| 00:01.52 | ping2921 | is there a way to switch realtime for all contexts in extensions.conf? |
| 00:06.37 | *** join/#asterisk fiber0pti (n=John@206-169-194-79.static.twtelecom.net) |
| 00:06.45 | *** join/#asterisk TI83Plus (i=x64pro@cpe-66-66-190-145.rochester.res.rr.com) |
| 00:07.29 | fiber0pti | I'm trying to do 10 digit dialing with a polycom 501. I have the following in my extensions.conf but the phone doesn't want to dial and there are no asterisk events in the cli: exten => _NXXNXXXXXX,1,Dial(${TRUNK}/1${EXTEN:${TRUNKMSD}}) |
| 00:08.07 | JT | phones have dialplans too, is your set correctly? |
| 00:08.19 | fiber0pti | I would have to say no since it's not working ;) |
| 00:08.37 | fiber0pti | do you know where the polycom dial plans would be? |
| 00:08.38 | JT | it's an issue in your phone if asterisk is getting nothing, especially with sip debug on |
| 00:08.43 | *** join/#asterisk xwire (n=akiru@pool-68-238-249-11.phlapa.fios.verizon.net) |
| 00:09.22 | Shaun2222 | bah... i really wish asterisk would disable all modules and let people enable the stuff they need.. |
| 00:09.43 | JT | Shaun2222: you can configure it to do that in modules.conf |
| 00:09.48 | Shaun2222 | ya i know.. |
| 00:09.59 | Shaun2222 | but the problem is that they dont list all the modules in the conf... |
| 00:10.08 | Shaun2222 | would be nice if they listed them all but commented out.. |
| 00:10.31 | Strom_C | Shaun2222: you could always write a new sample file and submit it in the bugtracker |
| 00:10.38 | xwire | omg.. jon teh! |
| 00:10.39 | JT | yeah but that would depend on you updating the config file from sample every time a new release of asterisk came out |
| 00:11.08 | *** part/#asterisk Ritalin2 (n=dave@74-34-103-241.dsl1.pwll.tn.frontiernet.net) |
| 00:11.24 | *** join/#asterisk znoG (n=gs@97-228-126-200.fibertel.com.ar) |
| 00:14.13 | Shaun2222 | JT: not that big of a deal, each upgrade could tell the user, hey these modules where installed that didnt exist... add them :) or somthing.. really wouldnt matter because if they dont need it, it doesnt matter if it's in the file or not.. at least for upgrades.. |
| 00:14.40 | *** join/#asterisk cslinuxboy (n=mlee@mail.biosourcefuels.com) |
| 00:15.18 | cslinuxboy | Does anyone know if connect.voicepulse.com supports the g729 codec? |
| 00:15.39 | dovid | Shaun2222: what modules do u not want loading and y ? for a general setup IMHO it would be a pain to start enabling everything |
| 00:15.52 | dovid | cslinuxboy: try to send a call and see if it goes thru |
| 00:16.16 | TI83Plus | where does one get a list of supported hardware? |
| 00:16.37 | JT | i don't think there's a single list |
| 00:16.38 | dovid | TI38Plus: what kind of hardware ? |
| 00:16.44 | cslinuxboy | dovid: well just wanted to check before I get an account with them. Their website does not list it as one but was hoping that they have not updated their site. |
| 00:16.46 | *** join/#asterisk shepimport (n=shepimpo@phoenix.u4eatechinc.com) |
| 00:16.48 | JT | just tell us what you need to do |
| 00:16.49 | TI83Plus | I am just starting |
| 00:16.56 | Shaun2222 | dovid: anything i'm not using... which makes sense... think about if you compiled apache or somthing and it enabled every module known... |
| 00:17.05 | TI83Plus | I want to replace our landline |
| 00:17.18 | Shaun2222 | also, my guess is that asterisk loads every module in the modules dir? |
| 00:17.18 | Shaun2222 | that sounds like a bad idea... |
| 00:17.49 | dovid | Shaun2222: not sure if it loads every one. i would say make gen configs that u wana use on installs and copy em over to evey new system that you create |
| 00:18.04 | dovid | Shaun2222: thats what i personally do |
| 00:18.09 | dovid | Shaun2222: can I pm u ? |
| 00:18.20 | shepimport | Helllooo all... does anyone know how to enable direct media between two SIP endpoints without using a REinvite... or if asterisk supports this without a SBC or SIP proxy??? |
| 00:18.21 | Shaun2222 | well i have a modules.conf i use but it was for 1.2... |
| 00:18.23 | xwire | jt: hows sydney? |
| 00:18.32 | Shaun2222 | now i guess i need to do the same on 1.4.x |
| 00:18.34 | *** join/#asterisk coppice (n=chatzill@55.157.17.210.dyn.pacific.net.hk) |
| 00:19.06 | Shaun2222 | one sec... |
| 00:19.12 | *** part/#asterisk cslinuxboy (n=mlee@mail.biosourcefuels.com) |
| 00:19.14 | dovid | Shaun222: its for a working enviroment i am still staying with 1.2.X, i have seen a few complaints on 1.4 and ths enough for me |
| 00:19.40 | JT | xwire: fine |
| 00:19.47 | xwire | haha.. it is you |
| 00:19.47 | fiber0pti | ok.. I've verified my polycom digitmaps are setup for 10 digit dialing yet asterisk won't dial 10 digits |
| 00:19.48 | hardwire | TI83Plus: don't replace your landline please |
| 00:19.51 | xwire | man, you still on austnet? |
| 00:19.53 | TI83Plus | why |
| 00:20.02 | hardwire | E911 |
| 00:20.07 | J4k3 | haha |
| 00:20.08 | TI83Plus | bah |
| 00:20.10 | J4k3 | e911 is a joke |
| 00:20.14 | J4k3 | if you want to get found, use a cellphone. |
| 00:20.15 | *** join/#asterisk Shaun2222 (n=Shaun@ip68-4-212-221.oc.oc.cox.net) |
| 00:20.22 | aptura | still best to have at least one land line in a small biz |
| 00:20.37 | J4k3 | lat/long >>>>> sometimes maybe getting your e911 info properly looked up from a database |
| 00:20.39 | *** join/#asterisk sivana[work] (n=richard@sivana-155-134.vianet.ca) |
| 00:20.40 | aptura | dont depend on a cell phone. Thay have legs and move around alot. |
| 00:20.41 | hardwire | J4k3: back in the day |
| 00:20.47 | hardwire | there used to be more phones in a house than cell phones |
| 00:21.05 | J4k3 | yeah... and back in the day cellular contracts cost a fortune and came with a few dozen minutes. |
| 00:21.12 | JT | aptura: in the us all new mobile phones have gps receivers |
| 00:21.16 | hardwire | 3 year olds weren't running around with micro brain tumors from chatting with their buddy next door on their cellies. |
| 00:21.18 | Shaun2222 | dovid, ya i guess if it needs to be private. |
| 00:21.23 | aptura | jt, im aware of that. |
| 00:21.41 | xwire | all of them? |
| 00:21.45 | aptura | JT, I have made GPS trasmitters 10 years ago for hamradio. |
| 00:21.47 | J4k3 | I've always paid right around $70/mo for my cellular contracts... I started out with 150 minutes in my region, now that same money gets me 1400 minutes and national calling. |
| 00:21.52 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
| 00:21.54 | JT | xwire: yeah, e911, it's law |
| 00:21.55 | kink0 | hello |
| 00:21.57 | J4k3 | oddly, still using the same company (GTE mobilenet -> Verizon) |
| 00:22.04 | JT | scary potential to be misused |
| 00:22.12 | xwire | no shit.. when did they implement that? |
| 00:22.22 | hardwire | JT: haha |
| 00:22.23 | dovid | i never rely on e911, also with voip u have many more points of failure |
| 00:22.24 | hardwire | you have no idea |
| 00:22.25 | J4k3 | so... just break your GPS |
| 00:22.27 | JT | couple years ago iirc |
| 00:22.30 | aptura | btw anyone here knowlegable on gsm cdma technoligies? |
| 00:22.33 | J4k3 | if you're paranoid about it |
| 00:22.35 | dovid | if some one needs medical attention i font wana take any chances |
| 00:22.35 | JT | hardwire: i'm sorry? |
| 00:22.43 | hardwire | so I called our CLEC that handles cellular on their own and told them hey, I want do to asset tracking and fleet management |
| 00:22.49 | hardwire | can I get a GPS feed of all the phones on our account? |
| 00:22.50 | J4k3 | GPS requires an antenna, in every handset I've seen torn down on fcc.gov, you could *easily* disable the GPS |
| 00:22.52 | kink0 | a quick cuestion: any way to set a call-id number when a call is originated from Asterisk and send to the PSTN, in a way the PSTN called party see these number in display, for call-back purposes ? |
| 00:22.53 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 00:22.56 | hardwire | "sure, I bet we could do that" |
| 00:23.08 | dovid | hardwire: i know nextel offers the option |
| 00:23.11 | aptura | I have a possible clinet that wants GPS tracking devices built for 4 assets and need a cell carrier to handle it. Analog is best for distance. |
| 00:23.12 | J4k3 | hardwire: verizon can do it now, I think cingular can too. |
| 00:23.21 | hardwire | yeh, scary potential |
| 00:23.38 | hardwire | actually I wanted to talk to them about offering an opt in service for cell phones |
| 00:23.42 | J4k3 | aptura: analog is dead next year.. if you want the best coverage in the USA, you're stuck with CDMA2000. |
| 00:23.44 | dovid | hardwire: i think they charge for it though |
| 00:23.56 | hardwire | yeh? |
| 00:23.58 | Strom_C | hardwire: what can the telco do? |
| 00:24.01 | aptura | J4 I am talking about Vancouver BC. |
| 00:24.01 | dovid | no more analog ?? :( |
| 00:24.11 | hardwire | well it would be a dream to give the traffic company GPS feeds from opt in users. |
| 00:24.17 | dovid | analog in the city i think is still better than digital in many areas |
| 00:24.21 | J4k3 | aptura: well, what are your choices... Telus (CDMA2000) or Rogers (GSM) |
| 00:24.28 | HushPe | naftali5: back now... do i copy all the libpri/zaptel/asterisk to the /usr/src dir without the versioning? |
| 00:24.33 | J4k3 | CDMA2000 beats GSM by about 10 dB in actual RF performance. |
| 00:24.34 | hardwire | even better to give those feeds to a private company that can suggest redirections for traffic. |
| 00:24.42 | aptura | With analog you can still hear somone even though it may be staticky. With digital, you get cut off if the Db gain drops to low |
| 00:24.50 | aptura | ohh really |
| 00:25.29 | J4k3 | I've made 1xRTT calls in places where my tri-mode phone couldn't hold an AMPS call. Using the same towers (Verizon SID12, 850 mhz) |
| 00:25.32 | shepimport | <PROTECTED> |
| 00:26.16 | J4k3 | aptura: I dunno... my new CDMA handset holds pretty good solid digital calls down to -108 dBm... around -110 dBm is the point where AMPS goes to the point of "painfully noisy" |
| 00:26.19 | aptura | I know little about AMPS is that for areas that are far and remote? |
| 00:26.27 | J4k3 | AMPS = analog |
| 00:26.30 | aptura | k |
| 00:26.35 | Strom_C | AMPS == Advanced Mobile Phone System |
| 00:26.39 | aptura | Right |
| 00:26.41 | perd | does asterisk have a sound file for a quick double beep |
| 00:26.48 | Strom_C | essentially the north american standard from 1983-1993ish |
| 00:26.50 | perd | like a hook flash beep |
| 00:26.57 | perd | USA hook flash at least |
| 00:27.04 | Strom_C | hookflashes don't beep |
| 00:27.07 | J4k3 | Strom_C: well, its still implemented on just about every 850 tower out there. |
| 00:27.16 | perd | well when i send a hookflash i get a 'beepbeep' |
| 00:27.23 | J4k3 | although that'll be rapidly changing this year |
| 00:27.24 | perd | that's the sound i want |
| 00:27.25 | Strom_C | J4k3: hence why I said "the" and not "a" |
| 00:27.56 | Strom_C | perd: which phone generates beeps when you hookflash? |
| 00:27.56 | JT | aptura: what do you need to know about gsm or cdma? |
| 00:28.14 | perd | phone on POTS line |
| 00:28.20 | J4k3 | JT: basically aptura was talking about asset tracking, and that AMPS would work better. |
| 00:28.30 | perd | i guess it's not so much a beep as it is a broken dialtone |
| 00:28.36 | Strom_C | perd: yes, I understand that, but what kind of station equipment are you using? |
| 00:28.36 | JT | hardwire: what did you mean "you have no idea"? |
| 00:28.40 | Strom_C | perd: you mean a recall dialtone? |
| 00:28.41 | perd | two breaks in audio from the DT |
| 00:28.47 | JT | amps is junk |
| 00:28.50 | perd | strom any PSTN phone i've used in the usa |
| 00:28.55 | JT | no security |
| 00:28.57 | JT | not good for data |
| 00:29.04 | perd | i just wanted a sound to use in this dialplan i have for turning on/off privacy |
| 00:29.11 | perd | i'll just have it hang up i guess |
| 00:29.14 | Strom_C | you hookflash and get a dial tone which stutters momentarily and then goes solid? |
| 00:29.23 | perd | yeah |
| 00:29.23 | J4k3 | AMPS was designed for a $1/minute airtime world |
| 00:29.28 | J4k3 | not 0.05c/minute airtime. |
| 00:29.28 | Strom_C | that's called recall dial tone |
| 00:29.33 | perd | ah |
| 00:29.51 | Strom_C | J4k3: yeah, but it sounds amazing compared to GSM :) |
| 00:29.55 | coppice | AMPS was also designed for a world without DSP or MCU chips |
| 00:30.04 | J4k3 | Strom_C: tin can and strings sounds amazing compared to GSM. |
| 00:30.05 | perd | cool, any way to have it do that after I am already floating around my contexts? |
| 00:30.07 | aptura | JT, I have a potential customer that wants to asset track his equipment and I have built GPS Analog tracking devices for hamradio 10 years ago. But in this case being it is a commerical product would need a air carrier to carry the NEMA signaling when the cell phone or tranciver is called then the base would hear the steaming data. I would like a carrier that can send/recive into the bush since this is where some of the equipment will |
| 00:30.11 | Strom_C | hahaha |
| 00:30.14 | Strom_C | touche |
| 00:30.53 | coppice | who considers GSM old and pathetic in the rest of the world? |
| 00:31.55 | xwire | cdma can suck, i like the fact I can change the sim card in my phone to switch between GSM providers |
| 00:31.55 | J4k3 | coppice: any country that has WCDMA deployed |
| 00:31.55 | J4k3 | like say... japan, sk, etc. |
| 00:31.55 | coppice | 1000M GSM subs. 5M UMTS subs. doesn't sound too old and pathetic |
| 00:32.01 | J4k3 | coppice: UMTS isn't GSM, it just happens to be pushed by GSM providers. |
| 00:32.01 | coppice | most people still consider UTMS phones suck |
| 00:32.16 | J4k3 | UTMS isn't suck, but the spectrum carriers are using for it does. |
| 00:32.20 | JT | UMTS is WCDMA |
| 00:32.24 | coppice | UMTS is the dominant WCDMA system |
| 00:32.51 | J4k3 | I don't care what protocol you use, spectrum above 1 ghz is sketchy, spectrum above 2 ghz is downright worthless. |
| 00:32.56 | JT | coppice: gsm is shit though, compared to newer telephony offerings |
| 00:33.15 | coppice | GSM works better than anything else I've seen |
| 00:33.38 | JT | maybe for a small densely populated european or asian nation |
| 00:33.49 | JT | it's a joke for nations with areas of low population density |
| 00:33.58 | J4k3 | the carriers that switched to GSM already had IS-136 in place... They gained very little and lost a whole lot in coverage. |
| 00:34.19 | J4k3 | JT: exactly. People fail to realize "GSM sucks from an RF performance aspect" |
| 00:34.28 | coppice | well, most countries have ripped out an IS-136 they had as obsolete |
| 00:34.28 | Mavvie | hmmm.. it would be nice if these cisco phones actually supported SNMP statistics... |
| 00:34.30 | J4k3 | CDMA sucks in a peak-time-urban-area aspect. |
| 00:34.38 | aptura | rogers voice recognition system works well |
| 00:34.46 | *** join/#asterisk zero-G (n=mark@69-2-64-170.wan.networktel.net) |
| 00:34.49 | J4k3 | coppice: yeah, they got sold a lot of hype from the GSM camp. |
| 00:35.11 | J4k3 | coppice: the customers didn't see any improvement. They see more signal issues and the prices staying the same. |
| 00:35.28 | coppice | no. the CDMA just sucked. China got suckered into a useless CDMA system the poor carrier can't sell as part of trade negotiatiosn with the US |
| 00:35.28 | JT | the audio quality is much worse with gsm |
| 00:35.49 | J4k3 | what china does has very little relevence to the rest of the world. |
| 00:35.59 | J4k3 | and I'm not going to cry for the communists, sorry. |
| 00:36.12 | dovid | what ports are needed for nat again ? |
| 00:36.17 | dovid | 5060 and 10k-20k ? |
| 00:36.33 | *** join/#asterisk GaVak (n=denniso@adsl-074-228-124-003.sip.sav.bellsouth.net) |
| 00:36.41 | J4k3 | all I want is some of that czech 450 mhz 1xEVDO action... |
| 00:37.02 | GaVak | Ok, my brains not working, whats the zap echo self-adjustment command? |
| 00:37.09 | GaVak | zttool -something? |
| 00:37.40 | xwire | btw, e911 doesnt mean every phone has a GPS in it, but the networks have pretty advanced triangulation technologies/etc |
| 00:38.00 | JT | old phones might not |
| 00:38.09 | J4k3 | xwire: afaik all the US carriers went GPS due to the liabilities of trying to use the cellular technology to triangulate. |
| 00:38.17 | coppice | xwire: I think the CDMA phone have GPS, and the GSM ones don't |
| 00:38.28 | J4k3 | GPS = "hey, its the military's system, if it doesn't work well - bitch at the defense department" |
| 00:38.41 | J4k3 | coppice: Cingular's phones are marked GPS now. |
| 00:39.06 | coppice | GPS only works outside, though |
| 00:39.09 | J4k3 | coppice: I believe the GSM handsets "AT&T Wireless" sold pre-merger were triangulation-based. |
| 00:39.19 | robin_sz | its pretty obvous why CDMA has to have GPS, and GSM doesnt |
| 00:39.20 | xwire | what about phones purchased ourside of the US? |
| 00:39.21 | J4k3 | my vx9800 has no problem getting a GPS fix inside my office |
| 00:39.24 | J4k3 | takes about 45 seconds |
| 00:39.44 | aptura | what audio ranges do cell phones transmit? |
| 00:39.52 | JT | robin_sz: it is possible to triangulate cdma signals, just harder |
| 00:40.07 | J4k3 | robin_sz: GSM-based triangulation takes a LOT of time off neighboring towers, and most american GSM networks lack the density to pull it off. |
| 00:40.19 | coppice | aptura: just a few centimetres, usually :-) |
| 00:40.39 | robin_sz | JT, quite. GSM calculates delay between handset and tower as part of the access scheme, so multiople towers in range make it a snap |
| 00:40.43 | aptura | what Audio ranges do cell phones transmit? 3-16 khz? |
| 00:40.45 | robin_sz | CDMA is much harder to do |
| 00:40.53 | robin_sz | aptura, nah, |
| 00:41.00 | J4k3 | its exactly equally as hard with CDMA... I can't see why you'd say it'd be somehow easier with GSM |
| 00:41.04 | robin_sz | aptura, try 300hz to 3khz |
| 00:41.04 | J4k3 | GSM just has fixed timing, thats it. |
| 00:41.14 | robin_sz | pah |
| 00:41.15 | JT | aptura: lol, 16kHz, who are you kidding? |
| 00:41.24 | coppice | aptura: most of the time is 3kHz bandwidth. some now do wideband audio, and go up to 8kHz |
| 00:41.28 | J4k3 | and to triangulate, you'll have to force all the calls on that channel on the neighboring towers to hop off, hope to god you can hear the handset that you're trying to triangulate, etc. |
| 00:41.28 | JT | aptura: landline phones are 300-3.4kHz |
| 00:41.32 | J4k3 | its just too sketchy. |
| 00:41.38 | JT | so normal bandwidth is 3.1kHz |
| 00:41.41 | robin_sz | J4k3, the timing is NOT fixed |
| 00:42.06 | robin_sz | J4k3, the timing varies to make sure the transmit burst arrives at the tower exactly in the right slot |
| 00:42.08 | JT | aptura: fm stereo radio only transmits 15kHz |
| 00:42.24 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 00:42.29 | J4k3 | robin_sz: yes, but its pre-negotiated. cdma is chaotic by comparison |
| 00:42.33 | coppice | aptura: GSM and UMTS have wideband options. I guess CDMA probably does too, these days |
| 00:42.47 | robin_sz | J4k3, so the handset has to knwo the tower<->handset delay, thus the distance |
| 00:42.57 | robin_sz | CDMA is quite differetn |
| 00:43.09 | J4k3 | robin_sz: yeah, but you can do the same with CDMA... and with CDMA you're already having conversations with the other towers... the bigger problem is the idea is just piss poor |
| 00:43.24 | aptura | coppice yea these are important questions to ask because if I solder my TNC to my gps and interface it with the cell phone it may not send the audio signaling if its outside the range of the phones Band Pass filters. |
| 00:43.33 | J4k3 | if you're in a place you can't get a 1.4ghz GPS signal, chances are your 850 (or more likely 1.9ghz) signal is getting multipath'd to death also. |
| 00:43.41 | JT | aptura: dude |
| 00:43.42 | test34 | Is there anyway to unlock a Motorola VT2442 that has recently been connected to the internet (ie today)? |
| 00:43.47 | JT | aptura: that is an insane idea |
| 00:43.52 | JT | aptura: use GPRS or similar |
| 00:43.52 | robin_sz | CDMA uses code sets with a high dgreee of orthogonality, so the timing between one handset and another is irrelevant |
| 00:44.00 | JT | and send the data digitally, shesh |
| 00:44.18 | JT | sending modem signals over mobiles is almost madness |
| 00:44.23 | coppice | aptura: use data, like GPRS. the compression codecs make any tone signalling iffy with any cellular system |
| 00:44.27 | aptura | JT and does it cover areas of 30 miles from the nearest cell site? |
| 00:44.36 | robin_sz | aptura, contact a professional to help you |
| 00:44.38 | J4k3 | f all this, dtmf over shortwave. |
| 00:44.43 | J4k3 | psk31 over shortwave. |
| 00:44.50 | aptura | robin thats what I am actually doing. |
| 00:44.56 | J4k3 | if you want to get into complex solutions for simple problems ;) |
| 00:45.02 | robin_sz | aptura, look in the phone book for "doctors, mental health" |
| 00:45.10 | aptura | psk31 is awsome. We made contact with france from Seattle once. |
| 00:45.45 | robin_sz | and ... how on earth is that connected to audio into cell phones huh? |
| 00:46.27 | JT | aptura: cdma will cover 30miles in the right terrain |
| 00:46.40 | aptura | Bc is mountainous. |
| 00:46.52 | JT | aptura: if it's a super remote application, you may need to use iridium |
| 00:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:46.57 | aptura | But anyway thats all the info I need. |
| 00:47.15 | coppice | its a while since i heard someone say iridium :-) |
| 00:47.20 | JT | :) |
| 00:47.21 | J4k3 | my CDMA record is 47 miles... the call was LOS with lousy fresnel |
| 00:47.32 | J4k3 | the phone bounced between 10-23 dBm for the call. |
| 00:47.35 | JT | iridium is definately the best choice atm for voice sat calls |
| 00:47.38 | robin_sz | nice |
| 00:47.55 | aptura | sat works part of the time. |
| 00:47.55 | J4k3 | iridium is quite awesome, just don't expect good battery life or cheap airtime |
| 00:47.58 | J4k3 | but it *works* |
| 00:48.11 | JT | J4k3: it's way cheaper than all the other sat phone networks |
| 00:48.17 | J4k3 | JT: yeah, but thats not saying much |
| 00:48.21 | JT | AUD$1-2/min |
| 00:48.23 | robin_sz | J4k3, ive managed over 150 miles with 200mw on 1.3ghz SSB |
| 00:48.24 | aptura | Depends where in the world you are the Sat phone may not work some times. |
| 00:48.49 | coppice | iridium sound quality isn't too hot, though |
| 00:48.53 | J4k3 | robin_sz: yeah... but did you do it with a handheld device with maybe an 1/8th wave antenna? |
| 00:49.01 | JT | aptura: umm, no, if you have clear view of the sky, and don't have bad weather, you are almost guaranteed iridium access |
| 00:49.16 | JT | aptura: iridium has 66 satellites, it covers every patch of sky exposed land |
| 00:49.20 | aptura | Depends where in the world you are the Sat phone may not work some times. |
| 00:49.26 | robin_sz | J4k3, well not quite ... that was with 4 x 96 ele loop yagis |
| 00:49.31 | JT | aptura: fucks sake, stop repeating, use brain pleas |
| 00:49.32 | JT | e |
| 00:49.36 | J4k3 | robin_sz: hehe, thats different ;) |
| 00:50.10 | JT | aptura: how much research have you done into iridium? |
| 00:50.24 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 00:50.33 | aptura | JT, I know Iridium went bankrupt in the 19990s |
| 00:50.34 | aptura | ;) |
| 00:50.41 | JT | thanks, and it's now 2007 |
| 00:50.44 | JT | get with the program |
| 00:50.45 | robin_sz | J4k3, it used to work even better with 250W from a water cooled 2C39 :) |
| 00:50.47 | coppice | JT: how much did motorola do before paying for it? :-) |
| 00:50.57 | JT | bought out by a private consortium/US DoD for USD$25M |
| 00:51.04 | JT | hence why it's so cheap now |
| 00:51.13 | J4k3 | yep |
| 00:51.17 | JT | coppice: i think they spent USD$6B making it |
| 00:51.21 | J4k3 | thank the stupid investors into the iridium program originally |
| 00:51.31 | J4k3 | who paid oodles of money for a business plan that could never work |
| 00:51.39 | J4k3 | that now does work thanks to the fact that it was purchased for pennies on the dollar. |
| 00:51.44 | aptura | I could care less. If china has its way thay will knock out all of the Iridium sats out of the sky if thay wish :) |
| 00:51.47 | robin_sz | are yo talkign about 3G here? |
| 00:51.56 | coppice | JT: yep. probably broke the company, and left it in the mess it is today. well, that and the waste on iDEN |
| 00:52.02 | JT | aptura: fuck you talk a lot of shit, live in blisful ignorance |
| 00:52.14 | J4k3 | aptura: yeah, but they might turn themselves into the "glass parking lot named china" |
| 00:52.17 | aptura | JT your ignorant on current events :) |
| 00:52.19 | JT | robin_sz: iridium is the largest sat phone network |
| 00:52.19 | sevard | gerble gerble angry words |
| 00:52.23 | JT | aptura: bullshit |
| 00:53.02 | JT | aptura: we try to educate you on how iridium currently is, and you just keep sprouting crap from 7 years ago? that's what i call ignorant |
| 00:53.46 | aptura | JT personally I could care less about your comments. You need to chill. I know Craig McCaw enough that my step dad worked for him in the early years and did test one of the first cells in the states in the 70s. |
| 00:53.52 | robin_sz | JT, sorry when you said "cost billions " and "business plan that could never work" i thught you must be talking about the UK/EU 3G system ;) |
| 00:54.01 | xwire | haha.. such an angry man JT |
| 00:54.06 | JT | robin_sz: J4k3 said could never work :) |
| 00:54.23 | sevard | TI83Plus: this is still #asterisk, regardless. what's your issue? |
| 00:54.28 | robin_sz | aptura, and ... stuff has moved on since the 70s ... |
| 00:55.01 | TI83Plus | I need help finding hardware |
| 00:55.02 | JT | aptura: hrm sorry, that name dropping failed to impress me, but really, we were giving you another option for your gps problem, trying to be helpful, and you just repeating < aptura> Depends where in the world you are the Sat phone may not work |
| 00:55.06 | JT | <PROTECTED> |
| 00:55.07 | JT | frustrating, a bit |
| 00:55.11 | aptura | of course. Dont you appricate history in the telecom sector? :) |
| 00:55.16 | JT | and iridium was launched in 1998, not 1970s |
| 00:55.16 | sevard | TI83Plus: http://www.digium.com |
| 00:55.18 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:55.20 | blitzrage | evening all! |
| 00:55.28 | robin_sz | is it? |
| 00:55.35 | JT | maybe you're talking about inmarsat |
| 00:55.36 | aptura | JT I am talking about the cellular network when it was created. |
| 00:55.44 | TI83Plus | •sevard• Too expensive, they have ot have something less than 200 |
| 00:55.45 | JT | which cellular network? |
| 00:55.46 | blitzrage | Got my 7970 working (w00t), but I'm looking for a guide from Cisco with the available options in the SEP<mac>.xml.cnf file |
| 00:55.57 | DonX | TI83Plus: lol |
| 00:55.58 | aptura | McCaw Cellular..the first one. |
| 00:56.02 | sevard | TI83Plus: what kind of hardware do you need? |
| 00:56.07 | JT | ah ok |
| 00:56.39 | TI83Plus | connect exsiting land line to server |
| 00:56.39 | TI83Plus | then out again |
| 00:56.39 | sevard | TI83Plus: do you care about quality? |
| 00:56.39 | TI83Plus | yea |
| 00:56.39 | xwire | aptura: you have to tiptoe around JT, he gets very angry when facts are not 100% factual |
| 00:56.44 | aptura | I could care less :) |
| 00:56.47 | sevard | then you'll be paying out cash, if you care about quality and echo. if you don't, pick up an x100 |
| 00:56.50 | xwire | and irc is a very serious place, just ask him |
| 00:56.58 | sevard | very, very serious place. |
| 00:57.00 | JT | xwire: not true, you just need to not ne a knob that keeps repeating the same wrong statement without discussing it :) |
| 00:57.04 | blitzrage | irc is the blind leading the blind :) |
| 00:57.15 | Jingles | shit! I'm being led somewhere? |
| 00:57.21 | xwire | with an angry JT in the middle |
| 00:57.22 | sevard | ohshitman |
| 00:57.22 | blitzrage | straight to hell! |
| 00:57.28 | coppice | JT: why? it usually works for politicians |
| 00:57.40 | JT | xwire: take your personal problems elsewhere |
| 00:57.59 | xwire | i did, i left #unix on austnet and came here |
| 00:58.02 | xwire | oh wait, that was you. |
| 00:58.08 | *** part/#asterisk TI83Plus (i=x64pro@cpe-66-66-190-145.rochester.res.rr.com) |
| 00:58.15 | JT | i'm still everywhere, and have been here for ages |
| 00:58.18 | JT | you're new here |
| 00:58.28 | xwire | you *think* that |
| 00:58.39 | JT | sure |
| 00:58.44 | robin_sz | aptura, Craig McCaw didnt start looking at cellular until '81 .. not the 70s, he was tunning a cable tv system then |
| 00:58.44 | xwire | actually, i couldnt remember my password for my account here, so regged this so I could get into a few rooms |
| 00:59.09 | JT | xwire: what was your other nick? |
| 00:59.34 | xwire | here? |
| 00:59.37 | JT | yes |
| 00:59.45 | JT | since you say you've been here for a while |
| 00:59.46 | aptura | robin, my step dad was with craig MCaw when thay turned on the first cell cite with the mayor next to them in the late 70s. |
| 01:00.23 | robin_sz | http://www.achievement.org/autodoc/page/mcc0bio-1 |
| 01:00.23 | aptura | That was in Chicogo |
| 01:00.34 | xwire | /whois xwire |
| 01:00.52 | aptura | Sorry robin but my step dad has pictures to prove it. |
| 01:01.22 | JT | xwire: already done that, thought you had another one, since you claim to have been here for so long |
| 01:01.39 | robin_sz | well, sorry, but the rest of your comments were not particualrly believable, I'll stick with the official biography version, thanks |
| 01:01.46 | xwire | and where was that claim made? |
| 01:01.55 | aptura | robin thats fine. |
| 01:01.55 | JT | 12:22 < JT> you're new here |
| 01:01.56 | JT | 12:22 < xwire> you *think* that |
| 01:02.03 | JT | i meant in this channel specifically |
| 01:02.27 | xwire | perhaps you should be clear in your facts jt, otherwise I may have to get angry and serious and correct you |
| 01:02.52 | xwire | no room for ambiguities |
| 01:03.08 | JT | you infered that you've been in this channel for longer than i think |
| 01:03.17 | xwire | perhaps on the network. |
| 01:03.23 | Strom_C | children, children |
| 01:03.29 | xwire | see, jt gets very angry |
| 01:03.35 | robin_sz | awww, it was just getting fun :) |
| 01:03.41 | xwire | and yeah, I am new to this room, just downloaded *now, and thought id see who the fanboys were |
| 01:03.53 | xwire | and trolls |
| 01:04.16 | JT | xwire: you're a shit stirrer, if you've been around for long enough, you'd realise i try to help people here as much as possible, instead of trying to stir shit |
| 01:04.20 | JT | but anyway |
| 01:04.30 | Jingles | poop |
| 01:04.31 | robin_sz | if any goats come past. I'm having them |
| 01:04.34 | JT | who has an asterisk problem? :) |
| 01:04.46 | blitzrage | JT: I do |
| 01:04.54 | robin_sz | JT, still me and mISDN :) |
| 01:04.56 | Shaun2222 | why am i seeing these warnings? http://rafb.net/p/VXDnC845.html |
| 01:05.03 | JT | robin_sz: given bristuff a go? |
| 01:05.29 | blitzrage | Shaun2222: don't worry about it -- something in the code, but no harm with it |
| 01:05.47 | aptura | robin see chicogo was a cellular test site. Not in full operation. I was living near seattle when I saw the first cell equipment as a kid in the early 80s. |
| 01:05.48 | robin_sz | JT, I am going to have to build another box for that ... so much stuff needed changing I feared screwing my install and thus the office phone system |
| 01:06.12 | blitzrage | JT: Quick!!! I need teh help!!! |
| 01:06.13 | Shaun2222 | blitzrage: i'm a bit confused as to why 1.4.0 which is suppose to be released would have warnings like that for somthing like iax |
| 01:06.17 | aptura | It was interesting and bizzare technoligy but soon it started to kick in. |
| 01:06.18 | *** join/#asterisk xwire (n=akiru@pool-68-238-249-11.phlapa.fios.verizon.net) |
| 01:06.21 | xwire | damn wireless |
| 01:06.26 | blitzrage | Shaun2222: because it's a point oh release |
| 01:06.28 | xwire | anyway, i had just said "fanboys" |
| 01:06.45 | Shaun2222 | what |
| 01:06.56 | JT | robin_sz: oh ok, didn't think it would've impacted that badly upon your existing * installation? |
| 01:07.02 | blitzrage | never trust a .0 release |
| 01:07.18 | blitzrage | Shaun2222: production should still be using 1.2.x |
| 01:07.43 | robin_sz | JT, its the zaphfc stuff thats painful at the moment |
| 01:07.46 | xwire | was actually here to ask a question, which is kinda newb |
| 01:07.55 | blitzrage | xwire: just ask |
| 01:08.18 | Shaun2222 | blitzrage: this isnt like a small bug that i discovered while diging deap into this thing... this is displayed like first thing... wtf says somthings released that has a obvious bug in it |
| 01:08.25 | xwire | * may be open source, but it seems (after speninging many seconds reading) that the guy who started the project has a monoply on the hardware for it |
| 01:08.31 | blitzrage | Shaun2222: ok |
| 01:08.33 | robin_sz | JT, I'll search out my Xorcom/Rapid install CD again ... seemed a quick and easy way of doing bristuffed |
| 01:08.34 | Shaun2222 | i mean it's right in front of their face. |
| 01:08.40 | blitzrage | xwire: totally un-true |
| 01:08.41 | Shaun2222 | are you sure this is a bug. |
| 01:08.43 | xwire | is that true... and has anyone else used other hardware, such as a basic modem to interface with pots |
| 01:08.49 | blitzrage | Shaun2222: I never said it was a bug |
| 01:08.49 | JT | not so, sangoma, openvox and junghanns and others have hardware for * |
| 01:08.49 | Shaun2222 | and not just a conf error on my part? |
| 01:08.55 | xwire | . nice |
| 01:09.09 | xwire | there is not toom uch mention of that on the *now page |
| 01:09.11 | Shaun2222 | jesus, do you just tell everybody who displays warnings/errors to ignore them.. |
| 01:09.13 | blitzrage | Shaun2222: just ignore it -- its probably because the CLI is going through a state of deprecation |
| 01:09.13 | xwire | go figure :P |
| 01:09.14 | JT | robin_sz: installing bristuff is very easy |
| 01:09.19 | JT | xwire: wonder why :) |
| 01:09.30 | blitzrage | xwire: uh... well DUH! |
| 01:09.40 | JT | the book mentions sangoma though |
| 01:09.41 | blitzrage | it's called being a company |
| 01:09.43 | JT | ~thebook |
| 01:09.46 | jbot | somebody said thebook was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:09.46 | robin_sz | JT with the download/build/install scripts? |
| 01:09.50 | JT | robin_sz: yes |
| 01:09.56 | xwire | anyone know a decent indepandant site which talks about/rates/lists/whatevers hardware for *, with lots of hacks and the like? |
| 01:09.59 | blitzrage | I heard the guys who wrote it get tons of chicks |
| 01:10.02 | robin_sz | JT, well, maybe |
| 01:10.10 | Corydon76-home | Wake me when somebody actually gets US ISDN working for Asterisk |
| 01:10.12 | blitzrage | xwire: yah, google does |
| 01:10.20 | robin_sz | blitzrage, chick with dicks ? |
| 01:10.23 | JT | xwire: |
| 01:10.23 | xwire | google also licks my penis |
| 01:10.25 | JT | ~thewiki |
| 01:10.27 | jbot | [thewiki] at http://www.voip-info.org/wiki-Asterisk |
| 01:10.27 | blitzrage | robin_sz: not so much |
| 01:10.41 | k-man__ | i'm trying to configure my linksys phone to talk to my isp's sip server.. do i need to do anything special on the firewall? like forward a port form the firewall to the phone? |
| 01:10.55 | blitzrage | k-man__: usually not |
| 01:11.05 | Corydon76-home | blitzrage: louder, some chicks might start hearing you |
| 01:11.06 | k-man__ | ok |
| 01:11.07 | k-man__ | thanks |
| 01:11.13 | blitzrage | Corydon76-home: yah right :) |
| 01:11.34 | Corydon76-home | Jim must be getting all the chicks |
| 01:11.42 | blitzrage | Corydon76-home: yah, he's the party animal |
| 01:12.52 | JT | xwire: what are you trying to interface * with anyway? |
| 01:13.32 | xwire | not sure yet, just bored, and seems something fun to play with |
| 01:13.37 | *** join/#asterisk demigod2k (n=joey@cpe-65-29-113-212.twmi.res.rr.com) |
| 01:13.47 | JT | most of the pci stuff isn't cheap |
| 01:13.49 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 01:13.51 | JT | ATAs are cheaper |
| 01:13.59 | demigod2k | anybody use the Polycom 4000? is it nice? recommend anything else instead? |
| 01:14.18 | Corydon76-home | JT: but you can't plug a T1 into an ATA |
| 01:14.26 | xwire | im behind a corporate firewall a lot of the time, and am wanting to setup an incoming voip service, so I dont have to pay the bullshit amounts we get charged |
| 01:14.33 | xwire | 4cents per minute for a local call is bull shit |
| 01:14.36 | JT | true, i doubt xwire is looking at a t1 now |
| 01:14.37 | robin_sz | demigod2k, i recommend almost anything other than Grandstream :) |
| 01:14.38 | Corydon76-home | JT: and fax machines don't exactly like voip, either |
| 01:14.53 | JT | Corydon76-home: i realise |
| 01:14.57 | demigod2k | robin_sz: ya, we've got all polycom otherwise. speakerphones are always touchy though |
| 01:14.58 | *** join/#asterisk ManxPower (n=manxpowe@77.sub-75-202-95.myvzw.com) |
| 01:15.00 | HushPe | i got my TDM400P working :) |
| 01:15.08 | Corydon76-home | xwire: that's what I pay for my toll free number |
| 01:15.12 | HushPe | it's a little crackly though, like dial tones |
| 01:15.27 | xwire | yeah... i know, its shit, and its a university too |
| 01:15.45 | demigod2k | robin_sz: I tried a grandstream first (tempted by all the buttons). It's like the walmart quality phone of voip |
| 01:15.52 | robin_sz | 4c minute sounds dirt cheap to me |
| 01:16.05 | xwire | for local calls? |
| 01:16.06 | Corydon76-home | robin_sz: for a tollfree number, sure |
| 01:16.07 | robin_sz | demigod2k, you mean yours actually worked? |
| 01:16.22 | Corydon76-home | robin_sz: for local calls, not so much |
| 01:16.30 | demigod2k | robin_sz: ya it worked but it looked like it would be super high maintenance |
| 01:16.36 | perd | astDB data survives shutdowns, etc, right? |
| 01:16.38 | JT | 4c/min for a local call is terrible |
| 01:16.43 | JT | perd: yes |
| 01:16.45 | Corydon76-home | perd: correct |
| 01:16.46 | perd | sweet |
| 01:16.53 | Qwell | 4c/min for int'l is some terrible :p |
| 01:16.54 | robin_sz | demigod2k, wow. did you get a phot of it working? |
| 01:16.55 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:16.55 | Qwell | erm |
| 01:16.59 | Qwell | 4c/min for some int'l is terrible :p |
| 01:17.02 | xwire | funny thing was... when I was living there (its in Australia), calling the US... my calling card was 1.5cents per minute (awesome), but the university charged me 4cents a minute just to call the local number |
| 01:17.03 | xwire | pfft |
| 01:17.13 | Corydon76-home | 4c/minute to Cuba is awesome |
| 01:17.21 | demigod2k | robin_sz: hahaha no :) I bought this off-the-shelf asterisk server. I love this as a hobby but for work I dont want to screw around |
| 01:17.27 | JT | maybe the uni makes a profit |
| 01:17.35 | Corydon76-home | Then again, calls to Cuba only have a 1/3 success rate |
| 01:17.36 | xwire | perhaps |
| 01:17.44 | demigod2k | robin_sz: it came with a ready-to-go configuration for the grandstream which is probably the only reason it worked |
| 01:17.45 | xwire | either that, or they get raped by telstra/govt contracts/whatever |
| 01:17.55 | xwire | perhaps an old ISDN contract |
| 01:17.57 | robin_sz | demigod2k, damn! ... a photo of a working Grandstream GXP2000 .. you could have sold that to fox news! |
| 01:18.15 | demigod2k | it didnt work well, if htat counts for something |
| 01:18.45 | JT | xwire: the uni in us or au charged 4c/min? |
| 01:18.46 | robin_sz | demigod2k, mine works great .. it keeps the door open, or can stop the truck rollign back on moderate slopes |
| 01:18.54 | robin_sz | not much use as a phone though |
| 01:19.06 | demigod2k | hahaha nice. ya we went all polycom 301/501 and I was looking at the 4000 speakerphone although it's $700 which is steep |
| 01:19.20 | robin_sz | Snom/Elmeg 290 here |
| 01:19.50 | demigod2k | I'm happy with everything except the message waiting indicator -- beeps every 30 seconds, no way to reduce the frequency of beeping |
| 01:20.57 | demigod2k | robin_sz: do you use any cordlesses? |
| 01:21.04 | robin_sz | yeah |
| 01:21.27 | robin_sz | sipura ATA and dect phones :) |
| 01:21.56 | robin_sz | we tired some wifi phones .. zyxel and one other ... shite. |
| 01:22.15 | robin_sz | afaik, there are no working wifi phones |
| 01:22.16 | demigod2k | what was the problem, echo? battery life? |
| 01:22.30 | robin_sz | inablity to connect to the base relaibly |
| 01:22.33 | xwire | anyways, ive been moving around the world a lot lately, and am looking at purchasing something similar to vonage (although, definately not vonage) so I can keep the same phone number whereever.. the problem is, lots of the time I am behind firewalls with no upnp, or forwarding ports/whatever... any advise from anyone? (besides to give up) |
| 01:22.37 | robin_sz | shit battery life |
| 01:22.42 | robin_sz | shit hardware |
| 01:22.49 | robin_sz | generally, shit. |
| 01:22.56 | xwire | *advice |
| 01:22.57 | xwire | heh |
| 01:22.59 | xwire | i can't spell |
| 01:23.03 | JT | xwire: get a server somewhere, and connect to it with IAX |
| 01:23.15 | robin_sz | i honestly would have prefered they sold me a block of wood |
| 01:23.27 | robin_sz | that way I wouldnt have wasted time trying to make it work |
| 01:23.38 | JT | xwire: or get a server somewhere, and connect to it using your mobile with ringback |
| 01:23.44 | demigod2k | ya. I thought about the ATA but then you lack the forwarding, other software features, without hitting # key combinations |
| 01:23.51 | JT | you could use a landline too |
| 01:24.06 | robin_sz | trust me ... its a better option |
| 01:24.12 | zero-G | from someone who has just enough telephony experience to be dangerous ... just what kind of hardware would you need to utilize asterisk with POTS lines if you weren't using digium's cards in a pc? |
| 01:24.17 | robin_sz | wifi dont work |
| 01:24.28 | JT | zero-G: SIP ATA |
| 01:24.47 | robin_sz | or an old intel modem |
| 01:25.02 | JT | robin_sz: he already said not using cards in pc though |
| 01:25.05 | demigod2k | zero-G: you can use an ATA (ethernet -> phone adapter). we actually use the digium PCI cards in my office which works great |
| 01:25.08 | robin_sz | no he didnt |
| 01:25.43 | robin_sz | he said " if you weren't using digium's cards in a pc" |
| 01:25.46 | JT | i'm pretty sure the X100P and clones count as "digiums cards in a pc" even if they're not really |
| 01:26.04 | demigod2k | robin_sz: ya it seemed like wifi wouldnt scale for a full office. I was hesitant to drop $300 on a phone to learn that lesson though. we switched from a panasonic full cordless system |
| 01:26.22 | JT | demigod2k: smart man :) |
| 01:26.36 | *** join/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net) |
| 01:26.37 | robin_sz | JT, I assumed he meant "I dont want to spend $300 on a card, what alternative is there" |
| 01:26.52 | JT | ah, well that's possible |
| 01:26.59 | JT | i wouldn't recommend X100P anyway |
| 01:27.00 | ManxPower | It really depends on how many people you want to be able to scale to |
| 01:27.03 | *** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net) |
| 01:27.03 | JT | i assume he wants it to work |
| 01:27.04 | demigod2k | JT, so far we love it. every phone system has its issues (ours included). but for the price it's a steal |
| 01:27.15 | zero-G | actually i was wondering was ther other equipment out there that could be controlled by * that wasn't necessarily inside the computer |
| 01:27.19 | robin_sz | JT, it does work .. sorta ;) |
| 01:27.30 | robin_sz | JT, but yeah the SIP ata is a better option |
| 01:27.32 | JT | zero-G: yeah, an ATA might be what you are after |
| 01:27.48 | robin_sz | right |
| 01:28.30 | zero-G | sorry, but what's ATA in plain english :) |
| 01:28.40 | JT | analogue telephony adapter |
| 01:28.50 | Strom_C | not quite |
| 01:28.50 | JT | a box that has an ethernet port and one or more phone ports |
| 01:28.52 | robin_sz | see ebay for details |
| 01:28.55 | Strom_C | analog terminal adapter :) |
| 01:29.03 | JT | right |
| 01:29.10 | zero-G | OK thnx! |
| 01:29.41 | robin_sz | analogue tlephoen line, analogue phone port and an ethernet |
| 01:29.52 | robin_sz | and a problem usually :) |
| 01:29.57 | JT | and power |
| 01:30.00 | zero-G | what exactly is a channel bank? |
| 01:30.03 | robin_sz | yeah |
| 01:30.19 | robin_sz | 20 phone ports in a box |
| 01:30.20 | JT | robin_sz: maybe you're describing a sipura 3102 |
| 01:30.23 | robin_sz | or line ports |
| 01:30.30 | robin_sz | JT, 2102 |
| 01:30.39 | JT | zero-G: a box that converts between a PRI like a T1 and lots of ports (usually) |
| 01:31.19 | robin_sz | JT, the 2102 is two phones, a "internet" port and a LAN port |
| 01:31.28 | robin_sz | JT, and a problem |
| 01:31.39 | Strom_C | I've never seen a channel bank that operates on a PRI |
| 01:31.48 | robin_sz | JT, it is incabable of believing your SIP server is on your LAN |
| 01:31.58 | mmlj4 | probably it's just a 2-port switch |
| 01:32.02 | Strom_C | you get a channelized T1 out one end and a bunch of analog circuits out the other |
| 01:32.03 | JT | robin_sz: the 2102 is 2 X FXS |
| 01:32.10 | JT | robin_sz: right |
| 01:32.11 | robin_sz | JT, regardless of the netmask |
| 01:32.27 | JT | robin_sz: 2102 doesn't connect to phone line though |
| 01:32.31 | robin_sz | true |
| 01:32.37 | robin_sz | 3102 ? |
| 01:32.40 | JT | yeah |
| 01:32.44 | *** join/#asterisk alrs (n=lars@dsl093-066-021.lax1.dsl.speakeasy.net) |
| 01:32.53 | robin_sz | will that believe yor * box is on your LAN? |
| 01:32.55 | JT | Strom_C: yeah, i meant CAS T1/E1 |
| 01:33.16 | JT | robin_sz: dunno, heard of plenty using it with * |
| 01:33.24 | *** part/#asterisk alrs (n=lars@dsl093-066-021.lax1.dsl.speakeasy.net) |
| 01:33.55 | robin_sz | yeah, my 2102 refused to send SIP lanwards .. regardless of network settings ... SIP was going "internet" port, but never LAN port |
| 01:34.25 | *** part/#asterisk generalhan (n=asd@67.90.64.2) |
| 01:34.26 | mmlj4 | interesting |
| 01:34.43 | robin_sz | got it working, but not between the router and the rest of the notwork .. sits as a device onthe net, whichj I prefer anyway |
| 01:34.49 | JT | robin_sz: if i bought one i'd probably never use the router function |
| 01:34.53 | robin_sz | quite |
| 01:34.56 | JT | but that's a good headsup anyway |
| 01:35.11 | robin_sz | right |
| 01:35.16 | robin_sz | bedtiem! |
| 01:35.23 | *** part/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 01:35.25 | JT | night |
| 01:35.54 | *** part/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 01:44.59 | J4k3 | whee... non-hiccupy asterisk |
| 01:45.20 | J4k3 | tomorrow the p-m 915 arrives to further insist in non-hiccupyness |
| 01:45.23 | J4k3 | err |
| 01:45.24 | J4k3 | p-d 915 |
| 01:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:46.57 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:47.19 | JT | J4k3: what is that? |
| 01:48.48 | *** join/#asterisk xpot (n=xpot@dsl093-228-250.slc1.dsl.speakeasy.net) |
| 01:49.36 | xpot | can somebody assist me with the following issue? : ERROR[21061]: pbx.c:1498 ast_func_read: Function Cut not registered |
| 01:50.01 | xpot | make menuselect shows that func_cut is enabled (with asterisk) |
| 01:50.07 | xpot | ?? |
| 01:50.07 | J4k3 | JT: Intel pentium-d dual core "915" chip, which is 2x2.8ghz w/ 2MB L2 per core (4mb total) |
| 01:50.16 | JT | ah |
| 01:50.43 | JT | J4k3: how much does that cost? |
| 01:51.15 | *** join/#asterisk niZx (n=bleh@voip.nizon.ca) |
| 01:51.44 | J4k3 | JT: 106 for the cpu, mobos can be had for $50 and up |
| 01:52.15 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 01:52.19 | J4k3 | "does geeksquad come in a box or can?" |
| 01:52.21 | JT | hmm, not bad |
| 01:52.26 | J4k3 | "CAN I GET GEEKSQUAD ON A DVD?" |
| 01:55.24 | xpot | ERROR[21061]: pbx.c:1498 ast_func_read: Function Cut not registered... any help? |
| 01:56.13 | J4k3 | now all I need is a headset that doesn't say "playstation" on it. |
| 01:57.49 | JunK-Y | xpot: load it? |
| 01:58.10 | Daveb21 | Has anyone got a recommendation for VoIP provider in Australia? Looking at about 4 VoIP lines i'd say. |
| 01:58.20 | sivana | xpot: load func_cut.so |
| 01:58.25 | JT | Daveb21: depends on your criteria |
| 01:58.27 | xpot | I am not sure how? I already checked the make menuselect |
| 01:58.35 | xpot | oh, I will try that |
| 01:58.45 | sivana | might be: module load |
| 01:59.07 | xpot | WARNING[20618]: loader.c:597 load_resource: Module 'func_cut.so' already exists. |
| 01:59.16 | xpot | ?? |
| 01:59.22 | sivana | there you go |
| 01:59.35 | xpot | I still get the error |
| 02:00.25 | Daveb21 | JT: Havent really narrowed that down too much yet - the guy I've got researching providers reckons that most of the ones hes found look like "home" 1 line services rather than business services |
| 02:01.02 | JT | Daveb21: well, price or quality |
| 02:01.07 | JT | any particular did requirements |
| 02:01.15 | JT | do you need 4 public numbers |
| 02:01.20 | Qwell | you all ready for the State of the Union drinking game? |
| 02:01.28 | Qwell | Take a drink every time Bush says something stupid |
| 02:01.29 | Daveb21 | JT: brb in 2 mins |
| 02:01.42 | JT | sounds like some sort os speech we ignore in Australia ;) |
| 02:01.45 | JT | s/os/of/ |
| 02:01.47 | ClydeGoffe | hey has anyone gotten asterisk to work with sipphone.com using virtual numbers? |
| 02:01.57 | ClydeGoffe | whenever i recieve calls the person can hear me but I can't hear them |
| 02:02.02 | ClydeGoffe | not sure what's causing that |
| 02:02.08 | ClydeGoffe | if i call out everything works fine |
| 02:02.11 | ClydeGoffe | any ideas? |
| 02:03.49 | Daveb21 | JT: Probably quality i'd say - its a business services - we'll only be using it for outgoing calls |
| 02:04.11 | JT | Daveb21: do you have any PSTN backup? |
| 02:04.30 | *** join/#asterisk dseeb_ (n=dcb@58.165.244.192) |
| 02:05.15 | Daveb21 | JT: We're looking at 4 incoming lines (ISDN) that we'll use as a backup in the event the VoIP provider or Internet link goes down. We'll also have a couple of PSTN lines for ADSL but those wont be linked to the Asterisk server |
| 02:06.23 | JT | Daveb21: personally, i don't think the provider needs top notch uptime since you have decent pstn backup |
| 02:06.34 | JT | 4 * BRI i assume (8 chans) |
| 02:06.50 | Daveb21 | JT: The catalyst for all this is we're moving offices and have to put in our own PBX system. And we're an IT company dealing mainly in linux so we figured we'd go an Asterisk solution with VoIP handsets and a VoIP provider for outgoing calls |
| 02:07.02 | JT | sounds like a plan |
| 02:07.03 | Daveb21 | JT: 2 * BRI |
| 02:07.07 | JT | ok |
| 02:07.23 | JT | yeah, engin is pretty good for outgoing only |
| 02:07.34 | Daveb21 | JT: Can you poke any holes in that theory :o) |
| 02:07.57 | JT | unlimited simultaneous inbound or outbound calls with engin, 20c/m to mobiles on business 50 plan |
| 02:08.09 | JT | sounds like a good plan |
| 02:08.09 | Daveb21 | JT: Thats not bad |
| 02:08.15 | JT | i've done similar stuff before |
| 02:08.52 | Daveb21 | JT: Excellent so im not completely crazy then hahah |
| 02:09.23 | JT | only thing is, which is really a matter of how fussy your users are, is that engin has network wide RTP silence supression on the leg of the call from their end to yours |
| 02:09.35 | JT | so for users it can sound unnervingly quiet when there's no talking |
| 02:09.55 | JT | and if the other end has low volume, it may clip a bit |
| 02:10.14 | Daveb21 | JT: I'll be sure to tell them not too be spooked by it - that would save a bit on bandwidth tho yeah? |
| 02:10.26 | JT | yes |
| 02:10.30 | Daveb21 | JT: Obviously they support Asterisk |
| 02:10.46 | perd | my silence sounds dont work! |
| 02:10.47 | JT | bandwidth is not much of a concern if you can get broadband |
| 02:10.50 | perd | what gives :( |
| 02:11.04 | JT | they don't support asterisk, you can connect it to asterisk if you're on a "voiper" plan |
| 02:11.28 | JT | but they won't tell you how to troubleshoot asterisk |
| 02:11.59 | Daveb21 | JT: Sorry meant traffic not bw - probably wont matter anyway as we'll be on a 20 or 60Gb plan |
| 02:12.20 | JT | Daveb21: also, you either need good qos (and upstream data), or a dedicated broadband link |
| 02:12.29 | JT | voip traffic is minimal, even with no compression |
| 02:12.33 | JT | whuch is how i prefer it |
| 02:13.05 | Daveb21 | JT: yeah definitely looking at a dedicated ADSL2 link, and will probably QoS anyway in case |
| 02:13.46 | JT | i'm curious what your average call volume/cost is |
| 02:13.53 | JT | you don't have to say of course |
| 02:14.12 | Daveb21 | JT: So basically with Engin you just configure Asterisk to "use this route/gateway/provider" (not sure on the proper term) and it will put all calls through there - you dont use IAX or anything funky like that |
| 02:14.26 | JT | Daveb21: you need to use SIP |
| 02:14.28 | Daveb21 | JT: Im trying to get that info now from our people |
| 02:14.40 | JT | don't have access to phone bills? |
| 02:15.23 | Daveb21 | JT: We dont control our PABX at the mo - we piggyback off another company upstairs and they charge us. Yep, but im the tech guy and I gotta get it off the admin ppl |
| 02:15.41 | JT | nasty |
| 02:16.10 | JT | piggybacking doesn't sound like fun |
| 02:16.21 | Daveb21 | apparently its $5 per extension per month and our call charges are about $600pm |
| 02:16.40 | JT | relatively low i guess |
| 02:16.44 | JT | for 4 lines anyway |
| 02:16.45 | Daveb21 | but they are using Nodephone for their STD calls already so were already on a cheap std rate |
| 02:16.52 | JT | ah |
| 02:17.12 | xpot | ERROR[21690]: pbx.c:1498 ast_func_read: Function Cut not registered => still having this issue... any other suggestions? |
| 02:17.13 | JT | do you do much std? |
| 02:17.47 | Daveb21 | apparently their std calls went from 1200pm to 55pm |
| 02:17.56 | JT | heh yeah |
| 02:17.58 | Daveb21 | nah not really - mostly local and mobile |
| 02:18.03 | JT | it really depends on the volume though |
| 02:18.10 | JT | do you get good rates on mobile now? |
| 02:18.28 | Daveb21 | dunno - trying to get a breakdown on call costs at the mo |
| 02:18.46 | Daveb21 | id suggest thats where most of our 600pm goes |
| 02:18.55 | JT | i built a spreadsheet app that can very accurately make call costs comparisons based on an actual bill from a typical month |
| 02:19.19 | JT | unlike the crappy guestimations that sales staff always try to give, that only look at totals |
| 02:19.19 | Daveb21 | cool |
| 02:19.25 | *** part/#asterisk zero-G (n=mark@69-2-64-170.wan.networktel.net) |
| 02:19.26 | JT | this works on the cost of every single call |
| 02:19.33 | JT | but you need the data in electronic format |
| 02:19.56 | JT | with duration, etc |
| 02:19.57 | Daveb21 | we're looking at this solution as not only for us but potentially for our customers too |
| 02:20.10 | Daveb21 | yeah - doubt id be able to get a hold of something like that |
| 02:20.26 | Daveb21 | ill just be guessing :) |
| 02:20.31 | JT | you could try scanning it in with ocr, probably wouldn't be fun |
| 02:20.39 | k-man__ | how can i tell if my linksys spa942 is connecting to the sip provider? |
| 02:20.44 | k-man__ | i can't make a call on it |
| 02:20.49 | k-man__ | and it doesn;t really tell me much |
| 02:21.26 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:21.59 | Daveb21 | JT: what about maximum uptime strategies such as clustering or failover/load balancing - have you dealt much with those? |
| 02:22.07 | perd | this IF statement is going to be true if ACTUAL_EXTEN is null, right? ${IF($["${ACTUAL_EXTEN}"=""]? |
| 02:22.23 | JT | Daveb21: for 4 lines, it seems like overkill to have a cluster |
| 02:22.27 | JT | how many extensions? |
| 02:23.10 | perd | oh nm i had a misplaced " |
| 02:23.10 | ManxPower | perd: in 1.2 and later, I believe so. you might need spaces around the = |
| 02:23.12 | perd | friggen a :) |
| 02:23.25 | Daveb21 | about 20 - i was thinking the same also but would like to have a strategy if the server falls over or if we grow it is easy to move to a clustered solution |
| 02:23.26 | perd | exten => s,1,Set(extension=${IF($["${ACTUAL_EXTEN}"=""]?${IF($["${LEN(${MACRO_EXTEN})}" = "7"]?${MACRO_EXTEN:3}:${MACRO_EXTEN})}:${ACTUAL_EXTEN})}) |
| 02:23.40 | perd | oh yeah, i love extensions.conf logic. |
| 02:23.43 | perd | WTS |
| 02:24.02 | perd | i think that syntax just broke my head |
| 02:24.21 | JT | Daveb21: pretty much all * redundancy and clustering solutions involve accepting that you will lose currently active calls |
| 02:25.30 | Daveb21 | Yeah I figured as much but even something that would failover within a couple of seconds so new calls could be made |
| 02:25.47 | JT | that's possible |
| 02:26.20 | Daveb21 | My initial thoughts are our worst case scenario we'll simply plug a phone into our PSTN line being used for ADSL and call forward the main number from ISDN |
| 02:26.46 | JT | yeah |
| 02:26.47 | Daveb21 | is it something that is done within asterisk or do you use Linux-HA or LVS style stuff |
| 02:26.57 | JT | that's pretty last ditch redundancy :) |
| 02:27.05 | JT | that can be done with isdn too |
| 02:27.31 | JT | Daveb21: it's up to you to work out a redundancy solution, taking care of your lines is usually the hardest bit |
| 02:27.43 | JT | also fallover proofing the box is a good idea |
| 02:27.44 | Daveb21 | need an ISDN phone tho dont ya? cant plug standard handset into ISDN line? |
| 02:28.05 | JT | isdn phone or an NT1+ |
| 02:28.13 | JT | and analogue phone |
| 02:28.37 | Daveb21 | as you can see my telephony knowledge aint flash :o) |
| 02:29.02 | Daveb21 | box will definitely by RAID disk, UPS |
| 02:29.32 | JT | RAID1 + redundant power supply + online ups preferably is my recommendation |
| 02:29.48 | JT | make sure the ups powers the PoE capable switch too |
| 02:30.29 | Daveb21 | yep got in that listed in my requirements - not sure if theyll go redundant power tho |
| 02:30.51 | J4k3 | UPSes are evil, but theres a general lack of decent pure-dc solutions. |
| 02:30.59 | J4k3 | thats my personal side project. |
| 02:31.04 | JT | it's pretty stupid not to go redundant power supplies for telephony or any servers, really |
| 02:31.37 | JT | in fact for asterisk, it's better to buy a second hand unit that meets all requirements than buy a bare bones brand new nox |
| 02:31.41 | JT | box |
| 02:31.59 | JT | J4k3: fabricating bus bars, too? |
| 02:32.04 | Daveb21 | yeah im in agreement there - I want to go Tier1 hardware (IBM,HP) preferably HP but we dont generally by brand hardware |
| 02:32.23 | JT | ibm is pretty good |
| 02:32.29 | JT | also, don't buy a pentium 4 |
| 02:32.42 | JT | get a xeon or even a PIII |
| 02:32.43 | Daveb21 | Ive had some bad experience with IBMs in the past so....... :) |
| 02:33.50 | J4k3 | JT: nah, texas metal casting is good for that sort of stuff ;) |
| 02:34.00 | JT | J4k3: heh |
| 02:34.07 | JT | they do cast copper? |
| 02:34.26 | J4k3 | yeah, they do all sorts of fun stuff with copper and brass |
| 02:34.32 | Daveb21 | what about HDD i'm thinking 2x72Gb should do the trick for a business of our size (20 extensions/users) |
| 02:34.43 | J4k3 | they also work alumimum out there, but thats less interesting. |
| 02:34.54 | JT | Daveb21: 2GB would be enough if you don't record calls |
| 02:35.07 | JT | recording calls is the only thing you need space for really with asterisk |
| 02:35.22 | Daveb21 | yeah but try getting HDDs with capacity below 40Gb nowadays :oP |
| 02:35.38 | JT | yeah i know, it's nice to have space to play with, anyway |
| 02:35.52 | JT | 72GB is more than enough unless you're recording a lot |
| 02:35.59 | Daveb21 | voicemail take up much space? i wouldnt have thought so |
| 02:36.05 | JT | oh, and system prompts take up space, but not that much |
| 02:36.10 | JT | voicemail takes a bit |
| 02:36.16 | JT | no, i mean recording calls |
| 02:36.29 | JT | like callcentres and 000 do |
| 02:36.39 | Daveb21 | yeah - we definitely dont have a requirement to record calls "for quality and coaching purposes" |
| 02:37.01 | Daveb21 | I |
| 02:37.07 | Daveb21 | damn |
| 02:37.10 | JT | 72GB might not be enough if you had 100lines and kept recordings for months |
| 02:37.51 | Daveb21 | im looking at integrating the voicemail with our messaging environment also once we finally figure out which one we're going with |
| 02:38.12 | JT | right |
| 02:38.35 | Daveb21 | Top stuff, thanks for all that info JT - back in 5 after i pop over the road and grab some lunch |
| 02:38.54 | JT | i should probably do similar at some stage |
| 02:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:57.22 | neddy | I'd like to set up an asterisk server so that my family can do VOIP to me -- don't care about landlines or much else -- what part of asterisk do I need for that? |
| 02:57.25 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 02:57.35 | neddy | what dialplan would that be? |
| 03:00.52 | pigpen | Does anyone know if the orphaned txt file issue has been resolved in recent ver's of Asterisk? (or still an issue?) |
| 03:00.58 | sevard | neddy: you'll want to figure out if you want to use analog lines in your home or voip phones, and how many, and how are you going to get service to your home |
| 03:01.06 | sevard | once you have those pinned down you're golden |
| 03:01.14 | neddy | sevard: no analog at all |
| 03:01.33 | neddy | I just want to install apps on their computers and use the mic and speakers on the computer to talk |
| 03:01.45 | sevard | alright, then you can get up and running right now |
| 03:01.48 | JT | softphone |
| 03:01.57 | neddy | softphone? |
| 03:02.00 | sevard | there's xlite or zoa's whatever client for windows, and there are softphones for linux aswell |
| 03:02.03 | sevard | software phones. |
| 03:02.07 | JT | tghat's what you're talking about |
| 03:02.15 | sevard | "install apps on their computers" == softphones |
| 03:02.20 | neddy | do you have a favorite MacOS softphone? |
| 03:02.44 | pigpen | idefisk |
| 03:02.46 | neddy | so, what part of asterisk needs setup for that? |
| 03:02.53 | pigpen | I use it allot. |
| 03:02.55 | neddy | idefisk? |
| 03:03.13 | neddy | I don't need a 'service provider' - do I ? |
| 03:03.13 | JT | /etc/asterisk/extensions.conf and /etc/asterisk/sip.conf or iax.conf |
| 03:03.20 | pigpen | http://www.asteriskguru.com/idefisk/ |
| 03:03.27 | JT | not if you don't want to talk to the phone network |
| 03:03.27 | sevard | you'll need to install asterisk on a server in your home, set up the sip clients, register them with the asterisk box, and build your dialplan. |
| 03:03.46 | JT | well, the server can be anywhere |
| 03:03.46 | sevard | if you want PSTN termination you'll need such a provider, if you just want to talk in house, then now. |
| 03:03.48 | sevard | no* |
| 03:04.01 | neddy | pigpen: very nice! |
| 03:04.01 | sevard | JT: of course the server can be anywhere, foolz. |
| 03:04.23 | JT | sevard: just saying, sometimes it needs to be spelt out to people |
| 03:04.25 | pigpen | neddy, very stable....at least on my quad mac, mac book pro and others... |
| 03:04.30 | neddy | what's the benefit of iax.,conf over sip.conf? |
| 03:04.49 | JT | iax protocol only uses a single port, so less issues with nat |
| 03:04.53 | sevard | they're different protocols, there's really no benifit if you're not doing NAT |
| 03:04.58 | neddy | if PSTN is Plain Standard Telephone Network, the answer is I don't care |
| 03:05.03 | neddy | I don't even have a landline |
| 03:05.14 | JT | PSTN is Public Switched Telephone Network |
| 03:05.22 | sevard | neddy: you can get VoIP service from a service providor to talk to the rest of the world on the telephone network |
| 03:05.31 | sevard | POTS would be plain old telephone service |
| 03:05.37 | sevard | erm, set |
| 03:05.52 | neddy | I see |
| 03:06.15 | neddy | Well, to my family, I will have to do NAT because they live all over the place |
| 03:06.17 | sevard | If you'd like a friendly contractor to do all of this for you, look no farther |
| 03:06.39 | JT | lol |
| 03:06.46 | neddy | sevard: I'm a geek trying to learn more, so I'm hoping to figure this out myself |
| 03:06.48 | *** join/#asterisk jlimb (n=user@networkv.dsl.xmission.com) |
| 03:06.52 | sevard | right on |
| 03:06.58 | JT | neddy: |
| 03:06.59 | neddy | I'm installing this on the machine that currently hosts just MythTV for me |
| 03:07.00 | JT | ~thebook |
| 03:07.02 | jbot | extra, extra, read all about it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:07.02 | neddy | it's always on |
| 03:07.29 | neddy | very nice |
| 03:07.41 | pigpen | anyone know if the voicemail app was overhauled in 1.4.x |
| 03:07.42 | pigpen | ? |
| 03:08.17 | pigpen | Either the documentation is well hidden, or I am too lazy.... :) |
| 03:08.48 | sevard | neddy: it's worth a read, it answeres most, if not all, questions you'll begin asking |
| 03:08.52 | neddy | So, which extensions do I need? |
| 03:08.58 | JT | extensions? |
| 03:08.58 | neddy | and, what dialplan do I need? |
| 03:09.08 | JT | the one that you write? |
| 03:09.15 | neddy | I built and installed the asterisk-gui yesterday and started the wizard |
| 03:09.24 | neddy | but I didn' t know what they mean by dialplan |
| 03:09.26 | sevard | neat, get rid of the gui and read the book. |
| 03:09.32 | JT | the dialplan is extensions.conf |
| 03:09.37 | neddy | I set up myself as a user with extension 6000 |
| 03:09.52 | neddy | I setup extension 7000 as the voicemail extension |
| 03:10.16 | sevard | sounds like logical digit maping :| |
| 03:10.16 | neddy | now, I'll try kphone or idefisk to see if I can check my voicemail |
| 03:10.33 | *** join/#asterisk k-man__ (n=jason@ppp244-232.static.internode.on.net) |
| 03:10.39 | neddy | What is the asterisk-lingo for a teamspeak-like server |
| 03:10.40 | neddy | ? |
| 03:10.46 | pigpen | yeah...learn it in console..then graduate to your own custom built gui.... |
| 03:10.49 | sevard | a conference? |
| 03:10.56 | neddy | One where everyone registers and dials in and you have a big conference room... |
| 03:10.59 | sevard | a conference? |
| 03:11.06 | neddy | ok |
| 03:11.10 | sevard | heh |
| 03:11.14 | JT | MeetMe |
| 03:11.21 | neddy | MeetMe? |
| 03:11.40 | JT | yes, that's the answer |
| 03:11.45 | pigpen | imapstorage.....hehe..... |
| 03:12.00 | JT | please either read the book, or search google, instead of repeating the answer i give :) |
| 03:12.46 | neddy | My google-fu is weak |
| 03:12.52 | neddy | lots and many hits for asterisk |
| 03:13.03 | JT | ~thewiki |
| 03:13.05 | jbot | rumour has it, thewiki is at http://www.voip-info.org/wiki-Asterisk |
| 03:13.12 | neddy | so I'm trying to narrow the scope so I can find individual terms to search for |
| 03:13.28 | sevard | neddy: google( site:voip-info.org <subject ) |
| 03:13.37 | dendrite | neddy: The book cited is an easy read. Really, plan to spend only a few hours, and you'll be MUCH clearer. |
| 03:13.46 | *** join/#asterisk JSabines (i=JSabines@189.158.190.152) |
| 03:14.18 | neddy | nice |
| 03:14.56 | neddy | did you say hours?? |
| 03:15.10 | JT | yes, i believe so |
| 03:15.13 | dendrite | It's not a very thick book. It's a good introduction. |
| 03:15.18 | neddy | <whiny voice>Dude, that is such a long time... |
| 03:15.26 | JT | you don't need to read the appendicies either |
| 03:15.40 | JT | welll if you don't have much experience in telecommunications, you'll need it |
| 03:16.03 | JT | i thought you said you wanted to learn :) |
| 03:16.18 | sevard | I believe I said.... |
| 03:16.22 | sevard | If you'd like a friendly contractor to do all of this for you, look no farther |
| 03:16.36 | sevard | Asterisk is a big scary beast and if you're not man enough... |
| 03:16.38 | JT | i'm sure a lot of us could set it up for you neddy |
| 03:16.40 | neddy | Learn? Does that mean reading? |
| 03:16.41 | neddy | bummer |
| 03:16.54 | JT | reading is the quick option |
| 03:16.58 | neddy | I'm just trying to be funny |
| 03:17.03 | sevard | you're failing. |
| 03:17.11 | JT | the slower one is to try random shit in configuration files and see if it does anything |
| 03:18.03 | neddy | My mythtv experience started off kinda weak too, because I didn't know anything about it and I started by installing just the driver for my tv tuner card and then by making it work with something simple like tvtime and then finally installing mythv and then a big logical volume |
| 03:18.18 | neddy | I'm having a harder time figuring out where to start with asterisk |
| 03:18.28 | neddy | so, I think I'll read the book |
| 03:18.45 | Daveb21 | just to clear up something, the PCI 2.2 standard used by most digium cards is NOT the same as PCI-x right? |
| 03:19.00 | JT | correct |
| 03:19.13 | JT | pci 2.2 specifices a bus voltage of 3.3v |
| 03:19.16 | JT | specifies |
| 03:19.24 | Daveb21 | damn servers all come with PCI-x nowadays |
| 03:19.28 | neddy | which is still different than PCI-e, right? |
| 03:19.35 | *** join/#asterisk grandy (n=chatzill@c-71-198-130-108.hsd1.ca.comcast.net) |
| 03:19.43 | JT | Daveb21: so? the cards should still work |
| 03:22.38 | Daveb21 | confused..... pci-e / pci-x / pci express - all the same? which ones can the digium cards (specifically B410P) plug into - forgive my ignorance been a while since i delved into hardware - normally its just gimme a server with X CPU, X RAM and X HDD |
| 03:22.41 | *** join/#asterisk ltd (n=z@202-161-1-26.dyn.iinet.net.au) |
| 03:23.01 | JT | pci-e != pci-x |
| 03:23.05 | *** join/#asterisk Trevor_B (n=tbenson@64-142-72-32.dsl.static.sonic.net) |
| 03:23.17 | JT | pci will go into pci-x slot |
| 03:23.42 | *** join/#asterisk Avochelm (n=damien@gw-morphett.koalatelecom.com.au) |
| 03:23.54 | pigpen | but, just because a PCI card fits into a PCI-X slot, does not mean it will work....but 90% of the time it works... |
| 03:24.06 | JT | if it's 3.3v it should work |
| 03:24.12 | Daveb21 | ah ok - so many standards makes my head hurt |
| 03:24.17 | JT | if you have anything else on the bus, it may slow it down |
| 03:24.29 | Corydon76-home | The nice thing about standards is that there are so many to choose from |
| 03:24.37 | JT | i was going to say that |
| 03:24.40 | Daveb21 | JT: card 3.3v or slot 3.3v? |
| 03:24.46 | JT | 3.3v card |
| 03:24.55 | pigpen | I have this issue with a video capture card, where the pci bridge isn't compatable...ie: cheap card. |
| 03:25.04 | J4k3 | wow.... why does a SIP call use 160kbit/sec of bandwidth? :| |
| 03:25.15 | JT | each direction, J4k3 ? |
| 03:25.18 | JT | or both |
| 03:25.23 | J4k3 | JT: total both ways |
| 03:25.28 | J4k3 | its right at 80k both ways |
| 03:25.30 | JT | that's normal with g.711 |
| 03:25.38 | J4k3 | between the voip provider and my asterisk |
| 03:25.40 | JT | i've found it closer to 85kbit/s each way |
| 03:25.47 | JT | if qualify is on |
| 03:26.06 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 03:26.09 | J4k3 | hrm |
| 03:26.14 | JT | codec is 64kbit/s, rest is overhead |
| 03:26.17 | J4k3 | yeah |
| 03:26.20 | Daveb21 | ok thanks guys - makes my job of identifying appropriate server hardware MUCH easier :) |
| 03:26.30 | J4k3 | 64kbit/sec sounds pretty well noncompressed :) |
| 03:26.32 | pigpen | how do you start an asterisk console logger again? |
| 03:27.06 | grandy | hello... how do i troubleshoot why i have poor quality iax calls in spite of a 60ms ping time |
| 03:27.23 | JT | J4k3: it is uncompressed, however is also companded, which can be argued is a form of compression |
| 03:27.39 | pigpen | grandy, you may want to define "poor quality" |
| 03:27.47 | JT | pigpen: /etc/asterisk/logger.conf? |
| 03:27.50 | Daveb21 | ah screw it ill just give the specs to the hardware supplier and say gimme a price on a whitebox and on a Tier1 machine hehe |
| 03:28.11 | grandy | pigpen: occasional jitter and missed words... |
| 03:28.21 | JT | Daveb21: it's not that hard really :) |
| 03:28.50 | JT | Daveb21: can you see my messages? |
| 03:28.53 | pigpen | JT, I thought I remembered a "side daemon" that dumps it out to a txt file.... |
| 03:29.08 | Daveb21 | JT: yeah im just lazy :P |
| 03:29.09 | JT | pigpen: you can make logger.conf do that |
| 03:29.13 | pigpen | hmm... |
| 03:29.28 | J4k3 | ahhh, with GSM its about 60kbit total, thats better. |
| 03:29.29 | pigpen | grandy, is this IAX going over dsl/cable modem? |
| 03:29.37 | JT | full log |
| 03:29.48 | JT | J4k3: if you have no bandwidth, i guess it's better :) |
| 03:29.59 | JT | g.729 sounds better than gsm and uses less |
| 03:30.01 | [TK]D-Fender | grandy : Your amazingly fast connection is capable of losing packets better that everyone else :) |
| 03:30.06 | J4k3 | JT: well, it sounds pretty awful :) |
| 03:30.33 | pigpen | [TK]D-Fender, well put....my line is the "King" of loss... |
| 03:30.37 | grandy | pigpen: the setup is: asterisk in a datacenter with a 20 megabit connection that is very fast to the backbones... and then i am using a cable modem to connect to that box and using idefisk on my laptop |
| 03:30.40 | J4k3 | JT: hrm... whats the setting to enable 729? |
| 03:31.02 | Corydon76-home | allow=g729 |
| 03:31.06 | J4k3 | ahh |
| 03:31.07 | JT | J4k3: go to digium.com and buy transcoding licences and then activate them, then it needs to be in a high priority i guess |
| 03:31.21 | grandy | pigpen: seems however that calls that don't go through voipstreet (the origination/termination provider) are of superb quality, but that the ones that go over pots via voipstreet are lousy and have lost syllables and jitter... |
| 03:31.37 | pigpen | grandy, it is more than likely a loss/latency upon load on your cable modem line, or a timing issue on your laptop.... |
| 03:31.37 | sevard | JT: wait, in a high priority? what |
| 03:31.46 | pigpen | grandy, ulaw or gsm? |
| 03:31.47 | JT | sevard: allow= |
| 03:31.57 | HushPe | is there i reason i can make outgoing calls on a FXO line, but the incoming one's aren't picked up? |
| 03:32.04 | grandy | pigpen: the connection to voipstreet is ulaw |
| 03:32.08 | JT | sevard: if both sides have other things with higher priorities, those codecs will be used instead |
| 03:32.12 | sevard | JT: I never ran into troubles as in where I placed the allow line |
| 03:32.23 | sevard | ah, yes |
| 03:32.25 | Corydon76-home | HushPe: are the FXO lines set up for a context which exists? |
| 03:32.33 | pigpen | grandy, try gsm...it may help... |
| 03:32.42 | grandy | pigpen: i did... it seemed worse actually |
| 03:32.58 | grandy | pigpen: as did g729 |
| 03:33.03 | pigpen | how does it do when you are at the datacenter? |
| 03:33.28 | JT | Daveb21: ? |
| 03:33.36 | HushPe | Corydon76-home: might have a valid point there, i'm still getting the hang of this... does each line need the context defined? or one for the lot? |
| 03:33.49 | grandy | pigpen: i haven't tried it from there... the datacenter is not local to me... but ping times are around 35-45ms from here... when i call in via the voipstreet pots line the jitter and lost packets are there on the moh |
| 03:33.49 | dseeb_ | can anyone tell me what this actually does ? |
| 03:33.50 | dseeb_ | ;exten => 6245,n(dial),Dial(${HINT},20,rtT) ; Use hint as listed |
| 03:34.03 | Corydon76-home | grandy: if available, you might try ilbc. It's designed to handle lost packets better |
| 03:34.16 | pigpen | also, if your laptop is wireless, you may be getting interference......try hardline. |
| 03:34.16 | Corydon76-home | HushPe: one for the lot |
| 03:34.27 | pigpen | ilbc is cool. |
| 03:34.35 | HushPe | ah ok, then yes... here is the warning i'm getting chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1 |
| 03:34.45 | grandy | Corydon76-home: it has improved the quality of calls from idefisk to asterisk, but voipstreet doesn't support it (unfortunately) |
| 03:34.51 | JT | ilbc sounds horrible though |
| 03:34.55 | HushPe | Corydon76-home: i've tried KS and LS to the same effect |
| 03:34.58 | Corydon76-home | HushPe: and since it's FXO, you need to have extension s in that context |
| 03:35.24 | HushPe | Corydon76-home: it's 'picking up' like playing my welcome message, but not actually picking up the line |
| 03:35.31 | JT | HushPe: fxs-ks? |
| 03:35.35 | Corydon76-home | JT: only if you have insufficient CPU. It's a hog. |
| 03:35.39 | grandy | so here's a question: anyone know a good iax or sip origination/termination company that has very low latency to the western US? The datacenter is in las vegas... |
| 03:35.51 | Corydon76-home | HushPe: then you need an Answer() in there. |
| 03:35.55 | JT | Corydon76-home: it doesn't sound "nice" though does it? |
| 03:35.58 | HushPe | JT: yes :) |
| 03:35.59 | [TK]D-Fender | dseeb_ : It dials whatever is in ${HINT} , which is either a constant defined in [globals] , a variable you should have set SOMEWHERE esle, or an outright conceptual flaw on your part |
| 03:36.07 | Corydon76-home | JT: yes, it sounds fine |
| 03:36.07 | sevard | <PROTECTED> |
| 03:36.21 | HushPe | i have answer too Corydon76-home |
| 03:36.30 | Corydon76-home | JT: iLBC is specially designed to lose up to 10% of packets without a corresponding loss of audio quality |
| 03:36.59 | JT | yeah i realise it was meant for bad connections |
| 03:37.03 | Corydon76-home | HushPe: I'd have to see your configs then |
| 03:37.21 | HushPe | zapata, extensions ? |
| 03:37.27 | Corydon76-home | HushPe: yes |
| 03:37.29 | Corydon76-home | ~pb |
| 03:37.30 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 03:37.33 | HushPe | any others? |
| 03:37.43 | Corydon76-home | That'll do it for now |
| 03:37.44 | HushPe | don't worry i know all about pastebin ;) |
| 03:38.11 | dseeb_ | [TK]D-Fender: i understand that bit that it dials ${HINT} its the lines bfore it i dont understand. |
| 03:38.38 | [TK]D-Fender | Corydon-w : ilbc loses 10% of your initial quality up-front as a down payment ;) |
| 03:38.44 | HushPe | Corydon76-home: extensions: http://pastebin.ca/326492 |
| 03:39.13 | HushPe | Corydon76-home: zapata: http://pastebin.ca/326493 |
| 03:39.15 | *** join/#asterisk infernix (n=nix@spirit.infernix.net) |
| 03:39.16 | [TK]D-Fender | dseeb_ : Well it'd be nice to SEE them now wouldn't it? pastebin the whole mess. www.pastebin.ca |
| 03:39.37 | HushPe | i've for the FXS cards working, a little crackly, but i think that's just a tweaking thing right? |
| 03:39.53 | dendrite | [TK]D-Fender: Wassamatta? You ain't psychic? |
| 03:40.23 | JT | HushPe: what's the average zttest score? |
| 03:40.40 | JT | crackling is indication of poor zttest scoor |
| 03:40.41 | Corydon76-home | HushPe: are you losing interrupts? |
| 03:40.41 | [TK]D-Fender | HushPe :Your dialplan is horribly insecure. |
| 03:41.13 | [TK]D-Fender | dendrite : load chan_psychic.so && chan_fluxcapacitor.so |
| 03:41.27 | HushPe | zztest = about 98-99 |
| 03:41.41 | Daveb21 | JT: Yeah sorry mate, got engrossed in reading some docco on voip-info.org |
| 03:41.44 | dendrite | [TK]D-Fender: ... Press 1 to hear what you're thinking... |
| 03:42.00 | HushPe | Corydon76-home: with an incoming line, it detects the ring, starts playing the message, but the phone isn't picking up |
| 03:42.33 | HushPe | 76 passed: Best: 99.291992 -- Worst: 76.464844 -- Average: 97.769165 |
| 03:42.34 | dseeb_ | [TK]D-Fender: its the standard extensions.conf.sample |
| 03:42.36 | Corydon76-home | HushPe: what do you mean, the phone isn't picking up? |
| 03:42.43 | JT | HushPe: fuck that's an awful score |
| 03:42.48 | HushPe | Corydon76-home: sorry i mean the line, i'll pase the debug |
| 03:42.51 | Corydon76-home | HushPe: holy |
| 03:42.55 | JT | HushPe: the score must be 99.97% or higher AT ALL TIMES |
| 03:43.03 | JT | or you will lose interrupts and it will sound shit |
| 03:43.08 | HushPe | is that my line thing or the computer thing? |
| 03:43.11 | Corydon76-home | HushPe: what else are you running in that machine? |
| 03:43.16 | HushPe | nothing |
| 03:43.19 | JT | check if it's interrupt sharing |
| 03:43.30 | [TK]D-Fender | dendrite : "If you have multiple personality disorder please press 4,5, and 6. If suffer from OCD, please press 2 repeatedly..." |
| 03:43.33 | Corydon76-home | Are you running an X server or text frame buffer? |
| 03:43.38 | HushPe | for sounding like a noob there abouts? |
| 03:43.48 | HushPe | Corydon76-home: none of the above |
| 03:43.51 | [TK]D-Fender | dseeb_ :scrap the "sample" and start from scratch |
| 03:44.14 | dendrite | [TK]D-Fender: <grin> |
| 03:44.23 | Corydon76-home | HushPe: so your screen is 25x80 and it has no faded penguin logo on the upper right |
| 03:44.47 | HushPe | nothing fancy, just plain old text console :) |
| 03:44.59 | dseeb_ | [TK]D-Fender: I'd love to, if i understood the concept |
| 03:45.05 | Corydon76-home | HushPe: then it's probably shared interrupts |
| 03:45.12 | JT | HushPe: cat /proc/interrupts |
| 03:45.42 | Corydon76-home | HushPe: grep wctdm /proc/interrupts |
| 03:45.51 | HushPe | http://pastebin.ca/326499 << debug from asterisk |
| 03:45.57 | Qwell | grep -ci mwar |
| 03:46.04 | Qwell | lspci | grep -ci mwar |
| 03:46.16 | HushPe | 22: 2053969 0 IO-APIC-fasteoi HDA Intel, wctdm << it's the sound card using the same interrupt i think |
| 03:46.34 | Corydon76-home | Ouch. Yeah, that'll do it |
| 03:46.40 | [TK]D-Fender | dseeb_ : .... |
| 03:46.41 | HushPe | where do i force it's own? |
| 03:46.42 | [TK]D-Fender | ~book |
| 03:46.44 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:46.51 | Corydon76-home | HushPe: your interrupts should all be less than 16 |
| 03:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:47.01 | Corydon76-home | 22 is a shared interrupt |
| 03:47.27 | [TK]D-Fender | HushPe : Time to start looking in your BIOS to see if you can dedicate it there, if no, try switching slots, and from there possibly disabling features you don't need. |
| 03:47.34 | HushPe | i've never had the need to change thing |
| 03:47.42 | HushPe | [TK]D-Fender: cheers will do :) |
| 03:48.06 | HushPe | would that be related to my incoming call problem too? or is that different again? |
| 03:48.19 | xpot | anyone know how to return to next priority after completing a GotoIf? |
| 03:48.32 | Corydon76-home | HushPe: highly likely, yes |
| 03:48.35 | [TK]D-Fender | HushPe : what happens on incoming? |
| 03:49.00 | HushPe | from what i see it's found that it's ringing, but doesn't actually pick up the line, it keeps ringing while i thinks it's picked up the line |
| 03:49.01 | jql | that's more of a gosubif |
| 03:49.06 | Corydon76-home | xpot: you want Gosub / Return |
| 03:49.08 | [TK]D-Fender | xpot : the point of doing Goto and GotoIf is to NOT continue where you are, but to go somewhere else. |
| 03:49.20 | HushPe | ring ring, pickup play message, ring ring, playing messsage, ring ring |
| 03:49.21 | xpot | rgr, thank you |
| 03:49.22 | HushPe | something like that |
| 03:49.45 | J4k3 | if I want to bring in g.729 calls, and have asterisk handle them, then feed out to g.729 based extensions, do I need to buy g.729 licenses for both sides of the connection? |
| 03:49.47 | JT | HushPe: pci slot swapping is usually easiest |
| 03:49.50 | [TK]D-Fender | HushPe : Do you SEE * trying to answer in the CLI? |
| 03:50.01 | Corydon76-home | [TK]D-Fender: yes, he does |
| 03:50.09 | HushPe | http://pastebin.ca/326499 << pasted ^^^ ;) |
| 03:50.10 | JT | J4k3: depends if asterisk just bridges or what |
| 03:50.10 | Corydon76-home | s |
| 03:50.11 | Corydon76-home | [21:45:51] <HushPe> http://pastebin.ca/326499 << debug from asterisk |
| 03:50.21 | [TK]D-Fender | J4k3 : No, but if at any time * has to play a sound (like voicemail, etc), then you will need one. |
| 03:51.03 | [TK]D-Fender | HushPe : Ok, I wouldn't NOT attibute your problem to being "chared" inteerupts... where are you located? |
| 03:51.05 | J4k3 | JT: I'd be bringing calls through an IVR then transfering them to various phones. most will be on ethernet and therefore work fine with g.711, but a couple will be portables and I think they support 729 |
| 03:51.23 | Corydon76-home | [TK]D-Fender: he's NZ |
| 03:51.24 | HushPe | [TK]D-Fender: i'm in nz |
| 03:51.26 | JT | J4k3: i think that would be straight bridging |
| 03:51.49 | JT | J4k3: straight bridging g.729 to g.729 = no licence needed |
| 03:51.52 | [TK]D-Fender | HushPe : Make sure your zaptel is set for your zone. I THINK "au" covers both areas.... |
| 03:51.53 | HushPe | we have bad telecom nz, and bad internet that stops voip LOL, but we just want a decent office pbx |
| 03:52.05 | JT | HushPe: internet that stops voip?? |
| 03:52.07 | HushPe | i'll paste that, there is one for nz :) |
| 03:52.15 | Corydon76-home | xpot: you have a knack for asking about things I wrote |
| 03:52.20 | HushPe | something they do with the packets, makes voip lag really bad |
| 03:52.52 | jql | deprioritizing UDP has that kind of effect |
| 03:53.02 | HushPe | that could be it |
| 03:53.04 | jql | just drop 50% of all udp packets, and most people won't even notice |
| 03:53.15 | jql | since most traffic is that vile tcp |
| 03:53.24 | Corydon76-home | jql: except for DNS queries |
| 03:53.25 | HushPe | zaptel for modprobe: http://pastebin.ca/326505 |
| 03:53.44 | jql | well, the ISP probably has internal DNS servers |
| 03:53.51 | jql | just have to... umm... whitelist them. :) |
| 03:54.34 | Corydon76-home | Remind me again why network neutrality is bad? |
| 03:55.07 | JT | if my isp fucked with udp in a bad way... bye bye isp |
| 03:55.07 | jql | I dunno. My cable/internet/phone company tells me it's evil |
| 03:55.15 | HushPe | at least it picks up the hangup on the other end, that's supposed to not happen very often here in nz (home biz) |
| 03:55.35 | HushPe | i'll reboot and see if i can't assign irqs :) |
| 03:55.56 | Corydon76-home | My favorite CS prof who has gotten a lot of grant money from Ma Bell thinks it's bad, too |
| 03:57.31 | *** join/#asterisk infernix (n=nix@spirit.infernix.net) |
| 04:01.43 | jlimb | Is there problems if two people page at once? |
| 04:02.14 | Corydon76-home | Well, kinda |
| 04:02.29 | Corydon76-home | The second person generally doesn't get through |
| 04:02.31 | jlimb | does app_page check for that at all? |
| 04:02.54 | Corydon76-home | Nope |
| 04:03.05 | jlimb | I will dialplan it then and see how it goes |
| 04:03.12 | jlimb | my users report ringing sometimes |
| 04:03.43 | jlimb | I think it is when a second person is paging but I havent tested yet |
| 04:08.41 | *** join/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 04:09.23 | docelmo | Does anyone know a way to make asterisk do a 302 reinvite |
| 04:11.22 | k-man__ | how do i configure asterisk to use a sip provider only? |
| 04:12.55 | [TK]D-Fender | k-man__ : Don't configure it to do anything else :) |
| 04:13.05 | k-man__ | ok |
| 04:13.31 | [TK]D-Fender | k-man__ : Its really easy to NOT do stuff..... I'll lend you a copy of my book on procrastination when I get around to finishing it.. |
| 04:14.21 | Shaun2222 | anybody know why when i upgraded to 1.4 that bridging of sip calls between phones fails.. |
| 04:14.23 | docelmo | [TK]D-Fender any ideas on making asterisk do a 302 redirect to another sip box? |
| 04:14.25 | Shaun2222 | [Jan 24 11:26:55] WARNING[18949]: res_features.c:1417 ast_bridge_call: Bridge failed on channels SIP/302-09c5ffb8 and SIP/301-09c61538 |
| 04:14.38 | docelmo | Shaun2222 check your codecs |
| 04:14.48 | [TK]D-Fender | docelmo : "show application transfer" ? |
| 04:14.51 | Shaun2222 | docelmo: i can use the phone to call real phones.. |
| 04:15.03 | k-man__ | [TK]D-Fender, yeah, ok |
| 04:15.07 | docelmo | thanks tk |
| 04:15.33 | [TK]D-Fender | docelmo : Any time... |
| 04:16.33 | [TK]D-Fender | Shaun2222 : That means nothing. turn up sip debug, and pastebin the ENTIRE failed cll from beginning to end. |
| 04:16.36 | [TK]D-Fender | ~pb |
| 04:16.47 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 04:28.30 | *** join/#asterisk Ritalin2 (n=dave@74-34-103-241.dsl1.pwll.tn.frontiernet.net) |
| 04:28.41 | Ritalin2 | anyone seen this when doing ztcfg? ZT_CHANCONFIG failed on channel 3: Inappropriate ioctl for device (25) |
| 04:32.20 | [TK]D-Fender | Ritalin2 : If you're going to ask why its complaingin about your config, perhaps you should consider SHOWING it to us instead of leaving us to guess.... |
| 04:32.21 | [TK]D-Fender | ~pb |
| 04:32.23 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 04:32.58 | docelmo | wow tk you're getting as bitchy as I do.. :) |
| 04:34.19 | Ritalin2 | the cfg is pretty simple. first line is fxsks=3-4 then loadzone=us and defaultzone=us |
| 04:34.42 | Ritalin2 | i dont know how it could be simplier |
| 04:34.48 | [TK]D-Fender | docelmo : Don't worry... I'm not after the crown :) |
| 04:34.59 | Ritalin2 | life's too short to have patience |
| 04:35.13 | grandy | Hello... is there any way to debug what aspects of tcp/ip are causing problems in an iax connection? in other words, can asterisk issue a warning when there is too much latency, etc...? |
| 04:35.24 | [TK]D-Fender | Ritalin2 : Are you really sure your modules are on the proper ports on the card? |
| 04:35.57 | Ritalin2 | fender: i'm doing this remotely for someone else but i tried changing it... to all possibilities |
| 04:36.05 | [TK]D-Fender | Ritalin2 : And we are indeed talking about 2 red modules on a TDM400P frame as well right? |
| 04:36.19 | Ritalin2 | i've never seen the card ;) |
| 04:36.39 | Ritalin2 | but dmesg looks right |
| 04:36.52 | [TK]D-Fender | Ritalin2 : "Double blind" is good for TASTE TESTS, not "hardware debugging" .... |
| 04:37.05 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com) |
| 04:37.05 | Ritalin2 | very true |
| 04:37.26 | Supaplex | hehe |
| 04:38.52 | Ritalin2 | fender: i'm pretty sure the kernel modules are jacked up |
| 04:40.34 | [TK]D-Fender | Ritalin2 : when in doubt, complete recompile... |
| 04:41.56 | *** join/#asterisk infernix (n=nix@spirit.infernix.net) |
| 04:42.25 | xpot | anyone know how to write to a db using func_odbc? I have: exten => s,n,Set(ODBC_CARDPIN(UPDATE demog SET pin=${NEWPIN} WHERE id=${IDNUM})) |
| 04:42.47 | xpot | does not work. anyone know what I am doing wrong? |
| 04:43.32 | [TK]D-Fender | xpot : what does "show function ODBC_CARDPIN" tell you? |
| 04:44.22 | xpot | ODBC_CARDPIN(<arg1>[...[,<argN>]]) |
| 04:44.32 | xpot | Read: |
| 04:44.32 | xpot | SELECT * FROM demog WHERE cardcode='${ARG1}' AND pin='${ARG2}' |
| 04:44.32 | xpot | Write: |
| 04:44.32 | xpot | UPDATE demog SET pin=${VAL1} WHERE id='${ARG1}' |
| 04:44.58 | Ritalin2 | fender: if this were my box i would do that. i think it's time for an email |
| 04:45.44 | [TK]D-Fender | xpot : Well that doesn't look like a standard app, so not sure what to say |
| 04:46.17 | grandy | hello, can anyone recommend a sip or iax origination company with good quality to the western US? |
| 04:46.19 | xpot | ok thanks for the attempt... anyone else have a suggestion? |
| 04:46.42 | xpot | grandy: try gafachi |
| 04:46.53 | grandy | xpot: saw their add... you usign them? |
| 04:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:47.08 | xpot | I have used them before |
| 04:47.14 | k-man__ | can i use the asterisk-sounds 1.2 package with 1.4? |
| 04:47.38 | grandy | xpot: cool i'll check 'em out |
| 04:47.50 | xpot | -=0) |
| 04:48.13 | grandy | oh here is one other question: Is there anything in asterisk 1.4 that would lead to better quality iax or sip callls? |
| 04:48.28 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-205-211.tx.res.rr.com) |
| 04:48.29 | xpot | g.729 I think |
| 04:48.32 | De_Mon | I have a dialplan that calls a numbers and puts the caller/callee in a conference. How do I determine when either person leaves the conference? |
| 04:48.39 | De_Mon | I tried adding a NoOp after the meetme command but it doesnt get executed, it says == Spawn extension (elephant-queues, goto-500, 3) exited non-zero on 'SIP/jon-08268540' |
| 04:48.57 | De_Mon | instead of going to priority 4 |
| 04:50.16 | Ritalin2 | thanks fender |
| 04:50.19 | *** part/#asterisk Ritalin2 (n=dave@74-34-103-241.dsl1.pwll.tn.frontiernet.net) |
| 04:52.46 | *** join/#asterisk ezw` (n=Ez@c66.203.210-59.clta.globetrotter.net) |
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| 05:13.26 | [TK]D-Fender | De_Mon : "h" |
| 05:15.46 | wunderkin | [TK]D-Fender: "i" |
| 05:15.47 | wunderkin | :D |
| 05:16.09 | [TK]D-Fender | wunderkin : u |
| 05:16.26 | wunderkin | 42! |
| 05:19.44 | neddy | ***[TK]D-Fender: dude, you just kicked off some major nostalgia |
| 05:22.08 | *** join/#asterisk infernix (n=nix@spirit.infernix.net) |
| 05:24.58 | De_Mon | [TK]D-Fender h seemed like overkill |
| 05:25.16 | De_Mon | I'd have to set a variable and check for it every time a channel hung up |
| 05:25.48 | De_Mon | or check the last extension for something, anyway itd happen for all channels that hung up |
| 05:29.26 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 05:43.03 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 05:46.17 | *** join/#asterisk MrTelephone (i=DeaLER25@bas13-toronto63-1096769727.dsl.bell.ca) |
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| 05:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 05:47.06 | MrTelephone | hey how do you get a list of the structures with suboptions and definitions? (for developers) |
| 05:47.32 | MrTelephone | for example ast_channel->tech_pvt |
| 05:47.43 | MrTelephone | what the heck is that? |
| 05:48.23 | Qwell | channel.h |
| 05:51.46 | MrTelephone | thanks qwell |
| 05:51.50 | MrTelephone | very nicely commented |
| 05:53.22 | docelmo | MrTelephone what is nicely commented |
| 05:53.30 | docelmo | dont you say asterisk.. |
| 05:54.25 | *** join/#asterisk asterisky (n=gimmesom@ip70-190-159-144.ph.ph.cox.net) |
| 05:55.03 | asterisky | Hi everyone, |
| 05:56.32 | MrTelephone | is asterisk called asterisk due to all the pointers in the source code? |
| 05:56.40 | asterisky | can some one help me with an iax trunk, im getting error: call rejected by: 200.134.2.23 No authority found |
| 05:57.37 | asterisky | and the same thing from the other end |
| 05:58.18 | asterisky | did an iax2 show peers and both show online, how ever making the call hell breaks loose |
| 05:59.27 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 06:00.19 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 06:01.02 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com) |
| 06:06.45 | *** part/#asterisk asterisky (n=gimmesom@ip70-190-159-144.ph.ph.cox.net) |
| 06:17.15 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 06:28.10 | docelmo | Anyone know where I can buy SMS numbers? |
| 06:41.19 | xpot | is there a way to bubble up a arg from Func_ODBC to see it on the screen while the dail plan is running |
| 06:41.33 | J4k3 | http://en.wikipedia.org/wiki/SMS_gateways#Carrier_Gateyways |
| 06:41.35 | J4k3 | that? |
| 06:41.48 | denon | lookie what the cat drug in |
| 06:42.20 | J4k3 | yay |
| 06:44.07 | xpot | ok |
| 06:44.12 | J4k3 | oh snap, the only incoming sms provider I've found yet |
| 06:44.24 | J4k3 | wants thirty five pounds sterling/month for unlimited in/out service |
| 06:44.45 | J4k3 | on *one* number |
| 06:44.56 | denon | pounds sterling .. |
| 06:45.03 | denon | sounds like you're buying a set of armor |
| 06:45.46 | denon | hehe |
| 06:45.53 | denon | a duel to the coredump! |
| 06:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 06:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 06:49.13 | docelmo | actually nevermind.. We will have SMS shortly.. :) Anyone interested in buying? |
| 06:49.56 | *** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) |
| 06:59.11 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkit.com.au) |
| 07:02.45 | *** join/#asterisk Emrah (n=Me@adslgva0491.worldcom.ch) |
| 07:02.51 | *** join/#asterisk azidenth (n=aby_azid@218.111.173.222) |
| 07:03.00 | azidenth | hello everyone.. |
| 07:03.10 | Emrah | Hello azidenth |
| 07:03.37 | azidenth | i need help |
| 07:04.37 | azidenth | using asterisk realtime sip and extension to make call (testing) |
| 07:05.12 | azidenth | trying to make sip call to another user---got error msg in asterisk CLI---no audio format to offer |
| 07:06.48 | Emrah | azidenth: what about the codecs you use? |
| 07:06.48 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:10.18 | azidenth | i load all from asterisk modules |
| 07:15.55 | *** join/#asterisk shinux__ (n=shinux@196.220.28.133) |
| 07:16.54 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 07:20.33 | docelmo | azidenth your codecs are incompatible.. |
| 07:26.25 | hads | Anyone know the deal with HPEC? |
| 07:28.39 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:29.16 | *** part/#asterisk MrTelephone (i=DeaLER25@bas13-toronto63-1096769727.dsl.bell.ca) |
| 07:30.28 | azidenth | docelmo |
| 07:30.33 | azidenth | :how to make it compatible? |
| 07:30.59 | hads | mitcheloc: Ping |
| 07:31.33 | azidenth | i used this codec g729;ilbc;gsm;ulaw;alaw |
| 07:35.37 | *** part/#asterisk Trevor_B (n=tbenson@64-142-72-32.dsl.static.sonic.net) |
| 07:36.40 | azidenth | asterisk CLI --> [Jan 24 15:44:16] WARNING[13837]: channel.c:3033 ast_channel_make_compatible: No path to translate from SIP/Aby-084ddf90(256) to SIP/abu-0849d2f8(1024) |
| 07:36.41 | azidenth | <PROTECTED> |
| 07:36.41 | azidenth | [Jan 24 15:44:20] WARNING[13810]: channel.c:2702 set_format: Unable to find a codec translation path from gsm to g729 |
| 07:36.42 | azidenth | <PROTECTED> |
| 07:36.44 | azidenth | <PROTECTED> |
| 07:36.53 | azidenth | anyone can help? |
| 07:37.45 | Emrah | azidenth: g729 isn't a free codec |
| 07:38.02 | Emrah | You should enable only g711 and ilbc... |
| 07:38.08 | Emrah | disallow=all |
| 07:38.10 | Emrah | allow=alaw |
| 07:38.12 | Emrah | allow=ulaw |
| 07:38.16 | Emrah | allow=ilbc |
| 07:38.19 | Emrah | allow=gsm |
| 07:38.34 | J4k3 | ;) |
| 07:38.43 | Emrah | and stop pasting things there :) |
| 07:38.45 | Emrah | www.pastebin.ca |
| 07:38.52 | azidenth | ok.. |
| 07:39.34 | *** join/#asterisk stephane_ (n=stephane@zafnib.cabale.net) |
| 07:40.09 | docelmo | Well this should be interesting.. I am about to launch a free calling service in the next few hours.. |
| 07:40.33 | jql | do I get to call belgium? |
| 07:40.48 | docelmo | Sorry US domestic calling only |
| 07:40.56 | jql | oh, darn |
| 07:41.03 | docelmo | international eventually depends.. |
| 07:41.21 | docelmo | its a service where you listen to a short advertisement then you make your calls |
| 07:45.05 | gfraysse | <PROTECTED> |
| 07:45.16 | *** join/#asterisk reber (i=reber@gateway/tor/x-1ef21d4b819a19a3) |
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| 07:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 07:47.01 | docelmo | What happened to absolutetimeout? |
| 07:47.06 | docelmo | was it replaced? |
| 07:47.26 | jql | TIMEOUT(xxx) ? |
| 07:48.25 | docelmo | nope in 1.4 its in the dial command now |
| 07:48.41 | jql | my version still has TIMEOUT(absolute) |
| 07:49.32 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 07:50.08 | docelmo | If your using 1.4 I dont know how.. I am looking at it now and it shows the L option in the dial command |
| 07:58.20 | oej | I think those are different timers |
| 07:58.34 | shimi | I recorded a call not a while ago. While listening to the recording - I heard only the remote side - not myself - any idea? |
| 07:58.47 | oej | L plays a lot of prompts, timeout(absolute) just kills the call |
| 08:07.41 | zoa | yo olle |
| 08:07.43 | zoa | damn |
| 08:07.44 | zoa | too late |
| 08:07.58 | zoa | docelmo |
| 08:07.59 | zoa | cool |
| 08:09.53 | Emrah | shimi: You should have two files. The input and the output, just mixe them |
| 08:10.24 | shimi | I think I have one, in the monitor directory. could be influence of trixbox? |
| 08:10.40 | *** join/#asterisk Mportnoy (n=test@200.122.158.88) |
| 08:10.41 | Emrah | no idea |
| 08:10.53 | Emrah | I don't use this soft |
| 08:11.37 | *** join/#asterisk poller (n=poller@poller.se) |
| 08:11.38 | Emrah | Does anyone know where I may find a doc to implement a missed call notification via sms? |
| 08:12.05 | Emrah | Currently I've implemented a first verison but I'm having a problem with the voicemail |
| 08:12.13 | Emrah | pfirst version* |
| 08:17.19 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 08:20.31 | zoa | olle!!! |
| 08:20.33 | Mportnoy | ANyone has a guide to enable CDR with mysql ? on Debian 3.1 ? |
| 08:20.53 | oej | zoa!!! |
| 08:21.42 | dezent | Mportnoy: maybe this will help http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql |
| 08:21.51 | Mportnoy | does anybody have a guide to enable CDR with MYSQL on DEbian 3.1 ? |
| 08:22.03 | dezent | ... |
| 08:26.54 | *** join/#asterisk s1gny|wrk (n=s1gny@p54914E79.dip.t-dialin.net) |
| 08:27.04 | *** part/#asterisk s1gny|wrk (n=s1gny@p54914E79.dip.t-dialin.net) |
| 08:27.56 | Mportnoy | more specifiq guide? |
| 08:29.47 | *** join/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net) |
| 08:29.55 | Emrah | dezent: I know something pretty good to find that kind of info |
| 08:30.02 | Emrah | It's called Google. |
| 08:30.10 | Emrah | And it's available on www.google.com |
| 08:30.28 | Emrah | Just have a look and tell me what you think |
| 08:30.35 | Emrah | :P |
| 08:30.44 | Emrah | dezent: |
| 08:30.45 | Emrah | hey sorry |
| 08:30.53 | Emrah | that was a message for Mportnoy |
| 08:30.59 | Emrah | excuse-me |
| 08:31.17 | Emrah | Your answer was quite helpful and efficient |
| 08:31.49 | Emrah | http://www.google.ch/search?hl=fr&q=asterisk+cdr+mysql+tutorial&meta= |
| 08:32.13 | Emrah | ow to make the asterisk to write the cdr in the table in the mysql. ... New tutorial on how to automatically test your extensions section: Asterisk ... |
| 08:32.34 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 08:32.38 | Mportnoy | thanks |
| 08:33.04 | Emrah | Mportnoy: http://forums.whirlpool.net.au/forum-replies-archive.cfm/601673.html |
| 08:33.05 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 08:33.07 | Emrah | that might also help |
| 08:36.09 | dezent | Emrah: ? |
| 08:36.14 | dezent | ah |
| 08:37.14 | Emrah | nJlol sorry dezent |
| 08:37.19 | dezent | jao |
| 08:39.25 | docelmo | sup Z |
| 08:40.27 | *** join/#asterisk profounded (n=pro@ool-44c4e6c0.dyn.optonline.net) |
| 08:40.55 | profounded | question, how do i convert a wav to gsm? |
| 08:41.29 | docelmo | sox |
| 08:41.32 | Emrah | profounded: install sox |
| 08:41.40 | docelmo | DOES NO ONE FUCKING USE THE WIKI ANYMORE?!?!?!? |
| 08:41.41 | Emrah | then sox file.wav file.gsm |
| 08:41.44 | profounded | does sox handle wav or is it a convertor |
| 08:41.53 | profounded | ok thank you emrah |
| 08:42.04 | Emrah | docelmo: that's a good question |
| 08:42.45 | JT | frustrated much, docelmo? ;) |
| 08:44.06 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
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| 08:45.09 | docelmo | JT of dumb newbie questions YES |
| 08:46.52 | Strom_C | hi |
| 08:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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| 08:55.35 | docelmo | sup sup |
| 09:03.54 | docelmo | anyone know the command to swap eth0 and eth1? |
| 09:06.03 | dongc | maybe u can check out /etc/sysconfig/network-scripts/ifcfg-ethx. Not sure will this help or not. |
| 09:06.28 | *** join/#asterisk Arnar (n=arnarb@landi.oddi.is) |
| 09:06.30 | docelmo | not really.. there is a command that can be used to swap them |
| 09:06.33 | docelmo | I just dont remember it. |
| 09:06.50 | docelmo | my servers are setup for eth1 and I need them setup for eth0 |
| 09:06.58 | docelmo | thanks tho |
| 09:07.07 | *** part/#asterisk Arnar (n=arnarb@landi.oddi.is) |
| 09:08.55 | hads | nameif |
| 09:11.55 | docelmo | thanks hads |
| 09:13.40 | hads | np |
| 09:15.16 | Rhizome | Anyone know if 1.2 AEL supports arrays? |
| 09:18.00 | profounded | im editing asterisk's dialplan and never worked with asterisk before: im editing /etc/asterisk/extensions.conf and i want to put a pause before: exten => s,4,Background(custom/welcome) |
| 09:18.11 | profounded | what is the command for a pause? |
| 09:18.24 | hads | Wait |
| 09:18.42 | profounded | got.. just found ty |
| 09:18.46 | profounded | got it* |
| 09:19.38 | *** join/#asterisk reber (i=reber@gateway/tor/x-1b3ecae8163f29a7) |
| 09:20.28 | profounded | im guessing wait is in seconds? |
| 09:21.34 | profounded | yep.. ty |
| 09:27.01 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 09:35.27 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 09:36.18 | docelmo | any debian guys in here? |
| 09:38.14 | hads | I use Debian... |
| 09:39.21 | docelmo | cool.. where does the config files for eth0 and eth1 reside? |
| 09:39.36 | docelmo | I changed the interfaces and now have no network.. |
| 09:40.54 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 09:41.20 | hads | /etc/network/interfaces |
| 09:41.23 | profounded | "SIP Supports text messaging during a call, but not outside of a call. " Im a bit confused with that statement, does it mean I can get text messages or is text something different, I want texts to go to my extension |
| 09:41.35 | docelmo | thanks.. Im a redhat guy.. Trying to learn this one also |
| 09:41.44 | hads | /etc/network/interfacesNo worries |
| 09:41.52 | hads | erm, no worries :) |
| 09:45.49 | profounded | when i dial an outgoing number im forced to add a 1 to the outgoing number if it doesnt exist (and is this a simple matter of programming or is their an option) |
| 09:45.53 | profounded | there* |
| 09:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 09:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 09:51.49 | Shaun2222 | whats the deal with this?... [Jan 24 17:04:39] WARNING[23297]: translate.c:600 __ast_register_translator: plc_samples 160 format 6 |
| 09:52.10 | *** join/#asterisk jm|laptop (n=jamie@dilbert.jamiem.com) |
| 09:52.10 | Shaun2222 | whats the deal with this?... [Jan 24 17:04:39] WARNING[23297]: translate.c:600 __ast_register_translator: plc_samples 160 format 6 |
| 09:52.16 | Shaun2222 | whoop |
| 09:52.17 | Shaun2222 | sorry |
| 09:56.30 | phearless | hello guys! |
| 09:56.55 | phearless | I am a bit confused to origanise the phone extensions |
| 09:57.08 | phearless | we are 3 sales and 10 IT people |
| 09:57.33 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 09:57.34 | phearless | should I use for ex : 4XX for IT and 5XX for IT ? |
| 09:58.10 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 09:59.39 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 10:01.49 | phearless | ? |
| 10:02.00 | *** join/#asterisk susinths (n=susinths@loven.ifi.uio.no) |
| 10:03.22 | *** join/#asterisk Dibbler_XP (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 10:10.22 | docelmo | in all honesty dude its up to you |
| 10:10.25 | docelmo | Your design |
| 10:11.18 | docelmo | Personally I segment 100 blocks for segmented locations.. For instance.. my delaware office is 1XX and our Orlando office is 2XX and mobile people are 3XX |
| 10:12.07 | *** join/#asterisk shinux_ (n=shinux@196.220.28.133) |
| 10:12.35 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 10:16.08 | J4k3 | if your company is large enough you may wish to designate your own internal area code ;) |
| 10:17.32 | Shaun2222 | anybody here seen a problem where when upgrading asterisk from 1.2 to 1.4 that when i dial another phone adn they connect that they cant hear eachother? |
| 10:17.43 | Shaun2222 | i can call out and into the phone from outside of asterisk and they work fine.. |
| 10:17.49 | Shaun2222 | it's just phone to phone |
| 10:20.12 | docelmo | Shaun2222 every heard of sip set debug ???? |
| 10:20.18 | Shaun2222 | ya |
| 10:20.29 | docelmo | ever heard of this one? |
| 10:20.30 | docelmo | ~pb |
| 10:20.38 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 10:22.49 | Shaun2222 | http://channels.debian.net/paste/5164 |
| 10:23.19 | phearless | ok docelmo |
| 10:23.24 | Shaun2222 | hmm |
| 10:23.29 | Shaun2222 | ok maybe this is the problem.. |
| 10:23.35 | phearless | J4k3: ?! we are 13, as I wrote |
| 10:23.43 | Shaun2222 | sip show peers says Nat N for those 2 phones.. |
| 10:24.10 | *** join/#asterisk jm|laptop (n=jamie@dilbert.jamiem.com) |
| 10:24.54 | docelmo | That could be a posibility |
| 10:25.03 | *** join/#asterisk telenieko (n=marc@167.Red-80-35-144.staticIP.rima-tde.net) |
| 10:25.07 | docelmo | in all honesty its simpler to look at the debug |
| 10:25.34 | telenieko | Hi, with func_odbc, is it possible to return more than one column and get each one assigned to a different variable? or inside an array. asterisk 1.2 ;) |
| 10:28.05 | Shaun2222 | docelmo: i pasted it.. |
| 10:28.25 | docelmo | Hold on |
| 10:29.18 | Chris-NB | hi |
| 10:29.40 | docelmo | Peer audio RTP is at port 192.168.1.104:2252 |
| 10:29.43 | Chris-NB | I've a 8port BRI card. 6 of these are in use (TE) and all connected to the pstn |
| 10:29.46 | docelmo | yes you have a NAT problem |
| 10:30.19 | Shaun2222 | with 1.2 these showed up with Nat = Y |
| 10:30.25 | Shaun2222 | with 1.4 now they are N |
| 10:30.46 | Chris-NB | how do I have to configure the 6 spans? |
| 10:30.53 | docelmo | You need to change the NAT setting.. I dont know if its changed.. I would check the sip.conf sample for the configuration of NAT |
| 10:31.04 | yacc | Hmm, how can one setup a 2.4 kernel to NAT SIP correctly? I've got a problem where when I want to talk with twinkle to sipgate.de, I hear the other side, but they don't hear me. And twinkle complains about lost RTP tx packets. |
| 10:31.06 | Shaun2222 | looks to be the same... |
| 10:31.07 | docelmo | Chris-NB to call the pstn |
| 10:31.15 | Shaun2222 | core it looks to be maybe ignoring it too |
| 10:31.17 | *** join/#asterisk jm|laptop (n=jamie@dilbert.jamiem.com) |
| 10:31.43 | Chris-NB | span=1,1,0,ccs,hdb3, span=2,1,0,ccs,hdb3 |
| 10:31.48 | Chris-NB | and so on |
| 10:32.04 | docelmo | Chris who made the card? Its not a digium board is it? |
| 10:32.09 | Shaun2222 | errr... did polycom change the reboot key's on the latest firmware for the 601's? |
| 10:32.14 | docelmo | Shaun2222 check into STUN |
| 10:32.32 | Chris-NB | or do I have to increase the 2nd parameter simiular to the 1st parameter? |
| 10:32.33 | docelmo | Shaun2222 thats the best I can offer you |
| 10:32.40 | Chris-NB | docelmo, it's a Beronet BN8S0 |
| 10:32.46 | JT | having span=1,1 and span=2,1 doesn't make sense |
| 10:33.05 | docelmo | yacc honestly dude.. Compile 2.6 make your life simpler |
| 10:33.07 | JT | the second parameter is timing source priority |
| 10:33.20 | JT | althought i don't know if it makes a difference on bri cards |
| 10:33.40 | docelmo | BRI's dont have timeing I dont think |
| 10:33.49 | docelmo | actually come to think of it I think they do |
| 10:33.51 | Chris-NB | JT, jep. and all ports should have the same clock (its pstn) |
| 10:34.03 | JT | probably internal to the kernel driver actually |
| 10:34.14 | JT | Chris-NB: no, it's priority, they should never have the same clock |
| 10:34.14 | yacc | docelmo: not an option. The box involved currently has no video/keyboard, that makes me really unhappy about any reboot, not to talk about a new kernel. |
| 10:34.28 | JT | Chris-NB: what kernel driver are you using? |
| 10:34.38 | Chris-NB | JT, qozap |
| 10:34.49 | JT | bristuff? |
| 10:34.56 | JT | did you modify qozap source? |
| 10:35.05 | Chris-NB | JT, they should have the same clock as it is pstn, the have to be syncron |
| 10:35.08 | Chris-NB | JT, jep |
| 10:35.11 | Shaun2222 | yacc: not video or keyboard??? what are you using? |
| 10:35.50 | JT | Chris-NB: you really don't understand how the parameter works, it's asking for a clocksource, once it successfully finds a clocksource, all zaptel stuff uses the same clock sync |
| 10:36.04 | yacc | Shaun2222: a standard PC, but because of moves around here, I've ended in a situation where I've got no display that I could connect. :( |
| 10:36.11 | JT | set one to priority one, one to priority 2, and so on, or set the rest to 0 |
| 10:36.13 | tzafrir | JT, actually not all of zaptel |
| 10:36.17 | yacc | Hmm, is there a way to convert an analog VGA port to HDMI? |
| 10:36.27 | tzafrir | JT, each device may use its own clock |
| 10:36.29 | Chris-NB | JT, ok. so I just define span1 as first clock .. ok! |
| 10:36.31 | JT | tzafrir: it probably does nothing for bri anyway |
| 10:36.40 | tzafrir | But conferences and such use the main clock |
| 10:36.45 | JT | Chris-NB: it doesn't matter which ones, as long as it's usually connected |
| 10:36.46 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
| 10:36.55 | tzafrir | All spans from the same device share the same clock |
| 10:37.05 | Chris-NB | ok. thanks |
| 10:37.07 | JT | well it is one device :) |
| 10:37.11 | *** join/#asterisk BugKhaM (n=LAMER@ppp-58.8.6.239.revip2.asianet.co.th) |
| 10:37.32 | JT | Chris-NB: does qozap load successfully? |
| 10:37.33 | tzafrir | Unless I misread Zaptel's source or miss a point there |
| 10:37.36 | BugKhaM | anyone knows how to define a key for a user to exit meetme? |
| 10:37.50 | Chris-NB | JT, jep. |
| 10:38.31 | JT | Chris-NB: can you paste zapata.conf and zaptel.conf and the bit of dmesg pertaining to loading qozap into pastebin.ca? |
| 10:39.09 | Chris-NB | JT, ähm, mom. |
| 10:39.39 | JT | i'm going to go out on a limb that you've set all the jumpers on the card correctly |
| 10:39.49 | JT | and have made the necessary cables/patch panels |
| 10:40.02 | Chris-NB | jep |
| 10:40.47 | BugKhaM | or simply how can a user exit meetme with "X" mode specified? |
| 10:41.38 | tzafrir | Chris-NB, what do you have connected to the card? any port connected to a telco? |
| 10:41.38 | phearless | if one person in the office is on the phone, how can somebody else "enter in the conversation", from another phone of the office ? |
| 10:42.24 | Chris-NB | tzafrir, first 6 ports |
| 10:42.41 | Chris-NB | tzafrir, 5 work fine, but I've problems with the 6th one |
| 10:42.49 | JT | 1-6 or first 3 physical ports? |
| 10:42.50 | tzafrir | make the first with priority 1, the second with priority 2, etc. |
| 10:42.55 | Chris-NB | that's why I asked for the priority |
| 10:43.04 | Chris-NB | JT, 1-6 |
| 10:43.09 | tzafrir | You should use the telco's timing |
| 10:43.22 | Chris-NB | JT, 1-6 physical ports |
| 10:43.22 | JT | Chris-NB: pastebin? |
| 10:43.27 | Chris-NB | JT, in progress |
| 10:43.29 | tzafrir | Do you have any devices connected to other ports? |
| 10:43.38 | JT | Chris-NB: isn't there only 4 physical ports? |
| 10:44.14 | Chris-NB | JT, ur right. hard to explain : ) port 1 - 6 or chan 1-2,4-5,7-8,10-11,13-14 |
| 10:44.31 | JT | yes i have one from junghanns, i know how the card works |
| 10:44.59 | *** join/#asterisk RoyK (n=roy@213.160.242.49) |
| 10:45.23 | JT | spans 4+5,3+6,2+7,1+8 |
| 10:45.23 | Chris-NB | ok |
| 10:45.32 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 10:45.36 | Chris-NB | jep. ... strange port numbering |
| 10:45.40 | Dovid | morning all |
| 10:45.54 | Dovid | i am trying to install asterisk addons 1.2.5 and i am getting an erorr |
| 10:46.00 | Dovid | http://pastebin.ca/326747 |
| 10:46.09 | JT | it makes sense, goes up then down |
| 10:46.35 | phearless | hello folks |
| 10:46.38 | phearless | :) |
| 10:46.43 | phearless | if one person in the office is on the phone, how can somebody else "enter in the conversation", from another phone of the office ? |
| 10:46.51 | Chris-NB | JT, http://pastebin.ca/326748 |
| 10:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:47.53 | Dovid | phearless: u can use chanspy |
| 10:48.04 | Dovid | although I dont think u can talk - u can just listen in |
| 10:48.12 | Dovid | unless u have all calls go thru a confrence |
| 10:48.18 | BugKhaM | From the wiki, 'X' — allow user to exit the conference by entering a valid single digit extension |
| 10:48.19 | JT | err MeetMe handles multiple users just fine |
| 10:48.41 | BugKhaM | what is this single digit extension? anyone? |
| 10:48.42 | JT | entering a meetme is easy, same as with the original extension in it |
| 10:48.45 | phearless | Dovid: yes I need t otalk |
| 10:48.49 | phearless | to talk* |
| 10:49.06 | phearless | I need to enter in any conversation |
| 10:49.17 | Dovid | phearless: do u want the option on the fly or for specific calls |
| 10:49.18 | Dovid | ? |
| 10:49.26 | phearless | err... |
| 10:49.33 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 10:49.36 | JT | Chris-NB: are the lines from the telco PTP or PTMP? |
| 10:49.36 | BugKhaM | JT: how to leave meetme then? |
| 10:49.43 | phearless | Dovid: not sure what do you mean |
| 10:49.46 | Chris-NB | JT, ptp |
| 10:49.56 | JT | Chris-NB: have you confirmed it with them? |
| 10:50.01 | Chris-NB | JT, jep |
| 10:50.19 | Dovid | phearless: how often do u think u will want to join the convo ? |
| 10:50.21 | phearless | Dovid: I just need to enter in the conversation if I want to |
| 10:50.25 | JT | you should probably also specify a default tonezone in zaptel.conf |
| 10:50.31 | phearless | Dovid: not for each calls |
| 10:50.36 | phearless | Dovid: just sometimes |
| 10:50.41 | Chris-NB | JT, as I said, span 1 to 5 come up and It's possible to place calls from asterisk or from outside |
| 10:50.57 | JT | what does span 6 do? |
| 10:50.58 | Chris-NB | JT, only span 6 comes up and down (cyclic) |
| 10:51.05 | JT | does asterisk spew errors? |
| 10:51.36 | Dovid | phearless: ok. so u can do 2 things |
| 10:51.55 | Dovid | 1) if u want to join a call have the current call confence u in |
| 10:51.59 | Chris-NB | span 6 is up (abaout 3 - 5 times) then span 6 is down and the thing that no available d channel 18 using it any way |
| 10:52.04 | Dovid | or have every call go thru a confrence room |
| 10:52.11 | JT | sounds about right |
| 10:52.11 | Chris-NB | and that repeats |
| 10:52.18 | JT | what version are you running |
| 10:52.19 | Chris-NB | no errs |
| 10:52.34 | Chris-NB | 1.2.13-BRIstuffed-0.3.0-PRE-1v |
| 10:52.44 | JT | use w or x |
| 10:52.48 | JT | i havent tried x yet |
| 10:52.53 | phearless | <Dovid> 1) if u want to join a call have the current call confence u in <---- how could I do this? |
| 10:52.59 | JT | w made significant changes to the qozap driver code |
| 10:53.01 | Dovid | on the phone |
| 10:53.05 | Dovid | what phones r u using ? |
| 10:53.14 | Dovid | anyone know the svn url for asterisk-addons ? |
| 10:53.18 | JT | which in my case, changed it from being completely unreliable and unusable, to quite stable |
| 10:53.28 | phearless | Dovid: I use Linksys/sipura SPA942 |
| 10:53.29 | yacc | so anyone got an idea how to setup NAT with basic iptables rules for SIP? |
| 10:53.51 | Dovid | never used it so i cant say for sure |
| 10:53.56 | JT | although i also use NT mode, that's generally poorly supported everywhere |
| 10:53.59 | Dovid | but what u would do is u would confrence in the new user |
| 10:54.21 | Dovid | something to the affect of looking for a confrence button, pressing it and entering the exten of the phone that u want to confrence with |
| 10:54.44 | phearless | okay Dovid |
| 10:54.44 | JT | Chris-NB: also, is that the version of your source? asking because it appears that release "w" forgot to change the version displayed in asterisk from "v" |
| 10:54.44 | phearless | I will have a lokk at this, thanks |
| 10:54.50 | Dovid | i never used the phone so i cant say for sure - poke around on it |
| 10:54.58 | *** join/#asterisk webmad (n=webmad@bkon.it) |
| 10:55.14 | Chris-NB | JT, mom. |
| 10:55.20 | webmad | g'day to al |
| 10:57.35 | Dovid | an anyone help me with an issue installing asterisk-addons1.2.5 ? |
| 10:58.16 | JT | CrossRoad: brb |
| 10:58.21 | JT | Chris-NB: brb, even |
| 10:58.27 | Chris-NB | ok |
| 11:00.09 | *** join/#asterisk fadey (n=fadey@239.Red-80-36-91.staticIP.rima-tde.net) |
| 11:05.22 | JT | back |
| 11:07.53 | JT | Chris-NB: any luck? |
| 11:08.06 | webmad | I have a question about Dialogic hardware, does anyone use it? |
| 11:08.48 | JT | not really |
| 11:09.01 | *** join/#asterisk dj015 (n=damjan@dsl-241-241-103.telkomadsl.co.za) |
| 11:09.21 | dj015 | hi, i'm getting the error misdn/isdn_lib_intern.h:14:83: missing binary operator before token "(" |
| 11:09.45 | Dovid | anyone out there for asterisk-addons ??? |
| 11:11.15 | Dovid | dj015: can u pb a lil more ? |
| 11:12.49 | *** join/#asterisk dimmik (i=dimmik@ios4.intranet.GR) |
| 11:13.04 | dj015 | http://pastebin.ca/326760 |
| 11:13.24 | *** part/#asterisk dimmik (i=dimmik@ios4.intranet.GR) |
| 11:21.09 | Dovid | anyone know how to apply a patch ? |
| 11:22.01 | dj015 | patch -p1 < file.patch |
| 11:27.38 | Dovid | its asking me for the file to patch and i dont know which one i am supposed to be patching |
| 11:27.48 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:28.29 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 11:28.33 | dj015 | Dovid, you're in the wrong directory, try going into the directory you're patching |
| 11:28.43 | Murdock__ | When dialing the pickupexten, is it possible to get the callerid of the call you're picking up onto the phone you picked it up from? |
| 11:28.45 | dj015 | and/or try -p0 |
| 11:29.18 | dj015 | why is my misdn failing to compile (http://pastebin.ca/326760) ? |
| 11:29.46 | Dovid | can't find file to patch at input line 5 |
| 11:29.46 | Dovid | Perhaps you used the wrong -p or --strip option? |
| 11:30.30 | dj015 | try -p0 instead of -p1 |
| 11:30.41 | Dovid | p0 worked |
| 11:30.50 | Dovid | but i still cant isntall the addons |
| 11:31.16 | Dovid | dj014: any expirience with addons ? |
| 11:31.50 | dj015 | they generally work out of the box :-) |
| 11:31.53 | dj015 | what's wrong? |
| 11:32.10 | Dovid | i keep getting an erorr |
| 11:32.14 | Dovid | let me pb it |
| 11:33.22 | Dovid | here ya go |
| 11:33.23 | Dovid | http://pastebin.ca/326747 |
| 11:33.59 | dj015 | did you patch your addons? |
| 11:34.26 | dj015 | because it looks like you patched it with the patch written for another version |
| 11:34.44 | dj015 | or you're using addons for a different asterisk version |
| 11:34.59 | Dovid | dj015 i got the same error b4 and after the patch |
| 11:35.07 | Dovid | i am using 1.2.5 for asterisk.1.2.14 |
| 11:35.31 | dj015 | did you install asterisk first? |
| 11:36.00 | Dovid | of course |
| 11:36.14 | Dovid | i am doing an upgrade |
| 11:36.18 | dj015 | try a few earlier/later versions of addons |
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| 11:45.06 | Dovid | dj015: dont know what i did but now it seems to be workign |
| 11:45.09 | Dovid | working* |
| 11:46.43 | Dovid | have a good one |
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| 11:48.44 | JT | Chris-NB: ? |
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| 12:24.39 | santibiotico | hi |
| 12:25.02 | santibiotico | i'm in trouble trying to send/receive faxes over VoIP, using sip channels |
| 12:25.17 | santibiotico | could anyone help me? |
| 12:26.16 | coppice | not really. you will never get it to work reliably |
| 12:26.16 | zoa | hmm its normal to give problems |
| 12:26.16 | zoa | you shouldnt fly using a car either |
| 12:27.05 | coppice | take no notice of zoa. he said there would never be a GPL'd T.38 :-) |
| 12:28.38 | *** join/#asterisk jontow (i=jontow@hijacked.us) |
| 12:29.50 | zoa | i didnt say that did i ? |
| 12:30.02 | zoa | btw, our commercial version is ready |
| 12:30.28 | coppice | what does it do? |
| 12:31.00 | dj015 | zoa, commercial version of what? |
| 12:31.35 | zoa | t.38 termination / gatewaying |
| 12:31.40 | zoa | + t.30 |
| 12:31.51 | coppice | what do you use for modems? |
| 12:31.52 | dj015 | he he, openpbx does that for free |
| 12:32.04 | zoa | our own modems |
| 12:32.26 | zoa | it does not depend on spandsp |
| 12:32.42 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 12:33.00 | zoa | i demoed it on astricon, but we didnt have gatewaying back then |
| 12:33.41 | coppice | has it been tested against many other implementations? |
| 12:33.56 | zoa | linksys, grandstream, allied telesys, mediatrix |
| 12:34.26 | coppice | mediatrix really sucks. I had to put in workarounds to tolerate their crap |
| 12:34.42 | zoa | we didnt have too much problem with the mediatrix, but had with the older grandstreams |
| 12:34.59 | santibiotico | but is there no way to work with t38 under asterisk? |
| 12:35.07 | coppice | grandstreams do things other people don't, but its all to spec |
| 12:35.21 | dj015 | santibiotico, t38 does work under openpbx |
| 12:35.24 | zoa | the older grandstreams seem to choke on big packets |
| 12:35.32 | coppice | mediatrix do crap like ending non-ecm data with and end of HDLC indication |
| 12:35.37 | zoa | not enough memory or so |
| 12:35.51 | coppice | the SDP should take care of that |
| 12:36.47 | coppice | i haven't tried allied-telesis yet. I've tried a few other now, though. |
| 12:37.01 | santibiotico | dj015 but any idea in how to integrate t38 under asterisk, not openpbx? |
| 12:37.05 | zoa | dunno about that, i know only a little of the t.38 stuff (i obviously didnt program it myself) |
| 12:37.10 | coppice | There are some things you actually can't get right. |
| 12:37.58 | coppice | there are too many commerical T.38 packages. I can't believe there is any real business in it |
| 12:38.22 | zoa | yeah it might be problematic |
| 12:38.34 | zoa | but we needed it before you had it so started it |
| 12:38.50 | zoa | then we lost 6 months with a fraud that pretended to write code but didnt |
| 12:38.53 | coppice | I've had mine for at least 18 months |
| 12:38.58 | *** join/#asterisk n0rus (n=linuxoid@210.85-200-239.bkkb.no) |
| 12:39.19 | zoa | so in the end we just nearly beated you to it :) |
| 12:39.26 | n0rus | Does X100P(clone) support the called ID function? |
| 12:39.30 | zoa | but we wouldnt have do it again |
| 12:39.30 | n0rus | caller |
| 12:39.41 | coppice | Only last summer did it get integrated with something and really exercised |
| 12:39.42 | zoa | i dont get how you could do that for free and release it as GPL |
| 12:39.43 | *** part/#asterisk Ebola (n=Ebola@host81-151-92-254.range81-151.btcentralplus.com) |
| 12:39.45 | zoa | its like a lot of work |
| 12:40.23 | coppice | well, if you've paid someone commerical rates and got a proper implementation it should have cost you >$100k |
| 12:40.37 | *** join/#asterisk sergee (i=opera@195.94.224.197) |
| 12:41.02 | zoa | we built it in house (All of it) |
| 12:41.33 | zoa | but needles to say that the bounty is not enough to actually open source it like that :) |
| 12:41.48 | *** join/#asterisk clorabit (n=eddysety@it.petra.ac.id) |
| 12:41.51 | coppice | I assume you only do V.29 and V.17 |
| 12:41.58 | clorabit | helloo |
| 12:42.06 | coppice | very few things do V.34, and you can't open source that |
| 12:42.17 | tzafrir | n0rus, yes, the x100p does |
| 12:42.23 | clorabit | anyone can assist me setup my fresh asterisk install |
| 12:42.29 | zoa | well there is something for v.38 and fax |
| 12:42.34 | n0rus | tazfrir: even the clone? |
| 12:42.36 | zoa | but indeed we don't do v34 |
| 12:42.45 | zoa | i meant t.38 and v.34 in the first line |
| 12:42.52 | tzafrir | n0rus, where are you at? to which telco do you connect? |
| 12:42.54 | zoa | so we dont have v34 |
| 12:43.05 | n0rus | tzafrir: I'm located in Norway |
| 12:43.09 | tzafrir | n0rus, most of the detection is done in Asterisk |
| 12:43.23 | coppice | yeah, the latest revision of T.38 has V.34 features, but very few T.38 implementation so it. actually, not that many FAX machines do V.34 |
| 12:43.31 | zoa | yeah indeed |
| 12:43.35 | tzafrir | n0rus, set in zapata.conf: |
| 12:43.42 | zoa | and i cant do it without some license again |
| 12:43.42 | tzafrir | usecallerid=yes |
| 12:43.43 | n0rus | tzfarir: it works with SIP calls. |
| 12:43.50 | tzafrir | callerid=asrecieved |
| 12:43.56 | n0rus | ok |
| 12:43.59 | n0rus | thank you |
| 12:44.00 | coppice | there are numerous patents on V.34 |
| 12:44.02 | n0rus | I'll try that |
| 12:44.12 | coppice | and they have a number of years to run |
| 12:44.27 | clorabit | tzafrir: where i can find tutorial to setup asterisk ? |
| 12:44.40 | tzafrir | clorabit, google? |
| 12:44.40 | zoa | our dsp guy also doesn't seem to like the idea to write it either :) |
| 12:44.46 | tzafrir | voip-info.org |
| 12:45.12 | coppice | a polished V.34 would be a lot of work |
| 12:45.34 | zoa | yeah probably |
| 12:45.55 | n0rus | tzfarir: do I have to reboot after making changes to zapata.conf or will a "reload" do it? |
| 12:46.09 | clorabit | tzafrir: i've install asterisk 1.2.13 what i should do next ? |
| 12:46.17 | coppice | V.17 to V.34 is a bit jump. A number of pretty neat things are needed to make theat step |
| 12:46.24 | coppice | s/bit/big |
| 12:46.46 | zoa | i'm confident the guy could do it, but there is just no point now :( |
| 12:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 12:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 12:47.05 | zoa | i only know 2 freaky guys, atanas and coppice :) |
| 12:47.29 | Chris-NB | JT, hi. I was at lunch |
| 12:47.49 | *** join/#asterisk jm|laptop (n=jamie@dilbert.jamiem.com) |
| 12:48.11 | Chris-NB | JT, I try the new settings (in zaptel.conf) and inform you about my results |
| 12:48.21 | Chris-NB | JT, but it could last a while. |
| 12:48.27 | zoa | coppice, what other ata's do you know that actually have t.38 ? |
| 12:48.36 | tzafrir | coppice, in your foip page you mentioned v.37 as simple and nice thing. Isn't it possible to use it at least as a bridge between different Asterisk/[rt]xfax-s (or openpbx, or iaxmodem) ? I figure you have given some minimal thought for that |
| 12:48.48 | oej | Bah. T.38 can be coded in day powered by redbull. By using cat > t38.c - no editor or anything cheesy. |
| 12:48.55 | zoa | hehe |
| 12:49.01 | sergee | can anybody give me a few TIPS about mISDN and H323? :) |
| 12:49.12 | tzafrir | Or is it basically an equivalent to sending the fax by email? |
| 12:49.21 | sergee | i have a problem when i call chan_h323 -> chan_misdn |
| 12:49.33 | clorabit | anyone can explain deference between fxs and fxo ? |
| 12:49.46 | sergee | if i call chan_sip -> chan_misdn everything OK |
| 12:49.53 | clorabit | sorry for basic question still new in telephony |
| 12:50.04 | tzafrir | clorabit, what hardware do you have? |
| 12:50.13 | clorabit | no hardware at all |
| 12:50.13 | sergee | clorabit: fxS generates power for a phone and fxO recieve power from PBX |
| 12:50.20 | *** join/#asterisk Nobbie (n=no@fwb003.fw.is.co.za) |
| 12:50.49 | *** join/#asterisk dasenjo (n=dasenjo@190.5.196.105) |
| 12:51.00 | Nobbie | eish, * .1.4.0 has me sweeting blood at the moment .... just stopped accepting SIP calls for some reason. how does one troubleshoot that ? |
| 12:51.00 | oej | mv chan_sip.c chan_misdn.c |
| 12:51.03 | tzafrir | You need an FXO adapter if you want to connect to a telco. You need an FXS adapter if you want to connect analog phones |
| 12:51.05 | oej | Should fix a lot of Isdn issues |
| 12:51.10 | clorabit | if i have pstn line which tipe interface should be connected fxo / fxs ? |
| 12:51.20 | coppice | zoa: lots of ATAs say they have T.38. Not all do. There is lots of cheating. A number do have at least semi working T.38 these days, though. |
| 12:51.58 | zoa | yeah i know |
| 12:51.58 | zoa | thomson is one of those fuckers |
| 12:51.58 | coppice | UT Starcom, any of the boxes using Myson Century chips, Most of the Welltech boxes |
| 12:51.58 | clorabit | tzafrir: ic ic |
| 12:52.04 | zoa | oh yes, welltech i tried those too |
| 12:52.16 | zoa | i've never seen a ut starcom gateway |
| 12:52.18 | clorabit | tzafrir: any idea cheap price for fxo adapter ? |
| 12:52.20 | coppice | The put T.38 on the nice colour printed boxes, when there is none inside |
| 12:52.25 | zoa | zyvel should also have some |
| 12:52.26 | sergee | oej: hi! :) |
| 12:53.06 | coppice | tfzfrir: T.38 is stupid. anyone with half a brain would use T.37. However, offices are filled with people who have less than half a brain |
| 12:53.13 | tzafrir | X100Ps are dirt cheap. If you really want to use them |
| 12:53.22 | coppice | yeah, zyxel |
| 12:53.35 | clorabit | X100Ps ?? |
| 12:53.38 | coppice | cisco, of course |
| 12:53.47 | zoa | Coppice: no they would use email :p |
| 12:53.55 | coppice | the sipura 2100, but none of their others |
| 12:54.05 | coppice | T.37 is basically e-mail |
| 12:54.06 | sergee | have some strange things in misdn log, no isdn knowledge though.., will dig source.. as always :) looks like irc is only for questions like "what is fxs/fxo"... |
| 12:54.07 | zoa | sipura 3102 does too |
| 12:54.29 | tzafrir | is there a specific mime type for a fax? |
| 12:54.32 | coppice | most of the sipuras lack the DSP to do it. they can't even do G.729 properly |
| 12:54.36 | merbzt | is sipura 3102 bad ? |
| 12:54.42 | clorabit | tzafrir: where i can buy it online ? |
| 12:54.44 | zoa | coppice: we blew up 3 grandstreams like that :) |
| 12:54.50 | zoa | they just stop working |
| 12:55.04 | coppice | tzafrir: see T.37. Its basically a spec for packing a fax in an e-mail between mailboxes or gateways |
| 12:55.36 | coppice | I've met a couple of boxes that lock solid as soon as they hear fax tone :-) |
| 12:55.44 | zoa | hehe |
| 12:55.52 | zoa | hard reset doesnt cure the grandstream |
| 12:55.57 | zoa | very nice |
| 12:56.54 | coppice | many of the boxes which have a fixed buffer mode and other features specially for fax over G.711 do things that mean it cannot possibly work. The industry standard for ATAs is unbelievably low |
| 12:59.51 | zoa | spa3102 is fine, but everything older than 2100 cannot do it |
| 13:00.15 | zoa | oops i replied to something lines higher |
| 13:00.50 | *** join/#asterisk Ebola (n=Ebola@host81-151-91-139.range81-151.btcentralplus.com) |
| 13:01.11 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 13:01.37 | susinths | which voice codec is used mostly in asterisk servers? |
| 13:01.38 | *** join/#asterisk Ebola (n=Ebola@host81-151-91-139.range81-151.btcentralplus.com) |
| 13:01.56 | zoa | gsm i think |
| 13:02.09 | zoa | or ulaw / alaw internally |
| 13:02.15 | susinths | gsm & G.711 the same? |
| 13:02.26 | susinths | i see |
| 13:04.18 | coppice | zoa: do you integrate your T.38 with *? |
| 13:04.29 | zoa | yes |
| 13:04.43 | zoa | well its a standalone library |
| 13:04.59 | coppice | then either you do quite a bit of work to it, or you have some way to go :-) |
| 13:05.09 | zoa | yeah we did quite a bit of work on it |
| 13:05.42 | coppice | you have to get RTP and UDPTL working on the same port to keep a number of things happy. |
| 13:05.47 | zoa | yes |
| 13:06.32 | coppice | i separated out an additional UDP port management layer in openpbx, so that worked smoothly |
| 13:07.38 | *** join/#asterisk reber (i=reber@gateway/tor/x-7b9ab47848c37f5f) |
| 13:08.07 | *** join/#asterisk neuwald (n=felipe@200.199.198.61) |
| 13:08.43 | neuwald | hi folks |
| 13:08.47 | neuwald | I have one extension like this: |
| 13:08.52 | neuwald | exten = _061.,2,Dial(SIP/tronco_enffbsb/${EXTEN:3},50) |
| 13:09.24 | neuwald | so, I wanna make a new extension, like this: _0XX.,2,Dial(SIP/tronco_enffbsb/...... |
| 13:09.46 | neuwald | but I wanna make the call like this: 025XXnumber |
| 13:09.52 | *** join/#asterisk AstaWerksDotCom (n=doug@63.161.96.170) |
| 13:10.14 | neuwald | Sometime ago I did ${EXTEN:3:5..... but I don't remember how to do this. Anyone can help ? |
| 13:10.49 | AstaWerksDotCom | i just logged in what were you trying to do ? |
| 13:11.30 | neuwald | I'll answer you in PVT to not repeat here |
| 13:11.58 | zoa | im off |
| 13:12.02 | zoa | be back later |
| 13:14.50 | clorabit | tzafrir: do you know digitnetworks.net products ? |
| 13:14.53 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 13:16.01 | tzafrir | clorabit, there are also cheaper X100Ps. But I'm not sure I would use an X100P in a production system |
| 13:16.34 | AstaWerksDotCom | yeah they dont work right with the newer versions of asterisk i tried it. |
| 13:16.48 | AstaWerksDotCom | get a tdm01b instead |
| 13:16.52 | coppice | it rather depends on whether the particular x100p like board is matched to your local line impedance |
| 13:16.53 | clorabit | tzafrir: what u mean with production system ? |
| 13:17.32 | coppice | if it is it works a lot better than something like a TDM400P |
| 13:17.35 | tzafrir | system where people actually expect a good voice quality (and pay for it) |
| 13:17.48 | *** part/#asterisk sergee (i=opera@195.94.224.197) |
| 13:18.29 | clorabit | tza |
| 13:18.33 | tzafrir | A lot better? |
| 13:18.35 | tzafrir | why? |
| 13:19.02 | clorabit | tzafrir: how about linksys spa3000 ? |
| 13:19.36 | coppice | the TDM400 can't seem to run on most machines without loosing data |
| 13:20.08 | clorabit | tzafrir: is it difficult to find voip adapter in my country so far i only find that product |
| 13:20.08 | zoa | for me the x100p was rubbish |
| 13:20.26 | *** join/#asterisk webmad (n=webmad@bkon.it) |
| 13:20.29 | coppice | the x100p is a modem card. modems need very solid operation to work |
| 13:21.04 | *** join/#asterisk ez` (n=Ez@c66.203.210-59.clta.globetrotter.net) |
| 13:21.37 | tzafrir | coppice, surprisingly, people do report succesfully faxing through TDM400P |
| 13:22.13 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 13:22.17 | zoa | yeah we fax through it |
| 13:22.20 | tzafrir | Actually in that thread noone has replied with "it does not work for me" |
| 13:22.23 | zoa | but im not sure if it works allt he time |
| 13:22.38 | coppice | sometimes. I did much of my early development with one. now the same card in the same box won't fax. either the newer linux kernel or the newer driver is probably at fault |
| 13:22.38 | tzafrir | which has surprised me |
| 13:23.12 | coppice | the TDM400 doesn't fax reliably in most machines these days |
| 13:23.14 | *** join/#asterisk jojo^ (n=jol@ph4.se) |
| 13:24.27 | jojo^ | I'm trying to build asterisk 1.4.0 and install it in a specific location, but ./configure --prefix= doesnt seem to do the trick. Any hints? |
| 13:25.49 | Nobbie | jojo: yeah, take baby steps with 1.4.0 |
| 13:27.09 | tzafrir | jojo^, what error message do you get? |
| 13:27.48 | tzafrir | the "permission denied" one |
| 13:29.24 | RoyK | does 1.4 still run as root 'out of the box'? |
| 13:30.42 | jojo^ | tzafrir, Yeah. It tries to install in / regardless of --prefix, or manually exporting INSTALL_PATH |
| 13:32.20 | tzafrir | jojo^, Please be more specific |
| 13:32.31 | tzafrir | Could you pastebin the relevant details? |
| 13:33.09 | tzafrir | RoyK, I figure. Adding a user is not the job of a program |
| 13:34.00 | jojo^ | tzafrir, I want asterisk to get installed in /home/asterisk (+/bin +/etc and so on). How do I tell the build-process that? |
| 13:34.52 | jojo^ | tzafrir, I've tried ./configure --prefix=/home/asterisk, and also export INSTALL_PATH=/home/asterisk and then make install, but it still tries to install in /. "mkdir -p /var/lib/asterisk/static-http" |
| 13:35.12 | RoyK | tzafrir: I'm aware of that, sir, but I was wondering if it did anything like chuser/chgroup like openpbx. openpbx us started as root, but changes later |
| 13:36.54 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 13:38.39 | tzafrir | RoyK, Asterisk has had -U since before the openpbx fork |
| 13:39.20 | tzafrir | in the Debain package, the Asterisk init.d script refuses to run asterisk without those parameters. |
| 13:40.04 | *** join/#asterisk nakee (n=nakee@grok.cs.huji.ac.il) |
| 13:40.21 | tzafrir | RoyK, in fact, you have to execute Asterisk as root and not as a user if you want -p (realtime scheduling priority) to work |
| 13:42.10 | yassine | tzafrir, i have enabled nat=yes for each sip user and forwarded calls comming from zap extention to a sip user now when a call comes in no one can hear eachother any ideas please ? |
| 13:42.58 | Gido-E | yassine check your tunnels |
| 13:42.59 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 13:43.32 | yassine | Gido-E, how can i achieve that ? |
| 13:44.14 | RoyK | tzafrir: I don't want asterisk with -p. suddenly there's a hang, and someone has to drive up the server farm.... |
| 13:45.12 | tzafrir | RoyK, keep the manager interface availble |
| 13:45.48 | tzafrir | RoyK, now you know why Digium developed the manager over HTTP ;-) |
| 13:46.48 | RoyK | tzafrir: I'd love to see the manager and the current cli go away. it's a security hole by design |
| 13:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:47.29 | tzafrir | RoyK, what alternative do you suggest? |
| 13:48.07 | RoyK | a real client/server approach |
| 13:48.09 | tzafrir | That is reasonably functional and resonably secure? |
| 13:48.17 | Gido-E | yassine: iptraf, tcpdump, ethereal |
| 13:48.30 | RoyK | a real client/server approach |
| 13:48.50 | yassine | Gido-E, i have mapped many ports from the router to my box and know that are well tunneled if thats what you mean ?? |
| 13:49.16 | tzafrir | RoyK, even the CLI uses a client-server approach. e.g: command-completion is done by the server... |
| 13:49.48 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 13:50.10 | RoyK | tzafrir: you probably know the difference between a server having a native cli/console and a true client/server architecture |
| 13:54.10 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 13:57.15 | *** join/#asterisk signius (n=IceChat7@dsl-217-155-69-102.zen.co.uk) |
| 13:58.24 | badcfe | hallo. is it possible to issue an outgoing call from an asterisk box. i mean without having an incoming call falling in some context triggering this. ? |
| 13:58.46 | badcfe | say from a crontab to take a concrete example.. |
| 13:59.37 | jojo^ | I'm trying to build asterisk 1.4.0 and install it in a specified location, but ./configure --prefix=/home/asterisk doesnt seem to do the trick. Any hints? |
| 14:01.09 | HarryR | jojo^, when you do 'make install' set the DESTDIR environment, e.g. 'DESTDIR=/home/asterisk make install' |
| 14:02.04 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 14:02.19 | jojo^ | HarryR, Thanks! |
| 14:04.57 | *** join/#asterisk davewise (i=icechat5@ofc.agcllc.net) |
| 14:06.54 | *** join/#asterisk davewise (i=icechat5@ofc.agcllc.net) |
| 14:08.09 | *** part/#asterisk davewise (i=icechat5@ofc.agcllc.net) |
| 14:08.49 | *** join/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net) |
| 14:16.43 | *** join/#asterisk mial (n=Semionsi@shound.org) |
| 14:16.49 | mial | 'morning |
| 14:17.11 | mial | is it possible to make asterisk spawn less threads at startup ? |
| 14:18.30 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:19.36 | jojo^ | Is it supported to run asterisk 1.4.0 as a regular user and not root? |
| 14:20.15 | mial | afaik asterisk runs as the 'asterisk' user and not root |
| 14:21.30 | *** join/#asterisk af_ (n=af@ip-173-157.sn1.eutelia.it) |
| 14:22.13 | *** join/#asterisk kgpsathish (n=sathish@61.246.251.18) |
| 14:22.58 | kgpsathish | hi |
| 14:23.23 | tzafrir | mial, why does it bother you? |
| 14:23.39 | mial | tzafrir: I run asterisk on a nslu2 |
| 14:23.44 | mial | with only ... 32 Mb of ram |
| 14:23.51 | HarryR | I'd be very scared of running asterisk as root |
| 14:23.52 | tzafrir | remove modules you don't need |
| 14:25.10 | tzafrir | mial, actually: disable autoload, and load explicitly only the modules you need |
| 14:25.25 | tzafrir | This can be a pain, but can save memory |
| 14:26.18 | mial | okay |
| 14:27.27 | Chris-NB | anyone knows what that mean: WARNING[30244]: chan_zap.c:8650 zt_pri_error: 9 !! Unexpected Channel selection 3 |
| 14:31.06 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:34.03 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 14:37.35 | *** join/#asterisk Tili (n=tili@87.219.93.228) |
| 14:37.48 | Tili | is it possible to select codec from extensions.conf |
| 14:37.56 | Tili | before answering a call |
| 14:38.21 | *** part/#asterisk mial (n=Semionsi@shound.org) |
| 14:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:46.56 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:52.05 | zoa | its easy to talk to mark on irc |
| 14:52.09 | zoa | he does not show up for 30 days |
| 14:52.11 | zoa | then he says hi |
| 14:52.14 | zoa | says 1 phrase |
| 14:52.18 | zoa | and is gone again for 30 days |
| 14:52.29 | Tili | zoa: what is he upto? |
| 14:52.56 | zoa | working i guess |
| 14:52.58 | mitcheloc | send him an e-mail or call him on the phone :) |
| 14:57.22 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:01.19 | webmad | I got a Dialogic D/4PCI card. Does it work with Asterisk? |
| 15:01.30 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 15:02.06 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 15:02.45 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:02.45 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:03.10 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 15:05.01 | *** join/#asterisk hohum (n=dcorbe@mercury.sunrocket.com) |
| 15:05.09 | e-ddie | webmad: ask google |
| 15:06.32 | mercestes | for that matter, ask Dialogic. lol |
| 15:10.39 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 15:11.09 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 15:13.08 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 15:13.27 | webmad | e-ddie: I found several pages on Google but I didn't find a clear answer. This page: http://www.asterisk.org/hardware doesn't talk about Dialogic hardware |
| 15:14.02 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 15:14.09 | webmad | mercestes: will you pay consulting fees to Dialogic? LOL |
| 15:14.19 | tzafrir | webmad, http://www.asterisk.org/hardware is obsolete. Ignore it |
| 15:14.46 | tzafrir | Some dialogic cards work with the proprietary Asterisk version from Digium |
| 15:14.59 | tzafrir | Others don't work at all |
| 15:15.55 | coppice | A D/4PCI won't work. its half duplex |
| 15:16.02 | webmad | tzafrir: AFAYK, do I need Dialogic software? Someone talks about "chan_dialogic.so". |
| 15:16.09 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 15:16.17 | webmad | coppice: thx |
| 15:16.52 | *** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) |
| 15:17.19 | emiquelito | hello all! Is there any problems one iax softphone trying to call a sip softphone? Both are extensions registered at the same server. Can asterisk handle this situation properly? |
| 15:17.36 | zoa | yes it works fine |
| 15:17.59 | *** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) |
| 15:18.23 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:18.24 | emiquelito | zoa, ok, but I'm having this debug message one trying to call the iax softphone using my registered sip extension: Unable to find key '5847' in family 'SIP/Registry' |
| 15:18.29 | emiquelito | I'm using realtime |
| 15:18.37 | emiquelito | s/one/on |
| 15:18.50 | *** join/#asterisk rrocha (n=brain@200.193.155.234) |
| 15:19.00 | zoa | wtf |
| 15:19.06 | zoa | aaaah |
| 15:19.12 | zoa | something is wrong in your realtime config then :) |
| 15:19.19 | emiquelito | I have looked at the table in mysql and everything is ok |
| 15:19.30 | emiquelito | there is a query in the debug message, which is also ok |
| 15:19.39 | zoa | cant help you then |
| 15:19.47 | zoa | without realtime it should work just fine |
| 15:20.06 | *** part/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) |
| 15:20.15 | emiquelito | zoa, look, I have one table called users for both SIP and IAX |
| 15:20.21 | *** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) |
| 15:20.32 | emiquelito | I think it should work |
| 15:21.00 | emiquelito | and the interesting thing is that the IAX extension is able to call the SIP one |
| 15:21.17 | emiquelito | but the opposite doesn't work |
| 15:25.51 | zoa | aha |
| 15:29.49 | *** join/#asterisk Weezey (n=ohno@lan6.LO.iasl.com) |
| 15:30.31 | clorabit | hello ... |
| 15:30.58 | clorabit | do i need install zaptel even i don't use any fxo / fxs device ? |
| 15:31.00 | Weezey | I'm running 1.2 on two machines, one has g729 licenses, the other doesn't. So when I send a g729 call to the iax2 connection between them it sets it up as ulaw, but i get a ton of "Jan 24 10:27:56 WARNING[13043]: chan_sip.c:2561 sip_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 64/64)" messages with no audio/rings/anything on the call. |
| 15:32.02 | zoa | so you have A calling B and B will pass on the call to C ? |
| 15:32.08 | Weezey | yes |
| 15:32.13 | zoa | A sends it to B in g729 |
| 15:32.18 | Weezey | no |
| 15:32.22 | Weezey | wait |
| 15:32.46 | tzafrir | clorabit, you may need zaptel for a timind source. If you don't have any zaptel hardware, you can use ztdummy |
| 15:33.52 | Weezey | Sip (g729) -> Asip (g729) --- Aiax2 (ulaw) --> Biax2 (ulaw) --- Biax2 (ulaw) --> Ciax2(ulaw) -- Csip(g729) |
| 15:34.10 | clorabit | tzafrir: ah ic ic.. thanks |
| 15:34.12 | Weezey | error messages show up on A |
| 15:34.39 | zoa | so what server has a g729 license > |
| 15:34.40 | zoa | ? |
| 15:34.44 | Weezey | A and C |
| 15:35.22 | zoa | and what server bitches about the codec ? |
| 15:35.27 | Weezey | A |
| 15:36.07 | Weezey | I tried installing codec_g729a.so on B because I thought that might have something to do with it (because even though it's not using it, it's still setting up the call) but that didn't help. |
| 15:36.18 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 15:36.52 | Weezey | I'm wondering if I should try allowing g729 for B instead and try to use it without a license |
| 15:37.17 | zoa | how about not using ulaw at all ? |
| 15:37.22 | zoa | and just go g729 all the way ? |
| 15:37.24 | *** join/#asterisk Mportnoy (n=test@201.199.68.150) |
| 15:37.32 | Weezey | yeah, that's next. |
| 15:37.48 | Weezey | I just wondered if there was something simple I was missing. |
| 15:37.56 | *** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) |
| 15:37.57 | Weezey | technically what I'm doing should be possible. |
| 15:38.02 | Mportnoy | does anybody knows how to install Asterisk 1.2.14 addons 1.2.5 Debian 3.1 with CDR MYSQL ? |
| 15:38.15 | *** join/#asterisk trixman (n=andy@rrcs-67-53-168-147.west.biz.rr.com) |
| 15:39.27 | trixman | can someone help me with a ringgroup problem |
| 15:39.49 | trixman | l |
| 15:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:47.18 | mercestes | Mportnoy: I know how to do it in gentoo. |
| 15:48.52 | Mportnoy | mercestes: how can I test |
| 15:48.57 | Mportnoy | if everything is working ? |
| 15:49.13 | mercestes | Mportnoy: read the manual. |
| 15:49.47 | mercestes | Mportnoy: by everything, do you mean the Cdr mysql, or do you mean literally everything? |
| 15:50.41 | Mportnoy | CDR MYSQL |
| 15:50.57 | mercestes | make a call. Log in to mysql. Do a use asterisk; and a select * from cdr; |
| 15:50.57 | *** join/#asterisk sudhir492 (n=sudhir@c-71-63-59-45.hsd1.va.comcast.net) |
| 15:51.38 | sudhir492 | hi all |
| 15:51.44 | *** join/#asterisk florz (i=nobody@2002:58c6:2592:1:0:0:0:2) |
| 15:51.45 | mercestes | hi sudhir492. |
| 15:52.15 | Mportnoy | mercestes: which files I need to modify for MYSQL CDR? |
| 15:52.31 | Mportnoy | so far I did modules.conf and add the cdr_mysql.conf |
| 15:52.38 | mercestes | res_mysql.conf |
| 15:52.43 | mercestes | I think |
| 15:53.09 | mercestes | yea, res_mysql.conf |
| 15:54.04 | Mportnoy | there is cdr_my and res_mysq |
| 15:54.11 | Mportnoy | I read that for cdr is only cdr |
| 15:54.41 | Mportnoy | http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql |
| 15:55.05 | sudhir492 | is it possible to put 2 quad T1 cards in a PSTN gateway? |
| 15:55.06 | mercestes | Mportnoy: I focused on editting res_mysql.conf |
| 15:55.24 | zoa | sudhir492: ues |
| 15:55.25 | zoa | yes |
| 15:55.31 | zoa | would i do that: no |
| 15:55.50 | mercestes | sudhir492: If you wanna use two cards go sangoma. They are mroe irq friendly. |
| 15:56.02 | mercestes | You can do it with diguim hardware if you are careful to check your IRQs and maek sure they are not shared. |
| 15:56.23 | mercestes | or are shared with something lazy that doesn't care if the IRQ tables are being raped by a t1-device |
| 15:56.51 | sudhir492 | Yes, I can go with Sangoma, (although people say that newer digium cards can match Sangoma's performance) |
| 15:57.44 | mercestes | Sangoma doesn't rape IRQs |
| 15:58.32 | HarryR | ;) |
| 15:58.43 | mercestes | other than that, I can't really say that Sangoma is superior, just IRQ and thus multicard friendly. |
| 15:59.28 | mercestes | Driver support is very anti-intuitive, the "stable" drivers are called "betas" |
| 16:00.00 | trixman | hi |
| 16:00.09 | coppice | installing the sangoma software sucks. other than that, their stuff is nice |
| 16:00.23 | sudhir492 | I plan to use the box strictly for a PSTN gateway, which will distribute the calls to other boxes |
| 16:01.34 | Mportnoy | Jan 24 10:01:05 WARNING[8816]: res_config_mysql.c:522 reload: MySQL RealTime: Couldn't establish connection. Check debug. |
| 16:01.37 | sudhir492 | coppice: You are right about Sangoma install. However I have done that for a few boxes |
| 16:03.56 | Mportnoy | <PROTECTED> |
| 16:06.09 | mercestes | Mportnoy: Turn on yoru debug |
| 16:07.22 | Mportnoy | mercestes: set debug 100 |
| 16:07.23 | Mportnoy | ? |
| 16:08.24 | sudhir492 | zoa: why would you not put 2 quad cards in a box. With 2 dual core processors, and faster bus these days, they seem to have more that double the performance of boxes a few years ago. I have a dual Xeon server running faithfully for 3 years with quad card, average 30 simultaneous calls but peak between 60 to 60 |
| 16:08.47 | sudhir492 | 50 to 50 |
| 16:08.54 | sudhir492 | oops. 50 to 60 |
| 16:09.52 | zoa | yes well you can |
| 16:09.54 | zoa | but i wouldnt |
| 16:09.55 | zoa | :) |
| 16:10.00 | zoa | if 1 port fucks up |
| 16:10.06 | zoa | chances are big everything would fuck yp |
| 16:10.07 | zoa | up |
| 16:10.08 | sudhir492 | others will run fine |
| 16:10.26 | sudhir492 | 8 ports are distributed over 2 cards |
| 16:10.56 | *** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) |
| 16:11.04 | sudhir492 | by the same logic, would you avoid 4-port cards also ?;-) |
| 16:11.21 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 16:11.51 | mercestes | Mportnoy: set debug with no arguements. make sure debug is under console => in logger.conf |
| 16:13.41 | Mportnoy | now i Have only this prob |
| 16:13.44 | Mportnoy | Jan 24 10:13:20 ERROR[8867]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on 192.168.1.151 (err 2003). Check debug for more info. |
| 16:13.44 | Mportnoy | Jan 24 10:13:20 WARNING[8867]: res_config_mysql.c:522 reload: MySQL RealTime: Couldn't establish connection. Check debug. |
| 16:13.44 | Mportnoy | <PROTECTED> |
| 16:14.17 | Mportnoy | Connected to asterisk@localhost, port 3306 using table cdr for 41 seconds. |
| 16:16.42 | mercestes | do a logger reload |
| 16:17.17 | mercestes | and try setting your "dbsock" variables and using "localhost" if you are using a Mysql server local to *. |
| 16:18.40 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 16:22.50 | ez` | asterisk support g729ab ? |
| 16:23.13 | Tili | ez`: yeah |
| 16:24.33 | Tili | g729a |
| 16:24.34 | ez` | g729ab ? |
| 16:24.34 | *** join/#asterisk stefmtl (n=stef@stef.istop.com) |
| 16:24.35 | Qwell[] | a |
| 16:24.51 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 16:25.25 | ez` | my polycom ip500 use g729ab ; asterisk will be able to play with this codecs ? |
| 16:25.52 | Qwell[] | I think I recall hearing that it was somehow compatible |
| 16:25.57 | stefmtl | Hello, is there a way to change RX and TX gain on a SIP channel, like we can do with a zaptel interface ? |
| 16:26.11 | Qwell[] | stefmtl: your phone should have a volume key... |
| 16:26.50 | stefmtl | Qwell : the calling flow is : user over pstn>quintum fxo switch> asterisk IVR |
| 16:27.09 | Qwell[] | stefmtl: then it would be a feature of the quintum |
| 16:27.45 | stefmtl | Qwell : on the asterisk side, there's nothing to do ? |
| 16:28.02 | Qwell[] | stefmtl: I don't think so, no |
| 16:29.02 | ez` | something funny about polycom ; i fix to g729 using web interface ; its dont use g729 ; but i edit manualy sip.cfg on my tftp server ; it use it ... |
| 16:29.47 | *** join/#asterisk CPSK (n=CPSK@c6.ars.ba.nextra.sk) |
| 16:34.11 | *** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com) |
| 16:34.51 | MRH2 | Hi can an NMI error be caused by a compatability issue with a specific kernel |
| 16:35.30 | *** join/#asterisk af_ (n=af@ip-173-157.sn1.eutelia.it) |
| 16:36.37 | MRH2 | (NMI with zaptel) |
| 16:37.45 | sweeper | is there a list somewhere of crazy shit people do with asterisk? |
| 16:38.12 | sweeper | or a "solutions looking for problems" list? :v |
| 16:38.56 | Qwell[] | sweeper: what, you don't subscribe to the -users mailing list? eh |
| 16:38.58 | Qwell[] | heh* |
| 16:39.03 | sevard | how about a wikipedia, voip-info.org |
| 16:39.22 | MRH2 | Is it possible an NMI error will be fixed by a distro upgrade? |
| 16:40.03 | Qwell[] | MRH2: kernel upgrade, perhaps? |
| 16:40.06 | sweeper | wikipedia article is fairly barren |
| 16:40.19 | sweeper | Qwell[]: mailing lists? :v |
| 16:40.25 | Qwell[] | lists.digium.com |
| 16:40.51 | coppice | MRH2: NMI errors are usually just to hardware problems |
| 16:41.00 | MRH2 | yeah i know |
| 16:41.19 | MRH2 | but could a kernel upg resolve it |
| 16:41.25 | MRH2 | (is it feasible) |
| 16:42.14 | sevard | ahh, another user looking for a magic bullet solution. |
| 16:43.06 | coppice | would a distro upgrade solve world poverty? |
| 16:43.17 | Qwell[] | coppice: yes, it could |
| 16:43.25 | Qwell[] | at least...that's what ubuntu thinks ;) |
| 16:44.50 | MRH2 | I've gone through the usual hw stuff, narrowed it down to a specific zaptel revision, posted to mantis referred to digium not getting ver far |
| 16:45.26 | ez` | i registered my g729 licence bought from digium ; do i need to compile a module ; or g729 is compiled / included by default ? |
| 16:45.45 | MRH2 | planning to move to move to cent-os and if it is possible that *could* fix it I will hold off |
| 16:46.38 | MRH2 | so... could a kernel update resolve a NMi error? |
| 16:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:47.15 | coppice | i think its more likely to solve poverty |
| 16:47.46 | sevard | CPSK: I've seen you ask the same question, over and over, and over.... and over.. and OVER. Why don't you just call Tenovis? |
| 16:48.58 | MRH2 | ok anyone know how many TE411P cards actually shipped before it got the new echo can? |
| 16:48.58 | zoa | not too much i think |
| 16:49.00 | zoa | i have 3 or so |
| 16:49.39 | Geert | Does anybody have more information about "zapata.conf -> privateprefix" |
| 16:49.41 | file | it's a different model number don't forget for the newer echo cancellation |
| 16:49.42 | MRH2 | that would explain why not many people will be complaining |
| 16:49.50 | zoa | the 412 is a lot better |
| 16:49.52 | Geert | and about unknownprefix too |
| 16:50.03 | zoa | i dont think they sell the 411 now |
| 16:50.17 | zoa | im very happy with the 412 |
| 16:50.20 | zoa | and was not with the 411 |
| 16:50.34 | MRH2 | any way to trade it up? |
| 16:50.36 | zoa | the 412 stops the dtmf talkoff |
| 16:50.41 | zoa | i dont know |
| 16:50.59 | MRH2 | i get probs with zaptel >=R1115 |
| 16:51.42 | MRH2 | i think that was when the 412 echo can was introduced |
| 16:51.57 | dhill | anyone know what this may mean from an adtran 900? |
| 16:52.07 | dhill | 16:03:07 TM.T01 64 SipTM_ReInviting Adding RTP Media Gateway Entry: 127.0.0.2:10000 -> 10.10.10.1:10000 |
| 16:52.07 | dhill | 16:03:07 TM.T01 64 SipTM_ReInviting No action taken, firewall traversal is not enabled |
| 16:52.07 | dhill | 16:03:07 TM.T01 64 SipTM_ReInviting call-leg-mod (0x2875228) -> Modify Re-Invite Sent |
| 16:52.07 | dhill | 16:03:07 TM.T01 64 SipTM_ReInviting ERROR! SipCallLegModifyStateChanged to ReinviteSent ignored |
| 16:52.07 | dhill | 16:03:07 TM.T01 64 SipTM_ReInviting sent: re-INVITE |
| 16:52.21 | dhill | 16:03:07 TM.T01 64 SipTM_ReInviting call-leg-mod (0x2875228) -> Modify Re-Invite Remote Accepted |
| 16:52.21 | dhill | 16:03:07 TM.T01 64 State change >> SipTM_ReInviting->SipTM_ReInvitingPassed |
| 16:52.21 | dhill | 16:03:07 TM.T01 64 SipTM_ReInvitingPassed sent: TA->ReInvite Response - PASS |
| 16:52.21 | dhill | 16:03:07 TM.T01 64 SipTM_ReInvitingPassed sent: TA->ReConnect |
| 16:52.39 | MRH2 | pastebin is ur friend |
| 16:52.44 | dhill | oh yea :) |
| 16:53.37 | phearless | hum... I got a little question |
| 16:53.49 | phearless | how can I redirect a caller to the voicemail of another extension ? |
| 16:54.51 | MRH2 | oh well .... i will check back next month lol |
| 16:56.09 | phearless | :) |
| 16:57.12 | *** join/#asterisk trixman (n=andy@rrcs-67-53-168-147.west.biz.rr.com) |
| 16:58.57 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:00.20 | sevard | GRAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAPOUHGIUEHGUOhoiushgs |
| 17:00.47 | coppice | CPSK: The rectangular plugs go in the rectangular holes |
| 17:01.00 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 17:01.03 | Qwell[] | coppice: can the square plug go into the rectangular hole? |
| 17:01.08 | Qwell[] | I mean...it'll fit |
| 17:01.19 | coppice | depends on the size of hammer you use |
| 17:01.22 | sevard | that's all we care about at this point. |
| 17:01.23 | zoa | :) |
| 17:01.30 | zoa | use lubricant |
| 17:01.50 | sevard | zoa knows all about lube, eh eh, eh zoa, eh? |
| 17:02.20 | coppice | you mean those KY-45s? |
| 17:02.25 | sevard | that was a good night. |
| 17:02.45 | CPSK | coppice: thanx :) |
| 17:03.09 | Qwell[] | zoa: stupid question... what is .be? |
| 17:03.26 | Qwell[] | Why do I want to say Belgium? |
| 17:03.26 | coppice | the worst place in the galaxy |
| 17:05.41 | zoa | http://adsoftheworld.com/files/images/manix_lubricant.jpg |
| 17:06.01 | zoa | i use to have www.2-be-or-not-2.be |
| 17:06.06 | *** join/#asterisk Murdock__ (n=brian@host217-40-20-25.in-addr.btopenworld.com) |
| 17:08.33 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 17:09.25 | in-pt | hi all |
| 17:09.34 | in-pt | anyone using cisco 7940G ip phone |
| 17:10.36 | *** join/#asterisk l2cache (n=ghansen@64.128.254.98) |
| 17:10.39 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 17:12.02 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 17:12.35 | zoa | everybody too shocked now ? :p |
| 17:13.05 | l2cache | i have a dialplan question : I want an agent to call extension 100 and prompt to add or remove their ext from a queue. I need to put their caller id to a variable so i can put AddQueueMember(queuename|SIP/$variable) ...any thoughts? |
| 17:15.19 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:15.19 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:17.05 | sevard | l2cache: look at the special variables on voip-info for asterisk |
| 17:17.17 | CPSK | coppice: thanx :) |
| 17:17.43 | *** join/#asterisk Vulpyne (n=na@sta-208-139-193-163.rockynet.com) |
| 17:18.13 | l2cache | can i use AddQueueMember(queuename|SIP/${CALLERID}) |
| 17:18.42 | coppice | ah, CPSK is using an "annoy everyone" bot |
| 17:19.02 | *** join/#asterisk Gankhuu (n=gankhuu@72-166-51-162.dia.static.qwest.net) |
| 17:20.28 | *** join/#asterisk s1gny|wrk (n=s1gny@p54914E79.dip.t-dialin.net) |
| 17:20.34 | *** part/#asterisk s1gny|wrk (n=s1gny@p54914E79.dip.t-dialin.net) |
| 17:26.00 | CPSK | coppice: thanx :) |
| 17:26.10 | sevard | OH MY GOD. |
| 17:26.15 | sevard | JESUS. |
| 17:26.33 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 17:26.46 | phearless | how can I redirect a caller to the voicemail of another extension ? |
| 17:26.49 | *** join/#asterisk dmaas (n=noone@h-69-3-134-254.lsanca54.dynamic.covad.net) |
| 17:27.22 | sevard | you can assign the voicemail box in sip.conf or you can create an extension |
| 17:28.49 | dmaas | i was wondering if anyone could tell me how reliable commercial non-asterisk pbxes are? Basically - I'm building an asterisk box and want to know how long it should be able to run for, how long the hardware should last for, and things like that. |
| 17:30.30 | phearless | sevard: not like this |
| 17:30.30 | phearless | I receive a call |
| 17:30.30 | phearless | I talk |
| 17:30.30 | phearless | and then the caller tell me that he want to let a message for Mr 205 |
| 17:30.31 | phearless | so I want to redirect the caller to the voicemail 205 |
| 17:30.31 | phearless | make sense? |
| 17:30.35 | mercestes | dmaas: As long as the hardware lasts. and when the hardware dies, ti's far easier/cheaper to replace. |
| 17:30.36 | *** part/#asterisk Vulpyne (n=na@sta-208-139-193-163.rockynet.com) |
| 17:30.37 | sevard | and you don't want the extension to ring |
| 17:31.08 | sevard | the only other way I can think of doing that is to set up a seperate extension array or context that kicks all straight to vm |
| 17:31.12 | phearless | no ring, that's right |
| 17:31.29 | phearless | or it can ring but I would not prefer |
| 17:31.41 | mercestes | dmaas: commercial hardware PBX's have this really annoying habit of never dying. They become giant piles of sticky corroded liquified oxidized metal deposits slowly leaking out this amber residue that will never wash away....but they keep churning along emitting this acrid sulfur-like "burnt electronics" smell. |
| 17:31.54 | sevard | which you could do with exten = 5,1,(5@context) |
| 17:32.10 | dmaas | mercestes: hmmm, thanks. |
| 17:32.13 | mercestes | dmaas: But they never "die" They just slowly add more and more humm, static, distortion and sound anamolies as they slowly crawl through the centuries. |
| 17:32.24 | yatesy | you just need something in your extensions to call Voicemail(uEXTENSION) surely? |
| 17:32.40 | dmaas | mercestes: hmmm, i didnt think of that... |
| 17:32.42 | phearless | I want to redirect to a voicemail during a call |
| 17:33.00 | phearless | but there is something that I do not get : |
| 17:33.03 | phearless | during a call, |
| 17:33.20 | sevard | erm, Voicemail* |
| 17:33.25 | sevard | left that out, scatterbrained this morning |
| 17:33.30 | phearless | I can't use any asterisk commands, like VoiceMail, right ? |
| 17:33.46 | yatesy | assign numbers that go directly to Voicemail(uUSER) then, slap it in another context or something |
| 17:33.51 | mercestes | dmaas: They're old skool electronics. That stuff just didnt' die. Now the *new* commerical PBX's are total trash. They are proprietary and insanely expensive (just like the old PBX's) and they still die. |
| 17:33.53 | phearless | I can't decide to redirect "*9 205" to the voicemail 205 |
| 17:33.53 | sevard | phearless: you'd transfer them to an xtension that went straight to the user's vm box |
| 17:34.13 | yatesy | so say you've got 100,1,Dial(SIP/100) then just set up *100,1,Voicemail(u100) |
| 17:34.29 | phearless | so how can I do *100 ? |
| 17:34.35 | phearless | during the call? |
| 17:34.45 | mercestes | There is a software based "metaswitch" that you could look into that's a software commercial switch solution that has nothign to do wtih * but I hear it's a hog to work with. |
| 17:34.47 | yatesy | yea just transfer the call to that extension |
| 17:35.01 | sevard | phearless: for the fourth time, you'd transfer to the extension |
| 17:35.01 | yatesy | blind transfer would obviously work best here |
| 17:35.22 | phearless | okay I see |
| 17:35.29 | wunderkin | exten => _#XXX,1,VoiceMail(${EXTEN:1}@vmcontext|u), if you are using sip, use the transfer button, if not, setup features.conf |
| 17:35.34 | phearless | the phones are Sipura/Linksys 942 |
| 17:35.57 | *** join/#asterisk frc11 (n=fribelle@212.145.178.3) |
| 17:36.01 | yatesy | perfect wunderkin :) |
| 17:36.05 | yatesy | ta for that |
| 17:36.14 | phearless | okay guys I will try that now |
| 17:36.26 | phearless | I will tell you if it worked.. |
| 17:37.32 | frc11 | hello... my name is Fer and I would like that someone could answer some "basic" questions |
| 17:37.47 | *** join/#asterisk xpotx (n=jim@71-213-32-194.slkc.qwest.net) |
| 17:38.18 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 17:38.23 | sevard | frc11: it's best if you just ask instead of asking to ask, unless of course if you're looking for a contractor |
| 17:38.51 | c4t3l | hello all |
| 17:39.16 | c4t3l | has anyone had any success with the asterisk-stat application |
| 17:39.35 | frc11 | the first one is about Digium. I'm evaluating in purchase one TDM04B but I would like to know if it will be full compatible with AMD |
| 17:39.50 | frc11 | yes, sevard, thnks |
| 17:39.51 | *** part/#asterisk webmad (n=webmad@bkon.it) |
| 17:40.05 | phearless | okay |
| 17:40.18 | phearless | when I receive a call, |
| 17:40.24 | phearless | I pressed "XFER" |
| 17:40.27 | *** join/#asterisk clorabit (n=eddysety@it.petra.ac.id) |
| 17:40.30 | clorabit | hello.. |
| 17:40.32 | FuriousGeorge | what a pain in the butt |
| 17:40.35 | phearless | then I was ready to press #408 |
| 17:40.36 | phearless | but ! |
| 17:40.38 | in-pt | does any one knows if asterisknow supports skinny channels |
| 17:40.42 | clorabit | got problem here anyone can help |
| 17:40.45 | phearless | when I pressed #, I had the busy tone |
| 17:40.58 | Strom_C | phearless: is # part of the extension number? |
| 17:41.19 | Strom_C | if it isn't, then don't dial it |
| 17:41.27 | FuriousGeorge | i have a location that wants a video voip, which i know is better supported in 1.4, by softphones (like eyebeam) need it to work with video for windows |
| 17:41.27 | Strom_C | just try the XFER key followed by 408 |
| 17:41.33 | *** join/#asterisk somethingbad (n=vladi@84.238.212.212) |
| 17:41.35 | FuriousGeorge | so im limited to fixed aperture usb cameras |
| 17:41.37 | clorabit | i trying to run safe_asterisk and failed with error message "Asterisk ended with exit status 127" any suggest |
| 17:41.52 | Strom_C | clorabit: what happens when you run "asterisk -cvvvvvg" |
| 17:41.57 | FuriousGeorge | ive seen a driver that will turn a camcorder into a usb camera |
| 17:42.07 | c4t3l | clorabit, what Strom_C said |
| 17:42.17 | FuriousGeorge | but i feel like that's a hack, that there should be a hardware solution, anyone know of one? |
| 17:42.17 | Qwell[] | Strom_C: "cutoff on disconnect" - ever heard of it? |
| 17:42.18 | phearless | I got : |
| 17:42.20 | phearless | ;DIRECT VOICEMAIL |
| 17:42.21 | phearless | exten => _#[45]XX,1,VoiceMail(${EXTEN:1}@default) |
| 17:42.38 | Strom_C | phearless: it's a baaaaad idea to start extensions with # |
| 17:42.45 | phearless | but when I press #, at any time, I got the busy tone |
| 17:42.49 | mercestes | c4t3l: S orry, I don't even have an asterisk-stat app? Where'd you get it? is that a 1.4 thing? |
| 17:42.49 | Strom_C | # means "I am finished dialing now; put the call through" |
| 17:42.50 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 17:42.55 | c4t3l | phearless dont use # it may be enables in features.conf |
| 17:42.57 | phearless | <wunderkin> exten => _#XXX,1,VoiceMail(${EXTEN:1}@vmcontext|u), if you are using sip, use the transfer button, if not, setup features.conf |
| 17:43.00 | phearless | I did this |
| 17:43.08 | phearless | ok Strom_C |
| 17:43.08 | Qwell[] | Strom_C: me, me, me! |
| 17:43.08 | clorabit | Strom_C: symbol lookup error: /usr/lib/asgterisk/modules/codec_speex.so: undefined symbol: speex_decode_int |
| 17:43.10 | Qwell[] | :D |
| 17:43.15 | FuriousGeorge | phearless: ive used * and ** in the past |
| 17:43.24 | Strom_C | Qwell[]: give me more context |
| 17:43.31 | c4t3l | mercestes, its supposed to be a web reporting app with graphs and such |
| 17:43.47 | Qwell[] | Strom_C: guy in Canada is looking to get disc sup on his line |
| 17:43.48 | wunderkin | yeah well * and # may not be the best but you can use whatever, i use it, because i dont use #, it is just an example |
| 17:43.51 | FuriousGeorge | is there a such thing as a usb camera that doesnt suck? |
| 17:44.09 | Qwell[] | is "cutoff on disconnect" the magic option he wants? |
| 17:44.16 | Strom_C | Qwell[]: battery drop at the end of the call? |
| 17:44.18 | phearless | ok FuriousGeorge |
| 17:44.21 | phearless | ok wunderkin too |
| 17:44.25 | phearless | I tried with : |
| 17:44.25 | clorabit | Strom_C: and other unresolve message error |
| 17:44.26 | Strom_C | and who's calling it "cutoff on disconnect"? |
| 17:44.31 | Qwell[] | telco, I guess? |
| 17:44.33 | phearless | exten => _*8[45]XX,1,VoiceMail(${EXTEN:1}@default) |
| 17:44.47 | phearless | but when I press *84.... i got the busy tone |
| 17:44.59 | phearless | I can not press *408 for example |
| 17:45.00 | mercestes | phearless: Anything show up on the CLI? |
| 17:45.01 | Qwell[] | ~lart Idle |
| 17:45.02 | *** join/#asterisk rrocha (n=brain@200.193.155.234) |
| 17:45.20 | Strom_C | phearless: dont' start extensions with * either; that conflicts with vertical service codes |
| 17:45.24 | wunderkin | now you are getting into built-in feature code phone dialplan crap too |
| 17:45.34 | phearless | <mercestes> phearless: Anything show up on the CLI? <--- no |
| 17:45.44 | mercestes | phearless: Using a polycom phone? |
| 17:45.48 | clorabit | Strom_C: what should i do ? |
| 17:45.54 | phearless | <Strom_C> phearless: dont' start extensions with * either; that conflicts with vertical service codes <--- I use * for special features, like direct voicemail |
| 17:45.59 | phearless | mercestes: sipura 942 |
| 17:46.08 | Strom_C | phearless: do you know what vertical service codes are? |
| 17:46.12 | phearless | Strom_C: no |
| 17:46.22 | mercestes | phearless: update your dialplan. You need to match for *xxx and #xxx in order to handle it correctly. |
| 17:46.23 | phearless | Strom_C: my extensions are 4XX and 5XX |
| 17:46.23 | Strom_C | Qwell[]: "cutoff on disconnect" sounds like it might be disconnect supervision |
| 17:46.40 | Strom_C | http://nanpa.com/number_resource_info/vsc_assignments.html |
| 17:46.42 | Strom_C | phearless: read that |
| 17:46.50 | *** join/#asterisk asdx (n=diego@200.61.236.33) |
| 17:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:47.07 | mercestes | phearless: Your device is looking at what your dialing, going "FU" and not even bothering to send it to * because it believes it's wrong. I agree wtih your device, but, hey, it's your dialplan, do it the way you wish. |
| 17:47.46 | phearless | ok Strom_C |
| 17:47.59 | phearless | ok mercestes , I have to modify the dialplan on the phone, not on asterisk, right? |
| 17:48.14 | Strom_C | there should be a list of things you are required to read before touching asterisk ;) |
| 17:49.09 | mercestes | phearless: correct. |
| 17:49.24 | pigpen | I have a dtmf tone issue using asterisk 1.2.9, on a PRI with polycom phones....seems I here a weak dtmf with a skip (like this: "b..b" where the tone should be "BBBB") |
| 17:49.31 | mercestes | Strom_C: lol. Linux for dummies would be a godo start. |
| 17:49.33 | FuriousGeorge | if you guys had to install a video-sip phone in a conference room, what kind of camera would you use? |
| 17:49.46 | dhill | so fax doesn't work on 1.4 at all.. not even with g711ulaw |
| 17:49.58 | pigpen | I am only running echocancel=yes and relaxdtmf=yes |
| 17:49.58 | mercestes | gah, I typo so much. I need to get back into programming so it matters again. I used to never typo. |
| 17:49.58 | dhill | 1.2 with same config works everytime |
| 17:51.30 | phearless | on http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Introduction it is written : "The Dialplan constists of collection of contexts. These context definitions are the most important part of the extensions.conf file and are the most important part of Asterisk configuration. " so the dialplan is in extensions.conf ? |
| 17:51.53 | phearless | but I have already seen a dialplan config on some phones |
| 17:51.58 | phearless | so I am confused |
| 17:52.19 | mercestes | phearless: For asterisk. your device has it's own internal "dialmap" as well, which could be called a "dialplan" that determines when it should collect digits and when it shoudl pass those digits onto it's supervisory device. |
| 17:53.00 | mercestes | phearless: Ever notice how if you dial 6 numbers your phone just sits there forever, and then goes, "beep beep BEEP! You are retarded. Please dial a valid number" but if you dial a full seven digits it "just goes?" |
| 17:53.02 | phearless | okay so in my case I have to define on the phone when the phone has to send the "numbers" |
| 17:53.17 | phearless | ok mercestes |
| 17:53.21 | mercestes | phearless: The secret to that "magic" is the phone "dialmap" that basically sets up valid patterns of numbers to let the deviec know when to "just go." |
| 17:53.22 | phearless | thanks |
| 17:53.27 | sevard | mercestes: hahahaha |
| 17:53.27 | phearless | okay |
| 17:53.28 | phearless | I see |
| 17:53.34 | phearless | why is this funny? |
| 17:53.49 | mercestes | phearless: *your* device is setup to not send any digits at all until you dial a correct series of digits, and what is considered "correct" is controlled by you in that digitmap in the config. |
| 17:54.26 | clorabit | Strom_C: my workaround is add noload=>codec_speex.so is this can cause other problem ? |
| 17:54.28 | mercestes | phearless: It's funny because...most of us have gotten "the call" where "the phone system is broken" and after 2 hours of troubleshooting we find out that they are dialing an invalid number. |
| 17:54.34 | phearless | ok I will investigate to find the dialplan/dialmap on the phone .. |
| 17:54.36 | phearless | okay |
| 17:54.45 | mercestes | phearless: So most of us have thought to ourselves, "you are retarded....learn how to dial and stop wasting our time." |
| 17:54.49 | *** join/#asterisk mrempire (n=trefpunt@mrempire.demon.nl) |
| 17:55.05 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 17:55.07 | mercestes | phearless: So the idea of a pre-recorded message going "you are retarded" instead of "The number you have dialed...." is a riot. |
| 17:55.27 | phearless | lol :) |
| 17:55.27 | *** join/#asterisk ezw` (n=Ez@c66.203.210-59.clta.globetrotter.net) |
| 17:56.25 | wunderkin | pigpen, not an expert on this but you probably don't want relaxdtmf on a pri, you may want to show us your zapata.conf, also no rxgain or txgain, umm there are some other things |
| 17:56.33 | phearless | (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 17:56.34 | phearless | got it |
| 17:57.14 | mercestes | phearless: Looks like it. |
| 17:57.17 | pigpen | wunderkin, yeah...the gain is set low to combat echo....but I plan to change the echo alogrythym to take care of that.... |
| 17:57.33 | pigpen | k..I will disable relaxdtmf and give it a shot. |
| 17:57.45 | phearless | okay thanks for the help |
| 17:57.48 | pigpen | I have had this prior with a bad echo cancel module on a 4 port PRI.... |
| 17:57.55 | pigpen | but not hardware modules here. |
| 17:58.11 | phearless | so, for special features, should I use *number or #number ? |
| 17:58.11 | Idle | Qwell[]: you suck |
| 17:58.23 | phearless | for example to listen to my voicemail *1 |
| 17:58.27 | phearless | is it ok? |
| 17:58.40 | *** join/#asterisk sharp (n=sharp@c-68-46-30-7.hsd1.pa.comcast.net) |
| 17:58.52 | phearless | I have seen the list of vertical things in the config of my phone so I know which ones to avoid |
| 17:59.07 | mercestes | phearless: It *can* be if you do a *xxxT and set a timeout. Or *xT. And don't use any *xx codes at all so you wont' be in conflict with Nanpa. |
| 17:59.15 | *** part/#asterisk frc11 (n=fribelle@212.145.178.3) |
| 17:59.35 | mercestes | phearless: The downside is, somewhere, you are going to end up with a 3 second wait state on a valid * code, either on *x, or *xx. |
| 17:59.59 | mercestes | phearless: unless you set the wait state down to 1 second instead of 3 seconds....but then old ppl will complain because they can't dial that fast. |
| 18:00.00 | phearless | okay |
| 18:00.02 | Strom_C | phearless: if you actually bother to read the vertical service code document I linked you to, you'll see that there are six codes reserved for you to assign as you see fit |
| 18:00.30 | mercestes | I've had old ppl complain that six seconds is too fast |
| 18:00.42 | phearless | Strom_C: right, I missed the 6 last lines ! |
| 18:01.05 | Strom_C | you didn't read the document |
| 18:01.06 | Strom_C | lame |
| 18:01.14 | mercestes | lol |
| 18:01.20 | phearless | but *94 ... *99 seems a bit complex, *1 ... *10 is easier |
| 18:01.33 | phearless | but I understand the timeout problem |
| 18:01.48 | sevard | NANP, BEOCH. |
| 18:02.03 | phearless | no I did not read each line of the doc .... 8-| |
| 18:02.09 | *** join/#asterisk inv_arp[work] (i=root@c-75-74-183-191.hsd1.fl.comcast.net) |
| 18:03.29 | phearless | NAMP = North American Numbering Plan |
| 18:03.32 | phearless | but I am in UK |
| 18:03.42 | phearless | it is used here too? |
| 18:05.18 | yatesy | heh i knew there was a reason why i didn't reconise that listing |
| 18:05.27 | *** join/#asterisk poppo (n=adas@S0106004063d8e527.ed.shawcable.net) |
| 18:05.38 | poppo | Anybody developing asterisk with rubyonrails ??? |
| 18:05.42 | phearless | http://en.wikipedia.org/wiki/North_American_Numbering_Plan |
| 18:05.54 | *** join/#asterisk Nobbie (n=corne@wbs-196-2-122-90.wbs.co.za) |
| 18:05.57 | Nobbie | damn what an awful day i've had after upgrading to * 1.4.0. SIP registrations failing, calls not being initiated, core dumps *argh* |
| 18:06.04 | phearless | poppo: http://anarchogeek.com/assets/2006/10/25/integrating_asterisk_and_rails_astricon_06.pdf |
| 18:07.19 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 18:08.42 | JunK-Y | 13:06 < Nobbie> damn what an awful day i've had after upgrading to * 1.4.0. SIP registrations failing, calls not being initiated, core dumps *argh* |
| 18:08.47 | *** join/#asterisk inv_arp[work] (i=root@c-75-74-183-191.hsd1.fl.comcast.net) |
| 18:08.52 | JunK-Y | did ya report these core dumps? |
| 18:08.58 | poppo | phearless: You have experience with this? |
| 18:09.24 | Nobbie | JunK-Y: 2 of them are 2GB |
| 18:09.43 | JunK-Y | read ur backtrace.txt in ur doc/ dir |
| 18:09.49 | Nobbie | JunK-Y: plus i don't have the DONT_OPTIMIZE and THREAD_DEBUG flags set, only read about that afterwards |
| 18:09.58 | JunK-Y | and report to bugs.digium.com if isnt there yet. |
| 18:10.04 | phearless | phearless: no but I have read this pdf a few days ago and it looks like a very good start for your question |
| 18:10.15 | *** join/#asterisk Aurs (n=Aurs@81.191.123.189) |
| 18:10.46 | pigpen | Why am I getting this: Jan 24 12:10:26 NOTICE[22016]: channel.c:1904 ast_read: Dropping incompatible voice frame on Local/5646121@from-sip-d4a7,2 of format ulaw since our native format has changed to slin |
| 18:11.08 | Nobbie | JunK-Y: you think they'll accept the backtrace's if i don't have the required Defines ? |
| 18:11.13 | pigpen | I have no slin defined. |
| 18:11.22 | pigpen | ulaw only. |
| 18:11.33 | JunK-Y | what do you have in ur bt full and thread apply all bt? |
| 18:11.35 | pigpen | I will get 300 of these from time to time. |
| 18:11.41 | Aurs | pigpen: transcode=no in asterisk.conf, i think |
| 18:11.44 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:11.47 | wunderkin | nobbie, no |
| 18:11.52 | Aurs | or transcode_to_slin=no or something |
| 18:12.08 | CrazyTux | can I set the hostname, of a DB to connect to in asterisk? |
| 18:12.11 | CrazyTux | using odbc? |
| 18:12.12 | Nobbie | JunK-Y: it seems pretty useful, function names, parameter names and values |
| 18:12.19 | pigpen | Aurs, thanks...I will google... |
| 18:12.29 | JunK-Y | sure, report it. |
| 18:12.40 | Nobbie | 11 core dumps, do i report it as 11 bugs ? |
| 18:12.47 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 18:13.04 | JunK-Y | no! |
| 18:13.08 | JunK-Y | just once. |
| 18:13.24 | JunK-Y | they are probably cause by the same thing, no? |
| 18:13.33 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 18:14.17 | Nobbie | the common part seems to be dummy_start(), but before that, there are all different functions |
| 18:14.50 | pigpen | transcode_via_sln=no |
| 18:15.41 | JunK-Y | nobbie: attach the infos requested in backtrace.txt for each coredumps (if segfault occured in different place) |
| 18:16.14 | *** join/#asterisk unik-rados (n=rados@c-68-62-71-239.hsd1.mi.comcast.net) |
| 18:18.52 | poppo | Is it possible to enter speak to txt on a call file? |
| 18:19.29 | *** join/#asterisk znoG (n=gs@97-228-126-200.fibertel.com.ar) |
| 18:19.48 | pigpen | so if a call was trying to transcode to slin from ulaw, could a call get borked? |
| 18:20.05 | *** join/#asterisk Ebola (n=Ebola@host81-151-91-139.range81-151.btcentralplus.com) |
| 18:20.41 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:20.56 | JunK-Y | pigpen: no. |
| 18:21.03 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
| 18:21.06 | JunK-Y | do a show translation, thats really cpu-eater. |
| 18:21.13 | JunK-Y | (not) |
| 18:21.20 | pigpen | k. |
| 18:21.32 | pigpen | I just have been getting annoyed seeing it. |
| 18:22.19 | fiber0pti | Is there any open source software that can make asterisk similar to vonage with a web front end and ability to edit functionality per subscriber? |
| 18:22.38 | Qwell[] | fiber0pti: the gui should be able to give you some basic functionality for that |
| 18:22.54 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 18:24.54 | fiber0pti | Qwell, sure. But I'm looking for more advanced functionality that can more mimic vonage. Like checking voicemail and changing call forwarding. |
| 18:25.11 | Qwell[] | So, you want to be the next vonage? |
| 18:25.23 | Qwell[] | step 1: Spend more than you make each quarter on advertising. |
| 18:25.25 | fiber0pti | Qwell, not at all.. but I want to provide some of that functionality |
| 18:25.27 | Qwell[] | That one is quite easy |
| 18:26.09 | pigpen | fiber0pti, many have written their own interfaces to meet their needs. |
| 18:26.29 | pigpen | We have written a gui to meet our needs, however it will more than likely not meet yours. |
| 18:26.55 | Qwell[] | and that's the problem with GUIs.. everybody needs something different |
| 18:27.07 | pigpen | as...probably most if all the pubilcly interfaces out there won't be to your liking either. |
| 18:27.28 | pigpen | Qwell[], agreed. |
| 18:28.18 | CrazyTux | [Jan 24 12:28:43] WARNING[28764]: res_odbc.c:511 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=2013 [unixODBC][MySQL][ODBC 3.51 Driver]Lost connection to MySQL server at 'reading initial communicatio |
| 18:28.25 | CrazyTux | Has anyone ever ran into that before? |
| 18:28.59 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 18:29.21 | pigpen | sorry, postgresql here...was it working prior? |
| 18:29.25 | pigpen | or new setup? |
| 18:29.30 | *** join/#asterisk svenna_ (n=svenna@p548D2A83.dip0.t-ipconnect.de) |
| 18:29.32 | wunderkin | 2. don't give out your business plan, 3. ???, 4. profit |
| 18:29.44 | CrazyTux | pigpen, new setup. |
| 18:29.47 | *** join/#asterisk rene- (n=rene-@200.34.66.137) |
| 18:29.50 | CrazyTux | pigpen, mysql |
| 18:29.50 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 18:29.54 | wunderkin | 5. buy some underpants |
| 18:30.52 | pigpen | CrazyTux, sounds like it is a basic setup issue, I would review the docs....ensure all requirements are met. |
| 18:31.10 | CrazyTux | pigpen, I'm trying to connect to a remote database |
| 18:31.15 | CrazyTux | I think I see whats wrong... |
| 18:33.19 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 18:37.30 | CrazyTux | rene-, lol |
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| 18:42.19 | *** part/#asterisk l2cache (n=ghansen@64.128.254.98) |
| 18:42.51 | rene- | ~seen [TK]D-Fender |
| 18:43.18 | jbot | [tk]d-fender <n=joe@64.235.216.2> was last seen on IRC in channel #asterisk, 13h 27m 9s ago, saying: 'wunderkin : u'. |
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| 19:01.04 | Idle | ~idle |
| 19:01.05 | jbot | Idle is a doodie head |
| 19:01.17 | Idle | ah |
| 19:02.38 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 19:06.46 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 19:10.41 | *** join/#asterisk clive- (n=pirch@dsl-242-131-152.telkomadsl.co.za) |
| 19:12.17 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 19:14.52 | mercestes | ~mercestes |
| 19:14.54 | jbot | it has been said that mercestes is is the almighty dark overlord. Worship him! Worship or lament and suffer! All hail Mercestes! Dark lord of existance. Mercestes is also my Evil Twin! |
| 19:15.13 | mercestes | wait.. |
| 19:15.14 | syzygyBSD | hi |
| 19:15.17 | mercestes | when that last part get in there? lol |
| 19:15.32 | mercestes | hi wysiwyg. |
| 19:15.37 | syzygyBSD | lol |
| 19:15.49 | *** join/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 19:17.11 | *** join/#asterisk infernix (n=nix@spirit.infernix.net) |
| 19:21.01 | *** part/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 19:22.23 | xpot | anyone know how to write to a db using func_odbc? |
| 19:25.29 | sweeper | wow, there seems to be a lot of angst between openpbx and asterisk XD |
| 19:26.24 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 19:26.33 | *** join/#asterisk nighty- (n=nighty-@66-163-28-100.ip.tor.radiant.net) |
| 19:26.53 | nighty- | hi |
| 19:27.15 | nighty- | I have a problem with mpg123 0.59s-r11 playing the moh too loud (distorted) |
| 19:27.23 | nighty- | anyone knows how to fix this ? |
| 19:30.46 | aydiosmio | does mpg123 not have a gain setting? |
| 19:31.33 | nighty- | aydiosmio: I believe this is only for hardware |
| 19:32.06 | nighty- | aydiosmio: is the zaptel board considered a hardware ? |
| 19:32.07 | *** join/#asterisk shy__guy (i=jeremy_g@c213-100-17-43.swipnet.se) |
| 19:32.54 | aydiosmio | what is only for hardware? |
| 19:33.05 | nighty- | aydiosmio: the -g |
| 19:33.23 | nighty- | aydiosmio: it sets the gain of the hardware I believe |
| 19:33.37 | nighty- | aydiosmio: the audio chip |
| 19:33.40 | aydiosmio | well, you could certainyl try it, couldn't you? |
| 19:33.46 | nighty- | aydiosmio: I tried |
| 19:33.52 | nighty- | aydiosmio: that is why I say this |
| 19:33.53 | aydiosmio | good deal |
| 19:34.05 | aydiosmio | maybe someone else can help you then |
| 19:34.12 | nighty- | aydiosmio: can't be sure though |
| 19:36.37 | nighty- | aydiosmio: I hope someone can help me , as I have researched google and the only thing I found what that 0.59s-r11 seems to be borked according to one person |
| 19:36.55 | nighty- | s/what/was/ |
| 19:37.21 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-21-116.red.bezeqint.net) |
| 19:37.38 | tzanger | hmm |
| 19:37.41 | tzanger | I ownder who I pissed off |
| 19:37.42 | tzanger | 13:53 [freenode] -ChanServ(ChanServ@services.)- You have been deleted from the access list for [#openpbx] |
| 19:37.44 | wunderkin | nighty-, i believe there is a problem using that version, is there a reason you are using mpg123? |
| 19:37.51 | *** join/#asterisk anthonyl (n=anthonyl@72.146.48.234) |
| 19:38.00 | nighty- | wunderkin: what else can I use ? |
| 19:38.10 | wunderkin | if you are using 1.2 or higher, use native |
| 19:38.11 | Corydon-w | 0.59r or native |
| 19:38.19 | nighty- | wunderkin: ok |
| 19:38.24 | nighty- | Corydon-w: thanks :) |
| 19:38.27 | nighty- | I'll try this |
| 19:40.32 | *** join/#asterisk topping (n=topping@204.152.96.50) |
| 19:40.37 | mercestes | tzanger: Lol |
| 19:40.50 | mercestes | I think RoyK is the openpbx bitch. Probably him. |
| 19:40.55 | mercestes | wonder if I'm deleted yet. |
| 19:41.03 | mercestes | nope, I'm not. |
| 19:41.03 | mercestes | :D |
| 19:42.50 | *** join/#asterisk RoyK (n=roy@ti211310a080-5551.bb.online.no) |
| 19:42.50 | nighty- | wunderkin: do you know where I can get the native moh files ? |
| 19:43.15 | nighty- | I am using 1.2.13 |
| 19:45.05 | *** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 19:45.35 | Hmmhesays | good afternoon |
| 19:46.35 | c4t3l | ~c4t3l |
| 19:46.43 | Hmmhesays | ~hmmhesays |
| 19:46.45 | jbot | extra, extra, read all about it, hmmhesays is not really here... |
| 19:46.52 | c4t3l | darmit! |
| 19:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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| 19:51.53 | mercestes | c4t3l: I could teach him if you want..;) |
| 19:57.25 | Hmmhesays | this episode of ds9 is a good one |
| 19:57.52 | Hmmhesays | the one where playdoh and kira finally hook up |
| 19:58.47 | yatesy | playdoh? :P |
| 19:58.56 | Hmmhesays | odo |
| 20:00.04 | yatesy | well yea, why playdoh?! |
| 20:00.24 | *** part/#asterisk CrossRoad (n=SilentVa@207.47.18.18.static.nextweb.net) |
| 20:01.42 | Hmmhesays | it was a joke |
| 20:01.54 | *** join/#asterisk Ebola (n=Ebola@host81-151-91-139.range81-151.btcentralplus.com) |
| 20:02.50 | *** part/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 20:03.05 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 20:07.20 | sweeper | zomg yate |
| 20:07.24 | sweeper | a TRAITOR |
| 20:07.49 | yatesy | just thought it was a bit random Hmmhesays! |
| 20:08.26 | *** join/#asterisk ucfMethod (n=ucfmetho@c2.efb7d1.client.atlantech.net) |
| 20:14.20 | CrazyTux | I'm having the hardest problem to get asterisk -> odbc, to connect to remote mysql server, any suggestions / help? |
| 20:14.46 | Corydon-w | Have you installed MyODBC? |
| 20:14.52 | CrazyTux | Corydon-w, yes. |
| 20:15.02 | CrazyTux | Corydon-w, Error SQLConnect=-1 errno=2013 [unixODBC][MySQL][ODBC 3.51 Driver]Lost connection to MySQL server at 'reading initial communicatio |
| 20:15.09 | Corydon-w | Have you configured /etc/odbcinst.ini and /etc/odbc.ini ? |
| 20:15.17 | CrazyTux | Corydon-w, yes. |
| 20:15.31 | CrazyTux | Corydon-w, I setup the DSN, etc, and referenced it in ref_odbc.conf |
| 20:15.42 | CrazyTux | res* |
| 20:15.51 | Corydon-w | CrazyTux: sounds like you're not permitted to connect from that host |
| 20:15.58 | Corydon-w | CrazyTux: or a firewall rule |
| 20:16.04 | CrazyTux | Corydon-w, thats what I was thinking |
| 20:16.33 | CrazyTux | Corydon-w, however I have replication setup between these two servers, and that works, so I don't see? |
| 20:16.59 | CrazyTux | However I try and connect from cli, mysql -u user -pPASS -h HOST and it connec. refuses |
| 20:17.01 | Corydon-w | CrazyTux: replication uses a different permission |
| 20:17.06 | CrazyTux | however I don't see how that is possible with replication |
| 20:17.09 | CrazyTux | Corydon-w, really? |
| 20:17.16 | Corydon-w | Yes |
| 20:17.18 | CrazyTux | Corydon-w, How can I go in and fix this. |
| 20:17.27 | CrazyTux | Corydon-w, In the mysql settings I have it to permit % |
| 20:17.29 | CrazyTux | hostname |
| 20:17.54 | Corydon-w | GRANT ALL PRIVILEGES ON . TO root@'%'; |
| 20:17.59 | Corydon-w | GRANT ALL PRIVILEGES ON asterisk.* TO root@'%'; |
| 20:18.10 | Corydon-w | or whatever your database is |
| 20:18.26 | CrazyTux | Corydon-w, ok actually, I just got rid of that theory |
| 20:18.32 | CrazyTux | Corydon-w, I connected from CLI with root mysql.... |
| 20:18.33 | *** join/#asterisk UVSoft (n=UVSoft@c7204-ge2-500.etelecom.ru) |
| 20:18.37 | CrazyTux | Corydon-w, permissions granted, etc.. to the DB |
| 20:18.52 | CrazyTux | Corydon-w, now I referenced the root information to connect via ODBC, same error. |
| 20:19.11 | Corydon-w | CrazyTux: try: isql -vvvv asterisk |
| 20:19.24 | *** join/#asterisk gammacoder (n=chatzill@64-132-192-33.static.twtelecom.net) |
| 20:19.44 | *** part/#asterisk gammacoder (n=chatzill@64-132-192-33.static.twtelecom.net) |
| 20:20.08 | CrazyTux | Corydon-w, [S1T00][unixODBC][MySQL][ODBC 3.51 Driver]Can't connect to MySQL server on '74.52.58.50' (111) |
| 20:20.11 | *** join/#asterisk gammacoder (n=chatzill@64-132-192-33.static.twtelecom.net) |
| 20:20.56 | CrazyTux | Corydon-w, however mysql -u root -pPASS -h HOST works... ? |
| 20:21.06 | Corydon-w | CrazyTux: 111 is connection refused |
| 20:21.21 | CrazyTux | Corydon-w, why would the mysql CLI work than? |
| 20:21.24 | Corydon-w | CrazyTux: on the same host? |
| 20:21.27 | CrazyTux | yes. |
| 20:21.37 | Corydon-w | Dunno |
| 20:21.38 | CrazyTux | Corydon-w, only difference would be the hostname |
| 20:21.42 | *** join/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com) |
| 20:21.52 | ctooley | I'm having some issues building zaptel on CentOS-4 |
| 20:21.53 | Corydon-w | CrazyTux: so change the hostname in odbc.ini |
| 20:22.15 | ctooley | WARNING: Error inserting zaptel (/lib/modules/2.6.9-42.0.3.ELsmp/extra/zaptel.ko): Invalid module format |
| 20:22.38 | Corydon-w | ctooley: version skew |
| 20:22.41 | CrazyTux | Corydon-w, hmm apparently it was having a problem resolving the hostname, that fixed it. |
| 20:22.52 | CrazyTux | Corydon-w, thankyou :) |
| 20:23.00 | Corydon-w | ctooley: your source and your kernel are probably not exactly the same string |
| 20:23.06 | Juggie | ctooley, do a uname -a |
| 20:23.08 | Juggie | paste it here. |
| 20:24.26 | UVSoft | hi there, there's a problem, i've got an FXS device, my dialplay looks like: 1. Answer() 2. Read(..) 3. Dial(..) and so on, so when i hang off i hear absolutely nothing, no dial tone, and asterisk offers me to dial the number (answer(), read()), so the question is how to make asterisk play dial tone before user press first button as the real telephone station does? |
| 20:24.48 | Corydon-w | UVSoft: sounds like a dead board |
| 20:25.00 | UVSoft | what do you mean |
| 20:25.11 | Corydon-w | unless you've set immediate=yes |
| 20:25.33 | Corydon-w | immediate=yes turns the phone into a hotline |
| 20:25.46 | CrazyTux | Corydon-w, have you any experiene with setting up asterisk voicemail routed through another softswitch (proxy) ? |
| 20:26.04 | Corydon-w | Nope |
| 20:26.39 | UVSoft | Corydon-w: thanks, I'll check it |
| 20:27.22 | Corydon-w | UVSoft: most fxs devices, you want immediate=no, which means offer dialtone and wait for the user to dial an initial extension |
| 20:27.42 | *** join/#asterisk gammacoder (n=chatzill@64-132-192-33.static.twtelecom.net) |
| 20:28.24 | *** join/#asterisk jaxxan (n=jaxxan@202.70.125.125) |
| 20:28.30 | UVSoft | as far as i remember i set it to yes) |
| 20:28.40 | jaxxan | hey guys |
| 20:29.42 | jaxxan | is there a console command i can issue to reset a voicemail password ? |
| 20:30.58 | UVSoft | and the second question, i've got another device, it's FXO one, after the dial is completed, asterisk (zaptel?) doesn't hang on for a long time... so who is responsible for it? and why can it be? |
| 20:32.02 | Corydon-w | jaxxan: no, but you can reset the password in voicemail.conf and issue a 'reload app_voicemail.so' |
| 20:32.25 | jaxxan | i'm trying to automate changing voicemail passwords so i dont get bothered |
| 20:32.27 | Corydon-w | UVSoft: it doesn't do what? |
| 20:32.37 | Corydon-w | jaxxan: use realtime |
| 20:32.47 | jaxxan | i haven't heard of that |
| 20:33.03 | Corydon-w | jaxxan: see the doc/ directory |
| 20:33.17 | jaxxan | k |
| 20:33.22 | variable_office | anyone using broadwing/level 3 for origination? |
| 20:33.33 | variable_office | (or termination) |
| 20:34.37 | mercestes | variable_office: Not atm but I have before. What's your question? |
| 20:34.50 | UVSoft | Corydon-w: it doesn't hand up |
| 20:35.05 | Corydon-w | UVSoft: are you using kewlstart? |
| 20:35.09 | UVSoft | yep |
| 20:35.11 | variable_office | mercestes whats the quality like? and does t.38 workk well with them? |
| 20:35.31 | Corydon-w | UVSoft: then your provider is probably not providing remote disconnect supervision |
| 20:35.45 | Corydon-w | UVSoft: other than bugging your provider, there's very little you can do |
| 20:38.22 | UVSoft | there's the _context_ options in zapata.conf, is it possible to seperate FXO and FXS devices between different contexts? |
| 20:40.36 | jaxxan | that's not exactly what i want to do though |
| 20:41.14 | jaxxan | but i see the power in it |
| 20:41.48 | anonymouz666 | anyone in here uses Nokia E61? |
| 20:41.55 | Corydon-w | jaxxan: well, you could write a parser for that file, if you liked |
| 20:42.25 | Corydon-w | jaxxan: and then rewrite the file. That's what changing voicemail passwords does if you're not using realtime |
| 20:43.27 | jaxxan | right now i have an oracle server that remotely connects, echo's a new voicemail line to voicemail.conf and then issues asterisk -rx 'reload' |
| 20:44.17 | simplexio | anonymouz666: i have one |
| 20:44.20 | jaxxan | i'd kill for an asterisk -rx 'voicemail extension password ####' command right about now haha |
| 20:44.21 | variable_office | mercestes i guess what was your experiene with them? |
| 20:44.28 | *** join/#asterisk neddy (n=js152033@192.18.43.225) |
| 20:45.27 | *** join/#asterisk |Rain| (i=rain@2001:440:eeee:fffb:42:0:0:2) |
| 20:45.38 | anonymouz666 | simplexio: can you register the phone to asterisk? I think this firmware is buggy |
| 20:45.49 | jaxxan | let me ask you a question, right now i have about 18,000 voicemail users, do you think moving all of that to realtime would be a better solution overall ? |
| 20:45.55 | simplexio | anonymouz666: yes |
| 20:46.12 | simplexio | anonymouz666: you using it throug wlan or 3g or edge ? |
| 20:46.22 | UVSoft | Corydon-w: could i ask you just have a look an my last question and tell me if that is possible or not? thanks |
| 20:46.32 | anonymouz666 | simplexio wireless |
| 20:46.58 | Corydon-w | UVSoft: yes |
| 20:46.58 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:46.58 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:46.58 | UVSoft | Corydon-w: so how to do it? |
| 20:47.13 | simplexio | anonymouz666: no problems with wlan here, few problem over 3g and edge but thats another story |
| 20:47.43 | anonymouz666 | how can I upgrade the firmware? |
| 20:47.43 | Corydon-w | UVSoft: you just described exactly how to do it |
| 20:48.03 | UVSoft | Corydon-w: did I? |
| 20:48.52 | simplexio | anonymouz666: http://newlc.com/Using-SIP-with-Nokia-Series60-and.html that one is used as ecample to put config to e61 |
| 20:50.09 | simplexio | anonymouz666 tcpdump -i eth0 -n -s0 -vv port 5060 show sip traffic on asterisk. do you even get traffic there ? |
| 20:50.20 | UVSoft | Corydon-w: so that option isn't for all the devices, is it? it's just for those which are after it... right? |
| 20:50.32 | Corydon-w | Correct |
| 20:50.34 | |Rain| | does anyone know of a way to fetch info about each leg of a transfered call (supervised, sip) in the dialplan? I'd settle for having to catch it at hangup time... |
| 20:51.39 | jaxxan | So if i used realtime, i wouldn't be able to edit voicemail.conf anymore huh |
| 20:51.43 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:51.47 | jaxxan | with vi that is |
| 20:52.12 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 20:52.21 | tzanger | hahahaha |
| 20:52.26 | tzanger | pirate day must be coming early |
| 20:52.28 | jaxxan | i'm sooooo not into databases )= |
| 20:52.30 | tzanger | I just typed "arsync" |
| 20:52.50 | simplexio | anonymouz666: far as i know both e61 which i use have "default" firmware |
| 20:56.19 | CrazyTux | Whats the codec that you need licensing for that I believe digum provides? |
| 20:56.36 | high-rez | g729? |
| 20:56.56 | CrazyTux | high-rez, how much bandwidth does that one use? |
| 20:57.04 | *** join/#asterisk lamer9 (n=anthony@175.21.188.72.cfl.res.rr.com) |
| 20:57.06 | lamer9 | ~wiki |
| 20:57.15 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 20:57.22 | high-rez | don't remember off the top of my head. |
| 20:57.35 | lamer9 | ~wikis |
| 20:57.45 | jbot | somebody said wikis was http://www.voip-info.org |
| 20:57.45 | lamer9 | where can I buy a toll free number? |
| 20:58.00 | Strom_C | CrazyTux: 8kbps per channel plus overhead |
| 20:58.11 | CrazyTux | Strom_C, and that is the one with the license fees? |
| 20:58.19 | Strom_C | yes |
| 20:58.38 | CrazyTux | Strom_C, Do you know where I can learn about all of the codecs, how much bandwidth etc, I'm looking at wikis although they provide irrelevant information |
| 20:58.47 | lamer9 | where i can buy a phone number for my asterisk server? |
| 21:00.12 | Strom_C | g711 - 64kbps; g726 - 32kbps; gsm - 13kbps; g729 - 8kbps |
| 21:02.27 | CrazyTux | Strom_C, thank you. |
| 21:02.38 | lamer9 | can someone pls help me? |
| 21:02.49 | Strom_C | lamer9: try teliax if you're in north america |
| 21:02.56 | Strom_C | or call your local telco |
| 21:03.00 | lamer9 | okay |
| 21:03.01 | lamer9 | so |
| 21:03.07 | CrazyTux | Strom_C, ever setup asterisk as voicemail only, which it is routed there through someother means (softswitch) ? |
| 21:03.26 | lamer9 | they can just give me a toll-free number |
| 21:03.28 | CrazyTux | lamer9, icall.net, teliax, a few others. |
| 21:03.30 | sevard | you can route voicemail whereever you want |
| 21:03.40 | Strom_C | CrazyTux: no, but I don't see how that would be any different than any other installation involving voicemail |
| 21:03.41 | lamer9 | i dont want them to host it |
| 21:03.41 | CrazyTux | sevard, I know, but as far as the handling of it with asterisk |
| 21:03.42 | *** join/#asterisk n0n0x (n=n0n0x@201-212-168-53.net.prima.net.ar) |
| 21:03.45 | lamer9 | i want to run it on my own machine |
| 21:03.49 | lamer9 | i just want to buy the number |
| 21:03.51 | lamer9 | and be able to run it |
| 21:04.05 | Strom_C | lamer9: you buy the number and the telco you buy it from becomes the resporg |
| 21:04.13 | Strom_C | they provide service to you at that number |
| 21:04.21 | Strom_C | you can port that number whenever you like |
| 21:04.44 | CrazyTux | Strom_C, how can I debug asterisk extensions I setup, I want it to be realtime integrated into the system? |
| 21:05.02 | Strom_C | CrazyTux: um, ask me a more specific question |
| 21:05.03 | lamer9 | the last time |
| 21:05.06 | lamer9 | i had free outgoing and such |
| 21:05.11 | CrazyTux | Strom_C, heard of ser/openser? |
| 21:05.12 | lamer9 | the only thing i was using was bandwidth |
| 21:05.23 | jaxxan | god that realtime just seems alot easier in the end. |
| 21:05.25 | CrazyTux | Strom_C, I want to route through ser/openser to voicemail asterisk |
| 21:05.34 | *** join/#asterisk J4k3 (i=jsuter@160.sub-70-216-100.myvzw.com) |
| 21:05.47 | CrazyTux | Strom_C, now I have my routing done fine, I just need asterisk (voicemail) to pick it up nopw |
| 21:05.52 | CrazyTux | s/nopw/now/ |
| 21:06.00 | UVSoft | modern telephone exchanges offers following feature: when you're talking to someone, and he hangs up, the exchange plays you a dial tone, and you can dial next number immediately. how to make asterisk do the same? |
| 21:06.01 | jaxxan | thanks for pointing me in the right direction Corydon-w |
| 21:06.21 | Strom_C | CrazyTux: that's simple |
| 21:06.27 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 21:06.31 | Strom_C | you'd do it just as you'd do any other voicemail implementation |
| 21:06.40 | Blackthorn | Hi, are keep-alive packets icmp? |
| 21:07.07 | *** join/#asterisk burus (n=burus@87.248.161.141) |
| 21:07.20 | *** join/#asterisk HushPe (n=arron@219-89-126-250.adsl.xtra.co.nz) |
| 21:07.43 | ucfMethod | can anyone give advice on to best setup DTMF tones, or more precisely how to insure that outbound tones are picked up correctly by receiving parties. I have dtmfmode=rfc2833 and progressinband=yes in sip.conf set |
| 21:08.05 | *** join/#asterisk darby_t (n=tom@aaqc78.neoplus.adsl.tpnet.pl) |
| 21:08.28 | ucfMethod | but people complain that some places pick up keys as duplicates etc, making it hard to use public conference call numbers and credit card balance info services |
| 21:08.46 | *** join/#asterisk docelm0 (n=vircuser@m215e36d0.tmodns.net) |
| 21:08.59 | Strom_C | ucfMethod: describe, in detail, the route the call takes from end-to-end |
| 21:10.12 | ucfMethod | Strom_C: Polycom 501s, LAN, Asterisk 1.2.14, Vitelity Termination using SIP, PSTN |
| 21:10.50 | Strom_C | ucfMethod: pastebin your sip.conf file |
| 21:11.03 | *** join/#asterisk CPSK (n=CPSK@c6.ars.ba.nextra.sk) |
| 21:11.28 | JunK-Y | some1 has already cross-compile * for powerpc ? |
| 21:11.40 | jaxxan | yeah |
| 21:11.55 | jaxxan | there's a mac version out, just google it JunK-Y |
| 21:12.02 | JunK-Y | jaxxan: any problem when compiling? |
| 21:12.17 | JunK-Y | jaxxan: i dont want a mac version, i want cross-compile. |
| 21:12.19 | jaxxan | just make sure you have the developer tools installed |
| 21:12.32 | jaxxan | oh, sorry, powerpc i instantly think mac (= |
| 21:12.36 | CrazyTux | Strom_C, care to give me a hand? :D |
| 21:12.40 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 21:12.45 | CrazyTux | Strom_C, I'm a first time asterisk user |
| 21:12.56 | CrazyTux | Strom_C, well not first time, i've just barely *dabbled* with it. |
| 21:14.15 | ucfMethod | Strom_C: http://rafb.net/p/KmUAUk95.html |
| 21:15.28 | *** join/#asterisk phatmonkey (i=nobody@81.2.121.150) |
| 21:17.01 | Strom_C | ucfMethod: and what about the provider and telephone set entries? |
| 21:17.36 | Blackthorn | Hi, are keep-alive packets icmp? |
| 21:17.44 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 21:18.40 | phatmonkey | calls from my internal phones go through various contexts which set which lines they can call out on by setting a variable with Set(). at the moment i'm using a _X. extension in that context, but it seems to override the _X. extension deep inside another including context that is a "catch all" for outgoing calls |
| 21:19.06 | *** part/#asterisk |Rain| (i=rain@2001:440:eeee:fffb:42:0:0:2) |
| 21:19.14 | phatmonkey | what's the best way to set a variable like that, or is there a better way to set outbound lines for internal phones? |
| 21:19.37 | anonymouz666 | If I have an asterisk in front a legacy PBX (digital) how can I detect in Asterisk if for example peer 8000 (tradional) is calling and will pass through asterisk ? |
| 21:19.41 | phatmonkey | *deep inside another includED |
| 21:19.42 | Strom_C | phatmonkey: um, jesus, stop kludging your dialplan :) |
| 21:20.00 | Strom_C | phatmonkey: learn to use pattern matches and includes |
| 21:20.02 | HushPe | which ways can i force the irq for my digium hardware? my bios doesn't let me do it, is there a way to do it via the kernel? |
| 21:20.33 | phatmonkey | Strom_C, explain? |
| 21:20.57 | Strom_C | phatmonkey: do you understand how to use pattern matches? |
| 21:21.48 | anonymouz666 | If I have an asterisk in front a legacy PBX (digital) how can I detect in Asterisk if for example peer 8000 (tradional) is calling and will pass through asterisk ? explaning better: i will know that theres a call from a tradional system, but how i will know that is 8001 and not 8000 for example? |
| 21:22.07 | anonymouz666 | or I wont be able to know ? |
| 21:22.38 | phatmonkey | Strom_C, i think so...! the wiki is very vague, i've been fiddling with asterisk systems for almost a year now but i still don't fully understand it |
| 21:23.06 | UVSoft | Hey guys! what about a dial tone after hand up on the other side and possibility to dial again immediately? |
| 21:23.14 | phatmonkey | if no patterns are matched in the current context, it goes through the includes |
| 21:24.08 | robin_sz | and .... importantly? |
| 21:24.33 | robin_sz | the matching rules DO NOT work down the current context in the order they appear |
| 21:24.47 | robin_sz | the matches are compiled and then hashed and then ... |
| 21:24.57 | robin_sz | well, the first match may not be what you expect! |
| 21:25.38 | *** join/#asterisk DaveCanoe (n=Dave@CPE000f3d61b549-CM0017ee549a8a.cpe.net.cable.rogers.com) |
| 21:25.41 | robin_sz | includes however ARE tested in the order they appear (obviously matching within the includes follows the same compiled rules as before) |
| 21:26.24 | phatmonkey | Strom_C, i have a bunch of contexts that internal phones go to that decide whether they can make outbound calls and on what medium (PSTN, SIP service etc). for the PSTN options, i then want to set what PSTN line they can call out at the very end of the PSTN context. how can i set a variable there that won't affect anything else? |
| 21:27.17 | wunderkin | fuckin a |
| 21:27.25 | wunderkin | /ignore |
| 21:27.46 | Strom_C | phatmonkey: pastebin your extensions.conf |
| 21:27.48 | phatmonkey | on an unrelated note... are there any plans for better asterisk documentation? the voip-info.org wiki is slow, unorganised and cluttered |
| 21:27.49 | Strom_C | ~pb |
| 21:27.51 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 21:28.18 | robin_sz | !book |
| 21:28.38 | robin_sz | !thebook |
| 21:28.42 | Crescendo | What ports need to be forwarded to the server in order for a WAN Cisco IP phone to work through NAT? |
| 21:29.03 | robin_sz | phatmonkey, there the book, when the bot wakes up and telss you the URL |
| 21:29.14 | phatmonkey | Strom_C, that'll take ages... it's a huge dialplan with loads of passwords |
| 21:29.35 | anonymouz666 | when Interconnecting two PBXs via ISDN the tradional system will send the callerid of a phone behind via ISDN to the asterisk? |
| 21:29.52 | phatmonkey | robin_sz, the oreilly book isn't really in depth or up to date! |
| 21:30.35 | dendrite | ~book |
| 21:30.36 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:30.37 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 21:31.10 | phatmonkey | what would be the best way to specify outgoing lines for internal phones? my way does seem a tad kludgy |
| 21:31.19 | robin_sz | phatmonkey, well, it is more organised ... |
| 21:31.27 | *** join/#asterisk J4k3- (i=jsuter@236.sub-70-216-192.myvzw.com) |
| 21:32.31 | phatmonkey | each phone has the same routing (various money saving routing), I just want to change the outgoing Zap line at the very end. variables with two underscores seemed to be the best way to do that, but I dunno how to set the variable at the start without affecting anything else |
| 21:32.58 | *** join/#asterisk jeffgus (i=[U2FsdGV@polly.zimage.com) |
| 21:33.54 | *** join/#asterisk signius (n=IceChat7@dsl-217-155-69-102.zen.co.uk) |
| 21:35.35 | *** join/#asterisk signius (n=IceChat7@dsl-217-155-69-102.zen.co.uk) |
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| 21:38.42 | *** join/#asterisk meshuga (i=meshuga@cpc3-cdif3-0-0-cust441.cdif.firenet.com.ru) |
| 21:39.15 | jaxxan | hey guys |
| 21:39.42 | phatmonkey | found a better way to do it, nm |
| 21:44.16 | *** join/#asterisk SomeOne1 (n=SomeOne1@pool-71-246-217-72.washdc.fios.verizon.net) |
| 21:44.34 | SomeOne1 | what kind of harware would i need to handle 2000 concurrent calls with no codec transcoding? |
| 21:44.47 | SomeOne1 | preferrably in one box, not two |
| 21:44.58 | SomeOne1 | like quad quadcore's? |
| 21:44.58 | sivana | commodore 64 |
| 21:45.08 | mercestes | I was thinking a high end TRS80 |
| 21:45.25 | SomeOne1 | i'll go with the commodore |
| 21:45.28 | tzanger | heh, heh... http://www.abandonia.com/games/en/331/AsterixOperationGetafix.htm |
| 21:45.47 | rudholm | Strom has a Linksys WRT54 that runs * |
| 21:46.25 | mercestes | rudholm: me too |
| 21:46.30 | *** join/#asterisk cbullock81 (n=cbullock@adsl-068-213-099-052.sip.jan.bellsouth.net) |
| 21:46.41 | cbullock81 | Hello everyone. |
| 21:46.48 | Strom_C | rudholm: http://www.stromcarlson.com/1d2/DSC03416.JPG |
| 21:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:47.05 | mercestes | SomeOne1: Seriously? Any processor, around a gb or RAM (overkill, but still) and SATA or SCSI. To run 2000 concurrent calls with no transcoding doesn't take a magical box to handle it. Running 2000 lines to it could be a trick. |
| 21:47.16 | rudholm | Strom_C: very nice |
| 21:47.49 | rudholm | it seems to have survived the trunk of my car nicely :) |
| 21:47.56 | rudholm | I need to get mine mounted in the den |
| 21:47.57 | cbullock81 | has anyone here used any linksys phones... i'm looking into them, and wanted to get some feedback |
| 21:47.57 | mercestes | SomeOne1: Dont' forget to remix your MoH, voicemail and recordings to ulaw or gsm or whatever you are using. |
| 21:48.12 | phatmonkey | still stuck. how can i send calls to different PSTN lines while having a common set of rules (adding prefixes etc) |
| 21:48.22 | phatmonkey | do I just have to write out all the rules 8 times? |
| 21:48.25 | mercestes | but once you eliminate cnoversions.....it's a smooth sea from there. |
| 21:48.33 | rudholm | Strom_C: phil is going to mail me his whole collection of payphone rate cards. I'm going to scan them and mail them back to him. |
| 21:48.52 | Strom_C | oh sweet |
| 21:49.04 | JunK-Y | Strom_C: nice rj-45 cable . |
| 21:49.05 | rudholm | Strom_C: that way I can just print out whichever one I want with my printer for my phones |
| 21:49.24 | Strom_C | JunK-Y: um, RJ-45 is a connector, not a cable |
| 21:49.33 | rudholm | what kind of cable is RJ-45? |
| 21:49.35 | rudholm | never heard of that |
| 21:49.37 | JunK-Y | ya know what i mean ;) |
| 21:49.42 | Strom_C | do I? |
| 21:49.51 | Strom_C | RJ-45 is an 8 position 8 conductor plug |
| 21:49.54 | Strom_C | er |
| 21:50.03 | Strom_C | plug and jack combination |
| 21:51.33 | Strom_C | rudholm: i have to get a new handset from phil; the one he gave me doesn't terminate in spade lugs like the one already on the phone |
| 21:52.00 | rudholm | Strom_C: I need to figure out how to polish the brushed stainless on my 2C2. it's not obvious what would work since it's not a flat finish. |
| 21:52.11 | rudholm | what kind of handset is it? |
| 21:52.17 | rudholm | is it armor cord? |
| 21:52.20 | rudholm | I didn't look in the box |
| 21:52.27 | Strom_C | it's armor-corded |
| 21:52.33 | rudholm | what kind of termination? |
| 21:52.42 | Strom_C | but instead of spade lugs, it terminates in pin receptors (or whatever those are called) |
| 21:52.43 | CrazyTux | Ok this might be a stupid question, but is there any official documentation on asterisk? |
| 21:53.13 | mercestes | voip-info.org |
| 21:53.23 | Strom_C | rudholm: I imagine a wire brush on the end of a rotating tool like a drill or something would be best |
| 21:53.28 | rudholm | yeah |
| 21:53.30 | CrazyTux | Wait that reminds me, I bought a book on asterisk, lol... brb |
| 21:53.43 | cbullock81 | anyone used linksys phones in here? |
| 21:53.48 | CrazyTux | cbullock81, I use pap2's |
| 21:54.15 | Strom_C | CPSK: how about you just ask a question instead of spamming the channel |
| 21:54.18 | rudholm | Strom_C: I think I might have phil mail me a 2C2 with the Bell logo (and the coin instruction plate). 'cause I really can't decide which I prefer so the clear answer is one of each. |
| 21:54.20 | x86 | cbullock81: i've got some 921's on order |
| 21:54.33 | Strom_C | rudholm: haha, that's one solution |
| 21:54.58 | rudholm | Strom_C: he has to mail me the coin return chute anyway, right? |
| 21:55.04 | cbullock81 | im trying to find out about the 942s. It says you have to upgrade it from a 2 line to 4 line phone... they dont tell you where or how much |
| 21:55.06 | Strom_C | exactly! |
| 21:55.15 | rudholm | Strom_C: might as well put a 2C2 in the box with it. |
| 21:55.50 | rudholm | but I should perhaps sell my NOS 2C2, since I don't need three (and that one has neither the clean coin slot instructions nor the Bell logo) |
| 21:56.06 | rudholm | but it *is* NOS |
| 21:56.08 | rudholm | which is cool |
| 21:56.08 | Strom_C | i thought your NOS one was a 2D2 |
| 21:56.13 | rudholm | nope |
| 21:56.14 | rudholm | 2C2 |
| 21:56.19 | Strom_C | mechanical totalizer? |
| 21:56.24 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com) |
| 21:56.25 | rudholm | it's actually a really good example of the phone |
| 21:56.31 | rudholm | yeah, the little wheel totalizer |
| 21:56.43 | Strom_C | drool. |
| 21:56.44 | rudholm | yeah |
| 21:56.45 | rudholm | exactly |
| 21:57.07 | rudholm | even phil said "wow, that's a nice one you got" when I read him the part numbers off the assemblies inside |
| 21:57.25 | rudholm | well, maybe 3 isn't too many. |
| 21:57.28 | Strom_C | impressing phil is...impressive :) |
| 21:57.36 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 21:57.36 | rudholm | yeah, that says something :) |
| 21:57.38 | Strom_C | you can never have too many phones |
| 21:57.45 | rudholm | maybe I could take four and make a "phone square" |
| 21:57.53 | Strom_C | haha yes! |
| 21:57.54 | phatmonkey | Strom_C, hokay, http://channels.debian.net/paste/5172 |
| 21:57.56 | CrazyTux | Anyone in here have problems with DTMF in asterisk? |
| 21:58.03 | phatmonkey | you probably get the idea of what i'm trying to do |
| 21:58.35 | rudholm | the panel phones are truncated 90-degree triangles, so they would form a proper square |
| 21:59.02 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 21:59.18 | Strom_C | phatmonkey: actually, no, it's clear as mud. |
| 22:00.34 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 22:00.45 | phatmonkey | internal SIP phones will be sent to context internal-outbound-pstn-1. i want a variable set there so at the last line, they're sent to the right context on the other server |
| 22:01.02 | Strom_C | phatmonkey: what exactly is your goal? |
| 22:01.23 | Strom_C | rudholm: my harris can wrench works perfectly for threading the enameled plaques |
| 22:01.30 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-130.mtco.com) |
| 22:01.38 | phatmonkey | Strom_C, assign PSTN lines (which happen to exist on another server) to certain phones |
| 22:01.53 | Strom_C | phatmonkey: actual POTS lines? |
| 22:02.03 | Strom_C | have you not heard of hunt groups? |
| 22:02.13 | phatmonkey | Strom_C, yeah, on a Zap card |
| 22:03.08 | phatmonkey | Strom_C, yes, but I want certain phones to be assigned to certain lines, and it's not practical to do that with contexts alone because prefixes are added/changed for all lines |
| 22:03.29 | Corydon-w | Comments help. A lot. |
| 22:03.42 | Strom_C | phatmonkey: you're doing the asterisk equivalent of duct tape, cardboard, and chewing gum |
| 22:04.07 | phatmonkey | Strom_C, that's why i'm asking what the best way to do it is! |
| 22:04.11 | Qwell[] | Strom_C: when all he really needs is WD-40 |
| 22:04.12 | phatmonkey | the documentation is no help |
| 22:04.18 | Strom_C | phatmonkey: HUNT GROUPS |
| 22:04.27 | Strom_C | and possibly.... |
| 22:04.28 | Qwell[] | make it not move, instead of fixing the hinge so it can move easily |
| 22:04.29 | Strom_C | ~hafc |
| 22:04.30 | jbot | it has been said that hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 22:05.04 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 22:05.14 | JT | it's about time CPSK gets the boot i say |
| 22:05.22 | JT | either trolling or incredibly thick |
| 22:06.21 | phatmonkey | Strom_C, how will hunt groups help me? |
| 22:06.42 | Strom_C | phatmonkey: ok, let me ask you several qualifying questions |
| 22:06.51 | Strom_C | 1. how many telephone sets are you supporting? |
| 22:06.59 | Strom_C | 2. how many telephone lines are you supporting? |
| 22:07.16 | Qwell[] | 90, 2 - in that order |
| 22:07.51 | phatmonkey | 1. 8 2. 8 |
| 22:08.05 | Qwell[] | it's always either a very high ratio, or 1/1 |
| 22:08.32 | Strom_C | phatmonkey: why is it imperative that you have a 1:1 relationship of lines to phones? and furthermore, what role does asterisk play in your setup? |
| 22:08.47 | phatmonkey | *but* two phones are assigned their own lines, the other 6 can use any of the spare lines. i've done that with groups in zapata.conf |
| 22:09.58 | phatmonkey | Strom_C, there were a bunch of PSTN lines on the old PBX, they need to be all hooked up. asterisk allows us to have employees around the world with SIP phones have access to the office lines |
| 22:10.01 | phatmonkey | and call each other and all that |
| 22:10.45 | Strom_C | something's not adding up |
| 22:10.52 | Strom_C | you're a worldwide organization with eight employees? |
| 22:11.24 | *** join/#asterisk jm|laptop (n=jamie@dilbert.jamiem.com) |
| 22:12.02 | robin_sz | makes no sense |
| 22:12.13 | robin_sz | so the first 2 phonbes have a line each ... |
| 22:12.16 | phatmonkey | 8 hardware sip phones in the office, planning to have more connect from around the world who aren't based in the office |
| 22:12.29 | robin_sz | and the last 6 phones have .. err any line from a group of 6 ... |
| 22:12.45 | robin_sz | 6/6 = 1 |
| 22:13.15 | Qwell[] | robin_sz: heh |
| 22:13.40 | mercestes | phatmonkey: What you are trying to do is setup a turnkey system based upon analog switching using a digital software solution. Not happening without a context for EACH line and EACH phone and yoru looking at a MAJOR pain to duplicate something that is officially outdated. |
| 22:13.41 | Qwell[] | 8/8 == 2/2 & 6/6 |
| 22:13.52 | Qwell[] | 8/8 == 1/1 & 1/1 & 6/6, rather |
| 22:14.04 | Strom_C | exactly |
| 22:14.16 | robin_sz | phatmonkey, ok, amaze us. the first two phones must use one perticular line each for waht reason exactly? |
| 22:14.17 | Strom_C | phatmonkey: it's seriously recommended that you hire a consultant at this point |
| 22:14.20 | mercestes | but if you wanted to do it.... |
| 22:14.26 | Corydon-w | What's with August 8th and June 6th? |
| 22:14.34 | wunderkin | 1+1=3 |
| 22:14.40 | mercestes | it would be a context for each PRI and phone you want to "dedicate". |
| 22:15.06 | robin_sz | mercestes, probably better to use the term "restrict" |
| 22:15.16 | mercestes | robin_sz: Exactly. |
| 22:15.24 | phatmonkey | robin_sz, the higher ups need their own phone lines ;) |
| 22:15.30 | phatmonkey | right, this is going to take some rethinking |
| 22:16.10 | mercestes | phatmonkey: And it *still* wouldn't be "turnkey"....it would just act that way because you walled it off that way. |
| 22:16.10 | robin_sz | "need their own lines"? for what reason? |
| 22:16.10 | Strom_C | wait wait! |
| 22:16.10 | Strom_C | I know what he's trying to do! |
| 22:16.11 | Strom_C | he's trying to do "direct inward dial" without actually knowing what it is! :D |
| 22:16.18 | mercestes | Strom_C: LMAO |
| 22:16.20 | robin_sz | ahh, |
| 22:16.38 | Qwell[] | save yourself some trouble, and get a partial PRI |
| 22:16.40 | *** join/#asterisk Ebola (n=Ebola@host81-151-91-139.range81-151.btcentralplus.com) |
| 22:16.45 | Qwell[] | that would completely solve that problem |
| 22:17.04 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:17.34 | robin_sz | phatmonkey, are you talking "need their own lines" outgoing, or incoming? |
| 22:18.28 | puzzled | hi |
| 22:18.54 | mercestes | robin_sz: I think he means both and I believe he thinks it will occupy the same line, as if these phones were running off of dedicated copper and plugging straight into an FXO with an analog "roll over" switch. |
| 22:19.08 | phatmonkey | maybe i should hire a freaking consultant... |
| 22:19.24 | robin_sz | phatmonkey, for outgoin, even then you would not allocate a specific line ... you would just have "non-managemtn can only get a line in the group if there are at leat 2 free" |
| 22:19.34 | mercestes | robin_sz: You know, stuff they used to do in the 60's when they first introduced "transferring" and "hold." Where the operator goes "Call on line 4" and she literally means, line #4 has a call on it and any phone can access it. |
| 22:19.52 | robin_sz | mercestes, yeah, it sounds a bit 1970 |
| 22:19.59 | JT | key systems can still do "call on line #" |
| 22:20.07 | JT | works for a small amount of lines |
| 22:20.40 | robin_sz | phatmonkey, the beauty of asterisk is it is flexible ... that means you can do a lot of the sort of thing you want to do like priority outgoing calls, with less lines than a 1:1 mapping ... |
| 22:20.55 | mercestes | or more than 1:1 mapping. |
| 22:21.00 | robin_sz | well yes |
| 22:21.25 | phatmonkey | robin_sz, these separate lines are just totally removed from everything else - management want their own phone lines with their own numbers which they call out and receive on |
| 22:21.26 | JT | arrrgh |
| 22:21.28 | mercestes | or phones with no mapping....or phones with no lines. |
| 22:21.43 | Grnd-Wire | and you CAN use GOTOIF logic to control which phones are dialing out on which Zap channels.. but it'd be some very scary dialplan code! |
| 22:21.43 | rudholm | is CPSK a cron job? |
| 22:21.46 | mercestes | phatmonkey: So set them up on their own group |
| 22:21.55 | JT | rudholm: or an irc client timer |
| 22:21.55 | robin_sz | phatmonkey, yes ... they get that |
| 22:21.58 | mercestes | one PRI. Chans 1-4 = management lines g1 |
| 22:22.08 | mercestes | Chans 5-23 is everything else, as g2 |
| 22:22.11 | rudholm | JT: that's kind of annoying |
| 22:22.23 | robin_sz | phatmonkey, but "own number in and own number out does NOT , i repeat NOT mean own line, physically |
| 22:22.24 | phatmonkey | having 8 hardware phones and 8 POTS lines is purely a coincidence! there will be more phones |
| 22:22.28 | JT | rudholm: no shit, and he doesn't reply when you ask him what his actual problem is |
| 22:22.37 | rudholm | nice |
| 22:22.41 | phatmonkey | robin_sz, yeah, but that's how it works at the moment |
| 22:22.50 | robin_sz | well, yes |
| 22:22.56 | robin_sz | but thats not an efficient solution |
| 22:23.13 | phatmonkey | robin_sz, they don't want to scrap the POTS lines, they just want a direct replacement for the crummy PBX they have there at the moment |
| 22:23.32 | *** join/#asterisk Skarmeth (n=Skarmeth@201009049189.user.veloxzone.com.br) |
| 22:23.41 | Grnd-Wire | It sounds like what he's asking for is what Avaya refers to as "Private Lines".. |
| 22:23.55 | robin_sz | a bit ... |
| 22:23.56 | Qwell[] | Grnd-Wire: yeah, but that's analog thinking |
| 22:23.56 | Grnd-Wire | So even when you're dealing with line pools - a user can actually have a line appearance on their phone that NOONE else can see.. |
| 22:24.11 | Strom_C | phatmonkey: that's a lot like saying "we want to replace the 1981 dodge omni with a 2007 Lexus, but we want to keep the same tires" |
| 22:24.14 | robin_sz | but its not necessary, as you dont do it that way, you just reserve space inthe pool |
| 22:24.32 | Grnd-Wire | Qwell: meh - When you're trying to communicate with someone, you need to figure out what he's asking for - then convince him there is a better way.. So far, you haven't convinced him you understand what he wants. |
| 22:24.57 | Qwell[] | Grnd-Wire: problem is, users often don't know what they want |
| 22:25.33 | rudholm | JT: yeah, a "/last CPSK" shows something of a pattern. |
| 22:25.56 | phatmonkey | Qwell[], i have absolutely no idea what i need or want now |
| 22:26.01 | J4k3 | is it a normal occurrance for vitelity incoming trunks to return "you have reached a non-working number" unless its an extremely off-peak time? |
| 22:26.22 | phatmonkey | Strom_C, what do you suggest I do then? |
| 22:26.23 | JT | rudholm: :/ |
| 22:26.40 | Strom_C | phatmonkey: I suggest you hire a competent consultant |
| 22:26.41 | robin_sz | phatmonkey, you should consider a multi-channel digital line, either partial PRI or multi BRI ISDN |
| 22:26.50 | Grnd-Wire | Qwell: Yeah, and sometimes they need to be educated enough to understand.. Alot of customers HATE being treated like they're stupid.. So the trick is to understand what they want, and use some analogies .. Then, once you're sure you're describing the samr thing, you go from there.. |
| 22:26.52 | mercestes | phatmonkey Hire a consultant. Or study Asterisk extensively and figure out what it does before you present what improvements it can offer. |
| 22:26.53 | robin_sz | easier to interface |
| 22:27.53 | JT | i see at least 20 of the same messages from CPSK with /last |
| 22:28.13 | rudholm | yeah |
| 22:29.02 | robin_sz | phatmonkey, and then consider everyting as a "pool" ... so long as you keep enough space in the pool for management you are fine ... and incoming can be routed direct to them by dialled number (regardless of which channel it comes in on) and outgoing can have the right CID number for them, regardless of which channel it goes out on |
| 22:29.10 | phatmonkey | i'll hire a consultant in the morning, right now i need to get some sleep |
| 22:29.27 | robin_sz | personally, I always reserve space for sales ... |
| 22:31.23 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 22:32.58 | JT | CTCP VERSION reply from CPSK: mIRC v6.21 Khaled Mardam-Bey |
| 22:33.06 | JT | i guess it's easy to setup timers with mirc |
| 22:33.28 | *** join/#asterisk dseeb_ (n=dcb@58.165.244.192) |
| 22:35.54 | *** join/#asterisk Splat (n=splat@220-253-136-53.TAS.netspace.net.au) |
| 22:37.31 | *** join/#asterisk sharp (n=sharp@c-68-46-30-7.hsd1.pa.comcast.net) |
| 22:38.53 | J4k3 | shit... they didn't tell me I was 80 cents from paying my monthly DID bill |
| 22:39.01 | J4k3 | thats just... ack... damn... scary. |
| 22:39.38 | robin_sz | 80cents from? |
| 22:41.07 | [hC] | perd: you alive? |
| 22:41.39 | sevard | JT: it _is_ very easy to setup timers with mIRC.. buttttt iirc that version is very exploitable and he IS extremly annoying so.... |
| 22:41.45 | sevard | just a suggestion. |
| 22:42.49 | [hC] | to anyone who may be using hylafax as their email-to-fax gateway here (i realize this is an off topic question, sorry) -- how on earth when attaching a pdf do i get it to KEEP the cover page (it does for every other doctype i try, but with pdf the cover page is gone) |
| 22:43.02 | JT | sevard: heh, it would be nice if CPSK took a 'holiday' from here ;) |
| 22:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:47.25 | JT | mercestes: pretty convincing |
| 22:47.29 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 22:47.31 | *** join/#asterisk bhrobinson (n=brobinso@northtx1-static.telwestonline.com) |
| 22:47.32 | mercestes | lol |
| 22:48.28 | bhrobinson | can anyone help me on how to set up a disa to record and email the recordings? |
| 22:50.00 | rudholm | 1 hour roundrip? |
| 22:50.01 | rudholm | that's nothing |
| 22:50.22 | mercestes | I can do that in 45 minutes |
| 22:51.14 | rudholm | I drove from http://www.museumofcommunications.org/ to http://www.yahoo.com/ in one go Monday. |
| 22:51.33 | JT | web addresses aren't very geographic to me |
| 22:51.45 | rudholm | Seattle to Sunnyvale |
| 22:51.46 | mercestes | JT: He meant he typed out both of those addresses all in one monday |
| 22:51.54 | JT | heh |
| 22:52.08 | rudholm | but in fact, I was actually travelling from that museum to yahoo |
| 22:52.50 | JT | bhrobinson: put a MixMonitor before the DISA priority, then umm, work out how to email it :), you could uses system or AGI for that i guess |
| 22:53.00 | Crescendo | What ports need to be forwarded to the server in order for a WAN Cisco IP phone to work through NAT? |
| 22:53.16 | mercestes | bhrobinson: Or a cron job. |
| 22:53.42 | bhrobinson | I am good either way... |
| 22:53.46 | JT | bhrobinson: you probably won't want to save in .wav if you're emailing the recordings, the space usage would be huge and email's not a file transfer protocol |
| 22:53.52 | *** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net) |
| 22:55.00 | *** join/#asterisk crich1999 (n=crich@port-212-202-210-130.dynamic.qsc.de) |
| 22:55.13 | bhrobinson | anyone have a template of how I can do it? |
| 22:55.21 | JT | a template |
| 22:55.27 | JT | are you asking someone to do it for you? |
| 22:55.57 | mercestes | bhrobinson: You got about $250.00 to give me to cover the template? :P |
| 22:59.00 | bhrobinson | wish I did.. |
| 22:59.02 | bhrobinson | :) |
| 22:59.33 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 23:01.14 | *** join/#asterisk rabelais (n=blank@hpolaris.Stanford.EDU) |
| 23:02.21 | Qwell[] | CPSK: enough |
| 23:02.38 | Strom_C | Qwell[]: it's an automated message |
| 23:02.48 | Strom_C | cpsk doesn't respond |
| 23:02.54 | Qwell[] | CPSK: repeat that once more, and you will be removed |
| 23:03.00 | anonymouz666 | hahahahaha |
| 23:03.29 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:06.43 | sevard | YES |
| 23:06.44 | sevard | FINALLY |
| 23:07.00 | sevard | Qwell[]: It's a timer, just remove him. He's been going on for two days. |
| 23:07.40 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 23:07.46 | Qwell[] | next time it happens, point it out to me |
| 23:08.25 | sevard | Qwell[]: if you check your logs you'll see it going at a regular interval for about 48 hours |
| 23:08.38 | Qwell[] | I don't have much scrollback here |
| 23:08.52 | rudholm | I just did a "/last CPSK" |
| 23:08.55 | mercestes | I think you should leave him. He makes a good metronome. |
| 23:08.58 | Qwell[] | and unless you can get me Qwell's IP, I can't ssh into my box at home to check logs :p |
| 23:09.11 | mercestes | ...I can |
| 23:09.11 | sevard | I don't generate logs either, if you grep them for 'Tenovis' though, you'll find lots. |
| 23:09.38 | Qwell[] | mercestes: I'm doubting that :) |
| 23:09.55 | sevard | Qwell[]: you should use Hamachi |
| 23:09.57 | mercestes | hrm |
| 23:10.01 | Qwell[] | sevard: eh? |
| 23:10.09 | sevard | Qwell[]: http://www.hamachi.cc |
| 23:10.16 | JT | if it's a timer, the interval is irregular |
| 23:10.29 | mercestes | Your right....weird hostname |
| 23:10.29 | mercestes | nice |
| 23:10.32 | JT | there might actually be a person there :/ |
| 23:10.38 | rudholm | all I see is 15:07 [freenode] -!- CPSK [n=CPSK@c6.ars.ba.nextra.sk] |
| 23:10.41 | sevard | yeah, how did you do that hostname? that's not a bnc |
| 23:10.51 | Qwell[] | sevard: standard freenode cloak |
| 23:11.00 | sevard | docs? |
| 23:11.00 | Qwell[] | well, project cloak, now |
| 23:11.06 | Qwell[] | RTFFAQ :p |
| 23:11.10 | sevard | :) |
| 23:11.18 | JT | quite bog standard |
| 23:11.18 | JT | :P |
| 23:11.52 | rudholm | c6.ars.ba.nextra.sk is in DNS |
| 23:11.52 | rudholm | that might be it |
| 23:11.57 | rudholm | 195.168.45.78 |
| 23:12.11 | JT | to get a standard cloak, you need to get a second nick, link it, then msg a staffer |
| 23:12.16 | mercestes | Qwell's IP address...not CPSK |
| 23:12.18 | JT | rudholm: talking about Qwell[]'s cloak |
| 23:12.19 | mercestes | QWELLL! |
| 23:12.20 | mercestes | He did it again! |
| 23:12.23 | JT | hi CPSK |
| 23:13.10 | *** mode/#asterisk [+b *!*n=CPSK@*.ars.ba.nextra.sk] by Qwell[] |
| 23:13.10 | *** kick/#asterisk [CPSK!i=qwell@pdpc/sponsor/digium/Qwell] by Qwell[] (constantly repeated messages - likely a scam) |
| 23:13.24 | sevard | you have sexy, firm legs. |
| 23:13.37 | Supaplex | for sure |
| 23:13.44 | sevard | made of blubber. |
| 23:13.50 | JT | bye CPSK |
| 23:14.04 | JT | pabx |
| 23:14.16 | Qwell[] | oh, sure, now he messages me |
| 23:14.16 | JT | it's a Tenovis Integral not a Tenovis Integra |
| 23:14.22 | rudholm | hahaha |
| 23:14.24 | rudholm | of course |
| 23:14.35 | rudholm | Qwell[]: what'd he say? |
| 23:14.39 | JT | saying what CPSK is looking for help with E1 interconnect Asterisk / Tenovis |
| 23:14.40 | JT | <PROTECTED> |
| 23:14.53 | sevard | holy crap, don't paste that message again, fool |
| 23:14.58 | rudholm | haha |
| 23:14.59 | rudholm | yeah |
| 23:15.01 | mercestes | rofl |
| 23:15.24 | *** join/#asterisk fetcher (n=jnh@ip-209-172-35-240.reverse.privatedns.com) |
| 23:15.29 | JT | <CPSK> Qwell[]: can you help with E1 interconnect??!11one |
| 23:15.35 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 23:15.36 | Supaplex | JT now has CPSK cooties |
| 23:16.04 | sevard | Supaplex: got any windex? hit him in the eyes! |
| 23:18.26 | *** mode/#asterisk [-b *!*n=CPSK@*.ars.ba.nextra.sk] by Qwell[] |
| 23:18.54 | Strom_C | sounds like you gave him a stern talking-to |
| 23:19.03 | JT | mr Tenovis is welcome back? :P |
| 23:20.59 | *** join/#asterisk Tili (n=tili@147.Red-88-14-88.dynamicIP.rima-tde.net) |
| 23:21.24 | *** mode/#asterisk [-b *!*n=ldm@*.apartia.fr] by Qwell[] |
| 23:21.35 | *** mode/#asterisk [-bbb *!*n=Limon@85.102.155.* *!*@190.48.132.* *!*@85.98.165.*] by Qwell[] |
| 23:21.51 | *** mode/#asterisk [-b *!n=Mr_DreaM@88.224.160.*] by Qwell[] |
| 23:21.56 | Qwell[] | meh, something like that |
| 23:25.38 | clorabit | helo anyone can help me |
| 23:26.06 | JT | clorabit: what? |
| 23:27.07 | *** join/#asterisk HushPe (n=HushPe@mail.kamar.co.nz) |
| 23:27.25 | clorabit | this is my first time using asterisk, i've install and add username in iax.conf file also add simple dial plan in extension.conf but i still can't connect to server using idefisk any suggestion ? |
| 23:28.02 | JT | yeah, you share what errors you are receiving :) |
| 23:28.39 | clorabit | where i can find error message ? |
| 23:28.51 | SomeOne1 | what kind of harware would i need to handle 2000 concurrent calls with no codec transcoding? |
| 23:28.51 | JT | idefisk must give an error |
| 23:29.11 | JT | SomeOne1: received over what tecnology? |
| 23:29.17 | JT | SomeOne1: a cluster is probably a good idea |
| 23:29.36 | clorabit | JT: when i click register button it's do nothing |
| 23:29.52 | JT | clorabit: have you tried to make calls |
| 23:29.55 | SomeOne1 | JT: gigabit ethernet and SIP |
| 23:30.10 | SomeOne1 | im not sure what you mean "received over what tecnology" |
| 23:30.13 | JT | SomeOne1: sip to sip? |
| 23:30.17 | SomeOne1 | yeah |
| 23:30.20 | JT | well there's POTS, BRI, PRI |
| 23:30.27 | file | carrying audio, or not? |
| 23:30.30 | SomeOne1 | ahh, no, SIP to SIP |
| 23:30.35 | JT | as well as SIP, IAX and H.323 |
| 23:30.44 | SomeOne1 | no POTS |
| 23:30.46 | SomeOne1 | or cards |
| 23:30.48 | SomeOne1 | or anything |
| 23:30.53 | clorabit | JT: yes i do, but acctualy i'm not sure that my additional config at iax.conf and extension.conf is right |
| 23:31.08 | Qwell[] | pfft, 2000 calls with audio is easy |
| 23:31.10 | HushPe | hi JT :) i'm still having problems with assigning an irq, my bios doesn't allow it :( is there something I can do at kernel level? |
| 23:31.25 | JT | clorabit: paste them into pastebin.ca (minus your actual password) |
| 23:31.32 | JT | and tell us the url |
| 23:31.42 | JT | HushPe: have you tried swapping pci slots? |
| 23:31.50 | HushPe | JT: and that :( |
| 23:31.55 | JT | ? |
| 23:32.05 | JT | removing all unnecessary pci cards, too? |
| 23:32.08 | hardwire | anybody else ignited an InterTel phone system lately? |
| 23:32.58 | HushPe | JT: there are no others, i only have 2 slots, one vacant, one with the card in it... i turned off a few onboard things like sound (which was sharing an irq), but it keeps taking irq 22 which i think is a 'virtual' type one |
| 23:33.52 | JT | HushPe: when you disabled sound, was the digium card sharing irq? |
| 23:34.23 | SomeOne1 | Qwell: what hardware? |
| 23:34.31 | SomeOne1 | im thinking dual quadcores 64 bits |
| 23:34.50 | SomeOne1 | two boxes |
| 23:34.52 | Qwell[] | SomeOne1: I was able to get around 2500 (with zero optimizations) on a Sunfire T2000... |
| 23:35.07 | SomeOne1 | one dual quadcore for 1000 calls |
| 23:35.08 | JT | does that run x86? |
| 23:35.17 | SomeOne1 | and another for 1000 |
| 23:35.18 | SomeOne1 | so 2 boxes |
| 23:35.24 | SomeOne1 | of course, the fastest system bus and all that |
| 23:35.26 | Qwell[] | JT: no |
| 23:35.51 | JT | sun hardware usually handles parallelism better than x86 |
| 23:36.02 | SomeOne1 | Qwell: how about in the x86 world? |
| 23:36.15 | clorabit | JT: http://pastebin.ca/327467 |
| 23:36.23 | SomeOne1 | im trying to get a general idea |
| 23:36.25 | HushPe | JT: it was irq 22, but i can't get it lower than that |
| 23:36.27 | SomeOne1 | i've been looking at: http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning |
| 23:36.33 | SomeOne1 | but, people usually have small setups |
| 23:36.35 | SomeOne1 | nothing big |
| 23:36.45 | JT | HushPe: but is anything else using the irq now? |
| 23:36.56 | HushPe | JT: from /proc/interrupts : 22: 389644 755654 IO-APIC-fasteoi wctdm |
| 23:37.12 | JT | HushPe: try booting with the kernel argument noapic |
| 23:37.16 | fetcher | high IO-APIC IRQs are usually 16-23 |
| 23:37.19 | JT | see if it makes a difference |
| 23:37.45 | HushPe | JT: doesn't appear so, but lspci -vb show it's sharing the video card (no vb or x), but really odd none the less |
| 23:37.52 | HushPe | JT: will do, be a few mins |
| 23:38.00 | clorabit | JT: but i just leave other default config is that ok ? |
| 23:38.40 | JT | clorabit: what is the default context in iax.conf? |
| 23:39.30 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 23:41.45 | clorabit | JT: i don't know how to check it ? |
| 23:41.59 | JT | clorabit: is that whole iax.conf? |
| 23:42.04 | JT | that you pasted |
| 23:42.08 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 23:42.15 | clorabit | JT: no |
| 23:42.23 | JT | why not |
| 23:42.36 | clorabit | JT: it just additional from default iax.conf when i install |
| 23:42.52 | JT | i have no idea what your default says |
| 23:43.06 | rene- | hey |
| 23:43.10 | clorabit | JT: i will repaste then |
| 23:43.15 | JT | okay |
| 23:43.23 | rene- | is anyone available for some test calls over g729/sip? |
| 23:43.51 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 23:43.59 | rene- | SomeOne1: how big do you want to go? |
| 23:44.01 | HushPe | JT: call me a little slow, but how do i tell lilo at the boot prompt to turn if off? |
| 23:44.21 | JT | clorabit: just a thought, it may be easier to do initial testing with no authentication |
| 23:44.24 | HushPe | JT: linux noapic irhgt? |
| 23:44.27 | HushPe | right* |
| 23:44.27 | JT | see if the problem doesn't lie elsewhere |
| 23:44.47 | JT | HushPe: if your kernel is called linux, then yes |
| 23:44.56 | JT | umm you may need to specify root and stuff |
| 23:45.06 | JT | been a while since i have used lilo |
| 23:45.11 | JT | i only use grub now |
| 23:46.00 | rene- | SomeOne1: if you dont do a lot of fancy stuff you could use a very simple box and do it with SER |
| 23:46.46 | JT | rene-: it's pushing audio so will still need the I/O grunt |
| 23:46.55 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:46.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:46.57 | rene- | audio playback? |
| 23:47.02 | JT | rtp |
| 23:48.36 | rene- | you mean between the phones ? or do you use something like sipp with prerecorded rtp streams? |
| 23:48.44 | clorabit | JT: how to do that ? |
| 23:49.32 | JT | remove the username and password, and a couple of options need to be added iirc, which i can't remember now |
| 23:49.35 | clorabit | JT: this is contents of my iax.conf http://pastebin.ca/327479 |
| 23:49.38 | JT | pretty sure the book says how |
| 23:49.40 | JT | ~thebook |
| 23:49.41 | jbot | [thebook] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:49.56 | anthonyl | computer != telephone |
| 23:50.48 | rene- | anthonyl: i believe that used to be the case a few years ago because now pretty much any cellphone ~ = computer |
| 23:51.08 | JT | anthonyl: being all philosophical now? |
| 23:52.13 | clorabit | JT: this is my extentions.conf contents http://pastebin.ca/327482 |
| 23:52.20 | SomeOne1 | Qwell: do you think a dual quad core 64bit will be able to handle 2000 calls on SIP? |
| 23:52.24 | SomeOne1 | over gigabit ethernet |
| 23:52.36 | mercestes | SomeOne1: With no transcoding? |
| 23:52.48 | SomeOne1 | no transcoding |
| 23:52.53 | SomeOne1 | what-so-ever |
| 23:53.02 | mercestes | I am pretty certain |
| 23:53.36 | mercestes | 64bit isn't going to help you much tho |
| 23:53.38 | SomeOne1 | 2000 concurrent calls |
| 23:53.51 | SomeOne1 | fastest system bus and everything |
| 23:53.54 | SomeOne1 | avaliable today |
| 23:54.45 | mercestes | won't hurt...but under linux I wouldn't install the 64 bit sources |
| 23:54.56 | mercestes | I'd stil go 32 bit |
| 23:54.59 | mercestes | just for stability |
| 23:55.16 | mercestes | go ahead and spot for the 2-4 gigs of ram, just to be safe. |
| 23:55.48 | SomeOne1 | cool |
| 23:55.50 | SomeOne1 | hmmm |
| 23:55.54 | SomeOne1 | bandwidth wise |
| 23:55.59 | SomeOne1 | what do you think i need |
| 23:56.00 | SomeOne1 | for ulaw |
| 23:56.03 | burus | we use two clustered x86_64 servers with asterisks and it's stable |
| 23:56.14 | clorabit | JT: what is default extention number to dial echo test |
| 23:56.16 | SomeOne1 | no silence suppressioj |
| 23:56.16 | SomeOne1 | n |
| 23:56.38 | *** join/#asterisk florz (n=florz@2002:58c6:2592:1:0:0:0:2) |
| 23:57.12 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 23:57.16 | *** join/#asterisk Daveb21 (n=daveb@eth2235.sa.adsl.internode.on.net) |
| 23:57.33 | HushPe | JT: ok apic off now, seems to be a great deal clearer even though it's sharing with 2 other items (network and something else) |
| 23:57.55 | JT | HushPe: zttest is more important first of all :P |
| 23:58.25 | *** part/#asterisk burus (n=burus@87.248.161.141) |
| 23:58.39 | HushPe | JT: just running now for a few passes, but looking much better |
| 23:59.04 | HushPe | JT: 31 passes = Best: 99.987793 -- Worst: 99.975586 -- Average: 99.975980 |
| 23:59.12 | mercestes | HushPe: ew. |
| 23:59.13 | JT | SomeOne1: for 2000 calls (in and out), 320Mbit/s |
| 23:59.18 | mercestes | HushPe: Is that for idle?? |
| 23:59.22 | JT | err |
| 23:59.26 | JT | that's a fine score mercestes |
| 23:59.26 | HushPe | no calls |
| 23:59.29 | JT | HushPe: excellent |
| 23:59.37 | mercestes | no it's not. |
| 23:59.45 | JT | mercestes: clearly you didn't see the last time he pasted |
| 23:59.47 | JT | yes, it is |
| 23:59.48 | mercestes | that would be great under a moderately load. |
