00:00.09 | `Sauron | That still doesn't answer the question. The broadcast has to come from somewhere... |
00:00.18 | mafkees | ping -b |
00:00.24 | *** join/#asterisk Ayano (~erik_leee@209.143.187.254) |
00:00.33 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
00:00.34 | ovidiu_25 | this is L3 |
00:00.35 | `Sauron | that's still layer3 |
00:00.57 | `Sauron | I need to send a packet to MAC FF:FF:FF:FF:FF:FF... |
00:01.07 | Luhiwu | what about arp query? |
00:01.43 | mafkees | use a bridging firewall with ebtables |
00:02.00 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
00:02.00 | *** mode/#asterisk [+o bkw_] by ChanServ |
00:02.07 | mafkees | reroute ur package to the mac u want |
00:02.11 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
00:03.00 | ovidiu_25 | but this is no broadcast |
00:03.18 | `Sauron | ovidiu's got the idea |
00:03.36 | mafkees | yeah, I'm lost |
00:03.45 | Ayano | does anyone have any good links on setting up IVR? |
00:03.46 | mafkees | what is it that you want to do ? |
00:03.53 | sivana | what kind of address is this: Mansarover Commercial Complex.near Habibganj Rly. Station |
00:03.57 | Nugget | Ayano: http://voip-info.org/ |
00:04.01 | Luhiwu | `Sauron, try an arp query, it does a L2 broadcast... |
00:04.24 | ovidiu_25 | but only in the broadcast domain |
00:04.40 | ovidiu_25 | the router break the broadcast domain |
00:04.46 | `Sauron | Humm. |
00:04.55 | mafkees | of course |
00:05.02 | `Sauron | ovidiu: That's alright. I'll never leave an L2 domain |
00:05.06 | `Sauron | s/an/the |
00:05.16 | mafkees | you don't want to address tho whole internet |
00:05.24 | harryvv | 142 is a little steep for a simple spa 1000 but thats the going rate for a reseller up here in BC canada |
00:05.25 | `Sauron | for my application, there will ever only exist a local subnet |
00:05.43 | `Sauron | harryv: you can get spa1k1's for less than that off the internet |
00:05.51 | mafkees | why stick to level 2 ? |
00:05.58 | `Sauron | Because I want to? :) |
00:06.00 | mafkees | lol |
00:06.03 | harryvv | yes...but there is the issue with gst/pst taxes |
00:06.18 | `Sauron | mafkees: It's part of the design requirement, as weird as that sounds. |
00:06.27 | mafkees | but if ur net is limited, the broadcast address is known |
00:06.32 | *** join/#asterisk sjaak538 (~sjaaknabu@d5c53145.dsl.concepts.nl) |
00:06.46 | harryvv | and anyway i dont want to wait. Wife is complaining about the IVR playing when she picks up the analog phone and cannot stop it. |
00:06.49 | ovidiu_25 | only for l3 |
00:07.09 | mikegrb | harryvv: so put asterisk in line with all the phones |
00:07.16 | mikegrb | one fxo, one fxs |
00:07.22 | mikegrb | asterisk wasn't designed to share the line |
00:07.25 | `Sauron | mafkees: I'm trying to broadcast data between wireless devices - and want to do as little configuration as possible |
00:07.37 | harryvv | mike, i know |
00:07.45 | mafkees | what kind of data ? |
00:07.52 | `Sauron | doesn't matter |
00:08.02 | ovidiu_25 | but wireless use diferent protocol |
00:08.24 | harryvv | to bad there was a dtmf code to shut off the ivr is my wife does not pickup the phone in time. |
00:08.25 | mafkees | layer 2 is the same, right ? |
00:08.33 | `Sauron | yes, it is |
00:08.37 | `Sauron | only layer1 is different |
00:08.41 | mafkees | yeah |
00:08.44 | mikegrb | harryvv: then add the dtmf code |
00:08.48 | mafkees | that's what I thought too |
00:09.04 | ovidiu_25 | yap |
00:09.05 | harryvv | mike, thats a idea |
00:09.16 | mikegrb | it's as simple as 9,1,hangup |
00:09.20 | mikegrb | and then she dials 9 |
00:09.23 | harryvv | anyway im commited to buy this thing :) |
00:09.49 | Beirdo | mikegrb: I am having fun with this :) got firefly installed on the work laptop now |
00:09.56 | mikegrb | :D |
00:10.07 | mikegrb | firefly is great for a free softphone |
00:10.10 | mikegrb | and it's iax! |
00:10.15 | Beirdo | precisely |
00:10.20 | Beirdo | only does one call instance though |
00:10.28 | Beirdo | and that kinda sucks |
00:10.31 | mikegrb | ja :/ |
00:10.39 | ovidiu_25 | but the wireless devices all the time make the broadcast |
00:10.45 | Beirdo | X-lite has the 3 lines, which is useful, too bad it's SIP |
00:11.27 | moonwick | what's wrong with SIP? |
00:11.39 | Beirdo | three letters |
00:11.41 | Beirdo | NAT |
00:11.45 | moonwick | ah |
00:11.49 | Nugget | that sounds like something wrong with NAT. |
00:11.50 | ovidiu_25 | all the clients near an AP are MAC known |
00:11.57 | moonwick | seems okay for me, behind nat |
00:12.03 | mafkees | ever tried to set it up between a server behind NAT and a client behind nat ? |
00:12.04 | moonwick | but my server's on a public IP |
00:12.07 | Beirdo | no, SIP is a NAT-unfriendly protocol by nature |
00:12.18 | moonwick | servers do not belong behind NAT. |
00:12.20 | Nugget | putting a server behind nat is just dumb. that's not SIP's fault. |
00:12.21 | Beirdo | you can hack it into behaving, but it wasn't designed for it |
00:12.27 | `Sauron | dum di dum |
00:12.30 | Nugget | nat makes all sorts of things break. |
00:12.47 | Beirdo | not many things I have problems with with NAT |
00:12.48 | Nugget | NAT is a protocol-unfriendly protocol. |
00:13.00 | greg_work | i thought there were problems even if just your client was behind nat? |
00:13.04 | Beirdo | and I'm not paying to get a second static IP on my DSL, thanks |
00:13.05 | `Sauron | most/many udp-based things break with nat |
00:13.07 | mafkees | Nugget: unless you have a state matching module in your firewall |
00:13.18 | moonwick | nat isn't really a protocol. :) |
00:13.37 | greg_work | moonwick: s/really// |
00:13.37 | *** part/#asterisk ovidiu_25 (~dd@80.96.223.40) |
00:13.42 | Beirdo | the only thing that breaks for me with nat are protocols that are stupid enough to send the IP address INSIDE the protocol |
00:13.44 | Nugget | the only thing that sucks more than nat is not having connectivity, so nat beats the alternative in many cases, but nat blows goats any way you look at it. |
00:13.47 | Beirdo | like SIP |
00:14.07 | moonwick | nugget's sitting pretty on his stable of IPs over there |
00:14.11 | moonwick | :P |
00:14.15 | Nugget | *shrug* |
00:14.21 | Nugget | I'm willing to pay to avoid nat suckage. |
00:14.23 | greg_work | Nugget: there are benefits to the way nat works. nat is like a condom for windows machines |
00:14.28 | mafkees | we just need ip v6 |
00:14.30 | mafkees | :) |
00:14.34 | `Sauron | hum di dum |
00:14.36 | Nugget | nat is not a security tool. |
00:14.44 | mafkees | nat sux |
00:14.45 | `Sauron | I doubt arp does any network requests |
00:15.03 | `Sauron | dum di dum |
00:15.43 | greg_work | Nugget: sure it is. the popularity of nat routers for residential users means less vulnerable windows machines on the internet |
00:15.59 | Nugget | only by accident and not very effectively. |
00:16.02 | Beirdo | say what you want, Nugget: at this point, the ONLY thing I have NAT issues with is SIP. everything else works fine |
00:16.24 | Beirdo | even FTP (with a helper) and H323 |
00:16.29 | greg_work | granted, it is simply making up for the absolute lack of even basic security that windows and most windows applications have .. |
00:17.01 | mafkees | not only windows |
00:18.27 | Beirdo | there are valid reasons for using NAT, and valid reasons not to. |
00:18.56 | mafkees | indeed |
00:19.24 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
00:19.47 | Beirdo | but as it *is* a fact of life that NAT will be around until IPv4 dies, designing protocols to be NAT-unfriendly is boneheaded. |
00:20.00 | mafkees | as long as ipv6 is not supported by 99% of all the services Out There (tm) nat is the only way for home users to have mone then 1 networked device hooked up to the internet |
00:20.29 | Beirdo | well, without paying loads of money to their ISP |
00:20.45 | SexyKen | Asterisk can process PHP files? |
00:21.08 | mafkees | SexyKen: yes, as agi scripts |
00:21.47 | mafkees | we use it here to lookup caller id info on ISDN BRI channels |
00:22.39 | mafkees | exten => our_phone_nr_1,1,agi(lookup.agi) |
00:22.53 | mafkees | that lookup.agi as a cli php script |
00:22.58 | greg_work | what makes you guys think ipv6 is going to solve the NAT problem? |
00:23.13 | `Sauron | dum di dum |
00:23.30 | SexyKen | •mafkees• So do I need actual PHP installed or no? |
00:23.31 | Sedorox | you don't need nat with IPv6.. every machine has a unique addy |
00:23.33 | mafkees | greg_work: cause every cable/dsl connection will have 64k routable ip addies |
00:23.44 | mafkees | SexyKen: yes |
00:23.46 | greg_work | specifically, what makes you think that ISPs will suddenly start giving out multiple addresses for free, when it's a fee-per-IP service now? |
00:23.57 | Beirdo | greg_work: IPv6 tunnelling |
00:24.05 | SexyKen | •mafkees• So once php is installed it'll work out of the box? |
00:24.08 | Sedorox | greendisease: fee's for ipv6? ahah |
00:24.10 | greg_work | Beirdo: to where? |
00:24.16 | mafkees | no, tunneling is as bad as NAT |
00:24.25 | Sedorox | no it isn't |
00:24.27 | mafkees | SexyKen: yes |
00:24.42 | greg_work | i mean, its nice to think that because its technically possible and numerically feasable, they'd do it |
00:24.46 | Sedorox | tunneling with IPv6 has just about, sometimes better, latency then native ipv4 |
00:24.52 | greg_work | but i highly doubt they'd give up a possible revenue stream |
00:25.05 | greg_work | why don't telco's stop charging for LD costs? |
00:25.20 | Beirdo | well, the reason you pay for extra IPs now is because they are a relatively rare commodity |
00:25.21 | greg_work | pratically, theres really not much difference anymore between a local and LD call on the PSTN |
00:25.33 | greg_work | yet, LD calls from ma bell on a POTS line are expensive |
00:25.41 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
00:25.49 | mafkees | greg_work: here in .nl they already have dsl connections that give you 1 ipv4 addy and a whole /64 ipv6 net |
00:25.52 | Beirdo | right, and we get around that by not getting service from them |
00:26.17 | greg_work | mafkees: ok, well, thats a selling point. once ipv6 is mainstream i doubt you'd see that happen as much. |
00:26.20 | Beirdo | same thing goes with the ISP, you can tunnel your IPv6 to somewhere that will route for you |
00:26.29 | neopher | <---- banging head against server |
00:26.37 | greg_work | i may be wrong. ISPs doing that now may mean everyone does it.. depends on what gets establish i guess |
00:26.51 | Beirdo | true |
00:26.55 | mafkees | uhhuh |
00:27.01 | greg_work | i'm just saying, most ISPs are used to charging more for extra IPs, I don't see any reason why they'd suddenly decide to give out extra IPs for free |
00:27.10 | Beirdo | because they can |
00:27.18 | Beirdo | and because it will get them customers |
00:27.23 | mafkees | but you can already get ur ipv6 /64 net for free @ 6bone |
00:27.25 | neopher | anyone here get a cisco 30VIP to work with chan_sccp? |
00:27.26 | Sedorox | they only charge because it costs them |
00:27.27 | greg_work | if it will get them customers - sure |
00:27.34 | greg_work | but "Because they can" is NOT a business case |
00:27.39 | Nugget | heh |
00:27.40 | Beirdo | Sedorox has hit it on the head |
00:28.17 | Nugget | When I moved froma /29 to a /28 it was only a one-time charge. I was amazed to hear that it wouldn't increase my monthly costs. |
00:28.18 | mafkees | neopher: I had some bad experience with chan_sccp |
00:28.29 | Beirdo | greg_work: if a SINGLE ISP in your area offered free IPv6 net and the others didn't, do you not think that would give them a lot more business? |
00:28.52 | greg_work | i can get additional IPs at my colo provider for free too, as long as i justify them |
00:28.56 | mafkees | I would go for the one with free ipv6 net for sure |
00:28.57 | Nugget | I'd love to be able to get legitimate ipv6 space, though. |
00:29.08 | Beirdo | mafkees: so would I. there is the business case |
00:29.14 | greg_work | Beirdo: right now? no, i don't care about ipv6 |
00:29.15 | mafkees | greg_work: colo providers are different |
00:29.16 | neopher | you still have to tunnel that IPv^ trough you isp and the isp must support it as well as your router |
00:29.20 | Beirdo | by giving them away, they can get more customers |
00:29.22 | harryvv | exten => 91,1,Hangup stops the IVR but does not hangup the zap |
00:29.26 | mafkees | I was talking about house dsl lines |
00:29.28 | *** join/#asterisk sezuan (sezuan@port-212-202-57-119.dynamic.qsc.de) |
00:29.33 | harryvv | ohh wait i think i can fix that |
00:29.40 | Beirdo | then once one does, the others will follow suit so they don't lose customers |
00:29.46 | greg_work | mafkees: i know, i was intending that for Nugget |
00:29.58 | mafkees | neopher: here in .nl there are providers that have native ipv6 stacks already |
00:30.14 | greg_work | Beirdo: yeah, but like i said, it depends on who gets in at what time |
00:30.40 | greg_work | Beirdo: if the aol's and whoever else is a big isp these days decides to start charging, other isps might follow suit |
00:30.44 | Beirdo | true, but ultimately, IPv6 will likely be the only thing that can or will eradicate NAT |
00:30.52 | Beirdo | nah |
00:31.01 | dsmouse | even that won't |
00:31.01 | greg_work | "hey look, that isp is making money off something we give away for free" |
00:31.10 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
00:31.12 | Beirdo | they will lose so many customers they will have no choice |
00:31.35 | mafkees | indeed |
00:31.37 | greg_work | maybe |
00:31.37 | neopher | wow, well your ahead of my isp, they do not distribut ipv6, have to get a block from HE (hurricain Electric) and then route it towards my ipv4 address |
00:31.47 | dsmouse | it'll solve it there, but a lot of companies do it for security too |
00:31.47 | Beirdo | customers go where they get more service for less money |
00:31.51 | dsmouse | it's a bad idea, but... |
00:31.54 | Nugget | with ipv6 there's really only one good reason remaining for nat -- isp independence, it makes swapping providers less painful because it allows you to avoid internal renumbering. |
00:32.00 | greg_work | if you can only get free additional addresses from hte fly-by-night ISP that goes down all the time and only has 9/5 tech support ...... |
00:32.07 | Beirdo | sigh |
00:32.12 | greg_work | i'd be willing to pay the additional fee. |
00:32.17 | Nugget | but I'm confident that we'll be able to correct that. |
00:32.31 | Beirdo | NAT for security is *not* a bad idea |
00:32.33 | *** join/#asterisk yurpls (~yurplsl@65.114.0.198) |
00:32.35 | Nugget | yes it is. |
00:32.39 | Beirdo | how so? |
00:32.45 | mafkees | it is |
00:32.48 | greg_work | Nugget: you can still run an internal network duplicated over top |
00:32.53 | Nugget | because it's based on the false premise that all attackers are "out there" |
00:32.54 | Sedorox | its security through obsecurity.. which is a bad idea |
00:33.06 | Nugget | which more often than not is incorrect. |
00:33.07 | Beirdo | Sedorox: partially |
00:33.15 | mafkees | and it's not even obscured |
00:33.17 | Nugget | and because nat isn't really a security mechanism. |
00:33.25 | greg_work | Nugget: its certainly not the be-all-end-all of security.. but it does protect against that thousands of attackers that are on the internet |
00:33.30 | mafkees | just have a peek at the tcp window scaling option |
00:33.36 | Beirdo | by port forwarding your incoming ports, you secure those ports on all the other machines |
00:33.40 | mafkees | and you know what os is in there |
00:34.03 | Nugget | for sufficiently laughable values of "protect" |
00:34.14 | Beirdo | that's not by obscurity, that's by proper setup |
00:34.17 | greg_work | Nugget: with windows, theres no good firewall. (i dunno, maybe with xp? i still use 2k, no compelling reasons to switch) |
00:34.29 | Nugget | so run a firewall. |
00:34.47 | greg_work | oh, you mean ,like a NAT router that doubles as a firewall because that's the way NAT works? |
00:34.54 | Nugget | no, I mean like a firewall. |
00:34.56 | mafkees | with windows, put a bridge between the internet routing hardware and your windows box |
00:34.56 | Beirdo | Nugget: in case you were unaware, most people who do NAT do it on a firewall |
00:34.58 | Nugget | the nat is unrelated. |
00:35.06 | welby | have an actual firewall running as well |
00:35.11 | Nugget | sure, there are devices that are firewalls and nat routers. |
00:35.12 | SexyKen | Instead of dialing one call at a time in a follow me sequence, how can I dial all phones at the same exact time? |
00:35.19 | Nugget | but that doesn't make nat a component of the firewall. |
00:35.27 | welby | the firewall may be the same package / software (like in pf or iptables) |
00:35.31 | Sedorox | SexyKen: queue... |
00:35.33 | Sedorox | ringall |
00:35.33 | dsmouse | SexyKen: and just have one pick up? |
00:35.33 | welby | but its still a different thing |
00:35.34 | Beirdo | if you are dumb enough to use NAT as the only security measure, you deserve what you get |
00:35.46 | SexyKen | •dsmouse• Yes. |
00:35.47 | greg_work | Nugget: find a firewall applicance near the price of a home/soho-targetted "router" |
00:35.56 | Nugget | nat is not a security measure. it's just a hacky routing trick. |
00:35.57 | SexyKen | •dsmouse• All would be cool as well. |
00:35.58 | Beirdo | but saying using NAT as part of your security is bad is just wrong |
00:36.01 | welby | greg_work: easy, wrt54g |
00:36.10 | dsmouse | SexyKen: queue... ringall |
00:36.11 | SexyKen | dsmouse, Here is current code: http://pastebin.ca/6095 |
00:36.19 | greg_work | welby: that is a nat/firewall/router .. whatever you want to call it. |
00:36.21 | greg_work | it does NAT |
00:36.29 | welby | yes |
00:36.32 | Sedorox | bah |
00:36.33 | *** part/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
00:36.38 | SexyKen | dsmouse, I'd have to set it up as a queue? |
00:36.48 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
00:36.48 | dsmouse | I *think* so |
00:36.59 | SexyKen | Can I have it ring all at once and have them all connect? |
00:37.01 | mafkees | nat = Network Address Translation |
00:37.02 | greg_work | Nugget: yes, i'll give you that .. it IS a hacky routing trick. but it adds some security |
00:37.05 | mafkees | that's all |
00:37.17 | dsmouse | SexyKen: not unless you interface it with a conference somehow |
00:37.18 | welby | however, even with the default firmware (ie linksys one) it does proper firewalling and natting |
00:37.30 | dsmouse | I wonder how that could work |
00:37.34 | Beirdo | either way, my whole point was that NAT is a necessity these days due to the way IPv4 works, and SIP sucks because they intentionally ignored that fact. |
00:37.41 | greg_work | hehe |
00:37.45 | greg_work | yeah that got a bit OT :) |
00:37.58 | mafkees | Beirdo: and that is true ;) |
00:38.10 | jets | Does * use tcp or udp generally for sip |
00:38.18 | Beirdo | UDP |
00:38.19 | greg_work | why DID they ignore that, anyways? |
00:38.19 | mafkees | udp |
00:38.20 | dsmouse | udp always |
00:38.33 | Beirdo | greg_work: no idea, but it is a PITA |
00:38.38 | greg_work | is there a reason, or is it "just the way it works(tm)" ? |
00:38.42 | SexyKen | How doe sone make astersik play back a number |
00:38.56 | Beirdo | they put the IP addresses inside the SIP packets |
00:39.00 | Beirdo | heheh |
00:39.14 | mafkees | that is bad anyways |
00:39.26 | mafkees | only the IP header should have that |
00:39.29 | Beirdo | now IPSEC I can forgive for being intentionally NAT unfriendly |
00:39.38 | Beirdo | it's an encrypted tunnel |
00:39.45 | Beirdo | SIP - there's no excuse |
00:39.48 | mafkees | IPSEC is NOT nat unfriendly |
00:39.56 | tzanger | mafkees: actually yes it is |
00:40.04 | tzanger | you need to use nat-t to get through it |
00:40.05 | mafkees | depends on the ipsec implementation |
00:40.05 | Beirdo | mafkees: quite so |
00:40.20 | tzanger | I mean think about it -- AH ensures that the headers weren't fucked with and that's exactly what NAT does |
00:40.35 | mafkees | it only needs 1 rdr rule here |
00:40.48 | Nugget | this whole issue just feels to me like the people that complain about licenses that are not compatible with the gpl. Lots of stuff is hard to accomodate with nat, just as many licenses can't coexist with the gpl. but in both cases it's the fault of nat and of the gpl that they're so unaccomodating. :) |
00:40.56 | mafkees | the rest is just plane state related stuff |
00:41.19 | Beirdo | how the hell is NAT unaccomodating? |
00:41.28 | mafkees | gpl is evil anyways |
00:41.32 | Beirdo | all it is is remapping IP addresses in the damned IP header |
00:41.39 | Nugget | because it turns the bidirectional nature of the internet into a fudamentally client-server model. |
00:41.43 | tzanger | Beirdo: it doesn't fix any addresses in the data payload |
00:41.47 | Nugget | anything that needs or wants to go both directions has problems. |
00:42.07 | Beirdo | tzanger: there should *BE* no addresses in the data payload, that's what the headers are for |
00:42.22 | mafkees | I agree Beirdo |
00:42.34 | mafkees | Data != routing info |
00:42.47 | tzanger | maksim: I agree but that is what makes NAT unaccomdating :-) |
00:42.56 | Beirdo | Nugget: a good NAT implementation has no bidirectional issues |
00:42.57 | tzanger | Beirdo: agree 100% |
00:43.03 | tzanger | Beirdo: bullshit |
00:43.16 | Nugget | Beirdo: well then, if that's truly the case then it's absolutely not adding any security either. :) |
00:43.17 | tzanger | a packet comes in to your NAT box, port 8342 |
00:43.18 | SexyKen | Is there anyway to make just a certain extension use a different hold file |
00:43.22 | neopher | i use nat and have no probs with sip, even remotely |
00:43.23 | tzanger | now which of your 100 "clients" is it for? |
00:43.46 | Beirdo | tzanger: it is for the one that sent to that socket, or it gets bounced |
00:43.47 | mafkees | tzanger: there should be a state for it |
00:43.49 | tzanger | NAT by its very design has issues with bidirectional transfers |
00:43.52 | Beirdo | or you port forward it |
00:43.55 | tzanger | Beirdo: as in client-server |
00:43.58 | tzanger | I had to make the request first |
00:44.14 | mafkees | of course |
00:44.17 | Beirdo | not if you port-forward the incoming ports you *want* |
00:44.19 | Beirdo | duh |
00:44.21 | mafkees | but that's not only with nat |
00:44.27 | mafkees | that's with all firewalling |
00:44.29 | SexyKen | Is there anyway to make just a certain extension use a different hold file |
00:44.33 | Ahewes | Got a problem with far end disconnect on an adtran ta750 |
00:44.40 | mafkees | all protocols that use dynamic ports |
00:45.01 | tzanger | Beirdo: but now you're patching up your NAT |
00:45.12 | Beirdo | tzanger: no I'm not |
00:45.13 | tzanger | Beirdo: and again, you can't have two clients share the same port |
00:45.17 | tzanger | sure you are |
00:45.19 | bkw_ | http://bkw.digiweb.com/conf.gs |
00:45.22 | bkw_ | http://bkw.digiweb.com/conf.gsm |
00:45.24 | tzanger | you're explicitly telling your NAT box what to do |
00:45.26 | bkw_ | for the dev conf recording |
00:45.28 | mafkees | if you close everything and only allow ftp-data, how will you handle passive ftp ? |
00:45.29 | bkw_ | just in case anyone wants it |
00:45.36 | tzanger | thanks bkw_ |
00:45.37 | mafkees | ever on your border firewall ?? |
00:45.38 | mafkees | not |
00:45.48 | bkw_ | cant' pasive ftp work on just one port? |
00:45.53 | bkw_ | wasn't that the point of pasv ftp? |
00:46.05 | *** join/#asterisk vooduhal (~christoph@pcp01069659pcs.andrsn01.tn.comcast.net) |
00:46.18 | Beirdo | mafkees: passive FTP opens the port from the client to the server, that's how |
00:46.27 | mafkees | bkw_: it supports multiple ftp clients shoring the same public ip |
00:46.32 | Beirdo | active FTP opens the data port from the server to the client |
00:46.40 | mafkees | 01:46 < Beirdo> mafkees: passive FTP opens the port from the client to the server, that's how |
00:46.43 | mafkees | indeed |
00:46.50 | mafkees | so if the server denies that |
00:47.03 | Beirdo | why would it be denying it? |
00:47.03 | mafkees | (as it should, cause it's a high port) |
00:47.08 | bkw_ | no isn't pasv ftp let the client tell you what port ot use? |
00:47.16 | bkw_ | not the otherway around |
00:47.19 | bkw_ | so you can TELL it what port to use |
00:47.26 | Beirdo | and who the hell runs an FTP server behind NAT? |
00:47.35 | mafkees | a lot of ppl |
00:47.36 | Luke-Jr | Active FTP -- Server connects to client on client-specified host and port |
00:47.49 | Luke-Jr | Passive FTP -- Client connects to server on server-specified port |
00:47.51 | bkw_ | doesn't make sense the other way |
00:47.57 | neopher | wow, when did they integrate ftp with VoIP, hehe |
00:48.07 | bkw_ | either way FTP sucks |
00:48.09 | sezuan | When I receive a call through a SIP/PSTN gateway (register => login:pass@gw), do I have access to the called number? |
00:48.10 | mafkees | neopher: tftp |
00:48.11 | Luke-Jr | neopher: no idea |
00:48.14 | mafkees | ;) |
00:48.16 | Luke-Jr | bkw_: agreed. I hate FTP |
00:48.39 | sezuan | Except SIPGetHeader |
00:48.51 | mafkees | Luke-Jr: but what's the alternative ? |
00:48.52 | Luke-Jr | sezuan: some providers use it instead of 's' for a starting ext |
00:48.54 | Nugget | sure, but ftp sucks for reasons which are unrelated to the difficulty of using ftp in a nat environment. :) |
00:48.55 | Beirdo | FTP is another NAT-unfriendly protocol, but it was invented way before NAT was thought of |
00:48.58 | Luke-Jr | mafkees: SSH |
00:49.06 | mafkees | yeah right |
00:49.09 | bkw_ | Luke-Jr you have it backwards |
00:49.14 | Nugget | mafkees: what's wrong with that? |
00:49.14 | bkw_ | pasv ftp the client says what ports |
00:49.25 | mafkees | overhead ? |
00:49.27 | dsmouse | mafkees: http? |
00:49.28 | Nugget | ssh is a perfectly usable and dramatically superior alternative to ftp. |
00:49.29 | Luke-Jr | bkw_: Nope |
00:49.38 | bkw_ | yes |
00:49.43 | bkw_ | PASV the client sets it up |
00:49.43 | sezuan | Luke-Jr: I should be able to use directly in the extensions.conf? |
00:49.43 | Beirdo | Nugget: ain't that the truth |
00:49.44 | Nugget | for many reasons. |
00:49.45 | Luke-Jr | mafkees: What overhead? |
00:49.47 | mafkees | Nugget: uhhuh, if you have enuf bandwidth |
00:49.54 | Nugget | bandwidth? |
00:49.55 | Luke-Jr | sezuan: sure... set a global var to the dialed # |
00:49.58 | mafkees | Luke-Jr: all the encryption |
00:50.04 | Nugget | I think you have an imperfect grasp of the issues. |
00:50.05 | bkw_ | Passive FTP (sometimes referred to as PASV FTP because it involves the FTP PASV command) is a more secure form of data transfer in which the flow of data is set up and initiated by the File Transfer Program (FTP) client rather than by the FTP server program. |
00:50.10 | Luke-Jr | mafkees: doesn't use much CPU, in my experience |
00:50.19 | mafkees | not much CPU |
00:50.31 | mafkees | but a LOT of tcp traffic |
00:50.31 | bkw_ | pasv = client side |
00:50.34 | bkw_ | NEXT!!! |
00:50.40 | Luke-Jr | bkw_: the SYN comes from the client |
00:50.44 | sezuan | Luke-Jr: How do I access the dialed number? |
00:50.45 | Luke-Jr | bkw_: not the port/host info |
00:50.56 | bkw_ | dumb ass don't argue with me |
00:51.05 | Qwell | bkw_: You are SO wrong. |
00:51.05 | Luke-Jr | sezuan: ${EXTEN} if they use it for your exten |
00:51.10 | Qwell | Don't ban me :P |
00:51.11 | bkw_ | no i'm not |
00:51.12 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
00:51.13 | bkw_ | i'm reading it |
00:51.21 | Qwell | dunno, I just got here |
00:51.24 | Luke-Jr | bkw_: whatever. that's what actually happens, in my experience |
00:51.29 | mafkees | a ftp link (without encryption) is around 30 to 60% faster than scp |
00:51.32 | bkw_ | riiight |
00:51.37 | Beirdo | mafkees |
00:51.39 | Luke-Jr | mafkees: over what connection? |
00:51.42 | SexyKen | Can I choose the music on hold to use for a certain extension??? |
00:51.42 | Beirdo | it shoudn't be |
00:51.46 | mafkees | dsl 8 mbit |
00:51.50 | bkw_ | Luke-Jr you're dyslexic |
00:51.58 | Luke-Jr | bkw_: ... |
00:52.00 | Qwell | SexyKen: like MusicOnHold(1234)? |
00:52.09 | Qwell | musiconhold.conf, or whatever. |
00:52.16 | sezuan | Luke-Jr: EXTEN is not the name/number in front of the @ in the To: header? |
00:52.17 | bkw_ | luke you're backwards |
00:52.20 | mafkees | SexyKen: yes: SetMusicOnHold |
00:52.28 | brc_ | Luke-Jr, wrong |
00:52.31 | SexyKen | •mafkees• So how would I set it ? |
00:52.33 | Silik0n | someone doesnt understand pasv mode obviously |
00:52.48 | Silik0n | PASV is needed because NAT breaks active FTP |
00:52.56 | Qwell | rtfrfc? |
00:53.07 | bkw_ | http://searchnetworking.techtarget.com/sDefinition/0,,sid7_gci512897,00.html |
00:53.18 | dsmouse | no, pasv is needed cause people still use ftp |
00:53.19 | Nugget | mafkees: so turn off encryption if it bugs you so much. |
00:53.28 | mafkees | SexyKen: on your asterisk console type: show application setmusiconhold |
00:53.29 | Nugget | you still get the other benefits of scp/sftp. |
00:53.43 | vooduhal | exit |
00:54.02 | mafkees | Nugget: no, it does't bug me, it's just the downside of using SSH |
00:54.03 | Silik0n | unless you have a active proxy that fixes the FTP protocol messages active mode does not work because the server will try to open aditioonal ports the nat box and the nat box will not know what to do with them |
00:54.13 | Nugget | mafkees: how can it be a downside if it can be disabled? |
00:54.18 | Nugget | it's just a factor. |
00:54.19 | Silik0n | thus the whole idea behind pasv where the client controls eeverything |
00:54.33 | Beirdo | http://www.faqs.org/rfcs/rfc1579.html |
00:54.40 | Luke-Jr | I can prove it :) |
00:54.43 | Beirdo | maybe we should all go read the RFC |
00:54.45 | mafkees | Nugget: speed is everything |
00:54.49 | Qwell | Beirdo: Thats what I already said, heh |
00:54.59 | Qwell | Beirdo: rtfrfc, ya know? ;] |
00:55.08 | Nugget | mafkees: so turn on ssh's compression (big win over ftp) and disable encryption. |
00:55.09 | Beirdo | yup |
00:55.10 | ManxPower | This E&M Wink problem is killing me. |
00:55.12 | Nugget | sounds like ssh is superior. |
00:55.14 | Beirdo | RTRFC1579 |
00:55.20 | Nugget | if you think that speed is the only thing that matters. |
00:55.23 | dsmouse | still, "mode i" therfore ftp sucks. |
00:55.28 | Silik0n | Luke-Jr what just cause the server says hey I can use ports 10000 to 11000 to the client doesnt mean the server controls it... all comms in PASV mode originate from the client NOT from the server |
00:55.29 | bkw_ | haha |
00:55.30 | mafkees | disabling encryption ??????? |
00:55.32 | mafkees | wtf |
00:55.36 | *** join/#asterisk florz (nobody@odnb-d9baa40a.pool.mediaWays.net) |
00:55.52 | Nugget | yes. you are aware that you can disable encryption for the scp transfer, right? |
00:56.00 | Nugget | since that's what you're complaining about. |
00:56.00 | mafkees | uhhuh |
00:56.02 | Luke-Jr | http://pastebin.ca/6096 |
00:56.07 | Beirdo | or use a less expensive one :) |
00:56.11 | Luke-Jr | ftp.gnu.org specifies host and port |
00:56.32 | mafkees | no, I'm saying it will lose from ftp when you look at the speed |
00:56.36 | Luke-Jr | 199.232.41.7 on port (134 * 256) + 185 |
00:56.52 | Luke-Jr | mafkees: if you want speed, use TFTP |
00:57.12 | mafkees | Luke-Jr: or a dvd ;) |
00:57.12 | dsmouse | what's quicker, ftp, gopher, or http |
00:57.33 | Luke-Jr | dsmouse: HTTP |
00:57.39 | Beirdo | none of the above |
00:57.41 | mafkees | dsmouse: gopher |
00:57.44 | florz | dsmouse: what's louder, a speaker or a human? |
00:57.48 | Beirdo | they are all TCP protocols |
00:57.56 | Silik0n | florz: a stupid human |
00:57.59 | *** part/#asterisk eKo1 (~bernd@207.42.191.66) |
00:58.00 | Luke-Jr | HTTP is faster only because the initial latency is lower |
00:58.01 | florz | Silik0n: =:-) |
00:58.05 | Beirdo | the ultimate speed will be determined by your TCP windowing |
00:58.06 | *** join/#asterisk Gator (~krp@adsl-068-209-187-058.sip.gnv.bellsouth.net) |
00:58.07 | Luke-Jr | and QoS is generally highest |
00:58.16 | Silik0n | hah |
00:58.17 | mafkees | Beirdo: no |
00:58.23 | Beirdo | initial latency has nothing to do with it |
00:58.30 | dsmouse | Silik0n: itym a stupid human that just screwed up his computer 20 mintues before a presentation to his biggest client |
00:58.37 | Beirdo | download a 1G file with all three :) |
00:58.38 | Luke-Jr | Beirdo: they all transfer the data the same way, so initial latency is all that's left |
00:58.51 | Silik0n | dsmouse hah |
00:59.01 | Beirdo | the initial latency will be in the range on ms on a several hour transfer |
00:59.08 | mafkees | Luke-Jr: no, you have to take the overhead in account |
00:59.09 | Beirdo | it's irrelevant |
00:59.12 | Luke-Jr | Beirdo: in which case, you won't notice any difference |
00:59.26 | Luke-Jr | mafkees: there is no overhead diff for HTTP and FTP |
00:59.32 | Luke-Jr | they send the data over TCP the same way |
00:59.32 | mafkees | there is |
00:59.47 | Beirdo | there can be if your TCP windowing is done differently in the apps |
00:59.59 | bkw_ | ok guys lets move on |
00:59.59 | Luke-Jr | Beirdo: that's the app then, not the protocol |
01:00.01 | bkw_ | you're all stupid |
01:00.02 | mafkees | my tcp windowing is fixed |
01:00.04 | bkw_ | NEXT!!! |
01:00.14 | bkw_ | and I can say that because its my birthday |
01:00.16 | mafkees | yeah |
01:00.18 | bkw_ | :P |
01:00.18 | mafkees | NEXT |
01:00.19 | Beirdo | bkw_: yeah, enough of this special olympics :) |
01:00.21 | Luke-Jr | bkw_: did you look at my proof against you, at least? |
01:00.32 | bkw_ | Luke-Jr yes dear I did |
01:00.34 | mafkees | NEXT == my naked wife in our bed ;) |
01:00.35 | bkw_ | but you still don't get it |
01:00.37 | mafkees | later all |
01:00.47 | Luke-Jr | bkw_: what don't I get? |
01:00.53 | dsmouse | GAH |
01:00.58 | bkw_ | Luke-Jr shut up now please |
01:01.03 | Beirdo | Luke-Jr: just drop it will ya? |
01:01.04 | Silik0n | hah |
01:01.10 | Luke-Jr | meh |
01:01.15 | Beirdo | sorry we even got into this in the first place |
01:01.17 | Luke-Jr | why did FTP come up in the first place? |
01:01.30 | bkw_ | no fuckin clue |
01:01.34 | bkw_ | but lets solve other issues |
01:01.37 | Beirdo | amen |
01:01.53 | Beirdo | happy birthday to bkw_ |
01:02.10 | Luke-Jr | bkw_: so how would I get started doing * devel? |
01:02.22 | bkw_ | open up app_skel.c |
01:03.15 | *** join/#asterisk dontmsgme (~none@adsl-68-121-22-193.dsl.irvnca.pacbell.net) |
01:03.29 | dontmsgme | How come with Nufone I bought a DID 3 weeks ago and whenever I call it it says "you are unreachable" |
01:03.37 | Beirdo | wow, look at the time... |
01:03.50 | greg_work | ugh, the only thing worse than spam, is spam from people who don't know how to set their computer's clock |
01:03.58 | Beirdo | heh |
01:04.21 | greg_work | just got one from 1/18/2001 |
01:04.22 | |Vulture| | dontmsgme: prolly didn't register it right |
01:04.27 | dontmsgme | in iax? |
01:04.42 | |Vulture| | dontmsgme: does it show on the * box when you call it at all? |
01:04.44 | greg_work | i suppose it IS the 18th today.. but even so |
01:04.54 | dontmsgme | No |
01:05.07 | bkw_ | paris hiltons birthday was yesterday |
01:05.09 | bkw_ | that whore |
01:05.14 | dontmsgme | But I put the correct Register syntax |
01:05.36 | |Vulture| | dontmsgme: do a "iax2 show registry" |
01:06.08 | dontmsgme | It says State: Registered |
01:06.41 | dsmouse | dontmsgme: did you do a "iax2 debug" and call the number? |
01:06.56 | |Vulture| | strange, if its setup right you should see the # trying to hit your server but not going anywhere if extensions.conf isn't setup right |
01:07.25 | |Vulture| | dsmouse:I thought he said there was nothing.. but he might not have been in debug |
01:09.27 | bjohnson | set verbose 3 should show it without the info overload of debug |
01:13.06 | SexyKen | Where is a good place to get sounds for asterisk? |
01:13.11 | SexyKen | "HOld up while we connect yo call" |
01:13.43 | SexyKen | Also can background play mp3s? |
01:14.19 | ManxPower | I think a telco problem has finally beaten me after all these years. |
01:14.27 | ManxPower | telco == telcom |
01:14.42 | Qwell | bkw_: So, you can say you come after Paris? |
01:14.56 | ManxPower | SexyKen, Did you try /var/lib/asterisk/sounds and the asterisk-sounds CVS? |
01:15.07 | SexyKen | I'll check it out. |
01:15.13 | SexyKen | Can Playback() play mp3's? |
01:15.31 | SexyKen | Sorry I mean Background |
01:15.31 | dontmsgme | It said rejected, no authority found |
01:15.48 | ManxPower | <PROTECTED> |
01:15.48 | ManxPower | <PROTECTED> |
01:15.48 | ManxPower | Feb 18 19:15:41 WARNING[2844]: chan_zap.c:4723 ss_thread: getdtmf on channel 9: Operation now in progress |
01:15.48 | ManxPower | <PROTECTED> |
01:16.04 | ManxPower | SexyKen, no |
01:16.21 | SexyKen | Is there a way to play an mp3's file out of no where? |
01:16.43 | ManxPower | SexyKen, "show applications" is your friend. |
01:16.45 | dsmouse | Mp3Player |
01:17.26 | dontmsgme | Feb 18 17:09:49 NOTICE[-235656272]: chan_iax2.c:5183 socket_read: Rejected connect attempt from 198.22.67.70 |
01:17.26 | dontmsgme | Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT |
01:17.26 | dontmsgme | <PROTECTED> |
01:17.26 | dontmsgme | <PROTECTED> |
01:17.36 | *** join/#asterisk ScythelX (Fleb@pc-24-181-176-153.sbi.ct.charter.com) |
01:18.26 | SexyKen | dsmouse, How would I answer the call and play an MP3 while they're being connected to the call? |
01:18.44 | ScythelX | hello all - there is a company near me thats going out of business that has a shit ton of cisco phones I wanted to take a few of their 7940's off their hands...I've never used the cisco series phones but I was wondering if someone knew if I would be able to use them...ie i dont know if you need a special license key to operate them |
01:18.55 | ManxPower | SexyKen, That is called MusicOnHold. Do I need to point you to "show applications" and the Wiki? |
01:19.14 | Qwell | ScythelX: How much they selling them for? heh |
01:19.26 | ScythelX | like 100 bucks with the power cubes |
01:19.28 | SexyKen | Manx, NO! I just dont know how to have it play while dialing! |
01:19.29 | Qwell | wow |
01:19.34 | ManxPower | ScythelX, Cisco will make you pay for SIP firmware. |
01:19.40 | Qwell | No, you don't need anything special, really |
01:19.48 | Qwell | SIP firmware isn't a requirement, afiak |
01:19.52 | ManxPower | SexyKen, "show application dial" Pay special attention to the "m" option. |
01:20.11 | ScythelX | i just want them for home use only gonna by 3, but I dont want to hook them up and not be able to use them |
01:20.12 | ManxPower | Qwell, You don't need a hemlet when racinh motorcycles either, but it's usually a good idea. |
01:20.30 | Qwell | ManxPower: I'm just commenting. You're right though |
01:20.54 | *** join/#asterisk IQ (~IQ@70-59-167-66.omah.qwest.net) |
01:21.11 | IQ | Hi, anyone using Avaya gateway with Asterisk? |
01:21.27 | ScythelX | ManxPower: so if I wanted to upgrade the firmware of the phone I would just have to pay for that? |
01:21.37 | ManxPower | ScythelX, Yes. Welcome to Cisco. |
01:21.39 | Qwell | its like $8 per phone or something, isn't it? |
01:21.58 | Qwell | at only $100 per phone, thats not bad at all |
01:21.58 | ScythelX | ManxPower: ok thank you for your input |
01:22.03 | ScythelX | Qwell: you as well |
01:22.04 | ManxPower | A support contract is $8/yr. I belive that gives you access to the firmware, but not the legal right to use it. |
01:22.08 | Qwell | hell, the phones may even HAVE the SIP images already |
01:22.29 | dontmsgme | Does vonage ring to your Xlite phone? |
01:22.34 | _Vile | I saw on the forums today that CDW can hook you up with a contract |
01:23.12 | _Vile | lists, rather |
01:23.13 | Qwell | _Vile: I heard that people had a difficult time getting them to give you one |
01:23.34 | _Vile | CDW? I've heard more of a hard time from people getting them from independant vendors |
01:24.04 | Qwell | dontmsgme: Vonage has a thing for like $13 a month, that lets you use a softphone. |
01:24.08 | Qwell | something like that |
01:24.19 | *** join/#asterisk mindCrime (~mindCrime@rrcs-24-106-188-6.se.biz.rr.com) |
01:25.01 | ManxPower | I'm getting "chan_zap.c:4723 ss_thread: getdtmf on channel 10: Operation now in progress" on incoming calls. Anyone have suggestions on how to fix that? |
01:25.02 | Qwell | ManxPower: So what gives you the legal right to use the firmware? |
01:25.07 | _Vile | bkw u wanna tweak the code to have newest messages play first in voicemail rather than oldest? |
01:25.13 | ManxPower | Qwell, US$130 I think. |
01:25.19 | Qwell | one time, or yearly? |
01:27.34 | ManxPower | Qwell, one time. |
01:27.40 | ManxPower | You are buying the software. |
01:27.49 | Qwell | ahh |
01:27.53 | ManxPower | OK, I posted a $100 bounty/bribe to get my problem fixed. |
01:27.56 | *** join/#asterisk jsolares (~jsolares@200.12.44.18) |
01:27.58 | Qwell | $130 each time you upgrade too, or? |
01:28.28 | ManxPower | Qwell, Yes, unless you have a $8/yr support contract. Gives you the right to .point upgrasdes. |
01:28.33 | ManxPower | Call Cisco and ask them |
01:28.45 | Qwell | I don't even have an IP phone, heh |
01:28.47 | Qwell | just curious |
01:31.16 | ariel_ | ManxPower, that is a strange statement you have thre about the getdtmf. |
01:31.28 | ariel_ | what were you trying to do? |
01:32.46 | ManxPower | ariel_, Trying to get incoming DID calls on T-1 E8M Wink channels |
01:32.49 | ariel_ | ManxPower, reason is I saw that once when were setup a pulse dial phone on a channel bank. |
01:33.14 | ManxPower | We confirmed that the telco is sending us DTMF. The think is that about %50 of the calls actually DO work. |
01:34.30 | Beirdo | ManxPower: go for both :) |
01:34.47 | ManxPower | Beirdo, I already know that is a bad combination |
01:35.07 | Beirdo | heh |
01:35.22 | ariel_ | ManxPower, have you tried the old style e&M wink from adtrans featd |
01:35.24 | Beirdo | I guess that would depend on intoxicant, but likely so |
01:36.05 | ManxPower | ariel_, Yes. Console said something like "got something that doesn't belong in FeatGroup D, assuming E&M Wink |
01:36.20 | ariel_ | ah I have seen that one. |
01:36.22 | ManxPower | A suprizingly usefull message, actually. |
01:36.58 | riksta | shit there's been another quake in indonesia |
01:36.59 | ManxPower | ariel_, Based on the symptoms I assume it's actually a wink or dtmf timeing problem |
01:37.00 | ariel_ | Is this going directly to the telco or an older pbx? |
01:37.04 | riksta | 6.9 on the richter |
01:37.15 | ariel_ | actually I think it's also a timing issue |
01:37.58 | ManxPower | ariel_, Telco -> Asterisk -> channel bank -> (analog) -> PBX. We are only having problems with the EM/W channels (there are a bunch of loopstart channels too that are non-DID and work just fine. |
01:38.00 | ariel_ | I had a similar problem with acepting dtmf on a Mitel which we had to upgrade there T1 to a PRI to get it working. |
01:38.14 | *** part/#asterisk jpablo (~jpablo@host-148-244-137-95.block.alestra.net.mx) |
01:39.23 | ManxPower | ariel_, I didn't really need to hear that. 8-) |
01:39.36 | ariel_ | ManxPower, you know me I don't lie. |
01:39.50 | ManxPower | The goal was to not make any changes on the T-1 or channel bank or the PBX for at least 30 days after getting it working. |
01:40.03 | ManxPower | Then we can look into doing interesting stuff. |
01:40.19 | ManxPower | The problem, of course, is that if I can't get it working by monday 6am we'll have to take asterisk out of the path. |
01:40.56 | ariel_ | ah I see. So the problem is mainly between the asterisk and the c/b or to the telco? |
01:41.25 | ManxPower | The problem is ONLY between the Telco and Asterisk |
01:41.40 | ariel_ | hummm which telco? |
01:42.09 | SexyKen | Anyway to convert wav to gsm? |
01:42.17 | ManxPower | I=55 Telecom, the ILEC is BellSouth. However, everything work fine if we take the T-1 and plug it directly into the channel bank |
01:42.46 | ariel_ | change the timing to the c/b as 1 and the t1 to 2. |
01:43.12 | ManxPower | ariel_, Huh? |
01:43.19 | ariel_ | B/S sometimes does not sync timing correctly. |
01:43.52 | ariel_ | your c/b is connected to the asterisk via a t1 correct? |
01:43.55 | ManxPower | ariel_, All the other 44 channels on the T-1 (all non-E&M) work fine. 8-) |
01:43.57 | ManxPower | Yes. |
01:44.06 | tzanger | ok |
01:44.16 | tzanger | how the fuck do you "accidentally" step into the path of a freight train? |
01:44.29 | Qwell | tzanger: I've seen it |
01:44.31 | ManxPower | span=1,1,0,esf,b8zs |
01:44.31 | ManxPower | span=2,2,0,esf,b8zs |
01:44.31 | ManxPower | span=3,0,0,esf,b8zs |
01:44.31 | ManxPower | span=4,0,0,esf,b8zs |
01:44.42 | tzanger | http://toronto.cbc.ca/regional/servlet/View?filename=to-train20050218 |
01:44.42 | ManxPower | span 1 and 2 are coming from the telco, span 3 and 4 are going to the channel bank |
01:44.43 | Beirdo | tzanger: it's called a walkman and headphones |
01:44.44 | tzanger | yeah so have they |
01:44.47 | `Sauron | Mmm. |
01:44.52 | `Sauron | <3 signaling |
01:45.27 | ManxPower | span 1, channels 9 - 12 are the E&M channels |
01:46.05 | ariel_ | change the timing on span 1 to 2 and span 2 to 1. |
01:47.10 | ManxPower | ariel_, done. restarting |
01:47.41 | ManxPower | ariel_, no difference |
01:47.57 | ManxPower | zttool shows syncing from span 2 |
01:48.10 | ManxPower | span2 is also coming from the telco |
01:48.19 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
01:48.36 | ariel_ | can you get to the configuration on the c/b |
01:48.51 | ariel_ | see what there set to for 9 - 12 |
01:48.58 | ManxPower | ariel_, no. |
01:49.48 | Chuji | ManxPower : my Bellsout t1 does wink with no problem |
01:49.59 | Chuji | ManxPower : I didn't have to do anything special |
01:49.59 | ManxPower | ariel_, Since the channel bank can no longer be managed via FDL by the telco..... |
01:50.18 | ManxPower | Chuji, wanna swap T-1s? 8-) |
01:50.36 | Chuji | Hah, not real bad |
01:50.54 | Chuji | but, fwiw, e&m wink is fine |
01:51.14 | Chuji | I saw your post earlier on -users and people were saying it couldn't be done |
01:51.39 | ariel_ | ManxPower, what c/b are they? |
01:51.49 | ManxPower | ariel_, Adtran I believe. |
01:52.05 | ariel_ | I guess your doing this remotely? |
01:52.24 | ManxPower | ariel_, Yes, but in theory I can be on-site at some point. |
01:52.49 | ManxPower | I was onsite last week, but we had....communications...issues with the people that manage the PBX. |
01:53.19 | ManxPower | My mind says "850" but I don't know if that's correct. |
01:54.02 | ManxPower | I would rather just have our CLEC convert all the channels we deal with to PRI 8-) |
01:54.02 | ariel_ | 750/850 to work correctly with asterisk newer builds needs firmware 36 or above. |
01:54.19 | ManxPower | But we don't have any available channels for a D-channel. |
01:54.37 | ManxPower | ariel_, yes, but the problem is telco -> Asterisk, not Asterisk -> channel bank. |
01:54.44 | riksta | anyone here using ADM? |
01:54.57 | ManxPower | I can make DID calls via the channel bank all day without problems as long as the call comes in via VoIP. |
01:55.31 | ariel_ | next put one line directly to the c/b with the problem and rest on the asterisk... |
01:55.59 | ManxPower | ariel_, That's what we did to make things work last week. |
01:56.07 | ManxPower | It works fine without asterisk there. |
01:56.14 | ariel_ | have you spoken with mark about this? |
01:56.19 | ManxPower | ariel_, nope. |
01:56.34 | ManxPower | I can't PROVE it's a bug and I know that I get bitchy when people bother me. |
01:58.16 | ManxPower | I can imagine mark gets a zillion requests a day for help. |
01:58.30 | Chuji | Mark would help you though man |
01:58.39 | Chuji | You give back to community more than most |
01:58.55 | ManxPower | ariel_, I just /msg'd him. He's away. |
01:58.57 | ariel_ | this is one that he just might help with. Or maybe get one of the other guys to talk to him. |
01:59.14 | ariel_ | He loves these type of problems. |
01:59.17 | riksta | out of interest, what is mark's nick? |
01:59.23 | ariel_ | kram |
01:59.25 | riksta | kool |
01:59.32 | riksta | obvious :) |
01:59.37 | ManxPower | I'm becoming rapidsly less than sober at the moment, however. |
01:59.39 | Chuji | ~kram |
01:59.40 | jbot | Looking for the elusive BishopChicken. |
02:00.03 | ariel_ | ManxPower, Martin P. is pretty good with T1's as well. |
02:00.05 | Chuji | ManxPower : Nobody around there has a T-Berd? |
02:00.26 | Chuji | Corydon too, if you can track him down |
02:00.57 | SexyKen | Is there anyway to change this: exten => _1XXXXXXXXXX,1,Dial(${CZ}/${EXTEN},60,H|g) so it automatically appends the one if it isn't entered |
02:01.26 | ManxPower | ariel_, exten => _XXXXXXXXXX,1,Goto(1${EXTEN},1) |
02:01.33 | ManxPower | ..er..that was for SexyKen |
02:04.05 | ariel_ | ManxPower, just a strange setup but have you tried the sf_w for inband signal for the ports your having the problem with? |
02:04.08 | ManxPower | ariel_, the problem MAY happen more often or less often depending on the digits they send us. |
02:04.18 | ManxPower | ariel_, I can try |
02:05.25 | ariel_ | looking at my notes I had to use that on a nortel with strange dtmf only 50% would work. |
02:05.52 | Chuji | We have to use feature group D with our Toshiba |
02:06.13 | Chuji | It sends ANI and DNIS as *xxxxxxxx*xxxx* |
02:06.25 | Chuji | but it still winks |
02:07.03 | *** join/#asterisk DHuang (~DHuang@203.49.132.48) |
02:07.19 | DHuang | hi |
02:07.22 | ManxPower | ariel_, Asterisk wants sf to be setup in /etc/zaptel.conf too and the config for that looks bizzare. |
02:08.23 | DHuang | I've put load => app_prepaid_auth_cid.so in the modules.conf but it's not showing up in the CLI > show modules ??? |
02:08.30 | ManxPower | ariel_, But the nortel accepts DID calls just fine. |
02:08.41 | ManxPower | ariel_, Thanks. |
02:08.55 | SexyKen | Thanks Manx. |
02:09.23 | DHuang | :-) |
02:09.42 | DHuang | any ideas? |
02:10.12 | ariel_ | DHuang, did you restart asterisk |
02:10.25 | DHuang | need to restart or reload? |
02:10.45 | ariel_ | if you did not load the model it will not work |
02:11.28 | ariel_ | you should be able to load it at the cli |
02:11.46 | DHuang | ariel: thanks.. checking now |
02:12.42 | dontmsgme | Is there any kind of expensive computer monitor |
02:12.49 | dontmsgme | on a laptop |
02:12.55 | dontmsgme | Where you wont get sick of looking at it |
02:13.07 | ariel_ | rofl |
02:13.13 | DHuang | plasma? |
02:14.09 | modulus_ | jbot g-g-g-g? |
02:14.10 | jbot | G-UNIT! |
02:14.11 | dontmsgme | Is there a plasma computer monitor? |
02:14.14 | modulus_ | jbot g-unit? |
02:14.15 | jbot | g-unit stands for "Guerilla Unit". It's members are Tony Yayo, Lloyd Banks, Young Buck, and the leader 50 Cent. Their official DJ is DJ Whoo Kid. Also see http://www.g-unitsoldier.com/ |
02:14.17 | DHuang | ariel: Thanks.. silly me, not restarting.... |
02:15.50 | ariel_ | DHuang, np |
02:15.59 | *** join/#asterisk jetx (~jetx@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
02:16.08 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
02:16.45 | fearnor | www.50shekel.com |
02:16.50 | fearnor | word up to j-unit |
02:17.48 | modulus_ | jew-unit |
02:17.50 | modulus_ | lol |
02:17.55 | fearnor | yes jew-unit! |
02:18.09 | fearnor | and lil' jap |
02:18.54 | riksta | jbot: g-g-g-g |
02:18.55 | jbot | G-UNIT! |
02:19.12 | riksta | :) |
02:21.14 | SexyKen | exten => _XXXXXXXXXXX,1,Dial(${CZ}/1${EXTEN},60,H|g) |
02:21.19 | brc_ | yo yo yo! |
02:21.21 | SexyKen | this doesn't automatically append a one does it? |
02:21.46 | Qwell | SexyKen: that works for me here |
02:22.05 | Qwell | I send my calls through the pstn like that, prepending a 9 |
02:22.15 | SexyKen | hm |
02:22.30 | SexyKen | So that should work if I dial just 6507841022? |
02:22.54 | jsolares | it dails ${CZ}/16507841022 |
02:23.00 | Qwell | should |
02:23.19 | *** join/#asterisk Brixius (Brixius@c-24-118-4-197.mn.client2.attbi.com) |
02:23.39 | Brixius | Hello |
02:24.41 | SexyKen | Is there anyway to make it so that a #asterisk moves to the next step |
02:26.15 | ariel_ | ManxPower, I don't have access to the old server. I just emailed there IS person to get me the settings. I got him the job there so he should send the info to me. |
02:26.33 | IQ | Hi, any windows SIP soft phone beside x-lite? |
02:26.36 | *** join/#asterisk _Raptor_ (~RaptorBlu@pD9E5AAAB.dip.t-dialin.net) |
02:26.38 | _Raptor_ | hi |
02:27.02 | ManxPower | ariel_, Thanks. |
02:27.15 | ariel_ | IQ, yes but why |
02:27.29 | _Raptor_ | another question: i only get as last message: |
02:27.37 | _Raptor_ | <PROTECTED> |
02:27.37 | _Raptor_ | <PROTECTED> |
02:27.37 | _Raptor_ | [1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323 v1.12.2, PWlib v1.5.2 |
02:27.37 | _Raptor_ | [1]WrapGatekeeperServer::WrapGatekeeperServer: Creating new gatekeeper. |
02:27.37 | _Raptor_ | PObject |
02:27.43 | IQ | ariel_: just looking for something with small footprint |
02:27.58 | _Raptor_ | what have i done wrong now? |
02:28.01 | ariel_ | small if you use asterisk get diax |
02:28.27 | IQ | ariel_: I am using asterisk but I need SIP based softphone |
02:28.50 | IQ | ariel_: which one do you know? |
02:28.56 | DHuang | IQ: try firefly... |
02:29.03 | SexyKen | Shoulnd't this play music: exten => _XXXXXXXXXX,2,Dial(${CZ}/1${EXTEN},60,H|g|m) |
02:29.09 | IQ | DHuang: thanks |
02:29.19 | DHuang | IQ: not a prob.. it also support G729a |
02:30.33 | IQ | DHuang: just looking for a link to download it - is this the one: http://www.virbiage.com/firefly/download/index.php |
02:30.53 | DHuang | IQ: yeap..download the 3rd party one |
02:31.03 | DHuang | IQ: http://www.virbiage.com/firefly/download/firefly-thirdparty.exe |
02:31.08 | ariel_ | IQ, firefly will work. But you asked for small footprint. |
02:31.38 | DHuang | :-) |
02:31.48 | IQ | ariel_: Yes, thats right. I dont know anything about firefly yet. |
02:32.02 | IQ | ariel_: you said you know something - could you tell me what is that? |
02:32.44 | ariel_ | I use xlite due to it works with most setups. And it's fairly easy to use it via a stun server. |
02:33.09 | IQ | ariel_: Yes, I know xlite. but your answer to my question was "yes but why" . :) |
02:33.29 | DHuang | xlite doesn't support G729 right? |
02:33.37 | dontmsgme | I want a laptop holster so i can walk and type, any URLs? |
02:33.41 | ariel_ | why is all the others I have tried have many issues. |
02:33.45 | ManxPower | SexyKen, No. You don't put | between options .... ,66,Hgm) |
02:33.52 | ariel_ | DHuang, your correct. |
02:34.21 | ariel_ | DHuang, it supports gsm |
02:34.24 | ManxPower | no free softphone can include G729 |
02:34.45 | Luke-Jr | ManxPower: even if they're out of the US? |
02:34.56 | DHuang | ManxPower: firefly can, if you get the g729.dll |
02:34.58 | ariel_ | Luke-Jr, hehehe |
02:35.05 | *** part/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
02:35.08 | ManxPower | Luke-Jr, That is up for debate, but since no free softphone currently includes G729...... |
02:35.16 | ManxPower | DHuang, Yes, but that's not "included" |
02:35.21 | Luke-Jr | ManxPower: I think MPlayer has the code for it :) |
02:35.25 | ariel_ | actually there is another sip software that was ok from a co. called SIPP in EU |
02:35.51 | DHuang | ManxPower: you are right, their www site gives you the LIBS + C code to compile it.. |
02:37.35 | Luke-Jr | ariel_: on principle |
02:37.44 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
02:38.02 | Luke-Jr | ariel_: also, if I pay $10 to Digium, do I get the source? |
02:38.02 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
02:38.24 | ariel_ | Luke-Jr, it's not gpl |
02:38.27 | DHuang | Luke-Jr: I wish.... |
02:38.47 | Luke-Jr | ariel_: that's why |
02:38.56 | ariel_ | Luke-Jr, there is a free one that works for 6 months from intel some where. But don't count on it too much for real production. |
02:38.58 | DHuang | Juke-Jr: you might be able to find source code for it.... as H232 is developed based on G729 |
02:39.02 | Luke-Jr | If I were to pay for any software, it would need to be open source |
02:39.14 | ManxPower | Luke-Jr, I heard a rumor that the patent holders refused to sell Digium a license unless the put copy protection in. |
02:39.36 | ManxPower | Hell, I have the source for the ITU G729 codec. It's not that hard to get. |
02:39.40 | Luke-Jr | ManxPower: Then I will gladly go with a non-US-regulated codec |
02:40.00 | mishehu | speex up, I can't here you |
02:41.44 | brc_ | here? |
02:44.54 | mishehu | hear |
02:44.56 | mishehu | whatever |
02:45.10 | mishehu | Mr. Bell, you are ringing... |
02:48.31 | DHuang | anyone know why the asterisk is not detecting Budget Tone-100 key press? |
02:48.59 | fearnor | budget |
02:49.16 | DHuang | :-p |
02:49.54 | DHuang | is it BT-100 setting or is it asterisk? |
02:50.24 | ariel_ | DHuang, I think that BT use info and not inband for there dtmf |
02:51.04 | DHuang | ariel: I see....I'll check the BT |
02:52.06 | DHuang | ariel: Send DTMF: in-audio via RTP (RFC2833) via SIP INFO , which to pick SIP? |
02:52.24 | *** join/#asterisk gabb0 (~gabb0@CPE0006258dff02-CM000a73661510.cpe.net.cable.rogers.com) |
02:53.26 | gabb0 | hello all |
02:53.49 | ariel_ | DHuang, your sip.conf for the bt need to have dtmfmode=info. But last time I used one was over 1 year ago. |
02:54.15 | DHuang | Ariel: I see.. :-) .. THanks |
02:55.04 | gabb0 | quick question, I recorded a call using Monitor and recorded it in pcm format. How do I convert that to a wav or mp3? |
02:55.17 | Strom_C | gabb0: lame? |
02:55.18 | *** join/#asterisk agave (phanop@216.81.43.75) |
02:55.31 | bjohnson | for what? playing on your mp3 player? |
02:55.31 | Strom_C | also, pcm == wav |
02:55.32 | gabb0 | won't sox do it |
02:55.51 | gabb0 | I know that but try playing it with itunes |
02:55.53 | agave | i want to have asterisk send a call to a new context if an incoming call doesn't have callerid, similar to privacy manager, how do I do that? |
02:55.57 | gabb0 | doesn't sound pretty |
02:56.12 | bjohnson | agave: gotoif() |
02:56.31 | agave | I can test on ${CALLERIDNUM} |
02:56.39 | agave | does it come in as null if it is blocked or not available? |
02:56.40 | bjohnson | exactly |
02:56.44 | bjohnson | null |
02:57.03 | gabb0 | Strom_C, what's the syntax for lame to convert |
02:57.04 | agave | okay that's what I thought... does asterisk have a NULL or should I just do if ${CALLERID} = "" |
02:57.05 | bjohnson | just do if = "" |
02:57.18 | agave | cool |
02:57.19 | agave | thanks |
02:57.25 | DHuang | ariel: I've set the BT to DTMF visa SIP..and works now... :-) |
02:57.27 | Strom_C | gabb0: I don't know. read the manpage. |
02:57.30 | bjohnson | there's an example on the wiki .. |
02:57.44 | bjohnson | that prompts a person to enter their callerid if it comes in as null |
02:58.14 | bjohnson | and prevents them from coming in otherwise .. not too customer friendly but might work in some cases |
02:58.19 | brc_ | ~seen atacomm |
02:58.24 | jbot | atacomm <~dan@69.54.45.98> was last seen on IRC in channel #asterisk, 16d 1h 1m 31s ago, saying: 'anyone want a IP 3000 conference phone? looking to replace ours with a IP 4000 model. Barely been used, in great condition.... looking for around $500'. |
02:58.52 | mishehu | the ip3000 doesn't have sip firmware though (last time I checked) |
02:59.08 | agave | the wiki si down :( |
02:59.31 | agave | damn, I tried to trap on $CALLERIDNUM but since it's a SIP did it is coming in as the SIP URL so it doesn't see that it is blocked |
03:00.03 | bjohnson | don't know how to help you on that one |
03:00.32 | bjohnson | all my dids are iax so the format is easier I guess .. I just check on callerid |
03:00.32 | agave | well, caller id name is set to asterisk... |
03:00.35 | bjohnson | like normal |
03:00.40 | agave | i guess i could trap on that |
03:06.40 | SexyKen | Hey guys, so I have a 'call through' system setup...so I can call into asterisk and dial a number and it calls out... |
03:06.55 | SexyKen | ....is there anyway to allow me to play music on the line if a button is pressed so both parties can hear? |
03:07.08 | IQ | ???? |
03:07.42 | *** join/#asterisk brookshire (~nobody@pcp01541028pcs.huntsv01.al.comcast.net) |
03:08.46 | SexyKen | ??????? |
03:08.53 | SexyKen | WHat do you need exlplained more |
03:10.56 | bjohnson | write a program to do it I guess |
03:11.13 | bjohnson | maybe some kind of 3 way call conferncing thing |
03:11.16 | bjohnson | or using meetme? |
03:11.44 | bjohnson | * doesn't monitor for much after a call has been successfully answered |
03:12.04 | bjohnson | and what it does monitor will usually take you out of the call |
03:19.44 | Moc | hi everyone |
03:21.43 | tzanger | evenign moc |
03:23.27 | _Raptor_ | anyone here who can explain this: http://rafb.net/paste/results/M5Alwm11.html ??? |
03:24.02 | brookshire | bah.. h.323 |
03:29.55 | doughecka | is voip-wiki down? |
03:30.30 | brookshire | looks like it |
03:31.04 | doughecka | crap |
03:31.06 | doughecka | :P |
03:31.31 | FaithX | voip-wiki goes out a fair bit |
03:32.16 | doughecka | they need a better host |
03:32.25 | doughecka | whats the backend? |
03:32.49 | brookshire | digium just needs to document stuff |
03:32.50 | doughecka | would it be simple to mirror the site? |
03:32.50 | brookshire | haha |
03:33.11 | doughecka | well I need info for cisco phones |
03:33.18 | doughecka | like how to get the external directory |
03:33.23 | doughecka | thing to work with asterisk |
03:33.23 | brookshire | oh |
03:34.01 | brookshire | like is said.. digium needs to document stuff :) |
03:34.17 | doughecka | lol |
03:34.37 | mtqh | doughecka: digium does not need to document stuff.....they charge for that |
03:35.08 | brookshire | heh.. are you complaining? |
03:35.20 | doughecka | document? |
03:35.28 | doughecka | they have AWESOME documentation |
03:35.29 | doughecka | :) |
03:41.26 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
04:04.27 | ScythelX | doughecka: the external directory is an xml file located on the websever that the phone accesses |
04:10.50 | shido6 | boink |
04:10.59 | shido6 | anyone seen Constantine? |
04:11.36 | *** join/#asterisk greendisease (~jack@greendisease.fedora) |
04:12.09 | *** join/#asterisk coppice (~chatzilla@245.195.17.210.dyn.pacific.net.hk) |
04:12.59 | *** join/#asterisk roamer323 (~sing@HSE-Toronto-ppp130885.sympatico.ca) |
04:13.35 | Beirdo | Hmmm |
04:13.46 | *** join/#asterisk TheEmperor (TheEmperor@218.111.51.3) |
04:13.48 | Beirdo | having a hard time getting Festival to behave |
04:14.58 | *** part/#asterisk SuperMMan (~graphic@d209-89-191-155.abhsia.telus.net) |
04:16.43 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
04:17.29 | doughecka | can I do something with mpg123 to make the MOH play louder? |
04:17.37 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
04:20.17 | shido6 | festival is fun |
04:20.30 | shido6 | heh, sorta |
04:20.44 | Beirdo | it doesn't seem to do squat |
04:22.00 | coppice | of course it doesn't do squat. only real peopel do that :-) |
04:22.03 | Qwell | Beirdo: the syntax is really weird |
04:22.20 | Qwell | festival (SayText "test") |
04:22.33 | coppice | its lisp |
04:22.35 | Qwell | probably have to quote it, and escape the quotes |
04:22.40 | shido6 | rerecord the mp3 louder |
04:23.18 | doughecka | shido6: btw the cisco phone works great :) |
04:24.29 | Beirdo | Qwell: is that a line for in extensions.conf? I'm looking for some online examples that use it |
04:24.52 | Qwell | Beirdo: Not sure how to use it in asterisk, I just remember screwing with it for a bit from CLI |
04:25.41 | shido6 | doughecka uhhh |
04:25.45 | shido6 | did we talk on the phone? |
04:25.49 | doughecka | nah |
04:25.52 | shido6 | k |
04:25.55 | shido6 | dont drop ur cisco |
04:25.58 | doughecka | you pointed me to voip-supply for the power supply |
04:26.00 | shido6 | u CAN drop the budgetone tho |
04:26.03 | shido6 | oh |
04:26.05 | shido6 | heh |
04:26.05 | *** join/#asterisk muesli (~muesli@mail.muehlhaeuser.de) |
04:26.06 | doughecka | hah |
04:26.07 | shido6 | did u tell them i sent ya |
04:26.08 | shido6 | ? |
04:26.24 | doughecka | actully I forgot, I ordered it when I woke up and wasnt exactly awake then |
04:26.26 | shido6 | they get criscos , I mean cisco's at stupid low prices |
04:26.36 | shido6 | in blocks |
04:26.44 | doughecka | I could email them and say greg sent me |
04:26.47 | doughecka | and make them all confused |
04:26.48 | doughecka | :P |
04:26.53 | shido6 | big phatass toomuchcashtothrowaround blocks |
04:27.01 | doughecka | hahaha |
04:36.55 | TheEmperor | how i call someone if their sip number is 12345@abc.com ? |
04:37.05 | TheEmperor | i mean, how do I call :) |
04:39.52 | loud | http://slacker.com/~nugget/asterisk7.php |
04:40.43 | TheEmperor | loud: thanks :) |
04:45.14 | ScythelX | finally got ser working correctly |
04:45.46 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
04:46.13 | neopher | anyone here have Firmware image for a cisco 30 vip? |
04:46.32 | *** join/#asterisk yashax (~yasha_x@c-24-98-23-73.atl.client2.attbi.com) |
04:46.46 | TheEmperor | loud: so i have to configure my exten file? i thought there is a softphone you can just put the address into |
04:48.06 | loud | no, its not that simple, unless you have a 7960. |
04:51.30 | wankel | yeah, well it's not that easy to type that on a 12-key pad either :P |
04:53.45 | Beirdo | OK, almost got it :) |
04:55.16 | Beirdo | Feb 18 23:54:32 NOTICE[12811]: rtp.c:430 ast_rtp_read: RTP: Received packet with bad UDP checksum |
04:55.20 | Beirdo | grrr |
05:00.56 | hmodes | wheee |
05:03.47 | *** join/#asterisk Defraz (~t0tal@65.103.222.4) |
05:10.07 | stepcutHM | Beirdo: Feb 18 18:11:23 NOTICE[65130]: rtp.c:452 ast_rtp_read: RTP: Received packet with bad UDP checksum |
05:10.10 | stepcutHM | :p |
05:11.04 | *** join/#asterisk eipi (~eipi@40-142-89-200.fibertel.com.ar) |
05:12.26 | *** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4116580.sympatico.ca) |
05:12.47 | NormAst | Hi all. |
05:13.05 | Beirdo | stepcutHM: got any ideas how to fix that? :) |
05:18.01 | *** join/#asterisk uberwolf (~djdjs@c-67-165-175-191.client.comcast.net) |
05:18.05 | uberwolf | hello all |
05:18.12 | *** join/#asterisk guugmember (~nramos@200.6.221.64) |
05:18.22 | NormAst | Hi. |
05:18.25 | uberwolf | trying to build my first * |
05:18.30 | NormAst | GREAT! |
05:18.38 | guugmember | is there a way my teleco can close my IAX comunication from my home to my Asterisk in my office? |
05:18.57 | uberwolf | got all the way through with at conf and I am looking at the console when you try to make a SIP to SIP call |
05:19.06 | uberwolf | and I get this on the screen |
05:19.19 | uberwolf | Feb 18 10:11:53 NOTICE[5454]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' |
05:19.19 | uberwolf | <PROTECTED> |
05:19.28 | JerJer | the type=peer is invalid |
05:19.49 | uberwolf | ah |
05:19.52 | uberwolf | ok |
05:20.04 | yashax | Guys, how can I force the ifconfig command to set the IP address to be a permanent change (static). Everytime when I issue the command, it get's the dynamic address from DHCP server after a while...? |
05:20.12 | uberwolf | I was using the sample config in mahler's book |
05:20.16 | *** join/#asterisk pawnbroker (~rstevensj@ca-santaanahead-cuda1-c5a-45.anhmca.adelphia.net) |
05:20.21 | ManxPower | yashax, Step 1: /join #linux |
05:20.51 | guugmember | who is going to go to Miami to the TMC telephony show next tuesday? |
05:20.51 | yashax | I have, have been there for 15 min with no help at all.. so thought to try it here.. I am here a lot, so.... |
05:21.02 | guugmember | jsolares, que onda chema |
05:24.13 | pawnbroker | is the gang up for a newby question? |
05:24.47 | sivana | yashax: try ##linux |
05:25.01 | sivana | #linux is read-only :P |
05:25.31 | ManxPower | Then try #linux-help |
05:25.37 | ManxPower | or #linux-newbiw |
05:25.41 | ManxPower | s/w/e |
05:25.54 | yashax | thank you.... |
05:27.32 | pawnbroker | can * simlt ring multiple remote extensions and a cell phone? |
05:27.40 | neopher | looking for cisco 30 vip Firmware image, anyone gots? |
05:27.52 | neopher | pawn: yes |
05:28.13 | ManxPower | neopher, There's no SIP firmware for that phone. |
05:28.37 | neopher | manx: not looking for sip, looking for the latest bootloader |
05:29.02 | Beirdo | I lost my phone line I was going to use to hook up the FXO |
05:29.08 | Beirdo | way to go, dumbass |
05:29.32 | pawnbroker | cool... I assume fist answer grabs the call. how well does it work with the ixay? |
05:29.51 | neopher | manx: when i hat *** the phone doesn't show firmware version, hence it is real old and i can not use it with chan_skinny |
05:29.56 | Inv_arp | hey who offers cheap local iax did's, not 800 besides VP connect |
05:30.01 | guugmember | have anybody experience that the fxo card is making calls every 15 minutes? |
05:30.33 | neopher | guug: had that one, had to restart the * box |
05:30.54 | guugmember | neopher, we have restarted, but also makes the call |
05:31.21 | neopher | is it just hooking or is it actually dialing |
05:31.32 | guugmember | just hooking |
05:31.52 | neopher | hmmm, line voltage variation |
05:31.55 | neopher | maybe |
05:32.02 | guugmember | neopher, how can I change that? |
05:32.03 | neopher | had that happen too |
05:32.18 | guugmember | change the pci slot? |
05:32.18 | neopher | you can't |
05:32.23 | *** join/#asterisk murangd (~nukaidc@pool-162-84-144-211.ny5030.east.verizon.net) |
05:33.02 | neopher | if it is a line voltage variation, it would be a prob at the CO which the telco would have to fix |
05:33.23 | guugmember | neopher, what is CO? |
05:33.41 | sivana | central office |
05:34.00 | neopher | if the voltage fluxtuates (can't spell) then the modem/asterick thinks a call is coming in |
05:34.21 | neopher | could also be a modem the is starting to go, seen that b4 |
05:34.32 | neopher | thanks siv |
05:35.45 | neopher | also , check for line noise, sometimes if the line sounds staticky it may mean a wireing prob somewhere along the line that could cause the prob |
05:37.00 | neopher | how many fxo card do you have, and what ones are you using |
05:37.58 | guugmember | neopher, I have a TDM04B, 4fxo, and just using 1 |
05:38.25 | Beirdo | I got one, a X100P clone, not hooked up as I can't find the f'ing phone cord I have for it |
05:38.55 | neopher | hmmm, you shouldn't have a prob with that card at all, digium makes good shizit. |
05:39.44 | neopher | i use 2 x100p clones, they work great, then again, there just an intel modem |
05:40.37 | Beirdo | if Digium still made the X100P, I might even feel guilty |
05:40.45 | *** join/#asterisk TheEmperor (TheEmperor@218.111.51.3) |
05:40.52 | TheEmperor | loud: where can i get a cheap one of those? :) |
05:41.13 | neopher | they stopped making them because the chipset (made by intel) was EOL |
05:41.17 | Beirdo | hmm? |
05:41.19 | guugmember | neopher, so it looks like a voltage fluctuation problem |
05:41.39 | neopher | guug: quite possibly |
05:41.50 | guugmember | neopher, and is exactlly every 15 mins |
05:41.54 | Beirdo | yeah, well, maybe some thought should be put into making more current cheap single-line cards :) |
05:42.05 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
05:42.08 | neopher | hmmm, thats a little weird |
05:42.27 | neopher | <PROTECTED> |
05:42.28 | Beirdo | if I had the time and the money, I'd be happy to make a DSP card that would do the trick. |
05:43.02 | mishehu | and what of the tigerjet chipset used on the tdm400's ? |
05:43.23 | Beirdo | isn't that the chipset used on the clones too? |
05:43.33 | neopher | beirdo: no need for single line cards, just get broadvoice, they are cheap and support * sip connections |
05:43.47 | Beirdo | neopher: yes there is a need |
05:43.56 | neopher | yes that chipset is used on the clones to |
05:44.03 | Beirdo | many of us have one or two phone lines that we want to hook up |
05:44.17 | mishehu | if a deal goes thru, I'll be using a t110p with a pri |
05:44.22 | guugmember | neopher, ok thnx a lot, will see the config files |
05:44.23 | neopher | there are actually 2 diff. chipsets that act exatly alike, one with more option then the other |
05:44.30 | Beirdo | mishehu: you are a bastard :) |
05:44.41 | mishehu | Beirdo: 2 phones, you might as well use a tdm02b |
05:44.48 | mishehu | save some interrupts |
05:45.00 | Beirdo | mishehu: for about $250? no thanks |
05:45.10 | mishehu | Beirdo: hey, I need to make sure the costs of the line are covered before the deal goes thru... |
05:45.38 | neopher | guug: if you catch BTW, ask him, he may have more insight with that card |
05:46.02 | mishehu | btw? or bkw? |
05:46.33 | Beirdo | mishehu: cool. you are making me jealous |
05:47.01 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
05:47.11 | _Vile | bkw u wanna tweak the code to have newest messages play first in voicemail rather than oldest? |
05:47.16 | neopher | how about this, ethernet ata fxo's , no irq usage, use sip, less driver headack |
05:47.29 | _Vile | asdf |
05:47.32 | neopher | mishehu: thanks, thats what i meant |
05:47.55 | *** join/#asterisk yasha (~yasha_x@c-24-98-23-73.atl.client2.attbi.com) |
05:48.01 | Beirdo | neopher: that's a possibility, but they aren't as easy to find, and I don't have IRQ issues |
05:48.04 | yasha | Guys, what/where is the best way to terminate * via IP to get unlimited local/long if possible? |
05:48.38 | ManxPower | yasha, pretty much none. Providers that give unlimited require you to use their box |
05:48.43 | neopher | i got four letters EBAY, there all over the place, at least when i was looking last week |
05:48.51 | guugmember | neopher, ok i will ask BTW |
05:49.08 | neopher | yasha: broadvoice |
05:49.18 | yasha | k... where would you recommend the best/cheapest place would be? |
05:49.21 | neopher | yasha: if you want to connect via sip |
05:49.44 | neopher | yasha: broadvoice www.broadvoice.com |
05:49.59 | neopher | yasha: i use them and love them |
05:50.08 | yasha | I can terminate * withit? How much is it? |
05:50.22 | Beirdo | neopher: fxo sip devices? I haven't seen terribly many of them |
05:50.45 | neopher | yasha: yes you can connect * via sip to them |
05:50.50 | Inv_arp | yasha: 9.95 activation plus 9.95 a month |
05:50.59 | guugmember | is there a way my teleco can close my IAX comunication from my home to my Asterisk in my office? |
05:51.05 | neopher | anywhere from $5 to $25 / month |
05:51.33 | Inv_arp | yasha: look at iax.cc, livevoip.com, voicepulse.com other good alternatives |
05:52.09 | yasha | neopher: unlimited? |
05:53.05 | Inv_arp | guugmember: you mean your isp stopping the packet transmission or somethin? |
05:53.36 | neopher | yasha: yes, unlimited us,ca and 21 contries, $20/month |
05:53.48 | neopher | yasha: yes, unlimited in state 9/mo |
05:53.59 | yasha | local DID? |
05:54.33 | Inv_arp | yasha: www.broadvoice.com explains all |
05:54.34 | neopher | if you isp is blocking VoIP packets, launch a complaint with the FCC as there are law that protect this |
05:54.36 | guugmember | Inv_arp, my ISP closing IAX port |
05:54.47 | Inv_arp | guugmember: use another port |
05:55.15 | guugmember | Inv_arp: like 80? |
05:55.27 | Inv_arp | guugmember: whatever your heart desires |
05:55.35 | guugmember | Inv_arp, great |
05:55.57 | guugmember | Inv_arp, so its almost impossible for your ISP to stop IAX packets |
05:56.34 | Inv_arp | guugmember: unless there running traffic analyzers... to see what type of data is floing thru ports |
05:56.35 | guugmember | Inv_arp, in my country you sing a contract saying that you are not allow to use the conection for VoIP |
05:56.40 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
05:57.15 | Inv_arp | guugmember: ahh sucks |
05:57.29 | guugmember | Inv_arp, totally |
05:57.52 | guugmember | Inv_arp, even thoug nobody respects that |
05:57.59 | Inv_arp | guugmember: heh |
05:58.23 | Inv_arp | i luv iax cause it just works |
05:58.54 | brookshire | sing a contract? |
05:58.54 | brookshire | that's awesome |
05:59.25 | guugmember | Inv_arp, yeah it works great in my country too |
05:59.31 | pawnbroker | is anyone here using the iaxy's? pro's con's? |
05:59.52 | JerJer | IAXy rules |
06:00.00 | guugmember | pawnbroker, i only have pros on the IAXy |
06:00.20 | guugmember | pawnbroker, a little con can be the price, close to US$100 |
06:00.31 | Inv_arp | i knew i sould have gotten one |
06:00.48 | pawnbroker | thanks |
06:01.27 | *** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4116580.sympatico.ca) |
06:03.32 | pawnbroker | forgive me i'm new to *.. :) do you need to subscribe to a service to use iaxy remotely? |
06:03.54 | *** join/#asterisk adker (~adker@70-97-138-2.dsl1.glv.ny.frontiernet.net) |
06:04.28 | guugmember | http://fp1.a2zinc.net/clients/fpvon/spring2005/flash/fp.aspx |
06:04.30 | guugmember | wow |
06:04.58 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
06:05.00 | guugmember | they should be using Apache at least |
06:05.10 | NormAst | Link doesn't work.. Microsoft. |
06:05.54 | guugmember | and its a von link |
06:06.02 | jetscreamer | Server Error in '/clients/fpVON/Spring2005/flash' Application. |
06:06.20 | neopher | linux = servers running microsoft = everyone running from the servers |
06:06.49 | NormAst | pawnbroker: http://www.digium.com/index.php?menu=iaxy |
06:06.49 | jetscreamer | vb |
06:07.02 | *** join/#asterisk Luhiwu (~marsosa@200.63.89.209) |
06:07.17 | guugmember | NormAst, i just love that pic |
06:07.18 | NormAst | vb for * |
06:07.47 | pawnbroker | Normast: thanks |
06:08.04 | guugmember | pawnbroker, better price i found http://voipstore.pulver.com/product_info.php?products_id=52 |
06:08.49 | NormAst | pawnbroker: so... IAXy is a little box that connects to your asterisk box. If you have two of them you can talk to your self.... You can use the echo test and talk to yourself... |
06:08.56 | neopher | anyone notice that in that pic ip phone entns should be ip phone extns |
06:10.15 | guugmember | neopher, entns is entrance? |
06:10.21 | neopher | NormAst: if i get three can i talk to the voices in my head? hehe |
06:10.39 | neopher | guug:ahhh |
06:11.05 | guugmember | neopher, what do you mean by ip phone ents should be ip phone extns |
06:11.09 | pawnbroker | guug: thanks |
06:11.09 | NormAst | hmm... I think so... I just hear voice... Never thought about talking to them. |
06:11.25 | *** join/#asterisk sudoer (~sudoer@65.75.148.190) |
06:11.53 | neopher | guug:i thought they meant to say ip phone extensions |
06:11.54 | sudoer | if I have an ethernet cable, can I convert it to rj11 for analog phones? |
06:12.11 | NormAst | You can.. Won't look nice. |
06:12.23 | guugmember | neopher, ahh ok |
06:12.41 | sudoer | I have cabling setup already and I dont want to rerun analog lines |
06:13.30 | neopher | sudoer: just use a pair in the center 2, no prob with that |
06:13.33 | guugmember | any one recommends me a good channel bank, and not that expensive |
06:14.36 | sudoer | neopher: is there a little converter plug I can buy or will I have to rewire the ends? |
06:15.36 | pawnbroker | smoking quells the voices... |
06:15.49 | neopher | sudoer: don't know of any converter, but you could essentually make one with an rj45 keystone and an rj-11 keystone |
06:16.41 | neopher | pawnbroker: haven't been to the movies since the doctor told me to stop entertaining the voices |
06:16.54 | mishehu | hmm... voip-info isn't responding to me. I tend to get users/peers mixed up... an iax peer can receive calls from me, and an iax user can make calls to me, right? |
06:17.36 | pawnbroker | neopher: I'm sitting here beside myself |
06:17.51 | sudoer | so neopher , you mean just cut the head off of rj45 cable, then replace it with rj-11 keystone? |
06:18.13 | neopher | you could do that |
06:18.38 | neopher | or crimp a rj11 end on |
06:19.00 | neopher | you ar only using 2 wire for analog telephone |
06:19.17 | Beirdo | mishehu: I believe that is correct |
06:19.33 | neopher | and it is the center 2 conductors that are used on the connectors |
06:19.40 | mishehu | Beirdo: my mind gets kind of polluted from all the crap I deal with day-to-day... |
06:19.44 | Beirdo | and friend can do both (as it's a combination) |
06:19.56 | mishehu | maybe I should install a garbage collector routine in my head ;-) |
06:20.04 | mishehu | yeah, friend I know can. |
06:20.21 | Beirdo | heh, vodka's my garbage collector tonight |
06:20.51 | neopher | no fair, you got vodka, i got relatives |
06:21.14 | Beirdo | hehe |
06:21.41 | neopher | they decided they were going to stop by and spend the night |
06:22.02 | Beirdo | so get fit-shaced |
06:22.45 | neopher | naaa, going to try to get up early go snowboarding, thats my outlet |
06:23.23 | pawnbroker | neopher: where do you board at? |
06:24.10 | neopher | in the poconos |
06:24.30 | neopher | i live like 15 min from the resort |
06:24.41 | pawnbroker | i'm on the west coast Mammoth every other week to decomp |
06:25.21 | neopher | i wish i was on the west coast, this ice shizit is really pizzing me off |
06:25.22 | pawnbroker | lucky you! |
06:26.16 | pawnbroker | its great here now with all this h20, a 4 hr drive for me |
06:26.33 | neopher | brb, crapper |
06:27.05 | pawnbroker | you have boilerplate to ski on and normally we get what we call sierra cement |
06:27.26 | *** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net) |
06:32.08 | pawnbroker | Thanks gang for all the * info, I'll be back |
06:32.12 | *** part/#asterisk pawnbroker (~rstevensj@ca-santaanahead-cuda1-c5a-45.anhmca.adelphia.net) |
06:36.27 | *** join/#asterisk DaLion (~Miranda@Quebec-HSE-ppp225437.qc.sympatico.ca) |
06:36.44 | DaLion | Hey all |
06:36.44 | DaLion | anyone tried teliax.com? |
06:37.23 | DaLion | and implicit u still drunk ? |
06:39.29 | *** join/#asterisk murangd (~nukaidc@pool-162-84-216-79.ny5030.east.verizon.net) |
06:40.03 | murangd | what's this IAX2/guest@66.250.68.194/996 |
06:40.22 | Inv_arp | DaLion: they seem good but they dont have local DID's in my area ... try also livevoip.com or iax.cc |
06:41.50 | *** join/#asterisk pranav (sameer@202-149-48-198.broadband.isp.exatt.net) |
06:45.26 | murangd | Inv_arp: livevoip.com charges like 1.2cents per minute for incoming calls to your did |
06:45.41 | murangd | is there like a provider of DID that doesn't chare any $$ for incoming calls to a did |
06:47.21 | *** part/#asterisk pranav (sameer@202-149-48-198.broadband.isp.exatt.net) |
06:47.24 | *** join/#asterisk pranav (sameer@202-149-48-198.broadband.isp.exatt.net) |
06:47.54 | Inv_arp | murangd: livevoip has an unlimited $6.99 plan iax.cc 10.49 unlimited .. think VP connect also |
06:48.11 | murangd | Inv_arp: not for commerical use |
06:48.33 | neopher | soft cap? |
06:48.51 | neopher | how can they call it unlimited if there is a soft cap |
06:49.01 | murangd | a message from livevoip.com |
06:49.02 | murangd | 1.1 cents per minutes everytime the number is called. We are forced to pay termination on DID's, everyone is. |
06:49.12 | murangd | guess they are not so unlimited |
06:49.28 | Inv_arp | murangd: http://connect.voicepulse.com/ or broadvoice or iax.cc |
06:49.32 | ManxPower | Talk to your locel CLEC. |
06:49.43 | murangd | ManxPower: CLEC? |
06:49.47 | ManxPower | We have totally free calling within Mississippi and Louisiana |
06:49.59 | neopher | i use broadvoice which is unlimited, but you connect via sip |
06:50.09 | ManxPower | CLEC = competitive Local Exchange Carrier. i.e. a company that competes with Bell. |
06:50.15 | ManxPower | We like our CLEC. |
06:50.36 | Inv_arp | neopher: how much is it? |
06:50.59 | Essobi | I know a few CLEC. :) |
06:51.14 | neopher | www.broadvoice.com |
06:51.42 | neopher | $20/mo for unlimited us canada and i beleive 21 other countries |
06:51.51 | murangd | does anyone the know the CLEC's for new york city |
06:51.57 | Inv_arp | bah i dont need unlimited outgoing, just unlim incoming |
06:52.07 | murangd | yeah me too |
06:52.12 | murangd | I just want unlimited incoming |
06:52.13 | ManxPower | unlimited incoming is pretty common |
06:52.14 | Inv_arp | i use voipjet for outgoing |
06:52.14 | neopher | then get the cheaper plan |
06:52.15 | murangd | forget about outgoing |
06:52.28 | murangd | Inv_arp: what you use for incoming |
06:52.50 | neopher | 5.95/mo unlimited incomming |
06:52.53 | murangd | I don't understand.. why do some companies provide unlimited INCOMING and others charge you for incoming |
06:52.57 | murangd | neopher: what provider? |
06:53.07 | neopher | broadvioce |
06:53.12 | Inv_arp | iax.cc for personal uses no business plan as of yet |
06:53.15 | murangd | connect.voiceplus.com offers unlimited uncoming for $8 per did |
06:53.23 | murangd | but $8 is to expensive |
06:53.40 | murangd | espically when services offer it for free |
06:54.01 | murangd | ManxPower: why do you suspects some providers charge you for incoming DID |
06:58.21 | Beirdo | hmmm. |
06:58.34 | Inv_arp | neopher: yea i might go with broad voice 5.95 plan .. u have good connection with them etc...? |
06:58.41 | Beirdo | is there a way to tell a particular extension NOT to use MOH? |
07:00.55 | ManxPower | murangd, Spend some time finding out pricing for all the IAX and SIP VoIP providers, you'll find a few that charge per min for DID. |
07:01.17 | murangd | ManxPower: yeah that's no good to charge for DID |
07:01.23 | murangd | they should give you free unlimited did's |
07:01.43 | ManxPower | I don't mind a monthly charge for a DID, just a per min charge. |
07:01.47 | murangd | I am looking for DID for NJ/NY |
07:01.59 | murangd | ManxPower: yeah me too, per min charges is what I was refering to |
07:02.09 | neopher | Inv_arp: haven't had one prob, i love them |
07:02.14 | murangd | ipkall.com = unlimited DID |
07:02.39 | Inv_arp | neopher: hmm that 5.95 plan is unlimited incoming? cant find where i says that |
07:02.40 | ManxPower | I just wish the cost of DIDs would come down. |
07:02.55 | ManxPower | We pay $20/month for 100 DIDs from our phone company. |
07:03.12 | *** join/#asterisk pfn (500@adsl-69-107-210-254.dsl.pltn13.pacbell.net) |
07:03.27 | pfn | , pfn\ |
07:03.28 | neopher | <PROTECTED> |
07:04.36 | Inv_arp | neopher: nice just hope its as easy to setup as iax since * is behind nat |
07:05.06 | *** join/#asterisk abombss (~abombss@c-67-175-115-51.client.comcast.net) |
07:06.29 | neopher | Inv_arp: broadvoice is not IAX, it is sip |
07:06.39 | ScythelX | broadvoice is cheap |
07:06.45 | neopher | Inv_arp: still very nice |
07:06.56 | murangd | ManxPower: $20 is not a bad price |
07:07.00 | murangd | ManxPower: you get free incoming? |
07:07.29 | Inv_arp | neopher: yea i know im hoping its as easy as my iax setup's since my * is behind a nat |
07:07.45 | neopher | Inv_arp: verry easy |
07:08.27 | Inv_arp | neopher: ahh k |
07:09.34 | ManxPower | murangd, $20/month for 100 DIDs, free unlimited calling withing Louisiana and Mississippi and free 256K internet (that we don't really use) |
07:09.42 | ManxPower | But this is a local phone company. |
07:11.01 | murangd | ManxPower: ah I see |
07:11.56 | ManxPower | VoIP is nice, it's useful, but compared to local PRI lines it's horribly complicated. |
07:14.05 | coppice | VoIP sucks, but its the future |
07:16.08 | ManxPower | We have a frame relay network with 0CIR so no VoIP over THAT. |
07:16.48 | JerJer | wholy fire that ISP |
07:16.54 | JerJer | batman |
07:26.46 | DaLion | trying to instal suse 9.2 |
07:26.55 | ScythelX | hey all looking for suggestions - not sure if I should just get a t1 line or use a datacenters interconnect for termination |
07:27.23 | DaLion | go for teliax.com |
07:27.25 | DaLion | i did |
07:28.36 | DaLion | tring to build a unix gateway |
07:28.36 | DaLion | between pci 802.11 and dsl |
07:28.48 | ScythelX | well im looking to reduce the amount of hops |
07:29.12 | ScythelX | either the t1 directly or host the box in the datacenter using their voice termination |
07:31.53 | neopher | what happened to voip-info.org |
07:32.26 | brookshire | died |
07:32.28 | brookshire | :) |
07:32.40 | Beirdo | fall down go boom |
07:34.37 | DaLion | ethernet everywhere nc100 v2.1 is what |
07:39.22 | neopher | i figured out that the best mtu for voip traffic with dsl using pppoe is 1492, just an addon for everyones notes incase you run into it in the future |
07:39.46 | jpayne | that may well depend on the dsl provider |
07:40.03 | Damin | neopher: That happens to be the best MTU for must PPPoE and PPPoA type services... |
07:40.15 | Beirdo | neopher: the best MTU for ANYTHING over PPOE is 1492 |
07:40.17 | neopher | yes |
07:40.29 | Damin | neopher: Leaves just enough room for the overhead of the PPPoE packets. :) |
07:40.30 | Beirdo | PPPoE rather |
07:40.48 | Qwell | http://ipv4.uuoc.com/?id=803 Is that fairly normal, for only one instance of * running? |
07:41.13 | neopher | but some people recommended 1428 which makes voip traffic scetchy |
07:41.31 | *** join/#asterisk pranav (sameer@202.149.48.198) |
07:43.00 | neopher | Qwell: going overboard with the v's ---> asterisk -vvvvvvvvvvvvc |
07:43.09 | DaLion | ok so a network everywhere nc100 .. i cant find in modules list of suse install ..whats is it like.. |
07:43.12 | neopher | i only use five, and thats plenty |
07:43.13 | Qwell | neopher: yeah, I hold it for a second or two :p |
07:43.31 | sivana | yes!! I found it |
07:43.44 | Qwell | But, is that normal? Having it show up so many times? |
07:43.45 | Damin | Qwell: Yeah.. that looks right.. |
07:43.51 | sivana | heh.. a bug in my app which would cause * to freeze up solid :P |
07:44.38 | neopher | qwell: I'm going to call you when i am ready for bed and you put me on hold so i can listen to you hold music |
07:44.43 | Damin | Qwell: If you only want to have one MPG123 process, then comment out all the other classes in /etc/asterisk/musiconhold.conf and add a -z to the end.. |
07:44.46 | Qwell | neopher: heh |
07:44.56 | Damin | Qwell: Ala default => quietmp3nb:/var/lib/asterisk/mohmp3,-z |
07:45.22 | Damin | Qwell: Then restart asterisk and kill all the mPG123 processes.. |
07:45.22 | Qwell | Damin: I was more worried about the multiple * processes |
07:45.30 | Beirdo | is there a way to tell one particular extension not to use MOH? |
07:45.32 | Qwell | I have 0 calls, and I've had 0 since I started it |
07:45.53 | Damin | Qwell: It's multithreaded.. |
07:45.56 | Qwell | ok |
07:45.59 | Damin | Qwell: That is good.. ;) |
07:46.03 | *** part/#asterisk brookshire (~nobody@pcp01541028pcs.huntsv01.al.comcast.net) |
07:47.23 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
07:48.09 | Beirdo | and lilo's back :) |
07:48.43 | Qwell | Damin: The only reason I ask, is I had some weird problems lately, I don't recall seeing that in the past |
07:49.22 | ScythelX | http://www.empirix.com/default.asp?action=article&ID=69 |
07:55.50 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
07:57.28 | ScythelX | can someone shed some light on this question for me - you have a company like voipjet.com that hosts their equip at a datacenter - where do they get their dids from? are they doing TDMoE? |
07:58.13 | JerJer | lol |
07:58.28 | JerJer | they buy them from some SIP provider |
07:58.51 | ScythelX | oh |
07:59.16 | Qwell | If you "buy" a bunch of DIDs, how long do you have them for? |
07:59.21 | Qwell | If you're a provider or something |
07:59.31 | JerJer | they are not a real telco or anything...they simply bought space in some IP colo facility |
07:59.31 | Qwell | Do you "own" them, or do you pay a yearly fee or something? |
07:59.39 | ScythelX | so there is still an extra network hop then |
08:00.15 | ScythelX | me --> voipjet.com --> their sip provider at another datacenter? |
08:00.39 | JerJer | yes |
08:01.35 | ScythelX | oh they make it sound deceiving |
08:01.47 | ScythelX | like they're the actual teleco |
08:01.52 | JerJer | hell no |
08:02.07 | ScythelX | so thats the same with nufone as well |
08:02.17 | JerJer | no |
08:02.24 | JerJer | absolutely not |
08:02.50 | JerJer | we have our own TDM network into an SS7 switch |
08:03.04 | JerJer | soon two switches |
08:03.31 | ManxPower | JerJer then get dids in a wide area of the country |
08:03.52 | JerJer | we have DIDs in many states, NuFone simply does not offer them |
08:03.54 | JerJer | yet |
08:04.23 | *** part/#asterisk DaLion (~Miranda@Quebec-HSE-ppp225437.qc.sympatico.ca) |
08:04.30 | Qwell | JerJer: Do you personally own NuFone, or are you just somebody higher up in the chain? |
08:04.40 | Qwell | "just"...not that thats bad |
08:04.43 | JerJer | i own NuFone |
08:04.57 | Qwell | s'what I thought, didn't want to sound like an idiot in the future though :p |
08:05.03 | ScythelX | I use nufone |
08:05.49 | ScythelX | so is your stuff onsite or do you have your equip in a datacenter |
08:06.28 | JerJer | data center |
08:06.30 | JerJer | most certainly |
08:06.48 | JerJer | our own in Southfield, then we are in Equinix Chicago |
08:07.09 | ScythelX | cool |
08:07.12 | JerJer | Southfield, MI |
08:07.14 | Qwell | out of curiousity, if you guys are in IL, why MI DIDs? |
08:07.18 | Qwell | nevermind, that answers that |
08:09.24 | JerJer | we have IL DIDs as well, just don't offer them |
08:09.34 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
08:10.30 | Qwell | any plans on a timeframe for offerings in other states? |
08:10.45 | JerJer | nope |
08:11.50 | Beirdo | www.nufone.net, right? |
08:14.52 | *** join/#asterisk Edgan (~edgan@okcforum.org) |
08:16.24 | *** join/#asterisk pranav (sameer@202-149-48-198.broadband.isp.exatt.net) |
08:16.39 | pranav | hello everyone |
08:17.04 | pranav | i am still facing a problem with the fwd stuff |
08:18.45 | pranav | it still says that "its an invalid extension" |
08:19.43 | pranav | i have made some changes in the sip.conf and i have pasted the sip.conf and the extensions.conf in the pastebin.ca/6112 |
08:20.17 | pranav | is there anyone else on the channel |
08:20.55 | *** join/#asterisk mbranca_home (~matteo@host-84-222-20-161.cust-adsl.tiscali.it) |
08:21.13 | pranav | hi mbracana |
08:23.07 | Qwell | pranav: I can't get to that domain right now for some reason. Mind repasting it at pastebin.com? |
08:23.33 | pranav | hello someone? there |
08:23.51 | pranav | ya sure |
08:24.02 | pranav | just a minute |
08:24.09 | *** join/#asterisk DaBigMac (~JJ@203-173-48-1.dyn.iinet.net.au) |
08:24.14 | DaBigMac | hello |
08:28.09 | pranav | ya i have pasted it in pastebin.ca/6113 |
08:30.57 | *** join/#asterisk inspired (mikael@host-81-191-119-90.bluecom.no) |
08:32.13 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
08:33.42 | *** join/#asterisk tecnico (~tecnico@user-24-236-123-31.knology.net) |
08:37.18 | *** join/#asterisk dawda (pranav@202-149-48-198.broadband.isp.exatt.net) |
08:37.30 | dawda | hello |
08:37.42 | dawda | qwell sorry i got disconnected |
08:38.01 | dawda | i am pranav-- registered with different name |
08:38.49 | dawda | did u check in pastebin.ca/6113 |
08:42.30 | DaBigMac | gents I have what I hope is a simple question answered a million times.............I have convered over to VoIP using an asterisk server.....what Id like to do is connect my previous landline to my asterisk box so I can receive calls on my old number via asterisk...that way I dont need 2 sets of phones in the house......whats the easiest cheapest card/way to do it |
08:42.38 | JerJer | Beirdo: yes that is our crappy website |
08:42.48 | Beirdo | Heh |
08:42.57 | JerJer | DaBigMac: TDM01B from Digium |
08:43.02 | Beirdo | OK, just wanted to make sure I was at the right site |
08:43.25 | DaBigMac | JerJer thanks for the info Ill go suss it out |
08:44.16 | Beirdo | Oooh, and you take PayPal. :) Excellent |
08:45.12 | herag | is the wiki down? |
08:49.22 | *** part/#asterisk dawda (pranav@202-149-48-198.broadband.isp.exatt.net) |
08:50.26 | ScythelX | herag: yeah |
08:54.55 | brc_ | JerJer! |
08:54.57 | brc_ | howdy |
08:59.14 | JerJer | mooo |
09:03.10 | mikegrb | ooooom |
09:03.43 | DaBigMac | JerJer : Just checked the Digium website hardware products section and TDM01B isnt listed |
09:04.05 | Qwell | DaBigMac: You have to go to the Yahoo store, and look at the tdm400p section |
09:04.12 | Qwell | Then you'll see the different configurations |
09:04.21 | DaBigMac | thanks qwell |
09:04.29 | *** part/#asterisk djin (~djin@gridfox.xs4all.nl) |
09:04.48 | Qwell | its probably 4-5 clicks from digium's site |
09:07.56 | JerJer | the TDM01B is a specific bundle of the TDM400P chassis and TDM Modules |
09:08.25 | Qwell | I would have said that, but was too lazy to read up a few lines to see if it was covered |
09:08.38 | Qwell | the 01 implies 0 FXS modules, 1 FXO |
09:16.17 | JerJer | i was trying to be nice |
09:16.36 | JerJer | some people seem to think I am always angry and/or hateful to people in the channel |
09:16.55 | DaBigMac | so the telco line to my house is designated FXO |
09:17.12 | *** join/#asterisk pranav (pranav@202-149-48-198.broadband.isp.exatt.net) |
09:17.36 | JerJer | you are at the station end of the line |
09:17.37 | pranav | hello everyone |
09:17.48 | JerJer | so you have to use FXS signalling, which means you need an FXO device |
09:17.54 | Qwell | JerJer: Who says that? |
09:17.56 | JerJer | since FXO uses FXS signalling |
09:18.03 | JerJer | and FXS uses FXO signalling |
09:18.15 | pranav | hi qwell , sorry i got disconnected |
09:18.40 | JerJer | wonderful circular dependency |
09:18.53 | pranav | i have pasted the extensions.conf and the sip.conf in the pastebin.ca/6113 |
09:19.15 | Qwell | pranav: And I said I couldn't get to pastebin.ca ...anyhow, bed time |
09:19.47 | pranav | thats ok |
09:21.04 | pranav | can anyone else check, i am able to call internally, mobile and to pstn, but i am not able to make fwd calls |
09:21.33 | JerJer | _. is EVIL |
09:21.33 | DaBigMac | ok so the tdm01b is a full height PCI card? do they come in half height? |
09:21.36 | JerJer | PURE EVIL |
09:21.45 | JerJer | DaBigMac: nope |
09:21.58 | DaBigMac | damn |
09:22.05 | DaBigMac | new PC time :-) |
09:22.25 | JerJer | and _7. is technically invlaid |
09:22.26 | JerJer | invalid |
09:22.32 | JerJer | use _7X. |
09:22.50 | JerJer | DaBigMac: yes you want to make sure it is newish |
09:22.56 | *** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au) |
09:23.03 | JerJer | they use the new PCI standard thingy |
09:23.28 | DaBigMac | 2.2 |
09:23.29 | pranav | i get an error saying that got sip response 404 "not found " back from 69.90.155.70 |
09:23.31 | DaBigMac | yup |
09:24.22 | pranav | and when i type http://69.90.155.70/ the fwd page opens |
09:24.30 | TheEmperor | JerJer: which softphone you recommend that has instant messenging? |
09:25.14 | jerlique | ~docs |
09:25.15 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
09:25.18 | JerJer | TheEmperor: none |
09:25.20 | JerJer | 7960 |
09:25.36 | TheEmperor | JerJer: 7960 has instant messenging? |
09:26.04 | JerJer | pranav: then whatever you are sending to that host is not finding anything that matches |
09:26.25 | JerJer | TheEmperor: not in the typical sense no |
09:26.39 | JerJer | but I have written an instant messaging service for those damn phones, yes |
09:26.58 | TheEmperor | JerJer: those phones are expensive :) |
09:27.19 | JerJer | yes, yes they are |
09:27.39 | TheEmperor | JerJer: any other recommendations aside from 7960 |
09:28.11 | JerJer | 7970 |
09:28.34 | TheEmperor | ok... |
09:30.05 | JerJer | That Hitatchi wifi phone looks promising |
09:30.49 | DaBigMac | thanks for the info guys, have a good evening/day/morning/night whatever :-) |
09:31.07 | TheEmperor | JerJer: what billing system do you use for NuFone? Mysql database? |
09:31.18 | JerJer | i wrote my own |
09:31.21 | *** part/#asterisk DaBigMac (~JJ@203-173-48-1.dyn.iinet.net.au) |
09:31.21 | JerJer | and mysql, yes |
09:32.20 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
09:34.07 | ScythelX | JerJer: how many t1s does nufone have |
09:36.06 | brc_ | http://despair.com/achievement.html |
09:37.17 | JerJer | ScythelX: t-1s lol |
09:37.21 | JerJer | we have three DS-3s |
09:37.28 | ScythelX | oh wow |
09:37.34 | ScythelX | $$ |
09:37.45 | JerJer | not $$ |
09:37.55 | JerJer | the loop is all of maybe 40 feet long |
09:37.57 | JerJer | :P |
09:38.22 | *** join/#asterisk SplasPood (~jwb@paravolve.net) |
09:42.19 | JerJer | i hope pranav doesn't try to run VoIP on the same connection he is IRC-ing from |
09:42.43 | modulus_ | i hope he does |
09:43.56 | jerlique | anyone help with agent logins/logouts? |
09:49.12 | *** join/#asterisk GMsoft (~r0_ot@gmsoft.developer.gentoo) |
09:49.14 | GMsoft | hi |
09:49.44 | *** join/#asterisk sysdef (~sysdef@pD9561EE0.dip.t-dialin.net) |
09:50.08 | sysdef | hi |
09:51.16 | *** join/#asterisk multrix (~chatzilla@ALyon-252-1-19-10.w82-122.abo.wanadoo.fr) |
09:53.49 | JerJer | hoe |
09:54.27 | modulus_ | eoh |
09:57.56 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
09:59.54 | jerlique | where is the structure/programming/information which describes how queues handle functions like queue-youarenext |
10:01.05 | GMsoft | anyone have experience with chan_bluetooth ? I'm getting SCO connection reset by peer each time I try |
10:06.15 | JerJer | i cannot even make my bluetooth dongle work in linux at all |
10:07.07 | JerJer | perhaps you need to pay the SCO license fee |
10:07.14 | GMsoft | haha :) |
10:09.01 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
10:20.17 | jerlique | is everyone in bed? this channel is quiet as |
10:24.27 | *** join/#asterisk darkpioneer (~Pioneer@spc1-hava1-4-0-cust101.cosh.broadband.ntl.com) |
10:25.47 | Zeeek | night |
10:26.04 | darkpioneer | i have an 2 x100 cards in my asterisk server and when im makeing calls with them i get like "pip" noises on the line |
10:26.22 | Zeeek | remove the seeds |
10:26.40 | darkpioneer | ? |
10:26.41 | Zeeek | darkpio have you checked IRQ sharing? |
10:26.52 | darkpioneer | ill have a look |
10:26.56 | Zeeek | I have 2 X100P |
10:27.05 | Zeeek | so I know of which I speaks |
10:28.37 | darkpioneer | also get the pip noises on incomeing calls |
10:30.43 | Zeeek | IRQ? |
10:32.23 | darkpioneer | i have to admint, i havenot cheqed irq befor so i dont know what to type |
10:33.25 | Zeeek | Either I have to go look or you have to go look to find that info. Which is more logical? Make sure your cards are not sharing IRQ and check some docs to find out more |
10:33.27 | Zeeek | Starter tutorial: |
10:33.27 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
10:33.27 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
10:33.27 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
10:33.27 | Zeeek | THE reference of the moment: |
10:33.29 | Zeeek | http://www.asteriskdocs.org |
10:33.49 | darkpioneer | roger |
10:33.51 | darkpioneer | thanks |
10:34.04 | Zeeek | ok - if I knew I'd tell you, but I can never remember :) |
10:34.14 | darkpioneer | k |
10:34.22 | Zeeek | OS dependent anyway I guess |
10:34.46 | *** join/#asterisk jofa (~jofa@a80-127-56-82.adsl.xs4all.nl) |
10:34.52 | darkpioneer | FC3 |
10:36.09 | Zeeek | I use slackware and I think the info in in /proc/interrupts |
10:37.11 | darkpioneer | <PROTECTED> |
10:37.36 | darkpioneer | ahh |
10:37.39 | darkpioneer | soundcard |
10:37.40 | darkpioneer | hmm |
10:37.43 | Zeeek | dump it! |
10:37.52 | Zeeek | and where's the second X100 ? |
10:38.03 | darkpioneer | <PROTECTED> |
10:38.16 | darkpioneer | hmm |
10:38.20 | darkpioneer | removeing soundcard.... |
10:38.28 | Zeeek | I'd pull the sound card to check, then if it is that, try moving it around or playing with BIOS |
10:38.34 | darkpioneer | yeah |
10:38.44 | darkpioneer | i dont need the soundcard anyway |
10:38.52 | Zeeek | I have a PIII-800 in an ASUS mobo and three digium cards |
10:39.04 | Zeeek | the hardest to install was the TDM400P |
10:39.13 | Zeeek | but that one finally settled down too |
10:39.42 | Zeeek | I disabled all interrupts like USB, etc. The box does nothing but asterisk - and I did pull the sound card as well |
10:40.01 | Zeeek | it still has an unneeded mouse which uses an IRQ but there are several free ones anyway |
10:40.32 | darkpioneer | i was trying to use the soundcard for a pa |
10:40.50 | Zeeek | you can maybe move it or change IRQ in BIOS |
10:40.55 | darkpioneer | yeah |
10:41.01 | darkpioneer | ill have to have a look |
10:42.23 | darkpioneer | damit, forgot the soundcard was onbord. ill have to dissable it in the bios |
10:42.39 | Zeeek | if you have to go to bios, see if you can change the IRQ |
10:43.00 | darkpioneer | yeah |
10:43.30 | darkpioneer | brb |
10:44.00 | GMsoft | I've got a voice modem. which chan should I use ? would chan_capi works ? |
10:44.07 | Zeeek | no idea |
10:55.09 | Zeeek | still trying to figure out why no callerid on ONE phone |
11:00.32 | zoa | chan_modem |
11:00.35 | zoa | but trust me |
11:00.38 | zoa | it wont work :p |
11:00.53 | GMsoft | heh |
11:01.42 | GMsoft | I'm trying with chan modem but asterisk doesn't seems to like my modem :) |
11:02.12 | zoa | thats normal |
11:02.18 | zoa | you need a wildcard |
11:03.05 | GMsoft | you mean for the extention ? that's not the problem |
11:03.23 | GMsoft | chan_modem fails to configure the modem and then stop asterisk |
11:06.40 | *** join/#asterisk HuangDi (TheEmperor@218.111.51.155) |
11:08.10 | darkpioneer | how do i start zaptel? |
11:08.20 | darkpioneer | seems i doesnt like starting automaticly |
11:09.57 | *** join/#asterisk rajo (~rajo@graphics.cs.uni-sb.de) |
11:10.47 | *** join/#asterisk maik_ (~maik@scumm.cs.uni-sb.de) |
11:11.15 | Zeeek | again |
11:11.16 | Zeeek | Starter tutorial: |
11:11.16 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
11:11.16 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
11:11.16 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
11:11.16 | Zeeek | THE reference of the moment: |
11:11.17 | Zeeek | http://www.asteriskdocs.org |
11:11.35 | Zeeek | look at automated - there is a complete guide from a-z for zaptel |
11:11.39 | *** join/#asterisk Fanguin (~Fanguin@p50819BF0.dip0.t-ipconnect.de) |
11:12.13 | *** join/#asterisk Tili (~Tili@202-133-65-239-dialup.sat.net.pk) |
11:13.12 | darkpioneer | right |
11:20.53 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
11:23.10 | *** join/#asterisk pranav (pranav@202.149.48.198) |
11:24.53 | pranav | hello everyon |
11:25.15 | Zeeek | hi pranav - I'm afraid to ask... |
11:28.44 | *** join/#asterisk pranav (pranav@202-149-48-198.broadband.isp.exatt.net) |
11:28.59 | pranav | sorry i got disconnected |
11:30.00 | pranav | i am still facing the same problem , the calls to fwd are not gong |
11:30.11 | pranav | going |
11:30.22 | Zeeek | how far have you gotten? |
11:30.28 | Zeeek | everything else works? |
11:30.32 | Zeeek | to SIP or IAX? |
11:32.06 | pranav | through sip |
11:32.14 | pranav | ya everything else works |
11:32.20 | Zeeek | you have other SIP providers working though? |
11:32.50 | *** join/#asterisk sysdef (~sysdef@pD9560EB9.dip.t-dialin.net) |
11:32.56 | pranav | i can call to pstn, mobile and also internally |
11:33.21 | Zeeek | none of that answers my question though which is: you never have made a SIP call work to a provider ? |
11:34.05 | pranav | ya have not made any call to sip provider |
11:34.26 | Zeeek | why don't you get another free account somewhere else and try it? It will only take 5 minutes |
11:34.55 | Zeeek | there's like2phone, gossiptel... |
11:35.14 | Zeeek | or if you speak German or Italian... providers in those places |
11:35.27 | pranav | no i know only english |
11:35.44 | Zeeek | and what else? |
11:35.45 | pranav | so i'll create another fwd ccount |
11:35.53 | Zeeek | no that won't prove anything |
11:35.58 | pranav | and other indian languages |
11:36.05 | Zeeek | hindi ? |
11:36.11 | pranav | yes ofcourse |
11:36.17 | Zeeek | and another 14 dialects ? :) |
11:36.28 | pranav | are you from india aswell |
11:36.45 | Zeeek | far from it! but I have worked with a few in Huntsvill, AL years ago |
11:36.56 | pranav | ok |
11:36.59 | Zeeek | that is ironically where asterisk is (digium) |
11:37.18 | Zeeek | anyway I suggest you try an account at a different provider first |
11:37.45 | Zeeek | in fact - you could get a test account at voipjet if they are still offering them |
11:37.51 | Zeeek | let me see |
11:38.18 | Zeeek | Yes, look here: https://www.voipjet.com/join.php |
11:38.22 | shido6 | u dont like NuFones sip termination? |
11:38.37 | Zeeek | talking about FWD |
11:39.32 | Zeeek | voipjet has a nice feature with that test account thing. You can get it working before paying :) |
11:39.57 | pranav | ok let me try with voipjet |
11:40.07 | Zeeek | pranav or try one of those other free ones. Also IAX with iaxtel |
11:40.16 | Zeeek | in fact IIRC voipjet is IAX |
11:40.40 | pranav | ok |
11:41.17 | pranav | ok see i have a sip provider in uk , if i want to connect to that then what should i do? |
11:41.26 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
11:41.37 | Zeeek | who are you talking about |
11:42.03 | pranav | see we have our own server in uk , i want to connect to that |
11:42.43 | Zeeek | which one? Free? |
11:43.07 | Zeeek | I just noticed I have a DID in UK from gisspitel |
11:43.13 | Zeeek | free. Gossiptel |
11:43.25 | pranav | ok gossiptel is free |
11:45.17 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
11:45.34 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
11:48.35 | Zeeek | heh I had to try about 80 emails before I could get my gossiptel password sent to me |
11:52.07 | Zeeek | I must have about 15 providers - gotta clean those files up some day |
11:55.55 | Mw3 | hm |
11:56.01 | Mw3 | i can't sign up to gossiptel |
11:56.14 | Mw3 | it said there were too many credit card fraud from hungary :( |
11:58.25 | Zeeek | awwww geee |
11:58.53 | Zeeek | there are about 90 others you may be able to sign up for |
11:59.14 | Zeeek | you need a UK DID? |
11:59.40 | Zeeek | does everyone realize that these free DID in the UK are VERY expensive for the person calling? |
12:02.52 | *** join/#asterisk Xenesis (~Xenesis@212.127.97-84.rev.gaoland.net) |
12:04.25 | murangd | Zeeek: do you know any cheaper providers than voipjet for call termination for the carbriean islands |
12:07.16 | Zeeek | murangd you have been asking the same two or three questions for several days here |
12:10.07 | Mw3 | Zeeek: why so expensive ? |
12:10.41 | Zeeek | what uk did? Dunno - but nothing is ever free - guess they have to make money, right? |
12:11.25 | *** join/#asterisk rkjpl (~rk@adsl-209-233-135-56.dsl.lsan03.pacbell.net) |
12:11.40 | Mw3 | that's right |
12:12.02 | Zeeek | so the best thing is to first get free accts with FWD and IAXTEL |
12:12.15 | Mw3 | by the way can i get somewhere DID from other countries than UK |
12:12.22 | Mw3 | for free |
12:12.25 | Zeeek | make sure your friends get on voIP and it'll be free |
12:12.40 | Zeeek | Bagdad? |
12:12.50 | Mw3 | no :), pragha |
12:12.55 | sysdef | germany |
12:13.07 | sysdef | nikotel ? |
12:13.12 | Zeeek | ah |
12:13.33 | Mw3 | i've uk did and when my brother was in .uk he could call us for about nothing (local call price) ... |
12:13.34 | sysdef | has also numbers from london |
12:13.43 | Mw3 | he will go to pragha soon :) |
12:13.48 | Zeeek | Mw3 where was that UK DID? |
12:13.52 | murangd | Zeeek: no one have provided me with a good answer |
12:13.58 | Mw3 | Zeeek: fwd |
12:14.12 | Zeeek | callUK? Far from free |
12:14.29 | Mw3 | cheaper than calling hungary from uk |
12:14.35 | Mw3 | much cheaper |
12:14.54 | Zeeek | there are very few good reasons to call hungry from anywhere that's why. |
12:17.31 | Mw3 | it's a good reason that i've been living there :) |
12:17.44 | Zeeek | even so, I wouldn't :) |
12:17.54 | Zeeek | pay to call u |
12:22.40 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
12:24.00 | *** join/#asterisk ckruetze (~ckruetze@i3ED61A1E.versanet.de) |
12:29.52 | Zeeek | . |
12:33.14 | *** join/#asterisk ZX81 (matt@222-153-114-115.jetstream.xtra.co.nz) |
12:33.40 | ZX81 | :) |
12:33.57 | ZX81 | hello all |
12:33.59 | RaYmAn-Bx | Zeeek: sipgate does provide free proper UK DID...admittedly they do require you to sign up from a UK ip address but still. |
12:34.22 | RaYmAn-Bx | and obviously they can't change the price of geographic DIDs like they can with some of the 08XX numbers |
12:34.37 | ZX81 | downloading message 114 of 582... |
12:34.41 | ZX81 | omg 1 day!!! |
12:34.43 | ZX81 | lol |
12:35.15 | ZX81 | ~ping |
12:35.35 | jbot | pong |
12:35.45 | ZX81 | heh it works lol |
12:36.07 | ZX81 | airpor (KL) -> home (NZ) -> irc |
12:36.08 | ZX81 | :) |
12:36.12 | ZX81 | ok brb~ |
12:40.44 | *** join/#asterisk postel (~jp@postel.user) |
12:44.46 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
12:47.32 | *** join/#asterisk zeedo (~notroot@www.bsrf.org.uk) |
12:47.47 | Zeeek | those numbers cost a huge amount to call - I have sipgate, calluk and gossiptel they are all "national" no's |
12:48.07 | Zeeek | go look at the rates - they are higher than calling a european num from a regular phone |
12:48.38 | *** part/#asterisk zeedo (~notroot@www.bsrf.org.uk) |
12:48.41 | *** join/#asterisk jhiver (~jhiver@ABoulogne-102-1-3-10.w193-253.abo.wanadoo.fr) |
12:52.33 | RaYmAn-Bx | Zeeek: outgoing or? They can't control the price on geographic DID's... |
12:53.21 | Zeeek | just call one of those numbers and find out - my wife called me from London once - it was outrageously expensive |
12:53.34 | Zeeek | like maybe 10c/min ? |
12:53.42 | Zeeek | look up BT UK national rate |
12:54.36 | RaYmAn-Bx | there's a difference between 08XX numbers are 01/02 numbers |
12:55.03 | Zeeek | give me an example of a free DID in London and the rate to call it? |
12:57.04 | Zeeek | I would love to be wrong - it would be nice for anyone who needs a FREE DID |
12:58.58 | RaYmAn-Bx | I have an oxford DID free from sipgate... |
12:59.18 | RaYmAn-Bx | it costs the same as it would cost to call any other oxford number |
12:59.19 | Zeeek | and what does it cost per mintue to call this thing from there? |
12:59.25 | Zeeek | which is? |
12:59.45 | RaYmAn-Bx | no idea. My point is that it cost EXACTLY the same as calling any other UK number... |
13:00.00 | Zeeek | maybe BT is just way expensive |
13:00.28 | RaYmAn-Bx | the price to call it depends on your phonecompany..with "BT Together Option 1" calls are 3p per minute for daytime calls |
13:00.38 | RaYmAn-Bx | that is stupidly expensive but that's just the price there |
13:00.47 | RaYmAn-Bx | I can call UK from denmark for around 2p/min any time of the day |
13:00.53 | Zeeek | by the way, nowadays there are several companies offering unlimited national dialing for insanely cheap rates |
13:01.27 | RaYmAn-Bx | but my point is rather that the dids aren't any more expensive than any other UK number |
13:01.37 | Zeeek | like $20/mo including 8Meg DSL |
13:02.03 | Zeeek | when I signed uop for a london number it was... I haven't needed one since but then it was only national rates |
13:02.21 | Zeeek | I still have a sipgate.de acct and maybe even calluk if they don't purge their files |
13:02.46 | Zeeek | the national numbers are "bend over!" |
13:03.13 | Zeeek | compared to say a nufone toll free at 2c/min |
13:04.06 | *** join/#asterisk Koshatul (~evangelio@202.9.38.223) |
13:04.06 | Zeeek | Does sipgate have an english lang site now? They didn't when I signed up. Maybe I pressed the "Bend over" button? |
13:04.15 | RaYmAn-Bx | I have heard of special london numbers that are national from anywhere |
13:04.33 | RaYmAn-Bx | but yeah, the 0870 and 0845 are generally extortionate prices |
13:04.49 | RaYmAn-Bx | bt even mentions them as premium rate on some parts of they website :> |
13:05.07 | Zeeek | they're INCREDIBLY expensive - anyone point taken |
13:05.32 | Zeeek | since I haven't needed the numbers. I wonder what voiptalk is offering these days? I have an acct there as well |
13:05.52 | RaYmAn-Bx | anything non 01xxx codes should be stayed clear off imho :> |
13:06.00 | Zeeek | I'l looking at voiptalk |
13:06.03 | RaYmAn-Bx | and yeah, sipgate has sipgate.co.uk now as well |
13:06.28 | RaYmAn-Bx | and they only give out proper DID's to people who connect from a UK ip address (And can give a UK address) |
13:06.45 | Zeeek | Telappliant can now provide you with an 0845 number, which can be configured to point to your IP phone or IP PBX, enabling callers to dial into your IP network via a conventional landline telephone number. |
13:06.53 | Zeeek | supposedly "local UK" |
13:06.58 | RaYmAn-Bx | yeah |
13:07.00 | RaYmAn-Bx | that's bullshit |
13:07.03 | RaYmAn-Bx | it's not nocal uk |
13:07.03 | Zeeek | charged to the caller at the same rate as a local rate UK telephone call. |
13:07.08 | RaYmAn-Bx | it's Lo-Call |
13:07.18 | RaYmAn-Bx | there's a big difference |
13:07.35 | Zeeek | one time 10£ charge for those |
13:08.00 | RaYmAn-Bx | do they claim it's charged at the same rate as local rate? |
13:08.08 | RaYmAn-Bx | if so then that's a gigantic lie |
13:08.19 | Zeeek | I'm quoting the site above |
13:08.49 | RaYmAn-Bx | 0870 is supposedly national rate as well...but generally it's a lot higher (and most companies are starting to offer same price for local and national as well...excluding 08XX) |
13:08.50 | Zeeek | fax to email. Nice |
13:09.29 | Zeeek | my cust in England moved to Geneva. As a bonus, callerID usually works now too. It never did from UK |
13:09.46 | Zeeek | so I need no UK no's which is why I have three I never use |
13:11.23 | Zeeek | 40£ /yr for a local UK no |
13:11.50 | Zeeek | 01865 |
13:11.55 | Zeeek | for oxford |
13:12.41 | *** join/#asterisk pranav (pranav@202.149.48.198) |
13:12.44 | RaYmAn-Bx | and sipgate gives that free. |
13:13.05 | Zeeek | can you have as many as you want? |
13:13.11 | RaYmAn-Bx | no |
13:13.15 | RaYmAn-Bx | 1 only I think |
13:13.29 | *** part/#asterisk pranav (pranav@202.149.48.198) |
13:13.36 | Zeeek | Were gonna be in Switzerland at some point too |
13:13.43 | Zeeek | They were |
13:29.37 | jhiver | arrrgh guyz I have something kinda strange |
13:29.54 | jhiver | when I place VoIP calls => no echo, but with X100P card => echo |
13:30.20 | jhiver | (well, X100P "clone" card in fairness...) |
13:30.23 | Zeeek | not that strange |
13:30.40 | jhiver | so... u know how to troubleshoot this? |
13:31.16 | Zeeek | read the wiki first |
13:31.28 | Zeeek | volumes written on ech |
13:32.29 | coppice | and a little of it is even accurate :-) |
13:32.41 | Zeeek | sturgeon's law |
13:32.48 | Zeeek | 90% of everything is crap |
13:32.59 | jhiver | also, how do I reload zaptel modules? do I need to reboot the box or is simple asterisk restart ok? |
13:33.14 | Zeeek | restart should do it |
13:33.24 | coppice | that's not true. at least 98% of everything is crap |
13:33.31 | Zeeek | heh |
13:33.48 | Zeeek | no that can't be because that self invalidates what you just said :) |
13:34.02 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
13:34.10 | coppice | how? |
13:34.41 | Zeeek | because you (or me) saying everyhting is crap invalidates 90% of what we are saying too |
13:35.02 | Zeeek | so only one in ten of my responses can ever be correct |
13:35.30 | coppice | ah, you bring this 90% measure right down the sentance level? |
13:35.43 | Zeeek | to the letter level and beyone! |
13:35.48 | postel | Am i the only one that cant get the damn BT line to Hangup on a X101P (NOT a clone) |
13:35.55 | Zeeek | there is a difference between é and e |
13:36.20 | Zeeek | postel usually it hangs up automatically right at the beginning of the call |
13:36.34 | postel | heh |
13:36.59 | Zeeek | I would take a gander at the zapata.conf sample file |
13:37.20 | postel | well, been there, sone that, got the tshirt and everything |
13:37.23 | Zeeek | and perhaps the mailing list for BT woes |
13:37.24 | postel | no cigar.. :/ |
13:37.33 | Zeeek | progress and all that? |
13:37.43 | postel | yeapers |
13:37.53 | postel | i have 64386434834638468 google tabs |
13:38.01 | Zeeek | so the $90 saved in not buying digium.... |
13:38.08 | postel | IT is digium |
13:38.23 | Zeeek | oh yeah that was jhiver that had the close |
13:38.28 | Zeeek | clone |
13:38.43 | coppice | so the $90 wasted in buying Digium....... |
13:38.52 | Zeeek | 90% 90% |
13:39.03 | postel | coppice: hehe |
13:39.20 | Zeeek | coppince out of nowhere, faxes are now being received! Mostly spam faxes unfortunately |
13:39.34 | Zeeek | but technology doesn't read them |
13:39.45 | Zeeek | just saves them and sends em on the me :) |
13:41.09 | Zeeek | I swear there are patches for UK polarity reverse shit or whatever it is? |
13:41.45 | postel | and for some stupid reason it Dials to SIP on incoming (as it should) but the damn analogs that are on the line go crazy, ringing like there's no tommorow, a constant pitching tone that blows your drums away |
13:42.12 | Zeeek | what line? |
13:42.17 | postel | BT |
13:42.31 | Zeeek | you left phones on the asterisk lines? |
13:42.44 | Zeeek | I've done it but ist isn't good |
13:43.08 | coppice | all telcos want to be like BT, but most just can't seem to drag themselves down to that level |
13:43.16 | postel | well, when * answers the call they should STFU right? |
13:43.40 | Zeeek | no you shouldn't have other shit on the X100 lines |
13:44.18 | postel | thats a new one to me |
13:44.27 | postel | danka Zeeek |
13:44.29 | Zeeek | it's common wisdom AFAIK |
13:44.44 | postel | well, aint |
13:44.48 | Zeeek | it should screw up the answering though, but who knows? |
13:45.09 | Zeeek | When I first began at the office we left the phones hooked up. Bad. |
13:45.17 | RaYmAn-Bx | does it actually answer before the SIP extension answers? |
13:45.21 | Zeeek | first they ring for several rings more after answered |
13:45.37 | Zeeek | second, asterisk doesn't knbow if someone is on the phone |
13:46.32 | postel | Zeeek: you're in the uk? |
13:46.42 | Zeeek | not if I can help it! |
13:46.46 | postel | heh |
13:46.56 | postel | thats my boy |
13:46.56 | Zeeek | weather is just as bad here, so what would the point be? |
13:47.05 | Zeeek | Paris, FR |
13:47.16 | postel | let me tell ppl to remove all analogs |
13:47.25 | postel | you're close enough |
13:47.32 | Zeeek | try it and see if it helps your problem |
13:47.43 | postel | im back in a sec |
13:47.53 | Zeeek | I have one phone that doesn't show callerid here |
13:53.43 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
13:58.03 | *** join/#asterisk kiran (~kiran@202.62.88.140) |
14:02.01 | kiran | hi all |
14:06.29 | ariel_ | morning all |
14:06.44 | kiran | morning......... |
14:18.48 | jhiver | more like 'nite over here :) |
14:18.56 | jhiver | well, getting late neway |
14:19.02 | Zeeek | cloudy? |
14:19.21 | jhiver | yup it's been pouring down for the last 3 weeks or so |
14:19.34 | Zeeek | did you try wengo btw? |
14:19.35 | jhiver | 'tropical depression' they call it over here... => lots and lots of rain |
14:19.51 | jhiver | didn't no |
14:19.59 | Zeeek | you have family or freinds here? |
14:21.40 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
14:22.15 | jhiver | here? where? |
14:23.37 | Zeeek | FR Métropole |
14:23.43 | jhiver | yeah |
14:23.55 | jhiver | got a few mates in bordeaux |
14:24.01 | Zeeek | we've talked before - you have a "slow" DSL yes? |
14:24.09 | jhiver | call them occasionally tru voipjet, works fine |
14:24.17 | jhiver | I used to have ISDN |
14:24.25 | Zeeek | wengo is 6eu/mo unlimited |
14:24.26 | jhiver | but yesterday finally got DSL |
14:24.33 | Zeeek | ah congrats ! |
14:24.44 | jhiver | 512/128, better than 64/64 :) |
14:24.53 | Zeeek | absolument |
14:25.04 | jhiver | but it really more like 300/120... |
14:25.11 | jhiver | ah well |
14:25.19 | jhiver | really much better for shaping |
14:25.23 | Zeeek | still... beteer than ISDN |
14:25.37 | jhiver | I have set up a linux box as ethernet bridge and I do shaping at that level |
14:25.43 | jhiver | so the whole LAN is shaped |
14:26.34 | jhiver | works nicely... I have bittorrent, ssh and the family surfing like crazy and it's nice sound quality for VoIP |
14:27.52 | jhiver | 6 EUR unlimited is pretty cool... I'll have to try that @ some point |
14:29.21 | Zeeek | You have a geo DID "free" with it |
14:29.42 | Zeeek | although I'm not sure about DOM-TOM |
14:29.44 | jhiver | nice |
14:29.54 | Zeeek | wouldn't be goe but 08 something |
14:29.57 | *** join/#asterisk doughecka (~Doug@doughecka.user) |
14:30.15 | Zeeek | http://www.wengo.fr/assistance/forum/viewforum.php?f=9 |
14:30.20 | doughecka | woot, its still my birthday |
14:30.32 | jhiver | too bad |
14:30.43 | jhiver | tomorrow you'll be ok for about a year |
14:30.48 | Zeeek | http://www.wengo.fr/assistance/forum/viewtopic.php?t=1793 |
14:30.53 | Zeeek | Martinique ^^^^ |
14:30.57 | doughecka | haha |
14:31.30 | Zeeek | better yet jhiver: http://www.wengo.fr/assistance/forum/viewtopic.php?t=1736 |
14:34.07 | *** join/#asterisk Mike_TK (~Mike_TK@212.165.78.5) |
14:41.32 | Zeeek | . |
14:41.40 | file[laptop] | .. |
14:41.51 | Zeeek | ... .... .. __ |
14:41.59 | file[laptop] | ..? |
14:42.04 | Zeeek | morse code |
14:42.15 | Zeeek | ... S |
14:42.16 | file[laptop] | ah |
14:42.18 | Zeeek | .... H |
14:42.23 | Zeeek | .. I |
14:42.26 | Zeeek | ___ T |
14:42.35 | *** join/#asterisk eKo1 (~bernd@207.42.191.66) |
14:43.04 | Zeeek | everyone should know morse code. What if you're lost in a cave? |
14:43.38 | file[laptop] | then I'll get out of the cave and yell, "LONELY UNGUARDED FEMALE SEEKING COMPANIONSHIP" and see who shows up |
14:43.46 | eKo1 | Zeeek: If you're lost in a cave, morse code is not going to get you out of there. |
14:44.13 | Zeeek | What you never saw any Sagal or STallone movies? |
14:44.24 | Zeeek | or submarine ones? |
14:44.28 | file[laptop] | so if it's an old movie, too bad |
14:44.32 | file[laptop] | I don't watch many movies either |
14:44.58 | Zeeek | those aren't that old - I didn't expect to discuss "It's a Wondeful Life" though |
14:45.24 | Zeeek | anyway in all these moves people are buried alive and tapping in morse |
14:45.49 | eKo1 | Zeeek: Yeah, but that doesn't help if nobody else understands it. |
14:46.05 | jhiver | this ART stating that it's illegal to redirect DIDs to elsewhere is entirely bollocks! |
14:46.05 | Zeeek | exactly! that's why it should be mandatory |
14:46.11 | jhiver | jesus |
14:46.20 | Zeeek | no one pays attention tot hose |
14:46.26 | Zeeek | to those |
14:46.30 | eKo1 | People will just think the tapping is arbitrary and ignore it. |
14:46.35 | file[laptop] | oh, hrm, I had more money then I thought |
14:46.40 | Zeeek | NO! Not if you're any good at it! |
14:47.07 | jhiver | di di di da da da di di di |
14:47.14 | Zeeek | there's only one flaw. Everyone will have their iPods on so they won't hear |
14:47.38 | Zeeek | those are dits, not di |
14:47.55 | Zeeek | dit dit dit, dah dah dah, dit dit dit |
14:48.06 | Zeeek | faster than callerid and SMS |
14:48.10 | eKo1 | What is that? Baby talk? |
14:48.35 | Zeeek | ..__. .._ _._. _._ |
14:48.50 | eKo1 | Much better. |
14:48.58 | Zeeek | a very poor dialplan IMO |
14:49.13 | eKo1 | But let's face it, if morse code was any good, then telegraphy would still be around. |
14:49.14 | Zeeek | ..__. .._ _._. _._,1,Dial(SIP/2000) |
14:49.22 | Zeeek | IT IS AROUND! |
14:49.53 | eKo1 | Zeeek: You need to get out of that 1890 time hole. |
14:50.06 | Zeeek | think of the joy of discussing like we do here, but at the lightening peeds of 20 wpm! |
14:50.30 | Luhiwu | eKo1, i know lot of examples where morse code is still in use here in Argentina |
14:50.33 | Zeeek | telegraphy is still a requirement for radio licenses even tho they have satellite |
14:50.42 | Zeeek | on ship |
14:50.55 | jhiver | cq cq dx guys.. |
14:51.03 | jhiver | 'nouf morse |
14:51.05 | Zeeek | I know a ship's radio op that has seamen on her passport :) |
14:51.20 | jhiver | not enough speed with morse... sucks |
14:51.23 | Zeeek | bug laugh in every port |
14:51.26 | jhiver | I have a much better protocol |
14:51.29 | eKo1 | They don't use morse on ships, they use telex. |
14:51.33 | Zeeek | RTTY |
14:51.39 | jhiver | tap = 1, no tap = 0 |
14:51.40 | jhiver | :) |
14:52.00 | Zeeek | I used to receive RTTY on my TRS80 thgru the casette port |
14:52.03 | file[laptop] | someone write app_morsecode! |
14:52.19 | Zeeek | too easy |
14:52.40 | Zeeek | for the deaf - they can hold the receiver and feel the message |
14:53.10 | Zeeek | . . . . __ . . __ |
14:53.14 | Zeeek | that is the answer |
14:53.14 | eKo1 | They can also look at it from their cell phone. |
14:53.42 | Zeeek | . . . . __ . . __,1, Dial(ZAP/1/42) |
14:54.05 | file[laptop] | woot 42 |
14:54.59 | Zeeek | when is digium gonna make a deal with motorola like skype has done? |
14:55.20 | Zeeek | motorola is building skype into cellphones |
14:57.24 | Zeeek | so supposedly when you're near a wifi point, you can call via skype |
14:57.47 | Zeeek | which is of no interest except the fact that the app will be built in |
14:57.54 | Zeeek | to the phones |
14:58.01 | Godsey | I would bet the driving force would be vonage :) |
14:58.16 | Zeeek | if only they were using IAX |
14:58.18 | Luhiwu | Zeeek, the application will be skype? i've heard about configurable voip, not exactly skype |
14:58.31 | Zeeek | this is a deal done a few days ago |
14:58.56 | Luhiwu | somebody calls Nokia now! :) |
14:59.32 | Zeeek | http://www.technewsworld.com/story/wireless/motorola-skype-mobile-voip-40622.html |
14:59.52 | Luhiwu | tnx for the link |
14:59.53 | Zeeek | in fact, http://news.google.com/news?q=motorola%20skype&hl=en |
15:00.21 | coppice | I wonder how commited anyone is to UMA |
15:00.44 | Luhiwu | i've made a iax softphone based on the iaxclientocx library, maybe i should convert it to java and upload it to some java based phone... |
15:00.59 | Zeeek | It would be worthwhile trying that |
15:01.22 | Zeeek | They're mostly commited to NBT |
15:01.35 | Zeeek | All business is looking at NBT |
15:02.03 | Luhiwu | what is NBT? |
15:02.05 | Zeeek | the NextBigThang (tm) |
15:02.11 | Luhiwu | thanks :) |
15:02.31 | Zeeek | WAP yesterday, GPRS today, UMA and then.... NBT! |
15:03.11 | coppice | WiMAX? WiNOT! |
15:03.52 | Luhiwu | maybe we all should put some money in an asterisk bounty to finance a java based softphone for cells... i don't think i could port it to java alone, but i'd help if the proyect begins... |
15:03.53 | Zeeek | Interesting: Spokespersons at both Skype and Siemens said there are no immediate plans to market the connector and phones in North America. |
15:04.05 | Zeeek | referring to the Siemens cordless USB handsets |
15:04.55 | coppice | people are moving on from the NBT to the one after next big thing. most 3G networks are not yet on the air, and some operators are already taking about a migration to flash-OFDM |
15:05.19 | Zeeek | you know that 3G backfired into a major fiasco anyway |
15:05.30 | Zeeek | they paid zillions of the licenses |
15:05.46 | coppice | well, the phones are networks are starting to trickle out now |
15:06.01 | coppice | still very troublesome, though |
15:06.32 | Zeeek | I've seen ads for video on cellphones, I spose that's using UTMS of whatever it is |
15:06.40 | coppice | they get a rush of early adoptors, then sales dry up. then they need to discount to below GSM |
15:07.05 | Zeeek | all that cell stuff has been a license to print money over here in EU |
15:07.14 | Zeeek | 8 year olds have 2 cells each |
15:07.22 | coppice | yeah, but the video over UMTS is a joke. the only reason they can do that is there are few subs. with more subs they won't have the capacity |
15:08.13 | Zeeek | bad enough housewives are talking on the bloody things in the supermarket aisles |
15:08.25 | coppice | they are signing up people for one year of unlimited video at a low price. this means they have no expectation of lots of subs within one year |
15:08.34 | Zeeek | 90% of the planet has never seen a doctor even once |
15:08.51 | coppice | wow, that healthy, eh? :-) |
15:08.59 | Zeeek | ya, amazing eh? |
15:09.15 | Zeeek | they should tax "where are you?" cell calls about 500% |
15:09.19 | coppice | 1 in 5 humans now carries a GSM phone. |
15:09.27 | Zeeek | got one in my pocket right now |
15:09.45 | Zeeek | and I have almost no need for it |
15:09.48 | Luhiwu | coppice, i know a lot of not-so-humans that carries a GSM phone :) |
15:10.05 | Zeeek | I refuse to sign a sub though |
15:10.57 | Zeeek | kinda nice now that we have SMS working on asterisk - we can send each other SMS from the office asterisk on any browser |
15:11.19 | coppice | really very poor people have them. the compulsion to have them seems really string |
15:11.19 | Zeeek | so for $15/mo it's like a fancy pager |
15:11.59 | Zeeek | As I walked by him, a beggar sitting on the sidewalk pulled one out to see if he had any messages! |
15:12.25 | coppice | for $15 we get 1500 minutes airtime, so few people send SMSes |
15:12.33 | Zeeek | they're powered byt he kids over here - I'd hate to be a parent at the moment |
15:12.59 | Zeeek | when I'm out on business I can't answer my phone but I can read SMS |
15:13.18 | coppice | people here don't care :-) |
15:13.19 | Zeeek | and all incoming are unmimited |
15:13.27 | Zeeek | unlimited even |
15:14.03 | coppice | we have the same deal. I can send and receive for free, but people still don't use them |
15:14.30 | Zeeek | I could send commands to asterisk from SMS |
15:15.01 | Zeeek | another kind of DISA :) |
15:15.34 | Zeeek | "With over 68 million downloads of their client in the last 18 months, we believe Skype is a natural fit with our vision of simple and seamless connectivity for our consumer customers around the globe." |
15:16.07 | Zeeek | I wonder how many asterisk downloads there have been in the last 18 mo? |
15:16.51 | Godsey | we pay cingular something like $40/line and each user has 200 minutes :) |
15:17.56 | Zeeek | you may as well pay a hooker and get 3 min for $100 |
15:18.29 | Zeeek | By making Voice over IP truly mobile and easily accessible, we can make |
15:18.29 | Zeeek | communications seamless for consumers as they travel throughout the |
15:18.29 | Zeeek | environments of their day - at work, at home, in the car, or out in the |
15:18.30 | Zeeek | world, |
15:18.31 | Zeeek | By making Voice over IP truly mobile and easily accessible, we can make |
15:18.50 | Zeeek | So why not put a SIP or IAX client then instead of Skype! |
15:19.07 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
15:23.04 | jhiver | thing is, skype sucks because it's proprietary but you have to admit, it just works... |
15:23.41 | jhiver | too bad it's so hard to work with and do cool stuff with it |
15:23.51 | Zeeek | it does work well. Too bad about prioritary |
15:23.54 | *** join/#asterisk mhnoyes (~mhnoyes@user-38lc08a.dialup.mindspring.com) |
15:24.22 | eKo1 | Well, it really all comes down to social networking. The people at Skype have good contacts so... |
15:24.29 | jhiver | it'll be good when somebody reverse engineers the protocol and we start to see alternative clients :) |
15:24.46 | Zeeek | ok, we all need this: |
15:24.46 | file[laptop] | yes, yes it wil |
15:24.47 | Zeeek | http://www.ubergizmo.com/15/archives/2005/02/clothes_that_bi.html |
15:25.52 | jhiver | lol |
15:25.58 | Zeeek | ok wait: |
15:25.59 | Zeeek | http://www.worthersoriginal.com/index.php?id=25 |
15:26.22 | Zeeek | ubergizmo is pretty crazy site |
15:26.29 | eKo1 | I don't need that. I just rub my feet on some carpet and ZAP! |
15:26.45 | eKo1 | But then again, I also get zapped. |
15:26.54 | Zeeek | here's a little more down to earth view: "Motorola and Skype hop in bed on devices and accessories" |
15:28.33 | coppice | jhiver: once you have reverse engineered it, you will hit patent problems |
15:29.02 | eKo1 | You mean license problems. |
15:29.24 | coppice | no, patents |
15:30.06 | Zeeek | Europe does not recognize software as patentable |
15:30.11 | Zeeek | does the USA? |
15:30.13 | jhiver | true |
15:30.24 | jhiver | but it will eventually |
15:30.27 | coppice | oh, don't start that crap. codecs are patented everywhere |
15:30.40 | Zeeek | but it isn't the codec that makes skype what it is |
15:30.41 | jhiver | there's massive lobbies and they just won't let go |
15:31.11 | jhiver | I mean the OEB has been accepting patents and know they're working to get them enforceable |
15:31.19 | coppice | but for compatibility you need to use the iLBC which is patented and not free. they don't use the free one, even for narrow band |
15:31.23 | Zeeek | the latest ruling is negative |
15:31.40 | coppice | codec patents are enforceable, and always have been |
15:31.49 | jhiver | compatibility isn't the issue |
15:31.50 | Zeeek | licenses aren't that expensive for each unit are they? |
15:32.25 | coppice | they only reason you reverse engineer is to achieve compatibility, isn't it? |
15:32.29 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
15:32.32 | Zeeek | after all the BT100 does all those |
15:32.41 | Zeeek | iLBC, 729 |
15:33.01 | coppice | no it doesn't. that's the narrow band iLBC. skype doesn't use that |
15:33.07 | jhiver | Well not necessarily... if you could make something that works just as well as skype but that's open then it would kick ass |
15:33.21 | jhiver | of course asterisk works well but it's damn hard to set up |
15:33.28 | jhiver | too powerful for m. joe sixpack |
15:33.30 | Zeeek | coppice ? I didn't knwo there were several iLBC? |
15:33.35 | coppice | lots of things are open and work as well as skype. |
15:33.44 | coppice | look on the GIPS site |
15:34.08 | Zeeek | the skype phenomenon is NOT from codecs though, it's that you start it up and it works |
15:34.17 | Zeeek | any idiot can use it |
15:34.17 | jhiver | well, the P2P telephony thing *is* a good idea and the fact that you can have 2 NATTed skype users call each other is pretty cool too |
15:34.34 | Zeeek | so they went the extra mile on that part |
15:34.47 | jhiver | I agree |
15:35.04 | Zeeek | the rest, no one gives a f what codec the software is using |
15:35.16 | coppice | yep, but you need to be compatible with it. the use of widebnd plays a strong part in people's perception that for the first time VoIP is actually better than PSTN calls |
15:35.22 | Zeeek | and now, if they do make deals right and left, the name has so much buzz value |
15:35.44 | Zeeek | they may be able to make a niche for themselves |
15:36.01 | Zeeek | also voIP will never be sold to the public as voIP - it's the ugly sister |
15:36.06 | coppice | there's no room for niches in telecoms. its all or nothing |
15:36.11 | Zeeek | the public isn't geeky - they want PnP |
15:36.51 | Zeeek | skype will likely be bought by a big op RealSoonNow, no? |
15:36.52 | ariel_ | actually the skype appeal is due to the kids using it. There the ones that use it most. |
15:37.03 | coppice | doubt it. |
15:37.05 | Zeeek | that is also true, they power the cell revolution too |
15:37.16 | jhiver | Well if skype manages to take over the world and all phone is free then I say why not. The problem is that if they do achieve monopoly then it won't stay free very long... |
15:37.16 | Zeeek | coppice why not? |
15:37.31 | Zeeek | jhiver skype phoning isn't free except to other PC |
15:37.40 | Godsey | it isn't free |
15:37.44 | Zeeek | they now sell minutes like everyone else |
15:37.49 | coppice | if a telco buys it, all the other telcos will run away. then it is worthless |
15:37.54 | Godsey | not free as in monitarialy or freedom :) |
15:37.54 | ariel_ | free I don't think that the bell's will allow that since they have there needs. |
15:38.06 | jhiver | yeah but I don't think their plan is to make money selling minutes |
15:38.15 | Zeeek | they are selling thelm now |
15:38.17 | jhiver | I think it's more like: |
15:38.24 | jhiver | 1. get everybody to use us |
15:38.30 | Zeeek | [and cool accessories] |
15:38.42 | jhiver | 2. argue that too many natted clients => need specialised servers |
15:38.42 | Godsey | I'm just waiting for a SBC Verizon merger :) |
15:38.52 | jhiver | 3. next version only connects to skype's servers |
15:39.07 | jhiver | 4. charge small amounts of money for calls |
15:39.25 | Godsey | I think phones are in a space easier to adopt ipv6 :) |
15:39.40 | jhiver | but maybe they really do want to make it free for everybody... don't know, i don't know them :) |
15:39.42 | coppice | if they have sense they won't charge for calls. they will charge a simple subscription |
15:39.55 | jhiver | yeah |
15:40.12 | jhiver | and premium rates for businesses of course... |
15:40.14 | eKo1 | Yeah, the charge per minute model is getting old. |
15:40.23 | ariel_ | subscriptions is the way that most will go just like the new napster. |
15:40.34 | eKo1 | It's old and complicates my programming substantially. |
15:40.58 | coppice | 90% of the code in a switch is doing accounting |
15:41.05 | eKo1 | Tell me about it. |
15:41.22 | file[laptop] | MOOOOOOOOOOOOOOOOO |
15:41.25 | *** join/#asterisk numBone (~numBone@c-24-129-204-233.se.client2.attbi.com) |
15:41.32 | Godsey | accounting records ware what I spend the most time dealing w/ as an isp too :) |
15:42.01 | Zaw | are there any TDM cards that are compatible with freebsd and asterisk? |
15:42.15 | Godsey | tho I like the flexability of charging for usage or flat for the customer |
15:42.27 | Godsey | people like both ways, and since it's $$ I do to |
15:42.28 | Godsey | :) |
15:42.31 | Godsey | too |
15:42.42 | dan2 | anybody have a sipura 2100 and know what the sipura code to turn on the wan side webserver is? |
15:42.47 | ariel_ | I wish that the colo were flat rates. |
15:42.51 | eKo1 | Yeah, but using a flat fee business model simplifies everything. |
15:42.55 | coppice | as soon as you break even a little bit from the subscription model you get 100% of the problems |
15:43.21 | *** join/#asterisk polymath (~jeffg@dsl027-163-129.atl1.dsl.speakeasy.net) |
15:43.28 | eKo1 | dan2: I don't think there is a wan side to it. |
15:43.32 | coppice | people will always go for subscription over pay as you go |
15:43.34 | jhiver | on the other hands you have to deal with abusers with flat rate model... |
15:43.36 | ariel_ | dan2, I know it's a web setting I did not know there was one for the ivr for the 2100 |
15:43.46 | jhiver | or special cases like "i have lots of family in india" |
15:43.48 | eKo1 | jhiver: yeah. |
15:43.57 | jhiver | so ATM it's kinda hard |
15:44.05 | plappy | thats why ya just buy an uncapped link. :) thats flat rate. |
15:44.28 | ariel_ | how can you abuse flat rate you charge for unlimited it should mean unlimited. |
15:44.40 | coppice | it makes no diffence to skype. they only do the call switching |
15:44.54 | jhiver | sure but termination isn't flat... you have to work out what customers are gonna cost you on average and if you underestimate your averages (or say your currency goes down) u r screwed |
15:45.30 | polymath | any chance somebody could help me troubleshoot a wcfxo problem? |
15:45.31 | jhiver | I'm talking from the business point of view, not the consumers' |
15:45.31 | eKo1 | jhiver: That's why you overcharge the customer so your ass is covered. |
15:45.33 | ariel_ | jhiver, that is due to the cash cows telco's |
15:46.00 | ariel_ | polymath, state your problem |
15:46.03 | jhiver | yeah but _practically speaking_, doing flat rate voip service at the moment is hard |
15:46.19 | jhiver | have you seen the prices on some mobile destinations? |
15:46.23 | ariel_ | jhiver, it's a mixed bag right now. |
15:46.23 | coppice | oh, for termination you have no choice but to work within the local framework. if local calls are charged you have no choice but to charge. you need to base a business model on increasing IP-to-IP and decreasing termination, though |
15:46.24 | eKo1 | jhiver: Yeah |
15:46.50 | polymath | i'm getting a red alarm when i plug in a real rboc line to my wcfxo, but if i plug in the tel port of my iaxy the alarm clears and * takes calls |
15:46.53 | eKo1 | But once those mobile destinations start using VoIP, it will all be cheaperl. |
15:46.58 | *** join/#asterisk otiske (~otiske@kauai.sys.pas.earthlink.net) |
15:46.59 | Zeeek | I go away for two minutes and people are discussing how to make money. SHameful! |
15:47.01 | ariel_ | that is why if something like dundi takes off would help us allot. |
15:47.36 | coppice | skype can only be an interim step. the long term has no need for a middle man |
15:47.40 | otiske | has anyone built asterisk with ICD on FreeBSD 5.3? |
15:47.50 | ariel_ | polymath, what is an rboc |
15:47.54 | polymath | to be up-front, the wcfxo is a generic (winmodem). i know that's a touchy topic, but i only need it for a month and will be buying more digium hardware |
15:48.06 | polymath | rboc == regional bell operating co (bellsouth in this case) |
15:48.19 | bjohnson | what is dundi anyway? |
15:48.25 | bjohnson | is it like e164.org? |
15:48.29 | polymath | bjohnson: http://dundi.info |
15:48.31 | ariel_ | polymath, I have mine here plugged into the bellsouth line works fine. |
15:48.53 | coppice | soon to be tonboc = the one nation bell operating co :-) |
15:48.59 | polymath | ariel_, is yours a generic or a supported wcfxo |
15:49.08 | ariel_ | the real thing |
15:49.19 | polymath | coppice, heh... i'm starting an office pool on how many years it takes to go from divestiture to 100% reconsolidation |
15:50.52 | eKo1 | The trick is, you only make it look like a divestiture so you never have to reconsolidate. |
15:51.05 | bjohnson | is dundi already in use in a public system? |
15:51.10 | polymath | ariel_, when i plug in the bell line, even with wcfxo and zaptel unloaded, it's as if the line loops up -- call it and it's busy |
15:51.41 | polymath | bjohnson, http://dundi.info/members.html |
15:52.04 | polymath | bjohnson, bellster^H^H^H^H^H^H^H^Hfwdout uses dundi in a limited capacity |
15:52.36 | ariel_ | polymath, use http://pastebin.ca and post your zaptel and zapata.conf files |
15:53.20 | coppice | polymath: it won't be reconcildation. it will be building a co strong enough to compete in the new deregulated global marketplace...... which just happens to have a total monopoly whereever it operates, as it owns all the coax, pairs and fibre |
15:53.39 | *** join/#asterisk loick (~loick@ATuileries-151-1-27-239.w82-123.abo.wanadoo.fr) |
15:54.00 | bjohnson | polymath: anyone else? I guess what I'm getting at is .. how usable is it currently? |
15:57.21 | *** join/#asterisk polymath (~jeffg@dsl027-163-129.atl1.dsl.speakeasy.net) |
15:57.38 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
15:57.52 | polymath | ariel_, sorry, had an xkill mishap |
15:58.57 | ariel_ | fxs_ks I use this instead of fxs_ls |
15:59.26 | ariel_ | bjohnson, dundi is still in testing |
15:59.52 | polymath | ariel_, i tried earlier with fxs_ks, but let me give it a shot again |
16:00.03 | ariel_ | bjohnson, but I have used it for an enterprise system. Which can be great for internal asterisks systems. |
16:00.52 | GMsoft | what's the status of asterisk on bug endian box ? is it working ? |
16:01.02 | GMsoft | err s/bug/big/ |
16:02.08 | polymath | GMsoft, * allegedly works on powerpc, which is normally big-endian |
16:02.33 | GMsoft | ok. I'm compiling on parisc right now |
16:02.54 | polymath | GMsoft, linux or hpux? |
16:03.09 | GMsoft | gento linux :) |
16:03.13 | GMsoft | +o |
16:03.20 | polymath | GMsoft, nice |
16:03.28 | GMsoft | my kbd is not cooperative today :) |
16:03.33 | Zeeek | polymath - a while back I had a power failure. The x100P shorted the phone line even when power was restored |
16:03.53 | Zeeek | but having removed power for 24hrs, they work again |
16:04.13 | Zeeek | it was depressing to shut down the asterisk box and plug in phones :( |
16:04.36 | ariel_ | Zeeek, he says it works off the iaxy as the a source. |
16:04.54 | polymath | ariel_, nod |
16:05.09 | Zeeek | ok, just though I'd chime in with that useless bit of info |
16:05.18 | polymath | Zeeek, thanks though |
16:05.23 | Zeeek | while I'm download 400 megs of music |
16:05.56 | ariel_ | polymath, here is my files. http://pastebin.ca/6129 |
16:06.06 | Zeeek | the Internet is a wonderful invention for instant satisfaction of many kinds :) |
16:06.47 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
16:07.02 | Zeeek | another example of "if you build it they will COME" |
16:07.15 | polymath | ariel_, about to try your configs now, thanks |
16:07.19 | ariel_ | don't you just hate it when you press the wrong key and you exit all the program instead of others. |
16:07.25 | *** join/#asterisk trym (~trym@linux.debian.us) |
16:07.55 | polymath | ariel_, yes -- i use evilwm, which puts "kill window" right next to "switch to desktop 1" =] |
16:08.51 | *** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4116580.sympatico.ca) |
16:09.16 | NormAst | Hi all. |
16:10.00 | polymath | ariel_, about to have to rmmod -f zaptel (accidentally tried to rmmod it while * was running) so may have to reboot... |
16:11.23 | ariel_ | try service zaptel restart |
16:11.39 | dan2 | hmm... Wonder how much longer I'm going to have to stay on hold... |
16:12.07 | ariel_ | dan2, is the moh good? |
16:12.09 | polymath | ariel_, no luck. reboot is in order i think |
16:12.13 | polymath | brb |
16:12.21 | dan2 | ariel_: sounds like elevator music, so no |
16:12.34 | dan2 | ariel_: its quire ironic I'm calling my own companies support line |
16:12.43 | ariel_ | heheeh |
16:12.44 | Silik0n | elevator music is the best |
16:12.58 | dan2 | Silik0n: it sucks compare to 32 channels of xm |
16:13.08 | Silik0n | hah |
16:13.13 | ariel_ | funny I dial my moh on my system allot just to listen to it. |
16:13.24 | Silik0n | same here |
16:13.44 | Silik0n | but I have all kindsa of crazyness on my moh |
16:14.24 | Silik0n | most of which I doubt any person in their right mind would put on a corporate MoH tho |
16:14.46 | Silik0n | dan2 headset |
16:14.56 | dan2 | Silik0n: don't have one of those either |
16:17.42 | *** join/#asterisk polymath (~jeffg@dsl027-163-129.atl1.dsl.speakeasy.net) |
16:17.51 | inspired | hmm, is it possible to convert a doc file to tif on linux? |
16:17.59 | inspired | or doc to pdf to tif |
16:18.16 | inspired | I know pdf to tif works, I just have to understand how to convert a word document to an image |
16:20.18 | *** part/#asterisk Kumbang (~ecvs@167.205.24.4) |
16:20.45 | RoyK | inspired: openoffice can do that |
16:20.47 | RoyK | hm |
16:20.55 | RoyK | why tif? fax? |
16:20.57 | inspired | openoffice requires X :p |
16:20.59 | inspired | yes, fax |
16:21.30 | polymath | ariel_, still same symptoms with your config (adapted to my environment)... |
16:21.35 | inspired | our users are probably too stupid to understand that they have to convert from doc to pdf on their own computer, so our machine has to do it for them |
16:22.23 | polymath | ariel_, got to run, thanks for your help! |
16:22.35 | ariel_ | polymath, sorry could not help more. |
16:23.45 | Zeeek | inspired sometimes a printer driver is the best solution |
16:23.54 | Zeeek | they may understand that |
16:24.22 | Zeeek | how are they getting their document to you? |
16:24.27 | inspired | our customers are not going to use any fax machines. we are doing web to fax |
16:24.30 | inspired | and fax to email |
16:24.43 | coppice | Lots of people are used to print to FAX on windows |
16:24.48 | Zeeek | why not Word to "print as fax"? |
16:25.02 | Zeeek | I think there are even free drivers |
16:25.03 | inspired | how will that help me? |
16:25.12 | Zeeek | print as fax and mail |
16:25.28 | Zeeek | or upload |
16:25.35 | inspired | uhm, print as fax = creates a file? |
16:25.40 | Zeeek | of course |
16:25.42 | Zeeek | a TIF |
16:25.42 | inspired | ah |
16:25.47 | inspired | nice |
16:26.12 | Zeeek | see somehow the average user will get that where as a convert or save as is seen as an extra step |
16:26.44 | Zeeek | for a quick idea try j2.com - I think you can download a free driver - or efax |
16:27.09 | Zeeek | or maybe there's open source stuff out there |
16:27.36 | inspired | ok |
16:28.15 | Zeeek | http://wwwi.efax.com/fr/efax/twa/page/download |
16:28.30 | Zeeek | oops they outsmarted themselves detecting Fren,ch |
16:28.52 | Zeeek | http://www.efax.com/en/efax/twa/page/download |
16:28.59 | inspired | I don't want our customers to use efax |
16:29.02 | inspired | I want them to use us ;) |
16:29.04 | *** join/#asterisk cybercron (~test@208-216-127-234.cust.gti.net) |
16:29.47 | Zeeek | yeah but the software is free - |
16:29.52 | Zeeek | not the faxing |
16:29.53 | ManxPower | Does eFax offer branded services? |
16:30.00 | Zeeek | they might? |
16:30.19 | inspired | well, seems that print to fax is the best idea |
16:30.30 | inspired | it will work well with our product |
16:30.56 | Zeeek | you can prolly find a driver out there somewhere - in fact maybe built in to Windoze |
16:31.05 | Zeeek | XP? |
16:31.32 | Zeeek | hehe, then there's Exchange :) |
16:31.38 | inspired | print to fax is not standard in word? |
16:31.39 | `Sauron | Is X-Lite the most popular softphone for winblows? |
16:32.10 | Zeeek | inspired I don't think so but XP is different it has a lot of that built in |
16:32.38 | |Vulture| | anyone ever get outbound fax working sucessfully with * via fax--fxs gateway--asterisk--(voip provider or POTS) |
16:32.55 | eKo1 | |Vulture|: not me. |
16:32.59 | `Sauron | hum, and voip-info seems to be down, bummer. |
16:33.12 | |Vulture| | yea I tried it awhile ago never got it to work |
16:33.30 | eKo1 | The problem is that your provider may not be using ulaw so... |
16:33.43 | |Vulture| | true.. but POTS shouldn't matter |
16:34.12 | eKo1 | Yeah, I was able to send faxes no problems but I could never receive any. |
16:34.21 | inspired | "Microsoft Office Document Writer" is default in MS Word 2003 |
16:35.19 | Godsey | vulture: yes we send faxes all the time |
16:35.36 | Godsey | you just have to disallow=all, allow=ulaw for the device and sip provider |
16:36.29 | ariel_ | |Vulture|, I have fax working via an iax provider. |
16:36.44 | Godsey | I wish winfax pro had sip support :) |
16:37.48 | ariel_ | |Vulture|, in fact I have fax working via sipura-2000 and sipura-2100. What problems are you getting? |
16:38.03 | |Vulture| | I think it might be our fax machines |
16:38.06 | |Vulture| | they are POSes |
16:38.09 | Godsey | we use linksys pap2 devices |
16:38.19 | Godsey | not sure what it is rebranded from :) |
16:38.26 | |Vulture| | ariel_: you use it over VPC? |
16:38.32 | ariel_ | Godsey, there sipura |
16:38.49 | ariel_ | |Vulture|, no actually i use it via race.com |
16:39.04 | ariel_ | voipjet is not good for it either. |
16:39.16 | |Vulture| | ariel_: do you know what fax machine you use? we have some all in wonder I was working with |
16:39.18 | |Vulture| | it sucked |
16:39.19 | ariel_ | And vpc I can get them inbound but only 50% of them out bound. |
16:39.41 | ariel_ | well here I am using hp officejets and internal modems. |
16:39.43 | |Vulture| | hmm race.com not found |
16:39.50 | ariel_ | www.race.com |
16:40.04 | |Vulture| | strange... nothing |
16:40.13 | ariel_ | There network is directly to tdm switches |
16:40.44 | ariel_ | yes your right. hummm I know why. |
16:41.03 | *** join/#asterisk drumkilla (~russell@12.21.241.80) |
16:41.03 | *** mode/#asterisk [+o drumkilla] by ChanServ |
16:41.19 | ariel_ | There moving this weekend to a new location for there colo. There going to offer service to the public in a week or so. |
16:41.41 | |Vulture| | oh oky, I think I am going to buy a nice fax to test with |
16:46.36 | `Sauron | Is there a simple SIP softphone where you can just put in username@domain.com for dialing someone who's got SRV records and al that set up properly? |
16:46.43 | *** join/#asterisk oej (~oej@apollo.webway.se) |
16:48.29 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
16:48.31 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
16:50.12 | BoRiS | !seen paulc |
16:50.23 | BoRiS | ~seen paulc |
16:50.25 | jbot | paulc <~paulc@S010600062586a0b4.vc.shawcable.net> was last seen on IRC in channel #asterisk, 7d 21h 36m 33s ago, saying: 'Firestrm: I gotta head out for lunch, got an appointment, I'll leave you in the capable hands of Dr B Johnson :-)'. |
16:53.14 | inspired | `Sauron: most/all should support that |
16:53.32 | *** join/#asterisk visik7 (~ciao@host11-39.pool80182.interbusiness.it) |
16:53.51 | visik7 | what's the * pastebin site ? |
16:53.56 | visik7 | that I forgot :) |
16:53.56 | Nivex | ~pastebin |
16:53.58 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
16:54.03 | visik7 | grazie |
16:54.09 | visik7 | sorry |
16:54.10 | visik7 | I mean |
16:54.12 | visik7 | Thank you |
16:54.17 | `Sauron | inspired: Hum. Maybe I didn't look enough at the config, then. |
16:54.36 | Nivex | visik7: You are welcome. |
16:55.13 | inspired | if the SRV records for domain.com are set up and pointed to a SIP server with the user "username", it will work |
16:55.28 | inspired | just add exten => username to the standard context on your sip server |
16:56.22 | inspired | i.e. [default] |
16:56.46 | inspired | exten => username,1,Goto(users,0001,1) |
17:02.26 | Zeeek | funny, I thought that worked without SRV |
17:04.01 | `Sauron | hum, I see. |
17:04.07 | RoyK | fsck |
17:04.13 | RoyK | voip-info.org is down again |
17:04.14 | Zeeek | tsk, tsk |
17:05.02 | Beirdo | it's making RTFM'ing harder :) |
17:05.05 | `Sauron | inspired: If I have [default] with include => from-sip |
17:05.25 | `Sauron | and [from-sip] does a bunch of exten => s,1,.... etcetc |
17:05.26 | Silik0n | dont you mean RTFW'ing harder ;) |
17:05.27 | `Sauron | that should work |
17:05.32 | ManxPower | This is funny: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=28048&item=5752360720&rd=1&ssPageName=WDVW |
17:05.50 | Beirdo | Silik0n: yeah, I kinda consider the W part of the M :) |
17:05.57 | ManxPower | Notice the half naked women in the listing pics, no aparent reason for her to be there, just to make make geeks look at the ad. |
17:06.41 | Silik0n | ManxPower yeah but i bet it does wonder for their sales |
17:06.42 | RoyK | ManxPower: hehe |
17:06.52 | RoyK | ManxPower: press 'play' below and you get the shots |
17:07.10 | *** join/#asterisk salmandr (~salmandr@66-188-101-214.mad.wi.charter.com) |
17:07.15 | Beirdo | maybe she comes with the PBX? |
17:07.17 | Silik0n | and do you really wanna buy a MICS anyway? |
17:07.34 | ManxPower | Silik0n, No, I want a PRI card for the existing MICS |
17:07.47 | Zeeek | http://web.archive.org/web/20040220020259/www.voipinfo.org/tiki-index.php |
17:08.21 | RoyK | Zeeek: what's that? |
17:08.41 | RoyK | just a copy? |
17:08.46 | Zeeek | cache |
17:08.57 | RoyK | k |
17:09.04 | RoyK | who's running the wiki? |
17:09.14 | Zeeek | the wikimasters |
17:09.17 | RoyK | s/s r/s supposed to be r/ |
17:09.36 | Zeeek | I think they had to shut down to add another 16megs of RAM |
17:10.02 | RoyK | lol |
17:10.04 | Zeeek | that chaches version is from Feb 11 |
17:10.08 | Zeeek | cache |
17:10.14 | Silik0n | their 486 dying under the load? |
17:10.29 | Zeeek | I mean it's not like NEW! H323 plugins |
17:10.47 | *** part/#asterisk maksim (~max@213.142.207.2) |
17:11.01 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
17:12.44 | Silik0n | ok anyone know anything about porting drivers to OSX? |
17:12.44 | Zeeek | 1:36 to go on 400megs |
17:12.57 | Silik0n | Zeeek whi |
17:13.07 | Silik0n | what movie you downloading? |
17:13.15 | Zeeek | saxophone lessaons |
17:13.20 | Silik0n | heh |
17:13.21 | Zeeek | and spelling lessons |
17:13.33 | Silik0n | send me the spelling lessons |
17:13.34 | RoyK | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=5752356642&ssPageName=MERC_VI_RSCC_Pr4_PcN__Stores |
17:13.46 | RoyK | seems to come with something similar as the other |
17:13.58 | Zeeek | if you build it they will come |
17:14.05 | Zeeek | 8 seconds |
17:14.27 | Zeeek | BAD CHECKSUM PLEASE BEGIN DOWNLOAD AGAIN!!!! |
17:14.30 | salmandr | i'll buy if that girl will deliver it :) |
17:15.11 | `Sauron | Grf. |
17:15.25 | `Sauron | So I set myself up in the address book on x-lite, with user@domain.com |
17:15.34 | `Sauron | and when I go to dial, it wants me to configure a default SIP proxy |
17:15.43 | `Sauron | but the point is not to have to set one up. Grf. |
17:16.30 | Beirdo | salmandr: she's likely even higher maintenence than the PBX |
17:16.41 | Zeeek | heh |
17:22.03 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
17:23.24 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
17:23.39 | `Sauron | X-Lite blows |
17:27.48 | *** join/#asterisk shayne (~shaynebat@ip68-100-97-241.dc.dc.cox.net) |
17:27.59 | shayne | ? |
17:28.35 | shayne | has anyone tried configuring asterisk on os x ? |
17:29.27 | Silik0n | you can build it on OSX and it runs just fine... |
17:29.35 | Silik0n | however there are no TDM drivers for OSX at this time |
17:29.44 | Nugget | yeah, I use asterisk on my powerbook for when I travel. |
17:29.47 | *** join/#asterisk dudewhere (~ashly@adsl-68-72-128-234.dsl.chcgil.ameritech.net) |
17:30.00 | Nugget | since sip is a pain in the ass through hotel nat hell, I use x-lite to a local asterisk and then iax to my main server. |
17:30.03 | *** join/#asterisk abombss (~abombss@c-67-175-115-51.client.comcast.net) |
17:30.04 | Silik0n | i have it compiled on my G4 |
17:30.37 | shayne | Thanks...I'm a complete newbie to this...have setup asterisk from sunrise-tel.com and wanting to make it work with a sipura 3000 |
17:30.48 | shayne | with FWD |
17:30.50 | Nugget | that should work just fine in os x. |
17:31.10 | dudewhere | Anyone know where I can find a list of packages need to install * FC2 I want the OS to bare bones, but I know I need NCurses, SSH, Bison and I'm not sure what else. |
17:31.18 | shayne | cant seem to find much documentation on configuring since I'm not a linux techo |
17:33.08 | *** join/#asterisk o-m-a-o-m-a (unknown@80.81.19.75) |
17:33.41 | o-m-a-o-m-a | Good evening |
17:34.18 | trym | good morning |
17:34.58 | o-m-a-o-m-a | I need some kind of hint. What means "chan_zap.c:7411 zt_pri_error: PRI: !! Got S-frame while link down |
17:35.06 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
17:35.14 | RoyK | inspired: ping |
17:36.28 | RoyK | ~seen inspired |
17:36.30 | jbot | inspired is currently on #asterisk (9h 5m 33s). Has said a total of 24 messages. Is idling for 39m 44s |
17:37.19 | Mw3 | ~seen P-Chan |
17:37.20 | jbot | p-chan <~pchan@68.142.66.200> was last seen on IRC in channel #asterisk, 6d 52m 8s ago, saying: 'oh'. |
17:44.21 | *** join/#asterisk santiago (~santiago@63.245.86.121) |
17:48.24 | *** join/#asterisk sysdef (~sysdef@pD9560EB9.dip.t-dialin.net) |
17:48.41 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
17:51.28 | Mw3 | hm, mohmp3s are gone from asterisk debian packages in sid :( |
17:54.21 | RoyK | don't use packages |
17:55.14 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
17:57.06 | *** join/#asterisk lyroy (~lyroy@modemcable117.123-202-24.mc.videotron.ca) |
17:57.56 | *** part/#asterisk santiago (~santiago@63.245.86.121) |
18:02.39 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
18:02.39 | *** mode/#asterisk [+o bkw_] by ChanServ |
18:04.05 | dan2 | could someone verify for me if there are issues on broadvoice dca? |
18:05.26 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
18:08.27 | bjohnson | dudewhere: there is some fedora info on the wiki |
18:08.31 | bjohnson | ~docs |
18:08.32 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
18:08.43 | bjohnson | always a good place to start looking for info |
18:09.04 | bjohnson | unfortunately, down today |
18:09.27 | Silik0n | yeah someone needs to whoever runs is and say "y0 WTF the wiki is down" |
18:10.13 | RoyK | who /is/ running the wiki? |
18:12.25 | eKo1 | Is the wiki down? |
18:12.45 | eKo1 | I can't seem access it. |
18:12.50 | Silik0n | what ever gave you that idea |
18:12.59 | RoyK | :) |
18:18.07 | *** join/#asterisk jhiver (~jhiver@ABoulogne-102-1-3-10.w193-253.abo.wanadoo.fr) |
18:18.14 | jhiver | hi lads |
18:18.23 | bkw_ | HEY HEY HEY |
18:18.28 | jhiver | silly question: how can you debug a Perl AGI script? |
18:18.38 | bkw_ | print to STDERR |
18:18.43 | bkw_ | or exec verbose |
18:18.52 | jhiver | OK so I can just use warn |
18:19.02 | jhiver | will it appear on the CLI? |
18:19.10 | bkw_ | use verbose then |
18:19.16 | bkw_ | but printing to STDERR is better |
18:19.28 | jhiver | ok |
18:19.43 | *** join/#asterisk ranliv (~ranliv@203.172.11.239) |
18:19.45 | jhiver | I'm tying together the stupidest script ever but it should do the work... |
18:20.11 | jhiver | basically using non-crap demo voice synthetisers rather than festival... horrible horrible stuff |
18:20.27 | jhiver | but it'll be better than mr. robocop talking :) |
18:21.00 | Mw3 | has anyone managed to get this work with capi driver: "Eicon Networks Corporation Diva 2.01 S/T PCI" ? |
18:21.21 | jhiver | it's not a server card |
18:21.25 | jhiver | I have the same at home |
18:21.33 | jhiver | won't work me thinks |
18:21.55 | Mw3 | ah :( |
18:21.56 | Mw3 | damn it |
18:22.05 | jhiver | too bad... cheap card... |
18:22.23 | Mw3 | works with i4l modem driver but that's not the best channel driver :) |
18:23.38 | ranliv | hello guys, I need help! My asterisk box is located behind a traditional pbx and I need to dial 9 first before dialing the destination number. how can i dial 9, put a 1 sec delay and dial the destination number from my Zap channel? |
18:24.27 | *** join/#asterisk lattice (~lattice@S010600045ad57bb6.vc.shawcable.net) |
18:24.42 | jhiver | Don't know about the 1 second delay but do you really need that delay? |
18:25.01 | ranliv | how do I do this? |
18:25.13 | jhiver | Otherwise it would be like Dial(9${EXTEN}) |
18:25.34 | bkw_ | is it Analog? |
18:25.49 | bkw_ | Dial(Zap/g1/9w${EXTEN}) |
18:25.58 | o-m-a-o-m-a | some hardware PBX need some time to catch the free line sign |
18:26.29 | o-m-a-o-m-a | w like in the good old modem times :-) |
18:26.56 | ranliv | what do w in 9w stands for wait? |
18:27.06 | bkw_ | dials 9 then waits |
18:27.06 | bkw_ | duh |
18:28.36 | *** join/#asterisk Rick_Hunter (~rhunter@04-158.008.popsite.net) |
18:29.25 | RoyK | ~seen inspired |
18:29.26 | jbot | inspired is currently on #asterisk (9h 58m 29s). Has said a total of 24 messages. Is idling for 1h 32m 40s |
18:29.54 | RoyK | Feb 19 19:29:42 WARNING[24545]: channel.c:1555 ast_prod: Prodding channel 'SIP/1001749-49a0' failed |
18:29.58 | RoyK | wtf does that mean? |
18:30.10 | bkw_ | prodding channel failed |
18:30.18 | bkw_ | look in channel.c line 1555 to see what it was doing |
18:30.30 | bkw_ | it prints line numbers for a reason :P |
18:33.02 | hermie | how do you send ps/ali over a PRI? Is there some part of the IE that has the unique station identifier? |
18:39.29 | ranliv | guys thank you very much! It worked already |
18:48.15 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
18:48.57 | [cc]smart | i'm using ztdummy on kernel 2.6, but having stutter in MOH et al... |
18:49.11 | [cc]smart | Opened pseudo zap interface, measuring accuracy... |
18:49.11 | [cc]smart | 97.558594% 97.460938% 97.460938% ... |
18:49.30 | [cc]smart | somebody has an idea what to do ? |
18:49.57 | [cc]smart | SMP system BTW |
18:50.55 | *** join/#asterisk Nivex (kjotte@user-0ce2jqe.cable.mindspring.com) |
18:51.17 | ManxPower | [cc]smart, Find a way to get this numbers to 99.7 or better. |
18:51.42 | ManxPower | Search the mailing list archives for HDLC Abort and you'll find suggestions for improving interrupt latency. |
18:51.45 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
18:54.12 | *** join/#asterisk delchi (delchi@amanda.dorsai.org) |
18:54.24 | delchi | mornin' |
18:54.39 | delchi | ( well in this part of the world at least ) |
18:55.48 | delchi | can anyone point me in the direction of information as to wether or not the el cheapo Linksys / other VOIP adapters are compadible with * ? |
18:56.08 | delchi | ( they coem bundled with vonage and/or other services, I was just wondering if I could hack one togeter to work with * ) |
18:56.24 | ManxPower | ~docs |
18:56.25 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
18:56.31 | Sedorox | they are locked to Vondage |
18:56.34 | Sedorox | vonage |
18:56.36 | delchi | thats what I needed to know |
18:56.38 | delchi | thanks |
18:56.45 | ManxPower | delchi, Those people lock their stuff up pretty tight. |
18:56.51 | delchi | yeah I suspected as much |
18:56.53 | ManxPower | Vonage will UNLOCK their box, but there are restrictions. |
18:56.59 | Sedorox | the linksys's you can't even buy w/o being a VOIP Provider |
18:57.01 | ManxPower | and it's not free. |
18:57.11 | delchi | Hm |
18:57.15 | JerJer | grrr |
18:57.21 | JerJer | they are not locked to vonage, just pre-configured |
18:57.22 | delchi | Ive seen a pile of carious VOIP > POTS adapters on the shelves |
18:57.33 | ManxPower | The PAP-NA is pretty much the same as the SIPura SPA-2k |
18:57.34 | file[laptop] | they become lock when they get personal with vonage |
18:57.36 | Sedorox | JerJer: I thought they were locked... hmmm |
18:57.43 | file[laptop] | they 'get jiggy with it' |
18:57.50 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
18:57.51 | JerJer | ManxPower: they 'are' a 2k |
18:57.56 | JerJer | just with linksys plastic |
18:58.01 | JerJer | and blue LEDs |
18:58.03 | Sedorox | lol |
18:58.04 | ManxPower | soundguy, "pretty much" |
18:58.07 | delchi | so they are not locked |
18:58.08 | delchi | Hm |
18:58.11 | *** join/#asterisk Derkommissar (~Derkkommi@66.100.55.66) |
18:58.13 | Derkommissar | Hello |
18:58.13 | ManxPower | ..er... So "pretty mcuh" |
18:58.18 | Derkommissar | i have a small question, |
18:58.21 | file[laptop] | oh, that reminds me |
18:58.25 | Derkommissar | why doesnt asterisk sends a=fmtp:18 annexb=no |
18:58.35 | Derkommissar | at the end of an invite when using g729a |
18:58.45 | *** join/#asterisk Darwin35 (~Darin@c-24-3-241-22.client.comcast.net) |
18:58.57 | ManxPower | Derkommissar, I THINK annexb is silence supressions / Vad |
18:59.10 | Darwin35 | hey manx |
18:59.22 | ManxPower | Hello, Darwin35 |
18:59.27 | Darwin35 | so what did I miss |
18:59.29 | delchi | basically what I need is something like the S100I |
18:59.43 | Derkommissar | ManxPower yes anexb is silence supprecion |
18:59.45 | ManxPower | delchi, Just buy a SIPura. |
18:59.48 | *** join/#asterisk WizzKid (~apryer@cpc3-lutn5-3-0-cust169.lutn.cable.ntl.com) |
19:00.03 | ManxPower | Darwin35, Too many nose rings? |
19:00.13 | Derkommissar | but without that line, the other party belives that the codec been used is g729ab istead of g729a |
19:00.14 | Darwin35 | no birth defect |
19:00.16 | Derkommissar | :-/ |
19:00.23 | Darwin35 | my rings are else where |
19:00.25 | delchi | ManxPower : perfect. |
19:00.32 | ManxPower | Derkommissar, So? They are functionally compatable |
19:00.37 | delchi | Thats pretty much what Im looking for. |
19:00.49 | Derkommissar | Yes |
19:00.51 | delchi | I was just hoping I coudl zip dowen to the store and buy one today, and not have to snail mail order it |
19:01.11 | delchi | Oh that sounds like fun |
19:01.16 | Derkommissar | but since the extra frames of Silence suprecion of the RTP is dropped, the quality comes out to be choppy |
19:01.21 | delchi | at least use a good car |
19:01.53 | ManxPower | Derkommissar, Then the far side is not honouring the request for no VAD |
19:02.16 | Derkommissar | Correct |
19:02.54 | Derkommissar | and its because they base themself to do audio based on the invite |
19:03.55 | Derkommissar | the rfc3555 says that its a standart to put wheather annexb= yes or no |
19:06.37 | Derkommissar | are we disconected ? |
19:06.43 | delchi | damn the spa-2000 is cheaper than the iaxy |
19:07.52 | Derkommissar | can we use g729ab with asterisk ? |
19:14.51 | RoyK | BREW pot HTCPCP/1.0 |
19:21.12 | Essobi | Mehe |
19:21.29 | Essobi | Quit talking to your coffee pot |
19:23.16 | Sedorox | errrr |
19:25.59 | dan2 | who is using broadvoice proxy dca here? |
19:27.45 | *** join/#asterisk file[laptop] (~file_lapt@mctn1-142166197096.nb.aliant.net) |
19:28.04 | Sedorox | ok.. all of the sudden.. when I dial 10-200, to login to a queue on a remote box.. and I press my password and press #, the local box wants to do a transfer... yet if I dial 8500, which is voicemail on another remote box, and I press #, it works fine... any clues? |
19:28.45 | JerJer | there is a T or t dial modifier |
19:28.49 | JerJer | in use |
19:28.51 | bjohnson | ManxPower: do you have spa 3ks? I have 2 probs I just can't nail down. |
19:28.59 | Sedorox | yes.. both.. |
19:28.59 | JerJer | bjohnson: i have spa 3k |
19:29.05 | JerJer | Sedorox: that is your problem |
19:29.08 | Sedorox | hmmm |
19:29.10 | bjohnson | I keep getting echo |
19:29.22 | JerJer | bjohnson: if caller*id isn't present it won't make a SIP call |
19:29.27 | JerJer | no problems with echo |
19:29.45 | JerJer | however my audio volume is a little low |
19:29.46 | Sedorox | I didn't have a problem with it till yesterday it seems... but hmmm |
19:29.56 | bjohnson | not every time, but often, a pstn call in the fxo that then goes to the fxs in the same unit or the fxs on another SPA .. get echo |
19:30.22 | JerJer | running the newest firmware? |
19:30.52 | *** join/#asterisk w0w0 (~apardo@80.26.166.71) |
19:31.00 | bjohnson | close to newest .. 2.0.11(GWg) |
19:31.15 | *** join/#asterisk bobx (~bobx@206.124.165.14) |
19:31.22 | JerJer | i would go up to the newest |
19:31.30 | bjohnson | it was newest about 3 weeks ago but I see a newer one there now |
19:31.39 | CoaxD | JerJer: Um, why would firmware version on spa-3000 matter for echo? |
19:31.48 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.rr.com) |
19:31.58 | JerJer | echo can tuned better |
19:31.59 | CoaxD | JerJer: I have spa-2000. they say not to upgrade to latest firmware unless there's a fix in the new one that specifically resolves the issue |
19:32.00 | bjohnson | perhaps a better echo cancellation routine |
19:32.00 | *** join/#asterisk welby (~welby@solas.plus.com) |
19:32.05 | CoaxD | jerjer: Ahhh. |
19:32.15 | CoaxD | bjohnson: Does your caller hear your echo or just you |
19:32.26 | bjohnson | just me |
19:32.44 | JerJer | then its near-side echo |
19:32.48 | bjohnson | <PROTECTED> |
19:32.52 | CoaxD | bjohnson: Pretty much a guarantee that your rxgain on your x100p is set too high |
19:33.01 | JerJer | its not an X100P |
19:33.02 | CoaxD | (or whatever your fxo is) |
19:33.07 | bjohnson | JerJer: yes .. but just on pstn calls. Calls coming voip are ok |
19:33.21 | bjohnson | it's a spa 3k |
19:33.35 | bjohnson | SPA to PSTN and PSTN to SPA gains were set to 0 |
19:33.41 | CoaxD | bjohnson: Yes, but your fxo doesnt connect to your spa, right? |
19:33.49 | CoaxD | (I forget what the difference between spa2k and spa3k is.) |
19:33.54 | bjohnson | it's happening on two of 3 spa 3ks I have |
19:33.57 | JerJer | FXO |
19:33.59 | Qwell | 3k has the fxo port, right? |
19:34.02 | CoaxD | ah |
19:34.04 | bjohnson | yes |
19:34.09 | CoaxD | prolly need to drop the rxgain somehow on fxo |
19:34.27 | JerJer | you can do it in the advanced config mode |
19:34.48 | CoaxD | jerjer: yeah i figured. prolly needs the new firmware to finely tune it, too |
19:34.48 | JerJer | under pstn tab, i think |
19:34.57 | CoaxD | i only have sp-2k's |
19:35.00 | Qwell | the spa3k has what, 2fxs, 1fxo, and an ethernet port? |
19:35.05 | CoaxD | no need for fxo on pstn thru sip adapter |
19:35.19 | bjohnson | yeah .. I dropped to -2 .. didn't want to go too low. Found some discussion on the web about line impedence being a possible echo source but it was all about UK lines |
19:35.22 | JerJer | now only if the spa 3ks talked IAX |
19:35.27 | bjohnson | Qwell: one of each |
19:35.29 | Qwell | ahh |
19:35.36 | Qwell | What do those generally run? |
19:35.38 | CoaxD | JerJer: Dont give up yet :) |
19:35.43 | JerJer | bjohnson: ahh yes!!! there is an option to change that |
19:35.49 | bjohnson | Qwell: $100 at voxilla |
19:36.03 | Qwell | I should have gotten one of those instead, heh |
19:36.09 | bjohnson | yes .. but all info I could find just listed 2 possibles for N.A. |
19:36.17 | CoaxD | bjohnson: Would you beleive that on one of my incoming POTS lines, on an X100P, i have to drop the gain to -7.5? |
19:36.26 | CoaxD | bjohnson: I am 200 feet from the telco. |
19:36.33 | CoaxD | bjohnson: And cabling to them that is less than 10 years old |
19:36.47 | Qwell | CoaxD: increase the loop length, heh |
19:37.01 | Qwell | just add like 1500 feet of cable in your walls :p |
19:37.04 | bjohnson | 600 or 900 are supposed to be NA standard and I have it set to factory default of 600 |
19:37.05 | CoaxD | qwell: Just think about the 10 miles of wrapped up telco wire in the server room. *g* |
19:37.11 | dan2 | I've got 'em all, sipura 1000,1001,2000,2100,3000 |
19:37.26 | CoaxD | dan2: Their ip phones dont look half bad, either |
19:37.26 | bjohnson | anyway I can check line impedance or should I just try the 900 option? |
19:37.47 | dan2 | CoaxD: heh, I get my voip stuff for free |
19:37.50 | CoaxD | bjohnson: There are ways to check line impedance. all requires the competent phone tech to test both ends. :P |
19:37.59 | Qwell | dan2: hook me up with a 7940 :p |
19:38.08 | CoaxD | bjohnson: Rest assured, if you can put a regular phone on it and it dont echo, your line is gonna be fine. :P |
19:38.14 | dan2 | Qwell: it sits on my desk |
19:38.26 | dan2 | Qwell: oh, this is 7960 |
19:38.50 | CoaxD | bjohnson: You can be almost 100% sure that your line is fine, and it is just your rxgain set too high |
19:38.56 | Qwell | Well, you get free hardware, right? Get a 7940, and send it on over. ;] |
19:39.01 | CoaxD | bjohnson: You might even notice callerid dont work right either |
19:39.15 | CoaxD | bjohnson: (or maybe only a portion of the time) |
19:39.16 | dan2 | Qwell: I'm waiting for the panasonic and uniden cordless phones to arrive |
19:42.13 | bjohnson | callerid is only a problem for the one that sits behind a fax/data/phone auto switch |
19:42.30 | bjohnson | so I guess I play with line impedence and gain until I get it to work |
19:42.39 | CoaxD | bjohn: yeah. just keep subtracting |
19:42.44 | CoaxD | bjohn: Your echo will go away. |
19:43.18 | bjohnson | if I switch line impedence to 900 and that isn't the correct one .. should I have consistant problems? |
19:43.31 | CoaxD | quit mucking with line impedance! |
19:43.38 | CoaxD | I dont know why you think its a line impedance problem |
19:43.43 | CoaxD | it has zero to do with line impedance :P |
19:43.47 | bjohnson | CoaxD: gain on the SPA to PSTN setting or the PSTN to SPA? I think SPA to PSTN. |
19:44.04 | CoaxD | bjohnson: pstn to spa, of course :P |
19:44.05 | bjohnson | CoaxD: just from googling other people trying to solve echo problems |
19:44.15 | CoaxD | bjohn: This is the difference between rxgain and txgain |
19:44.20 | CoaxD | rxgain would be 'pstn to spa' |
19:44.29 | CoaxD | txgain would be 'spa to pstn' |
19:44.38 | CoaxD | you dont need to transmit softer. you need to receive softer. :) |
19:46.01 | CoaxD | and if it gets too quiet, you can increase the txgain to your headset |
19:46.09 | CoaxD | i.e. your fxs |
19:46.23 | CoaxD | (Those are two different sets of settings.) |
19:46.50 | bjohnson | I don't think these units have fxs gain control |
19:49.09 | bjohnson | the other problem I have, is that the fxs ports are connected to the line in connectors to a Nortel CICS (I don't think this has anything to do with the echo). When a Nortel handset is hung up from a pstn call .. the line often rings back and noone is there. From the logs I can see it is actually the fxo port calling back into the system (ie it isn't hanging up fast enough). Any ideas on this one? |
19:49.36 | PatrickDK | they have gain control, it's default is -3db |
19:50.12 | CoaxD | bjohnson: I have nothing to offer on that issue. :/ |
19:51.03 | *** join/#asterisk vs_ (~vs@host-175.voip-gw.dial-pool.macomnet.net) |
19:52.00 | hermie | fearnor: nice drumroll on -dev :) |
19:52.17 | vs_ | howdy |
19:52.37 | bjohnson | btw, any technical or licensing reason why sipura couldn't make these support iax? |
19:53.08 | Qwell | lazyness, heh |
19:53.16 | hermie | bjohnson: they're fine as long as they do a clean-room implementation |
19:53.36 | hermie | bjohnson: which is hard because IAX is a moving target and not very well documented in writing |
19:53.51 | hermie | bjohnson: which we're trying to change over in ADP-land |
19:53.52 | bjohnson | well, I imagine they won't do it unless they think it will make enough increase in sales to justify the dev cost |
19:53.57 | ManxPower | IAX isn't really the problem, lack of docs for writing firmware for SIPuras is the problem. |
19:54.13 | o-m-a-o-m-a | I need some kind of hint. What means "chan_zap.c:7411 zt_pri_error: PRI: !! Got S-frame while link down" on HFC-S / ISDN? |
19:54.28 | hermie | what't with asking the mailing list if something's down? |
19:55.20 | hermie | like the wiki |
19:55.22 | hermie | and broadvoice |
19:57.09 | CoaxD | kinda lame, yeah, hermie |
19:57.24 | PatrickDK | bjohnson, you have to remember, only asterisk supports iax, and lots of people support sip |
19:57.53 | PatrickDK | and to produce a product with limited people that would be interested in it, taks time and money |
19:58.28 | *** join/#asterisk rodizump (~chatzilla@dsl-213-023-226-078.arcor-ip.net) |
19:58.46 | *** join/#asterisk Frantic (~ab@24-193-46-85.nyc.rr.com) |
19:58.52 | *** join/#asterisk Nukemizer (~Nuke@65.103.231.133) |
19:58.56 | *** join/#asterisk cjk (~cjk@80.92.75.91) |
19:59.08 | elric | do you reckon IAX will ever be made an IETF standard? |
19:59.10 | PatrickDK | if you could prove to sipura, that 50% of sipura users have asterisk, then they will probably consider supporting iax |
19:59.19 | PatrickDK | but otherwise you will probably have a tough time of it |
19:59.24 | rodizump | hi everyone, is the www.voip-info.org really down or my ISP can't route there ? can anybody confirm ? please |
19:59.38 | cjk | rodizump, i confirm and i really need to access the site |
19:59.47 | Frantic | <rodizump> confirmed |
19:59.57 | shido6 | i couldnt get to them lastnight either |
20:00.06 | PatrickDK | why can't people really believe that public access, none supported sites go down sometimes |
20:01.23 | Sedorox | ok... |
20:01.33 | Sedorox | what would cause Asterisk to lock up three totaly different boxes on stop now? |
20:01.51 | shido6 | same broken code on all three? |
20:02.06 | Sedorox | two run 1.0.3 one runs 10.0.5 |
20:02.08 | Sedorox | 1.0.5 |
20:04.07 | Sedorox | reverse that... |
20:04.13 | *** join/#asterisk darby_t (~tom@dol122.neoplus.adsl.tpnet.pl) |
20:05.11 | Sedorox | waiting on pastebin |
20:05.58 | Sedorox | http://www.pastebin.com/243834 |
20:06.05 | bjohnson | PatrickDK: where is the gain control on a spa 3k fxs? I can't find it |
20:06.10 | eKo1 | Has anyone had a problem with certain SIP entries in sip.conf having an accountcode not being logged properly in the CDRs? |
20:07.11 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
20:10.32 | Sedorox | anyone have any clues? |
20:10.43 | Nukemizer | I am looking for a Digium T1 card MASTERto hire .. second card card from digium still will not load properly and still get alarms . Any takers ? |
20:11.16 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-21-45.d4.club-internet.fr) |
20:15.59 | PoWeRKiLL | hi :) |
20:18.07 | bjohnson | well .. this is a nice tip about UPS shipments to Canada: Enjoy free customs clearance on UPS Express and UPS Expedited shipments |
20:20.32 | PatrickDK | ls |
20:21.34 | BoRiS | I wish Canada Post would ship on saturday |
20:21.56 | `Sauron | Anyone here with a working softphone? |
20:22.00 | Sedorox | nayone know what would cause theses lock ups? http://www.pastebin.com/243834 |
20:24.13 | ManxPower | Sedorox, You should be able to ignore those since you are shutting down Asterisk |
20:24.50 | ManxPower | "Yuck! Error in buffer handling...: Broken pipe " is an error from mpg123 |
20:25.23 | *** join/#asterisk kippi (chrisfrog@cpc4-hatf3-6-0-cust243.lutn.cable.ntl.com) |
20:26.07 | kippi | hey is voip-info.org down? |
20:26.12 | vs_ | hehe |
20:26.13 | vs_ | ye |
20:26.18 | kippi | damm |
20:26.36 | kippi | anyone used asterisk@home ? |
20:26.48 | ManxPower | Lots of people, but we don't talk to them. |
20:27.04 | kippi | how comes? it not worth using? |
20:27.05 | *** join/#asterisk NetworkStorm (~afernande@adsl-69-108-160-246.dsl.skt2ca.pacbell.net) |
20:27.13 | NetworkStorm | Hello |
20:28.32 | jetscreamer | is asterisk@home different from asterisk? |
20:28.43 | jetscreamer | or is that just where you use it |
20:28.44 | bjohnson | no |
20:28.47 | bjohnson | just uses amp |
20:28.48 | jetscreamer | o |
20:28.54 | Sedorox | ManxPower: the thing is.. after all that.. it crashed all the boxes |
20:28.55 | bjohnson | as a gui |
20:29.10 | bjohnson | and auto-installs asterisk and linux from cd |
20:29.22 | jetscreamer | ah the livecd |
20:29.24 | kippi | can you use linux from the bash? |
20:29.35 | jetscreamer | yes...? |
20:29.36 | NetworkStorm | I am having problems with asterisk not recognizing my TDM400P card. |
20:29.42 | NetworkStorm | Anymore |
20:29.48 | bjohnson | I think bkw_ has a different version of the same concept .. a prepared distro for asterisk server setup |
20:29.50 | NetworkStorm | Can someone help me? |
20:30.08 | shido6 | uhh |
20:30.09 | shido6 | yeah |
20:30.13 | shido6 | I can help ewe |
20:30.17 | NetworkStorm | Cool |
20:30.18 | bjohnson | jetscreamer: I don't think it will run from cd .. just auto install. but i haven't used it so might be wrong |
20:30.21 | kippi | so would poeple say to keep away from it? |
20:30.30 | NetworkStorm | I worked last night with no problems |
20:30.41 | jetscreamer | linux from bash. yes. learn concepts though. |
20:30.46 | NetworkStorm | I turned it off, and this morning asterisk would not start |
20:31.00 | bjohnson | kippi: don't worry about it. it's just hard for use to help you since it makes a very complicated setup to start |
20:31.14 | jetscreamer | saying yes to that question is like fingernails on a blackboard |
20:31.49 | Nugget | "can you use linux from the bash" <-- wow. I'm beyond words. This is so wrong on so many levels. |
20:32.02 | jetscreamer | but the answer he seeks is yes. |
20:32.07 | jetscreamer | jsut bad q. |
20:32.15 | kippi | so would you say I would be better of installing slackware and then installing asterisk? |
20:32.19 | bjohnson | I think he means access the asterisk cli from a bash prompt |
20:32.35 | bjohnson | kippi: define "better off" |
20:33.12 | kippi | is the only con to installing asterisk@home is that it uses the gui ? |
20:33.18 | bjohnson | if you can install ast@home and it does everything you want right away, then that might be best for you |
20:33.43 | bjohnson | if it doesn't do what you want and you have to troubleshoot .. maybe format disk and start with a typical install |
20:33.53 | kippi | hmm |
20:34.21 | bjohnson | no sense troubleshooting their system .. usually easier to start fresh in that case |
20:34.23 | rodizump | can anyone tell how to make asterisk send BYE after RTP timeout, ie. remote side hangs up and asterisk sends bye to originator ? anyone ? |
20:34.27 | jetscreamer | does this @home install on a preexisting linux install or does it need the 'distro' it comes with |
20:34.35 | eKo1 | Man...VoIP over 802.11b sucks orangutan nipples. |
20:34.45 | bjohnson | read it's faq .. I am not it's developer nor user |
20:34.54 | jetscreamer | k thx |
20:35.05 | bjohnson | eKo1: depends who you ask .. vaewynAFK loves it |
20:35.21 | eKo1 | Well, for short distances it should be no problem. |
20:35.41 | eKo1 | But after 5 miles, well...it stinks. |
20:36.47 | eKo1 | Then the customers complain about bad voice quality and shitty web surfing and blah, blah, blah.... |
20:36.57 | PatrickDK | hmm, voip over 802.11b works good for me |
20:37.15 | PatrickDK | I have only run it up to 1mile, with a voip phone |
20:37.30 | eKo1 | I guess there is too much interference. |
20:37.59 | *** join/#asterisk yashax (~yasha_x@69.15.218.218) |
20:38.02 | kippi | so then you are better of with a clean install and build the system up ur self |
20:38.04 | kippi | ? |
20:38.10 | rodizump | Does anyone know how to set RTP timeout in asterisk ? |
20:39.19 | NetworkStorm | ' |
20:39.24 | NetworkStorm | How can I verify that WBEL see my TDM400P card? |
20:39.52 | Moc | WBEL ? |
20:40.01 | `Sauron | Hum, so that worked, at least partially. |
20:40.16 | NetworkStorm | Whitebox linux |
20:40.26 | NetworkStorm | Its basically RHEL |
20:41.41 | NetworkStorm | be wctdm |
20:42.14 | vs_ | dropped calls with chan_oh323 |
20:42.20 | *** part/#asterisk eKo1 (~bernd@207.42.191.66) |
20:42.25 | vs_ | and cisco |
20:43.39 | Moc | I use TaoLinux |
20:47.04 | shido6 | Tao |
20:47.14 | shido6 | how is TaoLinux? |
20:50.57 | *** join/#asterisk file[laptop] (~file_lapt@mctn1-142166197096.nb.aliant.net) |
20:51.30 | ariel_ | NetworkStorm, zttool |
20:51.59 | *** part/#asterisk NetworkStorm (~afernande@adsl-69-108-160-246.dsl.skt2ca.pacbell.net) |
20:53.52 | GMsoft | mhh does fxs/fxo signaling is able to send the called and caller id ? |
20:55.18 | shido6 | yes |
20:55.39 | *** join/#asterisk jsolares (~jsolares@200.12.44.18) |
20:56.07 | GMsoft | so I could ask to have more than one number routed to my fxo line and match the called id and the route the call correctly ? |
20:56.28 | *** join/#asterisk zyke (~zakforeve@84.45.132.117) |
20:56.29 | *** join/#asterisk ryguillian (~ryguillia@c-24-12-96-52.client.comcast.net) |
20:58.54 | ariel_ | GMsoft, yes but you should check to see what the ${DNIS} says to make sure your provider is sending you info you can use. |
20:59.33 | GMsoft | ariel_: ok thanks. I'll ask them before subscribing :) |
21:01.06 | *** join/#asterisk zimdog (~zimdog@c-67-164-190-201.client.comcast.net) |
21:02.28 | *** join/#asterisk miguellinux (~miguellin@200.47.223.190) |
21:04.05 | Beirdo | I'll try again today :) |
21:04.24 | Beirdo | anyone know how I can tell a particular extension not to use music on hold? |
21:06.37 | Beirdo | ooh, the wiki's back. I'll try RTFW again :) |
21:06.42 | ariel_ | Beirdo, I don't understand your question? |
21:07.12 | Beirdo | I want to make it so if I call extension 502, then put it on hold that it doesn't play music on hold |
21:07.46 | Beirdo | as 502 goes to a meetme elsewhere, and if I put it on hold to answer another call, I don't need to be treating others on the conference to music :) |
21:07.51 | ariel_ | make different context for them set the variable in that context not to have moh. |
21:08.59 | Beirdo | now I'm the one not understanding :) |
21:09.10 | Beirdo | but I'll look around on the wiki a bit |
21:10.46 | `Sauron | is the wiki back up? |
21:11.09 | Beirdo | seems to be |
21:11.21 | *** join/#asterisk defian (ircuser@shakotay.alphanet.ch) |
21:12.33 | *** join/#asterisk zimdog (~zimdog@c-67-164-190-201.client.comcast.net) |
21:12.37 | defian | hello, I am quite new here. Anyone fond of chan_zap & ISDN issues ? :) |
21:13.31 | defian | (asterisk 1.0.5; one call is OK, a second call is OK; a third call comes in and is rejected (good); however after that point no second call will get answered) |
21:14.16 | roamer323 | is there any "production quality" iax2 softphones out there? similar to xten phones for sip? |
21:15.05 | defian | roamer323: I only know about the IAXy (analog-to-IAX2 adapter) |
21:15.12 | defian | ah sorry |
21:15.19 | defian | roamer323: IAXcomm works very well here |
21:15.31 | djin | firefly |
21:15.33 | defian | roamer323: both on GNU/Linux, Microsoft Windows and thrice on Mac OS X |
21:16.09 | roamer323 | djin - can firefly be configured for any IAX2 provider in and out? or is it hardwire to theirs? |
21:16.09 | djin | http://www.virbiage.com/firefly/ |
21:16.22 | defian | roamer323: with IAXcomm disable all automatic gain/filters and so on and use A-LAW or u-law and you get very good quality |
21:16.37 | djin | No, it' |
21:16.44 | defian | (only used it locally on Ethernet though) |
21:16.44 | djin | No, it's open |
21:17.39 | roamer323 | djin & defian - thanks, I'll try both of them out... the xten UI is really slick and responsive; too bad they're not talking IAX (yet) |
21:18.32 | empire667 | djin and all the others thanks for all the help, my asterisk box works great |
21:18.45 | empire667 | My compliments to you all |
21:21.47 | *** join/#asterisk jdg (~jdg@CA03F308.adsl.mana.pf) |
21:26.28 | *** join/#asterisk outsidefactor (barf@203-173-32-225.dyn.iinet.net.au) |
21:26.34 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
21:28.08 | bjohnson | couple of errors in the example here because the wiki removes the square brackets: http://www.voip-info.org/wiki-Asterisk+user+authentication |
21:28.18 | bjohnson | any idea how to show square brackets in the wiki? |
21:29.06 | `Sauron | umm |
21:29.07 | `Sauron | yeah |
21:29.13 | `Sauron | I think ~[ and ~] |
21:29.15 | `Sauron | or something |
21:29.43 | GMsoft | hehe asterisk compiled on my parisc box. it needs a little patch tho :) |
21:30.18 | *** join/#asterisk mooboi (~selfsck@silenceisdefeat.org) |
21:30.30 | mooboi | any asterisk at home Aficionado around ? |
21:30.40 | mooboi | i just finished installing an x100p fxo works fine, caller id work great too, i was just wondering what can beacomplished next, voice mail ? |
21:30.54 | defian | call transfer |
21:30.56 | defian | :) |
21:30.57 | GMsoft | conference room |
21:31.18 | *** join/#asterisk Rick_Hunter (~rhunter@06-166.008.popsite.net) |
21:31.28 | mooboi | conf room would be great feat, can it be acomplished with only one line ? |
21:31.46 | mooboi | i tought call trnasfert needed 2 pots lines |
21:32.06 | defian | no |
21:32.25 | defian | it depends :) |
21:32.30 | mooboi | defined in extensions.conf ? |
21:32.57 | defian | e.g.: call -> zap/1; then zap/1 can transfer to another extension (zap/2 or whatever) using Flash key |
21:33.36 | `Sauron | mooboi: A digium x100p, or a clone card? |
21:33.46 | mooboi | clone card, im poor ; / |
21:33.49 | defian | (in my case it's a TDM card) |
21:33.58 | `Sauron | Hum, nice. |
21:34.12 | mooboi | i would have gone with a tdm 1fxo 1fxs if it wanst for the 100$ |
21:34.16 | `Sauron | I've been thinking 'bout picking up a clone card, but people report mixed success with them |
21:34.16 | defian | analog works great, I have a few issues with ISDNs |
21:34.34 | mooboi | i guessed it like the loterry ... crossed with ebay |
21:34.57 | hermie | bjohnson: __~np~[whatever]~/np~__ |
21:35.02 | mooboi | mine seem to be fine so far , all for 12$ delivered to my door |
21:35.12 | defian | ugh |
21:35.20 | `Sauron | mooboi: That's nice. |
21:35.22 | mooboi | x100p//ebay |
21:36.00 | mooboi | now i neeed to figure out what else to do with it beside cid, i am looking for a more focused resource than voip-info.org |
21:36.17 | `Sauron | voip-info is very useful |
21:36.25 | `Sauron | just depends on what you're trying to do |
21:38.14 | `Sauron | set up a vru, set up meetme, set up voicemail |
21:38.16 | `Sauron | have fun |
21:38.23 | `Sauron | connect to FWD while you're at it |
21:38.24 | `Sauron | etc etc |
21:38.37 | mooboi | to FWD ? |
21:39.10 | `Sauron | www.freeworlddialup.com |
21:39.17 | mooboi | oh ok ok |
21:39.22 | tzanger | wow this beer is awful yeasty today |
21:39.32 | tzanger | It tastes like I'm eating sourdough |
21:39.33 | mooboi | i was thinking about buying terminaition to pstn from iax.cc later on |
21:39.42 | mooboi | tough luck |
21:39.49 | tzanger | ... which I don't particularly mind |
21:39.59 | riksta | tzanger: don't make homebrew then ;) |
21:40.03 | tzanger | this isn't |
21:40.06 | tzanger | it's Molson Ex |
21:40.12 | tzanger | my favourite brew |
21:40.38 | defian | BTW do you know if there is a way to interconnect with Skype? (just curious) |
21:40.43 | tzanger | defian: no |
21:40.48 | riksta | nop |
21:40.57 | defian | tzanger: it's still deadly proprietary? |
21:41.11 | tzanger | defian: more or less, yes |
21:41.29 | defian | tzanger: ok :) |
21:42.05 | *** join/#asterisk jtodd (~jtodd@h-67-103-42-29.snfccasy.covad.net) |
21:42.11 | *** join/#asterisk Legend (~legend@24.244.142.133) |
21:43.09 | *** join/#asterisk kFuQ (~somedude@c-24-17-224-78.client.comcast.net) |
21:46.28 | cjk | is it possible to connect with netcat to the asterisk manager so i can test a few things before coding them? |
21:47.10 | tzanger | cjk: why wouldn't it be? |
21:48.03 | cjk | tzanger, because i dont get any response when connecting to the asterisk manager and trying to do the login. im quite sure the username and password are working becaus an app not coded by me connects with this login |
21:48.41 | tzanger | cjk: hmm |
21:48.48 | tzanger | use some tcpdumpage and see what's really happenning |
21:48.49 | Mavvie | I have found out that asterisk doesn't work if the ISDN setup packet contain the exclusive dchannel bit. |
21:49.07 | Mavvie | that has caused a backout of the transfer of 800 numbers yesterday :-P |
21:49.30 | defian | Mavvie: great :) |
21:49.36 | Sedorox | hmmm |
21:49.41 | Mavvie | these two are related :-) |
21:49.48 | tzanger | Mavvie: that sounds bad |
21:50.05 | cjk | tzanger, i think im doing something wrong with nc when entering the commands, maybe you have a short example? |
21:50.53 | tzanger | echo "blah" | nc asterisk.box 5038 |
21:51.08 | defian | Mavvie: if you have experience with ISDN, did you also experience problems with call waiting (a BRI has 2 B channels used; a third call comes in; in Asterisk 1.0.5 it gets correctly ignored; however then no more than 1 call at a time are answered) |
21:51.45 | GMsoft | yay asterisk works on my hppa :) |
21:52.20 | cjk | tzanger, oh yes, but that wont give me a result |
21:52.24 | tzanger | cjk: well no ;-) |
21:52.50 | tzanger | nc is one-way |
21:53.09 | `Sauron | Nope |
21:53.10 | Mavvie | tzanger: if should write the output to stdout. |
21:53.12 | `Sauron | nc is two-way |
21:53.31 | GMsoft | mhh anyone have doc to nat correctly IAX calls ? |
21:53.39 | *** part/#asterisk djin (~djin@gridfox.xs4all.nl) |
21:53.43 | tzanger | `Sauron: not when used with | |
21:53.49 | cjk | yes nc normaly prints what it gets back |
21:53.56 | `Sauron | eh |
21:54.00 | `Sauron | I think you're on crack |
21:54.05 | Mavvie | [~] edwin@k7>echo hi | nc 0 22 |
21:54.05 | Mavvie | SSH-1.99-OpenSSH_3.8p1 |
21:54.05 | Mavvie | Protocol mismatch. |
21:54.45 | cjk | ok here is what i do: nc localhost 5038 [ENTER] Action: login[ENTER]Username: cjk[ENTER] Secret: **[ENTER] |
21:55.09 | Mavvie | cjk: you should use telnet for that. |
21:55.21 | `Sauron | :) |
21:55.38 | tzanger | `Sauron: well yeah but that's coming back to stdout |
21:55.46 | tzanger | you could write a perl script or something that worked with it I'm sure |
21:56.12 | cjk | Mavvie, ok i will try |
21:56.19 | *** join/#asterisk Nix (~Nix@81.213.125.220) |
21:56.51 | cjk | Mavvie, thanks its working |
21:57.09 | cjk | Mavvie, what is telnet sending different than netcat? |
21:57.42 | Nix | cjk: Telnet sends a buch of control chars on conntect... |
21:58.32 | Mavvie | cjk: telnet is a terminal application, netcat just sets up a tcp connection. |
21:58.51 | Mavvie | http://www.rhyshaden.com/voice.htm <- very interesting read for newbies. |
21:59.30 | cjk | Mavvie, yeah, but now when I will setup a php script, it will do only a tcp connection. nothing more |
21:59.45 | Mavvie | cjk: that's all what's needed. |
22:00.50 | Sedorox | Question.. why doesn't IAX have ipv6 support yet? or are they working on it? |
22:02.23 | ManxPower | Sedorox, Because nobody cares enough to add it. |
22:02.35 | Sedorox | hmm |
22:02.39 | Sedorox | I may look into it then... |
22:02.56 | Mavvie | Sedorox: several "I'll take it" have been shouted in the -dev mailinglist. |
22:03.02 | Sedorox | hmmm |
22:03.16 | Sedorox | but nothing has come as a result huh? |
22:03.46 | vs_ | better to have t.38 :) |
22:04.09 | Sedorox | lol |
22:05.46 | eaperezh | hi there, i have 128kbps adsl user that will connect to my 128kbps adsl asterisk, for fun....clients are IAX, should i use gsm or speex? |
22:06.29 | Zaw | are there any TDM cards that are compatible with freebsd and asterisk? |
22:07.00 | eaperezh | Zaw: sangoma if i recall correctly |
22:07.14 | Zaw | eaperezh: thanks |
22:08.10 | elric | hi how is the support for sangoma t1/e1 interfaces? i am thinking of using an embedded solution with soekris x86 boards. |
22:11.41 | kippi | hmm, been trying out asterisk@home and i see what you mean by the config is complex |
22:13.06 | mikegrb | overly complex |
22:13.11 | mikegrb | needlessly complex |
22:13.18 | mikegrb | just install asterisk like normal |
22:13.24 | mikegrb | asterisk@home breaks stuff |
22:13.42 | kippi | was trying to get the hold music to work and it was having none of it |
22:14.04 | mikegrb | and voicemail users can't change thier passcode |
22:14.10 | mikegrb | and probably loads of other stuff |
22:14.31 | mooboi | odd, all my call gets answered after 15sec and blank , nothing |
22:15.05 | kippi | mine where getting the voicemail and then the voicemail was sending me a email which was cool |
22:15.38 | cjk | Mavvie, ok then something like this should work? fputs($socket," |
22:15.39 | cjk | Action: login |
22:15.39 | cjk | Username: cjk |
22:15.39 | cjk | Secret: cjk111 |
22:15.39 | cjk | "); |
22:16.13 | Mavvie | if you google for asterisk nagios, you will find a script which connects to the manager interface. |
22:16.16 | Mavvie | perl script. |
22:16.44 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
22:17.54 | mikegrb | kippi: another fine example of asterisk@home |
22:17.55 | cjk | Mavvie, ok got it werking |
22:18.37 | sivana | ~google asterisk nagios |
22:18.54 | Mavvie | cjk: http://megaglobal.net/docs/asterisk/monitor_pbx.pl |
22:19.18 | Mavvie | it's not the best style of programming, but that's something you can overcome easily :-) |
22:20.01 | sivana | ~google 35c to F |
22:20.13 | sivana | heh |
22:29.09 | *** join/#asterisk djMax (~djMax@artsalliancelabs.com) |
22:30.34 | djMax | is there a way to tell if your wcfxs mo0dule is loaded with lowpower=1? |
22:31.32 | *** join/#asterisk mooboi (~selfsck@silenceisdefeat.org) [NETSPLIT VICTIM] |
22:31.32 | *** join/#asterisk sd-tux (user2267@emasq.stusta.mhn.de) [NETSPLIT VICTIM] |
22:33.22 | *** part/#asterisk sudoer (~sudoer@65.75.148.190) |
22:33.59 | djMax | another try, why would modprobe say it can't find wcfxs, but lsmod show it? |
22:35.29 | Corydon76-home | Might it actually say "No such device"? |
22:35.39 | *** join/#asterisk luke-jr_ (~luke-jr@207.192.221.172) |
22:36.01 | djMax | looks like the name is now wctdm |
22:36.04 | mooboi | WARNING[1367]: Channel 'Zap/1-1' sent into invalid extension 's' in context 'group-all', but no invalid handler |
22:36.10 | mooboi | what am i missing here ^ |
22:36.21 | defian | [group-all] |
22:36.33 | Corydon76-home | You're missing an s extension, perhaps? |
22:36.38 | *** join/#asterisk chetan (freetibet@24-193-188-21.nyc.rr.com) |
22:36.40 | defian | extension => s,1,Dial(demo,s,1) |
22:36.40 | mooboi | indeed |
22:36.42 | defian | for example :) |
22:36.52 | |Vulture| | defian: under [group-all] ? |
22:36.53 | defian | s/DIal/Goto/ |
22:37.05 | mooboi | damn, dinertime |
22:37.10 | mooboi | bbl |
22:37.22 | defian | |Vulture|: I don't know what is group-all |
22:37.39 | defian | |Vulture|: do you? |
22:37.40 | |Vulture| | seems to be the context he is feeding zaptel to |
22:38.02 | defian | |Vulture|: if those are internal lines, demo could be useful for testing |
22:38.10 | defian | |Vulture|: if those are external lines, well, it depends |
22:38.50 | defian | |Vulture|: can be unsafe |
22:38.50 | |Vulture| | looks like hes trying to call in, and he doesnt have a s,1, or he doesnt even have a [group-all] context |
22:39.14 | |Vulture| | yea but hes just testing so it shouldn't be a problem |
22:39.36 | defian | |Vulture|: yes. But remember that test config have the usual bad habits to stay |
22:39.55 | defian | |Vulture|: I was dialing at a friend's PBX from outside and I hit 7 instead of 8 for some reason and I got the demo :) |
22:42.07 | djMax | ideas? WARNING[731]: loader.c:509 load_modules: Loading module chan_h323.so failed! |
22:44.06 | djMax | or, alternatively, how can I disable this load attempt from h323 |
22:44.18 | |Vulture| | haha nice |
22:44.31 | |Vulture| | modules? |
22:44.35 | defian | djMax: noload => chan_h323.so in modules.conf |
22:45.09 | |Vulture| | defian: I breakup everything via contexts and have an admin context that allows me to use all the tests/demos |
22:45.42 | defian | good concept |
22:45.43 | djMax | whew. thanks, back up now. |
22:46.53 | |Vulture| | h323 is evil lol |
22:48.00 | vs_ | no shit |
22:48.04 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-21-45.d4.club-internet.fr) |
22:48.05 | zimdog | anyone setup NuFone with AMP? |
22:48.21 | vs_ | getting Payload type mismatch: expected PCMA, got CiscoCN. Ignoring packet. |
22:49.14 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
22:50.11 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
22:52.00 | *** join/#asterisk SuperMMan (~graphic@d209-89-191-155.abhsia.telus.net) |
22:52.30 | SuperMMan | question i`m trying to build a module for asterisk and i keep getting a core dump, can anyone recommend a way to find out what the core dump? |
22:52.51 | defian | you get the core dump when compiling the module? |
22:53.05 | SuperMMan | defian: no its when i go to use the module. |
22:53.12 | SuperMMan | and make a call |
22:53.18 | defian | ahem |
22:53.49 | SuperMMan | all i`m getting is a core dump, I would like to figure out why the core dump, but at this point i`m not getting enough information |
22:54.41 | defian | I have never debugged asterisk yet |
22:54.58 | defian | however gdb asterisk core might give some info (where command) |
22:54.59 | PoWeRKiLL | SuperMMan use gdb |
22:55.08 | file[laptop] | get a backtrace |
22:55.14 | file[laptop] | it'll tell you where/what is crashing it |
22:55.44 | SuperMMan | file[laptop]: thank you. |
22:56.57 | PoWeRKiLL | anyone have a good date for getting a ~stable CVS version I can't use stable cause I need voicemail feature database |
22:58.07 | djMax | what do you have to do to get rxgain to take effect, just restart asterisk? |
22:58.32 | |Vulture| | djMax: that will work |
22:58.47 | djMax | put the rxgain up to 6 but ztmonitor still not really budging |
22:59.20 | |Vulture| | do you have "echotraining=yes" |
22:59.24 | djMax | yes |
22:59.29 | |Vulture| | hmm strange |
22:59.41 | djMax | I can set rxgain per channel right? |
22:59.45 | |Vulture| | yup |
22:59.53 | |Vulture| | just have multiple entries |
23:00.03 | djMax | ok, 20dB here goes. |
23:00.44 | djMax | wow, now that's some serious echo. :) |
23:00.48 | SuperMMan | file[laptop]: do you know where i can get a copy of backtrace from? |
23:01.14 | file[laptop] | what you do is... |
23:01.18 | file[laptop] | well, use gdb |
23:01.23 | file[laptop] | to open the core file... then type "bt" |
23:01.27 | file[laptop] | and voila, you shall get what you seek |
23:01.42 | SuperMMan | file[laptop]: ya i don`t have bt at all And i can`t seem to find it |
23:01.52 | file[laptop] | it's a command in gdb |
23:02.19 | SuperMMan | oh ok |
23:02.57 | defian | good night :) |
23:04.59 | *** join/#asterisk abombss (~abombss@c-67-175-115-51.client.comcast.net) |
23:05.20 | |Vulture| | Anyone use a 7960? I have 7 lines registered and whenever a call comes in over any lines, it looks like it is coming in over line 1... any ideas? |
23:06.29 | Nugget | I have a 7960. |
23:06.50 | Nugget | I've never figured out how it handles that sort of thing. I just use three lines. |
23:06.57 | *** join/#asterisk SirPrize (~blah@83.146.62.181) |
23:08.59 | |Vulture| | Nugget: yea its kinda annoying because I can't see who is trying to call me |
23:10.58 | Frantic | <|Vulture|> had the same issues: I finally modified the caller id to show who it goes to. |
23:11.33 | djMax | 6dB gain a reasonable number or does that indicate something horribly wrong? |
23:12.41 | |Vulture| | wonder if the IP600 suffers from this infliction |
23:12.56 | hmmhesays | well my callback scripts are working nicely... i'll be the first to say... I rock |
23:12.59 | zimdog | anyone setup NuFone with AMP? |
23:13.05 | |Vulture| | Ive gone from 7940/60s to IP500s |
23:13.13 | *** part/#asterisk sysdef (~sysdef@pD9560EB9.dip.t-dialin.net) |
23:14.07 | SirPrize | My Asterisk server accepts incoming PSTN calls via SIP. If I call Asterisk SIP-to-SIP(Asterisk), it accepts DTMF tones (Using X-lite). PSTN-to-SIP(asterisk) DTMF tones don't work for the menus using normal UK phones. Any idea what I can do ? |
23:14.53 | SirPrize | I've registered incoming SIP line via register => sipinfo. I've seen dtmfmode modifiers, but am unaware how to make them work for "register"ed SIP channels |
23:15.23 | *** join/#asterisk abombss (~abombss@c-67-175-115-51.client.comcast.net) |
23:16.10 | hmmhesays | what kind of gateware are you using SirPrize |
23:16.17 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
23:17.04 | hmmhesays | *gateway, or are you using a tdm card |
23:17.34 | SirPrize | hmmhesays: My Asterisk configuration only has SIP inbounds and SIP registered clients. I use SipGate.co.UK who provide the PSTN number, which is registered to a SIP address, which I've registered into Asterisk, so not using a card |
23:17.56 | [TK]D-Fender | Anyone her have experience making gigabit patch cables? I'm going nust here trying to get them to work. |
23:19.04 | *** join/#asterisk hajekd (~hajekd@21.208.65.212.contactel.net) |
23:19.54 | hmmhesays | set your dtmfmode to something different for outgoing to them |
23:20.05 | hmmhesays | such as rfc2833 |
23:20.44 | hmmhesays | what codec are you sending them? |
23:20.45 | SirPrize | that's my problem. I know how to set the dtmfmode for the registered clients/peers, but don't know how to set the dtfmmode coming in via the registered SIP channels. :-( |
23:21.16 | SirPrize | hmmhesays: I haven't disallowed any codecs - all the codecs are currently enabled |
23:21.16 | hmmhesays | show me your registration line out of sip.conf you can paste it at pastebin |
23:21.19 | *** part/#asterisk WizzKid (~apryer@cpc3-lutn5-3-0-cust169.lutn.cable.ntl.com) |
23:21.33 | SirPrize | what's the complete address for pastebin? pastebin.com ? |
23:21.37 | hmmhesays | yeah |
23:22.24 | SirPrize | Is it visible here: http://www.pastebin.com/243885 ? |
23:22.47 | hmmhesays | yeah |
23:23.22 | SirPrize | I saw an example which specifies the SIP domain as a user, and thought I could specify the dtmfmode in that way. That's also included in the paste if it helps |
23:23.40 | hmmhesays | dtmfmode=inband? that'll only work for g.711, so fi they are not sending in g.711 that won't work |
23:23.55 | *** join/#asterisk sudhir492 (~sudhir@4.7.59.232) |
23:23.58 | sudhir492 | hi all |
23:24.06 | sudhir492 | Anyone from Pakistan here? |
23:24.06 | SirPrize | ah - I see..... what dtmfmode can I use for standard PSTN ? |
23:24.17 | hmmhesays | well first check what they are sending into you |
23:24.19 | `Sauron | Hum. |
23:24.24 | SirPrize | does the specification that I've done like this work at all ? |
23:24.26 | `Sauron | what linux command returns a sockaddr? |
23:24.28 | `Sauron | err |
23:24.30 | `Sauron | C function |
23:24.31 | `Sauron | blah |
23:24.34 | hmmhesays | call in and sip show channels |
23:25.07 | SirPrize | using "alaw" format |
23:25.12 | *** join/#asterisk verge (~jfargen@56-116.26-24.tampabay.res.rr.com) |
23:25.17 | hmmhesays | they are sending you alaw? |
23:25.23 | SirPrize | yeah |
23:25.51 | hmmhesays | and you are using alaw in xten? |
23:26.18 | verge | I am new to * and I am looking for some guidance. |
23:26.28 | Sedorox | ~docs |
23:26.30 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
23:26.38 | SirPrize | I checked just now - X-lite when it connects uses "gsm". Explains why that works and PSTN doesn't. I see now |
23:26.58 | hmmhesays | well |
23:27.11 | hmmhesays | in your sip.conf set the dtmfmode for xten to something other than inband |
23:27.17 | hmmhesays | then it will work |
23:27.22 | hmmhesays | and you can still use gsm |
23:27.40 | SirPrize | it's actually x-ten that DOES work, and PSTN incoming alaw lines that didn't work on the menu |
23:27.55 | SirPrize | ah - I see what you mean |
23:28.18 | hmmhesays | ok... so you call in from the pstn and asterisk picks up and gives you ivr? |
23:28.24 | SirPrize | yes |
23:28.27 | hmmhesays | and the buttons aren't working on the ivr menu's? |
23:28.36 | verge | Ok, I have * running and have looked at the docs. My questions are rather specific in regards to connecting to the PSTN through livevoip using IAX. |
23:28.40 | SirPrize | exactly - from PSTN doens't work. From XTen works |
23:29.55 | verge | I am not sure how to map my DID's to my extensions. |
23:30.04 | verge | Can anyone help me with this question? |
23:30.06 | hmmhesays | ~docs |
23:30.08 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
23:30.23 | *** join/#asterisk file[laptop] (~file_lapt@mctn1-142166197096.nb.aliant.net) |
23:30.38 | hmmhesays | Sirprize set an extension to call a sip endpoint, press buttons on the calling phone |
23:30.41 | hmmhesays | see if you can hear them |
23:30.58 | hmmhesays | verge that is a very vague question |
23:31.12 | SirPrize | mmmm.... I changed the dtmfmode in the [sipgate.co.uk] section to rfc2833, but that didn't make a difference. Is this the right way I am trying, to set the dtmfmode for incoming SIP registered channels? |
23:31.38 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
23:31.41 | SirPrize | hmmhesays: I did that before to check, yes, I hear the button presses. I've tried calling in from both mobile and land phone, and neither of them can access the IVR |
23:31.58 | *** part/#asterisk numBone (~numBone@c-24-129-204-233.se.client2.attbi.com) |
23:32.00 | hmmhesays | Sirprize dtmfmode has to be inband for alaw |
23:32.15 | hmmhesays | and the ivr menu's work using xten |
23:32.31 | SirPrize | hmmhesays: yes, menu works with xten, doesn't work with mobile or landphone |
23:32.38 | Sedorox | verge: I think what you want is to have exten => <didnumber>,1,stuff-here in your context that your line is set for |
23:32.38 | hmmhesays | hrm |
23:32.53 | verge | hmmhesays: I am not sure what I should use for context= in iax.conf. |
23:33.10 | hmmhesays | context is the context in your extensions.conf |
23:33.47 | hmmhesays | SirPrize that's a good one |
23:34.06 | SirPrize | ? |
23:34.47 | hmmhesays | how new is your build sirprize? |
23:35.44 | SirPrize | am using asterisk-1.0.5 built from source the day before |
23:35.50 | SirPrize | standard stable package |
23:36.05 | verge | sirprize: this is what I currently have as my exten "exten => 2000,1,Dial(SIP/2000,20)" |
23:36.12 | verge | should I change 2000 to my DID? |
23:36.29 | hmmhesays | register => 1433188:XXXXXXXF@sipgate.co.uk where does the call go when it comes in? |
23:36.34 | hmmhesays | you don't have an extension specified |
23:36.56 | SirPrize | hmmhesays: It goes to the 's' extension in the 'incoming' context. I can hear the IVR activate |
23:37.50 | SirPrize | verge: Sorry, I haven't been following your thread. Let me check if I know what the |
23:38.00 | hmmhesays | verge is just talking to anyone who is talking |
23:38.01 | hmmhesays | lol |
23:38.04 | Sedorox | verge: what you wanna do is on the context that you have for the incoming.. you want to set exten => <did>,1,Dial(SIP/2000,20) to have it dial that phone |
23:38.48 | Mavvie | ~q931 |
23:39.00 | Mavvie | jbot: q.931 ? |
23:39.03 | *** join/#asterisk eye69 (magnus@ipv6.upcore.net) |
23:39.35 | SirPrize | hmmhesays: could you please confirm whether the [sipgate.co.uk] entry I have does in fact affect the registered incoming SIP channel? Is there any way I can check what the dtmfmode on an incoming line is set to by default? |
23:39.51 | Mavvie | worthless bot. |
23:40.13 | hmmhesays | SirPrize ok, try this.... register => 1433188:XXXXXXXF@sipgate.co.uk/12345 in sip.conf and exten => 12345,1,SIPDtmfMode(inband) exten => 12345,2,goto(s,1) |
23:40.20 | hmmhesays | in extensions.conf and test |
23:40.30 | SirPrize | ok, let me try that now |
23:40.51 | eaperezh | hi people |
23:41.26 | eaperezh | one quick question. is there any ready available script/plugin for * to do voicemail to email? |
23:41.43 | Mavvie | eaperezh: it comes with the package. |
23:41.49 | ManxPower | ~docs |
23:41.50 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
23:41.50 | Sedorox | its already part of voicemail |
23:41.52 | hmmhesays | set up sendmail and you're set |
23:41.53 | Sedorox | look in voicemail.conf |
23:42.41 | Mavvie | hmmm... somebody here with a copy of the Q931 specification? |
23:42.55 | hmmhesays | it's on voip-info isn't it? |
23:43.06 | Mavvie | hmmhesays: only links to the ITU websites |
23:43.18 | hmmhesays | is that not what you want? |
23:44.40 | eaperezh | Mavvie: sorry but can you point me? i cant seem to find it |
23:45.05 | hmmhesays | vi /etc/asterisk/voicemail.conf |
23:45.06 | Mavvie | eaperezh: in voicemail.conf, there is a field for the email address of the voicemailbox |
23:45.21 | Mavvie | hmmhesays: oh, I can download three specs for free! |
23:45.45 | hmmhesays | as for setting up sendmail, grab one of the eleventy billion guides you can find on your favorite friend and mine... GOOGLE! |
23:46.06 | hmmhesays | could probably grab it off peer to peer also Mavvie |
23:46.50 | eaperezh | Mavvie: will the voicemail remain as voicemail or will it be deleted and only remain in the user's mbox? |
23:46.59 | Mavvie | it will also be there. |
23:47.10 | hmmhesays | eaperezh:rtfm man |
23:47.11 | eaperezh | Mavvie: nice |
23:47.34 | eaperezh | Mavvie: will check on that right away.....many thanks |
23:47.52 | Mavvie | eaperezh: but there is nothing new I told you which you could find in the voicemail configuration file. |
23:48.29 | Mavvie | which makes me wonder if you actually are skilled enough to get it all up and running. |
23:48.29 | SirPrize | hmmhesays: mmmmm......... am trying to implement that change, but now when I try to call in, Asterisk logs this message "Channel 'SIP/sip.gossiptel.com-081471a8' sent into invalid extension 's' in context 'incoming', but no invalid handler", and I get a 403 Forbidden ?! |
23:48.35 | eaperezh | hmmhesays: im new to *, im not a programmer and have u take a look at the manuals? explanation is below average |
23:48.44 | `Sauron | Anyone on a linux box, run the following for me: grep '_len' /usr/include/bits/sockaddr.h |
23:49.06 | `Sauron | Hurmph. |
23:49.11 | hmmhesays | 'Sauron: what is that going to do to my linux box? |
23:49.13 | Mavvie | eaperezh: ignorance is not an excuse. |
23:49.13 | SirPrize | `Sauron: No hit |
23:49.15 | `Sauron | Nothing |
23:49.21 | `Sauron | SirPrize: Thanks. Grr. |
23:49.24 | hmmhesays | eaperezh: most of us in here are not programmers |
23:49.42 | eaperezh | Mavvie: i do have i up and running with 4 iax phones and 2 port fxo card |
23:49.48 | hmmhesays | in fact, i'm an insurance salesman from idaho |
23:49.50 | `Sauron | According to UNPv1, sockaddr->sa_len should exists, but it's not created in sockaddr.h |
23:49.56 | eaperezh | Mavvie: im implementing new functions as i learn |
23:50.10 | Mavvie | eaperezh: how did you do that? not by just guessing I take it? |
23:50.14 | *** join/#asterisk Vulture- (~Vulture@109.238.204.68.cfl.res.rr.com) |
23:50.16 | hmmhesays | SirPrize: did that work? |
23:50.46 | SirPrize | hmmhesays: I'm trying to implement that, but Asterisk now gives me an error message of "Channel 'SIP/sip.gossiptel.com-081471a8' sent into invalid extension 's' in context 'incoming', but no invalid handler". Trying to figure out what's happening |
23:50.49 | eaperezh | Mavvie: well (gasp) by readin the .conf files but i was not clear about the voicemal thing |
23:51.04 | Mavvie | eaperezh: so you did read the voicemail configuration file? |
23:51.15 | Vulture- | SirPrize: you need a context [incoming] and s,1,(command) |
23:51.27 | Vulture- | in extensions.conf |
23:51.53 | eaperezh | Mavvie: kind of but i was thinking that the voicemail was going to be removed and placed only on the mbox |
23:51.55 | SirPrize | Vulture-: I do have both of those in my extensions.conf already. :-S |
23:51.58 | `Sauron | SirPrize: You don't have an extension s defined in [incoming] |
23:52.15 | Vulture- | SirPrize: pastebin your extensions.conf |
23:52.31 | hmmhesays | yeah I might have given you a bad command, or you didn't type it right |
23:52.31 | `Sauron | also, make sure you do an extensions reload, just in case. :) |
23:52.34 | eaperezh | Mavvie: thats the original question...but thanks for clarifying it for me |
23:52.34 | Chuji | or just 'show dialplan incoming' |
23:52.40 | Vulture- | `Sauron: nice call |
23:52.41 | Mavvie | eaperezh: maybe you should read the configuration file *again*, and look for something like "from the server" |
23:53.12 | eaperezh | Mavvie: will sure do. |
23:53.13 | SirPrize | `Sauron: Ok, I corrected it. I had a duplicate line in there. |
23:53.21 | Vulture- | ;) |
23:53.22 | hmmhesays | eaperezh: http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf |
23:53.40 | hmmhesays | there's some good reading |
23:54.33 | mooboi | in order, what files should be configured first ? |
23:54.46 | SirPrize | hmmhesays: Now when I call in via XTen, it says "Inband DTMF is not supported on codec gsm. Use RFC2833", and PSTN phone still doesn't work. Will try switching to rfc2833 and see what happens |
23:55.11 | hmmhesays | SirPrize, we're just testing something, call in from the pstn |
23:55.18 | hmmhesays | to your IVR menu |
23:55.29 | *** join/#asterisk kFuQ (~somedude@c-24-17-224-78.client.comcast.net) |
23:55.29 | SirPrize | hmmhesays: it still didn't work even with inband, calling in from pstn |
23:55.42 | hmmhesays | it should have |
23:55.47 | hmmhesays | are you sure it didn't work? |
23:55.54 | hmmhesays | because it should have |
23:56.03 | eaperezh | tomorrow i will learn how to put all this stuff in to a DB.......thanks all for you kind help and hmmhesays for the links |
23:56.27 | `Sauron | eaperezh: for the DB stuff, google for ast_data |
23:56.36 | hmmhesays | mooboi: whatever your heart desires |
23:56.43 | chipig | ast_data++ |
23:56.55 | SirPrize | hmmhesays: yeah, unfortunately it didn't work. :-( let me paste some things into pastebin |
23:57.06 | hmmhesays | ok |
23:57.15 | mooboi | but which confis executed as soon as the line rings ? |
23:57.17 | `Sauron | I need to reconfigure ast_data here |
23:57.26 | terrapen | brb |
23:57.38 | SirPrize | hmmhesays: http://www.pastebin.com/243897 |
23:57.39 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
23:57.52 | Chuji | mooboi : What kind of 'line'? |
23:57.58 | SirPrize | Line 2 & 3 are for the incoming XTen call. Everything below is for the PSTN incoming call |
23:57.59 | hmmhesays | haha I really have no life, I should be out snogging some hot chick right now |
23:58.04 | hmmhesays | instead i'm doing tech support |
23:58.07 | hmmhesays | k |
23:58.10 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
23:58.24 | SirPrize | hmmhesays: with the help you're giving all of us, you sure do deserve the hottest chick :) |
23:58.31 | hmmhesays | LOL |
23:58.34 | hmmhesays | wahoo! |
23:58.50 | SirPrize | *grin* |
23:58.50 | hmmhesays | Sirprize you got ssh access to your box? |
23:59.08 | terrapen | anybody here deployed a unified Asterisk PBX at multiple locations? |
23:59.10 | SirPrize | hmmhesays: yes, I do |
23:59.21 | hmmhesays | if you give me access i'll take a look |
23:59.30 | hmmhesays | i promise not to fuck it up |
23:59.45 | SirPrize | let me set up an account ... |
23:59.47 | hmmhesays | k |
23:59.53 | hmmhesays | terrapen: yes |