| 00:00.30 | adeel | vap0rtranz, Jason99 i think you're experiencing a similar thing that happens to a lot of 'The Clapper' users...if there is any real loud, sharp, short, noise the clapper will turn off/on...but in your case, DTMF tones are being sent |
| 00:00.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:00.49 | [TK]D-Fender | joobie, well typically you'd be paying for as many channels as you do phones.... quite rare. Does your deployment demand that kind of ratio? |
| 00:01.17 | [TK]D-Fender | joobie, in your typical company you might have 1 PSTN channel / 4 employees. |
| 00:01.40 | joobie | [TK]D-Fender, ya.. it's an outbound call center.. so if there's 10 phones and 10 bums on seats, there theoretically can be 10 outbound calls at any given time, given everyone is working |
| 00:01.47 | SteveTotaro | it is an outbound call center so they don't want people on hold |
| 00:02.15 | [hC] | mosty: do you know where i might find an example of using SIP/IAX headers/variables to achieve this? |
| 00:02.22 | SteveTotaro | no if you are running a predictive dialer, you actually want more channels than seats |
| 00:02.30 | [TK]D-Fender | joobie, well I'd still put * in front, because odds are you'll be wanting some kind of logging, etc and would use a provider that lets you rig a common outbound CALLERID, etc... makes sense to have * in there for that... |
| 00:02.32 | adeel | which I think the problem is probably in the first few hops of the chain....the ATA or the fist * box i think is where the problem could be |
| 00:02.43 | joobie | true |
| 00:02.44 | vap0rtranz | joobie: i luv the calculation for how you don't need that many lines. it's in Wallingford's book. nice statistical value for how often a line could be busy |
| 00:02.46 | [TK]D-Fender | joobie, Also yeah, it is good to have "internal" capabilities.... |
| 00:03.05 | SteveTotaro | lookup erlang tables |
| 00:03.09 | SteveTotaro | ~erlang |
| 00:03.10 | jbot | Full-featured programming language developed at the Ericsson CS Laboratory. URL: http://www.erlang.org/ |
| 00:03.18 | joobie | ok sounds good |
| 00:03.24 | joobie | i think i might go that path instead then |
| 00:03.27 | vap0rtranz | SteveTotaro: erlang. ich liebe Deutsch! |
| 00:03.32 | joobie | but err.. one thing that doesn't sit right |
| 00:03.39 | [TK]D-Fender | joobie, All just thorough food for thought. |
| 00:03.48 | [TK]D-Fender | joobie, namely? |
| 00:04.09 | SteveTotaro | http://www.kooltoolz.com/ccm.htm?gclid=CJ7MvemT8pECFT00FQodi2aExQ |
| 00:04.54 | joobie | i've been looking at the 330 polycoms and they seem to have some good features to help voip.. like noise reduction, etc.. im thinking if i ditch the digital handsets and go for analogue with the linksys, i will lose those "voice quality" features.. like i'm happy to trade off the digital handsets for analogue if it's just feature loss in terms of functionality (because it's literally all outbound calls, minimum features).. but if it's going to introduce voi |
| 00:04.54 | joobie | ce loss, then im reluctant |
| 00:05.04 | ManxPower | I don't see why you don't just connect DIRECT to the ITSP and not use Asterisk. |
| 00:05.08 | mosty | [hC], i don't know about IAX since that iaxvars thing only works in 1.4, for sip you can use SipGetHeader / SipAddHeader i think |
| 00:05.16 | ManxPower | You don't need inbound, you don't need extension dialing, you don't need voicemail, etc. |
| 00:05.21 | adeel | has anyone setup * on a box with multiple nic's? |
| 00:05.30 | joobie | ManxPower, the call logging would be nice.. |
| 00:05.39 | SteveTotaro | manx, how many call centers have you been the engineer of? how many seats? calls per day? |
| 00:05.41 | mosty | [hC], or you can encode the information in your dial string, eg by sending a prefix for long distance numbers |
| 00:05.46 | joobie | ManxPower, say for example the supervisor wants to llisten in on a call.. that would be good.. or even to track the call usage per user |
| 00:05.48 | SteveTotaro | average length of call? |
| 00:05.52 | ManxPower | joobie: since all calls are billed, the ITSP should provide the info. |
| 00:05.55 | [TK]D-Fender | joobie, honestly I've been perfectly happy with ATA + analog. For pure quality alone I might not bother with SIP hard-phones. |
| 00:06.01 | [hC] | mosty: yeah thats true. both good ideas. Thanks! |
| 00:06.15 | ManxPower | SteveTotaro: "call center" to me implies "not cheap bastard" |
| 00:06.20 | [TK]D-Fender | joobie, They're great for "normal" office use, but not esential. |
| 00:06.32 | SteveTotaro | not sure what that means |
| 00:06.32 | ManxPower | and it sounds like joobie is in the "cheap bastard" class. |
| 00:06.40 | Washy | what is asterisk for? |
| 00:06.52 | SteveTotaro | asterisk is a wildcard |
| 00:06.54 | [TK]D-Fender | Washy, you even have to ask that? |
| 00:07.02 | SteveTotaro | ~asterisk |
| 00:07.02 | jbot | hmm... asterisk is the best free PBX in the world, or #asterisk on irc.freenode.net, or http://www.asterisk.org |
| 00:07.08 | lunaphyte | it's an embedded operating system for high tech dishwashers. |
| 00:07.10 | [TK]D-Fender | Washy, go free your mind and read the book a bit... |
| 00:07.12 | *** part/#asterisk enjay5150 (n=chatzill@ip70-190-63-195.ph.ph.cox.net) |
| 00:07.12 | coppice | ManxPower: prudent and frugal might be nicer ways to descibe a tightass |
| 00:07.20 | ManxPower | I hate that. Asterisk isn't a PBX, it's a PBX toolkit. |
| 00:07.31 | [TK]D-Fender | ManxPower, Agreed.... |
| 00:07.33 | joobie | I see |
| 00:07.41 | ManxPower | coppice: I have no interest in sugar coating this. |
| 00:07.47 | joobie | thanks TK n Manx |
| 00:07.51 | joobie | i think ill try the linksys route |
| 00:07.53 | joobie | cheres |
| 00:07.57 | joobie | cheers even |
| 00:07.59 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 00:07.59 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:07.59 | SteveTotaro | manx, don't be a pr1ck though |
| 00:08.06 | joobie | u too steve |
| 00:08.08 | joobie | for the help:P |
| 00:08.10 | SteveTotaro | people have to start somewhere |
| 00:08.44 | joobie | ManxPower, the ITSP won't allow us to listen in on a call |
| 00:08.47 | joobie | as asterisk will |
| 00:08.48 | drmessano | <lunaphyte> it's an embedded operating system for high tech dishwashers. <---- ftfw |
| 00:08.53 | ManxPower | joobie: That is correct. |
| 00:08.57 | joobie | so i think it's worthwhile.. also we'll be limited by their accounting |
| 00:08.58 | SteveTotaro | i am defending you joobie, manx called you a "cheap bastard" |
| 00:09.08 | joobie | heeh yea i saw.. |
| 00:09.20 | joobie | but err, this is not coming out of my back pocket |
| 00:09.20 | ManxPower | joobie: Well, MOST ITSPs won't. Many will do a hosted Asterisk so they may have that feature. |
| 00:09.21 | SteveTotaro | talking about your mother..... |
| 00:09.24 | joobie | it's a client requirement.. |
| 00:09.27 | [TK]D-Fender | SteveTotaro, lol |
| 00:09.59 | lunaphyte | drmessano: :) |
| 00:10.02 | ManxPower | Heck, even New Orleans, which is in the stone age for many aspects of tech has at least 2 Asterisk based hosted solutions. |
| 00:10.17 | drmessano | joobie: I don't think you're a cheap bastard.. I think you're fatherless and ignorantly resistant to spending money... |
| 00:10.24 | drmessano | But not a cheap bastard |
| 00:10.24 | joobie | Manx, thanks.. it's a decent view to have. But I don't share it.. i think a local asterisk build will give more flexibility from that sense |
| 00:10.43 | joobie | ie. if you want to track realtime stats.. you can poll the asterisk box, rather than the TISP.. who may not update realtime |
| 00:10.45 | ManxPower | joobie: may or may not be a cheap bastard -- HOWEVER, he has all the problems of being a cheap bastard |
| 00:11.01 | SteveTotaro | yeah, i would just trust my itsp to do all my billing |
| 00:11.03 | joobie | your approach will put a lot more weight on the functionality of the TISP, which will cut down the choice available for a quality provider |
| 00:11.08 | SteveTotaro | with no way to reconcile |
| 00:11.17 | [TK]D-Fender | drmessano, Only fatherless person I know of was Jesus, the rest are just "absentee" :p |
| 00:11.25 | drmessano | hehe |
| 00:11.28 | ManxPower | joobie: you are using the internet for phone calls, just how reliable do you expect it to be? |
| 00:11.32 | joobie | i mean you could have a top notch provider in cost.. in performance.. but he could not support your accounting featres so he's out.. that's a big loss, given the vast amount of poor quality providers out there |
| 00:11.57 | SteveTotaro | just splice into a 200 pair, you are bound to have some dialtones |
| 00:12.05 | joobie | drmessano, get it right bro.. this system is not for me .. it's for a client. I just bought two polycom phones for myself today.. the analogue cheap-as-cheaps solution is not my doing |
| 00:12.16 | [TK]D-Fender | joobie, You seem to have a pretty good grasp of things so far. Go canvas your prospective internet & ITSP providers. |
| 00:12.29 | ManxPower | joobie: I've turned down customers for being cheap. |
| 00:12.29 | errr | I have an IAX trunk setup to teliax, when you make or recieve a call the person on the end out side the pbx hears all kinds of break up.. its really choppy. What could cause that?? I have a 12/1 connection so I would think that bandwidth wouldnt be an issue.. |
| 00:12.36 | joobie | Cheers TK |
| 00:12.42 | ManxPower | errr: turn off trunking and see if that helps. |
| 00:12.54 | drmessano | If you start off with the cheapest variety of any system, you will never come close to having the reliability and feature set required to jusitfy future upgrades.. Therefore, it will be doomed from the start and not even worth it. |
| 00:12.54 | SteveTotaro | so the question still remains, Manx, how many call centers have you engineered? |
| 00:12.56 | errr | ManxPower: what do you mean turn off trunking? |
| 00:13.03 | joobie | Manx, you yourself said you went with ONE analogue solution - you learnt from it and will never do it again.. |
| 00:13.07 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:13.08 | SteveTotaro | sizes, volumes, connectivity? |
| 00:13.35 | ManxPower | errr: You said you have an IAX trunk. An IAX trunk specifically puts voice from more than one phone call into the same UDP packet to save bandwidth. |
| 00:13.40 | joobie | i haven't been down that path. I've made my suggestoins to the client.. he still wants to push down that path, I get paid.. so i dont see the harm - he has been warned and in the process it can be a learning curve for me. |
| 00:13.51 | joobie | anywhoo |
| 00:13.53 | ManxPower | SteveTotaro: depends on how you define a call center, but the answer is probably "none". |
| 00:13.54 | joobie | back to work |
| 00:13.59 | joobie | thanks for the help guys - much appreciateed |
| 00:14.03 | ManxPower | ~trunk |
| 00:14.04 | jbot | from memory, trunk is is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 00:14.25 | vap0rtranz | jbot: sip channel? |
| 00:14.32 | ManxPower | errr: perhaps you are confused and you don't really have an IAX trunk? |
| 00:14.33 | drmessano | sip peer |
| 00:14.46 | drmessano | ~siptrunk |
| 00:14.47 | jbot | There is nothing special about a SIP trunk in the protocol like there is in the case of IAX2, for example. You set up a SIP trunk like a regular peer in sip.conf. |
| 00:14.56 | SteveTotaro | manx i think you are wrong, the udp packet does not contain voice from more than one call |
| 00:15.08 | SteveTotaro | i think iax trunking just cuts the overhead |
| 00:15.12 | ManxPower | SteveTotaro: No, YOU are wrong for IAX. |
| 00:15.20 | JT | SteveTotaro: you are not correct |
| 00:15.29 | JT | SteveTotaro: iax trunnking combines multiple calls |
| 00:15.35 | errr | ManxPower: maybe so.. I just know I have an account with teliax and they point an 800 number to me and the connection is made using IAX |
| 00:15.35 | JT | trunking |
| 00:15.45 | SteveTotaro | the signaling is combined |
| 00:15.47 | JT | errr: does trunk=yes? |
| 00:15.49 | ManxPower | errr: that's called an "iax peer" or "iax connection" |
| 00:15.52 | JT | SteveTotaro: and the payload |
| 00:16.00 | JT | SteveTotaro: signalling and payload is combined in iax |
| 00:16.02 | SteveTotaro | oh, ok i am wrong then |
| 00:16.14 | JT | signalling and payload is combined for multiple calls in iax with trunk=yes |
| 00:16.19 | SteveTotaro | i don't use iax because it causes so many problems |
| 00:16.22 | [TK]D-Fender | Tastes great! |
| 00:16.23 | JT | me too |
| 00:16.26 | [TK]D-Fender | Less filling! |
| 00:16.31 | [TK]D-Fender | TASTES GREAT! |
| 00:16.32 | SteveTotaro | even iax.cc says don't use it |
| 00:16.35 | [TK]D-Fender | LESS FILLING!!!!! |
| 00:16.35 | JT | i avoid iax except for testing or unimportant stuff |
| 00:16.57 | errr | ManxPower: ok so what can cause the choppy sound Im getting? |
| 00:17.01 | coppice | MGCP forever! |
| 00:17.01 | drmessano | [TK]D-Fender: Now lets talk about IAX phones |
| 00:17.12 | SteveTotaro | iax causes choppy sound |
| 00:17.21 | SteveTotaro | switch to sip and watch it go away |
| 00:17.48 | errr | SteveTotaro: we use iax at work and we dont have this problem.. |
| 00:17.49 | SteveTotaro | seen it in at least ten live production boxes |
| 00:18.19 | errr | I would rather have 1 port open on my firewall instead of all the ones needed for sip |
| 00:18.21 | drmessano | Since I started using IAX with my ITSP, my dishes seem to come out of the dishwasher cleaner.. not sure if there's a connection |
| 00:18.26 | SteveTotaro | i charged $750 to switch an ITSP from IAX to SIP |
| 00:18.34 | SteveTotaro | fixed their choppy problem |
| 00:18.43 | bkw__ | yah |
| 00:18.47 | bkw__ | IAX isn't great for an ITSP |
| 00:19.03 | SteveTotaro | their techs couldn't figure it out |
| 00:19.05 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.107.24) |
| 00:19.09 | bkw__ | great for smaller stuff... but sucks when you load 100's of people onto it |
| 00:19.18 | bkw__ | SteveTotaro: I know first hand about that problem ;) |
| 00:19.39 | drmessano | So what's the problem? IAX2, Asterisk, someone else implementation? |
| 00:19.39 | SteveTotaro | you do? do tell? |
| 00:19.46 | JT | iax is a joke |
| 00:19.57 | bkw__ | SteveTotaro: IAX mixing media and signaling in a large scale deployment will fail |
| 00:20.02 | bkw__ | many people registering |
| 00:20.07 | JT | errr: that is the stupidest reason ever for choosing IAX2 |
| 00:20.12 | bkw__ | and audio flowing.. gets hiccups in the audio |
| 00:20.14 | JT | errr: who cares how many ports are open? |
| 00:20.14 | bkw__ | due to that |
| 00:20.17 | SteveTotaro | oh yeah, i thought maybe you knew the ITSP |
| 00:20.21 | JT | errr: it's the same app listening on those ports/ |
| 00:20.23 | drmessano | The port argument sucks |
| 00:20.25 | bkw__ | SteveTotaro: Nope but I know of the problem |
| 00:20.29 | drmessano | JT: Exactly |
| 00:20.34 | bkw__ | drmessano: IAX has its place |
| 00:20.41 | bkw__ | an ITSP isn't really a good use of it |
| 00:20.47 | SteveTotaro | yes, iax is only good for funky nat problems with low call volumes |
| 00:21.03 | bkw__ | and good implementations can bust NAT without a problem |
| 00:21.12 | drmessano | "ZOMG, I HAVE 10000 ports open" <-- If Asterisk has a FAIL, it only takes ONE port |
| 00:21.13 | SteveTotaro | like getting voip in some african countries where their nat is behind another nat and so on |
| 00:21.23 | errr | guys I have 1 number and no more than 1 call at a time. |
| 00:21.48 | JT | errr: and it sounds like you have one poor quality call |
| 00:21.50 | errr | it cant believe iax is really *that* bad.. |
| 00:21.55 | SteveTotaro | i have seen five nats before hitting the other side |
| 00:21.56 | JT | would you rather one poor quality call |
| 00:21.56 | errr | I guess so |
| 00:21.59 | JT | or one good one? |
| 00:22.10 | SteveTotaro | hey if bkw is backing me up, then you know it's true |
| 00:22.15 | bkw__ | hehe |
| 00:22.20 | bkw__ | tripple nat is easy to bust |
| 00:22.24 | SteveTotaro | i might not always be right |
| 00:22.29 | coppice | theer is nothing in the IAX protocol which would make sound choppy. I assume it is problems in the implementation |
| 00:22.32 | SteveTotaro | but usually i am |
| 00:22.37 | bkw__ | coppice: BINGO |
| 00:22.54 | JT | *cough* zap timing |
| 00:23.02 | drmessano | Hmmm |
| 00:23.04 | bkw__ | you don't need no stinking timing |
| 00:23.14 | bkw__ | its voip.. perfection isn't a requirement |
| 00:23.21 | SteveTotaro | that is to sell zaptel cards |
| 00:23.21 | joobie | [TK]D-Fender, still around? just curious if that linksys device with analogue phone can handle on-hold? |
| 00:23.33 | drmessano | IAX uses zap timing? |
| 00:23.35 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 00:23.40 | bkw__ | drmessano: yep it can |
| 00:23.47 | mosty | drmessano, for trunking, yes |
| 00:23.56 | drmessano | I see |
| 00:24.10 | SteveTotaro | just forget iax |
| 00:24.15 | mosty | drmessano, and any channel that uses meetme |
| 00:24.15 | drmessano | Fascinating |
| 00:24.22 | drmessano | I knew about meetme |
| 00:24.25 | drmessano | Didnt know about IAX |
| 00:24.25 | bkw__ | meetme is just aweful |
| 00:24.26 | JT | and music on hold |
| 00:24.27 | SteveTotaro | go mgcp or h323 |
| 00:24.39 | coppice | MGCP rulez! |
| 00:24.45 | bkw__ | you dont need no hardware clocking for conference, moh or trunking |
| 00:25.01 | [TK]D-Fender | joobie, Yes, it can do "on-hold", but its not the friendliest thing to do it with.... |
| 00:25.07 | mosty | drmessano, i normally disable iax trunking anyway, it doesn't handle high loads very well |
| 00:25.16 | [TK]D-Fender | joobie, thats the price of analog. |
| 00:25.18 | drmessano | bkw__: Dare I ask this, why then is it that way? lol |
| 00:25.23 | SteveTotaro | iax is hype |
| 00:25.31 | bkw__ | drmessano: guess so they can sell more hardware |
| 00:25.32 | bkw__ | :P |
| 00:25.36 | drmessano | LOL |
| 00:25.42 | alrs | I'm no fan of IAX |
| 00:25.55 | bkw__ | IAX has its own set of problems |
| 00:25.55 | [TK]D-Fender | SteveTotaro, Not jsut hype, but the only times I validate it are if you NEED to trunk for BW, or a screwed by your firewall. |
| 00:26.05 | drmessano | If that's the case, chucking an X100P in there for timing solves that mess |
| 00:26.12 | bkw__ | what mess? |
| 00:26.13 | drmessano | For $30 |
| 00:26.14 | coppice | there seems to be some movement on standardising RTP trunking again |
| 00:26.16 | SteveTotaro | fix your firewall by your logic |
| 00:26.23 | bkw__ | coppice: yah that would be nice |
| 00:26.33 | drmessano | Well, not solves it.. |
| 00:26.46 | bkw__ | we do Conferences in FreeSWITCH without a hardware device |
| 00:26.51 | drmessano | Improves it over ztdummy, I guess |
| 00:27.00 | SteveTotaro | freeswitch needs docs |
| 00:27.06 | bkw__ | SteveTotaro: it has them. |
| 00:27.13 | alrs | mars needs women |
| 00:27.18 | bkw__ | SteveTotaro: I provide personal hands on assistance to anyone thats willing to wikify it |
| 00:27.27 | bkw__ | SteveTotaro: so far its a good trade |
| 00:27.33 | bkw__ | SteveTotaro: wiki.freeswitch.org |
| 00:28.04 | SteveTotaro | tell me about call setup in FW as opposed to * |
| 00:28.13 | SteveTotaro | raw setup speed |
| 00:28.19 | bkw__ | SteveTotaro: for SIP? |
| 00:28.32 | Jason99 | should RFC2833 payload type be 101 or 96 ? |
| 00:28.47 | bkw__ | Jason99: 101 is the standard location but can be anywhere in the dynamic range |
| 00:29.18 | Jason99 | bkw__: ok thanks, could it make a difference? |
| 00:29.33 | SteveTotaro | yes sip |
| 00:29.57 | SteveTotaro | i want to know * vs FS sip call setup raw numbers |
| 00:30.19 | coppice | RFC2833 needs casting into the cesspit of history |
| 00:30.26 | bkw__ | SteveTotaro: I know it can do way more than * |
| 00:30.34 | bkw__ | SteveTotaro: but it depends on setup... |
| 00:30.38 | bkw__ | SwK: you here? |
| 00:31.33 | bkw__ | SteveTotaro: I know it can exceed 200 cps in some cases. |
| 00:31.34 | [TK]D-Fender | alrs, old-school cool.... |
| 00:32.18 | SteveTotaro | what about * |
| 00:32.24 | SteveTotaro | where does that max out? |
| 00:32.41 | bkw__ | the last report that transnexus did.. went as far as 8 cps |
| 00:33.07 | bkw__ | SteveTotaro: I don't even begin to publish numbers ... a trusted third party would need to do the testing |
| 00:33.32 | bkw__ | SteveTotaro: because numbers all depend on situation.. usage.. and various other local variables |
| 00:34.18 | SteveTotaro | who would you consider a trusted 3rd party? |
| 00:34.23 | bkw__ | coppice: rfc4733 |
| 00:34.30 | SteveTotaro | not just for asterisk but for hardware as well |
| 00:34.31 | bkw__ | SteveTotaro: someone that isn't biased in either direction |
| 00:34.47 | SteveTotaro | with methodology that is strong |
| 00:35.03 | coppice | bkw__: duh, I do need to add a :-) to the end of everything? |
| 00:35.04 | SteveTotaro | i am asking for a reference here |
| 00:35.09 | drmessano | Microsoft |
| 00:35.17 | bkw__ | SteveTotaro: Ask SwK |
| 00:35.26 | JT | microsoft will help you VoIP As You Are |
| 00:35.31 | drmessano | HA |
| 00:35.35 | bkw__ | hehe |
| 00:35.35 | drmessano | Yes, they will |
| 00:35.39 | SteveTotaro | no because i have the opportunity to do hardware benchmarking and need help in this regard |
| 00:35.40 | drmessano | Don't throw away your PBX yet |
| 00:35.47 | bkw__ | oh |
| 00:35.52 | bkw__ | SteveTotaro: I have done it.. and can help you |
| 00:35.57 | bkw__ | but I don't publish numbers |
| 00:36.08 | drmessano | "Don't throw away your PBX yet, because our VoIP doesn't really handle the V part yet" |
| 00:36.32 | bkw__ | I'm not going to make any comments about the V part |
| 00:36.50 | tzafrir | bkw__, test what, exactly? What qualities / quantities? |
| 00:37.01 | coppice | V for vendetta? :-\ |
| 00:37.13 | bkw__ | tzafrir: all of the above |
| 00:37.35 | *** join/#asterisk GBR_ (n=gbr@201-67-16-229.gnace703.dsl.brasiltelecom.net.br) |
| 00:37.53 | drmessano | I see those M$ ads where they tell you not to throw your PBX out, and all I can think is "...for when our product crashes" |
| 00:38.30 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:38.32 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 00:39.41 | drmessano | "It's 3PM.... You're minutes away from closing the big deal.... The phone rin.... OH NOO A BSOD!!11!! FAIL!!11!!!! UR GOIN OT OF BUZNUSS LULZ FAIL FAIL" |
| 00:39.58 | drmessano | VoIP as you are |
| 00:40.01 | SteveTotaro | tzafrir is interested as well as many others, i think i need to seek a consultant outside of the asterisk world |
| 00:40.38 | tzafrir | drmessano, s/BSOD/BUG()/ |
| 00:40.53 | bkw__ | haha |
| 00:40.55 | SteveTotaro | i liked it when asterisk crashed during a commercial on judge judy |
| 00:41.04 | drmessano | lol |
| 00:41.09 | SteveTotaro | 400 callers dropped |
| 00:41.28 | Qwell | SteveTotaro: eh? |
| 00:41.55 | SteveTotaro | because my CTO decided to add queueprio= and not tell me and then leave to close the mortgage on his new home |
| 00:42.28 | SteveTotaro | a spot a judge judy isn't cheap |
| 00:42.36 | drmessano | WAIT |
| 00:42.37 | bkw__ | I like her |
| 00:42.41 | Qwell | oh, you did a commercial |
| 00:42.43 | drmessano | Dude! |
| 00:42.45 | bkw__ | and Judge Marilyn too |
| 00:42.51 | SteveTotaro | neither is dropping 400 suckers, i mean cutomers |
| 00:42.53 | drmessano | Was that for the Magic Brush? |
| 00:42.58 | drmessano | I SO WANTED TO ORDER ONE |
| 00:43.22 | SteveTotaro | no, i worked for a bad company |
| 00:43.25 | drmessano | Sorry, caffeine |
| 00:43.37 | drmessano | A Lawyer? |
| 00:43.38 | Qwell | ambulancechaser++ |
| 00:43.41 | drmessano | "One call, that's all" |
| 00:43.52 | Qwell | "Larry Parker got me ..." |
| 00:44.08 | drmessano | "Ken Nugent got my grandma paid" |
| 00:44.17 | SteveTotaro | the #1 most complained about company to the BBB |
| 00:44.23 | drmessano | We have this one guy in town, who insists he can "Supersize your settlement" |
| 00:44.32 | drmessano | I kid you not |
| 00:44.41 | tzafrir | sue him |
| 00:44.42 | Qwell | paypal? |
| 00:45.13 | SteveTotaro | i just got a paypal mastercard |
| 00:45.25 | drmessano | The law offices of dewey, cheatum, and Howe |
| 00:45.26 | SteveTotaro | figured i might as well get some points |
| 00:45.34 | coppice | drmessano: well, he's a lawyer. he needs to dumb things down until he can understand them |
| 00:45.41 | drmessano | lol |
| 00:46.28 | drmessano | Ken Nugent is great.. he's a slimeball... "One call..... that's all" in the same tone he uses to seduce his temps |
| 00:46.44 | SteveTotaro | temps rule |
| 00:47.00 | drmessano | "Im not married, thats a friendship ring" |
| 00:47.03 | Qwell | SteveTotaro: you should totally out the company |
| 00:47.22 | SteveTotaro | i am out of the company |
| 00:47.26 | drmessano | FlowBee ? |
| 00:47.35 | Qwell | SteveTotaro: hence the reason you should out it |
| 00:47.39 | SteveTotaro | do you know who the company is? |
| 00:48.00 | coppice | Honest Joe's VoIP Emporium? |
| 00:48.06 | Qwell | no idea |
| 00:48.13 | SteveTotaro | i know every piece of the operation |
| 00:48.22 | drmessano | LifeAlert? |
| 00:48.28 | drmessano | IVE FALLEN.... AND I CANT GET UP |
| 00:48.28 | SteveTotaro | they just settled with the FTC last week for $5mil |
| 00:48.41 | SteveTotaro | but are still in business |
| 00:48.55 | SteveTotaro | spend $30 million a year on advertising |
| 00:49.17 | Qwell | ...blue hippo? |
| 00:49.26 | drmessano | BlueHippo |
| 00:49.31 | SteveTotaro | i take the fifth |
| 00:49.32 | drmessano | Qwell: Googling SOB |
| 00:49.59 | SteveTotaro | http://www.edisonworldwide.com/ |
| 00:50.01 | drmessano | OHHHHHH |
| 00:50.12 | Qwell | uh huh |
| 00:50.12 | [TK]D-Fender | Got a quick question perhaps someone could help me with. I have a failry stock install of CentOS 5.1 brought current, and a process that is generating e-mails to be sent out. they get queued up, but don't see to transmit until I do a "sendmail -q" from CLI myself. |
| 00:50.16 | drmessano | I've seen BlueHippo |
| 00:50.20 | Qwell | wow |
| 00:50.23 | Qwell | they own bluehippo |
| 00:50.27 | Qwell | shaaaaddy |
| 00:50.27 | [TK]D-Fender | And the process is running. The message I get prior to processing the queue manually is "(host map: lookup (target domain for e-mail): deferred)" |
| 00:50.32 | SteveTotaro | it is a shell game |
| 00:50.40 | SteveTotaro | the whole thing is a shell game |
| 00:50.49 | SteveTotaro | bluehippo owns nothing |
| 00:50.55 | SteveTotaro | so sue them |
| 00:51.26 | drmessano | I always knew Edison was a fraud |
| 00:51.36 | SteveTotaro | corporate shell game at it's best devised by teams of high powered lawyers |
| 00:51.47 | coppice | is blue hippo supposed to be a play on pink elephant? |
| 00:51.59 | SteveTotaro | google joe rensin and credit trust |
| 00:52.37 | SteveTotaro | guess who built the relax phone system under edison's site? |
| 00:52.43 | Qwell | you? |
| 00:52.47 | SteveTotaro | not me, i plead the 5th |
| 00:53.27 | *** join/#asterisk kimosabe (n=nat@adsl-69-155-128-143.dsl.hstntx.swbell.net) |
| 00:53.31 | drmessano | BlueHippo has a foundation.. I bet they have an OLPC type program, but in there case, there's ONE laptop and ONE child |
| 00:53.50 | joobie | [TK]D-Fender, thanks.. just had a work call so had to duck out for a moment. Cheers |
| 00:53.59 | SteveTotaro | no, they made a marketing guy drive to the worst part of baltimore with crappy PCs |
| 00:54.17 | SteveTotaro | he was terrified |
| 00:54.21 | drmessano | Jeez |
| 00:54.31 | bkw__ | why was he scared? |
| 00:54.34 | bkw__ | pussy |
| 00:54.48 | SteveTotaro | west balto is place to be |
| 00:55.10 | SteveTotaro | especially with computers |
| 00:55.30 | kimosabe | does any one know any thing about t-3 conections |
| 00:55.35 | SteveTotaro | the foundation gave five or ten away and then released their own pr |
| 00:55.37 | drmessano | What is there to be scared of.. White guy in a suit, with a bunch of PCs loaded in his car...In the ghetto.. No cops around.. |
| 00:55.39 | drmessano | Wimp |
| 00:55.44 | SteveTotaro | i know about t3s |
| 00:56.05 | SteveTotaro | the sell crack in the open on the corners |
| 00:56.10 | coppice | t3 is the one where arnie plays the good guy, right? |
| 00:56.23 | SteveTotaro | if you stop for a stop sign they rush your car with little baggies |
| 00:56.29 | [TK]D-Fender | coppice, that'd be 2 & 3 |
| 00:56.32 | drmessano | Follow me if you vant to live |
| 00:57.10 | SteveTotaro | t3 uses two coax cables, one send one receive 672 channels |
| 00:57.37 | drmessano | Poor arnie.. now he just uses his puns in speeches.. and they're bad.. "I'll be back... for another term" |
| 00:57.38 | kimosabe | correct |
| 00:57.53 | kimosabe | we have 45 meg conection via coper pair tx rx |
| 00:58.28 | coppice | kimosabe: there, you're an expert |
| 00:58.41 | SteveTotaro | so did you have a question or just showing off? |
| 00:58.44 | SteveTotaro | either is cool |
| 00:59.19 | JT | i love it when people think "fibre optic" is a type of connection technology |
| 00:59.32 | kimosabe | no i want to sell t-1 now from this place im wrking at just looking for some one to lead me in the correct direction |
| 00:59.43 | SteveTotaro | i have a fiber optic plant looking decoration |
| 00:59.44 | drmessano | JT: It's a type of flashlight, duh |
| 00:59.57 | SteveTotaro | some stupid birthday present |
| 01:00.29 | kimosabe | does any one have a point to multipoint via a ds3 circuit want some advice please |
| 01:00.29 | SteveTotaro | you want to sell t1s off your t3? |
| 01:01.00 | kimosabe | i want to sell t-1s via the ds3 circuit yes |
| 01:01.02 | SteveTotaro | buy an adtran 2800 m13 |
| 01:01.10 | Qwell | I just wanna know where T2 went |
| 01:01.11 | SteveTotaro | 28 t1s |
| 01:01.15 | Qwell | Where's my T2 PRI? |
| 01:01.30 | drmessano | heh |
| 01:02.14 | coppice | Qwell it just didn't become popular. in ETSI land, E2 (8M) has been used quite a bit |
| 01:03.01 | kimosabe | stevetotaaro one sec please |
| 01:03.02 | Qwell | how many channels? |
| 01:03.16 | drmessano | We traded all our C3's for P0's back in the mid 90's |
| 01:03.33 | Qwell | or is it even channelized? |
| 01:03.57 | coppice | on a T2? I can't remember. the ETSI PDH stack goes up in steps of 4 each time, but the T ones group T1s in odd sized steps |
| 01:04.22 | vap0rtranz | *eek* telco speak! |
| 01:04.26 | coppice | T2 is a PDH stack of T1s |
| 01:04.33 | Qwell | vap0rtranz: in a PBX channel? never! |
| 01:04.33 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 01:05.10 | vap0rtranz | Qwell: sometimes i wish Mark would have written * for Winblows |
| 01:05.13 | drmessano | Pretty Damn High? |
| 01:05.16 | vap0rtranz | damn those closed api's |
| 01:05.16 | JT | an E2 is 4 * E1 iirc |
| 01:05.31 | drmessano | AsteriskWin32 FTW |
| 01:05.42 | JT | Plesiochronous Digital Heirarchy |
| 01:05.45 | JT | PDH |
| 01:05.48 | SteveTotaro | anyone know what an oc12 is? |
| 01:05.51 | vap0rtranz | JT: lol |
| 01:06.27 | Qwell | SteveTotaro: 12 oc's? |
| 01:06.27 | vap0rtranz | SteveTotaro: Nortel |
| 01:06.27 | drmessano | AsteriskWin64 <-- Gonna put * on the map |
| 01:06.27 | Qwell | drmessano: there's a thought |
| 01:06.27 | JT | SteveTotaro: 622Mbit/s |
| 01:06.27 | drmessano | OC12 is the stuff the cops spray in your eyes when you yell "Dont tase me, bro" way too much |
| 01:06.30 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 01:06.54 | JT | SteveTotaro: OC is the north american Sonet system |
| 01:07.00 | JT | most other places use SDH |
| 01:07.07 | vap0rtranz | drmessano: i honestly believe some of the closed thinking/speaking would have been a match out of heaven |
| 01:07.35 | coppice | sonet and SDH are almost the same thing |
| 01:07.45 | SteveTotaro | i know, i was just playing the kimosabe part |
| 01:07.47 | JT | almost |
| 01:08.19 | drmessano | You can happily run AsteriskWin32 (based on 1.2), right now.. Relive 2003 all over again |
| 01:09.25 | drmessano | I can't imagine anything running on Cygwin not being good enough for production |
| 01:10.23 | coppice | don't knock cygwin. its a great development tool for unix code when you are forced to travel with a windows notebook :-) |
| 01:10.39 | Qwell | no, it's really not |
| 01:10.46 | Qwell | vmware > cygwin |
| 01:11.02 | drmessano | coppice: It's datacenter ready |
| 01:11.23 | vap0rtranz | vmware >> wine < cygwin |
| 01:11.47 | drmessano | What about cygwin on wine? |
| 01:11.57 | vap0rtranz | it gets drunk |
| 01:12.07 | drmessano | lol |
| 01:12.09 | Qwell | I tried to wine cygwin once |
| 01:12.15 | Qwell | just to get that genuine experience |
| 01:12.17 | drmessano | Oh yeah? |
| 01:12.23 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:12.26 | coppice | well, I would miss cygwin a lot, if it disappeared |
| 01:12.34 | Qwell | no, not really, but give me about 2 minutes, because I'm going to |
| 01:12.38 | drmessano | LOL |
| 01:12.48 | Qwell | let's see what happens :P |
| 01:12.51 | drmessano | Anyone ever run VMware under VMware? |
| 01:13.13 | kimosabe | does any one here have a point to multipoint from a ds3 to x amount of conections |
| 01:13.24 | Qwell | fail |
| 01:13.28 | Qwell | drmessano: not possible |
| 01:13.33 | drmessano | :( |
| 01:13.35 | Qwell | they put in checks |
| 01:13.37 | *** join/#asterisk errr_ (n=errr@fedora/errr) |
| 01:13.45 | drmessano | There goes my future development plans |
| 01:13.48 | Qwell | loaded the cygwin setup.exe, and it came up...but with no buttons |
| 01:14.10 | drmessano | Hmm |
| 01:14.25 | drmessano | So wine IS just like windows.. it even has the "ZOMG, THIS DOESNT WORK" |
| 01:14.36 | drmessano | Damn, they're doing good |
| 01:14.45 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 01:14.57 | Qwell | heh, that's kinda like that samba thing... |
| 01:15.05 | drmessano | I hope it runs viruses as well |
| 01:15.07 | Qwell | there was a stupid bug in windows, and the same bug existed in samba |
| 01:15.19 | Qwell | I think it was samba anyways. might've been wine |
| 01:15.25 | drmessano | nice |
| 01:16.11 | drmessano | I think it's funny when someone codes an emulator BETTER than the original, and has to code in a throttle so the software doesn't puke from the lack of crappy hacks |
| 01:16.35 | drmessano | I bet Wine is full of those sort of things |
| 01:17.19 | SteveTotaro | i just like to run IE6 on FC8 |
| 01:17.37 | *** join/#asterisk nauCe (n=nauce@ip24-255-116-169.dc.dc.cox.net) |
| 01:18.01 | SteveTotaro | citibank won't let me bank online without ie |
| 01:18.10 | Qwell | time for a new bank |
| 01:18.27 | SteveTotaro | so i had to install wine and ie64linux |
| 01:20.01 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 01:21.03 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 01:21.38 | kimosabe | is there a ds3 card under digium |
| 01:21.50 | kimosabe | or anything of this sort |
| 01:21.54 | SteveTotaro | sangoma has one |
| 01:22.09 | SteveTotaro | digium announced one but i never saw it for sale |
| 01:22.53 | kimosabe | stevetotaro whats the site for this card |
| 01:23.00 | *** join/#asterisk BeeBuu (n=beebuu@219.135.42.4) |
| 01:23.05 | SteveTotaro | google |
| 01:23.26 | kimosabe | will asterisk suport this card |
| 01:23.42 | SteveTotaro | it is not for voice i do not believe |
| 01:23.58 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.or.comcast.net) |
| 01:24.00 | SteveTotaro | it would be stupid to put a ds3 into an asterisk box |
| 01:24.20 | obnauticus | that's a nice message to join to |
| 01:24.23 | JT | it doesn't do channelising |
| 01:24.30 | JT | the sangoma card |
| 01:24.43 | SteveTotaro | he said he just had a raw data ds3 |
| 01:24.44 | JT | it's pretty much just for data router use |
| 01:25.09 | obnauticus | get a juniper |
| 01:25.10 | obnauticus | :) |
| 01:25.24 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 01:25.39 | *** join/#asterisk Inssomniak (n=Dave@206-248-139-111.dsl.teksavvy.com) |
| 01:25.49 | SteveTotaro | i have a Juniper credit card |
| 01:25.57 | Inssomniak | Hey all.. has anyone heard of freephoneline.ca and know if I can use it with asterisk? |
| 01:26.04 | obnauticus | ... |
| 01:26.38 | obnauticus | Inssomniak do they provide sip termiantion>? |
| 01:26.51 | Inssomniak | obnauticus, Its so far hard to tell |
| 01:26.51 | drmessano | I like boxwoods better.. they really fill out the yard |
| 01:27.06 | kimosabe | stevetotaro you have a juniper card at the time |
| 01:27.08 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 01:27.52 | *** join/#asterisk bmg505 (n=leon@196-209-76-155-tbnb-esr-2.dynamic.isadsl.co.za) |
| 01:29.25 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@76-204-200-226.lightspeed.hstntx.sbcglobal.net) |
| 01:31.55 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 01:33.28 | generalhan | ok, i have been all over the internet to figure this one out and am coming up short. i followed the instructions at http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation to the letter and im having issues with the voicemail section when trying to record unavail messages, and when trying to leave messages for users |
| 01:33.43 | generalhan | can some one take a look at these WARNING messages and give me some insight please !? http://pastebin.com/d9fa90cc |
| 01:34.19 | vap0rtranz | <PROTECTED> |
| 01:37.22 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:37.23 | [hC] | what might be the cause, if calling someone on my pbx who's on a phone at home slowly gets more and more lagged (audio wise) as the call goes on, and we have to hang up and call back to start fresh? |
| 01:37.48 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 01:38.56 | vap0rtranz | but i do luv how Allison says "all lines are currently unavailable" |
| 01:38.59 | vap0rtranz | always wanted to do that |
| 01:40.30 | generalhan | i guess ill give it another shot tomorrow, my brain is fried ! see you all tomorrow ! |
| 01:42.21 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 01:42.42 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:42.42 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:44.39 | TJNII | [hC]: How are they connected? What devices sit between their phone and your PBX? |
| 01:45.47 | *** join/#asterisk xpotx (n=james@c-67-186-193-35.hsd1.ut.comcast.net) |
| 01:45.59 | BeeBuu | drmessano: hi,there? |
| 01:46.21 | xpotx | hello |
| 01:47.30 | *** join/#asterisk xpot (n=james@c-67-186-193-35.hsd1.ut.comcast.net) |
| 01:48.35 | [hC] | TJNII: well, one guy is on a sip phone on comcast(heh!) in florida, and he comes to me via sip in canada, and im locally connected. i realize anything can play a role but im curious how audio lag CAN be introduced |
| 01:51.40 | *** join/#asterisk SteveTotaro (n=root@pool-71-179-121-73.bltmmd.east.verizon.net) |
| 01:52.29 | SteveTotaro | ok anyone good at writing copy and web design? |
| 01:53.25 | *** join/#asterisk xpot (n=root@c-67-186-193-35.hsd1.ut.comcast.net) |
| 01:53.36 | SteveTotaro | good for a little extra ca$h (but not much) |
| 01:53.44 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 01:58.39 | *** join/#asterisk shmaltz (n=chatzill@mail2.dmaven.com) |
| 02:00.13 | SteveTotaro | is this thing on? |
| 02:01.00 | BeeBuu | ~rule |
| 02:01.41 | jbot | rule is probably play nice. |
| 02:01.41 | shmaltz | SteveTotaro, thanks for responding to my question about swtichvox |
| 02:02.14 | SteveTotaro | oh, no problem, i am a big fan |
| 02:02.34 | shmaltz | stevetotaro, the question was not really how to work around and get ssh access, but if it's supported. Knowing asterisk means that for troubleshooting I should be able to login to the CLI to see whats going on |
| 02:02.45 | SteveTotaro | a big fan of switchvox and of getting around "locks" |
| 02:02.51 | shmaltz | SteveTotaro, have you tried it yourself in production? |
| 02:03.04 | SteveTotaro | yes |
| 02:03.12 | SteveTotaro | long ago |
| 02:03.20 | shmaltz | AFAIK it's missing a provisioning system for phones |
| 02:03.46 | SteveTotaro | a year and a half ago, it must be really great now because it was just great then |
| 02:04.14 | *** join/#asterisk lka (n=anders@unaffiliated/lka) |
| 02:04.20 | SteveTotaro | not sure about that but switchvox will configure and ship the phones with the system |
| 02:04.25 | SteveTotaro | i was a var, not sure anymore |
| 02:06.36 | SteveTotaro | if you want a really plug and play system get a 3com v3000 as long as you don't need more than four lines it competes well with asterisk |
| 02:07.18 | SteveTotaro | plug in a new phone and it downloads it's firmware from the PBX and takes the next available extension automagically |
| 02:07.18 | shmaltz | SteveTotaro, this is for a business with a PRI |
| 02:07.31 | shmaltz | so I don't think it's an option |
| 02:07.42 | shmaltz | personally I am pushing for a Panasonic TDE system |
| 02:08.00 | SteveTotaro | well then 3com gets a little more expensive you would need to purchase an nbx chassis and a T1 card |
| 02:08.09 | shmaltz | for simple KEY System it's way better than asterisk |
| 02:08.34 | SteveTotaro | chassis is only ~$300 and the T1 card is ~$4k if memory serves me correctly |
| 02:08.55 | shmaltz | this is interersting: |
| 02:08.57 | shmaltz | http://www.google.com/search?hl=en&q=pbx&btnG=Google+Search |
| 02:08.59 | shmaltz | asterisk is 3rd answer |
| 02:09.04 | shmaltz | sorry 2nd answer |
| 02:09.29 | shmaltz | wow, 4k for a T1 card? |
| 02:09.33 | SteveTotaro | Qwell is in charge of SEO |
| 02:09.33 | shmaltz | how is that justified? |
| 02:09.43 | SteveTotaro | that is old skool |
| 02:09.46 | Qwell | huh? |
| 02:09.50 | SteveTotaro | onboard dsps |
| 02:09.56 | Qwell | I'm in charge of SEO? |
| 02:10.18 | SteveTotaro | yean, and doing a hell of a job look at the google search for pbx |
| 02:10.18 | shmaltz | are you not in charge of SEO? |
| 02:10.43 | Qwell | what's SEO? |
| 02:10.43 | shmaltz | well, I think it's the only article - other than PBX itself - that has PBX in the titile on wp |
| 02:10.47 | russellb | what is SEO? |
| 02:10.48 | Qwell | I mean...I might be in charge of it |
| 02:10.51 | russellb | heh |
| 02:10.59 | SteveTotaro | search engine optimization |
| 02:11.03 | russellb | oic |
| 02:11.21 | SteveTotaro | that is actually quite a feat |
| 02:11.31 | russellb | it was #1 for the longest time ... |
| 02:11.52 | Qwell | stupid wikipedia |
| 02:11.59 | SteveTotaro | Qwell, you are fired from SEO |
| 02:12.30 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 02:13.17 | SteveTotaro | i like how "phone system" always has buyerzone come up in one or two |
| 02:13.36 | SteveTotaro | buyerzone is a great source for leads but expensive |
| 02:13.46 | SteveTotaro | $24 per lead |
| 02:14.29 | *** join/#asterisk SomethingISOdd (i=TestMast@S010600a0d1757bfb.cg.shawcable.net) |
| 02:14.38 | lka | i have a situation (in the US) where my telco requires that i dial certain numbers in my area code as 7 digits, and others as 10 digits. the list of blocks of numbers where i have to do this is quite long. is there a graceful way to handle this in the dialplan? |
| 02:14.45 | SomethingISOdd | hello all question is there anyway to route all voip traffic using ip tables to another internal server? |
| 02:15.07 | lka | without making it 1000 lines long |
| 02:15.08 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 02:16.03 | rickross | hi all, I have a bizarre set of translation timings being returned by "core show translation" after installing asterisk 1.6b4 |
| 02:16.19 | rickross | is it OK to paste in here? (worried about flooding) |
| 02:16.22 | Qwell | rickross: yes, it changed. that's normal |
| 02:16.25 | shmaltz | 1ka, that is impossible |
| 02:16.51 | Qwell | ~pb |
| 02:17.02 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 02:17.02 | lka | shmaltz: sigh |
| 02:17.07 | shmaltz | meaning it's impossible that the provider should require in the same are code different ways of dialing and in fact it's illegal |
| 02:17.08 | Qwell | ^^ if you must |
| 02:17.27 | lka | people do illegal things every day |
| 02:17.33 | Qwell | shmaltz: how do you figure it's illegal? |
| 02:17.43 | SteveTotaro | i got a speeding ticket two weeks ago |
| 02:17.47 | Qwell | that's quite common |
| 02:17.54 | shmaltz | Qwell, there are FCC rules on 7 digit or 10 digit dialing |
| 02:17.58 | Qwell | show me |
| 02:18.17 | rickross | thx Qwell, even this crazy? http://pastebin.ca/927199 |
| 02:18.35 | Qwell | rickross: yep |
| 02:18.44 | Qwell | it's weird, but normal, if you understand how it works |
| 02:18.53 | SteveTotaro | you could try dialing the seven digit version first and if it fails then the second priority dial the ten digit |
| 02:19.16 | shmaltz | SteverTotaro, or get a provider thats sober |
| 02:19.18 | Qwell | lka: and to answer your question - no, not really... |
| 02:19.22 | rickross | hmm, ok, is there any secret to getting g.722 to translate correctly? |
| 02:19.35 | SteveTotaro | i have to send 1+10 |
| 02:19.35 | Qwell | rickross: what do you mean correctly? |
| 02:19.38 | rickross | I had it working fine with a backport into a 1.4.x version |
| 02:19.48 | shmaltz | rickross, is that on an 8 bit proccessor? |
| 02:19.57 | lka | SteveTotaro: well if the pstn line fails it just dials out over ip, which works fine with 1+10 every time, every number |
| 02:19.58 | rickross | but now it sounds like I am under water if I call a non-g.722 phone |
| 02:20.01 | lka | but local calls are free over the pstn |
| 02:20.27 | SteveTotaro | maybe you can escalate the issue with your provider |
| 02:20.38 | shmaltz | 1ka, where is this? |
| 02:20.44 | lka | north carolina |
| 02:20.46 | lka | usa |
| 02:20.49 | Qwell | shmaltz: many parts of the US? |
| 02:21.02 | SteveTotaro | but how many seven digit area codes can there be in NC? |
| 02:21.06 | shmaltz | 1ks, who is the provider? |
| 02:21.15 | rickross | Qwell, when I have g.722 enabled as a codec it sounds awful to people on the other end of the line (unless they also have a g.722 phone) - we're using Polycom 550s |
| 02:21.18 | lka | bellsouth |
| 02:21.50 | SteveTotaro | in the dc area before everything went ten digit there were only like five area codes that were seven digit |
| 02:21.59 | SteveTotaro | in a big metro area |
| 02:22.21 | rickross | Qwell, we have been using g.722 transcoding with good success under a 1.4.x backport |
| 02:22.38 | SteveTotaro | you know what i hate, i cannot set my caller id to my toll free and make certain calls |
| 02:22.44 | rickross | the problems seem to be connected to the 1.6.0b4 server |
| 02:23.02 | SteveTotaro | what does the b stand for? |
| 02:23.06 | shmaltz | SteverTotaro, use real provider and you'll be able to do that |
| 02:23.18 | SteveTotaro | it's not the provider |
| 02:23.20 | lka | (XXX) YYY-ZZZZ |
| 02:23.24 | lka | what is the YYY called? |
| 02:23.41 | lka | i have about 80 of those that must be dialed as 7 digits |
| 02:23.44 | SteveTotaro | it is the other side called, if i call another toll free they will block many times |
| 02:23.49 | SteveTotaro | not sure how to bill |
| 02:24.19 | SteveTotaro | i have qwest, they are decent |
| 02:24.27 | djs | lka - the prefix? |
| 02:24.29 | SteveTotaro | ~qwest |
| 02:24.30 | jbot | methinks qwest is a company with secksie backbones but lame peering (www.qwest.net). or a company that randomly scrambles routes and pisses off network engineers worldwide |
| 02:24.42 | lka | djs: thanks |
| 02:25.00 | SteveTotaro | ~verizon |
| 02:25.01 | jbot | Verizon is utter garbage. Do yourself a favor and stay away from that company. |
| 02:25.05 | djs | hah |
| 02:25.17 | SteveTotaro | ~ucn |
| 02:25.17 | djs | ~bellsouth |
| 02:25.33 | SteveTotaro | you guys need to check out UCN's in contact product |
| 02:25.38 | rickross | schmaltz - sorry - no, it is a Core2Duo |
| 02:25.44 | SteveTotaro | they put the PBX in the cloud |
| 02:25.50 | SteveTotaro | the call center in the cloud |
| 02:25.51 | lka | my current plan is to just have a shell script generate the same 10 lines in my dialplan like 80 times, substituting the different prefixes each time |
| 02:26.01 | lka | but its so ugly it offends me |
| 02:26.07 | djs | heh |
| 02:26.18 | SteveTotaro | macro |
| 02:27.04 | shmaltz | Steve, Macro will still require 80 lines |
| 02:27.10 | SteveTotaro | ever have a pimple inside your nose? MFer it hurts |
| 02:27.18 | lka | 800, actually |
| 02:27.30 | SteveTotaro | oh well, deal with it |
| 02:28.04 | shmaltz | 1ka, therer are only 800 exchanges in an area code, so how can it be 800 lines? |
| 02:28.16 | SteveTotaro | put it in an include so you don't have to see it |
| 02:28.30 | *** join/#asterisk jamesrdorn (n=jamesrdo@adsl-99-135-235-94.dsl.rcsntx.sbcglobal.net) |
| 02:28.33 | lka | well 80-100 prefixes need special consideration |
| 02:29.12 | lka | and i have maybe 10 lines for each (callerid, and some failover stuff) |
| 02:29.12 | shmaltz | 1ka, then just define those |
| 02:29.12 | SteveTotaro | making a mountain out of a molehill |
| 02:29.12 | shmaltz | 1ka, use macro for those 80-100 prefixes |
| 02:29.15 | SteveTotaro | that is what i suggested |
| 02:29.30 | SteveTotaro | he said it would take 800 lines |
| 02:29.51 | lka | can you give me a 10 second example? i looked at some documentation for macros but it didnt really get me anywhere |
| 02:30.09 | SteveTotaro | i charge by the hour |
| 02:30.14 | SteveTotaro | for the first hour |
| 02:30.20 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 02:30.30 | SteveTotaro | then half hour increments after that |
| 02:31.47 | *** join/#asterisk tobias (n=tobias@cpe-076-182-057-234.nc.res.rr.com) |
| 02:32.20 | SteveTotaro | ~jbot |
| 02:32.21 | jbot | i guess jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch, or suck |
| 02:32.32 | shmaltz | 1ka, look at this: |
| 02:32.34 | shmaltz | http://www.pastebin.ca/927208 |
| 02:33.02 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 02:33.19 | shmaltz | ~sex |
| 02:33.20 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/ |
| 02:33.23 | shmaltz | ~gender |
| 02:33.24 | jbot | I'm gay |
| 02:33.31 | shmaltz | ~pregnant |
| 02:33.31 | jbot | Yes, shmaltz, and it's your child. |
| 02:33.40 | SteveTotaro | ~fedora |
| 02:33.40 | jbot | somebody said fedora was stevetotaro is <action> tells stevetotaro that becoming RHCP certified, is The One True Way to Supreme Linux Knowledge |
| 02:33.45 | *** join/#asterisk orbi (n=orbi@68-119-116-31.dhcp.jcsn.tn.charter.com) |
| 02:34.07 | *** join/#asterisk lyroy (n=lyroy@bas1-montreal02-1096716687.dsl.bell.ca) |
| 02:34.21 | lka | ahh, i see what you guys mean |
| 02:34.50 | SteveTotaro | but do you mean what you see? |
| 02:35.08 | lka | thanks shmaltz |
| 02:35.26 | shmaltz | SteveTotaro, you can send him a bill now |
| 02:35.42 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 02:35.48 | lka | i could send you a check but i cant afford the postage |
| 02:36.04 | shmaltz | 1ka, ever heard of PayPal? |
| 02:37.14 | lka | i cant afford internet access either |
| 02:37.28 | SteveTotaro | who pays for internet access these days? |
| 02:37.49 | SteveTotaro | just get a yagi and aircrack |
| 02:37.50 | lyroy | I'm having an issue with the IAX Realtime module, the problem is that I have an entry like this in my database for the dialout of my users (...name=1112223333-TOPSTN).. asterisk always add a random number after the name (name=1112223333-TOPSTN-4)... so when the query is made in the database SELECT * FROM iax WHERE name = '1112223333-TOPSTN-4' AND host = 'dynamic' it cause an error because the real entry is 1112223333-TOPSTN and not |
| 02:38.08 | shmaltz | I like this one: |
| 02:38.09 | shmaltz | http://www.youtube.com/watch?v=uSmwJ6OgWko |
| 02:38.50 | shmaltz | or this one: |
| 02:38.52 | shmaltz | http://www.youtube.com/watch?v=J5z4Vs26-TI&NR=1 |
| 02:38.52 | SteveTotaro | i still can't get fc8 to play youtube on my core2duo |
| 02:39.50 | SteveTotaro | Hello, you either have JavaScript turned off or an old version of Adobe's Flash Player. Get the latest Flash player. |
| 02:39.58 | shmaltz | SteveTotaro, you have flash installed? |
| 02:40.09 | SteveTotaro | but there is no 64bit flash player |
| 02:40.33 | SteveTotaro | i tried some googling and howtos but nothing worked so i gave up |
| 02:40.52 | SteveTotaro | it actually makes me more productive not having youtube ;) |
| 02:41.10 | shmaltz | then try this site: |
| 02:41.11 | shmaltz | http://m.youtube.com/warning?next=/ |
| 02:41.13 | shmaltz | it shoudl work |
| 02:41.45 | shmaltz | just get an rtsp player |
| 02:42.01 | jameswf-home | ! WIZARD ! |
| 02:42.06 | SteveTotaro | XML Parsing Error: mismatched tag. Expected: </tr>. |
| 02:42.06 | SteveTotaro | Location: http://m.youtube.com/warning?next=/ |
| 02:42.06 | SteveTotaro | Line Number 19, Column 4: |
| 02:42.46 | SteveTotaro | </table> |
| 02:42.47 | SteveTotaro | ----------^ |
| 02:42.47 | *** join/#asterisk bkw_ (n=brian@adsl-70-234-168-136.dsl.tul2ok.sbcglobal.net) |
| 02:43.29 | shmaltz | SteverTotaro, yes thats a mozilla error hold on I'll give you a new link |
| 02:43.46 | shmaltz | here we go: |
| 02:43.48 | shmaltz | http://m.youtube.com/?warned=yes |
| 02:44.16 | *** join/#asterisk bkw___ (n=brian@adsl-64-149-54-142.dsl.tul2ok.sbcglobal.net) |
| 02:44.39 | SteveTotaro | 2g1cup |
| 02:45.07 | rickross | OK, here's someone else with my exact problem - http://www.spinics.net/lists/asterisk/msg84469.html - Transcoded G.722 calls unintelligible with recent SVN head |
| 02:45.08 | SteveTotaro | thanks for the link though |
| 02:45.19 | rickross | helps me feel like I'm not completely crazy ;) |
| 02:45.24 | SteveTotaro | stick with 1.2 |
| 02:45.44 | russellb | rickross: with trunk / 1.6? |
| 02:45.51 | SteveTotaro | 1.4 if you are feeling crazy |
| 02:46.09 | rickross | russellb - nope, with 1.6.0b4 |
| 02:46.14 | russellb | rickross: close enough |
| 02:46.16 | SteveTotaro | 1.6BETA if you want to live on bugtracker |
| 02:46.19 | rickross | should I pull it form svn and try again? |
| 02:46.20 | *** join/#asterisk letale (n=echosyp@75.111.172.173) |
| 02:46.24 | russellb | rickross: file a bug report and assign it to me, and i'll take a look and fix it |
| 02:46.41 | rickross | russellb - thx |
| 02:46.49 | shmaltz | this is funny: |
| 02:46.51 | shmaltz | http://www.youtube.com/watch?v=svEPX2GpoXY |
| 02:46.55 | russellb | rickross: i have a hunch ... |
| 02:47.06 | rickross | russellb - does this look right to you? |
| 02:47.07 | letale | guys, im really n00b, i want to setup asterisk, but i want a good frontend to help me manage things |
| 02:47.07 | rickross | http://pastebin.ca/927199 |
| 02:47.14 | lyroy | Please someone ... I'm having an issue with the IAX Realtime module, the problem is that I have an entry like this in my database for the dialout of my users (...name=1112223333-TOPSTN).. asterisk always add a random number after the name (name=1112223333-TOPSTN-4)... so when the query is made in the database SELECT * FROM iax WHERE name = '1112223333-TOPSTN-4' AND host = 'dynamic' it cause an error because the real entry is 1112223 |
| 02:47.18 | letale | what do you suggest |
| 02:47.32 | SteveTotaro | buy a switchvox system |
| 02:47.40 | SteveTotaro | great front end |
| 02:47.43 | russellb | rickross: i guess |
| 02:47.51 | letale | i have an unused box, and no money |
| 02:47.59 | letale | its running ubuntu 7.10 |
| 02:48.11 | SteveTotaro | an unused box is really a shame in more ways than one |
| 02:48.32 | letale | thats why i want to put it to use |
| 02:48.36 | letale | with asterisk |
| 02:48.57 | letale | but i can't stand sorting through a million conf files |
| 02:49.09 | letale | which im sure im about to get shit for |
| 02:49.40 | letale | but, never the less, id like a good front end |
| 02:49.41 | russellb | jblack: give letale shit |
| 02:49.46 | russellb | oh well. |
| 02:49.54 | russellb | jblack: not you, lol ... stupid tab completion |
| 02:50.00 | russellb | jbot: give letale shit |
| 02:50.01 | jbot | ACTION gives shit to letale |
| 02:50.04 | russellb | hehe. |
| 02:50.14 | letale | hah |
| 02:50.25 | russellb | letale: it's fine ... plenty of people aren't interested in configuring it manually. |
| 02:50.39 | letale | its so tedious |
| 02:50.40 | SteveTotaro | ~trixbox |
| 02:50.41 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 02:51.25 | letale | i may as well try it out |
| 02:51.31 | russellb | letale: i would recommend asterisknow or switchvox |
| 02:51.45 | russellb | switchvox has a free version, too if your install is small enough |
| 02:51.57 | russellb | or at least to try it out |
| 02:51.57 | letale | im all about free |
| 02:52.07 | russellb | asterisknow is another free option ... |
| 02:52.33 | *** join/#asterisk jamesrdorn (n=jamesrdo@adsl-99-135-235-94.dsl.rcsntx.sbcglobal.net) |
| 02:54.37 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 02:54.47 | letale | i'll try trixbox and asterisknow |
| 02:54.58 | BeeBuu | i can't find how to say how many people in room when get in meet,anyone help me please? |
| 02:55.54 | seanbright-home | BeeBuu: the 'c' option |
| 02:56.09 | russellb | letale: i would steer clear of trixbox... |
| 02:56.10 | BeeBuu | i got you,thanks |
| 02:56.16 | seanbright-home | BeeBuu: np |
| 02:56.53 | SteveTotaro | www.easyvoxbox.org is the shiznit |
| 02:57.06 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 02:57.48 | SteveTotaro | i can sell you an unlimited key to signate and give you their repos |
| 02:57.54 | rickross | russellb: apparently I don't have the required privs to assign the issue - http://bugs.digium.com/bug_view_page.php?bug_id=12130 |
| 02:58.01 | SteveTotaro | you just have to change your eth0 mac |
| 02:58.13 | *** part/#asterisk letale (n=echosyp@75.111.172.173) |
| 02:58.23 | russellb | rickross: assigned to myself |
| 02:58.27 | rickross | thx |
| 02:58.40 | rickross | if you'd like to hear it, I can give you a ring |
| 02:58.55 | rickross | I dunno if that would clue you into what is wrong? |
| 02:59.02 | SteveTotaro | it is 10 pm |
| 02:59.13 | J4k3 | SteveTotaro: bleh, easyvoxbox is commercial? |
| 02:59.18 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 02:59.24 | SteveTotaro | is it? |
| 02:59.24 | J4k3 | lame, it looks like trixbox-a-year-ago |
| 02:59.44 | J4k3 | err, I misunderstood |
| 02:59.51 | SteveTotaro | i thought is was just asterisk+freepbx without the bloat |
| 03:00.16 | J4k3 | you said something about selling a key right after ;) |
| 03:00.16 | SteveTotaro | oh, no signate |
| 03:00.24 | SteveTotaro | i have an unlimited key and it just checks the mac of eth0 |
| 03:00.57 | bkw___ | SteveTotaro: you used to work for signate? |
| 03:00.58 | SteveTotaro | then they used ioncube to scramble all their code |
| 03:01.22 | SteveTotaro | heck no, i started to work for a company that bought a signate system |
| 03:01.30 | bkw___ | hehe |
| 03:01.31 | SteveTotaro | paid $50k |
| 03:02.00 | SteveTotaro | $5k for answering machine detection which was exactly the dialplan stuff on the wiki |
| 03:02.00 | errr | well per y'alls suggestion of moving frmo iax to sip now my incoming calls are clear as can be. thanks |
| 03:02.21 | J4k3 | hmm EVB looks like what I've been looking for |
| 03:02.24 | SteveTotaro | told ya so |
| 03:02.30 | SteveTotaro | iax is teh suck |
| 03:02.36 | errr | well now I know |
| 03:02.42 | errr | it made a huge diff |
| 03:03.05 | bkw___ | SteveTotaro: how can you get away with saying that? I say anything sucks.. I get called a troll |
| 03:03.09 | SteveTotaro | j4k3, glad i showed you something useful |
| 03:03.30 | SteveTotaro | because i am SteveTotaro |
| 03:03.37 | SteveTotaro | and you are Brian K West |
| 03:03.50 | bkw___ | hehe |
| 03:03.52 | J4k3 | woah, brian k west... the guy from OK? |
| 03:03.59 | bkw_ | J4k3: yah |
| 03:04.00 | SwK | haha |
| 03:04.09 | bkw_ | this can't go well |
| 03:04.13 | SteveTotaro | sorry for blowing your cover |
| 03:04.15 | J4k3 | bkw: haha... old school. |
| 03:04.24 | SwK | bkw_, its cause everyone knows you are a troll |
| 03:04.32 | bkw_ | SwK: oh yes I'm such a troll |
| 03:04.33 | iamthelostboy | hi.. we're getting pretty bad echo over zap -> sip calls.. when i do a ztmonitor 1 -vv and make a call, i get a whole lot of nothing happening... what am i supposed to be seeing? our calls also seem to be very quiet to a lot of people, they complain they cant hear us... |
| 03:04.38 | jameswf-home | ~troll |
| 03:04.38 | jbot | extra, extra, read all about it, troll is a race on some muds; a guy under a bridge: an annoying robot; a port scanner, or somebody faking being clueless to be shown the One Linux Way so he can argue against it for his psychology thesis on linux advocates, or an employee of trolltech, or at http://www.kuro5hin.org/story/2001/7/27/51233/2979 or ... |
| 03:05.08 | bkw_ | J4k3: and you are? |
| 03:05.12 | jameswf-home | he prefers tee-roll |
| 03:05.22 | SteveTotaro | troll baiter |
| 03:05.29 | bkw_ | that don't sound right |
| 03:05.34 | bkw_ | sounds a bit dirty |
| 03:05.38 | jameswf-home | master-troll-bater |
| 03:06.04 | SteveTotaro | imagine your mother with two big black guys |
| 03:06.22 | SteveTotaro | if you think something dirty then you are a perv |
| 03:06.22 | jameswf-home | oh heck SteveTotarowe call that thursday night |
| 03:06.50 | SteveTotaro | i picture them helping her with her groceries |
| 03:07.03 | SteveTotaro | it is all in your mind |
| 03:07.09 | riddlebox | hey SteveTotaro |
| 03:07.16 | riddlebox | sorry I missed you |
| 03:07.23 | SteveTotaro | you didn't i am here |
| 03:07.27 | SteveTotaro | what's up? |
| 03:07.44 | riddlebox | nothing much, just getting a few things tidy'd up |
| 03:08.05 | SteveTotaro | any feedback on that thing? |
| 03:08.43 | J4k3 | bkw_: you were in... hrm... #inet-access? |
| 03:08.52 | bkw_ | oh yes |
| 03:08.56 | bkw_ | tiz me |
| 03:08.59 | J4k3 | yep |
| 03:09.08 | bkw_ | mrunix was mah boss around that time |
| 03:10.41 | drmessano | IAX is the next WWW |
| 03:10.59 | SteveTotaro | iax is the ipv10 |
| 03:11.11 | lyroy | Please someone ... I'm having an issue with the IAX Realtime module, the problem is that I have an entry like this in my database for the dialout of my users (...name=1112223333-TOPSTN).. asterisk always add a random number after the name (name=1112223333-TOPSTN-4)... so when the query is made in the database SELECT * FROM iax WHERE name = '1112223333-TOPSTN-4' AND host = 'dynamic' it cause an error because the real entry is 1112223 |
| 03:11.27 | drmessano | SIP is !!!11!!!!! and IAX is ^^^^^^6^^^^ |
| 03:11.50 | riddlebox | SteveTotaro, I have been so busy I havent been able to even look at it |
| 03:13.02 | *** join/#asterisk cowmix (n=mmarch@71-209-212-132.phnx.qwest.net) |
| 03:13.59 | SteveTotaro | it's cool, busy is good |
| 03:14.09 | SteveTotaro | just let me know when you get around to it |
| 03:14.16 | SteveTotaro | any feedback is good feedback |
| 03:14.19 | J4k3 | iax is the vista |
| 03:14.22 | J4k3 | VISTAR |
| 03:14.36 | J4k3 | I'm now refering to people who like vista as 'vistards' |
| 03:14.36 | drmessano | LOL |
| 03:14.51 | J4k3 | they're equally as annoying as 'mac weenies' |
| 03:15.00 | drmessano | Vista SP2 is gonna make all you haters eat teh dust |
| 03:15.04 | drmessano | MARK MAH WARDS |
| 03:15.05 | J4k3 | sp2? sheeit |
| 03:15.09 | J4k3 | 2011? 2012? |
| 03:15.10 | J4k3 | :) |
| 03:15.15 | bkw_ | oh what ever |
| 03:15.16 | J4k3 | sp1 is taking long enough |
| 03:15.17 | bkw_ | every OS sucks |
| 03:15.22 | bkw_ | just depends on to what degree |
| 03:15.29 | J4k3 | bkw_: this is true, but vista has some real show-stoppers. |
| 03:15.34 | drmessano | SP2 IS GONNA BE TEH UNIX OF WINDOZ |
| 03:15.39 | bkw_ | more like SLOW ASS stoppers |
| 03:15.47 | J4k3 | haha |
| 03:15.55 | bkw_ | drmessano: it will NEVER be the unix of doze |
| 03:16.01 | J4k3 | being slow is a feature. |
| 03:16.02 | bkw_ | they must rewrite it |
| 03:16.04 | J4k3 | it sells faster systems |
| 03:16.42 | J4k3 | who'd need a quad core P51-3.1337 processor if the base software was written halfway as efficiently as win98se or even NT4 |
| 03:16.45 | drmessano | Vista SP2 is gonna make Windows 2000 look like 3.1 |
| 03:16.56 | bkw_ | drmessano: sorry but windows 2000 was the best |
| 03:17.00 | Qwell | drmessano: going to make it awesome? |
| 03:17.09 | J4k3 | haha win2k is so old looking now |
| 03:17.16 | bkw_ | I like the look |
| 03:17.18 | J4k3 | the icons are oldschool |
| 03:17.20 | J4k3 | I like it |
| 03:17.24 | bkw_ | me too |
| 03:17.51 | drmessano | I'm trying to get a job as a core developer of Vista SP2, so suck it you guys |
| 03:18.12 | J4k3 | drmessano: hope it pays well, most jobs in futility are. |
| 03:18.13 | J4k3 | :) |
| 03:18.17 | J4k3 | err |
| 03:18.18 | J4k3 | are/do |
| 03:18.42 | *** join/#asterisk letale (n=echosyp@75.111.172.173) |
| 03:19.05 | drmessano | I'm hoping to get AsteriskWin32 added as a core app in Windows 2008 SP1 and Vista SP2 |
| 03:19.11 | drmessano | It will make telephony easier |
| 03:19.20 | letale | how do you change the admin password for asterisk |
| 03:19.29 | letale | cause i forgot it |
| 03:19.34 | Qwell | ~asterisknow |
| 03:19.35 | jbot | asterisknow is probably based on Asterisk, but it is not Asterisk, and it is unlikely to live up to Asterisk's standards. Only Asterisk is supported on #asterisk. Use #AsteriskNow instead. Even if the channel happens to be less helpful, support for systems other than Asterisk is offtopic on #asterisk |
| 03:19.40 | bkw_ | I would say something but people would say i'm a troll |
| 03:19.52 | Qwell | bkw_: you are a troll |
| 03:19.56 | bkw_ | no i'm not |
| 03:20.04 | Qwell | I'm a troll |
| 03:20.32 | Idle | yes you are |
| 03:20.47 | drmessano | bkw_: You are a troll in serious denial |
| 03:21.03 | Juggie | bkw_, might be a hard ass, but he is not a troll |
| 03:21.22 | drmessano | Hmm |
| 03:21.29 | drmessano | Now THAT sounds like trolling |
| 03:22.24 | letale | so how do i change the admin password? |
| 03:23.12 | drmessano | letale: Surely you have this room confused with one that deals with Asterisks with admin passwords |
| 03:23.23 | drmessano | #asterisknow may be a better choice |
| 03:23.40 | drmessano | Unless of course, you're using Trixbox.. in which case.. |
| 03:23.46 | letale | im not using asterisknow |
| 03:23.50 | drmessano | ~WIZARD |
| 03:23.51 | jbot | wizard is, like, enchancement to howto's |
| 03:24.10 | letale | i used the ubuntu repo to install asterisk awhile back |
| 03:25.04 | drmessano | Um |
| 03:25.12 | drmessano | Which admin password? |
| 03:25.14 | *** join/#asterisk b1ch0 (n=ralabiso@static-200-105-209-46.acelerate.net) |
| 03:26.00 | drmessano | ~uhtrixbox |
| 03:26.01 | jbot | WIZARD! |
| 03:26.19 | letale | idk what your getting at |
| 03:26.34 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 03:26.34 | letale | be blunt |
| 03:26.43 | letale | i installed gastman |
| 03:26.52 | letale | and its asking for a password |
| 03:26.55 | letale | but i forgot it |
| 03:27.03 | letale | sooo |
| 03:27.07 | letale | i need to reset it |
| 03:27.10 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 03:27.29 | jameswf-home | they rushed 2.6.0.2 out with zaptel still broke wtf |
| 03:27.41 | drmessano | lol |
| 03:27.52 | drmessano | They have NO idea WTF they're doing |
| 03:27.55 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 03:27.59 | drmessano | I very, truly believe that now |
| 03:28.38 | drmessano | They have a three stage beta cycle for every release |
| 03:28.41 | drmessano | 1. ZOMG |
| 03:28.44 | drmessano | 2. WTF |
| 03:28.47 | drmessano | 3. BBQ! |
| 03:28.49 | lyroy | Please someone ... I'm having an issue with the IAX Realtime module, the problem is that I have an entry like this in my database for the dialout of my users (...name=1112223333-TOPSTN).. asterisk always add a random number after the name (name=1112223333-TOPSTN-4)... so when the query is made in the database SELECT * FROM iax WHERE name = '1112223333-TOPSTN-4' AND host = 'dynamic' it cause an error because the real entry is 1112223 |
| 03:29.07 | jameswf-home | 4. RON PAUL 2008 |
| 03:30.20 | drmessano | I bet in 2.6.0.3 they'll forget to add Asterisk |
| 03:30.40 | letale | well i forgot my admin password |
| 03:30.42 | drmessano | "Heh, sorry guys.. we're only human... 2.6.0.3.1 will be released tomorrow" |
| 03:32.05 | ectospasm | 2.6.0.3?? |
| 03:32.53 | SteveTotaro | i want 2.8alpha |
| 03:33.02 | letale | so how do i change it |
| 03:33.11 | SteveTotaro | password for what? |
| 03:33.19 | SteveTotaro | trixbox? |
| 03:33.21 | letale | no |
| 03:33.26 | SteveTotaro | then what> |
| 03:33.28 | letale | admin pw for asterisk |
| 03:33.29 | drmessano | gastman |
| 03:33.34 | drmessano | Some GUI |
| 03:33.35 | letale | am i missing something here? |
| 03:33.39 | drmessano | not "ASTERISK" |
| 03:33.45 | letale | but it connects to asterisk |
| 03:33.46 | b1ch0 | guy i have a problem with a new pattern requirement ... user are dialing number starting by 07XXXXXXX or 0XX7XXXXXXX and i need to send just 7XXXXXXX to the trunk |
| 03:33.49 | SteveTotaro | what the heck is gastman |
| 03:33.53 | drmessano | Good god |
| 03:34.02 | b1ch0 | AND CANT DO 0|7 |
| 03:34.03 | drmessano | letale |
| 03:34.04 | SteveTotaro | that was me yesterday after eating chili |
| 03:34.09 | drmessano | LOL |
| 03:34.14 | jameswf-home | i use passizzlewordizzlefoshizzlemuhnizzle |
| 03:34.27 | drmessano | letale said he installed asterisk and soemthing called gastman |
| 03:34.52 | drmessano | Im assuming he needs the password for GASTMAN because Asterisk's admin password is _ |
| 03:35.20 | SteveTotaro | gastman looks cool, drag and drop |
| 03:35.48 | drmessano | Not enough chili |
| 03:36.06 | jameswf-home | i ate the chilli and got some bad gast man |
| 03:36.06 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 03:37.10 | drmessano | gastman looks too SMB for me.. Im waiting for the enterprise edition, gastmaster |
| 03:37.30 | SteveTotaro | check out UCN's in contact |
| 03:37.38 | SteveTotaro | it is better than aheeva |
| 03:37.55 | SteveTotaro | there is probably a .conf file for gastman somewhere |
| 03:38.13 | SteveTotaro | maybe in /etc/asterisk/gastman.conf? |
| 03:38.22 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 03:38.24 | SteveTotaro | or it may be in a mysqldb table |
| 03:38.31 | SteveTotaro | like freepbx |
| 03:39.15 | SteveTotaro | UCN has a great idea of offering PBX/Call center (very advanced) at the carrier level |
| 03:39.35 | *** part/#asterisk b1ch0 (n=ralabiso@static-200-105-209-46.acelerate.net) |
| 03:39.43 | *** join/#asterisk ectospasm (n=ectospas@c-71-207-229-248.hsd1.al.comcast.net) |
| 03:39.52 | *** part/#asterisk letale (n=echosyp@75.111.172.173) |
| 03:40.46 | SteveTotaro | i would call that the gastman enterprise edition |
| 03:41.27 | bkw_ | I hate broadvoice |
| 03:42.07 | SteveTotaro | i hate nufone |
| 03:42.25 | drmessano | I hate strawberry shortcake with not enough strawberries |
| 03:42.56 | SteveTotaro | ~hate |
| 03:42.57 | jbot | Oh, you hate your job? Why didn't you say so? There's a support group for that. It's called EVERYBODY, and they meet at the bar. --Drew Carey |
| 03:43.08 | drmessano | R O F L |
| 03:43.09 | jameswf-home | they removed the line from trixbox.conf to kill the audit tool... tisk |
| 03:43.36 | drmessano | So, you can't kill it now? |
| 03:43.42 | SteveTotaro | it was just a cron job right? |
| 03:44.07 | BeeBuu | drmessano: had you been auto call out? |
| 03:44.07 | drmessano | They probably have it compiled into the kernel now.. bastards |
| 03:44.25 | drmessano | What? |
| 03:44.33 | BeeBuu | auto call |
| 03:44.39 | drmessano | What about it |
| 03:44.53 | jameswf-home | you kill it the same way but if you dont know then it goes on living |
| 03:44.53 | BeeBuu | http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
| 03:45.05 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 03:45.31 | drmessano | Haven't messed with call files |
| 03:45.45 | BeeBuu | drmessano: how's your doing? |
| 03:45.53 | drmessano | Im ok |
| 03:46.21 | jameswf-home | drmessano: should play wistle blower |
| 03:46.23 | BeeBuu | how to do when you need auto call out? |
| 03:46.34 | SteveTotaro | use easyvoxbox, it is just asterisk and freepbx with no bloat or spy crap |
| 03:47.17 | jameswf-home | I use happyclownpbx... |
| 03:47.30 | jameswf-home | cause i am 1337 |
| 03:47.33 | drmessano | jameswf-home: link pls |
| 03:47.38 | drmessano | Can't find it on the forum |
| 03:47.46 | jameswf-home | find what? |
| 03:47.52 | drmessano | Is there a post for it? |
| 03:48.03 | ectospasm | BeeBuu: explain what you mean by "auto call out" |
| 03:48.12 | jameswf-home | no no one has caught it yet... I am just skimming a clean install |
| 03:48.19 | drmessano | ohhh |
| 03:48.33 | BeeBuu | ectospasm: i need call from asterisk |
| 03:48.33 | ectospasm | normally, that's done in some non-asterisk script, that generates a call file and dumps it into /var/spool/asterisk/outgoing |
| 03:48.38 | jameswf-home | I am not allowed to start flame wars |
| 03:48.54 | BeeBuu | ectospasm:i got that,but i got problem too |
| 03:49.03 | jameswf-home | ~ jameswf-home |
| 03:49.04 | jbot | when -home is added it means he is on his own time dont call his boss |
| 03:51.09 | *** join/#asterisk ahbritto (n=guest@adsl-68-125-197-181.dsl.pltn13.pacbell.net) |
| 03:51.17 | drmessano | So the line is not in the conf (which defaults the script to OFF) or the registry.pl isn't parsing that line anymore? |
| 03:51.54 | drmessano | BeeBuu: Have you read the book any? |
| 03:52.04 | BeeBuu | ectospasm: when a call file working and callee does not answer, i got this:http://rafb.net/p/Dd4j1b54.html |
| 03:52.26 | BeeBuu | drmessano: yes,thanks,but something beyond me,would you help? |
| 03:52.32 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 03:53.32 | BeeBuu | the callee do not answer,but why asterisk Playing 'conf-onlyperson'? |
| 03:53.43 | ectospasm | BeeBuu: because the calling channel is still on the call |
| 03:53.58 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 03:53.58 | drmessano | BeeBuu: No offense, but when I have tried to instruct you about different things, you don't seem to listen.. I don't lose weekends for anyone anymore :) |
| 03:54.06 | jameswf-home | no registry.pl but the whole /etc/trixbox/trixbox.conf is gone... |
| 03:54.20 | jameswf-home | s/registry.pl/registry.pl still looks/ |
| 03:54.33 | BeeBuu | drmessano: sorry,let me check above,too many msg... |
| 03:56.02 | drmessano | jameswf-home: The absense of the config item completely is supposed to = off |
| 03:56.10 | drmessano | The registry.pl script will check /etc/trixbox/trixbox.conf for AuditTool=yes, if this is set to ‘no’ or the value does not exist, no communication will take effect between the script and the Fonality servers. |
| 03:56.12 | BeeBuu | ectospasm: i want the asterisk system say how many people in current room when callee anwsered |
| 03:56.46 | ectospasm | Right... and it is! The call file's channel is the only person in the conference! |
| 03:56.53 | jameswf-home | previous versions had the file with audittool=on so it is on unless you change it... now they dont show you the path you have to google it |
| 03:56.53 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 03:57.25 | jameswf-home | hmm let me reread the perl |
| 03:57.37 | BeeBuu | ectospasm: only when callee anwser before ring 1 time |
| 03:57.55 | BeeBuu | OR callee hear nothing. |
| 03:58.28 | ectospasm | BeeBuu: I don't get it |
| 03:59.05 | BeeBuu | when the phone ringing.... |
| 03:59.25 | BeeBuu | if the callee not pick up right now |
| 03:59.52 | BeeBuu | can't get nothing voice.... |
| 04:00.13 | jameswf-home | if the file exists and there in no audittool line exit otherwise looks like it moves on and tries to create the file so they get atleast 1 phone home in... |
| 04:00.47 | drmessano | ah |
| 04:02.24 | jameswf-home | the blog was well worded that is why I rtfc |
| 04:03.23 | lanning | BeeBuu, you need to connect to a script that detects phone pickup before playing sound files. |
| 04:03.40 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 04:03.51 | BeeBuu | lanning: how? |
| 04:03.54 | drmessano | fascinating that they get in a phonehome before it disables itself |
| 04:04.00 | BeeBuu | lanning: any suggestion? |
| 04:05.00 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 04:06.02 | jameswf-home | well it doesnt disable its self simply adds a line to let you after round 1 |
| 04:06.24 | obnauticus | is it possible to have asterisk send a diffeernt client identifier? |
| 04:06.28 | obnauticus | or hardware identifier |
| 04:06.37 | drmessano | So it adds it as null, yes, or no? |
| 04:06.40 | obnauticus | i.e. change it from Asterisk PBX |
| 04:06.54 | jameswf-home | adds as yes |
| 04:07.04 | drmessano | Oh |
| 04:11.45 | obnauticus | anyway i need to make asterisk look liek it's registering to another peer as an ATA |
| 04:11.59 | obnauticus | because they make you pay less if you use an ATA with their services. |
| 04:13.45 | jameswf-home | someone gave kudos for slapping 1 broke over another |
| 04:13.58 | drmessano | lol |
| 04:15.14 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 04:15.46 | drmessano | useragent= |
| 04:16.01 | lanning | BeeBuu: you can try looking at DIALSTATUS variable or the DEVSTATE function |
| 04:16.12 | drmessano | http://www.voip-info.org/wiki/view/SIP+user+agent+identification |
| 04:17.41 | *** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net) |
| 04:19.47 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 04:21.26 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 04:21.33 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 04:22.48 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:23.01 | *** join/#asterisk vap0rtranz (n=jpittman@75.110.17.157) |
| 04:23.34 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:23.34 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:23.43 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 04:23.55 | *** join/#asterisk mir100 (n=mir100@adsl-76-193-179-244.dsl.chcgil.sbcglobal.net) |
| 04:24.19 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 04:24.26 | vap0rtranz | digium 2400 card has >3600 irq misses :( the pci bus must be to blame |
| 04:24.51 | drmessano | PCI is old and busted |
| 04:25.04 | bkw_ | no its not |
| 04:25.04 | vap0rtranz | *sniffles* |
| 04:25.24 | drmessano | Yeah it is |
| 04:25.51 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 04:26.07 | drmessano | PCI-E FTW |
| 04:26.46 | bkw_ | they'll change it again in a year or two |
| 04:26.51 | bkw_ | PCI-e |
| 04:26.56 | bkw_ | and PCI-X |
| 04:26.56 | drmessano | Doesn't matter |
| 04:27.01 | drmessano | PCI is dead |
| 04:27.07 | bkw_ | nice to call them the same name but they aren't the same |
| 04:27.24 | bkw_ | drmessano: oh shut up.. you know nothing if you think PCI is dead |
| 04:27.28 | bkw_ | its gonna be around for a LONG TIME |
| 04:27.44 | drmessano | First off.. I said PCI-E, not PCI-X |
| 04:27.44 | vap0rtranz | tape is dead! |
| 04:27.45 | vap0rtranz | lol |
| 04:27.50 | drmessano | and second, it's dead |
| 04:27.53 | bkw_ | its not PCI-E you must use the lower case e |
| 04:27.58 | drmessano | Troll |
| 04:28.06 | bkw_ | drmessano: if its dead why do people still produce products that use it? |
| 04:28.12 | *** part/#asterisk iamthelostboy (n=nathan@125-236-212-46.adsl.xtra.co.nz) |
| 04:28.29 | bkw_ | ISA is dead... PCI.. its got many more years left |
| 04:28.30 | drmessano | Same reason they produce ISA products.. Same reason we use Gasoline |
| 04:28.45 | bkw_ | its very hard to find mother boards with ISA slots |
| 04:29.05 | drmessano | Well, they still make them.. so that blows up your little theory |
| 04:29.20 | jameswf-home | bsd is da future |
| 04:29.25 | bkw_ | oh wtf ever |
| 04:29.35 | drmessano | bkw_: Are you a BSD user? |
| 04:29.48 | vap0rtranz | modprobe.conf:"alias eth1 wctdm24xxp". good lord! did kudzu do that?! |
| 04:29.49 | bkw_ | drmessano: I use all os's |
| 04:30.00 | drmessano | I'll take that as a "yes" |
| 04:30.09 | jblack | Yes! |
| 04:30.23 | jameswf-home | ron paul uses bsd |
| 04:30.28 | vap0rtranz | hah |
| 04:30.37 | jblack | Yes! |
| 04:30.50 | drmessano | Ron Paul maintains the ISA code for BSD |
| 04:30.51 | vap0rtranz | and i have a mboard with isa. 10+ years strong, yippie! |
| 04:30.55 | jblack | Yes! |
| 04:31.08 | drmessano | YADDIKI! |
| 04:31.16 | jblack | IDunno! |
| 04:31.41 | drmessano | ~idk |
| 04:31.42 | jameswf-home | IDK MY BFF Ron Paul |
| 04:31.50 | jblack | lol |
| 04:31.57 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 04:32.09 | jblack | The sequel to that commercial is running these days... Mom admits to having failed her child. |
| 04:32.23 | drmessano | Yep |
| 04:32.37 | jblack | Oh, I have an asterisk problem. |
| 04:32.52 | drmessano | Try Trixbox, n00b |
| 04:32.58 | bkw_ | drmessano: be nice |
| 04:33.01 | jameswf-home | its 3AM something has gone wrong.... when the phone rings do you want microsoft response point to answer |
| 04:33.04 | bkw_ | jblack: what kind of problem? |
| 04:33.05 | drmessano | bkw_: Piss off |
| 04:33.11 | jblack | It's an ironic one... "it just works". I have no problems to fix, no excuses for things to improve. |
| 04:33.18 | drmessano | lol |
| 04:33.27 | bkw_ | jblack: you haven't used it very hard yet |
| 04:33.28 | drmessano | jblack: Maybe you should try trixbox |
| 04:33.42 | drmessano | jblack: Would you like the URL? |
| 04:33.46 | jameswf-home | trixbox is 1337 yo |
| 04:33.48 | jblack | People call, my phone rings. I can out, their phone rings. What has started out exciting and amazing has turned into something exactly as dull as phone service. |
| 04:33.57 | jblack | basically, I feel robbed. |
| 04:34.09 | bkw_ | jblack: one phone ? |
| 04:34.09 | drmessano | jblack: Maybe you should beta test |
| 04:34.25 | vap0rtranz | actually, any dialed microsoft to move a license? she's all automated. recognized my voice entry for their damn long serial key. pretty nifty |
| 04:34.28 | jblack | A few phones. |
| 04:34.29 | drmessano | jblack: Trixbox is pretty |
| 04:34.31 | jameswf-home | someons should spoof the hillary/obama phone rings commercials with a voice over for asterisk |
| 04:34.31 | outtolunc | *=love |
| 04:34.37 | *** join/#asterisk PepOSX (n=angeldav@190.72.148.54) |
| 04:34.40 | jameswf-home | maybe allison |
| 04:34.44 | vap0rtranz | hah! |
| 04:34.48 | bkw_ | asterisk makes me sick |
| 04:34.54 | vap0rtranz | she didn't sound like allison; kinda older |
| 04:34.56 | russellb | bkw_: then leave, please. |
| 04:35.06 | bkw_ | russellb: how rude |
| 04:35.16 | drmessano | troll |
| 04:35.19 | russellb | bkw_: i said please, troll |
| 04:35.20 | jblack | Oh, another way that my life is b0rken.... |
| 04:35.20 | *** join/#asterisk ilowe (n=ilowe@modemcable014.189-201-24.mc.videotron.ca) |
| 04:35.25 | bkw_ | what ever |
| 04:35.39 | jblack | My 13 year old kid is at a bon jovi concert, while I sit here and watch Leno. |
| 04:35.41 | russellb | exactly my thoughts |
| 04:36.15 | jblack | I have the money; why is she having the fun? It's no fair! /me stomps his feet and pouts |
| 04:36.21 | drmessano | jblack: jon bon asterisk? |
| 04:36.31 | bkw_ | asterisk just doesn't fit every situation or solve every solution |
| 04:36.42 | jameswf-home | LIES |
| 04:36.47 | jblack | Nope. The one with the spikey hair... did ally mcbeal for a bit. |
| 04:36.51 | drmessano | bkw_: This is #ASTERISK you troll |
| 04:36.57 | bkw_ | jameswf-home: nothing ever solves every problem |
| 04:37.06 | russellb | nobody said it did. |
| 04:37.10 | jblack | bkw_: Hey, there's a rumor going around that you're a troll. There any truth to that? |
| 04:37.18 | bkw_ | jblack: no its not. |
| 04:37.24 | jameswf-home | LIES ron paul can cure cancer with chuck norris tears |
| 04:37.27 | drmessano | Asterisk doesn't make my bread toasty and crackly |
| 04:37.28 | jblack | Not even a little bit? |
| 04:37.44 | bkw_ | jblack: I am a bit abrasive at times |
| 04:37.51 | jblack | You Troll! |
| 04:37.57 | drmessano | jblack: He's a troll |
| 04:38.03 | bkw_ | what ever |
| 04:38.13 | drmessano | jblack: He's lives under a bridge |
| 04:38.16 | drmessano | Err |
| 04:38.17 | drmessano | He |
| 04:38.22 | bkw_ | worse.. I live in Oklahoma |
| 04:38.26 | jblack | I figured the green skin was just food poisoning |
| 04:38.36 | jblack | gnome sucks. |
| 04:38.43 | jblack | even travel gnomes. |
| 04:38.50 | jblack | hell. especially travel gnomes |
| 04:39.23 | drmessano | Go back to your bridge troll, you have no powers here! |
| 04:39.27 | bkw_ | russellb: I still have to support our Asterisk customers for now |
| 04:39.59 | russellb | ok. |
| 04:40.22 | bkw_ | too much has changed between 1.2 and the rest.. and I haven't followed those changes |
| 04:41.04 | drmessano | Well, you could spend your trolling time... reading |
| 04:41.11 | jameswf-home | you know what they say keep up or get thrown under the snow plow |
| 04:41.11 | russellb | can't help it that we have so much development going on. |
| 04:41.28 | drmessano | The snow gets deep under the troll bridge |
| 04:41.28 | jblack | drmessano: That's impossible. Trolls can only read 1/2 of a sentence at a time. |
| 04:41.50 | bkw_ | drmessano: I spent my time very well... |
| 04:42.07 | jameswf-home | ~trollbait |
| 04:42.14 | drmessano | Asterisk pancakes has shoe made armpit many frisbee advances grapefruit since china 1.2 |
| 04:42.42 | drmessano | Troll food |
| 04:42.58 | jblack | pancakes shoes armpit frisbee grapefruit china? |
| 04:43.04 | drmessano | Indeed. |
| 04:43.06 | bkw_ | drmessano: I dedicated many years to Asterisk |
| 04:43.10 | jameswf-home | thats an essay |
| 04:43.33 | jblack | bkw_: Congrats. You can now die a fullfilled man. |
| 04:43.48 | jameswf-home | jbot beer |
| 04:43.48 | jbot | ACTION has disconnected (Read error: 99 (Connection reset by beer)) |
| 04:43.49 | drmessano | So now you troll and bitch? |
| 04:43.58 | drmessano | Im confused |
| 04:44.01 | jblack | heh. jbot got drunk |
| 04:44.04 | bkw_ | drmessano: I still assist people with Asterisk |
| 04:44.13 | drmessano | Ohhhhh |
| 04:44.17 | jameswf-home | jbot: poo on the troll |
| 04:44.18 | jbot | ACTION summons a troop of flying monkeys to fling their poo at on the troll |
| 04:44.19 | drmessano | Intermixed with the endless bashing? |
| 04:44.35 | drmessano | and Freeswitch adsense |
| 04:44.40 | bkw_ | what ever |
| 04:44.56 | drmessano | Hey everyone, freeswitch I just had freeswitch a new baby boy! |
| 04:45.02 | bkw_ | haha |
| 04:45.13 | drmessano | You fail at google |
| 04:46.22 | drmessano | At least Asterisk has a kick ass win32 port |
| 04:46.35 | bkw_ | oh my |
| 04:46.45 | bkw_ | does asterisk compile native in MSVC? |
| 04:46.56 | bkw_ | MSVC is actually a damn good compiler |
| 04:46.59 | drmessano | Last time I tried to compile FS, neither did it |
| 04:47.04 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 04:47.08 | jblack | mingw I bet. |
| 04:47.16 | russellb | mingw, yes. |
| 04:47.31 | bkw_ | drmessano: it compiles in MSVC |
| 04:48.03 | drmessano | bkw_: Yeah, didn't work the 11 times I tried it under 3 diff versions... but het |
| 04:48.05 | drmessano | hey* |
| 04:48.10 | jblack | I dunno.... some people love .net... I think I'll hold out for .org |
| 04:48.20 | bkw_ | drmessano: must have been missing something related to the platform SDK |
| 04:48.21 | drmessano | .us is teh hot |
| 04:48.24 | *** join/#asterisk docelmo (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net) |
| 04:48.33 | jblack | Coming along any day now, thanks to our friends across the pond. |
| 04:48.37 | drmessano | bkw_: I dunno, Asterisk just works.. Seemed like too much work to me |
| 04:48.49 | bkw_ | drmessano: Could be |
| 04:48.58 | bkw_ | Asterisk does what you need then Asterisk is what you use then |
| 04:49.15 | jblack | At least until callweaver gets around to making a release. ;) |
| 04:49.23 | bkw_ | RC5 now? |
| 04:49.23 | drmessano | It does more than what I need actually, but I try not to troll other channels bashing the others like I do here ;) |
| 04:49.47 | bkw_ | drmessano: if stating facts are trolling then i'm guilty :P |
| 04:49.49 | jameswf-home | asterisk is only limited by the user |
| 04:49.54 | drmessano | CallWeaver, coming Febtober Epochity-twelve |
| 04:50.07 | jblack | At least rc5 |
| 04:50.08 | bkw_ | jameswf-home: not true |
| 04:50.19 | drmessano | brb |
| 04:50.38 | jameswf-home | I am alot of things but a lawyer i mean liar is not one of them |
| 04:50.55 | bkw_ | hehe |
| 04:51.07 | bkw_ | Asterisk does very well for what it was designed to do |
| 04:51.24 | bkw_ | but running an ITSP or CLEC wasn't really what it was designed for |
| 04:51.37 | russellb | yet there are thousands of people doing it |
| 04:51.39 | bkw_ | its really pushing what it was designed to do |
| 04:51.41 | jameswf-home | I had asterisk mking coffee and burning cd's I think that is well beyond design |
| 04:51.46 | russellb | guess it hasn't worked out _that_ bad, huh. |
| 04:51.53 | bkw_ | russellb: and its pushing it |
| 04:52.16 | bkw_ | you have to admit it wasn't designed to handle multiple DS3's worth of traffic |
| 04:52.28 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 04:52.34 | bkw_ | russellb: we just outgrew asterisk very rapidly |
| 04:52.40 | russellb | then i guess you don't need it, then |
| 04:52.49 | jameswf-home | yet your still here |
| 04:52.50 | ilowe | Hi guys, does anybody have a second for a quick question (and perhaps a longer answer)? |
| 04:52.52 | bkw_ | its still in limited use in our platform |
| 04:53.06 | jameswf-home | then your presence should be limited |
| 04:54.16 | denon | nothin wrong with bkw hangin around, just so long as he pays for the freeswitch ads placed in the channel :) |
| 04:54.30 | denon | or better yet, stick on topic to positive asterisk conversation, like the good ole days |
| 04:56.48 | jameswf-home | on+topic=buzz kill damn you denon |
| 04:57.31 | drmessano | lol |
| 04:57.51 | jblack | ohhh. I know who bkl is. He's one of those trixbox guys, right? |
| 04:58.03 | jameswf-home | fbosco |
| 04:58.56 | drmessano | I think it's funny when people come up with scenarios that this product or that product won't do.. because, you know, it does SUCH a bad job at it's core market that surely I can find scenario where it fails to scale up to |
| 04:59.04 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 04:59.28 | drmessano | I CAN RUN ASTERISK ON NETWARE.. YOU SUXORS |
| 04:59.28 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 04:59.29 | jblack | EOBVIOUS |
| 04:59.31 | ZaVoid | hey hey guys |
| 04:59.32 | drmessano | Errr |
| 04:59.33 | drmessano | Cant |
| 05:00.12 | ZaVoid | any ideas why exten => s,n,NoOp({$LANGUAGE}) doesn't work? |
| 05:00.20 | drmessano | Asterisk can't handle 75,000 concurrent calls on a PII 500.... Piece of crap |
| 05:00.20 | ZaVoid | it just nops the word $LANGUAGE |
| 05:00.21 | russellb | ZaVoid: invalid syntax. |
| 05:00.31 | ZaVoid | am i fat fingering russellb ? |
| 05:00.33 | russellb | it would be ${LANGUAGE} |
| 05:00.37 | ZaVoid | ah |
| 05:00.40 | ZaVoid | too much beer one second |
| 05:00.44 | russellb | or more likely ${CHANNEL(language)} |
| 05:00.50 | ZaVoid | thought i tried that |
| 05:00.51 | ZaVoid | one sec |
| 05:01.00 | ZaVoid | ahh |
| 05:01.04 | ZaVoid | i see what i did wrong i guess |
| 05:01.04 | ZaVoid | exten => s,n,NoOp(CHANNEL(language)) |
| 05:01.10 | ZaVoid | thats what i tried as well |
| 05:01.11 | ZaVoid | one sec |
| 05:01.56 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 05:01.58 | scooby2 | when is digium going to make a card so I can plug a ds3 into a quad quadcore xeon w/ 32gb of ram? |
| 05:02.16 | ZaVoid | so exten => s,n,NoOp${CHANNEL(language)} ? |
| 05:02.23 | russellb | ZaVoid: still wrong |
| 05:02.27 | russellb | missing a set of () |
| 05:02.27 | ZaVoid | yeah i thought so |
| 05:02.31 | ZaVoid | another () right? |
| 05:02.32 | jblack | scobby2: As soon as there's a bus that won't melt under the load? |
| 05:02.33 | ZaVoid | ok |
| 05:02.36 | drmessano | When is Apache gonna support IAX2 so I can run a webserver trunk |
| 05:02.36 | JT | scooby2: doesn't sound very redundant |
| 05:02.56 | scooby2 | JT thats when you get two of everything |
| 05:03.14 | JT | on a linux pc.... |
| 05:03.14 | ZaVoid | bingo that was it russellb |
| 05:03.31 | jameswf-home | asterisk 1.8 is built on web 4.0 |
| 05:03.53 | scooby2 | dot net |
| 05:04.01 | ZaVoid | so if i have the language set in my db how can i pull it via realtime russellb without doing a php/sql query? |
| 05:04.04 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 05:04.16 | ZaVoid | can't i force a realtime lookup somehow? |
| 05:04.26 | russellb | ZaVoid: let me ask you this question ... how do you look up something from a db without looking it up from the db? |
| 05:04.38 | ZaVoid | i phrased that wrong |
| 05:04.41 | drmessano | WIZARD! |
| 05:04.42 | russellb | :) |
| 05:04.43 | ZaVoid | i want to pull it from realtime |
| 05:04.48 | russellb | ok, there is a REALTIME() function |
| 05:04.53 | ZaVoid | vs calling a .php script and doing a sql command |
| 05:04.55 | Corydon76-dig | russellb: you could look it up from a webserver |
| 05:04.58 | ZaVoid | ok let me look at that |
| 05:05.02 | russellb | Corydon76-dig: nice :) |
| 05:05.04 | ZaVoid | i do most of it via sql right now |
| 05:05.07 | ZaVoid | buty you get what i'm saying? |
| 05:05.13 | russellb | yeah ... check REALTIME() |
| 05:05.15 | Corydon76-dig | russellb: res_config_curl |
| 05:05.17 | ZaVoid | gotcha |
| 05:07.10 | cowmix | weird problem: I have three SPA3102s.. Three phone numbers in hunt group.. Inbound works fine on two of my lines but on the 'lead' line i get "that number is not available".. |
| 05:07.29 | cowmix | [Mar 3 21:55:56] VERBOSE[31633] logger.c: -- Executing [600@from-sip-external:1] NoOp("SIP/10.10.10.130-b7701f58", "Received incoming SIP connection from unknown peer to 600") in new stack |
| 05:07.33 | scooby2 | now if only i could get timing and switchtype from level3 and global crossing. Both act like I am speaking klingon when I open tickets asking them. |
| 05:07.37 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 05:08.46 | drmessano | Bang rocks, make fire |
| 05:09.36 | *** part/#asterisk bkw_ (n=brian@adsl-64-149-54-142.dsl.tul2ok.sbcglobal.net) |
| 05:09.54 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 05:10.01 | jblack | 00:07 < scooby2> now if only i could get timing and switchtype from level3 and |
| 05:10.29 | jblack | uh, sorry. |
| 05:10.49 | scooby2 | guess i need framing, coding, and signalling as well |
| 05:10.57 | jameswf-home | what are there like 8 possible combinations just try em |
| 05:11.28 | scooby2 | i've tried them all but still have timing issues. after X period of time asterisk thinks all channels are in use while the provider thinks they are free |
| 05:11.41 | jameswf-home | scooby2: what ast ver |
| 05:11.49 | scooby2 | 1.2.26.2 |
| 05:11.58 | scooby2 | same with latest 1.4 |
| 05:12.38 | jameswf-home | sounds like a dead lock to me... |
| 05:12.45 | scooby2 | under 1.2.26.2 it just stops accepting incoming calls until a restart now when asterisk thinks all channels are full |
| 05:12.57 | scooby2 | under 1.4 when all channels are full it kernel panics |
| 05:13.13 | jblack | Hmm. 55% chance of a 1/2 point rate cut, 28% chance of a 1 point rate cut, 18% chance of a 3/4 rate cut. |
| 05:13.13 | jameswf-home | what does top say |
| 05:13.19 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 05:13.19 | jameswf-home | do you have swec |
| 05:13.47 | scooby2 | i thought so as well but according to the debug docs its not a deadlock or at least not the common deadlock |
| 05:14.28 | ZaVoid | and this syntax is wrong too |
| 05:14.29 | ZaVoid | exten => s,n,Set(CHANNEL(language)={$USERL}) |
| 05:14.43 | ZaVoid | exten => s,n,Set(CHANNEL(language)=$USERL) |
| 05:14.44 | ZaVoid | so is that |
| 05:14.46 | scooby2 | jameswf-home: swec? |
| 05:14.54 | jameswf-home | of your running a soft echo canceller on a t1 it can cause bad moojoo |
| 05:15.19 | jameswf-home | ${USER} |
| 05:15.38 | scooby2 | ahh, te212p is hardware i believe. sangoma 102 is software |
| 05:15.44 | ZaVoid | ah |
| 05:15.57 | ZaVoid | duh thanks man |
| 05:16.07 | scooby2 | no pthread_wait_for_restart_signal in gdb trace |
| 05:16.11 | *** join/#asterisk mazpe (n=mazpe@75.144.247.202) |
| 05:17.55 | drmessano | ISK BREST NAT TWO CRODE WHIAL DURNK |
| 05:18.03 | ZaVoid | nice perfect jameswf-home thanks |
| 05:18.40 | jameswf-home | mmmm breast crode |
| 05:19.13 | scooby2 | other thing that makes me think its something service provider related is all the playback() stuff starts 2-3 seconds in where as under 1.2.14 on the sangoma using the exact same code it plays the full messages |
| 05:19.45 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 05:19.52 | scooby2 | IE: "Thank you for calling..." is "<silence> for calling..." |
| 05:20.46 | jblack | Yay. First it was subprime.. then alt-a... then SIVs... CDOs...bond insurers.. VIEs... banks... now, the latest group to start running out of money is brokerage firms. |
| 05:21.07 | jblack | I think it's time to get to learning how to mend socks. That could a usefull skill. |
| 05:22.35 | drmessano | I'm gonna learn the ancient art of origami |
| 05:24.07 | ilowe | I get the "maximum retries exceeded" error when I don't dial out immediately to route a DID. Any ideas? |
| 05:24.33 | drmessano | Dial faster |
| 05:24.41 | ilowe | If I put just Dial(SIP/<my-DID>,20) it works perfectly |
| 05:24.50 | ilowe | But I want to collect some user input first |
| 05:24.53 | ilowe | And it chokes |
| 05:25.06 | ilowe | drmessano Thanks.... I think :) |
| 05:25.21 | jameswf-home | background +get digits |
| 05:25.33 | jameswf-home | ~book |
| 05:25.35 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 05:25.39 | jameswf-home | ch 5 |
| 05:28.03 | ilowe | jameswf-home: I've tried WaitExten and Read with and without Background, they both give the same result |
| 05:28.59 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
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| 05:33.25 | ZaVoid | i just bought the book this weekend :) |
| 05:33.33 | ZaVoid | figured i'd help support :) |
| 05:33.36 | jjg_ | anyone know of a decent sip applet or somthing else browser based? ... flash maybe? |
| 05:33.39 | drmessano | The book rocks |
| 05:33.51 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 05:33.58 | jameswf-home | ~buybook |
| 05:33.59 | jbot | You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 05:34.02 | ZaVoid | it does |
| 05:34.30 | ZaVoid | so i'm a bit confused here...(shocking i know right) |
| 05:34.34 | ZaVoid | looking at the realtime http://www.voip-info.org/wiki/index.php?page=Asterisk+func+realtime |
| 05:34.49 | ZaVoid | heres the sample right? |
| 05:35.12 | ZaVoid | <PROTECTED> |
| 05:35.12 | ZaVoid | <PROTECTED> |
| 05:35.22 | ZaVoid | but that grabs the whole row.. and the cuts crap out of it |
| 05:35.55 | ZaVoid | i just want to pull one variable from sipusers per say |
| 05:39.06 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 05:39.34 | ZaVoid | or do i have to get the whole row basically? |
| 05:41.26 | jameswf-home | Im not a realtime person but based on other languages that syntax looks like poo poo |
| 05:42.01 | ZaVoid | yeah thats why i do it via .php and sql normally |
| 05:42.17 | ZaVoid | was hoping not to just to pull one stinking variable this time but oh wells i guess |
| 05:48.02 | cowmix | more on my issue.. (if anyone cares..) if a call comes in with a 'fake' number.. (ie SkypeOut).. things work.. |
| 05:48.11 | cowmix | If its a real number.. like my cell no.. |
| 05:48.29 | cowmix | i get "Received incoming SIP connection from unknown peer to 600" |
| 05:48.41 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 05:52.07 | cowmix | ok... a little more.. my 2nd and 3rd lines don't do caller id at all |
| 05:52.19 | cowmix | so.. they work all the time |
| 05:52.24 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 05:52.29 | cowmix | if there is caller id.. i get that error |
| 05:57.12 | drmessano | ajohnson |
| 05:57.14 | drmessano | err |
| 05:57.19 | drmessano | ajohnstone |
| 06:00.00 | *** part/#asterisk supjigator (n=shanebur@152.53.16.10) |
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| 06:17.11 | [T]an1 | i am trying to understand how this transcoding stuff works...... i have a phone that dials to the asterisk server using the g729 codec which then connects to another asterisk server using the ulaw codec which then connects to the pri. when i do a show g729 i show 1 encoder and 3 decoders |
| 06:17.11 | [T]an1 | The exact opposite call path only yields 1 encoder and 1 decoder. can anyone explain how that works? |
| 06:20.52 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
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| 06:22.05 | AJayMN | So I purchased 4 g729 licenses. when I make a call from a desk phone out to a cell #. should i only be using 1 license if my provider is g729 aswell? |
| 06:22.08 | *** join/#asterisk jamesrdorn (n=jamesrdo@adsl-99-135-235-94.dsl.rcsntx.sbcglobal.net) |
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| 06:23.42 | [T]an1 | AJayMN: shouldnt need a codec at all unless you are transcoding to another codec. |
| 06:23.45 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 06:31.48 | b11d | ~book |
| 06:31.49 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 06:31.50 | b11d | ~thebook |
| 06:31.50 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 06:33.38 | b11d | grrr |
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| 07:28.58 | drmessano | yawn |
| 07:31.40 | b11d | i know |
| 07:31.43 | b11d | goodnighjt |
| 07:31.51 | drmessano | lol |
| 07:32.03 | b11d | im sleepy |
| 07:32.09 | b11d | dont want to even drive home |
| 07:32.12 | b11d | but.. oh well.. |
| 07:33.43 | obnauticus | this will be interesting, but is anyone here good with CiscoCM? |
| 07:33.51 | obnauticus | it seems..... interesting. |
| 07:34.28 | drmessano | CiscoCM? |
| 07:34.33 | obnauticus | Cisco Call Maanger |
| 07:34.34 | drmessano | Oh man |
| 07:34.35 | obnauticus | Manager* |
| 07:34.36 | drmessano | GTFO |
| 07:34.38 | obnauticus | lol |
| 07:34.39 | obnauticus | i hate it. |
| 07:34.39 | drmessano | G T F O |
| 07:34.41 | obnauticus | what? |
| 07:34.49 | drmessano | Get the F out |
| 07:34.53 | drmessano | GTFO |
| 07:34.55 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 07:34.59 | obnauticus | GTO |
| 07:35.01 | obnauticus | wait |
| 07:35.01 | drmessano | Door <------- |
| 07:35.02 | obnauticus | GTFO |
| 07:35.06 | obnauticus | -----> Door |
| 07:35.25 | drmessano | ^^^^^^6^^^^^ #Asterisk |
| 07:35.25 | obnauticus | man |
| 07:35.27 | obnauticus | CiscoCM is gay. |
| 07:35.51 | drmessano | ^^^^^^6^^^^ is the new !!!!11!!!!1!!! |
| 07:36.55 | drmessano | Ewww.. Just had an abandoned house burn down around the corner, apparently with some homeless dude in it |
| 07:37.06 | obnauticus | PWNED |
| 07:38.16 | drmessano | I shudda ran down there with my vid camera |
| 07:38.20 | obnauticus | ya |
| 07:38.22 | obnauticus | put it on liveleak |
| 07:38.57 | drmessano | You wanna know who is responsible for the era of posting videos of messed up crap online? |
| 07:39.02 | drmessano | Bob Saget |
| 07:39.06 | obnauticus | no shit. |
| 07:39.33 | drmessano | He invented "Videotape your brother getting his teeth knocked out and win $10000" |
| 07:39.40 | drmessano | We owe it all to him |
| 07:39.52 | obnauticus | cisco invented this utter crap. |
| 07:39.53 | obnauticus | i hate it |
| 07:40.02 | obnauticus | im just `trying it out' |
| 07:40.06 | drmessano | "Ciscos getting smashed" |
| 07:40.13 | obnauticus | i never paid for it |
| 07:40.16 | obnauticus | i got cisco hardware for free |
| 07:40.16 | obnauticus | :\ |
| 07:40.19 | drmessano | lol |
| 07:40.27 | obnauticus | well it's really just an HP ProLiant with a Cisco Systems logo on it |
| 07:40.39 | obnauticus | and they probably painted all the leads on the ethernet interface |
| 07:40.39 | obnauticus | lol. |
| 07:40.46 | drmessano | lol |
| 07:40.49 | obnauticus | you get that joke? |
| 07:41.09 | drmessano | Oh.. I dont think so |
| 07:41.13 | obnauticus | ... |
| 07:41.37 | obnauticus | im getting a picture for you |
| 07:42.14 | obnauticus | the interfaces on cisco hardwar |
| 07:42.17 | obnauticus | ethe leads are like painted red |
| 07:42.22 | obnauticus | http://files.quadrantcommunications.be/Quadrant.nsf/804ab887fef03a13c12566bb0030464c/1fd2741c10febd8ec1256c750072db3b/$FILE/Cisco%2011503,%2011506%20Content%20Service%20Switch%20-%201500x1200.jpg |
| 07:44.21 | BeeBuu | why my phone which connected to FXS port doesn't get dial tone? |
| 07:44.34 | drmessano | Darwin |
| 07:44.58 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.98.208) |
| 07:45.28 | BeeBuu | drmessano: would you help me? |
| 07:45.34 | Asterisk-nob | what's difference between callerid and sourceID? |
| 07:46.55 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 07:47.23 | *** join/#asterisk arooni (n=arooni__@c-24-19-232-203.hsd1.mn.comcast.net) |
| 07:47.25 | arooni | hey folks! |
| 07:47.29 | arooni | i keep getting: [Mar 4 07:46:45] NOTICE[2300]: pbx_spool.c:341 attempt_thread: Call failed to go through, reason 1 |
| 07:47.32 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:47.32 | arooni | why is that |
| 07:47.45 | obnauticus | BeeBuu is your phone providing ring voltage? |
| 07:47.50 | drmessano | BeeBuu: What kind of FXS port? |
| 07:47.57 | obnauticus | i don't care what your answer is, you need an FXO |
| 07:48.36 | drmessano | or skype |
| 07:48.45 | obnauticus | >;\ |
| 07:49.11 | obnauticus | did the skype-asterisk projects improve much? |
| 07:49.28 | arooni | i have files in /var/spool/asterisk/outgoing |
| 07:49.28 | drmessano | Like polishing a turd |
| 07:49.34 | arooni | that asterisk isnt calling |
| 07:49.35 | arooni | ideas? |
| 07:49.38 | obnauticus | drmessano i do not understand? |
| 07:49.46 | BeeBuu | drmessano: fixed.thanks.drmessano,how to reload zapata.conf? |
| 07:49.56 | obnauticus | how is it like polishing a turd drmessano? |
| 07:50.27 | obnauticus | BeeBuu thankyou. |
| 07:50.29 | drmessano | Getting skype to work on Asterisk will always be glorified turn polishing |
| 07:50.47 | obnauticus | in my experience asterisk is nothing like polishing turds. |
| 07:51.17 | arooni | [Mar 4 07:50:48] NOTICE[2358]: pbx_spool.c:341 attempt_thread: Call failed to go through, reason 1 ...... what is reason 1? |
| 07:51.27 | drmessano | The skype part of it is.. |
| 07:51.33 | cowmix | outbound issue... asterisk -> SPA3102.. log: http://pastebin.ca/927394 |
| 07:51.40 | obnauticus | drmessano how is skype like a turd? |
| 07:51.45 | obnauticus | i do not understand |
| 07:52.05 | drmessano | lol |
| 07:52.17 | obnauticus | Fail0r |
| 07:52.24 | drmessano | Too tired for a good line |
| 07:52.32 | cowmix | obnauticus: Skype is just pain funky in how it works.. they make it hard to interface with the VOICE part.. and their Linux port is very behind |
| 07:52.47 | obnauticus | cowmix i didn't want a smartass answer, smartass. |
| 07:52.48 | obnauticus | lol. |
| 07:52.53 | obnauticus | i know this already, im just being a smartass. |
| 07:53.02 | cowmix | hehe |
| 07:53.26 | arooni | what is reason1? |
| 07:53.27 | drmessano | obnauticus: You see, skype is like a gladiator movie.. A lot of people will sit thru it, but only some admit it |
| 07:53.29 | arooni | where can i find this |
| 07:53.56 | obnauticus | drmessano does this relate to gay porn? |
| 07:54.01 | obnauticus | because if it does im out. |
| 07:55.05 | drmessano | As if I would admit it.. See previous line |
| 07:55.13 | cowmix | anyone wanna take pitty on me? if I don't solve this outbound issue.. I'll go insane.. |
| 07:55.25 | obnauticus | drmessano helped the last dde |
| 07:55.26 | obnauticus | dude* |
| 07:55.28 | obnauticus | so ask him |
| 07:55.47 | drmessano | maybe I should try less and see if it helps more |
| 07:55.53 | obnauticus | lies. |
| 07:56.18 | drmessano | cheeseburger sneaker doorknob anvil |
| 07:57.15 | arooni | how can i find out what reason 1 means? |
| 07:57.17 | arooni | anyone? |
| 07:57.37 | drmessano | cowmix: You probably used one of the 120 crappy guides to setting up SPA-3102s out there |
| 07:57.40 | obnauticus | arooni did you figure out what reason 0 is? |
| 07:57.44 | arooni | no |
| 07:57.49 | obnauticus | find that out first |
| 07:57.49 | arooni | i dont see where any of this is documented either |
| 07:57.50 | obnauticus | then ask us |
| 07:57.59 | cowmix | drmessano: yes.. I did |
| 07:58.02 | obnauticus | Reason 0 is. |
| 07:58.08 | cowmix | I used them all.. :/ |
| 07:58.12 | arooni | obnauticus, what is reason 0 |
| 07:58.27 | obnauticus | you got it backwards. |
| 07:58.34 | cowmix | drmessano: not only that.. I'm using PBX in a Box.. I hang my head in shame. |
| 07:58.37 | *** join/#asterisk Daejeo (n=chatzill@211.177.189.62) |
| 07:59.07 | drmessano | Trixbox? |
| 07:59.14 | cowmix | oops.. /Box/Flash |
| 07:59.14 | obnauticus | EW |
| 07:59.17 | drmessano | Oh |
| 07:59.24 | arooni | obnauticus, where can i go to find out what the reasons mean |
| 07:59.29 | Daejeo | is there any one from states who can ping sip.vtwhite.com I am trying from my place but no response |
| 07:59.32 | obnauticus | what 0 reason means? |
| 07:59.35 | arooni | yes |
| 07:59.39 | arooni | google is not producing results |
| 07:59.44 | obnauticus | for reason 0? |
| 07:59.47 | arooni | yes |
| 07:59.48 | obnauticus | 0 reason* |
| 07:59.49 | yang | I am experiencing this NAT trouble, I cannot connect a client from outside the network - http://openpaste.org/en/5386/ |
| 07:59.51 | obnauticus | maybe you have no reason |
| 07:59.52 | arooni | not the ones i want |
| 07:59.57 | obnauticus | maybe you have `0 reason' |
| 07:59.58 | obnauticus | ooo!!! |
| 08:00.10 | Daejeo | s there any one from states who can ping sip.vtwhite.com I am trying from my place but no response |
| 08:00.14 | arooni | obnauticus, haha |
| 08:00.28 | obnauticus | hahah |
| 08:00.34 | obnauticus | hahahaa!! |
| 08:00.34 | drmessano | Daejeo: Bad paste, use up arrow next time |
| 08:00.37 | obnauticus | ohohoohoh |
| 08:00.39 | obnauticus | ahahaha |
| 08:00.40 | cowmix | Daejeo: http://pastebin.ca/927405 |
| 08:00.41 | drmessano | It makes for better repeating |
| 08:01.01 | obnauticus | Daejeo i'll try it |
| 08:01.04 | Daejeo | is there any one from states who can ping sip.vtwhite.com I am trying from my place but no response |
| 08:01.13 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 08:01.14 | drmessano | STOP REPEATING |
| 08:01.16 | obnauticus | here you go sexy |
| 08:01.17 | obnauticus | Ping statistics for 216.120.251.172: |
| 08:01.17 | obnauticus | <PROTECTED> |
| 08:01.21 | obnauticus | love me sexy |
| 08:01.29 | obnauticus | anyone here see semi-pro yet? |
| 08:01.33 | obnauticus | that movie sucked ass |
| 08:01.35 | drmessano | nope |
| 08:01.58 | cowmix | drmessano: I don't think it specific to the SPA3102.. i think its all Asterisk side |
| 08:02.30 | arooni | obnauticus, is reason 1 hangup? |
| 08:02.33 | drmessano | That link has Asterisk peer setup as well |
| 08:02.38 | obnauticus | arooni you are getting closer. |
| 08:02.51 | arooni | obnauticus, where is this documented???? |
| 08:02.53 | obnauticus | arooni if reason 3 is hangup |
| 08:02.56 | obnauticus | and reason 0 is pickup |
| 08:02.59 | obnauticus | what is reason 1? |
| 08:03.05 | obnauticus | sorry reason 2 is hangup |
| 08:03.14 | drmessano | reason 1 is "I have a headache" |
| 08:03.23 | obnauticus | keep guessing, you'll eventually find out |
| 08:03.24 | arooni | why couldnt they explain what reason 1 |
| 08:03.27 | arooni | is |
| 08:03.37 | arooni | 1 is never answered? |
| 08:03.38 | obnauticus | arooni they explained it just fine |
| 08:03.45 | drmessano | Is it bigger than a breadbox? |
| 08:03.48 | obnauticus | reason 1 is not equal to reason 2 and also not reason 0 |
| 08:03.52 | *** join/#asterisk steliosk (n=Stelios@85.75.211.185) |
| 08:03.52 | drmessano | sounds like...... |
| 08:03.53 | obnauticus | yes! |
| 08:04.00 | obnauticus | does it weigh more than a duck? |
| 08:04.00 | arooni | but i didnt hear my phone ring |
| 08:04.15 | obnauticus | can you fit it in your pocket? |
| 08:04.25 | drmessano | What is the square root of cry? |
| 08:04.27 | arooni | seriously though |
| 08:04.30 | arooni | where is this documented |
| 08:04.43 | obnauticus | i'll link you hold on |
| 08:05.03 | arooni | haha |
| 08:05.06 | obnauticus | shh! |
| 08:07.54 | arooni | a ha!!!!!!!!!!!!!!! |
| 08:07.55 | arooni | i fixed it |
| 08:07.57 | arooni | cuz i'm awesome |
| 08:08.01 | obnauticus | ... |
| 08:08.04 | arooni | i turned off my phone and then turned it back on again |
| 08:08.05 | arooni | problem fix |
| 08:08.11 | obnauticus | in8 |
| 08:08.13 | arooni | big FAIL for windows mobile 6 |
| 08:08.18 | drmessano | R O F L |
| 08:08.23 | obnauticus | even bigger fail for you... you use i. |
| 08:08.25 | obnauticus | it* |
| 08:08.30 | arooni | haha |
| 08:08.37 | arooni | obnauticus, you should read http://uncov.com |
| 08:08.39 | arooni | you'd like it |
| 08:08.45 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 08:08.51 | arooni | or write for them |
| 08:08.52 | obnauticus | i have something for you too |
| 08:08.54 | arooni | they like snarky guys |
| 08:09.00 | arooni | i mean that as a compliment |
| 08:09.04 | arooni | i laughed at your fail statement |
| 08:09.06 | obnauticus | stfu i don't care. |
| 08:09.49 | obnauticus | http://img151.imageshack.us/img151/5310/enjoyyourhatac1.jpg + Windows Mobile 6 = arooni. |
| 08:10.26 | arooni | haha |
| 08:10.34 | arooni | he kinda looks like me |
| 08:10.44 | arooni | except i buy my hats at garage sales |
| 08:10.44 | drmessano | Is that an asshat? |
| 08:11.48 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:11.51 | obnauticus | it's meh asshat |
| 08:11.57 | obnauticus | http://img187.imageshack.us/img187/365/1202613164922yx9.jpg == drmessano. |
| 08:12.16 | drmessano | arooni, I already hate the site you linked |
| 08:12.30 | arooni | hahah |
| 08:12.33 | drmessano | The pic of the naked fat dude in front of the computer.. Effin 1998 man |
| 08:12.36 | obnauticus | arooni i found a real picture of you |
| 08:12.36 | arooni | you should sell a fail detector |
| 08:12.40 | obnauticus | http://img187.imageshack.us/img187/2913/eurofagkr0.jpg |
| 08:12.44 | drmessano | ROFL |
| 08:12.50 | arooni | haha |
| 08:13.00 | arooni | wow he kinda looks like me too |
| 08:13.01 | drmessano | JA? |
| 08:13.07 | obnauticus | http://img249.imageshack.us/img249/7683/12044456991112a03870ho6wz0.jpg |
| 08:18.49 | tzafrir | What a useful support channel #asterisk is |
| 08:19.50 | tzafrir | Nice to know omeone can come here and actually get answers |
| 08:20.50 | obnauticus | <3 |
| 08:23.26 | joobie | tzafrir... u got lucky |
| 08:25.44 | drmessano | lucky? |
| 08:26.06 | obnauticus | yes. |
| 08:27.26 | *** join/#asterisk z3wb (n=zewb@c-76-31-96-238.hsd1.tx.comcast.net) |
| 08:27.29 | z3wb | hello |
| 08:28.46 | z3wb | im using 1.4.10 and im having some sound issues |
| 08:28.59 | z3wb | when i use chan_oss, i get garbled, distorted sound |
| 08:29.06 | z3wb | and when i use chan_alsa i get no sound at all |
| 08:29.17 | z3wb | but my alsamixer is all set up fine |
| 08:29.22 | z3wb | everything else that uses alsa works perfectly |
| 08:29.58 | z3wb | and there is nothing else using alsa when asterisk is running |
| 08:30.30 | *** join/#asterisk atop (n=user@oaktyres.force9.co.uk) |
| 08:31.49 | atop | I read advice that said for a partial E1 line, (we have 22 lines provisioned out of a possible 30) you should set zaptel.conf to the full range, and set zapata.conf to just the range you need. Is this right and if so, why? |
| 08:33.25 | *** join/#asterisk docelmo (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net) |
| 08:33.32 | *** join/#asterisk shinao1 (n=shinao1@35-230.rv.ipnxtelecoms.com) |
| 08:35.34 | z3wb | does anyone else know why chan_alsa would make it not produce any sound? |
| 08:35.53 | z3wb | it can't be the alsamixer settings, since all my other stuff that uses alsa works just fine |
| 08:36.32 | Daviey | z3wb: Does this box have X? |
| 08:36.36 | z3wb | yep |
| 08:36.40 | Daviey | ewwwww |
| 08:36.48 | z3wb | xubuntu |
| 08:37.18 | Daviey | Have you tried stopping that, and seeing if you get better performance? |
| 08:37.19 | drmessano | bless you |
| 08:37.46 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:37.51 | yang | I am experiencing this NAT trouble, I cannot connect a client from outside the network - http://openpaste.org/en/5386/ |
| 08:37.55 | dandre | Hello, |
| 08:38.18 | Daviey | ~hi |
| 08:38.18 | jbot | hello, daviey |
| 08:38.36 | dandre | Is there a way to globally set asterisk so that all messages are said in french? |
| 08:38.42 | Daviey | dandre: yes |
| 08:38.47 | drmessano | Yang, do you have 5060 and 10000-10500 open, UDP? |
| 08:38.54 | Daviey | (i guess you have french sound files? |
| 08:39.02 | dandre | yes I do |
| 08:39.03 | z3wb | i don't think X is causing the problem |
| 08:39.17 | yang | drmessano: right, I do |
| 08:39.38 | Daviey | z3wb: well you don't know whats causing the problem, so why not try it? |
| 08:39.47 | z3wb | im trying it right now |
| 08:40.02 | drmessano | yang: iptables maybe? |
| 08:40.08 | drmessano | on the box itself |
| 08:40.12 | yang | none |
| 08:40.15 | *** join/#asterisk Hyphenex (n=Hyphenex@60-241-72-151.tpgi.com.au) |
| 08:40.22 | Hyphenex | how do I fix this? [Mar 4 19:29:09] WARNING[3086]: chan_sip.c:3008 sip_call: No audio format found to offer. Cancelling call to 151\ |
| 08:40.23 | drmessano | I dunno... your config looks good |
| 08:40.24 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 08:40.36 | yang | drmessano: have you seen my sip.conf i think its set up correct |
| 08:41.00 | drmessano | The nat settings and all look good |
| 08:41.02 | yang | drmessano: yeah must be some firewall issue |
| 08:41.12 | drmessano | No typo on the externip? |
| 08:41.17 | z3wb | ok x is dead |
| 08:41.21 | z3wb | lets try again |
| 08:41.28 | yang | drmessano: nope |
| 08:41.32 | atop | I'm being told that people trying to call us are getting a busy tone when we do have enough available lines. This is a PRI with 22 channels, can asterisk 'lock' channels incorrectly sometimes? Is there a way to debug it? These call attempts dont show in the CDR so it looks like it's not reaching us at all |
| 08:41.35 | drmessano | I have nothing else then :) |
| 08:41.37 | drmessano | heh |
| 08:41.47 | z3wb | ok still no sound |
| 08:41.48 | drmessano | Gotta be on the other end perhaps |
| 08:41.51 | z3wb | im using chan_alsa |
| 08:41.56 | yang | drmessano: external ip is 86.61.78.105 while asterisk is on 192.168.1.5 |
| 08:41.58 | dandre | I have put language=fr in all my channel files (iax, sip and zap) so that it is ok but when I use the manager to issue a call the messages are in english Daviey |
| 08:42.00 | z3wb | and im doing console dial 1000 to test the sound |
| 08:42.18 | drmessano | yang: it looks all good to me.. i'd check the remote client |
| 08:43.41 | yang | drmessano: my client has sip=no in sip context, becouse its nto nat-ed |
| 08:43.52 | yang | in asterisk sip context |
| 08:44.04 | yang | its not nated |
| 08:44.40 | drmessano | But its outside? |
| 08:44.54 | z3wb | and chan_oss still produces garbled sound |
| 08:45.00 | z3wb | so it's not X doing it |
| 08:45.11 | z3wb | hmm |
| 08:45.40 | Daviey | dandre: can you pastebin "find /var/lib/asterisk/sounds" ? |
| 08:45.41 | dandre | so my previous question : Is there a way to globally set asterisk so that all messages are said in french? |
| 08:45.41 | Daviey | ~pb |
| 08:45.42 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 08:46.14 | dandre | this is quite huge |
| 08:46.34 | Daviey | sure, can you check there is a "fr" folder? |
| 08:47.03 | z3wb | yeah i was talking to a guy who told me about different folders with recordings for different languages |
| 08:47.35 | yang | drmessano: its across the internet different location |
| 08:47.47 | z3wb | usr/share/asterisk/sounds/fr |
| 08:47.52 | dandre | sorry, my sounds are in /usr/share/asterisk/sounds/ |
| 08:47.59 | yang | drmessano: do you think a sip debug would show you anything usefull ? |
| 08:48.09 | dandre | ls -ld /usr/share/asterisk/sounds/fr |
| 08:48.09 | dandre | drwxr-xr-x 6 root root 6144 Feb 13 11:21 /usr/share/asterisk/sounds/fr |
| 08:48.20 | tzafrir | If you set the language, the language directory will first be looked for sound files |
| 08:49.35 | tzafrir | So funny things happen if you only have half of the sound files in a language, and the original Allison's in the main sound dir |
| 08:49.35 | dandre | yes all is ok when dialed from an extension but not when I issue an originatecall command from the manager interface |
| 08:50.07 | *** join/#asterisk CaRb0n^ (n=playa@203.81.233.62) |
| 08:50.12 | z3wb | where exactly is the chan_alsa.so module? |
| 08:50.29 | CaRb0n^ | when i type sip show registry on CLI , it returns nothing |
| 08:50.38 | CaRb0n^ | it was showing till last month |
| 08:50.47 | CaRb0n^ | any idea any one? |
| 08:51.29 | dandre | see http://pastebin.com/d27dae07b |
| 08:51.39 | simbol76ss | /var/lib/asterisk/sounds |
| 08:52.01 | z3wb | no such directory |
| 08:52.15 | simbol76ss | mmmhh |
| 08:53.28 | z3wb | ok i found it |
| 08:53.33 | z3wb | /usr/lib/asterisk/modules |
| 08:53.59 | Hyphenex | Ahhhhhh, Gay. [Mar 4 19:53:24] WARNING[3113]: chan_sip.c:3008 sip_call: No audio format found to offer. Cancelling call to 121 -- Couldn't call 121@MyNetFone |
| 08:54.06 | Hyphenex | I don't get it |
| 08:54.43 | z3wb | the *.so modules just contain a bunch of garbled encrypted stuff so that doesn't help.. |
| 08:54.57 | tzafrir | Hyphenex, no matching codec |
| 08:55.11 | *** part/#asterisk simbol76ss (n=simbol@ip-212-26.sn1.eutelia.it) |
| 08:55.32 | tzafrir | Either that or they don't like your views about homosexuality |
| 08:55.38 | Hyphenex | tzafrir: That's what I thought, but I've enabled allow=729 and the web site says they support G729 40ms packet size |
| 08:55.43 | JT | z3wb: binaries, not sure if that's really encrypted |
| 08:55.49 | z3wb | yeah |
| 08:55.52 | *** join/#asterisk simbol76ss (n=simbol@ip-212-26.sn1.eutelia.it) |
| 08:55.56 | J4k3 | allow=g729 |
| 08:55.59 | z3wb | i can read some stuff in it though |
| 08:56.06 | Hyphenex | ahh, so I need the g? Awesome :) |
| 08:56.25 | JT | mynetfone is poop btw |
| 08:56.34 | CaRb0n^ | when i type sip show registry on CLI , it returns nothing |
| 08:56.46 | Hyphenex | JT: it is? |
| 08:57.03 | JT | last i checked, they can only do a single call per sip account |
| 08:57.06 | JT | which is pathetic |
| 08:57.07 | z3wb | i prefer using alsa over oss, so i'm going to just concentrate on getting chan_alsa to work |
| 08:57.18 | z3wb | im not sure why i'm not getting any sound |
| 08:57.23 | Hyphenex | JT: yep, all I require though |
| 08:57.39 | z3wb | i know it has to be something in asterisk itself, since alsa is working fine otherwise |
| 08:57.48 | *** join/#asterisk ArashHemmat (n=ArashHem@91.184.88.227) |
| 08:57.54 | *** join/#asterisk CrashSys (n=kumba@62-209.187-72.tampabay.res.rr.com) |
| 08:58.04 | JT | Hyphenex: useless for business though |
| 08:58.14 | CrashSys | Sox 14.0.1 doesn't have a soxmix app, do I just symlink soxmix to sox and call it a day? |
| 08:58.29 | Hyphenex | JT: yeah, I'm happy with it for my house though :) |
| 08:58.37 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
| 08:58.43 | Hyphenex | ... if it'd work, I'm still getting the same message with g729 |
| 08:58.48 | Hyphenex | does the g need to be a G? |
| 08:58.54 | JT | Hyphenex: what phone? |
| 08:59.28 | Hyphenex | JT: It's an Avaya 4610SW IP. (that does not matter if Asterisk supports g729 though, right? :S) |
| 08:59.50 | JT | Hyphenex: as long as there is no reason for asterisk to transcode the call |
| 09:00.02 | JT | how hard was the avaya to set up? |
| 09:00.20 | J4k3 | well |
| 09:00.21 | Hyphenex | JT: umm, it's annoying, not hard, but it has a lot of limitations |
| 09:00.24 | JT | hmm |
| 09:00.34 | J4k3 | you'll probably need a g729 license, unless the call is cut straight through |
| 09:00.39 | JT | i picked up an Avaya 4621SW IP for pretty cheap |
| 09:00.44 | JT | and have never used it |
| 09:00.48 | J4k3 | and I'm not sure if asterisk 1.2 supports unusual timeslot lengths on g729 |
| 09:00.55 | J4k3 | g729 is normally 20ms |
| 09:01.17 | Hyphenex | oahh crap, I need licences... I thought this was going to be easy to set up :S |
| 09:01.55 | JT | Hyphenex: you don't if the itsp and phone speak identical G.729 packet sizes |
| 09:01.58 | JT | and dont record calls |
| 09:02.07 | JT | and dont ever put calls through to IVRs, tones, MoH |
| 09:02.08 | JT | etc |
| 09:02.26 | Hyphenex | JT: but if they are not the same packet sizes then? I'm in trouble? |
| 09:02.42 | JT | probably, if the itsp doesn't support it |
| 09:02.43 | JT | try 20ms |
| 09:04.40 | Hyphenex | JT: I've tried allow=g729 and that does not work |
| 09:06.02 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 09:06.48 | z3wb | what format are the recordings used by asterisk? |
| 09:07.00 | z3wb | wav? |
| 09:07.15 | CrashSys | whatever format you specify in the record command |
| 09:07.33 | z3wb | ok they all end in gsm |
| 09:07.44 | z3wb | so my guess is that alsa doesn't know how to play gsm |
| 09:08.12 | CrashSys | try aplay file.gsm |
| 09:08.15 | CrashSys | see what happens |
| 09:08.41 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 09:08.44 | joelsolanki | Hi room |
| 09:08.53 | joelsolanki | i can passthru caller id in asterisk |
| 09:09.31 | joelsolanki | but CNAME means caller name is not passing thru my voip provider. |
| 09:09.31 | joelsolanki | is there anything needed to be done in sip.conf ? |
| 09:11.31 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 09:12.31 | z3wb | aplay demo-congrats.gsm, or any gsm for that matter, produces pure white noise |
| 09:12.34 | z3wb | i may be on to something here |
| 09:12.46 | z3wb | also my ears may be bleeding |
| 09:12.46 | z3wb | :D |
| 09:13.00 | nixguy | if i want statistics for calls etc, what good app can i use to collect and show this data? |
| 09:13.08 | nixguy | some nifty webapp people can recomend? |
| 09:14.00 | z3wb | so obviously, alsa can't play gsm properly |
| 09:14.02 | *** join/#asterisk MmixX (n=mmixx@202.124.138.69) |
| 09:14.13 | z3wb | but i didn't get any static noise in asterisk when it tried to play demo-congrats |
| 09:14.26 | z3wb | i didn't get any sound whatsoever |
| 09:14.32 | *** join/#asterisk ddunavant (n=David@pool-71-178-102-155.washdc.east.verizon.net) |
| 09:21.24 | dandre | Daviey, tzafrir, back with my language problem: if I put Set(CHANNEL(language)=fr) before the background(...), it works fine. How should I default to language=fr for all originated calls? |
| 09:26.11 | JT | z3wb: use sox |
| 09:26.30 | JT | z3wb: and if you want to actually use .wavs, download the .wav asterisk sounds package |
| 09:26.33 | tzafrir | you can set channel variables in the Originate command |
| 09:26.49 | Hyphenex | hmm, does this mean g729 did not work? http://paste2.org/p/15014 |
| 09:28.23 | dandre | I have tried language=fr in the originate commande whith no success |
| 09:29.27 | tzafrir | Hyphenex, you have no g729 codec. |
| 09:29.37 | tzafrir | You may use g729 only for pass-through |
| 09:30.09 | tzafrir | danalien, LANGUAGE=fr |
| 09:30.17 | Hyphenex | tzafrir: I don't get it :S |
| 09:30.36 | tzafrir | how exactly did you set it? What line in the originate command? |
| 09:31.17 | tzafrir | Hyphenex, if you happen to have a SIP phone that supports g729 that phone can call a g729 provider through you |
| 09:31.32 | tzafrir | you can send that call, but can't do anything with the content |
| 09:31.40 | Hyphenex | tzafrir: Sorry, don't have |
| 09:31.45 | tzafrir | e.g: can't even send it to voicemail |
| 09:32.05 | tzafrir | So you can't use g729 right now. |
| 09:32.14 | tzafrir | Use gsm , speex, or whatever |
| 09:32.54 | tzafrir | http://speex.org/comparison/ |
| 09:33.53 | CaRb0n^ | any one why sip registry not showing any results ? |
| 09:34.13 | Hyphenex | tzafrir: It does not support g729, I'm going to need to re-encode it (and pay for a right to do so I think) |
| 09:34.56 | Hyphenex | it says it supports G.729AB (whatever that is) |
| 09:35.03 | Hyphenex | http://www.epinions.com/pr-Avaya_4610sw_Ip_Handset_En_700326051_Modem/display_~full_specs |
| 09:35.05 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:36.05 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:39.17 | *** join/#asterisk shasta (i=shasta@bluzg.slackware.pl) |
| 09:41.33 | lnx | hi all |
| 09:41.51 | Hyphenex | I am soooo lost it's not funny :( |
| 09:41.58 | alexcf | http://maps.google.com |
| 09:43.27 | joobie | guys what's the average u/l d/l i need for a sip call? |
| 09:43.33 | joobie | with compression and without |
| 09:44.21 | tzafrir | it's not "average". voip codecs provide a steady rate |
| 09:44.43 | joobie | what is that rate? |
| 09:44.53 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 09:45.17 | tzafrir | http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 09:45.35 | tzafrir | Search I used: asterisk bandwidth calculator |
| 09:46.02 | joobie | thanks |
| 09:46.25 | tzafrir | ~g729 |
| 09:46.26 | jbot | somebody said g729 was an ITU-standard voice codec which operates at 8kbps and offers quality very similar to GSM. G.729 is patent-encumbered; those wishing to use it with Asterisk must buy a license from Digium. |
| 09:46.44 | tzafrir | Hyphenex, have you asked jbot? |
| 09:47.08 | z3wb | ok |
| 09:47.11 | z3wb | i installed sox |
| 09:47.20 | tzafrir | z3wb, what distro do you use? |
| 09:47.21 | zeeesh | i hv registered two sip peers like 100 and 200 ... both can call each other ... when they just logged in ... but after 30 minutes or later ... i can c both r logged in .. but unable to receive calls ... ? |
| 09:47.24 | z3wb | xubuntu |
| 09:47.34 | tzafrir | apt-get install sox, I hope |
| 09:47.40 | z3wb | yeah i did that |
| 09:47.42 | z3wb | sox is working fine |
| 09:47.55 | z3wb | but asterisk is still not playing sounds with chan_alsa |
| 09:48.05 | z3wb | and with chan_oss i get garbled sound |
| 09:48.32 | z3wb | now, im using the via82xx driver |
| 09:48.49 | tzafrir | if asterisk is using the sound card with chan_oss, that device is busy, and can't be used with chan_alsa at that time |
| 09:48.55 | tzafrir | and vice-versa |
| 09:48.55 | z3wb | and i had to disable dxs support in /etc/modprobe.d/alsa-base because it made the sound all scratchy |
| 09:49.27 | tzafrir | Which is why it is sane to set both to 'noload' and manually load the one you want for testing |
| 09:50.11 | z3wb | how do i manually load it? |
| 09:50.12 | tzafrir | I think that there's support for pulseaudio in 1.6, but I'm not sure |
| 09:50.24 | dandre | finally it's ok I had to put CHANNEL(language)=fr in the originate commande |
| 09:50.25 | z3wb | well i'm using 1.4.10 |
| 09:50.26 | tzafrir | module load chan_alsa.so |
| 09:50.30 | z3wb | ok |
| 09:51.18 | tzafrir | dandre, hmmm.... I don't think you can set a function. You can set a variable, |
| 09:52.00 | tzafrir | Variable: Channel variable to set, multiple Variable: headers are allowed |
| 09:52.21 | tzafrir | (from 'manager show command Originate') |
| 09:52.34 | z3wb | now asterisk won't even run |
| 09:52.35 | z3wb | fuck |
| 09:52.52 | tzafrir | Variable: LANGUAGE=fr |
| 09:52.54 | *** join/#asterisk giggham (n=giggham@203.110.178.130) |
| 09:53.07 | zeeesh | WARNING[31132]: pbx.c:2525 __ast_pbx_run: Timeout, but no rule 't' in context 'incoming' |
| 09:53.07 | zeeesh | <PROTECTED> |
| 09:53.42 | z3wb | i have to load one of them, otherwise asterisk won't even start |
| 09:54.03 | z3wb | im just going to concentrate on getting chan_alsa to work |
| 09:54.28 | Hyphenex | Damn, now I'm getting a segmentation fault trying to run asterisk |
| 09:54.45 | Hyphenex | I get this before hand though [Mar 4 20:54:32] WARNING[5341]: loader.c:620 inspect_module: Module 'codec_g723.so' was not compiled against a recent version of Asterisk and may cause instability. |
| 09:55.47 | dandre | tzafrir yes I have seen this help but I can say that LANGUAGE=fr doesn't work but CHANNEL(language)=fr works. |
| 09:55.53 | dandre | I am using 1.4.17 |
| 09:56.08 | z3wb | CLI> show version |
| 09:56.36 | z3wb | oh nvm i thought you said "am i using 1.4.17" |
| 10:00.07 | Hyphenex | how do I load a .so file manually? |
| 10:00.15 | z3wb | what the hell |
| 10:00.20 | z3wb | now console dial doesn't work |
| 10:00.54 | z3wb | console dial command not found |
| 10:02.12 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 10:04.47 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 10:04.58 | atop | asterisk crashed and core dumped yesterday at 09:30 and did the same today, at exactly the same time. I've checked for cronjobs and there's nothing at that time. Any suggestions on where to look, although as the time was exactly the same, it's likely to not be an asterisk issue :/ |
| 10:06.16 | *** part/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 10:10.54 | tzafrir | z3wb, so you have nither chan_alsa nor chan_oss loaded, right? |
| 10:11.03 | tzafrir | the dial command comes from them |
| 10:13.10 | *** join/#asterisk cowmix (n=mmarch@71-209-212-132.phnx.qwest.net) |
| 10:13.33 | cowmix | I can't get outbound to work to save my life.. |
| 10:14.10 | cowmix | <PROTECTED> |
| 10:14.16 | z3wb | asterisk won't start without at least one of them loaded |
| 10:17.31 | JT | cowmix: make sure the ip network between the sip destination and your asterisk system is functioning |
| 10:19.23 | cowmix | JT: inbound works fine |
| 10:19.31 | cowmix | I have a SPA3102 (three of them) |
| 10:19.43 | cowmix | the network is local |
| 10:20.11 | JT | cowmix: clearly something is not fine though |
| 10:20.18 | cowmix | JT: yup |
| 10:21.17 | *** part/#asterisk BeeBuu (n=beebuu@219.135.42.4) |
| 10:21.35 | cowmix | output of log: http://openpaste.org/en/5387/ |
| 10:21.46 | cowmix | JT: the end has the error |
| 10:23.10 | JT | [Mar 4 03:18:36] WARNING[4357] chan_sip.c: No such host: spa3 |
| 10:23.15 | JT | seems pretty obvious what the issue is |
| 10:23.38 | cowmix | JT: drum roll plz |
| 10:23.41 | cowmix | :) |
| 10:23.57 | JT | cowmix: i already told you |
| 10:24.04 | JT | btw, this is the wrong channel for freepbx |
| 10:24.28 | cowmix | 10-4.. sorry about that. |
| 10:25.30 | JT | spa3 does not resolve |
| 10:28.04 | *** join/#asterisk steliosk (n=Stelios@athedsl-288865.home.otenet.gr) |
| 10:28.50 | cowmix | i added entries into /etc/host |
| 10:28.51 | *** join/#asterisk docelmo (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net) |
| 10:28.59 | cowmix | JT: and.. they resolve |
| 10:29.18 | cowmix | but now I get this: [Mar 4 03:27:45] VERBOSE[4402] logger.c: -- SIP/spa3-08a49fa0 is circuit-busy |
| 10:36.48 | *** join/#asterisk shinao1 (n=shinao1@35-230.rv.ipnxtelecoms.com) |
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| 10:46.12 | *** part/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 10:46.49 | Hyphenex | JT: you still around? |
| 10:46.52 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:46.58 | agx | anyones knows how to call a number@ipaddress using a Lynksis adapter? |
| 10:47.45 | *** part/#asterisk CrashSys (n=kumba@62-209.187-72.tampabay.res.rr.com) |
| 10:48.20 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-4d157f46ab3b6995) |
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| 10:52.13 | Hyphenex | I'm trying to dial out through MyNetFone, I was just wondering what's wrong with this line: exten => _9x.,1,Dial(SIP/${EXTEN:1}@MyNetFone) |
| 11:09.15 | tzafrir | are there any known issues with a number that includes 00? |
| 11:09.31 | tzafrir | A number that begins with 0? |
| 11:09.37 | tzafrir | in chan_zap (analog) |
| 11:10.08 | tzafrir | I get funny dtmf detection problems with it |
| 11:10.25 | tzafrir | hmm... not exactly dtmf detection |
| 11:12.42 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 11:19.13 | lnx | omg perl still can't get variable with get_variable. But exten => s,n,NoOp(exten_Dialstatus ${DIALSTATUS}) shows it in the log |
| 11:19.28 | *** join/#asterisk hi365 (n=hi365@213.151.52.239) |
| 11:20.00 | lnx | there is no $AGI->verbose before $AGI->get_variable |
| 11:20.36 | *** join/#asterisk RypPn_OuT (i=TuMbL@rosscom.demon.co.uk) |
| 11:21.01 | lnx | it is weird |
| 11:24.03 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 11:24.32 | lnx | i call AGI script after exten => s,n,NoOp(exten_Dialstatus ${DIALSTATUS}) |
| 11:24.42 | lnx | via local channel |
| 11:24.54 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 11:25.12 | lnx | hell it is |
| 11:30.17 | z3wb | can someone give me some alternatives to fwd that allow you to call 800 numbers for free? |
| 11:30.25 | z3wb | im stuck behind NAT |
| 11:30.39 | z3wb | i tried voxalot, but the sound quality is horrible |
| 11:31.41 | *** join/#asterisk mattman99 (n=chatzill@ppp121-44-207-170.lns3.mel4.internode.on.net) |
| 11:32.08 | z3wb | are there any other providers besides freeworlddialup that let you call toll free numbers? |
| 11:34.38 | mattman99 | you can call some with sipphone |
| 11:38.09 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 11:41.15 | *** join/#asterisk steliosk (n=Stelios@athedsl-142674.home.otenet.gr) |
| 11:42.10 | z3wb | im trying to call goog411 with svsip on my ds |
| 11:42.11 | *** join/#asterisk nighty^ (n=nighty@p7154-adsau17honb13-acca.tokyo.ocn.ne.jp) |
| 11:42.39 | z3wb | when i use ideasip, goog411 can hear me perfectly but i can't hear it |
| 11:42.49 | z3wb | fwd doesn't work at all because im behind a nat |
| 11:43.04 | mattman99 | port forwarding? |
| 11:43.05 | z3wb | sipphone and voxalot work, but the sound quality is terrible |
| 11:43.21 | *** join/#asterisk BipBip (n=BipBip@194.65.5.235) |
| 11:43.27 | mattman99 | stunserver? |
| 11:43.36 | z3wb | so im just looking for any voip provider that lets you call 800 numbers |
| 11:44.34 | *** join/#asterisk Dovid (n=Dovid@bzq-79-179-14-220.red.bezeqint.net) |
| 11:47.26 | mattman99 | but you are asking that because you have an ausio problem so why not fix the audio problem? |
| 11:48.26 | mattman99 | then you can use the provider of your choice |
| 11:50.43 | *** join/#asterisk Floppe (i=muppet@crew.cluster-lan.org) |
| 12:00.20 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:08.04 | *** join/#asterisk mattman99 (n=matt@ppp121-44-207-170.lns3.mel4.internode.on.net) |
| 12:08.30 | mattman99 | ? |
| 12:12.25 | z3wb | the audio problem is on their side; i tested it out and even though i couldn't hear them when i was on ideasip, they could hear me perfectly |
| 12:13.12 | z3wb | but when i used voxalot or sipphone we could both hear each other, but it was all distorted |
| 12:16.24 | mattman99 | you cant hear them because your firewall is stopping the packet is my guess |
| 12:18.46 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
| 12:26.16 | *** part/#asterisk mattman99 (n=matt@ppp121-44-207-170.lns3.mel4.internode.on.net) |
| 12:28.04 | FlatFoot | afternoon all |
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| 13:15.36 | FlatFoot | seems very quiet today is everyone asleep |
| 13:16.09 | Mavvie | "stop snow" is not a valid asterisk CLI command. |
| 13:18.35 | *** join/#asterisk duckz (n=duckz@81.180.102.217) |
| 13:19.32 | FlatFoot | where is the snow ? |
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| 13:30.32 | *** part/#asterisk fugitivo (n=ajf@201-212-100-188.cab.prima.net.ar) |
| 13:30.57 | *** join/#asterisk ruied (n=ruied@89.181.119.28) |
| 13:32.00 | *** join/#asterisk ifnotwhynot (n=davidh@mail.nashua.co.za) |
| 13:34.16 | *** join/#asterisk sbrobou (n=sbrobou@201.48.5.105) |
| 13:35.02 | ifnotwhynot | is there any way of answering the channel on first ring because i have two ring delay before i can see the call in cli> using wctdm24xxp fxo? any help welcome, can't seem to google this one right?? |
| 13:37.12 | *** join/#asterisk ccvp (n=ax@66.0.46.210) |
| 13:38.33 | ccvp | hello fellow internet addicts - are we all looking forward to another long & glorious day of irc/internet addiction? :) |
| 13:40.31 | *** join/#asterisk af_ (n=getsmart@88-149-241-244.dynamic.ngi.it) |
| 13:41.02 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:42.21 | lesouvage | ifnotwhynot: use a answer() line. afaik that is the fastest way. |
| 13:42.50 | *** join/#asterisk beek (n=klinebl@65.211.106.243) |
| 13:46.18 | beasty_ | is it hard to setup a voicemail system ? |
| 13:46.28 | sbrobou | hi fellows. Im getting these errors from asterisk: "chan_zap.c: !! Unable to handle message of type 0xd" and " chan_zap.c: Received error from mtp3 layer: -1". Im using a digium card with 4E1. Asterisk is receiving this error from E1. Anybody know what does it mean? |
| 13:46.51 | [TK]D-Fender | beasty_: "core show application voicemail" , "core show application voicemailmain" |
| 13:47.36 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:48.46 | ccvp | hello fender |
| 13:48.51 | ccvp | are you ready for another long & glorious day of irc? :) |
| 13:48.56 | ccvp | Tha Killaz :) heh |
| 13:49.31 | lesouvage | la |
| 13:49.52 | [TK]D-Fender | ccvp: and it was not "Tha Killaz" |
| 13:52.17 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:54.50 | ccvp | jeeez, in #Politics is going in sane now |
| 13:54.52 | ccvp | saying that all people who support atheism support child abuse, and child molestation, because the evils are equal in nature |
| 13:55.16 | ccvp | wonder how long that channel has been on freenode |
| 13:57.25 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 13:58.01 | *** part/#asterisk ddunavant (n=David@pool-71-178-102-155.washdc.east.verizon.net) |
| 13:58.01 | coppice | oh. bummer |
| 13:58.26 | *** join/#asterisk ddunavant (n=David@pool-71-178-102-155.washdc.east.verizon.net) |
| 14:00.22 | *** join/#asterisk Cyorxamp (n=Cyorxamp@212.57.232.254) |
| 14:00.39 | Washy | Hi i'm looking for recs for SIP PSTN service providers |
| 14:01.13 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 14:01.52 | x86 | Washy: les.net |
| 14:02.14 | *** join/#asterisk RoyK_ (n=roy@ti200720a080-5936.bb.online.no) |
| 14:04.40 | *** join/#asterisk jeanmi_i_ (n=besnard@LMontsouris-152-61-20-55.w80-13.abo.wanadoo.fr) |
| 14:06.20 | lnx | [TK]D-Fender: i have made a call via local channel, but perl AGI still can't get value of ${DIALSTATUS} . Can you check http://pastebin.com/m6dbbf81d please :) |
| 14:06.33 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 14:06.35 | ifnotwhynot | [TK] help pleasis there any way of answering the channel on first ring because i have two ring delay before i can see the call in cli> using wctdm24xxp fxo? any help welcome, can't seem to google this one right?? |
| 14:07.32 | [TK]D-Fender | lnx: - Executing NoOp("Local/10@call-file-test-f802,2", "exten_Dialstatus BUSY") in new stack |
| 14:07.42 | [TK]D-Fender | lnx: looks like its reporting a status to me... |
| 14:07.53 | jeanmi_i_ | hi |
| 14:07.54 | [TK]D-Fender | lnx: and you are showing only useless tiny tidbits. |
| 14:08.08 | *** join/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 14:08.08 | [TK]D-Fender | lnx: show me the WHOLE picture |
| 14:08.09 | jeanmi_i_ | has asterisk 1.6 support for SIP over TLS ? |
| 14:08.23 | stansmith | wow there were a lot of hilary and obama supporters out today |
| 14:08.33 | cpm | run! |
| 14:08.35 | [TK]D-Fender | ifnotwhynot: And you have no idea why its waiting for 2 rings? |
| 14:08.54 | lnx | [TK]D-Fender: why logscript.pl: get_variable DIALSTATUS = ? |
| 14:09.28 | lnx | [TK]D-Fender: perl does not get the value of ${DIALSTATUS :(} |
| 14:09.29 | JayTee52 | jeanmi_i_, from what I was reading of the features in the SVN trunk notes for 1.6 it will have support for SIP over TCP and TLS |
| 14:10.29 | [TK]D-Fender | lnx: How would I know, you are hiding all the important parts and I can't see WHAT is calling any of that. You somehow felt you only had to show me 1 line of dialplan for which I don't even get a priority number. |
| 14:10.50 | ifnotwhynot | no using s,1,Answer() |
| 14:10.54 | jeanmi_i_ | JayTee52 thanks a lot |
| 14:11.02 | lnx | [TK]D-Fender: okay i'll paste all :) |
| 14:11.10 | tzafrir | ifnotwhynot, disable callerid detection? |
| 14:11.27 | [TK]D-Fender | ifnotwhynot: Why do you THINK * would not answer the call immediately? Come on... this is an easy one... what reason could * have to WAIT for the 2nd ring? |
| 14:11.54 | *** join/#asterisk agx (n=AGX@88.34.216.63) |
| 14:13.33 | beasty_ | mm |
| 14:14.52 | ifnotwhynot | its waiting for callerid i think but i have tis disabled TK? |
| 14:15.01 | *** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net) |
| 14:16.08 | ifnotwhynot | is it because * is answering the call first then does the routing? |
| 14:16.22 | beasty_ | anyone ever saw this ? |
| 14:16.24 | beasty_ | http://paste.ubuntu-nl.org/58367/ |
| 14:16.26 | ifnotwhynot | Are you saying remove the Answer? TK |
| 14:17.19 | ManxPower | Never Answer when you don't have to |
| 14:17.43 | ifnotwhynot | looks like you dont have extension added in your voicemail.conf for a specific user beasty |
| 14:17.53 | ifnotwhynot | thx let me try |
| 14:18.58 | lnx | [TK]D-Fender: http://pastebin.com/m6d46e69d |
| 14:20.10 | [TK]D-Fender | ifnotwhynot: No, its waiting for callerid |
| 14:22.37 | *** join/#asterisk shinao1 (n=shinao1@35-230.rv.ipnxtelecoms.com) |
| 14:23.20 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 14:24.14 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:24.14 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:24.42 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:24.45 | dandre | can I have more than one [globals] section in extensions.conf? |
| 14:25.11 | sbrobou | ?? |
| 14:25.15 | *** join/#asterisk af_ (n=getsmart@88-149-241-244.dynamic.ngi.it) |
| 14:25.20 | sbrobou | global is global |
| 14:25.28 | sbrobou | 1 is enough |
| 14:26.32 | lnx | :) |
| 14:26.33 | dandre | I intend to have 2 included files in my extensions.conf with one globals section in each. This doesn't seem to work |
| 14:27.40 | sbrobou | the word says: 'global'. Is not possible to add 2 sections of global 'cause global is global. |
| 14:27.55 | sbrobou | how asterisk will find what 'global' it needs to use? |
| 14:28.30 | beasty_ | mm |
| 14:28.41 | beasty_ | ifnotwhynot: http://paste.ubuntu-nl.org/58369/ |
| 14:29.20 | stansmith | ~hi all |
| 14:29.22 | jbot | Many greetings, all, most strange traveller, to this IRCdom of plenty. |
| 14:30.05 | sbrobou | dandre: it is not possible. You will need to find another way to follow |
| 14:31.10 | dandre | I was thinking of some concatenation mecanism |
| 14:31.24 | dandre | ok I try another way |
| 14:31.50 | sbrobou | :) |
| 14:32.35 | [TK]D-Fender | dandre: No. Don't include 2 files with the same context heading in it. just include both files UNDER the heading appearing under extensions.conf itself. |
| 14:32.41 | cmantito | can't you use like +[context] ? |
| 14:32.47 | lnx | [TK]D-Fender: i almost finish asterisk book :)) |
| 14:33.40 | tzafrir | cmantito, [context](+) |
| 14:33.50 | cmantito | would that work for his globals problem? |
| 14:33.52 | tzafrir | dandre, ==^ |
| 14:33.54 | cmantito | [globals]+ |
| 14:34.09 | cmantito | I've never used it, just remember reading bout it |
| 14:34.24 | tzafrir | the same syntax. You can continue any asterisk configuration section with the [section](+) syntax |
| 14:34.33 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:34.33 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:35.04 | [TK]D-Fender | lnx: Your Originate is RECURSIVE Go look at what you're asking it to do. |
| 14:35.59 | ManxPower | tzafrir: Yes. |
| 14:36.41 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 14:38.32 | beasty_ | [TK]D-Fender: can you tell me the default asterisk language ? |
| 14:38.43 | russellb | en |
| 14:38.50 | jeanmi_i_ | I am looking for a sip phone (softphone) that would support TLS (either for linux or windows) |
| 14:39.04 | tzafrir | russellb, explicitly "en", or empty? |
| 14:39.18 | russellb | well, probably empty, and empty is treated as "en" |
| 14:39.19 | lnx | [TK]D-Fender: umh where :/ |
| 14:39.36 | beasty_ | russellb: can you tell me why my asterisk takes 'nl' as default language ? |
| 14:39.52 | *** join/#asterisk ddunavant (n=David@pool-71-191-18-192.washdc.east.verizon.net) |
| 14:40.04 | russellb | beasty_: there are a lot of places where langues are set, so i don't know |
| 14:40.57 | beasty_ | not in /etc/asterisk/*.conf |
| 14:40.59 | [TK]D-Fender | lnx: You are originating as "Local/10@call-file-test" and look where you DUMP THEM after they answer |
| 14:41.31 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:43.37 | *** part/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net) |
| 14:45.40 | *** join/#asterisk RoyK_ (n=roy@box36.fortel.no) |
| 14:46.09 | *** join/#asterisk RoyK_ (n=roy@box36.fortel.no) |
| 14:47.56 | stansmith | beasty_: /etc/zaptel.conf ? |
| 14:49.10 | Washy | Hi i'm looking for recs for SIP PSTN service providers |
| 14:49.26 | *** join/#asterisk Schreiber1337 (n=chatzill@spectrumcontrol.com) |
| 14:51.11 | [TK]D-Fender | ~itsplist-us |
| 14:51.12 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, or http://www.jnctn.com, or http://www.bandwidth.com, or http://vitelity.net |
| 14:51.17 | *** join/#asterisk ruied (n=ruied@89.181.119.28) |
| 14:51.23 | stansmith | best comment ever ? ^^ |
| 14:51.24 | *** join/#asterisk tobias (n=tobias@cpe-076-182-087-105.nc.res.rr.com) |
| 14:51.34 | [TK]D-Fender | lnx: And while you're at it, enable agi debug when you do test calls |
| 14:51.37 | stansmith | oops sorry |
| 14:53.09 | Schreiber1337 | Has anyone had problems where cdr_addon_mysql.so doesn't exist after installing asterisk-addons-1.4.6 |
| 14:53.29 | stansmith | load module cdr_addon_mysql.so |
| 14:53.43 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 14:54.19 | lnx | [TK]D-Fender: the peroblem is in extensoins.conf? |
| 14:54.22 | beasty_ | stansmith: don't have a zaptel.conf |
| 14:54.52 | [TK]D-Fender | lnx: 1 problem is your originate (conceptual design flaw), 2nd I told you to enable agi debug so you can actually see whats going on. |
| 14:54.55 | Washy | Does anyone like FWD, InPhoneX, or SIPPhone? |
| 14:55.07 | drmessano | lol |
| 14:55.17 | stansmith | lol |
| 14:55.38 | Schreiber1337 | atansmith: WARNING[15727]: loader.c:363 load_dynamic_module: Error loading module 'cdr_addon_mysql.so': /usr/lib/asterisk/modules/cdr_addon_mysql.so: cannot open shared object file: No such file or directory |
| 14:56.02 | sbrobou | did you copy the file to this directory? |
| 14:56.11 | stansmith | Schreiber1337: recompile that module or make sure its in that directory |
| 14:56.35 | stansmith | Schreiber1337: `find / | grep cdr_addon_mysql.so` |
| 14:56.45 | stansmith | as root |
| 14:58.25 | *** join/#asterisk el_4_jinete (n=root@mail.pulxar.com.co) |
| 14:58.25 | *** join/#asterisk wmaulik (n=wmaulik@158.59.192.218) |
| 14:58.25 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.44.5) |
| 14:58.25 | el_4_jinete | Hi all |
| 14:58.25 | stansmith | ~hi el_4_jinete |
| 14:58.26 | jbot | Many greetings, el_4_jinete, most strange traveller, to this IRCdom of plenty. |
| 14:58.41 | *** join/#asterisk eharris (n=eharris@75-43-20-21.lightspeed.austtx.sbcglobal.net) |
| 14:59.01 | agx | Does Asterisk 1.6 still has the problem to freeze all the SIP phones due to a blocking DNS request (xDSL down, etc.) ? |
| 14:59.09 | el_4_jinete | Maybe someone could help me ... |
| 14:59.20 | stansmith | el_4_jinete: maybe you could ask a question ... |
| 14:59.34 | *** join/#asterisk Deeewayne (n=dwayne@216.207.245.1) |
| 14:59.34 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:59.59 | el_4_jinete | I've some errors related with zapata, but I dont know that means |
| 15:00.19 | stansmith | ~pb | el_4_jinete |
| 15:00.28 | stansmith | ~pb |
| 15:00.28 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:00.42 | el_4_jinete | callerid.c: fsk_serie made mylen < 0 (-1) |
| 15:00.46 | stansmith | ~pb |
| 15:00.46 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:01.37 | el_4_jinete | and this other |
| 15:01.58 | el_4_jinete | zaptel Disabled echo canceller because of tone (rx) on channel 7 |
| 15:02.02 | stansmith | hi el_4_jinete |
| 15:02.04 | stansmith | ~pb |
| 15:02.04 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:02.41 | el_4_jinete | and following the call hungs up |
| 15:02.41 | stansmith | (try pb-ing your zapata.conf) |
| 15:02.41 | el_4_jinete | hi stansmith |
| 15:03.02 | el_4_jinete | sorry I don't know the meaning of pb-ing |
| 15:03.06 | stansmith | ~pb |
| 15:03.07 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:03.25 | el_4_jinete | haa ok, paste bin |
| 15:03.38 | Washy | Does anyone like FWD, InPhoneX, or SIPPhone? |
| 15:03.41 | zeeesh | without using smtp how can asterisk send emails? |
| 15:03.48 | el_4_jinete | let me do that |
| 15:03.57 | tzanger | zeeesh: telnet? |
| 15:03.57 | Nugget | telnet is eeeeeeevil! |
| 15:04.36 | [TK]D-Fender | stansmith: You're just as bad for spamming the PB notice |
| 15:04.52 | *** join/#asterisk ddunavant (n=David@pool-71-191-18-192.washdc.east.verizon.net) |
| 15:05.28 | stansmith | [TK]D-Fender: i know... :-( |
| 15:05.55 | [TK]D-Fender | el_4_jinete: The "Disabled echo canceller because of tone (rx) on channel 7" warning is because zaptel detected a fax or modem tone on the line and disabled the EC so as not to disrupt it. |
| 15:07.04 | [TK]D-Fender | el_4_jinete: first looks like an incomplete CID |
| 15:07.07 | ifnotwhynot | is there anyway to play the whole Background(welcome-all) before accepting digits to route calls |
| 15:07.14 | ifnotwhynot | ? |
| 15:07.25 | [TK]D-Fender | ifnotwhynot: Yeah, use playback and not background. |
| 15:07.37 | [TK]D-Fender | ifnotwhynot: You seem to be missing the point of these applications. |
| 15:07.42 | stansmith | i knew that one |
| 15:07.47 | [TK]D-Fender | ifnotwhynot: "core show applications" |
| 15:07.50 | el_4_jinete | Thanks [TK]D-Fender |
| 15:08.03 | *** join/#asterisk joshaidan (n=joshaida@S0106001c1023e838.tb.shawcable.net) |
| 15:08.03 | el_4_jinete | My zapata.conf is in pb now |
| 15:08.06 | [TK]D-Fender | ifnotwhynot: You should stop and actually read the instructiosn for all of *'s applications... |
| 15:08.23 | stansmith | ifnotwhynot: http://www.voip-info.org/wiki/index.php?page=Asterisk+-+documentation+of+application+commands |
| 15:08.27 | stansmith | very useful ^ |
| 15:08.28 | [TK]D-Fender | el_4_jinete: Feel like sharing the LINK to your paste please? We aren't psychic you know... |
| 15:08.53 | [TK]D-Fender | stansmith: dated... better to view from CLI first and refer to the WIKI only when it isn't answered in CLI or the BOOK. |
| 15:09.09 | stansmith | :-/ |
| 15:09.22 | el_4_jinete | ok |
| 15:09.31 | el_4_jinete | http://pastebin.com/m710f993d |
| 15:11.30 | SteveTotaro | the wiki is still very helpful |
| 15:11.43 | stansmith | its a nice start to say the least |
| 15:12.15 | SteveTotaro | and if something is "dated" then the very nature of a wiki allows it to be 'updated" |
| 15:12.52 | Washy | Does anyone like FWD, InPhoneX, or SIPPhone? |
| 15:13.05 | stansmith | Washy: no |
| 15:13.07 | SteveTotaro | i am sure someone does |
| 15:13.14 | stansmith | we all hate it! |
| 15:13.18 | [TK]D-Fender | el_4_jinete: ">callerid.c: fsk_serie made mylen < 0 (-1)" You seem to be receiving an FSK and have disabled CID in zapata. Perhaps you should enable it and set it for your country's standard. |
| 15:13.20 | SteveTotaro | ~hate |
| 15:13.21 | jbot | Oh, you hate your job? Why didn't you say so? There's a support group for that. It's called EVERYBODY, and they meet at the bar. --Drew Carey |
| 15:13.29 | stansmith | lol |
| 15:13.44 | [TK]D-Fender | SteveTotaro: as in "Yes Steve you can go right ahead and fix it for us!" :) |
| 15:13.56 | SteveTotaro | no, i don |
| 15:13.58 | ifnotwhynot | [TK] just looking at my options the problem i am having is that i my pbx is sending me dtmf digits on channel answer i need these digits to route the call seeing * is waiting for these digits there is a 3 second pause, i relise now that is is not * delaying the dtmf digits sometimes i can get up to 20 digits, the problem i have is the customer is waiting(hearining nothing for 4 seconds and then hangs up the call, i need to eliminate that 4 second |
| 15:14.02 | SteveTotaro | 't RTMF |
| 15:14.08 | SteveTotaro | i figure it out on my own |
| 15:14.39 | Washy | you know what I mean |
| 15:14.45 | ifnotwhynot | good job SteveTotaro |
| 15:15.00 | [TK]D-Fender | ifnotwhynot: that is a nasty run-on-sentence and I can't follw you. Try again... |
| 15:15.01 | SteveTotaro | i try |
| 15:15.14 | ifnotwhynot | nooooooooooooo here goes |
| 15:15.29 | ifnotwhynot | what is that link to pastbin again please |
| 15:15.48 | SteveTotaro | ~pb |
| 15:15.49 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:15.53 | ifnotwhynot | thx |
| 15:16.06 | drmessano | Washy: Was is it you're looking for? |
| 15:16.09 | stansmith | i would like some senior advice - is it worth it to echo XML code in php or should i use SimpleXML ? (my php echos XML for use in asterisk) |
| 15:16.12 | el_4_jinete | [TK]D-Fender> is the country standard in zaptel.conf or in zapata.conf? |
| 15:16.30 | SteveTotaro | XML is teh suck |
| 15:16.44 | SteveTotaro | the one in /etc |
| 15:16.56 | [TK]D-Fender | el_4_jinete: both. This is worth looking up on the WIKI because of internationalization information |
| 15:17.13 | SteveTotaro | NOT the WIKI!!! for God's sake!!! |
| 15:17.23 | Washy | drmessano: Dirt cheap phone service |
| 15:17.26 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 15:17.26 | Daviey | SteveTotaro: /me cuddles xml |
| 15:17.27 | drmessano | lol |
| 15:17.29 | el_4_jinete | I have that in zaptel.conf loadzone = fr |
| 15:17.48 | SteveTotaro | frog, huh? |
| 15:18.10 | Washy | I'd also like voicemail if possible |
| 15:18.13 | SteveTotaro | for sale, used gun, never fired, dropped twice |
| 15:19.00 | SteveTotaro | washy, grandcentral is part of your puzzle |
| 15:19.37 | drmessano | Washy: Services like FWD and Sipphone work for FREE user <> user calling, but you can find a real ITSP with lower rates.. You're not gonna find "DIRT CHEAP" however.. |
| 15:19.38 | Washy | <PROTECTED> |
| 15:19.39 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, or http://www.jnctn.com, or http://www.bandwidth.com, or http://vitelity.net |
| 15:20.00 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 15:20.06 | Washy | Isn't SIP 2 SIP calling always free? |
| 15:20.06 | SteveTotaro | depends on definition of dirt cheap |
| 15:20.13 | drmessano | VoIP isn't about "ZOMG 1000 minutes for 3 cents" |
| 15:20.22 | stansmith | voip? |
| 15:20.29 | SteveTotaro | ~vonage |
| 15:20.29 | jbot | methinks vonage is a bunch of monkeys |
| 15:20.33 | Washy | It is to me |
| 15:20.38 | drmessano | SteveTotaro: Classic case of "1 billion minutes for 8 cents, pls" |
| 15:20.50 | SteveTotaro | i think they are relatively dirt cheap (vonage) |
| 15:20.57 | jameswf-home | I use PIOV its a little backwards but nat traversial kicks arse |
| 15:20.58 | Washy | I want a miniscule monthly rate |
| 15:21.06 | Washy | I don't talk much |
| 15:21.21 | drmessano | Washy: and I want a woman that does all the things the other ones didn't do.. Life is hard |
| 15:21.29 | SteveTotaro | lol |
| 15:21.40 | drmessano | If you don't take much, pay per minute and don't worry about it |
| 15:21.43 | drmessano | talk* |
| 15:21.45 | SteveTotaro | was trying to come up with a good one |
| 15:21.47 | stansmith | LOL |
| 15:22.11 | el_4_jinete | Thank you very much!! ;) |
| 15:22.15 | SteveTotaro | but openly hits on the women folks in irc.... |
| 15:22.28 | Washy | Take teliax's pay as you go |
| 15:22.38 | Washy | is that $10/month |
| 15:22.46 | SteveTotaro | i like vitelity |
| 15:23.10 | Washy | or 10 just for minutes |
| 15:23.13 | jameswf | I have spent about 10 years in telephony and have never met a female tech I would hit on.... |
| 15:23.15 | SteveTotaro | pre pay don't use, only monthly charge is for DIDs |
| 15:23.34 | SteveTotaro | i think i pay $.50/mo per toll free with vitelity |
| 15:23.39 | jameswf | alot of em are like carnies with manhands |
| 15:23.48 | stansmith | jameswf: my future jsut became very bleek. thanks guy |
| 15:23.52 | SteveTotaro | it's the sales reps |
| 15:23.53 | *** join/#asterisk RoyK (n=roy@ti200720a080-5936.bb.online.no) |
| 15:24.01 | Washy | what is Billed 60/6 |
| 15:24.13 | SteveTotaro | 3com has a very attractive one for the DC metro area |
| 15:24.27 | drmessano | stansmith: "Hot" "Telephony" "Chicks" <--- Pick 2 |
| 15:24.32 | jameswf | 1 out of how many lol] |
| 15:24.38 | JenniferAkemi | wow. |
| 15:24.40 | JenniferAkemi | you guys are harsh |
| 15:24.48 | jameswf | sorry |
| 15:24.53 | drmessano | Except for you JenniferAkemi, but you're too married |
| 15:24.57 | JenniferAkemi | heh |
| 15:25.04 | jameswf | drmessano: kiss ass |
| 15:25.10 | SteveTotaro | as opposed to not too married |
| 15:25.12 | JenniferAkemi | just in case right? ;) |
| 15:25.19 | JenniferAkemi | who knows. maybe i'm super hot |
| 15:25.22 | drmessano | I must be getting too mature.. I NEVER thought I would tell someone they're "too married" |
| 15:25.27 | drmessano | Blah, losing my touch |
| 15:25.35 | stansmith | drmessano: lol true |
| 15:25.38 | SteveTotaro | for asian women there is no middle ground |
| 15:25.39 | jameswf | I am a bit of a pessimest so I doubt it :) |
| 15:25.51 | JenniferAkemi | i'm only half asian |
| 15:25.56 | SteveTotaro | they are either very attractive or NOT |
| 15:25.57 | stansmith | hence the name |
| 15:26.00 | jameswf | I am married only one woman is hot caus thats what I am told |
| 15:26.03 | ccesario | hello, I make the follwoing agi script... http://pastebin.com/m50f51636 .... its work, http://pastebin.com/m30c7b3ca, but if execute verbose command before GET VARIABLE, its dont work... |
| 15:26.45 | ifnotwhynot | http://paste.lisp.org/display/56791 hope this make sence TK thanks for looking |
| 15:27.25 | jeanmi_i_ | hi |
| 15:27.31 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:27.43 | stansmith | ccesario: i read something about that once, im finding a link 1 sec |
| 15:27.46 | jeanmi_i_ | I have compiled asterisk 1.6 and configured it so that it will use TLS |
| 15:27.50 | drmessano | We had this woman who service our PBX a few years back that was so-so (A clear 5.5 on hotornot), and then I met her life partner |
| 15:27.57 | drmessano | I knew then.. Wrong occupation |
| 15:28.05 | *** join/#asterisk ruied (n=ruied@89.214.74.160) |
| 15:28.06 | [TK]D-Fender | ifnotwhynot: You should be listening for DTMF immediately so background is the way to go. |
| 15:28.07 | ccesario | stansmith, nice! |
| 15:28.09 | jameswf | ~wife |
| 15:28.10 | jbot | well, wife is the Wide Interface File Engine |
| 15:28.16 | stansmith | http://www.voip-info.org/tiki-index.php?page=Asterisk+perl+agi |
| 15:28.20 | jameswf | yes |
| 15:28.20 | stansmith | read the comments at the bottom |
| 15:28.21 | jeanmi_i_ | I was expecting to have asterisk listening on tcp port 5061, which it is not. when I start asterisk everything looks fine though |
| 15:28.25 | ccesario | see the results with and without verbose |
| 15:28.28 | ccesario | http://pastebin.com/m4ad5fd0c |
| 15:28.54 | drmessano | jeanmi_i_: What version of Asterisk? |
| 15:28.55 | SteveTotaro | change bindport to 5061 in sip.conf |
| 15:28.55 | sbrobou | i'm looking for Matthew Friedrik. Are you here, man? |
| 15:29.00 | ccesario | stansmith, going... |
| 15:29.12 | sbrobou | sarava |
| 15:29.16 | drmessano | SteveTotaro: tcp? |
| 15:29.19 | jeanmi_i_ | drmessano 1.6 |
| 15:29.24 | drmessano | Ah |
| 15:29.29 | drmessano | I will STFU now |
| 15:29.29 | ifnotwhynot | TK do you know of something else i can try? |
| 15:29.47 | drmessano | You know thats a BETA, right? |
| 15:29.53 | drmessano | Like... the fish |
| 15:30.06 | jeanmi_i_ | drmessano I know |
| 15:30.07 | jameswf | *google |
| 15:30.17 | SteveTotaro | they hide the fact it is beta by just using the letter b |
| 15:30.38 | SteveTotaro | i have 1.8.0.0.1a |
| 15:30.42 | drmessano | b doesn't mean bacon? |
| 15:30.46 | ifnotwhynot | b stands for beta |
| 15:30.52 | stansmith | wtf is beta |
| 15:30.53 | ifnotwhynot | or bear |
| 15:31.01 | ifnotwhynot | or beer |
| 15:31.01 | SteveTotaro | i am the alpha |
| 15:31.02 | drmessano | BROWNIES |
| 15:31.10 | stansmith | p=prodigal |
| 15:31.24 | drmessano | I am the alpha, I am the beta, coo coo cachoo?? |
| 15:31.30 | SteveTotaro | beta beat out vhs |
| 15:31.43 | SteveTotaro | although it was a better technology |
| 15:31.55 | drmessano | Now I have a Weird Al version of "I am the Walrus" stuck in my head |
| 15:31.58 | ifnotwhynot | you beta focus on asterisk |
| 15:32.01 | [TK]D-Fender | ifnotwhynot: Use background and make it a normal IVR. You need to start collecting digits the moment the start sending them. |
| 15:32.01 | SteveTotaro | i am the alpha and the omega |
| 15:32.19 | drmessano | You Beta, you beta, you bet |
| 15:32.26 | SteveTotaro | lol |
| 15:32.30 | jeanmi_i_ | SteveTotaro I have set tlsbindaddr to myIP:5061 (event though :5061 is default) and there is still nothing listening on port 5061 |
| 15:33.00 | SteveTotaro | i have not even thought about touching 1.6 yet |
| 15:33.26 | stansmith | is 1.4 out? |
| 15:33.37 | jeanmi_i_ | SteveTotaro ok I found an error regarding my cert in the logs so this is why .... |
| 15:33.39 | SteveTotaro | i did my first 1.4 install last week |
| 15:33.48 | SteveTotaro | yeah check your logs |
| 15:33.50 | [TK]D-Fender | stansmith: been out for over a year.... |
| 15:33.51 | *** join/#asterisk simbol76ss (n=simbol@ip-212-26.sn1.eutelia.it) |
| 15:34.14 | ccesario | stansmith, the "\n" appear the problem....... |
| 15:34.15 | jameswf | 1.4 with the exception of a hand full or subversions has been pretty good |
| 15:34.24 | tzanger | interesting, I just got my first hands-on with a macbook air |
| 15:34.26 | tzanger | ok, it's thin |
| 15:34.30 | tzanger | very thin |
| 15:34.30 | SteveTotaro | i was forced to do 1.4 because then new sangoma drivers don't have to mess with the 1.4 zaptel drivers |
| 15:34.32 | tzanger | I'd go as far as to say too thin, I'm afraid of bending/breaking it |
| 15:34.35 | tzanger | keyboard isn't bad for typing though |
| 15:34.36 | ccesario | stansmith, I'll try test ... |
| 15:34.40 | drmessano | macbook air: It fits in an envelope |
| 15:34.46 | tzanger | drmessano: yeah it definitely does |
| 15:34.57 | drmessano | Anyone see the comparison of the Commodore 64 and Macbook air? |
| 15:35.15 | Daviey | in 18months, some people will be using 1.6 |
| 15:35.20 | Daviey | :) |
| 15:35.21 | SteveTotaro | more like the timex sinclair |
| 15:35.25 | stansmith | guys, i was jk about 1.4 |
| 15:35.28 | tzanger | I don't really see the need for *that* thin though... make it a little bit (not much) thicker for more battery and a regular hdd... |
| 15:35.28 | drmessano | http://www.flickr.com/photos/ajaxed/2212112946/ |
| 15:35.31 | tzanger | the thin screen is nice |
| 15:35.32 | drmessano | There you go |
| 15:35.34 | drmessano | Go look |
| 15:35.35 | tzanger | the big big touchpad is nice |
| 15:35.36 | jameswf | as long as your not a bleading edge monkey who jumps on a release 2 seconds after its out you usualy know in a week what a version will look like |
| 15:35.46 | stansmith | woz was bashing the macbook air |
| 15:35.50 | stansmith | link on slashdot yesterday |
| 15:36.06 | tzanger | I actually had an sx64 |
| 15:36.11 | stansmith | see i call him woz cuz i know him like that |
| 15:36.13 | *** join/#asterisk hi365 (n=hi365@213.151.52.239) |
| 15:36.14 | SteveTotaro | http://www.troubleshooters.com/lpm/200610/timex.jpg |
| 15:36.16 | jameswf | I want a macbook air but I wont pay for 1... too short of a life span |
| 15:36.22 | drmessano | I wanted an SX-64.. But my parents were cheap |
| 15:36.33 | hi365 | how can iset the gender for Sayunixtime? |
| 15:36.44 | drmessano | Surgery? |
| 15:36.49 | stansmith | lol |
| 15:36.52 | jameswf | I am all ghetto cool I like to get 2-3 years from my laptops |
| 15:36.55 | tzanger | drmessano: oh I had a standard c64 growing up |
| 15:36.57 | tzanger | and tape drive |
| 15:37.02 | tzanger | didn't get a disk drive for many years |
| 15:37.02 | Daviey | an eee and lenovo tablet do me fine |
| 15:37.05 | drmessano | I still have my C64 |
| 15:37.05 | hi365 | lol, not what i meant though |
| 15:37.07 | tzanger | got hte sx64 at a surplus store |
| 15:37.10 | tzanger | like 10 years ago |
| 15:37.17 | tzanger | drmessano: I have one too (not my original) |
| 15:37.20 | jameswf | lenovo bah |
| 15:37.21 | tzanger | I had about 6 of them at one point |
| 15:37.35 | drmessano | I kept burning out power supplies |
| 15:37.37 | tzanger | jameswf: no bah... thinkpads are the best. the "low end" lenovo stuff I hate though |
| 15:37.40 | Daviey | jameswf: sturdiest laptop i've ever owned |
| 15:37.41 | SteveTotaro | i have two C=64s a floppy drive, tape drive, and abox of discs |
| 15:37.44 | tzanger | drmessano: I built my own :-) |
| 15:38.05 | SteveTotaro | i have the last ibm thinkpad before it went lenovo |
| 15:38.14 | SteveTotaro | 1.7ghz centrino |
| 15:38.17 | Daviey | "built" <-- bet no soldering iron came out |
| 15:38.21 | drmessano | The bricks that came with the C64s were actually built to handle less current than the C-64 needed.. by almost 1/2 amp |
| 15:38.29 | jameswf | I owned a few IBM made think pads and had mixed feelings... not sure I want to take the for lack of a better term thinkpad afterbirth |
| 15:38.57 | *** join/#asterisk RoyK_ (n=roy@ti200720a080-5936.bb.online.no) |
| 15:38.58 | SteveTotaro | i have a first get pentium thinkpad |
| 15:39.03 | Daviey | getting used to a nipple again was fun :) |
| 15:39.06 | tzanger | I have never had any issue with my thinkpads |
| 15:39.11 | tzanger | I swear by 'em |
| 15:39.14 | SteveTotaro | the black coating has come off and it is shiny metal underneath |
| 15:39.17 | drmessano | I got a nice aftermarket one that just needs the caps shotgunned in it every 15 years |
| 15:39.19 | jameswf | always wanted to try out the thinkpad airbags... |
| 15:39.26 | Daviey | lenovo run linux dandy |
| 15:39.40 | ifnotwhynot | same delay TK think i need to used normal ivr and get them to log onto cust database |
| 15:39.50 | jameswf | been looking at eepc those look popular for low power |
| 15:39.57 | SteveTotaro | i played with the nipple too much, it fell off |
| 15:40.06 | jameswf | dirty |
| 15:40.08 | Daviey | jameswf: i lie my eee |
| 15:40.10 | Daviey | like* |
| 15:40.29 | stansmith | lies |
| 15:40.34 | jameswf | my shnell computer only did 2 hours on my 3 hour flight |
| 15:40.52 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-58e7d18723c00b1f) |
| 15:41.18 | stansmith | hi, im new to the coffee drinking game, how many cups is too much for one morning? |
| 15:41.20 | Daviey | jameswf: and i bet there we no unsecured wifi's nearby either :/ |
| 15:41.28 | jameswf | shows how well apm has improved in linux though... think a year ago it would have been closer to an hour or 45 mon |
| 15:41.29 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 15:41.31 | jameswf | *min |
| 15:41.34 | [TK]D-Fender | ifnotwhynot: I just told you to do a normal IVR for this. |
| 15:41.55 | SteveTotaro | God has spoken |
| 15:42.10 | jameswf | we have a 1 pot per employee limit |
| 15:42.28 | SteveTotaro | i work in a drug free environment |
| 15:42.40 | stansmith | caffiene = worst drug |
| 15:42.41 | Daviey | but when you get home... |
| 15:42.45 | stansmith | kaffiene = worst media player |
| 15:43.01 | Daviey | kde = :( |
| 15:43.10 | SteveTotaro | i am so hooked on caffiene |
| 15:43.37 | SteveTotaro | has no effect unless i don't have any, then the worst throbbing headache you can imagine |
| 15:44.00 | stansmith | im trying not to get used to it |
| 15:44.02 | SteveTotaro | i can drink a as much as i want and go right to sleep |
| 15:44.09 | stansmith | is that healthy? |
| 15:44.16 | SteveTotaro | what is? |
| 15:44.28 | stansmith | drinking mad coffee then passing out |
| 15:44.46 | SteveTotaro | i take multivitamins to offset the caffeine and marlboros |
| 15:45.10 | stansmith | SteveTotaro: you never fail to amaze me with your wisdom |
| 15:45.17 | stansmith | great info for the budding mind |
| 15:45.20 | SteveTotaro | i can lucid dream, very practiced at it |
| 15:45.28 | stansmith | ive tried it, came close once |
| 15:45.33 | Daviey | ~SteveTotaro |
| 15:45.34 | jbot | rumour has it, stevetotaro is an IRC nub |
| 15:45.34 | ccesario | stansmith, hmmmm no success withou "\n" |
| 15:46.03 | stansmith | ccesario: what language are you using agi with? |
| 15:46.03 | SteveTotaro | ~Daviey |
| 15:46.17 | stansmith | ~stansmith |
| 15:46.19 | Daviey | hah |
| 15:46.22 | stansmith | <jbot> rookie of the year |
| 15:46.31 | stansmith | wow thanks |
| 15:46.36 | ccesario | stansmith, php |
| 15:46.41 | *** part/#asterisk simbol76ss (n=simbol@ip-212-26.sn1.eutelia.it) |
| 15:46.53 | stansmith | can you pastebin your code? |
| 15:47.06 | SteveTotaro | Washy is rookie of the year |
| 15:47.09 | stansmith | :-/ |
| 15:47.31 | stansmith | 3 years ago i was in high school |
| 15:49.08 | ccesario | stansmith, yes |
| 15:49.32 | ccesario | stansmith, http://pastebin.com/m50f51636 |
| 15:49.37 | stansmith | k |
| 15:49.55 | ccesario | the function console_write execute the verbose command |
| 15:49.59 | SteveTotaro | time sure does fly doesn't it stansmith? |
| 15:50.16 | SteveTotaro | you will have your 5 year reunion in two years |
| 15:50.49 | SteveTotaro | any college? |
| 15:50.58 | stansmith | yea but rather not say which one cause you guys will laugh at me |
| 15:51.07 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 15:51.21 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 15:51.31 | SteveTotaro | i went to WVU "Wet V***** University" |
| 15:51.55 | stansmith | devry... come on everyone at the count of three tell me thats not a real school |
| 15:52.36 | SteveTotaro | devry, i see commercials for it all the time |
| 15:52.43 | stansmith | yea...makes me feel cheezy |
| 15:52.52 | SteveTotaro | they teach you how to plug in cat3 cables |
| 15:53.03 | stansmith | they teach you how to write vb.net |
| 15:53.27 | *** join/#asterisk znoG (n=gs@130-215-114-200.fibertel.com.ar) |
| 15:53.48 | SteveTotaro | i am just messing with you, i am sure devry is great |
| 15:54.19 | stansmith | i wasnt disagreeing lol |
| 15:54.44 | stansmith | devry has a lot of resources...i was one of the few that actually used them |
| 15:54.51 | SteveTotaro | don't be ashamed of higher education |
| 15:55.18 | SteveTotaro | even that phoenix university online could be great |
| 15:55.34 | SteveTotaro | if students actually work at it |
| 15:56.03 | stansmith | online class is a joke..really, i just did the work to get the diploma and supplemented learning stuff i was actually interested in, in my free time |
| 15:56.11 | SteveTotaro | i am a fan of self study |
| 15:56.24 | drmessano | I took a two year electronics course from ICS about 10 years ago.. As much as I didn't want to admit I took the "One on TV", I learned a lot more than most other guys doing EE at the time |
| 15:56.56 | SteveTotaro | the only class in highschool that had any value was typing and half way through they took out the typewriters and put in computers and changed the name to keyboarding |
| 15:57.06 | stansmith | keyboarding, lol |
| 15:57.18 | drmessano | ha |
| 15:57.28 | drmessano | Keyboarding.. yes |
| 15:57.37 | SteveTotaro | then i learned "print screen" |
| 15:57.46 | stansmith | ccesario: i know in the perl AGI, you can create an AGI object, can you do that with the php ? |
| 15:57.50 | SteveTotaro | i would just keep pushing it over and over |
| 15:58.01 | SteveTotaro | and the teacher had no way of knowing who was doing it |
| 15:58.06 | stansmith | cause i do $AGI->verbose(...) all day and it works |
| 15:58.18 | SteveTotaro | back then printers were loud as heck |
| 15:58.54 | ccesario | stansmith, I don't using no agi library, only php functions to manage AGI |
| 15:59.40 | SteveTotaro | we don't need no stinking library! |
| 15:59.50 | stansmith | specific reason to that? i started out the same way but found using the AGI class much more easy |
| 16:00.37 | ccesario | stansmith, I can write one class to this, but I need solve this little problem :P |
| 16:01.00 | drfreeze | What can it mean when asterisk will not restart with 'restart when convenient', when all lines are not being used? |
| 16:02.15 | *** join/#asterisk fiXXXerMet (n=kjohnson@69.85.26.2) |
| 16:03.04 | SteveTotaro | one line is being used but not all |
| 16:03.15 | SteveTotaro | unless you only have one line |
| 16:03.22 | SteveTotaro | show channels |
| 16:04.16 | stansmith | ccesario: whats your native speaking language? |
| 16:04.37 | SteveTotaro | he is russian i bet |
| 16:04.48 | SteveTotaro | italian silly |
| 16:05.17 | SteveTotaro | maybe sicilian like myself |
| 16:05.48 | drfreeze | SteveTotaro: how can you tell from 'sip show channels' which lines are being used? |
| 16:06.07 | ccesario | stansmith, portuguese brazil :) |
| 16:06.23 | stansmith | haha your comments in your code threw me off |
| 16:06.25 | SteveTotaro | just do a show channels |
| 16:06.49 | ccesario | stansmith, ahahahahha |
| 16:06.55 | drfreeze | SteveTotaro: ok |
| 16:06.59 | SteveTotaro | or soft tab tab tab |
| 16:07.13 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-54-142.dsl.tul2ok.sbcglobal.net) |
| 16:07.14 | stansmith | ccesario: i think i might know the problem... |
| 16:07.18 | SteveTotaro | to hang one up or get a list of channels |
| 16:07.32 | drmessano | SteveTotaro: I think our families feuded once over some olives |
| 16:07.33 | SteveTotaro | to hangup |
| 16:07.56 | stansmith | exec_command recieves 2 strings, and it in turn, places those strings inside another stricng that gets executed |
| 16:07.59 | SteveTotaro | have you been to italy, sicily? |
| 16:08.28 | SteveTotaro | i went in 2000/2001 over Christmas and New Year |
| 16:08.44 | stansmith | try changing fwrite(STDOUT,"$COMMAND") to just fwrite(STDOUT,$COMMAND) on line 42 |
| 16:08.45 | SteveTotaro | I walked through the great door at the Vatican |
| 16:09.09 | SteveTotaro | but the line was too long going in so i walked through it backwards |
| 16:09.17 | stansmith | haha |
| 16:09.24 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 16:09.36 | SteveTotaro | supposedly, if you walk through the great door, you are cleansed of all of your sins |
| 16:09.50 | SteveTotaro | so walking backwards may be VERY BAD |
| 16:10.12 | jeanmi_i_ | has anyone here already configured asterisk 1.6 with TLS ? asterisk is complaining about my certificate (SSL cert error) and I have no idea what might be wrong with my cert |
| 16:10.28 | ccesario | stansmith, hmmmm |
| 16:10.36 | stansmith | ccesario: you know what im sayin? |
| 16:10.52 | SteveTotaro | you need an M$ or Digium signed cert |
| 16:10.56 | drmessano | I haven't been.. I'm told that there's two villages of Messano's over there.. and they both hate each other and claim to be unrelated |
| 16:11.00 | SteveTotaro | cannot be self signed |
| 16:11.02 | drmessano | Sounds like typical Messano's |
| 16:11.21 | SteveTotaro | same with the Todaro's |
| 16:11.23 | ccesario | stansmith, yes changing |
| 16:11.44 | jeanmi_i_ | SteveTotaro why can't the cert be selfsigned ? |
| 16:11.45 | SteveTotaro | name got changed at ellis island by immigration worker on paper work |
| 16:11.51 | drmessano | ROFL |
| 16:11.53 | SteveTotaro | i am just messing with you |
| 16:11.57 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 16:11.58 | drmessano | I was JUST going to ask that question.. |
| 16:12.00 | stansmith | ccesario: im not entirely sure though,but it doesnt make sense to me to do fwrite(STDOUT,"".."") |
| 16:13.06 | stansmith | changing line 40 to $COMMAND = $STR_CMD." ".$STR_PARAM; might help |
| 16:13.30 | SteveTotaro | stan, do you have steady work? |
| 16:13.46 | stansmith | what you mean |
| 16:13.49 | lirakis | hmm.. im trying to use the Transfer() application, but i keep getting 484 address incomplete. here is the line from extensions.conf |
| 16:13.49 | lirakis | exten => _*99XXX,1,Transfer(SIP/${EXTEN:3:}@dev) |
| 16:13.57 | lirakis | dev is a peer i have setup |
| 16:14.05 | SteveTotaro | do you have a steady jobby job |
| 16:14.10 | stansmith | yea im there right now |
| 16:14.15 | lirakis | and this is the cli output |
| 16:14.17 | lirakis | Executing Transfer("SIP/5000-09491c18", "SIP/333@dev") in new stack |
| 16:14.26 | lirakis | (truncated) .. i can pb the whole output |
| 16:14.34 | SteveTotaro | ~pb |
| 16:14.35 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:14.35 | [TK]D-Fender | lirakis: singular messages like the one you sent are nearly meaningless. please provide full CLI + SIP debug for these kinds of issues. |
| 16:15.10 | stansmith | SteveTotaro: you ask cause im chatty kathy in IRC today? |
| 16:15.31 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 16:15.45 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 16:16.29 | SteveTotaro | no, just wondering, if ccesario comes back and says, "it worked!" then i see a bright future for you |
| 16:16.41 | ccesario | stansmith, but the actual commands works |
| 16:16.42 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 16:16.46 | stansmith | d'oh |
| 16:16.47 | lirakis | <PROTECTED> |
| 16:17.09 | ifnotwhynot | lirakis how long is your extension numbers? |
| 16:17.11 | ccesario | stansmith, http://pastebin.com/m45d5fabf |
| 16:17.13 | [TK]D-Fender | lirakis: ....... |
| 16:17.16 | [TK]D-Fender | lirakis: ...... |
| 16:17.26 | ccesario | but I'll change as you mean.... |
| 16:17.26 | stansmith | ccesario: can u message me in a private chat? |
| 16:17.30 | [TK]D-Fender | lirakis: ----------> SIP DEBUG <--------------- |
| 16:17.34 | ifnotwhynot | _*99XXX,1,Transfer(SIP/${EXTEN:3:??????????????????????/}@dev) |
| 16:17.38 | ccesario | stansmith, yes |
| 16:18.05 | ifnotwhynot | must be _*99XXX,1,Transfer(SIP/${EXTEN:2:3/}@dev) |
| 16:18.16 | ifnotwhynot | must be _*99XXX,1,Transfer(SIP/${EXTEN:2:3}@dev) |
| 16:18.16 | lirakis | [TK]D-Fender: right... |
| 16:18.28 | [TK]D-Fender | ifnotwhynot: Apparently not. Look at its execution. |
| 16:18.43 | lirakis | ifnotwhynot: ? if only one arg is supplied it just strips the front |
| 16:18.46 | [TK]D-Fender | ifnotwhynot: -- Executing Transfer("SIP/5000-09491c18", "SIP/333@dev") in new stack <-- works jsut fine |
| 16:19.03 | ifnotwhynot | k |
| 16:19.36 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:19.56 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:19.56 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:21.52 | lirakis | [TK]D-Fender: full debug http://pastebin.ca/927852 |
| 16:22.15 | lirakis | [TK]D-Fender: fyi i am writing a simple redirect server in C .. and im using the transfer to test it... |
| 16:22.55 | ifnotwhynot | does NoOp application take any time to process meaning in seconds |
| 16:23.34 | [TK]D-Fender | lirakis: Doesn't look like the place you are redirecting to likes what you're sending them... |
| 16:23.43 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-78-61.vif.net) |
| 16:23.49 | [TK]D-Fender | ifnotwhynot: huh?! |
| 16:24.05 | [TK]D-Fender | ifnotwhynot: Normally NoOp Executespretty instantly |
| 16:24.07 | SteveTotaro | nanoseconds |
| 16:25.21 | lirakis | [TK]D-Fender: hmm .. right now its hardcoded to send back a responce.. regardless of the request |
| 16:25.42 | [TK]D-Fender | lirakis: well I don't see the response in there, do I? |
| 16:26.18 | agx | chan_misdn always return DIALSTATUS=CHANUNAVAIL instead of DIALSTATUS=BUSY... any idea why the BUSY from the Telco isn't handled? |
| 16:26.26 | lnx | [TK]D-Fender: if i originate Local/10@plan i can't dial(SIP/number)? in CLI out comes call(Local/10@plan, SIP/number). |
| 16:26.35 | lirakis | [TK]D-Fender: but i also dont see it hitting the redirect server |
| 16:26.57 | lirakis | [TK]D-Fender: .. let me take a sec to look at *'s side of the sip messaging |
| 16:27.02 | [TK]D-Fender | lirakis: did you do a global SIP debug or attempted only 1 peer? |
| 16:27.12 | [TK]D-Fender | lnx: PASTEBIN <-------- |
| 16:27.49 | lirakis | [TK]D-Fender: global |
| 16:28.09 | lnx | [TK]D-Fender: not modified, i'm searched relatien between Originate a channel and Dial(); |
| 16:28.15 | lnx | relation |
| 16:28.19 | lirakis | [TK]D-Fender: i can do 2 .. one for each side |
| 16:28.19 | fiXXXerMet | For the MeetMe() command, 'a' — set admin mode. What exactly does that do? I have an admin pin specified and I can log in as the admin already without that option. |
| 16:28.38 | lirakis | [TK]D-Fender: but i think i got it pretty clean.. not sure though.. looking at it now |
| 16:29.36 | lirakis | [TK]D-Fender: .. if i use Dial(SIP/dev/333) .. everything seems to work fine.. but .. dial doesnt deal with 300 redirect messages |
| 16:30.18 | [TK]D-Fender | lirakis: Go look at the receiving end. |
| 16:30.23 | lirakis | .. as ive heard the devs say a thousand times .. * is not a proxy server ;p |
| 16:31.22 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 16:31.26 | lirakis | [TK]D-Fender: i have.. its basically a udp socket on 5060.. i dump any incoming that i get to stdout.. and i dont see anything when the call is made |
| 16:31.51 | lirakis | [TK]D-Fender: but like i said.. if i do it with dial... i do see the messaging |
| 16:32.00 | generalhan | hey all ! |
| 16:32.12 | [TK]D-Fender | lirakis: Sorry, can't help you from here... |
| 16:32.57 | lirakis | [TK]D-Fender: np .. i really should be testing this on some other equipment any way.. * was just in the network so i figured id try and test it out |
| 16:33.06 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 16:33.17 | SteveTotaro | han solo is that you |
| 16:34.09 | generalhan | i seem to be having some issues with realtime and asterisk writting to the voicemail DB. i have followed the instructions for the CentOS 5 and Asterisk 1.4.x installation guide to the letter, but when i try to record unavail messages, or leave messages for a user i get some odd warning messages |
| 16:34.32 | generalhan | if any one has some experience with this and could take a look for me, i would really appreciate it !! http://pastebin.com/m21e1102c |
| 16:34.42 | fiXXXerMet | Trying to use the standard cdr to record conference billing information, but I am getting a lot of unnecessary stuff. For each caller, I get 3 entries in the database. What is a better way to do this? I need the callerID, date, duration, etc, |
| 16:34.52 | lnx | [TK]D-Fender: http://pastebin.com/m6d46e69d i cant find recursion, if i pick up the phone the server call me again in same time |
| 16:35.14 | *** join/#asterisk ddunavant (n=David@pool-71-191-18-192.washdc.east.verizon.net) |
| 16:35.51 | SteveTotaro | ddunavant, are you in |
| 16:36.01 | SteveTotaro | DC or PG county? |
| 16:36.56 | De_Mon | pg country? |
| 16:37.02 | De_Mon | oh nm, county |
| 16:37.15 | fiXXXerMet | Small world, in PG county myself ;) |
| 16:37.28 | SteveTotaro | my bro is a detective in PG county |
| 16:37.29 | [TK]D-Fender | lnx: Your channel and target are the same. Whats the point? Also you have not enabled AGI debug so we can see why DIALSTATUS isn't being picked up. You could also try copying it to another var for testing AFTER doing a first round debug. |
| 16:37.41 | fiXXXerMet | I just work there |
| 16:38.02 | SteveTotaro | i live in columbia sometimes and baltimore other times |
| 16:38.10 | *** join/#asterisk greekguy8888 (n=alex@c-76-118-204-95.hsd1.ma.comcast.net) |
| 16:38.13 | greekguy8888 | hey all |
| 16:38.17 | SteveTotaro | work wherever it takes me |
| 16:38.25 | fiXXXerMet | Live just outside of Baltimore in Brooklyn Park. Wish I lived in Baltimore though. |
| 16:38.51 | SteveTotaro | i am a mile from pimlico right now |
| 16:39.32 | greekguy8888 | when your cli becomes inaccessible and asterisk seems to still be running fine, is there anyway other than restarting to get back into it |
| 16:39.39 | SteveTotaro | so you commute 295 every day? |
| 16:41.03 | SteveTotaro | i stay off 295 since it is federal, i don't want to go to federal court for speeding ;) |
| 16:41.05 | fiXXXerMet | Yes, and it's a disaster every day. |
| 16:41.17 | fiXXXerMet | I didn't know that |
| 16:41.32 | SteveTotaro | not as bad as my old commute, columbia to vienna va |
| 16:41.37 | fiXXXerMet | gross |
| 16:41.46 | SteveTotaro | 40 miles could take more than three hours |
| 16:42.04 | SteveTotaro | rush hour started thrusday morning and ended friday 9pm |
| 16:42.12 | lnx | [TK]D-Fender: i have turned on, but not posted sorry, http://pastebin.com/m92fbb23 |
| 16:42.19 | fiXXXerMet | That's crazy... |
| 16:43.15 | [TK]D-Fender | lnx: its looking like you didn't reference your variable properly in your perls cript |
| 16:43.23 | JenniferAkemi | does anyone here do load balancing over multiple * boxes? |
| 16:43.35 | JenniferAkemi | I'm trying to figure out the best way to do it |
| 16:44.02 | *** join/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 16:44.12 | DrRighteous | what version of zaptel does asterisk 1.4.17 use? |
| 16:44.57 | SteveTotaro | whatever version you install |
| 16:45.55 | SteveTotaro | jennifer, i seem to recall a long thread on the user's list about load balancing |
| 16:46.08 | SteveTotaro | recently, should be some good info there |
| 16:46.34 | SteveTotaro | can i use zaptel 1.4 with asterisk 1.2? |
| 16:46.35 | JenniferAkemi | i'll look. thanks |
| 16:47.17 | SteveTotaro | that would be good with the new sangoma drivers |
| 16:47.26 | greekguy8888 | does anyone have a solution for a deadlocked cdr while asterisk is still running? |
| 16:47.44 | JenniferAkemi | is there a good way to search the list? or just use google |
| 16:47.55 | SteveTotaro | one second |
| 16:48.51 | SteveTotaro | are you thinking sip extensions? |
| 16:49.02 | lnx | [TK]D-Fender: i don't know how reference could be better than http://pastebin.com/m2feb09ea |
| 16:49.13 | *** join/#asterisk seanbright (i=seanbrig@65.207.74.18) |
| 16:49.17 | *** part/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 16:49.54 | SteveTotaro | ~pb |
| 16:49.54 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:49.54 | JenniferAkemi | SteveTotaro: is that question for me? |
| 16:50.05 | ifnotwhynot | can anyone please explain the timeout option in the application Read(variable[|filename][|maxdigits][|option][|attempts][|timeout]) timeout -- if greater than 0, that value will override the default timeout. i don't quite understand the concept please any help welcome |
| 16:50.08 | [TK]D-Fender | SteveTotaro: No. |
| 16:51.24 | SteveTotaro | no what? |
| 16:51.31 | SteveTotaro | i was asking jennifer |
| 16:51.57 | SteveTotaro | i was going to pb the email threads on load balancing but it didn't work so well |
| 16:52.29 | JenniferAkemi | SteveTotaro: if you want to email them to me that's cool too. |
| 16:53.15 | SteveTotaro | ur email? |
| 16:53.52 | ifnotwhynot | does the timeout imply that it will wait for x amount of seconds before coing to next exten=>?? please Tk help |
| 16:54.10 | JenniferAkemi | SteveTotaro: I msg'd you. |
| 16:54.17 | [TK]D-Fender | ifnotwhynot: I told you to forget "read" and make it a normal IVR. |
| 16:54.28 | fiXXXerMet | What is a good solution for conference cdr/billing information? |
| 16:55.16 | ifnotwhynot | can't need to use cli to route call for added security |
| 16:55.53 | ifnotwhynot | iso9002 must |
| 16:56.37 | SteveTotaro | http://lists.digium.com/pipermail/asterisk-users/ |
| 16:58.28 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:59.37 | adeel | if i've mapped port 5060 to my * box, and opened the appropriate rtp range, should i still set the nat=yes option? (i think i should, otherwise the receiving end will have the wrong ip address, but just double checking) |
| 17:00.39 | SteveTotaro | yes |
| 17:00.54 | SteveTotaro | as long as you are doing nat |
| 17:01.08 | adeel | that's what i thought |
| 17:01.32 | stansmith | language barries..yeesh |
| 17:01.45 | stansmith | i mean barriers |
| 17:03.55 | x86 | language barries sounds funnier ;) |
| 17:04.08 | stansmith | ya i shoulda left it |
| 17:04.12 | adeel | the -L option for * 1.4, is that a local or global loadaverage limit? |
| 17:06.27 | ifnotwhynot | if i record a channel its very soft can one increase the gain somewhere? |
| 17:07.07 | ifnotwhynot | TK took your advice made a normal IVR thx for your help |
| 17:08.37 | adeel | ifnotwhynot, the only channel i'm aware of that you can adjust gain is zapata |
| 17:09.18 | SteveTotaro | you can do it after the fact with sox |
| 17:09.25 | SteveTotaro | or lame or whatever |
| 17:10.16 | ifnotwhynot | thx adeel |
| 17:10.49 | x86 | man... I love these Adit 600 channel banks |
| 17:11.12 | SteveTotaro | starting simple switch |
| 17:11.59 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 17:12.21 | alrs | x86: i have one sitting right here on my desk |
| 17:12.34 | alrs | x86: are they still in production, or are they ebay-only at this point? |
| 17:12.35 | SteveTotaro | i slept so wrong my entire left side is hurting, damn 85# pit bull is a bed hog |
| 17:12.47 | x86 | alrs: I bought mine brand new from VoIP Supply |
| 17:13.13 | SteveTotaro | hope you don't have to RMA it... |
| 17:13.34 | x86 | alrs: I've got (3) Adit 600's with (3)8FXS cards, and (2) Adit 600's with (6)8FXS cards |
| 17:13.45 | Qwell | adeel: what do you mean local or global? |
| 17:13.48 | Qwell | there's only one average |
| 17:14.00 | x86 | SteveTotaro: VoIP Supply is a great vendor, they'll let me return anything for any reason... love working with them |
| 17:14.03 | alrs | x86: you might keep an eye on ebay |
| 17:14.13 | *** join/#asterisk supers (n=supers@animenfo.com) |
| 17:14.17 | x86 | alrs: why? we dont buy used stuff for production ;) |
| 17:14.26 | SteveTotaro | i like them but could not even get an RMA on a single Digium FXO module |
| 17:14.35 | *** join/#asterisk Buana (n=thomasn@p5B054C43.dip.t-dialin.net) |
| 17:14.38 | adeel | Qwell, ah, so it is using the system load average...i wasn't sure if it would use * load average or not |
| 17:14.43 | alrs | I've been using voiplink, but I haven't had to RMA anything |
| 17:14.46 | x86 | from VoIP Supply? hmm... I've got a dedicated account rep and everything ;) |
| 17:14.56 | [TK]D-Fender | adeel: read up : |
| 17:14.58 | [TK]D-Fender | ~sipnat |
| 17:14.59 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:15.00 | SteveTotaro | yeah, i know cory well |
| 17:15.06 | vap0rtranz | should you have to preload the zap module? i'm getting a 'no such command' from the cli for anything "zap" even though zt* tools work and had a chzap_zap error (that i fixed) |
| 17:15.13 | supers | i just recently upgraded to asterisk 1.4, i'm having an issue with one of my ATA's. when a 2nd person calls it, it automatically conferences both calls together, any idea? |
| 17:15.20 | x86 | SteveTotaro: Cory's cool... I'm dealing with Arthur Miller... great guy |
| 17:15.22 | adeel | [TK]D-Fender, yeah i just read Qwell's replay |
| 17:15.27 | adeel | s/replay/reply/ |
| 17:15.43 | adeel | that's an interesting feature |
| 17:16.03 | x86 | SteveTotaro: last year we spent about $100,000 with VoIP Supply, easy |
| 17:16.10 | x86 | so they like us :P |
| 17:16.29 | SteveTotaro | i spend a good deal with them too |
| 17:16.48 | vap0rtranz | or has the zap command been sucked into something weird like core ... |
| 17:16.53 | SteveTotaro | and very personal as well as b2b with them |
| 17:17.09 | x86 | vap0rtranz: no and no |
| 17:17.14 | SteveTotaro | but i could no get an RMA and i have heard that multiple times |
| 17:17.20 | x86 | vap0rtranz: if you have no zap command, chan_zap is not loaded |
| 17:17.21 | SteveTotaro | maybe they are better now |
| 17:17.35 | x86 | SteveTotaro: i've never had a problem doing an RMA with them... ever |
| 17:17.38 | SteveTotaro | module load chan_zap |
| 17:17.45 | vap0rtranz | x86: and it was suppose to have been autoloaded, right? so there's an error somewhere |
| 17:17.45 | x86 | load chan_zap.so |
| 17:17.55 | SteveTotaro | vi /var/log/messages |
| 17:17.58 | x86 | vap0rtranz: good detective skills ;) |
| 17:18.18 | x86 | vap0rtranz: turn up full logging in logger.conf, restart asterisk, tail -f /var/log/asterisk/full |
| 17:18.30 | x86 | vap0rtranz: might want debug too |
| 17:18.30 | vap0rtranz | x86: ty. i had "signaling" instead of "signalling". totally bombed out the zap stuff |
| 17:18.36 | SteveTotaro | ztcfg -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 17:18.41 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 17:18.47 | x86 | vap0rtranz: that'll do it :) |
| 17:18.53 | outtolunc | more v's move v's you need to reach the moon |
| 17:18.57 | SteveTotaro | but what is the correct spelling |
| 17:19.11 | vap0rtranz | SteveTotaro: hehe |
| 17:19.16 | x86 | vap0rtranz: i think you mean the other way around... pretty sure it only works with a single l |
| 17:19.21 | SteveTotaro | linux tells me signaling is correct |
| 17:19.33 | x86 | SteveTotaro: aspell? |
| 17:19.40 | SteveTotaro | no, two ll s |
| 17:19.49 | SteveTotaro | signalling in zaptel |
| 17:19.53 | vap0rtranz | SteveTotaro: correct |
| 17:19.54 | x86 | nope |
| 17:19.59 | x86 | needs only one l |
| 17:20.03 | SteveTotaro | no, i am correct |
| 17:20.03 | x86 | just checked my confs |
| 17:20.05 | vap0rtranz | x86: well two l's got it working |
| 17:20.09 | vap0rtranz | so me happy |
| 17:20.13 | Qwell | both work :p |
| 17:20.25 | x86 | :P |
| 17:20.30 | x86 | SteveTotaro: hah |
| 17:20.30 | SteveTotaro | qwell, fix the white space issues |
| 17:20.35 | Qwell | what issues? |
| 17:20.37 | vap0rtranz | ERROR[1794] chan_zap.c: Signalling must be specified before any channels are. |
| 17:20.44 | vap0rtranz | *cough* |
| 17:21.01 | x86 | vap0rtranz: pastebin zaptel.conf and zapata.conf |
| 17:21.05 | SteveTotaro | a little white space at the end of a line and zaptel bombs |
| 17:21.12 | Qwell | don't do that then |
| 17:21.25 | Qwell | and, I kinda doubt that would break anything |
| 17:21.31 | SteveTotaro | it does |
| 17:21.36 | Qwell | show me |
| 17:21.45 | SteveTotaro | what, white space? |
| 17:21.52 | SteveTotaro | or signaling? |
| 17:21.59 | SteveTotaro | both break stuff |
| 17:21.59 | vap0rtranz | x86: you don't believe that it's now working? *gasp* the audacity! |
| 17:22.41 | SteveTotaro | i also had an odd bug that took me a while to figure out that bombed asterisk |
| 17:22.54 | SteveTotaro | channel 1,2,4,5 |
| 17:23.07 | SteveTotaro | no, it had to be channel 1-2,4-5 |
| 17:23.16 | vap0rtranz | SteveTotaro: weird |
| 17:23.43 | SteveTotaro | asterisk has taken my troubleshooting skillz to a level of zen |
| 17:24.12 | vap0rtranz | SteveTotaro: so it flows now ... or is that tao. :) |
| 17:24.43 | SteveTotaro | i mix and match things i want to hear and drone out the other crap |
| 17:24.47 | SteveTotaro | ;) |
| 17:26.09 | SteveTotaro | in the channel bug defense of asterisk, this was on bristuffed |
| 17:30.51 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 17:31.28 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 17:32.43 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 17:35.21 | *** join/#asterisk gego (n=gego@host-091-097-121-244.ewe-ip-backbone.de) |
| 17:36.52 | lnx | anyone who get ${DIALSTATUS} succesfully with a perl AGI script, check http://pastebin.com/d42733e02 please :) |
| 17:38.16 | lnx | @ line 42 the var seems empty :7 |
| 17:39.44 | gego | hi everybody - could anyone give me a hint how to determine which (SIP) device accepted a multidial call so that I can trace it with GROUP() ? |
| 17:39.45 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 17:39.56 | stansmith | why are you making private variables outside a block liek that lnx |
| 17:40.46 | ZPertee | what os are you all running for asterisk? |
| 17:41.04 | stansmith | lnx plus there is some discrepancy about DIALSTATUS |
| 17:41.26 | ZPertee | asterisknow screwed me over and now I am trying to migrate to something else |
| 17:41.32 | *** part/#asterisk agx (n=AGX@88.34.216.63) |
| 17:41.43 | alrs | ZPertee: ran off with your girl? |
| 17:41.47 | *** join/#asterisk angom (n=angom@200.79.141.128.dsl.dyn.telnor.net) |
| 17:41.57 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@76-204-200-226.lightspeed.hstntx.sbcglobal.net) |
| 17:42.00 | alrs | ZPertee: "borrowed" your credit card? |
| 17:42.12 | [TK]D-Fender | ZPertee: Whatever OS you feel most comfortable maintaining. |
| 17:42.19 | lnx | stansmith: what kind of discrepancy? |
| 17:42.43 | ZPertee | all of the above. It doesn't like my tdm880B for some reason and Technical support was of no help |
| 17:42.50 | lnx | stansmith: please help me to solve it. |
| 17:44.40 | stansmith | im still chatting with my buddy ccesario 1 sec |
| 17:44.59 | lnx | stansmith: ok, thank you |
| 17:45.58 | flush | ahoy |
| 17:46.09 | flush | i just installed asterisk and my tdm400p card on ubuntu feisty |
| 17:46.19 | flush | any good place to have a good how to on how to place a call now |
| 17:48.10 | [TK]D-Fender | flush: ... |
| 17:48.12 | [TK]D-Fender | ~book |
| 17:48.13 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:48.14 | [TK]D-Fender | ^^^^^^^^^^ |
| 17:48.55 | [TK]D-Fender | ZPertee: And your description and backup provided (none, and none respectively) explain much. |
| 17:53.04 | flush | hrmm |
| 17:53.28 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 17:53.40 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 17:53.41 | ice_croft | hi all |
| 17:53.55 | ice_croft | need some help with 2* sip interconn |
| 17:53.55 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 17:54.15 | ice_croft | regs r well |
| 17:54.23 | ice_croft | some trouble with dialplab |
| 17:54.25 | ice_croft | some trouble with dialplan |
| 17:54.36 | ice_croft | anybody help |
| 17:54.45 | flush | hey |
| 17:55.02 | flush | am i supposed to have dial tone when i have my phone plugged in my tdm400p and the fxo module in the wall jack |
| 17:55.17 | flush | does it do like a 56k modem or im not sure.. |
| 17:56.23 | lesouvage | ice_croft: what is the problem? |
| 17:57.01 | ice_croft | lesouvage> wait a min for pastebin |
| 17:57.09 | lesouvage | ok |
| 17:57.23 | ice_croft | lesouvage> http://pastebin.ca/927956 |
| 17:58.07 | ice_croft | lesouvage> http://pastebin.ca/927958 |
| 17:58.28 | ice_croft | lesouvage> 912 is an phone on remote * |
| 17:59.01 | ice_croft | lesouvage> what did i do wrong? |
| 17:59.41 | *** join/#asterisk drmessano-LT (n=nonya@207.230.140.240) |
| 17:59.43 | ice_croft | lesouvage> sorry, it must be 212, not 912 |
| 18:00.11 | ice_croft | lesouvage> 212 same error |
| 18:00.13 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:02.42 | ice_croft | ~book |
| 18:02.43 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 18:03.45 | ice_croft | lesouvage> any issues? plz |
| 18:07.37 | lesouvage | ice_croft: you are missing a proper extension to handle the call. something like extend => 912,1,dial(<trunk>/912,40). I assume it is an outbound number |
| 18:07.50 | lesouvage | extra |
| 18:08.17 | lesouvage | extend=exten |
| 18:08.19 | ice_croft | lesouvage> maybe |
| 18:08.44 | ice_croft | lesouvage> so, '2XX' => 1. NoOp() [pbx_config] |
| 18:08.44 | ice_croft | <PROTECTED> |
| 18:09.00 | ice_croft | lesouvage> isnt the thin u sayin? |
| 18:09.03 | stansmith | this is crazy! |
| 18:09.28 | stansmith | JenniferAkemi: are you a developer of some kind or do you play around with asterisk for fun n giggles? |
| 18:10.30 | infinity3 | anyone know about the polycom LDAP suppot in v3 firmware? how to enable/license it etc? |
| 18:11.07 | ice_croft | lesouvage> i have this: |
| 18:11.07 | ice_croft | [remote] |
| 18:11.08 | ice_croft | exten => 2XX,1,NoOp() |
| 18:11.08 | ice_croft | exten => 2XX,n,Dial(SIP/sr/${EXTEN}, 30) |
| 18:11.08 | ice_croft | exten => 2XX,n,Hangup() |
| 18:11.19 | lesouvage | ice_croft: when you numbermatching it should look like this _9XX |
| 18:11.28 | ice_croft | oh |
| 18:12.08 | [TK]D-Fender | getting warmer... |
| 18:12.44 | ice_croft | yes. :)) now i have "frobidden" response. that's cool! thanx a lot |
| 18:12.51 | ice_croft | goona dig more |
| 18:12.53 | ice_croft | thanx |
| 18:13.05 | lesouvage | ice_croft: and when it is outbound you should use a trunk. just SIP is for internal numbers. |
| 18:13.17 | [TK]D-Fender | ... |
| 18:13.31 | lesouvage | for internal sip numbers |
| 18:13.48 | [TK]D-Fender | lesouvage: What on earth are you trying to say? |
| 18:14.23 | ice_croft | lesouvage> it kinda internal, really |
| 18:14.46 | ice_croft | lesouvage> inside my company :) inside ip-intranetwork |
| 18:15.17 | ice_croft | lesouvage> so sip should be enough |
| 18:16.15 | lesouvage | fender: he paste a dial line with just SIP and without the account info. |
| 18:17.03 | ice_croft | lesouvage> it's all about prefixes. 2 is the main office. |
| 18:17.18 | infinity3 | anyone know where i can download the polycom v3 firmware? |
| 18:17.18 | ice_croft | lesouvage> 2xx |
| 18:17.30 | lesouvage | fender: does it make sense now? |
| 18:19.27 | *** join/#asterisk Greek-Boy (n=email@41.221.58.4) |
| 18:19.58 | ice_croft | lesouvage> look more, plz |
| 18:20.05 | ice_croft | lesouvage> http://pastebin.ca/927985 |
| 18:20.15 | ice_croft | lesouvage> cant get it, really |
| 18:23.00 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 18:23.03 | teknoprep | hey all |
| 18:23.19 | teknoprep | hey with this ip 650 Backlit expansion module... is there a way to configure it through a TFTP boot server ? |
| 18:23.28 | teknoprep | or do i have to assign each key manually on the device itself ? |
| 18:24.43 | *** join/#asterisk Tako-san (n=jmkiffia@24.108.192.144) |
| 18:24.57 | ice_croft | [TK]D-Fender> check it through, please |
| 18:25.53 | gego | ice_croft> how about _2XX ? |
| 18:26.16 | fiXXXerMet | What is a good solution for recording conference cdr/billing information? The cdr.conf file is logging 3 entries for each call and that seems unnecessary |
| 18:26.37 | ice_croft | gego> excuse me? |
| 18:26.49 | ice_croft | gego> i did _2xx |
| 18:27.03 | ice_croft | gego> now i have "forbidden" error |
| 18:27.26 | [TK]D-Fender | ice_croft: and no sip debug or anything for us to see in there. Nice. |
| 18:28.08 | *** join/#asterisk Tako-san (n=jmkiffia@24.108.192.144) |
| 18:28.30 | *** join/#asterisk dlynes (n=dlynes@mail.247communications.com) |
| 18:28.42 | gego | I just thought that you needed the underscore for matching the XX. |
| 18:28.45 | teknoprep | [TK]D-Fender, you really like polycom.. you know anything about the ip 650 BEM ? |
| 18:28.51 | ice_croft | wait a min |
| 18:29.10 | *** join/#asterisk Hadi- (n=AlanS@cmr-208-124-195-107.cr.net.cable.rogers.com) |
| 18:29.13 | Hadi- | hello everyone |
| 18:29.26 | Hadi- | I have a major issue and I hope someone can help :) |
| 18:29.27 | gego | ice_croft> "forbidden?" - I do |
| 18:29.32 | ice_croft | [TK]D-Fender> http://pastebin.ca/927990 here is sip debug of the event |
| 18:29.46 | Hadi- | I'm using cisco 7950G phones with Asterisk - g729a codec |
| 18:29.55 | Hadi- | every once in a while.. I lose audio |
| 18:29.58 | Hadi- | right when i do that |
| 18:30.04 | Hadi- | I see the following on the asterisk CLI |
| 18:30.20 | Hadi- | 2008-03-04 13:27:36 NOTICE[16225]: rtp.c:415 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 209.47.1.157 |
| 18:30.26 | ice_croft | gego> please look at pastebin |
| 18:30.33 | Hadi- | I disable VAD in the IP phones but still no help |
| 18:30.44 | ice_croft | [TK]D-Fender> any directions? |
| 18:31.03 | ice_croft | "frobidden" whoa |
| 18:33.30 | Hadi- | anyone? :) |
| 18:35.10 | *** join/#asterisk mattchis (n=IceChat7@adsl-75-53-212-167.dsl.hstntx.sbcglobal.net) |
| 18:35.16 | ice_croft | oh, ppl |
| 18:35.24 | ice_croft | it's because of other side |
| 18:35.26 | ice_croft | bgg |
| 18:40.34 | ice_croft | [TK]D-Fender> other side's log : http://pastebin.ca/928000 |
| 18:43.12 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 18:43.31 | Yourname`` | Any major differences betweeb AgentCallbackLogin and AgentLogin? |
| 18:45.03 | SteveTotaro | agentlogin is what i like to use |
| 18:45.25 | SteveTotaro | less wasted time |
| 18:45.30 | SteveTotaro | and abandons |
| 18:46.19 | Yourname`` | I currently use AddQueueMember |
| 18:49.18 | mattchis | Does anyone know if there is a way for a person on hold in the queue to be able to drop from the queue and leave a voicemail? |
| 18:49.19 | Yourname`` | Hmm, I wonder how I can use AgentLogin to addqueuemember.. |
| 18:51.05 | jameswf | nice no mention in the sip RFC about paging so its total anarchy |
| 18:52.02 | SteveTotaro | http://forums.whirlpool.net.au/forum-replies-archive.cfm/658362.html |
| 18:52.07 | ice_croft | ~book |
| 18:52.07 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 18:52.31 | SteveTotaro | is for trixbox but tells you how to drop from a queue and go to exten |
| 18:53.12 | mattchis | SteveTotaro: Thanks I will take a look at that. |
| 18:54.14 | Yourname`` | SteveTotaro: I won't be able to use dynamic features to use AgentLogin, would I? I mean I will _have_ to add those agents as static agents on agents.conf, etc? |
| 18:54.16 | ice_croft | i'm fuckin best, that's for sure!!!!!! don't ban me for this |
| 18:54.44 | Yourname`` | ice_croft: No. [TK]D-Fender is THE. |
| 18:55.06 | ice_croft | Yourname``> no problem. i just fixed the_thing!!!!! |
| 18:55.16 | SteveTotaro | i am not sure why you wouldn't |
| 18:55.18 | ice_croft | Yourname``> im fuckin bruce willis |
| 18:55.31 | SteveTotaro | you are a kid |
| 18:55.44 | ice_croft | bgg, not really |
| 18:56.58 | JunK-Y | mattchis: use the context when defining the queue. |
| 18:57.27 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 18:57.57 | Yourname`` | SteveTotaro: How can I use AgentLogin to add agents dynamically like AddQueueMember? I'm just trying to avoid having to do agents.conf |
| 18:58.11 | mattchis | JunK-Y: Crap that looks to be my problem. LOL Thanks! I missed that |
| 18:58.20 | Yourname`` | !seen fujin |
| 18:59.07 | JunK-Y | Yourname``: you cant have agentlogin doing addqueuemember, theyre 2 different apps. |
| 18:59.26 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 18:59.58 | drmessano-LT | Trixbox user |
| 19:00.00 | Yourname`` | JunK-Y: I know.. but AddQueueMEmber was dynamic for me, so I didn't need to add agents to agents.conf/queues.conf. (Other than having to make the queues itself in queues.conf). I'm looking to get the functionality of AgentLogin with the dynamic ability of AddQueueMember. |
| 19:00.29 | SteveTotaro | just make a huge range in agents.conf |
| 19:00.37 | JunK-Y | adm has nothing to do with agents.conf |
| 19:01.03 | Yourname`` | JunK-Y: Are you even reading what I'm saying? Please read above to follow what I'm saying. |
| 19:01.06 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:01.40 | Yourname`` | SteveTotaro: I guess that's what I'll have to end up doing as there seems to be no way to do it dynamically. |
| 19:01.47 | Yourname`` | Atleast not one that I know of.. |
| 19:02.11 | juanjoc | file: Sorry to bother you, I've been experiencing the same problem that was reported on ticket 11491. I was wondering if there was any way I could help you fix this problem. |
| 19:03.23 | file | it's already fixed. |
| 19:03.40 | file | patch went into SVN earlier |
| 19:03.40 | flush | hrm |
| 19:03.48 | flush | how do i run commands in the asterisk console |
| 19:04.00 | flush | how can i enter console |
| 19:04.11 | fiXXXerMet | asterisk -r |
| 19:06.38 | juanjoc | file: Are you referring to the patch that fixed ticket #10355? |
| 19:07.03 | juanjoc | file: I tried that patch a few days ago and it wasn't solving this problem |
| 19:07.15 | file | I changed it slightly. |
| 19:08.26 | juanjoc | file: So, the patch currently present on #10355 should fix it? |
| 19:08.26 | file | if it still doesn't, then open a new issue |
| 19:08.41 | file | no, the changes I made in SVN should fix it |
| 19:08.46 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 19:09.11 | juanjoc | file: Is that commit safe to backport to 1.4.18 or should I use the 1.4 branch to test? |
| 19:09.43 | file | it should be safe, don't know whether it will cleanly apply. |
| 19:09.58 | file | 105674 and 105677 |
| 19:10.07 | file | er I mean 105676 |
| 19:16.05 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 19:18.43 | jer | if i've got a voicemail context, and i've got a: s,1,GotoIf($[${EXTEN} = 1234]?hangup) and then later on, i've got s,n(hangup),Hangup ... and the call isn't hanging up, any suggestions on what i might be doing wrong? |
| 19:19.37 | tzanger | http://www.cubis.ca/thumbs/192.jpg |
| 19:19.38 | tzanger | hahahahaha |
| 19:19.49 | jer | or is it simply enough to just nuke them from the voicemail users table? |
| 19:20.13 | x86 | jer: nuke them from the voicemail table, or put them in their own context where the voicemail app is never even called |
| 19:20.33 | jer | the former seems to be easier =] |
| 19:20.35 | jer | thanks |
| 19:20.36 | x86 | jer: I usually put voicemail users in one context, and non-voicemail users in their own context |
| 19:20.53 | x86 | not difficult either way |
| 19:21.01 | stansmith | that was mad crazy |
| 19:22.01 | jer | x86, right; i want to keep the work to this particular system to a minimum, the more problems that go wrong with it (not major problems obviously, but enough to cost the company more in labour), the more they'll consider upgrading (stuck at 1.2 and refusing to upgrade) |
| 19:22.08 | *** join/#asterisk willianmazzardo (n=willianm@201-41-29-25.smace701.dsl.brasiltelecom.net.br) |
| 19:22.31 | flujan | hi all. |
| 19:22.54 | ice_croft | does * connect to freepbx or trixbox? via modules? |
| 19:22.57 | stansmith | ~hi flujan |
| 19:22.58 | jbot | Many greetings, flujan, most strange traveller, to this IRCdom of plenty. |
| 19:24.49 | Jason99 | Is there a way to disable the ability to do 3-way calling on a SIP channel? |
| 19:26.20 | *** join/#asterisk ^scott^ (n=scott@stthom.org) |
| 19:26.53 | ^scott^ | Hi I'm trying to create an AGI script to do automated testing of a phone script. Is there a way to get Asterisk (via AGI I guess) to send touchtones? |
| 19:26.55 | flujan | I am having a issue with some sip peers... Asterisk is marking them as in Use. I disconnect the peer, reconnect it and it still appear as In Use. |
| 19:27.32 | GBR_ | if i call the trunk in extensions, the ring is send, but calling a2biling, dont send the ring to user!! |
| 19:28.50 | *** join/#asterisk cowmix (n=cowmix@204.235.245.20) |
| 19:30.29 | flujan | Let me explain it better... I have sip peers, one sip peers stop receiving calls from the queue. |
| 19:30.36 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 19:30.53 | flujan | I close and open the softphone, reboot the machine and when the sip register again. The queue shows it In Use. |
| 19:32.08 | flujan | I checked the AMI and the ExtensionStatus event, is return the code 1 for the Status. |
| 19:33.38 | *** join/#asterisk angryuser (n=nononon@df01t2-213-44-89-224.d4.club-internet.fr) |
| 19:33.51 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:34.13 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:35.30 | *** join/#asterisk asteriskmonkey (n=asterisk@69.77.169.14) |
| 19:36.19 | asteriskmonkey | has anyone used the asterisk manager api extensivly, im trying to track the status of calls and im not finding much info on the wikis |
| 19:36.29 | *** join/#asterisk Guggemand (i=Guggeman@80.198.131.46) |
| 19:39.28 | willianmazzardo | Hi all ... |
| 19:40.15 | Guggemand | can i somehow use my g729 licensed asterisk to convert some pcm files to g729 ? |
| 19:40.20 | teknoprep | [TK]D-Fender, you there ? |
| 19:41.28 | [TK]D-Fender | teknoprep: intermittently |
| 19:42.11 | *** join/#asterisk robmac67 (n=robmacle@82-35-181-109.cable.ubr03.croy.blueyonder.co.uk) |
| 19:45.29 | teknoprep | <tk: yo |
| 19:45.37 | teknoprep | <tk test |
| 19:45.50 | Yourname`` | Hi. Is there a way I can use a different musicclass for AgentLogin hold music? |
| 19:47.03 | *** join/#asterisk Rudolf (n=rodolfo@189.7.85.214) |
| 19:47.25 | Rudolf | hi there |
| 19:47.46 | Rudolf | i have seen a lot of errors like these: icmp v4 hw csum failure |
| 19:47.59 | Rudolf | with 01:04.0 Ethernet controller: Marvell Technology Group Ltd. 88E8001 Gigabit Ethernet Controller (rev 14) |
| 19:48.12 | Rudolf | on kernel Linux glixvoip.com.br 2.6.9-42.0.10.ELsmp #1 SMP Tue Feb 27 10:11:19 EST 2007 i686 i686 i386 GNU/Linux |
| 19:49.01 | Rudolf | i have a question about sip2sip call because the duration of calling are between 2 or 3 minutos and fall down |
| 19:49.32 | Rudolf | have relation the error on kernel(dmesg) and falling of inter sip callings? |
| 19:50.21 | *** join/#asterisk cleone (i=cleo@41.251.64.185) |
| 19:51.43 | cleone | any one here use iaxcomm? |
| 19:52.27 | x86 | Rudolf: upgrade to a real distro, then upgrade your kernel ;) |
| 19:52.42 | Rudolf | x86: yeah yeah, i know this |
| 19:52.54 | lnx | stansmith: are u there? :) |
| 19:53.01 | x86 | then why are you here asking? :) |
| 19:53.05 | Rudolf | x86: but you think that my think are correct? |
| 19:53.24 | Rudolf | x86: searching for another ideas |
| 19:55.18 | lnx | mmhm |
| 19:55.36 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 19:55.58 | teknoprep | [TK]D-Fender, is there a way to setup a file on a tftp server for the IP 650 Backlit expansion module |
| 19:56.52 | *** join/#asterisk roe_ (n=roe___@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 19:58.40 | *** join/#asterisk atis_home (n=chatzill@193.238.213.215) |
| 20:00.13 | roe_ | has anyone addressed a "click-to-call" option with asterisk and hard sip phones? I see there is a cockatoo project that builds click to call into thunderbird, but uses a softphone |
| 20:02.37 | GBR_ | if i call the trunk in extensions, the ring is send, but calling a2biling, dont send the ring to user!! |
| 20:04.35 | *** join/#asterisk cr0n (n=d@dsl-240-120-202.telkomadsl.co.za) |
| 20:06.08 | JenniferAkemi | I'm trying to eliminate single points of failure in my asterisk setup. I have multiple * boxes and am looking at ways to load balance them, but they are using realtime static and dynamic (for sip) so now i have a single point of failure of the database |
| 20:06.41 | JenniferAkemi | i have two database servers which are replicating, but is there a way that i'm missing to make realtime go to the replicant if the main db fails? |
| 20:06.51 | *** join/#asterisk jdspencer (n=jdspence@12.37.95.91) |
| 20:07.11 | jdspencer | hello there fine * people |
| 20:07.12 | cr0n | hi, i have a dtm400p with 2 fxo modules and installed correctly however, when running a genzaptelconf - it lists things in /etc/zaptel.conf under "span1" as "fxsks=1" and "fxsks=2" - any idea why it would do this? |
| 20:07.21 | jdspencer | Can anyone answer a PRI signalling question? |
| 20:08.17 | [TK]D-Fender | teknoprep: Should need any config, should be a slave to the host phone |
| 20:08.36 | JenniferAkemi | jdspencer: if you ask the question maybe someone can try |
| 20:08.54 | [TK]D-Fender | cr0n: thats fine |
| 20:09.00 | jdspencer | cr0n: I don't think it matters which line they are on, but it is specifying the signalling for each module |
| 20:09.05 | jdspencer | what he said |
| 20:09.14 | cr0n | ah |
| 20:11.02 | jdspencer | Okay, my PRI is connected to a Lucent 5ess switch with ni2 protocol |
| 20:11.17 | jdspencer | Zaptel seems to not know the difference between switchtype and protocol |
| 20:11.33 | jdspencer | AT&T is telling me that they're getting weird unknown commands from * |
| 20:11.43 | jdspencer | and our line keeps failing, they blame this on * |
| 20:11.55 | jdspencer | but it works just fine on other providers with ni2 |
| 20:12.18 | JenniferAkemi | but it works sometimtes? |
| 20:12.18 | jdspencer | Is there some option I'm missing to set switchtype and signalling protocol apart from each other? |
| 20:12.34 | jdspencer | most of the time -- it will go down for 1-5 minutes about twice a day |
| 20:12.53 | jdspencer | we've replaced all of the equipment last week, even the digium cards |
| 20:13.04 | jdspencer | still having the same issue |
| 20:13.08 | teknoprep | [TK]D-Fender, yeah i have to set it up on the phone itself.. pita... and i don't see an admin guide for the expansion module anywhere |
| 20:13.16 | *** join/#asterisk atis_home (n=chatzill@193.238.213.215) |
| 20:13.47 | *** join/#asterisk loompek (n=NoName@noname.rula.net) |
| 20:13.49 | loompek | evening |
| 20:14.01 | *** join/#asterisk angom_w (n=angom@200.79.141.128.dsl.dyn.telnor.net) |
| 20:14.15 | [TK]D-Fender | teknoprep: There is none. It isn't a separate device |
| 20:14.18 | cr0n | [TK]D-Fender: okay, i thought that would have been the problem but clearly not, ive configured my trunk for ZAP/g0 and a dialroute to use ZAP/g0 yet when dialing, its just dead.. no errors, just silence |
| 20:14.27 | teknoprep | [TK]D-Fender, thats rough |
| 20:14.28 | [TK]D-Fender | teknoprep: the phone controls it. contacts simply spill over |
| 20:15.09 | [TK]D-Fender | cr0n: Perhaps you're plugged into the wrong port on the card (the order may not be what you expect), or you may have another config error somewhere else |
| 20:16.25 | cr0n | [TK]D-Fender: this is a brand new install, and i have two lines plugged into each of the ports so no matter which one, it has somewhere to dial through |
| 20:17.31 | [TK]D-Fender | cr0n: the card has 4 ports.... like I said I'm not sure you can trust the order you assume them to be in, and for dialing out you could have a misconfigured group. |
| 20:17.43 | [TK]D-Fender | cr0n: then again, Tixbox is NOT supported here. |
| 20:18.10 | cr0n | [TK]D-Fender: def plugged into the right ones. okay.. |
| 20:21.04 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 20:23.38 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 20:24.00 | jdspencer | does libpri know the difference between switchtype and switch protocol? |
| 20:25.44 | *** join/#asterisk Bock (n=bock@dslb-084-057-002-183.pools.arcor-ip.net) |
| 20:26.18 | JenniferAkemi | i've only ever configured one jdspencer |
| 20:26.27 | JenniferAkemi | i've only ever configured one on my Harris switch too |
| 20:26.41 | JenniferAkemi | i would just go with the 5ess thing |
| 20:27.04 | jdspencer | it changes the protocol to ATT Custom... which won't work for us |
| 20:27.29 | stansmith | im just the first of my litter |
| 20:27.40 | Bock | Hello everyone, I am just making my first steps with asterisk and got 2 different sip-out providers working so far. I also configured that for some numbers one is used instead of the other. Now, how can I tell asterisk to use sip-provider-a for all unknown numbers? |
| 20:28.02 | jdspencer | bock: how are you defining "known" numbers? |
| 20:28.11 | JenniferAkemi | probably ni2 would be what you want. |
| 20:28.22 | JenniferAkemi | dunno why it is going down sometimes though sorry |
| 20:28.29 | jdspencer | that's what i'm thinking... it's worked perfectly for 3 years |
| 20:28.34 | Bock | jdspencer: the "known" ones are the ones, that already get routed through one of my sip providers |
| 20:29.02 | jdspencer | jennifer: all the sudden it got stupid -- i'm thinking a bad CSU/DSU unit possibly |
| 20:29.08 | jdspencer | jennifer: thanks for the help |
| 20:29.57 | jdspencer | bock: if i understand your question correctly, you could write a catch-all matching rule to get any of the numbers that aren't matched in your dialplan |
| 20:30.00 | Bock | jdspencer: there is a special case in germany where 01[5-7]x are mobile phone numbers, 0180, 0137x etc are different service numbers and 0[2-9] are local numbers |
| 20:30.04 | jdspencer | bock: something like _. => |
| 20:30.09 | Bock | jdspencer, yes, exactly that |
| 20:30.37 | jdspencer | bock: does that answer the question, or that IS the question? |
| 20:30.51 | Bock | jdspencer: the question is, how to do that |
| 20:31.00 | Bock | ah, is it ._? |
| 20:31.03 | jdspencer | bock: do you use AEL or extensions.conf |
| 20:31.09 | Bock | extensions.conf |
| 20:31.11 | loompek | is it possible for asterisk's cdr to write cdrs every 10 minutes in case a call is > 10 minutes? |
| 20:31.48 | jdspencer | bock: yes, _. => will get you a matchall -- more info here -- http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 20:32.35 | [TK]D-Fender | loompek: No. CRD is once a call is finished. |
| 20:32.56 | [TK]D-Fender | CDR* |
| 20:33.09 | Bock | jdspencer, so the extensions.conf will be read until a rule matches? so if I understand it right, adding this one at the very end of the section will catch all calls where no previous rule was applied? |
| 20:33.46 | loompek | [TK]D-Fender what in case i'd like to implement a pre-pay system? |
| 20:33.48 | jdspencer | bock: correct -- actually, i'm not sure it matters what order you apply them... asterisk will sort them internally if i understand it right |
| 20:34.08 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 20:34.20 | [TK]D-Fender | loompek: When you call dial, place a max limit based on the time remaining. |
| 20:34.48 | loompek | [TK]D-Fender you can do that? |
| 20:34.59 | [TK]D-Fender | jdspencer: Go read the book on the part about "extension sorting" |
| 20:35.23 | jdspencer | D-Fender: did i life to bock? |
| 20:35.27 | jdspencer | * lie |
| 20:36.04 | jdspencer | quoting voip-info --> "Because you may use patterns to define extensions, more than one extension pattern could match a given telephone number. Asterisk does not match against the extension patterns in the order you define them; the extension patterns are sorted first. Hence Asterisk may process a telephone number differently than you intended. " |
| 20:37.42 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 20:38.13 | [TK]D-Fender | Guess that says it all |
| 20:38.19 | Bock | jdspencer: 'exten => _.,1,Dial(SIP/0049${EXTEN:1}@dus.net_out,45,rtT)' now overrides all ohter rules... thats not what I wanted :/ |
| 20:38.23 | asteriskmonkey | how do you relate an event to an action with the asterisk manager api???? |
| 20:38.31 | Yourname`` | Hi. There's tons of voicemails in a mailbox, to delete them all.. is it as easy as going to the mbox dir and deleting all of those voicemails on command? |
| 20:39.04 | [TK]D-Fender | Yourname``: Yup |
| 20:39.17 | asteriskmonkey | yep |
| 20:39.20 | asteriskmonkey | rm -rf / |
| 20:39.28 | [TK]D-Fender | asteriskmonkey: BASTARD |
| 20:39.46 | cmantito | haha |
| 20:39.56 | Yourname`` | Also, [TK]D-Fender a whole queue is logged in, and there are agents logge dinto the queue, and still calls coming into the queue are going to the voicemail and I dont understand why as it always worked properly that calls will go to agents in the queue if they are logged in, and go to voicemail if they are not logged in. |
| 20:40.04 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 20:40.40 | *** join/#asterisk DJF5 (n=DJF5@84-105-201-37.cable.quicknet.nl) |
| 20:41.29 | cmantito | Bock: if you want a rule that will be matched failing all other rules |
| 20:41.37 | cmantito | my suggestion is to put it in it's own context |
| 20:41.42 | cmantito | and then include that context into the first |
| 20:41.48 | cmantito | since included contexts usually sort last |
| 20:42.15 | [TK]D-Fender | Yourname``: Queue's should no use agent VM's |
| 20:42.17 | [TK]D-Fender | not* |
| 20:42.44 | Bock | cmantito: thank you... I just read that on the website, but it feels good that it is confirmed :D |
| 20:43.07 | Yourname`` | I know, but it's a voicemail for the queue itself. |
| 20:43.20 | Yourname`` | :s |
| 20:43.24 | cmantito | Bock: example: http://pastebin.com/d252bd265 |
| 20:43.29 | Yourname`` | exten=> 200,n,Queue(frontier,tT,,,30,) |
| 20:43.31 | Yourname`` | exten=> 200,n,Voicemail(200,u) |
| 20:43.52 | [TK]D-Fender | Yourname``: then that is not an agent VM |
| 20:43.54 | Yourname`` | See,, |
| 20:43.59 | Yourname`` | I know, I guess I worded it wrong sorry |
| 20:44.03 | [TK]D-Fender | Yourname``: that implied it got there due to dialing an agent |
| 20:45.26 | jdspencer | bock: my apologies for being blind and leading you :) |
| 20:45.43 | Bock | jdspencer: nevermind... this way you didn't spoonfeed me :) |
| 20:45.47 | jdspencer | bock: i had used matching in that way with success, but in a limited context |
| 20:45.57 | cmantito | matching is a pain in my butt ;) |
| 20:46.22 | Bock | cmantito: I got it working now, I added a section [everything-else] with the _. Rule and uncluded it to the default set. |
| 20:46.33 | jdspencer | hooray! |
| 20:46.43 | cmantito | Bock: my suggestion is to avoid _. because while it's working, it may end up acting...weird |
| 20:46.54 | jdspencer | bock: Fender was right about taking a look at the book |
| 20:47.12 | cmantito | you can use _X! to match any string that is all numbers, and _*! to match any string starting with a star (*) |
| 20:47.25 | Yourname`` | [TK]D-Fender: Yeah, but in queues.conf I did joinempty=strict and leavewhenempty=strict |
| 20:47.29 | [TK]D-Fender | "_.,1" should only be used to Goto another fixed exten in another context immediately to avoid those hang-ups |
| 20:47.36 | jdspencer | bock: the section on pattern matching would likely be helpful, i just gave it a quick scan |
| 20:47.40 | Yourname`` | [TK]D-Fender: And it's still going to voicemail even though the queue is not empty. |
| 20:48.43 | [TK]D-Fender | Yourname``: and you've still shown me.... absolutely nothing. |
| 20:50.13 | Yourname`` | lol.. one sec, I did change one thing though, the way agents login.. so maybe that has to do something with it. Here's the pb http://pastebin.ca/928137 |
| 20:50.35 | Bock | can someone try to call 000387234170@voip.dus.net ? I can test from the local phone net, but not a direct voip call |
| 20:51.02 | [TK]D-Fender | Yourname``: Still little of value... |
| 20:51.27 | Yourname`` | [TK]D-Fender: What else would you like to see? |
| 20:51.46 | cmantito | Bock: number not avail |
| 20:52.00 | Bock | cmantito: thank you |
| 20:52.03 | cmantito | no prob |
| 20:52.28 | *** join/#asterisk k3mp (n=k3mp@pD9EBE53D.dip.t-dialin.net) |
| 20:52.38 | k3mp | hi @ all |
| 20:52.54 | vap0rtranz | queues aren't playing well with voicemail here. is there a better way to timeout the hold time to a voicebox? Allison starts talking over herself and it's bad |
| 20:53.46 | Yourname`` | [TK]D-Fender: I think because I added the AgentLogin procedure by adding the agents dynamically using agents.conf and queues.conf, it has done this for every queue. However, when I set queues.conf up by giving it the members, it was under just one queue.. not the other. So I dont understand why it's taking up for every other queue. |
| 20:53.50 | juanjoc | file: The fix for ticket #11491 does not seem to work. I'm trying to capture the RTP packets for a failed call? Should I create a new ticket or reopen #11491? |
| 20:53.53 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 20:54.18 | file | juanjoc: there were two commits, did you use both? |
| 20:54.49 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 20:54.49 | juanjoc | file: I grabbed branch/1.4 as of about 30 min ago |
| 20:55.02 | file | what revision number? |
| 20:55.17 | k3mp | Could someone help me reducing getting congestions when dialing out via voip? |
| 20:55.51 | juanjoc | file: The last commit was r105676 |
| 20:55.57 | juanjoc | file: From you |
| 20:55.58 | vap0rtranz | k3mp: is the channelstatus actually congestion? |
| 20:56.15 | Mavvie | k3mp: isn't that related to the amount of calls you are allowed to setup to a phone or SIP provider? |
| 20:56.18 | file | then grab the info and create a new issue |
| 20:56.55 | juanjoc | file: OK, I'll add the packet capture to the ticket |
| 20:57.11 | k3mp | Mavvie, vap0rtranz: i have a maximum of 30 sip calls simultanously, when i try to make more, they are rejected |
| 20:57.26 | vap0rtranz | Mavvie, k3mp: maybe, but i've seen * just sit there when a series of channels are not inuse. chan inuse 1/0 |
| 20:57.51 | Mavvie | k3mp: I think I found your problem. |
| 20:57.59 | vap0rtranz | Mavvie: hehe |
| 20:58.19 | k3mp | Mavvie, vap0rtranz: i can't stop doing more calls, but i could add a kind of latency? |
| 20:58.44 | vap0rtranz | k3mp: sure. * can wait forever for the other end to pickup |
| 21:00.20 | file | juanjoc: waittttttt, is it Packet2Packet bridging? |
| 21:00.37 | vap0rtranz | what is Packet2Packet? |
| 21:01.12 | juanjoc | file: What do you mean? This is a call from a SIP phone that goes through Asterisk to a PSTN phone over SIP via Level 3 |
| 21:01.28 | file | juanjoc: does it say on the screen "Packet2Packet bridging" |
| 21:01.42 | docelmo | Anyone know of issues with Asterisk 1.4.18 and RFC2833 DTMF? Im having some major issues.. it appears the RTP packet payload is being marked at 0 not 101 |
| 21:01.43 | juanjoc | file: You mean the Asterisk log? |
| 21:01.45 | k3mp | Mavvie: maybe you could explain me how ^^ |
| 21:01.54 | file | juanjoc: yes, on the CLI |
| 21:02.02 | *** join/#asterisk corrupt (i=81074dcb@gateway/web/ajax/mibbit.com/x-d7f2bb23d8c0a02c) |
| 21:02.18 | corrupt | does asterisk support speech recognition? |
| 21:02.30 | docelmo | yes w/ external applications like lumenvox |
| 21:02.42 | corrupt | lumenvox, aye... |
| 21:02.49 | docelmo | But only 1.4 not 1.2 |
| 21:03.02 | juanjoc | file: That string does not appear on the log. I can increase the log level if necessary |
| 21:03.11 | juanjoc | file: FYI, the SSRC did not change |
| 21:03.16 | corrupt | is lumenvox opensource? |
| 21:03.21 | docelmo | haha no |
| 21:03.39 | corrupt | is there any open source speech recognition software out there? |
| 21:03.53 | vap0rtranz | corrupt: hah |
| 21:03.55 | juanjoc | file: What log level is necessary for that message to show? |
| 21:04.09 | [TK]D-Fender | corrupt: CMU Sphinx |
| 21:04.09 | corrupt | what kind of speech recognition software does goog-411 or wachovia bank use? |
| 21:04.14 | file | verbosity of 3 or greater |
| 21:04.56 | vap0rtranz | careful. there's a difference b/w speech recognition and command recognition. command's can get numbers spoken |
| 21:05.50 | corrupt | does asterisk support command recognition as well? |
| 21:09.01 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.98.208) |
| 21:09.12 | *** join/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 21:09.20 | stansmith | o noes what happened to my pgadmin3? |
| 21:09.27 | stansmith | wrong channel, sorry |
| 21:10.27 | *** join/#asterisk edwin_quijada (n=m@25.116.88.200.m.sta.codetel.net.do) |
| 21:10.34 | edwin_quijada | Hi! |
| 21:10.48 | Yourname`` | [TK]D-Fender: Ok, now this is the deal. On queues.conf, under the queue name, I used member Agent/@1 and in agents.conf, I set a few agents under group=1. So when the call comes in to the queue, the calls are sent to the agents in group1. But the calls are going to the queue, and they think no one is logged in and so it goes to the voicemail. |
| 21:10.49 | *** join/#asterisk nirz (n=nir@89-138-84-162.bb.netvision.net.il) |
| 21:11.20 | edwin_quijada | I wanna know if I can receive faxes using asterisk but my line phone never be busy? |
| 21:11.36 | vap0rtranz | good question |
| 21:12.25 | edwin_quijada | I have a customer that wants a fax server receiver but their line is not busy |
| 21:12.35 | vap0rtranz | edwin_quijada: analog line? |
| 21:12.47 | edwin_quijada | vap0rtranz: digita |
| 21:13.27 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 21:14.09 | JerJer | I totally love the header graphic (and the content is actually relevant here, maybe :) http://tinyurl.com/24b8s2 |
| 21:14.10 | vap0rtranz | edwin_quijada: my guess is once * answers the inbound, then the did could be made free for the next fax ... at least as a roll-over |
| 21:14.11 | edwin_quijada | I thougth using HylaFax and Asterisk to redirect by VoIP |
| 21:14.12 | vap0rtranz | *shrug* |
| 21:16.18 | JT | err |
| 21:16.19 | edwin_quijada | vap0rtranz: I dont understand |
| 21:16.21 | JT | if it's PRI |
| 21:16.23 | edwin_quijada | Can u explain? |
| 21:16.30 | JT | there is no concept of "did being made free" |
| 21:17.06 | vap0rtranz | JT: i bastardize language. the "channel"; whatever can be made to not ring busy ... (dare i not say "trunk") |
| 21:18.13 | edwin_quijada | vap0rtranz: But u can do the channe no ring busy? |
| 21:18.32 | JT | vap0rtranz: i don't think you understand how BRI/PRI works |
| 21:18.43 | vap0rtranz | JT: he's all sip; mute |
| 21:19.10 | JT | vap0rtranz: where did he say that? |
| 21:20.53 | vap0rtranz | vap0rtranz: i misread "digita". how would that be done, i'm more curious than answers |
| 21:21.52 | JT | vap0rtranz: calls are setup via Q.931 over the D channel, which has the destination number, source number if not set to private, and the channel number |
| 21:22.05 | JT | the timeslot used for the call is arbitary |
| 21:22.37 | vap0rtranz | JT: how is that setup in *? |
| 21:22.52 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.131.143) |
| 21:22.57 | JT | what do you mean? |
| 21:23.54 | vap0rtranz | JT, edwin_quijada: the question was how a fax line could never ring busy (ideally, with one line); and how * could do that. |
| 21:24.26 | JT | vap0rtranz: forget about the old concept you may have of lines, if it's digital, there is no such thing as a dedicated channel for a DID |
| 21:24.54 | Bock | Thank you for your nice help and support, I'll be back soon :) |
| 21:25.01 | [TK]D-Fender | vap0rtranz: You can't and that idea is on crack. If you don't have free channels yuo can't take calls. |
| 21:25.14 | stansmith | 0wn3d! |
| 21:25.23 | seanbright | if i remove the 'span' line from zaptel.conf, that effectively disables the span, yes? |
| 21:25.37 | edwin_quijada | JT: So there is no way to take the fax |
| 21:25.47 | JT | edwin_quijada: how many channels do you have? |
| 21:26.04 | edwin_quijada | I have 3 lines |
| 21:26.15 | JT | what sort of lines? |
| 21:26.30 | edwin_quijada | digital? |
| 21:26.49 | *** join/#asterisk ccvp (n=ax@66.0.46.210) |
| 21:27.10 | edwin_quijada | So we need a pull modem to receive a few faxes |
| 21:27.34 | vap0rtranz | [TK]D-Fender: i remember reading somewhere (in this heap) that * could take control/release the other end. the subject wasn't faxing but was curious about how fax number could be made to ring almost at all times. maybe only via a callback that releases the original number? and the transmission occurs over another channel? |
| 21:27.37 | JT | edwin_quijada: what, do you have BRI, PRI, multiple BRI or what? |
| 21:27.50 | JT | digital is a class of lines |
| 21:27.53 | JT | not a specific type |
| 21:28.09 | edwin_quijada | I have 3 lines normal connect to asterisk |
| 21:28.11 | [TK]D-Fender | vap0rtranz: the only way to have it never it busy is to have more channels than you have calls. |
| 21:29.00 | JT | edwin_quijada: then they're not digital |
| 21:29.09 | JT | edwin_quijada: if they're "normal" POTS |
| 21:29.13 | JT | that's analogue |
| 21:29.18 | vap0rtranz | [TK]D-Fender: ok. but i wasn't on crack for thinking about call-back via a different channel ... was i? |
| 21:29.40 | JT | edwin_quijada: please work out what type of lines you actually have |
| 21:29.51 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 21:29.53 | [TK]D-Fender | vap0rtranz: The call is going SOMEWHERE. |
| 21:30.04 | *** join/#asterisk Wayhigh (i=noid@www.kevinlynn.com) |
| 21:30.08 | vap0rtranz | :) |
| 21:30.20 | fujin | anyone suggest the preferred method of achieving high-availibity with asterisk? I'm looking at using DNS SRV records and like mysql storage for voicemail etc |
| 21:30.25 | fujin | err, odbc storage** |
| 21:30.57 | JT | multimaster dns is good for dns HA |
| 21:31.04 | edwin_quijada | JT: are analogue |
| 21:31.06 | [TK]D-Fender | ok, BBIAB |
| 21:31.10 | fujin | JT: any docs? |
| 21:32.25 | ccvp | ?msg michelle_kat - well just come over around 9pm, after you goto macaroni grill w/ her, anal & 69 if you want :) |
| 21:32.37 | ccvp | ooooops, wtf |
| 21:32.45 | drmessano-LT | wtf |
| 21:32.48 | JT | edwin_quijada: are you sure about that now? you've said they're digital twice so far |
| 21:32.57 | *** join/#asterisk angryuser (n=nononon@df01t2-213-44-89-224.d4.club-internet.fr) |
| 21:33.27 | edwin_quijada | JT: yes, but i seeing again |
| 21:33.37 | edwin_quijada | there are normal POTS |
| 21:33.43 | ccvp | edwin |
| 21:33.44 | ccvp | ".do" ? |
| 21:33.51 | JT | edwin_quijada: it's good to be sure, otherwise you waste everyone's time giving inappropriate advice |
| 21:34.08 | edwin_quijada | 3 lines from POTS with openvox card |
| 21:34.13 | JT | sending /msgs from anything but the status window is not a smart thing |
| 21:34.31 | JT | edwin_quijada: in that case each did must be dedicated to a channel |
| 21:34.55 | edwin_quijada | ccvp: I think do u mean dominican republic? |
| 21:35.32 | edwin_quijada | JT: nothing to do? |
| 21:35.58 | edwin_quijada | There is any way to do that even change the line or move to another technology |
| 21:36.06 | JT | edwin_quijada: no |
| 21:36.55 | edwin_quijada | so the only way is one channel by line and if it is busy wait for this? |
| 21:37.48 | JT | you can have multiple lines dedicated to fax, and get the telco to put them in a line hunt group |
| 21:37.58 | JT | otherwise there is no other way normally with analogue. |
| 21:38.08 | MatBoy | ah nice, 2 4 port bri cards ordered :) |
| 21:38.11 | vap0rtranz | edwin_quijada: it sounds like the line/channel/trunk will ring busy until end of transmission; but i still think there would have been some call-back feature to use line/channel/trunk other than the one with the fax # |
| 21:38.22 | husimon | mmm my office brewed beer is almost ready to be bottled |
| 21:39.00 | MatBoy | husimon, you found some old stuff there ? |
| 21:39.02 | JT | vap0rtranz: how would that work? |
| 21:39.56 | edwin_quijada | JT: and if I use another lines type> Could be? |
| 21:40.04 | husimon | MatBoy, nope I brewed beer in my office |
| 21:40.12 | stansmith | illegal? |
| 21:40.16 | husimon | hehe |
| 21:40.20 | MatBoy | husimon, my having rotted appleas and so on ? |
| 21:40.22 | JT | edwin_quijada: with digital DIDs are not tied to channels, so as long as you have channels free, you're fine |
| 21:40.27 | MatBoy | stansmith, my cards were legal btw ;) |
| 21:40.31 | stansmith | haha nice |
| 21:40.38 | husimon | MatBoy, rotten apples? |
| 21:40.40 | MatBoy | stansmith, 2x 4pri cards |
| 21:40.52 | JT | pri or bri? |
| 21:40.53 | MatBoy | husimon, you can make alcohol out of rotten fruit :P |
| 21:40.54 | stansmith | where did you get them from? they were inexpensive, no? |
| 21:41.03 | MatBoy | JT, pri |
| 21:41.19 | JT | MatBoy: oh, i thought you said bri before |
| 21:41.40 | MatBoy | JT, I might need them too |
| 21:42.04 | husimon | MatBoy, yeah I just didn't follow, didn't see your earlier conversation if you mentioned apples |
| 21:42.32 | vap0rtranz | JT: there would only be momentary use of the channel with the fax #, just long enough to grab the inbound callerid. setting the callerid of the call-back channel to be the fax #; this all hinges on and the far end accepting the original dialed number and retransmitting. aka, too much work, but faxes are so old i almost think i've heard of this before |
| 21:42.59 | MatBoy | JT, why did you thought bri ? |
| 21:43.45 | JT | MatBoy: because of |
| 21:43.46 | JT | 08:41 < MatBoy> ah nice, 2 4 port bri cards ordered :) |
| 21:43.52 | *** join/#asterisk RoyK (n=roy@ip-216-4-149-91.dialup.ice.no) |
| 21:43.57 | MatBoy | JT, typo |
| 21:43.59 | MatBoy | sorry |
| 21:44.02 | MatBoy | cold hands :) |
| 21:44.03 | edwin_quijada | JT: check thsi scenario |
| 21:44.29 | Yourname`` | When using eyeBeam how can I make sure the external IP is used to foreverything with Asterisk? I hate seeing 192.169.* in the CLI. |
| 21:44.30 | JT | vap0rtranz: most fax machines probably are not setup for that to work |
| 21:44.49 | vap0rtranz | i know :( |
| 21:45.44 | ice_croft | i cant build asterisk_addons from freebsd ports. :( |
| 21:45.57 | ice_croft | http://www.mail-archive.com/freebsd-ports@freebsd.org/msg13100.html |
| 21:45.59 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 21:46.00 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 21:47.43 | stansmith | bakers dozen = 13 ? |
| 21:48.01 | JT | yes |
| 21:48.15 | stansmith | kthx |
| 21:48.32 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:48.34 | vap0rtranz | opinions on moh? as in, better than elevator but nothing fancy? |
| 21:48.50 | stansmith | vap0rtranz: p. diddy |
| 21:48.51 | MatBoy | JT, nice one you saw, no but PRI, PRI is nice :) |
| 21:49.06 | [TK]D-Fender | vap0rtranz, "Slayer" <- Accept no substitutes. |
| 21:49.07 | vap0rtranz | stansmith: hehe. meany |
| 21:49.10 | stansmith | haha |
| 21:49.38 | stansmith | vap0rtranz: im not sure if i am wording this correctly, but you need the license for the MOH |
| 21:49.42 | vap0rtranz | [TK]D-Fender: that new agey ;) |
| 21:50.18 | vap0rtranz | stansmith: depends on the artist's licensing, so i mean CC music |
| 21:50.59 | stansmith | just mentioning it, didnt want to see someone get sued |
| 21:51.06 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 21:52.31 | vap0rtranz | maybe psychedelic will brainwash customer's into forgetting their phone woes |
| 21:52.44 | *** part/#asterisk arooni (n=arooni__@c-24-19-232-203.hsd1.mn.comcast.net) |
| 21:55.12 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:55.56 | budol | hello |
| 21:55.57 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-195-124.tx.res.rr.com) |
| 21:56.25 | edwin_quijada | JT: i have t1 card and we want use it for receiving faxes the 24 lines. We can use asterisk and HylaFax to receive faxes by this way? |
| 21:56.40 | budol | how do I setup fax machine in asterisk? |
| 21:56.52 | stansmith | ~fax |
| 21:56.52 | jbot | Well, apperantly the fax was concieved of by Napoleon Bonaparte. He commissioned a system of devices that could transmit a traced image electrically over telegraph lines to a remote device that would redraw the image identically. |
| 21:57.01 | stansmith | oops |
| 21:57.08 | vap0rtranz | LOL |
| 21:57.21 | budol | aahh |
| 21:57.34 | vap0rtranz | budol: i'm right there with ya. fax 101 |
| 21:57.57 | edwin_quijada | budol: Hylafax and asterisk |
| 21:57.58 | JT | stansmith: yeah, you need to pay royalties for most music |
| 21:58.08 | JT | edwin_quijada: as long as you're using PRI signalling, yes |
| 21:58.10 | stansmith | royalties thats what it is, slipped my mind |
| 21:58.29 | JT | edwin_quijada: if you're using channelised RBS, then you must still dedicate channels to DIDs |
| 21:58.46 | vap0rtranz | JT: "most". i licensed mine with a liberal CC. a few people do also |
| 21:58.53 | budol | hylafax? |
| 21:59.13 | stansmith | 16:45 < edwin_quijada> JT: i have t1 card and we want use it for receiving faxes the 24 lines. We can use asterisk and HylaFax to receive faxes by this way? |
| 21:59.23 | stansmith | um |
| 21:59.43 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.98.208) |
| 22:00.11 | stansmith | sorry bout that |
| 22:00.17 | J4k3 | stansmith: I wouldn't use asterisk for that |
| 22:00.21 | J4k3 | err edwin_quijada: |
| 22:00.42 | J4k3 | personally I'd terminate my CT1 to an old portmaster 3 and use it for faxing |
| 22:00.46 | J4k3 | but, thats just me |
| 22:01.09 | budol | if im sending fax is it right? User => Analog Fax => SIP ATA => Asterisk => TDM Card => PSTN |
| 22:01.10 | edwin_quijada | J4k3: so it is not possible? |
| 22:01.12 | J4k3 | portmaster 3 = $10 to 50 |
| 22:01.22 | J4k3 | edwin_quijada: it may be possible but PCs kinda suck at 'realtime' jobs. |
| 22:01.29 | J4k3 | edwin_quijada: especially running non-rtos's |
| 22:01.29 | MatBoy | man, I can't wait to have my patch in the DC :) |
| 22:01.30 | budol | is it possible in viseversa? |
| 22:01.54 | JT | MatBoy: heh |
| 22:02.03 | MatBoy | JT, or the cards maybe more :) |
| 22:02.09 | JT | MatBoy: i'm waiting for my telco to install 2 * 20 pair cables to my rack |
| 22:02.12 | JT | (at their cost) |
| 22:02.33 | edwin_quijada | so we need a pull modem to do that |
| 22:02.33 | edwin_quijada | ? |
| 22:02.34 | J4k3 | JT: wow, *all* inside wiring in the USA costs money |
| 22:02.46 | *** join/#asterisk keith4 (n=keith@lust.CC.Lehigh.EDU) |
| 22:02.47 | JT | J4k3: same here, usually |
| 22:02.49 | MatBoy | JT, ah, the partches are alreayd in my suite, so that's not the issue... only the part from the carrier to the patchpanel need to be doen and so on |
| 22:02.55 | MatBoy | *done |
| 22:02.56 | ccvp | HAHAHAHHAHAHAHAHAH, someone posted on barak obamas forums |
| 22:02.57 | JT | J4k3: except when you harrass the telco enough |
| 22:03.00 | MatBoy | meetme-room |
| 22:03.02 | keith4 | is there an asterisk integration (click-to-call-esque) for thunderbird? |
| 22:03.05 | J4k3 | JT: I even got to the point of installing a 'test wire' under my carport so the telco would install/test T1s to that point, then I'd move the NIUs into my office. |
| 22:03.05 | ccvp | www.illegalalienreport.com |
| 22:03.06 | MatBoy | JT, nice setup :) |
| 22:03.08 | ccvp | that site is so illegal |
| 22:03.10 | ccvp | lol |
| 22:03.33 | MatBoy | ccvp, their name show the oposite :P |
| 22:03.48 | JT | MatBoy: well that's easy then |
| 22:03.54 | ccvp | that logo at that site |
| 22:03.54 | budol | tnx |
| 22:03.56 | ccvp | is hilarious |
| 22:03.56 | ccvp | lol |
| 22:03.58 | MatBoy | JT, indeed |
| 22:04.04 | ccvp | an alien in a sombrero |
| 22:04.04 | JT | MatBoy: i need cables run vertically up 2 floors, then horizontally 30metres |
| 22:04.05 | ccvp | hahaha |
| 22:04.11 | MatBoy | JT, on what cards do you attach those lines ? |
| 22:04.15 | MatBoy | E1's ? |
| 22:04.17 | JT | MatBoy: sangoma |
| 22:04.18 | JT | yes |
| 22:04.23 | MatBoy | nice |
| 22:04.32 | *** join/#asterisk l2cache (n=l2cache@m685e36d0.tmodns.net) |
| 22:04.35 | J4k3 | ahh, the USA is heading itself into admitting to the economic depression its in the middle of |
| 22:04.51 | *** join/#asterisk skyn3t (n=skyn3t@S0106006097940f68.vw.shawcable.net) |
| 22:04.54 | MatBoy | JT, ah maybe I need them once too... but this is a good start on those cards :) |
| 22:04.57 | J4k3 | and of course, its gotta get a scapegoat, and "terrorist ragheads" aren't working, so we're gonna go to our old 1920s scapegoat, the mexican. |
| 22:05.14 | JT | MatBoy: what do you have? |
| 22:05.27 | MatBoy | 2x 410P |
| 22:05.28 | l2cache | hi guys |
| 22:05.31 | JT | ah ok |
| 22:05.50 | MatBoy | should do the job |
| 22:05.58 | JT | do they have HWEC? |
| 22:06.58 | MatBoy | JT HWEC ? |
| 22:07.07 | MatBoy | HardWare... ? |
| 22:07.14 | JT | echo cancellation |
| 22:07.23 | MatBoy | yap |
| 22:07.25 | *** part/#asterisk IOscanner (n=IOscanne@cpe-76-187-195-124.tx.res.rr.com) |
| 22:07.26 | JT | pretty important |
| 22:07.55 | MatBoy | JT, they are new |
| 22:08.11 | JT | heh |
| 22:09.27 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 22:10.11 | MatBoy | JT, but why, you mean the type of card, or the type of the type I bought |
| 22:10.17 | MatBoy | version of the type |
| 22:12.29 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 22:15.03 | *** join/#asterisk Greek-Boy (n=email@41.221.58.4) |
| 22:15.06 | [hC] | Is there some known bug with asterisk 1.4.14 (?) where sometimes you get 0 byte voicemail files, which when played crashes the call and hangs up on the user? |
| 22:15.14 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:17.02 | fujin | awesome |
| 22:17.07 | fujin | haven't seen that |
| 22:17.24 | generalhan | omg, i cant believe they took the teletubbie-murder sound from the pack :( that was my favorite one |
| 22:18.28 | vap0rtranz | generalhan: hehe. mentioning that, is * really calling mpg123 per the music conf?? |
| 22:19.42 | [hC] | yeah, its weird. The customer will get msg0000.gsm = 0bytes in his 'old' folder, and when he tries to play them, the call drops |
| 22:20.35 | TJNII | [hC]: I had that happen once. Never figured out why. |
| 22:21.10 | *** join/#asterisk hi365_m (n=hi365@213.151.57.96) |
| 22:21.21 | JT | MatBoy: the TE410P has no HWEC btw |
| 22:22.17 | Yourname`` | fujin! |
| 22:23.04 | MatBoy | JT, doesn't matter for this test at the moment btw, and in price for this test it doesn't matter at all... |
| 22:23.11 | MatBoy | 410P |
| 22:23.15 | MatBoy | damn |
| 22:23.24 | generalhan | boo, im lightly upset ... there are only like 4 "funny" sound files ... in 1.2 there were TONS ! |
| 22:23.35 | generalhan | im gonna have to transfer them over from my old machine ! lol |
| 22:24.03 | hi365_m | anyone have any info wrt setting the mwi on cellphones when you have new voice mail (on the * server)? |
| 22:25.11 | MatBoy | JT, and you can upgrade them |
| 22:27.51 | husimon | is the default user to linksys pap2t : user? |
| 22:28.04 | MatBoy | JT, and it's not said that the EC will perform that good on those cards, I see people who have the license and complain... so that's what I'm going to test |
| 22:28.10 | MatBoy | so actually they (can) have it |
| 22:28.57 | MatBoy | JT, HPEC btw if you ask me |
| 22:29.36 | MatBoy | I wonder why people get better results for Software EC sometimes |
| 22:30.31 | husimon | does anyone know how the speed dial stuff with atas works? |
| 22:30.35 | husimon | the linksys pap2t |
| 22:30.44 | husimon | how do you dial them |
| 22:31.24 | *** join/#asterisk RoyK (n=roy@ip-216-4-149-91.dialup.ice.no) |
| 22:31.31 | MatBoy | JT, are you using 8port versions ? |
| 22:33.21 | husimon | is there anyway to make voicemail passwordless via the voicemail.conf? |
| 22:33.28 | husimon | for a given user |
| 22:33.37 | MatBoy | husimon, would be nice indeed |
| 22:34.04 | husimon | So you have to do it via a call to voicemailmain with a flag? |
| 22:34.14 | husimon | namely s |
| 22:34.55 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 22:35.07 | MatBoy | husimon, dunno, will look also for it |
| 22:36.30 | JT | MatBoy: what, HPEC is completely different to HWEC |
| 22:36.55 | MatBoy | JT, yeah, it makes the sound more clear too |
| 22:36.57 | *** join/#asterisk mattman99 (n=chatzill@203.171.196.209) |
| 22:37.02 | JT | MatBoy: i doubt that |
| 22:37.03 | MatBoy | I will test those cards later on |
| 22:37.03 | JT | anyway |
| 22:37.13 | JT | hwec is only meant to stop echo |
| 22:37.17 | JT | not make it "clear" |
| 22:37.17 | MatBoy | I have read that a lot |
| 22:37.29 | MatBoy | it should indeed |
| 22:37.31 | JT | hpec is useless for PRIs |
| 22:37.44 | MatBoy | indeed |
| 22:37.51 | MatBoy | that was what I was reading |
| 22:37.53 | JT | cpu time |
| 22:38.21 | MatBoy | and some people claim that they got better results with SW instead of HW, and that is what I want to test for this price :) |
| 22:38.55 | *** join/#asterisk l2cache (n=l2cache@m7c5e36d0.tmodns.net) |
| 22:39.09 | JT | they are crazy ;) |
| 22:39.26 | JT | or maybe they were using old digium HWEC that only has 64 taps |
| 22:39.30 | MatBoy | yeah, and I like to test it... you need to test anyhow |
| 22:39.41 | JT | new HWEC in digium and sangoma is 128taps |
| 22:40.54 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 22:44.56 | [hC] | Qwell: does chan_skinny support on-phone conferencing/3way calling? chan_sccp does not and customer is upset. :P |
| 22:45.33 | *** join/#asterisk angryuser[A] (n=nononon@df01t2-212-194-99-161.d4.club-internet.fr) |
| 22:45.51 | JerJer | [hC]: i am not sure about on phone conference |
| 22:45.55 | JerJer | but 3-way should be there |
| 22:46.03 | *** join/#asterisk Tebi (n=tebi@gw.aller.fi) |
| 22:46.15 | [hC] | jer: well, its the same thing really. I guess unless you consider conference to be able to do N people |
| 22:46.34 | [hC] | JerJer: you're saying 3-way should be there with chan_skinny? |
| 22:46.52 | [hC] | I gotta test chan_skinny out on my 7970 again. Last time I played with it there were a few key features that did not work. |
| 22:47.10 | JerJer | not sure any more - been so long since i paid attention |
| 22:50.37 | husimon | does anyone know how the speed dial on linksys atas works? |
| 22:50.41 | husimon | is it something like *<number> |
| 22:50.45 | husimon | or #<number> ? |
| 22:53.22 | hi365_m | how do you use the uds option in sms? |
| 22:53.30 | hi365_m | uds=udh |
| 22:57.11 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.wa.comcast.net) |
| 22:57.32 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:57.32 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:57.55 | *** join/#asterisk BipBip (n=BipBip@bl8-129-125.dsl.telepac.pt) |
| 23:02.18 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 23:02.36 | *** join/#asterisk quigon (n=matias@190.3.121.15) |
| 23:10.34 | *** join/#asterisk mindCrime (n=chatzill@64.241.37.140) |
| 23:10.47 | *** join/#asterisk angryuser (n=nononon@df01t2-213-44-144-45.d4.club-internet.fr) |
| 23:12.22 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:13.31 | *** join/#asterisk znoG (n=gs@130-215-114-200.fibertel.com.ar) |
| 23:16.18 | *** join/#asterisk znoG (n=gs@130-215-114-200.fibertel.com.ar) |
| 23:16.33 | Greek-Boy | I have an SQL query that returns a number with decimals. I want to strip the decimals. How do I go about it? I know this is not an asterisk query but I figured someone here should know... |
| 23:16.42 | znoG | hey .. does anyone know the status of using Asterisk with LDAP? (just for sip.conf and iax.conf, don't really care about the rest) |
| 23:17.00 | `Sauron | rewrite your query to not return decimals |
| 23:17.52 | *** join/#asterisk alerios (n=alerios@190.84.254.27) |
| 23:18.15 | Greek-Boy | lol `Sauron |
| 23:18.40 | alerios | Hi. Is there a way to prevent the Telco to timeout its own t309 timer on data link failure? |
| 23:18.55 | *** join/#asterisk cootoa (n=cootoa@maa78-2-82-241-93-36.fbx.proxad.net) |
| 23:19.02 | cootoa | hi everyone |
| 23:20.26 | Der-Tim | hi there |
| 23:20.40 | *** join/#asterisk jmesquita (n=jmesquit@200.170.114.149) |
| 23:20.55 | cootoa | i'm looking for some help |
| 23:21.07 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 23:21.23 | Der-Tim | everytime i try to reach a remote *, i get an error at the remote sides * cli saying, that the authority (context?!) wasn't found... what can i do to solve this problem? |
| 23:21.27 | cootoa | about iax configuration and outgoing calls |
| 23:22.11 | *** join/#asterisk l2cache (n=l2cache@97.101.178.117) |
| 23:22.16 | fujin | anyone got a script to do sip.conf -> odbc storage? |
| 23:22.58 | cootoa | i have 2 trixbox with an iax trunk |
| 23:23.07 | *** join/#asterisk angryuser[A] (n=nononon@df01t2-212-194-206-244.d4.club-internet.fr) |
| 23:23.25 | cootoa | the trunk works fine everyone can call everyone |
| 23:24.32 | cootoa | on of the trixbox has the oubounds to call the outside (pstn) |
| 23:24.43 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 23:25.19 | eric2 | sometimes, when I make a call, after the call bridge's, I can hear the other person, but they cannot hear me.... do I need to open some ports on my router? |
| 23:25.36 | eric2 | my phone is behind the NAT |
| 23:26.04 | cootoa | i am looking fo a configuration that will allow the ip phone behind the trixbox that does has the outbounds route to call the outside through the iax trunk |
| 23:26.14 | JT | eric2: and the phone connects to where? |
| 23:27.09 | mmlj4 | i can has outbound? |
| 23:28.27 | cootoa | i wasnt the one setting up the oubounds route |
| 23:29.18 | cootoa | only know that to make outgoing calls the phone needs to dial 9 before the number |
| 23:29.25 | cootoa | or 7 |
| 23:31.10 | *** join/#asterisk sergey_masushko (n=sergey@66.243.68.219) |
| 23:32.11 | cootoa | plz help |
| 23:33.41 | fujin | so anyone? script to go from sip.conf -> database? |
| 23:34.16 | *** part/#asterisk sergey_masushko (n=sergey@66.243.68.219) |
| 23:35.02 | outtolunc | you wiki broke? |
| 23:35.06 | outtolunc | http://www.krisk.org/asterisk/ast2sql.pl |
| 23:35.23 | outtolunc | +r |
| 23:36.04 | outtolunc | http://www.voip-info.org/wiki/view/Asterisk+RealTime+Static |
| 23:36.55 | fujin | you fail |
| 23:36.58 | fujin | that's for realtime static |
| 23:37.01 | obnauticus | LOL |
| 23:37.21 | outtolunc | what do you think putting your sip.conf in a db is |
| 23:37.39 | fujin | realtime buddy |
| 23:37.47 | fujin | sip(peers|users) |
| 23:37.50 | outtolunc | goodluck |
| 23:38.44 | cootoa | anyone... |
| 23:39.54 | *** join/#asterisk bkw__ (n=brian@adsl-64-149-54-142.dsl.tul2ok.sbcglobal.net) |
| 23:40.19 | hi365_m | tzafrir tzafrir_home ping |
| 23:42.23 | outtolunc | maybe bkw has his mysql_load_res_config.pl (because i'm not sharing) |
| 23:43.02 | outtolunc | now that is old <G> |
| 23:47.57 | [hC] | Anyone know how asterisk sorts the list of "show agents" ? |
| 23:48.10 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:48.39 | [hC] | FOP seems to get its data sorted the same way, and im looking to figure out how to sort it my own way. |
| 23:51.56 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-54-142.dsl.tul2ok.sbcglobal.net) |
| 23:53.40 | vap0rtranz | btw: about the signaling v. signalling conv earlier: the same typo is in the book (pg. 78 versus 80). *eek* |
| 23:53.50 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 23:54.00 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.101.126) |
| 23:54.14 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 23:55.47 | JT | vap0rtranz: which one is the typo? |
| 23:57.02 | vap0rtranz | well x86 says his works with signaling; i fixed mine by using signalling, so maybe this is version dependent?? |
| 23:57.11 | *** join/#asterisk xcompass (n=compass@sr-78.srsv01.resnet.ubc.ca) |
| 23:57.24 | JT | signalling is correct UK/AU/etc English |
| 23:57.30 | JT | signaling is US |
| 23:57.35 | JT | which isn't really english :P |
| 23:57.39 | vap0rtranz | hah |
| 23:57.46 | vap0rtranz | bloody hell it ain't |
| 23:58.02 | JT | they speak american ;) |
| 23:58.12 | vap0rtranz | we ?? |
| 23:58.51 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:59.04 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 23:59.17 | obnauticus | JT == AU |
| 23:59.17 | obnauticus | silly |
| 23:59.18 | obnauticus | nubtard |
