00:04.40 | thieumS | do you think I can handle 120 g729 SIp to zap transcoding with a bi-Xeon 3Ghz ? |
00:05.08 | thieumS | in the wiki, the guy succeed with 100 channels |
00:07.32 | hardwire | shit |
00:07.35 | hardwire | heh |
00:07.43 | hardwire | how big a machine for just 24 channels |
00:07.52 | hardwire | I think buying a dual xeon system for 96 channels is just insane |
00:08.06 | Grooby | HP IPAQ??!?! |
00:08.10 | thieumS | for transcoding ? |
00:08.12 | Grooby | 613mhz ARM |
00:08.13 | Grooby | :-D |
00:08.21 | stevekstevek | Timex Sinclair... |
00:08.26 | thieumS | it is said transcoding is a CPU killer |
00:08.27 | hardwire | http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-38356249088.htm |
00:08.29 | hardwire | now that would be nice |
00:08.37 | *** join/#asterisk atmel (~vlad@ruxi.dynamic.ucsd.edu) |
00:08.38 | stevekstevek | "near" real-time |
00:08.43 | stevekstevek | for some defintion of "near" |
00:08.45 | hardwire | you could use a 486 with that thing |
00:08.52 | hardwire | behold the power of basic DSPing! |
00:09.08 | hardwire | well |
00:09.13 | hardwire | 240 channels on a 2.0 ghz celeron |
00:09.25 | hardwire | works with *? |
00:10.04 | Nugget | http://bash.org/?464385 <-- heh |
00:10.35 | thieumS | do you mean Digium cards are shit ? |
00:11.14 | atmel | :) |
00:13.16 | wangster | Anyone know why ilbc -> anything is reasonable but anything -> ilbc is 13ms+ ? (according to "show translation") |
00:13.31 | wangster | Is this due to ilbc's built in jitter buffer? |
00:13.44 | *** part/#asterisk Grooby (~Grooby@66.160.105.186) |
00:14.13 | *** join/#asterisk Frantic (~ab@24-193-46-85.nyc.rr.com) |
00:14.18 | *** join/#asterisk tuxinator_linux (~anonymous@ip68-99-229-29.ph.ph.cox.net) |
00:15.11 | ariel_ | hardwire, what are you trying to do. |
00:15.44 | hardwire | http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-38356249088.htm |
00:15.45 | hardwire | !! |
00:15.50 | hardwire | thats what I am trying to do :) |
00:16.32 | KalD|Work | is there any header (cvs etc) that states that libiax is lgpl? |
00:16.46 | mtqh | pyrosteve: BV does not want to help you...they don't support you |
00:17.16 | DJ-Pyro | hardwire: that's a crazy card |
00:17.19 | ariel_ | hardwire, I would love to see that board actuall up and running. |
00:17.30 | PyroSteve | mtqh: why are you speaking for them |
00:17.44 | PyroSteve | mtqh: i support them, I pay them money everymonth |
00:17.55 | PyroSteve | mtqh: 30.00 a month for bussiness line |
00:18.06 | PyroSteve | mtqh: and more to come in the future |
00:18.13 | PyroSteve | mtqh: what kind of customer service is that |
00:18.19 | mtqh | pyrosteve; read there website....THEY DON'T SUPPORT ASTERISK AT ALL |
00:18.25 | mtqh | They give you a gudie |
00:18.27 | mtqh | *guide |
00:18.29 | mtqh | but thats it |
00:18.33 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
00:18.34 | yashax | GUYS: Grooby help me to get the * work with BV....... The issue for outgoing was that you HAVE to use sip.broadvoice.com and can not use their doc that says proxy.xxx.broadvoice.com |
00:18.54 | yashax | BTW, BV so far works great. Crystal Clear |
00:18.55 | PyroSteve | mtqh: I dont need them to support asterisk |
00:19.05 | PyroSteve | i already have asterisk working |
00:19.09 | PyroSteve | with BV |
00:19.51 | ariel_ | PyroSteve, here is there link for asterisk support: http://www.broadvoice.com/support_install_asterisk.html |
00:20.08 | *** join/#asterisk denon (denon@synapse.subneural.net) |
00:20.08 | *** mode/#asterisk [+o denon] by ChanServ |
00:20.12 | PyroSteve | i already have asterisk working |
00:20.13 | PyroSteve | with BV |
00:20.26 | ariel_ | so what type of support to you need? |
00:21.10 | PyroSteve | nahh... if i explain, ill just get shit for it |
00:22.22 | ariel_ | PyroSteve, so this was just to see if anyone from BV hangs out here: PyroSteve needs help from Broadvoice ... is there BV admin here ?? |
00:22.52 | PyroSteve | well almost |
00:22.53 | tuxinator_linux | Does anyone want to help a newb with a Digium T1 card? |
00:23.12 | PyroSteve | who is JerJer ? |
00:23.28 | ariel_ | tuxinator_linux, ask the question |
00:23.41 | wangster | All long distance VOIP providers will be out of business within 1.5 years. |
00:23.42 | ariel_ | ~jerjer |
00:23.43 | jbot | [jerjer] the guy who runs nufone |
00:23.58 | ariel_ | wangster, why? |
00:24.11 | *** join/#asterisk muesli (~muesli@mail.muehlhaeuser.de) |
00:24.30 | Beirdo | ariel_: because wangster said so |
00:24.47 | wangster | ariel_: because its a race to zero for long distance pricing. If you rely on long distance revenue then you are going to be in big trouble. |
00:25.09 | JerJer | word |
00:25.12 | ariel_ | wangster, but 1.5 years... |
00:25.36 | wangster | We already have a local Cable Co. doing residential voice now and offering a $40/mo package which includes unlimited long distance and all calling features. |
00:25.46 | JerJer | its not unlimited |
00:25.58 | wangster | jerjer: says who? |
00:26.02 | JerJer | as defined by webster: 'without limits, no boundries' |
00:26.08 | JerJer | read the terms and conditions |
00:26.12 | iceyp | i managed to get the prepaid calling card working to an extent, when i dial 1234 ( exten => 1234,2,DeadAGI(astcc.agi) ) i get a BING then it hangs up |
00:26.19 | iceyp | anyone know anything about this |
00:26.38 | tuxinator_linux | I don't know what service to order for use with a Digium T1 card. I want to have 6+ voice lines (one number), 1 fax number, and a toll-free number. My current setup is a 6 lines in a hunt-group, a fax line, and a the toll-free rings to the voice number. |
00:26.50 | JerJer | I guaruntee if you ran more than 4,000 minutes in 3 consectutive months, you would not be a customer for the 4th on a so called unlimited program |
00:26.52 | tuxinator_linux | all analog |
00:26.53 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
00:26.58 | JerJer | at this point in the ball game |
00:27.15 | iceyp | JerJer what prepaid system do you use? custom made one? |
00:27.22 | JerJer | yes |
00:27.28 | wangster | JerJer: it is true unlimited. But the price is actually $55/mo (my bad). |
00:27.38 | iceyp | care to opesource it :) |
00:27.50 | JerJer | wangster: and i'm saying it is not |
00:27.54 | JerJer | their costs are still per minute |
00:27.58 | iceyp | this astcc not working properly :( |
00:27.59 | *** join/#asterisk freddy (~jason@martin.pnc.com.au) |
00:28.03 | iceyp | i just get BING! |
00:28.29 | wangster | JerJer: $55/mo buys a LOT of minutes. So long as everyone isn't maxing it out they still make lots. |
00:28.44 | ariel_ | tuxinator_linux, if you have not ordered service from you telco yet what's you location? |
00:28.48 | Beirdo | ~bing |
00:28.49 | jbot | extra, extra, read all about it, bing is an Empirical stochastic bandwidth tester |
00:28.52 | JerJer | no they don't |
00:28.55 | Beirdo | hehe |
00:28.58 | JerJer | do the math |
00:29.03 | KalD|Work | wangster, 55/mon only gets me 450 min on my cell =) |
00:29.10 | wangster | In any case, that is beside the point. It won't be long before everyone has free long distance and then selling long distance only won't be a business. |
00:29.45 | Beirdo | not likely |
00:29.48 | DJ-Pyro | 55/mo gets us 5500 minutes of LD |
00:30.02 | wangster | JerJer: I've done the math. Big providers buy bulk LD for fractions of a cent. thats LOTS of minutes. |
00:30.13 | JerJer | iceyp: nope, not gonna open-source any part of my billing system |
00:30.15 | ariel_ | wangster, bellsouth offers a plan now here for adding to your service unlimited ld us calling for 29 dollars but you need there 30 dollar package before you can order it. |
00:30.29 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
00:30.32 | marc32344 | is there a smaller capacity line than a T1 PRI ? |
00:30.36 | JerJer | iceyp: my theory is if you are billing someone there is a resonable expectation that you are making money |
00:30.47 | ariel_ | wangster, and that is normal pstn service. But you don't see everyone dropping there service with everyone else and going with them. |
00:30.54 | KalD|Work | marc32344, frac t1 =) |
00:30.55 | tuxinator_linux | marc, partial T1 |
00:31.05 | wangster | ariel_: because of the limitations. |
00:31.05 | JerJer | marc32344: a single phone line |
00:31.05 | marc32344 | how do you get partial T1? |
00:31.13 | wangster | ariel_: its probably evenings only? |
00:31.16 | iceyp | jerjer i plan on it in a few months yes |
00:31.18 | tuxinator_linux | ask for it |
00:31.19 | marc32344 | no I want something like 6 lines.. |
00:31.20 | DJ-Pyro | marc32344: full t1 but they only turn up a few of the channels |
00:31.31 | tuxinator_linux | you usually have to pay for a full local loop |
00:31.33 | ariel_ | wangster, no it's not. |
00:31.50 | ariel_ | tuxinator_linux, do you have internet service? |
00:31.53 | marc32344 | dj-pyro -- but still have to pay the full T1 price? |
00:31.56 | yashax | guys, is it possible to have our caller id to show up on someone who is receiving our call from NAME instead of a number? |
00:31.56 | tuxinator_linux | yes |
00:32.05 | *** join/#asterisk pdracevich (~paul@smtp.aucklandtax.co.nz) |
00:32.15 | DJ-Pyro | marc32344: no, you pay the local loop charge, and your provider will give you a price for a fraction of the 24 channels |
00:32.15 | marc32344 | do telcos sell fractioanl T1? |
00:32.25 | wangster | ariel_: 1.5 years might be too quick but I don't see a good business in it. |
00:32.40 | JerJer | in my book Limitations means the same as restrictions which does not mean unlimited |
00:32.44 | *** part/#asterisk pdracevich (~paul@smtp.aucklandtax.co.nz) |
00:32.46 | tuxinator_linux | marc, just about everybody |
00:32.46 | *** join/#asterisk |neuro| (~|neuro|@212.176.51.231) |
00:32.46 | *** join/#asterisk pdracevich (~paul@smtp.aucklandtax.co.nz) |
00:32.50 | ariel_ | tuxinator_linux, there are other ways to do this you can get many different packages it depens on cost. |
00:33.04 | JerJer | read the legal document(s) behind that so-called unlimited service |
00:33.06 | marc32344 | dj-- whats the local loop chrage? |
00:33.08 | pdracevich | Rejected connect attempt from 210.54.x.x, request '00441344844717@bob' does not exist <----- help please please?? |
00:33.12 | tuxinator_linux | ariel, I'm listening |
00:33.18 | DJ-Pyro | marc32344: the cost of the physical line, usually around $200/mo |
00:33.20 | Beirdo | wangster: ma bell has made money for decades on long distance, people will always be able to make money on something people are willing to pay for |
00:33.23 | JerJer | pdracevich: make that extension exist and it will go away |
00:34.16 | JerJer | ...in context bob |
00:34.25 | ariel_ | tuxinator_linux, lets go to a our own window. |
00:35.19 | pdracevich | JerJer: AAAAA LIGHT BULB!!!!, i am calling from point a to point b, so at point be there will have to have a dial rule that tells it to go out into the worl... |
00:35.21 | marc32344 | dj-- how the fractional T1 price work? $x/24 * $(full T1)? |
00:35.44 | JerJer | marc32344: there are two separate charges |
00:35.46 | JerJer | a loop charge |
00:36.06 | JerJer | and the specific feature charge for which you want the T-1 for |
00:36.18 | JerJer | the loop costs will vary based on milage |
00:36.26 | marc32344 | specific feature charge? |
00:36.29 | marc32344 | what is that? |
00:36.36 | DJ-Pyro | the fractional voice service you want to order |
00:36.42 | pdracevich | Jerjer: am i right?? |
00:36.54 | KalD|Work | marc32344, the specail feature charge is for things like more DIDs and line features like CID etc |
00:37.06 | JerJer | exten => _00X.,1,DoSomethingHere |
00:37.23 | marc32344 | ok. so it possible to have a T1 but with only 6channels? |
00:37.25 | JerJer | marc32344: or a connection to the internet |
00:37.28 | JerJer | sure |
00:37.30 | KalD|Work | marc32344, special feature charges are the charges you get for special features =) classic 3rd grade answer =) |
00:37.35 | DJ-Pyro | marc32344: yes, talk to your provider |
00:37.46 | JerJer | the loop you get will be capible of all 24 channels, but only 6 ~can~ be lit up |
00:38.10 | wangster | Beirdo: the catch being "people are willing to pay for". I'm just saying the small guys doing VoIP for long distance only is not a long term viable business model IMHO. |
00:38.12 | pdracevich | exten => _00NXXNXXXXXX.,1,Dial(SIP/${EXTEN}@10.10.x.x) <--- how about that? |
00:38.27 | KalD|Work | typically the telco or provider won't do less than half tho - (or they will charge you for half) because they commonly pay for a full T1 from the upstream carrier |
00:38.28 | JerJer | if that mask is correct for your method dialing, sure |
00:38.38 | JerJer | and if that SIP carrier wants to see the 00 |
00:38.50 | marc32344 | anyone has a list of cheap T1 providers? |
00:38.57 | JerJer | and a period at the end?! how many digits you dialing man? |
00:39.11 | JerJer | what's wrong with simply _00X. ? |
00:39.13 | DJ-Pyro | marc32344: the physical line needs to come from your local telco |
00:39.20 | KalD|Work | marc32344, depends on your location... try eli.net or twtelecom.com or something |
00:39.24 | _Vile | pdr, yeah check Jer out |
00:39.25 | pdracevich | nothning *blush* |
00:39.25 | _Vile | eli sucks |
00:39.30 | _Vile | terrible billing |
00:39.31 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-243-cust.telepacific.net) |
00:39.33 | _Vile | dont go ELI |
00:39.44 | *** join/#asterisk chaoscon_ (~ph33r@chaoscon.user) |
00:39.53 | _Vile | their billing system simply sucks, their customer service is ass |
00:40.04 | JerJer | damn sounds like you are talking about NuFone |
00:40.07 | KalD|Work | i dunno they bill me every month - i'd say that is good billing |
00:40.08 | _Vile | almost |
00:40.16 | _Vile | KalD, check what you're billed for |
00:40.31 | marc32344 | whats the lowest T1 price one can expect? |
00:40.41 | JerJer | depends on the provider |
00:40.41 | KalD|Work | marc32344, ~500/mo |
00:40.44 | _Vile | at leas they suck for LD, dunno about T1 |
00:41.03 | _Vile | marc, I can deliver an LD T1 locally for 179$ for loop only |
00:41.10 | _Vile | i'm doing that right now anyway |
00:41.11 | JerJer | I can get loops for pocket change |
00:41.15 | KalD|Work | _Vile, we pay 590/mo for local and local ld w/ ELI - it is almost as cheap is TW |
00:41.15 | _Vile | central oregon area |
00:41.22 | PyroSteve | i cant seem to find chanspy, does anyone have a link > |
00:41.23 | PyroSteve | ? |
00:41.37 | _Vile | KalD, make sure you're paying them the right rate for LD |
00:41.37 | *** join/#asterisk km- (~km-@67.105.178.130) |
00:41.40 | km- | HI! |
00:41.43 | _Vile | I push 1.5m minutes a month |
00:41.46 | km- | TE405P rocks. :) |
00:41.47 | _Vile | and noticed many discrepancies |
00:41.51 | _Vile | w/ ELI |
00:42.07 | _Vile | as in 20% of their billing was wrong |
00:42.09 | pointer-gaim | JerJer: could you define pocket change? |
00:42.23 | DJ-Pyro | twtc came in at $1200/mo for the DS3 loop, I'm excited |
00:42.26 | KalD|Work | JerJer, who are you working with for your telco stuff? |
00:42.30 | km- | hey guys, if I wanted to connect a phone system to a te405p, when it was expecting e&m wink on an esf/b8zs t1, do I have to use a crossover cable from the te405p? |
00:42.33 | _Vile | I price $1099 for a full DS-3 |
00:42.36 | _Vile | loop |
00:42.44 | marc32344 | _vile -- with eli? |
00:42.45 | pointer-gaim | KalD|Work: jerjer == nufone |
00:42.48 | km- | or is it straight-through |
00:42.49 | _Vile | marc, hell no |
00:42.54 | _Vile | I don't deal w/ them anymore |
00:42.55 | Corydon76-home | Damn, that's cheap |
00:42.56 | km- | I cant seem to get this damn red alarm to clear either way |
00:42.57 | pointer-gaim | KalD|Work: oh, sorry...misread |
00:42.57 | _Vile | jumped carriers |
00:43.02 | DJ-Pyro | _Vile: that includes the cross connect from twtc to GC for the LD service |
00:43.06 | KalD|Work | pointer-gaim, yeah but doesn't nufone offer pstn? =) |
00:43.13 | _Vile | DJ, ahh |
00:43.27 | Corydon76-home | Around here, the ILEC charges ~$1200 for a T1 |
00:43.27 | _Vile | good deal then DJ |
00:43.42 | _Vile | we charge $1099 and buy it from the ILEC :) |
00:43.48 | pointer-gaim | KalD|Work: so find someone with a nufone did and look it up |
00:43.54 | DJ-Pyro | now they're trying to get down to 1c/min LD |
00:44.00 | _Vile | I' |
00:44.03 | Corydon76-home | _Vile: yeah, but for a DS3? |
00:44.07 | _Vile | m paying less than 1c/min |
00:44.12 | _Vile | Corydon, yes for DS-3 |
00:44.14 | KalD|Work | pointer-gaim, why not just ask JerJer? =) |
00:44.15 | DJ-Pyro | _Vile: we don't have the volumes right now |
00:44.19 | Corydon76-home | dayum |
00:44.21 | _Vile | gotcha |
00:44.23 | pointer-gaim | KalD|Work: yes, why not ;) |
00:44.34 | DJ-Pyro | starting at 500k/mo, working up to 10mil/mo |
00:44.50 | _Vile | What are you paying? |
00:44.50 | marc32344 | _vile-- what hardware you run? |
00:44.57 | _Vile | marc, depends, for what? |
00:45.08 | marc32344 | local loop |
00:45.08 | _Vile | DJ, I can blend you at 1.5 for outbound |
00:45.15 | Corydon76-home | _Vile: we're paying $500 for a T1 of data |
00:45.16 | _Vile | if you're doing SIP stuff |
00:45.18 | DJ-Pyro | _Vile: we farm it out to another company right now, we're bringing it back in house |
00:45.47 | DJ-Pyro | it's all PSTN termination, conference calling apps |
00:45.48 | _Vile | Corydon, we sell $99 Data Ts here |
00:46.11 | _Vile | DJ, colo? |
00:46.11 | marc32344 | _vile-- location? |
00:46.12 | DJ-Pyro | ugh, stupid probability and stats exam |
00:46.16 | _Vile | marc, Central Oregon |
00:46.32 | DJ-Pyro | _Vile: datacenter budget was approved last thursday, construction starts in a few weeks |
00:46.40 | *** join/#asterisk Legend (~legend@24.244.142.133) |
00:46.41 | _Vile | ok so you're doing your own colo |
00:46.49 | marc32344 | how many users/did can one typically run on a single T1? |
00:46.55 | _Vile | what are you bringing in facility wise, for voice? |
00:47.06 | _Vile | marc, depends on the type of traffic |
00:47.07 | DJ-Pyro | twtc extending their sonet mesh into the building |
00:47.16 | _Vile | DJ, cool |
00:47.27 | marc32344 | local loop in/out. |
00:47.32 | Trionnis | you're pulling upstream over twc sonet? |
00:47.43 | Trionnis | have fun with that |
00:47.44 | Trionnis | lol |
00:47.52 | DJ-Pyro | they're delivering tdm and ethernet over it |
00:48.12 | _Vile | marc, we charge $99 delivering 100GB/month max, $299 for 200GB, $450 for 300GB T1s |
00:48.14 | Trionnis | they can claim to deliver pizza over it too.... still doesn't matter when it goes down :) |
00:48.15 | Trionnis | lol |
00:48.29 | _Vile | or you can go $139 local loop w/ per MB 95th percentile billing |
00:48.36 | Trionnis | maybe they're better in your area |
00:48.37 | _Vile | s/MB/Mbps |
00:48.41 | Trionnis | their dc here sucks |
00:48.48 | _Vile | never had a problem w/ tw |
00:48.53 | _Vile | always had problems w/ eli |
00:48.55 | Trionnis | good... you're lucky |
00:48.58 | Trionnis | :) |
00:49.03 | DJ-Pyro | tw is good around here |
00:49.08 | coppice | why? the pizza will get cold, and the cheese al congealed and nasty |
00:49.16 | _Vile | yeah I hate that |
00:49.16 | Trionnis | hahaha |
00:49.22 | Trionnis | nice coppice |
00:49.23 | Trionnis | ;) |
00:50.06 | Beirdo | ~insult mikegrb |
00:52.15 | rvhi | has anyone implemented any paging function? |
00:52.32 | DJ-Pyro | what type of pages? |
00:53.01 | rvhi | call a number, it rings all phones |
00:53.06 | DJ-Pyro | yeah, we did that |
00:53.08 | rvhi | all phones have auto answer configed |
00:53.21 | rvhi | so all answered |
00:53.40 | rvhi | how was it done? |
00:53.45 | DJ-Pyro | create a meetme conference and join everyone to it |
00:54.01 | DJ-Pyro | we have an agi script that reads out our listing of extentions to join and it forces them all in |
00:54.28 | rvhi | is it in open source? |
00:54.37 | DJ-Pyro | no, but it's simple to do |
00:55.23 | rvhi | how do you join everyone? |
00:55.32 | rvhi | call them and auto answer? |
00:55.43 | DJ-Pyro | yes |
00:56.16 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
00:56.21 | rvhi | do i need agi? |
00:56.27 | DJ-Pyro | that's how we do it |
00:57.49 | rvhi | is it very similar concept? http://lists.digium.com/pipermail/asterisk-users/2004-March/040186.html |
00:58.05 | DJ-Pyro | yup, very close |
00:58.12 | DJ-Pyro | that's where we got the idea to do it |
00:58.33 | rvhi | cool, thx |
00:59.20 | *** join/#asterisk Brixius (Brixius@c-24-118-4-197.mn.client2.attbi.com) |
00:59.25 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
00:59.29 | Brixius | Hello |
00:59.42 | Brixius | Is there anyone here from voipjet? |
00:59.56 | BuckRogers | has anyone incorparated automated billing with automated creditcard processing |
00:59.57 | greg_work | rvhi: i do it just with computer speakers hooked up to audio out on my soundcard, then call Console/DSP |
00:59.59 | tzanger | there's a voipjet user (that's the nick) who is in here from time to time |
01:00.03 | tzanger | I think he got kicked out eventually |
01:00.12 | BuckRogers | this is more for the service providers |
01:00.58 | greg_work | rvhi: the spa-841's also have a SIP header they'll respond to specifically for paging, but you need the SipAddHeader() application (only in cvs head, afaik) to use it |
01:01.08 | Brixius | Dang, I wanted to talk to someone from there, just had some weirdness I thought they's like to know about |
01:01.29 | tclark | Brixius: like randon call drops on voipjet :) |
01:01.40 | Brixius | no, hearing someone elses call |
01:01.47 | tclark | heh even better |
01:01.52 | ManxPower | Someone claiming to be from VoipJet has been less than professional in public forums. That's why I won't use them. |
01:02.42 | Brixius | hmmm, I may have to find a replacement, overall I havn't had too many problems with them, but now I'm a little worried who's hearing my call. |
01:03.34 | Brixius | haha |
01:03.35 | tclark | yah its best to have a stable of iax providers they all have issue from time to time, ... |
01:04.40 | *** join/#asterisk mikes2277 (~mike@wireless-206.222.58.98.omnilec.com) |
01:04.44 | BuckRogers | yeah learn as u go |
01:04.46 | tzanger | yup I have two main ones, nufone and then I drop back to sixtel if nufone's down or I can't reach them |
01:04.50 | tzanger | but it's never come to htat :-) |
01:04.51 | BuckRogers | at the customers expenese |
01:05.07 | tzanger | and failing everything else I can always dial out my Bell Canada PRI at $0.05/min national |
01:05.26 | greg_work | tzanger: out of curiousity, how much does the PRI cost you? |
01:05.30 | tzanger | too much |
01:05.48 | tzanger | I am in a Tier 4 price group. That's the offical "bend you over and we won't use lube" group |
01:05.51 | mikes2277 | app_meetme seems to need a zaptel board, does anyone know of a conference server app that doesn't need a zaptel board? |
01:05.53 | greg_work | its a hard thing to find pricing on |
01:05.54 | greg_work | heh |
01:05.57 | tzanger | no it's not |
01:06.06 | tzanger | you call Bell, Telus, Primus, Allstream and so on and get quotes |
01:06.28 | greg_work | we only have 4 co's.. i'm just curious what the point is where it becomes cost effective |
01:06.36 | ManxPower | Then take the lowest quote, call all the others and tell them what your lowest qoute was. Rinse. Repeat. |
01:06.38 | tzanger | four central offices? |
01:06.40 | greg_work | i think they'd laugh if i asked for 4 channels :p |
01:06.46 | _Vile | oh 4 co lines |
01:06.50 | greg_work | no, sorry, i meant 4 Co lines :p |
01:06.59 | _Vile | yeah I'd laugh at you |
01:07.00 | Brixius | does nufone allow you to set ani information on outgoing calls? |
01:07.02 | tzanger | greg_work: yeah PRI is generally only cost effective about the 8-12 line mark |
01:07.07 | _Vile | yep 8 or so |
01:07.09 | tzanger | Brixius: why would I need to set ANI? |
01:07.13 | _Vile | depending on price |
01:07.22 | greg_work | what about BRI? |
01:07.32 | tzanger | I can set outgoing caller ID which does me just fine |
01:07.33 | ManxPower | BrianR___, I don't believe they do, but you can set Caller*ID |
01:07.33 | _Vile | why bother, more expensive |
01:07.37 | greg_work | i mean, all i really want is the call handling capabilites |
01:07.41 | Brixius | that's what I was wondering |
01:07.48 | _Vile | unless you want to set caller ID |
01:07.52 | Brixius | if I can set outgoing caller id |
01:07.59 | ManxPower | Brixius, Caller*ID is NOT NOT NOT the same as ANI. |
01:08.03 | _Vile | which you probably can't do unless your carrier lets you |
01:08.18 | *** join/#asterisk sysdef (~sysdef@pD9560C7A.dip.t-dialin.net) |
01:08.21 | _Vile | and even then you have to give a list of ANI's/Caller ID for you to be able to set |
01:08.21 | ManxPower | If you use the term ANI when you mean Caller*ID someone is going to smack you. |
01:08.30 | _Vile | unless they aren't hard asses |
01:08.32 | ManxPower | That's like saying "gasoline" when you mean "water" |
01:08.35 | mikes2277 | most carriers do, we let them set ANI on our PRI's |
01:08.35 | tzanger | ManxPower: with the mint from your own julip? |
01:08.43 | tzanger | ManxPower: hahahhaa |
01:08.45 | _Vile | We set ANI on our PRI |
01:08.47 | _Vile | s |
01:08.51 | _Vile | and Caller ID |
01:08.54 | nestAr | lol.. i set mine to whatever i want.. |
01:08.54 | mikes2277 | us too |
01:09.03 | tzanger | what does being able to set ANI on a VOIP provider get you? I've not understood that |
01:09.04 | _Vile | but we have 2 DS-3s |
01:09.10 | tzanger | 911 perhaps but that's a corner case |
01:09.26 | greg_work | it kinda sucks theres no viable in-between option. i want to be able to set callerid (only so when you make calls, it appears to come from the main number, as opposed to just random numbers from our hunt group), and just have it answer quicker .. the POTS line rings once before * picks up |
01:09.35 | _Vile | 911 is a problem, I always advise someone keeping a land line in the place for 911 |
01:09.38 | tzanger | greg_work: you can get your telco to do that on POTS |
01:09.46 | tzanger | _Vile: me too |
01:10.12 | greg_work | tzanger: do what? callerid, or sending the callerid info faster so * can answer immediately? |
01:10.14 | tuxinator_linux | Whats E911 then? |
01:10.18 | JerJer | 911 isn't really that big of a deal |
01:10.20 | _Vile | other than that, I just give my carrier 911 information just in case someone makes a call |
01:10.29 | marc32344 | can you legally operate without 911 capability? |
01:10.30 | _Vile | location details etc |
01:10.32 | tzanger | greg_work: you can ask your telco to set CID to your main # on all your POTS lines |
01:10.56 | _Vile | marc, no but you can tell the customer to use another line for 911 |
01:10.57 | tuxinator_linux | tzanger, I tried that, they said they couldn't |
01:11.12 | nestAr | they're fibbing |
01:11.14 | _Vile | and give your carrier 911 location details for that # |
01:11.14 | tuxinator_linux | putting the main num on all POTs |
01:11.15 | greg_work | tzanger: ah ok. next time i need to do something maybe i'll get them to do that. they charged me a $60 admin fee to add a line to our hunt group (was previously a modem line) |
01:11.16 | nestAr | can't = won't |
01:11.17 | _Vile | just in case |
01:11.44 | tzanger | JerJer: I've heard rumour (nobody seems to know at BCE) that when you're small potatoes (under a DS3 worth of channels) you can't set your DID 911 addresses to anything other than addresses that will be terminated at your local PSAP |
01:11.56 | tzanger | tuxinator_linux: that sucks, bell will do it for us IIRC |
01:12.05 | tzanger | greg_work: yes, I'd do the same thing :-) |
01:12.25 | greg_work | charge me you mean? :) |
01:12.30 | tuxinator_linux | I am changing providers ASAP, XO is killing me |
01:12.39 | tzanger | greg_work: yes |
01:12.44 | greg_work | bastard ;) |
01:13.07 | tzanger | you make it a nominal fee like $50 or something for any number of changes -- it keeps the dicks who call you every few days wanting a little tweak here and there from calling |
01:13.25 | tzanger | and if you save up a dozen things it's better for everyone |
01:13.35 | greg_work | fair enough |
01:13.51 | greg_work | but it also costs a lot to people that rarely need to change things |
01:13.52 | *** join/#asterisk yasha (~yasha_x@69.15.218.218) |
01:14.08 | tzanger | greg_work: nonsense.. if you rarely need to change things it costs you $50 what once or twice a year |
01:14.12 | greg_work | the sensible thing would be to allow one free change every x days (60 or 90 or something) and beyond that, charge |
01:14.27 | tzanger | greg_work: actually that's precisely how I do it |
01:14.39 | tuxinator_linux | greg, makes sense t me |
01:14.40 | tzanger | Bell Canada charges us $250 for any change to the PRI |
01:14.57 | greg_work | i guess it's more knowing the work they do. i'd guess it literally takes 2 minutes |
01:14.57 | JunK-Y | tzanger: like us. |
01:14.58 | nestAr | bell canada sounds like a bunch of pigfuckers |
01:15.00 | nestAr | ;) |
01:15.00 | *** join/#asterisk bjohnson_ (~bjohnson@ip226-181.tor.istop.com) |
01:15.05 | nestAr | oh wait.. it's a bell |
01:15.07 | nestAr | nvm |
01:15.11 | greg_work | well, thats why you don't use bell canada ;) |
01:15.18 | nestAr | lol |
01:15.22 | nestAr | that and not being in canada |
01:15.26 | Brixius | Isn't ani the # that is displayed as the caller id #. |
01:15.30 | tzanger | well when you're in a town of 5300 and not even allstream will return your calls about quoting a PRI... |
01:15.41 | tzanger | Brixius: no, ANI and CID are different |
01:15.46 | tzanger | or rather they can be |
01:15.48 | greg_work | tzanger: where are you from? |
01:15.52 | tzanger | Listowel, ON |
01:16.11 | greg_work | i used to be with allstream, but they couldn't get us service here. we have some centrex-style lines with primus now |
01:16.14 | km- | hey guys, if I have a RED/REC state, I see data on RxA,B,C, and D, but I can't get a green light |
01:16.16 | km- | what does that indicate? |
01:16.25 | km- | it's trying to recover but its not happening |
01:16.25 | tzanger | greg_work: where're you at |
01:16.27 | greg_work | sounds familliar, but i'm not sure where it is. western somewhere? i'm in kingston |
01:16.38 | tzanger | km-: bad framing? |
01:16.52 | km- | esf/b8zs is coming from the CO, that's what I'm sending to the remote side |
01:16.53 | greg_work | or odessa, more specifically (though i live in kingston) |
01:17.00 | tzanger | you're in RA which means you're transmitting YAI |
01:17.04 | _Vile | Brixius, ANI is how the call is rated.. what the original calling ID *is* |
01:17.10 | _Vile | Caller ID is what appears on the phone |
01:17.10 | tzanger | the other side shouldn't be sending you any normal data |
01:17.13 | km- | tzanger: I have it setup like this CO --->Asterisk----->Legfacy PBX |
01:17.27 | km- | tzanger: I have both spans setup as esf,b8zs |
01:17.28 | Brixius | So if I want the # I call to display the did # I'm calling from, don't I set the ANI prior to executing the Dial command. |
01:17.36 | km- | tazanger: are you saying span2 should be a different framing? |
01:17.38 | BuckRogers | i have tear1 and tear2 pricing |
01:17.40 | tzanger | km-: ok |
01:17.41 | _Vile | just use SetCallerID |
01:17.42 | tzanger | no |
01:17.46 | _Vile | it sets both ANI and Caller ID |
01:17.49 | tzanger | Asterisk has two spans |
01:17.58 | _Vile | prior to the Dial command |
01:17.59 | BuckRogers | spaning tree protocol? |
01:18.00 | tzanger | let's say span 1 is to Telco and span 1 is to PBX |
01:18.14 | km- | right, thats how I have it configured, span1=telco span2=pbx |
01:18.16 | BuckRogers | got some nortel baystack switches with that running |
01:18.19 | tzanger | set span=1,1,0,esf,b8zs and span=2,0,esf,b8zs |
01:18.31 | tzanger | er span=2,0,0,esf,b8zs |
01:18.47 | tzanger | you should get green light if all the wires are correct |
01:18.47 | BuckRogers | ooo nice |
01:18.50 | km- | span=1,1,0,esf,b8zs |
01:18.50 | km- | e&m=1-24 |
01:18.50 | km- | span=2,0,0,esf,b8zs |
01:18.50 | km- | e&m=25-48 |
01:18.54 | km- | that's what I have now |
01:18.59 | tzanger | ok |
01:18.59 | km- | ok |
01:19.00 | BuckRogers | got ot love that green light |
01:19.08 | tzanger | km-: you're using E&M with the telco? |
01:19.08 | BuckRogers | looking smooth |
01:19.10 | km- | tzanger: should I be using a crossover cable from asterisk to the PBX? |
01:19.18 | tzanger | km-: generally that is what you need |
01:19.19 | BuckRogers | cross over |
01:19.21 | km- | yeah, the signal comes in as e&m wink |
01:19.21 | tzanger | 1&4 -> 2&5 |
01:19.24 | km- | I tried using the crossover |
01:19.25 | _Vile | e&m is cool when you're not going PRI |
01:19.27 | BuckRogers | ether to ether, yeah |
01:19.32 | _Vile | more features |
01:19.37 | BuckRogers | comp to comp no router |
01:19.40 | _Vile | than standard DSS |
01:19.42 | tzanger | _Vile: yup our AS5248s were all E&M Wink |
01:19.43 | km- | tzanger: I trued using crossover, but the light stays permanent red in that case |
01:19.46 | tzanger | before we went to PRI |
01:19.51 | BuckRogers | roll over? |
01:19.54 | tzanger | km-: take a loopback cable and plug it in |
01:20.04 | tzanger | if it don't go green you have other problems |
01:20.10 | km- | loopback cable? I.e., a crossover? |
01:20.13 | tzanger | km-: no |
01:20.18 | _Vile | no, loopback |
01:20.19 | tzanger | take an RJ48 end |
01:20.22 | tzanger | take two wires |
01:20.22 | BuckRogers | when you went to pri is the zap configs simialar |
01:20.31 | _Vile | you need to take RX->TX TX->RX |
01:20.32 | tzanger | connect pin 1 to pin 4 and to to pin 5 |
01:20.46 | _Vile | pin 3 to pin 5 |
01:20.50 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
01:20.56 | tzanger | _Vile: no not pin 3 |
01:20.58 | km- | hmm |
01:21.06 | _Vile | ahh skip pin 3, pin 2 |
01:21.10 | km- | can I test this loopback cable by using spans 3 and 4 in a particular configuration? |
01:21.26 | tzanger | 4&5 is pair #1, 1&2 is pair 2, 3&6 is pair 3 and 7&8 is pair 4 |
01:21.34 | tzanger | ethernet uses pairs 2 and 3 |
01:21.39 | tzanger | T1 uses pairs 1 and 2 |
01:21.51 | tzanger | km-: it's not a loopback CABLE |
01:21.54 | tzanger | it's a loopback PLUG |
01:21.56 | neopher | tzanger: yep thats right |
01:21.59 | _Vile | yep |
01:21.59 | tzanger | it doesn't conenct to anything else |
01:22.08 | km- | sorry, I'm getting my nomenclature confused |
01:22.17 | km- | I created this crossover cable, I want to prove the crossover is wired right |
01:22.20 | iceyp | I got a problem using agi's when the call is connected it disconnects within a few seconds, sometimes it'll work for about 15 seconds, but hardly, i get the following error: Feb 23 14:19:19 WARNING[60894]: file.c:1058 ast_waitstream_full: Wait failed (Resource temporarily unavailable) |
01:22.23 | greg_work | tzanger: i thought 3&6 was pair 2? |
01:22.31 | tzanger | km-: do you have pair 1 and pair 2 crossed? If so, it's good |
01:22.35 | _Vile | http://www.cisco.com/warp/public/471/hard_loopback.html#lb_png |
01:22.36 | tzanger | greg_work: nope that's pair 3 |
01:22.37 | neopher | on ethernet yes |
01:22.47 | tzanger | blue is pair 1, orange is pair 2 green is pair 3 brown is pair 4 |
01:22.50 | km- | so, if I plug this crossover cable into spans 3 and 4, configure 3 and 4 both to have esf/b8zs |
01:22.53 | _Vile | km -> http://www.cisco.com/warp/public/471/hard_loopback.html#lb_png |
01:22.54 | km- | and set them both to e&m |
01:22.57 | km- | the line should go green |
01:22.58 | km- | right |
01:23.04 | tzanger | km-: technically yes but try the plug first |
01:23.08 | km- | ok |
01:23.09 | greg_work | tzanger: using 568B |
01:23.13 | tzanger | don't create your own experiments, this is basic troubleshooting |
01:23.13 | greg_work | :) |
01:23.21 | tzanger | greg_work: I use standard telco colouring |
01:23.24 | km- | tzanger: ok, make loopback plug, and plug it into span2 of the asterisk box? |
01:23.33 | tzanger | km-: yes, it should go yellow then green |
01:23.47 | km- | do I have to set the card in loopback mode? |
01:23.48 | greg_work | i used to use B, then i learned that only the US and old telco techs use B, so i switched to A :) |
01:23.49 | _Vile | I always screw up on using 3 instead of 2, I shouldn't. |
01:24.11 | *** join/#asterisk dsmouse (~mouse@rrcs-24-199-146-243.midsouth.biz.rr.com) |
01:24.14 | tzanger | _Vile: you're just a network weenie instead of a telco weenie |
01:24.17 | tzanger | that's all |
01:24.39 | _Vile | im telco, too... I just wire T-1s once a year |
01:24.50 | km- | I think my damn crossover cable is wrong, but lemme get this loopback plug fixed up |
01:24.54 | tzanger | _Vile: yeah but if you do any trunk wiring you know the color code |
01:24.57 | _Vile | I get to get my hands on a metaswitch in about 6 months |
01:25.06 | _Vile | tzang, I don't do enough you're right |
01:25.10 | tzanger | I'm also an electronics guy so I know that bad boys rape our young girls but violet goes willingly |
01:25.12 | _Vile | im more software |
01:25.13 | greg_work | tzanger: oh, T1 cabling is different from ethernet? |
01:25.18 | _Vile | greg, yes |
01:25.22 | tzanger | greg_work: yes and no |
01:25.37 | tzanger | the pairs are the same, it's just that ethernet uses pairs 2 and 3 and T1 uses pairs 1 and 2 |
01:25.38 | greg_work | i just assumed they'd use the same cabling .. on ethernet i was right, 3&6 is pair 2 |
01:25.47 | tzanger | so an ethernet crossover's totally different from a T1 crossover |
01:25.53 | greg_work | oh sorry, for 568A |
01:26.03 | neopher | you could use a straight through ethernet, but the tx and rx pins are different on T1 vs ethernet |
01:26.11 | ManxPower | But an Ethernet straight-thru cable can work for Ethernet, T-1, and POTS |
01:26.14 | _Vile | blue blue white red orange yellow green...? |
01:26.18 | greg_work | 568B it's 1&2 .. but orange on both |
01:26.21 | tzanger | neopher: yeah but the pairs are still right :-) |
01:26.35 | ManxPower | But, as someone pointed out you'll break the pairs and add potential problems |
01:26.50 | tzanger | split pairs are the cause of global warming |
01:26.53 | tzanger | and the iraq war |
01:27.00 | neopher | correct, but if you are making a loopback plug, you need to know the correct pins to jump and tx and rx are diff |
01:27.08 | tzanger | neopher: yes |
01:27.10 | _Vile | that and those damn cars |
01:27.18 | BuckRogers | o man i wish i had some cronic |
01:27.18 | tzanger | I've never seen a useful ethernet loopback plug :-) |
01:27.42 | BuckRogers | oooo thank god, (hands shaking) |
01:27.51 | _Vile | whiner |
01:27.53 | _Vile | :) |
01:28.02 | neopher | tzanger: very true |
01:28.10 | tzanger | Brixius: so what are they |
01:28.14 | BuckRogers | crazy day of agi perl scripting |
01:28.19 | tzanger | cid is for us wee plebs, ANI is for billing :-) |
01:28.20 | _Vile | Brix, you can just use SetCallerID and call it a day |
01:28.27 | _Vile | tzang is right |
01:28.34 | km- | ok |
01:28.37 | km- | loopback plug complete |
01:28.40 | _Vile | just remember the BTN will be the SetCallerID # |
01:28.41 | km- | and we have a green |
01:28.47 | tzanger | km-: so your crossover's not |
01:28.49 | _Vile | when you're dialing 800 #s etc |
01:28.50 | Brixius | From what I can tell ani is for billing, but you answered the q before I could. |
01:29.01 | tzanger | ANI can't be (easily) fucked with |
01:29.02 | tzanger | CID can |
01:29.04 | km- | tzanger: yeah, I had the plug in the wrong orientation, was counting from pin 8 as opposed to pin 1 |
01:29.12 | km- | tzanger: the cisco document proved that to me |
01:29.21 | tzanger | km-: always treat your RJ48 right |
01:29.23 | neopher | ANI.L |
01:29.26 | tzanger | pins up, facing you |
01:29.33 | tzanger | 1-8 is left-to-right |
01:29.59 | _Vile | tzang, I don't know if that's true or not, I can set ANI and Caller ID ?? |
01:30.06 | *** join/#asterisk trym (~trym@linux.debian.us) |
01:30.07 | tzanger | _Vile: depends on the telco |
01:30.08 | _Vile | and I'm on Override CPN |
01:30.11 | _Vile | w/ MCI |
01:30.13 | _Vile | outbound LD |
01:30.20 | _Vile | that's LD though, I guess and not 800 |
01:30.23 | tzanger | asterisk does allow you to set ANI too with another IE but the telco may just reject it |
01:30.25 | _Vile | so it doesn't matter |
01:30.41 | km- | ok |
01:30.43 | km- | for one more time |
01:30.46 | _Vile | I can call you right now w/ 666 as my phone number and the ANI will be 666 :) |
01:30.47 | km- | 1-4 2-5 right? |
01:31.00 | tzanger | km-: yup |
01:31.34 | tzanger | just look at it this way |
01:31.34 | _Vile | in fact your cdr's will show 666 :) |
01:31.36 | tzanger | take your cat5 |
01:31.40 | tzanger | bring out the orange pair |
01:31.48 | tzanger | orange-white and orange go right up the middle |
01:31.57 | tzanger | now take the blue pair |
01:32.02 | tzanger | bluewhite and blue go on the far left |
01:32.06 | _Vile | Brixius, but yes, tzanger is right, and you're right, ANI is for billing, CDR is for presentation |
01:32.10 | tzanger | now the othe rside is just the opposite |
01:32.25 | tzanger | take the greenwhite/green pair and "straddle" the middle pair |
01:32.32 | tzanger | take the brownwhite/brown pair and put them on the far right |
01:32.40 | _Vile | tzang, to correct myself, it'd be 000-000-0666 |
01:32.53 | tzanger | I don't even count anymore, it's just "right up the middle" and "to the left" or "straddle the middle" and "tot eh left" |
01:33.08 | tzanger | _Vile: my CDRs will? |
01:33.18 | _Vile | not LD, just inbound * CDRs |
01:33.51 | _Vile | speaking of which |
01:34.05 | _Vile | are standard csv CDRs for Asterisk reporting ANI or CID? |
01:34.09 | BuckRogers | dont forget to bring a towel |
01:34.11 | _Vile | I need to know that |
01:34.20 | tzanger | I'm confused |
01:34.27 | tzanger | why will my CDRs be 0000000666? |
01:34.49 | _Vile | If I call you from 666, it will show up in Asterisk as the number dialing being 666 |
01:34.55 | tzanger | ahhhhhh |
01:34.56 | _Vile | I can show you |
01:34.58 | tzanger | okay |
01:35.07 | tzanger | I understand now |
01:35.07 | _Vile | but I don't know if that's Caller ID or ANI |
01:35.16 | _Vile | in the Asterisk CDRs |
01:35.35 | _Vile | I think I'm supposed to be getting ANI reporting from my carriers and not CID |
01:35.36 | tzanger | I imagine asterisk gets it right although I haven't the need to worry about it |
01:35.50 | tzanger | and unless you'e got a PRI you'll never know if it's different |
01:35.56 | _Vile | it's all PRI |
01:36.24 | *** join/#asterisk jayden (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
01:36.26 | tzanger | _Vile: pri debug span 1 and place a few calls where you change the CID -- call yourself and see if you get ANI info |
01:36.27 | _Vile | I've asked for ANI reporting, but I dunno, I guess I could check by calling myself :) |
01:36.32 | tzanger | I'm doubting it but could be wrong |
01:36.41 | *** join/#asterisk sysdef (~sysdef@pD9560C7A.dip.t-dialin.net) |
01:36.48 | km- | if the line is in yellow alarm |
01:36.50 | km- | whats that mean? |
01:37.02 | km- | with my new crossover, the line is now in yellow as opposed to red |
01:37.04 | _Vile | signalling maybe? |
01:37.06 | JunK-Y | km: on PRI? |
01:37.10 | jayden | wassup... |
01:37.10 | km- | should I reboot the phone system? |
01:37.14 | tzanger | yellow alarm means that one end sees the other |
01:37.15 | _Vile | won't help |
01:37.15 | km- | em_w esf/b8zs |
01:37.22 | tzanger | basically take you and me. |
01:37.28 | JunK-Y | YEL means remote end is unplugged i think |
01:37.36 | JunK-Y | or the dchan is not alive. |
01:37.38 | _Vile | no red would be remote is unplugged |
01:37.40 | tzanger | I can't see you, so I will transmit red alarm. I CANNOT HEAR YOU |
01:37.56 | _Vile | yellow means there's signalling problems |
01:37.57 | tzanger | you hear me screaming at you, so you send back "YOU ARE SHOUTING IN MY EAR" |
01:38.00 | km- | ok, so, maybe one of the pins on my crossover isn't fully crimped? |
01:38.01 | _Vile | wrong setup, not esf when it should be |
01:38.05 | tzanger | _Vile: not quite |
01:38.09 | *** join/#asterisk convey (~test@208-216-127-234.cust.gti.net) |
01:38.09 | _Vile | not b8zs but is ami, when it should be reversed |
01:38.10 | _Vile | etc |
01:38.15 | tzanger | red alarm means I can't see the other side. it's an unframed all-1's pattern |
01:38.38 | _Vile | I've always experienced yellow when my signalling is not right on Digium cards |
01:38.45 | tzanger | yellow alarm means I can see a signal but it is unintelligible |
01:38.49 | tzanger | this is "lower" than signaling |
01:38.53 | convey | anyone using asterisk as an application server for Ser? |
01:39.00 | greg_work | testing with untested test equipment == fun |
01:39.06 | km- | tzanger: so, do you think my crossover may have a broken wire or something? |
01:39.06 | _Vile | so, if one side was setup for d4 |
01:39.11 | _Vile | and the other side was setup for esf |
01:39.13 | tzanger | km-: looks to be that way |
01:39.16 | jayden | yellow alarm... kinda like what usually goes on in #asterisk... |
01:39.18 | _Vile | that would be unintelligible, correct? |
01:39.19 | km- | tzanger: ok, I'll redo it again |
01:39.23 | tzanger | :-) |
01:39.27 | jayden | red alarm, like your network cable is unplugged |
01:39.29 | tzanger | _Vile: not necessarily |
01:39.30 | km- | hey, better I make progress |
01:39.40 | jayden | :) |
01:39.43 | _Vile | that's the cases where I've experienced a yellow, check it out |
01:39.49 | jayden | I'm gunna be good tonight :) |
01:39.56 | _Vile | I'm not :) |
01:39.58 | tzanger | D4/SF framing is perfectly legal ESF framing, but the extra bits in ESF can confuse SF |
01:40.08 | _Vile | thus causing a yellow |
01:40.28 | tzanger | _Vile: incorrect signalling *can* cause a yellow but doesn't ensure it |
01:40.41 | pr0m | anyone here have experience with the linksys pap2 ata? |
01:40.46 | tzanger | pr0m: not me |
01:40.48 | km- | besides vile |
01:40.59 | km- | in this case we've already verified I'm using the right signalling |
01:41.00 | pr0m | tzanger: ok. |
01:41.09 | Himeko | i use one, but that is about it |
01:41.18 | _Vile | km, ok, catch me up, what signalling are you using on each side? |
01:41.31 | _Vile | and have you restarted asterisk and re-ran ztcfg? |
01:41.36 | Himeko | a pap2 that is |
01:41.42 | pr0m | how about decrypting an rc4 encrypted file? |
01:41.42 | eKo1 | Dang it, why doesn't the src field in the cdr table have the proper source. Damn it to hell! |
01:41.57 | km- | _vile: esf/b8zs coming from the CO, esf/b8zs I'm sending to legacy pbx |
01:41.59 | jayden | km: are you using all 23 (or 24 if t1) channels? are you using external csu, if it is a smart csu, does it have the same settings |
01:42.00 | pr0m | Himeko: do you use it with asterisk? |
01:42.11 | tzanger | km-: that's framing and line coding, not signaling |
01:42.12 | jayden | or no csu |
01:42.16 | *** join/#asterisk Newbie___ (some@211.24.146.10) |
01:42.19 | _Vile | CO is sending esf/b8zs? |
01:42.23 | jayden | PRI or T1 or E1? |
01:42.38 | km- | yep |
01:42.38 | tzanger | _Vile: I've never seen a T1 in the last 10 years that was SF/AMI |
01:42.45 | tzanger | _Vile: not "in the wild" anyway |
01:42.46 | _Vile | I have 22 of them |
01:42.49 | _Vile | is it PRI or DSS, and what does your zapata.conf look like? |
01:42.50 | tzanger | _Vile: wow |
01:42.55 | Himeko | pr0m it is connected to * yes |
01:42.58 | Nugget | ms word uses rc4. |
01:43.16 | ManxPower | I didn't know you could run PRI over AMI |
01:43.23 | pr0m | i've got a pap2 that has vonage as the only voip provider.... and i'd like to "unlock" the device so i can point it towards an asterisk server. |
01:43.23 | _Vile | you can't |
01:43.28 | _Vile | I have 22 AMI/D4 |
01:43.33 | tzanger | I didn't thnk so either... technically feasable but why? |
01:43.35 | Himeko | oh, min eis a pap2-na |
01:43.41 | tzanger | _Vile: PRI or CAS T1 |
01:43.42 | _Vile | and 6 ESF/B8ZS PRI on the rest of the DS-3 |
01:43.47 | _Vile | cas |
01:43.48 | jayden | pr0m: I think vonage will unlock forr $15 |
01:43.49 | pr0m | Himeko: ok. hhhmmmm |
01:43.50 | jayden | call them |
01:44.02 | pr0m | jayden: really? huh. |
01:44.15 | JerJer | yeah 'unlock' ok |
01:44.17 | jayden | yes, they will for the cisco and motorolla ones |
01:44.23 | _Vile | it's only outbound stuff, calling card application w/ no caller id |
01:44.33 | pr0m | himeko can you download your firmware from the pap2-na? backup? |
01:44.38 | _Vile | gives me an extra channel per T, though I could option the whole thing w/ 6 signalling channels |
01:44.40 | _Vile | but im lazy |
01:44.45 | jayden | they reflash with a diff firmware I beleive |
01:44.45 | _Vile | and use all PRI |
01:44.48 | _Vile | but I'm lazy |
01:44.52 | jayden | that is the word on the street |
01:44.59 | JerJer | jayden: no |
01:45.08 | jayden | ok |
01:45.22 | jayden | hey, I am just repeating the bad info I got... :) |
01:45.25 | JerJer | they simply reconfigure the device not to ask for a password when going to the admin interface |
01:45.36 | pr0m | i've successfully upgraded the pap2 with a local tftpserver and vonage provided firmware package. .... |
01:45.40 | jayden | ^^ what he said |
01:45.56 | JerJer | pr0m: then you must pay vonage now |
01:46.01 | JerJer | that was your mistake |
01:46.05 | ManxPower | JerJer, Could you take over torturing users while I'm busy? 8-) |
01:46.11 | tzanger | I just discovered the uselessness of the smarthome X10 stuff that "auto-configures" their address |
01:46.19 | *** part/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
01:46.20 | jayden | JerJer says, use Nuphone :) |
01:46.23 | tzanger | power outage...everything is on the address the first command is sent to |
01:46.24 | pr0m | JerJer: i've reset the device several times. |
01:46.25 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
01:46.40 | JerJer | its locked down hard |
01:46.45 | BuckRogers | anyone use simpleconnect? |
01:46.49 | ariel_ | ManxPower, your busy.... |
01:46.50 | pr0m | that's not the problem. it's uploading firmware that will allow me admin access or get the passwd. |
01:46.57 | JerJer | now that you've let it communicate with vonage |
01:47.10 | greg_work | tzanger: which ones are those? i just started installing a bunch of the switchlinc dimmers at our cottage .. :p |
01:47.33 | pr0m | JerJer: i have access to the factory default web interface. just no passwd. |
01:47.34 | greg_work | they don't autoconfigure though (neither do the keypad 8's)... |
01:47.49 | tzanger | http://ericzorn.com/extra/dibs/ |
01:47.50 | tzanger | what the fuck |
01:47.55 | Beirdo | pr0m: welcome to the club |
01:47.56 | tzanger | you guys put trash in parking spaces to reserve them? |
01:48.07 | pr0m | Beirdo: what's your experience with the pap2? |
01:48.10 | tzanger | greg_work: these aren't dimmable but anything that autolearns will have this problem |
01:48.17 | Beirdo | it's locked |
01:48.34 | pr0m | hmmm. no success, eh? |
01:48.38 | JerJer | its not locked |
01:48.39 | cbachman | <PROTECTED> |
01:48.46 | JerJer | simply pre-configured to vonage |
01:48.48 | greg_work | tzanger: i had to send some weird sequence to program them |
01:48.48 | tzanger | cbachman: unreal |
01:48.55 | tzanger | greg_work: these auto-learn |
01:49.00 | JerJer | then vonage really locks it the first time it communicates with their TFTP server |
01:49.04 | tzanger | just press the button and send a command |
01:49.08 | tzanger | on or off |
01:49.13 | greg_work | tzanger: oh sorry, that was to do scenes.. to program all the ones i have you hold the set button for 3 seconds, then send the address |
01:49.14 | Beirdo | well... it's configured to do some funky MD5 auth on the admin interface too apparently |
01:49.18 | pr0m | Beirdo: do you know how to reset it to factory defaults? |
01:49.37 | greg_work | they come factory set at A1 |
01:49.40 | Beirdo | that info's out there, but I don't think it will get you very far |
01:49.45 | cbachman | tzanger: http://www.policyguy.com/pubs/ChicagoSnow.html |
01:49.46 | tzanger | pr0m: use the JTAG port and reflash that way :-) |
01:49.51 | JerJer | pr0m: it is never going to happen without involving vonage, since you let it communicate with them |
01:49.55 | sysdef | eh.. no u* -users here? lol |
01:50.12 | pr0m | jtag. i'll give that a googling. |
01:50.28 | pr0m | JerJer: i've heard otherwise on newsgroups. i |
01:50.37 | greg_work | which is a bit crazy at first .. i only put a few in so far (most of the elec box isn't wired up yet) and when i first tured it on, one button on the keypad would control EVERYTHING ;p |
01:50.38 | pr0m | i'll keep my hopes up for now. |
01:50.47 | tzanger | cbachman: that's unreal |
01:50.52 | tzanger | pr0m: I'm joking |
01:50.57 | tzanger | you won't flash it with a jtag easily |
01:51.08 | km- | OK |
01:51.09 | km- | that's odd |
01:51.12 | km- | green |
01:51.15 | pr0m | tzanger: ok then. thanks for... um nothing. ;-) |
01:51.15 | JerJer | the pap2's are trivial to reconfigure provided they never communicate with vonage's TFTP server |
01:51.18 | km- | but, I dont get a dialtone |
01:51.27 | tzanger | pr0m: sorry man, you let it talk to the world, go sell it on ebay |
01:51.32 | km- | the legacy pbx isn't getting a dialtone when I hit the t1 |
01:51.38 | Beirdo | JerJer: and if they have old enough firmware |
01:51.45 | km- | oh. |
01:51.46 | km- | fuck. |
01:51.49 | tzanger | km-: have you set your E&M signaling properly in zapata.conf (i.e. winktimes, etc.) |
01:52.08 | *** part/#asterisk sysdef (~sysdef@pD9560C7A.dip.t-dialin.net) |
01:52.21 | km- | asterisk wasnt running |
01:52.22 | km- | heh |
01:52.25 | km- | I'm such a moron |
01:52.34 | tzanger | km-: :-) |
01:52.36 | jayden | green is good |
01:52.44 | pr0m | back to genetic disorders. |
01:52.49 | Nugget | but it's not easy being green |
01:53.06 | Nugget | greeno shot first, anyway. |
01:53.17 | Himeko | after the pap2 has talked with vonage can't you fake their provisioning server to provide it with your own config? |
01:53.49 | JerJer | sure, provided you can provide the proper answer to the provided challenge |
01:53.50 | tzanger | Himeko: if you break the password |
01:53.58 | *** join/#asterisk {zombie} (zombie@soulasylum.penguincare.com.au) |
01:54.10 | *** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com) |
01:54.40 | Beirdo | mine's virgin and is still password locked. I think they got wise |
01:54.54 | km- | ok, the greens are flowin |
01:54.56 | km- | awesome |
01:55.03 | Himeko | if it's provosion tftp |
01:55.06 | Himeko | iwups |
01:55.07 | JerJer | Beirdo: then u don't have the right file |
01:55.14 | Beirdo | correct :) |
01:55.26 | Beirdo | I think mine's for an older version of firmware |
01:55.54 | Himeko | if it's provisioned tftp wouldn't the challange be plain text anyway |
01:56.19 | Himeko | they prolly use http provisioning though |
01:56.36 | Beirdo | if anyone has a clue, feel free to msg me :) |
01:56.37 | Beirdo | heh |
01:56.51 | *** join/#asterisk stilexip (~wow@pc-24-151-59-224.newt1.ct.charter.com) |
01:56.54 | stilexip | hello all |
01:57.15 | Frantic | guys- anyone has a problem that the show sip peers suddenly becomes empty- no sip reg is possible? (CVS from yesterday)? |
01:57.36 | Frantic | issueing stop now- doesn't do anything |
01:57.43 | Frantic | only way is to restart the machine |
01:59.08 | km- | tzanger: ok, I got the t1 up, now I've got a weird problem |
01:59.32 | km- | tzanger: when I try dialing a number, for instance, 14848759460, it only sees the 1 before it starts dialing |
01:59.36 | stilexip | can someone tell me why asterisk core dumps config.c:507 ast_internal_load: Loading Config extensions.conf via mysql engine |
01:59.44 | tzanger | what only sees the one |
01:59.44 | km- | tzanger: this is on a t1 -- shouldn't it be waiting for 3 seconds or something? |
01:59.49 | km- | tzanger: asterisk |
01:59.55 | tzanger | km-: what's your dialplan look like |
01:59.59 | stilexip | realtime works with dynmic extensions not the static file |
02:00.18 | km- | [incomingpbx] |
02:00.18 | km- | exten => _9XXX,1,Congestion |
02:00.18 | km- | exten => _1XXXXXXXXXX,1,Dial(Zap/g1/${EXTEN}) |
02:00.18 | km- | exten => _011XXXXXXXXXXXX,1,Dial(Zap/g1/${EXTEN}) |
02:00.27 | km- | calls from the legacy pbx go into incomingpbx context |
02:00.41 | tzanger | km-: ok, so no '.' use |
02:00.55 | km- | should immediate=no or immediate=yes be in the zapata.conf file for the legacy pbx's t1 line? |
02:00.56 | tzanger | that's all you have there? |
02:00.59 | tzanger | immediate=no |
02:01.02 | tzanger | you never want immediate |
02:01.08 | tzanger | unless you're wiring up a phone for BatMan |
02:01.17 | km- | group=2 |
02:01.17 | km- | immediate=no |
02:01.17 | km- | signalling=em_w |
02:01.17 | km- | context=incomingpbx |
02:01.17 | km- | channel=>33-48 |
02:01.19 | km- | that's from zapata.conf |
02:01.24 | jayden | I want immediate gratification |
02:01.47 | jayden | ok... I see an issue |
02:01.48 | tzanger | km-: that's fine, I meant what's show dialplan incomingpbx show fromthe CLI |
02:01.54 | km- | ok |
02:01.59 | jayden | dialing group 1, config you just sent was group 2 |
02:02.08 | tzanger | yeah this is true too |
02:02.12 | km- | jayden: I'm calling from g2 to g1 |
02:02.17 | jayden | ahhh |
02:02.17 | jayden | ok |
02:02.19 | km- | g1 is the CO T1 |
02:02.22 | km- | g2 is the PBX T1 |
02:02.31 | km- | *CLI> show dialplan incomingpbx |
02:02.31 | km- | [ Context 'incomingpbx' created by 'pbx_config' ] |
02:02.31 | km- | <PROTECTED> |
02:02.31 | km- | <PROTECTED> |
02:02.31 | km- | <PROTECTED> |
02:02.33 | km- | actually |
02:02.36 | km- | if I dial fast enough |
02:02.41 | *** join/#asterisk lilneon (~tj_r3@200.108.20.38) |
02:02.46 | lilneon | hi and good nigth everyone |
02:02.47 | km- | I'll get more chars |
02:02.51 | tzanger | km-: set a dial timeout |
02:02.55 | km- | on t1? |
02:02.59 | tzanger | now how to do it cleanly I'm not so sure |
02:03.00 | jayden | so, if you set somthing on incoming pbx s to just play a file, does that work (is it getting in ok?) |
02:04.11 | *** join/#asterisk xachen (xachen@edtntnt3-port-262.dial.telus.net) |
02:04.14 | lilneon | hey guys, anyone can point me in the direction of an installation for freetds with asterisk to use mssql server 2000? |
02:04.15 | jayden | then, noop(${EXTEN}) and see what your getting from pbx |
02:04.18 | km- | although I love listening to allison smith's voice say "the number you have diald is not in service" |
02:04.23 | km- | heh |
02:04.31 | jayden | so that works? |
02:04.48 | xachen | hmm |
02:04.53 | xachen | -> IAX2/guest@66.250.68.194/996 |
02:05.02 | xachen | how does a person setup direct SIP and IAX2 calls? |
02:05.07 | jayden | what about just a static dial back out zap/g2/some extn that works? |
02:05.09 | km- | *CLI> -- Starting simple switch on 'Zap/48-1' |
02:05.09 | km- | <PROTECTED> |
02:05.10 | xachen | I'm just familiar to using the PSTN |
02:05.18 | xachen | with IAX2 peer for VoIP |
02:05.20 | JerJer | lilneon: you are on your own for that one.... i have to believe not many here are going to even consider something like that much less do it |
02:05.22 | km- | that's with extension 1 setup |
02:05.38 | km- | jerjer: hey, do you know how to set a longer timeout on a dialplan? |
02:05.47 | km- | for instance, if you have a really slow typist |
02:05.49 | JerJer | which one? |
02:05.49 | km- | on the phone |
02:05.50 | tzanger | km-: DigitTimeout or something like that |
02:05.52 | JerJer | ResponseTimeout |
02:05.56 | JerJer | or DigitTimeout |
02:06.04 | tzanger | but I don't know how to do that for every single extension |
02:06.05 | tzanger | that seems wrong |
02:06.10 | jayden | bkw likes sql |
02:06.16 | km- | jerjer: is there a global responsetimeout or digittimeout? |
02:06.25 | jayden | I like sql, but have not played with it to * yet |
02:06.35 | shido6 | dialplan logic is what jerjer will learn you |
02:06.50 | tzanger | km-: I'm not aware of one |
02:06.51 | JerJer | km-: i have not seen one |
02:06.53 | tzanger | there should be one though |
02:06.59 | km- | ok |
02:06.59 | km- | so |
02:07.01 | tzanger | I mean it's absolute insanity to think you need that |
02:07.05 | km- | if I were to set it up for a particular constant |
02:07.06 | tzanger | for every single extension defined |
02:07.07 | km- | err context |
02:07.10 | tzanger | in fact you can't |
02:07.18 | JerJer | could be as simple as checking for the existance of a global variable being set |
02:07.22 | shido6 | can u rethink the dialplan to make that happen? |
02:07.23 | lilneon | jerjer: i did it before... doing the same exact thing but it wont wrk |
02:07.26 | tzanger | exten => _1NXXNXXXXXX,1,SetTimeout(10) own't work |
02:07.28 | JerJer | if so, use that value |
02:07.34 | tzanger | actually |
02:07.38 | tzanger | use immediat |
02:07.39 | tzanger | ee |
02:07.39 | km- | its gotta be defined in the source somewhere, this timeout |
02:07.43 | tzanger | immedate=yes |
02:07.45 | JerJer | lilneon: perhaps there is a reason |
02:07.45 | km- | immediate=yes? |
02:07.49 | stilexip | can you load a static config using the mysql res_mysql.so or is res_odbc required |
02:07.50 | tzanger | and s,1,SetTimeout(3) |
02:07.55 | tzanger | s,2,Read(DEST) |
02:08.05 | tzanger | s,3,Goto(dialout,${DEST},1) |
02:08.16 | tzanger | TO THE BATMOBILE |
02:08.40 | km- | theres got to be a place where that timeout is defined |
02:08.41 | jayden | everyone needs a batphone, |
02:08.43 | km- | ... |
02:08.44 | km- | I mean |
02:08.44 | lilneon | JerJer: care to share?? |
02:08.50 | km- | it's not just something the computer randomly decides |
02:08.54 | tzanger | km-: I just gave you the solution |
02:09.02 | km- | somewhere in a .c file there is "#define REALLY_SHORT_TIMEOUT" |
02:09.10 | km- | tzanger: but that's not the clean way to do it, wouldn't you agree? |
02:09.18 | lilneon | JerJer: is it that microsoft being stupid again? |
02:09.20 | jayden | timeout for what? I'm lost |
02:09.25 | tzanger | km-: actually it's the cleanest |
02:09.35 | km- | tzanger: you think so? Ok, I'll go with it then... |
02:09.36 | tzanger | and the only way I can forsee it working |
02:10.31 | *** join/#asterisk bkw_ (nobody@bkw.developer.and.friend.of.asterisk) |
02:10.31 | *** mode/#asterisk [+o bkw_] by ChanServ |
02:11.14 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
02:11.25 | jayden | bkw: somone was jsut looking to get mssql working w/ *, do you have it working, or were you just saying that ou like sql |
02:11.28 | hermie | perfect for circumventing spam filters! |
02:11.56 | km- | that Read(DEST) doesnt work |
02:12.36 | lilneon | bkw:yeah I think you wrote freetds? could you help me get it wrking? |
02:13.27 | lilneon | I keep getting a server is unavailable or does not exists when i try tsql.. and well asterisk just hangs an says unregistered mssql backend |
02:14.17 | km- | thats so weird |
02:14.22 | km- | I wish I could debug chan_zap |
02:14.24 | tzanger | km-: Read(DEST) followed by NoOp(${DEST}) -- what does the noop say |
02:14.28 | km- | matchdigit is 3000 |
02:14.51 | BuckRogers | zaptool? |
02:15.40 | km- | <PROTECTED> |
02:15.40 | km- | <PROTECTED> |
02:15.40 | km- | <PROTECTED> |
02:15.40 | km- | <PROTECTED> |
02:15.40 | km- | <PROTECTED> |
02:16.03 | tzanger | ... user disconnected? |
02:16.14 | km- | yeah |
02:16.18 | km- | the line was active the whole time |
02:16.24 | km- | but it said user disconnected after the 4 seconds |
02:16.40 | tzanger | show application read |
02:16.44 | tzanger | you might have to hit # |
02:16.49 | tzanger | or specify a max# of digits |
02:17.21 | tzanger | you may also need ||noanswer |
02:17.23 | km- | haha, I'm not going to make a whole company dial # when they've complteed the number |
02:17.31 | BuckRogers | haha |
02:17.35 | tzanger | well yeah I don't blame you |
02:17.41 | km- | shiesse |
02:17.42 | tzanger | you mught have to use multiple reads |
02:17.44 | BuckRogers | how many users |
02:17.45 | tzanger | read 3 |
02:17.45 | BuckRogers | ? |
02:17.48 | tzanger | then read 4 |
02:17.50 | tzanger | then read 3 |
02:17.56 | km- | buckroger: more than I want to have hit # :) |
02:18.03 | tzanger | then read maybe 10. :-) |
02:18.07 | tzanger | for international |
02:18.10 | BuckRogers | 20 -40 |
02:18.11 | km- | tzanger: oh my god, you cant possibly be suggesting that that's the easy way :P |
02:18.17 | km- | buckroger: yeah |
02:18.29 | km- | closer to 30 |
02:18.30 | tzanger | the timeout though should do it |
02:18.32 | BuckRogers | yeah doesnt look professonal |
02:18.50 | km- | someone else must have run into this problem before |
02:18.50 | tzanger | km-: I don't know -- there's gotta be an easier way but I'm not seeing it |
02:19.57 | km- | thats weird |
02:20.10 | km- | I just tried to put a static Dial(Zap/g1/14848759460) in there and I just got a dialtone |
02:21.06 | *** join/#asterisk verge (~jfargen@56-116.26-24.tampabay.res.rr.com) |
02:21.48 | km- | this shouldnt be this hard |
02:22.32 | tzanger | what about |
02:22.35 | tzanger | (and I cringe) |
02:22.44 | km- | well, if I have a group setup |
02:22.46 | tzanger | _.,1,DigitTimeout(4) |
02:22.57 | km- | tzanger: hmm. |
02:23.09 | ManxPower | NOOOOOOOOO!!!!!! Not _.! Anything but _.! |
02:23.11 | tzanger | and then have everything else as _NXXXXXX,**2**,Dial() |
02:23.18 | tzanger | ManxPower: well help us out here |
02:23.28 | km- | manx: do you have any ideas? |
02:23.31 | ManxPower | tzanger, What's his problem? |
02:23.34 | km- | manx: do you know the situation I'm in right now? |
02:23.35 | km- | ok |
02:23.38 | km- | lemme break it down |
02:23.43 | km- | I have a T1 from the CO going to Asterisk |
02:23.49 | km- | I have a T1 from Asterisk going to legacy PBX |
02:23.49 | ManxPower | km-, No, I've been cleaning the apartment. |
02:24.04 | km- | when I pick up the phone and dial 9 on the legacy pbx to open up a line on the t1 |
02:24.16 | km- | the timeout is set too high and the first digit I hit spawns an extension |
02:24.17 | ManxPower | km-, Loopstart? Groundstart? E&M? E&M Wink? |
02:24.20 | km- | instead of 14848759460 |
02:24.22 | km- | I get 1 |
02:24.25 | km- | as asterisk sees it |
02:24.31 | km- | em_w on both sides |
02:24.42 | *** part/#asterisk JoaoCorreia (~JoaoCorre@81.193.116.63) |
02:24.44 | ManxPower | km-, you didn't do something stupid like immediate=yes, did you? |
02:24.49 | km- | well |
02:24.51 | km- | we're trying that now |
02:24.56 | km- | immediate=no makes the 1 fire off |
02:24.59 | ManxPower | don't do immediate=yes |
02:25.02 | tzanger | ManxPower: we're trying that to read in the digits |
02:25.07 | tzanger | so we can set a digittimeout |
02:25.21 | ManxPower | I suspect you have wink problems. |
02:25.30 | km- | maybe the line isn't wink? |
02:25.43 | ManxPower | km-, What happens when you call into Asterisk? |
02:25.45 | km- | I tried em and it didnt work |
02:25.49 | km- | I tried em_w and it worjked |
02:25.57 | km- | calling into the Asterisk system from my cellphone works perfectly |
02:26.03 | km- | the IVR pops on, you can surf the directory, etc |
02:26.07 | ManxPower | km-, Have you confirmed with the telco that the line is E&M/Wink? |
02:26.13 | km- | I have not |
02:26.20 | ManxPower | km-, You don't have any DIDs? |
02:26.23 | km- | I do |
02:26.31 | km- | I tried dialing the did and I saw the correct extension |
02:26.37 | ManxPower | And do the DIDs work? i.e. do the DIDs come in as an extension? |
02:26.40 | km- | 14848759462 produced a call g1/462 |
02:26.42 | km- | yep |
02:26.47 | ManxPower | Good. |
02:26.55 | km- | I got what I was expecting on the DID dialing |
02:27.06 | ManxPower | Why isn't your PBX just collecting all the digits, stripping off the 9 and then sending them all to Asterisk? |
02:27.15 | km- | good question |
02:27.18 | km- | I can't ask the pbx to find out |
02:27.22 | ManxPower | That's the way you would want to do it with E&M/Wink. |
02:27.24 | km- | $700 worth of equipment to answer that question |
02:27.27 | tzanger | km-: if you plug the pbx directly inot the telco it works though |
02:27.30 | km- | yeah |
02:27.31 | ManxPower | km-, The answer is "lazy PBX admin" |
02:27.36 | km- | it works fine directly to the telco |
02:27.40 | km- | no |
02:27.54 | km- | this NEC electra system requires you to pay like $700 worth of manuals/software/cards so you can configure it yourself |
02:27.57 | km- | otherwise it's a service call |
02:28.05 | marc32344 | any other boards based on te410? |
02:28.19 | km- | ok, so, how come it works fine to the telco |
02:28.21 | tzanger | km-: does the electra have ATAs -- things you plug in that let you plug ordinary phones into it? |
02:28.22 | km- | but not to asterisk? |
02:28.34 | km- | tzanger: there are ata adapters for it, yes |
02:28.43 | tzanger | km-: got a part number or something I can go on in google? |
02:28.45 | km- | tzanger: but they're to feed extensions |
02:28.50 | tzanger | km-: yes exactly |
02:28.52 | km- | tzanger: the phone system? |
02:28.59 | tzanger | lets you use an ordinary cordless phone as an extension type of thing |
02:29.00 | km- | I think it's an Electra elite IPK 192 |
02:29.02 | km- | right |
02:29.27 | km- | but they're both in use |
02:29.27 | km- | what are you thinking? |
02:29.28 | tzanger | km-: totally separate problem, I'm just wondering what they're called (part number or name) in NEC land |
02:29.41 | scythelx | anyone know the cmd to perform an extensions reload through the manger interface |
02:29.50 | km- | manxpower: the weird thing is, the digittimeout in chan_zap is 3000ms, but, it's like it's only waiting 1000ms |
02:32.11 | scythelx | is it possible to issue an extensions reload command thru the manager interface |
02:34.33 | km- | thats weird |
02:34.39 | km- | I cant dial a 1 first |
02:35.08 | tzanger | km-: sounds like your phone system has a dialplan of its own doing |
02:35.20 | tzanger | km-: got a part number or model number of those NEC ATAs? |
02:35.33 | km- | just a sec, lemme see if I can rip one off the wall |
02:35.38 | km- | I think I may have mistakenly dialed 911 |
02:35.42 | km- | that'd be just great |
02:35.43 | km- | heh |
02:36.28 | tzanger | km-: hahaha |
02:37.36 | km- | SLTA-F-20 unit |
02:37.37 | km- | I think |
02:37.45 | km- | thats the only thing that looks like a part number on the PCB |
02:37.52 | BuckRogers | any one use the SBC t3 Cards |
02:38.13 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || Dev Conf 1PM CST FEB 24th -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf June 8-10th more coming soon.... |
02:38.38 | km- | SLT1-U-10ADP |
02:38.40 | km- | err |
02:38.48 | km- | SLT(1)-u10 ADP |
02:39.24 | tzanger | thanks km- |
02:40.21 | km- | thats really curious |
02:40.29 | *** join/#asterisk mikes2277 (~mike@wireless-206.222.58.98.omnilec.com) |
02:40.37 | km- | exten => s,1,Dial(Zap/24/4849191400) |
02:40.38 | km- | with that |
02:40.46 | km- | if I hit 9 on the legacy pbx, I get |
02:41.01 | km- | <PROTECTED> |
02:41.01 | km- | <PROTECTED> |
02:41.04 | mikes2277 | Does anyone know of a free activex or java phone for use in Internet Explorer? |
02:41.09 | km- | dialtone continues on the legacy pbx |
02:41.12 | km- | but my cellphone does ring |
02:41.30 | tzanger | km-: uh |
02:41.38 | km- | oh man that's odd |
02:41.45 | km- | if I dial a digit to get rid of the dialtone |
02:41.48 | km- | I can then hear the ringing |
02:42.17 | tzanger | is it the PBX dialtone or asterisk's dialtone |
02:42.26 | km- | I think it's the pbx's dialtone |
02:42.39 | ta[i]nted | you guys ever use DISA? |
02:43.04 | km- | [ Context 'incomingpbx' created by 'pbx_config' ] |
02:43.04 | km- | <PROTECTED> |
02:43.14 | km- | that should say that any invalid digit should dial that right |
02:43.24 | km- | I dialed 1, I got |
02:43.28 | km- | <PROTECTED> |
02:43.28 | km- | <PROTECTED> |
02:43.28 | km- | <PROTECTED> |
02:43.32 | ManxPower | km-, I belive Asterisk expects a DID on E&M/Wink |
02:43.46 | km- | hrm? |
02:43.59 | ManxPower | km-, That's what it looks like to me. |
02:44.08 | ManxPower | You may need to use DISA |
02:44.12 | marc32344 | any other cards that does the same as digium te410p? |
02:44.23 | JunK-Y | TE405P? |
02:44.29 | lilneon | alrighty then.. has anyone gotten the mbrola voices to wrk with festival? |
02:44.35 | ta[i]nted | sometimes i can't get DISA to return dialtone.. was wondering if carriers block it |
02:44.50 | tzanger | marc32344: what, *exactly* is your question or what are you trying ot do |
02:45.07 | tzanger | you are spitting out odd little questions and not getting any decent results, so how about some fresh tactics |
02:45.30 | ta[i]nted | tzanger it's a super secret squirrel project |
02:45.39 | tzanger | ta[i]nted: :-) |
02:45.40 | BuckRogers | yeah like repeating your self |
02:45.54 | BuckRogers | so any one use those sbe t3 cards that work with linux |
02:46.11 | tzanger | BuckRogers: IIRC they wont' work with * |
02:46.13 | tzanger | they're unchannelized |
02:46.17 | km- | manx: is it possible I need feature D or something? |
02:46.27 | km- | manx: feature D is em_w with goodies? |
02:46.31 | BuckRogers | they say they do |
02:46.43 | tzanger | they say they work with asterisk? |
02:46.51 | BuckRogers | yeah i was researching it |
02:46.56 | tzanger | BuckRogers: interesting |
02:47.03 | km- | uh. |
02:47.08 | BuckRogers | yeah i read it on a google news articel |
02:47.23 | BuckRogers | but ive yet to meet anyone using them |
02:47.34 | km- | holy shit dudes |
02:47.35 | km- | get this |
02:47.38 | BuckRogers | so what do u use for asterisk if you want t3 |
02:47.59 | km- | <PROTECTED> |
02:47.59 | km- | Feb 22 21:18:08 WARNING[1882]: chan_zap.c:4748 ss_thread: Got a non-Feature Group D input on channel 48. Assuming E&M Wink instead |
02:47.59 | km- | <PROTECTED> |
02:47.59 | km- | <PROTECTED> |
02:47.59 | km- | <PROTECTED> |
02:48.00 | BuckRogers | channelized t3 |
02:48.01 | km- | <PROTECTED> |
02:48.03 | km- | <PROTECTED> |
02:48.05 | km- | <PROTECTED> |
02:48.07 | km- | I switched it to feature d |
02:48.09 | km- | and it whines and switches to em_w |
02:48.11 | km- | but then it works fine |
02:48.20 | km- | if I say em_w specifically |
02:48.22 | km- | nothing seems to work |
02:48.38 | tzanger | BuckRogers: M13 and a bunch of systems with TE410? |
02:48.40 | BuckRogers | right on (km) |
02:49.01 | BuckRogers | te410 isnt that quad span t1 |
02:49.06 | tzanger | yes |
02:49.17 | BuckRogers | and split the t3 up |
02:49.24 | BuckRogers | thats alot of t1's |
02:49.25 | tzanger | BuckRogers: I did just say M13 |
02:49.30 | BuckRogers | ahh |
02:49.40 | tzanger | four decent systems should be able to handle it |
02:49.42 | BuckRogers | refresh my memory M13 |
02:49.42 | km- | tzanger: whatcha think about that? |
02:49.46 | tzanger | 4x8 = 32 |
02:49.47 | km- | tzanger: that's kinda weird, huh? |
02:49.51 | tzanger | km-: very |
02:51.24 | BuckRogers | my company is considering stepping up to t3 but their not sure what type of hard ware to get the engineers |
02:53.28 | BuckRogers | there talking dual 64 bit g5's unix systems |
02:53.42 | BuckRogers | possible quads |
02:53.47 | *** join/#asterisk Firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
02:53.49 | *** join/#asterisk lilneon_ (~tj_r3@200.108.20.38) |
02:54.01 | lilneon_ | hi again.. got booted |
02:54.20 | Firestrm | hi |
02:54.44 | lilneon_ | Firestrm: any experience getting festival and mbrola voices up and running? |
02:54.59 | Firestrm | none whatsoever.. |
02:56.10 | Firestrm | my struggles revolve around fighting with sipura for tech support on their product |
02:56.32 | Firestrm | i'll play with festival after i get hardware running.. |
02:56.41 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
02:56.46 | BuckRogers | sipura i emailed them over 2 weeks ago |
02:57.20 | BuckRogers | no response |
02:58.11 | Firestrm | BuckRogers, im glad i bought this one to play with before commiting it to my project.. i would NEVER use sipura gear for any commercial install.. |
02:58.49 | km- | tzanger: did you find any data on that NEC system? |
02:58.52 | BuckRogers | yeah really |
02:58.57 | km- | tzanger: it's doing more weird shit |
02:59.03 | tzanger | km-: not much but some |
02:59.03 | BuckRogers | there gsm is not there and they advertise it |
02:59.12 | tzanger | I'm looking for something that'll work with an electra elite 48 |
02:59.24 | km- | tzanger: that's the same hardware but smaller chasis I think |
02:59.29 | km- | I may have a 48, who knows |
02:59.36 | marc32344 | how many call mins can be terminated on a T1? |
02:59.37 | Firestrm | BuckRogers, i would be happy if i could get rid of the echo on the pstn line.. |
02:59.38 | km- | I know that the electra elite 48 sounds familiar |
02:59.51 | tzanger | baby brother to hte IPK I think |
03:00.01 | BuckRogers | did you go into the html interface |
03:00.10 | BuckRogers | i think there is echo cancelation |
03:00.16 | BuckRogers | on the ata |
03:00.56 | Firestrm | BuckRogers, yes.. and upgraded firmware... and read every article i can get on the matter... it seems that the echo cancelation on the s3k is just plain broken.. |
03:01.17 | tzanger | km-: hmm |
03:01.20 | BuckRogers | yeah they are really not reliable |
03:01.41 | Firestrm | BuckRogers, they are ok for residential toys.. but not for business.. |
03:01.48 | |Vulture| | has anyone tried using the TDM120? |
03:01.55 | tzanger | km-: you don't happen to have a manual for those ADPs do you? I'm trying to see if I can notify an extension of mail |
03:01.58 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
03:02.03 | tzanger | with Norstar you can flash, then *1+exten |
03:02.23 | BuckRogers | no they designed ciscos old ata and vonage had huge probs with it |
03:03.08 | BuckRogers | then they switched to motorala |
03:03.41 | Firestrm | BuckRogers, i didnt realize motorola was in the spa market.. |
03:04.31 | BuckRogers | yeah the vt1000 |
03:04.37 | BuckRogers | is their voip ata |
03:05.04 | *** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
03:05.10 | lilneon_ | hey guys, other than ebay anywhere someone can find a cheap channel bank? |
03:05.24 | lilneon_ | which works with asterisk |
03:05.48 | tzanger | lilneon_: ebay. :-) |
03:05.51 | km- | hmm |
03:05.59 | Firestrm | lilneon_, i saw one somewhere for 1000.00 |
03:06.05 | km- | when I try to dial the DID to the legacy pbx, I get congestion tone from the PBX |
03:06.06 | lilneon_ | tzanger: other than ebay? |
03:06.48 | *** join/#asterisk verge (~jfargen@56-116.26-24.tampabay.res.rr.com) |
03:07.11 | verge | can someone help me with a problem I am experiencing with caller-id? |
03:07.52 | km- | oik |
03:07.58 | km- | I can make the phone ring, but only if I have the line in em_w |
03:07.59 | km- | hahaha |
03:08.02 | km- | BALLS |
03:08.33 | km- | back to square one on this |
03:08.34 | tzanger | km-: that sounds one one amazingly frustrating problem |
03:08.36 | km- | it is definitely em_w |
03:08.46 | km- | because em_w is the only way I can get the extensions to ring |
03:08.56 | km- | but, that brings back the dial timeout problem |
03:08.58 | topping | Firestrm: http://www.channelbanks.com//pages/specifications.html ? |
03:09.34 | Firestrm | topping, ya i thin it was a rhino unit.. |
03:09.53 | topping | still pretty expensive at $1495... such a little turd of a box inside i'm sure |
03:10.07 | topping | probably costs $75 to make |
03:10.23 | Firestrm | topping, i know.. im tempted to devel an open hardware design.. |
03:10.25 | topping | most of that in connectors |
03:10.40 | km- | hmm, I've got 20 mins before my window is over |
03:10.46 | topping | there's probably a hundred people thinking the same thing tho lol |
03:11.09 | topping | i'm getting my dad to bring asterisk into his uni |
03:11.12 | Firestrm | topping, yes.. but i have the technology to do it... its just time and money i lack.. |
03:11.20 | topping | wow, that's nice |
03:11.50 | tzanger | those are nice |
03:12.02 | topping | whoooo |
03:12.09 | km- | tzanger: you think I need to tweak prewink/postwink times or anything like that? |
03:12.16 | *** join/#asterisk Sedorox (~Sed@Neptune.client.wlgrv.pa.Sed6.net) |
03:12.20 | Firestrm | cany you say.. multilayer PCB's in 1 hour? |
03:12.31 | topping | damn sam |
03:12.39 | tzanger | km-: you can try it but I'm at a total loss as to which direction to do it in |
03:12.47 | km- | damn |
03:12.48 | tzanger | like I said it almost sounds like your phone system is not doing its job |
03:12.50 | km- | I need kram here |
03:13.10 | km- | he'll probably know how to work around this without having to configure the pbx differently |
03:13.11 | topping | it's like that eyeglass place... "bifocals, in about an hour" |
03:13.34 | shido6 | luv Toonie Tuesday |
03:13.43 | Firestrm | topping, i can go right from protel to board, without even leaving my chair.. assembly requires that i walk the finished board over to the pick&place machine and then to the reflow oven |
03:13.58 | shido6 | 2 pcs of (say it with me.. ) Shicken and Fries for 2 dollars. |
03:14.12 | shido6 | 2 cdn dollars |
03:14.19 | topping | Firestrm: that's nuts... that's so nice |
03:14.24 | topping | no access any more? |
03:14.31 | topping | has now! |
03:14.33 | topping | wow |
03:14.34 | Firestrm | shido6 2 northen paso's |
03:14.45 | topping | can you use it for your own stuff? |
03:15.07 | tzanger | shido6: I am a fan of the KFC tuesday too |
03:15.09 | Firestrm | topping, im %49 partner in the company.... ummmm... yes.. |
03:15.14 | topping | hehehe |
03:15.16 | mikegrb | ~insult Beirdo |
03:15.33 | tzanger | shido6: it's $2.22 though is it not |
03:15.40 | shido6 | yeah yeah |
03:15.42 | shido6 | :) |
03:15.44 | tzanger | I get the 3 pcs of chicken + fries with gravy + drink for like $4 something |
03:16.12 | tzanger | Firestrm: ehat does your company do that it has its own SMT P&P line |
03:16.21 | verge | can anyone assist me with my callerid issue? |
03:16.38 | tzanger | verge: just spit it out and we'll do ewhat we can |
03:16.50 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
03:17.13 | Firestrm | tzanger, yes and we are probbly going to add VPR for bga an a month |
03:17.20 | km- | tzanger: did you find anything about configuring through the handset how the dialplan works? |
03:17.35 | km- | tzanger: for the electra elite systems |
03:17.40 | tzanger | Firestrm: nice, what are you, contract builder? |
03:17.41 | km- | there's a setup menu you can access from any phone |
03:17.48 | km- | where I can fiddle with DID's and shit |
03:17.51 | Firestrm | tzanger, aerospace r&d |
03:17.52 | tzanger | km-: uh, no... heh |
03:17.58 | tzanger | Firestrm: *nice* |
03:18.04 | km- | yeah, that is some cool shit |
03:18.10 | km- | make us some te405p's cheap! :) |
03:18.19 | topping | hey, so i'm wondering about cheap commercial-grade voip handsets |
03:18.19 | BrianR___ | Does anyone know if callprogress=yes can detect dialtones? |
03:18.26 | topping | any favorites? |
03:18.29 | Firestrm | km- im thinking about doing a run.. have the files.. need the time |
03:18.36 | km- | firestrm: hehe |
03:18.38 | *** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com) |
03:18.44 | tzanger | uh |
03:18.44 | km- | firestrm: if you're charging $50, I'm there! |
03:18.52 | tzanger | the TE405P is not an open design |
03:19.05 | Firestrm | km, how about parts and free beer at the next von? |
03:19.34 | km- | firestrm: hehe, I'd be buying it for my own fiddling, so, I don't think Mark would mind. Parts are expensive for it though I believe |
03:20.09 | Firestrm | km, i think the dsp is the biggie.. but we get very cut rate deals on parts.. |
03:20.32 | tzanger | ... I'm almost certain there is no DSP on TE405 |
03:20.34 | tzanger | or any Zapata |
03:20.39 | tzanger | it's the design of the beast |
03:21.16 | Firestrm | tzanger, whats the big bga chip on the board then? i was sure it was a dsp.. |
03:21.22 | tzanger | ... |
03:21.25 | tzanger | there isno BGA on the TE405P |
03:21.34 | kpfleming | there is no DSP on those boards |
03:21.44 | tzanger | http://www.mixdown.ca/~asterisk/ |
03:21.45 | Firestrm | tzanger, i must have different design docs.. |
03:21.46 | tzanger | that's the TE405P |
03:22.02 | km- | heh |
03:22.33 | tzanger | there's only two main chips on there, the quad framer and the PCI bridge/FPGA |
03:22.40 | tzanger | there's a configuration EEPROM too but that's it |
03:23.22 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
03:23.22 | *** mode/#asterisk [+o bkw_] by ChanServ |
03:23.36 | BrianR___ | poor jtn |
03:23.42 | km- | bkw_: dude! |
03:23.54 | km- | BrianR: are you the same brianr from #linuxos? |
03:24.07 | BrianR___ | km-: Yes. Haven't been there for many ages though./ |
03:24.16 | Firestrm | my mistake.. i mean the FPGA part.. |
03:24.18 | topping | hey, so i'm wondering about cheap commercial-grade voip handsets... any ideas? there's a hundred of them on voip-info, isn't there a new one that is really solid for about $100 |
03:24.34 | topping | (less is fine too :-) ) |
03:24.45 | tzanger | Firestrm: well as I said I'm fairly certain that the design is not open |
03:24.55 | tzanger | it's not the same card as on zapatatelephony.org |
03:25.07 | km- | brianr: I'm veneficus |
03:25.11 | kpfleming | it's not, but it would not be hard to replicate if you wanted to badly enough... but the parts are not going to be easily available |
03:25.33 | tzanger | kpfleming: the design between the T400P and TE405/TE410P are siginficantly different |
03:25.36 | BrianR___ | Considering the amount of time I spend on #linuxos, it's suprising I don't remember any of the people from there :( |
03:25.41 | kpfleming | yes, very very much so |
03:25.48 | km- | bkw_: dude, you gotta wake the hell up, I need your excellent tutiledge |
03:25.58 | tzanger | MOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOSE PENIS |
03:26.01 | km- | well |
03:26.02 | tzanger | is he awake yet? |
03:26.05 | km- | hahahaha |
03:26.08 | km- | the emergency wake up call |
03:26.14 | BrianR___ | km-: I've been doing some asterisk hacking lately though.. Thus my presence on this channel lately. |
03:26.20 | km- | MOOSE PENIS!MOOSE PENIS!MOOSE PENIS!MOOSE PENIS! |
03:26.26 | km- | brianr___: hehe, splaspood is here too |
03:26.29 | Trionnis | moose penis? |
03:26.31 | km- | brianr: he idles all the time though |
03:26.38 | km- | trionnis: it's the emergency bkw_ call |
03:26.38 | Trionnis | ... |
03:26.41 | Trionnis | ahhh |
03:26.45 | BrianR___ | km-: Heh. I remember that nick.. |
03:26.46 | km- | trionnis: if he's around, he'll respond to that :) |
03:26.52 | Trionnis | ;) |
03:26.58 | km- | well |
03:27.00 | BrianR___ | km-: I still idle on efnet... Usually in #978. |
03:27.02 | km- | I do have one thing on my side |
03:27.05 | BrianR___ | km-: RevThresh hangs there also. |
03:27.06 | km- | brianr: haha |
03:27.22 | km- | brianr: I saw a guy named "Thresh" on my server on world of warcraft, was wondering if it was the same guy |
03:27.32 | km- | brianr: my ex gf used to hang out in #978 too |
03:27.33 | BrianR___ | No idea if he plays warcraft... |
03:27.34 | Trionnis | anyone have an spa-1001 they want to sell cheap to a fellow * goon? :D |
03:27.41 | BrianR___ | km-: Who's that? |
03:27.45 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
03:27.48 | km- | tzanger: at least there's no echo on the T1 |
03:27.52 | km- | brianr: a chick named Meridith |
03:27.58 | km- | brianr: she was friends with another dude in #978 |
03:27.59 | tzanger | km-: give it a week |
03:28.02 | BrianR___ | km-: Heh. I remember her.. |
03:28.05 | tzanger | our echo is VERY destination dependent |
03:28.07 | BrianR___ | She has the biggest cans I've ever seen. |
03:28.08 | BrianR___ | Ever. |
03:28.14 | km- | yeah, she had huge boobs |
03:28.23 | tzanger | who |
03:28.24 | Firestrm | tzanger, not open design.. but also not hard to cleanroom reverse engineer either.. the boards im thinking of are the single t1/e1 tormenta |
03:28.26 | km- | my ex g/f |
03:28.29 | km- | brianr was aware of her |
03:28.37 | tzanger | Firestrm: ahh well yes that is true enough |
03:28.40 | km- | she was a damn slut though |
03:28.50 | tzanger | you can take apart a 6 layer board and make gerbers from them pretty easily |
03:28.52 | km- | although good in bed |
03:28.53 | km- | heh |
03:28.56 | Trionnis | you dated my ex wife? |
03:28.56 | km- | anyway |
03:28.58 | Trionnis | ?? |
03:29.01 | km- | trionnis: hahahaha |
03:29.03 | Trionnis | ;) |
03:29.17 | km- | damn |
03:29.19 | km- | there goes my window |
03:29.26 | tzanger | anyway |
03:29.28 | tzanger | I gotta get to bed |
03:29.30 | Trionnis | seriously tho |
03:29.34 | Firestrm | tzanger, no you dont need to copy the artwork, its usually easier to redesign based off of lessons learnt from the origial |
03:29.34 | tzanger | later all, sorry I couldn't be more help km- |
03:29.40 | Trionnis | anyone point me toward a cheapie sipura? |
03:29.44 | Trionnis | :) |
03:29.53 | tzanger | Firestrm: seems like too much of a pain in the ass |
03:29.56 | km- | thanks tzanger |
03:30.00 | km- | I'll hit kram up later I guess |
03:30.06 | tzanger | Firestrm: at the very least make the damn thing 3.3V/5V compatible |
03:30.20 | Firestrm | tzanger, hence the time and money problem between me and dsigning an open hardware design |
03:30.34 | *** join/#asterisk syslod (~yurplsl@65.114.0.198) |
03:30.34 | BrianR___ | km-: I'm doing a legacy pbx integration and the damn ATA's for the legacy PBX don't have disconnect supervision :( |
03:30.41 | tzanger | Firestrm: that and the fact that you're going out of your way to hurt asterisk and digium |
03:30.49 | km- | brianr: that sucks, I think I heard another person having that problem |
03:30.52 | syslod | Hello |
03:30.55 | ManxPower | BrianR___, That's pretty standard. |
03:30.59 | tzanger | from a business sense it might make sense but from a good karma sense it stinks |
03:31.03 | ManxPower | Plug an Asterisk FXS port into the PBX CO port. |
03:31.11 | tzanger | especially since you're not pushing the tech forward at all |
03:31.15 | km- | ok guys, going home, it's been 14 hours now I've been at this hell |
03:31.19 | km- | hehe |
03:31.21 | Firestrm | tzanger, i make sure to set 3.3/5v operation via a binary jumper set :) |
03:31.25 | Trionnis | only 14? |
03:31.26 | Trionnis | n00b |
03:31.36 | km- | brianr: good hearing from you again, hope to see ya around |
03:31.45 | km- | I'm going to start coming back here now since I'll actually have an active asterisk implementation |
03:31.50 | km- | trionnis: haha, you're cruisin buddy! |
03:31.55 | Trionnis | ;) |
03:31.56 | km- | don't make me get out the fishing pole! |
03:31.56 | tzanger | hahaha |
03:32.01 | Trionnis | OOH! |
03:32.05 | Trionnis | don't tease me like that |
03:32.07 | tzanger | "I don't use wearable computers because my tie keeps getting caught in the CPU fan!" |
03:32.09 | Trionnis | =| |
03:32.09 | km- | ok, that was a world of warcraft reference |
03:32.14 | Trionnis | owww |
03:32.16 | Trionnis | damn |
03:32.22 | BrianR___ | ManxPower: Won't do it for me... There's some special signalling I can put out on the pbx fxs ports that I can't on the fxo ports.. |
03:32.25 | km- | the fishing pole you can use for fishing, but you can hit people with it for like 5 damage |
03:32.30 | Trionnis | o |
03:32.33 | km- | and I beat the crap out of this guy with the fishing pole and won in a duel |
03:32.33 | Trionnis | didn't know that |
03:32.36 | Trionnis | hahaha |
03:32.37 | Trionnis | nice |
03:32.38 | Trionnis | :) |
03:32.44 | Firestrm | tzanger, reverse engineering allmost allways brings the design forward.. its just part of the deal.. think of it as stepping on other to get a leg up on the competition.. Standard business practice |
03:32.53 | km- | yeah, so whenever I threaten people now I say "don't make me get out the fishing pole!" |
03:32.58 | Trionnis | noted |
03:32.59 | Trionnis | ;) |
03:33.00 | BoRiS | ~seen paulc |
03:33.02 | jbot | paulc <~paulc@S010600062586a0b4.vc.shawcable.net> was last seen on IRC in channel #asterisk, 22h 36m 1s ago, saying: 'Is it me, or are there a handful of guys in the final 12 who are just fecking awful?'. |
03:33.02 | km- | hahaha |
03:33.06 | BrianR___ | ManxPower: I think I may have to do some hacking on the code surrounding the progress stuff... |
03:33.11 | syslod | Firestrm: U building a DS3 card with DSPs? |
03:33.13 | km- | ok, later crew |
03:33.16 | *** part/#asterisk km- (~km-@67.105.178.130) |
03:33.17 | Trionnis | adios |
03:33.17 | tzanger | Firestrm: meh. I've been reverse engineering for over a decade professionally and probably half that again as a hobby |
03:33.34 | BrianR___ | km-: Yes.. good seeing some old nicks... Smallw orld.. |
03:33.38 | BrianR___ | hmm.. missed him.. |
03:33.39 | tzanger | unless there's SERIOUS coin to be made at it it's not generally worth starting lower than ground zero to get the design out |
03:33.40 | syslod | meetme brokey in head??? |
03:34.00 | Firestrm | syslod, im concidering designs.. i feel that i may need an open hardware design project to play with... |
03:34.05 | ta[i]nted | tzanger how do u start lower than ground zero? u mean moral grounds? |
03:34.10 | tzanger | no |
03:34.19 | tzanger | ground zero means you know the concepts and base ideas |
03:34.32 | tzanger | below ground zero you're starting with the card and none of the "higher level" data |
03:34.34 | ta[i]nted | no ground zero means ground zero |
03:34.47 | tzanger | ta[i]nted: not in design. :-) |
03:34.51 | BrianR___ | It can send a DTMF tone to signal disconnect inband. I guess there's some code in CVS to take advantage of that... |
03:35.00 | BrianR___ | Dialtone detection would make my life easiest, but.. |
03:35.15 | ta[i]nted | then u mean clean slate? |
03:35.19 | tzanger | there's a lot of art and black magic in design that unless you're there for it, you're below ground zero |
03:35.23 | ta[i]nted | square one? |
03:35.25 | tzanger | ta[i]nted: kind of, yeah |
03:35.41 | syslod | Hell, Now I'm getting "that is not a vaild conference" |
03:35.48 | ta[i]nted | there is a lot of cut n paste in design |
03:35.56 | tzanger | yes |
03:36.02 | Firestrm | tzanger, ground zero kind implies that you tried it and i blew up in your face :) |
03:36.23 | Trionnis | can anyone suggest a cheap fxs solution? |
03:36.30 | *** join/#asterisk file (~file@mctn1-8179.nb.aliant.net) |
03:36.32 | Trionnis | i.e. ~50usd? |
03:36.36 | tzanger | Firestrm: ;-) |
03:36.40 | tzanger | wouldn't be the first time |
03:36.46 | ta[i]nted | Trionnis PSTN? |
03:36.50 | BrianR___ | $10 ebay fx0 card and a battery? :) |
03:36.54 | tzanger | I'm actually starting project where I get to parallel up VFD power sectiosn |
03:37.00 | Trionnis | ? |
03:37.02 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
03:37.05 | Trionnis | don't know that method apparently |
03:37.11 | Firestrm | tzanger, its the smoke that makes the chips work.. once you let the smoke out.. they dont work anymore... |
03:37.13 | tzanger | thinking of using serdes chipsets to syncrhonize everything |
03:37.16 | greg_work | if i have NUM = 1 in [globals], is SetGroup(OUT_${NUM}) going to make the group OUT_1 ? |
03:37.25 | tzanger | We're talking 250kW and up here, lots of smoke |
03:37.32 | greg_work | or do I have to use $[OUT_${NUM}] ? |
03:37.34 | tzanger | except it's not in the chips per se but moreso in the caps and the IGBTs |
03:37.59 | Trionnis | BrianR___: got a link for some info on that "trick" ? |
03:38.00 | tzanger | it'll be an interesting project, and I have my OHSA-approved PPE on hand |
03:38.07 | tzanger | I have a feeling I'll need it for this |
03:38.08 | *** join/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net) |
03:38.20 | ta[i]nted | PPE = Power Point Expert? |
03:38.22 | BrianR___ | Trionnis: search for fxo to fxs adaptor for ready-made solutions |
03:38.24 | ta[i]nted | lol |
03:38.30 | tzanger | anyway |
03:38.31 | tzanger | bedtime |
03:38.32 | tzanger | later all |
03:38.37 | *** join/#asterisk seawolf_ (~seawolf_@sea.slackwolf.com) |
03:38.40 | Firestrm | tzanger, if you want real fun.. try accidentally pointing a weather radar at the com antenna of another aircraft and turning it on for 1/2 hour.. |
03:38.55 | Firestrm | S.M.O.K.E |
03:39.04 | BrianR___ | You actually need slightly more than just a battery if you want real 90v style ringing |
03:39.08 | tzanger | Firestrm: gimme a job in aerospace and I bet I can find more fun than that with RADAR |
03:39.14 | Firestrm | lol |
03:39.28 | Firestrm | why do i feel warm all of a sudden :) |
03:39.39 | Firestrm | ive had that happen to me as well.. |
03:39.47 | tzanger | Firestrm: well yeah... a system expecting -100dBm getting a few hundred Watts of RF directed at it might not take it so nicely |
03:39.48 | Trionnis | the commercial stuff is putting the price up around the same as a new spa-1001 on ebay |
03:39.50 | Trionnis | :( |
03:39.53 | Trionnis | oh well |
03:40.20 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
03:40.21 | tzanger | especially if you're emitting at the same wavelength the system's tuned to sniff out of noise |
03:40.37 | ta[i]nted | that'd break stuff |
03:40.44 | ta[i]nted | expensive stuff |
03:40.48 | Firestrm | tzanger, not at all... also makes a good way of killing Photoradar sets.. just need a traveling wave tube, waveguide, power supply = fun |
03:41.06 | tzanger | ta[i]nted: yes, but if we knew what we were doing, it wouldn't be called research |
03:41.29 | tzanger | <-- R&D manager for a power electronics Mfg |
03:41.41 | ta[i]nted | what's that one field of RF.. TEMPEST? |
03:41.51 | Firestrm | tzanger, high voltage? |
03:41.53 | tzanger | fuck tempest |
03:41.57 | tzanger | UWB is where the cool shit's at |
03:42.03 | tzanger | medium voltage |
03:42.10 | Trionnis | doh |
03:42.11 | ta[i]nted | UWB is UWGay |
03:42.20 | Trionnis | BPL stuff? |
03:42.24 | *** part/#asterisk seawolf_ (~seawolf_@sea.slackwolf.com) |
03:42.28 | tzanger | low/medium is where we play mos to fhte time but we do some 13kV+ stuff on occassion |
03:42.42 | ta[i]nted | tzafrir_home over what medium |
03:42.45 | tzanger | yes I have HF HV stories to tell from my youth :-) |
03:42.47 | ta[i]nted | powerline? |
03:42.50 | ta[i]nted | wireless? |
03:42.51 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
03:42.53 | tzanger | ta[i]nted: yes this is industrial control |
03:43.00 | tzanger | motor starters and VFDs |
03:43.05 | tzanger | up to 22000HP |
03:43.08 | jayden | ~Guatamala |
03:43.09 | jbot | methinks guatamala is where the examples end |
03:43.16 | *** join/#asterisk sricard (sricard@HSE-Montreal-ppp133166.qc.sympatico.ca) |
03:43.38 | BrianR___ | is there a cvsweb for the asterisk source code? |
03:43.46 | Firestrm | tzanger, my two favorite things to play with as youth were,, baloons filled with hydrogen and tesla coils.. hmm i wonder what you could do with those two things? :) |
03:43.58 | jayden | ~Galapagos |
03:44.01 | Trionnis | win a darwin award? |
03:44.03 | Trionnis | => |
03:44.08 | ta[i]nted | lol |
03:44.14 | sricard | can someone help me with tdm400? |
03:44.27 | jayden | jbot, Galapagos is an island, with turtles and stuff |
03:44.28 | jbot | okay, jayden |
03:44.33 | Firestrm | Trionnis, lol.. some close calls that did allmost result in a nomination :) |
03:44.38 | Trionnis | laf |
03:44.40 | Trionnis | :) |
03:44.58 | sricard | it was working for a while, then installed openldap and other stuff on my gentoo and now, it does not work anymore |
03:45.02 | *** join/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net) |
03:45.08 | Firestrm | as it was.. our neibours cat was highly tramatized... |
03:46.01 | sricard | refreshed zaptel and asterisk from cvs, recompiled the kernel, zaptel and asterisk and still the same |
03:46.13 | Firestrm | there is NOTHING on this planet that is louder than a garbage bag filled with acetoline and oxygen, ignited by spark.. |
03:46.41 | Firestrm | at 3:00 am ... im rotfl just remembering it .. |
03:46.46 | sricard | i get may errors starting with -> chan_zap.c:769 zt_open: Unable to specify channel 1: No such device or address |
03:46.50 | syslod | sricard: zttool? |
03:47.43 | Firestrm | sricard, i get that on occasion.. lsmod to make sure wcfxo is there... |
03:48.03 | Firestrm | for some reason it occasionaly does not start properly |
03:48.31 | sricard | zttool says the card is not configured |
03:49.00 | sricard | lsmod show both zaptel and wctdm loaded |
03:49.19 | Damin | Firestrm: Try inflating a 4 foot tall Truck intertube to 80 PSI w/ Acete/O2 and throwing it onto a bonfire. ;) |
03:49.36 | Firestrm | sricard, mosprobe wcfxo then ztcfg -vvvvv |
03:49.37 | ManxPower | Um, not thanks |
03:49.51 | Firestrm | Damin. that work too.. |
03:50.15 | sricard | Channel map: |
03:50.16 | sricard | Channel 01: FXO Kewlstart (Default) (Slaves: 01) |
03:50.16 | sricard | Channel 04: FXS Kewlstart (Default) (Slaves: 04) |
03:50.16 | sricard | 2 channels configured. |
03:50.17 | Damin | Firestrm: Better stand back though.. ;) We put an 8 foot wide hole in the ground with that little trick, and completely blew the fire out. |
03:50.46 | Firestrm | Damin, with the garbage bag method you can use a model rocket igniter on a timer, and be a safe (read away from arrest) distance away when it goes off |
03:51.14 | sricard | all look fine there right? |
03:51.42 | Firestrm | sricard, now try zttool |
03:52.02 | Trionnis | (read away from arrest) <-- lol |
03:52.21 | Trionnis | "who gives a shit if I lose an arm... as long as I don't get arrested!!" |
03:52.30 | sricard | Firestrm, it sees it fine |
03:52.54 | sricard | it says 4 channels and 2 configured |
03:53.13 | Sedorox | channel one and four |
03:53.14 | Sedorox | thats two |
03:53.27 | sricard | now asterisk started...... man, i hate those weird one |
03:53.29 | sricard | s |
03:53.53 | sricard | zttool says 4 channels for a tdm400 |
03:54.02 | Firestrm | sricard, it does that to me once in a while.. for some reason wcfxo doesnt want to start the first time.. |
03:54.07 | Sedorox | how many modules you have on it? |
03:54.16 | sricard | 2 |
03:54.26 | Sedorox | then you only have two channels.. there are 4 available... |
03:54.47 | sricard | understand... |
03:54.51 | Sedorox | hehe |
03:55.10 | Sedorox | because your not using channels 2 and 3 |
03:56.08 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
03:56.26 | *** part/#asterisk lilneon_ (~tj_r3@200.108.20.38) |
03:56.28 | sricard | for a tdm400, do you use wctdm or wcfxo? |
03:56.34 | Firestrm | Trionnis, this is what happens when you have neibourhood firecracker wars evolving into an arms race :) |
03:56.40 | Sedorox | I guess wctdm |
03:57.04 | Firestrm | sricard, wcfxo |
03:57.28 | Firestrm | wctdm is t1/e1... i think.. |
03:57.34 | sricard | Sedorox -> was just reading zttool which says: Span 1: 4 total channels, 2 configured |
03:57.49 | Sedorox | there ya go ;-) |
03:58.19 | sricard | i don't have wcfxo and have wctdm and it is wokring.... |
03:59.00 | kpfleming | wctdm is the new name for the old wcfxo driver |
03:59.18 | sricard | wctdm 30752 2 |
03:59.18 | sricard | zaptel 182272 8 [wctdm] |
03:59.36 | sricard | kpfleming, so i am ok then? |
03:59.40 | Firestrm | time to make sup... b.b.l8tr |
03:59.48 | Sedorox | looks like it |
04:00.00 | sricard | thanks folks |
04:00.10 | kpfleming | sricard: probably so, yes |
04:00.16 | |Vulture| | Anyone purchse a ipVolution T1 card yet? |
04:00.41 | sricard | spent about 2-3 hours researching on the net and try stuff and i come here,i get it to work in 5 :-) |
04:00.55 | Sedorox | hehe |
04:01.09 | Sedorox | sricard: depends.. sometimes the other way is better.. sometimes not |
04:01.19 | sricard | thanks kpfleming,Sedorox,Firestrm |
04:02.44 | sricard | ttyl everyone |
04:03.24 | Sedorox | enjoy |
04:04.16 | *** join/#asterisk Inv_arp (junya@adsl-8-230-5.mia.bellsouth.net) |
04:07.01 | *** join/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net) |
04:07.16 | Firestrm | np sricard |
04:08.53 | convey | Anyone have a sample config for a Sipura 841 phone? |
04:09.24 | sricard | convey: for the sip channel? |
04:09.46 | convey | lol |
04:09.49 | sricard | convey: 841 are configured in asterisk as sip channels |
04:09.55 | convey | I am a serious newbie |
04:10.08 | *** join/#asterisk astermex (~mmg@einstein.transtelco.com.mx) |
04:10.19 | convey | I was following the quick start guide. |
04:10.28 | convey | I defined the phone inthe sip.conf file |
04:10.49 | convey | then added the dial plan for the estension in the extensions.conf file |
04:11.21 | sricard | convey: and configured the 841 by using its web interface? |
04:11.26 | convey | when I type sip show peers in the CLI it shows the phone as offline |
04:11.43 | astermex | After upgrading kernel to support data in zaptel and recompiling asterisk I always get a seg fault when I try to start asterisk. Zaptel module is loading witout any problems. Do I need to recompile something else in linux. I am using RH ES 3 |
04:11.51 | sricard | convey:: did you configure the phone itself via it's web interface |
04:12.03 | JunK-Y | convey: past all ur output at pastebin.ca |
04:12.30 | convey | sricard: the web interface did not reveal any configurable settings. ie sip server/ proxy, user name etc. |
04:12.55 | sricard | convey: i configured mine that way in 2 minutes |
04:13.30 | convey | sricard: all I see are netwokr settings, DNS, gateway, etc. |
04:13.43 | sricard | go to ext 1 tax |
04:13.46 | sricard | tab |
04:13.53 | convey | sricard: ok |
04:13.57 | sricard | in proxy, put the ip of asterisk |
04:14.28 | sricard | in user id, put the section/user id define in sip channel |
04:14.36 | *** part/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net) |
04:14.52 | convey | sricard: found it |
04:14.56 | sricard | in password put the password defined in the sip channel |
04:15.14 | sricard | all you need to get it to work and register |
04:15.19 | sricard | to asterisk |
04:15.41 | convey | now here is where we have a disconnect. When you say sip channel, which config file is that information located? |
04:15.51 | sricard | sip.conf |
04:16.08 | sricard | i can paste my sip.conf section for my 841 here |
04:16.12 | sricard | if you want |
04:16.22 | convey | sricard: that would be great |
04:16.50 | sricard | [220] |
04:16.50 | sricard | type=friend |
04:16.50 | sricard | host=dynamic |
04:16.50 | sricard | context=toll-access |
04:16.50 | sricard | secret=qazwsx20 |
04:16.51 | sricard | mailbox=220 |
04:16.53 | sricard | dtmfmode=rfc2833 |
04:16.55 | sricard | disallow=all |
04:16.57 | sricard | allow=ulaw |
04:16.59 | sricard | that's it |
04:17.19 | sricard | it is extension 220 in my house |
04:17.27 | |Vulture| | pastebin.ca pls |
04:17.46 | sricard | what is pastebin.ca? |
04:18.08 | convey | the [220] is equivilant to my [phone1] correct? |
04:18.45 | sricard | if you put a name there, you'll need alias in your extensions.conf |
04:19.18 | sricard | but it should still work |
04:19.20 | Inv_arp | sricard: pastebin.ca is a site to paste stuff |
04:19.42 | sricard | Inv_arp: never used that one |
04:20.30 | sricard | Inv_arp: i get it now, thx will do next time |
04:21.14 | convey | sricard: [phone1] is not aliased |
04:21.52 | convey | sricard: I am just using phone1 as a device |
04:22.22 | sricard | convey: you could still use it but you will have to define an extension with a number and use DIAL(SIP/Phone1) to reach it |
04:22.51 | sricard | convey: its fine, it should work |
04:23.03 | *** join/#asterisk jayden (~ircatjerr@adsl-69-209-134-225.dsl.sfldmi.ameritech.net) |
04:23.15 | sricard | convey: i always use numbers (right or wrong) |
04:23.41 | convey | sricard: yes the example you have stated DIAL(SIP/Phone1) is exactly waht I am using |
04:24.38 | sricard | convey: so if you put phone1 as the user id and what ever password you define in sip.conf, your 841 should register with asterisk |
04:25.14 | *** join/#asterisk letherglov (~letherglo@8036aa5e.resnet.ucsd.edu) |
04:25.32 | sricard | convey: all in the Ext 1: tab on the Web interface (Advance mode) of the SPA-841 |
04:25.50 | *** join/#asterisk lildivil (user@ool-18bc24d7.dyn.optonline.net) |
04:26.36 | convey | sricard: we have a registration |
04:26.45 | convey | sricard: THanks for the guidance |
04:27.19 | sricard | convey: np my friend, happy to help someone else for once :-) |
04:27.32 | sricard | bye for now |
04:27.43 | convey | sricard: thanks again, bye |
04:28.20 | lildivil | good evening all... |
04:31.34 | *** join/#asterisk jayden (~ircatjerr@adsl-69-209-134-225.dsl.sfldmi.ameritech.net) |
04:31.35 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
04:32.22 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
04:46.02 | *** join/#asterisk marc32344 (~marc32344@69-28-224-214.dsl.teksavvy.com) |
04:48.10 | jayden | Manx |
04:50.01 | ManxPower | Nobody has any problems? Cool! My work here is done! |
04:50.24 | jayden | nice |
04:50.39 | mrgoby | Manx: my asterisk dont work none, what's wrong with it ? |
04:50.41 | jayden | I am fighting with screwed up header files |
04:50.53 | topping | OT: http://www.bash.org/?301963 |
04:51.03 | ManxPower | jayden, You are using RedHat FC3, huh? |
04:51.05 | jayden | and a guy in #asterisk-dev that can't figure out how to msg nickserve |
04:51.10 | jayden | ummmmm |
04:51.16 | jayden | not on this box |
04:51.16 | ManxPower | jayden, That's pretty sad. |
04:51.22 | ManxPower | jayden, I don't really do developer questions. |
04:51.35 | ManxPower | mrgoby, First you need to find a goat. |
04:51.47 | jayden | hehe |
04:52.04 | jayden | ~Gatamala |
04:52.23 | jayden | sigh |
04:52.27 | jayden | ~mbtt |
04:52.28 | jbot | well, mbtt is Mavis Beacons Teaches Typing |
04:52.31 | ManxPower | mrgoby, The next step is kind of complicated. It's best to contact your local Satanist group for the proper prodecure. |
04:52.33 | jayden | ~Guatamala |
04:55.22 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
04:55.23 | *** topic/#asterisk is Asterisk: The Open Source PBX || Dev Conf 1PM CST FEB 24th -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf June 8-10th more coming soon.... |
04:55.32 | ManxPower | ~clec |
04:55.33 | jbot | extra, extra, read all about it, clec is Created by the Telecommunications Act of 1996, a CLEC is a service provider that is in direct competition with an incumbent service provider. CLEC is often used as a general term for any competitor, but the term actually has legal implications. To become a CLEC, a service provider must be granted "CLEC status" by a state's ... |
04:56.12 | ManxPower | jbot needs a bigger buffer. |
04:56.46 | mrgoby | jayden: was that 'sure' directed at me ? |
04:56.50 | jayden | y |
04:57.10 | iceyp | ManxPower how do you add it for your own extensions? |
04:57.13 | mrgoby | hints ?? |
04:57.22 | jayden | ummmm |
04:57.25 | ManxPower | iCEBrkr, I don't use CDRs. |
04:57.30 | ManxPower | I don't bill users for calls. |
04:57.31 | iceyp | I've added username, callerid, fromuser and none of them do it |
04:57.38 | ManxPower | The LD company does that. |
04:57.41 | jayden | I can't think of an app that will do that.... |
04:57.42 | iceyp | I'm trying to build a web interface |
04:57.44 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
04:58.07 | bjohnson | how can I get * to not use chan_oss (and not use my /dev/dsp) |
04:58.20 | PyroSteve | hey guys |
04:58.20 | PyroSteve | i just got my iaxy |
04:58.20 | PyroSteve | its just simply got a fxs module |
04:58.20 | PyroSteve | can I insert an fxo moudule instead |
04:58.22 | ManxPower | bjohnson, noload => chan_oss.so in /etc/asterisk/modules.conf. |
05:16.38 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
05:16.38 | *** topic/#asterisk is Asterisk: The Open Source PBX || Dev Conf 1PM CST FEB 24th -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf June 8-10th more coming soon.... |
05:16.44 | shepherd | i've heard soooo many stories |
05:16.47 | shepherd | :D |
05:16.51 | bkw_ | shepherd, about me? |
05:16.58 | shepherd | mmmhmm |
05:17.05 | bkw_ | their is a video out there of me drunk as hell at astricon |
05:17.08 | shepherd | <-- matt |
05:17.11 | bkw_ | i tend to repeat myself when i'm drunk |
05:17.14 | bkw_ | shepherd, you ninny |
05:17.16 | brc_ | *there* |
05:17.18 | mikegrb | bkw_: no, they grow right by my porch, just for you |
05:17.18 | bkw_ | haha |
05:17.19 | mrgoby | http://www.voip-info.org/wiki-Asterisk+cmd+Rpt |
05:17.21 | mikegrb | and maybe Beirdo |
05:17.29 | bkw_ | haha |
05:17.45 | mikegrb | where is next astricon going to be and what month |
05:17.47 | mrgoby | ~cluebat jayden |
05:17.49 | jbot | ACTION pulls out a ClueBat (tm) and thwaps jayden. |
05:17.50 | bkw_ | oct 2005 |
05:17.55 | mikegrb | may be able to get new employer to foot the bill |
05:17.56 | shepherd | chan_ham ? |
05:17.56 | bkw_ | cluecon is june 2005 |
05:17.57 | shepherd | haha |
05:18.00 | mikegrb | that would be most excelent |
05:18.00 | mrgoby | :-) that's more like it |
05:18.09 | bkw_ | shepherd, you have to write about cluecon when we get it all lined up fully |
05:18.24 | bkw_ | thats our task this week to get it more filled in and lined up |
05:18.38 | jayden | hey bkw, who from asterlink is in detroit... I saw the address was here |
05:18.42 | mikegrb | where are each going to be? |
05:18.48 | mrgoby | i'm in ann arbor |
05:18.51 | bkw_ | jayden, the payroll lady |
05:18.56 | bkw_ | the one that PAYS ME |
05:18.58 | bkw_ | ;) |
05:19.10 | mrgoby | BIG UPS TO DA U.P. ! |
05:19.18 | mikegrb | hmm california |
05:19.20 | bkw_ | UPS? |
05:19.22 | bkw_ | I hate UPS |
05:19.28 | bkw_ | they play football with my loot all the time |
05:19.33 | shepherd | what is cluecon? |
05:19.41 | bkw_ | shepherd, a more hardcore dev conf |
05:19.56 | mrgoby | they played flippy floppy with my servers |
05:20.02 | bkw_ | ya |
05:20.04 | bkw_ | I hate that |
05:20.06 | bkw_ | pisses me off |
05:20.14 | mishehu | I can't stand UPS. even if I order for 10am delivery, they'll still show up at 18:00 and tell me that I am in a residential area, thus the 10:00 delivery doesn't apply |
05:20.14 | mrgoby | love to see them rack ears all bent |
05:20.17 | bkw_ | they left 600 bucks worth of shit on my porch one friday |
05:20.19 | bkw_ | I was out of town |
05:20.24 | bkw_ | came home on sunday it was gone |
05:20.24 | mishehu | fedex on the other hand tends to arrive on time. |
05:20.25 | bkw_ | the best part is |
05:20.29 | bkw_ | they said I signed for it |
05:20.32 | Beirdo | <PROTECTED> |
05:20.32 | bkw_ | funny part is |
05:20.40 | bkw_ | they infact did have "Brian West" on the sig |
05:20.47 | mikegrb | Beirdo: dunno, what? |
05:20.48 | bkw_ | but I never sign UPS with my name |
05:20.50 | shepherd | fedex loses a lot of packages though |
05:20.54 | Beirdo | heh |
05:20.55 | bkw_ | I always sign "bugs bunny" |
05:20.57 | shepherd | TRUST ME, I KNOW |
05:21.03 | mrgoby | that is funny as hell |
05:21.13 | bkw_ | so I told them I didn't sign for that |
05:21.16 | bkw_ | they were like we have your sig |
05:21.18 | mrgoby | so your only defense is that it was wrong because it was right |
05:21.19 | mrgoby | haha |
05:21.20 | bkw_ | i'm like HELL YOU DO |
05:21.29 | mishehu | bkw_: oooh you wascilly wabbit |
05:21.39 | bkw_ | well I sign all credit card stuff with looney toon charaters |
05:21.47 | bkw_ | so if someone has my real sig |
05:21.49 | bkw_ | I know its fake |
05:21.51 | bkw_ | ;) |
05:21.52 | mrgoby | my servers are named after muppets |
05:21.59 | mrgoby | :-) |
05:22.05 | bkw_ | infact the bugs bunny thing with UPS |
05:22.07 | bkw_ | saved my ass |
05:22.14 | mrgoby | yeah |
05:22.17 | bkw_ | I pulled up the last 10 tracking numbers on shit I ordered |
05:22.19 | shepherd | that's better than choochies |
05:22.20 | bkw_ | all "Bugs Bunny" |
05:22.22 | *** join/#asterisk t3t (~t3t@bar.pangalacticgargleblaster.com) |
05:22.42 | mishehu | mrgoby: which one did you name as "swedishchef" or "borkborkbork" ? |
05:22.45 | bkw_ | t3t so when are we gonna go out for sone Pan Galactic Gargle Blasters? |
05:22.47 | bkw_ | you buying or me? |
05:22.54 | bkw_ | s/sone/somme/ |
05:22.57 | bkw_ | fuck I can't type |
05:23.01 | bkw_ | I think I had too many already |
05:23.21 | t3t | have another http://pangalacticgargleblaster.com/ |
05:23.27 | mrgoby | neither of them exist yet :0) but soon to come |
05:23.29 | bkw_ | LOVE IT |
05:23.35 | bkw_ | 42 42's |
05:23.35 | bkw_ | haha |
05:23.39 | bkw_ | LOVE IT |
05:23.42 | bkw_ | can't wait for the movie |
05:24.01 | t3t | I was bored a couple of years ago... |
05:24.14 | bkw_ | 3 letters baby |
05:24.16 | bkw_ | THREE |
05:24.22 | t3t | that's hard to come by |
05:24.23 | bkw_ | check the expire date baby |
05:24.28 | bkw_ | had it since 98 |
05:24.30 | t3t | i had to settle for a 3-letter .us name |
05:24.36 | Firestrm | mikegrb, http://www.vrl.ca/pic19.jpg that was taken over my shoulder as i was running from the fire |
05:24.39 | bkw_ | didn't think they alloed 3 letters in .us |
05:24.44 | bkw_ | thought it was a 4 letter min |
05:24.49 | bkw_ | I want fuck.us |
05:24.53 | t3t | I'm an idiot for not getting more domains in '94 when they were free |
05:24.58 | shepherd | bkw_: i have hijacked.us |
05:25.05 | bkw_ | eww debian |
05:25.06 | bkw_ | get out |
05:25.08 | trym | debian owns |
05:25.15 | bkw_ | no asterlinux ownz |
05:25.15 | trym | so.. what is the thing.. redhat? haha |
05:25.18 | t3t | The rule may be 4, but i have a 3... |
05:25.21 | bkw_ | roothat |
05:25.21 | mikegrb | Firestrm: wow |
05:25.29 | bkw_ | or root that |
05:25.29 | mrgoby | you a gentoo man, bkw ? |
05:25.29 | bkw_ | haha |
05:25.32 | trym | i also own whiteho.us |
05:25.34 | t3t | shepherd: nice catch |
05:25.37 | mrgoby | or woman ? |
05:25.40 | bkw_ | mrgoby, not exclusivly |
05:25.43 | bkw_ | but I do like it |
05:25.47 | bkw_ | I like solaris and freebsd more |
05:25.53 | t3t | trym: you missed the 'e' :) |
05:25.57 | mrgoby | yikes ... solaris ? |
05:26.01 | bkw_ | ya |
05:26.10 | mrgoby | i guess i cant speak on it... i only ever used it in school |
05:26.18 | t3t | slowlaris, bkw? |
05:26.18 | bkw_ | as a server its great |
05:26.19 | Firestrm | mikegrb, and this http://www.vrl.ca/pic13.jpg is what it looked like at night.. not a trick photo.. this is real.. very eery |
05:26.20 | bkw_ | as a desktop it sucks |
05:26.23 | loud | HINT: do not type reset on the CISCO 7960 cli .. thinking its "reboot" .. |
05:26.24 | mrgoby | yes |
05:26.24 | bkw_ | it might be slow but tell ya |
05:26.35 | bkw_ | it rocks |
05:26.38 | bkw_ | and is solid |
05:26.52 | mrgoby | and expensive |
05:27.00 | bkw_ | shit I have seen solaris boxes sail along on shit that would make a linux box blow up |
05:27.04 | bkw_ | linux will not scale |
05:27.11 | bkw_ | neither will asterisk |
05:27.14 | Firestrm | mikegrb, its very disconcerting to have a mushroom cloud over your home town |
05:27.15 | bkw_ | unless you throw more boxes at it |
05:27.25 | t3t | bkw: I agree... most of my boxes are fbsd |
05:27.32 | mrgoby | hehe |
05:27.52 | Firestrm | i tried fbsd once..... once.. |
05:27.53 | bkw_ | haha |
05:27.56 | bkw_ | my address is on my whois |
05:27.59 | bkw_ | ship away baby |
05:28.02 | bkw_ | Firestrm, you gave up too soon |
05:28.03 | mrgoby | UPS |
05:28.07 | bkw_ | asterlinux is very BSDish |
05:28.07 | mrgoby | send em UPS |
05:28.08 | t3t | slowlaris is too expensive for what you get... besides it took them almost 15 years just to get most of the init scripts to work |
05:28.25 | Firestrm | bkw_, i felt like a fish out of water.. |
05:28.40 | bkw_ | Firestrm, you will for about a week |
05:28.53 | bkw_ | I switched from windows to OSX and guess what i didn't miss a beat |
05:29.04 | shepherd | assturdlinux |
05:29.08 | shepherd | :D |
05:29.09 | t3t | looking at making the leap soon... |
05:29.10 | bkw_ | shepherd, be nice |
05:29.16 | bkw_ | t3t, you won't regret it |
05:29.16 | Firestrm | bkw_, it was like trying to drive a right hand steer car for the first time |
05:29.28 | t3t | that's what people keep telling me |
05:29.42 | t3t | I used to admin about 15 macs at a newspaper |
05:29.45 | bkw_ | t3t it took me less than a day to get used ot it |
05:29.47 | freddy | Firestrm: all the cars here are right hand steer! |
05:29.52 | mrgoby | i'm out.... power to the people ! |
05:29.54 | t3t | os[7-9] suck |
05:29.56 | *** part/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net) |
05:29.57 | bkw_ | yes |
05:30.01 | bkw_ | OS X is the only reason I own a mac |
05:30.03 | freddy | Firestrm: its not that hard really |
05:30.04 | bkw_ | if it were not for that |
05:30.07 | bkw_ | i wouldn't even think of it |
05:30.08 | shepherd | yeah |
05:30.11 | bkw_ | but Jobs did it right |
05:30.14 | shepherd | i refuse to run x11 on my mac |
05:30.18 | bkw_ | haha |
05:30.19 | shepherd | though |
05:30.22 | bkw_ | its actually cool |
05:30.30 | shepherd | NO! |
05:30.31 | bkw_ | rootless X |
05:30.33 | t3t | I just wish that apple didn't mess with the fbsd core as much |
05:30.43 | bkw_ | t3t, well its all good |
05:30.47 | bkw_ | atleast they have jordan hubbard |
05:30.47 | Firestrm | freddy, when i visited uk, i rented a motorcycle, at least then all i had to do is get out of the way of ppl driving towards me in my lane :) |
05:31.12 | bkw_ | when he left the freebsd project it kinda isn't as good as it used to be |
05:31.12 | t3t | I guess it's just about getting used to it |
05:31.42 | bkw_ | but i'm sooooooo geeky I painted a white apple on my luggage |
05:31.49 | bkw_ | ya know |
05:31.58 | bkw_ | dont wanna loose it when he goes to VON in two weeks |
05:32.15 | shepherd | heh |
05:32.16 | t3t | better than a beastie... never know what dhs would do with that... |
05:32.25 | *** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg) |
05:32.37 | shepherd | bkw_: do you have a powerbook? |
05:32.46 | bkw_ | ibook and imac |
05:32.58 | shepherd | k |
05:33.01 | letherglov | hahha |
05:33.02 | bkw_ | in all honesty the system profiler says this is a powerbook |
05:33.11 | letherglov | von's in san jose yeah? |
05:33.15 | bkw_ | yep |
05:33.18 | letherglov | goodtimes |
05:33.22 | t3t | I wish that I could justify the $$ for VON... it should be a good show this year |
05:33.31 | letherglov | I volunteered at the tech museum (across the street) for a couple years |
05:33.36 | bkw_ | you'll love it |
05:33.39 | shepherd | m$ is going security crazy |
05:33.44 | bkw_ | haha |
05:33.45 | t3t | Firestrm: they lost their mind 5 years ago when they were contemplating how to do updates |
05:33.47 | shepherd | but they are doing it all about the wrong way |
05:33.52 | bkw_ | the fact that OS X gets the fuck out of my way |
05:33.56 | bkw_ | and lets me do what I wanna get done |
05:34.04 | letherglov | bkw_, enough mac talk |
05:34.08 | bkw_ | no no |
05:34.08 | shepherd | i like their creative top down approach |
05:34.11 | letherglov | :-P |
05:34.12 | bkw_ | MORE MAC talk |
05:34.25 | bkw_ | I have apples |
05:34.27 | bkw_ | you ninny |
05:34.30 | bkw_ | a pair of them |
05:34.34 | bkw_ | and you know you wanna lick em |
05:34.41 | *** join/#asterisk _chad (~Chad@c-24-6-142-55.client.comcast.net) |
05:34.48 | Firestrm | t3t, sadly i have to agree, but the latest, were not going to patch unauthorized boxes stupidity had put a nail in the M$ coffin.. |
05:35.03 | t3t | Firestrm: they'll change their tune... again |
05:35.16 | t3t | This idea was kicked around two years ago |
05:35.42 | t3t | I was one of a few people who convinced those in charge not to do it |
05:35.44 | Firestrm | t3t, only after there is a sea of rootkited unpatched machines |
05:35.48 | shepherd | well.. i heard somewhere that m$ signed away their rights to use any type of unix based system |
05:35.51 | t3t | sounds like they changed the guard again... |
05:36.44 | t3t | Firestrm: then is now... the sea is churning and bubbling as we speak |
05:37.13 | Firestrm | t3t, yes, but if you can even imagine it... its going to get worse.. |
05:37.22 | t3t | shepherd: haven't heard that... they did stop development of the POSIX subsystem of NT a few years back though |
05:38.12 | t3t | Firestrm: I don't think that there is currently an avenue for them to get better... if MS gives up, the consumer network providers are are only hope at containing the problem |
05:39.19 | |Vulture| | anyone seen any info on the new ipVolution TDMs? |
05:39.34 | *** join/#asterisk Tough_Nuts (~Tough_Nut@m19105e42.tmodns.net) |
05:39.36 | t3t | Nope, what do they do? |
05:39.46 | DJ-Pyro | |Vulture|: someone brought it up earlier tonight |
05:40.29 | |Vulture| | they seem pretty good... in concept |
05:40.29 | |Vulture| | http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-38356249088.htm |
05:40.50 | shepherd | we'll see :) |
05:40.58 | |Vulture| | onboard echo cancel and codec DSP |
05:40.59 | Firestrm | t3t, the only way to really fix this problem, is to remove the right of software companies to disclaim liability for damages due to poor coding.. Software is the only place that one can currently get away with this. If an auto manufacturer built a car with a flaw that caused it to randomly swerve for example, they would have to recall it and fix it. |
05:41.06 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-86-218.sentechsa.net) |
05:41.37 | t3t | Firestrm: I thought that the insurance companies would have had that changed a long time ago... |
05:41.45 | DJ-Pyro | with an add-on daughter card that adds another 4 T1 ports |
05:41.52 | t3t | ... clearly my crystal ball is broken. |
05:42.28 | |Vulture| | DJ-Pyro: doesn't that seem too good to be true? |
05:43.01 | Firestrm | t3t, personal injury lawyers prevented that.. problem is that software bugs do cause injury in the form of time and money, but because it isnt a broken leg or somthing physical, judges dont see that.. |
05:43.07 | ManxPower | I don't know if those cards have shipped yet or not. |
05:43.10 | |Vulture| | wonder if it will run off the same zaptel drivers |
05:43.21 | t3t | |Vulture|: It looks like a rendering of a pcb with some rj11 connectors on it... I wonder what the finished product will actually do |
05:43.23 | shepherd | Firestrm: it's not the car manufacture's fault if your car gets stolen |
05:43.31 | |Vulture| | ManxPower: it says the duals were suppose to ship in Jan.. but they are still in Pre-Order |
05:43.50 | greg_work | Firestrm: i've always thought that would be done well with with engineering.. have a policy where you require software to be approved by a P.Eng. If it fails, the P.Eng. is held liable, and depending on the circumstances may face harsh punishments (not unlike what happens to hte engineer that approves a bridge that collapses under its own weight) |
05:43.54 | |Vulture| | t3t: yea the picture is a joke, I was talking about the actual stats |
05:44.04 | ManxPower | Hardware design, as Digiumn found out, is much harder than it looks. |
05:44.05 | `Sauron | Murf. |
05:44.10 | t3t | Firestrm: maybe that's a good thing... have you ever seen a federal judge try to figure out what a trojan horse controlling a pc can do |
05:44.24 | shepherd | they would have to rewrite most of asterisk to get it to work too |
05:44.29 | greg_work | of course, the chances of finding a P.Eng to sign off on windows are about as good as microsoft licencing it as GPL |
05:44.38 | Firestrm | shepherd, yes and no. If they design a lock that is so flimsey that you can break it with a plastic knife (see toyota) i think they should at least be partially responsible.. its all about due diligence |
05:44.51 | bkw_ | windows is missing something |
05:44.53 | `Sauron | Anyone know if there's integer versions of most voice codecs out there? |
05:45.00 | Firestrm | t3t , now that would be amusing :) |
05:45.00 | shepherd | i can't use windows anymore |
05:45.05 | shepherd | i tried the other day |
05:45.06 | `Sauron | My sparse google attempts earlier today didn't find much information |
05:45.22 | shepherd | i get pissed waiting on it |
05:45.22 | shepherd | haha |
05:45.33 | t3t | |Vulture|: what actual stats? all i see is some bullet points beneath a picture and some marketing-speak. Until someone unrelated to the company gets to pound on it, it's vaporware for me... |
05:45.50 | |Vulture| | t3t: good point |
05:46.11 | `Sauron | greg |
05:46.13 | `Sauron | what up |
05:47.32 | greg_work | |Vulture|: you know what's funny about that? asterisk was originally created (once CPUs were fast enough) so you didn't require onboard DSPs (because they're expensive) :) |
05:47.45 | greg_work | `Sauron: not much. about to go home |
05:48.07 | `Sauron | You just work there, or you own the place? |
05:48.09 | shepherd | but now.. we're gonna have to move towards them again |
05:48.18 | `Sauron | late work hours is generally a sign of owning a shop. :) |
05:48.30 | greg_work | my dad owns it |
05:48.48 | Firestrm | greg_work, even worse... slave labour.. |
05:48.49 | `Sauron | Er, probably nevermind.. I think I remember.. yeah. Your lastname ~ company name |
05:48.53 | `Sauron | grin |
05:49.20 | greg_work | tonight i was mostly working on AMP stuff though .. so it guess you could consider i'm doing it partially on company time, partly personal |
05:49.55 | greg_work | i didnt even get doing ANY programming till 4 though :p i hate days like that |
05:51.09 | greg_work | (actually if any of you use AMP, go read my last couple posts on amportal-users and give me suggestions ;) ) |
05:51.43 | t3t | Does anyone know how 800# calls from payphones are billed. Is there still a ~$.25 surcharge in the US? |
05:52.54 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
05:52.54 | greg_work | (if you don't use amp, basically i added some stuff to make pattern-based routing that can use various trunks with a configurable priority .. ie, for local calls, use ZAP lines then try voip, and for long distance, use voip then zap) |
05:53.29 | shepherd | sweet |
05:54.22 | greg_work | t3t: by law, toll-free, operator, and 911 calls from payphones are free. (i dunno if the payphone owner pays or not though) |
05:55.03 | `Sauron | I'd imagine receiving 800 owner pays |
05:55.05 | t3t | thanks greg, I meant to ask if I (as the 800# owner) get charged extra for the calls |
05:55.32 | *** join/#asterisk MarkK (~got@65-100-56-164.ptld.qwest.net) |
05:55.35 | t3t | `Sauron: that makes sense, but how does a provider like iaxtel or nufone pass it on? |
05:55.57 | `Sauron | Hum, dunno :) |
05:56.04 | t3t | I'll just have to wait until jj is on and ask him |
06:03.19 | clive- | does anyone have any pointers on iax2 registrations with iaxclient? I get "Inappropriate authentication received |
06:03.20 | clive- | " |
06:04.37 | t3t | clive-: is your password blank? |
06:05.51 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
06:08.41 | t3t | clive-: what version of * are you running? |
06:09.52 | clive- | t3t I have a password, I am thinking I should have auth=rsa rather than auth=plaintext |
06:10.03 | t3t | give it a try |
06:10.24 | t3t | from the source it looks like the md5 password or the plaintext password is 0 length |
06:11.33 | clive- | my client is iaxclient |
06:11.55 | clive- | rsa gives the same eroror |
06:12.07 | t3t | clive-: I haven't used it in a while.. I use firefly currently |
06:12.17 | t3t | what version of asterisk do you have? |
06:12.26 | clive- | I woudl expect firefly works the saem |
06:12.32 | clive- | verison..hmm, old,,,lets check |
06:13.02 | t3t | probably over a year, right? |
06:14.01 | clive- | eek |
06:14.02 | clive- | CVS-06/11/04-03 |
06:14.19 | t3t | you may want to think about updating :) |
06:14.31 | clive- | let me do it quick |
06:14.32 | clive- | :) |
06:14.44 | clive- | will that fix my login trouble? |
06:14.54 | t3t | this is gonna be good... a quick update of * from 2003 to present :) |
06:15.14 | t3t | I assume... or at least you'll get a more descriptive error message |
06:15.21 | clive- | lol |
06:15.53 | clive- | let me quickly figure out how to co version 1.0.2 or something |
06:16.25 | `Sauron | just co cvs-head |
06:16.27 | `Sauron | ;) |
06:16.38 | clive- | is head stable (ish) |
06:16.40 | t3t | http://www.voip-info.org/tiki-index.php?page=Asterisk+Download |
06:17.00 | `Sauron | stable enough for me at home |
06:17.24 | t3t | clive-: It's been stable for me at home and work (2/11/05 and 12/25/04) |
06:17.51 | t3t | clive-: The bad part of head is you don't know exactly what you'll get until you try it |
06:18.39 | djin | What's the difference between 'friend' and 'peer' in sip.conf?? |
06:19.07 | t3t | a friend is someone you trust, a peer is just someone the same age as you... oops. wrong context again... |
06:19.16 | t3t | friend: both user and peer |
06:19.36 | t3t | peer: outgoing connection to another server |
06:19.37 | shepherd | oh peer me! |
06:20.05 | t3t | user: incoming connection |
06:20.34 | Beirdo | ~httpdtype www22.verizon.com |
06:20.40 | t3t | djin: be careful with the names in []. Some services expect them to be an EXACT string (capitolization is important) |
06:20.46 | Beirdo | ahhh, that explains a lot |
06:21.11 | djin | what setting is used to link two * servers to each other? |
06:21.26 | t3t | you could use friend on both |
06:21.51 | djin | ok, thanks. |
06:21.55 | `Sauron | capitAlization is important ;) |
06:22.24 | clive- | oh boy I am rusty at this...since 2003:) |
06:22.56 | t3t | `Sauron: Nothing like interactive live spell checking |
06:23.00 | t3t | thanks |
06:23.10 | `Sauron | you're welcome :) |
06:23.25 | t3t | how do you know I wasn't talking about a governmental unit? |
06:23.34 | bkw_ | t3t talkin about Mac's again? |
06:23.40 | t3t | sure |
06:23.52 | bkw_ | live spell checking happens in just about every application |
06:23.54 | t3t | macworld is only a few months away |
06:23.54 | bkw_ | its in the API |
06:24.00 | t3t | no way |
06:24.03 | bkw_ | yes way |
06:24.07 | bkw_ | my xchat spell checks on me |
06:24.09 | bkw_ | its nice |
06:24.20 | bkw_ | everything thats cocoa baseed |
06:24.23 | t3t | what does it do if you mess up a word? |
06:24.28 | bkw_ | red underline |
06:24.39 | bkw_ | you use emacs? |
06:24.41 | t3t | that's scary-useful |
06:24.52 | t3t | never got into it, unfortunately |
06:25.12 | t3t | I haven't had anyone able to articulate its advantages clearly enough |
06:25.26 | bkw_ | I would have to show you |
06:25.30 | bkw_ | I wouldn't knwo if someone didn't show me |
06:25.37 | bkw_ | I thought emacs was evil |
06:25.41 | bkw_ | till i was showed the light |
06:25.44 | shepherd | yahoo needs to make a decient port of messenger for the mac :( |
06:25.47 | t3t | I just don't know what it can do |
06:26.05 | bkw_ | hehe |
06:26.05 | clive- | t3t haa, got it, auth needs to be md5 meanwhile I have a new asterisk versuion |
06:26.08 | t3t | someone must have created a 'this is why you need emacs' page for the laxy |
06:26.22 | t3t | da###it.. ^lazy |
06:26.28 | bkw_ | haha no it turns in to an editor war |
06:26.39 | bkw_ | even with live spell checking I still fuckup because i'm lazy |
06:26.39 | bkw_ | haha |
06:26.44 | bkw_ | fuck sdfow |
06:26.46 | bkw_ | haha |
06:26.53 | shepherd | vi4life! |
06:28.11 | t3t | Other than context-highlighting, regex find/replace, and line numbering what does it do that's so special? |
06:28.26 | bkw_ | its just nice |
06:28.33 | t3t | simple |
06:28.35 | bkw_ | I use maybe a tiny bit of it |
06:28.39 | bkw_ | but its easy |
06:28.46 | t3t | not very informative, but simple |
06:28.59 | bkw_ | well simple to me is hard as hell for others it seems |
06:29.00 | bkw_ | haha |
06:29.12 | bkw_ | when I first meet anthm before I worked here for him |
06:29.15 | bkw_ | I was using pico |
06:29.17 | bkw_ | eww ewww eww |
06:29.22 | t3t | didn't you learn c to mess with *? |
06:29.26 | bkw_ | he showed me emacs.. and I LOVE IT |
06:29.29 | bkw_ | t3t yes |
06:29.42 | bkw_ | jumped right in |
06:29.44 | drumkilla | well damn, for code, almost anything would be better than pico |
06:29.47 | t3t | not many out there like you, bkw_... |
06:29.57 | bkw_ | I still have stuff I dont know in C |
06:29.59 | bkw_ | but I learn |
06:30.05 | t3t | I generally use joe or vi for farting around |
06:30.21 | drumkilla | i'm a fan of vim ... |
06:30.21 | bkw_ | I like emacs becuase I can do emacs /usr/src/asterisk/ |
06:30.36 | bkw_ | it shows me a directory.. or emacso /usr/src/asterisk/*.c |
06:30.40 | bkw_ | and it will open every C file |
06:30.46 | bkw_ | er emacso.. haha |
06:30.48 | bkw_ | funny me |
06:31.02 | t3t | you had me.. I thought it was a special emacs command |
06:31.11 | drumkilla | sounds like a super hero |
06:31.16 | bkw_ | type'o |
06:31.25 | bkw_ | emacso the super editor |
06:31.37 | shepherd | bkw_: you should irc from emacs |
06:31.44 | t3t | I use homesite for most moderate lifting |
06:31.44 | bkw_ | you can play tetris in emacs |
06:31.48 | bkw_ | meta-x |
06:31.50 | bkw_ | tetris |
06:31.52 | bkw_ | it starts |
06:32.00 | DJ-Pyro | someone was running an irc bot in emacs on another channel |
06:32.02 | shepherd | emacs should replace hurd in a few years |
06:32.31 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
06:32.46 | rvhi | polycom uses standard sip call park |
06:33.04 | t3t | at any rate, I shoudl check it out... care to put up a screenshot of how you have it laid out, bkw? |
06:33.04 | rvhi | anyone is working on making this work? |
06:33.06 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
06:33.12 | kpfleming | bkw_: i have been using since before there was GNU... before Linux was even a dream, before there were PCs... it's a wonderful editor in innumerable ways |
06:33.16 | t3t | ^ should check it out that is... |
06:33.18 | DJ-Pyro | rvhi: you refering to the park soft key not actually parking? |
06:33.29 | rvhi | yes |
06:33.51 | DJ-Pyro | ok, that means that I can stop messing with the configs trying to find out why it doesn't work :) |
06:35.28 | bkw_ | kpfleming, its nice isn't it |
06:35.37 | kpfleming | yep |
06:35.44 | bkw_ | i'm glad tony made me learn to use it |
06:35.49 | bkw_ | he said it was a requirement for me to use |
06:35.53 | bkw_ | hehe |
06:36.12 | bkw_ | kpfleming, we need to get you, tony, mark, me and twisted in the conf and smack these patches around on the bug trakcer |
06:36.18 | bkw_ | before mark closes the darn thing down |
06:36.20 | kpfleming | used to work entirely in emacs on a green screen terminal all day, never left it for any reason... even had a forum (bbs) interface in an emacs buffer, along with email and other cool stuff |
06:36.39 | kpfleming | the conf calls do not seem to accomplish much at all |
06:37.25 | bkw_ | well we have in the past |
06:37.32 | bkw_ | we have had 8 hour conf calls before |
06:37.36 | kpfleming | i wish we could figure out how to get the conference calls to work better, the delays in conversation are just so long |
06:37.37 | bkw_ | and cleared out 30-40 bucks |
06:37.40 | kpfleming | yeah, i know, i've been there |
06:37.44 | bkw_ | ya we are working on that too |
06:37.49 | bkw_ | with stevekstevek |
06:37.53 | bkw_ | we might use his conf thingy next week |
06:37.54 | t3t | bucks? |
06:37.56 | kpfleming | but these are not simple patches we are talking about here, they require a lot more thought and discussion |
06:38.01 | bkw_ | er bugs |
06:38.13 | t3t | need a word context checker |
06:38.15 | bkw_ | kpfleming, exactly maybe we can line out a few of the major ones next week |
06:38.40 | bkw_ | kpfleming, we could hog tie mark at VON |
06:38.52 | DJ-Pyro | new conference app? |
06:39.06 | kpfleming | i hope so, we need a very significant shift in the way things get done if we are going to accomplish what we are all trying to do |
06:39.17 | t3t | assuming that you could touch him.. I have a feeling that mark will be a little busy with the minions at VON |
06:39.26 | bkw_ | t3t, RIIIGHT |
06:39.27 | bkw_ | haha |
06:39.28 | kpfleming | i could spend the next three months cleaning and optimizing code, but i don't, because it would take _forever_ to get the patches merged |
06:39.35 | bkw_ | kpfleming, yep |
06:39.41 | bkw_ | we have got to speed this along |
06:39.48 | bkw_ | that is what caused asterisk to fork in the first place |
06:40.17 | kpfleming | yeah, we've talked about all this before, including two more people have commit privs for stuff that they didn't write, and nothing has changed yet :-( |
06:43.25 | `Sauron | j/k |
06:44.16 | t3t | ok, it's late and I'm getting giddy. Y'all have a good night now. |
06:44.21 | `Sauron | I understand their licensing thingy before you can submit code, but my understanding is, that even after going through all the paperwork, code doesn't get added even after it was submitted. |
06:44.47 | `Sauron | And I really need to go to bed, even though I just got some coding inspiration. |
06:44.49 | `Sauron | Ugh. |
06:51.20 | *** join/#asterisk akrall (~akrall@201.128.92.118) |
06:51.35 | akrall | Guys.. anybody using festival on asterisk? and using the text2wave method? |
06:53.26 | Beirdo | Jeez, US local calling is so fucked up |
06:55.13 | goatmilk | language, language... |
06:57.05 | *** join/#asterisk odie_flocon (~chatzilla@S01060011953994ee.cg.shawcable.net) |
06:58.02 | odie_flocon | anybody use IAXtalk.com products? |
06:58.09 | odie_flocon | ie the AT-320ED |
06:58.15 | Beirdo | goatmilk: what you disagree? |
06:59.19 | akrall | Guys.. anybody using festival on asterisk? and using the text2wave method? |
06:59.19 | *** join/#asterisk tandrews (~tandrews@mail.grok.co.za) |
06:59.27 | goatmilk | agree or disagree i'm not going to talk like that in here... not very respectful at all. |
06:59.34 | odie_flocon | I have never used it. |
06:59.50 | goatmilk | akrall: we saw you, and it's late that's why no one is responding. I have heard of people using festival |
07:00.50 | tandrews | morning |
07:00.53 | Beirdo | uhhh, and who is it I'm supposed to be respecting in the US telco market? the local calling areas are borked beyond belief |
07:01.21 | goatmilk | Beirdo: i'm talking about cursing, not your opinions on telephone service. |
07:01.27 | akrall | goatmilk: sorry ofr repeating... too late and lack of cofee |
07:01.29 | akrall | :) |
07:01.36 | goatmilk | akrall: np buddy |
07:02.06 | Beirdo | meh, whatever |
07:02.23 | Beirdo | cursing is not disrespectful. Crude and stupid perhaps. |
07:02.27 | goatmilk | akrall: there is a script out there that does weather info over the phone.. you dial your zip code and it'll talk it out to you. maybe this will help you? |
07:02.43 | Beirdo | but anyways, I apologize if it offended |
07:03.11 | odie_flocon | interesting |
07:03.23 | goatmilk | Beirdo: I do myself, but not in this channel. |
07:04.16 | odie_flocon | is there any good Voip Providers in the States? |
07:04.16 | tandrews | quick h323 question - What's the difference between the h323 implementation on inaccessnetworks and the one included in asterisk in channels/h323 ? (And which is the most sensible to use?) |
07:04.24 | odie_flocon | I need a Utah number. |
07:05.04 | odie_flocon | who is inaccessnetowrks? |
07:05.33 | tandrews | http://www.inaccessnetworks.com/projects/asterisk-oh323 |
07:05.49 | tandrews | "H.323 support for ASTERISK PBX using the OpenH323 library" |
07:06.37 | tandrews | They seem to provide a module which is in effect a wrapper for the C++ stuff |
07:07.25 | odie_flocon | the h323 implementation should be close to the same. |
07:07.43 | odie_flocon | since H.323 is a standard. |
07:08.24 | odie_flocon | imho you should use *'s H.323 |
07:09.16 | odie_flocon | tandrews are you in NZ? |
07:09.32 | tandrews | the documentation on inaccess networks on how to install looked much better though ;) |
07:09.43 | tandrews | no odie_flocon, I'm in ZA |
07:09.52 | odie_flocon | ohh sorry. |
07:10.11 | tandrews | similar accent though, easy mistake ;) |
07:10.32 | odie_flocon | my friend used to work for Siemens in ZA |
07:10.58 | tandrews | k |
07:11.44 | odie_flocon | the H.323 implementation is already built into * |
07:11.54 | odie_flocon | so you should need much to install. just configuration. |
07:11.58 | *** part/#asterisk akrall (~akrall@201.128.92.118) |
07:13.05 | tandrews | well, the README says "You must run Open H.323 v1.15.1 and PWLib v1.8.1. All other versions are not supported." |
07:13.11 | tandrews | yuk |
07:13.22 | odie_flocon | ahh |
07:13.39 | odie_flocon | hmmm. |
07:13.48 | odie_flocon | I honestly can't tell you |
07:14.03 | odie_flocon | I have only used SIP, and IAX. |
07:14.50 | tandrews | They're simple in comparison by the looks of things |
07:14.56 | *** join/#asterisk TheEmperor (~mattn@203.121.47.100) |
07:16.43 | clive- | tandrews howzit |
07:16.58 | tandrews | hi clive- |
07:18.14 | tandrews | JerJer are you here ? |
07:18.14 | clive- | nice to find anothetr african in here:) |
07:18.21 | tandrews | :) |
07:19.29 | *** join/#asterisk schurig (~schurig@p54B2818C.dip0.t-ipconnect.de) |
07:21.52 | odie_flocon | so tandrews? do you have to pay by the minute to be online? |
07:22.15 | tandrews | no odie_flocon I have a leased line at home |
07:22.27 | odie_flocon | you have DSL? |
07:22.41 | tandrews | nah, 28.8 modems :) |
07:22.49 | tandrews | dsl next month |
07:22.57 | odie_flocon | yeah. |
07:23.05 | odie_flocon | how many modems u got? |
07:23.08 | tandrews | but it's expensive here compared to say the UK |
07:23.26 | odie_flocon | I'm in Canada. |
07:23.53 | tandrews | ah, k |
07:24.18 | odie_flocon | it kinda sux |
07:24.46 | tandrews | just the one modem odie_flocon |
07:24.47 | odie_flocon | cuz here we have T1's, and in Europe etc. they use E1's. |
07:25.09 | odie_flocon | I wish they used E1's here in Canada. |
07:28.47 | `Sauron | Whee. |
07:28.56 | `Sauron | I ordered a full set of embedded linux toys. |
07:28.59 | `Sauron | www.gumstix.com |
07:37.08 | odie_flocon | so Sauron, you gonna install * on these boxes? |
07:37.15 | `Sauron | Hehe |
07:37.45 | `Sauron | Doubtful. But I do plan to play around with some audio stuff. We were talking here earlier about making an iax2 phone device or something. |
07:37.50 | schurig | `Sauron: please note that those devices have an ARM based chip without FPU, so some voice codecs won't work |
07:38.01 | `Sauron | I know :) |
07:38.49 | `Sauron | as far as I saw, there's integer versions of 711, 723 and gsm |
07:38.52 | `Sauron | possibly others |
07:39.01 | schurig | `Sauron: do they have an sound interface? The XScale has AC97 and SPI, but without some proper chip (e.g. Wolfson Micro, UCB1x00) you won't have any audio sound at all ... |
07:39.20 | `Sauron | the audiostix add-on board has an UCB1400 on it |
07:40.26 | `Sauron | http://www.gumstix.org/tikiwiki/tiki-index.php?page=Schematics |
07:40.35 | `Sauron | http://www.gumstix.org/tikiwiki/tiki-index.php?page=Audiostix+Schematics |
07:41.22 | `Sauron | I'm mainly picking up the stuff for a different project. But I went ahead and added the audiostix add-on board so I could play around with some audio stuff. |
07:41.25 | odie_flocon | would be interesting. |
07:42.01 | odie_flocon | I'm looking for a small footprint * box. |
07:42.30 | odie_flocon | was thinking about using an Xbox. |
07:42.40 | odie_flocon | and calling it the "pbXbox |
07:42.43 | odie_flocon | and calling it the "pbXbox" |
07:42.46 | `Sauron | hehe |
07:42.54 | `Sauron | how about the *box? |
07:42.55 | `Sauron | ;) |
07:43.18 | kpfleming | be extremely careful what you call anything... Asterisk is a registered trademark |
07:43.21 | *** join/#asterisk dg1_work (~schulte@gate.sympat.de) |
07:43.33 | odie_flocon | so it the Xbox. :D |
07:43.57 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
07:44.03 | `Sauron | Mmm. |
07:44.09 | `Sauron | Look, there's an IAXclient library. |
07:44.22 | odie_flocon | ??? |
07:44.28 | odie_flocon | what do you mean? |
07:44.46 | `Sauron | Earlier, there was talk about there not being any iax2 phones. |
07:44.52 | `Sauron | Just the IAXy |
07:45.36 | `Sauron | Since there's an iaxclient library, it should be (relatively) trivial to get an app compiled on the gumstix that connects to * using iax, and turns it into audio, etc. |
07:46.27 | odie_flocon | hmm |
07:46.32 | `Sauron | http://www.voip-info.org/wiki-IAXClient |
07:46.46 | `Sauron | And, it looks like someone's done all the hard work to verify iax runs on arm processors: |
07:46.47 | odie_flocon | why don't you just get a hold of an existing phone. and use it. |
07:46.50 | `Sauron | http://www.kauss.org/Stephan/ziaxphone/ |
07:47.00 | `Sauron | ARM softphone |
07:47.15 | `Sauron | the only existing phones, are all SIP |
07:48.08 | odie_flocon | not true. |
07:48.15 | `Sauron | okay |
07:48.28 | `Sauron | so there's sccp, mgcp and h.323 phones |
07:48.43 | `Sauron | blah blah blah |
07:49.15 | `Sauron | It's a challenge. Something interesting to do. |
07:50.53 | odie_flocon | http://www.iaxtalk.com/ |
07:50.54 | kpfleming | all the PA1688 phones support IAX2 now as well |
07:50.56 | odie_flocon | not true |
07:51.16 | odie_flocon | I'm sure you could write code for the snom phones as well. |
07:51.20 | *** join/#asterisk eivindtr (~Eivind@193.91.146.34) |
07:51.26 | odie_flocon | since they are linux based. |
07:52.16 | odie_flocon | and the A320AD is still less then 100.00 |
07:52.51 | `Sauron | google can't find anything about a320ad |
07:53.07 | odie_flocon | look at www.iaxtalk.com |
07:53.08 | *** part/#asterisk Firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
07:54.57 | `Sauron | Hehn. I stand corrected. |
07:55.02 | `Sauron | Oh well |
07:55.22 | `Sauron | I'll find something to do with my audio board... |
07:55.24 | `Sauron | :) |
07:57.31 | odie_flocon | although I can't verify this..... becuase I've never used it. |
07:59.05 | odie_flocon | if I use an X-box, I can use Sip phones internally, and use IAX to go outside of the network. |
07:59.53 | odie_flocon | or if I get something small enough with Dual nic's. |
08:00.09 | odie_flocon | I can eliminate the router. |
08:00.50 | odie_flocon | the thing that I liked about the gumstix. is that they can be battery powered. |
08:01.58 | odie_flocon | which would be nice for power outage. |
08:02.16 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
08:16.07 | *** part/#asterisk djin (~djin@gridfox.xs4all.nl) |
08:17.31 | *** join/#asterisk Mike_TK (~Mike_TK@212.165.78.5) |
08:21.13 | *** join/#asterisk shaZwaz (~chatzilla@203.81.196.167) |
08:21.23 | shaZwaz | hi room |
08:23.59 | *** join/#asterisk qiu (~andrei@home-073519.b.astral.ro) |
08:24.00 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
08:26.13 | *** join/#asterisk inticonnet (~nick@118.68.233.220.exetel.com.au) |
08:26.20 | inticonnet | Hey Peoples |
08:28.40 | shaZwaz | hi inticonnet |
08:28.43 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
08:30.47 | inticonnet | Hey Manx if ur around :) Im not in the mood for asking questions so you should feel safe for a while. I do have 2morow off work tho sooo :P |
08:38.52 | *** join/#asterisk Firebird_ (~xxx@130.40.39-62.rev.gaoland.net) |
08:41.08 | Firebird_ | Hi, can someone tell me why I have poor audi quality on a wav file coming from the monitoring option ? |
08:42.17 | JerJer | cuz its a german car? |
08:43.22 | zoa | haha |
08:43.26 | zoa | german cars are the best |
08:43.31 | zoa | JJ go to bed |
08:43.33 | *** join/#asterisk djin (~marius@62.58.40.196) |
08:43.41 | zoa | its not good for your attitude to stay up this long :p |
08:44.08 | Firebird_ | sorry, I wanted to say audio quality... |
08:45.16 | *** join/#asterisk shaZwaz (~chatzilla@203.81.196.167) |
08:45.38 | *** join/#asterisk Shrink (~tgb@cpc1-cwma1-6-0-cust233.swan.cable.ntl.com) |
08:46.49 | Firebird_ | euh...did I say something wrong or nobody has knowledge of the monitoring option ? |
08:52.49 | zoa | nobody is awake |
08:52.55 | zoa | i have perfect quality from the wav file |
08:53.01 | zoa | so it should work normally |
08:55.22 | JerJer | bed? |
08:55.33 | JerJer | does that have something to do with that sleep thing I keep hearing about? |
08:56.33 | shaZwaz | where can I find a good comparison of Yate and * ? |
08:58.10 | inticonnet | Sleep is over rated. Then again it is only 8pm here :) |
08:58.41 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
08:58.51 | shaZwaz | ~sleep |
08:58.52 | jbot | i heard sleep is overrated, and a poor substitute for caffeine |
08:59.14 | inticonnet | Jbot is so smart :P |
08:59.22 | inticonnet | ~jbot |
08:59.23 | jbot | well, jbot is the shipboard computer, but you may call me eddie if it helps you relax |
09:02.33 | *** join/#asterisk meppl (~mephisto@p3E9E220E.dip.t-dialin.net) |
09:04.01 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
09:04.37 | meppl | guten morgen |
09:05.50 | djin | ohne sorgen |
09:09.32 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
09:11.59 | Firebird_ | zoa : when using monitoring, I have two wav files, both have the same problem...it's like the background sound is on foreground and the conversation is background.... I can't hear what it is being said... Any idea of what I should check ? |
09:13.37 | zoa | no sorry |
09:13.40 | zoa | try recording to gsm |
09:13.43 | zoa | i never had that problem |
09:17.57 | zoa | oh i did around 1 million recordings so far :) |
09:18.16 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
09:18.23 | Firebird_ | ok, I will have a try but it's very strange.... |
09:22.49 | shaZwaz | hi Zeeek |
09:27.02 | *** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
09:29.40 | scythelx | is regexten the same as mailbox in sip.conf |
09:30.29 | Zeeek | hi shaZwaz |
09:31.45 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
09:31.56 | neopher | gmorning everyone |
09:33.43 | neopher | i'm looking for a peice of hardware that will allow you to connect old tdma phones to a PBX |
09:33.52 | neopher | any idea |
09:35.25 | *** join/#asterisk pjm_uk (~pjm_uk@cpc1-pool3-3-0-cust116.sot3.cable.ntl.com) |
09:38.26 | Zeeek | . |
09:55.20 | *** join/#asterisk visik7 (~ciao@host178-39.pool80182.interbusiness.it) |
09:55.34 | *** join/#asterisk visik7 (~ciao@host178-39.pool80182.interbusiness.it) |
10:02.35 | *** join/#asterisk Newbie___ (some@211.24.146.10) |
10:02.59 | Newbie___ | hi, can cisco 186 work with * ? |
10:03.00 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
10:06.29 | *** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl) |
10:10.22 | Firebird_ | cisco 186 works great ! |
10:10.40 | Newbie___ | tks, Firebird_ |
10:11.54 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
10:15.23 | Zeeek | . |
10:17.45 | JerJer | , |
10:17.56 | Newbie___ | - |
10:18.01 | JerJer | ` |
10:18.18 | Newbie___ | :D |
10:19.54 | Newbie___ | can * reroute calls to my codes? ie. all UK fixed lines to broadvoice and UK cell to another provider? |
10:20.13 | shaZwaz | will there be any jitter buffer settings in upcoming chan_sip module ? |
10:23.42 | *** join/#asterisk pr0m (~pr0metheu@ip-wv-68-187-250-031.charterwv.net) |
10:24.09 | *** join/#asterisk datareactor (datareacto@203.81.192.33) |
10:24.48 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
10:24.48 | datareactor | how can i implement fax over ip though asterisk ? |
10:25.48 | Newbie___ | i dont think FoIP is any good, datareactor |
10:26.04 | Zeeek | Fsck over IP ? |
10:26.17 | JerJer | fax over ip is very possible |
10:26.30 | JerJer | we use app_tx and rx fax like they are going out of style |
10:26.54 | JerJer | it is all in how you implement the solution |
10:27.39 | Newbie___ | foip is heavily dependent of isp bandwidth, am i right ? |
10:27.47 | JerJer | not necessarily |
10:27.57 | Newbie___ | hrm |
10:28.16 | JerJer | don't attempt to actually make a fax VoIP call over the 'net and you will be fine |
10:28.23 | JerJer | find a better way |
10:28.23 | datareactor | what is best solution for it . we have plenty of bandwith |
10:28.48 | Newbie___ | my last experience with T.38 failed, we are using fax - email instead |
10:29.15 | JerJer | notice I didn't say T.38 |
10:29.28 | Newbie___ | oh, ok |
10:29.30 | Newbie___ | sorry |
10:30.09 | Newbie___ | i was told that T.38 is out now, need to buy some gadget |
10:30.16 | JerJer | bleh |
10:30.19 | datareactor | JerJer what do you recommend for running FAXOverip |
10:30.19 | JerJer | not necessary |
10:30.28 | JerJer | app_txfax and app_rxfax |
10:30.38 | JerJer | and don't do it over ip |
10:30.57 | Zeeek | JerJer don't you ever have problems with certain fax machines? |
10:31.14 | JerJer | not so far |
10:31.30 | Zeeek | I have and a lot of other people do |
10:31.38 | Newbie___ | yeah, i also tried * fax some 8 mths ago, it only work on certain fax machine |
10:31.49 | JerJer | don't do it over IP |
10:31.51 | Zeeek | it has been working better recently |
10:32.05 | JerJer | going out our PRIs it has been 100% solid |
10:32.06 | Zeeek | I'm talking FXO spandsp |
10:32.11 | JerJer | hell no |
10:32.13 | JerJer | not gonna happen |
10:32.22 | Newbie___ | care to elaborate more on 'dont do it over ip' |
10:32.34 | JerJer | don't make a call over IP and expect fax to work |
10:32.58 | Zeeek | I'm talking PSTN |
10:33.04 | datareactor | JerJEr how you billing for foip services |
10:33.13 | JerJer | datareactor: per call |
10:33.21 | JerJer | nothing different whatsoever |
10:33.29 | Newbie___ | Zeeek: my experince was PSTN -> * but failed on most fax machines |
10:33.34 | *** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au) |
10:33.36 | JerJer | Zeeek: FXO is not going to be reliable |
10:33.37 | datareactor | JerJer OK |
10:33.44 | JerJer | PRI |
10:33.46 | JerJer | digital |
10:33.49 | JerJer | all the way |
10:33.57 | Zeeek | JerJer why not? |
10:34.29 | Zeeek | oh, ok, digital |
10:35.01 | JerJer | no DAC going on |
10:35.02 | Zeeek | anyway, fax isn't much to us, better to have a $50 fax machine than dick around for days with spandsp |
10:35.04 | datareactor | JerJer what services you acquire for international faxes |
10:35.13 | JerJer | acquire ? |
10:35.26 | JerJer | we place a call out of our PRIs |
10:35.30 | JerJer | if a fax answers, it works |
10:35.40 | shaZwaz | :) |
10:35.42 | datareactor | use i mean |
10:36.05 | JerJer | a PRI, TE410P, a ds-3 mux and a telica switch |
10:40.00 | Zeeek | JerJer you receive faxes with spandsp? |
10:40.13 | zoa | goddamn jj |
10:40.14 | zoa | go to bed |
10:40.37 | sambal | :D |
10:40.56 | datareactor | JerJer how you do international fax ? |
10:41.24 | JerJer | make a phone call |
10:41.30 | JerJer | using app_txfax |
10:41.35 | JerJer | next question |
10:41.50 | datareactor | JerJer but i think it will be costly |
10:41.54 | JerJer | Zeeek: app_rxfax |
10:42.00 | zoa | jj, but how can your users do a fax ? |
10:42.01 | zoa | :) |
10:42.02 | JerJer | datareactor: how do you figure? |
10:42.03 | Zeeek | and it works? |
10:42.10 | JerJer | Zeeek: sure |
10:42.23 | zoa | anyone feels like sponsoring the sip jitter buffer ? :( |
10:42.43 | *** join/#asterisk teemu-x (~tnurmine@tuomi.oulu.fi) |
10:42.45 | datareactor | JerJEr it will all going though Pstn and Telco will charge you international call |
10:42.51 | shaZwaz | JJ Is there a way that is cheaper than that :) |
10:42.52 | JerJer | and this is a problem how? |
10:43.11 | JerJer | we are all about the PSTN here |
10:43.18 | zoa | yeah |
10:43.23 | zoa | dont trust a carrier using voip |
10:43.25 | JerJer | its cheaper for us to call Russia than my parents up the road |
10:43.28 | zoa | voila |
10:43.30 | zoa | same here |
10:43.30 | Zeeek | I have had several cases where rx_fax refuses to work where windows shareware does on the same line and same fax sender |
10:43.44 | JerJer | ? |
10:43.49 | shaZwaz | same here Zeeek |
10:43.49 | JerJer | don't use analog |
10:44.17 | JerJer | coppice's DSP code is only so good |
10:44.20 | Zeeek | you aren't addressing the problem which is why does it work on the same line, same sender with different software? |
10:44.20 | JerJer | thus far |
10:44.32 | JerJer | um better DSP code |
10:44.53 | Zeeek | I'd give him $25 for spandsp if it worked perfectly :) |
10:45.05 | Zeeek | that's what the shareware cost |
10:45.15 | JerJer | add a few decimal places and i'm sure he would be interested |
10:45.27 | Zeeek | nah, we don't get that many faxes |
10:45.27 | JerJer | then run the goddamn shareware |
10:45.32 | Zeeek | we do |
10:45.35 | JerJer | next |
10:45.43 | zoa | :) |
10:45.44 | zoa | hehe |
10:45.58 | Zeeek | next waht, I'm just sharing my experience |
10:47.08 | shaZwaz | jerjer how about a jitter buffer in Chan_sip ? |
10:47.22 | datareactor | what is other choices if i dont want to use spandsp |
10:47.34 | JerJer | see bug 2532 and implement accordingly |
10:47.44 | JerJer | datareactor: buy a fax machine |
10:47.56 | shaZwaz | :) |
10:48.00 | JerJer | and plug it in to a telephone line |
10:48.17 | zoa | shaZwaz: we have a jitter buffer in chan_sip |
10:48.21 | zoa | but its not finished yet |
10:48.32 | zoa | and we are spending way too much money onit |
10:48.36 | shaZwaz | JerJer sh'd I send you my faxes and you forward them :) |
10:48.49 | JerJer | i guess that means chan_h323 will need to be updated soon as well |
10:49.08 | zoa | yes probably |
10:49.19 | zoa | we could also do that |
10:49.28 | zoa | but dunno if we will want to |
10:49.32 | zoa | as we no longer use it |
10:49.40 | zoa | we already spent 2 months on jitter buffer crap |
10:49.48 | JerJer | mkay? why |
10:50.06 | zoa | reading books, trying different approaches etc :( |
10:50.07 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02v-22-254.d4.club-internet.fr) |
10:50.11 | JerJer | um |
10:50.23 | JerJer | stevekstevek has a jitterbuffer and plc code that works |
10:50.38 | zoa | well the jitter buffer in chan_sip is using his now |
10:50.49 | zoa | we are also doublechecking his work |
10:50.53 | JerJer | not the newest i don't thikn |
10:51.03 | JerJer | i'm only patching chan_iax2 on the devel box |
10:51.17 | zoa | yeah its too soon to do it anywhere else |
10:51.22 | zoa | we found some bugs in there so far |
10:51.30 | zoa | i think he already patched them now |
10:51.56 | zoa | we first were going for a different approach, but then we dediced to team up |
10:52.00 | zoa | and started over again |
10:52.23 | zoa | anyway the quality is a huge improvement |
10:52.53 | zoa | i really want it in v1.2 |
10:53.01 | zoa | but dunno if we will make it |
10:53.14 | JerJer | lite a fire under some asses |
10:53.26 | Zeeek | that wouldn't smell so good |
10:53.34 | zoa | hehe |
10:53.37 | JerJer | Want me to bring in Sargent Hartman? |
10:55.23 | shaZwaz | that works with SIP ? |
10:55.27 | shaZwaz | 2532 |
10:55.41 | JerJer | when u implement it into chan_sip, it will |
10:55.42 | JerJer | sure |
10:57.01 | *** join/#asterisk dg1nsw (~schulte@gate.sympat.de) |
10:57.09 | teemu-x | Hi. If user B has redirected his incoming calls from number B1 to B2, and user A calls B1, then A is charged for A -> B1 and B for B1 -> B2, right? |
10:57.34 | shaZwaz | zoa: does't work with 729 ? |
10:58.02 | JerJer | not until mark/digium updates their shared object |
10:58.38 | shaZwaz | and when will that happen ? |
10:58.57 | shaZwaz | this thing must be on top priority |
10:59.10 | JerJer | and the code is ready |
10:59.13 | JerJer | and? |
10:59.17 | JerJer | when the code is ready |
11:00.00 | shaZwaz | so one can patch it safely |
11:00.59 | JerJer | there is nothing to patch |
11:01.09 | JerJer | codec_g729a.c is not available |
11:01.09 | zoa | i think mark should have it ready very soon |
11:04.04 | JerJer | Real Soon Now(tm) |
11:04.13 | zoa | hehe yeah |
11:04.21 | zoa | :) |
11:04.33 | zoa | it will be ready as soon as someone nagged enough |
11:04.46 | zoa | its like the overflow of nagging to his ears + 2 minutes |
11:06.27 | PoWeRKiLL | Hello :) |
11:07.02 | PoWeRKiLL | I got cvs on sunday on a production server it's very very unstable |
11:07.15 | JerJer | make clean install |
11:07.19 | zoa | wait till you add a jitter buffer to it :p |
11:07.31 | JerJer | i am running cvs -head as of a few hours ago and it is fine |
11:07.53 | zoa | ive also seen no issues with non patched -head |
11:07.58 | PoWeRKiLL | I already make a clean install I see that some function completely change I got to update asterisk-addons also |
11:08.12 | PoWeRKiLL | JerJer on a production server ? |
11:08.21 | JerJer | absolutely |
11:09.06 | PoWeRKiLL | How many peer do you have on it |
11:09.12 | JerJer | all of them |
11:09.21 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
11:09.21 | *** mode/#asterisk [+o bkw_] by ChanServ |
11:09.28 | zoa | look |
11:09.30 | PoWeRKiLL | I think since I upgrade I loose about 30% of peers |
11:09.30 | zoa | its the other guy |
11:09.39 | JerJer | then you have bigger problems |
11:10.33 | *** join/#asterisk eipi (eipi@136-218-114-200.fibertel.com.ar) |
11:10.45 | zoa | newsflash of the day |
11:12.02 | zoa | some bastard broke the heating and hot water supply for all houses in my region |
11:12.22 | zoa | they have like hot water systems in the city, people dont need to have a boiler |
11:12.35 | zoa | luckily i have a backup system for hot water |
11:12.40 | zoa | which is also not working :(((((((( |
11:12.48 | zoa | brian |
11:12.50 | zoa | you bastard |
11:12.52 | zoa | reply to me |
11:15.08 | JerJer | ugg |
11:15.24 | JerJer | you let a municipality provide you HOT water? |
11:15.28 | JerJer | how evil |
11:16.05 | PoWeRKiLL | is it possible that my problem come from asterisk-head and asterisk-stable discussing over IAX ? |
11:16.52 | shaZwaz | is that due to the jitter buffer ? |
11:17.09 | zoa | thats how they do it here |
11:17.14 | zoa | also never saw it beforfe |
11:17.14 | JerJer | zoa: that's tough |
11:17.16 | zoa | before |
11:17.18 | zoa | yeah |
11:17.21 | zoa | and its freezing here |
11:17.42 | zoa | and every year they start cleaning these pipes |
11:17.44 | zoa | during the summer |
11:17.55 | zoa | no hot water for 2 months or so |
11:19.02 | Newbie___ | why would u need hot water in summer? |
11:19.10 | Zeeek | JerJer out of curiosity, what countries have you visited? |
11:19.33 | JerJer | USA |
11:19.35 | JerJer | Canada |
11:19.52 | Zeeek | Mexico? |
11:21.04 | PoWeRKiLL | why I got lot of this error Got SIP response 415 "" back from ? |
11:21.20 | zoa | because thats what the phone sent |
11:21.27 | zoa | or whatever device you are calling to |
11:22.06 | PoWeRKiLL | I think it's a GS device I don't understand I got this error since I got last *-head |
11:22.23 | zoa | what does 415 mean ? |
11:23.09 | zoa | unsupported media type |
11:23.12 | zoa | = 415 |
11:23.33 | PoWeRKiLL | I don't know what that mean |
11:23.51 | zoa | it means get a packet dump :p |
11:23.57 | PoWeRKiLL | I'm not trying to insert a floppy disk in the GS :) |
11:24.00 | zoa | probably the codec is incompatible |
11:24.25 | zoa | and asterisk fallsback to some codec |
11:24.28 | zoa | and gs doesnt accept it |
11:24.36 | zoa | thats my guess anyway |
11:30.42 | *** join/#asterisk pjm_uk (~pjm_uk@cpc1-pool3-3-0-cust116.sot3.cable.ntl.com) |
11:33.53 | shaZwaz | PoWeRKiLL: are u using the jitter buffer in SIP ? |
11:36.05 | modulus_ | got some thc |
11:36.08 | modulus_ | now time to code.. |
11:37.48 | modulus_ | i wish i had some beef jerkey |
11:38.48 | JerJer | ahhh Vitamin T, H and C |
11:38.51 | Delvar | i wish i could jerk off a cow |
11:39.28 | modulus_ | daddy would you like some sausage? |
11:42.04 | bkw_ | haha |
11:42.06 | bkw_ | naughty boi |
11:42.17 | modulus_ | hi bkw |
11:42.24 | bkw_ | hi modulus_ |
11:42.29 | bkw_ | so you gonna come to cluecon? |
11:42.49 | modulus_ | only if i can get stoned and drink dangerous amounts of alcohol |
11:42.55 | bkw_ | beginners corse... look app_skel.c.. look make it say hello world.. |
11:42.59 | bkw_ | haha |
11:43.08 | bkw_ | modulus_, haha you can do what you wish |
11:43.08 | teemu-x | n |
11:43.14 | modulus_ | which means 'increased chance of getting laid' |
11:43.22 | bkw_ | for who? |
11:43.24 | bkw_ | me or you? |
11:43.37 | modulus_ | me duh |
11:43.43 | bkw_ | you sure about that |
11:43.49 | modulus_ | everyone around me gets ass except me |
11:43.57 | bkw_ | can't really get laid when you're face down in a pillow |
11:44.19 | Zeeek | why not? |
11:44.33 | bkw_ | you have to ask? |
11:44.39 | bkw_ | Have I not warped you enought? |
11:44.45 | modulus_ | bkw how old are you? |
11:44.49 | bkw_ | 28 |
11:44.53 | modulus_ | cool |
11:44.57 | modulus_ | let's party together |
11:45.03 | bkw_ | ok now i'm scared. |
11:45.06 | modulus_ | and make fools of ourselves in front of women |
11:45.08 | bkw_ | :P |
11:45.11 | modulus_ | hot women |
11:45.16 | JerJer | LOL |
11:45.17 | bkw_ | Oh hell I don't need to be drunk to do that |
11:45.23 | modulus_ | i do |
11:45.26 | modulus_ | it's much more entertaining |
11:45.35 | bkw_ | I'm like a broken record when drunk |
11:45.37 | bkw_ | I repeat |
11:45.38 | modulus_ | after the fact too |
11:45.46 | Zeeek | only when drunk? |
11:45.47 | modulus_ | i grow gimongous testicles |
11:45.50 | modulus_ | when drunk |
11:45.56 | bkw_ | haha |
11:46.00 | bkw_ | someone has issues then |
11:46.03 | modulus_ | yeah |
11:46.18 | *** join/#asterisk jluk (~jluk@pl6.lawrence.org.uk) |
11:46.20 | bkw_ | so getting drunk automagically straps on a huge pair of balls? |
11:46.32 | modulus_ | for me it does |
11:46.40 | bkw_ | haha |
11:46.51 | bkw_ | I don't have to be drunk for that |
11:46.58 | modulus_ | i'm a regular don juan under the influence |
11:46.59 | bkw_ | I'll do all kinds of stupid shit without it |
11:47.07 | bkw_ | say anything |
11:47.09 | bkw_ | do anything |
11:47.10 | modulus_ | usually the girls look better too |
11:47.13 | bkw_ | make a fool of myself |
11:47.17 | bkw_ | modulus_, or guys |
11:47.19 | bkw_ | har har har |
11:47.25 | modulus_ | i'm hetero |
11:47.44 | Zeeek | not a crime |
11:47.57 | Zeeek | yet |
11:48.25 | modulus_ | jbot cluecon? |
11:49.44 | JerJer | ClueCon Talk: How not to create a VoIP signalling method: H.323 and SIP :) |
11:50.17 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-86-218.sentechsa.net) |
11:50.26 | modulus_ | hi jerjer |
11:50.37 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
11:51.11 | clive- | does anyone know what VNACK's are ? |
11:51.34 | clive- | something to do with chan iax |
11:51.37 | *** join/#asterisk smurfix (~smurf@smurfix.developer.debian) |
11:52.55 | modulus_ | jbot vnack? |
11:53.19 | modulus_ | jbot hug clive- |
11:53.21 | jbot | ACTION hugs clive- |
11:53.25 | modulus_ | awwwww |
11:55.13 | clive- | thanks guys:), now I have a hug and thousands of VNAcks too :) |
11:55.21 | clive- | lol |
11:55.27 | modulus_ | clive-, where'd you see that? |
11:55.47 | clive- | chan_iax2.c:5837 socket_read: Sending VNAK |
11:56.04 | modulus_ | vnack is kinda different than vnak |
11:56.19 | clive- | my typing needs work:) |
11:56.34 | codebreaker | hello i have language=de and exten => _cause_0,3,Playback(demo-congrats,skip) but asterisk still plays t he sound in english. from cli Executing Playback("SIP/7304910-69a6", "demo-congrats|skip") in new stack Playing 'demo-congrats' (language 'en') what have i done wrong? |
11:56.52 | JerJer | clive-: smells like your asterisk box(s) need to be upgraded |
11:57.08 | clive- | Jerjer , hi, I just upgraded |
11:58.09 | JerJer | make clean install |
11:58.16 | JerJer | run cvs code |
11:58.21 | JerJer | -head |
11:58.22 | bkw_ | also |
11:58.28 | bkw_ | check your modules |
11:58.31 | bkw_ | for zaptel |
11:58.47 | bkw_ | their is a issue if you have say a 410 card loaded but ZERO spans configured |
11:58.51 | bkw_ | asterisk can't play sound files |
11:58.55 | bkw_ | have you seen that one jerjer? |
11:59.02 | clive- | Jerjer thanks, I'll try that |
12:00.54 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
12:01.24 | RoyK | what is "Distinctive Ringing" |
12:02.10 | Delvar | a ring that is distinctive? |
12:02.45 | Delvar | isnt it a cisco thing?, the ringtone sounds sligtly different depending on what value you set? |
12:03.05 | RoyK | nfi |
12:03.57 | JerJer | bkw_: hmm |
12:04.29 | JerJer | i'll have to fire up another devel box and try that (all of the ones i have online are being utilized) |
12:05.12 | RoyK | anyone that knows how I can simulate packet loss? |
12:05.39 | modulus_ | man ping |
12:05.50 | RoyK | modulus_: ? |
12:05.55 | JerJer | RoyK: i think u can do it with iptables |
12:06.03 | JerJer | and the tc stuff |
12:06.09 | modulus_ | royk, have someone ping flood you |
12:06.20 | RoyK | JerJer: there's a "-m random" and "-m nth" but those aren't supported in 2.6 |
12:06.28 | RoyK | modulus_: I said simulate...... |
12:06.31 | JerJer | ahh sucky |
12:06.39 | tzanger | JerJer: what're you building? |
12:06.42 | RoyK | also, it's on a gigabit link to the norwegian internet hub..... |
12:06.57 | modulus_ | damn |
12:07.07 | modulus_ | straight into the machine's nic? |
12:07.13 | *** join/#asterisk Tili (~Tili@202-133-65-33-dialup.sat.net.pk) |
12:07.21 | RoyK | hm |
12:07.24 | RoyK | nth is ported :) |
12:08.22 | JerJer | there ya go |
12:08.44 | *** join/#asterisk __pbx__ (~strace@ADSL-F49-S197-critical-coi.nortenet.pt) |
12:09.44 | __pbx__ | http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold&comments_threshold=0&comments_offset=0&comments_sort_mode=commentDate_desc&comments_maxComments=10&comments_parentId=202#threadId352 |
12:09.48 | __pbx__ | any comments on this? :( |
12:10.12 | __pbx__ | manual? :> |
12:10.28 | tzanger | JerJer: using my rc.tc script are ya :-) |
12:10.35 | *** join/#asterisk micos (~micosat@host217-44-194-118.range217-44.btcentralplus.com) |
12:11.06 | micos | Hi All |
12:11.36 | micos | Just compiled Asterix on Solaris and did make install |
12:11.50 | micos | And make samples |
12:12.00 | micos | is there a guide somewhere? |
12:12.44 | JerJer | oh god make samples is so evil |
12:13.00 | tzanger | JerJer: yes it is |
12:13.08 | JerJer | tzanger: nope...Royk asked about simulating packet loss |
12:13.35 | tzanger | JerJer: with the new jitter buffer there's the set losspct command |
12:13.45 | JerJer | Lets play the how we spell A S T E R I S K game |
12:13.57 | clive- | tzanger not for sip, only iax |
12:14.08 | JerJer | tzanger: killer |
12:14.27 | tzanger | clive-: true enough but you can adapt that very easily |
12:14.51 | *** join/#asterisk Mother_ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
12:14.52 | Mother_ | hi all |
12:15.21 | Mother_ | is there a way to return the call to the original pickup SIP phone after a blind transfer ends on a busy phone? |
12:15.58 | Mother_ | i.e. instead of the call going to voicemail |
12:15.58 | JerJer | don't beat me Mother_ |
12:16.04 | Mother_ | lol |
12:16.23 | RoyK | how should the new jitterbuffer be configured? |
12:16.27 | RoyK | automagically? |
12:16.47 | Mother_ | right now these people are complaining that they transfer calls but would like to talk to the caller if the extension to which it was transfered is busy |
12:16.54 | Mother_ | rather than sticking them into VM |
12:18.05 | *** join/#asterisk marcel_ (~marcel@cpc1-shep4-3-0-cust235.leic.cable.ntl.com) |
12:18.06 | JerJer | Asterisk made slashdot again |
12:18.52 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
12:21.14 | *** join/#asterisk pbxjunkie (~stormtroo@videocomputer.gr) |
12:22.11 | pbxjunkie | guys, does anyone know how to stop asterisk from flashing my grandstream phones whenever there is voicemail pending? |
12:22.25 | pbxjunkie | I'm pretty happy with just e-mailing the messages as attachments |
12:22.37 | Mavvie | pbxjunkie: how to get asterisk down it would also be interesting :-) |
12:22.54 | *** join/#asterisk phreak (~phreak@ua-83-227-137-86.cust.bredbandsbolaget.se) |
12:22.58 | Mavvie | or how to see it. |
12:23.07 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-60.client.comcast.net) |
12:24.29 | modulus_ | asterisk can down voicemail? |
12:24.42 | JerJer | down? |
12:24.43 | RoyK | down? |
12:24.58 | modulus_ | i must be stoned |
12:25.10 | modulus_ | i'm eating rice |
12:25.11 | JerJer | pbxjunkie: don't specify a mailbox in the sip.conf stanza for that fone |
12:25.15 | modulus_ | with this korean meat dish |
12:25.28 | JerJer | modulus_: you mean unhatched maggots? |
12:25.40 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
12:25.42 | modulus_ | jerjer, you tried those before? |
12:26.11 | JerJer | um no |
12:26.11 | pbxjunkie | JerJer: thank I got it. |
12:26.21 | JerJer | Thank you, drive-thru |
12:27.07 | JerJer | omg sipx is pure evil |
12:27.11 | JerJer | jboss ?! |
12:27.31 | JerJer | WTF were they smoking when they wrote that crap |
12:27.36 | JerJer | I want some |
12:27.45 | *** join/#asterisk Blackvel (~blackvel@dsl-213-023-033-142.arcor-ip.net) |
12:28.01 | Blackvel | hiya, anyone using latest bristuff RC7f? DISA still works? |
12:28.14 | Blackvel | Accepting voice call from '90' to 's' on channel 0/2, span 1 |
12:28.37 | Blackvel | exten => s/90,1,DISA,no-password|ctx_blackvel |
12:28.49 | Blackvel | does that still work with asterisk 1.0.5? |
12:29.20 | Blackvel | i forward to the correct context, but then I get a hangup, not a dail tone |
12:29.40 | JerJer | how about keeping it simple? |
12:29.54 | Blackvel | it is simple |
12:29.54 | JerJer | don't try to match callerid when testing |
12:30.00 | Blackvel | and worked all the time |
12:30.08 | Blackvel | but not with new version :( |
12:30.35 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-218-37.dsl.scarlet.be) |
12:30.39 | Blackvel | I used bristuff RC2 with asterisk 1.0.2 and it run really fine |
12:30.49 | Blackvel | maybe there are some big changes in asterisk |
12:31.14 | JerJer | um yeah, it is called progress |
12:31.23 | Blackvel | hehhe |
12:31.53 | Blackvel | are there any informations on voip-info.org if there is something I would have to change? |
12:32.03 | Blackvel | does anything sound familar to you, JerJer? |
12:33.16 | JerJer | nope |
12:33.25 | JerJer | don't run 1.0.2 or bristuff |
12:33.33 | Gh0sty | which type of isdn cards (chipset?) which support NT mode do i need for an asterisk ? |
12:34.03 | Gh0sty | i just read an article that says the hfc card driver is still beta? :s |
12:35.04 | Blackvel | hm |
12:35.04 | Blackvel | no |
12:35.07 | Blackvel | its 1.0.5 now |
12:35.20 | Blackvel | and does not work anymore, but with 1.0.2 everything have been fine |
12:35.41 | Blackvel | Gh0sty: how many lines? 2? |
12:35.51 | Gh0sty | ? |
12:35.55 | Mother_ | any good examples on forwarding a call on busy? |
12:36.02 | Blackvel | how many NT lines do you want to connect? |
12:36.12 | Gh0sty | not sure, start with 1 :) |
12:36.29 | Mother_ | hmmm |
12:36.33 | Gh0sty | i start next week with an asterisk setup |
12:36.44 | Blackvel | hfc isdn card (NT mode) can work with 2 lines |
12:36.57 | Blackvel | otherwise you would have to buy a quadbri or some other cards |
12:37.12 | Gh0sty | its a first test with asterisk for a company, its my school working experience ... |
12:37.31 | Gh0sty | well, all i know is: the company got 3 bri lines |
12:37.33 | clive- | ghosty I never had success wth hfc-s cards, too wacky for a production situation, so I went with eicons |
12:37.42 | Gh0sty | which is connected to an existing pbx |
12:38.35 | Gh0sty | all the phones are isdn phones |
12:38.35 | ManxPower | "the fax machine is nothing but a waffle iron with a phone attached to it." - Grandpa Simpson |
12:38.35 | clive- | oh NT, you need the quad bri then |
12:38.35 | ManxPower | Blackvel, What is your problem? |
12:38.36 | Gh0sty | and i have to fix an asterisk in trial to replace the existing pbx over time |
12:39.02 | Gh0sty | but first stage i should need to show em how all of it works on 1 line with 1 phone ... |
12:39.19 | Gh0sty | and if possible link the asterix up to the existing pbx for now |
12:39.50 | Gh0sty | so they can start buying voip phones and use the existing setup for a while |
12:40.06 | Gh0sty | to come to full asterisk over time |
12:40.30 | *** join/#asterisk benno2 (~benno2@host153-15.pool80182.interbusiness.it) |
12:41.05 | Gh0sty | perhaps i don't even need nt-mode, if they can afford 1 voip phone to start with :) |
12:41.08 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
12:41.32 | Gh0sty | if i can get the asterisk linked up to the existing pbx in a descent way too ... |
12:41.54 | rajo | Gh0sty: does the existing pbx have an S0? |
12:42.07 | Gh0sty | not sure, i'll look into that next week |
12:42.12 | Gh0sty | possibly |
12:42.21 | Gh0sty | its an expesive and large thing |
12:42.35 | rajo | if so, you can hook asterisk up with probably any linux-supported isdn card for first trials |
12:42.37 | Gh0sty | i believe they have around 30 phones and with 3 bri lines ... |
12:42.46 | benno2 | modem connection over an ATA. Impossible even with g711 ? http://forums.speedguide.net/archive/index.php/t-161277.html |
12:43.17 | Gh0sty | well, the most important stuff would be that they can make a call from the asterisk to the internal phones on the existing pbx |
12:43.53 | rajo | Gh0sty: should be rather easy then :) |
12:43.57 | Gh0sty | cause the asterisk would also serve for a daughter company to connect stuff trough vpn connection :) |
12:44.06 | Gh0sty | to make it a simple setup ... |
12:44.45 | Gh0sty | (still wondering if i made the right choice for my school work assignement :s ) |
12:45.03 | ManxPower | Can anyone think of a reason I would get a red alarm only on TWO channels, and it only happens ocasionally. |
12:47.51 | Mother_ | anyone know what happens to a blind transfered call that goes to a busy SIP extension? |
12:48.03 | Mother_ | i.e. how can I get it to transfer/forward back to the caller |
12:48.15 | Mother_ | there are some really vague comments on this on list archives |
12:49.20 | Mother_ | someone said "I got it working" but didn't really say how |
12:49.37 | *** join/#asterisk sysdef (~sysdef@pD9561FF1.dip.t-dialin.net) |
12:49.42 | *** join/#asterisk soulz- (~Soulz-@cm252.sigma237.maxonline.com.sg) |
12:49.48 | soulz- | hello all |
12:56.23 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
12:57.00 | *** join/#asterisk sysdef (~sysdef@pD9561FF1.dip.t-dialin.net) |
13:02.19 | *** join/#asterisk ToyMan (~stuq@user-12lcqq2.cable.mindspring.com) |
13:02.41 | *** join/#asterisk kamranahmad (~root@mbl-82-51-9.dsl.net.pk) |
13:03.51 | kamranahmad | hello |
13:06.33 | kamranahmad | hello i am in |
13:06.41 | kamranahmad | any developer |
13:07.23 | *** join/#asterisk didz_ (didz_@200.218.192.52) |
13:10.00 | Zeeek | Mother_ doesn't it ring back to the phone you tried to transfer from? |
13:10.13 | Zeeek | ooops that was a whaile ago |
13:21.49 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
13:24.55 | BrianR___ | benno2: Not at any reasonable speed. |
13:27.51 | brc_ | http://slashdot.org/article.pl?sid=05/02/23/0246246&tid=215&tid=95 |
13:28.13 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
13:29.58 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com) |
13:33.34 | benno2 | BrianR___: 9600baud would be ok for me |
13:37.14 | *** join/#asterisk ^login^ (~avl@star.ukr.net) |
13:39.35 | *** join/#asterisk montoya (~montoya@200.195.90.104) |
13:40.06 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-86-218.sentechsa.net) |
13:43.02 | *** join/#asterisk B0ngFrOg (~wsmith@c-24-9-253-203.client.comcast.net) |
13:49.43 | *** join/#asterisk lyroy (~lyroy@picachou.csaffluents.qc.ca) |
13:49.46 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-218-37.dsl.scarlet.be) |
13:58.41 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-70-18-115-202.buff.east.verizon.net) |
13:59.26 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
14:00.03 | lyroy | IS there a good guide where I can find information about Netmeeting and Asterisk for videoconferencing.. |
14:00.26 | bjohnson | no |
14:00.38 | *** join/#asterisk znoG (gs@200.115.216.109) |
14:00.45 | bjohnson | MS doesn't want to admit anyone but MS exists |
14:00.57 | bjohnson | asterisk doesn't do videoconferencing |
14:01.03 | Moc | lyroy, netmeeting use h323 :( that in itself is a problem |
14:01.16 | bjohnson | I don't think it will even pass it through .. but I haven't tried |
14:01.28 | Moc | it should pass video packet between phone |
14:01.37 | bjohnson | I thought there was a version that did SIP |
14:04.42 | *** join/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net) |
14:06.38 | pointer-gaim | anyone found a clean fix for the VM "low volume when played on a pc" issue? |
14:07.09 | pointer-gaim | it seems like scaling the volume up before saving the file would make it really loud for people listening to it over the phone |
14:08.03 | *** join/#asterisk eipi (eipi@136-218-114-200.fibertel.com.ar) |
14:08.45 | *** join/#asterisk [ro]nic3try (~iancu@81.181.199.39) |
14:08.50 | [ro]nic3try | re all |
14:11.53 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com) |
14:15.28 | [ro]nic3try | i can't connect mysql cdr to mysql.. i have an 30420 error |
14:15.36 | [ro]nic3try | help ! |
14:15.54 | Blackvel | sorry have been away |
14:16.14 | Blackvel | ManxPower: I have upgraded bristuff to latest (RC7f) |
14:16.20 | Blackvel | which is based on asterisk 1.0.5 |
14:16.47 | Blackvel | DISA accepts the call but does not give me a dail tone (which worked before wtih asterisk 1.0.2) |
14:16.50 | Blackvel | this is the message: |
14:16.51 | Blackvel | Accepting voice call from '90' to 's' on channel 0/2, span 1 |
14:17.14 | Blackvel | but then I get a busy, instead of a dial tone (well, I use DISA for analog pbx inward dailing) |
14:20.10 | [ro]nic3try | i have installed asterisk addons, and i'm tring to use cdr_mysql.. but, when i start asterisk cannot connect to mysql database cdr |
14:21.12 | [ro]nic3try | where should i look for the problem ? |
14:21.23 | *** join/#asterisk lyroy_ (~lyroy@picachou.csaffluents.qc.ca) |
14:27.00 | *** join/#asterisk Guyo (~chatzilla@www.paneura.com) |
14:27.46 | Guyo | i have a problem with configuring zaptel.conf for an hfc bri card and a tdm400p. I am using bristuff... is this the right channel ? |
14:31.02 | tzanger | yes it is the right channel,but I know nothing about the BRI stuff |
14:31.33 | Blackvel | what problem guyo? |
14:32.13 | Guyo | i'm using zaptel and zaphfc at the same time, but can't get zaptel.conf right to make them work at the same time, always getting ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
14:32.20 | *** join/#asterisk thefallen (PolarBear@thefallen.user) |
14:32.22 | Blackvel | maybe to configure both cards? :) |
14:32.31 | Guyo | span #1 is the HFC card |
14:32.43 | Blackvel | had you tried each card alone? |
14:32.45 | *** join/#asterisk JohnnyC (~JoaoCorre@81.193.116.63) |
14:32.48 | Guyo | yep |
14:32.50 | Guyo | both work |
14:33.04 | Blackvel | ah, so patching libpri package with bristuff does not fail working for tdm400p then |
14:33.26 | *** join/#asterisk cbachman (~cbachman@129.105.7.250) |
14:33.32 | Guyo | :-| I need to get it work :-| |
14:33.38 | Blackvel | too bad, but I have not yet the experience about zaptel.conf and configuring both cards at the same time :( |
14:33.59 | Blackvel | what bristuff btw? |
14:34.02 | Blackvel | RC7f? |
14:34.04 | Guyo | rc7f |
14:34.14 | Guyo | yes, just got it... i'm gonna try rc5 |
14:34.15 | Blackvel | I even can not make it work with DISA anymore :) |
14:34.32 | Guyo | .... |
14:35.55 | Hmmhesays | hmm I'm having trouble getting "read" to work from an agi |
14:35.57 | jobi | hi all |
14:36.48 | jobi | I'm trying to configure an PRI/E1 between an TE410P and a E1 over IP equipment |
14:36.51 | *** join/#asterisk coppice (~chatzilla@245.195.17.210.dyn.pacific.net.hk) |
14:37.05 | Hmmhesays | nevermind I figured it out |
14:37.21 | jobi | but whatever I try the equipment reports a LOF alarm, and zaptel reports a yellow alarm |
14:37.38 | Hmmhesays | ("EXEC READ var|soundfile") for anyone that cares |
14:38.04 | Guyo | Blackvel: Thanks, i solved that |
14:38.21 | Guyo | seems it's new of rc7f |
14:38.23 | jobi | is there a way I can ask zaptel why it raised a yellow alarm? |
14:38.31 | Guyo | i'll notify the author |
14:38.42 | Blackvel | Guyo: what is new? the problem? |
14:38.51 | Blackvel | rc5 works? |
14:38.58 | Guyo | yes, using rc5 it works, must be the rework of zaphfc code |
14:39.08 | didz_ | jobi you configured the flag yellow for this span in /etc/zaptel.conf |
14:39.43 | jobi | didz_: no |
14:39.56 | jobi | span=1,0,0,ccs,hdb3 |
14:40.54 | *** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
14:41.32 | Blackvel | hmmm RC7f uses new zaptel.conf and zapata.conf? |
14:42.20 | Guyo | don't think so |
14:42.37 | *** join/#asterisk djin (~marius@62.58.40.196) |
14:43.04 | HitTop | I just got a polycom ip500~ I want to set it to register to my asterisk server. but under Sip Conf., there's no field for inputing userid and password. Can anybody help me? |
14:44.48 | |Vulture| | HitTop: you talking about the web interface? |
14:44.54 | |Vulture| | its under Register or something |
14:45.02 | |Vulture| | far right link |
14:45.26 | _Brian | HitTop: in your sip.conf you will define the server to register to....in the phone.conf file, you define what username/password to utilize |
14:49.54 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
14:50.53 | sivana | anyone using a channel bacnk with an MICS? |
14:52.06 | ManxPower | sivana, I think so. |
14:53.47 | clive- | does anyone know what this means : Received iseqno 10 not within window 11->11 |
14:58.25 | stevekstevek | clive-: probably means you got a duplicate IAX2 full frame. |
14:59.09 | *** join/#asterisk fwittekind (rom@pcp0010183025pcs.columbus.in.indy.comcast.net) |
15:00.43 | *** part/#asterisk fwittekind (rom@pcp0010183025pcs.columbus.in.indy.comcast.net) |
15:05.47 | *** join/#asterisk Frantic (~ab@TechnologicPartners35.dsl.concentric.net) |
15:06.19 | Frantic | hi all- anyone had a problem with current CVS that the sip peers list is suddenly empty? |
15:09.14 | *** join/#asterisk isamar (~isamar@p8131-ipadfx21sasajima.aichi.ocn.ne.jp) |
15:09.17 | isamar | Hi folks |
15:09.30 | isamar | having problems with oh323 and a Cisco 2600... anybody using oh323? |
15:09.49 | isamar | Q.931 error cause 24 |
15:10.39 | clive- | stevek thanks...its on a very high jitter connection, I thought maybe I was like way off on a late packet |
15:12.56 | stevekstevek | clive-: no, it's really not a problem, I think.. |
15:13.26 | stevekstevek | clive-: basically, IAX2 will retransmit full (reliably-sent) frames if it doesn't receive an ACK in 2*RTT. |
15:14.04 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
15:14.04 | *** mode/#asterisk [+o anthm] by ChanServ |
15:14.09 | stevekstevek | clive-: so, in this case, the receiver actually got it, but either the ACK got lost, or the ACK took longer than expected to get back to the sender. |
15:14.22 | stevekstevek | clive-: so it retransmitted. The receiver will just ignore the duplicate. |
15:15.10 | bjohnson | sivana: I'm using fxo/fxs devices with a CICS |
15:15.55 | Frantic | anyone had a problem with current CVS that the sip peers list is suddenly empty? |
15:16.11 | *** join/#asterisk |Barcode (~uid@h-68-165-204-41.chcgilgm.covad.net) |
15:16.34 | *** join/#asterisk dsmouse (~mouse@rrcs-24-199-146-243.midsouth.biz.rr.com) |
15:16.42 | sivana | bjohnson: I'm working on it. Issue with the channel bank or MICS not hanging up |
15:17.09 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com) |
15:18.22 | |Barcode | I have two TDM400P cards with FXO ports. I can not figure out how to determine which line a call came in on in my extensions.conf. |
15:18.52 | bjohnson | sivana: same here. the fxo reinitiates a call into the CICS .. I call it a ghost call |
15:19.06 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
15:19.26 | bjohnson | sivana: I've been playing with the delay before the fxo gets answered on my SPA 3000 units .. hoping to fix that problem |
15:19.27 | drumkilla | |Barcode: an easy way is to put each FXO in a different context |
15:19.27 | sivana | bjohnson: my issue is that they have 19 lines. We've moved line 19 to port 1 on the channel bank |
15:19.38 | sivana | and forwarded that number to a temp number we assigned |
15:20.26 | sivana | <PROTECTED> |
15:20.48 | dsmouse | I saw the licensplate NXX7823 on the way to work this morning. I though it was a odd patern |
15:20.49 | |Barcode | drumkilla: Do you mean in the zapata.conf file, or in extensions.conf? |
15:21.30 | drumkilla | |Barcode: well, in zapata.conf you say a different "context=blah" for each channel, and that corresponds to different contexts in extensions.conf |
15:21.46 | drumkilla | |Barcode: or, if you want the channel in a variable, it's available as ${CHANNEL} |
15:22.03 | drumkilla | but the context thing is probably what you want |
15:23.09 | |Barcode | Ok, let me give this a shot. Different contexts does seem what I want. |
15:28.39 | |Vulture| | whats the "show database" command? |
15:28.49 | |Vulture| | brain fart... |
15:29.05 | bjohnson | hahahaha - Toronto1-Suncall is launching a Canada-wide service (only in selected cities) |
15:29.26 | |Vulture| | oh "database show" lol |
15:30.05 | *** join/#asterisk jcims (~jcims@cpe-69-135-121-57.columbus.rr.com) |
15:36.17 | bkw_ | WHATS UP PEOPLE!!!!!!! |
15:36.43 | bkw_ | sivana, well stop calling yourself :P |
15:36.49 | mikegrb | bkw_: http://thegrebs.com/~michael/rasterbate.jpg <-- I will put your picture next to it, OK? |
15:38.49 | *** join/#asterisk sabre (~urfos@69.149.209.83) |
15:39.54 | sivana | bkw_: kewlstart solved the problem |
15:40.17 | *** join/#asterisk km- (~km-@brdgw1.rttx.com) |
15:40.31 | km- | Howdy! |
15:42.40 | goatmilk | km-: lo. |
15:42.49 | km- | goatmilk: how's it goin? |
15:42.53 | km- | ~seen bkw_ |
15:42.59 | jbot | bkw_ is currently on #asterisk (4h 33m 38s). Has said a total of 38 messages. Is idling for 6m 16s |
15:43.02 | goatmilk | km-: peachy. yourself? |
15:43.25 | km- | goatmilk: it's 20 of 11:00 and I'm at work.. That means, 1:20 till lunch, so, I'm rather happy :) |
15:43.31 | km- | ~seen kram |
15:43.32 | jbot | kram is currently on #asterisk |
15:43.58 | mmlj4 | um, kram is mark spencer? |
15:44.11 | km- | yeah, why? |
15:44.36 | mmlj4 | because I met him last weekend in mobile, and wanted to say "hi" |
15:44.36 | goatmilk | km-: class for me in 30 min :( |
15:44.49 | km- | hehe, yeah, he's mark spencer |
15:44.56 | km- | or rather, "The Mark Spencer" |
15:44.57 | goatmilk | mmlj4: he's a nice guy, isn't he? |
15:45.04 | mmlj4 | yes, very |
15:45.07 | km- | kram is awesome |
15:45.08 | ariel_ | morning all |
15:45.20 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
15:45.54 | goatmilk | km-: agreed :) |
15:47.03 | km- | ah shoot, he's at a conference? |
15:47.11 | goatmilk | ariel_: now I am jealous |
15:47.37 | km- | maybe twisted knows |
15:47.47 | goatmilk | yeah km- |
15:47.51 | goatmilk | he's down there for something |
15:48.16 | km- | I've got a T1 problem that I fear only Mark has the knowledge to solve |
15:48.28 | km- | because I think it's a bug in chan_zap |
15:48.32 | goatmilk | you callin everypne stupid ;-) |
15:48.36 | km- | not at all |
15:48.38 | ariel_ | It's the km- what is the problem. |
15:48.41 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
15:48.48 | km- | I'll be happy to go over what the problem is in case anyone else wants to take a shot at it |
15:48.48 | goatmilk | i know i am teasin' |
15:49.09 | twisted[work] | Maybe I know what? |
15:49.13 | km- | I'm having some issues with e&m wink on a NEC PBX, talking to Asterisk. |
15:49.14 | km- | The NEC system is sending the digits to asterisk as the user dials them, and Asterisk |
15:49.14 | km- | is timing out within 1 second of silence, as opposed to the nominal 3 that you usually |
15:49.14 | km- | get with a normal zap channel. Is there a way to up that timeout? |
15:49.16 | goatmilk | twisted[work]: ho hum |
15:49.18 | km- | twisted: ^^^^ |
15:49.42 | twisted[work] | what does your dialplan look like? |
15:49.52 | twisted[work] | If you're using a catchall, the timeout will be much shorter than if you burst the digits |
15:49.53 | km- | lemme log in and do some cuttin and pastin |
15:50.01 | dsmouse | DON'T PASTE THE DIALPLAN |
15:50.07 | km- | hahahaha |
15:50.08 | km- | dude |
15:50.12 | km- | I've been using asterisk for like 2 years now |
15:50.14 | km- | I know better |
15:50.16 | dsmouse | :) |
15:50.16 | twisted[work] | dsmouse, calm down.. I'm the one that will deal with that. |
15:50.18 | km- | I know what he's asking for |
15:50.41 | twisted[work] | see that pretty @ symbol? If someone gets out of hand, I have the authoritah |
15:50.44 | codebreaker | question: if i use PSTN(ISDN)--CApicard/zaptelcard-->[asterisk]-->othercapicard/zaptelcard--->ISDN phone. can i still see the number whois calling me from PSTN? |
15:50.47 | twisted[work] | ;) |
15:50.48 | km- | here's the entry from zapata.conf |
15:50.54 | km- | group=2 |
15:50.54 | km- | immediate=no |
15:50.54 | km- | signalling=em_w |
15:50.54 | km- | context=incomingpbx |
15:50.54 | km- | channel=>36-48 |
15:51.01 | *** join/#asterisk __Sparks_ (ringding@bb-194-6-118-37.ukonline.co.uk) |
15:51.01 | twisted[work] | mmkay |
15:51.05 | twisted[work] | i meant in the dialplan |
15:51.12 | *** join/#asterisk thieumS (~darkmind@nanterre-7-82-229-210-142.fbx.proxad.net) |
15:51.14 | km- | just covering all the bases |
15:51.16 | km- | just a second |
15:51.21 | twisted[work] | ie, what's the first 3-4 lines of [incomingpbx] |
15:51.34 | __Sparks_ | Am i correct in thinking, any US number beginning with an 8 is toll-free? |
15:51.38 | km- | [incomingpbx] |
15:51.38 | km- | exten => _9XXX,1,Congestion |
15:51.38 | km- | exten => _1XXXXXXXXXX,1,Dial(Zap/g1/${EXTEN}) |
15:51.38 | km- | exten => _011XXXXXXXXXXXX,1,Dial(Zap/g1/${EXTEN}) |
15:51.39 | twisted[work] | __Sparks_, no. |
15:51.43 | twisted[work] | hmm |
15:51.48 | km- | _Sparks: 814 is an area code in PA |
15:51.50 | twisted[work] | that SHOULD be okay.. |
15:52.08 | twisted[work] | you might try using immediate=yes |
15:52.08 | km- | twisted: yeah, it's really weird, if I dial the number quickly, I can get it all out before the timeout occurs |
15:52.11 | __Sparks_ | okay, is there a list somwhere! - or is it just 800 and 888? |
15:52.13 | km- | twisted: but, if I dial like a user |
15:52.17 | jsolares | __Sparks_: afaik 888, 877, 866 and 800 are toll free |
15:52.21 | km- | _Sparks: 800, 888, 877, 866 |
15:52.27 | __Sparks_ | thanks! |
15:52.36 | twisted[work] | and using s,1,DISA(no-password|myrealcontext) |
15:52.42 | km- | twisted; if I dial 1(pause)484(pause) like most people do, I'll just get the 1 |
15:52.47 | twisted[work] | right |
15:52.48 | km- | OMG |
15:52.49 | twisted[work] | see my suggestion |
15:53.00 | ariel_ | km-, you have to add a longer wait on the wink side. manxpower hd something similar there is a setting for that. |
15:53.01 | km- | Why the hell didnt tzanger and I think of that last night |
15:53.10 | twisted[work] | km-, *shrug*.. this is what I get paid to think of :) |
15:53.14 | km- | ariel: you mean postwink and prewink? |
15:53.36 | twisted[work] | ariel_, that may work too, although, the winking is a line state change, irrc. |
15:53.39 | ariel_ | km-, don't remember which one he did but he added =270 to it. I think. |
15:53.39 | km- | hmm, Manx was in the conversation last night, maybe he just didnt relate the problems |
15:53.40 | twisted[work] | iirc, rather. |
15:53.56 | bjohnson | __Sparks_: in NA |
15:54.03 | *** join/#asterisk nix000 (~nixman@CPE0006256d190c-CM0011aeff5db6.cpe.net.cable.rogers.com) |
15:54.04 | twisted[work] | or, it could be that the winktime is actaully like I said - line state change |
15:54.10 | twisted[work] | dtmf != wink |
15:54.14 | mtqh | Anyone use agent groups? |
15:54.25 | twisted[work] | although It may work.. *shrug* |
15:54.28 | km- | twisted: you know what's weird, if I set the line to featured, the dialing works fine. |
15:54.34 | twisted[work] | ahhhhh |
15:54.35 | twisted[work] | well |
15:54.36 | twisted[work] | there ya go then |
15:54.37 | km- | twisted: but, asterisk complains that the line really isn't feature d |
15:54.43 | km- | twisted: and switches back to em_w |
15:54.44 | twisted[work] | e&m w/feat_d |
15:54.48 | km- | and if I set it to featd, the DID's dont work |
15:54.53 | km- | so, it's really em_w |
15:54.54 | twisted[work] | oh |
15:54.55 | twisted[work] | hmm. |
15:55.01 | *** join/#asterisk ManxPower (~eric@24-116-82-96.cpe.cableone.net) |
15:55.05 | km- | but for some reason, switching to featd improves the problem |
15:55.05 | goatmilk | problem solved.. |
15:55.08 | km- | Manx! |
15:55.10 | twisted[work] | I know e&m wink is meant to get it's digits in a burst |
15:55.12 | km- | but causes another |
15:55.21 | twisted[work] | *CLID*NUMBER* |
15:55.38 | km- | Manx: ariel says that you might have had a problem similar to the one I was explaining last night? Something about upping one of the wink times to improve the timeout? |
15:55.47 | twisted[work] | anywho, i'm off to my task list. |
15:56.01 | km- | twisted: thanks for the info, I think the DISA hack will work beautifully with immediate=yes |
15:56.13 | *** join/#asterisk Derkommissar (~Loving@fl-southhub-u1-c6-0a-41.miamfl.adelphia.net) |
15:56.13 | twisted[work] | it should |
15:56.25 | Derkommissar | when i type show translation i get this. |
15:56.26 | km- | that may be the answer I finalize on, but I have all day before my next maintenance window |
15:56.33 | Derkommissar | <PROTECTED> |
15:56.33 | Derkommissar | <PROTECTED> |
15:56.43 | km- | so I'll keep asking in case some people know the "Right Way" to fix it |
15:56.55 | Derkommissar | does that mean that i can translate from g729 to ulaw ? |
15:57.01 | jsolares | yes |
15:57.39 | mtqh | I can't get agent groups to work.....It will only call the first member of the group |
15:57.48 | Derkommissar | what is the 3 suposed to mean ? |
15:58.15 | km- | [ Context 'preincomingpbx' created by 'pbx_config' ] |
15:58.15 | km- | <PROTECTED> |
15:58.17 | km- | word |
15:58.17 | km- | hehe |
15:58.34 | *** join/#asterisk gonzo- (~gonzo@SIRIUS-ats227-UTC.ukrtel.net) |
15:58.57 | jsolares | Derkommissar: the lag in ms the transcoding adds |
15:59.01 | jsolares | i think |
15:59.18 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
15:59.21 | Hmmhesays | good lord I hate people that think static == public and dynamic == private in regards to ip addresses |
15:59.51 | jsolares | hehe |
16:00.04 | *** part/#asterisk djin (~marius@62.58.40.196) |
16:00.05 | goatmilk | off to class. good luck km- ! |
16:00.06 | twisted[work] | Derkommissar, the 3 is the amount of time it takes to translate between the two |
16:00.08 | Damascene | idon't follow. static is just static nat, one to one mapping. |
16:00.23 | Damascene | so what do most people think about static nats? |
16:00.23 | twisted[work] | 3ms is pretty damn good. What kind of system is this you have? |
16:00.45 | Damascene | that is 'has' to be public->private mappings? |
16:01.03 | twisted[work] | you're thinking of PAT, right? |
16:01.16 | Hmmhesays | heh, most of the people i work are idiots |
16:01.23 | Hmmhesays | and they think exactly what I said |
16:01.45 | twisted[work] | Hmmhesays, work at an ISP do ya? |
16:01.49 | *** join/#asterisk bill522 (~bill522@182-30.201-68.swfla.res.rr.com) |
16:01.55 | Hmmhesays | unfortunately not |
16:02.08 | twisted[work] | ah... most ISP's i've dealt with think that way, believe it or not |
16:02.31 | twisted[work] | in fact, my ISP told me I had a "randomly changing static address" once. |
16:02.35 | Damascene | OH, they think static nat == PAT (many to one?) |
16:02.57 | Hmmhesays | no... they think static addresses are always public |
16:03.03 | km- | goat: thanks man! |
16:03.04 | Hmmhesays | and dynamic addresses are always private |
16:03.06 | twisted[work] | Damascene, no, i was making a joke :P |
16:03.07 | Damascene | Hmmhesays: OH MY |
16:03.20 | twisted[work] | unfortunately humor is lost on irc |
16:03.29 | Hmmhesays | then I point them to a guide of ip addressing... and it melts their brain |
16:03.38 | twisted[work] | anywho |
16:03.42 | Hmmhesays | lol |
16:03.58 | Damascene | Hmmhesays: melt them more by asking them to subnet anythin beyond a /24 or... as they say two-five-five dot two-five-five dot two-five-five dot zero |
16:04.21 | Hmmhesays | heh... i'd rather reach through the screen and strangle them |
16:04.35 | twisted[work] | nah, be nice to them, they control the intarweb |
16:04.43 | twisted[work] | :P |
16:04.46 | Hmmhesays | the intardweb? |
16:04.52 | twisted[work] | that too |
16:04.55 | Hmmhesays | lol |
16:05.09 | Hmmhesays | did you renew your tfark account this month? |
16:05.14 | twisted[work] | yep |
16:05.17 | twisted[work] | it's on automatic renewal now |
16:05.22 | twisted[work] | ;) |
16:05.29 | Hmmhesays | haha, it keeps me from going insane at work |
16:05.34 | twisted[work] | same here |
16:06.17 | twisted[work] | speaking of which |
16:15.43 | *** join/#asterisk isamar (~isamar@p8131-ipadfx21sasajima.aichi.ocn.ne.jp) |
16:15.46 | isamar | hi folks |
16:15.59 | isamar | having problems to compile latest chan_oh323 |
16:16.01 | isamar | chan_oh323.c:5192: warning: passing arg 4 of `ast_channel_register' from incompa |
16:16.02 | isamar | tible pointer type |
16:16.57 | *** join/#asterisk gr0mit (~gr0mit@router1.txrx.org.uk) |
16:24.29 | bill522 | can anyone tell me how I can call into asterisk by outside PSTN and transfer to SIP? |
16:24.44 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
16:24.47 | mutilator | hey all |
16:25.03 | Beirdo | heh |
16:25.29 | mutilator | having a problem, i assume related to bandwidth or maybe latency? when this person calls, you can hear him great, but after ~3 minutes of talking all he hears is breaking up and sounds like crap |
16:25.39 | mutilator | and i can still hear him perfectly fine |
16:26.07 | gr0mit | bill522, we need a bit more info on your configuration before we can answer your question |
16:26.16 | mutilator | going from ata -> g729 -> asterisk -> ulaw -> as5350 -> pstn |
16:27.06 | bill522 | gr0mit, okay, I just have a X100P with my analog line, I would like to dial in on it, and transfer to my FWD or something like that |
16:27.12 | mutilator | was limited to 384kbit, where the 729 codec was used, i just bumped to 512kbit to see if it solved problem |
16:27.39 | gr0mit | k. do u have a SIP account already with fwd? |
16:27.46 | km- | hrm |
16:28.00 | bill522 | yes gr0mit, but I would like to choose, I have many SIP accounts |
16:28.03 | km- | I don't think I can transfer calls from the nec pbx to the asterisk system |
16:28.42 | gr0mit | cos then you can take your inbound call from your X100p and dump it into a context which has a Dial(mysipprovider) line |
16:28.54 | gr0mit | with whatever dial command options you want |
16:29.24 | bill522 | yes, preferably my voip phone dialplan |
16:29.27 | *** join/#asterisk Slainte (~Slainte@66.55.112.13.ppp.northrock.bm) |
16:29.38 | gr0mit | you can send the call to multiple SIP channesl, plus your voip phone. |
16:29.42 | *** join/#asterisk kamran (~kamran@mbl-82-51-9.dsl.net.pk) |
16:29.53 | kamran | hello |
16:30.02 | bill522 | yes gr0mit, but need to autheenticate first, not just transfer every1 |
16:30.25 | Slainte | Anyone know how to setup the dialplan on a polycom IP600. W=users need to be able to dial the missed call list, but cant because there is no 9 in front of the number. |
16:30.57 | *** join/#asterisk MicH323 (~micosat@host217-44-194-118.range217-44.btcentralplus.com) |
16:31.06 | gr0mit | please describe exactly what you are trying to do, bill522 |
16:31.14 | MicH323 | Hi All |
16:32.10 | bill522 | okay, call into * with pstn, maybe press 5, get prompt for pin, then dialtone to use current sip dialplan like dial 8{FWD}, 7{IAXTEL}, etc |
16:32.42 | mutilator | meh |
16:32.54 | mutilator | searching cisco's website sucks |
16:32.56 | MicH323 | Just setup Asterisk on Solaris. Everything running. Configured to FWD via IAX, Now trying to attach an Cisco ATA to it, its failing with registration failed |
16:32.57 | __Sparks_ | I am after a little help with my sip.conf file! - this line in sip.cong "exten => _8.,1,Dial(SIP/${EXTEN:1}@sipgate1)" seems to make calls prefixed with a 8 go to sipgate- stipiing off the 8 - what bit of this line stripps off the 8? |
16:33.02 | mutilator | can't find anythin |
16:33.23 | km- | __Sparks: the $EXTEN:1 part strips off the 8 |
16:33.27 | bill522 | the EXTEN:1 _Sparks_ |
16:33.28 | km- | __Sparks: chnage it to ${EXTEN} |
16:33.45 | kamran | any one know how to call Dial or HangUp application from one application (maybe changing priority) |
16:33.51 | km- | I need to get me a 7960 again |
16:33.51 | km- | hehe |
16:34.00 | __Sparks_ | km- thanks! |
16:34.06 | km- | no problem |
16:34.08 | bill522 | gr0mit? |
16:34.29 | bjohnson | bill522: look at the user authentication wiki page |
16:34.42 | bjohnson | linked to from the tips and tricks page |
16:34.43 | MicH323 | Simle question. Where do you configure users? |
16:35.04 | bjohnson | I think it has examples for what you want to do |
16:35.10 | bjohnson | MicH323: voicemail.conf |
16:35.15 | bill522 | ty bjohnson, will that help with what I am trying to do? |
16:35.35 | MicH323 | will voicemail.conf have ATA registration passwords? ? ? |
16:35.41 | bjohnson | MicH323: it's about the only place where the concept of 'users' is used |
16:35.50 | bjohnson | MicH323: you want the same page that bill wants |
16:35.54 | gr0mit | ok, i see what you are trying to do...not sure i undersatnd why. |
16:36.04 | bjohnson | bill522: I think so, dial in, authenticate, dial out |
16:36.18 | gr0mit | if you tell me why you are trying to do it there may be a more elegant sln we can suggest |
16:36.23 | kamran | hi developres any one know how to call Dial or HangUp application from one application (maybe changing priority) |
16:36.43 | MicH323 | So voicemail.conf has phonenumbers mapped to password? |
16:37.23 | bjohnson | MicH323: no |
16:38.04 | bjohnson | gr0mit: I use it to allow cell phone users to dial in, authenticate, and dial out = 3c/minute for LD instead of 25c/min |
16:38.12 | bjohnson | (using a toll free number) |
16:38.32 | MicH323 | ok, which file does the Authenticate commands go into? |
16:38.42 | bjohnson | MicH323: extensions.conf |
16:39.20 | bjohnson | MicH323: So voicemail.conf has phonenumbers mapped to password? .. nothing is that direct .. you have to config you dialplan to do something like that |
16:39.46 | MicH323 | lolz! Thans bjohnson |
16:40.40 | bjohnson | gr0mit: I use the same concept for all incoming calls to authenticate who gets direct access to dial internal extensions |
16:40.42 | BuckRogers | hello hello |
16:41.02 | MicH323 | So something like: USERA=SIP/6601 and then exten => 6601,1,Dial(SIP/6601) |
16:41.14 | bjohnson | MicH323: no |
16:41.19 | MicH323 | ops! |
16:41.22 | greg_work | can someone read this: http://sourceforge.net/mailarchive/forum.php?thread_id=6634469&forum_id=42627 and suggest the best way to implement it? (really, just the part at the bottom about replacing Dial() ) .. can i do a macro, or would a goto work? |
16:41.28 | bill522 | ty gr0mit lol searching for info now |
16:41.31 | BuckRogers | we also do a simialar dial in dail out method] |
16:42.05 | bjohnson | greg_work: you want the superdial macro from the wiki I think |
16:42.12 | greg_work | no |
16:42.34 | greg_work | basically i'm trying to "fix" a dialstring |
16:42.38 | tzanger | ugh |
16:42.49 | tzanger | I fucking hate automated dialers with AUTOMATED FUCKING MESSAGES!!! |
16:43.09 | greg_work | if i dial 5551234, i want it to dial "5551234" on my ZAP trunk, but "16135551234" on my voip trunks |
16:43.20 | bjohnson | greg_work: yes |
16:43.26 | Nugget | greg_work: that's easy enough. |
16:43.36 | greg_work | similarly, if i dial 16135551234, i want it to dial 5551234 on my ZAP trunk, but 16135551234 on my voip trunk |
16:43.36 | bjohnson | use the superdial macro to try different channels in sequence |
16:43.50 | bjohnson | then use 1613${EXETN} for the voip one |
16:43.53 | greg_work | i already have done the sequence thing, and its more powerful and easier to use than superdial |
16:44.00 | bjohnson | instead of just ${EXTEN} |
16:44.12 | greg_work | and i'm writing this into AMP so it's actually easy to manage |
16:44.14 | Derkommissar | can someone tell me whats wrong with this invit |
16:44.17 | Derkommissar | invite |
16:44.18 | Derkommissar | http://www.pastebin.com/245447 |
16:44.32 | Derkommissar | its not sending the audio request |
16:44.42 | Derkommissar | in sip.conf i configured it for ulaw |
16:44.50 | bjohnson | greg_work: append the 1 and area code to the local number depending on what channel it goes out on |
16:45.02 | greg_work | that's what i'm trying to do ;) |
16:45.13 | bjohnson | greg_work: don't change the pattern match |
16:45.23 | bjohnson | greg_work: I'm trying to tell you how I have it working |
16:45.28 | bjohnson | WORKING |
16:45.50 | greg_work | i know its easy to do by just hardcoding into the dialplan |
16:46.29 | greg_work | i'm trying to write it so you can put "1613+NXXXXX" in a web-based interface, then it makes exten=>_NXXXXXX,1,Dial(1613${EXTEN}) |
16:46.54 | greg_work | or 1613|NXXXXX makes exten=>_1613NXXXXXX,1,Dial({$EXTEN:4}) |
16:47.04 | bjohnson | ok..then have the user enter the 1613 somewhere are their local area code |
16:47.11 | bjohnson | check for it, and remove it |
16:47.19 | bjohnson | (for local pstn calls) |
16:47.19 | greg_work | ok but here's the complicated part |
16:47.26 | Juggie | bj, why not just check and see if the call is local |
16:47.38 | bjohnson | Juggie: how? |
16:47.40 | greg_work | my area code is 613. i can dial 613-544-xxxx locally.. but 613-789-xxxx is long-distance |
16:47.51 | Beirdo | greg_work, and what about 819, any of it local? |
16:47.54 | greg_work | no |
16:48.01 | Beirdo | ahh, so not Ottawa. |
16:48.06 | Beirdo | that makes life simpler |
16:48.06 | bjohnson | greg_work: yes .. easier to go other way and append rather than remove |
16:48.08 | gr0mit | bill522, why don't you use caller id on the inbound call to send you into a special context |
16:48.16 | greg_work | 1613+789XXXX |
16:48.17 | gr0mit | and then use app_DISA |
16:48.18 | Derkommissar | can someone take a look at this? |
16:48.22 | greg_work | 1613|544XXXX |
16:48.36 | Juggie | greg, i had to solve this issue just the other day |
16:48.51 | codebreaker | how can i check if asterisk has sucesfully registered at iaxtel.com? |
16:48.56 | Juggie | i am in ottawa however so we have local dialing across two area codes |
16:49.04 | Juggie | and u dial them like they are in the same exchange. |
16:49.06 | bjohnson | greg_work: I do a pattern match on the 7 digit number and append the 1613 to outgoing if through a voip channel |
16:49.19 | bjohnson | I do not append it if going out pstn directly |
16:49.22 | gr0mit | exten => s/yourcellphonecallerid,1,Goto(cellphone,s,1) |
16:49.24 | greg_work | yeah thats fine |
16:49.31 | greg_work | here, hold on, i'll show you my implementation of superdial |
16:49.45 | greg_work | and EXACTLY what i'm trying to do here |
16:49.53 | Juggie | greg_work, use http://members.dandy.net/~czg/search.html to implement a database of local areacodes/exchanges |
16:50.14 | Juggie | then use some agi be it perl/php etc. to do a lookup and see if its a local call or not... |
16:50.19 | greg_work | http://pastebin.ca/6340 |
16:51.06 | *** part/#asterisk sysdef (~sysdef@pD9561FF1.dip.t-dialin.net) |
16:51.28 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
16:51.45 | bkw_ | Juggie, so does telcodata.us |
16:52.37 | Juggie | bkw, thats all i did, we needed 10 digit dialing for a conference app.... so that was all i did to implement that.... just lookup see if its local or no and then handle it in extensions.conf |
16:52.38 | bjohnson | greg_work: looks like it does the same thing but hides all the macro args in external arrays .. I guess it depends what you mean by easier to use when comparing to superdial .. but you do whatever works for you |
16:52.41 | Juggie | works perfect. |
16:52.50 | codebreaker | how can i call from PSTN in Germany to an iaxtelaccount/number |
16:53.03 | greg_work | bjohnson: well, thats more suited to my use where it's built automatically from the web interface |
16:53.15 | bjohnson | codebreaker: make a gateway or check iaxtel or fwd to see if they have one |
16:53.25 | bjohnson | codebreaker: might be other ways in as well |
16:53.31 | greg_work | Juggie: thanks... how do i use this thing though? :p |
16:53.55 | greg_work | oh nm, ok .. very non intuitive interface ;P |
16:54.29 | codebreaker | bjohnson: the talk on the website about voicepulse. i dondt know what this is. the numbers there i ve tried but fails for me |
16:54.50 | bjohnson | voicepulse is a voip provider |
16:54.56 | km- | anyone here have an international dialing extension that they'd be willing to share with me? |
16:54.59 | Juggie | greg_work, are you in ottawa? |
16:55.05 | greg_work | Juggie: kingston |
16:55.15 | codebreaker | ah. ok. thanks |
16:55.30 | greg_work | hm.. this is very interesting. i wonder if i could build this into the interface so it could automatically populate.... |
16:55.34 | bjohnson | greg_work: you don't like the idea of allowing the dialing user to dial the 7 digit number? |
16:55.42 | Derkommissar | what does m=audio 10840 RTP/AVP |
16:55.48 | *** join/#asterisk devel (~devel@wiggum.digitalcoven.com) |
16:55.56 | Juggie | greg, all you want to do is decide if a num is local or not, if it is dial on a zap chan, and if its not, dial on iax/sip or whatever? |
16:56.07 | greg_work | bjohnson: hm? yes, thats what i'm trying to allow. if you dial 7 digits now, and it tries to dial out on voip, it will fail |
16:56.08 | bjohnson | why decide? |
16:56.42 | bjohnson | greg_work: so append the 1613 to ${EXTEN} when going out through voip? |
16:56.54 | greg_work | Juggie: no.. i have a routing screen, where you can set patterns and then a sequence of trunks to try |
16:57.05 | bjohnson | add the option to append the 1613 |
16:57.10 | Juggie | greg, thats a little holey :) |
16:57.16 | Juggie | what if someone dials 16135551212 |
16:57.20 | Juggie | but the call is local |
16:57.29 | bjohnson | what does Bell do? |
16:57.30 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
16:57.30 | greg_work | thats the exact problem i'm trying to fix ;) |
16:57.34 | Juggie | its gonna go through your ld when it didnt have to |
16:57.34 | greg_work | bjohnson: complain ;) |
16:57.44 | bjohnson | if it goes out pstn and it's local, Bell tells you it's a local call |
16:57.56 | Juggie | greg, all you have to do is this. |
16:57.58 | bjohnson | if it goes out voip and it's a local call, it works |
16:58.00 | nestAr | match your local AREA codes |
16:58.15 | nestAr | exten => _613NXXNXXX |
16:58.36 | Juggie | greg, take this http://members.dandy.net/~czg/lprefix.php?exch=155000&dir=1 make a database out of it. |
16:58.48 | greg_work | ok the routing is like (we'll do this for the general case of all of 613 being local) .. NXXXXX is the pattern, trunk priority is: ZAP/g0, IAX2/voip second route, 1NXXNXXXXXX, trunks: IAX2/voip, ZAP/g0 |
16:58.59 | Juggie | then write an agi script, to look up the number dialed against that database, if it returns a record the number is local |
16:59.05 | bjohnson | all of this to avoid a dialing user getting Bell telling them that it's really a local call? |
16:59.06 | bjohnson | wow |
16:59.07 | Juggie | if it doesnt, then its long distance. |
16:59.15 | greg_work | sigh |
16:59.22 | greg_work | i'm making a general-case web-interface here |
16:59.28 | Juggie | bj, its called proper routing, maby they dont know the call is local |
16:59.35 | Juggie | why waste money on LD if they dont have to. |
16:59.35 | greg_work | if it was just hardcoded, it wuold be very simple |
16:59.45 | bjohnson | Juggie: ? waste money on LD? |
16:59.59 | bjohnson | the call won't be placed if it's a local call!! |
17:00.02 | Juggie | bj, if someone dials 16135551212 and the number is local |
17:00.08 | Juggie | it will go through iax/sip when it shoudnt have |
17:00.09 | bjohnson | Bell gives a message |
17:00.10 | Juggie | smarten up. |
17:00.42 | greg_work | Juggie: that would be fixed by making a route like this: patterns: 1613NXXXXXX, NXXXXXX and then trunks: ZAP/g0, IAX2/voip |
17:00.57 | greg_work | 16135551234 will match the pattern, and try to dial on ZAP/g0 first |
17:01.14 | greg_work | however.. before ZAP/g0 can dial that number, it needs to drop the 1613 |
17:01.20 | greg_work | that's the part i'm working on now |
17:01.20 | Juggie | greg_work, thats fine for pri, but not analog as they will hear the bell message. |
17:01.26 | greg_work | yes |
17:01.37 | greg_work | if you had a PRI trunk, you wouldn't need to put any patterns in for the "fixing" part |
17:01.49 | greg_work | it would just dial as-is |
17:02.19 | codebreaker | greg_work: what du you mean with drop? cut off and dial the number after the 1613 |
17:02.49 | greg_work | REALLY, myquestion is how can i do pattern matching from within that macro i posted (http://pastebin.ca/6340) .. i want to replace exten => s,8 with something that can fix the number |
17:03.06 | greg_work | codebreaker: yes |
17:03.28 | codebreaker | greg_work: do ${EXTEN:4} the 4 will cut off the 4 digits and dial the rest |
17:04.49 | codebreaker | greg_work: at home i do exten => _99.,1,Dial(CAPI/9420576:${EXTEN:2},30,r) so i call 99andtherestofthenumber to get calls routet via PSTN and dial everything after the 99 |
17:05.06 | greg_work | codebreaker: yes. i actually have code that lets you enter "1613|NXXXXXX" that creates exten=>_1613NXXXXX,1,Dial(...${EXTEN:4}) |
17:05.16 | JohnnyC | Anyone whats a good board to buy to use with Asterisk, I just need a conection to a S0 line with 10 numbers |
17:05.23 | bjohnson | greg_work: http://www.voip-info.org/wiki-Asterisk+E164+Call+Routing ?? |
17:05.50 | codebreaker | JohnnyC: multiplex or only 2 channels? |
17:06.00 | JohnnyC | codebreaker: whats the diference ? |
17:06.16 | greg_work | bjohnson: hm, maybe useful but i'm not sure thats simplifying things ;) |
17:06.21 | greg_work | oh crap |
17:06.25 | greg_work | maybe i can just use chan_local |
17:06.46 | JohnnyC | its a ISDN line , now I have 3 numbers but I'll have 10 very soon |
17:06.55 | codebreaker | JohnnyC: multiplex is 10 lines can be used simultanous and the other is only 2 users can dial simoultanous |
17:07.37 | gr0mit | johhnyc - best cheap card is an hfc chipset card |
17:07.53 | gr0mit | Made by Billion or Asustek |
17:07.55 | codebreaker | full ack. |
17:08.02 | gr0mit | costs eur 15-20 |
17:08.13 | gr0mit | i have 3 of them in my home * box |
17:08.48 | JohnnyC | well can you advise me in these two cases |
17:09.04 | JohnnyC | because now only 2 users can call simultaneous using that ISDN line |
17:09.12 | gr0mit | yup |
17:09.14 | Derkommissar | why when i allow = ulaw it doesnt send a=rtpmap:0 PCMU/8000 |
17:09.17 | JohnnyC | but with the 10 lines it should be diferent |
17:09.28 | codebreaker | JohnnyC: or buy Acer ISDN 128 Surf PCI (in germany so called) |
17:09.30 | Derkommissar | but when i allow all it sends all and one of them is a=rtpmap:0 PCMU/8000 |
17:09.41 | gr0mit | no. if you get 10 numbers on a BRI you will stil only get 2 voice channels |
17:10.04 | Derkommissar | is the setting allow=PCMU |
17:10.19 | codebreaker | JohnnyC: then a "normal" isdn card is enough |
17:10.25 | *** join/#asterisk devel (~devel@wiggum.digitalcoven.com) |
17:10.48 | JohnnyC | hmm ok and in the multiplex case where I can use the 10 lines simultaneous ? |
17:11.03 | gr0mit | on a single BRI? not possible. |
17:11.21 | codebreaker | JohnnyC: if you order a multiplex anschluss from your pstn provider :) |
17:11.30 | gr0mit | you would need 5 x bri to make 10 simul calls |
17:11.54 | gr0mit | configured in ptp mode |
17:12.02 | gr0mit | as a single group |
17:12.06 | codebreaker | and the right hardware. if you have euroISDN |
17:12.07 | JohnnyC | so I can use the 10 lines simultaneous if I ask the provider ? I dont need another card ? |
17:12.15 | *** join/#asterisk christo (~chris@office.enovi.com) |
17:12.34 | *** join/#asterisk marc_c (~marc32344@69-28-224-214.dsl.teksavvy.com) |
17:12.40 | gr0mit | if u want 10 calls you need 5 x BRI or a single PRI |
17:12.52 | JohnnyC | whats 5 x BRI ? |
17:12.57 | gr0mit | don't confuse 'numbers' with 'channels' |
17:13.06 | JohnnyC | ok |
17:13.16 | gr0mit | 5x bri is 5 seperarate Basic Rate ISDN lines. |
17:13.30 | JohnnyC | hmm ok |
17:13.40 | marc_c | partial T1? |
17:13.44 | codebreaker | JohnnyC: for 10 lines simultaneous you need 1. a multiplexline from your PSTN provider 2. 5 BRIcards o a quadbricard and a singlebricard |
17:14.26 | JohnnyC | hmm so to have 10 lines I need at least 5 cards ( ACER ISDN ......) |
17:14.28 | JohnnyC | ? |
17:14.42 | codebreaker | JohnnyC: not so expensive. i thnk a quadbri now costs about 600¤ |
17:14.55 | gr0mit | but in some countries, e.g. for 8 or more chans you can use a sub-provisioned PRI |
17:14.59 | JohnnyC | so each 2 lines / BRI Card only ? |
17:15.18 | codebreaker | JohnnyC: you have a mainboard with >5 pcislots? |
17:15.23 | gr0mit | one BRI from your telco will give you 2 voice channels |
17:15.40 | JohnnyC | no I dont |
17:15.47 | gr0mit | two bri will give you 4 channels |
17:16.10 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
17:16.14 | gr0mit | in UK, if you want more than 8 voice channels an E1 PRI is cheaper |
17:16.37 | codebreaker | in germany i think too |
17:16.57 | gr0mit | you can get a subequpped E1 |
17:17.32 | gr0mit | which country are you in, JohnnyC ? |
17:17.37 | outtolunc | sounds like something that has bubbles and a bath tub <G> |
17:17.45 | coppice | here its cheaper to get 24 analogue pairs than a T1 :-\ |
17:18.10 | JohnnyC | Portugal ? |
17:18.21 | gr0mit | are you not certain?! |
17:19.53 | gr0mit | cu codebreaker |
17:20.31 | *** part/#asterisk Moc____ (~mochouina@64.235.210.66) |
17:20.32 | Blackvel | who is using DISA, bristuff RC7f and asterisk 1.0.5? for some reason it doesn't work anymore |
17:21.38 | gr0mit | johhnyC what is your application? do you really need 10 simultaneuos calls? |
17:21.46 | *** join/#asterisk Moc____ (~mochouina@64.235.210.66) |
17:21.48 | JohnnyC | no |
17:21.50 | JohnnyC | I dont |
17:21.52 | JohnnyC | :) |
17:21.59 | JohnnyC | maybe I just need 2 |
17:22.05 | JohnnyC | so a BRI Card is enought |
17:22.09 | gr0mit | yup. |
17:22.10 | Moc____ | cluecon / |
17:22.26 | gr0mit | Get a zaphfc card for 15 or 20 euros |
17:22.32 | thieumS | is it possible to use libpri and zaptel CSV with * 1.0.5 ? |
17:22.35 | JohnnyC | but when I ask more BRI to a telco , they give me the same or another cable ? |
17:22.41 | thieumS | CVS sorry |
17:22.54 | gr0mit | they will run another pair into your house. |
17:23.01 | gr0mit | (if you are lucky) |
17:23.16 | gr0mit | if they have no spare copper you are unlucky. |
17:23.16 | JohnnyC | hmm oki |
17:23.29 | *** join/#asterisk adjacent (~scott@office.bftwave.com) |
17:23.42 | gr0mit | in our office in Lisbon Portugal Telecom took AGES |
17:23.53 | JohnnyC | AGES ? |
17:23.57 | gr0mit | 2 months |
17:24.13 | JohnnyC | ages = time |
17:24.22 | gr0mit | yes. ages = a very long time |
17:24.35 | JohnnyC | hehe I tought AGES was some acronim ! |
17:24.39 | gr0mit | hehe! |
17:24.43 | JohnnyC | hehe Im really traumatized |
17:24.54 | JohnnyC | But I was able to put Asterisk working with IP phones |
17:24.58 | JohnnyC | yesterday night |
17:25.07 | gr0mit | so, get 2 zaphfc cards |
17:25.25 | JohnnyC | so this means that if you have 100 lines you have 50 cables ? |
17:25.37 | gr0mit | configure one in NT mode, the other in TE mode, and place your Asteroisk box between them |
17:25.38 | JohnnyC | can they pass signal into the same cable ? |
17:25.39 | gr0mit | yup. |
17:25.42 | gr0mit | no. |
17:25.48 | gr0mit | it is called PRI |
17:26.07 | JohnnyC | PRI ? |
17:26.09 | gr0mit | you get 30 voice channels on an E1 PRI EuroISDN. |
17:26.15 | gr0mit | Primary Rate ISDN |
17:26.22 | JohnnyC | hmm ok a PRI |
17:26.32 | JohnnyC | and then I supose you have a PRI card also |
17:26.37 | gr0mit | yup. |
17:26.49 | JohnnyC | that suports 30 voice channels |
17:26.57 | gr0mit | Digium make a few for about 1000 dollars |
17:27.20 | tzanger | ?? |
17:27.24 | tzanger | TE110P is $499 |
17:27.55 | gr0mit | in EU after import and tax, about 1000 dollars |
17:28.06 | outtolunc | ouch |
17:28.58 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-70-18-115-202.buff.east.verizon.net) |
17:29.08 | thieumS | i just bough TE405P for 1100 euros in France |
17:29.09 | __Sparks_ | in this line I have in sip.conf, "exten => _001800.,1,Dial(SIP/*${EXTEN:2}@fwd-outgoing)" what is the first _ for? |
17:29.09 | gr0mit | GBP359 + VAT. |
17:29.55 | gr0mit | 'about' means give or take 20-50% !!!! |
17:30.05 | *** join/#asterisk zeek (~zeekk@gw.dhivehinet.net.mv) |
17:30.07 | gr0mit | and the dollar is getting weaker by the hour.... |
17:30.21 | SuPrSluG | _Sparks_:pattern match |
17:30.29 | gr0mit | bbl. |
17:30.50 | __Sparks_ | thanks! - is there a document somewhere that explains it all....well!? |
17:30.54 | coppice | actually the dollar strengthened a little recently |
17:31.39 | SuPrSluG | _Sparks_:voip-info.org look up extensions and dial plans |
17:31.42 | zeek | I have a TDM40B with rj45 sockets. Can anyone tell the the pinouts? |
17:32.47 | bjohnson | greg_work: put it on the wiki if you get it to work |
17:32.56 | *** join/#asterisk zno (~zeno@ip-160-79-174-99.autorev.intellispace.net) |
17:34.44 | zeek | I have a TDM40B with rj45 sockets. Can anyone tell the the pinouts? |
17:35.00 | SuPrSluG | i just got dundi working thru openvpn. anyone piping sip thru a vpn? |
17:35.26 | Blackvel | hm |
17:35.38 | zno | is it possible to do use exten => XXX,hint for queues? |
17:36.04 | Blackvel | externip=xxx.dyndns.org still works with asterisk 1.0.5? |
17:36.55 | *** part/#asterisk Moc____ (~mochouina@64.235.210.66) |
17:38.58 | *** join/#asterisk ^HeLL^ (~admin@217.11.115.168) |
17:39.06 | __Sparks_ | Another question! - this line from sip.conf routes all calls to the POTS line when prefixed with a 9, removing the 9. - if I wanted it to prefix the number with somthing, then pause, then dial the number origanally dialed, how do T do it? - Do i use commas for pauses? |
17:39.13 | ^HeLL^ | hello all... |
17:39.16 | __Sparks_ | "exten => _9.,1,Macro(dialout,${TRUNK},${EXTEN:1})" |
17:40.48 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
17:41.33 | Blackvel | is it true you can set with asterisk 1.0.5 only externip = xxx.xxx.xxx.xx instead of xxx.dyndns.org? externhost has just been added to CVS head? what should/can I do now? I need 1.0.5 but with dyndns addy |
17:41.38 | __Sparks_ | Blackvel - it does for me! (Not DynDNS, but another DNS Name in there) |
17:41.42 | zno | can you define an agent without a password? |
17:42.10 | Blackvel | __Sparks_: ah great |
17:42.14 | Blackvel | do you use DISA too? |
17:42.32 | __Sparks_ | DISA? |
17:42.37 | __Sparks_ | (I guess not then!) |
17:42.38 | Blackvel | yeah |
17:42.41 | Blackvel | m |
17:42.42 | Blackvel | hm |
17:42.44 | bjohnson | __Sparks_: that line isn't from sip.conf |
17:43.00 | bjohnson | Blackvel: have you looked at the user authentication page in the wiki? |
17:43.43 | __Sparks_ | bjohnson, You are quite correct. I mean extensions.conf |
17:44.14 | bjohnson | __Sparks_: read the extensions.conf wiki page .. and the info about the dial command .. they will help you understand it (and yes commas are usually pauses) |
17:44.33 | __Sparks_ | bjohnson, Okay, I will sit and read! |
17:44.35 | *** join/#asterisk mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
17:44.49 | Blackvel | bjohnson: not for 1.0.5, I upgraded and DISA does not work anymore. what should I look for? |
17:45.56 | __Sparks_ | is this error bad? - chan_skinny.c:2584 reload_config: Unable to get our IP address, Skinny disabled |
17:46.55 | SuPrSluG | __Sparks_:yes. if your using a cisco phone and using skinny protcol |
17:47.13 | __Sparks_ | okay, then I will ignore it! |
17:48.12 | __Sparks_ | If I have more than one Sip registration in sip.conf, i seem to be getting errors - do I need to define different ports for different SIP profiders to stop this? |
17:49.55 | __Sparks_ | For example - NOTICE[1833]: chan_sip.c:6801 handle_response: Failed to authenticate on REGISTER to '<sip:265532@fwd.pulver.com> |
17:50.04 | zeek | I have a TDM40B with rj45 sockets. Can anyone tell the the pinouts? |
17:50.42 | roamer323 | __Sparks_ I have 5 registrations , no problem |
17:50.47 | Delvar | zeek: what you pluging it into? |
17:51.15 | roamer323 | __Sparks_ check for tyeps |
17:51.23 | roamer323 | typos :-D |
17:51.27 | zeek | Delvar: telephone line |
17:51.33 | Delvar | zeek: if its into the wall socket then its a standard stright cable |
17:51.36 | *** join/#asterisk rontecxt44 (~rontecxt4@dsl9-173.rb.comporium.net) |
17:52.13 | zeek | Delvar: its a rj11 thats going to the socket |
17:52.18 | __Sparks_ | roamer323 - the account info is correct, as if I only have one in there , I dont get errors (And I dont always get the errors on the same accounts! |
17:52.55 | zeek | Delvar: TDM04B bord has RJ45 sockets |
17:53.04 | Delvar | zeek: ah.. ill see if i can find one of my links with the exatct pinouts |
17:53.14 | tzanger | Delvar: no |
17:53.19 | *** join/#asterisk dsmouse (~mouse@rrcs-24-199-146-243.midsouth.biz.rr.com) |
17:53.20 | tzanger | you just plug RJ11 into the RJ45 |
17:53.30 | tzanger | it fits, I think the RJ specification demands that actually |
17:53.39 | Delvar | cool |
17:53.44 | Delvar | never tried that |
17:53.59 | MicH323 | Help Please... getting Feb 23 17:36:07 NOTICE[7062]: chan_sip.c:8404 handle_request: Registration from '<sip:6601@asterisk.itsp.net;user=phone>' failed for 'xxx.xx.xxx.xxx' |
17:54.11 | Delvar | i usualy use an adapter, rj45 one end rj11 the other |
17:54.13 | greg_work | my office is wired only with rj45 and cat5e .. but we still use regular phones on them |
17:54.16 | MicH323 | Trying to setup user on ATA |
17:54.22 | tzanger | MicH323: well there's your problem right there... xxx.xx.xxx.xxx isn't a valid domain name. :-) |
17:54.44 | roamer323 | __Spraks_ : the only registrar I get problem with is iaxtel, but that server is hopelessly overloaded ; others may choke once a while, but never consistent |
17:54.46 | MicH323 | Thats the IP of my ATA |
17:54.58 | *** join/#asterisk Gerrath (Gerrath@shanev.lifecor.com) |
17:55.07 | zeek | Delvar: I just want to find out the 2 pins that is used in the TDM04B board |
17:55.12 | __Sparks_ | roamer323 - I am behind a router - would that possibly be the problem? |
17:55.33 | tzanger | zeek: the middle two |
17:55.34 | tzanger | pair 1 |
17:55.40 | __Sparks_ | roamer323 - I tried putting my asterisk box inot the DMZ zone, but I still get the errors |
17:55.43 | Delvar | its the middle ones |
17:56.24 | roamer323 | __Sparks_ is it consistent with a fixed sip.conf ? i.e. the same provider will give the same error given a fixed sip.conf? |
17:56.41 | zeek | Delvar: you sure? |
17:56.47 | Delvar | zeek: 100% |
17:56.59 | zeek | Delvar: I trust you |
17:57.01 | Delvar | i just dont know the exact pinouts |
17:57.16 | Delvar | just remember to expect a loud band when you plug it in |
17:57.21 | Delvar | bang* |
17:57.51 | __Sparks_ | roamer323 - The error is always the same, but it isn't always the same SIP account that throws the errors |
17:58.09 | __Sparks_ | roamer323 - I am also seeing this - WARNING[1833]: chan_sip.c:6786 handle_response: Got 200 OK on REGISTER that isn't a register |
17:58.42 | *** part/#asterisk Cresl1n (~matt@216.207.245.23) |
17:59.29 | rontecxt44 | hi...has anyone dealt with error "Avalible sdfsdf |
17:59.29 | rontecxt44 | The Spirit of Tulsa |
17:59.29 | rontecxt44 | Learn about Avenue One |
17:59.32 | rontecxt44 | Leasing News |
17:59.36 | rontecxt44 | Request Info |
17:59.48 | rontecxt44 | hi...has anyone dealt with kernel error "TDM PCI Master abort" |
17:59.50 | rontecxt44 | sorry |
17:59.52 | rontecxt44 | bad past |
17:59.54 | *** join/#asterisk okieplaya (~okieplaya@ip68-229-252-53.ok.ok.cox.net) |
17:59.54 | rontecxt44 | :( |
18:00.01 | rontecxt44 | paste |
18:00.10 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
18:00.29 | roamer323 | __Sparks_ that's a strange one... you're not running any other SIP thing on the same box, are you (i.e. a softphone) ? |
18:00.46 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
18:01.38 | __Sparks_ | roamer323 - nope, I have 1 asterisk box, just doing asterisk, and two hard SIP Phonbes on the LAN connected to it |
18:01.41 | okieplaya | hey someone have some time to help me setup my new S100I im sorry i just dont under stand the pdf info ? |
18:01.57 | __Sparks_ | the router is connected to this same LAN |
18:02.03 | adjacent | anyone used asterisk@home ? |
18:02.06 | rontecxt44 | let me try again without the annoying extra crap |
18:02.22 | rontecxt44 | anyone run into "TDM PCI Master abort" ? |
18:02.30 | adjacent | more precisely, anyone willing to give it a thumbs up or thumbs down =) |
18:02.33 | rontecxt44 | then the card shuts down |
18:02.48 | roamer323 | __Sparks : and you have taken at least 1 incoming call from each of your registration in the past? |
18:03.41 | roamer323 | adjacent - one thumb down |
18:03.46 | *** join/#asterisk didz_ (didz_@200.218.192.52) |
18:03.46 | SuPrSluG | __Sparks_:you need to port forward |
18:04.10 | SuPrSluG | __Sparks_:better off w/ iax |
18:04.44 | adjacent | roamer323: reason? |
18:05.05 | __Sparks_ | SuPrSluG - I have the followinf forwarded UDP ports - 10000->20000 4569 5060->5070 |
18:05.37 | __Sparks_ | SuPrSluG - I tried putting the asterisk box in the DMZ zone, and still got the same errors |
18:05.45 | roamer323 | adjacent - you get 10+ users coming on here with the same questions on the same bugs every single day - enough reasons |
18:06.04 | __Sparks_ | I have just turned off all the accounts except one that wes erroring, and now the errors have stopped |
18:06.09 | adjacent | i hear ya. |
18:06.27 | SuPrSluG | __Sparks_: i think fwd allows iax. use that to register. examples on wiki |
18:06.29 | adjacent | bugs is a good reason for a thumbs down |
18:06.55 | outtolunc | .. |
18:07.00 | okieplaya | is there somewhere i can read more on how to setup the digium s100i other than the pdf they have ? |
18:07.08 | adjacent | but probably 9/10 users coming here with problems are too dumb to use the product, or read the manual |
18:07.14 | okieplaya | or there i tech number i can call |
18:07.15 | __Sparks_ | SuPrSluG, it isnt only FWD I use - i have Sipgate x3 - FWD and voiptalk |
18:07.36 | adjacent | its downloading now. ill give it a look anyway |
18:08.45 | roamer323 | adjacent - good luck :-) |
18:09.02 | adjacent | heh. thanks =) |
18:09.11 | *** join/#asterisk zpn (~xpn@dhcp-152.digium.com) |
18:09.26 | *** part/#asterisk numBone (~numBone@c-24-129-204-233.se.client2.attbi.com) |
18:09.29 | adjacent | imho, its going to be better than VOCAL any way you look at it |
18:09.59 | roamer323 | __Sparks_ you're best off turning on "sip debug" and do a message by message trace to see what's going on |
18:11.57 | zpn | i'm having problems dialing out on my sip. i get a 'Everyone is busy/congested at this time' with a warning that 'No channel type is registerd for 'Zap', then "Unable to create channel of type Zap' any ideas on how to fix this? |
18:12.07 | SuPrSluG | __Sparks_: which one works? |
18:13.34 | __Sparks_ | <SuPrSluG> - whick what works? - SIP account? |
18:14.20 | SuPrSluG | __Sparks_: voiptalk has asterisk setup |
18:15.21 | Blackvel | oh my dear |
18:15.30 | Blackvel | now I even understand my DISA 1.0.5 problem |
18:15.42 | __Sparks_ | <SuPrSluG> - all the accounts seem to work ok, just the errors are shown - and when it is erroring there seems to be big delays in the system when dialing out (including if I am using a POTS line |
18:16.08 | Blackvel | my analog pbx is dailing too fast, DISA does not pick up fast enough so the rest of the extensions don't get used for extensions dialplan |
18:16.27 | Blackvel | that can be everyething |
18:16.31 | Blackvel | new DISA code in 1.0.5 |
18:16.35 | Blackvel | new zaphfc code |
18:16.42 | Blackvel | too bad |
18:17.07 | *** part/#asterisk zpn (~xpn@dhcp-152.digium.com) |
18:17.14 | SuPrSluG | zpn:check lsmod ? is it loaded? |
18:17.16 | Blackvel | how can I find that out? |
18:17.31 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
18:21.41 | *** part/#asterisk rontecxt44 (~rontecxt4@dsl9-173.rb.comporium.net) |
18:22.44 | *** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
18:24.11 | *** join/#asterisk trym (~trym@linux.debian.us) |
18:24.46 | Slainte | How do I get * to have a cold beer ready for me by the time I get home from work? |
18:25.22 | zno | Try exten => _XXX,s,Beer(${num_cans{) |
18:25.31 | Slainte | its about time someone asks a question with some importance |
18:25.43 | Pinhole | Slainte, I'm pretty sure you'll need some AGI code for that. |
18:26.09 | *** join/#asterisk PTG123 (~PTG123@ip68-106-17-54.ph.ph.cox.net) |
18:26.13 | Slainte | Pinhole, I think so. I wonder what variable I use for the temp of the beer. |
18:27.09 | Pinhole | You should be able to have it order over the net when supplies are low. Some low level robotics could also help. I think that's in the home automation stuff. |
18:27.35 | *** part/#asterisk PTG123 (~PTG123@ip68-106-17-54.ph.ph.cox.net) |
18:27.49 | Slainte | While thirsty=yes;do pour_beer |
18:28.18 | Slainte | simple loopk, yet so powerfull :) |
18:28.37 | Pinhole | I thought it would be useful to have sphinx2 and festival create a gateway to this IRC channel, but sphinx2 does not perform well enough. |
18:29.22 | Beirdo | Pinhole: then we could all curse you out for the fun of it |
18:29.45 | Beirdo | curse you in German and see how well festival handles it, etc :) |
18:30.22 | Pinhole | sphinx: THE CAN PHONE FIT |
18:30.34 | adjacent | while (thristy) { if (glass_empty) pour_beer(); else drink_beer(); } |
18:30.34 | adjacent | otherwise you would have a big mess to clean up |
18:30.57 | Slainte | yipppy I just won the porn lottery!!, What a day of good news. First I qualify for a home loan, then I win the porn loterry, and then I get an offer to help some african government funnel out money |
18:30.59 | SuPrSluG | grandstream has a new firmware update. |
18:31.03 | CoaxD | whee |
18:31.14 | *** join/#asterisk [cc]smart (~smart@62.65.149.158) |
18:31.21 | Slainte | who says the internet is not full of good stuff |
18:31.56 | Pinhole | while(thirsy) { if(glass_empty && !drinking_beer) pour_beer(); if(!beer_pouring && ! drinking_beer) drink_beer() } |
18:32.05 | tzanger | Slainte: heh |
18:32.17 | __Sparks_ | anyone here know about Xorcom Rapid? |
18:32.26 | Slainte | __Sparks yes |
18:32.29 | CoaxD | pinhole: that would be a scarey robot. *lol* |
18:32.36 | Slainte | Pinhole, nice code |
18:32.49 | adjacent | now write a peripheral driver to control a device that interfaces with a 12 pack, and use it in pour_beer. then set an extention in * to start it up ;) |
18:32.55 | tzanger | doesn't work worth a shit since ther's no flags to control the pouring |
18:33.00 | Beirdo | change it to "while(!passed_out)" |
18:33.04 | *** join/#asterisk Fanguin (~Fanguin@p508187ED.dip0.t-ipconnect.de) |
18:33.19 | *** join/#asterisk trym (trym@linux.debian.us) |
18:33.25 | Pinhole | which is more complicated, call routing or pouring beer? |
18:33.29 | Beirdo | tzanger: that's all in the pour_beer routine |
18:33.35 | Slainte | Sparks what is your question about Rapid |
18:33.35 | adjacent | tzanger: use mcdonalds "idiot-proof" style pouring |
18:33.38 | CoaxD | pinhole: Hmmm. Pouring beer would be far more complex |
18:33.43 | tzanger | Beirdo: that only gets executed if the glass is empty |
18:33.51 | CoaxD | pinhole: (to a robot who thinks in digital terms, doing an analog action.) |
18:33.53 | tzanger | once you start pouring the glass is no longer empty even though there's only a little beer in it |
18:34.13 | adjacent | tzanger: output 12 oz and receck. |
18:34.17 | Beirdo | but the routine doesn't return immediately :) |
18:34.17 | __Sparks_ | Slainte - I keep trying to do the Maintanance - Update Software Inventory, but it keeps failing |
18:34.17 | CoaxD | tzanger: Of course, you'd have to write a glass_is_half_empty_or_glass_is_half_full() routine |
18:34.18 | Beirdo | hehe |
18:34.32 | tzanger | CoaxD: yeah but you'd get bogged down in that routine with semantics |
18:34.40 | Slainte | Sparks throw me a bone, what does "it" do |
18:34.42 | CoaxD | tzanger: Sadly enough, you're correct :/ |
18:34.49 | __Sparks_ | Slainte - I just tired pinging updates.xorcom.com, then doing the updates worked!? |
18:34.54 | adjacent | #include <philosohy.h> |
18:35.09 | km- | coaxd: sup. |
18:35.11 | Pinhole | while(true) { offer_beer_bottle(); while(!beer_taken); } |
18:35.12 | CoaxD | adjacent: #error i cant parse this data so i'll stop compiling |
18:35.16 | __Sparks_ | Slainte - usually sticks at - 40% [Connecting to updates.xorcom.com (1.0.0.0)] |
18:35.21 | CoaxD | km: Nada, pete. what you up to? :) |
18:35.26 | *** join/#asterisk ScarletCrusader (~GMMiller@wsip-66-210-74-254.mc.at.cox.net) |
18:35.35 | km- | coaxd: we just got a te405p in the office, I'm 20 mins from my changeover :) |
18:35.45 | km- | asterisk++ |
18:35.45 | CoaxD | km: OOOH!!! |
18:35.52 | CoaxD | ^5 |
18:36.05 | km- | Got a T1 from the CO running to asterisk, then another T1 running to the NEC system |
18:36.10 | CoaxD | km; We're running * inhouse here on a single POTS line, and also, i have a voip install in the same building. I like *. :) |
18:36.13 | km- | gonna transition off the NEC gradually |
18:36.16 | Slainte | Sparks, check your /var/log/messages |
18:36.22 | Slainte | What are you trying to update? |
18:36.23 | km- | yeah, I've got some work ahead of me here :) |
18:36.57 | adjacent | http://www.telegraph.co.uk/news/main.jhtml?xml=/news/2005/02/18/wpill18.xml&sSheet=/news/2005/02/18/ixworld.html |
18:37.01 | km- | luckily I've already got experience running asterisk in full voip and POTS/voip configs so this is just adding T1 knowledge to the pile! |
18:37.30 | *** join/#asterisk mud (~mud@bestekdsl.customer.sentex.ca) |
18:37.37 | km- | slainte: Amen. That's like, what you do when you're a hobbyist. Can't do that in a business setting! |
18:37.38 | __Sparks_ | Slainte - I am just checking for avalable updates |
18:37.52 | km- | if you're a hobbyist, who cares if it breaks, it's a learning experience |
18:37.53 | __Sparks_ | Slainte - what am I looking for in /var/log/messages? |
18:38.01 | Slainte | Sparks, problems :) |
18:38.23 | Slainte | Sparks, do you have much *nix experience? |
18:38.32 | __Sparks_ | Slainte - the last load of messages are all Feb 23 18:24:05 localhost -- MARK -- |
18:38.44 | Slainte | ok try the /var/log/syslog |
18:38.44 | __Sparks_ | Slainte - no, cant you geuess :- |
18:38.57 | Slainte | Sparks, heheh no biggy, a few things you need to know about unix. |
18:39.09 | Slainte | 1. You rarely need to reboot it |
18:39.24 | Slainte | 2. dont listen to km- |
18:39.26 | Slainte | :) |
18:39.31 | km- | HAHAHAHA |
18:39.32 | __Sparks_ | lol |
18:39.34 | tzanger | km-: I do that too except it's |
18:39.37 | tzanger | *.* -/var/log/all |
18:39.41 | km- | tzanger: it fills up a drive quickly :) |
18:39.42 | Slainte | 3. Logging will set you free. |
18:39.45 | tzanger | km-: nonsense |
18:39.53 | tzanger | only if you've got something spewing so badly |
18:40.00 | __Sparks_ | in syslog, the last three messages are |
18:40.00 | __Sparks_ | Feb 23 17:44:05 localhost -- MARK -- |
18:40.00 | __Sparks_ | Feb 23 18:04:05 localhost -- MARK -- |
18:40.00 | __Sparks_ | Feb 23 18:17:01 localhost /USR/SBIN/CRON[1910]: (root) CMD ( run-parts --report /etc/cron.hourly) |
18:40.09 | km- | tzanger: run snmpd with a monitoring system like Nagios, and you'll see! :) |
18:40.10 | __Sparks_ | there is a lot like that |
18:40.11 | Slainte | DONT PASTE to #* |
18:40.12 | adjacent | 3a: Remote logging will keep you free =) |
18:40.21 | __Sparks_ | sorry! |
18:40.29 | Slainte | pastebin.ca is your friend |
18:40.31 | Slainte | :) |
18:40.39 | km- | __Sparks: don't worry about Slainte, he's just trying to sound intelligent and in control :P |
18:40.41 | trym | or pastebot.org |
18:40.42 | trym | ;) |
18:41.01 | Slainte | hahaha km- |
18:41.16 | Slainte | I get very few chances a week to do this, so dont piss on my parade please :) |
18:41.24 | km- | hahahaha |
18:41.25 | km- | word :) |
18:41.36 | goatmilk | if this is your parade you should get out more often |
18:42.07 | km- | hmm |
18:42.21 | __Sparks_ | anyway, the apt-get update seems to work after i did a ping to updates.xorcom.com |
18:42.35 | Slainte | thats wierd |
18:42.45 | Slainte | nameservers are in /etc/resolv.conf |
18:42.56 | Slainte | check to see if anything is listed |
18:43.00 | __Sparks_ | ...he says, as it is now stuck on "40% [Connecting to updates.xorcom.com (1.0.0.0)]" |
18:43.16 | __Sparks_ | is the 1.0.0.0 the IP it is trying to get to!?! |
18:43.45 | __Sparks_ | DNS is fine, i can ping anything i have tried |
18:44.28 | *** join/#asterisk lyroy (~lyroy@picachou.csaffluents.qc.ca) |
18:44.35 | Slainte | 1.0.0.0 is not the IP no. |
18:44.47 | Slainte | do a netstat -a | grep xorcom and see where it is connected |
18:44.57 | __Sparks_ | in /etc/resolv.conf i have the IP of my router and 212.135.1.36 |
18:45.17 | lyroy | Does someone can tell me how can i change the tftp server of my phone in telnet... |
18:45.42 | __Sparks_ | I did a "netstat -a | grep xorcom" but it just returned a prompt |
18:45.56 | Slainte | netstat -a | more |
18:46.06 | km- | to pastebin please |
18:46.09 | km- | dont want to see that netstat |
18:46.10 | km- | :P |
18:46.20 | Slainte | hehehe very true |
18:46.39 | goatmilk | how about we just take all of our chit chat to pastebin |
18:47.15 | Slainte | I am still waiting for you spin tops to sort out my beer code for * |
18:47.26 | __Sparks_ | http://pastebin.ca/6342 |
18:47.40 | marc_c | is redund abs nec? |
18:48.11 | Beirdo | WTF? |
18:48.39 | Slainte | Sparks. its not running anymore. Does it freeze at 40% or did you close it? |
18:48.52 | __Sparks_ | I colosed it :) |
18:48.58 | __Sparks_ | Closed it |
18:49.16 | *** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
18:49.19 | Slainte | put your game hat on. |
18:49.27 | __Sparks_ | I will do it again, sorry!! |
18:50.14 | Slainte | An Irishman and Englashman and a Welshman walk into a pub. The bartender looks up at the three of them, and says.. is this a joke? |
18:50.59 | __Sparks_ | done, and updated pastebin |
18:51.03 | Slainte | I will be here all week. Please dont put your smokes out on the floor, and please tip your waitresses |
18:51.59 | Slainte | Sparks same pastebin number? |
18:52.15 | __Sparks_ | yep |
18:52.21 | Slainte | I dont see any whanges |
18:52.40 | __Sparks_ | sorry - http://pastebin.ca/6343 I thought it just updated it |
18:52.53 | Slainte | I am getting frustrated with you Sparks |
18:53.08 | __Sparks_ | sorry, I'm new to all this :-S |
18:53.34 | Slainte | I allready told you to put your gamehat on. |
18:53.39 | Slainte | Line 7 is a problem. |
18:53.46 | Slainte | pastebin your /etc/hosts file |
18:54.40 | __Sparks_ | 127.0.0.1 localhost.localdomain localhost tsbserver02 |
18:54.40 | __Sparks_ | # The following lines are desirable for IPv6 capable hosts |
18:54.40 | __Sparks_ | ::1 ip6-localhost ip6-loopback |
18:54.40 | __Sparks_ | fe00::0 ip6-localnet |
18:54.40 | __Sparks_ | ff00::0 ip6-mcastprefix |
18:54.41 | __Sparks_ | ff02::1 ip6-allnodes |
18:54.43 | __Sparks_ | ff02::2 ip6-allrouters |
18:54.45 | __Sparks_ | ff02::3 ip6-allhosts |
18:54.47 | __Sparks_ | shit, sorry! |
18:55.17 | km- | 5 mins to changeover! |
18:55.18 | km- | hehe |
18:55.55 | __Sparks_ | http://pastebin.ca/6344 - not that it matters now |
18:56.43 | *** part/#asterisk Fanguin (~Fanguin@p508187ED.dip0.t-ipconnect.de) |
18:56.54 | __Sparks_ | ...someone at the door now, BRB |
18:57.55 | Slainte | dont let it hit you in the ass on the way out |
19:00.36 | km- | thats weird |
19:00.40 | km- | where's twisted |
19:00.40 | km- | hehe |
19:02.46 | km- | how do I store a digit? |
19:03.08 | km- | I want to save ${EXTEN} as a variable |
19:03.10 | km- | how do I do that? |
19:03.37 | km- | I'll use dbget/dbput |
19:05.18 | km- | shit |
19:05.35 | bjohnson | setvar |
19:06.00 | RoyK | show application setvar |
19:06.20 | km- | theres something wrong with this fucker |
19:06.20 | km- | dammit |
19:06.42 | RoyK | copulator? |
19:07.01 | km- | there goes my window again |
19:07.09 | km- | its this darned NEC system, it just doesnt play right |
19:07.19 | RoyK | running windows? |
19:07.22 | km- | no |
19:07.46 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
19:08.07 | shido6 | anyone have a serial ata poweredge box and needed to install redhat 9? |
19:10.04 | trelane | nope |
19:10.22 | RoyK | shido6: use gentoo, or debian, or slackware, or even SuSE, but forget redhat 9 |
19:11.30 | km- | ~seen twisted[work] |
19:11.31 | jbot | twisted[work] is currently on #asterisk. Has said a total of 49 messages. Is idling for 3h 5m 14s |
19:12.07 | km- | Asterisk just doesnt want me to do this |
19:12.08 | km- | hehe |
19:12.34 | Delvar | nn all |
19:13.27 | shido6 | RoyK thanks |
19:16.04 | *** join/#asterisk XeNoSiS (user@216.234.145.18) |
19:16.11 | XeNoSiS | Hello Hello |
19:16.21 | XeNoSiS | Just the people I was looking for. |
19:16.58 | RoyK | shido6: gentoo will be my first choice |
19:16.59 | RoyK | btw |
19:17.02 | XeNoSiS | I am trying to setup Asterisk with a Cisco 2600 as the PSTN to the gateway. Inbound / Outbound calls appear to setup properly but there is no audio either way. |
19:17.32 | XeNoSiS | Cisco2600 gateway to the PSTN that is. Sorry. I am tired. |
19:19.09 | XeNoSiS | Softphone (Xten Xlite) <---> Asterisk <---> Cisco 2600 <---> Nortel PBX <----> Phone |
19:19.38 | XeNoSiS | any sample configs for Cisco 2600 Gateway configuration? or ideas on why i get no audio? |
19:20.01 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
19:20.49 | *** join/#asterisk stepcut (~user@207.67.194.2) |
19:21.18 | *** join/#asterisk sonic_br (~asimoes@ip-64-32-179-115.dsl.nyc.megapath.net) |
19:21.24 | yasha | I have seen some references of integrating * with Microsoft CRM. Does anyone have any info on that or can point me in the right direction? Thank you. |
19:26.21 | XeNoSiS | I don't think anyone is really here. :) |
19:26.34 | km- | eh, there are people here |
19:26.35 | km- | hehe |
19:27.09 | XeNoSiS | So no one has done anything with using cisco for the PSTN gateway off an asterisk system? |
19:27.17 | XeNoSiS | Or just don't feel like helping right now? |
19:28.48 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
19:28.48 | *** mode/#asterisk [+o bkw_] by ChanServ |
19:29.05 | nestAr | heh |
19:29.17 | *** join/#asterisk Graphikos (~Graphikos@71-32-6-49.spkn.qwest.net) |
19:29.28 | RoyK | yasha: there's something like asterisk tapi support |
19:29.33 | RoyK | yasha: that'll be all I know |
19:29.38 | bjohnson | somehow I've got colours showing now on my cli and it's leaving codes in my message log .. making it harder to read. How can I correct what goes into the logs? |
19:29.53 | nestAr | colors are fun! |
19:29.59 | Graphikos | anyone brave enough to take a complete asterisk newbie under thier wing a bit? |
19:30.06 | *** join/#asterisk ACiDV (~joel@69.156.197.246) |
19:30.06 | RoyK | nestAr: colours? |
19:30.12 | Graphikos | I could use a bit of direction... |
19:30.20 | nestAr | RoyK: depending on your locale.. yes. |
19:30.21 | nestAr | :) |
19:30.30 | RoyK | nestAr: what about them? CoVoIP? |
19:30.32 | RoyK | :P |
19:30.38 | nestAr | lol |
19:30.57 | nestAr | i sometime use colourful language oIP |
19:31.00 | MicH323 | Anyone explain what this means: Rejected connect attempt from 65.39.205.121, requested/capability 0x4/0x4 incompatible with our capability 0xff03 |
19:31.17 | bjohnson | looks like ansi colours on the cli |
19:31.18 | RoyK | MicH323: codec problems |
19:31.22 | ACiDV | =) Does it's possible to get the "original phone context" ? I have sip/zap phones that have context=default, others have context=full, etc... in a macro, does it's possible to get the context of the caller ? |
19:31.29 | MicH323 | Ah, Thans Roy |
19:31.33 | bjohnson | they save codes into the messages log file |
19:32.15 | MicH323 | Graphikos: I am a total newbie myself... Just up today! :) |
19:32.31 | *** join/#asterisk km- (~km-@67.105.178.130) |
19:32.50 | nestAr | anyone know how to supress the channel reset messages from the CLI? |
19:33.20 | XeNoSiS | Does the gateway have to be SIP? or can i setup an h323 cisco gateway? |
19:33.55 | ACiDV | To create a macro like: [macro-lastredial] exten=> s,1,DBGet(last=...) s,2,Dial(Local/${last}@${ORIGINALCONTEXT} ? |
19:34.03 | MicH323 | XeNoSiS: You can setup Cisco gateway to do SIP, but it cant register |
19:34.17 | ACiDV | hmmm not sure if I self answer... I think I've see a variable for this... go to wiki :| |
19:34.40 | XeNoSiS | We setup the gateway as SIP and calls work in / out but there is no audio. |
19:34.49 | MicH323 | XeNoSiS: You have to compile the H323 into Asterisk |
19:35.25 | MicH323 | Are you behind NAT? |
19:35.32 | XeNoSiS | no |
19:35.44 | MicH323 | What GW do you have? |
19:36.17 | *** join/#asterisk tufone_ny (~asimoes@ip-64-32-179-115.dsl.nyc.megapath.net) |
19:36.28 | XeNoSiS | Cisco 2600 with an NM-HDV-T1 PRI connected to a Nortel PBX. |
19:36.47 | MicH323 | So you are using sip-ua? |
19:37.09 | MicH323 | And dial-peer xx voip? |
19:37.41 | tclark | ok who knows how to interface these FRS GMRS walkie talkie to * ? what h/w to wee nned to make chan_gmrs ? |
19:37.48 | *** join/#asterisk harryvv (~comming@S010600055d210201.vs.shawcable.net) |
19:38.23 | harryvv | anyone here running a spa1k? need to ask a quick config question. |
19:38.49 | XeNoSiS | yeah I am using sip-ua and dial-peer voip |
19:39.04 | XeNoSiS | can you send me a /msg with your email and I will send the config |
19:40.15 | *** join/#asterisk dstevens_ (~dstevens@cpc3-ches1-4-0-cust87.lutn.cable.ntl.com) |
19:41.00 | Pinhole | Is there an automated way of testing the call quality through asterisk? |
19:41.12 | nix000 | anyone tried running asterisk with linksys ata ? |
19:42.21 | RoyK | Pinhole: no |
19:42.30 | RoyK | Pinhole: ethereal can do some, though |
19:42.55 | harryvv | Are there free bandwith measuring tools on the market? |
19:43.11 | bjohnson | for? |
19:43.26 | bjohnson | dsl speed? |
19:43.44 | harryvv | Say I want to see what a customers bandwith on there network before sugesting the idea of voip |
19:44.03 | harryvv | and measure the use dsl speed to. |
19:44.13 | *** join/#asterisk cp5 (~samy@dsl081-232-019.lax1.dsl.speakeasy.net) |
19:44.15 | cp5 | hola |
19:44.15 | bjohnson | I get mrtg graphs on my ipcop box |
19:44.23 | cp5 | anyone know how to make hints work for outgoing calls? |
19:44.27 | t3t | harryvv: It's more about latency and jitter than bandwidth... you need to measure the connection over time |
19:44.35 | bjohnson | mrtg just makes the graphs though .. I think they get the data from iptables |
19:44.44 | harryvv | t3t i know over time. |
19:45.05 | t3t | harryvv: like over a few weeks, not just a b/w test once or twice |
19:45.27 | Pinhole | What I'm really trying to do is detect * crashes. Some crashes leave * working, but badly distorting calls. |
19:45.50 | ionix | use ser *cough* for sip server |
19:45.50 | bjohnson | nix000: yes |
19:46.01 | harryvv | well, If a voip consultant walks into the biz he wants to keep the delay of measuing and providing a package service to a min or the company president might change his mind. |
19:46.02 | t3t | harryvv: With dsl/cable type services, unless there is an SLA in place for prioritized voice traffic, the connection probably won't always be voice-clean |
19:46.28 | harryvv | SLA? |
19:46.39 | t3t | Service Level Agreement |
19:46.46 | Pinhole | It's *not* a bandwidth issue. It did this on the local network while we were testing. |
19:46.51 | harryvv | ohh i see yea, set aside some bandwith for voice only traffic. |
19:47.11 | t3t | part of the contract that specifies how they will handle the connection specifics and what they pay you when they don't |
19:47.41 | yasha | RoyK: Thank You....... I was hoping that someone would have already looked at it... |
19:48.10 | t3t | harryvv: tcptraceroute is a good tool to do external latency monitoring |
19:48.15 | harryvv | what POe router would prioritize network traffic based on what port number? |
19:48.24 | bjohnson | any linux one |
19:48.29 | t3t | POe? |
19:48.32 | loud | i cant believe broadvoice only supports g711. |
19:48.36 | nix000 | bjohnson, was it working out of the box ? |
19:48.36 | bjohnson | oh .. power over ethernet? |
19:48.37 | PatrickDK | harryvv, any real l3 router |
19:48.43 | harryvv | level3 |
19:48.44 | *** join/#asterisk greendisease (~jack@greendisease.fedora) |
19:49.26 | t3t | Level3 is a communications company, L3 is ISO network model layer 3 |
19:49.35 | bjohnson | nix000: 1. I don't have one 2. you will of course have to configure it 3. only the PAPs marked -NA will not be locked and can therefore be used |
19:49.36 | harryvv | yea I know |
19:50.28 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
19:50.28 | *** mode/#asterisk [+o bkw_] by ChanServ |
19:50.29 | dstevens_ | Hi all, has anyone tryed compiling asterisk on debian/ubuntu as its failing on ssl, even though " dpkg --get-selections | grep openssl " shows openssl installed. |
19:50.29 | t3t | harryvv: what do you mean by a 'POe router' |
19:50.37 | t3t | what's up, bkw |
19:50.48 | bkw_ | not much |
19:50.59 | t3t | just wake up ;) |
19:51.09 | DJ-Pyro | dstevens_: openssl-dev |
19:51.11 | bjohnson | dstevens_: a third of the people here run debian (I do not) |
19:51.35 | dstevens_ | DJ-Pyro, Will check that out thanks back in a bit. |
19:51.38 | cp5 | anyone use hints in asterisk? any docs on using them for outgoing calls? |
19:51.50 | DJ-Pyro | err, nevermind |
19:51.52 | marc_c | any cnd here? |
19:51.57 | stevekstevek | hmm, outgoing calls.. |
19:51.59 | DJ-Pyro | you need the development libraries, I foret what they are |
19:52.00 | stevekstevek | can it do that? |
19:52.13 | DJ-Pyro | dstevens_: libssl-dev |
19:52.14 | DJ-Pyro | sorry |
19:52.39 | t3t | cp5: are you referring to registration hints? |
19:52.51 | *** part/#asterisk ACiDV (~joel@69.156.197.246) |
19:53.04 | dstevens_ | bjohnson, Hi ubuntu is debian based, i use deian but prefer a humane interface |
19:54.04 | shido6 | dstevens_ |
19:54.07 | shido6 | how IS ubuntu |
19:54.10 | shido6 | Ive seen it |
19:54.13 | shido6 | never used it tho |
19:54.56 | dstevens_ | Very nice, gnome only based only at the moment. |
19:56.18 | dstevens_ | Although Kubuntu is on its way and looking good if your a kde' a |
19:56.34 | cp5 | t3t, well, call hints. so when someone calls an extension, a special phone configured (such as a secretary's phone) can see that line is getting a call |
19:58.16 | *** join/#asterisk rephorm (~rephorm@ip67-95-13-62.z13-95-67.customer.algx.net) |
19:59.24 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
20:01.02 | rephorm | hello, i'm setting up call recording (via monitor) in asterisk, and was wondering if it were possible to initiate it while in a conversation from the phone keypad (without transfering or switching to a conference) |
20:02.04 | *** join/#asterisk bobx (~bobx@lowfreq.trancemitter.org) |
20:02.19 | `Sauron | ls |
20:02.41 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02v-22-254.d4.club-internet.fr) |
20:03.27 | rephorm | i.e. can i have an agi script sit an wait for dtmf during a call? |
20:03.39 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
20:09.00 | bjohnson | how is ubuntu different from the hundred other distros? |
20:09.24 | tzanger | bjohnson: it's got a cool name? <rolls eyes> |
20:09.43 | Sedorox | bjohnson: its a love CD that actually runs Gnome instead of KDE |
20:09.49 | Sedorox | otherwise its like knoppix |
20:09.54 | tzanger | a love CD? |
20:09.55 | tzanger | oh live |
20:09.56 | tzanger | haha |
20:09.57 | bjohnson | hehe |
20:10.09 | tzanger | I'm like "whoa they're REALLY taking this tribal theme too far" |
20:10.18 | Sedorox | yea live.. sorry |
20:10.21 | bjohnson | let's vote them off |
20:10.29 | Sedorox | if you really wanna use a cd for love... ummm... thats your thing |
20:10.57 | *** join/#asterisk pdracevich (~paul@smtp.aucklandtax.co.nz) |
20:11.18 | aminorex | there was a pakistan-made IAX phone in the works. what's the link? |
20:11.42 | *** join/#asterisk lorion (~van@63.115.106.66) |
20:12.07 | gr0mit | www.farfon.com |
20:12.16 | lorion | Are there any good docs regarding NAT? |
20:12.21 | bjohnson | yes |
20:12.41 | Logan | Has anyone here had any problems with Sipura's phones? |
20:12.42 | bjohnson | but basically it's a system to reuse IP addresses |
20:12.42 | aminorex | anyone seen the faron product in the wild yet? |
20:12.54 | gr0mit | logan, yes |
20:13.03 | bjohnson | aminorex: zeeek is waiting for one |
20:13.28 | *** join/#asterisk dsmouse (~mouse@rrcs-24-199-146-243.midsouth.biz.rr.com) |
20:13.29 | gr0mit | sipura 841 has strange behaviour with tx audio |
20:13.32 | dsmouse | 'lo y'all |
20:13.49 | harryvv | gr0mit im not impressed with the looks of the sipura 841 |
20:14.03 | gr0mit | sticky buttons |
20:14.13 | gr0mit | rx audio is very good |
20:14.34 | gr0mit | but tx seems to have strange muting if the audio is too quiet. |
20:15.00 | lorion | How would you configure a phone that was nat'd.. For example the Asterisk server is on one network and the phone is nat'd on another. |
20:15.18 | lorion | It is a remote sales agent config. |
20:15.28 | gr0mit | depends on the phone, lorion! |
20:15.31 | modulus_ | anyone from broadvoice here? |
20:15.50 | lorion | I am testing with X-lite |
20:15.59 | modulus_ | broadvoice is down again |
20:16.09 | gr0mit | x-lite just worked for me |
20:16.11 | modulus_ | this is like number 3 |
20:16.23 | modulus_ | when receiving calls, they hear nothing |
20:16.24 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
20:16.24 | gr0mit | seems to have its own stun server |
20:16.31 | pdracevich | aaaaaaaaarrrrrrrrrgggggggg!!!!!!!!1 can some one have a look and give me some advice! http://pastebin.ca/6345 |
20:16.46 | lorion | are there ports I need to open? |
20:16.53 | bjohnson | lorion: add nat=yes to sip.conf |
20:16.54 | gr0mit | have you set nat=yes in your sip.conf file? |
20:17.15 | bjohnson | lorion: is the * server behind nat? |
20:17.15 | tzanger | pdracevich: interesting |
20:17.19 | tzanger | how are you getting that |
20:17.20 | lorion | my xlite connects, I have nat=yes, but if I try to dial an extension it give me a 404 |
20:17.34 | lorion | my x-lite can pick up vm |
20:17.56 | pdracevich | tzanger: I had the two sip boxes talking and rules donw so the calls are being placed went home and now this |
20:18.04 | *** join/#asterisk [Outcast] (~knoppix@h0006259a2649.ne.client2.attbi.com) |
20:18.12 | pdracevich | tzanger: iax2 debug |
20:19.19 | *** join/#asterisk visik7 (~ciao@visik7.user) |
20:21.13 | *** part/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
20:22.48 | ScarletCrusader | is there a newbi cnannel I can ask easy question that i cannot figure out? |
20:22.57 | pdracevich | tzanger: any idea? |
20:23.38 | nestAr | anyone know how i can supress these " B-channel 0/2 successfully restarted on span 2" messages? |
20:23.44 | modulus_ | OMG broadvoice sucks |
20:23.58 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-199-86.dsl.scarlet.be) |
20:24.31 | [Outcast] | modulus_: what is wrong with broadvoice? |
20:24.40 | BrianR___ | Mmm... 12648430 |
20:24.58 | modulus_ | outcast, all calls are answered by * |
20:24.58 | PatrickDK | wrong? broadvoice? heh |
20:25.07 | modulus_ | but the caller can't hear anything |
20:25.07 | PatrickDK | broadvoice won't follow sip standards correctly |
20:25.16 | modulus_ | patrickdk, i'm starting to realise that too |
20:25.24 | modulus_ | nothing changed in my asterisk box |
20:25.28 | modulus_ | it just stopped working |
20:25.50 | modulus_ | and their number on their site just has you hold for 15 minutes then says "your call cannot be transfered. thankyou" |
20:26.04 | [Outcast] | modulus_: I have been using broadvoice for over six month without a problem |
20:26.17 | modulus_ | outcast, this is the 3rd time broadvoice is down |
20:26.36 | modulus_ | i'm on the west coast |
20:26.37 | [Outcast] | modulus_: mine is working fine right now. |
20:26.58 | [Outcast] | modulus_: i am using the lax server as well |
20:27.14 | modulus_ | outcast, what version of *? |
20:27.28 | ScarletCrusader | is there a documentation site which goes through the procedure of adding a sip phone to asterisk? |
20:27.41 | dsmouse | ~rtfw |
20:27.42 | jbot | from memory, rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
20:27.48 | [Outcast] | modulus_: 1.0.5 |
20:28.01 | [Outcast] | modulus_: without the broadvoice patch |
20:28.06 | modulus_ | outcast, yesterday no problems |
20:28.12 | modulus_ | i don't use the broadvoice patch either |
20:28.21 | [Outcast] | modulus_: no problems yesterday either |
20:28.27 | modulus_ | this morning, all calls coming in via broadvoice can't hear anything |
20:28.30 | modulus_ | it's just silence |
20:28.32 | modulus_ | dead silence |
20:28.38 | modulus_ | i watch the cli and asterisk picks up fine |
20:30.08 | modulus_ | on hold... (again) |
20:31.41 | tzanger | pdracevich: not offhand |
20:31.45 | tzanger | get a packet cap and post it |
20:32.43 | dsmouse | modulus_: what did it say was the codec it was useing? |
20:32.49 | modulus_ | ulaw |
20:33.00 | modulus_ | did they stop using ulaw? |
20:33.12 | jaiger | tzanger, have you got the echo canceller yet? |
20:33.15 | tzanger | tcpdump -s0 -w blah.dump host 1.2.3.4 and host 5.6.7.8 and port 4569 |
20:33.25 | dsmouse | beats me, but that sure could be a codec mismatch |
20:33.27 | tzanger | jaiger: yes but I need pinouts :-) |
20:33.28 | terrapen | can anyone recommend a good German voip provider? |
20:33.29 | [Outcast] | broadvoice will stop using ulaw |
20:33.38 | [Outcast] | never |
20:33.46 | terrapen | i'm going to visit a friend in Germany and i'd like to set his business up with an asterisk pbx |
20:33.53 | jaiger | tzanger, I upgraded mine with a daughter card today - sounds pretty good now |
20:34.03 | jaiger | tzanger, didn't I get that to you? |
20:34.11 | dsmouse | terrapen: you've written a app_ before, right? |
20:34.43 | tzanger | daughter card? |
20:34.47 | tzanger | jaiger: I have a different echo can |
20:34.58 | jaiger | tzanger, ahh, what'd you get? |
20:35.18 | tzanger | uh |
20:35.19 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
20:35.22 | jaiger | tzanger, yeah the 2572 can be upgraded with 2 daughter cards |
20:35.35 | tzanger | trying to think what it is now |
20:35.39 | tzanger | I can't remember offhand |
20:36.23 | shido6 | shido6 |
20:36.41 | jaiger | tzanger, one card adds time - up to 128ms tail - and the other adds some other dsp features like audio level adjustment |
20:36.43 | cp5 | has anyone used extension 'hint's and noticed that they stop working after an asterisk reload? |
20:36.46 | cp5 | for outbound alls |
20:36.47 | cp5 | calls |
20:36.47 | tzanger | jaiger: ahhhhh |
20:37.10 | *** join/#asterisk ReVoK (ReVoK@82.224.60.46) |
20:37.12 | ReVoK | hi |
20:37.18 | tzanger | this is bugging me now |
20:37.19 | tzanger | wtf was it |
20:37.22 | jaiger | and there are multiple versions of the 'feature' card with more/less features |
20:38.28 | tzanger | sounds hideous |
20:38.36 | tzanger | so what is the total cost of your echo can now? |
20:38.45 | tzanger | and did you get the upgrade cards off ebay or from tellabs themselves? |
20:38.56 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
20:39.11 | km- | tzanger: hey dude |
20:39.15 | tzanger | km-: |
20:39.21 | km- | tzanger: twisted had an excellent idea for my problem, till it didnt work |
20:39.29 | jaiger | tzanger, I got the upgrade card from ebay but also found some vendors as backup |
20:39.33 | tzanger | km-: hahaha what was his idea? |
20:39.39 | km- | tzanger: he said do incoming=yes and then set extension s,1 as DISA |
20:39.43 | km- | tzanger: with no password |
20:39.47 | tzanger | you mean immediate=yes |
20:39.50 | tzanger | ? |
20:39.54 | km- | yeah, immediate=yes |
20:40.04 | jaiger | tzanger, the upgrade card came mounted on another 2572 for ~$80 shipped |
20:40.06 | tzanger | km-: why didn't that work? |
20:40.11 | tzanger | jaiger: nice |
20:40.13 | km- | tzanger: but for some reason, the digits dialed weren't being sent over the T1 |
20:40.13 | tzanger | very nice |
20:40.18 | tzanger | ... |
20:40.20 | jaiger | and a vendor quoted me $60 for the card alone - not too expensive |
20:40.22 | tzanger | you have some very strange problems |
20:40.22 | km- | tzanger: I think that the NEC does have some sort of dialplan problem |
20:40.39 | tzanger | km-: and those fuckers are so hard to configure compared to my norstar |
20:40.48 | km- | tzanger: hehe |
20:40.50 | [Outcast] | I don't understand why you guys have so much trouble with broadvoice. I use to the asterisk support for them. If there is any way i can help let me know |
20:40.58 | km- | I'm thinking of requiring everyone to dial 9 twice before dialing |
20:41.03 | km- | but thats not much better than dialing # at the end |
20:41.11 | tzanger | yeah? |
20:41.26 | Nugget | I think that dialing prefixes are a curious relic from a simpler time. I think they're avoidable with modern technology and that avoiding them is good. |
20:41.39 | km- | because I can always set an exten => 9 to get them in the context waiting to do some magic |
20:41.45 | tzanger | Nugget: not with his NEC it seems -- not unless he can reprogram it |
20:41.58 | km- | yeah, I need the special programming card to fix this problem |
20:42.00 | km- | which is $$$ |
20:42.00 | tzanger | my norstar can't be told to avoid 9 either |
20:42.03 | Nugget | oof |
20:42.17 | tzanger | since you can't select a PRI channel like you can an analog or even CAS T1 trunk |
20:42.20 | tzanger | so you have to route it |
20:42.29 | tzanger | km-: you do? You can't do it through a handset? |
20:42.33 | km- | ariel was saying that Manx had a problem similar to this that he fixed by changing his wink times |
20:42.44 | km- | tzanger: I can't change the dialplan settings without the programming card I think |
20:42.48 | km- | tzanger: I can change DID's |
20:42.51 | tzanger | owie |
20:42.53 | km- | tzanger: but I dont know about dialplan |
20:43.51 | km- | it sucks because, Asterisk is working fine, but the PBX is what's not working, but the company will naturally implicate Asterisk, because "it works fine otherwise" |
20:44.09 | terrapen | y |
20:44.12 | Nugget | heh, yep. |
20:44.37 | km- | unless the wink timing stuff is the magic |
20:44.56 | km- | oh. |
20:45.04 | jaiger | tzanger, I've probably put out < $300 for the echo can hardware I use. I have purchased another $150 in 'spare' parts while dicking around |
20:45.31 | tzanger | jaiger: *nods* |
20:45.41 | harryvv | When doing a sip show peer on a peer I get this info and mabey it is interfearing with athentication? Expire : -1 |
20:45.42 | harryvv | <PROTECTED> |
20:45.43 | km- | tzanger: I could setup extensions 1 and 0 in preincomingpbx, then create seperate contexts for "toll dial' and "international dial" and then force everyone to dial the 1 for calls |
20:45.45 | tzanger | km-: have you tried playing with the wink settings? |
20:45.59 | tzanger | km-: you could, but that's painful too |
20:46.21 | km- | tzanger: no, I don't have the knowledge to know what I'm doing with that, I'm afraid that if I fiddle with settings I may damage something? |
20:46.30 | modulus_ | ugh i hate broadvoice |
20:46.35 | km- | tzanger: or do you think it's fine to fiddle with those? Like I said, I'm a T1 n00b |
20:46.46 | modulus_ | i've never had any problems with any voip providers |
20:46.49 | modulus_ | broadvoice is the first |
20:46.53 | km- | voicepulse |
20:46.59 | km- | I've had problems with voicepulse before |
20:47.07 | dsmouse | modulus_: I've had problems with vonage. |
20:47.09 | tzanger | km-: damage it: no... but fuck it up and not know how to fix it: likely. That's why I was so hesitant about doing anything with the norstar but now I understand it to a very good level and I'm fearless |
20:47.10 | km- | only provider I trust anymore is NuFone |
20:47.15 | bjohnson | modulus_: I rmember you bitching about nufone |
20:47.29 | modulus_ | bjohnson, they just have shitty bandwidth |
20:47.30 | tzanger | km-: you can fiddle with the settings and you will not break anything, it's all just bits after all |
20:47.41 | tzanger | you can make it stop working but you can comment out your changes and get it back to the level of not working you have today :-) |
20:47.42 | modulus_ | broadvoice just has shitty services |
20:47.43 | moonwick | heh |
20:47.45 | Logan | gr0mit: We've had three phones fail in a manner that suggests a bad NIC or something. |
20:47.49 | km- | yeah, definitely |
20:48.26 | modulus_ | i still get those "timed out lagged" messages for nufone |
20:48.36 | tzanger | modulus_: packet cap |
20:48.45 | modulus_ | and i'm on a gig-e link upstream via global crossing |
20:48.51 | modulus_ | tzanger, what? |
20:48.52 | tzanger | sounds like you just stop receiving data for 2s and asterisk auto-congests |
20:49.01 | tzanger | my ADSL line has that problem from time to time |
20:49.02 | km- | bbiafm |
20:49.03 | dsmouse | modulus_: /me is envious |
20:49.29 | tzanger | it's not nufone's issue... I have some jitter packet caps that make me cry with respect to how deterministic their network is |
20:49.47 | modulus_ | they could put a proxy in LA |
20:49.59 | modulus_ | on real bandwidth |
20:50.02 | modulus_ | somewhere close |
20:50.10 | tzanger | I'm talking nominal jitter of 500us (half a millisecond) with peaks of maybe 5-10ms |
20:50.31 | modulus_ | i offered to put a box on global crossing for them |
20:50.32 | tzanger | my ADSL line is nominally about 50ms of jitter with peaks up to 1500ms (1.5s) |
20:50.37 | tzanger | now |
20:50.39 | tzanger | having said that |
20:50.39 | modulus_ | as a poc |
20:50.49 | tzanger | 50ms of jitter is *NOTHING* in real-world |
20:50.49 | Nugget | after 2.5 months of bitching I still cannot get acceptable quality from my area code 512 DID with voicepulse. |
20:50.54 | modulus_ | tzanger, you're talking about adsl |
20:50.57 | Nugget | it's totally unusable. |
20:51.13 | tzanger | Nugget: so cancel, have your credit card charges reversed and find someone else :-) |
20:51.15 | modulus_ | i'm talking about gig-e capacities |
20:51.24 | Nugget | I can't find anyone else that offers local DIDs |
20:51.27 | Nugget | any suggestions? |
20:51.27 | harryvv | I have checked and rechecked the settings between spa1000 sip.conf and extensions.conf reloaded and still getting a authenticaion message. What else needs to be done. |
20:51.27 | tzanger | modulus_: at gig-e you should have no issues unless you have a bad driver |
20:51.30 | *** join/#asterisk pdracevich (~paul@smtp.aucklandtax.co.nz) |
20:51.33 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
20:51.34 | tzanger | Nugget: iax.cc seems to have DIDs everywhere |
20:51.38 | Nugget | I'll try them. |
20:51.47 | modulus_ | tzanger, nufone always times out still |
20:51.49 | tzanger | Nugget: they're not perfect but they certainly seem acceptable |
20:52.07 | tzanger | modulus_: have you tried using switch-2 in lieu of switch-1.nufone.net? |
20:52.13 | tzanger | (it's in Michigan as opposed to Chicago) |
20:52.19 | tzanger | could just be a route path problem |
20:52.20 | modulus_ | tzanger, yeah that's worse |
20:52.25 | shido6 | who |
20:52.27 | shido6 | whoa |
20:52.31 | shido6 | nufone times out? |
20:52.35 | PBXtech | can you wisper page a IP connection? |
20:52.37 | tzanger | modulus_: I'd open up a ticket with your provider |
20:52.46 | modulus_ | global crossing? |
20:52.47 | modulus_ | hahahaha |
20:52.55 | modulus_ | i'm the network admin here |
20:52.58 | tzanger | I have been using nufone for the past year and any connectivity issues have been on my end |
20:53.08 | modulus_ | i _promise_ you it's not global crossing's issue |
20:53.12 | ionix | beside the huge long distance charge, nufone is ok |
20:53.17 | tzanger | modulus_: can you set up a packet capture at both ends of your network and see what they say when a timeout occurs? |
20:53.37 | modulus_ | tzanger, i'd love to but jerjer didn't want to even talk to me |
20:54.01 | tzanger | modulus_: if you can get that data together and find jerjer I am willing to be he will help you by packet cap'ing at his end (I'm assuming this problem is easy to reproduce) |
20:54.03 | modulus_ | i guess he's a networking guru as well as a nufone voip guru |
20:54.17 | tzanger | jerjer's very curt unless you have some solid data to prove the issue might be around his side of the ship |
20:54.26 | modulus_ | i've shown traces |
20:54.35 | modulus_ | where the last hop is his router interface |
20:54.45 | tzanger | you have packet caps on his side of the router? |
20:54.50 | modulus_ | where the ms times shoot up 150 ms |
20:55.01 | PBXtech | can you wisper page a IP connection? |
20:55.04 | modulus_ | every link before is ok |
20:55.04 | ionix | I had some issues with Nufone too |
20:55.09 | ionix | his routing jumping to 150ms |
20:55.09 | pdracevich | can some one have a look, it is an iax2 connection that was working perfectly, I think it has something to do with lag http://pastebin.ca/6346 |
20:55.10 | tzanger | PBXtech: not sure |
20:55.22 | Graphikos | I'm can't get my IP phone to register... complete newbie here... can someone point out my stupidity? |
20:55.40 | ionix | Hey, anyone ever used the Nortel Symposium system with asterisk ? |
20:55.45 | ionix | Trying to make it work with TAPI here. |
20:55.54 | *** join/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net) |
20:55.54 | tzanger | pdracevich: get me a packet capture of that with -s0 and -w |
20:56.14 | tzanger | I'm not sure what "INVAL" means (invalid,yes, but invalid what, auth?) |
20:56.20 | *** join/#asterisk pUmkInhEd (nobody@s142-179-184-59.ab.hsia.telus.net) |
20:56.22 | pdracevich | that was a iax2 debug log |
20:56.35 | tzanger | pdracevich: I want a tcpdump or ethereal packet capture with -s0 and -w |
20:56.52 | tzanger | pdracevich: in other words, I want a copy of the actual bits on the wire :-) |
20:57.08 | tzanger | ionix: odd |
20:57.13 | pUmkInhEd | hi, hopefully a quick q for you guys, does asterisk provide functionality for bridging two norstar systems? I checked asteriskdocs.org and found a blank section in the current docs on this area |
20:57.26 | tzanger | pUmkInhEd: bridging two norstar systems? |
20:57.34 | tzanger | pUmkInhEd: explain, I have done some norstar/asterisk integration |
20:57.46 | pUmkInhEd | well i currently have an FXO/FXS setup |
20:57.59 | dstevens_ | Does somebody know how to fix a broken package, i do not understand any more. |
20:58.07 | pUmkInhEd | and it feeds into an ATA which we access as an outside line (ie line pool b) |
20:58.38 | tzanger | pUmkInhEd: eh? |
20:58.40 | tzanger | that's not making sense |
20:58.47 | tzanger | you have a norstar system with analog trunk lines, ok |
20:59.09 | dstevens_ | {sorry wrong tab ignore previous} |
20:59.11 | *** join/#asterisk Ayano (~erik_leee@209.143.187.254) |
20:59.14 | pUmkInhEd | tzanger: sorry if I am confusing |
20:59.14 | tzanger | it's feeding into asterisk over an FXS interface (TDM400 or channel bank + T1) |
20:59.17 | tzanger | right? |
20:59.30 | pUmkInhEd | tzanger: neg, i currently do not use asterisk software |
20:59.35 | tzanger | ok |
20:59.43 | PatrickDK | tzanger, using like a sipura device he means |
20:59.45 | Ayano | what is the best way to achieve load balance and failover with *? |
20:59.54 | pUmkInhEd | PatrickDK exactly |
20:59.57 | tzanger | so you have a norstar system with analog trunk lines that plug into the telco jacks then? |
21:00.49 | shido6 | ionix |
21:00.52 | shido6 | what issues? |
21:01.09 | pUmkInhEd | yes but the analog jacks plug into another device which digitizes the sound, sends it to another unit connected to another norstar system |
21:01.13 | *** join/#asterisk _Raptor_ (RaptorX@dsl-082-083-172-119.arcor-ip.net) |
21:01.16 | _Raptor_ | hi |
21:01.35 | tzanger | pUmkInhEd: hmm what does that get you right now? |
21:01.40 | pUmkInhEd | the other unit decodes and outputs the audio to the norstar |
21:01.48 | tzanger | normal way to join to norstar systems is to use a DTI with MCDN keys |
21:01.48 | modulus_ | OMG |
21:01.58 | pUmkInhEd | tzanger I have a bridge between two norstars, if I press 9 I get local dial tone as expected |
21:01.59 | modulus_ | Peer 'broadvoice' is now UNREACHABLE! Last qualify: 6 |
21:02.13 | pUmkInhEd | if I press 8 I get dialtone from the other norstar |
21:02.16 | modulus_ | WTF ARE THEY DOING? |
21:02.21 | pUmkInhEd | but its feature set is somewhat limited |
21:02.23 | tzanger | pUmkInhEd: ok |
21:02.40 | tzanger | and your question with asterisk integration is what now |
21:03.55 | tzanger | Moc: holy disclaimer signature line, batman! |
21:03.55 | harryvv | I have a question about sip.conf and sip show peer. If for example a username and password is mismatched on either a spa1000 or sip.conf would this interfear with the * getting a Addr-> IP address ? |
21:03.59 | tzanger | quelle fromage! |
21:04.27 | PatrickDK | harryvv, hmm, yes, that is the point of username/password |
21:04.32 | modulus_ | OMG BROADVOICE IS A PEICE OF SHIT |
21:04.35 | modulus_ | fuckin' eh |
21:04.44 | modulus_ | someone report them to bbb |
21:04.53 | modulus_ | someone start a hate/fan site |
21:05.02 | djin | please take that somewhere else |
21:05.07 | lorion | what is an alternative to Broadvoice? |
21:05.08 | CoaxD | modulus: Um, you might want to reconsider that statement |
21:05.19 | CoaxD | modulus: Already, you have lorion, here, who is actually stupid enough to listen to you |
21:05.24 | harryvv | Patrick, the user/pass are the same on sip.conf and the spa1000. doing a sip show peer spa_01 I get a Addr->IP : (Unspecified) Port 0 |
21:05.25 | harryvv | <PROTECTED> |
21:05.28 | modulus_ | ok sorry lorion |
21:05.37 | pUmkInhEd | tzanger: my intention is to have a larger feature set than the ata can provide |
21:05.45 | PatrickDK | harryvv, do you have host=dynamic? |
21:05.48 | modulus_ | lorion, they all kinda suck |
21:05.57 | modulus_ | i haven't found 1 voip provider that i like yet |
21:05.58 | CoaxD | modulus: Everybody bitches about non-regulated telco services |
21:06.01 | modulus_ | and i've tried about 10 |
21:06.08 | CoaxD | modulus: er |
21:06.15 | CoaxD | modulus: Everybody bitches about regulated telco services |
21:06.21 | tzanger | pUmkInhEd: ok |
21:06.21 | harryvv | :) |
21:06.21 | CoaxD | modulus: They always cost 10x that of a non-regulated |
21:06.46 | CoaxD | modulus: But you know what? |
21:06.46 | tzanger | pUmkInhEd: that's an awfully broad statement |
21:06.46 | CoaxD | modulus: Regulated telephone service is WORTH IT sometimes. |
21:06.48 | modulus_ | yeah you're telling me |
21:07.05 | tzanger | CoaxD: I happen to agree with modulus_ ... just from watching interaction with people with broadvoice |
21:07.05 | modulus_ | even with all the 7 plus taxes per channel |
21:07.12 | CoaxD | tzanger: Heh :) |
21:07.20 | *** join/#asterisk TrevMeister (~thammonds@ip68-4-223-70.oc.oc.cox.net) |
21:07.20 | CoaxD | tzanger: (So do I. But thats not the point.) |
21:07.29 | tzanger | :-p |
21:07.44 | modulus_ | i've been on hold for over an hour |
21:07.53 | WGFreewill | broadvoice has always worked great for me |
21:07.57 | WGFreewill | i just made a call |
21:08.00 | WGFreewill | worked great |
21:08.40 | WGFreewill | (its only been a few months that its been up) |
21:08.52 | bjohnson | lorion: there are about 200 alternatives to broadvoice listed on the wiki |
21:09.12 | shido6 | modulus_ |
21:09.16 | shido6 | u need to make a call? |
21:09.33 | CoaxD | the thing is, broadvoice DOESNT suck. And if you actually have a reaosn to call their technical support, you obviously didnt read what was on the website |
21:09.41 | CoaxD | when broadvoice goes DOWN, however, yeah, they SUCK |
21:09.53 | CoaxD | the thing is, regulated telcos CANT go down. Thats the whole point of a regulated telco. |
21:09.55 | modulus_ | shido, i have a DID that's half down |
21:10.02 | modulus_ | my asterisk box picks up for the DID |
21:10.06 | modulus_ | but the caller hears nothing |
21:10.12 | shido6 | ouch |
21:10.14 | modulus_ | and i see my menu system exec playbacks() |
21:10.15 | CoaxD | modulus: Sure sounds like a firewall proiblem to me |
21:10.21 | modulus_ | omg |
21:10.26 | WGFreewill | or maybe some hung asterisk |
21:10.32 | WGFreewill | have you restarted asterisk? |
21:10.48 | modulus_ | haha coaxd and wgfreewill are so cute trying to help me |
21:10.50 | modulus_ | but uh |
21:10.54 | CoaxD | modulus: lameriferous |
21:11.06 | pcm | coaxD: who's regulated telco ? |
21:11.10 | modulus_ | nothing changed network-wise nor system-wise |
21:11.15 | WGFreewill | rboc |
21:11.22 | modulus_ | incoming calls just stopped hearing anything |
21:11.28 | Ayano | what is the best way to achieve load balance and failover with *? |
21:11.30 | CoaxD | pcm: Any pstn telco in the united states is regulated. :) |
21:11.44 | CoaxD | pcm: voip, on the other hand, is not regulated |
21:11.49 | WGFreewill | Ayano: thats an interesting question |
21:11.55 | WGFreewill | PSTN failover |
21:11.57 | WGFreewill | SIP failover |
21:11.58 | CoaxD | pcm; If I am a voip telco and i sell you an 800 number, that 800 number belongs to ME. |
21:12.17 | CoaxD | pcm: If my business goes away, so does your 800 number - unless I decide to transfer it to you |
21:12.22 | CoaxD | pcm: Same goes with any DID |
21:12.34 | CoaxD | pcm: regulated telco service? That phone number belongs to YOU. |
21:12.34 | pcm | CoaxD: so any voip service can go down ... since they won't get any penalties for that .... |
21:12.41 | modulus_ | jesus the last time i remember talking to broadvoice, they had to replace the whole LAX proxy |
21:12.42 | CoaxD | pcm: You got it |
21:12.48 | Ayano | WGFreewill: either. |
21:12.50 | CoaxD | pcm: It isnt about penalties, btw. its about law |
21:12.56 | modulus_ | they sure don't sound like they got a clue |
21:13.02 | pcm | CoaxD: law is penalties when you brake it |
21:13.07 | CoaxD | pcm: For sure |
21:13.20 | bjohnson | how can I tell what extension a call came in on? I'm trying to get a voip call to come in on an extension that matches the DID called .. but I can't tell from the messages whether it is working or just coming in on the s,1 in that context |
21:13.31 | CoaxD | bjohnson: set verbose 5 |
21:13.47 | pcm | CoaxD: but redundancy costs x 2 or even more ... :) |
21:13.55 | CoaxD | pcm: *far* more than x2 |
21:13.59 | plappy | any ideas as to why I keep getting "Maximum retries exceeded on call..." after 5-10 seconds on local sip to sip calls? |
21:14.01 | pUmkInhEd | gah, must fix printer, bbiab, srry tzanger |
21:14.04 | CoaxD | pcm: (For GOOD redundancy) |
21:14.17 | CoaxD | plappy: set verbose 5 |
21:14.24 | pcm | CoaxD: so how much redundant is the regualar telco ? 3 x ? |
21:14.34 | CoaxD | pcm: Well, keep this in mind: Most telcos only have 1 switch |
21:14.43 | CoaxD | pcm: Although the switch can provide redundancy within itself, chances are, it doesn't |
21:15.04 | CoaxD | pcm: The redundancy in question needs to be things like. power, extra patch panels, wiring, etc |
21:15.41 | lorion | what would cause one asterisk extension from being able to dial another? |
21:16.00 | Ayano | extensions.conf? |
21:16.31 | PatrickDK | context :) |
21:16.37 | lorion | do I have to state in extensions.conf whether extension1 can call extension2? |
21:16.49 | bjohnson | lorion: yes |
21:17.00 | bjohnson | not directly though |
21:17.00 | lorion | do you have an example? |
21:17.12 | bjohnson | lorion: thousands on the wiki |
21:17.19 | bjohnson | ~docs |
21:17.20 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
21:17.29 | bjohnson | start on the extensions.conf page |
21:17.54 | harryvv | bj, I made some changes to make the auth simpler. Well that did not work. |
21:17.55 | bjohnson | each device has to be configured with an extension to ring that device |
21:18.04 | bjohnson | usually done with a mcro |
21:18.26 | bjohnson | lorion: try making some extensions that do not call another device |
21:18.32 | bjohnson | try from all devices |
21:18.39 | bjohnson | then you know incoming is working |
21:18.41 | lorion | I can call VM |
21:18.47 | bjohnson | ok |
21:18.52 | lorion | I can cal my extension and it rings on line 2 |
21:19.00 | bjohnson | make an exten definition for each of you devices |
21:19.00 | Ayano | Okay guys, I see your point about redundancy. So how do you do load balancing? |
21:19.04 | lorion | when I call another extension i get 404 |
21:19.37 | harryvv | bj, are you talking to me? |
21:19.43 | bjohnson | harryvv: no |
21:19.44 | Ayano | lorion: make sure that both are authenticated too. |
21:19.46 | WGFreewill | keepalived |
21:19.46 | harryvv | k |
21:19.51 | WGFreewill | vrrp |
21:19.54 | pcm | CoaxD: so you can realize redundancy using 2 or 3 voip providers |
21:20.01 | WGFreewill | but its failover not active-active |
21:20.02 | sung | tards. |
21:20.09 | WGFreewill | for inbound |
21:20.38 | WGFreewill | exten => s,1,ChanIsAvail(SIP/Provider1&SIP/Provider2) |
21:20.42 | WGFreewill | for outbound |
21:21.09 | bjohnson | WGFreewill: the superdial macro is a little more straght forward |
21:21.18 | bjohnson | and easier to add/remove voip providers |
21:21.25 | tzanger | bjohnson: what are you collecting |
21:21.27 | WGFreewill | what is superdial macro |
21:21.33 | tzanger | bjohnson: and are you going to torastricon on friday? |
21:21.47 | Nugget | http://www.google.com/search?q=asterisk+superdial+macro |
21:21.49 | bjohnson | for inbound I've been asking voip providers to forward to my telco pstn if my * is not answering them |
21:22.05 | bjohnson | tzanger: I've been collecting voip provider accounts |
21:22.11 | bjohnson | new hobby I guess |
21:22.27 | Nugget | the superdial macro ought to do enumlookup. |
21:22.28 | bjohnson | tzanger: I'm not going |
21:22.31 | WGFreewill | (Miami VON, on now) |
21:22.46 | bjohnson | Nugget: the enumlookup is done and then fed to superdial |
21:22.54 | _Raptor_ | cu |
21:23.00 | Nugget | you could do it that way, yes. |
21:23.09 | tzanger | bjohnson: I'll trade you a VPC for a livevoip |
21:23.09 | tzanger | heh |
21:23.18 | Nugget | but without enum it's not as "super" as it could be. :) |
21:23.20 | bjohnson | exten => s,4,EnumLookup(${tfnumber}) |
21:23.20 | bjohnson | exten => s,5,GotoIf($[${ENUM:0:3} = IAX]?6:7) ; SIP behind NAT not working |
21:23.20 | bjohnson | exten => s,6,Macro(superdial,${ENUM},,,,voip,${MAXVOIPCALLS},Johnson Engineering Consultants,519-271-9923,e164) |
21:23.40 | bjohnson | the absolute nicest thing about superdial ... |
21:23.48 | bjohnson | is that it returns to the context that called it |
21:24.00 | bjohnson | (if the call is not connected) |
21:24.03 | Nugget | that's spiffy. |
21:24.21 | Nugget | I don't use superdial, but it doesn't look very different from what I built myself, in ignorance of superdial's existence. |
21:24.21 | bjohnson | rather than a speghetti string of goto's all over the place |
21:24.35 | bjohnson | it's based off of examples on the wiki |
21:24.39 | modulus_ | bjohnson that's almost as nice as having one of those reliable broadvoice DIDs |
21:24.43 | Nugget | ${DIALSTATUS} is tasty. |
21:24.47 | *** join/#asterisk Alejandriax26 (~nurbina23@proxy.more.cl) |
21:25.07 | *** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
21:25.12 | *** part/#asterisk djin (~djin@gridfox.xs4all.nl) |
21:25.14 | bjohnson | so to try another channel out .. |
21:25.19 | bjohnson | exten => s,7,Macro(superdial,IAX2/iaxnboom/1519${tfnumber},,,,voip,${MAXVOIPCALLS},Johnson Engineering Consultants,519-271-9923,nboom) |
21:25.23 | WGFreewill | the dialparties.agi |
21:25.25 | WGFreewill | in the AMP package |
21:25.29 | WGFreewill | is the superdial marco |
21:25.31 | WGFreewill | to the next level |
21:25.38 | *** part/#asterisk XeNoSiS (user@216.234.145.18) |
21:25.44 | *** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com) |
21:25.50 | mishehu | what is AMP? asterisk made painful ? |
21:25.51 | mishehu | heh |
21:25.54 | ariel_ | argh people will never learn. Get this a person wants to get a sipura working via not one but 3 NATs over a sat link and do faxing. ARgh |
21:25.58 | tzanger | mishehu: actually that's not far from the truth :-) |
21:25.59 | WGFreewill | asterisk more pretty |
21:26.15 | roamer323 | whatz the detail on the torastrcon? |
21:26.23 | tzanger | roamer323: opensource.meetup.com/42 |
21:26.29 | loud | ariel_, wait until they ask for 4 simultaneous calls through sat link. |
21:26.38 | bjohnson | ariel_: faxing is tricky .. but I have mine working through 3 nats |
21:26.40 | WGFreewill | if your not scared to go edit a few .conf files, it works great, but its lacking a few pieces |
21:26.44 | ariel_ | bjohnson, I dont' use amp for there conf I use them for the reports. |
21:26.47 | bjohnson | ariel_: (not with fax though) |
21:26.48 | roamer323 | thx tzanger |
21:27.07 | [Outcast] | does anyone have some good res_perl examples? |
21:27.25 | ariel_ | bjohnson, I have already gotten voice calls working for them. Now they want faxing too. ARgh this is not going to work. |
21:27.45 | tzanger | ariel_: use some app_rxfax/txfax and scp magic |
21:28.02 | ariel_ | bjohnson, actually what I do is install asterisk@home and change the conf files to my use. |
21:28.05 | modulus_ | broadvoice could use a little "magic" |
21:28.07 | loud | ariel_, which sat platform, gilat, ipsat ? |
21:28.56 | WGFreewill | yep I use amp and @home pieces |
21:28.56 | bjohnson | ariel_: vpn? |
21:28.56 | ariel_ | loud, don't know trying to find out now. |
21:29.09 | WGFreewill | faxing needs 0% packet loss g711, and latency isnt going to help any |
21:29.21 | modulus_ | i think broadvoice just power cycled their lax proxy |
21:29.32 | ariel_ | WGFreewill, yes sir your correct. |
21:29.36 | modulus_ | but the calls coming in still can't hear anything |
21:29.40 | WGFreewill | rxfax at the edge |
21:29.49 | loud | broadvoice service last night like at 11 pm pst was awful. |
21:30.02 | ariel_ | tzanger, spandsp is not working for it either. |
21:30.09 | modulus_ | loud, it hasn't worked for me since then |
21:30.14 | tzanger | ariel_: damn that's not good then |
21:30.23 | loud | it works now, althought i have one way audio though |
21:30.24 | tzanger | perhaps a pair of t.38-aware ATAs |
21:30.58 | [Outcast] | modulus_: what do you have your externip set to in your general section? |
21:31.10 | modulus_ | the public ip address |
21:31.14 | fearnor | wg: even with 0% packet loss, 1ms latency, 1ms jitter, you aren't guaranteed 100% working fax ;) |
21:31.17 | modulus_ | of their lax server |
21:31.29 | modulus_ | x.x.8.x |
21:31.35 | fearnor | it'll be 90% or so. |
21:31.50 | bjohnson | ariel_: what about just digitizing faxes locally, emailing them as an attachment elsewhere, and reprocessing them? |
21:31.54 | fearnor | i had moderate luck with fax over TDMoE though. |
21:31.58 | [Outcast] | are you be hind nat? |
21:32.03 | tzanger | fearnor: me too |
21:32.05 | modulus_ | ooh broadvoice is now UNREACHABLE again! |
21:32.05 | fearnor | bjohnson: bad for the end-to-end. |
21:32.11 | tzanger | oh tdmoe |
21:32.13 | tzanger | sorry I haven't tried that |
21:32.17 | modulus_ | no outcast |
21:32.31 | [Outcast] | modulus_ : turn qualify off for broadvoice, just causes problems |
21:32.32 | fearnor | bj: imagine if phone number is busy etc, you need to implement redelivery logic etc |
21:32.43 | Blackvel | who has working FWD with asterisk behind NAT? does that work? |
21:32.43 | [Outcast] | modulus are you using srvlookup? |
21:32.48 | ariel_ | bjohnson, it's a customer that wants to be able to use a fax machine in his office. There not too up on the computer stuff there yet. |
21:32.56 | loud | modulus_, do they support g729 ? bv i mean |
21:33.08 | modulus_ | loud, i kinda doubt it |
21:33.09 | [Outcast] | Blackvel: i do you need to but in the DMZ |
21:33.13 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
21:33.19 | modulus_ | loud, they can't even support sip |
21:33.19 | shmaltz | hi everybody |
21:33.20 | [Outcast] | loud: no they do not |
21:33.23 | loud | so just g711 :/ |
21:33.32 | Blackvel | loud: i think g711 |
21:33.33 | modulus_ | ulaw |
21:33.34 | [Outcast] | loud: yes |
21:33.42 | loud | i think i screwed it up buying an international plus blah plan |
21:33.43 | WGFreewill | wish efax was cheaper |
21:33.45 | Blackvel | modules: they do not support SIP? hehehhe....are you sure? |
21:33.48 | modulus_ | registrations are timing out |
21:33.55 | modulus_ | blackvel, they try to |
21:33.57 | Essobi | Umm. I've got a problem.. My 5400's send SIP calls to my * box with a random port.. and specifingy port= or port=0 on my peers list won't match them.. anyone know how to rectify that? |
21:34.00 | modulus_ | but they fail |
21:34.12 | shmaltz | does anybody know if it is possible (and how) to show the status of anohter extension (SIP device) on the Cisco 7960 or Cisco 7914? |
21:34.25 | loud | yes |
21:34.26 | modulus_ | wow their lax proxy is completely down now |
21:34.27 | modulus_ | neat |
21:34.33 | [Outcast] | modulus_: do you want my config |
21:34.36 | Blackvel | modulus_: my sip show registry shows: "Registered". but i am unable to call echo |
21:34.38 | Beirdo | mikegrb: http://homokaasu.org/rasterbator/galleryimage.gas?2232 |
21:34.39 | modulus_ | maybe they got a tech there punching the copper back down |
21:34.51 | Beirdo | what some people will put on their walls |
21:34.59 | loud | shmaltz: show reg through CLI, on the 7960 |
21:35.03 | modulus_ | outcast, thank you for your samaritan antics but i know it's nothing on my end. |
21:35.15 | modulus_ | outcast, nothing has changed on my end for over a week |
21:35.20 | modulus_ | and last night it stopped working |
21:35.27 | tzanger | and he's sitting on gige, the bastard |
21:35.35 | modulus_ | yes i'm a bastard |
21:35.42 | modulus_ | actually i have 3 gig-e uplinks |
21:35.42 | WGFreewill | modulus_: maybe the tech set his crack pipe on the copper your DID comes in on |
21:35.44 | tzanger | see, he doesn't know who his father is |
21:35.48 | modulus_ | global crossing/level3/cogent |
21:35.51 | shmaltz | loud, this will show on a Cisco 7960 the status of another cisco 7960? using one of the line buttons? |
21:36.01 | [Outcast] | modulus_: i have heard that many time when i did support for them, i sure it just a simple thing. |
21:36.03 | tzanger | modulus_: you have a link to cogent, switch-2 should be amazingly close to you |
21:36.10 | modulus_ | omg lax sip proxy is down HARD for broadvoice |
21:36.15 | tzanger | cogent -- the Wal-Mart of bandwidth providers |
21:36.18 | loud | no, this will show the reg status of an extension on another voip gateway. |
21:36.31 | modulus_ | tzanger, but for the other voip providers, using cogent would suck |
21:36.50 | tzanger | modulus_: true, but you said you had pauses going to switch-2 too (actually you said they were worse) |
21:37.07 | shmaltz | loud, somthing like the hint priority |
21:37.12 | tzanger | what's a traceroute to switch-2.nufone.net look like to you |
21:38.15 | shmaltz | is there anyway to show on a sip phone that a call is parked? |
21:38.17 | modulus_ | they both get around 60 ms pings |
21:38.23 | modulus_ | but they both spike and lagg out |
21:38.25 | modulus_ | a lot! |
21:38.39 | *** join/#asterisk kuonSama (kuon@alragore.goyman.com) |
21:38.44 | kuonSama | hello everybody |
21:38.54 | Blackvel | hmm with 1.0.5 DISA only accepts one dial in, but not the extensions from my analog pbx behind the inward number (no extension can be executed) |
21:38.59 | WGFreewill | modulus: try another BV proxy |
21:39.03 | *** join/#asterisk zapa (zapa@200.92.147.148) |
21:39.12 | Blackvel | what is going on? is that a problem of DISA or bristuff? |
21:39.13 | tzanger | modulus_: I said traceroute not ping, I was just curious as to the path |
21:39.16 | modulus_ | wgfreewill, the only one that authenticates my sipuser seems to be the lax one |
21:39.25 | terrapen | http://www.foxnews.com/images/154561/0_25_jackson_michael_legal.jpg |
21:39.31 | terrapen | that spot on his chin |
21:39.38 | terrapen | i think that's where his soul drained out |
21:39.59 | *** join/#asterisk Umaro (~umaro@209.140.74.64) |
21:40.03 | modulus_ | tzanger, gblx->alter.net->garage-webhosting-company-> nufone.net |
21:40.09 | WGFreewill | hmm i dont seem to have provlems using chicago or lax |
21:40.20 | WGFreewill | i am on chicago now |
21:40.37 | modulus_ | when i use other sip proxys, authentication fails |
21:40.44 | modulus_ | lax is the only one that seems to work for me |
21:40.48 | Umaro | Hey guys, anyone setup their spa-2000 with broadvoice manually? I have asterisk connecting to broadvoice, but I want to try configuring my sipura directly to broadvoice to see if the quality problems I am having are related to asterisk or just broadvoice |
21:41.02 | loud | exactly. |
21:41.03 | Umaro | if I auto config it with broadvoice, they'll lock me out of my own sipura |
21:41.08 | modulus_ | umaro, it's broadvoice |
21:41.17 | bjohnson | Umaro: you just missed a gang rape on bv |
21:41.21 | *** join/#asterisk Rith (nobody@66.142.28.35) |
21:41.23 | modulus_ | i'd bet my 2nd nutsack on it |
21:41.27 | Umaro | bjohnson: oic |
21:41.28 | terrapen | haha |
21:41.28 | tzanger | garage-webhosting-company? |
21:41.45 | modulus_ | tzanger, did you look at who nufone hosts with? |
21:41.54 | bjohnson | it's better use the the crap I throw into my garage |
21:41.55 | tzanger | modulus_: that's through global crossing, I wanted to see what your path to switch-2 was like through cogent |
21:42.02 | modulus_ | hold up |
21:42.30 | tzanger | modulus_: switch-2 is through 123.net I think |
21:42.41 | tzanger | but switch-1's through scnet |
21:43.10 | tzanger | I'm 36 and 25ms to him (switch-2,switch-1) and I'm in Canada through Ikano |
21:43.41 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
21:43.42 | *** join/#asterisk bsenicar (~bsenicar@BSN-77-155-238.dsl.siol.net) |
21:44.00 | zapa | hi all, i have e1 with digial trunk to pstn, i have 10 DID, is there any way to change |
21:44.00 | zapa | the caller id to specific one, when i make call to pst. |
21:44.41 | marc_c | ikano?? |
21:44.43 | [Outcast] | There is nothing wrong with BV as long as you configure your box rightg |
21:44.43 | tzanger | zapa: if your provider lets you set CID on outgoing calls, sure |
21:44.46 | tzanger | marc_c: yeah |
21:45.17 | modulus_ | tzanger, cogent->switch-2 |
21:45.19 | marc_c | T1's? |
21:45.20 | Blackvel | whats the best flat provider? broadvoice, nufone, something else? |
21:45.24 | modulus_ | they use cogent as an upstream |
21:45.26 | tzanger | modulus_: so a few hops in cogent and that's it? |
21:45.33 | modulus_ | few? |
21:45.33 | Umaro | [Outcast]: yes.. their hold time sucks though |
21:45.34 | tzanger | and you still have shitty connectivity to them?? |
21:45.35 | bjohnson | Blackvel: why flat? |
21:45.36 | modulus_ | more like GANG |
21:45.36 | loud | nufone = flat ? international ? |
21:45.48 | modulus_ | 14 hops to switch-2 only going through cogent |
21:45.57 | modulus_ | cogent needs help |
21:45.59 | PBXtech | can asterisk do whisper paging? |
21:46.06 | Blackvel | my german providers provides a german flat |
21:46.09 | Blackvel | you pay 20EUR |
21:46.11 | [Outcast] | Umaro: that is because it don't work there any more |
21:46.16 | Blackvel | and you can call germans for free |
21:46.26 | bjohnson | Blackvel: no .. you call them for 20EUR |
21:46.37 | bjohnson | is that per month? |
21:46.47 | Blackvel | but i think there are also providers offering for 20$ germany,uk,usa,etc? |
21:46.50 | Rith | Can anyone point me in the right direction for what equipment I'd need (and what costs I'd run, USD) to set up a small office w/ a few phones and voicemail boxes. the docs on the asterisk.org leave me with a lot of questions |
21:46.50 | Blackvel | yeah per month |
21:46.59 | tzanger | modulus_: I have 4 hops in cogent, 4 in peer1 and 2 in ikano |
21:47.01 | Blackvel | its great if your call volume is always 30EUR and above |
21:47.11 | modulus_ | 12 Internet123.demarc.cogentco.com (66.250.4.86) 68.221 ms 68.642 ms 68.254 ms |
21:47.11 | modulus_ | 13 vl119.lodden.sfld-mi.123.net (216.234.104.114) 68.023 ms 68.077 ms 67.555 ms |
21:47.11 | modulus_ | 14 198.22.67.70 (198.22.67.70) 70.605 ms 68.745 ms 69.363 ms |
21:47.27 | modulus_ | hop 13-14 used to show 200ms increase |
21:47.27 | tzanger | Rith: your post leaves us with just as many. |
21:47.29 | heison | anyone here got a Canadian toll free number ? |
21:47.30 | modulus_ | a month ago |
21:47.38 | tzanger | heison: yes |
21:47.42 | tzanger | modulus_: hmm |
21:47.46 | zapa | tzanger: where i change CID ? in zapata.conf? |
21:48.01 | tzanger | zapa: there (on a channel basis) or using SetCIDNum() in the dialplan |
21:48.26 | Rith | tzanger: understandably. would like to take POTS incoming lines and run them to VoIP phones w/ voicemail and such... mostly confused as to what equipment I'd need |
21:48.44 | shmaltz | does anybody here use a Cisco 79xx? |
21:48.45 | bjohnson | other than a server you don't need any hardware |
21:48.57 | tzanger | Rith: what VOIP phones are you interested in |
21:48.57 | bjohnson | you may "choose" to add hardware |
21:49.07 | tzanger | bjohnson: he will need FXO interfaces |
21:49.09 | ionix | Hey, anyone has ANY IDEA on how I can fill in a name when I have a phone number ? Trying to find a way to access the RBOC database |
21:49.14 | ionix | or query it at least |
21:49.21 | bjohnson | tzanger: not with voip pstn DIDs |
21:49.23 | Rith | don't know yet, that's part of what i'm trying to determine |
21:49.38 | shmaltz | ionix, use a script to query anywho reversi lookup |
21:49.43 | tzanger | he said "POTS incoming lines" -- that indicates the need for FXO interfaces :-) |
21:50.07 | tzanger | Rith: well you don't need any hardware aside from the netwroking infrastructure to connect SIP phones with asterisk |
21:50.07 | bjohnson | he likely doesn't know an alternative exists |
21:50.12 | tzanger | you want decent network infrastructure though |
21:51.32 | bjohnson | Rith: there isn't a magic formula .. you likely want a fxo for each pstn line you want and a voip phone or a fxs+analog phone for each phone you want |
21:51.37 | heison | ~seen sivana |
21:51.40 | jbot | sivana is currently on #asterisk. Has said a total of 6 messages. Is idling for 6h 11m 46s |
21:52.00 | bjohnson | Rith: however, that is not required .. just what I think you will want |
21:52.22 | dsmouse | ~seen jbot |
21:52.23 | jbot | jbot is currently on #ipaq (16h 33m 55s) #how (16h 33m 55s) #bzleague (16h 33m 55s) #storm (16h 33m 55s) #orkut (16h 33m 55s) #asterisk-doc (16h 33m 55s) #uphpu (16h 33m 55s) #va (16h 33m 55s) #asterisk (16h 33m 55s) #nslu2-linux (16h 33m 55s) #magnia (16h 33m 55s) #aegis (16h 33m 55s) #ol ... |
21:52.41 | tzanger | no wonder he's so slow |
21:53.24 | shmaltz | anybody using a cisco 79xx? |
21:53.35 | tzanger | shmaltz: not I |
21:53.41 | Rith | so just a few FXO cards on incoming POTS and that's it for specialized hardware? |
21:53.50 | shmaltz | tzanger, realy? |
21:54.01 | tzanger | I don't do SIP |
21:54.12 | shmaltz | tzanger, so what do you do? |
21:54.13 | tzanger | Rith: for what you just described, yes |
21:54.19 | tzanger | shmaltz: IAX2 all the way |
21:54.22 | Rith | alright, thanks |
21:54.28 | tzanger | I have a TDM430P at home for 3 analog phones |
21:54.36 | shmaltz | tzanger,what hard phones support IAX2? |
21:54.43 | tzanger | shmaltz: farfon does if you can find one |
21:54.55 | tzanger | shmaltz: there are a few others too based on some weird chipset but I've no experience with them |
21:55.04 | tzanger | PA1688 or something like that |
21:55.22 | shmaltz | why souldn't i be able to fine one farfon? tzanger. |
21:55.34 | tzanger | shmaltz: because wasim's still working on it |
21:55.34 | bjohnson | not available to the public yet |
21:55.46 | tzanger | citats is supposed to be the north american contact but he's been gone for a dog's age |
21:55.49 | tzanger | ~seen citats |
21:55.51 | jbot | citats <~james@duff.gnuinter.net> was last seen on IRC in channel #asterisk, 72d 22h 24m 47s ago, saying: 'and i gotta go back, so i'll catch ya'll later'. |
21:55.58 | bjohnson | haha |
21:56.00 | bjohnson | much later |
21:56.04 | tzanger | ok |
21:56.06 | tzanger | 10 dog's ages |
21:56.09 | shmaltz | so tzanger, meanwhile I can't talk to you unless i'm your girlfriend or using IRC? |
21:56.24 | tzanger | shmaltz: I just said I use a TDM430P and regular old analog phones |
21:56.26 | bjohnson | tzanger: <- needs to teach jbot how to count in dog ages |
21:56.46 | bjohnson | shmaltz: he has * server at home |
21:56.47 | tzanger | a panasonic cordless, a cheapo but pretty one in my bedroom and a PT450 in the kitchen |
21:57.20 | bjohnson | shmaltz: and a PCI card with fxs ports |
21:57.24 | Juggie | anyone know of a good call back script? like if the pri is full, on the next chan, call me then give me that channel, etc... or if someone is busy with no call waiting/vm then offer callback? |
21:57.40 | shmaltz | I understand but if he doens't use sip and only iax2 and iax is not available he has no phones (he didn't mention analog) |
21:57.41 | tzanger | Juggie: huh? |
21:57.49 | tzanger | shmaltz: yes I did |
21:57.57 | Juggie | eg, i make a call, the pri is full... |
21:57.58 | shmaltz | ok, giving up |
21:58.06 | plappy | okay, this makes no sense at all... Both of my clients can use the demo and dont have any issues, client to client through * gives me the "maximum retries exceeded..." error after a few seconds. is it possible its just not solid on FBSD yet? |
21:58.13 | Juggie | when theres a free channel, asterisk calls me back, then calls the number from before |
21:58.13 | tzanger | 17:02 < tzanger> shmaltz: I just said I use a TDM430P and regular old analog phones |
21:58.27 | tzanger | plappy: yes, very possible |
21:58.33 | Juggie | its one of our requirements here for work... i'm evaluating * against our RFP. |
21:58.38 | bjohnson | shmaltz: anyway .. people are using cisco 79xx phones |
21:58.42 | plappy | bugga me. |
21:58.42 | plappy | heh |
21:58.45 | bjohnson | but not tzanger |
21:58.53 | tzanger | bjohnson: precisely |
21:59.00 | bjohnson | and not me :( |
21:59.16 | shmaltz | I know bjohonson, and I'm trying to figure out what is the best way using the multiple lines, using multiple registrations? or just one registration? |
21:59.23 | tzanger | Juggie: I don't have a script handy but it shouldn't be too difficult |
21:59.26 | tzanger | asterisk has all the info |
21:59.41 | Juggie | yeah, but how to loop until theres a channel free |
21:59.45 | Juggie | thats the issue... |
21:59.45 | tzanger | shmaltz: bkw_ might have the low-down, he's a know cisco sympathizer. :-) |
21:59.51 | tzanger | Juggie: you don't |
21:59.55 | tzanger | you put it in a 5min cron job |
21:59.59 | tzanger | so it just checks every 5 min |
21:59.59 | shmaltz | bkw_, you around? |
22:00.00 | WGFreewill | shmaltz |
22:00.07 | WGFreewill | like the cisco multiple lines |
22:00.08 | tzanger | I do that with my voicemail callback |
22:00.09 | tzanger | s |
22:00.17 | Juggie | tzanger, the requirement is instant call back when the line is free |
22:00.17 | tzanger | since the callback as it's currently coded sucks |
22:00.23 | shmaltz | WGFeewill, yes |
22:00.30 | tzanger | Juggie: IIRC there is also a way to loop inside of asterisk |
22:00.33 | tzanger | in the dialplan |
22:00.37 | *** part/#asterisk ReVoK (ReVoK@82.224.60.46) |
22:00.38 | WGFreewill | if you are using the AMP packages |
22:00.41 | WGFreewill | dialparties.agi |
22:00.41 | tzanger | anyway |
22:00.45 | tzanger | I gotta go get the kids |
22:00.46 | tzanger | ttyl |
22:00.50 | roamer323 | anyone here from the tmc miami show? |
22:00.51 | shmaltz | WGFreewill, I'm not. |
22:01.03 | WGFreewill | eh good to have a look |
22:01.09 | WGFreewill | you can register all cisco lines |
22:01.12 | harryvv | anyone here running a spa1000 |
22:01.15 | WGFreewill | to one sip entry |
22:01.41 | WGFreewill | change the 0 in that agi |
22:01.42 | WGFreewill | <PROTECTED> |
22:01.57 | WGFreewill | to the number of lines you want to support system wide (unfortunately) |
22:02.08 | Mavvie | sometimes, when I call overseas, it looks like asterisk doesn't bridge two channels until I have really spoken with a loud voice. (this bridge is on two FXO cards) |
22:02.12 | shmaltz | I know, but I'm trying to figure out if this is realy the best way. Currently I'm using six different sip accounts, it is better b/c it allows one to pick up any line to make a call. |
22:02.15 | Mavvie | is that normal? |
22:02.19 | WGFreewill | and your snom and cisco phones will ring through each available line |
22:02.33 | WGFreewill | same with this |
22:02.59 | WGFreewill | MWI only comes on the first line |
22:03.33 | WGFreewill | one extension per phone |
22:03.38 | WGFreewill | one user / pass |
22:03.45 | WGFreewill | but maybe there is a better way |
22:04.08 | *** join/#asterisk file (~file@mctn1-8179.nb.aliant.net) |
22:04.24 | WGFreewill | if there is maybe we'll hear about it right now |
22:05.58 | shmaltz | Thanks, WGFreewill, I just tested it and it makes much more sense. |
22:06.16 | *** join/#asterisk eipi (eipi@136-218-114-200.fibertel.com.ar) |
22:06.36 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
22:06.57 | shmaltz | WOW: |
22:06.58 | shmaltz | http://www.surfsantamonica.com/ssm_site/the_lookout/news/News-2005/Feb-2005/02_23_05_Cell_Phone_Virus_Strikes_Santa_Monica.htm |
22:07.08 | *** join/#asterisk bsenicar (~bsenicar@BSN-77-155-238.dsl.siol.net) |
22:08.40 | shmaltz | Just wondering when the first Cisco phone virus will enter the internet |
22:08.49 | Blackvel | what isn't DISA accepting in asterisk 1.0.5 anymore the numbers after the dailin? that worked before with bristuff and asterisk 1.0.2? my analog pbx does not support blockmode, so I have to use DISA to send extensions behind the inward-number into asterisk |
22:08.51 | shmaltz | or maybe Polycom |
22:08.58 | terrapen | cisco phone virus, heh |
22:09.53 | WGFreewill | people are going to realize one day that talking on the phone sucks anyways |
22:10.03 | file | telepathy, that's where it's going |
22:10.03 | WGFreewill | the cisco virus will help us get there |
22:10.12 | terrapen | my girlfriend will never realize that. |
22:10.42 | Rith | ok, coming in from pots lines, we can use a wildcards + FX0 bundle into the PBX. coming out to VoIP phones, we just plug straight into NICs? and if so, can we put them through a switch for more ports, or do they need to go straight into each NIC port? |
22:11.05 | WGFreewill | the new voice communication platform will be ale at the pub |
22:11.06 | *** part/#asterisk gr0mit (~gr0mit@router1.txrx.org.uk) |
22:11.13 | WGFreewill | Rith: no switch the ethernet is fine |
22:11.19 | terrapen | i love that i can set my cisco handset off the hook |
22:11.20 | Mavvie | calltone detection is really a POS when you try it outside your progzone. |
22:11.22 | terrapen | and the phone stays on the hook |
22:11.23 | WGFreewill | Rith: just like always |
22:11.29 | Rith | alright, great, thanks |
22:11.32 | terrapen | i just wish there was a little button on the handset to pick up the hook |
22:12.34 | terrapen | need to figure out how you set the 7960 to do not disturb |
22:14.06 | Nugget | unplug it. :) |
22:14.27 | heison | ~seen Jerjer |
22:14.31 | jbot | jerjer <~JerJer@dsl-107-24.che.centurytel.net> was last seen on IRC in channel #asterisk, 9h 41m 6s ago, saying: 'don't run 1.0.2 or bristuff'. |
22:14.52 | heison | ~seen shido6 |
22:14.54 | jbot | shido6 is currently on #asterisk (3h 7m 8s). Has said a total of 15 messages. Is idling for 1h 4m 42s |
22:15.34 | terrapen | Asterisk CVS-v1-0-01/10/05-03:06:46 |
22:15.37 | terrapen | is that way old? |
22:15.55 | Nugget | 10-Jan-2005 was about a month and 10 days ago. |
22:16.01 | Nugget | was that a trick question? |
22:16.04 | terrapen | :P |
22:16.17 | terrapen | this day has turned out bad ass |
22:16.21 | terrapen | i should go outside and enjoy this |
22:16.25 | terrapen | im sick of being in this office |
22:16.31 | Mavvie | I don't want to do FSX/FXO anymore.... |
22:17.56 | Juggie | anyone have an example of a looping dialplan? one where if all the channels where busy on zap, i could call the user back and connect their call when one became available. |
22:18.09 | terrapen | much less stuck here coding a fucking shopping cart |
22:18.37 | harryvv | man... this is truely wierd. I have not been able to configure my ata because of a sinfigure mismatch and now after not touching it for 30 min it works? i have all valid info in sip show peer and have a dialtone. |
22:18.52 | harryvv | I was ready to pull my hair out. |
22:18.53 | *** join/#asterisk jero (~SFLphone@199.243.85.90) |
22:18.58 | jero | hello |
22:18.59 | terrapen | harry, its the gypsies |
22:19.56 | harryvv | terrapen, I dont know but I even simplified the authentication and that did not work. Gave up and did more searching on google to find out what was going on and subconciosly picked up the reciver. |
22:22.04 | harryvv | So what can ya say ;) |
22:25.33 | *** join/#asterisk xeet2 (~xeet2@es.jsci.net) |
22:26.43 | Essobi | Anyone know why I can't have a dynamic src port on a sip peer calling into me? |
22:27.17 | shmaltz | WGFreewill, I'm getting 403s in the log if I do it this way, what Can I do to avoid this? |
22:27.35 | Essobi | I got a cisco 5400 that's picking random ports and they won't land in a peer context, only a default context. |
22:27.53 | Trionnis | well there's your first problem |
22:27.55 | Essobi | due to the ports src ports changing every call |
22:27.56 | Trionnis | you're using a Cisco |
22:28.08 | Essobi | Uhh. Yea. |
22:28.25 | Trionnis | I'm just yankin' yer chain man ;) |
22:28.30 | Essobi | I worked fine in 1.0 but in -head it's broke. |
22:28.30 | WGFreewill | doing it using the dialparties.agi from the AMP package? |
22:28.34 | Trionnis | we've all had to put up with 'em |
22:28.37 | Trionnis | :) |
22:29.10 | Essobi | It's agravating me. |
22:29.30 | xeet2 | essobi: what protocol? |
22:29.59 | harryvv | Essobi well persistance pays off eventually. I am a bit relieved a little happier this problem this problem of mine is over. |
22:31.08 | *** part/#asterisk Grooby (~Grooby@12.22.232.212) |
22:31.35 | Trionnis | you could diff the sources for the sip stuff, and find out what was changed I suppose |
22:31.39 | Trionnis | that sounds like a PITA tho |
22:33.36 | *** join/#asterisk |Vulture| (~Vulture@109.238.204.68.cfl.res.rr.com) |
22:37.57 | jalsot | hi |
22:38.32 | jalsot | does anybody use HFC-PCI ISDN card in NT mode here? |
22:38.38 | km- | is there an application Dialtone()? |
22:39.13 | km- | ... |
22:40.30 | BrianR___ | Is there anything in the zaptel package I might use to figure out why I'm not getting caller id from my X100P? |
22:41.18 | km- | hmm |
22:41.20 | km- | I dont think so |
22:41.26 | tclark | clidtest |
22:41.38 | BrianR___ | clidtest? |
22:41.49 | *** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com) |
22:41.51 | BrianR___ | Which zaptel version has this program? |
22:41.54 | xeet2 | does the x100p have to wait for 2 rings to receive all the callerid info? or can it answer right away? |
22:42.10 | BrianR___ | xeet2: No idea. Asterisk is having it wait two rings. |
22:42.15 | tclark | hmm looks like from the digium ftp site |
22:42.41 | BrianR___ | The PBX I have it connected to actually sends the full callerid in two bursts before the first ring, then again .5 seconds after each ring. |
22:42.46 | tclark | hell is just a few line i could put it in pastebin.ca if ya want |
22:42.55 | BrianR___ | tclark: Ok. |
22:43.30 | BrianR___ | Aah. Found it. ftp://ftp.linux-support.net/pub/zaptel/clidtest.tgz |
22:43.44 | BrianR___ | hmm.. No.. It's not there anymore.. :( |
22:43.54 | |Vulture| | What is a good LD rate on a PRI? |
22:44.03 | tclark | http://pastebin.ca/6348 |
22:44.07 | |Vulture| | all the places keep quoting me ~5cents |
22:44.13 | shido6 | how do u make a single module instead of make modules? |
22:44.56 | Essobi | this is pissing me off |
22:44.57 | xeet2 | vulture: if you're looking at a pri for just long distance, you would be better off getting a t1 to a good isp and using voip |
22:45.08 | Essobi | no matter what I do.. my sip peer lands in [default] |
22:45.13 | |Vulture| | xeet2: oky |
22:46.10 | WGFreewill | Essobi |
22:46.18 | WGFreewill | you need to match |
22:46.20 | WGFreewill | IP |
22:46.21 | WGFreewill | and port |
22:46.26 | WGFreewill | check sip debugs |
22:47.51 | xeet2 | vulture: I only ever use a pri or a voice t1 from a lec for local calls and inbound |
22:47.52 | BrianR___ | clidtest exits right away with "Error getting Caller*ID..." |
22:47.55 | *** join/#asterisk zno (~zeno@ip-160-79-174-101.autorev.intellispace.net) |
22:48.06 | shmaltz | WGFreewill, I'm getting my logs full with "Registration From ..... Failed for .... |
22:48.45 | tclark | think you are supposed to run it w/o * up |
22:49.16 | BrianR___ | same deal |
22:49.25 | BrianR___ | as soon as the line rings it exits... |
22:51.20 | |Vulture| | xeet2: oky thanx |
22:51.36 | |Vulture| | xeet2: on a PRI you can get DIDs in different areacodes other than yours.. right? |
22:52.30 | xeet2 | vulture: yes, depending on your provider |
22:53.06 | |Vulture| | xeet2: oky thanx |
22:53.12 | km- | man this problem just evades capture all the time |
22:53.14 | xeet2 | surprisingly intermedia/allegiance/whatever other name they want seem to be pretty good about that in the US |
22:53.33 | xeet2 | we're in baltimore and we have dids from every major city in the us, and local calling to them too |
22:53.44 | |Vulture| | Any comments about Xspedius? |
22:54.27 | |Vulture| | xeet2: how much do you pay for a PRI with multiple local calling areas? |
22:55.06 | xeet2 | we pay a set fee for each lata we have access too, I don't remember exactly what that fee is |
22:56.32 | xeet2 | and for anything else outside of those areas we send the calls to a number of sip and iax providers |
22:56.39 | |Vulture| | xeet2: which provider do you use? |
22:57.15 | xeet2 | vulture: for what? voip? quite a few, no single one meets all our requirements in all areas |
22:57.16 | km- | ah |
22:57.20 | km- | so that what prewink does |
22:58.09 | tzanger | km-: what's it do |
22:58.22 | km- | tzanger: if I kick everything up to 3000 |
22:58.32 | km- | tzanger: the t1 takes 3 seconds to realize the phone is off hook |
22:58.42 | km- | doesn't change anything about how long I have to hit digits |
22:59.00 | km- | i.e., as soon as I hit 1 |
22:59.05 | km- | it's there |
22:59.16 | xeet2 | hmmmm |
23:00.50 | tzanger | km-: |
23:00.51 | tzanger | ; emdigitwait: Time to wait for DID digits on E&M links (default 250ms) (Increase to 500 |
23:00.55 | tzanger | ; or so if you are not getting all DID digits on your E&M link) |
23:01.03 | tzanger | set that to 3000 |
23:01.19 | tzanger | and set it back to 250 on the telco-facing T1 |
23:01.44 | km- | damn people trying to call at 6oclock |
23:01.44 | km- | I did |
23:01.47 | BrianR___ | hmm.. this is going to be difficult to debug.. a google search for my error finds only one other person and they didn't get it resolved either :( |
23:01.56 | km- | emdigitwait? |
23:01.58 | km- | I didnt see that |
23:02.33 | km- | no change |
23:02.38 | tzanger | ...?? |
23:02.43 | tzanger | that seems odd |
23:02.49 | tzanger | I would have thought that did it |
23:02.52 | km- | emdigitwait=3000 does not change the digit wait |
23:02.56 | tzanger | that was in CVS HEAD, btw, not sure if that's in stable |
23:02.57 | km- | is that a feature only in cvs? |
23:03.01 | km- | I downloaded 1.0.5 |
23:03.11 | km- | eh, lemme pull cvs |
23:03.30 | km- | because I obviously love running HEAD in a full-on production environment |
23:03.34 | km- | hehee |
23:03.45 | tzanger | km-: look around line 156 in channels/chan_zap.c and see if it's there |
23:04.05 | km- | just a min, letting apt-get finish installing \cvs |
23:04.11 | tzanger | km-: -HEAD is often perfectly stable |
23:04.17 | km- | I know |
23:04.21 | km- | I used to run HEAD before |
23:04.25 | km- | but it's just the idea of it |
23:04.32 | tzanger | km-: pish tosh |
23:04.33 | km- | I'm forced to think like a narrowsighted IT admin :) |
23:05.06 | km- | line 156 of chan_zap.c is a #define |
23:05.15 | tzanger | look around there |
23:05.17 | km- | #define DCHAN_UP (1 << 2) |
23:05.17 | km- | #define DCHAN_AVAILABLE (DCHAN_PROVISIONED | DCHAN_NOTINALARM | DCHAN_UP) |
23:05.17 | km- | #define zt_close(fd) if(fd > 0) close(fd); |
23:05.18 | tzanger | or just search for emwinktime |
23:05.45 | km- | negative on emwinktime, emdigittime |
23:05.50 | tzanger | good |
23:06.41 | tzanger | I think extensions.conf sould look like httpd.conf |
23:06.48 | tzanger | or hell everything like that |
23:06.51 | tzanger | I really like that setup |
23:07.02 | tzanger | easily machine-parsed too |
23:07.26 | terrapen | once you get a hang of extensions.conf, its not hard at all |
23:07.35 | tzanger | terrapen: i agree |
23:07.40 | tzanger | I'm just saying I like how apache.conf is |
23:07.44 | tzanger | er httpd.conf |
23:07.44 | Beirdo | I think it should all be in MySQL/Postgres :) |
23:07.45 | Beirdo | hehe |
23:07.49 | terrapen | oh lord no |
23:07.51 | tzanger | Beirdo: that too |
23:07.55 | tzanger | bt that's a little overkill |
23:08.04 | Beirdo | meh |
23:08.05 | terrapen | and when i was starting with asterisk, i thought it should be in Apache config format too |
23:08.06 | tzanger | so long as the DB is allowed to cache results |
23:08.09 | terrapen | but now that i understand it |
23:08.11 | tzanger | and then everything is in realtime |
23:08.13 | terrapen | i like it just the way it is |
23:08.17 | tzanger | if you need to update several things, use a transaction |
23:08.20 | Nugget | I like it just the way it is. |
23:08.22 | Beirdo | it works good the way it is |
23:08.29 | terrapen | fuck databases |
23:08.30 | Nugget | extensions.conf is a scripting language, not a data store. |
23:08.32 | Beirdo | but it would be nice in SQL :) |
23:08.37 | km- | its just like the old tired debate about rewriting asterisk in C++ |
23:08.39 | terrapen | you should not need a DB to configure a PBX |
23:08.40 | km- | it works fine the way it is |
23:08.45 | km- | no point in reinventing the wheel |
23:08.49 | Beirdo | terrapen: true |
23:08.55 | terrapen | DBs are so overused |
23:09.02 | Beirdo | be nice to have it as an option though |
23:09.13 | tzanger | what advantage would a rewrite have? |
23:09.19 | tzanger | terrapen: but for a phone system it makes a lot of sense |
23:09.23 | Beirdo | which it seems it soon will be with Asterisk RealTime |
23:09.27 | tzanger | especially for iax.conf, sip.conf and extensions.conf |
23:09.29 | terrapen | you only need a DB if a) you are storing massive amounts of information and need to access a small peice quickly and b) you need to run specialized queries against that database |
23:09.32 | terrapen | err data |
23:09.40 | terrapen | tzanger, no, it doesn't |
23:09.43 | terrapen | it does not make sense |
23:09.57 | Beirdo | if you say so :) |
23:10.00 | terrapen | a phone system is a critical peice of infrastructure |
23:10.03 | Nugget | it makes more sense for iax.conf and sip.conf, but I don't see how it makes much sense at all for extensions.conf |
23:10.11 | terrapen | complexity decreases stability |
23:10.13 | Beirdo | fair enough |
23:10.17 | tzanger | terrapen: true |
23:10.17 | Nugget | the benefits sure wouldn't outweigh the disadvantages |
23:10.18 | terrapen | and adding a DB increases complexity |
23:10.23 | Beirdo | extensions.conf is good as it is |
23:10.35 | terrapen | and the DB bits are there, if you want them |
23:10.45 | Beirdo | iax.conf, sip.conf, and especially voicemail.conf. nice to have in DB |
23:10.48 | terrapen | but they should not be required, as most people do not (and should not) use them |
23:10.51 | Nugget | fundamentally, extensions.conf is not a database. to make it fit in a database would require compromises to functionality. |
23:11.03 | km- | yeah |
23:11.04 | Beirdo | agreed |
23:11.06 | tzanger | I've changed my mind about extensions.conf |
23:11.07 | km- | extensions.conf is fine the way it is |
23:11.08 | tzanger | you're right |
23:11.13 | km- | hahaha |
23:11.17 | Beirdo | or a complete rewrite, and that would be silly |
23:11.19 | km- | "I dont want to have this conversation anymore!" |
23:11.21 | tzanger | just need to get the "reload" command working better |
23:11.23 | Beirdo | :) |
23:11.24 | terrapen | beirdo, any novice perl programmer can generate any one of those config files based on information pulled from a database |
23:11.31 | tzanger | it has on more than one occassion hung the box for a few seonds |
23:11.41 | *** join/#asterisk svantuil (~svantuil@054.209-89-66-0.interbaun.com) |
23:11.45 | km- | these mp3's need to get the hell out of CVS |
23:11.52 | Beirdo | hehe |
23:11.55 | tzanger | km-: or get moved to asterisk-sounds with the rest of 'em |
23:11.59 | dsmouse | but why isn't it called "dialplan.conf"? |
23:12.11 | *** part/#asterisk MicH323 (~micosat@host217-44-194-118.range217-44.btcentralplus.com) |
23:12.13 | Beirdo | because it isn't |
23:12.15 | tzanger | dsmouse: why is 'signaling' spelled wrong in zapata.conf |
23:12.25 | dsmouse | ahhh |
23:12.44 | dsmouse | tzanger: it's a industry term |
23:12.46 | dsmouse | SHHHH |
23:12.52 | tzanger | heh |
23:13.19 | terrapen | "to keep the noobs away" |
23:14.52 | *** join/#asterisk agave-txlink (phanop@216.81.43.75) |
23:15.01 | dsmouse | terracon: it didn't work. I was a noob sunday a weekago |
23:16.11 | *** join/#asterisk buddah (~hnic@67.110.253.129) |
23:16.11 | svantuil | noob question: Is there any decent softphones available for XP that will interact directly with asterisk? My boss is pro MS, and pro opensource (oxymoron). |
23:16.12 | *** part/#asterisk bsenicar (~bsenicar@BSN-77-155-238.dsl.siol.net) |
23:16.40 | dsmouse | svantuil: almost all of them talk SIP, which can work with asterick |
23:16.49 | zno | svantuil: avoid softphones altogether |
23:16.51 | km- | mmm one of the ladies at work bought a box of cow tails |
23:16.51 | km- | you know, those caramel candies with cream in the middle |
23:16.51 | km- | dinner is served! |
23:16.58 | buddah | question, is there anyway to assign 2 DIDs to 1 sip phone, and manipulate the sip configuration so that when either # is called it rings the same phone? |
23:17.02 | dsmouse | be wary of NAT with with sip too |
23:17.07 | buddah | or would i have to get a 2 port ATA for that? |
23:17.21 | terrapen | goddamn you, internet explorer |
23:17.27 | *** join/#asterisk file (~file@mctn1-8179.nb.aliant.net) |
23:17.28 | dsmouse | buddah: that should be easy |
23:17.29 | terrapen | i hate coding around your bugs |
23:17.32 | buddah | i figured |
23:17.38 | km- | hahahaha |
23:17.40 | dca[laptop] | buddah: the DID provider should be able to send both to the same device |
23:17.42 | km- | terrapen: amen to that |
23:17.53 | km- | terrapen: you should try programming for Pocket PC's, they're a hoot. |
23:17.54 | dsmouse | buddah: when you have a spot for the extension that recives the line, make it dial the phone's extension |
23:17.59 | dca[laptop] | buddah: should be invisible to you |
23:18.03 | buddah | so like say 5142718929,1,DIAL(SIP/9378322738) to get the 514 to dial same phone as the 937? |
23:18.10 | km- | terrapen: I set the z-order through a wm_message and the device went nuts |
23:18.33 | greg_work | whats the proper dial pattern for an area code? ZXX ? |
23:18.41 | terrapen | its hard to make things look nice in IE and make them work and do it without JavaScript |
23:18.43 | km- | NXX? |
23:18.51 | buddah | think its N |
23:19.27 | terrapen | ok im bored |
23:19.31 | terrapen | im going downstairs |
23:19.36 | km- | asterisk compiles slowly on a P2 266 |
23:19.45 | greg_work | yeah i guess N would make more sense. thanks |
23:19.49 | km- | amazingly though, it runs pretty snappy on the box |
23:19.49 | buddah | ok, next question. my boss does the DID thing, like he orders them and what not, and i dont know how providers work, when he orders another batch, can he get like 1 in 937 and 1 in 514, or do they usually make you get blocks? |
23:19.54 | dsmouse | terracon: don't type and walk down stairs as the same time... |
23:19.55 | svantuil | zno: it's just for testing anyways. So I can convince him to let me buy some digium hw and interface (attempt) to our analog ksu. |
23:19.58 | *** join/#asterisk r1_ (~erwan@www.thiscow.com) |
23:20.07 | Juggie | its N, Z includes 1 |
23:20.27 | buddah | then it should be Z, some area codes are 1xx right? |
23:20.32 | km- | are there any good manager gui apps available to the open source community? |
23:20.32 | buddah | some east coast |
23:20.33 | buddah | or no? |
23:20.35 | km- | buddah: negative |
23:20.37 | buddah | oh |
23:20.42 | Juggie | no, because long distance starts in 1. |
23:20.42 | km- | buddah: never heard of a 1xx areacode |
23:20.44 | buddah | err i'm thinking of zip codes sorry |
23:20.56 | dsmouse | mmmm |
23:21.00 | dsmouse | dialing a zip code. |
23:21.01 | km- | buddah: yeah, 1xxxx is in PA (zip code) |
23:21.05 | buddah | yup |
23:21.10 | km- | sounds like a rainbow box to me |
23:21.12 | buddah | 15044 was mine in pittsburgh |
23:21.14 | stevekstevek | huh? |
23:21.17 | buddah | thats why i was thiking about that |
23:21.22 | tzanger | km-: did it work? |
23:21.23 | km- | hehe, I live in philadelphia |
23:21.25 | stevekstevek | 10001 is New York City.. |
23:21.30 | buddah | so is 10101 |
23:21.37 | buddah | brooklyn i believe |
23:21.38 | km- | tzanger: asterisk isn't even close to finished compiling yet |
23:21.40 | buddah | or part of |
23:21.40 | stevekstevek | I think all of 10xxx is manhattan. |
23:21.44 | km- | tzanger: I'm running asterisk on a p2 266 |
23:21.45 | tzanger | km-: heh |
23:21.47 | tzanger | no worries |
23:21.52 | tzanger | I ran it on a P90 (no MMX) |
23:21.57 | km- | only to indications.c |
23:21.58 | stevekstevek | 112 = brooklyn, etc. PA doesn't get all of 1XXXX :) |
23:22.03 | *** join/#asterisk lyroy (~lyroy@modemcable117.123-202-24.mc.videotron.ca) |
23:22.03 | buddah | yeah |
23:22.11 | km- | ok, I know 17xxx 18xxx and 19xxx are in PA |
23:22.17 | km- | so I stand corrected |
23:22.20 | buddah | isnt some ny 0xxxx? or is that NJ? |
23:22.29 | dsmouse | maine? |
23:22.30 | km- | NJ has some 0xxxx |
23:22.30 | buddah | 15xxx is PA |
23:23.04 | tzanger | 17xxx? |
23:23.12 | tzanger | oh zip codes |
23:23.17 | buddah | yeah |
23:23.18 | tzanger | 15xxx is PA too |
23:23.20 | buddah | yup |
23:23.24 | buddah | pittsburgh area |
23:23.26 | Beirdo | pah |
23:23.26 | buddah | or north of |
23:23.27 | tzanger | yup |
23:23.29 | Beirdo | :) |
23:23.33 | buddah | i miss pittsburgh |
23:23.34 | tzanger | glenshaw, etc |
23:23.36 | lyroy | I'm in montreal and I want to know if there is any 514 or 450 DID provider, does someone can help me? |
23:23.37 | buddah | yeah |
23:23.45 | buddah | my dad worked in glenshaw |
23:23.54 | buddah | lyroy: lemme know if you find any, i need a 514 |
23:24.08 | buddah | i feel stupid looking for just 1 DID |
23:24.18 | tzanger | buddah: our parent company's right on rte 8 |
23:24.22 | buddah | nice |
23:24.29 | buddah | i lived right off rte 8 in gibsonia |
23:24.32 | Beirdo | sixtel maybe? |
23:24.33 | lyroy | well I'm looking for a couples of 514 DID |
23:24.34 | Beirdo | iax.cc |
23:24.40 | Beirdo | they might offer something |
23:24.54 | lyroy | alright thanx ill check this out |
23:25.03 | Beirdo | they might also suck rocks, but worth looking at |
23:25.08 | mikegrb | lyroy: check the asterisk-biz mailing list |
23:25.10 | buddah | do profiders typically allow a request of like just 1 DID? |
23:25.19 | Beirdo | there's a Canadian VOIP providers page on the wiki |
23:25.19 | buddah | or is there a min.? |
23:25.21 | tzanger | they're not the greatest but they're not bad |
23:25.24 | Beirdo | min 1 :) |
23:25.42 | buddah | awesome |
23:25.45 | Beirdo | for VOIP providers, I'd expect 1 is a common request |
23:26.19 | buddah | ok, now i dont feel bad |
23:26.25 | km- | we buy our DID's in blocks of 20, but we do have some 1-off numbers which lead me to believe you can get them one at a time |
23:26.28 | Nugget | like good. |
23:26.37 | buddah | i need one for montreal, and one for dayton, so people can call me without long distance |
23:26.39 | *** join/#asterisk hcclNoodles (~hcclnoodl@hcclnoodles.plus.com) |
23:26.42 | Beirdo | for any home geek users, 1 would be the norm |
23:26.50 | buddah | figured i might as well since the boss is paying for my usage |
23:26.52 | tzanger | VOIP providers allow onesie-twosies |
23:26.57 | tzanger | telcos generally sell in blocks |
23:27.02 | tzanger | Bell Canada, for example, sells in blocks of 30 |
23:27.13 | Beirdo | Bell Canada sucks :) |
23:27.14 | Blackvel | what place in asterisk code (zaptel, libpri) can I change, so DISA picks up some extensions faster? |
23:27.24 | km- | tzanger: chan_oss |
23:27.32 | km- | tzanger: hehe, this thing sure does cook! woo doggie! |
23:28.00 | tzanger | Blackvel: what do you mean |
23:28.22 | lyroy | you can buy DID from Bell Canada? |
23:28.26 | km- | so, my question remains, anyone know of a good graphical manager interface? One that preferably works in windows? |
23:28.53 | Blackvel | tzanger: i mean i type this in my analog pbx telephone: #91309110000 |
23:28.58 | tzanger | lyroy: if you have a PRI, yes |
23:29.04 | Blackvel | and asterisk does not get 9110000 as extension |
23:29.05 | tzanger | km-: not offhand no |
23:29.10 | Blackvel | but that had worked some time ago |
23:29.25 | Blackvel | BUT, if I call #9130 |
23:29.35 | Blackvel | DISA picks up, gives me dail tone, and I can dail 9110000 |
23:29.45 | tzanger | Blackvel: interesting |
23:29.50 | tzanger | brain's not functioning well enough to help yo uhere though |
23:30.00 | Blackvel | well its not blockmode, but I used DISA all the time to get that #91309110000 scenario working |
23:30.01 | Blackvel | hehehe |
23:30.04 | Blackvel | neither mine :( |
23:30.13 | km- | ut oh |
23:30.16 | km- | pbx_dundi died |
23:30.19 | Beirdo | why would you want a number starting with 911? |
23:30.19 | km- | ah I need to get libzlip |
23:30.39 | Blackvel | but asterisk 1.0.2 and 1.0.5 app_disa.c has not changed, so its some zaphfc code |
23:30.49 | Blackvel | _91 is sipgate, _92 nikotel, etc |
23:30.58 | Beirdo | bad idea |
23:31.04 | tzanger | yeah I agree |
23:31.14 | Beirdo | don't use something that will start with 911 |
23:31.15 | km- | so, can somoene explain in 100 words or less what DUNDi is, and what one might use it for? |
23:31.22 | Beirdo | not in North America |
23:31.26 | Blackvel | hehe |
23:31.28 | Beirdo | that's just asking for trouble later |
23:31.29 | Blackvel | i am in germany |
23:31.31 | Blackvel | i dont care |
23:31.38 | Blackvel | its my home system |
23:31.39 | tzanger | :-) |
23:31.40 | Beirdo | Ah, don't use 112 then :) |
23:31.45 | Blackvel | hehe |
23:31.49 | drumkilla | km-: check out the whitepaper on dundi.com |
23:31.51 | mutilator | my phone is like instant dial soon as 911 is hit dtmf timeout is like 2ms or something |
23:32.11 | Blackvel | well this * is not connected to pstn, its parallel to my pbx |
23:32.12 | Beirdo | I'd still suggest avoiding 911 |
23:32.15 | *** join/#asterisk cbachman (~cbachman@129.105.7.250) |
23:32.26 | Blackvel | tell me why i cant dail in, thats more important! :) |
23:32.28 | Beirdo | it could well have special code somewhere that you aren't expecting |
23:32.41 | Beirdo | there's likely a reason |
23:32.43 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
23:32.46 | Beirdo | anyways, home time |
23:32.50 | km- | ooh, WAMI |
23:32.55 | Blackvel | yeah, it is, new bristuff version :( |
23:33.42 | buddah | good lord. 2.50 a month and 1.1c a min |
23:33.44 | buddah | and thats in canadian |
23:33.47 | buddah | thats like nothing in US |
23:34.00 | *** join/#asterisk grailink (~grailink@adsl-66-143-140-135.dsl.stlsmo.swbell.net) |
23:34.03 | km- | I need to get some asterisk stickers for this box |
23:34.04 | km- | hehe |
23:34.16 | grailink | hey guys i have a quick question... |
23:34.20 | Blackvel | i am off |
23:34.21 | Blackvel | cu |
23:34.45 | grailink | with meetme the ztdummy is causing serious pausing/delay issues... is there anything that I need to adjust to fix this? |
23:34.47 | Juggie | anyone know of any free SMS services? |
23:35.09 | grailink | anyone? anyone? |
23:35.11 | grailink | :) |
23:35.17 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
23:35.32 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
23:35.40 | Grooby | anyone here w/ experience setting up speex? |
23:35.42 | buddah | anyone know a provider with cheap 937 (ohio) did's? |
23:35.46 | tzanger | ok I'm an idiot |
23:35.56 | tzanger | I figured it'd be time-saving to skip the 'l' diskset in slackware |
23:35.59 | km- | tzanger: hmm? |
23:36.06 | grailink | anyone here have meetme working? |
23:36.07 | tzanger | of course that includes skipping all the development libs |
23:36.11 | tzanger | *rolls eyes* |
23:36.12 | km- | oops |
23:36.16 | Trionnis | buddah: check www.myphonecompany.com... they look kinda low rent, but they have a ton of did's for 4.95/mo |
23:36.21 | buddah | k |
23:36.29 | km- | oooh codecs |
23:36.30 | Trionnis | use level3 for dids |
23:36.31 | km- | we're almost there! |
23:36.35 | Trionnis | so they have a lot |
23:36.58 | tzanger | now to see if I can remember how to do globbing iwth wget so I can get *.tgz in the /l/ directory |
23:37.06 | Grooby | i compiled speex from speex.org |
23:37.08 | *** join/#asterisk MicH323 (~micosat@host217-44-194-118.range217-44.btcentralplus.com) |
23:37.09 | Grooby | and recompile my asterisk |
23:37.27 | Grooby | but it's crashing when trying to load codec_speex.so |
23:37.29 | wankel | level3 is great. they just don't talk to you for under like $15k/mo :P |
23:37.37 | hermie | anybody here ever lease dark fiber |
23:37.37 | MicH323 | Anyone using Ast* with BroadVoice? |
23:37.39 | Nugget | Trionnis: is there a voip termination firm that doesn't look low rent? :) |
23:37.40 | Trionnis | anyone have some real-life asterisk + broadvoice stories they'd like to share? thinking about a did from them, but I'd like to hear from the guys that use it :) |
23:37.48 | Trionnis | lol @ Nugget |
23:37.51 | Grooby | Mich, I use * w/ BV |
23:37.52 | Trionnis | this is true |
23:38.37 | km- | trionnnis: someone earlier was saying broadvoice was the only company they ever complained about |
23:38.46 | *** join/#asterisk GodThor (~ninja@212.110.67.6) |
23:38.48 | MicH323 | I am having registration problems, tried BroadVoice Support mages. but get message -- Called 17182500199@sip.broadvoice.com |
23:38.48 | MicH323 | <PROTECTED> |
23:38.48 | MicH323 | <PROTECTED> |
23:38.48 | MicH323 | <PROTECTED> |
23:38.55 | Trionnis | I've heard their support really just doesn't exists |
23:38.59 | Trionnis | er |
23:39.00 | Trionnis | s |
23:39.03 | Trionnis | -s |
23:39.09 | Grooby | mich, change your host to sip.broadvoice.com |
23:39.14 | Grooby | don't use proxy.dca.broadvoice.com |
23:39.17 | Grooby | see if that helps |
23:39.21 | MicH323 | ok |
23:39.33 | Grooby | no one has problem with speex huh? |
23:39.44 | loud | suffered those broadvoice problems last night .. |
23:39.55 | km- | almost done! yay |
23:39.57 | grailink | are the any alternatives to meetme for conferences that work well? |
23:39.58 | Trionnis | well, I'm mostly just wanting them for EU calls |
23:40.04 | km- | ok, moment of truth |
23:40.07 | grailink | that don't need the zaptel timer |
23:40.14 | Poincare | hmmm, there was a page somewhere with the 'costs' for transcoding from one to another codec... anyone knows where to look for it? |
23:40.16 | Trionnis | I call .de a lot, so 20/mo ain't bad really |
23:40.21 | loud | Trionnis, only g711 |
23:40.25 | Trionnis | ewww |
23:40.27 | loud | yeah, i do .ar and .br all the time |
23:40.27 | Trionnis | :( |
23:41.28 | tzanger | myphonecompany.com doesn't have any canadian DIDs, at least that's my impression since you can't select a province |
23:41.29 | grailink | poincare: i saw a page on the wiki that mentioned that transcoding is pretty expensive. do a search on it and it should come up. i'm not sure if it came up with specifics |
23:41.41 | Trionnis | they might not tz |
23:41.42 | tzanger | grailink: depends on the codecs |
23:41.46 | Trionnis | he asked for ohio tho |
23:41.47 | Trionnis | ;) |
23:41.50 | tzanger | oh |
23:41.54 | tzanger | I thought he was looking for 514 |
23:41.56 | tzanger | which is Montreal |
23:42.06 | grailink | tzanger: gsm |
23:42.12 | Trionnis | <buddah> anyone know a provider with cheap 937 (ohio) did's? |
23:42.15 | Trionnis | ;) |
23:42.21 | km- | tzanger: that did it! YAY |
23:42.25 | tzanger | gsm is pretty cheap |
23:42.28 | tzanger | km-: excellent :-) |
23:42.52 | MicH323 | On the broadvoice. sip.broadvoice.com did the trick!!! Many Thans |
23:43.00 | Grooby | yup yup |
23:43.03 | km- | tzanger has saved my voip project! |
23:43.05 | GodThor | who knows hot to installed h323 on asterisk |
23:43.08 | km- | all hail tzanger! |
23:43.10 | tzanger | km-: hahaha |
23:43.14 | mishehu | gsm is blah. |
23:43.23 | tzanger | mishehu: I disagree, it sounds *great* to me |
23:43.29 | grailink | tzanger: i need a dynamic conference room that you can setup by calling an xtension. i got it working up till the point where the conf starts and now it just stutters like an idiot. I think its the zaptel timer ztdummy |
23:43.33 | tzanger | mishehu: I want to like ilbc but every time I use it I get complaints about the quality |
23:43.34 | km- | tzanger: dude, ya need some back patting for that :) |
23:43.39 | grailink | but I can't find an alternative thats worthadamn |
23:43.41 | Trionnis | what stutters? |
23:43.48 | *** join/#asterisk [Outcast] (~knoppix@h0006259a2649.ne.client2.attbi.com) |
23:43.48 | hcclNoodles | hi there, new to the channel, is there anybody from digium on here, i have a question re TDM400p in the UK |
23:43.50 | Trionnis | I'm using meetme with ztdummy |
23:43.52 | Trionnis | works ok |
23:44.01 | tzanger | grailink: it could be -- grab yourself an x101P or a clone and try it |
23:44.10 | Trionnis | 2.6.10 kernel |
23:44.15 | tzanger | km-: hey I am just glad it is working for you :-) |
23:44.21 | MicH323 | I will be attempting to compile h323 on Asterisk tonite :) |
23:44.34 | grailink | trionnis: what hardwaer? |
23:44.38 | tzanger | when I get it in my head that something should work, I generally find a way to make it work. This time it was very easy :-) |
23:44.42 | Trionnis | ;) |
23:44.42 | km- | YES! |
23:44.43 | MicH323 | Need to sort out the libs!# |
23:44.45 | km- | that is so awesome |
23:44.51 | Grooby | ok...i compiled speex and make clean; make; make install asterisk |
23:44.53 | zapa | thanks to all |
23:44.55 | Trionnis | grailink: ?? |
23:44.57 | Trionnis | no hardware |
23:45.08 | Trionnis | 2.6 kernel has a high-res timer built in |
23:45.10 | grailink | the hardware you're running asterisk on |
23:45.13 | Trionnis | ah |
23:45.21 | Grooby | now i am getting [codec_speex.so] ouch: error while writing audio data:: broken pipe |
23:45.25 | Trionnis | Athlon 2400xp, 512mb |
23:45.28 | tzanger | km-: now install the patches from bug 2532 |
23:45.33 | Grooby | can someone point me to the right place to start debugging? |
23:45.40 | tzanger | give them the highly experimental (but very stable and VERY GOOD) jitter buffer |
23:45.53 | tzanger | I've been running it for the last two or three weeks |
23:45.59 | tzanger | and switch-3.nufone.net is running it too |
23:46.05 | tzanger | so you can have end-to-end groovy jitter buffer |
23:46.14 | grailink | hmm... this thing should work. well i'm running 2.6.1 and when I try to use meetme without ztdummy it gives me the /proc error but with it it stutters. |
23:46.15 | tzanger | IAX2 only at the moment but zoa's getting it to work on SIP |
23:46.23 | *** join/#asterisk paulc (paulc@S010600062586a0b4.vc.shawcable.net) |
23:46.25 | tzanger | "/proc error" ?? |
23:46.29 | shmaltz | anybody here using a polycom phone? |
23:46.39 | *** join/#asterisk eivindtr (~Eivind@062016241059.customer.alfanett.no) |
23:46.44 | tzanger | shmaltz: no, but I hear they're very good |
23:46.44 | Trionnis | did you do "make linux26" when you compiled ztdummy? |
23:46.58 | Trionnis | that's what hooks it into the kernel timer instead of the usb one |
23:47.04 | hcclNoodles | anybody regarding TDM400P ???? |
23:47.10 | shmaltz | wow, tzanger. Do the polycom's support IAX2? :) |
23:47.12 | tzanger | hcclNoodles: what's up |
23:47.15 | tzanger | shmaltz: no :-) |
23:47.27 | grailink | correction Unable to open '/dev/zap/pseudo': No such device |
23:47.37 | Trionnis | insmod ztdummy |
23:47.39 | Trionnis | er |
23:47.42 | Trionnis | modprobe, rather |
23:47.50 | *** join/#asterisk guugmember (~nachoramo@168.234.226.39) |
23:48.01 | guugmember | hello, can we edit the code that writes the CDR´s so we can save a new field? after creating it in the table in mysql of course |
23:48.17 | guugmember | if yes, where? |
23:48.20 | mishehu | guugmember: you got the source... |
23:48.22 | grailink | Unable to open pseudo channel for timing... Sound may be choppy. |
23:48.23 | grailink | what's that? |
23:48.23 | Trionnis | of course you can, it's open source |
23:48.24 | km- | ugh, that sucks |
23:48.26 | *** join/#asterisk Ahewes (~rsb@adsl-69-107-39-45.dsl.pltn13.pacbell.net) |
23:48.30 | Trionnis | ./fanboy |
23:48.33 | Trionnis | ;) |
23:48.41 | km- | tzanger: I can't transfer phone calls across the two pbx's |
23:48.42 | grailink | i did that and this is the new error: Unable to open pseudo channel for timing... Sound may be choppy. |
23:48.47 | grailink | insmod that is |
23:48.47 | Trionnis | ok |
23:48.53 | hcclNoodles | in the uk , the card doesnt detect the british telecom hangup tone, and as such when in IVR /voicemail (when the line is not quiet) the card does not end the call |
23:49.10 | Trionnis | grailink: take this to pm? easier to keep track of |
23:49.21 | grailink | sure |
23:49.46 | km- | small price to pay I guess |
23:50.10 | hcclNoodles | everybody in the UK with this card is getting this apparantly, and we dont know what to do, digium support have suggested we try a few settings but they do not work, |
23:50.10 | tzanger | km-: could you before? |
23:50.21 | km- | tzanger: I hadn't gotten far enough to test it |
23:50.37 | km- | tzanger: I can still transfer intra-legacy-pbx, but I can't transfer from legacy pbx out to asterisk |
23:50.40 | tzanger | hcclNoodles: can you get disconnect supervision on UK phone lines? i.e. battery drop or battery reversal? |
23:50.42 | km- | I can transfer, naturally, from asterisk into the pbx |
23:50.54 | tzanger | km-: well no, the PBX sees * as trunk lines |
23:50.56 | |Vulture| | Anyone know of a Firwall/Router for ~$500 that has an ISP failover feature? |
23:51.01 | tzanger | km-: and it's absurd to transfer a call to a trunk line |
23:51.05 | guugmember | Trionnis, so I can, what file does that? |
23:51.12 | km- | kinda sucks because I can't park people on asterisk from the pbx |
23:51.21 | km- | I bet I could come up with a zapbarge app or something |
23:51.25 | Trionnis | haven't a clue |
23:51.26 | km- | there's a ZapRedirect type thing? |
23:51.29 | Grooby | :( |
23:51.32 | guugmember | km-, what are the two pbxs? both *? |
23:51.39 | Grooby | no one have any experience with speex?!?! |
23:51.45 | hcclNoodles | i dont know, can you elaborate tzanger |
23:51.50 | tzanger | km-: yeah -- that's one of the reasons I was looking at the ADPs for the NEC |
23:51.54 | Trionnis | find the code that handles CDR and start digging |
23:52.00 | tzanger | km-: it's also why I used PRI -- I *can* do that to a limited degree with the norstar |
23:52.00 | Trionnis | only thing I know to tell ya |
23:52.02 | km- | guugmember: one's a NEC system, the other's a Asterisk |
23:52.14 | tzanger | km-: and I could *own* the Norstar if I could get MCDN reverse-engineered |
23:52.15 | shmaltz | anybody (but tzanger) using polycom phones? |
23:52.16 | guugmember | Trionnis, ok, thnks |
23:52.22 | km- | tzanger: I suppose I could steel the adp for an FXO |
23:52.32 | km- | but that'd mean at max only one call could be transferred back to the system at once |
23:52.45 | guugmember | km- have you tried to pass calls from * to Avaya? I will have to do that soon |
23:53.18 | tzanger | km-: you wouldn't happen to have a manual for those ADPs would you? SPecifically flash codes or *-codes to do things like transfer or call forard and stuff? |
23:53.20 | hcclNoodles | i know that in the US the hang up signal is a voltage change or polarity reversal, but in the UK it is continuous tone from the teco for a period between 15-30 seconds |
23:53.42 | Graphikos | does a "sip reload" from CLI reload everthing including extensions.conf? |
23:53.43 | km- | tzanger: hmm, no, I dont believe they ever left us a manual on it |
23:53.48 | *** join/#asterisk calvinhp (~calvinhp@rrcs-24-123-25-236.central.biz.rr.com) |
23:53.56 | km- | tzanger: I'll dig around |
23:53.57 | tzanger | km-: damn, ok |
23:54.02 | hcclNoodles | */TDM400P doesnt detect this and waits for silence on the line |
23:54.22 | *** join/#asterisk |neuro| (~|neuro|@212.176.51.231) |
23:54.45 | Essobi | Mmm. |
23:54.49 | guugmember | what do you think of varion, V400P-E 4 Port E1 Digital Interface Card |
23:55.02 | Essobi | Graphikos no |
23:55.08 | Essobi | that's what "reload" does |
23:55.09 | guugmember | where can I see the diff between it and TDM400P |
23:55.13 | Essobi | sip reload only reloads sip.conf |
23:55.27 | Graphikos | ok.. so reload by itself... |
23:55.32 | Essobi | What? |
23:55.35 | Essobi | yea |
23:55.44 | Essobi | just reload loads all the "supported" configs |
23:55.50 | km- | first phone call |
23:55.52 | km- | worked good |
23:55.56 | km- | tzanger: I've got local echo problems |
23:56.00 | Essobi | nice |
23:56.06 | Essobi | love the echos |
23:56.21 | km- | I kicked rxgain up a bit to make the audio coming in a bit hotter |
23:56.21 | *** join/#asterisk ChrisRouse (~crouse@67.131.247.187) |
23:56.28 | km- | is it possible that I kicked it up too far? |
23:56.31 | Essobi | make sweeeet sweeet love to them, they'll leave you eventually. |
23:56.40 | Essobi | km- maybe |
23:56.42 | ChrisRouse | Good afternoon. |
23:57.33 | tzanger | km-: meaning you hear your own voice |
23:57.35 | tzanger | ? |
23:57.58 | ChrisRouse | I have a question about Cisco Call Manager integration if anyone is available... |
23:58.02 | km- | tzanger: yeah |
23:58.20 | km- | I just shut off the rxgain and the problem appears to have gone away |
23:58.21 | tzanger | km-: you can try killing txgain too |
23:58.26 | KalD|Work | ChrisRouse, we've done that at my place - what's the q? |
23:58.28 | tzanger | km-: also what do you have for echocancel= settings? |
23:58.34 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
23:58.38 | km- | no echocancel settings |
23:59.02 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
23:59.10 | tzanger | km-: try echocancel=32 and restart (not reload) asterisk |
23:59.10 | km- | should I set echocancel=yes and echotraining=yes? |
23:59.16 | tzanger | km-: don't use echotraining yet |
23:59.24 | tzanger | and dont' use yes, it's 128 and that's awfully long |
23:59.28 | ChrisRouse | Kald: I am working on integrating Asterisk with all manager. I am having a problem associating Cisco extensions with Asterisk. For instance when I attempt to login as an Agent and type in my extension asterisk tells me that it is an invalid extension. |