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00:02.53 | leifmadsen | DLNoah: odd -- I can't find any issues like that in the bug tracker right now |
00:03.12 | leifmadsen | doesn't mean the issue doesn't exist -- just that it hasn't been reported before :) |
00:03.22 | DLNoah | yeah, I hadn't seen anything, didn't know if it was a bug or a PEBKAC |
00:03.57 | leifmadsen | ya... could be a layer 8 issue, but not 100% sure of that :) |
00:04.14 | leifmadsen | DLNoah: oh -- ast_streamfile |
00:04.26 | leifmadsen | DLNoah: this may be related then: https://issues.asterisk.org/view.php?id=15224 |
00:06.20 | DLNoah | well, if I'm reading that report correctly, Playback() from within the dialplan is working for him. |
00:06.23 | DLNoah | it is not for me. |
00:06.53 | DLNoah | is there a config file somewhere for format_mp3.so that I'm missing? I've never really had to burrow into it b/c someone else set this server up and MP3 playback has "Just worked" since I took over maintenance |
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00:17.15 | vegbox | Hi, I have a NEC Electra phone system with a ghetto cd player attached to it that plays our music on hold. It has a regular stereo cable coming out of the NEC system and in to the cd player. Is it possible to replace this cd player with asterisk? |
00:19.18 | telnettech | anyboady know of any company in US looking for an Asterisk technician? |
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00:21.03 | leifmadsen | DLNoah: you're recompiling addons once you've upgraded asterisk right? |
00:21.08 | DLNoah | yes. |
00:21.18 | leifmadsen | DLNoah: and format_mp3.so is loading? (core show modules like mp3) |
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00:21.20 | DLNoah | asterisk refuses to load format_mp3.so if I don't |
00:21.24 | leifmadsen | right |
00:21.25 | DLNoah | and yes. |
00:21.29 | leifmadsen | hmmm |
00:21.50 | leifmadsen | I've never used it, so I'm not 100% sure |
00:22.39 | DLNoah | ok, thanks for the attempt |
00:23.03 | boch | how can an agent log out after AgentLogin() without hanging up the phone ? |
00:23.33 | DLNoah | I also kinda have a feature request / suggestion for Park(), though I suppose 8pm at night isn't the best time to catch whoever I'd need to talk to for that. |
00:23.33 | leifmadsen | boch: I don't think you can.... what are you actually trying to do? |
00:23.52 | leifmadsen | DLNoah: heh, not really -- asterisk-dev mailing list is probably your best bet. There are not really any devs in here. |
00:24.05 | DLNoah | ok, thanks. |
00:24.35 | boch | leifmadsen, i want my agents to be able to log in to a queue, or to dial manually the customers by themselves |
00:24.58 | leifmadsen | boch: then use AddQueueAgent() |
00:25.01 | leifmadsen | errr |
00:25.03 | leifmadsen | AddQueueMember() |
00:25.24 | leifmadsen | and RemoveQueuMember() to logout |
00:25.37 | boch | leifmadsen, already tried, but calls does not goes to agents |
00:25.44 | leifmadsen | then you didn't set it up right |
00:26.09 | leifmadsen | runs off to try and get this PXE server setup so he can load his new server since the CDROM appears to be dead |
00:26.12 | boch | leifmadsen, CLI reports agent added to queue, and i see the inbound call running Queue() app |
00:26.37 | boch | leifmadsen, but i cant hear "beep", neither the customer |
00:26.45 | leifmadsen | boch: status probably is set to unknown or something -- use a real channel, not an Agent channel |
00:27.02 | leifmadsen | hear beep? the queue() would call you back with AddQueueMember() |
00:27.07 | leifmadsen | it'd be more like a ring |
00:27.13 | boch | leifmadsen, im using an IAX channel, but maybe the status is the problem, ill check that, thank you |
00:27.17 | leifmadsen | and anyways, you just need a 2nd line on your phone if you wanna make a call |
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00:27.44 | leifmadsen | AgentLogin() is designed specifically to stay on-line while you're logged in. Hanging up is how you logoff |
00:31.21 | boch | leifmadsen, i can see the problem: IAX/1901 (dynamic) (Invalid) has taken no calls yet |
00:31.28 | leifmadsen | right |
00:31.38 | boch | do you know how to set the interface state in 1.4 ? |
00:31.51 | leifmadsen | boch: setup hints |
00:32.12 | boch | excuse me ? |
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00:33.44 | leifmadsen | boch: I don't use IAX (just SIP), but typically you need to enable hints in the dialplan for presence |
00:33.53 | leifmadsen | exten => 9001,hint,IAX2/9001 |
00:34.11 | leifmadsen | *CLI> core show hints |
00:34.22 | boch | i see, thanks again |
00:36.30 | KavanS | can someone give me an idea as to the term I'm looking for? |
00:36.59 | KavanS | I'm trying to setup a system where someone can sit down at a different extension/station each day and then when someone were to dial ex: 954, they would reach the same person each time |
00:37.13 | KavanS | would that be an "agent" ? |
00:37.20 | WindowsUser | followme |
00:37.25 | leifmadsen | KavanS: yes -- typically hot-desking |
00:37.35 | KavanS | ok, hot-desking...let me google a little |
00:37.38 | leifmadsen | don't call it an extension -- it is a device |
00:37.42 | KavanS | thanks for the wording |
00:38.01 | leifmadsen | extensions are numbers, assigned to users, who can dynamically be assigned to any device that they login to (hot-desking) |
00:38.16 | leifmadsen | they are all mutually exclusive of one another |
00:38.27 | KavanS | nice, that's pro as shit |
00:38.42 | leifmadsen | o.O |
00:38.44 | KavanS | ok, looks liek I have a short term asterisk project tonight |
00:38.50 | leifmadsen | short term? heh. |
00:39.01 | KavanS | cool...I figured there was a defacto solution to said problem |
00:39.07 | leifmadsen | took me 3 days to build a simple hot-desking platform :) |
00:39.20 | leifmadsen | and I've done it more than once |
00:39.35 | leifmadsen | although honestly most of that time was spent with stupid sangoma hardware |
00:39.57 | KavanS | ok...hrm, well found a post that looks as if it has the logic needed |
00:40.06 | KavanS | just need to make the additions to my asterisk box... |
00:40.07 | KavanS | http://www.757.org/~joat/wiki/index.php/Simple_hot-desking |
00:41.17 | leifmadsen | http://astbook.asteriskdocs.org has some hot-desking stuff too |
00:41.34 | KavanS | ok cool |
00:41.54 | DLNoah | leifmadsen: what about if I had a hot-desking situation, but also a shop where each person had a DID assigned to them... the customer wants the incoming callers to ring the designated agent if available, or to hear a message if the agent is unavailable (busy, etc) |
00:42.21 | DLNoah | and if the agent is unavailable, the caller would be presented with an option to go to VM, wait on hold (if the agent was merely busy), or talk to a different agent. |
00:42.37 | DLNoah | would that be just a simple hot-desk thing, or would some sort of queue setup be more appropriate? |
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00:43.57 | leifmadsen | DLNoah: that is just an IVR basically that rings the device the agent has "logged in" to |
00:44.06 | leifmadsen | queue would be unnecessary there |
00:44.11 | rockhard1981 | tzafrir_laptop: yes, any stream of sound to a channel like SIP/trunk-idid |
00:44.36 | DLNoah | ok. and basically a Background or Read for the "your agent isn't available" message, then? |
00:44.52 | WindowsUser | DLNoah: you could queue for if they choose a different agent |
00:45.44 | rockhard1981 | tzafrir_laptop: like a reverese mixmonitor, where you put audio to a channel, instead of grabbing it. |
00:51.18 | rockhard1981 | anybody knows if manger interface has a limitation to connections? |
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00:58.32 | WindMaker | :( |
00:58.37 | WindMaker | i have a problem :( |
00:58.55 | WindMaker | but my english it's bad :( |
00:59.54 | ManxPower | WindMaker: Try this if you need to: http://babelfish.yahoo.com/ |
01:00.06 | WindMaker | thks |
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01:19.38 | WindowsUser | MusicOnHold doesn't react to dtmf does it? |
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01:29.07 | leifmadsen | no |
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01:51.05 | Pan3D | WindowsUser: if you think about it, that's a good thing. Otherwise, people who are supposed to be on hold could accidentally/intentionally bounce themselves to another extension while on hold. |
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01:55.56 | WindMaker | why -> error 408 ? |
01:56.05 | WindMaker | ? |
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01:58.00 | VoipForces | Memory hole... In my brain... was the s,n(label) introduced in 1.2 or in 1.4 ? |
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02:01.01 | VoipForces | [TK]D-Fender: Memory hole... In my brain... was the s,n(label) introduced in 1.2 or in 1.4 ? |
02:01.35 | leifmadsen | 1.2 I think |
02:02.55 | VoipForces | leifmadsen: That is what i tought, was not sure. |
02:03.21 | leifmadsen | it's been years since I've used 1.2 :) |
02:03.26 | VoipForces | leifmadsen: Writing code that must run on 1.2 and I'm so used under 1.4 now |
02:03.34 | leifmadsen | but ya.... I still can't believe people use priority numbers |
02:04.07 | VoipForces | leifmadsen: Indeed, feels like doing Basic under DOS LOL |
02:04.57 | VoipForces | And for younger guys/gals, DOS meant Disk Operating System back then, not Denial Of Service |
02:05.11 | leifmadsen | heh |
02:05.17 | leifmadsen | DOS 5.0 ftw! |
02:05.29 | VoipForces | Ahh the memories of dos 3.3 on 1 5.25" floppy, using edlin and debug |
02:05.38 | leifmadsen | is tweaking his X00 fossile driver for Frontdoor |
02:06.09 | leifmadsen | goes to hang out with the g/f |
02:06.42 | VoipForces | Here everyone is asleep, gf, 3 kids, even the cat |
02:09.50 | eppigy | no need for other entities |
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02:18.07 | WindowsUser | Pan3D: yea, i'm just probing around Wake-up calls |
02:19.10 | nkohh | leifmadsen!! long time no see man! how are you? what've you been up to? |
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02:22.05 | CryWolf | Apologies for using a gui, but I'm trying to figure out why accountcode is not being set for outgoing calls. And as near as I can see, that is (or should be) a simple asterisk configuration, just one line in the appropriate sip peer definition. |
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02:23.22 | CryWolf | Well, I can set it for an individual extension, but I'm trying to set it for the trunk. |
02:27.16 | WindowsUser | so accountcode=xxxxx doesn't work? |
02:27.49 | CryWolf | Correct. It remains blank in the cdr. |
02:31.21 | VoipForces | Anyone knows if the syntax Goto(${last_question_context},s,1) is valid? Variable ${last_question_context} containing a valid context. |
02:35.35 | VoipForces | CryWolf: What gui? |
02:35.54 | CryWolf | VoipForces: freepbx |
02:36.21 | CryWolf | I've removed the 'accountcode=' from each of the extensions, so it's only defined for the trunks now. |
02:37.17 | VoipForces | CryWolf: I don't this it's valid to put it in the trunk definition. |
02:38.35 | CryWolf | It doesn't seem to be well documented, other than to say that it is used to populate the "accountcode" field of the CDR. |
02:39.23 | CryWolf | It works for the inbound, just not the outbound. I guess the inbound isn't a trunk, as such. |
02:39.25 | VoipForces | CryWolf: thing is that the variable neds to be set dialplan wise and if you put it in the trunk definition it will not be populated. |
02:39.40 | VoipForces | CryWolf: Exact. |
02:40.18 | CryWolf | VoipForces: Okay, I was beginning to suspect as such. So I just need to find the correct place to put Set(CDR(accountcode)=...) |
02:41.26 | VoipForces | CryWolf: something like that, but might be somewhat difficult if you do not want to break freepbx. |
02:42.06 | CryWolf | I've already modified it somewhat. What's one more mod? :) |
02:42.21 | VoipForces | CryWolf: True. |
02:42.32 | VoipForces | CryWolf: Just remember when you do an update :-) |
02:43.10 | CryWolf | VoipForces: Trust me, it's all documented. |
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03:01.05 | modex | Does anybody have any experience getting the Line1 button to work on the Grandstream GXP2000? |
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03:07.35 | Jumpie | hey guys |
03:07.43 | Jumpie | im having a particularly problematic 'gremlin' |
03:07.58 | Jumpie | i have a bunch of aastra 55i that i was deploying, went smooth |
03:08.09 | Jumpie | im using the dnd.php script on the phone |
03:08.52 | Jumpie | i have copied the same line of code, path, etc to each phone and they all work fine, changing of course each ?user=xxxx for each person..but just one particular person i get 'authentication error, you are not authorized to use this application" on the phone |
03:08.57 | Jumpie | NOTHING is different |
03:09.07 | Jumpie | shes registered, can do anything/everything everybody else can, the path is fine |
03:09.25 | Jumpie | if i take out the ?user= portion, then she can activate it..but can't deactivate it unless i reboot phone....any ideaS? drivin myself nuts here |
03:30.42 | CryWolf | Woohoo! Success at last! |
03:38.59 | WindowsUser | and odd characters in her name? |
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03:45.38 | xflavio | list |
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04:11.07 | taxilian | 'evening, all |
04:14.14 | taxilian | any experts here who can tell me if there is a way to create a script (possibly AGI?) that can be used for outgoing calls that will trigger an incoming call from another system and connect them? |
04:14.52 | taxilian | i.e., call a PHP script with an extension number that uses a 3rd-party interface to instruct my work phone system to call me and connect the call to the line that is already open |
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04:21.13 | taxilian | is there a different channel that would be better for this question? |
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04:21.36 | WindowsUser | ~idle |
04:21.37 | infobot | Idle is a doodie head |
04:21.49 | WindowsUser | aww i was hoping it'd have something useful |
04:22.45 | WindowsUser | taxilian: people will only respond if they have something to say thats not "i don't have a clue" :) |
04:23.04 | taxilian | =] yeah, that's why I was asking if there is somewhere better to go =] |
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04:23.12 | taxilian | seems like there ought to be some way of doing it |
04:23.16 | taxilian | but it's kinda a weird problem |
04:23.43 | taxilian | I seem to mess with a lot of those :-/ |
04:24.19 | WindowsUser | why are you triggering a call from remote? |
04:25.02 | taxilian | basically, I can use the remote phone system to make a call |
04:25.09 | taxilian | but I can only control it via HTTP |
04:25.20 | taxilian | so I send an HTTP command with the number to call and the number to call from |
04:25.31 | taxilian | and then it calls me and when I answer, calls the remote |
04:25.43 | taxilian | somewhat similar to google voice |
04:25.46 | WindowsUser | horray for call files |
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04:26.37 | WindowsUser | i wonder if you can silently enter a conference |
04:26.54 | taxilian | but since I have this nice asterisk system barely being used at my place, I thought it should be possible to make a script that asterisk runs when you call out |
04:27.09 | taxilian | yeah, I was wondering the same thing; then have the incoming call automatically routed to the same conference |
04:27.19 | taxilian | it'd be max one call at a time, but I don't really need more than that anyway =] |
04:28.41 | WindowsUser | well you'd be limited to starting a call one at a time |
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04:30.16 | taxilian | right |
04:30.35 | taxilian | I'm still relatively new to asterisk, so I haven't played with conferences much yet |
04:30.48 | WindowsUser | it might also be called meetme |
04:30.50 | taxilian | the other problem would be getting it to hang up on the incoming call when I disconnected from the conference |
04:31.37 | WindowsUser | kill the conference from the h extension if possible from dial plan? |
04:31.41 | taxilian | hmm. that is pulling up some useful search results |
04:31.44 | jaytee | you can turn off the announcement for when new people enter a conference but I'm pretty sure it affects all conference members. |
04:31.59 | taxilian | that would be fine; I don't need any announcements |
04:33.50 | WindowsUser | taxilian: 1.4 or 1.6? |
04:33.53 | taxilian | 1.4 |
04:34.03 | taxilian | running on freebs |
04:34.05 | taxilian | freebsd |
04:34.30 | taxilian | I'm using freepbx as an admin tool, but I'm roughly familiar with the config files as well |
04:34.43 | taxilian | still learning advanced dialplans (trying, anyway =]) |
04:36.00 | WindowsUser | in the h extension you could DeadAGI a script that uses the Asterisk Management Interface and kicks people out of the conference, probably very hacky, but the whole concept is ;) |
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04:37.14 | taxilian | lol. you have a point. this is probably a stupid question, but what is the h extension? |
04:37.24 | carrar | hangup |
04:37.33 | taxilian | ahh |
04:37.36 | carrar | ~book |
04:37.37 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
04:37.52 | taxilian | thank you. good resource |
04:38.08 | taxilian | I promise to read carefully before bringing too many more questions to the channel =] |
04:38.20 | taxilian | well, that gives me some ideas; I sure appreciate the tips, gusy |
04:38.28 | taxilian | even if I cant' tpye |
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04:52.31 | taxilian | 'nite all |
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04:52.34 | WindowsUser | huh asterisk not running future dated callfiles must be a new thing, I'm finding wakeup call scripts from 2004 that use cron |
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05:08.30 | NewtoPBX | so 1000 question time |
05:08.32 | NewtoPBX | haha |
05:08.35 | NewtoPBX | anyone up for it ? |
05:08.54 | [TK]D-Fender | NewtoPBX: 1000? No, try your luck and see how many you get |
05:09.04 | NewtoPBX | ok |
05:09.31 | NewtoPBX | so i have installed asterisk to my linux machine |
05:09.37 | NewtoPBX | now what do i do ? |
05:09.43 | NewtoPBX | i have no idea where to start |
05:09.53 | NewtoPBX | since there is like 30 configure files |
05:09.59 | NewtoPBX | i have a sip provider |
05:10.09 | [TK]D-Fender | ~book |
05:10.10 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
05:10.33 | NewtoPBX | that is the one i am reading |
05:10.34 | NewtoPBX | haha |
05:10.36 | WindowsUser | what provider? |
05:10.57 | NewtoPBX | iptel.org |
05:10.59 | [TK]D-Fender | NewtoPBX: Here is a SAMPLE to use as inspiration : |
05:11.00 | WindowsUser | some of them tell you what to put in sip.conf :) |
05:11.01 | [TK]D-Fender | ~jerjerguide |
05:11.02 | infobot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
05:11.14 | NewtoPBX | sweet |
05:11.15 | NewtoPBX | thanks |
05:12.15 | NewtoPBX | so here is a question to. can i just push this thing using tcp/udp using the nick on the pc or do i have to have the hard ware and phones ? or can i just call into it using the nic? |
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05:12.43 | WindowsUser | ~softphone |
05:12.44 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
05:14.11 | NewtoPBX | ok so i guess the next question is..i am doing this for a project to see if i can make a conferance call server where you call in and ther is options like press 1 to talk to bla bla bal press 2 for bla bla. can asterisk do this ? |
05:14.38 | [TK]D-Fender | NewtoPBX: Yes |
05:15.45 | NewtoPBX | sweet |
05:15.54 | NewtoPBX | then i will keep messing with it |
05:16.12 | NewtoPBX | infobot that is a good link |
05:16.13 | infobot | NewtoPBX: I think you lost me on that one |
05:16.14 | NewtoPBX | thanks |
05:16.43 | NewtoPBX | like.. say i want to talk to 10 people at once using asterisk |
05:17.29 | WindowsUser | the bot is a bot btw |
05:18.39 | NewtoPBX | haha |
05:18.58 | WindowsUser | NewtoPBX: they'd need headsets or handsets, 10 people all using the equiv to speakerphone would be really ugly under any circumstances |
05:19.29 | NewtoPBX | ok |
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05:48.37 | [TK]D-Fender | Checkout time, later all |
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06:18.07 | Stdht | Hi! Could someone please help me. Asterik 1.4 under ubuntu. I try to connect mobigater via chan_celliax. I built chan-celliax.so from celliax_stuf.tgz from celliax.org |
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06:20.59 | Stdht | I get segfault in app_userevent.so when incoming call |
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06:26.54 | NewtoPBX | sweet i got it working |
06:26.56 | NewtoPBX | thanks guys |
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06:36.28 | Stdht | How to ask for Help ?!! |
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06:38.46 | Stdht | Anyone hears me?? |
06:43.25 | Stdht | PLease HELP |
06:43.51 | tzafrir_laptop | infobot, tell Stdht about ask |
06:44.41 | tzafrir_laptop | Stdht, start by installing asterisk-dbg |
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06:45.00 | tzafrir_laptop | with this you'll be able to make your core files meaningful |
06:45.11 | tzafrir_laptop | (assuming you use asterisk from packages) |
06:45.42 | WindowsUser | infobot, tell WindowsUser about ask |
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06:46.12 | WindowsUser | i see, I'm used to doing those on channel |
06:48.33 | tzafrir_laptop | WindowsUser, you could simply use: /msg infobot whatever |
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06:48.54 | tzafrir_laptop | Or, alternativly, if you want this to show on the channel: |
06:48.57 | tzafrir_laptop | ~whatever |
06:48.58 | infobot | somebody said whatever was an expression of confusionambivalence, or something else |
06:49.37 | [T]ank | so what would cause a peer behind nat to register, but still show unavailable when I do a sip show peers? |
06:51.00 | [T]ank | I see what is going on... but not sure how to correct... |
06:51.17 | [T]ank | externhost... looks like |
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06:54.24 | Stdht | before I tried asterisk-dbg - I have asterisk1.4(from sources) under ubuntu8desktop. I have chan-celliax.so built from sources from celliax_stuff.tgz from celliax.org. I am testing asterisk with softphone on another PC. When I call extension which makes Dial(Celliax/mobi1:....) I receive "Unable to creare celliax channel cause-0 -Unknown" when I call from GSM to my celliax device (mobigater) - I get segfault in app_user_event.so |
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06:55.06 | Stdht | and asterisk crashes |
06:56.53 | Stdht | how to install asterik_dbg |
06:56.56 | tzafrir_laptop | asterisk-dbg is indeed only relevant if you use packaged debs |
06:56.58 | WindowsUser | tzafrir_laptop: I was just testing if it'd work |
06:57.22 | Stdht | i can run asterisk -cvvvvvdddddddddd |
06:57.36 | Stdht | but there no any more specific info about error |
06:58.02 | tzafrir_laptop | Stdht, can you get a core file? |
06:58.17 | Stdht | what is it- core file ? |
06:59.24 | Stdht | dump? |
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07:12.05 | Stdht | sorry I was disconnected |
07:12.20 | Stdht | which pid to use in /usr/sbin/asterisk pid |
07:13.51 | Stdht | or more generally hot to get core file |
07:20.18 | tzafrir_laptop | for starters, run asterisk with -g |
07:21.05 | Stdht | jr |
07:21.08 | Stdht | ok |
07:21.10 | Stdht | started |
07:21.36 | Stdht | and what to do now |
07:24.11 | Stdht | I tried asterisk -g ... it crashed as expected and what to do now? |
07:24.30 | yang | helloes tzafrir_laptop |
07:24.42 | tzafrir_laptop | hi |
07:25.03 | tzafrir_laptop | did it leave a file called "core" ? |
07:25.19 | Stdht | where? |
07:25.59 | Stdht | oh yeas |
07:26.06 | Stdht | it left |
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07:26.26 | Stdht | core.1387 |
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07:27.59 | tzafrir_laptop | next, get yourself a gdb (aptitude install gdb #or whatever) |
07:28.20 | Stdht | I have gdb installed |
07:28.32 | tzafrir_laptop | and run: gdb -c core.1387 /usr/sbin/asterisk |
07:28.46 | tzafrir_laptop | in the gdb prompt, get a backtrace: |
07:28.49 | tzafrir_laptop | bt |
07:29.03 | tzafrir_laptop | bt full #more details, more noise |
07:30.09 | Stdht | ok I typed |
07:30.25 | Stdht | dgb -c -core ... then bt full |
07:30.52 | tzafrir_laptop | if the output does not make sense to you at this point, pastebin it |
07:30.53 | tzafrir_laptop | ~pb |
07:30.54 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
07:32.06 | tzafrir_laptop | infobot, pb is also apt-get install pastebinit |
07:32.07 | infobot | tzafrir_laptop: okay |
07:32.11 | tzafrir_laptop | ~pb |
07:32.12 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
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07:36.53 | Stdht | something with souund system .... one mometn |
07:37.33 | Stdht | http://pastebin.com/m4d5aed1f |
07:40.24 | Stdht | dst_exten - is smothing that I can't understand |
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07:45.39 | tzafrir_laptop | Stdht, as you can see, the crash is somewhere inside chan_celliax.so . But there does not appear to be debug information for it |
07:45.57 | tzafrir_laptop | Did you get it as aseparate binary? or build it on your own? |
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07:46.48 | Stdht | I got it from http://www.celliax.org/celliax_stuff.tgz |
07:46.59 | Stdht | than I maked it |
07:47.23 | Stdht | I changed in make file just the path to asterisk 1.4 include |
07:50.17 | Stdht | what could I do next, |
07:50.18 | Stdht | ? |
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07:59.21 | Stdht | celliax.org is down with all of its forums.....:(( |
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08:27.51 | jerryeguru | recently i fitted an old degium pci card, not installed asterisk or anyother related software and lspci does not show me the card! |
08:28.16 | jerryeguru | could that mean, without related software the card is not visible or it is damaged |
08:28.34 | jerryeguru | I would like to get information about this interface card |
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08:56.18 | WindowsUser | jerryeguru: if its pci and its plugged in properly it should show up in lspci |
09:05.53 | WindowsUser | sleepytime |
09:06.16 | WindowsUser | all i can suggest is reseat it (while powered down!) |
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09:26.42 | jmls | morning all |
09:26.46 | jmls | using 1.4, is there anyway of lighting a BLF on cisco/aastra phones from the dialplan or AMI ? |
09:26.56 | jmls | except for using voicemail |
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09:31.27 | Zhad | Can you do it using SIP Notify? |
09:31.55 | jmls | I really don't know ;) |
09:32.13 | jmls | can you construct your own SIP notify message ? |
09:32.25 | jmls | or are you limited to the existing ones ? |
09:32.36 | Zhad | Probably, using Add-Header, I've never tried it. |
09:33.53 | Zhad | Sorry, SIPAddHeader |
09:34.51 | Zhad | although that only adds a header to the next call you make afaik. |
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09:41.10 | mattboll | hi |
09:42.18 | mattboll | I've got the "ringing but no voice" problem, but it happens only when I redirect an extern call to an other extern call |
09:42.47 | mattboll | I've got voip phones, i can call extern phones like a cellular |
09:43.03 | mattboll | I can call from a cellular to my voip phone |
09:43.33 | mattboll | when I receive a phone from a cellular, I can "transfer" it |
09:44.23 | mattboll | but when I received an external call with this property : |
09:44.24 | mattboll | exten => 0974534610-0952044141,3,Dial,SIP/0632882022@ovh-3612 |
09:44.41 | mattboll | then it rings but we don't hear each other |
09:44.52 | mattboll | (btw : sorry for my english) |
09:45.53 | mattboll | So, it seems to be a firewall problem but I don't know what I should search for because everything else work |
09:46.28 | mattboll | firewall/nat… |
09:49.48 | Zhad | have you tried turning can reinvite off? |
09:49.57 | Zhad | ie. making sure that you stay in the loop. |
09:52.09 | Zhad | I didn't know you could do exten => 0974534610-0952044141 |
09:54.18 | mattboll | I'll try the reinvite off |
09:54.29 | mattboll | the exten is called by Goto(filter,0974534610-${CallerIDString},1) |
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10:08.32 | mattboll | I had tTr so canreinvite was off, so I tried to turn it on but it didn't work |
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10:32.33 | Zhad | what's wrong with using exten => 0974534610/0952044141 ? |
10:32.49 | Zhad | unless 0952044141 is not the CallerID(num). |
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11:33.21 | zeeesh | sip to yahoo voice chat ... is there any posibility with asterisk ? |
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11:54.11 | mattboll | I didn't know the 0974534610/0952044141 syntax, but it works so I won't change now |
11:54.32 | mattboll | well it works unless I call an external number :/ |
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12:16.26 | ariel_ | morning |
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12:36.27 | mattboll | ariel_: it's 2pm here >_< |
12:37.14 | ariel_ | mattboll: sorry about that, but it's just 8:37 am and just starting.... |
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12:50.56 | Pan3D | exten => _1NXXNXXXXXX,1,Dial(goodmorning@you,300,r) |
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12:54.18 | mattboll | geek ! >_< |
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13:26.08 | Stdht | Hi. Does asterisk passes dtmf through by default... For instalce I call from sip softphone to asterisk and asterisk calls external line... how could I pass dtmf to external ivr |
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13:33.51 | minotaur01 | hello everyone |
13:34.57 | Aiatek | Stdht thats call disa |
13:36.42 | Stdht | yes Disa I've found that asterisk reproduces dtmf |
13:36.50 | Stdht | to other side |
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13:38.12 | minotaur01 | im having a problem with audio playback (IVR/Directory) over IAX and im wondering if anyone can help me |
13:39.33 | Aiatek | minotaur01 describe your problem and maybe someone here can help you |
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13:40.49 | minotaur01 | when i use a SIP connection the audio plays fine but over IAX it's choppy and it's as if it's in slow motion |
13:40.54 | Great_Anta_Baka | sup all!!! |
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13:41.49 | SuPrSluG | minotaur01:do you have jitter buffer enabled? |
13:41.56 | Great_Anta_Baka | please tell me if i am laughing for nothing.. this company has come into this business park which has 19 offices and is running an asterisk server in each of the 19 virtual machines!!! |
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13:43.04 | minotaur01 | SuPrSluG: it's a clean install let me check |
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13:52.03 | minotaur01 | SuPrSluG: i belive it's not on i checked the iax conf files there is no jitter buffer setting |
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13:54.34 | leifmadsen | minotaur01: see https://issues.asterisk.org/view.php?id=15337 |
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14:10.16 | Katty | peeks in |
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14:14.07 | Amorsen | Is there a way to detect whether a card is TE410P (3.3V) or TE405P (5V) without driving to the server room and opening the case? |
14:14.45 | *** join/#asterisk galeras (n=galeras@186.80.186.118) |
14:14.48 | *** join/#asterisk davevg (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
14:15.41 | galeras | Semi * question: what mail client do you suggest to implement a mail to fax solution, i f mail server is exchange? |
14:16.05 | mog | Amorsen, its probably easier to figure out what kind of server it is in |
14:16.12 | mog | as most servers have 3.5 or 5 v slots |
14:16.29 | Amorsen | mog: Dual voltage, unfortunately |
14:16.41 | mog | aww |
14:16.51 | Amorsen | Handle 0x0018, DMI type 9, 13 bytes |
14:16.56 | mog | there is no way to tell that i know of |
14:16.56 | Amorsen | <PROTECTED> |
14:16.56 | Amorsen | <PROTECTED> |
14:16.56 | Amorsen | <PROTECTED> |
14:17.06 | Amorsen | That's dmidecode |
14:17.35 | mog | Amorsen, digium might have started giving them different sub ids recently, but as long as i have known they look the same |
14:17.37 | Amorsen | There's no identifier anywhere on the card? |
14:17.40 | Amorsen | Ok |
14:18.29 | Amorsen | Hmm, maybe iLO knows. |
14:19.27 | Amorsen | Nope. |
14:20.58 | *** join/#asterisk Vec (n=Vector@host-87-74-7-50.dslgb.com) |
14:21.29 | Amorsen | The other card is easy: |
14:21.31 | Amorsen | 05:01.0 Communication controller: Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev 02) |
14:21.45 | Amorsen | So that probably means card 1 is 5V |
14:25.31 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
14:27.37 | Vec | I am trying this: but its not working, any ideas: |
14:27.38 | Vec | exten => s,n,ExecIf($[${LEN(ARG5)} = 0]|Set|TIME=4) |
14:27.49 | Vec | I would like to keep it to 1 line |
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14:29.43 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
14:30.08 | beek | Vec: more like: ExecIf($[${LEN(ARG5)}=0]?Set,TIME=4) |
14:30.20 | beek | Vec: which version of asterisk? |
14:30.45 | Vec | beek : 1.4 latest |
14:33.04 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
14:33.44 | Vec | beek : not working, the variable is not being set although ${LEN(ARG5)} does = 0 |
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14:34.09 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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14:37.24 | Vec | beek : found the prob: I was looking at the length of ARG4, for some reason even though it is not SET it still has a length of 4, for some reason it contains 4 spaces. |
14:38.17 | wdoekes | Vec: isn't LEN(ARG4)=4 ? |
14:38.30 | wdoekes | or does it expand $ARG4 ? |
14:38.37 | wdoekes | *${ARG4} |
14:39.03 | [TK]D-Fender | ^^ |
14:39.15 | [TK]D-Fender | You don't pass it a var nam, you pass it TEXT |
14:39.22 | beek | Vec: Missed that... it IS LEN(${ARG4}) |
14:40.39 | leifmadsen | [TK]D-Fender: I can understand that confusion since there are several functions (like CUT()) that don't want the text -- they want the variable name |
14:41.56 | [TK]D-Fender | leifmadsen: Yeah... but they both have instructions.... |
14:42.04 | [TK]D-Fender | leifmadsen: But seriously... who reads those? ;) |
14:42.28 | leifmadsen | no effin' idea |
14:44.05 | beek | It's not a lack of RTFM, it's more that your eyes start to go buggy looking at that melange of ({[]}) characters. The dialplan instruction looks more like transmission line noise than code. |
14:45.19 | [TK]D-Fender | beek: /me squelches beek |
14:45.23 | [TK]D-Fender | :O |
14:47.00 | brah | what is this i don-t even |
14:47.02 | brah | channel.c:2401 set_format: Unable to find a codec translation path from 0x4 (ulaw) to unknown |
14:50.17 | *** join/#asterisk [T]ank (n=ckwall@132.sub-75-216-112.myvzw.com) |
14:51.46 | [T]ank | so I have a server I set up last night which is remotely hosted. actual phone calls work perfectly. But any playback of files, for example Playback(tt-monkeys) doesn't play back very well. The server shows it working... and every once in a while you can hear the file clip in and out... the are playing at normal speed and such, you just cant hear it. any ideas on what could cause that? |
14:51.53 | [T]ank | I am not getting any errors. |
14:52.01 | [T]ank | just not hearing more than about 1% of the total file |
14:52.10 | *** join/#asterisk ManxPower (n=manxpowe@69.73.94.162) |
14:52.13 | [T]ank | called from multiple cell carriers to test |
14:52.19 | [T]ank | tried multiple files. |
14:52.36 | maddog01 | [T]ank: what version are you using |
14:52.37 | [T]ank | the voicemail messages are the same quality as playback |
14:52.55 | ManxPower | [T]ank: What is the problem? |
14:53.12 | [T]ank | maddog01: 1.6.1.1 |
14:53.20 | maddog01 | [T]ank: and are you using iax |
14:53.27 | [T]ank | yes |
14:53.30 | [T]ank | well... |
14:53.37 | [T]ank | from the carrier it is iax. |
14:53.37 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
14:53.39 | maddog01 | same problem i just fixed |
14:53.52 | [T]ank | carrier to the server... then from the server to the phone it is sip |
14:53.52 | maddog01 | down grade to 1.6.0.1 |
14:54.01 | [T]ank | really?!! |
14:54.02 | maddog01 | all will be fixed |
14:54.04 | [T]ank | just a bug? |
14:54.08 | maddog01 | it a bug |
14:54.12 | [T]ank | ok |
14:54.13 | maddog01 | wait one sec |
14:54.14 | [T]ank | good to know. |
14:54.28 | maddog01 | https://issues.asterisk.org/view.php?id=15337 |
14:54.37 | maddog01 | there thats the info |
14:54.57 | Vec | wdoekes, [TK]D-Fender, beek : for some reason Arguments that are passed into a Macro even when they are empty eg, ${ARG9} seem to consist of 4 spaces " ", so LEN returns 4, is this normal ? |
14:55.21 | [T]ank | looks a little different... mine you just cant hear... that looks more like the issue that the files are played back super slow. |
14:55.26 | [T]ank | am I wrong? |
14:55.30 | [TK]D-Fender | Vec: not AFAIK, show me |
14:55.39 | [T]ank | my issue is that I cannot hear the file even though it is being played. |
14:55.51 | [T]ank | easy fix for me... just change to sip :-D |
14:55.53 | maddog01 | i had the same problem |
14:56.01 | [T]ank | I will try sip first. |
14:56.08 | [T]ank | thanks so much for the reply |
14:56.11 | ManxPower | I like SIP for inter-asterisk stuff. |
14:56.11 | maddog01 | i couldn't hear anything from the outside |
14:56.34 | maddog01 | [T]ank: i couldn't hear anything from the outside either |
14:57.03 | [T]ank | thanks... now I know how to fix. two options. wanted to move to sip anyway |
14:57.07 | [T]ank | thanks a bunch |
14:57.25 | maddog01 | np |
14:57.49 | ManxPower | Vec: Paste the exact Macro(.... line you are using |
14:57.57 | Vec | [TK]D-Fender : not sure how to show u, but here is the code and the output: http://pastebin.com/d672439a2 |
14:58.47 | [TK]D-Fender | Vec: I don't see you executing the macro <----- |
14:58.49 | ManxPower | that's not the macro(... line. |
14:59.29 | [TK]D-Fender | exten => s,n,Verbose(1|The Length of ARG5 is ${LEN(ARG9)}) <---- SAYS 5 BUT YOU'RE LOOK AT ***9*** |
14:59.45 | [TK]D-Fender | And without reverenceing the car |
14:59.55 | [TK]D-Fender | And without referencing the variable |
14:59.56 | Vec | yes I know 9 is not in use, the same thing happens with 5!!! |
15:00.01 | [TK]D-Fender | Vec: We just went through this!@ |
15:00.09 | leifmadsen | ${LEN(${ARG9})} |
15:00.12 | Vec | I will get the macro line |
15:00.15 | ManxPower | Vec: less talk, more pasting of the actual calling of the macro |
15:00.18 | leifmadsen | the length for ${LEN(ARG9)} is 4 |
15:00.21 | [TK]D-Fender | Vec: NO |
15:00.22 | Vec | ok ok 1 sec |
15:00.28 | [TK]D-Fender | Vec: You pass LEN **test**, not a var name! |
15:00.31 | [TK]D-Fender | text |
15:00.44 | ManxPower | [TK]D-Fender: good call! |
15:01.09 | ManxPower | ARG9 is 4 chars. Of course ${ARG9} would be however many chars the value is. |
15:01.16 | *** join/#asterisk comradeb14ck (n=comradeb@72.37.252.50) |
15:01.30 | Vec | oh oops |
15:01.35 | comradeb14ck | hi all |
15:01.52 | Vec | LOL that is dumb |
15:01.53 | Vec | ok thanks |
15:01.54 | Vec | get it |
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15:08.47 | boch | is it possible to continuing executing dialplan after Park() is called? I am taking the call with another call but when the second one ends, it hangs up the first one too |
15:10.22 | [TK]D-Fender | boch: Show us |
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15:21.59 | boch | [TK]D-Fender, http://pastebin.com/m5286df7b |
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15:27.54 | [TK]D-Fender | boch: How are you transferring the call? |
15:27.54 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
15:27.58 | [TK]D-Fender | boch: And next time, show the ENTIRE call |
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15:28.52 | boch | [TK]D-Fender, with Park() app |
15:28.58 | amazinzay | Is there any way to disconnecte manager connections from the CLI? |
15:29.18 | [TK]D-Fender | boch: You don't transfer with park, you transfer with your PHONE |
15:29.58 | boch | [TK]D-Fender, so im not transfering the call, im running Park() in dialplan |
15:30.37 | [TK]D-Fender | boch: You do an attended transfer of a call you're on to Par(), You don't jsut start a new call and hang up |
15:32.12 | korcan | can I use open source Asterisk with a PRI card? OpenVox D410E ? |
15:32.35 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
15:32.50 | korcan | Someone told me I have to pay for a business license to use a PRI card... |
15:32.57 | russellb | korcan: that is not true. |
15:33.03 | russellb | that person is full of lies. :-) |
15:33.39 | korcan | Thanks :) thats what I needed to know!!! |
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15:33.54 | korcan | russellb, any experience with OpenVox cards? |
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15:34.21 | russellb | I have no experience with them. I also do not recommend them. I recommend the use of Digium cards (I work for Digium). |
15:34.33 | russellb | I do not support cheap chinese clones :-p |
15:34.41 | comradeb14ck | korcan, the older models of openvox cards (like the d410e) is an exact clone of digium |
15:34.49 | Joel | don't use a clone |
15:34.55 | Joel | anyone who says a clone is ok is retarded |
15:35.03 | comradeb14ck | the newer openvox models, the 800 and 1200 are not clones (they have their own drivers) |
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15:35.56 | comradeb14ck | If you want something stable that will last you a while, I recommend digium/sangoma/rhino cards. + those companies all have English tech support. OpenVOX has no english tech support. |
15:36.05 | comradeb14ck | So you'll be kinda screwed if you need assistance with it. |
15:36.13 | amazinzay | Does anyone know how to disconnect a manager from the CLI? |
15:36.19 | *** join/#asterisk IBC_jkenney (n=jkenney@99.23.50.73) |
15:36.23 | comradeb14ck | restart now? =p |
15:36.29 | [TK]D-Fender | korcan: The OpeVox clones are the crappy old PCI design, and their support sucks, so if you have problem, you're asking for trouble |
15:36.50 | ManxPower | Digium moved away from their original design for a reason. PCI issues. |
15:36.52 | jameswf | korcan: And they hate kittens... |
15:37.06 | jameswf | just sayin |
15:37.23 | amazinzay | anything that doesn't take the phone sytem down? |
15:37.32 | korcan | I need a 4 port PRI Card, and a 12 Port FXS Card |
15:37.36 | denon | ManxPower: it's a pity USB was designed so poorly |
15:37.51 | jameswf | USB 3.0 <3 |
15:38.01 | coppice | if they hate kittens they can't be all bad |
15:38.01 | korcan | any recommendations? |
15:38.26 | korcan | I must have missed something.... What do kittens have to do with anything? |
15:38.27 | denon | jameswf: I was thinking something more like ePCIe :) or Infiniband on commodity PCs |
15:38.43 | korcan | PCI Express |
15:40.00 | jameswf | korcan: you havent been on the internet verry long kittens have everything to do with everything |
15:40.05 | jameswf | http://icanhascheezburger.com/ |
15:40.58 | IBC_jkenney | I am also confused on the kittens |
15:41.10 | russellb | <3 kittens |
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15:42.05 | korcan | so any PCI-E card recommendations? |
15:43.35 | Qwell | korcan: Digium, naturally |
15:43.55 | Qwell | I think all of our current hardware has PCIe models |
15:44.14 | russellb | korcan: http://www.digium.com/en/products/digital/te420.php |
15:44.25 | IBC_jkenney | I heard a rumor that if you have a Digium card you need to use a business license of asterisk |
15:44.27 | IBC_jkenney | is that true |
15:44.43 | Qwell | IBC_jkenney: no.. |
15:45.00 | IBC_jkenney | so that is just an ugly rumor? |
15:45.09 | Qwell | IBC_jkenney: one I've never heard before |
15:45.10 | russellb | korcan: http://www.digium.com/en/products/analog/aex2400.php ... it's a 24 port analog card, the next smaller is 8 |
15:45.14 | denon | probably heard it from you sangoma rep? |
15:45.21 | russellb | IBC_jkenney: where did you hear that? that has never been the case |
15:45.33 | denon | you/your |
15:46.32 | jameswf | adds rumor to sales schpeel |
15:46.38 | jameswf | :) |
15:47.11 | Qwell | jameswf: write me a script to automatically connect to all my VMs to build new packages when CentOS releases a new kernel update, k? |
15:48.10 | Joel | haha |
15:48.10 | [TK]D-Fender | Qwell: Outsourcing to the competition now? ;) |
15:48.10 | [TK]D-Fender | *ouch* |
15:48.10 | Joel | keep dreaming Qwell |
15:48.10 | jameswf | holy netsplit batman |
15:48.10 | Qwell | [TK]D-Fender: it's in his best interest! |
15:48.12 | denon | Qwell: PXE boot all your VMs, and NFS all your configs to one place :) |
15:48.12 | Joel | Qwell just got told about cobbler, would have been very handy a few days ago :\ |
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15:48.51 | Qwell | Joel: meh |
15:48.51 | [TK]D-Fender | Qwell: He doesn't charge any, damn sub-prime loans! |
15:48.51 | Qwell | denon: persistent VMs + svn :D |
15:48.51 | jameswf | Qwell: you know you don't actualy have to run a kernel to build against it... |
15:48.51 | Qwell | jameswf: I know |
15:48.51 | Qwell | but I have to install it, and build the packages |
15:49.01 | *** part/#asterisk Amorsen (n=Amorsen@94.127.50.7) |
15:49.01 | jameswf | I has a robot that syncs to the repos and spits out RPM's he is a neat little fella |
15:49.22 | denon | Qwell: I find it's best to let your kernels age .. like a fine wine .. few new security advisories every few months gives them flavor |
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15:50.28 | jameswf | denon: Security is over rated... chmod -R denons_mom |
15:50.46 | jameswf | crap |
15:50.58 | jameswf | chmod -R 777 denons_mom |
15:51.14 | denon | yeah chmod requires some args .. it's not funny once you screw up the insult :) |
15:51.20 | jameswf | lol |
15:52.06 | IBC_jkenney | <russellb> a co worker brought it up plus i read it on a few lists |
15:52.18 | russellb | weird. |
15:53.02 | jameswf | Probably an old troll steve post |
15:53.17 | [TK]D-Fender | [11:32]<korcan>Someone told me I have to pay for a business license to use a PRI card... |
15:53.18 | jameswf | what happened to steve the sangoma troll |
15:53.21 | [TK]D-Fender | And this poor chap :) |
15:53.38 | beek | Can anyone here give me the definitive answer to the question: Does ADA (Asterisk Desktop Assistant) have a plugin that works with Internet Explorer like it does with Firefox? |
15:53.42 | IBC_jkenney | see i am not the only one |
15:54.18 | denon | [TK]D-Fender: I heard that digium cards all call home to activate, and digium is going to start charging ongoing license fees ... </rumor type='foolish'> |
15:55.05 | Joel | I heard Qwell was going to propose to me |
15:55.52 | jameswf | I heard digium is being bought by Microsoft |
15:56.19 | Qwell | jameswf: old rumor |
15:56.26 | russellb | we're rewriting Asterisk in bash |
15:56.26 | IBC_jkenney | I heard Microsoft is buying everyone |
15:56.38 | Joel | I heard a rumor asterisk was getting more stable |
15:56.40 | Joel | </burn> |
15:56.41 | Joel | ;) |
15:56.43 | Qwell | Joel: FALSE |
15:56.45 | Qwell | err, wait |
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15:56.46 | jameswf | Except freeswitch that was bought by baracuda |
15:57.24 | denon | MORE stable? you imply that there's something more stable than perfectly stable? |
15:57.31 | russellb | slaps Qwell |
15:57.32 | Joel | speaking of bs, global warming, it was 60F for the high yesterday, week before it was 85 out |
15:57.40 | Qwell | russellb: what? :( |
15:57.51 | russellb | Qwell: you said Asterisk wasn't getting more stable! |
15:58.09 | Qwell | I misread! |
15:58.29 | jameswf | okay lets try this on |
15:58.48 | jameswf | Digium is reported to be in talks to buy Fonality for $50 |
15:58.53 | russellb | lol |
15:58.53 | Qwell | pfft |
15:58.56 | Qwell | clearly false |
15:59.02 | Qwell | who would pay $50 for that? |
15:59.03 | denon | I doubt fonality would bring 50 bucks |
15:59.04 | leifmadsen | what a rip off that'd be |
15:59.31 | denon | unless the sale included some cheeeezburgers for kitty |
16:00.31 | Joel | I love how everyone knocks fonality, work environment wise it's probably the most fun I've ever had |
16:00.47 | Joel | read: awesome when not dealing with chris 'cry baby' lyman |
16:00.52 | *** join/#asterisk ManxPower (n=manxpowe@69.73.94.162) |
16:02.09 | leifmadsen | hmmm... putting a DROP rule in iptables just before the 2 rules you want to allow traffic with is a sure fire way to make them not work :) |
16:02.19 | Joel | leifmadsen :P |
16:02.26 | jameswf | oh crap Fonality troll in here... |
16:02.36 | leifmadsen | is an asterisk troll |
16:02.39 | leifmadsen | so it balances out |
16:02.44 | leifmadsen | s/troll/advocate |
16:03.00 | jameswf | !!!!!!!! |
16:03.08 | leifmadsen | I don't want to meet your mom..... |
16:03.11 | jameswf | s/!/poo /g |
16:04.00 | leifmadsen | nice :) |
16:04.07 | Joel | jameswf how's rhino's deal with fonality going? |
16:04.17 | jameswf | ? |
16:05.03 | Joel | oh oops, nm. |
16:10.50 | jameswf | wants a Jalipeno cheese bagel... |
16:11.04 | thehar | mmmm me 3 |
16:11.25 | ariel_ | oh boy, it's hot pockets for lunch today....;-( |
16:11.36 | Joel | ariel_ barf pockets? |
16:11.51 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
16:11.53 | ariel_ | well yes but it's cheap and it's in the vending machine |
16:12.33 | *** join/#asterisk Boingo (n=malachi@mail.techglia.com) |
16:13.31 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.82.28) |
16:13.34 | DelphiWorld | hello |
16:13.36 | Boingo | Hello everyone, I have been looking for hard drive sizing suggestions for Asterisk. I found CPU/RAM/etc but not much for hard drive. I intend to save voicemails for < 10 users, only a few voicemails a day. |
16:14.06 | DelphiWorld | Qwel: what is the included asterisk version in asterisknow 1.5? |
16:14.37 | Qwell | Boingo: assume each of those 10 users have 100 VMs saved.. thats 1GB *at most* |
16:14.52 | ManxPower | Boingo: What format will your voicemail be in and how many mins of messages do you want to store (not number of messages, number of mins) |
16:15.51 | leifmadsen | 10 users? any modern HD is going to be fine |
16:16.05 | Boingo | Dont really care which format. Figure 2/3 minutes per message? 3 per day? So 9ish minutes per day * < 10 users. 90 minutes per day? |
16:16.25 | ManxPower | Boingo: I suggest using WAV49 |
16:16.29 | leifmadsen | 120GB will be more than enough |
16:16.33 | Boingo | That is way on the high side of what will actually happen.. but it should be good for an estimate. |
16:16.39 | ManxPower | As leifmadsen said, any modern drive will be fine. |
16:16.41 | leifmadsen | it's not gonna be like 1MB a minute or anything :) |
16:16.47 | leifmadsen | and even then... :) |
16:16.55 | ManxPower | leifmadsen: WAV would be like 1 mb/min wouldn't it? |
16:16.58 | Boingo | Why Wav49? |
16:17.15 | leifmadsen | ManxPower: yes, unless you use wav49 like others use :) |
16:17.27 | ManxPower | Boingo: because if at some time in the future you want the messages e-mailed to the user, WAV49 is the only small format WMP knows about without extra software. |
16:17.41 | Boingo | Oh. |
16:17.48 | Boingo | I do want to email the voicemails. |
16:17.52 | ManxPower | and you do NOT want to change formats on your voicemail store in the middle. |
16:17.57 | DelphiWorld | qwel: tel me please |
16:18.08 | Boingo | WMP doesnt know about mp3? |
16:18.10 | DelphiWorld | qwel: what is the included asterisk version in asterisk now 1.5? |
16:18.15 | Boingo | Or is Wav49 smaller? |
16:18.28 | Boingo | Or is MP3 generally frowned upon for some other reason? |
16:18.30 | ManxPower | Boingo: What makes you think Asterisk can (out of the box) save voicemails in MP3 format? |
16:18.41 | Boingo | No particualr reason. |
16:18.57 | ManxPower | MP3 is patent encumbered. Any MP3 support would have to be added from an outside package/source |
16:19.03 | Boingo | I guess somewhere in the back of my head, that was what I was aiming for. |
16:19.08 | Boingo | Ah, ok. |
16:19.12 | Boingo | No biggie. |
16:19.15 | Joel | Qwell ok for the record, you don't need a comps.xml in the yum repo. so yum repo solves the issue for me |
16:19.15 | Boingo | Whatever works. |
16:19.20 | leifmadsen | Boingo: wav49 is basically GSM with a WAV wrapper on it so Windows users can play it. |
16:19.36 | Qwell | Joel: you do for groups |
16:19.38 | leifmadsen | Boingo: i.e. it is a compressed format |
16:19.41 | Joel | Qwell correct |
16:19.42 | Boingo | I assume Mac/Linux as well? |
16:19.51 | Joel | Qwell but who needs those in a headless install? |
16:19.53 | *** join/#asterisk phurl (n=mdupont@82.114.94.9) |
16:19.58 | Boingo | I have users on all OSes. |
16:20.01 | leifmadsen | Boingo: correct |
16:20.04 | Boingo | Cool. |
16:20.08 | Boingo | Sounds like a winner. |
16:20.54 | Boingo | Is there anything else that is hard drive consuming besides voicemail? |
16:21.02 | ManxPower | Boingo: logs |
16:22.06 | Boingo | Any ideas on sizing for that? |
16:22.20 | Boingo | I am assuming I don't need to keep logs until the end of time. |
16:22.34 | Boingo | Rotate them out after a certain amount of time. |
16:23.05 | ManxPower | Boingo: it all depends on how much logging you enable and how many calls you get. |
16:23.23 | ManxPower | On some systems I've installed logs take up like 100K/day. On other systems it's like 5GB/day |
16:23.42 | Boingo | Well, I am probably a lot closer to the bottom ned. |
16:23.44 | Boingo | *end |
16:23.49 | ManxPower | Do you really think a modern HD would not meet your needs? |
16:24.01 | Boingo | Yes, I think it will. |
16:24.06 | Boingo | Just trying to get an idea. |
16:24.50 | Boingo | Does Asterisk play nice in a VM? All VoIP, no cards. |
16:25.03 | Boingo | Mainly my reason for asking about the HD I guess. |
16:25.06 | ManxPower | Boingo: no. |
16:25.10 | Boingo | No? |
16:25.16 | Boingo | I am surprised. |
16:25.21 | ManxPower | Boingo: Asterisk is a pesudo realtime system. |
16:25.28 | Boingo | Looking at the CPU/RAM specs.... I figured. |
16:25.54 | ManxPower | Boingo: will you be using any conferencing? |
16:26.01 | Boingo | Doubtful. |
16:26.14 | ManxPower | try it and see, just don't expect it to work well under load. |
16:26.29 | Boingo | Where would the bottleneck be? |
16:26.37 | ManxPower | Boingo: latency and jitter |
16:27.23 | Boingo | So I am better off dropping it on an old 486 than I am putting it in a VM with 10x the horsepoweR? |
16:27.49 | ManxPower | Boingo: a 486 may not have enough cpu power to do much better. |
16:28.21 | Boingo | I was reading a page that had all sorts of really old computers listed that were running it. Or was I missing something? |
16:28.35 | ManxPower | Remember Asterisk is a SOFTWARE PBX Toolkit. It takes all the stuff that used to be done on expensive telecom cards' and does it in software on the PC. |
16:29.07 | ManxPower | Boingo: I once ran Asterisk on a low end Pentium. It could handle 1 call at a time and the codec has to be ulaw. otherwise there was not enough CPU to keep up |
16:29.09 | thehar | Is anyone familiar with Queuemetrics? |
16:29.41 | Chainsaw | Boingo: Virtual machines fall down on one specific thing. |
16:29.46 | Chainsaw | Boingo: Timing. |
16:30.10 | ManxPower | Chainsaw: But applications should not care about timing! *grin* |
16:30.27 | Chainsaw | Boingo: When handling voice packets that are so latency-sensitive that even TCP is regarded as overhead, you can not afford the non-deterministic timing that a VM will give you. |
16:31.15 | Boingo | So, essentially, older harderware still trumps my much better (spec wise) VM? |
16:31.33 | ManxPower | Boingo: for the most part yes. |
16:31.45 | Boingo | Interesting. |
16:32.06 | ManxPower | Just like older hardware would be better at realtime data capture than the same software running in a VM. |
16:32.24 | Boingo | Is there any harm in trying it in a VM? Or is it a waste of time? |
16:32.24 | ManxPower | when you go realtime or pseudo-realtime all the rules change. |
16:32.42 | ManxPower | Boingo: I expect it will work in a VM with low usage. |
16:32.50 | ManxPower | once you ramp up is another story. |
16:32.56 | Boingo | I mean, is it the sort of thing where %100 of everyone thinks it sucks and goes back? |
16:33.03 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
16:33.09 | Boingo | Or is it the sort of thing where it has worked for some people? |
16:33.29 | Boingo | If I ramp up, then I can figure out moving it later. |
16:33.39 | ManxPower | Boingo: and some people win the lottery. |
16:33.44 | [TK]D-Fender | Boingo: Has worked for quite a few. YMMV, don't get your hopes to high and be prepared for great frustration and the possibility you'll never get it wuite right |
16:33.50 | Boingo | Not what I ment. :-) |
16:34.18 | Boingo | It cant be that hard to move it to a different box later. |
16:34.24 | Boingo | Right? |
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16:37.49 | [TK]D-Fender | Boingo: Move what? * configs? jsut a bunch of text files... |
16:38.02 | Boingo | Thats what I figured. |
16:38.24 | Boingo | Install it on the other end, copy over the configs (and possibly VMs) and start it up. |
16:38.48 | Boingo | Change a few IPs/DNS and should be all good. |
16:39.08 | Joel | recordings, prompts, call records |
16:39.12 | Boingo | But then again, I have never used Asterisk so maybe I don't know what I am talking about. |
16:39.20 | jameswf | ~nowwhat |
16:39.21 | infobot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=FJ3oHpup-pk |
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16:44.44 | [TK]D-Fender | TALKIN' BOUT FLEA MARKET! |
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16:46.37 | russellb | lol |
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16:57.40 | phurl | what is the best open source database for a call center with many users and replicatoin |
16:59.25 | bmoraca | assuming all things are equal (service costs, capabilities, etc), which would be the trunk mechanism you would use: SIP or PRI? is the hardware timing source that a PRI gives you very important if we're talking about, say, a 50 phone deployment? |
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17:01.58 | jameswf | bmoraca: I say PRI.. sip is okay but alot of break points |
17:02.55 | bmoraca | well, technically, the PRI is delivered via SIP...I'm more curious about just how important the external hardware timing source is |
17:03.35 | *** part/#asterisk Connor (n=billy@75.76.32.44) |
17:04.42 | [TK]D-Fender | bmoraca: easy... PRI |
17:05.02 | [TK]D-Fender | and a PRI is not deliveered over SIP |
17:05.15 | bmoraca | it's an emulated PRI, so, yes, it does get delivered over SIP |
17:05.22 | bmoraca | like a media gateway |
17:05.28 | [TK]D-Fender | bmAnd what does "emulated PRI" mean? |
17:05.29 | carrar | then it's not a PRI |
17:05.30 | carrar | it's SIP |
17:05.45 | [TK]D-Fender | bmoraca: You can't emulate a clock sync. You can't get telco monitoring like PRI. |
17:06.14 | [TK]D-Fender | bmoraca: And the SIP is never equal. In the case of ITSP's it goes over the internet where "random shit" happens. constantly. |
17:06.25 | carrar | Hows that Q.931 over SIP |
17:06.27 | carrar | heh |
17:06.45 | [TK]D-Fender | carrar: Would be more viable to sugget H.323 for that ;) |
17:06.48 | rene- | PRI is very stable, actually it remained the ruling party in my country for well over 70 years |
17:06.50 | carrar | haha |
17:07.16 | carrar | PRI Party sounds fun |
17:07.28 | carrar | no loopbacks required |
17:07.33 | [TK]D-Fender | rene-: Cell party gained some ground I head, but became diconnected from the populace outside of tower range... |
17:07.40 | [TK]D-Fender | heard* |
17:08.03 | bmoraca | [TK]D-Fender: the device is a T1 CSU/DSU with a DSX-1 port that emulates PRI functionality. the device itself trunks out via SIP. so calls placed through its PRI terminate via SIP to the provider network (whoever that may be). it uses the clock source of the inbound T1 (or it can generate its own internal clock source) and uses that as the clock for the PRI. |
17:08.15 | bmoraca | my question is not so much about this device, as I know it works. |
17:08.42 | carrar | Yes lets move on to the BEEF of your question |
17:08.42 | bmoraca | my question is about asterisk and whether or not its extremely important to have an external timing source, rather than relying on ztdummy or dahdidummy |
17:08.51 | [TK]D-Fender | bmoraca: Well the sync to the local box may be solid, but the guarantees on the packets out of it ... well thats another matter |
17:10.39 | bmoraca | [TK]D-Fender: i realize that, but 9 times out of 10, the SIP provider and the T1 provider are the same company. my concern is for asterisk...as it would be just as easy for me to simply provide a SIP trunk to asterisk from that same provider...but if I'm going to have issues with conference and playback and what not due to not having an external timing source, then I would prefer to go the... |
17:10.41 | bmoraca | ...route of the PRI |
17:11.48 | [TK]D-Fender | bmoraca: For conferencing and playback you only are recommended to have a timing SOURCE. That would be the card itself,k not so much being used as the link to a telco |
17:12.01 | ManxPower | With a PRI if you have problems you can call the telco and say "The PRI is down" and they might fix it. If you call up your ISP and say my SIP connection is down, the telco will laugh at you. |
17:12.10 | [TK]D-Fender | bmoraca: Card prividers mixer & timing, not the telco |
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17:13.04 | bmoraca | [TK]D-Fender: so the card doesn't just take the timing from the line? interesting. I thought it required the link to provide accurate timing. |
17:13.29 | [TK]D-Fender | bmoraca: No, which is why many people bought blank TDM cards for this |
17:13.58 | bmoraca | hmmm...learn something new every day... |
17:16.07 | ManxPower | bmoraca: there are two types of "timing". There is timing for a T-1 (which I call "sync source") this "sync" comes from the telco usually. and timing for meetme and IAX2 trunking, that timing comes from a zaptel device, |
17:17.01 | bmoraca | ManxPower: I realize that, but I always thought that the zaptel device got its timing from its PSTN link (atomic PSTN clock or T1 sync) |
17:17.20 | ManxPower | bmoraca: the telco sync has no time info in it. |
17:17.43 | bmoraca | no time info, no, but there is definitely clock sync |
17:17.56 | ManxPower | Zaptel cards can receive their sync from the T-1(PRI) or it can provide sync to a device connected to the T-1/PRI. |
17:18.13 | ManxPower | bmoraca: I don't usually call things that don't measure time "clocks". 8-) |
17:18.41 | [TK]D-Fender | bmoraca: That sync is only to align communication with the telco. The alignment has nothing to do with the interval being constant <- |
17:18.44 | ManxPower | But everyone else seems to. |
17:19.52 | bmoraca | interesting |
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17:20.12 | ManxPower | there is no "atomic clock" involved in t-1 sync, AFIK |
17:20.18 | bmoraca | i still think I'll push customers toward the PRI...couple thousand extra on the bill and all that |
17:20.34 | ManxPower | bmoraca: it will save them time in the long run. |
17:20.36 | bmoraca | ManxPower: not T1 sync, no, but on bare copper analog, there is |
17:20.45 | ariel_ | most PRI's end up being cheaper here in the states... |
17:20.56 | coppice | ManxPower then you don't know much about telephony |
17:21.17 | [TK]D-Fender | rut-roh! |
17:21.22 | ManxPower | coppice: I know much less about the internals of the telco. |
17:21.35 | ManxPower | coppice: so correct me. |
17:21.55 | ManxPower | You *are* the expert (no sarcasm intended) |
17:22.01 | ariel_ | sit's back and watches and takes notes |
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17:22.14 | coppice | the T1 clock is atomic sourced. they need to do this to keep the entire net nicely synced without requiring some vulnerable national master clock source |
17:22.42 | coppice | every public exchange contains a rhubidium clock |
17:23.05 | ManxPower | I sit corrected. It still doesn't send "time info", just ticks, right? |
17:23.08 | bmoraca | ooo...throw some water on that thing and watch sparks fly! |
17:23.20 | bmoraca | ManxPower: what is time but a series of ticks? |
17:23.41 | coppice | A timing source is also known as a "clock". A "clock" |
17:23.42 | Qwell | bmoraca: tocks too |
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17:23.59 | ManxPower | bmoraca: time is a count of ticks. If you don't count them you don't know the time, do you? |
17:24.41 | ManxPower | coppice: so what should we call zaptel "timing" for things like meetme and iax2 trunking. |
17:25.08 | ManxPower | To avoid newbie confusion? |
17:25.19 | coppice | time is all relative. times of day are just as relative to midnight as digit clock pulses are relative to each other |
17:25.59 | ManxPower | newbies think they need NTP installed for their T-1 "clock" |
17:26.20 | ManxPower | OK, SOME newbies at least |
17:26.36 | [TK]D-Fender | ManxPower: Without a timing source it takes newbs 2 hours to watch "60 Minutes" :p |
17:26.48 | jaytee | and if you use the better atomic clock in Colorado then your T1 will be accurate to a microsecond :-) |
17:27.20 | coppice | more often people think they need a more accurate clock in absolute terms, rather than one that's just precisely in sync with the far end |
17:27.58 | nkohh | [TK]D-Fender!! man! how's it going broheim? i haven't seen you in forever! |
17:28.02 | nkohh | what are you up to these days? |
17:28.30 | *** join/#asterisk pulpster (n=pulpster@p16.eregie.pub.ro) |
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17:30.09 | [TK]D-Fender | ~cluebat nkohh |
17:30.10 | infobot | ACTION pulls out a ClueBat (tm) and thwaps nkohh. |
17:31.06 | nkohh | if you keep that up, im going to have to consult with my lawyer, Bob Loblaw |
17:31.54 | nkohh | you may recognize my lawyer from his blog, Bob Loblaw's Law Blog |
17:31.57 | [TK]D-Fender | nkohh: I can swing harder and spare you the need for more than a final set of visitors :D |
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17:32.47 | [TK]D-Fender | nkohh: Seriously that random nag script of your is obnoxious, please turn it off in here. |
17:33.04 | nkohh | it's not random or nagging |
17:33.09 | nkohh | and it most certainly is not a script |
17:33.26 | pulpster | hello |
17:33.28 | [TK]D-Fender | nkohh: Fine, "slightly targeted, one shot annoyance the same as the rest" script |
17:33.49 | pulpster | anybody has a configuration of skinny.conf for a 7911G cisco phone or a config that works for that |
17:39.35 | eppigy | TRABAJO |
17:39.47 | [TK]D-Fender | eppigy: Holiday here :p |
17:39.51 | eppigy | YES |
17:40.00 | eppigy | I am off thursday |
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17:44.06 | admin0 | hi .. is anyone using this ? http://www.voip-info.org/wiki/view/Vovida.org+load+balancer .. |
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17:52.49 | Boingo | Will having MySQL for CDR on a different computer affect Asterisk to any real degree? |
17:53.06 | [TK]D-Fender | Boingo: No. |
17:54.14 | hydra__ | Hi. I have 2 phones and 2 fixed phone lines. What is the quickest way to create a "load balancer" for those 2 fixed lines? |
17:55.01 | ManxPower | hydra__: call the telco, tell them you want "longest idle hunting" between the two lines. |
17:55.06 | hydra__ | Example: if someone is using a phone line and I make a simultaneous call, the other line would be used |
17:55.18 | ManxPower | hydra__: um, Asterisk. |
17:55.59 | hydra__ | ManxPower: I got to have PCI cards for in/out rs232? |
17:56.22 | [TK]D-Fender | hydra__: RS-232? |
17:56.24 | [TK]D-Fender | HUH!? |
17:56.40 | bmoraca | hydra__: staples has 2-line phones for like $99 that don't require any PBX and will allow you to do what you want to do |
17:56.43 | ManxPower | hydra__: Perhaps you should go to digium.com and asterisk.org and learn a little about Asterisk |
17:56.59 | [TK]D-Fender | hydra__: You need a proper FXS/FXO interface to plug all those in. Ones like the Digium TDM410P, Sangoma A200, etc |
17:57.33 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
17:57.37 | hydra__ | bmoraca: how do they work? |
17:57.46 | bmoraca | you plug them in... |
17:57.58 | ManxPower | this is too painful to watch. |
17:58.00 | *** part/#asterisk ManxPower (n=manxpowe@69.73.94.162) |
17:58.00 | hydra__ | i have to chose which line to use? |
17:58.12 | bmoraca | if a line is busy, it shows up as red instead of clear and then you use the other one |
17:58.38 | [TK]D-Fender | bmBut taht isn't automatic! OH NOES!!!! |
17:58.44 | bmoraca | ~rtfm |
17:58.45 | infobot | rtfm is probably Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://en.wikipedia.org/wiki/RTFM |
17:59.16 | bmoraca | you know what I always liked? CyberData VoIP paging speakers and amplifiers has a button on them that's called the "RTFM Button" |
17:59.33 | bmoraca | that's litterally how they label it on the device and what they refer to it as in the manual |
18:02.18 | hydra__ | So, the reasonable way to do this install asterisk on a box with TDM410P? |
18:02.27 | *** join/#asterisk mltlnx (n=mltlnx@mail.rssalesco.com) |
18:02.49 | IBC_jkenney | If i have a digium PRI card and a Digium FXS card and want to use the FXS card for analog devices can i Push calls thru the PRI interface with the dialplan? |
18:03.25 | IBC_jkenney | (I have read i can but the people at digium are making me wonder) |
18:04.30 | IBC_jkenney | I am sure its just a matter of Device A using Context to place call thru DeviceB |
18:04.45 | [TK]D-Fender | IBC_jkenney: You can process your calls however you want, its your dialplan |
18:05.02 | hydra__ | [TK]D-Fender: a box with one TDM410P card + 2 phones with 2 fixed lines is a good combination? |
18:05.41 | IBC_jkenney | I am just checking the digium person i got on the phone stated they didn't know if that would work |
18:05.44 | hydra__ | (for line balancing) |
18:05.45 | jameswf | still wonders how Digium cards are NOT F.A.C.E. certified.... |
18:05.49 | IBC_jkenney | so I am now just double and tripple checking |
18:06.08 | [TK]D-Fender | hydra__: Sure Actually, Id get the TDM410P with 2 FXO modules, and get a 2-port ATA for the phones. |
18:06.24 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
18:06.36 | coppice | jameswf: just another faceless company |
18:06.43 | [TK]D-Fender | IBC_jkenney: You dial withever you want in your dialplan <- |
18:07.08 | jameswf | oh snap... coppice that should be a tag line |
18:07.22 | hydra__ | what is 2-port ATA? |
18:07.25 | jameswf | we can all be faceless certified |
18:08.07 | coppice | we can all be certified |
18:09.29 | *** join/#asterisk pepe (n=pepesz@77-120.ipact.nl) |
18:09.31 | pulpster | anyone knows what is the Firmware version identifier for a 7911 to place in skinny.conf ? |
18:10.42 | *** join/#asterisk arthax0r (i=arthax0r@gateway/shell/blinkenshell.org/x-1bcf6bf4e4736918) |
18:14.39 | *** join/#asterisk Dwayne__ (n=Dwayne@64.42.227.97) |
18:14.51 | ariel_ | pulpster: you mean the version=P002F202 this should be the one you have on the tftp for upgrading or keeping the devices with same version of software. |
18:15.01 | Dwayne__ | how can i check if my asterisk box has spandsp installed properly |
18:15.15 | pulpster | yes |
18:16.17 | pulpster | should I leave that one ? is this version the same that should I put in OS79XX.TXT file ? |
18:16.57 | ariel_ | it's the version you have for the device in your tftp for upgrading |
18:17.06 | ariel_ | or to maintain a stable build |
18:17.16 | pulpster | I don't have one |
18:17.33 | *** join/#asterisk intralanman (n=lanman@173-133-135-140.pools.spcsdns.net) |
18:17.39 | *** join/#asterisk mltlnx (n=mltlnx@mail.rssalesco.com) |
18:17.41 | *** join/#asterisk ccesario (n=ccesario@200.183.211.66) |
18:17.46 | ariel_ | then your device will not get upgraded or updated. |
18:18.22 | Deeewayne | Dwayne__, look for the spandsp libs in your lib dir |
18:19.02 | pepe | Hi, can someone tell me how to implement "callback on busy" (http://www.voip-info.org/wiki/view/Asterisk+tips+Call+Back) into my system? see http://pastebin.com/d1ef94229. Thanks for help:) |
18:19.20 | pulpster | P003-08-4-00 is the version to load into memory of phone (found it in asterisk book) - is this the same for all cisco phones - in particular for 7911 ? |
18:20.01 | ariel_ | pulpster: each phone has a diff file, if you have a cal from cisco you can go to there site and get the one you need |
18:20.49 | *** join/#asterisk layne (n=layne@c-68-37-72-150.hsd1.nj.comcast.net) |
18:26.41 | *** join/#asterisk rudeboy_xix (n=rudeboy@ded-139-109.eglobalreach.net) |
18:30.54 | *** join/#asterisk davevg (n=davevg-b@75.97.64.33.res-cmts.senj.ptd.net) |
18:32.08 | [TK]D-Fender | pepe: The sample shows you dialplan that allows you to choose to activate the callback |
18:32.19 | [TK]D-Fender | pepe: And your dialplan looks nothing like it. |
18:32.50 | [TK]D-Fender | pepe: Also, all they do is issue an Originate which calls the target 100 times which, if they have call-waiting-like ability will nag the shit out of them |
18:33.11 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-d373974d215267e5) |
18:33.17 | IBC_jkenney | is there a digium employee in here? |
18:33.18 | Katty | grooves |
18:33.26 | Katty | i'm never gonna dance again! |
18:33.40 | Katty | [TK]D-Fender: waste the chance that i'd been given |
18:33.52 | Katty | IBC_jkenney: probably. |
18:34.03 | IBC_jkenney | i'd like to speak to one if possible |
18:34.11 | Katty | IBC_jkenney: i would recommend calling them. |
18:34.16 | [TK]D-Fender | Katty: I liek the new hard-rock version better.... |
18:34.16 | IBC_jkenney | i just did |
18:34.24 | Katty | IBC_jkenney: call them back. |
18:34.33 | Katty | [TK]D-Fender: oh? linky link? |
18:34.34 | [TK]D-Fender | IBC_jkenney: About what? |
18:34.45 | IBC_jkenney | Its a really long story |
18:34.54 | Katty | right. so. |
18:34.56 | IBC_jkenney | but in a nutshell information on a product they carry |
18:34.57 | Katty | call them back. |
18:35.07 | Katty | hit the redial button. |
18:35.12 | eppigy | Katty: not the way I danced with you |
18:35.27 | [TK]D-Fender | Katty: http://www.youtube.com/watch?v=I7imqO-OBVk |
18:35.41 | [TK]D-Fender | IBC_jkenney: Just ask your question already |
18:35.41 | Katty | listens |
18:35.51 | Katty | hmm. |
18:36.26 | Katty | that's like listening to another person sing never gonna give you up |
18:36.30 | Katty | it's just not the same |
18:36.37 | [TK]D-Fender | Katty: THANK GOD! :p |
18:36.56 | Katty | i prefer the george michael version. |
18:37.15 | IBC_jkenney | its not important i found my answer. |
18:37.22 | IBC_jkenney | i just expected better people over there |
18:37.23 | IBC_jkenney | thats all |
18:37.55 | Katty | that's funny. |
18:38.05 | Katty | everyone i talk to over there is an absolute joy to work with |
18:38.11 | Katty | maybe it's all about your attitude. |
18:38.32 | Katty | eppigy: ATTITUDE |
18:38.36 | eppigy | i have it |
18:38.39 | eppigy | I HAVE IT |
18:38.42 | Katty | you are it. |
18:38.47 | eppigy | oh true |
18:39.11 | Katty | also |
18:39.17 | Katty | in a completely unrelated topic. |
18:39.31 | Katty | i need some sort of 'exchange' and 'sql' aware backup software. |
18:39.33 | Katty | that does not suck. |
18:39.45 | Katty | which is hard, considering exchange is the suck of all sucky |
18:39.47 | eppigy | netbackup |
18:39.52 | [TK]D-Fender | IBC_jkenney: You're the one being all mysterious about what you actually want and expecting "any random Digium employee" to have whatever answer it is you're looking for. |
18:39.59 | Katty | eppigy: orly |
18:40.12 | pepe | H[TK]D-Fender: I did modified the extentions.conf - changing it to my two digit dialplan. http://pastebin.com/d512b7495 However this doesn't work for me. When line is busy it goes to voicemail directly. If this solution is crapy can you suggest better one |
18:40.17 | eppigy | or backup exec |
18:40.17 | Katty | eppigy: is that like symantec backup exec? |
18:40.21 | Katty | ohiseethenk |
18:40.22 | eppigy | if it is not a huge enviroment |
18:40.23 | IBC_jkenney | i am already on the phone |
18:40.29 | Katty | eppigy: it's like ummm 8 servers? |
18:40.32 | eppigy | yeah for like a crap load of jobs |
18:40.35 | eppigy | o lol |
18:40.38 | eppigy | then backuop exec |
18:40.42 | Katty | mmmmk |
18:40.42 | eppigy | would eb the way to go |
18:40.50 | Katty | does it do like...non compression stuff |
18:41.00 | Katty | where you could actually browse backup directories |
18:41.13 | eppigy | Well you have catalogues |
18:41.16 | eppigy | you browse |
18:41.22 | [TK]D-Fender | pepe: And after your 2-digit exten YOU dump them into voicemail and choose not to do anything with them. |
18:41.22 | Katty | but you're like... |
18:41.23 | eppigy | that list every file and dir |
18:41.28 | Katty | not able to browse the external hard drive |
18:41.33 | Katty | and copy a file, and paste it somewhere |
18:41.44 | Katty | you must use some sort of restore wizard thingamaboberoonie |
18:41.52 | eppigy | well we use a tape library |
18:41.56 | eppigy | if you backup to disk |
18:41.57 | Katty | oh gods. |
18:41.59 | eppigy | you may be able to |
18:42.04 | Katty | i have such horrible horrible luck with tapes. |
18:42.09 | Katty | it's like i'm magnetic or something. |
18:42.12 | eppigy | lol |
18:42.20 | Katty | i'm serious :< |
18:42.25 | eppigy | yeha tapes suck |
18:42.31 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
18:42.33 | eppigy | I do not touch the backups |
18:42.36 | Katty | we've been using external hard drives and REV disks. |
18:42.44 | Katty | and cobian backup |
18:42.48 | eppigy | I replicate my mysql db's and rsync my filez |
18:42.50 | Katty | and batch scripts |
18:43.11 | Katty | i will pastebin my asterisk backup. |
18:43.12 | Katty | it is fun. |
18:43.17 | eppigy | funzors |
18:43.39 | *** join/#asterisk wpbrown (n=wpbrown@wh-gtw-0001.woolfharris.com) |
18:43.51 | pepe | [TK]D-Fender: so what should be done - the lines #16 & #19 should be removed? I'm just newbie in asterisk world |
18:43.56 | wpbrown | hey guys |
18:44.37 | wpbrown | I have Asterisk with a Sangoma t1 card feeding it with a PRI. |
18:44.49 | wpbrown | I just got this error. Jul 1 13:41:34 pbx kernel: [652407.149878] dahdi: Disabled echo canceller because of tone (rx) on channel 2 |
18:44.54 | wpbrown | and it dropped the call. |
18:44.57 | wpbrown | any pointers? |
18:45.00 | [TK]D-Fender | pepe: You aren't calling a script |
18:45.03 | [TK]D-Fender | Katty: http://www.youtube.com/watch?v=j7r7L4eTXAI&feature=PlayList&p=8C8A7C32138FDDB6&playnext=1&playnext_from=PL&index=22 |
18:45.22 | [TK]D-Fender | Katty: on the topic of other people's songs ;) |
18:45.34 | Katty | eppigy: http://pastebin.ca/1481061 |
18:45.55 | Katty | oh god. |
18:45.56 | Katty | family guy |
18:46.14 | Katty | :> |
18:47.23 | pepe | [TK]D-Fender: from where/how should be the script called |
18:47.24 | pepe | ? |
18:47.50 | [TK]D-Fender | pepe: Apparently you didn't even read the WIKI page's samples which show you the kind of place they would put it. |
18:48.15 | Katty | eppigy: you like? |
18:48.40 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
18:49.40 | *** join/#asterisk aces1up (n=Now@ip70-173-52-152.lv.lv.cox.net) |
18:49.42 | pepe | [TK]D-Fender: I know I'm missing something byt |
18:50.11 | aces1up | would anyone have the sip firmware files for the cisco 7960 phones? |
18:50.21 | pepe | I cant see the example, I'm reading http://www.the-asterisk-book.com/unstable/bk01-toc.html and trying to understand how the scripting works |
18:50.43 | [TK]D-Fender | pepe: They GIVE YOU the damn sample that uses the script |
18:50.51 | [TK]D-Fender | pepe: Wake up and read it |
18:51.01 | Katty | aces1up: i only have polcyom stuffs :< sorry |
18:51.30 | aces1up | any idea where i can find it? i really don't want to sign up as a cisco partner |
18:51.43 | *** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
18:51.47 | Qwell | aces1up: you don't |
18:52.18 | aces1up | qwell i don't have to signup to get the firmware? |
18:52.30 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
18:52.38 | Katty | hugs [intra]lanman |
18:52.43 | Qwell | aces1up: you don't find the firmware. your only option is to follow their procedures |
18:52.54 | aces1up | hrmm. |
18:52.55 | *** join/#asterisk davevg (n=davevg-b@75.97.64.33.res-cmts.senj.ptd.net) |
18:53.04 | Katty | hi Qwell |
18:53.08 | Qwell | waves |
18:53.13 | Katty | how're you today dear |
18:54.26 | *** join/#asterisk mythicalbox_desk (n=mythical@rrcs-64-183-110-250.west.biz.rr.com) |
18:57.47 | *** join/#asterisk dfaulk3 (n=dfaulk3@nat/digium/x-8895fc6ea63f4ef3) |
19:02.54 | *** part/#asterisk errr (n=errr@fedora/errr) |
19:08.13 | pepe | [TK]D-Fender: I DID letter by letter what was on the wiki and still cant see whats wrong. |
19:08.14 | *** join/#asterisk bbryant (n=brett@74.222.102.238) |
19:08.25 | *** join/#asterisk mltlnx (n=mltlnx@mail.rssalesco.com) |
19:09.14 | [TK]D-Fender | pepe: Really? Where are you calling the AGI in your code? |
19:10.44 | [TK]D-Fender | pepe: http://pastebin.com/d512b7495 <- I see it in this exten called "callback" How the hell does any call GET THERE? |
19:11.03 | Jumpie | hey guys, in an IVR timeout, does that timer start ticking when the announcement start? or does it wait until the audio announcement file is completeD? |
19:11.06 | *** join/#asterisk mags2 (n=mags2@ampulex.whoi.edu) |
19:11.08 | [TK]D-Fender | pepe: Your 2-digit extension do not do anything with regards to starting a call-back |
19:15.32 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
19:15.51 | [TK]D-Fender | Jumpie: After audio completed |
19:18.06 | jameswf | tzafrir_laptop: do you have a SRPM for SRPM 1.4.12.9.svn.r4590-Xorcom-trunk-r7049 |
19:19.33 | *** join/#asterisk apeiron_ (i=apeiron@isuckatdomains.net) |
19:20.29 | *** join/#asterisk mltlnx (n=mltlnx@mail.rssalesco.com) |
19:20.48 | pepe | [TK]D-Fender: ok - after exten => _5[0-46-9],1,Dial(SIP/${EXTEN},30,r) should be exten => _5[0-46-9],2,AGI(callback) ?? is that what I'm missing? |
19:21.33 | Jumpie | fender, thanks |
19:21.38 | [TK]D-Fender | pepe: You aparently have no grasp on what their sample is doing. Go read it again a few hundred more times. |
19:23.43 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
19:29.11 | *** join/#asterisk aiksa[LV] (n=aiksa[LV@mx.fiveplus.lv) |
19:29.15 | aiksa[LV] | Hi everyone |
19:29.54 | aiksa[LV] | where could I find full documenation of AMI interface changes in 1.6 both - commands and events? Please dont tell me taht the only source is the source code :P |
19:33.15 | *** join/#asterisk blkry (n=blkry@64.147.222.130) |
19:43.59 | Katty | [TK]D-Fender: i started doing pushups. |
19:44.08 | Katty | [TK]D-Fender: i got to 20 before falling on my face. |
19:44.15 | Katty | [TK]D-Fender: :< |
19:44.33 | jaytee | I like to do at least 10 pushups a year just to maintain my present level of fitness |
19:44.43 | Katty | giggles |
19:44.54 | Katty | sadly, i've let myself decline horribly in the last 4 years. |
19:46.11 | Joel | bike riding 20 miles every other day is what I do |
19:46.12 | Joel | works wonders |
19:46.31 | Katty | sadly, i've no bike. |
19:46.37 | Katty | but i have jogging shoes. |
19:46.39 | Katty | and a yoga mat. |
19:46.55 | jaytee | Ommmmmmm |
19:46.55 | Katty | maybe i'll do some sitsup. |
19:47.13 | Katty | jaytee: i don't use it for yoga. just padding on the floor |
19:47.15 | Joel | vigerous masturbation was my workout method for a long time, but I apparently eat too much for that to be enough |
19:47.30 | Katty | i hear sex burns a lot of calories. |
19:47.36 | Katty | i've also been lacking in that lately too. |
19:47.38 | Katty | sadface |
19:48.15 | jaytee | I loved it in Good Will Hunting when Matt Damon lights a cigarette and Robin Williams says, "Those damn things'll kill ya!" and Matt says, "Yeah, they're really getting in the way of my yoga" |
19:48.36 | Katty | hehe |
19:49.34 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
19:49.43 | pulpster | I get only dial tone when calling from a cisco to a sip phone using chan_skinny.so - anybody knows what might be wrong - all phones are registering ok |
19:50.24 | pulpster | I dial the number and the tone is still present, during the call (I can transfer no voice payload though) |
19:50.39 | pulpster | any ideas are greatly wellcomed |
19:52.11 | pepe | [TK]D-Fender: unless you're willing to help - I'm stuck :( |
19:52.42 | [TK]D-Fender | pepe: If you can't even tell where they are calling it in their sample and which you've got an even bigger problem |
19:53.47 | [TK]D-Fender | pepe: If you can't even tell where they are calling it in their sample and *why* you've got an even bigger problem |
19:54.02 | aiksa[LV] | i have found another way to stay fit w/o exercises which bore me |
19:54.46 | aiksa[LV] | kitesurfing - a fun way to switch off my brain, and have some really good excersies |
19:55.27 | jaytee | hmmm, Karl Malden died today @ 97 |
19:55.50 | Qwell | jaytee: he was almost your age! |
19:55.59 | jaytee | LOL |
19:56.24 | Joel | karl who? |
19:58.05 | pepe | [TK]D-Fender: PLS, don't judge only because your're experienced... |
19:58.24 | [TK]D-Fender | pepe: Can you read priory steps in an exten? |
19:58.39 | [TK]D-Fender | pepe: Do they call Voicemail if the call doesn't get answered? No, they do something else. |
19:59.04 | [TK]D-Fender | pepe: This has nothing to dow ith experience. I READ the code and followed the steps it takes. You don't seem to notive the steps. |
19:59.08 | [TK]D-Fender | notice* |
20:01.00 | pa | i have a small problem with asterisk.. I have an HFC ISDN TA, and i configured it in zapata.conf. i then added an extension with my phone number as extension id, and it gets correctly invoked when i receive a call through the ISDN TA. Allright. |
20:02.14 | *** join/#asterisk mltlnx (n=mltlnx@mail.rssalesco.com) |
20:02.16 | pa | now i tried to add an extension to call regular phones through the ISDN TA . i did something like this (to test): exten => _8XX.,2,Dial(${LINEA}/${MSN1}:${EXTEN:1},60,Tr) |
20:02.22 | pa | $LINEA is Zap/g1 |
20:02.29 | pa | $MSN1 is my phone number |
20:03.15 | pa | when i try to dial 8somenumber via some sip client, i get the phone ringing sound, but in fact it doesnt work |
20:03.31 | pa | (im calling some telco number which automatically answer at once) |
20:03.41 | tzafrir_laptop | jameswf, http://updates.xorcom.com/astribank/elastix/repo/zaptel-1.4.12.9.svn.4590.xpp.r7049-1.src.rpm |
20:03.57 | pa | im not sure what im doing wrong |
20:05.30 | [TK]D-Fender | pa : Zap doesn't use MSN's |
20:05.54 | [TK]D-Fender | pa "Za/[channel or group]/numbertocall" |
20:06.03 | Qwell | Zap* |
20:06.08 | [TK]D-Fender | ^ |
20:06.38 | *** part/#asterisk boch (n=fran@200.61.191.9) |
20:07.34 | pa | ah! thanks! |
20:08.19 | pa | <PROTECTED> |
20:08.59 | [TK]D-Fender | pa / |
20:09.32 | pfn | I forget, cdr's are stored in localtime, right? |
20:09.32 | *** join/#asterisk MindTheGap (n=MindTheG@187.20.141.72) |
20:09.33 | pfn | or gmtime? |
20:09.38 | pa | thanks! i think it's correct : ) now i got answered, but i cant hear anything.. |
20:09.42 | pa | mmmh.. |
20:10.16 | Katty | mmm, walnuts |
20:10.19 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
20:10.31 | pa | my MP3Player() extension instead works fine |
20:10.34 | pa | (i can hear) |
20:11.10 | pa | probably i screwed after my dial command.. |
20:11.46 | WindowsUser | odd |
20:11.58 | WindowsUser | my wakeup calls aren't generating cdr's |
20:12.31 | WindowsUser | pfn: my Master.csv is in gmt |
20:12.56 | pa | this is my extension: http://pastebin.com/m5e84e50e |
20:13.08 | pfn | WindowsUser, thanks |
20:13.32 | pa | i think it was working back in the years when i was using some isdn4linux driver |
20:13.49 | pa | (instead of zaptel) |
20:13.49 | pfn | <PROTECTED> |
20:13.49 | pfn | <PROTECTED> |
20:13.49 | pfn | hmm, sounds like localtime |
20:15.53 | *** join/#asterisk dwschool (n=Dwayne@64.42.227.97) |
20:16.15 | dwschool | textfmt: No font metric information found for "Courier-Bold". i'm getting this error when trying to send test fax |
20:18.05 | dwschool | with hylafax or should i ask elsewhere |
20:21.12 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
20:23.42 | [TK]D-Fender | dwschool: What is generating that error? |
20:23.47 | aiksa[LV] | asterisk manager event *AgentCalled* gets raised when an Agent is called or queue member as well? |
20:24.03 | aiksa[LV] | what exactly is a scope of the Agent here? |
20:24.55 | WindowsUser | dwschool: is that an error or a warning? |
20:25.36 | aiksa[LV] | dwschool: I guess this error comes from a ghostscript |
20:25.55 | aiksa[LV] | dwschool how are you trying to send the fax? |
20:27.09 | aiksa[LV] | if you submited it to hylafax as a ps file, it shouldnt have those problems |
20:27.21 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
20:28.17 | aiksa[LV] | however if you are trying to convert it to image from some format which only identifies fonts w/o including them, that would be the prblm. |
20:30.20 | *** join/#asterisk illizit (n=cengroba@c-76-109-84-129.hsd1.fl.comcast.net) |
20:30.59 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
20:32.28 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
20:35.07 | *** join/#asterisk pcmedic (n=pcmedic@213.63.137.210) |
20:35.22 | *** join/#asterisk callenb (n=callenb@host86-168-62-181.range86-168.btcentralplus.com) |
20:35.28 | *** join/#asterisk aces1up (n=hey@76.3.153.228) |
20:37.10 | aces1up | i have a cisco 7960 connected to my asterisk box, and it says proxy unavailable but i have the right ip in the .cnf files that the cisco phone loads.. |
20:37.29 | aces1up | is there a way i can check to make sure asterisk is running correctly as in receiving SIP requests? |
20:38.12 | pcmedic | ls |
20:39.02 | *** join/#asterisk awk_r (n=awk@nat/digium/x-092d80a6f9163896) |
20:41.26 | bmoraca | aces1up: enable sip debugging |
20:44.05 | dwschool | aiksa yes i am trying to send a fax |
20:44.36 | dwschool | [TK]D-Fender: I am trying to send a fax |
20:44.47 | dwschool | WindowsUser: its an error |
20:45.54 | dwschool | aiksa[LV]: sendfax -n -d <faxnumber> <file.txt> |
20:50.44 | *** join/#asterisk aces1up (n=hey@76.3.153.228) |
20:50.59 | aces1up | dumb question, but where are the asterisk config files typically located? |
20:51.37 | dwschool | what are u running? |
20:52.55 | dwschool | /etc/asterisk |
20:53.36 | [TK]D-Fender | dwschool: Yeah, that part was very clear. And I asked you WHAT piece of software spat out that message |
20:54.43 | aces1up | is there a way to see what extensions are currently connected to asterisk? |
20:54.53 | dwschool | hylafax |
20:55.06 | dwschool | aceslup: you mean sip? |
20:56.00 | aces1up | i guess yeh, so if i have an extension configured as 100 how can i tell if the cisco phone is connecting to my box on that extension. |
20:56.27 | aiksa[LV] | dwschool: ok, got it |
20:56.34 | awk_r | aces1up: what technology are you using to connect the phone to Asterisk? if its sip then do "sip show peers" |
20:56.43 | aces1up | sip sorry |
20:56.43 | aiksa[LV] | dwschool: install ghostscript font metrics package |
20:56.46 | aces1up | ok will do. |
20:57.27 | aces1up | ok so i got this for my extension |
20:57.29 | aces1up | 100 (Unspecified) D N 0 UNKNOWN |
20:57.29 | aces1up | 1 sip peers [Monitored: 0 online, 1 offline Unmonitored: 0 online, 0 offline] |
20:57.36 | awk_r | aces1up: also, to help "extensions" is usually a reserved word meaning a dialplan extension, a "peer" is what you were trying to refer to :-) |
20:57.37 | aces1up | so 0 are online. |
20:57.52 | aces1up | where do i start troubleshooting to get it to go online? |
20:57.57 | awk_r | aces1up: also, don't mass paste, use something like pastebin |
20:57.58 | awk_r | ~pastebin |
20:57.59 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
20:58.02 | aiksa[LV] | dwschool: the point is hylafax tries to convert txt document into something sendable. for that it needs fonts to represent chars in your text file |
20:58.04 | aces1up | awk ok sorry. |
20:58.11 | awk_r | aces1up: np :-) |
20:58.15 | aces1up | it was only 2 lines so didn't think it was a prob. |
20:58.20 | aiksa[LV] | dwschool: what linux distro are you using? |
20:58.32 | dwschool | aiksa[LV]: i know, what package would it be for debian? |
20:58.50 | aces1up | awk where can i start troubleshooting as to why my phone isn't online? i am also using freepbx BTW. |
20:59.31 | aiksa[LV] | lemme consult google |
20:59.31 | dwschool | aiksa[LV]: debian |
20:59.44 | dwschool | aces1up: what phone make? |
20:59.46 | awk_r | aces1up: turn on sip debug ("sip set debug on") and see if your phone's registration is actually making it to Asterisk. If so, sift through and see if it is getting rejected |
20:59.49 | aces1up | cisco 7960 |
20:59.52 | aiksa[LV] | ghostscript-fonts perhaps |
21:00.14 | aiksa[LV] | i dont have any debian family linuxes at my disposal at the moment so cant check this |
21:00.28 | awk_r | aces1up: change that...first uninstall FreePBX and install Asterisk-GUI (lol kidding...<for those that know me>) |
21:01.50 | aiksa[LV] | dwschool: where you able to locate it? |
21:02.06 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
21:02.12 | dwschool | no - i know ghostscipt is installed |
21:02.20 | dwschool | package is gs |
21:02.36 | aces1up | hrmm ok so im getting a buncha sip messages most of em say sip/2.0 401 unauthorized |
21:02.41 | aces1up | as the header |
21:02.49 | aces1up | that can't be good i suppose. |
21:03.07 | pepe | [TK]D-Fender: exten => _5[0-46-9],1,Dial(SIP/${EXTEN},30,r) and exten => _5[0-46-9],2,Goto(s-${DIALSTATUS},1) ?? where DIALSTATUS will be BUSY when callee is busy |
21:03.52 | aiksa[LV] | hmm, the just follow this guide |
21:04.00 | aces1up | when i register a 7960 and editing the sip<mac>.cnf file is ther a authorization line i need to put in there? is this the correct place to be putting it? |
21:04.02 | aiksa[LV] | http://linux.about.com/library/howto/font/blfont5.htm |
21:04.18 | *** join/#asterisk mltlnx (n=mltlnx@mail.rssalesco.com) |
21:04.49 | aiksa[LV] | perhaps you will need to move/copy some of the existing font files and convert them or just change a path to exact location |
21:06.17 | aiksa[LV] | dwschool: try this one: gsfonts |
21:06.45 | dwschool | already installed |
21:07.31 | aiksa[LV] | strange. never the less the error states that gs is unable to locate these files. could be path issues |
21:07.35 | awk_r | aces1up: usually Asterisk sends 1 unauthorized...if it sends it twice in a conversation its bad yes. I'm not familiar with Cisco phones, so someone else will have to chime in |
21:08.43 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
21:09.06 | aiksa[LV] | dwschool: enter #gs --help and pastebin the output |
21:10.29 | aiksa[LV] | in that output you should see a folder in which gs is looking for fonts |
21:10.33 | dwschool | http://pastebin.com/m4d164aee |
21:10.58 | dwschool | i see it and its there when i browse for it |
21:12.04 | dwschool | aces1up: is this the first cisco phone u are setting up? |
21:12.09 | aiksa[LV] | you have a number of pfb and afm files there? |
21:12.17 | aiksa[LV] | in one of those folders i mean? |
21:12.42 | aces1up | dwschool yes |
21:13.11 | dwschool | is the [device] and username in sip.conf the same? |
21:13.52 | dwschool | aiksa[LV]: yes tons |
21:14.21 | rudeboy_xix | when using the AMI of asterisk, and then i execute a command like 'Originate' then the other party answered, can i use extenspy on it? is it possible? |
21:14.22 | aces1up | dwschoool, well i am using freepbx to configure the extensions. |
21:14.31 | aces1up | and i don't see in what sip.conf file it is defined., |
21:15.21 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:b01e:f686:4124:ac1d) |
21:15.26 | dwschool | aces1up: there are three file containing sip in /etc/asterisk open them and check. I always have to go back and redit the sip.conf |
21:15.28 | aces1up | is there a way in asterisk cli to see what sip extensions are registered but not connected? |
21:15.40 | dwschool | sip show peers |
21:16.04 | dwschool | if address is unspecified, its not registered |
21:16.11 | aces1up | so when it show extension 100 in that list... that means it is configured somewhere is some .conf file/ |
21:16.19 | dwschool | yes |
21:16.33 | dwschool | does address say unspecified? |
21:16.43 | aces1up | yes |
21:16.46 | rudeboy_xix | can anyone answer me? |
21:17.05 | aces1up | i dunno i have the secret and everything matching in the sip<mac>.cnf file as well. |
21:17.30 | dwschool | ok, go to sli - set verbose to 10 and see if your phone is trying to auth |
21:17.48 | dwschool | cli i mean |
21:18.05 | Katty | [TK]D-Fender: :< |
21:18.12 | Katty | [TK]D-Fender: my sister was considering taking lipozene. |
21:18.22 | Katty | [TK]D-Fender: my family is a bunch of dingbats. |
21:18.45 | Katty | [TK]D-Fender: i told her to take the money she was going to spend on lipozene, and go see a doctor. |
21:18.58 | aces1up | dwschool it is. |
21:19.15 | dwschool | find it in sip.conf |
21:19.17 | aiksa[LV] | dwschool: - then sorry |
21:19.41 | aiksa[LV] | I dont have any idea why it would not find font metrics file |
21:20.22 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
21:20.27 | aiksa[LV] | thats it - I am leaving now. it is over midnight here already |
21:20.36 | aiksa[LV] | dwschool: good luck! |
21:20.44 | dwschool | thanks |
21:21.44 | aiksa[LV] | dwschool: just last thing |
21:21.50 | aiksa[LV] | in those dirs |
21:21.56 | aiksa[LV] | there should be two additional files |
21:21.58 | Katty | file: OH EM GEE |
21:22.12 | Katty | file: so pretty :> |
21:22.25 | aiksa[LV] | fonts.dir && fonts.scale |
21:22.28 | Katty | file: will there be cats, or dogs? |
21:22.41 | aiksa[LV] | if i am not mistaken fonts.dir acts as a roadmap |
21:22.55 | aiksa[LV] | links specific font name to a pfb file |
21:23.02 | aiksa[LV] | check if you have that file |
21:23.26 | aiksa[LV] | and if it lists the Courier-Bol in there |
21:23.44 | [TK]D-Fender | Katty: Agree... Wingbats...... |
21:23.50 | dwschool | dont have both |
21:24.04 | dwschool | aiksa[LV]: don't have both |
21:24.49 | Katty | [TK]D-Fender: you see file's new house? |
21:24.59 | [TK]D-Fender | Katty: Nope.... |
21:25.06 | Katty | [TK]D-Fender: you still on facebook? |
21:25.40 | [TK]D-Fender | Katty: Extrememly rarely, and I remove almost all material about myself.... Facebook = EVIL |
21:25.50 | *** join/#asterisk pha3drs (i=nobody@static-96-254-70-2.tampfl.fios.verizon.net) |
21:26.01 | [TK]D-Fender | kicks Novation's MIDI implementation in the ndas |
21:26.06 | [TK]D-Fender | nads* |
21:26.16 | aiksa[LV] | dwschool: cd into that directory |
21:26.42 | [TK]D-Fender | kicks Roland's MIDI implementation in the nads for not letting him cheat with SysEx either |
21:26.46 | pulpster | hello |
21:26.49 | aiksa[LV] | and perform #mkfontdir |
21:26.50 | pulpster | please help |
21:26.55 | pulpster | my problem is here: http://pastebin.com/m268ac740 |
21:27.09 | Katty | [TK]D-Fender: well you should go have a look anyway. |
21:27.14 | pulpster | I have tried anything can't seem to figure out what I do wrong |
21:27.17 | aiksa[LV] | dwschool: documentation says it should rebuild the fonts.dir and fonts.scale |
21:27.49 | dwschool | ok,ok |
21:28.14 | aiksa[LV] | I`ll hang in to see if you suceed |
21:28.15 | pha3drs | Hello... I am using Trixbox v2.6.1.10 - I was wondering what controls my ability to call Canada, as everytime I try to call any number in CA I get a busy.... |
21:28.15 | aiksa[LV] | :) |
21:28.45 | aiksa[LV] | pha3drs: your telco controlls your ability to call Canada |
21:29.09 | *** join/#asterisk intralanman (n=lanman@70-10-178-143.pools.spcsdns.net) |
21:29.19 | Katty | hugs intralanman |
21:29.24 | dwschool | ok, i fixed the path in hyla.conf was incorrect |
21:29.24 | aiksa[LV] | dwschool: did it work? |
21:29.29 | pha3drs | right, I called Voicepulse and they didn't see anything in the logs when I dialed |
21:29.41 | dwschool | but u pointed me in the right direction - thanks |
21:29.55 | aiksa[LV] | dwschool: take care |
21:30.01 | dwschool | :D |
21:30.09 | aiksa[LV] | missing font packs is a traditional issue with slackware |
21:30.31 | aiksa[LV] | they doesnt have this as dependancy for GS, so everytime I setup another hylafax - oh the fun ... |
21:30.41 | pha3drs | Calling any number in the US works, but not CA and voicepulse said that it was on my end, is that not the case? |
21:30.53 | aiksa[LV] | pha3drs: check your logs |
21:30.59 | pha3drs | ok |
21:31.03 | aiksa[LV] | see how call gets procesed |
21:31.09 | aiksa[LV] | and if it even leaves your server |
21:31.15 | aiksa[LV] | thats it for now |
21:31.18 | aiksa[LV] | end of show |
21:32.00 | *** join/#asterisk [acer]lanman (n=lanman@70-10-178-143.pools.spcsdns.net) |
21:32.42 | pulpster | anybody configured cisco phones + asterisk - need someone with experience to ask her/him some questions |
21:32.45 | pulpster | please help |
21:33.05 | pulpster | I will soon have no hair left on my head |
21:33.20 | pha3drs | here is the log: http://pastesite.com/8449 |
21:33.21 | dwschool | pulpster - whats the problem? |
21:33.26 | pulpster | http://pastebin.com/m268ac740 |
21:33.32 | dwschool | i have quite a few - what model? |
21:33.36 | pulpster | 7911 |
21:33.39 | pulpster | 7911G |
21:33.59 | pulpster | I have tryed both chan_sccp.so and chan_skinny.so |
21:34.07 | pulpster | I cannot send voice payload |
21:34.13 | pulpster | this is wher I am stuck |
21:34.15 | dwschool | well i iuse sip |
21:34.32 | pulpster | you've changed the firmware of your phone to sip ? |
21:34.56 | pulpster | I cannot change anything in my phone.. |
21:35.05 | *** join/#asterisk errr (n=errr@fedora/errr) |
21:35.08 | pulpster | I'll just have to use it as it is I guess |
21:35.29 | pulpster | ...or tear it appart..eventually |
21:36.05 | *** join/#asterisk mltlnx (n=mltlnx@mail.rssalesco.com) |
21:36.40 | dwschool | i've decided to switch to mitel phones - i like them way better |
21:37.08 | pulpster | I will switch to any other vendor next after I finish this job configuring the cisco |
21:38.20 | pulpster | do you guys have any ideas, what might cause this behavoiur ? |
21:39.24 | pulpster | am I stupid and do smtg obviously wrong ? or is it because skinny.so / sccp.so are not well suited to my 7911 phone ? |
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21:42.12 | Katty | eppigy: hungry :< |
21:44.00 | pulpster | do you know anyone who might solve my problem ? |
21:44.02 | *** join/#asterisk watchy (n=dolphinr@76.196.98.139) |
21:44.07 | watchy | hi |
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21:44.27 | pulpster | an email / site or a developer from sccp who might know these stuffs ? |
21:44.57 | watchy | whats a easy way to send a cell SMS messages from asterisk? |
21:45.42 | WindowsUser | watchy: via a phone over bluetooth or via an email gateway? |
21:48.31 | watchy | well a customer wants a text sent to his cell everytime he gets a new voicemail on his landline |
21:49.01 | dwschool | i am getting hylafax error no local dialtone |
21:49.31 | IBC_jkenney | watcy thats easy |
21:49.36 | IBC_jkenney | watchy thats easy |
21:49.48 | IBC_jkenney | why send sms get his cell phone e-mail address |
21:50.03 | IBC_jkenney | most cell providers let you send number@carrier. and it goes right to the phone as sms |
21:50.06 | IBC_jkenney | i use it all the time |
21:50.49 | IBC_jkenney | you can do it in voicemail.conf add pager=emailaddress |
21:50.54 | IBC_jkenney | to the line and you should be all set |
21:51.23 | watchy | oh |
21:51.25 | watchy | good call |
21:52.04 | IBC_jkenney | want an example? |
21:52.21 | watchy | nah i just never thought about that now that you brought it up |
21:52.31 | watchy | thats a very easy way without actually sending an SMS |
21:52.54 | IBC_jkenney | Yeah i fiddled with it for myself |
21:53.01 | watchy | did it work well? |
21:53.38 | IBC_jkenney | Sure did if you have a smart phone with windows on it you can also attach the VM as a wave and send it to the phone |
21:54.50 | watchy | wow |
21:55.48 | *** join/#asterisk Dr1 (n=AXON@pool-71-105-96-98.lsanca.dsl-w.verizon.net) |
21:56.07 | Dr1 | are there any free ITSPs? |
21:56.30 | watchy | i guess it wont work with a iphone? |
21:56.43 | IBC_jkenney | yeah it will work with a iphone |
21:56.50 | IBC_jkenney | as long as you can open a wav file |
21:57.02 | IBC_jkenney | make sure to specify in voicemail.conf wav not wav49 |
21:57.11 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
21:57.13 | watchy | well you can't send MMS to a iphone |
21:57.34 | IBC_jkenney | but you can send e-mails to the I phone |
21:58.39 | watchy | they arent allowed to use email |
21:58.47 | watchy | thats why they need it to be an SMS |
22:01.31 | WindowsUser | watchy: cellphone companies usually have a email to txt gateway |
22:01.44 | IBC_jkenney | that was what i said |
22:01.46 | watchy | yea, thats what i m gonna have to use |
22:01.54 | watchy | but i cant attach a wav to that |
22:02.09 | IBC_jkenney | the pager= will only send the text of the email |
22:03.32 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
22:03.32 | *** mode/#asterisk [+o lmadsen] by ChanServ |
22:03.49 | watchy | yea thats what i need |
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22:23.02 | Shazaum | any doc srtp in asterisk 1.4 |
22:23.02 | Shazaum | ? |
22:24.17 | leifmadsen | Shazaum: just the issue open in the issues tracker |
22:25.44 | *** part/#asterisk squish102 (n=squish10@cpe-075-181-098-059.carolina.res.rr.com) |
22:25.54 | Shazaum | leifmadsen, so |
22:26.06 | Shazaum | leifmadsen, srtp only in asterisk 1.6? right? |
22:26.20 | leifmadsen | Shazaum: yes, 1.6.3, or whenever it gets merged |
22:26.27 | leifmadsen | it has not been merged -- see the issue |
22:28.26 | Shazaum | leifmadsen, ok, tks 4ur attention |
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23:03.14 | galeras | little help: why call is not starting after mv sendfax.call /var/spool/asterisk/outgoing/ |
23:05.09 | drmessano | galeras: I just looked in /var/spool/asterisk/outgoing/ and it's empty |
23:06.09 | *** join/#asterisk securevoip (n=securevo@c-76-123-20-170.hsd1.va.comcast.net) |
23:06.28 | galeras | drmessano: mine is not empty |
23:07.36 | drmessano | galeras: I wouldn't know.. you've shown us NOTHING yet |
23:09.38 | galeras | never mind, i missed autoload=yes in modules.conf |
23:13.39 | *** join/#asterisk lasko (n=verilan@66.178.162.34) |
23:14.09 | lasko | Could any tell me why I would be getting "SIP response 488 - Not Acceptable Here" when I'm making a call? |
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23:17.39 | jaytee | lasko, is it all calls or just some? |
23:20.11 | [TK]D-Fender | lask488 = codec mismatch |
23:20.17 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
23:20.28 | [TK]D-Fender | Darn, missed the exit |
23:22.26 | jaytee | same here |
23:40.59 | Dr1 | Is there a free ITSP that you can use to test your asterisk VoIP connection? |
23:43.18 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
23:44.27 | securevoip | http://www.sipgate.com/ |
23:44.58 | securevoip | OR http://www.ipcomms.net/html/freedid.html |
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23:51.04 | [TK]D-Fender | Dr1: www.ekiga.net |
23:53.24 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
23:57.21 | Dr1 | If I'm reading this right, it looks like ipcomms is free dial-in only. Do SIPGate and ekiga both do free inbound and outbound? What are the limitations (cap on minutes per day, etc)? The Quality? |