| 00:00.29 | CrashSys | I still like the idea of it being a "defined standard" when it has nice large openly interpretive sections in it... |
| 00:00.52 | CrashSys | reminds me of goverment law using terms like "Quick" or "Expediantly" or "Within reason" |
| 00:01.12 | jaytee | [TK]D-Fender, I think I'm at the point where I'm going to give up on the Valcom PA system and try Viking's product. It works with FXS from an ATA. I got the SPA3102 working with a POTS line and could dial out fine. I got the FXS port working fine for an extension but the damn FXO port when it's plugged into the PA acts like it's always off-hook and throws a 503 error - circuit busy. Since it works with a Digium FXO port on a TDM04B card I have to |
| 00:01.12 | jaytee | <PROTECTED> |
| 00:01.22 | [TK]D-Fender | CrashSys: Where each is completely flexible in definition |
| 00:02.17 | drmessano | jaytee: What about just checking the line voltage in the SPA3102s status page? |
| 00:02.27 | drmessano | It can be tempermental |
| 00:02.39 | jaytee | it's at 19V when it indicates on-hook |
| 00:02.46 | drmessano | That's no goo |
| 00:02.47 | drmessano | That's no good |
| 00:03.36 | *** part/#asterisk harryjr (n=harryjr@67-207-147-205.slicehost.net) |
| 00:03.49 | CrashSys | 48v = talk |
| 00:03.52 | jaytee | drmessano, that's when it's plugged into a Valcom V-2001A PA paging amplifier system which works just fine with a Digium FXO port. |
| 00:04.13 | drmessano | Great, except the SPA-3102 will have a problem with that |
| 00:04.21 | drmessano | You need to adjust it in the SPA-3102 |
| 00:04.29 | drmessano | Trust me.. been there done that |
| 00:04.33 | jaytee | yeah, when I try to call the PA it acts like it's busy |
| 00:05.12 | CrashSys | well, it could be trying to do 24v talk battery, but that's pretty old-school/low/unsupported by most stuff :) |
| 00:05.37 | drmessano | I had a fax switch running 29 volts that I had fits with |
| 00:06.00 | drmessano | 3102 wouldn't see the line do shit |
| 00:06.37 | drmessano | I never did get it perfect... just better |
| 00:06.51 | CrashSys | and ringer voltage is 70-100v AC :D |
| 00:07.01 | jaytee | drmessano, I see alot of configurable parameters for the FXO port in the International section of the PSTN line tab and line in use voltage is set to 30 but I can't see anything to adjust for on-hook voltage. |
| 00:07.17 | CrashSys | that's always fun working in a BET and get shocked by ringer |
| 00:07.27 | drmessano | What is the off-hook? |
| 00:07.49 | jaytee | off-hook isn't on the config page |
| 00:07.56 | drmessano | No |
| 00:08.18 | drmessano | Can you put a splitter on the line, pick up a pots phone on the circuit, and watch the config page? |
| 00:08.46 | jaytee | drmessano, yeah but I'd have to drive back to the office to do it. I got fed up and came home. |
| 00:08.56 | drmessano | Oh ok |
| 00:09.19 | CrashSys | what is the run length between the ATA and phone? |
| 00:09.20 | drmessano | I'm guessing you're gonna see it reverse polarity at 19 volts |
| 00:09.31 | drmessano | which means your 30 wont work |
| 00:09.52 | CrashSys | or ATA and PA |
| 00:10.01 | CrashSys | not trying to pass it through 200' of house wiring are you? |
| 00:10.44 | jaytee | CrashSys, nope. Using a 12' line cord |
| 00:10.50 | CrashSys | Hmmm |
| 00:10.59 | *** join/#asterisk imcdona (n=imcdona@imcdona.broker.freenet6.net) |
| 00:11.11 | jaytee | loop current for the port is set to the default of 10ma |
| 00:13.17 | jaytee | tip and ring voltage adjust is 3.5v. On the info page where it shows On-hook and 19V it also shows loop current at 0.0 |
| 00:14.29 | CrashSys | Try increasing the loop current to 20ma |
| 00:14.46 | CrashSys | maybe you are sending out enough power to trigger the FXO on the PA |
| 00:15.00 | CrashSys | that would cause a low loop current too |
| 00:15.29 | jaytee | what about adjusting the Line In Use voltage from 30? |
| 00:15.51 | CrashSys | If the PA is using SLT ports then it probably has an old-school transformer hybrid and you have to send enough juice to saturate it before it'll work right |
| 00:16.22 | jaytee | hmmm, now I wanna go back in the office to try that :-) |
| 00:16.47 | *** join/#asterisk mateo_au (n=chatzill@12.144.159.231) |
| 00:17.00 | CrashSys | http://www.rane.com/note150.html |
| 00:17.30 | drmessano | jaytee, you said it thinks its always off-hook, right? |
| 00:18.00 | CrashSys | 10ma is pretty small for loop current |
| 00:18.13 | drmessano | and when it's on-hook you only get 19v |
| 00:18.16 | CrashSys | lack of current could definately cause on-hook VDC to drop |
| 00:18.20 | drmessano | and the linksys is set for 30 |
| 00:18.30 | drmessano | You need to drop the threshold a bit |
| 00:18.34 | jaytee | drmessano, I believe so. At least when I make a call it from a SIP phone I get call progress indication on the CLI in Asterisk and it then indicates 503 Service Unavailable - circuit busy |
| 00:18.44 | drmessano | I had the same problem |
| 00:21.28 | jaytee | drmessano, so CrashSys suggests bumping the loop current to 20ma instead of 10 but how much should I drop the Line In Use voltage from 30? |
| 00:22.07 | drmessano | So less than 19 :) |
| 00:23.05 | drmessano | I would leave the loop current alone |
| 00:24.08 | jaytee | the PA unit has a battery switch to supply loop current that needs to be ON when plugged into a Digium FXO but I tried it both ON and OFF with the SPA3102 with no change. |
| 00:24.19 | drmessano | Its the voltage |
| 00:24.24 | drmessano | Forget the loop current |
| 00:24.31 | drmessano | The linksys expects 30+ |
| 00:24.35 | jaytee | so 16V sound good? |
| 00:24.35 | drmessano | You have -30 |
| 00:24.38 | drmessano | yes |
| 00:25.10 | jaytee | ok, I'm definitely gonna drive back in and give that a shot. |
| 00:25.46 | jaytee | drmessano and Crashsys, thanks both of you for your suggestions. I'll let ya know if it works if you're still here in an hour or so. |
| 00:25.57 | CrashSys | What makes you think the low VDC isn't cause by underpowering the line? |
| 00:26.03 | conathan | alright, /exit |
| 00:26.14 | drmessano | Current doesnt make voltage |
| 00:26.20 | drmessano | and there's little load |
| 00:26.55 | CrashSys | 10ma is still below norm's for loop current |
| 00:28.03 | drmessano | He already said the "supply loop current" doesn't change a thing with the SPA-3102 |
| 00:28.17 | CrashSys | Ahhh, missed that part |
| 00:28.40 | drmessano | It's a simple issue of there not being enough voltage |
| 00:28.50 | CrashSys | Some PA's also never go on/off hook too ya know :) |
| 00:28.53 | drmessano | and the SPA-3102 has a readily accessible adjustment for it |
| 00:29.35 | CrashSys | in which case he may have problems if the SPA cant be set to ignore on/off hook... |
| 00:29.54 | drmessano | He said it works fine with his Digium FXO |
| 00:29.55 | jaytee | CrashSys, I get the feeling this may be one of those but it works fine with any of the 4 FXO modules on my TDM04B card. Why can't Linksys make FXO ports that work as well as Digium's? (shameless plug) |
| 00:30.38 | drmessano | wonders why people can't just help fix a problem and need to keep looking for something deeper to be right about |
| 00:30.54 | *** join/#asterisk sucituanbo (n=free@c-24-21-121-148.hsd1.wa.comcast.net) |
| 00:30.57 | CrashSys | who knows |
| 00:31.00 | drmessano | Ohh oh.. maybe the PA is using PAL and not NTSC |
| 00:31.17 | drmessano | Have you checked the frame buffers? |
| 00:31.28 | [TK]D-Fender | woah... now thats deep... |
| 00:31.39 | CrashSys | wonders why some people think their answer is right and that there can be no other possible answers to the solution |
| 00:31.53 | jaytee | ok, how did we go from analog audio to TV signalling? |
| 00:32.15 | CrashSys | jaytee: Aren't you supposed to be in yoru car driving to the office? |
| 00:32.30 | jaytee | yes, thanks for reminding me :-) |
| 00:32.31 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 00:32.31 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:32.34 | jaytee | be back later |
| 00:32.40 | drmessano | Raise your hand if you've had the same problem with an SPA-3102 with the off-hook setting being too high for the connected line? |
| 00:33.19 | jaytee | \\\\ / |
| 00:34.15 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 00:34.39 | CrashSys | Never used a SPA, but have used PAP2's |
| 00:34.54 | CrashSys | and have had to increase loop current to 20ma to get lines to signal correctly :) |
| 00:35.20 | drmessano | Which is not an FXO on a SPA-3102 |
| 00:37.22 | CrashSys | Ahhh, ok, I thought he said he had tried an FXO and was using FXS |
| 00:37.29 | CrashSys | nevermind |
| 00:38.51 | CrashSys | I'm currently trying to find an SLT to Headset adaptor box that avaya makes but I cant remember their non-intuitive naming of the damn thing |
| 00:40.07 | *** join/#asterisk coolthreads (n=shane@203-97-238-71.cable.telstraclear.net) |
| 00:40.21 | coolthreads | micky do bro |
| 00:41.22 | CrashSys | ahhh, when in doubt, ebay :) |
| 00:47.02 | coolthreads | Hey guys, im a newbie to asterisk and would like to say hi :) I have read a little bit on asterisk and im hooked on the concept. look forward to learning with everyone. |
| 00:47.46 | russellb | yay |
| 00:47.52 | russellb | coolthreads: I hope Asterisk treats you well |
| 00:48.04 | russellb | coolthreads: be sure to pick up a copy of the book, it's the best documentation out there IMO |
| 00:48.06 | russellb | ~thebook |
| 00:48.07 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 00:48.10 | drmessano | "I have read a little bit on asterisk" <-- Now THAT is a good start |
| 00:48.18 | russellb | drmessano: :) |
| 00:48.52 | coolthreads | Thanks for the warm welcome |
| 00:49.13 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 00:49.49 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 00:49.53 | coolthreads | I am enjoying that book, well really the electronic book lol very easy reading |
| 00:49.55 | jeev | russel |
| 00:50.02 | russellb | jeev |
| 00:50.07 | jeev | hi. |
| 00:50.13 | drmessano | Dang.. you're already reading the book |
| 00:50.29 | seanbright | russellb: it's a trap! |
| 00:50.40 | russellb | seanbright: eep |
| 00:51.25 | jaytee | drmessano, dude!!!! I owe you a case of beer. :-) |
| 00:51.49 | drmessano | jaytee: That makes the week I wasted figuring that out myself ALLL worth it |
| 00:52.03 | drmessano | Damn fax switches |
| 00:52.10 | drmessano | Mine was a fax switch |
| 00:52.36 | russellb | faxing is lame |
| 00:52.42 | drmessano | lol |
| 00:52.59 | drmessano | I put a SPA-3012 at a friend of mines house.. and he had a fax switch.. needed it behind it |
| 00:53.02 | *** join/#asterisk dlewis (i=4576958c@about/security/staff/dlewis) |
| 00:53.18 | coolthreads | yup already reading the book lol I found some good video clips on asterisk, through youtube. |
| 00:53.30 | jaytee | Damn PA systems, although I'm more inclined to blame Linksys for 1) the fact they won't provide telephone or web support for the SPA line and 2) because they've got the Line In Use voltage set so damn high. |
| 00:53.30 | drmessano | He was getting 29v out of the faux phone port and the SPA-3102 thought the line was always off hook |
| 00:53.48 | jeev | jaytee |
| 00:53.51 | jeev | wanna screw linksys |
| 00:53.53 | jeev | file a bbb claim. |
| 00:53.58 | jeev | they'll suck your cock |
| 00:54.18 | drmessano | Will they send you a Linksys Black Card? |
| 00:54.46 | drmessano | or one of the cool Linksys WRT54GB black edition routers? |
| 00:54.57 | jaytee | my experience is once lawyers get involved it's all pain from there on in. |
| 00:55.41 | jeev | bbb is easy, fast, free. |
| 00:55.47 | jeev | within a week, you'l get someone contacting you |
| 00:55.57 | jeev | i had their corporate support # |
| 00:55.59 | jeev | maybe i could find it |
| 00:56.26 | drmessano | Because the FXO port voltage detection defaulted to 30? |
| 00:56.29 | drmessano | wow |
| 00:56.32 | drmessano | Damn them |
| 00:56.53 | CrashSys | Actually it's the PA's fault for not putting out proper voltage |
| 00:56.59 | drmessano | Indeed |
| 00:57.03 | CrashSys | So technically you need them to suck yer cock |
| 00:57.19 | CrashSys | and you should be happy linksys had it as a changable setting :) |
| 00:57.21 | jaytee | drmessano, I set the Line in Use voltage to 16 from 30 and when I go offhook now on the Info page it shows Off hook and 7V with 34.1ma loop current. |
| 00:57.28 | dlewis | anyone know how to get the SIP info. out of Optimum Voice? |
| 00:57.35 | drmessano | jaytee: Sounds correct |
| 00:57.37 | jaytee | those values look ok to me |
| 00:57.41 | drmessano | Yes |
| 00:57.46 | drmessano | 5 to 15 is nominal |
| 00:57.52 | drmessano | 7V nails it |
| 00:57.59 | drmessano | 34ma loop current is good |
| 00:58.09 | CrashSys | that all looks fine |
| 00:58.13 | drmessano | Just needed to set the threshold |
| 00:58.17 | jaytee | great!!! now I can go home and get into some more comfortable clothes and chill. |
| 00:58.18 | drmessano | Easy as pie |
| 01:00.05 | CrashSys | It's probably a 25-volt PA and the power-supply is 30v |
| 01:00.45 | dlewis | guess not |
| 01:08.16 | *** part/#asterisk tristanbob (n=tristanb@ubuntu/member/tristanbob) |
| 01:10.06 | drmessano | I've found that most equipment that has to emulate a line tried to do it as cheaply as possible, skimping out on putting out true battery as much as possible |
| 01:10.52 | drmessano | Analog devices don't care.. Some like that Digium card, apparently, probably have better logic for sensing a variable state range |
| 01:11.09 | drmessano | The Linksys just happens to be a little too stupid to auto detect when it's low |
| 01:12.10 | nr4q | i used a sipura 3k for a little bit. couldn't stand it |
| 01:12.18 | nr4q | well, linksys branded |
| 01:12.40 | mchou | nr4q: what? what was wrong with it? |
| 01:14.03 | drmessano | Adjusting the gain and impedence always help if the audio has problems |
| 01:14.15 | nr4q | mchou: there was always a problem with echo if the call went through both the FXO and FXS ports. and even then if i just used one side the gain was never right. one party could never hear the other or they were too loud |
| 01:14.54 | nr4q | as soon as i thought i got it right a few calls later it'd be a problem again |
| 01:15.01 | mchou | <PROTECTED> |
| 01:15.06 | drmessano | Oh, and turning off the echo canceller helps too |
| 01:15.40 | nr4q | i think i tried with it both on and off. i just gave up and went with something else |
| 01:15.56 | mchou | nr4q: so what are you using now for fxo/fxs? |
| 01:16.12 | *** part/#asterisk DigitalIrony (n=eric@nat/digium/x-afd23d80abdba488) |
| 01:16.33 | nr4q | mchou: fxs is all SIP phones and FXO is a digium tdm400 |
| 01:16.48 | mchou | ok |
| 01:16.48 | nr4q | mchou: before that i used a fxs daughter card on the digium card with good results |
| 01:20.44 | *** join/#asterisk synchris (n=synchris@athedsl-4386346.home.otenet.gr) |
| 01:22.32 | dlewis | quick question |
| 01:22.45 | CrashSys | 3...2...1... |
| 01:22.51 | dlewis | since Optimum Voice doesn't provide SIP info, I can essentially use my OV via POTS, correct? |
| 01:22.56 | dlewis | and still have the same features... |
| 01:23.39 | dlewis | is that an accurate statement? |
| 01:24.35 | *** join/#asterisk DigitalIrony (n=eric@nat/digium/x-b02655ecd29d51db) |
| 01:25.47 | [TK]D-Fender | dlewis: What kind of service are they providing you? And why won't they give you SIP conenct info? |
| 01:26.17 | *** join/#asterisk frogonwheels_ (n=michaelg@203.59.141.93) |
| 01:26.52 | frogonwheels_ | I was assuming that doing stop gracefully - would stop when calls were idle.. |
| 01:27.18 | frogonwheels_ | but it seems to be waiting for something else .. does it need to wait till all SIP dialogs (and therefore registrations..) have been closed or something/ |
| 01:27.43 | [TK]D-Fender | frogonwheels : Should only wait for end of calls. |
| 01:28.01 | frogonwheels_ | :| it's not working for me *sigh* |
| 01:28.16 | [TK]D-Fender | frogonwheels : pastebin is your friend.. |
| 01:29.53 | *** join/#asterisk imcdona (n=imcdona@imcdona.broker.freenet6.net) |
| 01:30.59 | dlewis | [TK]D-Fender: they provide VOIP via my cable modem |
| 01:31.14 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 01:31.19 | dlewis | www.optimumvoice.com |
| 01:31.33 | Qwell | DigitalIrony: O.o |
| 01:31.59 | frogonwheels_ | argh zombie asterisk processes |
| 01:32.00 | frogonwheels_ | brb |
| 01:33.08 | [TK]D-Fender | dlewis: Ok, great odds that you have no choice and are locked. You can always use an FXO interface to take in the "line" they provide, but no, you will lose a lot of functionality |
| 01:33.32 | dlewis | damn |
| 01:34.15 | [TK]D-Fender | dlewis: Ditch their service and pick your own then |
| 01:34.45 | [TK]D-Fender | Qwell: Is who? |
| 01:34.54 | Qwell | dunno |
| 01:35.24 | dlewis | [TK]D-Fender: yea, the wife wants to keep there "triple play" package |
| 01:35.37 | dlewis | it's "cheaper" to get cable, internet, and phone at the same time from Optimum |
| 01:35.45 | frogonwheels_ | [TK]D-Fender: hmm. ok - it seems to be stopping a bunch of services, but not finally quitting. |
| 01:36.06 | frogonwheels_ | if I attach, there's not SIP command any more.. but iax2 is still there. |
| 01:36.26 | DigitalIrony | Qwell ? |
| 01:36.35 | Qwell | just looking at your hostmask |
| 01:36.42 | Qwell | (and the time) |
| 01:37.27 | DigitalIrony | oh yeah, I work third shift |
| 01:37.44 | Qwell | ahh, at the office? |
| 01:38.04 | DigitalIrony | yeah |
| 01:39.52 | dlewis | DigitalIrony: call center/tech support? |
| 01:40.06 | DigitalIrony | yeah I do the Tech support for digium |
| 01:40.11 | Qwell | I didn't realize we had a third shift.. |
| 01:40.14 | dlewis | right |
| 01:40.18 | dlewis | figured as such |
| 01:40.22 | DigitalIrony | Qwell only 5 days a week |
| 01:40.26 | riddlebox | mchou, hrmm have been sifting through all of these ITSP |
| 01:40.33 | Qwell | makes sense |
| 01:40.42 | DigitalIrony | Qwell and only me and one other |
| 01:40.43 | riddlebox | and only broadvoice can port my number |
| 01:40.44 | Qwell | DigitalIrony: I suppose last night wasn't one of those nights? |
| 01:41.01 | dlewis | another question |
| 01:41.12 | dlewis | the Switchvox SMB tower, what are the tech specs? |
| 01:41.14 | DigitalIrony | Qwell: we are open sunday at 8 - friday at 5am |
| 01:42.35 | dlewis | is there one set of specs for the Switchvox SMB? |
| 01:42.39 | dlewis | (tower) |
| 01:43.43 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:44.01 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177583722.dsl.bell.ca) |
| 01:44.32 | [TK]D-Fender | DigitalIrony: I always wanted to meet one of those support dornes for whom I've saved so much grief over the years ;) |
| 01:44.40 | DigitalIrony | lol |
| 01:45.07 | jeev | i need to change my nick to DigitalFart |
| 01:45.39 | dlewis | guess not |
| 01:45.39 | dlewis | lol |
| 01:45.59 | jeev | huh |
| 01:46.01 | jeev | what you mean |
| 01:46.17 | dlewis | jeev: was referring to my Switchvox SMB Tower question |
| 01:46.28 | jeev | ahh |
| 01:47.31 | dlewis | I know of company that's going belly up |
| 01:47.35 | [TK]D-Fender | drones* |
| 01:47.45 | dlewis | and they recently (6 months ago) bought a Switcvox SMB tower |
| 01:47.50 | dlewis | and they're looking to get rid of it |
| 01:47.59 | dlewis | wanted to know the specs, but they couldn't provide it |
| 01:48.12 | DigitalIrony | What kind of specs? |
| 01:48.20 | [TK]D-Fender | dlewis: Call up support. I'm sure they'll have the answers |
| 01:48.24 | DigitalIrony | They are kind of closed on parts of it |
| 01:48.43 | [TK]D-Fender | "Open" is a scrwedriver away! |
| 01:48.48 | DigitalIrony | yeah |
| 01:49.07 | dlewis | DigitalIrony: hardware specs |
| 01:49.11 | DigitalIrony | sec |
| 01:49.15 | dlewis | does it defacto come with FXO/FSO |
| 01:49.22 | dlewis | i don't have physical access to the box |
| 01:49.23 | DigitalIrony | SMB? |
| 01:49.27 | dlewis | (unfortunately) |
| 01:49.35 | dlewis | SMB, yes |
| 01:49.59 | DigitalIrony | give me a moment I can get you a link to what's publicly available |
| 01:50.17 | dlewis | ok, thanks! |
| 01:50.20 | dlewis | I really appreciate it |
| 01:51.38 | DigitalIrony | Do you know anymore other than just SMB? |
| 01:51.52 | mchou | riddlebox: what?? DC can port your #. Write them an email |
| 01:52.12 | dlewis | unfortunately, no |
| 01:52.22 | *** join/#asterisk Chicago (n=Chicago@c-98-223-62-171.hsd1.in.comcast.net) |
| 01:52.42 | dlewis | I can possibly get a serial number of some sort |
| 01:52.52 | dlewis | is there an identification number on the tower that would be helpful? |
| 01:53.17 | mchou | riddlebox: DC doesnt say it on their web sites but # porting is routine for them. Costs 1 time fee of $10 |
| 01:53.41 | DigitalIrony | well its comes like this. There are different SMB's available. You pick one and can have it outfitted with the hardware you choose |
| 01:53.49 | DigitalIrony | Alot like buying a dell or something |
| 01:53.55 | dlewis | ah, ok |
| 01:53.59 | dlewis | (makes sense) |
| 01:54.19 | dlewis | so, it could have anything... |
| 01:54.20 | DigitalIrony | Except its really more like telephony hardware and not stuff like procs and dvd drives |
| 01:54.26 | DigitalIrony | more or less. |
| 01:54.27 | dlewis | right |
| 01:54.44 | DigitalIrony | http://www.switchvox.com/catalog/smb_bundles.php |
| 01:54.51 | DigitalIrony | there is a good place to start though |
| 01:54.54 | dlewis | thanks |
| 01:54.57 | dlewis | appreciate the info |
| 01:55.00 | DigitalIrony | np |
| 01:55.51 | dlewis | also, although I don't think we have sales guys here, I was told that there unlimited extension plan ends in February. Would you happen to know if there is a way to transfer over this unlimited extension service plan to a new or existing plan? |
| 01:58.53 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 02:04.20 | DigitalIrony | I do not know sorry |
| 02:04.28 | DigitalIrony | I only help work on them |
| 02:04.40 | dlewis | cool |
| 02:05.25 | dlewis | DigitalIrony: have you guys worked on getting a Cisco 7970 working with Asterisk? |
| 02:05.31 | dlewis | I've seen/read all of the tutorials |
| 02:05.36 | dlewis | just wanted to get your opinion |
| 02:07.13 | DigitalIrony | I haven't personally. Some one here probably has though |
| 02:08.41 | dlewis | ok |
| 02:08.46 | *** join/#asterisk nix8n82 (n=nate@mo-65-41-196-62.sta.embarqhsd.net) |
| 02:09.26 | dlewis | for the group, what are your thoughts on the Cisco 7970 and asterisk? |
| 02:10.01 | [TK]D-Fender | BLEH |
| 02:10.15 | [TK]D-Fender | Cisco = overpriced SIP-Crippled trouble |
| 02:10.55 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ip68-111-67-4.oc.oc.cox.net) |
| 02:11.09 | DigitalIrony | I don't really have any, I don't work with cisco much, I mostly work with Pure asterisk implementations |
| 02:11.19 | dlewis | ok |
| 02:11.26 | dlewis | a friend of mine has a couple and was wondering if he can use them with asterisk |
| 02:11.31 | dlewis | i've pointed him to a few sites |
| 02:11.39 | CrazyTux[m] | Is there a way I can make asterisk do a "quiet reload" |
| 02:11.40 | jblack | Cisco... Crisco... both of them cause heart problems when used too much. |
| 02:11.48 | dlewis | lol |
| 02:11.54 | DigitalIrony | Well from what I have done in the past it should, as long as they support the same standards |
| 02:12.14 | DigitalIrony | CrazyTux: why? |
| 02:12.53 | dlewis | a lot of websites have said, besides the Aastra phones, the Cisco ones are the best |
| 02:12.55 | [TK]D-Fender | CrazyTux[m]: "quiet" in what way? |
| 02:12.59 | dlewis | even for * |
| 02:13.16 | DigitalIrony | well you can do a "set verbose 0" in CLI |
| 02:13.22 | DigitalIrony | and it won't display anything |
| 02:13.35 | jaytee | reload -shhhhh! |
| 02:13.46 | jaytee | :-) |
| 02:13.50 | DigitalIrony | lol |
| 02:13.52 | CrazyTux[m] | [TK]D-Fender, in the way that it dosent spit out all of the reload information |
| 02:14.13 | DigitalIrony | CrazyTux: "set verbose 0" |
| 02:14.16 | [TK]D-Fender | CrazyTux[m]: kill your versbose first |
| 02:14.45 | CrazyTux[m] | [TK]D-Fender, theres no command to throw in to do it as well? I'd like to do it from asterisk manager |
| 02:15.00 | CrazyTux[m] | [TK]D-Fender, i.e. only for an instance, of reloads through x gateway |
| 02:15.33 | [TK]D-Fender | CrazyTux[m]: you've been answered twice now |
| 02:15.59 | CrazyTux[m] | setting verbose to 0, I can do this, but it still does not silence |
| 02:16.01 | CrazyTux[m] | everything |
| 02:16.08 | *** join/#asterisk sakajawebe (n=chazz@nat/digium/x-7f8e2ddc2d25cf9f) |
| 02:16.31 | DigitalIrony | CrazyTux[m]: it will silence the reload |
| 02:16.47 | CrazyTux[m] | DigitalIrony, I'm getting, Notices within the relad |
| 02:16.49 | CrazyTux[m] | s/relad/reload/ |
| 02:16.54 | CrazyTux[m] | DigitalIrony, with a verbose of 0 |
| 02:17.26 | DigitalIrony | CrazyTux[m]: Thats the best you can get |
| 02:17.32 | DigitalIrony | Well not really |
| 02:17.40 | DigitalIrony | actually /etc/asterisk/logger.conf |
| 02:17.48 | DigitalIrony | console='' |
| 02:18.10 | DigitalIrony | make console= (to nothing) |
| 02:18.12 | sakajawebe | heh |
| 02:18.16 | DigitalIrony | and save then logger reload |
| 02:18.51 | CrazyTux[m] | yup, just did that |
| 02:19.28 | DigitalIrony | still not what you want? |
| 02:19.32 | *** join/#asterisk DAHDI (n=dahdi@m760e36d0.tmodns.net) |
| 02:20.11 | sakajawebe | so you keen on the upgrade then dahdi? |
| 02:20.45 | MAHMI | :P |
| 02:20.57 | dlewis | lol |
| 02:21.05 | sakajawebe | awww |
| 02:21.29 | sakajawebe | dahdi, meet mahmi. mahmi, meet dahdi |
| 02:21.50 | sakajawebe | you remember that night you told me about where you passed out and didn't really remember anything when you woke up? |
| 02:22.12 | MAHMI | nope, as you said i didn't remember it |
| 02:22.19 | *** join/#asterisk jameswf-home (n=james@ip68-2-99-240.ph.ph.cox.net) |
| 02:23.19 | sakajawebe | whirr, so how is everyone tonight? |
| 02:23.30 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 02:23.33 | dlewis | doing well sakajawebe |
| 02:23.37 | *** join/#asterisk techie (n=techie@m760e36d0.tmodns.net) |
| 02:23.39 | dlewis | I assume you are also tech support? |
| 02:24.37 | *** part/#asterisk korihor (n=korihor@201.211.168.130) |
| 02:26.37 | jameswf-home | ~assume |
| 02:26.42 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 02:27.45 | Digitalirony | yes he is dlewis |
| 02:28.04 | dlewis | ok, cool |
| 02:55.44 | jblack | BofA owns Merrill Lync. Lehman to go bankrupt. |
| 02:56.32 | *** join/#asterisk Entr4nced (i=IMG001@244.sub-75-218-107.myvzw.com) |
| 02:57.40 | dlewis | yup |
| 02:57.48 | dlewis | John Thain is very smart |
| 02:57.49 | [TK]D-Fender | Yippy-kai-yay, the Fed is FUBAR-ing America. |
| 02:57.52 | sapere | evening guys ... anyone have experience with call terminations thru kall8? |
| 02:57.58 | dlewis | comes in for a REDONCULOUS salary |
| 02:58.12 | dlewis | then, helps to sell |
| 02:58.15 | dlewis | eh |
| 02:58.17 | dlewis | it is what it is |
| 02:58.41 | dlewis | they had to do what was needed |
| 02:58.47 | dlewis | which was best for the firm |
| 03:00.50 | jblack | weird. we're talking about asterisk in #mythtv-users |
| 03:00.52 | jaytee | maybe 12 to 18 months and the economy will implode, there'll be riots in the streets, 30% unemployement, Helter Skelter, dogs sleeping with cats and a bowl of Ramen will cost 80 bucks. |
| 03:01.05 | jameswf-home | no no the government is taking over they are here to help all will be okay |
| 03:01.06 | jblack | jaytee: That's what I think. |
| 03:01.23 | jblack | Not tonight, they're not. They're letting Lehman go bankrupt. |
| 03:01.28 | jameswf-home | is well stocked on Ramen |
| 03:01.34 | dlewis | jameswf-home: government is not bailing out anymore banks |
| 03:01.46 | jaytee | whenever a politician stands up at a microphone and says, "There is no cause for alarm!" I immediately panic. |
| 03:01.47 | jblack | Oh, don't believe that. That's a lie. |
| 03:01.51 | dlewis | (according to fed meetings today) |
| 03:02.04 | dlewis | they're letting wall street work things out this time |
| 03:02.12 | jblack | This time, yes. |
| 03:02.35 | jblack | They had to let them sink, otherwise the market would take it as proof that the fed will always do so. It's a sacrificial lamb. |
| 03:02.48 | [TK]D-Fender | RP2008!!!!! |
| 03:02.52 | *** join/#asterisk SteveTotaro (n=Administ@pool-141-157-95-245.balt.east.verizon.net) |
| 03:02.52 | jblack | Just like when they sent a message with Bear that don't wont force all of them to sink. |
| 03:02.55 | jblack | RP2008!!! |
| 03:03.07 | CrashSys | I just want a "None of the above" option on the ballot |
| 03:03.08 | jameswf-home | the feds hrow around money like crazy cause well you know they print i. graned the dollar will be worth .10 cents but the people demand bail outs and its an election year so let them eat cake |
| 03:03.33 | jblack | well, it depends on how far ahead you're looking. |
| 03:03.42 | *** join/#asterisk coppice (n=chatzill@177.162.17.210.dyn.pacific.net.hk) |
| 03:04.21 | jblack | We're likely to get a healthy dose of deflation before we get the zimbabwe style inflation of death. |
| 03:05.05 | jameswf-home | yay sociallist goverment we all can work for the governmen and tehy will in turn provide all they feel we need, substandard health care for all government cheese mmmmm |
| 03:05.54 | CrashSys | socialist medicine/healthcare works in a lot of places |
| 03:05.58 | jaytee | I liked the government cheese they gave out in the 80's. Kinda bound you up though. |
| 03:06.02 | jameswf-home | I cant wait for my free health care.... 6 month to see a doctor who cant help me |
| 03:06.06 | Nugget | your pick: socialist where we all just get money for the state, of fascist where the only good jobs are with halliburton. |
| 03:06.12 | [TK]D-Fender | jameswf-home: Easy to throw around money when you print it and invent its value ;) |
| 03:06.14 | Nugget | er, from the state. |
| 03:06.37 | jblack | Heh. The US is one of the few industrialized economies that didn't socialize health care. And look waht happened to the cost. |
| 03:07.06 | [TK]D-Fender | Nugget: Unless you're female & don't like the thought about getting raped, maybe ;) |
| 03:07.10 | jameswf-home | Do you want good expensive healthcare or bad free health care |
| 03:07.28 | coppice | free and paid health care both suck. with free health care you can't always get enough. with paid health care you can't get an honest opinion of how little you need |
| 03:07.29 | jblack | I can understand why someone wouldn't want to socialize welfare, education, job hunting, etc.... But not socializing health care means that you're surrounded by disease spreading sick people that are too ill to work. |
| 03:07.34 | CrashSys | We have expensive bad healthcare :) |
| 03:07.38 | jaytee | right now I get expensive shitty healthcare |
| 03:07.39 | [TK]D-Fender | lives in healthy happy Canuckianland |
| 03:07.48 | jameswf-home | Look at the projects in chicago that is how well socialist housing works |
| 03:08.04 | jaytee | 300 bucks to get my cholesterol and triglycerides tested. |
| 03:08.06 | Chicago | It is a pretty ugly scene. |
| 03:08.10 | jblack | Housing and health care are different issues. |
| 03:08.18 | jameswf-home | Jaytee learn spanish it will be free |
| 03:08.48 | jameswf-home | No habla english = free health care |
| 03:08.49 | jaytee | and I paid 4.15 a gallon for gas yesterday. two days before it was 3.56 a gallon. |
| 03:08.55 | nr4q | if you are using Dial() to ring multiple extensions at once, is there a way to run a command only if a specifc extension picks up? |
| 03:09.03 | jameswf-home | I paid 3.37 today for gas |
| 03:09.09 | jblack | nr4q: Good question. |
| 03:09.34 | jameswf-home | We get our gas from cali |
| 03:09.49 | jblack | Usually, if a call goes through, you don't get to the next step in the dialplan at all. |
| 03:09.52 | nr4q | jameswf-home: my mom told me gas is $5.50/gal where she lives |
| 03:10.20 | jameswf-home | nr4q: I bet when this is all over it won drop much under 5 |
| 03:10.46 | [TK]D-Fender | "over"? |
| 03:10.46 | jaytee | one thing I always wondered about. When I drove cross country I went through Nevada on Interstate 80 and every gas station I filled at only had 85 octane. |
| 03:10.52 | [TK]D-Fender | lol |
| 03:10.55 | jameswf-home | once they get that much from a market they dont let go |
| 03:11.15 | [TK]D-Fender | Of course not. The price of oil drops, the price at the pump doesn't. |
| 03:11.16 | nr4q | jblack: i have an internal SIP extension and a ZAP extension (to ring a cell phone) at once. first that answers gets the call but I want to run a System() command if the Zap channel gets it |
| 03:11.23 | jameswf-home | [TK]D-Fender: hurricane crap |
| 03:12.03 | nr4q | in some areas it's over $6. apparently the area my parents live in is just out of gas |
| 03:12.10 | [TK]D-Fender | jameswf-home: **BS** America's production is an insignificant portion of its consumption. |
| 03:12.16 | *** join/#asterisk irieKEN (n=irieken@netblock-68-183-146-204.dslextreme.com) |
| 03:12.32 | *** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net) |
| 03:12.33 | jaytee | if only we could invent a car that runs on bullshit. we could build pipelines to the Capitol building in D.C. and we'd have an unlimited fuel supply for future generations. |
| 03:12.40 | [TK]D-Fender | FTW! |
| 03:12.52 | [TK]D-Fender | jaytee: Its called "natural gas" ;) |
| 03:12.58 | jaytee | lol |
| 03:13.06 | jameswf-home | I dunno AZ as of yet is uneffected |
| 03:13.10 | [TK]D-Fender | jaytee: "passing-wind-fall tax" |
| 03:14.28 | nr4q | wind for oil |
| 03:15.02 | *** join/#asterisk korihor (n=korihor@190.78.32.60) |
| 03:15.11 | jameswf-home | some folks are running natural gas in their cars and have a tap in their garage...... I cant imagine paying my gas bill in a lump sum would probably make me not drive |
| 03:15.49 | [TK]D-Fender | Electricity / air should be the future... |
| 03:16.07 | nr4q | children are our future. teach them well and let them lead the way |
| 03:16.18 | jameswf-home | i say methane and hydrogen |
| 03:16.33 | [TK]D-Fender | jameswf-home: Anything hydrocarbon based = ew |
| 03:17.22 | [TK]D-Fender | jameswf-home: fire, pressurization issues, poitn of origin, etc. |
| 03:17.22 | jblack | I don't understand. Is consumer grade natural gas liquid or solid? |
| 03:17.22 | jaytee | in the Phillipines they have small busses and cars that run on burning pig shit and coconut husks. |
| 03:17.22 | [TK]D-Fender | jameswf-home: non-renewable <- |
| 03:17.22 | jblack | Pardon, liquid or gas. |
| 03:17.22 | [TK]D-Fender | McCain is a dumb-ass with his "drill now" policies. |
| 03:17.27 | coppice | jaytee: well .ph definitely has shitty traffic |
| 03:17.32 | CrashSys | McCain = Bush v.2.0 |
| 03:17.33 | jblack | If it's a gas, how would one achieve the energy density required to make it more than 500' down the road? |
| 03:17.43 | *** join/#asterisk Entr4nced (i=IMG001@244.sub-75-218-107.myvzw.com) |
| 03:17.55 | coppice | jblack compression |
| 03:17.56 | jaytee | McCain = Bush wearing Depends with a full load |
| 03:18.07 | [TK]D-Fender | CrashSys : more like Bush v2.0 Alpha |
| 03:18.09 | jameswf-home | wtf did we buy alaska for if not oil |
| 03:18.47 | jblack | coppice: So... gas liquification plants in every home? |
| 03:18.56 | CrashSys | We bought alaska for the aliens... didn't you know that's where they all come from in the movies? |
| 03:19.07 | jaytee | I say we just build huge submarine tankers, underwater horizontal drilling rigs and go over to the gulf offshore and drill under their land to the oil, pump it all out and bring it back here. |
| 03:19.10 | jblack | We bought alaska for polar bear meat. |
| 03:19.20 | nr4q | jaytee: in the philippines they also have open sewers |
| 03:19.23 | jblack | bets polar bear tastes like pork. |
| 03:19.48 | jaytee | mmmmm, bacon wrapped bacon sprinkled with bacon bits. |
| 03:19.55 | [TK]D-Fender | jblack: Nope, much more pungent. Its a VERY red meat. |
| 03:20.11 | jblack | Have you eaten polar bear? |
| 03:20.19 | jaytee | I've eaten caribou |
| 03:20.31 | jameswf-home | club some baby seals? |
| 03:20.52 | jameswf-home | pray for whirled peas |
| 03:20.53 | jaytee | there's a restaurant in Boston called Durgin Park that serves bear meat on occassion. |
| 03:21.08 | jameswf-home | I like elk |
| 03:21.18 | jaytee | Yak aka dak! |
| 03:22.28 | jaytee | we have one of Ted Turner's restaurants here that serves Buffalo. Buffalo is pretty tasty but the meat seems kinda grainy in texture. |
| 03:23.44 | jameswf-home | we have alot of mexican restraunts here all the dog and horse meat you can eat |
| 03:23.58 | jaytee | El Taqueria Rapido |
| 03:24.29 | coppice | donkey is better than horse |
| 03:24.51 | jameswf-home | mule the perfect blend? |
| 03:25.09 | jaytee | the area of Indianapolis I live in qualifies as "Little Tijuana". We even have mexican "roach coaches" |
| 03:25.14 | coppice | dunno. don't think I've tried crossbreeds |
| 03:25.46 | jameswf-home | Jack in the box serves Kangaroo mea |
| 03:25.50 | jameswf-home | *meat |
| 03:26.04 | jaytee | eventually all we'll be able to get is soy/plankton substitutes and then finally Soylent Green. |
| 03:26.22 | coppice | kangaroo is good |
| 03:26.32 | jaytee | Skippy burgers! |
| 03:26.47 | coppice | kiwi is yummy |
| 03:26.52 | jaytee | "He's got a gun!" |
| 03:27.12 | jblack | Aspergers video: http://www.youtube.com/watch?v=6jkBOU9etRA |
| 03:27.44 | [TK]D-Fender | Horse is great... I do it in fondue all the time. |
| 03:28.13 | [TK]D-Fender | jblack: Asperagus? |
| 03:28.34 | jameswf-home | those 4 for a dollar burritos from the grocery store says on the ingriedience list "textred protienn product" which i say is delicious |
| 03:30.12 | jameswf-home | you know what else is good, jack in the box tacos, i dont know what that meat like paste is just that it is not meat and is delicious |
| 03:30.12 | jaytee | I gotta crash. Nite all |
| 03:30.12 | coppice | "textred protienn product" == tufo? |
| 03:30.15 | jaytee | chicken lips |
| 03:30.15 | *** join/#asterisk devhen (n=devhen@66.236.68.230.ptr.us.xo.net) |
| 03:30.28 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-20-76.revip2.asianet.co.th) |
| 03:30.33 | [TK]D-Fender | coppice: repacked meat. You see it in frozen prepared foods all the time. |
| 03:30.37 | jaytee | they actually put cow lips in Slim Jims |
| 03:30.53 | [TK]D-Fender | coppice: They can make chicken breast meat you would really thing is "real" yet isn't |
| 03:30.53 | jameswf-home | cow lips yummm |
| 03:31.33 | coppice | [TK]D-Fender: I guess they use silicone |
| 03:31.47 | jameswf-home | my mother in law went to the dimsum restraunt in china town sanfrancisco said they kept coming up pushing the chicken feet |
| 03:32.04 | [TK]D-Fender | coppice: Corn starch & wheat glutan primarily |
| 03:32.34 | coppice | chicken feet are yummy, especially chillied ones |
| 03:33.07 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:33.07 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:33.42 | coppice | [TK]D-Fender: they learned most of those fake meat tricks from this part of the world. Many Buddhists are vegetarian, but may others eat meat yet have veggie days when they eat mostly fake meat |
| 03:33.44 | jameswf-home | shhhh hes here |
| 03:34.01 | *** join/#asterisk Levonk (n=lk@adsl-76-227-119-47.dsl.lsan03.sbcglobal.net) |
| 03:35.15 | [TK]D-Fender | coppice: Well here it isn't to be "fake" so much as to be preservable longer, etc and save on cost |
| 03:35.58 | coppice | the cost saving comes mostly from a cow taking 10kg of food to make 1kg of meat |
| 03:38.48 | *** join/#asterisk xuser (i=jaood@unaffiliated/xuser) |
| 03:39.16 | nr4q | how obvious is the BLI on polycom phones? I can't seem to find any photos on google images |
| 03:44.06 | [TK]D-Fender | nr4q: does "clear red light" say it for you? |
| 03:45.22 | nr4q | fender: er... I meant BLF |
| 03:45.37 | nr4q | i need more coffee :( |
| 03:45.44 | *** join/#asterisk irieKEN_ (n=irieken@netblock-68-183-146-204.dslextreme.com) |
| 03:49.39 | [TK]D-Fender | nr4q: And the answer isn't any different |
| 03:50.32 | x86 | '8 |
| 03:50.47 | nr4q | fender: oh i know which light you are talking about now. thanks |
| 03:54.14 | *** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net) |
| 03:57.54 | roe_ | has anyone had recent success using asterisk (with the dummy driver) in a xen domu? |
| 03:58.20 | CrashSys | virtualized Asterisk = Hell warmed over |
| 03:58.47 | CrashSys | Although, Switchvox has their own proprietary ztdummy written for their virtualized hosting that they wont share/sell |
| 03:59.03 | roe_ | so not a good idea? |
| 03:59.20 | CrashSys | Not a stable idea to the general asterisk user yet :) |
| 03:59.27 | roe_ | "Hell warmed over" doesn't seem too far from reality. Hell is generally considered pretty warm |
| 03:59.36 | CrashSys | It's a good idea, just not stable... |
| 03:59.48 | roe_ | so still has some timing issues |
| 03:59.51 | CrashSys | ztdummy cant maintain 98%+ accuracy |
| 04:00.20 | CrashSys | So yes, it can cause audio problems, specially if you use IAX Trunking or meetme or anything that requires audio processing |
| 04:00.45 | CrashSys | If you are basically using asterisk as a packet router it's fine :) |
| 04:02.21 | roe_ | well suxors |
| 04:18.13 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 04:23.18 | jameswf-home | Dont call me dummy |
| 04:23.40 | jameswf-home | should put that in a printk when ztdummy loads |
| 04:40.37 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:40.37 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:54.24 | *** join/#asterisk moy (n=moy@189.169.91.147) |
| 05:23.48 | frogonwheels_ | trying to debug problem with -rx "stop gracefully" not actually stopping. |
| 05:24.30 | frogonwheels_ | if I have a console open on another terminal, (verbose 10) and send asterisk -rx "stop gracefully" .. then it seems to stop at.. |
| 05:24.37 | frogonwheels_ | <PROTECTED> |
| 05:25.01 | frogonwheels_ | (I'm trying to get a general mechanism for gracefull restart into openwrt asterisk) |
| 05:28.10 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 05:29.51 | frogonwheels_ | anybody got any ideas on what I should be looking for? |
| 05:30.45 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ip68-111-67-4.oc.oc.cox.net) |
| 05:32.06 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 05:36.54 | paulproteus | frogonwheels_, Is it possible to just send Asterisk a signal? |
| 05:36.58 | paulproteus | frogonwheels_, Also, what Debian does is: |
| 05:37.14 | paulproteus | asterisk -rx reload |
| 05:38.14 | paulproteus | http://paulproteus.acm.jhu.edu/asterisk-debian-rc fwiw |
| 05:38.17 | frogonwheels_ | paulproteus: I missed that one - that will help one case. |
| 05:38.17 | paulproteus | dives into bed |
| 05:38.49 | frogonwheels_ | Problem is I want to gracefully restart and then also reboot the ata |
| 05:39.06 | frogonwheels_ | - but the stop gracefully doesn't seem to work. |
| 05:39.51 | paulproteus | Does it stop? |
| 05:39.57 | frogonwheels_ | no |
| 05:39.58 | paulproteus | http://bugs.digium.com/view.php?id=8897 indicates to me it might print that out even if stop would succeed. |
| 05:40.02 | frogonwheels_ | it stops some stuff |
| 05:40.22 | paulproteus | is out of ideas and falls asleep |
| 05:40.37 | frogonwheels_ | yeah- it's just after SIPshowpeer that has the problems |
| 05:40.43 | frogonwheels_ | gn |
| 05:41.53 | *** join/#asterisk sergey (n=Sergey@sergey.iks.ru) |
| 05:44.47 | sergey | Hi. Have trunk (asterisk, dahdi), use dahdi-dummy (dahdi show status - DAHDI_DUMMY/1 (source: HRtimer) 1). But chan_iax2.c: Unable to support trunking on user 'iax' without DAHDI timing |
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| 06:08.14 | mosty | is it possible to start asterisk without putting an asterisk console on tty9? |
| 06:09.16 | frogonwheels_ | mosty: that sounds like a distro thing |
| 06:09.50 | frogonwheels_ | check the init script |
| 06:10.58 | mosty | frogonwheels_, i compiled asterisk myself, and there is no mention of tty9 (or any other tty) in my init script |
| 06:12.29 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 06:14.37 | frogonwheels_ | oh |
| 06:18.32 | *** join/#asterisk jeev (n=email@unaffiliated/jeev) |
| 06:24.35 | mosty | ahh, found it (in safe_asterisk) |
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| 06:49.20 | implicit | pl |
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| 06:51.51 | dominic1 | hi folks |
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| 07:08.09 | reallost1 | Anyone awake here? I'm stumped trying to fix one way audio on iax2 calls. |
| 07:09.16 | mchou | woah |
| 07:09.31 | mchou | one way audio on iax?? |
| 07:09.39 | reallost1 | yeah, sip has no audio at all. |
| 07:09.45 | mchou | man that's bad |
| 07:09.54 | reallost1 | I can hear them, but they can't hear me... |
| 07:09.58 | mchou | check your firewall or NAT |
| 07:10.21 | reallost1 | I've been going over my firewall/nat trying to figure out what the problem could be. |
| 07:10.44 | mchou | where is your firewall wrt to local phone and asterisk? |
| 07:11.24 | reallost1 | pstn -> firewall -> asterisk -> (dial) iax2 provider -> callee |
| 07:11.25 | mchou | and what firewall are you using? |
| 07:12.06 | mchou | shit, you're that same guy who has been asking this question for 1 wk |
| 07:12.12 | frogonwheels_ | reallost1: have you tried 2 iax2 providers inside the firewall? |
| 07:12.30 | reallost1 | mchou, no I haven't asked this question here at all. |
| 07:13.19 | mchou | reallost1: you sure? Cause someone else was whining about this |
| 07:13.30 | reallost1 | positive. |
| 07:13.37 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 07:13.59 | mchou | reallost1: where are you on that graph? (left or rigt?) |
| 07:14.16 | reallost1 | I usually figure out my own problems. This started on friday when I moved the server behind the firewall. |
| 07:14.36 | reallost1 | I'm calling in from the pstn. |
| 07:14.50 | reallost1 | so that would be left. |
| 07:15.00 | mchou | ~sipnat |
| 07:15.01 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 07:15.26 | mchou | reallost1: you set extenip? |
| 07:15.31 | mchou | externip* |
| 07:15.35 | obnauticus | heh i just use a sip proxy at my router :\ |
| 07:15.41 | obnauticus | a lightweight one |
| 07:15.44 | obnauticus | that does it real fast. |
| 07:15.56 | obnauticus | works out pretty well :\ |
| 07:16.06 | mchou | obnauticus: which sip proxy? siproxd? |
| 07:16.44 | reallost1 | yeah, externip is set. |
| 07:17.04 | mchou | is it set correctly? :) |
| 07:17.49 | mchou | oh, wait a sec |
| 07:18.06 | mchou | heh, I got you reversed |
| 07:18.19 | obnauticus | mcab ya i use siprox |
| 07:18.19 | obnauticus | d |
| 07:18.21 | *** join/#asterisk L-info (n=L-info@80.169.36.194) |
| 07:19.29 | reallost1 | externip is for sip |
| 07:19.34 | reallost1 | and its set correctly. |
| 07:19.38 | mchou | reallost1: when you say pstn->firewall->asterisk, I'm not sure what you mean. PSTN is not IP |
| 07:20.06 | mchou | so why would firewll even be involved? |
| 07:20.11 | mchou | firewall* |
| 07:20.50 | reallost1 | oh, pstn I mean I dial a real number on my phone -> iax2 -> firewall -> asterisk |
| 07:21.19 | reallost1 | I can play audio files and hear them just fine with that setup. |
| 07:22.11 | reallost1 | When I continue the call and dial out and then bridge the call, the person I call can't hear me. |
| 07:22.11 | mchou | you have root priv on this asterisk box? |
| 07:22.11 | reallost1 | yeah |
| 07:22.11 | reallost1 | and on the firewall. |
| 07:22.52 | mchou | so what relevant iax ports are open on firewall? |
| 07:23.22 | reallost1 | I've even turned off the firewall rules, just leaving the redirects and get the same results. |
| 07:23.43 | mchou | nah, that wont work |
| 07:23.53 | mchou | there is still nat |
| 07:24.44 | mchou | that pretty much means your redirects are foobar |
| 07:25.23 | mchou | redirect meaning firewall redirects |
| 07:25.43 | mchou | not sip reinvite or some such |
| 07:26.06 | reallost1 | right firewall redirects... |
| 07:26.45 | reallost1 | k, I'll go back to my firewall... |
| 07:26.53 | mchou | reallost1: no funky deny/permit rules? |
| 07:27.06 | mchou | on firewall and * config? |
| 07:27.10 | frogonwheels_ | reallost1: possibly you should pastebin your f/w config? |
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| 08:17.19 | awk | hmm, anyone could explain this |
| 08:17.24 | awk | "Skip Busy Agents = Yes" I see this on thenet at places |
| 08:17.30 | awk | is this really a queue.conf option? |
| 08:17.38 | awk | why is it not linked eg: skip-busy-agents? |
| 08:17.43 | awk | and is this a realy option? |
| 08:27.55 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 08:32.09 | kaldemar | awk: no, it's not. |
| 08:32.56 | awk | kaldemar then what is this nonsense |
| 08:33.06 | awk | http://forum.queuemetrics.com/index.php?topic=273.0 |
| 08:33.13 | awk | the bottom page in blue... and i've seen it in other posts? |
| 08:34.49 | kaldemar | not asterisk anyway. |
| 08:36.56 | kaldemar | maybe freepbx, trixbox or queuemetrics stuff. look into ringinguse if you need something like that. |
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| 08:58.43 | ttf | when I connect an ATA (spa3102) to my pstn line and to my switch I will be able to configure the ata so that if a call comes in through my pstn line it will be forwarded to my notebook (which is on the same network like my switch) - correct? |
| 08:59.58 | awk | kaldemar yes ringinuse would be great if app_queue.so wasn't broken |
| 09:00.16 | awk | but it still duplicates ringing when busy... sip or local channels |
| 09:02.06 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 09:04.38 | awk | out of intrest how many people here are having queue issues? |
| 09:04.47 | awk | ring in use/ wrong states , etc etc |
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| 09:46.23 | jblack | OMG The banks are all gonna die. We're dooomed! |
| 09:46.51 | jblack | "Lehman is the biggest bankruptcy filing in history" |
| 09:47.26 | *** join/#asterisk admin0 (n=admin@bb121-7-190-232.singnet.com.sg) |
| 09:47.53 | jblack | omg omg. Lehman bonds to get $0.60 on the dollar. |
| 09:50.11 | *** join/#asterisk smach (n=smach@guy78-3-82-239-225-173.fbx.proxad.net) |
| 09:50.40 | smach | Hi guys, I have some trouble loading the skinny module |
| 09:51.02 | jblack | What sort of trouble? |
| 09:51.22 | smach | it says 'Unable to register channel class skinny' |
| 09:52.04 | *** join/#asterisk kraypius (n=dasdas@c-67-167-121-213.hsd1.il.comcast.net) |
| 09:52.34 | mchou | I blame Greenspan |
| 09:52.40 | kraypius | has anyone here successfully used this? http://revver.com/video/1083088/iphone-app-store-asteriskc2d/ |
| 09:53.12 | kraypius | mt asterisk is on linux but im thinking of trying to get that script running |
| 09:53.30 | mchou | Greenspan is responsible for the financial mess |
| 09:53.31 | smach | more accurately, the log says 'already have a handler for type 'Skinny' and then 'Unable to register channel class skinny' |
| 09:54.27 | kraypius | if i could do that I could reduce my iphone bill |
| 10:00.46 | frogonwheels_ | kraypius: I have my nokia connecting to Asterisk via sip - and it's soo great |
| 10:01.03 | kraypius | using c2d? |
| 10:01.09 | mchou | kraypius: lol. iphone click 2 dial |
| 10:01.16 | frogonwheels_ | no |
| 10:01.32 | mchou | kraypius: nokia is a sip client |
| 10:01.46 | mchou | s/is/has |
| 10:02.00 | kraypius | which nokia? just curious. i used to be a big nokia fanatic |
| 10:02.52 | mchou | E61,E70 |
| 10:03.27 | linuxstb | uses the E60, also very happily |
| 10:05.23 | mchou | kraypius: is asteriskc2d 'legit?' |
| 10:05.38 | mchou | or is it jailbreak? |
| 10:05.44 | kraypius | legit |
| 10:06.39 | mchou | frigging istore and Steve Jobs |
| 10:06.47 | kraypius | dont remember if i paid to get it. i think it was free |
| 10:07.24 | kraypius | yeah they have went the extra mile to make people pay for shit. i hate it |
| 10:07.28 | mchou | kraypius: just philosophically opposed to iStore is all |
| 10:07.53 | dlewis | they need SIP for the blackberry... |
| 10:08.00 | *** join/#asterisk nacer (n=God@l.alcolo.a.mpl.pastIX.net) |
| 10:08.07 | *** join/#asterisk nacer (n=God@l.alcolo.a.mpl.pastIX.net) |
| 10:08.07 | mchou | how come Stallman doesnt rail against iStore? :) |
| 10:08.07 | kraypius | you even have to use itunes and buy music to make ringtones unless you download an itunes hack |
| 10:09.33 | linuxstb | kraypius: Doesn't that app just use the iphone as a dialer app - i.e. you need a second (VOIP) phone to actually make the call? |
| 10:09.48 | kraypius | idk. it sucks because you pay so much for a powerful piece of hardware but its like you dont even own it for all its capable of |
| 10:09.48 | mchou | linuxstb: no |
| 10:10.14 | kraypius | i believe its a true sip client |
| 10:10.38 | linuxstb | mchou: That's what the commentary says though - i.e. it allows you to make calls from your desktop phone. |
| 10:10.41 | mchou | linuxstb: it essentially uses iax to talk to asterisk, I think |
| 10:10.57 | frogonwheels_ | kraypius: mines E65 |
| 10:11.07 | mchou | asterisk rings any of your defined extensions |
| 10:11.16 | mchou | linuxstb: ^^^ |
| 10:11.34 | linuxstb | mchou: Where did you find that information? |
| 10:11.37 | mchou | linuxstb: well, the exten defined on the video |
| 10:12.00 | mchou | in the setup screen |
| 10:12.06 | kraypius | does it have to be aix? |
| 10:12.36 | mchou | kraypius: no |
| 10:12.49 | kraypius | o |
| 10:12.50 | mchou | kraypius: SIP would also work |
| 10:12.58 | kraypius | i dont have any aix trunks defined as of yet |
| 10:13.08 | mchou | nono |
| 10:13.20 | mchou | you dont need to define trunks |
| 10:13.31 | mchou | betw. the iphone and asterisk |
| 10:14.17 | mchou | think of it this way. *c2d enters you dial plan to dial an extension |
| 10:14.23 | mchou | your* |
| 10:14.58 | mchou | the extension you specified on that first screen shown in the video |
| 10:15.11 | mchou | that's all it's doing really |
| 10:16.07 | linuxstb | Yes, it just looks like it uses the AMI to originate a call - i.e. it's not using the iphone as a VOIP client, just as a phone directory. |
| 10:16.09 | kraypius | i just asked because i assume it has to communicate the call using whatever protocol the trunk for the extension is using |
| 10:16.31 | mchou | linuxstb: yup, you got it |
| 10:17.19 | *** join/#asterisk alibb (n=ali@196.203.178.164) |
| 10:17.26 | mchou | you still talk on another phone (i.e. not the iphone) |
| 10:17.26 | linuxstb | mchou: So pretty pointless compared to our Nokias... |
| 10:17.41 | mchou | linuxstb: yeah |
| 10:17.43 | kraypius | so wait what is the difference? |
| 10:18.03 | linuxstb | What mchou just said - you need a second phone to actually talk. You don't talk into the iphone. |
| 10:18.16 | mchou | kraypius: what?? I dont understand what you're even asking |
| 10:18.49 | mchou | kraypius: you ever use Grandcentral or facebook click to dial? |
| 10:18.57 | kraypius | no |
| 10:19.02 | mchou | kraypius: the concept is the same |
| 10:19.08 | kraypius | you have to use a second phone? |
| 10:19.14 | mchou | yup |
| 10:19.23 | kraypius | the concept leads you to believe you can use the phone to talk |
| 10:19.29 | mchou | lol |
| 10:19.34 | mchou | no it does not |
| 10:19.52 | kraypius | so whats the point of the app then? |
| 10:20.10 | mchou | I mean you could ring your own iphone but I'm not sure how that saves you minutes |
| 10:20.13 | linuxstb | kraypius: Exactly what I was thinking... |
| 10:20.53 | mchou | does iphone charge for inbound minutes? |
| 10:21.08 | kraypius | yeah i think so |
| 10:21.11 | kraypius | at&t |
| 10:21.33 | mchou | so it wont save you anything if you ring your iphone |
| 10:21.46 | linuxstb | mchou: That would depend on your cellular provider - in the UK (and I think all of Europe), the caller pays 100% of the cost when calling a mobile. |
| 10:22.03 | mchou | granted inbound minutes are probably cheaper than outbound |
| 10:22.22 | mchou | linuxstb: agreed |
| 10:22.36 | kraypius | well, for at&t i believe they are the same cost |
| 10:22.44 | kraypius | minutes are minutes |
| 10:23.37 | mchou | so you could ring (and talk) on iphone but it's no toll saver |
| 10:23.50 | mchou | at least for at&t |
| 10:24.14 | kraypius | if you're using the c2d it wont be using the data network? |
| 10:24.37 | mchou | only if your iphone supports sip :) |
| 10:24.44 | mchou | lol |
| 10:24.46 | frogonwheels_ | kraypius: .. and doesn't help if they ring your mobile |
| 10:24.54 | mchou | chicken and egg :) |
| 10:24.56 | frogonwheels_ | kraypius: only helps if they ring your sip # |
| 10:25.06 | frogonwheels_ | I mean .. you know what I mean. |
| 10:25.15 | mchou | frogonwheels_: haha :) |
| 10:25.50 | mchou | kraypius: check out iCall and try to get in on that |
| 10:25.55 | kraypius | so.. if they ring my sip# i can receive the call on my iphone... |
| 10:26.05 | mchou | that's really what you need |
| 10:26.13 | mchou | kraypius: NO dude |
| 10:26.42 | mchou | kraypius: you need a sip or iax client on iphone to be on data net |
| 10:26.51 | mchou | PERIOD |
| 10:26.59 | mchou | asterisk or no asterisk |
| 10:27.00 | frogonwheels_ | kraypius: yeah, that _is_ the idea.. as long as you have a sip/iax client on the iphone to connect to asterisk |
| 10:27.28 | kraypius | OH THATS FINE |
| 10:27.29 | kraypius | #(_JIKOPE#I |
| 10:27.32 | kraypius | FINE |
| 10:27.49 | kraypius | :P |
| 10:28.31 | mchou | kraypius: iCall is your savior man |
| 10:28.46 | mchou | but that looks suspiciously like Vaporware |
| 10:28.49 | kraypius | heh i dont think icall will run on my iphone |
| 10:29.04 | kraypius | oh wait. i see it |
| 10:29.30 | mchou | http://icall.com/iphone/ |
| 10:29.47 | kraypius | they're talkin bout wifi |
| 10:29.55 | kraypius | 3g would be nice |
| 10:29.57 | mchou | No, you can not have a copy. We are not physically capable of distributing the application to any phones except our development phones. If Steve-ness himself wanted a copy we could not provide it to him. Sorry. |
| 10:30.06 | mchou | lol |
| 10:30.25 | mchou | classic vaporware come-on |
| 10:30.59 | mchou | "Apple has explicitly stated that VoIP is allowed, just not over Edge networks. Steve himself answered this question in the Q&A session after the last keynote speech." |
| 10:31.14 | mchou | I dont see how you gonna overcome that |
| 10:31.37 | kraypius | yeah what about 3g |
| 10:31.46 | kraypius | edge is crap anyway |
| 10:32.23 | kraypius | though if call quality on edge was good it would obviously be awesome |
| 10:32.36 | mchou | I thing by "Edge networks" Steve included 3g. Beats me though |
| 10:32.56 | mchou | think* |
| 10:33.02 | kraypius | if so then steve is so much more of a fagtron |
| 10:33.15 | mchou | haha |
| 10:33.40 | mchou | dude, that man is wasting away. It's no laughing matter |
| 10:33.42 | kraypius | yeah, you can do voip, but not anywhere that it will matter |
| 10:33.54 | mchou | looks like he's been on chemo |
| 10:34.37 | mchou | either that or he contracted something BAD |
| 10:34.46 | *** join/#asterisk angryuser (n=aster@78.115.228.63) |
| 10:35.06 | kraypius | good maybe his iphone gave him cancer |
| 10:35.14 | mchou | lol |
| 10:35.31 | mchou | iphone hasnt been around long enough to give him cancer |
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| 10:51.35 | Devyll | hello all. can you guys please tell me what kind of device do I need for beeing able to connect a e1 flux to the asterisk server ? . It'll probably be a kind of e1 flux card (like a eth card). can anybody confirm ? |
| 10:51.50 | JT | flux? |
| 10:52.16 | disposable | do polycom phones (ip650, ip430) do any QoS magic on the cable if i daisychain a computer through them? |
| 10:54.14 | angryuser | disposable, im not sure if the phone manages the data flow going through it |
| 10:54.35 | Rico29 | i don't think |
| 10:54.45 | Rico29 | i think they don't |
| 10:54.47 | Rico29 | sorry :) |
| 10:55.56 | Devyll | e1/t1 (something that I will have from the telephone provider) 30 phone lines which will probably come by wire (optic fiber) that is a e1 (europe) t1 (us) .. |
| 10:56.04 | disposable | thanks |
| 10:56.12 | angryuser | disposable, normaly if it a windows station you got Qos in network params, and the pc will use 80% of bandwith |
| 10:56.57 | *** join/#asterisk steliosk (n=Stelios@79.107.51.60) |
| 10:58.15 | disposable | angryuser: it's not that important. it's just for a client who wants everything for nothing. (as in full call quality with no dedicated voice adsl connection) |
| 11:01.23 | *** join/#asterisk Cy (n=Cy@109.Red-83-57-65.dynamicIP.rima-tde.net) |
| 11:02.31 | Cy | Does anybody know how to add a SIP extension XXX@domain.com to a 7900 series phone? |
| 11:02.31 | Cy | It rejects the @ sign |
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| 11:08.59 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 11:10.28 | *** join/#asterisk Ast001 (n=uros@81.18.55.102) |
| 11:11.31 | Ast001 | hi I just bought digium TE121 card and I want to make it ready for E1. I can see that area for that in card has 2 pins and one jumper over only 1 pin. To make it ready for E1 I need to put jumper on both pins right ? |
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| 11:13.12 | BrokenNoze | Hi all, has anyone found a workaround for the attended transfer with MixMonitor? |
| 11:13.51 | BrokenNoze | all my recordings are being cut off when a transfer occurs!!! |
| 11:17.35 | JT | Devyll: i know what it is, i have no idea how flux relates |
| 11:18.50 | Ast001 | jumper on both pins for E1 on TE121 right ? |
| 11:19.35 | JT | Ast001: read the TE121 docs |
| 11:19.42 | JT | there is documentation you know |
| 11:19.54 | flenders | JT: hey |
| 11:19.56 | Ast001 | I read the docs and it said jumper must be on for E1 |
| 11:20.00 | flenders | JT: long time no see |
| 11:20.02 | flenders | :D |
| 11:20.15 | Ast001 | but docs did not say is that mean jumper on both pins or just on one like it is on card now |
| 11:20.56 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
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| 11:22.53 | JT | Ast001: what is a jumper that is not connected? off |
| 11:22.58 | JT | flenders: hey |
| 11:23.10 | flenders | JT: how's it going mate? |
| 11:23.26 | JT | not bad |
| 11:23.27 | Ast001 | i am not sure did you understand jumper is on one pin but another in free |
| 11:23.40 | Ast001 | another pin is free |
| 11:23.45 | JT | Ast001: what is a jumper that is on one pin? OFF |
| 11:24.51 | Ast001 | I dunno if you say so ok I will put jumper on both pins then I don't know why beroNet did not put that already or maybe custom office or DHL did play with my card |
| 11:25.18 | JT | Ast001: it's pretty obvious, if a jumper is not completing a circuit, it is off |
| 11:25.34 | Ast001 | <PROTECTED> |
| 11:25.52 | Devyll | JT: I actually need to know how can I connect E1 from the phone company to my pc/server with asterisk ? |
| 11:26.10 | JT | with an ethernet cable |
| 11:26.11 | *** part/#asterisk Ast001 (n=uros@81.18.55.102) |
| 11:26.21 | JT | or a T1/E1 crossover cable |
| 11:26.27 | JT | depending on how the outlet is pinned |
| 11:29.31 | Devyll | JT , so you are saying that if the phone company brigs the e1 through an optic fiber cable, there should be a divice such as a mediaconvertor , from which I will have ethernet cable to the server. is that correct ? |
| 11:29.41 | JT | yes |
| 11:30.03 | JT | the standard handoff is generally an RJ-48C connector |
| 11:31.38 | Devyll | JT, isn't rj-48c for t1 ? I need e1 (europe). |
| 11:32.40 | JT | same thing. |
| 11:32.43 | JT | pins are the same |
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| 11:39.26 | coppice | Devyll: you might have a pair of coax cables terminating an E1 in some places. You'll need a converter if you do, and few cards now offer coax connectors |
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| 11:56.12 | *** join/#asterisk sircco (n=sircco@dh207-69-209.xnet.hr) |
| 11:56.30 | sircco | how can i see type of user agent i.e. xlite in asterisk |
| 11:56.53 | sircco | astersk says this [Sep 15 13:56:08] -- Saved useragent "Grandstream GXP2000 1.1.6.16" for peer 40 |
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| 12:00.42 | *** part/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 12:01.32 | yang | Is there a function in Voicemail which enables you to delete all mesages at once? |
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| 12:06.22 | yang | sircco: you see it when the phone connects |
| 12:06.40 | yang | sircco: unplug it or disconnect and it will come back |
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| 12:19.42 | gego | Hello, is there anyone who could help me with AstManProxy? |
| 12:20.13 | gego | I use standard format and never get anything more than "Asterisk Call Manager Proxy/1.21" back |
| 12:20.57 | gego | sending Actions work - but i don't recv any Events |
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| 12:24.16 | angryuser | disposable, i have installed pfsense and set Qos queues for voip, most of the time it is working fine, they dont use internet that much, but in case of BIG load i still git the qulity problems |
| 12:24.30 | angryuser | *got |
| 12:24.48 | BrokenNoze | Hi, anyone any clues on MixMonitor and transfer, recordings being cut off when a call is transferred? |
| 12:25.49 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
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| 12:25.54 | dandre | <PROTECTED> |
| 12:26.20 | dandre | Is there anyone who knows the inner of activa tsp driver? |
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| 12:43.19 | Dr-Linux|home | App SpeechBackground doesn't detect my voice, wondering what could be the issue. |
| 12:45.21 | kkrzys | hello |
| 12:45.44 | kkrzys | sometimes i'm getting this during calls: [Sep 15 13:49:51] WARNING[14896]: chan_zap.c:8074 pri_fixup_principle: Can't fix up channel from 210 to 196 because 196 is already in use |
| 12:45.44 | kkrzys | [Sep 15 13:49:51] WARNING[14896]: chan_zap.c:8127 pri_fixup_principle: Call specified, but not found? |
| 12:45.44 | kkrzys | [Sep 15 13:49:51] WARNING[14896]: chan_zap.c:9245 pri_dchannel: Hangup on bad channel 0/31 on span 7 |
| 12:45.58 | kkrzys | does anyone know how to fix it? |
| 12:46.36 | [TK]D-Fender | Ahhh, no real loss in that mini-netsplit :) |
| 12:46.41 | [TK]D-Fender | Well... maybe Strom_C |
| 12:47.20 | Maliuta | yeah, I'm still here. So the party can continue ;) |
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| 12:57.55 | paulproteus | yes |
| 12:58.28 | frogonwheels_ | hey, somebody in #openwrt wants to know why we should use asterisk over freepbx.. |
| 12:59.14 | frogonwheels_ | anybody got any responses / experience with using freepbx |
| 12:59.15 | frogonwheels_ | ? |
| 12:59.34 | *** join/#asterisk stencil (n=stencil@206-248-132-65.dsl.teksavvy.com) |
| 12:59.59 | tzafrir_laptop | frogonwheels_, freepbx 2.5 has better support for sqlite |
| 13:00.22 | [TK]D-Fender | frogonwheels_: That funny... I didn't know you installed Asterisk OVER FreePBX. What do you do when you want to run FreePBX without *? |
| 13:00.38 | [TK]D-Fender | frogonwheels_: And you are in the wrong channel to ask that. |
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| 13:00.58 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:01.02 | [TK]D-Fender | frogonwheels_: And you want to try to dump a GUI, web & DB server onto a WRT? lol |
| 13:01.12 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 13:01.25 | tzafrir_laptop | [TK]D-Fender, who said anything about a DB server? |
| 13:01.48 | yang | new openwrt version does not have GUI |
| 13:01.50 | tzafrir_laptop | and any mini gadget nowadays has a web server |
| 13:01.59 | Dr-Linux|home | file: around? |
| 13:01.59 | frogonwheels_ | no - I mean prefer one over the other.. not having both .. |
| 13:02.00 | frogonwheels_ | :) |
| 13:02.05 | tzafrir_laptop | freepbx requires a CGI-capable web server |
| 13:02.08 | [TK]D-Fender | tzafrir_laptop: Well FreePBX typically saves all its stuff to DB and generates static configs from there but still uses the DB for other run-time stuff... |
| 13:02.17 | tzafrir_laptop | (as php can run as a GCI program) |
| 13:02.27 | yang | frogonwheels_: Kamikaze doesn't have GUI ! |
| 13:02.39 | tzafrir_laptop | [TK]D-Fender, db, but not db server. |
| 13:02.40 | frogonwheels_ | yeah - it certainly can |
| 13:02.53 | tzafrir_laptop | In other words: sqlite |
| 13:02.54 | frogonwheels_ | yang: it actually has a couple of choices |
| 13:03.08 | tzafrir_laptop | (one of the main sponsors of sqlite is Nokia) |
| 13:03.53 | [TK]D-Fender | tzafrir_laptop: Ah, I did mean some DB storage. Not meant to imply a secondary "server" |
| 13:03.58 | yang | running asterisk on openwrt must be fancy task |
| 13:04.07 | [TK]D-Fender | tzafrir_laptop: If it can run off SQLite, then more power to them |
| 13:04.09 | tzafrir_laptop | [TK]D-Fender, we actually managed to run freepbx once with the busybox httpd |
| 13:04.26 | [TK]D-Fender | tzafrir_laptop: * should take more advantage of SQLite IMO |
| 13:04.46 | frogonwheels_ | yang: the basics wasn't so hard - just avoid transcoding at all costs. |
| 13:05.17 | frogonwheels_ | and the performance is fine - I've got a device with a usb so storage for voicemail is easy enough |
| 13:05.29 | yang | :) |
| 13:06.11 | frogonwheels_ | I've had 2 incoming calls and one internal call in a meeting, while another incoming call was going on. |
| 13:06.15 | tzafrir_laptop | (and that said: freepbx is still a badly-designed program) |
| 13:06.18 | frogonwheels_ | which is not bad for a router. |
| 13:06.30 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:07.53 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:08.27 | mchou | frogonwheels_: you running asterisk on openwrt? |
| 13:08.35 | frogonwheels_ | yep |
| 13:08.50 | mchou | frogonwheels_: I'd like to pick yer brain. What HW? |
| 13:09.08 | mchou | meaning router |
| 13:09.11 | frogonwheels_ | Asus wl 500g premium v1 |
| 13:09.16 | mchou | bah |
| 13:09.32 | mchou | how much RAM does that thing have? |
| 13:10.13 | mchou | frogonwheels_: know anyone running it on wrt54g? |
| 13:10.33 | frogonwheels_ | no afaik |
| 13:10.33 | mchou | dont want any OOM 'issues' |
| 13:10.41 | frogonwheels_ | OOM ? |
| 13:10.49 | mchou | out of mememory |
| 13:10.54 | mchou | memory* |
| 13:11.10 | mchou | i.e. router crash |
| 13:11.37 | mchou | how much ram on asus? |
| 13:11.54 | frogonwheels_ | oh.. umm. |
| 13:12.53 | frogonwheels_ | 8mb |
| 13:12.54 | mchou | frogonwheels_: log in to your router, type 'free' lol |
| 13:12.57 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 13:13.01 | frogonwheels_ | no 32mb |
| 13:13.07 | mchou | sigh |
| 13:13.07 | frogonwheels_ | ahh free.. always forget that. |
| 13:14.13 | frogonwheels_ | mchou: and it has a usb port for swap if needed |
| 13:14.23 | mchou | yeah yeah |
| 13:14.27 | frogonwheels_ | though I'm not using any swap. |
| 13:14.48 | mchou | pastebin output from free |
| 13:14.53 | yang | frogonwheels_: freepbx would probably be more resource hungry than asterisk |
| 13:14.57 | frogonwheels_ | but with XMail and asterisk (meetme, voicemail loaded) |
| 13:15.29 | frogonwheels_ | I've not got much free |
| 13:15.52 | mchou | sigh. how much you got free? |
| 13:16.01 | frogonwheels_ | 1126400 bytes |
| 13:16.15 | yang | frogonwheels_: you can manage Asterisk via CLI don't need a httpd which freepbx requires |
| 13:16.40 | frogonwheels_ | oh yeah, and I'm running busybox httpd. |
| 13:16.55 | frogonwheels_ | yang: yep. but the nice thing will be that I have a 'uci' configuration loader for asterisk. |
| 13:17.19 | frogonwheels_ | which means you can set up asterisk in uci with either web (luci) or your fav editor. |
| 13:17.47 | mchou | frogonwheels_: pls pastebin your free output |
| 13:18.04 | mchou | frogonwheels_: getting data from you is like pulling teeth |
| 13:18.23 | yang | mchou: you will probably have a hard time on wrt54G as it has only 16MB of memory |
| 13:18.29 | yang | and no usb slots to expand |
| 13:18.42 | mchou | bah. who needs USB |
| 13:18.44 | dandre | <PROTECTED> |
| 13:18.52 | mchou | when you have NASD :) |
| 13:19.34 | yang | good solution |
| 13:20.08 | yang | mchou: do you also experience timeuts with wrt54g? |
| 13:20.28 | mchou | yang: I havent loaded asterisk on my router |
| 13:20.39 | yang | mchou: I mean with wireless connection? |
| 13:20.45 | mchou | that's why I'm asking ppl with experience |
| 13:20.46 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:20.51 | frogonwheels_ | http://pastebin.com/d7659b729 |
| 13:20.52 | mchou | yang: nope. never |
| 13:21.39 | mchou | yang: but I only use wireless on my lappy (I rarely us that, iow) |
| 13:21.49 | mchou | frogonwheels_: thank you |
| 13:22.54 | mchou | well, that pretty much means wrt54g is a no go |
| 13:23.11 | *** join/#asterisk dlewis (i=c7340d67@about/security/staff/dlewis) |
| 13:23.16 | mchou | ram is limiting factor |
| 13:24.14 | mchou | frogonwheels_: I thought you said you didnt have swap |
| 13:24.19 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 13:24.30 | frogonwheels_ | no - I said not much is in use. |
| 13:24.34 | *** join/#asterisk arpu (n=arpu@chello084113208186.3.14.vie.surfer.at) |
| 13:24.45 | mchou | frogonwheels_: ok, my bad |
| 13:25.23 | mchou | sigh. too bad |
| 13:26.17 | mchou | I had such high hopes |
| 13:26.49 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 13:27.05 | hi365 | (how) can I set more than one file in Dial with the A option? |
| 13:28.26 | *** join/#asterisk Corydon76-dig (i=gold@pdpc/supporter/bronze/Corydon76-home) |
| 13:28.26 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 13:29.05 | mchou | hi365: why not just concat the files? |
| 13:29.29 | hi365 | mchou: cause they get updated for every call (think: doorbell) |
| 13:30.08 | mchou | agi will do it :) |
| 13:30.45 | mchou | preprocess the files |
| 13:31.04 | hi365 | i guess, was hoping for something like & |
| 13:31.34 | mchou | well, I'm not saying & wont work. But I'm dubious |
| 13:31.50 | hi365 | I AM. it doesnt |
| 13:32.03 | hi365 | whishes thing were a bit more uniform in asterisk |
| 13:32.08 | hi365 | it doesnt work for read either |
| 13:32.15 | hi365 | but it works in playback |
| 13:33.05 | mchou | hi365: you ever heard of astycrapper? |
| 13:33.21 | mchou | it's marvelous |
| 13:33.30 | hi365 | looks at mchou with a puzzled look |
| 13:33.59 | mchou | http://www.linuxsystems.com.au/astycrapper/ |
| 13:34.14 | mchou | check out sample 1. It's HILARIOUS |
| 13:34.20 | *** join/#asterisk dlewis (i=c7340d67@about/security/staff/dlewis) |
| 13:34.45 | mchou | you can use a somewhat similar approach :) |
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| 13:36.18 | *** part/#asterisk dwayne (i=dwayne@76.29.245.9) |
| 13:39.08 | paulproteus | mcab, Does it work by detecting silence / pauses by the other person? |
| 13:39.24 | paulproteus | Oh, a link to the conf. |
| 13:39.29 | mchou | paulproteus: exactly |
| 13:39.30 | *** join/#asterisk wiscados (n=mint@81.25.184.155) |
| 13:39.37 | paulproteus | That's so sad! |
| 13:39.55 | mchou | why is that sad?? I think it's hilarious |
| 13:40.13 | mchou | telemarketer talks to machine |
| 13:40.15 | paulproteus | It exploits this crucial weakness in human communication. |
| 13:40.32 | paulproteus | It's also hilarious. |
| 13:40.34 | paulproteus | Don't get me wrong. |
| 13:41.13 | mchou | Kristi is so focused on her goal she misses the big picture :) |
| 13:41.17 | *** join/#asterisk stimpie (n=stimpie@84-104-5-14.cable.quicknet.nl) |
| 13:42.32 | paulproteus | I also *love* how the robotic Asterisk dude doesn't even know she's talking over him when she does. |
| 13:43.26 | mchou | "Come back to telstra!" |
| 13:43.32 | mchou | "Do it!" |
| 13:43.35 | paulproteus | NEVER!!!eleven |
| 13:43.46 | paulproteus | I'm having trouble hearing you there, can you speak up a little bit? |
| 13:43.50 | mchou | lol |
| 13:45.12 | Katty | morning |
| 13:45.13 | mchou | I love Grandpa Jordan |
| 13:45.52 | *** join/#asterisk SteveTotaro (n=Administ@pool-141-157-95-245.balt.east.verizon.net) |
| 13:48.49 | hi365 | heres: what im trying to do: have the caller record their name, and then hear ringing. at the same time have an extensions calld and play the recording when answered |
| 13:49.05 | hi365 | then offer the aption to bridge the two |
| 13:49.34 | hi365 | if i were to do it with a .call file it seems kinds simple, but I belive there is a way to do it directly from the dialplan... |
| 13:49.38 | mchou | hi365: copy GrandCentrals asterisk configuration :) |
| 13:49.54 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 13:50.05 | mchou | hi365: that's exactly what GrandCentral does |
| 13:50.19 | mchou | well, more or less |
| 13:50.35 | hi365 | assuming that they use asterisk... |
| 13:50.45 | mchou | they use asterisk |
| 13:50.58 | hi365 | so where's the dialplan? :) |
| 13:51.00 | mchou | that's the user agent ID |
| 13:51.18 | hi365 | they probably use an agi... |
| 13:51.18 | mchou | hi365: go ask grandcentral |
| 13:52.16 | hi365 | let me rephrase my question: how can i start another leg of a call without bridging it with the first? |
| 13:54.57 | mchou | Dial with M option? |
| 13:57.37 | *** join/#asterisk Psychobilly (n=moi@adsl59-158.kln.forthnet.gr) |
| 13:57.45 | *** join/#asterisk Danskmand (n=danskman@p4FD3C7C7.dip.t-dialin.net) |
| 13:58.16 | *** join/#asterisk DarylVOIP (n=daryl@75.147.121.177) |
| 13:58.33 | Danskmand | Glut ? - are you there ? |
| 13:58.53 | *** join/#asterisk fogo (n=Paul@69.169.132.200) |
| 14:00.40 | *** join/#asterisk rgsteele||work (n=rgsteele@75.147.74.137) |
| 14:00.51 | rgsteele||work | Erm, what's the name for PBX cards that accept POTS lines? |
| 14:01.28 | Danskmand | ,or, maybe someone else that has realized a faxserver (send/receive and convert to email) ? |
| 14:02.27 | jefftucson | rgsteele: FXO cards.. |
| 14:04.00 | Dr-Linux|home | anybody use LumenVox SR with asterisk? |
| 14:05.02 | *** join/#asterisk Steak__ (n=Alex@pub212004070190.fx-hfc.datazug.ch) |
| 14:05.05 | Steak__ | hello |
| 14:05.45 | *** join/#asterisk dlewis (i=c7340d67@about/security/staff/dlewis) |
| 14:06.02 | Steak__ | a quick question... I am going to set up a PBX at my new home, I would like to get an FXO gateway so that I can connect the PSTN line.. any advice on what to buy? I would like something standalone... thanks! |
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| 14:08.58 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 14:09.07 | jefftucson | Steak__: i like th Rhino R8FXX-EC series of cards. |
| 14:09.25 | Steak__ | is that a PCI card or a standalone box? |
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| 14:09.26 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:09.43 | Steak__ | because I would prefer a selfstanding device |
| 14:09.47 | *** join/#asterisk dramman (n=Miranda@122.111.56.35) |
| 14:10.10 | dramman | What's the default password for the "asterisk" user? |
| 14:10.38 | jefftucson | Steak__: it's pci, i've never used a standalone fxo device |
| 14:10.44 | Steak__ | ok thanks |
| 14:11.23 | Steak__ | I am trying to avoid PCI cards because this would mean to have a server in the basement, which is something I would like to avoid - but thanks anyways :) |
| 14:11.35 | RoadKillDK | is ther anybody that know how to pickup at Queue with out joining it |
| 14:11.39 | Steak__ | (or to move the POTS outlet somewhere else :P) |
| 14:12.31 | tzafrir_laptop | Steak__, hmmm... you can send telephony wires there just as much as you can send ethernet wires there |
| 14:12.53 | Steak__ | well the fact is that I want to reduce the wired network to a real minimum |
| 14:13.01 | Steak__ | and try to get wireless at every floor |
| 14:13.22 | Steak__ | because this house hasn't got a cable conduit |
| 14:15.59 | Steak__ | otherwise I think I will play it dirty and get a FXO/FXS converter |
| 14:17.22 | [TK]D-Fender | Steak__: Linksys SPA-3102 <- |
| 14:17.38 | [TK]D-Fender | dramman: No such thing |
| 14:18.19 | Steak__ | [TK]D-Fender, does the SPA-3102 plays nicely with asterisk? |
| 14:18.27 | [TK]D-Fender | Steak__: Yes |
| 14:18.30 | Dr-Linux|home | App SpeechBackground doesn't detect my voice, wondering what could be the issue. |
| 14:18.33 | Dr-Linux|home | -- Executing [s@lumenvox:5] SpeechBackground("SIP/4092-b7d3da |
| 14:18.47 | Dr-Linux|home | it gets stuck here , doesn't go to next priority |
| 14:19.18 | Steak__ | because I do not have particular needs, I only have 1 PSTN line, but I would like to have all the phones at home to be SIP, so that I can setup dial plans/routes that act differently depending on the destination... |
| 14:20.52 | [TK]D-Fender | Steak__: So you have 1 line, and a bunch of phones? |
| 14:20.58 | Steak__ | it would be like that, yes |
| 14:21.12 | [TK]D-Fender | Steak__: SPA-3102 is all you need for your whole home then |
| 14:21.21 | Steak__ | but the phones/PABX can decide which route to take (internet/PSTN) |
| 14:21.40 | Steak__ | and of course I want to handle incoming calls and the ability to forward those calls to all the internal phones |
| 14:21.44 | [TK]D-Fender | Steak__: Plug it in at the demarc for the PSTN (FXO) port, and the rest of yourhome gang-piled onto the FXS port |
| 14:22.23 | Steak__ | and the SIP phones through the network, I guess :D |
| 14:22.30 | Steak__ | there will be only one analog phone in the whole system |
| 14:22.37 | Steak__ | all the others will be SIP |
| 14:22.42 | *** join/#asterisk bkw_ (n=bkw_@freeswitch/developer/bkw) |
| 14:22.46 | bkw_ | ev |
| 14:22.54 | bkw_ | woops |
| 14:23.00 | Danskmand | Noone can help me ? |
| 14:23.16 | bkw_ | Danskmand: what was your problem again? |
| 14:23.31 | Danskmand | (16:01:28) Danskmand: ,or, maybe someone else that has realized a faxserver (send/receive and convert to email) ? |
| 14:23.50 | [TK]D-Fender | Danskmand: Go lookup Hylafax & IAXmodem on the WIKI |
| 14:23.52 | [TK]D-Fender | ~wikis |
| 14:23.53 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 14:23.53 | bkw_ | isn't their the astfax project? |
| 14:24.24 | Danskmand | I've got the feeling that Hylafax is pretty old, isnt it ? |
| 14:24.29 | bkw_ | nope |
| 14:24.31 | bkw_ | http://www.voip-info.org/wiki/view/astfax |
| 14:24.39 | bkw_ | Hylafax is rather awesome |
| 14:24.41 | Danskmand | Plus, I would like to use mISDN... |
| 14:25.05 | [TK]D-Fender | Steak__: The method I described would let you use all of your existing wiring on a singe ATA port. You could of course add other SIP hard phones etc into the mix as well, but not necessary |
| 14:25.13 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 14:25.32 | [TK]D-Fender | Danskmand: Hylafax doesn't care what your call comes in on. It'd just better have good integrity |
| 14:25.44 | Steak__ | [TK]D-Fender, thanks... The fact is that I will try to avoid analogue phones all around, just keeping one in the basement near the FXO for emergencies |
| 14:26.12 | *** join/#asterisk tobias (n=tobias@user-0c2hj2f.cable.mindspring.com) |
| 14:26.13 | [TK]D-Fender | Steak__: thats what the 3102 is good at as well. If the poewr goes out it will bridge the FXS & FXO ports directly |
| 14:26.15 | Steak__ | so the thing will be : 1x POTS line - 1x analogue phone - 4-5x VoIP phones (either wifi or wired) |
| 14:26.31 | [TK]D-Fender | Steak__: Ok, then this is definitely the solution for you |
| 14:26.41 | [TK]D-Fender | @ 75$ USD |
| 14:26.44 | Steak__ | looks like :) and it is even quite cheap :D |
| 14:26.48 | [TK]D-Fender | (maybe 70 if you look real hard) |
| 14:26.51 | Steak__ | no need for 300$+ cards |
| 14:26.56 | [TK]D-Fender | Steak__: nope. |
| 14:27.05 | Steak__ | well I am in Switzerland, but the price is more or less the same |
| 14:27.08 | Steak__ | but it's ok |
| 14:27.26 | *** join/#asterisk Laureano (n=Laureano@190.245.108.2) |
| 14:27.44 | [TK]D-Fender | Steak__: Best option no matter where you are really.. |
| 14:27.55 | Steak__ | yep :) thanks for the help |
| 14:27.59 | Steak__ | was really appreciated :D |
| 14:29.53 | Steak__ | then ok it will be dialplan nightmare... but that I will leave when I will have moved to the new house (read: next week) |
| 14:30.29 | [TK]D-Fender | Steak__: For a HOME? Dialplan is hardly a "nightmare" unless you're hooking up a few dozen ITSP, etc |
| 14:31.24 | Steak__ | well let me sum it up... normal POTS/PSTN, skype (with a converter), VoIPbuster, direct calls... it's more a directory work than anything else... |
| 14:31.52 | Steak__ | then maybe a GSM gateway in the future |
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| 14:32.31 | Steak__ | don't know how it will be anyways :P |
| 14:33.16 | [TK]D-Fender | Steak__: I think you'll do jsut fine, though it may seem a little cludgy during the initial setup |
| 14:33.45 | Steak__ | Since I do not know Asterisk so much, I will anyways go in steps.. the first one will be using the phone line as the default route for external calls |
| 14:34.25 | Steak__ | so that you have to dial 0 to get the tone and then you can call wherever you like, plus receiving calls (maybe on a hunt group schema) |
| 14:34.33 | Steak__ | then I will add the other functionalities one by one |
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| 14:37.23 | gego | I'd like to access AMI via astmanproxy - but i don't get much action response to the client |
| 14:38.05 | gego | I always receive "Asterisk Call Manager Proxy/1.21" - nothing more - any hints? |
| 14:38.07 | *** join/#asterisk stencil (n=stencil@206-248-132-65.dsl.teksavvy.com) |
| 14:39.59 | gego | I can originate a call that way - but I'd like to see what's happening on client side - and monitor other SIP-accounts on the same * |
| 14:41.36 | [TK]D-Fender | gego: And if you poiont it directly to * it works, just not via the proxy? |
| 14:43.34 | gego | [TK]D-Fender - I haven't tried - wanted to avoid http-requests |
| 14:43.56 | [TK]D-Fender | gego: AMI is not HTTP. |
| 14:44.21 | [TK]D-Fender | gego: astmanproxy is for direct TCP, not HHTP Web2.0. Forget AJAM |
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| 14:45.00 | CrashSys | AJAM? |
| 14:46.34 | gego | [TK]D-Fender - I access astmanproxy by direct sockets in python - wouldn't know how to do this with AMI direct |
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| 14:50.23 | Steak__ | ok bye |
| 14:50.27 | *** part/#asterisk Steak__ (n=Alex@pub212004070190.fx-hfc.datazug.ch) |
| 14:51.02 | jblack | Wow. Maybe google is getting too big. |
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| 14:53.10 | tzafrir_laptop | gego, telnet to it |
| 14:53.10 | Nugget | telnet is eeeeeeevil! |
| 14:53.16 | mort_gib | Quick question. My clients are asking for "Call manages" I have found Asterisk Dialer from www.voip.com.sg but are there any others?? |
| 14:54.21 | mort_gib | I'm looking at Tikal's offering... |
| 14:54.32 | [TK]D-Fender | gego: AMI is direct TCP, not via HTTP-faked out junk |
| 14:55.56 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 14:56.30 | gego | tzafir_laptop in telnet i get a response |
| 14:58.33 | tzafrir_laptop | gego, see example session in http://voip-info.org/wiki/view/Asterisk+Manager+API |
| 14:58.39 | tzafrir_laptop | well, example login |
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| 15:00.19 | gego | tzafir_laptop I can login, send Actions - but don't get much response - maybe my python code sucks. |
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| 15:01.09 | gego | tzafir_laptop Anyway - thanks for your help - I've got go now |
| 15:01.31 | tzafrir_laptop | gego, then don't trust it. There should already be some python implementation of access to the AMI, IIRC |
| 15:02.41 | tzafrir_laptop | http://www.voip-info.org/wiki/view/Asterisk+manager+Examples |
| 15:03.11 | sant0sk1 | I am trying to get sms working w/ my asterisk box (1.4.21.2) and I've compiled the smsq utility for testing. When I generate an sms from the command line I get the following output "No call scheduled as already sending". The sms does not send. I have 'load=app_sms.so" in modules.conf... anything else I'm missing? |
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| 15:07.28 | bkw_ | sant0sk1: you do know thats fixed line sms and not mobile sms? |
| 15:07.49 | *** join/#asterisk Rico29 (n=Rico@static-120-146.blueline.mg) |
| 15:07.50 | sant0sk1 | bkw_: no I didn't know that. |
| 15:08.02 | bkw_ | if you're wanting to send to mobile phones.. thats not the app you wanna use ;) |
| 15:08.16 | sant0sk1 | ok, what app do I want to use? |
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| 15:11.11 | sant0sk1 | is there a de facto? |
| 15:11.28 | tzafrir_laptop | sant0sk1, sms got a major fix a while after trunk was forked from 1.4 . The one in 1.4 has indeed many problems, IIRC |
| 15:11.59 | tzafrir_laptop | app_sms and smsq are for fixed-line |
| 15:12.33 | tzafrir_laptop | I'm not sure they use that in the US |
| 15:12.43 | sant0sk1 | tzafrif_laptop: yah fixed line was not what I was looking for... |
| 15:13.48 | sant0sk1 | does 1.6 have mobile sms built in? |
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| 15:14.17 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:15.40 | *** join/#asterisk ManxPower (n=manxpowe@53.sub-75-251-89.myvzw.com) |
| 15:16.57 | glut | Danskmand: present |
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| 15:31.44 | jesselang|laptop | Hello folks. Can I use Dial to send a call to a particular context on the target box? |
| 15:32.25 | *** join/#asterisk imcdona (i=imcdona@imcdona.broker.freenet6.net) |
| 15:32.26 | ManxPower | jesselang|laptop: only with IAX2 |
| 15:32.34 | jesselang|laptop | I seem to recall that it's possible, but I can't find details anywhere. |
| 15:32.44 | ManxPower | jesselang|laptop: for all other protocols, the context is not set in the Dial, only in the sip peer/user/friend |
| 15:32.47 | jesselang|laptop | ManxPower, yes, this is IAX2. |
| 15:33.16 | jesselang|laptop | How can I accomplish this using Dial to an IAX2 peer? |
| 15:33.41 | ManxPower | jesselang|laptop: Dial(IAX2/iax2conf/number@remotecontext |
| 15:33.58 | ManxPower | the remotecontext must match what is on the iax2.conf user on the far box. |
| 15:34.43 | ManxPower | jesselang|laptop: almost nobody specifies the remote context on the local Dial line. |
| 15:34.46 | jesselang|laptop | ManxPower, in this case, I'm trying to send the call to a non-default context. |
| 15:35.19 | jesselang|laptop | Can I specify multiple contexts in the target system's iax.conf? |
| 15:35.21 | ManxPower | jesselang|laptop: there is nothing different here bot IAX2 .vs. other protocols. |
| 15:35.52 | ManxPower | jesselang|laptop: you do not ACTUALLY specify the remote context, you request the remote context, if the far end allows the incoing user to access that context then you can, if not it will be rejected. |
| 15:36.05 | jesselang|laptop | Or do I need to create a separate peer connection? |
| 15:36.21 | jesselang|laptop | Can the remote system allow multiple contexts? |
| 15:36.32 | ManxPower | jesselang|laptop: yes, a seperate peer on the remote box with the the context= matching the incoming requested context. |
| 15:36.45 | ManxPower | jesselang|laptop: Have you read ATFOT? |
| 15:37.08 | jesselang|laptop | I've browsed it. |
| 15:37.15 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 15:37.17 | ManxPower | go read it. |
| 15:37.54 | jesselang|laptop | Thanks for the help. |
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| 15:58.14 | errr | is there any messages in logs or something that would let me know if I am using all my channels on my pri? |
| 15:59.00 | *** join/#asterisk dlewis (i=c7340d67@about/security/staff/dlewis) |
| 15:59.06 | ManxPower | errr: Dial will return CONGESTION if you try dialing out and all the channels in that group are in use. |
| 15:59.28 | errr | ManxPower: what about incoming calls? |
| 15:59.43 | ManxPower | errr: what happens is TOTALLY up to the telco. |
| 15:59.49 | errr | hmm |
| 16:00.31 | ManxPower | I usually set my group= to be a few channels less than I have, that way there is (almost) always some channels available for incoming calls. |
| 16:00.44 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 16:01.06 | errr | ManxPower: see the problem we are having is, thanks to ike we had to do an emergency reroute of our Houston PRI to our San Antonio office. Sometimes we are getting (when calling in) this call cant be completed as dialed or number not in service |
| 16:01.38 | ManxPower | errr: that would not be an "out of channels on your PRI" problem. |
| 16:01.49 | errr | and I cant tell if we are using all our channels or if its something with the telco just being under such a load |
| 16:01.53 | errr | ok |
| 16:02.00 | ManxPower | That would be "there was a hurricane -- all the telcos are screwed up. |
| 16:02.08 | errr | ok |
| 16:02.19 | *** join/#asterisk Zenos (n=chatzill@66.155.138.67) |
| 16:02.25 | ManxPower | normally "all circuits are busy" message is what the caller would get if they dialed in and your pri was full. |
| 16:02.44 | errr | ok |
| 16:02.57 | *** part/#asterisk sant0sk1 (n=sant0sk1@wsip-68-15-235-89.om.om.cox.net) |
| 16:03.00 | errr | thanks |
| 16:03.31 | errr | since we are an insurance company our phones are ringing off the hook with people filing claims :( |
| 16:03.58 | Zenos | owch bummer |
| 16:04.04 | ManxPower | errr: can you send most outgoing calls over IP to an ITSP, that will free up channels too. |
| 16:04.16 | errr | ah good idea |
| 16:04.38 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:04.38 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:04.51 | *** part/#asterisk jesselang|laptop (n=jesse@h75-100-162-159.mntimn.dsl.dynamic.tds.net) |
| 16:06.07 | ManxPower | errr: My largest customer has offices all over Louisiana. I've don't quite a bit of rerouting of calls over the past couple of years. |
| 16:06.54 | *** join/#asterisk wacky_ (n=abourget@mtl.savoirfairelinux.net) |
| 16:07.09 | wacky_ | is it possible to have IAX2 trunk=yes and also have jitterbuffer enabled ? |
| 16:07.18 | russellb | yes |
| 16:07.25 | wacky_ | since when version ? |
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| 16:08.23 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 16:08.50 | wacky_ | is it possible it was working clunkily in Asterisk 1.4 ? |
| 16:09.03 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 16:09.28 | ManxPower | wacky_: 1.2 was the first version to have an iax2 JB, IIRC, so it would be working since then. |
| 16:09.51 | ManxPower | what SPECIFIC problem are you having? |
| 16:11.29 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
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| 16:12.24 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 16:12.46 | wacky_ | hmm.. I don't have a specific problem yet. We have servers with trunk disabled because of jitter, as we thought they were mutually exclusive or something.. |
| 16:13.07 | wacky_ | I'll have a look into it.. if you say it should work :) thanks a lot.. |
| 16:14.46 | wacky_ | JT: are you the one who reported about the IAX2 trunk performances and bandwidth usage on voip-info ? |
| 16:14.55 | Katty | stretchy |
| 16:15.00 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:15.08 | Katty | [TK]D-Fender: what do you think about the name Gotham |
| 16:15.14 | Katty | [TK]D-Fender: possibly Gotem. |
| 16:15.19 | Katty | [TK]D-Fender: Gothem |
| 16:15.45 | wacky_ | I'm looking for that `rate` program you seem to have used, is it a sub-command of tcpdump or something ? |
| 16:16.06 | Katty | jbot: Gothem? |
| 16:16.11 | Katty | jbot: Gotham? |
| 16:16.25 | Katty | jbot: hopeless. |
| 16:17.02 | viraptor | is there any reason, I could get dtmf logged correctly for a channel (rfc2833), but nothing goes into Read() result? it happens from time to time... |
| 16:17.44 | Katty | ManxPower: what do you think of the name Gotham? |
| 16:18.56 | Zenos | This question is for anybody, I am running Asterisk 2.6 on a 2.0 GHz Pentium 4 with 1Gig of ram and 100 Gig HDD. Ubuntu 8.04 is the base OS and I really am using it in a SMB environment so no more than two concurrent calls max. No matter the load on the box the voicemail prompts sound weak and distorted. Is their any way to improve this? |
| 16:18.57 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.13) |
| 16:19.54 | tzafrir_laptop | Zenos, what phones do you use? |
| 16:20.15 | tzafrir_laptop | and you're not running asterisk 2.6 :-) |
| 16:20.56 | tzafrir_laptop | 1.6.0-rc<?> ? |
| 16:21.05 | *** part/#asterisk korihor (n=korihor@190.78.32.60) |
| 16:21.20 | Zenos | sorry typo 1.6 with nortel I2004 phones |
| 16:21.31 | wacky_ | ManxPower: I read on the wiki "IAX2 jitter buffer (when turned on) doesn't currently work well with trunking" |
| 16:21.36 | tzafrir_laptop | Are those SIP phones? |
| 16:21.38 | wacky_ | and then "(2008-07-16 update: jitter buffering works much better these days - JT) " |
| 16:21.39 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:21.49 | Zenos | no they use UNISTIM |
| 16:22.09 | wacky_ | so I was wondering what was true and which version of Asterisk have those better working jitterbuffer stuff ? |
| 16:22.58 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 16:23.20 | ManxPower | wacky_: well stop reading a page with old and outdate information |
| 16:23.30 | wacky_ | JT: found the rate program! thanks (it was linked at the bottom) |
| 16:24.29 | ManxPower | wacky_: in 1.4.?? (maybe .12?) major IAX2 improvements happened. But you are not running an old version of 1.4 with all the unpateched security holes, right? |
| 16:24.38 | Zenos | the phones don't have anything to do with what an incoming caller hears when interacting with voicemail as far as I understand am I wrong? |
| 16:25.08 | ManxPower | Zenos: correct |
| 16:25.13 | wacky_ | of course not :) |
| 16:26.35 | ManxPower | Zenos: the only "phone issue" could be a DTMF mode mismatch, but that's only for DTMF, not audio. |
| 16:27.35 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 16:27.56 | Zenos | ok, so would the voicemail prompts the sound files that asterisk is playing to the caller be improved if I modified some setting or maybe I should just record new voicemail IVR files? |
| 16:29.01 | [TK]D-Fender | Zenos: There is a known issue with the compiling of the GSM codec which is quite possibly your problem. This occurs with GCC 4.+ |
| 16:29.06 | [TK]D-Fender | 4.2+ |
| 16:29.55 | jeev | fender, gimme high 5 |
| 16:30.02 | Zenos | Really, ok, I will look into the version of GCC I used to compile asterisk |
| 16:30.13 | Zenos | thanks D-Fender |
| 16:33.50 | *** join/#asterisk tvirus (i=TheVirus@c-68-54-165-28.hsd1.md.comcast.net) |
| 16:34.27 | tvirus | Is it possible to do reporting in Asterisk? Such as calls per user, total minutes used, etc? |
| 16:34.55 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
| 16:36.11 | ManxPower | tvirus: You must have missed the CDR information in ATFOT |
| 16:36.58 | rene- | hello, is it possible to produce auto answer of an x-lite softphone from the dial plan using sip info headers? is it is possible? what sip info one must use? |
| 16:37.25 | Qwell | anybody happen to know what key on a cell phone gives you a 'p'? |
| 16:37.37 | tvirus | Ah, that's what it's called, CDR. |
| 16:37.41 | tvirus | Thanks ManxPower :) |
| 16:37.43 | Qwell | nevermind, it's under the menu |
| 16:38.44 | *** join/#asterisk kovger (n=kovger@217.27.219.70) |
| 16:38.48 | jablko | is there a command line tool which can play .gsm files? |
| 16:39.00 | viraptor | jablko: `play` |
| 16:39.10 | viraptor | (sox packag) |
| 16:39.19 | *** join/#asterisk dlewis (i=c7340d67@about/security/staff/dlewis) |
| 16:39.56 | *** join/#asterisk javb (n=javb@190.80.203.124) |
| 16:40.38 | javb | hello, i planning to take the dCAP exam, without taking the bootcamp, and i wonder if i could find some advices around here on how to succesful pass the exam.. |
| 16:41.08 | Qwell | javb: take the bootcamp first :) |
| 16:41.09 | russellb | javb: do you already know a lot about asterisk? |
| 16:41.10 | outtolunc | i hear muffins help <G> {just kidding} |
| 16:41.20 | russellb | if not, you're going to fail ... it's not a trivial exam |
| 16:41.22 | Qwell | to be honest, I don't know if I could pass the dCAP... |
| 16:41.28 | russellb | Qwell: no kidding |
| 16:42.01 | *** join/#asterisk L-info (n=L-info@80.169.36.194) |
| 16:42.05 | javb | It is not that i ` am totally a GURU, but yes, i have 5 years dealing with Linux and around 2.5 with Asterisk . . . |
| 16:42.16 | javb | I dont work with AGI and aMI, and DUNDI .. |
| 16:42.28 | javb | But, i think it is not in the PRACTICAL exam. |
| 16:42.33 | javb | But in the written |
| 16:44.39 | jablko | viraptor: @sox awesome - works great - thanks! |
| 16:45.09 | rene- | written is harder, i took it in 2005 |
| 16:45.13 | rene- | read the book a lot |
| 16:45.15 | rene- | really |
| 16:45.39 | rene- | they ask ridiculous questions sometimes but if u study the book really hard you should be ok |
| 16:46.43 | rene- | no agi, or ami in the DCap certification |
| 16:46.54 | javb | rene- sure sure? :p |
| 16:47.10 | rene- | a coworker just took the 2008 version of the exam |
| 16:47.41 | rene- | they still ask tricky questions so you must really study the asterisk book |
| 16:47.50 | rene- | agi and ami |
| 16:47.57 | javb | rene- perfect. what about the practical exam? |
| 16:48.25 | rene- | they depend on an external language, and that could be anything really |
| 16:48.30 | *** join/#asterisk soro (n=soro@201009072042.user.veloxzone.com.br) |
| 16:48.30 | rene- | practical exam is not hard |
| 16:48.44 | rene- | if u have done asterisk for 2.5 years you should pass it easily |
| 16:48.57 | rene- | most people will fail the written one so study hard |
| 16:49.11 | javb | Well, thanks for you info. :p |
| 16:49.25 | rene- | sure |
| 16:50.25 | *** part/#asterisk soro (n=soro@201009072042.user.veloxzone.com.br) |
| 16:53.07 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 16:53.13 | *** join/#asterisk fakhir (i=a7ce8021@gateway/web/ajax/mibbit.com/x-2679e0878fbe0008) |
| 16:55.01 | viraptor | anyone? dtmf present in verbose/dtmf output (along with channel name) but missing from "Read()" result?... help? :) |
| 16:58.18 | *** part/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 17:00.22 | [TK]D-Fender | viraptor: make sure you Answer the call first and do at least a Payback of silence minimum |
| 17:01.00 | viraptor | [TK]D-Fender: it's Answer() | Playback(conf-getpin) | Read() |
| 17:01.24 | [TK]D-Fender | viraptor: PASTEBIN your attempt with full debug. |
| 17:03.41 | viraptor | [TK]D-Fender: just a sec... |
| 17:04.10 | viraptor | I know it works in general, because it works in 95% - fails only sometimes |
| 17:05.05 | *** join/#asterisk tristanbob_ (n=tristanr@ubuntu/member/tristanbob) |
| 17:05.12 | tristanbob_ | anyone live taking astricon? |
| 17:05.27 | tristanbob_ | I'll go to #astricon |
| 17:05.37 | *** join/#asterisk jtodd (i=_squid@c-76-27-193-118.hsd1.or.comcast.net) |
| 17:08.14 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 17:09.50 | The-Bat | Hie |
| 17:09.56 | The-Bat | Is there any command which would give me the name of the queue from where the call has come ..?? |
| 17:11.16 | viraptor | [TK]D-Fender: http://gradwell.pastebin.com/d577fb817 <- it's that person entering 2414 |
| 17:11.31 | viraptor | but there's 'User entered nothing' at the end |
| 17:11.53 | viraptor | and in dtmf they're actually connected to the jittermeetme-a2ea |
| 17:12.21 | *** join/#asterisk remibemol (n=remibemo@was59-1-82-225-136-24.fbx.proxad.net) |
| 17:12.58 | *** join/#asterisk fakhir (i=a7ce8021@gateway/web/ajax/mibbit.com/x-991425205c97f481) |
| 17:17.17 | Zenos | [TK]D-Fender, I checked the version of GCC and sure enough it is 4.2 so that is why my voicmail prompts sound bad. So in your opinion if I downgrade the version of GCC and recompile asterisk it should be ok? |
| 17:18.14 | *** join/#asterisk moy (n=moy@189.169.91.147) |
| 17:19.11 | rene- | hey moy |
| 17:19.13 | rene- | chidos ese |
| 17:20.01 | moy | rene-: whos this? |
| 17:20.10 | rene- | user of openr2 |
| 17:20.16 | rene- | maybe u are not that moy |
| 17:20.18 | moy | ah :) |
| 17:20.33 | moy | yup, I am that moy ... where are you using it? |
| 17:20.38 | rene- | cancun mx |
| 17:20.46 | rene- | i have like 6 trunks |
| 17:20.48 | moy | ah, vientos, mejor aun heh |
| 17:20.49 | rene- | works fine |
| 17:20.51 | rene- | si |
| 17:21.16 | moy | did I talk to you before on google talk or msn? or is it this the first time we talk? |
| 17:21.27 | rene- | i think we talked before over msn |
| 17:21.36 | rene- | we wanted some chan_spy fix |
| 17:21.44 | moy | ah, yeah, I remember that now |
| 17:21.48 | rene- | we found a workaround eventually |
| 17:22.24 | moy | nice to hear that, well, don't hesitate on contacting me if you have any issue with openr2 |
| 17:22.31 | rene- | thanks i will |
| 17:29.30 | [TK]D-Fender | Zenos: Yup. |
| 17:34.48 | javb | Zt-dummy gives clock to asterisk in the absent of a card? But, ztdummy gives clock taken from? what does asterisk needs clock for? Whats the meaning of needing a clock? |
| 17:35.53 | [TK]D-Fender | javb: Meetme, page, IAX2 Trunking all need a timing source |
| 17:36.05 | [TK]D-Fender | javb: ZTDUMMY runns off linus 2.6 RTC |
| 17:36.52 | Zenos | [TK]D-Fender, Thanks again! |
| 17:38.20 | javb | [TK]D-Fender, "javb: ZTDUMMY runns off linus 2.6 RTC" ... i`m sorry, you mean it is not longer needed in 2.6? (sorry, english is not my mother tongue) |
| 17:41.41 | [TK]D-Fender | javb: Linux Kernel 2.6+ Reat Time Clock |
| 17:41.52 | [TK]D-Fender | Real* |
| 17:42.20 | [TK]D-Fender | javb: Pis je le sait... |
| 17:44.12 | *** join/#asterisk bradleyprice861 (n=bradleyp@fw.datafax.net) |
| 17:44.16 | viraptor | [TK]D-Fender: any ideas on dtmf? |
| 17:44.32 | *** part/#asterisk bradleyprice861 (n=bradleyp@fw.datafax.net) |
| 17:44.50 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
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| 17:52.07 | *** part/#asterisk Zenos (n=chatzill@66.155.138.67) |
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| 17:52.56 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
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| 17:57.47 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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| 17:59.30 | *** join/#asterisk dlewis (i=c7340d67@about/security/staff/dlewis) |
| 18:07.15 | justdave | Do Polycom phones have something that lets you change which server settings are used based on a dial prefix or something? |
| 18:08.16 | *** join/#asterisk dlewis (i=c7340d67@about/security/staff/dlewis) |
| 18:08.28 | [TK]D-Fender | justdave: HUH? |
| 18:08.37 | ManxPower | justdave: no |
| 18:08.54 | jjshoe | anyone get counterpath's softphone to auto-off hook from adding a sip header or anything of the sort? |
| 18:08.58 | [TK]D-Fender | viraptor: Nope, but I don't see the original inbound channel. I suspect that though you are receiving rfc2833 it might not be coming in on a peer set to care about it |
| 18:09.01 | ManxPower | but you can per line appearance and there is a failover server setting too. |
| 18:09.10 | justdave | asterisk says my phone is getting an authfail when I try to make a call |
| 18:09.18 | justdave | but only when I call certain numbers |
| 18:09.20 | [TK]D-Fender | justdave: Then you set it up wrong |
| 18:09.28 | justdave | calling other numbers works fine |
| 18:10.35 | dlewis | [TK]D-Fender: how easy is it to hook up a door phone to asterisk? |
| 18:10.38 | dlewis | one of these: http://www.smarthomeusa.com/ShopByManufacturer/Panasonic/Item/KX-T30865/?adwatcher=3 |
| 18:11.16 | [TK]D-Fender | dlewis: Requires KX-TA82461 Door/opener card and the KX-TA624-5 or KX-TA824 phone system <--- you tell me. I call this "big print" |
| 18:11.35 | justdave | config on this phone hasn't changed in a year or two, and it's always worked fine |
| 18:11.44 | justdave | never tried calling this particular number from it though |
| 18:11.46 | ManxPower | [TK]D-Fender: it means it only works with Toshiba (or is it Panasonic?) PBXs |
| 18:11.59 | [TK]D-Fender | justdave: That comment doesn't add much of value. debug info please... |
| 18:12.03 | justdave | *81 with 11 more digits after it |
| 18:12.09 | ManxPower | justdave: it is all in the dialplan on the phone. start by pasting that one line. |
| 18:12.11 | justdave | starting with an * shouldn't break it would it? |
| 18:12.30 | [TK]D-Fender | justdave: Stop wasting time with empty descriptions and show the DEBUG |
| 18:12.32 | ManxPower | justdave: the answer to that question is "it depends on the phone dialplan" |
| 18:12.43 | justdave | if the phone's dialplan was blocking it I'd think it would never try to connect to the server |
| 18:12.44 | dlewis | [TK]D-Fender: I didn't send that link to say THAT specific one will work, I was wondering if a door phone would work with Asterisk... I assume it would just be another extension, but wanted to get your opinion. |
| 18:12.55 | justdave | looks for the phone config |
| 18:13.17 | justdave | that would make sense except for the authfail message on the asterisk console when I try it :) |
| 18:13.26 | [TK]D-Fender | dlewis: Don't assume, and thats like saying "can a phone work with *"? The answer is a giant DEPENDS. |
| 18:14.30 | dlewis | gotta love google... |
| 18:14.31 | dlewis | http://nerdvittles.com/wp-content/doorbell.pdf |
| 18:15.52 | justdave | and I try to test it again by hitting redial and it works :P |
| 18:16.25 | viraptor | [TK]D-Fender: well - it get's accepted into the channel - same as the rest of dtmf-s that are handled correctly - here's dtmf log: http://gradwell.pastebin.com/d52623816 |
| 18:16.27 | justdave | direct dialing it fails, but I immediately hit redial and it works. |
| 18:16.32 | justdave | that's strange. |
| 18:16.42 | [TK]D-Fender | justdave: Dialplan error on the phone |
| 18:16.51 | [TK]D-Fender | justdave: dial ON-HOOK and see |
| 18:17.22 | justdave | yeah, dialing onhook works |
| 18:17.23 | *** join/#asterisk dlynes (n=dlynes@S01060016b68219f1.vs.shawcable.net) |
| 18:17.43 | [TK]D-Fender | justdave: Good, now go fix your dialplan |
| 18:17.44 | justdave | dialplan does indeed not have any allowance for this dial pattern |
| 18:17.56 | ManxPower | justdave: You have two choices at this point. You can either start providing the requested information and stop arguing with the experts or you can not have your problem fixed. Your choice. |
| 18:18.18 | justdave | ManxPower: problem's already fixed, you're lagged. ;) |
| 18:18.23 | ManxPower | You just tell us when you are ready to start working on fixing your problem. |
| 18:19.10 | ManxPower | justdave: good for you. |
| 18:19.46 | justdave | takes time to find some of said information that was requested, and by the time I found it the problem was obvious (so requesting said info got me looking in the right place) |
| 18:19.50 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 18:21.09 | ManxPower | justdave: We usually assume that if you configured the phone you know where the setting is. |
| 18:21.39 | justdave | yes, but I hadn't thought of that setting being the problem (the error message on the asterisk console was pointing me in the wrong direction) |
| 18:22.18 | justdave | the dialplan was the issue, but it seems strange that the polycom would transmit the call anyway and not send a password or something :) |
| 18:22.34 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
| 18:22.44 | justdave | sip debug on the console actually showed the phone number being sent intact, but it was getting an authfail |
| 18:23.02 | ManxPower | justdave: and that is why I think your problem is not solved. |
| 18:24.26 | *** join/#asterisk newmember (n=chatzill@static-66-11-81-77.ptr.terago.net) |
| 18:24.45 | justdave | could be, I'm on a call now (using said phone) so I'll have to play more when I'm off the phone. |
| 18:26.12 | brodiem | ~centos52bug |
| 18:26.13 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889, or trouble compiling x86_64 packages? Report a bug to CentOS! ".i386 packages should not satisfy dependencies for .x86_64 packages." |
| 18:26.49 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 18:27.26 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 18:27.59 | cmw_ | Can anyone help me with this IAX2 connection. It was working perfectly then it stopped and now it shows this in the logs |
| 18:28.00 | cmw_ | Unable to create channel of type 'IAX2' (cause 3 - No route to destination |
| 18:28.46 | cmw_ | i tested both ways and they are able to connect.. so i don't think it's a connectivity issue |
| 18:28.51 | ManxPower | cmw_: There is no route to that destination (ip/hostname). IAX2 show peers will show you the registration status of all your peers. |
| 18:29.42 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 18:29.49 | justdave | ok, off the phone call. rebooting the phone to pick up the revised dialplan |
| 18:29.52 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
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| 18:37.15 | cmw_ | ManxPower: yep it shows unreachable but like i said.. there is no firewall issue i can see the traffic going and coming back fine.. I can even kill asterisk and nc -u bind to that port and hit it from the other side |
| 18:37.16 | *** join/#asterisk arpu (n=arpu@chello084113208186.3.14.vie.surfer.at) |
| 18:37.33 | wacky_ | Asterisk is cool |
| 18:37.54 | ManxPower | cmw_: so you are sending a packet from the asterisk host to the IP of the remote side, UDP port 5060? |
| 18:38.13 | newmember | Where doed asterisknow get the GUI for the IVR? |
| 18:38.25 | ManxPower | newmember: ask on the AsteriskNOW channel |
| 18:38.33 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 18:38.46 | cmw_ | ManxPower: show peers says port 4569 but it doesn't matter since there is no firewall between the two boxes |
| 18:39.11 | ManxPower | cmw_: You know more about this than I do. Best of luck with your problem. |
| 18:39.49 | newmember | I will rephrase, does asterisk have a plugin or add on that is like the asterisknow ivr gui? |
| 18:39.52 | cmw_ | ManxPower: i doubit it :) i know very little.. but i'm sure its not network related |
| 18:39.53 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 18:39.57 | *** join/#asterisk dlewis (i=c7340d67@about/security/staff/dlewis) |
| 18:40.01 | ManxPower | newmember: we don't do Asterisknow stuff |
| 18:40.10 | ManxPower | cmw_: Best of luck. |
| 18:40.11 | cmw_ | clearly it looks that way but i cant see how it could be.. |
| 18:40.12 | cmw_ | thanks |
| 18:41.12 | [TK]D-Fender | cmw_: pastebin your iax2 peer entry |
| 18:41.14 | [TK]D-Fender | ~pb |
| 18:41.14 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:42.18 | *** join/#asterisk EI5GTB (n=Paul@78.16.170.190) |
| 18:42.19 | *** join/#asterisk jnials (n=jnials@cuervo.unwiredbuyer.com) |
| 18:42.36 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 18:43.26 | cmw_ | [TK]D-Fender:http://pastebin.ca/1203575 |
| 18:43.55 | ManxPower | [TK]D-Fender: good luck with this one. |
| 18:44.03 | cmw_ | [TK]D-Fender: i even see this in the logs http://pastebin.ca/1203576 |
| 18:44.28 | *** join/#asterisk tomcontr3 (n=gcontrer@41-144-222-201.adsl.terra.cl) |
| 18:44.41 | [TK]D-Fender | cmw_: Not what I asked for. iax.conf <- |
| 18:44.53 | cmw_ | [TK]D-Fender: sorry sec |
| 18:44.58 | tomcontr3 | hi, does anyone knows if 3 TDM400 cards can be installed on a same server to have 12 FXO modules? |
| 18:45.02 | [TK]D-Fender | masking only passwords |
| 18:45.10 | [TK]D-Fender | tomcontr3: Unadvisable |
| 18:45.19 | [TK]D-Fender | tomcontr3: Buy a more appropriate solution |
| 18:45.31 | tomcontr3 | any example? |
| 18:46.17 | ManxPower | tomcontr3: an example would be the Digium 24 port analog card with 12 modules. |
| 18:46.36 | ManxPower | you will have 1/3 the interrupt load using 1 card .vs. 3 cards. |
| 18:46.55 | cmw_ | [TK]D-Fender: http://www.pastebin.ca/1203578 |
| 18:48.06 | [TK]D-Fender | cmw_: If this is supposed to be iax.conf why and I seeing the AMI port up top referenced, SIP DTMF mode settings and other garbage? |
| 18:48.36 | ManxPower | [TK]D-Fender: "bandwidth=low" I've not seen someone use that in YEARS |
| 18:48.59 | [TK]D-Fender | Somebody needs to get their head screwed on straight... |
| 18:49.08 | [TK]D-Fender | this file is a hodge-podge mess |
| 18:49.27 | cmw_ | [TK]D-Fender: its generated by the fonality interface |
| 18:49.39 | russellb | LOL |
| 18:49.48 | russellb | that is classic |
| 18:50.11 | ManxPower | cmw_: we don't really do GUIs here, as they screw up your config files |
| 18:50.16 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 18:50.23 | ManxPower | you might consider contacting Fonality tech support. |
| 18:50.24 | CrashSys | mmm... fonality... same great taste, less filling... |
| 18:50.33 | [TK]D-Fender | cmw_: if you're telling me thats iax.conf then you are just screwed |
| 18:50.42 | [TK]D-Fender | cmw_: Go ask for their support. |
| 18:50.53 | cmw_ | ManxPower: since the other end is just asterisk.. fonality won't touch it |
| 18:51.07 | cmw_ | [TK]D-Fender: thanks anyway |
| 18:54.24 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 18:54.50 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 18:54.55 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 18:55.25 | *** join/#asterisk soulfreshner (n=D@dsl-243-37-15.telkomadsl.co.za) |
| 18:55.40 | lmadsen | laughs along with russellb |
| 18:56.52 | soulfreshner | I've set up a test asterisk box on a laptop I wasn't using - my sound quality is terrible. I'm only using sip - where can I set start tweaking? |
| 18:57.11 | EI5GTB | evening guys, ok. Sip phone (SIP/100) is on an external network, * is behind nat locally with sip port (5060) and rtp ports (as per rtp.conf) forwarded (udp) when the remote caller calls in, it connects up just fine, but no sound is heard, the following appears on the *cli http://pastebin.com/m2a03c31e |
| 18:57.38 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 18:58.53 | [TK]D-Fender | ~sipnat |
| 18:58.54 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:58.55 | [TK]D-Fender | ^^^^^^^^^^^ |
| 18:59.18 | jaytee | wow, boss told me to go home early |
| 19:00.06 | EI5GTB | [TK]D-Fender, i have followed those guides, and refollowed them twice |
| 19:00.15 | [TK]D-Fender | EI5GTB: PASTEBIN |
| 19:00.24 | EI5GTB | ? |
| 19:00.28 | jaytee | be back later |
| 19:00.31 | EI5GTB | pastebin what? |
| 19:00.36 | EI5GTB | conf? |
| 19:00.41 | [TK]D-Fender | EI5GTB: Show us your configs and realize you only described ONE END of the call. |
| 19:01.03 | [TK]D-Fender | mask only passwords |
| 19:01.29 | EI5GTB | ok so.. sip.conf .. rtp.conf .. extensions.conf? |
| 19:02.19 | *** join/#asterisk bkw__ (n=brian@freeswitch/developer/bkw) |
| 19:03.34 | [TK]D-Fender | EI5GTB: rtp & sip |
| 19:04.08 | ManxPower | [TK]D-Fender: I'll bet you a newbie.smackdown that he missed "canreinvite=no" |
| 19:05.22 | *** part/#asterisk rene- (n=renemend@200.34.66.137) |
| 19:06.40 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 19:12.24 | [TK]D-Fender | *crickets* |
| 19:12.55 | *** join/#asterisk bradleyprice861 (n=bradleyp@fw.datafax.net) |
| 19:15.23 | soulfreshner | I get very bad sound quality testing on a LAN with asterisk running on a pIII 1000MHz (sip.conf: http://pastebin.com/m579073b) |
| 19:15.35 | EI5GTB | [TK]D-Fender, http://pastebin.com/m5ee62037 |
| 19:15.38 | EI5GTB | sorry for the delay |
| 19:15.56 | EI5GTB | doing homework and running a business while trying to have fun with hobies |
| 19:15.58 | EI5GTB | :P |
| 19:16.16 | [TK]D-Fender | EI5GTB: Your sip.conf is woefully incomplete |
| 19:16.45 | [TK]D-Fender | EI5GTB: Right now I'd say you didn't follow much at all. |
| 19:17.15 | [TK]D-Fender | EI5GTB: including the list of things I actually wanted to see. |
| 19:17.50 | ManxPower | EI5GTB: It is considered rude to expect us to wait around for you to provide the requested information. We are not paid to be here. |
| 19:20.02 | ManxPower | But I see that does not matter to you. |
| 19:20.41 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 19:20.50 | justdave | that's right, you're not being paid to be here, so there's no reason to expect an answer from you to come immediately when he finally answers either. You could have walked off just as easily as he did, and taken a while to come back and see it, but instead you complain about having to wait for him. |
| 19:20.51 | soulfreshner | [TK]D-Fender: EI5GTB did provide sip, rtp and extensions, didn't he? |
| 19:21.19 | ManxPower | justdave: and yet we don't normally do that. |
| 19:21.58 | [TK]D-Fender | soulfreshner: I didn't want extensions, and I get a broken sip.conf |
| 19:22.23 | [TK]D-Fender | soulfreshner: And thent here is the thought of waiting for it... |
| 19:23.22 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 19:23.23 | soulfreshner | [TK]D-Fender: oh - didn't mean to butt in - just thought you may have clicked on my paste by accident... we posted close together |
| 19:23.39 | [TK]D-Fender | soulfreshner: for yours, what is on eaxh end of the call? |
| 19:23.58 | *** join/#asterisk bradleyprice861 (n=bradleyp@fw.datafax.net) |
| 19:24.06 | ManxPower | soulfreshner: are you using Linksys/SIPura devices? |
| 19:24.33 | soulfreshner | x-lite sofphone on my main windows PC and an atcom phone on the same network... asterisk is running on an ubuntu machine |
| 19:24.46 | ManxPower | not the problem I was thinking of then |
| 19:25.05 | [TK]D-Fender | soulfreshner: try another softphone |
| 19:25.21 | [TK]D-Fender | soul then try between 2 softphones |
| 19:25.37 | *** part/#asterisk cmw_ (n=ice@167.206.176.233) |
| 19:25.40 | [TK]D-Fender | soulfreshner: this will help isolate if its software to blame |
| 19:25.49 | soulfreshner | [TK]D-Fender: I tried just playing tt-weasels by calling on the atcom |
| 19:25.49 | The-Bat | Hie |
| 19:25.52 | The-Bat | Is there any command which would give me the name of the queue from where the call has come ..?? |
| 19:25.57 | [TK]D-Fender | soulfreshner: Or your Atcom. Next, does it only sound bad for ONE party? |
| 19:26.17 | EI5GTB | [TK]D-Fender, what else do you want? i have provided even pre than what you asked |
| 19:26.17 | [TK]D-Fender | soulfreshner: Does phone-phone sound bad? |
| 19:26.22 | EI5GTB | i have axplained the situation |
| 19:26.35 | EI5GTB | i have given a pastebin of errors |
| 19:26.46 | [TK]D-Fender | EI5GTB: if thats your SIP.CONF as claimed you don't even have a heading for [general] OR a ton of the other essential settings |
| 19:26.48 | EI5GTB | i realise your not being payed for being here |
| 19:27.07 | EI5GTB | and im not getting payed to have a little fun with * |
| 19:27.13 | ManxPower | EI5GTB: maybe you should read ATFOT and rebuild your sip.conf that is reasonably valid. |
| 19:27.23 | soulfreshner | [TK]D-Fender: ok - so it seems it's only the playback of recordings - the sound quality seems fine between phones |
| 19:27.36 | *** part/#asterisk jnials (n=jnials@cuervo.unwiredbuyer.com) |
| 19:27.40 | EI5GTB | i dont expect a whole hearted answer, just a little help, to boost me onto the first stap that is the * learning curve |
| 19:27.45 | [TK]D-Fender | EI5GTB: Don't confuse ManxPower's statements about "not being paid to help" with anything I have said. If I have something to say, I speak for myself. |
| 19:27.55 | [TK]D-Fender | EI5GTB: IS that your complete sip.conf? |
| 19:28.00 | EI5GTB | cheks |
| 19:28.16 | ManxPower | [TK]D-Fender: you're paid to be here? |
| 19:28.36 | soulfreshner | thinks [TK]D-Fender is swell :) |
| 19:28.37 | EI5GTB | [TK]D-Fender, well spotted, its not |
| 19:28.40 | [TK]D-Fender | soulfreshner: Then the problem is that your * was compiled with a GCC of 4.2 or higher which corrupts GSM playback. Recompile with a lower version. |
| 19:28.46 | EI5GTB | i missed a chunk at the top |
| 19:28.53 | EI5GTB | excuse me for jumping the gun |
| 19:29.10 | CrashSys | d-fender: or patch the Makefile and the private.h in the inc directory... :) |
| 19:29.14 | nr4q | contemplates going back to sleep |
| 19:29.15 | [TK]D-Fender | EI5GTB: Please pay close attention when talk to you. I made specific notice of this twice and is something you should have caught onto and checked already |
| 19:29.17 | soulfreshner | thanks [TK]D-Fender ... |
| 19:29.23 | bkruse | newmember: Need some GUI help |
| 19:29.40 | ManxPower | [TK]D-Fender: he's obviously distracted. 8-| |
| 19:29.41 | [TK]D-Fender | CrashSys: That will help the GSM issue? |
| 19:29.53 | *** join/#asterisk unpaidbill (i=bill@420nugs.info) |
| 19:30.20 | CrashSys | d-fender: yeah... http://azrael.crashsys.com/conf/asterisk-gsm-1.4.patch and http://azrael.crashsys.com/conf/asterisk-gsm-asm.patch |
| 19:30.34 | EI5GTB | http://pastebin.com/m4352fd50 |
| 19:30.54 | newmember | bkruse: Thanks for the offer, I am looking at sipx today to complete my comparison, then we will see what the ourcome will be. But thanks very much for reaching out. |
| 19:31.01 | EI5GTB | ffs, i missed a [ as the very first char |
| 19:31.03 | ManxPower | EI5GTB: I hope you have an [ as the first char of your file. |
| 19:31.06 | EI5GTB | but it is in my real conf |
| 19:31.22 | ManxPower | I give up. He's ALL yours, [TK]D-Fender |
| 19:31.25 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 19:31.55 | CrashSys | Fender: Both of those patches are needed and they work with GCC 4.2 and 4.3 :) |
| 19:31.55 | *** join/#asterisk seanmh (i=seanmh@216.31.101.89) |
| 19:32.08 | [TK]D-Fender | CrashSys: Mergable as a global fix? |
| 19:32.16 | bkruse | newmember: No problem, let me know when you need some help :) |
| 19:33.03 | soulfreshner | how do I select the compiler to use for compiling asterisk? |
| 19:33.09 | [TK]D-Fender | EI5GTB: you should have an "externrefresh" set under [general] and please verify that your system can properly get DNS for your host. Also what router are you using? What exactly have your forwarded to *> Is anything forwarded on yor remote phone's side? |
| 19:33.21 | [TK]D-Fender | CrashSys: Can you help him on that one? |
| 19:33.29 | soulfreshner | I have both gcc 4.2 and 3.4 installed |
| 19:34.16 | [TK]D-Fender | soulfreshner : I'd suggest trying CrashSys's patch as linked ablove |
| 19:35.29 | EI5GTB | shit, right, i have to leave now.. chat soon hopefully |
| 19:36.17 | nr4q | EI5GTB can't wait |
| 19:37.24 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 19:37.38 | *** join/#asterisk n9urk (n=IceChat7@rrcs-70-63-204-248.midsouth.biz.rr.com) |
| 19:38.03 | [TK]D-Fender | is used to waiting... |
| 19:38.34 | nr4q | hurry up and wait... that's what I do a lot |
| 19:38.49 | n9urk | anyone know of any good code samples for dynamically routing calls based on area code (like in a sqlite table or mysql table)? |
| 19:39.48 | n9urk | area code from the ANI |
| 19:39.51 | [TK]D-Fender | n9urk: Any simple call in your dialplan to match an area-code with a provider name would do. |
| 19:40.08 | [TK]D-Fender | n9urk: If you're referring to LCR for dialout. |
| 19:40.12 | *** join/#asterisk LemensTS (i=LemensTS@adsl-75-41-210-200.dsl.stlsmo.sbcglobal.net) |
| 19:40.19 | n9urk | [TK]D-Fender: I mean for dial in |
| 19:40.22 | [TK]D-Fender | n9urk: if not please be more specific with the in's and out's |
| 19:40.24 | *** join/#asterisk demonist (n=demonist@CPEbaddeadfed08-CM0019474d4b20.cpe.net.cable.rogers.com) |
| 19:40.38 | n9urk | [TK]D-Fender: yeah on looking back I wasn't too specific |
| 19:40.43 | LemensTS | hey all, what is your prefered click to call method? i see several here http://www.voip-info.org/wiki/view/Asterisk+click+to+call |
| 19:40.43 | [TK]D-Fender | n9urk: you can do any DB lookup you want based on CID. What you do is up to you |
| 19:40.54 | [TK]D-Fender | LemensTS: web + AMI |
| 19:41.21 | LemensTS | TK: thanks |
| 19:41.26 | n9urk | [TK]D-Fender: yeah, I know, but do you know of some good code samples? Sould I just look up AGI? |
| 19:41.34 | [TK]D-Fender | n9urk: no need ofr AGI |
| 19:41.46 | [TK]D-Fender | n9urk: go read the BOOK for func_odbc |
| 19:42.05 | n9urk | [TK]D-Fender: thanks, I wasn't sure what func to use |
| 19:42.29 | [TK]D-Fender | n9urk: There is a nice chapter in the book for this stuff... you should consider actually reading it. |
| 19:42.45 | unpaidbill | i'm trying to tune my pri tx/rx gain but not having any luck. i have a # from my telco for the milliwatt tone but no matter what i put in rx/tx gain the quantitative indicator is always the same. i adjust rxgain= and restart asterisk. am i doing something wrong? |
| 19:43.09 | *** join/#asterisk sack (n=sack@132.Red-79-153-66.staticIP.rima-tde.net) |
| 19:43.38 | unpaidbill | oh jesus i put it after the channel => line |
| 19:45.43 | *** part/#asterisk wacky_ (n=abourget@mtl.savoirfairelinux.net) |
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| 19:50.59 | soulfreshner | cool - recompile fixed the sound quality issue... |
| 19:51.20 | soulfreshner | thanks TK |
| 19:51.49 | CrashSys | fender: I would say those GSM patches are mergable as a global fix... |
| 19:51.56 | CrashSys | they came from the digium bug tracker you know :) |
| 19:52.22 | CrashSys | well, other then the gsm-1.4 makefile patch I removed the attempt to be graceful and just made it forcefully set the optimization to -O2 |
| 19:52.29 | CrashSys | works in 1.2 as well |
| 19:52.51 | CrashSys | Need them both tho |
| 19:55.27 | CrashSys | Ohh, and I got sidetracked with realworld, sorry I couldn't help him, but it seems he just read the metadata in the patch and figured out where to go :) |
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| 20:16.33 | soulfreshner | how do I make my queue exit? I've set timeout to 20 and put a t extension after my call to queue, but the queue never exits... |
| 20:16.52 | [TK]D-Fender | soulfreshner: "core show application queue" |
| 20:18.43 | soulfreshner | [TK]D-Fender: I don't understand it :( |
| 20:19.14 | soulfreshner | well - I don't understand what to do with the variables that's been set and how it helps me out the queue |
| 20:19.48 | [TK]D-Fender | soulfreshner: this is a single line of dialplan. If you have trouble reading the order of parameters and what they mean on that single page of instructions there is a problem... |
| 20:20.21 | [TK]D-Fender | soulfreshner: I have pointed towards what exectly needs to be changed. The relevent bit stands out like a sore thumb |
| 20:21.10 | soulfreshner | n? |
| 20:21.42 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:22.45 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 20:29.25 | CrashSys | Remember noobs, the Fender is as scared of you as you are of it... |
| 20:29.42 | [TK]D-Fender | GNRRRAAAAARRRHH! |
| 20:29.46 | jeev | fender is scared of noobs |
| 20:29.54 | CrashSys | noobs ask noob questions |
| 20:29.58 | soulfreshner | I still don't get it ... http://pastebin.com/d11f9a3e0 |
| 20:30.06 | soulfreshner | what am I missing? |
| 20:30.09 | [TK]D-Fender | whispers "I hear dumb people..." |
| 20:30.10 | *** join/#asterisk tessier_ (n=treed@kernel-panic/sex-machines) |
| 20:30.12 | *** join/#asterisk L-info (n=L-info@host86-163-171-189.range86-163.btcentralplus.com) |
| 20:30.17 | tessier_ | Wow, 1.7 supports T.38 faxing! Sweet! |
| 20:30.30 | tessier_ | I should upgrade my system. I just hate doing so because it is a bit complicated. |
| 20:30.54 | tessier_ | Do any of the ATA's that you might plug into a fax support T.38? Or perhaps most of them do. I never looked into it much because asterisk didn't do T.38. |
| 20:31.15 | tessier_ | I actually had surprising success just sending fax via VOIP with no T.38 but I know it's a roll of the dice. |
| 20:31.27 | [TK]D-Fender | I need to trash Windows on my primary home PC and go Linux so I can cross-backup my destop & server systems |
| 20:31.37 | tessier_ | [TK]D-Fender: Yes, you do. |
| 20:31.51 | tessier_ | A shame you can't get back that $100 even though you don't use the license anymore. |
| 20:32.03 | [TK]D-Fender | tessier_: What $100? :) |
| 20:32.23 | [TK]D-Fender | tessier_: Work-loaned license :) |
| 20:32.27 | tessier_ | uh huh ;) |
| 20:32.31 | tessier_ | Tell it to the judge. |
| 20:32.43 | *** join/#asterisk LND (n=lee@fazer1-adsl.demon.co.uk) |
| 20:32.48 | [TK]D-Fender | tessier_: Oh it's legit, just didn't cost me a penny personally, and is entirely reusable |
| 20:33.21 | [TK]D-Fender | I'll wait for Ubuntu 8.10 to release and a week or two for the bugs to come out |
| 20:34.18 | [TK]D-Fender | I have little need for MS really... |
| 20:34.24 | [TK]D-Fender | ok, checkout time. heading home. |
| 20:34.33 | ManxPower | Um, OEM Windows licenses are like $24 for large vendors aren't they? |
| 20:37.05 | soulfreshner | nvm - it turns out the timeout has to be specified in the queue context and cannot be specified globally |
| 20:41.23 | soulfreshner | why would you ever use queue() without an 'n'? |
| 20:41.59 | *** join/#asterisk paulproteus (n=paulprot@2002:cbb2:8293:0:0:0:0:1) |
| 20:42.01 | soulfreshner | what would the point of a timeout be then? |
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| 20:54.00 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
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| 21:09.49 | esaym | I want to do a project with asterisk. Can someone point me in the right direction? I want the phone to ring for 10 seconds or so and then prompt the caller to press 1 to leave voice mail or press 2 to transfer to an external number (ie. my cell phone) |
| 21:09.59 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
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| 21:16.08 | [TK]D-Fender | esaym: that isn't a "prject", that is about 5 lines of dialplan. |
| 21:16.24 | [TK]D-Fender | esaym: Go read up on IVR basics in teh BOOK & WIKI |
| 21:16.26 | [TK]D-Fender | ~book |
| 21:16.27 | jbot | somebody said book was Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 21:16.27 | [TK]D-Fender | ~wikis |
| 21:16.28 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 21:17.15 | *** join/#asterisk dryrot (i=10539@tsunami.OCF.Berkeley.EDU) |
| 21:17.35 | dryrot | i can't get DTMF working, what should i do? i use ekiga and asterisk |
| 21:17.40 | esaym | [TK]D-Fender: that easy? sounds good thanks |
| 21:18.08 | [TK]D-Fender | dryrot: make sure to set "dtmfmode=rfc2833" for your peer entry |
| 21:18.22 | dryrot | [TK]D-Fender: I did. i tried that, and 'auto'. doesn't work for me. |
| 21:18.45 | [TK]D-Fender | dryrot: pastebin the CLI output of the failed call at verbose 10 along with your sip.conf entry for it. |
| 21:18.46 | [TK]D-Fender | ~pb |
| 21:18.47 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
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| 21:47.21 | [TK]D-Fender | well times up, rebooting to work on other stuff... |
| 21:47.23 | [TK]D-Fender | back later |
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| 22:55.03 | dryrot | which version of asterisk should i install, in linux ? |
| 22:55.29 | *** join/#asterisk Levonk (n=lk@adsl-75-62-142-245.dsl.lsan03.sbcglobal.net) |
| 22:59.34 | *** join/#asterisk savaticus (n=savaticu@sta-206-168-231-41.rockynet.com) |
| 23:00.13 | savaticus | Hello, does anyone know why or if Digium stopped making the IAXy devices. |
| 23:01.48 | outtolunc | http://store.digium.com/products.php?category_id=9 |
| 23:01.51 | [TK]D-Fender | savaticus: http://www.digium.com/en/products/analog/s101i.php |
| 23:01.56 | [TK]D-Fender | savaticus: what do YOU think? |
| 23:02.36 | [hC] | [TK]D-Fender: they discontinued it, and it doesnt say on that page, you know. |
| 23:02.51 | savaticus | Yeah, trying to figure out the why |
| 23:03.02 | [hC] | no interest, probably. |
| 23:03.04 | savaticus | I love those things |
| 23:03.22 | savaticus | We where buying 10 of them a month lol |
| 23:03.37 | savaticus | best home user IAD on the market |
| 23:03.43 | [hC] | you can still buy them until stock runs out, but they have effectively discontinued production |
| 23:03.44 | [TK]D-Fender | iaxy = bleh |
| 23:03.49 | [hC] | and are not planning on a replacement product |
| 23:03.51 | savaticus | just a PiTA to configure |
| 23:04.08 | [TK]D-Fender | PITA to configure, low on options, overpriced, etc |
| 23:04.25 | savaticus | I have searched with no luck to find any other IAX hard phone |
| 23:04.47 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 23:04.56 | savaticus | the joy of it is that it works behind a firewall without having an end use muss about the firewall settings. |
| 23:06.15 | [hC] | http://www.digium.com/en/products/eol.php |
| 23:06.20 | [hC] | there is all the end of life announcements |
| 23:07.33 | outtolunc | i thought they eol'd the *old one, not the new S101 series |
| 23:08.07 | [hC] | it just says 'IAXy' so i presume it means all of em |
| 23:08.16 | outtolunc | yep i see that. |
| 23:08.24 | outtolunc | fun fun |
| 23:08.36 | outtolunc | get'em while they last <G> |
| 23:09.35 | EI5GTB | [TK]D-Fender, re our erlier discussusion, let me know if you cant remember, i had to run off, but in reply to yourt questions. If you mean can the dns address be sucsessfully resolved to the wan ip of my router, then yes. I have forwarded 5060 udp and the port range specified in rtp.sip and the same is forwarded on the phones side |
| 23:10.10 | [TK]D-Fender | EI5GTB: NO forwarding on the remote side. |
| 23:10.18 | *** part/#asterisk bradleyprice861 (n=bradleyp@fw.datafax.net) |
| 23:10.38 | EI5GTB | it wont work with them forwarded? |
| 23:10.47 | [TK]D-Fender | EI5GTB: it can screw things up |
| 23:11.05 | EI5GTB | this was one of the first things i tried... then the guide said you dont have to, i assumed it would still work tho |
| 23:12.01 | [TK]D-Fender | EI5GTB: rule of thumb : follow the instructions to the letter, THEN ask why things don't work |
| 23:12.36 | EI5GTB | ok, well, thats the only deviation i have made, but refering to my knowledge of porsts, and port forwarding, it should meke no difference |
| 23:12.48 | EI5GTB | man my spelling sucks.. |
| 23:13.43 | [TK]D-Fender | EI5GTB: Now pastebin an inbound call from the device with SIP debug enabled |
| 23:14.13 | EI5GTB | ok# |
| 23:14.32 | EI5GTB | (it may take a minute, i have to phone someone to do it for me :P) |
| 23:14.47 | [TK]D-Fender | EI5GTB: understandable |
| 23:15.16 | *** join/#asterisk arpu (n=arpu@chello084113208186.3.14.vie.surfer.at) |
| 23:15.25 | EI5GTB | and considering its nearly 00:15 here, heh |
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| 23:16.21 | [TK]D-Fender | EI5GTB: Silly Brit.... |
| 23:16.38 | EI5GTB | IM NOT BRITISH |
| 23:16.45 | EI5GTB | im irish |
| 23:16.55 | rasterix | is the help on core show application kept up to date? i.e. Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]) |
| 23:16.57 | EI5GTB | dont get em mixed up :P |
| 23:17.11 | [TK]D-Fender | I didn't say ENGLISH. Brit AKA "of the Brittish Isles" |
| 23:17.20 | *** join/#asterisk talntid (n=talntid@c-67-185-208-137.hsd1.wa.comcast.net) |
| 23:17.22 | rasterix | pipe delimeters? |
| 23:17.22 | [TK]D-Fender | And an isle it is... |
| 23:17.24 | EI5GTB | I AM NOT OF THE BRITISH ISLES |
| 23:17.33 | EI5GTB | we are an independant state |
| 23:17.35 | *** join/#asterisk nicudotro (n=nicu@91.123.7.67) |
| 23:17.38 | EI5GTB | known as irelans |
| 23:17.40 | EI5GTB | ireland* |
| 23:17.57 | [TK]D-Fender | EI5GTB: Choas is not a politcally accepted entity ;) |
| 23:17.58 | seanbright | i thought ireland was a country, not a state |
| 23:18.00 | seanbright | giggles |
| 23:18.03 | trnzmeta | sure to be sure? |
| 23:18.07 | rasterix | oh no and just when we thought that war was over |
| 23:18.20 | EI5GTB | im not even gonna get started |
| 23:18.26 | [TK]D-Fender | lol |
| 23:18.31 | EI5GTB | we fought for indapendance, and got it in 1916 |
| 23:18.34 | EI5GTB | thats all im going to say |
| 23:18.35 | [TK]D-Fender | EI5GTB: Ok, fun times over, hop to it! |
| 23:18.40 | EI5GTB | :P |
| 23:18.47 | rasterix | sigh i asked an asterisk question back there |
| 23:19.00 | rasterix | it got lost in the political infighting *sigh* |
| 23:19.05 | seanbright | rasterix: in 1.4 it was | |
| 23:19.14 | [TK]D-Fender | rasterix: "|" is valid through 1.4 1.6 uses "," exclusively |
| 23:19.15 | *** join/#asterisk simNIX (n=simNIX@82-204-21-111.dsl.bbeyond.nl) |
| 23:19.25 | rasterix | ah ok |
| 23:19.27 | *** join/#asterisk StephenF (n=stephen@c-67-188-58-4.hsd1.ca.comcast.net) |
| 23:19.35 | EI5GTB | right, i cant get any people to answer any phoines or radios |
| 23:19.43 | EI5GTB | so time to go do homework instead |
| 23:19.54 | EI5GTB | notes how it takes 3 days to solve a 5 min problem |
| 23:20.01 | rasterix | so the core show application documentation is kept up to date? |
| 23:20.16 | nicudotro | Hello, Did anyone try to install Wanpipe from Sangoma with DAHDI ? |
| 23:20.19 | seanbright | rasterix: should be, yes. |
| 23:20.25 | StephenF | anyone know why I might be getting a garbled connection when dialing into a local extension that is configured to play back an audio file, but when I call my extension and put myself on hold the music is fine? |
| 23:20.27 | rasterix | k thnx |
| 23:20.41 | seanbright | StephenF: i think there is a bug in mantis about that. |
| 23:20.53 | StephenF | seanbright: for 1.4? |
| 23:20.55 | seanbright | yes |
| 23:21.01 | StephenF | ok i'll check, thanks |
| 23:21.04 | rasterix | i installed Wanpipe from Sagoma but with mah MAHMI |
| 23:21.22 | rasterix | she loves her asterisk |
| 23:21.26 | EI5GTB | [TK]D-Fender, im off, thanks for your time. sorry im a bit... sparse in the concentration end of things :P |
| 23:22.01 | nicudotro | :)) |
| 23:23.16 | [TK]D-Fender | EI5GTB: Positively bucolic..... |
| 23:23.16 | rasterix | uhoh... dictionary time |
| 23:23.45 | rasterix | define: positively |
| 23:24.12 | [TK]D-Fender | StephenF: Can happen if you compiled * with GCC 4.2+. GSM codec gets misoptimized and causes nasty noise in playback |
| 23:24.35 | nicudotro | I have asterisk 1.6 beta9 installed and I was considering installing the latest version which I noticed does not have chan_zap anymore |
| 23:24.36 | StephenF | [TK]D-Fender: i think im using g711 codec |
| 23:24.49 | StephenF | similar problem with that codec or no? |
| 23:24.58 | [TK]D-Fender | rasterix: "you positively should grab that active cathode" ;) |
| 23:25.18 | [TK]D-Fender | StephenF: Good odds the sound file being transcoded back to you is in GSM <- |
| 23:25.35 | [TK]D-Fender | StephenF: transcoding with GSM is what fails |
| 23:25.50 | [TK]D-Fender | StephenF: *'s stock sounds come as GSM by default |
| 23:26.16 | StephenF | [TK]D-Fender: there are .ulaw and .alaw files in my sounds folder, wouldnt it just use those? |
| 23:26.16 | rasterix | "GSM the darling of *" |
| 23:26.31 | StephenF | Or do i need to specify to use those |
| 23:26.44 | [TK]D-Fender | StephenF: the right files in the right place, yes. But your description fits this common scenario to a tee |
| 23:27.05 | StephenF | weird, I wonder why i would be transcoding |
| 23:27.20 | StephenF | Do I have to recompile inorder to fix that issue? |
| 23:28.10 | StephenF | Im definetly on GCC 4.2 |
| 23:29.59 | *** join/#asterisk Danskmand (n=danskman@p4FD3C7C7.dip.t-dialin.net) |
| 23:30.16 | russellb | nicudotro: please read the latest release announcements. they explain why chan_zap is not there. |
| 23:30.42 | Danskmand | Hi :-) - Has anyone of you installed asterisk as a faxserver ? - Send AND receive ? |
| 23:31.18 | Danskmand | Maybe with mISDN ? |
| 23:31.37 | *** join/#asterisk coppice (n=chatzill@177.162.17.210.dyn.pacific.net.hk) |
| 23:33.12 | ManxPower | Danskmand: Your extensive search using Google was not helpful? |
| 23:33.26 | ManxPower | Nor was your voip-info.org search? |
| 23:33.48 | russellb | this channel might as well be #justfuckinggoogleit |
| 23:33.50 | russellb | :-p |
| 23:34.00 | rasterix | lol |
| 23:34.27 | Danskmand | Well, I have tried google....but the info is mostly from 2004 or 2005....kinda old... |
| 23:34.45 | Danskmand | Especially with mISDN.... |
| 23:34.49 | rasterix | danskmand dont be put off keep going someone will help you |
| 23:37.38 | coppice | Danskmand: you are right. googling tells you little about the current state of mISDN. all you'll find is a lot of old and conflicting information. the latest news is mISDN stinks and always has. |
| 23:37.48 | Danskmand | rasterix: Thank you ! - I have tried several things, asked many ppl, getting agreed that there are mostly older info on the net, even several ppl telling me they ran into a "wall" for info's (incomplete or outdaed infos) - they gave up on it and now use old-style faxes (hardware), hooked up to a linux-system somehow..... |
| 23:39.05 | coppice | Danskmand: that seems to be the general conclusion. mISDN cannot supply a sufficiently reliable audio stream to a software FAX implemenation. |
| 23:39.17 | [TK]D-Fender | Danskmand: And I have already answered your question to the point you required. |
| 23:39.18 | StephenF | are the sample files transcoded when compliling asterisk or do they come pre-transcoded? |
| 23:39.24 | StephenF | sample sounds |
| 23:39.39 | [TK]D-Fender | StephenF: the ones that are selected by default are GSM |
| 23:40.07 | StephenF | [TK]D-Fender: ok, I selected the G711 when compiling. So shouldnt that bypass the transcoding problem? |
| 23:40.34 | StephenF | I dont understand why I would be transcoding if the sounds are G711 and the phone is using G711 |
| 23:41.23 | rasterix | stephenf: why not try recompiling see if it fixes the problem and then if you still care go back to it? |
| 23:41.39 | coppice | StephenF: because they think working a CPU hard to reduce voice quality is cool? |
| 23:41.41 | Danskmand | coppice: Well, right now I am using capi4linux and capisuite...works lovely, but is not supported on newer Linux'es....So I am searching for a good substitude....But I have read sooo much about different sollutions so I am totally confused... |
| 23:42.34 | coppice | Danskmand: BRI on linux is in a sad state. There are several options, but none of them are reliable. |
| 23:43.00 | rasterix | BRI is dying anyway... |
| 23:43.10 | [TK]D-Fender | StephenF: go verify it |
| 23:43.18 | Danskmand | Coppice: RIGHT ! - I think so too ! |
| 23:43.20 | [TK]D-Fender | StephenF: And what are you listening on? |
| 23:43.56 | StephenF | [TK]D-Fender: verify what? Im using X-Lite softphone, set to use ulaw or alaw |
| 23:44.10 | coppice | Danskmand: it is especially sad when you consider how similar BRI and PRI are, and that PRI is quite well supported these days |
| 23:44.36 | Danskmand | But its not dying....Many company's are using it because it is right there wih minimum risk of being faked... |
| 23:44.54 | coppice | faked? |
| 23:45.54 | rasterix | bri was a failure it never lived up to the hype... it will die |
| 23:46.06 | Danskmand | well, you can write in an email what you want, but faking a fax by putting in different information is hard to do so it doesnt show... |
| 23:46.14 | [TK]D-Fender | StephenF: pastebin the cli output of your distorted call at verbose 10 |
| 23:46.38 | rasterix | thats -vvvvvvvvvv |
| 23:46.38 | Danskmand | What is BRI ? |
| 23:46.40 | rasterix | :) |
| 23:47.08 | Danskmand | IS it VOIP ? |
| 23:47.16 | coppice | Danskmand: er, no it isn't. FAX is very fakable. However, FAXes are accepted by courts, which may be bogus but is very very useful |
| 23:47.38 | StephenF | http://pastebin.com/m603d60ad |
| 23:48.13 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 23:48.13 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:48.19 | Danskmand | coppice: Well, its at least accepted as being hard to fake... |
| 23:48.39 | Danskmand | Plus, companys |
| 23:48.56 | rasterix | < off to send mass bogus fax from everyone in the uk making a donation to rasterisks new charity |
| 23:49.14 | Danskmand | accept it as a "pre-sign" of a contract.... |
| 23:49.40 | Danskmand | Sending it takes 3-4 days via snail-mail.... |
| 23:49.58 | rasterix | still perhaps you ought to know what BRI is before you think about writing a fax server |
| 23:50.00 | rasterix | :) |
| 23:50.17 | coppice | Danskmand: because of things like Sarbanes Oxley FAX is actually growing in many areas. strange world we live in :-) |
| 23:50.33 | *** join/#asterisk mirrorcolor (n=mirrorco@unaffiliated/mirrorcolor) |
| 23:52.26 | ManxPower | rasterix: You must be an USAian |
| 23:52.56 | ManxPower | Because in much of the world BRI was and continues to be a success. |
| 23:53.12 | ManxPower | It's the USA market where the telcos really screwed up BRI |
| 23:53.15 | coppice | not much of the world. just europe |
| 23:53.18 | rasterix | nope uk |
| 23:53.36 | ManxPower | coppice: "much of the world" is Europe 8-) |
| 23:54.01 | coppice | we 40% of the human race living in Asia might argue with that |
| 23:54.23 | ManxPower | coppice: what percentage of the world phone lines does Asia have? |
| 23:54.48 | coppice | lines or phones? asia is heavy with GSM |
| 23:55.11 | ManxPower | Lines, we are not really talking about mobile here. |
| 23:55.41 | Danskmand | Rasterix: Ooh....Now I see...You are talking about the s0 and the s2m standards..,,, |
| 23:55.47 | ManxPower | In any case, BRI is thriving in Europe. |
| 23:55.53 | rasterix | I quote "BRI was intended to service terminal devices and samller sites... but BRI has laregely been deprecated in favour of faster, less expensive technologies..." < the book says u r blasphemers! |
| 23:56.09 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 23:56.18 | ManxPower | rasterix: perhaps the book is talking about DATA on BRI? |
| 23:56.28 | coppice | Chinese makers say almost all BRI hardware goes to .eu |
| 23:57.12 | coppice | rasterix: I wonder what the author's blinkers look like? |
| 23:57.23 | ManxPower | rasterix: What technology is replacing BRI. I'm not awate of anything that transports plain voice calls any faster than a BRI. |
| 23:57.57 | coppice | nothing is replacing BRI. the author is clearly too internt-centric to see the wood for the trees |
| 23:58.09 | ManxPower | If you mean that BRI is pretty much a failure for DATA, then I totally agree. |
| 23:58.22 | coppice | BRI was always primarily a voice system |
| 23:58.26 | ManxPower | coppice: that is the point I was trying to make. |
| 23:58.37 | Danskmand | Well, BRI is replaced....In data tings....by DSL.... |
| 23:58.45 | Danskmand | But in Voice... |
| 23:59.31 | coppice | interestingly, although BRI was successful for lines, it was a total failure for phones. very few BRI phones are shipped |
| 23:59.45 | Danskmand | And in the US its harder to realize because of the huge size of the country... |
| 23:59.51 | coppice | hence G.722 never took off |
| 23:59.55 | StephenF | I had this same problem with my last config, and I changed something and then the audio was fine. But I cant remember what it was |