| 00:00.10 | *** part/#asterisk danielqb2 (n=danielqb@69.79.225.244) |
| 00:04.05 | *** join/#asterisk ctp (n=ctp@brsg-d9bef4d0.pool.mediaWays.net) |
| 00:05.41 | kn0x | my client needs capacity for 12 channels origination... any suggestions for SIP trunking? |
| 00:05.52 | kn0x | I was looking into SIP |
| 00:06.00 | kn0x | trunking from bandwidth.com |
| 00:06.23 | leifmadsen | kn0x: I've used bandwidth.com, they are ok |
| 00:06.43 | kn0x | leifmadsen: do they offer inbound-only channels, and how much/ channel? |
| 00:06.45 | paulius | kn0x: I fell in love with www.voip.ms |
| 00:06.53 | kn0x | paulius: are the reliable>? |
| 00:07.01 | leifmadsen | kn0x: I have no idea. I just implemented them, I didn't deal with them directly. |
| 00:07.08 | kn0x | oh |
| 00:07.09 | paulius | kn0x: So far, oh yes. And they have two different tiers. |
| 00:07.25 | paulius | And they actually have multiple SIP locations which I found great for latency. |
| 00:07.38 | paulius | They don't look like a fly by night company like most SIP trunk providers do. |
| 00:08.20 | kn0x | yeah thats what im affraid of |
| 00:08.42 | kn0x | if i port the clients phone number that they have for 25 years and leave it in the hands of some fucktards |
| 00:08.43 | paulius | Apparently they've been doing vo-ip since 2004 and they actually have offices, lol. |
| 00:08.56 | kn0x | yeah i l;ike the fact they have an address |
| 00:09.00 | paulius | AND their website doesn't look like some spyware site template taken from templatemonster. |
| 00:10.30 | kn0x | haha |
| 00:10.43 | paulius | I think their DIDs aren't too bad. Usually a few bucks per month for just the DID. And $5 for 2 channel unlimited. |
| 00:10.56 | kn0x | paulius: they dont seem to offer inbound trunking tho... just 2channel or per-minute billling |
| 00:11.05 | kn0x | i need at least 10 channels |
| 00:11.18 | paulius | Email their sales? |
| 00:11.28 | paulius | I'm sure they're more than willing to work with you if you're actually interested. |
| 00:11.43 | paulius | More $$$ = people usually willing to work |
| 00:11.44 | paulius | lol |
| 00:12.14 | *** join/#asterisk M1s3ry (n=M1s3ry@boromir.api-digital.com) |
| 00:12.16 | kn0x | yeah |
| 00:12.59 | kn0x | thats the problem, client is a jew... i would reccommend vitelity's virtual pri |
| 00:13.06 | kn0x | but they dont like $20/channel |
| 00:13.45 | paulius | Well if it's $5 for 2 channels, maybe they'll give it at 5 or less per channel. |
| 00:14.11 | paulius | And that's quite a not nice prejudice that you said. |
| 00:18.07 | *** join/#asterisk bbryant (n=brett@c-68-59-21-152.hsd1.sc.comcast.net) |
| 00:39.49 | KyleK | hehe I love south park |
| 00:40.32 | *** join/#asterisk jbjapan (n=jerome@mail.biosjp.com) |
| 00:41.45 | jbjapan | hi |
| 00:42.52 | KyleK | yo |
| 00:44.57 | KyleK | Unlimited channels DID numbers from $0.99 per month <-- wouldn't that be more than 10 incoming calls? |
| 00:46.12 | *** join/#asterisk Shinu (n=goodbyef@unaffiliated/shinu) |
| 00:46.44 | KyleK | ~tier |
| 00:48.39 | KyleK | whats meant by "premium tier-1" for calling people? |
| 00:49.41 | dshap | kn0x: voip.ms is a crazy-good deal, i just signed up for their services and have been using them to test my Asterisk box |
| 00:50.12 | dshap | kn0x: unfortunately for me they don't support a particular feature that I need and i've searched far and wide (well, not really) and have only found bandwidth.com to support the feature that i need |
| 00:50.21 | dshap | but bandwidth.com wants $30 PER CHANNEL for unlimited |
| 00:50.33 | dshap | or $17.50 PER CHANNEL + metered rate |
| 00:50.52 | dshap | i really don't get how voip.ms can offer $1 per DID + metered rate and give you UNLIMITED channels |
| 00:50.54 | dshap | but they say they can |
| 00:50.57 | KyleK | ILEC voip? |
| 00:51.42 | KyleK | a high price like that sounds like its a voip company ran by an incombent carrier somewhere |
| 00:51.52 | dshap | h |
| 00:51.54 | dshap | hm* |
| 00:52.18 | dshap | well i can say that i'm a huge n00b with this stuff and have consequently had to deal with voip.ms's support over the past couple days |
| 00:52.20 | dshap | and they are great |
| 00:52.39 | drmessano | voip.ms, how ironic |
| 00:52.41 | dshap | for the tiny amount of money i've given them, they've given me a great deal of attention and decent support |
| 00:52.49 | dshap | but i can't stay with them |
| 00:52.54 | *** join/#asterisk wwalker (n=wwalker@mailbox.eclipsing.com) |
| 00:53.15 | wwalker | is there a debug option that will show me when asterisk sees DTMF? |
| 00:54.14 | paulius | dshap: I've seen waaaaaay cheaper pricing man. |
| 00:54.27 | dshap | paulius: cheaper than voip.ms? |
| 00:54.35 | paulius | voip.ms is quite expensive compared to the other providers. |
| 00:54.41 | dshap | paulius: they have a 6-second billing increment on $0.01 per minute |
| 00:54.45 | paulius | But at least they have a decent website and are reliable. |
| 00:54.53 | paulius | dshap: Try 0.003 per minute. |
| 00:54.58 | dshap | where? |
| 00:55.03 | paulius | www.google.com |
| 00:55.06 | paulius | they're not hard to find |
| 00:55.22 | paulius | I saw tons of them when trying to find a provider. But I wouldn't recommend using those. |
| 00:55.26 | dshap | i am going to send e-mails to every VOIP provider I can find until i find an affordable one that offers the feature i need |
| 00:55.38 | dshap | yea honestly i thought voip.ms prices were a steal |
| 00:55.46 | dshap | i would be glad to pay them for a long time |
| 00:55.53 | paulius | They're great for the quality that you get. |
| 00:55.59 | paulius | I won't switch providers for a long while now. |
| 00:56.18 | KyleK | dshap: what are you using rdnis for? |
| 00:56.55 | dshap | KyleK: haha some people in here such as drmessano are going to get annoyed by me vaguely explaining this again... |
| 00:57.01 | dshap | but okay |
| 00:57.14 | dshap | i essentially want to duplicate a service that i currently use |
| 00:57.17 | dshap | www.youmail.com |
| 00:57.29 | dshap | they serve as an alternative to your major mobile carrier voicemail (i.e. AT&T voicemail) |
| 00:57.39 | dshap | you set your phone to forward missed calls to their PBX |
| 00:57.40 | *** join/#asterisk Deeewayne (n=dwayne@76.29.245.9) |
| 00:57.40 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 00:58.00 | dshap | when people call you and you don't pick up, it goes to them and they are able to determine both who is calling AND which mailbox to deliver to (i.e. rdnis) |
| 00:58.12 | dshap | i would think that many more people would need this feature |
| 00:58.13 | dshap | but i guess not |
| 00:58.18 | paulius | *caugh* Cisco CallManager had that for years *caugh* |
| 00:58.31 | dshap | still has to use RDNIS |
| 00:58.49 | dshap | voip.ms doesn't pass the CID of the forwarding phone in the SIP headers |
| 00:59.07 | dshap | bandwidth.com claims they do |
| 00:59.13 | dshap | but they charge more than i can afford right now |
| 00:59.16 | dshap | hence i will continue to search |
| 00:59.16 | paulius | And their support said they can't do it? |
| 00:59.17 | KyleK | huh, what are you going to use for transcriptioning of the messages? |
| 00:59.18 | dshap | and chat |
| 00:59.36 | dshap | paulius: yes, they flat out told me that they do not support it |
| 00:59.48 | KyleK | dshap: how many people are you planning on doing this for? if its less than 30 just get 30 dids ;) |
| 00:59.58 | drmessano | What does CCM supporting a feature have to do with a provider not supporting it? |
| 01:00.25 | paulius | drmessano: Just trying to irritate you... As usual :-P |
| 01:00.28 | dshap | KyleK: you're probably going to laugh just like everyone else did, but i actually have something else in mind. |
| 01:00.43 | paulius | dshap: Microsoft Bob? |
| 01:00.45 | dshap | it's kind of thing that i'll want to send invites to everyone on my facebook list to test it out |
| 01:00.51 | dshap | ~500 people or so |
| 01:01.19 | *** join/#asterisk propellerhead (n=yogurt2u@host237.190-31-75.telecom.net.ar) |
| 01:01.25 | drmessano | paulius: You're not clever enough to irritate me. It takes someone truly gifted to get to that point. But I admire your false sense of self worth. |
| 01:01.27 | dshap | i just need 1 DID with multiple channels and with RDNIS |
| 01:01.41 | dshap | i don't know any other way of doing it other than getting DIDs for every user which is too complicate |
| 01:01.43 | dshap | complicated* |
| 01:01.54 | KyleK | so whats the idea everyones laughed at? |
| 01:02.46 | dshap | it's more the fact that i'd rather not share my specific/full-detailed idea in a public place at this point |
| 01:02.47 | dshap | lol |
| 01:03.14 | dshap | and i understand the fact that i'm new to all of this and maybe if i told you guys everything about it you could come up with some other way that i'm not thinking about to implement it |
| 01:03.34 | dshap | but it will definitely involve doing what Youmail does |
| 01:03.39 | dshap | and for that i need RDNIS |
| 01:03.44 | dshap | my first obstacle |
| 01:03.47 | paulius | It's a good idea. Google Voice has that, I think, but it's a very limited beta/invite system. |
| 01:04.02 | dshap | Google Voice gives you a DID |
| 01:04.13 | dshap | and is way more complicated than what i plan to do |
| 01:04.26 | KyleK | well google could pay $30 for a DID with RDNIS |
| 01:04.32 | KyleK | but thats besides the point |
| 01:04.43 | dshap | true |
| 01:05.05 | dshap | i just wish there was some setting on my phone that would change the call forwarding CID |
| 01:05.11 | dshap | but that would be too simple |
| 01:05.35 | KyleK | bbiab dinner time |
| 01:05.36 | dshap | POTS lines cost a lot, don't they? |
| 01:05.44 | paulius | yes |
| 01:05.48 | KyleK | you'd need like isdn or something |
| 01:05.53 | paulius | which is why vo-ip is such a big deal lol |
| 01:05.57 | dshap | yea |
| 01:06.07 | dshap | i know if i got a PRI i could use RDNIS |
| 01:06.27 | dshap | but bandwidth.com's whole story is that they provide you with everything a PRI does and more, for cheaper |
| 01:06.32 | KyleK | so how many places have you emailed yet? |
| 01:06.32 | dshap | so that's out of the question |
| 01:06.39 | dshap | about 12 |
| 01:06.44 | dshap | there are probably a lot more huh |
| 01:06.45 | dshap | lol |
| 01:06.58 | dshap | there ARE a lot of sketchy ones though |
| 01:07.19 | *** join/#asterisk leif[mobile] (n=leifmads@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:07.19 | *** mode/#asterisk [+o leif[mobile]] by ChanServ |
| 01:07.32 | KyleK | dshap: I'm interested in the voice to text stuff |
| 01:07.38 | KyleK | dshap: les.net? vitelity.com? |
| 01:07.38 | dshap | totally unrelated to my idea |
| 01:07.42 | dshap | but yes it is interesting |
| 01:07.45 | dshap | i sent one to vitelity |
| 01:07.51 | dshap | still waiting on respone |
| 01:08.02 | dshap | not les |
| 01:08.06 | dshap | i do remember hearing about them |
| 01:08.12 | dshap | will add it to my e-mail list |
| 01:08.26 | dshap | i'm basically copying/pasting a "prospective customer" e-mail and changing the company name lol |
| 01:09.57 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:10.47 | dshap | les.net charges $10 per month for each additional channel |
| 01:10.56 | dshap | still a steal compared to bandwidth.com |
| 01:11.05 | dshap | and i have heard they are reliable even though their website sucks |
| 01:11.07 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 01:11.15 | dshap | *sends e-mail* |
| 01:11.26 | KyleK | my asterisk has been bitching about lag but that might be the router its behind |
| 01:11.45 | paulius | lol it's not too bad of a site |
| 01:11.47 | paulius | just random |
| 01:11.55 | KyleK | spikes between 3000ms and 40ms for the SIP registers |
| 01:11.56 | paulius | like some shitty building on the front page |
| 01:12.12 | KyleK | thats oddly enough marketing related |
| 01:12.33 | KyleK | but yea I lol'd |
| 01:12.59 | paulius | speaking of marketing, this thing is pretty cool: http://ge.ecomagination.com/smartgrid/?c_id=FM#/augmented_reality |
| 01:14.38 | *** join/#asterisk dshap (n=IceChat7@ip70-181-91-110.oc.oc.cox.net) |
| 01:15.03 | dshap | paulius: i blindly clicked that link and it took me out of the channel |
| 01:15.06 | dshap | haha wtf |
| 01:15.11 | paulius | wow how |
| 01:15.16 | paulius | irc in browser? lol |
| 01:15.19 | dshap | no |
| 01:15.22 | dshap | IceChat |
| 01:15.23 | paulius | It's like this virtual reality thing. |
| 01:15.29 | paulius | Oh okay, not familiar with that one. |
| 01:15.33 | dshap | neither am i |
| 01:15.33 | dshap | haha |
| 01:15.37 | KyleK | dshap: hopefully its not a socially distributed answering service ;) |
| 01:15.38 | paulius | I meant IceChat |
| 01:15.49 | dshap | KyleK: what's that? |
| 01:16.14 | KyleK | refering to your idea |
| 01:16.18 | dshap | yea |
| 01:16.24 | dshap | i don't know what you mean by that |
| 01:16.29 | dshap | "socially distributed answering service" |
| 01:17.53 | dshap | but it sounds like it could be :-p |
| 01:18.31 | leif[mobile] | sounds like a call centre |
| 01:18.40 | paulius | LOL |
| 01:18.48 | dshap | okay guys |
| 01:18.53 | dshap | 1 of 2 things will happen in the future |
| 01:19.03 | paulius | alright, go ahead with crazy predictions. |
| 01:19.03 | leif[mobile] | But thats what i'm calling them from now on! |
| 01:19.09 | paulius | We'll all have jet packs? |
| 01:19.22 | dshap | 1: i will implement my idea, come back here and show you guys |
| 01:19.30 | leif[mobile] | Either the future will exist, or it won't. |
| 01:19.41 | dshap | 2: i will never find an affordable service that has RDNIS, and thus i will come here and tell you guys my idea |
| 01:19.47 | paulius | lol okay |
| 01:19.54 | drmessano | 3. You will end up an entry on newbipedia |
| 01:20.05 | paulius | dshap: If it's some game changing idea, try to form a company around it or something. |
| 01:20.12 | leif[mobile] | 3! 3! 3! |
| 01:20.15 | dshap | haha |
| 01:20.17 | paulius | Get some VC money, move to san francisco, and live like a king... |
| 01:20.20 | dshap | paulius: i don't think it's HUGE |
| 01:20.21 | paulius | until the bubble bursts. |
| 01:20.29 | dshap | i just know it's something i would use every day |
| 01:20.33 | dshap | and some friends have told me they would as well |
| 01:20.37 | dshap | it's definitely a new-gen kind of idea |
| 01:20.47 | *** join/#asterisk juanIMP (n=Juancho@200.26.152.222) |
| 01:20.47 | KyleK | every day? wow |
| 01:20.59 | paulius | KyleK: We all use voice mail every day. |
| 01:21.11 | paulius | From all he has told, it's an extension to voice mail or call receival |
| 01:21.11 | dshap | my parents would probably never use it |
| 01:21.25 | drmessano | Yes, you will raise $1000 in VC before your mom realizes she gave you $1000 to invent voicemail, after which she will beat your ass and send you back to your basement apartment |
| 01:21.33 | dshap | lol |
| 01:22.28 | dshap | it's also not a 100% original idea (but then again what is) |
| 01:22.36 | dshap | i've seen it thrown out there on a few blogs |
| 01:22.43 | jaytee | voicemail has already been invented. the man who invented it is named Scott Jones and he lives here in Indianapolis. |
| 01:22.54 | dshap | people with no technical background come out and say "wouldn't it be cool if...._____" |
| 01:23.00 | *** join/#asterisk securevoip (n=securevo@c-76-123-20-170.hsd1.va.comcast.net) |
| 01:23.01 | paulius | dshap: Then continue your search for providers and try doing it. |
| 01:23.10 | dshap | i'm the guy who comes and says "yea, that would be fuckin awesome. also im gonna make it" |
| 01:23.12 | paulius | Maybe at least you can make some extra beer money with it by selling your service. |
| 01:23.20 | dshap | very true |
| 01:23.25 | dshap | im a 21 year old college student |
| 01:23.29 | dshap | at this point i have very little to lose |
| 01:23.36 | dshap | :-p |
| 01:23.37 | paulius | well... |
| 01:23.43 | paulius | <5 weeks later> |
| 01:23.46 | dshap | lol |
| 01:23.54 | dshap | only time will tell |
| 01:24.07 | paulius | <dshap> I'm trapped on a deserted island with sharks with friggin' lasers all around me. |
| 01:24.27 | dshap | i guess i have more to lose than i initially thought |
| 01:24.30 | paulius | lol |
| 01:24.34 | paulius | just like thisL |
| 01:24.43 | Qwell | wait, wait... is your idea to be the next Vonage? |
| 01:24.47 | Qwell | please say it is |
| 01:25.13 | dshap | Qwell: haha i can't remember, but it was either drmessano or carrar who beat you to that one last night |
| 01:25.15 | securevoip | anybody monkied with the Citel IP-Phone 4110 yet? IAX codec negotiation doesn't work |
| 01:25.31 | Qwell | dshap: I *invented* that one. |
| 01:25.34 | paulius | dshap: http://xkcd.com/349/ |
| 01:25.37 | Qwell | okay, I didn't really, but you get the idea |
| 01:25.39 | dshap | haha ok u got me there |
| 01:26.20 | dshap | lol |
| 01:27.12 | Qwell | paulius: a classic |
| 01:27.17 | paulius | indeed |
| 01:28.24 | dshap | Qwell: i'm not the kind of guy who wastes his time. if i didn't think this could be worth my time i wouldn't be on here every day asking people about RDNIS and Asterisk |
| 01:28.33 | Qwell | heh... |
| 01:28.41 | Qwell | neither would lots of people |
| 01:29.07 | paulius | Hahah. I just discovered www.omegle.com.... uhhm nevermind. |
| 01:29.28 | dshap | i've had nothing better to do for the past 2 weeks and still nothing to do for the next week until i start my summer internship |
| 01:29.37 | dshap | paulius: i've heard of that before |
| 01:29.47 | paulius | dshap: Except now it's getting strange. |
| 01:29.56 | KyleK | Stranger: hello |
| 01:29.56 | dshap | yea weird ass people go on there |
| 01:30.09 | paulius | I think I just discovered the to find a predator stinger... |
| 01:30.09 | KyleK | did it get goonswarmed yet? |
| 01:30.27 | paulius | what's that? |
| 01:31.09 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 01:31.17 | paulius | http://img132.imageshack.us/img132/3600/picture3wpz.png |
| 01:31.28 | paulius | You'd think there could be some moderation going on. |
| 01:38.32 | KyleK | paulius: something awful forum members refer to themselves as goons |
| 01:38.46 | paulius | KyleK: I still think 4chan is probably worse. |
| 01:38.51 | dshap | ahah |
| 01:38.57 | dshap | i'm an SA forum member |
| 01:39.01 | paulius | At least some of those awful forums members are intelligent. |
| 01:39.04 | paulius | Yeah, case in point. |
| 01:39.29 | dshap | i even made a post about my RDNIS bullshit |
| 01:39.30 | KyleK | so 4chan is the new SA because the goons grew up? :) |
| 01:39.31 | dshap | no respone yet though |
| 01:39.38 | dshap | 4chan is just a mess |
| 01:39.48 | dshap | there's some really interesting stuff that people post on SA |
| 01:39.49 | dshap | in the right forum |
| 01:53.53 | dshap | probably a dumb question: when I call tech support for some product/service and they put me on hold beause all of their agents are helping other customers...i'm using one of their channels/concurrent calls, right? |
| 01:54.10 | dshap | if a company wants to be able to have 5 people on the line with agents and 10 people waiting in a queue, they need 15 channels |
| 01:54.11 | dshap | correct? |
| 01:54.28 | paulius | yes |
| 01:54.48 | dshap | wow |
| 01:55.07 | dshap | so i bet some of these companies have expensive phone bills |
| 01:55.07 | dshap | hah |
| 01:56.21 | paulius | That's why companies have something called a profit margin. |
| 01:56.26 | dshap | does anyone here know anything about in-call advertising? like if you want to make money by playing advertisements to people waiting on hold? |
| 01:56.29 | dshap | is that feasible? |
| 01:56.48 | *** join/#asterisk classicmac (n=michaelm@udp223660uds.hawaiiantel.net) |
| 01:56.53 | paulius | I think some startups tried that. |
| 01:56.55 | Qwell | dshap: is your customer calling to complain about something? |
| 01:57.00 | dshap | no |
| 01:57.03 | *** join/#asterisk MrNaz (n=mrnaz@ppp121-44-214-193.lns10.mel4.internode.on.net) |
| 01:57.09 | Qwell | dshap: is your customer calling to buy something? |
| 01:57.09 | dshap | it's actually part of the service that they'll be using |
| 01:57.20 | dshap | and the free service is ad-supported |
| 01:57.24 | dshap | but they can pay to remove the ads |
| 01:57.28 | dshap | a small subscription fee |
| 01:57.42 | dshap | im just wondering how hard it is to find advertisers willing to pay for that |
| 01:58.22 | dshap | and what the payment structure is |
| 01:58.36 | dshap | by the minute? by the # of plays? |
| 01:58.39 | classicmac | I'm using trixbox with an X100P but keep getting an error when the kernel module is being loaded saying NOTICE-wcfxo: WCFXO/0: Unknown DAA chip revision: REVB=0. Does anyone have any ideas on how to fix it? |
| 02:02.25 | Qwell | ~cheap |
| 02:02.26 | infobot | it has been said that cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 02:02.27 | Qwell | classicmac: ^^^ |
| 02:03.04 | Qwell | You bought a clone of a crappy (to begin with) card, that isn't compatible with the real drivers. |
| 02:03.24 | classicmac | I understand that the card isn't the greatest |
| 02:03.37 | Qwell | It isn't supported. Talk to the manufacturer if you can find them. |
| 02:03.47 | classicmac | I have it though so if it is possible I would like to try to get it working. |
| 02:04.01 | classicmac | I did actually get the digium card though. Not a clone. |
| 02:04.15 | Qwell | Did you buy it 6 years ago? |
| 02:04.21 | classicmac | A few weeks ago. |
| 02:04.26 | Qwell | Was it $10? |
| 02:04.32 | classicmac | $30 |
| 02:04.35 | Qwell | mmhmm |
| 02:04.36 | paulius | lol |
| 02:04.38 | Qwell | unsupported |
| 02:04.42 | paulius | Whta is that card supposed to do? |
| 02:04.45 | Qwell | (and you overpaid..lol) |
| 02:04.52 | classicmac | Yeah I understand that it is unsupported and junk. |
| 02:04.59 | classicmac | Just one FXO port. |
| 02:05.03 | Qwell | Then why are you still asking? |
| 02:05.12 | paulius | wow that's cheap for FXO |
| 02:05.19 | Qwell | paulius: no, it's just cheap. |
| 02:05.22 | classicmac | I would still like to try to get it working. Because it replaced a generic X100P that was working but unstable. |
| 02:05.26 | Qwell | (expensive for that card though) |
| 02:05.32 | Qwell | classicmac: I told you what you need to do. |
| 02:05.59 | classicmac | ok, thanks |
| 02:06.27 | Qwell | actually, you know what.. you bought it off ebay, right? |
| 02:06.31 | classicmac | Yes |
| 02:06.33 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 02:06.34 | Qwell | got the link? |
| 02:06.40 | classicmac | Let me see if I still have it. |
| 02:06.57 | paulius | Qwell: What'cha gonna do? Those clones are illegal? |
| 02:07.02 | Qwell | paulius: :) |
| 02:07.10 | classicmac | Also I bought two cards from the same auction and the other one works fine on another system. |
| 02:08.36 | *** join/#asterisk s14ck (n=s14ck@190-76-79-45.dyn.movilnet.com.ve) |
| 02:09.24 | classicmac | The item number on ebay was 120404635229 |
| 02:09.36 | Qwell | US? |
| 02:09.54 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 02:10.03 | classicmac | No |
| 02:10.10 | Qwell | you, not them |
| 02:10.18 | classicmac | Oh yes I am in the US. |
| 02:10.19 | classicmac | Hawaii |
| 02:11.16 | Qwell | ebay's URLs are neat.. |
| 02:11.17 | Qwell | http://cgi.ebay.com/_W0QQitemZ120404635229QQcmdZViewItem |
| 02:11.42 | paulius | "The X100P SE is The Defacto Standard Single Port FXO Interface for Asterisk" |
| 02:11.43 | paulius | haha |
| 02:13.06 | paulius | Qwell: They keep putting emphasis that the card is authentic. |
| 02:13.09 | paulius | But it's not, right? |
| 02:13.11 | dshap | anyone here ever used Vonage as a SIP provider? |
| 02:13.15 | Qwell | no, it's not. |
| 02:13.36 | classicmac | How can you tell it is not authentic? |
| 02:13.44 | paulius | classicmac: Price. Location. |
| 02:13.45 | Qwell | how can I tell? |
| 02:13.54 | Qwell | The fact that it hasn't been made/sold in...6 years? |
| 02:13.58 | paulius | LOL |
| 02:14.09 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 02:14.29 | Qwell | the fact that it *never once* says that it's made by Digium? |
| 02:14.38 | Qwell | (learn to read auctions, people) |
| 02:15.20 | Qwell | to be honest, I can't believe you paid $80 incl shipping for 1 working FXO |
| 02:15.21 | classicmac | All it said was authentic X100p |
| 02:15.25 | classicmac | Not authentic digium. |
| 02:15.28 | Qwell | for...that, anyhow |
| 02:15.47 | Qwell | <classicmac> I did actually get the digium card though. Not a clone. |
| 02:15.49 | ricko73 | Qwell: funny they don't don't claim it was made by Digum or some other spelling variation on Digium |
| 02:15.49 | classicmac | It is a lot better than the other cheap X100P card. |
| 02:16.03 | Qwell | classicmac: ...IT'S A CLONE *OF THE CRAP ONE* |
| 02:16.08 | paulius | LOOOOL |
| 02:16.10 | paulius | gotta love that. |
| 02:16.11 | ricko73 | "Reel card made bi Deegium" |
| 02:16.21 | classicmac | Ok that was my mistake. |
| 02:16.29 | classicmac | I was misled by the auction. |
| 02:16.32 | *** part/#asterisk classicmac (n=michaelm@udp223660uds.hawaiiantel.net) |
| 02:16.40 | *** join/#asterisk classicmac (n=michaelm@udp223660uds.hawaiiantel.net) |
| 02:17.04 | classicmac | But aside from that does anyone have any constructive suggestions. |
| 02:17.27 | classicmac | I hate to put the other card back in. The other card is even worse. |
| 02:17.28 | Qwell | I told you what to do. |
| 02:17.35 | classicmac | Contact the manufacturer? |
| 02:17.44 | Qwell | That would be a good start. |
| 02:18.15 | classicmac | ok, thanks |
| 02:19.08 | paulius | Yeah, getting part of your money back would be the best start. |
| 02:19.15 | classicmac | Nah, the other one works. |
| 02:19.22 | Qwell | "works" |
| 02:19.39 | classicmac | I'm not running a business with it. |
| 02:19.45 | classicmac | It is just to experiment with. |
| 02:20.13 | classicmac | I'd definitely get a better card if I needed it to be reliable. |
| 02:20.19 | ricko73 | an empty Campbell's soup can might be more reliable |
| 02:20.19 | classicmac | The PC it is in isn't the greatest either. |
| 02:20.57 | classicmac | If I could use a soup can I would try it. This is just to experiment. |
| 02:21.03 | bkw_ | Qwell: my man how are you doing? |
| 02:21.03 | paulius | lol |
| 02:21.17 | Qwell | bkw_: got fillings today, and my jaw is f'ing killing me.. |
| 02:21.24 | paulius | :-( |
| 02:21.27 | Qwell | other than that... not bad :D |
| 02:22.12 | *** join/#asterisk ctp (n=ctp@brsg-d9bee414.pool.mediaWays.net) |
| 02:22.12 | bkw_ | Qwell: I got all four of my wisdom teeth cut out in one day.... and I was AWAKE when they did it... top that! |
| 02:22.12 | Qwell | bkw_: my appointment was Tuesday |
| 02:22.12 | bkw_ | I almost beat the shit out of the lady at walmart |
| 02:22.12 | bkw_ | she wouldn't let greg pick up my pain killers |
| 02:22.12 | Qwell | heh |
| 02:22.13 | bkw_ | so it was 4 hours after the fact when I finally got them |
| 02:22.13 | paulius | LOL |
| 02:22.18 | bkw_ | I was in so much pain I thought I was gonna die |
| 02:22.21 | paulius | :-( |
| 02:22.27 | Qwell | bkw_: just recently? |
| 02:22.31 | bkw_ | it was a few years ago |
| 02:22.42 | Qwell | bkw_: I went and saw the surgeon the other day to talk about getting mine out |
| 02:22.50 | bkw_ | Qwell: just do them all at the same time |
| 02:22.53 | bkw_ | don't pussy foot it |
| 02:22.55 | Qwell | can't ;/ |
| 02:22.58 | bkw_ | why? |
| 02:23.04 | bkw_ | don't wanna look like a chimpmunk? |
| 02:23.09 | bkw_ | cuz I sure as hell did |
| 02:23.21 | Qwell | insurance stuffs.. meh. rather not get into it |
| 02:23.35 | ricko73 | bkw_: I ate steak the same night they took out my wisdom teeth |
| 02:23.45 | bkw_ | Qwell: chances are the fillings are hitting your upper or lower teeth |
| 02:24.01 | ricko73 | I didn't have much swelling though so I feel lucky. |
| 02:24.38 | Qwell | the fillings were on the back side of the last teeth. there was some stretching involved. that's why the jaw hurts |
| 02:24.57 | Qwell | it was just this morning, so.. |
| 02:25.16 | bkw_ | bet they need some shaving |
| 02:25.31 | bkw_ | that is where mine was when they didn't do it right and it caused my jaw to be out of whack by about a mm |
| 02:25.36 | bkw_ | which hurts liek a bitch |
| 02:25.44 | *** join/#asterisk orpheee (n=u-applet@tra78-6-88-186-13-251.fbx.proxad.net) |
| 02:25.46 | bkw_ | just get the dremel tool out and do it yourself |
| 02:25.57 | Qwell | he said to expect it, and give it a few days. if it still hurts, I'm gonna go back |
| 02:26.12 | bkw_ | sounds like an asshat to me if it hurts something is wrong |
| 02:26.38 | ricko73 | sounds like something fun for Astricon |
| 02:26.51 | ricko73 | drill Qwell's teeth for fun a profit |
| 02:26.57 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 02:27.02 | Qwell | it's all just pressure pain. feels like it's on muscle or something where it's hurting |
| 02:27.24 | *** join/#asterisk orpheee (n=u-applet@tra78-6-88-186-13-251.fbx.proxad.net) |
| 02:27.27 | orpheee | hi |
| 02:27.34 | bkw_ | low |
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| 02:29.03 | *** join/#asterisk nephfl (n=none@wsip-98-175-64-147.ga.at.cox.net) |
| 02:29.19 | nephfl | anyone have a link to a reference for off-hook auto-dial for polycom phones? |
| 02:29.56 | drmessano | Qwell: Never use "stretching" and "jaw hurts" in the same sentence |
| 02:30.21 | drmessano | Technically, that was two sentences. Fine. |
| 02:34.39 | *** join/#asterisk bryanfe2 (n=chatzill@pool-71-102-158-60.snloca.dsl-w.verizon.net) |
| 02:35.45 | bryanfe2 | I have a small asterisk module I wrote for my SIP clients. Most of them are fine, but one SIP client when it hits my app, asterisk sends me tens of thousands of frames of type 5 ("An empty, useless frame"). Does anyone have any idea why that would be? My CPU is pegged having to process them all. |
| 02:36.06 | bkw_ | bryanfe2: sounds like you need to yield somewhere |
| 02:36.09 | bryanfe2 | whats frame type 5 for and why would asterisk send me 150,000 of them? |
| 02:36.21 | bkw_ | bryanfe2: what are yo doing exatly? |
| 02:36.29 | bryanfe2 | waiting for a frame type of DTMF |
| 02:36.38 | bkw_ | then doing? |
| 02:37.03 | bryanfe2 | I'm just calling ast_waitfor, then ast_read, and looping forever until the frame which ast_read returns is DTMF |
| 02:37.16 | bkw_ | did you check out app_read does it? |
| 02:37.23 | bryanfe2 | or until frame type of AST_FRAME_CONTROL (hangup) |
| 02:37.45 | bryanfe2 | or until it returns null |
| 02:37.51 | bryanfe2 | I guess I should look at app_read then |
| 02:55.04 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-2114bd2d109f368f) |
| 02:58.26 | bryanfe2 | I'm not sure what I'm supposed to do with AST_FRAME_NULL |
| 02:58.52 | bryanfe2 | so on bkw's idea, I'm going to try to yield with ast_safe_sleep(chan, 100) whenever I get a AST_FRAME_NULL and see if that helps. |
| 03:03.32 | *** part/#asterisk securevoip (n=securevo@c-76-123-20-170.hsd1.va.comcast.net) |
| 03:05.44 | bkw_ | brb |
| 03:16.48 | leifmadsen | dances |
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| 03:42.18 | *** join/#asterisk Aiatek (n=Alfio@190.94.56.221) |
| 03:42.51 | Aiatek | hi, im having a little issue with an analog card configuration |
| 03:43.12 | Aiatek | i will give you the paste bi |
| 03:43.23 | Aiatek | bin* |
| 03:45.11 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 03:46.55 | Aiatek | http://pastebin.com/m785413ab |
| 03:47.27 | Aiatek | here it is i can get tone from my fxs station, i got the molex cable conected |
| 03:48.08 | Aiatek | yesterday i was in asterisk 1.4 with zaptel and all worked fine |
| 03:48.39 | Aiatek | right now im in asterisk 1.6.0 and dahdi 2.1.0.4 |
| 03:49.22 | Aiatek | ? |
| 03:57.42 | *** join/#asterisk Asteriskdom (n=Alfio@190.94.56.221) |
| 03:57.45 | Asteriskdom | hi |
| 03:59.12 | Asteriskdom | http://pastebin.com/m785413ab |
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| 04:32.47 | carrar | WHAT |
| 04:33.55 | *** join/#asterisk dshap (n=IceChat7@ip70-181-91-110.oc.oc.cox.net) |
| 04:39.58 | *** join/#asterisk demonist (n=hi@CPE0004dd6b8001-CM0019474d4b20.cpe.net.cable.rogers.com) |
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| 05:11.22 | dshap | anyone here ever heard aanything about NexVortex? |
| 05:15.16 | *** join/#asterisk rashed2020 (n=asda@67.205.245.208) |
| 05:15.41 | rashed2020 | What's up guys. I'm writing an article, I'm trying to get a sense of what the asterisk community thinks of freeswitch. Anyone care to comment? |
| 05:17.09 | dshap | i've been working with asterisk for about 1 week. when i started looking into open source PBX options, asterisk was the only thing i ever came across. i have never even heard of freeswitch. |
| 05:17.11 | dshap | that is my 2 cents |
| 05:17.13 | ricko73 | rashed2020: probably not the best place to ask |
| 05:17.33 | rashed2020 | ricko73: Suggestions? |
| 05:18.10 | rashed2020 | dshap: You should look into it. From what I get so far is that it's better when it comes to performance. But I could completely wrong, I'm not sure. |
| 05:18.20 | ricko73 | ~book |
| 05:18.21 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 05:18.35 | dshap | i'm pretty impressed with how configurable Asterisk is once you set it up |
| 05:19.00 | ricko73 | rashed2020: perhaps a location that's more neutral |
| 05:19.20 | rashed2020 | ricko73: I'm not trying to figure out the techincal side of stuff ATM. Just what the communities think of each other. Or is there no connection at all? |
| 05:19.22 | ricko73 | something like #voip-users-conference |
| 05:19.33 | rashed2020 | Oh, gotcha |
| 05:19.41 | ricko73 | but that channel is mostly only active on Fridays |
| 05:19.47 | ricko73 | in about 10 hrs |
| 05:20.06 | ricko73 | there's a weekly conference call |
| 05:20.16 | rashed2020 | Cool. Thank you. |
| 05:20.24 | dshap | ricko73: do you know of a channel where people discuss different SIP providers? |
| 05:20.45 | ricko73 | not off hand |
| 05:20.57 | dshap | do you use a SIP provider yourself |
| 05:20.57 | dshap | ? |
| 05:21.32 | ricko73 | yes, I use one myself and have experience using a few others with my clients |
| 05:21.41 | ricko73 | all of them have their pluses and minuses |
| 05:21.53 | ricko73 | I don't believe there is one perfect provider |
| 05:22.04 | ricko73 | some are better than others |
| 05:22.39 | dshap | what's the deal with some that charge like 20 times as much as others for seemingly the same service? |
| 05:22.48 | dshap | voip.ms vs. bandwidth.com |
| 05:22.59 | ricko73 | never heard of voip.ms |
| 05:23.03 | dshap | ic |
| 05:23.16 | dshap | well bandwidth.com charges $30 per month PER CHANNEL |
| 05:23.26 | dshap | it seems like any other providers offer more channels per month for a fraction of that |
| 05:23.28 | ricko73 | no they don't |
| 05:23.29 | dshap | just trying to understand it |
| 05:23.33 | dshap | hu |
| 05:23.36 | dshap | huh* |
| 05:23.37 | ricko73 | well they might if you have a short term contract |
| 05:23.59 | dshap | on their pricing PDF it says a 1,2,or 3 year contract is necessary for $30/month for unlimited calling |
| 05:24.02 | ricko73 | You also don't get charged for incoming or local calls with bandwidth.com |
| 05:24.02 | dshap | for a single channel |
| 05:24.14 | dshap | unlimited inbound/outbound ^ |
| 05:24.27 | ricko73 | you also can pick up the phone and talk to an engineer pretty much 24/7 |
| 05:24.35 | dshap | hm |
| 05:24.41 | dshap | yea |
| 05:24.43 | ricko73 | so you get what you pay for |
| 05:24.54 | dshap | as of now, bandwidth.com is the only provider i've found that supports RDNIS |
| 05:25.04 | ricko73 | right now I'm using the voip carrier who is going to carry me to my bed |
| 05:25.04 | dshap | but im still looking for a cheaper one |
| 05:25.11 | dshap | wow |
| 05:25.13 | ricko73 | night |
| 05:25.15 | dshap | and who might that be? |
| 05:25.43 | ricko73 | beatfeet |
| 05:25.56 | dshap | =\ |
| 05:28.46 | *** join/#asterisk {Repelex} (n=STRESSAD@201.86.177.249.dynamic.adsl.gvt.net.br) |
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| 06:55.20 | *** join/#asterisk ck_28 (n=CK@212.98.141.199) |
| 06:55.27 | ck_28 | Morning All |
| 06:55.53 | ck_28 | i have a problem with GOTOIF condition |
| 06:55.58 | ck_28 | exten => h,n,GotoIf($["${FAXOPT(status)}}" = "SUCCESS"]?success:failed) |
| 06:55.59 | ck_28 | exten => h,n(success),Goto(mysqlcal,555,1) |
| 06:55.59 | ck_28 | exten=> h,n(failed),Goto(mysqlcal2,666,1) |
| 06:57.06 | ck_28 | the debug is at http://pastebin.com/d7d7a52f3 |
| 06:57.54 | ck_28 | why when "${FAXOPT(status)} ="SUCCESS" it goes to h,n(failed), ?????? |
| 06:59.08 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
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| 07:12.48 | *** join/#asterisk SebastianS (n=schu@dsl-static-111.212-5-200.telecom.sk) |
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| 07:22.31 | miloux | ck_28: Print the FAXOPT(status) var to confirm its set to SUCCESS...Noop(${FAXOPT(status)}) |
| 07:23.36 | ck_28 | miloux -- Executing [h@fax-tx:15] NoOp("SIP/msx-09b3aa88", "---------------------------SUCCESS----------------------") in new stack |
| 07:29.15 | ck_28 | miloux any idea |
| 07:29.24 | miloux | ck_28: can you paste more of the dialplan? |
| 07:29.27 | miloux | this part: |
| 07:29.32 | miloux | -- Executing [h@fax-tx:11] NoOp("SIP/msx-09b3aa88", "FAXOPT(status) : SUCCESS") in new stack |
| 07:29.32 | miloux | <PROTECTED> |
| 07:29.32 | miloux | <PROTECTED> |
| 07:29.32 | miloux | <PROTECTED> |
| 07:29.32 | miloux | <PROTECTED> |
| 07:29.38 | kaldemar | ~pb |
| 07:29.39 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 07:30.28 | kaldemar | ck_28: "${FAXOPT(status)}}" should be "${FAXOPT(status)}" |
| 07:30.46 | ck_28 | Micc http://pastebin.com/d48072a25 |
| 07:31.12 | ck_28 | kaldemar thanks i will try that :) |
| 07:37.11 | ck_28 | kaldemar thanks for your accurency :) |
| 07:38.50 | kaldemar | np, those brackets own everyone at some point. :) |
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| 07:58.45 | CrashSys | Anyone ever gotten cause code 99 if you attempt to place a call on a PRI with caller-id-name preasent? |
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| 08:38.23 | dandre | Hello, |
| 08:38.54 | dandre | How can I display a short message on my sip phone while it is called? |
| 08:41.47 | *** join/#asterisk my007ms (i=master@217.139.17.150) |
| 08:42.01 | kaldemar | depends on the phone, but for example sipsak can be used to send messages to some phones. |
| 08:42.37 | dandre | I am using thomson st2060 |
| 08:44.06 | CrashSys | I've got an NI2 PRI and get cause code 99 if anything is defined in callerid(name)... is there anyway to tell zaptel to NOT send out callerid(name) other then within extensions.conf? |
| 08:46.25 | dandre | I have tried to send Alert-Info: sometext header but it doesn't work |
| 08:49.18 | kaldemar | have you asked thomson? i recall snom people being very helpful with that when i asked about it years ago. |
| 08:50.45 | *** join/#asterisk cool^tom (n=thomas@122.166.46.215) |
| 08:50.58 | cool^tom | Hi |
| 08:53.46 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.96.203) |
| 08:53.50 | cool^tom | I have a Digium TE121B Pri Card. It used to work properly however now the booting the OS stops at udev. Is there a problem with the card? |
| 08:53.53 | DelphiWorld | hello |
| 08:54.02 | cool^tom | Hi DelphiWorld. |
| 08:55.31 | ectospasm | cool^tom: you should call Digium tech support to troubleshoot that further |
| 09:00.07 | Kevin` | would an ambient md5628d-based modem work as an fxo port for asterisk? I also have an agre, lucent, and conexant card |
| 09:06.24 | Zhad | Just as a curiousity, has a channel driver been written that allows you to use a mobile as an FXO device using a datacable? |
| 09:07.11 | Kevin` | hm, i'm just gonna stick them all in and see if any drivers pick them up |
| 09:07.21 | Zhad | I know it can be done using bluetooth (Which I played with in callweaver). |
| 09:08.01 | Zhad | Kevin> afaik the only modem that can be used is the X100/X101 (which iirc is a motorola). |
| 09:08.03 | *** join/#asterisk stijnbe (n=stijnbe@d54C16246.access.telenet.be) |
| 09:08.11 | Zhad | (which I probably don't). |
| 09:08.28 | Zhad | if you look at the sources, you will see the PCI IDs. |
| 09:09.11 | \void\ | cool^tom, do you see any comment when udev stops loading? |
| 09:09.37 | \void\ | cool^tom, comment=system message* |
| 09:09.54 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 09:10.11 | *** join/#asterisk dinhtrung (n=dinhtrun@118.71.94.128) |
| 09:10.21 | dinhtrung | hi all |
| 09:10.34 | dinhtrung | i'm testing OpenVOX 400 |
| 09:10.37 | dinhtrung | using dahdi |
| 09:10.46 | dinhtrung | and i can't manage to get the caller ID from PSTN |
| 09:10.54 | dinhtrung | i'm from vietnam |
| 09:11.37 | dinhtrung | in my country, telco send DTMF Caller ID, then a reversal polarity, then ringing |
| 09:11.53 | dinhtrung | is there any patch or some configuration i could use to get caller ID? |
| 09:12.33 | dinhtrung | i read about zaptel's patch for uk, and DTMF callerID without reversal polarity for brazil and costa rica, but can't find one to handle my situation |
| 09:12.37 | dinhtrung | plz help :( |
| 09:14.35 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 09:14.37 | tzafrir_laptop | dinhtrung, do you actually get a polarity reversal event? |
| 09:14.49 | dinhtrung | yep |
| 09:14.57 | dinhtrung | I'll pastebin some from full log |
| 09:15.48 | tzafrir_laptop | dtmf and only then polarity reversal? that good does that do? |
| 09:15.52 | tzafrir_laptop | :-( |
| 09:15.58 | Zhad | din> I don't know if it's the same, but on the 800P cards and the 1200P cards callerid only works through dtmf and then that's only if you have their very latest patches. |
| 09:16.10 | dinhtrung | the last warning message is something like DTMF CID Timed Out |
| 09:16.22 | Zhad | although the 400P cards may just be TDP400P clones. |
| 09:16.27 | Zhad | s/TDP/TDM/; |
| 09:16.44 | tzafrir_laptop | there's a patch to try to listen all the time for dtmfs and then fake a polarity reversal event up |
| 09:16.59 | dinhtrung | yep, it's the patch for brazil telco |
| 09:17.14 | dinhtrung | i tried that, and the card can receive DTMF CID |
| 09:17.25 | dinhtrung | but after the polarity, it timeout waiting for ringing |
| 09:17.29 | dinhtrung | so it hangup the channel |
| 09:17.34 | dinhtrung | and start a new one |
| 09:17.43 | dinhtrung | the dialplan works here, not the first one |
| 09:17.59 | dinhtrung | e.g 2 calls for just 1 call from PSTN |
| 09:18.21 | dinhtrung | https://issues.asterisk.org/view.php?id=9096 |
| 09:18.52 | dinhtrung | here is the patch that help recognize DTMF |
| 09:18.54 | Zhad | takes it that there isn't a channel driver that works using mobiles as premicells then. |
| 09:19.31 | Zhad | vaguely remembers reading something years ago, probably wasn't finished/started. |
| 09:28.22 | *** join/#asterisk mahiti-irc (n=mahiti@203.145.183.210) |
| 09:28.31 | mahiti-irc | hi |
| 09:28.34 | mahiti-irc | has anyone here configured asterisk and freepbx for a hypermedia gsm gateway? |
| 09:28.42 | mahiti-irc | i require some help with that |
| 09:34.21 | *** join/#asterisk MrNaz (n=mrnaz@203.214.68.222) |
| 09:38.22 | mahiti-irc | i am actually using asterisknow with ooh323 addon |
| 09:40.30 | kaldemar | #asterisknow could be more helpful |
| 09:43.14 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 09:48.30 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:56.41 | defswork | wants asterisktomorrow |
| 09:56.56 | defswork | or is it asterisdontknow |
| 10:03.13 | Zhad | homes chan_mobile is as simple(ish) to set up as chan_bluetooth was. |
| 10:04.23 | Zhad | wonders if dahdi will ever make it into the mainline kernel. |
| 10:06.14 | dandre | What is the meaning of CALLERID type? I understand "name" and "num" but not "ANI", "DNID", "RDNIS". |
| 10:07.06 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 10:09.14 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 10:18.10 | xrmx__ | hi, how do you use qos? do you mangle dscp with iptables on the asterisk machine or let routers / firewall do the job? |
| 10:19.05 | andresmujica | asterisk server must mark the packets |
| 10:19.36 | andresmujica | if asterisk is compiled with libpcap support it would do it by itself with the corresponding confs at iax and sip.conf |
| 10:19.44 | andresmujica | if not you must mark those packets at iptables |
| 10:20.17 | xrmx__ | andresmujica, thanks for the answer, will check that |
| 10:20.20 | andresmujica | ok |
| 10:23.27 | xrmx__ | andresmujica, do you kinow if it /usr/bin/asterisk linked against libpcap or is a particular module? |
| 10:23.38 | Zhad | would be nice to not need to recompile dahdi every time I make a small chanhe to the kernel. |
| 10:25.14 | *** join/#asterisk tarik (n=chatzill@41.248.238.217) |
| 10:25.41 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 10:25.42 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 10:26.14 | brunner | Isn't there a way to allow queue agents to acknowledge a call before they're automatically connected to the next person in a queue? |
| 10:27.36 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 10:27.46 | andresmujica | xrmx: it depends a lot on distribution, and it's after an asterisk version i don't recall which one... check your logs and if you find some error about asterisk couldn't set the dscp something you'll have to go iptables route |
| 10:29.45 | xrmx__ | andresmujica, yeah, my asterisk is older than that so i have to do with iptables |
| 10:32.01 | phix | hi |
| 10:32.05 | phix | lets asterisk |
| 10:35.59 | *** join/#asterisk uluatu (n=uluatu@189.32.44.21) |
| 10:39.21 | phix | uluatu: you like to asterisk? |
| 10:41.14 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 10:41.31 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
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| 10:47.33 | *** join/#asterisk Jurian (n=magic@217-208-153-205-no26.tbcn.telia.com) |
| 10:49.52 | brunner | what's the cheapest sip phone on the market? |
| 10:50.27 | Jurian | Hi, I need some help finding what I messed up in my asterisk config; I have several SIP phones, they can call each other just fine, they can also call out over several SIP trunks. The problem is with incoming calls over those same SIP trunks, I can hear the caller, but they cannot hear me. |
| 10:50.32 | Jurian | The server is not behind any NAT or firewall. allowreinvite is off, and tcpdump shows RTP traffic is in fact flowing both ways. |
| 10:50.34 | Chainsaw | brunner: You're probably going to end up with some nasty GrandStream then. Please reconsider. |
| 10:50.59 | brunner | it's not for me, so no, I won't be using it |
| 10:51.09 | Jurian | the problem occurs both on 1.4 and 1.6 so I'm sure it's a config problem on my end :| |
| 10:51.39 | Jurian | anyone have any idea what could be wrong? |
| 10:51.45 | phix | brunner: asterisk |
| 10:51.46 | phix | fag |
| 10:51.59 | Jurian | the phones all have nat=yes and host=dynamic |
| 10:52.11 | phix | Jurian: firewall |
| 10:52.12 | phix | RTP |
| 10:52.13 | *** join/#asterisk intralanman (n=lanman@70-10-38-150.pools.spcsdns.net) |
| 10:52.17 | phix | man rtp |
| 10:52.21 | Jurian | no firewall on the server |
| 10:52.25 | phix | man rtp |
| 10:52.35 | phix | NAT == firewall |
| 10:52.44 | Jurian | there is only nat on the client side |
| 10:52.49 | phix | you need to map port |
| 10:52.50 | phix | s |
| 10:52.56 | phix | you setup a proxy on the server |
| 10:52.57 | phix | fag |
| 10:52.58 | phix | <3 |
| 10:53.01 | phix | sex |
| 10:53.25 | kaldemar | ~sipnat |
| 10:53.26 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 10:53.59 | Jurian | weird, the same phones work fine with my other asterisk 1.2 server without any special nat settings |
| 10:54.33 | Jurian | but I'll forward the ports just to be sure |
| 10:56.47 | phix | hi |
| 11:00.44 | Jurian | forwarding the rtp ports to my phone doesn't change anything; the rtp traffic goes to and from the phone just fine |
| 11:04.03 | phix | Jurian: cunt |
| 11:04.19 | Jurian | again, the server is NOT behind nat |
| 11:04.25 | Jurian | only the phones are |
| 11:06.20 | phix | Jurian: that is your problem |
| 11:06.22 | phix | cunt |
| 11:06.34 | phix | lets talk about vagina |
| 11:06.42 | phix | NAT is boring me |
| 11:06.43 | Jurian | http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions ; 9: Asterisk as a SIP server outside nat, clients on the inside connecting to Asterisk |
| 11:06.46 | Jurian | #9 is solved with nat=yes and qualify=xxx in sip.conf for the client in most cases. Some clients (X-lite) assist themselves by using STUN and sending UDP keep-alive packets. Qualify sends keep-alive packets from Asterisk to the client on the inside. |
| 11:06.48 | Jurian | I have that |
| 11:06.53 | phix | no, URLs are boring me too |
| 11:06.55 | Jurian | and it doesn't help |
| 11:07.05 | phix | Jurian: you don't help |
| 11:07.13 | Jurian | neither do you, obviously |
| 11:07.15 | phix | where is your sisteR? |
| 11:07.26 | phix | she can help me more |
| 11:07.28 | Jurian | she's been dead for a few weeks, you can have her |
| 11:07.35 | phix | great |
| 11:07.37 | Jurian | if you're into that sort of thing |
| 11:07.42 | Jurian | I have some sheep for you as well |
| 11:07.46 | phix | I respect if you want a few weeks with her first |
| 11:07.53 | phix | she doesn't haveb to be freah |
| 11:07.55 | phix | fresh |
| 11:08.00 | phix | I can compansate |
| 11:08.13 | phix | Jurian: oh no, I am not from NZ, I dont require sheep |
| 11:08.35 | phix | I am from AU |
| 11:08.39 | phix | I like black cunts |
| 11:08.47 | Jurian | so, is there anyone with a clue here? :| |
| 11:09.48 | kaldemar | this is the moment when you should be pasting configurations and a cli output of a failed call. :P |
| 11:10.03 | phix | Jurian: I has one |
| 11:10.11 | phix | only one clue though |
| 11:11.35 | phix | fail |
| 11:11.41 | phix | that you are still aive |
| 11:11.45 | phix | alive tat is |
| 11:11.53 | phix | I will retify this mistake soon |
| 11:11.58 | Jurian | kaldemar: well, the CLI doesn't show any problems with the calls, the phones ring and when I pick up: SIP/1002-007686c0 answered SIP/31<number> |
| 11:12.01 | phix | where do you live again btw ? |
| 11:12.10 | phix | I want ti give you lots of money |
| 11:14.23 | kaldemar | Jurian: sure the CLI doesn't say anything, but the sip debug might give hints. |
| 11:17.48 | Jurian | kaldemar: here's the relevant parts of my sip.conf and extensions.conf : http://www.fluidic.net/voip.txt |
| 11:19.20 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
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| 11:24.31 | Jurian | added sip debug from the phone and tcpdump of the rtp stream taken at my asterisk server |
| 11:31.15 | Jurian | kaldemar: do you see anything wrong with my config so far? :| |
| 11:39.58 | kaldemar | sip debug from asterisk is the interesting part. |
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| 12:00.25 | dinhtrung | okay, i 've got how to get callerid from PSTN telco in Vietnam |
| 12:00.46 | Jurian | wtf.. with my old (working on another server) asterisk 1.2 config, I have the same problem on this 1.4 :o |
| 12:01.06 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-034817bebd532707) |
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| 12:01.40 | dinhtrung | in my country, the telco gave us DTMF CID |
| 12:01.50 | dinhtrung | then a reverse polarity |
| 12:01.58 | dinhtrung | later, a ringing signal |
| 12:02.12 | dinhtrung | I changed the wait time between DTMF CID and ring signal |
| 12:02.17 | dinhtrung | but this is hard coded |
| 12:02.41 | dinhtrung | is there anyway to configure chan_dahdi to wait a bit longer before decide the channel is hang up? |
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| 12:46.27 | Zhad | I wonder if there's a symbian app that'll let me use my spare handset as an ATA. |
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| 12:57.32 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:58.06 | lftsy | Yello, I'm using the Application ForkCDR to try to save rtcp stats from 2 bridged channels in CDR |
| 12:58.45 | *** part/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:58.47 | jaytee | mornin [TK]D-Fender |
| 12:59.03 | lftsy | so on the first channel at Hungup, I add a Set(CDR(userfield)=${CDR(userfield)}${CHANNEL(rtpqos|audio|all)}\;) |
| 12:59.04 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 12:59.25 | lftsy | but I yould like to do it on the bridged channels too |
| 12:59.37 | lftsy | have you got any idea how to do it please? |
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| 13:04.30 | prxtien | hey all... im having an issue where sometimes people i am calling OR people that call me (including when i call into my menu system) are recieving really bad quality calls... the next call will be perfect, im not sure where to start with the troubleshooting |
| 13:05.25 | [TK]D-Fender | jaytee: mornin' |
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| 13:52.23 | stope | http://pastebin.ca/1439523 line 91, any reason why the sip header is being refused? It's causing the caller id not to show up |
| 13:52.39 | rue_mohr | is there a sip_monitor equiv to the dahdi_monitor ? |
| 13:53.24 | iratik | I need a little help. I have a client who has about 16 users on asterisk 1.2.30.2, they are getting intermittent hangups mid-call. Using the CLI and pri debug i've collected some debug information out of the output -- can you guys see anything that jumps out at you as odd out of the lines I have chosen? http://etherpad.com/yMaS2kb7uG |
| 13:53.30 | KavanS | rue_mohr, chanspy |
| 13:53.38 | KavanS | maybe... |
| 13:53.41 | rue_mohr | hmm |
| 13:53.43 | rue_mohr | k |
| 13:53.48 | rue_mohr | iratik, |
| 13:53.58 | rue_mohr | I recall this problem... |
| 13:54.40 | rue_mohr | its something about signaling and the co not understanding that you picked up the call, so they cut if after an expiry time |
| 13:54.41 | iratik | Can you provide any insight? |
| 13:54.51 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
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| 13:55.17 | iratik | That is something I have not heard about before , what can I do to confirm this is the issue and if I confirm it .. how can I troubleshoot this? |
| 13:55.30 | rue_mohr | I'm tryingto remember anything about it |
| 13:55.37 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
| 13:55.39 | rue_mohr | its happening on a T1 right? |
| 13:55.42 | iratik | Yes |
| 13:55.48 | rue_mohr | k |
| 13:55.59 | rue_mohr | with some calls and not others or all calls? |
| 13:56.05 | iratik | Do you remember some key words from the issue that I can use to seek out the issue on my own? Is there a name for this phenomena ? |
| 13:56.09 | iratik | Its some calls |
| 13:56.22 | rue_mohr | ok, are they long distance calls? |
| 13:56.35 | iratik | Maybe 8-10 out of 100-150 calls/hour and they are long distance calls throughout the US |
| 13:56.45 | rue_mohr | cause i recalls the story goes it was a fault with particular long distance carriers |
| 13:57.21 | rue_mohr | but honestly I'm having a really hard time remembering the details |
| 13:57.22 | iratik | That may explain why it happens in spurts ... more in certain areas of the country than others |
| 13:57.56 | iratik | The clients call geographic regions sequentially, they will call southern florida ... then east north carolina ... so on.. there are periods of time when it constantly happens |
| 13:58.27 | rue_mohr | now you know I dont want to help you if your running a call shop right? :) |
| 13:58.40 | iratik | Does anyone else in the room know what he is talking about? About the carrier not receiving acknowledgement that the call has been picked up. |
| 13:59.37 | rue_mohr | this is a guess, their end dosn't tell your end the call was picked up, asterisk defaults and hangs up. thats a guess, kb1 is the one who experianceed the issue and I dont remember the solution |
| 13:59.45 | rue_mohr | kb1kanobe |
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| 14:00.58 | iratik | ~seen kb1Kanobe |
| 14:01.00 | infobot | i haven't seen 'kb1kanobe', iratik |
| 14:01.08 | rue_mohr | hah |
| 14:01.09 | *** join/#asterisk juanIMP (n=Juancho@200.71.41.254) |
| 14:01.17 | rue_mohr | ~seen kb1canobe |
| 14:01.17 | infobot | rue_mohr: i haven't seen 'kb1canobe' |
| 14:01.23 | rue_mohr | bots too new |
| 14:01.55 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 14:02.08 | rue_mohr | KavanS, no I need a realtime level monitor |
| 14:02.49 | iratik | ztmonitor? |
| 14:03.00 | rue_mohr | unless maybe I record to /dev/dsp and use an analog meter |
| 14:03.13 | rue_mohr | no I need to know the audio levels going ina nd out of the sip phones |
| 14:03.54 | rue_mohr | I'm working on a pile of problems |
| 14:04.47 | [TK]D-Fender | rue_mohr: SIP is expected to be normalized by endpoints so there is no reason for * to do this |
| 14:05.13 | rue_mohr | AND what if the phone ISN"T sending the levels it shoudl!? |
| 14:05.21 | rue_mohr | how the heck am I to know |
| 14:05.39 | rue_mohr | symptom: users complain they cant hear the other end. |
| 14:06.10 | rue_mohr | so I dial up gains, and have all sorts of hwec issues, including it CLIPPING OUT ACTIVE AUDIO |
| 14:06.15 | rue_mohr | !!!?!?! |
| 14:06.23 | anonymouz666 | rue_mohr: use func_volume and be happy. the end. |
| 14:06.31 | rue_mohr | ? |
| 14:07.09 | rue_mohr | whats this!? |
| 14:07.43 | rue_mohr | 1.6.... |
| 14:07.47 | anonymouz666 | no. |
| 14:07.53 | anonymouz666 | I use it in 1.4. |
| 14:07.59 | rue_mohr | Oooo.. |
| 14:08.06 | iratik | Wow i've never seen kb1kanobe in the irclogs, you are always talking about him .. for the last few years... but have never seen him actually in the logs |
| 14:08.36 | rue_mohr | he wrote a good part the wiki article on echo |
| 14:08.54 | rue_mohr | he "wrote" the kb1 echo module |
| 14:09.14 | rue_mohr | (he fixed some major oversights) |
| 14:09.55 | *** join/#asterisk maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 14:10.26 | rue_mohr | I suspect he's still subscribed to some of the mailing lists |
| 14:10.58 | *** join/#asterisk spck (n=spck@unioncab.com) |
| 14:11.01 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 14:11.14 | rue_mohr | ok, you know, being able to adjust gains is great and all, BUT I need to know what the level IS to adjust it properly |
| 14:11.32 | rue_mohr | if you go out to your co too loud on a pots line, you are gonna have major pain |
| 14:11.58 | rue_mohr | I wouldn't be hounding this if I wasn't having a legit issue |
| 14:12.45 | rue_mohr | with a registered tdm800P card, I presume I can call digium and have them help address any audio issues? |
| 14:13.51 | rue_mohr | and having it take over * and # is a problem, I have to use those as keys to control flashing the dahdi channel to get the call waiting calls |
| 14:14.22 | rue_mohr | funny enough, it seems to me that voip is not fit for telephony |
| 14:15.18 | rue_mohr | I dont know one voip set that will show you digits you have dailed while a call is established |
| 14:16.38 | rue_mohr | but in th meantime there are a few basics I have to work out, like levels |
| 14:17.23 | rue_mohr | I'm gonna take in my meter, its got dbm on it, have asterisk send a 1mw to the co's dead end and see what the line levels look like |
| 14:17.24 | mort_gib | rue_mohr: "voip is not ready for telephony" ?? You are having issues with your connection to an Telco, not voip |
| 14:18.04 | rue_mohr | the abscence of a means to 'flash' your call I can understand |
| 14:18.23 | spck | i'm having a helluva time with sip registration |
| 14:18.26 | spck | this is getting annoying |
| 14:18.39 | rue_mohr | the way all the sets and applications I'v seen do not show you digiits you dial after a call has been established is REALLY annoying |
| 14:18.57 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 14:18.58 | rue_mohr | spck, sip show peers |
| 14:19.07 | rue_mohr | see if they have ip addresses beside them |
| 14:19.23 | rue_mohr | if they dont, thats your problem, settings on the phone not registering it right |
| 14:20.01 | rue_mohr | if their aastra phones, the labels on all the settings are effectivly 'random' they mean nothing |
| 14:20.50 | rue_mohr | login name might be call display info, password might be login name, and call display name might be used as host ip |
| 14:20.57 | rue_mohr | as a random example |
| 14:21.21 | rue_mohr | dunno what kinda whackos made the labels on those settings |
| 14:21.58 | rue_mohr | I ended up puting a b c d e f... in for each setting and looking at how the phone tried to register with asterisk |
| 14:22.03 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 14:22.24 | rue_mohr | and the polycom manual sucks |
| 14:23.00 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:23.00 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:23.08 | rue_mohr | "this is our phone, these are things that could be changed, dont touch them" |
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| 14:24.03 | [TK]D-Fender | [10:05]<rue_mohr>AND what if the phone ISN"T sending the levels it shoudl!? <- very low odds |
| 14:24.03 | [TK]D-Fender | rue_mohr: Whats ont he OTHER side of the call? |
| 14:24.03 | rue_mohr | I think their high odds, did you know that a number of remote ivr's cant make out our dtmf |
| 14:24.22 | rue_mohr | I THINK I should move the dtmf generation from cahnnel to info and have asterisk generate it |
| 14:24.30 | rue_mohr | or have it let the card generate it |
| 14:24.43 | [TK]D-Fender | rue_mohr: You mean with the fact you've have psycho gains since forever and jsut can't get over the mental block that that card is a flaming piece of crap and just replace it? |
| 14:24.43 | rue_mohr | on a nortel phone, you just hear "blip" when you dial |
| 14:25.08 | rue_mohr | you dont eralize why I have those gains |
| 14:25.08 | [TK]D-Fender | rue_mohr: You spend all your time compensating for it |
| 14:25.23 | [TK]D-Fender | rue_mohr: does a normal phone work on their lines? |
| 14:25.26 | rue_mohr | and their more reasonable now, 9db and -9db (which is still stupid) |
| 14:25.42 | rue_mohr | yes, standard phones work fine |
| 14:25.49 | rue_mohr | they can hear and dtmf works |
| 14:25.58 | [TK]D-Fender | "more reasonable" = "TRAGIC - 5%" |
| 14:26.16 | rue_mohr | and I can send faxes to associated telephones inc |
| 14:26.16 | [TK]D-Fender | rue_mohr: if normal phones work on the lines then your card is GARBAGE. Deal with it |
| 14:26.20 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 14:26.31 | [TK]D-Fender | rue_mohr: Look at what this is causing. Its ridiculous. |
| 14:26.32 | rue_mohr | do I really shoudl call digium on this re the card |
| 14:27.15 | rue_mohr | besides, if it wasn't the card, how the hell would I know, I have no way to know if its calibrated properly |
| 14:27.32 | rue_mohr | cause dahdi_monitor has no real units |
| 14:27.35 | [TK]D-Fender | rue_mohr: We KNOW your card is crap. |
| 14:27.48 | rue_mohr | so all TDM800P cards are crap? |
| 14:27.48 | [TK]D-Fender | rue_mohr: This has been established for what, almost a year now? |
| 14:27.57 | [TK]D-Fender | rue_mohr: Yours at the very least |
| 14:28.39 | rue_mohr | ok, what if the problem is the polycom phones, if you search you WILL find a PILE of people compaining about the audio being too quiet on the 601's |
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| 14:29.16 | rue_mohr | I need to know more about "chassis gain" on the polycoms |
| 14:29.30 | rue_mohr | I think its the dtmf/busy etc |
| 14:29.37 | rue_mohr | but the manual dosn't say |
| 14:29.45 | [TK]D-Fender | rue_mohr: this is just wrong. We've known the culpit and you are desperately trying to poitn the finger everywhere else. |
| 14:29.54 | rue_mohr | they dont say anything about the gains on the phone at all |
| 14:30.02 | [TK]D-Fender | rue_mohr: And yes... they do. |
| 14:30.19 | [TK]D-Fender | rue_mohr: the gains are prtty evident in the provisioning. |
| 14:30.46 | rue_mohr | no, there is analog and digital for transmitt and recieve, and theirs chassis |
| 14:31.00 | jaytee | personally I enjoy watching a dead horse being beaten repeatedly :-) |
| 14:31.02 | rue_mohr | that dosn't form a picture in my mind thats verry clear |
| 14:31.05 | [TK]D-Fender | rue_mohr: And the only time things like side-tone have acted up (was the IP 430), new defaults & firmware came corrected stock from the provisioning |
| 14:31.20 | [TK]D-Fender | jaytee: unload chan_brokenrecord.so |
| 14:31.34 | jaytee | :-) |
| 14:31.46 | rue_mohr | i9f it worked I wouldnt be here |
| 14:32.01 | rue_mohr | if I could diagnose it using whats available I wouldn't be there |
| 14:32.03 | rue_mohr | or here |
| 14:32.24 | [TK]D-Fender | rue_mohr: I've never heard any complain on IP 601's ever. Your card is faulty. Please get some "help" and replace it. |
| 14:32.37 | [TK]D-Fender | OH LOOK, THERE'S AN ELEPHANT IN THE ROOm! |
| 14:32.40 | rue_mohr | I cant say what eh levels are comming off the adc of that card |
| 14:32.52 | rue_mohr | I cant say what the levels are comming out the hwec on the card |
| 14:33.01 | rue_mohr | I cant say what the levels are comming from the sip phones |
| 14:33.23 | rue_mohr | I cant be sure the level comming out of the hwec of the card is the same as the level going to the sip phone |
| 14:33.29 | [TK]D-Fender | rue_mohr: You admit the gains are still nuts. What is it going to take? A giant flashing neon sign? |
| 14:33.37 | rue_mohr | I cant say what hte level is going out the card to the pots |
| 14:33.45 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 14:33.47 | rue_mohr | I want to know how to fix it |
| 14:33.51 | coppice | [TK]D-Fender: nah. the big grey things are clouds. we've had them all week |
| 14:33.57 | jaytee | giant flashing neon sign? hmmmmm.....brb |
| 14:33.58 | rue_mohr | the gains are 'educated guesses' |
| 14:34.21 | coppice | actually, the educated wouldn't be guessing |
| 14:34.22 | rue_mohr | I want to know what number relates to 0dbm on the dahdi_monitor |
| 14:34.52 | rue_mohr | I could dial the pots 1mw and see if the level is right |
| 14:35.39 | rue_mohr | I can hook my meter to the telco line and know how far off the 1mw is comming in |
| 14:35.50 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 14:35.55 | rue_mohr | in my house its out by about 2dbm |
| 14:36.00 | [TK]D-Fender | rue_mohr: How much anguish is that busted-up card worth? |
| 14:36.19 | rue_mohr | $800 |
| 14:36.41 | *** join/#asterisk wpbrown (n=wpbrown@wh-gtw-0001.woolfharris.com) |
| 14:37.06 | [TK]D-Fender | Whats your "pay-tp-self" value for time you throw away futilely trying to make it work? |
| 14:37.11 | coppice | 2dB is what's technically known as bugger all |
| 14:37.11 | [TK]D-Fender | to* |
| 14:37.21 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
| 14:37.28 | rue_mohr | yup, and at my house it is |
| 14:37.51 | jaytee | http://tinyurl.com/ltaowx |
| 14:37.54 | rue_mohr | I also know I have a bit of loss, casue my house is all analog and when bridged the signal at the set is a little lower than the signal on the pots |
| 14:39.11 | rue_mohr | my house is another story though, its problems aren't * at all |
| 14:40.18 | rue_mohr | in chan_dahdi, the gains are not done within asterisk are they, their passed on to the interface, T1 card, or tdm card yes? |
| 14:40.28 | wpbrown | I have a quick question. I have a asterisk box with a Sangoma PRI card. Using 23b and a d for signaling. I am seeing a problem with cell users. It is like Asterisk isn't reconizing the touch tone from cell phones. Especially Blackberry and Iphone. Anyone else had any experience with this? |
| 14:40.29 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:40.57 | rue_mohr | aka, if the gain in asterisk are all set to 0db and there is no volume() used, asterisks gain in 0db right? |
| 14:41.09 | wpbrown | Think it could be weak cell signal? It isn't this wall ALL of the time but I would say 50/50 |
| 14:41.25 | rue_mohr | hehe level problems |
| 14:41.30 | rue_mohr | wpbrown, sip phones? |
| 14:41.33 | rue_mohr | with agc? |
| 14:41.50 | rue_mohr | use dahdi_monitor to see what your incomming levels are like |
| 14:41.52 | rue_mohr | :) |
| 14:42.06 | wpbrown | I am running SIP internal yes.. but I am refering to cell phone users calling in across the PRI |
| 14:42.12 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 14:42.33 | rue_mohr | dont ask what is quiet and what isn't, casue there is really no way to know |
| 14:42.33 | wpbrown | when they dial a extension Asterisk says invalid extention or transfers to zero by default |
| 14:42.50 | rue_mohr | but it gives you a vu meter you can get an idea from |
| 14:42.58 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 14:43.02 | rue_mohr | all cell phones |
| 14:43.04 | [TK]D-Fender | wpbrown: What do you see on core debug for their entries? |
| 14:43.05 | rue_mohr | ? |
| 14:43.38 | wpbrown | I haven't ran it in a while. As of now debugging is disabled. In the past everything seems normal. |
| 14:43.49 | wpbrown | in the debug logs |
| 14:44.31 | wpbrown | Do you think it could be a issue with the PRI itself? |
| 14:45.19 | [TK]D-Fender | wpbrown: It hink there is a certain bare minimum you could do to see how many digits on a call get lost |
| 14:45.57 | [TK]D-Fender | wpbrown: Very easy to make a test IVR that reads back DTMF as entered so you can collect stats and test. Maybe your telco is running hot/cold.... |
| 14:46.06 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
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| 14:48.09 | afink | hello everyone, I have a strange problem. Every time I reboot asterisk the T1 doesn't come back online. The light on the card shows green but I always get circuit channel congestion. I am using asterisk-gui, and it seems deleting re-adding works sometimes. Output from the cli -> http://pastebin.com/m573c0e07 |
| 14:49.07 | afink | I have asked in #asterisk-gui already and am waiting for an answer, but if someone here knows a quick fix it would be appreciated since I have a phone server that is currently not working. |
| 14:49.35 | [TK]D-Fender | afink: I've seen several PRI's return ISDN 34 when the number you are calling is busy |
| 14:49.42 | [TK]D-Fender | afink: Could be perfectly normal |
| 14:50.16 | *** join/#asterisk jon_farmer (n=chatzill@195.74.96.119) |
| 14:50.24 | eppigy | It can also be a capacity issue |
| 14:50.29 | eppigy | on the telco end |
| 14:50.45 | afink | wow....I restarted dahdi and it started working. |
| 14:50.48 | eppigy | I had that happen using GC ld in St Louis |
| 14:52.41 | afink | thanks everybody. |
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| 15:19.16 | nephfl | which is better cisco/linksys ata or grandstream? |
| 15:19.58 | bmoraca | is it possible to have multiple SIP peers register with the same credentials? which value under that sip peer would I use for that? |
| 15:20.40 | srf21c | nephfl: hard to make a blanket statement like that. I think it would depend somewhat on the models being compared and what criteria are most important to you. |
| 15:21.02 | srf21c | I would imagine a top of the line cisco, will be "better" than a top of the line grandstream. |
| 15:21.25 | srf21c | but how much is the price premium worth to you? That's in individual decision. |
| 15:21.29 | bmoraca | bottom of the barrel Cisco is superior to top of the line grandstream |
| 15:22.01 | Chainsaw | Linksys is being included here. |
| 15:22.15 | Chainsaw | Note that. Actual Cisco will be better then Grandstream for sure. Linksys, not so sure. |
| 15:22.33 | nephfl | i need autodial on pickup as well, dont know if one is better than the other in any applicaton |
| 15:22.52 | [TK]D-Fender | [11:19]<bmoraca>is it possible to have multiple SIP peers register with the same credentials? which value under that sip peer would I use for that? <- yes, but the last one to reg "wins" |
| 15:23.00 | rbd | hey guys... having problems playing mp3s in asterisk via AGI STREAM FILE, GET DATA, etc.... via Playback in the dial plan it works fine though.... anyone had this problem? |
| 15:23.06 | srf21c | nephfl: my personal favorite brand of IP phone is Snom. |
| 15:23.09 | [TK]D-Fender | Linksys is a ton better than GS |
| 15:23.10 | rbd | seems like format_mp3 is broken in some regards |
| 15:23.15 | [TK]D-Fender | Polycom > All |
| 15:23.35 | *** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1096579290.dsl.bell.ca) |
| 15:24.18 | bmoraca | [TK]D-Fender: I have a SIP client that needs to register to multiple peers, but I can only specify a single username and password without jumping through hoops. is there no way to have all of those peers register with those same username and password? looking at the "username" attribute with the type set to friend or user seems like it might work... |
| 15:24.51 | bmoraca | reason i ask is because there could potentially be hundreds of peers |
| 15:24.52 | [TK]D-Fender | bmoraca: you can't have devices finght over a registration. * Will not track multiple enpoints. |
| 15:24.52 | *** part/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1096579290.dsl.bell.ca) |
| 15:25.11 | [TK]D-Fender | bmoraca: I think I've heard SEr doing this |
| 15:25.14 | [TK]D-Fender | SER* |
| 15:25.19 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:25.33 | bmoraca | [TK]D-Fender: I don't want multiple devices to register to the same peer. I want a single device to register to multiple peers with a single username and password. |
| 15:25.34 | [TK]D-Fender | bmoraca: At which point you could use that in front |
| 15:25.42 | bmoraca | yeah...i was afraid i might have to do that |
| 15:25.59 | [TK]D-Fender | bmoraca: Umm.. single device tor eg to multiple peers? Like what kind of device? |
| 15:26.09 | bmoraca | it's an Adtran media gateway |
| 15:26.37 | [TK]D-Fender | bmoraca: I am confused.... if you what THAT to reg to multiple devices.. well then read its damn manual :) |
| 15:26.58 | [TK]D-Fender | bmoraca: this has nothing to do with * |
| 15:27.12 | bmoraca | it's strange the way it works...every phone number you configure it with registers as a peer to the SIP gateway, but you can only specify a single username and password for ALL registrations |
| 15:27.44 | *** join/#asterisk CrashSys (n=james@azrael.crashsys.com) |
| 15:28.10 | bmoraca | [TK]D-Fender: it's not the device that I'm having problems with. it's Asterisk's configuration that I'm not sure of. I've got the Adtran configured fine, but I cannot figure out how to get asterisk to accept a username that's not the peer name |
| 15:28.17 | bmoraca | let me pastebin an example |
| 15:29.22 | nephfl | anybody know the difference betwen spa2102 and pap2t-na? |
| 15:29.38 | CrashSys | I'm having an issue with Asterisk 1.4 where if I have anything set in caller-id-name when I try to dial out my PRI is returning cause code 99 which causes asterisk to hang-up... Is there any way to configure the PRI to just not sent caller-ID-name without blanking that variable in the dialplan? |
| 15:31.01 | [TK]D-Fender | bmoraca: Accept un-authed calls, and validate that its from your gateway in dialplan. Or auth by IP only. |
| 15:31.19 | [TK]D-Fender | nephfl: The featuts list is pretty clear |
| 15:31.23 | [TK]D-Fender | features* |
| 15:31.34 | nephfl | i dont see the difference |
| 15:31.36 | nephfl | sorry |
| 15:32.22 | CrashSys | Would calling SetCallerPres(allowed) before the dial do what I need? |
| 15:32.31 | [TK]D-Fender | nephfl: then either you're not looking at a good list or you're jsut not looking... |
| 15:32.33 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 15:32.40 | coppice | nephfl: T.38 support and a router are the key differences |
| 15:32.54 | [TK]D-Fender | nephfl: T.38 support, Acts as router, bgiger CPU, etc... |
| 15:33.45 | nephfl | if im not using it as a router is there any advantage to the spa? |
| 15:34.20 | coppice | let's see. T38 + router - router => ??? |
| 15:34.24 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 15:34.44 | [TK]D-Fender | nephfl: If I'm not driving 200mph on the autobahn in my Ferrari is it still a better choice to use for my grocery runs? |
| 15:35.15 | coppice | Ferraris suck for grocery shopping. no luggage space |
| 15:35.15 | [TK]D-Fender | nephfl: I think you need to be able to look at the fature list and make up your own mind |
| 15:35.37 | [TK]D-Fender | coppice: Yes, but at 200mph think about how many trips you could do :) |
| 15:35.51 | *** join/#asterisk iratik (n=ctatechs@74-84-99-12.client.mchsi.com) |
| 15:37.25 | CrashSys | I find going 140 in my 4-door family sedan on the interstate to be much more effective |
| 15:37.45 | CrashSys | comfortable seats, gentle ride, and cops are never looking for you to be speeding in that kind of car :) |
| 15:38.22 | CrashSys | They're usually looking for the ferrari's |
| 15:38.26 | mnicholson | some ferraris have luggage space |
| 15:38.43 | coppice | you mean their station wagons? |
| 15:39.26 | CrashSys | The passenger seat is nog luggage space |
| 15:39.31 | mnicholson | the 612 scaglietti, 599 fiorano, and the ferrari california all have some luggage space |
| 15:39.37 | iratik | Can anyone help me figure out this pri debug information, it is at the end of the pastie and it is a disconnect message - i cannot tell if it is incoming or outgoing and i am having trouble understanding what it means... (Selected debug lines , with the pri packet at the end) http://pastie.org/494029 |
| 15:40.30 | CrashSys | Cause code 16, normal cleaning |
| 15:40.37 | CrashSys | Typically means the remote end hung up |
| 15:40.45 | CrashSys | as in went on-hook |
| 15:41.17 | iratik | For sure? "Private network serving the local user" .. that couldn't mean that the hangup was caused by a local netork disruption? |
| 15:41.23 | bmoraca | [TK]D-Fender: here's a PB: http://pastebin.com/m7eb7d82d . If I send the credentials for 2095545245, that peer registers, but 2095545246 also attempts to register with those credentials and gives me back a "SIP/2.0 403 Authentication user name does not match account name". |
| 15:41.28 | iratik | > Call Ref: len= 2 (reference 104/0x68) (Originator) |
| 15:42.04 | iratik | Does that mean that the remote end hung up for sure ? Or is there any other possibilities, ... this is the greatest amount of debug i can do for the dropped calls |
| 15:42.08 | CrashSys | Ahhh, Vicidial :) |
| 15:42.37 | iratik | You saw the "Local/8600052" |
| 15:42.43 | CrashSys | well, maybe not, but something similar |
| 15:42.49 | CrashSys | Yeah |
| 15:42.57 | iratik | Any help , ideas? directions to explore? |
| 15:43.09 | [TK]D-Fender | bmoraca: Inded the name does NOT match in their own register. Look at it. |
| 15:43.18 | [TK]D-Fender | bmoraca: Looks like the gateways is configured wrong |
| 15:43.35 | [TK]D-Fender | bmoraca: 19 vs 30 |
| 15:44.01 | CrashSys | that's the whole d-channel debug? |
| 15:44.32 | iratik | The whole d-channel debug would be 20+ pages for the length of the entire call as this is happening during 12-16 simultaneous calls |
| 15:44.34 | *** join/#asterisk ingenius (n=alektro@netsolution.com.ar) |
| 15:44.42 | bmoraca | [TK]D-Fender: this is the only way to configure the gateway. what I need asterisk to do is allow me to specify a different auth name from the peer name... |
| 15:44.55 | CrashSys | Yeah, but the PRI debug looks truncated |
| 15:45.17 | bmoraca | if asterisk can't do that, I'll use SER in front of it...but I'd prefer not to do that |
| 15:45.37 | [TK]D-Fender | bmoraca: I can't really advise on any way for * to do this. |
| 15:45.38 | iratik | CrashSys: I have another PRI debug set, let me paste it |
| 15:45.49 | CrashSys | What version of stuff is this? |
| 15:46.38 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 15:46.45 | [TK]D-Fender | [11:39]<CrashSys>The passenger seat is nog luggage space <- transporter 3 begs to differ ;) |
| 15:47.14 | CrashSys | Ok, if you are Jason Stratham then yes, it's a viable option... |
| 15:47.16 | *** join/#asterisk b14ck (n=comradeb@72.37.252.50) |
| 15:47.17 | b14ck | hi all :) |
| 15:47.25 | CrashSys | BTW, go see crank 2... the godzilla scene is worth the money... |
| 15:47.43 | luckyaba | Has anyone had any trouble hooking Nortel phones up with Asterisk? |
| 15:48.07 | CrashSys | iratik: All the messages that have a > as the first character in a debug means that message is being delivered to the asterisk server... |
| 15:48.19 | luckyaba | specifically the Nortel 1140E |
| 15:48.21 | [TK]D-Fender | CrashSys: a match for #1's "Chinatown" scene? ;) |
| 15:48.38 | CrashSys | And inversely, all messages that have a < as the first character in a debug are messages going out from asterisk |
| 15:49.00 | CrashSys | D-Fender: Better in my opinion... |
| 15:49.01 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 15:49.14 | [TK]D-Fender | luckyaba: Never heard of anyone with those loking to use * |
| 15:49.20 | CrashSys | Although, they duplicate the chinatown scene at a horse race track :) |
| 15:49.21 | *** join/#asterisk jnfuller (n=jnfuller@99.199.170.110) |
| 15:49.30 | CrashSys | The movie makes fun of how ridiculous it is... |
| 15:49.33 | [TK]D-Fender | CrashSys: I'm sure its craptastic, but hopefully not overly painful to watch |
| 15:50.13 | CrashSys | It was seriously pretty funny... the first one attempted to be somewhat serious... the second one gives up and just makes fun of itself the whole time... |
| 15:50.18 | iratik | Here is the _full_ cli output , the sections involving the progress of the call that dropped are marked (where the line numbers reset) http://pastie.org/494072 |
| 15:50.24 | jnfuller | are the dahdi-$foo-current.tar.gz on downloads.asterisk.org being refreshed right now? the tarballs seem to be hosed/ |
| 15:50.45 | iratik | (The call starts on 269/1) |
| 15:52.39 | *** join/#asterisk JackEStorm (n=no@ip24-252-118-155.no.no.cox.net) |
| 15:53.26 | *** join/#asterisk Black_L (n=chatzill@wsip-98-175-64-147.ga.at.cox.net) |
| 15:53.26 | CrashSys | So what asterisk version is this? |
| 15:53.33 | iratik | 1.2.0.3 ? |
| 15:53.41 | Black_L | Can i install Asterisk under Ubuntu? |
| 15:53.43 | iratik | Its 1.2.30.2 |
| 15:53.51 | CrashSys | ohh, ok, I was about to say |
| 15:54.07 | jaytee | Black_L, no but you can install Asterisk on top of Ubuntu |
| 15:54.19 | Black_L | jattee : Expalin |
| 15:54.23 | Black_L | jaytee* |
| 15:54.29 | bmoraca | damnit...i didn't want to have to dedicate another server to this, but it looks like I don't have a choice |
| 15:54.38 | jnfuller | they are hosed. asterisk-dev says someone is fixing them |
| 15:54.43 | Black_L | bmoraca : Virtualize it |
| 15:54.46 | bmoraca | too difficult to maintain otherwise |
| 15:55.07 | bmoraca | Black_L: probably not going to be a possibility with how much traffic I hope to have on it |
| 15:55.10 | jaytee | Black_L, just a figure of speech, yes you can run Asterisk ON ubuntu OR under it it if you prefer :-) |
| 15:55.19 | CrashSys | looks like it hangs-up when it tries to bridge it to the local channel |
| 15:55.22 | CrashSys | is meetme working? |
| 15:55.27 | iratik | Yes. |
| 15:55.27 | Black_L | jaytee : Thank you |
| 15:55.33 | iratik | But the call is already in progress |
| 15:55.59 | jaytee | lunchtime, bbiab |
| 15:56.12 | CrashSys | You can dial 8600051 on a phone and get into a conference with no issues? |
| 15:57.06 | iratik | Yeah |
| 15:57.19 | CrashSys | This a new issue? |
| 15:57.21 | iratik | I mean there are 1,000s of calls takign places over 20+ conferences every day |
| 15:57.27 | CrashSys | or you just setting the system up? |
| 15:57.34 | iratik | its been a persistent issue since the beginning |
| 15:57.47 | CrashSys | Verified PRI protocol with carrier? |
| 15:58.03 | *** join/#asterisk chendy (n=chatzill@59.40.164.130) |
| 15:58.12 | iratik | 8-10 calls out of 100 calls/hour get "dropped" like this.. but from the pri debug it looks like a remote hangup |
| 15:58.19 | *** part/#asterisk jnfuller (n=jnfuller@99.199.170.110) |
| 15:58.21 | iratik | thats why i was verifying if you are sure it was a remote hangup |
| 15:58.38 | iratik | AT&T is the carrier |
| 15:58.50 | *** join/#asterisk mchou_ (n=mchou@unaffiliated/mchou) |
| 15:59.56 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 16:00.05 | *** part/#asterisk lanning (n=lanning@173.8.187.197) |
| 16:00.09 | iratik | AT&T has never had an open-source friendly tone when we've brought up asterisk. They wwill basically only talk to us about their certified telephony solutions .. avaya, cisco etc... |
| 16:00.26 | CrashSys | Is switchvox certified? |
| 16:00.28 | CrashSys | or trixbox? |
| 16:00.31 | CrashSys | if so, say it's that |
| 16:00.42 | CrashSys | volunteering information willingly is never a good idea :) |
| 16:00.56 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 16:01.10 | JackEStorm | hey how do I turn on sql tracing for res_odbc? it keeps crashing here randomly once a week (with a stuid sql error:: ERROR: value "<CID NUMBER>" is out of range for type integer;) |
| 16:01.23 | *** join/#asterisk Joe_CoT (n=joecot@ubuntu/member/pdpc.bronze.joeterranova) |
| 16:01.36 | Joe_CoT | ok. So yesterday, the power went out. Today, meetme won't work |
| 16:01.57 | Joe_CoT | i have ztdummy loaded into the kernel, and I just recompiled asterisk. no dice |
| 16:02.52 | Black_L | Ok i could use some help here. |
| 16:03.18 | Black_L | I have no idea how to use Linux. Downloaded Asterisk in my virtualized Ubuntu box and now i would like to know how to install the app. |
| 16:03.32 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 16:03.41 | Black_L | Also is there any way to replace this horrific GUI Ubuntu has going on by default? |
| 16:04.36 | timeshell_atwork | Yep |
| 16:04.38 | timeshell_atwork | Use Fedora |
| 16:04.51 | timeshell_atwork | :D |
| 16:04.52 | Joe_CoT | Black_L, http://www.joeterranova.net/wiki/index.php?title=Install_Asterisk |
| 16:05.04 | Joe_CoT | you can stop before the setup freepbx part if you're just doing asterisk |
| 16:05.20 | Joe_CoT | Black_L, if you don't want a gui, you probably wanted ubuntu server |
| 16:05.40 | Black_L | I wanted a GUI |
| 16:05.43 | Black_L | But i want it to not suck |
| 16:06.03 | Joe_CoT | Black_L, well what do you want? KDE? XFCE? |
| 16:06.20 | Black_L | I know very little about Linux. But i am quite partial to the Vista or 7 theme. |
| 16:06.32 | Black_L | Black glass is something i wish the entire world was made of lol. |
| 16:06.55 | Joe_CoT | look at screenshots of Kubuntu and Xubuntu. Let me know if either of those look better to you |
| 16:07.03 | Black_L | I've used Kubuntu |
| 16:07.05 | Black_L | Horrible |
| 16:07.09 | Black_L | But i'll look at Xubuntu |
| 16:07.45 | Black_L | They all look the same except Kubuntu |
| 16:08.30 | CrashSys | iratik: Which call should I be looking at? |
| 16:08.37 | CrashSys | the one for conference 8600051? |
| 16:09.10 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:09.15 | Joe_CoT | Black_L, ok. Kubuntu and Xubuntu are the supported ones. If neither of those work for you, there are others, mostly minimalist ones. Regarding guis, you should probably PM me, it's not an asterisk thing |
| 16:09.33 | Black_L | Back to Asterisk then |
| 16:09.47 | Black_L | To install it i would use what command now? |
| 16:10.09 | Black_L | "sudo make samples" |
| 16:10.11 | Black_L | ? |
| 16:10.43 | Joe_CoT | Black_L, there's a package built in to Ubuntu, but it's not kept up to date. To compile it from source, run through everything here: http://www.joeterranova.net/wiki/index.php?title=Install_Asterisk#Install_Asterisk |
| 16:11.05 | Joe_CoT | if you want to have a go with the package, sudo apt-get install asterisk |
| 16:11.38 | Black_L | Where do i download the latest package? I think i have source. |
| 16:12.07 | *** join/#asterisk youngproguru (n=mm@74.10.229.45) |
| 16:12.16 | Black_L | Also, i am curious. Why doesn't Linux adopt an installer system utilizing executables like Windows? |
| 16:12.32 | nephfl | like rpms? |
| 16:12.39 | youngproguru | I have such a quick question I hope someone can help with |
| 16:12.46 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 16:12.57 | youngproguru | I have installed Asterisk Now 1.5 and Asterisk doesn't seem to start at system boot? |
| 16:13.05 | youngproguru | <PROTECTED> |
| 16:13.12 | youngproguru | <PROTECTED> |
| 16:13.19 | youngproguru | and I am not using the Gui's in any way. |
| 16:13.26 | youngproguru | Thoughts? |
| 16:13.28 | Joe_CoT | Black_L, what I just gave you is the equivalent of an install. If you want to install the latest package on Ubuntu: |
| 16:13.31 | Joe_CoT | sudo apt-get update |
| 16:13.35 | Joe_CoT | then, sudo apt-get install asterisk |
| 16:13.37 | Joe_CoT | that's it |
| 16:13.41 | Black_L | Ok |
| 16:14.03 | youngproguru | Of Course, I could add the init myself, but I have to assume that Asterisk Now would start asterisk at boot? |
| 16:14.06 | *** join/#asterisk mog (n=mog@c-68-62-174-19.hsd1.al.comcast.net) |
| 16:14.06 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:14.15 | youngproguru | At least It did in the past |
| 16:14.25 | Black_L | ...How do i open the command window? |
| 16:14.59 | Joe_CoT | Black_L, Applications -> Accessories -> Terminal |
| 16:15.10 | Black_L | Thank you very much |
| 16:16.34 | *** join/#asterisk smultron (n=smultron@cpe-67-9-150-163.austin.res.rr.com) |
| 16:17.23 | *** join/#asterisk nkohh (n=justin@unaffiliated/kohh) |
| 16:17.35 | Black_L | Is there a way to have Ubuntu automatically download the driver for my GPU? |
| 16:18.20 | nkohh | that question might be better placed in a channel that is NOT #asterisk |
| 16:18.46 | Joe_CoT | Black_L, you can try #ubuntu. Or I can help you in PM, or in #ubuntu-us-nj |
| 16:19.21 | Joe_CoT | #asterisk is about asterisk. Speaking of which, any help with my meetme problem would be appreciated |
| 16:19.27 | nkohh | whats your meetme problem |
| 16:20.28 | Joe_CoT | power went out last night, now meetme won't work at all. I just get kicked out (or disconnected) as soon as I call it. usually I know this is a zaptel problem, but I have ztdummy loaded, and I just recompiled asterisk, but no dice |
| 16:20.49 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 16:22.34 | Black_L | Ok i used the command "sudo apt-get install asterisk" |
| 16:22.52 | Black_L | Is it installed? How do i access it? |
| 16:22.54 | nkohh | Joe_CoT: any logged messages? |
| 16:24.21 | Joe_CoT | nkohh, http://asterisk.pastebin.com/m3b80d6ea |
| 16:24.31 | Joe_CoT | Black_L, yes it should be installed, it should even be running |
| 16:24.36 | Joe_CoT | you can access it with sudo asterisk -r |
| 16:24.46 | Joe_CoT | my question is, do you know what do to with it |
| 16:25.13 | nkohh | Joe_CoT: thanks, I'll take a look |
| 16:25.23 | Black_L | Unable to connect to remote asteris |
| 16:25.25 | Black_L | asterisk |
| 16:25.28 | Black_L | I know little |
| 16:25.32 | [TK]D-Fender | Joe_CoT: Since not knowing how to even open a terminal window or start it... I'd bet on "no" |
| 16:25.41 | Joe_CoT | Black_L, try sudo /etc/init.d/asterisk start |
| 16:26.01 | Joe_CoT | [TK]D-Fender, yeah. So he'd probably be better suited with one of the specialized graphical setups |
| 16:26.07 | Joe_CoT | or reading, a lot |
| 16:26.12 | Black_L | Says it is already running |
| 16:27.54 | *** join/#asterisk nny_1 (n=scott@64.203.244.146) |
| 16:27.58 | Joe_CoT | nkohh, think I found it in the full log: app_meetme.c: Unable to open pseudo device |
| 16:28.56 | nny_1 | still working on possible creative solutions to my experimental personal macro for calling my desk/cell phone. Right now I mangle the CID to be a specific number before using Dial to diall my deskphone&cellphone@SIPProvider. |
| 16:29.13 | nny_1 | <PROTECTED> |
| 16:29.25 | nny_1 | wondering if there is a way to only set the CID for the cellphone part |
| 16:29.26 | VaGoNeTaS | y |
| 16:29.39 | nny_1 | VaGoNeTaS: me? |
| 16:30.12 | VaGoNeTaS | shit still got the same issue |
| 16:30.26 | VaGoNeTaS | maybe you could help me |
| 16:30.38 | nny_1 | possible but not likely, i burn toast, but shoot |
| 16:30.46 | VaGoNeTaS | hahaha, k |
| 16:31.09 | VaGoNeTaS | i got an asterisk server running on the ip address 192.168.1.27 , and for the people on the same subnet, it works properly |
| 16:31.17 | VaGoNeTaS | (with wired connection) |
| 16:31.34 | Joe_CoT | VaGoNeTaS, but outside that it can't get the public ip address? |
| 16:31.38 | VaGoNeTaS | but i also have an wireless running on the subnet 192.168.2.xx |
| 16:31.47 | VaGoNeTaS | at the same office |
| 16:31.51 | Joe_CoT | oh, nm |
| 16:31.59 | VaGoNeTaS | and when they try to talk, people cant listen them |
| 16:32.14 | nny_1 | how are you routing the two subnets together? |
| 16:32.18 | nny_1 | at the gateway? |
| 16:32.31 | VaGoNeTaS | what u mean? |
| 16:32.40 | VaGoNeTaS | wanna see my SIP.conf file ? |
| 16:32.42 | VaGoNeTaS | just a sec |
| 16:32.43 | nny_1 | no |
| 16:32.47 | VaGoNeTaS | im gonna pb it |
| 16:32.48 | Joe_CoT | VaGoNeTaS, so it connects, but voice doesn't happen? The connection is through SIP, the voice is through RTP. The RTP packets probably aren't making it through |
| 16:32.56 | VaGoNeTaS | Joe_CoT : they can listen |
| 16:33.00 | nny_1 | VaGoNeTaS: are you using the wireless as a router? |
| 16:33.04 | VaGoNeTaS | but people at the other side cant listen them |
| 16:33.22 | nny_1 | VaGoNeTaS: cause if you have ROUTER --> ASTERISK --> ROUTER WITH WIFI --> CLIENT< it's a NAT issue |
| 16:33.29 | VaGoNeTaS | or if they do, people listen them breaking up |
| 16:33.41 | nny_1 | VaGoNeTaS: why do you have a diff subnet for wifi? |
| 16:33.44 | VaGoNeTaS | we have, a fw, then a switch |
| 16:33.56 | VaGoNeTaS | ok, 24 ports switch (subnet 192.168.1.xx) |
| 16:34.01 | nny_1 | ok but how is the 2.X subnet talking to the .1 subnet? |
| 16:34.11 | VaGoNeTaS | and coming out of that router we have connected to, the wireless router |
| 16:34.16 | nny_1 | bad juju |
| 16:34.24 | nny_1 | is the wireless router routing? |
| 16:34.31 | VaGoNeTaS | i've tried to change the subnet of that router but it doesnt works |
| 16:34.38 | nny_1 | as in did you plug the 1.X network into the WAN port!?: |
| 16:34.43 | nny_1 | do this: |
| 16:35.02 | VaGoNeTaS | the worse part of this story is that i have to fix it TODAY |
| 16:35.08 | VaGoNeTaS | or my ass will be burned slowly |
| 16:35.14 | nny_1 | VaGoNeTaS: if you listen it will work |
| 16:35.23 | VaGoNeTaS | k, im listening |
| 16:35.35 | nny_1 | VaGoNeTaS: first of all the NAT --> NAT thing is never a good idea really |
| 16:35.51 | *** join/#asterisk viraptor (n=viraptor@87-194-164-154.bethere.co.uk) |
| 16:35.54 | nny_1 | VaGoNeTaS: right now you have a NAT routrer behind a NAT router |
| 16:35.56 | VaGoNeTaS | i know, but is the boss requirement |
| 16:36.11 | nny_1 | VaGoNeTaS: so you have to seperate the wireless from the main network? |
| 16:36.12 | VaGoNeTaS | nat router behind a nat switch |
| 16:36.22 | nny_1 | yeah thats not a good idea, but eh |
| 16:36.46 | VaGoNeTaS | if i were the boss, ill have it different but im not |
| 16:37.27 | Black_L | Ok i have Asterisk running under Ubuntu. How do i access Asterisk? |
| 16:37.35 | nny_1 | VaGoNeTaS: well. you can try to forward the RTP ports on the wireless AP |
| 16:37.42 | nny_1 | VaGoNeTaS: it may work, but not sure |
| 16:38.03 | nny_1 | VaGoNeTaS: but if you have more than one client on the second NAT router, it wont work |
| 16:38.12 | nny_1 | VaGoNeTaS: tell your boss this setup isn't correct |
| 16:38.15 | VaGoNeTaS | http://pastebin.ca/1439770 |
| 16:38.26 | nny_1 | VaGoNeTaS: the issue isn't asteriskl |
| 16:38.43 | VaGoNeTaS | the issue is the NAT i know that |
| 16:38.43 | nny_1 | VaGoNeTaS: the issue is the incoming RTP packets aren't making it across NAT to the sip client |
| 16:38.54 | JackEStorm | VaGoNeTaS: turn reinvites off |
| 16:39.08 | VaGoNeTaS | JackEStorm : what reinvites? |
| 16:39.12 | VaGoNeTaS | wtf |
| 16:39.34 | nny_1 | so think of it logically, if you cant mangle the packets to their destination(s) you won't get two way audio |
| 16:39.49 | nny_1 | VaGoNeTaS: and having local=192.168.2.X is pointless |
| 16:39.50 | JackEStorm | VaGoNeTaS: in sip.conf |
| 16:40.03 | VaGoNeTaS | reinvites=off |
| 16:40.03 | VaGoNeTaS | ? |
| 16:40.10 | viraptor | is there a way to jump to a context only if it exists? I mean something like GotoIfExists(context-if-exists,context-else,${EXTEN},1) ? |
| 16:40.12 | nny_1 | since all packets from the wireless router are forwarded as the single IP the router has as a WAN address |
| 16:40.15 | nny_1 | gives up |
| 16:40.22 | nny_1 | have fun with that |
| 16:40.29 | JackEStorm | VaGoNeTaS: canreinvite=no |
| 16:40.38 | VaGoNeTaS | nny_1 : belive me im having fun with this since 1 week and a half |
| 16:40.48 | VaGoNeTaS | and my ass is getting burned really really painfully |
| 16:40.53 | nny_1 | VaGoNeTaS: your issue is a bad network setup |
| 16:41.14 | nny_1 | VaGoNeTaS: deal with that and you have your solution. I might add you may wanna read up on how a NAT network works |
| 16:41.17 | VaGoNeTaS | JackEStorm : what's that for? |
| 16:41.35 | Joe_CoT | nny_1, if i'm not mistaken, isn't it 1 rtp port per stream? If he has 100 ports, I don't see the problem |
| 16:41.56 | JackEStorm | VaGoNeTaS: tells the phone and asterisk not to try a direct connection between the two peers |
| 16:41.57 | Joe_CoT | he could just need to configure his sip nat settings |
| 16:42.07 | nny_1 | Joe_CoT: he did |
| 16:42.10 | *** join/#asterisk dbcooper1 (n=User@64.203.244.146) |
| 16:42.13 | JackEStorm | (look in the sample for sip.conf, it's well documented) |
| 16:43.21 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 16:43.38 | Joe_CoT | nny_1, so he set up the rtp ports, and set an externip in sip.conf, and it isn't working still? |
| 16:43.48 | nny_1 | Joe_CoT: you would think that 192.168.1.(WIRELESS):RTPPORT would get NAT'd to the right client behind the router, but not the case it seems |
| 16:44.23 | nny_1 | Joe_CoT: he has asterisk on the first NAT network, and the client behind a router with NAT within that network |
| 16:44.30 | VaGoNeTaS | somebody told me something about using IAX instead of SIP |
| 16:44.35 | VaGoNeTaS | for the wireless connections |
| 16:44.47 | VaGoNeTaS | what about that? |
| 16:44.48 | nny_1 | i dunno off the shelf hardware does strange things when the WAN is a PFC |
| 16:46.13 | VaGoNeTaS | ? |
| 16:46.13 | Joe_CoT | VaGoNeTaS, AIX is a special protocol for asterisk that works better over NAT setups, but is rarely supported by phone hardware, and has few soft clients. Don't know if that would fix your rtp issue or not |
| 16:46.37 | VaGoNeTaS | we dont need it to be working under phone harware |
| 16:46.48 | VaGoNeTaS | we need it working under wi fi with laptops and softphones |
| 16:47.05 | Joe_CoT | ok, well you'd need a soft client that supports AIX. there aren't very many |
| 16:47.25 | Joe_CoT | and you could give that a shot, i guess? I've never used it, i don't know if it would solve your rtp problem |
| 16:48.42 | *** join/#asterisk Alborracho (n=chatzill@190.25.135.1) |
| 16:48.43 | nny_1 | VaGoNeTaS: so your issue is that the client on the wifi can't hear the other party? |
| 16:49.00 | viraptor | what is the best way to jump to a context that may or may not exist? (and recover properly) |
| 16:49.01 | Alborracho | hi everyone |
| 16:49.27 | Alborracho | i need to chance asterisk so it can accept rcf2833 inband, any pointers i should look for? |
| 16:49.30 | Alborracho | *change |
| 16:49.32 | Black_L | How do i access Asterisk inside Ubuntu? |
| 16:49.56 | nny_1 | Black_L: google for the Asterisk book |
| 16:50.03 | nny_1 | ~book |
| 16:50.04 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:50.10 | Joe_CoT | Black_L, it should just be sudo asterisk -r |
| 16:50.13 | SuPrSluG | viraptor:http://www.voip-info.org/wiki/index.php?page=Asterisk+func+exists |
| 16:50.27 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 16:50.28 | Black_L | Joe : Says unable to connect to remote asterisk |
| 16:50.39 | *** join/#asterisk tokozedg (n=CCNA@89.232.24.53) |
| 16:50.57 | jplank | anyone know of a good wholesale SIP provider that could provide INTL origination? (INTL as in not the US) |
| 16:51.19 | nny_1 | Joe_CoT: tbh if one party can't hear the other, it's normally a network issue. The rtp packets just aren't making it across whatever hardware is in the way |
| 16:51.39 | nny_1 | Joe_CoT: this is based on my experience setting up funky things, so eh |
| 16:51.49 | viraptor | SuPrSluG: I meant when I don't have any more information about the current situation... there might be a context and if there is, I want to jump there - I can't query any variable to get that info |
| 16:51.58 | VaGoNeTaS | JackEStorm |
| 16:52.01 | nny_1 | Black_L: than asterisk is not running |
| 16:52.04 | VaGoNeTaS | i did the change u told me |
| 16:52.08 | VaGoNeTaS | and we made a test |
| 16:52.11 | Black_L | How do i run it then? |
| 16:52.25 | VaGoNeTaS | and the phone call went through properly |
| 16:52.47 | SuPrSluG | Black_L:asterisk -vvvc |
| 16:52.54 | nny_1 | Black_L: in Ubuntu it would be /etc/init.d/asterisk start |
| 16:53.02 | Joe_CoT | he already did that |
| 16:53.12 | Joe_CoT | Black_L, do this please: ps aux | grep asterisk |
| 16:53.14 | Joe_CoT | do you see it running? |
| 16:53.19 | tokozedg | hi, i`m trying to configure call forwarding and http://pastebin.com/m71e36a60 is this enought for that, it didn`t worked, can anyone give me a little manual? |
| 16:53.53 | Black_L | Joe : a bunch of text came up |
| 16:54.05 | nny_1 | Joe_CoT: awesome, you must be clarivoyant |
| 16:54.17 | *** join/#asterisk ctp (n=ctp@brsg-d9bee414.pool.mediaWays.net) |
| 16:54.36 | Joe_CoT | nny_1, ? I told him to do that previous, he said the output was asterisk already started |
| 16:54.58 | Black_L | So why can't i access it then? |
| 16:55.22 | SuPrSluG | Black_L:it's not a GUI |
| 16:55.45 | Black_L | SuPrSluG : What about it? |
| 16:57.00 | Joe_CoT | Black_L, can you pastebin the output of "ps aux | grep asterisk" and "sudo asterisk -r" ? |
| 16:57.11 | Black_L | aye |
| 16:57.25 | Black_L | Wait |
| 16:57.27 | Black_L | It just connected |
| 16:57.58 | Joe_CoT | are you sure you were doing sudo asterisk -r before, and not asterisk -r ? |
| 16:58.12 | Black_L | I was probably typing sudo |
| 16:58.50 | Black_L | Now my line says vUbuntu*CLI? |
| 16:58.58 | Black_L | vUbuntu is the virtualized machine's name |
| 16:59.00 | nny_1 | VaGoNeTaS: so your problem is solved? |
| 16:59.20 | [TK]D-Fender | nny_1: For your cell question, dial a local channel to place that out-call and set the CID in there |
| 16:59.21 | nny_1 | VaGoNeTaS: sounds like the router was breaking the peer to peer rtp stuff. |
| 16:59.38 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.83) |
| 16:59.41 | nny_1 | [TK]D-Fender: hmm gotcha |
| 17:00.16 | nny_1 | [TK]D-Fender: that's genius ha |
| 17:05.46 | tokozedg | anyone help :| |
| 17:06.37 | VaGoNeTaS | nny_1 : yep, my problem is solved |
| 17:06.41 | nny_1 | tokozedg: i would try but i seem to be zero for two ha |
| 17:06.54 | VaGoNeTaS | now, i have to cancel the echo , coz at the office with the SIP Phones, they sometimes have echo |
| 17:07.02 | VaGoNeTaS | i have an Redfone Quad as PRI Device |
| 17:07.17 | nny_1 | VaGoNeTaS: ha cool. Not sure why peer to peer would break though. Maybe the router is confused as to who the originating IP is sending the packets |
| 17:07.20 | VaGoNeTaS | do you know if that device has an hardware echo cancel or i have to set it up ? |
| 17:07.26 | nny_1 | VaGoNeTaS: guess PFM |
| 17:07.32 | VaGoNeTaS | PFM? |
| 17:07.32 | nny_1 | VaGoNeTaS: not sure |
| 17:07.34 | VaGoNeTaS | wtf is that |
| 17:07.39 | nny_1 | VaGoNeTaS: Pure F-ing Magic |
| 17:07.56 | nny_1 | VaGoNeTaS: it's the source of 90% of my success. The other 10% is dumb luck |
| 17:07.59 | tokozedg | VaGoNeTaS: for what? |
| 17:08.16 | VaGoNeTaS | nny_1 hahahah |
| 17:08.23 | VaGoNeTaS | tokozedg Redfone Quad |
| 17:09.08 | nny_1 | here http://pastebin.com/m398adb23 someone have a laugh at my avoid the cost if incoming cell phone smilu ring |
| 17:09.11 | nny_1 | of* |
| 17:09.23 | nny_1 | i am testing now so any glaring errors shall be fixed |
| 17:09.33 | nny_1 | simu-ring* |
| 17:10.43 | VaGoNeTaS | now i have |
| 17:10.53 | VaGoNeTaS | some static calling from softphone to softphone |
| 17:11.05 | VaGoNeTaS | but, im in another place |
| 17:11.12 | VaGoNeTaS | connecting to 192.168.1.27 with a VPN |
| 17:11.24 | VaGoNeTaS | it might be coz of that |
| 17:12.31 | *** join/#asterisk neurosys (n=neurosys@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 17:13.38 | bmoraca | i can't believe that asterisk does not have a way to specify a different auth name than the peer name for incoming registrations...that seems like a big shortcoming! |
| 17:14.07 | neurosys | bmorac: wanna use encrypted keys? |
| 17:14.19 | bmoraca | no, just a different auth name |
| 17:14.46 | neurosys | Hmm yeah i never noticed that |
| 17:15.39 | nkohh | feel free to implement it yourself and submit the patch |
| 17:18.02 | nny_1 | hmm is it just me or does using Local not pass the ARG variables passed to the macro? |
| 17:18.57 | [TK]D-Fender | nny_1: Local is a completely separate channel... they MIGHT be inherited |
| 17:19.36 | nny_1 | I guess I can set a global long enough to do it, although it would barf it happened at the same time |
| 17:20.53 | [TK]D-Fender | nny_1: How so? |
| 17:22.09 | nny_1 | [TK]D-Fender: actually ha I can pull the channel info to do what I want, since the Local channel it is sent to can be used to invoke the global variable. Um I'll post it in pastebin when I get it working and you can tell me how I could have done it easier hehe |
| 17:23.01 | [TK]D-Fender | nny_1: Wait.. I said inhereited, I did not say "use a global variable" |
| 17:23.20 | [TK]D-Fender | nny_1: and yes, that is a race-condition disaster waiting to happen |
| 17:23.48 | nny_1 | [TK]D-Fender: it doesn't appear that a macro argument is inherited |
| 17:24.06 | [TK]D-Fender | nny_1: which is why i said to allocate another var prior to the call. |
| 17:24.37 | nny_1 | [TK]D-Fender: yeah that's what I meant by setting a global, my bad |
| 17:28.07 | *** join/#asterisk seanmh (i=seanmh@c-69-254-131-168.hsd1.nm.comcast.net) |
| 17:30.35 | nny_1 | [TK]D-Fender: behold the monster http://pastebin.com/m72d7f6d5 |
| 17:31.32 | nny_1 | [TK]D-Fender: changed the 190 exten to a 3 digit wildcard, but other than that it works |
| 17:33.28 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:35.44 | *** join/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 17:36.36 | nny_1 | [TK]D-Fender: hmm it works, but the latency from when it says dialing cell@sipprovider to when the cell phone rings changes. Not sure why, it consistently takes a bit longer, so may have to scrap it anyways |
| 17:39.44 | [TK]D-Fender | nny_1: Local should not take more than an extra second to do the dial. For the cell to actually start ringing is a telco/antenna issue |
| 17:40.13 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:41.10 | nny_1 | [TK]D-Fender: yeah i figured as much, I suspect something with the sip provider, although the extra 10 seconds (so far) only manifest themselves when I call the provider from the local channel. I changed it back and consistently got less latency. I am gonna look more, just think it's odd |
| 17:41.46 | [TK]D-Fender | nny_1: makes no sense to be an * issue |
| 17:42.29 | tokozedg | as i guessed i was searching another function and this one is for another one, i want when A answers call from B, A dial a number and conncet B to C, any ideas? |
| 17:43.01 | nny_1 | [TK]D-Fender: i don't think so either, i think it's an issue with the sip provider, since the cell phone normally takes 35 seconds except when this happens. I am gonna keep trying to reproduce the issue and do a sip debug on the provider. It take slonger to "make progress/ pass the call" it seems, i'll let ya know what it ends up being |
| 17:43.37 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 17:44.13 | Zeeek | you should join us at #voip-users-conference for an intercontinental discussion |
| 17:44.37 | *** join/#asterisk WHYS (n=drumm@137.28.94.209) |
| 17:44.59 | Zeeek | http://tr.im/voip to call in and join the VoIP Users Conference discussion |
| 17:45.37 | timeshell_atwork | Anyone here familiar with customizing polycom's sip.cfg file? I'm looking at A-76 in the Admin Guide for customizing a key and where it refers to attribute key.x.y.function.prim, can someone tell me what prim represents? |
| 17:46.28 | WHYS | has anyone successfully ordered ABE? - having vendor problem - they can't seem to deliver a downloadable product |
| 17:47.14 | WHYS | I have to order through state contract, and the vendor had to buy from a distributor who says digium is laying off people and can't deliver |
| 17:50.13 | timeshell_atwork | nm... this isn't what I'm looking for. |
| 17:50.17 | WHYS | No one at Digium to answer? :) I would have thought someone would have balked. |
| 17:51.36 | coppice | WHYS: well, if they've all been laid off......... |
| 17:51.44 | WHYS | :) |
| 17:53.56 | Corydon76-dig | WHYS: what distributor is saying that? |
| 17:54.05 | nny_1 | is there a way to put timestamps in console? |
| 17:54.12 | nny_1 | nm heh |
| 17:54.14 | nny_1 | ignore me |
| 17:55.07 | WHYS | Vendor doesn't say. I've asked under threat of recalling the order. It's been over a week now with no download info |
| 17:55.36 | Corydon76-dig | WHYS: then it's probably your vendor that is laying off people and can't deliver, not Digium or the distributor |
| 17:55.58 | Corydon76-dig | Shifting blame is the oldest trick in the book |
| 17:56.14 | WHYS | I'd guessed that. Just thought I would check. |
| 17:56.17 | nny_1 | [TK]D-Fender: yeah this is just irregular network time from the cell provider, off to figure out how to lower the ring to VM on the cell from the provider |
| 17:57.15 | [TK]D-Fender | nudges the blame over 3" with his foot and whistles innocently |
| 17:58.10 | Corydon76-dig | WHYS: if you find out who, let me know. I can push back on the other end. |
| 17:58.19 | nny_1 | [TK]D-Fender: heh |
| 17:58.37 | WHYS | will do. thanks. |
| 18:01.04 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:01.04 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:01.07 | leifmadsen | anyone know if there is a dialplan application or function that will send a voicemail notification? (i.e. MWI) |
| 18:01.25 | leifmadsen | (any version of asterisk, currently on 1.6.0 branch) |
| 18:03.02 | juanIMP | leifmadsen: voicemail ( voicemail at voicemail.conf ) |
| 18:03.14 | leifmadsen | juanIMP: no, from the dialplan |
| 18:03.21 | leifmadsen | I realize I can use mailbox= in sip.conf |
| 18:03.31 | leifmadsen | but it doesn't work well if you're creating a hot-desking feature |
| 18:03.40 | leifmadsen | since the user is not tied to the device |
| 18:04.49 | [TK]D-Fender | leifmadsen: Scripts are in order. Would ahve to parse out sip.conf, etc and edit the box & apply |
| 18:05.00 | Aiatek | anyone knows if there is a problems with DAHDI Linux 2.1.0, asterisk 1.6.1 and the openvox a400p pci cards |
| 18:05.01 | Aiatek | ? |
| 18:05.06 | leifmadsen | [TK]D-Fender: I will figure out something better |
| 18:05.22 | leifmadsen | openvox is a clone card if I remember right |
| 18:05.24 | leifmadsen | unsupported |
| 18:05.49 | Aiatek | ok |
| 18:06.14 | [TK]D-Fender | leifmadsen: leifmadsen Sure they are.. jsut not by Digium :) |
| 18:06.36 | leifmadsen | [TK]D-Fender: you find me someone actually supporting those clone cards, and then I'll believe that is a true statement |
| 18:06.44 | Aiatek | because they used to work with zaptel |
| 18:07.02 | [TK]D-Fender | leifmadsen: guess ti depends what kind of "support" is needed. |
| 18:07.15 | leifmadsen | [TK]D-Fender: btw -- file just told me minivm has what I need in 1.6.2 and trunk |
| 18:07.16 | coppice | leifmadsen: openvox make somewhat original cards, and support them with their own drivers |
| 18:07.28 | leifmadsen | coppice: ah, gotcha |
| 18:07.32 | leifmadsen | Aiatek: there ya go :) |
| 18:07.37 | Aiatek | ok |
| 18:07.45 | Aiatek | thx |
| 18:12.45 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-112b92d9a84bf674) |
| 18:12.54 | *** join/#asterisk vi390 (n=fc@unaffiliated/vi390) |
| 18:15.27 | *** join/#asterisk ghenry (n=ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 18:15.31 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 18:16.45 | vi390 | hej, how can I catch outgoing connections? (need a working concept) I get that the extension with outgoing number is not found. It should not look for the number as extension, but dial out |
| 18:17.15 | vi390 | thats what I have in the outgoing: exten => _0.,1,Dial(SIP/${EXTEN:1}@10) |
| 18:17.38 | vi390 | incoming is working, thats where "10" points |
| 18:17.54 | [TK]D-Fender | viwhat do you mean "10" "points there"? Huh? |
| 18:18.19 | vi390 | the context 10 is defined in sip.conf |
| 18:18.24 | [TK]D-Fender | vi390: Do you have a SIP peer named [10]? |
| 18:18.39 | vi390 | yes |
| 18:18.52 | [TK]D-Fender | v3Show us the failed call with SIP debug, and the sip.conf |
| 18:18.55 | [TK]D-Fender | ~pb |
| 18:18.55 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 18:18.57 | [TK]D-Fender | ^^^^^^^^ |
| 18:20.58 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 18:21.48 | Zeeek | oh, am I still here? |
| 18:21.58 | *** part/#asterisk srf21c (n=seth@ip98-165-60-42.ph.ph.cox.net) |
| 18:22.11 | Zeeek | sorry |
| 18:22.13 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 18:22.14 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
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| 18:33.08 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:35.45 | *** join/#asterisk spck (n=spck@unioncab.com) |
| 18:36.02 | spck | ~ldirectord |
| 18:36.25 | spck | anyone have any docs on configuring * with ldirectord to do the load balancing? |
| 18:36.32 | *** join/#asterisk farkus (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
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| 18:39.45 | nny_1 | adding 2 g729 codec channels to my dev box here. Stupid question, but if a third sip channel opens to the sip provider will it fallback if i run out of licenses or just complain the codec conversion won't happen (assuming my sip.conf has the codecs listed in order) |
| 18:39.46 | *** join/#asterisk seb- (n=seb@li30-51.members.linode.com) |
| 18:40.14 | [TK]D-Fender | nny_1: run out = DOA |
| 18:40.14 | seb- | [TK]D-Fender: ping |
| 18:40.22 | [TK]D-Fender | seb-: y0 |
| 18:40.35 | seb- | [TK]D-Fender: can you try to chat w/ me one more time? |
| 18:40.45 | seb- | [TK]D-Fender: just for a minute |
| 18:41.13 | [TK]D-Fender | seb-: @work. nope |
| 18:41.52 | seb- | [TK]D-Fender: any hope you'll be @home in next 6 hours? |
| 18:41.52 | nny_1 | [TK]D-Fender: lame |
| 18:42.05 | nny_1 | [TK]D-Fender: so if the licenses get used up, it won't drop back to GSM? |
| 18:42.09 | nny_1 | (or w/e i use) |
| 18:47.36 | *** join/#asterisk yacc__ (n=andreas@188-23-73-70.adsl.highway.telekom.at) |
| 18:51.41 | [TK]D-Fender | seb-: yup |
| 18:51.50 | [TK]D-Fender | nny_1: DOA <- |
| 18:53.43 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 18:54.54 | *** part/#asterisk wwalker (n=wwalker@mailbox.eclipsing.com) |
| 18:59.44 | nny_1 | [TK]D-Fender: nice. Wonderful limitation. I assume I can just have the outbound channel set to a call limit and write failover in the dialplan, but sounds like garbage. Nice of them to not give me a reason to use it. |
| 19:00.50 | *** join/#asterisk jnfuller (n=jnfuller@99.199.170.110) |
| 19:00.57 | nny_1 | from what I just read this is a limitation in the codec itself |
| 19:01.15 | jnfuller | does anyone know if the syntax for include => context changed in 1.6.0? |
| 19:02.18 | ectospasm | nny_1: it should fall back to any secondary codecs you have set up. |
| 19:02.38 | ectospasm | jnfuller: to my knowledge it hasn't |
| 19:03.23 | nny_1 | ectospasm: from what I read, it's the codec stopping it. It basically sends the conversion to the codec, and the codec fails because of licensing, as opposed to asterisk itself having knowledge of the limit |
| 19:03.27 | jnfuller | I'm trying to include a context and instead of showing up in the dialplan the include statement looks like an extension |
| 19:04.31 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 19:05.28 | ectospasm | nny_1: so you're saying it's not falling back? You should probably call Digium about that... |
| 19:06.15 | ectospasm | jnfuller: that's what "dialplan show <context>" does, it doesn't "dereference" or follow the includes... |
| 19:11.43 | jnfuller | ok so then the include is not actually including the context info |
| 19:12.18 | jnfuller | something is broken |
| 19:15.52 | nny_1 | ectospasm: nm i lied |
| 19:16.10 | nny_1 | ectospasm: [TK]D-Fender i get an error in console, but it falls back to ulaw (or gsm) after |
| 19:17.08 | b14ck | hey guys, anyone here using cepstral? |
| 19:17.41 | b14ck | i'm having a bit of an issue getting the app_swift program to work with asterisk 1.6, im following some nerd vittles instructions that i found here: http://nerdvittles.com/index.php?p=202 |
| 19:17.57 | b14ck | when I get to the part where I download and make app_swift, i get compile errors, i'll paste them in a moment |
| 19:18.01 | nny_1 | http://pastebin.com/m623b046b (this is from my sip client (ulaw) to the server and then from server to vitelity |
| 19:18.10 | b14ck | i'm using app_swift.1.6.2 |
| 19:18.35 | *** join/#asterisk edibrac (n=elusive4@206.173.193.34.ptr.us.xo.net) |
| 19:22.11 | nny_1 | ectospasm: it is possible once the codec is installed though it can still fail, installing it now fwiw |
| 19:23.09 | nny_1 | ectospasm: it is so rare [TK]D-Fender is wrong. Last time he was wrong the world fell into the Dark Ages and havoc ensued |
| 19:23.20 | leifmadsen | whats a dialplan variable/application/function to get the VM pin? |
| 19:23.26 | [TK]D-Fender | nny_1: Or they fixed it. It used to jsut bomb |
| 19:23.38 | [TK]D-Fender | leifmadsen: vmauthenticate |
| 19:23.46 | nny_1 | [TK]D-Fender: yeah going to test now |
| 19:23.47 | edibrac | i had some dropped calls on a conference line this morning - asterisk logs mention "span 1 got hangup request, cause 16" - though, those have showed up for months and we haven't had dropped calls until now ..here's my logs w/ full debugging enabled: http://pastebin.com/m14ab06e4 |
| 19:23.52 | [TK]D-Fender | leifmadsen: Well doesn't GET it, just auths by it |
| 19:24.07 | leifmadsen | [TK]D-Fender: that might work... let me see.... |
| 19:24.47 | leifmadsen | aha, for this application, yes, it will work |
| 19:24.48 | leifmadsen | thanks |
| 19:25.00 | edibrac | if the cause of a dropped call was the other side's pbx, what would show up on my end in the logs? |
| 19:25.05 | leifmadsen | (I was searching for 'like VM' in functions, and didn't try that with applications) |
| 19:25.46 | [TK]D-Fender | leifmadsen: np |
| 19:26.25 | [TK]D-Fender | leifmadsen: IMO there should be a function for this. it the natural way |
| 19:26.44 | leifmadsen | [TK]D-Fender: I would agree |
| 19:26.58 | leifmadsen | funny such a thing doesn't exist actually |
| 19:27.01 | leifmadsen | I'm even using 1.6.2 right now |
| 19:29.19 | leifmadsen | maybe I'll look at the code for VMauthenticate one of these days and see if I can figure out how to make that function |
| 19:30.44 | leifmadsen | seems like it should be pretty simple task -- will be a good homework assignment to learn some C |
| 19:40.36 | *** join/#asterisk jicksta (n=jicksta@67.164.0.78) |
| 19:41.28 | edibrac | are "chan_dahdi.c: Not yet hungup... Calling hangup once with icause, and clearing call" normal? |
| 19:41.43 | edibrac | i don't have a h exten defined |
| 19:42.03 | edibrac | ..seems like they might be related? |
| 19:42.11 | [TK]D-Fender | edibrac: nope. |
| 19:42.25 | *** join/#asterisk dshap (n=IceChat7@ip70-181-91-110.oc.oc.cox.net) |
| 19:44.44 | edibrac | i guess it's a dahdi thing that happens when there's a (intentional or unintentional) hangup |
| 19:44.52 | edibrac | when it talks about "icause" |
| 19:46.15 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:49.28 | [TK]D-Fender | edibrac: Go lok at the debug for your channel |
| 19:50.09 | dshap | does anyone know of any Level3 SIP Trunking resellers to use with Asterisk? |
| 19:51.10 | *** join/#asterisk Marquis42 (n=bfhbmw0@65-127-126-34.dia.static.qwest.net) |
| 19:51.13 | leifmadsen | Marquis42: yo :) |
| 19:51.16 | leifmadsen | Marquis42: so this is my idea |
| 19:51.17 | Marquis42 | hi! :) |
| 19:51.29 | ajohnson | Asterisk is stopping dialplan execution in the middle of a context unexpectedly. Was wondering if anyone else could take a look? http://pastebin.com/d642a6ad0 |
| 19:52.23 | leifmadsen | Marquis42: I'm building a hot-desking feature, so the user is not tied to a device. Thus my MWI must follow the user around, not the device. What I'm thinking of doing is making a separate <mac>@mwi context in voicemail.conf for those devices, then modifying the MWI based on the users actual mailbox of <extension>@company_context |
| 19:52.41 | ajohnson | There's a GotoIfTime and it appears that whether the applications matches the time requirements or not, it stops at that line |
| 19:52.53 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 19:53.11 | Marquis42 | leifmadsen: Oooh... Clever. I think that would work well. |
| 19:53.28 | leifmadsen | ajohnson: try $[ ] around the statement in GotoIfTime.... |
| 19:53.34 | leifmadsen | Marquis42: ok, I'm going to try it |
| 19:54.23 | Marquis42 | leifmadsen: Good deal, let me know how you get on with it. |
| 19:54.29 | barbacha | is it possible to make it that every extention is abble to receive voice AND fax calls and to diverts faxes to hylafax or similar while letting the voice ones through ? |
| 19:54.33 | leifmadsen | Marquis42: will do! |
| 19:54.45 | edibrac | [TK]D-Fender: are you talking about the PRI debug or DEBUG entries in my /var/log/asterisk/full ? |
| 19:55.31 | ajohnson | leifmadsen: Not working, and core show application gotoiftime shows it being used without $[] as well as voip-info and the same line works in 1.4 |
| 19:55.39 | edibrac | [TK]D-Fender: wait i think not "pri debug span" because you're talking about loooking at the channel. |
| 19:55.46 | leifmadsen | ajohnson: gotcha..t hen I'm not sure |
| 19:55.53 | ajohnson | cool, thanks |
| 19:56.18 | ajohnson | And since it's stopping in the middle of a context and not following either path and not showing an error, I'm assuming it's a bug |
| 19:58.56 | *** join/#asterisk esaym (n=user@cpe-24-174-186-34.satx.res.rr.com) |
| 19:59.44 | leifmadsen | Marquis42: wow, that totally worked :) |
| 20:01.01 | *** part/#asterisk youngproguru (n=mm@74.10.229.45) |
| 20:03.49 | Marquis42 | leifmadsen: Excellent! |
| 20:05.48 | *** part/#asterisk dbcooper1 (n=User@64.203.244.146) |
| 20:09.24 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 20:09.50 | leifmadsen | Marquis42: ya, was pretty surprised when it worked :) |
| 20:15.39 | Marquis42 | leifmadsen: I can imagine. It works well for my uses, but that's a creative way to use the app. Definitely follows the Asterisk spirit there. :) |
| 20:16.25 | leifmadsen | Marquis42: ya, I find all sorts of ways to use apps the developers never imagined :) |
| 20:16.47 | leifmadsen | Marquis42: I tried using Originate() along with the Calender API stuff to create an auto-dialer for meetings :) |
| 20:16.56 | leifmadsen | Almost worked too.... damn you Local channels |
| 20:17.02 | leifmadsen | *shaky fist* |
| 20:17.53 | Marquis42 | lol... I've been poking around the Calender API since it was merged (well, since this morning since I was off for a couple of days). Not to come up with my own nefarious uses.... Mwahahaha ;) |
| 20:18.02 | Marquis42 | s/Not/Now/ |
| 20:18.21 | leifmadsen | Marquis42: ya, I was testing it when it was in a branch. I showed it off at IT360 in April |
| 20:19.02 | ajohnson | leifmadsen: The reason it is failing is because 1.6.2 considers sunday the beginning of the week, not monday |
| 20:19.16 | leifmadsen | ajohnson: neato |
| 20:19.26 | ajohnson | however any invalid gotoif time criteria will cause asterisk to just stop handling the call |
| 20:19.51 | Marquis42 | leifmadsen: Excellent. It definitely looks cool. I have some of the basic stuff going (calling me when I have a meeting, etc.). |
| 20:19.52 | leifmadsen | heads home |
| 20:20.06 | leifmadsen | Marquis42: ya, it is really quite cool -- I need to get my PBX at home running trunk or something :) |
| 20:20.16 | leifmadsen | well, I'm outta here for the day, peas out! |
| 20:20.18 | Marquis42 | leifmadsen: Definitely, that's what I try to do. |
| 20:20.21 | Marquis42 | OK, see you later! |
| 20:20.37 | *** part/#asterisk Marquis42 (n=bfhbmw0@65-127-126-34.dia.static.qwest.net) |
| 20:22.43 | VaGoNeTaS | does anyone have an Redfone here' |
| 20:22.45 | VaGoNeTaS | ? |
| 20:22.48 | VaGoNeTaS | Redfone Quad box |
| 20:25.44 | jameswf | I have a phone that is red ! |
| 20:26.00 | VaGoNeTaS | hhahaa |
| 20:26.11 | VaGoNeTaS | i have echo on a device with "Hardware Echo Cancellation" |
| 20:26.12 | VaGoNeTaS | :S |
| 20:26.23 | VaGoNeTaS | calling from softphone to local |
| 20:26.33 | VaGoNeTaS | and sometimes calling from SIP Phones to Local |
| 20:26.59 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 20:28.48 | VaGoNeTaS | how do i cancel the echo ? |
| 20:28.56 | VaGoNeTaS | echocancel=yes on sip.conf =? |
| 20:29.20 | nny_1 | anyone know if the polycom 330 supports distinctive ring? |
| 20:29.37 | nny_1 | found docs that say you can do it with the 501/601 configs, nothing yet on the 330 |
| 20:30.56 | nny_1 | nm found my info |
| 20:32.20 | *** join/#asterisk propellerhead (n=yogurt2u@host151.190-31-150.telecom.net.ar) |
| 20:33.04 | VaGoNeTaS | nny_1 |
| 20:33.18 | VaGoNeTaS | i need your 90% lucky here |
| 20:33.23 | *** join/#asterisk bbryant1 (n=brett@c-68-59-21-152.hsd1.sc.comcast.net) |
| 20:34.35 | *** part/#asterisk jnfuller (n=jnfuller@99.199.170.110) |
| 20:38.47 | nny_1 | VaGoNeTaS: echo? |
| 20:47.50 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 20:49.36 | *** part/#asterisk nny_1 (n=scott@64.203.244.146) |
| 20:49.41 | SuPrSluG | VaGoNeTaS:EC should be done on the redphone box not in sip.conf |
| 20:49.53 | *** join/#asterisk smps (n=smps@83-65-116-98.liesing.xdsl-line.inode.at) |
| 20:52.37 | justdave | is there a way to tell from a dialplan if MeetMeCount() crashes? |
| 20:55.38 | justdave | meetme bailed for lack of a timing source because I didn't have dahdi loaded |
| 20:57.08 | justdave | and of course the dialplan following that wound up telling the users confusing things. :) trying to figure out how to get it to cluefully tell the user it's broken instead of telling them the conference number is invalid |
| 20:57.27 | SuPrSluG | apparently with 1.6 you don't need it anymore |
| 20:58.47 | justdave | hmm, actually, looks like it didn't get that far. the "conference number is invalid" appears to have come from MeetMe before it crashed, and it hung up and never returned to my dialplan. |
| 20:58.52 | justdave | that complicates things |
| 20:59.21 | justdave | ah well, people knew something wasn't right anyway, since it was supposed to be a valid conf number :) |
| 21:08.08 | VaGoNeTaS | SuPrSluG , so i have to disable the echocancel=yes on the sip.conf and the echocancel=yes on zapata.conf file? |
| 21:08.20 | VaGoNeTaS | i have on zapata, echocancel=yes and echocancelwhenbridge=yes |
| 21:08.36 | VaGoNeTaS | and on the sip.conf file i have echocancel=yes under each SIP/Acc |
| 21:08.36 | *** part/#asterisk Joe_CoT (n=joecot@ubuntu/member/pdpc.bronze.joeterranova) |
| 21:08.59 | *** join/#asterisk ZenBSDi (n=ZenBSDi@unaffiliated/ZenBSDi) |
| 21:11.33 | timeshell_atwork | Is there any way to monitor what res_phoneprov is doing when a phone asks for files? |
| 21:15.36 | *** part/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 21:18.35 | *** join/#asterisk smps (n=smps@83-65-116-98.liesing.xdsl-line.inode.at) |
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| 21:39.23 | *** join/#asterisk Zock (n=zock@190.158.58.8) |
| 21:39.25 | Zock | Hello. |
| 21:40.46 | Zock | I am doing some dialplan-magic today and i am wondering how to achieve that during a call a certain "check" is done.... 2 Parties talk, and every minute a database-check should be done to look if there is enough credit for the call...if not, the call should be disconnected. |
| 21:40.55 | Zock | What command should i take a closer look at? |
| 21:43.53 | _ShrikE | Zock: I would calculate the available credit or time available for the call via func_odbc adn then limit the call accordingly with the L option in dial. |
| 21:44.19 | Zock | _ShrikE: I cant do that because the credits can be decreased by other sources. |
| 21:44.30 | Zock | _ShrikE: So a pre-calculation of time is not the answer. |
| 21:44.39 | _ShrikE | Zock: I see |
| 21:44.44 | Zock | _ShrikE: AbsoluteTimeout() ? |
| 21:44.54 | Zock | _ShrikE: Or does this disconnect in case of the timeout? |
| 21:45.18 | Zock | Whops...removed in 1.4 :D |
| 21:45.34 | Zock | But TIMEOUT(absolute) seems to be the follow-up. |
| 21:45.42 | _ShrikE | correct |
| 21:46.12 | _ShrikE | But im not sure that will do what you are looking for either. |
| 21:46.58 | Zock | Hm, in my Mind i have this idea..... TIMEOUT(absolute) / Dial() .... at priority t i do the db-checks.. |
| 21:47.49 | Zock | _ShrikE: Do you know if at the time the timeout is hit, is the Timout discarded or keeps running again? |
| 21:47.54 | hardwire | anybody got an XO sales contact for me? |
| 21:49.17 | Zock | "The absolute maximum amount of time permitted for a call." sounds quite wrong for my needs :-/ |
| 21:50.05 | profXavier | guys, I have to edit the Polycoms to use 9250*******, for local #s |
| 21:50.37 | profXavier | i -believe- asterisk manages any call that is 9******* and local, by adding on the 250 |
| 21:51.24 | profXavier | but when I use the Polycom, I have to dial 9*******, whereas the 9250******* doesn't work (which is what people are used to dialing |
| 21:51.28 | KyleK | asterisk will do whatever you configure it for |
| 21:51.41 | KyleK | check the polycoms dialplan? |
| 21:51.54 | profXavier | i want to edit the Polycom to handle it using the Degitmap |
| 21:52.08 | Zock | G(context^exten^pri) of Dial could be an idea. |
| 21:52.10 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:52.18 | KyleK | I recently set my SPA3102 to try dialing after only 7 digits ;) |
| 21:52.43 | KyleK | (whoops) |
| 21:52.47 | profXavier | so if I want it to handle 9250******* numebrs... |
| 21:53.17 | profXavier | then I need to add that exactly, in the Digitmap ? |
| 21:53.23 | KyleK | profXavier: is asterisk bitching about a lack of an extension for like 92506355428? |
| 21:53.45 | profXavier | i cannot see the output from * atm |
| 21:53.52 | KyleK | oh is this a case where you cant just edit the asterisk box? |
| 21:53.54 | profXavier | the room that the phone is in, is in use |
| 21:53.58 | [TK]D-Fender | profXavier: Doesn't have to be exact, jsut has to match |
| 21:54.11 | *** join/#asterisk MrNaz (n=mrnaz@203.214.68.222) |
| 21:54.45 | profXavier | so if user does 92505551234, and I have 9250xxxxxxx in the phone's Digitmap, then the calls should complete? |
| 21:55.35 | KyleK | if its what i think it is, yes |
| 21:55.45 | profXavier | ok, let me try that |
| 21:55.54 | profXavier | i just wasnt sure how it would be handled |
| 21:56.12 | profXavier | as * manages to add on the 250 to all the numbers, when 9 + 7 digits are dialed |
| 21:56.48 | profXavier | im a little rusty on contexts :D |
| 21:57.35 | KyleK | ah, my personal asterisk box doesn't do 7 digit dialing, but my itsp supports it, I'd just have to fill the default area code on thier website |
| 21:57.58 | profXavier | we have our own box, locally |
| 21:58.11 | profXavier | i personally think its more of a hassel |
| 21:58.23 | profXavier | but it saves us money *rolls eyes* |
| 21:58.31 | hardwire | hmm.. anybody have PRI with XO in Cali?? |
| 22:00.01 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 22:01.10 | rue_mohr | if I have a T1 card, using a zaptel driver, if I specify a gain, is that a digital gain between the pcm and the T1? |
| 22:03.29 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) [NETSPLIT VICTIM] |
| 22:03.29 | *** join/#asterisk ruben23 (n=AGENT@124.107.3.178) [NETSPLIT VICTIM] |
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| 22:09.05 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:09.37 | VaGoNeTaS | have any1 installed FreePBX? |
| 22:11.49 | VaGoNeTaS | http://pastebin.ca/1440220 |
| 22:11.52 | VaGoNeTaS | :s |
| 22:13.53 | VaGoNeTaS | i dont understand coz im doing the installation as root |
| 22:13.53 | VaGoNeTaS | :s |
| 22:14.08 | Qwell | VaGoNeTaS: This is #asterisk. Try elsewhere, like #freepbx. |
| 22:14.23 | VaGoNeTaS | shit dude, it have to do with asterisk |
| 22:14.39 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 22:15.06 | VaGoNeTaS | it has* |
| 22:15.46 | timeshell_atwork | Hrmmm |
| 22:15.49 | VaGoNeTaS | forget it, u wont understand ure never useful for noone, i've neever seen u helping ppl |
| 22:16.13 | timeshell_atwork | boots VaGoNeTaS into next week (and #freepbx) |
| 22:16.18 | VaGoNeTaS | but telling Get outta here, or go find help somewhere else |
| 22:16.38 | florz | puts up a large "don't feed the trolls" sign |
| 22:17.04 | *** mode/#asterisk [+b *!*@fine.] by Qwell |
| 22:17.07 | Qwell | err |
| 22:17.19 | *** mode/#asterisk [-b *!*@fine.] by Qwell |
| 22:17.25 | *** mode/#asterisk [+b *!*n=debian@*.datapartner.cl] by Qwell |
| 22:17.25 | *** kick/#asterisk [VaGoNeTaS!i=north@pdpc/sponsor/digium/Qwell] by Qwell (fine. bye.) |
| 22:18.14 | timeshell_atwork | gives Qwell a standing ovation |
| 22:18.28 | jaytee | way to go, ace! |
| 22:18.39 | *** part/#asterisk bionoid (i=terje@mesyah.org) |
| 22:18.48 | Qwell | I fail at bans occasionally. Don't tell anybody. |
| 22:19.27 | drmessano | fuk u gyus i wil set halp somewear else!!$ |
| 22:19.36 | jaytee | hehehe |
| 22:19.36 | Daviey | :o |
| 22:19.37 | timeshell_atwork | I gotta learn how to use if statements in * |
| 22:19.42 | *** join/#asterisk my007ms (i=master@botmaster.x86.be) |
| 22:20.08 | [TK]D-Fender | Qwell: S'ok, a few more swings and I might have helped your aim :) |
| 22:20.14 | jaytee | I've got the If statements down pat, it's the Maybe statements that keep tripping me up |
| 22:20.19 | drmessano | part #akerisk u r suk so bed i hat u all |
| 22:20.20 | Daviey | timeshell_atwork: for dialplan, a nice example is the time condition |
| 22:20.21 | [TK]D-Fender | Qwell: He's really more clueless and frustrated than troll. |
| 22:20.22 | drmessano | :( |
| 22:20.30 | drmessano | \part #akerisk u r suk so bed i hat u all |
| 22:20.32 | drmessano | :( |
| 22:20.43 | *** mode/#asterisk [-b *!*n=debian@*.datapartner.cl] by Qwell |
| 22:21.19 | *** join/#asterisk brookshire (i=mbrooks@hijacked.us) |
| 22:21.32 | *** part/#asterisk my007ms (i=master@botmaster.x86.be) |
| 22:22.04 | *** join/#asterisk VaGoNeTaS (n=debian@xen.datapartner.cl) |
| 22:22.10 | timeshell_atwork | heh |
| 22:26.29 | [TK]D-Fender | timeshell_atwork: there is absolutely nothing to "if statements" |
| 22:26.41 | timeshell_atwork | I know |
| 22:26.46 | timeshell_atwork | I just haven't looked at em yet |
| 22:26.47 | timeshell_atwork | :p |
| 22:27.25 | timeshell_atwork | was thinking out loud when he said that |
| 22:27.30 | drmessano | IF you do THEN you will know |
| 22:27.46 | jaytee | ELSE you will forever remain ignorant |
| 22:27.58 | drmessano | ELSEIF <--- |
| 22:28.08 | timeshell_atwork | nm |
| 22:28.09 | timeshell_atwork | lol |
| 22:28.34 | timeshell_atwork | It wasn't really a request for help in other words |
| 22:28.43 | jaytee | oh thank god! |
| 22:28.52 | jaytee | cuz I really suck at that |
| 22:29.09 | drmessano | We dont do that here: |
| 22:29.10 | drmessano | [18:16] <VaGoNeTaS> forget it, u wont understand ure never useful for noone, i've neever seen u helping ppl |
| 22:29.12 | rue_mohr | what data rate does the isdn run at on nortel sets? |
| 22:29.52 | jaytee | what? |
| 22:30.02 | KyleK | are enum lookups used in practise? like dundi/e164.org/e164.arpa |
| 22:30.09 | drmessano | ~asteriskhelp |
| 22:30.11 | infobot | <VaGoNeTaS> forget it, u wont understand ure never useful for noone, i've neever seen u helping ppl |
| 22:30.13 | rue_mohr | did you know nortel digital sets use isdn? |
| 22:30.37 | jaytee | like the M3900 series phones? |
| 22:30.51 | rue_mohr | like the M7208 phones |
| 22:31.06 | KyleK | so from the pbx to the handset is an isdn line of sorts? |
| 22:31.08 | timeshell_atwork | Why isn't it picking up my entries when I have it sitting in WaitExten(3)? |
| 22:31.15 | profXavier | ok, that didnt work |
| 22:31.15 | timeshell_atwork | Keeps timing out |
| 22:31.19 | jaytee | rue_mohr, most like 64kbps |
| 22:31.34 | profXavier | I did this --> so if user does 92505551234, and I have 9250xxxxxxx in the phone's Digitmap, then the calls should complete? |
| 22:31.40 | rue_mohr | ah |
| 22:31.41 | rue_mohr | hmm |
| 22:31.50 | [TK]D-Fender | rue_mohr: its an ISDN influenced protocol, but no actual telcom standard |
| 22:32.13 | [TK]D-Fender | profXavier: From the phone, yes. Whether * accepts the call is another matter entirely |
| 22:32.16 | rue_mohr | I have a digital card for my channelbank, and they dont say much that makes it sound like, i could plug one of them in |
| 22:32.24 | profXavier | tried to do 9xxxxxxx |
| 22:32.32 | profXavier | where its 9250xxxx |
| 22:32.53 | profXavier | so I need to adjust the context on *, not on the phone then ? |
| 22:33.12 | [TK]D-Fender | profXavier: Don't know. I don't see you showing us any SIP debug for the attempt. |
| 22:33.32 | profXavier | u mean, from asterisk -r ? |
| 22:33.41 | [TK]D-Fender | profXavier: As always |
| 22:33.52 | [TK]D-Fender | profXavier: and with SIP DEBUG enabled |
| 22:33.56 | Zock | Bye |
| 22:34.42 | profXavier | <PROTECTED> |
| 22:35.31 | [TK]D-Fender | profXavier: Looks like its executing dialplan if I'm to take that line at face value |
| 22:35.35 | jaytee | rue_mohr, I connected M2616 phones via channel banks to a couple of remote Nortel Meridian Option 11C PBX systems but that was back in 1994-1995. I can't even remember the equipment brand I used. |
| 22:35.40 | profXavier | Executing [9*******@pbxinternal-zap:1] Set("SIP/boardroom.waikik1-c40b166 0", "CALLERID(all)=Neverblue Media<250*******>") in new stack |
| 22:35.52 | profXavier | the first: when I place the numbers in, before I hit dial |
| 22:36.09 | profXavier | the second: when I press dial, have a dial tone, then enter the # |
| 22:36.12 | rue_mohr | jaytee, ah, to interconnect systems? |
| 22:36.21 | jaytee | rue_mohr, yes |
| 22:37.01 | [TK]D-Fender | profXavier: those are 2 different lengths. What are we supposed to be seeing here? |
| 22:38.15 | rue_mohr | my channelbank co card picks up radio stations, sorta, I ran a rf jammer thru its paces in the telco closet and wasn't able to block out the main station comming in, so its not being picked up locally, I just tried an old nortel 616 on my line and cant hear it, I dont hear much of anything with a standad phone |
| 22:38.39 | rue_mohr | I'm still confused about what causes the radio station to come up, |
| 22:39.05 | rue_mohr | the channelbank has a solid state line interface that I'm suspicious of |
| 22:39.26 | drmessano | Put a few ferrite chokes on the lines coming in |
| 22:39.29 | drmessano | See what happens |
| 22:39.32 | rue_mohr | I know its got a perculier impedence |
| 22:39.54 | rue_mohr | I tried a crude choke, about 20 turns on a 1"^3 core, didn't change anything |
| 22:40.26 | rue_mohr | I'v got trying a 1:1 600R:600R transformer in series with the line on my list |
| 22:40.37 | rue_mohr | you use it as a balanced choke |
| 22:40.49 | drmessano | Ah |
| 22:40.57 | drmessano | Ive used something similar |
| 22:40.58 | rue_mohr | drmessano, have you ever had such an encounter? |
| 22:41.03 | rue_mohr | hmm |
| 22:41.06 | rue_mohr | radio station? |
| 22:41.17 | drmessano | I worked in radio engineering and IT for 12 years |
| 22:41.43 | jaytee | he's seen it all |
| 22:41.56 | rue_mohr | ah |
| 22:41.57 | [TK]D-Fender | rue_mohr: If you see him around, talk to tzanger about the Nortel protocol, he's done a lot of work with them |
| 22:42.04 | drmessano | I had a server room 100ft from a 4000watt AM tower |
| 22:42.14 | rue_mohr | you know, if it wasn't CBC, I woldn't even mind |
| 22:42.15 | [TK]D-Fender | rue_mohr: Or do yourself a favor and forget Nortel tech altogether |
| 22:42.26 | drmessano | heh |
| 22:42.34 | drmessano | So yeah.. Dealt a LITTLE with RFI |
| 22:42.57 | rue_mohr | [TK]D-Fender, I just used the 616 to see if it also picked up the radio station, with the volume all the way up I cant hear it |
| 22:43.03 | jaytee | I don't miss my Nortel switch. it's been gone since end of February. Everyone loves Asterisk with Exchange UM so much more than the old system. |
| 22:43.15 | drmessano | Ground the everloving shit out of everything.. Bond it all to one ground.. Put chokes on whatever is being most annoying |
| 22:43.17 | [TK]D-Fender | drmessano: I love the grand irony that FCC approval hinges on devices' ability to be disrupted ;) |
| 22:43.27 | rue_mohr | I wish my users loved the asterisk system |
| 22:44.00 | drmessano | Good grounding rivals chokes any day |
| 22:44.15 | drmessano | But chokes can supplant where ground has no place |
| 22:44.23 | rue_mohr | I'v grounded it every way east west and south, no change at all |
| 22:44.56 | drmessano | Then it may be induced inside the equipment itself... is it shielded? |
| 22:44.58 | rue_mohr | I put two meters on it and can confirm the tip and ring currents are pretty close, within .1ma |
| 22:45.06 | rue_mohr | yes |
| 22:45.09 | jaytee | we had such bad grounding at one facility in Oklahoma we had to drive copper stakes into the ground and add quicklime for some reason, not sure what the chemisty was all about there. |
| 22:45.43 | rue_mohr | BUT I ran a rf gentorator in the room and swept the stations, it didn't block it, and it would have, so I really dont think its being picked up locally |
| 22:46.20 | rue_mohr | but it only seems to occur when the channelbank is on there |
| 22:46.28 | drmessano | rue_mohr: What sort of RF generator? |
| 22:46.38 | rue_mohr | I know I'm behind a few chokes on the lines |
| 22:46.54 | rue_mohr | a nice big old tube one, its great for silencing neighbours radios |
| 22:47.22 | profXavier | sorry |
| 22:47.28 | profXavier | had a few talkers |
| 22:47.55 | profXavier | Fender, I want to do the following: allow 9******* and 9250******* to work on the Polycom phones |
| 22:48.03 | rue_mohr | I'm also heavily considering, because I only have 1 line, to get a usb pots interface |
| 22:48.19 | profXavier | currently, from the previous output I showed, the 9******* works |
| 22:48.49 | drmessano | That really doesnt tell me a whole lot about its effectiveness.. A wideband noise generator is only going to silence a device intending proper reception of an RF signal.. Incidental oscillation, by its very nature, may not be affected at all by the same device |
| 22:48.54 | profXavier | whereas, the 9250******* fails, when I enter the digits, as it only reads 9 + 7_digits |
| 22:49.20 | rue_mohr | O/C on my pots line is 55v! |
| 22:49.30 | rue_mohr | thought its supposed to top out at 48... |
| 22:50.15 | rue_mohr | runs about 25mA which, looking it up, seems a little high, 20 sounds like the proper current |
| 22:50.24 | seb- | [TK]D-Fender: @home yet? :) |
| 22:50.48 | drmessano | If you had access to an optillator, you could isolate the line |
| 22:50.58 | drmessano | That would be interesting |
| 22:51.40 | [TK]D-Fender | profXavier: Show a complete dialplan sample and SIP debug for each attempt |
| 22:51.50 | [TK]D-Fender | profXavier: Including the actual number dialed. |
| 22:51.55 | rue_mohr | yea, its been a thought, how to try isolating it |
| 22:52.11 | [TK]D-Fender | seb-: yup |
| 22:52.16 | rue_mohr | I'm sure if the pots interface wasnt solid state, this woulnd't be an issue |
| 22:52.22 | drmessano | They make lightning protectors that have a fiber optic link inside them |
| 22:52.28 | [TK]D-Fender | seb-: I'm in |
| 22:52.30 | rue_mohr | give me relays and transformers any day |
| 22:52.33 | drmessano | That would isolate the copper |
| 22:52.35 | [TK]D-Fender | (conference) |
| 22:57.11 | rue_mohr | the pots line interface is on a subcard of the fxo card, I almost think I know enough about it now that I could remove it, build a relay/transofrmer one, and have it work |
| 22:57.32 | rue_mohr | or I can go buy a usb one, and return it if I dont get any difference |
| 22:57.52 | rue_mohr | er, usb? no those ethernet things, the pap2 but withteh co jack |
| 22:58.06 | drmessano | SPA-3102 |
| 22:58.06 | rue_mohr | who makes them now? |
| 22:58.12 | rue_mohr | they any good? |
| 22:58.19 | drmessano | They work for me |
| 22:58.31 | drmessano | Theres some troll in here who cant get them to work |
| 22:58.34 | drmessano | Ignore him |
| 22:58.44 | drmessano | Many of us have used them |
| 22:58.46 | rue_mohr | great, I'll end up being the second |
| 22:58.53 | rue_mohr | can I buy one at london drugs I wonder |
| 22:59.06 | rue_mohr | they current? |
| 22:59.10 | drmessano | yeah |
| 22:59.40 | rue_mohr | do I get to have fun with tftp servers uploading new firmware to them? |
| 22:59.53 | drmessano | No, web based UI |
| 22:59.59 | rue_mohr | aaaawe... |
| 23:00.19 | rue_mohr | hmm so it just uses a sip account? |
| 23:00.33 | drmessano | yes |
| 23:00.41 | rue_mohr | hmm that would make it awefull unchallenging to set up... |
| 23:00.58 | rue_mohr | it have built in echo can? |
| 23:01.28 | rue_mohr | "just works"? |
| 23:01.31 | drmessano | Somewhat.. get it tuned correctly and it wont be a problem |
| 23:01.38 | drmessano | Gain, Impedence |
| 23:01.45 | rue_mohr | hehehe |
| 23:02.03 | drmessano | http://wiki.2l2o.com/index.php/SPA-3102 |
| 23:02.06 | rue_mohr | you know I been fighting with that tooth and claw with my tdm800p? |
| 23:02.20 | drmessano | If that guide doesnt work for you, i dont give a fuck <--- Standard disclaimer |
| 23:02.26 | drmessano | But it works fine |
| 23:02.29 | rue_mohr | hahah |
| 23:02.46 | drmessano | Anyone who has used my guide (except that one guy) have got them working |
| 23:02.58 | [TK]D-Fender | drmessano: Actually I'm pretty sure that if it DOES work for his you still don't give a fuck ;) |
| 23:03.08 | drmessano | Very true |
| 23:03.13 | rue_mohr | and that 1fxo and 1 fxs yes? |
| 23:03.18 | drmessano | yes |
| 23:05.36 | rue_mohr | hmm |
| 23:06.31 | rue_mohr | $50! |
| 23:06.44 | rue_mohr | why the hell did we pay $800 for a tdm800 |
| 23:06.55 | rue_mohr | damnit |
| 23:07.12 | rue_mohr | 16 freaking lines for that |
| 23:08.21 | *** join/#asterisk SaiSoma|AFK (n=SaiSoma@74.167.136.30) |
| 23:08.40 | rue_mohr | grinds his teeth |
| 23:22.10 | profXavier | Fender |
| 23:22.18 | profXavier | ill give it to you now.. |
| 23:22.32 | profXavier | [2-9]11|0T|100|101|011xxx.T|9011xxx.T|1[2-9]xxxxxxxxx|91[2-9]xxxxxxxxx|9[2-9]xxxxxx|9250xxxxxxx|*xx|[8]xxx|[2-7]xx :: Polycom |
| 23:23.19 | *** join/#asterisk Greek-Boy (n=greek@41.222.89.77) |
| 23:23.20 | profXavier | (I just added the 9250xxxxxxx) |
| 23:23.52 | profXavier | not sure what else you want to see |
| 23:24.08 | profXavier | and yes, I didnt enter the actual # I dialed, as its my own # |
| 23:24.22 | *** join/#asterisk MikeJ_ (n=MikeJ@freeswitch/developer/mikej) |
| 23:24.24 | profXavier | but its a 7 digit number, starting with (2-9) |
| 23:27.39 | [TK]D-Fender | profXavier: You showed us dialplan executing with a 9+7 exten. |
| 23:27.52 | [TK]D-Fender | profXavier: Does that somehow not work now? |
| 23:28.02 | profXavier | correct, it doesn't |
| 23:28.13 | [TK]D-Fender | profXavier: What did you have before when it worked? |
| 23:28.24 | profXavier | when the receiver is down, i can enter digits, then hit -dial- |
| 23:28.39 | profXavier | but when I have a dialtone..., it will only allow 9 + 7 digits |
| 23:29.21 | [TK]D-Fender | profXavier: Try reversing the order of those 2 paramteres |
| 23:29.57 | profXavier | so # (7 digits), then 9 ? |
| 23:30.33 | [TK]D-Fender | [2-9]11|0T|100|101|011xxx.T|9011xxx.T|1[2-9]xxxxxxxxx|91[2-9]xxxxxxxxx|9250xxxxxxx|9[2-9]xxxxxx|*xx|[8]xxx|[2-7]xx |
| 23:31.51 | profXavier | sorry, thats the same as I posted... did you change anything ? |
| 23:32.12 | SaiSoma|AFK | he changed the order |
| 23:32.47 | SaiSoma|AFK | i think, bah, too tired to count |
| 23:34.02 | profXavier | oh |
| 23:34.04 | profXavier | i see now |
| 23:34.06 | profXavier | my bad |
| 23:34.08 | profXavier | sorry |
| 23:38.05 | profXavier | trying now |
| 23:38.06 | profXavier | brb |
| 23:38.20 | *** join/#asterisk MmixX (n=mix@61.14.191.137) |
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| 23:48.38 | *** part/#asterisk lanning (n=lanning@nat/yahoo/x-112b92d9a84bf674) |
