal astdb be the mysql db? |
| 04:18.32 | FuriousGeorge | nm, i was thinking about extensions.conf |
| 04:18.37 | FuriousGeorge | in a db |
| 04:18.45 | harryvv | Fastest way to turn off a voip customer is a shitty connection when demonstrating it. |
| 04:19.02 | harryvv | who here uses xo? |
| 04:19.02 | FuriousGeorge | harryvv: a lot or pressure on one call |
| 04:19.38 | harryvv | FuriousGeorge no just talked to people that I could sell the service to and thay affiliate voip with unreliability. |
| 04:19.47 | *** part/#asterisk AAH_USER (n=none@c-67-186-207-234.hsd1.ut.comcast.net) |
| 04:19.54 | *** join/#asterisk p0g0_ (n=pogo@madwifi/support/p0g0) |
| 04:20.25 | FuriousGeorge | harryvv: i guess that all depends on your ISP |
| 04:20.36 | harryvv | So when I advertise my system, the word voip is not going to be included. |
| 04:20.38 | harryvv | yes |
| 04:20.48 | harryvv | and there iax/sip service. |
| 04:20.58 | trixter | I wouldnt include the word voip either but I might include it as an acronym |
| 04:21.51 | *** join/#asterisk Olobola (n=casper_s@216.218.221.166) |
| 04:23.31 | Olobola | dudes, my outgoing calls through provider A are fine, incoming through provider B is choppy. This started after we moved, I'm on a symetrical connection. |
| 04:25.59 | jontow | FuriousGeorge: res_odbc :) |
| 04:29.44 | bewest | does directoryinfo directive in a context in voicemail.conf change the greeting for the voicemail? |
| 04:38.31 | *** join/#asterisk hellop (n=hellop@cpe-72-130-252-68.hawaii.res.rr.com) |
| 04:44.00 | mogorman | who is aah_user |
| 04:44.22 | bewest | how can I tell asterisk to use a sound other than vm-intro when I send someone to voicemail? |
| 04:46.48 | *** join/#asterisk livinded (n=livinded@cpe-24-24-190-252.socal.res.rr.com) |
| 04:47.37 | russellb | bewest: open up apps/app_voicemail.c |
| 04:47.46 | russellb | bewest: you will see "#define INTRO "vm-intro" |
| 04:47.55 | russellb | change that to whatever you want, recompile, and reinstall |
| 04:48.07 | bewest | oh |
| 04:48.18 | bewest | thanks |
| 04:48.28 | bewest | I thought it was a config thing |
| 04:48.31 | russellb | you're welcome |
| 04:48.31 | livinded | what causes "Received mini frame before first full voice frame" and how do i stop it from happening? |
| 04:48.36 | wunderkin | is anyone here familiar with app_changrab? apparantly it was written by anthm, i was just wondering how hard it would be for it to obtain the variables of the channel it grabbed |
| 04:48.40 | russellb | bewest: no option for that one |
| 04:49.18 | bewest | russellb: I thought there was a way to play an arbitrary sound when sending someone to voicemail |
| 04:49.19 | *** join/#asterisk SPoon_TSX (n=Kit@S0106001422e182f9.vc.shawcable.net) |
| 04:49.44 | bewest | I attempted recording an unavailable message |
| 04:50.38 | SPoon_TSX | hi there, I got a quick questions. When I talk from the IP Phone to PSTN, as soon as I talk, the other side will hear some statics at the ackground of my voice. I am currently have rx=5, tx=1. May I know what may possible the reason? |
| 04:51.13 | hellop | I have an older ver of * 1:1.0.7 I have no agi-bin. What is the recommended way to get it? I installed via apt-get. |
| 04:51.17 | hellop | In Debian sarge. |
| 04:52.15 | hellop | Should I A: re-install * from source. B: Look for deb package for agi. C: Get a source package for AGI. D: make sure the AGI package matches my asterisk version. |
| 04:52.56 | *** join/#asterisk angom_h (n=angom@red-corp-200.38.17.111.telnor.net) |
| 04:53.02 | hellop | I should probably apt-get remove asterisk and get a new version... |
| 04:53.07 | litage | what do the "h" and "i" in these 2 dialplan entries mean?: exten => h,1,Hangup exten => i,2,Hangup |
| 04:53.36 | wunderkin | h is hangup, you dont want to call hangup in h, i is invalid extension |
| 04:54.39 | livinded | what causes "Received mini frame before first full voice frame" and how do i stop it from happening? |
| 04:55.19 | hellop | tzafrir, what do you think, is trying to get AGI on the * included with Sarge a lost cause? |
| 04:55.35 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 04:57.51 | jontow | woo |
| 04:58.01 | jontow | got sip config from odbc / mysql working |
| 04:58.04 | jontow | took a bit of hacking for sure :) |
| 04:58.12 | *** join/#asterisk outtolunc (n=me@adsl-69-110-29-140.dsl.pltn13.pacbell.net) |
| 05:00.08 | wunderkin | ~striplast |
| 05:00.41 | jontow | curious, though, does 'deny' / 'permit' work in the sip peers table(s)? |
| 05:04.14 | Snake-Eyes | Does any one know if you can have a set of commands that are genric to a whole section and then in between set extensions eg: exten => s,1,Answer |
| 05:04.14 | Snake-Eyes | exten => 400,2,Dial(SIP/100) |
| 05:04.15 | Snake-Eyes | exten => 401,2,Dial(SIP/101) |
| 05:04.15 | Snake-Eyes | exten => s,3,Hangup |
| 05:06.07 | Snake-Eyes | s or X or something not sure if theres a char i can use for this |
| 05:06.14 | outtolunc | s is s, everything else is everything else |
| 05:06.44 | outtolunc | so you need multiple 'sets' |
| 05:07.12 | *** join/#asterisk SPoon_TSX_ (n=Kit@S0106001422e182f9.vc.shawcable.net) |
| 05:07.29 | outtolunc | s,1 thru whatever, then _X,1 thru whatever, then _X.,1 thru whatever |
| 05:07.49 | *** join/#asterisk angom_h (n=angom@red-corp-201.143.96.47.telnor.net) |
| 05:08.02 | SPoon_TSX_ | Hi there, May I know what would be the possible reason I will hear static via my PSTN line from SIP Phone? |
| 05:08.04 | outtolunc | there should be info like this in the README's |
| 05:08.49 | Snake-Eyes | outtolunc, hmm ok |
| 05:09.01 | outtolunc | SPoon_TSX: moisture? <G> |
| 05:09.55 | outtolunc | Snake-Eyes: you just have to release that s is a 'fall-thru' nothing more |
| 05:11.04 | litage | thanks wunderkin |
| 05:14.03 | Snake-Eyes | outtolunc, so i have _XXX,1Answer 400,2,Dial() 401,2,Dial() _XXX,3,Hangup ? |
| 05:14.07 | FuriousGeorge | so am i supposed to use chan/dsp for the intercom? |
| 05:14.07 | *** part/#asterisk bewest (n=bewest@ool-435012e6.dyn.optonline.net) |
| 05:14.21 | SPoon_TSX_ | outtolunc: Moisture? You means bad line? |
| 05:14.25 | FuriousGeorge | the old system is getting totally phased out, i suppose i gotta figure out the intercom now |
| 05:15.21 | outtolunc | no, moisture means moisture, which doesn't NOT equate to 'bad line' |
| 05:15.52 | outtolunc | there are alot of reasons |
| 05:16.16 | Snake-Eyes | outtolunc, which readme are you reffering to? one in src ? (looked at voip-info extensions.conf) |
| 05:16.20 | outtolunc | i mentioned moisture because of the shitload of rain (hense the <G> i put on the end) we are getting in this area lately |
| 05:17.31 | outtolunc | for those of you that *might* have missed (or forgot) there are a bunch of docs IN the doc dir |
| 05:17.39 | outtolunc | asterisk/doc |
| 05:17.43 | outtolunc | hmm |
| 05:19.51 | outtolunc | the 2 you really need to read first are extensions.txt and README.variables |
| 05:24.52 | *** join/#asterisk hcatlin (n=hcatlin@host6614614774.dsl.res.tor.fcibroadband.com) |
| 05:25.15 | hcatlin | Alright, anyone feel like helping someone who is 4 hours into upgrading the firmware on a Cisco 7960 to SIP? |
| 05:26.48 | harryvv | !seen ariel |
| 05:26.56 | harryvv | !seen arial |
| 05:27.06 | harryvv | mabey im not typing his nick right. |
| 05:27.23 | Beirdo | ~seen ariel |
| 05:27.31 | jbot | ariel <n=kvirc@host224.201-252-221.telecom.net.ar> was last seen on IRC in channel #debian, 5d 14h 32m 25s ago, saying: 'pipeline: thanks, gonna try it. be right back'. |
| 05:27.31 | hcatlin | Hrrrmmm... no one else fought with the 7960 |
| 05:27.32 | hcatlin | ? |
| 05:27.42 | harryvv | Well at least he was on 5 days ago |
| 05:27.45 | Beirdo | harryvv, wrong command char :) |
| 05:27.55 | Snake-Eyes | outtolunc, thanks :) |
| 05:28.06 | harryvv | yea, I gave him a call to see what he was up to and he did not get back. |
| 05:28.11 | Beirdo | "be right back" means 5.5days, I guess |
| 05:28.22 | harryvv | Did you talk to him? |
| 05:28.29 | Beirdo | not me, no |
| 05:28.33 | harryvv | k |
| 05:38.56 | *** join/#asterisk mrzip (i=mrzip@cpe-24-193-115-20.nj.res.rr.com) |
| 05:39.11 | mrzip | can someone help me with a quick question.. |
| 05:39.11 | mrzip | <mrzip> I have a ZAP clone card, connected to a phone line that you need to dial 9 first, when i check the log i noticed that some of the digits are getting cut off.. tried adding a few more XX to the dial plan, but no luck |
| 05:39.30 | mrzip | oh, im using AAH |
| 05:41.38 | *** join/#asterisk hardwire (n=hardwire@209.112.194.39) |
| 05:42.21 | *** join/#asterisk xtrvd (n=j@d209-121-36-44.bchsia.telus.net) |
| 05:44.30 | xtrvd | Has anybody run into this problem before while trying to play standard gsm files on an IVR: Unable to find a codec translation path from ulaw to gsm |
| 05:45.33 | *** join/#asterisk clive- (n=pirch@dsl-145-49-228.telkomadsl.co.za) |
| 05:46.41 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 05:47.51 | livinded | what causes "Received mini frame before first full voice frame" and how do i stop it from happening? |
| 05:48.03 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:50.56 | wasim | livinded: by receiving packets in the correct order |
| 05:51.18 | livinded | wasim, how can i force it to recieve them in the correct order? |
| 05:51.32 | wasim | livinded: you can't |
| 05:51.49 | wasim | livinded: it'll receive them in the order received |
| 05:52.00 | trixter | has anyone had problems with asterisk in fbsd 6? does anyone even run asterisk in fbsd6? |
| 05:52.04 | livinded | doesn't iax2 use tcp? |
| 05:52.21 | wasim | livinded: no |
| 05:52.22 | trixter | no |
| 05:52.23 | livinded | and don't tcp packets always get recieved in order |
| 05:52.24 | wasim | livinded: heavens forbit |
| 05:52.25 | livinded | oh |
| 05:52.50 | livinded | it always cuts off the first couple words of my ivr |
| 05:53.03 | livinded | even with a wait of 3 seconds before it starts |
| 05:53.04 | trixter | maybe you should answer(); wait(2); |
| 05:53.18 | wasim | livinded: put a Wait(), or better Playback(muzik-intro-3-seconds) |
| 05:53.19 | trixter | maybe it only starts the stream when you send audio |
| 05:53.26 | trixter | so try answer(); playback(silence/3); |
| 05:53.38 | trixter | that will have the same effect as a wait but will force audio to be sent |
| 05:55.25 | livinded | trixter, thanks |
| 05:55.30 | livinded | that works so much better |
| 05:56.17 | *** part/#asterisk propagandhi (n=opera@CPE-144-131-132-107.nsw.bigpond.net.au) |
| 05:56.22 | mrzip | <mrzip> I have a ZAP clone card, connected to a phone line that you need to dial 9 first, when i check the log i noticed that some of the digits are getting cut off.. tried adding a few more XX to the dial plan, but no luck |
| 05:56.44 | mrzip | anyone have any idea, how to fix this? |
| 06:00.36 | xtrvd | how many digits are being cut off? |
| 06:00.48 | asterisk99 | mrzip: sounds like a timing problem... and I've read that you have to make darned sure your Zap card isn;t sharing and IRQ with another device otherwise the damn things start fouling up |
| 06:01.22 | asterisk99 | kinda like an electronic version of Abbot & Costello's "Who's on First?" |
| 06:01.50 | asterisk99 | Ther's also kooky things like having to turn of hyperthreading on some PCs |
| 06:02.38 | asterisk99 | Hyperthreading if not properly supported by the motherboard can cause all sort weird timing problems |
| 06:03.24 | msw | hyperthread support? on the motherboard? huh? |
| 06:03.34 | mrzip | no, i checked the log file and its cut off there missing the last few digits, has to be a dial plan thing |
| 06:04.04 | asterisk99 | mrzip: it'll be in your CMOS... it's a feature of the Pentium chips |
| 06:04.31 | xtrvd | asterisk99: With all due respect, that seems to be a very specific solution to such a broad problem. |
| 06:05.01 | xtrvd | mrzip: How many digits? Is it the same every time? |
| 06:05.28 | clive- | mrzip frmo my expereince, hyperthreading and asterisk dont mix] well |
| 06:05.32 | mrzip | yes, the last 2 |
| 06:05.39 | msw | mrzip: what kinds of phones are you using |
| 06:05.51 | mrzip | sipura 2002 |
| 06:06.20 | asterisk99 | xtvrdI agree. But timing problems are like that. It's always some dumb little thing that goes wroing. I have somewhere a write-up from Digium (I'm looking now)... in it they list all sorts of itsy-bitsy things that can affect timing |
| 06:06.51 | mrzip | pc is old, might be a p2 even.. dont think it has multithreading |
| 06:07.37 | xtrvd | asterisk99: Well then I admit, you know a lot more about timing than I do. =) |
| 06:07.37 | msw | mrzip: if you originate a call from the asterisk console, do you miss digits? |
| 06:07.44 | asterisk99 | xtvrdI: I found it... in an old email |
| 06:08.11 | mrzip | never did it from the consol |
| 06:08.18 | msw | mrzip: what's the current dial plan setting in the sipura |
| 06:08.19 | mrzip | from command line? |
| 06:08.42 | xtrvd | asterisk99: Do you think that you could try solving this little bugger if you're so quick on the draw =) : "channel.c:2333 set_format: Unable to find a codec translation path from ulaw to gsm" |
| 06:08.45 | kll | mrzip: dial extension@context |
| 06:08.50 | asterisk99 | xtvrdI: Naaaaaaaa. I only seem smart cuz I've made all the mistakes before... I started out life as a operating systems developer back 30 years ago ... when men were men, and computers were rooms size behemoths |
| 06:09.11 | xtrvd | Ahh yes. When men were men, and the sheep were scared. |
| 06:09.14 | mrzip | (0|00|011,xx.|1xxxxxxxxxx|xxxxxxxxxx|*xx|[3469]11) |
| 06:09.24 | asterisk99 | xtvrdI: Baaaaa-aaaaa-aaaaa |
| 06:09.45 | mrzip | cant test it right now, but im going to leave the dial plan blank on the sipura and try it tomorrow |
| 06:09.50 | msw | mrzip: you would use "dial (whateverphonenumber)" on the console |
| 06:10.02 | msw | mrzip: and "hangup" to hang up |
| 06:10.25 | msw | mrzip: at least then you can see if it's a problem on the voip handset end or the zaptel end |
| 06:10.43 | asterisk99 | xtvrdI: Are we talking IAX here? |
| 06:11.03 | xtrvd | Yes, incoming from an IAX line onto an IVR |
| 06:11.23 | msw | mrzip: but first guess is that the dialplan on the voip device is wrong |
| 06:11.39 | asterisk99 | xtvrdI: Read page at http://www.digitnetworks.com/forums/showthread.php?t=112 |
| 06:12.14 | xtrvd | asterisk99: <3 |
| 06:12.14 | mrzip | ok, ill give that a shot tomorrow |
| 06:12.18 | kll | anyone knows what this means: 2006-02-24 09:26:04 WARNING[712] frame.c: Unable to calculate sample length for format unknown |
| 06:12.41 | asterisk99 | xtvrdI: The solutin there was: disallowed=all and allow=glaw,ulaw in my sip and iax.conf |
| 06:12.55 | xtrvd | Hmm, I thought I had those already... *double checking now* |
| 06:13.52 | iaxy | xtrvd: look here. |
| 06:13.56 | mrzip | also, i got this ht-488 that id like to use the FXO port.. from what I understand it connects to asterisk and registers as an Extention . would that be the same with those 4port fxo hubs? |
| 06:13.57 | iaxy | http://bugs.digium.com/view.php?id=4825&nbn=49 |
| 06:14.26 | xtrvd | iaxy: Thank you, I'll have a look. |
| 06:15.11 | xtrvd | iaxy: I'm sorry, I don't speak greek. |
| 06:15.14 | xtrvd | Crap... |
| 06:16.44 | iaxy | Then, I'll translate.... recompile with that patch or follow asterisk99's link and rearrange your codecs.... |
| 06:17.11 | *** join/#asterisk damania2 (n=heh@adsl-70-141-253-163.dsl.irvnca.sbcglobal.net) |
| 06:17.14 | damania2 | hello>? |
| 06:17.29 | damania2 | i need to setup a system for a motel 40 rooms |
| 06:18.11 | xtrvd | iaxy: Thank you for the translation. I do appreciate it. |
| 06:19.03 | mrzip | well thanks for the help guys, ill be back im sure.. |
| 06:19.33 | damania2 | can anyone help me give me an idea on what hardware i need |
| 06:19.52 | damania2 | i've never setup a phone system before |
| 06:20.01 | Qwell | damania2: a computer |
| 06:20.06 | Qwell | beyond that...it depends |
| 06:20.20 | xtrvd | damania2: Perhaps a phone or two to go with that computer. |
| 06:20.20 | kll | anyone knows what this means: 2006-02-24 09:26:04 WARNING[712] frame.c: Unable to calculate sample length for format unknown |
| 06:20.27 | damania2 | i'm trying to get an idea for all the parts that are needed so i can get an idea for the budget |
| 06:20.32 | damania2 | it's for a 40 unit motel |
| 06:20.37 | livinded | damania2, do you want to put ip phones in all the rooms? |
| 06:20.37 | Qwell | what type of phones? |
| 06:20.43 | damania2 | cheapest phones |
| 06:20.52 | damania2 | it's a really cheap client |
| 06:21.08 | Qwell | analog would probably be best. stolen phones cost less |
| 06:21.25 | Qwell | how many trunks? |
| 06:21.42 | Qwell | and what type? voip? analog? pri? |
| 06:21.45 | damania2 | they will ahve 3 incoming lines |
| 06:21.51 | Qwell | for 40 guests? |
| 06:21.54 | damania2 | yeah |
| 06:21.58 | damania2 | it's an el cheapo motel |
| 06:22.00 | Qwell | o...k... |
| 06:22.13 | damania2 | u didn't believe me when i said cheap hehehe |
| 06:22.21 | damania2 | they will put in more lines when they get more funding |
| 06:22.41 | livinded | theres going to be a lot of all circuits are busy messages being sent |
| 06:22.53 | Qwell | if you got a dualspan T1 card, you could get 40 phones, and up to 6 lines |
| 06:23.03 | Qwell | you'd need that, and two channel banks with mostly fxs |
| 06:23.36 | damania2 | so i need 2 24port fxs cards? |
| 06:23.38 | livinded | or put ip phones in all the rooms and get a big switch |
| 06:23.55 | Qwell | damania2: well...you could, sure |
| 06:24.08 | Qwell | but a dualspan T1 card + two channelbanks would probably be better |
| 06:24.13 | damania2 | how much are ip phones compared to regualr phones |
| 06:24.23 | Qwell | depends |
| 06:24.27 | livinded | depends where and what you buy |
| 06:24.32 | Qwell | You can get piles of shit for $80 |
| 06:24.38 | livinded | buying in bult you may be able to get a discount |
| 06:24.43 | livinded | bulk* |
| 06:24.47 | damania2 | i have to order pbx lines from the phoen company? am i able to use regular pots lines? |
| 06:24.53 | Qwell | damania2: a line is a line |
| 06:25.18 | damania2 | ok |
| 06:25.50 | damania2 | so give me an idea of how much an ip phone costs vs analog |
| 06:25.57 | damania2 | new |
| 06:26.03 | Qwell | You can get piles of shit for $80 |
| 06:26.09 | damania2 | $80 each? |
| 06:26.09 | Qwell | read: grandstream |
| 06:26.11 | *** join/#asterisk Rhizome (n=rhizome@tor/session/x-b6d8e2c33c8c1ebb) |
| 06:26.19 | Qwell | BUT...I can guarantee they'll be stolen |
| 06:26.27 | damania2 | so $80 each |
| 06:26.49 | damania2 | so would be the cost savings with ip phones |
| 06:26.54 | xtrvd | Or if you connect $10 from china analog phones to a FXS, you can save a headache... |
| 06:27.03 | Corydon76-home | $80 each, but before you compare to analog, note that SIP phones are standalone |
| 06:27.20 | Corydon76-home | Analog phones need to be attached to a special interface |
| 06:27.26 | damania2 | would digital phones use cat3 or cat5 |
| 06:27.41 | Corydon76-home | digital phones? |
| 06:27.44 | Qwell | they aren't digital |
| 06:27.45 | Corydon76-home | Like what? |
| 06:28.14 | iaxy | Do you have a bunch of meridian phones lying around? |
| 06:28.21 | damania2 | i don't have anything |
| 06:28.22 | Qwell | analog != digital != IP |
| 06:28.27 | iaxy | connect them to a citel box... |
| 06:28.40 | Corydon76-home | Actually, IP phones are digital |
| 06:28.43 | damania2 | sorry i am using the wrong terms. i meant ip phones |
| 06:28.45 | Qwell | well.. |
| 06:29.01 | Qwell | when people say "digital phone" they almost always mean a proprietary phone |
| 06:29.13 | Corydon76-home | They just tend not to be proprietary pieces of crap, unless you're using Cisco |
| 06:30.03 | iaxy | I just did an upgrade on a norstar switch, ripped out the switch put an asterisk box in, connected the meridian mdf to the citel box and it was up and running. |
| 06:30.09 | livinded | Corydon76-home, whats wrong with cisco phones? |
| 06:30.39 | Corydon76-home | Well, what's not wrong with Cisco phones? |
| 06:30.54 | kll | I think they work great |
| 06:31.04 | kll | they are a tad large though |
| 06:31.05 | Corydon76-home | They're expensive, they're annoyingly difficult to upgrade, to configure. |
| 06:31.20 | iaxy | don't you need a liscense for Cisco phones? |
| 06:31.21 | Corydon76-home | Sure, once you configure them and get them to the right firmware level, they work fine |
| 06:31.27 | kll | no they are not, it'a a bit of a threshold |
| 06:31.38 | livinded | i was thining about buying one for my house |
| 06:31.40 | kll | tftp is great for the "enterprise" |
| 06:32.05 | Corydon76-home | tftp is great when you don't have phones that can do FTP |
| 06:32.20 | Corydon76-home | Ooops, we have phones that can do FTP. So much for Cisco |
| 06:32.45 | kll | so your telling me cisco is crap based on the fact they don't do ftp |
| 06:32.51 | Corydon76-home | Ooops, we have phones that will directly upgrade to the latest firmware. So much for Cisco |
| 06:33.04 | damania2 | which ones are they |
| 06:33.09 | Qwell | Corydon76-home: cisco phones can upgrade to the latest directly |
| 06:33.11 | Corydon76-home | Ooops, we have phones that are easy to configure. So much for Cisco |
| 06:33.20 | kll | the upgrade path is a tad cumbersome, yes. |
| 06:33.23 | Qwell | My cisco (on sccp) is dead simple |
| 06:33.33 | kll | configuration is easy |
| 06:33.39 | Corydon76-home | Qwell: only if the previous version on the Cisco is only 1 major version down |
| 06:33.45 | Qwell | Corydon76-home: not true :) |
| 06:33.53 | Qwell | You can easily go sccp 3.2 > 7.x |
| 06:34.02 | Corydon76-home | Qwell: I mean SIP |
| 06:34.11 | Qwell | then sip is a step away. You can set it up once, and it'll "Just Work" |
| 06:34.31 | Corydon76-home | Qwell: yeah, see, that's why we use Polycom phones. |
| 06:34.45 | Corydon76-home | Because we don't care to go through that shit again with Cisco |
| 06:34.57 | Qwell | They've made upgrades dead simple |
| 06:35.07 | Qwell | one line in two files... |
| 06:35.11 | Qwell | and you're at the latest SIP |
| 06:35.15 | Corydon76-home | Qwell: oh, they have, finally? |
| 06:35.45 | Corydon76-home | Well, they're still more expensive than Polycom phones |
| 06:36.14 | Qwell | well worth it, imo |
| 06:36.29 | Corydon76-home | Qwell: are you all paid up on your Cisco licenses, btw? |
| 06:36.33 | Qwell | nope |
| 06:36.36 | iaxy | I think we can safely say that Corydon is anti-Cisco...:-) |
| 06:37.03 | Corydon76-home | Qwell: it woiuld be a shame if the BSA dropped by and did an audit on your Cisco licenses, wouldn't it? |
| 06:37.10 | Qwell | nope |
| 06:37.13 | Qwell | the license is for CCM |
| 06:37.23 | Qwell | so says all of their documentation |
| 06:37.29 | livinded | is there a flag for the new meetme that wont prompt a user to record a name? |
| 06:37.39 | Qwell | livinded: "new meetme"? |
| 06:37.50 | Corydon76-home | iaxy: not entirely undeserved by Cisco, either, though |
| 06:37.51 | livinded | well the newest version |
| 06:38.04 | livinded | the version i used to run didn't prompt |
| 06:38.15 | Qwell | livinded: it only prompts if you set the flag to prompt |
| 06:38.22 | Qwell | i I believe it is |
| 06:38.36 | livinded | i didn't set a flag to prompt |
| 06:39.08 | iaxy | So if you connect Cisco phones to *, you don't have to pay liscense? |
| 06:39.17 | Qwell | iaxy: as far as I understand it |
| 06:39.25 | Qwell | talk to your cisco rep |
| 06:39.56 | livinded | the only flags i have set are icp and none of those prompt to record a name |
| 06:40.19 | Qwell | Didn't I just say it was i? |
| 06:40.22 | Corydon76-home | Even better, talk to a lawyer |
| 06:40.26 | Qwell | could've sworn I did... |
| 06:40.37 | Qwell | Corydon76-home: indeed |
| 06:40.42 | livinded | doe si play the sound when someone joins or leaves? |
| 06:40.47 | livinded | doesn't* |
| 06:40.53 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 06:41.01 | Corydon76-home | Cisco rep can say whatever; if the lawyers say different later, you're going to pay... |
| 06:41.48 | livinded | ok cool i was the flag that did it |
| 06:45.57 | *** part/#asterisk justinu (n=justin@eowyn.blacksun.net) |
| 06:46.30 | Ukyo | okay. so if I have asterisk sitting behinid a nat, and a cisco phone on public IP. calls work, etc. the phone can receive audio, but cant send it. that means the phone cant connect-out via rtp on the rtp ports to the asterisk server. right? |
| 06:46.40 | Ukyo | works when its behind the nat with the server |
| 06:49.35 | *** join/#asterisk MGSsancho (n=user@adsl-67-125-157-194.dsl.irvnca.pacbell.net) |
| 06:49.40 | xtrvd | Generally that could be a RTP issue, yes. |
| 06:50.08 | Ukyo | is there anything else it can be? |
| 06:50.14 | Ukyo | I have tried setting the * server as the dmz |
| 06:50.20 | Ukyo | but that does not seem to help either |
| 06:50.35 | *** join/#asterisk MatsK (n=mk@141.221.181.62.in-addr.dgcsystems.net) |
| 06:50.39 | Ukyo | so it's making my head hurt a bit. :) |
| 06:51.18 | Ukyo | unless charter cable is now blocking those specific ports coming in >,> |
| 06:51.21 | Ukyo | trying to squeeze voip users |
| 06:51.25 | Ukyo | but I dont think they are |
| 06:51.44 | xtrvd | At the moment, I couldn't help ya.... I'm not proficient enough. Perhaps somebody else in the channel may be able to help though. |
| 06:55.30 | xtrvd | asterisk99: I found the problem.... it was because I was upgrading from * 1.0.4 to 1.2, and I neglected to use 'codecpriority-disabled' in my iax.conf |
| 06:58.25 | Ukyo | i recently did somethign similar |
| 06:58.31 | Ukyo | 1.0.6 -> 1,2 |
| 06:58.36 | Ukyo | what problem did that cause ? |
| 07:00.02 | xtrvd | It was in the IVR, my sound files would not play back |
| 07:00.44 | Ukyo | ah |
| 07:04.52 | Qwell | damania2: please don't msg me |
| 07:04.59 | MGSsancho | lol |
| 07:05.02 | damania2 | sorry |
| 07:05.11 | Qwell | digium has the dualspan card, and the channelbanks can be found elsewhere |
| 07:05.28 | damania2 | can u please provide me urls to the channelbanks. i know nothing about phone systems |
| 07:05.47 | Qwell | http://google.com/ |
| 07:05.52 | MGSsancho | lol |
| 07:06.01 | Qwell | adtran, rhino... |
| 07:06.11 | Qwell | ~google asterisk channelbank T1 |
| 07:06.19 | Qwell | ~wikis |
| 07:06.20 | jbot | i heard wikis is http://www.voip-info.org |
| 07:06.35 | MGSsancho | http://froogle.google.com/froogle?q=channel+bank+phone+cards&hl=en&lr=&safe=off&sa=N&tab=ff&oi=froogler |
| 07:06.53 | Qwell | oh, carrier access too |
| 07:06.55 | Qwell | adit |
| 07:09.11 | Grizzy-jbug | So, if one of us knew what we were doing with a generic FPLA, we'd have a T1/E1 - USB card going by now. :o) |
| 07:09.31 | Qwell | why? |
| 07:09.39 | Qwell | that would be silly |
| 07:09.40 | *** join/#asterisk EnErGy[CSDX] (n=energy@193.242.114.14) |
| 07:09.54 | *** part/#asterisk EnErGy[CSDX] (n=energy@193.242.114.14) |
| 07:14.16 | FuriousGeorge | so are these sangoma cars as wonderful as everyone is saying |
| 07:14.33 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:16.36 | FuriousGeorge | ~snom320 |
| 07:17.00 | FuriousGeorge | wondering if the only difference from the 360 is the ability to add the extra LEDs |
| 07:17.08 | Snake-Eyes | dont know they ship them here ;( |
| 07:17.24 | Snake-Eyes | *wont |
| 07:17.48 | tzafrir | trixter, here? |
| 07:18.02 | trixter | sometimes |
| 07:18.11 | *** join/#asterisk Fedoracore6 (n=FC$@60.50.138.230) |
| 07:18.21 | Fedoracore6 | hai all |
| 07:18.25 | trixter | whats up? |
| 07:18.46 | Fedoracore6 | i wann ask about my .. error |
| 07:19.57 | *** join/#asterisk EnErGy[CSDX] (n=energy@193.242.114.14) |
| 07:21.46 | Fedoracore6 | trixter : http://pastebin.com/577997 |
| 07:21.49 | Fedoracore6 | http://pastebin.com/577997 |
| 07:21.53 | Fedoracore6 | this my error |
| 07:22.10 | Fedoracore6 | i try use connection to database |
| 07:23.40 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 07:24.03 | *** part/#asterisk EnErGy[CSDX] (n=energy@193.242.114.14) |
| 07:25.17 | *** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net) |
| 07:25.42 | firestrm | man what i night im having.. |
| 07:26.17 | harryvv | wow, firestrm how are you doing? |
| 07:26.42 | firestrm | not bad now.. was a bit of a ride 30 min ago though |
| 07:26.54 | harryvv | did you leave that company on vancouver island? I called months and months ago and I think the line was disconected? |
| 07:27.26 | firestrm | ya, im working for motorola now |
| 07:27.35 | harryvv | I wondered what happened to you. DId you install a asterisk system for the BCDOE? |
| 07:27.48 | harryvv | ohh really doing what? |
| 07:27.58 | harryvv | moe |
| 07:28.00 | firestrm | radio engineer |
| 07:28.04 | harryvv | I mean MOE |
| 07:28.22 | firestrm | public safety stuff.. |
| 07:28.24 | harryvv | Really, did not know you are a radio engineer. |
| 07:29.02 | harryvv | I have two years collage in Aviation electronic and communication electronics my self. Have my FCC licence if I ever want to work on the equipment in the united states. |
| 07:29.26 | firestrm | i have electronics engineeing, mostly avionics eng exp.. but lots of radio exp as a result |
| 07:29.33 | harryvv | Do you repair to the component level? |
| 07:29.39 | harryvv | thats cool |
| 07:29.56 | firestrm | yes, to the nanoscopic component level.. |
| 07:30.00 | harryvv | I guess the IT/asterisk was just a side job then. |
| 07:30.06 | harryvv | yea no kidding, smt |
| 07:30.29 | firestrm | well, wasnt getting enough hours at AC, so i had to pad the paycheque somehow.. |
| 07:31.09 | delmar | need a simple solution to add an "0" to the front of the callerID for an incoming call... whats a good method.. anyone? |
| 07:31.29 | harryvv | Right |
| 07:31.41 | harryvv | did thay go under or change there number? |
| 07:31.55 | delmar | so.. the telco passes 1234567 but i need to see 01234567 on the SIP phones. |
| 07:31.58 | FuriousGeorge | ${EXTEN} = 0${CALLERIDNUM}? |
| 07:32.02 | delmar | no not a number change... |
| 07:32.09 | [av]bani | ~seen qwell |
| 07:32.12 | jbot | qwell <n=north@unaffiliated/qwell> was last seen on IRC in channel #asterisk, 22m 33s ago, saying: 'that would be silly'. |
| 07:32.27 | delmar | FuriousGeorge yep that. |
| 07:32.29 | harryvv | still flying? |
| 07:32.43 | firestrm | no, not lately.. more money in eng |
| 07:32.51 | FuriousGeorge | well your gonna set ${CALLERIDNUM} |
| 07:32.53 | harryvv | yea, it can get expensive |
| 07:33.23 | firestrm | AC slowly trickled my hours down to nothing.. all the senior pilots get the first crack at work, i get the hand me downs |
| 07:33.42 | harryvv | I see |
| 07:33.48 | harryvv | yea, aviation is so unstable |
| 07:34.24 | firestrm | harryvv, plus we have a baby on the way, so i want to stick around home more anyways.. |
| 07:35.16 | harryvv | I graduated in Avionics tech at the height of the aviation layoffs. Imagine a new grad appling for a job when in a 3 year period a million aviation personal across the united states are let go because of deregulation and fierce comp. |
| 07:37.09 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 07:38.42 | Fedoracore6 | http://pastebin.com/578003 |
| 07:39.45 | Grizzy-jbug | grrr, about all the code with MySQL hardwired in. I want to use SQLite. |
| 07:39.58 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 07:40.43 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 07:41.09 | *** join/#asterisk razu_ (n=razu@tln-kontor.norby.ee) |
| 07:41.44 | Nugget | yuck. I hate when developers assume that everyone loves mysql. |
| 07:42.12 | Nugget | "It's all I know, so clearly mysql is the best!" :) |
| 07:42.20 | RoyK | lol |
| 07:42.35 | Grizzy-jbug | Yes, re-grrrrr. |
| 07:42.59 | RoyK | but then, afaics mysql 5 seems to kick ass |
| 07:43.00 | Grizzy-jbug | add some generic database routine layer. |
| 07:43.25 | RoyK | it even has writable views :) |
| 07:43.43 | Grizzy-jbug | initialize, submit sql, submit and gather result... |
| 07:43.57 | Nugget | the mysql guys do indeed seem to be beginning to address the more critical failures in mysql, but they're still managing to hose it up pretty well. |
| 07:44.04 | RoyK | mysql < v5 sucks |
| 07:44.57 | Grizzy-jbug | generally, the "real" database based websites I see are insanely slow. |
| 07:44.59 | Nugget | a lot of the "fixes" in mysql have the same teeth as "php safe mode" in that they can be locally disabled at any time by any code in a connection. |
| 07:45.16 | Nugget | which makes them worse than useless, imho. |
| 07:45.54 | Grizzy-jbug | I was told the WikiPedia dumped MySQL for the plain old unix filesystem. |
| 07:46.19 | RoyK | Nugget: what's so bad about recent versions of mysql? it seems quite a bit better than even most of what postgresql has |
| 07:46.32 | RoyK | Grizzy-jbug: unix filesystem _is_ faster, of course, but requires more coding |
| 07:46.37 | Nugget | RoyK: it's still a total failure when it comes to data integrity enforcement. |
| 07:46.57 | RoyK | Nugget: even v5? |
| 07:47.03 | Nugget | yes, even in v5. |
| 07:47.07 | RoyK | Nugget: i thought v5 had all that :P |
| 07:47.18 | Nugget | it has it, optionally, and not in a form that can be enforced or relied on. |
| 07:47.41 | RoyK | nice |
| 07:47.59 | Nugget | to their credit, it's a lot better about warning you when it does bad things. |
| 07:48.18 | Nugget | but it still has that fun habit of taking 10000 rows on an insert, then warning you that 4 didn't fit. |
| 07:48.22 | Nugget | but not letting you know which 4. |
| 07:48.35 | RoyK | :) |
| 07:48.48 | Nugget | and, of course, only if you took the time to enable warnings |
| 07:49.59 | *** join/#asterisk af_ (n=af@ip-172-156.sn1.eutelia.it) |
| 07:50.45 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 07:56.03 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:58.05 | *** part/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 07:58.44 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 08:01.01 | *** join/#asterisk vgster (n=vg@host217-45-221-53.in-addr.btopenworld.com) |
| 08:01.54 | Grizzy-jbug | (deep yuk, about MySQL) |
| 08:05.18 | xtrvd | Does anybody have any experience in codec translation in Asterisk when it comes to voicemail? "Unable to find a codec translation path from gsm to slin" |
| 08:07.23 | Fedoracore6 | http://pastebin.com/578017 |
| 08:07.52 | Fedoracore6 | i didint know to slove this problem .... from last night i try but still fail |
| 08:07.59 | firestrm | gnite all.. im done lurking for the night.. and thanks to whom eve saved my a$$ on the nick glitch.. you know who you are ;) |
| 08:11.19 | diLLec | Fedoracore6 |
| 08:11.34 | diLLec | what does /var/log/asterisk/debug say ? |
| 08:12.15 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 08:15.34 | Fedoracore6 | yes |
| 08:16.15 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-219.claranet.co.uk) |
| 08:17.14 | Fedoracore6 | in var/log asterisk just have fail masseg queue_log |
| 08:17.23 | Fedoracore6 | what fail i must read |
| 08:17.54 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 08:18.51 | Fedoracore6 | Check debug. |
| 08:18.51 | Fedoracore6 | Mar 1 02:53:29 NOTICE[6245] config.c: Registered Config Engine mysql |
| 08:18.51 | Fedoracore6 | Mar 1 02:53:30 WARNING[6245] pbx_config.c: No closing parenthesis found? 'MYSQL(Query resultid ${connid}' |
| 08:18.51 | Fedoracore6 | Mar 1 02:53:31 WARNING[6245] cdr_addon_mysql.c: Unable to load config for mysql CDR's: cdr_mysql.conf |
| 08:19.36 | FuriousGeorge | i dont know, but it seems to want a ')' somewhere |
| 08:20.07 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 08:20.27 | Fedoracore6 | <PROTECTED> |
| 08:22.14 | Fedoracore6 | ddiLLec : in var/log /asterisk just have file masseg queue_log |
| 08:25.49 | Fedoracore6 | didLLec: what i must do i already setting in res_mysql.conf and cdr_mysql.conf |
| 08:26.02 | Fedoracore6 | but still cannot connect in database |
| 08:29.21 | Fedoracore6 | lor... |
| 08:37.33 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 08:38.28 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:40.01 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 08:41.13 | *** join/#asterisk TallAndy (n=sdad@82.163.6.213) |
| 08:41.44 | TallAndy | Hi does anyone have experience with PHPAGI using phpagi-asmanager.php and Originate? |
| 08:43.34 | Fedoracore6 | lor |
| 08:48.38 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.181) |
| 08:50.20 | *** join/#asterisk liew123 (n=goh_mail@60.49.6.190) |
| 08:50.40 | liew123 | hello |
| 08:51.05 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 08:51.30 | liew123 | it in asterisk group liao? |
| 08:51.50 | liew123 | why no person chat here? |
| 08:52.58 | real-dev | maybe everybody fallen asleep, or not yet awaken yet ;-) |
| 08:53.43 | TallAndy | :) |
| 08:53.51 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 08:53.52 | Kernel_core | liew123: hi |
| 08:55.59 | liew123 | hi |
| 08:56.18 | Nugget | http://flightaware.com/ <- new site's live (for those who had asked earlier) |
| 08:56.28 | liew123 | I think may be is the registration problem. It make me whole day to get in |
| 08:58.26 | *** join/#asterisk sack (n=sack@127.Red-81-38-35.dynamicIP.rima-tde.net) |
| 08:58.41 | *** join/#asterisk _Vile (n=vile@90.b160.bendtel.net) |
| 09:07.27 | MGSsancho | asterisk should get ported to this http://www.linuxdevices.com/news/NS3880195342.html |
| 09:07.30 | MGSsancho | id laugh |
| 09:08.12 | *** join/#asterisk nagl (n=nagl@137.208.4.162) |
| 09:09.02 | *** join/#asterisk vinsik (n=vinsik@gw-ff.verkkokauppa.com) |
| 09:12.34 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 09:15.34 | xtrvd | Does anybody know what a screen full of this means: WARNING[XXXX]: chan_sip.c:2527 sip_write: Asked to transmit frame type 2, while native formats is 4 (read/write = 4/4) |
| 09:16.00 | xtrvd | It occurs after attempting to dial an IP phone after connecting via the PSTN to an IVR on Asterisk. |
| 09:17.11 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 09:21.09 | *** join/#asterisk propagandhi (n=opera@d220-236-171-251.dsl.nsw.optusnet.com.au) |
| 09:21.43 | liew123 | where can find the most complete step to install asterisk |
| 09:21.55 | *** join/#asterisk Fedoracore6 (n=FC$@60.50.138.230) |
| 09:22.10 | xtrvd | ~docs |
| 09:22.16 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 09:22.16 | liew123 | xtrvd:you can run IVR on asterisk |
| 09:22.30 | xtrvd | liew123: Is that a question? |
| 09:23.00 | liew123 | xtrvd:em.. your IVR is writen by AGI or others language? |
| 09:23.55 | xtrvd | Standard dialplan markup, what ever that is. |
| 09:23.56 | propagandhi | liew123: you will find this isn't a very helpful channel methinks |
| 09:24.05 | liew123 | jbot:thanks |
| 09:24.05 | jbot | liew123: sure thing |
| 09:24.43 | heroine | hi pl |
| 09:24.51 | propagandhi | normally they just tell you to look at voip-info.org or other documentation, lads in here couldnt be stuffed twitching a brain cell |
| 09:27.03 | liew123 | now I install asterisk in fedora core 3 |
| 09:27.27 | liew123 | for the package group I sould select with package to install |
| 09:27.28 | xtrvd | propagandhi: I'm not sure you're entirely correct with that statement. It's only when people come here looking to install asterisk that we immediately throw the book(s) at them. We simply don't have the time to lead somebody through 'make install'.... If you can't take the time to learn how to read simple documentation, why should we take the time to help you? |
| 09:28.05 | liew123 | xtrvd:sorry I have try a lot of time. |
| 09:28.25 | xtrvd | liew123: Try this page: http://www.asterisk.org/download You'll get the correct syntax and you can download the latest 1.2 version of Asterisk. |
| 09:28.54 | propagandhi | xtrvd: I've been here on hundreds of occasions, and inevitably I got somebody do this |
| 09:28.56 | propagandhi | ~docs |
| 09:28.58 | jbot | somebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 09:29.05 | liew123 | xtrvd: I dun want be dumm to use software like asterisk@home. thanks |
| 09:29.57 | xtrvd | liew123: I didn't make any reference to Asterisk@home. I made reference to the packages that you need to install. |
| 09:31.02 | xtrvd | liew123: Use subversion to download the 1.2 branch, and you're set. |
| 09:31.08 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:37.55 | *** join/#asterisk mrkyr (n=bviitane@h24-207-83-214.cst.dccnet.com) |
| 09:38.55 | *** join/#asterisk Scum-Person (n=sdfsdf@scumperson.eu.org) |
| 09:43.49 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 09:46.01 | *** join/#asterisk Igbothom (n=HiltonT@203.206.170.99) |
| 09:50.20 | Scum-Person | anyone alive ? |
| 09:51.12 | RoyK | <PROTECTED> |
| 09:51.39 | Scum-Person | <PROTECTED> |
| 09:51.42 | Scum-Person | oops |
| 09:52.42 | trixter | what does the S stand for? |
| 09:52.51 | trixter | I get the rest of it, just not the 's' |
| 09:53.34 | Scum-Person | old school nick from years ago |
| 09:54.07 | Scum-Person | Scum-Bag was taken already :) |
| 09:54.42 | dpryo | Scum-Person: Where you an esp guy? |
| 09:54.51 | dpryo | s/h// |
| 09:54.52 | Scum-Person | yup |
| 09:55.13 | dpryo | I knew I'd seen that nick before :) |
| 09:55.21 | Scum-Person | hehe, still in #esp |
| 09:55.31 | Scum-Person | at work atm, i need some help with * |
| 09:55.54 | Scum-Person | can't coax it into connecting with sipgate :/ |
| 09:56.13 | trixter | whty not its easy |
| 09:56.30 | Scum-Person | cos i suck at it atm :) |
| 09:57.01 | *** join/#asterisk Gennaro (n=Gennaro@ppp-62-10-136-66.dialup.tiscali.it) |
| 09:57.02 | Gennaro | di |
| 09:57.03 | Gennaro | fi |
| 09:57.07 | Gennaro | hi |
| 09:57.22 | Gennaro | some one can say me something in * |
| 09:57.23 | Gennaro | ? |
| 09:57.25 | Scum-Person | i made a sip trunk for sipgate, using asterisk@home |
| 09:57.40 | Scum-Person | i made a few extensions which work fine etc local |
| 09:58.08 | trixter | http://pastebin.ca/44086 |
| 09:58.11 | Scum-Person | made a dial 9 to outside rule thing |
| 09:58.35 | trixter | what you meant to say is that you created a trunk in AMP becuase that is the web interface that is bundled with A@H |
| 09:58.44 | Scum-Person | ta |
| 09:58.52 | trixter | AMP does strange stuff even if not used with A@H |
| 09:59.07 | trixter | and that pastebin is not likely to help you becuase of it |
| 09:59.15 | RoyK | a@h is childish :) |
| 09:59.19 | Scum-Person | yeah i know, i can't find where it's generating .conf files to debug the dam thing :( |
| 09:59.26 | trixter | a@h isnt so much, amp however is |
| 09:59.40 | Scum-Person | gotta start somewhere, thought i'd start with noob stuff :) |
| 09:59.44 | trixter | odds are its the included files sip_additional.conf and extensions_additional.conf |
| 10:00.06 | Scum-Person | they are commented out in sip.conf mmm |
| 10:00.19 | trixter | the include? |
| 10:00.27 | trixter | what is the comment char? I bet you get it wrong |
| 10:00.30 | trixter | but that is just a guess |
| 10:00.30 | Gennaro | i have 2 question... |
| 10:00.31 | Scum-Person | # |
| 10:00.38 | trixter | that isnt a comment char |
| 10:00.41 | trixter | ; is a comment char |
| 10:00.45 | trixter | #include means to include a file |
| 10:00.46 | Scum-Person | mmm |
| 10:00.49 | Gennaro | how can i do to let musiconold play?!? |
| 10:00.52 | Scum-Person | good point |
| 10:00.52 | trixter | where-as include without a # means to include a context |
| 10:01.09 | trixter | Gennaro: perhaps calling the application musiconhold ? |
| 10:01.19 | Gennaro | ? |
| 10:01.23 | Gennaro | i see in sip.conf |
| 10:01.33 | trixter | exten s,1,musiconhold(default) |
| 10:01.34 | Gennaro | that is default or native etcc.. |
| 10:01.36 | trixter | or whatever you want |
| 10:01.38 | trixter | and toss in a => |
| 10:01.42 | Gennaro | oh.. |
| 10:01.54 | trixter | well if you want native you can specify the class as a parameter |
| 10:02.05 | trixter | or you can rename 'native' to be 'default' and comment out 'default' |
| 10:02.05 | Gennaro | i have an extension like 100,(SIP/100,20) |
| 10:02.11 | trixter | then all is native by default |
| 10:02.26 | Scum-Person | thats a bit better, irc = dose of common sense |
| 10:02.26 | trixter | how do you mean you have an extension like that? |
| 10:02.30 | Gennaro | if press hold on telephon |
| 10:02.30 | trixter | that isnt valid formatting |
| 10:02.52 | Gennaro | console say me that "name file" doesnt exist. |
| 10:03.52 | Gennaro | <PROTECTED> |
| 10:04.16 | acidchild | does it exist? |
| 10:04.29 | Gennaro | name was loaded.. |
| 10:04.35 | trixter | and does asterisk have permissions to read it - ie is asterisk running as a non-root user |
| 10:04.44 | Gennaro | root |
| 10:04.58 | acidchild | surely running asterisk as root is abit dodgy? ;\ |
| 10:04.59 | Gennaro | shuld i try with gsm file?!? |
| 10:05.06 | trixter | make sure that /var/lib/asterisk/mogmp3/fpm-world-mix* exists |
| 10:05.19 | Gennaro | exist! |
| 10:05.28 | trixter | yes perhaps with the ls command |
| 10:05.41 | acidchild | ls -la /var/lig/asterisk/mogmp3/fpm-world-mix |
| 10:05.42 | trixter | such as ls -l /var/lib/asterisk/mohmp3/fpm-world-mix.* |
| 10:05.58 | trixter | it looks for .format so you have to add a * |
| 10:06.00 | acidchild | and tell us the drwxr-xr-x |
| 10:06.03 | acidchild | part |
| 10:06.14 | *** part/#asterisk propagandhi (n=opera@d220-236-171-251.dsl.nsw.optusnet.com.au) |
| 10:06.16 | *** join/#asterisk Igbothom (n=HiltonT@office.quarkit.com.au) |
| 10:06.25 | Gennaro | -rw-r--r-- |
| 10:06.33 | acidchild | errrrr, weird |
| 10:06.35 | trixter | what is the last 4 or so letters |
| 10:06.40 | trixter | such as everything after the . |
| 10:06.49 | acidchild | what dot? lol |
| 10:06.53 | acidchild | oh |
| 10:06.58 | trixter | becuase its looking for a format |
| 10:07.05 | trixter | like playback(beep) will goto beep.gsm |
| 10:07.07 | trixter | not just 'beep' |
| 10:07.11 | acidchild | yeppers, i am totaly new to asterisk |
| 10:07.20 | acidchild | 100% newbie |
| 10:07.24 | acidchild | :) |
| 10:07.26 | trixter | or whatever you have it converted to (beep.gsm for example is a default, but can be transcoded into something else) |
| 10:07.48 | Scum-Person | Wed Mar 1 05:07:28 EST 2006 <-- quality ntp ? :/ |
| 10:07.58 | Gennaro | so i want to play an mp3 i need to convert i in gsm?!? |
| 10:07.59 | acidchild | Wed Mar 1 10:07:59 UTC 2006 |
| 10:08.02 | acidchild | errr |
| 10:08.07 | trixter | Gennaro: please show me the whole line from ls -l /var/lib/asterisk/mohmp3/fpm-world-mix.* |
| 10:08.09 | Gennaro | and quality?!? |
| 10:08.14 | acidchild | woah, this machine is in the whole wrong time zone!! |
| 10:08.16 | Scum-Person | server 0.pool.ntp.org |
| 10:08.28 | acidchild | mit.edu i would sync off |
| 10:08.30 | Nugget | UTC is never the wrong timezone. |
| 10:08.32 | Gennaro | pastebin? |
| 10:08.42 | trixter | its one line or should be so only paste one line |
| 10:08.47 | acidchild | Nugget: it is whem i aint an american |
| 10:08.52 | acidchild | and beleive in GMT |
| 10:08.55 | trixter | that ist he same as a pastbin url so its not bad, its when people paste more than one line it casues problems |
| 10:08.59 | Gennaro | <PROTECTED> |
| 10:08.59 | Gennaro | -rw-r--r-- 1 root root 2217563 27 feb 11:03 fpm-world-mix.mp3 |
| 10:09.00 | RoyK | don't paste > 3 lines.... |
| 10:09.05 | acidchild | the 'generating' of the UTC standard time was stupid |
| 10:09.06 | Nugget | "GMT" as a term is deprecated, and that has nothing to do with Americans. |
| 10:09.19 | Scum-Person | i think daylight time saving is stupid too |
| 10:09.21 | trixter | ok, do you have format_mp3 loaded? |
| 10:09.24 | RoyK | GMT is still descriptive |
| 10:09.25 | trixter | its from asterisk-addons |
| 10:09.26 | acidchild | Scum-Person: true |
| 10:09.26 | trixter | I bet you dont |
| 10:09.27 | acidchild | thats BST |
| 10:09.29 | iDunno | GMT is Greenwich Mean Time - and is the one true time ;) |
| 10:09.34 | trixter | and that is why it doesnt know that file format so it wont play it |
| 10:09.38 | RoyK | ~lart iDunno |
| 10:09.41 | acidchild | iDunno: damn right ;) |
| 10:09.42 | Scum-Person | british stupid time :) |
| 10:09.50 | Nugget | UTC is the one true time. GMT is a curious, vestigial phrase. |
| 10:09.51 | iDunno | UTC isn't far out, though. |
| 10:09.58 | acidchild | hah, atleast we didn't make utc just so we could be diffren't |
| 10:10.01 | acidchild | utc is like |
| 10:10.07 | *** join/#asterisk mzo (n=moz@ool-435193b3.dyn.optonline.net) |
| 10:10.10 | trixter | Gennaro: d oyou understand? |
| 10:10.13 | acidchild | 0.0000028ms off gmt |
| 10:10.14 | Nugget | "GMT" is a colloquialism. |
| 10:10.15 | iDunno | UTC isn't true time - it's broken and trying to be a replacement for something that actually works. |
| 10:10.22 | mzo | is there anyone who is an expert with FWD and aah? I'm lost with trying to figure out what i broke :P |
| 10:10.27 | mzo | nugget lives! |
| 10:10.38 | *** join/#asterisk X-Rob_ (n=Rob@dsl-220-235-91-96.vic.westnet.com.au) |
| 10:10.43 | mzo | we should all live on switch time |
| 10:10.44 | acidchild | :p |
| 10:10.46 | mzo | er, swatch |
| 10:10.49 | Nugget | swatch internet beats! |
| 10:10.52 | mzo | that way it'd be like 11359385.10 |
| 10:10.53 | denon | or unix time |
| 10:11.02 | mzo | YES! swatch beats, and if you don't like someone you can beat them |
| 10:11.08 | acidchild | unix time :P |
| 10:11.16 | Nugget | unix epoch time is in UTC ;) |
| 10:11.20 | Nugget | so there. |
| 10:11.20 | trixter | ok well I am going out for a minute, if gennaro comes back tell him to get asterisk-addons and build format_mp3 and install that (its generally safe to make all install) |
| 10:11.23 | mzo | can you imagine a unix time watch? It'd be an armored armband that's like, 8 inches long |
| 10:11.31 | RoyK | lol |
| 10:11.31 | trixter | once he does that he can natively play mp3 files |
| 10:11.39 | Gennaro | i not loaded asterisk addon |
| 10:11.52 | trixter | to play mp3s you must load format_mp3 |
| 10:11.52 | mzo | but really is anyone an fwd expert? I can't figure out this mysterious code 29, and no one seems to know either |
| 10:11.57 | trixter | or asterisk has no idea how to process an mp3 |
| 10:11.57 | Gennaro | maybe crush?!? |
| 10:12.04 | Gennaro | ah.. |
| 10:12.08 | Gennaro | i'm going... |
| 10:12.27 | acidchild | <PROTECTED> |
| 10:12.30 | acidchild | better :P |
| 10:12.33 | trixter | get it, untar, cd, make all install, restart asterisk, poof it should work :) |
| 10:12.35 | acidchild | Wed Mar 1 10:12:35 GMT 2006 |
| 10:12.40 | mzo | trixter rocks :P |
| 10:12.57 | trixter | nah I pebble |
| 10:12.58 | trixter | much smaller |
| 10:13.16 | mzo | haha. i'm dialing all my 800 calls thanks to your mad pebblz skills :P |
| 10:13.39 | trixter | thanks |
| 10:14.02 | *** join/#asterisk Tili (n=Tili@82-217-236-131.cable.quicknet.nl) |
| 10:14.05 | mzo | now if i can get fwd to work :P |
| 10:14.39 | trixter | if only amp was as good as the os X guis |
| 10:14.45 | trixter | then no one would have any questions |
| 10:14.49 | trixter | those are VERY slick |
| 10:14.50 | acidchild | shame OSX stinks :P |
| 10:14.51 | mzo | how easy are the osx guis? |
| 10:14.55 | trixter | ~praise benjk |
| 10:14.57 | jbot | All hail benjk! |
| 10:15.02 | mzo | i have osx sitting around. :P would i be able to get it working easily? |
| 10:15.09 | trixter | yes |
| 10:15.10 | trixter | VERY |
| 10:15.13 | Nugget | asterisk runs just fine in OS X. |
| 10:15.15 | mzo | i'll do that then. |
| 10:15.21 | mzo | heh, what about zaptel cards? |
| 10:15.22 | trixter | www.astmasters.net |
| 10:15.26 | Nugget | nope, no zaptel. |
| 10:15.27 | trixter | get their distro if you wanna do mac |
| 10:15.30 | denon | why .. would you want to run asterisk in a box with a gui |
| 10:15.35 | trixter | yeah no hardware or timers afaik |
| 10:15.38 | denon | why would you want to run * in anything but a stripped down appliance |
| 10:15.39 | mzo | this is to configure the box |
| 10:15.45 | mzo | i'm using aah, and amp is uh, bad :P |
| 10:15.57 | trixter | denon: most of the asterisk installs are small home installs basically glorified answering machines |
| 10:16.06 | denon | yick |
| 10:16.07 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 10:16.26 | trixter | yeah well ... I think more calls are processed by higher end asterisk servers but more installs are lower end glorified answering machines |
| 10:16.27 | mzo | trixter, yeah, but im a noob, and i need something to learn from. I'm also buying voip service to call some places, too. :P |
| 10:16.52 | mzo | i've also gotten really good at breaking asterisk :P |
| 10:18.13 | mzo | and now if i can make a lot of 800 calls... |
| 10:18.36 | Gennaro | i installed addon |
| 10:18.43 | Gennaro | and error is no more.. |
| 10:18.45 | Gennaro | argh... |
| 10:19.00 | Gennaro | is the same i hear nothing |
| 10:21.49 | trixter | did you answer the channel? if its voip that should be taken care of but its always good to answer the call first |
| 10:21.59 | trixter | if its zap you must answer it first |
| 10:22.10 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 10:22.39 | Gennaro | i installed addon for mp3 an now i havent errors in console... but i don't hear music i hear silence |
| 10:22.48 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 10:22.50 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 10:22.54 | Gennaro | why? |
| 10:23.07 | mzo | that's the kind of stuff that scares me :p |
| 10:23.11 | trixter | do you have a sip or iax2 guest account I can call in and you can put me on hold? |
| 10:23.33 | Gennaro | i have 2 sip phone |
| 10:23.44 | Gennaro | if i call exten 101 |
| 10:23.49 | Gennaro | call is done |
| 10:24.34 | Gennaro | i dont know how |
| 10:24.38 | trixter | do you have reinvite=yes ? |
| 10:24.40 | trixter | for those phones |
| 10:24.43 | Gennaro | no |
| 10:24.49 | Gennaro | do u want it? |
| 10:25.17 | trixter | do you see anything like -- Started music on hold, class 'default', on SIP/25-c727 |
| 10:25.25 | trixter | whati s your verbosity level? |
| 10:25.34 | Gennaro | verbosity? |
| 10:25.41 | Gennaro | what is it?!? |
| 10:25.42 | Gennaro | :) |
| 10:25.44 | trixter | how many 'v' did you specify when you did asterisk -r ? |
| 10:25.45 | Gennaro | sorry.. |
| 10:25.52 | Gennaro | vvvg |
| 10:26.06 | trixter | ok, your verbosity is 3 you should see something like what I pasted earlier |
| 10:26.11 | trixter | 'started music on hold ...' |
| 10:26.32 | mzo | heh, i couldn't put my call on hold, unless i call someone |
| 10:26.42 | Gennaro | if u say me how can i get my ip.. |
| 10:26.51 | Gennaro | i let you register |
| 10:26.53 | trixter | www.whatismyip.con |
| 10:28.03 | Gennaro | in linux isnt something like ipconfig |
| 10:28.04 | Gennaro | ? |
| 10:28.20 | real-dev | Gennaro: sure |
| 10:28.26 | trixter | in your 'default' dontext add a extension for your phones |
| 10:28.29 | trixter | then I can call you |
| 10:28.37 | trixter | probably ifconfig ipconfig is windows |
| 10:28.40 | trixter | I have your IP{ |
| 10:28.56 | trixter | but I cant call you becuase in [default] you dont have an extension mapping to your phones |
| 10:29.00 | real-dev | Gennaro: ifconfig -a |
| 10:29.08 | trixter | so add exten => 101,1,dial(SIP/101,90) |
| 10:29.10 | trixter | then I can |
| 10:29.48 | *** join/#asterisk EnErGy[CSDX] (n=energy@193.242.114.14) |
| 10:30.10 | *** part/#asterisk EnErGy[CSDX] (n=energy@193.242.114.14) |
| 10:30.58 | trixter | let me know when you add that to default and do an 'extensions reload' |
| 10:31.38 | Gennaro | 62.10.136.66 reg @5060 rtp 10000 |
| 10:32.04 | Gennaro | 100, 100, "" |
| 10:32.05 | trixter | just add the extension like I requested and I can call you |
| 10:32.09 | trixter | that is the only thing I need :) |
| 10:32.28 | Gennaro | register a phone on me |
| 10:32.49 | acidchild | yay my voice over IP stuff came :> |
| 10:32.49 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 10:32.50 | Gennaro | i dont know how can i do an extension to let call in.. |
| 10:32.55 | trixter | I'd rather not, but I will call you if you set up an extension to route the call |
| 10:33.06 | Gennaro | ah.. i'm surfing @56 |
| 10:33.08 | trixter | in [default] add exten => 101,1,dial(SIP/101,90) |
| 10:33.11 | trixter | then do an extensions reload |
| 10:33.24 | Gennaro | already |
| 10:33.29 | trixter | in extensions.conf that is |
| 10:33.45 | trixter | Call rejected by 62.10.136.66: No such context/extension |
| 10:35.22 | *** join/#asterisk pawal (n=pawal@212.247.14.36) |
| 10:35.25 | trixter | did you add the extension in extensions.conf to [default] ? |
| 10:35.26 | Gennaro | if you need context "sales" |
| 10:35.56 | Gennaro | but u dont need it. |
| 10:36.24 | trixter | check the context 'guest' is in in iax.conf |
| 10:36.28 | trixter | that is what I am coming in as |
| 10:36.43 | Gennaro | what i should check? |
| 10:36.50 | Gennaro | i'm going to open iax.conf |
| 10:37.04 | trixter | if its not default then the extension needs to be added to whatever context it is in, remember guest doesnt require a password so it should only be able to dial local entities and nothing else or people can make calls on your service and you get charged |
| 10:38.22 | Gennaro | [guest],type=user,context=default,callerid="Guest IAX User" |
| 10:38.42 | trixter | ok, so in extensions.conf you need to locate the [default] context |
| 10:38.54 | Gennaro | i'm going... |
| 10:38.58 | trixter | then add exten => 101,1,dial(SIP/101,90) in that context |
| 10:39.13 | Gennaro | ok just a minute |
| 10:40.11 | acidchild | anyone use vonage here? |
| 10:40.29 | denon | haha |
| 10:40.37 | acidchild | ;( |
| 10:40.42 | acidchild | it works! :P |
| 10:40.54 | denon | acidchild: yeah .. so does a hammer on your thumb |
| 10:41.02 | denon | www.nufone.net - grow up :) |
| 10:41.02 | acidchild | hey... |
| 10:41.04 | trixter | unless you opt into the higher paid business plan vonage isnt BYOD so it tends to be a little more difficult to work with |
| 10:41.26 | acidchild | denon: it came out cheaper to go with vonage... cos of the kit required. |
| 10:41.29 | acidchild | plus i aint in the US |
| 10:41.43 | denon | why would you need to be in the US? |
| 10:41.56 | acidchild | i donno, i aint sure what it is |
| 10:42.01 | acidchild | nufone.net is down |
| 10:42.04 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 10:42.10 | denon | acidchild: no its not .. |
| 10:42.14 | acidchild | yes it is |
| 10:42.18 | denon | they're just not accepting new registrations for a short while |
| 10:42.26 | denon | try back in a couple days, well worth the wait |
| 10:42.27 | acidchild | yeah, it has no information about it though |
| 10:42.33 | Gennaro | trixter, so i do |
| 10:42.37 | trixter | its been that way for a few days |
| 10:42.39 | denon | take my word for it, try back in a couple days |
| 10:42.39 | acidchild | just got vonage so i am going to have a play |
| 10:42.51 | trixter | Gennaro: ok, now in the asterisk console type extensions reload |
| 10:43.13 | Gennaro | asterisk ready |
| 10:43.39 | mzo | Please bear with us during this transition period. We expect to launch an entire new website on or about March 1st. |
| 10:43.42 | mzo | IT's 3/1!! :P |
| 10:43.43 | trixter | I hear music |
| 10:43.52 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 10:44.03 | Gennaro | why i not?!? |
| 10:44.12 | trixter | so if you dont that means that your phones are directly talking to each other (most likely) or you have audio issues with those phones |
| 10:44.31 | Gennaro | i hang up |
| 10:44.41 | Gennaro | so |
| 10:44.48 | trixter | my guess is that the RTP streams are direct to each other, possibly a 'canreinvite' or a 'reinvite' being set to 'yes' |
| 10:44.50 | acidchild | denon: could i keep the same thing that vonage spent me? |
| 10:44.54 | Gennaro | what i should do to hear music on sip |
| 10:44.55 | trixter | make sure both are no in your sip.conf definitions |
| 10:44.57 | Gennaro | not on IAX |
| 10:45.05 | acidchild | denon: if i went with nufone? |
| 10:45.11 | denon | acidchild: you mean you ATA? |
| 10:45.14 | acidchild | yep |
| 10:45.15 | denon | with nufone you wouldnt need to |
| 10:45.19 | denon | asterisk does it all |
| 10:45.23 | acidchild | err? |
| 10:45.24 | denon | vonage just locks you into their ATA |
| 10:45.29 | trixter | no |
| 10:45.35 | acidchild | no it doesn't |
| 10:45.36 | Gennaro | i can load a firmware to let phone in IAX2 |
| 10:45.37 | trixter | vonage has a higher paid business plan they dont advertise that lets you BYOD |
| 10:45.40 | mzo | can you call vonage using an iax phone? |
| 10:45.41 | denon | well, it does on the consumer plans |
| 10:45.50 | trixter | they have 3 plans, residential, business and big business |
| 10:45.51 | mzo | like is there an iax or sip way of calling them? |
| 10:45.54 | Mavvie | hmmm... sometimes you want a "sip ping" command |
| 10:45.54 | trixter | only big business lets you BYOD |
| 10:45.59 | denon | mzo: SIP yes |
| 10:46.02 | trixter | its in their TOS if you doubt :P |
| 10:46.05 | acidchild | denon: asterisk? |
| 10:46.07 | acidchild | how so? |
| 10:46.07 | mzo | is there a config page for that? |
| 10:46.16 | acidchild | that means getting new hardware i am imagining :P |
| 10:46.21 | Gennaro | so.. |
| 10:46.23 | mzo | like make it so i dial extension whatever and then dial a number to get them without using pstn? |
| 10:46.23 | denon | acidchild: no hardware, just over ethernet |
| 10:46.28 | acidchild | oh? |
| 10:46.34 | denon | that's the whole idea |
| 10:46.34 | acidchild | how does it link to the phone then? |
| 10:46.41 | denon | oh, in that respect .. |
| 10:46.41 | trixter | Gennaro: generally no, make sure in sip.conf you have both canreinvite=no and reinvite=no |
| 10:46.51 | Gennaro | i can send u with paste bin configuration of Sip |
| 10:46.52 | denon | yeah, you need an fxs device or sip phone, or at least a softphone |
| 10:46.58 | Gennaro | i'm going to see... |
| 10:46.58 | acidchild | exacly |
| 10:46.59 | trixter | edit out your passwords |
| 10:47.06 | denon | but well worth it |
| 10:47.10 | denon | to have a real pbx :) |
| 10:47.14 | acidchild | :) |
| 10:47.20 | acidchild | i will consider it |
| 10:49.05 | acidchild | denon: and i am sure there is away of flashing the firm ware on the box's |
| 10:49.09 | acidchild | even if they do lock it :) |
| 10:49.30 | denon | depending on the device .. |
| 10:49.31 | denon | not easily |
| 10:49.39 | acidchild | they are now made by cisco, it can't be that secure :P |
| 10:49.41 | denon | ie: much more complicated than you could pull off |
| 10:49.54 | acidchild | you judgeing my capabilitys? :> |
| 10:49.56 | acidchild | though so |
| 10:50.04 | denon | acidchild: well, by what you said .. |
| 10:50.08 | denon | yes |
| 10:50.12 | acidchild | I know _nothing_ by voip |
| 10:50.13 | acidchild | :) |
| 10:50.16 | acidchild | i can learn. |
| 10:50.23 | denon | because the cisco ATAs have a very advanced locking mechanism now |
| 10:50.42 | acidchild | yeppers. |
| 10:51.03 | Scum-Person | 'advanced' *is* a relative term though :) |
| 10:51.17 | Gennaro | trixter i do so and i reload asterisk |
| 10:51.21 | Gennaro | -vvvg |
| 10:51.24 | denon | eh, if it involves breaking a relatively secure encryption to recover a $75 device .. |
| 10:51.26 | Gennaro | but is the same... |
| 10:51.28 | denon | I'd say thats advanced |
| 10:51.29 | Gennaro | ARGHH |
| 10:52.02 | acidchild | denon: it can be done |
| 10:52.15 | denon | google about it |
| 10:52.19 | denon | this isnt a new discussion |
| 10:52.26 | Scum-Person | but is it worth the bother? i bet not.. :/ |
| 10:52.29 | *** join/#asterisk mrdigital (n=Mrdgitia@pool-68-163-50-110.phil.east.verizon.net) |
| 10:52.34 | mrdigital | Hello |
| 10:52.41 | acidchild | denon: its just a disussion at the moment, |
| 10:52.52 | acidchild | i am not going to run off and research the matter this second |
| 10:52.55 | Fedoracore6 | hai all |
| 10:52.56 | mrdigital | is anyone here good with programming Asterisk? |
| 10:53.04 | Fedoracore6 | i already follow step bye step in http://www.voip-info.org/wiki-Asterisk+cdr+mysql |
| 10:53.13 | Scum-Person | nope, so i'll be not helpful whasoever yay |
| 10:53.13 | Fedoracore6 | but still have same error |
| 10:53.16 | heroine | mrdigital: programming asterisk ? in the core program ? or agi stuff ? |
| 10:53.26 | mrdigital | anything in * |
| 10:53.42 | acidchild | thats a bit of a big question |
| 10:53.51 | acidchild | :P |
| 10:53.55 | mrdigital | well lemme rephrase who can code what |
| 10:53.57 | mrdigital | :) |
| 10:54.09 | mrdigital | brb |
| 10:54.17 | acidchild | -.- |
| 10:54.34 | mrdigital | man that was fun |
| 10:54.34 | Fedoracore6 | res_config_mysql.so] => (MySQL RealTime Configuration Driver) |
| 10:54.35 | Fedoracore6 | <PROTECTED> |
| 10:54.35 | Fedoracore6 | Mar 1 03:44:22 ERROR[6846]: res_config_mysql.c:615 mysql_reconnect: MySQL RealTime: Failed to connect database server ivr on localhost. Check debug for more info. |
| 10:54.38 | heroine | i've only done some minor patch in app_voicemail .. and AGI's :) |
| 10:54.39 | mrdigital | drink went down the wrong pipe |
| 10:54.57 | Fedoracore6 | plase some budy help me try to slove |
| 10:55.00 | acidchild | your not ment to sit on the bottle |
| 10:55.06 | Mavvie | mrdigital: yeah, channel mis-connections are very tricky. |
| 10:55.07 | mrdigital | ????? |
| 10:55.10 | Rhizome | Fedoracore6: enable debug in /etc/asterisk/logger.conf and check the log files. |
| 10:55.15 | mrdigital | heroine: pm? |
| 10:55.39 | heroine | mrdigital: pm ? |
| 10:55.44 | trixter | Gennaro: where is that pastebin? |
| 10:55.45 | mrdigital | ok |
| 10:55.48 | Gennaro | http://pastebin.ca/44088 |
| 10:55.53 | acidchild | query i think i beleive he is asking for. |
| 10:55.54 | acidchild | :P |
| 10:56.17 | mrdigital | pm = irc code for privte message aka query |
| 10:56.28 | Fedoracore6 | ok |
| 10:56.39 | mover | lalalala |
| 10:56.40 | heroine | aaahh :) |
| 10:56.45 | mover | aka mgs |
| 10:56.52 | mover | mag :P |
| 10:57.00 | mover | msg :-P |
| 10:57.04 | heroine | i was thinking about something like PostMaster or .pm for perl modules :) |
| 10:58.01 | Fedoracore6 | Rhizome: hemm i already find the logger.conf |
| 10:58.06 | Fedoracore6 | so what should i do |
| 10:58.07 | Fedoracore6 | ;debug => debug |
| 10:58.07 | Fedoracore6 | console => notice,warning,error |
| 10:58.07 | Fedoracore6 | ;console => notice,warning,error,debug |
| 10:58.07 | Fedoracore6 | messages => notice,warning,error |
| 10:58.07 | Fedoracore6 | ;full => notice,warning,error,debug,verbose |
| 10:58.09 | Fedoracore6 | ;syslog keyword : This special keyword logs to syslog facility |
| 10:58.11 | Fedoracore6 | ; |
| 10:58.13 | Fedoracore6 | ;syslog.local0 => notice,warning,error |
| 10:58.13 | acidchild | heroine: YOu don't go on AIM enough!!! |
| 10:58.14 | trixter | Gennaro: that should work, you dont use nat? given the IP that you have I would think that you do ... |
| 10:58.15 | Fedoracore6 | ; |
| 10:58.18 | acidchild | STOP flooding |
| 10:58.50 | trixter | ~pb |
| 10:58.51 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 10:58.51 | heroine | acidchild: irc is too much time consumming to allow me to loose more time on aim :) |
| 10:58.57 | FuriousGeorge | Fedoracore6: has been trying to get this going all night :) |
| 10:59.06 | acidchild | heroine: oh noes! |
| 10:59.13 | acidchild | download msn aim and yahoo today! |
| 10:59.31 | acidchild | emurse youself in the wonderful world of camwhores and retards |
| 10:59.36 | acidchild | =\ |
| 10:59.45 | Gennaro | what |
| 10:59.46 | Gennaro | ? |
| 10:59.57 | acidchild | i am being sarcastic |
| 10:59.57 | Fedoracore6 | yess |
| 10:59.59 | Gennaro | i need to set nat = yes?!? |
| 11:00.05 | trixter | I dont know |
| 11:00.06 | heroine | acidchild: noway ! :) |
| 11:00.08 | trixter | do you use nat? |
| 11:00.10 | acidchild | heroine: haha :P |
| 11:00.36 | Fedoracore6 | FuriousGeorge i must try |
| 11:00.38 | Gennaro | shuod i set default ip for phone?!? |
| 11:00.47 | Rhizome | Fedoracore6: uncoment the line starting with full |
| 11:00.50 | trixter | what ip is one of your phones? |
| 11:00.52 | Fedoracore6 | to prove my lecture i not stupid student |
| 11:00.53 | acidchild | Gennaro: go back to msn |
| 11:00.59 | Rhizome | Fedoracore6: then tail -f /var/log/asterisk/full or something, while you watch asterisk |
| 11:01.00 | acidchild | please stop pleaking ??!?!?!? is annoying as hell |
| 11:01.10 | acidchild | :P |
| 11:01.15 | FuriousGeorge | Fedoracore6: hope you didnt wait till the last minute |
| 11:01.36 | acidchild | wooops |
| 11:01.47 | Rhizome | eww :/ |
| 11:01.55 | Fedoracore6 | Fedoracore6: then tail -f /var/log/asterisk/full or something, while you watch asterisk |
| 11:02.08 | Fedoracore6 | i didint understand waht rizome say |
| 11:02.13 | acidchild | Rhizome: indeedie |
| 11:02.18 | Fedoracore6 | any simple word can make me understand |
| 11:02.25 | FuriousGeorge | Fedoracore6: man tail |
| 11:02.48 | FuriousGeorge | <PROTECTED> |
| 11:03.11 | *** join/#asterisk puzzled (n=yeahrigh@62.45.11.228) |
| 11:03.51 | Fedoracore6 | didint have file full |
| 11:04.02 | puzzled | morning |
| 11:04.23 | Rhizome | Fedoracore6: if you read logger.conf it sayes how to reload the logging system, or just restart asterisk :P |
| 11:04.59 | Fedoracore6 | only have cdr-csv file cdr custom even_log massages queue_log |
| 11:05.13 | FuriousGeorge | Fedoracore6: he just told you, set it up in logger.conf |
| 11:05.14 | Fedoracore6 | that file have in /var/log/asterisk |
| 11:05.43 | *** join/#asterisk frenzy (n=frenzy@196.45.144.41) |
| 11:05.54 | frenzy | 11:05:02 ERROR[18708]: pbx.c:1406 ast_func_write: Function CDR not registered |
| 11:05.59 | frenzy | what does that mean? |
| 11:06.41 | acidchild | you know there is a coffee pepsi |
| 11:06.42 | acidchild | haha |
| 11:06.48 | acidchild | i got given some for my birthday today |
| 11:06.49 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 11:07.15 | frenzy | ? |
| 11:07.17 | Mavvie | bugs.digium.com seems euhm... down |
| 11:07.28 | Fedoracore6 | ok ok try read about longger.conf |
| 11:07.44 | acidchild | thats fucking fowl |
| 11:07.45 | acidchild | lol |
| 11:07.51 | agkhram | frenzy: chech cdr function was loaded. asterisk -vvvc and scroll up |
| 11:08.13 | Fedoracore6 | ${connid}' |
| 11:08.14 | Fedoracore6 | Mar 1 03:44:24 WARNING[6846] cdr_addon_mysql.c: Unable to load config for mysql CDR's: cdr_mysql.conf |
| 11:08.22 | Fedoracore6 | i find this i debug |
| 11:08.28 | puzzled | Mavvie: couldn't get to it either |
| 11:09.01 | puzzled | Fedoracore6: it's onl;y a warning. if you don't use mysql just ignore it. warnings are only informational |
| 11:09.05 | Mavvie | puzzled: and that while I have an important one line patch :-P |
| 11:09.22 | puzzled | Mavvie: for 1.2? |
| 11:09.30 | Mavvie | puzzled: no, for my mute-patch. |
| 11:09.32 | Mavvie | :-) |
| 11:09.51 | Fedoracore6 | ok .. puzzled acctully my system wanna me connect in my database |
| 11:09.52 | mrdigital | Pepsi Kona |
| 11:10.15 | mrdigital | acidchild: its discontinued |
| 11:10.41 | frenzy | <PROTECTED> |
| 11:10.43 | frenzy | <PROTECTED> |
| 11:10.43 | frenzy | <PROTECTED> |
| 11:10.48 | RoyK | "The good thing about standards is that there are so many of them to choose from" |
| 11:11.05 | mrdigital | http://en.wikipedia.org/wiki/Pepsi_Kona |
| 11:11.15 | mrdigital | all the different pepsi varityes |
| 11:11.38 | mrdigital | hey acidchild: you from PA? |
| 11:11.59 | RoyK | a quick test with four ATAs and a softphone shows only one ATA and the softphone (x-pro) did regular lookups in case sip server's IP address was changed :( |
| 11:12.35 | Fedoracore6 | http://pastebin.com/578160 |
| 11:12.41 | Fedoracore6 | this is my extensions fail |
| 11:12.59 | Fedoracore6 | i wanna make this system touch tone work |
| 11:13.42 | Fedoracore6 | FuriousGeorge: i dont know what i must doing in logger.conf |
| 11:13.43 | mrdigital | Fedoracore6: what you tryinng to accommplish? |
| 11:13.54 | Fedoracore6 | hemm first |
| 11:13.54 | agkhram | frenzy: asterisk>show functions |
| 11:14.06 | Fedoracore6 | i must build one syste, its call touch tones sytem |
| 11:14.11 | mrdigital | Explain |
| 11:14.25 | Fedoracore6 | ( sorry my english not so good but i try) |
| 11:14.27 | Fedoracore6 | ok |
| 11:14.45 | Fedoracore6 | for next semester .. my student |
| 11:14.54 | mrdigital | ok |
| 11:14.54 | Fedoracore6 | just call extensions 110 |
| 11:15.09 | Fedoracore6 | anthen can register the subject |
| 11:15.16 | frenzy | CHECKSIPDOMAIN CHECKSIPDOMAIN(<domain|IP>) Checks if domain is a local domain |
| 11:15.17 | frenzy | SIPCHANINFO SIPCHANINFO(item) Gets the specified SIP parameter from the current channel |
| 11:15.17 | frenzy | SIPPEER SIPPEER(<peername>[:item]) Gets SIP peer information |
| 11:15.17 | frenzy | SIP_HEADER SIP_HEADER(<name>) Gets or sets the specified SIP header |
| 11:15.17 | frenzy | IAXPEER IAXPEER(<peername|CURRENTCHANNEL>[: Gets IAX peer information |
| 11:15.21 | Fedoracore6 | ttuely the phone |
| 11:15.28 | mrdigital | hey frenzy: use pastebin |
| 11:15.36 | frenzy | *sorry* |
| 11:15.41 | Fedoracore6 | truely the phone |
| 11:16.02 | Fedoracore6 | so when some call useing X-lite press 110 |
| 11:16.31 | Fedoracore6 | the say system say " welcome to touch tones system plase press thepassword" |
| 11:16.52 | Fedoracore6 | when student press the password correct |
| 11:17.27 | Fedoracore6 | then the TROS sya again " press 1 tu registration and 2 for drop subject" |
| 11:17.31 | *** join/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at) |
| 11:17.35 | nicox | Hello |
| 11:17.41 | Fedoracore6 | hemm that all mrdigital |
| 11:17.49 | mrdigital | so you want |
| 11:17.55 | mrdigital | a student to call 110 |
| 11:18.07 | mrdigital | enter the right password and press 1 to reg a phone or 2 to drop the phone? |
| 11:18.29 | nicox | Did anyone seen a problem with faxes on asterisk 1.2.x? |
| 11:18.55 | Fedoracore6 | yes |
| 11:19.14 | Fedoracore6 | the passwors i save i database |
| 11:19.43 | nicox | how to solve this problem? |
| 11:21.03 | nicox | if i fax over asterisk, there are always someline which are not seen |
| 11:22.19 | *** join/#asterisk digime (n=digime@ip68-101-196-93.sd.sd.cox.net) |
| 11:24.45 | *** join/#asterisk NassoR (n=root@step.funcitec.rct-sc.br) |
| 11:25.11 | X-Rob_ | mrdigital, he wants to write an AGI, but doesn't know how to code. I spent 3 days with him a couple of weeks ago. |
| 11:25.18 | X-Rob_ | (this is what he wants to achieve) |
| 11:25.39 | X-Rob_ | he wants to let students register for courses at uni |
| 11:26.06 | X-Rob_ | for some reason he thinks that realtime is a way to achieve this. |
| 11:27.30 | Fedoracore6 | yes that right what the X-rob say |
| 11:27.35 | Fedoracore6 | he alot help me |
| 11:28.06 | Scum-Person | 3 days is a lot of time spent :/ |
| 11:28.27 | Scum-Person | whats wrong with a website to reg poeple on courses :) |
| 11:28.37 | Scum-Person | or old fashioned paper even |
| 11:29.51 | acidchild | mrdigital: PA no |
| 11:30.18 | nicox | is there any known problem with fax over asterisk 1.2.x? |
| 11:30.40 | mrdigital | x-rob: pm? |
| 11:30.54 | Fedoracore6 | hemm ... i wanna try using voip application ... |
| 11:31.18 | Fedoracore6 | i wanna explorer and teach other people about this technologi |
| 11:32.26 | Scum-Person | ya plenty of neat things u can do, remember always keep it simple though |
| 11:32.48 | Scum-Person | simple = good, not simple = confused poeple = poeple not impressed |
| 11:32.56 | Scum-Person | and poeple confuse easy :/ |
| 11:33.51 | Fedoracore6 | oic .. ok |
| 11:34.19 | Fedoracore6 | i must do a manual .. to guide user use the system first |
| 11:35.16 | *** join/#asterisk mut (n=animenod@65.111.201.79) |
| 11:37.06 | Scum-Person | a good system needs no manual |
| 11:37.13 | mut | argh |
| 11:37.18 | mut | whats this positon worth |
| 11:37.22 | mut | It looks like the Web Developer position in Traverse City is going to open up. Are you still available and interested? It is a full time permanent position doing web development with ASP and SQL. They also do quite a bit with PHP but PHP is not required. ASP and SQL are required for this position. They are a solid growing organization. They are small but on a pretty agressive growth track. This position would be part of a team. |
| 11:37.40 | Scum-Person | m$ job yay |
| 11:38.41 | *** join/#asterisk fulgas (n=fulgas@209.8.233.247) |
| 11:41.04 | mut | yea |
| 11:41.07 | mut | i dunno |
| 11:41.38 | mut | traverse city is a more expensive place to live, dunno what that kinda job would pay there tho |
| 11:41.42 | mut | any estimations? |
| 11:41.47 | Fedoracore6 | ;D |
| 11:44.09 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 11:44.48 | Skid | i wouldn't get out of bed for less than 30-35k |
| 11:44.51 | Skid | (GBP) |
| 11:45.13 | nicox | <PROTECTED> |
| 11:46.34 | mut | ya think so eh? |
| 11:46.43 | mut | i was going for like 40k USD |
| 11:46.45 | Fedoracore6 | my friends good in asp |
| 11:46.59 | Fedoracore6 | try lokk in www.musleh.com.my |
| 11:46.59 | mut | i didn't know if that'de be high or low tho |
| 11:47.10 | Fedoracore6 | but in php lor |
| 11:49.13 | Scum-Person | i do get out of bed for less than £30-35k .. doh :p |
| 11:49.22 | mut | fuk |
| 11:49.28 | mut | i make $9.50/hr right now |
| 11:49.42 | mut | i do enough to be making 50k |
| 11:50.16 | Skid | 40k usd = approx 20k |
| 11:50.19 | Scum-Person | IT isn't exactly high value anymore |
| 11:50.22 | mut | yea |
| 11:50.23 | Skid | of which is a starting wage here really |
| 11:50.24 | RoyK | $9.5 per hour - where? mut? |
| 11:50.29 | Scum-Person | wtf is .com.my anyway |
| 11:50.32 | mut | m33access |
| 11:50.40 | RoyK | mut: what country? |
| 11:50.41 | Skid | mylassis |
| 11:50.42 | Skid | (sp) |
| 11:50.43 | Skid | ? :p |
| 11:50.44 | mut | usa |
| 11:50.51 | x86 | malaysia |
| 11:50.53 | x86 | fools ;) |
| 11:50.54 | mut | wiress/voip/hosting/dialup/dsl/telco in northern michigan |
| 11:51.17 | Scum-Person | why couldn't they use .co.my like (almost) everywhere else |
| 11:51.18 | RoyK | you make > $9.5 per hour at mac donald's in .no :P |
| 11:51.30 | mut | i could move to detroit and do it too |
| 11:51.38 | mut | but i wouldn't be gettin this on my resume |
| 11:51.39 | x86 | RoyK: > $9.50 USD per hour? |
| 11:52.17 | RoyK | $9.5 =~ NOK 60 |
| 11:52.20 | RoyK | yes |
| 11:53.58 | mut | so ya think a |
| 11:54.06 | mut | "They are small but on a pretty agressive growth track." comapny will pay $40k for that? |
| 11:55.31 | mut | Median household income: $37,330 (year 2000) |
| 11:55.40 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 11:56.53 | Fedoracore6 | lor |
| 11:57.06 | Fedoracore6 | its fool |
| 11:58.51 | *** join/#asterisk Tili (n=Tili@82-217-236-131.cable.quicknet.nl) |
| 12:00.33 | *** join/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br) |
| 12:01.37 | *** join/#asterisk kink0 (n=kinko@pluton.interec.com) |
| 12:01.41 | kink0 | hello |
| 12:02.51 | kink0 | if I set a external gateway where to route calls from my asterisk when I have congestion in local asterisk, calls comming from outside to my asterisk, and now re-route to other gw, would pass the traffic throug my asterisk ? |
| 12:03.47 | *** join/#asterisk propagandhi (n=opera@d220-236-171-251.dsl.nsw.optusnet.com.au) |
| 12:04.15 | kink0 | or will connect the originate gw to the second gw and RTP traffic will not pass by my gw once the originate and external gw does the SIP negotiation ? |
| 12:07.55 | *** join/#asterisk jbenson (n=jbenson@87.194.2.120) |
| 12:08.15 | jbenson | Hi, has anyone created any telephone survey systems using Asterisk please? |
| 12:09.08 | x86 | i have a bunch of extensions that all behave the same, but ring different locations... is there a way to define them in a batch, or do i really have to type all of them out? |
| 12:09.19 | _Paulo_ | jbenson, seems trivial. |
| 12:10.03 | _Paulo_ | x86, use a macro. |
| 12:10.03 | tuxinator_linux | jbenson: I'm sure it's been done |
| 12:10.08 | x86 | _Paulo_: example? |
| 12:10.57 | tuxinator_linux | x86: Asterkast, episode 2 |
| 12:11.25 | tuxinator_linux | x86: it goes into macros |
| 12:11.48 | x86 | wtf is asterkast? |
| 12:11.51 | tuxinator_linux | x86: It is not the most correct place for information, but its a start |
| 12:12.13 | x86 | i'd rather have a link to a place that explains them well ;) |
| 12:12.48 | tuxinator_linux | asterkast is found at http://www.asterikast.com/ |
| 12:13.13 | tuxinator_linux | x86: the benifit with asterkast is it is a video |
| 12:13.21 | tuxinator_linux | helps you see it being done |
| 12:13.46 | tuxinator_linux | you might also find some info in the book |
| 12:13.50 | tuxinator_linux | ~book |
| 12:13.51 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 12:14.02 | tuxinator_linux | and the wiki |
| 12:14.06 | tuxinator_linux | ~wiki |
| 12:14.14 | *** join/#asterisk Inkubot (n=inkubot@200.74.170.218) |
| 12:14.37 | tuxinator_linux | voip-info.com |
| 12:14.53 | cpm | got that book, it isn't very good. The Oreilly book is better. |
| 12:14.59 | x86 | voip-info seems to be down |
| 12:15.07 | x86 | there it goes, nvm |
| 12:15.17 | cpm | try voip-info.org |
| 12:15.17 | tuxinator_linux | cpm: you're silly, it's the same book |
| 12:15.20 | x86 | http://www.voip-info.org/wiki-Asterisk+cmd+Macro |
| 12:15.24 | cpm | hehe |
| 12:15.28 | x86 | this is the info i'm looking for tuxinator_linux :P |
| 12:15.40 | jbenson | wiki is working okay here |
| 12:15.45 | tuxinator_linux | x86: glad you found out |
| 12:15.56 | tuxinator_linux | s/out/it |
| 12:16.26 | x86 | thanks :) |
| 12:16.59 | x86 | <PROTECTED> |
| 12:16.59 | x86 | exten => 1234,1,Macro(stdexten,1234,SIP/7960) |
| 12:17.26 | x86 | err sorry for double paste... why would someone use extension 1234 to ring extension SIP/7960 ? |
| 12:18.07 | RoyK | well. to tell they love the person at that extension? to tell her she's fired? |
| 12:18.07 | tuxinator_linux | I'm sure it is not an extension but it is referring to a cisco 7960 |
| 12:18.10 | RoyK | dunno |
| 12:18.40 | kaldemar | x86: 7960 has a scent of a certain cisco device. maybe hence the name? |
| 12:20.35 | tuxinator_linux | x86: in my dial plan I did 'exten => 101,1,Dial(SIP/cisco796001) |
| 12:20.36 | tuxinator_linux | ' |
| 12:21.16 | tuxinator_linux | and cisco79600X for each of the 6 lines |
| 12:23.07 | cpm | Is there a iax2 friendly termination provider who supports the speex codec? |
| 12:23.21 | tuxinator_linux | I only have one cisco 7960 that I use for testing, so I named each phone as cisco9760XX |
| 12:23.34 | tuxinator_linux | cpm: not sure, may iaxtel.cc? |
| 12:23.51 | tuxinator_linux | oops, iaxtel.com |
| 12:23.58 | cpm | aren't they the same? |
| 12:24.36 | tuxinator_linux | iaxtel.cc didn't come up for me |
| 12:25.03 | tuxinator_linux | iaxtel.com is in my history, but I can't seem to get the one up either |
| 12:25.35 | cpm | iaxtel is very sick right now. Probably has that flu that is going around. |
| 12:25.48 | RoyK | bitd flu |
| 12:26.13 | fugitivo | morning |
| 12:26.21 | tuxinator_linux | morn fugitivo |
| 12:26.21 | RoyK | afternoon |
| 12:26.34 | _Paulo_ | morning |
| 12:26.39 | fugitivo | ~seen coppice |
| 12:26.43 | jbot | coppice <n=chatzill@212.197.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 3d 3h 28m 30s ago, saying: 'you mean packets are delivered by a cron job? :-\'. |
| 12:27.28 | *** join/#asterisk frenzy (n=frenzy@196.45.144.41) |
| 12:27.36 | frenzy | pbx.c:1406 ast_func_write: Function CDR not registered |
| 12:28.12 | cpm | I've already been through the bird flu, I've been through the Avian Spongiform Ebola Antrax weaponized flu |
| 12:28.26 | cpm | they call me CiproMan! |
| 12:28.58 | tuxinator_linux | I was on cipro for a while |
| 12:29.11 | cpm | ick |
| 12:29.20 | cpm | glad that's over. |
| 12:29.56 | tuxinator_linux | a cat died this week from the bird flu, saw a headline in the news yesterday |
| 12:30.25 | iDunno | silly cat should be more choosy about what it eats ;) |
| 12:30.32 | tuxinator_linux | yep |
| 12:30.33 | cpm | Yeah, but was it the special weaponised strain |
| 12:31.06 | tuxinator_linux | don't know, doubt it |
| 12:31.14 | cpm | It's Obviously the HANDICRAFTS of FRACTIONAL FANATICISM and RADICALISM |
| 12:31.20 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 12:31.26 | cpm | The anagram seeds of , , , , , |
| 12:31.29 | *** join/#asterisk Bambr (n=Bambr@213-35-232-241-dsl.end.estpak.ee) |
| 12:31.40 | cpm | NILIHISTARMYOFCADIZ |
| 12:31.52 | cpm | The Nilihist Peoples Army of Cadiz, I might have known. |
| 12:31.58 | cpm | it was a sad day. |
| 12:31.59 | *** join/#asterisk NotFreak (n=extmail@cp12193-e.tilbu1.nb.home.nl) |
| 12:32.32 | tuxinator_linux | cpm: I'm not familiar with what you are talking about |
| 12:34.35 | cpm | We had a homing pidgeon show up one day. hung around for a day. We called in the tag, fellow said it was doing a hop from Cadiz. We thought that was pretty cool. Fed it, looked after it, it left after 3 days. Then we all got sick as dogs. In my fever, I realized that we had been exposed to a nilihist weapon |
| 12:34.59 | cpm | Amazing what you can learn in a deep fever :) |
| 12:35.46 | fugitivo | birds are dangerous, they transport a lot of diseases |
| 12:35.56 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 12:35.59 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 12:36.17 | Mavvie | Mar 1 23:35:53 DEBUG[18874]: pbx_spool.c:319 scan_service: Delaying retry since we're currently running 'Øз ¸h,' |
| 12:38.07 | *** join/#asterisk Gennaro (n=Gennaro@ppp-62-10-136-50.dialup.tiscali.it) |
| 12:38.19 | Gennaro | Hi |
| 12:38.37 | Gennaro | is included in asterisk flop or only add? |
| 12:38.45 | frenzy | ERROR[19725]: pbx.c:1406 ast_func_write: Function CDR not registered |
| 12:39.16 | tuxinator_linux | cpm, fugitivo: pigeons are roaches with wings |
| 12:39.34 | *** join/#asterisk ibob63 (n=hp@bb-87-82-15-125.ukonline.co.uk) |
| 12:39.43 | fugitivo | i don't like birds |
| 12:39.56 | tuxinator_linux | nasty little birds, crap on everything |
| 12:40.02 | fugitivo | yeah |
| 12:40.24 | tuxinator_linux | I hang CD's to keep them away |
| 12:40.32 | fugitivo | cds?? |
| 12:40.42 | fugitivo | does that work? |
| 12:40.49 | tuxinator_linux | the reflect the light and they move around in the breeze |
| 12:41.09 | tuxinator_linux | yep, they don't like the light shining on them, scares them |
| 12:41.13 | ibob63 | one of my sip trunk registration keeps timing out. i there a way I can debug this? |
| 12:41.22 | tuxinator_linux | sip debug |
| 12:41.25 | fugitivo | ibob63: sip debug |
| 12:41.37 | fugitivo | ibob63: but if it's a timeout, maybe it's a network problem |
| 12:41.47 | ibob63 | i've tried sip debug - but all it says it is keeps retrying |
| 12:41.59 | tuxinator_linux | did you ping the server? |
| 12:42.02 | fugitivo | ibob63: check if you reach the remote server |
| 12:42.07 | tuxinator_linux | traceroute? |
| 12:42.10 | *** join/#asterisk cj-rm (n=cjrm@81-178-22-214.dsl.pipex.com) |
| 12:42.17 | tuxinator_linux | homing pigeon? |
| 12:42.40 | ibob63 | I have pinged the server and its there |
| 12:42.40 | cj-rm | Hey people, I have Ringing() specified in my dialplan, but it doesn't play the ringing tone... Any idea why?? |
| 12:42.54 | fugitivo | ibob63: are sip ports on the remote server open? |
| 12:43.00 | Scum-Person | http://www.theregister.co.uk/2006/03/01/ofcom_voip/ <-- half ? lol |
| 12:43.39 | ibob63 | I think so, they are a gateway for us. |
| 12:43.51 | ibob63 | is there a simply way I can tell if there ports are open? |
| 12:43.52 | Scum-Person | i'd be impressed if 1/8 of ppl even heard of voip, never mind understanding what it is |
| 12:44.09 | fugitivo | that's a LOT |
| 12:44.26 | ibob63 | whats traceroute? |
| 12:44.51 | fugitivo | a program that traces routes to reach a server |
| 12:45.04 | *** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl) |
| 12:45.10 | fugitivo | you can see all hops you need to reach the destination |
| 12:45.23 | tuxinator_linux | ibob63: updatedb; locate traceroute |
| 12:45.49 | tuxinator_linux | you may need to install it |
| 12:45.49 | Tagor | I give my question a second try; I am trying to get MOH on 1.2.4 working. However I just don't hear anything. Can someone tell me how to get this working? |
| 12:46.06 | Tagor | I am using the native solution of Asterisk 1.2 |
| 12:46.24 | fugitivo | Tagor: what do you have in musiconhold.conf? |
| 12:46.47 | Tagor | fugitivo; |
| 12:46.47 | Tagor | [classes] |
| 12:46.47 | Tagor | [moh_files] |
| 12:46.47 | Tagor | default => /var/lib/asterisk/moh-native,r |
| 12:47.08 | fugitivo | if you're using native moh, the correct config should be: [default] mode=files directory=/var/lib/asterisk/mohmp3 |
| 12:47.23 | Tagor | That doesn't work either |
| 12:47.24 | tuxinator_linux | Scum-Person: I think people are wary if its anything like it is hear, the broadband connections at home are not very reliable, mine goes out daily |
| 12:47.31 | fugitivo | it should |
| 12:47.41 | Tagor | Well, here it doesn't :p |
| 12:47.52 | fugitivo | what do you see on the cli? |
| 12:47.52 | Tagor | Tried that several times |
| 12:48.07 | Tagor | It just says that it sets the MOH to default |
| 12:48.11 | Tagor | No errors at all |
| 12:48.15 | Scum-Person | broad band isn't sold as very reliable |
| 12:48.32 | fugitivo | Tagor: make an extension that calls musiconhold |
| 12:48.39 | fugitivo | call that extension and see the cli |
| 12:49.08 | Tagor | You mean just an extension that plays the MOH? |
| 12:49.32 | Gennaro | where i can get flop? |
| 12:49.37 | ibob63 | okay, I have installed traceroute :) What is the best to use it to test why the sip registration is failling? |
| 12:49.58 | kll | Tagor: if you're trying to play .mp3 files.. have you installed mpg123? |
| 12:50.09 | kll | just do a: make mpg123 && make install in the asterisk dir |
| 12:50.37 | Tagor | kll >> I have tried mpg123 too. But it doesn't load according to ps aux. However the native MOH of 1.2 should also be able to play mp3 files |
| 12:51.37 | Gennaro | i installed all mp123 and asterisk from bin in usr src |
| 12:51.52 | kll | alright, that I didn't know. perhaps I'm old fashioned but I'm using mpg123 with 1.2 |
| 12:52.25 | Gennaro | i used ztdummy & zaptel |
| 12:52.31 | kll | Tagor: if you add an exten and set verbose 100 in the console, what does it tell you? |
| 12:52.33 | *** join/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at) |
| 12:52.44 | Gennaro | works |
| 12:52.44 | Tagor | Let me try, kll |
| 12:53.10 | *** join/#asterisk bronze (n=clark@c-24-91-157-212.hsd1.ma.comcast.net) |
| 12:53.24 | nicox | Hello, did anybody know a problem with faxes over asterisk 1.2.4? |
| 12:53.36 | Gennaro | where i can get a flash operator pannel? |
| 12:53.59 | denon | you could try googling for it |
| 12:54.04 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 12:54.29 | tuxinator_linux | ibob63: traceroute addressOfServer |
| 12:54.56 | tuxinator_linux | ibob63: you want to see the ping times for each hop |
| 12:56.01 | tuxinator_linux | ibob63: what's is the time returned on pinging the server? |
| 12:56.25 | Tagor | kll >> It says: -- Stopped music on hold on ......... directly after it starts MOH |
| 12:56.44 | Mavvie | wonder why AGI->get_data does only play the wave file but not does do anything with my dtmf keys. |
| 12:56.47 | kink0 | I receive call from outside gw to my asterisk, if I set my asterisk to route calls to a third external gateway on congestion, will all traffic pass trhough my asterisk ? or your will negotiate SIP between originate and end gw ? |
| 12:57.03 | kll | Tagor: it's prbably due to that asterisk cannot spawn mpg123, ie it is not installed. |
| 12:57.25 | Tagor | Well, I am now using the native MOH |
| 12:57.32 | Tagor | And yes, it is installed |
| 12:57.34 | kll | kink0: sip signalling will pass through your asterisk, though you can make the media path go directly |
| 12:57.35 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 12:57.36 | Tagor | I can run it manually |
| 12:57.44 | mut | what causes a double ring? |
| 12:58.02 | warthawg | of all the possible causes of stutter, how large a part does the physical phone play, if any? |
| 12:58.29 | kink0 | k11 : any special configuration to cause RTP goes directly ? or just I set a Dial priority as normal in extensions.conf ? |
| 12:58.31 | nicox | Hello, did anybody know a problem with faxes over asterisk 1.2.4? |
| 12:59.04 | _Paulo_ | nicox, I can receive well, but not send |
| 12:59.08 | kll | Tagor: okey, could you do a `which mpg123` and a ls -la `which mpg123` |
| 12:59.18 | mut | anyone know |
| 12:59.22 | kll | Tagor: and what does you class look like in musiconhold.conf? |
| 12:59.31 | mut | all i'm doing is dialing out a zap chan |
| 12:59.32 | ibob63 | tuxinator_linux: Here is the report for the trace route - http://pastebin.com/578270 I think the time are around 10 - 100 ms - how does that sound? |
| 13:00.10 | tuxinator_linux | ibob63: that's high, my ping time is 10ms |
| 13:00.30 | kink0 | kll : any special configuration to cause RTP goes directly ? or just I set a Dial priority as normal in extensions.conf ? |
| 13:00.37 | _Paulo_ | nicox, what is your problem? |
| 13:00.41 | Tagor | kll; |
| 13:00.41 | Tagor | which mpg123: /usr/local/bin/mpg123 |
| 13:00.41 | Tagor | ls -la `which mpg123`: -rwxr-xr-x 1 root root 131028 2006-02-28 22:24 /usr/local/bin/mpg123* |
| 13:00.51 | tuxinator_linux | ibob63: the most important one is the server, sip.jnctn.net, which is 112, that's high |
| 13:01.03 | kll | kink0: no, it should be done by default. make sure to have nat=no and reinvite=yes |
| 13:01.11 | ibob63 | <PROTECTED> |
| 13:01.19 | kll | kink0: look out for firewalls and the alike, they can mess up things |
| 13:01.20 | Tagor | And my musiconhold.conf, kll; [default] mode=files directory=/var/lib/asterisk/mohmp3 |
| 13:01.20 | tuxinator_linux | maybe |
| 13:01.20 | kink0 | kll: ok thanks, I will try . |
| 13:01.29 | kll | Tagor: alright, that looks good |
| 13:01.40 | kll | Tagor: have you tried mode=mp3 ? |
| 13:01.45 | tuxinator_linux | ibob63: are you behind a NAT? |
| 13:01.49 | Tagor | No, not yet, I will try that, kll |
| 13:02.08 | ibob63 | tuxinator_linux: no. my server is in a dmx and not behind NAT. |
| 13:02.12 | kll | Tagor: when I play from a directory I just go with mode=mp3 and directory=<dir> |
| 13:02.39 | kll | Tagor: then there are shoutcast, in which case I do: mode=custom application=/usr/local/bin/mpg123 -q -r8000 -f 8192 -b 2048 --mono -s http://64.236.34.196:80/stream/1026 |
| 13:02.42 | *** join/#asterisk bronze (n=clark@c-24-91-157-212.hsd1.ma.comcast.net) |
| 13:03.25 | kink0 | other question... I have my asterisk connected to a PRI, the caller gw told me he receive a CAUSE 63 on congestion, where they expected to see a CAUSE 34 ... but I only see that my asterisk send SIP code 503 |
| 13:03.27 | Tagor | kll; I again get the 'stopped MOH' message |
| 13:03.31 | tuxinator_linux | ibob63: is it easy to pastebin your sip debug? |
| 13:03.42 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:03.57 | kink0 | so, I supusse the translation from SIP to ISDN CAUSE is done at her end, or may be we have sending the code 63 ? |
| 13:04.14 | kll | Tagor: try restarting asterisk and use STOP NOW and then start it over again |
| 13:04.58 | Tagor | What do you mean with 'STOP NOW'? Should I use that in CLI of asterisk? |
| 13:05.11 | ibob63 | tuxinator_linux: here it is - http://pastebin.com/578280 |
| 13:05.54 | ibob63 | tuxinator_linux: It atempts the registration multiple times and then gives up |
| 13:06.18 | kll | Tagor: yes, to stop it |
| 13:06.30 | kink0 | in other words, Asterisk never sends ISDN CAUSE, right ? is translated to SIP and original ISDN CAUSE is not sent, right ? |
| 13:06.34 | tuxinator_linux | ibob63: let me compare to mine |
| 13:07.14 | kink0 | so the remote peer is unable to see my original ISDN CAUSE, since they got only a SIP 403 for many ISDN causes, right ? |
| 13:07.32 | kink0 | sorry, SIP 503 |
| 13:07.47 | Tagor | This is what I now noticed, kll; |
| 13:07.49 | Tagor | <PROTECTED> |
| 13:07.49 | Tagor | <PROTECTED> |
| 13:07.49 | Tagor | <PROTECTED> |
| 13:07.50 | kll | Tagor: normally you do just 'restart now' and if I recall correctly music on hold doesn't reload properly. so you need to stop asterisk and then start it again. |
| 13:07.55 | Tagor | <PROTECTED> |
| 13:07.55 | Tagor | <PROTECTED> |
| 13:07.55 | Tagor | <PROTECTED> |
| 13:08.06 | Tagor | I get the 'stopped MOH' after putting down the phone |
| 13:08.12 | kll | Tagor: do Answer the extension before? |
| 13:08.14 | kll | ie, |
| 13:08.16 | tuxinator_linux | ibob63: before we dive to deep into this, are you sure your ISP doesn't block port 5060? |
| 13:08.20 | kll | exten => 212,1,Answer |
| 13:08.28 | kll | exten => 212,n,MusicOnHold(default) |
| 13:08.50 | *** join/#asterisk robbie2 (n=rob@CPE-60-231-50-21.qld.bigpond.net.au) |
| 13:09.09 | robbie2 | is it possible to have a queue where it only rings agents who are not already in a call ? |
| 13:09.22 | ibob63 | tuxinator_linux: no they don't block any ports |
| 13:09.24 | tuxinator_linux | Tagor: pastebin please |
| 13:09.28 | tuxinator_linux | ~pb |
| 13:09.30 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 13:09.40 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 13:09.44 | Tagor | I've to go for an hour now, kll. Thanks a lot for your help :) |
| 13:09.52 | Tagor | tuxinator_linux >> Yeah, sorry |
| 13:09.54 | ibob63 | tuxinator_linux: the strange thing is everything was working when I left the server last night. |
| 13:10.03 | kll | Tagor: don't mention it, hope you get it running |
| 13:10.07 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 13:10.22 | FlyboySR22 | good morning everyone |
| 13:10.23 | *** join/#asterisk Navman_La (n=R2D2@62.108.221.102) |
| 13:10.35 | Scum-Person | afternoon |
| 13:10.59 | FlyboySR22 | Ah yes - afternoon somewhere :-) |
| 13:11.48 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:12.26 | FlyboySR22 | .msg [TK]D-Fender Good Morning..I am still gathering info from the potential customer for our project..I will let you know if they decide to move ahead :-) |
| 13:13.10 | Scum-Person | :p |
| 13:13.17 | tuxinator_linux | ibob63: it was working, and not it's not? |
| 13:13.27 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com) |
| 13:13.47 | mut | anyone know what causes a double ring? i call the phone on my desk thru and it rings same time as in my ear, but right after it's done ringing the first time it rings again right way |
| 13:13.54 | mut | and does it til someone answers |
| 13:13.59 | tuxinator_linux | ibob63: Did you pay your bill? |
| 13:14.06 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 13:14.50 | ctooley | mut, sounds like you're using the wrong ringone. I'm not sure whre that's changed anymore |
| 13:14.58 | ibob63 | tuxinator_linux: yep, have got lots of money in their account - but I like you thinking :) |
| 13:15.09 | ctooley | s/ringone/ringtone |
| 13:15.10 | tuxinator_linux | mut: sounds like a ring style, you can set ring types somewhere, I think |
| 13:15.11 | mut | ? |
| 13:15.18 | tuxinator_linux | mut: distinctive ring? |
| 13:15.22 | mut | no |
| 13:15.27 | mut | it's what i hear when calling someone |
| 13:15.37 | mut | and it's only when i use my zap chan |
| 13:15.51 | tuxinator_linux | mut: hmm, not sure |
| 13:15.55 | Druken | wut ya hearing? a double ring? |
| 13:16.00 | mut | yes |
| 13:16.03 | Druken | hehehe |
| 13:16.13 | Druken | one is from asterisk, one is from the telco |
| 13:16.22 | tuxinator_linux | ibob63: is it possible to do a reload, and sip debug.. and then pastebin more of the output? |
| 13:16.25 | mut | right so how do stop it |
| 13:16.47 | Druken | when ya figure it out... let me know :) aside from using ,r, in your dialstring |
| 13:16.49 | mut | do i* |
| 13:17.32 | Gennaro | sorry, someone sayme how to create IVM and install GUI and FLOP |
| 13:17.47 | tuxinator_linux | FLOP? |
| 13:17.54 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 13:18.01 | Gennaro | FLASH OPERATOR PANNEL |
| 13:18.03 | _Sam-- | flash losers operator panel |
| 13:18.07 | Druken | flop.... for lazy operators? |
| 13:18.07 | fugitivo | IVM? |
| 13:18.11 | tuxinator_linux | ahh |
| 13:18.19 | fugitivo | Interactive voice mail? |
| 13:18.25 | Gennaro | so |
| 13:18.33 | Gennaro | fugitivo, i installed 2 phone |
| 13:18.39 | Gennaro | i tried incoming from iax |
| 13:18.54 | Gennaro | and now id like to do a voice menu |
| 13:19.02 | Gennaro | to get call from SIP |
| 13:19.06 | tuxinator_linux | Gennaro: you won't receive much help in here.. most of us don't use the gui stuff |
| 13:19.14 | Druken | IVR = interactive voice RESPONCE |
| 13:19.36 | fugitivo | Druken: responCE or responSe? :) |
| 13:19.38 | Gennaro | i dont wanto to use gui |
| 13:19.40 | tuxinator_linux | se |
| 13:19.43 | Druken | fugitivo, eat me :) |
| 13:19.47 | Gennaro | i wanto to know how works.. |
| 13:20.02 | heison | ~seen coppice |
| 13:20.09 | jbot | coppice <n=chatzill@212.197.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 3d 4h 21m 56s ago, saying: 'you mean packets are delivered by a cron job? :-\'. |
| 13:20.13 | tuxinator_linux | Gennaro: have you looked at the wiki? voip-info.com |
| 13:20.16 | fugitivo | Gennaro: check the wiki |
| 13:20.28 | Gennaro | ok i'm going.. |
| 13:20.31 | fugitivo | heison: heh, i'm looking for him too |
| 13:20.48 | tuxinator_linux | what is so special about coppice? |
| 13:20.52 | Gennaro | but i cant understand the mechanism.. i'm going... |
| 13:21.06 | Druken | tuxinator_linux: he's made of sugar and spice and everything nice?? |
| 13:21.14 | fugitivo | that's because you use asterisk@home |
| 13:21.22 | fugitivo | you don't understand how asterisk really works |
| 13:21.23 | tuxinator_linux | Druken: hmm, interesting ;-) |
| 13:21.35 | mut | argh |
| 13:21.40 | mut | why is this double ringing! |
| 13:21.45 | fugitivo | we should kill a@h |
| 13:21.50 | Druken | because it likes you mut |
| 13:21.59 | Druken | it wants to make sure you know it's ringing :) |
| 13:22.03 | fugitivo | mut: what do you mean by doble ringing?? |
| 13:22.19 | tuxinator_linux | *@~ is good for getting your feet wet, but not good for learning or production use |
| 13:22.22 | Druken | he's getting the double telco ring, |
| 13:22.26 | mut | it rings, then rings again fast after that |
| 13:22.31 | mut | when calling someone |
| 13:22.32 | _Paulo_ | coppice is the man. |
| 13:23.14 | mut | voip-info is gettin on me nerves |
| 13:23.37 | tuxinator_linux | mut: it doesn't like you much either ;-) |
| 13:23.41 | Druken | voip-info was my saviour when i was first starting out |
| 13:24.00 | tuxinator_linux | I find voip-info helpful |
| 13:24.08 | tuxinator_linux | as well as asterisk-guru |
| 13:24.13 | fugitivo | only when the info is correct... |
| 13:24.14 | Druken | but i must admit.... the google search pisses me off... i like it better with it's own search |
| 13:24.25 | fugitivo | i agree |
| 13:25.00 | tuxinator_linux | I don't like how returns the results for the same page, but different version |
| 13:25.26 | Druken | i say we all donate like 10 bux each, and buy it a google appliance and have it integrated :) |
| 13:25.33 | tuxinator_linux | ibob63: how you doing over there? |
| 13:26.10 | *** join/#asterisk walhala (i=walhala@213.161.208.11) |
| 13:26.12 | tuxinator_linux | Druken: hmm, maybe |
| 13:26.12 | walhala | hi all |
| 13:26.14 | walhala | i have this error "failed to grab lock" what does it mean ? |
| 13:26.23 | tuxinator_linux | it failed to grab lock |
| 13:26.27 | Druken | fugitivo: remind me to never unpack a tar over nfs |
| 13:26.38 | walhala | tuxinator_linux: yeah of course but exactly ? |
| 13:26.39 | tuxinator_linux | walhala: can you explain more |
| 13:26.52 | tuxinator_linux | walhala: when does it say this? |
| 13:26.52 | fugitivo | Druken: why not? nfs is fast and furious |
| 13:27.12 | walhala | when i have a lot of call (~100) asterisk crash with this error |
| 13:27.16 | *** join/#asterisk bronze (n=clark@c-24-91-157-212.hsd1.ma.comcast.net) |
| 13:27.19 | Druken | then one of my connections is pooched |
| 13:27.33 | Druken | cause this is taking forever.... |
| 13:27.35 | tuxinator_linux | ~wiki pooched |
| 13:27.37 | Druken | brings me back to my dialup days |
| 13:27.39 | mut | fugitivo: there a way to get * not to generate a ring? |
| 13:27.52 | fugitivo | Druken: lan nfs or wan nfs? |
| 13:27.56 | Druken | wan |
| 13:28.00 | fugitivo | well |
| 13:28.17 | walhala | tuxinator_linux: any idea ? |
| 13:28.21 | Druken | across the country wan.. hehe |
| 13:28.26 | tuxinator_linux | walhala: I'm searching |
| 13:28.36 | fugitivo | that's the problem |
| 13:28.41 | fugitivo | not nfs |
| 13:28.44 | walhala | tuxinator_linux: ok :) |
| 13:28.57 | Druken | :) |
| 13:29.11 | fugitivo | mut: ring tone or ring a phone? |
| 13:29.26 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 13:29.30 | mut | tone, i'm calling the phone on my desk from an ata |
| 13:29.35 | kll | I'm looking at the output of show channel SIP/lalalala-a4fd and I find NativeFormat/WriteFormat/ReadFormat a bit confusing, which one says what? |
| 13:29.39 | mut | i hear 2 rings on the ata |
| 13:29.42 | mut | one ring on my desk |
| 13:29.45 | Druken | i wonder when the city is going to start making contractors run fiber into the new homes... :) |
| 13:30.19 | fourcheeze | mut: your phone is probably slow |
| 13:30.30 | Druken | mut: i thought you said this was only with your ZAP channels... or is the phone on your desk a phoneline? |
| 13:30.31 | mut | huh? |
| 13:30.41 | mut | it's a phoneline |
| 13:30.45 | tuxinator_linux | walhala: the error can be found in /channels/chan_sip.c of the source |
| 13:30.46 | Druken | okie |
| 13:30.58 | mut | goes from ata -> * -> zap -> echo can -> adtran -> pbx -> desk |
| 13:31.06 | fugitivo | mut: use the r flag |
| 13:31.07 | Druken | i get that shit all the time too... i just ignore it... |
| 13:31.31 | mut | is that going to mess up dialing ata -> * -> ata? |
| 13:31.40 | Druken | shouldn't |
| 13:32.31 | tuxinator_linux | walhala: something to do with a 'netlock' but I am not sure what that is |
| 13:32.31 | Druken | all r does it make asterisk do the ring, instead of trying to get it from the telco... |
| 13:32.32 | Druken | it's all fuct up |
| 13:32.39 | Druken | also blocks the audio from the telco till the end party answers |
| 13:32.39 | mut | reload |
| 13:32.46 | mut | doh wrong window |
| 13:32.47 | tuxinator_linux | walhala: looks like '/* Lock the network interface */' |
| 13:33.13 | [TK]D-Fender | FlyboySR22 : Sounds good, keep me posted |
| 13:33.25 | FlyboySR22 | [TK]D-Fender, Will DO :-) |
| 13:33.50 | tuxinator_linux | walhala: also '/*! \brief Protect the monitoring thread, so only one process can kill or start it, and not |
| 13:33.51 | tuxinator_linux | <PROTECTED> |
| 13:33.55 | tuxinator_linux | AST_MUTEX_DEFINE_STATIC(netlock); |
| 13:33.56 | tuxinator_linux | ' |
| 13:33.57 | walhala | tuxinator_linux: so this mean my network interface is full ? |
| 13:34.40 | tuxinator_linux | walhala: do you have any other programs using the network card? |
| 13:34.58 | walhala | tuxinator_linux: yeah a wan nfs share |
| 13:35.06 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:35.07 | walhala | and a connection to mysql |
| 13:35.37 | x86 | anyone ever setup a macro to handle a bunch of inbound providers? |
| 13:36.08 | Druken | i simply have a incoming context and don't give a shit where the call came from :) |
| 13:36.13 | tuxinator_linux | walhala: possible could be the problem, maybe fixable by adding a network card just for * |
| 13:36.42 | x86 | hmm |
| 13:36.43 | walhala | tuxinator_linux: ok i'll try :) but another way is may be ram ? |
| 13:37.24 | x86 | Druken: all the inbound providers are defined in a single context, but i want to keep them seperate so i can re-label the caller ID to display which provider they came in on... |
| 13:37.32 | tuxinator_linux | walhala: how much do yo have? RAM that is |
| 13:37.54 | Druken | x86: why would you want to do that ? |
| 13:38.03 | walhala | tuxinator_linux: only 1go "Mem: 1138 1120 17" |
| 13:38.19 | walhala | tuxinator_linux: so only 17 mo free at the moment |
| 13:39.31 | tuxinator_linux | walhala: I suppose it could be either... make sure to check voip-info and asteriskguru, and google before you do anything like adding a NIC or RAM |
| 13:40.34 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 13:41.04 | tuxinator_linux | walhala: are you using the lastest trunk? |
| 13:41.11 | walhala | tuxinator_linux: i have it |
| 13:41.17 | walhala | tuxinator_linux: i use it |
| 13:42.24 | tuxinator_linux | walhala: may have to do with mysql |
| 13:43.41 | *** join/#asterisk Xen^ (i=linux@203.101.160.180) |
| 13:44.03 | walhala | tuxinator_linux: so i your opinion what should I have to do ? |
| 13:44.16 | tuxinator_linux | walhala: what is the whole error message? |
| 13:45.08 | *** join/#asterisk docelm0 (n=docelmo@66.239.192.34.ptr.us.xo.net) |
| 13:45.16 | tuxinator_linux | walhala: are you using mysql for *? |
| 13:45.27 | fugitivo | evil |
| 13:45.30 | walhala | tuxinator_linux: Mar 1 10:04:33 DEBUG[18363] chan_sip.c: Failed to grab lock, trying again... |
| 13:45.46 | fugitivo | evil wannabe database |
| 13:45.47 | walhala | tuxinator_linux: yeah i use it |
| 13:46.04 | tuxinator_linux | walhala: http://72.14.207.104/search?q=cache:3l_H2FGPufQJ:bugs.digium.com/view.php%3Fid%3D6181%26nbn%3D14+asterisk+%27failed+to+grab+lock%27&hl=en&gl=us&ct=clnk&cd=1&client=firefox |
| 13:46.07 | Druken | pgsql |
| 13:46.25 | fourcheeze | fugitivo: don't hold back - tell us what you really feel |
| 13:46.49 | fugitivo | i feel good! |
| 13:46.53 | tuxinator_linux | fugitivo: MySQL works well, so does Postgres... whatever fits your needs |
| 13:47.18 | fugitivo | yeah yeah |
| 13:47.22 | fourcheeze | mysql ruuuulez |
| 13:47.27 | tuxinator_linux | fugitivo: MySQL has always been nice to me |
| 13:47.40 | fourcheeze | put it this way: I trust mysql more than I trust asterisk |
| 13:48.24 | walhala | tuxinator_linux: the page is very very slow ! |
| 13:48.24 | fugitivo | put it this way: some people uses realtime! |
| 13:48.40 | fourcheeze | realtime has never been a problem |
| 13:48.50 | fourcheeze | not for me anyway |
| 13:49.11 | fourcheeze | realtime makes asterisk usable |
| 13:49.24 | *** join/#asterisk kristinG (n=kristin@gentoo/user/kristinG) |
| 13:49.26 | fugitivo | ok, maybe we are from different production environments |
| 13:49.27 | kristinG | hi |
| 13:49.31 | fugitivo | i have different opinions |
| 13:49.47 | tuxinator_linux | walhala: the site is not up, so I gave you the google cache version |
| 13:49.58 | kristinG | i need help please with a pstn --> tnt -> digium card -> asterisk config |
| 13:50.10 | fourcheeze | fugitivo: one thing I want to be able to do is have different * boxen find each others users |
| 13:50.15 | fourcheeze | try doing that without a database |
| 13:50.17 | walhala | tuxinator_linux: ok :) |
| 13:50.25 | tuxinator_linux | kristinG: tnt? sounds like an explosive solution |
| 13:50.41 | kristinG | :p |
| 13:50.53 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 13:51.02 | fugitivo | fourcheeze: for my work, mysql is a joke, and realtime is something i can't trust for production |
| 13:51.16 | walhala | tuxinator_linux: in all case thanks a lot tuxinator_linux :) |
| 13:51.31 | tuxinator_linux | walhala: no problem |
| 13:51.42 | kristinG | then what do you use fugitivo ? |
| 13:51.44 | fourcheeze | fugitivo: mysql isn't as good as postgres , this is true, but it's far from a joke |
| 13:51.45 | kristinG | mssql? |
| 13:52.03 | tuxinator_linux | walhala: my recommendation, for a production system, * only on the machine, and everything else on another |
| 13:52.07 | fugitivo | kristinG: postgresql, mssql sometimes or oracle |
| 13:52.24 | fugitivo | that depends on the budget of the company i work for |
| 13:52.25 | kristinG | mysql 5.x versions blow postgres away |
| 13:52.33 | kristinG | imho |
| 13:52.35 | fugitivo | kristinG: too late |
| 13:52.39 | webmind | mysql, postgre, just no mssql |
| 13:52.59 | kristinG | anyways |
| 13:53.13 | fourcheeze | fugitivo: of course if you have a solution of how to route dynamically between many asterisks please let me know :-) |
| 13:53.16 | tuxinator_linux | kristinG: ask you question |
| 13:53.24 | fugitivo | just wait for version 5 of a program to be something i can use, is a joke |
| 13:53.27 | kristinG | can anyone offer some help in dumping a call from the pstn to the digium card? |
| 13:53.47 | tuxinator_linux | is TNT a channel bank? |
| 13:53.49 | fourcheeze | fugitivo: what does mysql5 have that earlier versions don't that you need? |
| 13:54.01 | fugitivo | fourcheeze: i don't know, i don't use mysql 5 |
| 13:54.01 | kristinG | the tnt is a lucent TNT |
| 13:54.16 | kristinG | i have a ds3 card and a 8 port ds1 card in it |
| 13:54.18 | tuxinator_linux | I'm not a telco guy, don't know what the TNT does |
| 13:54.20 | fourcheeze | mysql was very reliable in version 3 |
| 13:54.25 | kristinG | as well as madd2 cards |
| 13:54.34 | fourcheeze | it just didn't have many features |
| 13:54.48 | fugitivo | fourcheeze: maybe you didn't find the reason to use another database, yet |
| 13:54.59 | fourcheeze | sure I have |
| 13:55.06 | fourcheeze | I know about stored proceedures etc |
| 13:55.15 | fugitivo | fourcheeze: some complex apps need complex databases |
| 13:55.21 | fourcheeze | of course |
| 13:55.22 | kristinG | perhaps this is not the best forum to discuss the merits of a dbs |
| 13:55.31 | fourcheeze | kristinG: why not? |
| 13:55.32 | tuxinator_linux | kristinG: agreed |
| 13:55.54 | fourcheeze | fugitivo: just because something is better suited to a simple solution doesn't make it a joke |
| 13:55.56 | fugitivo | kristinG: this is not a forum, and #asterisk is always a channel to discuss about everything |
| 13:55.59 | kristinG | why not because it will just trun into a pissing contest |
| 13:55.59 | tuxinator_linux | fourcheeze: unless it is relating to its use with * |
| 13:56.08 | fourcheeze | which it does |
| 13:56.18 | walhala | tuxinator_linux: that's what is in production with a distant mysql server and a nfs share to put voicemail into |
| 13:56.19 | fourcheeze | we're talking about using asterisk with realtime and different DBs |
| 13:56.30 | _Paulo_ | mysql have issues with gpl |
| 13:56.32 | mut | Druken |
| 13:56.33 | mut | sip.conf |
| 13:56.38 | mut | progressinband=no |
| 13:56.48 | fugitivo | _Paulo_: great, one person with feet on the ground |
| 13:56.57 | Zeeek | I'm using asterisk while eating two different brands of chocolate cookies. |
| 13:57.06 | mut | fixes |
| 13:57.08 | fourcheeze | fugitivo: it seems that your asterisk needs are not complicated enough to require a RDBMS and that your other applications require more than mysql can offer |
| 13:57.14 | tuxinator_linux | kristinG: you have a DS3 coming in with your voice channels? |
| 13:57.22 | _Paulo_ | mysql is evil in the sense that it encourages bad pratices and non standard sql |
| 13:57.24 | kristinG | tuxinator_linux, yes |
| 13:57.26 | fugitivo | fourcheeze: i don't use realtime |
| 13:57.33 | fourcheeze | I know |
| 13:57.35 | kristinG | i have data and voice coming in on the ds3 |
| 13:57.48 | fourcheeze | fugitivo: if you were in my position you might |
| 13:57.58 | kristinG | some voice is sent to my sip gateways |
| 13:58.09 | kristinG | some voice is to be sent to a digium |
| 13:58.16 | fourcheeze | _Paulo_: you have to be a particular kind of anal retentive to be worried about that |
| 13:58.18 | kristinG | and the modem calls terminate on the tnt |
| 13:58.26 | tuxinator_linux | kristinG: which digium card? |
| 13:58.26 | kristinG | i have my trunk-groups set |
| 13:58.33 | iDunno | tnt and termination - sounds explosive. |
| 13:58.40 | fugitivo | fourcheeze: actually, i work with crm and some other complex systems that need interaction with asterisk, mysql is not good for that |
| 13:58.45 | tuxinator_linux | iDunno: old joke ;-) |
| 13:58.53 | kristinG | it is just a single t1 card |
| 13:58.56 | iDunno | tuxinator_linux: but still good :) |
| 13:59.10 | kristinG | the t1 is up between the tnt and the asterisk |
| 13:59.15 | tuxinator_linux | kristinG: okay, so where are you having trouble/ |
| 13:59.16 | kristinG | as is the d-channel |
| 13:59.16 | _Paulo_ | <PROTECTED> |
| 13:59.32 | *** join/#asterisk NDT (n=noone@cpe-24-195-218-134.nycap.res.rr.com) |
| 13:59.33 | kristinG | it is not routing calls to the correct ds1 |
| 14:00.06 | mut | fugitivo: progressinband = no fixes the double ring issue |
| 14:00.09 | mut | in the sip.conf |
| 14:00.13 | tuxinator_linux | kristinG: getting to much into telco land for me |
| 14:00.27 | fourcheeze | _Paulo_: which is essentially what I use it for |
| 14:00.48 | fourcheeze | to replace sip.conf, cdr_csv etc |
| 14:00.51 | kristinG | ok thanks |
| 14:00.57 | tuxinator_linux | kristinG: 'it' being what? |
| 14:01.05 | kristinG | i'll go back to screwing with call-routes |
| 14:01.14 | fourcheeze | fugitivo: which crm do you use? |
| 14:01.15 | fugitivo | fourcheeze: well, that's the reason of mysql being a joke |
| 14:01.20 | kristinG | it being the tnt |
| 14:01.26 | _Paulo_ | fourcheeze, so few other tools will perform better for these tasks. |
| 14:01.47 | tuxinator_linux | kristinG: the tnt, yep, sorry, can't help you there |
| 14:01.48 | fourcheeze | fugitivo: using mysql for config and logging doesn't make it a joke |
| 14:02.04 | fugitivo | fourcheeze: i use text files for that |
| 14:02.29 | fourcheeze | fugitivo: you feel free to use text files if you want - mysql is more efficient |
| 14:02.39 | fourcheeze | unless you're generating your text files from SQL |
| 14:02.39 | _Paulo_ | fourcheeze, postgresql plays another league. |
| 14:02.57 | Tagor | kll; have you got any other idea's how to fix the MOH problem? |
| 14:02.58 | fourcheeze | _Paulo_: I think a few years back that was true - and I'm a postgres fan |
| 14:03.02 | fugitivo | fourcheeze: a database is not for config files or logging only |
| 14:03.14 | Tagor | kll; I use the same extension as you mentioned above |
| 14:03.23 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 14:03.25 | fourcheeze | fugitivo: it's a subset of what a database can be for |
| 14:03.36 | NDT | config is easier to just put in your conf files and database interaction through agi |
| 14:03.36 | fourcheeze | I never said it was the limit of mysql |
| 14:04.12 | fourcheeze | I may yet move over to postgres if the need arrives |
| 14:04.27 | fugitivo | if mysql and postgresql are both free |
| 14:04.32 | fourcheeze | NDT: I do use AGI but I try not to |
| 14:04.32 | fugitivo | why should i pick mysql? |
| 14:04.37 | fugitivo | if postgresql is superior? |
| 14:04.43 | fourcheeze | postgres has a larger footprint |
| 14:05.04 | NDT | storing your config in the database though is just a bunch of useless queries in most cases |
| 14:05.08 | mut | is there any reason why my b channels restart all the time? |
| 14:05.15 | NDT | supposed to |
| 14:05.17 | fourcheeze | NDT: if you use realtime a lot is cached |
| 14:05.23 | mut | thats normal |
| 14:05.23 | mut | ? |
| 14:05.25 | mut | hm |
| 14:05.26 | NDT | yeah |
| 14:05.27 | mut | k |
| 14:05.33 | NDT | incase any hung channels |
| 14:05.41 | mut | ah |
| 14:05.47 | mut | my cisco had one open all weekend once |
| 14:05.52 | NDT | restarts any channels that haven't been used in like an hour or somethign |
| 14:05.53 | fugitivo | fourcheeze: actually, from what i could see, realtime code is far from optimized in asterisk |
| 14:05.57 | mut | see if the zap does the same |
| 14:05.57 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 14:06.00 | fugitivo | fourcheeze: a lot of queries are bad written |
| 14:06.07 | mut | its more than an hour |
| 14:06.09 | mut | er |
| 14:06.09 | mut | less |
| 14:06.14 | mut | seems like every 20 min |
| 14:06.21 | NDT | forget what the default is |
| 14:06.25 | mut | maybe even less |
| 14:06.34 | mut | it a setting somewhere? |
| 14:07.26 | fourcheeze | fugitivo: I think it should be possible to write your own queries for realtime |
| 14:07.32 | fourcheeze | having fixed ones makes no sense |
| 14:07.33 | NDT | heh yeah don't remember where...the default is fine |
| 14:07.40 | fourcheeze | why should I have to use someone else's schema? |
| 14:07.53 | fourcheeze | but I guess if I bothered to hack the code I could |
| 14:08.46 | fourcheeze | fugitivo: so I should be able to give it a query which returns something if a username and password are correct, for instance |
| 14:09.06 | NDT | mut: it won't restart a channel that is in use...if that was your worry heh |
| 14:09.17 | mut | yeh i know |
| 14:09.33 | *** join/#asterisk ruza (n=ruza@holly.cervenytrpaslik.cz) |
| 14:09.49 | NDT | you can pop bchannels all day long...long as the D is up who cares heh |
| 14:09.55 | _Paulo_ | hum... I would like to hack the app_authenticate to use an sql query... |
| 14:09.59 | kFuQ | http://www.pcpowercooling.com/products/viewproduct.php?show=TC1KW <-- hmmm 1KW power supply |
| 14:11.16 | *** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 14:13.55 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 14:15.24 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:17.08 | *** join/#asterisk mkl1525 (n=daniel@82.100.204.242) |
| 14:17.58 | Dr-Linux | hi |
| 14:18.14 | _Paulo_ | hi |
| 14:18.16 | Dr-Linux | anybody can help me to read my one line from Queue logs? |
| 14:18.18 | Dr-Linux | woww |
| 14:18.25 | Dr-Linux | _Paulo_: how are you ? :) |
| 14:18.26 | Dr-Linux | 1141063442|1141063411.648|fc-support|NONE|ENTERQUEUE||3102880422 |
| 14:18.39 | _Paulo_ | Dr-Linux, fine, and you? |
| 14:18.45 | Dr-Linux | _Paulo_: i'm fine thanks |
| 14:18.48 | x86 | Mar 1 08:18:32 NOTICE[704]: chan_iax2.c:6775 socket_read: Rejected connect attempt from 192.246.69.186, who was trying to reach '750159@' |
| 14:18.56 | x86 | <PROTECTED> |
| 14:19.06 | x86 | what does this mean? |
| 14:19.15 | Dr-Linux | _Paulo_: we had recived a call last night, caller said he waited in queue for 10 minutes, |
| 14:19.22 | x86 | when calls come in via FreeWorldDialup this is what shows up in the CLI |
| 14:19.26 | Dr-Linux | _Paulo_: i wanna find that call in queue |
| 14:19.34 | Dr-Linux | _Paulo_: can you help me with that? |
| 14:19.35 | Dr-Linux | 1141063442|1141063411.648|fc-support|NONE|ENTERQUEUE||3102880422 |
| 14:19.39 | Dr-Linux | what does this mean |
| 14:19.50 | Dr-Linux | i just know fc-support is a queueu |
| 14:20.09 | x86 | Dr-Linux: 310xxxxxxx sounds like his/her CID |
| 14:20.24 | x86 | Dr-Linux: the first two look like seconds since the epoch |
| 14:21.10 | Dr-Linux | x86: should i show you another line? |
| 14:21.23 | Dr-Linux | 1141063442|1141063411.648|fc-support|NONE|ENTERQUEUE||3102880422 |
| 14:21.23 | Dr-Linux | 1141066567|1141063411.648|fc-support|NONE|ABANDON|1|1|3125 |
| 14:22.04 | _Paulo_ | the second is the call unique id, isnt it? |
| 14:22.10 | mkl1525 | Hi, I'm using mysql to log cdr. For external calls there is our pbx prefix to get an external line + called number - is there any option to get rid of the prefix before it is written into the database? |
| 14:22.28 | Dr-Linux | yes |
| 14:22.40 | Dr-Linux | _Paulo_: but what's the first one? |
| 14:23.12 | *** join/#asterisk bigjb (n=bigjb@195.60.10.113) |
| 14:23.28 | Zeeek | x86 you have an entry for FWD in iax.conf with a context ? |
| 14:23.33 | _Paulo_ | total numbe of seconds since 1/1/1970? |
| 14:23.35 | bigjb | whats the best hardware to use to bring a single analogue line into a asterisk box? |
| 14:23.48 | Zeeek | x86 and you installed the FWD RSA key? |
| 14:24.19 | *** join/#asterisk SibRw0rk (n=DaPhrek@66.234.235.84) |
| 14:24.44 | [TK]D-Fender | Dr-Linux : that is the line that shos a call first entering a queue. the 2nd line shows that they hung up after 1 second waiting. |
| 14:26.13 | Dr-Linux | [TK]D-Fender: thanks but sir how can i see the call time ? |
| 14:27.11 | _Paulo_ | bigjb, TDM400P or TE210P |
| 14:27.25 | [TK]D-Fender | the very first value on the lines is the UNXTIME of the event, and the "1" after abandon tells you how long. There is a readme file with the sourece. |
| 14:27.25 | _Paulo_ | bigjb, from Digium... |
| 14:28.13 | _Paulo_ | Dr-Linux, use a 1liner perl script to convert the timestamp |
| 14:28.38 | [TK]D-Fender | bigjb : Not the TE210P, thats a 2 port digital card. For analog its either a TDM400P w/ FXO card, X100P single port FXO PCI card, A200 card with FXO Module, or a VoIP gateway like the SPA-3000 |
| 14:28.46 | Dr-Linux | :S |
| 14:29.11 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 14:29.28 | bigjb | ahh, i was slightly confused by the te210p i was pretty sure that was bri |
| 14:29.56 | Dr-Linux | but it could be the time? |
| 14:29.57 | Dr-Linux | 1141066567|1141063411.648|fc-support|NONE|ABANDON|1|1|3125 |
| 14:30.12 | Dr-Linux | 1141066567 <<< this is very first .. :S |
| 14:30.22 | [TK]D-Fender | Yes, thats the time for hte event |
| 14:31.46 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:32.40 | *** join/#asterisk siim (n=siim@213-35-232-241-dsl.end.estpak.ee) |
| 14:33.27 | _Paulo_ | Dr-Linux, perl -e 'use POSIX qw(strftime); print strftime("%a %b %e %H:%M:%S %Y", localtime(1141066567)); print "\n"' |
| 14:35.19 | Tili | hey what is the command for perl shell |
| 14:35.25 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 14:35.30 | Hmmhesays | god its sad, i'm filling out one of those stupid survey emails.... "name 4 websites you visit daily" "um.. totalfark.com, voip-info.org" |
| 14:35.47 | mkl1525 | is there a way to avoid the # press of an agent when he gets the call from a queue? |
| 14:35.54 | *** join/#asterisk RoyKa (n=roy@80.239.107.70) |
| 14:36.04 | _Paulo_ | Tili, /usr/bin/perl ??? |
| 14:36.16 | x86 | Zeeek: yes to the context, not sure about the key... |
| 14:36.24 | Tili | _Paulo_: no i was talking about perl -MCPAN -eshell |
| 14:36.26 | Tili | got it now |
| 14:36.30 | x86 | Zeeek: do you have an example of how to install the key? |
| 14:38.16 | Chotaire | morning all.. I wonder if it's possible to use MeetMeAdmin to kick specifically flagged users (like un-marked or marked users), or to set a variable that could be used instead of "user". Or if there exists any code to read the user number out of meetme list. |
| 14:38.39 | siim | hi, I got this error msg > Started music on hold, class 'default', on channel 'SIP/term_default_555-9491' |
| 14:38.39 | siim | <PROTECTED> |
| 14:38.39 | siim | Mar 1 16:34:20 NOTICE[24356]: res_musiconhold.c:507 monmp3thread: Request to schedule in the past?!?! how can I make it work? |
| 14:39.08 | Zeeek | x86 the key is explained on teh FWD site. It's just a short text file you put in the asterisk install |
| 14:39.11 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 14:39.27 | xtrvd | Greetings asteriskmonkey |
| 14:39.48 | Chotaire | from what I know, the "user" in a meetme will be dynamic. or is it possible to statically set the user number for a specific user upon transfer? |
| 14:40.31 | Zeeek | x86 in /var/lib/asterisk/keys you add the file they send (or you download it frolm the FWD site) |
| 14:40.57 | Zeeek | and you use auth=rsa |
| 14:41.18 | Chotaire | morning Zeeek ;) good to see an old face.. you think you can assist me with my little dream? ;) |
| 14:41.38 | tzafrir | the FWD key is distributed with asterisk, right? |
| 14:41.40 | Zeeek | Hi Chotaire... I remember you had a weird problem years ago |
| 14:41.45 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:41.46 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:41.53 | Zeeek | tzafrir you may be right |
| 14:42.12 | Zeeek | can't be hard to find, google for freeworlddialup.pub |
| 14:42.13 | Chotaire | Zeeek: I always come up with weirdo stuff.. the easy is not for me ;) |
| 14:42.17 | *** join/#asterisk Winkie (n=urmom@cpc2-stre1-6-0-cust119.bagu.cable.ntl.com) |
| 14:42.35 | tzafrir | Any other IAX service providers use rsa authentication? |
| 14:42.50 | Zeeek | iaxtel and vopicepulse |
| 14:43.06 | Zeeek | jeeze does iaxtel even exist anymore? |
| 14:43.32 | Zeeek | Chotaire in reading your question I have to step back into the "useless" area |
| 14:43.48 | *** join/#asterisk tracinet (n=tracinet@24-50-29-205.atlsfl.adelphia.net) |
| 14:44.03 | Chotaire | it always depends on your goal ;) |
| 14:44.03 | Zeeek | tzafrir seems like the public key method would be great between your own multiple IAX servers |
| 14:44.16 | Chotaire | I could explain the goal and you would understand. |
| 14:44.17 | Zeeek | I meant me, I'm useless |
| 14:44.25 | Chotaire | ah ok ;) |
| 14:44.45 | Chotaire | I just recently fell in love with meetme "X" mode and dynamic features... |
| 14:44.48 | Druken | Zeeek: anyone who spends 10 mins in here already knows that... |
| 14:44.54 | Zeeek | hahah |
| 14:44.57 | Druken | :) |
| 14:45.08 | Zeeek | All my redeeming qualities are elswhere |
| 14:45.23 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
| 14:45.26 | Chotaire | so anyone know if it would be able to give someone a static user number upon join of a meetme? |
| 14:45.47 | Zeeek | I'll never forget the first time I tried to get help here |
| 14:45.48 | Chotaire | that would already make it with just a bit of creativity. |
| 14:46.07 | Zeeek | Chotaire you promised to explain the goal |
| 14:46.36 | Druken | oh god, my memory isn't that good |
| 14:47.02 | tracinet | anyone know how to return all available channels instead of just the first one when using ChanIsAvail ? |
| 14:47.28 | Chotaire | goal is easy explained.... imagine a developer meetme with guests joining... if there is chaos in a meetme, any admin (e.g. marked user) could hit dtmf "7" and thereby disconnect all non-admins or even specific users. |
| 14:47.42 | Chotaire | or move them to another guest meetme. |
| 14:47.44 | heroine | hmm .. did somebody manage to get spandsp working with asterisk-1.2.4 ? |
| 14:48.01 | _Paulo_ | ~seen coppice |
| 14:48.13 | jbot | coppice <n=chatzill@212.197.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 3d 5h 50m ago, saying: 'you mean packets are delivered by a cron job? :-\'. |
| 14:48.13 | Chotaire | the problem is, how could I tell MeetmeAdmin to only disconnect marked users... or how would meetmeadmin find out the correct user number? |
| 14:48.18 | MikeJ[Laptop] | _Paulo_, coppic is out of town |
| 14:48.27 | Chotaire | the only useful option I know is to kick the last user who joined... that won't help... it would only be a bad hack. |
| 14:48.30 | MikeJ[Laptop] | he won't be back for another couple days |
| 14:48.41 | _Paulo_ | Thanks MikeJ[Laptop] |
| 14:48.53 | *** join/#asterisk DarKnesS_WolF (n=sherif@212.103.170.135) |
| 14:49.30 | Zeeek | Chotaire my lizard brain summons a mental pitcher of a console for meetme |
| 14:49.36 | _Paulo_ | ~seen caio1982 |
| 14:49.38 | jbot | caio1982 <i=caio1982@CAcert-br/caio1982> was last seen on IRC in channel #debian-br, 12h 44m 1s ago, saying: 'HugLeo: puta merda, mesmo segundo'. |
| 14:49.40 | Zeeek | someone did this |
| 14:49.55 | DarKnesS_WolF | i have a fast question now i have zap 3 FXO channle and i want asterisk to ask for a password when any call coming from zap 3 trying to get 9XXXXXXXX how can i do that ? |
| 14:50.06 | Chotaire | zeeek: yes that was me ;) the problem is, I want anyone with dtmf be able to do that... not through a vt100 console. |
| 14:50.22 | Chotaire | anyone with a phone must be able to |
| 14:50.42 | _Paulo_ | DarKnesS_WolF, show application authenticate |
| 14:51.14 | Chotaire | and that's where option "X" comes in.. if I just knew how to kick specifically marked users only ;( |
| 14:51.36 | DarKnesS_WolF | _Paulo_: this should be where ? the dialplan ? |
| 14:52.35 | Chotaire | <PROTECTED> |
| 14:52.48 | Chotaire | if there was something like that with "kick", that would be just perfect. |
| 14:53.04 | _Paulo_ | DarKnesS_WolF, yes, in the dialplan |
| 14:53.09 | exonic | What's it going to take to get T38 in the main source tree of asterisk? |
| 14:53.30 | exonic | apparantly they have completed some patches that have been "rather successful" in the last week |
| 14:53.36 | _Paulo_ | DarKnesS_WolF, put a _9XXXXXXXXX extension in the context of zap 3. |
| 14:54.26 | _Paulo_ | DarKnesS_WolF, you can define the incoming context for zap3 in the zapata.conf file. |
| 14:54.39 | siim | anyone, how to resolve error: monmp3thread: Request to schedule in the past?!?! |
| 14:55.23 | _Paulo_ | siim, your machine is smp? |
| 14:55.52 | DarKnesS_WolF | _Paulo_: this is too much info for me ;-) i'm a newbie in this Asterisk |
| 14:56.03 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 14:56.34 | brad_mssw | exonic: most of those patches are pass-thru only ... |
| 14:57.30 | siim | _Paulo_ : its not smp, it is xen |
| 14:57.33 | brad_mssw | exonic: nothing to do actual T38 conversion (or whatever the heck you want to call it [e.g. from a SIP channel to a Zap channel]) |
| 14:57.50 | brad_mssw | exonic: though steveu is getting close |
| 14:57.54 | exonic | brad_mssw, Yes but I view it as more progress than leaving it out for patches |
| 14:58.30 | brad_mssw | exonic: well, yeah, the problem is I had the T38 patches on my system, and it broke a few things ... namely SIP transfers |
| 14:59.01 | brad_mssw | exonic: don't know why, I backed out the patch, and transferring calls with SIP started working again :/ |
| 14:59.02 | exonic | brad_mssw, ugh, I've got it running on a test system now. |
| 14:59.07 | _Paulo_ | siim, weird stuff... I think * is more for a realtime os than for virtualization ones... |
| 14:59.17 | MikeJ[Laptop] | anyone know what these are: 8? Feb 23 18:48:41 DEBUG[13835] chan_zap.c: Exception on 21, channel 6 |
| 15:00.15 | exonic | brad_mssw, care to talk about any other problems? Just so I know what i'm getting into? |
| 15:00.48 | *** join/#asterisk maayani (i=hidden-u@fw-int.transbeam.com) |
| 15:00.55 | brad_mssw | exonic: well, that was the main one ... I can't necessarily confirm any other issues were directly related to that patch (like dropped calls) |
| 15:01.39 | cj-rm | hey people... |
| 15:02.13 | *** join/#asterisk fulgas (n=fulgas@209.8.233.247) |
| 15:03.02 | siim | _Paulo_: well, other functionalities are working without errors |
| 15:03.05 | *** join/#asterisk af_ (n=af@ip-172-156.sn1.eutelia.it) |
| 15:03.12 | *** join/#asterisk Fedoracore6 (n=FC$@60.50.141.168) |
| 15:04.00 | siim | _Paulo_ : only if using MusicOnHold() fuction in dialplan, I get this error |
| 15:04.18 | _Paulo_ | siim, there are some kind of sync problem with threads... |
| 15:04.31 | exonic | siim, experiment with other moh classess |
| 15:04.38 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 15:05.00 | exonic | I am running native MoH w/o any trouble on SMP |
| 15:05.26 | _Paulo_ | siim, Its not my business but I'm curious... why are you using xen? |
| 15:05.58 | _Paulo_ | siim, rented server? |
| 15:06.51 | *** join/#asterisk bigjb_ (n=bigjb@195.60.10.114) |
| 15:07.05 | siim | _Paulo_, it is easier to me to use xen, its not rented |
| 15:08.47 | Fedoracore6 | Mrdigital: are u there |
| 15:09.25 | siim | _Paulo_, but how to solve these sync problems? |
| 15:09.46 | *** part/#asterisk maayani (i=hidden-u@fw-int.transbeam.com) |
| 15:09.53 | Hmmhesays | yes the newest incarnation of amp is pretty kickass |
| 15:09.58 | Hmmhesays | i'm liking it... a lot' |
| 15:09.59 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 15:10.00 | _Paulo_ | patch sched.c to check what is going on |
| 15:10.35 | _Paulo_ | siim, are you into C programming? |
| 15:10.47 | Blackthorn | Hello, what does "Zaptel Disabled Echo canceller because of tone (tx) on channel 2" mean? |
| 15:11.16 | *** join/#asterisk maayani (i=hidden-u@fw-int.transbeam.com) |
| 15:11.40 | siim | _Paulo_, no |
| 15:12.28 | _Paulo_ | there is a line like "if (ast_tvcmp(*tv, now) < 0) {" in sched.c |
| 15:13.14 | _Paulo_ | try another value instead of zero (-1, -2, -3). |
| 15:13.27 | Druken | mut: who told ya about that inprogress thingy? |
| 15:13.41 | Hmmhesays | god i love gmail |
| 15:13.48 | Hmmhesays | fan-farking-tastic |
| 15:14.14 | _Paulo_ | Blackthorn, do you have faxdetect=both in your zapata.conf? |
| 15:14.32 | mut | just reading |
| 15:15.01 | _Paulo_ | faxdetection will turn echo cancelling off, I think. |
| 15:15.02 | *** join/#asterisk Seldon1975 (n=someone@CPE0013105d0913-CM0014e8b6162c.cpe.net.cable.rogers.com) |
| 15:15.09 | Seldon1975 | hey y'all |
| 15:15.19 | Seldon1975 | does asterisk have it's own logging mechanism |
| 15:15.35 | *** join/#asterisk Craziman2 (n=Craziman@110-host62.planetc.com) |
| 15:15.35 | *** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41) |
| 15:15.37 | Seldon1975 | i cant finsd any Asterisk entries in the syslog configuration file |
| 15:15.38 | blitzrage | yes -- logger.conf |
| 15:15.51 | Zeeek | blitzrage LONG time no C |
| 15:15.57 | blitzrage | :)_ |
| 15:16.06 | GerbilWrk | Can anyone recommend a VoIP provider that can port numbers other then Teliax? |
| 15:16.07 | blitzrage | I've been away working :) |
| 15:16.08 | Zeeek | coming to astricon Eu ? |
| 15:16.15 | Seldon1975 | blitzrage: that doesn't specify how log rolling should happen though |
| 15:16.20 | Seldon1975 | blitzrage: does it? |
| 15:16.20 | blitzrage | GerbilWrk: MixNetworks (I work for them) |
| 15:16.32 | blitzrage | Seldon1975: I think you can tell it to output to syslog if you want |
| 15:16.59 | *** join/#asterisk Maveric (i=maveric@ip68-3-248-136.ph.ph.cox.net) |
| 15:17.01 | blitzrage | Seldon1975: and it should be able to roll logs -- at least you can from the CLI under the 'logger' command |
| 15:17.05 | Seldon1975 | blitzrage: the issue is this: Asterisk's log files are rolling with inordinate frequency; where should I look to change the rolling policy? |
| 15:17.12 | Zeeek | Seldon1975 /var/logs/asterisk |
| 15:17.17 | Seldon1975 | yes |
| 15:17.21 | Seldon1975 | thats where the logs are |
| 15:17.29 | Zeeek | you asked if there wrere any |
| 15:17.31 | Seldon1975 | but where is the rolling policy specified? |
| 15:17.36 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 15:17.50 | Zeeek | there is none. Asterisk just stops if theyget too big :) |
| 15:18.06 | Seldon1975 | but its rolling too often on my machine |
| 15:18.11 | Seldon1975 | like the log files are tiny |
| 15:18.14 | Seldon1975 | 200b |
| 15:18.20 | Zeeek | weird |
| 15:18.24 | Seldon1975 | yes indeed |
| 15:18.30 | Zeeek | what file name? |
| 15:18.45 | Seldon1975 | event_log , messages, verbose - all the ones mentioned in logger.conf |
| 15:18.54 | Zeeek | what user is * running as? |
| 15:18.59 | Seldon1975 | root |
| 15:19.10 | GerbilWrk | anyone else with any VoIP provider recommendations? |
| 15:19.11 | Zeeek | plenty of disk space avail? |
| 15:19.15 | Seldon1975 | yep |
| 15:19.30 | Zeeek | what time period? |
| 15:19.34 | Zeeek | by size only? |
| 15:19.39 | Craziman2 | Question: With Cisco 7960's and Asterisk is there a way to feed back a messge to a caller that the called party is on the phone? |
| 15:19.45 | Seldon1975 | yes thats what it looks like |
| 15:19.51 | Seldon1975 | they are all about the same size |
| 15:19.56 | Zeeek | interesting... |
| 15:20.18 | Druken | GerbilWrk: for where? |
| 15:20.20 | Zeeek | would seem to point to an allocation issue. Have you rebooted since? |
| 15:20.34 | Seldon1975 | i have to go to a meeting but if you have any ideas please write them and I'll scan the channel when I get back |
| 15:20.35 | Blackthorn | paul: not that i know off but i will check righ tnow. |
| 15:20.40 | blitzrage | Craziman2: use the ${DIALSTATUS} variable to check if a line is busy, then playback a file |
| 15:20.43 | Seldon1975 | zeek: reboot = yes |
| 15:20.54 | Zeeek | shit, really odd |
| 15:21.11 | blitzrage | I've never looked for logging rollover -- and I don't see a command for it in logger.conf |
| 15:21.21 | Zeeek | I didn't think there was any |
| 15:21.37 | Craziman2 | blitzrage: will that work if there are multiple lines allocated on the 7960? |
| 15:22.01 | iCEBrkr | err |
| 15:22.09 | Blackthorn | my zapdata.conf does not have any line faxdetect |
| 15:22.37 | blitzrage | Craziman2: multiple lines should equal multiple users -- unless you have it setup differently |
| 15:22.47 | blitzrage | iCEBrkr: is that in trunk -- because its not in 1.2.4 |
| 15:22.57 | iCEBrkr | blitzrage: Huh, it's in linux :P |
| 15:23.12 | blitzrage | iCEBrkr: oh -- but asterisk writes it's own logs |
| 15:23.13 | *** join/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 15:23.26 | iCEBrkr | blitzrage: So? |
| 15:23.28 | blitzrage | does it still read that file and use it? |
| 15:23.46 | *** part/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 15:23.46 | iCEBrkr | blitzrage: Maybe I'm misunderstanding what you wanna do |
| 15:23.51 | Craziman2 | blitzrage: each 7960 has to of the base extenstion (example) 101 and an auto answer intercom extenstion example (4101) |
| 15:24.03 | blitzrage | iCEBrkr: control asterisk log rotation when its not being sent to syslog |
| 15:24.14 | kristinG | can anyone offer some help in dumping a call from the pstn to the digium card via a lucent tnt? |
| 15:24.14 | iCEBrkr | blitzrage: I'd use logrotate. |
| 15:24.19 | Craziman2 | wish I could type... mean two :) |
| 15:24.20 | Zeeek | static char logger_rotate_help[] |
| 15:24.24 | [TK]D-Fender | blitzrage : So.... got your IP 500 running yet? ;) |
| 15:24.26 | iCEBrkr | blitzrage: But you're going to have a split-second of 'down time' |
| 15:24.35 | blitzrage | [TK]D-Fender: no -- I had to give it away :( |
| 15:24.42 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 15:24.55 | iCEBrkr | blitzrage: logrotate is used for rotating Apache's log files, along with mail, etc, etc, etc. |
| 15:24.59 | [TK]D-Fender | blitzrage : Damn, I'd have forked over $20 for it! |
| 15:25.17 | blitzrage | [TK]D-Fender: I didn't want to give it away -- I had to give it to someone who has been helping me with code |
| 15:25.45 | blitzrage | iCEBrkr: I'm not really the one who has to rotate his logs -- I'm just curious :) |
| 15:25.51 | siim | _Paulo_, I tried it, but there was no change |
| 15:25.51 | iCEBrkr | oh |
| 15:26.08 | visba | okay, i've search high and low for this one, seems simple...and important but i can't find the answer. how do i generate a second cdr for a hairpinned call that comes inf from the pstn and get's diverted back out to the pstn? |
| 15:26.14 | iCEBrkr | blitzrage: So ok, yeah.. I'd use logrotate to manage any log rotation/log management :P |
| 15:26.17 | blitzrage | but logrotate.conf doesn't sound like it'll do anything unless you direct Asterisk's logs to syslog |
| 15:26.41 | Zeeek | CLI> logger show channels |
| 15:26.41 | *** join/#asterisk p0g0_ (n=pogo@madwifi/support/p0g0) |
| 15:26.52 | Zeeek | I wonder if this displays something about rotation? |
| 15:26.56 | iCEBrkr | blitzrage: Nope. You can make your own definitions.. I managed over a gig worth of system logs generated from PHP scripts that ran this huge website |
| 15:27.03 | Zeeek | I do mine at the first of each month |
| 15:27.15 | _Paulo_ | siim, did you rebuild * ? |
| 15:27.20 | siim | _Paulo_, by the way, do I need to compile the sched.c or just edit it |
| 15:27.28 | blitzrage | Zeeek: doesn't look like it |
| 15:27.37 | siim | _Paulo_, yes I restarted asterisk |
| 15:27.48 | iCEBrkr | blitzrage: I had /etc/logrotate.d directory full of the definitions of what what/when to rotate |
| 15:28.04 | _Paulo_ | siim, make; make install |
| 15:28.10 | blitzrage | iCEBrkr: yep -- looks like you can specify a dir to control |
| 15:28.15 | iCEBrkr | Hell, you can even make logrotate FTP the logs offsite if you want |
| 15:28.37 | blitzrage | iCEBrkr: not sure that helps though since asterisk seems to be rotating the logs on its own -- too often |
| 15:28.45 | iCEBrkr | hrrrm. |
| 15:29.21 | Zeeek | the only thing I can see in source is that there is SIG that forces the rotate and a CLI command to do same |
| 15:30.20 | iCEBrkr | Yea, there doesn't appear to be any rotation control in logger.conf |
| 15:30.44 | Chotaire | oh yeah, I'm facing occasional deadlocks, probably caused by chan_capi-cm... how much I hate this. |
| 15:31.06 | *** join/#asterisk Defraz (i=t0tal@tim.mychoice.cc) |
| 15:31.47 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 15:31.50 | PakiPenguin | evening |
| 15:32.01 | *** join/#asterisk DeeJay[2] (n=bleh@office.abi.ca) |
| 15:32.28 | *** join/#asterisk rpm (i=russell@S010600111155e117.cg.shawcable.net) |
| 15:32.35 | DeeJay[2] | What could be a good motherboard to run asterisk with a TE410P (PCI-X) and socket 478 (P4). |
| 15:32.36 | DeeJay[2] | ? |
| 15:33.06 | blitzrage | ok -- I'm off to shower then work -- lates! |
| 15:33.20 | blitzrage | Zeeek: I think I'll be at AstriCon EU |
| 15:33.30 | blitzrage | not yet confirmed though |
| 15:33.45 | Zeeek | come to Paris |
| 15:33.53 | blitzrage | Zeeek: I think I'll be at all of them :) |
| 15:34.08 | Zeeek | that's extravagant |
| 15:34.20 | blitzrage | Zeeek: I work with the company that puts on the AstriCon events, so I'll be working -- but it won't cost me anything to go :) |
| 15:34.32 | Zeeek | I offered to speak |
| 15:35.14 | blitzrage | Zeeek: cool -- I don't have any say in that stuff -- I just make sure the network is up and running, and whatever errands need to be done :) |
| 15:35.18 | blitzrage | I'll probably end up speaking as well |
| 15:35.48 | Zeeek | I wonder where it will be? The web site doesn't say yet |
| 15:36.05 | Zeeek | anyway, you'd better shower before astricon :) |
| 15:36.12 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:36.14 | blitzrage | totally :) I have to go work on E911 |
| 15:36.15 | blitzrage | lates! |
| 15:36.49 | *** join/#asterisk Pupeno (n=Pupeno@19-161-126-200.fibertel.com.ar) |
| 15:37.11 | *** join/#asterisk Xen^ (i=linux@203.101.168.233) |
| 15:38.37 | visba | anyone know how to generate a second cdr on a hairpinned call? |
| 15:39.21 | Katty | Zeeek: ! |
| 15:39.32 | Katty | Zeeek: i've not bugged you in ages! |
| 15:40.09 | Zeeek | so I bought some rice milk |
| 15:40.12 | Katty | :> |
| 15:40.18 | Katty | did you make rice pudding with it? |
| 15:40.19 | Zeeek | and we both love it! |
| 15:40.22 | Katty | yay! |
| 15:40.43 | Zeeek | yeah rice milk is better than attended transfer |
| 15:40.58 | Katty | but the question is, is it better than blind transfer. |
| 15:41.05 | Katty | cause blind transfer is pretty darn hottt |
| 15:41.08 | Hmmhesays | I look better to blind girls |
| 15:41.08 | Zeeek | no, soy milk is though |
| 15:41.11 | _Paulo_ | coppice is at astricom? |
| 15:41.26 | Katty | Hmmhesays: i dunno about that. |
| 15:41.31 | kristinG | can anyone offer some help in dumping a call from the pstn to the digium card via a lucent tnt? |
| 15:41.37 | Katty | Hmmhesays: you look pretty good, for a scrawny little thing. |
| 15:42.15 | Hmmhesays | I'll be putting on some more muscle now that i'm done boozing |
| 15:42.15 | Hmmhesays | i've been eating like a freaking horse |
| 15:42.15 | Katty | i'm bigger than you, and i'm a vegan! |
| 15:42.27 | Katty | or, i claim to be bigger, anyway |
| 15:42.44 | *** join/#asterisk compuwizz (n=compuwiz@blacksburg-bsr1-69-174-71-216.chvlva.adelphia.net) |
| 15:42.50 | Zeeek | I lost my veganity when I ate meat |
| 15:42.57 | Katty | Zeeek: it happens. |
| 15:43.01 | synthetiq | body image problems, that asterisk story |
| 15:43.06 | Zeeek | I was only 1 years old! |
| 15:43.11 | Katty | Zeeek: horrors! |
| 15:44.01 | Pupeno | compuwizz: ok... tell me what you did and what versions you were using. |
| 15:44.07 | synthetiq | you set switch type to 5ess kristin? |
| 15:44.48 | compuwizz | I am using SpanDSP 0.0.2pre20 and asterisk 1.2.4 |
| 15:45.16 | compuwizz | I've installed libtiff and I assume that is the only other requirement for tx_fax and rx_fax |
| 15:45.26 | compuwizz | I applied the makefile patch from spandsp |
| 15:45.39 | Pupeno | compuwizz: you installed spandsp I supouse... where ? |
| 15:45.59 | Katty | bkw_: you around, deary? |
| 15:46.13 | Zeeek | is he back? |
| 15:46.18 | Zeeek | gone for years |
| 15:46.28 | _Paulo_ | compuwizz, sometimes you will have to unload the zaptel kernel modules |
| 15:46.43 | Pupeno | _Paulo_: what for ? |
| 15:47.11 | _Paulo_ | Pupeno, if he is using libunicall, for example... |
| 15:47.30 | compuwizz | I used the default spandsp settings, so I'm not sure exactly where it might be |
| 15:47.44 | Pupeno | _Paulo_: compuwizz's problem is compiling, not runnig. |
| 15:47.44 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 15:48.04 | compuwizz | I have the error message, shall I paste it in here? |
| 15:48.33 | _Paulo_ | compuwizz, did you run "./configure --prefix=/usr" ??? |
| 15:48.42 | compuwizz | yex |
| 15:48.44 | compuwizz | *yes |
| 15:48.45 | Pupeno | compuwizz: if it is less than three lines, yes, if not, use http://paste.lisp.org and paste the url. |
| 15:49.15 | Pupeno | compuwizz: then you installed on /usr, that's good. |
| 15:50.27 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfl2j.dialup.mindspring.com) |
| 15:50.40 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfl2j.dialup.mindspring.com) |
| 15:51.00 | compuwizz | http://paste.lisp.org/display/17372 |
| 15:52.37 | synthetiq | yiu know there was a krisining at my previeous job, had her ccnp, i never saw her tho, first off she claimed sexual harassment, so got comp time deal or soemthing, then when she was supposed to come back magically breaks her foot, in a car accident on the way, getting more comp time some how |
| 15:52.45 | Pupeno | compuwizz: I remember we had trouble with 0.0.2pre20 regarding a re-factoring not finish (I am not 100% sure it was that version), why don't you try a newer one ? http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre25/ |
| 15:53.05 | compuwizz | ok, I shall |
| 15:53.24 | Pupeno | be sure the patch applies correctly or apply it by hand (it's not hard) |
| 15:54.13 | _Paulo_ | compuwizz, are you using Debian? |
| 15:54.18 | compuwizz | Gentoo |
| 15:55.14 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 15:55.16 | _Paulo_ | caio1982 has a nice set of * .debs Debian/Stable. |
| 15:55.38 | *** join/#asterisk corruptor (n=andrew55@www.tae.ru) |
| 15:55.46 | cpm | ping doesn't really tell you enough, try mtr -r -c 10 addressOfServer |
| 15:55.53 | cpm | errrp! |
| 15:56.35 | *** join/#asterisk Utah_Dave (n=boucha@c-67-172-255-244.hsd1.ut.comcast.net) |
| 16:01.54 | *** join/#asterisk mikefoo (n=mikefoo@64.124.169.2) |
| 16:02.14 | mikefoo | Anyone interested in buying a cisco 7960? Works without a problem. |
| 16:03.35 | Katty | i'm not. |
| 16:04.32 | kpettit | I'm using a Sangoma a200. Trying to configure it and it keeps wanting to define a span. |
| 16:04.41 | kpettit | but with a a200 Ishouldn't have too. Any idea? |
| 16:06.16 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 16:07.21 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 16:08.58 | brad_mssw | compuwizz: if you're using Gentoo, just use the gentoo-voip overlay to get the latest spandsp, and rx/txfax |
| 16:09.09 | brad_mssw | compuwizz: a lot easier than manually compiling it |
| 16:09.21 | compuwizz | ok, thank you |
| 16:09.40 | Tagor | How can I play the onhold music for 20 seconds? |
| 16:09.47 | brad_mssw | compuwizz: http://svn.netdomination.org/gentoo-voip/wiki/OverlayRsync |
| 16:12.47 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 16:13.10 | compuwizz | I will look into these later in the day. Thank you everyone for your suggestions |
| 16:13.40 | jbenson | Hi there - has anyone any experience with the Splicecom Maximiser PBX please? |
| 16:13.50 | Hmmhesays | i wonder if ser's lcr module can be modified to do load balancing |
| 16:14.57 | GerbilWrk | has anyone had experience running asterisk on a 2.6Ghz Celeron processor? |
| 16:15.10 | Skid | i've ran it off a 1.4 celeron |
| 16:15.18 | Skid | infact, still am doing i guess unless someone's wiped hte box :P |
| 16:15.27 | Skid | doesn't do much though, I moved it to a p4 |
| 16:15.42 | GerbilWrk | Did you run a T1 card in it by chance? |
| 16:16.04 | Skid | nope, just pure voip |
| 16:16.26 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 16:16.28 | GerbilWrk | who are you using for the provider? |
| 16:16.33 | cpm | WTF happened to the mailing list archives? |
| 16:16.54 | cpm | Or the mailing lists for that matter? |
| 16:16.58 | Skid | GerbilWrk: provider? |
| 16:17.00 | Skid | outbound proxy? |
| 16:17.10 | GerbilWrk | sure |
| 16:17.19 | Skid | mmmmm voip company i know of here |
| 16:17.22 | Skid | voip.co.uk |
| 16:17.34 | GerbilWrk | ahh, .uk, scratch that idea then |
| 16:17.36 | RoyK | hi |
| 16:18.10 | tracinet | anyone notice that priority jumping still works even if you don't specify it at the top of extensions.conf? |
| 16:18.32 | Skid | i hate to be off topic, but check this dog out: http://video.google.com/videoplay?docid=-8025218193636377093 |
| 16:18.46 | RoyK | it seems to me ss7 can be compared to the whole ipv4 suite |
| 16:18.47 | *** join/#asterisk redax (n=redax@r6.hu) |
| 16:18.49 | redax | hi |
| 16:19.29 | redax | what to do with this: (incoming SIP call) |
| 16:19.29 | redax | WARNING[3224]: chan_sip.c:3492 process_sdp: Unknown SDP media type in offer: video 30002 RTP/AVP 34 |
| 16:19.43 | Hmmhesays | every time I see AVP i think alien versus predator |
| 16:19.58 | RoyK | wtf is avp? |
| 16:20.10 | Abydos313 | a good movie |
| 16:20.16 | redax | kaspersky antivirus or what |
| 16:20.32 | Zeeek | anal video packet |
| 16:20.34 | Katty | Abydos313: and a good game. |
| 16:20.45 | Abydos313 | never played the game |
| 16:20.52 | Katty | Hmmhesays: how's you? |
| 16:21.18 | Hmmhesays | attribute value pairs |
| 16:21.31 | Hmmhesays | Katty: fine, starting to play with openser's LCR module |
| 16:21.32 | Abydos313 | it's so cold her this morning |
| 16:21.36 | austinnichols101 | avp = active virus protection in the mcafee world |
| 16:21.38 | Hmmhesays | they game kicked ass, the movie sucked it |
| 16:21.43 | [TK]D-Fender | Hmmhesays : AVP = Action Velo-Plus (a quebec bike manufaturer... MINE actually!) |
| 16:21.46 | Katty | Hmmhesays: and chica? |
| 16:21.48 | redax | please help with that SDP media stuff ;-) |
| 16:21.50 | [TK]D-Fender | Katty: mew. |
| 16:22.04 | Hmmhesays | Her and the fiance got in a heated arguement over me last night, HAHA |
| 16:22.18 | Katty | [TK]D-Fender: mew. |
| 16:22.24 | Katty | Hmmhesays: shame on you. |
| 16:23.00 | Katty | Hmmhesays: and the outcome? |
| 16:23.20 | Hmmhesays | same as before, now he'll be all nice to her for awhile and she won't call |
| 16:23.28 | Katty | RoyK: uh, thanks. |
| 16:23.40 | mikefoo | [TK]D-Fender: hey.. |
| 16:23.41 | Katty | Hmmhesays: :< |
| 16:23.59 | kristinG | can anyone offer some help in dumping a call from the pstn to the digium card via a lucent tnt? |
| 16:24.08 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 16:24.10 | Katty | hmm, tnt. |
| 16:24.14 | Katty | tnt is good for sploding things. |
| 16:24.18 | Katty | including calls. |
| 16:24.26 | kristinG | :p |
| 16:24.35 | fugitivo | tnt is a tv channel |
| 16:24.58 | Hmmhesays | some dude went up and played that on jam night |
| 16:25.01 | tuxinator_linux | kristinG: Did you check the TNT page on voip-info.org? |
| 16:25.33 | cpm | kristinG, huh? |
| 16:26.08 | kristinG | yes |
| 16:26.25 | kristinG | you are everywhere cpm |
| 16:27.26 | *** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 16:27.37 | cpm | kristinG, cluelessness is universal |
| 16:27.50 | *** join/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
| 16:28.04 | mroth_imm | got a question for you all about load averages |
| 16:28.05 | kristinG | apparently so |
| 16:28.18 | cpm | :p |
| 16:28.31 | websae | anyone know how to get yellow page ad ??? i live in the milwaukee area, i know they have the yellow page one book by SBC, and want to know how to still be in that switching my system over to voip...any ideas? |
| 16:28.37 | mroth_imm | we currently have 100 concurrent calls with digital recording via monitor |
| 16:28.38 | cpm | So, you are pulling a T1 *from* your tnt to a digium card? |
| 16:28.43 | mroth_imm | on a 4 processor box |
| 16:28.54 | cpm | I'm trying to get my head around what you are doing. |
| 16:29.05 | mroth_imm | curious to know when you'd start worrying about load averages on that machine |
| 16:29.16 | mroth_imm | what number would be the "oh shit" number : ) |
| 16:29.21 | Hmmhesays | file oh file |
| 16:29.26 | file[laptop] | yes? |
| 16:29.39 | Hmmhesays | you still have that link to gemini, i can' find it anywhere |
| 16:29.43 | cpm | mroth_imm, when they hit 40 or so |
| 16:29.47 | file[laptop] | Hmmhesays: oh, yeah |
| 16:30.50 | mroth_imm | cpm: thank you for your answer...do you have any background info i could look at (links, etc) that i could review? |
| 16:31.38 | cpm | mroth_imm, do you have a problem? Or are you looking for a problem to have? |
| 16:32.04 | cpm | meaning, are you seeing really high loads, and freaking out? |
| 16:32.35 | mroth_imm | adding load to the box...trying to see *when* we'll have problems |
| 16:32.48 | *** join/#asterisk Rowter (n=Silver@201.133.210.220) |
| 16:32.52 | mroth_imm | basically trying to extrapolate when our 100 calls go up to 200...300...400...when things will break down |
| 16:32.56 | Rowter | join #openpbx |
| 16:32.59 | Rowter | ups |
| 16:33.11 | Rowter | <PROTECTED> |
| 16:33.23 | mroth_imm | at 100 calls i see the load average bouncing between 1.5 and 4 (all digitally recorded via monitor) |
| 16:33.25 | cpm | what's the box? |
| 16:33.34 | mroth_imm | dell poweredge 6850 |
| 16:33.45 | cpm | your ram sockets will melt off first :) |
| 16:34.27 | mroth_imm | 4x3.16 Xeon processor...20 gigs of ram (16 ram disk for monitor, 4 for the system itself |
| 16:34.36 | websae | anyone know...if you can stil have a yellow page ad in SBC with a VoIP DID ???? :) thanks |
| 16:34.37 | Chotaire | zeeek: thanks to a bug, I got it to work ;) |
| 16:34.38 | cpm | I expect if that is your nominal load, your are underpowered. Might start looking to see where your bottlenecks are. |
| 16:34.45 | Chotaire | I gotta patch some meetme source though... bbl |
| 16:35.00 | RoyK | mroth_imm: wtf do you need that shite for? |
| 16:35.17 | mroth_imm | cpm: we aren't having problems now, but i just wanted to know when the load average should start to scare me |
| 16:35.34 | cpm | if you are seeing loads of 4, with that much box, for a hundred calls, something is amiss |
| 16:35.39 | RoyK | mroth_imm: use sysstat and so on to monitor it |
| 16:35.43 | Zeeek | Chotaire you should publish the solution |
| 16:35.51 | cpm | what RoyK said |
| 16:36.03 | RoyK | mroth_imm: carefully monitor it with oprofile as well if the kernel time starts growing |
| 16:36.21 | RoyK | mroth_imm: i've seen horrible kernel time with just sip/sip bridging, nothing more |
| 16:36.30 | Chotaire | Zeeek: I also found bad documentation... yup, I will think about how to publish it. |
| 16:36.32 | mroth_imm | RoyK: one central asterisk server for all of our queues...400-500 concurrent calls with dig rec + all of our agents (200 - 250) spread across 5 offices |
| 16:36.33 | Chotaire | fixing.... |
| 16:36.43 | Zeeek | publish at least on the wiki |
| 16:36.49 | RoyK | mroth_imm: sounds scary |
| 16:36.50 | mroth_imm | yeah, all calls are sip to sip on the box |
| 16:36.59 | Chotaire | " 'e' -- Eject last user that joined (except admin)\n" |
| 16:37.07 | RoyK | mroth_imm: how's the I/O subsystem on the system? how many drives? what kind of drives? raid level? |
| 16:37.07 | Chotaire | that one will do the trick ;) it's undocumented |
| 16:37.12 | mroth_imm | no kidding, that's why i'm asking here and now before the number of calls shoots up |
| 16:37.32 | cpm | kristinG, you still there? |
| 16:37.34 | mroth_imm | RoyK: 2 scsi drives in a hardware raid 1 |
| 16:37.41 | mroth_imm | but all of our digital recordings are going to a ram disk |
| 16:37.45 | RoyK | mroth_imm: sounds quite low |
| 16:37.47 | RoyK | ah |
| 16:37.47 | RoyK | ok |
| 16:38.01 | RoyK | mroth_imm: my testing showed a single xeon was maxed out with _only_ SIP/SIP bridging about 400 calls |
| 16:38.12 | RoyK | mroth_imm: asterisk really doesn't scale that well |
| 16:38.13 | RoyK | also |
| 16:38.15 | mroth_imm | monitor dies at 60 calls otherwise, regardless of the rest of the box - i/o bottleneck |
| 16:38.20 | Chotaire | Zeeek: now all I gotta make sure is that admins will also have enter/leave sounds.... that's the only patch I have to do to app_meetme.c (besides missing documentation) |
| 16:38.38 | mroth_imm | royk: single cpu...do you remember the load average at the time? |
| 16:38.41 | RoyK | orprofile showed mostly kernel time was the problem - mainly i/o related |
| 16:38.41 | Chotaire | Zeeek: I also did a script on how to change the hardcoded enter/leave sounds ;) |
| 16:38.52 | sevard | mroth_imm: which xeon? how much cache? what's the clock speed? |
| 16:38.53 | RoyK | mroth_imm: load avg isn't really relevant imho |
| 16:39.12 | RoyK | sevard: cache amount is not really relevant for this app |
| 16:39.18 | mroth_imm | 1024 kb on each |
| 16:39.23 | RoyK | quite low |
| 16:39.23 | Zeeek | Chotaire all this stuff should go in the wiki. Generations to follow will revere you for it |
| 16:39.34 | mroth_imm | 3.16 GHz |
| 16:39.34 | sevard | I'm just curious about the max ammount of sip calls per cycle |
| 16:39.42 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 16:39.50 | mroth_imm | did i mention the calls are all u-Law (lots of frames!!) |
| 16:39.59 | RoyK | Zeeek: perhaps i should write an article about my testing as well? 1000 calls sip/sip bridging and so on? |
| 16:40.03 | sevard | I'm thinking about building a server that would service many customers, or is it better to build serveral and load balance across them |
| 16:40.14 | RoyK | mroth_imm: just as many frames as with g.729 |
| 16:40.20 | Zeeek | everyone should write articles on everything |
| 16:40.22 | RoyK | mroth_imm: 50 frames per second per direction |
| 16:40.28 | RoyK | per call |
| 16:40.29 | mroth_imm | once we hit the ceiling on our architecture, it's all going out to the wiki for consumption and criticisim |
| 16:40.35 | Chotaire | Zeeek: I totally agree.. |
| 16:40.40 | cpm | no one should write articles on nothing |
| 16:40.44 | mroth_imm | oh, is that so...i though uncompressed meant more frames...mistaken there...thanks! |
| 16:40.52 | Chotaire | <PROTECTED> |
| 16:40.56 | Chotaire | i think that's the line... |
| 16:41.07 | Chotaire | ;) |
| 16:41.14 | sevard | nioooce |
| 16:41.32 | heroine | is asterisk-1.0 still supported or it's definitively out of date ? |
| 16:41.44 | RoyK | heroine: it's never been really supported :) |
| 16:41.52 | mroth_imm | RoyK: what do you consider relevant...with 90 calls right now we are at 84.6 percent idle...1.76 2.02 1.89 loads |
| 16:42.13 | RoyK | what does sar / vmstat say? |
| 16:42.14 | heroine | RoyK: tss tss :) |
| 16:42.36 | RoyK | imho what's important is not the number of processes waiting for cpu but the total effective load |
| 16:42.44 | RoyK | 'load' only shows the former |
| 16:42.47 | mroth_imm | 11:42:27 AM CPU %user %nice %system %iowait %idle |
| 16:42.54 | RoyK | ~pb |
| 16:42.55 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 16:42.57 | mroth_imm | 11:42:28 AM all 1.75 0.00 12.03 0.00 86.22 |
| 16:43.10 | heroine | i shouldn't tell that to my boss who keep thinking that 1.0.7 is more stable than the 1.2.4 in test ... |
| 16:43.23 | Zeeek | go to 1.2 |
| 16:43.24 | mroth_imm | processor 0 is at 73...1, 2, and 3 around 90 |
| 16:43.26 | RoyK | mroth_imm: see? all is system load. |
| 16:43.38 | RoyK | mroth_imm: do an oprofile readout |
| 16:43.41 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 16:43.43 | Chotaire | recompiling, testing.. wish me luck. |
| 16:44.14 | Tagor | Has someelse here also problems with MOH native? I notice a lot of shocking while playing the sound file |
| 16:44.15 | Chotaire | if that works, I managed to complete a full partyline system on asterisk, all administratable by dtmf ;) |
| 16:44.20 | cpm | as does me |
| 16:44.47 | RoyK | brb |
| 16:44.49 | mroth_imm | RoyK: thanks, i will look into it... |
| 16:46.29 | mroth_imm | RoyK: i wonder if load average is somewhat overstated on the box, given that there are 87 processes, but only one is not sleeping (asterisk) |
| 16:47.20 | *** part/#asterisk mover (n=dlu@gw-dus-net.dus.de.ncore.net) |
| 16:47.39 | *** join/#asterisk Rhizome (i=_tor@shodan.nognu.de) |
| 16:48.45 | Chotaire | YES! |
| 16:50.09 | sevard | Chotaire: wiki it up. |
| 16:52.29 | Druken | anyone here on rogers and had problems connecting to a foreign mailserver? |
| 16:53.41 | *** join/#asterisk Fedoracore6 (n=FC$@60.50.141.168) |
| 16:53.59 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 16:54.00 | mut | they probly block it |
| 16:54.04 | mut | less spammination |
| 16:54.20 | RoyK | mroth_imm: ping |
| 16:54.26 | SplasPood | hrm... wonder what's so exciting.. http://newyork.craigslist.org/mnh/sad/137889552.html |
| 16:54.27 | mroth_imm | RoyK: i wonder if load average is somewhat overstated on the box, given that there are 87 processes, but only one is not sleeping (asterisk) |
| 16:55.01 | RoyK | mroth_imm: i beleive loadavg shows waiting threads as well |
| 16:55.20 | RoyK | mroth_imm: man ps, there are flags to show all threads as well |
| 16:55.42 | RoyK | or perhaps even top can do that - dunno |
| 16:56.13 | *** join/#asterisk skkip (n=Skipper@216.160.91.91) |
| 16:56.15 | mroth_imm | ps -eLf | grep asterisk | wc -l |
| 16:56.20 | mroth_imm | 89 |
| 16:56.23 | *** join/#asterisk apardo (n=apardo@87.218.44.213) |
| 16:56.28 | RoyK | sounds reasonable |
| 16:56.52 | mroth_imm | i really appreciate your input...i've kind of been thrown into this position...was doing windows desktop programming before this : ) |
| 16:57.31 | RoyK | lol |
| 16:57.32 | RoyK | ok |
| 16:57.33 | RoyK | :) |
| 16:57.34 | *** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com) |
| 16:57.47 | mroth_imm | i will look into what is causing such high system resource utilization...in your opinion, is this intrinsic to asterisk, or is there an obvious configuration tweak that i should start with |
| 16:57.59 | RoyK | mroth_imm: man opcontrol |
| 16:58.07 | RoyK | mroth_imm: using 2.6? |
| 16:58.12 | RoyK | linux 2.6? |
| 16:58.14 | RoyK | uname -r |
| 16:58.17 | mroth_imm | yes...2.6 on FC3 |
| 16:58.29 | Abydos313 | so what other software is in copetition with asterisk? |
| 16:58.32 | RoyK | opcontrol allows you to start oprofiled |
| 16:58.34 | *** join/#asterisk trelane_ (n=trelane@mail.allthingsit.com) |
| 16:58.35 | mroth_imm | abe...requires FC3 or RHEL3 for full support :) |
| 16:58.51 | mroth_imm | and this will allow me to research where the resources are being spent? |
| 16:58.54 | RoyK | Abydos313: perhaps openpbx, but that's about it. or yate... freeswitch is not finished yet |
| 16:59.10 | Abydos313 | so asterisk is the farthest along? |
| 16:59.26 | mroth_imm | we tossed aside sipx for asterisk in the VERY beginning...don't know where that is now |
| 16:59.28 | RoyK | Abydos313: depends what you need |
| 17:00.01 | RoyK | mroth_imm: oprofile output will tell you what kernel threads/calls are using cpu |
| 17:00.12 | RoyK | and currently it's only kernel threads using time |
| 17:00.31 | RoyK | or almost only those according to your vmstat/sar output |
| 17:00.50 | Abydos313 | well actually my interest is in a call center with 30 users. i talked to my ISP rep and he kinda said the free stuff like asterisk is OK but no where near ready for his clients. |
| 17:00.56 | mroth_imm | i wonder...we are using nfs to move files from the ram disk to a remote machine for mixing, etc... |
| 17:01.07 | RoyK | Abydos313: asterisk can do that - so can yate - so can openpbx |
| 17:01.28 | mroth_imm | nfs hooks into the kernel...could be the culprit |
| 17:01.38 | Abydos313 | i know it can, but how well can it do it |
| 17:02.01 | RoyK | mroth_imm: doubt it. my testing showed LOTS of kernel time with just sip/sip bridging with no other activity |
| 17:02.25 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 17:02.38 | RoyK | mroth_imm: the oprofile output may perhaps not be intresting to you if you don't know kernel internals, but it may give the asterisk coders an idea of how to fine-tune the code |
| 17:02.39 | Abydos313 | we have 'televantage' now.. its' ok. but i would never use it again. |
| 17:02.48 | mroth_imm | RoyK: okay...i would say that indicates that it is intrinsic to asterisk...i will research it further, but it may just be *the way it is* |
| 17:03.01 | Abydos313 | we are currently knocking down 4 t1's to 2.. |
| 17:03.23 | Druken | mut: yeah they do block it... the stupid fucks |
| 17:03.24 | RoyK | mroth_imm: nothing is 'the way it is' to a good developer :) |
| 17:03.29 | mroth_imm | RoyK: yes, i would like to make this pertinent to the dev list, so i will really look into that suggestion |
| 17:03.41 | RoyK | thanks |
| 17:03.44 | mut | heh |
| 17:03.46 | *** part/#asterisk Utah_Dave (n=boucha@c-67-172-255-244.hsd1.ut.comcast.net) |
| 17:03.47 | mut | saved on spam from them |
| 17:03.53 | RoyK | mut: wtf? |
| 17:03.56 | mroth_imm | RoyK: i am speaking relative to my configuration with the current asterisk installation |
| 17:04.05 | mut | RoyK: ftw? |
| 17:04.12 | RoyK | :P |
| 17:05.01 | tzafrir | heroine, if you go for 1.0.x, go at least for 1.0.10 |
| 17:05.53 | RoyK | heroine: why would you use 1.0? |
| 17:05.53 | _Paulo_ | openpbx or yate can receive fax like *??? |
| 17:05.54 | mut | w000t |
| 17:06.11 | RoyK | _Paulo_: opbx, yes, yate, no |
| 17:06.12 | RoyK | iirc |
| 17:06.29 | mroth_imm | RoyK: once again, thank you...it'll take a while before i digest everything you've said, but i truly appreciate it |
| 17:06.38 | tzafrir | faxing is not exactly built into asterisk. you can use rx_fax/tx_fax |
| 17:07.00 | RoyK | mut: 35k wot? icelandic krónur? |
| 17:07.02 | RoyK | :) |
| 17:07.03 | _Paulo_ | RoyK, thanks |
| 17:07.15 | mut | usd |
| 17:07.19 | RoyK | :) |
| 17:08.00 | tzafrir | mroth_imm, on 2.6 you get get a list of threads of the process in /proc/<num>/tasks |
| 17:09.02 | mroth_imm | tzafrir: awesome...this whole irc session is going into my notes : ) |
| 17:09.03 | RoyK | tzafrir: there's some t.38 stuff in trunk, and even more of it in opbx since some of it is pure gpl |
| 17:09.45 | mroth_imm | there's a lot of good stuff in that dir tzafrir |
| 17:09.48 | RoyK | mroth_imm: tried oprofile yet? |
| 17:10.38 | mroth_imm | RoyK: will do...kinda frantic here today : ) |
| 17:10.44 | RoyK | :) |
| 17:10.57 | _Paulo_ | tzafrir, app_txfax is not working for my setup, so I looking for alternatives. |
| 17:11.23 | _Paulo_ | (app_rxfax is ok) |
| 17:11.32 | RoyK | it certainly is..... |
| 17:11.52 | websae | is it still possible to be in the yellow page adds with a voip number??? |
| 17:11.55 | websae | anyone know? |
| 17:12.20 | cthompson | websae: I can't imagine why it wouldn't be allowed |
| 17:12.27 | cthompson | they just want a number that works |
| 17:12.34 | RoyK | websae: there is no good answer, but at least here, in .no, it's quite possible |
| 17:13.05 | RoyK | cthompson: it might vary between countries |
| 17:13.19 | cthompson | it might |
| 17:13.34 | cthompson | wait |
| 17:13.40 | *** join/#asterisk Huynh (n=chatzill@w034.z064001163.sjc-ca.dsl.cnc.net) |
| 17:13.41 | cthompson | you mean the US isn't the only country in the world? |
| 17:13.46 | cthompson | :) |
| 17:13.53 | Huynh | hello |
| 17:14.12 | *** join/#asterisk lazzarello (n=lee@dsl254-077-209.nyc1.dsl.speakeasy.net) |
| 17:14.22 | cthompson | heh |
| 17:15.08 | Huynh | anyone know when mailing list will be back up? |
| 17:16.05 | Huynh | link on the digium site returns 404 |
| 17:16.10 | lazzarello | is it possible to change the umask asterisk uses to write voicemail files? I need the group read bit set so another application not running as asterisk can access the recordings. |
| 17:17.06 | RoyK | Huynh: that's not the mailing list. it's the archive.. |
| 17:17.27 | Huynh | ok, but it's still down |
| 17:17.29 | RoyK | lazzarello: just rtfs :) |
| 17:17.33 | lazzarello | google has a cache: http://www.google.com/search?as_q=group+read+bit+voicemail&num=10&hl=en&btnG=Google+Search&as_epq=&as_oq=&as_eq=&lr=&as_ft=i&as_filetype=&as_qdr=all&as_occt=any&as_dt=i&as_sitesearch=lists.digium.com&as_rights=&safe=off |
| 17:17.33 | Fedoracore6 | hai i still have problem i my configuretion witd asterisk code |
| 17:17.42 | Fedoracore6 | http://pastebin.com/578643 |
| 17:17.54 | _Paulo_ | Can bayonne do fax like *? |
| 17:18.14 | lazzarello | RoyK, besides patching the source code... |
| 17:18.27 | RoyK | lazzarello: i don't know. i don't think so...... |
| 17:19.02 | Fedoracore6 | i alredy try a lot configuration in extensions.conf res_mysql_conf,extconfig.conf , cdr_mysql.conf |
| 17:19.13 | RoyK | Fedoracore6: sir, as always, an error message itself does not really mean much without the configuration/dialplan/whatever in place |
| 17:19.21 | lazzarello | I'd imagine making /var/spool/asterisk/voicemail setuid is a Bad Thing^tm |
| 17:19.27 | RoyK | Fedoracore6: pastebin the config as well |
| 17:19.39 | Fedoracore6 | http://pastebin.com/578643 |
| 17:19.41 | lazzarello | I already made it setgid |
| 17:20.09 | RoyK | Fedoracore6: that only shows verbose output |
| 17:20.32 | RoyK | lazzarello: create a crontab as root to chmod it :) |
| 17:21.09 | *** part/#asterisk maayani (i=hidden-u@fw-int.transbeam.com) |
| 17:21.31 | lazzarello | was thinking about that...there could be a small gap between voicemail left and user's ability to play files via said external application. |
| 17:21.59 | trelane_ | what is wrong with this line? exten => 2,2,Set(${DB( forward/${CALLERIDNUM}=${FORWARD} )}) asterisk is claiming it can't pull variables from it but I think the brackets are right and the format is correct. |
| 17:22.06 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 17:22.42 | lazzarello | it's funny cause the .txt description of the .wav file is umask 022 but the .wav files are 077 |
| 17:23.25 | Zeeek | trelane_ Set(DB(RFLAGS/DND)=1) |
| 17:23.54 | trelane_ | Zeeek, right, unless I missed something what was pasted follows that example just using variables |
| 17:24.02 | Fedoracore6 | hemm i try |
| 17:24.10 | Zeeek | note the equal sign position |
| 17:24.13 | Fedoracore6 | but this error |
| 17:24.14 | Fedoracore6 | Mar 1 12:21:44 ERROR[4188]: res_config_mysql.c:615 mysql_reconnect: MySQL RealTime: Failed to connect database server ivr on localhost. Check debug for more info. |
| 17:25.30 | lazzarello | Fedoracore6, can you connect via a shell with the same user as asterisk? |
| 17:25.30 | *** join/#asterisk ckruetze (n=ckruetze@i577A7125.versanet.de) |
| 17:26.22 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 17:26.27 | Zeeek | trelane_ I think you want |
| 17:26.30 | Zeeek | exten => 2,2,Set(${DB(forward/${CALLERIDNUM})=${FORWARD} }) |
| 17:26.33 | Zeeek | <PROTECTED> |
| 17:27.08 | Fedoracore6 | hemm.... i dont understand |
| 17:27.18 | Fedoracore6 | what lazzarello mean |
| 17:27.33 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 17:27.48 | trelane_ | Zeeek, no I'm still not getting it there, set isn't seeing anything to set |
| 17:28.02 | Fedoracore6 | <PROTECTED> |
| 17:28.06 | Zeeek | well at least you have the correct syntax now |
| 17:28.14 | Fedoracore6 | so my english no good |
| 17:28.25 | lazzarello | Fedoracore6, mysql -u asteriskdbuser -p |
| 17:28.28 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 17:30.04 | Fedoracore6 | mysql -u ivr -p |
| 17:30.11 | Fedoracore6 | my database name ivr |
| 17:30.17 | Fedoracore6 | so i put ivr right |
| 17:31.55 | Fedoracore6 | yes Lazzarello |
| 17:32.03 | lazzarello | Fedoracore6, no. mysql -u <username> -p |
| 17:32.04 | Fedoracore6 | its right i put ivr -p |
| 17:32.13 | lazzarello | then enter the password asterisk is using to connect |
| 17:32.28 | Fedoracore6 | username i root |
| 17:32.33 | lazzarello | if you can't connect, your DB permissions are messed up. |
| 17:32.43 | lazzarello | do not let asterisk connect to the DB as root |
| 17:33.40 | Fedoracore6 | hemm i put the username root password 1234 |
| 17:34.53 | Fedoracore6 | oic |
| 17:34.53 | tzafrir | Fedoracore6, give that system a different user |
| 17:35.07 | Fedoracore6 | ok like user name edifier |
| 17:35.20 | Fedoracore6 | if wanna i put the password or not |
| 17:36.53 | Fedoracore6 | so i must set the diffrent user i mysql data base |
| 17:36.57 | Fedoracore6 | its right |
| 17:38.27 | tzafrir | It's not a "must". It will work perfectly well as "root", just as Asterisk will run perfectly well as root |
| 17:39.28 | fugitivo | nothing should be run as root |
| 17:39.44 | fugitivo | only sshd :) |
| 17:40.04 | _Sam-- | do radio waves in the atmosphere travel at near the speed of light? |
| 17:40.08 | lazzarello | oh hell! why does asterisk not obey the umask when writing voicemail sound files, only the .txt description? |
| 17:40.33 | fugitivo | _Sam--: hell no |
| 17:40.43 | _Sam-- | fugitivo: wrong! |
| 17:40.50 | Fedoracore6 | so tzafrir i cannot use the username"root' |
| 17:40.53 | _Sam-- | hold on i'll give you the URL i just found |
| 17:40.58 | Fedoracore6 | so i must change the username |
| 17:41.01 | _Sam-- | "Yes, all electromagnetic radiation -- from radio waves to x-rays -- travel at the speed of light. In empty space this speed is approximately 300,000 kilometers per second" |
| 17:41.09 | *** join/#asterisk razu_ (n=razu@ip228.cab74.mus.starman.ee) |
| 17:41.27 | iDunno | it's. all. broken. |
| 17:41.31 | _Sam-- | an expert on wireless network implentation just told me the same thing...i couldnt beleieve it |
| 17:41.47 | _Sam-- | that wireless radio signals he is using to create a 111km wireless bridge travel at the speed of light |
| 17:41.51 | _Sam-- | and that his latency is 3ms |
| 17:42.02 | _Sam-- | end to end |
| 17:42.12 | Zeeek | all radio signals are wireless |
| 17:42.30 | *** join/#asterisk trym (n=trym@194.63.254.6) |
| 17:42.36 | _Sam-- | sorry, i think think you knew what i meant....but i could not believe 3ms on a wireless network over 111kms |
| 17:42.42 | digime | looking for asterisk developer in san diego |
| 17:42.59 | _Sam-- | and sure enough, he was right...the radio waves go at close to the speed of light |
| 17:43.19 | Curus | We have several customers linked with FWA or Wimax through a provider |
| 17:43.20 | Zeeek | latency comes more from routers and switches than from distance of wire, fibre or radio |
| 17:43.29 | Curus | Some of them have latencies below 3ms |
| 17:43.57 | _Sam-- | thats the same thing he said, he has some that are below 1ms |
| 17:44.09 | fugitivo | just add a satellite and you'll add 500ms of latency |
| 17:44.32 | _Sam-- | must only be related to the shear distance of the sat. |
| 17:44.33 | Curus | Yes, our latencies include the service provider's network, I bet if you had the link yourself you could do quite a bit better |
| 17:44.51 | Curus | 36000km*2 just sucks. |
| 17:45.49 | _Sam-- | thanks for the info, always good to learn something new. |
| 17:46.02 | lazzarello | alright, asterisk royally loses. where is it getting it's umask from? I'm setting it but it's overwriting that somewhere after it's init script. |
| 17:46.20 | Curus | (Our Wimax links are faster than FWA, which I find odd) |
| 17:46.42 | Curus | lazzarello: That has been discussed on the mailing list |
| 17:46.51 | Curus | Asterisk sets its own umask, apparently |
| 17:47.07 | lazzarello | which google's cache is only partially saving. |
| 17:47.22 | Curus | gmail keeps everything |
| 17:47.25 | lazzarello | Curus, does this qualify as "sucks"? |
| 17:47.56 | Curus | I read all my mailing lists as news from gmail |
| 17:48.06 | Zeeek | http://threebit.net/mail-archive/asterisk-dev/msg00598.html |
| 17:48.11 | lazzarello | Curus, thanks for the 411. go gmail! |
| 17:48.31 | lazzarello | Zeeek, thanks |
| 17:48.46 | Zeeek | np |
| 17:49.36 | lazzarello | what's the logic of the application taking over the sysadmin's umask? seems a bit odd. |
| 17:50.39 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:50.50 | fugitivo | lazzarello: don't run asterisk as root |
| 17:51.02 | lazzarello | fugitivo, what makes you think I am? |
| 17:51.17 | fugitivo | "application taking over the sysadmin's umask" |
| 17:52.17 | lazzarello | fugitivo, it's running as asterisk, and it doesn't matter what the sysadmin sets if the application is setting the umask in the source code. |
| 17:52.41 | lazzarello | the only way to change it is to edit the source code or write a script external to the application. |
| 17:52.51 | fugitivo | some applications need to set their own umask |
| 17:52.51 | lazzarello | pretty suck if you ask me. |
| 17:52.58 | lazzarello | why is that? |
| 17:53.08 | lazzarello | seriously, I can't think of a reason. |
| 17:53.10 | fugitivo | i don't know, join #asterisk-dev |
| 17:53.44 | fugitivo | some apps like ftp servers do that also |
| 17:53.46 | Curus | Well asterisk hasn't always been developed according to the pure Unix spirit |
| 17:53.57 | lazzarello | Curus, it appears so :) |
| 17:54.27 | fugitivo | it runs as root by default... |
| 17:54.40 | lazzarello | It's annoying to collect a bunch of patches for production installations but whatever, that's what packages are for, right? |
| 17:55.20 | Hmmhesays | file o file, i have a simple question for you |
| 17:55.22 | *** join/#asterisk gbodemantv (n=gbodeman@mail.televerde.com) |
| 17:55.29 | gbodemantv | hey there |
| 17:55.34 | gbodemantv | big problem here |
| 17:55.37 | gbodemantv | as always |
| 17:56.09 | *** join/#asterisk Rhizome (n=rhizome@tor/session/x-594ea636a4ffe8b7) |
| 17:58.52 | *** join/#asterisk sack (n=sack@44.Red-83-32-164.dynamicIP.rima-tde.net) |
| 18:08.18 | *** join/#asterisk stoffell (n=stoffell@d51A58027.access.telenet.be) |
| 18:08.29 | gbodemantv | so who knows about Ramdisks? |
| 18:08.39 | gbodemantv | here is the situation |
| 18:08.40 | *** join/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br) |
| 18:08.42 | stoffell | goodevening |
| 18:08.46 | gbodemantv | we record all calls |
| 18:08.58 | gbodemantv | and when using a ramdisk the call quality is ok |
| 18:09.04 | gbodemantv | but without it they are bad |
| 18:09.24 | gbodemantv | but...the ramdisk wa scrashing theserver |
| 18:09.32 | gbodemantv | we had a 2.3GB ramdisk |
| 18:09.46 | gbodemantv | it was crashing after the ramdisk got fille dto 720MB |
| 18:09.54 | gbodemantv | even with asterisk turned off |
| 18:10.08 | gbodemantv | What is the best way to create a ram disk |
| 18:10.14 | fafnir | dynamically? |
| 18:10.20 | mut | by good ram? |
| 18:10.27 | gbodemantv | ?have 4 gb |
| 18:10.34 | mut | buy* |
| 18:10.40 | gbodemantv | fafnir, how is that? |
| 18:11.08 | fafnir | i have no idea, but maybe you could figure out how to demount or destroy a ramdisk and make one for every instance, instead of having one for all |
| 18:11.11 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 18:11.14 | fafnir | and having it grow and grow |
| 18:11.20 | fafnir | just make a new one for each instance |
| 18:11.31 | fafnir | and then ermove it when its usefulness is over |
| 18:11.51 | mroth_imm | gbodemantv: take a look at this |
| 18:11.55 | gbodemantv | we record all calls and then move them to another server for compression |
| 18:11.59 | fafnir | yeah |
| 18:12.05 | asteriskmonkey | andone ever had a sip->sip or iax connection where its too quite on the same machine |
| 18:12.10 | fafnir | im saying dynamically create them |
| 18:12.19 | fafnir | instead of making one that grows |
| 18:12.37 | gbodemantv | the concept is great but I have no idea how |
| 18:12.48 | fafnir | whats the enviroment? |
| 18:13.00 | mroth_imm | gbodemantv: http://64.233.179.104/search?q=cache:vlI2KQh_bj4J:lists.digium.com/pipermail/asterisk-users/2005-October/127919.html+%22512+simultaneous+calls+with%22&hl=en&gl=us&ct=clnk&cd=3 |
| 18:13.13 | mroth_imm | sorry for the cached version (list servers look to be down) |
| 18:13.27 | mroth_imm | right now we're recording 100 concurrent calls to a ram disk using that setup |
| 18:13.46 | mroth_imm | will experiment with moving to tmpfs in the future, but stuck with what we know works for now |
| 18:14.14 | gbodemantv | fedora core 3, ast 1.2.4 |
| 18:14.21 | gbodemantv | HP DL360 |
| 18:14.25 | gbodemantv | 4 GB ram |
| 18:14.27 | fafnir | #fedora :P |
| 18:14.33 | fafnir | but check out that setup |
| 18:14.56 | fafnir | <PROTECTED> |
| 18:14.56 | fafnir | <PROTECTED> |
| 18:14.56 | fafnir | <PROTECTED> |
| 18:14.56 | fafnir | <PROTECTED> |
| 18:14.56 | fafnir | <PROTECTED> |
| 18:15.01 | fafnir | they have a bit better stats |
| 18:15.07 | fafnir | <PROTECTED> |
| 18:15.25 | fafnir | mroth_imm: thats yours? |
| 18:15.52 | *** join/#asterisk xmark (n=brg@c-68-43-129-244.hsd1.mi.comcast.net) |
| 18:16.00 | mroth_imm | yeah...don't need that much juice...monitor taps out at 60 regardless of hardware if you go straight to disk from everything else i've read and seen |
| 18:16.22 | mroth_imm | looking for a replacement to nfs eventually too, but fighting fires in production gets my focus |
| 18:16.23 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 18:16.24 | *** join/#asterisk VirTERM (n=VirTERM@shiva.kanatek.com) |
| 18:16.47 | *** part/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 18:17.24 | mroth_imm | as far as the RAM disk goes though, we've never had a problem with it |
| 18:17.55 | xmark | I have a POTS line (PSTN) attached to a FXO port. How can I route calls that come in on that PSTN number to one particular extension? I guess this would be like a psuedo DID. |
| 18:17.59 | fafnir | you have a 16 gig ram disk though |
| 18:18.13 | [av]bani | \o> |
| 18:19.03 | fafnir | morning sunshine |
| 18:19.13 | _Paulo_ | xmark, in zapata.conf you should setup the context for that channel. |
| 18:19.33 | gbodemantv | the way we had it setup |
| 18:19.36 | asteriskmonkey | does rx and tx gains in the zapata take effect on sip/iax phones on the same box? if so how to avoid in circumstanes where gains are set in the high - |
| 18:19.38 | gbodemantv | we had a 2.3 GB ramdisk |
| 18:19.39 | Curus | Perhaps going 64-bit would be good |
| 18:19.40 | mroth_imm | true, but i've also filled it completely with no crashes |
| 18:19.54 | gbodemantv | which was locking the server when it got filled to 720MB |
| 18:19.55 | mroth_imm | so size shouldn't be pertinent to gbodemantv's problem |
| 18:19.56 | justinu | asteriskmonkey: no, the gains only affect calls that use the zap interfaces |
| 18:19.56 | Curus | Those Xeons sound new enough to be EM64T |
| 18:19.58 | [av]bani | 64-bit wouldnt help, almost always it hurts |
| 18:20.12 | [av]bani | the xeons dont do 64bit as well as amd... |
| 18:20.20 | Curus | [av]bani: With 20GB memory a 32-bit kernel is struggling |
| 18:20.27 | asteriskmonkey | justinu: any idea why a sip to sip call on the same * box would be really quite ? |
| 18:20.31 | Curus | You risk running out of low mem |
| 18:20.41 | mroth_imm | we are running 64 bit... |
| 18:20.42 | [av]bani | Curus: if its memory then yes 64bit will help, but 64bit for the sake of 64bit will not do anything |
| 18:20.55 | [av]bani | Curus: eg calculating pi will not be any faster in 64bit than 32bit |
| 18:21.12 | asteriskmonkey | clustering quad xeon boxes is the best solution to date :D |
| 18:21.18 | [av]bani | Curus: and in many cases its slower in 64bit |
| 18:21.24 | gbodemantv | mroth_imm: and idea why the server would crash when the ramdisk was only 1/3 full |
| 18:21.33 | Curus | mroth_imm: Ah, sorry for the assumption |
| 18:21.46 | mroth_imm | none at all...give me a moment and i'll show you a nice way to fill it easily for testing |
| 18:22.09 | Curus | [av]bani: I know about the disadvantages of 64 bit for user space, but for kernel space 64-bit is almost always a win these days |
| 18:22.41 | asteriskmonkey | you might aswell get a real ramdisk like a ramsan or something :) oober hardcore |
| 18:22.48 | mroth_imm | gbodemantv: it's actully in the earlier link |
| 18:23.06 | justinu | asteriskmonkey: what are the endpoints of the calls? |
| 18:23.11 | justinu | what devices |
| 18:23.11 | mroth_imm | you can use dd to create arbitrarily sized files on the ram disk for testing |
| 18:23.15 | mroth_imm | 2 GB - time dd if=/dev/zero of=/digrec-nfs/testfile bs=16k count=131072 |
| 18:23.17 | asteriskmonkey | justinu: 2 sip phones |
| 18:23.28 | xmark | gbod, any clue on the config options? |
| 18:23.42 | mroth_imm | once i completely filled ours, nothing nasty happened at all, just reported disk full |
| 18:23.56 | brad_mssw | [av]bani: well, for most processors, that's true ... but with x86-64, you have more GPRs that may be used, but only in 64bit mode, so all [properly designed] apps _should_ be faster on x86-64 running in 64bit mode |
| 18:24.01 | justinu | asteriskmonkey: it would have to be a problem with one of the devices. if the codec is the same on each device, asterisk isn't mucking with the RTP stream at all, just proxying it |
| 18:24.17 | mroth_imm | and currently we are using over a gig of ours for live calls, so we have no 720 meg limit |
| 18:24.30 | [av]bani | brad_mssw: it's overall a loss because the instruction width is larger, it bashes memory harder, eats more cache for the same amount of processing |
| 18:24.31 | asteriskmonkey | justinu: thanks :) must be one of the sip phones then.. |
| 18:24.36 | Curus | Anyway, disk I/O shouldn't interrupt asterisk these days |
| 18:24.36 | justinu | np |
| 18:25.16 | [av]bani | brad_mssw: it is not only true for most processors, it is also true of x86-64. just about the only clear cut case where x86-64 is a win is for memory copies |
| 18:25.17 | brad_mssw | [av]bani: I don't know about the 'instruction width' being larger, each instruction is still 1 byte afaik, granted, each reference to a memory address is twice the size |
| 18:25.29 | GerbilWrk | will a 64bit motherboard and processor load the 32bit version of slackware without issue? |
| 18:25.51 | Curus | Use of a 64-bit register costs an 8-bit prefix I think, but that's still cheap |
| 18:25.51 | Beirdo | umm |
| 18:26.02 | Beirdo | instructions aren't 1 byte in i386 |
| 18:26.07 | [av]bani | brad_mssw: there's more registers... look at the amd processor manual |
| 18:26.10 | Beirdo | one word maybe |
| 18:26.11 | mroth_imm | Curus: you are right, ideally i/o shouldn't be the bottleneck |
| 18:26.17 | brad_mssw | (granted the data passed to the instruction may be larger in 64bit mode, but most apps should use precise data types) |
| 18:26.30 | brad_mssw | [av]bani: right, like I said, there's more GPRs in 64bit mode |
| 18:26.31 | mroth_imm | but if you can get past 60 recordings with monitor straight to disk, please share, it'd be a valuable contribution |
| 18:27.02 | [av]bani | brad_mssw: and uh, might want to actually benchmark... i have. in 95% of cases x86-64 is slower. this is well known. |
| 18:27.28 | gbodemantv | mroth_imm: if we have 4GB of RAM what would you a size for ramdisk |
| 18:27.30 | Curus | [av]bani: Which compiler? gcc sucks on 32-bit x86 |
| 18:27.32 | [av]bani | brad_mssw: it is usually only a few % but it is slower |
| 18:28.05 | brad_mssw | [av]bani: from my benchmarks, x86-64 was always faster, especially databases, etc ... though if you test JAVA apps, you've got to realize that only the JAVA Server model was ported to 64bit mode, the Client model was not, so it's not a apples to apples comparison |
| 18:28.11 | heroine | nice evening ppl |
| 18:28.16 | mroth_imm | gbodemantv: it depends on how many concurrent calls you'll be recording, their codec and duration |
| 18:28.37 | mroth_imm | currently we're recording 240 legs of pcm audio and we're using a gig of our space |
| 18:28.44 | brad_mssw | [av]bani: /me used to be the gentoo amd64 lead 2 years ago ... most of my benchmarks are from that timeframe |
| 18:28.47 | [av]bani | brad_mssw: databases use mostly memory moves. so it is not suprising it would be faster. but calc heavy stuff is generally slower. |
| 18:28.48 | mroth_imm | they get transferred IMMEDIATELY after the call |
| 18:29.01 | Curus | 60 recordings at alaw, 8kB/s, that's only 480kB/s |
| 18:29.14 | Curus | If asterisk can't handle that, something really needs fixing |
| 18:29.28 | mroth_imm | we're also using only 527 megs of system memory with 240 dynamic agents loaded |
| 18:29.30 | Curus | Ok 960kB/s for duplex |
| 18:29.32 | [av]bani | brad_mssw: look at xmame... calc heavy, and it is about 50/50 faster or slower in various cases |
| 18:29.53 | mroth_imm | i'd say you can split 50/50 (2 gigs system, 2 gigs ram disk) as a start...analyze from there and tweak to your pleasure |
| 18:29.58 | [av]bani | brad_mssw: or povray, where its slower but the advantage you get is 64bit accuracy |
| 18:30.46 | [av]bani | Curus: gcc4 compares quite favorably with msvc for 32bit |
| 18:30.49 | brad_mssw | [av]bani: err, povray was faster last time I checked ... never ran xmame though |
| 18:31.04 | Curus | [av]bani: icc beats the pants off gcc still, unfortunately. |
| 18:31.06 | mroth_imm | Curus: i understand what you are saying, but the reality is that 60 makes things get nasty...go to ram disk and the issue goes away...maybe when all of my fires are out i'll know exactly why but for now i have to run with what works |
| 18:31.41 | Curus | mroth_imm: It would be nice to see vmstat 1 while the recordings are going on |
| 18:32.45 | mroth_imm | a representative sample: 0 0 208 45752 1883624 18106476 0 0 0 948 6571 28779 2 15 83 0 |
| 18:33.20 | mroth_imm | when we did our initial tests (prior to ram disk) at around 60 calls the numbers of blocks written to disk per second were pretty high |
| 18:34.05 | mroth_imm | don't have the actual data at hand but they were high enough to make us go "hmmmmmm" |
| 18:34.46 | [av]bani | Curus: gcc4 is quite favorable with icc too these days |
| 18:34.54 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 18:35.41 | [av]bani | Curus: gcc still "sucks" on ppc and sparc, but i dont know how many people care about that anymore :)) |
| 18:36.21 | brad_mssw | wow, didn't realize gcc 4.1 was released yesterday |
| 18:37.40 | Curus | mroth_imm: Is that with ramdisk or real disk? |
| 18:38.06 | Curus | cs is high, the rest looks excellent |
| 18:38.44 | Curus | (Admittedly I've never recorded calls with asterisk, only with tcpdump, so I haven't seen the problem) |
| 18:38.59 | mroth_imm | with ram disk |
| 18:39.26 | mroth_imm | honestly, there is almost no physical disk activity on that box |
| 18:39.27 | Curus | Is the bo rate stable around 948? |
| 18:39.44 | Curus | I can't remember if linux counts ram disk I/O |
| 18:39.56 | [av]bani | all block i/o is counted |
| 18:40.25 | Curus | Yes but I don't think a ram disk is actually a ram disk, I think it's a filesystem which happens to not have backing store |
| 18:40.50 | [av]bani | one way to find out |
| 18:40.57 | Tagor | Has someone here fax with SIP working? |
| 18:41.15 | mroth_imm | no, bo fluctuates pretty wildly |
| 18:41.24 | Curus | Tagor: When the wind is easterly and the moon is waxing |
| 18:41.34 | mroth_imm | 23201 one second, 52 the next |
| 18:41.34 | [av]bani | yep it counts |
| 18:42.07 | [av]bani | any blocks which go through the fs layer is counted, regardless, period, end of story |
| 18:42.09 | Tagor | Curus >> :P, that's a 'no'? |
| 18:42.19 | [av]bani | backing store or not |
| 18:42.35 | mroth_imm | kernel sees the ramdisk as an ext2...as i said, tmpfs is an option in the future (tweak time) |
| 18:42.44 | rayvd | for your ramdisk are you using tmpfs or the actual ramdisk that you can set up via a kernel boot option? |
| 18:42.52 | Curus | Tagor: Sometimes it works. Pretty often it works, actually |
| 18:42.53 | mroth_imm | kernel boot option ramdisk |
| 18:43.04 | [av]bani | tmpfs is better |
| 18:43.11 | Tagor | Curus >> Can you tell me what program you use? rxfax? |
| 18:43.23 | Curus | Tagor: Physical faxes |
| 18:44.24 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:44.30 | mroth_imm | [av]bani: i imagine you are right...gotta benchmark a little and will probably move over eventually...worked at the time all hell broke loose and we stuck with it |
| 18:44.37 | Curus | Oh, hmm, I wonder if the ramdisk option picks memory below 4G |
| 18:44.44 | mroth_imm | all hell broke loose = our scalability tests failed miserably |
| 18:45.09 | Tagor | Curus >> So you use a fax to handle the incomming fax messages? |
| 18:45.24 | [av]bani | mroth_imm: i dont think its faster, tmpfs just doesnt take a fixed block of ram.. it can grow and shrink |
| 18:45.43 | Curus | Tagor: Yes, that and an ATA. It sucks, but most of the time it works. Except for those billion-dollar order faxes, they never get through. |
| 18:45.45 | *** join/#asterisk kukko (n=kukko@host213-77.pool82104.interbusiness.it) |
| 18:45.47 | Tagor | Curus >> I actually mean if someone has a incomming SIP line from a VOIP provider as fax line |
| 18:45.52 | mroth_imm | okay...we are so overconfigured on ram i don't know it'll make a difference then : ) |
| 18:45.57 | kukko | hello |
| 18:46.06 | kukko | there is somebody that would like to help me to configure asterisk? :-) |
| 18:46.14 | kukko | please |
| 18:46.37 | [TK]D-Fender | kukko : My rates are very affordable ;) |
| 18:46.48 | Curus | mroth_imm: If tmpfs picks memory above 4G and ramdisk picks memory below 4G, it could make a difference. |
| 18:47.03 | kukko | :-) |
| 18:47.08 | cpm | mine aren't, AND I won't get it right, So, I am a first class consultant! |
| 18:47.11 | Curus | Linux's handling of bounce buffer allocations for 32-bit hardware is not always stellar |
| 18:47.41 | Curus | (And not all Linux drivers allow the hardware to do 64-bit transfers, even if they are capable) |
| 18:47.49 | mroth_imm | Curus: so putting the ram disk above 4 gigs satisfies that, putting it below 4 gigs forces bounce buffering all over the place |
| 18:48.17 | mroth_imm | or do i have that backwards : ) |
| 18:48.24 | Curus | mroth_imm: Yes, except on perfect servers with all hardware and drivers allowing 64-bit transfers. If you find one I'd like to hear |
| 18:48.26 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 18:48.53 | mroth_imm | ram disk above 4 gigs-good...below 4 gigs-bad ... got it! |
| 18:49.06 | mroth_imm | i remember when i thought computers would make my life easy ; ) |
| 18:49.16 | cpm | No way, what liar told you that! |
| 18:49.23 | Curus | Obviously you had never tried PC's then |
| 18:49.55 | mroth_imm | it musta been some politician or salesman, or possibly satan himself |
| 18:49.57 | Curus | The problem is that if you reserve all the <4G memory for ram disk, Linux has no memory to allocate bounce buffers in, and dies |
| 18:49.58 | *** part/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 18:50.07 | mroth_imm | hard to tell the difference |
| 18:50.15 | Curus | But I still don't know whether ramdisk picks high mem |
| 18:50.40 | mroth_imm | is there an obvious way to tell by observing a running system? |
| 18:51.35 | Curus | Probably, but I don't know of it |
| 18:51.38 | mroth_imm | from /proc/meminfo |
| 18:51.45 | mroth_imm | LowTotal: 20573600 kB |
| 18:51.55 | mroth_imm | LowFree: 40264 kB |
| 18:52.07 | mroth_imm | means nothing to me, to be completely honest : ) |
| 18:52.27 | Curus | Does it mention a HighFree as well? |
| 18:52.35 | Curus | Nope, it counts it all as low |
| 18:52.40 | mroth_imm | yeah, both of the High #s are 0KB |
| 18:53.18 | Curus | Right, because memory is (almost) uniform on a 64-bit machine. Only matters for 32-bit hardware, of which there is much |
| 18:53.23 | mroth_imm | but i understand the DMA bounce buffering issue and i don't think it's at all reflected in those numbers |
| 18:53.39 | mroth_imm | as you said, the 4GB limit is the domain of 32bit cards |
| 18:53.44 | kristinG | does anyone know of a irc channel for lucent tnt users? |
| 18:54.11 | mroth_imm | and all of our current cards are 64 bit so i think we're okay |
| 18:54.39 | Curus | Excellent then |
| 18:55.42 | mroth_imm | thanks Curus...i've gotten a ton of helpful info from this channel today : ) |
| 18:55.54 | mroth_imm | maybe i'll actually get some work done soon : ) |
| 18:58.08 | Curus | Thanks too, I'll certainly keep this in mind if we start really recording calls |
| 18:58.11 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:58.27 | [av]bani | mroth_imm: bounce buffering shouldnt be needed if you have iommu enabled |
| 18:58.38 | Curus | [av]bani: Good point |
| 18:58.44 | [av]bani | mroth_imm: thats what its _there_ for :)) |
| 18:59.06 | mroth_imm | [av]bani: filed for our next foray into 32 bit hardware |
| 18:59.10 | xmark | How can I configure a channel (In my case FXO) for PSTN to always ring extenstion 1000? |
| 19:00.23 | _Sam-- | [av]bani : does the 7960 support SIP out of the box with no extra firmware? |
| 19:00.37 | [av]bani | _Sam--: no, they always come sccp |
| 19:00.45 | [av]bani | afaik _all_ cisco phones come sccp out of the box |
| 19:00.55 | [av]bani | you have to buy a sip image |
| 19:00.57 | _Sam-- | i had a meeting today with an ISP that does alot of hosted PBX stuff for cisco phones.... |
| 19:01.07 | _Sam-- | and they told me it came out of the box ready for SIP...i was like 'huh'? |
| 19:01.20 | _Sam-- | i guess they lie |
| 19:01.25 | [av]bani | _Sam--: it is "ready for sip" as in, the hardware supports it and a sip image is available |
| 19:01.31 | mroth_imm | Curus: if you get past 60 (say ~80) concurrent recordings right to disk with no call quality issues, share it, it would be a significant achievement |
| 19:01.39 | cpm | or, if they get a *bunch* of 'em, they can probably gettem boxed however they want |
| 19:02.24 | [av]bani | _Sam--: tis probably easier to stick with sccp, since cisco "punishes" sip users with poorer audio quality |
| 19:02.29 | Chotaire | dudes, I seem stuck with something... Can anyone assist with some probably very easy problem? On Keypress, I want do Dial a phonenumber and after connect, the callee shall be transferred to an extensions and me (the caller) shall continue in context, NOT be connected to the callee. |
| 19:03.01 | Chotaire | Dial M(x[^arg]) comes up with something completely unexpected (record your name) |
| 19:03.19 | Chotaire | and what is "x" by the way? the number of the macro? |
| 19:03.32 | SplasPood | Anyone here have any experience with either hylafax or iaxmodem? |
| 19:03.49 | *** join/#asterisk SPoon_TSX (n=Kit@h24-83-96-211.sbm.shawcable.net) |
| 19:04.32 | SPoon_TSX | hello experts.... I need help.... My asterisk zap channel have no echo BUT static..... Anyone have the same problem? |
| 19:05.13 | Fedoracore6 | hai all |
| 19:05.33 | Fedoracore6 | i already change my username data base and password |
| 19:05.39 | Fedoracore6 | but still have same problem |
| 19:05.41 | Fedoracore6 | http://pastebin.com/578798 |
| 19:07.18 | SibRw0rk | Fedoracore6: you have mysql running? |
| 19:07.41 | SibRw0rk | do you have the proper username and password to write to a dbase setup in your cdr_mysql? |
| 19:08.08 | Fedoracore6 | yes running |
| 19:08.09 | Fedoracore6 | [root@localhost ~]# service mysqld status |
| 19:08.09 | Fedoracore6 | mysqld (pid 2139) is running... |
| 19:08.18 | SibRw0rk | can you query the proper dbase |
| 19:08.33 | x86 | brad_mssw: who uses java? |
| 19:08.50 | SibRw0rk | Fedoracore6: i get can't connect to database on local host on my machine as wel - but it still works |
| 19:09.35 | Skumling | SplasPood: nope... installed hylafax and iaxmodem but thought it was just too complex, and dropped it again... are now using rx_fax |
| 19:09.35 | Fedoracore6 | oic |
| 19:10.07 | SplasPood | Skumling: does it work well? I'm totally SIP here, no zaptel |
| 19:10.32 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
| 19:10.44 | Skumling | SplasPood: it works well with zaptel ;) |
| 19:10.48 | Fedoracore6 | http://pastebin.com/578828 |
| 19:10.54 | Fedoracore6 | this is my cdr_mysql |
| 19:10.55 | SplasPood | Skum: :P |
| 19:11.04 | Skumling | SplasPood: but I don't see why you shouldn't be able to use it with SIP only? |
| 19:11.06 | Fedoracore6 | oic |
| 19:11.53 | Skumling | SplasPood: maybe you'll need to compile zaptel and use ztdummy, but that shouldn't hurt either? |
| 19:11.55 | Fedoracore6 | SibRw0rk: when i run the asterisk .... i try using the softphone (x-lite) put the password |
| 19:12.04 | Fedoracore6 | but aserisk say error |
| 19:12.29 | SplasPood | Skumling: oh, I didn't know that it'd work /wo proper hardware |
| 19:13.03 | SPoon_TSX | Hi there, May i know if anyone here also experience some static problem on their Asterisk Zap Channel? |
| 19:13.03 | Skumling | SplasPood: oh yes... ztdummy is a dummy device for use when you need zaptel without having any zaptel-compliant hw |
| 19:13.12 | *** join/#asterisk dArF_AST (n=dArF_AST@63.144.116.5) |
| 19:13.13 | SPoon_TSX | Since I got no echo but Static issue. |
| 19:13.17 | dArF_AST | hello |
| 19:13.54 | SplasPood | Skum: yea I know, I just thought the fax piece required more than ztdummy would offer |
| 19:14.19 | *** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net) |
| 19:14.21 | kippi | hi |
| 19:14.38 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.76.208) |
| 19:14.57 | Skumling | SplasPood: not AFAIK... had some problems with it at first, but turned out that 1) recompile got a few things cleared up and 2) it's really really good idea to load the paper in the sending fax... damnit |
| 19:15.33 | Skumling | SplasPood: wasted 4-5 hours being angry on rx_fax without reason... |
| 19:16.02 | SplasPood | heh, I've got iaxmodem/hylafax working great, but at 14400 I seem to be having issues, so I wanna force it to 9600... just don't seem to know how to do that :) |
| 19:16.46 | Skumling | SplasPood: okay cool... people say that hylafax is great... but for receiving approx 5 faxes/month it just seemed overkill to me |
| 19:17.03 | Chotaire | sorry to say but it seems option "M" in Dial command is completely fucked. |
| 19:17.32 | *** join/#asterisk ToTo (n=ToTo@host43-130.pool874.interbusiness.it) |
| 19:18.15 | SplasPood | Skumling: whole reason I'm doing this is cause we use efax now and they... suck. |
| 19:18.29 | Chotaire | option M doesn't seem what is documented. |
| 19:18.30 | kippi | How can i make sure that my te110p card is up |
| 19:18.45 | Chotaire | it actually asks for recording a name instead of executing a macro |
| 19:18.47 | Hmmhesays | ahh ser diaplan just enough to give me a freaking headache |
| 19:20.47 | trelane_ | hrm I have a snom360 during testing I seem to have lost dial tone and all but the default ring. The phone works fine otherwise, but a factory reset doesn't seem to solve this issue |
| 19:21.21 | Chotaire | (and no, I have no demo/sample macros defined that might hit this behaviour) |
| 19:21.51 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 19:22.06 | Chotaire | still investigating... |
| 19:22.35 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 19:22.51 | *** join/#asterisk jldb (n=2070D58E@adslfixo-b3-123-7.telepac.pt) |
| 19:23.06 | jldb | hello, anyone from digium? |
| 19:23.11 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 19:23.57 | mogorman | yes jldb |
| 19:24.30 | jldb | i'm initiating my study about asterisk pbx |
| 19:24.35 | jldb | i'm a student |
| 19:24.56 | jldb | asterisk is easy to install in linux? |
| 19:25.06 | mogorman | sure |
| 19:25.17 | jldb | and about the hardware |
| 19:25.19 | mogorman | asterisk is about as difficult to understand as apache |
| 19:25.20 | lazzarello | yes. apt-get install asterisk in Debian. |
| 19:25.30 | mogorman | ewww lazzarello |
| 19:25.33 | mogorman | build from source |
| 19:25.36 | mogorman | you will live happier |
| 19:25.40 | jldb | i will install it in fedora core 4 |
| 19:25.45 | lazzarello | mogorman, that's the easiest you'll find. |
| 19:25.50 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 19:25.58 | lazzarello | with no judgments for functionality |
| 19:26.00 | mogorman | okay taht should be fine |
| 19:26.21 | jldb | i don't understand the hardware |
| 19:26.32 | lazzarello | my mantra for noobs is that compiling is a lot more difficult than installing a binary. |
| 19:26.33 | jldb | i have at home a isdn line |
| 19:26.57 | jldb | can i join asterisk to my line |
| 19:27.18 | *** join/#asterisk Fedoracore6 (n=FC$@60.50.141.168) |
| 19:27.29 | jldb | ? |
| 19:27.58 | Fedoracore6 | hai all |
| 19:28.02 | [TK]D-Fender | I never had real pkg mgmt and never had any problems compiling everything from source and I suck at linux :) |
| 19:28.30 | [TK]D-Fender | jldb : Yes, you just need an appropriate BRI card like the AVM Fritz |
| 19:28.37 | lazzarello | [TK]D-Fender, good for you. |
| 19:28.54 | *** join/#asterisk lo_tech (n=lo_tech@209.36.181.24) |
| 19:29.50 | justinu | i got my 4 X100Ps working, woot |
| 19:29.58 | justinu | sounds fine too |
| 19:30.49 | Fedoracore6 | ok i try put in the password for x-lite phone pass:810325045093 |
| 19:31.08 | Fedoracore6 | and error out like that |
| 19:31.08 | Fedoracore6 | http://pastebin.com/578851 |
| 19:31.12 | Fedoracore6 | waht error mean ... |
| 19:32.18 | *** join/#asterisk kukko (n=kukko@host213-77.pool82104.interbusiness.it) |
| 19:32.30 | Fedoracore6 | any suggestion |
| 19:32.31 | SPoon_TSX | Anyone from Toronto here? |
| 19:33.18 | justinu | probly |
| 19:33.32 | jldb | i need to ports one for my switch that have several ip phones and another for my external isdn line, am i right? |
| 19:33.40 | fugitivo | justinu: where did you get the cards? |
| 19:33.51 | [TK]D-Fender | jldb : Something like that, yes |
| 19:34.13 | justinu | someone gave them to me |
| 19:34.19 | *** join/#asterisk crich1999 (n=crich@port-212-202-198-154.dynamic.qsc.de) |
| 19:34.22 | [av]bani | \o/ |
| 19:34.32 | jldb | i need a pci card that have that ports, witch one? |
| 19:34.33 | [av]bani | justinu: four x100p's in a single pc? o_O |
| 19:34.50 | justinu | yep |
| 19:35.04 | [av]bani | O.o |
| 19:35.11 | justinu | getting the interrupt sharing worked out was challenging, but it's done, and working. |
| 19:35.19 | [av]bani | heh.. i can imagine |
| 19:35.27 | [av]bani | its like playing rubiks cube sometimes |
| 19:35.31 | jldb | anyone from Portugal? :( |
| 19:35.33 | justinu | turns out you need an APIC motherboard, and you need to run the SMP kernel |
| 19:35.41 | justinu | so once I got those things, all was well. |
| 19:35.48 | [av]bani | apic mobo, and pci irq routing |
| 19:36.46 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 19:36.47 | fu3 | hello everyone |
| 19:37.17 | fu3 | does anyone here run FreeBSD as their OS for Asterisk? I want to badly, but I heard something about PRI not working in FreeBSD. Can someone please elaborate, or at least, confirm or deny this? |
| 19:37.17 | [av]bani | PakiPenguin: you there? |
| 19:37.18 | SPoon_TSX | Hello everyone, May I know anyone may have the Bell Canada Milliwatt phone number please? |
| 19:37.32 | jldb | where can i get a cheap card to teste asterisk? |
| 19:38.18 | _Paulo_ | jldb, http://www.digium.com |
| 19:38.18 | jldb | cheap??? |
| 19:38.18 | _Paulo_ | U$ 200,00 I think |
| 19:38.25 | _Paulo_ | its not cheap enough? |
| 19:38.29 | [av]bani | _Paulo_: we use . in this channel :)) |
| 19:39.00 | *** join/#asterisk VirTERM (n=VirTERM@shiva.kanatek.com) |
| 19:39.04 | _Paulo_ | oh, yes. decimal separator in english is "." |
| 19:39.07 | [TK]D-Fender | jldb : You need a BRI card... that would not be Digium gear, but rather the AVM Fritz card I mentioned earlier. |
| 19:39.09 | kippi | when i run modprobe wcte11xp i get this error line 0: Unable to open master device '/dev/zap/ctl' |
| 19:39.17 | jldb | but i need a port for my ip phones and another for my isdn line, it's 200 € |
| 19:39.19 | kippi | anyideas how to sort this? |
| 19:39.20 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
| 19:39.59 | [av]bani | _Paulo_: :)) |
| 19:41.09 | _Paulo_ | jldb, you need 1 fxs and 1 fxo? |
| 19:41.14 | jldb | what is zaptel? |
| 19:41.21 | jldb | yes |
| 19:41.54 | jldb | fxs is for ethernet? |
| 19:42.04 | digime | anyone know a good place to buy ip phones? esp. polycom 501 |
| 19:42.09 | AndyCap | jldb: umm, get a normal ethernet card for ip phones. |
| 19:42.16 | lo_tech | kippi: new install, kernel 2.6? |
| 19:42.33 | [av]bani | ~fxs |
| 19:42.35 | jbot | methinks fxs is foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 19:42.39 | [av]bani | ~fxo |
| 19:42.40 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 19:42.52 | [av]bani | thanks, jbot. |
| 19:43.05 | [av]bani | ~zaptel |
| 19:43.06 | jbot | it has been said that zaptel is zapata telephony interface. A low level interface designed to abstract hardware access to a variety of devices for BRI, PRI or analogue access. |
| 19:43.14 | jldb | but i need also to join my external digital(isdn) line its a fxs/fxo? |
| 19:43.19 | [av]bani | no |
| 19:43.29 | [av]bani | thats just ISDN |
| 19:43.35 | [av]bani | fxs/fxo only apply to analogue phones |
| 19:43.41 | [av]bani | for ISDN its CPE/CO |
| 19:43.57 | jldb | digium have a card for that? |
| 19:44.08 | [av]bani | for PRI yes, for BRI no |
| 19:44.19 | lo_tech | BRI is for mortals |
| 19:44.34 | [av]bani | lo_tech: then DS3 is for immortals? :)) |
| 19:44.47 | *** part/#asterisk Pupeno (n=Pupeno@19-161-126-200.fibertel.com.ar) |
| 19:44.52 | lo_tech | s/mortals/mere mortals/ |
| 19:45.02 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 19:45.10 | lo_tech | OC3! |
| 19:45.11 | [av]bani | Interface ATM1/0 is up |
| 19:45.11 | [av]bani | Hardware is ENHANCED ATM PA - DS3 (45000Kbps) |
| 19:45.13 | [av]bani | :)) |
| 19:45.18 | jldb | ok i'm confused |
| 19:45.20 | *** join/#asterisk GoRK (n=GoRK@amarillo.energynet.com) |
| 19:45.36 | AndyCap | Heh, this was old: http://www.intel.com/network/csp/products/7007web.htm |
| 19:46.00 | [av]bani | jldb: http://www.digium.com/index.php?menu=product_category&category=hardware |
| 19:46.15 | jldb | i have a isdn card is that enougth? |
| 19:46.31 | GoRK | Does anyone know if it is possible with polycom phones -- whenever I dial an extension that my phone will display the callerID of the person i am *calling* (if asterisk knows it).. seems like it could be set in the response to the INVITE |
| 19:46.36 | lo_tech | jldb: hint, google/froogle/ebay search 'asterisk bri' |
| 19:46.44 | gbodemantv | mroth: why when i create a 2 ramdisk does it say 3086 is in use |
| 19:46.54 | gbodemantv | when no one is using it |
| 19:46.58 | AndyCap | [av]bani: besides, for isdn TE/NT is the equivalent of FXS/O iirc. |
| 19:47.02 | justinu | gork: polycom supports it. asterisk doesn't. :( |
| 19:47.21 | jldb | i need to go now but i am pleased to meet you , see 'u' around |
| 19:47.22 | [TK]D-Fender | GoRK : nope, and I've never heard of an IP phone that will... |
| 19:47.23 | *** part/#asterisk jldb (n=2070D58E@adslfixo-b3-123-7.telepac.pt) |
| 19:47.40 | GoRK | justinu: I figured that much; do you know how the protocol works by chance? |
| 19:48.03 | Zodiacal | anyone know why i can't load my sip firmware on my cisco 7960 phone? it says File not found P0S3-07-4-00. but that file is there. i have done this a few months ago but i must have forgoten a step or somthing. i have the OS79XX.TXT showing that file name too. and the SIP<mac>.cnf file configured with that file name too. any suggestions? |
| 19:48.05 | justinu | yes, i do |
| 19:48.25 | GoRK | [TK]D-Fender: i had a sprint-logo'd IP phone in my hotel room last weekend that did it |
| 19:48.31 | lo_tech | gbodemantv: the filesystem (depending on which you have) uses a bit to store inode/dir structure, even if it's empty |
| 19:48.34 | [TK]D-Fender | Zodiacal : Perhaps an attribute problem... |
| 19:48.56 | Zodiacal | tkd-fender nope chmod 777 the whole folder |
| 19:49.04 | GoRK | justinu: is it proprietary or would it be something that I could see? i'd maybe like to try making an asterisk patch for it... |
| 19:49.16 | AndyCap | GoRK: and you didn't open it to figure out the oem? :P |
| 19:49.19 | justinu | no, it's in the SIP RFC that describes "remote-party-id" |
| 19:49.27 | Skumling | Zodiacal: I had _lots_ of problems getting the SIP image on a 9760 |
| 19:49.30 | Skumling | 7960 even |
| 19:49.32 | justinu | the RPID tag should be in the 180 Ringing, or 183 progress |
| 19:49.37 | Zodiacal | skumling what was yer solution? |
| 19:49.38 | justinu | that shows you who you've dialed |
| 19:50.01 | justinu | i wrote a somewhat funky patch to asterisk that implements it |
| 19:50.03 | justinu | but it's a total hack |
| 19:50.13 | justinu | because asterisk doesn't support the concept of "Called party ID" |
| 19:50.19 | Skumling | Zodiacal: if you're using the "template"-files from Cisco.com, I found out that they were fscked up... a big mess of version-numbers, SIP/SCCP filename references etc. |
| 19:50.21 | justinu | err "Connected Party ID" |
| 19:50.24 | GoRK | justinu: could you send it to me or is it posted somewhere? |
| 19:50.35 | justinu | i gave it to stkn on #openpbx |
| 19:50.43 | justinu | you might ask him about it |
| 19:50.54 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 19:51.10 | GoRK | andycap: no i dont think they would have liked me hacking it apart... it did end to end encryption and everything.. had about 30 line keys too |
| 19:51.14 | GoRK | justinu: thanks |
| 19:51.18 | Zodiacal | skumling nope the versions are pretty clean in my files.. |
| 19:51.32 | Skumling | Zodiacal: humm okay... |
| 19:51.34 | AndyCap | GoRK: hehe, got to take some pictures next time. :P |
| 19:51.38 | justinu | i would have been more interested in getting that patch into asterisk if anyone gave a damn |
| 19:51.41 | warthawg | iz hard being a noobie. what do it mean, i dial from one sip extension to another, but asterisk, she say 'no route to destination' ? |
| 19:52.06 | GoRK | andycap: inn at the ballpark in houston, tx |
| 19:52.16 | Zodiacal | skumling i mean, its only my version listed... |
| 19:52.18 | Skumling | Zodiacal: have you tried checking the logs from your tftp server? |
| 19:52.19 | Zodiacal | i don't see what could be wrong |
| 19:52.25 | Zodiacal | nopers |
| 19:52.27 | Zodiacal | ill check |
| 19:52.42 | Skumling | Zodiacal: to check if the 7960 requests some non-existent files or something like that |
| 19:53.07 | justinu | warthawg: it means she can't figure out how to ring that phone, because she doesn't know the IP of the phone |
| 19:53.12 | GoRK | justinu: well maybe I can figure out a way that it makes more sense to implement it; maybe an extension to the hint context is what i was thinking.. ie if we have a called id set in the hint we send it when necessary |
| 19:53.13 | warthawg | i spent all morning fighting atftpd |
| 19:53.24 | warthawg | justinu, thanks |
| 19:53.26 | justinu | gork: if you're serious, i'll lend you technical assistances |
| 19:53.35 | justinu | gork: pm me for an email address |
| 19:54.01 | justinu | it's a kick ass feature |
| 19:54.02 | GoRK | justinu: well, lets see if i can look at your initial patch so i can see how the sip packets should go and then i can see about working it into the code better |
| 19:54.11 | justinu | and asterisk not having it is pretty lame |
| 19:54.11 | warthawg | justinu, i can use each of the extensions to dial out |
| 19:54.26 | warthawg | doesnt that mean asterisk, she know the ip? |
| 19:54.40 | justinu | warthawg: no, that means your phones know the IP of asterisk |
| 19:54.45 | warthawg | ah |
| 19:54.56 | digime | 10am 3/1 wed. Meet Ops Mgr. Cindy Howard at Wardell jobsite where vpn system then go to corp hdqts. ask for Ken Underhill, Malcolm Rosenberger is IT mgr.jobsite phone number is 858-622-9131. |
| 19:54.56 | digime | Jobsite: |
| 19:54.57 | digime | 2681 Idlehour Lane |
| 19:54.57 | digime | La Jolla, CA 92037 |
| 19:54.57 | digime | Corp. office: |
| 19:54.58 | digime | 646 Valley Ave., Ste C |
| 19:55.00 | digime | Solana Beach, CA 92075 |
| 19:55.11 | [av]bani | ?? |
| 19:55.15 | digime | sorry |
| 19:55.19 | warthawg | thanks, gigime, but i am not looking for a job |
| 19:55.22 | digime | haha |
| 19:55.23 | Chotaire | exten => 5,2,Dial(SIP/soft-chotaire,10,gD(www0)M(tp-locked1)G(tp-pool1-control^5^3)) |
| 19:55.28 | digime | hey does anyone here live in sa diego |
| 19:55.41 | [av]bani | for h0t ph0n3 s3xx0r call 858-622-9131 and ask for Cindy Howard |
| 19:55.42 | Chotaire | how could I make "www0" dialed? it won't work neither with M or G option. |
| 19:56.55 | Chotaire | is it not possible to send dtmf dial with M or G option? |
| 19:57.01 | websae | does anyone here deal with E911 services at tall? |
| 19:57.03 | warthawg | justinu, where would a fella tell asterisk the ip address? or is that done when the phone registers? |
| 19:57.22 | Chotaire | please someone help, that's the only problem left... |
| 19:58.14 | justinu | warthawg: that's right, registration... |
| 19:58.24 | warthawg | thanks again, justinu |
| 19:58.38 | *** join/#asterisk gaspiz (n=gaspiz@86.34.6.164) |
| 19:58.38 | justinu | np |
| 19:59.56 | gaspiz | hi, i am tryiing to dial an asterisk servere from another asterisk server and keep getting Failed to authenticate on INVITE |
| 20:00.17 | gaspiz | how should I set up the 2 asterisk so they accept eachothers calls? |
| 20:00.27 | *** part/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 20:01.27 | *** join/#asterisk Eggplant (i=No@40.193.217.216.cascadeaccess.com) |
| 20:01.32 | *** join/#asterisk rene- (n=rene-@201.127.101.127) |
| 20:01.42 | *** join/#asterisk jbenson (n=jbenson@87.194.2.120) |
| 20:03.09 | rene- | hello, i am in FC4, i have installed glibc kernheaders and i am running kernel 686 SMP, i know i need kernel-devel 686, i do not know wheather kernel-devel works for both SMP and plain kernels since the one available in yum doesnt say anything about it |
| 20:03.24 | kippi | can someone help me with this error? |
| 20:03.25 | kippi | http://pastebin.com/578906 |
| 20:05.23 | Skumling | kippi: are you using zaphfc ? |
| 20:05.43 | kippi | Skumling: nope |
| 20:06.19 | Skumling | kippi: humm okay. well, maybe you have to do a ztcfg -vv anyways |
| 20:06.23 | Skumling | did you run ctcfg? |
| 20:07.04 | rene- | i think you could do lsmod | grep zaptel to see if you are loading the driver |
| 20:07.21 | Skumling | ztcfg even... |
| 20:07.36 | *** join/#asterisk kietlak (n=kietlak@11-mo3-6.acn.waw.pl) |
| 20:07.43 | kippi | 31 channels configured. |
| 20:07.43 | kippi | ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 20:07.56 | kippi | and |
| 20:07.57 | kippi | zaptel 196612 0 |
| 20:07.57 | kippi | crc_ccitt 6081 1 zaptel |
| 20:08.37 | SPoon_TSX | Hi there, May I know if I have 2 network card which all on it owns IRQ. Would it prossibly causing the echo or static problem with my TDM400P card? |
| 20:08.39 | Skumling | kippi: after doing ztcfg, can't you load asterisk then? |
| 20:09.17 | justinu | SPoon_TSX: if your TDM400 card is on the same interrupt as either of those network cards, yes. |
| 20:09.38 | rene- | kippi i assume you have an E1 card? |
| 20:09.47 | kippi | ztcfg |
| 20:09.47 | kippi | ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 20:10.05 | rene- | is the driver loading? lsmod | grep wct |
| 20:10.15 | kippi | TE110P |
| 20:10.16 | SPoon_TSX | Nope, they are all on their own. HELPPPPPP.... I got no echo but static on the phone..... I put up $100USD who can help me to fix the static problem. Anyone? |
| 20:10.40 | justinu | i'll give it a whirl |
| 20:10.42 | justinu | pm me |
| 20:10.46 | kippi | rene- got nothing when i did that |
| 20:10.59 | rene- | what model of card you own |
| 20:11.04 | SPoon_TSX | jsutinu: How can I pm? |
| 20:11.12 | justinu | <PROTECTED> |
| 20:11.13 | kippi | TE110P |
| 20:11.40 | redax | -quit |
| 20:12.12 | *** join/#asterisk peted20 (n=chatzill@71.39.93.58) |
| 20:12.21 | rene- | the 4 port models have to be dip switched for E1 or T1 operation, i dont know about your model, what are you using to load your card driver if anything at all |
| 20:12.51 | kippi | jumper is right |
| 20:13.12 | rene- | do you have any lines that start with modprobe besides modprobe zaptel? |
| 20:14.18 | rene- | run this kippi: modprobe wct1xxp |
| 20:14.33 | rene- | and then ztcfg -vv |
| 20:14.45 | kippi | [root@voip ~]# modprobe wct1xxp |
| 20:14.45 | kippi | ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 20:14.45 | kippi | FATAL: Error running install command for wct1xxp |
| 20:15.18 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 20:15.33 | rene- | what OS your running? |
| 20:16.04 | kippi | redhat |
| 20:16.14 | rene- | redhat what |
| 20:16.29 | kippi | RHEL |
| 20:16.33 | mikefoo | uughh.. |
| 20:16.38 | mikefoo | uninstall that asap |
| 20:16.48 | kippi | wh |
| 20:16.50 | kippi | why |
| 20:16.56 | Chotaire | anyone here who is good in Dial command? |
| 20:16.59 | Chotaire | maybe a dev? |
| 20:17.01 | rene- | were you able to install zaptel using that? |
| 20:17.09 | mikefoo | if you don't know, thats more of a reason why to uninstall rhel |
| 20:17.16 | *** join/#asterisk Grizzy (i=Generic@ppp-69-238-229-72.dsl.pltn13.pacbell.net) |
| 20:17.22 | rene- | :-D |
| 20:17.28 | kippi | but its supported |
| 20:17.40 | Chotaire | will option D not work if option G used? And will option M not work if option G used? |
| 20:17.56 | mikefoo | install fedora/debian/gentoo |
| 20:18.12 | lo_tech | leaving the Holy OS wars aside for a moment, kippi... are you using 2.6 kernel? |
| 20:18.21 | kippi | lo_tech yeah |
| 20:18.29 | *** join/#asterisk ManxPower (n=ewieling@stirprop-S4-0-0-21.ndcr2.datasync.net) |
| 20:18.33 | grandy | hello... does anyone know what ports need to be open in a firewall in order for sip to work? |
| 20:18.41 | lo_tech | kippi: and checked out the README.udev, right? |
| 20:18.48 | [TK]D-Fender | grandy : 560, 10000-20000 UDP |
| 20:18.49 | kippi | yeah |
| 20:18.50 | ManxPower | *sigh* |
| 20:18.57 | ManxPower | grandy, ALL OF THEM. |
| 20:19.00 | kippi | and added the lines |
| 20:19.09 | ManxPower | Well UDP 5060 PLUS whatever ports are used for audio |
| 20:19.10 | Hmmhesays | anyone know how to drop back to a failure_route[x] if(!lookup("location")) in SER? |
| 20:19.11 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 20:19.12 | *** join/#asterisk dpolitech (n=Owner@c-67-172-132-21.hsd1.co.comcast.net) |
| 20:19.15 | rene- | kippi: how did you compiled zaptel |
| 20:19.26 | grandy | [TK]D-Fender: do u mean 5060? |
| 20:19.28 | mikefoo | if you don't have a licensed rhel with up2date account do NOT use rhel |
| 20:19.39 | kippi | make clean ; make install |
| 20:19.43 | kippi | mikefoo: I do |
| 20:19.44 | [TK]D-Fender | ManxPower : We don't need no stiiiiiinking audio! |
| 20:19.46 | ManxPower | Our Bell account manager AND our Bell technical contact are on vacation, we have a cutover scheduled for tonight. |
| 20:19.51 | [TK]D-Fender | grandy : Yeah, 5060 |
| 20:20.14 | mikefoo | kippi: then my question to use you, why the hell are you spending redhat buck? |
| 20:20.19 | grandy | thanks guys |
| 20:20.41 | kippi | its what my boss wants so he gets it, not my money |
| 20:20.48 | ManxPower | And based on a message from our account rep one of two things are happening 1) lines are moved over tonight, but the number port won't happen until friday or 2) the cutover will happen on friday. I'm not available on friday evening. |
| 20:21.27 | rene- | kippi: there most likely reason is that your RHEL using udev is not creating the zaptel files, i m no expert but the wiki has a wealth of info on udev and asterisk |
| 20:21.28 | lo_tech | kippi: not make linux26, make install? |
| 20:21.56 | [av]bani | ManxPower: welcome to ilec hell |
| 20:22.00 | Fedoracore6 | i already tried |
| 20:22.14 | Fedoracore6 | and cannot the connection the database |
| 20:22.21 | kippi | lo_tech: so do make ; make linux26? |
| 20:22.27 | ManxPower | [av]bani, It's still better than the CLEC hell we went thru with this office after Katrina. |
| 20:22.28 | rene- | how do i fetch a specific version of a package using yum |
| 20:22.37 | *** part/#asterisk skkip (n=Skipper@216.160.91.91) |
| 20:22.40 | Curus | rene-: Specify the version? |
| 20:22.43 | Fedoracore6 | sleep and cotinued tomorrow |
| 20:22.50 | Chotaire | manxpower: could you assist me with one command? I am totally stuck. |
| 20:22.51 | Fedoracore6 | bye all |
| 20:22.54 | [av]bani | ManxPower: well, ilec have the fed govt in their pocket... so of course they get preferential treatment |
| 20:23.06 | ManxPower | I've not seen my Significant Other since Dec 15th, we have a hotel room, a spa tub, and wine for Friday - tuesday. there's no way in hell I'm going to do a T-1 install on friday. |
| 20:23.24 | Beirdo | yeah, screw the T1 |
| 20:23.31 | rene- | Curus: how can i do that? |
| 20:23.40 | lo_tech | kippi: cd /usr/src/zaptel; make clean; make linux26; make install |
| 20:23.40 | *** join/#asterisk Dr-Linux (n=Dreamer@host202-147-168-130.lhr.dancom.net.pk) |
| 20:23.41 | ManxPower | Beirdo, that would not be nearly as fun. |
| 20:23.48 | Beirdo | heh, so true |
| 20:23.59 | ManxPower | Chotaire, make it fast, I'm in a bad mood. |
| 20:24.04 | Chotaire | Dial(SIP/test|10|gD(www0)M(tp-locked1)G(tp-pool1-control^5^3)) |
| 20:24.38 | ManxPower | Chotaire, other than using old style | in a new style Dial line, what's the problem? |
| 20:24.42 | Chotaire | I want "www0" to be dialed, the callee to be put into macro tp-locked and the caller to continue with tp-pool1-control,5,3 after that connection been established. |
| 20:24.49 | xmark | How can I configure a channel (In my case FXO) for PSTN to always ring extenstion 1000? |
| 20:24.51 | cthompson | ok, so I wanted to sign up at iaxtel.com |
| 20:24.59 | cthompson | but the site has been down for three days |
| 20:25.03 | cthompson | anyone know what happened to them? |
| 20:25.05 | Chotaire | D won't work when option M or G is in use and M doesn't seem to work when option G is in use. |
| 20:25.10 | lo_tech | kippi: that and the manual entries to /etc/udev/rules.d/50-udev.rules and /usr/share/festival/festival.scn were the only things that were different about setting * up on RH |
| 20:25.15 | ManxPower | Chotaire, can't help you with that |
| 20:25.15 | Chotaire | or I am totally f*cking that up. |
| 20:25.28 | ManxPower | I used M() for the first time the other night. |
| 20:25.43 | rene- | Curus: the wanted package is in base, but im getting the ones in updates released... how do i get the one i want? |
| 20:25.44 | Chotaire | any idea how I could get that to work without M? |
| 20:26.06 | Chotaire | it's important the callee gets jumped to a different context than myself, and it's important there will be no bridging between caller and callee |
| 20:26.21 | x86 | hmm |
| 20:26.32 | kippi | arrrrrrrrggggggggghhh |
| 20:26.48 | x86 | i have a group of extensions that i don't want to be able to dial through a particular dialplan... is there a way i can prevent this? |
| 20:27.03 | lo_tech | getting kinda piratical in here |
| 20:27.12 | ManxPower | x86, put them in a different contexts, that's what contexts are for |
| 20:27.26 | Chotaire | manxpower: you know anyone who could assist me with that? |
| 20:27.32 | ManxPower | Chotaire, nope. |
| 20:27.49 | ManxPower | If I was bored, not in crisis mode, and not on a business trip, I might have time to try it myself. |
| 20:28.13 | lo_tech | x86: well, 'exten => 19005555555, Congestion' will do |
| 20:29.18 | Chotaire | manxpower: ok thanks anyway. |
| 20:29.54 | X-Rob_ | lo_tech, you forgot the priority. |
| 20:30.21 | lo_tech | s/, C/,1,C/ |
| 20:32.35 | Dr-Linux | when asterisk had first version? |
| 20:32.49 | Dr-Linux | when asterisk released for the first time? |
| 20:33.02 | Dr-Linux | 1999 ? |
| 20:33.43 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 20:33.51 | shmaltz | whats wrong with the list? |
| 20:34.02 | brookshire | they are broken :) |
| 20:34.30 | x86 | ManxPower: so make a context for outbound PSTN, a context for extensions allowed to go out via PSTN, and a context for extensions who can not go out PSTN.... |
| 20:34.31 | cpm | they are flat gone! |
| 20:34.52 | x86 | ManxPower: i figured this much, but how do i allow the extensions out via the outbound context? |
| 20:35.34 | *** join/#asterisk seele2 (n=_seele@200.124.172.72) |
| 20:36.20 | seele2 | Hi... is there anyone that could help me with this please i beg you: None of my extensions are hearing Asterisk default recordings such as the clock.. or any of the menus.. The CLI Console shows that there is already the correct File playing to the extension.. but i hear nothing.. doesn't matter what phone is is not the phone.. is the asterisk... Can any one offer me a little help with this please?? |
| 20:37.27 | seele2 | _Sam--, hi, im just checking if you can see me or if i'm not registered. |
| 20:37.43 | seele2 | _Sam--, Answer me just to see please |
| 20:37.46 | ManxPower | x86, Better yet, read the Asterisk book |
| 20:38.04 | x86 | ManxPower: url? |
| 20:38.08 | cpm | that's a good idea. Best bucks you'll spend on your asterisk system |
| 20:38.09 | ManxPower | ~docs |
| 20:38.10 | jbot | from memory, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 20:38.17 | ManxPower | http://www.oreilly.com/catalog/asterisk |
| 20:38.51 | seele2 | nickserv identify 123456 |
| 20:39.32 | Abydos313 | haha |
| 20:39.35 | seele2 | ManxPower, Are you registered?? is anyone Chatting in this channel already? |
| 20:39.42 | Abydos313 | seele2 you forgot the /msg part |
| 20:39.50 | seele2 | i know lol.. |
| 20:40.17 | seele2 | Abydos313, Typo |
| 20:40.36 | Abydos313 | ;) |
| 20:40.48 | fu3 | hey, when I hear of "D4 Channel Bank" whats up with the D4? |
| 20:40.57 | fu3 | is that a specification, or a standard of some sort? |
| 20:41.04 | x86 | fu3: framing type? |
| 20:41.09 | fu3 | ahhh |
| 20:41.17 | fu3 | ok. |
| 20:41.20 | cpm | same as ami |
| 20:41.21 | x86 | fu3: i know of a D4 T1 circuit... it could be different though |
| 20:41.25 | x86 | cpm: right |
| 20:41.32 | fu3 | yeah well I have a T1 and I guess it's D4. |
| 20:41.35 | fu3 | so that makes sense. |
| 20:41.50 | x86 | fu3: more common is ESF / B8ZS though ;) |
| 20:41.55 | *** join/#asterisk NovceGuru (n=asdf@oh-71-53-48-11.dhcp.sprint-hsd.net) |
| 20:42.01 | fu3 | ok, i've heard of those too. |
| 20:42.02 | cpm | yeah, my channel bank claims it does ami/esf, but it is a liar |
| 20:42.04 | x86 | fu3: AMI / D4 is oldschool :) |
| 20:42.20 | x86 | cpm: you can do ESF with AMI? |
| 20:42.24 | fu3 | I just found out that my ex boss ordered a T1 like 3 years ago, and it's been idle ever since. |
| 20:42.36 | x86 | fu3: oh wow ;) |
| 20:42.37 | fu3 | so I'm getting a line swung onto it, and am going to begin testing * across the T1 |
| 20:42.37 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 20:42.43 | fu3 | but dont know what kind of questions to ask |
| 20:42.44 | Abydos313 | NICE! |
| 20:42.46 | fu3 | this is my first T1 :) |
| 20:42.49 | cpm | nope, they are exclusive, It says it is dip switch selectable, but it isn't. |
| 20:43.11 | x86 | fu3: so it's a T1 to the internet? |
| 20:43.16 | fu3 | its a T1 to the telco. |
| 20:43.21 | x86 | err |
| 20:43.21 | cpm | good start |
| 20:43.22 | fu3 | thats about ALL i know |
| 20:43.26 | x86 | with nothing running on it? :P |
| 20:43.30 | fu3 | nothing at all. |
| 20:43.31 | fu3 | it's 100% idle |
| 20:43.36 | Abydos313 | voice or data t1? |
| 20:43.40 | gbodemantv | mroth: so we went by your document, and still the ramdisk fails and locks up the server hard |
| 20:43.45 | fu3 | the qwest guys were here troubleshooting another issue, when they asked about 'why this t1 isnt hooked up to anything' |
| 20:43.46 | gbodemantv | even if asterisk is not running |
| 20:43.52 | fu3 | and I was like 'what t1?' and we went from there. |
| 20:44.01 | x86 | fu3: so it's voice or data? |
| 20:44.05 | fu3 | Abydos313.. it's a T1, and we're GOING to be pushing voice over it |
| 20:44.07 | gbodemantv | get an out of memory error , killing process (hhtp) |
| 20:44.12 | gbodemantv | or (asterisk) |
| 20:44.23 | fu3 | weather or not it's hooked up to some kind of data service on the CO side or not, i do not know. |
| 20:44.24 | x86 | fu3: i'm assuming that if it's D4, it's data :P |
| 20:44.44 | fu3 | well I dont *know* that it's D4. |
| 20:44.46 | fu3 | for certain. |
| 20:44.49 | fu3 | so, sorry about that. |
| 20:44.52 | cpm | look at the smartjack |
| 20:44.56 | fu3 | ? |
| 20:45.04 | fu3 | this T1 is punched into a 66 block and thats where it sits. |
| 20:45.07 | x86 | fu3: smartjack is what the T1 runs into |
| 20:45.13 | *** join/#asterisk mcreedjr (n=mcreedjr@cblmdm72-240-21-51.buckeyecom.net) |
| 20:45.25 | x86 | fu3: smartjack turns 2-pair into RJ48C |
| 20:45.38 | fu3 | well, i dont have a smartjack then. |
| 20:45.44 | MikeJ[Laptop] | fu3, it's somewhere |
| 20:45.47 | x86 | fu3: RJ48C runs into CSU, DSX, or PBX |
| 20:45.52 | *** join/#asterisk Seyr (n=Seyr_@cpe-67-10-139-141.houston.res.rr.com) |
| 20:45.54 | mcreedjr | Does anyone have any success running VoIP over the internet? I'm having a real tough time with voice quality |
| 20:45.54 | MikeJ[Laptop] | maybe not in your office area |
| 20:45.59 | x86 | fu3: you dont have a T1 unless you have a smartjack ;) |
| 20:46.03 | MikeJ[Laptop] | mcreedjr, sometimes |
| 20:46.05 | fu3 | like I said, it goes right into a 66 block. |
| 20:46.05 | mcreedjr | I'm not sure if its packet loss, or something else. |
| 20:46.10 | fu3 | oh ok.. well i'll trace the wire |
| 20:46.15 | fu3 | make sure I know where it's coming from |
| 20:46.24 | MikeJ[Laptop] | fu3, where are you? |
| 20:46.25 | MikeJ[Laptop] | US? |
| 20:46.27 | fu3 | yheah |
| 20:46.30 | fu3 | yeah |
| 20:46.32 | syzygybsd | mcreedjr: it works fine for me |
| 20:46.33 | MikeJ[Laptop] | big building? |
| 20:46.34 | x86 | fu3: smartjack is usually right at the demarc |
| 20:46.38 | fu3 | it's a college |
| 20:46.44 | x86 | fu3: unless they extend the demarc |
| 20:46.48 | cpm | my smartjack is in my bedroom |
| 20:46.49 | MikeJ[Laptop] | heh... who knows then.. follow the wires.. |
| 20:46.50 | mcreedjr | MikeJ[Laptop]: I would say that 90% of the time the voice quality is fine, other times there are long pauses and breaks in the voice stream and things like that |
| 20:46.56 | Dr-Linux | when asterisk released for the first time? |
| 20:46.58 | Dr-Linux | 1999 ? |
| 20:47.00 | fu3 | hmm.. all I saw was the wire being terminated in a 66 block, I did not verify the path of that wire. |
| 20:47.03 | x86 | fu3: have you no toner / prober? |
| 20:47.04 | Seyr | anyone know where to set the timeout for digits for the # transfer? Example: If I am on a call and press #, the Asterisk box says "transfer" and waits for me to dial an extension.. but if I wait more than 2 or 3 seconds, it gives an invalid. |
| 20:47.06 | fu3 | so, it COULD be coming from somewhere else. |
| 20:47.07 | MikeJ[Laptop] | mcreedjr, bad connection probably |
| 20:47.18 | fu3 | x86.. yeah I have cable location gear. |
| 20:47.20 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:47.25 | x86 | fu3: tone it out then ;) |
| 20:47.29 | mcreedjr | MikeJ[Laptop]: On one end, or transient? |
| 20:47.35 | fu3 | I will, now that I know I should be finding a smartjack. |
| 20:47.38 | x86 | fu3: when you find the smartjack, come back to us... :) |
| 20:47.42 | syzygybsd | mcreedjr: what kind of connection do you have? |
| 20:47.44 | fu3 | ok.. i'll go right now :) |
| 20:47.45 | fu3 | brb |
| 20:47.54 | x86 | fu3: write down everything you see on the smartjack |
| 20:47.58 | fu3 | thanks by the way. |
| 20:48.00 | fu3 | ok.. will do. |
| 20:48.07 | mcreedjr | syzygybsd: full T1 on one side, Cable on the other |
| 20:48.20 | gbodemantv | having a ramdisk issue still |
| 20:48.23 | x86 | fu3: if it's a multi-spot smartjack, write down all of the info on all of the cards, and record it in a safe place ;) |
| 20:48.26 | gbodemantv | anyone...can you help? |
| 20:48.34 | x86 | mcnobody: what codec? |
| 20:48.37 | syzygybsd | ok, just a question, but is there any p2p application running on either connection? |
| 20:48.48 | x86 | mcreedjr: even |
| 20:48.57 | mcreedjr | x86: u-law |
| 20:49.04 | x86 | mcreedjr: hmm.. should be fine |
| 20:49.19 | x86 | mcreedjr: switch to GSM (also low-bandwidth) and see if it makes a difference |
| 20:49.21 | mcreedjr | x86: my thoughts exactly.. i have outbound prioritization setup on both sides... |
| 20:49.34 | x86 | mcreedjr: you cant prioritize across the internet silly ;) |
| 20:49.50 | x86 | mcreedjr: you can police, but not prioritize... |
| 20:49.53 | mcreedjr | x86: no, i know, i'm talking about prioritizing it out of my link |
| 20:50.06 | mcreedjr | x86: just so it doesn't ahve to contend with other traffic at my locations |
| 20:50.10 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-190-62.red.bezeqint.net) |
| 20:50.15 | x86 | mcreedjr: once it hits the Internet it's fair game for thrashing ;) |
| 20:50.16 | mcreedjr | x86: and you're silly. |
| 20:50.21 | *** join/#asterisk Ukyo (n=Kami-Sam@68-113-211-246.dhcp.ftwo.tx.charter.com) |
| 20:50.41 | justinu | not as silly as file |
| 20:50.52 | Ukyo | alright, so I have reda the voip-info pages on using nat. it tells you what ports to open when the phone is behind nat, and the * ispublic. I need the other way around. |
| 20:50.53 | x86 | what did file do now? heh |
| 20:50.54 | mcreedjr | x86: right, i realize that... |
| 20:50.59 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 20:50.59 | syzygybsd | hey, he isn't here to defend himself |
| 20:51.03 | Ukyo | my * is behind nat, and the phone is public. |
| 20:51.08 | justinu | he wouldn't defend himself even if he was |
| 20:51.11 | justinu | he'd agree ;) |
| 20:51.29 | Ukyo | i have a phone that works perfectly with the server behind the nat, but if I take it to public ip, * cant receive any audio. |
| 20:51.30 | mcreedjr | syzygybsd: what kind of connections do you have without a problem? |
| 20:51.33 | syzygybsd | Ukyo: same ports |
| 20:51.34 | x86 | Ukyo: so forward ports 5060 (UDP) and 9000-20000 into your * box |
| 20:51.40 | Ukyo | already done |
| 20:51.46 | Ukyo | I even set DMZ to the asterisk servers ip's |
| 20:51.51 | Ukyo | but it still does not work |
| 20:52.03 | Ukyo | I tested the dmz, I _know_ its forwarding |
| 20:52.06 | x86 | look at the logs on the firewall? |
| 20:52.12 | syzygybsd | mcreedjr: cable and > 10mbit trunk |
| 20:52.12 | Ukyo | crappy linksys >,> |
| 20:52.32 | x86 | Ukyo: you sure it's forwarding UDP and not TCP? |
| 20:52.37 | Ukyo | I set it to both |
| 20:52.41 | Ukyo | just to be safe |
| 20:52.43 | mcreedjr | syzygybsd: hrm.. |
| 20:52.47 | mcreedjr | thanks for the help all |
| 20:52.53 | syzygybsd | Ukyo: if they are on the smae network does it work? |
| 20:52.59 | Ukyo | yes |
| 20:53.14 | Ukyo | if i move the phone behind the nat with the * server, it works fine |
| 20:53.22 | syzygybsd | well, the question is then, what does tcpdump say? |
| 20:53.35 | Ukyo | the only thing anyone can come up with, is that charter is blocking the ports. |
| 20:53.53 | syzygybsd | Ukyo: change the ports adn see if it works |
| 20:54.04 | Ukyo | where do I change them? |
| 20:54.10 | Ukyo | the only changeable ports on the cisco is 5060 |
| 20:54.12 | Ukyo | but thats not rtp |
| 20:54.22 | Ukyo | cisco lists media ip range.. but you cant change those |
| 20:54.32 | *** join/#asterisk Dr-Linux (n=Dreamer@host202-147-168-130.lhr.dancom.net.pk) |
| 20:54.45 | syzygybsd | sip.conf |
| 20:55.24 | *** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl) |
| 20:55.41 | Ukyo | thats just the control prot. |
| 20:55.41 | Ukyo | 5060 |
| 20:55.59 | fu3 | ok |
| 20:56.00 | fu3 | im back |
| 20:56.02 | Ukyo | othing for the voice ports |
| 20:56.04 | Ukyo | nothing even |
| 20:56.27 | syzygybsd | the control port is what they would block |
| 20:56.38 | syzygybsd | I couldn't see them blocking a random range |
| 20:56.45 | fu3 | from the 66 block, the cat5 cable plugs into an "ADTRAN" with no model #. From there, it goes into one of two 25-pair wires, that go into a Nortel OC-3/12 Express HX box, and from there, into fiber and out to the CO. |
| 20:56.55 | Ukyo | if the control port was blocked, I would'nt be able to connect at all |
| 20:56.57 | Ukyo | which is not the case |
| 20:57.17 | syzygybsd | oh, just voice issues then? |
| 20:57.43 | Ukyo | yes |
| 20:57.47 | fu3 | x86? |
| 20:57.53 | syzygybsd | can you hear one way but not the other |
| 20:58.04 | Ukyo | the nat docs say, if the phone is behind nat, and the server is public ip, then you cant hear anything, but can talk |
| 20:58.08 | Ukyo | in this case, its the oppposite |
| 20:58.13 | Ukyo | since the server is behind nad |
| 20:58.18 | Ukyo | at the phone, i can hear |
| 20:58.20 | Ukyo | but cant talk |
| 20:59.17 | Druken | well, who in their right mind would put a server being a nat ? |
| 20:59.26 | Druken | er... behind |
| 20:59.41 | Ukyo | its an off-site server |
| 20:59.50 | Ukyo | either wya |
| 20:59.50 | syzygybsd | lol, anyone who uses it at home |
| 21:00.04 | syzygybsd | and it is DMZed anyway |
| 21:00.22 | austinnichols101 | druken: servers behind nat is done all the time. |
| 21:00.33 | Ukyo | yeah, so I am pulling my hair out |
| 21:00.35 | Druken | austinnichols101: not in my world :) |
| 21:00.42 | syzygybsd | do you still have the ports forwarded and the dmz? |
| 21:00.45 | Ukyo | Druken: your worlds getting shattered then. :P |
| 21:00.51 | *** part/#asterisk Seyr (n=Seyr_@cpe-67-10-139-141.houston.res.rr.com) |
| 21:00.55 | Ukyo | syzygybsd: yes |
| 21:01.06 | Druken | yup, keep this up and i'll be crying in the corner in the fetal position |
| 21:01.19 | Ukyo | Druken: been there, done that |
| 21:01.23 | syzygybsd | Druken: are you on IPv6 only and have no issue with the number of ipaddreses you have? |
| 21:01.32 | fu3 | I think instead of crying, you should create an invisible world of perfection all around you. |
| 21:01.33 | Hmmhesays | damn its easy to send SER into a nasty loop |
| 21:01.48 | Ukyo | fu3: hey, there's a good one |
| 21:01.55 | Druken | syzygybsd: no... i have a block of ip's... no shortage |
| 21:01.57 | GerbilWrk | ok, Teliax is loosing my service, and voipreach.net is not impressing me, any other providers yall would recommend? |
| 21:02.01 | Ukyo | "Let's all live in happy land." -Homer Simpson |
| 21:02.07 | fu3 | :) |
| 21:02.19 | Ukyo | I have a nice.. large... block of IPv6 :) |
| 21:02.42 | fugitivo | i want one |
| 21:02.43 | Druken | i got ipv6 too... |
| 21:02.46 | austinnichols101 | druken: just hope that you don't ever have to switch blocks of Ips. With NAT is a piece of cake... |
| 21:02.51 | Ukyo | Druken: mine is bigger. |
| 21:02.52 | Ukyo | :P |
| 21:03.03 | Beirdo | IPv6 penis wars now? |
| 21:03.05 | fugitivo | Ukyo: give me a piece |
| 21:03.07 | Ukyo | hell yeas |
| 21:03.12 | Ukyo | ePenis! |
| 21:03.24 | Beirdo | ewww |
| 21:03.24 | mcreedjr | ePenis V6 |
| 21:03.36 | Ukyo | EPv6 |
| 21:03.44 | [av]bani | o_O |
| 21:03.44 | Beirdo | I'm sure others might enjoy that, but not me :) |
| 21:03.46 | mcreedjr | I'm only runnin' V4 over here... I think it makes me less of a man. |
| 21:03.51 | Druken | :) |
| 21:03.53 | mcreedjr | :( |
| 21:03.58 | *** join/#asterisk mrempire (n=trefpunt@mrempire.demon.nl) |
| 21:04.02 | fugitivo | mcreedjr: gay |
| 21:04.09 | Ukyo | mcnobody: well, hopefully I will get time to setup the tunnel server soon |
| 21:04.14 | Ukyo | and I will give out ipv6 blocks |
| 21:04.20 | mcreedjr | fugitivo: erm.. creative. |
| 21:04.32 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-46.west.biz.rr.com) |
| 21:04.58 | Ukyo | alright then, how about this.. |
| 21:05.03 | Ukyo | any of you ever done failover? |
| 21:05.19 | docelm0 | ya why? |
| 21:05.21 | docelm0 | HA |
| 21:05.22 | Dr-Linux | HeartBeat? |
| 21:05.29 | Ukyo | if I have * running primarily at the office, if the office drops offline, an offsite server kicks in and handles incoming calls |
| 21:05.43 | Ukyo | any good docs on setting it up ? |
| 21:06.01 | Dr-Linux | linux-ha.org |
| 21:06.02 | docelm0 | Setting up HA um, search google for Linux HA |
| 21:06.06 | docelm0 | ya thats it.. |
| 21:06.14 | Ukyo | I run * on FreeBSD >,> |
| 21:06.49 | mcreedjr | Ukyo: Multihome and run BGP :) |
| 21:07.01 | Dr-Linux | how does * work on Solaris? |
| 21:07.05 | mcreedjr | Ukyo: You didn't say it had to be cheap |
| 21:07.16 | stoffell | lol mcreedjr |
| 21:07.24 | glm2k | OT: hmmm Ukyo, you a fan of Samurai 7? |
| 21:08.06 | Ukyo | mcnobody: I do run BGP, I do have my own /19 |
| 21:08.10 | Druken | mcreedjr: he's bitching about his lack of ipv4, you think he's going to be multihomed? |
| 21:08.15 | Ukyo | mcnobody: I am talking, mejor meltdown JIC situation |
| 21:08.18 | Dr-Linux | nobody answer me |
| 21:08.24 | Dr-Linux | when asterisk released for the first time? |
| 21:08.32 | Ukyo | Druken: I never said I had a lack of IPv6 |
| 21:08.35 | Ukyo | er IPv4 even |
| 21:08.49 | Druken | then i must have read something wrong :) |
| 21:09.00 | mcreedjr | Druken: Take it back. |
| 21:09.01 | mcreedjr | lol |
| 21:09.02 | Ukyo | mebbe it was someone else. :) |
| 21:09.08 | Ukyo | mcreedjr: lol |
| 21:09.12 | mcreedjr | And everyone stop autocompleting my name wrong |
| 21:09.14 | mcreedjr | heh |
| 21:09.17 | Ukyo | I have a /19, a /8 (IPv6) |
| 21:09.37 | mcreedjr | Sweet. I have a /25 w00... |
| 21:09.49 | synthetiq | i have a /0 |
| 21:09.51 | synthetiq | what what |
| 21:09.58 | stoffell | Dr-Linux, it's on asterisk.org somewhere |
| 21:09.58 | Ukyo | "mcnobody": yeah, but mine is mine. you have to give yours up :P |
| 21:10.07 | synthetiq | i own all fo you! |
| 21:10.17 | synthetiq | of |
| 21:10.26 | mcreedjr | Ukyo: True.. not a big enough EP to keep mine |
| 21:10.47 | Ukyo | I was just happy I get my IPv6 for free |
| 21:10.54 | Ukyo | because I have teh IPv4 block. |
| 21:11.54 | Dr-Linux | stoffell: one of our client complaint, he made a call and did stay in a queue for 10 minutes, i confess there was no one available, all i want to see his call logs .. |
| 21:11.55 | Druken | uhmm... ya can't have a 255.... |
| 21:12.02 | Druken | ya could have a 254.... |
| 21:12.07 | Ukyo | 1 sec |
| 21:12.15 | Dr-Linux | stoffell: but i'm unable to see his call info in queue logs beside all others |
| 21:12.24 | Dr-Linux | even i can see all other log .. |
| 21:12.31 | *** join/#asterisk Ukyo (i=br4d@66.207.160.128) |
| 21:12.32 | Ukyo | :) |
| 21:12.33 | synthetiq | like a 68.113.255.246 |
| 21:12.42 | Dr-Linux | does it mean, if the call is not answered queue doesn't log? |
| 21:12.45 | mcreedjr | Druken: you could have a 255 as long as its not the broadcast.. he could be supernetting. |
| 21:13.09 | stoffell | Dr-Linux, unanswered calls are also logged.. |
| 21:13.17 | Ukyo-tmp | are we happey now synthetiq? |
| 21:14.01 | synthetiq | what is special about that ip |
| 21:14.05 | Dr-Linux | stoffell: but what could be happend, i can't see his call logs ? :S |
| 21:14.10 | Ukyo-tmp | look up ownership |
| 21:14.17 | Ukyo-tmp | its part of my /19 |
| 21:14.29 | Ukyo-tmp | let me know when you come up with the company name |
| 21:14.30 | Ukyo-tmp | :) |
| 21:14.34 | Beirdo | that's a /20 |
| 21:14.42 | Ukyo-tmp | Beirdo: well, i am claiming /19 |
| 21:14.47 | stoffell | Dr-Linux, no idea, if logging is verbosely enough, you should see them in your logs.. |
| 21:14.48 | Ukyo-tmp | cause they have the 2nd /19 reserved |
| 21:14.58 | Ukyo-tmp | been meaning to request the upgrade |
| 21:15.01 | Ukyo-tmp | er |
| 21:15.03 | Ukyo-tmp | 2nd /20 |
| 21:15.08 | Beirdo | ah, you have the adjacent /20 |
| 21:15.10 | Beirdo | gotcha |
| 21:15.11 | Dr-Linux | stoffell: i have an other question about queue .. |
| 21:15.12 | Ukyo-tmp | yeah |
| 21:15.13 | *** join/#asterisk gnosys_ (n=gnosys_@griffin2.GnoSys.us) |
| 21:15.24 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 21:15.24 | stoffell | Dr-Linux, i'm no queue expert :( |
| 21:15.53 | MikeJ[Laptop] | Dr-Queue, i'm no Linux expert :D |
| 21:15.55 | Dr-Linux | does asterisk queue have option, "call should be route for most idle member" ? |
| 21:15.59 | Katty | MikeJ[Laptop]: ! |
| 21:16.01 | Ukyo-tmp | synthetiq: you found the company name yet ? |
| 21:16.01 | Dr-Linux | opss |
| 21:16.11 | Katty | MikeJ[Laptop]: by request of my accounting department, i'm bugging you |
| 21:16.15 | Dr-Linux | MikeJ[Laptop]: lolzzzzzzz youuuuu :@ :P |
| 21:16.16 | gnosys_ | could someone point me to a howto for setting up the Snom 360 hardphone with asterisk? I see one on the wiki for the 360 softphone but not for the hardphone... |
| 21:16.18 | Katty | MikeJ[Laptop]: bugbug, paperwork, bugbug, hotel, bugbug, etc. |
| 21:16.21 | Katty | MikeJ[Laptop]: kthx, that is all. |
| 21:16.37 | MikeJ[Laptop] | no prob.. |
| 21:16.40 | MikeJ[Laptop] | happy to help |
| 21:16.49 | Ukyo-tmp | synthetiq: mebbe if you ask Beirdo he can tell ya :P |
| 21:16.52 | Dr-Linux | MikeJ[Laptop]: then help me :P |
| 21:16.56 | MikeJ[Laptop] | ok |
| 21:16.59 | synthetiq | cologuys |
| 21:16.59 | MikeJ[Laptop] | with what? |
| 21:17.04 | Ukyo-tmp | good |
| 21:17.05 | Ukyo-tmp | http://www.cologuys.com/ast.php |
| 21:17.13 | Ukyo-tmp | I think that should be sufficient proof. :) |
| 21:17.25 | Beirdo | heh |
| 21:17.35 | Dr-Linux | MikeJ[Laptop]: i have 4 queue members .. i want incoming call should be route to most idle member :) |
| 21:17.39 | Beirdo | I think people need to learn how to use ARIN |
| 21:17.47 | MikeJ[Laptop] | what is "most idle" |
| 21:17.48 | Ukyo-tmp | or even whois -a |
| 21:17.57 | *** join/#asterisk apardo (n=apardo@87.218.44.213) |
| 21:18.40 | Dr-Linux | MikeJ[Laptop]: hhm.. i mean there are 4 queue members .. 1st one took a call, then 2nd one then 3rd one .. |
| 21:19.02 | Dr-Linux | so in this case 4th one is most time idle :S |
| 21:19.19 | Beirdo | and a /16 in IPv6 according to ARIN |
| 21:19.20 | Dr-Linux | 4th member did get a call since long .. |
| 21:19.27 | Ukyo-tmp | is there any modules or way 2 asterisks servers can be linked, and if a server sees the other drop, it enables an iax connection for inbound ? |
| 21:19.32 | Dr-Linux | so call should be route to him/her :S |
| 21:19.36 | Dr-Linux | sorry for my bad english |
| 21:19.37 | Ukyo-tmp | was it 16? |
| 21:19.47 | Ukyo-tmp | got it like 2 months ago, and have not had time to touch it |
| 21:19.55 | Ukyo-tmp | been busy with all the new server setups and the new gige |
| 21:19.59 | Beirdo | 2001:49E8:... |
| 21:20.06 | Ukyo-tmp | yeah, 16 then. :) |
| 21:20.16 | Beirdo | sufficiently large for now |
| 21:20.21 | mcreedjr | and busy with the chatting in the #asterisk room? :) |
| 21:20.35 | Ukyo-tmp | mcreedjr: supposed to be making progress on this blasted phone :/ |
| 21:20.38 | Dr-Linux | MikeJ[Laptop]: you still don't understand my queustion? |
| 21:20.47 | mcreedjr | Ukyo-tmp: just bustin' your chops |
| 21:21.01 | Katty | what room does 888 go to? |
| 21:21.29 | *** part/#asterisk Ukyo (i=br4d@66.207.160.128) |
| 21:21.56 | mcreedjr | Ukyo: I know how those problems are :) A pain in the ass... |
| 21:22.07 | Ukyo | been fighting it for days now. |
| 21:22.17 | Ukyo | I'd be happy just to make the server local |
| 21:22.23 | Ukyo | but I want backup. |
| 21:22.37 | Ukyo | brb, snack |
| 21:22.47 | Katty | oh. snack. |
| 21:22.48 | Katty | hmm. |
| 21:22.54 | mcreedjr | Ukyo: Couldn't you just script it somehow, doesn't seem like it'd take much |
| 21:22.55 | iDunno | lalala |
| 21:23.02 | iDunno | today is mostly over - hoorah! |
| 21:23.06 | Katty | yay! |
| 21:23.10 | iDunno | tomorrow is yet to come ;) |
| 21:23.13 | Katty | :< |
| 21:23.25 | iDunno | but we're getting close to a weekend again :) |
| 21:23.26 | mcreedjr | Ukyo: Use ping as a heartbeat, then if it fails, have the script setup IAX |
| 21:23.28 | Katty | :> |
| 21:23.32 | MikeJ[Laptop] | Dr-Linux, sorry.. I'm back... |
| 21:23.36 | MikeJ[Laptop] | the answer is kinda |
| 21:23.45 | Dr-Linux | MikeJ[Laptop]: WB |
| 21:23.48 | Katty | iDunno: mahna mahna! |
| 21:23.57 | MikeJ[Laptop] | look at roundrobin and rrmemory |
| 21:24.07 | MikeJ[Laptop] | or fewestcalls |
| 21:24.30 | Katty | iDunno: does that not parse? |
| 21:24.44 | Dr-Linux | MikeJ[Laptop]: yes, but what's the option for "idle" ? |
| 21:24.46 | Dr-Linux | is there any? |
| 21:24.47 | MikeJ[Laptop] | you can also do some magic with penalties on agents |
| 21:24.53 | MikeJ[Laptop] | idle? |
| 21:24.54 | iDunno | Katty: well, I'm just getting it as song lyrics, but I can't get the song ;) |
| 21:25.03 | Katty | iDunno: http://video.google.com/videoplay?docid=-5694538330599665511 |
| 21:25.03 | MikeJ[Laptop] | there really isn't a longest idle perse |
| 21:25.16 | MikeJ[Laptop] | it's based on who the calls are presented too |
| 21:25.26 | MikeJ[Laptop] | not who actually gets the calls |
| 21:25.49 | Dr-Linux | MikeJ[Laptop]: what's fewestcalls for? :S |
| 21:25.53 | iDunno | ahh :) |
| 21:25.56 | MikeJ[Laptop] | fewestcalls is pretty close to what you want I think |
| 21:26.11 | Katty | iDunno: today is mahnamahna |
| 21:26.22 | MikeJ[Laptop] | http://svn.digium.com/view/asterisk/trunk/configs/queues.conf.sample?rev=10163&view=markup |
| 21:26.29 | MikeJ[Laptop] | they are all listed in there |
| 21:26.30 | iDunno | Katty: I did not know that :) |
| 21:27.23 | Dr-Linux | MikeJ[Laptop]: my browsing :( |
| 21:27.49 | GerbilWrk | has anyone gotten queue announcements working in 1.2.4? |
| 21:27.51 | MikeJ[Laptop] | iDunno, don't beleive Katty |
| 21:27.57 | MikeJ[Laptop] | it's all a big ruse |
| 21:28.39 | Katty | yes. |
| 21:28.46 | Katty | to distract you from the muffins. |
| 21:30.06 | Beirdo | mmm, muffin |
| 21:30.24 | mcreedjr | Is there a "G" number which corresponds to GSM? |
| 21:31.13 | xmark | exit |
| 21:31.30 | Chotaire | dudes.. only one little little problem left.. someone please help... |
| 21:31.38 | Chotaire | exten => 5,2,Dial(SIP/soft-chotaire,10,gG(tp-pool1-control,5,3)D(www0)) |
| 21:31.55 | Chotaire | why would Dial command ignore D (dial dtmf ,,,0) when G is in use? |
| 21:32.12 | Chotaire | the order doesn't matter. |
| 21:32.48 | iDunno | MikeJ[Laptop]: but Katty is always honest and truthful. And pure as snow. (or something) |
| 21:33.53 | Katty | iDunno: i dunno about /always/ but i try to be. |
| 21:34.07 | Katty | iDunno: pure is snow is off just a smidgen. |
| 21:34.17 | Katty | s/is/as/ |
| 21:34.54 | iDunno | Katty: just how big a smidgen? ;) |
| 21:35.00 | *** join/#asterisk Qber (n=Qbera@natint3.juniper.net) |
| 21:35.04 | Katty | iDunno: probably more pure than you |
| 21:35.06 | Qber | hello...test |
| 21:35.12 | Beirdo | probably depends on how dirty the snow is... |
| 21:35.13 | greendisease | i keep getting an "Ouch, part reset, quickly restoring reality (0)" after using my zap channel |
| 21:35.19 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 21:35.19 | Qber | Okay, looks liek i am conncted now. |
| 21:35.20 | greendisease | whats causing it |
| 21:35.23 | greendisease | and how od i fix it |
| 21:35.28 | iDunno | Katty: hang on, I think we did this before... ;) |
| 21:35.34 | Katty | iDunno: k |
| 21:35.49 | Qber | Folks, i am looking for a speech recog system for asterisk. any pointers suggestions, success stories would be appreciated |
| 21:35.53 | kpettit | anybody have a startup script for wanrouter I can look at? |
| 21:35.53 | iDunno | Katty: I'm not entirely sure that you ever admitted how pure you weren't ;) |
| 21:36.04 | Katty | iDunno: not surprising. |
| 21:36.06 | Qber | perhaps, this has been discussed many time but the subject is very interesting. |
| 21:36.08 | *** join/#asterisk p0g0_ (n=pogo@madwifi/support/p0g0) |
| 21:36.09 | Katty | iDunno: it is none of your concern afterall |
| 21:36.28 | Qber | any pointers on open source or low cost ASR for asterisk??? |
| 21:36.44 | mcreedjr | After doing some diags, it appears that jitter is part of my call problem. I can adjust the jitter buffer bigger on my phone, but there is not equivalent on the * side of things is there? |
| 21:36.48 | Dr-Linux | Qber: phinx |
| 21:36.52 | mcreedjr | Using SIP/RTP, not IAX |
| 21:37.04 | Dr-Linux | Qber: but i never get it to work |
| 21:37.05 | iDunno | Katty: well, exactly - so you should obviously just admit to being pure ;) |
| 21:37.11 | Katty | k |
| 21:37.46 | Beirdo | ?? |
| 21:37.54 | Beirdo | slapping women is bad, mmmkay? |
| 21:38.18 | Katty | Dr-Linux: sorry, i only like girls. |
| 21:38.23 | Katty | Dr-Linux: go find another female to slap on |
| 21:38.33 | x86 | can someone tell me what i'm doing wrong here: http://www.shellshark.net/pub/ast-ext.conf-20060301.txt |
| 21:38.35 | *** join/#asterisk ibob63 (n=hp@bb-87-82-15-125.ukonline.co.uk) |
| 21:38.46 | Dr-Linux | Katty: sorry i didn't know you are a girl ;) |
| 21:38.57 | Katty | heh |
| 21:39.10 | Katty | why is it everyone thinks i'm a male? |
| 21:39.26 | Beirdo | because you have the courage to hang out in IRC? :) |
| 21:39.28 | Dr-Linux | Katty: i just slaped you bcoz your chat was kinda different :) |
| 21:39.31 | x86 | from the [local] context, i can call anything in the [tollfree-outbound] context, the [siptosip] context, or the [friends] context, but can not call the [toll] context... |
| 21:39.36 | Katty | Dr-Linux: well get used to it mister. |
| 21:39.36 | Ukyo | back |
| 21:39.39 | iDunno | Katty: I didn't! (but then, I couldn't think of any bloke that would have a nick of Katty...) |
| 21:39.44 | Katty | Dr-Linux: i've been on irc for 10 years, aint' changing now |
| 21:39.47 | Dr-Linux | Katty: its freenode though .. not dalnet :P |
| 21:39.54 | Katty | i've never been on dalnet. |
| 21:39.58 | Katty | my home is slashnet.org |
| 21:39.59 | fu3 | Dont |
| 21:40.04 | fu3 | it's DumbAssLamerNET |
| 21:40.08 | fu3 | :| |
| 21:40.08 | iDunno | dalnet kinda sucks, as does efnet. |
| 21:40.13 | Dr-Linux | Katty: then where have you been? |
| 21:40.14 | fu3 | efnet used to rule |
| 21:40.18 | iDunno | oftc is sanity |
| 21:40.18 | Katty | Dr-Linux: hmm? |
| 21:40.22 | fugitivo | dalnet is l4m3, efnet is 37337 |
| 21:40.23 | Katty | iDunno: i'm there too |
| 21:40.34 | iDunno | feenode is just, well, it's feenode |
| 21:40.36 | Katty | iDunno: moocows and linuxhelp |
| 21:41.01 | rene- | what was the make target for getting init.d scripts with zaptel? |
| 21:41.03 | iDunno | ohh, I'm in alug, debian-uk, and debian-devel on oftc ;) |
| 21:41.17 | Katty | all i can say is if anyone questions me being female, just show up at the cluecon convention |
| 21:41.33 | iDunno | and they'll get a clue? :) |
| 21:41.41 | Beirdo | hehe |
| 21:41.43 | gnosys_ | anybody here use snom hard phones? |
| 21:41.47 | Katty | and a slap across the face if they don't mind their manners, too |
| 21:41.52 | Dr-Linux | Katty: you are asterisk guru? :) |
| 21:42.00 | Katty | Dr-Linux: uhh, why else would i be in here? |
| 21:42.12 | Katty | Dr-Linux: for the hot guys and stimulating conversation? |
| 21:42.23 | Beirdo | hehe |
| 21:42.29 | Katty | Dr-Linux: yes, i run asterisk servers. |
| 21:42.33 | Dr-Linux | Katty: cool, i just said, bcoz i didn't see a word from you regarding asterisk |
| 21:42.35 | Katty | Dr-Linux: and occasionally kill them |
| 21:42.40 | Dr-Linux | i'm sorry, actually i'm newbie :) |
| 21:42.43 | *** join/#asterisk retroneo_ (n=retroneo@m234.net81-66-39.noos.fr) |
| 21:42.45 | Katty | Dr-Linux: i don't ask in here, it's futile |
| 21:42.54 | Katty | Dr-Linux: if i need answers, i go straight to the people that i know have them |
| 21:43.04 | iDunno | Katty: mmhmm. I can think of worse things ;) |
| 21:43.23 | Dr-Linux | Katty: yeah you can go to them .. but i can't :( :) |
| 21:43.24 | Katty | iDunno: deadlocking asterisk is always fun |
| 21:43.30 | Katty | what with co-workers running about panicing and such |
| 21:43.34 | rene- | Katty: were you at astricon anaheim? |
| 21:43.39 | iDunno | heh |
| 21:43.46 | Katty | rene-: no |
| 21:43.49 | Dr-Linux | Katty: bcoz the country i'm living no one knows about asterisk :) |
| 21:43.55 | *** join/#asterisk ToTo (n=ToTo@host43-130.pool874.interbusiness.it) |
| 21:43.57 | Katty | Dr-Linux: k |
| 21:44.10 | Katty | iDunno: i do too, if it's something simple |
| 21:44.11 | iDunno | (but google and the book are quicker, generally ;) |
| 21:44.15 | Katty | like....what's this trying to tell me, etc. |
| 21:44.32 | iDunno | uh huh |
| 21:44.33 | Katty | otherwise i'm bugging anthm or twisted :) |
| 21:44.45 | iDunno | that's cheating ;) |
| 21:44.55 | Katty | i never said i played fair. |
| 21:45.07 | iDunno | true, true :) |
| 21:45.15 | Katty | today i have a sick laptop. |
| 21:45.17 | Katty | it needs hugging |
| 21:45.24 | Katty | :< |
| 21:45.29 | Katty | i prescribed a format. |
| 21:45.35 | iDunno | but I get shiny new one tomorrow :) |
| 21:45.37 | Katty | :> |
| 21:45.42 | Katty | file as a nice laptop |
| 21:45.44 | Katty | izzoshiny. |
| 21:45.49 | synthetiq | how do u deadlock a server with out ruinging the code |
| 21:45.51 | iDunno | ahh - mines decided that it doesn't want power, which is lovely of it ;) |
| 21:45.54 | synthetiq | ruining |
| 21:46.01 | iDunno | so I bought a new one - a shiny new one :) |
| 21:46.09 | Katty | whoo! |
| 21:46.12 | Katty | file: give me your laptop |
| 21:46.42 | file | nope! |
| 21:46.45 | Katty | :< |
| 21:46.48 | Katty | file: give me a hug! |
| 21:46.53 | Katty | iDunno: nite. |
| 21:46.53 | Hmmhesays | we were parked down by the tracks and we just started gettin' bizzay when she whispered what was that |
| 21:47.01 | iDunno | Katty: night night :) |
| 21:47.20 | Katty | Hmmhesays: new song? |
| 21:47.24 | Hmmhesays | the wind I think cause no one else knows where we are, thats when started screamin' thats my dad out side the car |
| 21:47.24 | Chotaire | let me repost my question... sorry, I must get this stuff fixed, it's only one command I'm stuck with... |
| 21:47.28 | GerbilWrk | has anyone gotten queue position announcements working in 1.2.4? |
| 21:47.29 | Chotaire | exten => 5,2,Dial(SIP/soft-chotaire,10,gG(tp-pool1-control,5,3)D(www0)) |
| 21:47.32 | Chotaire | why would Dial command ignore D (dial dtmf ,,,0) when G is in use? |
| 21:47.38 | *** join/#asterisk dgorski (n=dgorski@c-69-245-111-167.hsd1.mi.comcast.net) |
| 21:48.11 | Hmmhesays | the keys oh please they're not in the ignition, musta wound up on the floor while we were switchin are position |
| 21:49.18 | grandy | hello... i set up * with nufone.net (sip) and for some reason inbound calling isn't working... i see no debug info when i call the number... i am successfully registered as a sip user and sip peer with nufone.... there appears to be no debug output from asterisk when i attempt to dial the number... watching the output with asterisk -vvvr |
| 21:49.39 | grandy | (any suggestions for how to debug this?) |
| 21:49.49 | justinu | grandy: turn on sip debug |
| 21:49.52 | Hmmhesays | sip debug |
| 21:49.56 | Hmmhesays | send in a call |
| 21:50.02 | grandy | did that |
| 21:50.20 | grandy | there doesn't appear to be any output in asterisk... only nat keepalives |
| 21:50.21 | Hmmhesays | you probably don't have anything in your dialplan for the context its getting dumped into |
| 21:50.59 | grandy | Hmmhesays: i register the sip user as context inbound and i have an inbound context in my dialplan that's supposed to saydigits(123) after it answers... |
| 21:51.56 | Hmmhesays | turn sip debug on |
| 21:51.59 | Hmmhesays | if you haven't already |
| 21:52.03 | grandy | it's on |
| 21:52.15 | Hmmhesays | ok, then its a nufone problem if you don't see anything the call isn't getting to you |
| 21:52.18 | GerbilWrk | is it possible nufone isn't sending it to you yet? |
| 21:52.21 | Druken | anyone has jitter problems on rogers express ? |
| 21:52.33 | grandy | GerbilWrk: it's possible, i guess, but i think they instantly provision stuff |
| 21:52.39 | Hmmhesays | sip express router is pissing me off today |
| 21:52.40 | Hmmhesays | ARGH |
| 21:52.50 | synthetiq | newbie |
| 21:53.04 | Hmmhesays | indeed |
| 21:53.07 | Katty | we were all newbies at one time |
| 21:53.12 | synthetiq | nextone resells ser with a rpetty gui for 40k$ |
| 21:53.18 | synthetiq | pretty |
| 21:53.25 | Hmmhesays | i don't want a pretty gui |
| 21:53.31 | Katty | why not? |
| 21:53.36 | Katty | they're curvacious! |
| 21:53.40 | glm2k | i just want things to work |
| 21:53.42 | Hmmhesays | I want to understand how it works |
| 21:53.43 | Katty | and resource hoggy! |
| 21:53.56 | Katty | and have wizards that don't work! |
| 21:54.01 | synthetiq | web gui |
| 21:54.02 | _Paulo_ | does * support the brooktrout tr114 board? |
| 21:54.10 | docelm0 | Anyone know of a way to get rid of the resource limit in linux? I keep running out of open files |
| 21:54.11 | Hmmhesays | i dunno the latest amp is pretty tyte |
| 21:54.25 | Hmmhesays | go down to home depot and buy some more file descriptors |
| 21:55.11 | grandy | btw just noticed that it * is saying Destroying call ..... after I place a call... is there a way to find out what is happening to it? |
| 21:55.12 | docelm0 | Hmmhesays, dude.. you are just not funny.. |
| 21:55.25 | docelm0 | Seriously.. I tried the whole ulimit deal and it didnt seem to do much for me |
| 21:55.29 | Hmmhesays | docelm0: I thought it was pretty funny |
| 21:55.38 | Hmmhesays | anyhoo, are you sure you did it right? |
| 21:56.34 | Katty | hi docelm0 (= |
| 21:57.27 | Hmmhesays | yeah my |
| 21:57.34 | Hmmhesays | my SER is seriously freaking out |
| 21:57.48 | FlyboySR22 | docelm0, WHat kernel..? |
| 21:58.07 | dgorski | <PROTECTED> |
| 21:58.07 | dgorski | <PROTECTED> |
| 21:58.07 | dgorski | <PROTECTED> |
| 21:58.07 | dgorski | <PROTECTED> |
| 21:58.15 | dgorski | that's how I did it |
| 21:58.18 | docelm0 | FlyboySR22, byte me |
| 21:58.43 | docelm0 | dgorski, where did you put this information? |
| 21:58.49 | docelm0 | the safe_asterisk? |
| 21:58.58 | dgorski | I run gentoo, it's in my /etc/init.d/asterisk startup script |
| 21:59.03 | *** part/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
| 21:59.06 | FlyboySR22 | docelm0, OK, but if you want to know how to open up the number of files Linux can use, I need to know the kernel.... |
| 21:59.14 | dgorski | it inherits ATERISK_NOFILES from /etc/conf.d/asterisk |
| 21:59.15 | dgorski | I think |
| 21:59.15 | docelm0 | What did you specify for the number of open files? |
| 21:59.22 | dgorski | yep |
| 21:59.29 | dgorski | # |
| 21:59.29 | dgorski | # open file descriptors |
| 21:59.29 | dgorski | # |
| 21:59.29 | dgorski | ASTERISK_NOFILES="65535" |
| 21:59.44 | docelm0 | ok kewl.. I will make this change.. |
| 22:00.09 | dgorski | let me know if it worked for you. |
| 22:00.19 | dgorski | the other ways I tried it are supposed to work but did not |
| 22:01.08 | ibob63 | newbie problem: I am stuck at the beginning and can't work out why this config doesn't work - http://pastebin.com/579155 can someone put me out of my misery. |
| 22:01.38 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 22:01.42 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 22:02.29 | Hmmhesays | ibob63: you can't directly dial an s extensions dude |
| 22:03.01 | Hmmhesays | add this to your internal context exten => _X.,1,Goto(s,1) |
| 22:03.39 | docelm0 | FlyboySR22, I know how to open them.. Im more interested in knowing the correct procedure to set it so asterisk can use the new setting |
| 22:03.57 | FlyboySR22 | docelm0, here is a good article on Increasing the number of open file descriptors.....http://bcr2.uwaterloo.ca/~brecht/servers/openfiles.html |
| 22:04.18 | Katty | i love it when people never say hi back. |
| 22:04.20 | FlyboySR22 | docelm0, Sorry man, thought you were talking about open file descriptors in general... |
| 22:04.49 | dgorski | I wasn't able to get sysctl.cf work work |
| 22:05.03 | ibob63 | HmmHesays: thanks - tried that and the phone still just beeps at me :( |
| 22:05.30 | *** join/#asterisk konfuzed (n=Konf@H135.C72.B0.tor.eicat.ca) |
| 22:05.37 | konfuzed | slePP, yo |
| 22:05.37 | docelm0 | Im not a novice.. I am dCAP.. :) |
| 22:06.07 | docelm0 | MEW!!!! |
| 22:06.35 | Katty | docelm0: what took you so long? |
| 22:07.21 | *** join/#asterisk saftsack (n=saftsack@p54A7F36B.dip.t-dialin.net) |
| 22:07.23 | saftsack | hi |
| 22:07.29 | docelm0 | Im pullin my hair out.. busy as hell |
| 22:07.32 | saftsack | are some isdn experienced people here? |
| 22:07.46 | Katty | docelm0: :< |
| 22:07.56 | docelm0 | yes.. Im going home however.. will be back on in about 30 minutes |
| 22:07.58 | Katty | docelm0: i'd recommend getting some tlc from the wife. |
| 22:08.05 | docelm0 | Not married.. |
| 22:08.12 | docelm0 | or getting laid.. TOOO DAMN busy.. |
| 22:08.13 | docelm0 | :( |
| 22:08.16 | Katty | then the girlfriend |
| 22:08.21 | Hmmhesays | argh, why is SER sending back an unauthorized, i'm not trying to auth anything |
| 22:08.23 | docelm0 | dont have one |
| 22:08.26 | Katty | what?! |
| 22:08.28 | justinu | ack |
| 22:08.29 | Katty | slacker, hop to it |
| 22:08.49 | Hmmhesays | you can be the ugliest moron in the world and still get laid |
| 22:08.54 | synthetiq | newb |
| 22:09.10 | synthetiq | yea u just got to read the don juan manual like hmmhesays did |
| 22:09.16 | Hmmhesays | so synthetiq, help me out |
| 22:09.26 | Katty | synthetiq: he doesn't need a manual, deary |
| 22:09.27 | dgorski | saftsack: ask away |
| 22:09.27 | Hmmhesays | i'm getting the 10lb sledge ready for this thing |
| 22:09.30 | synthetiq | 150$/hr sir |
| 22:09.32 | dgorski | you'll see |
| 22:09.45 | Hmmhesays | heh |
| 22:09.54 | saftsack | dgorski, ask away? |
| 22:09.56 | Katty | Hmmhesays: i wish you came with a readme file |
| 22:10.10 | Hmmhesays | not sure what it would say Katty |
| 22:10.13 | Hmmhesays | run awayyy |
| 22:10.22 | Katty | probably ;) |
| 22:10.38 | synthetiq | you are not having the too many hops probkem |
| 22:10.42 | *** join/#asterisk mzo (n=moz@ool-435193b3.dyn.optonline.net) |
| 22:10.44 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 22:10.48 | mzo | now paging websae :P |
| 22:10.50 | synthetiq | ? |
| 22:12.20 | *** join/#asterisk drbrown_ (n=keith@65.121.240.66) |
| 22:13.37 | Hmmhesays | I got something funky in my setup here |
| 22:13.41 | Hmmhesays | I'll find it |
| 22:14.56 | *** join/#asterisk kratzers (n=kratzers@65.119.217.4) |
| 22:15.29 | Flauto | hmmhesays, i got the fwd peering thing to work now |
| 22:15.30 | Flauto | thanks |
| 22:15.38 | *** part/#asterisk synthetiq (n=roger@64.201.13.50) |
| 22:15.39 | kratzers | is there a way to use variables in queues.conf? |
| 22:15.45 | Hmmhesays | np Flauto |
| 22:16.03 | Flauto | it was my spa 3k setting |
| 22:17.10 | Hmmhesays | yeah |
| 22:17.12 | Hmmhesays | i know |
| 22:17.21 | drbrown_ | does anyone know if there is a way to detect all the sip phones in use for use in paging????? |
| 22:17.39 | pifiu | anyone know much about ISDN? |
| 22:18.09 | pifiu | whats the difference in between ISDN BRI service and ISDN PRI service? |
| 22:18.13 | kippi | if I want to re-install zaptel, do i have to remove it first? |
| 22:18.22 | kratzers | the number of channels |
| 22:18.28 | kratzers | and the bandwidth of the D channel |
| 22:19.06 | pifiu | ok good thats what iw ant to get at |
| 22:19.10 | pifiu | the D channel is usually how big? |
| 22:19.24 | dgorski | 16k |
| 22:19.26 | pifiu | im looking at some questions and it says |
| 22:19.30 | dgorski | http://www.cisco.com/univercd/cc/td/doc/cisintwk/ito_doc/isdn.htm |
| 22:19.33 | kratzers | yup, 16k |
| 22:19.35 | justinu | one full DS0 on PRI |
| 22:19.45 | dgorski | yep, PRI is 64k |
| 22:19.49 | pifiu | which of hte following technologies is characterized as having two 64Kbps B channels and one 16Kbps D channel? |
| 22:19.56 | dgorski | BRI |
| 22:20.06 | pifiu | so PRI will have a 64k channel and BRI will have a 16? |
| 22:20.15 | dgorski | yep |
| 22:20.21 | dgorski | you should read this, it's pretty good |
| 22:20.23 | kratzers | for D, yes |
| 22:20.23 | pifiu | and the D channel does what? |
| 22:20.23 | dgorski | http://www.cisco.com/univercd/cc/td/doc/cisintwk/ito_doc/isdn.htm |
| 22:20.31 | kratzers | typically used for signaling purposes |
| 22:20.34 | dgorski | signalling ("out-of-band") |
| 22:20.38 | pifiu | damn thats long |
| 22:20.39 | kratzers | but can often be configured for data if need be |
| 22:20.44 | pifiu | ill minimize it and read it later |
| 22:20.54 | pifiu | so its not used, its like a "management" port |
| 22:21.02 | kratzers | no |
| 22:21.05 | kratzers | to manage what? |
| 22:21.13 | dgorski | that's fair - it's how CallerID and ANI/DNIS/etc are passed |
| 22:21.17 | pifiu | link management? |
| 22:21.54 | kratzers | so... variables in queues.conf? anybody? |
| 22:21.54 | pifiu | but BRI would be used when? ive never heard of it, just always hear PRI |
| 22:22.01 | dgorski | yep, RING/ANSWER/DISCONNECT/ etc. too |
| 22:22.06 | kratzers | when you only need two channels |
| 22:22.15 | kratzers | and don't want to pay for a full DS1 |
| 22:22.16 | pifiu | is that often used? ive never seen it |
| 22:22.20 | pifiu | oh like home ISDN? |
| 22:22.27 | dgorski | usually used in the old days for dial-up |
| 22:22.31 | pifiu | yeah ok |
| 22:22.31 | dgorski | some use it for voice |
| 22:22.32 | pifiu | got it |
| 22:22.37 | pifiu | and basically its two phone lines |
| 22:22.46 | dgorski | yes, "home ISDN" is almost always BRI |
| 22:22.55 | dgorski | video-conf systems use it too |
| 22:22.57 | pifiu | i assume it would be cheaper to get just two phone lines? |
| 22:22.59 | pifiu | oh |
| 22:23.00 | pifiu | hmm |
| 22:23.03 | pifiu | ok ok |
| 22:23.07 | dgorski | probably, ISDN has costs |
| 22:23.10 | pifiu | but its really not used much now? |
| 22:23.14 | kippi | if I want to re-install zaptel, do i have to remove it first? |
| 22:23.15 | *** join/#asterisk angom (n=Administ@red-corp-200.38.16.10.telnor.net) |
| 22:23.20 | dgorski | but you can't get 64k out of a POTS line... |
| 22:23.26 | pifiu | you cant? |
| 22:23.26 | tzafrir_laptop | kippi, why? |
| 22:23.30 | dgorski | I don't use it... ;) |
| 22:23.33 | Hmmhesays | yeah i'm flarkin' retarded |
| 22:23.35 | pifiu | i thought POTS was 64? |
| 22:23.39 | pifiu | oh no its 53 |
| 22:23.42 | kratzers | and most ISPs don't multiplex dial-up PPP links |
| 22:23.44 | kratzers | that I know of |
| 22:23.50 | tzafrir_laptop | kippi, why re-install? for an upgrade? |
| 22:24.01 | dgorski | no gurantee - you're 56k modem doesn't really get 56k |
| 22:24.06 | kippi | tzafrir_laptop: my zaptel is playing up, its not picking up my card |
| 22:24.06 | dgorski | it's asymmetric |
| 22:24.14 | kippi | and asterisk won't start |
| 22:24.18 | pifiu | well the maximum line transfer on a POTS is 53 |
| 22:24.20 | tzafrir_laptop | kippi, what card? |
| 22:24.25 | kippi | TE110P |
| 22:24.45 | tzafrir_laptop | no problem building from the same dir |
| 22:24.50 | dgorski | kippi: what does ztcfg -vv do? |
| 22:24.55 | GerbilWrk | has anyone gotten queue position announcements working in 1.2.4? |
| 22:25.06 | tzafrir_laptop | Just make sure you stop asterisk before you rmmod modules |
| 22:25.24 | kratzers | queue announcements work for me |
| 22:25.49 | GerbilWrk | they worked for me till i upgraded |
| 22:25.51 | kippi | dgorski: Notice: Configuration file is /etc/zaptel.conf |
| 22:25.51 | kippi | line 0: Unable to open master device '/dev/zap/ctl' |
| 22:25.51 | tzafrir_laptop | Also keep in mind that after you have installed the new modules, they will only be used once loaded, so unload old ones |
| 22:25.54 | *** join/#asterisk YoMama (n=tchen@d14-69-186-121.try.wideopenwest.com) |
| 22:26.01 | YoMama | son of a motherless beyotch |
| 22:26.19 | kratzers | kippi, make sure /dev/zap/ctl exists |
| 22:26.21 | dgorski | kippi: sounds like the zaptel driver isn't loded (lsmod | grep zaptel) |
| 22:26.25 | YoMama | can someone please explain to me why my voicemail email notification would all of a sudden stop working? |
| 22:26.35 | kratzers | I have a problem after an install that made them /dev/zapctl or something |
| 22:26.37 | tzafrir_laptop | kippi, is zaptel loaded? |
| 22:26.38 | dgorski | look for the |
| 22:26.44 | kratzers | had* |
| 22:26.45 | dgorski | module for your card |
| 22:26.52 | pifiu | thanks for the help on ISDN, i will read that cisco document |
| 22:26.55 | kratzers | it misplaced the / in the pathname |
| 22:27.02 | YoMama | anyone tell me how i can see what * is doing with the voicemail notification? I logged into the CLI and didn't see any messages regarding VM notification |
| 22:27.03 | dgorski | pifiu: enjoy |
| 22:27.05 | dgorski | # lsmod | grep zaptel |
| 22:27.05 | dgorski | zaptel 180388 105 wct4xxp |
| 22:27.05 | dgorski | crc_ccitt 2176 1 zaptel |
| 22:27.12 | kippi | nothing |
| 22:27.14 | kratzers | I think I created the /dev/zap directory and created sym links under it |
| 22:27.30 | kippi | and /dev/zap/ctl no such dir |
| 22:27.39 | tzafrir_laptop | kratzers, do you use udev/devfs? |
| 22:27.54 | tzafrir_laptop | kippi: if you use udev, that is expected. et zaptel loaded |
| 22:28.31 | tzafrir_laptop | modprobe zaptel, that is |
| 22:28.33 | kratzers | yes, I use udev |
| 22:28.37 | kratzers | but that wasn't the issue |
| 22:28.55 | dgorski | modprobe wcte11xp maybe |
| 22:29.21 | kippi | wooo |
| 22:29.39 | dgorski | what OS ? |
| 22:29.46 | kippi | it works :) |
| 22:29.55 | kippi | now i'll reboot and make sure it works again |
| 22:30.04 | dgorski | before you do that |
| 22:30.19 | dgorski | check /etc/modules.conf and make sure you have the zaptel stuff in there |
| 22:30.26 | dgorski | otherwise you'll need to load by hand all the time |
| 22:30.27 | kippi | its rebooting |
| 22:30.32 | dgorski | something like: |
| 22:30.36 | dgorski | options torisa base=0xd0000 |
| 22:30.36 | dgorski | alias char-major-196 torisa |
| 22:30.36 | dgorski | alias wctdm wcfxs |
| 22:30.41 | tzafrir_laptop | dgorski, modules.conf is for kernel 2.4 . |
| 22:30.47 | dgorski | (at least that's what's on my box) |
| 22:30.58 | tzafrir_laptop | Most of it is old cruft |
| 22:31.02 | dgorski | advantage1 etc # uname -a |
| 22:31.02 | dgorski | Linux advantage1 2.6.12-gentoo-r9 #1 SMP Thu Sep 15 03:50:00 EDT 2005 i686 AMD Athlon(tm) 64 Processor 3000+ AuthenticAMD GNU/Linux |
| 22:31.13 | tzafrir_laptop | toris is practically not used anymore |
| 22:31.24 | dgorski | sure, some of that is old, but |
| 22:31.29 | tzafrir_laptop | the wctdm alias is not needed in 1.2 |
| 22:31.30 | dgorski | alias char-major-196 torisa |
| 22:31.35 | dgorski | that's the trigger |
| 22:31.37 | *** part/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
| 22:31.47 | *** join/#asterisk Dandan (i=dandan@ellie.pacanka.com) |
| 22:32.02 | kratzers | a/clear |
| 22:32.03 | *** join/#asterisk VirTERM (n=VirTERM@204.225.113.91) |
| 22:32.11 | tzafrir_laptop | dgorski, why do you need torisa? for what card? |
| 22:33.26 | *** part/#asterisk rene- (n=rene-@201.127.101.127) |
| 22:33.35 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 22:34.26 | YoMama | baaah |
| 22:34.37 | kippi | I don't have a /etc/modules.conf |
| 22:34.53 | dgorski | how about /etc/modprobe.conf |
| 22:34.54 | kratzers | /etc/modprobe.conf |
| 22:34.57 | kippi | yeah |
| 22:35.07 | SkramX | hey all |
| 22:35.18 | kippi | should i add it to that? |
| 22:35.24 | dgorski | # grep 196 modprobe.conf |
| 22:35.24 | dgorski | alias char-major-196-* torisa |
| 22:35.24 | tzafrir_laptop | All the aliases there are useless at best and harmful at worst |
| 22:35.43 | YoMama | hey SkramX |
| 22:35.59 | tzafrir_laptop | that should never be necessary with 2.6, unless the driver is really dumb |
| 22:36.03 | dgorski | please explain then - how does the module get loaded on your system |
| 22:36.08 | Chotaire | dudes... please help me finally.. I'm so stuck... Dial(SIP/soft-chotaire,10,gG(tp-pool1-control,5,3)D(www0)) <- is this the right format? |
| 22:36.22 | Chotaire | like G(x)D(x)? or would that be GD(x^x)? |
| 22:37.00 | dgorski | the kernel will not just try to load every module under /lib/modules |
| 22:37.06 | dgorski | so something triggers it |
| 22:37.51 | tzafrir_laptop | modules declare what PCI IDs they support. The kernel then has a list of modules for PCI IDs. And then the module ges loaded automatically |
| 22:38.12 | dgorski | I don't believe in automatically |
| 22:38.41 | tzafrir_laptop | If you're not using hotplug or similar, you can modprobe some modules explicitly. In debian you list them in /etc/modules |
| 22:38.57 | dgorski | hotplug |
| 22:39.00 | dgorski | there you go |
| 22:39.03 | dgorski | gentoo here |
| 22:39.09 | tzafrir_laptop | A simple mechanism that is sadly lacking from other distro |
| 22:39.10 | tzafrir_laptop | s |
| 22:39.14 | SkramX | go gentoo :) |
| 22:39.34 | dgorski | fair enough, I stand corrected |
| 22:40.03 | kippi | on redhat were do i need to add modprobe xxxx so it loads on bootup? |
| 22:40.07 | Chotaire | tzafrir: don't you have any idea regarding app_dial? |
| 22:40.13 | dgorski | so back to kippi's concern |
| 22:40.45 | dgorski | redhat |
| 22:40.55 | dgorski | sorry |
| 22:41.15 | dgorski | let me look at something |
| 22:41.52 | Dr-Linux | kippi: same problem with me, i'm using RH, it doesn't load modules on bootup |
| 22:42.02 | YoMama | anyone got experience troublehsooting voicemail? |
| 22:42.09 | Dr-Linux | i added a script, that runs on boot to load zaptel modules |
| 22:42.25 | SkramX | YoMama: whats the problem? |
| 22:42.30 | tzafrir_laptop | Chingetas, maybe options are separated with commas as well? |
| 22:42.40 | Dr-Linux | YoMama: ask your questio if someone knows, will answer you |
| 22:42.44 | YoMama | SkramX: my email voicemail notification just stopped working all of a sudden |
| 22:42.51 | tzafrir_laptop | Dr-Linux, that's what init scripts are for |
| 22:42.58 | SkramX | Dr-Linux: add it to /etc/modules.autoload.d/kernel-2.X |
| 22:43.03 | YoMama | i wanna turn some debugging on so i can see if it's actually sending out the email..i know it's not a firewall issue because i tested email sending manually and it worked |
| 22:43.06 | kratzers | maybe it's your mail server? |
| 22:43.12 | YoMama | kratzers: nope |
| 22:43.17 | YoMama | already tested that |
| 22:43.50 | SkramX | YoMama: mail server problem maybe? |
| 22:43.52 | SkramX | yeah- |
| 22:43.53 | SkramX | hrmm. |
| 22:44.03 | tzafrir_laptop | YoMama, did you check the logs of your local MTA? |
| 22:44.09 | tzafrir_laptop | which one is it? |
| 22:44.14 | YoMama | SkramX: did the voicemail.conf format change between 1.2.x and 1.1.x? |
| 22:44.24 | Chotaire | tzafrir, let me try. |
| 22:44.28 | SkramX | YoMama: basic stuff should still work |
| 22:44.38 | SkramX | did it stop working right after an upgrade/ |
| 22:45.04 | tzafrir_laptop | hmmm, an "svk mirror" command takes time... |
| 22:45.16 | YoMama | SkramX: it's hard to say since i don't get many calls..lemme monitor the maillog...and see what happens |
| 22:45.46 | dgorski | kippi: maybe you need to run depmod after building zaptel? |
| 22:45.49 | SkramX | aiight |
| 22:47.03 | dgorski | I don't see why it should need to be hardcoded it hotplug is supposed to handle it |
| 22:47.10 | dgorski | it->if |
| 22:47.21 | kippi | dgorski: trying that, just rebooting |
| 22:47.46 | Dr-Linux | SkramX: what should i put there in path you told to load modules? |
| 22:48.08 | *** join/#asterisk apardo (n=apardo@87.218.44.213) |
| 22:48.11 | Dr-Linux | SkramX: whats differnce if i put that in /etc/rc.local ? |
| 22:48.17 | x86 | can someone tell me what i'm doing wrong here: http://www.shellshark.net/pub/ast-ext.conf-20060301.txt |
| 22:48.20 | x86 | from the [local] context, i can call anything in the [tollfree-outbound] context, the [siptosip] context, or the [friends] context, but can not call the [toll] context... |
| 22:48.36 | SkramX | eh |
| 22:48.45 | *** join/#asterisk Psykick (n=anon@203.167.226.250) |
| 22:48.47 | SkramX | The requested URL /pub/ast-ext.conf-20060301.txt was not found on this server. |
| 22:48.49 | Psykick | hi guys |
| 22:48.53 | SkramX | x86: ? |
| 22:48.55 | x86 | err |
| 22:48.56 | x86 | hold |
| 22:49.05 | *** join/#asterisk iaxy (n=iaxy@modemcable236.55-131-66.mc.videotron.ca) |
| 22:49.13 | kippi | dgorski: nope that didn't work |
| 22:49.59 | *** join/#asterisk needs_help (i=Gir@67.189.110.174) |
| 22:50.06 | SkramX | haha. |
| 22:50.17 | gnosys_ | anyone here use snom hard phones? |
| 22:50.34 | Psykick | hey is there a channel for asterisk development? |
| 22:50.42 | x86 | err |
| 22:50.56 | x86 | http://www.shellshark.net/pub/ast-ext.conf.20060301.txt |
| 22:50.59 | justinu | i bet file wrote this line of code: |
| 22:51.00 | x86 | SkramX: |
| 22:51.05 | justinu | <PROTECTED> |
| 22:51.23 | SkramX | x86: one sec |
| 22:51.29 | Psykick | hi justinu |
| 22:51.31 | SkramX | Psykick: #asterisk-dev ? |
| 22:51.47 | justinu | yeah, asterisk-dev |
| 22:51.50 | Psykick | It's ok I'm in there ... |
| 22:52.13 | Psykick | I think there is a bug with the queues linked list |
| 22:52.23 | x86 | SkramX: anything in the [toll-outbound] context is supposed to match 8XXXXXXXXXX or 8XXXXXXXXXXX |
| 22:52.31 | Psykick | not sure its iterating through each member properly |
| 22:52.33 | x86 | SkramX: but i think it's not matching |
| 22:52.51 | SkramX | hrmmm |
| 22:52.53 | SkramX | ill look in a sec |
| 22:53.02 | SkramX | just got like 3 support tickets within a minute. |
| 22:53.11 | x86 | ouch heh |
| 22:53.16 | SkramX | yeah kinda. |
| 22:53.22 | *** join/#asterisk lorinc (n=ang@caracas-1562.adsl.interware.hu) |
| 22:53.39 | [av]bani | hmm.. digium ml archives are busted :| |
| 22:55.29 | *** join/#asterisk lorinc (n=ang@caracas-1562.adsl.interware.hu) |
| 22:55.32 | Chotaire | tzafrir: I really dunno what to do... |
| 22:55.34 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 22:55.42 | Chotaire | tzafrir: I tried everything... |
| 22:58.46 | Chotaire | isn't anyone here who has really played around with dial command? |
| 22:59.02 | Psykick | Chotaire: what's wrong? |
| 22:59.14 | Chotaire | exten => 5,2,Dial(SIP/soft-chotaire,10,gG(tp-pool1-control,5,3)D(www0)) |
| 22:59.20 | YoMama | bizarre |
| 22:59.21 | Chotaire | D will be ignored when G is in use |
| 22:59.29 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 22:59.30 | Chotaire | bad format? |
| 22:59.35 | Psykick | possibly |
| 22:59.50 | Psykick | let me look at the docs a lil more closely just to double check |
| 22:59.54 | Chotaire | k |
| 23:00.18 | Chotaire | D alone will work... when D and G are in use, no matter what the order is or whatever strange format I tried, D will be completely ignored. |
| 23:01.50 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-62-67.cybersurf.com) |
| 23:02.37 | Psykick | Chotaire: I suggest you read the G parameter again |
| 23:02.46 | Psykick | Chotaire: If the call is answered, transfer both parties to the specified priority; however it seems the calling party is transferred to priority x, and the called party to priority x+1 (new in v1.2) |
| 23:03.13 | *** join/#asterisk Micc (n=dotirc@c-24-16-228-130.hsd1.wa.comcast.net) |
| 23:03.51 | mzo | can someone help me debug an outbound dialing problem? |
| 23:04.07 | mzo | i have a trunk defined over iax, and it's connected, but when i dial it just gives me a 404, did i bug the contexts somewhere? |
| 23:04.17 | Psykick | I'm assuming that your D option is supposed to be your what is executed after if G doesn't respond |
| 23:04.18 | Chotaire | psykick: that is not the problem. |
| 23:04.34 | Psykick | ok ... can you go into more detail |
| 23:04.37 | *** join/#asterisk _Raptor_ (i=sirasenn@faui08r.informatik.uni-erlangen.de) |
| 23:04.39 | _Raptor_ | hi |
| 23:04.48 | Chotaire | the problem is D is not executed at all.. and why should D be executed if no party answers? that makes no sense sending dtmf when nobody picks up |
| 23:05.20 | Psykick | that's what I thought ;) |
| 23:05.25 | Chotaire | after pickup, it shall send the DTMF and THEN transfer both parties to the appropiate contexts (while priority+1 is just perfect btw) |
| 23:06.17 | *** join/#asterisk WolfsDen (n=hawkinss@c-68-48-88-52.hsd1.md.comcast.net) |
| 23:06.28 | Chotaire | D will not be used if M or G is in use.. and M will not be used if G is in use... is that a bug? |
| 23:06.38 | WolfsDen | is there a way to make an outbound call from a meetme conference? |
| 23:06.40 | Chotaire | or am I too stupid for the foramtting? |
| 23:06.47 | Chotaire | wolfsden: yes |
| 23:06.57 | _Raptor_ | i have the problem with my sip.conf. i want my asterisk to act as sip proxy. for instance i have in my extensions.conf exten => raptor,1,Dial(SIP/raptor) but this gives me a 404 with xtenlite. 123,Dial(SIP/raptor) works. any ideas? |
| 23:06.59 | WolfsDen | how would i set that up in the dialplan? |
| 23:07.11 | syzygybsd | WolfsDen: yes, originate a local call |
| 23:07.20 | Chotaire | why local? |
| 23:07.28 | Chotaire | you can trigger anything you like. |
| 23:07.48 | WolfsDen | chotair. how would i trigger the outbound call? |
| 23:07.53 | syzygybsd | yes, I did local so i could manage the call seperatly from the other members of the conference easily... |
| 23:07.56 | *** join/#asterisk fjean (n=fjean@201009186209.user.veloxzone.com.br) |
| 23:08.04 | fjean | hello! |
| 23:08.05 | Chotaire | wolfsden: |
| 23:08.05 | Chotaire | <PROTECTED> |
| 23:08.05 | Chotaire | <PROTECTED> |
| 23:08.05 | Chotaire | <PROTECTED> |
| 23:08.07 | Chotaire | that is your friend |
| 23:08.31 | rayvd | Friends are my foes! |
| 23:08.45 | syzygybsd | keep you friends close and your enemies closer |
| 23:08.55 | Chotaire | and Dial definitely IS my enemy. |
| 23:09.03 | dArF_AST | hello |
| 23:09.08 | Chotaire | and it seems nobody can help. |
| 23:09.22 | WolfsDen | I don't understand. right now i have it setup so that i have users dial an extension for a conference. |
| 23:09.33 | *** join/#asterisk Tamarisk (n=adrian@user-5384.lns5-c8.dsl.pol.co.uk) |
| 23:09.35 | Psykick | Chotaire: try doing this -> exten => 5,2,Dial(SIP/soft-chotaire,10,gG(tp-pool1-control,5,3) & D(www0)) |
| 23:09.36 | fjean | anybody knows how to find out why an IAX2 route/agi script doesn't give ring back ? I am out of clues here, hehe |
| 23:09.36 | dArF_AST | i have question about making conference call on ip telephone |
| 23:09.43 | Chotaire | psykick: trying... |
| 23:09.50 | Chotaire | wolfsden: wait, I'll get back to you |
| 23:09.58 | Chotaire | wolfsden: you just need a bit creativity |
| 23:10.01 | WolfsDen | i would like to be able to press a key, say *, followed by the outbound number. |
| 23:10.10 | *** join/#asterisk jbenson (n=jbenson@87.194.2.120) |
| 23:10.12 | dArF_AST | every ip telephones and adapters must use this same codec ? |
| 23:10.15 | WolfsDen | this is all in meetme ofcourse |
| 23:10.18 | Chotaire | look at option X then. |
| 23:10.54 | Chotaire | if you don't find out yourself how to do that I will give you a hint |
| 23:11.02 | Mavvie | hah! they will call back tomorrow! |
| 23:11.56 | dArF_AST | true or false? |
| 23:11.57 | dArF_AST | :) |
| 23:12.16 | Chotaire | psykick: no effect |
| 23:12.19 | Chotaire | D is ignored |
| 23:12.20 | Psykick | hmmm |
| 23:12.39 | Psykick | perhaps specify it in step 3 |
| 23:12.54 | WolfsDen | chotaire, i need that hint |
| 23:13.17 | Psykick | chotaire: can you paste dial plan to pastebin.ca |
| 23:13.23 | dArF_AST | or what should i do to make 3-way call on phone when every person use different codec? |
| 23:14.16 | *** join/#asterisk veepster (i=veepster@c-69-143-163-86.hsd1.va.comcast.net) |
| 23:14.47 | *** join/#asterisk ManxPower (n=ewieling@stirprop-S4-0-0-21.ndcr2.datasync.net) |
| 23:15.04 | syzygybsd | WolfsDen: why don't you set it up so the user exits to conference, dials the number, then enters the conference after the number is entered |
| 23:15.41 | Psykick | chotaire ... by the looks of things ... that g option will continue to execute further commands ... so what I'm assuming would be to use the g option ... step 3 specify your G(tp-pool1-control,5,4) with the D(www0) |
| 23:15.58 | Chotaire | Psykick: http://pastebin.com/579288 |
| 23:16.11 | WolfsDen | syzygybsd, how would i do it that way? |
| 23:16.28 | Chotaire | ..and reload... |
| 23:19.10 | gnosys_ | anyone here use Snom hardphones? |
| 23:19.12 | *** join/#asterisk Vyeperman (n=Vye@ip68-8-174-154.sd.sd.cox.net) |
| 23:19.16 | Chotaire | psykick: reload again.. I added a comment |
| 23:19.17 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 23:19.20 | Psykick | Chotaire: why not just move the D(www0) to the tp-pool1-control context? |
| 23:19.50 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 23:20.26 | Chotaire | psykick: check the pastebin please.. I dunno what you're saying since we ARE in that context, which is now called "whatever-context" |
| 23:20.37 | warthawg | anyone using zultys phones? I have a config issue that makes them seem busy when called by other extensions. |
| 23:22.23 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 23:23.08 | Chotaire | psykick: transferring call BEFORE sending dtmf is no option, it will break. |
| 23:23.45 | Psykick | ok |
| 23:23.52 | Chotaire | call may not be transferred before dtmf is sent. and both caller and callee must be transferred to different priorities, I dunno any other way besides Dial with G to do that. |
| 23:24.11 | Psykick | Chotaire: just noticed someone else is trying to do something similar but without trying to send DTMF tones |
| 23:24.20 | Chotaire | dtmf tones are implicit |
| 23:24.24 | Psykick | apparently they have a workaround |
| 23:24.39 | Chotaire | workaround? dtmf is the only thing that doesn't work. |
| 23:24.58 | Chotaire | it's all about the dtmf ;) |
| 23:25.14 | Psykick | where are the DTMF tones supposed to be getting sent to? |
| 23:25.21 | Psykick | both parties? |
| 23:25.34 | Chotaire | to the callee |
| 23:25.41 | Chotaire | just the way D() works |
| 23:25.47 | Chotaire | D() will not work with G() |
| 23:25.57 | warthawg | what happens on rtp ports? |
| 23:26.20 | Chotaire | ? |
| 23:26.21 | warthawg | can multiple extensions use the same starting rtp port? |
| 23:26.33 | YoMama | k..what the hell....somehow...my voicemail email notifiers are ending up in root's mailbox |
| 23:27.30 | Chotaire | psykick: if we get rid of G option, everything is perfect.. just that I need to transfer both the caller and callee to different extensions afterwards. |
| 23:27.53 | Chotaire | so if anyone would know a trick about that without using Dial's G(), then someone please give me a hint. |
| 23:28.16 | Psykick | Chotaire: well why not do exten => 5,2,Dial(SIP/soft-chotaire&SIP/callee,10,gG(tp-pool1-control,5,3)) |
| 23:28.43 | Chotaire | a) where is the dtmf? |
| 23:28.49 | Psykick | oh sorry |
| 23:29.12 | *** part/#asterisk Homer99 (n=homer@67.128.26.42) |
| 23:29.15 | WolfsDen | can someone post their dialplan so i can see how to dial out on a meetme conference? |
| 23:29.19 | Chotaire | b) why call two people if only one person is to be called? ;) |
| 23:29.44 | Psykick | ok how bout you layout what you want to do and maybe I can be of more help .. |
| 23:30.06 | Psykick | cos this is getting a lil confusing ... ... I'm in 4 conversations |
| 23:30.12 | warthawg | i want to be able to dial my zultys phone extensions |
| 23:30.14 | Psykick | 3 different channels |
| 23:30.16 | Chotaire | number gets dialed, after connection dtmf string is sent, and both caller and callee get transferred to different extensions. |
| 23:30.59 | ManxPower | Chotaire, Why? If there is no communication between the two calls then there is no reason to dial them as one call. Use a .call file. |
| 23:31.18 | Chotaire | a call file can be triggered by an extension? |
| 23:31.40 | Chotaire | hm yes... |
| 23:31.46 | Chotaire | system command or whatever that was... |
| 23:32.30 | Psykick | Chotaire: perhaps your just missing a , (comma) before the D(www0) |
| 23:32.38 | syzygybsd | WolfsDen: http://pastebin.ca/44161 |
| 23:32.57 | Psykick | because that is sent to both caller and callee |
| 23:33.09 | Psykick | which is what I'm assuming your wanting to do |
| 23:33.49 | Chotaire | no, D is always sent to callee only |
| 23:34.10 | Psykick | so is the optional URL parameter |
| 23:34.24 | Psykick | Chotaire: The optional URL parameter will also be sent to the called party upon successful connection, if the channel technology supports the sending of URLs in this way. |
| 23:34.57 | Chotaire | putting a comma infront of D(www0) makes no sense, it's not a URL ;) |
| 23:35.08 | Chotaire | anyway... .call file time... |
| 23:35.11 | Psykick | try it |
| 23:35.22 | _Raptor_ | can someone plz call sip:test@icip.de |
| 23:35.24 | Chotaire | I did |
| 23:35.28 | Chotaire | didn't work ofcourse |
| 23:35.31 | Psykick | ok |
| 23:35.42 | *** part/#asterisk Tamarisk (n=adrian@user-5384.lns5-c8.dsl.pol.co.uk) |
| 23:36.33 | grandy | Question: what are the sections in [brackets] called in sip.conf? contexts? channels? |
| 23:36.50 | _Raptor_ | account names |
| 23:37.05 | Chotaire | manxpower: how can I make the callee (dialed through call file) get transferred to a specified extension after the dtmf diggits were sent? is that app_transfer ? |
| 23:37.32 | grandy | _Raptor_: i see... i'm trying to set up asterisk with nufone.net via SIP and the register directive they provide appears only to handle one inbound number... |
| 23:38.19 | *** part/#asterisk fjean (n=fjean@201009186209.user.veloxzone.com.br) |
| 23:38.19 | grandy | _Raptor_: i can't seem to figure this out... nufone.net has "devices" and "numbers"... I don't understand what "devices" is for... any ideas? |
| 23:38.27 | *** join/#asterisk Eggplant (i=No@dsl-859.cascadeaccess.com) |
| 23:38.32 | _Raptor_ | grandy: nope |
| 23:38.38 | grandy | anyone? |
| 23:38.44 | glm2k | grandy: devices might be an ATA |
| 23:39.01 | grandy | glm2k: ATA? |
| 23:39.08 | Chotaire | manxpower: because that is actually the problem.. after the callee picks up and the dtmf string is sent the callee must be transferred to an extension. |
| 23:39.12 | glm2k | grandy: that's a unit that sits at your place, is connected to your broadband and has a DID assigned from nufone |
| 23:39.33 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 23:39.33 | Chotaire | and voila.. I'm with the same shit... use option G and D at the same time. |
| 23:39.41 | grandy | glm2k: hmm... i don't think that's it... maybe if i paste in the sample config for a "device" it will make sense... hang on one second |
| 23:39.50 | glm2k | not here |
| 23:39.57 | glm2k | pastebin.ca |
| 23:40.01 | grandy | i know... pulling that up |
| 23:40.04 | glm2k | kk |
| 23:40.45 | WolfsDen | syzygybsd, 1 question. what does the orginate line in addmember context do? |
| 23:40.53 | grandy | http://pastebin.ca/44163 |
| 23:40.57 | grandy | glm2k: there it is |
| 23:41.29 | Psykick | sorry I couldn't help Chotaire |
| 23:41.38 | syzygybsd | after you change it, it would originate the call, ie dial the caller and put one end in the meetme room |
| 23:42.15 | glm2k | grandy: ah. ok. i don't use the register line |
| 23:42.37 | grandy | glm2k: you'll notice that there's no register line in there... |
| 23:43.04 | grandy | glm2k: oddly under the "numbers" section it has other config stuff that does work for inbound, but i'm trying to figure out what the difference between "devices" and "numbers" is... |
| 23:43.20 | glm2k | grandy: lol! i must stilll be groggy. lol. yep. stand corrected. |
| 23:43.48 | grandy | glm2k: np... do you have any idea what the devices config is actually doing? |
| 23:44.26 | Chotaire | psykick: np, it seems nobody can help.- |
| 23:44.42 | *** join/#asterisk stormfr (n=StorM@bluegix-213-161-221-2.adsl.frontier.fr) |
| 23:44.53 | glm2k | grandy: peer fields are used to authenticate so you can make a call |
| 23:45.22 | grandy | glm2k: ahh... ok... |
| 23:45.33 | ManxPower | Chotaire, Um, you can do that with .call files. |
| 23:45.52 | stormfr | hello, is anybody know what this means : "chan_sip.c: Unable to build sip pvt data for" |
| 23:46.15 | ManxPower | Heck, the custom voicemail notification script I wrote ages ago, generates a call file to call an agent's cell phone, when the agent picks up the call is sent to an extension in the dialplan to let them check their voicemail |
| 23:46.46 | Chotaire | and that would be able to send dtmf to the CALLEE? |
| 23:47.23 | Chotaire | SendDTMF will do that trick? |
| 23:47.36 | ManxPower | Chotaire, Um, a .call file can dial a number, but it can also send the call to a specific extension when the call picks up. senddtmf should work, |
| 23:48.03 | ManxPower | only 10 more mins until I can take down the office phones to replace the channel banks. |
| 23:48.11 | ManxPower | maybe I won't be here for hours and hours afterall. |
| 23:48.12 | *** join/#asterisk naturalblue (n=Administ@87.192.100.109) |
| 23:48.38 | naturalblue | hey there |
| 23:48.55 | naturalblue | i got an interesting problem if anyone might give a hand out |
| 23:48.58 | *** part/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 23:49.28 | naturalblue | the text in most of my CLI is garbled |
| 23:49.44 | naturalblue | giving weird characters intstead of actual text |
| 23:50.03 | *** part/#asterisk angom (n=Administ@red-corp-200.38.16.10.telnor.net) |
| 23:51.24 | Chotaire | manxpower: how should it send the dtmf if it transfers the call to another extension before doing so? |
| 23:51.51 | *** join/#asterisk lesouvage (n=lesouvag@82.74.19.41) |
| 23:51.55 | Chotaire | ehrm wait... |
| 23:52.00 | Chotaire | let me try before I talk.. |
| 23:52.19 | Mavvie | hmmm... wonder if the refusal of AGI->get_data to read my DTMF keys is related to the problem I have with a SIP uplink to recognize my DTMF keys. |
| 23:53.10 | ManxPower | Mavvie, Prolly |
| 23:53.41 | Mavvie | ManxPower: that's what I think, but the first one is zap-to-zap, while the second one is zap-to-sip. |
| 23:53.58 | Mavvie | this is very tricksy |
| 23:54.10 | ManxPower | zap to zap DTMF issues are almost ALWAYS 1 of 2 things |
| 23:54.20 | naturalblue | has anyone any idea why the text would have be garbled/have foreign characters |
| 23:54.47 | Mavvie | naturalblue: make a screenshot and post it somewhere so we can see what you mean. |
| 23:54.47 | ManxPower | 1) the gains are too loud or too soft or 2) the DTMF length Asterisk uses is too short (defaults to 100ms) |
| 23:55.28 | ManxPower | The same issues with gains apply to CLID in additon to DTMF |
| 23:55.35 | ManxPower | But that's less common. Usually it either works or it doesn't. |
| 23:56.00 | naturalblue | whats the pastebin website again |
| 23:56.10 | Mavvie | ManxPower: I don't have gains (at least I don't think I need to worry about them) with my PRIs. |
| 23:56.28 | Chotaire | manxpower: goddamnit, it works. |
| 23:57.02 | Chotaire | now I have to automate it... I can do that myself. |
| 23:57.07 | Chotaire | if it really works the way it should, I will obey! |
| 23:57.11 | ManxPower | Mavvie, Weird, usually it's an issue with analog. |
| 23:57.34 | ManxPower | Chotaire, Feel free to send some cash via paypal to eric@fnords.org |
| 23:57.36 | Mavvie | I'll fire up a test machine first to find out what happens. |
| 23:57.49 | Mavvie | can't be bothered waiting for 10 hours until it's after hours :-) |
| 23:58.20 | naturalblue | Mavvie: sorry about this but it turns out to be only on the tty9 screen, if i log in with another terminal its fine |
| 23:58.23 | ManxPower | Mavvie, Is the problem with receiving DTMF from somewhere to Asterisk, or the problem is sending DTMF from Asterisk to someplace. Also what veriosn of Asteirsk? |
| 23:58.24 | naturalblue | i can live with that |
| 23:58.37 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 23:58.45 | *** join/#asterisk mcreedjr (n=mcreedjr@cblmdm72-240-21-51.buckeyecom.net) |
| 23:59.07 | ManxPower | naturalblue, perhaps tty9 (serial?) has the wrong parity ot stop/start bits? |
| 23:59.16 | *** join/#asterisk R3DB0x (i=nobody@66.142.28.36) |
| 23:59.22 | Mavvie | ManxPower: it's both: one case is an AGI script which needs to wait for them, the second one is a SIP uplink which doesn't accept them (but sees them in the RTP stream as telephony-events) |
| 23:59.36 | Mavvie | it's 1.2, of last tuesday. |
| 23:59.38 | mcreedjr | Hey all, I'm having some problems with voice quality over the 'net.. I tested my connection at www.testyourvoip.com and had a sizeable amount of packet discards to the destination. Is this a result of jitter? |
| 23:59.42 | *** join/#asterisk DeeJay[2] (n=bleh@37-179.sh.cgocable.ca) |
| 23:59.45 | ManxPower | Maveric, I was referring to the zap/zap stuff |
| 23:59.52 | Mavvie | ManxPower: don't worry too much about it, I'll figure it out. |
| 23:59.57 | naturalblue | mavvie: im actually at a physical connection to it so i thought it should be fine but obviously theres an error in its config or something |