00:01.39 | jeev | if dtmfmode isn't set, what is the default method ? |
00:02.32 | chkngumbo | am i right in thinking that with asterisk i can have 4 phones talking to 4 different people through a single phone line? |
00:03.04 | jeev | that, i dont know how many channels are available.. |
00:03.10 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
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00:18.34 | jeev | [Aug 14 17:17:36] DTMF[22791] channel.c: DTMF begin ignored '4' on SIP |
00:18.35 | jeev | hmm |
00:30.02 | jeev | ok, so apparently i can't get incopming DTMF if i'm set to inband, not sure if bug or provider but provider will check it out.. i will try teliax later, so i added SIPDtmfMode(rfc2833) into the call for accepting and booya! oh, i disabled rfc2833compensate too :) |
00:30.04 | jeev | wooooooooo |
00:30.10 | chkngumbo | can anyone tell me if asterisk can accept calls on a single phone line, and carry multiple conversations over that single phone line at the same time? |
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00:34.22 | chkngumbo | i haven't seen that functionality clearly described as a function of a pbx or asterisk in particular. closest i've found is that its one of a pbx's most basic functions to "Establish connections (circuits) between the telephone sets of two users. (e.g. mapping a dialled number to a physical phone, ensuring the phone isn't already busy)" |
00:47.11 | chkngumbo | nobody? |
00:47.44 | jeev | i guess not now |
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00:52.35 | qorky | need some help please... with a legacy PABX and PRI card. |
00:53.08 | qorky | when dialing from the PABX. asterisk picks up the extension and start trying to use it before I dial all my digits on the PABX system. |
00:53.21 | qorky | I believe i need to set the TIMEOUT digit etc. |
00:53.27 | qorky | but it doesnt seem to be working. |
00:54.03 | qorky | if i use a speed dial on the PABX it works fine. I guess that is because it is passing all the digits to asterisk within the time it is expecting. |
00:54.16 | qorky | can anyone help? |
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01:02.14 | Johnakabean | hey guys the interdigit short timer and interdigit long timer are what? |
01:02.35 | Johnakabean | i know they are the timout before sending the SiP message but what is the difference |
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01:32.01 | Johnakabean | hey guys the interdigit short timer and interdigit long timer are what? |
01:32.03 | Johnakabean | i know they are the timout before sending the SiP message but what is the difference |
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01:39.39 | _-Jon-_ | Hey all, I have a call forwarding question.. |
01:40.03 | _-Jon-_ | Basically, I'm wondering it it's possible to activate call forwarding remotely |
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02:01.19 | flujan | hello guys |
02:01.31 | flujan | i am having a problem with the blindxfer and the # key. |
02:01.48 | flujan | When i use the transfer button from the softphone or the hardphone it works |
02:02.02 | flujan | but it does not work when i use the # key and type the extension/phone number |
02:02.06 | flujan | here goes the output |
02:02.14 | flujan | http://pastie.org/253375 |
02:02.16 | flujan | any ideas? |
02:05.55 | jaytee | wow, no solution in 3 minutes = quit |
02:06.06 | jaytee | with patience like that you'll never fix anything |
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03:01.29 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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03:02.28 | lmadsen | howdy! |
03:10.58 | jeev | my body shut down |
03:11.00 | jeev | i was so tired |
03:11.59 | jeev | i just woke up from a wack nap |
03:12.49 | jeev | damn |
03:12.55 | jeev | almost had a full on conversation with myself |
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03:14.43 | AndyML | What is the default behavior for a blind transfer in asterisk 1.4? does it eventually go to voicemail? |
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03:15.27 | _ShrikE | AndyML: that entirely depends on what you have your dialplan configured to do. |
03:16.33 | AndyML | ok - that is what i was thinking as I was typing the question. |
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03:18.30 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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03:22.57 | AndyML | any idea if there is a channel variable that you could use to get the channel of the device that transfered a call into a particular extension? for example - you have a custom pattern matching extension in the dialplan that rings a device for 20 seconds, then you want it to sent the call back to the person that initially transferred it (like how call parking timeouts go back to the original parker.)\ |
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03:26.46 | AndyML | oops - did anyone answer me? i got disconnected... |
03:26.54 | jeev | nop |
03:27.18 | AndyML | darn. i was optimistic... |
03:27.24 | jeev | if you transfer a call |
03:27.28 | jeev | it'll come back anyway |
03:27.31 | jeev | i dont get what you're trying to do |
03:27.44 | AndyML | why would it come back anyway? |
03:28.09 | AndyML | when you blind transfer a call, its gone. |
03:28.42 | AndyML | sure - if you do an attended transfer you can get it back anytime, but if you just want to transfer and get out, its essentially gone. |
03:29.31 | jeev | is call parking considered a blind transfer? |
03:29.33 | AndyML | these people want to transfer the call, move onto another call, and if the person they transfered it to isn't there, they want the call back, rather than it going to voicemail. |
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03:30.17 | jeev | my call parking does that |
03:30.35 | AndyML | problem with call parking is that you then have to tell the person that the call is for that someone called them and is waiting on line 701. blind transfer takes them to a different part of the dialplan - to ring a different phone for X seconds. |
03:31.05 | AndyML | but i want to configure that part of the dialplan to send the call back 20 seconds or so later - JUST like call parking. |
03:31.15 | AndyML | or rather, just like call parking's timeout response |
03:31.46 | jeev | so if you dont want to do call parking |
03:31.50 | jeev | what do you want to do? |
03:31.55 | jeev | just send it over and never see it again? |
03:31.57 | AndyML | transfer |
03:32.17 | jeev | ahh |
03:32.27 | AndyML | if you get a call on extension 101, and want to send it to the person at extension 102, you transfer it. |
03:32.47 | jeev | man, my friends office is full of the most pretentious people. if i can manage to make them work with call parking, anyone else can |
03:33.22 | AndyML | usually the dialplan is configured for that call to ring for a while, then go to voicemail. This customer absolutely demands that, instead of going to voicemail, the call ring back to 101 |
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03:33.57 | jeev | http://forums.whirlpool.net.au/forum-replies-archive.cfm/680210.html |
03:34.37 | AndyML | my thought was to create a section in the dialplan for like #XXX or 9XXX - something other than the regular extension - that if transfered to, would ring XXX for a while, then callback the person that transfered the call, but I can't seem to figure out what channel variable the original person was at to return the call to them... |
03:35.05 | AndyML | wow jeev - nice find. |
03:36.01 | jeev | hope it works for you |
03:36.32 | AndyML | that is pretty much exactly what I was looking for. Looks like instead of using a channel variable that is already defined, he creates one before making the transfer. |
03:36.43 | AndyML | thanks man. |
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03:36.56 | jaytee | AndyML, check in the asterisk docs, there is a file called channelvariables.txt that explains the use of variables in the dialplan and has a list of them towards the end of the file. |
03:37.00 | ppyy | ~centos52bug |
03:37.00 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
03:37.10 | jeev | kills jaytee with Fender's katana |
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03:37.39 | jeev | jaytee, i fixed my dtmf issues. |
03:38.42 | jaytee | thought he heard something but he must have been imagining it cuz he's dead. |
03:39.15 | jeev | takes a crap on jaytee's body to give flies incentive. |
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03:47.19 | jaytee | [TK]D-Fender, PING? |
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03:49.15 | HaMYaI | I 'm trying to detect hangups on my TDM400 + 4 FXOs using "hanguponpolarityswitch=yes ; busydetect=yes ; busycount=2 ; busypattern = 500,499 ; callprogress = yes" |
03:50.15 | HaMYaI | but it just doesn't work for me, is it because the amplitude of the hangup tones is low? |
03:50.55 | HaMYaI | I also tried busypattern = 500,500 but still doesn't work |
03:51.22 | hoegaatit | I would like to call AddQueueMember when a voip client registers with my pbx. I see you can trap the PeerStatus AMI event with PyStar and then issue an AddQueueMember call through that, but is there a cleaner 100% asterisk config way of doing this? |
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04:11.55 | jeev | ~book |
04:11.55 | 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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04:33.42 | LoOoD | There any sip tool/client I can run from the command line?.. |
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05:08.09 | oilinki | is there an restiction of the default context name? can it include dots? sip.conf:context=foo.bar.com |
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05:37.06 | route | Hi, I have a brand new PIAF install, and Asterisk isn't playing any system recordings... Well, if you stay on the line for over 4 minutes you'll hear about a quarter of a second of the main greeting, and about 4 minutes later another quarter of a second of it, and so on. |
05:39.25 | jeev | i dunno what a PIAF install is |
05:39.32 | jeev | have you watched the console with debugging on ? |
05:40.15 | route | PBX in a Flash |
05:40.20 | route | Yes I have. |
05:40.32 | route | It says it's playing, but it doesn't play correctly. No errors. |
05:41.21 | route | Files are .wav recorded at 8k 16bit just like they should be. In fact, they are the exact same files from our old Asterisk box. |
05:41.37 | route | THey worked fine on there... Even the pre-recorded system sounds don't play back correctly. |
05:43.14 | jeev | so you've checked the codec |
05:43.16 | jeev | it's the same codec? |
05:43.28 | jeev | or at least the proper codec? |
05:43.47 | route | Where do I check that? |
05:43.49 | jeev | pastebin.com your allow and disallow |
05:43.53 | jeev | show me your sip.conf in pastebin.com |
05:43.59 | jeev | hide your passwords and crap |
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05:44.43 | route | Uhm, every line in sip.conf is commented out. |
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05:45.27 | route | sip_additional.conf has a lot of stuff in it, but I can't edit that with FreePBX. It doesn't list that file at all. |
05:45.51 | jeev | grep allow sip_additional.conf |
05:46.00 | jeev | pastebin.com |
05:46.01 | jeev | dont paste it here |
05:46.44 | route | disallow=all, and allows are ulaw alaw gsm and g729... Not a paste, I typed that. :-P |
05:47.11 | jeev | find your sounds directory and see what files are in it |
05:47.12 | route | All are there twice. |
05:47.21 | jeev | what are the extensions |
05:47.22 | jeev | are you behind nat ? |
05:47.36 | route | sounds directory has all the pre-recorded stuff, sounds/custom has the 2 custom greetings we recorded, and that worked on the old box. |
05:47.50 | route | in sounds/custom there are closed.wav and main.wav |
05:48.13 | route | The box isn't behind a nat, but the phones are. |
05:48.13 | jeev | what are the extension |
05:48.15 | jeev | extensions |
05:48.22 | route | .wav .wav |
05:49.07 | jeev | hmm, i'm not too sure how this works.. but when i was having audio issues.. i had only gsm files in my sounds directory and ulaw was passing it.. so it was sounding nasty |
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05:49.17 | jeev | so installed the sounds and got the .ulaw extension sounds and all is well |
05:50.55 | route | I don't understand why the pre-recorded sounds won't work on a fresh install though. |
05:51.07 | jeev | well, i had built mine from source and had only installed .gsm and i didn't realize that |
05:51.24 | route | I even did update-scripts and update-fixes |
05:51.34 | jeev | well, i dunt know how that works.. |
05:52.13 | jeev | asterisk-core-sounds-en-ulaw-1.4.9.tar.gz |
05:52.15 | jeev | that may be your issue man |
05:52.21 | route | ok, my custom recordings are .wav, but the rest of the sounds are .gsm. I just looked at them again. I thought it had both in there. |
05:52.25 | jeev | i'm not that great with asterisk.. but look for that and empty it into the sounds folder |
05:52.39 | jeev | debug it and see what codec it's playing it at |
05:52.48 | jeev | i'm sure ulaw would play wav though.. that's how my voicemail is recorded |
05:53.45 | route | -- <Local/1908XXXXXXX@from-sip-external-1ed9,2> Playing 'custom/closed' (language 'en') |
05:54.58 | jeev | ah |
05:55.03 | jeev | so it's your custom that's being stupid ? |
05:55.11 | jeev | not the default sounds? |
05:55.41 | route | No, all sounds. |
05:56.03 | route | -- <SIP/1337-0c98d190> Playing 'ss-noservice' (language 'en') |
05:56.28 | route | -- Executing [s@from-pstn:4] Playback("SIP/1337-0c98d190", "ss-noservice") in new stack |
05:57.17 | route | After 1 minute and 7 seconds it said "The Nu" |
05:57.38 | route | I believe it's supposed to say "The number you have dialed....." |
05:57.53 | jeev | just *try* getting those files, untarring it in the folder |
05:57.55 | jeev | it wont hurt. |
05:58.06 | jeev | i'm in no position to help anyone anyway but that's what my sound issue was. |
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06:03.31 | route | ok, ulaw files are in the directory now. |
06:04.39 | route | Do I need to change anything anywhere for it to use those files? |
06:06.08 | HaMYaI | I've unsuccessfully detected hangups on my TDM400 + 4 FXOs using "hanguponpolarityswitch=yes ; busydetect=yes ; busycount=2 ; busypattern = 500,500 ; callprogress = yes" |
06:06.51 | HaMYaI | what else should I consider? |
06:11.47 | route | jeev ? |
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06:13.07 | jeev | ah sorry |
06:13.09 | jeev | well |
06:13.19 | jeev | just change a sound to that |
06:14.14 | jeev | try setting one to 'privacy-your-callerid-is' |
06:14.16 | jeev | and see what happens |
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06:15.41 | jeev | or vm-goodbye |
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06:18.46 | route | Still not playing. |
06:19.26 | jeev | but it shows it ? |
06:19.41 | jeev | sip debug and pastebin the whole thing |
06:19.50 | jeev | including the call |
06:23.32 | route | genzaptelconf -svdM |
06:23.41 | route | That's all I needed to do to solve the problem. |
06:23.56 | route | Now all sounds seem to be playing correctly. YAY! |
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06:27.06 | jeev | oh heh |
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06:27.53 | jeev | sleep time. night |
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06:33.11 | pputman | Has anyone heard of the G.CLEAR codec, and know if it has another name that would be supported by asterisk? |
06:33.42 | Corydon76-dig | pputman: sounds like a marketing gimmick |
06:34.17 | pputman | Corydon76-dig, a customer is telling me it's the codec used for an ISDN line, so I assumed he meant ulaw, but now I'm thinking he means the equivalent of setting clear=1-23 in zaptel.conf |
06:36.12 | Strom_M | lol. |
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06:38.26 | Corydon76-dig | ulaw, alaw, or G.721 (ADPCM) |
06:39.02 | Corydon76-dig | Those are the only codecs specified in the ISDN spec |
06:39.05 | pputman | right, I think he's just confused though |
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07:42.13 | obnauticus | Corydon76-dig: >:| |
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07:47.42 | dominic1 | hi folks |
07:48.57 | dominic1 | can anybod tell me why the second channel in the asterisk parkaction ist not dialed after giving up the call? |
07:52.23 | *** join/#asterisk E-bola (i=psybnc@ip181.rev112.brygge.net) |
07:52.45 | E-bola | Where there any big changes in regards to conferences/meetme from Asterisk 1.2 to 1.4? |
07:53.40 | poller | Is there any cmd that lets me listen in on ongoing calls? |
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08:04.16 | E-bola | Hmm can you change the pincode for a meetme room with a command? |
08:07.25 | mort_gib | E-bola: not without reloading |
08:08.15 | mort_gib | You want to write a small ivr that only allows the user to enter based on a key fetched from a database |
08:08.16 | E-bola | Hmm i just found the Web-MeetMe util |
08:08.22 | E-bola | that seems to make it possible, via the web |
08:08.37 | E-bola | mort_gib: Ahh your right, that would work just as well |
08:09.10 | E-bola | I was just a bit worried that if the password stayed the same always, anybody who had ever been on a conference could join any future conferences |
08:09.35 | mort_gib | Yes that is true... |
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08:10.58 | mort_gib | You have different options, a perl script that changed the pwd every night, and reloads, finally sending the new pwds to a reception... |
08:12.07 | E-bola | Yep, but web meetme seems as the most userfriendly |
08:12.13 | mort_gib | sure :-) |
08:12.16 | E-bola | if it works etc.... |
08:12.26 | E-bola | mort_gib: have you ever tried it? Web meetme |
08:12.32 | mort_gib | Yeah,. but you want it to be automatic right |
08:12.51 | mort_gib | I must admit I haven't had a look at web-meetme |
08:13.01 | E-bola | Nah, optimally the conference organiser can set it up before the conference and give it to the attendees |
08:13.13 | E-bola | Which seems preicely what web meetme allows you to do |
08:13.39 | mort_gib | So after the conference has stopped the password is revoked?? |
08:15.51 | E-bola | The whole conference room is deleted |
08:16.11 | E-bola | or maybe the password is just left untill next time a room is needed, and its changed then |
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09:06.09 | HaMYaI | in the Asterisk's Makefile, there's an option BUSYDETECT+= #-DBUSYDETECT_COMPARE_TONE_AND_SILENCE |
09:06.50 | HaMYaI | I tried to uncomment that and compiled but it shows errors |
09:10.44 | dominic1 | hi, short question. Why do I need the extension on the manager redirectaction, when I bridge two channels? |
09:12.27 | _E-bola | mort_gib: You still here? |
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09:13.46 | mort_gib | Yeah, I'm here :-) |
09:14.26 | _E-bola | I drove out to the client needing the conference setup. And they actualy had something else in mind. The boss wants to be able to dial each participant in the conference 1 by one to "let them into the room" |
09:14.35 | _E-bola | IS this sort of setup possible at all with asterisk? |
09:17.14 | mort_gib | Eh, my clients do that, they ask the receptionist to do it :-) |
09:17.28 | mort_gib | -You want to have a look at creating dial files... |
09:17.34 | dominic1 | how can I redirect a parked call to my current active call? |
09:17.43 | dominic1 | does anybody know that? |
09:18.08 | mort_gib | dominic: you want a three way join?? |
09:18.13 | _E-bola | mort_gib: no i think u missunderstand. Lets say u got a conference going with 2-4 people |
09:18.46 | _E-bola | mort_gib: And suddenly need the advice of a 5'th person. How can you ring this person, and get him joined to the conference call? |
09:18.51 | dominic1 | no I want that my call to a specific person is hung up and the person I called to speaks to a person which was in parkposition |
09:18.54 | mort_gib | Yes, so when J.Doe.1 calls 8000 (meetme) * automatically calls 4 other numbers and join them to 8000 |
09:19.27 | dominic1 | bridgeaction is only available in 1.6 |
09:19.35 | dominic1 | is there anything else in 1.4 |
09:19.59 | _E-bola | mort_gib: cant it be done more manualy? Like dial something while in the room, and then you "privately" talk to the 5th guy, and if he agrees to join the room. You press a button and he's joined to the room |
09:20.51 | mort_gib | Ah, You could 1. Leave the conference, call the 5th participant, transfer to the conf romm and join yourself 2. put the conf on hold on your handset, call 5th and hava a little prep talk transfer to meetme and go back to meetme |
09:21.28 | _E-bola | ahh right |
09:21.47 | _E-bola | thats what i want. What happens if you transfer a call to a conf room? They get asked for the pin code or? |
09:22.59 | mort_gib | Normally I don't put PIN codes on conf rooms |
09:23.25 | _E-bola | I guess if i set it up so all conf rooms are only accisible locally i dont have to either |
09:23.35 | _E-bola | Great, thanks alot for your help mort_gib :) |
09:23.41 | mort_gib | But if you do the person calling the conf room gets the request, in this case the you would get the request as you are dialing BEFORE transferring |
09:24.01 | mort_gib | ;-) Your welcome, where are yo in the world?? |
09:24.12 | _E-bola | Denmark (North europe) |
09:24.32 | obnauticus | drmessano: are you there -- honey? |
09:24.41 | mort_gib | -This is why my clients ask their PA's or reception staff to handle conf calls |
09:24.51 | mort_gib | -Yeah :-) Hvordan er vejret?? |
09:24.58 | _E-bola | lol |
09:25.12 | _E-bola | Fint fint Morten :P |
09:25.20 | mort_gib | Her er det 30+ |
09:25.28 | _E-bola | bastard :) |
09:25.32 | _E-bola | Where's here? |
09:25.43 | mort_gib | Sorry... |
09:28.22 | mort_gib | Gibraltar/Spain |
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09:49.47 | obnauticus | bridgeaction? |
09:49.50 | jazzmann | Unable to read config file mobile.conf I am trying to add chan_mobile |
09:49.50 | obnauticus | what is this business dominic1 ? |
09:50.04 | obnauticus | jazzmann: have fun with that, i hate chan_mobile, it was mean to me :| |
09:50.07 | obnauticus | crosses arms |
09:51.16 | jazzmann | http://www.geek-pages.com/articles/asterisk/howto_build_and_configure_chan_mobile_on_trixbox.html |
09:51.17 | jazzmann | this is the tutorial I was following in mandriva |
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09:52.09 | jazzmann | I really need some help in installing it |
09:59.03 | dominic1 | obnauticus: it's atxfer but initiated by a software |
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11:23.50 | mocker | I'm having a problem with a Rhino channel bank that I'm getting configured. |
11:24.12 | mocker | I have the T1 sending signal and framing is good, but when I press digits on the phone nothing seems to be detected. |
11:24.28 | mocker | So.. it's analog phone connected to channel bank which is connected to asterisk |
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11:38.00 | bartpbx | hello |
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12:01.47 | J4zen | Does anyone happen to have implemented HylaFax into Asterisk? |
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12:01.58 | bartpbx | ys |
12:02.11 | J4zen | Did you ever get an error such as "Failure to train remote modem at 2400 bps or minimum speed" ? |
12:02.15 | J4zen | resulting in fax failure |
12:02.24 | J4zen | i cant recieve faxes because of that |
12:02.31 | bartpbx | no, never seen this before |
12:02.38 | bartpbx | but looks like a problem on the remote side |
12:02.53 | J4zen | yes, but in this case the remote modem is my PBX |
12:05.52 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
12:07.47 | [netman] | so , u should try to send a fax to a different number to check it out |
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12:10.12 | Wayhigh | sup all |
12:11.56 | J4zen | [netman]: Yeah i have, outgoing fax works fine to all numbers |
12:12.07 | J4zen | its the incoming faxes on my PBX that don't work |
12:12.27 | J4zen | if someone faxes to me, in this case i used HylaFax to fax myself.. they will get a repsonse stating "Failure to train remote modem at 2400 bps or minimum speed" |
12:12.40 | J4zen | so outgoing works just fine |
12:12.43 | J4zen | incoming is broken |
12:14.38 | mocker | J4zen: Welcome to faxing w/ asterisk :) |
12:16.07 | [TK]D-Fender | J4zen: incoming and outgoing over WHAT? And last I recall 9600 was the minimum, no? |
12:17.08 | mocker | opens a support ticket w/ Rhino. |
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12:22.18 | keith4 | 9600 used to be the maximum |
12:22.28 | keith4 | i hear tell of some fancy ones that can do 14400, though |
12:22.48 | J4zen | [TK]D-Fender: In more detail > Fax going out through SIP-trunk(sending) -> PSTN -> SIP-trunk(recieving) -> PBX -> Inbound Route -> HylaFax |
12:23.03 | keith4 | ugh |
12:23.11 | keith4 | it's a miracle it works at all |
12:23.32 | [TK]D-Fender | keith4: 33.6kbps |
12:23.35 | J4zen | it sounds edgy, it might just be due to the fact that im both sending and recieving the fax on the same SIP-trunk |
12:23.44 | [netman] | J4zen: so *everybody* who wants to fax you generetes that error msg, I see |
12:23.46 | keith4 | [TK]D-Fender: incredible. soon we'll have flying cars! |
12:24.15 | keith4 | (shows when the last time I used a modem was, I guess) |
12:24.24 | Wayhigh | ?? P-Asserted-Identity |
12:24.27 | [netman] | J4zen: hylafax doesn't see a SIP trunk..., only a iaxmodem, I don't think that could be the reason |
12:24.38 | J4zen | [netman]: Well i tested it on one external analogue fax machine and from my own SIP-trunk.. seeing as we're not activley using FAX thru VoIP right now; Yes, *everyone*. |
12:24.52 | J4zen | i see |
12:25.16 | J4zen | ill run some more tests, gotta love faxing |
12:25.49 | Wayhigh | is still looking for multiple trunks that pass P-Asserted-Identity |
12:26.30 | [TK]D-Fender | Wayhigh: flowroute.com |
12:26.59 | Wayhigh | yeah.. I know.. I want to find more of them that pass it |
12:27.01 | keith4 | is that the one that was used to demonstrate "unmasking" hidden CID? |
12:27.34 | [TK]D-Fender | keith4: Yup. All the kiddies want to be the shiznit y0 |
12:27.41 | Wayhigh | I figure they can't be the only one and it is an interesting technique. I know that a number of the wholesalers are requiring p-asserted-identity headers now |
12:28.15 | keith4 | I just don't accept private calls. problem solved |
12:28.34 | Wayhigh | I want it for something different though.. I'm testing something I discovered yesterday |
12:28.56 | keith4 | [TK]D-Fender: maybe someday I'll get a flowroute.com account, so I can h4x CID too! |
12:29.03 | [TK]D-Fender | Wayhigh: Namely? |
12:30.26 | Wayhigh | fender: a method of screenpopping |
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12:32.10 | [TK]D-Fender | Wayhigh: ummm, you want to get what from the telco exactly? |
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12:37.01 | *** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv) |
12:37.07 | aiksa[LV] | hi everyone. |
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12:37.36 | aiksa[LV] | where could I find full list of asterisk events (sent to manager interface) |
12:37.51 | [TK]D-Fender | ~book |
12:37.51 | 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 |
12:37.53 | [TK]D-Fender | aiksa[LV]: ^^^^^^^^ |
12:38.22 | aiksa[LV] | [TK]D-Fender: hi. somehow i overlooked that |
12:38.44 | aiksa[LV] | http://tfot.leifmadsen.com,thanks |
12:38.54 | [TK]D-Fender | aiksa[LV]: thats like asking where you jacket is with your eyes closed. |
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12:45.30 | aiksa[LV] | :) |
12:46.11 | aiksa[LV] | iiiiihj |
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12:51.53 | aiksa[LV] | thats more like looking for the glasses when you have them on your eyes already |
12:52.09 | aiksa[LV] | [TK]D-Fender: thanks nevertheless |
12:52.29 | [TK]D-Fender | aiksa[LV]: I was being kind... |
12:53.42 | aiksa[LV] | i just somehow thought that was not covered in the book |
12:54.35 | aiksa[LV] | dont know why |
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13:02.17 | zamba | what can i do on my router (software router running linux) to give voip higher priority? |
13:02.27 | zamba | i'm talking about the outgoing stream now |
13:02.49 | zamba | i'm perfectly aware there's little i can do with throttling the incoming |
13:03.24 | [TK]D-Fender | zamba: http://www.google.ca/search?hl=en&q=linux+bandwidth+management&btnG=Google+Search&meta= |
13:04.40 | zamba | [TK]D-Fender: but what software tool do you guys recommend? |
13:05.02 | zamba | i've heard about wondershaper |
13:05.13 | zamba | but i need something -simple- to set up |
13:05.28 | zamba | i see m0n0wall, but that's a complete distribution, isn't it |
13:05.29 | zamba | ? |
13:05.39 | [TK]D-Fender | zamba: I have also heard of Wondershaper, and it was simple when I looked at it back then |
13:05.42 | mocker | zamba: Outgoing to the Internet? |
13:05.45 | zamba | mocker: yeah |
13:05.52 | [TK]D-Fender | zamba: Yes, m0n0wall is a distro |
13:05.59 | mocker | So high priority until it leaves your home/office, then nobody cares. :) |
13:06.00 | zamba | so that's not an option |
13:06.25 | zamba | mocker: yeah, exactly.. but i need a way to shape it on my router and am thus looking for a software package that can help me fix this |
13:07.32 | keith4 | monowall is a mean and lean little bugger |
13:07.44 | zamba | well, reinstalling the router isn't exacly an option |
13:07.50 | keith4 | i had it running for years, on an 8M kodak CF card from an ancient digital camera |
13:08.27 | luxxx77 | Hello! Can anyone tell me if i can configure asterisk 1.4.21.2 with an SVN install of asterisk-gui/branches/2.0/? Is 2.0 compatible with 1.4.? |
13:08.32 | zamba | especially not since the router is around 400 km away from me :) |
13:08.41 | zamba | keith4: i'm running the router on a 1 GB CF atm :) |
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13:08.48 | zamba | keith4: did you use flashybrid then? |
13:08.54 | keith4 | what what? |
13:09.11 | zamba | keith4: to prevent writes to the flash? |
13:09.25 | keith4 | monowall doesn't really work that way |
13:09.34 | zamba | how does monowall work? |
13:09.37 | keith4 | it only writes if you change the config |
13:09.44 | keith4 | (as long as you log so a syslog server) |
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13:10.09 | zamba | ah, so it's embedded per design? |
13:10.13 | keith4 | yep |
13:10.15 | zamba | cool |
13:10.40 | keith4 | doubtful that it will help your issue. but, whatever |
13:11.28 | luxxx77 | does somebody konw here a bit about asterisk-gui? |
13:13.44 | keith4 | points at the topic |
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13:24.56 | Cheap_tdm400 | :) |
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13:26.47 | JenniferAkemi | Morning! |
13:30.15 | mgroman | omg hi2u!!!! |
13:31.13 | [TK]D-Fender | Wayhigh: Just F'N ebay it |
13:31.33 | Wayhigh | fender: I may do that.. I just didn't wanna go through that kinda hassle :P |
13:31.47 | mgroman | ... stoner ... |
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13:34.36 | Wayhigh | I think I may sell my S100think I may sell a S100-FX ATA as well |
13:34.45 | Wayhigh | err.. heh.. |
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13:40.11 | aiksa[LV] | [TK]D-Fender: are you sure The Book had a list of asterisk manager events? |
13:40.25 | [TK]D-Fender | aiksa[LV]: Yes. |
13:40.45 | aiksa[LV] | all i found was manager actions |
13:42.28 | [TK]D-Fender | aiksa[LV]: And have you tried the WIKI? |
13:42.29 | aiksa[LV] | the unlink as a word was mentioned in three places in the book, but none of them was even closely related to that specific event |
13:42.37 | aiksa[LV] | you mean voip-info? |
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13:43.26 | [TK]D-Fender | aiksa[LV]: Clearly. |
13:43.32 | aiksa[LV] | of course I have. blame on me - voip-info is the first resource i go to for a quick reference |
13:43.46 | aiksa[LV] | but I have a suspicion that their event list is not complete |
13:43.50 | [TK]D-Fender | aiksa[LV]: Seems to be a list there. |
13:44.08 | [TK]D-Fender | aiksa[LV]: Why don't you leave a listener on AMI and log the events it sees? |
13:44.55 | aiksa[LV] | [TK]D-Fender: what I was looking for - a documentation of specific events and circumstances under which they are triggered |
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13:45.31 | aiksa[LV] | i have already logged the events and i am not sure about one thing - thats was reason why I wanted to read up on it |
13:46.03 | [TK]D-Fender | aiksa[LV]: Maybe you should have come right out with this specific thing in the first place. |
13:46.50 | aiksa[LV] | [TK]D-Fender: I will come out - but I first wanted to know if there is any other information resource which I didnt consult in the first place |
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13:48.44 | aiksa[LV] | [TK]D-Fender: the question was - If I have a queue served by a couple of SIP extensions as queueMembers. then why doesnt dial event happens before link event for an incomming call from queue |
13:49.37 | aiksa[LV] | and are there alternatives to detect that specific call member have been called by a person waiting in the queue. |
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13:50.14 | Iptime | hi |
13:50.32 | aiksa[LV] | as far as I can see for now - the Dial event gets launched only when there is a Dial command in the dialplan, but not in the applications |
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13:52.39 | [TK]D-Fender | aiksa[LV]: Guess thats just the way it is.. |
13:53.11 | Iptime | hello |
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13:54.22 | *** part/#asterisk luxxx77 (n=luxxx77@f051064033.adsl.alicedsl.de) |
13:54.45 | Iptime | I am very new to the Asterisk IP telephony. i got a small and stupid question like you can say, boff my question is simple, i got asterisk server at home, and create lots of sip extensions, the problem is all my sip extension users not "online" all the time, if i want to join one my sip user, i have to check each time from my server by doing this command line : " sip show peers " and from... |
13:54.46 | Iptime | ...my work i can't check if the user is online or not, to understand well my question, i just need the same functionality as skype,( when a user come online u heard a bip ) like i want my phone ring twice time. In hope of gotting best answer here |
13:55.45 | [TK]D-Fender | Iptime: Go read about "presence" on the WIKI. |
13:55.48 | [TK]D-Fender | ~wikis |
13:55.49 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
13:55.50 | [TK]D-Fender | ^^^^^^^^^^^ |
13:56.36 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
13:57.50 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
13:58.08 | Iptime | is it this one the answ<wer ? |
13:58.09 | aiksa[LV] | [TK]D-Fender: - so you dont know of any other source for more detailed explanation on asterisk manager interface events? |
13:58.10 | Iptime | http://www.voip-info.org/wiki/view/Asterisk+presence |
13:58.31 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
13:58.37 | [TK]D-Fender | aiksa[LV]: Probably on Mantis where it is explained. Just Goolge whatever evernt you're wondering about. |
13:59.12 | [TK]D-Fender | Iptime: Does it LOOK like what you want? |
13:59.33 | *** join/#asterisk luxxx77 (n=luxxx77@f051064033.adsl.alicedsl.de) |
14:00.23 | Iptime | what i want is exactly same as msn messenger |
14:00.39 | Iptime | when a user present on my server |
14:00.53 | Iptime | want to have alert |
14:01.28 | *** join/#asterisk oilinki (n=oil@ppp-124-120-251-104.revip2.asianet.co.th) |
14:03.52 | [TK]D-Fender | Iptime: use presence on a soft-phone, or FOP, or any of the other pile of status viewers out there. |
14:06.36 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
14:15.34 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
14:18.02 | mgroman | Do people still use nano to code at digium? |
14:18.18 | Nugget | oh dear. |
14:18.54 | Nugget | http://macnugget.org/photos/strange/curves |
14:19.04 | mgroman | virus? |
14:19.08 | [TK]D-Fender | fires up the old flammenwerfer |
14:19.13 | russellb | has never used nano to code |
14:19.13 | russellb | heh |
14:19.28 | russellb | points at Qwell |
14:19.30 | russellb | only he does |
14:19.48 | russellb | most are using vim or emacs ... |
14:19.55 | mgroman | I remember a long time ago, like 6 months, one of the admins here suggested I try vim, and i have been using it since, and im starting to get fast at it, so whoever it was, thanks |
14:20.01 | Nugget | russell uses a magnetized needle and a steady hand. |
14:20.08 | russellb | <3 vim !!! |
14:20.13 | *** join/#asterisk km- (n=pgrace@fmservices.v6.fierymoon.com) |
14:20.15 | d3wayne | Qwell is silly with his nano |
14:20.29 | Nugget | using nano to code is like using scissors to mow the lawn. |
14:20.37 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
14:20.44 | Nugget | sure, the UI is simple, but it's certainly not easier. |
14:20.49 | km- | ooh did I join into a vi/nano penis contest?! |
14:20.51 | *** join/#asterisk ManxPower (n=manxpowe@45.sub-75-248-217.myvzw.com) |
14:20.54 | russellb | km-: yes |
14:21.10 | km- | lemme throw it in that I love nano until I need vi's feature set. |
14:21.20 | russellb | alright fence sitter |
14:21.22 | Iptime | thcs Tk |
14:21.26 | russellb | :-p |
14:21.29 | Nugget | heh |
14:21.45 | Iptime | the problem is : i use linksys adaptor |
14:21.56 | Iptime | and there there is no that option |
14:22.03 | seanbright | emacs <3 |
14:22.29 | seanbright | i have to admit though... the *only* reason i don't use vim is because... |
14:22.31 | russellb | emacs is for noobs |
14:22.33 | km- | russel: gotta play the politics, man! |
14:22.35 | seanbright | i have a penis and testicles |
14:22.40 | Iptime | ok |
14:22.42 | russellb | km-: not on IRC you don't! |
14:22.48 | Iptime | i got an other question |
14:22.55 | mgroman | is sorry for starting a text editor crusade |
14:23.04 | russellb | mgroman: ;) |
14:23.04 | km- | russellb: bah! |
14:23.06 | Iptime | i got an asterisk server |
14:23.21 | km- | Me too!!! |
14:23.27 | russellb | doesn't |
14:23.28 | russellb | :( |
14:23.34 | Iptime | is it possible to listen what r they talking using my server for example ? |
14:23.43 | km- | In fact, I'm about to have an entire ESX cluster of them |
14:23.49 | russellb | Iptime: *CLI> core show application ChanSpy |
14:23.53 | [TK]D-Fender | Iptime: "core show applications like spy" |
14:24.05 | russellb | awww, [TK]D-Fender had to be more complete than me |
14:24.07 | russellb | I see how it is |
14:24.22 | km- | that was pretty sneaky with that "like" thrown in there. |
14:24.22 | seanbright | ExtenSpy is for noobs |
14:24.33 | Iptime | my question is : asterisk is it secure or not .?4 |
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14:24.48 | km- | Yeah, so get this. I'm getting promoted at my company to R&D, I get to hack on asterisk all day every day! |
14:24.49 | russellb | Iptime: depends how you set it up, and what you mean by secure, i guess |
14:24.50 | [TK]D-Fender | seanbright: Yeah and its not like we have any of THOSE around here.... |
14:24.56 | seanbright | [TK]D-Fender: :) |
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14:25.32 | Iptime | is it possible to listen what ppl r talking ? |
14:25.32 | km- | now the question is whether I can give patches back to the community for stuff I write. |
14:25.44 | Iptime | with 2 extension ? |
14:25.53 | russellb | damnit, Iptime, two people already answered that question |
14:25.57 | russellb | go read about what those apps do |
14:26.11 | russellb | km-: a question of your employer, you mean? |
14:26.18 | km- | yeah. |
14:26.24 | russellb | km-: well I sure hope so. :) |
14:26.33 | Iptime | if i type this from asterisk cli |
14:26.34 | km- | I'm already signed up as a committer for asterisk, the question is whether I'll be allowed to commit it |
14:26.36 | Iptime | core show applications like spy |
14:26.37 | seanbright | most employers have this nagging concept of "intellectual property" |
14:26.39 | russellb | km-: If you have any questions about how the process works, feel free to contact me directly. russell@digium.com |
14:26.52 | Iptime | it can only show they are online or not |
14:26.54 | russellb | already signed up as a committer? |
14:26.56 | km- | russellb: I think I was actually one of the first couple hundred people to sign mark's agreement |
14:27.02 | [TK]D-Fender | Iptime: READ AGAIN |
14:27.08 | Iptime | my question is how to pick a line and listen |
14:27.08 | km- | russellb: you know the, disclaim all rights to what you submitted |
14:27.16 | Iptime | what they r talking |
14:27.21 | russellb | km-: Oh, that. Yeah, we have a new license agreement in place now that is electronic. |
14:27.23 | km- | russellb: they were doing that wayyy back in the day |
14:27.29 | [TK]D-Fender | Iptime: And we just gave you the command list to use to DO IT |
14:27.29 | russellb | nods |
14:27.30 | km- | oh, yeah, I had to fax mine in |
14:27.45 | russellb | km-: You'll have to electronically agree to the new one via your account on bugs.digium.com |
14:27.53 | km- | ahhh |
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14:27.58 | km- | lemme go see if I remember my password |
14:28.09 | russellb | km-: once you log in, you should see "Sign License" at the top, I think |
14:28.14 | Iptime | logically for me there must be have a number to connect the line |
14:28.15 | km- | I havent had much time to hack lately, my job has been 60 hours a week of operations |
14:28.38 | russellb | km-: if you get into coding, hang out in some of the dev channels ... #asterisk-dev, -bugs, -commits |
14:28.38 | Iptime | and the command not help |
14:29.11 | [TK]D-Fender | Iptime: Go read each applications instructions |
14:29.12 | km- | russellb: yeah, I occasionally pop into -dev to harass corydon, et al |
14:29.15 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
14:29.19 | russellb | cool. |
14:29.44 | km- | wow, up to 11 windows in irssi heh |
14:29.59 | km- | hmm, how do I hit alt+11. :P |
14:30.04 | Dr-Linux|home | I'm using Queues with asterisk, but i don't want it should play the message "You are caller number 2 ..." what should i do to disable on this message? |
14:30.11 | Dr-Linux|home | i mean position |
14:30.19 | Iptime | ok thx |
14:30.19 | km- | there's probably a flag on the app for it |
14:30.29 | Nugget | alt-q through alt-p will get you windows 11-19 or whatever. |
14:30.35 | km- | nugget: hot! |
14:30.45 | km- | nugget wins for fun fact of the day |
14:30.51 | Dr-Linux|home | km-: talking to me? |
14:31.07 | km- | dr-linux: yeah, I'd have to assume there's gotta be an option on the app for it. |
14:31.23 | km- | since way back it didn't used to do that, so it'd be a feature you should really allow to be turned on and off |
14:31.30 | Dr-Linux|home | km-: yeah but i'd like to know that option |
14:32.01 | Dr-Linux|home | km-: I made search on the web, but can't get appropirate solution for get if off |
14:32.07 | Dr-Linux|home | s/if/it |
14:32.38 | km- | curious |
14:32.53 | [TK]D-Fender | Dr-Linux|home: Read the sample config. |
14:33.17 | Dr-Linux|home | [TK]D-Fender: already did |
14:33.37 | Qwell | russellb: pfft, you kids these days and your regex and syntax highlighting |
14:33.37 | [TK]D-Fender | Dr-Linux|home: then keep reading it till your eyes bleed |
14:34.11 | seanbright | nano has syntax highlighting, hater. |
14:34.49 | Dr-Linux|home | [TK]D-Fender: hhm... can you give me hint for that option where i can disable the position? |
14:35.26 | km- | [tk]d-fender: are you referring to voip-info or something? I'm curious on this now and want to read up :) |
14:35.29 | file | Dr-Linux|home: http://pastebin.com/m3f919d14 copy and pasted right from the queues.conf.sample file |
14:35.54 | [TK]D-Fender | Dr-Linux|home: READ THE DAMN SAMPLE CONFIG FILE |
14:35.57 | km- | file: wow, that is extremely obvious. |
14:36.05 | km- | dr linux should have his phd stripped |
14:36.22 | seanbright | invents a way to stab people in the face over the internet |
14:36.49 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
14:37.10 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
14:37.21 | [TK]D-Fender | seanbright: License it to me! |
14:37.30 | km- | yeah I'd totally license that patent. |
14:37.44 | seanbright | on a per use basis |
14:37.46 | seanbright | :) |
14:37.51 | km- | finally a software patent I can agree with. |
14:37.58 | km- | though something as useful as that really should be open-sourced. |
14:38.04 | seanbright | pfft |
14:38.05 | Dr-Linux|home | file: I aready read that on wiki but when i do announce-frequency=0 it disable time announcement as well |
14:38.08 | Dr-Linux|home | which i don't want |
14:38.49 | file | doubt you can get that granularity |
14:39.15 | Dr-Linux|home | as i mentioned before i only want to disable the the message like "you are caller number ..." |
14:39.36 | seanbright | Dr-Linux|home: going to have to edit the source |
14:39.39 | ManxPower | Dr-Linux|home: it's an all or nothing thing. |
14:39.51 | seanbright | Dr-Linux|home: what version of asterisk are you running? |
14:39.53 | ManxPower | Why don't you just record an file and use it as MoH for those calls. |
14:40.19 | Dr-Linux|home | seanbright: the server where i want to deploy it is 1.2.17 |
14:40.28 | seanbright | jeeeeeeeeeeeeeebus |
14:40.47 | seanbright | it's going to be one of those days, huh? |
14:40.55 | km- | hahaha |
14:41.10 | km- | I'm running asterisk v0.9.8 and I hear there's this app called valet parking that I MUST HAVE. |
14:41.18 | ManxPower | seanbright: every day is like that here, why do you think I hang out on #asterisk-cli mostly? |
14:41.26 | Dr-Linux|home | seanbright: rest of all servers having 1.4.xx .... however never tried 1.6 |
14:41.40 | seanbright | Dr-Linux|home: vanilla 1.2.17? or do you have custom patches and such floating around in there? |
14:42.04 | km- | the documentation in that config file is wrong. and/or is wrong, it should just be "and" if it turns off both. |
14:42.33 | ManxPower | In any case, the answer to Dr-Linux|home's question has been answered. |
14:43.24 | Dr-Linux|home | what is vanilla |
14:43.31 | seanbright | a flavor |
14:43.32 | Dr-Linux|home | Vanilla is IceCream here |
14:43.44 | ManxPower | vanilla = no chances made |
14:43.50 | seanbright | ^^^ |
14:43.51 | *** join/#asterisk ElSonico (n=tav@xdsl-179-7.nblnetworks.fi) |
14:43.51 | ManxPower | no changes, that is. |
14:43.53 | km- | dr-linux: un-bespoke |
14:44.00 | ManxPower | vanilla also means "not kinky" |
14:44.39 | Dr-Linux|home | seanbright: I don't knwo C .. so can't modify app_queue |
14:44.52 | Dr-Linux|home | however i'd like to know can i do that using 1.4 what i want? |
14:45.03 | ManxPower | Dr-Linux|home: then you are out of luck if my suggestion is not workable for you. |
14:45.45 | Dr-Linux|home | ManxPower: sorry sir, but your suggestion was? |
14:46.05 | ManxPower | Dr-Linux|home: 9:39:52, go read the scrollback |
14:46.13 | *** join/#asterisk ichverstehe (n=harry@67-207-147-205.slicehost.net) |
14:46.46 | ichverstehe | Dial(Zap/g2/12341234||o) doesn't send the original callerid to 1234 1234 |
14:46.50 | *** join/#asterisk sakajawebe (n=chazz@nat/digium/x-9e4dc16c9671847f) |
14:47.18 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
14:47.41 | Dr-Linux|home | ManxPower: i don't get time in my irc client but: |
14:47.45 | Dr-Linux|home | <ManxPower> Dr-Linux|home: it's an all or nothing thing |
14:47.56 | Dr-Linux|home | is that? |
14:48.01 | ManxPower | ichverstehe: your carrier lets you set the callerid? |
14:48.02 | ManxPower | (9:39:52 AM) ManxPower: Why don't you just record an file and use it as MoH for those calls. |
14:48.20 | ManxPower | I don't know why I get so sick and tired of me making suggestions and people not even READING them. |
14:48.21 | seanbright | Dr-Linux|home: i have a patch for you |
14:48.49 | ManxPower | ichverstehe: also 12341234 is not a valid callerid |
14:48.59 | ManxPower | at least in the USA, since you didn't specify. |
14:49.14 | ichverstehe | ManxPower: it was an example as i don't want to reveal my id's etc .. and it's denmark. |
14:49.18 | Dr-Linux|home | ManxPower: good suggestion, i'm already using that method, but my question was just to understand/learn |
14:49.31 | ManxPower | ichverstehe: the more you mask the harder it is to help you. mask only PASSWORDS |
14:49.41 | [TK]D-Fender | ichverstehe: NoOp your callerID in your dialplan and pastebin the complete CLI output of the call with PRI debug enabled. |
14:49.42 | Dr-Linux|home | seanbright: can you share? |
14:49.45 | ichverstehe | ManxPower: hm. might be they don't .. there goes a great deal of using asterisk to redial incoming calls to out of the house cellular .. it's too annoying not to be able to see who's calling |
14:49.46 | [TK]D-Fender | ~pb |
14:49.47 | 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 |
14:49.48 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
14:49.54 | seanbright | Dr-Linux|home: no, i'm just going to tease you with it ;) |
14:50.24 | ichverstehe | 2min |
14:50.28 | Dr-Linux|home | seanbright: okey no problem, |
14:50.29 | ManxPower | For example if the callerid has a leading 0 then your carrier might reject the call or the callerid |
14:50.38 | seanbright | Dr-Linux|home: cd /path/to/1.2.17/source/ |
14:50.40 | seanbright | Dr-Linux|home: wget -q -O - "http://pastebin.ca/raw/1173004" | patch -p0 |
14:50.46 | ManxPower | but until I see some real data I can't do ore. |
14:50.51 | seanbright | Dr-Linux|home: recompile and reinstall and such |
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14:53.10 | ichverstehe | [TK]D-Fender: http://pastie.org/253586 |
14:53.18 | Dr-Linux|home | seanbright: you had it already? |
14:53.25 | seanbright | Dr-Linux|home: no, i just wrote it |
14:53.30 | seanbright | it's a 2 line change :) |
14:53.52 | Dr-Linux|home | seanbright: great that what i was thinking :) |
14:54.37 | ichverstehe | [TK]D-Fender: and that is some other phone calling 96528888 which is then dialing 21437412 |
14:54.39 | seanbright | Dr-Linux|home: once you apply that and install that, you won't hear the position part. |
14:55.01 | Dr-Linux|home | seanbright: this will work only with 1.2? |
14:55.26 | ichverstehe | asterisk 1.4.18 |
14:55.30 | seanbright | Dr-Linux|home: it will probably only work with 1.2.17 |
14:56.20 | seanbright | Dr-Linux|home: i didn't even test it, so it might not work. but it should. |
14:56.26 | seanbright | goes to smoke and such |
14:57.11 | Dr-Linux|home | seanbright: I also want to apply it on Asterisk 1.2.24 |
14:59.58 | ManxPower | ichverstehe: What info did [TK]D-Fender ask for? |
15:00.16 | [TK]D-Fender | ichverstehe: What do they see? |
15:00.28 | ManxPower | ah, he's back. |
15:00.41 | [TK]D-Fender | ichverstehe: And what happens if you don't use "o"? |
15:00.54 | ManxPower | [TK]D-Fender: I suspect his carrier is not allowing custom callerid. |
15:01.15 | ManxPower | or he is adding a leading 0 or 00, but with all that crap he posted it's hard to tell. |
15:01.38 | ichverstehe | [TK]D-Fender: the caller id shows as '99400300' which is the base number in the series we've got .. |
15:01.42 | ichverstehe | same without 'o' |
15:01.48 | [TK]D-Fender | ManxPower: No, its actually pretty clear and super minimalist |
15:02.12 | [TK]D-Fender | ichverstehe: a thought : Answer the call first, then issue Dial. |
15:02.28 | ManxPower | [TK]D-Fender: maybe to a Q.931 guru. |
15:02.58 | [TK]D-Fender | ichverstehe: If that fails, it could be that your telco blocks #'s that they have not assigned to you. |
15:03.16 | [TK]D-Fender | ManxPower: well... it IS exactly waht we need to see :) |
15:03.21 | ManxPower | ichverstehe: "(10:01:43 AM) ichverstehe: same without 'o'" contact your carrier then and tell them to allow to send the callerid info. |
15:03.42 | ManxPower | [TK]D-Fender: no what you wanted to see. I could have seen the same thing with a simple Noop. |
15:04.32 | [TK]D-Fender | ManxPower: but here we should see if it was indeed being passed on, a confirmation that this IS PRI we're looking at, and response accept/deny, etc. |
15:04.42 | [TK]D-Fender | ManxPower: Don't get all crotchety on us now! |
15:04.53 | [TK]D-Fender | ManxPower: Fo once they comply quickly and accurately! |
15:05.58 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
15:06.07 | ManxPower | [TK]D-Fender: well other than there not being a single Noop in that paste.... |
15:06.36 | ManxPower | which you did ask for. |
15:06.42 | [TK]D-Fender | ManxPower: Yeah, I know.... but the PRI debug showed it all anyways... |
15:06.55 | [TK]D-Fender | ManxPower: Not going to complain... |
15:07.03 | ManxPower | [TK]D-Fender: after 5 mins of reading it instead of the 5 seconds for a noop 8-| |
15:08.06 | [TK]D-Fender | ManxPower: Took you 5 minutes? Convalescence is upon you! |
15:08.19 | ManxPower | But you solved his problem, I guess that's what's important. |
15:08.29 | ManxPower | [TK]D-Fender: I never need to look at PRI debug. |
15:08.59 | [TK]D-Fender | ManxPower: well... I've SUGGESTED something... no confirmation of "solved" yet. |
15:09.09 | [TK]D-Fender | ManxPower: Credit where credit is due |
15:09.09 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
15:09.24 | *** join/#asterisk bad_duck (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni) |
15:09.27 | ManxPower | Gotta love the mailing list: My Astra MWI doesn't work unless I use qualifty=yes". |
15:09.59 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
15:10.02 | ManxPower | [TK]D-Fender: when the answer is "contact your carrier", as far as I'm concerned that is the end of needing support here. |
15:10.54 | bad_duck | Hi, I have just installed asterisk on debian linux, and asterisk-gui, the installation seems to be ok but the web interface refresh itself all the time when i log me in |
15:11.16 | ariel_ | has anyone worked on a load balance for the app_queue.so, Seems asterisk has a hard time with having more then 100 agents to a box. |
15:11.17 | [TK]D-Fender | ManxPower: That was your answer. Was more of a "response" than an answer actually since you didn't go through other avenues and have no confirmation. |
15:11.50 | [TK]D-Fender | ManxPower: and that ML post was understandable (minus the typo) |
15:13.49 | seanbright | Dr-Linux|home: you can try using the same patch, you'll get an error if it won't apply cleanly |
15:15.54 | x86 | http://farm4.static.flickr.com/3283/2762173260_b67720e0d0_o.jpg |
15:15.57 | x86 | this is the best "toilet humour" i think I've ever seen ;) |
15:16.39 | Iptime | ( ¨ | ¨ ) |
15:18.41 | x86 | hah |
15:18.51 | x86 | 8===D ~~ |
15:19.45 | seanbright | wow |
15:19.54 | seanbright | _slightly_ OT |
15:19.58 | *** part/#asterisk Iptime (n=chatzill@119.82.102-84.rev.gaoland.net) |
15:20.54 | x86 | perhaps |
15:20.56 | x86 | :p |
15:21.53 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
15:21.57 | *** join/#asterisk bsdwarrior (n=bashguar@70.44.84.87.res-cmts.sm.ptd.net) |
15:22.50 | bsdwarrior | once a parked call has timed out it rings back to the ext that parked the call. If they do not answer, it hangs up on them. Is there any way to have the call return to park if this happens (I know I can increase the timeout setting) |
15:23.58 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
15:24.30 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |
15:26.02 | Chotaire | hi guys, I have a question, how can I have a command executed after a user hangs up in a meetme? I don't seem to have an idea how to do continue in an extension when user exits non-zero. |
15:26.33 | Chotaire | i am sure there is a trick, I am just stuck on this. would be so great if you give me a tip. |
15:27.17 | ManxPower | bsdwarrior: that is all dialplan stuff |
15:27.25 | Chotaire | hey there manxpower... ;) |
15:27.29 | Chotaire | ltns |
15:27.37 | ManxPower | hello Chotaire |
15:27.41 | *** join/#asterisk chkngumbo (n=warble@adsl-69-232-228-26.dsl.pltn13.pacbell.net) |
15:27.42 | chkngumbo | hi |
15:28.07 | bsdwarrior | manxpower ok, I have no idea where to even start |
15:28.25 | ManxPower | bsdwarrior: start with the extension the call gets sent to on a timeout. |
15:28.47 | [TK]D-Fender | Chotaire: "h" <- * Standard extension. |
15:28.59 | Chotaire | exten => h,* ? |
15:29.13 | [TK]D-Fender | Chotaire: exten => h,1,..... |
15:29.31 | Chotaire | fender: give me a second, I will try it out... |
15:29.36 | Chotaire | does that work with asterisk v1.2? |
15:30.06 | Chotaire | (just don't ask, I haven't found time to port all my code to v1.4) |
15:30.11 | [TK]D-Fender | Chotaire: Yes |
15:30.14 | Chotaire | ok sec. |
15:30.15 | ManxPower | bsdwarrior: different versions handle parking timeouts differently |
15:30.34 | bsdwarrior | the problem is I want the call to go back to park if the person that parked it doesnt answer. |
15:30.36 | chkngumbo | using asterisk, can i carry on multiple phone calls at once through a single phone line from my phone company? |
15:30.45 | Chotaire | fender: also.. hi fender, long time no see... good to see most old faces still here. |
15:30.52 | ManxPower | bsdwarrior: that can be done too |
15:31.12 | Chotaire | mark and kapejod still around? |
15:31.16 | bsdwarrior | manxpower, I know very little, would appreciate a kick in the right direction |
15:32.38 | ManxPower | bsdwarrior: start with the cli output of a failed call on pastebin |
15:33.17 | Chotaire | fender: perfect... |
15:33.18 | *** part/#asterisk bad_duck (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni) |
15:33.44 | Chotaire | fender: thanks so much... I was just totally slammed against my head.. I've been coding all night and just had a mental outage... couldn't have been easier. |
15:34.20 | chkngumbo | i'm confused about whether its such a basic feature of a pbx that nobody's mentioning it or if its an exotic feature that nobody's doing |
15:34.30 | bsdwarrior | manxpower, this will be hard, these people are trained to pickup the parks (but yet they bitch about this problem) lol |
15:34.41 | Qwell | chkngumbo: No |
15:34.47 | Qwell | it's not possible |
15:34.52 | ManxPower | bsdwarrior: I'm sorry, but unless you do what I ask I cannot help you. |
15:35.04 | bsdwarrior | manxpower, ill see what I can do |
15:35.07 | ManxPower | I'm waiting for a pastebin of a failed call. |
15:35.14 | bsdwarrior | gimme a few |
15:35.24 | heedly | chkngumbo: no, one phone line = one call |
15:35.32 | ManxPower | bsdwarrior: If it's too hard to do we don't have to work on this. |
15:35.33 | heedly | but you can put several calls over an interenet link. |
15:35.53 | ManxPower | heedly: unless the line it a T-1/E-1/PRI |
15:36.01 | Qwell | but then it's 23 lines |
15:36.07 | heedly | right |
15:36.13 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
15:36.13 | ManxPower | well, 23, 23, 30, or 31 |
15:36.17 | Qwell | whatever |
15:36.21 | heedly | lol |
15:36.28 | chkngumbo | how do call centers redirect a single phone number to a large number of seperate conversations? |
15:36.29 | Qwell | you know what I mean |
15:36.40 | Qwell | chkngumbo: a phone number is NOT the same as a phone line |
15:36.44 | ManxPower | chkngumbo: they use multiple lines. |
15:36.53 | heedly | chkngumbo: line hunting it's called. |
15:36.56 | Qwell | a single phone number can go to multiple lines |
15:36.59 | Qwell | heedly: no it isn't :p |
15:37.12 | Qwell | (and a single line can also have multiple numbers) |
15:37.19 | heedly | how about you two just answer his questions instead of correcting me. |
15:37.19 | bsdwarrior | manxpower trying now |
15:37.28 | heedly | I'm sure your both very smart. |
15:37.30 | heedly | *you're |
15:37.38 | Qwell | I did answer his questions. |
15:38.09 | heedly | \o/ |
15:38.22 | ManxPower | heedly: how about we keep correcting you so the 279 people on this channel don't get wrong information? |
15:38.32 | heedly | ManxPower: how about not. |
15:38.39 | heedly | since I've really said nothing wrong. |
15:38.46 | ManxPower | heedly: I guess you could start using the right terms. |
15:38.54 | heedly | or I could just ignore you! |
15:39.03 | ManxPower | (10:36:53 AM) heedly: chkngumbo: line hunting it's called. |
15:39.27 | ManxPower | heedly: all people are welcome to /ignore anyone they want. |
15:39.31 | Qwell | this is just silly... |
15:39.53 | chkngumbo | all this on behalf of poor little me? heavens! |
15:40.02 | ManxPower | But I'm not going to participate in some poor sod asking for "line hunting " on a PRI because he heard it here. |
15:40.06 | Qwell | chkngumbo: no, ask away if you've got more questions :D |
15:40.11 | heedly | #asterisk is full of ass hats, so it's not just because of you. |
15:41.16 | Katty | more dots more dots more dots!!! |
15:41.39 | Katty | now stop dots! |
15:41.46 | Qwell | mutters at Katty |
15:41.56 | Chotaire | ok I'll be busy coding extensions, I'll stay tho. good luck with your projects. |
15:41.58 | jeev | ManxPower, thanks for answering my question yesterday. i had gone afk. |
15:42.01 | Katty | hugs Qwell |
15:42.07 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
15:42.09 | chkngumbo | my dream is to be able to have a customer call a single phone number, for a system to ring all the phones in the office (4 of them) which aren't currently busy, and be able to maintain 1 individual conversation per phone |
15:42.11 | Katty | hai Zeeek! |
15:42.27 | ManxPower | chkngumbo: then you must have more than 1 line. |
15:42.39 | Zeeek | Preconference seats available right now at #voip-users-conference and call us http://bit.ly/voip |
15:42.42 | Katty | Qwell: hit it like you mean it! |
15:42.49 | Zeeek | {{{Katty}} |
15:42.58 | Katty | Qwell: now stop dots!!! |
15:43.18 | Zeeek | DOnner, Blitzen,Russell,Qwell,File, MOH pleaseeeeeeese join us now |
15:43.24 | ManxPower | chkngumbo: the best would be to have a PRI line, but it would be too expensive for just 4 calls |
15:43.27 | Qwell | Zeeek: topic? |
15:43.35 | Zeeek | oh, that again? |
15:43.39 | Katty | Qwell: there is no aggro reset :< |
15:43.40 | Zeeek | Ummmmmm |
15:43.46 | Katty | Qwell: except for invisible |
15:43.54 | Katty | Qwell: can you invisible in raid? |
15:44.01 | Zeeek | the the topic is uh.... it's... tooo sensitive to be mentioned aloud hre |
15:44.08 | Qwell | Katty: 5 seconds. |
15:44.20 | Katty | Qwell: disgusting. |
15:44.20 | Zeeek | but there are FREE Astricon passes to be given away, I can mention that |
15:44.28 | Katty | Qwell: i have a solution. |
15:44.31 | Katty | Qwell: bring a better tank. |
15:46.50 | *** join/#asterisk jazzmann (n=chatzill@cpc1-lutn9-0-0-cust163.lutn.cable.ntl.com) |
15:47.41 | jazzmann | mobile search coming as no suc command in chan_mobile |
15:47.45 | chkngumbo | a T1 line is a sort of PRI? |
15:47.58 | Qwell | a PRI is over a T1 |
15:48.05 | jazzmann | please help |
15:48.40 | Qwell | jazzmann: is it loaded? |
15:48.53 | jazzmann | yes |
15:48.57 | chkngumbo | how much is a T1 line likely to cost, and how many voice connections would i let me have? |
15:49.16 | jazzmann | http://www.geek-pages.com/articles/asterisk/howto_build_and_configure_chan_mobile_on_trixbox.html |
15:49.17 | Qwell | it varies wildly.. between ~$300 and...~$800 |
15:49.30 | ManxPower | chkngumbo: $300 - $800/month, up to 23 calls |
15:49.34 | ManxPower | looks at Qwell |
15:49.38 | Katty | chkngumbo: you can have lots of DIDs tho |
15:49.39 | Qwell | ManxPower: was I actually..right? |
15:49.43 | jazzmann | I followed this walkthrough .I do not have tribox it is mandriva linux |
15:49.49 | jeev | chkngumbo, 300-800 bux.. about 23- MAYBE 24 calls. |
15:49.51 | Katty | DIDs++ |
15:49.54 | jeev | lol |
15:50.09 | Qwell | You can also get partial PRIs - I think the min you can usually get is about...8 channels? |
15:50.13 | jeev | russelb? |
15:50.13 | ManxPower | Qwell: no way to know about prices, I was just surprised you estimated the same as me. |
15:50.22 | Qwell | ManxPower: yeah, we must be right then |
15:50.30 | russellb | ~8ball does jeev have a good reason to be messaging me privately? |
15:50.30 | jbot | I'm sure of it. |
15:50.30 | Katty | our company is paying 500 something a month |
15:50.33 | russellb | damn |
15:50.37 | jeev | hahahah |
15:50.40 | jeev | and you aw, it's a damn good one |
15:50.41 | Qwell | ~8ball does jeev want his +q yet? |
15:50.42 | jbot | Absolutely. |
15:50.51 | Qwell | russellb: You heard the bot! |
15:50.56 | heedly | Katty: do you get free local calls wiht that? |
15:51.10 | Katty | heedly: i believe so. |
15:51.38 | Qwell | jazzmann: how do you know it's loaded? |
15:51.39 | russellb | Qwell: k! |
15:51.40 | ariel_ | Prices for PRI and T1 are depended on locations. I have 12 PRI lines setup and I am paying 3,500 for all of them. But I also have 3 DS3's for data with same provider |
15:51.40 | jeev | ~8ball if qwell goes to astricon, should i give him a real +q ? |
15:51.41 | jbot | Absolutely. |
15:51.56 | *** mode/#asterisk [+b %jeev!*@*] by russellb |
15:52.06 | Katty | hehehe |
15:52.07 | *** join/#asterisk coppice (n=chatzill@175.202.17.210.dyn.pacific.net.hk) |
15:52.30 | jazzmann | module load chan_mobile.so when I enter this command asterisk disconnects |
15:52.40 | Zeeek | http://bit.ly/voip to join us. See you all later. Have a nice, California-like day! |
15:52.46 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
15:52.48 | Qwell | so what's the topic ze |
15:52.49 | Qwell | ... |
15:52.51 | Qwell | whatever |
15:53.03 | heedly | ariel_: what do you pay then for usage on the PRI's? |
15:53.05 | *** join/#asterisk astassistant (n=jtknapp@h-72-244-204-146.sttnwaho.dynamic.covad.net) |
15:53.35 | jazzmann | Nw when I trying to connect it is giving this error.Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
15:53.45 | jazzmann | i.e asterisk |
15:54.07 | ariel_ | heedly, all inbound traffic is included, I don't use them for outbound. I pay per 100 did's permonth at a rate of $ 3.20 per 100 |
15:54.24 | russellb | Qwell: I'm working on a bribe to get the +q removed |
15:54.43 | Qwell | pallet of sugarfree redbull? |
15:54.48 | Qwell | something along those lines? |
15:54.50 | chkngumbo | right now i have 2 voice lines, 1 fax line, and 1 dsl line. something like $160/mo. does that sound like a reasonable setup unless i feel like spending a bit more for a nicer system? |
15:54.51 | jazzmann | when I close the command prompt and open it again asterisk is connecting but again mobile search not working |
15:55.12 | jazzmann | at cli command |
15:55.19 | russellb | Qwell: i told him i'd settle for a single case (24 cans) |
15:55.21 | Qwell | chkngumbo: that's pretty expensive, though I'm sure the DSL line is a large part of that |
15:55.44 | chkngumbo | i'm in a pretty expensive area, so that's probably part of it |
15:56.03 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
15:56.24 | JayTee52 | which is a better choice with Asterisk for an ATA adapter, Linksys PAP2T or Linksys SPA2102? |
15:56.35 | [TK]D-Fender | JayTee52: Clearly the latter |
15:56.59 | JayTee52 | [TK]D-Fender, then if that's your opinion that's what I'm going to order :-) Thanks! |
15:59.24 | *** mode/#asterisk [-b %jeev!*@*] by russellb |
15:59.28 | jeev | gasps for air |
15:59.31 | jazzmann | I need to connect my cell to asterisk ,please help me in providing some solution |
15:59.59 | jazzmann | it is now a week since trying to achieve this |
16:00.41 | *** join/#asterisk draygon-w (n=draygon@gateway5-pnap.exigo.com) |
16:00.44 | draygon-w | HeyHey |
16:00.45 | draygon-w | Hey* |
16:00.58 | draygon-w | Does anyone have Allison's phone number? I remember she used to have it on the site but I dont see it there anymore |
16:02.19 | chkngumbo | can i make it so that a land line and a few cell phones ring at the same time, except the ones that are already busy? |
16:03.04 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
16:03.13 | draygon-w | that's possible |
16:03.23 | draygon-w | chkngumbo look up follow me |
16:03.25 | draygon-w | i think |
16:04.34 | jazzmann | localhost*CLI> mobile search |
16:04.35 | jazzmann | No such command 'mobile search' (type 'help' for help) |
16:04.54 | jazzmann | localhost*CLI> module load chan_mobile.so |
16:04.56 | jazzmann | localhost*CLI> |
16:04.58 | jazzmann | Disconnected from Asterisk server |
16:05.09 | jazzmann | this is the output |
16:05.35 | ManxPower | jazzmann: chan_mobile is crashing your Asterisk |
16:06.00 | jazzmann | so what should I do |
16:06.07 | ManxPower | Don't ask me why. I just know it is based on your paste of the 5 lines |
16:06.35 | jazzmann | I mean do you have any suggestions |
16:06.39 | draygon-w | ManxPower do you know a quick way to get in touch with allison or a number i can reach her at? |
16:06.42 | JayTee52 | did anyone hear the news that Nortel bought Pingtel? |
16:06.51 | brodiem | draygon-w: theivrvoice.com |
16:06.53 | ManxPower | jazzmann: No. |
16:07.06 | ManxPower | draygon-w: No. |
16:07.40 | jazzmann | anyone on this channel has any suggestion please |
16:07.42 | draygon-w | I know she used to have a number on her site |
16:07.44 | draygon-w | I guess she took it out |
16:08.11 | *** join/#asterisk moy (n=moy@nat/ibm/x-998d8b2c4d5f3c52) |
16:08.19 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
16:08.47 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
16:08.52 | jazzmann | any forum or other channel who can help me in chan_mobile |
16:08.53 | jazzmann | thanks |
16:10.05 | chkngumbo | it sounds like what i want is "Multi-line hunting" |
16:10.50 | heedly | hehe |
16:11.00 | heedly | they don't call it that anymore I guess. |
16:11.08 | Qwell | not on PRI they don't |
16:11.18 | heedly | it's just channels now? |
16:11.26 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
16:11.29 | heedly | and DID's associated with them? |
16:11.33 | chkngumbo | i got the term from wikipedia: http://en.wikipedia.org/wiki/Hunting_(telephony) |
16:13.16 | draygon-w | anyone here can recommend someone who does good voice overs? |
16:13.36 | x86 | Allison |
16:13.43 | x86 | I forget her last name.... |
16:14.06 | draygon-w | Allison smith |
16:14.10 | draygon-w | yes, other than her |
16:14.14 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
16:14.18 | draygon-w | She seems hard to get through with these days |
16:14.35 | x86 | http://store.digium.com/productview.php?category_id=8&product_code=8IVRPROMPT&main_category_id=8 |
16:15.19 | x86 | draygon-w: there is also June |
16:15.31 | draygon-w | ahh |
16:16.05 | draygon-w | whats the turn around with these? |
16:16.26 | *** join/#asterisk bad_duck (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni) |
16:16.32 | x86 | not sure, i closed the page |
16:16.36 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
16:16.47 | draygon-w | whats june's last name? |
16:16.52 | draygon-w | any idea? |
16:17.02 | [TK]D-Fender | Wallach |
16:17.02 | x86 | nope |
16:17.17 | *** join/#asterisk Nate187 (n=Nate187@gw.bigrivertel.net) |
16:17.23 | [TK]D-Fender | something like that |
16:17.24 | x86 | Note: Orders, not prompts, are typically processed within 24 hours of receipt, on the next business day. Allow 2-3 days for voice prompts to be created. |
16:17.29 | Qwell | June Wallack |
16:19.25 | draygon-w | thanks qwell |
16:19.32 | draygon-w | anyone in here do voice overs? heh |
16:19.41 | Qwell | draygon-w: sure, I'll do voiceovers |
16:19.45 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
16:19.53 | Qwell | I offer a 0% quality guarantee |
16:19.57 | draygon-w | no thanks |
16:19.59 | draygon-w | heh |
16:20.01 | Qwell | 2% |
16:20.08 | draygon-w | getting there.. |
16:21.36 | *** join/#asterisk Firass-z0r (n=Firass@ead224-222.housing.wwu.edu) |
16:22.10 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
16:23.29 | *** join/#asterisk steliosk (n=Stelios@athedsl-285035.home.otenet.gr) |
16:27.56 | *** join/#asterisk ar3dam (n=ar3dam@189.156.217.142) |
16:28.22 | ar3dam | hello there, how i can see what channel is busy with the cli? |
16:28.36 | ariel_ | core show channels |
16:29.12 | ar3dam | thks, ariel_ |
16:31.00 | chkngumbo | does anybody know how this might be done: all cell phones in a group of 4 cell phones ring at the same time (unless if they're busy) |
16:31.25 | Qwell | chkngumbo: sure, the Dial application in Asterisk can dial multiple devices simultaneously |
16:31.35 | Qwell | whichever device answers, gets the call |
16:32.01 | ManxPower | Well, unless any of the calls go via a FXO port, of course |
16:32.14 | chkngumbo | would that require 1 phone line per simultaneous call you want to make to a cell phone? |
16:32.23 | Qwell | chkngumbo: yes |
16:33.19 | ManxPower | chkngumbo: You should expect to pay about $700 for all the cards and devices you need |
16:36.32 | seanbright | Dr-Linux|home: did it work? |
16:36.57 | Dr-Linux|home | seanbright: i didn't try yet it's production time |
16:37.03 | Dr-Linux|home | many calls bridged |
16:37.06 | seanbright | ah |
16:37.12 | seanbright | i say try anyway |
16:37.14 | seanbright | they'll call back. |
16:37.41 | Dr-Linux|home | ok |
16:38.30 | chkngumbo | do any phone companies offer the ability to have one number ring a group of numbers at the same time? (whether or not each number is busy might not matter) |
16:39.37 | ManxPower | chkngumbo: maybe you stop asking and start reading The Book |
16:39.39 | ManxPower | ~book |
16:39.40 | 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 |
16:39.48 | ManxPower | this is not a telecom tutorial channel |
16:40.15 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
16:41.44 | *** join/#asterisk discHead (n=larry@wsip-70-183-82-162.sd.sd.cox.net) |
16:42.17 | Dr-Linux|home | seanbright: when i can see you here again |
16:42.24 | chkngumbo | i'm not sure if asterisk is the way to go if a phone company can ring 4 phones at once for cheaper than they'll sell me 5 phone lines so that i can do it myself |
16:42.48 | Qwell | ~101 |
16:42.49 | jbot | methinks 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 |
16:42.56 | Qwell | chkngumbo: That may help you a little bit as well |
16:43.39 | [TK]D-Fender | chkngumbo: NO. |
16:47.02 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
16:48.25 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
16:50.53 | jameswf-home | I think i shal start a new blog... I will post on craigslist as an underage girl and post all the replies to said blog... |
16:52.38 | Strom_M | you could be extra clever and call it "The Pedometer" |
16:54.00 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
16:54.00 | *** mode/#asterisk [+o lmadsen] by ChanServ |
16:55.08 | *** join/#asterisk matt_keys (i=smatt@usaregs.com) |
16:55.17 | *** join/#asterisk luckyaba (n=lucky@ip68-6-98-146.sb.sd.cox.net) |
16:57.07 | chkngumbo | Qwell: that book was very helpful. thank you |
16:57.15 | Qwell | chkngumbo: thank Strom |
16:57.21 | chkngumbo | thank you, strom |
16:57.36 | Strom | you're welcome |
16:58.31 | chkngumbo | did you suggest the book or write it or? |
16:58.53 | Qwell | he hosts it and made it available |
16:59.19 | chkngumbo | cool |
16:59.35 | lowtek | Well so much for satisfiying both my inner-geek and inner-child, apparently clone-wars is a bust ... |
16:59.47 | Qwell | lowtek: boo |
16:59.57 | luckyaba | I just got a DID and want to make sure 5060 is the only port i need to forward on my router to my asterisk box? |
16:59.57 | matt_keys | I'm trying to create a MixMonitor macro (named wiretap) so I can monitor by extension when needed. The macro has the line "exten => _X.,1,MixMonitor(${EXTEN}.wav|a)" , and I'm trying to include it in the regular extensions list (i.e, exten => 0154,1,Macro(wiretap,stdexten,0154,sip/0154) ) |
17:00.11 | matt_keys | When I do so and place a test call, the line is busy... what am I doing wrong? |
17:00.16 | *** part/#asterisk korihor (n=korihor@190.78.32.60) |
17:03.31 | matt_keys | <-- asterisk newb. |
17:03.39 | luckyaba | haha, ditto |
17:03.53 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
17:04.08 | matt_keys | anyone? |
17:04.40 | lowtek | matt_keys: pastebin your console output ... |
17:04.58 | matt_keys | ok just a sec.. |
17:05.23 | *** join/#asterisk jayrod422 (n=jayrod42@node2.164.136.64.1dial.com) |
17:06.34 | jayrod422 | does anyone know how i can read the number from where a call was forwarded from using astiersk? i thinking about installing a pri card into a * box and use the asterisk to read the ani of the originall caller not the did that the call came in on... |
17:06.55 | Strom | jayrod422: lol. |
17:07.08 | Strom | you don't get separate ANI and CPN on a PRI |
17:07.54 | matt_keys | lowtek : http://pastebin.com/d6f245cf6 |
17:09.16 | lowtek | matt_keys: Ok, now please pastebin your macro ... |
17:09.59 | [TK]D-Fender | Context 'macro-wiretap' for macro 'wiretap' lacks 's' extension, priority 1 <- like this doesn't say it all? |
17:10.13 | jeev | damn, nakee pics of swedish olympian leaked, she's pretty good. |
17:10.38 | jameswf-home | okay first ad up :)) lets see if it gets flagged off |
17:10.40 | lowtek | jeev: url? |
17:11.29 | matt_keys | lowtek: http://pastebin.com/d3b17ebfc |
17:11.45 | lowtek | Replace your _X. with s |
17:12.23 | lowtek | Also, just advice, if this is a business, you may want to change the word 'wiretap' to something else .. just in case |
17:12.52 | chkngumbo | "QualityAssurance" |
17:13.24 | *** join/#asterisk MrNaz (n=naz@ppp121-44-245-95.lns4.mel4.internode.on.net) |
17:13.48 | matt_keys | lowtek: http://pastebin.com/d5301b6ea |
17:14.02 | lowtek | I personally don't see anything wrong with the naked olympian pics .. she's worked hard to look that way, they are tasteful, and she is beautiful. |
17:15.10 | Wayhigh | naked olympian pics? |
17:15.13 | jeev | yea |
17:15.19 | jeev | i'm not gonna paste here |
17:15.21 | lowtek | matt_keys: Did you reload? |
17:15.23 | jeev | get my ass kicked |
17:15.30 | matt_keys | lowtek : yes. something to do with queues? |
17:15.33 | jeev | her last pic is nice.. nice boobs and pootang |
17:15.51 | [TK]D-Fender | matt_keys: because your macro has 1 line in it, thats all it does, and then it stops |
17:15.54 | lowtek | lol, very nice grooming ... |
17:16.21 | matt_keys | [TK]D-Fender : ok, so what else should be included to keep it going throug hthe other macros? |
17:16.55 | [TK]D-Fender | matt_keys: what "other macros"? "s" runs out! If you wanted to do more go make more priorities |
17:17.40 | matt_keys | [TK]D-Fender : I want it to continue on to stdexten|0154|sip/0154 |
17:18.01 | matt_keys | [TK]D-Fender : I'm assuming that's why it's not ringing? |
17:18.02 | lowtek | matt_keys: Do an s,n,GoTo() |
17:18.14 | Wayhigh | ok.. I got one complaint about the pics.. |
17:18.24 | Wayhigh | them tan lines are totally blinding.. |
17:18.25 | lowtek | Wayhigh: ?? |
17:18.28 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
17:18.28 | jeev | lol |
17:18.37 | [TK]D-Fender | matt_keys: Macro does not jsut call a bunch of other macros. Who said you could next them like that? |
17:18.46 | [TK]D-Fender | nest* |
17:19.11 | [TK]D-Fender | matt_keys: You want to call 2 macros, call them back to back. You can't cram them onto the same line |
17:19.17 | jameswf-home | holy crap NSFW |
17:19.39 | lowtek | jeev: I'm sure it's some ugly chicks making a big deal out of it... always is... |
17:19.48 | jeev | yea |
17:20.40 | matt_keys | ok, so the best thing would be to remove the macro all together, then put it above the origional extension list line (exten => 0154,1,Macro(stdexten,0154,sip/0154) )? |
17:20.41 | lowtek | Yowzers! Did you look at the third section down o nthat page??? I need to go shower ... |
17:21.08 | lowtek | matt_keys: Or just call the mixmonitor first, second do a Goto() to your stdexten macro ... |
17:21.10 | [TK]D-Fender | matt_keys: Of call the other macro from within this one, but that can get messy. |
17:21.45 | jameswf-home | here lizard lizard lizard |
17:22.19 | matt_keys | [TK]D-Fender : so how is stdexten,0154,sip/0154 being called all at once? |
17:22.27 | matt_keys | or are they arguments? |
17:23.05 | luckyaba | anyone know where i can look to get info on mapping my DID to ring to all phones? |
17:23.19 | luckyaba | google isn't being nice to me at the moment |
17:23.33 | lowtek | luckyaba: Dial(SIP/peer&SIP/peer&SIP/peer|options) |
17:23.47 | jeev | jameswf-home, not safe? you'd poke that till it was unsafe to move |
17:24.16 | jameswf-home | is married therefore doesnt poke anything |
17:24.19 | lowtek | I wouldn't touch that with your d... |
17:24.28 | matt_keys | sorry guys, i'm pretty new at this! |
17:24.32 | lowtek | I had to empty my browser cache ... |
17:24.36 | lowtek | For fear of disease |
17:24.40 | coppice | come on. olympic athletes are supposed to complete naked. its traditional |
17:24.42 | jeev | i iwould hit that.. but i love my girl so i wouldn't poke anything either. she doesn't have diseases! |
17:25.05 | *** part/#asterisk bad_duck (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni) |
17:25.17 | jameswf-home | how can you truely reach the olympic spirit with clothes on |
17:25.18 | *** join/#asterisk jayrod422 (n=jayrod42@node2.164.136.64.1dial.com) |
17:25.18 | lowtek | jeev: Are we talking about the same pic? Third section down? |
17:25.37 | jayrod422 | <PROTECTED> |
17:25.40 | matt_keys | lowtek : I've got the Mixmonitor line above the stdexten line, should I remove the s and just have 0154,1,MixMonitor(.... |
17:25.41 | jeev | oh, no idea who you're talking about |
17:25.43 | lowtek | jeev: I'm not talking about the hot olypian.. |
17:25.49 | jeev | ohhhhhh |
17:25.51 | lowtek | Look at the third section down |
17:26.10 | jeev | i can't look now |
17:26.48 | lowtek | There's the olypian, then some other girl who looks pretty mean, then the third section there's some ragged looking thing .. |
17:27.39 | jameswf-home | i must be looking at the wrong site... |
17:27.39 | [TK]D-Fender | matt_keys: stdexten,0154,sip/0154 <- stdexten is the macro, the rest ar arguments |
17:27.46 | lowtek | coppice: agreed |
17:28.01 | matt_keys | [TK]D-Fender : thank you, that's what I was confused about |
17:28.58 | coppice | gymnasium comes from the greek word meaning naked. the original olympians competed naked. folk are just so screwed up these days |
17:29.14 | jeev | james, are you looking at that swede with the stockings? |
17:32.37 | matt_keys | [TK]D-Fender : Ok great it works (and no one liner macros!) but there's a new problem. It only works on inbound and not outbound |
17:33.00 | [TK]D-Fender | matt_keys: Show us that you're USING it on "outbound" |
17:33.20 | [TK]D-Fender | matt_keys: "monitor" works... your DIALPLAN... well that you don't my opinion on ;) |
17:33.45 | matt_keys | oic, so it needs to be on the trunk also |
17:33.55 | matt_keys | listening for outbound on 0154 |
17:34.31 | [TK]D-Fender | matt_keys: "trunk"? No, this is DIALPLAN. |
17:34.40 | [TK]D-Fender | matt_keys: And never use that word again. |
17:36.03 | jayrod422 | anyone have any idea what it is needed to read a pri's q931 debug info for call forwarding info and then gen a sip call with the info prepened on the invite? im trying to build a vmail app using asterisk but cant figure out the call forward part so asterisk knows what phone forwarded the call |
17:36.46 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
17:37.53 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
17:37.59 | luckyaba | when i call my number i don't hear anything. Doesn't ring but it does say its connected |
17:38.26 | *** join/#asterisk mercutioviz (n=chatzill@freeswitch/developer/msc) |
17:38.52 | luckyaba | ohhh |
17:39.03 | luckyaba | now it says the call didn't go through... |
17:39.05 | luckyaba | lol |
17:43.24 | matt_keys | [TK]D-Fender so I'm assuming I need to place the MixMonitor line before the Dial command, but if that's included in the stdexten macro, how can I narrow it down to the extension of my choice? |
17:46.49 | *** join/#asterisk talntid (n=eric@66.208.251.170) |
17:47.48 | [TK]D-Fender | matt_keys: Change your macro so you pass it a parameter that will indicate if it should record or not and just do it all in one macro |
17:52.31 | *** join/#asterisk Entr4nced (n=IMG001@cpe-76-190-141-153.neo.res.rr.com) |
17:52.54 | *** join/#asterisk til_ (n=til@ns5.okg-computer.de) |
17:53.57 | til_ | hello all |
17:54.10 | til_ | i am new to asterisk |
17:54.19 | matt_keys | [TK]D-Fender : I created another stdexten macro and placed the MixMonitor line before Dial in priority, but it didn't kick it off. Do I need to change something in sip.conf, or am I on the right track? |
17:54.23 | [netman] | hi til_ |
17:54.34 | til_ | and i have troubles with the dial plan |
17:54.36 | matt_keys | [TK]D-Fender : we're going sip -> pri for outbound |
17:55.01 | til_ | i managed to activate the sip account of my voip provider |
17:55.19 | til_ | and i can connect via softphone to asterisk |
17:55.38 | til_ | but how do i route the calls? |
17:55.43 | [TK]D-Fender | matt_keys: Dialplan is dialplan. If it didn't kick off, you did it wrong. |
17:56.14 | [TK]D-Fender | til_: Make a peer entry for your ITSP and you'll probably be wanting to register to them as well |
17:59.26 | til_ | [TK]D-Fender: well sounds so easy ;) |
17:59.47 | til_ | but i am a complete asterisk noob |
18:00.57 | til_ | and if im right, i only need a connection between my sip account at my provider and my local sip accounts for the soft phones |
18:02.30 | [TK]D-Fender | til_: there is no connection betwen your softphone & your ITSP |
18:03.00 | [TK]D-Fender | til_: Each connects to * and your DIALPLAN tells * to take a call from A and call out to B. |
18:03.13 | [TK]D-Fender | til_: Time to go read the BOOK |
18:03.23 | [TK]D-Fender | til_: and here's a "sample" for you |
18:03.25 | [TK]D-Fender | ~jerjerguide |
18:03.26 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
18:03.28 | [TK]D-Fender | ~book |
18:03.29 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
18:07.27 | til_ | [TK]D-Fender: i try my best, but the syntax is very confusing when you look at it for the first time :( |
18:07.36 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
18:08.13 | [TK]D-Fender | til_: Go try stuff. |
18:10.49 | til_ | i do, i am really happy, that i can talk between my two pc |
18:12.08 | jeev | hmm |
18:12.12 | jeev | til.... same til i know ? |
18:12.27 | mercutioviz | are there any fonality users here? I was wondering if there's a discussion forum or IRC channel for things like PBXtra, HUD, etc. Thoughts? |
18:13.33 | til_ | jeev: dont think so |
18:14.02 | [TK]D-Fender | mercutioviz: I think that if there were you'd have seen it on their respective web-sites |
18:14.32 | mercutioviz | I thought so... all I could find was a wiki |
18:14.44 | mercutioviz | Well, I'll ask my question here, just in case anyone knows... |
18:14.54 | matt_keys | [TK]D-Fender : can you point me to some conditional parameter examples? I'm having a hard time figuring out what to do here |
18:15.54 | seanbright | [TK]D-Fender: thought you might appreciate this -> http://www.penny-arcade.com/comic/2008/8/1/ |
18:17.06 | ManxPower | I wish the devs would make "cat /path/to/src/asterisk/UPGRADE.TXT" as the last line of "make install". NOBODY seems to read the damn thing. |
18:19.11 | jayrod422 | any know how to read the redirecting number in a dial plan for a pri using libpri? so when i a call come in i can assign the redirecting num to a var... |
18:19.36 | [TK]D-Fender | matt_keys: "core show application gotoif". Go read about expressions and how to use variables on the WIKI |
18:20.22 | [TK]D-Fender | jayrod422: I know its visible in debug, but I don't believe * imports it or give you access to it like you can with SIP headers |
18:20.51 | jayrod422 | can you buy sip orig with that info in the header? |
18:22.06 | *** join/#asterisk rs^ (n=irc@84-75-214-211.dclient.hispeed.ch) |
18:22.12 | rs^ | hello there |
18:22.37 | rs^ | does anybody have an idear how to configure, how long it should ring until voicemail comes up ? |
18:23.03 | matt_keys | [TK]D-Fender thanks |
18:23.17 | *** join/#asterisk angom (n=angom@201.170.65.143) |
18:23.28 | [TK]D-Fender | jayrod422: Depends on the ITSP |
18:23.51 | [TK]D-Fender | rs^: extensions.conf. its YOUR dial statement, go change it |
18:25.06 | rs^ | [TK]D-Fender: great one second |
18:26.17 | jameswf-home | awesome http://rocketgirl1993.blogspot.com |
18:26.56 | rs^ | [TK]D-Fender: is it this ? "4:5)" for 45 seconds ? |
18:26.59 | draygon-w | Wow. |
18:27.08 | draygon-w | June has a pretty good turn around prices |
18:27.13 | draygon-w | time* |
18:27.14 | *** join/#asterisk Paige_ (n=Paige@208.89.241.31) |
18:27.39 | [TK]D-Fender | rs^: thats 6 chars without any proof that it is your DIAL COMMAND. |
18:27.58 | [TK]D-Fender | rs^: does it say that its calling the DIAL application? |
18:28.41 | rs^ | hmm where exactly do i configure it ? its saying |
18:29.04 | ManxPower | rs^: you edit extensions.conf, copy the part you have a problem with, copy it up to pastebin.ca |
18:29.05 | rs^ | exten => asterisk,3,GotoIf .... /voicemail/${PHONE}/vm1) 4:5) |
18:29.22 | rs^ | may i query one of u ? |
18:29.29 | rs^ | [TK]D-Fender: or ManxPower ? |
18:29.47 | [TK]D-Fender | rs^: that is GOTOIF, not DIAL |
18:29.47 | ManxPower | rs^: Only if you have a credit card with a high limit. I don't do personal consulting for free. |
18:30.00 | ManxPower | ~manxpower |
18:30.01 | jbot | extra, extra, read all about it, manxpower is NOT an employee of Digium. He is looking for a training/teaching job in networking and/or Asterisk. Contact: eric@fnords.org |
18:30.03 | talntid | anyone want to buy a 3 month old R1T1? |
18:30.13 | jameswf-home | ~me |
18:30.13 | jbot | no u |
18:30.21 | rs^ | [TK]D-Fender: ok so it has to be DIAL command |
18:30.22 | [TK]D-Fender | ~i |
18:30.23 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
18:30.40 | jameswf-home | ~i |
18:30.41 | jbot | rumour has it, jameswf-home is a tool |
18:30.42 | [TK]D-Fender | rs^: How many more times should I have to say it? |
18:30.47 | jameswf-home | damn bot |
18:30.50 | rs^ | [TK]D-Fender: sorry |
18:30.53 | fogo | ~i |
18:31.02 | jameswf-home | dude wtf |
18:31.07 | [TK]D-Fender | ~ou812 |
18:31.20 | rs^ | I dont find the word "DIAL" in extensions.conf |
18:31.28 | rs^ | [TK]D-Fender: there is none |
18:31.47 | rs^ | It's always getting into voicemail after to time ringing [TK]D-Fender |
18:31.52 | talntid | Had to replace the ever-defective r1t1 with a sangoma card... anyone had good luck with these r1t1 cards? |
18:31.55 | ManxPower | rs^: I'm starting to think that you did NOT set up this Asterisk server or that you used a GUI to set it up. |
18:32.12 | [TK]D-Fender | ManxPower: that's what I'm thinking. |
18:32.20 | ManxPower | ~gui |
18:32.21 | jbot | hmm... gui is (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, ... or at http://pla-netx.com/linebackn/guis/index.html. Of course Real Programmers use the command line interface. See cli |
18:32.27 | ManxPower | eeek! That was not what I wanted. |
18:32.30 | ManxPower | ~zeeek |
18:32.31 | jbot | extra, extra, read all about it, zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
18:32.35 | *** join/#asterisk oej (n=olle@ns.webway.se) |
18:32.36 | ManxPower | That's the one I wanted. |
18:32.51 | rs^ | ManxPower: [TK]D-Fender Thats right I have only basic knowledge up to now |
18:33.16 | [TK]D-Fender | rs^: What exactly are you using to configure your system? |
18:34.24 | ManxPower | ~employees |
18:35.14 | ManxPower | rs^: I'm sorry, but you don't even know enough about Asterisk for us to help you. You might consider taking some time right now to read the asterisk book. |
18:35.16 | ManxPower | ~book |
18:35.16 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
18:36.28 | jblack | so, Sangoma cards have this really neat feature that the Rhino cards lack. They work. |
18:36.33 | rs^ | [TK]D-Fender: Im using the config files directly |
18:36.57 | talntid | jblack: be fair, the r1t1 worked........ for about 2 days.... at a time.... |
18:37.01 | ManxPower | employees |
18:37.23 | rs^ | ManxPower: so point me at least to where i have to read for solving the problem this would be fine for me too |
18:37.35 | talntid | what really makes the sangoma cards worth it though, is they actually have documentation. |
18:38.05 | ManxPower | talntid: what in the world made you go with someone other than Digium or Sangoma? |
18:38.45 | ManxPower | rs^: You're going to make me go find the book, find the section and give you a page number, right? |
18:38.49 | jblack | jameswf-home in the world. |
18:38.57 | jameswf-home | huh? |
18:39.28 | talntid | supposedly, ManxPower, the rhino r1t1 was a good product. |
18:39.41 | [TK]D-Fender | rs^: only reason not to see a dial command in extensions.conf is A ) devices are configured with users.conf (flaming piece of shit) B ) it is #include from another file. |
18:39.51 | [TK]D-Fender | rs^: or C ) from AEL |
18:39.51 | talntid | buy it seems poorly engineered.. at least, the driver side of it... |
18:39.56 | ManxPower | ?4:5 means "Jump to priority 4 if true or priority 5 if false (I would have to do "core show application gotoif" to know for sure). Of course we don't use priority numbers in Asterisk anymore, so who knows even if changing it would work. |
18:40.25 | jameswf-home | thinks the r1t1 is good of course I am biased :) |
18:40.32 | jblack | or drugged. |
18:40.35 | [TK]D-Fender | ManxPower: Speak for yourself... explicit line-numbering FTW! |
18:40.56 | ManxPower | [TK]D-Fender: you luddite you, move in to the 1.2 world from your 1.0 world! |
18:41.09 | talntid | jameswf-home: why? have you had good luck putting LOTS of calls through it? like in a call center enviorment? |
18:41.15 | talntid | whats the trick to making it reliable? |
18:41.16 | [TK]D-Fender | ManxPower: Ok, I'll be the pot this time, you can be the kettle. |
18:41.22 | ManxPower | or in shorter terms "pot. kettle. black." 8-) |
18:41.28 | *** join/#asterisk nighty^ (n=nighty@host217-37-109-17.in-addr.btopenworld.com) |
18:41.28 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
18:41.30 | [TK]D-Fender | ManxPower: Beat you to it :p |
18:42.40 | talntid | I was putting 2k calls per day through it, and it was as unreliable as a windows ME machine. |
18:42.46 | ManxPower | talntid: I'm tempted to try Digium cards again. They have been totally redesigned at least once, maybe twice since I used them. However, the last Digium card I used cost me $1,200 of my own money and almost cost me a client. |
18:43.18 | ManxPower | the $1,200 was for a server that did not generate HDLC Abort errors anytime the disk was used. |
18:44.54 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
18:45.11 | ManxPower | But I was using 2nd generation cards, not the 4th generation cards they have now |
18:47.00 | jblack | Does digium do the same object file blob that rhino does? |
18:47.27 | ManxPower | jblack: you mean for the kernel module? |
18:47.36 | jblack | Correct. |
18:47.37 | ManxPower | or firmware to upload to the card? |
18:47.55 | ManxPower | Neither Digium nor Sangoma have binary blob kernel modules. |
18:48.07 | Qwell | I would highly doubt Rhino does either. |
18:48.08 | jblack | Nope, not firmware. Object files. The bulk of the driver comes as an object file. |
18:48.13 | ManxPower | Sangoma has a binary blob for firmware that is field upgraded. |
18:48.33 | jblack | No, no, not firmware. That's a universal thing. |
18:48.37 | ManxPower | I assume Digium does to if they have any firmware upgradable cards. |
18:49.29 | *** join/#asterisk MarkJenks (n=MarkJenk@yosemite.cellcom.com) |
18:50.13 | MarkJenks | Anyone out here have knowledge of getting ss7 working w/ dahdi? |
18:50.15 | jblack | Let me grab their tallball, now that they have a working ftp site |
18:50.56 | lowtek | tallball? Some kind of drink? Like a 40? |
18:51.01 | tzafrir_laptop | MarkJenks, I suspect libss7 will require dahdi (or rather: versions of asterisk that support libss7) |
18:51.11 | jblack | um, whoops. I'm sorry. I retract that statement about the rhio. |
18:51.37 | jblack | All I see there now is firmware |
18:51.39 | tzafrir_laptop | MarkJenks, I saw your(?) changes in the dahdi page on voip-info |
18:51.52 | tzafrir_laptop | To unload all modules: /etc/init.d/dahdi stop |
18:51.53 | MarkJenks | yes, I have libss7 and dahdi loaded already. If I put a loopup back on it, I can see it come up with ss7linktest. |
18:52.27 | tzafrir_laptop | You can't tell in advance which modules will be loaded. e.g: which echo canceller module, xpp module, etc. |
18:52.30 | MarkJenks | I never saw that out there. I will take a look at it and remove the wiki if it's the same. |
18:52.34 | *** join/#asterisk bram247 (n=bram@96.28.114.46) |
18:52.48 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
18:53.12 | rs^ | [TK]D-Fender: what's AEL ? |
18:53.21 | MarkJenks | hey, it's about the same! :) |
18:53.25 | [TK]D-Fender | rs^: Google-able |
18:53.41 | rs^ | [TK]D-Fender: can you show me or point me to an example conf ? |
18:53.56 | [TK]D-Fender | rs^: http://www.google.ca/search?hl=en&q=asterisk+AEL&btnG=Google+Search&meta= |
18:53.56 | MarkJenks | If we set the PRI to AMI, all I get from the DMS500 is HDLC Abort. When it' |
18:54.02 | ManxPower | rs^: example conf included in /path/to/src/asterisk/configs |
18:54.12 | jblack | wonders how he screwed up thinking the rhino driver was loaded with sourceless object files. there's clearly none there now. |
18:54.12 | *** join/#asterisk Op3r (n=Edwin@ded-139-109.eglobalreach.net) |
18:54.23 | MarkJenks | s set for B8, it just doesn't want to come up on the switch side. But the switch sees that I'm up. |
18:54.57 | Op3r | hello, Is it possible to create a caller id on per area code on extensions.conf? does anyone knows any script that does this?> |
18:55.00 | MarkJenks | What it looks like I need, is a way to change the dchan to 56k and leave the PRI at 64. |
18:55.04 | ManxPower | MarkJenks: SS7 and PRI require cleanchannel, and ESF/B8ZS provides that. AMI/D4 does not |
18:55.19 | rs^ | ManxPower: sorry but your no help for me you dont even want to I'm happe that there are not so many people around like you ... and thank you very much [TK]D-Fender for trying to help me !! |
18:55.40 | ManxPower | rs^: Best of luck with that. |
18:56.04 | MarkJenks | okay, that explains why AMI isn't working. But something must be missing.. |
18:56.08 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
18:56.17 | MarkJenks | should it be mtp2, dchan, or ? |
18:56.49 | Kobaz | jblack: i've recently dumped the last of my rhino cards |
18:56.52 | Qwell | MarkJenks: there is a README in the libss7 source... |
18:57.00 | Qwell | mtp2=24, bchan=1-23 |
18:57.15 | [TK]D-Fender | rs^: its the 1st link is a silly Google search that I handed you. |
18:57.16 | Qwell | MarkJenks: take a look at that - it's got a bunch of config options there |
18:57.23 | [TK]D-Fender | rs^: You sho no evidence of trying at all |
18:57.25 | MarkJenks | okay, I'll take a quick look. |
18:57.34 | MarkJenks | I'll be right back in 5...... |
18:57.40 | tzafrir_laptop | so it's like pri but with mtp2 instead of dchan/hardhdcl, right? |
18:57.46 | Qwell | *shrug* |
18:57.52 | tzafrir_laptop | Same on E1-s? |
18:58.06 | Qwell | probably replace whatever the D channel would be, with mtp2 |
18:58.27 | Qwell | hell, might even work on BRI...who knows |
18:58.42 | Qwell | err |
18:58.52 | *** join/#asterisk balzac (n=chatzill@199.99.96.227) |
18:58.53 | Qwell | not BRI, because SS7 would replace BRI |
18:59.03 | tzafrir_laptop | the HFC bri chips do the hdlc encoding in hardware. mtp2 uses hdlc as well? |
18:59.09 | Qwell | tzafrir_laptop: no idea |
18:59.36 | *** join/#asterisk jpastore (n=jpastore@69.65.65.40) |
18:59.46 | coppice | mtp2 uses HDLC, but to do it properly you need a couple of features most HDLC channels don't have |
19:01.24 | tzafrir_laptop | well, I figure using ss7 over bri is not such a great idea anyway |
19:02.07 | coppice | what's wrong with SS7 over BRI? it seems a pretty good idea for a lot of people |
19:02.40 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
19:03.09 | jeev | if you use a company like callcentric and some other providers with unlimited calling.. what do they usually provide? 2 outgoing calls at once max? |
19:03.10 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
19:04.12 | Op3r | jeev, really depend on how many channels they will open up for you |
19:04.18 | jeev | hmm |
19:04.24 | ManxPower | jeev: there are no flat monthly rate services or providers, their "unlimited" always has limits |
19:04.49 | jeev | yea, i know manx.. im thinking hrmf.. what's the best way to make sure that if those two channels (assuming they give 2), if they're being used, to use another ITSP? |
19:05.32 | [TK]D-Fender | was wondering what that burning smell was |
19:06.09 | jeev | it's your feet, i set them on fire. |
19:06.35 | km- | you can limit it on your side I think |
19:06.43 | km- | set your account to call-limit=2 |
19:06.56 | km- | and then attempt to dial out on it and then if it fails you can make extension 101 or whatever be your next itsp |
19:07.01 | Op3r | or just use congestion goto another provider |
19:08.07 | rs^ | [TK]D-Fender: thats the only dial I have -> exten => _22X.,1,Dial(IAX2/blah/${EXTEN},,T) |
19:08.22 | km- | ah that reminds me |
19:08.27 | [TK]D-Fender | rs^: Funny last time you said you didn't have a dial. |
19:08.40 | km- | I want to write a transfer-to-blackhole thing for my wife for these damned 800 number callers |
19:08.51 | [TK]D-Fender | rs^: perhaps you should pastebin an entire call. |
19:08.52 | rs^ | [TK]D-Fender: I said i have no DIAL but a Dial ... |
19:08.53 | [TK]D-Fender | ~pb |
19:08.54 | 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 |
19:08.56 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
19:09.09 | jeev | they only speak of incoming channels here: http://www.callcentric.com/faq.php?s_go=1&search=channel&go=Search#172 |
19:09.11 | [TK]D-Fender | rs^: Don't play capitalization games |
19:09.38 | rs^ | [TK]D-Fender: sorry I dont play games I mean it like that ... |
19:09.54 | rs^ | [TK]D-Fender: thast was meant seriously |
19:10.25 | [TK]D-Fender | rs^: pastebin the complete call attempt |
19:11.37 | *** join/#asterisk Gershwin (n=fake@63.250.233.162) |
19:12.49 | km- | d-fender: didn't yaknow? Clinton had "Relations" with monica, but not, RELATIONS. |
19:13.03 | MarkJenks | Sorry about that, had to go talk to the electrician for a project..... |
19:13.42 | tzafrir_laptop | km-, dialplan apps are case insensetive |
19:14.22 | tzafrir_laptop | (as opposed to dialplan functions, whose names are case sensitive and are always CAPITAL) |
19:14.23 | ManxPower | tzafrir_laptop: But FUNCTIONS are not. |
19:14.23 | km- | tzafrir: I think that's bound to rs^, I was being snarky ;) |
19:14.40 | km- | haha |
19:14.46 | km- | so, does nufone still exist? |
19:14.52 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
19:15.05 | [TK]D-Fender | tzafrir_laptop: personally I find that a complete waste. Why make functions case sensitive? Stupid parser.... |
19:15.27 | seanbright | maybe for BC |
19:15.46 | seanbright | less likely to collide with existing variable names |
19:15.47 | MarkJenks | I just looked at README for dahdi, and it looks like I'm all set. I am trying to to it outside of asterisk for testing, using just the ss7linktest. Should I just do it inside of asterisk with chan_dandi? |
19:19.36 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
19:20.08 | MarkJenks | The guys at the switch say that the pri has to be b8.. but the ss7 mtp2 has to be 56 |
19:20.22 | MarkJenks | is there a place to set it up like that? |
19:24.02 | *** join/#asterisk MrNaz (n=naz@122.110.231.8) |
19:27.13 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
19:29.46 | *** join/#asterisk nny_2 (n=Scott_My@64.203.244.146) |
19:30.33 | nny_2 | Anyone who has experience with building in call center requirements to an asterisk system lemme know. May have some work coming up, and I have only set up asterisk for soho |
19:32.10 | Op3r | Hello, here's a question. For example I have 100 did onmy disposal that I want to be used as a Caller ID, then I want it to reflect when I dial a certain area code for example 212, I will also show that my caller ID starts with 212 too. Is there an easy way to do it? |
19:33.05 | [TK]D-Fender | Op3r: its your dialplan, do whatever you want with it |
19:34.31 | Op3r | [TK]D-Fender, yeah I know but its 100 Caller ID numbers I want to reflect. Is there any easy way to just implement it than having to type exten => 91212NXXXXXXXX,1,SetCallerID per caller ID and area code? |
19:34.35 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
19:35.02 | [TK]D-Fender | Op3r: Base your CID on the # dialed. |
19:36.04 | Op3r | [TK]D-Fender, any idea on how? I have 0 clue when it comes to creating macros on asterisk :( |
19:36.52 | *** join/#asterisk Jacco (n=root@unaffiliated/jacco) |
19:36.53 | Jacco | Hey guys. |
19:36.55 | [TK]D-Fender | Op3r: I never said anything about "macros". And you'd better learn to master the dialplan, but that's 95% of Asterisk |
19:37.07 | Jacco | I have two zap lines. One of them will show the caller's number but the other won't. :( |
19:37.09 | [TK]D-Fender | Op3r: Go read the WIKI page on variables, and expressions. |
19:37.15 | Jacco | How do I figure out what's wrong? |
19:37.52 | Op3r | ~wiki |
19:37.56 | [TK]D-Fender | Jacco: Physically verify with a separate analog phone that both lines DO have CID functional |
19:37.58 | [TK]D-Fender | ~wikis |
19:37.59 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
19:38.01 | [TK]D-Fender | ^^^^^^ |
19:38.22 | [TK]D-Fender | Jacco: Then if that checks out, pastebin your zapata.conf. |
19:38.24 | [TK]D-Fender | ~pb |
19:38.25 | 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 |
19:38.27 | [TK]D-Fender | ^^^^^^^^^^ |
19:40.17 | nny_2 | I am working on setting up a quote for an * system with advanced call queue features, for a call center. Their old system was just* a phone system, so I don't need to engineer the complete solution, and at first only want to offer them the basics. (no experimental bs) |
19:40.17 | Jacco | [TK]D-Fender: so... I get an analog phone with callerid and call both lines? |
19:40.19 | Jacco | Hokay, thanks. |
19:40.25 | Jacco | I'll go test that. |
19:40.33 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
19:40.38 | nny_2 | They only need 30 phones fwiw |
19:40.43 | Jacco | although.. oh crap. The phone server might collapse if I disconnect the lines. Probably not though. |
19:41.05 | [TK]D-Fender | nny_2: Clarify "advanced" |
19:41.15 | [TK]D-Fender | Jacco: nope. |
19:41.51 | mgroman | vimperator |
19:44.52 | nny_2 | [TK]D-Fender: hmm comparing this to their old system, so advanced would be queue control with round robin, etc strategies as options. I am still researching what other options would be good to offer them, vs here's everything available. |
19:45.10 | nny_2 | [TK]D-Fender: agent log in / out etc, (afaik this is all native to asterisk) |
19:45.52 | [TK]D-Fender | nny_2: thats actually all basic stuff so far |
19:46.50 | nny_2 | [TK]D-Fender: yeah i need to go poke around and figure out what features a small call center would expect |
19:48.28 | *** join/#asterisk blq (n=Bl@dslb-088-066-229-038.pools.arcor-ip.net) |
19:48.39 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
19:49.14 | *** join/#asterisk heedly (n=heedly@purplehaze.lamedomain.net) |
19:50.09 | *** join/#asterisk VaNNi (n=VaNNi___@38.98.61.143) |
19:50.51 | obnauticus | [TK]D-Fender: know of any projected release date of 1.6 stable? |
19:50.53 | obnauticus | or 1.6rls |
19:51.39 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
19:53.41 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
19:54.09 | jeev | will the passthru on a polycom 330 also distribute PoE? i REALLY need it to! |
19:55.39 | JayTee52 | jeev, NO |
19:56.45 | *** join/#asterisk washburnello (n=johnsy@h23.238.30.71.dynamic.ip.windstream.net) |
19:57.02 | [TK]D-Fender | obnauticus: "when its ready" |
19:57.20 | obnauticus | Who said that? |
19:58.11 | [TK]D-Fender | obnauticus: that the stock answer for anyone crazy enough to ask for dealines on an open-source project. |
19:58.26 | obnauticus | [TK]D-Fender: I know there are no deadlines. |
19:58.28 | obnauticus | lol |
19:58.44 | obnauticus | I'm just wondering if you may have some idea...well maybe with bug reports and stuff it's hard to predict |
19:58.44 | obnauticus | nm |
19:58.47 | obnauticus | I'm just being retarded |
19:58.57 | [TK]D-Fender | obnauticus: I'm guessing 2008, but don't quote me on it. |
19:59.06 | JayTee52 | if you want a reliable release date on open source software try Ubuntu. The release date is reliable but usually the software isn't. |
19:59.20 | obnauticus | Oh god |
19:59.22 | bram247 | 1.6.0-beta9 works pretty well for me ;) |
19:59.25 | obnauticus | I was just makign fun of ubuntu |
19:59.27 | bram247 | as is |
19:59.30 | obnauticus | I made this quote: |
19:59.32 | obnauticus | `Make this channel more sexy. `Trying to gain Linux experience on Ubuntu is like trying to gain BSD experience on a Mac.` |
19:59.39 | obnauticus | oh shi- |
19:59.45 | obnauticus | well... it did make this channel more sexy. |
19:59.54 | JayTee52 | I'm running Gutsy which works very well for me but I've been really disappointed with Hardy on the two machines I've tested so far. |
20:00.01 | obnauticus | eh |
20:00.04 | obnauticus | I'm running Slackware 12.1 :\ |
20:01.12 | *** join/#asterisk deeperror (n=deeperro@76.226.177.255) |
20:02.02 | jeev | crap JayTee52. |
20:02.12 | JayTee52 | I got really spoiled by Ubuntu but I'm now using RHEL 5 64 bit for production * and I much prefer that to running * on 'buntu. If I was going to choose a different distro for production other than RH or one of it's offshoots I'd stick with Debian itself. |
20:02.35 | JayTee52 | jeev, they have pills for that now. Hope everything comes out alright |
20:02.48 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
20:02.50 | jeev | obnauticus, you can't tell people your own quote |
20:02.58 | obnauticus | jeev: how do you know it's mine. |
20:03.01 | jeev | JayTee52, i've actually been having pooping problems.. but oh well |
20:03.06 | obnauticus | I can quote myself |
20:03.07 | jeev | cause you said i made this quote. |
20:03.14 | JayTee52 | jeev, TMI |
20:03.17 | obnauticus | made/found |
20:03.18 | obnauticus | whatever. |
20:03.29 | jeev | you have less credibility than the average gardner |
20:03.34 | obnauticus | I'm quoting myself from another channel. |
20:03.36 | obnauticus | rofl. |
20:03.56 | jeev | JayTee52, know any unlimited outgoing providers with lots of channels ? |
20:04.22 | jeev | ~itsp-us |
20:04.23 | obnauticus | There are none :\ |
20:04.26 | obnauticus | heh |
20:04.27 | jeev | ~us-itsp |
20:04.28 | JayTee52 | jeev, told you before I don't do ITSP's |
20:04.28 | obnauticus | wouldn't they loose money? |
20:04.30 | jeev | ~itsp |
20:04.31 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
20:04.33 | [TK]D-Fender | jeev: "Would you like fries with that, sir?" |
20:04.33 | jeev | bastards |
20:04.39 | JayTee52 | PRI to PSTN FTW!!!! |
20:04.42 | jeev | ~itsplist-us |
20:04.42 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
20:04.48 | obnauticus | ya I have a PRI |
20:04.51 | obnauticus | hehehehheh |
20:04.52 | jeev | fender, dood, seriously. go to astricon, i will send you a shemale hooker |
20:05.09 | obnauticus | jeev: i think if I goto astricon Corydon76-dig will molest me |
20:05.16 | JayTee52 | jeev, how old are you? like 14 or something? |
20:05.26 | bkruse | jeev: you kind find them all over the forums :/ |
20:05.27 | jeev | lol |
20:05.33 | jeev | thanks bkruse |
20:05.37 | jeev | bkruse, you work at digium? |
20:05.42 | bkruse | No. |
20:05.43 | obnauticus | ROFL |
20:06.20 | jeev | what's so funny |
20:06.28 | obnauticus | LFOR |
20:06.46 | obnauticus | Ricers racing on my street just got in a big car accident |
20:06.51 | obnauticus | fail. |
20:07.04 | bkruse | wonders if it's collin |
20:07.40 | jeev | what the hell kind of street do you live on |
20:07.43 | JayTee52 | "Dude! watch my Tokyo drift!!! Oooops!!!! [CRASH!!!!] Waaahhh, waaaahhhhh, MOMMY!!!!" |
20:07.55 | obnauticus | jeev: im in a neighborhood, but close to the outside jeev. |
20:07.57 | bkruse | races on the track |
20:08.05 | obnauticus | so there is a like 4 lane street outside my neighborhood |
20:08.06 | jeev | where do you live |
20:08.07 | obnauticus | that people alwas race on |
20:08.09 | jeev | vegas? |
20:08.12 | obnauticus | SW Washington. |
20:08.19 | jeev | ah |
20:08.29 | JayTee52 | the state or the district |
20:08.35 | obnauticus | it's like a 2.5mi long semiflat road |
20:08.39 | jeev | i was in the district 2 weeks ago.. was cool |
20:08.48 | jeev | humid like a mofo |
20:08.49 | obnauticus | Washington...state. |
20:08.58 | JayTee52 | near Camas or Vancouver? |
20:09.05 | obnauticus | IN vancouver. |
20:09.16 | JayTee52 | used to go there alot on business. |
20:09.22 | obnauticus | heh |
20:09.27 | obnauticus | To centrytel? |
20:09.33 | JayTee52 | got to stay at the Residence Inn on the company dollar |
20:09.35 | obnauticus | I would guess because you are in #asterisk-dev |
20:09.39 | obnauticus | asterisk*** |
20:09.47 | Corydon76-dig | obnauticus: I already told you you aren't getting into my pants |
20:09.53 | obnauticus | Corydon76-dig: ... |
20:09.59 | JayTee52 | no, this was back when I worked for US Cellular |
20:10.02 | obnauticus | Corydon76-dig: I told you, i wasn't expecting you to not put up a fight. |
20:10.14 | obnauticus | JayTee52: ya this area is pretty nice |
20:10.16 | jeev | JayTee52, residence innn sounds like it's 10 bux a night |
20:10.28 | obnauticus | jeev: the operative word is `dollar` |
20:10.30 | obnauticus | company DOLLAR |
20:10.34 | obnauticus | $1 a night. |
20:10.57 | JayTee52 | nope, it's Marriot upscale. It's like getting an apartment. Comes with living room, kitchen and they'll even shop for groceries for you while you're working. |
20:11.01 | obnauticus | Eh, the residence inn by PDX is pretty nice, if that's where you stayed. |
20:11.10 | jeev | ahh |
20:11.18 | *** join/#asterisk pascal_alm (n=pascal_a@static-66-11-93-129.ptr.terago.net) |
20:11.19 | jeev | thought you said their PBX is pretty nice |
20:11.19 | bkruse | jeev: I am waiting on my redbull |
20:11.24 | jeev | it's on the way |
20:11.25 | JayTee52 | It's just down the street from Tony Roma's |
20:11.26 | jeev | assmunch |
20:11.32 | obnauticus | PDX is an airport.. |
20:11.33 | jeev | you guys are extorting me |
20:11.42 | bkruse | notices the star next to his name, which has the power to ban. |
20:11.52 | obnauticus | JayTee52: If you are firmilliar with the salmon creek area. that's whre I am |
20:11.53 | obnauticus | :\ |
20:11.57 | pascal_alm | is there known problem for asterisk voicemail accepting dmtf from blackberry? |
20:12.00 | obnauticus | Corydon76-dig: is probably gmapsing it right now. |
20:12.03 | jeev | i dont see a star, i see an at sign though and i know the capabilities.. so i'll shut my mouth! |
20:12.04 | JayTee52 | been through it. |
20:12.10 | Corydon76-dig | obnauticus: do what? |
20:12.14 | obnauticus | rofl. |
20:12.24 | jeev | bkruse, i told russell that it may come with some anthrax. |
20:12.25 | obnauticus | pascal_alm: not really. |
20:12.39 | obnauticus | pascal_alm: rfc2833 is pretty darn clear |
20:12.39 | bkruse | jeev: I will personally, kill you. |
20:12.46 | jeev | haha |
20:12.53 | russellb | jeev: I have that logged for reference by the FBI as needed |
20:12.57 | JayTee52 | there's a nice jazz club in downtown PDX called Jazz Uno (at least that's what I think it's called, been 8 years) that had good food too. |
20:12.57 | jeev | hahahah |
20:13.03 | jeev | bastages |
20:13.16 | obnauticus | JayTee52: that's where you pick up your bitches when on business eh? |
20:13.16 | pascal_alm | oh... blackberry seems to go through my app ok, but once it gets inside voicemail, app no longer recognizes key presses. |
20:13.35 | russellb | jeev: I'll make you a deal. I'll get you a copy of this _awesome_ telephony toolkit for _FREE_ if you hook me up with a case of sugarfree redbull |
20:13.45 | jeev | haha |
20:13.48 | JayTee52 | obnauticus, no but I did see alot of hookers in that neighborhood. think it's right on the outskirts of Portland's Chinatown |
20:14.00 | bkruse | heck, you know what russellb, I'll even throw in zaptel |
20:14.05 | obnauticus | JayTee52: probably, Portland is pretty fscked up. |
20:14.06 | russellb | oh snap |
20:14.07 | bkruse | out of _my_ pocket |
20:14.12 | bkruse | russellb: not that though :X |
20:14.19 | jeev | russellb, i'll go hijack a truck containing pallets of red bull and bring it to digium |
20:14.20 | bkruse | lol |
20:14.27 | jeev | infact, i'll drive it straight through the doors. |
20:14.33 | bkruse | k |
20:14.41 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
20:14.42 | JayTee52 | obnauticus, I probably wouldn't be picking up hookers even if I could afford them since I just found out from jeev the other day that I'm gay because I rollerblade. |
20:14.47 | obnauticus | Speaking of redbull I went to the portland Flugtag. |
20:14.55 | jeev | you guys will probably be so messed up on redbull, wont even realize what happened |
20:14.58 | obnauticus | ROFL you rollerblade>?! |
20:14.59 | JayTee52 | all this time I thought I preferred women |
20:15.06 | obnauticus | ya me too. |
20:15.07 | jeev | hahahaha |
20:15.09 | russellb | rollerblading is fun |
20:15.10 | obnauticus | Why do you rollerblade? |
20:15.16 | russellb | i haven't done it in years ... |
20:15.16 | JayTee52 | low impact |
20:15.17 | jeev | where is that link that fender sent!!! |
20:15.20 | JayTee52 | I can't jog |
20:15.25 | jeev | "the hardest thing about rollerblading is telling your parents you're gay" |
20:15.32 | JayTee52 | almost lost a leg once in an accident |
20:15.33 | obnauticus | ROFL |
20:16.01 | russellb | #asterisk is off the hook right now |
20:16.03 | *** join/#asterisk GhOnDiE (n=ghondie@dsl78-143-210-38.in-addr.fast.co.uk) |
20:16.04 | bkruse | ok losing a leg in a motorcycle accident, tight, in a rollerblade accident, lame |
20:16.08 | obnauticus | JayTee52: how do you rollerplade with one leg? |
20:16.15 | jeev | hahahah |
20:16.18 | obnauticus | rollerplade? |
20:16.20 | obnauticus | what the hell |
20:16.22 | obnauticus | how;'d i even do that |
20:16.23 | JayTee52 | It's a good workout and I really like it because it reminds me of ice skating. I used to ice skate and play hockey all the time growing up. |
20:16.35 | obnauticus | Do you wear hot shorts too? |
20:16.43 | obnauticus | with large headphones |
20:16.44 | obnauticus | in miami |
20:16.50 | JayTee52 | obnauticus, I still have both legs. I just don't jog because of the constant impact bothers my right leg and knee. |
20:16.59 | russellb | JayTee52: they're all jealous |
20:17.03 | obnauticus | JayTee52: WD40 |
20:17.16 | russellb | too much time on their asses at the compter, rollerblading is too much physical exertion :-p |
20:17.19 | Corydon76-dig | I don't jog, because it's nauseatingly hot outside |
20:17.26 | JayTee52 | I wear earbuds with a SanDisk R250 MP3 Player and I wear cargo shorts. I have very sexy legs. |
20:17.33 | jeev | lol |
20:17.34 | JayTee52 | except for the scar |
20:17.35 | jeev | ew. |
20:17.36 | russellb | i don't jog because it's boring as hell |
20:17.43 | jeev | running is gay too |
20:17.45 | JayTee52 | it's bad for your ankles |
20:17.49 | jeev | i can't even breathe anymore, i've broken my nose a lot. |
20:17.59 | jeev | i just play basketball and i'm not in any tournaments right now |
20:18.02 | russellb | jeev: huh, i wonder why |
20:18.10 | jeev | actually, all sports. |
20:18.18 | JayTee52 | jeev, everything is gay to you because........well.....because you're from california and you're in denial. |
20:18.20 | jeev | i've taken elbows and basketballs in my face |
20:18.24 | obnauticus | I jog because usually when I jog ridiculous amounts of wemon come out of their houses wanting to get with me |
20:18.40 | russellb | "I've taken ... balls in my face". --jeev |
20:18.44 | obnauticus | ROFL |
20:19.32 | obnauticus | the same thing happens to JayTee52 when he rollerblades... except with....men |
20:20.04 | *** part/#asterisk deeperror (n=deeperro@76.226.177.255) |
20:20.51 | JayTee52 | ok, joke if you will but at 40 I was living with a 21 year old hottie and that lasted 3 1/2 years before I dumped her. |
20:21.00 | obnauticus | ... |
20:21.04 | obnauticus | you were 19 when she was born |
20:21.09 | obnauticus | that makes you almost a pedophile. |
20:21.21 | JayTee52 | almost, but still within the legal boundary |
20:21.25 | obnauticus | ... |
20:21.30 | obnauticus | that is disturbing. |
20:21.49 | obnauticus | I am still living with my hand |
20:21.54 | obnauticus | If that's what you wanna call it |
20:21.57 | JayTee52 | no, she was an adult. hell in Arkansas she'd have been considered a spinster |
20:21.58 | jeev | ha ha ha |
20:22.23 | Corydon76-dig | obnauticus: not nearly as disturbing as people who sleep together who are wide enough in age to be grandfather/grandson |
20:22.27 | obnauticus | JayTee52: I bet thta's what you said about all the other 13yr. olds |
20:22.31 | obnauticus | no they were adults! |
20:22.47 | bkruse | what does jeev like about 28 year olds? |
20:22.53 | bkruse | those 20 of them! |
20:23.02 | bkruse | there's * dangit I messed that up. |
20:23.02 | jeev | whatcha mean |
20:23.13 | [TK]D-Fender | lol |
20:23.19 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
20:23.28 | blebleble | i'm looking to have in our ivr ask the user for their account # and if they have it have it intergrate with some type of screen pop to a csr's desk when they take the call, has anyone done something like this or can point me in the correct direction |
20:24.07 | [TK]D-Fender | blebleble: "core show application read" , "core show application dial" |
20:24.18 | [TK]D-Fender | blebleble: "core show application system" |
20:24.54 | blebleble | TK: thank you, ill start reading |
20:25.04 | [TK]D-Fender | ok, checkout time. heading home. |
20:25.30 | obnauticus | bkruse: that is a verbal pun |
20:25.46 | obnauticus | it doesn't work when you type 28 and therefore is clearly a differetn number than 20 & 8 |
20:26.46 | jeev | i'm lost |
20:26.56 | obnauticus | jeev: in reality |
20:26.58 | obnauticus | when someone says |
20:27.11 | obnauticus | What does jeev like about `twenty-eight` year-olds |
20:27.15 | obnauticus | and you say there is 20 of them |
20:27.23 | obnauticus | it changes the last statment so that it means |
20:27.32 | obnauticus | What does jeev like about twenty, eight year olds |
20:27.40 | jeev | oh |
20:27.42 | jeev | lameeeeeeeeeeeeeeeeeee |
20:27.44 | jeev | l4m3 |
20:27.46 | bkruse | obnauticus: Right, but I thought since most people knew the joke, they would get it. |
20:27.47 | obnauticus | doesn't work on IRC. |
20:27.57 | obnauticus | bkruse: canadians dont joke about that sort of thing |
20:27.59 | obnauticus | they actually do it. |
20:28.02 | bkruse | jeev: the fact the you spelled lame "l4m3" says a lot. |
20:28.05 | obnauticus | explains jeev. |
20:28.11 | jeev | haha |
20:28.17 | jeev | i can't believe i did that. sorry |
20:28.29 | jeev | i dont send text messages with 'u' and stuff |
20:28.40 | Corydon76-dig | Just wait until you get old and crothety. You'll understand. |
20:34.31 | *** join/#asterisk deeperror (n=deeperro@76.226.177.255) |
20:36.53 | mgroman | jeev: how old are you? |
20:37.08 | jeev | i'm not sure, like the chinese olympians |
20:37.40 | *** part/#asterisk nny_2 (n=Scott_My@64.203.244.146) |
20:39.14 | russellb | the chinese woman gymnasts looked like they were 9 years old |
20:40.00 | jeev | seriously |
20:40.20 | mgroman | guys, there might be some java programmers here that are offended by that |
20:40.32 | jeev | heh |
20:44.41 | *** part/#asterisk mgroman (n=miles@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
20:45.10 | *** join/#asterisk nighty^ (n=nighty@host217-37-109-17.in-addr.btopenworld.com) |
20:50.12 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
20:52.28 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
20:56.27 | exothermc | Where do I start to debug when my music on hold starts to breakup |
20:57.22 | [TK]D-Fender | exothermc: What source? Where is the caller? |
20:57.49 | exothermc | [TK]D-Fender: Just called into an empty conf bridge (99.999% sure it isn't network related) |
20:57.54 | exothermc | so on net |
20:58.00 | [TK]D-Fender | exothermc: DETAILS DAMMIT |
20:58.18 | exothermc | SIP to SIP g.711u |
20:59.35 | *** join/#asterisk alancio (n=Alancio@10.Red-80-38-197.staticIP.rima-tde.net) |
21:01.03 | kfife | Speaking of MOH, is there a way to have the caller be placed RANDOMLY into a point within one of the music files in the MOH class? If we have 3 professional recordings in the class, a regular customer always hears one of three musicified sales pitches, and never hears the rest. Ideas? |
21:01.27 | kfife | ...hears the BEGINNING and never the END |
21:01.29 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
21:01.41 | EmleyMoor | Has support for rotary phones on FXS ports been dropped? |
21:02.01 | *** part/#asterisk deeperror (n=deeperro@76.226.177.255) |
21:02.18 | kfife | Rotary as in "turn the magneto crank?" :-) Just kidding |
21:02.28 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
21:02.48 | Strom_M | EmleyMoor: not AFAIL |
21:02.50 | Strom_M | er, AFAIK |
21:02.53 | EmleyMoor | Rotary as in loop disconnect signalling by rotary dial |
21:02.58 | EmleyMoor | My dial no longer works |
21:03.05 | Strom_M | EmleyMoor: what kind of card? |
21:03.11 | [TK]D-Fender | Strom_M: looks like a "fail" to me :) |
21:03.15 | EmleyMoor | TDM400P |
21:03.19 | Strom_M | [TK]D-Fender: I'm captain fail |
21:03.32 | Strom_M | EmleyMoor: you may need to tweak the debounce setting in wctdm.c |
21:03.37 | [TK]D-Fender | Strom_M: Aye aye cap'n! |
21:03.56 | bkruse | Strom_M knows all |
21:04.02 | EmleyMoor | Strom_M: Tweaked it. Dial is now "dead", rather than going wrong |
21:04.06 | bkruse | along with [TK]D-Fender, of course |
21:06.35 | EmleyMoor | makes sure the tweaked version is installed |
21:08.19 | jeev | i wonder who changes extensions and sip files at digium |
21:08.21 | jeev | do people fight over it ? |
21:09.15 | bkruse | no. |
21:09.15 | jeev | ahh |
21:09.15 | russellb | we used to have pretty open access to it, heh |
21:09.15 | russellb | i had root on the pbx for a long time ... |
21:09.17 | russellb | but not anymore :( |
21:09.40 | EmleyMoor | Hmmm... dial is now going "wrong" rather than not doing anything |
21:09.42 | russellb | IT manages it. |
21:10.17 | EmleyMoor | Tone dial still works# |
21:10.21 | jeev | ahaha |
21:10.27 | jeev | does it talk shit or what |
21:10.28 | jeev | IT |
21:10.29 | bkruse | russellb: ohrly? I thought you still had root |
21:10.38 | EmleyMoor | will look into this one further |
21:10.38 | russellb | negative |
21:10.52 | russellb | bkruse: our PBX is switchvox now, d00d |
21:10.59 | russellb | web admin FTW |
21:11.03 | bkruse | russellb: Oh right, nvm |
21:11.34 | [TK]D-Fender | EEK |
21:12.00 | russellb | [TK]D-Fender: :-p |
21:12.35 | [TK]D-Fender | That's like commissioning Picasso to draw stick-people in a "colour by numbers" book. |
21:12.47 | russellb | blinks |
21:13.44 | *** join/#asterisk lzhang (n=lzhang@rrcs-67-78-33-170.sw.biz.rr.com) |
21:13.59 | *** join/#asterisk Zemmad (n=calibisi@cpe-66-69-164-90.sw.res.rr.com) |
21:14.05 | EmleyMoor | I've also lost caller ID on my FXO line |
21:14.23 | Zemmad | I'm in need of a solving a AGI problem that i am having |
21:14.52 | lzhang | question: I have 2 pri's from my provider, and they have been trunked together. Does this mean I need to make 1 channelgroup instead of 2? |
21:16.16 | *** join/#asterisk washburnello (n=johnsy@h23.238.30.71.dynamic.ip.windstream.net) |
21:18.12 | *** part/#asterisk washburnello (n=johnsy@h23.238.30.71.dynamic.ip.windstream.net) |
21:18.28 | *** join/#asterisk Entr4nced (n=IMG001@h160.145.31.71.dynamic.ip.windstream.net) |
21:18.58 | *** join/#asterisk wonderworld (n=ww@ip-62-143-216-14.hsi.ish.de) |
21:19.29 | [TK]D-Fender | lzhang: Clarify your definition of "trunked |
21:20.15 | EmleyMoor | I don't seem to be able to call festival any more |
21:22.10 | EmleyMoor | Any clues on that? |
21:22.38 | lzhang | [TK]D-Fender: apparanted they are merged in some way, so they can roll over to each other |
21:23.17 | [TK]D-Fender | lzhang: "in some way". Doesn't say much. You can have roll-over on otherwise separate PRI's. |
21:23.24 | EmleyMoor | Just done a Debian upgrade and it's now failinng to configure zaptel - what should I do? |
21:25.19 | EmleyMoor | Caller ID is coming in, but not being processed correctly, so that's OK |
21:25.37 | EmleyMoor | What's gone wrong with festival, though? |
21:27.27 | *** join/#asterisk justmehere (n=justmehe@24-176-158-178.dhcp.kgpt.tn.charter.com) |
21:28.19 | *** part/#asterisk justmehere (n=justmehe@24-176-158-178.dhcp.kgpt.tn.charter.com) |
21:28.29 | lzhang | [TK]D-Fender: the actual problem I am having is that some outbound calls are hanging (dead air)... then I end up seeing this in the error log chan_zap.c: Ring requested on channel 0/10 already in use on span 2. Hanging up owner. |
21:28.52 | lzhang | this only started happening after this morning, when my provider came in and added a second pri |
21:31.10 | lzhang | when I check the active channels while one of these problem calls are going on, I see Local Extension -> (the outbound number) and then a call in from (the outbound number) -> internal |
21:31.39 | lzhang | this leads me to believe that there is some sort of mixup when it's trying to rollover, or that I've misconfigured my channel groups somehow |
21:32.19 | *** join/#asterisk barakuda (n=baraka@78.158.192.203) |
21:35.18 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
21:38.09 | jeev | a |
21:46.08 | jeev | damn, cdrtool is a little headache isn't it. i have to start that tonight |
21:49.05 | jeev | http://www.gamma.ru/~avk/ anyone use that? |
21:49.07 | jeev | ~cdr |
21:49.08 | jbot | methinks cdr is Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw |
21:51.07 | *** join/#asterisk Entr4nced (n=IMG001@h160.145.31.71.dynamic.ip.windstream.net) |
21:51.25 | TJNII | Hey, according to Moscow the US started the war in Georgia! Go USA! |
21:52.53 | *** join/#asterisk duncan_16 (n=richard@pool-141-158-248-35.phil.east.verizon.net) |
21:53.36 | drmessano | I live in Georgia, and I have seen no war |
21:53.40 | duncan_16 | hi all.. was wondering if you could assist me with a quick query. I'm trying to install an RPM into the asteriskNOW system, and cannot... Any suggestions? It was installed from the ISO that we downloaded at the asteriskNOW site. |
21:53.45 | drmessano | I'm calling it Shenanigans |
21:53.50 | errr | lol |
21:54.03 | jeev | i believe russia TJNII. |
21:54.07 | drmessano | "unable to reproduce" |
21:54.28 | jeev | maybe russia should've warned the US not to invade iraq |
21:54.55 | drmessano | "user reported war in georgia" |
21:55.04 | duncan_16 | to be more specific, we are trying to get the HUDLite server running on the asteriskNOW system. |
21:55.04 | drmessano | "went outside, observed no war." |
21:55.12 | drmessano | Ticket Closed: Unable to reproduce |
21:55.37 | drmessano | Hudlite sucks |
21:55.40 | drmessano | Use astassistant |
21:55.56 | duncan_16 | ok. We looked at HudLight... seemed like what we wanted. |
21:56.07 | drmessano | Suit yourself |
21:56.15 | drmessano | You'll be sorry |
21:57.08 | duncan_16 | hrm. Why would we be "Sorry"? Need to ask the noob questions on this... so that I can get learned on the system ;) |
21:58.41 | drmessano | Hudlite uses some pseudo XML crap spewed forth by a perl bot running on an IRCd.. your clients are connecting to the IRC backend and attempt to interface to it |
21:58.54 | drmessano | It's some shit I would write on a bored weekend, not professional software |
21:59.08 | drmessano | It's buggy as hell, almost unsupported, and GREEN |
21:59.45 | drmessano | When a feature breaks in Hudlite, fonality removes the feature |
21:59.49 | drmessano | Forget fixing it |
22:00.15 | drmessano | Its a crap application that the devs know is crap that even they won't use |
22:00.19 | duncan_16 | ok... I just downloaded the ASTassist. Looking at how it configs. What would we need to do with a stock install of the asteriskNOW server to run ASTassistant? |
22:00.40 | heedly | wow.... I thought you were joking at first. |
22:00.52 | heedly | and then I See ircd-hybrid as a requirement |
22:00.53 | drmessano | Set up manager accounts in manager.conf, assign them user rights, let the client connect |
22:00.56 | drmessano | Yes |
22:01.01 | drmessano | Its a HORRID hack |
22:01.12 | heedly | gui looks pretty |
22:01.30 | duncan_16 | we were partially interested because of the integration with Outlook. |
22:01.48 | drmessano | hudtapi freezes outlook |
22:01.53 | drmessano | Doesnt work half the time |
22:02.03 | drmessano | Or sends crap dial strings |
22:02.05 | duncan_16 | ahh. what about ASTAssistant? |
22:02.13 | drmessano | It's rock solid |
22:02.41 | drmessano | Works with AMI, so no need for some crap middleware to make it work |
22:03.08 | *** part/#asterisk Assimilate (n=Assimila@216.83.78.108) |
22:03.09 | duncan_16 | oh... ok. So you can do the same functionality that you could with their piece, minus the outlook integration then, correct? |
22:03.44 | drmessano | AstAssistant works with your Outlook Address book |
22:03.53 | drmessano | Just doesnt have a tapi connection |
22:03.58 | drmessano | tapi sucks for the most part |
22:04.04 | drmessano | Well I take that back |
22:04.11 | drmessano | Theres few good implementations |
22:04.49 | drmessano | But most work for one specific Outlook version.. and then the app becomes buggy when a new MS Office is released |
22:05.13 | drmessano | "Oh, it installs on Outlook 2009.. Hmm.. but it send my credit card number to thailand" |
22:05.16 | drmessano | Shit like that |
22:06.01 | drmessano | I seriously doubt the Outlook functionality in Hudlite works worth a crap in Outlook 07 |
22:06.17 | drmessano | It barely worked in 2003.. I doubt they put the resources into updating it |
22:06.22 | duncan_16 | heh. |
22:07.12 | drmessano | AstAssistant has chat functionality built in that can use an IRC server.. but thats all it uses it for.. the chat backend |
22:07.22 | drmessano | Not as the API |
22:07.39 | duncan_16 | Here's the deal. I do consulting for a P.O.S. company... they are implementing it in their office, and we are working close with them, since some of our product offerings are in-line with theirs. We all just wanted to have a decent way of doing call switching/routing... it seemed as if the HUD piece would have done what we wanted. |
22:07.40 | drmessano | Not sure I have ever seen a Telephony Internet Relay Chat API before |
22:08.05 | duncan_16 | but... as I'm evaluating this piece a bit... I think it may be the solution. |
22:08.27 | drmessano | AstAssistant is a great app, and the developer is very responsive |
22:12.28 | duncan_16 | I guess I'll have to drop a line to the dev guy and get some info on how to connect with the outlook contacts. |
22:12.49 | duncan_16 | we were also considering configuring up with LCS '07, since we're a M$ Gold Partner, but dunno if we want to go that far. |
22:15.52 | alancio | what is LCS? |
22:16.15 | scooby2 | Whats the best place to get * documentation? The wiki is quite out of date as is the documentation on digium.com. |
22:17.17 | ManxPower | scooby2: /path/to/src/asterisk/doc |
22:17.22 | ManxPower | It's a secret, so don't tell anyone! |
22:17.33 | scooby2 | lol |
22:17.38 | alancio | http://downloads.oreilly.com/books/9780596510480.pdf |
22:17.59 | ManxPower | Tha asterisk book is a littel out of date, but still the best reference outside of the doc dir. |
22:18.11 | ManxPower | Also, I can't seem to type today. |
22:18.33 | alancio | why? you only misstyped a littel |
22:18.50 | ManxPower | But ANY time docs conflict with the info in /doc then the info in /doc should be considered correct. It is THE place for Asterisk doc. |
22:19.15 | scooby2 | I was just looking for some more explanation on the new queue/agent system. the doc dir explanation is pretty tough to understand |
22:19.54 | scooby2 | queues-with-callback-members.txt |
22:20.00 | ManxPower | scooby2: The queue/agent system is hell to understand regardless of the Asterisk version 8-| |
22:20.17 | duncan_16 | alancio: Live Communications Server 2007 from Microsoft. |
22:20.23 | barakuda | scooby2, what version ? |
22:20.35 | ManxPower | We have almost totally eliminated queues from my systems, but we are not a call center and don't have complex queue needs. |
22:21.18 | scooby2 | 1.4.21.2 I'm getting the warning about "AgentCallbackLogin is deprecated and will be removed in a future release." |
22:22.42 | scooby2 | so i figure that I might as well fix it now. Trying to upgrade from 1.2 |
22:23.35 | barakuda | does your agents sip peers ? |
22:24.26 | scooby2 | yes |
22:24.35 | barakuda | we wrote down in our configs that the members of the queue is directly sip peers of operators extension |
22:24.39 | barakuda | like tha |
22:24.40 | barakuda | t |
22:24.42 | barakuda | member => SIP/402 |
22:25.13 | barakuda | so we have extension registered as queue member, just right after it's registration on asterisk |
22:25.14 | scooby2 | ahh, no |
22:25.27 | scooby2 | member = Agent/535 |
22:25.52 | barakuda | we wiped this out of queue.conf |
22:26.14 | scooby2 | can they move to different locations that way? |
22:26.21 | barakuda | <PROTECTED> |
22:26.44 | barakuda | yes, they just enter their sip login each time at differnet places |
22:26.58 | scooby2 | that sounds nice |
22:27.24 | barakuda | i've had same warnings |
22:28.00 | barakuda | and i've had to interact with * when i had to register as agent |
22:28.09 | barakuda | now this problems went away |
22:28.43 | barakuda | i register extension, and autmatically, asterisk pushes queue calls for me |
22:32.10 | scooby2 | sounds like what I need |
22:32.26 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
22:33.28 | *** join/#asterisk Greek-Boy (n=email@41.221.58.7) |
22:34.40 | EmleyMoor | I'm in the middle of applying a patch to make UK caller ID work |
22:37.06 | EmleyMoor | wonders how long this package is going to take to build |
22:39.38 | *** join/#asterisk icel (n=mail@c-76-113-115-132.hsd1.nm.comcast.net) |
22:40.13 | EmleyMoor | I wanted to test my upgraded asterisk out but found a showstopper I needed to patch for |
22:40.25 | *** part/#asterisk lzhang (n=lzhang@rrcs-67-78-33-170.sw.biz.rr.com) |
22:41.52 | EmleyMoor | Is there a list anywhere of what Asterisk 1.4 allows but considers deprecated? |
22:42.11 | [TK]D-Fender | lzhangUPGRADE.TXT |
22:42.15 | [TK]D-Fender | EmleyMoor: rather |
22:42.31 | EmleyMoor | I will have a look |
22:43.29 | EmleyMoor | I know my dialplan is 1.4 safe, and some "deprecated in 1.4" stuff that wasn't even needed in 1.2 is already clear |
22:44.18 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
22:45.21 | *** part/#asterisk mercutioviz (n=chatzill@freeswitch/developer/msc) |
22:47.18 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
22:52.35 | ManxPower | [TK]D-Fender: make install needs to just cat UPGRADE.txt |
22:59.35 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
23:06.52 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
23:08.53 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
23:13.50 | *** join/#asterisk Greek-Boy (n=email@41.221.58.7) |
23:17.20 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
23:19.17 | EmleyMoor | Found my first deprecated thing |
23:22.15 | EmleyMoor | My FXO line is staying open if someone calls it and a SoftHangup occurs during that time |
23:22.33 | EmleyMoor | (and they don't hang up in consequence) |
23:31.16 | *** join/#asterisk deeperror (n=deeperro@d149-67-49-94.try.wideopenwest.com) |
23:51.19 | *** join/#asterisk Johnakabean (n=Johnakab@pool-72-82-114-148.nrflva.east.verizon.net) |
23:52.18 | Wayhigh | is having a really hard time finding a local milliwatt line |
23:52.47 | Johnakabean | wonders why wayhigh needs 1000 killowatts |
23:54.02 | drmessano | milliwatt is not 1000 kilowatts |
23:54.05 | Johnakabean | <PROTECTED> |
23:54.08 | seanbright | it's 1002 |
23:54.31 | Wayhigh | john: a milliwatt test line is a line that puts out 1004Hz tone at 0dBm.. it's used for testing line loss |
23:54.33 | drmessano | Always high on my list of qualifications for a tech |
23:54.37 | drmessano | "Bad at math" |
23:54.47 | seanbright | 1011 i mean |
23:55.04 | Wayhigh | they're otherwise known as type 102 |
23:55.21 | Wayhigh | 11? |
23:55.33 | seanbright | 17, then. |
23:55.36 | Johnakabean | pulls out a crash cart, charges to 500, and hands it to wayhigh |
23:55.48 | Johnakabean | you want two to make 1000? |
23:56.23 | deeperror | 1000? |
23:57.15 | Johnakabean | anyway, do i have to specify the DiD along with the cid or can i just use the cid |
23:57.32 | Johnakabean | it straight up ignores everything except for DID's |