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00:28.44 | Iskorptix_ | *problem with flower fixed by resetting config through the phone* |
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00:38.58 | ChannelZ | sounds more like a weed |
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01:50.11 | cstk421 | hello all |
01:50.25 | cstk421 | anyone decent with Hylafax ? |
01:50.33 | cstk421 | basic modem setup |
01:57.47 | phix | hey cstk421 |
01:57.52 | cstk421 | sup |
01:58.02 | phix | I have setup Hylafax, with aix-modem |
01:58.33 | cstk421 | ok i have 2 types of modems conexant and trendnets both internal pci |
01:58.42 | cstk421 | cant get them to work for anything |
01:59.28 | lanning | if they are "winmodems" good luck... |
01:59.56 | phix | hmmm what chip is on it? |
01:59.59 | phix | lspci to find out |
02:00.29 | cstk421 | k |
02:00.36 | phix | cstk421: Most manufactorers of "winmodems" do not create drives for operating system other than windows |
02:00.45 | phix | drivers* even |
02:01.04 | cstk421 | <PROTECTED> |
02:01.04 | cstk421 | <PROTECTED> |
02:01.05 | cstk421 | <PROTECTED> |
02:01.05 | cstk421 | <PROTECTED> |
02:01.05 | cstk421 | <PROTECTED> |
02:01.11 | cstk421 | thats the trendnet |
02:02.25 | phix | ok apparatnly there is support for the Agere systems lucent v.92, however only 32bit support, not 64 from what I can tell |
02:03.24 | phix | There are some deb packages for it too |
02:03.27 | phix | what distro you using again? |
02:03.35 | cstk421 | centos 5 |
02:03.45 | phix | ok there are RPMs for it too |
02:03.53 | cstk421 | 5.7 to be specific |
02:03.55 | cstk421 | k |
02:04.13 | phix | google it, first result is from a mailing list with links to the drivers |
02:04.22 | phix | you can also try the manufactorers website |
02:06.03 | cstk421 | manufacturer has only windows drivers as far as i see |
02:06.31 | cstk421 | although its 251mb |
02:06.46 | cstk421 | would that have linux drivers if the manufacturers site sais its supported you think ? |
02:07.23 | phix | not sure |
02:07.31 | phix | is it a 251mb exe or zip file? |
02:07.47 | cstk421 | zip |
02:07.59 | cstk421 | pullling it anyway to see |
02:07.59 | phix | then maybe |
02:14.07 | cstk421 | nope |
02:14.10 | cstk421 | all windows drivers |
02:14.16 | cstk421 | where did you find the other one ? |
02:21.54 | phix | from google search |
02:21.55 | phix | http://linmodems.technion.ac.il/packages/ltmodem/11c11040/ |
02:22.05 | phix | google ltmodem linux |
02:22.09 | phix | and you should get the official site |
02:24.10 | cstk421 | and what defines which package is needed ? |
02:26.42 | phix | well you would want the rpm, probably the src so you can build it yourself for your system |
02:30.35 | cstk421 | seems more complicated then i expected. so based on what i am seeing there is no driver to just install and get this up and running / |
02:30.36 | cstk421 | ? |
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02:48.01 | itbroke | ack |
02:48.26 | itbroke | can't load dahdi or sip after recompiling and installing asterisk... complained about not being able to find modules.conf |
03:04.55 | phix | cstk421: of course it is :) blame the modem manufactors for not creating a GUI for linux to install it like they did for windows |
03:05.23 | phix | itbroke: nice, does the file exist? |
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04:10.27 | phix | hey I want a 4 port TDM card, should I get AEX431 (PCI-E) or TDM431 (PCI)? |
04:12.35 | ectospasm | phix: they're both functionally equivalent |
04:12.47 | ectospasm | phix: does your target system have PCI or PCIe slots? |
04:14.38 | phix | yes both |
04:14.52 | [TK]D-Fender | What's your future system likely to have? |
04:15.12 | phix | both |
04:15.13 | [TK]D-Fender | My money's on PCIe |
04:15.20 | phix | ok |
04:15.24 | [TK]D-Fender | for if you'll only have one |
04:15.54 | phix | mightput another card in the future, but atm 1 is all I need |
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04:46.25 | bluregard | good evening all |
05:02.25 | p3nguin | wants to build a fire. |
05:02.38 | p3nguin | -12 degrees tonight |
05:04.51 | ectospasm | eventually PCI will be very hard (and very expensive) to find. |
05:10.34 | [TK]D-Fender | And servers are very hard to find WITH PCI these days. Prepare your cards for the future |
05:10.47 | [TK]D-Fender | Which is actually.. the NOW when you think about it |
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06:14.24 | *** join/#asterisk Chorca (~chris@230.88.204.68.cfl.res.rr.com) |
06:16.00 | Chorca | heyhey, having an odd issue.. wonder if anyone here's seen it before.. My asterisk default vm prompts play back awesome to internal phones, yet lock up randomly while playing back over an IAX channel. Audio is great all day long over the channel, but playing back the prompts will just stop randomly, no errors, just dead air. |
06:18.06 | ectospasm | Chorca: what codec(s)? |
06:18.26 | Chorca | gsm and ulaw |
06:18.58 | Chorca | tried forcing the iax to both, i'm able to get clear, no-jitter audio on both of them just fine over the trunk for minutes at a time, but the prompts stall about 5 seconds in. |
06:19.44 | Chorca | this is Asterisk 1.6.2.5-0ubuntu1.4 |
06:20.15 | Chorca | not sure how to get more debugging than it showing it playing the prompts, other than packet-level captures |
06:23.17 | Chorca | http://lists.digium.com/pipermail/asterisk-users/2010-September/254319.html got posted awhile ago, but no resolution :/ |
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06:26.42 | ectospasm | Chorca: no telling. Asterisk 1.6.2.5 is WAAAAAAY out of date. |
06:27.20 | ectospasm | only distro I know of that keeps things truly up to date is AsteriskNOW. PBX in a Flash does OK, too |
06:27.24 | ectospasm | but then again, I'm biased |
06:27.57 | _omer | Hello, I need to install a PBX for my office for internal calls, conference, calls transfers, etc .... so which PBX is good ... (I have never used any asterisk based PBX but worked on Asterisk for couple of years) ... |
06:32.33 | Chorca | Hmm, alright. |
06:32.38 | Chorca | I can bump it up from source, no biggie |
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06:47.24 | _omer | trixbox or freepbx? |
06:47.52 | Chorca | they're both pretty good |
06:48.08 | Chorca | i haven't used either for awhile, but they are both great frontends |
06:49.08 | _omer | and they both have same features? |
06:49.21 | _omer | I think I should go for trixbox .... |
06:49.28 | _omer | or may be freepbx :s |
06:50.34 | Chorca | heh, just install one and see how you like it, they're free, no reason not to try both |
06:51.11 | Chorca | i think trixbox has traditionally had a bit more development because it was a company selling hardware, but like I say, I haven't touched either for awhile, so FreePBX may have caught up |
06:52.07 | min3r | PBX In a Flash is excellent as welll |
06:52.27 | min3r | GV for outbound right out of the box |
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07:24.51 | bluregard | does asterisk have issues with sip softphones on the same machine asterisk is running on? |
07:25.15 | _omer | freepbx or trixbox ?? which one is recommended ? |
07:26.26 | bluregard | never used freepbx, but I really hate trixbox. |
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07:26.32 | kaldemar | _omer: you're on a channel were people would like to answer "none", but choose silence instead. ask in #freepbx. |
07:26.48 | _omer | ok |
07:26.57 | bluregard | _omer: don't cheat, learn asterisk. |
07:27.25 | _omer | I cant develop a pbx in asterisk manually :s |
07:27.40 | bluregard | why not? |
07:28.19 | _omer | thats gonna take month or months and then QA also needed |
07:28.42 | _omer | but I need a pbx for my client in a week |
07:29.19 | bluregard | how big of a phone system is this going to be? |
07:29.29 | _omer | 30 - 40 |
07:29.46 | _omer | it will be a replace of CISCO CALL MANAGER |
07:30.39 | bluregard | good deal |
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07:31.29 | bluregard | for what it's worth, Asterisk really isn't as hard as it seems. The first time I started playing with it I had a working system up in about 2 hours. |
07:32.14 | _omer | but you know client also needs an interface to create extensions, IVR, recording etc .... I cant develop interface that quickly .... |
07:32.23 | _omer | if I do, then it's gonna cost him alot |
07:33.01 | bluregard | I understand. |
07:33.46 | _omer | so freepbx includes all the features ? like call conference, call transfer, recording , etc etc |
07:34.33 | bluregard | no idea. never used that one. I was strong-armed into using trixbox once and I'll never do that again. |
07:35.21 | _omer | ok :) |
07:35.32 | _omer | anyhow, thanks alot |
07:35.40 | bluregard | sure thing |
07:35.43 | bluregard | good luck |
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07:55.14 | Shazzamy | ~book |
07:55.14 | 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 |
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08:33.37 | ChannelZ | *asterisk* contains these features. FreePBX and the like are just obfuscation GUIs that sit on top |
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09:59.30 | KNERD | I kicked the power off on my phone and now it cannot register again |
09:59.52 | KNERD | I keep seeing a SIP/2.0 401 Unauthorized message |
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13:22.43 | Cadey | Hi guys, anyone in here got a couple of mins to explain how we can setup a customer with an analog fax machine up on asterisk with a FXO media gateway |
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13:23.38 | Cadey | we are looking for more of a "fax pass though" than asteisk handeling the faxes I guess |
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13:48.37 | ketas | hmm |
13:48.59 | ketas | how about doing dial-up over voip |
13:49.26 | Cadey | well I guess im trying to find the simplest way tbh |
13:49.54 | Cadey | we are not talking VOIP at the moment because we have PRI's on site not a SIP trunk |
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14:25.12 | twanny796 | when I am in cli, and I do a typo I cannot backspace and the cli freezes?? |
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15:26.16 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
15:27.05 | EmleyMoor | Any known problems with thi native SIP client on Galaxy Nexus and Asterisk? |
15:27.30 | EmleyMoor | the, even... |
15:27.55 | EmleyMoor | Have been trying out all the VoIP clients I could find for Android... |
15:30.04 | fireman_biff | EmleyMoor: I've used CSipSimple successfully on a tablet |
15:30.09 | fireman_biff | note very much, but its worked |
15:30.13 | fireman_biff | *not |
15:30.57 | fireman_biff | it integrates with your contacts and pops up a picture when you get a call, which is nice |
15:34.58 | EmleyMoor | fireman_biff: CSipSimple is the most promising of the ones from Android Market that I've tried - though I did have it stick ringing once, and it won't dial with Bluetooth on |
15:36.40 | fireman_biff | EmleyMoor: hmm... I never tried it with the bluetooth on |
15:37.04 | EmleyMoor | Acknowledged bug - they're trying to fix it |
15:37.45 | EmleyMoor | It seems that even the native client is working over 3G - the manual implies it's WiFi only |
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16:15.18 | mateu | I'm using postfix with asterisk (on same machine) and I'd like to have voicemail emails be sent out on port 587. I have postfix listening on port 587 (via submission service). |
16:16.00 | mateu | However, I haven't figured out how to get the voicemail emails to use port 587 |
16:16.38 | Gugge | setup a smarthost in postfix, using whatever smtp server you want to sent to on port 387 |
16:16.40 | Gugge | 587 |
16:16.45 | Gugge | its a postfix thing |
16:16.50 | Gugge | has nothing to do with asterisk |
16:17.33 | mateu | I want the localhost to be the sender, but yeah seems like it's postfix specific question |
16:18.11 | mateu | Gugge: thanks. I'll dig deeper (every example I've come across doesn't force sending on 587) |
16:18.45 | Gugge | normally it sends from a random port, to port 25 on whatever mailserver the receiver uses |
16:18.58 | Gugge | unless you specify a smarthost, then it sends everything to the smarthost |
16:19.21 | mateu | is trying to deal with port 25 being blocked outbound for verizon fios |
16:20.04 | Gugge | and they dont provide a smarthost on port 25 for the customers? |
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16:53.53 | lauris | hi, is asterisk 1.6.2.x configuration/dialplan compatible with 10.0.0 ? |
16:56.12 | [TK]D-Fender | lauris: Pretty much |
17:05.54 | lauris | is it possible to disable Network configuration menu in Cisco/Linksys SPA phones? |
17:06.26 | lauris | or to limit access to it, setting admin password doesn't help |
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18:06.23 | p3nguin | lauris: You should only be able to see the config, not change it, if you don't supply the password. |
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18:07.48 | lauris | i was able to change web server setting without providing a password |
18:07.52 | lauris | through the menu |
18:10.33 | DennisG | hi everyone |
18:10.51 | DennisG | are here people with a large CDR set? |
18:11.04 | DennisG | in file or SQL format |
18:11.37 | ChannelZ | I have my csv log going back to 2007 when I brought the system up for the first time |
18:12.08 | lauris | DennisG, define large |
18:12.11 | DennisG | oke nice ChannelZ, do you have also a backup of that csv log? |
18:12.33 | DennisG | lauris, like 1 mil+ records |
18:12.49 | ChannelZ | yes |
18:12.49 | lauris | ok, then I have one |
18:12.49 | lauris | | 2704589 | |
18:13.06 | lauris | this is a record count of cdr table |
18:13.07 | DennisG | what's the best way to create proper backups of the CDR data? |
18:13.16 | DennisG | haha nice lauris, good job |
18:13.39 | ChannelZ | well it depends, where is yours? A file or a database? |
18:13.48 | lauris | i use binlog on the mysql side |
18:13.49 | DennisG | a MySQL database |
18:14.00 | lauris | and do incremental sql backups every weekend |
18:14.14 | ChannelZ | you can use mysqldump |
18:14.26 | DennisG | but via SQL statements or do you just copy some files? |
18:14.31 | lauris | via SQL |
18:14.47 | lauris | to be more precise - via mysqldump |
18:15.09 | DennisG | oke |
18:15.18 | DennisG | i'm checking the documentation of mysqldump right now |
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18:16.31 | ChannelZ | mysqldump --complete-insert -u username database > /somewhere/mydatabase.sql |
18:16.51 | DennisG | thank you ChannelZ! |
18:16.52 | lauris | why would you need complete inserts? |
18:17.05 | DennisG | i have the same question hehe |
18:17.07 | lauris | it's waste of space |
18:17.14 | ChannelZ | Fine, do whatever you want. Sheesh. |
18:17.22 | DennisG | i believe that the data is more then enough |
18:17.23 | ChannelZ | (ever heard of compression?) |
18:17.40 | DennisG | and how can mysqldump create incremental backups? |
18:18.12 | lauris | it can't |
18:18.41 | DennisG | hmm.. but IF i have a dataset of 10M records then i have to wait for a long time... |
18:19.25 | ChannelZ | Then you're stuck using binlogs in mysql which are basically transaction logs |
18:19.42 | lauris | DennisG, if you use binlog you don't have to do full backup very often |
18:19.56 | lauris | and you can also have slave mysql server to take load of primary server |
18:20.08 | ChannelZ | You just have to mind to make a backup of the database and clear the log which is like a starting point.. then the binlog can "replay" everything that happened to the database since the snapshot |
18:20.28 | lauris | as for incremental backups |
18:20.33 | lauris | I use rdiff-backup for this purpose |
18:20.38 | lauris | this is kind of *diff* tool |
18:20.54 | DennisG | hmm... so the best way is like: scp (or something else) the binlogs, delete them and repeat this every week? |
18:21.21 | lauris | DON'T DELETE IT MANUALLY |
18:22.07 | ChannelZ | have fun. I'm going gun shopping. |
18:22.15 | DennisG | oke, with the tool mysqlbinlog then? |
18:22.31 | lauris | mysqldump can do this for you in a proper way |
18:22.34 | lauris | just read the manual |
18:23.31 | DennisG | i reading this: http://dev.mysql.com/doc/refman/5.6/en/mysqlbinlog-backup.html |
18:23.36 | DennisG | i'm * |
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18:25.29 | DennisG | creating incremental backups like: mysqlbinlog --read-from-remote-server --host=host_name --raw binlog.000130 binlog.000131 binlog.000132 |
18:25.40 | DennisG | restoring backups like: mysql --host=host_name -u root -p < dump_file |
18:26.08 | DennisG | it looks promising |
18:26.55 | DennisG | it's even possible to create backups via a slave SQL server to offload the master |
18:27.19 | lauris | that's what i was telling |
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18:27.42 | DennisG | haha oke sorry, but it's not the mysqldump program |
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18:28.40 | ven0m | Question - does asterisk support "on-the-fly" codec change? meaning - can it handle situations where client switch codec in the middle of coversation for example |
18:29.33 | DennisG | ven0m i think not.. SIP use a triple handshake system for using a correct codec |
18:29.40 | DennisG | eg Asterisk uses that |
18:30.40 | ven0m | hmm |
18:31.17 | ven0m | so basically the codec which the client will use is determined once he connects to astersk (event before he makes teh call) ? |
18:31.21 | DennisG | lauris, thank you for everything! |
18:31.48 | DennisG | yup, if Asterisk and the other end use the same codec |
18:32.04 | DennisG | aka you can use multiple codecs in a specific order |
18:32.19 | DennisG | first codec have then the highest priority |
18:33.03 | ven0m | So even though client/server list the same codecs, i cannot switch between them, even through both client and server support it? |
18:34.39 | DennisG | change the order of the codecs |
18:34.48 | DennisG | but it will work after a new call |
18:35.16 | ven0m | I see. what i wanted to do is build an iphone client, with abiliy to switch the codec it uses |
18:35.22 | ven0m | I guess its impossible |
18:37.31 | DennisG | i think it is with SIP |
18:37.36 | DennisG | but i'm not a SIP-expert.. |
18:38.31 | ven0m | I see |
18:38.32 | ven0m | thanks |
18:45.22 | [TK]D-Fender | [13:31]ven0mso basically the codec which the client will use is determined once he connects to astersk (event before he makes teh call) ? <- there is no "codec" before a call |
18:45.31 | [TK]D-Fender | codec offerings are part of the call |
18:45.50 | [TK]D-Fender | Check the SDP offerings in the INVITE |
18:46.11 | ven0m | SDP? (sorry i am a new in the space) |
18:47.13 | ven0m | Is there some guide you could recommend which describes how asterisk determines which codecs both parties will be using throughout the call? |
18:50.33 | [TK]D-Fender | ven0m: Enable SIP DEBUG at * CLI and watch the invite process |
18:51.12 | ven0m | Thank you |
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21:06.34 | Cadey | will Digium still release asteriskNow now that FreePBX have there Distro sorted? |
21:08.26 | KNERD | What could cause a phone to get a "SIP/2.0 401 Unauthorized" error while functioning just fine for 2 months, then power is acctidentally kicked out, now gets that error. |
21:08.46 | Cadey | power out of the phone? |
21:08.51 | Cadey | or out of the asterisk box? |
21:08.57 | KNERD | phone |
21:09.24 | Cadey | was the extension/secret on the phone only in memory and not in its config file |
21:09.34 | Cadey | check the config file first is what I would do |
21:09.53 | KNERD | config is fine, passwors is still in phone |
21:09.58 | KNERD | nothing has been changed |
21:10.33 | Cadey | can you run a sip trace on the box whiile the phone is trying to register |
21:11.08 | KNERD | I have never done this |
21:11.44 | KNERD | sip set debug on? |
21:12.07 | Cadey | sip set debug on <--- in console |
21:12.25 | Cadey | you can pipe all the output from verbose and debug into a text file also |
21:12.25 | KNERD | yeah I did this, and how I am telling what error I am getting |
21:12.47 | Cadey | well can you see it authenticating |
21:13.20 | Cadey | what messages is the phone sending before asterisk returns the 401 |
21:13.23 | KNERD | I can see it trying to register, but the system is refusing it |
21:13.41 | Cadey | can you paste the trace? |
21:13.45 | Cadey | pastebin.com or somthing |
21:14.31 | Cadey | you can pipe the verbose+debugger output to a file from the /etc/asterisk/logger.conf config |
21:14.50 | Cadey | with the -> /var/log/my/file/path/file.txt |
21:15.16 | Cadey | its been a while since I did it so that may be incorrect syntax but its as simple as that :) |
21:17.21 | KNERD | http://pastebin.com/xKShBipm |
21:17.28 | KNERD | 1002 is the one |
21:19.20 | Cadey | hum |
21:20.35 | KNERD | exactly..nothign given |
21:20.46 | Cadey | can you check the extensions.conf |
21:21.30 | KNERD | check it for what? |
21:23.45 | Cadey | its registering with out the extension |
21:23.54 | Cadey | 51.REGISTER sip:localhost |
21:25.06 | KNERD | acutally I chnaged that in the paste because I dont want it to be part of Google history.. it's a domain name in there |
21:25.08 | Cadey | oh no sorry my bad |
21:25.17 | Cadey | :) |
21:25.55 | KNERD | yeah this setup had been working fine for 2 months, then all of a sudden it's not working? |
21:26.46 | Cadey | well a 401 is because the credentials are wrong normaly |
21:27.09 | Cadey | so triple check the secret in the config file of the phone and the secret against 1002 in the extensions.conf |
21:27.17 | KNERD | if they were wrong, my fail2ban would have banned the IP |
21:27.39 | Cadey | check anyway |
21:27.42 | KNERD | i did |
21:27.45 | KNERD | only 6 times |
21:28.04 | Cadey | stupid question but have you tried rebooting asterisk |
21:28.16 | KNERD | "amportal restart" |
21:28.33 | Cadey | asteirsk-CLi> Core restart now ? |
21:28.35 | KNERD | but not a complete system reboot |
21:28.42 | KNERD | yeah I did that too |
21:28.49 | Cadey | do a sip show peers |
21:28.52 | Cadey | do you see 1002 |
21:29.25 | KNERD | yes |
21:29.35 | Cadey | ok, try this |
21:29.48 | Cadey | log out 1001, change its config to 1002 and see if that phone work |
21:30.03 | KNERD | ok |
21:30.34 | Cadey | also, are you authing with scretMD5 or normal secret |
21:30.35 | KNERD | mayb eyou mean 1000 |
21:31.14 | Cadey | because the 401 is telling the phone to use MD5 for auth |
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21:32.03 | KNERD | hmm...let me recheck the phone |
21:33.38 | KNERD | i dont even see a whay to see on th ephone..see no MD5 |
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21:38.12 | Cadey | well |
21:38.29 | Cadey | check in extension.conf a sec and see if secretMD5 is set for 1002 |
21:38.37 | Cadey | or is it just Secret |
21:39.45 | KNERD | it seems the phone does have the ability |
21:41.39 | Cadey | well I may be wrong so check the extension.conf and see if its asking for SecretMD5 |
21:42.03 | Cadey | as the phone may simply have reset its self to open text instead of MD5 :) |
21:44.02 | KNERD | looking |
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21:48.25 | Cadey | any good? |
21:53.16 | KNERD | no |
21:56.31 | wdoekes2 | KNERD: looks like your phone doesn't get the answer from your asterisk |
21:56.38 | wdoekes2 | iptables? |
21:57.46 | wdoekes2 | you're seeing retransmissions of the original register.. no amount of password setting will fix that |
21:59.17 | KNERD | as I mentioned.been functioning fien for 2 months then I kicked the power out by accident |
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21:59.36 | KNERD | that should be no reason to stop functioning properly |
21:59.39 | wdoekes2 | hm.. the problem is probably with the nat router on 99.27.165.224 |
21:59.57 | KNERD | i can try to power cycle it |
22:00.19 | wdoekes2 | yea.. or switch local port in the linksys to something other than 5061 |
22:02.43 | KNERD | why would it be on 5061? |
22:03.47 | KNERD | well..power cycle time...brb |
22:05.20 | *** join/#asterisk KNERD (~KNERD@99.27.165.224) |
22:05.28 | KNERD | and it worked |
22:05.42 | KNERD | wdoekes2: thankgs for poitning that out |
22:05.58 | wdoekes2 | np |
22:06.13 | KNERD | Cadey: and thanks too |
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23:34.11 | blognewb | hey guys |
23:35.08 | blognewb | i hope this is not a stupid question: is there a way to use an android phone without being signed up to a network/carrier? just use google voice and obi? |
23:36.12 | [TK]D-Fender | blognewb: over what? |
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23:37.00 | blognewb | hey tk sorry i know its off topic, I forgot the name of the pbx channel |
23:37.04 | raggi | the Global Satellite Meganet, duh! |
23:37.24 | [TK]D-Fender | blognewb: That's fine. So wover what? |
23:37.27 | [TK]D-Fender | over* |
23:38.33 | blognewb | i dont even know what im talking about lol i guess i need to connect a network still, i just find everyone else expensive except for PlatinumTel but they don't carry the galaxy s2 |
23:39.17 | [TK]D-Fender | then buy an unlocked phone |
23:40.03 | [TK]D-Fender | Or a locked one and unlock it |
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23:46.28 | brainiac | I am running Asterisk v. 1.8.7 and am trying to use the MeetMeCount app. When dialing a conference room ext, the system complains that there is no such application. Can anyone provide some insight? |
23:53.31 | p3nguin | Show us. |
23:53.33 | p3nguin | ~pb |
23:53.33 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
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