| 00:00.10 | file | docelm0: ask Mohammed who Joshua Colp is... :) |
| 00:00.30 | docelm0 | We had this discussion already |
| 00:00.32 | docelm0 | who are you? |
| 00:00.41 | docelm0 | ohh chris |
| 00:00.42 | docelm0 | t |
| 00:00.46 | aaronz | can i have another thread inside my app, or does each app have to have 1 thread? |
| 00:00.50 | docelm0 | your colp |
| 00:00.54 | file | yes, yes I am |
| 00:00.58 | X-Files | denon: i see ;) |
| 00:01.18 | lidl | what kind of gsm gateway would you recommend for 8 gsm channels? |
| 00:01.39 | denon | X-Files: I would send a message to the mailing list |
| 00:01.40 | tzanger | why, an 8-port one, of course |
| 00:01.44 | denon | that's the fastest way |
| 00:02.03 | X-Files | denon: But it my last hope |
| 00:02.18 | denon | I think that's the least of your worries |
| 00:03.33 | X-Files | denon : I shall try, but I will not be assured that result |
| 00:05.18 | greyhound4334 | anyone seen this: zap channel is answering on OUTBOUND analog calls from phone connected to same pstn line as x100p card? |
| 00:06.22 | *** join/#asterisk GXTi (n=omgwtfbb@freenode/developer/GXTi) |
| 00:06.26 | aaronz | can i simultaneously read & write to an asterisk channel or are they not threadsafe? |
| 00:08.44 | *** join/#asterisk Smi|k (n=smilk@adsl-66-159-200-157.dslextreme.com) |
| 00:09.16 | Smi|k | what is the best solution for a small company with limited resources who needs a custom crm app? |
| 00:10.09 | denon | notepad |
| 00:10.15 | liran_ | lol |
| 00:10.19 | Smi|k | I considered that |
| 00:10.25 | liran_ | but its not open source, nor free denon :) |
| 00:10.28 | denon | vi |
| 00:10.31 | Smi|k | like a google mini + notepad and save each contact as its own file |
| 00:10.36 | liran_ | yeah, thats more like it |
| 00:10.40 | Smi|k | then just use the google search to find crm search results |
| 00:10.46 | liran_ | Smi|k, thats actually a great idea |
| 00:10.50 | Smi|k | using custom modifiers |
| 00:10.50 | tzanger | mmmm ice cream and tuna |
| 00:11.17 | liran_ | uhmmm, i need some perl tool to parse all the Master.csv file and also do a little billing |
| 00:11.38 | tzanger | liran_: so build it |
| 00:11.41 | Smi|k | [leads](CA)(18-24)Josh |
| 00:11.41 | konfuzed | liran_: hey there |
| 00:11.51 | liran_ | dont have the skills |
| 00:11.53 | konfuzed | tzanger: it always the easy way with you |
| 00:11.57 | tzanger | konfuzed: :-) |
| 00:12.01 | liran_ | or alteast, it would take me some time to accomplish |
| 00:12.05 | Smi|k | once people get good with the google search strings and the server is set up with all the custom fields it will be lightning fast and easy |
| 00:12.10 | liran_ | konfuzed, hey man, how're you? |
| 00:12.23 | Smi|k | or imagine "call Josh Porter cell" |
| 00:12.26 | Smi|k | in google mini |
| 00:12.36 | file | another Josh? yeekz |
| 00:12.37 | Smi|k | then the result is parsed and asterisk calls etc..etc.. |
| 00:12.41 | liran_ | tzanger, just give me a link for one heh |
| 00:12.58 | tzanger | liran_: if I had a link to one, i'd have done that |
| 00:13.12 | liran_ | tzanger, you're ok :) |
| 00:13.23 | tzanger | something along the lines of Parse::CSV or DBD::CSV would get you far in a hurry |
| 00:14.02 | Smi|k | really though |
| 00:14.09 | Smi|k | I'm very confused with the whole CRM thing |
| 00:14.14 | konfuzed | liran_: have you seen perl-agi |
| 00:14.21 | Smi|k | every time an open source project starts developing into something good it goes commercial (like sugarcrm_ |
| 00:14.25 | wunderkin | file: imagine that, eh? |
| 00:14.27 | tzanger | konfuzed: that's for AGIs not for Master.csv parsing |
| 00:14.45 | liran_ | konfuzed, no |
| 00:14.57 | Dr_Ray | perl sed/awk are pretty good at Master.csv parsing |
| 00:15.07 | tzanger | "Grep," sed awk. |
| 00:15.21 | file | tzanger: is that going to be your new MSN tagline? |
| 00:15.28 | tzanger | file: nah |
| 00:15.32 | file | bah |
| 00:15.34 | tzanger | I've got Bender in there right now |
| 00:15.54 | konfuzed | tzanger: im often a little konfuzed but doesnt perl-agi provide perl modules for passing commands to * |
| 00:16.04 | konfuzed | as in basically any command |
| 00:17.12 | tzanger | konfuzed: yes |
| 00:17.14 | konfuzed | as in could (should )easily pass commmands to make sql calls to read the CDRs |
| 00:17.21 | tzanger | but there are no commands for viewing/manipulating Master.csv |
| 00:17.25 | konfuzed | or may be that is some other "API" |
| 00:17.26 | tzanger | konfuzed: no |
| 00:17.52 | konfuzed | perhaps not master.csv but what about CDR |
| 00:18.13 | tzanger | asterisk generates CDRs it does not really manipulate them in any way |
| 00:18.40 | InfraRed | i just realised why my phone banking wasnt working |
| 00:18.43 | InfraRed | i was typing my icq number instead of my account number |
| 00:18.45 | InfraRed | :/ |
| 00:18.52 | tzanger | hahahahh |
| 00:18.58 | konfuzed | yeah * generates cdr and rights them to (one of a few destinations) prefereably sql database and can also generate cdr reports |
| 00:20.23 | konfuzed | so my thinking ( Presuming the above reports are doable ) is use perl-agi to access the cdrs and then have perl pass the results to your billing system |
| 00:21.19 | tzanger | again, AGI is not the way to do it |
| 00:21.22 | tzanger | AGI is for call handling |
| 00:21.27 | tzanger | not post-call reporting |
| 00:21.38 | konfuzed | i see |
| 00:21.40 | tzanger | if you want as-the-call-happens CDR updates then sure you can use AGI |
| 00:21.40 | konfuzed | or hear ya |
| 00:21.46 | konfuzed | ah type ya I guess |
| 00:22.04 | file | you could always jump through the list of CDRs on the channel while the call is going on ... freaks! |
| 00:22.29 | file | how random of me |
| 00:22.49 | konfuzed | the AGI does not implement a full set of command handling but was built with the idea of call handling and doesnt facilitate much else |
| 00:23.42 | konfuzed | admittedly I have not done much more than skim over perl-AGI as I dont do any perl programming |
| 00:24.02 | konfuzed | but a developer I know was all excited about the functionality provided |
| 00:24.54 | konfuzed | but then perl for opening and reading any.csv should be a piece of cake right |
| 00:25.16 | konfuzed | its all in how you use the data |
| 00:25.23 | konfuzed | not how big your data is |
| 00:25.30 | konfuzed | or how big your code is |
| 00:25.33 | konfuzed | ;^) |
| 00:26.37 | konfuzed | liran_: are you going to be coding in perl yourself? |
| 00:27.06 | liran_ | konfuzed, yes but i dont want to |
| 00:27.07 | konfuzed | or are you wanting to inform some other perl programmer what to use to do this |
| 00:27.15 | liran_ | konfuzed, i started but its getting too complicated |
| 00:27.36 | liran_ | konfuzed, id be very happy to get a tool that does it from someone who already worked on it |
| 00:28.10 | *** join/#asterisk test34 (i=1000@unaffiliated/test34) |
| 00:28.23 | konfuzed | liran_: do you remember when I said, at first it starts with a simple inquiry about making a bill or an invoice and then leans into all kinds of other considerations and things that have to be looked after |
| 00:28.30 | konfuzed | well I said something like that |
| 00:28.42 | liran_ | more or less |
| 00:29.20 | konfuzed | has there been any indication of using some other formal accounting system that will need reports from the billing system |
| 00:29.45 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 00:29.50 | liran_ | nope, nothing |
| 00:30.02 | L|NUX | hey |
| 00:30.02 | konfuzed | or do they still want you to come up with the magic solution that does everything before they know it needs to be done |
| 00:30.11 | konfuzed | ;^) |
| 00:30.14 | L|NUX | when i try to dial using iax2 to fwd then i got this message one console |
| 00:30.15 | L|NUX | Nov 15 16:25:51 WARNING[2392]: chan_iax2.c:6885 socket_read: Call rejected by 192.246.69.187: No authority found |
| 00:30.15 | L|NUX | <PROTECTED> |
| 00:30.20 | L|NUX | what does it means ? |
| 00:30.22 | liran_ | all that is needed is some tool to parse the Master.csv tool and do some basic billing upon caller destination numbers |
| 00:30.23 | konfuzed | ive only been asked to do that once or twice |
| 00:30.42 | liran_ | i mean, its not a magic solution |
| 00:30.47 | liran_ | i understand what they want |
| 00:31.00 | konfuzed | so youve done billing systems before? |
| 00:31.08 | liran_ | they want a semi-billing system. just some tool to parse the master.csv and produce a report of how many seconds it used etc |
| 00:31.11 | liran_ | no, never konfuzed |
| 00:31.37 | konfuzed | master.csv is a simple report |
| 00:31.44 | konfuzed | you just need to make it more pretty |
| 00:31.49 | konfuzed | and add a few words |
| 00:32.05 | liran_ | yeah, thats easy, but what about the rates billing? |
| 00:32.07 | konfuzed | or rather you start with an out put invoice template |
| 00:32.38 | *** join/#asterisk bmg505 (n=leon@rndf-146-57-40.telkomadsl.co.za) |
| 00:32.50 | konfuzed | and your perl reads in the master.csv plus contact and billing address and then spits them in to an email template |
| 00:32.51 | konfuzed | done |
| 00:33.15 | Dr_Ray | or html template |
| 00:33.18 | konfuzed | you have a plain text rate card that can be updated when ever desired |
| 00:33.37 | *** join/#asterisk Rowter (n=SilverDr@201.135.26.195) |
| 00:33.52 | Rowter | I could call out with manager and detect a fax? |
| 00:33.55 | liran_ | for an example |
| 00:33.56 | konfuzed | your perl charges based on volume of service use indicated by master.csv times the rate card category |
| 00:33.58 | konfuzed | done |
| 00:34.27 | konfuzed | out put to html template that is attached to an email |
| 00:35.02 | liran_ | right. so say if someone calls to a 212 destination then the rate is 0.2usd/m and if it's 514 destination call then the rate is 0.4usd/m |
| 00:35.27 | konfuzed | you're either gonna keep at this kind of level or you're gonna have to get into a more dedicated real integrated billing system |
| 00:35.39 | liran_ | right. ok |
| 00:35.54 | liran_ | i think i actually got the idea, i want to try and move on with the perl program |
| 00:36.06 | liran_ | thanks konfuzed, i think you had it organized in my head just now |
| 00:36.42 | konfuzed | i need real integrated billing because it has to do more than deal with asterisk it also has to deal with radius accounts for various indpendent agents |
| 00:37.17 | konfuzed | liran_: No probs I have a love hate relationship with accounting/billing so |
| 00:37.55 | konfuzed | it always helps to bridge the GAAP of applied billing accounting to any one |
| 00:38.04 | konfuzed | ;^) |
| 00:38.51 | Druken | konfuzed: if your doing a billing suite... i'm intrested :) |
| 00:39.04 | konfuzed | lawyers wives doing the billing and accounting has to be the worst case scenario ever |
| 00:39.07 | *** join/#asterisk tessier (n=treed@wsip-68-15-4-13.sd.sd.cox.net) |
| 00:39.59 | konfuzed | Druken: I'm co-ersing real developers to facilitate a real billing accounting setup |
| 00:40.32 | konfuzed | I can only pay in chocolate though |
| 00:40.52 | konfuzed | high rate cacao to boot |
| 00:40.54 | konfuzed | ;^) |
| 00:41.04 | Druken | konfuzed: geez... i can only try to convince my wife to give a lil... hehehe |
| 00:41.16 | *** part/#asterisk patpatnz (n=pjs@unaffiliated/patpatnz) |
| 00:42.35 | konfuzed | Druken: try these chocolates http://e3live.com/productpages/truffles.htm |
| 00:42.45 | konfuzed | you can guarantee the health benefits |
| 00:42.54 | Druken | i'm sure my wife would like them better :) |
| 00:43.03 | *** join/#asterisk bweschke (n=bweschke@wsip-24-120-60-190.lv.lv.cox.net) |
| 00:43.12 | konfuzed | and check out the pepper chocolate |
| 00:43.34 | konfuzed | you would be quite happy if your wife had a half dozen too |
| 00:43.43 | konfuzed | ;^) |
| 00:43.51 | Druken | probably |
| 00:44.19 | konfuzed | these puppies are better than gin |
| 00:44.41 | tzanger | better than gin? surely you jest |
| 00:46.19 | konfuzed | tzanger: I never jest about chocolate |
| 00:47.17 | konfuzed | they've got these chocolates there that actually have the same chemicals the brain uses to express the emotion of love |
| 00:47.59 | konfuzed | horny awake women are way better then drunk numb women |
| 00:48.08 | konfuzed | at least for me any way |
| 00:48.35 | tzanger | well hell yeah |
| 00:48.49 | konfuzed | i know too much about neurotransmitters thats my problem |
| 00:49.12 | Dr_Ray | Druken |
| 00:50.03 | konfuzed | uhm the active alkaloyd in chocolate will relax the muscles in throat and supress the cough/choke reflex |
| 00:50.18 | konfuzed | you can extrapolate the rest |
| 00:50.37 | Smi|k | any suggestions for crm? |
| 00:50.42 | konfuzed | compiere |
| 00:51.17 | konfuzed | and make it run on mysql or postgres |
| 00:51.28 | *** part/#asterisk greyhound4334 (n=john@adsl-69-106-241-168.dsl.pltn13.pacbell.net) |
| 00:52.32 | *** join/#asterisk zigman (i=zigman@irc.zigman.de) |
| 00:52.34 | konfuzed | my favourite is moodle.org but thats not much of a sales environment |
| 00:52.39 | Dr_Ray | Druken |
| 00:52.45 | konfuzed | moodle is all about crm |
| 00:52.49 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 00:52.54 | konfuzed | and is in perl |
| 00:53.35 | konfuzed | im gonna see about getting a module into moodle that can pull in account info about asterisk |
| 00:53.37 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
| 00:55.24 | konfuzed | maybe with perl-agi |
| 00:55.34 | konfuzed | but sounds like more will be needed |
| 00:58.01 | *** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com) |
| 01:01.39 | jeffik | can anybody answer a couple questions about sipura 1001 |
| 01:02.34 | *** join/#asterisk oldave (i=dbs@h245.189.141.67.ip.alltel.net) |
| 01:02.47 | oldave | how's the clue level tonight? |
| 01:03.27 | [TK]D-Fender|AFK | Asterisk doesn't have account info last I checked.... |
| 01:03.30 | X-Files | drumkilla: u there ? |
| 01:03.31 | konfuzed | konfuzed: what Klue ? |
| 01:03.37 | hhoffman | how does one answer that question? |
| 01:03.38 | *** join/#asterisk Gourou_fou (n=x@ACaen-151-1-25-53.w86-195.abo.wanadoo.fr) |
| 01:03.43 | konfuzed | 8^) ;^) |
| 01:03.43 | Gourou_fou | hello :) |
| 01:03.50 | oldave | heh... just checking |
| 01:04.01 | oldave | got a voicetronix card issue |
| 01:04.10 | Gourou_fou | i've a problem... |
| 01:04.18 | Gourou_fou | ("oh noo") |
| 01:04.55 | Gourou_fou | i've convert an mp3 in wav format 16Kb 8KHz mono with sox |
| 01:05.08 | Gourou_fou | and wav to gsm, with sox |
| 01:05.37 | Gourou_fou | but when i use "background(myfile.gsm)" |
| 01:05.48 | Tclp | hey all, I have a fairly unrelated question -- we have a Vonage line here at the office and I'm hoping to be able to drop into this line via some sort of PC-to-PC call (Home to Office to the VOIP Line), I'm not sure what type of hardware/software would be required for something like this ... we do have some old unused Asterisk boxes that are no longer being used bu I'm hoping to not have ot goto that extent |
| 01:05.56 | Gourou_fou | Nov 16 02:07:32 WARNING[14289]: file.c:493 ast_openstream_full: File carioca.gsm does not exist in any format |
| 01:06.08 | file[laptop] | don't specify the file extension |
| 01:06.13 | file[laptop] | just use background(myfile) |
| 01:06.26 | Tclp | looking to do a pc to pc call / pc that I call auto answers and drops me to dial tone via a modem or something |
| 01:06.28 | Gourou_fou | but format ulaw if defaut and my file is in gsm format ..? |
| 01:06.34 | Gourou_fou | i try :) |
| 01:07.21 | Tclp | anyone have any idea ? :) |
| 01:07.25 | Gourou_fou | yeaaaaaaaaaaaaaaaah |
| 01:07.30 | Gourou_fou | it's done |
| 01:07.43 | IronHelix | tclp- vonage is a bitch |
| 01:07.50 | Gourou_fou | i've try with the extension because i've an other error in the past |
| 01:07.54 | IronHelix | the only way you'll get vonage into * is with some sort of fxo port |
| 01:07.57 | Tclp | Iron .. yeah but it has nothing to do with vonage |
| 01:08.34 | IronHelix | vonage also does not originate or terminate calls over anything other than the pstn |
| 01:08.45 | IronHelix | or for that matter, do anything particularly useful |
| 01:08.46 | Tclp | I'm basically just looking for a software (perferably win32) that I can call via IP from another PC that allows me to drop to the modem to dial out |
| 01:08.48 | IronHelix | like reply to their email |
| 01:09.10 | Rowter | its possible to detect outgoing fax ? |
| 01:09.11 | IronHelix | so you jsut want to share your line |
| 01:09.30 | Tclp | in a sense I suppose |
| 01:09.45 | Tclp | I want to be able to use the line over the internet from remote location |
| 01:09.48 | justinu | vonage is a bitch, but vonage works very well |
| 01:09.51 | IronHelix | first- you cant use a modem, you'll need a real fxo port. digium x100 clone will do nicely |
| 01:10.02 | IronHelix | load asterisk and configure the x100 card |
| 01:10.06 | IronHelix | plug x100 into vonage |
| 01:10.19 | IronHelix | then connect your pc's to asterisk with softphones |
| 01:10.27 | IronHelix | creating entries in sip.conf for each softphone |
| 01:10.31 | justinu | or you could buy the "vonage softphone" account |
| 01:10.32 | Tclp | I'm hoping to not have to use asterisk |
| 01:10.37 | justinu | and connect asterisk to them via sip |
| 01:10.48 | IronHelix | why no *? |
| 01:10.53 | Tclp | right now the vonage lines are hooked into an Analog PBX |
| 01:11.00 | oldave | TcIp... you can use any of several softphone clients... and just let the provider handle dialing out... |
| 01:11.09 | Tclp | modems do support POTS calling don't they ? |
| 01:11.17 | IronHelix | call yes, voice no |
| 01:11.22 | oldave | TcIp... yeah, but they don't do the voice out bit |
| 01:11.32 | Gourou_fou | thanks for help file[laptop], good night :) ++ |
| 01:11.37 | oldave | sure, they can be answering machines, listening to voice... |
| 01:11.38 | Tclp | voice modems do ? |
| 01:11.47 | IronHelix | not with anything useufl that i've seen |
| 01:12.00 | oldave | but not enough to really be useful... |
| 01:12.02 | konfuzed | IronHelix: im debating what to do with this voinage ata service account |
| 01:12.08 | Tclp | hmmm |
| 01:12.15 | oldave | if voicemodems as answering machines worked well, they'd be everywhere by now |
| 01:12.15 | konfuzed | good ideas above |
| 01:12.25 | Tclp | how about a usb>RJ11 adapter ? |
| 01:12.37 | IronHelix | my advice- kill it before it becomes at all useful, because if you ever need to get your number out of vonage you might as well tear your eyes out with a spork |
| 01:12.45 | oldave | TcIp... kill the Vonage account |
| 01:12.56 | justinu | lol |
| 01:13.02 | konfuzed | i mostly agree |
| 01:13.06 | IronHelix | tcip, why dont you want to use *? |
| 01:13.09 | oldave | USB>RJ11 doesn't seem to exist |
| 01:13.15 | konfuzed | its been in use for months because of a bad setup |
| 01:13.24 | konfuzed | dont care about keeping the phone number though |
| 01:13.33 | justinu | then just turn it off |
| 01:13.34 | IronHelix | oldave- it exists, theres a gadget called the internet phonejack or internet linejack |
| 01:13.36 | IronHelix | not widely used tho |
| 01:13.41 | konfuzed | but I like having more than one did provider and incoming pipe |
| 01:13.46 | justinu | true |
| 01:14.02 | IronHelix | Vonage once took no less than (this is not an exaggeration or a joke) 8 MONTHS to reply to an email of mine |
| 01:14.05 | oldave | odd... can't imagine the utility |
| 01:14.15 | oldave | I've stayed away from Vonage... |
| 01:14.28 | oldave | did get a Broadvoice BYOD account... |
| 01:14.35 | IronHelix | i wish i'd learned about * before i learned about vonage, not the other way around :( |
| 01:14.37 | oldave | and promptly set up Asterisk for it |
| 01:14.45 | konfuzed | i cant use this linksys ata unless voanage unlocks it so I want to send it back |
| 01:14.46 | justinu | broadvoice seems to work well |
| 01:14.57 | justinu | IronHelix: yeah, vonage is what convinced me voip could work well |
| 01:15.00 | oldave | I want a linksys ATA unlocked |
| 01:15.02 | konfuzed | id be happ y to keep a few softphone accounts on the go and pipe them into asterisk |
| 01:15.36 | oldave | right now, just doing XLite into Asterisk here |
| 01:15.46 | oldave | was kinda wild last night... had the FWD account call the Broadvoice account |
| 01:15.50 | *** join/#asterisk blop (i=blop@VoIP-with-Asterisk.mgcp.h323.sccp.sip.iax.be) |
| 01:15.54 | oldave | quality was decent |
| 01:15.59 | konfuzed | now ive got a HandyTone 486 to configure and setup |
| 01:16.00 | konfuzed | ;^) |
| 01:16.08 | konfuzed | low end but better than a locked linksys |
| 01:16.20 | oldave | yeah, undecided on which ATA to get here |
| 01:16.28 | justinu | sipuras are nice |
| 01:16.34 | oldave | just want it to work, and not be dropping SIP registration every little bit |
| 01:16.39 | konfuzed | check out the PA1688 models |
| 01:16.41 | IronHelix | sipuras are good, i have yet to hear any major problems with them |
| 01:16.49 | konfuzed | find something with iax |
| 01:16.52 | oldave | got a Soyo phone at the office, trying it out, not impressed |
| 01:16.53 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:16.55 | IronHelix | and they are configuraable up the wazoo |
| 01:16.55 | konfuzed | handytone doesnt have it yet |
| 01:16.59 | justinu | sipura is also super customizeable |
| 01:17.01 | justinu | yeah |
| 01:17.07 | Rowter | I need to detect when a fax machine answeres an outgoing call. NV_FaxDetect and the zaptel fax detect seem to only work in calls originated FROM a fax machine, not for calls ANSWERED by a fax. |
| 01:17.26 | oldave | at the office, I've got a PRI into a Digium 110... then 2 Openswitch 12 cards... |
| 01:17.28 | oldave | a note... |
| 01:17.34 | Rowter | maybe background detect? |
| 01:17.37 | oldave | don't call one port from another port |
| 01:17.39 | konfuzed | i just got this HT 486 for 99$cdn |
| 01:18.29 | oldave | here at the house, it's just the Solaris box running * |
| 01:19.01 | oldave | what's that work out to, about $1.78 US? |
| 01:19.02 | oldave | heh |
| 01:19.15 | justinu | sipuras are 60 bucks here |
| 01:19.16 | *** join/#asterisk test34 (i=1000@unaffiliated/test34) |
| 01:22.16 | *** join/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 01:22.18 | konfuzed | oldave: if the economy keeps going in this directio it'll be 110 USD |
| 01:22.33 | oldave | yeah, yeah... was nice while it lasted |
| 01:22.47 | oldave | I'm guessing no voicetronix wizards in this evening? |
| 01:24.49 | tzanger | tainted_: werd? |
| 01:25.42 | tzanger | werd to the kram |
| 01:25.45 | tzanger | kay-ram |
| 01:26.02 | tzanger | kay-to-tha-are-to-tha-aye-to-tha-em |
| 01:26.08 | tzanger | sure YOU have a cool nick |
| 01:26.12 | tzanger | you just can't do that to mine |
| 01:26.17 | file[laptop] | kram: off IRC muffin man, before they attack! |
| 01:26.17 | IronHelix | lol |
| 01:26.54 | tzanger | tee-to-tha-zed-to-tha-aye-to-tha-enn-to-tha-oh-bollocks-screw-this-where's-the-booze |
| 01:27.25 | hhoffman | can you do multiple SayXXX in a exten? |
| 01:28.16 | hhoffman | like exten =>123,1,(SayAlpha(A),SayPhonetic(test)) |
| 01:28.35 | *** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc) |
| 01:29.45 | tzanger | no |
| 01:29.50 | tzanger | do it in two steps |
| 01:29.54 | tzanger | 123,1 and 123,2 |
| 01:30.00 | hhoffman | ah, ok |
| 01:30.29 | *** join/#asterisk tessier (n=treed@wsip-68-15-4-13.sd.sd.cox.net) |
| 01:32.36 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 01:33.01 | *** join/#asterisk Luke-Jr (n=luke-jr@user-0c938qu.cable.mindspring.com) |
| 01:35.19 | *** join/#asterisk bweschke_ (n=bweschke@24-234-5-214.ptp.lvcm.net) |
| 01:35.59 | hhoffman | ah! SayPhonetic() isn't at all what I thought it was |
| 01:37.16 | shido6 | Cepstral |
| 01:37.18 | shido6 | might |
| 01:39.25 | hhoffman | anyone familiar with: Nov 15 20:37:34 ERROR[24614]: chan_modem.c:1023 load_module: Failed to load driver chan_modem_aopen.so |
| 01:42.59 | oldave | familiar with that error... see it frequently |
| 01:43.04 | kram | back |
| 01:43.08 | oldave | don't care, since there's no aopen gear anywhere in the machine |
| 01:43.30 | oldave | my main concern is getting the vpb 3.0 beta driver working with * |
| 01:43.33 | justinu | noload chan_modem.so |
| 01:43.55 | oldave | justin... if you do that, * won't start, 'cause something's trying to force the aopen one |
| 01:44.04 | justinu | weird |
| 01:44.24 | oldave | at least with chan_modem loading, it handles the error and continues, rather than just shutting down |
| 01:45.21 | *** join/#asterisk oogle (n=jart@justin.ctlinc.com) |
| 01:47.16 | tzanger | is there something obviously wrong with this? |
| 01:47.17 | tzanger | exten => s,10,GotoIf($[${AUTH_TRIES} > 0],4) |
| 01:47.26 | tzanger | AUTH_TRIES=2 when it hits that |
| 01:47.31 | tzanger | the CLI says 1|4 |
| 01:47.34 | tzanger | but it drops to the next line |
| 01:48.06 | tzanger | <PROTECTED> |
| 01:48.21 | *** join/#asterisk tessier (n=treed@wsip-68-15-4-13.sd.sd.cox.net) |
| 01:48.25 | tzanger | that's telling me that $[${AUTH_TRIES} > 0] evaluates true, which should jump to priority 4 |
| 01:48.31 | justinu | is there a 4? |
| 01:48.33 | tzanger | yes |
| 01:48.42 | tzanger | exten => s,4,Playback(vm-password) |
| 01:48.45 | justinu | weird |
| 01:48.50 | tzanger | this is 1.0.7 btw |
| 01:48.53 | ManxPower | tzanger, add in the no-match priority too. |
| 01:48.54 | justinu | oh |
| 01:48.59 | tzanger | ManxPower: you need both? |
| 01:49.02 | tzanger | ugh |
| 01:49.06 | tzanger | gotta love old software :-) |
| 01:49.09 | ManxPower | tzanger, no, but it can help diagnose things. |
| 01:49.16 | tzanger | I miss the 'n' and n(label) dearly |
| 01:49.36 | ManxPower | tzanger, and it generates a debug message at some log levels |
| 01:49.37 | oldave | is there a reason you wouldn't upgrade the box? |
| 01:49.48 | tzanger | oh wait |
| 01:49.51 | justinu | probably because it works |
| 01:49.52 | tzanger | oldave: yeah |
| 01:50.02 | tzanger | I can't upgrade for specific reasons I can't go in to at this point |
| 01:50.03 | tzanger | it's not my box |
| 01:50.05 | ManxPower | oldave, IAX2 trunking problems between 1.2 and 1.0, slightly different behavoir. |
| 01:50.46 | oldave | ok |
| 01:51.02 | oldave | now someone 'splain why I don't get busy tones on my Voicetronix cards :) |
| 01:51.35 | tzanger | gotoif **REQUIRES* condition?true |
| 01:51.38 | tzanger | not condition,true |
| 01:52.01 | ManxPower | oldave, make sure you have an /etc/asterisk/indications.conf |
| 01:52.18 | ManxPower | tzanger, Ah. |
| 01:52.28 | ManxPower | tzanger, want some example GotoIf's? |
| 01:52.44 | tzanger | nope |
| 01:52.44 | tzanger | got it |
| 01:52.49 | tzanger | it was just braindeadness on my part |
| 01:52.51 | oldave | Manx... do... and ringback works, dialtone, too... but no busy tones |
| 01:52.52 | tzanger | it's working now |
| 01:53.00 | oldave | annoying when a number is actually busy and all I get is silence |
| 01:53.04 | *** join/#asterisk emturan (n=sadasd@85.97.68.144) |
| 01:53.13 | tzanger | Ringing() isn't working though, ugh |
| 01:53.27 | ManxPower | Complex gotoif example: http://pastebin.ca/28860 |
| 01:53.33 | IronHelix | dial(whatever,,r) try that |
| 01:53.51 | tzanger | IronHelix: no no |
| 01:53.54 | ManxPower | oldave, Generally if you don't get ringing automatically, you frequently won't get it by using .r on dial |
| 01:53.57 | tzanger | you NEVER do that unless you specifically need it |
| 01:54.06 | oldave | I get ringing :) |
| 01:54.13 | oldave | tzanger is having *that* issue |
| 01:54.16 | IronHelix | i've specifically needed it a bunch of times |
| 01:54.19 | tzanger | I'm calling an IAX2 peer and the peer has "Ringing()" as the first priority (no answer yet) |
| 01:54.30 | IronHelix | hmmm |
| 01:54.31 | tzanger | I should get ringtone but don't |
| 01:54.38 | IronHelix | yeah you should even w/out ,,r |
| 01:54.38 | tzanger | I was at one point I have to see waht it was I buggered up |
| 01:55.58 | ManxPower | Also ,r can MASK problems. |
| 01:56.01 | oldave | I get ringing, but when it drops through to Busy, I don't get busy tones |
| 01:56.27 | ManxPower | In some situations ,r will make the caller hear a ringing tone instead of a message like "the number you have dialed has been disconnected" message. |
| 01:56.34 | IronHelix | yeah i know ,,r is 'bad' it just is required sometimes |
| 01:57.21 | IronHelix | it wouldnt apply in his situ, but from reading some of it it looked like it might help |
| 01:58.14 | ManxPower | tzanger, you saw my pastebin? |
| 01:58.30 | tzanger | no |
| 01:58.47 | tzanger | ManxPower: that's specifically what I use 'r' for too (for the cell) |
| 02:04.52 | asterboy | Boolpool.com $21 for Asterisk...anyone else find a better price? |
| 02:06.44 | tzanger | dammit |
| 02:06.52 | [TK]D-Fender | Dunno, I thought the CVS was still free :) |
| 02:07.00 | tzanger | the channel used for an outgoing call cannot receive any variables |
| 02:07.12 | [TK]D-Fender | what kind of channel? |
| 02:07.20 | tzanger | however you can SetVar any variable you want in the channel that the call connects to when it's answered |
| 02:07.24 | oogle | [TK]D-Fender: but you don't get that cool asterisk-shaped cd! |
| 02:07.25 | tzanger | Local/ |
| 02:08.06 | [TK]D-Fender | tzanger : ythere are ways, though a pain at time to get around that... |
| 02:08.27 | asterboy | Grandstream FXS 486 $25, anyone have other prices? |
| 02:08.43 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 02:08.47 | tzanger | [TK]D-Fender: how do you get a var in the *outgoing* channel? |
| 02:08.49 | [TK]D-Fender | tzanger I was working on the same to pass info from IVR in 1 channel to the channel created by AgentCallbackLogin |
| 02:08.51 | tzanger | I can set them in the connecting channel |
| 02:09.08 | [TK]D-Fender | give me a sample |
| 02:09.19 | tzanger | <PROTECTED> |
| 02:09.49 | Qwell | twisted: You rock. |
| 02:10.16 | [TK]D-Fender | oogle : I hope its a well balanced CD otherwise its confetti :D |
| 02:10.29 | [TK]D-Fender | And doubles as shuriken ;) |
| 02:10.36 | tzanger | there's an asterisk-shaped CD? |
| 02:10.41 | Qwell | tzanger: there is |
| 02:10.49 | tzanger | damn, I suck |
| 02:10.57 | tzanger | [TK]D-Fender: well I can cheat |
| 02:11.03 | Qwell | I didn't get one with my tdm400p when I bought mine... :( |
| 02:11.06 | tzanger | I guess I can embed the var in the extension to call and then chop it off |
| 02:11.11 | tzanger | but that's *really* ugly |
| 02:11.13 | Qwell | I managed to get one another way though |
| 02:11.23 | tzanger | I never got one with ANY of my asterisk purchases |
| 02:11.38 | tzanger | and that includes a T100P, two TE405s and a TE406 |
| 02:11.54 | *** join/#asterisk greyhound4334 (n=john@adsl-69-106-241-168.dsl.pltn13.pacbell.net) |
| 02:14.46 | *** join/#asterisk konfuzed (n=KonfuzeD@H129.C72.B0.tor.eicat.ca) |
| 02:14.50 | *** join/#asterisk JunK-Y (n=junky@MTL-HSE-ppp197246.qc.sympatico.ca) |
| 02:14.59 | docelm0 | Yippie! |
| 02:15.38 | [TK]D-Fender | tzanger : You can push a single database entry over which can retreive a whole bunch too... |
| 02:15.48 | [TK]D-Fender | which is what I was looking to do with queues |
| 02:15.50 | tzanger | yeah I guess... eww |
| 02:16.33 | tzanger | bah the wiki mentions a "failed" extension but that doesn't work |
| 02:16.56 | tzanger | I guess I could use Application: and Data: |
| 02:17.11 | *** join/#asterisk konfuzed (n=KonfuzeD@H129.C72.B0.tor.eicat.ca) |
| 02:17.15 | [TK]D-Fender | I was going to collect IVR data in one channel, push it with the uniqueID attached, change the callerID to the uniqueID, have the queue place the call to the agent. In that dial-out it would retreive the original caller ID from the DB and other vars, then do a screen pop with the customer file before dialing |
| 02:17.33 | tzanger | yeah |
| 02:17.53 | konfuzed | hm |
| 02:17.57 | konfuzed | well that crap shouldnt happen again |
| 02:18.00 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 02:18.42 | tzanger | ahh |
| 02:18.48 | tzanger | failed exten must be in the connecting context |
| 02:19.23 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 02:20.31 | tzanger | sweet |
| 02:20.35 | *** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net) |
| 02:20.38 | tzanger | hook up the failed and the timeouts to voicemail and I'm done |
| 02:20.54 | Flauto | is anyone indeed using enumlookup here? |
| 02:21.27 | [TK]D-Fender | tzanger : I'd be interested in seeing if possible when you're done.. |
| 02:21.42 | tzanger | absolutely |
| 02:21.57 | tzanger | part of the contract was that I retained copyright and the ability to release GPL |
| 02:22.12 | konfuzed | i happy to have what I consider to be stuck with a crappy setup in this particular location |
| 02:22.37 | konfuzed | that'll change when i get all the right pieces on hand ann dhave the time to mess with it |
| 02:22.54 | tzanger | eh? |
| 02:23.08 | konfuzed | but right now ive got this linksys voipgateway as the router connected to the dsl modem |
| 02:23.18 | tzanger | [TK]D-Fender: it required a small patch to ParkAndAnnounce() though |
| 02:23.24 | tzanger | and another patch for 1.0.x |
| 02:23.28 | tzanger | but I think my stuff made it in to 1.2 |
| 02:23.39 | konfuzed | and I wanna make this HT486 work on the lan sode of the linksys |
| 02:24.01 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 02:24.27 | konfuzed | the ht486 actually works if plugged into the dslmodem directly but not when getting dhcp from linksys box |
| 02:24.52 | konfuzed | I have the suspision that the linksys is hording traffic but dont know for sure |
| 02:26.04 | tzanger | hording traffic? |
| 02:26.18 | konfuzed | the asterisk box im connecting to is at the DSL providers data center |
| 02:26.19 | [TK]D-Fender | eek |
| 02:26.58 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
| 02:27.27 | konfuzed | im suspiscious that teh linksys will direct all the VoIP ports to it self and not pass them off to other devices on LAN |
| 02:27.42 | [TK]D-Fender | *greed* |
| 02:27.52 | oldave | why would it pass those ports to another device? It's not a router |
| 02:28.00 | konfuzed | can anyone confirm that or confirm it is not the case |
| 02:28.06 | konfuzed | teh one i have is |
| 02:28.16 | konfuzed | linksys RTP300 |
| 02:28.17 | oldave | then turn off the VOIP portion of it |
| 02:28.36 | oldave | and set it to forward those ports to whatever device you want on the LAN side |
| 02:28.57 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 02:30.29 | Damin | Morning.. |
| 02:30.57 | konfuzed | when the RTP300 connects to vonage it does so from the wan interface, then when the ht486 connects to remote asterisk pbx - it should do nat to other than vonage and be happy but |
| 02:31.30 | docelm0 | evening! |
| 02:31.37 | konfuzed | but I dont know if linksys does that intentionally or not or not |
| 02:32.21 | konfuzed | has anyone had an ATA connect through a Vonage GateWay to another asterisk box? |
| 02:33.36 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 02:36.42 | *** join/#asterisk Uberbot (n=Uberbot@c-69-252-219-76.hsd1.nm.comcast.net) |
| 02:36.47 | Uberbot | Hi all. |
| 02:36.52 | Uberbot | Any AGI programmers on? |
| 02:37.44 | Uberbot | I'm trying to set a variable with: $agi->set_variable("name", "\"$name\""); |
| 02:38.10 | Uberbot | But $name has spaces in it and I'm only getting the first work actually put into the variable. |
| 02:38.49 | *** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net) |
| 02:38.58 | Flauto | is there anyone using enumlookup? i tried, but it is not working though it tells me the look up was successful but when it dials enum, it got a busy tone |
| 02:41.33 | alephcom | UberBot: set_variable does not work on recent version of asterisk |
| 02:41.38 | hhoffman | so, on my POTS line asterisk answers the phone even if I've already picked up a handset (analog). Is there anyway to make sure it doesn't do that? |
| 02:42.10 | Uberbot | Grrrr |
| 02:42.26 | Uberbot | alephcom, any hints on how to pass values BACK from an agi script, then? |
| 02:42.45 | [TK]D-Fender | hhoffman : You have an analog phone that isn't BEHIND asterisk? |
| 02:43.02 | alephcom | Uberbot: I'll post a sample |
| 02:43.11 | Uberbot | Thank you much. |
| 02:43.27 | alephcom | Uberbot: $AGI->exec('Set',"LCRSTRING$count=$dialstring"); |
| 02:43.47 | Uberbot | Copied. Thanx. |
| 02:44.46 | Nugget | hhoffman: buy one of those modem line sharing plug things from radio shack |
| 02:44.54 | hhoffman | [TK]D-Fender: correct... it's plugged into the FXS card |
| 02:45.12 | hhoffman | asterisk doesn't seem to detect that I've picked up the handset |
| 02:45.24 | Katty | Nugget: :< |
| 02:45.30 | Netgeeks | AGI's are the tool of the devil /duck |
| 02:45.33 | Katty | Nugget: oh, is this hide and seek? |
| 02:45.42 | Nugget | heh |
| 02:46.34 | Uberbot | alephcom, are you sure you don't mean setvar instead of Set? |
| 02:46.35 | Katty | Nugget: i'll get you yet! |
| 02:46.46 | [TK]D-Fender | hhoffman ... I'm sure it DOES... was the phone RINGING? |
| 02:46.58 | Uberbot | Nov 15 19:44:56 WARNING[2408]: res_agi.c:829 handle_exec: Could not find application (Set) |
| 02:47.46 | ManxPower | Uberbot, SetVar = 1.0, Set = 1.2 |
| 02:48.01 | docelm0 | uberbot um, take out EXEC :) |
| 02:48.09 | docelm0 | are you using PHP? |
| 02:48.14 | alephcom | Yes, thanks. I'm not sure on SetVar in 1.0.9 though.. |
| 02:48.16 | Uberbot | Perl. |
| 02:49.42 | Uberbot | Seems to obe working: $agi->exec('Setvar',"number=$number"); |
| 02:50.14 | Uberbot | But when I upgrade to 1.2, I'll need to change it back to "set". I'll comment my code... |
| 02:50.15 | Uberbot | Thanx. |
| 02:54.51 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
| 02:57.14 | tzanger | ?? |
| 02:57.18 | tzanger | Does _X. not work with 1.0.7? |
| 02:58.03 | docelm0 | should |
| 02:58.10 | *** join/#asterisk test34 (i=1000@unaffiliated/test34) |
| 02:58.13 | docelm0 | I was using that in version 1.0 |
| 03:00.49 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 03:00.59 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 03:01.03 | Ariel_ | hello folks |
| 03:02.48 | Katty | :> |
| 03:03.27 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 03:03.45 | Katty | Nugget: i'm too tired to chase. |
| 03:04.56 | rajiv | anyone seen this while trying modprobe cls_u32 ? /lib/modules/2.4.31/kernel/net/sched/cls_u32.o: unresolved symbol register_tcf_proto_ops |
| 03:06.57 | konfuzed | which public STUN server is is free and good to use for a home user device |
| 03:13.20 | alephcom | Uberbot: Did you get it working? |
| 03:16.32 | kuku5 | katty: do you use FOP |
| 03:16.33 | kuku5 | ? |
| 03:16.55 | *** join/#asterisk asteriskmonkey (n=phil@HSE-Toronto-ppp300017.sympatico.ca) |
| 03:17.07 | asteriskmonkey | hey everybody |
| 03:17.15 | Ariel_ | konfuzed, you can use stun.xten.com |
| 03:17.34 | asterboy | hey, there no more room in here for aster prefixed names! |
| 03:18.03 | Ariel_ | why do you want an asterisk pre-fix anyway |
| 03:18.05 | Nugget | I'll take aster prefix names over all the lame Tux nicks any day of the week. |
| 03:18.13 | asterboy | hey! stop monkeying around! |
| 03:18.18 | asteriskmonkey | :P |
| 03:18.37 | asteriskmonkey | ok question for people out there good question to add to chat log here :) |
| 03:18.44 | Ariel_ | why be different then who you really are? |
| 03:19.19 | tzanger | hmm |
| 03:19.20 | asteriskmonkey | case a) i call a number that is not in sevice on a landline and get message not in services b) i call same number using asterisk with a t1 card and get nothing |
| 03:19.24 | asteriskmonkey | what am i missing? |
| 03:19.38 | tzanger | how do I define *NO* music on hold? commenting out 'default' in musiconhold.conf doesn't seem to quite do it |
| 03:19.49 | Katty | kuku5: yes, when it doesn't deadlock the server |
| 03:19.56 | Ariel_ | tzanger, did you restart |
| 03:20.02 | tzanger | asteriskmonkey: are you calling on a PRI or T1? |
| 03:20.08 | tzanger | Ariel_: restart yes? stop and restart, no |
| 03:20.36 | [TK]D-Fender | tzanger : tried doing NOLOAD for that module? |
| 03:20.55 | tzanger | I think that fucks a lot up :-) |
| 03:20.56 | Ariel_ | you can unload the module as well |
| 03:21.09 | asteriskmonkey | yes unload then load again :) |
| 03:21.19 | tzanger | it has a use count of 1 |
| 03:21.40 | tzanger | I think that's what's messing up my ringing |
| 03:21.48 | Ariel_ | wow tornado's are still active this late in the season. Strange |
| 03:21.53 | tzanger | yeah it's strange |
| 03:21.55 | tzanger | it's bloody windy here |
| 03:22.03 | asterboy | global warming |
| 03:22.13 | tzanger | hahahahahahahhhaha |
| 03:22.18 | tzanger | 22:07 < skolnick> this shit is bananas |
| 03:22.18 | tzanger | 22:09 < cwarner> skolnick, are you wearing panties too? |
| 03:22.29 | asteriskmonkey | if i call a busy number and just get dead air it becaise i dont have priindication=outofband set right? |
| 03:22.42 | tzanger | asteriskmonkey: are you on PRI or CAS T1? |
| 03:22.46 | asteriskmonkey | pri |
| 03:22.52 | tzanger | you want out of band signaling |
| 03:23.01 | tzanger | and then YOU are responsible for inband notification to your people |
| 03:23.20 | asteriskmonkey | tzanger: thanks i put your name in a comment card to digium :D |
| 03:23.26 | tzanger | otherwise Asteirsk will pollute your audio stream with inband notification (busy/congestion tones, SIT, etc) |
| 03:23.29 | asterboy | Tornado in Halifax, (unheard of), especially in November, Hurricans into the greek alphabet, major ice shelf break up, here the weather is in record temps when its suppose to be cold. |
| 03:23.44 | tzanger | it ain't no record temps here |
| 03:23.55 | tzanger | it's unseasonably warm but not record-settingly so |
| 03:23.59 | tzanger | (midwestern ontario) |
| 03:24.16 | asterboy | (alberta bound) |
| 03:24.21 | asteriskmonkey | ive been fighting with that damn pri of mine for months :) what i learned the other day seriously helped with my echo thanks :) |
| 03:24.22 | morale | alberta yay |
| 03:24.25 | Ariel_ | actually it's nice outside temps are normal here for this time of year. |
| 03:24.37 | alephcom | asterboy: I'm sorry for you. It's cold here! Albert Rules though. :-) |
| 03:24.43 | asterboy | were all beer drinking red necks here. |
| 03:25.01 | tzanger | ok wtf |
| 03:25.03 | tzanger | <PROTECTED> |
| 03:25.03 | tzanger | <PROTECTED> |
| 03:25.08 | tzanger | why is one var set and not the other? |
| 03:25.12 | morale | gotta love our co-op gold beer |
| 03:25.17 | oogle | i <3 agi |
| 03:26.01 | tzanger | oogle: so how come I can't see my ${EXTEN} ? |
| 03:27.06 | oogle | what kind of channel is it? |
| 03:27.13 | tzanger | IAX2 |
| 03:27.16 | tzanger | I'm parking it |
| 03:27.21 | docelm0 | Hay file Mo says HI! |
| 03:27.31 | oogle | can you paste in the extension that calls the agi script? |
| 03:27.40 | tzanger | and it seems to try to call up MOH for it even with default commented out |
| 03:27.50 | tzanger | there's nothing much to it |
| 03:28.00 | tzanger | exten => _X.,1,SetVar(ORIG_EXTEN=${EXTEN}) |
| 03:28.00 | tzanger | exten => _X.,2,AGI(incoming-cidcheck.agi) |
| 03:28.35 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
| 03:28.43 | tzanger | I haven't even attempted to see if I can see the caller ID yet :-) |
| 03:29.21 | tzanger | oogle: all the AGI's doing right now is this |
| 03:29.22 | tzanger | my $AGI = new Asterisk::AGI; |
| 03:29.22 | tzanger | my $exten = $AGI->get_variable("EXTEN"); |
| 03:29.22 | tzanger | $AGI->verbose("AGI: EXTEN is $exten\n",1); |
| 03:29.29 | tzanger | This is 1.0.7 |
| 03:29.41 | oogle | listen |
| 03:29.54 | oogle | you know how agi passes a bunch of vars to you when it calls your script? |
| 03:30.01 | oogle | the extension is in agi_extension |
| 03:30.07 | tzanger | oh really |
| 03:30.10 | oogle | yep |
| 03:30.16 | tzanger | still doesn't explain why I can't call it up this way but ok |
| 03:30.35 | oogle | grabbing variables using AGI is a bit flakey |
| 03:30.59 | oogle | they say stuff about if it's set with an app you can't read it |
| 03:31.06 | tzanger | so my $exten = $AGI->agi_extension; ? |
| 03:31.38 | oogle | i'm not sure where the Perl AGI thing puts them, want me to check for you? |
| 03:31.55 | tzanger | I can see if I can find it here |
| 03:32.56 | asteriskmonkey | damn... this is a good read with explanation if you have a pri http://www.asteriskguru.com/tutorials/pri_zaptel.html |
| 03:32.58 | oogle | but i like your approach, i do it the same way... the only purpose of extensions.conf is to call DeadAGI |
| 03:33.17 | tzanger | :-) |
| 03:34.14 | asteriskmonkey | deadagi is the best :) |
| 03:36.39 | tzanger | it does not appear to be in Perl at all, unless it's in the ReadParse() hash |
| 03:36.41 | [TK]D-Fender | Damn, neither Sipura no Linksys seem to have the SPA-941 listed on their sites.... |
| 03:37.04 | [TK]D-Fender | nor* |
| 03:38.27 | tzanger | yup that's where it is |
| 03:38.28 | oogle | tzanger: http://lobstertech.com/jukebox.agi that's a Perl AGI script I wrote several months ago |
| 03:38.43 | oogle | tzanger: scroll down to where it says while (<STDIN>)... that reads the vars it sends you |
| 03:39.20 | oogle | tzanger: but it doesn't use that fancy class you use so if you can find code in the source to Asterisk::AGI that looks like that part of code, you might be able to track where it's putting those vars |
| 03:39.45 | asteriskmonkey | damm it... where do i go to look at handling my inside indications...? i have it set to priindication=outband.. so now i have to set up something somewhere that tells clients whats going on right? currently now i have restarted with priindication=outband when i call a number thats not in services i get some time of silence then a busy signal .. where do i look to make it spit out a proper line notification? |
| 03:39.46 | tzanger | yup I pulled it out already, thanks |
| 03:39.54 | tzanger | I appreciate your help :-) |
| 03:40.28 | oldave | most telcos are going to want inband on the PRI indications |
| 03:40.30 | oogle | tzanger: no prob |
| 03:40.45 | tzanger | oldave: how do you figure? |
| 03:40.55 | oldave | well, let's put it this way.. ALLTEL does |
| 03:41.01 | tzanger | oldave: bell canada don't care |
| 03:41.03 | tzanger | they just go by teh IEs |
| 03:41.05 | tzanger | it's far easier |
| 03:41.15 | asteriskmonkey | i have mci if that helps |
| 03:41.17 | oldave | ALLTEL's odd |
| 03:41.22 | tzanger | :-) |
| 03:41.28 | astcryz | oldave: whats that agi script doing? :) |
| 03:41.46 | asteriskmonkey | i have it set to outband now but i dont understand why i am still getting dead air then busyt |
| 03:41.46 | oldave | NI-2 on the switch... |
| 03:41.54 | asteriskmonkey | yes ni-2 on mine |
| 03:41.55 | oldave | but calls will NOT complete if I set out of band |
| 03:42.25 | asteriskmonkey | really? .. i can complete calls but any usualyl errors like busy or dead numbers etc.. i get silence then busy |
| 03:42.26 | oldave | and here's an interesting tidbit... * keeps sending national for type... had to get 'em to do translations for the local NXXS |
| 03:42.40 | asteriskmonkey | wha thats whacks |
| 03:42.44 | asteriskmonkey | ill try inband sec... |
| 03:43.05 | oldave | monkey... you may have a similar issue with local vs national |
| 03:43.11 | hhoffman | are there any iax soft phones that will build under linux? specifically fedora core 4? |
| 03:43.56 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 03:44.01 | asteriskmonkey | idefsk i think |
| 03:44.21 | asteriskmonkey | oldave: i need help i feel clueless |
| 03:44.27 | asteriskmonkey | do i set to inband then? |
| 03:44.55 | oldave | setting inband can't hurt |
| 03:45.02 | oldave | if that doesn't work for you, set it back |
| 03:45.15 | oldave | is there any rhyme or reason to the numbers you call that give you trouble? |
| 03:45.52 | asteriskmonkey | same result |
| 03:45.53 | asteriskmonkey | odd.. |
| 03:46.09 | asteriskmonkey | anyone know next step on this? |
| 03:46.15 | oldave | is there any rhyme or reason to the numbers you call that give you trouble? |
| 03:46.27 | oldave | in other words, do long distance numbers work, but not local numbers? |
| 03:46.57 | asteriskmonkey | no |
| 03:47.01 | asteriskmonkey | local works great |
| 03:47.06 | asteriskmonkey | i have no problem with local |
| 03:47.29 | oldave | ok... couple of thoughts... first, is your pridialplan set to local? |
| 03:47.39 | asteriskmonkey | no .. i dont have that line at all |
| 03:47.52 | oldave | dunno what the default is... I suspect national |
| 03:48.02 | asteriskmonkey | national is set |
| 03:48.12 | kuku5 | anyone using FOP ? |
| 03:48.13 | oldave | have you done: pri debug span <number> |
| 03:48.34 | asteriskmonkey | my issue is that a number that is out of service i get blank air then bust signal not the appropriate this number is no longer in servies yadaa. |
| 03:48.36 | kuku5 | katty: How often does it crash it ? |
| 03:48.47 | oldave | (and if you're really masochistic: pri intense debug span <number>) |
| 03:49.14 | oldave | you have ,r in the _NXXXXXX dial command? |
| 03:49.15 | Katty | kuku5: whenever i make a change to the config files andsomething is screwed up |
| 03:49.27 | asteriskmonkey | no |
| 03:49.38 | asteriskmonkey | why would i have an r? |
| 03:49.40 | tzanger | now to tackle this stupid MOH |
| 03:49.41 | oldave | tonezone=us in /etc/zaptel.conf? |
| 03:49.44 | *** join/#asterisk graphyx_home (n=mike@c-67-169-246-4.hsd1.ut.comcast.net) |
| 03:49.51 | kuku5 | katty: but other than that its cool? |
| 03:49.54 | oldave | tzanger... good luck... can't make it happen on Solaris 8 :) |
| 03:49.59 | tzanger | no I want it *gone* |
| 03:50.04 | oldave | oooh |
| 03:50.04 | graphyx_home | is asterisk talking to asterisk on SIP the same as a phone talking to a phone? |
| 03:50.09 | graphyx_home | I mean a phone to asterisk? |
| 03:50.14 | graphyx_home | configuration wise |
| 03:50.24 | tzanger | chan_agent needs res_moh |
| 03:50.47 | Katty | kuku5: sure, but i'd recommend not making it interactive |
| 03:50.51 | tzanger | now chan_mgcp |
| 03:50.55 | tzanger | I have a feeling they all need it |
| 03:50.55 | asteriskmonkey | just added tonezone=us |
| 03:51.17 | oldave | monkey... do the pri debug and see what you're getting back on the out of service number |
| 03:51.35 | asteriskmonkey | k |
| 03:51.41 | *** join/#asterisk bmg505 (n=leon@rndf-146-20-160.telkomadsl.co.za) |
| 03:51.58 | docelm0 | I have a really dumb question but I have done everything I can think of.. What would keep DTMF tones from working? |
| 03:52.10 | Nugget | sunspot activity. |
| 03:52.15 | oldave | Planet X |
| 03:52.27 | oldave | what phone? |
| 03:52.29 | *** join/#asterisk justinnn (n=justinnn@61.95.68.85) |
| 03:52.29 | docelm0 | sure.. No seriously.. I am racking my brain here |
| 03:52.30 | oldave | soft phone? |
| 03:52.30 | justinnn | hello ppls |
| 03:52.38 | justinnn | anyone no when cisco 7.6 sip firmware will be released:) ? |
| 03:52.49 | oldave | justinnn... 11/22 |
| 03:52.50 | docelm0 | No.. I am using Soyo -> asterisk -> asterisk -> PSTN |
| 03:53.01 | oldave | Soyo has some interesting issues with levels |
| 03:53.11 | asteriskmonkey | i get this |
| 03:53.13 | asteriskmonkey | NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null |
| 03:53.13 | asteriskmonkey | NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null |
| 03:53.14 | oldave | make sure it's set to inband for DTMF |
| 03:53.25 | [TK]D-Fender | docelm0 : And sound is on through both sides EXCEPT DTMF? |
| 03:53.34 | docelm0 | yep |
| 03:53.35 | justinnn | anyone have any issues with ATA's and faxs |
| 03:53.38 | oldave | monkey... before that, there'll be more info on what the switch sent back |
| 03:53.43 | justinnn | where the fax dies after like 5 pages inbound ??? |
| 03:53.44 | docelm0 | Just DTM is absent.. But it does it on a cisco also.. |
| 03:53.49 | asteriskmonkey | Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] |
| 03:53.49 | docelm0 | this is the buggy part |
| 03:53.53 | *** join/#asterisk zamsler (n=zamsler@c-67-184-232-149.hsd1.il.comcast.net) |
| 03:54.07 | [TK]D-Fender | I've always used RFC2833 for my SIP phones.... |
| 03:54.13 | docelm0 | ya same.. |
| 03:54.23 | docelm0 | and it works.. but this issue is driving me nuts |
| 03:54.29 | justinnn | anyone using faxing / ataing ?? |
| 03:54.33 | docelm0 | Im a dCAP for pete sake |
| 03:54.39 | oldave | monkey... ok, so you got what's expected... that comes in, and * hangs up as requested |
| 03:54.41 | *** join/#asterisk Math` (n=math@modemcable148.4-81-70.mc.videotron.ca) |
| 03:54.52 | asteriskmonkey | oldave: so... why dont i get an appropraite msg instead of dead air then busy? |
| 03:55.15 | asteriskmonkey | where do i set a msg to the * user stating that oh crap that number dont exist like.. the regular pstn spits at us |
| 03:55.15 | oldave | monkey... because the switch said "this isn't a valid number, disconnect" |
| 03:55.27 | oldave | and * did exactly what it was told |
| 03:55.29 | justinnn | hey olddave how did u find out it comes out 11/22 :) ? |
| 03:56.00 | asteriskmonkey | when i call land line i get not in service... where do i set that in asterisk to do the same? or shoudl i say how? |
| 03:56.02 | oldave | monkey... frankly, I'm not sure on that |
| 03:56.24 | oldave | You'll have to do some checking on the call status returned... |
| 03:56.25 | tzanger | asteriskmonkey: with Bell Canada you simply do *NOT* have a matching exten => line |
| 03:56.32 | tzanger | and Bell takes care of it because * returns an invalid * IE |
| 03:56.39 | justinnn | ??? |
| 03:56.40 | tzanger | asteriskmonkey: but if you want to do it yourself |
| 03:56.49 | oldave | justinnn... PFM |
| 03:57.05 | tzanger | exten => i,1,Zapateller |
| 03:57.12 | justinnn | sorry whats pfm mean :) ? |
| 03:57.15 | asteriskmonkey | thanks |
| 03:57.21 | oldave | pure f'in' magic |
| 03:57.24 | tzanger | haha |
| 03:57.29 | asteriskmonkey | lol |
| 03:57.32 | *** join/#asterisk shmooz (n=shmooz@H142.C72.B0.tor.eicat.ca) |
| 03:57.39 | shmooz | yo |
| 03:58.40 | justinnn | ooh ok.. |
| 03:58.42 | justinnn | magic u say |
| 03:58.43 | hhoffman | is libiax still used? it was last updated in 2004 |
| 03:59.02 | hhoffman | or has it all moved under the asterisk tarballs? |
| 04:00.18 | asteriskmonkey | tzanger : i put exten => i,1,Zapateller under my context.. and i got nothing same result? |
| 04:01.08 | tzanger | oh yeah that's right |
| 04:01.10 | *** join/#asterisk hellop (n=hellop@cpe-70-95-165-136.hawaii.res.rr.com) |
| 04:01.13 | tzanger | mark kaiboshed my patch that fixed that |
| 04:01.18 | tzanger | you need to use _X.,1,... |
| 04:01.28 | tzanger | oh for fuck sakes |
| 04:01.31 | hellop | suddenly * won't pick up.. can't dial out. my phone is ringing... |
| 04:01.34 | tzanger | if I can't have MOH I can't have anything apparently |
| 04:01.40 | hellop | tried reloading zap and * |
| 04:01.49 | hellop | sip debug shows nothing |
| 04:01.53 | asteriskmonkey | use a blank mp3 |
| 04:02.10 | hellop | I have a 100p card.. |
| 04:03.09 | asterboy | If the 100p is an Intel v92 winmodem, is it possible to convert other pci modem cards to FXO devices Asterisk can use? |
| 04:03.11 | *** join/#asterisk Peaceful (n=Peaceful@67.50.46.118) |
| 04:03.20 | asteriskmonkey | damnit ... where do i set the thing to happen on non exsisting numbers... its not the exten => i,1,Zapateller cause that checks local digits... the number gets dialed and bell spits back the infor |
| 04:03.31 | asteriskmonkey | so where to i hand the message back tot he client? |
| 04:03.44 | Peaceful | Can asterisk use the modem built-into Apple PowerMac G5's or Apple Powerbooks? |
| 04:04.00 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 04:04.04 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 04:04.06 | TheCops | Hi |
| 04:04.22 | TheCops | Someone is using Rhino Channel banks with FXO ?! |
| 04:05.05 | asterboy | wow, same type of questions all in a row. |
| 04:05.39 | Nugget | asterisk doesn't use modems, Peaceful. |
| 04:05.45 | Nugget | apple modems or otherwise |
| 04:05.54 | asterboy | oh/ |
| 04:06.29 | konfuzed | hm |
| 04:06.42 | asterboy | so apple modems work? |
| 04:06.47 | Nugget | no. |
| 04:06.54 | Nugget | no modems work. asterisk doesn't use modems at all. |
| 04:07.37 | *** join/#asterisk rene- (n=rene-@201.154.240.182) |
| 04:07.41 | *** part/#asterisk rene- (n=rene-@201.154.240.182) |
| 04:08.05 | Nugget | the x100p card (discontinued) is one very specific flavor of one specific softmodem with a specific firmware version and some people have had luck buying cards that are that same firmware, version and pretending that they're x100p cards. |
| 04:08.25 | Nugget | but that shouldn't be extrapolated to mean that "modems" work or that asterisk has any idea about modems at all |
| 04:09.06 | Nugget | and the x100p itself is deprecated by digium and no longer for sale |
| 04:09.55 | *** join/#asterisk PBXtech (i=nik@91.sub-70-213-180.myvzw.com) |
| 04:09.59 | asterboy | Interesting...I'm thinking a driver and patch could be written at a lower level to incorp modems. |
| 04:10.11 | *** join/#asterisk asteriskgeeks (n=SIPdawg@pbxtech.com) |
| 04:10.12 | asteriskgeeks | <PROTECTED> |
| 04:10.26 | Peaceful | Soo, if I'm getting this correctly....I'm confusing "analog interface card" with a modem? |
| 04:10.29 | asterboy | another aster prefixed name! noooooo |
| 04:10.32 | Nugget | Peaceful: yeah |
| 04:12.01 | Nugget | Peaceful: http://www.sipura.com/products/spa3000.htm is a good solution if you need an FXO port. |
| 04:12.26 | Nugget | or you can buy a tdm400p, but I'd recommend the sipura over that, and the sipura will work with a mac. |
| 04:13.00 | Math` | the sipura will work with an embedded coffee machine (with ethernet support) |
| 04:13.03 | TheCops | sipura own |
| 04:13.04 | TheCops | :) |
| 04:13.17 | *** part/#asterisk graphyx_home (n=mike@c-67-169-246-4.hsd1.ut.comcast.net) |
| 04:13.34 | Peaceful | So, if my old modem on my PII-233 under windows could be used to make phone calls with a proprietary Windows app, what's stopping asterisk from doing the same with modern modems?? |
| 04:14.14 | oldave | woohoo... so I can call my toaster from work... |
| 04:14.19 | oldave | I knew this day would finally arrive |
| 04:14.24 | asterboy | its a driver function. |
| 04:14.28 | Nugget | that's a voice modem, which really has not much to do with what asterisk does. |
| 04:14.44 | Nugget | and there's virtually zero interest in the asterisk community in extending asterisk in that direction, I think. |
| 04:15.08 | asterboy | If I was building asterisk with a business model to sell digium equipment...I certainly would not want to provide such access to other hardware. |
| 04:15.28 | Nugget | it's more than just that, though |
| 04:15.44 | Math` | voice modems are.... pretty much shit compared to FXOs :P |
| 04:15.47 | TheCops | always a money history |
| 04:16.18 | Nugget | even the good FXO hardware pretty much sucks, and voice modems are several orders of magnitude worse. |
| 04:16.25 | asterboy | I would agree IF * could interface better with other fax devices. |
| 04:16.51 | Nugget | asterisk isn't all that great unless you're doing non-internet voip traffic or real, honest-to-god pri termination |
| 04:16.51 | Math` | asterboy: T38 is coming :) |
| 04:17.14 | Nugget | there's little motivation to extend asterisk to use hardware that we already know will suck to use |
| 04:17.14 | Math` | a fax protocol |
| 04:17.36 | Math` | "realtime" faxing, as opposed to T30 which is "store & forward" |
| 04:17.40 | asterboy | http://www.vocal.com/data_sheets/t38.html |
| 04:17.49 | asterboy | interesting. |
| 04:17.59 | Math` | many ATAs are supporting T38 |
| 04:18.08 | *** join/#asterisk d-tech (n=dtc@node-423a1ebb.cle.onnet.us.uu.net) |
| 04:18.11 | Math` | allowing you to put a real fax machine connected to an * server |
| 04:18.23 | asterboy | of course this won't be very compatible with older faxes |
| 04:18.32 | Math` | asterboy: oh yes it is |
| 04:18.38 | Nugget | it's like complaining that linux doesn't support the $8 tape drive you bought at hamvention. sure, with enough effort you could make it work, but it's really only of use to people who have exactly $8 worth of data. |
| 04:18.55 | Math` | heh |
| 04:20.13 | asterboy | ya, what Math said |
| 04:20.38 | Peaceful | Fascinating. Well, glad that I just wanted it for grins while learning asterisk at home. |
| 04:20.55 | Nugget | asterisk can be fun even without a connection to your voice line. |
| 04:20.59 | Math` | Peaceful: uhm, you should install a "real" asterisk tho |
| 04:21.08 | Math` | Nugget: thats so true |
| 04:21.11 | Nugget | install it on the powermac and get a voip phone number from asterlink or voicepulse connect. |
| 04:21.26 | Nugget | you'll be out $8 or so and you can play with asterisk to your heart's content |
| 04:21.28 | Math` | I just got a SIP provider for my home line |
| 04:22.02 | oldave | now, now... I've bought plenty of boat anchors at hamfests |
| 04:22.14 | *** join/#asterisk ptiggerdine (n=ptiggerd@c210-49-98-194.rochd1.qld.optusnet.com.au) |
| 04:22.22 | *** join/#asterisk Knight_DKN (n=knight_d@61.95.68.85) |
| 04:22.31 | asterboy | The cheapest way to build an * box is to get clone x100p cards and external ATA for FXS |
| 04:22.45 | asterboy | unless of course you have IP phones. |
| 04:22.54 | *** join/#asterisk hellop (n=hellop@cpe-70-95-165-136.hawaii.res.rr.com) |
| 04:22.57 | asterboy | But then you can connect to a service without * |
| 04:23.03 | Peaceful | It's ok, I've got linux servers and Cisco IP phones to play with at work. |
| 04:25.10 | *** join/#asterisk Ayano (n=erik_lee@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 04:25.22 | Ayano | does the 7910 have a sip image yet? |
| 04:25.58 | Peaceful | Glad to hear that asterisk works well weth PRI termination / LAN VOIP, though, since that's what we're going to be doing. |
| 04:26.11 | Math` | heh |
| 04:26.16 | Math` | that's what I'd use asterisk for too |
| 04:26.20 | Peaceful | Isn't IAX2 between remote sites any good? |
| 04:26.26 | Math` | in fact, thats what I use it for |
| 04:26.30 | Math` | Peaceful: yeah its good |
| 04:27.28 | *** join/#asterisk hhoffman (n=hhoffman@tor/session/x-fc038bd4a259168b) |
| 04:32.16 | ManxPower | To become One with Asterisk, you must work with Asterisk's oddities, not against it, grasshopper. |
| 04:32.45 | Sedorox | yes oh master |
| 04:33.30 | ptiggerdine | and then once inside, change the oddities. |
| 04:33.54 | ManxPower | ptiggerdine, only if you can write decent code 8-) |
| 04:37.03 | asteriskmonkey | is there anyway to make an exten read froma db line? |
| 04:37.16 | Qwell | asteriskmonkey: realtime |
| 04:37.19 | asteriskmonkey | yes |
| 04:37.30 | Qwell | or do you mean a custom query from the dialplan? |
| 04:37.36 | asteriskmonkey | no realtime |
| 04:38.02 | oldave | just today I figured out how to get a DND to work (non Zaptel on the FXS ports, so no, it's not in the channel driver) |
| 04:38.23 | oldave | incidentally, I'm less than a month into this whole * thing |
| 04:38.28 | oldave | learning more every day |
| 04:39.25 | asteriskmonkey | my issue is this i want a customer to be able to update there number forwarding from a webpage so i need the ext to read the number fromt he database |
| 04:39.41 | Qwell | asteriskmonkey: so you do want a custom query |
| 04:41.19 | Qwell | asteriskmonkey: you probably want the Realtime function. show application Realtime |
| 04:43.08 | *** join/#asterisk YoMama (n=tchen@pcp02689850pcs.roylok01.mi.comcast.net) |
| 04:43.40 | X-Files | Me interception of a call what is necessary for this purpose interests? |
| 04:47.14 | YoMama | X-Files: u wanna intercept calls? |
| 04:47.19 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 04:52.17 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-15-19-142.sd.sd.cox.net) |
| 04:53.06 | asteriskmonkey | Qwell: Your application(s) is (are) not registered |
| 04:53.40 | asteriskmonkey | is a documentation on how to do this somewhere? |
| 04:54.02 | Math` | you put the app_something.so file in your module dir |
| 04:54.46 | asteriskmonkey | is there actuallt and app call app_something.so or is it something i have to write ? |
| 04:56.28 | X-Files | YoMama: yes |
| 04:57.09 | asteriskmonkey | realtime is a function of asterisk 1.2 not 1.0.9 :( |
| 04:57.16 | YoMama | X-Files: u could "record" every single conversation to a file and then stream it to a speaker i suppose |
| 04:57.20 | Qwell | asteriskmonkey: upgrade |
| 04:57.27 | Qwell | or you'll have to use an AGI or something |
| 04:57.52 | X-Files | this not actual |
| 04:58.21 | asteriskmonkey | gah... what a pain |
| 04:58.41 | X-Files | YoMama: this is not actual |
| 04:58.48 | asteriskmonkey | is there an agi anyone knows of to this effect? |
| 04:58.54 | *** join/#asterisk Igbothom_III (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) |
| 04:59.33 | Qwell | X-Files: You might want to brush up on your English a little bit... |
| 04:59.57 | X-Files | :)))) |
| 05:03.55 | oldave | somehow, listenin' in on other's conversations seems... well... boring |
| 05:04.25 | Qwell | oldave: even if it's a phonesex line? |
| 05:04.34 | Qwell | okay, that probably would be boring |
| 05:05.20 | oldave | now why would I wanna listen to other people gettin' off? |
| 05:05.41 | oldave | and frankly, phone sex seems silly to me |
| 05:06.43 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 05:08.17 | Math` | uhm if you have sex over phone over ip, it could be abbreviated as Sex over IP or SoIP |
| 05:08.31 | Math` | and I hope you don't get packet loss |
| 05:08.36 | ptiggerdine | ROFL LMAO |
| 05:08.37 | ptiggerdine | !!! |
| 05:09.17 | Math` | could support features such as MAD (Movement Activity Detection) |
| 05:10.51 | Flauto | what is agi.pm? |
| 05:11.05 | Math` | Asterisk::AGI perl module I guess |
| 05:11.23 | oldave | A Girl In: Period Monthly? |
| 05:11.28 | Flauto | is there anywhere that i can get it? |
| 05:12.03 | oldave | the girl or the Perl module? |
| 05:12.46 | hypa7ia | psysterisk? |
| 05:13.10 | Flauto | oldave, hehe, the module |
| 05:13.11 | hardwire | the zaptel source really needs a better make system |
| 05:13.35 | Math` | Flauto: should've answered both |
| 05:13.55 | oldave | gotta give the shy ones a chance |
| 05:14.45 | *** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au) |
| 05:15.05 | oldave | y'know what's sad? I have a local # (Broadvoice), a UK number, a German number and a Toronto number... along with the FWD number... and nobody calls |
| 05:15.06 | oldave | heh |
| 05:16.00 | Flauto | hehe |
| 05:16.24 | Flauto | i can call you. oldave |
| 05:16.28 | oldave | good thing I'm only spending $$ on the Broadvoice |
| 05:16.37 | oldave | actually, I have all those numbers for my 'net radio station |
| 05:16.49 | Flauto | oldave, are you using vbuzzer? |
| 05:16.50 | oldave | which simply says to me that nobody's listening |
| 05:17.03 | oldave | Flauto... yeah... never received a call on it... and it keeps lagging out |
| 05:17.21 | oldave | Nov 15 23:48:38 NOTICE[23629]: chan_sip.c:9688 handle_response_peerpoke: Peer 'vbuzzer' is now TOO LAGGED! (2570ms / 200ms) |
| 05:17.21 | oldave | Nov 15 23:49:01 NOTICE[23629]: chan_sip.c:9682 handle_response_peerpoke: Peer 'vbuzzer' is now REACHABLE! (47ms / 200ms) |
| 05:17.39 | Flauto | have your tired if it can recevie or call out at all? oldave |
| 05:17.50 | oldave | nope, never tried it |
| 05:17.51 | hardwire | oldave: pathetic :) |
| 05:17.57 | Flauto | hehe |
| 05:18.03 | Flauto | i have the same problem |
| 05:18.12 | Flauto | never can get calls from vbuzzer |
| 05:18.22 | oldave | I never tried |
| 05:18.25 | oldave | no clue if it works |
| 05:18.32 | Flauto | give me that number |
| 05:18.36 | Flauto | i will cal you now |
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| 05:19.55 | oldave | 416-273-1924 |
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| 05:20.47 | Math` | oldave: you used * with vbuzzer? |
| 05:21.22 | oldave | sure |
| 05:21.46 | oldave | why not? It's a SIP system |
| 05:22.06 | oldave | doing IAX2 with FWD |
| 05:22.33 | Flauto | same here |
| 05:22.35 | oldave | I'm pretty sure that works... |
| 05:22.37 | Flauto | wait |
| 05:22.45 | oldave | outbound from FWD to my Broadvoice worked |
| 05:22.51 | oldave | haven't tried it the other way |
| 05:24.29 | wasim | hahah!! 3 wickets in 2 overs |
| 05:24.37 | oldave | actually, I just did outbound on Broadvoice to the vbuzzer number... went through fine |
| 05:24.58 | oldave | so yes, * works fine with vbuzzer |
| 05:26.26 | Flauto | really |
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| 05:26.28 | YoMama | blah |
| 05:26.28 | Flauto | great |
| 05:26.38 | Math` | oldave: yeah its unencrypted SIP authentication so you can config it :) |
| 05:26.46 | Flauto | would you show me the setting for vbuzzer |
| 05:26.57 | YoMama | Flauto: i figured it otu :) |
| 05:26.59 | YoMama | out |
| 05:27.06 | YoMama | Flauto: I'm using vbuzzer with asterisk |
| 05:27.11 | Flauto | yomama would youuu please |
| 05:27.21 | YoMama | Flauto: sure...lemme look up the config..wait a sec |
| 05:27.29 | Flauto | great |
| 05:27.31 | Flauto | thanks |
| 05:27.44 | YoMama | np |
| 05:27.48 | YoMama | i'll put it on pastbin for ya |
| 05:27.53 | oldave | prolly gonna flood me out |
| 05:27.56 | oldave | but here |
| 05:28.00 | oldave | [vbuzzer] |
| 05:28.00 | oldave | type=peer |
| 05:28.00 | oldave | host=vbuzzer.com |
| 05:28.01 | oldave | port=80 |
| 05:28.01 | oldave | nat=yes |
| 05:28.01 | oldave | username=<username> |
| 05:28.03 | oldave | secret=<password> |
| 05:28.05 | oldave | fromuser=<username> |
| 05:28.07 | oldave | restrictcid=yes |
| 05:28.09 | oldave | fromdomain=vbuzzer.com |
| 05:28.11 | oldave | context=vbuzzer |
| 05:28.13 | oldave | insecure=very |
| 05:28.15 | oldave | dtmfmode=rfc2833 |
| 05:28.17 | oldave | disallow=all |
| 05:28.19 | oldave | allow=gsm |
| 05:28.22 | oldave | allow=ulaw |
| 05:28.23 | oldave | allow=alaw |
| 05:28.25 | oldave | qualify=200 |
| 05:28.28 | oldave | useragent=vbuzzer/1.0 |
| 05:28.31 | oldave | and on that note, I will wish you all a good evening... bedtime for Bonzo... and I'm gonna head that way myself |
| 05:28.32 | wasim | jeesux |
| 05:28.37 | wasim | err sus |
| 05:28.43 | Flauto | what about register |
| 05:28.48 | Flauto | and exteions.conf |
| 05:28.48 | YoMama | Math`: i'm gonna just buy one of these grandstream things if i don't get the mediatrix |
| 05:28.53 | YoMama | Flauto: yeah..he forgot some shit |
| 05:28.56 | YoMama | and u don't need all that |
| 05:28.57 | YoMama | sheesh |
| 05:29.12 | YoMama | Flauto: i'll put exactly what works on pastebin for ya |
| 05:29.15 | YoMama | hold on man |
| 05:29.18 | Math` | YoMama: well I'll be getting those if you don't |
| 05:29.26 | Qwell | ~pb |
| 05:29.27 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 05:29.32 | Qwell | oh, he left...meh, wtf |
| 05:29.40 | YoMama | Math`: it supports 802.11q? |
| 05:29.45 | YoMama | err..802.1Q |
| 05:29.47 | Math` | yeah it does |
| 05:29.53 | YoMama | sweet |
| 05:29.53 | Qwell | gs ata? |
| 05:29.55 | Math` | I've an snmp option for it |
| 05:32.28 | Flauto | have you posted? yomama |
| 05:32.54 | YoMama | Flauto: go and get it -> http://pastebin.ca/28872 |
| 05:33.00 | YoMama | Flauto: and have fun :) |
| 05:33.38 | YoMama | Flauto: i haven't tried makign multiple calls at once..but i wouldn't recommend it or they might be on to us not using their client :) |
| 05:35.28 | konfuzed | yeh |
| 05:35.49 | konfuzed | i got the two ht486s working on lan side of vonage gateway |
| 05:36.28 | konfuzed | didnt even touch the vonage linksys rtp300 |
| 05:37.39 | BleedingMe | was the LEN function removed from CVS HEAD... or did something change with it's parameters? Anybody know? |
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| 05:40.12 | YoMama | Flauto: is it working? |
| 05:42.31 | *** join/#asterisk implicit (n=implicit@216.13.124.132) |
| 05:42.40 | Flauto | not yet |
| 05:42.43 | Flauto | let me try |
| 05:43.21 | Flauto | context=outgoing? |
| 05:43.48 | Flauto | where do set for incoming call |
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| 05:44.22 | Flauto | yomama |
| 05:44.35 | Flauto | would you give me your number so i can try to call? |
| 05:45.53 | YoMama | Flauto: just call your cell phone |
| 05:46.09 | Flauto | i am not in toronto |
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| 05:49.56 | joelsolanki | Hi all, anybody used dlink dph 70 dialup box ? i need to disable silence suppression ..i dont find any option :( |
| 05:50.19 | Flauto | yomama, what do you do for your incoming call |
| 05:50.32 | YoMama | incoming? i use a regular phoneline |
| 05:50.38 | Flauto | no |
| 05:50.40 | YoMama | i set mine up so it only does outbound with vbuzzer |
| 05:50.47 | Flauto | i mean vbuzzer |
| 05:50.49 | Flauto | oh |
| 05:50.57 | YoMama | Flauto: if you want to accept vbuzzer inbound..then set up another one where it's of type user instead |
| 05:51.04 | joelsolanki | anybody has any idea? |
| 05:51.08 | Flauto | you don't use inbound |
| 05:51.08 | YoMama | so then when someone calls your vbuzzer #..u can have it connecte dto an extension |
| 05:51.09 | Flauto | okay |
| 05:51.12 | Flauto | let me try |
| 05:51.41 | Flauto | is 1800 numbers free to vbuzzer? |
| 05:52.15 | wasim | hehe ... its fun to see pk sysadmins try to bypass PTCL port block on 5060 |
| 05:56.16 | Damin | Hehehehe.. |
| 05:56.25 | Flauto | yomama, do i have to open ports for vbuzzer? |
| 05:56.43 | Damin | Despite the lack of a manual with this Audiocodes gateway, I have mastered it's intricacies and have beaten it into submission! |
| 05:56.51 | Damin | ALL YOUR FXS ARE BELONG TO US! |
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| 06:01.38 | *** join/#asterisk [MUPPETS]Gonzo (i=gonzo@80.69.47.16) [NETSPLIT VICTIM] |
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| 06:03.25 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 06:03.30 | wasim | haha! 6 wickets down! |
| 06:03.33 | Flauto | yomama, it seems the phone would ring when i call out |
| 06:03.37 | Flauto | but no audio |
| 06:10.05 | *** join/#asterisk fragmastr (n=fragmast@202.83.81.122) |
| 06:18.25 | *** join/#asterisk YoMama (n=tchen@pcp02689850pcs.roylok01.mi.comcast.net) [NETSPLIT VICTIM] |
| 06:18.34 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 06:18.36 | YoMama | argh |
| 06:19.33 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:22.44 | Flauto | i dont' know what is wrong |
| 06:22.48 | Flauto | i called to my cell phone |
| 06:22.54 | Flauto | my cell phone was ringing |
| 06:23.06 | Flauto | but i can not hear anything from my cell phone |
| 06:23.15 | Flauto | also, inbound is not working |
| 06:23.50 | YoMama | Flauto: hmm...well, i called several people using that setup i put on pastebin |
| 06:24.31 | Flauto | did you need to open port 443? |
| 06:25.00 | YoMama | Flauto: so u can't hear voice on which end? |
| 06:25.13 | Flauto | on my vbuzzer end |
| 06:25.31 | Flauto | i can hear from cell |
| 06:25.38 | YoMama | so u can hear the voice on your cell |
| 06:25.47 | Flauto | yes |
| 06:25.53 | YoMama | but when u talk on your cell..u can't hear it |
| 06:26.01 | YoMama | is your asterisk server behind a firewall? |
| 06:26.24 | Flauto | no |
| 06:26.35 | YoMama | umm |
| 06:26.41 | Flauto | i put the linux machine in front of the firewall already |
| 06:26.44 | YoMama | u got your linux box blocking certain ports? |
| 06:26.46 | *** join/#asterisk johnrage (n=jabetong@212.93.201.89) |
| 06:26.52 | Flauto | no |
| 06:26.56 | Flauto | opened everything |
| 06:27.05 | YoMama | well, that's kinda dangerous |
| 06:27.12 | YoMama | umm |
| 06:27.13 | Flauto | i know |
| 06:27.17 | Flauto | i am just trying for now |
| 06:27.19 | YoMama | i dunno...works for me |
| 06:27.26 | YoMama | the way i put it up there |
| 06:27.33 | ptiggerdine | wasim, the poms will make 87 is win |
| 06:27.39 | Flauto | would you let me to call you on your vbuzzer number? |
| 06:27.40 | johnrage | Hello Asterisk guru, am always pasting this inquiry. I need Phil and India DID |
| 06:27.53 | YoMama | Flauto: i don't have inbound set up |
| 06:27.53 | Flauto | oh, you dont' use inbound |
| 06:27.56 | YoMama | right |
| 06:27.58 | Flauto | i forgot |
| 06:27.59 | Flauto | sorry |
| 06:28.16 | *** join/#asterisk bassig (n=track_ui@75.179.uio.satnet.net) |
| 06:28.18 | ptiggerdine | Flauto, what's te pastebin url again? |
| 06:28.22 | Uberbot | Can anyone tell me why this doesn't work if the variable "number" has spaces in it? exten => s,3,setcidnum(${number}) |
| 06:28.42 | Flauto | http://pastebin.ca/28872 |
| 06:29.18 | *** join/#asterisk axscode (n=paranoid@203.213.217.123) |
| 06:29.56 | Flauto | yomama, would you try to call my vbuzzer number then? |
| 06:30.14 | wasim | ptiggerdine: hahaha!! go shoaib! |
| 06:30.17 | *** join/#asterisk sbingner (n=thanotos@pdpc/supporter/sustaining/sbingner) |
| 06:31.09 | wasim | we should wrap 'em up before lunch |
| 06:31.33 | axscode | anyone running asterisk on CentOS? |
| 06:32.22 | justinu | lots of people run asterisk on centos |
| 06:32.31 | ptiggerdine | wasim, u wish! |
| 06:33.08 | *** join/#asterisk hellop (n=hellop@cpe-70-95-165-136.hawaii.res.rr.com) |
| 06:33.55 | hellop | Any idea why, suddenly, when I dial nothing happens? I see "Executing Dial..." in the CLI, but then nothing. How can I diagnose this? |
| 06:34.13 | hellop | It's like the Zap card is broken. |
| 06:34.50 | wasim | ptiggerdine: as a consolation, udal scored a run :P |
| 06:35.28 | ptiggerdine | kewl, I'm seening web-updates so I can "really follow" the match. |
| 06:36.09 | hellop | I go to voicemail, my phone is sending data to server, but server not responding.. |
| 06:36.50 | wasim | ptiggerdine: i'll update you as soon as the next wicket falls |
| 06:36.54 | hellop | welp, third time rebooting * fixed it.. |
| 06:37.00 | justinu | <PROTECTED> |
| 06:37.11 | justinu | nothing like reliable telco gear :P |
| 06:37.38 | wasim | hellop: rebooting just * or the entire box? |
| 06:37.43 | hellop | hey man, it's not the 80's anymore |
| 06:38.05 | hellop | wasim just * I dunno why woking one sec, then nada |
| 06:38.39 | hellop | oh, now it's not working again.. |
| 06:39.14 | *** part/#asterisk bassig (n=track_ui@75.179.uio.satnet.net) |
| 06:39.44 | hellop | If I go to voicemail, I hear nothing, but can access the menus. |
| 06:39.48 | justinu | what causes ethernet errors in ifconfig |
| 06:40.45 | wasim | bad cable, cheap switches, 220v on the wire |
| 06:40.50 | hellop | bugs |
| 06:40.58 | hellop | geckos |
| 06:41.19 | YoMama | and do a: sip show registry |
| 06:41.22 | YoMama | oops |
| 06:43.05 | *** join/#asterisk dippo (n=cwage@quietlife.net) |
| 06:44.01 | justinu | any idea what the frame: stat means in ifconfig? |
| 06:44.10 | hellop | wonder why my * is breaking all a sudden... |
| 06:47.03 | JunK-Y | some can help me for a cool app? |
| 06:47.15 | justinu | what cool app? |
| 06:51.05 | JunK-Y | the voicechanger patched. |
| 06:51.31 | Qwell | JunK-Y: I kinda wanted to set that up too...haven't had time |
| 06:52.15 | JunK-Y | Qwell: want a live demo? |
| 06:52.19 | YoMama | why the hell can't i get ignorepat to work properly? |
| 06:52.55 | FuriousGeorge | what you are hearing is a C flat on your cello |
| 06:52.58 | FuriousGeorge | not dialtone |
| 06:53.03 | drumkilla | YoMama: because you're probably not using it the way that it is supposed to be used :) |
| 06:53.14 | Qwell | JunK-Y: another time |
| 06:53.15 | JunK-Y | drumkilla: have u tried that app? |
| 06:53.17 | drumkilla | JunK-Y: nope |
| 06:53.19 | YoMama | drumkilla: ok..so how am i supposed to use it |
| 06:53.21 | JunK-Y | thats fucking great. |
| 06:53.28 | drumkilla | YoMama: are you using it with Zap? |
| 06:53.37 | drumkilla | I think that's the only channel that does anything with it |
| 06:55.35 | *** join/#asterisk snitt (i=snitt@snitt.info) |
| 06:56.50 | drumkilla | YoMama: ignorepat doesn't strip anything from the extension, just instructs the channel driver to not break dialtone on that pattern |
| 06:57.11 | drumkilla | you probably want ${EXTEN:1} or something ... |
| 06:57.51 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:57.52 | YoMama | drumkilla: yeah..no |
| 06:57.54 | YoMama | drumkilla |
| 06:58.02 | YoMama | drumkilla: does x-lite pay attention to it? |
| 06:58.14 | YoMama | drumkilla: maybe that's the problem...when i dial 9..dialtone stops on x-lite |
| 06:58.19 | YoMama | it's probably just the damn client |
| 06:58.48 | drumkilla | no, it has no effect on a SIP channel. |
| 07:00.29 | YoMama | drumkilla: oh...duh |
| 07:00.40 | Math` | YoMama: the dialtone is handled by x-lite |
| 07:00.56 | YoMama | Math`: stupid thing |
| 07:01.04 | Math` | it just sends out the SIP message after you hit send, or after a timeout |
| 07:01.09 | Math` | YoMama: all ATAs work like that too |
| 07:01.27 | YoMama | Math`: yeah..'cause it's SIp..not an FXS |
| 07:01.59 | drumkilla | IAX has support for sending partial numbers, but SIP does not |
| 07:02.00 | Math` | then why do you say its a stupid thing? :P |
| 07:02.01 | drumkilla | :-p |
| 07:02.11 | Math` | drumkilla: iax has that? nice feature |
| 07:02.19 | Math` | you can match the dialplan without config'ing the ata |
| 07:02.28 | drumkilla | Math`: exactly |
| 07:02.34 | drumkilla | and that's how the IAXy works |
| 07:02.43 | Math` | oh |
| 07:02.45 | drumkilla | there is no dtmf timeout |
| 07:02.53 | drumkilla | it's all handled by the server |
| 07:02.55 | Math` | I heard the 1st version was pretty shit... you've tried the last one? |
| 07:03.05 | drumkilla | yes, I have. |
| 07:03.14 | drumkilla | I work for Digium :) |
| 07:03.27 | YoMama | drumkilla: get me a free TDM400 :) |
| 07:03.29 | YoMama | j/k |
| 07:03.32 | drumkilla | heh |
| 07:03.49 | YoMama | drumkilla: my friend gave me his devkitlite from a ways back |
| 07:03.53 | YoMama | the S100U sucks my butt |
| 07:04.48 | YoMama | Math`: on the mediatrixes..how do u tell the damn thing you're done dialing? u dont' haveta hit # do you? |
| 07:04.57 | YoMama | on x-lite..u gotta hit enter |
| 07:05.16 | Math` | YoMama: you can config a digitmap |
| 07:05.27 | YoMama | k |
| 07:05.42 | YoMama | does x-lite have the ability to know when u got voicemail? |
| 07:05.50 | Math` | yeah |
| 07:06.00 | Math` | uh I think so |
| 07:06.03 | Math` | eyeBeam does |
| 07:08.49 | Callum | drumkilla, what do you do @ Digium ? |
| 07:10.12 | Qwell | JunK-Y: I'm not sure I like the way that guy went about the voicechanger stuff. |
| 07:10.27 | drumkilla | Callum: part-time software development |
| 07:10.31 | drumkilla | I'm also a full-time student |
| 07:10.33 | Qwell | would be a hell of a lot more useful if it was an app that didn't actually dial... |
| 07:10.58 | Qwell | drumkilla: so, when'd that happened? |
| 07:11.18 | JunK-Y | Qwell: huh? |
| 07:11.20 | drumkilla | Qwell: well, I have been doing some stuff off and on since January, actually |
| 07:11.29 | drumkilla | Qwell: worked there full-time this past summer |
| 07:11.32 | Qwell | ahh |
| 07:11.40 | *** join/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 07:11.52 | drumkilla | i still do plenty of stuff on my own time ... |
| 07:12.01 | wasim | Qwell: thats why we need app_jack |
| 07:12.07 | Qwell | JunK-Y: something like 5551212,1,VoiceChange(-2) 5551212,2,Dial(IAX2/blah) |
| 07:12.16 | Qwell | wasim: something like that |
| 07:12.25 | JunK-Y | Qwell: my patch change the voice live. |
| 07:12.32 | Qwell | JunK-Y: oh? |
| 07:12.35 | JunK-Y | ;) |
| 07:12.41 | Qwell | well why didn't you say so?! :P |
| 07:12.44 | JunK-Y | hehhe |
| 07:12.58 | YoMama | voicechange? |
| 07:15.28 | Math` | YoMama: changes the pitch |
| 07:15.39 | Math` | some people are having fun :P |
| 07:16.30 | Math` | hooked up a mic to my mixer when I got my dj gear and a lot of my friend were having fun using the pitch bend effect |
| 07:18.58 | JunK-Y | Qwell: sounds great? |
| 07:19.06 | Qwell | JunK-Y: much better than the original |
| 07:19.24 | Math` | JunK-Y: now implement a flanger :P |
| 07:19.28 | Qwell | gonna contrib your changes back to him? |
| 07:19.39 | JunK-Y | sure |
| 07:19.41 | JunK-Y | app_voicechanger.c Revision: 1.01 |
| 07:19.41 | Qwell | would be great to see that as part of it too |
| 07:19.48 | konfuzed | oh yeah these atas do the trick |
| 07:19.54 | *** join/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr) |
| 07:19.55 | wasim | see, if we have app_jack, then we have all the plugins in the world |
| 07:19.56 | JunK-Y | ive added stuff for Rev. |
| 07:20.08 | konfuzed | my buddy is dieing to get one now |
| 07:20.19 | Math` | or... what about an xmms visualization plugin support for voice, thats even more unuseful |
| 07:20.30 | lme | yop all |
| 07:20.32 | JunK-Y | Math`: just need something to that app to be awesome. |
| 07:20.40 | *** join/#asterisk lubomier (n=lubomier@217.118.109.179) |
| 07:20.42 | Math` | JunK-Y: what? :P |
| 07:20.54 | JunK-Y | passing dtmf. |
| 07:21.06 | Math` | send it out of band |
| 07:21.19 | lubomier | hi guys, i compiled from sources asterisk, but i haven't socket file /var/run/asterisk.ctl how it is possible? or, how do I create it? |
| 07:21.31 | Math` | asterisk isn't running? |
| 07:21.33 | Qwell | JunK-Y: dtmf gets pitch shifted too? |
| 07:21.40 | lubomier | no |
| 07:21.49 | Math` | then start it |
| 07:21.59 | JunK-Y | Qwell: with my patch, * is to lower, # is to higher the voice. |
| 07:22.08 | lubomier | [root@*@~.P4_fritz]:(/etc/asterisk)$ asterisk -vvc |
| 07:22.08 | lubomier | Segmentation fault |
| 07:22.17 | Math` | uh oh |
| 07:22.21 | Math` | what version's that |
| 07:22.23 | Qwell | lubomier: add a g, then run gdb on the core |
| 07:22.24 | lubomier | strace shows this |
| 07:22.25 | lubomier | connect(3, {sa_family=AF_FILE, path="/var/run/asterisk/asterisk.ctl"}, 110) = -1 ENOENT (No such file or directory) |
| 07:22.34 | JunK-Y | lubomier: read README.backtrace and provide a backtrace |
| 07:22.43 | Math` | lubomier: it wants to see if another asterisk is running |
| 07:22.49 | *** part/#asterisk bartpbx (n=bartpbx@p54B0360E.dip0.t-ipconnect.de) |
| 07:23.03 | Math` | gdb asterisk |
| 07:23.07 | Math` | run -vvvvvvvvvvvvgc |
| 07:23.29 | lubomier | still segfaulting ;[ |
| 07:23.37 | Math` | thats the point |
| 07:23.38 | Qwell | you want it to segfault in gdb |
| 07:23.44 | Math` | we wanna know where the segfault is |
| 07:23.45 | JunK-Y | gdb -se "asterisk" -c /tmp/core.xyz |
| 07:23.53 | JunK-Y | can i login to that machine? |
| 07:24.12 | lubomier | it is possible, can I trust you? |
| 07:24.14 | lubomier | ;> |
| 07:24.20 | wasim | yep |
| 07:24.21 | JunK-Y | no, im in prison. |
| 07:24.34 | JunK-Y | shit, 2:24 ive to wake up in 4 hours |
| 07:24.51 | Math` | where do u work? |
| 07:25.00 | lubomier | :} |
| 07:25.09 | wasim | Math`: at the prison laundry |
| 07:25.35 | JunK-Y | bingo. |
| 07:25.36 | JunK-Y | :) |
| 07:25.38 | Math` | lol |
| 07:29.15 | YoMama | Math`: is eyebeam better? |
| 07:29.31 | Qwell | YoMama: eyebeam doesn't completely suck |
| 07:29.46 | YoMama | is it better than x-lite? |
| 07:29.51 | Math` | YoMama: well... http://www.xten.com/index.php?menu=eyeBeam |
| 07:29.59 | Qwell | well, x-lite does completely suck, so, yes |
| 07:30.09 | Math` | lol |
| 07:30.31 | JunK-Y | Math`: go get a polycom bro. |
| 07:30.35 | LostFrog | ok.. I must be dumb, I can't get auto-dial to work. |
| 07:30.39 | Math` | the only thing I don't like about eyebeam is.... when SIP reg fail, it doesnt retry! |
| 07:30.56 | Math` | JunK-Y: maybe one day heh |
| 07:31.25 | JunK-Y | go get a PAP2-CA in ur wait time. |
| 07:31.30 | LostFrog | I tried with both an Application and a Context/Extensio/Priority |
| 07:31.33 | YoMama | oh u gotta buy it |
| 07:31.37 | JunK-Y | fucking cheap, work not so bad so far. |
| 07:31.49 | Math` | JunK-Y: a pap2 is an ata... |
| 07:32.08 | JunK-Y | will be much better then ur current mic. |
| 07:32.29 | Math` | I don't use a mic |
| 07:32.33 | Math` | I use a 2 port FXS |
| 07:32.41 | YoMama | wow..my wireless at my house blows ass |
| 07:32.47 | YoMama | i'll brb..i'm gonna use my desktop machine |
| 07:32.52 | *** part/#asterisk YoMama (n=tchen@pcp02689850pcs.roylok01.mi.comcast.net) |
| 07:33.15 | orlock | Hmm. |
| 07:33.21 | Math` | Juggie: actually I use a mediatrix 2102 |
| 07:33.26 | orlock | goddamn op in #php is a scientologist i think |
| 07:33.43 | orlock | he's kicking people for reasons that only a scientologist would care about |
| 07:33.51 | Math` | like? |
| 07:34.03 | orlock | having the nick "clambaker" |
| 07:34.07 | Dr_Ray | irc channels are personality cults |
| 07:34.12 | orlock | pointing a "thetan ray" at preople |
| 07:34.19 | orlock | yeah, i know |
| 07:34.44 | Math` | some people feel powerful of being a channel operator |
| 07:34.54 | Dr_Ray | or webforum queen bee |
| 07:34.54 | orlock | i know :) |
| 07:35.07 | orlock | yup |
| 07:36.15 | Dr_Ray | it makes me cheer the trolls and channel take over artists |
| 07:37.56 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
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| 07:44.17 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 07:44.43 | puzzled | morning all |
| 07:45.52 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 07:46.00 | *** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg) |
| 07:53.59 | *** join/#asterisk YoMama (n=tchen@pcp02689850pcs.roylok01.mi.comcast.net) |
| 07:55.33 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
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| 07:57.01 | yxa | anyone using dundi? |
| 07:58.45 | Math` | yxa: teag |
| 07:58.50 | Math` | uh, that meant "yeah" |
| 07:59.10 | Math` | (right hand not aligned) |
| 07:59.33 | *** join/#asterisk Pikoro (n=pikoro@db.sunny-net.ne.jp) |
| 07:59.37 | yxa | Math` don't quite understand it |
| 07:59.51 | Callum | c0w, are you there ? |
| 08:00.24 | Pikoro | is there a walkthrough somewhere on how to set up a TDM400P for just incoming calls? |
| 08:00.47 | yxa | Math` if i have 5 * servers in my org serving different departments, should i use dundi? |
| 08:00.58 | Pikoro | all i get when i do a zap show channels is: pseudo from-outside |
| 08:01.39 | Pikoro | the kernel module is loaded and /etc/zaptel.conf seems to be configured correctly... |
| 08:03.56 | yxa | Pikoro how abt your zapata.conf? |
| 08:04.20 | Pikoro | well... i figure its not :D |
| 08:05.55 | Pikoro | i have czeched it out, and as near as I can figure, it _seems_ to be correct... |
| 08:06.10 | Pikoro | but i don't receive anything on the console when i try to dial in |
| 08:07.00 | yxa | Pikoro if zap show channels doesnt show them, you wont be able to do anything |
| 08:07.10 | Pikoro | yah, that's what i figured... |
| 08:07.35 | yxa | did you compile asterisk after your compile and install zaptel? |
| 08:07.37 | Pikoro | i was just looking for a sample zapta.conf for that card.. 4 fxo ports |
| 08:07.43 | Pikoro | yes, the module is loaded |
| 08:07.48 | Pikoro | and the kernel module is loaded |
| 08:07.56 | yxa | ztcfg -vv? |
| 08:08.16 | Pikoro | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 08:08.16 | Pikoro | Channel 02: FXS Kewlstart (Default) (Slaves: 02) |
| 08:08.16 | Pikoro | Channel 03: FXS Kewlstart (Default) (Slaves: 03) |
| 08:08.16 | Pikoro | Channel 04: FXS Kewlstart (Default) (Slaves: 04) |
| 08:08.41 | yxa | any alarms in zttool? |
| 08:09.32 | yxa | if not i'm pretty sure its your zapata.conf |
| 08:09.44 | Pikoro | let me czech |
| 08:10.28 | Pikoro | no alarms |
| 08:12.22 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
| 08:13.33 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 08:14.46 | yxa | might be IRQ assignment but my experience is that they are pretty tolerent. |
| 08:15.16 | Pikoro | yah, everything seems to be working |
| 08:15.28 | Pikoro | but no zap channels |
| 08:16.48 | yxa | Pikoro pastebin your zapata.conf |
| 08:16.55 | Pikoro | k |
| 08:23.24 | *** join/#asterisk mcn (n=mcn@ext-gw.newtoncomputing.co.uk) |
| 08:23.29 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:23.39 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 08:24.58 | Telamon | Does anyone have any oppinions on the gnet vp104 series of phones? I'm looking for an IAX alternative to the Grandstream 2000 just to test things out. I hear the VP320 isn't very good, but nothing about the 104. |
| 08:27.00 | *** join/#asterisk Math[laptop] (n=math@modemcable148.4-81-70.mc.videotron.ca) |
| 08:27.33 | Pikoro | http://pastebin.ca/28883 |
| 08:27.39 | Pikoro | sorry, had to cut out all the comments |
| 08:27.46 | Pikoro | basically it's a default zapta.conf |
| 08:29.53 | yxa | Pikoro you dont have your channel => 1-4 declaration and grouping (if needed)? |
| 08:30.08 | Pikoro | that's in /etc/zaptel.conf correct? |
| 08:30.27 | yxa | and zapata.conf :) |
| 08:30.30 | YoMama | la de da |
| 08:30.44 | Pikoro | fxsks=1-4 |
| 08:30.49 | Pikoro | that's in zaptel.conf |
| 08:31.05 | Pikoro | shouldn't that be fxoks=1-4? |
| 08:31.40 | yxa | just add this line: channel => 1-4 |
| 08:31.48 | Pikoro | to zapta or zaptel? |
| 08:31.52 | yxa | zapata |
| 08:32.09 | Pikoro | ok, and do a reload? |
| 08:32.12 | yxa | yep |
| 08:32.32 | Igbothom_III | Telamon; have a look at the http://iaxtalk.com/index.php?main_page=product_info&products_id=2&zenid=2e4c4fce1ce10b4a609452ecf28a0b54 <-- not used them myself, hoping to test one soon |
| 08:32.45 | Pikoro | i get the same thing... pseudo from-outside |
| 08:33.00 | Math[laptop] | is realtime pretty stable? |
| 08:36.29 | Telamon | Igbothom_III: Yeah, I can find where to buy them, I just don't want to waste hundreds of bucks on a phone and find out it's a cheap piece of garbage. Can't seem to find any actual *reviews* of any of the IAX phones though. |
| 08:37.49 | Vhata | where does 'make progdocs' put the stuff it creates/installs? |
| 08:38.02 | *** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg) |
| 08:38.10 | *** join/#asterisk JimmyCarter (n=del@213083175015.sonofon.dk) |
| 08:38.38 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
| 08:41.41 | mmmToop | guys...any ideas if there are limits to meetme? |
| 08:42.05 | mmmToop | just got a request for a 1500 + conference call facility... |
| 08:42.18 | YoMama | anyone know of a way for mpg123 to read a shoutcast stream? |
| 08:42.38 | puzzled | mmmToop: depends on the amount of transcoding and the specs of the box. dunno about 1500+ though |
| 08:42.54 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
| 08:43.01 | puzzled | YoMama: read the musiconhold.conf file in /etc/asterisk |
| 08:43.26 | mmmToop | throw a couple of quad Xeons at it ; ) |
| 08:43.44 | puzzled | I think you will be better of with operterons |
| 08:43.57 | mmmToop | sure... |
| 08:44.05 | puzzled | and save a villages of powerusage at the same time :) |
| 08:44.11 | mmmToop | I know junipernetworks does it.. |
| 08:44.12 | puzzled | s/a/a few/ |
| 08:44.27 | puzzled | with asterisk? |
| 08:44.58 | mmmToop | no they offer a conference call services |
| 08:45.01 | mmmToop | for big setups |
| 08:45.33 | puzzled | ah yes, AT&T does that too. no idea what they use though. prolly Lucent 5E with some additional kit |
| 08:45.41 | Telamon | mmmToop: I think you'd probably want to use multiple servers with peering. I've got no idea how you'd do the software-based load ballancing though. |
| 08:46.11 | mmmToop | sure...as "one" machine has to do the transcoding... |
| 08:46.16 | puzzled | and sync the conferences from multiple servers to appear as one without the possible delays... |
| 08:46.20 | YoMama | puzzled: stramplayer eh? |
| 08:46.20 | mmmToop | I would have no idea how to distribute that load. |
| 08:46.30 | YoMama | streamplayer |
| 08:46.35 | mmmToop | yes...timing would be an issue |
| 08:46.37 | puzzled | YoMama: there is some stuff in there how to stream stuff |
| 08:46.47 | YoMama | puzzled: not that i can see |
| 08:47.28 | puzzled | YoMama: I'm using 1.2.0-rc2 |
| 08:47.37 | YoMama | i'm still using rc1 |
| 08:48.29 | mmmToop | didn't even know r2 was out... |
| 08:49.40 | asterboy | You joined: #asterisk |
| 08:49.40 | asterboy | (01:42) ›› Topic: Asterisk 1.2.0 RC2 has been released! Please try it out! || http://www.asterisk.org |
| 08:49.50 | asterboy | ;) |
| 08:50.22 | mmmToop | oops...shows how often I read the Topic! ; ) |
| 08:51.22 | YoMama | ha |
| 08:51.31 | asterboy | ya, I'm bad for that too...that's why I get those pesky parking tickets. |
| 08:56.47 | *** part/#asterisk lubomier (n=lubomier@217.118.109.179) |
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| 09:23.23 | *** join/#asterisk d-tech (n=dtc@node-423a1ebb.cle.onnet.us.uu.net) |
| 09:28.13 | *** join/#asterisk Tili (n=Tili@213.132.60.182) |
| 09:28.55 | *** join/#asterisk Abbas (n=Abbas@203.81.194.242) |
| 09:29.01 | Abbas | hello |
| 09:29.40 | Abbas | can we do something to let SIP users register on 2 different ports other than the standared one |
| 09:31.58 | Tili | Abbas: this is a bit of problem. Because asterisk will need to send the responses using same port. |
| 09:32.19 | Tili | again it cannot bind itself to 2 different ports at one time |
| 09:36.48 | *** join/#asterisk JimmyCarter (n=del@213083175015.sonofon.dk) |
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| 10:03.02 | *** join/#asterisk clive- (n=pirch@ndn-165-155-199.telkomadsl.co.za) |
| 10:04.08 | clive- | does anyone know of a good place to purchase linksys pap2-na's ? |
| 10:05.43 | *** join/#asterisk ful|work (n=fulgas@209.8.233.68) |
| 10:06.07 | ful|work | hey |
| 10:06.30 | puzzled | hi |
| 10:25.16 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:27.02 | X-Files | Hello users ! I need intercept calls, what me need for this ? I use Gateway ADAPTER |
| 10:27.18 | zoa | look monitor |
| 10:27.28 | X-Files | ;)))) |
| 10:27.36 | *** join/#asterisk Abbas_ (n=Abbas@203.81.194.242) |
| 10:27.40 | zoa | dialplan application monitor |
| 10:28.05 | zoa | http://www.asteriskguru.com/tutorials/monitor.html |
| 10:28.20 | X-Files | tnky :) |
| 10:28.38 | *** join/#asterisk Abbas (n=Abbas@203.81.194.242) |
| 10:29.43 | zoa | ~ ptiggerdine |
| 10:29.46 | wasim | :) |
| 10:30.08 | zoa | <PROTECTED> |
| 10:31.10 | RoyK | zoa: ding |
| 10:31.10 | wasim | zoa: he doubted we'd beat the english |
| 10:31.17 | RoyK | zoa: any news aobout the jb y et? |
| 10:31.23 | RoyK | s/y e/ye/ |
| 10:33.32 | zoa | dong |
| 10:33.41 | zoa | working hard on it royk |
| 10:34.45 | *** join/#asterisk Cybernetics (n=bharatsa@210.211.246.47) |
| 10:34.46 | Cybernetics | hello all |
| 10:36.46 | Cybernetics | i am executing the Queues on real time, though the Queue is getting exevuted successfullly, I am not able to hear the sound which plays the position of the call... |
| 10:37.48 | zoa | royk, nothing new, cleaning up according to coding guidelines now |
| 10:39.26 | JimmyCarter | Anyone know how to periodically reset the queues? |
| 10:40.15 | RoyK | zoa: oki |
| 10:41.58 | X-Files | zoa: this not needed, record a conversation ! |
| 10:42.09 | X-Files | It is necessary to me if someone calls to whom (number 204) to intercept a call and obshchatsja with it |
| 10:42.26 | X-Files | br |
| 10:42.33 | X-Files | and talk with it |
| 10:51.26 | *** join/#asterisk testmachine (n=user@81-171-6-142.dsl.fiberworld.nl) |
| 10:52.35 | Cybernetics | JimmyCarter: what do you mean by periodically resetting the Queues? |
| 10:52.46 | Cybernetics | will you plese explain me |
| 10:52.49 | Cybernetics | whats that |
| 10:53.44 | testmachine | anybody here worked with amp? |
| 10:56.24 | *** join/#asterisk zobia (n=laura_sh@218.6.242.212) |
| 10:56.31 | zobia | Hello everyone |
| 10:56.58 | JimmyCarter | I mean that the data in the queues are reset once in a while. every 30 mins for example |
| 10:57.10 | zobia | i want to translate one line in the dialplan to the database . |
| 10:57.32 | zobia | the line is exten => 1000,1,Dial(${TRUNK}/14558555243,,G(phrase-menu^s^1)) |
| 10:58.01 | JimmyCarter | the number of completeted and abandonded calls are mostly interesting if it is per day.' |
| 10:58.03 | zobia | if i use realtime table to make this extension. in the app field i enter dial |
| 10:58.49 | *** join/#asterisk testmachine (n=user@81-171-6-142.dsl.fiberworld.nl) |
| 10:58.51 | JimmyCarter | I think it's pretty standard in most PBX's |
| 10:58.53 | zobia | in the appdata field i enter ${TRUNK}/14558555243||G|phrase-menu|s|1| |
| 11:00.05 | zobia | ok. thanks. i got the answer |
| 11:03.35 | Cybernetics | JimmyCarter: I didnt get by what do you mean by "data in the Queues"; Queues is just calls waiting in a Queue for thier turn to be answered by an Agent .. Is it? |
| 11:06.00 | *** join/#asterisk GordonF (i=hixscrip@wbs-196-2-112-26.wbs.co.za) |
| 11:07.11 | GordonF | Hi all. I'm having an issue connecting IAX2 to SIP. I get dropped calls with Asterisk 1.9 saying that it could'nt make the link between IAX2 and SIP. Any ideas? |
| 11:07.25 | JimmyCarter | Cybernetics: By data I mean, ex. num of answered, completed and abandoned calls, service level, avg holdtime etc. |
| 11:07.33 | X-Files | PPls, It is necessary to me if someone calls to whom (number 204) to intercept a call and to talk to it (pressing *8). I use asterisk from cvs. |
| 11:07.51 | clive- | gordon howzit |
| 11:07.58 | JimmyCarter | Cybernetics: Its that information I'd like to be able to reset once in a while. |
| 11:08.14 | clive- | gordon sounds like a codec mismatch to me |
| 11:09.44 | Cybernetics | ok |
| 11:09.51 | Cybernetics | I got you JimmyCarter... |
| 11:10.16 | GordonF | clive- Any ideas on resolving it? I have enabled all my codecs and the ISP has enable 723 on their side |
| 11:11.28 | clive- | eugh....G.723??.....you prolly need to use g729 and install a licnece on your asterisk box |
| 11:13.54 | RoyK | there isn't any support for G.723.1 in * |
| 11:14.51 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
| 11:14.59 | cjk | hi, is there a way to set custom cdr fields? |
| 11:15.35 | GordonF | oh dear well that would explain why we can't get it right. A-Law and U work? |
| 11:15.56 | clive- | RoyK...correction..."No legal support"..:)...lol |
| 11:16.17 | RoyK | lol |
| 11:16.41 | *** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 11:21.02 | johnrage | ANYONE here offer a VONAGE like service. I want to have a Philippine Number. PM |
| 11:23.49 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 11:23.58 | Samoied | Hello all! |
| 11:24.11 | Samoied | Is possible to use a 2nd line in FXS port |
| 11:24.20 | Samoied | without threewaycalling? |
| 11:31.42 | GordonF | Thanks for the help guys gotta shoot off quick but I'll be back :) |
| 11:31.42 | wasim | Samoied: no |
| 11:34.10 | *** join/#asterisk coppice (n=chatzill@40.199.17.210.dyn.pacific.net.hk) |
| 11:38.24 | d-tech | Samoied: you just trying to use a two line POTS phone? |
| 11:38.37 | Samoied | d-tech: no |
| 11:38.39 | Samoied | d-tech: I want to make a 2nd call |
| 11:38.48 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 11:39.01 | Samoied | d-tech: but no threewaycalling |
| 11:39.31 | Samoied | d-tech: I want to press Flash, and talk to first caller, no make a conference |
| 11:39.59 | *** join/#asterisk ard (n=ard@2001:7b8:32d:0:20c:6eff:fe18:d11f) |
| 11:40.20 | *** join/#asterisk t0p (i=lamer@203.152.43.215) |
| 11:40.42 | *** part/#asterisk t0p (i=lamer@203.152.43.215) |
| 11:41.48 | d-tech | Samoied: I'm gonna say it is possible, but not aware of any proven solution |
| 11:42.12 | *** join/#asterisk t0p (i=lamer@202.8.86.162) |
| 11:42.25 | *** part/#asterisk t0p (i=lamer@202.8.86.162) |
| 11:43.02 | *** join/#asterisk [Lamer] (i=lamer@202.8.86.162) |
| 11:44.11 | Samoied | d-tech: OK, for now I was using IPPhones for that. But I have user connected in FXS lines, and they want same functionality. |
| 11:45.10 | [Lamer] | Hi there, is the type=peer workable in 1.2.0 RC2? |
| 11:45.11 | mut | just turn off threeway calling in the config.. |
| 11:45.47 | mut | threewaycalling=no |
| 11:45.51 | mut | in zapata.conf |
| 11:46.02 | Samoied | mut: But with this, I dont have 2nd call |
| 11:46.08 | [Lamer] | I had the "returned 0: Invalid argument" error after the upgrade |
| 11:46.27 | mut | huh? |
| 11:46.36 | mut | you're talkin ya want to switch between calls right? |
| 11:46.43 | Samoied | mut: yes |
| 11:46.49 | mut | so just turn it off.. |
| 11:46.53 | mut | and it should switch |
| 11:47.04 | mut | just make sure call waiting is on |
| 11:47.17 | Samoied | mut: I want to make a 2nd call. Not only receive. |
| 11:47.23 | mut | yes |
| 11:47.38 | mut | should be able to make a second call doing that.. |
| 11:47.53 | Samoied | mut: Ok, I try this. |
| 11:48.09 | mut | it just won't connect the two when ya press flash again |
| 12:03.08 | Samoied | mutilator: I have tried this, but when I press Flash, not occurs. |
| 12:04.02 | mutilator | does the cli show anything when ya press flash? |
| 12:04.06 | mutilator | verbose 5 it |
| 12:04.32 | mutilator | i don't have an fxs hooked up to test it out |
| 12:06.00 | Samoied | mutilator: show nothing |
| 12:06.55 | Samoied | <PROTECTED> |
| 12:06.55 | Samoied | <PROTECTED> |
| 12:06.55 | Samoied | <PROTECTED> |
| 12:06.55 | Samoied | <PROTECTED> |
| 12:06.55 | Samoied | <PROTECTED> |
| 12:07.07 | mutilator | hm just ignores it alltogether eh |
| 12:07.49 | mutilator | not sure then |
| 12:07.59 | mutilator | never actually done it myself |
| 12:08.57 | Samoied | mutilator: thanks |
| 12:09.06 | mutilator | sry :P |
| 12:09.14 | *** join/#asterisk razu (n=razu@tln-kontor.norby.ee) |
| 12:15.11 | {zombie} | Samoied: you need threewaycalling=yes for the hookflash to work |
| 12:15.34 | {zombie} | and I don't think threewaycalling in zapata.conf means 3way conference |
| 12:16.19 | {zombie} | I think they mean you can talk to two different people and switch between them with a hook flash |
| 12:16.47 | {zombie} | see http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zapata.conf |
| 12:17.31 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 12:18.12 | mutilator | it conferences when ya flash |
| 12:18.34 | mutilator | if i call bob |
| 12:18.37 | mutilator | then flash and call mary |
| 12:18.45 | mutilator | then flash again, i'll be talkin to bob and mary both |
| 12:18.54 | mutilator | won't go back to just bob |
| 12:22.09 | christo | I'm using IO::Socket::INET to write to the manager interface on a local * box, but I can't seem to capture what comes back. Can anybody see the obvious mistake in this please? http://pastebin.ca/28899 |
| 12:24.54 | Samoied | {zombie}: the mutilator is correct |
| 12:25.26 | Samoied | {zombie}: I want to talk to only 1 person, and switch with hook flash |
| 12:25.28 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
| 12:25.58 | Samoied | {zombie}: But with threewaycalling=yes, I talk to 2 at same time |
| 12:32.25 | *** join/#asterisk stoffell (n=stoffell@241.43-201-80.adsl.skynet.be) |
| 12:32.41 | stoffell | hi all |
| 12:32.59 | mutilator | O_o |
| 12:40.05 | *** join/#asterisk Teeli (n=Tili@213.132.60.182) |
| 12:40.19 | *** part/#asterisk Teeli (n=Tili@213.132.60.182) |
| 12:42.36 | stoffell | anyone knows the big (quality? echo?) diff on using chan_mISDN or junghanns bristuff? |
| 12:47.25 | *** join/#asterisk echion (n=rickard@83.140.44.242) |
| 12:47.43 | echion | is there a way to specifiy what SIP Reponse should be used as an answer? |
| 12:49.05 | echion | i.e I'd like to send back 484 or something? |
| 12:54.27 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 13:02.34 | *** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com) |
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| 13:06.18 | *** join/#asterisk _GiGi_ (i=gigi@disc.more.pl) |
| 13:06.28 | _GiGi_ | hello |
| 13:07.24 | Druken | anyone have a NPANXX rate center database they are willing to share? |
| 13:08.01 | _GiGi_ | how i can transfer connection on ZAP channel ? |
| 13:08.26 | fugitivo | the duration= in the voicemail txt, is milisec? |
| 13:14.14 | *** part/#asterisk billatq (i=bill@aggienerds.org) |
| 13:15.04 | *** join/#asterisk t0p (i=lamer@tech-mgr.chatri.com) |
| 13:15.36 | X-Files | i need help, i want configure asterisk to answering other line (press *8)? |
| 13:15.46 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:15.48 | t0p | what is the better way to setup dual * servers? iax or sip? |
| 13:16.26 | fugitivo | i like iax |
| 13:17.28 | zoa | iax2 |
| 13:17.31 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:17.35 | Vhata | t0p: iax2 |
| 13:17.36 | zoa | do you mean interconnect ? |
| 13:17.41 | zoa | just 2 servers ? |
| 13:18.15 | t0p | zoa: yeah, only start with 2 servers |
| 13:19.22 | zoa | no no i mean those 2 servers are interconnected ? |
| 13:19.27 | zoa | or they are just standalone ? |
| 13:19.53 | X-Files | need help, I want configuring asterisk to answering other line (press *8), if not answering other line? protocol SIP |
| 13:21.34 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 13:21.45 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:21.48 | t0p | zoa: by what means? |
| 13:21.55 | fourcheeze | is there a way to increase the frequency of the keepalives * sends with qualify=yes ? |
| 13:22.14 | zoa | qualify=20 |
| 13:22.20 | zoa | i think the default is 500 or so |
| 13:22.22 | zoa | dunno |
| 13:22.23 | zoa | check sources |
| 13:22.28 | t0p | zoa: they are both connected to the internet, one with real ip and the other behind NAT |
| 13:22.34 | fourcheeze | no, Ithink that just tells it how long to wait before giving up |
| 13:22.42 | fourcheeze | zoa: default is 2000 |
| 13:22.54 | fourcheeze | but I think it sends a keepalive every minute |
| 13:23.10 | mutilator | Druken? |
| 13:23.22 | mutilator | http://members.dandy.net/~czg |
| 13:23.25 | mutilator | just rip that |
| 13:26.40 | _GiGi_ | how i can do native transfer on zaptel (zaphfc) channel ? |
| 13:27.06 | t0p | zoa: i linked them using sip so that users on the one behind NAT could call out using fxo on the real ip |
| 13:27.23 | _GiGi_ | i connect my asterisk to another PBX and i must transfer connection to another number. |
| 13:28.09 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 13:29.34 | *** join/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 13:29.52 | *** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 13:31.00 | QbY | I have an IAX Connection to a telco provider--I want to start recording the moment a call comes in, and continue until its done.. Where do I put that at? |
| 13:31.23 | wasim | QbY: in the first priority |
| 13:31.43 | QbY | in extensions.conf or iax.conf? |
| 13:31.49 | wasim | QbY: extensions.conf |
| 13:32.52 | ManxPower | ~docs |
| 13:32.56 | jbot | rumour has it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 13:33.14 | ManxPower | fourcheeze, Why do you want to do that. They are sent every 30 seconds or so. |
| 13:34.03 | ManxPower | qualify=20 means the device has to take longer than 20ms to respond to a OPTIONS (SIP) or PING (IAX2) packet in order to be considered "LAGGED" |
| 13:34.05 | fugitivo | the duration= in the voicemail txt, is milisec or what?? |
| 13:34.16 | ManxPower | it does NOT indicate how often to send those packets |
| 13:34.25 | fourcheeze | ManxPower: yeah, that's what I thought |
| 13:34.35 | ManxPower | fugitivo, I always assumed it was in seconds, but I'd have to look it up. |
| 13:34.43 | fourcheeze | I want to send them faster becuase my snoms are still losing registration :-( |
| 13:34.54 | ManxPower | fourcheeze, that is not the cause. |
| 13:35.01 | fourcheeze | ManxPower: ok, any other ideas? |
| 13:35.08 | ManxPower | fourcheeze, see the mailinglist for info on SNOMs losing registration |
| 13:35.13 | fourcheeze | yeah, done that |
| 13:35.18 | fugitivo | ManxPower: i think it's not seconds :( |
| 13:35.20 | fourcheeze | trried a load of things |
| 13:35.45 | fourcheeze | gotta find a solution though, so I'm happy to search again |
| 13:35.51 | ManxPower | fugitivo, What NAT router are the SNOMs behind? Have you upgraded the firmware on the SNOMs? |
| 13:36.12 | fourcheeze | ManxPower: assume you mean fourcheeze |
| 13:36.23 | fourcheeze | ManxPower: I've tried 3 different routers |
| 13:36.27 | fugitivo | ManxPower: me? |
| 13:36.41 | fourcheeze | vigor, belkin, cheap ebuyer router |
| 13:36.49 | fourcheeze | it is certainly worse on the belkin |
| 13:37.02 | fourcheeze | ManxPower: I've also tried different firmwares on the snoms |
| 13:37.09 | fourcheeze | 3.57, 3.60 and 4.0 |
| 13:37.17 | fourcheeze | no change with any of those |
| 13:37.30 | *** join/#asterisk nick125 (n=nick@unaffiliated/nick125) |
| 13:37.59 | fourcheeze | ManxPower: it only seems to happen when more than 1 snom is behind the same NAT router |
| 13:38.00 | ManxPower | fourcheeze, sounds like you tried everything I would have suggested other than to set qualify=no and setting the reginstration interval on the SNOM to be 1 min |
| 13:38.18 | fourcheeze | ManxPower: that's pretty well where I started |
| 13:38.24 | ManxPower | fourcheeze, Ah. That is a CLASSIC problem with crappy NAT routers, but if you tried 3 of htem....... |
| 13:38.39 | QbY | So to record everything coming in on my IAX connection -- would i put monitor() in the from-pstn of my setup ?? http://pastebin.ca/28905 |
| 13:38.43 | fourcheeze | well strangely it's much better on the cheaper router |
| 13:39.06 | fourcheeze | the unbranded router doesn't seem to have a problem |
| 13:39.16 | ManxPower | Try Cisco 8-) |
| 13:39.19 | fourcheeze | hmm |
| 13:39.29 | fourcheeze | is that what you use? |
| 13:39.36 | ManxPower | I could tell you how to configure a Cisco 17xx router for NAT/SIP 8-) |
| 13:39.41 | fourcheeze | ManxPower: sipuras seem tohave no problem |
| 13:39.46 | fourcheeze | only snoms |
| 13:40.00 | ManxPower | fourcheeze, IF the router has options for NAT SIP, DISABLE IT. If the phones have options for NAT TURN THEM OFF. |
| 13:40.08 | *** join/#asterisk lehel (n=lehel@82.79.20.17) |
| 13:40.12 | *** join/#asterisk Nivex (i=kjotte@user-0c8hq5r.cable.mindspring.com) |
| 13:40.12 | lehel | hello |
| 13:40.32 | ManxPower | they way many phones handle NAT (the ones that have special options for that) can cause issues with Asterisk. |
| 13:40.48 | ManxPower | fourcheeze, Ah, so it's phone specific. |
| 13:40.56 | fourcheeze | certainly seems to be |
| 13:41.04 | fourcheeze | it seems to be a combination of things |
| 13:41.20 | fourcheeze | things I try seem to help |
| 13:41.28 | fourcheeze | when I started out they would only register for a few minutes at a time |
| 13:41.32 | fourcheeze | and now I'm up to 2 hours |
| 13:42.03 | fourcheeze | in the snom gui it says "Registration failed" which seems odd |
| 13:43.14 | lehel | because i don't have my channels configured in zapata.conf, but in zapata_additional < that's why there is no effect if i change the rx/txgain? |
| 13:43.16 | X-Files | need help, I want configuring asterisk to answering other line (press *8), if not answering other line? protocol SIP |
| 13:43.36 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net) |
| 13:44.29 | *** join/#asterisk remibreval (i=Remek@pro75-3-82-234-175-208.fbx.proxad.net) |
| 13:44.40 | remibreval | Hello everyone ! |
| 13:45.52 | remibreval | Which command do you use in the CLI to force register again IAX ? It seems that when I change parameters it and re-read conf file, it does not try to register again and stay with last parameters |
| 13:48.51 | fourcheeze | ManxPower: do snoms actually need to register - if I poked a hole in the NAT and forwarded a sip port per phone would that work? |
| 13:49.07 | ManxPower | remibreval, use IP addresses in the register line, not DNS names |
| 13:49.26 | ManxPower | fourcheeze, do the SNOMs have the option to change the SOURCE PORT of their SIP traffic? |
| 13:49.33 | fourcheeze | yes |
| 13:49.43 | ManxPower | fourcheeze, if so, that might help |
| 13:49.43 | remibreval | Manxpower, I'm not sure, because when I use iax2 show registry it shows the IP (so DNS works). |
| 13:50.09 | ManxPower | remibreval, all it takes is ONE DNS failure asterisk will stop working for that device/protocol |
| 13:50.13 | remibreval | It worked before, but I ask to change my password (fucking bad copy/paste in pastebin....) and now I have troubles |
| 13:50.22 | fourcheeze | ManxPower: yes they have an option "Network Identity" which is for a fixed port |
| 13:50.37 | ManxPower | remibreval, reload and reload_chan_iax2.so will restart the reg process. |
| 13:50.38 | fourcheeze | do with a static IP I could do away with registrations altogether |
| 13:50.49 | fourcheeze | ManxPower: do I need to use insecure=very if I do that? |
| 13:51.00 | ManxPower | fourcheeze, no. |
| 13:51.10 | remibreval | Ok, I try with IP -just back in 2 mins |
| 13:51.22 | fourcheeze | ManxPower: do I use a blank secret? |
| 13:51.31 | ManxPower | fourcheeze, no. |
| 13:51.45 | X-Files | need help, I want configuring asterisk to answering other line (press *8), if not answering other line? protocol SIP |
| 13:51.50 | ManxPower | registration ONLY tells the remote device what the IP/port of the local device is. |
| 13:52.14 | fourcheeze | ManxPower: if I set the host as a fixed IP it tells me that something tried to register but wasn't set with host=dynamic |
| 13:52.47 | remibreval | ManxPower, I know it. It works well for peer but not for user. I still have 60 Rejected |
| 13:52.54 | remibreval | what could be reasons ? |
| 13:53.47 | remibreval | Before I had success with it... :-( |
| 13:55.08 | *** join/#asterisk Patrick^ (n=patrick_@pc-0-34.mountaincable.net) |
| 13:58.16 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 14:00.41 | X-Files | need help, I want configuring asterisk to answering other line (press *8), if not answering other line? protocol SIP |
| 14:01.27 | *** join/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net) |
| 14:01.42 | obsidian-studios | morning all |
| 14:02.02 | obsidian-studios | is there anyway to get sip registration debugging or logging info? |
| 14:02.52 | ManxPower | obsidian-studios, you mean like "sip debug"? |
| 14:04.49 | obsidian-studios | well done all that and still not getting registration info. I am not really asking for me, I have not really had issues with SIP stuff except for softphones. A person on the LUG is having issues with Xlite and a Sipurar ATA registered with * |
| 14:05.20 | *** join/#asterisk fri (n=fri@port84.ds1-sdb.adsl.cybercity.dk) |
| 14:05.23 | obsidian-studios | covered all the basics with them, and it really looks like a client side issue, because * is configured correctly at least in sip.conf |
| 14:06.21 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 14:06.25 | ManxPower | There is 1 hotel with rooms available out of 37 BestWestern brand hotels within 100 miles of New Orleans. *sigh* |
| 14:07.31 | ManxPower | obsidian-studios, if it's not showing up on sip debug then it's a client or network or router issue. |
| 14:07.48 | obsidian-studios | yeah that's what I thought and have been telling them |
| 14:08.03 | obsidian-studios | everything looks good when doing a sip show peers or users |
| 14:08.29 | obsidian-studios | pretty sure it's Xlite bitching about the registration, the Sipura ATA was what shocked me |
| 14:08.54 | ManxPower | obsidian-studios, it's prolly a router problem |
| 14:09.24 | obsidian-studios | they were doing all the tcpdump and etc looking at packets seeing them traverse |
| 14:09.38 | obsidian-studios | but bottom line it's anything but * :) |
| 14:13.00 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 14:13.29 | *** join/#asterisk pmr (n=chatzill@kmwenergy.ody.ca) |
| 14:14.32 | *** join/#asterisk Aurix (n=r@CPE-61-9-212-60.qld.bigpond.net.au) |
| 14:14.53 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:15.49 | Aurix | hey, sorry for a possibly stupid question, but I used to use Traverse Technology's NetJet-S using the hisax module under 2.4, with a voice patch and i4l. I've been unable to find a voice patch for 2.6, wondering if the voice changes have been merged into 2.6, or what I need to do to get this card working under asterisk under 2.6? |
| 14:15.56 | *** join/#asterisk jimmy_deanPB (n=jhodapp@72.244.232.226) |
| 14:16.48 | Damin | There is no Asterisk 2.6 |
| 14:16.53 | obsidian-studios | ManxPower: should sip debug show registration attempts and successes? If it should and does not it's because the request is not making it to *? |
| 14:16.56 | Aurix | 2.6 kernel |
| 14:18.19 | Aurix | atm, it's able to pick up calls, but it doesn't seem to be able to do anything. line is silent. |
| 14:18.43 | Aurix | i'm leaning towards it being a problem with the driver for the card, and lacking voice support in 2.6 |
| 14:20.26 | ManxPower | obsidian-studios, sip debug will show ALL SIP traffic, including registratation (not audio, since that's RTP, not SIP) |
| 14:20.47 | *** join/#asterisk tamp4x (n=kkdkdkkk@204.124.238.248) |
| 14:20.51 | obsidian-studios | ManxPower: ok, so if we get nothing it's because * see nothing ;) cool |
| 14:21.05 | ManxPower | obsidian-studios, correct. |
| 14:21.09 | tamp4x | is anyone having problems with snom 360s holding registration? |
| 14:21.16 | X-Files | need help, I want configuring asterisk to answering other line (press *8), if not answering other line? protocol SIP |
| 14:21.28 | ManxPower | tamp4x, fourcheeze is |
| 14:21.34 | obsidian-studios | ManxPower: ty |
| 14:21.41 | fourcheeze | tamp4x: join the club |
| 14:22.13 | fourcheeze | free Snom 360 with every new subscription |
| 14:22.32 | fourcheeze | you can use as a doorstop or how you like |
| 14:23.19 | *** part/#asterisk pmr (n=chatzill@kmwenergy.ody.ca) |
| 14:24.24 | fourcheeze | tamp4x: what have you tried so far? |
| 14:24.37 | tamp4x | what do u mean |
| 14:24.56 | tamp4x | as in settings or other phones |
| 14:24.57 | Katty | Damin: but there's a 2.6 kernel |
| 14:25.04 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 14:25.14 | fourcheeze | tamp4x: I mean what have you tried with the snoms |
| 14:25.17 | Katty | Damin: cause i'm on 2.6.8 |
| 14:25.19 | fourcheeze | to get them to work |
| 14:25.42 | tamp4x | changing qualify time, changing timing on the phones |
| 14:25.48 | stoffell | anyone else used kirk phones on * ? :) |
| 14:25.52 | Damin | Katty: Of course there is a 2.6 kernel! |
| 14:26.00 | tamp4x | updating software version dont help |
| 14:26.02 | Damin | Katty: Now go back to bed.. |
| 14:26.05 | Katty | Damin: kbi |
| 14:26.06 | tamp4x | im losing m ymind here |
| 14:26.14 | tamp4x | out of ideas |
| 14:26.23 | tamp4x | about to grab cvsehad see if that does anything |
| 14:26.26 | [TK]D-Fender | Funny Voicemail question : I have users update their VM pass in CM, I see an update to voicemail.conf to reflect it, but it doesn't seem to go into effect. Any ideas as to what could block it? I have found that doing a "reload" at CLI seems to do the job, but should not be necessary (as this has other negative effects as well). |
| 14:26.27 | tamp4x | head |
| 14:26.31 | Damin | tamp4x: How about changing the phone? ;) |
| 14:26.51 | tamp4x | these people have 8 lines on each phone |
| 14:26.59 | tamp4x | no other phone supports that many |
| 14:27.20 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfj32.dialup.mindspring.com) |
| 14:27.25 | tamp4x | i _HAVE_ to make this work |
| 14:27.30 | fourcheeze | tamp4x: I find I get best results using qualify=yes, snom session set to 1min, support broken registrar=yes |
| 14:27.45 | *** join/#asterisk hhoffman (n=hhoffman@asylum.afflictions.org) |
| 14:27.46 | Damin | tamp4x: Then call SNOM! :) |
| 14:27.53 | fourcheeze | tamp4x: likewise - I have a site that just went all snom after trying out one and now it doesn't work |
| 14:28.06 | Damin | I dumped SNOM as a vendor last year.. |
| 14:28.09 | fourcheeze | tamp4x: which router are they behind? |
| 14:28.29 | *** join/#asterisk znoG (n=gs@OL101-122.fibertel.com.ar) |
| 14:28.33 | znoG | hi |
| 14:28.34 | clive- | Damin do you sell linksys apa2's |
| 14:28.38 | fourcheeze | tamp4x: I also found that using a stun server helped a little |
| 14:28.38 | clive- | pap2 |
| 14:28.39 | znoG | just wondering, where is it exactly that BKW has gone off to? |
| 14:28.46 | Damin | clive-: Not really.. |
| 14:28.51 | fourcheeze | Damin: what would you suggest as an alternative to the 360 |
| 14:28.51 | Katty | znoG: asleep? |
| 14:28.52 | hhoffman | hi, I'm trying to setup a IAX softphone and the phone auth's just fine but when I dial a extension I get: chan_iax2.c: Rejected connect attempt from 192.168.4.12, request '1000@incoming' does not exist |
| 14:28.57 | clive- | I am looking for a good supplier |
| 14:29.02 | Katty | znoG: he's not usually awake right now..or at least talking |
| 14:29.07 | znoG | Katty: i meant which project |
| 14:29.12 | Katty | oh |
| 14:29.16 | Katty | no clue |
| 14:29.16 | Damin | fourcheeze: Polycom IP 601 |
| 14:29.20 | Katty | he's working on a lot of things, znoG |
| 14:29.24 | znoG | Katty: according to Allison, he doesn't develop for Asterisk anymore, unless I misinterpreted |
| 14:29.26 | fourcheeze | Damin: do they support BLF? |
| 14:29.30 | Damin | fourcheeze: Only 6 line appearanced, but they work. |
| 14:29.36 | Damin | What is BLF? |
| 14:29.42 | fourcheeze | busy lamp field |
| 14:29.49 | fourcheeze | i.e. you can see if another person is busy |
| 14:29.55 | Damin | Nope.. |
| 14:29.56 | tamp4x | they are behind 3com 4400 se switches going to a cisco 3800 |
| 14:30.13 | Katty | Damin: what have you done?! |
| 14:30.24 | Damin | Katty: What do you mean? |
| 14:30.25 | Katty | Damin: yesterday it was 68F over here. and now.. /now/ it's 34F |
| 14:30.31 | tamp4x | and when i put stun they wont register at all |
| 14:30.33 | *** part/#asterisk stoffell (n=stoffell@241.43-201-80.adsl.skynet.be) |
| 14:30.48 | [TK]D-Fender | hhoffman : you don't have your extensions.conf set up to match what you dialed. You need an exten => 1000,1, whatever and more in that context in extensions.conf |
| 14:30.48 | Damin | Katty: Ohh.. THAT... Sorry.. |
| 14:30.54 | fourcheeze | tamp4x: check that the stun server is actually working |
| 14:31.04 | Katty | hmm, glasses. |
| 14:31.09 | Katty | i only wear those while driving. |
| 14:31.15 | Damin | Anyone know how to make an Audiocodes FXS gateway hunt? |
| 14:31.20 | tamp4x | i am currently at the site that has 500 students and i have 8 administrative snom 360s that must be up 24/7 |
| 14:31.26 | file | Damin: tell it to hunt or you'll shoot it? |
| 14:31.27 | Katty | Damin: have you tried hugging them? |
| 14:31.32 | hhoffman | [TK]D-Fender: I have exten => 1000,1,VoiceMail(b1000@default) |
| 14:31.35 | Katty | Damin: also, they sell magic wands on ebay, i hear |
| 14:31.43 | tamp4x | four cheeze i use stun.fwdnet.net ...do you have any better suggestion? |
| 14:31.50 | Damin | Wow.. the technical nature of this conversation is blowing me away! |
| 14:31.51 | mutilator | they don't work katty |
| 14:31.56 | mutilator | i tried |
| 14:31.59 | Katty | sad. |
| 14:32.01 | file | Damin: reflash it with newer firmware that turns it into a toaster |
| 14:32.05 | mutilator | i was floored |
| 14:32.08 | mutilator | cried for a week |
| 14:32.11 | Katty | Damin: and a brave little toaster at that! |
| 14:32.12 | mutilator | i spent like $500 on it |
| 14:32.14 | tamp4x | or does anyone know of a good stun server |
| 14:32.21 | [TK]D-Fender | hhoffman : if you do its not in a context named [incoming] |
| 14:32.25 | Damin | tamp4x: Just download one and install it.. |
| 14:32.33 | Katty | Damin: besides, i can't have technical conversations this early in the morning. |
| 14:32.34 | Damin | tamp4x: Look on sourceforge.. |
| 14:32.39 | fourcheeze | tamp4x: try stunserver.org |
| 14:32.41 | Katty | Damin: if it's before 9, all i comprehend is that black holes suck. |
| 14:32.44 | hhoffman | [TK]D-Fender: oh, I see what I did... thanks! |
| 14:32.48 | [TK]D-Fender | np |
| 14:32.59 | Damin | Katty: And the black hole is the lack of coffee in my coffee jar.. |
| 14:33.09 | hhoffman | is there a way to have a default for s without having asterisk answer my POTS line? |
| 14:33.10 | fourcheeze | tamp4x: also if you have a unix box you can test the stunserver with the stun client |
| 14:33.12 | Katty | Damin: you have a coffee jar? |
| 14:33.17 | Damin | Katty: I nearly cried when I went to make a pot last night.. |
| 14:33.22 | Katty | aww. |
| 14:33.24 | malverian[work] | What's the lowest number you can adjust txgain value to in zapata.conf ? |
| 14:33.27 | Damin | Katty: Yes.. to store coffee beans in.. |
| 14:33.31 | Katty | Damin: oooh. |
| 14:33.35 | Katty | Damin: you're one of /those/ people |
| 14:33.46 | Damin | Katty: Yes.. THOSE people.. |
| 14:33.55 | tamp4x | did the stun server solve your problems fourcheze? |
| 14:34.04 | Damin | Katty: I also cook with something other than a microwave! ;) |
| 14:34.07 | skyen | ah, stun |
| 14:34.08 | fourcheeze | tamp4x: it helps a little behind some firewalls |
| 14:34.11 | Katty | also! on an unrelated note... can itunes do something with oggs? |
| 14:34.16 | Katty | file would know. |
| 14:34.17 | skyen | how can i make my asterisk stun klients? |
| 14:34.23 | Katty | Damin: neat, so do i. |
| 14:34.29 | skyen | what do I google? |
| 14:34.30 | Katty | Damin: in fact, my entire blog is dedicated to cooking. |
| 14:34.31 | file | Katty: it needs a plugin to do it I believe |
| 14:34.36 | Damin | Oh yeah? |
| 14:34.37 | fourcheeze | tamp4x: how long do your snoms stay registered for? |
| 14:34.38 | Katty | file: do you know the name of said plugin? |
| 14:34.39 | ^Howler | Damin: what? like a friends microwave? |
| 14:34.48 | file | Katty: unfortunately no, I don't have any oggs |
| 14:34.53 | Katty | file: k |
| 14:34.56 | Katty | file: i shall consult google |
| 14:35.30 | *** part/#asterisk skyen (n=rickard@skalleper.ostman.net) |
| 14:35.47 | Katty | file: oh, is that just osx? |
| 14:35.57 | tamp4x | one building 1-15 minutes, another building hours |
| 14:36.00 | file | Katty: and Windows I think |
| 14:36.06 | Katty | file: ooh. |
| 14:36.06 | Katty | k |
| 14:36.13 | file | Gooooooogle is thy friend |
| 14:36.16 | [TK]D-Fender | hhoffman : PM me, you seem to be a little lost and quite new to * and I'm willing to help you outside of channel |
| 14:36.17 | Katty | file: what about ipod? |
| 14:36.19 | file | and if it isn't, you make me sad :( |
| 14:36.23 | file | iPod doesn't play oggs |
| 14:36.29 | MikeJ[Laptop] | ogg |
| 14:36.48 | fourcheeze | tamp4x: are they both the same router? |
| 14:36.48 | Katty | file: can it be flashimicated? |
| 14:36.54 | tamp4x | yup |
| 14:36.58 | file | Katty: to do oggs? nah |
| 14:37.08 | [TK]D-Fender | iPod : VERY nice mechanics, DUMB management |
| 14:37.14 | tamp4x | different switches tho |
| 14:38.23 | Katty | file: k |
| 14:38.53 | yxa | why would anyone buy ipods when a mobile phone nowadays can accomplish everything and more like err talking on the phone? |
| 14:39.03 | *** join/#asterisk _Sam-- (n=sam@phone2.kneedraggers.com) |
| 14:39.04 | Vhata | because ipods are *sexy* |
| 14:39.13 | ManxPower | yxa, Do you HAVE an MP3 cell phone? |
| 14:39.22 | ManxPower | If you did, you would know why. |
| 14:39.30 | yxa | yeah i do but i seldom use that |
| 14:39.46 | _Sam-- | if i have no calls happening on my system, why would i see this message: Nov 16 09:33:08 WARNING[27824]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 57b787c9504ad37e7938994c08624e5f@192.168.0.7 for seqno 102 (Non-critical Request) |
| 14:40.12 | ManxPower | _Sam--, it means the remote device stopped accepting SIP traffic |
| 14:40.32 | ManxPower | _Sam--, Even if there are no calls SIP is still a chatty protocol |
| 14:40.35 | [TK]D-Fender | MP3 players suck up battery and we'd rather not be empty when we need it? |
| 14:40.49 | fourcheeze | tamp4x: I find it hard to believe that a switch makes any difference |
| 14:40.50 | _Sam-- | is it only showing because i have too many -vvvvvv's at startup? |
| 14:40.59 | fourcheeze | tamp4x: what expiry do you have on your sessions? |
| 14:41.08 | ManxPower | _Sam--, no. It's a bad message |
| 14:41.09 | tamp4x | whatever is default |
| 14:41.21 | tamp4x | i even lowered it before etc |
| 14:41.30 | _Sam-- | how can i track which client/user/anything that message goes with? |
| 14:41.52 | fourcheeze | tamp4x: try 1 minute |
| 14:41.59 | tamp4x | ok |
| 14:42.00 | ManxPower | _Sam--, 192.168.0.7 is the IP address of your Asterisk server? |
| 14:42.03 | fourcheeze | and check on the console that it's using that |
| 14:42.03 | _Sam-- | yep |
| 14:42.19 | tamp4x | ok |
| 14:42.37 | ManxPower | _Sam--, add qualify=yes to all the entries in sip.conf, then you should see one device become unreachable |
| 14:42.57 | _Sam-- | thank you, i'll give it a shot |
| 14:43.04 | fourcheeze | tamp4x: and you're also using qualify in sip.conf ? |
| 14:43.09 | tamp4x | yes |
| 14:43.19 | tamp4x | qualify=5000 |
| 14:43.51 | fourcheeze | tamp4x: I don't think the actual time makes any difference to our problem |
| 14:43.58 | fourcheeze | as long as qualify is on |
| 14:44.41 | tamp4x | i talked to other peopel and they said to set it high |
| 14:45.04 | fourcheeze | tamp4x: I've played with high and low and not found any difference yet |
| 14:45.50 | fourcheeze | tamp4x: the other suggestion I had was to do a pcap dump |
| 14:46.05 | fourcheeze | I'm not near the phones right now so I can't try it, but it might throw up something |
| 14:46.07 | *** join/#asterisk hikaro (n=devil@202.53.235.51) |
| 14:46.23 | *** join/#asterisk josue_m (n=joss@200.30.173.114) |
| 14:47.21 | tamp4x | would max fowards be an issue |
| 14:49.00 | *** join/#asterisk mutilator (n=animenod@65.111.201.79) |
| 14:51.13 | [TK]D-Fender | Another nifty problem : I have 2 Uniden UIP-200's here and am having trouble getting them running properly. I have nat=never as the WIKI suggests, and they previously worked. They list as "unreachable", but can dial in just fine, just not OUT. |
| 14:52.02 | tamp4x | as soon as people call out with the snom 360 it resets |
| 14:52.44 | tamp4x | the registration |
| 14:53.35 | fourcheeze | tamp4x: I'm not sure but that sounds like a problem I had with an old vgor router |
| 14:53.59 | fourcheeze | except in that instance the router reset itself every time a call was made |
| 14:54.24 | fourcheeze | tamp4x: do you have the option to put an asterisk server on each site and terminate via those? |
| 14:54.31 | josue_m | hello: where can I find a very basic tutorial for newbies to asterisk? |
| 14:54.49 | [TK]D-Fender | ~docs |
| 14:54.50 | jbot | docs is probably probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 14:55.03 | [TK]D-Fender | check out the handbook draft. Its a good primer |
| 14:55.25 | [TK]D-Fender | the Oreilly book is a big heavy for n00bs. |
| 14:55.27 | josue_m | k, thanks |
| 14:55.30 | [TK]D-Fender | np |
| 14:56.38 | LostFrog | Who thinks a WRT54G would handle one sip line with no translation (except for prompts)? |
| 14:57.04 | LostFrog | Not a critical system.. personal phone number. |
| 14:57.06 | [TK]D-Fender | Using Re-invite I hear 4 could work. |
| 14:57.44 | LostFrog | But, then I could transfer calls between extensions. |
| 14:57.51 | [TK]D-Fender | I've got a WRT54GS here I'm going to mod to hell |
| 14:57.59 | LostFrog | Albeit, I only want to use one extension at a time |
| 14:58.06 | [TK]D-Fender | Sure, just make sure that * isn't in the RTP path I guess.... |
| 14:58.18 | [TK]D-Fender | avoid wherever possible. What kind of endpoints? |
| 14:58.48 | LostFrog | snom and ATA. |
| 14:58.53 | [TK]D-Fender | How do I change verbosity in CLI again? |
| 14:58.59 | [TK]D-Fender | Which ATA? |
| 14:59.09 | LostFrog | grandstream |
| 14:59.23 | [TK]D-Fender | Hmmmm, if they support re-invite you should be fine.. |
| 14:59.48 | *** join/#asterisk mashedpotats (n=potats@pool-151-203-73-60.bos.east.verizon.net) |
| 15:00.44 | *** join/#asterisk starsoft (n=starsoft@66.36.22.52) |
| 15:00.53 | starsoft | good morning everyone |
| 15:01.10 | josue_m | hello |
| 15:01.11 | *** join/#asterisk gvag11 (n=g@84.254.12.133) |
| 15:01.26 | gvag11 | hi everybody |
| 15:01.39 | c0w | just a question |
| 15:01.46 | c0w | does asterisk work ok with gcc4 |
| 15:02.59 | *** join/#asterisk thefergus (n=fergus@ip-152010154003.ess.appstate.edu) |
| 15:03.07 | azzie | c0w, does anything work with gcc4 ? :-) |
| 15:03.26 | Dibbler | Anything special with regards Arch module compilation on an AM64 system need to be done? |
| 15:03.26 | Math[laptop] | yes it does |
| 15:03.40 | Math[laptop] | gcc4 is fine :P |
| 15:04.00 | gvag11 | using spandsp 0.0.2pre21 and asterisk 1.2rc2 and TE205P, i am getting broken pages . Not all of the faxes i sent or receive although. Any idea? Is that because of frame slips? |
| 15:04.13 | Dibbler | Was that meant for me? |
| 15:04.32 | azzie | Math[laptop], too much stuff does not compile with it :-( |
| 15:04.34 | Math[laptop] | Dibbler, no, was meant for gcc4 |
| 15:04.46 | Math[laptop] | really? |
| 15:04.48 | hhoffman | I compile lots of stuff with gcc4 |
| 15:04.54 | Math[laptop] | so do I |
| 15:04.59 | starsoft | I have two grandstream sip phones behind 2 different nats that can call out using a sip proxy and they work fine, although when i try to call one phone from the other all i get is dead air, can someone point me in the direction of why this might be? |
| 15:05.31 | starsoft | oh, and when both phones werebehind the same nat (at my office) they talked to eachother perfectly |
| 15:05.33 | gvag11 | using spandsp 0.0.2pre21 and asterisk 1.2rc2 and TE205P, i am getting broken pages . Not all of the faxes i sent or receive although. Any idea? Is that because of frame slips? |
| 15:05.49 | *** part/#asterisk thefergus (n=fergus@ip-152010154003.ess.appstate.edu) |
| 15:06.07 | azzie | Math[laptop], I had fedora core 4 and I had to reinstall it to FC3 because FC4 had gcc4 :( |
| 15:06.31 | *** join/#asterisk P4C0 (i=1000@201.224.107.47) |
| 15:06.37 | gvag11 | i have FC4 and i compiled Asterisk 1.2rc2 just fine... |
| 15:06.59 | hhoffman | azzie: what sort of things did you have problems with? my asterisk-1.2 box is FC4 |
| 15:07.22 | Vhata | my FC4 box also compiled it perfectly |
| 15:07.27 | azzie | hhoffman, for example try compiling festival |
| 15:07.28 | Aurix | hey, sorry for a possibly stupid question, but I used to use Traverse Technology's NetJet-S using the hisax module under 2.4, with a voice patch and i4l. I've been unable to find a voice patch for 2.6, wondering if the voice changes have been merged into linux 2.6, or what I need to do to get this card working under asterisk under 2.6? |
| 15:07.38 | *** join/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
| 15:08.13 | hhoffman | azzie: I haven't tried to compile it... but there are packages so that means that someone did |
| 15:09.35 | azzie | hhoffman, it does not mean gcc4 compiles festival as gcc3 does. Maybe somebody spent his vacation making it work with gcc4 :) |
| 15:10.14 | hhoffman | azzie: I certainly won't argue that ;-) |
| 15:10.44 | c0w | just zaptel isn't working on gcc4 |
| 15:10.49 | gvag11 | using spandsp 0.0.2pre21 and asterisk 1.2rc2 and TE205P, i am getting broken faxes(cut short pages) . Not all of the faxes i sent or receive although. Any idea? Is that because of frame slips? |
| 15:10.49 | c0w | thats rc2 |
| 15:11.01 | tmccrary | anyone have problems with comedian mail? |
| 15:11.13 | LostFrog | Other than the name, no. :) |
| 15:11.15 | hhoffman | zaptel compiled fine for me under FC4/gcc4 |
| 15:11.27 | tmccrary | my dial plan works great until the user is transffered to comedian mail, at that point I get all kinds of audio glitches and stutters |
| 15:11.37 | c0w | root@voicemail:/usr/src/asterisk/zaptel-1.2.0-rc2# modprobe wct4xxp |
| 15:11.37 | c0w | WARNING: Error inserting zaptel (/lib/modules/2.6.14.1/misc/zaptel.ko): Invalid module format |
| 15:11.40 | LostFrog | I'm still waiting for my users to notice the name. |
| 15:11.40 | c0w | WARNING: Error inserting zaptel (/lib/modules/2.6.14.1/misc/zaptel.ko): Invalid module format |
| 15:11.44 | c0w | FATAL: Error inserting wct4xxp (/lib/modules/2.6.14.1/misc/wct4xxp.ko): Invalid module format |
| 15:11.44 | tmccrary | hehe |
| 15:11.47 | c0w | FATAL: Error running install command for wct4xxp |
| 15:11.53 | LostFrog | c0w: ~pb |
| 15:11.57 | LostFrog | ~pb? |
| 15:11.58 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 15:11.58 | c0w | soz |
| 15:11.58 | hhoffman | o_0 |
| 15:12.01 | c0w | is only a couple of lines |
| 15:12.04 | Math[laptop] | c0w, get the new insmod/modprobe |
| 15:12.05 | ikarus | LostFrog: they won't with that pronounciation |
| 15:12.07 | P4C0 | hello guys, one quick question, I'll have a VoIP telephone provider, so, he's going to give me a sip phone, can I plug that "line" into asterisk and build my own phone network? like using a voip provider as an fxo line? |
| 15:12.08 | hhoffman | 2.6.15.1?? |
| 15:12.19 | Math[laptop] | c0w, 2.6 doest have the same kernel module format as 2.4 |
| 15:12.21 | ikarus | P4C0: ofcourse |
| 15:12.27 | P4C0 | ikarus: thank you :D |
| 15:12.27 | c0w | thing is |
| 15:12.33 | c0w | i have it running on 2.6.14.1 |
| 15:12.38 | c0w | using gcc3.5 |
| 15:12.44 | hhoffman | but he got .ko modules which should indicate a 2.6 module |
| 15:13.30 | c0w | and if i copy across the kernel modules to the gcc4 box |
| 15:13.33 | starsoft | I have two grandstream sip phones behind 2 different nats that can call out using a sip proxy and they work fine, although when i try to call one phone from the other all i get is dead air, when both phones were behind the same nat device (at my office) they talked to each other perfectly, can anyone suggest what i may need to investigate here? |
| 15:13.33 | c0w | it will load them |
| 15:14.03 | hhoffman | c0w: are you running the same kernel as the headers you are compiling against? |
| 15:14.07 | ikarus | starsoft: other then IPv6 so NAT can die a painful death, I would not know |
| 15:14.32 | c0w | should be |
| 15:14.38 | *** join/#asterisk testmachine (n=user@81-171-6-142.dsl.fiberworld.nl) |
| 15:15.14 | hhoffman | uname -r match /lib/modules/`uname -r`/ ? |
| 15:15.29 | hhoffman | that's the only thing that comes to mind |
| 15:15.52 | *** join/#asterisk razu (n=razu@tln-kontor.norby.ee) |
| 15:15.59 | hhoffman | especially if they are working on another box |
| 15:16.46 | ikarus | hmmmmm, that would be such a pleasure, IPv6 |
| 15:17.35 | starsoft | so |
| 15:17.41 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 15:17.44 | starsoft | we'll give ipv4 the hose |
| 15:17.49 | starsoft | but in the mean time |
| 15:18.01 | starsoft | what might cause this squirrely business |
| 15:18.04 | *** join/#asterisk lilwookie (n=lilwooki@modemcable155.254-70-69.mc.videotron.ca) |
| 15:18.21 | ikarus | hmmmmm, asterisk always also routes the RTP stream, right |
| 15:18.31 | starsoft | ive debugged the packets |
| 15:18.38 | ManxPower | ikarus, Not always |
| 15:18.46 | starsoft | and they are going from phone1 -> asterisk -> phone2 |
| 15:18.49 | starsoft | and vise versa |
| 15:19.16 | ikarus | starsoft: SIP or RTP ? |
| 15:19.16 | starsoft | ive trued dtmfmode, inband and info |
| 15:19.39 | starsoft | i see both sip and rtp traffic in the tcpdump |
| 15:20.18 | ManxPower | Asterisk will handle the RTP if it needs to listen to DTMF on the call, if the two devices are using 2 different codecs, if there is NAT involved. |
| 15:20.44 | starsoft | both phones are setup identically |
| 15:21.00 | ikarus | starsoft: my guess would be that Asterisk is unable to outbound connect RTP to the SIP phone in the NATted network |
| 15:21.01 | ManxPower | starsoft, what is the actual Dial line you are using? PASTE it. |
| 15:21.05 | mutilator | in the dialplan |
| 15:21.15 | ikarus | But I might be totallty wrong |
| 15:21.17 | mutilator | does . match no characters and any character? |
| 15:21.22 | mutilator | exten => _.9895632470,1 |
| 15:21.24 | *** join/#asterisk loud (n=ariel@cypher.punk.net) |
| 15:21.32 | mutilator | would match 19895632470 or just 9895632470 |
| 15:21.32 | ManxPower | mutilator, . must be the LAST character in a match |
| 15:21.40 | Vhata | ; . - wildcard, matches anything remaining (e.g. _9011. matches |
| 15:21.40 | Vhata | ; anything starting with 9011 excluding 9011 itself) |
| 15:21.48 | Vhata | 'anything remaining', except nothing |
| 15:21.53 | mutilator | so i'de need extens for that then |
| 15:22.07 | ManxPower | mutilator, what stuff do you want to match? |
| 15:22.08 | starsoft | [default] |
| 15:22.08 | starsoft | exten => _9.,1,Dial(SIP/${EXTEN:1}@1231231234,50) |
| 15:22.08 | starsoft | exten => _9.,2,Congestion |
| 15:22.08 | starsoft | exten => _9.,102,Busy |
| 15:22.08 | starsoft | exten => _500X,1,Dial(SIP/${EXTEN}) |
| 15:22.08 | starsoft | exten => _500X,2,Hangup |
| 15:22.24 | starsoft | but like i was saying |
| 15:22.32 | starsoft | when both phones were at my office they worked |
| 15:22.44 | ManxPower | starsoft, and are both legs of the call using the same codec, as shown by "sip show channels" when there is an active call between the two devices? |
| 15:22.46 | YoMama | anyone know how to get the MOH to be a Shoutcast stream? |
| 15:22.54 | starsoft | now that they are at diffrent locations only outbound calls work, phone to phone gets dead air |
| 15:22.59 | Dibbler | Anybody got openssl-devel for AM64? |
| 15:23.14 | Dibbler | Preferably .deb |
| 15:23.15 | ManxPower | Results 1 - 10 of about 37 from lists.digium.com for MOH shoutcast. (0.46 seconds) |
| 15:23.26 | YoMama | Dibbler: just download the source |
| 15:23.33 | starsoft | sip show channels |
| 15:23.33 | starsoft | Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Msg |
| 15:23.33 | starsoft | 192.168.0.5 5002 3be006935cf 00105/00000 ulaw No Tx: ACK |
| 15:23.33 | starsoft | 192.168.9.105 5001 cb0431891ec 00102/30587 ulaw No Tx: ACK |
| 15:23.33 | ManxPower | Results 1 - 10 of about 54 from lists.digium.com for music on hold shoutcast. (0.51 seconds) |
| 15:23.46 | YoMama | ManxPower: i can't find the "search" engine for the listserv |
| 15:23.58 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 15:24.04 | ManxPower | starsoft, as long as you don't have "canreinvite=no" it should work |
| 15:24.04 | kink0 | good morning |
| 15:24.06 | ManxPower | ~mailinglist |
| 15:24.07 | jbot | mailinglist is probably Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 15:24.27 | *** join/#asterisk john8675309tm (i=1001@207.177.124.3) |
| 15:24.30 | YoMama | ManxPower: duh..thanks :) |
| 15:24.35 | kink0 | I bougth two separate g729 licenses from Digium, both are sussefully installed on both linux PBX boxes |
| 15:24.50 | kink0 | but, now here is the problem , how to do both uses g729 ? |
| 15:24.51 | john8675309tm | is there any way to get a 410p to answer all calls handed to it? |
| 15:25.11 | mutilator | ManxPower i just want the exten to match a 10 or 11 digit dial |
| 15:25.12 | kink0 | I set disallow=all + allow=g729 at both sip.conf |
| 15:25.16 | ManxPower | john8675309tm, you mean like exten => _X.,1,Answer ? |
| 15:25.24 | mutilator | 11 digit being a 1 at the beginning |
| 15:25.34 | [TK]D-Fender | john8675309tm : yeah, use a catch-all exten for it like _x. |
| 15:25.35 | john8675309tm | ManxPower I will try that |
| 15:25.43 | kink0 | but that does not word and show g729 reports none in use |
| 15:25.57 | ManxPower | mutilator, exten => _1NXXNXXXXXX,1,NoOp(FNORD) |
| 15:26.13 | mutilator | ya manx |
| 15:26.14 | john8675309tm | that is awesome thank you!!!! |
| 15:26.19 | YoMama | ManxPower: u posted an answer 11/24/2004 :) |
| 15:26.24 | mutilator | i was just looking for the same exten to match both |
| 15:26.29 | mutilator | that won't match a 10 digit dial |
| 15:26.40 | mutilator | _1NXXNXXXXXX and _NXXNXXXXXX, |
| 15:26.43 | ManxPower | mutilator, not going to happen. |
| 15:26.58 | Dibbler | YoMama: Got a src, for a tar,gz src? ;-) |
| 15:27.00 | }cytrak{ | hm I want to use stream file $file,$digit .. does the digit need to be sent in asci ? |
| 15:27.36 | ManxPower | }cytrak{, I would have to check the AGI docs to know. |
| 15:27.51 | mutilator | didn't think so |
| 15:27.51 | }cytrak{ | it doesn't say anything |
| 15:28.07 | }cytrak{ | it just mentions enter a digit |
| 15:28.10 | c0w | uname -r match doesn't work ? |
| 15:28.25 | ManxPower | }cytrak{, based on my vague memory if the agi docs say "digit" they mean "0-9" |
| 15:28.32 | ManxPower | if it says ASCII, it means....ASCII |
| 15:28.37 | YoMama | Dibbler: if u don't wanna compile it, try finding it on rpmfind...otherwise, just go to www.openssl.org |
| 15:28.47 | kink0 | I got: chan_sip.c:3569 process_sdp: No compatible codecs! |
| 15:28.57 | starsoft | So the TX/RX should not be 000000 ? |
| 15:28.58 | kink0 | when I try to use g729 at both PBX |
| 15:29.06 | starsoft | that would mean that 5002 isnt sending data? |
| 15:29.09 | ManxPower | kink0, sounds like the remote device is not saying it supports G729 |
| 15:29.13 | [TK]D-Fender | kink0 : You have licences for G729? |
| 15:29.30 | }cytrak{ | anyone know a sound studio kind of app that i can use on linux to merge some wav files ? |
| 15:29.32 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 15:29.34 | kink0 | ManxPower, g729 have been sussefully registered in both, local and remote |
| 15:29.40 | ManxPower | kink0, you should prolly do a "sip debug peer sipconfpeername" and then try a call. |
| 15:29.51 | kink0 | [TK]D-Fender, yes, of course, in both, I bought TWO separate licenses to do this. |
| 15:30.09 | ManxPower | then pay attention to the rtpmap= which I think lists the codec Asterisk supports in 1 packet and the codecs the phone says it supports in another packet |
| 15:30.53 | starsoft | ame/username Host Dyn Nat ACL Mask Port Status |
| 15:30.53 | starsoft | 5002/5002 69.207.189.56 D N 255.255.255.255 5060 OK (61 ms) |
| 15:30.53 | starsoft | 5001/5001 66.67.243.13 D N 255.255.255.255 1026 OK (80 ms) |
| 15:31.05 | starsoft | why is one 1026 and the other 5060? |
| 15:31.16 | starsoft | if they are configured the same |
| 15:31.36 | [TK]D-Fender | Different SPORT vs DPORT? |
| 15:31.45 | [TK]D-Fender | Nat scenario? |
| 15:32.08 | starsoft | both phones were behind the same nat and they worked fine |
| 15:32.15 | starsoft | now one phone is behind another nat |
| 15:32.29 | starsoft | and now 5001 cant call 5002 |
| 15:32.32 | starsoft | and vise versa |
| 15:32.41 | starsoft | im going to dump my arp cache |
| 15:32.48 | starsoft | the interweb is stupid |
| 15:33.14 | kink0 | ManxPower, pls, help, what is sipconfpeername ? |
| 15:33.27 | kink0 | No such peer 'asterisk.interec.org' |
| 15:33.36 | ManxPower | kink0, the [whatever] name in sip.conf for that device. |
| 15:33.52 | kink0 | ahh ok , thanks |
| 15:33.57 | kink0 | I will try to debug now |
| 15:36.12 | kink0 | I have place disallow and allow for codecs in [general] in sip.conf at both ends |
| 15:37.34 | ManxPower | kink0, I don't recommend that. in [general] do an allow=all and then in EACH sip.conf section do a disallow=all and allow=whatever for the codec you want |
| 15:38.15 | *** join/#asterisk hikaro (n=devil@202.53.235.51) |
| 15:40.03 | Abbas | hello all |
| 15:43.00 | X-Files | People, probably such in asterisk. At present I speak by with the person to phone and I wish it to send in a mode hold line, and to call to other person or to answer other call after end of conversation I want will return to the first call (with whom before communicated) |
| 15:43.27 | Dibbler | DOH!!, Invalid module format |
| 15:44.37 | docelmo | Yippie! |
| 15:45.17 | lme | plop |
| 15:45.42 | Katty | yuppie? |
| 15:47.00 | }cytrak{ | the asterisk gsm files what are their common sample rate and channel ? anyone know that ? |
| 15:47.19 | *** join/#asterisk stars0ft (n=starsoft@cpe-66-67-243-13.rochester.res.rr.com) |
| 15:47.38 | zoa | 20ms |
| 15:47.52 | ManxPower | }cytrak{, define "sample rate" |
| 15:48.08 | echion | if I have one ast_channel *ast is there anyway I can find the other leg of the call? (it is not setup yet, but supposed to start dial) |
| 15:48.19 | ManxPower | ALL Asterisk codecs require 20ms paket size (except for iLBC?) |
| 15:48.58 | stars0ft | damn the interweb |
| 15:49.02 | coppice | and lpc10 |
| 15:49.08 | coppice | and g723.1 |
| 15:49.08 | Dibbler | Anybody know if there is an mpg123 for AM64? |
| 15:49.18 | zoa | Dibbler: yes probably |
| 15:50.38 | Dibbler | So you don't know |
| 15:50.55 | X-Files | People, probably such in asterisk. At present I speak by with the person to phone and I wish it to send in a mode hold line, and to call to other person or to answer other call after end of conversation I want will return to the first call (with whom before communicated) |
| 15:50.59 | zoa | i am 99,99% sure |
| 15:51.08 | zoa | just take a 64 bit distro |
| 15:51.19 | fourcheeze | Dibbler |
| 15:51.21 | fourcheeze | file /usr/bin/mpg123 |
| 15:51.21 | fourcheeze | /usr/bin/mpg123: symbolic link to `mpg123-x86_64' |
| 15:51.24 | ManxPower | coppice, asterisk doesn't use G723.1 and nobody actually USES LPC10 |
| 15:51.30 | Dibbler | I've got a 64 bit distro, hence the question, it has mpg321 |
| 15:51.41 | Dibbler | I don't even know if that is * compat |
| 15:51.54 | ManxPower | Dibbler, mpg321 will not work with Asterisk |
| 15:52.01 | Dibbler | That is whay I wa sasking |
| 15:52.19 | fourcheeze | Dibbler: file /usr/bin/mpg123-x86_64 |
| 15:52.19 | fourcheeze | /usr/bin/mpg123-x86_64: ELF 64-bit LSB executable, AMD x86-64, version 1 (SYSV), for GNU/Linux 2.4.1, dynamically linked (uses shared libs), stripped |
| 15:52.30 | fourcheeze | does that answer your question? |
| 15:52.41 | Dibbler | It does indeed |
| 15:52.43 | Dibbler | Thank you |
| 15:52.48 | }cytrak{ | ManxPower: sample rate could be 8000Hz for example |
| 15:53.12 | ManxPower | }cytrak{, all telcom codecs are 8000Hz |
| 15:53.35 | ManxPower | The only exception is some of the codecs that support Wideband mode, Speex and iLBC, but Asterisk does not support those modes. |
| 15:53.58 | }cytrak{ | I'm trying to open the gsm files with sweep |
| 15:56.44 | ikarus | ManxPower: which is annoying |
| 15:56.49 | coppice | ManxPower: and G..722, and G.722.1 and G.722.2 and WB-AMr |
| 15:57.42 | zoa | does somebody know what ilbc broadband costs ? |
| 15:58.28 | ikarus | ManxPower: G.722 is used |
| 15:58.34 | ikarus | heck, my BudgeTones support it |
| 15:58.55 | coppice | zoa: they only deal with volume licencing |
| 15:59.02 | ManxPower | copantl, not with Asterisk 8-) |
| 15:59.55 | zoa | yeah but still |
| 15:59.59 | zoa | what do they ask ? |
| 16:01.24 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
| 16:01.49 | *** join/#asterisk viperdude (n=jon@84.45.193.6) |
| 16:02.05 | viperdude | hi guys |
| 16:02.09 | loud | havent seen him in a couple of weeks |
| 16:02.17 | viperdude | anyone care to help me with a dialplan problem |
| 16:03.19 | _Sam-- | i have a bunch of SIP users that get dialed from an extension like exten => 1,1,Dial(SIP/1&SIP/2&SIP/3...etc)....is there a way to make it only dial an extension if that extension's channel is available? |
| 16:03.21 | Qwell | viperdude: if you ask a question |
| 16:04.09 | _Sam-- | so if SIP/2 is on the phone it wont try SIP/2 |
| 16:04.09 | Dibbler | fourcheeze: Worked a treat thnx |
| 16:04.09 | Qwell | _Sam--: I think if any of the channels in the list, it'll skip it |
| 16:04.09 | Qwell | in the list are busy* |
| 16:04.09 | ikarus | I would love Asterisk to be able to handle G.722 it would make it that much more useful, but I looked into it, Asterisk makes too many assumptions |
| 16:04.09 | _Sam-- | thats what i wanted, so perfect...thanks. |
| 16:04.13 | _Sam-- | i hadnt tested it |
| 16:04.33 | viperdude | i have asterisk answering a call and playing a message using BackGround(). The message plays ok but when I press 1 nothing happens even though I have a exten=> 1,1,Goto,default,450,1 in the same context. Any ideas whats wrong? |
| 16:05.00 | coppice | ikarus: people seem to have fudged g.722 passthrough, never telling * is idn't running at 8000 sample/second |
| 16:05.17 | Qwell | viperdude: Do you perhaps also have an exten => 1NXXNXXXXXX ? |
| 16:05.18 | _Sam-- | extenyour goto is wrong |
| 16:05.33 | viperdude | aha let me look |
| 16:05.37 | _Sam-- | goto (Default,450,1) |
| 16:05.46 | ikarus | coppice: I want real support, so my internal calls, etc sound better |
| 16:06.09 | coppice | ikarus: what does real support mean? |
| 16:06.36 | viperdude | Qwell: I dont have any other extens in the context starting with 1 |
| 16:06.43 | ikarus | coppice: rewrite asterisk to be samplerate agnostic and simply convert when needed |
| 16:06.51 | Qwell | viperdude: anything that starts with N or X perhaps? |
| 16:07.03 | coppice | ikarus: that's pointless and inefficient |
| 16:07.05 | Qwell | anything that could potentially match a 1, and wait for more digits |
| 16:07.11 | viperdude | Qwell are you talking in the same context as the BackGround? |
| 16:07.16 | Qwell | yes |
| 16:07.20 | ikarus | coppice: not really |
| 16:07.21 | viperdude | nope |
| 16:07.21 | Qwell | or anything included by it |
| 16:07.25 | _Sam-- | its the goto() |
| 16:07.47 | viperdude | nothing just a s, i, t and the 1 which has the goto command |
| 16:07.49 | ikarus | coppice: with the spread of VoIP it means that more and more we get freed from the stupid 8000hz limit |
| 16:08.10 | viperdude | nothing included in the context |
| 16:08.11 | ManxPower | ikarus, only for people not using actual phones |
| 16:08.14 | *** join/#asterisk fishboy1669 (i=proxyuse@62.69.81.129) |
| 16:08.15 | Qwell | viperdude: try what _Sam-- said |
| 16:08.21 | fishboy1669 | hi |
| 16:08.25 | coppice | ikarus: this is why you have negotiation. converting on the fly is wasteful |
| 16:08.27 | fishboy1669 | hows things |
| 16:08.29 | viperdude | Qwell you mean use () |
| 16:08.32 | LostFrog | I noticed something weird with 1.2 vs. 1.0.X |
| 16:08.33 | Qwell | sure |
| 16:08.43 | Qwell | LostFrog: lots of "weird" stuff between them |
| 16:08.46 | ManxPower | viperdude, i.e. people using softphones. In the future there may be IP phones that suport wideband, but I'm not aware of any at the moment |
| 16:08.50 | _Sam-- | are contexts case sensitive? ie does default == Default? |
| 16:08.51 | *** join/#asterisk frenzy (n=frenzy@193.220.82.108) |
| 16:08.53 | LostFrog | If I have 1100 and _1X. in the same context, it actuallys works correctly. |
| 16:09.01 | ikarus | ManxPower: my BudgeTone does G.722 |
| 16:09.01 | ManxPower | LostFrog, Correct. |
| 16:09.09 | Qwell | Things that are supposed to happen... |
| 16:09.13 | ikarus | ManxPower: and it is not exactly rare |
| 16:09.15 | coppice | ManxPower: lots of IP phones support wideband |
| 16:09.21 | LostFrog | Which I don't think worked right in 1.0.X |
| 16:09.25 | coppice | most only do G.722, though |
| 16:09.29 | Qwell | LostFrog: pretty sure it did |
| 16:09.38 | viperdude | Qwell: that made no difference |
| 16:09.50 | frenzy | hi all |
| 16:09.56 | Qwell | viperdude: Do you get anything on the CLI with the verbose up? |
| 16:09.56 | frenzy | can some one help me with this warning |
| 16:09.57 | frenzy | channel.c:1449 ast_indicate: Unable to handle indication 3 for |
| 16:09.57 | _Sam-- | viperdude: what phone are you using, maybe its not sending the tones right |
| 16:09.59 | viperdude | I am using a cisco 7940 series phon |
| 16:10.04 | ikarus | coppice: negotiation is fine, unless in mid communication you need to switch |
| 16:10.06 | Qwell | ah hah |
| 16:10.08 | Qwell | dtmfmode |
| 16:10.20 | viperdude | CLI shows Background plays but nothing when I press keys |
| 16:10.26 | ikarus | coppice: or you have a conference with people on both traditional and wideband |
| 16:10.31 | *** join/#asterisk Uther_P (n=uther_p@66.180.120.82) |
| 16:10.31 | Qwell | viperdude: make sure the dtmfmode in sip.conf and on the cisco are the same |
| 16:10.35 | viperdude | dtmf works fine with voicemail |
| 16:10.43 | _Sam-- | yeah i had the same problem with a grandstream last week using call parking, wouldnt recognize the # sign |
| 16:10.55 | viperdude | alreasy tested dtmf with voicemail |
| 16:11.02 | frenzy | ? |
| 16:11.03 | Qwell | _Sam--: # usually means "okay, I'm done dialing. Send the call now" |
| 16:11.04 | coppice | ikarus: conferencing is a little different |
| 16:11.07 | frenzy | channel.c:1449 ast_indicate: Unable to handle indication 3 for... |
| 16:11.22 | fishboy1669 | anyone here ever set up loadballanceing on asterisk boxes |
| 16:11.24 | fishboy1669 | ha |
| 16:11.27 | ikarus | coppice: true, but I also mean being able to handle a asterisk side transfer |
| 16:12.03 | coppice | for a transfer you should renegotiate |
| 16:12.15 | *** part/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 16:12.27 | ikarus | coppice: I don't think SIP really supports that out of the box |
| 16:12.47 | coppice | yes it does. reinvite |
| 16:13.07 | coppice | that is how you do things like switch into t.38 mode |
| 16:13.20 | _Sam-- | viperdude, i know this sounds stupid, but your background(message), message is playing? |
| 16:13.34 | *** join/#asterisk Conductor (n=thomas@62.8.240.185) |
| 16:13.34 | viperdude | yes background is playing |
| 16:13.49 | _Sam-- | so it has to be something with the way the phone is sending the tones, in my opinion |
| 16:14.04 | _Sam-- | if you dont see it receiving the input from the CLI |
| 16:14.24 | viperdude | <_Sam-->: How would voicemail work if it was DTMF problem? |
| 16:14.31 | _Sam-- | do you have a softclient or anything else you can test with? |
| 16:14.38 | viperdude | yes x-lite |
| 16:14.41 | viperdude | hang on |
| 16:14.44 | _Sam-- | worth a shot |
| 16:15.17 | *** join/#asterisk Sobakai (n=jmwoodga@45.e6.d12c.cidr.airmail.net) |
| 16:15.54 | viperdude | x-lite doesn't work either |
| 16:16.08 | *** join/#asterisk gvag11 (n=g@84.254.12.133) |
| 16:16.11 | gvag11 | hi |
| 16:16.24 | viperdude | this is really frustrating lol |
| 16:16.35 | *** join/#asterisk b000m (n=boom@opencode.tea.bg) |
| 16:16.40 | _Sam-- | sorry , im out of options |
| 16:16.59 | *** join/#asterisk toddf (n=toddf@net-66-210-104-120.theshop.net) |
| 16:17.03 | gvag11 | having cut short pages using spandsp mean there is a problem with frame slips? |
| 16:17.03 | kink0 | puffff,. unable to get running my g729 |
| 16:17.23 | Sobakai | g'morning everyone =) |
| 16:17.25 | *** join/#asterisk JASON-0 (n=jason@jason.unitz.ca) |
| 16:17.26 | kink0 | I will come later, need to read a bit more about to config it. |
| 16:17.32 | kink0 | cu later |
| 16:17.51 | b000m | kink0 you have to applay patch |
| 16:18.09 | viperdude | is it possible for DTMF to work for Voicemail but not for a IVR app? |
| 16:18.33 | JASON-0 | Does anyone know how to set the distintive ring options in the extensions.conf for a SIP device? |
| 16:18.50 | ManxPower | viperdude, yes. |
| 16:19.07 | gvag11 | having cut short pages using spandsp mean there is a problem with frame slips? |
| 16:19.26 | viperdude | how would i get it to work with Background cmd ManxPower? |
| 16:19.26 | ManxPower | viperdude, but the answer is usually "no" IF you are calling Voicemail and the IVR using the same phone using the same codec, using the same DTMF mode. |
| 16:19.39 | viperdude | yes same phone |
| 16:19.56 | ManxPower | If the calls into the IVR are coming from the PSTN and calls to the voicemailmain are via a SIP phone then you have some other problem |
| 16:20.08 | viperdude | aha this could be it... |
| 16:20.08 | ManxPower | viperdude, SIP phone? |
| 16:20.24 | viperdude | I am dialling the IVR via our SIP provider |
| 16:20.38 | ManxPower | if you have DTMF problems for calls coming in over a Zaptel card, then it's usually a volume problem |
| 16:20.43 | Conductor | i cannot set the callerId to 0 when using SetCallerID |
| 16:20.58 | ManxPower | Conductor, try setcalleridnum |
| 16:21.42 | JASON-0 | Does anyone know how to set the distintive ring options in the extensions.conf for a SIP device? |
| 16:21.56 | ManxPower | JASON-0, That TOTALLY depends on the SIP device |
| 16:22.00 | ecto | I have a TE210p card, with 2 PRI lines attached to it. Each span has 23 voice channels on it, numbered 1-23 for each span. How do I configure zapata.conf to use both spans? Since I have two channels for each number, how do I get Asterisk to distinguish between the two? |
| 16:22.12 | tzanger | JASON-0: repeating every few minutes tends to get you ignored. Besides you should be consulting the manual for your SIP device |
| 16:22.13 | ManxPower | you would SetVar(_ALERT_INFO=SOMETHING) where SOMETHING depends on your phone |
| 16:22.20 | JASON-0 | ManxPower: Using a Cisco 7960, but what I need to know is how to configure it |
| 16:22.41 | ManxPower | JASON-0, the mailing list archive will tell you, as will the Wiki |
| 16:22.43 | ManxPower | !mailinglist |
| 16:22.48 | ManxPower | ~mailinglist |
| 16:22.49 | jbot | i heard mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 16:22.50 | ManxPower | ~docs |
| 16:22.51 | jbot | [docs] probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 16:22.58 | tzanger | ~thebook |
| 16:23.00 | jbot | well, thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 16:23.23 | *** part/#asterisk johnrage (n=jabetong@212.93.201.89) |
| 16:23.33 | JASON-0 | Thanks |
| 16:23.44 | ManxPower | gvag11, that is the most common cause of failed faxes with spandsp |
| 16:24.04 | *** join/#asterisk jon335 (n=jon335@ottawa-hs-64-26-167-13.d-ip.magma.ca) |
| 16:24.19 | *** join/#asterisk Rowter (n=SilverDr@201.135.26.195) |
| 16:24.32 | zoa | ManxPower, can you add the asteriskguru search thing to that mailinglist line in the bot ? |
| 16:24.51 | _Sam-- | zoa: nice job on the guru, i like that place |
| 16:24.53 | zoa | http://www.asteriskguru.com/archives/search.php |
| 16:24.56 | zoa | thanks! |
| 16:25.04 | Rowter | there is any way to detect when a fax machine answeres an outgoing call. NV_FaxDetect and the zaptel fax detect seem to only work in calls originated FROM a fax machine, not for calls ANSWERED by a fax. |
| 16:25.10 | zoa | we just added a search for the mailinglists |
| 16:25.17 | zoa | updated every 5 minutes or so |
| 16:25.19 | _Sam-- | do you have anything to do with the idefisk phone? |
| 16:25.22 | zoa | yes |
| 16:25.28 | _Sam-- | i figured, i like that too |
| 16:25.36 | ManxPower | zoa, I don't know how to do that without screwing up the existing info |
| 16:25.37 | zoa | linux version is coming in a few days |
| 16:25.49 | zoa | dont touch it then :) |
| 16:26.14 | zoa | ~mailinglist is i heard mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search it at http://www.asteriskguru.com/archives/search.php |
| 16:26.15 | jbot | ...but mailinglist is already something else... |
| 16:26.34 | zoa | ~mailinglist |
| 16:26.35 | jbot | rumour has it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 16:26.39 | ManxPower | zoa, you can take out the google info if you are SURE your search works well |
| 16:26.41 | fishboy1669 | any ideas on the asterisk loadballanncing? |
| 16:26.53 | ManxPower | fishboy1669, Yes. Don't bother. 8-) |
| 16:26.57 | *** part/#asterisk jon335 (n=jon335@ottawa-hs-64-26-167-13.d-ip.magma.ca) |
| 16:27.01 | zoa | manx, how the hell does that both work :) |
| 16:27.08 | zoa | jbot, no |
| 16:27.10 | jbot | YES |
| 16:27.10 | zoa | p |
| 16:27.11 | zoa | :p |
| 16:27.12 | zoa | now behave |
| 16:27.20 | zoa | somebody knows how that thing works ? |
| 16:27.34 | zoa | ~mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search it at http://www.asteriskguru.com/archives/search.php |
| 16:27.35 | jbot | ...but mailinglist is already something else... |
| 16:27.38 | gvag11 | manxpower, how can i detect frame slips and i suppose that can happen because of IRQ sharing or latency, right? |
| 16:27.47 | zoa | jbot, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search it at http://www.asteriskguru.com/archives/search.php |
| 16:27.48 | jbot | ...but mailinglist is already something else... |
| 16:27.56 | zoa | aaarghl |
| 16:27.57 | ManxPower | ~mailinglist |
| 16:27.58 | jbot | somebody said mailinglist was Search Asterisk mailing lists at http://www.asteriskguru.com/archives/search.php or Browse the mailing list archive at http://lists.digium.com/ |
| 16:28.10 | ManxPower | There. |
| 16:28.33 | ManxPower | gvag11, frame slips are usually a timing problem in the span= line of /etc/zaptel.conf |
| 16:28.38 | Conductor | ManxPower, the problem is, that this is overridden by our default number |
| 16:28.51 | ManxPower | IRQ misses would show up as HDLC aborts on the CLI if you are using PRI |
| 16:29.08 | ManxPower | Conductor, you mean for outgoing calls to the PSTN? |
| 16:29.48 | *** join/#asterisk marc324 (n=marc3234@206-248-128-159.dsl.teksavvy.com) |
| 16:30.25 | Conductor | ManxPower, outgoing calls through pmx |
| 16:30.40 | ManxPower | Conductor, I have no idea what a "pmx" is. |
| 16:30.48 | InfraRed | http://www.sonystyle.com - Do a search for USB (case sensitive). |
| 16:30.55 | gvag11 | manxpower, i have a a TE205P and the span 1 connects to span 2 (with E1 crossover cable) the timing there is for the first span to be master and slave... |
| 16:31.20 | Conductor | ManxPower, pbx... |
| 16:31.22 | ManxPower | gvag11, paste the two span= lines (and only those two lines) |
| 16:31.41 | Conductor | ManxPower, and yes, to the PSTN (in this case via German Telekom) |
| 16:31.55 | ManxPower | Conductor, So it's Call -> Asterisk -> PBX -> phone or is it Call -> Asterisk -> PBX -> PSTN? |
| 16:32.29 | ManxPower | Conductor, what the PSTN displays on the far end as callerid is up to your carrier. Many carriers don't let you to set your outgoing callerid number to anything except a DID/DDI you have. |
| 16:32.47 | Conductor | ManxPower, Call -> Asterisk -> PBX -> PSTN |
| 16:32.56 | cpatry | ~mailinglist |
| 16:32.57 | jbot | well, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search it at http://www.asteriskguru.com/archives/search.php |
| 16:33.03 | cpatry | like that? |
| 16:33.07 | zoa | how the fuck do you do that |
| 16:33.09 | zoa | :) |
| 16:33.19 | Conductor | ManxPower, that could be it. how do i find out? call them? |
| 16:33.40 | ManxPower | Conductor, that's the only way to know. |
| 16:33.42 | gvag11 | manxpower, here it is - span=1,1,0... & span=2,0,0.... |
| 16:33.52 | Conductor | ManxPower, ok thanks. |
| 16:33.54 | ManxPower | gvag11, that would work fine. |
| 16:34.09 | ManxPower | Conductor, Of course the PBX may not allow setting the callerid number. |
| 16:35.04 | gvag11 | manxpower, i know but still have problems with still active channels after sending and receiveing faxes and cut short pages (not all of them) |
| 16:35.13 | zoa | ~mailinglist |
| 16:35.15 | jbot | hmm... mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php |
| 16:35.20 | zoa | k |
| 16:35.22 | ManxPower | gvag11, Don't know what to suggest. |
| 16:35.35 | zoa | the search thing on asteriskguru works fine, but dont want to remove someone elses entry |
| 16:35.43 | gvag11 | manxpower, thanks anyway... |
| 16:37.15 | LostFrog | is ',' not a valid character in caller ID? |
| 16:37.25 | ManxPower | LostFrog, in the name it is. |
| 16:38.02 | LostFrog | Hmm.. then the guy at broadvoice is a poofdah. |
| 16:38.16 | LostFrog | I told him to make it "Sris, P.C." and it came out as "P.C. ." |
| 16:38.17 | Samoied | anyone have tested the new digium card - TDM2400? |
| 16:39.05 | LostFrog | Samoied: people outside of Digium have them? |
| 16:39.11 | Nugget | I remember when 2400 baud was cool. |
| 16:40.12 | Samoied | LostFrog: Its not for sale? |
| 16:40.26 | LostFrog | Samoied: I think it hasn't been officially released yet. |
| 16:40.31 | LostFrog | I could be wrong. |
| 16:40.37 | LostFrog | It's not in the digium store. |
| 16:41.23 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 16:41.42 | Samoied | LostFrog: Ok, in the preesrelease: "will be available from Asterisk resellers and distributors worldwide beginning December 2005." |
| 16:41.51 | Samoied | LostFrog: :) |
| 16:42.48 | LostFrog | voipsupply says 11/18/05 |
| 16:42.56 | LostFrog | Which would be friday. |
| 16:43.08 | TheCops | I'm using asterisk at my business and when I'm calling at a specific place (my ISP) and I'm going to leave a message to a mailbox at this place, The line hang up after I'm hearing the *beep* sound. |
| 16:43.10 | *** join/#asterisk soop (n=soop@CPE00055d221a57-CM0014048df602.cpe.net.cable.rogers.com) |
| 16:43.33 | Samoied | LostFrog: Its much time for me :) I WANT THIS CARD! :) |
| 16:43.47 | LostFrog | Samoied: suck it up. :) |
| 16:43.48 | Samoied | LostFrog: I hate channel-banks |
| 16:44.03 | LostFrog | TheCops: probably a *feature* of the ISP. <Grin> |
| 16:44.10 | TheCops | LostFrog, lol |
| 16:44.16 | LostFrog | I'm sure if it was sales, it would work. |
| 16:44.23 | *** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca) |
| 16:44.26 | yxa | Max retries exceeded to host 218.104.198.138 on IAX2/abcd-4 (type = 6, subclass = 11, ts=330028, seqno=59) |
| 16:44.35 | Samoied | LostFrog: My only question is the price... |
| 16:44.39 | TheCops | Samoied, a new card will replace channel banks ? |
| 16:44.50 | Qwell | Samoied: voip-info has pricing. voip-info is usually a bit high |
| 16:44.55 | LostFrog | Samoied: |
| 16:44.58 | Samoied | TheCops: yep, This card up to 24 FXS or FXO ports |
| 16:45.02 | *** join/#asterisk rjuan (n=rjuan@233.Red-217-127-61.staticIP.rima-tde.net) |
| 16:45.03 | TheCops | LostFrog, This is weird, anywhere else I'm calling and leaving a message is working. |
| 16:45.04 | LostFrog | oops. |
| 16:45.15 | TheCops | Samoied, PCI ? or an external box ? |
| 16:45.22 | Qwell | pci |
| 16:45.22 | Samoied | TheCops: PCI |
| 16:45.27 | TheCops | Where can I see specs ? |
| 16:45.29 | queuetue | How would I "add an person" to the current call? IE, I'm on a call and I want to include someone from the design department on the current call... (I'm using asterisk@home/amportal, but a general answer is fine...) |
| 16:46.02 | rjuan | hi |
| 16:46.08 | Qwell | TheCops: digium.com |
| 16:46.12 | LostFrog | look at www.voipsupply.com |
| 16:46.35 | *** join/#asterisk funxion (n=nunya@mtnuser.icgws.com) |
| 16:46.50 | Samoied | Qwell: pricing, Where? I dont see anything abou TDM24XX in voip-info |
| 16:47.16 | rjuan | anybody knows how to play mp3 files in asterisk? |
| 16:47.31 | rjuan | I'm using debian |
| 16:47.39 | rjuan | and mpg123 |
| 16:47.42 | funxion | on an inbound zap e1 cas call is there a way to not send answer sup till call is answered on the outbound side? |
| 16:48.04 | fourcheeze | rjuan: show application MP3Player |
| 16:48.05 | funxion | rjuan I think you need mpg321 |
| 16:48.28 | funxion | i went through it before but cat remember what I did atm |
| 16:49.17 | Samoied | $1,699.95 - for 24FXS with echo cancelation |
| 16:49.38 | rjuan | it uses mpg123 |
| 16:49.55 | funxion | I know mpg123 doesnt werk ryte with debian for some reason you need mpg321 |
| 16:49.55 | Samoied | $1,469.95 - w/o EC |
| 16:50.20 | funxion | I cant remember why but I know Im using mpg321 to play mp3 on hold music |
| 16:50.30 | funxion | and it werx most importantly |
| 16:50.43 | rjuan | Have I to configure something |
| 16:50.52 | funxion | you have to install and configure |
| 16:51.07 | rjuan | i have both |
| 16:51.15 | rjuan | mpg123 and mpg321 |
| 16:51.21 | funxion | is this for moh? |
| 16:51.31 | Samoied | Its much for Latin-America market.... |
| 16:52.01 | lunk | if you have placed someone on hold, can you still process any input they send? Like, could someone on hold press a number to select a music genre? |
| 16:53.20 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 16:53.27 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 16:54.00 | LostFrog | It seems you should be able to dial 0 to get an operator while you are on hold. |
| 16:54.17 | LostFrog | I don't know whether it works in *.. |
| 16:54.26 | justinu | someone else here was allowing voting on MoH w/ * |
| 16:54.35 | queuetue | Does digium still do ad-hoc IVR recordings as a service? I seem to recall they used to, but can't find it now. |
| 16:54.48 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 16:55.09 | perd | has anyone attempted faxing over voip |
| 16:55.10 | funxion | does anyon know the answer to my e1 cas question |
| 16:55.14 | funxion | perd yes |
| 16:55.19 | funxion | got it to werk once |
| 16:55.28 | funxion | but only one pag fax |
| 16:55.47 | funxion | since then there hass been improvments in t38 pass-through and I have not tried the new code |
| 16:55.54 | MRH2 | quick ? - in telco speak a "line plant" = line to the exchange? |
| 16:56.14 | justinu | MRH2: line plant is the entire wiring system to the exchange |
| 16:56.32 | JASON-0 | I'm trying to search for ways to make the phones ring differently depending on what number is dialed, but I'm not sure the wording to search for? Can someone assist me on finding what I'm looking for? |
| 16:56.34 | justinu | from CPE to CO |
| 16:56.36 | perd | so it's pretty unreliable unless t38 is a large improvement? |
| 16:56.57 | justinu | JASON-0: distinctive ring |
| 16:57.19 | MRH2 | ok thanks... as long as it doesn't need watering.. |
| 16:57.25 | justinu | nope :) |
| 16:57.31 | funxion | so no on knows a way of not passing answer sup on e& wink circuit until the other side of the call is connected? |
| 16:57.42 | justinu | in telco terms plan == physical infrastructure |
| 16:57.54 | justinu | plant |
| 16:58.20 | MRH2 | cool thanks |
| 16:58.28 | justinu | funxion: that shouldn't be an issue |
| 16:58.34 | funxion | what is the config |
| 16:59.03 | funxion | is it in the zapata or do I just not answer() the xcall |
| 16:59.06 | justinu | i dunno about zap, but in your dialplan don't answer the channel |
| 16:59.07 | TheCops | Samoied, this is really nice this card... |
| 16:59.19 | funxion | ok |
| 16:59.24 | justinu | funxion: that's right, let dial pass the answer sup onto the caller |
| 17:00.01 | Conductor | how do i dial a *31# before a number? |
| 17:00.21 | syzygyBSD | queuetue: http://store.digium.com/product_view.php?category=8&product_code=IVR50 |
| 17:00.21 | Conductor | Dial(ZAP/g1/*31#${EXTEN}) does not work |
| 17:00.41 | TheCops | Samoied, the card are so tall |
| 17:00.53 | TheCops | There's no info on the lenght |
| 17:01.06 | Samoied | TheCops: full-length PCI |
| 17:01.19 | TheCops | yeah, but what that mean, this is a standard ? |
| 17:01.31 | Samoied | TheCops: Yep |
| 17:01.39 | syzygyBSD | Conductor: try to put a pause in there |
| 17:01.41 | TheCops | ok, in inch, do you know what is mean ?! :) |
| 17:01.45 | _Sam-- | i have a bunch of SIP users that get dialed from an extension like exten => 1,1,Dial(SIP/1&SIP/2&SIP/3...etc)....is there a way to make it only dial an extension if that extension's channel is available? |
| 17:01.52 | LostFrog | Lazy question: is there a way to write to a file from the dialplan? |
| 17:01.53 | _Sam-- | so if SIP/2 is on the phone, it wont try SIP/2? |
| 17:02.08 | justinu | LostFrog: system? |
| 17:02.16 | yxa | iax to zap is very soft. should i adjust the rxgain/txgain? |
| 17:02.25 | Samoied | _Sam--: Its implemented in client |
| 17:02.32 | *** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net) |
| 17:02.39 | LostFrog | justinu: That would work, I was just looking for a more elegant solution, one that doesn't fork a process. |
| 17:02.40 | Conductor | syzygyBSD, how do i pause? |
| 17:03.02 | perd | does a 'virtual modem' exist for SIP? by this i mean.. a kernel module or software that creates a tty that hylafax or wahtever could connect to and emulates a standard AT command set, but connects to asterisk for dialing out and making fax connections/. |
| 17:03.05 | justinu | Lostfrog: might look into the asterisk logger system |
| 17:03.13 | justinu | not sure if that can do what you want tho |
| 17:03.28 | perd | err dialing into the asterisk system.. dunno where the hell logger came from |
| 17:03.36 | syzygyBSD | Conductor: I think it is a 'w' |
| 17:03.45 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
| 17:03.58 | syzygyBSD | so Dial(ZAP/g1/*31#ww${EXTEN}) |
| 17:03.58 | shmaltz | ~seen tzafrir |
| 17:04.02 | jbot | tzafrir <n=tzafrir@local.xorcom.com> was last seen on IRC in channel #asterisk, 37d 23h 47m 46s ago, saying: 'quasi2k, try #asterisk-de (is there such a channel?)'. |
| 17:04.39 | shmaltz | tzfrir_laptop, you have a min? |
| 17:04.49 | InfraRed | xorcom |
| 17:05.05 | syzygyBSD | each w is 1/2 second |
| 17:05.50 | Rawplayer | re |
| 17:07.15 | Conductor | syzygyBSD, no, does not help... |
| 17:07.29 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 17:08.06 | syzygyBSD | ok, somewhat off question, why do you want to dial that first? |
| 17:08.43 | lehel | guys, you know how to hack a PDF document? pls |
| 17:08.59 | syzygyBSD | uh, what do you mean hack? |
| 17:09.14 | lunk | lehel: with adobe acrobat. |
| 17:09.46 | syzygyBSD | do you mean create, view, unpassword protect, or take an axe to it... |
| 17:09.47 | shmaltz | ~seen tzafrir_laptop |
| 17:09.49 | jbot | tzafrir_laptop is currently on #asterisk (9h 31m 53s) |
| 17:10.23 | lehel | syzygyBSD: unpassword it, i need to be able copy/paste -ing |
| 17:12.50 | lehel | syzygyBSD, give me a hint please |
| 17:13.28 | lehel | lunk, very smart;) |
| 17:13.29 | syzygyBSD | do a google search for pdf password free |
| 17:13.54 | lehel | k;) |
| 17:14.43 | zoa | olle!!!! |
| 17:15.01 | zoa | :) |
| 17:15.12 | perd | damnit |
| 17:15.23 | perd | i needa sip fax client so i can ship out faxes over voip :( that would be sexy |
| 17:16.46 | syzygyBSD | I think all my company has been able to do is the first page over voip, but we didn't really try that hard |
| 17:17.10 | syzygyBSD | have 3 other ways to send them so it wasn't really worth the time |
| 17:17.10 | LostFrog | Hmm.. I should be using ChanIsAvailable |
| 17:17.26 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 17:18.09 | perd | syzygy did you guys use some sort of sip client to perform the fax transfer or did you use hylafax and a modem connected to the asterisk server? |
| 17:18.44 | perd | i want to use a sip fax client basically.. punch a number in, it conects to my asterisk pbx via sip, dials, gets a fax machine on pstn and then i go :D |
| 17:18.57 | perd | but it looks like everyone else that asked this question got the same response.. none :P |
| 17:19.02 | perd | damn the forums |
| 17:19.07 | *** part/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr) |
| 17:20.11 | syzygyBSD | on one server we had hylafax, we also have a fax gateway we can email too |
| 17:20.24 | fishboy1669 | hi manx why do u say dont bother to ha? |
| 17:21.40 | perd | hrm |
| 17:21.52 | syzygyBSD | perd: have you read http://www.voip-info.org/wiki-Asterisk+fax |
| 17:22.09 | perd | yeah, i've been poking around t.38 info |
| 17:22.18 | syzygyBSD | specifically the section "Sending a fax to a SIP device" |
| 17:23.33 | perd | that isnt what i want to do though.. i want a sip client to originate the fax, and send it out to pstn |
| 17:23.45 | perd | app_rxfax is possibly what i need udnno |
| 17:23.48 | *** join/#asterisk Elven (i=elven@ptr-42.fw.swordcoast.net) |
| 17:23.54 | Elven | hi there |
| 17:24.24 | InfraRed | hai2u |
| 17:24.47 | ikarus | funxion: mpg123 works fine in Debain |
| 17:24.50 | syzygyBSD | perd you have a sip fax machine? or a fax - > ATA -> asterisk? |
| 17:24.50 | ikarus | wow |
| 17:24.55 | ikarus | I was up in scroll back |
| 17:24.56 | ikarus | lol |
| 17:25.18 | LostFrog | Debain? lol |
| 17:25.38 | fishboy1669 | night |
| 17:25.47 | syzygyBSD | i love debian for servers |
| 17:25.53 | perd | i have no fax machine |
| 17:26.10 | perd | i'm looking computer with sip client -> Asterisk -> PSTN |
| 17:26.14 | perd | for fax |
| 17:26.19 | LostFrog | perd: SpanDSP? |
| 17:26.30 | perd | i was just looking at that lastfrog :) |
| 17:26.35 | perd | txfax :) |
| 17:26.36 | Elven | are there any guides on how to handle the Dial command properly? im doing local -> SIP forwarding now. i'd like to inform the local user if the call succeeded, failed or otherwise; and i want to continue with the local user if the remote party hangs up (now the Dial extension returns -1 and this kills the local channel |
| 17:26.56 | LostFrog | Elven: show application dial |
| 17:27.06 | Elven | oh, ah, thanks |
| 17:27.08 | Elven | will read that |
| 17:27.11 | LostFrog | There is a setting to continue the call after remote hangup. |
| 17:27.35 | LostFrog | I believe some variables are set as to the result of the call as well. |
| 17:27.44 | syzygyBSD | Elven: http://www.voip-info.org/wiki/index.php?page=Asterisk%20cmd%20Dial |
| 17:28.02 | Elven | found those, but i couldnt evaluate those because of the dial cmd terminating :) |
| 17:28.24 | LostFrog | ,g |
| 17:28.28 | perd | thanks for the help, i believe i have my answers now! |
| 17:29.22 | *** join/#asterisk rjuan (n=rjuan@233.Red-217-127-61.staticIP.rima-tde.net) |
| 17:30.59 | rjuan | hi |
| 17:30.59 | rjuan | i've got a problem |
| 17:30.59 | rjuan | when I play a mp3 file |
| 17:30.59 | rjuan | in asterisk |
| 17:30.59 | rjuan | i've got this message |
| 17:31.01 | rjuan | Nov 16 18:26:30 WARNING[13021]: chan_sip.c:1836 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/64) |
| 17:31.04 | rjuan | any idea? |
| 17:31.21 | Elven | ok LostFrog, thanks :) found it |
| 17:31.25 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-100.rockynet.com) |
| 17:31.43 | Elven | whats the meaning of "transfer" (dial option T,t) in that cmd context? |
| 17:31.53 | Conductor | has any1 ever implemented a dial execution with a *XX# prefix? |
| 17:32.02 | Conductor | i mean, there has to be one? |
| 17:32.34 | *** join/#asterisk insurin1 (n=root@82-42-19-166.cable.ubr01.knor.blueyonder.co.uk) |
| 17:33.07 | *** part/#asterisk shmooz (n=shmooz@H142.C72.B0.tor.eicat.ca) |
| 17:33.09 | LostFrog | Elven: Now.. I have to tell you to RTFW. |
| 17:33.11 | LostFrog | ~wiki? |
| 17:33.13 | jbot | i heard wiki is http://www.voip-info.org |
| 17:33.19 | Elven | okay |
| 17:33.29 | LostFrog | No offense. |
| 17:33.37 | rayvd | None taken! =-o =-o |
| 17:33.42 | InfraRed | the wiki offends me |
| 17:34.07 | *** join/#asterisk Nivex (i=kjotte@user-0c8hq5r.cable.mindspring.com) |
| 17:34.26 | syle | to much manhood for you? |
| 17:34.37 | LostFrog | ewoks offend me. :) |
| 17:34.40 | *** part/#asterisk oej (n=oej@apollo.webway.se) |
| 17:34.50 | syle | lol |
| 17:34.50 | LostFrog | emacs just confuses me. |
| 17:35.08 | syle | i use nano, can't help you there :) |
| 17:35.20 | LostFrog | ewww.. Might as well use a GUI, syle. |
| 17:35.34 | *** join/#asterisk ManxPower (n=eric@adsl-67-65-233-194.dsl.lgvwtx.swbell.net) |
| 17:35.44 | syle | i don't need a gui |
| 17:35.46 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 17:36.09 | rjuan | anybody knows what does mean this error: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/64)??? |
| 17:36.33 | ManxPower | rjuan: "show codecs" will tell you what formats 4 and 64 is |
| 17:36.37 | ManxPower | and 1 |
| 17:36.38 | syzygyBSD | ewoks give me nightmares |
| 17:36.51 | syzygyBSD | crazy little people that eat you! |
| 17:37.18 | ^Howler | syzygyBSD: are you thinking of clowns? |
| 17:37.33 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 17:37.50 | rjuan | MaxPower thanks |
| 17:37.50 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-55-91.38-151.net24.it) |
| 17:38.01 | syzygyBSD | those are only in 'It'. Those gave me nightmares too, but ewoks are worse because they really exist |
| 17:38.24 | shido6 | LOL |
| 17:38.29 | LostFrog | ewoks are real? |
| 17:38.35 | syle | what planet you from? |
| 17:38.35 | LostFrog | Ewww. |
| 17:38.53 | LostFrog | syle: it was a moon. |
| 17:39.09 | LostFrog | Endor |
| 17:39.22 | LostFrog | He is obviously from the Moon of Endor. |
| 17:39.52 | syle | does the moon of psychiatry revolve around that one? |
| 17:40.12 | *** part/#asterisk rculp (n=rculp@66.173.240.20) |
| 17:40.14 | LostFrog | I've never mooned a psychiatrist. |
| 17:40.34 | LostFrog | I suppose that is a good way to get institutionalized. |
| 17:40.53 | ManxPower | rjuan: 1) never /msg me. 2) you never want allow=all You want disallow=all and allow= lines for each codec you want |
| 17:41.31 | rjuan | ok |
| 17:41.34 | syle | 1) use /msg 2) wtf are you taking about |
| 17:41.36 | rjuan | thanks |
| 17:43.21 | obsidian-studios | great time 4 me to start paying attention :) |
| 17:43.37 | mutilator | http://www.local6.com/health/5302544/detail.html WTF |
| 17:44.15 | ManxPower | obsidian-studios: why pay attention? It's always codec problems, registration problems, problems with crappy hardware, softphones, etc. |
| 17:44.39 | obsidian-studios | ManxPower: for all the other stuff, this channel is very entertaining :) |
| 17:44.42 | lunk | mutilator: FEAR |
| 17:45.01 | obsidian-studios | ManxPower: so newb problems ;) |
| 17:45.12 | InfraRed | mutilator: ! |
| 17:45.20 | ManxPower | Random Person: What's the best softphone to use with Asterisk? Me: All softphones suck. Buy a hardphone, you cheapass. |
| 17:45.37 | Sedorox | lol |
| 17:45.50 | syzygyBSD | lol |
| 17:45.55 | obsidian-studios | ManxPower: really not sure what the obsession is with softphones? do they need viagra or something |
| 17:46.18 | ManxPower | obsidian-studios: they don't cost money. |
| 17:46.28 | syzygyBSD | ok, here is a shitty hardware question, I have a SPA-841 that keeps reseting randomly, has anyone had issues with this? |
| 17:46.48 | ManxPower | syzygyBSD: no. Are you running the latest firmware for the phone? |
| 17:46.48 | obsidian-studios | ManxPower: true |
| 17:46.55 | Sedorox | I think its also kinda like where its one less thing on the desk |
| 17:47.10 | obsidian-studios | well I am pretty happy about PoE :) one less cord |
| 17:47.17 | ManxPower | Oh no! My PC crashed again and I can't call 911!!! |
| 17:47.29 | syzygyBSD | ManxPower: it just started happening 2 weeks ago, I think some settings changed on the router, I did a factory reset, I will see if i have any more issues |
| 17:47.36 | syzygyBSD | I will update to the latest firware |
| 17:47.42 | ManxPower | User dies, family sues the softphone company |
| 17:47.53 | Sedorox | lol |
| 17:47.59 | ManxPower | syzygyBSD: I have an 841 and do not have that problem |
| 17:48.08 | Sedorox | unfortinatly I could see that happening |
| 17:48.19 | *** join/#asterisk [Boriz] (n=Boris@64-191-230-114.eqx.chi.sparkplugbb.com) |
| 17:48.29 | obsidian-studios | ManxPower: which is some poor skilled coder in a third world country that does not have much, but owns an uses a hardphone :) |
| 17:48.30 | syzygyBSD | ya, I thought it was strange that it just started happening |
| 17:48.37 | ManxPower | Family wins, court orders the softphone company to include "no 911" stickers in their product. |
| 17:48.41 | X-Files | ManxPower: Prompt please why when to me speak there are losses voices (sounds) use codec ulaw |
| 17:48.42 | syzygyBSD | probably has something to do with the nat settings here |
| 17:48.54 | *** part/#asterisk [Boriz] (n=Boris@64-191-230-114.eqx.chi.sparkplugbb.com) |
| 17:49.00 | syzygyBSD | oh! I know what it is, makes sense |
| 17:49.42 | obsidian-studios | people just want it all, it's not enough to have * for free, and save thousands of $'s there, they want it all free, easily configurable, and work just as a hardphon |
| 17:49.47 | ManxPower | syzygyBSD: well don't keep us waiting |
| 17:49.53 | syzygyBSD | behind a nat with a cheap router it only allows one outgoing connection per local port, so if any other computer/device uses that port it will overwrite the mapping and lose the connection! |
| 17:50.09 | hhoffman | hmm, my X100P card is answering the phone even after I pick it up :-( |
| 17:50.21 | hhoffman | can you use n, in the s, i , and t values? like exten => t,1,... exten =>t,n,... ? |
| 17:50.22 | obsidian-studios | hhoffman: right on |
| 17:50.27 | X-Files | Pipls, Prompt please why when to me speak there are losses voices (sounds) use codec ulaw ? |
| 17:50.27 | hhoffman | even after reading the book asterisk isn't exactly straightforward ;-) |
| 17:50.31 | ManxPower | syzygyBSD: that's why the SIPuras default to using a different SOURCE port for each line. |
| 17:50.35 | hhoffman | softphones are nice if you don't (can't) spend a shitload of money on hardphones :-( Especially if just starting out |
| 17:50.52 | mutilator | er |
| 17:50.55 | mutilator | shitload? |
| 17:50.57 | mutilator | they like $50 |
| 17:50.59 | ManxPower | hhoffman: If you can't spend the money you shuold not be using Asterisk |
| 17:51.07 | obsidian-studios | or VOIP |
| 17:51.11 | obsidian-studios | VOIP is not cheap |
| 17:51.15 | obsidian-studios | even with * |
| 17:51.22 | hhoffman | ManxPower: geez, that's a pretty harsh view... |
| 17:51.34 | hhoffman | I'm using asterisk b/c it sounded cool and I wanted to learn it |
| 17:51.36 | ManxPower | hhoffman: not really. You WILL have to spend money on telecom |
| 17:51.37 | [TK]D-Fender | I dunno... Voip can be pretty cheap..... |
| 17:51.39 | X-AFK | ManxPower: Prompt please why when to me speak there are losses voices (sounds) use codec ulaw |
| 17:52.00 | Sedorox | voip is cheap... depending what you wanna do with it |
| 17:52.06 | ManxPower | X-AFK: I don't know, If I knew I would have answered you. |
| 17:52.07 | syzygyBSD | ManxPower: not if your company is a CLEC, then they have to spend money |
| 17:52.14 | obsidian-studios | hhoffman: yes, and that brings you across those that make a living in the telco world, harsh interactions at times |
| 17:52.17 | Sedorox | if you wanna add channels banks... multi-line phones.. then yes.. it gets up there |
| 17:52.18 | hhoffman | ManxPower: right... but if I use it at home and then am called to use it at some job... I've at least got an inkling of experience with it |
| 17:52.22 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 17:52.35 | yxa | iax to zap is very soft. should i adjust the rxgain/txgain? |
| 17:52.43 | ManxPower | yxa: yes |
| 17:52.44 | obsidian-studios | I have one of the cheapest * deployments and still cost like $500 or so |
| 17:52.51 | X-AFK | ManxPower: tnky |
| 17:52.54 | obsidian-studios | clone card, cheap phone, used computer etc |
| 17:53.02 | [TK]D-Fender | 1 port ATA for you home > plug onto line ; disconnect @ demarc, install * ; start with VoIP DID provider ; ATA pays for itself in 3 months. |
| 17:53.13 | [TK]D-Fender | TOPS |
| 17:53.21 | file | but when you touch my body, and close your eyes |
| 17:53.21 | yxa | ManxPower any rule of thumb? i read the wiki but i'm not so sure abt telco test tones |
| 17:53.23 | file | my little baby I don't realize |
| 17:53.28 | file | if you can say goodbye tell me why |
| 17:53.28 | ManxPower | I think my last Asterisk install was something lke $2,000 in hardware and that is excluding my fees |
| 17:53.49 | hhoffman | wow! |
| 17:53.56 | hhoffman | see, I could never spend that just to d/l and learn how it works |
| 17:54.02 | Sedorox | hehe |
| 17:54.02 | [TK]D-Fender | ManxPower : What kind of setup? Sure if you buy all new sip pphones the cost goes up... |
| 17:54.06 | yxa | i agree. voip is not cheap but the features and flexibility that comes with it is limitless |
| 17:54.06 | ManxPower | The one before that was closer to $7,000 |
| 17:54.12 | obsidian-studios | yes, when I presented to the LUG my rule of thumb was * hardware alone will run close to $1k |
| 17:54.22 | Sedorox | let me guess.. bunch of T1 cards and stuff too? |
| 17:54.26 | obsidian-studios | if you are buying decent stuff, not top of the line, but not clones or generic stuff |
| 17:54.28 | [TK]D-Fender | Wow... I am a budget king here! |
| 17:54.41 | syzygyBSD | ManxPower: we just bought a $5400 asterisk server |
| 17:54.48 | hhoffman | that being said i did just spend a shit load of money on a brand new switch... but that's also how I pay the bills |
| 17:54.53 | syzygyBSD | some clients want more then they need |
| 17:55.32 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 17:55.33 | syzygyBSD | that was without any cards |
| 17:56.02 | [TK]D-Fender | Yeah I for instance for my company chose Polycom IP 600's. I mean I could have gotten them el-cheap-o Grandstream's, or SPA-841's or something, but I wanted PoE, multiple lines, the micro-browser, and something SOLID. |
| 17:56.11 | ManxPower | [TK]D-Fender: Dual Xeon Server, w/1 CPU, 5 polycom phones, TSU 120 |
| 17:56.30 | [TK]D-Fender | TSU? Channel banked? |
| 17:57.01 | obsidian-studios | [TK]D-Fender: or you could have gone Cisco for twice the price of Polycoms |
| 17:57.04 | file | POTATOES |
| 17:57.14 | ManxPower | [TK]D-Fender: no a device to split the PRI and the DATA into our Asterisk server and our Cisco router. The stuff all comes in on one T-1 |
| 17:57.15 | obsidian-studios | only with gravy |
| 17:57.23 | Sedorox | I think they are about the same |
| 17:57.34 | obsidian-studios | or butter, sour cream, and chives |
| 17:57.41 | [TK]D-Fender | obsidian-studios : I DID look at them, and thats exactly why I chose Polycom. The products have parity on features, but Polycom's pricing kills Cisco. |
| 17:57.41 | yxa | grandstream blow |
| 17:58.05 | Sedorox | aparently the GXP-2000 is nice |
| 17:58.08 | obsidian-studios | [TK]D-Fender: yep, and good quality, Poly is on my recommend list now |
| 17:58.13 | Sedorox | but the budgetones are cheap..... in all aspects.. :p |
| 17:58.14 | shido6 | hehe |
| 17:58.16 | [TK]D-Fender | ManxPower : Ok, well that is a very special scenario. When you need that kind of stuff it costs what it costs... |
| 17:58.42 | syzygyBSD | ya, my sipura needed a firmware upgrade, was .91 now is 3.1.4 |
| 17:58.43 | obsidian-studios | ManxPower: * can terminate a PRI right and split the voice and data itself? |
| 17:58.44 | yxa | Sedorox don't get it. the gxp-2000. |
| 17:58.50 | Sedorox | hmmm |
| 17:58.50 | [TK]D-Fender | I'm going to grab an SPA-941 & SPA-3000 for home now, and find a local vendor for a Sangoma S518 ADSL card. |
| 17:59.09 | Sedorox | I heard they were good |
| 17:59.20 | [TK]D-Fender | Grandstream = Cheap ship I wouldn't touch with a 10' pole... |
| 17:59.29 | Sedorox | hehe |
| 17:59.34 | obsidian-studios | was talking to a client about that a while ago, was about to order the PRI, then BellSouth started offering business lines with unlimited ld for $24 a month, which is really over $50 when all is said and done |
| 17:59.37 | yxa | Sedorox seriously, if you wanna get grandstream, spend like 20 bucks more and get a Polycom 301 |
| 17:59.41 | Sedorox | the gxp I heard was good.. the BT100 I have.. yea.. sucks |
| 17:59.51 | [TK]D-Fender | There's a reason they're called "Barbie-tones" |
| 17:59.59 | Sedorox | yxa: I got a cheap bt100 right now.. I'm trying to save up for either a poly or cisco |
| 18:00.06 | [TK]D-Fender | the GXP is just BETTER crap, but crap just the same |
| 18:00.14 | obsidian-studios | so grandstreams are on the shit list now huh? last I heard they were ok, nothing great, but nothing bad either? |
| 18:00.22 | ManxPower | If you want to go grandstream just cancel the project -- the customer is too cheap. |
| 18:00.32 | obsidian-studios | prefer not to pay for crap, I can make it for free at least once a day ;) |
| 18:00.34 | ManxPower | obsidian-studios: um, they are always on the shitlist |
| 18:00.36 | _Sam-- | if i add a new user to IAX.conf, how do you reload that? |
| 18:00.50 | obsidian-studios | so glad I never bought one |
| 18:00.56 | InfraRed | _Sam--: reload in CLI ? |
| 18:00.57 | InfraRed | :) |
| 18:00.59 | [TK]D-Fender | _Sam-- : RELOAD in CLI |
| 18:01.03 | InfraRed | snap |
| 18:01.05 | [TK]D-Fender | ;) |
| 18:01.05 | InfraRed | i win |
| 18:01.07 | InfraRed | \o/ |
| 18:01.20 | _Sam-- | hah, i know im a tard, but i know "extensions reload" "sip reload".... |
| 18:01.23 | _Sam-- | but no iax reload? |
| 18:01.29 | InfraRed | 'reload' |
| 18:01.38 | _Sam-- | thanks you |
| 18:01.40 | _Sam-- | -s |
| 18:01.43 | oej | reload chan_iax2.so |
| 18:01.43 | InfraRed | np |
| 18:01.53 | file | reload oej.so |
| 18:02.01 | syzygyBSD | lol |
| 18:02.03 | oej | reload res_file.so |
| 18:02.08 | InfraRed | shutdown |
| 18:02.11 | InfraRed | halt |
| 18:02.17 | MikeJ[Laptop] | unload res_file.so |
| 18:02.18 | oej | No, "stop now" :-) |
| 18:02.21 | syzygyBSD | this isn't a console... |
| 18:02.26 | cpatry | file is so slow to start, leave it started please! :P |
| 18:02.32 | file | LOL |
| 18:02.34 | MikeJ[Laptop] | syzygyBSD, shush you |
| 18:02.42 | ManxPower | _Sam--: How about reload chan_iax2.so |
| 18:02.43 | syzygyBSD | where did you come from? |
| 18:02.53 | [TK]D-Fender | file is NOT slow! "Challenged" man... get withthe "now"! ;) |
| 18:02.58 | cpatry | did? from my mother! |
| 18:02.59 | queuetue | How do I mage a digital receptionist go into voicemail automatically if someone holds long enough? |
| 18:03.26 | ManxPower | queuetue: You start out by reading the following URLs. |
| 18:03.27 | ManxPower | ~docs |
| 18:03.28 | jbot | [docs] probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 18:03.29 | ManxPower | ~mailinglist |
| 18:03.31 | jbot | it has been said that mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php |
| 18:03.31 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 18:04.14 | [TK]D-Fender | queuetue : hint : look for the "t" extension in dialplan logic (extensions.conf) |
| 18:04.22 | queuetue | ManxPower: Thanks - I wil read them, but I meant to send that message to the amportal channel. (I was hoping AMP and AAH have a canned mechanism.) |
| 18:04.37 | queuetue | [TK]D-Fender: Thanks. |
| 18:05.05 | [TK]D-Fender | AMP? nevermind what I said.... its too late for you now ;( |
| 18:05.23 | ManxPower | Yes, queuetue has joined the dark side |
| 18:05.54 | queuetue | [TK]D-Fender: It's a little weird they way everyone is against using a tool to manage asterisk... Is it "geek bravado" that makes you want it to be hard on everybody? |
| 18:05.55 | [TK]D-Fender | If you're at all competent avoid * GUI's at all costs... |
| 18:06.32 | syzygyBSD | what about a custom gui you programed for internal use? |
| 18:06.38 | ManxPower | queuetue: I'm not against a tool to manage Asterisk. The problem is that ALL AVAILABLE TOOLS SUCK. |
| 18:06.38 | [TK]D-Fender | queuetue : * is very powerful and believe me, actually pretty simple. GUI's take away power from doing even some simple stuff just to sugar coat it with ITS call-fllow concepts... |
| 18:06.39 | obsidian-studios | even worse ;) |
| 18:07.47 | [TK]D-Fender | ManxPower : I'm on ScopServ's GUI here at work, and its what I needed to get * in the door, but naturally its not doing much that I couldn't do myself with time (I'm not an "experienced" linux user and would find integrating SQL and a few other things tricky) |
| 18:07.47 | obsidian-studios | if * ever has a gui, it needs to be embedded and not rely on third party stuff, or be a service like Samba's swat |
| 18:08.19 | queuetue | obsidian-studios: I think SWAT was probably the worst idea the samba team ever had. |
| 18:09.07 | [TK]D-Fender | webmin > swat, and I don't even want to hear about an * module for it.... *shudder* |
| 18:09.15 | ManxPower | [TK]D-Fender: I need to write my own GUI. I need a simple web page where the onsite person enters the MAC address of the phone, the model of the phone, the person's CallerID for each line, and it creates the polycom boot files, sip.conf entries, assigns an available extension from extensions.conf and sends a reboot packet to the phone |
| 18:09.49 | *** join/#asterisk supaigtr (n=yurplsl@152.53.16.10) |
| 18:09.50 | [TK]D-Fender | ManxPower : doesn't sound that hard... |
| 18:10.09 | ManxPower | [TK]D-Fender: it is when everything is a text file 8-) |
| 18:10.20 | queuetue | I guess it's hard to conceive that someone would prefer to use thier phone system than spend time hacking it. :) |
| 18:10.32 | file | I hate phones. |
| 18:10.38 | ManxPower | I'm waiting until we get our LDAP server is up and running, if the weasel that's installing it ever does so, then I'll do the rest in Realtime |
| 18:10.45 | [TK]D-Fender | Database the web setup (SQLite), and rebuild .conf files that get INCLUDED into your * config. |
| 18:10.59 | tzanger | nice, realtime ldap |
| 18:11.35 | [TK]D-Fender | queuetue : if you want a "toaster" then thats what you should buy. just keep in mind that pay-off for having control. |
| 18:11.36 | *** join/#asterisk tamp4x (n=kkdkdkkk@204.124.238.248) |
| 18:11.39 | obsidian-studios | queuetue: yes, I am not a swat fan, it's done more bad than good for me |
| 18:11.48 | ManxPower | tzanger: I'm not planning on doing realtime ldap, but I do want to put some stuff in LDAP |
| 18:11.52 | perd | fender cant you just store all that in a postgres database and dynamically update it all fender? |
| 18:11.53 | file | if you want a toaster that toasts, it costs extra |
| 18:11.59 | perd | or at least you can for extensions |
| 18:12.05 | obsidian-studios | however for those that are aware of it, ASSP has a great GUI, web based, and you can do all the same stuff with a text editor on the config file |
| 18:12.45 | *** part/#asterisk darius_ (i=darius@integrity.bourg.net) |
| 18:13.47 | *** join/#asterisk mut (n=animenod@65.111.201.79) |
| 18:13.51 | obsidian-studios | ASSP uses the philosophy of putting the web server in the app, not the app in a web server |
| 18:13.54 | mut | suck to #asterisk-unregistered |
| 18:14.05 | obsidian-studios | most likely would cause to much bloat to * |
| 18:14.54 | queuetue | [TK]D-Fender: For some reason, you have "interested in ease" confused with "scared of complexity." I'm pretty technical - I'm a Linux Kernel contributor, and write fairly hairy code for a living. But, when I can pop in a cd and have a working asterisk system, I'm going to try and work with that, instead of starting from scratch. Please - stop abusing people because they put different priorities on things you are emotionally inv |
| 18:16.33 | *** join/#asterisk Sobakai (n=jmwoodga@45.e6.d12c.cidr.airmail.net) |
| 18:16.58 | mut | oh oh what'de i miss |
| 18:17.39 | queuetue | mut: Nothing, just got sick of being treated like a second class citizen because I chose to install using AAH. |
| 18:17.59 | mut | oh you n00bz0r |
| 18:18.00 | mut | ;P |
| 18:19.48 | [TK]D-Fender | queuetue : Sorry if I sounded a bit harsh there. Just that I've never seen anyone get into * and expect things to be "easy". |
| 18:20.15 | *** join/#asterisk Mike (n=mike@201.135.48.190) |
| 18:20.36 | mut | er i see that everyday |
| 18:20.45 | Mike | guys if i need more than 30 lics of g729 should i buy intel or digium? |
| 18:20.57 | InfraRed | digium |
| 18:20.59 | Mike | i got the evaluation but its only a 30 day evaluation |
| 18:21.05 | obsidian-studios | * is a world unto its own |
| 18:21.08 | InfraRed | if you plan to use asterisk |
| 18:21.32 | Mike | InfraRed, yes, but intel sells it all for 199? |
| 18:21.42 | Mike | InfraRed, digium sells it for 10 each? |
| 18:21.50 | InfraRed | Mike: does it work with *? :) |
| 18:22.03 | Mike | intel does |
| 18:22.08 | InfraRed | buy intel then |
| 18:22.09 | Mike | digium also works tho |
| 18:22.18 | InfraRed | and if it fails |
| 18:22.19 | shido6 | building from scratch and witnessing an AMP dialplan are so different its not even funny, the learning curve is steep. you dont want *that* kind of help tho |
| 18:22.20 | InfraRed | ask intel for support |
| 18:23.16 | Mike | InfraRed, its for a lanparty who cares? |
| 18:23.17 | Mike | :) |
| 18:23.40 | InfraRed | not me \o/ |
| 18:23.50 | X-Files | ManxPower: I apologize that has disturbed, but I here have checked up: I Lift a tube and I dial the number 301 and I get on Asterisk and there to me speak constantly and here sounds constantly vanish. I have still checked up without Asterisk, have called on local number 204 and here all ideally sounds also is lost nothing, can eat any washed? Please answer. |
| 18:24.54 | file | my parser failed on that sentence |
| 18:25.03 | mut | as did mine |
| 18:25.14 | obsidian-studios | I got a null pointer error |
| 18:25.27 | file | obsidian-studios: atleast you didn't try to use and abuse the pointer |
| 18:25.49 | obsidian-studios | nope, just my own :) |
| 18:26.00 | LostFrog | "can eay any wash?" |
| 18:26.04 | LostFrog | "can eat any wash?" |
| 18:26.20 | obsidian-studios | I myself have never tried to eat that ;) |
| 18:26.31 | obsidian-studios | where's Mikey |
| 18:31.37 | *** topic/#asterisk by drumkilla -> Asterisk 1.2 will be released today!!! || http://www.asterisk.org |
| 18:31.48 | *** topic/#asterisk by drumkilla -> Asterisk 1.2.0 will be released today!!! || http://www.asterisk.org |
| 18:31.56 | obsidian-studios | whoot |
| 18:33.17 | [hC] | so... was 1.2.0 a branch off cvs from a while ago, or has it been kept up to date with cvs from a few days ago? just trying to figure out if id rather run 1.2.0 stable or keep my cvs head from a few days ago |
| 18:34.22 | *** join/#asterisk distortion (i=distorti@junipero.3sheep.com) |
| 18:34.22 | drumkilla | there has been no branch yet |
| 18:34.31 | drumkilla | and probably won't be for a couple more weeks |
| 18:34.37 | drumkilla | to let things settle down a bit |
| 18:34.39 | X-Files | drumkilla I apologize that has disturbed, but I here have checked up: I Lift a tube and I dial the number 301 and I get on Asterisk and there to me speak constantly and here sounds constantly vanish. I have still checked up without Asterisk, have called on local number 204 and here all ideally sounds also is lost nothing, can eat any washed? Please answer. |
| 18:35.01 | obsidian-studios | X-Files: when I lift a tube, I light it and smoke it ;) |
| 18:35.32 | X-Files | ;/ |
| 18:35.41 | Nugget | I quit smoking 8 years, 6 months, 1 week, 4 days, 13 hours, 35 minutes, and 41 seconds ago. During that time, I would have smoked 68,509 cigarettes. (That's like smoking a 3.24 mile-long cigarette) By quitting, I've saved $11,989.07! I've avoided inhaling 1.78 kg of tar, 109 grams of nicotine, and 1.10 kg of carbon monoxide. |
| 18:36.16 | LostFrog | X-Files: I don't think that is english. |
| 18:36.24 | obsidian-studios | X-Files: I assume you are having language translations issues? might want to find someone to translate for you, or speaks your native tongue. Your posts do not make sense in english? |
| 18:36.41 | LostFrog | He is probably using babelfish. :) |
| 18:36.53 | X-Files | ;) |
| 18:37.03 | obsidian-studios | Nugget: you could have avoided all that by just smoking WEED :) |
| 18:37.06 | LostFrog | Babelfish-IRC. |
| 18:37.19 | LostFrog | Neat idea. |
| 18:37.25 | *** join/#asterisk gorauskas (n=gorauska@66-224-20-131.atgi.net) |
| 18:38.46 | ikarus | hmmm, translation, reminds me, anyone know of a site with a few other languages (either brittish english or dutch) for atleast the voicemail app ? |
| 18:39.08 | Nugget | I quit smoking pot 11 years, 6 months, 1 week, 5 days, 13 hours, 39 minutes, and 7 seconds ago. During that time, I would have toked 2,105 joints. By quitting, I've saved $4,210.00! (partially by not accidently overtipping pizza delivery guys) I've avoided eating 16,840 starburst fruit chews, I've managed to remember 842 phone numbers, and I've missed watching The Wall 421 times. |
| 18:39.55 | zoa | ikarus |
| 18:39.56 | zoa | just a sec |
| 18:40.01 | *** join/#asterisk bweschke (n=bweschke@204.96.162.40) |
| 18:40.09 | obsidian-studios | Nugget: lol |
| 18:40.20 | zoa | kijk hier eens: http://www.asteriskguru.com/board/viewtopic.php?t=81 |
| 18:40.53 | ikarus | zoa: great |
| 18:41.16 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 18:41.36 | ikarus | Not entirely sure if we'll actually use it, because my co-worker seems to prefer to keep the answering machine traditional (which means it would hang of an ATA) |
| 18:41.48 | obsidian-studios | Nugget: what about ether |
| 18:42.21 | LostFrog | Damn.. the token fell out of our network again. :( |
| 18:42.28 | X-Files | I call from analog phone to asterisk, To me there speak any nonsense, but a problem in the lost sound for second and back a sound . I use codec u-law. How me understand ? |
| 18:42.33 | ikarus | (for enhanced luser friendlyness, annoying) |
| 18:43.00 | LostFrog | X-Files: that is closer. |
| 18:43.23 | X-Files | ^) |
| 18:43.53 | UlbabraB | sera |
| 18:44.34 | obsidian-studios | I think what X-Files: is trying to say is they are experiencing a choppy call |
| 18:44.35 | ikarus | I myself am leaning more towards simple voicemail via asterisk (dial an extension that automaticly plays back the messages, no prompting just an option to replay and to save and at the end to delete all unsaved), but I am not the only one to decide on that |
| 18:44.41 | Nugget | ik ben een vliegende koe. |
| 18:44.44 | zoa | haha |
| 18:44.53 | X-Files | and I try connect to local phone (not connecting to asterisk) , working perfect .. How me understand ? |
| 18:44.54 | zoa | ik geloof er niets van |
| 18:45.01 | Nugget | boe boe |
| 18:45.06 | zoa | :) |
| 18:45.31 | X-Files | obsidian-studios: ok wait, i try check.. |
| 18:45.43 | ikarus | zoa: Nugget is dat echt hoor |
| 18:46.31 | obsidian-studios | X-Files: huh, you have an analog phone that sounds choppy when talking to *, but if you call another analog phone it sounds fine? |
| 18:47.12 | X-Files | obsidian-studios: yes |
| 18:47.34 | obsidian-studios | how do the analog phones connect to *? how do they connect to each other? |
| 18:47.37 | *** join/#asterisk Igbothom (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) |
| 18:48.26 | X-Files | obsidian-studios: i call to 301 connect to asterisk |
| 18:48.40 | X-Files | obsidian-studios: call to 201 this local phone |
| 18:48.59 | X-Files | obsidian-studios: i call from 202 |
| 18:49.28 | obsidian-studios | how do the phones connect to each other? if you are dialling extensions and it rings another phone? Are those calls going through * |
| 18:50.08 | obsidian-studios | what is 301? You can't call *, it's not a person or device, you can place calls through * |
| 18:50.48 | Katty | obsidian-studios: i call * all the time |
| 18:50.52 | X-Files | obsidian-studios: i call to 201 and its not going trouth asterisk :( |
| 18:51.03 | Katty | obsidian-studios: in fact, like......for every call |
| 18:51.16 | Katty | X-Files: direct ip connection? |
| 18:51.17 | obsidian-studios | Katty: calls are placed through *, but not to it? |
| 18:51.22 | X-Files | obsidian-studios: after asterisk sound not fine ... |
| 18:51.40 | obsidian-studios | Katty: a call to * would only map you to an extension no? to do something else? or directly |
| 18:51.47 | Katty | obsidian-studios: a number, in extensions.conf can do anything |
| 18:51.51 | obsidian-studios | context, I meant not extension |
| 18:52.03 | Katty | obsidian-studios: i can call 200, and it will run rsync if i want it to |
| 18:52.06 | obsidian-studios | ok, just trying to figure out what should happen when they dial 301? |
| 18:52.16 | obsidian-studios | Katty: nice |
| 18:52.30 | Katty | obsidian-studios: i can setup asterisk backups on extension 5 using a shell script |
| 18:52.35 | Katty | obsidian-studios: asterisk does anything. |
| 18:52.45 | obsidian-studios | Katty: thanks you just took this to a whole other level for me |
| 18:52.47 | obsidian-studios | damit |
| 18:52.53 | obsidian-studios | damn * |
| 18:53.05 | *** join/#asterisk twisted[mobile] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 18:53.05 | *** mode/#asterisk [+o twisted[mobile]] by ChanServ |
| 18:53.10 | Katty | personally, i spam 3 of our computers using smbclient with date and callerid info everytime a call comes in |
| 18:53.12 | obsidian-studios | it's very hard to learn about something that has no limits :) |
| 18:53.20 | Katty | yay, twisted! |
| 18:53.22 | X-Files | Katty: yep ;) |
| 18:53.30 | Katty | obsidian-studios: you're telling me |
| 18:53.34 | obsidian-studios | Katty: yeah, I get CID stuff emailed all the time |
| 18:53.36 | Katty | obsidian-studios: i had to learn linux and asterisk all at once |
| 18:53.47 | obsidian-studios | Katty: OMG, glad I am not ue |
| 18:54.01 | Katty | twisted[mobile]: you really need to get some sleep |
| 18:54.05 | twisted[mobile] | i got sleep |
| 18:54.08 | Katty | twisted[mobile]: stop poking about the casino and /sleep/ more |
| 18:54.08 | obsidian-studios | Katty: I had many years of Linux experience, but no telco, so I got slapped around a bit by the * community, most of it here |
| 18:54.18 | twisted[mobile] | but i was up long last night partying on the roof of the rio |
| 18:54.23 | file | meep |
| 18:54.26 | Katty | obsidian-studios: maybe i have an advantage over you |
| 18:54.37 | obsidian-studios | Katty: I am sure lots :) |
| 18:54.38 | Katty | obsidian-studios: very few guys in here will tell a female to rtfm ;) |
| 18:54.46 | X-Files | obsidian-studios: you can't me help ? |
| 18:54.55 | obsidian-studios | Katty: yeah I just got sktn |
| 18:55.09 | obsidian-studios | sktn -> swift kick to the nutsack |
| 18:55.12 | Katty | luckily, there are nice people here |
| 18:55.19 | Katty | like twisted and hmmhesays and spackle |
| 18:55.30 | Katty | and little by little, i learned linux and asterisk |
| 18:55.35 | Katty | or.....still am |
| 18:55.39 | twisted[mobile] | teehee |
| 18:55.39 | obsidian-studios | X-Files: trying man, but I have no clue what you are trying to do |
| 18:55.57 | Katty | and file! |
| 18:56.00 | obsidian-studios | X-Files: what is supposed to happen when you call 301? is * playing a recording? |
| 18:56.02 | Katty | who provides magic wands, once in awhile |
| 18:56.09 | twisted[mobile] | file, huzzah |
| 18:56.09 | file | and magic muffins |
| 18:56.12 | Katty | or at least sarcasm |
| 18:56.30 | twisted[mobile] | whoa |
| 18:56.33 | twisted[mobile] | he brokw out the sw |
| 18:56.34 | ManxPower | ..er.. stroopfaffel |
| 18:56.38 | obsidian-studios | yes, once I did my homework, and showed I could take a beating I was some what welcomed it ;) |
| 18:56.56 | Katty | ManxPower: i'll pass. |
| 18:56.58 | obsidian-studios | ManxPower: 3rd times a charm |
| 18:57.00 | Katty | ManxPower: i'm just not feeling like cookies |
| 18:57.02 | X-Files | obsidian-studios: i call to 301 this is asterisk, asterisk answer and play BackGround(vm-options) |
| 18:57.28 | Katty | twisted[mobile]: ooh, i discovered something! |
| 18:57.33 | Katty | twisted[mobile]: two things, actually |
| 18:57.35 | obsidian-studios | X-Files: ok, and that sounds choppy, and you are using ulaw, progress. Possibly network congestion or bogged down machine? |
| 18:57.46 | Katty | twisted[mobile]: abcde and gkrellm |
| 18:57.53 | Katty | twisted[mobile]: they're both hotttt. |
| 18:58.05 | twisted[mobile] | heh |
| 18:58.48 | X-Files | obsidian-studios: and in this time there are some sound missing.. yes i use ulaw , this server link 100mbit my home network |
| 18:59.14 | obsidian-studios | X-Files: if it's an analog phone could be the fxs devices problem |
| 19:02.16 | obsidian-studios | ManxPower: nice, do that to the bosses |
| 19:02.22 | X-Files | from FXS port if without asteriks .. to connect without him call to another analog telephone trough gateway, then the qualitty is perfect |
| 19:02.33 | obsidian-studios | ManxPower: I left BadMoFo on clients 7960 :) They like it |
| 19:02.36 | *** join/#asterisk Renacor (n=kvirc@ip21.farheap.net) |
| 19:02.55 | Renacor | anybody have problems with manager apps when asterisk is under load? |
| 19:03.16 | cpatry | Renacor: more details? |
| 19:03.50 | obsidian-studios | X-Files: parsing and failing, totally confused man |
| 19:04.30 | obsidian-studios | X-Files: what are the other phones connected to? What gateway? They should also be connected to FXS devices if they are analog phones |
| 19:05.48 | X-Files | obsidian-studios yep.. they are there :( bough port 3 and 4 |
| 19:06.45 | obsidian-studios | X-Files: ok, so you have 3 analog phones connected to fxs devices. You can call from a phone to a phone and sound is good. You make a call from any phone to *, and it's choppy? |
| 19:07.47 | X-Files | yep! |
| 19:08.02 | X-Files | that`s it! :) |
| 19:08.26 | obsidian-studios | X-Files: possible a problem with playback of the recording, doubt it's a problem with the recording itself |
| 19:08.31 | obsidian-studios | or a bogged down machine |
| 19:09.05 | syle | lets say someone calls your asterisk box, and in that context you use dial command to bridge to somewhere else, does callerid number change? |
| 19:09.09 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 19:09.37 | file | syle: if you change it. |
| 19:09.47 | X-Files | registration is fine and it's shows that hey are on-line and registred |
| 19:10.32 | syle | #DEFINE changeit ... you mean cdr->src stays as original caller right? |
| 19:11.02 | X-Files | obsidian-studios registration is fine and it's shows that hey are on-line and registred |
| 19:11.52 | obsidian-studios | X-Files: sounds like playback issues |
| 19:14.20 | X-Files | obsidian-studios there are something like when you are talking some letters are going down and after again everything is ok for 5 seconds and again :/ |
| 19:15.24 | obsidian-studios | X-Files: problems with the fxs device, network issues, playback issues, or load on machine, it's all I can say man I am not * expert or guru |
| 19:15.31 | obsidian-studios | just an * wannabe and poser |
| 19:16.32 | X-Files | ;/ |
| 19:16.39 | X-Files | but who can help ? |
| 19:16.46 | X-Files | do you know ? |
| 19:17.42 | X-Files | becouse it`s very strange that sound are missing just fof second after 5 seconds .. |
| 19:17.48 | obsidian-studios | X-Files: well I have identified where the problems could lie, you will have to research and provide more info for others to help out |
| 19:18.11 | obsidian-studios | X-Files: also I know it's hard, but you got to formulate better and clearer sentences and phrases |
| 19:18.30 | obsidian-studios | or teach me what language u speak natively ;) |
| 19:18.50 | X-Files | :))) |
| 19:19.04 | X-Files | no now are writing my girlfriend :P |
| 19:19.16 | X-Files | or may be you would like to learn russian ?? :) |
| 19:19.44 | obsidian-studios | yes, women that serve, we all need a few |
| 19:19.51 | X-Files | :) |
| 19:19.55 | X-Files | just few? |
| 19:19.58 | obsidian-studios | yes Russian, I might order a Russian bride so that would help :) |
| 19:20.19 | X-Files | obsidian-studios o'k :) i have some friends :) which age do you need???? :))) |
| 19:20.40 | kink0 | hello, I have a little problem in g729 implementation |
| 19:21.06 | tzafrir_laptop | I try to use Originate from the manager interface. In the full log I get "Manager recieved command "Originate" but nothing more. Any idea what's goint on? |
| 19:21.09 | kink0 | ManxPower, still you there ? after debuging I got : Capabilities: us - 0x100 (g729), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x100 (g729) |
| 19:21.09 | kink0 | Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) |
| 19:21.18 | tzafrir_laptop | IS it incomplete? |
| 19:21.31 | X-Files | obsidian-studios order :) we`ll pakage and delliver her to you :) |
| 19:21.39 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
| 19:21.55 | obsidian-studios | X-Files: all ages, so long as they are legal in the US, hot, have no morals, willing to serve, and no will |
| 19:22.08 | X-Files | :) |
| 19:22.17 | *** join/#asterisk Daniel28 (n=daniel_t@141.85.0.66) |
| 19:22.20 | obsidian-studios | X-Files: I will get back to you with some more requirements and criteria |
| 19:22.35 | X-Files | obsidian-studios o`k .. but when? |
| 19:22.40 | obsidian-studios | X-Files: try some other sound files, if the problem continues, make sure the machine can play sounds normally |
| 19:22.50 | obsidian-studios | X-Files: when I have $, and want a wife |
| 19:22.58 | X-Files | o`k :)))))))) |
| 19:22.59 | obsidian-studios | X-Files: for now I just want whores |
| 19:23.41 | *** join/#asterisk Assid (n=assid@59.183.57.221) |
| 19:23.44 | paryl | will all of my config files transfer seamlessly to 1.2.0? |
| 19:24.17 | X-Files | obsidian-studios xmmm ... :) but don't take it too long :P |
| 19:24.21 | Assid | hrmm.. i got type=friend on both my * boxes.. but i still cant get it to authenticate to each other |
| 19:24.21 | kink0 | 1/1 encoders/decoders of 1 licensed channels are currently in use !! great !! I did, but I am not sure how to success |
| 19:24.30 | obsidian-studios | X-Files: why I am sure you all will make more :) |
| 19:24.59 | X-Files | :) |
| 19:27.11 | X-Files | obsidian-studios o`k how he is telling me .. if he calls frome one phone to another (port phones, analogs ) trouth asterisk thet the problem is the same as if calling to * |
| 19:27.16 | X-Files | :( |
| 19:28.13 | _Sam-- | anyone know how to make a grandstream phone not ring in your ear if you already on a call? |
| 19:28.14 | Assid | anyone have 2 boxes tlaking to one another? |
| 19:28.45 | paryl | Assid: of course |
| 19:28.50 | LostFrog | _Sam--: I just read that that was a lost cause. |
| 19:28.56 | Assid | paryl: want to help me with this? |
| 19:28.59 | Assid | i cant seem to do it |
| 19:29.09 | paryl | sure.. what's your problem? |
| 19:29.36 | Daniel28 | has anyone tested T38 using spandsp? |
| 19:29.40 | Assid | i have type=friend on both .. and host=dynamic on the main one which has a static ip.. and host=dynamic for the one that does |
| 19:29.43 | Assid | doenst |
| 19:29.49 | *** join/#asterisk justinu (n=j2@72.18.13.48) |
| 19:29.57 | _Sam-- | LostFrog: the only way ive been able to is to turn off the call-waiting on the phone, but then it is limited to 1 incoming....my employees are going nuts, just ported our DID to asterisk today |
| 19:30.14 | obsidian-studios | X-Files: ty, that makes sense. so all calls through * are choppy, if it's not network congestion, it's got to be the load on the machine? What processor is in the machine, and how much ram? |
| 19:30.15 | Assid | but it doresnt connect |
| 19:30.24 | _Sam-- | and if they are on a call on the grandstream, and another line appearance rings, it rings in their ear and cuts off the conversation |
| 19:31.22 | InfraRed | cool |
| 19:31.28 | InfraRed | use it as a prank |
| 19:31.29 | X-Files | obsidian-studios Server: Intel Server MB , 2 CPU Xeon 2.4Ghz and 1Gb Mem. lan 100Mbit in one net with me |
| 19:32.14 | obsidian-studios | X-Files: should be fine, what FXS device are you using? |
| 19:32.58 | obsidian-studios | anyone else got any thoughts now that the problem is a bit clearer? I am leaning toward problems with the FXS device, since all calls through * are choppy? |
| 19:33.56 | X-Files | Eusso UTG7104-22 the same as Yoda VG-400 and planet vip-000(400,800) |
| 19:34.03 | X-Files | obsidian-studios Eusso UTG7104-22 the same as Yoda VG-400 and planet vip-000(400,800) |
| 19:34.14 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 19:34.24 | obsidian-studios | not familiar with those fxs devices at all |
| 19:35.02 | obsidian-studios | X-Files: are you sure there is not transcoding taking place? |
| 19:36.46 | X-Files | transcoding ? can you translate? :( I don't know what thise word meens and translater too :((( |
| 19:37.00 | Daniel28 | can anyone help me with t38 over *,please? |
| 19:37.32 | *** join/#asterisk rajiv_ (n=irc@gentoo/developer/rajiv) |
| 19:38.19 | [TK]D-Fender | t38 is a lie made up to scare little children :) |
| 19:38.27 | Daniel28 | :) |
| 19:38.38 | [TK]D-Fender | (translation - not supported yet. Pass-through is in devel) |
| 19:38.51 | Daniel28 | ok...that makes it more clear |
| 19:38.55 | Connor | I'm having problems with a SNOM 360 behind nat.. I've specified the stun server.. but,, I'm only getting 1 way audio.. |
| 19:38.59 | [TK]D-Fender | ywc .... |
| 19:39.15 | Daniel28 | thanks |
| 19:39.17 | [TK]D-Fender | Connor : Which side(s) are NAT'd? |
| 19:39.28 | Connor | just the phone.. not asterisk. |
| 19:39.35 | [TK]D-Fender | keeping in mind * doesn't support STUN IIRC... |
| 19:39.49 | Daniel28 | but if using spansdp only pass-throudh is supported,right? |
| 19:40.04 | X-Files | obsidian-studios: you can check http://download.eusso.com:8080/Manual/VoIP/ITG_Command_Ref.zip |
| 19:40.11 | [TK]D-Fender | you should need anything really dor basic NAT. set your qualify for keep-alive, and let * know the ext is behind NAT. |
| 19:40.12 | obsidian-studios | X-Files: transcoding is using different codecs, like ulaw and alaw, right now you are transcoding this conversation :) |
| 19:40.27 | obsidian-studios | X-Files: in a bit, got to work on some other stuff atm |
| 19:40.36 | X-Files | :) |
| 19:40.41 | [TK]D-Fender | SpacnDSP is a live fax receipt, not T37/38 |
| 19:43.52 | *** part/#asterisk SplasPood (i=nobody@paravolve.net) |
| 19:44.24 | X-Files | obsidian-studios when you`ll come back tell me plz. |
| 19:45.48 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 19:48.52 | *** join/#asterisk emakris2 (n=emakris@c-24-128-56-2.hsd1.ma.comcast.net) |
| 19:48.57 | hhoffman | is a cisco ata186 worth $40 for use with * ? |
| 19:50.58 | *** join/#asterisk docelmo (n=docelmo@66.237.242.41.ptr.us.xo.net) |
| 19:51.12 | docelmo | sup sup |
| 19:51.17 | *** join/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com) |
| 19:52.58 | InfraRed | hhoffman: it works |
| 19:55.05 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 19:55.25 | hhoffman | InfraRed: the ata186? |
| 19:55.25 | hhoffman | or your config? |
| 19:55.51 | InfraRed | ata186 |
| 19:56.16 | hhoffman | cool, thanks :-) |
| 19:56.25 | InfraRed | check the wiki |
| 19:56.31 | InfraRed | it's full of info bout the 186 |
| 19:57.35 | Renacor | anybody have problems with manager apps when asterisk is under load? as in taking forever to respond back |
| 19:57.42 | Renacor | or not responding at all |
| 19:58.27 | *** join/#asterisk Nivex (i=kjotte@user-0c8hq5r.cable.mindspring.com) |
| 19:59.21 | Renacor | version : CVS-Nv1-2-0-beta1 |
| 20:01.45 | docelmo | Wow.. Dead room...... |
| 20:02.17 | docelmo | need I say more? |
| 20:02.25 | X-Files | obsidian-studios: u there ? |
| 20:03.09 | obsidian-studios | X-Files: yes, but I am starting to think I can't help much, I would like to , to an extent, but not a guru |
| 20:03.53 | X-Files | obsidian-studios: ok, but maybe in rxgain or txgain ? |
| 20:03.56 | *** join/#asterisk Uther_P (n=uther_p@66.180.120.82) |
| 20:05.01 | kink0 | I got executing MusicOnHold on the remote CLI, but sounds ( nor traffic ) arrives to the local CLI, any idea ? |
| 20:05.02 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 20:06.15 | obsidian-studios | X-Files: choppiness does not sound like a volume or gain issue |
| 20:07.03 | *** join/#asterisk DrDeke (i=dekemar@auriaria.engin.umich.edu) |
| 20:07.11 | DrDeke | 120 eh? |
| 20:07.14 | DrDeke | 1.2.0* |
| 20:09.26 | X-Files | obsidian-studios :( there not the volume there like tve all connect are missing for one sec after each 5 sec. |
| 20:10.29 | *** join/#asterisk bryanh2 (n=bryanh@mybox.ngworld.net) |
| 20:10.45 | obsidian-studios | <PROTECTED> |
| 20:10.58 | obsidian-studios | <PROTECTED> |
| 20:11.17 | obsidian-studios | did you compile * from source or install a binary, you might want to try compiling if you used a binary |
| 20:11.22 | bryanh2 | anyone have nat problems with 1.2.0 tree? seems to detect my correct src port but transmits back to default sip port, still going thru chan_sip.c |
| 20:12.13 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
| 20:12.36 | X-Files | obsidian-studios another words the problem can be in machine, network or programm which are installed on machine .. am I right? |
| 20:14.44 | X-Files | now there are the programm asterisk 1.0.9. but i can install from cvs ... |
| 20:14.46 | bryanh2 | strange.... debug Sending to 24.166.178.102 : 3094 (NAT) and then Transmitting (NAT) to 24.166.178.102:5060: dono why it uses 5060, it needs to use 3094 |
| 20:18.06 | queuetue | Can anyone point me towards a tutorial on asterisk scripting? Not a big book that tasks me through the history of telephony equipment and partitioning a T1, just simple scripting... :) |
| 20:18.29 | lunk | queuetue: www.voip-info.org has some good succinct docs |
| 20:18.42 | bryanh2 | my slave is falling behind my master but i am not sure if it's the master i/o problem or the sla |
| 20:18.47 | bryanh2 | sorry, wrong window |
| 20:19.04 | queuetue | lunk: Spread in a massive amount of other stuff... I was hoping for a link to the actual succinct stuff... |
| 20:19.33 | X-Files | obsidian-studios now there are the programm asterisk 1.0.9. but i can install from cvs ...can it help ?? |
| 20:19.40 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
| 20:19.49 | lunk | queuetue: try Asterisk Expressions |
| 20:20.32 | Katty | expressions? |
| 20:20.38 | Katty | what is this, a new seasonal line? |
| 20:20.54 | DrDeke | lol |
| 20:21.07 | Katty | let me guess, a new promotional fragrance |
| 20:21.30 | *** join/#asterisk echion (n=rickard@c-e7c6e255.010-56-73746f39.cust.bredbandsbolaget.se) |
| 20:21.30 | lunk | haw |
| 20:21.44 | echion | anyone using sipaddheader? |
| 20:21.49 | Katty | echion: hi |
| 20:21.59 | echion | Katty, hi |
| 20:22.03 | Katty | echion: how're you today? |
| 20:22.08 | Katty | echion: besides impatient |
| 20:22.12 | Renacor | does anybody here use flash operator panel? |
| 20:22.16 | Katty | Renacor: yes |
| 20:22.34 | Renacor | Katty: is it taking up 100% cpu usage for the server and client on your system?? |
| 20:22.40 | echion | Katty, well good question and you? |
| 20:22.50 | Katty | Renacor: nope, but it likes to cause deadlocks. |
| 20:23.00 | Katty | echion: i'm always patient, except when i'm not. |
| 20:23.10 | Renacor | Katty: what version? |
| 20:23.25 | echion | Katty, I've been patient today, just now I'm getting annoyed as I can't figure the problem |
| 20:23.29 | Katty | Renacor: saying, i would know. do not know, therefore, cannot say. |
| 20:23.43 | Katty | echion: nice of you to at least say hi when you walk in (= |
| 20:23.56 | Katty | Renacor: let me dig up the folder, hold on |
| 20:24.21 | Katty | Renacor: op_panel-0.24.tar.gz |
| 20:24.43 | Renacor | Katty: Thanks, I got op_server.pl running on an amd 2800+ and its using up 99% cpu, plus on some machines the client side doesn't even load all the info |
| 20:24.55 | Renacor | Katty: yeah Im using the same one |
| 20:24.57 | Katty | Renacor: that's not good. |
| 20:25.02 | Katty | Renacor: i took mine down. |
| 20:25.16 | Katty | Renacor: after 3 or 4 changes and stopping the op_server.pl file, the server would deadlock |
| 20:25.29 | Katty | Renacor: and insane, to the point of having to reboot |
| 20:25.34 | Katty | Renacor: i don't have time to mess with it right now |
| 20:25.37 | [TK]D-Fender | queuetue : what kind of scripting do you have in mind? Basic dial-plan stuff, or something you think would be more involvoing (database lookups, etc)? |
| 20:25.49 | Renacor | Katty: hmm don't have that problem, but the cpu usage is insane |
| 20:25.57 | Katty | Renacor: dunno |
| 20:27.04 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 20:27.20 | TheCops | Hi |
| 20:27.31 | lunk | TheCops: AHOY |
| 20:27.35 | TheCops | What's the maximum of PCI card (TDM2400) I can put in a computer? |
| 20:28.23 | InfraRed | two meeelion |
| 20:28.33 | tamp4x | looks like it can only fit in a 3u |
| 20:29.10 | DrDeke | 0 if you are as poor as i am ;) |
| 20:29.53 | [TK]D-Fender | TheCops : I don't think you want more than 1. If thats the case you may be better off with channel banks. |
| 20:29.53 | *** join/#asterisk liran_ (n=liran@80.178.123.120.adsl.012.net.il) |
| 20:29.54 | TheCops | [TK]D-Fender |
| 20:30.00 | queuetue | [TK]D-Fender: Eventually, some pretty intense stuff, but just basic dialplans right now... |
| 20:30.03 | TheCops | I need to plug 200 POTS lines |
| 20:30.05 | TheCops | into asterisk |
| 20:30.11 | *** join/#asterisk razu_ (n=razu@ip61.cab74.mus.starman.ee) |
| 20:30.14 | TheCops | and I can't transform POTS line into PRI |
| 20:30.19 | [TK]D-Fender | 200 POTS?! dear god why?! |
| 20:30.21 | TheCops | this is a business limitation |
| 20:30.35 | Katty | [TK]D-Fender: to make you /cringe/ |
| 20:30.55 | DrDeke | Well, you could buy several 24 port POTS-to-T1 banks, and some quad T1 cards |
| 20:30.58 | *** join/#asterisk brent21 (n=Brent21@70.88.149.221) |
| 20:31.01 | [TK]D-Fender | Ok, you NEED a channel bank solution then. Even then, thats a serious load you might even want to spread across 2+ servers |
| 20:31.04 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 20:31.15 | TheCops | DrDeke, TDM2400 card is to replace that |
| 20:31.18 | DrDeke | What would be the best way to interconnect the servers then? TDMoE? |
| 20:31.21 | TheCops | and is less expensive |
| 20:31.28 | docelmo | YIPPIE! |
| 20:31.39 | brent21 | I see there are a couple LCR modules and rate engines out there that still appear to be in beta mode, is there any sense of feeling of which one is most reliable? |
| 20:31.50 | X-Files | obsidian-studios are you here ? can you answer? |
| 20:31.53 | TheCops | [TK]D-Fender, that's why I'm asking what's the maximum allowed PCI card for TDM in a PC |
| 20:32.02 | TheCops | channel banks card from TDM will be out in about 2 weeks |
| 20:32.06 | docelmo | brent21 code your own.. I did.. |
| 20:32.10 | TheCops | 2 days |
| 20:32.10 | TheCops | sorry |
| 20:32.27 | Flauto | hi all |
| 20:32.27 | DrDeke | well, a TDM2400 can only handle 24 channels, so to plug in 200 lines you would need 9 of them |
| 20:32.28 | [TK]D-Fender | TheCops : Well Digium says if you have to pass 2 cards, think twice.... |
| 20:32.31 | brent21 | docelmo, ok, you do it in AGI? |
| 20:32.32 | Flauto | is yomama here? |
| 20:32.32 | DrDeke | I have never seen a machine with 9 PCI slots. |
| 20:32.48 | TheCops | [TK]D-Fender, why ?! |
| 20:33.00 | docelmo | Yep |
| 20:33.01 | [TK]D-Fender | Since 2 cards = 48 ports for POTS, that simply won't do for you. You'd need 2 x 4port T1 cards and channel banks to match |
| 20:33.04 | docelmo | PHP actually |
| 20:33.10 | brent21 | nice |
| 20:33.13 | DrDeke | TheCops: Digium manufactures the cards and knows a lot about Asterisk. They evidently recommend not exceeding two line cards per machine. |
| 20:33.27 | DrDeke | Although you might be able to get more to work, they probably recommend that for a reason. |
| 20:33.28 | TheCops | DrDeke, that's what I want to hear |
| 20:33.29 | TheCops | :) |
| 20:33.34 | docelmo | I have 10 carriers I use and load up the LCR and let er rip. Once I get all the dust settled I am moving it to a C object. |
| 20:33.34 | [TK]D-Fender | TheCops : Well Digium's TDM cards are IRQ / PCI picky and don't play nice, plus the CPU overhead they demand. |
| 20:33.43 | TheCops | DrDeke, Magma compagny is doing PCI slot expansion for 2000$ (13slot) |
| 20:34.15 | DrDeke | You can try that, but chances are that it won't work well if at all. |
| 20:34.26 | DrDeke | Since the company that makes the cards says it won't. |
| 20:34.39 | TheCops | [TK]D-Fender, it was more for a money problem, I calculated around 37k CND for physical infrastructure...and with the TDM2400, about 27k |
| 20:35.32 | [TK]D-Fender | TheCops : that only for the POTS part? |
| 20:35.34 | DrDeke | I mean, I think it would be interesting to see how many cards of various types you could get to work in a really beefy, modern, well-designed server. But if you are asking what the recommendation is... Well, there you have it. |
| 20:35.48 | Katty | how many of those TDM400 can a serverhandle? |
| 20:35.51 | DrDeke | (bbiaf) |
| 20:35.51 | TheCops | [TK]D-Fender, server, channels banks, T1 card, all stuff like that |
| 20:35.53 | Katty | as many as you can stick in the motherboard? |
| 20:36.03 | TheCops | Katty, did you know how to read a buffer? |
| 20:36.06 | TheCops | do you know sorry |
| 20:36.15 | Katty | TheCops: i've no clue what a buffer even is |
| 20:36.28 | TheCops | just scroll up, you will have your answer |
| 20:36.32 | InfraRed | if it's cheaper and wont work, it'll be more expensive |
| 20:36.32 | [TK]D-Fender | TheCops : well the TDM2400 setup won't cut it for you anyways. and this is for 200 LINES (not phone extensions) right? |
| 20:36.32 | InfraRed | :P |
| 20:36.34 | Katty | i see. |
| 20:36.42 | TheCops | 200 LINES |
| 20:36.43 | Katty | TheCops: i call that backlog |
| 20:37.00 | [TK]D-Fender | EEK.... ok, let me seew what that will run you... |
| 20:37.07 | TheCops | IRC client call it a buffer in programmation hehe |
| 20:37.11 | Katty | TheCops: I'm not talking about those |
| 20:37.16 | Katty | TheCops: i am talking about TDM400s |
| 20:37.23 | Katty | TheCops: ANALOG cards, not TDM2400s |
| 20:37.30 | *** join/#asterisk msw (n=msw@rdu-nat.rpath.com) |
| 20:37.36 | TheCops | This is the same rules |
| 20:37.39 | tamp4x | <Katty> TheCops: i've no clue what a buffer even is <- must be a woman |
| 20:37.46 | TheCops | lol |
| 20:37.52 | tamp4x | =] |
| 20:38.03 | TheCops | [TK]D-Fender, I asked twice to my client if there's a way to transform all shit (200 lines) to PRI |
| 20:38.10 | Katty | tamp4x: i wouldn't recommend insulting me ;) |
| 20:38.16 | TheCops | This is a special services that he can't do that |
| 20:38.20 | Katty | tamp4x: because, like a woman, i can be a bitch ;) |
| 20:38.28 | TheCops | lol |
| 20:38.30 | TheCops | hahahahaha |
| 20:38.37 | tamp4x | aka she has an irc bf with ops |
| 20:38.37 | Katty | TheCops: i gather it's not recommend for more than two |
| 20:38.47 | tamp4x | =D |
| 20:38.49 | TheCops | hahahaha |
| 20:38.52 | Katty | actually, i prefer females |
| 20:38.53 | Katty | kthxbi |
| 20:38.56 | TheCops | ho! |
| 20:39.19 | [TK]D-Fender | TheCops : 21000$CDN |
| 20:39.26 | TheCops | For ? |
| 20:39.35 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:39.38 | Katty | i see my question is not getting answered |
| 20:39.39 | TheCops | caliss |
| 20:39.40 | TheCops | Fender |
| 20:39.41 | Katty | this makes me unhappy |
| 20:39.44 | TheCops | jai comme pas allumer |
| 20:39.48 | [TK]D-Fender | full setup, 2 x 4-port T1 Cards, and 8 24 FXO channel banks. |
| 20:40.05 | [TK]D-Fender | m'excuse? |
| 20:40.08 | InfraRed | what is your question |
| 20:41.54 | *** join/#asterisk copantl (n=galel@205.240.205.192) |
| 20:42.18 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:48.11 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 20:48.41 | P4C0 | is it normal that my voip provider don't want to let me connect from asterisk to their sip (proxy/server/switch whatever), insted they force me to use their analog device? (forcing me to buy a fxo card...) |
| 20:48.46 | *** part/#asterisk brent21 (n=Brent21@70.88.149.221) |
| 20:48.49 | *** join/#asterisk copantl (n=galel@205.240.205.192) |
| 20:48.49 | bjohnson | Katty: depends on the mobo, but most max out at 2 pci cards |
| 20:49.23 | bjohnson | Katty: I think I read that on the wiki, but ManxPower is always saying that too |
| 20:49.31 | hypa7ia | P4C0: not really, that's dumb |
| 20:49.35 | InfraRed | P4C0: some companies do that |
| 20:49.45 | InfraRed | don't like it? change provider |
| 20:49.51 | [TK]D-Fender | P4C0 : Yup a LOT of places force you to use their gear to avoid abuse of service and for liability reasons when they quote 911 service. using * you could screw your setup up and not be able to dial 9111. in case that fails they don't want you suing them :) |
| 20:49.53 | P4C0 | hypa7ia: yep that's what I think... |
| 20:49.53 | InfraRed | it's not like you're lacking choice |
| 20:49.55 | Katty | bjohnson: i've been informed you're limited to pci slots, irqs, processing powerrrr, and how the chipset handles routing stuffs. |
| 20:49.57 | bjohnson | P4C0: it's so they don't have to answer questions |
| 20:50.00 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 20:50.26 | P4C0 | but here I only have 2 providers... |
| 20:50.34 | InfraRed | dude |
| 20:50.43 | InfraRed | "here" you;re on the internet |
| 20:50.50 | bjohnson | Katty: i don't know why, just that there is a limit to ow many can run well even if there are more slots |
| 20:50.50 | InfraRed | you can select from * |
| 20:51.08 | InfraRed | have a look at voip-info.org |
| 20:51.09 | Netgeeks | TDM400 cards scare me |
| 20:51.13 | bjohnson | he might be referring to did providers |
| 20:51.23 | InfraRed | did sucks |
| 20:51.24 | DrDeke | Netgeeks: Why's that? |
| 20:51.27 | InfraRed | use 1800 :) |
| 20:51.36 | DrDeke | InfraRed: that's still a did |
| 20:51.45 | P4C0 | InfraRed: yes but I want connections to the PSTN |
| 20:51.54 | InfraRed | boring |
| 20:51.55 | InfraRed | :) |
| 20:52.01 | InfraRed | glad i am in the uk |
| 20:52.01 | bjohnson | P4C0: damn near every voip provider will let you call out to the pstn |
| 20:52.11 | InfraRed | i can select any area code from many providers |
| 20:52.23 | bjohnson | P4C0: the trick is finding one with a did that will work for your purposes |
| 20:52.34 | hypa7ia | P4C0: do you need incoming? you ip indicates you're in panama, i can see why you might have a limited choice of providers :/ |
| 20:52.38 | bjohnson | P4C0: of course, it doesn't have to be just one voip provider |
| 20:52.46 | P4C0 | bjohnson: yes, but it's cheap if it's local... right? I mean I live in panama, how much will it cost to call the girl next door? |
| 20:52.51 | Netgeeks | I've never had alot of luck with analog cards & asterisk. Channels get hung, echo is a moving target, I always confuse fxo and fxs and order the wrong modules... stuff like that |
| 20:52.59 | DrDeke | heheh |
| 20:53.02 | bjohnson | P4C0: depends on the voip provider |
| 20:53.15 | bjohnson | P4C0: most will charge the same to panama, even if based in panama |
| 20:53.36 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-79.cpe.netcabo.pt) |
| 20:53.46 | bjohnson | P4C0: if you want cheap local calling, get a regular phone line |
| 20:53.47 | P4C0 | bjohnson: and will i get a panama phone number? for incoming calls? |
| 20:53.59 | bjohnson | P4C0: totally depends on what you buy |
| 20:54.38 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 20:54.40 | P4C0 | bjohnson: regular phone lines here sucks... (the requirements for new lines are worst that for taking a trip to the moon) |
| 20:55.05 | hypa7ia | so then you're probably stuck with getting an ATA |
| 20:55.10 | bjohnson | so you need to find a voip provider that offers panama dids or toll free numbers tat work in panama |
| 20:55.11 | hypa7ia | from one of the two providers |
| 20:55.56 | bjohnson | P4C0: you can still choose another voip provider for outgoing, but if you're locked to certain hardware, you may not be able to use multiple voip providers |
| 20:56.11 | P4C0 | hypa7ia: ATA? no, they have their own ata, I just want to avoid having to buy a fxo card... |
| 20:56.21 | hypa7ia | nah, you'll prolly have to |
| 20:56.43 | P4C0 | I'll take a visit to ebay... :p |
| 20:56.48 | DrDeke | That's the sound of the TDM400 police! :p |
| 20:57.00 | [TK]D-Fender | P4C0 : You don't need an FXO card, by getting a VoIP carrier, you need an ATA (fxs) or similar |
| 20:57.00 | hypa7ia | bjohnson: how would that be the case? if he/she gets an ata from them, just plugs into an fxo, and then uses something else for outgoing |
| 20:57.36 | hypa7ia | [TK]D-Fender: he/she will need an FXO if all that the provider gives is an FSX ATA |
| 20:58.21 | P4C0 | [TK]D-Fender: no, the voip provider will came here and give a little black box where I plugged my wan and eth networks, then it have a little phone jack for my phone... |
| 20:58.25 | bjohnson | hypa7ia: some provider's ata's will still allow custom configuration of the second fxs port (if there is a second fxs port) |
| 20:58.31 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
| 20:58.52 | P4C0 | that's all |
| 20:59.02 | Daniel28 | anyone has MGCP softphone? |
| 20:59.19 | bjohnson | P4C0: you're limited by what is available in your area for incoming |
| 20:59.37 | hypa7ia | but you can still use someone else for outgoing |
| 20:59.37 | bjohnson | P4C0: you could still by non-locked hardware and use someone else for outgoing |
| 21:00.14 | InfraRed | SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM |
| 21:00.18 | bjohnson | ok |
| 21:00.22 | DrDeke | wtf |
| 21:00.31 | InfraRed | </bored> |
| 21:00.47 | hypa7ia | if you do panamanian_voip_provider <> ATA <> fxo <> asterisk <> IP phones there's nothing preventing you from using another outgoing provider |
| 21:00.49 | DrDeke | me too; i want to go home and find out if this guy's advice has my TDM400 working with dial pulse phones correctly :) |
| 21:00.54 | P4C0 | and I can't use a regular modem with asterisk like fxo? |
| 21:01.01 | queuetue | InfraRed: What you say intrigues me. Can I subscribe to your newsletter? |
| 21:01.19 | hypa7ia | P4C0: nope |
| 21:01.48 | InfraRed | queuetue: available from all good newsgroups |
| 21:01.54 | P4C0 | hypa7ia: and what will be the reason for using a different provider for outgoing calls? |
| 21:02.04 | DrDeke | P4C0: To get cheaper rates possibly. |
| 21:02.08 | hypa7ia | P4C0: probably cheaper... though not necessarily |
| 21:02.25 | hypa7ia | your panamanian provider will (hopefully) give you unlimited local |
| 21:02.26 | P4C0 | 6 cent the minute... |
| 21:02.37 | hypa7ia | and then you can get way cheaper than that for north american |
| 21:02.42 | P4C0 | hypa7ia: yep, for 20 |
| 21:02.59 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 21:03.12 | *** join/#asterisk tracinet (n=tracinet@64.139.141.22) |
| 21:03.19 | DrDeke | yeah you can easily get IAX2 calls to the USA for $0.024 per minute with no monthly fees |
| 21:03.27 | ^Howler | tracinet: welcome =) |
| 21:03.28 | DrDeke | (or less) |
| 21:03.37 | tracinet | Howler - thanks for all your help |
| 21:03.53 | P4C0 | DrDeke: cool :) |
| 21:04.17 | DrDeke | Do any of you know whether VoIP is legal in Peru? |
| 21:05.08 | P4C0 | DrDeke: yes... In my last company we used it... but well... without any voip provider... |
| 21:05.14 | hhoffman | hmm, I seem to have screwed up my extensions.conf... "I'm getting the error: Can't find extension '1000' in context 'incoming'. Did you pass the wrong context to Directory?" I thought that this would make it work "exten => 2120,1,Directory(default,incoming)" |
| 21:05.14 | DrDeke | ok |
| 21:05.40 | DrDeke | I have family in Peru and in the USA; they both have cable modems at home, and I am thinking of having them each get a SIP phone or gateway and connect through my Asterisk server... I mean, they talk for at least an hour a day. |
| 21:05.44 | DrDeke | I just don't want to get them in any trouble. |
| 21:06.19 | P4C0 | DrDeke: we had an asterisk server in panama, and clients in peru and miami... in peru was really cool, cause we get an ATA with a sim card... to make local calls to mobile from mobile... (cheaper than to fixed lines) so yes, I think it should be legal... |
| 21:06.30 | LostFrog | DrDeke: VPN? |
| 21:07.03 | DrDeke | LostFrog: Sure, there are always things like that. But not being familiar with the legal system there nor the laws, I just kind of wanted to know. |
| 21:07.11 | *** join/#asterisk kn0x (n=root@adsl-68-77-35-143.dsl.emhril.ameritech.net) |
| 21:07.12 | LostFrog | It would be hard to prove what traffic was going behind a VPN. |
| 21:07.24 | perd | attention everyione, i just got a boner from spandsp, that is all. |
| 21:07.35 | kn0x | does anyone run ztdummy on 2.6.13 (specifically the gentoo release) |
| 21:07.38 | P4C0 | LostFrog: hard? even impossible |
| 21:07.46 | *** join/#asterisk webmind (n=webmind@feather.perl6.nl) |
| 21:07.53 | LostFrog | P4C0: give the NSA 5 minutes.. :) |
| 21:08.04 | DrDeke | P4C0: But on the other hand it might be pretty obvious to an observer; EXACTLY 20 packets per second of 172+vpn_overhead bytes each... |
| 21:08.20 | DrDeke | On the other hand, the NSA cares very little as to whether VoIP is legal in Peru or not ;) |
| 21:08.22 | kn0x | dmesg spits back this after a load of ztdummy fails |
| 21:08.23 | kn0x | ztdummy: Unknown symbol rtc_unregister |
| 21:08.23 | kn0x | ztdummy: Unknown symbol zt_register |
| 21:08.25 | kn0x | ztdummy: Unknown symbol rtc_control |
| 21:08.26 | DrDeke | i guess it'd be the equivalent of the PSA |
| 21:08.53 | LostFrog | I wrote a new game, DrDeke. It's simulation engine runs at 30/sec. |
| 21:08.59 | DrDeke | Yeah, yeah, I know :) |
| 21:09.13 | LostFrog | Pure coincidence on the packet size. :) |
| 21:09.26 | DrDeke | I am familiar with how these things work; I just wanted to know that's all ;0 |
| 21:10.04 | Rawplayer | re |
| 21:10.07 | kn0x | can anyone help me |
| 21:10.07 | kn0x | ? |
| 21:10.25 | DrDeke | kn0x: Yes, are you running Linux 2.6? If so, you need to recompile your kernel to add real time clock suppot. |
| 21:10.26 | DrDeke | support* |
| 21:10.28 | LostFrog | I've heard talk of termination in China.. I can't imagine if it's legal in China, it would be illegal in Peru. |
| 21:10.35 | DrDeke | lol |
| 21:10.45 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 21:10.45 | DrDeke | it's cheap as all get out to call China on VoIP |
| 21:10.47 | DrDeke | from many providers |
| 21:10.50 | DrDeke | 2-3c/min |
| 21:10.52 | tracinet | Trying to understand how/when bug fixes are updated into the CVS source - I reported a bug today (ID 0005766) regarding an issue on v.1.2rc2 and it was resolved VERY quickly (very impressed with the turn around) - my question is this - The note attached to the bug says Committed to CVS HEAD - does that mean that it will not make it into the CVS for rc2 if I download it again? |
| 21:11.25 | LostFrog | rc2 is not CVS HEAD. |
| 21:11.39 | DrDeke | CVS head is kind of the ongoing-project |
| 21:11.48 | tracinet | that's what i wanted to know - gotcha |
| 21:11.55 | DrDeke | question is, will it make 1.2.0? :) |
| 21:12.02 | tracinet | so that means the bug still exists in 1.2 as of now |
| 21:12.03 | *** join/#asterisk gr0mit_home (n=wendolen@extrt.txrx.org.uk) |
| 21:12.05 | LostFrog | It very well could. |
| 21:12.07 | Blackthorn | How come I get the ip address of my wireless router instead of the ata? When doing show sip peers? router --- Wireless --- wet11 wireless bridge --- spa 2000 |
| 21:12.20 | DrDeke | No release (or rc) changes once it is released. So yes, if you download 1.2rc2 the bug would be there. |
| 21:12.46 | tracinet | thanks - makes sense - just need to wait for next release to see if it got fixed |
| 21:12.49 | DrDeke | yup |
| 21:12.54 | DrDeke | which is supposed to be today i guess |
| 21:13.00 | tracinet | that would be timely LOL |
| 21:14.56 | kn0x | no one has anyideas? |
| 21:15.16 | DrDeke | kn0x, I told you how to fix it a few minutes ago |
| 21:15.34 | DrDeke | <DrDeke> kn0x: Yes, are you running Linux 2.6? If so, you need to recompile your kernel to add real time clock suppot. |
| 21:15.34 | DrDeke | <DrDeke> support* |
| 21:16.02 | kn0x | dr. dreke ive done that |
| 21:16.11 | DrDeke | if you built it as a module, did you load it? |
| 21:16.12 | kn0x | im running genrtc as a module |
| 21:16.18 | kn0x | and its loaded |
| 21:16.27 | DrDeke | in lsmod, the name of the module is genrtc? |
| 21:16.29 | LostFrog | Now, recompile zaptel |
| 21:16.40 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 21:16.47 | *** join/#asterisk bumblefsck (n=bumblefs@69-160-145-156.ontrca.adelphia.net) |
| 21:16.57 | LostFrog | I don't believe 1.2.0 will be released today. |
| 21:17.01 | kn0x | lsmod shows genrtc |
| 21:17.06 | DrDeke | interesting |
| 21:17.09 | DrDeke | on my machine, it shows 'rtc' |
| 21:17.11 | LostFrog | It doesn't seem like RC2 has been out log enough. |
| 21:17.38 | X-Files | Is there a setting in the * that controls the audio for the canned voice prompts (voice mail, meet me conf). Anytime the canned voice comes on it sounds like stuttering. This is happening on internal calls in *. Voice quality between phones internally sounds fine in gateway. |
| 21:17.59 | kn0x | DrDeke yes.... would this mean anything to you http://lists.digium.com/pipermail/asterisk-dev/2005-August/014293.html |
| 21:18.14 | ManxPower | X-Files: No. There should be no stuttering with the sounds files in Asterisk |
| 21:18.36 | DrDeke | kn0x: *shrug* |
| 21:18.56 | DrDeke | kn0x: I have a digium card in there now, so I don't use it any more :) |
| 21:18.56 | kn0x | but http://bugs.digium.com/view.php?id=4301 says it was fixed in zaptel months ago |
| 21:19.59 | X-Files | ManxPower: really have stuttering or lost sound |
| 21:20.28 | *** join/#asterisk Timoti (n=asqsa@85.99.166.94) |
| 21:20.33 | Timoti | hi |
| 21:20.44 | *** join/#asterisk robtro (n=lol@unaffiliated/robtro) |
| 21:21.07 | DrDeke | hi |
| 21:21.11 | Timoti | are there any good and user friendly dynamic ( with Mysql connection ) Least cost routing stuff |
| 21:21.14 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 21:21.15 | Timoti | Hi Dr |
| 21:21.21 | robtro | has anyone here successfully (or know anywhere to start looking) used php to `explode' the results of 'asterisk -r -x 'sip show XXX'' ? |
| 21:22.03 | CunningPike | Greetings - exciting news in the topic |
| 21:22.05 | LostFrog | OMFG.. I just found a Bt-chip based TV card in my old G3 mac.. |
| 21:22.14 | LostFrog | DVR, here I come. :) |
| 21:22.24 | Timoti | no ??? |
| 21:22.59 | *** join/#asterisk Astinus (i=iBook@freenode/staff/gentoo.astinus) |
| 21:23.24 | Blackthorn | How come I get the ip address of my wireless router instead of the ata? When doing show sip peers? router --- Wireless --- wet11 wireless bridge --- spa 2000 |
| 21:23.26 | fulgas | ( robtro ): better with a preg_match ? or * manager ? |
| 21:23.35 | CunningPike | Does anyone know what this means in /var/log/asterisk/messages: Nov 16 08:36:06 WARNING[13204]: Unable to handle indication 3 for 'SIP/2420-2a55' |
| 21:23.42 | robtro | "* manager" ? |
| 21:23.45 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 21:24.08 | *** join/#asterisk convey (n=test@66.55.43.2) |
| 21:24.31 | CunningPike | Oh, on 1.0.9 |
| 21:25.43 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 21:25.57 | hhoffman | Why can't Directory find my vmail box? Can't find extension '1000' in context 'incoming'. Did you pass the wrong context to Directory? exten => 1000,1,VoiceMail(b1000@default) exten => 2120,1,Directory(default,incoming) |
| 21:26.32 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 21:26.52 | Netgeeks | it's bugged in the version you have |
| 21:27.02 | Netgeeks | upgrade to 1.2 |
| 21:27.07 | Druken | hhoffman: why do you have an incoming context for voicemail ? |
| 21:27.21 | LostFrog | I thought it was just supposed to be VoiceMail(b1000) |
| 21:27.46 | Druken | LostFrog: you can specify a voicemail context |
| 21:28.18 | hhoffman | Drunken: [incoming] includes all incoming calls, which go to a [auto-attendent] which includes [extensions] |
| 21:28.58 | Druken | ok, that's dialplan... directory don't care about dialplan, it wants voicemail |
| 21:29.00 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 21:29.33 | hhoffman | oh, so I don't need to include the context it's coming from? |
| 21:29.49 | Druken | no... |
| 21:30.03 | hhoffman | shouldn't it use default since I specify that as the vm context? |
| 21:30.06 | Druken | you need to tell directory what VOICEMAIL context you want |
| 21:30.42 | hhoffman | right, doesn't exten => 2120,1,Directory(default,incoming) do that? |
| 21:31.06 | hhoffman | book says Directory(vm-context[,dial-context[,options]]) so I thought that was right? |
| 21:31.45 | CunningPike | hhoffman: The second context is the one in which to place the eventual call |
| 21:32.17 | CunningPike | I use exten => 1234,1,Directory(default,${CONTEXT}) which places the eventual call in the users current context |
| 21:32.28 | hhoffman | hmm, ok.. but exten => 2120,1,Directory(default) doesn't work either... |
| 21:32.36 | hhoffman | Can't find extension '1000' in context 'default'. |
| 21:33.02 | ManxPower | hhoffman: do you have an extenson 1000 in the context default? |
| 21:33.06 | *** join/#asterisk gvag11 (n=g@ppp37-adsl-107.ath.forthnet.gr) |
| 21:33.08 | CunningPike | Right - because 1000 doesn't exist in the default context |
| 21:33.12 | Druken | for dialplan :) |
| 21:33.17 | X-Files | My problem is that I am having Audio Quality issues within my local network. If I call straight from X-lite to Asterisk (let's say to record VM msg or listen to it) the quality is very bad. Lots of Crackling, hissing, etc. That result is both with 711u or a. |
| 21:33.18 | *** part/#asterisk rculp (n=rculp@66.173.240.20) |
| 21:33.38 | hhoffman | but voicemail.conf shows... [default] 1000 => xxxx |
| 21:33.40 | DrDeke | X-Files: Sounds like you need to fix your network. |
| 21:33.40 | gvag11 | Hi guys |
| 21:33.42 | LostFrog | Is that a timing issue? |
| 21:33.56 | X-Files | DrDeke: network lan 100mbit ! |
| 21:34.03 | CunningPike | hhoffman: That's the voicemail context - where is 1000 in your extensions.conf |
| 21:34.05 | X-Files | i can show ping :) |
| 21:34.13 | *** part/#asterisk Timoti (n=asqsa@85.99.166.94) |
| 21:34.19 | hhoffman | ManxPower: I don't have a default context in extension.conf :-( |
| 21:34.38 | hhoffman | 1000 is in the incoming context |
| 21:34.48 | gvag11 | How can i detect frame slips on a TE210P? And is there a way to resolve this? |
| 21:34.50 | Blackthorn | How come I get the ip address of my wireless router instead of the ata? When doing show sip peers? router --- Wireless --- wet11 wireless bridge --- spa 2000 |
| 21:34.58 | hhoffman | so, I have to change the voicemail to incoming context then? |
| 21:35.33 | CunningPike | What context is the person who dialed the directory in? |
| 21:35.50 | CunningPike | incoming as well? |
| 21:35.54 | hhoffman | CunningPike: incoming |
| 21:36.13 | CunningPike | Hmm: well then Directory(default,incoming) should work |
| 21:36.20 | CunningPike | But I guess you already knew that ;) |
| 21:36.24 | hhoffman | incoming points to auto-attendent which sets context to incoming and includes extensions |
| 21:36.44 | hhoffman | it was working :-( I screwed something up, just don't know what it was |
| 21:36.54 | hhoffman | shoulda svn'd this shit :-/ |
| 21:37.16 | gvag11 | How can i detect frame slips on a TE210P? And is there a way to resolve this? |
| 21:37.19 | *** join/#asterisk br00ksh1r3 (n=matt@wsip-24-120-60-190.lv.lv.cox.net) |
| 21:37.57 | LostFrog | Usually if the picture is crooked on the wall, the frame slipped. <Grin> |
| 21:38.23 | DrDeke | LostFrog: Sounds like something 1-800-TEAM-DATA would tell me regarding my PRI lines :\ |
| 21:39.12 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 21:39.43 | *** join/#asterisk tclark_ (n=TC@S0106000f66c5d294.gv.shawcable.net) |
| 21:40.05 | CunningPike | gvag11: when we were getting slips, we got message in the console |
| 21:40.14 | CunningPike | Now, my turn |
| 21:40.33 | CunningPike | Does anyone know what this means in /var/log/asterisk/messages: Nov 16 08:36:06 WARNING[13204]: Unable to handle indication 3 for 'SIP/2420-2a55' |
| 21:41.29 | hhoffman | is that distinctive ring? |
| 21:41.33 | CunningPike | hhoffman: if you pastebin your extensions.conf, I can take a look at it if you want |
| 21:41.39 | CunningPike | I have no idea |
| 21:41.42 | CunningPike | :) |
| 21:41.47 | hhoffman | CunningPike: ok, thanks |
| 21:42.24 | CunningPike | I'm not experiencing any problems, I just noticed these messages since we updated to 1.0.9 |
| 21:43.55 | gvag11 | CunningPike: Is there a way to resolve this? |
| 21:44.04 | CunningPike | Yes :) |
| 21:44.12 | CunningPike | We got messages like PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 21:44.24 | CunningPike | And other stuff |
| 21:44.34 | ManxPower | CunningPike: Your extensive search of the Wiki and mailing list archives were not helpful? |
| 21:44.39 | CunningPike | We fixed it with the old IRQ shuffle |
| 21:44.54 | CunningPike | ManxPower: If they were, do you think I'd be here? |
| 21:45.17 | LostFrog | I found one mail list posting from a person with the same problem, but there was no response. |
| 21:45.27 | ManxPower | CunningPike: indication messages are caused by the lack of /etc/asterisk/indications.conf |
| 21:45.40 | CunningPike | Ah - thanks |
| 21:45.46 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:45.47 | CunningPike | I'll follow that |
| 21:45.53 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 21:46.30 | hhoffman | here you go: http://pastebin.ca/28955 |
| 21:46.50 | gvag11 | CunningPike: What are the ways to resolve the frame slips? |
| 21:46.52 | ManxPower | CunningPike: HDLC errors are caused by 1 of 2 things. 1) your T-1/PRI is getting errors. 2) high interrupt lacency on your PCI bus is causing corrupted data from the card, common causes of this are SATA, RAID, onboad LAN, graphics mode, and sometimes that specific brand/model of motherboard just has very high interrupt latency. |
| 21:47.15 | CunningPike | ManxPower - I was helping gvag11 with his errors, not mone |
| 21:47.17 | CunningPike | mine |
| 21:47.26 | ManxPower | CunningPike: Ah. Well now he knows 8-) |
| 21:47.34 | CunningPike | I rarely come here just to ask questions |
| 21:47.40 | DrDeke | Speaking of latency, is it best (in linux 2.6) to set your kernel to non-preemptible, medium-preemption or always-preemption, for an asterisk system with one or more TDM400s? |
| 21:47.44 | CunningPike | <PROTECTED> |
| 21:48.17 | ManxPower | DrDeke: the traditional answer is "Digium cards don't work well with preemptive set on". I don't know if that's true for 2.6 or not. |
| 21:48.24 | DrDeke | thakns |
| 21:48.27 | gvag11 | Manxpower : Thanks.. So i should try to disable some devices to get a better IRQ? |
| 21:48.39 | DrDeke | I don't have any preference for which setting to use; I just want to use what is recommended for Asterisk. |
| 21:48.41 | r0d3nt|m | test the irq availability |
| 21:48.50 | ManxPower | gvag11: always disable all onboard devices you don't need even before you bother to install the OS. |
| 21:48.56 | ManxPower | You can do that after, of course. |
| 21:49.48 | ManxPower | We disable onboard LAN, USB, SATA. |
| 21:49.59 | ManxPower | We disable the printer ports, and sometimes the serial ports. |
| 21:49.59 | gvag11 | manxpower, everything is disabled just the requireds one are on but still i have problem. should i check on an other mainboard maybe? |
| 21:50.16 | ManxPower | gvag11: Are you using RAID, SATA, or graphics? |
| 21:51.05 | gvag11 | manxpower, RAID onboard controller Intel ICH5 (or something) |
| 21:51.23 | ManxPower | gvag11: don't use RAID. See if that fixes it. |
| 21:51.54 | ManxPower | Things like RAID and graphics want to provide the absolute highest performance, which means pushing other devices out of the way when it comes to servicing interrupts |
| 21:52.18 | *** join/#asterisk rontecxt44 (n=rontecxt@ns1.dreamdeferred.org) |
| 21:52.39 | gvag11 | manxpower , i will try this |
| 21:53.14 | rontecxt44 | hi all....does anyone have any experience with getting snom phones to answer call waiting? |
| 21:54.28 | gvag11 | Manxpower, do you think that i can use a RAID controller (not on-board) without problems with Asterisk? |
| 21:54.46 | *** join/#asterisk bbz (i=badboyz@adsl-70-128-78-21.dsl.stlsmo.swbell.net) |
| 21:55.01 | ManxPower | gvag11: maybe. Get it working reliable without RAID. Why do you want RAID? Asterisk is not normally disk intensive. |
| 21:55.40 | ManxPower | gvag11: It's not actually RAID that's the problem, it's the actual specific make/model of the raid card / raid chip and the drivers. |
| 21:55.51 | gvag11 | mkanxpower, i want the box for send and receive faxes .... |
| 21:56.13 | ManxPower | gvag11: lots of faxes using spandsp? |
| 21:57.03 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 21:57.19 | gvag11 | manxpower, i am using spandsp and its doing really fine BUT after 30 active calls on the TE205P things going crazy... |
| 21:58.14 | hhoffman | hmm, I can login to voicemail just fine too :-( |
| 21:58.44 | juanjoc | Can anyone tell me if Asterisk supports the RTP payload type 100 for DTMFs? |
| 21:58.55 | *** join/#asterisk svenna_ (n=svenna@p548D289C.dip0.t-ipconnect.de) |
| 22:01.31 | *** join/#asterisk bweschke_ (n=bweschke@wsip-24-120-60-190.lv.lv.cox.net) |
| 22:02.51 | ManxPower | juanjoc: Asterisk supports RFC2833 OOB DTMF |
| 22:02.58 | ManxPower | I don't what the payload type value is. |
| 22:04.09 | *** part/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 22:05.54 | juanjoc | Supposedly the payload type for DTMFs is 101 |
| 22:06.02 | ManxPower | Log Message: don't build chan_modem by default |
| 22:06.17 | ManxPower | Yay! Another %10 reduction in questions here! |
| 22:06.18 | *** join/#asterisk docelmo (n=docelmo@66.237.242.41.ptr.us.xo.net) |
| 22:06.45 | juanjoc | But in some tests with one of our providers they are sending 100 as payload type for the DTMFs and it works, but I cannot see how it works |
| 22:07.07 | ManxPower | juanjoc: Asterisk may look at the payload type, not the number |
| 22:07.16 | juanjoc | Supposedly Asterisk only supports payload type 100 and 121 (Cisco DTMF) for this |
| 22:07.17 | file | juanjoc: like Kevin said that'll be negotiated dynamically on inbound in the SDP |
| 22:07.21 | file | it'll look for telephone-event |
| 22:07.23 | juanjoc | Like telephone-event |
| 22:07.32 | juanjoc | Ah, I understand |
| 22:07.34 | file | but on outgoing it'll send rfc2833 as payload 101 |
| 22:07.45 | juanjoc | OK, thanks, now I see |
| 22:07.59 | gvag11 | manxpower, thanks a lot for your advices.... |
| 22:08.01 | gvag11 | Bye |
| 22:09.05 | *** part/#asterisk rontecxt44 (n=rontecxt@ns1.dreamdeferred.org) |
| 22:13.57 | *** join/#asterisk tracinet (n=tracinet@216.242.235.2) |
| 22:15.21 | *** join/#asterisk sapo_original (n=Fenix@200.138.76.58) |
| 22:15.39 | sapo_original | hi .. i need a little help.... |
| 22:17.12 | sapo_original | anywhere knows the ata or some else for i plug the pstn line and voip line in one telephone? |
| 22:18.00 | *** part/#asterisk InfoLibre[Frank] (n=ncc@modemcable228.85-70-69.mc.videotron.ca) |
| 22:20.26 | ManxPower | sapo_original: you are looking for a SIPura SPA3000, but the PSTN port is NOT easy to set up |
| 22:20.29 | *** join/#asterisk BladeRunner05 (n=gianni@adsl-ull-8-64.44-151.net24.it) |
| 22:20.33 | *** join/#asterisk tracinet (n=tracinet@216.242.235.2) |
| 22:21.31 | Assid | i like the linksys pap2 |
| 22:21.48 | ManxPower | Assid: Other than I don't think the PAP2 has a PSTN port, only FXS ports |
| 22:22.04 | Assid | yep |
| 22:22.20 | Assid | i was just making a statement really |
| 22:23.08 | tzafrir_laptop | is sethdlc-new used for digium pri cards? if so: what do I use for INTERFACE? |
| 22:23.14 | hugo-v6 | gd evening. |
| 22:25.30 | *** join/#asterisk Kyreeth (n=ashley@aquila.feathers.net) |
| 22:26.18 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 22:26.26 | hugo-v6 | q: sip-phone rings, want to pick up call on other sip-phone. is this the function configured in features.conf or is this possible? |
| 22:27.06 | tzafrir_laptop | ManxPower, I take it that the answer is negative |
| 22:27.16 | ManxPower | hugo-v6: see callgroup= and pickupgroup= in sip.conf.sample The correct term is "call pickup", Asterisk does not support DIRECTED call pickup |
| 22:27.38 | ManxPower | tzafrir_laptop: I use Ciscos for routing, rather than Linux |
| 22:28.13 | hugo-v6 | Manx: what means directed call pickup? thanks for the hints. |
| 22:28.14 | hhoffman | openbsd has been showing some promise for ability to route packets in a timely manner :-) |
| 22:28.48 | tzafrir_laptop | ManxPower, Asterisk's main use of PRI is for voice, rather than data |
| 22:29.01 | ManxPower | hugo-v6: Asterisk's callpick up will pick up ANY ringing phone. directed pickup allows you to specify the ringing phone you want to pickup. 1.2/CVS-HEAD has I thnik some directed call pickup features |
| 22:29.35 | ManxPower | tzafrir_laptop: I assume you have a T-1 coming in with both PRI B and D channels and other channels are just plain data channels? |
| 22:29.48 | Netgeeks | Hrm, I would hope PRI's are not used for data. thier data channel would be somewhat limiting... |
| 22:30.24 | infinity1 | is there a problem with having multiple phones attached to the same context in sip.conf? |
| 22:30.40 | ManxPower | Netgeeks: No. The carrier assigns some of the channels on your T-1 to voice/PRI (say, channels 1-17 and 24) and some channels for data (say frame relay on channels 18-23) |
| 22:30.54 | ManxPower | infinity1: Other than the fact that you cannot do it, no. |
| 22:31.13 | infinity1 | ManxPower: oh :) ..it seemed to work. but okay :) |
| 22:31.51 | ManxPower | infinity1: phone 1 registers to [happyphone] section, then phone 2 registers using the same username and password, asterisk will forget about phone 1 and send calls to phone 2 |
| 22:32.17 | hugo-v6 | ManxPower: oh ok thank you. but directed pickup is not needed atm. (the last thing i hope is i get it to work. rumors say, it wont work with snom phones) |
| 22:32.19 | infinity1 | ManxPower: that makes sense. i didn't test it much. |
| 22:32.19 | ManxPower | then phone 1 registers again and asterisk sends calls to phone 1, then phone 2 registers and calls go to phone 2 |
| 22:32.22 | *** join/#asterisk SERGEUS (i=sergey@195.112.98.13) |
| 22:32.40 | infinity1 | ManxPower: whats the best way to get two phones to be the same extenion then? |
| 22:32.43 | SERGEUS | hi! is there anybody from voipjet stuff? |
| 22:32.50 | Assid | this is weird |
| 22:32.58 | Assid | i get a call on my phone.. but the caller id is wrong |
| 22:33.04 | Assid | shows my extension instead |
| 22:33.08 | ManxPower | infinity1: Don't have them register as the same userid, then use Dial(SIP/phone1&SIP/phone2) |
| 22:33.23 | ManxPower | In fact we don't have ANY association between the extension and the sip userid on our systems. |
| 22:33.40 | Assid | but i have it set to Set(CALLERIDNUM=${CALLERIDNUM}) |
| 22:33.46 | Assid | as i get from the provider |
| 22:33.51 | ManxPower | Each phone registers using it's MAC address (all lower case) as it's userid. If a phone supports more than 1 line, we add a -a -b -c, etc to the username |
| 22:34.01 | hugo-v6 | damn shot googling said it will work. |
| 22:34.20 | hugo-v6 | s/ot/ort/ |
| 22:34.34 | infinity1 | ManxPower: what do you use for the password? blank? |
| 22:34.46 | ManxPower | infinity1: the password is the same as the userid |
| 22:35.07 | ManxPower | Since, in theory, you can't find out the MAC address unless you are on the local lan or have physical access to the phone. |
| 22:35.23 | infinity1 | so you have [205] context (extension #) and under that user and secret with the MAC |
| 22:35.50 | ManxPower | infinity1: no, I have a [0004f20189eb-a] section |
| 22:36.05 | infinity1 | ManxPower: what it is a softphone? |
| 22:36.14 | infinity1 | s/it/if/ |
| 22:36.27 | ManxPower | infinity1: Softphones suck. They are only used by people too cheap to buy a real phone. |
| 22:36.41 | infinity1 | ManxPower: i realize that :) ..what "if" |
| 22:36.42 | ManxPower | infinity1: we don't use softphones. If we did, we would use the MAC of the PC |
| 22:37.02 | infinity1 | interesting ideas. |
| 22:37.20 | ManxPower | infinity1: It helps us remember than a DEVICE IS NOT AN EXTENSION |
| 22:37.24 | ManxPower | than == that |
| 22:37.52 | infinity1 | ManxPower: yea. i think that should be bolded somehwere in the voip-info . heh |
| 22:38.06 | ManxPower | an extension is an abstraction layer, accessed using DTMF, to provide 1 or more destinations for a call to ring at. |
| 22:38.11 | *** join/#asterisk cp5 (n=samy@69.111.181.112) |
| 22:38.30 | *** join/#asterisk Lurr (n=pr0ph3t@63.69.20.3) |
| 22:39.29 | ManxPower | The other nice thing is that since all our phones have the MAC printed on a sticker on the bottom, we can find the phone by asking the user what that number on the bottom of the phone is. |
| 22:39.57 | ManxPower | Rather than asking them what their extension is and having them tell us one (randomly) of the three extension buttons on thier phone, each with a different extension on it. |
| 22:40.18 | infinity1 | we have mobile users and have actually started using softphone so they can be in the office while at a client site. |
| 22:40.33 | ManxPower | i..e button 1 is the person's personal extension, button 2 is their boss's extension, and button 3 is the main switchboard extension |
| 22:40.48 | Assid | hrmm anyoe have this linksys pap2? |
| 22:41.04 | ManxPower | Assid: only idiots. All the cool kids use the PAP2-NA |
| 22:41.19 | justinu | yeah, dumbass |
| 22:41.20 | ManxPower | And the UberCool kids use SIPura. |
| 22:41.28 | Assid | na? |
| 22:41.40 | ManxPower | PSP2s are locked to a provider and almost impossible to unlick |
| 22:41.43 | ManxPower | unlock too |
| 22:41.50 | Assid | hrmm.. this aint locked |
| 22:41.58 | ManxPower | Assid: then it might be a PAP2-NA |
| 22:42.07 | Assid | weird.said pap2-as |
| 22:42.36 | *** part/#asterisk SERGEUS (i=sergey@195.112.98.13) |
| 22:42.47 | Assid | anyways, everytime i get a call. it shows the wrong caller id.. it shows my extension on my caller id instead of the calling party |
| 22:42.57 | ManxPower | Assid: weird |
| 22:43.14 | ManxPower | I wonder why so many people have problems that I've never, ever experienced. |
| 22:43.17 | bbz | i recommend formatting. |
| 22:43.28 | Assid | formatting? |
| 22:43.33 | bbz | formatting. |
| 22:43.50 | Assid | i even have Set(CALLERIDNUM=${CALLERIDNUM}) |
| 22:44.11 | Assid | i even set the name.. but did it do it. no.. |
| 22:44.13 | ManxPower | Assid: Um, if the calleridnum is wrong, setting it to the wrong value won't do any good. |
| 22:44.14 | justinu | that won't work |
| 22:44.18 | ManxPower | That's like setting 1=1 |
| 22:44.29 | Assid | the verbose shows it correct |
| 22:44.33 | ManxPower | or if $BOB = 1 and then setting $BOB = 1 |
| 22:44.40 | Assid | verbose shows it to be correct |
| 22:44.52 | *** join/#asterisk kiwnix (n=egarcia@82.158.153.4) |
| 22:44.53 | justinu | Set(callerid(number) = ${CALLERIDNUM}) |
| 22:45.02 | justinu | or maybe it was Set(CALLERID(num) = |
| 22:45.27 | ManxPower | justinu: only if you are using CVS-HEAD or 1.2 |
| 22:45.45 | ManxPower | Assid: put your sip.conf entry for that device on pastebin.ca |
| 22:46.09 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-60-57.cybersurf.com) |
| 22:46.33 | ManxPower | Assid: regardless, that's not the right way to set the calleridnum. Try SetCIDNum(${CALLERIDNUM}) |
| 22:47.13 | Assid | okay.. set(callerid(name) fixed it little bit.. num i gotta work on |
| 22:47.15 | ManxPower | But that's still the same as setting 1 = 1 |
| 22:47.41 | IronHelix | why is it that the only decent sip voice/video softphone is eyebeam |
| 22:47.50 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 22:48.11 | IronHelix | and/or does anybody know of any decent free/OSS voice/vid SIP softphone? |
| 22:48.14 | syzygyBSD | grr, VPN'ing ruins all my connections |
| 22:48.21 | IronHelix | (not including ripped off eyebeam) |
| 22:48.32 | ManxPower | syzygyBSD: that should NOT suprize you. |
| 22:48.48 | ManxPower | syzygyBSD: what SPECIFIC VPN software are you using? |
| 22:48.49 | syzygyBSD | oh, not at all, just annoying |
| 22:48.58 | syzygyBSD | my computer should know what to VPN and what not to |
| 22:49.01 | syzygyBSD | lol... |
| 22:49.32 | syzygyBSD | ManxPower: just windows |
| 22:49.42 | ManxPower | syzygyBSD: so PPTP then. |
| 22:49.42 | IronHelix | syz if you're using windows, theres an option in vpn connection setup for 'use default gateway on remote network', if you turn that off it wont route every single socket through vpn |
| 22:50.04 | hardwire | a beep before channel redirecting would be cool |
| 22:50.07 | IronHelix | if you use dial up networking / network connections to make pptp that is |
| 22:50.13 | ManxPower | Of course Windows supports IPSec, PPTP, L2PT, and I'm sure it supports others |
| 22:50.44 | syzygyBSD | IronHelix: so there is.. thanks a ton |
| 22:50.57 | bbz | has anyone had success with using Windows DHCP server & setting the tftp server, with PolyCom phones? |
| 22:50.58 | IronHelix | no problem. i assume you found it? its like 6 screens in |
| 22:51.23 | syzygyBSD | yup, there is only 1 place to configure a tcp/ip gateway i know of.. easy to find |
| 22:51.27 | ManxPower | bbz: If anyone did the universe would implode and we would all die. |
| 22:51.35 | bbz | :) |
| 22:51.45 | justinu | bbz: me |
| 22:52.12 | infinity1 | ManxPower: awesome. i like this better. |
| 22:52.22 | ManxPower | infinity1: like what better? |
| 22:52.26 | bbz | justinu: how've you accomplished it? my polycoms refuse to communicate w/ the bootserver |
| 22:52.39 | infinity1 | so in the extensions.conf i put EXT207=SIP/00047641d8e3&SIP/00047641d800 |
| 22:52.52 | ManxPower | infinity1: Yup. |
| 22:53.01 | ManxPower | We actually have scripts that handle lots of that stuff for us. |
| 22:53.04 | infinity1 | and now i do exten => 207,1,Macro(stdexten3,207,${EXT207}) |
| 22:53.10 | justinu | bbz: i'm running bootrom 3.1 and sip application 1.6.2 |
| 22:53.11 | syzygyBSD | nice! still connected! |
| 22:53.16 | IronHelix | naming sip channels by mac address? thats not a bad idea... |
| 22:53.31 | infinity1 | ManxPower: yea. that makes sense. but we have like 10 phones. this is an improvement. |
| 22:53.33 | syzygyBSD | that is too nice, thanks a ton IronHelix |
| 22:53.34 | bbz | justinu: so i've gotta get those updated, before i try to use tftp? |
| 22:53.41 | IronHelix | no prob :) |
| 22:53.42 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 22:53.45 | justinu | bbz: i also only use http for provisioning, not tftp |
| 22:53.56 | justinu | bbz: are you having trouble assigning IP address to the phones? |
| 22:54.10 | RoyK | i don't get this |
| 22:54.18 | IronHelix | sup roy |
| 22:54.18 | bbz | justinu: nope, they grab ip's fine -- but when it tries to grab the tftp info from dhcp -- it wont grab it |
| 22:54.36 | justinu | bbz: option 66? |
| 22:54.40 | bbz | just: correct |
| 22:54.46 | ikarus | hmmmmm, the incoming CID can you do anything with it other then matching (like modifying it) ? |
| 22:54.56 | IronHelix | maybe its looking for a SRV entry for tftp? |
| 22:55.03 | RoyK | earlier tonight my asterisk server suddenly started to report TOO LAGGED and then REACHABLE and then flip-flopping between those two |
| 22:55.07 | RoyK | on iax2 |
| 22:55.08 | bbz | IronHelix: SRV? |
| 22:55.10 | justinu | bbz: you set it up as a "scope option"? |
| 22:55.19 | IronHelix | ikarus- sure you can |
| 22:55.23 | RoyK | restarting asterisk helped |
| 22:55.26 | RoyK | sick asterisk |
| 22:55.31 | IronHelix | you can add/remove digits on either end |
| 22:55.52 | bbz | justinu: option 66 - Boot Server Host Name -> String Value: 192.168.x.x |
| 22:55.57 | M-I-A | is there a site with a list of popular applications people have made for *? |
| 22:56.09 | RoyK | M-I-A: voip-info.org? |
| 22:56.09 | justinu | bbz: hmm, i didn't have trouble with that... which bootrom? |
| 22:56.11 | RoyK | asterisk.org? |
| 22:56.13 | ikarus | IronHelix: which functions/commands/website should I look for for info on it ? |
| 22:56.17 | bbz | justin: lemme double check. |
| 22:56.22 | RoyK | M-I-A.org/asterisk/? :) |
| 22:56.28 | IronHelix | http://www.voip-info.org/wiki-Asterisk+variables scroll down to substrings |
| 22:56.39 | RoyK | M-I-A: the official ones are there with a 'show applications' |
| 22:56.46 | M-I-A | come on guys lets not be too funny now :) |
| 22:56.46 | RoyK | the unofficial are all over |
| 22:56.47 | IronHelix | use the variable ${CALLERIDNUM} and you can use the syntaxes there to drop digits on either side or by pattern |
| 22:56.59 | *** join/#asterisk pcm (n=pcm@user-69-73-0-22.knology.net) |
| 22:57.29 | bbz | justinu: BootRom |
| 22:57.29 | bbz | 2.5.0 |
| 22:57.32 | RoyK | M-I-A: take a look at voip-info.org and asteriskdocs.org. if you're looking for a particular asterisk appp, just ask |
| 22:57.35 | ikarus | IronHelix: so you can modify ${CALLERIDNUM} ? |
| 22:57.36 | M-I-A | yeah unofficial is what i am interested in |
| 22:57.49 | IronHelix | yes, do SetCallerIDNUM(whateveryouwant) |
| 22:57.51 | RoyK | M-I-A: what sort of functinality is it you need? |
| 22:57.55 | justinu | bbz: hmm... i'd tell you to upgrade to at least 2.6.2, but you might have a tough time doing that if you can't get your phone to talk to tftp |
| 22:58.03 | RoyK | IronHelix: cvs head or 'stable'? |
| 22:58.14 | RoyK | 2.6.2 is, what, two years old? |
| 22:58.22 | bbz | justinu: i can manually set the ftp, and get it that way -- i was hoping to not have to touch the phones initially |
| 22:58.26 | IronHelix | stable. whatever you want can include ${CALLERIDNUM} which will put in the caller id as it currently is set. you can use the substrings thing i linked above to modify how it will be set |
| 22:58.31 | IronHelix | eh |
| 22:58.38 | IronHelix | yeah |
| 22:58.48 | M-I-A | a while back i saw a web based config util, but i cant remember the name nor where i saw it... |
| 22:58.48 | IronHelix | i think it might be a bit different in HEAD/1.2x) |
| 22:58.49 | justinu | bbz: understand... my suggestion might be to look at the release notes for 2.5 and see if it's looking in a different dhcp option for the boot server |
| 22:58.58 | bbz | justinu: good call -- will do that. |
| 22:59.13 | RoyK | M-I-A: check voip-info.org |
| 22:59.19 | RoyK | M-I-A: best place to look |
| 22:59.27 | M-I-A | ok will do, thanks |
| 22:59.30 | ikarus | IronHelix: hmmmm, no regexing, a shame, ah well, |
| 22:59.43 | IronHelix | i think theres also a way to regex it |
| 22:59.48 | lesouvage | Ikarus: I will paste an example to pastebin. wait a moment |
| 23:00.15 | justinu | bbz: if you can upgrade to 3.1, you can use http provisioning, which IMO is much easier |
| 23:00.22 | justinu | https even |
| 23:00.37 | ikarus | But there are some other things I want to try |
| 23:00.49 | justinu | royk: you a polycom expert? |
| 23:00.55 | bbz | justinu: essentially in the tftp string you just pass https:// rigt? |
| 23:01.01 | justinu | bbz: correct. |
| 23:01.16 | justinu | https://username:password@server.com/polycom |
| 23:01.18 | justinu | or something like that |
| 23:01.31 | RoyK | góða kvöldið |
| 23:01.35 | bbz | thats what i was even trying w/ ftp, ftp://xxx and wasnt picking up on it |
| 23:01.36 | ikarus | Reminds me, need to fire of an e-mail to Grandstream |
| 23:01.37 | RoyK | justinu: not at all |
| 23:02.13 | justinu | bbz: yeah, the older boot roms don't understand URI's like that |
| 23:02.27 | lesouvage | ikarus: this is an example I'm using myself. exten => s,1/0[1-9].,SetCallerID(00031${CALLERIDNUM:1});national dutch number |
| 23:02.43 | RoyK | three zeros leadingÞ |
| 23:02.50 | RoyK | ? |
| 23:02.53 | RoyK | seems a lot |
| 23:02.54 | justinu | bbz: i'm desperatly trying to figure out how to get the "Connected Party Information" feature on the polycoms going. |
| 23:02.58 | bbz | justinu: /sniff i was hoping to get away with not having to set the tftp/ftp info on each phone |
| 23:03.05 | RoyK | SetCallerID(00000000000000000000000000031${CALLERIDNUM}) |
| 23:03.12 | bbz | Connected Party? |
| 23:03.19 | IronHelix | ikarus- the example will dump the first 1 digit off calleridnum and add 00031 to the beginning |
| 23:03.34 | justinu | bbz: when you dial an extension, it should show you via the display who you're ultimately connected to |
| 23:03.39 | lesouvage | RoyK: one to let asterisk know what to do with it and two extra because a complete number starts with two zero's. |
| 23:03.51 | justinu | bbz: like if I call ext 2, and 2 is forwarded to 3, it should show I'm connected to 3 |
| 23:03.58 | bbz | ahhh interesting. |
| 23:04.13 | justinu | the docs all claim they support it, but I can't figure out why SIP RFC/Draft that's discussed in. |
| 23:04.13 | RoyK | lesouvage: my * boxes just sends whatever I dial further onto pstn..... |
| 23:04.24 | justinu | s/why/what |
| 23:04.31 | lesouvage | This way I can use the caller ID to call back |
| 23:04.46 | justinu | bbz: have you confirmed the phone isn't trying to contact your tftp server w/ a packet sniffer? |
| 23:05.41 | IronHelix | hahaha |
| 23:05.54 | justinu | royk: you've got a little on your nose still |
| 23:06.00 | lesouvage | I have a fwd acount registered, I dial an 8 to use it and a normal pstn line with 9 prefix and a interconnected voip line with a 0 prefix |
| 23:06.05 | RoyK | lol |
| 23:06.36 | ikarus | IronHelix: ah, the exact opposite of what I need, I need to match if it contains 0031 and if it does drop it, ah well, I should be able to figure it out |
| 23:06.45 | bbz | justinu: i was just reading how its a good idea to ensure that its not w/ Etherreal |
| 23:06.57 | bbz | justinu: cant say that ive used Etherreal enough though to understand what ill see |
| 23:07.04 | IronHelix | so if it starts with 0031 you want to dump the 0031, otherwise leave it alone? |
| 23:07.17 | bbz | juanjoc: |
| 23:07.18 | bbz | BR 2.6.1 is recommended in all cases for both FTP and TFTP. BR 2.5.0 does not mix well with TFTP, nor is it compatible with SIP 1.5 and later. |
| 23:07.26 | ikarus | IronHelix: yes, I need to dump it and replace it with 0 |
| 23:07.26 | bbz | justinu: ^^ that was meant to be for you |
| 23:07.32 | IronHelix | thats pretty easy |
| 23:07.34 | IronHelix | gimme a sec |
| 23:07.56 | justinu | bbz: uhh, if it doesn't mix well with TFTP, what does it work with? FTP? |
| 23:08.06 | IronHelix | also- are you running 1.2.x/head or stable/1.0.x? |
| 23:08.09 | RoyK | http://thedailywtf.com/forums/50765/ShowPost.aspx |
| 23:08.22 | bbz | justinu: id assume tftp includes ftp/http/etc .. |
| 23:08.27 | *** part/#asterisk myke420247 (n=myke@69.29.2.130) |
| 23:08.34 | lesouvage | ikarus: why, the result is the same. It's not a problem to dial the international number of Holland in Holland. |
| 23:08.51 | ikarus | lesouvage: it confuses the people using it |
| 23:09.26 | justinu | bbz: even 2.6.2 doesn't work with HTTP/HTTPS... only FTP/TFTP |
| 23:09.44 | ikarus | lesouvage: they rely alot on CID to weed out if a call is worth handling |
| 23:09.53 | bbz | justinu: which is the most optimal to use at this time? |
| 23:10.06 | *** join/#asterisk royth1 (n=royth1@201.230.161.178) |
| 23:10.07 | lesouvage | ikarus: I think this will do the trick exten => s,1/0031.,SetCallerID(0${CALLERIDNUM:4}) |
| 23:10.34 | justinu | bbz: if you can get 2.5 to work with plan old FTP, i'd use that to upgrade the phones to 3.1 |
| 23:10.39 | justinu | 3.1 and SIP 1.6.2 |
| 23:10.46 | ikarus | lesouvage: no, because not all calls have 0031 prepended |
| 23:10.47 | bbz | gotcha -- let me give this a go |
| 23:11.01 | bbz | justinu: thanks for giving me some help :) |
| 23:11.01 | ikarus | lesouvage: only those from a few telecom co's have that |
| 23:11.15 | justinu | bbz: sure... also remember that etherreal is your friend :) |
| 23:11.20 | bbz | hehe will do |
| 23:11.29 | lesouvage | ikarus: this line will only effect the cid's that actually start with 0031 because of the 1/0031. part |
| 23:11.35 | ikarus | lesouvage: ah |
| 23:11.40 | ikarus | lesouvage: missed that |
| 23:12.35 | ikarus | Right, I am gone, need to implement this stuff in the morning |
| 23:12.57 | lesouvage | ikarus: tot ziens |
| 23:13.14 | IronHelix | ikarus |
| 23:13.16 | IronHelix | before you go |
| 23:13.19 | IronHelix | heres a simple thing |
| 23:13.55 | IronHelix | match: if ${CALLERIDNUM:0:4} = 0031, then setcalleridnum(${CALLERIDNUM:4}) |
| 23:14.42 | lesouvage | (0${CALLERIDNUM:4}) |
| 23:15.07 | IronHelix | or that to add a 0 |
| 23:15.10 | IronHelix | i think he left :( |
| 23:15.25 | lesouvage | a thank you will do |
| 23:15.32 | bbz | thank you guys :) |
| 23:15.43 | IronHelix | :) |
| 23:15.46 | dippo | do you set your callerid for an IAX service with callerid= in aix.conf? |
| 23:16.02 | dippo | and if so does it take a while to propagate or something |
| 23:16.10 | dippo | any references to documentation on this would be appreciated |
| 23:16.16 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 23:16.20 | IronHelix | changes take effect as soon as you save the file and reload asterisk |
| 23:16.23 | IronHelix | 'reload' at CLI |
| 23:16.32 | dippo | hm |
| 23:16.38 | dippo | it's still showing up as "unknown" on my cellphone when i call |
| 23:16.42 | CunningPike | So, ManxPower: I traced my indications error |
| 23:16.42 | dippo | asterisk -> teliax -> my cellphone |
| 23:16.52 | BladeRunner05 | IronHelix: Hi iron |
| 23:17.04 | IronHelix | keep in mind when you use that in iax.conf it takes effect always |
| 23:17.33 | IronHelix | you might want to just put in your extensions.conf irght before it dials iax, something like SetCallerID("Caller ID Name" <CallerIDNUM>) |
| 23:17.35 | dippo | can you set it elsewhere on a per-handset basis or something? |
| 23:17.37 | IronHelix | replace those of course |
| 23:17.40 | dippo | interesting |
| 23:17.46 | IronHelix | sup blade |
| 23:17.51 | dippo | well i'll just set it hardcoded in iax.conf for now |
| 23:17.52 | IronHelix | dippo- if you do nothing at all |
| 23:17.58 | dippo | so i can test |
| 23:18.00 | IronHelix | it will use the caller id sent from the handset |
| 23:18.05 | dippo | it's still showing UNKNOWN for me right now |
| 23:18.10 | IronHelix | there is usually a place in the handset setup to put that |
| 23:18.20 | *** join/#asterisk many (i=many@krikkit.ukeer.de) |
| 23:18.38 | sahafeez | if i am using a TDM400 only for fax/modem should i turn off echo cannel in zapta.conf |
| 23:18.45 | Druken | dippo: are you restarting after changing iax.conf ? |
| 23:18.54 | sahafeez | s/cannel/cancel |
| 23:18.54 | dippo | yeah |
| 23:19.04 | Druken | reSTART or reLOAD ? |
| 23:19.12 | dippo | restart |
| 23:19.17 | IronHelix | dippo |
| 23:19.17 | Druken | k |
| 23:19.22 | IronHelix | try fromuser= |
| 23:19.24 | IronHelix | instead of callerid= |
| 23:19.51 | IronHelix | IIRC, fromuser sets for outgoing calls, callerid= sets for all incoming calls which is less useful |
| 23:19.56 | dippo | the calls I am making are being routed via IAX to teliax.com first, does that matter? |
| 23:20.04 | dippo | oh, hm |
| 23:20.14 | IronHelix | i mean put that in iax.conf for the teliax section |
| 23:20.51 | dippo | still shows UNKNOWN |
| 23:21.10 | nick125 | well, probably should be at my firewall.. |
| 23:21.27 | IronHelix | should be in the form fromuser="calleridname" <calleridnum> |
| 23:21.28 | IronHelix | it hink |
| 23:21.39 | dippo | yeah that's what i have |
| 23:21.51 | dippo | teliax has a configuration section for callerID |
| 23:21.58 | dippo | i originally set it there, but then realized you could set it in asterisk |
| 23:21.59 | nick125 | for some reason, with my asterlink account, i keep getting incoming TCP 5060 ports, however, 5060 is the source port, not hte destination |
| 23:22.01 | dippo | so I blanked it out again and saved |
| 23:22.06 | dippo | I wonder if that takes a while to take effect or something |
| 23:22.32 | IronHelix | nick- thats normal |
| 23:22.41 | nick125 | IronHelix: its being blocked though |
| 23:22.45 | nick125 | and i cant get calls in |
| 23:22.48 | IronHelix | with SIP, port 5060 is usually both the source and destination |
| 23:22.57 | nick125 | i havent tried to get it working outbound :/ |
| 23:23.51 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 23:24.35 | nick125 | IronHelix: well, im behind a nat, maybe that could be the issue? |
| 23:24.52 | IronHelix | it doesnt help |
| 23:25.21 | IronHelix | you need to in sip.conf set externip= and localnet=. then you need to unblock and forward port 5060 and a range of ports spec'd in rtp.conf |
| 23:25.43 | IronHelix | you dont need to do 10k-20k, a range of 250 ports will be more than you need |
| 23:25.44 | Druken | 9000 - 10,000 |
| 23:26.05 | IronHelix | then forward 5060/udp and whatever range you set/udp to your * box |
| 23:26.09 | *** join/#asterisk YoMama (n=tchen@pcp02689850pcs.roylok01.mi.comcast.net) |
| 23:26.19 | SkramX | Momma! |
| 23:26.22 | nick125 | i already tried that :( |
| 23:26.25 | YoMama | SkramX: yo :) |
| 23:26.44 | IronHelix | also- if you have mroe than one SIP device behind your NAT they can conflict with each other |
| 23:26.55 | IronHelix | if you have a vonage box or something, it may be using upnp or something to grab 5060 |
| 23:28.07 | Druken | then you can use 5061 :) |
| 23:28.12 | YoMama | vonage...*barf* |
| 23:28.30 | IronHelix | yes but you also have to set port=5061 in sip.conf :) |
| 23:28.33 | IronHelix | and yeah vonage blows |
| 23:28.46 | dippo | is fromuser only for SIP? |
| 23:28.50 | IronHelix | speaking of vonage- anybody have tips for LNPing a vonage number out of vonage? |
| 23:28.51 | nick125 | i wonder if my speakeasy voip is taking 5060.. |
| 23:28.58 | IronHelix | no i think it works for iax too |
| 23:29.09 | dippo | hm |
| 23:29.11 | dippo | i am not havin much luck here |
| 23:29.25 | redax | hi |
| 23:29.26 | IronHelix | are you reloading asterisk after you make changes? |
| 23:29.29 | IronHelix | hi red |
| 23:29.30 | dippo | yes |
| 23:29.30 | nick125 | yah |
| 23:29.34 | Druken | IronHelix: no you don't... |
| 23:29.40 | nick125 | this is odd.. |
| 23:29.54 | dippo | i haven't really made any changes though |
| 23:29.57 | redax | may I have a stupid question? |
| 23:30.07 | IronHelix | i dont what? port my number? i'm slowly figuring that out, |
| 23:30.08 | Druken | i have clients connected with 5060, 5061, 5062 and 1024 |
| 23:30.25 | IronHelix | red- sure |
| 23:30.27 | IronHelix | just remember |
| 23:30.28 | redax | what is the best way to "check" or rather "show |
| 23:30.38 | IronHelix | theres no such thing as a supid question, only stupid people :) |
| 23:30.48 | redax | 'the incoming calls in asterisk? |
| 23:30.51 | redax | using AGI? |
| 23:31.05 | Druken | redax: like show channels ? |
| 23:31.29 | redax | well. rather the events like the incoming calls. |
| 23:31.52 | IronHelix | you want your agi to be notified if thers an incoming call? |
| 23:31.57 | *** part/#asterisk Elven (i=elven@ptr-42.fw.swordcoast.net) |
| 23:32.20 | redax | IronHelix: yes. that should be more important |
| 23:32.31 | IronHelix | put the agi in the dialplan |
| 23:32.41 | IronHelix | exten => s,1,AGI(youragi,params) |
| 23:32.51 | IronHelix | if the agi isnt going to do anythign with it it doesnt have to |
| 23:33.03 | IronHelix | maybe just read some stuff off the channel |
| 23:33.14 | IronHelix | then have s,2,dial(people) |
| 23:33.29 | redax | this should be a simple app which should lookup the phonenum at the local telco provider's CD |
| 23:33.37 | redax | and show who's calling |
| 23:34.01 | IronHelix | well the incoming call should have a cidname attached to it |
| 23:34.03 | redax | somewhat a simple ncurses application |
| 23:34.06 | IronHelix | are you trying to supplement that? |
| 23:34.27 | redax | basicly yes. |
| 23:34.29 | *** part/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 23:34.47 | IronHelix | cool |
| 23:34.55 | IronHelix | also- this is definately not a stupid question |
| 23:35.02 | redax | just the destination should be not an ip phone but a simple ISDN phone |
| 23:35.30 | redax | that's why I'd like to have a "console" |
| 23:35.38 | IronHelix | ah, so call comes in over isdn with number and no name attached |
| 23:35.40 | IronHelix | you want name too |
| 23:35.41 | IronHelix | right? |
| 23:35.46 | redax | to display who the "fuck" is calling |
| 23:35.59 | redax | yep. |
| 23:36.16 | TheCops | Someone know something about RDNIS ? |
| 23:36.28 | IronHelix | quick q, your trunk (call comes in on) is isdn, correct? |
| 23:36.31 | Druken | why not just noop the CIDNAME ? |
| 23:36.32 | bbz | redax: there is an agi already developed to do that |
| 23:36.35 | IronHelix | and the call goes out to another isdn phone? |
| 23:36.39 | redax | just there's no way do display the caller information on the phone |
| 23:36.42 | nick125 | i can shoot this damn thing |
| 23:36.54 | redax | that;s why I'd like tohave an ncurses console |
| 23:36.55 | nick125 | it suddenly stops working for no reason :/ |
| 23:37.13 | IronHelix | because redax if * already has the cidname (ie if the call comes IN with caller id name attached) there are a few apps that will do that quite nicely |
| 23:37.25 | IronHelix | theres one i saw a while ago that gives you a SAMBA winpopup message with the name |
| 23:37.33 | redax | IronHelix: well. no. it comes from pstm. |
| 23:37.53 | IronHelix | im saying theres two problems here- getting the cidname to * and getting the cidname to the user |
| 23:38.00 | IronHelix | getting the cidname from * to the user is exceedingly easy |
| 23:38.07 | IronHelix | and there are numerous apps and scripts that do just this |
| 23:38.13 | redax | one zaphfc for the telco line.... just another for local S0 |
| 23:38.50 | redax | simple installation... |
| 23:39.16 | redax | 2 HFC cards... 1for telco, 1for local |
| 23:39.35 | IronHelix | so you have pstn -> isdn -> zaphfc -> * -> other zaphfc card -> phone |
| 23:39.36 | IronHelix | right? |
| 23:39.40 | redax | and I'd like to inform the local ppl, who's calling |
| 23:39.51 | infinity1 | ManxPower: doing this has created a callerid issue. |
| 23:39.59 | redax | yeah. right IronHelix |
| 23:40.52 | IronHelix | and you get no cidname from the pstn, ie pstn -> isdn -> zaphfc -> * doesnt have cidname. i ask because if it does, and your problem is the phone or whatever, the solution is easy |
| 23:41.02 | redax | but AGI is suitable for "watching" incoming calls? |
| 23:41.03 | *** part/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 23:41.40 | redax | IronHelix: the number is there.. I just wanna to append the caller info's |
| 23:41.47 | IronHelix | ok so you get number but not name |
| 23:41.50 | IronHelix | ok |
| 23:41.52 | IronHelix | yeah |
| 23:41.54 | IronHelix | agi is what you need |
| 23:41.59 | IronHelix | in extensions.conf |
| 23:42.15 | redax | cool.. |
| 23:42.17 | IronHelix | for the incoming line you'll have to do s,1,agi(youragi) |
| 23:42.19 | redax | but how? :) |
| 23:42.23 | IronHelix | s,2,dial(people) |
| 23:42.30 | IronHelix | so when a call comes in |
| 23:42.36 | IronHelix | you agi script will be run |
| 23:42.48 | redax | right. |
| 23:42.54 | IronHelix | it has to then look up the channels calleridnum |
| 23:43.03 | IronHelix | match that to the CO database |
| 23:43.06 | IronHelix | get the name |
| 23:43.19 | IronHelix | and set ${CALLERIDNAME} to be the name |
| 23:43.31 | _Sam-- | we do that |
| 23:43.40 | redax | hmmm... |
| 23:43.47 | _Sam-- | if the caller is in our database, we set callerid = order number |
| 23:43.58 | IronHelix | then it can exit status 0 (everything is peachy) and when s,2 runs (dial people) the caller id channel will have the right name |
| 23:44.01 | IronHelix | THEN (part 2) |
| 23:44.10 | IronHelix | you will need some way of getting the cidnum on a console |
| 23:44.39 | IronHelix | if you want to leave * console running, you can just do NoOp(${CALLERIDNAME} is calling!) |
| 23:44.51 | IronHelix | which will print out their name and 'is calling!' |
| 23:45.14 | _Sam-- | that doesnt look so pretty though on the console |
| 23:45.19 | _Sam-- | it would look something like this: |
| 23:45.19 | _Sam-- | <PROTECTED> |
| 23:45.32 | _Sam-- | except it would say is calling in there |
| 23:45.38 | *** join/#asterisk xbmodder (i=nobody@unaffiliated/xbmodder) |
| 23:45.41 | infinity1 | _Sam--: thats beautiful :) |
| 23:46.00 | IronHelix | BUT the agi script would also have the name in it |
| 23:47.25 | IronHelix | here http://www.voip-info.org/wiki-Asterisk+call+notification are a bunch of useful ways of reading off the caller id |
| 23:47.30 | *** join/#asterisk twisted[mobile] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 23:47.30 | *** mode/#asterisk [+o twisted[mobile]] by ChanServ |
| 23:47.47 | *** join/#asterisk xtr (i=01928375@S01060012174cc0e1.vf.shawcable.net) |
| 23:47.57 | nick125 | anyone here have an asterisk box working with a pfsense box/ |
| 23:48.23 | IronHelix | in the above link there is an app called YAACID which will sit in a widnows tray and pop up when it gets a call with CIDname/num |
| 23:48.45 | IronHelix | nick- pastebin your sip.conf (minus any passwords of course) |
| 23:49.21 | _Sam-- | what do people use to monitor asterisk...in that i mean monitor the service to make sure it is constantly up and running? i use a unix tool called 'mon' to monitor other services... |
| 23:49.36 | nick125 | can i just pastebin the top 25 lines or so (the rest are the asterlink lines)? |
| 23:49.38 | *** join/#asterisk vitaminmoo (n=vitaminm@wza.us) |
| 23:49.40 | IronHelix | sure |
| 23:49.41 | vitaminmoo | Hello. |
| 23:49.52 | IronHelix | sam- asterisk has a tool called safe_asterisk that runs asterisk with realtime priority whiel watching to make sure it doesnt screw up |
| 23:50.11 | nick125 | http://pastebin.com/432477 |
| 23:50.14 | _Sam-- | no i am looking for a tool to monitor the service from another box...and to alert me if the service is down |
| 23:50.17 | IronHelix | easy to use- in asterisk source dir after make install do make config and it sets up safe_asterisk as a service so you can do service asterisk stop and service asterisk start |
| 23:50.23 | IronHelix | ah |
| 23:50.30 | IronHelix | safe_asterisk can email you if it fails |
| 23:50.39 | _Sam-- | what if it doesnt know it failed? |
| 23:50.49 | vitaminmoo | Anyone know why the other party would hear full volume echo, but only when using a headset device on a snom 320? |
| 23:50.50 | _Sam-- | im looking for something that check the ports somehow |
| 23:50.59 | vitaminmoo | An echo of themself, that is |
| 23:51.16 | IronHelix | but any network tool that tries to connect on udp/5060 and sends something that looks like a sip message (and gets a reply back) should do |
| 23:51.18 | IronHelix | hmmm |
| 23:51.34 | hardwire | can I bitch about the make system? |
| 23:51.41 | hardwire | or is that kind of talk untollorable? |
| 23:52.34 | IronHelix | sam try this http://www.voip-info.org/wiki/view/Asterisk+monitoring |
| 23:52.46 | IronHelix | well you can try, but i dont think we have any real power over the make system |
| 23:52.53 | IronHelix | i've never had any real issues with it |
| 23:57.03 | *** join/#asterisk konfuzed (n=KonfuzeD@H129.C72.B0.tor.eicat.ca) |