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| 00:10.01 | deeperror | skyggen, 41? |
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| 00:23.53 | *** join/#asterisk nobesnickr (n=pmccaffr@ip72-201-157-30.ph.ph.cox.net) |
| 00:24.02 | nobesnickr | hello all |
| 00:25.16 | nobesnickr | i seem to be having a weird issue with meetme that i cant put my finger on. I have used it quite a bit in the past but this is my first attempt to use meetme with ztdummy. everything seems to work well for the first minute or so but after that everyone in the conference gets disconnected and the cli reports "Quitting Time..." |
| 00:26.21 | nobesnickr | has anyone had this issue or be able to help me diagnose? |
| 00:28.51 | nobesnickr | anyone? |
| 00:29.50 | JT | pretty patient there |
| 00:30.11 | nobesnickr | sorry, my mirc doesnt always connect |
| 00:30.17 | jaytee | sorry, but I just started messing with meetme and I have a Digium TE-212 PRI card in my server so I don't use ztdummy. |
| 00:30.28 | nobesnickr | happened the other day and i was talking to myself for 10 minutes |
| 00:30.49 | JT | i suggest not using mirc |
| 00:30.59 | nobesnickr | what do you use? |
| 00:31.11 | JT | all sensible irc clients have client/server lage detection and can detect if they're not connected to the server |
| 00:31.16 | JT | s/lage/lag/ |
| 00:31.26 | jaytee | the timing from the card's driver or the kernel timing with zaptel |
| 00:31.27 | JT | mirc is about the only one with no lag detection |
| 00:31.30 | JT | which is insane |
| 00:31.38 | JT | i use irssi myself |
| 00:31.42 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 00:31.43 | jaytee | I use xchat |
| 00:31.52 | nobesnickr | lol, well lucky me, ill give take a look at both of them |
| 00:32.14 | drmessano | mIRC can't detect if it's not connected to the server? |
| 00:33.21 | drmessano | Funny.. it's worked fine for me since they fixed it like 18 months ago |
| 00:33.26 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 00:33.41 | JT | drmessano: maybe he has an old version |
| 00:33.47 | nobesnickr | yes |
| 00:33.49 | nobesnickr | very old |
| 00:34.07 | drmessano | ZOMG |
| 00:34.11 | drmessano | 6.31 is not VERY old |
| 00:34.14 | drmessano | WTF |
| 00:34.16 | JT | that must be like the second coming of christ, mirc actually adding lag detection |
| 00:36.10 | drmessano | mIRC has always used pings and pongs to determine connectivity.. it was just horribly broken |
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| 00:43.20 | jaytee | anyone use SIP TAPI for click to dial in Outlook with Asterisk? |
| 00:48.15 | deeperror | jaytee, i've played with it some |
| 00:49.43 | deeperror | jaytee, it seems very limited but it does get our calls to connect |
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| 00:55.16 | [hC] | Ive used TAPI for click to dial in outlook |
| 00:55.24 | [hC] | i had to add a ton of shit to my dial plan to make the tapi dialer play nice |
| 00:55.30 | [hC] | and even still it can be a bit iffy |
| 00:55.58 | [hC] | there's a program out there called outcall (outcall.sf.net) that is much nicer to do the same sort of thing, but development on it seemed to stop a few months ago and it has a couple bugs connecting to exchange 2007, etc i think. |
| 00:55.59 | *** join/#asterisk obnauticus (n=obnautic@c-71-193-156-193.hsd1.wa.comcast.net) |
| 00:56.03 | jaytee | drmessano, I got the siptapi.tsp from Klaus Darilion of of Sourceforge and got it working so I can click to call someone from Outlook and it rings my Polycom and then calls their phone and everything works fine except after I hangup I get Incoming call: 408 Request timed out from xxx.xxx.xxx.xxx with the address being my sipX server. I know the problem something between * and sipX but if I restart sipX the messages keep coming until I restart *. |
| 00:57.05 | jaytee | as an FYI, I'm using sipX as a sipproxy to Exchange UM since Exchange UM speaks SIP tcp and Asterisk 1.4 only speaks SIP udp. |
| 00:58.03 | [hC] | why all the confusion? |
| 00:58.22 | [hC] | this is just simply to dial sip extensions on Exchange UM? |
| 00:58.24 | JT | isn't sipX a B2BUA and not a Proxy? |
| 00:58.35 | jaytee | damn thing works but the 408 messages never stop until Asterisk is restarted. It's like the call through sipX never gets completely torn down when the callers hangup. |
| 00:58.47 | jaytee | JT, nope, sipX is not a B2BUA |
| 00:58.56 | [hC] | jaytee: asterisk 1.6 supports SIP TCP |
| 00:59.50 | jaytee | hC, yeah and I've messed with the beta a little but I've had too many other things I've had to work on but eventually I want to move to 1.6 and eliminate sipX |
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| 01:00.24 | [hC] | jaytee: did you have to put a bunch of TAPI events in your dialplan to get the tapi tsp to work right? you know, telling you that the call is going through, connected, hung up, etc |
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| 01:01.28 | jaytee | hC, nope. I plugged in the TAPI module in Windows and configured it in Outlook to use sipX and it worked. |
| 01:01.29 | [hC] | jaytee: i used the one from star2star and without all those events (which are poorly documented to begin with) the tapi dialer wouldnt tell me the call states like Ringing, busy, answered, connected, dialing, hung up, etc... |
| 01:01.41 | [hC] | jaytee: have a URL for the TAPI tsp? |
| 01:01.50 | jaytee | hC, just a sec |
| 01:02.00 | [hC] | i think i have it |
| 01:02.02 | jaytee | http://sipx-wiki.calivia.com/index.php/Click-to-Dial_for_Outlook%2C_CardScan%2C_ACT%21_using_SIP_TAPI#Configuration_and_Use_of_SIP_TAPI_with_sipX |
| 01:02.11 | jaytee | that's from the sipX wiki |
| 01:02.12 | [hC] | ah ok |
| 01:02.38 | [hC] | I wonder if the tapi events are already in sipX |
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| 01:03.48 | jaytee | here's a pdf from the site of the guy who wrote the tsp http://www.enum.at/fileadmin/public/SIPTAPI-Tutorial-0.1.pdf |
| 01:05.38 | jaytee | I'm thinking it would probably work directly with Asterisk 1.6 instead of using sipX but I haven't tested it yet. |
| 01:07.55 | jaytee | it's strange though, because if I call Exchange UM I can use voice recognition to go to Contacts and tell Exchange to dial another user and it will connect us just fine with no errors going through sipX but when I use TAPI it works but then I get the continuous 408 errors after the call. I'm thinking of just running wireshark to capture the sip traffic to analyze it but it's not a high priority right now. |
| 01:08.09 | mmartinn | You don't need much in your dialplan for SIPTAPI |
| 01:08.33 | jaytee | I didn't add squat to my dialplan to make it work |
| 01:08.47 | mmartinn | You just need to ensure you can receive more than one call, as it calls yourself and transfers you to your destination. |
| 01:09.18 | mmartinn | I use it in a ~100 station outbound public opinion research call center |
| 01:09.52 | jaytee | yeah, it's using SIP REFER |
| 01:10.01 | mmartinn | yup. |
| 01:10.11 | jaytee | but I've used it and had another call come in fine on my second line key. |
| 01:10.53 | mmartinn | Yeah, I don't think REFER takes up its own line or anything. It is simply all part of one call. |
| 01:11.09 | jaytee | now if I can just figure out how the LCS 2005 stuff for Polycom phones translates to OCS 2007 I'll be a happy camper. |
| 01:11.21 | mmartinn | I'm no help there :( |
| 01:11.44 | jaytee | mmartinn, I think you're right but I can't remember the specifics from skimming through the RFC |
| 01:12.11 | jaytee | I usually only read the RFC's when I'm suffering from insomnia or terribly constipated. :-) |
| 01:12.11 | deeperror | mmartinn, what do you use for crm? |
| 01:12.12 | mmartinn | It's supervised... the original leg only hangs up if the REFER was successful. |
| 01:12.51 | mmartinn | deeperror: It's outbound survey research, so there's special sample management software for our studies. The software uses Windows TAPI, though, so that's how we do sip tapi. |
| 01:15.15 | *** join/#asterisk nobesnickr (n=pmccaffr@ip72-201-157-30.ph.ph.cox.net) |
| 01:15.26 | jaytee | this tsp I'm using was based on asttapi which used the * manager interface with telnet and the guy stripped all that out and replaced it with SIP. |
| 01:15.37 | skyggen | Has anyone had a problem when setting caller id on certain area codes in long distance calls they get fast busy and 41? |
| 01:16.09 | jaytee | what's a 41? |
| 01:17.09 | skyggen | Progress 41 or temporarly out of service status |
| 01:17.36 | lanning | 41? isn't it 401? |
| 01:17.51 | lmadsen | all SIP msg's should be 3 digits.... |
| 01:18.20 | nobesnickr | greetings everyone |
| 01:18.20 | lmadsen | You are not a winner. Better luck next time! |
| 01:18.36 | nobesnickr | anyone have any experience with meetme? I am having STRANGE issues |
| 01:19.11 | skyggen | no, 41 is what gets sent back from the pri |
| 01:19.22 | skyggen | has nothing to do with sip |
| 01:19.49 | lmadsen | ahh... I didn't see a mention of debugging PRI |
| 01:19.53 | lmadsen | doesn't do PRI |
| 01:20.09 | tzanger | lmadsen doesn't do chicks, either |
| 01:20.19 | nobesnickr | lol ouch |
| 01:20.19 | lmadsen | tzanger: chicks would disagree with that statement |
| 01:20.23 | lmadsen | I have references :) |
| 01:20.26 | tzanger | hahaha |
| 01:20.32 | tzanger | references? I shall have to check up on some of these |
| 01:20.45 | tzanger | reminds me of something I read today, and I want desperately to try |
| 01:20.47 | lmadsen | I'm so good I break up engagements |
| 01:20.49 | skyggen | its progress with cause code 41 |
| 01:20.49 | nobesnickr | lol, please post a list |
| 01:20.51 | nobesnickr | ill check also |
| 01:21.06 | tzanger | "You have the right to remain silent. Anything you say will be used against you" ... "ok, tits!" |
| 01:21.11 | tzanger | er held against you |
| 01:21.14 | tzanger | dammit, I fucked it up |
| 01:21.18 | tzanger | *sigh* |
| 01:21.24 | nobesnickr | HA HA HA HA |
| 01:21.54 | tzanger | s/used/held/ |
| 01:22.09 | skyggen | without the callerid set it completes long distance calls normally and completes certain area codes correctly as well. |
| 01:22.29 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552398.dsl.bell.ca) |
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| 01:22.36 | ManxPower | What is the EXACT Callerid setting you are using |
| 01:22.37 | nobesnickr | how many digits are you sending for caller id |
| 01:22.53 | nobesnickr | yes, i had a similar issue |
| 01:22.53 | dieno | does any one know how to limit call duration while originating |
| 01:23.05 | skyggen | exten => 1,n,SetCallerID(DNL <14148582299>) |
| 01:23.19 | nobesnickr | what version did u say u are using? |
| 01:23.21 | ManxPower | dieno: A Manager Origination or .call file? |
| 01:23.31 | tzanger | skyggen: don't put the 1 there |
| 01:23.32 | skyggen | 1.4.11 |
| 01:23.33 | ManxPower | skyggen: the leading 1 is NOT valud |
| 01:23.46 | nobesnickr | correct |
| 01:23.53 | ManxPower | Also, you cant send calleridname so don't bother to set it |
| 01:23.53 | dieno | nobesnickr manager Origination :) |
| 01:23.54 | nobesnickr | some voip companys need it for some reason |
| 01:24.16 | nobesnickr | no sorry, i was answering skyggen |
| 01:24.23 | skyggen | awesome that worked |
| 01:24.23 | nobesnickr | i think we both were |
| 01:24.24 | ManxPower | dieno: Can you Originate a call to a Local/ channel and then just handle it in the dialplan? |
| 01:24.46 | dieno | yes |
| 01:24.51 | JT | skyggen: SetCallerID, that has been deprecated for a very very long time |
| 01:24.58 | lmadsen | 1 is the country code |
| 01:24.59 | dieno | Local/1NXXNXXXX@from-internal |
| 01:25.12 | dieno | this is how i originate acll |
| 01:25.19 | dieno | call* |
| 01:25.23 | lmadsen | dieno: yes |
| 01:25.31 | lmadsen | uhhh.... |
| 01:25.38 | lmadsen | 1NXXNXXXX is a pattern match |
| 01:25.50 | lmadsen | you would have the _1NXXNXXXX in the dialplan, not in the originate request |
| 01:25.54 | lmadsen | you would have a real number there |
| 01:25.55 | dieno | hmm yes i know i need to limit it |
| 01:26.00 | dieno | while i am generating |
| 01:26.12 | dieno | like using T or AbsoluteTimeout |
| 01:26.17 | skyggen | the current caller id set is Set(CallerID(all)="Name" <number>)? |
| 01:26.22 | lmadsen | Local/15195915119@from-internal would be valid |
| 01:26.23 | ManxPower | dieno: or the options to Dial to let you do that. |
| 01:26.28 | dieno | or L(x) |
| 01:26.35 | ManxPower | skyggen: never ever use " in callerid |
| 01:26.40 | dieno | yes i can dial using this pattern |
| 01:26.46 | nobesnickr | skygen: just set the number |
| 01:26.49 | lmadsen | skyggen: Set(CALLERID(all)=...) |
| 01:26.52 | dieno | ManxPower but dont know how to limit call |
| 01:27.00 | lmadsen | dialplan functions need to be all uppercase |
| 01:27.01 | ManxPower | dieno: The L option of Dial |
| 01:27.14 | dieno | ok how do i put this |
| 01:27.20 | ManxPower | lmadsen: I missed that. Good catch |
| 01:27.33 | dieno | Manxpower or use it with Manager |
| 01:28.01 | ManxPower | exten => _1NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN},,L(...... |
| 01:28.24 | dieno | hmm it will liimit all calls |
| 01:28.51 | dieno | ManxPower i need to limit the call when i originate |
| 01:28.54 | nobesnickr | can anyone help me with a meetme issue i am having? the conf dies for no apparent reason after a minute |
| 01:29.11 | ManxPower | dieno: Do you mean you need to SPECIFY the limit" |
| 01:29.35 | ManxPower | lmadsen: Can you set channel variables during an originate? |
| 01:29.44 | lmadsen | yes, I think so |
| 01:29.54 | lmadsen | actually, yes, you can |
| 01:29.56 | lmadsen | I just did that actually :) |
| 01:30.52 | lmadsen | the parameter is 'Variable' |
| 01:32.50 | ManxPower | lmadsen: and here I was going to do something like Local/30^15045551212@happycontext and then match on exten _XX^1NXXNXXXXXX,1,Set(LIMIT=${CUT....... |
| 01:33.04 | lmadsen | heh :) |
| 01:33.14 | ManxPower | and a Goto(${EXTEN:3},1) |
| 01:33.38 | NovceGuru | vitelity doesn't have mwi :\ |
| 01:33.52 | ManxPower | lmadsen: I have a voicemail notification system that used to pass variables that way, before you could have __ variariables in .call files |
| 01:35.30 | lmadsen | ahhhh, yep, I've done something similar as well |
| 01:36.04 | *** part/#asterisk mmartinn (n=martins@n128-227-41-215.xlate.ufl.edu) |
| 01:37.27 | nobesnickr | has anyone set up meetme conferences? |
| 01:37.52 | jaytee | I have, works like a champ! |
| 01:38.15 | nobesnickr | besisdes the extensions.conf to transfer caller into to meetme room |
| 01:38.20 | nobesnickr | and the meetme.conf to set up the room |
| 01:38.28 | nobesnickr | is there anything else that needs to be set up? |
| 01:38.37 | nobesnickr | i seem to be missing something |
| 01:38.56 | ManxPower | nobesnickr: you need a zaptel timer like a zaptel compat card or ztdummy |
| 01:38.56 | jaytee | that should be all you need |
| 01:39.03 | nobesnickr | i have ztdummy |
| 01:39.12 | nobesnickr | meetme seems to work great |
| 01:39.15 | nobesnickr | for like a minute |
| 01:39.28 | nobesnickr | between about 45-75 seconds from the conf start |
| 01:39.31 | nobesnickr | it just dies |
| 01:39.44 | jaytee | no errors on the console? |
| 01:40.05 | nobesnickr | none that i can see |
| 01:40.36 | jaytee | well, most errors aren't invisible. they either show up or they don't. |
| 01:41.12 | nobesnickr | Quitting time... |
| 01:41.13 | nobesnickr | <PROTECTED> |
| 01:41.13 | nobesnickr | <PROTECTED> |
| 01:41.31 | nobesnickr | thats what im getting, no module warnings or errors |
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| 01:48.53 | nobesnickr | any ideas? |
| 02:11.10 | jaytee | guess no one had any ideas |
| 02:12.50 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 02:18.40 | jaytee | is listening to Yes - It can happen [33:00 (1%)] |
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| 02:19.38 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:20.18 | *** join/#asterisk mw-home (n=mw-home@adsl-99-161-87-221.dsl.bcvloh.sbcglobal.net) |
| 02:21.06 | mw-home | Hi -- my office just got a T1 line put in. Could I use asterisk to make a whole bunch of parallel voice calls that play an mp3? |
| 02:21.20 | lmadsen | yes |
| 02:22.06 | mw-home | lmadsen: how many parallel calls could I make? We're also considering using an IVR provider, but I would prefer using a T1. |
| 02:22.27 | lmadsen | with a T1 with PRI signalling, 23 in North America, 30 in Europe |
| 02:23.00 | mw-home | 23 parallel calls? |
| 02:23.09 | jaytee | 23 channels |
| 02:23.11 | mw-home | is this a good use for asterisk? |
| 02:23.16 | lmadsen | it's a use |
| 02:23.26 | JT | 23 B channels |
| 02:23.31 | *** join/#asterisk stybba (n=alpha@201.193.44.196) |
| 02:23.34 | mw-home | whats a channel vs a call? |
| 02:23.40 | JT | and it's an E1 in europe :) |
| 02:23.42 | jaytee | each call uses a channel |
| 02:23.56 | JT | and everywhere else in the world almost outside the us/canada and japan |
| 02:24.33 | JT | most calls use 2 channels to be technical, but usually 1 on the PRI side unless the call goes back out |
| 02:24.51 | drmessano | Japan is a J1, because, they're japan |
| 02:24.52 | mw-home | thanks! |
| 02:25.13 | mw-home | so, a T1 buys me about 20 parallel calls, but then the line is probably 100% used |
| 02:25.25 | Juggie | no, 23 or 30 |
| 02:25.35 | jaytee | are you in the US? |
| 02:25.35 | Juggie | theres no about, its a defined number. |
| 02:25.48 | Strom_M | 23 == ISDN PRI in north america |
| 02:25.54 | Strom_M | 24 == channelized T1 |
| 02:26.04 | mw-home | yeah, I'm in USA |
| 02:26.05 | Strom_M | 30 == ISDN PRI / channelized E1 (everywhere else) |
| 02:26.05 | stybba | hi all |
| 02:27.05 | *** join/#asterisk dieno (i=771e68ea@gateway/web/ajax/mibbit.com/x-49651a1d87257ddc) |
| 02:27.21 | dieno | ManPower still there |
| 02:27.25 | dieno | ManxPower still there |
| 02:27.33 | stybba | is posible transfer calls from CLI? or EXEC "COMMAND"? |
| 02:28.01 | mw-home | so, if i need to play a 60-second mp3 to 46 different people, it would take about 46 / 23 * 60 seconds? |
| 02:29.50 | Strom_M | plus call setup time |
| 02:30.23 | Strom_M | plus, you're better off converting the mp3 to telephone-quality audio first |
| 02:30.26 | mw-home | Strom_M: right. So, are other people using asterisk for this kind of thing? |
| 02:30.32 | dieno | can any tell me how to limit call while using orignate command of manager |
| 02:32.26 | Strom_M | mw-home: i'm sure |
| 02:32.34 | Strom_M | what kind of 60 second recording are you playing? |
| 02:32.42 | stybba | can any tell me how to transfer a call from one AGI script? i tink is posible with "EXEC" but i dont now how |
| 02:33.05 | Juggie | dieno, dial on a local channel which sends the call to a context which then does your DIAL() with the limit option |
| 02:33.14 | JT | Strom_M: New from Hormel: It's SPAM! |
| 02:33.24 | mw-home | Strom_M: it is a text-to-speech deal of a message from employers to employees. for example, "Today, the office is closed. stay home" |
| 02:33.57 | mw-home | JT: oh, yeah, absolutely :) No, really, it's all opt-in. |
| 02:34.13 | Juggie | mw-home, woudnt it be more effective to just have them call in to find out? :) |
| 02:34.32 | Juggie | if theres 50cm of snow on the ground and i know i'm not going to work i dont want to be woken up ;) |
| 02:34.33 | mw-home | Juggie: how can I charge them for that? |
| 02:34.47 | dieno | Juggie sorry about that buyt i am new b i am Using this patter Local/1NXXNXXXXX@from-internal to orignate call can you tell me where should i put this |
| 02:35.01 | Juggie | mw-home, your still providing a service the employer can update but i see your point |
| 02:35.30 | Juggie | in your Dial() .. which is in [from-internal] add the limit option |
| 02:35.42 | dieno | ok |
| 02:35.46 | Juggie | show application dial at the console |
| 02:35.49 | Juggie | to see details |
| 02:35.57 | dieno | if i want to change each time when i originate call then what should i do |
| 02:36.00 | Strom_M | Juggie: we've been telling him that all afternoon |
| 02:36.05 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 02:36.06 | Strom_M | he doesnt listen |
| 02:36.13 | dieno | :D hes right but not precisely |
| 02:36.18 | mw-home | Besides, some of the other outbound calls will be surveys, like "Can you come in today?" |
| 02:36.21 | Juggie | er, core show application dial |
| 02:37.08 | dieno | hmm storm_M was sayin this thing from last 10 hours but still didnt get it how to use it with PHP |
| 02:37.21 | dieno | i can orignate call using PHP but cant limit the time |
| 02:37.30 | Juggie | php has nothing to do with limiting the time |
| 02:37.38 | Juggie | you do that in your dialplan |
| 02:37.48 | dieno | i know but i want to do this using PHP :) |
| 02:38.17 | mw-home | hey, thanks for all the help. is the o'reilly book PDF a good place to start learning about asterisk? |
| 02:38.18 | Juggie | Dial(Zap/g1/somenumber, L(100000)) |
| 02:38.29 | Juggie | would be 1000000 ms |
| 02:38.29 | nick125 | wonders why he waited so long to get an IP Phone |
| 02:38.47 | mw-home | nick125: what do you use for your phone number? |
| 02:39.09 | Juggie | dieno, you do your action originate, your action originate dials on the local channel |
| 02:39.18 | nick125 | mw-home: I'm using Vitelity for my termination/origination |
| 02:39.30 | Juggie | then the dialplan kicks in, parses the number your trying to dial, and dials with any options you want |
| 02:39.50 | dieno | hmmm |
| 02:39.57 | mw-home | nick125: so, Vitelity maps IP stuff to a landline number? |
| 02:40.01 | Juggie | the reason you want to use the local channel is to push the call through the dialplan, else you just dial directally on zap, you have no control |
| 02:40.30 | dieno | oke |
| 02:40.36 | dieno | thats better |
| 02:40.39 | nick125 | mw-home: Other way around. |
| 02:40.50 | Juggie | Action: Originate |
| 02:40.50 | Juggie | Channel: Zap/g2/8135551212 |
| 02:40.54 | Juggie | so instead of doing like that |
| 02:40.58 | Juggie | which gives you no control |
| 02:41.12 | nick125 | mw-home: Vitelity sends calls from my DID to my Asterisk server over SIP |
| 02:41.43 | Juggie | so, you would do this instead. |
| 02:41.43 | Juggie | Action: Originate |
| 02:41.43 | Juggie | Channel: Local/8135551212@mycontext |
| 02:41.47 | dieno | hmm ok if i Add one more command liek Timeout: 30/r/n; will it going to drop wihtin 30s |
| 02:41.55 | Juggie | then in your extensions.conf you would have [mycontext] |
| 02:42.21 | mw-home | nick125: yeah, i don't understand that sentence. is the o'reilly PDF still a good up-to-date way to learn asterisk? |
| 02:42.47 | Juggie | and then pattern match the incomming number eg, exten => NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN},L(10000000)) |
| 02:43.04 | Juggie | and thats how you make action orignate use any dial variables you want |
| 02:43.11 | Juggie | very simple |
| 02:43.17 | Juggie | ~rtfm |
| 02:43.19 | jbot | i guess rtfm is Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://en.wikipedia.org/wiki/RTFM |
| 02:43.27 | Juggie | :) |
| 02:43.39 | nick125 | mw-home: I believe it covers Asterisk 1.2, which should get you through the basics with Asterisk 1.4 and 1.6 (there might be minor changes here and there) |
| 02:43.42 | dieno | hmm rite means that will quit call in 1XXXseconds as you mentikons |
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| 02:44.17 | dieno | haha :) |
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| 02:44.38 | Juggie | and if you want php to control the limit |
| 02:44.51 | Juggie | then you can set a variable in your action orignate and use it in the dialplan |
| 02:45.32 | dieno | now again need to be explain :) please can you tell me how do i put variable and call it using originate |
| 02:45.44 | mw-home | where is the best place to start learning asterisk? i'm not seeing a big DOCUMENTATION link on asterisk.org |
| 02:45.52 | russellb | ~book |
| 02:45.53 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
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| 02:46.19 | Juggie | Variable: myvar=value|myvar2=value2 |
| 02:46.35 | mw-home | ok, so, the book is good |
| 02:46.44 | Juggie | buy a copy and support the authors |
| 02:46.58 | JT | nick125: release 2 of the book is out, that covers 1.4 |
| 02:47.04 | mw-home | Juggie: ok |
| 02:47.05 | jaytee | mw-home, you can download the pdf version for free but the print version is handier to bookmark plus you can read it on the crapper. Plus it kills trees and that really pisses off the druids which is a good thing. |
| 02:47.08 | Strom_M | here's an esoteric one: is there any way to get the TDM800 FXS port to reverse polarity when the far side sends answer supervision? |
| 02:47.20 | dieno | means something like this in extension.conf eg, exten => NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN},L(value)) |
| 02:47.36 | Juggie | yes |
| 02:47.44 | Juggie | in a context which you point to when using the local channel |
| 02:47.55 | Juggie | go read up on local channels, action originate using local channels. |
| 02:48.03 | Juggie | theres nothing else i can say |
| 02:48.07 | Juggie | your going to have to figure it out |
| 02:48.20 | dieno | ok let me make a script let you show something |
| 02:48.21 | nick125 | JT: Oooh, awesome |
| 02:48.34 | jaytee | you can lead a horse to water............ |
| 02:48.49 | Maliuta | and then shoot it in the head? |
| 02:48.57 | jaytee | might as well! |
| 02:49.00 | dieno | Juggie i really appreciate your support |
| 02:49.01 | dieno | thnx |
| 02:49.02 | dieno | :) |
| 02:49.10 | Juggie | dieno, i dont need to see a script |
| 02:49.15 | Juggie | i know exactally what you are doing |
| 02:49.24 | dieno | ok :) |
| 02:49.30 | Juggie | given that i've written it all myself a million times |
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| 02:50.27 | mw-home | what's a DID? Is that a globally unique phone number? |
| 02:50.48 | Strom_M | mw-home: direct inward dial |
| 02:50.52 | Strom_M | perhaps you should read this |
| 02:50.55 | Strom_M | ~1012 |
| 02:50.56 | Strom_M | er |
| 02:50.57 | Strom_M | ~101 |
| 02:50.57 | jbot | i guess 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 02:51.38 | mw-home | Strom_M: thanks! |
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| 03:04.12 | mw-home | ok, i need to read all these docs. night all. thanks for the help |
| 03:10.06 | NovceGuru | So how unconvential would it be for a company to just rent a dedicated server and run a pbx on it? I guess no failover is a main issue |
| 03:12.55 | JT | dedicated servers are so yesterday, virtual servers are the go... except not always with asterisk... |
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| 03:15.47 | styelz | heh |
| 03:16.18 | styelz | i could never get over this school that ran win2k server and vmware for nix with services |
| 03:17.27 | JT | why is that? |
| 03:17.40 | styelz | it just seems odd |
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| 03:18.00 | styelz | they spent $1000's on a xeon to do it. |
| 03:18.01 | JT | how was it set up? |
| 03:18.41 | styelz | windows server was proxying the services were on nix vmware. not sure thats all i was told |
| 03:19.01 | JT | that sounds extremely vague |
| 03:19.08 | styelz | yea |
| 03:19.32 | JT | they could've been doing something smart, or extremely stupid, hard to tell from this info though |
| 03:19.36 | JT | what do you mean proxy? |
| 03:20.13 | styelz | from what i gathered, they were just using the firewall and proxying of win2k.. and then running services.. sql apache etc from vmware |
| 03:20.45 | styelz | i guess its normal these days |
| 03:21.08 | JT | most companies have way more servers than they need |
| 03:23.16 | drmessano | Sounds pretty smart to me |
| 03:23.30 | drmessano | Sounds like they were ahead of the curve on virtualization |
| 03:24.21 | LiNeTuX|Home | I 'proxy' an iSCSI box back to VMWare over CentOS for storage... because VMWare doesn't like the cheap iSCSI box, but Cent doesn't care. Works great. |
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| 03:25.08 | JT | LiNeTuX|Home: iSCSI enclosure > Centos running iscsi stuff > VMWare ESX? |
| 03:25.22 | LiNeTuX|Home | JT: Yeah, the Cent box shares it as NFS |
| 03:25.30 | LiNeTuX|Home | so ESX can use it |
| 03:25.55 | JT | cool |
| 03:26.14 | JT | is esx like its own OS these days? |
| 03:26.26 | LiNeTuX|Home | I can pull 100MB/sec over a single GigE port |
| 03:26.33 | drmessano | ESX is bare metal.. thats the idea of that product |
| 03:26.36 | LiNeTuX|Home | JT: if you ask VMWare, the answer is a resounding "yes" |
| 03:26.36 | JT | nice |
| 03:26.48 | JT | it's not an embedded linux or something? |
| 03:26.49 | LiNeTuX|Home | but it uses RHEL to boot |
| 03:26.51 | JT | ah |
| 03:27.11 | LiNeTuX|Home | they usurp all the os, then virtualize the RHEL into a 'console' OS |
| 03:27.20 | drmessano | I need to find a good open source bare metal package |
| 03:28.30 | drmessano | First priority is finding a cheap KVM that I can VNC or RDP into |
| 03:29.06 | JT | cheapest ip kvm i've found were around AUD$700 for a single port (can connect to standard KVM) |
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| 03:31.52 | luke-jr | drmessano: qemu |
| 03:32.20 | luke-jr | drmessano: or openvz |
| 03:32.50 | JT | qemu isn't very fast on its own |
| 03:33.07 | luke-jr | JT: he said KVMoIP, not fast |
| 03:33.41 | hsv-al | heh, i was reading farther in the book, and i think thats awesome how you can retrieve xml data |
| 03:33.59 | hsv-al | from weather services, and listen to thel ocal weather by calling your phone up, and having some text to speech engine say it back |
| 03:34.49 | drmessano | I thought QEMU requires a host OS? |
| 03:34.50 | luke-jr | whispers about VoiceXML 2.1 being capable of that as well. |
| 03:34.56 | luke-jr | drmessano: everything requires a host OS |
| 03:35.14 | drmessano | wow |
| 03:35.17 | luke-jr | drmessano: ESX is just a messy hacked up Linux with VMWare's crap on it |
| 03:35.26 | drmessano | I forgot this is IRC |
| 03:35.33 | LiNeTuX|Home | luke-jr: now now. |
| 03:35.45 | JT | luke-jr: virtualisation is totally different to a hardware kvm |
| 03:36.06 | drmessano | Hi, I am looking for a bare metal open source Virtualization solution that doesn't require an ENTIRE FUCKING OS TO BE INSTALLED ON WHICH TO RUN |
| 03:36.08 | drmessano | How is that? |
| 03:36.27 | LiNeTuX|Home | drmessano: very clear. |
| 03:36.42 | luke-jr | drmessano: no such thing exists |
| 03:36.51 | luke-jr | there is always a host needed to handle the real hardware |
| 03:37.13 | drmessano | so I need to install Windows Vista, or CentOS and then run some crap software on top of it? |
| 03:37.29 | drmessano | Hmmm |
| 03:37.32 | luke-jr | drmessano: or more ideally Debian |
| 03:37.49 | luke-jr | that can be made to use only about 100 MB or so |
| 03:38.05 | drmessano | luke-jr: I think you need to read up on virtualization solutions |
| 03:38.35 | luke-jr | drmessano: just because ESX hides the host OS from you doesn't change the fact that there is a host OS |
| 03:39.26 | LiNeTuX|Home | luke-jr: ever heard of ESX 3i? |
| 03:39.54 | luke-jr | nope |
| 03:39.58 | drmessano | I never said their wasn't a host OS.. but you're being overly anal fucking retentive about the fact that there is indeed an OS involved.. No shit, everything has an OS.. the difference between installing a 100MB base OS along with app X and the entire Fedora DVD with QEMU are you know, kinda different |
| 03:40.11 | hsv-al | drmessano, are you powered by guarana |
| 03:40.14 | hsv-al | to be typing 200wpm rebuttals |
| 03:40.15 | hsv-al | :) |
| 03:40.37 | luke-jr | drmessano: a minimal Debian wouldn't have X etc ⺠|
| 03:41.08 | LiNeTuX|Home | luke-jr: ESX 3i boot off of a flash device on the host - it's usually built in by the mfg... so it's not really the os. think PXE for ESX. |
| 03:41.49 | luke-jr | LiNeTuX|Home: I don't consider an OS to be any less an OS if it's netbooted |
| 03:42.01 | drmessano | luke-jr: Do you REALLY, REALLY think that enterprises rollout out bare metal virtualization are sitting there installing custom debian installs to make it work? Come on man |
| 03:42.51 | luke-jr | sigh |
| 03:43.30 | Juggie | ESXi is a 32mb footprint |
| 03:44.16 | LiNeTuX|Home | and ESX3i gives you full insight into the hosts' resources .. kind of like an HP Insight Manager or something |
| 03:45.29 | JT | i agree, an ESX style "bare metal" system is the way to go |
| 03:46.05 | Juggie | bingo |
| 03:46.10 | Juggie | if your gonna virtualize, do it right |
| 03:46.16 | drmessano | Exactly |
| 03:46.21 | LiNeTuX|Home | if you're crazy enough, you can also boot off a SAN... so there's no real configin' to do |
| 03:47.03 | drmessano | Throwing Ubuntu or Vista on a box and the running some VM app on top of it.. is.. well... pointless |
| 03:47.35 | hsv-al | drmessano, , , all of dynetics.com is run on virtualization |
| 03:47.48 | hsv-al | near billion dollar defense company |
| 03:48.01 | hsv-al | roughly 8 dell 2950's decked out, all running vmware products, 150+ virtual servers |
| 03:48.20 | Juggie | there is a point to it, if you do it right you can reduce the amount of hardware you need |
| 03:48.44 | Juggie | because usually systems to not get put to 100% load, and even some times when they are, there is another system which is not getting any load. |
| 03:48.51 | hsv-al | visited the noc, and its highly optimized |
| 03:48.56 | hsv-al | but they have skilled people running the show there |
| 03:49.10 | Juggie | i like how vmware can move a running vm from one server to another |
| 03:49.11 | Juggie | thats hot |
| 03:49.38 | LiNeTuX|Home | i also like how it'll tell me to move vm's around because resources are overused on one host and underused on another |
| 03:49.57 | Juggie | yeah |
| 03:50.03 | Juggie | its pretty hot |
| 03:50.09 | Juggie | they say it can do it without loosing a packet |
| 03:50.44 | LiNeTuX|Home | say, yes. do, no. :) |
| 03:50.50 | hsv-al | juggie |
| 03:51.09 | hsv-al | this guy teaches the masters in MIS at the university I goto, built their vmware infrastructure, view these powerpoints |
| 03:51.15 | hsv-al | http://www.utilitytechnology.org/conference/spring%202008%20presentations/DR%20Planning.ppt |
| 03:51.21 | JT | Juggie: almost all half decent virtualisation systems can move hosts live btw |
| 03:51.41 | LiNeTuX|Home | like MS Virtual Server? <cough> |
| 03:51.47 | Juggie | ya they can now but vmware was first |
| 03:51.52 | LiNeTuX|Home | oh wait, you said 'decent' |
| 03:52.06 | JT | i dunno |
| 03:52.18 | JT | i think qemu and xen have been able to do it for a long time now |
| 03:53.09 | hsv-al | juggie |
| 03:53.14 | hsv-al | vmotion is what makes drs possible |
| 03:55.01 | hsv-al | makes maintenance of an ESX server possible, again, without any downtime for the users of those virtual guests...... what is required is a shared SAN storage system between the ESX Servers and a VMotion license |
| 03:57.28 | LiNeTuX|Home | You can also use something like LeftHand's VSA on 2 hosts with local storage to create a virtual SAN and do the same thing. Pretty cool stuff. |
| 03:59.29 | hsv-al | linetux, their info earlier: http://h18013.www1.hp.com/products/servers/software/vmware-esx3i/index.html |
| 04:00.42 | LiNeTuX|Home | hsv-al: yeah, I don't run 3i (on 3.0.x now) but we've thought about it |
| 04:02.09 | LiNeTuX|Home | looks around and wonders when #Asterisk turned into #Virtualization |
| 04:04.36 | hsv-al | everyone is passed out from their taco bell and soda , digesting for the night, re-energinzing for another day of irc tomorrow :) |
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| 04:04.51 | Juggie | if anyone had taco bell they are not passed out |
| 04:04.57 | Juggie | more like praying to the porcelain god. |
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| 04:05.17 | nick125 | haha |
| 04:05.29 | Juggie | i would expect a FSB |
| 04:10.09 | delparnel | I'm trying to get this so that when I call a DID, it picks up... waits for the number I dial, hangs up on me, calls me back, and connects the call I dialed... Can anyone see what i'm doing wrong here? http://pastebin.org/43045 |
| 04:11.30 | Strom_M | delparnel: sorry, completely wrong |
| 04:11.34 | Strom_M | look at generating call files |
| 04:11.36 | Strom_M | Hanguo() |
| 04:11.37 | Strom_M | er |
| 04:11.44 | Strom_M | Hangup() destroys the channel |
| 04:12.57 | delparnel | h |
| 04:12.59 | delparnel | ah* |
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| 04:17.50 | [TK]D-Fender | delparnel: Next, Setvar no longer exists in 1.4. Then exten => s,5,Dial(${CALLERIDNUM}) <- this has no Tech in it (SIP/ZAP, etc), and you can't kill the call and continue on like that. For a call-back script you'll need to use a "call file" or "AMI Originate" Go read up on them on the WIKI |
| 04:17.52 | [TK]D-Fender | ~wikis |
| 04:17.52 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 04:26.13 | drmessano | There is an almost blinding flash of light as the spell book begins to |
| 04:26.13 | drmessano | glow! It slowly fades to a less painful level, but the spell book is now |
| 04:26.13 | drmessano | quite usable as a light source. |
| 04:26.23 | drmessano | yay |
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| 04:46.14 | jblack | Froboz, right? |
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| 05:19.21 | drmessano | Another Visitor |
| 05:19.42 | drmessano | Stay awhile |
| 05:19.46 | drmessano | Stay Forever! |
| 05:22.20 | FreedomBI | impossible |
| 05:23.41 | drmessano | Yep |
| 05:24.27 | FreedomBI | been a few months since I've played that game. |
| 05:25.38 | drmessano | It's been about 12 years for me |
| 05:25.50 | pputman | hello |
| 05:26.29 | drmessano | Hmm.. maybe longer |
| 05:27.25 | FreedomBI | I was testing a bunch of Commodores. |
| 05:27.50 | FreedomBI | Yeah, testing. Not just playing around with them. :) |
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| 05:41.04 | nick125 | Set(CALLERID(num)=5555551212) < That should work, right? |
| 05:46.10 | nick125 | (with the number being a real telephone number, of course) |
| 05:47.09 | [hC] | indeed that will work. |
| 05:47.21 | nick125 | Okay, I think my provider is not allowing me to set my caller ID to something other than one of my DIDs. |
| 05:47.36 | drmessano | Probably not |
| 05:47.38 | [hC] | A likely cause, yep. |
| 05:47.42 | nick125 | They used to. |
| 05:48.05 | drmessano | Shit happens? |
| 05:48.14 | nick125 | gets really annoyed |
| 05:49.01 | nick125 | Anyone else here with Vitelity that is having the same issue? |
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| 06:13.48 | pputman | Does anyone know what zaptel loadzone to use for callerid in malaysia perhaps? or any experience with malaysian callerid? |
| 06:16.41 | pputman | oops, didn't see the .my, nevermind |
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| 07:14.16 | johndbritton | anyone have a recommendation for a SIP provider in the USA? |
| 07:14.27 | johndbritton | I've been using broadvoice, but I'm not too happy with them |
| 07:15.12 | johndbritton | they don't need to be in the usa, just need a phone number in the usa |
| 07:16.07 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 07:17.28 | L|NUX | i have compiled chan_oh323 when i load this module it get following error |
| 07:17.28 | L|NUX | [Jun 11 03:17:23] WARNING[32562]: loader.c:376 load_dynamic_module: Module 'chan_oh323.so' did not register itself during load |
| 07:17.28 | L|NUX | [Jun 11 03:17:23] WARNING[32562]: loader.c:649 load_resource: Module 'chan_oh323.so' could not be loaded. |
| 07:17.33 | L|NUX | any one have any diea |
| 07:17.38 | L|NUX | *idea* |
| 07:22.15 | Maliuta | compile it again, properly this time |
| 07:22.50 | L|NUX | i did that brother |
| 07:23.03 | L|NUX | first pwlib then openh323 and then chan_oh323 |
| 07:23.09 | L|NUX | :( |
| 07:25.56 | *** join/#asterisk taner_fh (n=taner@gutenberg.dn.FH-Koeln.DE) |
| 07:26.01 | taner_fh | hi |
| 07:26.53 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
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| 07:28.10 | *** part/#asterisk johndbritton (n=john@210.82.36.181) |
| 07:28.44 | taner_fh | can anybody help me about ilbc codec ? how to install ... |
| 07:31.24 | kaldemar | have you compiled asterisk yourself? |
| 07:33.25 | taner_fh | kaldemar: i have only to install ilbc on a linux system, i dont have asterisk |
| 07:34.43 | kaldemar | so you come to an asterisk channel to ask for general linux help? |
| 07:35.36 | kaldemar | well, use your package manager to install ilbc if it exists in the distro repository. |
| 07:35.38 | taner_fh | not linux help, codec help.. ok :-) thx |
| 07:35.43 | Maliuta | I think that is installed by dd if=/dev/random of=/dev/sda |
| 07:35.52 | Maliuta | as root, then you reboot |
| 07:36.16 | kaldemar | that's just plain evil. |
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| 08:09.29 | Prog-JavApplet | i can make call between 2 sip client (X-Lite) but i hear nothing |
| 08:09.41 | Prog-JavApplet | can someone help please? |
| 08:09.57 | *** join/#asterisk bootc (n=bootc@78-33-44-114.no-dns-yet.enta.net) |
| 08:10.48 | pputman | Trying to solve a callerid problem in malaysia from a zaptel card, I'm getting an error with "No start bit found in fsk data". Any ideas? |
| 08:10.51 | bootc | hey folks, I'm having lots of trouble with my Snom m3 and putting it on hold |
| 08:11.00 | bootc | when you put it on hold it loses inbound audio |
| 08:11.17 | pputman | And the loadzone is already set to my |
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| 08:14.04 | bootc | if I dissect the stream using wireshark I can see the audio coming into the phone, but using rtpbreak I only see 2 streams (before hold music and hold music, not afterwards) |
| 08:14.17 | bootc | if I do the same with my 300 I can see 3 streams including the audio afterwards |
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| 08:21.26 | *** join/#asterisk Tond (n=s@ner-b116942.alshamil.net.ae) |
| 08:23.22 | Tond | Hi is there a way for me to force an IAX peer or friend's registry to expre every 30 seconds. so basically get the device or phone to keep registering itself every 30 seconds? |
| 08:23.47 | Prog-JavApplet | i can make call between 2 sip client (X-Lite) but i hear nothing |
| 08:23.52 | Prog-JavApplet | <PROTECTED> |
| 08:24.54 | kaldemar | Tond: try minregexpire and maxregexpire. you'll find them in the sample config. |
| 08:25.36 | Tond | tnx |
| 08:26.37 | *** join/#asterisk codestr0m (n=asura@76.74.174.194) |
| 08:30.49 | Prog-JavApplet | i can make call between 2 sip client (X-Lite) but i hear nothing |
| 08:30.50 | Prog-JavApplet | <PROTECTED> |
| 08:31.19 | codestr0m | I'm on a short trip to china and experiencing some interesting new /things/errors) "chan_sip.c:1972 retrans_pkt: Hanging up call xoorqmfpiekfdsx@chaos - no reply to our critical packet" |
| 08:38.42 | oej | Packet loss hits you in the back |
| 08:39.18 | codestr0m | oej: yeah. I enabled jitterbuffer and gsm codec and was able to make a 30 second echo test |
| 08:39.29 | codestr0m | , but that's about as good as it gets for me |
| 08:39.33 | codestr0m | I may try iax2 |
| 08:39.41 | codestr0m | , but not sure that jitterbuffer is better or will help |
| 08:41.53 | florz | Meanwhile anyone got any ideas as to how to (easily) limit a sip client to calling a certain PSTN prefix? |
| 08:42.48 | nick125 | codestr0m: Checked for NAT? |
| 08:43.34 | codestr0m | florz: in sip.conf context=out which is the context that will be used in your extensions.conf you can specify another one there. that's about the lowest threshold way I can recommend. .not sure your current setup |
| 08:45.02 | florz | codestr0m: well, yeah, that far it's obvious - but how construct the extensions in that context? |
| 08:45.03 | nick125 | er |
| 08:45.05 | nick125 | Prog-JavApplet: ^^ |
| 08:45.26 | nick125 | notes this as reason #46273 not to IRC past 3am. |
| 08:45.35 | bootc | Prog-JavApplet: are the two X-Lites on the same physical network? |
| 08:45.38 | bootc | same LAN |
| 08:47.18 | *** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 08:53.46 | codestr0m | florz: not to be rude, but there's a host of resources on how to construct a dialplan |
| 08:54.23 | codestr0m | nick125: yeah. checked for nat.. oej is 99.9% right. just packet loss.. |
| 08:55.01 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 08:55.09 | codestr0m | at this point I just don't know if I should try to switch over to iax2 and test that or if I'll get similar results.. the jitterbuffer is supposed to be better, but would it still make a difference |
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| 08:57.27 | florz | codestr0m: Yeah, sure, plenty of, but I haven't found any as to how to easily implement said requirement - as in, without lots of string manipulation. |
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| 09:08.47 | Prog-JavApplet | <PROTECTED> |
| 09:08.49 | Prog-JavApplet | i can make call between 2 sip client (X-Lite) but i hear nothing |
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| 09:14.15 | bootc | Prog-JavApplet: you haven't answered the two questions you have been asked, is there NAT involved and are the X-Lites on the same network |
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| 09:16.10 | awk | hmm, anyone here know what curses package needs to be installed |
| 09:16.11 | awk | *** Install ncurses to use the menu interface! *** |
| 09:16.29 | awk | can't use menuconfig on addons,etc.. I have ncurses and ncurses-devel installed, using centos |
| 09:21.20 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 09:23.55 | jblack | awk: that should be what you need |
| 09:24.13 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 09:24.30 | s0ck | what you guys use to backup your pbx's |
| 09:24.36 | s0ck | im thinking disaster recovery... |
| 09:26.49 | *** join/#asterisk Daejeo (n=chatzill@211.211.234.62) |
| 09:26.59 | awk | jblack hmm, strangly enoigh it isn't and I have perl-curses installed too |
| 09:27.10 | awk | s0ck our GUI does it all |
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| 09:29.19 | tzafrir_laptop | awk: ncurses-dev[el]? |
| 09:29.23 | awk | lol |
| 09:29.35 | awk | needed to do a make distclean coz i configured before i had ncurses packages installed |
| 09:30.05 | tzafrir_laptop | thinks curses-devil |
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| 09:40.52 | *** join/#asterisk kombi (n=kombi@port-213-160-14-18.static.km-it.de) |
| 09:41.24 | kombi | what is a good sip gateway that forwards all callerids? |
| 09:41.43 | *** part/#asterisk codestr0m (n=asura@76.74.174.194) |
| 09:42.59 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:44.35 | kombi | or in other words, which US based aix gateway would you recommend? |
| 09:49.02 | kombi | was it ever this quiet in here? ;) |
| 09:52.05 | awk | tzafrir_laptop: hmm, mybe you could tell me, I need to compile 'MYSQL_LOGUNIQUEID' into asterisk-addons, where would I specify this |
| 09:52.37 | tzafrir_laptop | Is this still needed in 1.4? |
| 09:52.50 | tzafrir_laptop | I really have no idea |
| 09:53.04 | awk | tzafrir_laptop yes, needed by 1 of my php scripts |
| 09:53.31 | awk | something like CFLAGS+=-DMYSQL_LOGUNIQUEID ? |
| 09:55.11 | kombi | awk: can't you use an auto_increment field also? |
| 09:57.15 | kombi | ..gives you a pretty unique id for each row |
| 09:57.46 | awk | kombi def something that needs to be changed |
| 09:58.41 | kombi | or try php's uniqid() perhaps? |
| 09:59.10 | kombi | salt it with last inserted id and your ready to go |
| 09:59.40 | awk | thanks |
| 10:04.18 | *** join/#asterisk knobo (n=bohmersp@148.122.202.168) |
| 10:06.24 | Daejeo | anyone awake in US? . i want to make a test call |
| 10:06.52 | kombi | the entire us is asleep at this hour..,) |
| 10:07.46 | styelz | feels like a slave all of a sudden |
| 10:11.12 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:13.46 | Daejeo | kombi :) |
| 10:16.57 | Daejeo | kombi: do u have a line line number? |
| 10:17.05 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:17.14 | Daejeo | kombi: do u have a land line number? |
| 10:17.16 | kombi | not in the us i don't.. |
| 10:20.00 | Daejeo | i know u r in germany now |
| 10:21.13 | Prog-JavApplet | <PROTECTED> |
| 10:21.20 | Daejeo | i can call ur landline in germany if u have one |
| 10:21.22 | Prog-JavApplet | can someone help |
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| 10:34.32 | Daejeo | i upgraded my cisco phone |
| 10:34.52 | Daejeo | i would like to make a test call |
| 10:35.03 | Daejeo | anyone awake? |
| 10:35.03 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.120) |
| 10:37.15 | *** join/#asterisk dieno (i=771e682e@gateway/web/ajax/mibbit.com/x-d5949b7469fc0324) |
| 10:37.55 | dieno | does any one know how to use absolutetimeout in freepbx |
| 10:40.42 | *** join/#asterisk ikevin (n=Informat@ANancy-256-1-145-123.w90-33.abo.wanadoo.fr) |
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| 10:40.57 | ikevin | hello |
| 10:41.21 | ikevin | i've some registration timeout problem while using asterisk |
| 10:41.54 | ikevin | i've things like that: Registration for 'xxxxxxxx@voip.kiwak.net' timed out, trying again (Attempt #32) in the asterisk console |
| 10:42.14 | ikevin | i've see on google that a frequent' |
| 10:42.24 | ikevin | does it because i'm behind a nat? |
| 10:47.38 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 10:48.01 | dieno | if its IAX |
| 10:48.10 | dieno | then download iaxping and check is your port open |
| 10:49.22 | Daejeo | ikevin: make sure your registration parameters are correct |
| 10:50.04 | ikevin | they are correct |
| 10:50.12 | ikevin | i can make call before the timeout |
| 10:50.54 | ikevin | and while i start asterisk i've a message who said me i'm registered |
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| 11:45.01 | *** join/#asterisk DonAlex (n=DonAlex@mirafiori.demon.co.uk) |
| 11:45.11 | DonAlex | afternoon peeps.. |
| 11:48.05 | DonAlex | I just compiled Asterisk from SVN seems to compile ok.. but when I run it asterisk uses up 97% of one CPU with this bogus behaviour. http://pastebin.com/m47062026 |
| 11:48.10 | DonAlex | Anyone else come across this.. |
| 11:48.35 | DonAlex | For the record it is is a Quad CPU proliant with no sound card |
| 11:49.06 | DonAlex | And playing back any voicemail effect are all staccato.. and I am wondering if the two are related? |
| 11:49.35 | DonAlex | other than that there are not errors really |
| 11:51.54 | DonAlex | I mean I have seen this is so many different context on google it is hard to say why it is happening in asterisk? |
| 11:51.56 | DonAlex | ioctl(0, SNDCTL_TMR_TIMEBASE or TCGETS, 0xbf8c8d98) = -1 ENOTTY (Inappropriate ioctl for device) |
| 11:51.57 | DonAlex | write(1, "\0", 1) = 1 |
| 11:52.06 | DonAlex | repeated over and over again.. |
| 11:52.24 | DonAlex | making the CPU run at 97-99% |
| 11:57.16 | DonAlex | For the record using Asterisk SVN-trunk-r121716 on Linux 2.6.22-3-vserver-686 #1 SMP |
| 12:04.08 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 12:10.55 | tzafrir_laptop | DonAlex, ls -l /proc/PID_OF_ASTERISK/fd/1 |
| 12:11.05 | tzafrir_laptop | DonAlex, ls -l /proc/PID_OF_ASTERISK/fd/0 |
| 12:14.30 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
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| 12:16.50 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:19.04 | dominic1 | hello Is it possible to customize meetme,that a user will be asked for his name and if he joines the conference the name is played back for all users in the conference? |
| 12:20.20 | _ShrikE | dominic1: core show application meetme |
| 12:20.44 | dominic1 | okay, then it's not possible |
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| 12:25.25 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:27.21 | remibemol | hello all, I need a little help. I have configure a sipline and I can call my computer with any phone. But I want to call it by a softphone localy. I want to have the same that when I call with a phone. do you know how do it ? |
| 12:28.53 | _ShrikE | dominic1: option i |
| 12:29.00 | *** join/#asterisk jack_sparo (n=eddy@dxb-b13928.alshamil.net.ae) |
| 12:29.55 | jack_sparo | what is the best codec that works with zap channels? |
| 12:30.16 | DonAlex | tzafrir_laptop: ls -l /proc/32155/fd/1 |
| 12:30.16 | DonAlex | lrwx------ 1 asterisk asterisk 64 2008-06-11 13:27 /proc/32155/fd/1 -> /dev/null |
| 12:30.40 | DonAlex | tzafrir_laptop: ls -l /proc/32155/fd/0 |
| 12:30.41 | DonAlex | lrwx------ 1 asterisk asterisk 64 2008-06-11 13:27 /proc/32155/fd/0 -> /dev/null |
| 12:30.58 | [TK]D-Fender | remibemol: ... |
| 12:31.00 | [TK]D-Fender | ~book |
| 12:31.01 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 12:31.18 | [TK]D-Fender | remibemol: All calls getting to * are just calls. they go where YOU send them in your dialplan. |
| 12:32.10 | DonAlex | tzafrir_laptop: Make any sense? |
| 12:32.18 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 12:32.57 | dominic1 | thanks a lot shrike |
| 12:33.13 | *** join/#asterisk RoyK (n=roy@ip-80-5-149-91.dialup.ice.no) |
| 12:33.47 | remibemol | [TK]D-Fender, I must call the number "*" ? |
| 12:34.08 | [TK]D-Fender | remibemol: No... "8" = ASTERISK <- |
| 12:34.12 | [TK]D-Fender | * |
| 12:34.13 | [TK]D-Fender | rather |
| 12:35.02 | jack_sparo | ~book |
| 12:35.02 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 12:35.33 | DonAlex | tzafrir_laptop: hmmm interestingly enough that is the same for 0,1 & 2 |
| 12:35.36 | hsv-al | Hello fellow internet addicts. Are we all looking forward to another long & glorious day of internetisseriousbusiness addiction? :) |
| 12:36.04 | remibemol | [TK]D-Fender, ok, thank, I will try |
| 12:36.24 | [TK]D-Fender | hsv-al: yes, which also goes by the name of runonsentencesarenotreallycoolokplzthxbibi :) |
| 12:36.37 | hsv-al | d-fender, my boy rick astley is coming over tonight |
| 12:36.58 | [TK]D-Fender | hsv-al: You know what I'm talkin' 'bout... |
| 12:37.01 | [TK]D-Fender | ~nowwhat |
| 12:37.02 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 12:37.11 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 12:37.21 | hsv-al | d-fender, rick astley coming over, were gonna chill and drink 40's |
| 12:38.46 | [TK]D-Fender | hsv-al: At least you know he'll never give them up... |
| 12:38.53 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 12:39.39 | hsv-al | he gave an interview recently, he actually looks normal |
| 12:39.51 | hsv-al | probably got facelift, botox and the works, doesnt even look like him in the video |
| 12:42.57 | [TK]D-Fender | hsv-al: Good... that was over 20 years ago. |
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| 12:52.50 | jack_sparo | what is the best codec that works with zap channels? |
| 12:53.55 | mwalling | is going to not abuse the action command, since using it for a *THIRDPERSON* self refrence is proper |
| 12:56.18 | [TK]D-Fender | jack_sparo: unload chan_brokenrecord.so |
| 12:56.25 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.140.193) |
| 12:56.41 | styelz | hides |
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| 13:07.40 | *** join/#asterisk ViKing78 (n=ViKing78@cerberus.franklinamerican.com) |
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| 13:11.46 | *** join/#asterisk ManxPower (n=manxpowe@181.sub-75-202-46.myvzw.com) |
| 13:12.12 | dominic1 | hello, hope somebody can help me . I changed something in app_meetme.c. Can I recompile this application only? |
| 13:12.20 | dominic1 | how does that work? |
| 13:12.34 | *** join/#asterisk jetlagmk2 (n=jetlag@70.17.48.245) |
| 13:12.46 | lmadsen | dominic1: if you already compiled asterisk, and just changed that file, running 'make' again will just recompile the changed modules |
| 13:12.46 | ManxPower | dominic1: just rebuild |
| 13:14.15 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
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| 13:27.49 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 13:28.07 | hackeron | hey, what 16 port switch would you recommend to use on a network with both PCs and Sip phones plugged in? (snom320) -- I tried dlink which gave me awful sound quality on the snom320, switched to netgear which gives great sound quality but phone freezes every few days. |
| 13:29.16 | [TK]D-Fender | hackeron: these all sound like SNOM problems. Never heard of a switch nuking sound, and Snom has a long history of flakey firmware and freezing. |
| 13:30.23 | hackeron | [TK]D-Fender: yes, probably but the strangest thing, if I connect my snom320 directly to my dlink switch I get crackly terrible sound, if I connect it to a netgear switch which I connect to the same dlink switch - perfect sound quality |
| 13:31.07 | ViKing78 | I had a problem with sound quality on a Netgear 8 port POE switch. I spent a lot of time hashing out why I had so much packet loss and was very upset when I found it was the switch. I vowed to never buy another Netgear switch again. |
| 13:31.30 | hackeron | ViKing78: what switches do you buy now? |
| 13:32.29 | ViKing78 | I've used Dell with good success for 48 port models. I haven't used any of their POE stuff yet but I would probably buy Linksys next for small installs with POE. |
| 13:33.15 | ManxPower | I use Cisco switches. Used they are very affordable. |
| 13:33.41 | [TK]D-Fender | I've never had a problem with D-Link PoE switches personally... |
| 13:33.45 | ManxPower | If I need PoE I can get a PoE injector (1-port or multi-port) |
| 13:33.55 | ViKing78 | ManxPower: Who do you use as a source? |
| 13:34.36 | ManxPower | ViKing78: I would have to look at my files. Initally we got them from eBay, now we go direct to the vendor (the vendor only puts a small percentage of their inventory on eBay). |
| 13:35.14 | ManxPower | Our main general computer vendor also works with a couple of company for used equipment. |
| 13:37.03 | ManxPower | If you use used Cisco switches you should go plenty of research to make sure that exact model supports the features you want. |
| 13:37.36 | ManxPower | There are several fastethernet cards for the modular switch models that don't support VLAN's, for example |
| 13:38.10 | ViKing78 | You do have to be careful about used Cisco gear because there are a lot of counterfeit gear out there. |
| 13:38.26 | ManxPower | ViKing78: most of that seems to be sold as new |
| 13:39.10 | ManxPower | All of the stuff we use was discontinued sales several years ago. |
| 13:39.49 | ManxPower | Catalyst 55xx switches, they were VERY popular and there are many, many used ones for sale |
| 13:40.00 | hackeron | thanks for the input :) -- I'll have a look at cisco and PoE D-link and Dell -- guess the cheap £50 non PoE D-Link isn't very good :) |
| 13:40.22 | ManxPower | I would not use netgear, dlink, or linksys in a corporate enviroment |
| 13:40.41 | hackeron | ManxPower: what about dell? |
| 13:41.49 | ViKing78 | ManxPower: Since your are buying used Cisco gear, do you get it re-certified? If not how do you get IOS updates for security and bug fixes? |
| 13:42.18 | *** join/#asterisk delparnel (n=delparne@KTNRON06-1168103470.sdsl.bell.ca) |
| 13:42.32 | ManxPower | hackeron: Dell has not screwed me yet, but I still don't recommend them. |
| 13:42.41 | ManxPower | Dell switches seem to be made by SMC |
| 13:43.27 | hackeron | hmm, so what's a cheap 16 port switch from cisco? -- I don't need any router features |
| 13:43.42 | ViKing78 | There's nothing checp from cisco |
| 13:43.49 | ViKing78 | It's called Linksys |
| 13:44.07 | ManxPower | ViKing78: You can't get the hardware we use re-certified as they are no longer supported. We have a CCO contract for our new cisco hardware (23 2621/2621XM routers) and we just got the latest firmware and install it on the switches. Yes, IOS can be an issue |
| 13:44.18 | hackeron | ViKing78: I mean like in the <$600 range |
| 13:44.56 | ManxPower | We spend about $2,000 for a 5505 w/96 ports |
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| 13:45.07 | hackeron | ManxPower: I need 16 ports |
| 13:45.25 | ManxPower | It's so inexpensive we just keep a spare chassis, supervisor engine, and cards |
| 13:45.34 | ManxPower | hackeron: do you need VLANs? |
| 13:45.39 | hackeron | ManxPower: no |
| 13:45.50 | ManxPower | Then almost any switch will work for you |
| 13:45.51 | hackeron | ManxPower: I need a switch, level 3 |
| 13:46.03 | ManxPower | no, LAYER 3 means "VLAN" |
| 13:46.04 | ViKing78 | level 3 is routing for IP |
| 13:46.10 | hackeron | sorry, level 2 :) |
| 13:46.15 | ManxPower | There you go. |
| 13:46.20 | ManxPower | LAYER, not LEVEL. |
| 13:46.36 | ManxPower | Specfically OSI LAYER |
| 13:46.38 | ViKing78 | VLANs are on layer 2 |
| 13:46.47 | defswork | [Jun 11 14:42:15] VERBOSE[28976] logger.c: -- Executing [s@macro-record-enable:4] AGI("Zap/6-1", "recordingcheck|20080611-144215|1213191735.37280") in new stack |
| 13:46.47 | defswork | [Jun 11 14:42:15] WARNING[28976] res_agi.c: Failed to fork(): Cannot allocate memory |
| 13:46.49 | defswork | :o |
| 13:46.54 | ManxPower | ViKing78: you are prolly correct |
| 13:47.20 | hackeron | ViKing78: oh, well, I don't need those :) |
| 13:47.24 | ManxPower | A "layer 3 switch" is also called a router. |
| 13:47.42 | ManxPower | hackeron: "layer 2" does not mean VLAN. 802.1q means VLAN |
| 13:47.45 | hackeron | ManxPower: yes, sorry, I meant a layer 2 switch with no vlans |
| 13:47.54 | [TK]D-Fender | A Layer 3 Switch is NOT a "router". So many articles written on this its almost funny. |
| 13:48.04 | ManxPower | [TK]D-Fender: might as well be. |
| 13:48.19 | ManxPower | if it routes packes in my book it is a router. |
| 13:48.32 | ManxPower | Much like a linux box routing packets is a router. |
| 13:48.58 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 13:49.06 | ManxPower | Heck, Cisco even calls their Layer 4 addon a "Routing Engine" |
| 13:49.08 | ViKing78 | It's not a router in the sense that it understands routing protocols but ManxPower is right in that it "routes" packets. |
| 13:49.34 | ManxPower | We don't use that because we already have routers |
| 13:49.51 | hackeron | right, so I need a cheap reliable layer2 802.3-2005 1GigE switch |
| 13:50.18 | ManxPower | In any case, when using eBay you need to make sure you buy from a good vendor and you need to make sure that what you are getting is what you need. |
| 13:50.33 | *** join/#asterisk coppice (n=chatzill@174.202.17.210.dyn.pacific.net.hk) |
| 13:51.41 | ManxPower | I spent at least a week researching it |
| 13:51.42 | LiNeTuX | hackeron: look at this one... great switch for the price: http://www.newegg.com/Product/Product.aspx?Item=N82E16833316053 |
| 13:52.11 | hackeron | LiNeTuX: yeah, fantastic price, is it reliable though? :) |
| 13:52.37 | LiNeTuX | hackeron: Extremely from what I've seen. Much higher quality than the Netgears or D-Links. |
| 13:52.52 | hackeron | LiNeTuX: thanks, looks good :) |
| 13:52.52 | LiNeTuX | And it's got a 'real' backplane to it |
| 13:53.05 | LiNeTuX | for that price range, that is |
| 13:53.05 | hackeron | LiNeTuX: now to find one in the UK |
| 13:53.21 | ViKing78 | HP makes pretty good switches. Even cicso nuts will tell you they are at least competition. |
| 13:53.54 | LiNeTuX | I've got some higher-end HP's as well - all Cisco ISO commands work in 'em, so you can take Cisco ACL's and dump 'em into the HP's. |
| 13:55.11 | hackeron | LiNeTuX: this look right? < http://www.lambda-tek.com/componentshop/index.pl?origin=gbase10.2&prodID=1012163 |
| 13:55.28 | *** join/#asterisk jackson__ (n=jackson@96.42.220.89) |
| 13:55.33 | *** join/#asterisk jack_sparo (n=eddy@pptp03.witopia.net) |
| 13:55.41 | hackeron | oh wait, that's 10/100 ports :( |
| 13:55.42 | jack_sparo | what is the best codec that works with zap channels? |
| 13:55.46 | LiNeTuX | hackeron: That's an even higher end model than the 8-port |
| 13:56.10 | hackeron | LiNeTuX: but 10 times slower :) -- and yes, I need a 16 port switch |
| 13:56.41 | LiNeTuX | http://www.lambda-tek.com/componentshop/index.pl?prodID=B60782 |
| 13:56.44 | ManxPower | jack_sparo: that makes no sense. in USA/Canada the PSTN uses ulaw, most of the rest of the world uses alaw. You don't set the codec for Zap. It uses whatever one it needs to use.; |
| 13:56.56 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 13:56.57 | ViKing78 | Here's a 24 port HP 10/100/1000 http://www.newegg.com/Product/Product.aspx?Item=N82E16833316077 |
| 13:57.35 | LiNeTuX | the 1400 series is the low-end of HP's managed line |
| 13:58.03 | ViKing78 | He already said he didn't even need vlan support. That's pretty low end |
| 13:58.11 | LiNeTuX | but they're still a far cry from most other brands stuff |
| 13:58.51 | LiNeTuX | ViKing78: I saw he wanted a L2 switch. |
| 13:59.38 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 14:00.37 | *** join/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 14:00.55 | jack_sparo | the idea ManxPower is that iam having hardtime calling |
| 14:01.05 | *** part/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 14:01.19 | jack_sparo | the bandwidth on box boxes are not so good, and when i allow gsm on the trunks to gave a better quality |
| 14:01.27 | jack_sparo | i cant use zap then |
| 14:01.33 | jack_sparo | so i need something in between |
| 14:01.44 | ManxPower | jack_sparo: Asterisk will automatically convert between codecs |
| 14:01.58 | ManxPower | ~trunk |
| 14:01.58 | jbot | i guess trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 14:01.59 | jack_sparo | let me explain the case ok? |
| 14:02.32 | [TK]D-Fender | ManxPower: box boxes? |
| 14:02.42 | [TK]D-Fender | ManxPower: new term quota hit! |
| 14:02.49 | ManxPower | jack_sparo: let me explain. One of the core features of Asterisk is the ability to convert between any supported to/from any supported codec. The exception to this is G729 which requires a license to convert, and g723.1 which asterisk cannot convert at all. |
| 14:03.15 | [TK]D-Fender | ManxPower: On G.723.1 yes.. and no on several counts. |
| 14:03.35 | ManxPower | [TK]D-Fender: Simple people require simple questions. |
| 14:03.48 | ManxPower | and simple answers too |
| 14:04.11 | jack_sparo | i have 3 pbxs, pbx main that has sip account, pbx1 and pbx2 that are connected via iax2 to main pbx , the reason we do this os that because sip is blocked. so what i do is that i send all traffic to main pbx and it makes everything, so if i want to call pbx2 from pbx1, the call goes to main pbx and then from there it routes, and it is working perfectly |
| 14:04.12 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 14:04.21 | ManxPower | [TK]D-Fender: jack_sparo lost karma when he /msg'd me privatly for help. |
| 14:04.25 | jack_sparo | ManxPower when u read it let me continue please |
| 14:04.31 | [TK]D-Fender | ManxPower: TC400 only = simple answer. "largely illegal 3rd party codecs available if you look hard and don't care" = slightly less simple, but not really. |
| 14:05.04 | ManxPower | jack_sparo: where are you located? |
| 14:05.18 | jack_sparo | gulf |
| 14:05.29 | ManxPower | jack_sparo: Just set codec=gsm in /etc/asterisk/zaptelconf |
| 14:05.34 | ManxPower | then don't worry about it. |
| 14:06.20 | ManxPower | jack_sparo: "gulf" is not a location. |
| 14:06.22 | jack_sparo | on all the boxes right? |
| 14:06.40 | [TK]D-Fender | ManxPower: lol |
| 14:06.45 | ManxPower | jack_sparo: only with boxes with zap cards. |
| 14:06.54 | jack_sparo | all does though |
| 14:07.01 | ManxPower | [TK]D-Fender: hush you, don't fight someone on things they are convinced of. |
| 14:07.15 | [TK]D-Fender | ManxPower++ |
| 14:07.16 | ManxPower | Where in the Gulf of Mexico are you located? |
| 14:07.35 | ManxPower | [TK]D-Fender: much like.... |
| 14:07.39 | ManxPower | ~siptrunk |
| 14:07.39 | jack_sparo | arabian gulf dude |
| 14:07.59 | ManxPower | jack_sparo: you could have just said "I don't want to tell you where I am located." |
| 14:08.06 | jack_sparo | [root@asterisk1 asterisk]# nano zapata |
| 14:08.06 | jack_sparo | zapata_additional.conf zapata-auto.conf.bak zapata.conf.template |
| 14:08.06 | jack_sparo | zapata-auto.conf zapata.conf |
| 14:08.13 | jack_sparo | sorry for the paste |
| 14:08.14 | [TK]D-Fender | LOL!!! |
| 14:08.16 | jack_sparo | i apolpgize |
| 14:08.29 | [TK]D-Fender | ManxPower: Fire up the oven! |
| 14:08.33 | ManxPower | I THINK all countries in the Arabian Gulf region use the same codec. |
| 14:08.44 | ManxPower | alaw. |
| 14:09.01 | ManxPower | jack_sparo: we cannot help you with GUI versions of Asterisk |
| 14:09.49 | jack_sparo | :( |
| 14:09.58 | ManxPower | It says this in the /topic of this channel. |
| 14:10.06 | ManxPower | You should go to the support forums/channels for your Asterisk GUI |
| 14:10.10 | ManxPower | ~trixbox |
| 14:10.11 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 14:10.17 | jack_sparo | i understand ManxPower, thanks anyway |
| 14:10.49 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 14:11.28 | ManxPower | thanks for wasting the time of all of us |
| 14:12.48 | ManxPower | [TK]D-Fender: do a /msg jbot_ siptrunk |
| 14:13.13 | *** part/#asterisk Aurs (n=Ove_Aurs@ap39pb.ip.ssc.net) |
| 14:13.30 | [TK]D-Fender | ManxPower: ...please undo that... |
| 14:13.44 | ManxPower | [TK]D-Fender: it is a harmless thing |
| 14:14.18 | ManxPower | It's easier than trying to convince the idiots that think "SIP trunk" is a valid thing. |
| 14:14.48 | ManxPower | I get virtually no support from the channel on this issue. |
| 14:16.31 | [TK]D-Fender | jack_sparo: I'll leave you with this thought : There is no "codec" to use with Zap. Whatever codec a VoIP call comes into * as will get translated as it gets bridged to a Zap channel. There is NOTHING to set in * for this. You are lokoking for, and arguing about something that DOES NOT EXIST. If something failes because you switch from one codec to another, then its because of your... |
| 14:16.32 | [TK]D-Fender | ...misconfiguration of your VoIP peers, or lack of licensing if a non-included codec is used. |
| 14:17.41 | [TK]D-Fender | looking* |
| 14:18.17 | *** part/#asterisk harryv (n=harry@67-207-147-205.slicehost.net) |
| 14:18.39 | tzanger | ManxPower: haha nice siptrunk entry |
| 14:18.51 | *** join/#asterisk JenniferAkemi (n=akemi@69-196-131-228.dsl.teksavvy.com) |
| 14:19.05 | ManxPower | tzanger: sometimes I think I'm the only one that thinks this is an important issue. |
| 14:19.25 | tzanger | ManxPower: that's probably because you get hit witht he brunt of request for it |
| 14:19.31 | ManxPower | or maybe are just getting lazy. All the time I hear people use a term that is not valid. |
| 14:19.32 | *** join/#asterisk coppice (n=chatzill@174.202.17.210.dyn.pacific.net.hk) |
| 14:19.50 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:19.57 | tzanger | kind of like lcd display or led diode |
| 14:20.21 | ManxPower | No! I just bought a high end LED Plasma! |
| 14:20.32 | tzanger | LCD plasma? heh |
| 14:21.00 | ManxPower | at least Asterisk ignores invalid config options |
| 14:21.08 | tzanger | ManxPower: that's a blessing and a disguise |
| 14:21.22 | tzanger | wtf |
| 14:21.23 | ManxPower | I like it, I use the feature for my scripts. |
| 14:21.23 | tzanger | I cannot type |
| 14:21.26 | tzanger | a blessing and a curse |
| 14:21.28 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 14:21.28 | [TK]D-Fender | It lets stupid people remain stupid. |
| 14:21.37 | [TK]D-Fender | it SHOULD whine. |
| 14:21.54 | ManxPower | It also lets people write 3rd party utilitys configured in the standard asterisk config files. |
| 14:22.29 | tzanger | ManxPower: true, but that could be worked around with comments |
| 14:22.41 | tzanger | ;=thirdparty_opt=ooga |
| 14:22.49 | tzanger | ;=thirdparty_flag=no |
| 14:22.51 | tzanger | type of thing |
| 14:22.55 | ManxPower | *nod* |
| 14:22.59 | [TK]D-Fender | I love people who can't read a parameter list on the "show applicatio" and "show function" info screens. My take is, if you can't read that "man page" version of *'s dialplan apps/functions, you don't deserve to be running * in the first place. |
| 14:24.00 | ManxPower | I agree |
| 14:24.27 | ManxPower | You elitist you!!! |
| 14:24.30 | ManxPower | 8-) |
| 14:24.55 | ManxPower | Heck, I think people should be required to take IQ tests before being allowed to reproduce. |
| 14:25.05 | lmadsen | I agree |
| 14:25.21 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-85-197.vif.net) |
| 14:25.31 | *** join/#asterisk igascream (n=igascrea@80.179.192.178.cable.012.net.il) |
| 14:25.37 | ManxPower | welfare moms with 5 kids .vs. college educated couple with 1 child. This is not the right balance. |
| 14:26.05 | ManxPower | ZPG! ZPG! |
| 14:26.28 | florz | Now anyone got any hints as to how to easily limit a sip client to calling a certain PSTN prefix? Or if someone can say that with some certainty there is no easy way, that would kindof help me too, I guess ;-) |
| 14:26.45 | ManxPower | florz: we already told you -- yesterday -- twice. |
| 14:26.51 | *** join/#asterisk brendan_ (n=brendan@72.15.28.7) |
| 14:26.59 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 14:27.14 | florz | ManxPower: eh? I only saw one, by you, which doesn't work!? |
| 14:27.26 | ManxPower | it works if you understand it. |
| 14:27.29 | igascream | HI need hellp with * DB. |
| 14:27.31 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 14:27.42 | ManxPower | Since we use that exact same design for 5 sites w/300 users |
| 14:27.53 | florz | ManxPower: Well, I tried it, it didn't work, so I guess I am rather sure that it doesn't work ;-) |
| 14:27.57 | brendan_ | Hello, i have an ip phone that has a phone directory, and dials numbers in the form 1112223333 ext 123 |
| 14:28.17 | anonymouz666 | ManxPower: 300 users using IP Phones? |
| 14:28.23 | ManxPower | florz: And I'm sure you did it wrong. |
| 14:28.29 | brendan_ | of course, i din't want the extension dialed with the number, is there a way to get asterisk to ignore the extension? |
| 14:28.31 | ManxPower | anonymouz666: correct |
| 14:28.39 | dominic1 | I made changes to app_meetme.c, now I want to contribute my changes. Will I need to create a patch - file? How does that work? |
| 14:29.05 | ManxPower | anonymouz666: ALL of the phones are Polycom |
| 14:29.41 | florz | ManxPower: And I am pretty sure you don't understand what the dial plan you proposed actually does. After all, what I see happen is exactly what I'd expect after reading the source. |
| 14:30.32 | [TK]D-Fender | dominic1: Go look at Mantis |
| 14:31.05 | ManxPower | You put the phone in a context by itself, then you only have extension patterns matching only what that phone is allowed to dial. This isn't rocket science, this is a BASIC, FUNDAMENTAL thing in Asterisk |
| 14:31.06 | *** join/#asterisk chewie__ (n=nick@nickswks.nasl.co.uk) |
| 14:31.08 | igascream | I can't receive some of Ast vars like TIMESTAMP and DATETIME and CALLERID from SIP phone. What's the reason for this? |
| 14:31.24 | ManxPower | igascream: that is correct. |
| 14:31.24 | [TK]D-Fender | brendan_: If its passed with the extension dialed, then parse it out in your dialplan yourself. |
| 14:31.26 | anonymouz666 | ManxPower: Polycom's are great. Did you ever use the SPA8000? |
| 14:31.31 | chewie__ | hi all, just wondering if there is any way i have over looked for running a script when a call comes in even if it isnt picked up |
| 14:31.45 | igascream | ManxPower: what do you mean? |
| 14:31.51 | dominic1 | what's the url? |
| 14:31.58 | ManxPower | anonymouz666: we used the 941 linksys, grandstream, ATAs, Uniden, etc. We standardized on Polycom |
| 14:32.05 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-85-197.vif.net) |
| 14:32.12 | ManxPower | igascream: SIP phones can't send variables. |
| 14:32.17 | igascream | ManxPower: why it's correct |
| 14:32.20 | florz | ManxPower: Well, yeah - but then, how would I go about writing a pattern matching such that only a particular PSTN prefix can be called? The one proposed by you obviously doesn't do that. |
| 14:32.42 | ManxPower | I can't tell you that without knowning what they are allowed to dial. |
| 14:32.42 | brendan_ | [TK]D-Fender, i would, but the users of the phones need to see the extension |
| 14:32.55 | anonymouz666 | ManxPower: I can understand the reason for that :) |
| 14:33.17 | brendan_ | [TK]D-Fender, do you mean that i can remove the extension in the dialplan in asterisk? |
| 14:33.34 | [TK]D-Fender | brendan_: doesn't matter what the SEE, its what YOU do with the #. |
| 14:33.37 | ManxPower | brendan_: what exactly is asterisk receiving as the dialed number? |
| 14:34.01 | igascream | ManxPower: for example I receive the fax and I want to name it like ${TIMESTAMP}_${CALLERID(num)} it doesn't set the TOMESTAMP params |
| 14:34.16 | ManxPower | igascream: that is NOT what you said. |
| 14:34.32 | [TK]D-Fender | igascream: go read channelvariables.txt |
| 14:34.33 | ManxPower | you said "I can't receive some of Ast vars like TIMESTAMP and DATETIME and CALLERID from SIP phone. What's the reason for this?". SIP phones don't send variables. |
| 14:34.50 | ManxPower | Now, if this is not what you are trying to do, then my answers are not valid. |
| 14:35.30 | ManxPower | igascream: it sounds like you have simply having trouble using CHANNEL VARIABLES in your DIALPLAN |
| 14:35.38 | igascream | ManxPower: the problem with SIP is just CALLERID |
| 14:35.45 | ManxPower | These variables are NOT SET by the phone. |
| 14:35.55 | igascream | ManxPower: rest of vars in any case |
| 14:36.11 | ManxPower | no variables are set by the phone. |
| 14:36.19 | florz | ManxPower: Well, take as an example the one from yesterday - ^0800.* (this being regex syntax) should be possible to be called on Zap/g1, nothing else. For any strings that are not valid phone numbers on the PSTN, some predictable action should be taken, that does not result in any outbound call. |
| 14:36.30 | ManxPower | Asterisk creates those variables, and it does not matter if it's a SIP phone, and MGCP gateway, or a ZAP port. |
| 14:37.09 | ManxPower | florz: now stop using regex and start using correct dialplan patterns. |
| 14:37.44 | ManxPower | Your "regex" says "dialed number starts with 0800 followed by 1 or more digits, no max length" |
| 14:37.45 | igascream | ManxPower: I understand in case of fax i don't use telephone at all so what could be the problem with TIMESTAMP for example |
| 14:38.08 | ManxPower | igascream: you are referencing it wrong. pastebin the CLI output of a failed call. |
| 14:38.25 | florz | ManxPower: Well, yeah, if I knew how to express that set of constraints in dialplan patterns - after all, I am telling you in regex syntax because I know how to express it that way, and I assume you will be able to understand that syntax, too |
| 14:38.45 | [TK]D-Fender | florz: Go read chapter 5 of THE BOOK again. |
| 14:38.46 | [TK]D-Fender | ~book |
| 14:38.48 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 14:38.50 | [TK]D-Fender | ^^^^^^^ |
| 14:38.51 | ManxPower | You need to understand the Asterisk pattern matching. |
| 14:39.10 | *** join/#asterisk coppice (n=chatzill@174.202.17.210.dyn.pacific.net.hk) |
| 14:39.21 | florz | ManxPower: and no, that regex includes 0800 itself, without any further digits - even though ^0800.+ would be OK, too, in this case, yeah |
| 14:39.55 | ManxPower | florz: Why are you not reading chapter 5 of the book rather than being here? |
| 14:40.16 | igascream | ManxPower: asterisk*CLI> console dial time |
| 14:40.16 | igascream | <PROTECTED> |
| 14:40.16 | igascream | <PROTECTED> |
| 14:40.16 | igascream | <PROTECTED> |
| 14:40.16 | igascream | <PROTECTED> |
| 14:40.18 | igascream | <PROTECTED> |
| 14:40.20 | igascream | <PROTECTED> |
| 14:40.38 | ManxPower | igascream: if you flood the channel again I will never ever ever help you again. PUT THIS STUFF ON PASTEBIN.CA |
| 14:40.56 | brendan_ | ManxPower, it recieves it as 1112223333%20ext%201234 |
| 14:40.59 | ManxPower | igascream: now put the dialplan that generates that on pastebin.ca |
| 14:41.06 | brendan_ | ManxPower, i can change the format of the number/extension |
| 14:41.37 | florz | ManxPower: and you think after reading that I could arrive at some single pattern that matches any valid PSTN number starting with 0800, and nothing else? |
| 14:41.39 | igascream | ManxPower: sorry I didn't understand you right |
| 14:41.43 | florz | +that |
| 14:41.59 | [TK]D-Fender | brendan_: YOU need to read chapter 5 of the book as well, and learn to use the CUT function <- |
| 14:42.35 | ManxPower | florz: if you can't arrive at the pattern after reading the book then you should not be using Asterisk. |
| 14:43.19 | ManxPower | igascream: do not expect any of these variables to work with a console dial |
| 14:43.49 | [TK]D-Fender | igascream: go read channelvariables.txt <------------ |
| 14:43.59 | ManxPower | florz: perhaps you would have better luck asking on the Asterisk-Users mailinglist. |
| 14:45.52 | ManxPower | brendan_: go read channelvariables.txt |
| 14:46.03 | brendan_ | thanks |
| 14:46.09 | *** join/#asterisk PepOSX (n=angeldav@200.90.126.130) |
| 14:46.16 | ManxPower | THEN go read Chapter 5 of the BOOK |
| 14:46.41 | florz | ManxPower: Well, I just re-read that pattern-matching part of that chapter - and I am sorry, but I don't see any way to construct such a pattern using the meta characters listed there ... |
| 14:46.42 | igascream | ManxPower: http://pastebin.ca/1044995 this is for fax |
| 14:47.08 | florz | ManxPower: Any hint how you would go about doing that?! |
| 14:48.03 | ManxPower | florz: You download the book, then you make sure you have a PDF reader, install it if you have to. Then double click on th book PDF. |
| 14:48.20 | florz | ManxPower: Well, I used wget+xpdf, if that's fine with you? =:-) |
| 14:48.40 | [TK]D-Fender | florz: exten => _XXXXX.,12,NoOp(Yay, 6 or more digts, now parse the stupid ext chars off! |
| 14:48.54 | ManxPower | The leading _ indicates this is a pattern match. X means [0-9], N means [2-9], Z means [1-9] and . means one or more chars. |
| 14:49.14 | florz | ManxPower: yeah, exactly, that's what's listed there |
| 14:49.15 | [TK]D-Fender | florz: And that will accept spaces, etc. |
| 14:49.21 | ManxPower | florz: I have never in my 6 years of using asterisk, ever met someone that simply could not understand Asterisk mattern matches. |
| 14:49.27 | s0ck | any ideas where i can get uk english voices for * ? |
| 14:49.34 | [TK]D-Fender | s0ck: ... |
| 14:49.36 | [TK]D-Fender | ~wikis |
| 14:49.36 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 14:49.38 | [TK]D-Fender | ^^^^^^^^^ |
| 14:49.52 | ManxPower | florz: Why don't you just TRY to make an Asterisk pattern match and I can tell you what is wrong with it. |
| 14:50.35 | florz | [TK]D-Fender: Erm, how ya mean? that's supposed to match only syntactically valid PSTN numbers? |
| 14:51.02 | ManxPower | florz: What is the syntax for PSTN numbers for your country? |
| 14:51.15 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 14:51.15 | florz | ManxPower: Because I don't see any way to construct a pattern matching my requirement that I couldn't tell you why it doesn't work? |
| 14:51.15 | ManxPower | no regexes, write it in english |
| 14:51.20 | [TK]D-Fender | florz: You are being passed #'s with CRAP IN THEM. So clearly you need to accepts all of that crap, parse it out, and decide for YOURSELF after manipulating it if it is valid or not. |
| 14:51.34 | [TK]D-Fender | florz: *'s pattern matching is not regex, GET OVER IT. |
| 14:51.41 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:52.04 | *** join/#asterisk kamh (n=q@host-81-190-236-85.wroclaw.mm.pl) |
| 14:52.06 | ManxPower | exten => _1800NXXNXXXXXX,1,,1,Whatever would only match USA 800 numbers |
| 14:52.24 | s0ck | ta |
| 14:53.08 | florz | [TK]D-Fender: yeah, of course, it's not a regex - after all, all I was asking for, was, whether there is a simple way of limiting a sip client to calling PSTN numbers via a Zap interface that start with a certaing sequence of digits - be it by using an appropriate pattern or any other way ... |
| 14:53.09 | ManxPower | florz: and no, the regex you gave earlier will match many many many things that are not valid PSTN numbers |
| 14:53.34 | florz | ManxPower: which is why I further restricted in the text after that regex, yeah |
| 14:53.54 | ManxPower | florz: Unfortunatly you don't know what IS a valid format for your country. You need to find that out before going any further. |
| 14:54.02 | florz | ManxPower: Well, yeah, IC - problem being that it should be able to cope with variable-length numbers |
| 14:54.08 | [TK]D-Fender | florz: You clearly want to accept crap and filter out the bad part. |
| 14:54.21 | ManxPower | florz: does your country have variable length 0800 numbers? |
| 14:54.28 | florz | ManxPower: yep |
| 14:54.40 | ManxPower | and that variance is? |
| 14:54.45 | [TK]D-Fender | florz: So make a pattern that accepts more that you might actually need and verify it in your dialplan. |
| 14:54.46 | ManxPower | this is like pulling teath |
| 14:54.56 | ManxPower | [TK]D-Fender: he understands nothing |
| 14:55.04 | [TK]D-Fender | ManxPower: No, that usualy takes a quick yank with a pair of pliers. |
| 14:55.29 | florz | [TK]D-Fender: well, yeah, that's pretty much the strategy I see - but I thought there might be a simple way, given that that's pretty much a standard scenario!? |
| 14:56.03 | ManxPower | florz: you need to hire a consultant |
| 14:56.12 | [TK]D-Fender | florz: Standard? BS. Since when do "normal" scenarios send TEXT along with an EXTENSION merged into a PHONE NUMBER to be dialed? No, this is just YOU. |
| 14:56.23 | ManxPower | [TK]D-Fender: that was not florz |
| 14:56.25 | igascream | ManxPower: I found my mistake sorry for disturbing |
| 14:56.28 | [TK]D-Fender | florz: And you seem to be completely lost on the concept of * pattern matching. |
| 14:56.56 | ManxPower | [TK]D-Fender: florz wants to limit calls for a specific device to only 0800 numbers |
| 14:57.02 | florz | [TK]D-Fender: Well, it's normal that you want to limit certain clients to certain numbers, right? |
| 14:57.21 | [TK]D-Fender | florz: Yes. Once again, Dialplan Patterns 101 |
| 14:57.42 | florz | [TK]D-Fender: Now, I was expecting that there might be a simpler way than tons of string manipulation and conditional jumping ... |
| 14:57.47 | *** join/#asterisk PepOSX (n=angeldav@200.90.126.130) |
| 14:57.50 | [TK]D-Fender | florz: If it doesn't fit a perfect nice & neat pattern match, then you have to widen your pattern a bit and parse it inside o fthe extension. |
| 14:57.53 | florz | [TK]D-Fender: For limiting to a given prefix, that is |
| 14:58.15 | [TK]D-Fender | florz: If the prefix is fixed then YES, it is easy. If you need MORE afterwards, thats another matter. |
| 14:58.35 | ManxPower | [TK]D-Fender: you are never going to succeed in helping this guy, even thought his question seems to brain dead simple my cat could figure out the answer. |
| 14:58.47 | Rico29 | hi |
| 14:58.51 | [TK]D-Fender | florz: exten => _0800.,1,NoOp(Yay, I start with 0800 and have 1 or more characters folloing and am OK with that!) |
| 14:59.01 | florz | [TK]D-Fender: Yes, that's all I am asking for - limiting a sip client to being able to call only numbers starting with a particular prefix on the PSTN line |
| 14:59.06 | Rico29 | i get trouble with Thomson ST2030S provisioning |
| 14:59.18 | ManxPower | [TK]D-Fender: that's EXACTLY what I gave him almost exactly 24 hours ago. |
| 14:59.20 | Rico29 | the phone doesn't doqnload the ST2030S_xxxxx.txt file |
| 14:59.29 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 14:59.39 | Rico29 | and I don't understand why |
| 14:59.46 | [TK]D-Fender | florz: So now ManxPower and I have both given you the same line, 24hrs apart. What don't you get? |
| 14:59.46 | florz | [TK]D-Fender: Well, that only accepts numbers starting with 0800[0-9], yeah - now, how do I do the (complete) limiting? |
| 14:59.49 | ManxPower | Rico29: this isn't really an Asterisk question |
| 14:59.59 | ManxPower | florz: NO IT DOES NOT.l; |
| 15:00.01 | [TK]D-Fender | florz: And what does "complete" limiting mean? |
| 15:00.20 | Rico29 | yes but I thought somebody used to have the same problem |
| 15:00.24 | ManxPower | That accepts dialed strings of 0800 plus any number of any number, alpha, or other char |
| 15:00.39 | [TK]D-Fender | florz: and no, the "." does NOT mean followed by DIGITS, it means followed by 1 or mor CHARACTERS of any kind (alpha, digit, etc) |
| 15:00.51 | ManxPower | so if you could dial 0800thecatjumpedoverthefox! that pattern would match that. |
| 15:00.54 | florz | [TK]D-Fender: Well, such that that sip client can't call anything but 0800 numbers on the PSTN side - after all, I have to dial out some way after accepting the input ... |
| 15:00.55 | [TK]D-Fender | ManxPower: Feel the echo in this room today... |
| 15:01.08 | florz | [TK]D-Fender: erm, yeah, sorry, mixed that up @. |
| 15:01.22 | [TK]D-Fender | florz: Fine, NOW whats wrong with the sample we just gave you? |
| 15:01.24 | *** join/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 15:01.28 | ManxPower | [TK]D-Fender: It's like a million brains suddenty stopped thinking. |
| 15:01.56 | florz | [TK]D-Fender: you simply can dial 0800&Zap/g1/whateveryoulike and thus call anything, without any limit!? |
| 15:02.01 | [TK]D-Fender | ManxPower: I heard each cell scream out in agony as it died of oxygen deprivation. |
| 15:02.01 | ManxPower | [TK]D-Fender: make tell you the rules for 0800 in his country. He seems to think this is secret information |
| 15:02.24 | [TK]D-Fender | florz: Holy. Crap. |
| 15:02.30 | ManxPower | florz: that is because you have a . in your patterm, that would be expected. |
| 15:02.46 | igascream | ManxPower: You said that I can't receive callerid for sip calls ,so how can I set call forwarding from SIP phones? |
| 15:02.51 | jaytee | two Putty windows open to SSH sessions on 2 * boxes, VMWare VM running XP open, VNC into Exchange 2007 open, Firefox and Xchat. This desktop is getting a bit cluttered. |
| 15:03.03 | florz | [TK]D-Fender: hmm? |
| 15:03.09 | jaytee | I need a 40" HDTV as a monitor |
| 15:03.13 | [TK]D-Fender | florz: exten => _0800.,1,Dial(Zap/g1/${EXTEN}) ; I can only call numbers starting with 0800 + 1 or more chars OUT MY STUPID LINE. |
| 15:03.26 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:03.55 | ManxPower | igascream: no, I said the phone can't set channel variables. It can't. ASTERISK sets those variables. |
| 15:04.09 | florz | [TK]D-Fender: Now, try sending it a SIP request where in the place where the SIP URI should be you put "sip:0800&Zap/g1/1234" |
| 15:04.19 | ManxPower | [TK]D-Fender: have you been reading the logs, thats what I gave florz yesterday |
| 15:04.44 | [TK]D-Fender | florz: You just don't friggen get it. * dialplan apps as part of a stupid URI to AAsterisk! |
| 15:04.56 | [TK]D-Fender | florz: You jsut pass the damn number! |
| 15:05.14 | [TK]D-Fender | florz: the NUMBER is your EXTENSION, and thats what gets matched in your dialplan. |
| 15:05.26 | ManxPower | [TK]D-Fender: he's trying to hack the dialplan from the phone |
| 15:05.41 | [TK]D-Fender | florz: You clearly understand aabsolutely nothing about how to use *. Go read the book. |
| 15:05.42 | ManxPower | He never told us that, but that's what I think. |
| 15:05.43 | florz | [TK]D-Fender: "* dialplan apps as part of a stupid URI to AAsterisk!"?! |
| 15:05.49 | ManxPower | In fact he's pretty much not telling us anything |
| 15:06.01 | florz | [TK]D-Fender: sorry, don't get that one ... |
| 15:06.12 | igascream | ManxPower: Thats quite clear ,but i still can't receive callerid for sip calls what could be a problem? |
| 15:06.33 | florz | [TK]D-Fender: erm, sorry, maybe I confused you - what I meant was: |
| 15:06.42 | [TK]D-Fender | florz: "sip:0800&Zap/g1/1234 <- What the hell are you doing MENTIONING Zap in a URI you are dialing? That is *'s dialplan syntax. All you dial is a stupid NUMBER |
| 15:06.44 | ManxPower | florz: if you send Asterisk "0800&Zap/g1/1234" then Asterisk will Dial(Zap/g1/0800&Zap/g1/1234) |
| 15:06.49 | florz | [TK]D-Fender: Now, try sending it a SIP request where in the place where the SIP request URI should be you put "sip:0800&Zap/g1/1234@asteriskhost" |
| 15:07.10 | igascream | ManxPower: I use ${CALLERID(num)} |
| 15:07.23 | ManxPower | igascream: Asterisk sets that, not the phone |
| 15:07.40 | [TK]D-Fender | florz: What the are you doing embedding the word "zap" in there in the first place? |
| 15:07.56 | florz | [TK]D-Fender: Well, I am mentioning that because I can dial that, obviously. And because that gets me a call to a destination you claim I wouldn't be able to reach. |
| 15:08.09 | ManxPower | The phone just provides it's callerid as a SIP header, then, unless overrided in sip.conf, asterisk will populate that variable with the callerid info |
| 15:09.26 | [TK]D-Fender | florz: The person who came up with "sip:0800&Zap/g1/1234@asteriskhost" as a URI is clearly on crack. What are you doing passing * app_dial tech formatting as part of your URI? |
| 15:09.43 | *** join/#asterisk southtel (n=southtel@24-240-24-20.dhcp.gwnt.ga.charter.com) |
| 15:09.49 | ManxPower | [TK]D-Fender: HE IS TRYING TO HACK THE DIALPLAN TO BYPASS THE SECURITY. |
| 15:09.53 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 15:10.00 | florz | [TK]D-Fender: because it demonstrates that what you say isn't the case? |
| 15:10.07 | ManxPower | He does not want to be able to hack the dialplan, |
| 15:10.39 | *** join/#asterisk Dovid (n=Dovid@bzq-79-178-105-250.red.bezeqint.net) |
| 15:10.45 | ManxPower | florz: until you know the 0800 dialing rules and tell them us for your country we will NEVER EVER be able to help you. Now either proivide the requested information and I will NOT ask again, or you are on your own. |
| 15:11.09 | [TK]D-Fender | florz: if you want to accept an arbitrary number of extra chars and want to prevent abue, the like I told you before PARSE THE EXTEN. You are being a complete retard about this. |
| 15:11.17 | igascream | ManxPower: So if I don't set callerid in sip.conf I wouldn't receive even a extension number ,so if I have 200 extensions I have to set callerId for each of them in sip.conf? |
| 15:11.21 | [TK]D-Fender | abuse* |
| 15:11.34 | Dovid | anyone here have an issue with a softphpone on a Dell Optiplex 1720L. I have tried multiple softphones, audio cards and i can hear fine but the person i am calling cant hear me at all. bandwidth isnt an issue cause other computers there work fine. |
| 15:11.48 | ManxPower | igascream: no, if you don't set it in sip.conf then asterisk will take the callerid provided by the phone and put it in the CALLERID variable. |
| 15:11.52 | Dovid | wondering if dell has something yummy on the machine that would be making it works so well ;0 |
| 15:12.15 | florz | ManxPower: well, 0800 is just an example - and in general, there are no rules in germany as to how long a number is - it's just historically grown, so fixed-length patterns are not a way that will work |
| 15:12.41 | ManxPower | florz: then we can't help you. |
| 15:13.05 | igascream | ManxPower: Ok so thats what i am trying to say it doesn't do it |
| 15:13.06 | ManxPower | Normal people use multiple patterns |
| 15:13.25 | [TK]D-Fender | florz: use "_0800.", and verify for YOURSELF that the extra chars are legit. You'll have to do this in your DIALPLAN, not in the pattern itself. as I said countless times before, * apttern matching is NOT REGEX. So get used to GotoIf's, CUT's, etc. |
| 15:13.27 | lmadsen | can't you just use . then? |
| 15:13.33 | ManxPower | igascream: then your phone is not sending valid callerid |
| 15:13.36 | lmadsen | didn't have time to read the scrollback |
| 15:13.51 | [TK]D-Fender | lmadsen: Stand back, you really don't want in on this... |
| 15:13.52 | florz | [TK]D-Fender: Well, yeah, as I said, that was clear to me, too - and still, what I was asking for, was a simple way to achieve that limiting - given that it's pretty much a standard scenario that you want to do such limiting. |
| 15:13.52 | tzanger | lmadsen: you just can't read, period :-) |
| 15:13.58 | igascream | ManxPower: so the problem is in phone you say |
| 15:14.00 | lmadsen | stands back |
| 15:14.08 | lmadsen | tzanger: what?! |
| 15:14.15 | tzanger | see, you couldn't read what I said |
| 15:14.21 | ManxPower | florz: as you did not provide me the information I requested I cannot help you further. Best of luck trying to change a duck into a squid. |
| 15:14.24 | lmadsen | I don't understand the characters you're typing |
| 15:14.45 | florz | ManxPower: you want to say that asterisk is not a pbx? =:-) |
| 15:14.48 | [TK]D-Fender | florz: OR, you can make a PILE of patterns. _0800X , _0800XX , _0800XXX , _0800XXXX , etc |
| 15:15.07 | [TK]D-Fender | florz: There... no more GotoIf's, no more alpha checks. Only #'s allowed. |
| 15:15.28 | florz | [TK]D-Fender: well, that would be conceptually simple, I guess - but easy maintainability looks different, I think =:-) |
| 15:15.43 | ManxPower | florz: no, you are confused about so many basic things in Asterisk that you are simply beyond my help. |
| 15:15.58 | s0ck | Use language= in a .conf file, or use the SetLanguage() application in extensions.conf |
| 15:16.01 | s0ck | which conf...? |
| 15:16.05 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.134) |
| 15:16.15 | s0ck | from http://www.voip-info.org/wiki/view/Asterisk+sound+files+international btw |
| 15:16.23 | ManxPower | s0ck: which ever conf file the device is using |
| 15:16.38 | [TK]D-Fender | florz: We are now beating a dead horse and I'm about done. If you don't like the way *'s pattern matching works, go change it yourself. You've got the code like the rest of us. |
| 15:16.47 | ManxPower | if it's a sip device, then sip.conf, if it's a zap port then zapata.conf, etc |
| 15:17.00 | florz | ManxPower: I doubt that I am much confused - after all, you still haven't shown me any way to do that limiting easily - without tons of redundancy and without much of manual sanitation/filtering |
| 15:17.05 | s0ck | im trying to change the us english voice prompts |
| 15:17.08 | s0ck | the wiki says the above ^ |
| 15:17.14 | florz | And it well may be that there is nont |
| 15:17.16 | florz | none |
| 15:17.47 | ManxPower | [TK]D-Fender: I wish had a "one or more numbers" and "zero or more numbers" pattern chars |
| 15:17.49 | florz | but that would be good to know, too, since then I don't feel like doing something quite as stupid |
| 15:17.50 | [TK]D-Fender | florz: There is no easy way and what we've shown you is what you've got as options. You are looking for something that doesn't exist. there is no easier way and you are wasting time looking for something that doesn't exist. |
| 15:17.56 | s0ck | my /var/lib/asterisk/sounds is full of .wav and .gsm |
| 15:17.58 | [TK]D-Fender | florz: TFB <----- |
| 15:18.12 | s0ck | the new audio files i have are .g711u |
| 15:18.22 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 15:18.27 | s0ck | can i just drag and drop the fuckers in? :) |
| 15:18.40 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 15:18.45 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:18.45 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:18.51 | [TK]D-Fender | s0ck: put them in the proper folders, and yes. |
| 15:19.12 | florz | [TK]D-Fender: Well, thanks, that's pretty much what I was looking for. Even though I certainly would prefer an actual solution if there was one ;-) |
| 15:19.47 | ManxPower | [TK]D-Fender: if he had told me the number length of 0800 numbers we could use ${EXTEN:0:maxnumberof0800digits}, that would at least make the hack not work. |
| 15:19.49 | s0ck | ok thanks fender. if i end up with vm-friends.g711u and vm-friends.wav, which one is it gonna use? |
| 15:19.54 | s0ck | do i have to set it somewhere |
| 15:19.55 | ManxPower | But since he refused to tell us that information....... |
| 15:20.01 | s0ck | or just make sure only the former exists |
| 15:20.31 | [TK]D-Fender | florz: Thanks for the collosal waste of time. |
| 15:20.32 | ManxPower | too bad florz was so secretive and refuse to tell us his country dialing rules. |
| 15:20.50 | ManxPower | [TK]D-Fender: I told you he was beyond help. |
| 15:21.03 | florz | ManxPower: _there_ _are_ _no_ _dialing_ _rules_, really - or at least close to it |
| 15:21.11 | *** part/#asterisk Oy90 (n=ivan@213.187.111.94) |
| 15:21.15 | ManxPower | florz: ALL COUNTRIES HAVE DIALING RULES> |
| 15:21.30 | ManxPower | For example 0800 numbers are never more than 45 digits is a dailing rule |
| 15:21.39 | florz | ManxPower: all you get to know is lower and upper limits to the length per prefix (which in turn are variable length) |
| 15:21.44 | [TK]D-Fender | florz: And then you want on some "vulnerability hunt" without mentioning a new question about it and no-one can follow what the hell you're going on about. |
| 15:22.13 | ManxPower | florz: Do you really think that me knowing the min and max is NOT useful information??? Do you really think that????? |
| 15:22.20 | florz | [TK]D-Fender: Eh? I think I asked a pretty clear question, no!? |
| 15:22.27 | ManxPower | It's the most important information I have ever asked you for. |
| 15:22.49 | ManxPower | florz: what country are you located in? |
| 15:22.55 | florz | ManxPower: .de |
| 15:23.04 | florz | ManxPower: As I said earlier, BTW =:-) |
| 15:24.56 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 15:26.15 | s0ck | that didn't work ;/ |
| 15:26.32 | *** join/#asterisk PepOSX (n=angeldav@200.90.126.130) |
| 15:27.07 | southtel | Has anyone out there run polycom sip behind a netopia router (with * on the other side, but everyone is on public IPs). |
| 15:27.20 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:27.35 | puzzled | hi |
| 15:28.08 | [TK]D-Fender | s0ck: Remeber, PROPER folders, and make sure you specified the language properly on your device or int he dialplan. |
| 15:28.10 | puzzled | anyone know what "Media type not available" means in a 488 Not Acceptable Here? |
| 15:28.23 | puzzled | is that a codec issue or something else? |
| 15:28.34 | [TK]D-Fender | puzzled: Codec mismatch <- |
| 15:29.11 | puzzled | [TK]D-Fender: thanks. weird cause the N95 supports the amr codec. Time to reboot the phone :) |
| 15:29.13 | southtel | puzzled: you're trying to use a codec that * doesn't know how to use. |
| 15:29.47 | puzzled | southtel: thanks. it's a Nokia N95 that sends it back to Asterisk |
| 15:30.14 | [TK]D-Fender | puzzled: Go look at the complete SIP debug of the call attempt. |
| 15:30.19 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 15:30.38 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.149) |
| 15:31.25 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 15:31.25 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:38.01 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.140.193) |
| 15:39.41 | *** join/#asterisk loompek (n=NoName@noname.rula.net) |
| 15:39.43 | loompek | hello |
| 15:40.18 | loompek | what softphone do you suggest... i'd like both audio (alaw) and video (h263) support... x-lite freezes all the time :@ |
| 15:42.05 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:45.32 | chewie__ | Hi all, im using an auto call file to place an outbound call and connect it to a local channel when the call is answered, my problem is that the dialplan isnt processed until this call is answered, my question is how can i run a script with knowledge of the asterisk uniqueid for each call even if it isnt answered? |
| 15:47.55 | Juggie | use a local channel to place the outbound call |
| 15:48.23 | chewie__ | now thats not a bad idea |
| 15:48.25 | chewie__ | d'oh |
| 15:48.27 | chewie__ | :-) |
| 15:48.29 | Juggie | instead of using Zap/g1/7775551234 in your .call file, use Local/7775551234@mycontext |
| 15:49.01 | Juggie | then stick at pattern match in mycontext for the number, and do a real Dial(Zap/g1/${EXTEN}...) in there |
| 15:49.04 | Juggie | and you'll be good to go |
| 15:49.14 | Juggie | at=a |
| 15:49.43 | chewie__ | i'll give that a go cheers |
| 15:49.46 | chewie__ | genious |
| 15:49.48 | chewie__ | :p |
| 15:52.03 | s0ck | so |
| 15:52.08 | s0ck | so i was getting: -- <SIP/101-082aa1c8> Playing 'vm-theperson' (language 'en') |
| 15:52.16 | s0ck | now i get -- <SIP/101-08eb94e8> Playing 'vm-theperson' (language 'uk') |
| 15:52.34 | s0ck | doesn't appear to be playing the new sounds tho |
| 15:52.34 | RoyK | s0ck: shouldn't that be en_UK? |
| 15:52.47 | s0ck | RoyK: i was wondering that too |
| 15:53.04 | RoyK | s0ck: rename the directory to en_UK and change language accordingly |
| 15:53.36 | RoyK | it shouldn't matter, though - AFAICR asterisk just uses /var/lib/asterisk/sound/$lang as the path |
| 15:53.50 | *** join/#asterisk MaartenB_ (n=Maarten@195-241-32-141.ip.telfort.nl) |
| 15:54.11 | s0ck | -- <SIP/101-b7900708> Playing 'vm-theperson' (language 'en_UK') | still playing the default us ones, argh |
| 15:54.19 | Qwell | it can do dialect if you give it one, and it exists |
| 15:54.23 | jaytee | man, * with Exchange UM totally rocks!!!!!!! |
| 15:55.07 | southtel | We have a remote office that just changed to DSL, and I'm pretty sure that we're having NAT issues with the new router... |
| 15:55.14 | Qwell | and of course, en_UK doesn't exist... |
| 15:55.30 | southtel | ...everything worked fine before, so I'm wondering if we could change out the telco's router for a different model... |
| 15:55.32 | Qwell | UK is not an ISO3166 country code |
| 15:55.45 | southtel | ...does anyone have any suggestions for an AT&T DSL router that we could use? |
| 15:55.46 | Qwell | you probably want en_GB |
| 15:57.12 | s0ck | no joy with en_GB ;/ |
| 15:57.17 | [TK]D-Fender | southtel: Sangoma S519 :) |
| 15:57.19 | Qwell | s0ck: What is the actual problem? |
| 15:57.39 | s0ck | i want to change the default US voice prompts to englishy sounding UK ones |
| 15:57.53 | Qwell | 1.4? |
| 15:58.05 | s0ck | Connected to Asterisk 1.4.18.1-2 |
| 15:58.16 | Qwell | The files are in /var/lib/asterisk/sounds/en_GB/ ? |
| 15:58.18 | RoyK | Qwell: GB is a country - not a language - ever been to scotland? |
| 15:58.19 | s0ck | yup |
| 15:59.01 | Qwell | and you have languageprefix=yes in asterisk.conf? |
| 15:59.20 | RoyK | s0ck: btw - the i18n term for British English is en_UK, not GB |
| 15:59.30 | Qwell | RoyK: You are wrong. |
| 16:00.37 | s0ck | neither of them work, chaps, so let's move on :) |
| 16:00.41 | RoyK | Qwell: hm. you may be right |
| 16:00.53 | RoyK | s0ck: seems asterisk ignores it then :P |
| 16:00.55 | Qwell | southtel: is that option set? |
| 16:00.59 | Qwell | s0ck: ^^ |
| 16:01.09 | s0ck | southtel? |
| 16:01.14 | Qwell | and you have languageprefix=yes in asterisk.conf? |
| 16:01.53 | southtel | Sorry...was on froogle looking for sangomas... |
| 16:02.16 | dominic1 | does anybody know what is the filename of the beep sound when somebody joins a conference? |
| 16:02.50 | *** join/#asterisk uTx (n=unix@modemcable232.79-58-74.mc.videotron.ca) |
| 16:02.53 | Qwell | dominic1: it isn't a file by itself |
| 16:03.02 | Qwell | it's actually compiled in, for some silly reason.. |
| 16:03.06 | southtel | Qwell, are you asking me about my languageprefix setting? |
| 16:03.13 | Qwell | southtel: no |
| 16:03.23 | southtel | What option then? |
| 16:03.30 | Qwell | none, it was a typo |
| 16:03.37 | southtel | Gotcha...sorry. |
| 16:04.02 | dominic1 | I am currently editing the meetme application and what that the software plays a beep before it anounces a new user |
| 16:04.16 | *** join/#asterisk raytruz` (n=raytruz_@74-129-178-146.dhcp.insightbb.com) |
| 16:04.37 | raytruz` | Anyone know of a good device to turn a regular coordless pots phone into an IP device so I can use it with asterisk? |
| 16:05.11 | [TK]D-Fender | raytruz`: Linksys PAP2T-NA |
| 16:05.18 | ManxPower | raytruz`: the devices are called ATAs, SIPura are good ones |
| 16:05.31 | raytruz` | Thanks |
| 16:05.32 | [TK]D-Fender | ManxPower: Shouldn't use that name anymore.... |
| 16:05.39 | Qwell | dominic1: it's in enter.h and leave.h, if you feel like trying to modify them.. |
| 16:05.52 | southtel | D-Fender: that was a bit of a red herring...if I could do this myself, I'd love to use the internal PC card option (that just seems kinda cool)... |
| 16:06.05 | southtel | ...but alas, I need something more "pre-packaged". |
| 16:06.24 | s0ck | Qwell: no |
| 16:06.31 | Qwell | s0ck: set it |
| 16:06.35 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 16:06.40 | dominic1 | okay a conf_play(chan, conf, ENTER); |
| 16:06.43 | [TK]D-Fender | southtel: DSL modems are a dime a dozen. Alcatel's used to be decent. |
| 16:06.49 | dominic1 | should do what I want, thanks qwell |
| 16:07.29 | Qwell | under [options] |
| 16:07.34 | southtel | D-Fender: Thanks...I just wanted to check here before I went out and got the first one I found...we're down to our last strike with this setup. |
| 16:07.51 | s0ck | Qwell: no joy |
| 16:08.00 | raytruz` | [TK]D-Fender: on that linksys, what is this business about it being "unlocked"? |
| 16:08.28 | [TK]D-Fender | raytruz`: Some ATA's are sold locked to a particular provider like Vonage and prevent you from configuing it yourself. |
| 16:08.47 | raytruz` | Ah, so linksys sells it like that out of the box? |
| 16:10.18 | [TK]D-Fender | raytruz`: If you find it in a retail store, odds are its locked. If you find it at a typical voip-store online, odds are it isn't. |
| 16:10.39 | [TK]D-Fender | raytruz`: www.telephonydepot.com |
| 16:10.40 | raytruz` | I see. |
| 16:10.55 | raytruz` | Well I guess that rules out me going and buying one today :-) |
| 16:11.48 | [TK]D-Fender | raytruz`: From Best Buy anyways ;) |
| 16:11.55 | [TK]D-Fender | raytruz`: TD there ships FAST... |
| 16:12.05 | [TK]D-Fender | raytruz`: US48 is usually next day. |
| 16:12.14 | hsv-al | d-fender |
| 16:12.19 | hsv-al | thats where I bought my tdm411P from |
| 16:12.22 | hsv-al | good price |
| 16:12.50 | raytruz` | Btw, there is no cheaper way to connect my pots phone as ANY type of device to asterisk right? (like a fxo card) |
| 16:13.01 | raytruz` | I wouldn't care if it was a zap device :-) |
| 16:13.13 | raytruz` | Too bad I can't do that with the x100p card |
| 16:13.14 | *** join/#asterisk coppice (n=chatzill@199.204.17.210.dyn.pacific.net.hk) |
| 16:13.27 | s0ck | forums are full of posts with people unable to get this working |
| 16:14.06 | hsv-al | raytruz |
| 16:14.07 | hsv-al | http://store.digium.com/productview.php?product_code=SOLOFXO |
| 16:16.05 | raytruz` | yeah, that is just the module :-) |
| 16:16.23 | raytruz` | So the linksys is defn the cheapest way to get it done |
| 16:16.24 | hsv-al | hmmm, |
| 16:16.25 | hsv-al | $3000 |
| 16:16.30 | hsv-al | Asterisk Bootcamp |
| 16:16.31 | hsv-al | A Five-Day Ultra-Intensive Course Huntsville, AL USA |
| 16:16.31 | hsv-al | Accommodations English 2008-Jul-07 - |
| 16:16.31 | hsv-al | 2008-Jul-11 $3000.00 |
| 16:16.35 | raytruz` | LOL |
| 16:16.59 | raytruz` | Would be cool to meet some people there, but only on your company's dime |
| 16:20.15 | [TK]D-Fender | raytruz`: Yes, you could get a 1-port usint from GrandSuck or similar, but that is the minimum recommended unit. |
| 16:20.38 | *** join/#asterisk gardo (n=gardo@121.97.140.126) |
| 16:21.06 | s0ck | (NOTE: The structure for 1.4 is diferent but it requires a configuration change) |
| 16:21.12 | s0ck | so what is it...? |
| 16:23.54 | Qwell | the one I told you to set |
| 16:24.31 | Qwell | where exactly is vm-theperson? |
| 16:24.46 | Qwell | wait, did you say 1.4.18? |
| 16:24.54 | Qwell | upgrade |
| 16:25.41 | *** join/#asterisk juice_d (n=juice_d@gozur.sunflowerbroadband.com) |
| 16:26.58 | s0ck | is there a known voice prompt bug in this distro then? |
| 16:27.28 | ManxPower | s0ck: distro? |
| 16:27.44 | [TK]D-Fender | s0ck: Use packages, and you get what you deserve. |
| 16:27.48 | s0ck | 1.14.18 |
| 16:27.55 | ManxPower | we don't support distros or packages or guis |
| 16:28.08 | ManxPower | if you have any of those go to the correct support forum. |
| 16:28.15 | Qwell | Changes since asterisk Version 1.4.18/ - svn revision 101648 |
| 16:28.15 | Qwell | 441 |
| 16:28.40 | *** join/#asterisk laichzeit (n=santa@dsl-241-153-236.telkomadsl.co.za) |
| 16:29.32 | *** join/#asterisk deeperror (n=deeperro@adsl-76-234-138-5.dsl.sfldmi.sbcglobal.net) |
| 16:30.20 | *** join/#asterisk tobias (n=tobias@cpe-069-134-205-184.nc.res.rr.com) |
| 16:30.24 | laichzeit | hi all, I'm having a problem detecting incoming calls on a tdm400, I see the ring on ztmonitor 1 -vv, but asterisk does not start a ring event (tail -f full.log), can anyone tell me what I should be looking at to get this working? |
| 16:30.42 | s0ck | bizarrely, it mentions something about english prompts on the changelog |
| 16:38.40 | Qwell | Idle: . |
| 16:39.29 | *** join/#asterisk codestr0m (n=asura@76.74.174.194) |
| 16:40.21 | s0ck | laichzeit: can you dial out? |
| 16:40.28 | laichzeit | s0ck, yes |
| 16:42.13 | laichzeit | also, if I do a "zap show channel 1" it says "Hookstate: Onhook", don't know if that is right or not. |
| 16:43.36 | *** join/#asterisk af_ (n=getsmart@88-149-240-186.dynamic.ngi.it) |
| 16:45.05 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 16:45.14 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.151) |
| 16:45.38 | ManxPower | I thougt it said right there that hookstate is only valid for FXS? |
| 16:48.08 | *** join/#asterisk arctic_import (n=jasonj@mail.uui-alaska.com) |
| 16:48.57 | arctic_import | I just build asterisk 1.4.20.1, how come I don't have any iax2 commands anymore? I used to use 1.2, and use the iax2 show peers command. Now it doesn't exist. |
| 16:48.59 | codestr0m | ok. I know I shouldn't ask here, but anyone have for free or pay an updated sip based firmware for a Cisco 7960g.. feel free to pm me since this is certainly ot.. I'm on a very old sip version.. thanks |
| 16:49.22 | arctic_import | codestr0m, buy smartnet. |
| 16:49.55 | s0ck | arctic_import: some of the commands must now be prefixed with 'core' |
| 16:50.01 | *** join/#asterisk znoG (n=gs@host226.190-30-156.telecom.net.ar) |
| 16:51.00 | znoG | hey, i've a problem where I can dial some numbers just fine (the majority), but for some specific ones Asterisk doesn't detect when the call has been answered, so it drops the call after the timeout is reached (as it doesn't detect the call has been answered) |
| 16:51.04 | znoG | any ideas what switches I can play with? |
| 16:52.40 | arctic_import | s0ck: is there a cheat sheet somewhere on the net? the help core command is useless |
| 16:52.49 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 16:54.04 | [TK]D-Fender | arctic_import: make sure chan_iax2.so is loaded. |
| 16:54.18 | arctic_import | arctic_import: how do I do that? |
| 16:54.19 | *** join/#asterisk CVirus (n=GoD@41.233.145.155) |
| 16:54.52 | arctic_import | [TK]D-Fender, How do I do that? My fist time using 1.4. Do I load them in some config file? |
| 17:00.22 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:00.50 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 17:03.07 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 17:03.20 | hsv-al | are there 'any' software sip , or software iax clients |
| 17:03.24 | hsv-al | for blackberries, ie: 8703e? |
| 17:03.33 | s0ck | updated *, same bloody prompts :| |
| 17:04.07 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 17:05.09 | lesouvage | I have installed asterisk 1.4.18.1 a while ago and now I want to add the asterisk add-ons that fit with the 1.4.18.1 asterisk version but I can't find any info about this. Can anybody please point me to the proper asterisk add-ons? |
| 17:05.30 | Daejeo | [TK]D-Fender:) |
| 17:05.37 | Daejeo | [TK]D-Fender :) |
| 17:06.06 | [TK]D-Fender | lesouvage: look at the changelogs, do the math |
| 17:06.08 | hsv-al | just found this |
| 17:06.11 | hsv-al | on a blackberry forum |
| 17:06.12 | hsv-al | No. Currently there are NO J2ME SIP clients. There are a few people working on a true SIP VOIP client and a handful of programs that CLAIM to be to VOIP (but they use a call-back scheme that ends up using your cell plan minutes...this includes Gizmo). Estimates are about another 1-2 years before someone finally comes up with a workable SIP client for the Blackberry. |
| 17:06.16 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 17:06.37 | hsv-al | in reference to blackberry 8xxx's |
| 17:07.51 | nny_1 | anyone know of any bad effects that could arise from setting Subscribe Expires: 10 Subscribe Retry Interval: 5 (i believe these are in seconds) on a linksys962/932 combo? It seems to help with the phone re-associating on a service interruption |
| 17:08.59 | *** join/#asterisk sack (n=sack@249.Red-81-32-160.dynamicIP.rima-tde.net) |
| 17:09.11 | RoyK | lesouvage: just grab the latest -addon |
| 17:09.21 | lesouvage | [TK]D-Fender: there isn't a table somewhere with a column <asterisk version>, <zaptel version>, <libpri version>, <asterisk add-on version>? That would be helpfull for lots of people. |
| 17:09.33 | lesouvage | RoyK: thanks. |
| 17:09.51 | [TK]D-Fender | lesouvage: Dunno, never seen one. tons of places to look. |
| 17:09.57 | RoyK | lesouvage: most versions of libpri/zaptel/asterisk/-addons work with another |
| 17:10.09 | RoyK | at least within major releases |
| 17:10.23 | RoyK | minor that is |
| 17:10.46 | RoyK | 1.2 addons may not work with 1.4, but 1.4.something should work with asterisk 1.4.something |
| 17:11.05 | RoyK | zaptel usually works across versions |
| 17:16.01 | loompek | i have problems connecting sjphone to asterisk.. even though asterisk's sip show peers says OK, sjphone says Not Registered... any ideas? |
| 17:17.11 | *** join/#asterisk tobias (n=tobias@cpe-069-134-205-184.nc.res.rr.com) |
| 17:18.07 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 17:19.03 | s0ck | RoyK: have you managed to change the sound files in * at all? |
| 17:19.39 | s0ck | has anyone... |
| 17:20.03 | nny_1 | anyone see anything funky with this code: http://pastebin.com/m4377ac23 |
| 17:20.04 | [TK]D-Fender | s0ck: Yes. You managed to discover pastebin yet? |
| 17:20.09 | nny_1 | testing it now |
| 17:20.40 | nny_1 | er i see an error |
| 17:20.45 | nny_1 | Busy = "0" |
| 17:21.07 | nny_1 | er 3 not 0 |
| 17:22.37 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:23.16 | nny_1 | hmm seeing some goofs here, i think i can make this work |
| 17:24.27 | s0ck | [TK]D-Fender: thank you for your input but i find you largely unhelpful so please don't feel obliged to respond to me in future. |
| 17:24.54 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-198-98.rgv.res.rr.com) |
| 17:26.03 | TrentCreek | who was touting their hosting services a while back? |
| 17:27.30 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 17:28.48 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 17:29.06 | nny_1 | i am at http://pastebin.com/m63b9eacf now but still learning how to use chanisavail. I assume in that code I am asking it to check and in turn set the variable ${AVAILSTATUS} to 5 (unavailable) based on the fact that with the "s" parameter I have told it to assume unavailable *at all* if any channels are in us |
| 17:29.17 | nny_1 | use |
| 17:29.30 | nny_1 | however judging by the output from cli i am still not doing it right |
| 17:30.18 | *** join/#asterisk snapple42 (n=snapple4@h216-18-80-132.gtconnect.net) |
| 17:32.27 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 17:32.41 | a1fa | are there any other brave souls running * in XEN? |
| 17:35.32 | [TK]D-Fender | s0ck: well you ask generic questions without showing us exactly what you've done. Are we supposed to guess where you might have gone wrong? |
| 17:35.53 | a1fa | waves at D-Fender |
| 17:35.53 | a1fa | :P |
| 17:36.30 | *** join/#asterisk freezey (i=hidden-u@gw.mypublisher.com) |
| 17:38.35 | *** join/#asterisk thomas (i=tm@tm.muc.de) |
| 17:38.37 | thomas | hola! |
| 17:39.07 | [TK]D-Fender | nny_1: Look at this, tell me what you see as being odd with it : -- Executing [*26@sip:2] GotoIf("SIP/20-09745d28", "0 ?10") in new stack |
| 17:39.26 | thomas | is it posible the format for "Monitor" as "mp3" ? |
| 17:39.31 | Qwell | thomas: no |
| 17:39.37 | thomas | Qwell: hm. only as WAV? |
| 17:39.42 | thomas | Qwell: Hello. |
| 17:39.45 | Qwell | any supported format |
| 17:39.57 | thomas | Qwell: which format is supported? |
| 17:40.22 | [TK]D-Fender | thomas: "show codecs" |
| 17:40.32 | thomas | ah, ok. |
| 17:41.10 | *** join/#asterisk eharris (n=eharris@75-43-20-21.lightspeed.austtx.sbcglobal.net) |
| 17:41.26 | a1fa | blah |
| 17:41.30 | a1fa | i need to buy a book on * |
| 17:41.34 | a1fa | soo many new commands |
| 17:41.40 | Qwell | ~book |
| 17:41.41 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:41.49 | a1fa | sweet |
| 17:41.53 | a1fa | free download |
| 17:41.53 | a1fa | :p |
| 17:41.59 | a1fa | ~wsteal |
| 17:42.02 | a1fa | i mean wget :P |
| 17:42.04 | kamh | handbook is better :P |
| 17:42.11 | *** join/#asterisk bootc (n=bootc@78-33-44-114.no-dns-yet.enta.net) |
| 17:42.30 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 17:42.35 | [TK]D-Fender | handjob is better :p |
| 17:42.37 | bootc | can one remove a SIP header added using SIPAddHeader ? |
| 17:42.39 | kamh | :] |
| 17:42.49 | a1fa | bj > hj |
| 17:42.59 | kamh | hehhehe |
| 17:43.09 | [TK]D-Fender | bootc: Can one install a bottle-cap with a bottle opener? |
| 17:43.29 | [TK]D-Fender | a1fa: Indeed. Leaving room for the continued progression ;) |
| 17:43.42 | bootc | g2g now, but I don't mean to remove the header using SIPAddHeader :-P |
| 17:44.36 | a1fa | beats developers |
| 17:44.52 | [TK]D-Fender | ~developers |
| 17:44.53 | jbot | developers is probably http://www.youtube.com/watch?v=KMU0tzLwhbE |
| 17:45.08 | a1fa | seriously ;( |
| 17:45.15 | a1fa | i am pissed off about clicking in playback |
| 17:45.46 | a1fa | timing mining |
| 17:46.22 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-149.hsi.ish.de) |
| 17:50.36 | Qwell | Strom_M: ping |
| 17:52.46 | a1fa | so.. ;) no answer on * in XEN |
| 17:52.53 | a1fa | everybody is a chicken in here to talk about it ;P |
| 17:55.20 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 17:55.24 | nny_1 | [TK]D-Fender: i am re-reading the documentation on the variable. It read like AVAILSTATUS would return a numerical value, but obviously that is incorrect if it is returning the channel |
| 17:55.55 | [TK]D-Fender | nny_1: No, just look at the result of your GOTO and tell me what you see. |
| 17:57.34 | a1fa | [TK]D-Fender : you recommend IP320, right? |
| 17:57.40 | a1fa | err |
| 17:57.42 | a1fa | IP330 |
| 17:57.51 | [TK]D-Fender | a1fa: IP 320 typically. |
| 17:58.56 | nny_1 | I think i have a space somewhere i shouldn't in my gotoifstatement, after the = |
| 17:59.05 | [TK]D-Fender | nny_1: :) |
| 17:59.08 | nny_1 | :D |
| 17:59.40 | [TK]D-Fender | nny_1: Good. Next, learn to NoOp variables before using them in a GotoIf. Thats what you call a "sanity check" |
| 18:00.37 | *** join/#asterisk deeperror (n=deeperro@adsl-76-234-138-5.dsl.sfldmi.sbcglobal.net) |
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| 18:01.44 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 18:02.00 | lmadsen | prefers Verbose() |
| 18:03.35 | *** join/#asterisk eharris (n=eharris@75-43-20-21.lightspeed.austtx.sbcglobal.net) |
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| 18:03.35 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) [NETSPLIT VICTIM] |
| 18:03.35 | *** join/#asterisk disposable (i=disposab@blackhole.sk) [NETSPLIT VICTIM] |
| 18:03.35 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) [NETSPLIT VICTIM] |
| 18:03.35 | *** join/#asterisk esk (i=esk@esk.cs.usu.edu) [NETSPLIT VICTIM] |
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| 18:03.36 | *** join/#asterisk yang (i=yang@CAcert/Assurer/yang) |
| 18:06.06 | a1fa | hey is waitexten significant? |
| 18:06.52 | Qwell | it wouldn't exist if it wasn't |
| 18:07.04 | a1fa | i mean it works without it |
| 18:07.05 | a1fa | :P |
| 18:07.24 | *** join/#asterisk smach (n=smach@207.35.173.122) |
| 18:07.42 | smach | HI guys |
| 18:08.25 | *** join/#asterisk plik (i=gorph@phalse.2600.COM) |
| 18:08.34 | smach | I'm having some trouble with the line registration on asterisk |
| 18:09.14 | [TK]D-Fender | a1fa: marginal value |
| 18:09.16 | smach | I can make asterisk register line against a Mitel 3300, and receive calls from that 3300 |
| 18:09.21 | [TK]D-Fender | smach: Do tell... |
| 18:10.22 | a1fa | heh |
| 18:10.27 | a1fa | i dont understand marginal value :( |
| 18:10.58 | [TK]D-Fender | a1fa: very little. As you said, you don't need it. |
| 18:11.14 | a1fa | also answer() is pointless i think |
| 18:11.24 | [TK]D-Fender | a1fa: Not in 1.4 and under anyways. Can't vouch for 1.6+ changes |
| 18:11.32 | a1fa | awaits for a beat-down |
| 18:11.32 | [TK]D-Fender | a1fa: No, not pointless. |
| 18:11.38 | a1fa | why not? |
| 18:11.51 | smach | [TK]D-Fender: sorry, didn't get you |
| 18:12.03 | [TK]D-Fender | a1fa: Sometimes you need to answer the line right away and not vai some sort of playback that force-answeres |
| 18:12.10 | a1fa | ah |
| 18:12.12 | a1fa | cool |
| 18:12.21 | [TK]D-Fender | smach: I asked you to continue to tell us the problem you've got. |
| 18:12.32 | [TK]D-Fender | via* |
| 18:12.50 | smach | [TK]D-Fender: ok thanks |
| 18:13.00 | nny_1 | hmm need to keep trying this till i get it, i have Gotoif($["${AVAILSTATUS}" =5]?10) but it says GotoIf("SIP/20-0974e790", "0?10") which i believe reads as SIP/20-0974e790 is what AVAILSTATUS is, which doesn't match (0 being no?) and doesn't jump to 10. The docs reads as though AVAILSTATUS should return 0-6. Am i reading the output right? |
| 18:13.14 | *** part/#asterisk codestr0m (n=asura@76.74.174.194) |
| 18:13.30 | smach | well basically what I'm trying to do is to make the asterisk behave like multiple user agents |
| 18:14.05 | smach | I have cisco set registered against my asterisk and I want asterisk to register multiple lines against the Mitel 3300 |
| 18:14.06 | s0ck | well, i got it working but had to cheat |
| 18:14.10 | nny_1 | er no i am not eh |
| 18:14.14 | s0ck | converted all my wav to gsm and overwrote the existing sound files |
| 18:14.23 | *** join/#asterisk SamuraiDio (n=diovani@201.41.41.235) |
| 18:14.25 | a1fa | i need to put some sound "bling bling bling" after answer ;) kind of like what ATT does.. anybody suggest a sound? |
| 18:14.28 | nny_1 | the SIP/etc is just stating the channel, as it always does |
| 18:14.31 | a1fa | i was thinking monkey sound may do |
| 18:14.36 | SamuraiDio | how do i allow anonymous sip accounts to register? |
| 18:15.14 | smach | so the 3300 accepts the registration from Asterisk |
| 18:15.37 | smach | it sends calls to Asterisk whenever one of the extensions registered is called |
| 18:16.25 | [TK]D-Fender | nny_1: Now you are failing to match both sides of an evaluation. Quote characters are LITERAL <- |
| 18:16.31 | a1fa | Playtones(425/50,0/50) |
| 18:16.37 | smach | which is half of what I need, the other part is to be able to call the 3300 from the cisco sets |
| 18:17.19 | *** join/#asterisk tobias (n=tobias@cpe-069-134-205-184.nc.res.rr.com) |
| 18:17.29 | [TK]D-Fender | smach: and what happens when you try to call to the 3300 right now? |
| 18:17.41 | a1fa | hm |
| 18:17.45 | a1fa | playtones dont work crap |
| 18:17.47 | smach | 404 from the 3300 |
| 18:18.13 | [TK]D-Fender | smach: pastebin the SIP debug of a failed attempt |
| 18:18.15 | [TK]D-Fender | ~pb |
| 18:18.15 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:18.16 | [TK]D-Fender | ^^^^^^^^^^ |
| 18:19.19 | a1fa | hm |
| 18:19.32 | a1fa | why doesnt playtones work i wonder |
| 18:22.23 | smach | here is the pastebin of my wireshark capture http://pastebin.com/d1799e593 |
| 18:22.47 | *** join/#asterisk bkruse (n=bkruse@phoenixii.total-vision.net) |
| 18:22.47 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 18:23.20 | *** join/#asterisk MaartenB_ (n=Maarten@84-105-197-212.cable.quicknet.nl) |
| 18:23.52 | *** join/#asterisk shido6 (n=shido6@74-130-224-188.dhcp.insightbb.com) |
| 18:24.10 | nny_1 | [TK]D-Fender: so i should have quotation marks around the expected value? I only changed that based on an example i saw |
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| 18:25.04 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) [NETSPLIT VICTIM] |
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| 18:25.04 | *** join/#asterisk _mm_ (n=mmclain@cpe-67-49-233-178.dc.res.rr.com) [NETSPLIT VICTIM] |
| 18:25.04 | *** join/#asterisk yang (i=yang@CAcert/Assurer/yang) |
| 18:25.04 | nny_1 | it's too bad i can't use the same states the blf registration uses to see if the channel is "in use" |
| 18:25.05 | [TK]D-Fender | nny_1: each side of the = is literal. if you use quotes on one side, it'd better have quotes on the other. |
| 18:25.12 | nny_1 | [TK]D-Fender: gotcha |
| 18:27.13 | a1fa | [TK]D-Fender : got a second |
| 18:28.07 | a1fa | i want to give you * # |
| 18:28.09 | [TK]D-Fender | a1fa: maybe even two |
| 18:28.19 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 18:28.30 | a1fa | and see if you can tell me if this clicking is related to timing :( |
| 18:28.47 | [TK]D-Fender | smach: can you do sip debug via * instead? from the beginning please. |
| 18:29.01 | smach | sure |
| 18:29.17 | *** join/#asterisk Falle (n=falle@diana.falle.se) [NETSPLIT VICTIM] |
| 18:29.48 | a1fa | anybody running * in XEN enviroment? |
| 18:31.02 | [TK]D-Fender | smach: Actually... based on the 100 Trying, I'd say that the Mitel doesn't like that # you dialed. Either "6001" isn't legit, or * doesn't have access to it. |
| 18:31.33 | [TK]D-Fender | smach: because getting a 100 should mean it accepted the call auth at least AFAICT |
| 18:32.03 | nny_1 | hmm i may be doing something wrong before that goto |
| 18:32.28 | nny_1 | i don't think AVAILSTATUS is returning what I expect at all |
| 18:32.33 | nny_1 | the c code states 5 AST_DEVICE_UNAVAILABLE |
| 18:32.38 | nny_1 | as the return value |
| 18:33.58 | nny_1 | still wondering if there is a variable somewhere that reflects the hint status for that extension |
| 18:37.23 | jblack | I'm having problems with poor call quality on a PRI, and I don't know how to diagnose it, or where to learn how to |
| 18:37.53 | arctic_import | I can't get my IAX2 trunk to work. I'm geting No Authority errors. I'm using a type=friend, user/secret are correct on both sides. Any ideas? |
| 18:38.35 | jblack | It sounds like... well, like as if packets were being dropped, but afaik, pri's don't have packets |
| 18:47.02 | *** join/#asterisk ajricoveri_ (n=ajricove@190.37.169.212) |
| 18:48.11 | [TK]D-Fender | arctic_import: make sure you specify the target context when you dial. |
| 18:48.51 | ajricoveri_ | hi, i'd like to know if i can divide sip.conf user entries on different files like sip_client1.conf and sip_client2.conf and include them on asterisk ... =) |
| 18:48.51 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.140.193) |
| 18:49.16 | *** join/#asterisk af_ (n=getsmart@88-149-240-186.dynamic.ngi.it) |
| 18:50.01 | [TK]D-Fender | ajricoveri_: Yes. do "#include myotherfile.conf", etc |
| 18:50.07 | smach | [TK]D-Fender: sorry I was installing asterisk on another server |
| 18:50.31 | smach | member:[TK]D-Fender: I'll take a look at what you said |
| 18:51.25 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 18:51.43 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 18:52.03 | ajricoveri_ | [TK]D-Fender, thx a lot =) ... i need to modularize context users |
| 18:52.17 | smach | [TK]D-Fender: 6001 is the voicemail extension, where do you see a # dialed ? |
| 18:52.20 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 18:52.29 | ajricoveri_ | [TK]D-Fender, btw, i readed the book yesterday as u told me =) and gave me all the answers =) thank u again |
| 18:52.30 | [TK]D-Fender | smach: in the "To:" header. |
| 18:55.28 | smach | [TK]D-Fender: I must need glaces :) , I see no # in the To header -> "To: <sip:6001@192.168.215.20>;tag=0_3079386432-60271647" |
| 18:55.43 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 18:55.51 | SamuraiDio | ~test |
| 18:55.51 | jbot | Failed! |
| 18:55.52 | *** join/#asterisk NovceGuru (n=NovceGur@oh-65-40-70-180.sta.embarqhsd.net) |
| 18:55.53 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 18:55.59 | [TK]D-Fender | smach: Correct... you probably do. its the "6001" |
| 18:56.36 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) [NETSPLIT VICTIM] |
| 18:56.36 | *** join/#asterisk Falle (n=falle@diana.falle.se) [NETSPLIT VICTIM] |
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| 18:59.44 | nny_1 | [TK]D-Fender: shouldn't NoOp show me what AVAILSTATUS is reporting in console ? |
| 18:59.44 | nick125 | 10 |
| 18:59.46 | nick125 | er |
| 18:59.48 | *** join/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 18:59.49 | *** part/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 19:00.07 | [TK]D-Fender | nny_1: yup.... if your verbose is high enough. I suggest "10" |
| 19:00.37 | smach | [TK]D-Fender: Sorry I just got what you meant by # and * :( |
| 19:00.38 | nny_1 | [TK]D-Fender: yeah not seeing 5 or 5 AST_DEVICE_UINAVAIL etc |
| 19:00.44 | nny_1 | hrrm |
| 19:00.51 | [TK]D-Fender | nny_1: Thanks for showing me exactly what you're doing... |
| 19:02.22 | nny_1 | [TK]D-Fender: roger one sec |
| 19:03.17 | Titanous | Can I have some recommendations for toll-free DID providers in Canada? |
| 19:03.17 | *** part/#asterisk Rem| (n=remi@bg-fw2out.monmouth.com) |
| 19:05.57 | *** join/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 19:06.01 | ThoMe | emm |
| 19:06.08 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) [NETSPLIT VICTIM] |
| 19:06.08 | *** join/#asterisk Falle (n=falle@diana.falle.se) [NETSPLIT VICTIM] |
| 19:06.08 | *** join/#asterisk bminish (n=bminish@brenbox.home.minish.org) [NETSPLIT VICTIM] |
| 19:06.08 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) [NETSPLIT VICTIM] |
| 19:06.08 | *** join/#asterisk keith4 (n=keith@lust.CC.Lehigh.EDU) [NETSPLIT VICTIM] |
| 19:06.18 | ThoMe | have in extensions.conf |
| 19:06.18 | ThoMe | this |
| 19:06.18 | ThoMe | [globals] |
| 19:06.18 | ThoMe | MONITOR_EXEC=/var/lib/asterisk/agi-bin/wav2mp3 |
| 19:06.24 | ThoMe | are this ignored on asterisk 1.2X ? |
| 19:06.56 | nny_1 | [TK]D-Fender: http://pastebin.com/m50a1e092 |
| 19:07.17 | nny_1 | [TK]D-Fender: pretty sure i am doing something wrong with chanisavail or it's variable |
| 19:07.26 | nny_1 | since i dont see any of the expected values |
| 19:08.06 | [TK]D-Fender | nny_1: You really aren't thinking too straight... don't NoOp an EVALUATION, NoOp the damn VARIABLE. ALONE <- |
| 19:08.21 | nny_1 | ok |
| 19:09.11 | smach | [TK]D-Fender: Here is the SIP set debug output http://pastebin.com/d5f5e0b25 |
| 19:09.43 | nny_1 | hmm that's better |
| 19:09.48 | nny_1 | NoOp("SIP/20-0974e790", "2") |
| 19:10.03 | nny_1 | wow that explains a hell of a lotrr |
| 19:10.05 | nny_1 | lot |
| 19:10.54 | smach | [TK]D-Fender: When I have phone registered directly to the 3300 (and using the same extension= 7003), I can call the ext=6001 and receive calls |
| 19:11.04 | [TK]D-Fender | smach: Looks like the Mitel doesn't match that # up ery well... |
| 19:11.27 | geoffl | part |
| 19:11.32 | [TK]D-Fender | full |
| 19:11.43 | nny_1 | [TK]D-Fender: is the SIP/20 part of the variable returned from _*XX,2,NoOp(${AVAILSTATUS}) |
| 19:11.43 | nny_1 | <PROTECTED> |
| 19:12.03 | smach | [TK]D-Fender: I can't see why he wouldn't do so when the invite is sent by Asterisk |
| 19:12.08 | [TK]D-Fender | nny_1: You already should know the answer to that... |
| 19:12.34 | *** part/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 19:12.35 | nny_1 | [TK]D-Fender: yes i do, thanks |
| 19:12.40 | nny_1 | [TK]D-Fender: it seems to work now |
| 19:12.45 | *** join/#asterisk James|TCC (i=James@87-194-161-247.bethere.co.uk) |
| 19:13.06 | nny_1 | although i was expecting "5" since i told chanisavail to use unavail if any lines were in use |
| 19:13.17 | nny_1 | however 2 (busy) seems to be true when 1 sip line is in use |
| 19:14.20 | *** join/#asterisk l2cache (n=chatzill@179.190.204.68.cfl.res.rr.com) |
| 19:15.08 | nny_1 | ok it seems busy = unavail |
| 19:15.20 | nny_1 | very confusing docs for that app heh |
| 19:15.32 | nny_1 | i mean |
| 19:16.05 | nny_1 | s - Consider the channel unavailable if the channel is in use at all = 2 AST_DEVICE IN USE - "In use"; channel is in use. NOT 5 AST_DEVICE_UNAVAILABLE - "Unavailable"; channel is unavailable which actually means NOT REGISTERED |
| 19:16.09 | nny_1 | my fault all the same |
| 19:17.03 | l2cache | I'm having a very weird problem. I have asterisk 1.4.20 running at two separate locations, at random intervals (maybe once a week, maybe 2x) calls stop coming in and no one can dial any extensions. When you log in to the CLI "sip show peers" and "show channels" and "sip show registry" all do not return any values. The only way for functionality to return is to restart the asterisk... |
| 19:17.05 | l2cache | ...service. Has anyone experienced this? |
| 19:17.44 | ThoMe | is in asterisk 1.2 no globals? |
| 19:17.50 | ThoMe | [globals] |
| 19:17.52 | ThoMe | <PROTECTED> |
| 19:17.52 | l2cache | I then installed FREEPBX on one location, and we haven't had that problem at all. |
| 19:17.53 | ThoMe | ? |
| 19:18.26 | keith4 | l2cache: anything in the log? |
| 19:18.38 | SamuraiDio | is there some configuration to allow/deny anonymous sip calls (when callerid=anonymous)? |
| 19:18.43 | keith4 | ThoMe: globals work fine in 1.2 |
| 19:19.18 | ThoMe | keith4: hm, but where? |
| 19:19.18 | l2cache | nothing in "var/log/messages" |
| 19:19.18 | l2cache | abnormal anyway |
| 19:19.18 | ThoMe | in extentins.conf or features.conf ? |
| 19:19.18 | ThoMe | keith4: ? |
| 19:19.54 | ThoMe | keith4: i would like set MONITOR_EXEC=/var/lib/asterisk/agi-bin/wav2mp3 |
| 19:20.20 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 19:20.30 | l2cache | if freepbx didn't have the problem on 1.2 I'm guessing that it is either the ver of asterisk I am running, or some obscure sip option that I am missing |
| 19:20.37 | James|TCC | Hi, I'm trying to configure an asterisk, with a TDM800P card. Ports 1-4 are FXS and unused atm, ports 5-8 are FXO with 5 and 6 connected to lines. Calls are working inbound, but i cant work out how to get groups working so all outbound calls are on 1 line only, if you try making 2 the system errors with " app_dial.c:1196 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)" |
| 19:20.55 | arctic_import | I cannot get my iax2 trunk working correctly. no matter what I do it gives me a No Authority. |
| 19:20.56 | James|TCC | from the info ive read, it seems really simple so i'm not sure what im doing wrong |
| 19:21.00 | keith4 | l2cache: what happens when you try to make a call, and it's in that state? what about verbose 10 output? |
| 19:21.04 | James|TCC | i'll post configs if anyone eants them |
| 19:21.34 | keith4 | James|TCC: are you using zap channel groups? |
| 19:21.35 | l2cache | with verbose 10, there is no scrolling in the CLI, inbound calls are dropped and registered extensions cannot make calls |
| 19:21.46 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 19:21.54 | l2cache | the CLI will scroll 1 line on every command you input, but no output at all, or functionality |
| 19:22.13 | keith4 | what distro? compiled yourself? |
| 19:22.38 | James|TCC | keith4: trying to i think |
| 19:22.38 | keith4 | James|TCC: paste zaptel.conf |
| 19:22.51 | [TK]D-Fender | James|TCC: pastebin your zapata.conf and the CLI output of your failed attempt at verbose 10 |
| 19:22.54 | [TK]D-Fender | ~pb |
| 19:22.54 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:22.56 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 19:22.57 | James|TCC | using asterisk now, but thinking of moving onto a regular distro tbh |
| 19:23.03 | James|TCC | yeah ta :P |
| 19:23.06 | [TK]D-Fender | James|TCC: PASTEBIN, not PASTE. |
| 19:23.19 | James|TCC | yes thanks [TK]D-Fender lol |
| 19:23.31 | James|TCC | (im an irc admin - i KNOW how annoying that is) :P |
| 19:23.41 | *** join/#asterisk ajricoveri_ (n=ajricove@201.248.93.18) |
| 19:23.48 | [TK]D-Fender | James|TCC: just saving myself the effort of having to clean up a mess... |
| 19:24.26 | l2cache | So no one has experience the CLI not responding to your commands? The only remedy being to restart asterisk. |
| 19:24.51 | smach | is there a sip header in the sip invite that will tell a proxy that the UA is a sip client or an asterisk ? |
| 19:24.52 | arctic_import | the trunk can be the same [name] and username/secret on both sides correct? |
| 19:25.39 | *** join/#asterisk fogo (n=fogo@72.8.104.15) |
| 19:26.02 | James|TCC | zaptel.conf: http://www.pastebin.ca/1045306 |
| 19:26.33 | smach | when I use Asterisk as a UA registering lines against a mitel ip pbx, I receive a 404 not a 401 or 407 as a reponse to the invite Asterisk sends |
| 19:26.38 | *** join/#asterisk Yourname`` (i=chatzill@unaffiliated/yourname/x-837320) |
| 19:27.06 | keith4 | James|TCC: sorry, I meant zapata.conf |
| 19:27.11 | James|TCC | ah 2 secs |
| 19:27.23 | James|TCC | i did wonder - ive been playing with zapata all day lol |
| 19:28.14 | Yourname`` | Is there a way in Asterisk 1.4* that I can rotate the Master.csv in /vsr/log/asterisk/cdr-custom nightly? |
| 19:28.31 | James|TCC | easier to paste that tho lol, has an empty [trunkgroups] then |
| 19:28.32 | James|TCC | [channels] |
| 19:28.32 | James|TCC | signalling => fxs_ks |
| 19:28.32 | James|TCC | group = 1 |
| 19:28.32 | James|TCC | channel => 5-6 |
| 19:28.42 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 19:28.59 | *** join/#asterisk tobias (n=tobias@cpe-069-134-205-184.nc.res.rr.com) |
| 19:29.20 | James|TCC | in extension.conf ive setup it to use Zap/g1 |
| 19:29.32 | James|TCC | it works, we can dial out - on the first line |
| 19:29.50 | James|TCC | the phones we're using are Flexor 500's, which have 4 line buttons |
| 19:30.05 | keith4 | tell me something |
| 19:30.10 | James|TCC | can someone point me at a readme which might help me configure the line buttons to work with a particular line? |
| 19:30.17 | keith4 | do you notice anything different about "group = 1", compared with "channel => 5-6" ? |
| 19:30.22 | [TK]D-Fender | James|TCC: "button 2" will NOT pick your 2nd "line" |
| 19:30.28 | James|TCC | *apparantly* it worked this morning |
| 19:30.41 | James|TCC | yeah whats with the different syntaxes |
| 19:30.47 | keith4 | heh |
| 19:30.49 | [TK]D-Fender | James|TCC: g1 will pick the FIRST available line in that group |
| 19:31.00 | James|TCC | ive noticed all over the web half the sites use .. = ... the rest use .. => ... |
| 19:31.04 | *** join/#asterisk km2 (n=x@c-24-23-255-173.hsd1.mn.comcast.net) |
| 19:31.08 | [TK]D-Fender | James|TCC: See above |
| 19:31.21 | James|TCC | right , so if i make a call |
| 19:31.25 | James|TCC | picks line 5, |
| 19:31.29 | James|TCC | someone else makes a call |
| 19:31.36 | James|TCC | the first available line is 6 |
| 19:31.42 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 19:31.43 | [TK]D-Fender | James|TCC: Correct. |
| 19:31.43 | James|TCC | but instead they get error |
| 19:31.50 | James|TCC | all lines busy |
| 19:31.57 | keith4 | odd |
| 19:32.01 | [TK]D-Fender | James|TCC: pastebin the actual calls as I originally requested please. |
| 19:32.08 | James|TCC | lemme pastebin the lot |
| 19:32.09 | smach | hey guys, any change someone could help me with my issue, I ve been workin on it for a while |
| 19:32.30 | smach | I'd just like to know if I misconfigured Asterisk or is it coming from the 3300 side |
| 19:32.44 | [TK]D-Fender | smach: is the "from" header looking right to you? I thought I saw it mark is as if it were the Cisco... |
| 19:32.54 | James|TCC | Verbosity is at least 3 |
| 19:32.54 | James|TCC | Core debug is at least 10 |
| 19:33.03 | James|TCC | that right or do you want different debug output? |
| 19:33.17 | James|TCC | i cant see a command to increase verbosity |
| 19:33.23 | keith4 | you can never have too much verbosity |
| 19:33.29 | James|TCC | how do i increase it? |
| 19:33.40 | keith4 | what version? |
| 19:33.43 | James|TCC | might help lol |
| 19:34.12 | James|TCC | 1.4.18.1 |
| 19:34.51 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 19:35.04 | keith4 | uh, "core set verbose 10" ? |
| 19:35.58 | smach | [TK]D-Fender: the invite sent to the 3300 has a from header: "username" <sip:ext@Asterisk>;tag=TAG, does it look weird ? |
| 19:36.36 | James|TCC | right |
| 19:36.41 | James|TCC | core debug off too? |
| 19:37.47 | keith4 | why don't you start with verbose 10, and nothing else |
| 19:38.32 | James|TCC | http://www.pastebin.ca/1045317 |
| 19:38.38 | James|TCC | yeah have done, just checking |
| 19:38.48 | James|TCC | i turened on debug earlier (instead of verbose) |
| 19:39.35 | keith4 | are you able to call out of both of those channels, individually? |
| 19:39.41 | keith4 | like, not using Zap/g1 ? |
| 19:39.42 | *** join/#asterisk B1ST (i=overdose@d51A41676.access.telenet.be) |
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| 19:40.20 | B1ST | hi guys, i still have connection problems during the registering proces, here is the output, hope someone can help me out this .. |
| 19:40.34 | B1ST | [Jun 11 21:40:26] NOTICE[3465]: chan_sip.c:7511 sip_reg_timeout: -- Registration for 'ZHY3J169DIM2SR0LW5X@voipsolutions.be' timed out, trying again (Attempt #321) |
| 19:42.04 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 19:42.06 | James|TCC | if i update asterisk now's config, it will let me use either yeah |
| 19:42.12 | James|TCC | but they never work together |
| 19:42.51 | James|TCC | seems a bit of a flak in asterisknow to me, it will let me setup a dialplan, and assign a line to it, but nowhenre does it even mention groups |
| 19:43.10 | keith4 | tsk tsk tsk |
| 19:43.12 | James|TCC | hence me thinking about installing asterisk on its own on a 'normal' os |
| 19:43.33 | keith4 | you're lucky [TK]D-Fender didn't notice that you're using asteriskNOW, earlie |
| 19:43.34 | keith4 | r |
| 19:43.46 | James|TCC | i did say lol |
| 19:43.49 | keith4 | ~asterisknow |
| 19:43.49 | jbot | well, asterisknow is based on Asterisk, but it is not Asterisk, and it is unlikely to live up to Asterisk's standards. Only Asterisk is supported on #asterisk. Use #AsteriskNow instead. Even if the channel happens to be less helpful, support for systems other than Asterisk is offtopic on #asterisk |
| 19:44.20 | Strom_M | life would be so wonderful if we could just kill all the zealots |
| 19:44.23 | James|TCC | right ok lol |
| 19:44.43 | James|TCC | off to installing a distro to load asterisk onto then |
| 19:44.55 | James|TCC | any recommendations as to which distro works 'best' |
| 19:45.05 | Strom_M | whichever one you're most comfortable administering |
| 19:45.18 | James|TCC | ok fedora it is then |
| 19:45.33 | James|TCC | any issues with the latest versions i should know about? |
| 19:45.41 | Strom_M | not afaik |
| 19:45.58 | James|TCC | cool, right i'll be back when we have fc9 and asterisk on the machine :) |
| 19:47.28 | keith4 | does *NOW really not have support for zap groups? |
| 19:47.33 | keith4 | seems... shitty |
| 19:48.20 | tzafrir_laptop | edit config files, then |
| 19:48.44 | James|TCC | well keith4 |
| 19:49.03 | James|TCC | wanna try help me get it running, give it 5 mins before we say categorically "no it doesnt" lol |
| 19:49.16 | James|TCC | seems pretty shit to me too tbh |
| 19:49.17 | keith4 | not really |
| 19:49.20 | James|TCC | hehe |
| 19:49.34 | keith4 | i mean, zap groups are one of those things that just sort of "work", in my experience |
| 19:49.46 | keith4 | there isn't really any magic to making that work |
| 19:50.00 | James|TCC | idd, from what ive read of various help sites / blogs, the lines i have should be working |
| 19:51.43 | ThoMe | how i can record a call ondemand? |
| 19:51.51 | ThoMe | <PROTECTED> |
| 19:51.55 | ThoMe | is it correct? |
| 19:52.18 | l2cache | Has anyone had the asterisk CLI stop responding after a while. you can get in the CLI and type commands, but they do not output any results? |
| 19:58.02 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:00.18 | anonymouz666 | holy cow. Never saw something so bugged like chan_gtalk and jabber |
| 20:00.55 | *** join/#asterisk didz_ (n=hoje@201.19.140.193) |
| 20:05.26 | Daejeo | holy ox:) |
| 20:05.26 | Daejeo | holy ox :) |
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| 20:11.19 | fogo | I believe I've encounterd a bug in chanspy after upgrading to 1.4.20.1 - what's the best way to go about sending in a bug report? |
| 20:12.13 | Dovid | bugs.digium.com |
| 20:12.16 | Dovid | #asterisk-dev |
| 20:12.23 | Dovid | speak to a bug marshal there |
| 20:12.53 | fogo | is there any special info I should gather previous? |
| 20:12.59 | ThoMe | with "mixmonitor start" how i can move the file after this call? |
| 20:13.08 | Dovid | core dumps if any |
| 20:13.24 | Dovid | ThoMe: you can use the system command |
| 20:13.35 | ThoMe | Dovid: how? |
| 20:13.44 | fogo | no crashes, it just goes out of sync when people are put on hold |
| 20:13.50 | Dovid | http://www.voip-info.org/wiki/view/Asterisk+cmd+System |
| 20:13.54 | Dovid | use mv with it |
| 20:14.07 | ThoMe | Dovid: no, i mean automatic after the call |
| 20:14.08 | Dovid | fogo: ask the guys on #asterisk-dev |
| 20:14.16 | ThoMe | Dovid: if i press "mixmonitor start" and hangup |
| 20:14.20 | fogo | Dovid: ok. Thanks! :) |
| 20:14.27 | ThoMe | Dovid: then i would like "mv" |
| 20:15.12 | ThoMe | Dovid: understand?! |
| 20:17.12 | Dovid | ThoME: You can set the name of the file that record say to a variable and then use the h extension to move it with the system command |
| 20:18.07 | ThoMe | Dovid: and how can i do it "only then set the variable name X" ? |
| 20:18.34 | Dovid | you co do |
| 20:18.57 | Dovid | Set(foo=${EPOCH})) |
| 20:19.05 | Dovid | mixmonitor(${Foo}) |
| 20:19.11 | Dovid | then in the h extension |
| 20:19.28 | Dovid | system("mv ${Foo} /home/blah) |
| 20:19.31 | Dovid | system("mv ${Foo} /home/blah") |
| 20:19.45 | ThoMe | Dovid: hm, but i use "mixmonitor start" only the CLI |
| 20:19.59 | Dovid | u use it fromt eh CLI ? |
| 20:20.00 | ThoMe | not good? |
| 20:20.13 | ThoMe | jep. wrong? |
| 20:20.37 | Dovid | y would u do it there and not in the dial plan. i mus be missing something here |
| 20:21.45 | ManxPower | Dovid: I think you are missing a bunch of letters. This is not TXT 2 ur bff jill |
| 20:22.08 | Dovid | ManxPower: Must be sleep deprivation |
| 20:22.09 | seanbright | ManxPower: for the record, if that was my daughter, i would slap her right in the face |
| 20:22.13 | Dovid | sits in the corner |
| 20:22.17 | RoyK | Too high IQ to be a cop... http://query.nytimes.com/gst/fullpage.html?res=9A06E2DB143DF93AA3575AC0A96F958260 |
| 20:23.35 | Dovid | why would they do that ? gona be too smat and figure out the shinanigans that r going on ? |
| 20:26.44 | ManxPower | *** Dovid added to /ignore list |
| 20:27.01 | l2cache | Has anyone had the asterisk CLI stop responding after a while. you can get in the CLI and type commands, but they do not output any results? |
| 20:27.02 | ManxPower | go to bed, Dovid |
| 20:27.13 | ManxPower | l2cache: yes. |
| 20:27.24 | Dovid | l2cache: http://bugs.digium.com/view.php?id=11181 ? |
| 20:27.27 | l2cache | Did you figure out why it was doing that? |
| 20:27.49 | ManxPower | it was shortly before it crashed |
| 20:28.06 | ManxPower | I would reboot the server as soon as you can without disrupting call |
| 20:28.33 | l2cache | All i have to do is "service asterisk restart" and then the server will work fine for 2 - 22 days |
| 20:28.36 | l2cache | very frustrating |
| 20:29.50 | *** join/#asterisk zeniffty2002 (n=zeniffty@mail.revenueworx.com) |
| 20:30.04 | *** part/#asterisk zeniffty2002 (n=zeniffty@mail.revenueworx.com) |
| 20:30.06 | kamh | bye |
| 20:30.26 | *** join/#asterisk resin0008 (n=resin000@7.218.204.68.cfl.res.rr.com) |
| 20:30.39 | resin0008 | got question regarding hints |
| 20:30.50 | resin0008 | anybody here understand how hints work |
| 20:31.24 | resin0008 | i don't understand when the "hints" priority is executed |
| 20:31.26 | l2cache | Jerry...... |
| 20:31.32 | resin0008 | HA lol |
| 20:31.36 | l2cache | lol |
| 20:31.52 | resin0008 | nice username ll2cache |
| 20:31.57 | l2cache | its my handle |
| 20:32.17 | l2cache | What sort of problems are you having with hints? |
| 20:32.19 | resin0008 | i like it when the one person who responds is the person who doesnt know the answer to my question |
| 20:32.41 | jaytee | well hell! if I knew that I'd have chimed right in! |
| 20:32.49 | resin0008 | hahahahaaa |
| 20:32.53 | resin0008 | l2, what you doin here |
| 20:33.15 | l2cache | ^^^^^ working with 1.4 ...CLI commands do not respond |
| 20:33.22 | resin0008 | oh |
| 20:33.25 | l2cache | Dovid found the digium ticket/patch to fix though |
| 20:33.40 | resin0008 | hermn, well, for hints, here's what i have.... |
| 20:33.45 | resin0008 | Hints exist as priorities in the dialplan, so based on what I know of priorities and asterisks simple step-based diaplan concept, I imagine the following scenario: |
| 20:33.51 | Dovid | l2cache: don't think there is one at the moment |
| 20:33.53 | ThoMe | why is exten => 444,1,ChanSpy(SIP/82) wrong? |
| 20:34.02 | ThoMe | i would like SPY the chan "82" |
| 20:34.05 | l2cache | ahh, ok. Thanks Dovid |
| 20:34.07 | Dovid | l2cahce: r u using the AMI ? |
| 20:34.10 | ThoMe | any ideas? |
| 20:34.23 | l2cache | No, but I am running a few scripts that do a show channels |
| 20:34.29 | l2cache | and show queues every 20 secs or so |
| 20:34.45 | Dovid | l2cahce: the one who cubmited the bug wasnt able to re-produce it, something about his clients. read the entire ticket |
| 20:35.04 | l2cache | it looks like downgrading to 1.2 will fix it though |
| 20:35.42 | resin0008 | Why are you running scripts that show channels? don't do it that way. |
| 20:35.53 | l2cache | <arg> |
| 20:35.57 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 20:36.00 | jaytee | when does 1.4 come out of beta? |
| 20:36.02 | resin0008 | classic |
| 20:36.15 | Dovid | hehe |
| 20:36.18 | ManxPower | ThoMe: and you have an [82] section of sip.conf? |
| 20:36.39 | ThoMe | ManxPower: hm. no |
| 20:36.46 | ManxPower | ThoMe: then you can't spy it can you? |
| 20:37.01 | ManxPower | 82 is not an EXTENSION, it is a CHANNEL, and the CHANNEL is based on the SIP USER ID |
| 20:37.05 | ThoMe | ManxPower: h hmmm |
| 20:37.21 | ThoMe | ManxPower: and how i can spy extentino? |
| 20:37.25 | ManxPower | jaytee: I've been asking myself that for several years. |
| 20:37.29 | ManxPower | ThoMe: I don't know if you can. |
| 20:37.30 | jaytee | lol |
| 20:38.17 | ManxPower | ThoMe: "core show application chanspy" does not tell you something useful about spying on extensions? |
| 20:38.29 | *** join/#asterisk sniper_sniper (i=michofr@62.84.92.31) |
| 20:38.47 | sniper_sniper | Hi all...Did someone works with verso soft switch? |
| 20:38.48 | ManxPower | jaytee: It's sort of moot now, but I plan on skipping 1.4 and going from 1.2 to 1.6 if 1.6 ever becomes stable |
| 20:39.05 | ManxPower | sniper_sniper: I think you are on the wrong channel |
| 20:39.45 | sniper_sniper | ManxPower, I know but only trying to see if someone has any info about it |
| 20:39.50 | Dovid | ManxPower: I think a lot of lessons were learnt from 1.2 -> 1.4 |
| 20:40.01 | ThoMe | ManxPower: hm. have only " -- Playing 'beep' (language 'de') |
| 20:40.01 | ThoMe | " |
| 20:40.07 | jaytee | ManxPower, I'm really curious how long that will be. I'd love to be able to get rid of the sipX proxy I'm using and go native * 1.6 with SIP tcp |
| 20:40.29 | spokra | if you buy a cepstral voice (allison) for $30 do you have to buy the channels? what happens if you try to play more then one channel at a time. does it wait until the first is done or fail? |
| 20:41.19 | *** join/#asterisk PodMan99a (n=keith@77-101-121-169.cable.ubr02.maid.blueyonder.co.uk) |
| 20:41.31 | PodMan99a | hey all ... im getting this error when making calls out |
| 20:41.31 | PodMan99a | channel.c:3059 set_format: Unable to find a codec translation path from g729 to slin |
| 20:41.47 | *** join/#asterisk merlinn (n=merlin@bramble.vostron.net) |
| 20:41.48 | PodMan99a | any ideas...? |
| 20:42.06 | Dovid | PodMan99a: Do you purchase any g729 codecs ? |
| 20:42.21 | PodMan99a | no i dont want to use them at all.... |
| 20:42.42 | Dovid | dont want to use g729 ? |
| 20:43.08 | Dovid | do u have allow=g729 ins sip.conf ? |
| 20:43.12 | *** join/#asterisk smace (n=IceChat7@189.84.255.23) |
| 20:43.15 | PodMan99a | well... i dont want to buy it |
| 20:43.35 | PodMan99a | no to the allow statement |
| 20:44.20 | jaytee | quittin time, be back later from the homefront |
| 20:45.02 | PodMan99a | Dovid, would i therefor set disallow=g729 in my sip.conf then? |
| 20:46.38 | Kobaz | hmm |
| 20:46.51 | Kobaz | what does record_in and record_out do in sip.conf... i can't seem to find any docs on those |
| 20:47.02 | PodMan99a | Dovid, answered that my self.... |
| 20:47.11 | PodMan99a | although for 10$ i shouldnt moan really |
| 20:47.37 | Dovid | PodMan99a: yes |
| 20:47.53 | Dovid | PodMan99a: depends on the ammount of channles u have. i think it is worth it. lowers bandwitch a lot |
| 20:48.38 | PodMan99a | only using about 3 extensions so not really a need... but suppose would be good |
| 20:49.38 | keith4 | bandwitch? hmmm |
| 20:49.42 | PodMan99a | the translator is free though is it not? |
| 20:49.47 | keith4 | Dovid: are you IRCing from a cell phone, or something? |
| 20:51.19 | Dovid | no. just doing lots of other work. tryign to help as i am able |
| 20:51.33 | Dovid | and i have a spelling issue some times |
| 20:54.15 | smace | Hi. I'm in trouble. I try to login myself as one agent, * asks for one Agent ID and password. After it I only hear Goodbye and call is ended. But I do not see myself logged in "show agents". I am wondering what I've done wrong. Or even where I could get the error message related to this login try. |
| 20:55.35 | ManxPower | Kobaz: some guis create their own options in sip.conf |
| 20:55.47 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 20:56.40 | Kobaz | ManxPower: mmm |
| 20:57.04 | Kobaz | so it's just a custom thinger? |
| 20:57.11 | ManxPower | if the option is not listed in sip.conf.sample then it is not a valid sip.conf option |
| 20:57.17 | Kobaz | k |
| 20:57.27 | Kobaz | for some reason we have it in our configs |
| 20:57.46 | smace | where is T-KFender or something like it. |
| 20:57.49 | ManxPower | Remember, Asterisk SILENTLY IGNORES any invalid option in it's config files. |
| 20:57.56 | Kobaz | yeah |
| 20:58.16 | Kobaz | except the chan_zap module |
| 20:58.27 | Kobaz | it will complain about ignored options |
| 20:59.08 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 21:00.20 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 21:03.47 | ManxPower | That is true |
| 21:03.56 | ManxPower | well, actually maybe. |
| 21:04.07 | *** join/#asterisk s0lid (n=s0lid@124.106.141.127) |
| 21:06.57 | smace | How do I debug one Agent login ? I'm tired of that Goodbye :( |
| 21:08.24 | *** join/#asterisk andrew[andrboot] (n=andrboot@unaffiliated/andrewandrboot/x-689432) |
| 21:08.30 | Kobaz | heh |
| 21:08.34 | Dovid | smace: This is what i use http://pastebin.ca/1045402 |
| 21:08.49 | Kobaz | it would be nice if it said agent logged in rather than just hanging up |
| 21:09.16 | ThoMe | hmmis it correct? |
| 21:09.16 | ThoMe | exten => 1433461444,3,Read(pass,agent-pass) |
| 21:09.16 | ThoMe | exten => 1433461444,4,GotoIf(pass "124"]?6:5) |
| 21:09.30 | andrew[andrboot] | shiney |
| 21:13.04 | resin0008 | Asterisk is stepping through the diaplan and the agent presses #0# which parks a call in slot 1 because of my dialplan. |
| 21:13.12 | resin0008 | This will change the state of the parking space "1" from "available" to "in use". |
| 21:13.40 | resin0008 | I want to make my light for my second line key on all my phones blink by subscribing to a hint for this parking space... should be simple with hints right? |
| 21:13.43 | smace | Dovid: @ah-queue is the context ? |
| 21:14.09 | smace | Dovid: Or the name of the queue in queue.conf ? |
| 21:14.46 | Dovid | smace: name of the cotext |
| 21:14.54 | Dovid | that the call should go to |
| 21:14.59 | Dovid | have a look at the wili |
| 21:15.28 | Dovid | http://www.voip-info.org/wiki/view/Asterisk+cmd+AgentCallbackLogin |
| 21:15.52 | Dovid | i use their CID to log them in and out |
| 21:16.28 | resin0008 | dovid, is there a place in the asterisk manual that talks about hints? |
| 21:16.32 | Dovid | so fi the cid of the user is 100 asterisk will ring 100@ah-queue |
| 21:16.52 | Dovid | resin0008: have not read the new one. dont use them much to talk about em |
| 21:17.16 | resin0008 | u iknow where to get it? |
| 21:17.18 | Dovid | i know u need to set the context that u use for the hints in sip.conf and then have hint extensions for ev1 |
| 21:17.23 | ThoMe | its workd not |
| 21:17.24 | ThoMe | exten => 1433461444,3,Read(pass,agent-pass) |
| 21:17.25 | ThoMe | exten => 1433461444,4,GotoIf("${pass}" = "123"]?6:5) |
| 21:17.28 | ThoMe | always wrong pass |
| 21:17.30 | ThoMe | but why? |
| 21:17.35 | Dovid | http://h6315.com/ast_docs/Asterisk%20TFOT%20v2.pdf |
| 21:17.41 | ThoMe | if pass = 123 then goto X |
| 21:17.45 | ThoMe | Dovid: ideas? |
| 21:18.21 | resin0008 | oh yah i was looin at this the other day |
| 21:19.07 | Dovid | ThoMe: what r u trying to do ? why dont you jsut use Authenticate ? |
| 21:19.13 | ThoMe | jep |
| 21:19.21 | ThoMe | Dovid: but i would test wirh read.. how? |
| 21:20.08 | smace | Dovid: I have done the same of you and the call only hangs up. in "show agents" I still not logged in. if I've done one wrong attempt to login (I hope I've done one attempt :) it should be stored or logged somewhere. Any sugesstion of how debugging it? |
| 21:20.35 | Dovid | ThoMe: try taking out the spaces |
| 21:20.48 | Dovid | exten => 1433461444,4,GotoIf("${pass}"="123"]?6:5) |
| 21:21.14 | ManxPower | For one thing you don't have the opening $[ |
| 21:22.01 | ManxPower | It does help if you use the right syntax, see channelvariable.txt in the Asterisk source "doc" directory |
| 21:24.31 | resin0008 | ThoMe: I'm also not sure you need the "" around ${pass}" |
| 21:25.59 | ManxPower | you don't in 1.4, but it does not hurt, as long as you have them on both sides of the = |
| 21:26.16 | ManxPower | but since his syntax is totally screwed up, it doesn't matter at this point |
| 21:27.59 | resin0008 | give him the complete statement |
| 21:28.07 | resin0008 | corrected |
| 21:31.16 | smach | hey guys, is it possible to dial out from an asterisk to another SIP proxy without having sip trunks between them ? |
| 21:31.50 | resin0008 | yes |
| 21:31.57 | seanbright | except asterisk isn't a SIP proxy |
| 21:32.18 | *** join/#asterisk s0lid (n=s0lid@124.106.141.127) |
| 21:32.38 | *** part/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 21:32.38 | resin0008 | thank you for that seanbright |
| 21:32.45 | seanbright | i added no value |
| 21:32.48 | seanbright | but that's kinda my thing |
| 21:32.54 | resin0008 | Hahahahahaa allaoaolaooal |
| 21:32.58 | seanbright | is a troll |
| 21:33.03 | resin0008 | classic, awesome |
| 21:33.14 | smach | can you please tell me how, I ve been playing with line registration but cant get more than incoming calls to my asterisk !!! |
| 21:33.33 | resin0008 | it depends on the proxy you're trying to send to |
| 21:33.41 | resin0008 | is it a SIP service provider? |
| 21:33.53 | smach | it's a Mitel 3300 which talks sip |
| 21:34.28 | resin0008 | did you configure that? |
| 21:34.29 | smach | either I set up sip trunks or I use line registration between my asterisk and the 3300 |
| 21:35.23 | smach | does it make sense to try recieve inbound calls trough line registration ? |
| 21:35.47 | resin0008 | you can, but for internal traffic, a sip trunk seems logical and simple |
| 21:35.55 | resin0008 | sip trunks are really just entries in sip.conf that provide authentication and parameter controls (codecs, etc) |
| 21:36.50 | smach | I want my sip sets to be registered against the 3300 so that I could access the services of the 3300 |
| 21:37.01 | resin0008 | ooooohhhhh |
| 21:37.05 | smach | not that I think Asterisk services are not good |
| 21:37.17 | smach | just a customer requirement |
| 21:37.53 | resin0008 | and what are you using asterisk for that the mitel can't do |
| 21:38.11 | smach | Asterisk talks sip better than Mitel |
| 21:38.22 | smach | specially with cisco sets !!! |
| 21:38.41 | resin0008 | what services on the mitel are required by customer and why |
| 21:39.16 | resin0008 | obviously, im asking this because your configuration is going to be convoluted as hell if you want to use both |
| 21:40.36 | smach | all the services that can be delivered by mitel using sip should be provided to the sets by the 3300 |
| 21:40.56 | resin0008 | basically, you can have the phones register to the mitel, and then trunk to asterisk for extra feature |
| 21:41.16 | smach | Asterisk will be used for all the other features such as paging |
| 21:41.17 | resin0008 | OR, you can have the phones register to asterisk, and then trunk to the mitel for extra features |
| 21:42.28 | smach | but then, the extensions wont exist on the 3300 and the mitel ip-pbx will have no control/view on what's going on |
| 21:43.23 | smach | the product I'm working on is aimed to use the 3300 first and then switch features/services to Asterisk |
| 21:43.30 | smach | does it make sense ? |
| 21:44.41 | *** join/#asterisk smace (n=IceChat7@189.84.255.23) |
| 21:47.22 | smach | so my question is, if I have a sip set using a 7004 extension with an asterisk and asterisk registering the extension 7004 against the 3300 |
| 21:47.39 | *** join/#asterisk deeperror (n=deeperro@d149-67-253-63.try.wideopenwest.com) |
| 21:47.53 | smach | would I be able to make calls from my sets and have them forwarded (in the dialplan) to the 3300 ? |
| 21:48.41 | smach | until now I have Asterisk registering successfully the 7004 extension against the 3300 |
| 21:49.34 | smach | but whenever I make a call from my set, the invite is rewritten by asterisk before being sent to the 3300 and I get a 404 from the 3300 |
| 21:52.18 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 21:52.47 | smach | if you want to take a look the sip debug output is on: http://pastebin.com/d5f5e0b25 |
| 21:52.55 | *** join/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 21:52.56 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 21:53.47 | resin0008 | i can imagine the diaplan on asteerisk |
| 21:53.48 | *** join/#asterisk smace (n=IceChat7@189.84.255.23) |
| 21:54.45 | *** join/#asterisk qdk (n=qdk@195.242.194.41) |
| 21:54.46 | resin0008 | saying basically _700X,1,Dial(SIP/mitel330/${exten}) |
| 21:54.48 | smach | I can pastebin it too, but it's just a exten => _7XXX,Dial(${EXTEN}@3300,15) |
| 21:54.50 | resin0008 | or whatever the syntax is |
| 21:55.01 | resin0008 | right ok |
| 21:55.05 | smach | there you go I forgot to put the SIP |
| 21:55.24 | resin0008 | ok, so, then the mitel has to have a corresponding diaplan to handle it |
| 21:55.33 | resin0008 | what do you expect it to do |
| 21:56.26 | lmadsen | Dial(SIP/${EXTEN}@3300 would send to a hostname of '3300' |
| 21:56.26 | smach | well 7004 (for instance) is a registered extension, it should be able to dial 6001 (voicemail), it does when I use xlite directly with the 3300 |
| 21:57.02 | resin0008 | lmadsen, thats his mitel |
| 21:57.08 | smace | Dovid: Now after I type the password, * hangs up the call. And then I'm not logged in (again) ... any idea? |
| 21:57.34 | smach | actually I have the 3300 defined as a peer in the sip.conf |
| 21:57.46 | resin0008 | right smach |
| 21:59.12 | resin0008 | if you have xlite register to mitel as extension 7004, and then place a call to 6001 it works |
| 21:59.18 | smach | so the only pb I could think of is if the 3300 sees "the asterisk that register" and "the asterisk that sends the invite" as 2 different UA |
| 21:59.30 | resin0008 | correct |
| 21:59.31 | smach | yes it works |
| 21:59.51 | resin0008 | the registration really only says send the calls here |
| 22:00.03 | smach | oh ok, good to know |
| 22:00.33 | smach | that's why when I do it with an unregistred extension I got the same response |
| 22:01.10 | resin0008 | so in asterisk, do you have a diaplan statement for the 6001? |
| 22:01.38 | smach | oh yes sure a _6XXX dial plan the same than the one you suggeste |
| 22:01.40 | smach | d |
| 22:02.16 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:02.38 | resin0008 | so you have: exten => _6XXX,1,Dial(SIP/${EXTEN}@3300,15) .... and it don't work |
| 22:03.18 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 22:03.49 | smach | exactly and the 3300 is configured as a peer in the sip.conf |
| 22:04.12 | smach | no username or password in the 3300 section in the sip.conf |
| 22:04.13 | resin0008 | it's type=friend? |
| 22:04.34 | resin0008 | put the same username/secret that you use with the xlite |
| 22:04.37 | resin0008 | and in the registration |
| 22:04.50 | resin0008 | into sip.conf |
| 22:04.52 | resin0008 | should fix it |
| 22:05.03 | lmadsen | you realize that sending a call to a configured peer/friend in sip.conf is of the format Dial(SIP/peer_friend_in_sip_conf) yes? |
| 22:05.24 | resin0008 | theres 2 accesptable syntax |
| 22:05.30 | lmadsen | and if you want to request a certain extension from that peer, it is Dial(SIP/peer/${EXTEN}) |
| 22:05.46 | lmadsen | Dial(SIP/${EXTEN}@ip_addr) is the other format |
| 22:05.52 | lmadsen | which bypasses sip.conf |
| 22:06.23 | smach | lmadsen: I didnt know that, thx |
| 22:06.39 | resin0008 | Dial(SIP/${EXTEN}@peerinsip.conf) works as well |
| 22:07.01 | lmadsen | you're positive of that? I've never seen that format used to call a peer in sip.conf |
| 22:07.02 | smace | I have trouble to login as one agent. It just does not work and I do not get any message from asterisk. I'm not sure how to debug it. |
| 22:07.05 | resin0008 | in either case an IP Address or a peername in sip.conf are interchangable |
| 22:07.31 | resin0008 | no i'm not positive of that |
| 22:07.43 | lmadsen | now i'm curious to see if you're right... |
| 22:07.49 | lmadsen | is taking bets... with odds! |
| 22:08.01 | smace | it seems that the agent login does not succed. it should be simple, but it is not. I need some information to find out how to solve it. |
| 22:08.09 | smace | !logurl |
| 22:08.21 | resin0008 | please test that, i'm curious to hear the answer |
| 22:08.37 | resin0008 | smace, did you put the username/secret in the sip.conf a? |
| 22:09.28 | smace | resin0008: yes. |
| 22:10.23 | resin0008 | try adding authusername or whatever |
| 22:11.06 | lmadsen | wow... learn something new everyday |
| 22:11.17 | smace | resin0008: I have already. Atm I'm looking for logs of attempts to login. To find out why it does not succed. |
| 22:11.18 | lmadsen | I've been using asterisk for 5 years and never seen anyone use that format for calling a peer in sip.conf |
| 22:11.32 | lmadsen | and it apparently works |
| 22:12.36 | smach | I changed the format in the dialplan, and added a username secret to the 3300 section in the sip.conf, still have a 404 from the 3300 |
| 22:12.56 | lmadsen | what device are you calling? |
| 22:12.57 | lmadsen | a mitel? |
| 22:13.06 | lmadsen | you don't request an extension number from a phone.... |
| 22:13.15 | lmadsen | you just do Dial(SIP/mitel3300) |
| 22:13.55 | smach | lmadsen: didnt get you |
| 22:13.57 | lmadsen | Dial(SIP/peerinsipconf/${EXTEN}) is only for calling other PBXs/switches (like another asterisk box, or an ITSP) |
| 22:14.15 | lmadsen | you calling the former, or the latter? |
| 22:14.33 | lmadsen | 404 Not Found means the extension you are requesting is not available on that device/swtich |
| 22:14.35 | lmadsen | switch* |
| 22:15.11 | smach | I'm calling from 7004(cisco set) that is registred against an asterisk, and Im calling the voicemail of the 3300 => 6001 |
| 22:15.34 | lmadsen | does not compute |
| 22:15.44 | lmadsen | 3300 => 6001 ? |
| 22:16.07 | smach | sorry, 6001 is the voicemail extension on the 3300 |
| 22:16.12 | lmadsen | what is a 3300? |
| 22:16.33 | smach | the 3300 is a Mitel IP-PBX that talks sip |
| 22:17.11 | lmadsen | the 3300 requires authentication? |
| 22:17.21 | lmadsen | I'm assuming that part is working if you're getting the 404 |
| 22:17.43 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 22:17.43 | *** mode/#asterisk [+o denon] by ChanServ |
| 22:18.20 | smach | yep the 3300 does requires authentiation |
| 22:18.37 | lmadsen | if you're getting 404, then you're calling the wrong extension number on that box |
| 22:18.44 | lmadsen | 404 means, "I don't know about that extension" |
| 22:18.55 | smach | but in my case instead of getting 401 I have a 404 |
| 22:19.00 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 22:19.42 | smach | lmadsen: agree on that, but I'm calling I tried with a couple of working extension on the Mitel 3300 |
| 22:19.48 | lmadsen | that means authentication must be working. Progress is probably: INVITE --> <-- 401 INVITE (w/auth) --> <-- 404 Not Found |
| 22:20.56 | smach | The sip log and the wireshark show no 401 from the 3300, only a 100 trying and a 404 not found |
| 22:21.09 | *** join/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 22:21.09 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 22:21.19 | smach | the pastebin : http://pastebin.com/d5f5e0b25 |
| 22:21.27 | lmadsen | weird... I'd have thought SIP would have auth'd before reporting an unknown extension, but I don't have the spec memorized |
| 22:22.08 | lmadsen | more than likely a configuration error on the 3300 then -- your peer is probably not dropping into the right "context" on the 3300, and thus, the extension you're requesting isn't being found -- or your peer isn't authorized to access it |
| 22:24.12 | smach | my peer is my asterisk box, when I use the same extension 7004 on a xlite connected directly to the 3300 it does work |
| 22:24.55 | smach | with xlite: Invite --> <---401 Invite---> <---- 200 ok |
| 22:29.33 | *** join/#asterisk jets (n=brian@pdpc/supporter/active/jets) |
| 22:29.37 | ThoMe | is it posible, IF the user has forward to my voiceback, i can catch this call back? |
| 22:29.39 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 22:30.12 | Strom_M | what's a voiceback? |
| 22:30.16 | ThoMe | aaeh |
| 22:30.17 | ThoMe | voicebox |
| 22:30.21 | ThoMe | mailbox |
| 22:30.42 | Strom_M | you want to know the number the calling party called from? |
| 22:30.58 | smach | lmadsen: would the section [authentication] of the sip.conf be helpful in my case ? |
| 22:32.21 | ThoMe | Strom_M: hm. i want if you call me, and i'm not availible and you are forward to my voicebox then i Would like if you speak to my box, i in this moment call with you |
| 22:32.45 | Strom_M | grumbles about terrible English grammar |
| 22:32.51 | ThoMe | mh |
| 22:33.11 | Strom_M | you want to be able to yank the calling party out of Voicemail() and speak with them on the same call? |
| 22:34.36 | ThoMe | Strom_M: jep |
| 22:34.42 | ThoMe | Strom_M: sorry, my english |
| 22:34.44 | Strom_M | I believe that's not possible |
| 22:34.49 | Strom_M | Asterisk is not an answering machine |
| 22:34.59 | ThoMe | Strom_M: hm. ok |
| 22:35.35 | Strom_M | if callers are going to voicemail too quickly for you to be able to answer the call personally, make the call ring the station for a longer duration of time before going to voicemail |
| 22:41.30 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 22:43.31 | brendan_ | hi, i'm trying to strip the extension from a dialed number (1112223333 ext 123) |
| 22:43.55 | Strom_M | is the "extension" the number of the station placing the call? |
| 22:44.01 | brendan_ | i think i can use CUT to get everything before ext ..., but i'm not sure where to do this |
| 22:44.21 | brendan_ | no, it would be the extension of the person called |
| 22:44.25 | ManxPower | Strom_M: his phone dials numbers as the string "15045551212 ext 1234" He wants to strip off the garbage at the end |
| 22:44.46 | Strom_M | it actually dials the " ext " as part of the number? |
| 22:44.54 | Strom_M | including the spaces and the letters? |
| 22:45.37 | ManxPower | Strom_M: URL encoded spaces even |
| 22:45.52 | Strom_M | huh. |
| 22:45.57 | Strom_M | that's...um...odd. |
| 22:46.07 | ManxPower | Strom_M: one of the stupidest thing I've ever seen a phone do. |
| 22:46.14 | Strom_M | what phone is this? |
| 22:46.20 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 22:46.22 | brendan_ | its a sisco 7960 |
| 22:46.35 | nick125 | uhh |
| 22:46.39 | ManxPower | brendan_: The phone should not let you put letters in the phone number |
| 22:46.41 | Strom_M | that's odd -- ive never had my 7960s do that |
| 22:46.41 | smace | I've learned how to add verbosity to asterisk. LOL. Now I have more details. I'm getting the following error when trying to log in my agent: http://pastebin.com/m6bf3a6a6 |
| 22:46.43 | ManxPower | brendan_: and it's Cisco |
| 22:46.45 | nick125 | My 7940 doesn't do that. |
| 22:46.58 | ManxPower | I'll bet it would if you put that info into the phone number |
| 22:46.59 | brendan_ | i'm pulling the number from a service |
| 22:47.17 | brendan_ | so the ext 1234 is not typed into the phone |
| 22:47.33 | Strom_M | brendan_: sounds like You've Got Problems (tm) |
| 22:47.34 | Strom_M | anyway |
| 22:47.37 | Strom_M | use substrings |
| 22:48.04 | brendan_ | the problem is i'm not quite sure where to do it |
| 22:48.16 | Strom_M | with the EXTEN variable |
| 22:48.18 | brendan_ | i know i need to modify a variable in the dialplan, but i'm not sure what variable or where |
| 22:49.04 | *** join/#asterisk arosen (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 22:49.45 | brendan_ | isn't the EXTEN variable the current extension? |
| 22:49.59 | *** join/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 22:50.23 | Strom_M | yes, which will be whatever number the originating station dialed |
| 22:50.52 | ManxPower | brendan_: channelvariables.txt was not helpful? |
| 22:52.21 | brendan_ | ManxPower, it was |
| 22:52.46 | ManxPower | then you should be able to figure out how to remove everything except the first 11 digits of EXTEN |
| 22:54.05 | brendan_ | ahh, right |
| 22:54.39 | smace | == Spawn extension (sumicity, 2011, 2) exited non-zero on 'SIP/1024-08412c70' ... what does spawn extension means > |
| 22:57.49 | brendan_ | so, i should be able to put exten => s,1,SET(EXTEN=${EXTEN:0:11}), in my trunk statement and it will work? |
| 22:58.06 | ManxPower | NEVER set EXTEN |
| 22:58.07 | smace | please, help me login my agent, I get no error but It does not succed. http://pastebin.com/m306e1212 |
| 22:58.19 | ManxPower | You could Goto(${EXTEN:0:11},1) |
| 22:58.47 | ManxPower | smace: that message means the call ended at priority 2, extension 2011, context sumcity |
| 22:59.13 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:00.56 | smace | ManxPower: 2011 is the extension I login. I've set in extensions.conf: exten => 2011,2,AgentCallbackLogin(|${CALLERIDNUM}@sumicity) |
| 23:01.13 | brendan_ | how does Goto help here? |
| 23:01.37 | ManxPower | smace: I CANNOT help you with queues, agents, or callback |
| 23:02.02 | ManxPower | brendan_: It jumps to an extension that matches the dialed number without all the crap at the end. |
| 23:02.29 | ManxPower | also you calls will NEVER EVER match extension s |
| 23:02.36 | brendan_ | ahh |
| 23:02.48 | ManxPower | "s" means "device too stupid to send number" |
| 23:02.51 | brendan_ | i nead to read more |
| 23:02.54 | ManxPower | your device is too smart. |
| 23:02.58 | Strom_M | ~book |
| 23:02.59 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 23:03.00 | Strom_M | ~101 |
| 23:03.00 | jbot | well, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 23:03.01 | ManxPower | brendan_: yes, you do. so does smace |
| 23:03.25 | brendan_ | i tried that, but the pdf seemed corrupt, i couldn't open it in kpdf or acrobat readoer |
| 23:03.32 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-149.hsi.ish.de) |
| 23:03.32 | drmessano | ~s |
| 23:03.32 | jbot | "s" means "device too stupid to send number" |
| 23:03.34 | ManxPower | contrary to popular belief, the developers, and the docs, "s" means "stupid" not "start" |
| 23:04.12 | ManxPower | drmessano: add a note this is mainly for FXO ports (analog or T-1) |
| 23:04.20 | smace | ManxPower: I think you did not understood me. My problem is logging one agent. I've read a lot already, and also followed some tutorials. I just want to "show agents" and see my agent "logged in". |
| 23:04.41 | ManxPower | smace: I don't think you understand me. I have never used queues, I have never used agents, and I have never used callbacklogin |
| 23:04.47 | *** join/#asterisk _henrique (n=henrique@unaffiliated/henrique) |
| 23:04.47 | wonderworld | what would be a realistic cpu hardware setup for a digium card handling 30 channels? |
| 23:05.28 | wonderworld | 4 of the channels will be monitored all the time. the rest would just receive or orginate calls |
| 23:05.40 | smace | henrique: fala portugues ? :) |
| 23:05.59 | henrique | smace, falo :) |
| 23:06.36 | smace | henrique: rapaz to passando raiva com os gringos aqui. me salva ai. to tentando fazer meu agent se logar. mas ele nao se loga. e o pior nao da erro nem nada so na hora de dizer sucess ele diz hangout :( |
| 23:06.40 | wonderworld | the box would be a dedicated asterisk-box so i am wondering what kind of hardware would meet the requirements |
| 23:07.20 | smace | henrique: http://pastebin.com/m18573358 |
| 23:08.45 | henrique | smace, tem o #asterisk-br também, se ajudar em algo :) |
| 23:08.56 | *** part/#asterisk andrew[andrboot] (n=andrboot@unaffiliated/andrewandrboot/x-689432) |
| 23:09.12 | smace | henrique: vou la hehe. mas se vc tiver ideia do que to enrolado aqui, tbm vale :) |
| 23:14.10 | *** join/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 23:14.10 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 23:14.47 | *** join/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 23:14.47 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 23:21.43 | *** part/#asterisk resin0008 (n=resin000@7.218.204.68.cfl.res.rr.com) |
| 23:23.36 | brendan_ | ThoMe, i just stumbled across this, it may help you http://www.voip-info.org/wiki/view/Asterisk+tips+voicemail+live |
| 23:24.22 | ThoMe | brendan_: ah, muy bien. gracias! (very good, thank you / sehr gut, danke dir!) |
| 23:34.57 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 23:36.49 | ThoMe | brendan_: hm. its a old article, to be out of date. newsworthy is not posible? |
| 23:38.23 | *** join/#asterisk _mm_ (n=mmclain@cpe-67-49-233-178.dc.res.rr.com) |
| 23:42.15 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552398.dsl.bell.ca) |
| 23:43.32 | brendan_ | ThoMe, i just found it in the hints section, and though you'd be interested |
| 23:43.39 | brendan_ | ThoMe, i don't really know anything about it |
| 23:44.06 | ThoMe | brendan_: ok. |
| 23:44.06 | ThoMe | debian2:/usr/src/voip/misdn/mISDNuser-1_1_7_2# /etc/init.d/misdn-init start |
| 23:44.07 | ThoMe | ----------------------------------------- Loading module(s) for your misdn-cards: |
| 23:44.10 | ThoMe | ----------------------------------------- |
| 23:44.12 | ThoMe | /sbin/modprobe --ignore-install hfcmulti type=0x8 protocol=0x2,0x2,0x2,0x2,0x2,0x2,0x2,0x2 layermask=0xf,0xf,0xf,0xf,0xf,0xf,0xf,0xf poll=128 debug=0 |
| 23:44.16 | ThoMe | /sbin/modprobe mISDN_dsp debug=0x0 options=0 poll=128 dtmfthreshold=100 |
| 23:44.18 | ThoMe | [i] creating device node: /dev/mISDN |
| 23:44.21 | ThoMe | debian2:/usr/src/voip/misdn/mISDNuser-1_1_7_2# |
| 23:44.23 | ThoMe | ups |
| 23:44.26 | ThoMe | sorry, wrong channel |
| 23:59.18 | arctic_import | is native bridge a good thing? I"m attempting to take in ulaw connections and trunk them between asterisk servers trunk is using gsm. I get a "cant' native bridge" error message. what is native bridge? |