| 00:06.47 | *** join/#asterisk ryang (sid10904@gateway/web/irccloud.com/x-ggxvvatcoezcvyot) |
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| 00:35.20 | prelude2004c | hey, anyone know if IAX supports TCP ? |
| 00:35.29 | prelude2004c | probably UDP only right? |
| 00:35.53 | prelude2004c | problem i am having with ios, if you close or minimize a softphone that runs on UDP , you can't get inbound calls :( |
| 00:36.27 | WIMPy | Realtime and TCP don't mix. |
| 00:37.00 | WIMPy | Err. That's an interesting "feature". |
| 00:37.40 | WIMPy | So what was the result of your Echo() test? |
| 00:38.11 | prelude2004c | didn't need it :) channelZ pointed me in the right direction |
| 00:38.16 | prelude2004c | tried a new phone and voala !! |
| 00:38.21 | prelude2004c | stupid app :( |
| 00:38.27 | prelude2004c | wasted like 2 hours |
| 00:39.00 | WIMPy | So you used another app and that fixed it? |
| 00:39.08 | prelude2004c | well the issue i am having now is... the call is good, most of the time works well... only draw back now is that while you are talking sometimes it looses data and i hear a bit of a cut out |
| 00:39.17 | prelude2004c | another phone |
| 00:39.27 | prelude2004c | went to S5 and it fixed it |
| 00:39.38 | prelude2004c | problem must be with the nubia Z9 |
| 00:40.13 | prelude2004c | so anyone have any ideas how to better improve relability of the stream |
| 00:40.25 | prelude2004c | the reason i wanted tcp is because if it can't get some packets, it would be nice if it retransmitted it |
| 00:40.29 | WIMPy | Fix the network. |
| 00:40.35 | prelude2004c | i know it uses a bit more bw but if packets are not lost.. that would be awsome |
| 00:40.36 | WIMPy | There's no other way. |
| 00:40.42 | prelude2004c | can't fix the network... its on the other side of the world |
| 00:40.51 | prelude2004c | server is in china and i am in Canada |
| 00:41.00 | WIMPy | Then don't use it for VOIP. |
| 00:41.10 | prelude2004c | but it is for voip :) |
| 00:41.19 | prelude2004c | how do cell phone companies do it |
| 00:41.27 | prelude2004c | the quality on my cell phone is soooo much better |
| 00:41.29 | WIMPy | And BTW: Yest IAX can set up a direct link betwenn the endpoints. |
| 00:41.31 | prelude2004c | even when i am on VOIP using wifi |
| 00:41.38 | WIMPy | They don't use the internet. |
| 00:42.06 | prelude2004c | yes i know they dont' use the internet but..... geez if you have the bw , it should not be a problem |
| 00:42.09 | WIMPy | If you use wifi you can run in to the same trouble if others use it as well. |
| 00:42.38 | WIMPy | Yes, and you obviousely don't. |
| 00:42.49 | prelude2004c | :) |
| 00:43.28 | prelude2004c | hey , iax supports media from one user to another direct don't they? |
| 00:43.38 | WIMPy | yes |
| 00:43.40 | prelude2004c | like when the call RTP takes place.. doesn it go from custoemr A > B ? |
| 00:43.57 | prelude2004c | eg.. registraiton is in china but... if 2 are in Canada, it should only go back to initiate the call |
| 00:44.02 | WIMPy | There's no RTP. |
| 00:44.12 | WIMPy | It can. |
| 00:44.16 | prelude2004c | so do both customers in canada have to go all the way to china ? |
| 00:44.27 | prelude2004c | to chat to eachother because i tested on locally and its still cutting out |
| 00:44.32 | prelude2004c | so clearly the packets are traveling |
| 00:44.43 | WIMPy | It has to be enabled and your dialplan must not make use of DTMP in the bridged call. |
| 00:45.17 | prelude2004c | how do you enable that ? isn't it the notransfer=no |
| 00:45.50 | WIMPy | transfer=yes |
| 00:46.24 | WIMPy | And if you need CDRs, there's transfer=mediaonly, but that seems to be broken. :-( |
| 00:46.28 | prelude2004c | isn't transfer=yes for transfering calls capabiltiy |
| 00:46.48 | WIMPy | Nope. |
| 00:46.55 | WIMPy | Not in IAX. |
| 00:47.20 | prelude2004c | ahh i had taken that out becuase i was worried about the security |
| 00:47.24 | prelude2004c | :) i am normally working in SIP |
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| 00:51.19 | prelude2004c | so media only doesn't work :( |
| 00:51.24 | prelude2004c | does it still connect with it? |
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| 00:52.10 | WIMPy | The CDR part doesn't work. |
| 00:52.46 | prelude2004c | oh that.. |
| 00:52.58 | prelude2004c | it's ok... just for testing right now anyways :) |
| 00:53.08 | prelude2004c | i can have someone fix the CDR if i decide to go with it |
| 00:53.28 | prelude2004c | someone working on it ? |
| 00:55.01 | WIMPy | I don't think so. |
| 00:58.31 | prelude2004c | k, if i proceed i will get it fixed and will send out the patch |
| 00:59.09 | prelude2004c | still trying to decide the best case scenario for long distance ( over 300 - 500ms ) latency and low BW |
| 00:59.10 | WIMPy | That would be usefull. |
| 00:59.10 | prelude2004c | :( |
| 00:59.18 | prelude2004c | there should be a good way to do it |
| 00:59.21 | prelude2004c | or at least the best way to do it |
| 00:59.38 | WIMPy | Last time I go stuck in trying out when it fails. - It seams to always fail if the transfer succeeds. |
| 01:00.57 | prelude2004c | hey, i am using speex |
| 01:01.02 | prelude2004c | do you think that is the best codec ? |
| 01:01.08 | prelude2004c | low bw and high latency |
| 01:07.48 | *** join/#asterisk bkruse (~Adium@user-24-96-51-167.knology.net) |
| 01:08.03 | prelude2004c | hey speex Wideband seems to do variable |
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| 01:17.18 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.4.0 (2015/06/04), 11.18.0 (2015/06/04), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 06:09.15 | branden2 | Hello? I'm new to irc. and I have a question about fpbx.. |
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| 07:06.53 | ChannelZ | #freepbx |
| 07:12.07 | janicez | ChannelZ: hi |
| 07:12.15 | ChannelZ | ahoy |
| 07:12.48 | janicez | ChannelZ: Are you interested in a casual chat with some random user of an Asterisk competitor? :P |
| 07:13.47 | ChannelZ | not really. You're the Freeswitch evangelist yeah? |
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| 07:19.16 | ChannelZ | shrugs |
| 07:27.28 | janicez | ChannelZ: I was the yate evangelist lately |
| 07:28.26 | ChannelZ | oh right. I couldn't remember. (I actually didn't even remember yate at all) |
| 07:28.34 | janicez | ChannelZ: then i came back to *, then after deadlock-after-deadlock I moved to anthm's rewrite |
| 07:30.00 | janicez | Also fs has support for skype, meaning now i run * and fs in tandem |
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| 07:31.16 | janicez | ChannelZ: I also use kamailio |
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| 09:20.10 | ruied | Hello. I am trying to make a system with asterisk to make emergency notifications being passed to asterisk. Ex someone in a bathroom that pressed the emergency button and than send notification to asterisk that a certain room pressed the emergency button. I'm thinking with analog gateway like grandstream with 24fxs. Has anyone made something like this that? |
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| 09:21.54 | ruied | I would like to use the existing wiring and keeping the analog fxs phones, do not know if with some aditional electronic in each fxs (ex: loop) it can make an event to asterisk.... |
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| 12:38.42 | Mub | I am so dumb |
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| 12:39.17 | Mub | It's a really bad idea to chown -R asterisk:asterisk /* |
| 12:39.39 | [TK]D-Fender | I've done something similar once a long time ago. |
| 12:39.44 | [TK]D-Fender | killed the FS |
| 12:45.29 | WIMPy | In that case it would be a lot better to run Asterisk as root ;-) |
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| 13:19.28 | dan_j | Where can I find the sample config files without building them from the source? I dont have a server that I can run it on without overriding the existing files. |
| 13:19.55 | WIMPy | Just read them in the source tarball. |
| 13:19.57 | dan_j | Specifically, i want to look at the options for NAT |
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| 13:21.53 | johnny_|_ | nat=force_rport,comedia |
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| 13:24.28 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/branches/13/configs/samples/ |
| 13:24.50 | [TK]D-Fender | Haven't gotten to github yet. |
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| 13:59.19 | Mub | Alright. I have a dialplan. This dialplan: http://pastebin.com/HKgp0XXq |
| 13:59.41 | Mub | This is the SIP carrier template that comes with vicidial. |
| 14:00.06 | Mub | Is it supposed to just hang up immediately and play a busy tone? |
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| 14:07.19 | [TK]D-Fender | Mub: If does whatever that AGI does... Then if the AGI doesn't end the call it continues on to dial out using whatever is set in that var and the original number less 2 digits, then hang up |
| 14:07.39 | [TK]D-Fender | There is no "busy tone" in the dialplan. |
| 14:07.42 | [TK]D-Fender | Maybe there is in the AGI |
| 14:07.44 | [TK]D-Fender | We don't know |
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| 14:09.18 | stefan27 | is there any time estimate for version 13.5 yet? as in more likely to be within 0-3 weeks than 1-2 months? |
| 14:09.54 | [TK]D-Fender | When whatever they are working on is more stable I guess |
| 14:10.12 | [TK]D-Fender | Since it's been 2 months I'd bet on sooner than later |
| 14:11.27 | dan_j | johnny_|_: Thanks but that doesnt seem to be working for a client. Suddenly, they are having random calls dropping within a few seconds of starting. Only outgoing calls. It's most certainly a NAT issue but can't quite pin point it. |
| 14:11.33 | file | 0-3 weeks |
| 14:11.51 | dan_j | All incoming calls are fine. Occasional outgoing calls drop due to a SIP packet not being received. |
| 14:13.04 | stefan27 | cool - gonna upgrade when it comes, ive skipped a few major versions |
| 14:13.23 | [TK]D-Fender | stefan27: What doesn't work in 13.4.0 for you? |
| 14:14.25 | mjordan | stefan27: We release every 4 - 8 weeks. In this case, there's one or two last issues we want to get fixed before we get an RC out. |
| 14:14.48 | mjordan | the last one, I believe, is up for review on Gerrit right now, actually |
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| 14:15.17 | file | indeed! a minor comment on it, should be resolved and merged today |
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| 14:15.36 | stefan27 | D-Fender - I have 13.1.0 with manual patches extracted from various jiras. with 13.5.0 out i can throw those patches away and get better aligned! |
| 14:16.22 | [TK]D-Fender | So what about 13.4.0? |
| 14:18.09 | stefan27 | most of the issues we had with 13.1 was with dtls which were supposedly properly solved between 13.4.0 and 13.5.0. Didnt have enough incentive to download 13.4.0 :) |
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| 14:21.15 | stefan27 | I guess I should read all change-logs summarys from 13.2 to 13.5 to see what else happend |
| 14:22.59 | stefan27 | Thanks for the info you 3 - have a nice weekend everyone |
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| 14:30.53 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.4.0 (2015/06/04), 11.18.0 (2015/06/04), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 15:12.56 | hkraal | I'm trying to get Call files working on a system that has been thrown into my lap using http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out as my source. The hello-world example works but I want to call outbound when the user picks up his phone which lead me to http://pastie.org/pastes/10309976/text?key=xkn5nenqgnpktcgjoejneg but now Asterisk is congratulating me all the time instead of dialing outbound. What is my error, wrong context? (don't |
| 15:13.20 | [TK]D-Fender | Extension: SIP/316123456789 |
| 15:13.27 | [TK]D-Fender | that is not an extension |
| 15:13.40 | [TK]D-Fender | you have context,extension, priority to set for this |
| 15:13.50 | [TK]D-Fender | that is where in the DIALPLAN you send the caller once they answer |
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| 15:21.55 | hkraal | [TK]D-Fender: Not sure if I understand fully what you said but I gathered; I've to define an context containing some sort of autodial extention which actually run the Dial Application, is that more in the direction? |
| 15:22.36 | [TK]D-Fender | yes |
| 15:22.51 | [TK]D-Fender | this is just the reverse of a normal call. |
| 15:23.01 | [TK]D-Fender | * calls OUT to Channel: then dumps them into the dialplan |
| 15:23.10 | [TK]D-Fender | instead of call coming IN from a device, and then hitting the dialplan |
| 15:23.33 | hkraal | Right... thats works... awsome... some puzzle pieces are falling into place, thanks! :) |
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| 15:48.41 | Ice_Strike | On the dialer system web backend, I can create an outbound queue with name, ring times, etc. It will then insert into MYSQL. Should it also add addiontal outbound queue in the asterisk extensions dynamically or how it should work? |
| 15:49.19 | [TK]D-Fender | GUI's not supported here.... |
| 15:50.21 | Ice_Strike | I am not asking that. |
| 15:51.16 | [TK]D-Fender | Ok, there is no such thing as an "outbound queue" really either... |
| 15:51.34 | [TK]D-Fender | so I'm not quite sure what your direction is on this |
| 15:51.37 | WIMPy | And why extensions? |
| 15:51.40 | [TK]D-Fender | rephrase that a little |
| 15:59.02 | Ice_Strike | I will try my best to rephrase. I am trying to understand the logic. If I develop my own predictive dialer system (web base) - I can create a number of Inbound or Outounds queue/campaign. Basically I enter a campaign such as "Technical Support" and choosed Outounds type. That information get added to MySQL database. Now how does Asterisk dialer plan come to play for new added queue/campaign? |
| 15:59.13 | Ice_Strike | I have tried my best to rephrase. |
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| 16:02.01 | WIMPy | The dialplan coes to play when you originate a call. |
| 16:02.22 | WIMPy | That's somethign you will have to do. |
| 16:04.43 | Ice_Strike | Indeed I understand that. Should every new Outounds or Inbound queues/campaigns get dynamically added into dialplan? |
| 16:07.03 | WIMPy | What dou you want to add to your dialplan? |
| 16:13.20 | Ice_Strike | WIMPy Well If I have add mulitple Outounds queues/campaigns via web backend. Then I want all added campaigns can originates the outbound calls. Each campaign have unique ID from from MySQL, and each campaign may have different max ring times, max call attempts. |
| 16:14.22 | WIMPy | And why do you need to add/modify dialplan for that? You have to do the queueing anyway. |
| 16:15.13 | Ice_Strike | Ah ok, just trying to understand the logic how it work. |
| 16:15.39 | WIMPy | It is (or will be) YOUR logic. |
| 16:15.57 | Ice_Strike | How does queueing come to play? |
| 16:16.49 | WIMPy | That's what it's all about, isn't it? |
| 16:18.57 | Ice_Strike | WIMPy http://pastebin.com/19TDPEBv |
| 16:19.37 | WIMPy | What's that? It says absolutely nothing. |
| 16:19.57 | Ice_Strike | I have seen a predictive dialer system that everytime I add a new outbund campaign, it get added in the extention.conf |
| 16:20.26 | Ice_Strike | This pastebin example |
| 16:20.58 | WIMPy | You surely could do so, but I have no idea why you would. |
| 16:21.07 | Ice_Strike | Also campaign-queue-c is definded in campaign table in mysql database. |
| 16:21.38 | Ice_Strike | WIMPy Ah ok. That is why I am asking is there is better approch or how can it be done without this. |
| 16:22.34 | WIMPy | Create an extension for your calls and pass variables as needed. |
| 16:22.35 | Ice_Strike | Looking at in queues.conf - [campaign-queue-c] and [campaign-queue-a] |
| 16:22.42 | Ice_Strike | the block is 100% identical |
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| 16:22.53 | Ice_Strike | It has campaign-queue-a to campaign-queue-z |
| 16:23.05 | Ice_Strike | in queues.conf |
| 16:23.16 | Ice_Strike | Proberly badly designed. |
| 16:23.51 | WIMPy | So it's limited to annoying 26 groups of people simultaneousely? |
| 16:24.32 | Ice_Strike | yes |
| 16:27.26 | Ice_Strike | WIMPy So there is no need for mulitple queues for each campaigns? |
| 16:29.07 | Ice_Strike | or how queues suppose to work for each campaign |
| 16:29.39 | WIMPy | The queues are NOT in the dialplan. They are in whatever you're going to write. |
| 16:32.56 | Ice_Strike | write via? |
| 16:33.10 | WIMPy | ??? |
| 16:34.03 | Ice_Strike | Ignore me :) I need to read about Queue. |
| 16:35.37 | WIMPy | It is something you have to do. Asterisk doesn't have any such functionality included. |
| 16:35.43 | Ice_Strike | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id285928.html |
| 16:35.47 | Ice_Strike | I am reading that page |
| 16:35.54 | Ice_Strike | It has |
| 16:35.55 | Ice_Strike | [sales](StandardQueue) |
| 16:35.59 | Ice_Strike | [support](StandardQueue) |
| 16:36.09 | Ice_Strike | StandardQueue is a template |
| 16:36.47 | Ice_Strike | Can [something via database](StandardQueue) be added dymaically via AMI or something? |
| 16:37.54 | WIMPy | Those are incomming cueues AKA ACD. |
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| 16:38.08 | WIMPy | That has nothign to do with outgoing calls. |
| 16:38.12 | Ice_Strike | Ahh |
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| 16:52.16 | Ice_Strike | Hmmm What you think of Dynamic realtime + queues? |
| 16:52.24 | Ice_Strike | http://lists.digium.com/pipermail/asterisk-users/2013-April/278638.html |
| 16:55.05 | [TK]D-Fender | I fail to see why I'd even care. I'd just generate mine on demand like FreePBX does if I wanted to do it myself |
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| 17:02.56 | [TK]D-Fender | Ice_Strike: Also templates do not exist in DB |
| 17:03.02 | [TK]D-Fender | that is text-only |
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| 17:05.59 | Ice_Strike | [TK]D-Fender Yep/ |
| 17:06.00 | Ice_Strike | . |
| 17:07.22 | Ice_Strike | [TK]D-Fender With Asterisk, how to create new Queue name on demand? |
| 17:07.39 | [TK]D-Fender | make the queue. reload your config |
| 17:07.45 | [TK]D-Fender | ~book |
| 17:07.45 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:07.46 | [TK]D-Fender | ^^^^^ |
| 17:07.56 | [TK]D-Fender | there is a wonderful BOOK that says how * works..... |
| 17:14.17 | Ice_Strike | Yes I know, I meant is there a way to make the queue "dynamically" and reload config. |
| 17:15.04 | Ice_Strike | Via web -> Enter a new queue name and submit -> add to a queue |
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| 17:23.16 | [TK]D-Fender | It';s either realtime, or flat + reload |
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| 18:08.59 | joker_89 | hi |
| 18:09.16 | joker_89 | i have a wrt54gl router, do anyone knows if that router will work with asterisk with some firmware? |
| 18:11.10 | robmal | dd-wrt if any |
| 18:12.38 | joker_89 | and openwrt? |
| 18:13.27 | [TK]D-Fender | It'll work fine stock |
| 18:13.51 | joker_89 | which one [TK]D-Fender ? |
| 18:13.59 | [TK]D-Fender | that router |
| 18:14.04 | [TK]D-Fender | with what it COMES with |
| 18:14.16 | joker_89 | with dd-wrt it comes with asterisk? |
| 18:14.18 | joker_89 | and with openwrt no? |
| 18:14.39 | [TK]D-Fender | Three is a difference between "work with asterisk" and "RUNS asterisk" |
| 18:14.47 | [TK]D-Fender | Your question was poorly worded |
| 18:15.34 | [TK]D-Fender | And I'm sure Google can turn up plenty of results for the latter |
| 18:17.46 | joker_89 | mmm |
| 18:25.05 | lordvadr | So I've got an interesting problem--older system (1.8.7.0) getting upgraded but I need to find out if this is either a bug (fixed or not), or if this is a config issue. I've got several instances of iaxmodem running over localhost, and * appers to hang momentairly every so often for something to the tune of ~130ms. Here's part of the stream analysis for one of the calls. Take a look at packets 48-5 |
| 18:25.06 | lordvadr | 4...everything stops for 151ms, then 5 packets come out nearly instantly, and then everything goes back on it's merry way: http://paste.ubuntu.com/11931691/ |
| 18:26.18 | lordvadr | I've seen something similar before, but it involved debug turned on in a DAHDI kernel module where the interrupt handler was spinning and starving the NIC interrupt. This, however, is on the lo interface, so it appears the process or thread is hanging. I can find nothing in the debug logs that shed any light on it. |
| 18:26.47 | lordvadr | It also appears to be happening on RTP streams setup via SIP as well, but the jitter buffers seem to be handling them fairly well. |
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| 19:49.47 | Dovid | Othern than using MusicOnHold is there any way to play an online music stream? The issue I have is I need to have 200+ streams but only pull them when needed |
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| 22:04.01 | westondennis | hello |
| 22:05.57 | pjensen00 | wut up |
| 22:07.45 | westondennis | I have a Digium TE133 Wildcard. Can someone tell me what exactly it is used for? |
| 22:08.25 | WIMPy | Connecting to PSTN equipment. |
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| 22:08.47 | WIMPy | MAybe you should take a look at what's connected to it. |
| 22:10.22 | westondennis | It connects VoIP with traditional telephone systems right? |
| 22:11.16 | WIMPy | There doesn't have to be VOIP involved. |
| 22:12.28 | westondennis | Could you possibly explain what it does in more detail? |
| 22:13.54 | WIMPy | You can connect one E1/T1/J1 line to it or some equipment like a PBX with the same interface. |
| 22:15.54 | westondennis | Then what would it do? |
| 22:17.35 | WIMPy | Enable you to sen and receive calls. |
| 22:18.32 | WIMPy | (or data) |
| 22:18.37 | westondennis | Traditional telephone calls? |
| 22:18.52 | [TK]D-Fender | calls over a T1/E1/J1 type interface |
| 22:18.54 | WIMPy | yes |
| 22:19.25 | [TK]D-Fender | This is a kind of link offered by telcos |
| 22:19.37 | [TK]D-Fender | or it could used to interface with other gear that uses the same. |
| 22:20.12 | westondennis | Can you explain what exactly T1 is? |
| 22:20.15 | [TK]D-Fender | You could for intance use it to connect an existing PBX and then use * to send those calls over some other technology that the PBX didn't natively support |
| 22:20.18 | [TK]D-Fender | ~t1 |
| 22:20.18 | infobot | [~T1] T1 is the basic digital telephony circuit used in North America. T1 runs at 1.544 Mbps. It can be an unstructured channel for data. It can be channelized, to provide 24 time slots of voice or data, each of 64kbps. Time slot 24 is used for D-Chan when used with PRI signalling. |
| 22:20.24 | westondennis | Sorry if I'm slow I'm new to Asterisk and telephony in general |
| 22:20.35 | [TK]D-Fender | T2 = Digital phone circuit |
| 22:20.37 | [TK]D-Fender | T1* |
| 22:21.23 | westondennis | This is making more sense to me now. |
| 22:21.26 | westondennis | Thanks guys! |
| 22:23.42 | westondennis | So if I have a server running AsteriskNOW as my IP PBX would I be able to connect the card to it? |
| 22:24.02 | WIMPy | To what? |
| 22:24.07 | westondennis | The server |
| 22:24.21 | WIMPy | You need to be more specific. |
| 22:24.56 | WIMPy | Are we talking about interfaces or multiple servers or what? |
| 22:25.37 | westondennis | Just a single server. |
| 22:25.41 | [TK]D-Fender | yes |
| 22:25.44 | [TK]D-Fender | its a card |
| 22:25.47 | [TK]D-Fender | you put it in a server |
| 22:25.52 | WIMPy | And what do you want to connect where? |
| 22:25.52 | [TK]D-Fender | * can talk to the card |
| 22:26.06 | [TK]D-Fender | Do you have, or are you planning on getting a T1 circuit? |
| 22:26.24 | westondennis | I don't have one currently. |
| 22:26.54 | WIMPy | What do you plan to do? |
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| 22:27.21 | westondennis | I'm not sure I just have this card. |
| 22:27.42 | WIMPy | So you have no use for it? |
| 22:28.07 | westondennis | Not at the moment just looking to put it to use I guess. |
| 22:28.37 | westondennis | This really isn't my field but I was handed this card and I'm supposed to figure out what to do with it. |
| 22:28.57 | WIMPy | If you don;t have anything you could connect to it ... |
| 22:29.02 | [TK]D-Fender | It can hold down a few sheets of paper from a mild breeze then.... |
| 22:29.16 | WIMPy | That would work. |
| 22:29.35 | WIMPy | Or you put it on ebay and hope someone offers a few bucks. |
| 22:29.57 | westondennis | Hmm our offices are indoors, breezes don't blow through too often. |
| 22:30.38 | westondennis | So what is something I could connect the card to? |
| 22:30.56 | [TK]D-Fender | a telco circuit or hardware PB that has such an interface itself |
| 22:31.01 | [TK]D-Fender | I just told you that... |
| 22:31.11 | WIMPy | A telco line or a (not too small) PBX would be the most obvious ones. |
| 22:32.33 | janicez | What's a J1 |
| 22:33.22 | [TK]D-Fender | Japanese version of T1 |
| 22:33.27 | janicez | aha |
| 22:33.28 | WIMPy | The japanese version of T1. |
| 22:34.37 | WIMPy | A point where the North Americans aren't the only odd ones :-) |
| 22:35.04 | janicez | LOL |
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