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| 01:21.31 | fornax | hi, is somewhere there that already added a cisco phone via sccp to asterisk? |
| 01:22.23 | fornax | I have trouble configuring my 7914 expansion module for my 7960 but cannot find the right documents |
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| 03:55.57 | volga629 | When web meet me installed where I can find info what need to be added to dial plan |
| 03:56.25 | volga629 | to create dynamic conference room |
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| 05:15.24 | Hyphenex | hey everyone. I've not had the best of luck with the cards to put in the computers, so I'm just wondering what those boxes are that terminate a PSTN line to SIP just with an Ethernet port |
| 05:18.57 | carrar | ATA |
| 05:19.28 | Hyphenex | do they work ok? Is there a recommended one I can get and hopefully not have much/any echo on it? |
| 05:19.50 | carrar | AudioCodes makes nice ones |
| 05:20.44 | Hyphenex | Thanks carrar, I'll check them out. Any models I should be looking for? |
| 05:21.28 | carrar | http://www.audiocodes.com/products/mediapack-1xx# |
| 05:22.07 | volga629 | Is web meet me should insert conference number in asterisk config ? |
| 05:25.28 | p3nguin | I have a feeling that no one here knows what web meet me is. |
| 05:25.51 | carrar | double checks the channel name |
| 05:26.27 | volga629 | yes exactly I will glad do some reading and understand how it should work |
| 05:26.40 | volga629 | the right way |
| 05:27.00 | carrar | ~book |
| 05:27.01 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 05:27.04 | carrar | start there |
| 05:30.04 | volga629 | thanks, my pbx is running but web meet me just can't get How exactly it should work what web meet me doing when create the conference in web gui |
| 05:33.12 | carrar | might talk to the people who wrote web meetme |
| 05:33.45 | volga629 | I have web met me setup with mysql, web interface running no issue, but interaction between we meetme and asterisk ? |
| 05:33.54 | carrar | might talk to the people who wrote web meetme |
| 05:39.05 | p3nguin | Is assembly considered a programming language? |
| 05:39.34 | carrar | yes |
| 05:39.51 | carrar | low level |
| 05:39.53 | carrar | but yes |
| 05:40.06 | carrar | I was a mad 6502 assembly programmer |
| 05:40.17 | p3nguin | Can AGI be written in assembly? |
| 05:40.34 | alucardx-matt | there's a 6502 in the Atari 2600 isn't there? |
| 05:40.36 | carrar | be written in anything |
| 05:40.57 | carrar | alucardx-matt, I believe so |
| 05:41.02 | carrar | I learned on the Atari 800 |
| 05:41.31 | alucardx-matt | which is very similar to the atari 5200 |
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| 05:46.22 | p3nguin | I think the correct answer was that assembly is not a programming language... it is machine code and is needed to run code written in programming languages. |
| 05:46.25 | Hyphenex | carrar: The AudioCodes are pretty expensive, is there a single FXO model out there that is known to have little echo? |
| 05:51.36 | p3nguin | I have pretty good luck with an SPA-3102. |
| 05:52.32 | Hyphenex | p3nguin: I'm looking at that, about $60 and not any echo from the forums. Can I use the FXO port so when my home phone number calls, I can divert it over SIP? I don't think I'd have any use for the FXS port |
| 05:53.36 | p3nguin | Yes. You can use the FXS to connect a phone to it, the FXO to connect your telco's line to it, or both. |
| 05:54.06 | p3nguin | It can also serve as a router (internet gateway). |
| 05:54.36 | Hyphenex | sweet, might just get one then :) Have you found there's any echo on the FXO port? (I'll be using this for all inbound calls to my SIP phones) |
| 06:00.23 | ectospasm | ctrl-M /input return |
| 06:00.56 | ectospasm | Hyphenex: there are many factors that determine whether you have echo, not just the FXO module... |
| 06:01.36 | Hyphenex | ectospasm: If I don't have any echo on my normal phone, can't I assume I could get a module to replace the phone and I shouldn't get any echo? |
| 06:02.15 | ectospasm | no, that's not a safe assumption. |
| 06:02.45 | Hyphenex | darn, what do they do differently? :P |
| 06:02.53 | ectospasm | the connection could be totally different, even though not functionally different |
| 06:03.14 | ectospasm | Hyphenex: different hardware involved |
| 06:03.47 | Hyphenex | so it's not the case of just buy expensive enough equiptment that should have the smarts to just "deal with it"? |
| 06:03.51 | ectospasm | it can be impossible to tell if you'll have echo when you connect an FXO port to a POTS line, until you actually plug it in and test. |
| 06:04.09 | ectospasm | Hyphenex: well, if it comes with an echo canceler, that will help... |
| 06:05.01 | ectospasm | the software echo cancelers that ship with Asterisk may work well enough for you... |
| 06:06.21 | ectospasm | and with that, I must go to bed... |
| 06:08.48 | Hyphenex | Thanks ectospasm |
| 06:13.18 | p3nguin | Using an ATA, you won't be using Asterisk's echo cancellation. |
| 06:13.53 | Hyphenex | yeah I was thinking that. but I wish there was something you'd know would just work |
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| 06:40.45 | doolittlework | moning people |
| 06:41.42 | doolittlework | can anyone poin tme in the right direction to reset the web admin password for a snom 812? |
| 06:42.30 | doolittlework | sorry 821 |
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| 09:45.02 | bitwize | Question: If I have a couple of managers connected through AMI and all of them listening for originate-response events, do all managers get the event when it occurr or just the manager who initiated the originate-action? |
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| 10:34.53 | AliRezaTaleghani | I don't know why, or how to debug it... |
| 10:35.20 | AliRezaTaleghani | but suddenly, my "ReadExten" Application had stopped to work |
| 10:35.52 | AliRezaTaleghani | all the server is ok, and all other Dialplan parts are working properly.... |
| 10:36.11 | AliRezaTaleghani | but as the dialplan, reach to "ReadExten" application, it will stop! |
| 10:36.50 | AliRezaTaleghani | ** I test the Media file, with "Read"and "PlayBack"... both are work as usuall |
| 10:37.16 | AliRezaTaleghani | but ReadExten dosen't work and just go to silent... |
| 10:39.59 | bitwize | Does the ReadExten-application catch the DTMF-signal and lets you go further in the dialplan? |
| 10:41.10 | bitwize | If not, how are you transmitting the DTMF from the phone? Inband, 2833 or both? |
| 11:21.27 | AliRezaTaleghani | bitwize: the mather is that, i am not able to hear any thing on, caller Channel to mention the call he should input digits... |
| 11:21.37 | AliRezaTaleghani | bitwize: all just go silent.. |
| 11:22.52 | AliRezaTaleghani | ofcourse if the caller input anything... they will not be catch in dialplan... |
| 11:23.25 | AliRezaTaleghani | after a while, channel will time out and go to 't' (timeout) extention... |
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| 12:20.25 | bitwize | AliRezaTaleghani: ok, have you looked at the output in logfiles/console? Any warnings/errors? |
| 12:20.54 | bitwize | enable loglevels and output targets in logger.conf |
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| 13:15.22 | AliRezaTaleghani | bitwize: i had changed to verbose mode 15 and nothing special has been catched |
| 13:16.52 | bitwize | AliRezaTaleghani: ok, have you enabled errors and warnings to be printet in console? |
| 13:17.47 | bitwize | look in logger.conf, you should specify something like: console => error,warn,dtmf etc.... |
| 13:18.37 | bitwize | get the correct syntax at voip-info.org |
| 13:27.59 | AliRezaTaleghani | bitwize: yep.. i had done them all and ofcourse enable debug level for the calls which fail on problem |
| 13:28.07 | AliRezaTaleghani | just 10 min ago i found the problem... |
| 13:28.30 | AliRezaTaleghani | i the matter was that we disable the module pthream timming |
| 13:28.36 | AliRezaTaleghani | and it was the reason |
| 13:28.40 | AliRezaTaleghani | :) |
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| 14:27.39 | euphor][a | hi guys |
| 14:29.42 | euphor][a | Thinking of virtualising asterisk by creating a vm. Currently I have two asterisk servers with 4 EuroISDN circuits in each. Does it sound reaosnable to terminate the circuits using some hardware, delivering the calls to asterisk via IAX2, and if so, what hardware might do this? Thanks |
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| 14:48.27 | WIMPy | You want to virtualize two boxes and additinally replace them bu two others? |
| 14:50.13 | WIMPy | And protocoll conversions are always lossy. |
| 14:54.00 | euphor][a | I have two boxes I want to replace with two virtual machines |
| 14:54.18 | euphor][a | and some hardware, not a full server, to terminate PRI lines and connect them to the VM |
| 14:54.32 | WIMPy | And two additional pieces of hardware. |
| 14:54.49 | euphor][a | whatever hardware is needed |
| 14:55.11 | WIMPy | You could go to xorcom and use USB. |
| 14:55.33 | euphor][a | Something like this http://www.beronet.com/product/berofix-gateways/#1 ? |
| 14:55.48 | euphor][a | ok let me look at Xorcom |
| 14:55.49 | WIMPy | YOu could use gateways, but I woudn't recommend that. |
| 14:56.11 | WIMPy | Or use just another Asterisk as gateway. |
| 14:56.58 | euphor][a | I could use another asterisk box, but then I might as well have that instead of the virtual machine |
| 14:57.13 | WIMPy | yes |
| 14:57.19 | euphor][a | trying to use the most basic equipment I can to do the job rather than a custom asterisk server |
| 14:57.34 | euphor][a | only throwing the idea out there |
| 14:57.38 | euphor][a | may be it isn't feasible |
| 14:57.59 | WIMPy | It's surely possible. |
| 14:58.11 | WIMPy | You have to devide if it makes sense to you. |
| 14:58.41 | euphor][a | looing to virtualise my current environment, and would like to virtualise the asterisk servers too |
| 14:58.55 | euphor][a | but they are core to the business, so it needs to be a viable solution |
| 14:59.28 | euphor][a | You're think Xorcom Astribank? |
| 14:59.44 | WIMPy | Is it two servers at different locations? |
| 14:59.56 | euphor][a | no |
| 15:00.25 | euphor][a | just two servers for redundancy |
| 15:00.29 | euphor][a | at one location |
| 15:00.35 | WIMPy | Ah |
| 15:03.23 | euphor][a | astribank looks good, although the USB would only be connected to one server, providing a single point of failure |
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| 15:04.09 | WIMPy | Yes, if you want redundancy, you need two of them. |
| 15:04.18 | euphor][a | aah, I see it has two USB ports, one for backup |
| 15:04.28 | WIMPy | So I guess you've got two PCI cards now? |
| 15:04.40 | WIMPy | Still a SPOF. |
| 15:05.12 | euphor][a | yes, and yes, still SPOF with current setup |
| 15:05.17 | euphor][a | but trying to eliminate with new one |
| 15:05.42 | euphor][a | virtual machine enviromnent has no SPOF from end to end |
| 15:05.49 | WIMPy | So it;s going to be an expensive virtualisation. |
| 15:05.56 | euphor][a | yes |
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| 15:06.22 | euphor][a | 3 servers, 2 switches, 1 SAN |
| 15:07.13 | WIMPy | Could be interesting to connect a pice of hardware anyway. |
| 15:09.22 | euphor][a | yep |
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| 16:02.06 | luckman212 | Is there any way to have Asterisk start playing back Voicemail messages from the NEWEST message to the oldest? |
| 16:09.32 | *** join/#asterisk [ProB]CrazyMan (~chatzilla@mx40.roterschnee.com) |
| 16:13.10 | [ProB]CrazyMan | hi, I have an short question, if I have two asterisk boxes and want to test dahdi communication threw some BRI cards, and one card is signalling = bri_net, what signalling do the receiving side should have? bri_cpe? |
| 16:14.03 | WIMPy | yes |
| 16:15.29 | [ProB]CrazyMan | and if i have on one site bri_net_ptmp I should have on the other side bri_cpe_ptmp? what happens if i have this combination: bri_net --> bri_cpe_ptmp ? |
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| 16:15.56 | WIMPy | You won;t be able to obtain a TEI. |
| 16:16.25 | WIMPy | The other way round will probably work partially. |
| 16:17.18 | WIMPy | Err, and yes to the first question. |
| 16:17.29 | [ProB]CrazyMan | so next question if I run dahdi_genconf, how does it detect the the modes ? |
| 16:18.37 | WIMPy | It doesn't |
| 16:19.16 | [ProB]CrazyMan | ok thx |
| 16:19.20 | WIMPy | Well, I think the latest version detects the TE/NT jumpers, but ptp/ptmp can't be auto detected. |
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| 16:37.22 | timbreck | hi, i'm reading conference quota information from mysql (using func_odbc), and am trying to compare it to the result of MeetmeCount : exten -> _ZXXXX,n,GotoIf($[${CONF_CNT} >= ${CONF_QUOTA}]?i,1) |
| 16:37.22 | luckman212 | sorry to repeat but, nobody knows if it's possible to have Ast playback voicemail starting with the newest messages first? |
| 16:37.39 | timbreck | doesn't seem to work, i'm guessing because func_odbc returns a string ? |
| 16:38.03 | timbreck | or am I writing things wronf? |
| 16:38.08 | timbreck | wrong* |
| 16:38.23 | WIMPy | luckman212: You could always do your own VM. |
| 16:39.19 | luckman212 | WIMPy: I could? |
| 16:40.41 | timbreck | is there anything blatantly wrong about my snippet? |
| 16:41.13 | WIMPy | luckman212: Wee, I don;t know if you could, but it's vertainly posible. |
| 16:41.32 | luckman212 | WIMPy: heh. that's more like it |
| 16:42.04 | WIMPy | And obviousely I can't use a keyboard. |
| 16:42.14 | luckman212 | WIMPy: would probably be easier for me to dig through the c code and change the existing behavior to a LIFO |
| 16:42.44 | WIMPy | That's another possibility |
| 16:43.12 | timbreck | ah sh*t yes, -> instead of => |
| 16:43.17 | timbreck | so sorry .. :p |
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| 17:13.54 | naif | I'm selling OpenBTS hardware http://pastebin.com/sbu86z8d - anyone interested? |
| 17:14.37 | fenrus | that would be so awesome |
| 17:15.02 | naif | i used it for experiments, it's working, but now i'm not using it anymore |
| 17:15.13 | naif | it works with asterisk to make GSM 900mhz network |
| 17:19.19 | fenrus | this would have been so awesome to build for a scout centre in the archipelago |
| 17:20.01 | WIMPy | It's hard to get a test licence for 900. |
| 17:20.22 | fenrus | yea, pretty much what i recon |
| 17:20.42 | fenrus | well see what the swedish goverment says :) |
| 17:27.26 | naif | i just didn't care about the license |
| 17:27.40 | naif | as long as you don't bother other GSM users by using your own MCC/MNC |
| 17:28.01 | naif | and filter your allowed IMSI-only |
| 17:28.21 | naif | it may still be "slightly illegal" but for sure will not disturb anyone |
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| 17:30.18 | WIMPy | I've been told the transmission itself is no longer illegal here. But it you accidentally disturb something you're not only an official terrorist but also liable for damages. |
| 17:31.42 | WIMPy | A test licence isn't expensive, but hard to get for 900. |
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| 17:36.58 | twanny796 | exten => s,1,Dial(SIP/asterisk1/1001,30 & SIP/asterisk1/1000,30) ... I'm doing this to call two stations at the same time, but I don't think it's correct,?? |
| 17:37.37 | WIMPy | No, don't repeat timeout and options. |
| 17:38.03 | WIMPy | (dest1&dest2&dest3,time,options) |
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| 17:52.34 | twanny796 | WIMPy: ok, ;) |
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| 22:42.42 | Rene | hi all! |
| 22:43.44 | Rene | is anyone using the findayly+fincheck scripts to retreive their credit-balance from the voipbuster/voipdiscount servers? |
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| 22:56.14 | fornax | i have two cisco 7914 on my 7960 and my second does not show any entries. does anyone has the same setup and can help me? |
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| 23:18.00 | ChannelZ | I don't even understand the question |
| 23:24.37 | fornax | ChannelZ: The CISCO 7960 has the ability to add to expansion modules 7914. These can be configured using the /etc/asterisk/sccp.conf. You can add speed dial, lines etc as buttons that should show up on both devices. The configuration works so that the buttons are set and the lcd shows the lines on the first 7914 but the second does nothing. So my question is if someone else already got two 7914 working with a cisco 7960? |