| 00:00.23 | underdog | ye I am....trying to learn the system, so I just watned to get it up and running...I'm digging through the conf files now figuring out how things works |
| 00:00.51 | ManxPower | underdog, you will not learn asterisk by using a prebuilt system like FreePBX/TrixBox, etc |
| 00:00.58 | ManxPower | you will learn THOSE systems, but not Asterisk |
| 00:01.29 | underdog | no I'm digging though the asterisk configuration files now |
| 00:01.52 | ManxPower | you won't learn much asterisk from those files. |
| 00:04.20 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 00:04.43 | russellb | Besticles: did you have another question? |
| 00:04.48 | underdog | well, reading the O'Rielly book may help also |
| 00:04.56 | Besticles | Yeah, I just don't know how to word it. |
| 00:05.05 | Besticles | I definitely have a question, though. :x |
| 00:05.08 | russellb | Besticles: hehe |
| 00:05.17 | Besticles | Okay |
| 00:05.22 | Besticles | So I understand what FastAGI is. |
| 00:05.31 | russellb | that's a good start :-) |
| 00:05.32 | Besticles | And I fully understand the limitation that I just ran into with the Dial Exec |
| 00:05.56 | Besticles | Even if I override my code to ignore the result code, it wont do me any good because Asterisk will ignore any future commands |
| 00:06.02 | Besticles | Until the call finishes, correct? |
| 00:06.22 | Besticles | If DeadAGI allows me to execute code while the channel is on hook... |
| 00:06.25 | Besticles | THen whats the opposite |
| 00:06.38 | ManxPower | beardy, have your script either use .call files or Manager Originate |
| 00:06.54 | Besticles | Well |
| 00:06.55 | russellb | Besticles: you can use the Asterisk manager interface from your application, as well |
| 00:06.56 | ManxPower | those are the two main ways to make asterisk call something/somewhere without blocking |
| 00:07.00 | russellb | if you want to do some asynchronous operations. |
| 00:07.05 | russellb | such as getting/setting some variables. |
| 00:07.14 | beardy | looks around |
| 00:07.19 | russellb | what would you like to do while the call is up between the 2 channels that you feel limited from doing? |
| 00:07.39 | Besticles | Well, let me paint the picture of what I am doing exactly. |
| 00:07.46 | Besticles | I am using AMI to originate the call to DAHDI |
| 00:08.23 | Besticles | oh wait. |
| 00:08.24 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 00:08.26 | Besticles | I dont have a problem. |
| 00:08.31 | russellb | heh |
| 00:08.33 | Besticles | Listen |
| 00:08.34 | russellb | yay! |
| 00:08.41 | Besticles | I am exec dialing My 2nd box |
| 00:08.47 | Besticles | Then using FastAGI again on the second box |
| 00:08.58 | Besticles | To dial the agent that resides on that second box. |
| 00:09.02 | Besticles | But I might have a problem. |
| 00:09.06 | Besticles | Like recording the call |
| 00:09.14 | russellb | you set up recording before the Dial |
| 00:09.21 | Besticles | Cooool. |
| 00:09.35 | russellb | you tell Asterisk to plz record this call kthx, and then proceed to Dial |
| 00:09.49 | russellb | you can allow the agent to start/stop recording dynamically during the call, too, if you'd like |
| 00:10.20 | Besticles | Cool. |
| 00:10.31 | Besticles | I haven't dipped into call monitoring.. |
| 00:10.32 | russellb | but in either case, you set it up before dial |
| 00:11.58 | russellb | or if you really had to (though this would be rare, i would think) you could asynchronously start call recording from your application using the AMI |
| 00:12.17 | Besticles | I think I am going to do that. |
| 00:12.31 | Besticles | You know the only other thing I ran into that I am queezy with Asterisk |
| 00:12.39 | Besticles | is that I send the Originate with AMI with a timeout |
| 00:12.52 | Besticles | The timeout starts as soon as I send the cmd. |
| 00:13.17 | Besticles | I am basically creating a dialer / ivr solution, so I dont want the timeout to be too long. |
| 00:13.23 | Besticles | But I've noticed that if I make it too short, |
| 00:13.32 | Besticles | And if the box is running slow, that it won't even make the call. |
| 00:13.50 | Besticles | Have you noticed that? |
| 00:13.56 | russellb | I can't say that I have. |
| 00:14.10 | russellb | I'd have to look in more detail at how your application works and such ... |
| 00:14.22 | Besticles | It's all Async AMI/AGI. lol. |
| 00:14.37 | russellb | you're using Async AGI? |
| 00:14.49 | Besticles | Honestly Russel, you are the man. You put my mind at ease. For a second there I was worried that I was writing code that I couldn't use. |
| 00:14.54 | Besticles | No, FastAGI. |
| 00:14.58 | russellb | ok. |
| 00:15.00 | Besticles | Async on my end. |
| 00:15.00 | russellb | heh, thanks. |
| 00:15.14 | russellb | it's a useful discussion to have, actually |
| 00:15.23 | russellb | i'm working on some AGI documentation at the moment ... |
| 00:15.32 | Besticles | Ah cool. |
| 00:15.53 | Besticles | I went to Jared's class over here in Las Vegas. We covered alot, but AGI/AMI was very little in scope. |
| 00:16.08 | russellb | cool. Yeah, there is quite a bit in Asterisk |
| 00:16.14 | russellb | AGI/AMI is one of the more advanced topics. |
| 00:16.21 | Besticles | Yeah. |
| 00:16.50 | russellb | at least anything you learn about the dialplan still applies to AGI programming |
| 00:16.54 | russellb | so you know what applications do what |
| 00:16.59 | russellb | AMI is its own beast, that's for sure |
| 00:17.19 | Besticles | Yeah. I log the communication back and forth of AMI. |
| 00:17.30 | Besticles | Alot of information going back and forth. |
| 00:17.32 | *** join/#asterisk guilhermebr (~Guilherme@189.63.50.214) |
| 00:18.03 | russellb | quite |
| 00:18.10 | russellb | especially if you turn on all the events ... |
| 00:18.24 | Besticles | lol |
| 00:18.50 | Besticles | The only other thing that I need to figure out was that he had his boxes preconfigured for TTS already setup, and it appears my installation doesn't have it installed, I am sure I can figure it out with alittle googling. |
| 00:19.25 | russellb | I would recommend using Cepstral. |
| 00:19.54 | russellb | and if you want it to sound like the rest of the prompts included with Asterisk, http://www.digium.com/en/products/software/cepstral.php |
| 00:20.20 | russellb | you can play with some samples on that site |
| 00:20.24 | Besticles | Nice thanks. |
| 00:20.54 | Besticles | I just have one last question, I promise. |
| 00:21.08 | Besticles | Is EXEC DIAL application the only application that does not return a result code? |
| 00:21.15 | Besticles | (it does, just not immediate, i know) |
| 00:21.21 | russellb | no no, it does return a result |
| 00:21.24 | ManxPower | Besticles, if it blocks in the dialplan it will block in AGI |
| 00:21.25 | russellb | oh ok, sorry |
| 00:21.41 | russellb | it just takes however long it takes the app to run ... lots of things take a while |
| 00:21.52 | russellb | MeetMe(), Queue(), Voicemail(), a huge list |
| 00:22.16 | Besticles | Thanks guys. |
| 00:24.39 | russellb | you're quite welcome. |
| 00:24.42 | russellb | good luck to you. |
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| 00:33.20 | Besticles | Crap I lied, question. Originating the call using AMI, is there a way I can get Asterisk to determine a Answering Machine? |
| 00:34.29 | [TK]D-Fender | Besticles: "core show application AMD" |
| 00:34.35 | MRH2 | hi if i have a live sip call that shows "Scheduling destruction of call..." on the CLI does theat mean the peer is not reachable - even though the media is still live |
| 00:35.02 | MRH2 | this is a call via a sip gateway with qualify =bla |
| 00:36.06 | ManxPower | MRH2, turn off debugging and I don't think you will see that message anymore. |
| 00:36.11 | Besticles | Thanks Fender |
| 00:36.20 | MRH2 | aye i am trying to get to the bottom of a dropped call |
| 00:39.09 | russellb | Besticles: reliability will kind of suck, no matter what you do. detecting answering machines is an unwinnable battle. |
| 00:39.32 | ManxPower | People have gotten rich by writing better AMD |
| 00:39.42 | *** join/#asterisk imroot (~imroot@201.86.172.223.dynamic.adsl.gvt.net.br) |
| 00:40.08 | Besticles | Yeah, we've experience that with our current Dialer/IVR application made by Dialogic/Envox. Some is better than nothing. |
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| 01:30.32 | underdog | can you wildcard the blacklist "list"...i.g. _222XXXXXXX to block all calls from 222 area code? |
| 01:30.55 | underdog | seen examples of people saying you can do this...but tried it and it didn't appear to work |
| 01:36.11 | [TK]D-Fender | underdog: What "blacklist"? Where do you have this implemented? |
| 01:36.35 | p3nguin_ | I know you can use it in caller ID matching, but if you're talking about BLACKLIST(), I have no idea. |
| 01:39.34 | underdog | database blacklist put <CID> 1 |
| 01:39.37 | underdog | in the AstDB |
| 01:40.26 | [TK]D-Fender | underdog: That doesn't support patterns, and is near useless. |
| 01:40.37 | [TK]D-Fender | underdog: And you have to actually call it from the dialplan yourself |
| 01:41.06 | [TK]D-Fender | underdog: Better for you to do yourself |
| 01:41.30 | Heldwin | I am looking to buy some equipments, and look with my ISP to get the caller phone number, but is there a way to use asterisk as an intercepter for analogic calls ? line -> something -> asterisk -> ATA -> anloagic phone ? |
| 01:41.36 | Heldwin | analogic* |
| 01:42.15 | [TK]D-Fender | Heldwin: Yes, there are tons of devices to let * use analog lines |
| 01:42.33 | [TK]D-Fender | Heldwin: Lots of PCI type calds, Analog FXo ? SIP gateways, etc |
| 01:42.37 | [TK]D-Fender | > |
| 01:44.51 | Heldwin | ok thank you :) I found a few ata box for the asterisk -> analogic phone, but not sure yet what I need for the line -> asterisk |
| 01:45.18 | Heldwin | an ata box will work too ? |
| 01:45.43 | [TK]D-Fender | Heldwin: Typically the term ATA is reserved for devices you plug phones into, but not lines. |
| 01:45.48 | p3nguin_ | You can use any ATA that provides at least one FXO port plus Ethernet. |
| 01:46.09 | Heldwin | I cannot seem to be able to buy a pci card, because site like digium, but it seems they ask for a credit card, and I don't have |
| 01:46.15 | [TK]D-Fender | Heldwin: Opposite purpose. Linksys SPA-3102 is a device that gives you 1 FXS, and 1 FXO. Might do the job for you |
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| 01:46.41 | Heldwin | ok thanks. I saw a few linksys on a reseller around here |
| 01:48.52 | Heldwin | they have the spa 3102 |
| 01:51.13 | [TK]D-Fender | . |
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| 02:00.02 | exothermi | Is there anyway to set how long between hanging up a dahdi channel and when it becomes available to the group dialing pool? |
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| 02:11.04 | oshkosh | Hi everyone hoping for assistance. I enabled a pinset on my only outbound route, and enabled pinless dialing on each of my extensions except for 2. Problem is if I call forward or use follow me now, when people call in, they're prompted to enter their password followed by pound since I guess it's forwarding using the outbound route... |
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| 02:55.47 | corey` | Hi, I need to create a dial plan where by a user can set the destination of an extention by dialing another extention proceeded by the the digits of the desired destination. which dial plan application command would I need to use? |
| 02:57.38 | ChannelZ | I think you'll have to roll that yourself |
| 02:57.43 | ChannelZ | ~book |
| 02:57.43 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 02:57.55 | ChannelZ | see the "hot-desking" section if memory serves |
| 03:00.14 | ChannelZ | (but depending on the devices you use, they might have built-in 'call forward' features of their own |
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| 03:02.20 | corey` | thank you ill check out the book |
| 03:02.38 | underdog | does zapateller play a 3 tone SIT? |
| 03:02.54 | underdog | mine is playing a single beep...not sure that is correct in researching SITs |
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| 03:18.55 | b14ck | Hi all. Can someone tell me how I might tell if an AMI originated call was: BUSY, NOANSWER, or some other condition code? I'm trying to figure out how to get the call status. |
| 03:35.15 | ManxPower | you don't want to use the built in call forward or dnd of the phones. |
| 03:35.39 | ManxPower | users will set those options, then they will forget they set them and call into the helpdesk screaming about your stupid phone system being broken AGAIN. |
| 03:38.21 | Nugget | ManxPower is wise. |
| 03:38.26 | Nugget | ignore him at your peril |
| 03:38.31 | underdog | figured it out...needed to Wait() after the answer prior to playing SIT |
| 03:39.25 | underdog | you just want to round robin call forwarding...keeps things interesting |
| 03:44.34 | ManxPower | corey`, exten => _303XXXX,1,Dial(Zap/g1/{EXTEN:3}) user dials 303 + number to dial, the dial strips off the first three digits |
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| 04:13.24 | underdog | can see why asterisk GUIs are such a pain to work with....the more I start trying to customize asterisk and the GUIs keep wanting to overwrite my changes |
| 04:14.07 | ManxPower | underdog, now you understand why you should be on #FreePBX or whatever gui you are using. |
| 04:14.23 | underdog | or both |
| 04:14.37 | underdog | just have to know which one is having the "issue" |
| 04:14.42 | ChannelZ | or not using it in the first place |
| 04:14.45 | underdog | heh |
| 04:14.56 | underdog | touche |
| 04:14.58 | underdog | touche |
| 04:15.50 | underdog | it's all good and I can take the "you idiot, that's not an asterisk" comments |
| 04:16.14 | ChannelZ | People who want to learn Asterisk shouldn't be downloading AsteriskNOW which sadly is not really stated anywhere |
| 04:16.51 | ManxPower | underdog, but you don't understand enough at this point to even know WHICH part, Asterisk or your GUI is having the problem. |
| 04:17.09 | underdog | i think most of the time 99% will be the GUI |
| 04:17.33 | ChannelZ | goes to watch Rubicon |
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| 06:47.37 | schmidts | good morning |
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| 06:53.35 | ChannelZ | good? it's monday. |
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| 07:06.48 | schmidts | :D thats right |
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| 07:37.09 | elliot98 | where can I search the irc archives? |
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| 07:39.06 | ijpalmer | Hi all, I've got forced answering setup using SipAddHeader but when the agent is using a headset I need it to beep in their ear a couple of times before a call is presented to her, does anyone know how to do this. Thanks |
| 07:42.09 | jql | see M(x) for Dial |
| 07:43.42 | jql | err, according to TFM, the A(x) option would be easier |
| 07:43.42 | *** join/#asterisk |StyleDim| (~StyleDim@217.15.188.82) |
| 07:43.46 | jql | *shrug* |
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| 07:50.30 | ijpalmer | Thanks, jql, sometimes when it's staring you in the face you don't see it |
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| 08:41.57 | elliot98 | where are the irc archives kept? is it in searchable format? |
| 08:59.16 | schmidts | we still dont know elliot ;) |
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| 09:15.02 | russ | the irc archives |
| 09:15.06 | russ | that is awesome |
| 09:15.25 | russ | unless of course he just means the #asterisk archives, then it is just a regular boring question |
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| 09:29.58 | elliot98 | I seem to recall a website with all the irc archives... |
| 09:35.17 | elliot98 | russ: hehe, just the asterisk archives, plus asterisk-dev |
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| 09:37.08 | kaldemar | elliot98: from uncle google: http://ibot.rikers.org/%23asterisk/ |
| 09:37.27 | elliot98 | that's it! |
| 09:37.29 | elliot98 | thanks! |
| 09:38.39 | Jasnejac | that's a useful thing to know, cheers |
| 09:39.16 | elliot98 | just wondering if it is searchable too! |
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| 09:49.11 | *** join/#asterisk stix (~stix@firewall.o4.dk) |
| 09:50.08 | stix | Hi guys. How is it possible to put a call on "hold"? As I see it, the term doesn't even exist on Asterisk. Should I use the "park" in stead? |
| 09:52.14 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 10:08.04 | Chainsaw | stix: Going on hold is generally a signal that your SIP (soft) phone sends out. |
| 10:08.15 | Chainsaw | stix: It is supported; complete with music. |
| 10:08.16 | stix | yes I see |
| 10:09.02 | stix | so how can I do it from the AMI? |
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| 10:54.55 | schmidts | ARGS what could be the reasion, i cant register from one asterisk to another after i have done a upgrade vom 1.6.2 to trunk? |
| 10:55.06 | schmidts | i allways get an 401 unauthorized |
| 10:55.15 | Chainsaw | schmidts: Trunk is not supported, go for the 1.8 branch instead. |
| 10:56.09 | schmidts | :P my problem is, i had to develop my patch for trunk, but i cant test it without registering a user (or more) to it |
| 10:57.15 | elliot98 | when I upgraded from 1.4.18 to 1.4.33, it seems fax passthrough started to fail |
| 11:02.05 | *** join/#asterisk timahvo1 (~rogue@41.191.224.178) |
| 11:04.48 | Naikrovek | yawns |
| 11:05.42 | *** join/#asterisk WWGD (~WWGD@208.79.14.130) |
| 11:05.42 | *** join/#asterisk coppice (~chatzilla@m121-202-83-188.smartone-vodafone.com) |
| 11:05.47 | Chainsaw | elliot98: Try to bisect it. Make smaller steps. |
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| 11:23.04 | schmidts | chainsaw, same in 1.8 ;) |
| 11:28.12 | schmidts | thats my sip config on the 1.8 system |
| 11:28.13 | schmidts | http://pastebin.com/RKquh6yw |
| 11:28.40 | schmidts | the real strange this about this, a cisco spa phone can register against 1.8 but a 1.6 asterisk cant :( |
| 11:31.04 | elliot98 | Chainsaw: I remember I needed to apply a patch that worked |
| 11:31.35 | elliot98 | when I upgraded one time before from 1.4.18 to 1.4.33.1 |
| 11:31.58 | elliot98 | basically, another server techie made some changes overwriting the patch |
| 11:32.17 | elliot98 | so I'm trying to find it again |
| 11:34.10 | TobSnyder | where can I find a list of all voiceprompts with text |
| 11:34.43 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
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| 11:44.26 | tzafrir_laptop | TobSnyder, the Digium ones have a text file with the prompts in the tarball |
| 11:44.35 | tzafrir_laptop | Not sure about the others |
| 11:46.31 | coppice | tzafrir_laptop: the xorcom.com site still seems to have leftover stuff about Howler |
| 11:49.06 | TobSnyder | ok I will check them |
| 11:49.23 | TobSnyder | when tying to make an outoging call I get Destination out of order (27) |
| 11:49.32 | TobSnyder | but the number dialed work 2 hours ago |
| 11:49.39 | TobSnyder | so what does this mean? |
| 11:56.34 | TobSnyder | ok tried with mobile phone, does not work either, seems that the destination is indeed not reachable |
| 11:56.49 | TobSnyder | but than the "Error" message of asterisk is a bit confusing |
| 11:57.23 | TobSnyder | (all circuits bus now) |
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| 13:12.45 | *** join/#asterisk Apocalipse (~Daniel@200.198.204.68) |
| 13:12.51 | Apocalipse | hello people... |
| 13:13.09 | Apocalipse | how can i change codec definition in asterisk? |
| 13:13.37 | Naikrovek | codec definition? |
| 13:13.41 | Naikrovek | you want to change which codec is used? |
| 13:14.02 | Apocalipse | yes |
| 13:14.09 | Naikrovek | 1) from asterisk console "core show translation" to see which codecs are available |
| 13:14.41 | Naikrovek | do you want to change the codec that the phones use or that goes out the trunk to the provider (in the case of the voip trunk) |
| 13:15.16 | Apocalipse | change codec for phones... |
| 13:15.24 | Naikrovek | what do you want to switch from / to |
| 13:15.33 | Apocalipse | 729 to 711 |
| 13:16.29 | Naikrovek | i believe 711 is the default for asterisk |
| 13:16.58 | Naikrovek | somewhere in your sip.cfg (or iax.cfg if you're using iax phones) there will be something like "allow=g729" -- get rid of that |
| 13:17.05 | Naikrovek | make sure g711 is not in the disallow list |
| 13:17.29 | Apocalipse | in sip.conf i got that... allow=g729 |
| 13:17.36 | Naikrovek | okay |
| 13:17.44 | Naikrovek | does it say "disallow=g711" |
| 13:17.57 | Naikrovek | or actually |
| 13:18.02 | Naikrovek | just change that g729 to g711 |
| 13:18.04 | Naikrovek | oh wait |
| 13:18.06 | Naikrovek | to ulaw |
| 13:18.09 | Naikrovek | make it say |
| 13:18.11 | Naikrovek | allow=ulaw |
| 13:18.15 | Naikrovek | if you're in the US or japan |
| 13:18.21 | Naikrovek | europe should be "allow=alaw" |
| 13:18.23 | Apocalipse | disallow=all |
| 13:18.23 | Apocalipse | allow=g729 |
| 13:18.23 | Apocalipse | allow=alaw |
| 13:18.23 | Apocalipse | qualify=yes |
| 13:18.23 | Apocalipse | directrtpsetup=yes |
| 13:18.23 | Apocalipse | t38pt_udptl=yes |
| 13:18.24 | Apocalipse | t38pt_usertpsource=yes |
| 13:18.24 | Apocalipse | dtmfmode=rfc2833 |
| 13:18.27 | Naikrovek | ack |
| 13:18.29 | Naikrovek | ~pb |
| 13:18.30 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 13:18.36 | Naikrovek | okay |
| 13:18.42 | Naikrovek | then just remove the g729 line |
| 13:18.46 | Naikrovek | and don't paste in here |
| 13:18.47 | Naikrovek | please |
| 13:18.48 | Apocalipse | only that? |
| 13:18.52 | Apocalipse | sorry |
| 13:18.55 | Naikrovek | doesn't annoy me much but freaks others out |
| 13:18.56 | Naikrovek | yeah |
| 13:18.57 | Naikrovek | that's it |
| 13:19.01 | Naikrovek | remove the g729 line |
| 13:19.08 | Naikrovek | and reload asterisk |
| 13:19.14 | Naikrovek | you may need to restart |
| 13:19.17 | Naikrovek | don't remember |
| 13:19.32 | Apocalipse | ok... |
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| 13:19.38 | Apocalipse | thx |
| 13:19.43 | Naikrovek | sounds like you don't believe me |
| 13:19.43 | Apocalipse | 1 min... let me try... |
| 13:19.52 | Naikrovek | what kind of phones do you have |
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| 13:20.57 | Naikrovek | why are you switching away from g729 |
| 13:21.08 | Naikrovek | license thing or just voice quality |
| 13:21.43 | Apocalipse | not working... |
| 13:22.30 | Naikrovek | did you restart asterisk? |
| 13:22.32 | Naikrovek | or do a sip reload |
| 13:22.39 | Apocalipse | Naikrovek, my problem here its a little bit complex... |
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| 13:23.09 | Naikrovek | alright what's the real problem then |
| 13:23.32 | Apocalipse | my scenario [phone] <-> [hipath 8000 siemens] <-> [trixbox] <-> PSTN |
| 13:24.02 | Naikrovek | ah |
| 13:24.06 | Apocalipse | when i made a outbound call to an internal extension its ok |
| 13:24.07 | Naikrovek | heh |
| 13:24.24 | Naikrovek | but when you want to call out out, it fails because of codec problem |
| 13:24.38 | kaldemar | Apocalipse: ask in #freepbx, if you use trixbox, it generates your configurations and will most likely override any modifications by hand. |
| 13:24.44 | Apocalipse | when i tranfer this to another internal extension i got no RTP |
| 13:24.45 | Naikrovek | yup |
| 13:25.02 | Naikrovek | Apocalipse: let's take this over to #trixbox |
| 13:25.07 | Apocalipse | yes |
| 13:25.19 | Apocalipse | 0 answer there... |
| 13:25.22 | Apocalipse | but ok |
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| 13:25.44 | Naikrovek | i gotta stop lurking in there and i gotta knock this "be helpful" instinct out |
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| 13:44.02 | _zoom_ | hi, can i ask asterisk to force certain frame size for rtp? |
| 13:47.27 | [TK]D-Fender | _zoom_: normally like "allow=ulaw:30" IIRC |
| 13:48.00 | _zoom_ | <PROTECTED> |
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| 13:48.46 | kaldemar | _zoom_: doc/rtp-packetization.txt in the source package contains some documentation. |
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| 13:58.51 | MRH2 | Hi if I am sending sip traffic to a gateway/provider and qualify = x ms, if a call has already been handed off to a media gateway at the provider and the main sip gateway/peer at the provider becomes unavail or too slow to respond would that cause a call to be dropped? |
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| 14:00.12 | MRH2 | ( I am assuming no) |
| 14:02.22 | [TK]D-Fender | MRH2: Qualify has no impact on calls in progress |
| 14:02.31 | MRH2 | cool thanks |
| 14:02.51 | [TK]D-Fender | MRH2: If you drop a ton of RTP taht could kill a call, but then you'd notice to moment that started happening |
| 14:06.08 | MRH2 | media is fine and then the call just cuts, intermittent problem (which I guess means it is likely a network issue) |
| 14:07.22 | [TK]D-Fender | MRH2: Time to hit the SIP debug and confirm which side calls it quits first |
| 14:09.11 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 14:11.30 | MRH2 | <---trying to understand the sip stuff - although i do see "Asterisk-HangupCause: Normal Clearing" |
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| 14:16.15 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 14:24.19 | *** join/#asterisk WindBack (~quassel@200-122-74-15.cab.prima.net.ar) |
| 14:25.32 | stix | I can set up call forwarding for an extension by updating the asteriskdb's CF family. Can I do something like it if I want to forward calls to queues? |
| 14:30.12 | *** join/#asterisk b11d` (~no@234-200-29-134.hcc.mnscu.edu) |
| 14:31.08 | [TK]D-Fender | stix: What isn't "forwarding". |
| 14:31.12 | [TK]D-Fender | That* |
| 14:31.13 | MRH2 | thanks for the confirmation about qualify - i gtg unfortunately |
| 14:31.26 | [TK]D-Fender | stix: That is a DB value which is meaningless without dialplan that cares about it |
| 14:31.59 | stix | [TK]D-Fender, apparently something in the freepbx dialplan cares about it then |
| 14:32.18 | stix | I'll figure something else out |
| 14:32.22 | [TK]D-Fender | stix: Which is something you shouldn't be asking about in here... |
| 14:32.24 | [TK]D-Fender | ~freepbx |
| 14:32.24 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 14:32.29 | [TK]D-Fender | stix: Second door to the left |
| 14:32.33 | stix | yes yes I know |
| 14:33.54 | *** part/#asterisk _zoom_ (~user@41.218.32.13) |
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| 14:39.00 | WindBack | I need to connect an * server to a Ericcson PBX using an h.323 trunk. The Ericcson technicina says that I need to authenticate in his pbx sending some key. SOmbebody know which option should I use for this in h323.conf.? Anybody has experience on chan_h323 in asterisk? |
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| 14:51.51 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 14:58.33 | kevcox | What is a reasonable rate for managing up to a four port Asterisk box? I have a school wanting us to maintain their system. We would provide the computer and any changes made to the system but they will buy the phones and install cables where needed. |
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| 15:06.03 | kevcox | Anyone? |
| 15:06.41 | Qwell | What do you feel your time is worth? |
| 15:06.44 | leifmadsen | kevcox: I just charge an hourly rate |
| 15:06.52 | Chainsaw | kevcox: Generally you would charge for your time, yes. |
| 15:07.07 | Chainsaw | kevcox: Unless you want to organise this like a flat-fee subscription? |
| 15:07.13 | kevcox | I was thinking of a flat $85 a month. |
| 15:07.28 | leifmadsen | Chainsaw: I believe he wants a flat-rate subscription type maintenance contract |
| 15:07.30 | Chainsaw | kevcox: Just about two hours of your time, okay. |
| 15:07.43 | leifmadsen | $85 wouldn't even cover an hour of my time :) |
| 15:08.00 | Chainsaw | leifmadsen: Yes, but celebrities always charge extortionate rates :) |
| 15:08.06 | leifmadsen | Chainsaw: haha |
| 15:08.15 | leifmadsen | I'd probably charge at least $2000 a year |
| 15:08.17 | kevcox | It is a school... |
| 15:08.26 | leifmadsen | right... |
| 15:08.53 | russellb | public school or private? |
| 15:09.13 | kevcox | Private |
| 15:09.16 | kevcox | Catholix |
| 15:09.19 | kevcox | Sorry |
| 15:09.21 | kevcox | No offence |
| 15:09.21 | Qwell | extortion rate then |
| 15:09.33 | kevcox | can't spell today |
| 15:10.45 | leifmadsen | I charge the same rate whether the client is a non-profit or a megalomaniacal corporation |
| 15:11.32 | kevcox | I typically give 10% off rate |
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| 15:14.18 | kevcox | We typically manager servers and firewalls so this is a new area for us the closer that telephony merges with IP traffic. |
| 15:14.26 | kevcox | The input was a good help. |
| 15:14.39 | Chainsaw | will now forever envisage leifmadsen going "one... million... dollars" when asked for a quote |
| 15:15.06 | leifmadsen | Chainsaw: that's where I start the negotiations |
| 15:15.09 | Kyosh | $2k/yr is not enough |
| 15:15.27 | kevcox | Come on how many changes will one make |
| 15:15.27 | WindBack | Somebody knows a good documentation about chan_h323 for asterisk please? |
| 15:15.40 | leifmadsen | kevcox: that's exactly how you get burned (with statements like that :)) |
| 15:15.42 | Kyosh | for $167/mo? nopes |
| 15:16.08 | kevcox | For four ports and about 20-25 stations |
| 15:16.12 | Kyosh | gotta start with at least $500/mo |
| 15:16.16 | leifmadsen | Kyosh: I forgot to state that over the year I'd limit the number of "incidents" and not unlimited add/move/changes |
| 15:16.28 | kevcox | I figure you have to stay close to the market |
| 15:16.36 | Kyosh | $2k a year, 5 incidents |
| 15:16.43 | kevcox | I may feel one way but it doesn't mean I'll get it |
| 15:16.52 | leifmadsen | then I'd find a new client |
| 15:17.27 | leifmadsen | I've had people come to me and say, "I'll pay you $X instead of $Y dollars" and I usually ask them to find someone else. |
| 15:17.39 | mmattice | where $X < $Y |
| 15:18.59 | leifmadsen | yes :) |
| 15:19.12 | leifmadsen | $X never ends up being > $Y in the offer for some reason |
| 15:19.19 | leifmadsen | "You don't charge enough! How about $500 an hour?!" |
| 15:19.31 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 15:19.46 | kevcox | I guess it might also help to limit types of changes by the box I run |
| 15:20.08 | leifmadsen | don't get stuck in a contract where you're managing the system and have to add new features and such for a flat amount |
| 15:20.45 | kevcox | If they need the basics I could use AskoziaPBX |
| 15:20.56 | kevcox | Very basic on top of Asterisk |
| 15:23.00 | Naikrovek | is rewatching You Suck at Photoshop and is reliving hilarity |
| 15:25.46 | Nugget | heh, I forgot about those |
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| 15:29.07 | Naikrovek | ooh postgresql 9 is out. nice. |
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| 15:38.25 | leifmadsen | russellb: http://leifmadsen.wordpress.com/2010/09/20/asterisk-imap-and-gmail/ |
| 15:38.31 | russellb | hot |
| 15:38.59 | leifmadsen | Naikrovek: damnit! I hope I don't have to redo the Relational Database Integration chapter now! :) |
| 15:39.19 | Naikrovek | i doubt it |
| 15:39.21 | leifmadsen | :D |
| 15:39.22 | Naikrovek | also |
| 15:39.25 | Naikrovek | i can't tell if you're joking |
| 15:39.32 | leifmadsen | heh, either can I |
| 15:39.33 | Naikrovek | i wonder if i'm autistic or something |
| 15:39.38 | Naikrovek | i can never tell |
| 15:39.45 | Naikrovek | and being around people makes me nervous |
| 15:39.49 | Naikrovek | hmm |
| 15:39.53 | Naikrovek | fine on irc tho |
| 15:39.59 | leifmadsen | you'd likely have aspergers syndrome rather than being autistic |
| 15:40.28 | leifmadsen | if you were truly autistic, you'd have been diagnosed long ago |
| 15:40.53 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 15:41.17 | *** join/#asterisk Lantizia (~lantizia@erebus.seaquake.net) |
| 15:41.39 | Lantizia | Lo are there special version of dahdi/libpri/addons/etc... that should be used with 1.6.2.13? or just whatever the latest is? |
| 15:42.08 | [TK]D-Fender | Lantizia: Latest |
| 15:43.32 | leifmadsen | Lantizia: should always be the latest -- there wouldn't be special editions for versions |
| 15:44.03 | Lantizia | ok I'll get just 1.6.2-current of asterisk and addons and current of dahdi |
| 15:44.20 | Naikrovek | leifmadsen: i dunno. i only got diagnozed with narcolepsy and heamochromatosis last year. although autism does show itself a hell of a lot easier |
| 15:44.24 | Naikrovek | s/easier/earlier/ |
| 15:44.27 | Lantizia | if I want dahdi linux and tools... do I just get complete instead? |
| 15:44.50 | leifmadsen | Lantizia: yes |
| 15:45.19 | Naikrovek | aspergers would be nice though, i guess |
| 15:45.23 | Naikrovek | ability to concentrate |
| 15:45.23 | leifmadsen | Naikrovek: ya, aspergers is like a very high functioning level of autism |
| 15:45.37 | leifmadsen | very hard to diagnose though |
| 15:46.29 | Naikrovek | yeah |
| 15:54.56 | tzafrir_laptop | Actually from what I heard it is too easy to diagnose... |
| 15:55.40 | Naikrovek | too easy to misdiagnose probably |
| 15:55.44 | Naikrovek | hell i dunno |
| 15:56.22 | coppice | the main problem to diagnosing most of these kinds of issue is parents in denial |
| 15:57.15 | *** join/#asterisk n3hxs (~HAMming@63.68.135.4) |
| 15:57.22 | Naikrovek | hmm |
| 15:58.06 | coppice | http://www.visual6502.org/JSSim/index.html |
| 16:04.22 | fauxalliance | http://www.pbs.org/wgbh/pages/frontline/medicatedchild/ |
| 16:04.39 | *** join/#asterisk ghenry (~ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 16:05.09 | ghenry | hi, with AMI, what's the best way to get the status of my extension to show me what call I'm on etc. so I can see the callerid of who's called me? |
| 16:07.42 | leifmadsen | coppice: that's crazy |
| 16:08.03 | coppice | isn't HTML5 fun? |
| 16:08.12 | leifmadsen | and scary |
| 16:08.52 | *** join/#asterisk pabelanger (~pabelange@2607:f2c0:a000:166:218:f3ff:fe51:c71) |
| 16:14.19 | Naikrovek | html5 is kinda awesome |
| 16:14.28 | Naikrovek | as much as HTML can be "awesome" |
| 16:14.41 | Lantizia | Hey since I'm compiling asterisk from the original source it is thus "outside" of my distribution and as I understand it should be compiled in /usr/local/src not /usr/src... but since the very asterisk binary installs to /usr/sbin not /usr/local/sbin.... should I use /usr/local/src or /usr/src to compile it in? |
| 16:14.48 | coppice | now they are cranking up JS speed, there are some impressive HTML5 demos |
| 16:14.54 | Naikrovek | yeah |
| 16:16.23 | Lantizia | I know there is no hard right or wrong answer here... just opinion |
| 16:17.04 | Lantizia | Do I compile what is an outside of distro thing in /usr/local/src when it won't actually install under /usr/local |
| 16:17.37 | Qwell | Lantizia: it doesn't matter where it's compiled. |
| 16:17.46 | Lantizia | Qwell, I'm well aware of that :) |
| 16:17.58 | drmessano | So why did you ask> |
| 16:18.08 | Lantizia | because I'm after an opinion on which of the two locations |
| 16:18.09 | drmessano | The answer is: It doesn't matter |
| 16:18.21 | Lantizia | I'll take a rain check on that answer |
| 16:18.48 | drmessano | I think you should compile it in /tmp/etc/local/src/asterisk/exports/src/local/etc/temp |
| 16:18.58 | p3nguin | lantizia: Things you install should be installed with a prefix of /usr/local. It doesn't matter where it is compiled. |
| 16:19.18 | Lantizia | p3nguin, but asterisk does not install to /usr/local |
| 16:19.39 | p3nguin | Then you've ./configure-ed it wrongly. |
| 16:20.02 | Lantizia | p3nguin, it's ./configure-ed as it was extracted... shouldn't be ready for /usr/local already? |
| 16:20.07 | p3nguin | man hier |
| 16:20.32 | Lantizia | I know the FHS already - whats your point? |
| 16:20.41 | Lantizia | *FSH |
| 16:20.55 | drmessano | Asterisk installs to /usr/sbin on CentOS by default |
| 16:20.59 | p3nguin | If you're installing it in the wrong place, you've configured it incorrectly. |
| 16:21.01 | drmessano | I compile in /usr/src |
| 16:21.18 | drmessano | There, real world example |
| 16:21.21 | Lantizia | p3nguin, or it comes misconfigured is my point |
| 16:21.33 | fauxalliance | drmessano, your |
| 16:21.39 | fauxalliance | 'r real? |
| 16:21.52 | fauxalliance | stutters |
| 16:22.07 | drmessano | No, I am a facsimile |
| 16:22.14 | Qwell | The configure script is there so you can configure it how you choose. The options you've chosen (ie; the defaults) install into /usr/. I fail to see why this is such an issue. ./configure --help |
| 16:22.16 | fauxalliance | my clone is on stress leave.. |
| 16:22.29 | drmessano | Which is why it's so important to get T.38 working. I need to fax myself back to my home planet |
| 16:22.49 | Lantizia | Qwell, why is /usr/local not the default? |
| 16:22.56 | fauxalliance | drmessano, in colour? |
| 16:22.57 | Qwell | Because it's not. |
| 16:22.59 | drmessano | Qwell, because p3nguin says it's wrong, n00b |
| 16:23.01 | Lantizia | because... |
| 16:23.14 | Qwell | Why isn't the default /var/tmp/bacon/? Because it just isn't. |
| 16:23.19 | *** join/#asterisk ahowlader (~Adnan@119.30.38.44) |
| 16:23.23 | Lantizia | I proclaim bug |
| 16:23.24 | drmessano | HAHAH |
| 16:23.35 | Qwell | rolls his eyes and goes back to doing something useful |
| 16:23.48 | p3nguin | Just because it is wrong by default does not prevent the user from configuring it correctly. That's what the configure script is for. |
| 16:23.58 | Lantizia | well if /usr is for the distribution to use... and your provided tar.gz's are clearly outside of that scope |
| 16:24.05 | Lantizia | then why would /usr be the default! |
| 16:24.26 | ghenry | Sorry, what's the best way to get the status of my extension to show me what call I'm on etc. so I can see the callerid of who's called me? Can anyone point me to the right command/s? |
| 16:24.28 | Lantizia | "Because it just is" is just a crap answer lol |
| 16:24.40 | ghenry | I'm reading http://www.voip-info.org/wiki/view/Asterisk+manager+API |
| 16:25.17 | p3nguin | lantizia: That's how some people say "We did it wrong, sorry." |
| 16:25.39 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 16:25.40 | Lantizia | p3nguin, you're making me feel better :) aaaaw I'm all blushy lol |
| 16:26.36 | drmessano | Why is apachectl and httpd in /usr/sbin ? |
| 16:26.38 | Qwell | Call me when FSH mandates that we use /usr/local/ |
| 16:26.39 | cusco | hi... |
| 16:26.52 | Lantizia | drmessano, you get them via your distro? |
| 16:26.59 | drmessano | ... |
| 16:27.04 | cusco | how can I diagnose why a call is on a queue... and it is not ringing in memberes registered in the queue...? |
| 16:27.09 | cusco | http://paste.debian.net/90590/ |
| 16:27.10 | p3nguin | Install it from source and it won't be in that location. |
| 16:27.30 | p3nguin | I think the default for apache2 is /usr/local/apache, if I remember right. |
| 16:27.41 | Lantizia | hurrah for apache lol |
| 16:27.58 | tzafrir_laptop | Qwell, it doesn't. However the FHS recommends so |
| 16:28.09 | p3nguin | Of course it's been like 10 years since I cared how to install apache from sources. |
| 16:28.11 | drmessano | If you're so sure this is a bug, submit a bug report.. let the masses decide |
| 16:28.15 | *** join/#asterisk makafre (40568d85@gateway/web/freenode/ip.64.86.141.133) |
| 16:28.40 | Lantizia | drmessano, nah it's only a matter of time FreeSWITCH takes over anyway and it installs to the right place lol |
| 16:28.59 | drmessano | FreeSWITCH taking over? Laughable |
| 16:29.02 | Qwell | tzafrir_laptop: actually, it doesn't even recommend that. |
| 16:29.02 | p3nguin | Yeah, there's no mandate, but it does say THIS IS THE STANDARD, PLEASE OBEY IT. |
| 16:29.11 | makafre | Hey guys, how are you; quick question, I have 2 IPs on my server (same nic), but SIP users can't register on both, just the primary IP, any hint ? |
| 16:29.20 | Qwell | http://www.pathname.com/fhs/pub/fhs-2.3.html#USRLOCALLOCALHIERARCHY |
| 16:29.24 | Lantizia | well this is #asterisk - I wasn't expecting agreement :) |
| 16:30.11 | p3nguin | Locally installed software must be placed within /usr/local rather than /usr unless it is being installed to replace or upgrade software in /usr. [27] |
| 16:30.17 | tzafrir_laptop | Qwell, "Locally installed software must be placed within /usr/local rather than /usr unless it is being installed to replace or upgrade software in /usr. " |
| 16:30.20 | p3nguin | "must be placed within /usr/local" |
| 16:30.23 | p3nguin | MUST |
| 16:30.28 | *** join/#asterisk ethix (~ethix@rrcs-24-39-101-66.nys.biz.rr.com) |
| 16:30.32 | drmessano | Lantizia: Yeah, and you seem like a newb, so I would expect a baseless, uneducated comment about FreeSWITCH vs Asterisk.. So :) |
| 16:30.52 | Lantizia | drmessano, I can be expertly yet whimsical... doesn't make me a newb :) |
| 16:31.04 | Lantizia | you just seem like a bit of an asshat tbh |
| 16:31.19 | cusco | I really cannot understand what is wrong with queue() that it does not dial to members in the queue |
| 16:31.22 | Lantizia | everyone is entitled to label lol |
| 16:31.38 | Qwell | tzafrir_laptop: "locally installed software" refers to stuff that can't be mounted elsewhere |
| 16:31.51 | tzafrir_laptop | Huh? |
| 16:31.57 | drmessano | Lantizia, an educated asshat who understand the space the two products compete in? Yes. |
| 16:32.04 | Qwell | tzafrir_laptop: ie; over NFS |
| 16:32.09 | tzafrir_laptop | Actually traditionally /usr/local was a potentiall shared mount point |
| 16:32.11 | p3nguin | Great, now the FHS is open to interpretation. |
| 16:32.13 | Qwell | exactly |
| 16:32.18 | Qwell | well, no. opposite |
| 16:32.25 | Lantizia | drmessano, and you're thinking behind the idea I don't? you've no idea what I do and do not know about! |
| 16:32.32 | Qwell | /usr/ was potentially shared. /usr/local/ is strictly not shared. |
| 16:32.52 | Lantizia | drmessano, still this is #asterisk... if you want a FreeSWITCH why don't you PM me so I can ignore you |
| 16:33.04 | fauxalliance | Lantizia, asterisk consultant by any chance? |
| 16:33.11 | tzafrir_laptop | Qwell, wrong |
| 16:33.34 | drmessano | Lantizia, I don't PM without asking.. It's IRC etiquette.. but I wouldn't expect a newb to know that :) |
| 16:33.37 | Qwell | http://www.pathname.com/fhs/pub/fhs-2.3.html#THEUSRHIERARCHY |
| 16:33.46 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 16:33.48 | Lantizia | drmessano, I do but it's not without asking is it... it's with invite :) |
| 16:33.48 | Qwell | "That means that /usr should be shareable between various FHS-compliant hosts and must not be written to. " |
| 16:35.13 | *** join/#asterisk BANSAL (~bansal@117.199.114.58) |
| 16:35.28 | tzafrir_laptop | Qwell, the point is that when Asterisk is installed from source, it is a "Locally installed software" |
| 16:36.08 | *** join/#asterisk drudge` (tacos@unaffiliated/drudge/x-837452) |
| 16:39.30 | p3nguin | Of all the times I ever compiled asterisk, I never knew it was installing into the wrong place... I ALWAYS use ./configure --prefix=/usr --sysconfdir=/etc because I ALWAYS roll it into a package. |
| 16:41.36 | drmessano | Now you know... and knowing is half the battle |
| 16:41.37 | Lantizia | p3nguin, oh sure if it's going to be packaged then it makes sense lol |
| 16:43.04 | p3nguin | But it should never be assumed that it will be packaged, and therefore should require setting the prefix for the purpose of packaging. |
| 16:43.15 | Lantizia | exactly |
| 16:44.24 | drmessano | But are you installing on behalf of the local user or on behalf of the distro? In that scenario, you're both.. |
| 16:45.00 | drmessano | So is that where /var/tmp/bacon comes into play? |
| 16:45.11 | p3nguin | When packaging, it is for the distro only. |
| 16:47.57 | drmessano | I guess it's sorta like if you roll a bunch of joints, you pretty much become a prime candidate for "intent to distribute" which labels you a "dealer", but if you're going to smoke them yourself, you're the "customer".. which brings up the problem as illustrated with schrodinger's cat.. you simultaneously exist in both states |
| 16:48.21 | Lantizia | tzafrir_laptop, please can I call him names again ? :D |
| 16:48.48 | drmessano | Lantizia: Sorry if I am talking over your head.. Maybe you can google some of this |
| 16:49.26 | fauxalliance | gets all puffy eyed at drmessano's expert analogy. |
| 16:49.47 | Lantizia | he's single too lol :P |
| 16:49.56 | Lantizia | later |
| 16:49.58 | *** part/#asterisk Lantizia (~lantizia@erebus.seaquake.net) |
| 16:49.59 | drmessano | I am? Not quite |
| 16:50.00 | fauxalliance | and the moral of the story drmessano ? Allways keep one rolled in the holster. |
| 16:50.07 | tzanger | http://i.imgur.com/AxxfN.png |
| 16:50.07 | leifmadsen | ya! drmessano is mine! |
| 16:50.44 | *** part/#asterisk kevcox (~User@208.66.88.194) |
| 16:51.03 | leifmadsen | tzanger: nice |
| 16:51.27 | tzanger | leifmadsen: that's in case someone tries to pull a fast one on you |
| 16:51.34 | tzanger | I know you're a little gullible and all, but I like you |
| 16:51.42 | tzanger | wouldn't want you to get shanghai'd |
| 16:51.42 | leifmadsen | tzanger: yay! I like you too |
| 16:51.51 | drmessano | I guess I could have told him about my gorgeous fiancee but talking to him about girls was probably as bad as talking to him about any subject that requires more than a 4th grade education. *sigh* |
| 16:52.04 | leifmadsen | tzanger: sometimes I lie though |
| 16:52.17 | p3nguin | Her wife Jennifer... what kind of freaky people are they?! |
| 16:53.04 | tzanger | leifmadsen: everyone lies sometimes |
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| 16:53.41 | *** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 16:55.31 | WWGD | references to rolling joints and schrodinger's cat in the same sentence... i love this channel |
| 16:55.58 | drmessano | bows and burps |
| 16:57.48 | tzanger | oh man, this year's the 200th anniversary of oktoberfest! |
| 16:58.00 | fauxalliance | what fun... |
| 16:58.01 | drmessano | May we burn a witch? |
| 16:58.23 | fauxalliance | how about an effigy of Guy Fawkes... that comes next. |
| 16:58.38 | drmessano | Maybe a "Strange Brew" marathon? |
| 17:01.05 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 17:03.36 | leifmadsen | tzanger: beer! |
| 17:05.21 | *** part/#asterisk WWGD (~WWGD@208.79.14.130) |
| 17:08.20 | tzanger | where? |
| 17:08.53 | fauxalliance | Munich |
| 17:09.14 | leifmadsen | tzanger: over there! |
| 17:09.37 | bougyman | any vici users/admins about atm? |
| 17:20.32 | *** join/#asterisk sol (~sol@unaffiliated/sol) |
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| 17:35.08 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 17:35.11 | kristianpaul | Hello |
| 17:35.48 | kristianpaul | Wha's the digit i should use to add a delay after a call is stablished? |
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| 17:46.22 | exothermi | How do you control the time between when a channel hangs up and when it becomes available to the dial group? |
| 17:50.12 | Naikrovek | hangup = available for next call, I think |
| 17:50.15 | Naikrovek | not sure if it's configurable |
| 17:50.19 | Naikrovek | what are you trying to do |
| 17:50.37 | Naikrovek | oh you want a wait time after a queue member hangs up from a call. to let them log what happened or whatever |
| 17:51.06 | p3nguin | wrap-up time? |
| 17:51.23 | Naikrovek | i'm thinking that's what he wants but we'll never know until he replies :) |
| 17:53.11 | Naikrovek | anyone know any good configuration management software? free as in beer would be best, if possible |
| 17:53.23 | Naikrovek | otherwise i'll write one |
| 17:53.41 | Qwell | vim |
| 17:53.45 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 17:53.48 | exothermi | Naikrovek: actually these are analog channels, and it take the CO about 6 seconds to reset their gear. |
| 17:54.12 | Naikrovek | Qwell: my supervisor suggested Excel but neither VIM nor Excel beat paper and I want to implement some process |
| 17:54.14 | exothermi | Naikrovek: So after I hang up a channel I want to ensure that it isn't attempted again for 6 seconds. |
| 17:54.24 | Naikrovek | gotcha |
| 17:56.26 | Naikrovek | no idea |
| 17:56.31 | Naikrovek | probably a way though |
| 17:59.03 | *** join/#asterisk zxvff (shahid@dev.hockingits.com) |
| 18:13.06 | exothermi | [TK]D-Fender: You have any idea how to delay that? |
| 18:13.46 | carrar | Keep track of channels in a db |
| 18:14.07 | carrar | for a outgoing call query a free channel based on last used time |
| 18:14.42 | leifmadsen | when you hang up the call, in the 'h' extension, set the value of EPOCH to the AstDB, then when placing a call, check the EPOCH and find the diff and delay for the remainder of the time |
| 18:15.30 | exothermi | ok thanks |
| 18:15.41 | leifmadsen | Set(DB(last_used/channel/1)=${EPOCH}) |
| 18:16.22 | leifmadsen | Set(DELAY=$[${EPOCH} - ${DB(last_used/channel/1)}]) |
| 18:16.24 | leifmadsen | something like that |
| 18:16.39 | leifmadsen | my math is wrong I'm sure |
| 18:17.06 | leifmadsen | GotoIf($[${DELAY} < 6]?wait,1) |
| 18:17.32 | [TK]D-Fender | leifmadsen: Nifty idea. unfortunately he said dial group. If that means he doing "g0" etc, this itea won't work since you can't lok a channel out of a group |
| 18:17.33 | leifmadsen | exten => wait,1,Wait($[6 - ${DELAY}]) |
| 18:17.55 | leifmadsen | [TK]D-Fender: I didn't read back that far, but you should still be able to find what channel the call actually went out on |
| 18:18.05 | leifmadsen | yes, he'll need to be clever and figure something out |
| 18:18.43 | [TK]D-Fender | leifmadsen: You could put maybe a global dial-colkout in effect... |
| 18:18.47 | [TK]D-Fender | lockout* |
| 18:18.50 | exothermi | ya I need to do this across 70 channels |
| 18:19.25 | p3nguin | Maybe you could get the phone company to make some adjustments. |
| 18:19.30 | carrar | haha |
| 18:19.35 | exothermi | p3nguin: lmas |
| 18:19.37 | carrar | like fix their crap |
| 18:19.56 | exothermi | carrar: ya like that is going to happen. |
| 18:20.03 | p3nguin | If you tell them "Fix your crap," they might be less likely to comply. |
| 18:20.18 | carrar | Got a ticket open with them on it? |
| 18:20.59 | exothermi | p3nguin: "Hey Qwest, would you mind replacing your switches in you CO for me? |
| 18:21.04 | *** part/#asterisk kristianpaul (~kristianp@unaffiliated/kristianpaul) |
| 18:21.44 | carrar | I've gotten Qwest to do a lot |
| 18:22.05 | carrar | but you have to keep on them |
| 18:22.15 | [TK]D-Fender | That's what SHE said |
| 18:22.22 | exothermi | carrar: You've had them go as far as to replace a full switch? |
| 18:22.33 | carrar | if it's the problem |
| 18:22.52 | carrar | replace cards, move DSLAMS, fix their ATM |
| 18:23.06 | carrar | come out and re-wite the nbeighborhood |
| 18:23.09 | p3nguin | You know they'll question your qualifications when you ask for such extreme measures. |
| 18:23.10 | carrar | re-wire |
| 18:23.18 | carrar | They are responsibile |
| 18:23.31 | p3nguin | Doesn't mean they are accountable, though. |
| 18:23.39 | exothermi | From the people I have talked to they just say that it is the norm for a carrier's gear to leave the channel open for a few seconds (since there is no disconnect supervision) |
| 18:23.41 | carrar | Up to you to make them accountable |
| 18:23.53 | p3nguin | At which time they question your qualifications. |
| 18:23.55 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 18:24.08 | exothermi | These are CAS T1s using loopstart |
| 18:24.37 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 18:25.02 | carrar | With so many lines, why are you not using straight T1's to begin with |
| 18:25.51 | carrar | or am I miss reading your stuff |
| 18:27.22 | exothermi | carrar: They are straight T1s |
| 18:27.47 | carrar | can you switch to Koolstart |
| 18:27.59 | carrar | or something else |
| 18:28.06 | exothermi | That may help. Let me check. |
| 18:28.57 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 18:29.48 | carrar | alternativly you could switch to pri |
| 18:29.54 | carrar | but probably cost more |
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| 18:46.54 | nny | hrmm |
| 18:47.08 | nny | anyone ever dealt with a dialer that uses tones to alert the system of a new call>? |
| 18:47.36 | nny | have a client with a "strata dialer" working on a recording solution.. wondering if I can use the tones + features.conf and some sorcery to start a new recording |
| 18:47.41 | nny | (sounds like heavy sorcery) |
| 18:48.31 | leifmadsen | nny: yes |
| 18:48.36 | leifmadsen | as long as the tones are DTMF |
| 18:50.20 | nny | leifmadsen: yeah I need to do a test call and see what asterisk thinks of the tones with debug. I assume at the point I can do some channel magic, start a new MixMonitor and create additional magic as needed |
| 18:50.56 | leifmadsen | well I've used features.conf to cancel a call recording, so I see no reason it couldn't start one too :) |
| 18:51.13 | nny | leifmadsen: nice thanks |
| 18:52.07 | nny | another stupid question I guess since I am here and elgible. I have a bunch of MCF files that were originally g729 that someone tried to record with orecx's GPL version. I was told it's possible to use transcode http://www.transcoding.org/ but don't see support for that specific item. Am i missing something? |
| 18:56.32 | *** part/#asterisk shayy (~shayy@109.160.132.106) |
| 19:07.28 | *** join/#asterisk Letoric (~Letoric@tpi-dfw-uc-f1-69-94-238-22.totalprocess.net) |
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| 19:21.16 | Letoric | Afternoon folks |
| 19:21.40 | Letoric | Anybody know of a way to tell mixmonitor to spool files in 1 location, but when finished, move them to a final location? |
| 19:22.44 | Letoric | Our CFO wants access to the monitor files, to give you an idea of what I'm trying to achieve. The thought was using a script to copy the completed files to a location he can access |
| 19:23.17 | Letoric | mixmonitor doesn't seem to spool in a tmp file, it seems to constantly write to an appropriately named file for whatever call it's working on |
| 19:23.17 | tzanger | Letoric: do the move in a hangup handler |
| 19:23.27 | [TK]D-Fender | Letoric: Use Monitor() and mix/move them afterwards yourself |
| 19:24.52 | Letoric | [TK]D-Fender: I'm trying to put as much automation into this system as possible so that when they replace me (it's inevitable, us in IT are never permanent) they won't have to struggle too much |
| 19:26.02 | Letoric | trying not to use samba, as we already have unix servers exporting shares with nfs, so that's the preferred dump location |
| 19:26.22 | Letoric | didn't want to have it write the files to a network location while recording, in case of any failures, thus the need for a script |
| 19:28.00 | *** part/#asterisk frigidzephyr (~rnewton@nat/digium/x-ykjqxoszcaybkigf) |
| 19:30.05 | *** join/#asterisk Lantizia (~lantizia@erebus.seaquake.net) |
| 19:34.25 | *** join/#asterisk WWGD (~WWGD@208.79.14.130) |
| 19:41.17 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-dfgoftvtmeavzbdo) |
| 19:41.19 | [TK]D-Fender | Letoric: Go for it. You have our permission. |
| 19:42.59 | KavanS | Letoric, hire....a....consultant |
| 19:43.20 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 19:46.15 | Letoric | KavanS: Thanks for your opinion, I was looking for advise on how to handle the issue myself. |
| 19:46.40 | Letoric | [TK]D-Fender: uhm...thanks? You're usually pretty helpful, bad day? |
| 19:48.02 | KavanS | Letoric, the amount of education necessary to bring you in tune with this - and to accomplish it effectively, you will want a consultant |
| 19:48.15 | KavanS | or that CFO of yours might just say "hey this outsourcing thing, way better than that half assed job Letoric did for us" |
| 19:48.30 | [TK]D-Fender | [15:23]<[TK]D-Fender>Letoric: Use Monitor() and mix/move them afterwards yourself <-- I did tell you to use Monitor for this... |
| 19:48.34 | citywok | Letoric: monitor writes to a file in the monitor folder. you cna do whatever you want with it via a script after the call has ended by using the hangup extension. |
| 19:48.38 | [TK]D-Fender | Letoric: "core show applciation Monitor" |
| 19:48.55 | [TK]D-Fender | And NO, you will NOT use the "hangup" extension for this. |
| 19:49.12 | [TK]D-Fender | Letoric: There is a nifty parm you should have read up on in there already for this |
| 19:49.24 | [TK]D-Fender | Letoric: It's all hidden in the big print :) |
| 19:49.45 | Letoric | [TK]D-Fender: I did read up on that, it's just that there was also some commentary about monitor being somewhat buggy when using mixing |
| 19:49.48 | citywok | whoops, that's right i forgot you can pass the post-execution arguments in to the monitor command directly. been a while since i wrote all that. |
| 19:50.20 | [TK]D-Fender | Letoric: Monitor isn't going to do the mixing. You are in your script as well as moving the file. |
| 19:50.44 | Letoric | [TK]D-Fender: Something about using the same threads? Sound familiar? Again..I am rather new, I was just being precautious because of that |
| 19:50.49 | citywok | yea, mixmonitor is what i'm using at the moment but it sometimes muxes the halves incorrectly and the calls overlap. |
| 19:51.03 | [TK]D-Fender | citywok: And is a command I am not advising |
| 19:51.11 | Letoric | also mixmonitor allowed me to use something that allows passage between channels while maintaining the recording |
| 19:51.29 | Letoric | forgot the exact command, but I can find it if it doesn't ring a bell already for you |
| 19:51.31 | citywok | [TK]D-Fender: yea, are there any other reasons to avoid it? |
| 19:51.55 | [TK]D-Fender | citywok: I don't know all the fine points on this, jstu those big overview ones that really differentiate them |
| 19:52.12 | [TK]D-Fender | citywok: Monitor() opens more doors |
| 19:53.40 | Letoric | [TK]D-Fender: Can you still use AudioHook_INHERIT with monitor? |
| 19:53.57 | [TK]D-Fender | Letoric: That would be one of those points I don't know so well |
| 19:54.11 | Letoric | ;( That's a critical function for us |
| 19:54.20 | Letoric | All calls have to be fully recorded :/ |
| 19:55.32 | *** join/#asterisk Dovid (Dovid@213.8.121.90) |
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| 20:41.27 | *** join/#asterisk rdircio (~rdircio@201.137.141.217) |
| 20:41.34 | rdircio | hi guys |
| 20:41.34 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 20:42.09 | rdircio | do you know of a working callprogress tone detection configuration for Mexico? |
| 20:45.51 | *** join/#asterisk simond (~simon@syria.uc.org) |
| 20:46.24 | simond | Does anyone happen to know why I can't close a terminal after running some asterisk console commands? |
| 20:46.28 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 20:47.40 | frigidzephyr | simond: huh? |
| 20:47.56 | frigidzephyr | simond: like the actual terminal window? |
| 20:48.00 | p3nguin | press Ctrl+Z and see what it says. |
| 20:48.01 | simond | like I do something like 'asterisk -rx "sip show peers"', and then I try to exit out of the terminal window |
| 20:48.29 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 20:48.48 | frigidzephyr | are you clicking a GUI close button on the window? or are you just typing exit to get out the CLI, then exit again to get out of the terminal ? |
| 20:48.48 | simond | I come back to the shell just fine, but something is stoping the terminal from ending when I log out |
| 20:48.54 | p3nguin | Usually when a terminal doesn't want to "exit," it is because there is a child process still running. |
| 20:52.13 | simond | It's a problem I've run into consistently across centos and debian, from asterisk 0.7 to 1.4 |
| 20:53.07 | simond | I mean intermittently |
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| 20:54.51 | simond | although this time it's during an upgrade process in a virtual machine snapshot, so maybe I'll finally figure out what it is. |
| 20:59.18 | *** join/#asterisk rlankfo (~areohbee@hahainyourface.com) |
| 20:59.42 | rlankfo | hello, i have a sip phone registered but to register i had to open ports 5060 tcp/udp on the firewall, now when i make a call i can't hear the other person and they can't hear me |
| 20:59.54 | rlankfo | are there more ports that i would need to open? inbound or out? i only opened 5060 inbound |
| 21:00.55 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:01.01 | b14ck | http://3.bp.blogspot.com/_RZa9ISY8vY8/TGR9SR63gZI/AAAAAAAAEmQ/pBjDmgcan7A/s1600/more_bear_i_fucking_love_cocaine.jpg |
| 21:01.45 | Dougy | rlankfo: tcpdump it |
| 21:01.54 | Dougy | http://www.techbytes.ca/techbyte136.html |
| 21:01.57 | Dougy | or do something like that |
| 21:02.53 | p3nguin | rlankfo: You have to forward/open the RTP ports, too. |
| 21:03.31 | p3nguin | SIP is only the signaling, RTP carries the audio. You can find your RTP port range in rtp.conf (probably UDP 10000-20000). |
| 21:03.48 | rlankfo | inbound and out? |
| 21:03.56 | rlankfo | i almost opened 10000-20000 UDP |
| 21:04.19 | p3nguin | The ports have to be allowed in both directions. |
| 21:04.50 | rlankfo | thanks |
| 21:10.55 | trelane | I'm taking some errors on a T1, other than accusing the provider of outright lying is there anything useful in pri debug? |
| 21:11.48 | *** join/#asterisk joren (~j@66.206.86.190) |
| 21:13.03 | joren | hey, I'm hoping somone could give me some advice.. I'm trying to setup a voicemenu in asterisk-gui, but when I add a "playback" event, it says "& donnot listen for keypress events" |
| 21:13.20 | joren | is there any way to get it to listen for keypress events from the start? |
| 21:15.04 | p3nguin | Use BackGround() instead of Playback(). |
| 21:16.00 | beardy | nods |
| 21:16.11 | joren | p3nguin, ahha, thank you very much! |
| 22:00.30 | Kobaz | man, grandstream phones suck so bad |
| 22:01.27 | *** join/#asterisk jinxed (~chatzilla@CPE0016b6eddb9a-CM001404dc5032.cpe.net.cable.rogers.com) |
| 22:01.45 | jinxed | is there a channel variable for the extension the caller entered the system from? |
| 22:02.49 | Dougy | i want to set my pbx to ring the extensions for 30sec (done), then call my cell phone, ring for 15sec, ifn o answer, go to phone again, then voicemail |
| 22:02.52 | Dougy | is that possible |
| 22:04.14 | p3nguin | jinxed: ${EXTEN} is the extension. |
| 22:06.22 | jinxed | p3nguin: isn't that for the current context? what about after a goto? |
| 22:06.40 | [TK]D-Fender | <PROTECTED> |
| 22:07.02 | exothermi | [TK]D-Fender: would it be possible to create a fifo using astdb? |
| 22:07.24 | [TK]D-Fender | exothermi: It holds data like anything else |
| 22:08.33 | exothermi | [TK]D-Fender: you see i'm thinking about replacing the dial groups with a fifo of channels |
| 22:09.08 | [TK]D-Fender | exothermi: "the dial groups"? Pardon? |
| 22:09.19 | [TK]D-Fender | exothermAls I have no idea what feeds your FIFO list. |
| 22:10.26 | exothermi | [TK]D-Fender: instead of round robin (r0 etc) where it moves through a set list incrementally and picks any available channel, it would move though a queue choosing the channel that has been unused the longest. |
| 22:14.28 | [TK]D-Fender | exothermi: There is a queue strategy for that |
| 22:14.56 | exothermi | [TK]D-Fender: for dahdi channels? |
| 22:15.57 | [TK]D-Fender | exothermi: No. Now you've gone from ring-groups to DAHDI grouped channels... |
| 22:16.04 | [TK]D-Fender | exothermDoes the channel really matter? |
| 22:17.34 | exothermi | [TK]D-Fender: my plan is to solve my "channels aren't available until x seconds after hanging up" problem. |
| 22:19.26 | exothermi | [TK]D-Fender: So if I create a fifo, and put the channel into it x seconds after it hangs up. Or the more approximate version of having total call limit set to y less than total channels and using the fifo it is probable that the next channel assigned will have been unused for at least x seconds. |
| 22:19.54 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 22:20.15 | exothermi | I guess what would happen if I set the dahdi channels as queue members? |
| 22:26.05 | *** join/#asterisk GlobeTrotterz (chatzilla@pool-96-232-156-186.nycmny.fios.verizon.net) |
| 22:40.13 | *** join/#asterisk p3nguin_ (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 22:48.36 | drmessano | Got an email from Flowroute stating they had added a "maximum outbound rate" fraud control under the user account settings. Very nice! |
| 22:50.03 | drmessano | I guess it doesn't help if you have a callcenter and turn it to 11, but then again, if you run a callcenter and get exploited for tens of thousands because of an improperly configured Asterisk box, I want the address of your HR person so I can apply for your job. |
| 22:54.36 | Dougy | <PROTECTED> |
| 22:54.39 | Dougy | is that correct |
| 22:57.32 | *** join/#asterisk [cannibalera] (~cannibale@201-14-150-105.fnsce703.dsl.brasiltelecom.net.br) |
| 22:57.56 | drmessano | Well, assuming those Xs are your number and that ,5, is the line of 1000 where it forwards to your cell, and vitel-outbound is the proper peer, and that you're using a gregorian calendar where 60 seconds is approximately one minute, and that your name is Dougy, I would probably say "yes" |
| 22:58.31 | drmessano | But that's taking a lot into account from one line.. and while I may love CSI, I am no Gil Grissom |
| 22:58.59 | Dougy | ok let me pastebin a bit |
| 22:59.20 | Dougy | <PROTECTED> |
| 22:59.45 | Dougy | everything works fine, it comes in, rings my 7960, after 5 secs it calls my cell (or at least asterisk shows the dial the same way it is if i dial it by hand), but my cell never rings |
| 22:59.51 | Dougy | then after 60 secs, my phone rings again and then goes to VM |
| 23:08.00 | p3nguin_ | Get rid of the Ringing(). |
| 23:08.08 | Dougy | will it do it automatically? |
| 23:08.19 | p3nguin_ | Get rid of the numbered priorities, in favor of using n instead. |
| 23:08.23 | Dougy | what is n |
| 23:08.34 | p3nguin_ | Get rid of the spaces in the app data. |
| 23:08.39 | Dougy | meaning |
| 23:08.43 | Dougy | Dial(SIP/1000,5) |
| 23:08.45 | Dougy | ionstead of , 5) |
| 23:08.46 | Dougy | ? |
| 23:08.50 | p3nguin_ | right |
| 23:08.55 | Dougy | ok |
| 23:09.00 | Dougy | what is the benefit of n over number |
| 23:10.09 | p3nguin_ | http://paste2.org/p/996053 |
| 23:10.39 | p3nguin_ | But here's the catch... your SIP/1000 phone is going to ring for ONLY 5 seconds. Is that really what you wanted? |
| 23:10.54 | Dougy | for the moment yes, i want it to quickly ring here once so i know it works, then redirect to my phone, and come back |
| 23:11.03 | Dougy | it will eventaully be more like 60 sec instead of 5 |
| 23:11.08 | Dougy | its just for testing sake |
| 23:11.28 | Dougy | p3nguin_: why n instead of number |
| 23:12.07 | p3nguin_ | Do you think people are going to let the phone ring for that long? 60 seconds is a long time to wait... 10 rings. I don't know anyone that waits that long before assuming there is no one to answer and hangs up. |
| 23:12.21 | p3nguin_ | Why n? Because this is not Asterisk 1.2 anymore. |
| 23:12.21 | Dougy | well, 60 was hypothetical |
| 23:12.27 | Dougy | and oh? |
| 23:12.29 | p3nguin_ | n means NEXT. |
| 23:12.30 | Dougy | what does 'n' do exactly |
| 23:12.30 | Dougy | oh |
| 23:12.36 | Dougy | well that makes perfect sense then |
| 23:12.36 | Dougy | heh |
| 23:12.40 | Dougy | lets try it |
| 23:13.16 | paulc | I have a dial to my desk phone & a local channel.. that delays a few secs, then goes to my cell.. specifically for my front door buzzer system thing.. works great (using Local/... + delay) |
| 23:13.19 | Dougy | seems to work profectly |
| 23:13.24 | Dougy | perfectly rather |
| 23:13.27 | Dougy | thank you :D |
| 23:13.47 | Dougy | so ,Answer and ,Ringing are useless these days? |
| 23:13.59 | leifmadsen | no |
| 23:14.06 | Dougy | oh, hi leifmadsen |
| 23:14.10 | leifmadsen | ohai :) |
| 23:14.13 | Dougy | how are you |
| 23:14.18 | p3nguin_ | Useless in general? No. Useless in your specific given circumstance, yes. |
| 23:14.25 | leifmadsen | ^^^ |
| 23:14.36 | Dougy | where are they practical |
| 23:14.40 | Dougy | just for argument's sake |
| 23:14.43 | Dougy | is trying to learn a lot here |
| 23:15.01 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 23:16.02 | p3nguin_ | Let's say you wanted to Answer the channel, bringing it Up. You would use Answer(). |
| 23:16.27 | Dougy | seems simple |
| 23:16.29 | p3nguin_ | Most people don't want the channel Up when it is still ringing. |
| 23:16.34 | Dougy | i owe you guys a lot for the help |
| 23:16.36 | leifmadsen | it affects CDRs |
| 23:16.52 | leifmadsen | Dougy: beers can be sent to 50 Losino St., Caledon, ON, Canada, L7C 3N5 |
| 23:16.56 | Dougy | i just need to figure out now how to record a .mp3 and upload it so it will use that for a VM greeting instead of using a SIP phone and recording it |
| 23:17.00 | Dougy | leifmadsen: sadly i cannot buy you beer |
| 23:17.01 | Dougy | i am 17 |
| 23:17.09 | leifmadsen | Dougy: money is also acceptable :) |
| 23:17.19 | Dougy | tehe |
| 23:17.23 | Dougy | paypal got disabled for not being 18 |
| 23:17.26 | leifmadsen | Dougy: that, or you could grab a copy of the book from http://www.asteriskdocs.org |
| 23:17.28 | Dougy | but since i turn 18 tomorrow...... |
| 23:17.34 | leifmadsen | I don't need your money :) |
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| 23:17.49 | Dougy | hehe |
| 23:17.49 | exothermi | is there anyway to implement a queue (from a data-structure perspective) or linked list in asterisk dialplan? |
| 23:17.50 | leifmadsen | keep it for college :D |
| 23:17.53 | leifmadsen | it's very expensive |
| 23:17.53 | Dougy | leifmadsen: what is your role in the project |
| 23:17.59 | Dougy | and im actually off to college in about 10 minutes |
| 23:18.10 | leifmadsen | Dougy: I write documentation and am the primary bug marshal and release manager |
| 23:18.14 | Dougy | ooh |
| 23:18.17 | Dougy | :) |
| 23:18.37 | p3nguin_ | Don't let him fool you, he's only here for decoration. |
| 23:18.37 | leifmadsen | exothermc: I don't quite understand the question, but I could probably build at least a simple queue from dialplan |
| 23:18.41 | leifmadsen | it's true |
| 23:18.44 | leifmadsen | I just talk big |
| 23:19.32 | Dougy | lol |
| 23:19.37 | Dougy | thx guys, bbl |
| 23:19.43 | exothermi | leifmadsen: basically I need a simple queue (basically a dahdi trunk group that would do leastrecent) |
| 23:20.02 | p3nguin_ | Oddly, Queue() provides simple queues. |
| 23:20.26 | Dougy | my queue was simple |
| 23:20.31 | Dougy | call main #, ring all phones, goto voicemail |
| 23:20.32 | Dougy | wo0t |
| 23:20.37 | Dougy | downloads asteirsk book |
| 23:20.38 | Dougy | asterisk |
| 23:21.03 | p3nguin_ | You could also throw some sound files in front of the Dial() commands and add MoH rather than ringing sounds. |
| 23:21.14 | exothermi | leifmadsen: so instead of r,R,G,g (which keep track of where it left off (versions of roundrobin)) I need leastrecent which would be more like a queue/fifo |
| 23:21.32 | Dougy | p3nguin_: my next task is conferencing and maybe some MOH |
| 23:22.08 | p3nguin_ | For home, I use Playback(silence/1&vm-dialout). |
| 23:22.32 | leifmadsen | exothermc: so why not enable leastrecent method for Queue()? |
| 23:22.59 | Kobaz | mostlymuchrecent |
| 23:22.59 | leifmadsen | the dialing groups do no have the method you are looking for. You'd have to create some clever dialplan to do it. |
| 23:23.12 | leifmadsen | mostrecentlyouttolunch |
| 23:23.27 | exothermi | leifmadsen: this is simply for dialing pstn and selecting a channel, full queues are overkill. Plus I can't answer the A-Leg until the B-leg answers. |
| 23:23.43 | leifmadsen | more information was required then to answer your question properly |
| 23:24.03 | leifmadsen | just keep track of the information in the AstDB and program the dialplan to do what you want |
| 23:24.07 | Kobaz | what about answering improperly |
| 23:24.15 | leifmadsen | Kobaz: that is what I did apparently |
| 23:24.19 | *** join/#asterisk joren (~j@97-120-211-143.ptld.qwest.net) |
| 23:24.20 | Kobaz | heh |
| 23:24.22 | p3nguin_ | dougy: MusicOnHold is pretty simple. You can use the included music files, download some free-to-use mp3s, or even stream from the internet. |
| 23:25.00 | exothermi | leifmadsen: I'm just looking for a way to do this in the dialplan, but the data-structures I'm thinking to do it with (an array) don't seem to be there. |
| 23:25.27 | leifmadsen | i could program it, but I don't have time to do it right now |
| 23:25.29 | leifmadsen | I'm heading out |
| 23:25.31 | leifmadsen | lates! |
| 23:25.40 | exothermi | leifmadsen: ok thanks |
| 23:25.46 | leifmadsen | you don't need an arrow |
| 23:25.48 | leifmadsen | array* |
| 23:25.53 | exothermi | what would you use? |
| 23:25.54 | leifmadsen | just use a loop and some astDB information |
| 23:26.08 | exothermi | leifmadsen: ahh I guess that would work. |
| 23:26.17 | leifmadsen | of course it would |
| 23:26.36 | exothermi | leifmadsen: not exactly efficient, but it would work :) |
| 23:26.47 | exothermi | heads off to code it up. |
| 23:27.03 | leifmadsen | if you need something complicated then use an AGI |
| 23:27.16 | exothermi | leifmadsen: oh ya, that is even better. Thanks |
| 23:27.21 | leifmadsen | use a real programming language if you need programming language type things |
| 23:27.38 | leifmadsen | this is not rocket science :) |
| 23:27.42 | leifmadsen | and now I'm really out |
| 23:36.36 | paulc | Screenpops: JabberSend can send a URL to the end user, they click the link, tada - web page with all the relevant details (that page built in house). Seems easy enough. Versus some kind of "soft phone with built in URL passing for screen popping" if such a thing even exists? Thoughts? |
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| 23:40.29 | *** join/#asterisk Woody2143 (~Woody2143@machine76.Level3.com) |
| 23:48.20 | GlobeTrotterz | Hi DFender, thanks fro helping me out with the agents log ins. I am now passing the value with gosub. Now i am able to log the agents in and out of the queue. But i set the calls to be recorded. but its not recording. In queues.conf i have .. monitor-format = gsm & monitor-type = MixMonitor |
| 23:48.37 | GlobeTrotterz | does anyone else had that issue before? |
| 23:48.43 | exothermi | is there a way to call a shell program from asterisk and get the return value as a variable? Or set a variable to the value the program returns? |
| 23:50.26 | exothermi | ahh the SHELL function |
| 23:52.17 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
