| 00:01.40 | p3nguin | katty: I don't really know, but I'm sure worn out. |
| 00:04.13 | hardwire | I'm using ResetCDR in the application portion of an outgoing call origination and it appears to be resetting the CDR (and then turning off CDR) for the destination I'm originating to. |
| 00:04.15 | hardwire | it's funkay. |
| 00:04.59 | Katty | p3nguin: :< |
| 00:05.04 | p3nguin | katty: Could be all that dancing you subjected me to, ya know? |
| 00:05.30 | Katty | highly doubtful |
| 00:07.02 | raden_work | hi Katty |
| 00:07.27 | raden_work | tired to |
| 00:07.38 | raden_work | beating groupwise with a stick |
| 00:08.40 | Katty | hugs raden_work |
| 00:09.13 | raden_work | all warm and fuzzy now |
| 00:13.21 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 00:15.56 | raden_work | sniffles my boss took my logitech G11 *cry* |
| 00:16.44 | *** join/#asterisk Mango (n=Mango@96.49.69.137) |
| 00:18.20 | Mango | My VoIP provider will play music when I put a call on hold if my IP phone is connected directly to them, but not if it's connected via Asterisk. How can I do this? |
| 00:18.23 | Mango | I realize I can make my Asterisk play music but I'd like to save the bandwidth. |
| 00:18.58 | *** join/#asterisk ming_zym (n=ming_zym@124.127.101.0) |
| 00:21.56 | raden_work | Mango, how are you saying bandwith with your provider doing it |
| 00:22.13 | raden_work | youd save bandwith having asterisk do it |
| 00:22.19 | raden_work | unless im severly missing something |
| 00:22.20 | Mango | Do explain. |
| 00:22.54 | raden_work | how many phones you have ? |
| 00:22.59 | Mango | 2 |
| 00:23.04 | raden_work | what do you connect with besides asterisk ? |
| 00:23.19 | Mango | It goes IP Phone -> My Asterisk -> Provider's Asterisk |
| 00:24.17 | p3nguin | mango: You just need to configure music on hold. |
| 00:24.38 | Mango | p3nguin: I realize that. How? :) |
| 00:24.58 | Mango | I checked the docs for musiconhold.conf, but it wasn't there, unless I'm missing something. |
| 00:24.59 | p3nguin | Did you even TRY to figure out how before you asked me? |
| 00:25.05 | Mango | Yes of course. |
| 00:25.33 | ruben23 | hi how do i correct problem compiling zaptel and having error about not having kernel source. |
| 00:26.03 | p3nguin | Make sure you have musiconhold.conf configured with a context and a directory to the music files. Make sure there are suitable music files in that directory. Load the musiconhold module. |
| 00:26.19 | ruben23 | http://pastebin.com/m24d01b3b |
| 00:26.24 | Mango | p3nguin: Did you even READ my question before you answered it? ;) |
| 00:26.43 | *** part/#asterisk jelly-bean1 (n=jelly@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 00:27.11 | p3nguin | Are you trying to get my help or what? |
| 00:28.10 | p3nguin | 'Cause my music on hold works, and I could easily go work on something else. |
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| 00:28.11 | *** join/#asterisk _bradk (n=brad@unaffiliated/-bradk/x-9249860) |
| 00:28.19 | Mango | Okay. That's exactly what I don't want to do. |
| 00:28.27 | Mango | I want my VoIP provider to handle music on hold. |
| 00:28.58 | p3nguin | Why would you want that? |
| 00:29.05 | Mango | To save bandwidth. |
| 00:29.06 | zamba | what is really the register option for in sip.conf? |
| 00:29.13 | zamba | is it related to incoming or outgoing, or both? |
| 00:29.22 | p3nguin | How is sending more packets (music) going to SAVE bandwidth? |
| 00:29.32 | Mango | zamba: register => username:password@server:5060 |
| 00:29.44 | zamba | Mango: yeah, i know that.. but what does it *do*? |
| 00:29.59 | Mango | It registers with the SIP provider so that they may send incoming calls to you. |
| 00:30.01 | zamba | Mango: do i need it to be able to receive calls from that peer or do i need it to make outgoing calls over that peer |
| 00:30.04 | zamba | ah |
| 00:30.05 | p3nguin | zamba: Register is mostly for outgoing. Sometimes you will be required to AUTH in order to make calls. |
| 00:30.12 | zamba | ok, two different answers |
| 00:30.20 | p3nguin | You don't register to get calls. |
| 00:30.22 | zamba | which one is it? |
| 00:30.24 | raden_work | WTF do you want you VOIP providecr to handle music on hold ? |
| 00:30.29 | Mango | p3nguin: If I don't have to send the packets, then it saves bandwidth. |
| 00:30.42 | raden_work | Mango, then get g.729 |
| 00:30.45 | p3nguin | mango: Your thinking is flawed. |
| 00:30.50 | Mango | Oh? |
| 00:31.01 | raden_work | agress with p3nguin |
| 00:31.29 | raden_work | Mango, if you have them on hold you still have a connection to them |
| 00:31.37 | raden_work | doesnt matter who plays the darn music |
| 00:32.08 | Katty | unless you use silence supression |
| 00:32.13 | Katty | does asterisk support silence supression stuff? |
| 00:32.18 | *** join/#asterisk obnauticus (n=l@about/windows/regular/obnauticus) |
| 00:32.20 | Mango | No, no silence suppression afaik. |
| 00:32.35 | Mango | So you guys are saying even when I put a call on hold, my phone is still using 80Kbit/sec up and down? |
| 00:32.51 | Katty | yes. |
| 00:33.01 | p3nguin | zamba: Let me give you an example. I have a DID with an ITSP, but I have no reason to register to them. They just deliver SIP to my IP address, where I have a peer context for them. |
| 00:33.02 | Katty | well, depending on yoru codec |
| 00:33.06 | raden_work | Mango, YES |
| 00:33.14 | raden_work | what codec ? |
| 00:33.16 | zamba | p3nguin: DID? |
| 00:33.26 | p3nguin | zamba: phone number for incoming calls |
| 00:33.27 | Katty | the better the codec, the more bandwidth you need to accomodate |
| 00:33.31 | zamba | p3nguin: ok |
| 00:33.41 | Mango | raden_work: What is it sending then? Dead air? Why would it do that? |
| 00:33.43 | raden_work | zamba, direct inward dial |
| 00:33.50 | zamba | p3nguin: so you don't have any register line for them? |
| 00:33.51 | p3nguin | zamba: You'll use a register string when you call out. |
| 00:33.55 | p3nguin | zamba: That's right. |
| 00:33.57 | raden_work | Mango, you need to understand network protocals |
| 00:34.00 | zamba | but you don't *need* to |
| 00:34.05 | zamba | sometimes it'll work anyway, right? |
| 00:34.14 | Katty | that's why silence supression was invented. |
| 00:34.19 | raden_work | an empty packet just as full as a full one and a full one as empty as a empty one |
| 00:34.29 | p3nguin | zamba: I do not have any register string for my origination service (the phone number where people call me). |
| 00:34.47 | raden_work | Katty, to my knowledge asterisk does not have silence suppression cause thats usually taken care of on client side |
| 00:35.07 | raden_work | Mango, what codecs u using |
| 00:35.12 | p3nguin | zamba: On the other hand, I do have a register string on my termination provider (the way I make calls). |
| 00:35.20 | Katty | raden_work: silence supression was invented to cut down on bandwidth |
| 00:35.33 | Katty | raden_work: if the System(tm) doesn't Hear(tm) anything, it doesn't send the full audio stream |
| 00:35.37 | raden_work | Katty, yes that i know :) |
| 00:35.52 | zamba | p3nguin: i'm able to dial without using a register line |
| 00:35.58 | raden_work | i just saying the way he wants to use it asterisk does not support it |
| 00:36.01 | Mango | raden_work: G.711 |
| 00:36.11 | p3nguin | zamba: They must not be requiring authentication, then. |
| 00:36.33 | hardwire | anybody come up with a clever way of detecting and ignoring a false answer over SIP? |
| 00:36.45 | raden_work | so 90 kbps |
| 00:36.46 | zamba | p3nguin: but that's perfectly valid and common, right? |
| 00:36.47 | Mango | p3nguin: I also dial without a register line. My SIP provider requires defaultuser= and secret= in the relevant context in sip.conf. |
| 00:36.53 | hardwire | do you have to loop it back around through PSTN channels? |
| 00:37.00 | hardwire | just to use libtonezone? |
| 00:37.14 | raden_work | Mango, why not use a diffrent codec ? |
| 00:37.24 | raden_work | how little bandwith do you have ? |
| 00:38.03 | Mango | I like the sound of G.711. |
| 00:38.06 | Mango | 512Kbit |
| 00:38.31 | p3nguin | zamba: If it works, then there is no problem. If you can't make calls, then you should look into why not. The obvious reason, in that scenario, might be that you are required to register and you aren't doing it. |
| 00:40.11 | Mango | logs into the router to check bandwidth usage |
| 00:40.51 | raden_work | Mango, 512 async ? |
| 00:41.05 | Mango | 7.5Mbit down, 512Kbit up |
| 00:41.16 | raden_work | Mango, whats the problem then ? |
| 00:41.38 | Mango | When I place a call on hold, bandwidth usage for my phone drops to 0. |
| 00:42.01 | raden_work | 768 / 6.0 we can hold 8 calls on 711 plus some people on hold no issues |
| 00:42.11 | p3nguin | If you play music, it'll go back up. |
| 00:42.47 | Mango | Asterisk was playing music. The phone wasn't. |
| 00:43.02 | p3nguin | It plays music to the phone. |
| 00:43.09 | p3nguin | The phone receives it. |
| 00:43.19 | p3nguin | Creating a usage of bandwidth. |
| 00:43.22 | Mango | No, my phone has placed the call on hold. |
| 00:43.30 | Mango | the other phone receives the music. |
| 00:43.40 | p3nguin | Oh, yeah... duh. |
| 00:43.42 | Mango | ;) |
| 00:43.54 | Mango | So back to my original question... |
| 00:44.05 | p3nguin | I'm finally on the same page. |
| 00:44.08 | Mango | hehe |
| 00:44.25 | p3nguin | I thought you were worried about your phones having music playing when you were on hold. |
| 00:44.31 | Mango | ah, no no no. |
| 00:44.55 | raden_work | no he wants people when they are on hold to get the music from his ITSP instead of his internal asterisk server |
| 00:45.00 | Mango | Exactly. |
| 00:45.16 | zamba | hm.. i have a strange problem here.. i'm able to receive incoming calls from my ITSP (?) if i configure the account in x-lite.. but not when i set the same account up in asterisk |
| 00:45.16 | p3nguin | Yeah, I'm with him now. I wasn't understanding the direction of the call and who was on hold. |
| 00:45.31 | raden_work | zamba, errors ? |
| 00:45.45 | zamba | raden_work: well, i don't know, since i can't see what happens at the ITSP end |
| 00:45.53 | zamba | raden_work: when i call it just goes silent.. nothing happens |
| 00:45.56 | raden_work | p3nguin, i still think it ridiculous for the ITSP to handle music |
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| 00:46.09 | raden_work | zamba, you in asterisk CLI ? |
| 00:46.14 | zamba | raden_work: yup |
| 00:46.23 | Mango | Raden: Why? |
| 00:46.24 | raden_work | is anything showing up in there ? |
| 00:46.39 | zamba | nope, nothing at all |
| 00:46.42 | Mango | zamba, is verbose on? |
| 00:46.43 | raden_work | Mango you still have a datastream connecting to your asterisk box |
| 00:46.46 | zamba | so the call doesn't reach my asterisk |
| 00:46.49 | zamba | Mango: yup, set to 9 |
| 00:46.52 | raden_work | I dont even know howd youd pass that to make it happen |
| 00:46.56 | p3nguin | raden_work: I do too, but at least he has presented a valid reason for it. |
| 00:47.08 | zamba | raden_work: so the problem is probably related to how i register at the other end |
| 00:47.11 | raden_work | he has plenty of bandwith for 2 phones on 711 |
| 00:47.24 | raden_work | zamba, firewall ? |
| 00:47.24 | Mango | Ok |
| 00:47.28 | zamba | raden_work: the stuff inside the brackets ([ ]), is that relevant for anything? |
| 00:47.34 | p3nguin | I would let my box handle it and not worry about bandwidth usage, but that's just me. |
| 00:47.35 | Mango | Let's say I have exactly 180Kbit/sec upstream |
| 00:47.36 | Mango | lol |
| 00:47.44 | raden_work | port 5059-5061 forward and 10000-2000 |
| 00:47.59 | raden_work | zamba, pastebin |
| 00:48.26 | raden_work | 90 kbps total up and down |
| 00:48.31 | p3nguin | I never actually graphed a phone's usage, but I estimate I have enough bandwidth for a couple hundred simultaneous calls using ulaw. |
| 00:48.59 | Mango | >.< |
| 00:49.05 | Mango | I envy you. |
| 00:49.33 | zamba | raden_work: http://pastebin.com/d63e63d2a |
| 00:50.08 | raden_work | Mango, u have silence suppression on your phone ? |
| 00:50.28 | Mango | raden, Asterisk does not support silence suppression. |
| 00:50.49 | raden_work | zamba, show me your whole sip.conf |
| 00:51.21 | raden_work | Mango, asterisk does not, but like my aastra 9133i phones do so i can use it |
| 00:51.24 | zamba | the only other relevant bits are the global stuff, right? |
| 00:51.31 | raden_work | yeah |
| 00:51.43 | raden_work | and what provider ? |
| 00:51.47 | Mango | On that topic, how do you like the 9133i? |
| 00:52.04 | Mango | I've heard good things about htem but never actually used one. |
| 00:52.07 | raden_work | i tested 5 phones under $100 it stomped the hell outta them |
| 00:52.24 | raden_work | better sound Q than polycoms i tried for 180 something |
| 00:52.41 | zamba | raden_work: http://pastebin.com/d6f17046a |
| 00:52.45 | raden_work | loud, clear, easy to configure , fast setup |
| 00:52.51 | zamba | raden_work: that's all i've got apart from the relevant peer |
| 00:52.51 | raden_work | takes me about 2 min per phone |
| 00:53.29 | raden_work | zamba, what country u i n ? |
| 00:53.37 | zamba | raden_work: norway |
| 00:54.29 | raden_work | how is asterisk registering to your ITSP ? |
| 00:54.37 | fuxu2 | where in norway? |
| 00:54.39 | zamba | raden_work: the peer |
| 00:54.47 | zamba | fuxu2: up north |
| 00:54.56 | zamba | raden_work: which i pastebin-ed earlier |
| 00:54.56 | raden_work | zamba, im confussed |
| 00:55.12 | zamba | http://pastebin.com/d63e63d2a |
| 00:55.13 | raden_work | asterisk needs to register with your itsp to recieve calls |
| 00:55.27 | zamba | .. register => ..? |
| 00:55.48 | fuxu2 | zamba: anywhere near Lillehammer? |
| 00:56.00 | zamba | fuxu2: hehe, you're definitely not from norway ;) |
| 00:56.22 | zamba | fuxu2: nope.. and lillehammer is not north.. not by far |
| 00:56.28 | raden_work | <PROTECTED> |
| 00:56.39 | fuxu2 | zamba: lillehammer is north of Stavanger though, right? |
| 00:56.59 | zamba | raden_work: now i'm confused.. you said earlier that register mostly was for outgoing calls? |
| 00:57.08 | zamba | raden_work: but it now looks like it's pretty vital to incoming calls as well..? |
| 00:57.19 | zamba | fuxu2: not by far, i'd imagine |
| 00:58.30 | zamba | fuxu2: about 2 degrees |
| 00:59.00 | zamba | fuxu2: i'm about 5-6 degrees north of that again :) |
| 00:59.00 | raden_work | zambra when and where did i say that ? |
| 00:59.21 | fuxu2 | ahh ok.. ya know.. I meant to say Trondheim instead of Lillehammer.. |
| 00:59.27 | fuxu2 | but yeah.. I get ya |
| 00:59.40 | zamba | raden_work: sorry.. i didn't mean "you" as you, but you as in "you people" :) |
| 00:59.41 | zamba | raden_work: all the helpful ones :) |
| 01:00.04 | zamba | so then.. register is definitely for incoming calls.. and maybe only that? |
| 01:00.05 | raden_work | you need to register if you dont register they have to idea where to send the call to the peer part for outgoing |
| 01:00.30 | raden_work | zamba, some providers require you be registered for outgoing all matters |
| 01:00.50 | fuxu2 | zamba: my family is from Var Haug |
| 01:01.12 | fuxu2 | which I guess is near Sandnes which is near Stavanger |
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| 01:02.27 | zamba | raden_work: go it |
| 01:02.33 | zamba | fuxu2: ok.. cool.. ever been to norway, then? |
| 01:04.01 | fuxu2 | yeah back in '84 |
| 01:05.52 | zamba | ah, i was barely born then |
| 01:06.02 | zamba | raden_work: hm.. i'm totally messing up stuff now.... |
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| 01:07.09 | p3nguin | zamba: I have several origination providers (incoming calls), and ZERO of them require me to have a register string. |
| 01:07.37 | p3nguin | zamba: I have multiply termination providers (allowing me to make outgoing calls), and they all want me to use a register string. |
| 01:08.10 | zamba | p3nguin: well.. when i set up register i was able to get incoming calls, so :) |
| 01:08.18 | p3nguin | But registration is not directly related to making nor receiving calls. |
| 01:08.37 | zamba | but i have another problem here now.. i guess i can't have two registrations at the same provider? |
| 01:09.03 | p3nguin | Bah, I need to go home. |
| 01:09.18 | p3nguin | Why do you need to register twice? |
| 01:09.20 | Mango | p3nguin: Do you have a static IP? |
| 01:09.45 | p3nguin | mango: nah, I'm talking about dynamic peers. |
| 01:10.14 | zamba | p3nguin: two different sets of users |
| 01:10.19 | p3nguin | SIP delivery certainly doesn't require registration. Now if the provider does in order to deliver SIP to you, that's a different story. |
| 01:10.20 | zamba | p3nguin: but both are using the same provider |
| 01:11.31 | Mango | The advantage to registering for me is that it keeps the NAT hole open so I don't need to forward ports. |
| 01:12.08 | p3nguin | I'm not NATing, so I don't worry with that. |
| 01:13.49 | zamba | but not possible to register to the same provider twice from the same machine, but with different users? |
| 01:14.26 | p3nguin | I never tried. |
| 01:14.48 | p3nguin | I don't think it would work out very well if you're running on the same port. |
| 01:15.11 | Mango | Bah |
| 01:15.14 | p3nguin | You would get mixed traffic. |
| 01:15.18 | Mango | How do you set verbosity in 1.6? |
| 01:17.30 | zamba | p3nguin: can't i just set a different source port? |
| 01:25.18 | zamba | are the register statements and the [peer] declaration in any way related? |
| 01:26.20 | zamba | when you set up a register statement you end up with the extention to dial (after the '/').. in what context will it dial this extention? |
| 01:26.35 | zamba | extension* |
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| 02:15.40 | p3nguin | zamba: When you create the register string, you use whatever credentials the peer requires you to use. If the username is after the / and the username just so happens to be your phone number, great. It'll be easy to remember that way. |
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| 03:34.39 | levy | Hello Channel, Im looking for large call volume call recording, is Oreka GPL the recommended solution for this? |
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| 04:12.21 | g-ram | random question -- what's the most creative music on hold you've used? |
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| 04:46.42 | g-ram | c'mon -- anyone out there? |
| 04:47.02 | g-ram | if you could set up a playlist for MOH, what would you use? |
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| 04:48.17 | [8none1] | g-ram: use a live mic on a busy city street |
| 04:48.20 | p3nguin | the file system |
| 04:48.27 | g-ram | :) |
| 04:48.55 | p3nguin | Put your files into a directory and then configure the context to play from that directory. |
| 04:49.05 | g-ram | I'm thinking of either Boards of Canada "Hi Scores" |
| 04:49.18 | g-ram | Rachels "Systems/Layers" |
| 04:49.26 | p3nguin | Oh, you want to know what things we would put on our playlist? |
| 04:49.35 | g-ram | or Max Richter "The Blue Notebooks" |
| 04:49.46 | g-ram | yep, that's what I'm asking |
| 04:49.49 | p3nguin | I use piano and dulcimer music. |
| 04:49.59 | g-ram | someone suggested Kraftwerk earlier -- not a bad idea |
| 04:51.01 | g-ram | I want something different; ideally the customer will hear the hold music and first think "what the hell?" |
| 04:51.13 | g-ram | and then think "this is different, this is nice" |
| 04:51.15 | p3nguin | You want the shock factor? |
| 04:51.39 | g-ram | slight shock |
| 04:51.46 | p3nguin | NiN |
| 04:51.56 | g-ram | that might be too much shock |
| 04:51.59 | g-ram | :p |
| 04:52.22 | p3nguin | Puddle of Mudd |
| 04:52.26 | g-ram | that's why Boards of Canada is at the top of my list |
| 04:52.34 | g-ram | never heard Puddle of Mudd |
| 04:53.01 | p3nguin | Never heard "She fucking hates me"? |
| 04:53.21 | g-ram | that's still too much shock |
| 04:53.32 | p3nguin | Spineshank |
| 04:53.39 | g-ram | if you've got an e-mail i'll send you the track I'm thinking of |
| 04:53.46 | p3nguin | Slipknot |
| 04:54.06 | g-ram | slipknot? no way! |
| 04:54.14 | p3nguin | On a more serious note, consider The Fray. |
| 04:55.02 | *** join/#asterisk thansen (n=thansen@d137-186-151-2.abhsia.telus.net) |
| 04:55.27 | p3nguin | or Pearl Jam |
| 04:55.33 | g-ram | gewwww |
| 04:55.37 | p3nguin | No? |
| 04:55.38 | g-ram | I like pearl jam |
| 04:55.48 | g-ram | but it's too mainstream |
| 04:56.00 | p3nguin | Pick something not to popular from them. |
| 04:56.24 | p3nguin | Maybe even Coldplay |
| 04:56.33 | g-ram | I'm looking for something that will make at least 1/2 of our customers ask "what the hell was that, and where can I find it" |
| 04:56.39 | g-ram | coldplay is way too mainstream |
| 04:56.58 | g-ram | seriously, send me your e-mail and I'll send a few of the tracks I'm considering |
| 04:57.28 | p3nguin | Shattersphere ... they'll ask what the hell was that, but they probably won't ever want to hear it again. |
| 04:57.41 | g-ram | heh |
| 04:58.03 | g-ram | that's not quite what I'm going for, but you're half way there :p |
| 04:58.41 | p3nguin | If you are very selective, Flogging Molly. |
| 04:59.06 | g-ram | again, no way |
| 04:59.13 | p3nguin | They might not have much left after the final cut, though. |
| 04:59.30 | p3nguin | You're familiar with that one? |
| 04:59.34 | p3nguin | surprising |
| 05:01.20 | p3nguin | Evanescence? |
| 05:01.37 | g-ram | baah, not what I'm going for at all |
| 05:01.58 | g-ram | let me send you 3 clips |
| 05:02.02 | g-ram | they're short |
| 05:02.15 | g-ram | criticize them if you like, and if you enjoy them, keep them |
| 05:02.43 | g-ram | but I want music without words (that makes for better MOH, IMO) |
| 05:07.52 | tlarsen | g-ram: Maybe some sort of ambient? |
| 05:08.14 | tlarsen | g-ram: Lots of that has no words, and it is engaging but not too edgy. |
| 05:08.43 | g-ram | yep, I'm thinking ambient |
| 05:08.51 | g-ram | indie ambient |
| 05:11.53 | tlarsen | g-ram: Another sort of on-hold music I like is some of the classical-influenced anime soundtracks. |
| 05:12.02 | tlarsen | g-ram: When they do have lyrics, it is in Japanese. |
| 05:12.10 | g-ram | :) |
| 05:12.23 | g-ram | sounds good |
| 05:12.29 | *** join/#asterisk JakFrost (i=43f3a43f@gateway/web/freenode/x-sftgieffwanpdxqr) |
| 05:12.42 | g-ram | might I send you a track to review? |
| 05:12.49 | g-ram | or, may I? |
| 05:14.05 | tlarsen | g-ram: I'm not sure I'd be much of a judge. |
| 05:14.20 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-us/x-ndnuwvnlbygjskxv) |
| 05:14.42 | g-ram | I don't care much; would just be nice to bounce the idea off of someone before putting it into production |
| 05:14.46 | drmessano | prefers vintage green jelly for his MOH |
| 05:16.37 | g-ram | crap, I need to be at work in 6 hours :( |
| 05:16.52 | g-ram | should probably go to bed |
| 05:17.09 | JakFrost | I want to implement a simple menu system (IVR) for a PSTN (POTS) line just to offer options like (1) Location (2) Hours (3) Info (0) Operator and then act as a fall through to the old analog POTS phones hooked up to the single line. No VoIP/SIP required, just IVR menu functionality with PSTN in and PSTN out. Asterisk for PBX and hardware X100P for FXO to PSTN) is required. |
| 05:17.55 | JakFrost | Does the X100P card can handle PSTN out through the pass-through port or do I require a FXS to connect the old analog POTS phones to the PBX for output. |
| 05:19.19 | JakFrost | The setup is simple, 1 PSTN in for FXO and output would be 1 PSTN out with two phones sharing the line on it. |
| 05:20.08 | drmessano | No, the passthru is a passthru, not an FXS |
| 05:20.10 | JakFrost | I can't find info on how to do this setup without having to use VoIP output or with FXS interface for PBX to PSTN. |
| 05:20.35 | JakFrost | I was affraid that the pass-through would be no good. I read that there is a "delayed" pass-through but I don't think that would help me either. |
| 05:20.52 | drmessano | Um no |
| 05:21.12 | drmessano | Its all soldered together.. there is no delay or anything other than it being a passthru |
| 05:21.16 | JakFrost | The whole setup is just to get a simple menu based IVR for store info. I think that a PBX like Asterisk might offer way too many features. |
| 05:21.52 | JakFrost | So I need a FXS interface to connect the old analog phones then. Any recommendations for el Cheapo one? Internal would be enough. |
| 05:22.30 | drmessano | If youre gonna get an internal FXS, you need to chuck that shit X100P and get a real TDM card with daughterboards |
| 05:22.59 | JakFrost | Any recommendations for a 1 FXO / 1 FXS on the budget? |
| 05:23.18 | drmessano | How much of a budget? |
| 05:23.51 | JakFrost | Well, the cheapest is the best considering that this is a 1 PSTN setup for just one feature... IVR menu. |
| 05:25.12 | drmessano | TDM400P with 2 modules is about $200 I guess |
| 05:25.28 | JakFrost | I might have to go back and offer up some more features to the folks if they want to actually use a PBX but they would have to switch from POTS phones to SIP phones also. The issue is that I don't know if they need any other features and frankly voicemail will be probably ignored most of the time anyway. |
| 05:27.26 | *** join/#asterisk Tim_Toady (n=moi@adsl272-87.kln.forthnet.gr) |
| 05:27.31 | JakFrost | Maybe there is another solution less powerful than Asterisk PBX that could do their simple IVR menu. |
| 05:28.19 | drmessano | If they/you aren't willing to spend the money for even the analog card, you should probably look for something at Radio Shack or Walmart |
| 05:31.07 | JakFrost | They are willing to spend the money as long as it is reasonable, probably a few hundred or a K. I will have to go back now and price out a full PBX setup that would include 2 to 4 SPI phones. |
| 05:31.50 | p3nguin | You can get used SIP phones and save a substantial amount. |
| 05:32.46 | drmessano | Well, the fact that the X100P card is $25 compared to the $150+ for the next best solution should have tipped you off a bit |
| 05:32.57 | drmessano | I suggest doing some reading and getting more familiar |
| 05:33.17 | JakFrost | The whole situation is more of a question of how much of a setup to provide if they aren't going to be using any of the features except for the simple IVR menu. They have a single POTS line and they are fine with it now, but they want an IVR menu and they don't know if they care about anything else. |
| 05:33.52 | JakFrost | I just though I could use the X100P as a cheap solution but getting output back into POTS is the problem. |
| 05:34.24 | JakFrost | I could get the TDM400P for $100-$150 USD which is acceptable. |
| 05:34.24 | drmessano | The X100P is a winmodem that someone once made work with Asterisk.. its not a good solution for a production sysem |
| 05:34.37 | drmessano | ~x100p |
| 05:34.38 | infobot | it has been said that x100p is an obsolete card. You don't want to bother trying to make it (or any of the "digium compatible" clones) work. Get a TDM01B, and you will save your sanity, your hair, and countless other things. |
| 05:36.49 | *** join/#asterisk fiddur (n=fiddur@dhcp08.textalk.com) |
| 05:39.14 | JakFrost | So a TDM400P with one X100M (FXO) and one S110M (FXS) interface is what I would need. |
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| 06:02.05 | JakFrost | Thanks for your help. Good night. |
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| 06:19.34 | bipz | hello anybody have experience in lib-ss7 |
| 06:21.54 | bipz | anybody there? |
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| 07:59.19 | MWE | morning all (hi all) |
| 07:59.54 | MWE | can someone help me out with setting up a originate step by step for a meetme? |
| 08:03.58 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 08:07.47 | MWE | can someone help me out with setting up a originate step by step for a meetme? |
| 08:11.22 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 08:17.27 | kaldemar | MWE: still having problems with originate? what have you done so far? |
| 08:17.53 | MWE | kaldemar, I wanna make an originate for a meetme |
| 08:18.34 | MWE | but |
| 08:18.38 | MWE | I have 2 problems |
| 08:18.47 | MWE | 1. how can I execute some kind of originate |
| 08:19.07 | MWE | is it always in a AGI/SCRIPT or is it also possible to do that in the dialplan |
| 08:19.49 | MWE | 2. I don't know where I had to enter the roomnumber when the application is meetme |
| 08:20.01 | Chainsaw | Asterisk 1.6 & Patton gateways seem capable over SIP over TCP; does anyone know whether Cisco 7960G handsets can do this on P0S3-08-11-00 firmware? |
| 08:23.25 | MWE | kaldemar, even copy paste the script ^^ brb |
| 08:24.26 | MWE | http://pastebin.com/d219f85cb kaldemar |
| 08:24.52 | MWE | kaldemar, is it possible to execute an originate in the dialplan? |
| 08:27.27 | kaldemar | MWE: 1.6.2 will have an application to originate from dialplan. i thought we went through this already. |
| 08:27.47 | MWE | problem 3... Asterisk 1.4.23.1 |
| 08:28.44 | kaldemar | that's no problem, just use another origination method. |
| 08:30.08 | kaldemar | MWE: http://www.the-asterisk-book.com/unstable/asterisk-manager-api.html |
| 08:30.21 | kaldemar | MWE: http://www.voip-info.org/wiki/view/Asterisk+manager+Examples |
| 08:30.32 | MWE | the voip I already found |
| 08:30.39 | many | call Manager API from AGI |
| 08:30.41 | MWE | i've got a book: the future of telephony :) |
| 08:31.01 | kaldemar | many: what's the benefit of using AGI for that? |
| 08:31.05 | *** join/#asterisk Grof (n=dule@89.201.165.226) |
| 08:31.08 | Grof | need help |
| 08:31.23 | MWE | many, is that with a fsocket? |
| 08:31.54 | Grof | channel.c: Set channel Local/XXX to write format g723channel.c: Set channel Local/XXXto read format g723 |
| 08:32.13 | Grof | and after i try to Dial through DAHDI interface |
| 08:32.28 | Grof | channel.c:4106 ast_request: No translator path exists for channel type DAHDI (native 0x4c) |
| 08:32.51 | Grof | "core show translation" really does show that there is no translator path from g723 to other codecs |
| 08:33.22 | Grof | am i missing something? |
| 08:35.08 | kaldemar | asterisk supports g.723 in pass-through mode only |
| 08:35.31 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 08:36.57 | MWE | but is there another function that you can use to make a call and put the callee in a meetme room... or is it originate... but is it possible to use originate in a dialplan without any AGI script |
| 08:37.06 | Grof | how can i disable it? |
| 08:37.25 | Grof | why is Local channel defaulting to g723? |
| 08:39.04 | kaldemar | MWE: 1. no. 2. yes, by using a shell script or a call file. |
| 08:39.13 | *** join/#asterisk BrokenNoze (n=BrokenNo@host81-137-172-233.in-addr.btopenworld.com) |
| 08:39.38 | MWE | I read the first link, but I had to set some other variables in manger.conf is that always needed? |
| 08:39.38 | *** join/#asterisk misteranonymous (n=mister@udp115943uds.hawaiiantel.net) |
| 08:39.43 | BrokenNoze | hi, has anyone here had any experience using sangoma cards with qsig? |
| 08:39.53 | kaldemar | MWE: what other variables? ask specific questions. |
| 08:40.56 | MWE | euhm the first link says that I had to edit manager.conf. I've no idea what kind of effect it shall have on the other scripts. so is that change in manager.conf (enable=yes) really needed? |
| 08:41.48 | kaldemar | MWE: if you intend yo use the manager interface, yes. |
| 08:42.01 | kaldemar | what other scripts do you mean? |
| 08:42.49 | misteranonymous | hi, when i try to 'make' dahdi i get a floating point exception error, here is the make output http://pastebin.com/mba100cf |
| 08:42.51 | MWE | there are some PHP scripts in the agi-bin what I didn't make |
| 08:43.13 | MWE | so I don't know how the effect will be if I make a change in the manager.conf |
| 08:43.24 | MWE | and the originate function need a change in manager.conf? |
| 08:44.01 | kaldemar | MWE: enabling the manager interface won't harm anything. |
| 08:44.47 | kaldemar | and you don't have a function to do the origination, it's a manager interface command. |
| 08:45.53 | bipz | hello all anybody worked in libss7 |
| 08:46.18 | *** join/#asterisk MT`AwAy (n=MagicalT@shigoto.ookoo.org) |
| 08:47.26 | *** join/#asterisk Polysics (n=luca@host113-41-static.25-87-b.business.telecomitalia.it) |
| 08:47.29 | Polysics | hello |
| 08:47.52 | Polysics | can i have a condition to place answring users in a queue |
| 08:47.53 | Polysics | ? |
| 08:49.17 | Polysics | let me explain: i have answering SIP users that can obviosuly log in to the system |
| 08:49.26 | Polysics | but they also have "working hours" |
| 08:49.49 | Polysics | outside those "working hours" they can't receive calls even if they are logged to the system |
| 08:53.27 | Polysics | i suppose i should use some AGI script |
| 08:59.02 | troffasky | you can use time conditions inside the dialplan |
| 08:59.14 | troffasky | should be no need to go out to AGI for a simple time condition |
| 08:59.28 | troffasky | I use a time condition at home so I don't get calls after 11 at night :-) |
| 09:00.02 | Polysics | i will have time conditions on EACH user though, and quite complicated too |
| 09:00.13 | Polysics | like "monday from 8.00 to 12.00" |
| 09:00.20 | Chainsaw | Cisco 7960G, SIP-over-TCP support, Y/N? |
| 09:00.51 | troffasky | Polysics, are you saying every user has a different time condition? |
| 09:01.08 | Polysics | troffasky, yes, and different for days of the weeks and months of the year |
| 09:01.32 | troffasky | oh well, have fun doing that then ;-) |
| 09:02.15 | Polysics | that brings AGI back into the picture? |
| 09:02.35 | Polysics | it's basically a db query that says "yes, call me" or "no, don't call me" |
| 09:03.24 | troffasky | it sounds like you want to build a queuing system inside a queuing system |
| 09:03.58 | troffasky | but yeah, you can do DB queries from dialplan too |
| 09:04.19 | troffasky | but if you're more proficient in whatever AGI language you want to use, then might as well use that instead |
| 09:08.46 | Polysics | yeah, so far i have had more success with AGI logic |
| 09:08.57 | Polysics | i feel more at ease coding in PHP :-) |
| 09:09.18 | troffasky | so how does your queuing logic decide which agent to send a call to? |
| 09:09.29 | troffasky | I guess that would be the best place to add the time query |
| 09:14.13 | MWE | many do you have experience with originate in the AGI? |
| 09:15.11 | *** join/#asterisk kmate (n=mkoch@195.70.53.252) |
| 09:15.59 | Polysics | troffasky, the big picture is: each user is added on a queue based on which language he speaks |
| 09:16.29 | Polysics | so, when a call comes, an IVR asks the caller which language he wants to speak to |
| 09:16.39 | Polysics | from that on it is simple queue operation |
| 09:16.54 | Polysics | but there is this "working hours" problem |
| 09:18.43 | Polysics | the logic is "the first free person that speaks your language" |
| 09:18.56 | troffasky | right, but there must be some sort of logic that determines who is logged into a given queue? |
| 09:22.14 | Polysics | that is another thing i need to figure out yet :-P |
| 09:24.22 | kaldemar | MWE: there is no magical way of doing the origination in AGI. you have yo make a tcp connection to the manager interface and use it, or do it with a call file. |
| 09:24.58 | troffasky | Polysics, well my money is on, that being the right place to put your time check, whether you do it with dialplan logic or an external check |
| 09:25.03 | kaldemar | MWE: you can't do this with a copy & paste. you have to know what you're doing. |
| 09:25.29 | MWE | is a call file difficult to make? |
| 09:25.41 | *** join/#asterisk darkskiez (n=mhb@86.4.210.148) |
| 09:28.02 | kaldemar | MWE: it's just writing to a file. |
| 09:33.10 | kmate | hello all! i'm about to start a medium sized pbx project, and i need to plan the hw/sw setup for the system. if i tell some details could any of you help me with some advice, suggestion? |
| 09:34.41 | troffasky | irc is usually better suited to answering specific questions, but you could try asking anyway |
| 09:36.47 | kmate | okay, thx. so the task is that there is an existing network with a custom voip communication. there are about 50-100 clients and they are communicating 1 to 1 or in conference groups. |
| 09:37.30 | kmate | this existing stuff has to be connected to a SIP protocoll so that normal SIP clients can connect |
| 09:38.14 | kmate | so actually 1 conference server is needed which can handle ~100 clients |
| 09:39.41 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 09:40.09 | kmate | i was thinking about asterisk with a hw voice processing card. my first question can a single pc handle more then 1 of that cards? |
| 09:41.55 | kmate | (like TCE400B) |
| 09:42.39 | kmate | what kind of PC do I need for this? |
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| 09:48.42 | zamba | should the source port for asterisk be fixed? |
| 09:49.11 | zamba | i have this problem with incoming traffic being routed together, since i've registered twice (two different accounts) at a provider |
| 09:49.15 | Chainsaw | Cisco 7960G, SIP-over-TCP support, Y/N? |
| 09:58.29 | many | MWE: theres no originate in agi. however, calling an agi script which does originate to manager api or placing a call file aint difficult |
| 09:58.33 | many | now |
| 09:59.07 | many | if you dont know how to do it, you might be better off getting some tailored asterisk gui which does the magic for you |
| 09:59.49 | MWE | I just wana learn things. I'm a php-programmer so I 've got the exp. with php and with normal agi-bin scripts in phhp |
| 09:59.58 | MWE | so this should be done I guess:P |
| 10:01.08 | many | php i'd suggest callfiles... fopen(); fprint(yadayada); fclose(); |
| 10:01.18 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 10:01.51 | MWE | and asterisk will check for an call file, if there is a call file execute and remove? |
| 10:02.55 | many | yes |
| 10:03.00 | many | or you use http://www.straw-dogs.co.uk/asterisk-api-php/ or something |
| 10:03.11 | MWE | omg thatś easy :X |
| 10:16.04 | BrokenNoze | anyone used qsig with sangoma? seem to be getting t203 counter reset and link drop followed by reestablish? |
| 10:16.11 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
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| 10:21.03 | fofware | Hello guys, Is possible set different languages in mails that Voicemail send? |
| 10:28.37 | *** join/#asterisk jcape (n=jcape@75.21.66.12) |
| 10:36.28 | MWE | how can you give a variable with the call file like the meetme room? |
| 10:42.46 | MWE | o// |
| 10:46.31 | kmate | troffasky: as noone seems to answer my question, could you recommend a better place to ask it? maybe some forum or dedicated mailing list? |
| 10:47.21 | troffasky | could be a matter of timing |
| 10:47.31 | MWE | kmate, i wanna help you but my experience with asterisk is not really good :P |
| 10:47.34 | troffasky | this channel is a lot busier later on |
| 10:47.44 | troffasky | and I don't know anything about voice hardware |
| 10:48.04 | troffasky | you might be better off asking the vendor of said hardware what they recommend |
| 10:48.56 | troffasky | eg they might know it won't work with certain mobo chipsets |
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| 11:01.45 | BrokenNoze | anyone know why my q921.c might keep reporting a release followed immediately by a established message? |
| 11:02.38 | *** join/#asterisk dustybin (n=dustybin@thinkdebian.org) |
| 11:08.01 | kmate | troffasky, MWE i see, thanks. maybe i will try once more later |
| 11:08.28 | MWE | good luck m8 |
| 11:08.40 | *** join/#asterisk cuco (n=elcuco@bzq-79-182-99-45.red.bezeqint.net) |
| 11:11.20 | *** join/#asterisk rethus (n=rethus@xdsl-81-173-176-105.netcologne.de) |
| 11:12.07 | rethus | i have suse 11.1 with kernel 2.6.27.29. ztdummy is loaded, but i get allways "That is not a valid Confernce Number". |
| 11:12.13 | rethus | whats wrong here? |
| 11:12.44 | *** join/#asterisk cuco (n=elcuco@bzq-79-182-99-45.red.bezeqint.net) |
| 11:13.21 | MWE | many kaldemar both thnx alot. I fixed this problem with a call file :) |
| 11:13.27 | cuco | tzafrir_laptop: ping |
| 11:13.29 | MWE | it was easier than I thought :) |
| 11:13.51 | MWE | but still have one question... |
| 11:14.27 | MWE | when the call fails or something, the script will give +101? |
| 11:16.23 | zamba | does asterisk always use 5060 as source port when registering with providers? |
| 11:16.57 | *** join/#asterisk cuco (n=elcuco@bzq-79-182-99-45.red.bezeqint.net) |
| 11:30.08 | *** join/#asterisk icyValk77 (n=icyValk7@213.129.64.4) |
| 11:35.37 | MWE | is there a way to kick all the users when somebody hangsup with the meetme? |
| 11:40.24 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 11:42.06 | leifmadsen | happy friday! |
| 11:52.10 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:52.10 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:02.02 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 12:05.43 | *** join/#asterisk dijungal (n=kdaniel@199.85.237.63) |
| 12:05.52 | dijungal | good day.... |
| 12:06.50 | kaldemar | MWE: how did you execute the script? |
| 12:06.57 | dijungal | i've applied Deny= 0.0.0.0/0.0.0.0, allow=10.68.0.0/255.255.248.0 to an extension yes i can still register from a public address to that extension and make calls.... what's i;m i missing here |
| 12:06.58 | dijungal | ? |
| 12:07.33 | MWE | kaldemar, I've made a call file. Asterisk wil run that file and everything is working. Now the only problem is, when the callee hangs up the caller had to kicked out... |
| 12:07.38 | kaldemar | MWE: and hopefully you used an atomic file operation (=move) to put the call file in the spool dir. |
| 12:07.51 | *** join/#asterisk greysd (n=oae2@ns.plasma-prospect.com) |
| 12:07.58 | MWE | not directly in the spooldir? |
| 12:08.36 | *** join/#asterisk superciuc (n=alexandr@ip62.trivenet.it) |
| 12:08.51 | kaldemar | writing it directly in the spooldir is bad, asterisk might read it when it's not completely written, which leads to unexpected behavior. |
| 12:09.09 | MWE | that way :) |
| 12:09.48 | MWE | for the test it's good now... just fix that last problem and it works fine for what I want :P |
| 12:11.12 | dijungal | i've applied Deny= 0.0.0.0/0.0.0.0, allow=10.68.0.0/255.255.248.0 to an extension yes i can still register from a public address to that extension and make calls.... what's i;m i missing here? |
| 12:12.55 | superciuc | Is there someone here who tried t38 passtrough with patton devs? |
| 12:12.57 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 12:13.25 | *** join/#asterisk regan40 (n=regan@c122-106-234-217.belrs3.nsw.optusnet.com.au) |
| 12:13.29 | Naikrovek | omfg i finally got my cert errors in exchange resolved |
| 12:13.30 | Naikrovek | woot |
| 12:15.28 | dijungal | cert erros in exchange? |
| 12:15.29 | regan40 | hi |
| 12:15.47 | Naikrovek | dijungal: yes, was offtopic but i'm so glad i got them fixed |
| 12:15.59 | Naikrovek | regan40: hi. just ask your question |
| 12:16.19 | *** join/#asterisk kaii (n=kai@ciphron.de) |
| 12:16.34 | *** join/#asterisk yang (n=yang@freenode/sponsor/cacert.assurer.yang) |
| 12:16.51 | dijungal | oooh i thought there was some asterisk exchane integration |
| 12:16.52 | dijungal | lol |
| 12:17.03 | Naikrovek | there is actually |
| 12:17.09 | Naikrovek | it's called unified messaging |
| 12:17.14 | dijungal | yea... just googled it.. :s |
| 12:17.16 | dijungal | nice |
| 12:18.16 | regan40 | has a Dialogic D41 and wrote some code for it under windoze but is thinking about getting a new machine and a another card that asterisk support.. must be analog.. what is cheap on ebay...? |
| 12:18.21 | kaldemar | dijungal: based on that, no one can give you an answer. show the configuration and a sip debug of the registration. |
| 12:18.35 | *** join/#asterisk superbeef (n=superbee@74.84.194.4) |
| 12:19.05 | superbeef | If I have an FXS card directly in my PBX, will modem speeds still only be 9600bps? |
| 12:19.40 | kaii | for debugging purposes i would love to set a CDR(userfield) to indicate wether the caller or the callee initially hung up. but i have no idea how to achieve it. ideally, in my mind, this would be done in the hangup extension. any ideas? |
| 12:21.10 | kaldemar | kaii: or with Dial option g |
| 12:21.10 | Naikrovek | superbeef: modem has nothing to do with fxs |
| 12:21.31 | Naikrovek | dial option g, eh |
| 12:21.39 | Naikrovek | never heard of that |
| 12:21.50 | kaii | Naikrovek: # g: When the called party hangs up, exit to execute more commands in the current context. |
| 12:22.12 | superbeef | Naikrovek: if I plug an analog modem into an FXS port it would seem related |
| 12:22.31 | kaii | kaldemar: i already thought of that, but that would have huge effects on my dialplan and would mean i have to rewrite 50%+ of it. |
| 12:22.38 | dijungal | kaldemar: thanks for the hint.... the extension i was testing didn't have restrictions on it... :o) |
| 12:22.48 | Naikrovek | superbeef: ah yes, sorry. i missed that. you'd be limited to whichever is slower, i would think by the modem |
| 12:23.01 | kaii | kaldemar: dialplan should only continue if Dial() is not answered |
| 12:23.28 | superbeef | Naikrovek: so i'll have better performance than using one of the Network based analog adapters? I've only hit 9600bps with those |
| 12:23.58 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:38e2:590a:bfa9:772e) |
| 12:24.14 | kaii | kaldemar: i was hoping for some "noninvasive" solutions |
| 12:24.26 | Naikrovek | superbeef: i'm not an fsx expert, but i would imagine that if your modem can reach 33.6k or whatever it is to a telco, that it could do the same to an fxs port. |
| 12:26.29 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 12:29.11 | *** join/#asterisk scalex000 (n=chatzill@180.120.88.200.f.sta.codetel.net.do) |
| 12:29.16 | scalex000 | good morning |
| 12:29.31 | *** join/#asterisk jcape (n=jcape@adsl-99-132-248-218.dsl.chcgil.sbcglobal.net) |
| 12:30.27 | scalex000 | Hi have question. when I dial in another system and the number I dial need number 1 to go through, the another system only ringing but not give the messages back to asterisk |
| 12:30.46 | *** join/#asterisk Grof (n=dule@89.201.165.226) |
| 12:32.04 | kaldemar | scalex000: what is the quoestion? |
| 12:33.19 | troffasky | good luck with that superbeef, I've never had much luck with modems thru IP PBXes |
| 12:35.34 | superbeef | troffasky: I've had decent luck using USR couriers on Analog adapters at 9600.... I'm hoping that taking IP out of the equation with an FXS card will give better performance |
| 12:37.25 | Naikrovek | how are you going to take the IP out of the question AND use an FXS port |
| 12:37.28 | *** join/#asterisk tnt_ (n=tnt_@212.166.48.236) |
| 12:38.02 | tnt_ | Hi. Is there somewhere I can find infos about building modules outside the main tree ? |
| 12:38.11 | troffasky | how would that *not* take IP out of the equation? |
| 12:38.25 | troffasky | ie using FXS instead of an ATA |
| 12:39.07 | Naikrovek | FXS ports are necessarily connected to IP PBXs, yes? otherwise it's just an analog POTS port |
| 12:39.17 | superbeef | Naikrovek: an FXS PCI card directly in the PBX |
| 12:39.35 | *** join/#asterisk Nasra (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 12:40.07 | Naikrovek | yes but the PXS PCI card is in an IP PBX, right? |
| 12:40.15 | troffasky | lol go back to sleep Naikrovek |
| 12:40.26 | Naikrovek | piss off |
| 12:40.36 | Naikrovek | explain it to me if i've got it wrong |
| 12:40.51 | superbeef | Naikrovek: PBX also has T1 for voice, so if I go through hte local FXS card out the local t1 on the same box I'll never hit IP |
| 12:41.30 | Naikrovek | superbeef explained it, why couldn't you, troffasky |
| 12:41.32 | xrmx__ | does anybody know a german did provider? |
| 12:41.37 | troffasky | sipgate |
| 12:41.43 | troffasky | I thought it was obvious Naikrovek |
| 12:41.54 | Naikrovek | troffasky: you assume incorrectly |
| 12:42.15 | Naikrovek | who uses a voice T1 with an IP PBX? they're more expensive than data T1s |
| 12:42.55 | superbeef | Naikrovek: my company does..... |
| 12:43.11 | troffasky | makes more sense to me than using 24 analog FXOs FFS |
| 12:43.16 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 12:43.49 | Naikrovek | troffasky: yes it does but if you have an IP PBX, you don't NEED 24 FXOs |
| 12:43.57 | Naikrovek | unless you have all analog phones |
| 12:44.24 | Naikrovek | again, why? you wouldn't need a freaking IP PBX if you had a voice T1 anyway |
| 12:44.32 | superbeef | Naikrovek: So you'd rather just have a SIP trunk over a T1 from a shitty provider than a T1 with 23 crystal clear digital voice channels from the Local provider? |
| 12:45.20 | Naikrovek | i have 40 crystal clear voice channels over a T1 from a local SIP provider |
| 12:45.25 | Naikrovek | so yeah it makes no sense to me |
| 12:46.28 | Naikrovek | whatever. you two have a special relationship or something and i'm apparently the enemy |
| 12:46.35 | superbeef | hahaha |
| 12:46.51 | MWE | kaldemar, it's now workin this way: somebody calls, create a meetme room, save some variables, run another AGI script which makes a call file and give the variables. put the caller in the room and call out the callee |
| 12:46.56 | troffasky | its possible to enjoy the benefits of an IP PBX without using SIP PSTN interconnect |
| 12:47.38 | Naikrovek | it's also possible to take the bus when you own an aston martin |
| 12:48.04 | Naikrovek | (makes no sense to me) |
| 12:48.19 | Naikrovek | you guys do whatever the hell you wanna do, i'll do the same |
| 12:48.35 | troffasky | what if you wanted to go to the pub in the evening? you wouldn't drive |
| 12:48.39 | troffasky | it's horse for courses |
| 12:48.43 | superbeef | haha |
| 12:48.58 | superbeef | I have 2 cars, but i still ride my bike places..so.. T1 voice for life |
| 12:49.53 | Naikrovek | fair enough |
| 12:50.53 | superbeef | Naikrovek: The real reason we don't do SIP trunks is because we have a big contract with ATT for all our services, and they dont do SIP trunks right now |
| 12:51.08 | creativx | ~siptrunk |
| 12:51.09 | infobot | i heard siptrunk is To set a SIP peer/friend/user as a trunk add either trunk=yes or wombat=yes (they both do the same thing) in the peer/friend/user definition in sip.conf |
| 12:51.10 | Naikrovek | so how do you tie the voice t1 into an ip pbx. is there a digium card for that |
| 12:51.19 | Naikrovek | ah yeah there is |
| 12:51.21 | Naikrovek | i see now |
| 12:51.25 | superbeef | Naikrovek: yeah that's their cash cow |
| 12:51.32 | Naikrovek | wtf troffasky was right. i do need more sleep |
| 12:51.46 | troffasky | yeah, you know you need sleep when I start being right about stuff ;-) |
| 12:51.55 | Naikrovek | i don't think each half of my brain was talking to the other for a while there |
| 12:52.07 | Naikrovek | that's what i get for fighting MS Exchange first thing in the morning i guess |
| 12:52.31 | Naikrovek | sorry everyone |
| 12:52.35 | Naikrovek | huuuuge brain fart |
| 12:52.38 | *** join/#asterisk garymc (n=garymc@host86-163-43-91.range86-163.btcentralplus.com) |
| 12:53.34 | troffasky | some SPs will let you mix them as well, buy ISDN voice off them and use SIP as well for the same numbers |
| 12:54.05 | troffasky | so you can fail over in either direction or use SIP to add more channels, cheaper than renting more ISDN channels |
| 12:54.11 | Naikrovek | yeah |
| 12:55.30 | fofware | Hello guys, How i can send mail from voiceMain in different languages? |
| 12:56.05 | fofware | the notification mail for each user language |
| 12:57.18 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 12:59.42 | Katty | GOOD MORNING :> |
| 13:01.06 | fofware | Katty: mornig |
| 13:01.30 | Katty | fofware: hello. |
| 13:02.59 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 13:03.12 | MWE | somebody an idea how to kick every one when one of the parties hang up in a meetme room? |
| 13:03.52 | Katty | hugs jaytee |
| 13:04.07 | jaytee | morning Katty *hugs* |
| 13:04.28 | Katty | hmm. there was a web app...that the rhino box had. |
| 13:04.37 | Katty | that let you interact with the meetme conference room |
| 13:05.31 | MWE | yeah I know there were some webapps, but this had to be done when a party hangs up.. |
| 13:05.34 | scalex000 | kaldemar: sorry Im back |
| 13:05.38 | MWE | I thought it was x but that will close the room when the last participant is gone |
| 13:05.42 | Naikrovek | well FOP can kick everyone out I think, but you gotta do a double click on the green bubble, as I recall |
| 13:06.04 | Katty | yeah not exactly what he wants tho |
| 13:06.05 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 13:06.35 | MWE | no because you will be called and when you picks up you enter the room |
| 13:07.09 | *** join/#asterisk mumtazah1 (n=mumtazah@124.82.79.96) |
| 13:07.09 | MWE | you can sit anywere you want... *were the phone is* |
| 13:08.53 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:09.35 | kaldemar | MWE: not with MeetMe, but newer versions have an application called Page which does just that. |
| 13:09.41 | Grof | NOTICE[13100]: channel.c:2946 __ast_read: Dropping incompatible voice frame on Local/xxx@test-e9ca;2 of format ulaw since our native format has changed to 0x3fff0001 (g723|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140) |
| 13:09.42 | Grof | ? |
| 13:09.58 | Grof | why is native format being changed? |
| 13:10.03 | Grof | on RetryDial |
| 13:10.05 | Grof | ? |
| 13:10.31 | kaldemar | MWE: http://www.voip-info.org/wiki/view/Asterisk+cmd+Page |
| 13:11.50 | *** join/#asterisk Carlos_Tico (n=grillo_v@c-98-201-162-34.hsd1.tx.comcast.net) |
| 13:12.17 | kaldemar | MWE: seems to be in 1.4 too. |
| 13:12.31 | MWE | I reading how I can do it.. |
| 13:13.04 | MWE | maybe I will add a d to the meetme insert of the callee and do it with a while loop... |
| 13:13.33 | Carlos_Tico | got a question to ring 2 extensions simultanly |
| 13:13.44 | *** join/#asterisk dymaxion (n=dymaxion@host86-172-50-54.range86-172.btcentralplus.com) |
| 13:14.11 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 13:14.53 | Naikrovek | Carlos_Tico: ask. it's pretty easy to ring two at once |
| 13:15.36 | Carlos_Tico | lets see |
| 13:15.39 | Carlos_Tico | Naikrovek |
| 13:15.43 | kaldemar | Carlos_Tico: Dial(Tech/first&Tech/second) |
| 13:15.50 | Carlos_Tico | exten=s,n,Dial(Zap/g1/8323401414&Zap/1,20,i) |
| 13:15.54 | Carlos_Tico | whats wrong on that ? |
| 13:16.05 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:16.06 | MWE | g1? |
| 13:16.07 | troffasky | xrmx__, that's sipgate.de btw, not sipgate.com |
| 13:16.20 | kaldemar | Carlos_Tico: nothing, as itself |
| 13:16.43 | dijungal | g1 - HTC :D |
| 13:16.43 | Carlos_Tico | they ring together but only a couple of times |
| 13:17.03 | Carlos_Tico | g1 i think is the fxo |
| 13:17.39 | troffasky | so if you ring them each individually, do they ring the number of times you would expect? |
| 13:17.46 | Carlos_Tico | yes |
| 13:17.46 | kaldemar | Carlos_Tico: g1 is a group of channels defined in zapata.conf below "group => 1" |
| 13:18.02 | Carlos_Tico | oh ok |
| 13:19.44 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:19.44 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:19.49 | Carlos_Tico | well |
| 13:19.57 | Carlos_Tico | maybe i can show you the CLI |
| 13:21.52 | *** join/#asterisk spck (n=spck@unioncab.com) |
| 13:21.53 | Carlos_Tico | http://pastebin.com/d7adf9e4f |
| 13:22.43 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 13:24.07 | *** join/#asterisk Chesther (n=cam2@cam2-mac.cit.cornell.edu) |
| 13:30.14 | Carlos_Tico | any idea |
| 13:30.36 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:31.42 | Katty | http://upload.wikimedia.org/wikipedia/commons/4/49/Fredmeyer_edit_1.jpg |
| 13:32.58 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 13:33.02 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
| 13:33.23 | jaytee | I used to have the Fred Meyer jingle get stuck in my head as an "earworm" all the time when living in Oregon |
| 13:33.42 | Katty | jaytee: http://www.geekologie.com/2009/09/02/tactical-bacon.jpg <- i saw this and thought of you. |
| 13:34.09 | Katty | jaytee: i never really thought about all the products on the market. |
| 13:34.20 | Katty | jaytee: and that's just a Fredmeyer...imagine walmart :/ |
| 13:34.22 | jaytee | tactical bacon! love it |
| 13:35.02 | creativx | man that is a lot of crap there Katty |
| 13:36.21 | garymc | anyone in the uk got a decent digit map i could use. Ive got 3 digit extensions and I want to dial 9 to get an outside number. My emergency number is 999 |
| 13:36.44 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:40.42 | garymc | well im just having problems understanding digitmaps. I suppose ill just do the old "trial and Error" technique |
| 13:42.20 | troffasky | yeah I'm sure 999 will love that :-) |
| 13:43.33 | garymc | ?? |
| 13:43.38 | garymc | Maybe its a dial plan i need? |
| 13:44.06 | garymc | oh yeah 999 wont like me testing them lol troffasky |
| 13:46.38 | troffasky | not sure what a digit map is, but a dialplan determines what goes where when you dial something |
| 13:46.44 | troffasky | so I guess a dialplan is what you want |
| 13:47.21 | *** join/#asterisk Polysics (n=luca@host113-41-static.25-87-b.business.telecomitalia.it) |
| 13:47.22 | Polysics | hello |
| 13:47.37 | Polysics | i have been trying to make a custom sound for my IVR, but so far i only hear gibberish |
| 13:47.51 | Polysics | what's a recommended way of recording files for Asterisk? |
| 13:48.17 | Chesther | Set up an extension you can call into that fires off the recording app? |
| 13:49.12 | *** join/#asterisk voipmonk (n=voipmonk@67.204.45.155) |
| 13:49.53 | *** join/#asterisk flujan (n=flujan@201-68-49-1.dsl.telesp.net.br) |
| 13:50.01 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 13:55.25 | Polysics | i don't like the results, audio comes out noisy |
| 13:55.38 | Polysics | if all else fails, that already works for me |
| 13:55.51 | Polysics | but there must be a proven way to record a sound for asterisk :-) |
| 13:56.11 | Chesther | What kind of handset are you using to record? |
| 13:56.41 | Chesther | (In my experience so far, nothing is proven in Asterisk until you've done it yourself.) |
| 13:58.05 | scalex000 | kaldemar: I need help |
| 14:01.01 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 14:01.51 | *** join/#asterisk psilikon (n=psilikon@static-173-65-4-24.tampfl.fios.verizon.net) |
| 14:02.24 | psilikon | p3nguin, what up homey. thanks for the help the other day |
| 14:02.57 | Polysics | Chesther, using Pc headphones |
| 14:03.17 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 14:03.27 | Polysics | i know i should probably have something better, but recording locally allows me to use noise reduction and similar stuff |
| 14:04.08 | Chesther | Do you have a SIP phone that you can register to Asterisk? |
| 14:04.14 | garymc | Polysics, try using a polycom phone plugged into your server, recordings sound clear as a whistle |
| 14:04.35 | Chesther | That's what I was thinking. |
| 14:04.51 | garymc | troffasky : where can i find a good explanation on dial plans then? |
| 14:06.01 | troffasky | ~dialplan |
| 14:06.02 | infobot | i guess dialplan is the thing configured in extensions.conf |
| 14:06.10 | troffasky | hmm, http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 14:07.23 | Polysics | i don't have a polycom phone :-) |
| 14:07.26 | *** join/#asterisk moy (n=moy@74.12.127.128) |
| 14:07.34 | Polysics | but what if i need moh stuff? |
| 14:07.51 | Polysics | can't record everything on the phone... need to find out the proper compression for * |
| 14:09.11 | *** join/#asterisk Gumug (n=Gumug@nmd.sbx09566.joplimo.wayport.net) |
| 14:09.35 | superbeef | i like infobot's lack of confidence |
| 14:10.41 | Chesther | Polysics: Yeah, if all you're looking to record is voice prompts, then doing it from a phone is the easiest. If you don't have a standalone SIP phone, a good headset and a softphone will do it. |
| 14:11.10 | Chesther | If you've got existing recordings for MOH and you want to translate it so * can use it, that's a diffrent problem to be solved. |
| 14:11.34 | Chesther | You can build .mp3 support in to *. That may be the easiest way. |
| 14:11.36 | *** join/#asterisk coppice (n=chatzill@157.202.17.210.dyn.pacific.net.hk) |
| 14:12.18 | *** part/#asterisk icyValk77 (n=icyValk7@213.129.64.4) |
| 14:13.10 | garymc | yeah i wanna know how i build mp3 support into asterisk |
| 14:13.21 | garymc | cos these .wav files do my head in |
| 14:13.43 | garymc | i know i can add an mp3 track for music on hold but it converts it to wav |
| 14:14.55 | Chesther | Well, it'll have to translate it to whatever codec the phone is using anyway. |
| 14:15.15 | Chesther | It's a matter of balancing CPU and disk space. |
| 14:15.34 | *** join/#asterisk Belgarath (i=belgarat@banda.pl) |
| 14:15.36 | Chesther | If you've got plenty of CPU, store the moh files in the smallest format, and let * translate it on the fly each time. |
| 14:16.02 | Chesther | If you've got gobs of disk, make copies in every codec format you're likely to use, and * can just feed the right one to the channel. |
| 14:16.10 | Naikrovek | Chesther: asterisk can transcode the mp3s for you, as can sox |
| 14:16.18 | Chesther | Right. |
| 14:16.44 | Chesther | As long as you've got enough CPU, that's fine. |
| 14:17.01 | Polysics | i have more disk than cpu |
| 14:17.01 | Naikrovek | well asterisk can do the one-time conversion of filetypes, just like sox |
| 14:17.18 | Naikrovek | or you can do nothing and let it do real-time conversion |
| 14:17.30 | Polysics | what would be the best format with infinite space and limited cpu? |
| 14:17.33 | Naikrovek | but it sounds like that's not what you want to do |
| 14:17.41 | Naikrovek | Polysics: infinite space? G711u |
| 14:17.41 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:18.00 | Naikrovek | almost all phones speak G711 |
| 14:18.02 | Polysics | how do i convert existing wav files to that? |
| 14:18.06 | Naikrovek | i can't think of one that doesn't |
| 14:18.16 | *** join/#asterisk _trine (n=psybnc_a@trine1-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 14:18.17 | Naikrovek | what are the wav files? 44100, stereo? |
| 14:18.22 | Polysics | yes |
| 14:18.44 | Polysics | recorded with audacity on linux |
| 14:18.45 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 14:19.00 | *** part/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 14:19.10 | Polysics | i was thinking of ffmpeg, but i don't know the parameters |
| 14:19.41 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
| 14:19.50 | Belgarath | Polysics: what format you want them in ? |
| 14:19.52 | Naikrovek | Polysics: sox would probably be better, dunno if ffmpeg supports ulaw |
| 14:20.07 | troffasky | audacity can save directly in Ulaw |
| 14:20.08 | *** join/#asterisk greysd (n=oae2@ns.pallada.ru) |
| 14:20.11 | troffasky | and Alaw |
| 14:20.14 | *** part/#asterisk greysd (n=oae2@ns.pallada.ru) |
| 14:22.45 | Naikrovek | nice |
| 14:22.47 | Naikrovek | there you go |
| 14:22.51 | Naikrovek | if you have a lot of files though... |
| 14:22.57 | Naikrovek | maybe there's a sox commandline you can use |
| 14:23.53 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 14:24.34 | Naikrovek | µlaw |
| 14:24.40 | Naikrovek | hrm. |
| 14:26.30 | Polysics | troffasky, i can't seem to find the option there |
| 14:26.46 | kaldemar | scalex000: describe you system and tell what protocols you're using. no one can help you without any information. |
| 14:27.53 | Chainsaw | Cisco 7960G, SIP-over-TCP support, Y/N? |
| 14:28.25 | troffasky | Polysics, File > Export > OK > Other uncompressed files > Options > Encoding: |
| 14:29.35 | Naikrovek | Polysics: sox whatever.wav -r 8000 whatever.ul |
| 14:30.07 | Polysics | troffasky, Header: WAV Microsoft, Encoding: u-law is ok? |
| 14:30.17 | Polysics | Naikrovek, that easy? good :-) |
| 14:30.20 | *** join/#asterisk geneticx (n=chatzill@host-208-88-126-198.biznesshosting.net) |
| 14:30.34 | troffasky | if you've got many to convert, just do what Naikrovek says |
| 14:30.51 | Naikrovek | Polysics: not sure if that converts from stereo to mono, may need to add another option for that |
| 14:30.54 | Naikrovek | but that's the meat of it |
| 14:32.41 | Naikrovek | ah, -c 1 for 1 channel |
| 14:32.58 | troffasky | but does that mix them down, or just throw one away? |
| 14:33.05 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:33.05 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:33.38 | scalex000 | kaldemar: Asterisk to BCM, using VOip Trunk protocol SIP |
| 14:34.25 | kaldemar | scalex000: and asterisk is not getting any SIP responses to an invite? |
| 14:35.04 | kaldemar | scalex000: you might want to grab a SIP debug of a call and pastebin it. |
| 14:35.18 | Polysics | ok, i exported in ulaw using audacity |
| 14:35.27 | Polysics | now * says the audio file is not there |
| 14:35.44 | Polysics | no, sorry |
| 14:36.15 | Polysics | says it is not a wav file |
| 14:36.19 | Polysics | wrong extension? |
| 14:36.20 | Naikrovek | don't specify the filetype when you tell asterisk about the file. |
| 14:36.26 | Polysics | i didn't |
| 14:36.27 | scalex000 | kaldemar: http://pastebin.ca/1579175 |
| 14:36.35 | Naikrovek | if your file name is menu.ul, tell asterisk it's just called menu |
| 14:36.36 | Naikrovek | k |
| 14:36.48 | Polysics | Playback(custom/welcome-prompt) |
| 14:37.07 | Polysics | the file is named welcome-prompt.wav |
| 14:37.09 | Naikrovek | if you dump the raw stereo wav in there, does it complain |
| 14:37.57 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 14:39.19 | Polysics | would that be the WAV (Microsoft) 16-bit signed PCM format? |
| 14:39.48 | Naikrovek | probably yeah |
| 14:39.48 | kaldemar | scalex000: you asterisk seems to be getting responses from somewhere. |
| 14:39.48 | Naikrovek | i know it's 16-bi |
| 14:39.48 | Naikrovek | t |
| 14:40.06 | scalex000 | kaldemar: this is a interconection between 2 pbx |
| 14:40.31 | scalex000 | kaldemar: so, the another pbx not recognize asterisk I think so |
| 14:40.48 | Polysics | error is "not in mono 2" now |
| 14:41.55 | Naikrovek | <PROTECTED> |
| 14:42.00 | Naikrovek | wonder what that is |
| 14:42.28 | kaldemar | scalex000: show a whole call next. |
| 14:42.54 | Belgarath | moni is os version of .net framework |
| 14:42.57 | *** join/#asterisk jlnt (n=jlnt@adsl-99-57-151-117.dsl.rcsntx.sbcglobal.net) |
| 14:43.05 | Belgarath | mono* |
| 14:43.17 | Polysics | converted to mono, then error is "Unexpected frequency 44100" |
| 14:43.43 | Naikrovek | you gotta downsample |
| 14:43.47 | Naikrovek | to 8000Hz |
| 14:44.01 | Naikrovek | you gotta resample, not just change the sample rate |
| 14:44.13 | *** join/#asterisk fuxu2 (i=iconicfl@www.kevinlynn.com) |
| 14:45.04 | troffasky | or record them through a handset ;-) |
| 14:45.10 | Naikrovek | yes |
| 14:45.29 | scalex000 | kaldemar: http://pastebin.ca/1579201 |
| 14:45.36 | scalex000 | kaldemar: see it again |
| 14:46.30 | scalex000 | kaldemar: I use a monitor in BCM to see if asterisk dial in, asterisk dial but not ring the extension not ring |
| 14:47.09 | Chainsaw | Right, anyone using Cisco 7960G handsets on SIP firmware? |
| 14:47.33 | Naikrovek | yeah lots of people are in here. but not me. i shouldn't have even mentioned it really |
| 14:47.33 | kaldemar | scalex000: you're still not showing a whole call. |
| 14:47.51 | Polysics | i have resampled but it's still not correct, probably |
| 14:48.00 | Polysics | should resampling change the file size, btw? |
| 14:48.01 | *** join/#asterisk Subdolus (i=dexterit@creep.bur.st) |
| 14:48.13 | Naikrovek | Polysics: play it and make sure it sounds right |
| 14:48.16 | Naikrovek | at the right speed |
| 14:48.28 | Naikrovek | Polysics: audacity doesn't modify the original file, you'll have to export i think |
| 14:48.48 | Naikrovek | been a while since i've done that with audacity |
| 14:48.52 | scalex000 | kaldemar: this is the all debug, what can I do |
| 14:48.55 | troffasky | yeah, if you Save in Audacity, all you save is the .aup project file |
| 14:49.03 | Polysics | i'm using Export |
| 14:49.08 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:49.11 | Polysics | i'll try the Sox route |
| 14:49.16 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 14:49.20 | Naikrovek | well downsampling should change the filesize if you do it right |
| 14:49.58 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:50.00 | kaldemar | scalex000: well, then this is all the help i can give you. based on that, there's no issue between asterisk and bcm. the problem is elsewhere. |
| 14:50.01 | Naikrovek | you need to resample via a filter rather than just changing the sample rate. if you change it from 44100 down to 8000 without resampling, all it'll do is play at 1/5th speed |
| 14:50.47 | Polysics | i used Tracks > Resample |
| 14:50.53 | Naikrovek | but, Polysics, now that i think about it, converting from one raw wav to another raw wav won't use much cpu anyway, even in asterisk |
| 14:50.59 | Naikrovek | Polysics: okay that's cool |
| 14:51.55 | Polysics | then i am doing something wrong when i save |
| 14:52.08 | Zeeek | what a busy day |
| 14:52.35 | kaldemar | Naikrovek: lowering sample rate doesn't affect playback speed, it just cuts frequencies. |
| 14:53.12 | Naikrovek | well with audacity you can change the sample rate, after you record sound. so you can record at 44100 and playback at 8000, and all it does is slow the file down |
| 14:53.26 | *** join/#asterisk riksta (n=rick@92.63.131.41) |
| 14:53.38 | Naikrovek | but yes, if you resample properly all it does is change the sample rate and maintain playback speed |
| 14:54.28 | riksta | Hi we are using asterisk 1.6 and we are experiencing dropped SIP-SIP calls at exactly 15 minutes regularly. I have found two bugs https://issues.asterisk.org/view.php?id=15922 and https://issues.asterisk.org/view.php?id=15270 but they seem to be related to FAX... does anyone know about this problem? |
| 14:55.26 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:55.34 | *** join/#asterisk jcape (n=jcape@155.sub-75-207-61.myvzw.com) |
| 14:57.20 | Naikrovek | riksta: related to FAX how |
| 14:57.56 | riksta | second bug URL i posted says it's to do with a reinvite that happens when theres T.38 involved |
| 14:58.06 | riksta | as far as I can see Naikrovek |
| 14:58.44 | riksta | I am still working on getting a sip trace and media dump for a dropped call |
| 14:59.01 | SuPrSluG | Polysics: go to Edit -> Preferences -> Qualtity -> Default Sample Rate = 8000Hz , Default Sample Rate= 16bit |
| 14:59.03 | Naikrovek | we'll need that to see what's up |
| 14:59.28 | riksta | Naikrovek: yeah, sure...i am not sure if it is our trunk provider's issue or asterisk |
| 14:59.35 | SuPrSluG | Polysics: then export as wav |
| 15:00.10 | Naikrovek | riksta: does it happen when a fax is being sent to you? i suspect the 15 minute thing is a remote fax machine or something autodialing you |
| 15:00.15 | Zeeek | In an hour, we start the live VoIP Users Conference with the authors of Asterisk 1.4 Professionals Guide. YOu can join us and ask questions or maybe win the free ebook. All the details to call in are http://VUC.me - go to #voip-users-conference IRC any time. |
| 15:00.51 | riksta | Naikrovek: no - we do not even do any fax handling, these are two standard alaw sip channels which are bridged |
| 15:00.57 | Naikrovek | riksta: ah |
| 15:01.00 | Naikrovek | nevermind me then |
| 15:01.08 | riksta | Naikrovek: also i have canreinvite=no and the udptl=no |
| 15:01.13 | Naikrovek | get a sip debug of when things fail and we can probably help |
| 15:01.19 | Naikrovek | where is [tk]d-fender anyway |
| 15:01.20 | riksta | Naikrovek: working on it, cheers |
| 15:02.24 | Naikrovek | i'm so hungry my water is starting to taste like ice cream |
| 15:02.32 | *** join/#asterisk tgunr (n=tgunr@cust-66-249-166-11.static.o1.com) |
| 15:03.14 | troffasky | Naikrovek has finally lost it |
| 15:03.14 | Polysics | SuPrSluG, do i need to re-record my audio? or will that export correctly? |
| 15:03.22 | Naikrovek | perhaps :) |
| 15:03.34 | Naikrovek | insanity is pretty strong as i understand it |
| 15:04.03 | SuPrSluG | Polysics: no i'll save the file with those attributes. which should work with * |
| 15:06.32 | Naikrovek | or use sox |
| 15:06.36 | Naikrovek | or use a handset :) |
| 15:06.45 | Naikrovek | i need to rerecord my menus .. that reminds me |
| 15:06.56 | Naikrovek | festival sounds like dookie compared to a real human |
| 15:07.34 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 15:08.06 | Polysics | Naikrovek, but waht about music? how do i record that with a handset? |
| 15:08.18 | troffasky | get the band in the room and put it on speaker phone |
| 15:08.21 | troffasky | easy |
| 15:08.51 | psilikon | Is there a way to use Zap channels that are on one asterisk server from another asterisk server. Perhaps thru and iax trunk or something? |
| 15:09.46 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 15:10.15 | superbeef | psilikon: yep IAX trunk |
| 15:10.16 | Chesther | psilikon: yes, going though an IP trunk would be the way to do that. |
| 15:15.15 | psilikon | Good, now that I know there is a way to do it I am off to google |
| 15:15.40 | superbeef | psilikon: its not too tough, you build the trunk, then add it to yoru dialplan |
| 15:16.09 | Polysics | yay! it works! |
| 15:16.13 | psilikon | superbeef, do you have to define the trunk in iax.conf on both machines? |
| 15:16.15 | Polysics | thanks to all |
| 15:17.16 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 15:17.22 | Chainsaw | Has "sip show registry" changed from 1.2 to 1.6? |
| 15:17.23 | psilikon | Actually I realized that there are a couple of pages on it in the Asterisk Future of Telephony book |
| 15:17.36 | Chainsaw | In 1.2, it showed devices registering with Asterisk *and* Asterisk registering with other devices. |
| 15:17.49 | Chainsaw | Now, in 1.6, it only shows the latter, not the former. |
| 15:17.59 | Chainsaw | Can I change that behaviour, as there is a custom web interface that depends on this? |
| 15:18.21 | *** join/#asterisk tris (n=tristan@camel.ethereal.net) |
| 15:18.48 | troffasky | psilikon, more here too: http://www.voip-info.org/wiki-IAX |
| 15:19.06 | psilikon | troffasky, thanks |
| 15:20.30 | *** join/#asterisk intralanman (n=lanman@67.76.163.226) |
| 15:20.40 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:20.40 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:21.33 | *** join/#asterisk dajhorn (n=dajhorn@206.16.96.160) |
| 15:21.49 | superbeef | psilikon: yes..... are using a gui or anything for asteirsk or just edintg configs by hand |
| 15:21.52 | psilikon | by hand |
| 15:24.17 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 15:24.25 | rue_mohr | IIII'm back! |
| 15:25.04 | rue_mohr | see if I can do this upgrade prove it works, submit the ticket and have them get back to me before they release another version |
| 15:26.49 | Gumug | in a federated Multi-layered peering system, using DUNDI, can i transfer calls via a HUD like its possible using a centralized system? |
| 15:27.57 | psilikon | I got this message: Unable to support trunking on peer '4Agent1' without zaptel timing. I think I need to get ztdummy running right? |
| 15:33.11 | Chainsaw | psilikon: Yes. Or DAHDI pseudo timing, if you're on Asterisk 1.4 or 1.6 |
| 15:33.20 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 15:33.52 | xrmx__ | i've installed latest svn asterisk-gui, make checkconfig reports everything is fine but every page request is a 404, any hint? |
| 15:34.03 | psilikon | yeah i am using asterisk 1.4.26. I was unable to load zaptel and ztdummy and I do not have a zaptel.conf in /etc/asterisk/. So has it been replaced by DAHDI? |
| 15:34.55 | Chainsaw | psilikon: Correct. DAHDI is what you want. |
| 15:35.04 | Chainsaw | psilikon: Zaptel as a name (and as a technology) is being phased out. |
| 15:35.05 | SuPrSluG | ~asterisk-gui |
| 15:35.16 | infobot | [~asterisk-gui] The Asterisk-GUI is an open source project lead by Digium. It's client-driven, and its only dependency is Asterisk itself. Check out the new version from svn here: $ svn co http://svn.digium.com/svn/asterisk-gui/branches/2.0 asterisk-gui-2.0. For support go to #asterisk-gui |
| 15:35.28 | Chainsaw | psilikon: It offers the same hrtimer-based dummy timer, so you will not have to install a PCI/PCIe telephony adapter if you don't want to. |
| 15:36.04 | Chainsaw | psilikon: It is worth making sure that your kernel has hrtimers support and that there is a high-resolution timing source (such as HPET) available on the underlying hardware. |
| 15:36.12 | psilikon | Now... how do I install it ;) |
| 15:36.20 | tzafrir_laptop | xrmx__, which page? |
| 15:36.38 | Chainsaw | psilikon: How did you install Asterisk? From sourcE? |
| 15:36.39 | psilikon | Should I be able to grep HPET out my .config? |
| 15:36.52 | Chainsaw | psilikon: On most distributions you can grep -i hpet /var/log/dmesg |
| 15:36.54 | psilikon | Nah through a ubuntu package |
| 15:37.03 | xrmx__ | tzafrir_laptop, http://foo:8088/asterisk/static/config/index.html |
| 15:37.13 | Chainsaw | psilikon: Then there's likely a Ubuntu package for DAHDI as well. |
| 15:37.35 | SuPrSluG | apt-cache search dahdi |
| 15:37.47 | *** join/#asterisk LiNeTuX (n=LiNeTuX@rrcs-71-43-123-202.se.biz.rr.com) |
| 15:38.04 | Chainsaw | psilikon: If not, they're behind, and you'll need to alert them of that through their BioPod. HeliPad. Whatever it is. |
| 15:39.12 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
| 15:40.04 | tzafrir_laptop | xrmx__, http show status |
| 15:40.04 | psilikon | yeah I already did a apt-cache search for the dahdi stuff... no luck. Maybe I need to add an Asterisk/telephony repo like I did with suse |
| 15:40.26 | troffasky | rt |
| 15:40.29 | psilikon | Looks like HPET is good to go |
| 15:40.31 | Polysics | i'd suggest compiling * and dahdi, always worked for me, while debs never did |
| 15:40.33 | tzafrir_laptop | psilikon, which version of Ubuntu do you use? |
| 15:41.06 | psilikon | It is actually crunchbang which is like ubuntu 8.1 |
| 15:41.07 | tzafrir_laptop | I think DAHDI just got into what will become their 9.10 release |
| 15:41.22 | tzafrir_laptop | that one has zaptel |
| 15:41.53 | rue_mohr | going from dahdi 2.1.0.4 to 2.2.0.2 do I need new init scripts? |
| 15:41.56 | xrmx__ | tzafrir_laptop, Server Enabled and Bound to 0.0.0.0:8088, looks it's only the static stuff that does not work |
| 15:41.59 | psilikon | Yeah funny you mention compiling *. I was talking with a dude in #suse who told me I was wasting my time compiling * from source and anything else for that matter. So he swayed me and I just went with the binaries |
| 15:42.15 | psilikon | so will zaptel work? |
| 15:42.43 | tzafrir_laptop | psilikon, what do you need it for? |
| 15:42.59 | superbeef | psilikon: the suse guy was on glue..... the asterisk stack is pretty low maintenance as far as compiling is concerned |
| 15:43.06 | Polysics | another thing i have learned with * is that basically no 2 installs are alike, for reasons i can't fathom |
| 15:43.09 | psilikon | I am thinking I need a timing source for the iax trunk |
| 15:43.10 | rue_mohr | hmm I hate it when the office is down and cant get simple answers to critical questions |
| 15:43.15 | tzafrir_laptop | rue_mohr, I think there were minor changes only |
| 15:43.25 | rue_mohr | ok.. |
| 15:43.37 | Polysics | i mean, things like apache are as complex as asterisk, yet doing A results in B every time |
| 15:43.44 | Polysics | that is not the case with * |
| 15:43.57 | rue_mohr | make config overwrites theconfig files though, right? |
| 15:44.06 | rue_mohr | its not just init scripts |
| 15:44.07 | Polysics | doing A on setups that look identical results in B, C, D, nothing, potatoes, at random |
| 15:45.09 | florz | Polysics: you do know some C? |
| 15:45.42 | psilikon | modprobe zaptel is a no go |
| 15:46.01 | rue_mohr | dahdi |
| 15:46.23 | Polysics | florz, i actually do :-) |
| 15:46.26 | rue_mohr | zaptel is (c), they had to change it |
| 15:47.44 | psilikon | I must be missing something. I installed the ubuntu zaptel package yet modprobe can't find the zaptel nor the ztdummy module. |
| 15:47.47 | rue_mohr | what kinda card you using? |
| 15:48.20 | psilikon | No card. I only need zaptel for ztdummy which is needed for the iax trunk |
| 15:48.44 | psilikon | Oh... face palm. |
| 15:48.55 | *** join/#asterisk Pazzo (n=ugelt@195.254.225.136) |
| 15:48.59 | psilikon | I need to build the driver. That is why there is the zaptel-source in the repo |
| 15:49.11 | *** join/#asterisk Skrusty (n=root@83.166.170.138) |
| 15:50.03 | Skrusty | does anyone know why, when using getoption (through AGI) the timeout seems to be ignored? In the console i can see it has a timeout, but as soon as streaming ends, it returns timeout. |
| 15:50.16 | troffasky | psilikon, shurely its in the packages? |
| 15:51.26 | florz | Polysics: Well, then the answer isn't that far ;-) |
| 15:51.51 | Polysics | florz, i don't follow you :-) |
| 15:52.08 | florz | Polysics: well, the explanation, rather |
| 15:52.12 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 15:52.58 | Polysics | the explanation of why every * install behaves differently? |
| 15:53.07 | florz | that's what I mean |
| 15:53.07 | Polysics | i don't think it's C's fault :-) |
| 15:53.23 | florz | no, but knowing C could help you understanding it |
| 15:54.03 | Polysics | i use about 5 languages daily and am proficient in a couple more, yet some things still escape me |
| 15:54.14 | Polysics | not saying * isn't a great piece of software, mind you |
| 15:54.23 | Polysics | just that it is pretty erratic |
| 15:54.42 | florz | which is pretty bad for software, isn't it? =:-) |
| 15:56.56 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 16:00.27 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 16:02.57 | rue_mohr | is there a way to get the tdm800P card to recognize a MWI signal properly and not think its the line ringing/ |
| 16:02.58 | rue_mohr | ? |
| 16:03.33 | psilikon | This might be real simple but how could set it up so if a sip extension rings more than 5 times another sip extension is dialed? |
| 16:04.02 | rue_mohr | have it fail after 5, and then it'll go to the next line |
| 16:04.35 | kaldemar | psilikon: there's no ring times, but you can use a timeout for app Dial. |
| 16:05.00 | *** part/#asterisk superciuc (n=alexandr@ip62.trivenet.it) |
| 16:05.17 | psilikon | kaldemar, could you pastebin something? |
| 16:05.28 | rue_mohr | its odd, on all the phone systems I'v worked on, its always been how long to wait, and not how many rings |
| 16:05.37 | rue_mohr | everyone always wants it set by rings tho.. |
| 16:05.47 | [TK]D-Fender | psilikon: Dial(SIP/100,15) <- dial for 15 seconds |
| 16:06.17 | [TK]D-Fender | psilikon: So just dial the firt then dial the second |
| 16:06.20 | [TK]D-Fender | first |
| 16:06.25 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:06.30 | psilikon | So then after that dial another. Nice. |
| 16:06.54 | psilikon | Sorry for the spoon feed there I am just new with this stuff and was making it harder then it needed to be. |
| 16:07.25 | [TK]D-Fender | psilikon: Yup. Dialplan just processes 1 step at a time |
| 16:07.57 | [TK]D-Fender | psilikon: Do something. maybe check some status. Do comething else, etc |
| 16:08.11 | psilikon | I need a Dial(SIP/XXX,15) followed by a Hangup then another Dial? |
| 16:09.01 | kaldemar | psilikon: no hangup |
| 16:09.45 | kaldemar | hangup would hang up the caller's channel. you don't have to end the first Dial by any means, it does it by itself. |
| 16:11.20 | [TK]D-Fender | psilikon: If the first dial is answered then at the end of the call the dialplan will halt |
| 16:11.48 | psilikon | Gotcha |
| 16:12.09 | [TK]D-Fender | psilikon: if the call does not get answered for any reason, DIALSTATUS will be set (which we don't care about here), and execution will continue to the next priority |
| 16:12.27 | p3nguin | You could even make it ring both lines at the same time. Whoever answers the phone gets the call. |
| 16:15.32 | p3nguin | rue_mohr: You just have to do the math when they want rings instead of time. |
| 16:15.37 | troffasky | or if you don't like the caller, you could make it ring nowhere :-) |
| 16:15.59 | Gumug | alright, i think i'm just going to do a central location in the beginning |
| 16:16.12 | p3nguin | I send people that I don't like into the talking clock context. |
| 16:16.35 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 16:17.12 | p3nguin | Answer(), SayUnixTime(,,ABdY \'digits/at\' IMp), then Hangup(). I'm sure they LOVE it! |
| 16:17.15 | wonderworld | i play them the normal sounds as if a phone was ringing. after a minute i play comgestion and hang up. |
| 16:19.08 | p3nguin | Congestion is fast busy or regular busy tone? |
| 16:19.24 | wonderworld | i think it depends on the country you are in |
| 16:19.41 | p3nguin | Which one are you in? I'm in the US. |
| 16:19.48 | wonderworld | germany |
| 16:19.54 | psilikon | So i've got ztdummy loaded but when I do a 'iax2 reload' I get the message : Unable to support trunking on peer '4Agent1' without zaptel timing. Maybe ztdummy wasn't the issue |
| 16:19.59 | kaldemar | usually it's fast busy. |
| 16:20.07 | p3nguin | I've never called Germany before. |
| 16:21.08 | p3nguin | So you just do Playtones(congestion)? |
| 16:21.28 | wonderworld | it's this one in germany: http://upload.wikimedia.org/wikipedia/commons/1/13/1TR110-1_Kap8.4_Teilnehmerbesetztton.ogg |
| 16:23.34 | [TK]D-Fender | psilikon: Odds are you didn't compile * AFTER Zaptel |
| 16:23.55 | [TK]D-Fender | psilikon: Zaptel needs to be in and configured first for support to be compiled into apps |
| 16:23.57 | psilikon | Should I need to? |
| 16:24.06 | [TK]D-Fender | psilikon: redo the * install process |
| 16:24.14 | vader-- | have any of oyu guys configured a polycom soundstation ip 7000? |
| 16:24.15 | [TK]D-Fender | psilikon: minus building samples |
| 16:25.13 | psilikon | Oh. So is there anyway to use PRI's connected to another asterisk box? I was hopping to send out bound calls to an asterisk server with a sangoma and 2 t1's |
| 16:25.29 | troffasky | with an IAX trunk |
| 16:25.33 | Naikrovek | vader--: not a 7000, but a 6000, yes |
| 16:25.35 | troffasky | didn't you already ask that? |
| 16:25.49 | psilikon | er hoping |
| 16:25.54 | [TK]D-Fender | psilikon: you can call whatever you want with your * sever... including another server and you can configure that one to also do whatever you want with the calls it was passed over |
| 16:26.08 | vader-- | naikrovek ive never worked with polycom, im really confused by the configuration files |
| 16:26.18 | Naikrovek | vader--: what do you want to know |
| 16:26.37 | Naikrovek | vader--: [TK]D-Fender and i are both versed in polycom |
| 16:26.56 | p3nguin | I tested it, it's a fast busy. (for anyone that was wondering) |
| 16:26.59 | psilikon | troffasky, to use an IAX trunk i need timing on both ends. I just learned that since I compiled zaptel an ztdummy after asterisk was loaded there is no support for ztdummy. So I can't use an iax trunk |
| 16:27.03 | vader-- | well im having issues with atftpd serving up the new sip image |
| 16:27.17 | vader-- | also im not sure how to setup the cfg files |
| 16:27.29 | vader-- | i have a sip.cfg, phone1.cfg, 000000000.cfg |
| 16:27.40 | psilikon | What about a sip trunk? |
| 16:27.48 | Naikrovek | vader--: you have just the default configs that are in the firmware download, right? |
| 16:27.55 | vader-- | ya |
| 16:27.56 | [TK]D-Fender | psilikon: as I said, just reinstall * |
| 16:28.05 | superbeef | psilikon: I'd trunk with IAX if you can |
| 16:28.20 | wonderworld | psilikon: you'll have less problems with IAX |
| 16:28.41 | *** part/#asterisk raspi (i=raspi@62.204.2.215) |
| 16:28.55 | Naikrovek | vader--: copy your 000000000000.cfg to a filename with the MAC address of the phone, .cfg. for example: 0004d21abcdef.cfg |
| 16:29.52 | Naikrovek | eh this'll take forever |
| 16:29.52 | Naikrovek | vader--: pm me your email |
| 16:29.52 | [TK]D-Fender | vader--: There is a rather comprehensive guide on the WIKI already. |
| 16:29.52 | vader-- | i tried to follow it |
| 16:30.33 | [TK]D-Fender | vader--: http://www.google.ca/#hl=en&q=polycom+phone+provisioning&meta=&fp=58658b2190507a24 |
| 16:31.12 | spck | easiest way i found was generating and tracking everything in a db or spreadsheet |
| 16:31.21 | vader-- | the 000000000.cfg file has alot of lines for other polycom models, do i need them i.e <APPLICATION_SPIP300 |
| 16:31.24 | spck | spreadsheet is obviously easier |
| 16:34.28 | *** join/#asterisk tris (i=tristan@207.241.238.17) |
| 16:41.01 | geneticx | asterisk is the shit! no matter what they tell me. |
| 16:42.31 | psilikon | [TK]D-Fender, Reinstall through my repos? If so do I need to bother backing up my conf files? |
| 16:43.37 | SuPrSluG | vader: no, it's not going to harm anything. Those lines are for the legacy models |
| 16:45.22 | *** join/#asterisk anwoke (n=A@75-145-57-202-utah.hfc.comcastbusiness.net) |
| 16:45.52 | kc8pxy | i need some trying to setup a small business asterisk server how i want. i think i know what i'm trying to do, but I'm not sure where the config lies. |
| 16:46.06 | [TK]D-Fender | psilikon: Compile again as before |
| 16:46.33 | psilikon | aww man. I don't wanna have to compile asterisk right now :) |
| 16:49.16 | *** part/#asterisk Hatrix (n=Hatrix@213.201.24.127.static.user.ono.com) |
| 16:52.53 | *** join/#asterisk Joel (n=jjshoe@wsip-70-183-82-162.sd.sd.cox.net) |
| 16:53.07 | Chainsaw | There used to a be a CLI dial command in 1.2 |
| 16:53.11 | Chainsaw | Can I emulate this in Asterisk 1.6? |
| 16:53.25 | Chainsaw | (It'll probably involve originate, but I don't want to make that call on a real phone) |
| 16:53.27 | Joel | Chainsaw, maybe it's called originate now? |
| 16:54.23 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 16:54.28 | Chainsaw | Joel: originate isn't quite the same though. Originate expects you specify a source channel and call a command. |
| 16:54.44 | Chainsaw | Joel: There doesn't need to be a source channel, I just want to call this one magic number. |
| 16:55.08 | Joel | Chainsaw, you have a number and no source? sounds very magical. |
| 16:55.20 | Joel | Chainsaw, can I have your unicorn? :) |
| 16:55.31 | Chainsaw | Joel: Yes. The magic number is *LI. |
| 16:55.45 | Joel | chainsaw what context? |
| 16:55.49 | Chainsaw | Joel: It's called with an agent number and a SIP station ID. So something along the lines of LI123#12345 |
| 16:56.02 | Chainsaw | Joel: It triggers a login from a web management interface. |
| 16:56.06 | Joel | chainsaw what context? |
| 16:56.17 | Joel | you can always try Local/number@internal |
| 16:56.28 | Joel | internal being the context for internal calls, for ex. |
| 16:56.30 | Chainsaw | Joel: [local], yes. |
| 16:57.58 | Chainsaw | Joel: Hmm. Okay. That still rings on my phone though. |
| 16:58.41 | Joel | so don't tell it to? |
| 16:58.59 | kc8pxy | i have 2 POTS plugs on my ata/broadband modem. client wants line1 to roll over to line2 when the did on the line1 is in use. where do i need to get that setup? at the voip/internet provider side, on the ata, or on my asterisk server, currently planned to connect to the ata on line1 with an fxs card. |
| 16:59.14 | Joel | kc8pxy, provider |
| 16:59.24 | Chainsaw | Joel: originate Local/0 application Dial Local/*LI123#1234 |
| 17:01.03 | [TK]D-Fender | kc8pxy: You should be using * to take in calls from your provider. Your conversion for D>A, then A>D will lose you functionality, audio quality, reliability, and cost you money |
| 17:01.10 | Chainsaw | Joel: More like originate Local/*LI123#12345@local application NoOp |
| 17:01.13 | Chainsaw | Joel: Thanks :) |
| 17:01.15 | Joel | Chainsaw, if you don't want this action to ring any phones then originate is not what you want. |
| 17:01.15 | Gumug | anyone done Amazon EC2 + asterisk? |
| 17:01.32 | Chainsaw | Joel: I do indeed not want it to ring any phones. The above command seems to work. |
| 17:01.43 | Chainsaw | Joel: (It's mostly to accomodate this awful web interface until I have time to write something better) |
| 17:01.44 | Joel | Chainsaw, waste of a call channel. |
| 17:01.59 | Joel | Chainsaw, your web interface should do whatever that dialplan does, directly. |
| 17:02.07 | Chainsaw | Joel: A properly written one would, yes. |
| 17:02.30 | Joel | Chainsaw, a proper admin would fix it, yes. |
| 17:02.42 | Joel | mess + more mess != better |
| 17:02.56 | anwoke | Joel, our provider is comcast |
| 17:02.58 | Chainsaw | Joel: There is such a thing as making it work before you replace it. |
| 17:03.11 | Joel | anwoke, ? |
| 17:03.29 | anwoke | kc8pxy and I are working on the same asterisk server |
| 17:03.33 | Joel | Chainsaw, I'm a believer in doing it right the first time, every time. |
| 17:03.38 | Joel | saves lots of valuable time |
| 17:04.39 | Chainsaw | Joel: Yes, one day you'll get to do this in the real world. And you'll understand what I mean. Thanks for your help. |
| 17:04.42 | fofware | Hello, How I can read mailbox settings for one extension? |
| 17:05.04 | Joel | Chainsaw, if only you knew what I did for a living :) |
| 17:05.12 | Joel | fofware, which settings? |
| 17:05.21 | fofware | sip.fonf |
| 17:05.25 | fofware | [2000] |
| 17:05.39 | fofware | mailbox=2000@something |
| 17:05.59 | Naikrovek | Gumug: nope, not used ec2 |
| 17:06.34 | fofware | Joel: I want redirect to diferent contex in mailbox but I don't find how read mailbox setting |
| 17:06.53 | Joel | fofware, I'm sorry, I just don't understand your question. |
| 17:07.05 | Naikrovek | [TK]D-Fender: polycom 3.2 firmware is out looks like |
| 17:08.42 | [TK]D-Fender | fofware: "core show function SIPPEER" |
| 17:08.48 | levy | Hello Channel, Im looking for large call volume call recording, is Oreka GPL the recommended solution for this? |
| 17:09.01 | fofware | Joel: exten => NOANSWER,1,Voicemail(${MACRO_EXTEN}@${MACRO_CONTEXT},u) but this MACRO_CONTEXT = context of channel not from mailbox setting |
| 17:09.10 | Naikrovek | levy: how large is "large" |
| 17:09.28 | Joel | fofware, so you want to automatically get the voicemail context? |
| 17:09.29 | kc8pxy | [TK]D-Fender: then what i need to do is get the info for 2 lines, and have asterisk take over as the ATA, not the modem, yes? can i make the rollover work from there? |
| 17:09.38 | fofware | Joel: yes |
| 17:09.45 | levy | 10 concurent throughout 9-5 5 days a week for 3 years |
| 17:10.00 | [TK]D-Fender | kc8pxy: If you hveq multiple DID's, the rollover is at the TELCO, not you |
| 17:10.01 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 17:10.02 | levy | Large for me lol... good point |
| 17:10.11 | Naikrovek | levy: asterisk can record all on its own |
| 17:10.31 | levy | Naikrovek I have read issues with max file count on EXT3 |
| 17:11.21 | [TK]D-Fender | levy: 10 concurrent isn't "large" |
| 17:11.22 | Naikrovek | levy: so offload once per month or whatever |
| 17:11.30 | Joel | fofware, I'm not sure how you would do that from within the dialplan, someone else here might know though. |
| 17:11.33 | [TK]D-Fender | Naikrovek: I'd say daily to a DB would be more than easy |
| 17:11.34 | levy | Naikrovek can CDR read files off another loaction? |
| 17:11.43 | Naikrovek | [TK]D-Fender: yes |
| 17:11.53 | [TK]D-Fender | Joel: and its not like I didn't jsut HAND HIM the answer |
| 17:11.57 | fofware | Joel: If I can do that if the only way that I find to set emailbody for different languages |
| 17:12.22 | fofware | Joel: Ok, thankz anyway |
| 17:12.24 | levy | [TK]D-Fender Agree its not large, large for me |
| 17:12.41 | Naikrovek | levy: not sure what you're asking, a CDR is a record, not a program |
| 17:12.48 | Joel | levy use asterisk and rsync. |
| 17:13.10 | Naikrovek | you want to record the audio, yes, not just the CDR |
| 17:13.29 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 17:13.36 | kc8pxy | [TK]D-Fender: unless i upgraded my PSTN connecting to a bri/pri, right? i don't think we're gonna do that. looks like a call to the provider. |
| 17:14.23 | [TK]D-Fender | kc8pxy: You are already getting service delivered over VoIP. What are you looking to add fixed wireline costs and hard cards to the mix? |
| 17:15.24 | kc8pxy | [TK]D-Fender: exactly what i thought. but that's the only way i could take over controll of the rollover.. yes? |
| 17:16.09 | levy | Naikrovek sorry, I wish to record CDR records audio and have a nice web GUI for a customer to pull audio based on extention date time and the other sides phone number |
| 17:16.22 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 17:16.33 | [TK]D-Fender | kc8pxy: NO. I just told you... you can't make 1 line roll over to another on inbound. the TELCO HAS TO DO IT |
| 17:18.29 | Corydon76-dig | Whoa, tiger. I think that's what he just said |
| 17:18.49 | Naikrovek | levy: you want to record the Call Data Record (CDR) AND the audio, and you want to provide a UI for a customer to be able to query that and replay the call |
| 17:19.41 | levy | Naikrovek, yes. |
| 17:19.58 | Naikrovek | levy: i don't know about the UI, but you can put the CDR records into a database, along with the path to the recorded file in another table probably, then you can find a UI for that |
| 17:20.14 | Joel | kc8pxy, my origional answer to you is correct, tkd-fender is (poorly) trying to tell you that while you are calling your provider you should see if they offer straight sip service so you don't have to use the analog lines. |
| 17:22.03 | [TK]D-Fender | Joel: No, that was very clear and a comment I haven't havd to make again |
| 17:22.13 | kc8pxy | Joel: we already have it, don't we? it's simply that the modem/ata is connecting to it, yes? |
| 17:22.35 | [TK]D-Fender | Joel: I am making sure he;s clear that one DID won't roll over to another any other way than telling the telco to do it. |
| 17:23.09 | Joel | kc8pxy, correct, if you can cut out that ata, and do direct sip service, you'll get more bang for your buck. |
| 17:23.13 | p3nguin | Why not just get multiple channels and not worry about the one being busy? |
| 17:23.40 | *** join/#asterisk wtca (n=wtca@williamt.noc.sonic.net) |
| 17:23.47 | *** part/#asterisk wtca (n=wtca@williamt.noc.sonic.net) |
| 17:24.17 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 17:24.58 | Joel | p3nguin, smidge irrelevant, if you check his original question. |
| 17:25.08 | p3nguin | perhaps |
| 17:27.09 | *** join/#asterisk HeMan (n=jimmy@ssh.southpole.se) |
| 17:27.21 | Anth8708 | Afternoon guys. Question. I have asterisk 1.6.1.1 running on redhat 4 with a digium card tied to a nortel option 11c via PRI. We have intermittent issues where the PRI goes down, but asterisk and dahdi still see the link as up |
| 17:28.09 | Anth8708 | the only way to bring the circuit back online is to reboot the asterisk box. restarting dahdi will then show the pri span as down |
| 17:29.43 | Anth8708 | er. ..restarting dahdi without a reboot that is. intermittent means this may work for a week or more solid, but within 10 days we get the "lock ups" on the PRI. they may happen 3-4 times a day as well . .very intermittent. i'm thinking hardware issues, perhaps on the box itself and am thinking about changing it out. Any other suggestions to try first? |
| 17:30.39 | Joel | pri intense debug ? |
| 17:30.44 | wonderworld | Anth8708: did you talk with your telco already? maybe the problem is on their side. a PRI shouldn't just "go down" |
| 17:31.04 | Anth8708 | both of these boxes are under my control |
| 17:31.23 | *** join/#asterisk Greek-Boy (n=greek@41.188.154.137) |
| 17:31.55 | Chesther | What happens if you just yank the PRI cable and re-plug it? |
| 17:32.35 | Anth8708 | wonderworld: nothing significant on the nortel side, it just shows the pri as "down." forcing a download (hard reboot almost) to the PRI card on the nortel side doesn't work either, just a reboot of the linux box |
| 17:32.43 | Anth8708 | Chesther: no go. I've tried that as well. |
| 17:33.15 | Anth8708 | i'm trying to see what debugs I can enable on the asterisk side to catch this when it happens. |
| 17:33.37 | wonderworld | i'd first automate the process of rebooting the box, so you have the most minimal downtime possible (if this is for a company). |
| 17:34.10 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:34.36 | Anth8708 | wonderworld: that's what I'd like to do, at a minimum. But I can't tell when the circuit goes down until a user calls or we test it |
| 17:34.45 | Chesther | wonderworld: ack. That's a total kludge, and is just begging for the real problem to turn from intermittent failure to hard failure at the most inconvenient time. |
| 17:35.30 | Anth8708 | Chesther: right. we only have a few phones on asterisk now, but we're planning on moving wholesale in a couple of months |
| 17:35.42 | HeMan | If I have to run a soft phone on a Windows machine, which do you recomend? |
| 17:36.04 | wonderworld | you could auto call to one of your numbers. cell-phone for example. if the call fails, reboot the box. |
| 17:36.12 | Anth8708 | Chesther: and doing away with the nortel. If the problem is on the nortel, then we'll have to deal with it for at least another 60 days or so. |
| 17:36.19 | p3nguin | heman: I like zoiper. |
| 17:36.28 | wonderworld | you'll sure have to examine whats really going on and don't rely on that quick and dirty "fix" |
| 17:36.37 | Anth8708 | wonderworld: initiate the call from command line is what you're saying? |
| 17:36.59 | wonderworld | Anth8708 look into call-files |
| 17:36.59 | HeMan | is IAX-phones prefered? |
| 17:36.59 | *** join/#asterisk ber_ (i=brad@neu.cow.org) |
| 17:37.13 | ber_ | does anyone know how to read the Remote-Party-ID from a SIP header via AGI? |
| 17:37.14 | Anth8708 | wonderworld: rgr. I'll google right now. Thanks |
| 17:37.18 | ber_ | or any other method |
| 17:37.24 | p3nguin | originate |
| 17:37.31 | fofware | Is there some way to get in dialplan the info that giveme sip show peer 2000 |
| 17:37.43 | *** join/#asterisk samy^ (n=samy@cpe-76-166-215-193.socal.res.rr.com) |
| 17:38.12 | p3nguin | heman: I like SIP just fine. Although zoiper does SIP and IAX. |
| 17:38.31 | HeMan | ok, I'll try zoiper then |
| 17:38.32 | HeMan | thanks |
| 17:38.47 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 17:40.05 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 17:40.09 | Joel | x-lite for windows gets the job done, it's ugly, but it works |
| 17:40.13 | [TK]D-Fender | fofware: [13:08]<[TK]D-Fender>fofware: "core show function SIPPEER" |
| 17:40.23 | [TK]D-Fender | fofware: I gave you this answer over half an hour ago |
| 17:40.35 | samy^ | joel's hot body gets the job done |
| 17:40.39 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 17:40.39 | fofware | sorry [TK]D-Fender |
| 17:40.50 | fofware | [TK]D-Fender: tanks |
| 17:41.06 | ber_ | another method is there a way to read the sip header and I can parse myself for remote-party-id |
| 17:41.16 | raden_work | Im trying to find a 10 ft VGA cable and all of them i find online are missing pin 7 ( green ground ) is this normal ? |
| 17:43.15 | ber_ | ahh there seems to be a SIPGetHeader function in 1.2 but not 1.4 |
| 17:43.17 | p3nguin | joel: I experience choppy audio (possibly due to resource consumption) using X-Lite, so I switched strictly to Zoiper. It always works. |
| 17:44.08 | ber_ | and SIP_HEADER function |
| 17:44.17 | levy | Normal being the norm, if all stores are missing then its normal |
| 17:45.25 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 17:47.38 | [TK]D-Fender | ber_: "core show function SIP_HEADER" |
| 17:48.21 | *** join/#asterisk mumtazah1 (n=mumtazah@202.98.48.60.wmu01-home.tm.net.my) |
| 17:49.57 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-69-213.link.net.pk) |
| 17:50.23 | *** part/#asterisk HeMan (n=jimmy@ssh.southpole.se) |
| 17:51.24 | ber_ | thanks TK |
| 17:52.44 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
| 17:57.34 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 17:59.31 | *** join/#asterisk errotan (n=errotan@62.201.122.236) |
| 18:01.33 | Joel | samy^, <3 <3 |
| 18:02.51 | Joel | samy^, next time you get community service shoot for the park and rec department, if you "volunteer" 400 hours you get free lifetime entrance into any park |
| 18:03.03 | *** part/#asterisk mumtazah1 (n=mumtazah@202.98.48.60.wmu01-home.tm.net.my) |
| 18:03.31 | Chainsaw | raden_work: Confirmed, this is expected: http://www.hardwarebook.info/VGA_(15) |
| 18:03.52 | Chainsaw | raden_work: Pin 9 is the 'key' pin. If implemented properly, the video card & monitor side do not have a hole for a pin 9. |
| 18:04.12 | samy^ | joel, hahaha nice! |
| 18:05.03 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
| 18:05.52 | Joel | samy^, take mental note how I said next time, I'm cheering you on, but I know it's just a matter of itme. |
| 18:06.22 | samy^ | Joel, time is on my side...wait..no, no it's not |
| 18:09.21 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 18:12.17 | raden_work | Chainsaw, thank you |
| 18:12.34 | Naikrovek | man wtf is up with my vsftpd... login with correct username & password, it pauses, then "login failed" grr |
| 18:13.03 | raden_work | Naikrovek, howdy |
| 18:13.03 | Naikrovek | and yes, local_enable=YES |
| 18:13.07 | Naikrovek | howdy raden_work |
| 18:13.37 | raden_work | your trying to login vsftpd via local lan ? |
| 18:13.42 | Naikrovek | yeah |
| 18:13.47 | *** join/#asterisk wr| (n=wr@p54BE3347.dip.t-dialin.net) |
| 18:14.03 | Naikrovek | worked a couple days ago, and I don't remember touching anything |
| 18:14.09 | raden_work | can u pasty your conf file |
| 18:14.15 | carrar | You've been OWNED! |
| 18:14.27 | Naikrovek | no not owned, i'm sure it's something i did. |
| 18:14.33 | raden_work | Naikrovek, you restart the service ? |
| 18:14.33 | carrar | heh |
| 18:14.39 | Naikrovek | raden_work: yeah |
| 18:14.46 | carrar | check your firewall settings |
| 18:14.46 | Naikrovek | hope i'm not pwned at least |
| 18:15.14 | Naikrovek | i can connect, i just get logiin denied even though i KNOW the username & pass are correct |
| 18:15.20 | raden_work | Naikrovek, still logged in somewhere else ? |
| 18:15.25 | Naikrovek | something to do with PAM I think |
| 18:15.40 | Naikrovek | i'm logged into the server via ssh |
| 18:15.40 | raden_work | Naikrovek, is this internal or external server |
| 18:15.44 | Naikrovek | internal |
| 18:15.49 | p3nguin | If Hangup() doesn't hang up, is there some other type of command to exit the dialplan? |
| 18:17.05 | raden_work | Naikrovek, ldd vsftpd |
| 18:17.27 | *** join/#asterisk Grof (n=dule@89.201.165.226) |
| 18:17.31 | Grof | hi guys |
| 18:17.35 | Grof | need help |
| 18:18.01 | Grof | "No translator path exists for channel type DAHDI (native 0x4c) to 0x3fff0001" |
| 18:24.29 | *** join/#asterisk jlnt (n=jlnt@adsl-99-57-151-117.dsl.rcsntx.sbcglobal.net) |
| 18:25.02 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
| 18:25.15 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 18:25.16 | *** part/#asterisk jcape (n=jcape@209.120.251.81) |
| 18:25.55 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
| 18:26.57 | Corydon76-dig | Grof: You need the transcoder card if you're seeking to do ANY transcoding of G.723.1 |
| 18:30.57 | Grof | i'm not trying to transcode |
| 18:31.05 | Grof | this is what i'm doing: |
| 18:31.13 | Grof | i call into PBX with IAX (alaw) |
| 18:31.38 | Grof | then i make a local call into PBX, and bridge those 2 calls with manager Bridge action |
| 18:31.53 | Grof | then, i try to call RetryDial on the local channel |
| 18:32.09 | Grof | so that i get IAX -> local -> local -> DAHDI |
| 18:32.30 | Grof | local -> dahdi fail because second local channels somehow defaults to g723 (!?) |
| 18:32.48 | Grof | and i have no idea why |
| 18:33.14 | Grof | 0x3fff0001 is a preffered format for second local channel |
| 18:34.16 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:34.30 | Grof | channel.c: Dropping incompatible voice frame on Local/2323@2323¸-6dd1;2 of format ulaw since our native |
| 18:34.30 | Grof | format has changed to 0x3fff0001 (g723|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140) |
| 18:39.20 | Anth8708 | wonderworld: Thanks again. I have a script up and monitoring call status with auto restart. It also gathers basic data for me to use to troubleshoot the REAL issue. Thanks again |
| 18:44.20 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 18:51.00 | [TK]D-Fender | Grof: I fail to see any debug or configs |
| 18:51.51 | vader-- | For some reason this Polycom soundstation IP7000 does not like my atftpd 0.7 server |
| 18:51.53 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 18:51.56 | vader-- | it won't grab anything from it |
| 18:52.11 | vader-- | i see the request on the tftp server and i see the request in my syslog on the phone |
| 18:52.22 | vader-- | i can use a tftp client and get the files |
| 18:52.22 | Naikrovek | did you try ftp? |
| 18:52.29 | vader-- | i don't have ftp setup |
| 18:52.32 | Naikrovek | k |
| 18:52.34 | vader-- | i use tftp for all my phones |
| 18:52.46 | vader-- | i setup a tftp server on my desktop and redirected the phone to that and it worked |
| 18:52.49 | vader-- | it's weird |
| 18:52.52 | Naikrovek | yeah i do too, well, i did, but only because i remember seeing this once or twice |
| 18:53.05 | Naikrovek | ah must be the server then |
| 18:53.06 | vader-- | My computer and Polycom Phone Worked |
| 18:53.15 | Naikrovek | it's probably trying to transfer in ascii mode or some BS like that |
| 18:53.23 | vader-- | The atftp server and my tftp client worked |
| 18:53.42 | vader-- | atftp server and polycom no work |
| 18:56.23 | aiksa[LV] | heelo everyone |
| 18:56.50 | aiksa[LV] | any idea how to transfer a call which has been pickuped? |
| 18:56.51 | *** join/#asterisk riksta (n=rick@5e00a756.bb.sky.com) |
| 18:57.00 | *** join/#asterisk Gumug (n=Gumug@nmd.sbx09566.joplimo.wayport.net) |
| 18:57.14 | [TK]D-Fender | aiksa[LV]: Press the transfer button on your phone |
| 18:57.22 | aiksa[LV] | as i understand t and T switches available for Dial command are lost when pickup happens right> |
| 18:57.24 | scalex000 | hello there? how asterisk send through sip trunk private |
| 18:57.35 | riksta | Hi i have posted a bug with asterisk 1.6.1 dropping SIP->SIP calls at EXACTLY 900 seconds and attached a pcap for both legs with full RTP media, I hope someone could help https://issues.asterisk.org/view.php?id=15966 |
| 18:57.35 | aiksa[LV] | [TK]D-Fender: :))) if that was so obvious ... |
| 18:57.37 | scalex000 | private URI |
| 18:58.16 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 18:58.28 | vader-- | any thoughts why the polycom would not like atftd server but works with another server? |
| 18:58.43 | aiksa[LV] | [TK]D-Fender: this is strange happens on SIP native attended transfer (from Snom) after a call has been pickuped from other recipient |
| 18:59.20 | aiksa[LV] | I googled up a few maillist posts regarding the same issue yet all of them end nowhere |
| 18:59.32 | aiksa[LV] | just a question asked and dead silence afterwards |
| 18:59.47 | Naikrovek | vader--: now that you mention (or rather, I notice) atftpd I remember a few people coming in here with problems with it |
| 19:00.21 | vader-- | atftp works fine with my cisco phones, which is just weird |
| 19:00.26 | Naikrovek | yeah |
| 19:00.37 | Naikrovek | works for most things |
| 19:00.37 | vader-- | it's driving me nuts |
| 19:00.45 | Naikrovek | works fine for my phones |
| 19:00.48 | vader-- | and polycom support won't talk to me unless im a certified polycom voip person |
| 19:01.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:01.03 | Naikrovek | really? |
| 19:01.10 | riksta | Naikrovek: hey, we talked earlier, i created a bug with the pcap dumps you asked for... #15966 |
| 19:01.17 | aiksa[LV] | [TK]D-Fender: if instead of standart res_features funcionality from features.conf (*8) i would rather ha a Pickup() application in dialplan and then access it indirectly through Dial to Local channel and append t option to DIal application, should that do the trick? |
| 19:01.20 | Naikrovek | that is really lame of polycom |
| 19:01.23 | Naikrovek | riksta: okay |
| 19:01.35 | Naikrovek | vader--: try to make sure that it always defaults to binary mode |
| 19:01.46 | riksta | Naikrovek: it may or may not be related to #15270 but i do not believe so. |
| 19:02.02 | vader-- | the phone or atftpd? |
| 19:02.08 | Naikrovek | atftpd |
| 19:02.42 | [TK]D-Fender | aiksa[LV]: WTF are you using DTMF transfers for anyway? |
| 19:03.10 | aiksa[LV] | [TK]D-Fender: not DTMF transfer |
| 19:03.15 | aiksa[LV] | pickup through DTMF |
| 19:03.35 | [TK]D-Fender | aiksa[LV]: you asked how to transfer. I said just hit the transfer button on your phone. |
| 19:03.48 | aiksa[LV] | I asked how to transfer Pickuped call :) |
| 19:04.08 | aiksa[LV] | well now i see the two meanings of "Pick up" ... sorry |
| 19:04.12 | [TK]D-Fender | aiksa[LV]: Be on a call. Hit "transfer" |
| 19:04.32 | aiksa[LV] | [TK]D-Fender: works perfect for a call which has not been pickuped |
| 19:04.37 | aiksa[LV] | has always workes |
| 19:04.40 | Naikrovek | pickuped |
| 19:04.53 | Naikrovek | if i pick up a call, i can transfer it |
| 19:04.55 | [TK]D-Fender | aiksa[LV]: No, the transfer button on my phone does not care where the call came from |
| 19:05.22 | aiksa[LV] | [TK]D-Fender: ok. this is strange then. |
| 19:05.34 | [TK]D-Fender | aiksa[LV]: What are you doing to transfer the call, and on what phone? |
| 19:05.48 | aiksa[LV] | snom 300 , latest firmware |
| 19:05.55 | [TK]D-Fender | aiksa[LV]: and HOW? |
| 19:05.58 | zamba | i have a problem with two different registrations to the same provider.. only one of them works for incoming calls at any given time |
| 19:06.45 | aiksa[LV] | ok, two lines activated on the first I have a pickuped call, I choose the second line, dial to the other party, make announcemnt regarding the caller and hit transfer twice |
| 19:06.56 | aiksa[LV] | has worked this way with snom for ages |
| 19:07.11 | [TK]D-Fender | aiksa[LV]: If your transfer fails, you're doing it wrong or your phone is flakey |
| 19:07.24 | [TK]D-Fender | aiksa[LV]: A call is a call is a call. |
| 19:07.38 | [TK]D-Fender | aiksa[LV]: If a SIP transfer fails, its the phone, or the user |
| 19:07.43 | aiksa[LV] | [TK]D-Fender: and tT options are only for DTM transfers? |
| 19:07.51 | [TK]D-Fender | aiksa[LV]: YES |
| 19:08.03 | aiksa[LV] | [TK]D-Fender: ok, then i can drop them alltogeather . |
| 19:08.10 | [TK]D-Fender | aiksa[LV]: YES |
| 19:08.43 | aiksa[LV] | [TK]D-Fender: 5 years mingling with asterisk and somehow has always thought that they are necesary for both :P |
| 19:09.39 | citywok | if i have a queue with no available members (logged in but autopaused), why does it leave the caller on hold and not kick them out with leaveonempty or something? How do i make it automatically send the caller to voicemail? |
| 19:10.07 | [TK]D-Fender | citywok: Show us the queue, and the call. |
| 19:10.23 | aiksa[LV] | citywok: wasnt paused memebers considered as active? |
| 19:10.24 | [TK]D-Fender | citywok: And your configs |
| 19:10.38 | bmoraca | does anyone have a good resource for a voicemail callback feature? |
| 19:10.49 | aiksa[LV] | [TK]D-Fender: I suppose its because queue with paused members is not considered empty |
| 19:10.53 | [TK]D-Fender | bmoraca: as in? |
| 19:10.55 | citywok | aiksa[LV]: yes i am pretty sure paused members are considered actgive |
| 19:11.26 | aiksa[LV] | citywok: thats the reason why the caller stays in the queue, it i not considrered empty |
| 19:11.35 | bmoraca | [TK]D-Fender: when a voicemail is left in a mailbox, asterisk calls the owner of that mailbox and logs them into their voicemail |
| 19:11.48 | citywok | http://pastebin.com/d1453f7b5 |
| 19:12.13 | TJNII | bmoraca: that seems .. counter intuitive. |
| 19:12.13 | *** join/#asterisk batphone (n=will@rrcs-24-153-211-180.sw.biz.rr.com) |
| 19:12.18 | batphone | what causes clicking sounds on voip calls? |
| 19:12.30 | citywok | is there an option i have not yet found which sets a maximum time a user can sit in a queue on hold before being kicked out and sent to voicemail? |
| 19:12.39 | bmoraca | TJNII: not if the person is not at their desk half the time and has a blackberry, which has no codec to listen to wav files |
| 19:12.51 | superbeef | Do the polycom 501 DHCP client's have the ability to pass a host paramter to the DHCP server (dyndns) |
| 19:12.55 | aiksa[LV] | citywok: yes |
| 19:12.57 | TJNII | bmoraca: Why not send the call to the blackberry? |
| 19:13.09 | [TK]D-Fender | batphone: packet loss or the audio starteds that way on one end |
| 19:13.10 | vader-- | WTF why won't this phone work with atftpd |
| 19:13.11 | aiksa[LV] | you can even then add another queue as a next step in dial plan |
| 19:13.12 | Naikrovek | superbeef: for dhcp reservations? |
| 19:13.12 | bmoraca | TJNII: because he may not want to actually talk to the person. |
| 19:13.30 | batphone | [TK]D-Fender: can you clarify? |
| 19:13.32 | citywok | aiksa[LV]: yep, i just kick it to voicemail for now and email the voicemail to everybody that is a member of that queue |
| 19:13.36 | [TK]D-Fender | bmoraca: there are scripting hook in voicemail.conf. that + Originate |
| 19:13.49 | [TK]D-Fender | batphone: what is there to clarify? |
| 19:13.50 | aiksa[LV] | city wok: on asterisk CLI> show application queue |
| 19:13.59 | aiksa[LV] | should do the trick |
| 19:13.59 | batphone | [TK]D-Fender: starteds? |
| 19:14.03 | superbeef | Naikrovek: so that my DHCP/DNS server assigns pretty hostnames in DNS |
| 19:14.08 | [TK]D-Fender | started* |
| 19:14.11 | [TK]D-Fender | batphone: ^^ |
| 19:14.14 | bmoraca | [TK]D-Fender: ahh...so i can configure a script to launch when a voicemail shows up in a box? i'll look into it |
| 19:14.42 | batphone | [TK]D-Fender: ok. what do you mean by the audio "starts that way" on the other end? |
| 19:14.45 | zamba | what's wrong here: http://pastebin.com/d1ceb695f ? |
| 19:14.48 | Naikrovek | superbeef: don't know, i assume so. polycom has documents on it i think |
| 19:14.51 | batphone | [TK]D-Fender: just a bad connection in general? |
| 19:15.05 | zamba | i have two registrations at the same provider, with two different accounts, but only one of them work for incoming calls |
| 19:15.21 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
| 19:15.23 | riksta | Naikrovek: i wish I could make more sense of the pcap's i linked to in the bug ... I am no expert at these things |
| 19:15.24 | [TK]D-Fender | batphone: Packet loss = bad connection. "Started that way" = they encode audio. what they are encoding FROM is flakey |
| 19:15.28 | superbeef | Naikrovek: I'm going through the docs now.... not looking too hopeful.. I'd love to see ext.blah.blah in my dns |
| 19:15.34 | Naikrovek | riksta: it's cool, they'll figure it out |
| 19:15.48 | aiksa[LV] | citywok: Queue(queuename[|options[|URL][|announceoverride][|timeout][|AGI]) |
| 19:15.52 | batphone | [TK]D-Fender: like a crappy phone using some codec that conflicts with cadences on a PRI? |
| 19:15.55 | Naikrovek | vader--: maybe try increasing the timeout on the server, if you've not done that |
| 19:16.13 | Naikrovek | vader--: on the phone as well |
| 19:16.15 | aiksa[LV] | [|timeout] is the param you are looking for |
| 19:16.19 | riksta | Naikrovek: sweet, i hope it contains enough info |
| 19:16.21 | [TK]D-Fender | batphone: or their PRI is noisy. or their hardware. Or "insert other act of God" |
| 19:16.33 | batphone | [TK]D-Fender: much appreciated! |
| 19:16.33 | citywok | aiksa[LV]: i'm using a timeout right now. http://pastebin.com/d1453f7b5 |
| 19:16.34 | Naikrovek | riksta: they may ask you for a sip debug of the moment the call fails |
| 19:16.45 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:16.48 | riksta | Naikrovek: that will be in the pcap |
| 19:16.49 | citywok | aiksa[LV]: i've been on hold with myself for 9 minutes now, and it has yet to kick me out of the queue |
| 19:16.55 | TJNII | zamba: Since you have the same provider for both * is probably only using one of the contexts for both peers. Try creating one common incoming context for sip.provider.com. |
| 19:16.57 | Naikrovek | riksta: that's true |
| 19:17.11 | aiksa[LV] | citywok: thats completely other timeout |
| 19:17.15 | TJNII | zamba: extensions.conf contexts, that is. |
| 19:17.31 | aiksa[LV] | that timeout considers how long should agents phone ring before giving up |
| 19:17.40 | citywok | okay thats what i thought it was |
| 19:17.50 | citywok | so the only way to define the queue timeout is to do it in the dialplan? |
| 19:17.58 | aiksa[LV] | timeout passed to Queue application from dialplan is a completely different story |
| 19:18.01 | citywok | as long as i know that, i'm totally okay with that |
| 19:18.15 | Naikrovek | vader--: i'm reading that atftpd has block size problems; everything i'm reading shows to use ftp :/ |
| 19:18.17 | zamba | TJNII: how do you mean? both in the same context? |
| 19:18.22 | zamba | TJNII: context= in sip.conf? |
| 19:18.57 | aiksa[LV] | atis_work: :)) fellow Latvian overe here :))) |
| 19:18.59 | aiksa[LV] | nice |
| 19:19.09 | TJNII | zamba: Try creating a common context in extensions.conf with both the 33333333 and 44444444 extensions and then set the context= for both peers to that context in sip.conf |
| 19:19.12 | aiksa[LV] | let the headhunting begin. |
| 19:19.31 | TJNII | zamba: I've run into this problem with Broadvoice, and this was my solution. It seems to work. |
| 19:20.15 | TJNII | [TK]D-Fender: (Since you like to jump in with corrections) if there is a better way, I'd love to hear it. |
| 19:20.16 | zamba | TJNII: ah, that worked |
| 19:20.24 | zamba | TJNII: thanks a bunch! |
| 19:20.28 | TJNII | zamba: np. |
| 19:20.37 | aiksa[LV] | citywok cascading queues can be used to do some nice spillover controlls |
| 19:20.37 | zamba | this is just for the incoming calls, right? |
| 19:20.42 | TJNII | zamba: Yes |
| 19:20.47 | aiksa[LV] | thats justfor future reference |
| 19:21.03 | citywok | yep, i've used second tier queues to run to receptionist before |
| 19:21.11 | aiksa[LV] | much more transparaent IMHO than agent penalties |
| 19:21.15 | citywok | it works pretty well when you want to do that |
| 19:21.28 | citywok | we dont run any agent penalties, everybody should be answering the calls equally |
| 19:22.02 | aiksa[LV] | citywok penalties can be used as a no-disturb-if-not-end-of-wold switch for some queue members |
| 19:22.13 | *** join/#asterisk staykov (n=staykov@pdpc/supporter/active/staykov) |
| 19:23.03 | aiksa[LV] | lets say your reeptionist would chime in if a call was not answered in 60 seconds, yet if she is not avaialble/busy...etc. you still have your agent pool waiting to take the call |
| 19:23.13 | citywok | yea, i can see when they would be useful. for agents answering an inbound line the way we've worked is just everybody gets to answer it, the shorter the hold time the happier the client |
| 19:23.37 | citywok | and we avoid sending calls to the receptionist for project inbounds because the receptionist doesnt' know what do do, lol |
| 19:23.49 | aiksa[LV] | :) |
| 19:23.59 | aiksa[LV] | send them to CEO in that case |
| 19:24.10 | citywok | hahaha, oh boy would he kill me for that |
| 19:24.24 | aiksa[LV] | any calls with a timeout setting of lets say minutes, he should call a lot nice things about the company. |
| 19:24.25 | psilikon | Call rejected by 10.11.12.2: No authority found <---- does this mean there is a username/password issue? |
| 19:24.33 | aiksa[LV] | 5 minutes |
| 19:24.40 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 19:25.08 | *** join/#asterisk voipmonk (n=voipmonk@67.204.45.155) |
| 19:25.48 | aiksa[LV] | respect my authoritaah ! |
| 19:26.35 | *** part/#asterisk staykov (n=staykov@pdpc/supporter/active/staykov) |
| 19:30.08 | aiksa[LV] | now on to tmrws. wind report, oh boy this is going to be good ! |
| 19:31.15 | aiksa[LV] | hell yeah. 8m/s straight for two days; this means my fam. wont be very happy :))) |
| 19:36.23 | psilikon | Ok now that I clear the username and password issue I keep getting an unable to negotiate codec error. |
| 19:36.49 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
| 19:36.57 | superbeef | psilikon: what codecs do you have installed on those boxes |
| 19:37.31 | timeshell | Has anyone an alternative or solution to mpg123's losing a stream and not trying to get it back again so that MOH ends up being blank air. |
| 19:37.36 | psilikon | i am not sure. How do I know. I have dissallow=all and allow=ulaw in both iax.conf files |
| 19:37.45 | timeshell | This would be on a internet radio stream |
| 19:37.53 | superbeef | psilikon: ubuntu boxes? |
| 19:38.06 | psilikon | One ubuntu one opensuse 11.0 |
| 19:38.28 | psilikon | is there an asterisk console comman to show what codecs like show codecs??? |
| 19:38.33 | citywok | aiksa[LV]: thanks, i figured there was another timeout setting somewhere that i was missing |
| 19:40.48 | superbeef | "chow codecs" |
| 19:40.55 | superbeef | psilikon: "show codecs" |
| 19:41.41 | psilikon | it has gsm, alaw, ulaw and all the usual suspects |
| 19:43.44 | aiksa[LV] | citywok, the one for application which I showedd you |
| 19:43.45 | Qwell | umm, yeah. see the first line of output? |
| 19:43.56 | Qwell | how it has nothing to do with your system or configuration? |
| 19:45.22 | psilikon | yep |
| 19:45.26 | Joel | umm, yeah. I'm going to need you to come in on saturday. |
| 19:45.57 | aiksa[LV] | bye everyone |
| 19:46.00 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:46.21 | aiksa[LV] | off for a wonderfull weekend :) |
| 19:46.38 | aiksa[LV] | gotta try that 9m furia finally :)) |
| 19:46.58 | TJNII | needs to rewrite his dialplan...... |
| 19:53.00 | Naikrovek | timeshell: use streamripper or something to prerecord the stream before you put it on as MOH |
| 19:53.19 | Naikrovek | timeshell: unless that's against the ToS for the stream |
| 19:54.11 | timeshell | How much of a delay would it create on playback? |
| 19:54.27 | [TK]D-Fender | timeshell: Use another streaming app that can recover, output as audio, and stream from audio-in locally |
| 19:54.27 | timeshell | Does it restart the stream automagically if there's a disconnect? |
| 19:55.19 | timeshell | I don't believe I have an audio-in on this server and I don't have any free PCI slots for a soundcard |
| 19:55.44 | [TK]D-Fender | timeshell: Well, it was a thought |
| 19:55.56 | timeshell | Thank you, I had actually considered it. |
| 19:56.08 | [TK]D-Fender | timeshell: I'm pretty sure there are several ways to fake this out |
| 19:56.23 | timeshell | I'm still looking for a way to disable buddies on the Polycom phones using SIP 3.1.3 btw |
| 19:56.37 | timeshell | That's why I'm asking ;) |
| 19:56.50 | timeshell | I'm not familiar with the ways to "fake this out" |
| 19:57.50 | [TK]D-Fender | timeshell: Me neither, but sound interfaces should be abstractable to a degree... depends on how cooperative the apps may be to devices you point them to. |
| 20:00.38 | *** join/#asterisk [8none1] (n=[8none1]@c-68-52-23-77.hsd1.tn.comcast.net) |
| 20:12.13 | p3nguin | chan_iax2.c:3253 __auto_congest: Auto-congesting call due to slow response <-- what action should I take about this? |
| 20:32.25 | *** join/#asterisk engrxyz (n=engrxyz@92.237.248.183) |
| 20:35.37 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
| 20:40.12 | *** join/#asterisk fofware (n=fofware@host188.190-136-191.telecom.net.ar) |
| 20:42.08 | fofware | any one can helpme I trying to send mail notification of new message in mailbox in many languages but I can't |
| 20:42.26 | fofware | any idea? |
| 20:50.30 | [TK]D-Fender | fofware: Notification how? |
| 20:51.25 | *** join/#asterisk andres833 (n=andres83@166.238.42.235) |
| 20:51.33 | *** join/#asterisk tamiel (n=tamiel@ip-7.net-81-220-254.rev.numericable.fr) |
| 20:52.43 | fofware | [TK]D-Fender: I trying to send one emailbody for each language, but look like I can not set it inside of contex |
| 20:53.12 | [TK]D-Fender | fofware: Then make your own e-mail script. |
| 20:53.41 | [TK]D-Fender | fofware: * just calls a standard sendmail shell script. make your own script instead and do whatever you want |
| 20:53.43 | fofware | [TK]D-Fender: the only emailbody that work Is that i define in general |
| 20:54.03 | [TK]D-Fender | fofware: MAKE. YOUR. OWN. SCRIPT. |
| 20:54.22 | fofware | [TK]D-Fender: ok, thank I will do |
| 20:54.57 | fofware | [TK]D-Fender: thanks |
| 20:58.27 | fofware | [TK]D-Fender: do you know one Howto or gide to do that? |
| 20:58.51 | [TK]D-Fender | fofware: #sendmail |
| 20:59.41 | fofware | [TK]D-Fender: ok, thanks |
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| 21:06.04 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 21:10.29 | Joel | fofare the easiest way is to have the voicemail system execute an external script for you |
| 21:12.36 | Joel | mailcmd would be one way |
| 21:13.01 | *** join/#asterisk davidandgoliath (n=David@216.198.139.38) |
| 21:13.13 | Joel | ic ould have sworn there was a more general excute after receiving an email |
| 21:13.20 | Joel | err voicemail. |
| 21:16.34 | p3nguin | chan_iax2.c:3253 __auto_congest: Auto-congesting call due to slow response <-- what action should I take about this? |
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| 21:46.00 | `paul | have you guys used the sangoma call analyser with vicidial... wahts the limit of the trial download? |
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| 22:08.48 | GGD | question i don't know if this has been asked before or not but can you install asterisk on a fedora box? |
| 22:08.56 | GGD | instead of using a prebuilt one? |
| 22:09.14 | giovani | sure ... |
| 22:09.18 | giovani | any distro is suitable |
| 22:09.29 | GGD | ok... |
| 22:09.32 | GGD | is therfe a guide? |
| 22:09.42 | [TK]D-Fender | GGD: in the source tarball |
| 22:09.42 | giovani | sure ... the asterisk documentation |
| 22:09.50 | nextime | GGD : why you need to install it from source instead of using the prepackaged? |
| 22:09.58 | GGD | ahh |
| 22:10.09 | nextime | GGD : if you ask how you can do it, i dubt you really need it |
| 22:10.36 | GGD | well i have a system already up and running and i would like to add it |
| 22:10.40 | GGD | if that makes sense |
| 22:10.55 | GGD | or would QOS take a crash to combine it/ |
| 22:11.23 | nextime | GGD : if you don't have particular needs, it is generally a good idea to use what your distro give to you |
| 22:11.38 | GGD | what do you mean nextime? |
| 22:12.37 | nextime | GGD : i mean that if you don't need to recompile asterisk with different build options and/or patch and/or use a different version, it is better if you will use the pre-packaged binary from your distro instead to use the source upstream version |
| 22:12.55 | GGD | understood |
| 22:13.07 | GGD | i have seen centos used but not fedora |
| 22:13.24 | GGD | rhel, suse |
| 22:13.25 | GGD | etc |
| 22:13.48 | nextime | ther's no difference |
| 22:13.55 | GGD | ok... |
| 22:14.00 | nextime | linux is linux, and gnu userland is gnu userland |
| 22:14.05 | GGD | understood |
| 22:14.11 | GGD | i thought there was.... |
| 22:15.06 | GGD | i could be wrong |
| 22:15.08 | GGD | thou |
| 22:15.14 | GGD | its happned before |
| 22:15.22 | nextime | GGD : the only differences between major distros are some specific distro maintenance packages, some scripts, the software in the default install, some configs, and also software versions |
| 22:15.38 | nextime | but basically they are all a linux kernel with a bunch of gnu and other softwares on it |
| 22:15.45 | GGD | ahh ok |
| 22:15.52 | GGD | does what i am asking make sense? |
| 22:17.46 | nextime | GGD : if you mean the question "can i install asterisk from source on fedora?" yes, make sense. You can. You can install from source in more or less every distro you want ( with some exceptions, but only on particular distros ) |
| 22:18.06 | nextime | but the real question is in my opinion "why you need to install it from source?" |
| 22:18.30 | GGD | i would like to combine with what i already have on the distro/ box |
| 22:18.38 | GGD | so its all in one |
| 22:18.45 | GGD | what i already have on the box |
| 22:20.15 | nextime | GGD : you mean that you want to install 2 different asterisk? |
| 22:20.22 | GGD | no |
| 22:20.30 | GGD | one |
| 22:20.39 | nextime | so, just install the package from your distro |
| 22:20.47 | GGD | i have an established fedoira box |
| 22:20.50 | GGD | fedora |
| 22:20.54 | GGD | ok... |
| 22:21.05 | nextime | for our luck a linux distro permit to install a lot of different things concurrently :) |
| 22:21.10 | raden_work | whats dahdi ? |
| 22:21.19 | GGD | yea |
| 22:21.21 | nextime | raden_work : the "new" version of the zaptel drivers |
| 22:21.29 | GGD | instead on one at a time? |
| 22:21.37 | raden_work | so if im using just SIP im fine ? |
| 22:21.44 | nextime | raden_work : yes |
| 22:21.55 | nextime | GGD : sure! |
| 22:22.14 | nextime | do you know any modern OS that permit to install just one things and nothing more? |
| 22:22.16 | GGD | LOL long live linux |
| 22:22.25 | GGD | oh yea |
| 22:22.28 | GGD | m$ |
| 22:22.53 | nextime | GGD : even on windows you can install different things concurrently.... the difference is that even if you install just one, it will not work good :P |
| 22:23.13 | GGD | exaCTLY |
| 22:23.17 | GGD | sorry for the caps |
| 22:24.37 | *** part/#asterisk `paul (n=kutimoy@asterisk.techvenueworld.com) |
| 22:27.04 | [TK]D-Fender | raden_work: depends |
| 22:27.24 | raden_work | howso ? |
| 22:28.11 | [TK]D-Fender | raden_work: without it : no MeetMe, no IAX2 trunk-mode |
| 22:28.56 | raden_work | I will keep that in mind |
| 22:29.11 | raden_work | It starts on boot then shutsdown i dont know what the deal isa |
| 22:36.15 | *** join/#asterisk Sandheaver (n=jeremiah@98.214.112.102) |
| 22:41.09 | raden_work | Sandheaver, you around ? |
| 22:41.19 | Sandheaver | yup |
| 22:45.33 | raden_work | how well does that virtual box work like running it on windows with like ubuntu or opensuse in it ? |
| 22:45.49 | raden_work | or running linux with windows in it on a dual core |
| 22:45.57 | raden_work | cause that way i could eliminate a computer at home |
| 22:47.11 | Sandheaver | should be native performance either way, really |
| 22:47.37 | Sandheaver | the big thing is disk, two (or more) operating systems have to share one disk, but virtual machines can be pinned to just one core if you like |
| 22:48.05 | Sandheaver | or if you have multiple physical disks that won't even be a problem |
| 22:48.31 | raden_work | I can put 2 drives in the machine would that help |
| 22:48.35 | Sandheaver | the 64-bit processors have virtualization extensions, which means that the virtual machines can just call the cpu directly. |
| 22:48.41 | raden_work | only reason i have windows at home is for gaming |
| 22:48.44 | Sandheaver | so there's no loss in translation |
| 22:49.00 | Sandheaver | well i'd make Windows the host then, and make linux the guest |
| 22:49.06 | raden_work | Sandheaver, so if i build like a core 2 duo 2.8 Ghz i wont have issues ? |
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| 22:49.21 | raden_work | I like linux for console utils |
| 22:49.21 | Sandheaver | 3D performance in a virtual machine can be difficult to get right |
| 22:49.30 | raden_work | thats what i was wondering |
| 22:49.46 | raden_work | does linux work well in a virtual machine ? |
| 22:49.55 | Sandheaver | if you're going to use linux mostly for console, definitely put it in a virtual machine |
| 22:50.01 | Sandheaver | yes it works perfectly |
| 22:50.18 | Sandheaver | the only thing you lose with a virtual machine is the ability to interface directly with hardware (other than the cpu or hard disk) |
| 22:50.25 | Sandheaver | or USB sometimes |
| 22:50.35 | Sandheaver | but you can't directly access a PCI card, for example |
| 22:50.47 | Sandheaver | (this includes graphics cards) |
| 22:51.04 | Sandheaver | so what happens is that the HOST machine emulates a video card for the guest machine(s) |
| 22:51.10 | raden_work | so id be fine |
| 22:51.13 | Sandheaver | yes |
| 22:51.36 | Sandheaver | just download virtualbox, download an .iso file of your favorite linux distro, and get to testin' |
| 22:52.13 | Sandheaver | oh yes, a core 2 duo 2.8ghz would be plenty good |
| 22:52.46 | Sandheaver | set the virtual network adapter type to bridged |
| 22:52.51 | Sandheaver | NAT is a PITA |
| 22:53.32 | raden_work | gotcha |
| 22:53.37 | raden_work | I have 2 nics |
| 22:53.45 | raden_work | can i have win use one and linux use the other ? |
| 22:53.50 | Sandheaver | you don't need two (you're not thinking virtually :)) |
| 22:54.00 | raden_work | i dont need , but i do have |
| 22:54.01 | Sandheaver | you can use one for both |
| 22:54.05 | Sandheaver | couldn't hurt i guess |
| 22:54.07 | raden_work | I get where your coming from |
| 22:54.43 | Sandheaver | just make sure the virtual network adapter is bridged, so it gets its own ip address on the network |
| 22:54.53 | Sandheaver | (so you can SSH into it) |
| 22:55.04 | Sandheaver | i gotta go |
| 22:55.07 | nextime | never used linux inside a windows host |
| 22:55.14 | nextime | just the opposite |
| 22:55.15 | nextime | :) |
| 22:55.45 | raden_work | nextime, i normally never would but i get sick of having 2 boxes at home |
| 22:56.25 | nextime | raden_work : i understand, but if i have just 1 box, i will use linux as host and windows in theh virtual |
| 22:56.35 | nextime | or better no windows at all :P |
| 22:57.11 | raden_work | nextime, i agree but linux dont run WOW , AOE3 , or unreal |
| 22:57.21 | raden_work | not that i play that often but i do play |
| 22:57.28 | nextime | raden_work: i have 5 boxes in front of me on my home desk, but the only windows in on the more powerfull one... inside a kvm virtual machine, booted just when i need to deploy something to be tested under windows |
| 22:57.29 | nextime | :P |
| 22:57.53 | nextime | raden_work ; i think you can run both under linux, anyway, i don't play so i don't know |
| 22:57.57 | raden_work | i normally run 2 boxes and rlogin to windows from linux |
| 22:58.04 | fofware | [TK]D-Fender: sorry, one more questions mailcmd=/usr/share/asterisk/sendmail.lua that is all to call this script? |
| 22:58.07 | raden_work | you can just PITA |
| 22:58.26 | raden_work | and my buddy got banned from wow from running it on linux voids there terms of service BS agreement |
| 22:58.47 | dustybin | rings himself for the 16th time today |
| 23:00.06 | fofware | how I can debug if the call to mailcmd=/usr/share/asterisk/sendmail.lua work? |
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| 23:45.21 | raden_work | wowie did raid 10 make a diffrence |
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