| 00:07.11 | paulc | With an Aastra 6731i - can you have a single registration on multiple keys? (Kind of like Polycom does with "X keys per registration" configuration) |
| 00:09.46 | Dovid | anyone know how to dial chan also? |
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| 00:23.46 | ChannelZ | Dial chan also huh?> |
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| 01:04.21 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 01:05.43 | nveid | hey |
| 01:06.01 | nveid | I'm curious.. Do I need a sip trunk for every virtual extension(when I have numbers forwarded) |
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| 01:07.44 | nveid | anyone alive? |
| 01:07.51 | jpsharp | You at least need enough available channels for the number of concurrent calls you'll be handling. |
| 01:09.37 | nveid | So.. for one incoming call, and forwarding out.. Would that require 2 siptrunks then? |
| 01:10.01 | nveid | hrm.. Whats the best provider trunks? |
| 01:10.06 | nveid | provider of even |
| 01:10.16 | jpsharp | Or at least 2 channels on one trunk. |
| 01:10.29 | jpsharp | ~itsp-us |
| 01:10.46 | jpsharp | There's a bunch of providers. Pick one :) |
| 01:11.06 | nveid | oh.. You can get multiple channels on one trunk? |
| 01:11.07 | nveid | interesting |
| 01:11.11 | jpsharp | Yes. |
| 01:11.17 | jpsharp | If the provider allows it. |
| 01:12.01 | jpsharp | I have a trunk from my provider to the house and I can run up to 3 concurrent calls on it. |
| 01:12.26 | nveid | whos that from? |
| 01:12.33 | jpsharp | gafachi.com |
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| 01:14.02 | jpsharp | http://www.voip-info.org/wiki/view/VOIP+Service+Providers |
| 01:14.14 | nveid | yes.. so far I spoke to nextiva on that list.... |
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| 01:14.38 | nveid | they told me I need to pay $25 for each line and/or virtual extension |
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| 01:15.04 | jpsharp | I use gafachi.com, my biggest client uses vitelity.net and callwithus. |
| 01:15.30 | jpsharp | You're going to either pay a fixed amount per "channel" per month or a per minute usage rate. |
| 01:16.22 | nveid | garachi looks decent |
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| 01:17.02 | jpsharp | I use them mostly because they're the ones I could get fax working with. |
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| 01:18.20 | nveid | do they support direct fax machine connections or something? |
| 01:18.28 | nveid | with some adapter of course.. it would have to be |
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| 01:19.31 | jpsharp | They support T.38 faxing, which makes sending fax over voip work much better. |
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| 03:09.12 | ruben23 | hi guyshi guys anyone can help me with this issue please ----> http://pastebin.com/FnsEz65D |
| 03:09.52 | ruben23 | and as i can see i already have a kernel headers correctly set ------> 2.6.32-42-server |
| 03:11.46 | ruben23 | any guys please |
| 03:12.43 | sruffell | ruben23: how do you know you have the kernel headers installed? |
| 03:15.01 | ruben23 | with this ----> http://pastebin.com/zeMLAcFY |
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| 03:15.41 | ruben23 | i base it on my kernle version ---> 2.6.32-42-server ----> so i do apt-get install linux-headers-2.6.32-42-server |
| 03:15.51 | sruffell | Yeah..it doesn't look like you installed one |
| 03:15.52 | ruben23 | and its say you have the latest kernel version |
| 03:16.15 | ruben23 | but i did this apt-get install linux-headers-2.6.32-42-server |
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| 03:16.37 | sruffell | try: sudo apt-get install linux-headers-`uname -r` and paste the output |
| 03:16.43 | ruben23 | <PROTECTED> |
| 03:17.19 | sruffell | normally you want to install the headers for the kernel you're currently running with the command like the above…unless you're trying to compile for a kernel you're about to boot into |
| 03:18.36 | ruben23 | <PROTECTED> |
| 03:19.08 | sruffell | what is the output of: uname -r |
| 03:20.18 | ruben23 | root@p2914644:/usr/src/asterisk/dahdi-linux-complete-2.6.1+2.6.1# uname -r ---> 2.6.32-42-server |
| 03:21.55 | ruben23 | help me please, badly need |
| 03:22.09 | jpsharp | Don't you have to install the kernel source, too, not just the headers? |
| 03:22.45 | sruffell | No…just the headers, and the .config file are all that's needed. |
| 03:23.15 | jpsharp | been a while since I've had to build hardware drivers :) |
| 03:24.01 | sruffell | what is the output of: ls -la /lib/modules/`uname -r`/build |
| 03:24.15 | sruffell | and do it *just* like that. For example on my Ubuntu test server: |
| 03:24.21 | sruffell | $ ls -la /lib/modules/`uname -r`/build |
| 03:24.21 | sruffell | lrwxrwxrwx 1 root root 43 Jul 27 13:38 /lib/modules/3.2.0-29-generic-pae/build -> /usr/src/linux-headers-3.2.0-29-generic-pae |
| 03:32.53 | ruben23 | root@p2914644:/usr/src/asterisk/dahdi-linux-complete-2.6.1+2.6.1# ls -la /lib/modules/`2.6.32-42-server`/build |
| 03:32.53 | ruben23 | 2.6.32-42-server: command not found |
| 03:32.54 | ruben23 | ls: cannot access /lib/modules//build: No such file or directory |
| 03:33.10 | ruben23 | sruffell: thats the output |
| 03:33.51 | sruffell | Why do you have "ls -la /lib/modules/`2.6.32-42-server`/build" instead of "ls -la /lib/modules/`uname -r`/build" ? |
| 03:34.18 | sruffell | just cut and past the command as I posted it above |
| 03:37.51 | ruben23 | lrwxrwxrwx 1 root root 39 2012-09-13 18:37 /lib/modules/2.6.32-42-server/build -> /usr/src/linux-headers-2.6.32-42-server |
| 03:39.33 | sruffell | and is there a .config file in that folder? |
| 03:40.18 | ruben23 | sruffell: what folder..? folder name build. |
| 03:40.21 | ruben23 | ..? |
| 03:40.52 | sruffell | yes…for example: |
| 03:40.52 | sruffell | <PROTECTED> |
| 03:40.52 | sruffell | -rw-r--r-- 1 root root 147351 Jul 27 13:38 /lib/modules/3.2.0-29-generic-pae/build/.config |
| 03:41.35 | sruffell | The presence of that file is what determines whether you get the error on make about not having the sources installed. |
| 03:42.14 | ruben23 | root@p2914644:/lib/modules/2.6.32-42-server# ls -la /lib/modules/`2.6.32-42-server`/build |
| 03:42.14 | ruben23 | 2.6.32-42-server: command not found |
| 03:42.15 | ruben23 | ls: cannot access /lib/modules//build: No such file or directory |
| 03:43.00 | sruffell | ruben23: do you know what the backticks do in your shell? |
| 03:43.39 | sruffell | In case I have to logoff suddenly, you may want to look at the DAHDI section in Chapter 3 of the book: http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html |
| 03:43.41 | sruffell | ~book |
| 03:43.41 | 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 |
| 03:44.58 | ruben23 | <PROTECTED> |
| 03:45.38 | sruffell | it means that you're not putting "uname -r" in the backticks…but are instead puting the *output* of uname -r in backticks |
| 03:47.02 | sruffell | the backticks mean "run the command and put the output here", and the shell doesn't know what command "2.6.32-42-server" is, which is why your getting the error: /lib/modules//build: No such file or directory |
| 03:49.32 | ruben23 | root@p2914644:/lib/modules/2.6.32-42-server# ls -la /lib/modules/`uname -r`/build/.config |
| 03:49.35 | ruben23 | ls: cannot access /lib/modules/2.6.32-42-server/build/.config: No such file or directory |
| 03:49.56 | ruben23 | thats the output |
| 03:50.18 | sruffell | ok…then that is the problem. Try uninstalling and re-installing the kernel headers like: |
| 03:51.54 | sruffell | https://gist.github.com/3719688 |
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| 03:53.49 | carrar | h4X0r5 |
| 03:54.09 | ruben23 | <PROTECTED> |
| 03:54.27 | sruffell | You shouldn't need to. You're just installing the headers for the currently running kernel. |
| 03:54.46 | sruffell | If you've *installed* a different kernel version and haven't rebooted….then…you'll need to repeat this process when you reboot. |
| 03:55.50 | ruben23 | ok done, so ill try installing dahdi now..? |
| 03:56.00 | ruben23 | adn let see if it push through |
| 03:56.02 | sruffell | yes…should just be able to run make |
| 03:59.32 | ruben23 | <PROTECTED> |
| 04:02.27 | sruffell | woot! And be sure to check out the ~book |
| 04:02.32 | sruffell | If you have any other issues. |
| 04:02.48 | ruben23 | sruffell: your a dcap..? |
| 04:03.21 | sruffell | :) Actually no. I work for Digium and maintain the DAHDI drivers. |
| 04:05.04 | sruffell | Digium's training department has a goal that all technical staff get their DCAP certifications….I've just been dragging my feet taking that test. |
| 04:06.15 | ruben23 | sruffell: wow been my dream to work on digium..ven atted a dcap traiing, but just a dream for sure..miracles might make it a reality |
| 04:14.56 | jpsharp | Do you have to go thorugh all the training for DCAP or can you just take a test? |
| 04:15.30 | sruffell | ruben23: Heh cool! Good luck with your current install. |
| 04:17.27 | sruffell | jpsharp: We can just sit for the test. I know…no excuse for me not to, but I also don't pretend to be an Asterisk expert. There are areas that I would need to study in order to pass since I primarily only deal with the parts that interface with the drivers. |
| 04:21.09 | *** join/#asterisk cyborg-one (~cyborg-on@31.31.116.183) |
| 04:28.24 | fling | kaldemar: hello! :p |
| 04:29.37 | jpsharp | I had applied at Digium, instead ended up working for an ITSP in Florida, which was bought by another company and I ended up working for John Todd, who now works at Digium. It's a weirdly twisted family tree of voip people :) |
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| 04:31.38 | sruffell | heh..although John Todd's been gone for over a year. He went to work for a startup in CA as far as I know. That's when Bryan Johns came on to take his role as community directory, which now has been taken over by David Duffett a month or so ago. |
| 04:32.09 | fling | kaldemar: I have logged my problem about asterisk not sending a fromdomain after a while |
| 04:33.09 | fling | kaldemar: after error occured I have set verbosity to 10 and enabled sip debug |
| 04:33.41 | fling | kaldemar: started logging with 'asterisk -vvvr | tee /home/asterisk_broken.log' and tried to call |
| 04:34.39 | fling | kaldemar: after this I restarted asterisk with old config file and restarted it back with my current config, error gone as expected! |
| 04:34.53 | fling | (but idk why it helps, sounds stupid) |
| 04:35.37 | fling | kaldemar: I have enabled verbosity and debug again, started logging with 'asterisk -vvvr | tee /home/asterisk_working.log' and called again without error |
| 04:36.02 | fling | kaldemar: now I need to compare my logs somehow, but idk where to look in the log and what should I see |
| 04:36.19 | fling | so what should I do next? |
| 04:42.26 | fling | same => n,Dial( ChannelZ ) |
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| 05:26.24 | sruffell | g |
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| 06:19.54 | fling | beebeeep: beep |
| 06:21.32 | beebeeep | wazzup |
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| 06:30.23 | deepgray | Good day, folks!, could you recommend opensource web phone (need run in browser) WEBRTC in draft now and not stable. May be Java Applet or Flash? |
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| 06:44.30 | kresp0 | good day deepgray, try this: http://www.sipml5.org/ |
| 06:44.44 | kresp0 | http://code.google.com/p/sipml5/ |
| 06:45.02 | deepgray | that use WEBRTC and works only in experimental mode in Google Chrome |
| 06:45.50 | kresp0 | oh sorry I didnt get your question right |
| 06:46.48 | deepgray | no problem that anyway! |
| 06:46.52 | deepgray | *thanks |
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| 10:19.09 | zamba | how do i turn on moh from an internet radio source? (shoutcast/icecast) |
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| 10:25.22 | v0lZy | hm... interesting |
| 10:25.49 | v0lZy | zamba: i think that moh support using an external program to play it |
| 10:26.14 | v0lZy | i guess u need a player that can connect to internet radio and play something |
| 10:26.28 | v0lZy | and specify it as a source |
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| 10:53.35 | oddball_bfi | Hi folks, does anyone know which modules I need to include for SLIN192 -> alaw translation? |
| 10:53.51 | oddball_bfi | I know its a missing module because autoloading everything fixes my lack of a path |
| 10:57.01 | oddball_bfi | Frustratingly I can translate to and from the un-numbered SLIN no problem |
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| 11:05.17 | oddball_bfi | Random trial and error seems to have shown it to be codec_resample :S |
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| 12:01.34 | atan | ~itsp |
| 12:01.34 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 12:01.45 | atan | ~itsplist-us |
| 12:01.46 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 12:01.47 | atan | ~itsplist-ca |
| 12:01.47 | infobot | somebody said itsplist-ca was Here are some popular Canadian ITSPs: http://www.les.net , http://www.babytel.ca , http://www.voip.ms, http://unlimitel.ca |
| 12:11.07 | atan | Most of the providers I just clicked on have per-minute rates. Anyone have a favorite provider to purchase unlimited inbound? |
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| 12:16.43 | newtonr | atan: i used Teliax for a little bit at home, they have an unlimited inbound plan |
| 12:16.50 | newtonr | had no problems |
| 12:20.49 | v0lZy | all inbound should be unlimited, no' |
| 12:20.56 | v0lZy | the caller should pay not the callee... |
| 12:24.09 | carrar | All inbound (except tollfree) and local outbound should be free. |
| 12:24.27 | carrar | but that really only applies to the US and some places in EU |
| 12:29.09 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 12:44.09 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 12:47.02 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 12:51.50 | atan | Yeah it's so hit or miss... I find there's fine print on a lot of the unlimited stuff. "Fair use" I suppose you could say |
| 12:55.44 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 12:56.08 | Kobaz | what's the max characters in a line of dialplan |
| 12:58.58 | gavimobile | folks I'm having the exact issue "No audio, phone keeps ringing" which is listed on this page. http://www.voip-info.org/wiki/view/Asterisk+Google+Talk . I tried the solution listed on the page but my problem still has not been solved |
| 12:59.05 | gavimobile | any solutions would be great |
| 12:59.38 | *** join/#asterisk bchia (~Adium@nat/digium/x-nkonvzizmjpzndnx) |
| 13:00.12 | *** join/#asterisk serafie (~erin@nat/digium/x-ygzlmimciohoblfp) |
| 13:01.35 | gavimobile | to be more accurate http://www.voip-info.org/wiki/view/Asterisk+Google+Talk#Noaudiophonekeepsringing |
| 13:03.23 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 13:04.41 | *** join/#asterisk wonderworld (~ww@dsdf-4db5e775.pool.mediaWays.net) |
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| 13:06.39 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:08.22 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-tnyocwuyiarmgkov) |
| 13:08.22 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:16.33 | *** join/#asterisk w9sh (~chatzilla@64.238.96.125) |
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| 13:26.56 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 13:27.55 | leifmadsen | gavimobile: if you're not using Asterisk 11 beta for Google Talk, then there is likely nothing to be done. The way Google Talk works has constantly changed over time, and the channel drivers prior to Asterisk 11 do not handle Gtalk well |
| 13:28.02 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:28.33 | leifmadsen | you need to use res_xmpp and chan_motif in Asterisk 11 for anything remotely close to stable |
| 13:29.27 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 13:29.28 | *** join/#asterisk ta^3 (~tacvbo@fixed-203-101-206.iusacell.net) |
| 13:29.46 | gavimobile | leifmadsen: thanks for the detailed answer |
| 13:30.11 | gavimobile | I think I will highly consider upgrading |
| 13:30.36 | gavimobile | on a different note, why doesn't it mention anything about that in the asterisk definitive guide? |
| 13:31.13 | leifmadsen | gavimobile: because that is based on 1.8 |
| 13:31.21 | gavimobile | I have 1.8 |
| 13:31.27 | leifmadsen | and I can't change the text after it is has been printed or predict the future |
| 13:31.46 | leifmadsen | at the time of writing, that was accurate |
| 13:31.49 | leifmadsen | it is no longer accurate |
| 13:32.03 | leifmadsen | 4th edition based on Asterisk 11 documented chan_motif and res_xmpp |
| 13:32.05 | gavimobile | leifmadsen: that's understandable but the guide should prove that it should work on 1.8 |
| 13:32.21 | leifmadsen | gavimobile: it did work on 1.8 at the time, then resources outside the control of Asterisk changed |
| 13:32.23 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 13:32.28 | leifmadsen | e.g. Google changed their implementation |
| 13:32.45 | gavimobile | leifmadsen: so googletalk changed their setup which effected asterisk users |
| 13:32.47 | gavimobile | I see |
| 13:32.48 | *** join/#asterisk Nugget (nugget@carrera.macnugget.org) |
| 13:32.56 | gavimobile | I have a few more questions if that's alright with you |
| 13:33.23 | gavimobile | will saving my config files for use on 11 be useful? |
| 13:34.06 | leifmadsen | I don't know what config files you have |
| 13:34.16 | leifmadsen | look at the UPGRADE.txt and CHANGES file in Asterisk 11 |
| 13:34.23 | leifmadsen | that'll help you determine what is still useful and what needs to be changed |
| 13:34.46 | gavimobile | I assume the upgrade.txt file from asterisk 11 beta |
| 13:35.22 | leifmadsen | right |
| 13:35.34 | leifmadsen | the file included in the version you plan to upgrade to |
| 13:35.34 | gavimobile | great. regarding beta version of 11 |
| 13:35.40 | gavimobile | any news on a stable release |
| 13:35.59 | leifmadsen | if you look at the Asterisk Versions page on the wiki it'll detail the standard release timeline |
| 13:36.11 | leifmadsen | (hint, usually released around the time of AstriCon) |
| 13:36.14 | leifmadsen | and it's not called stable |
| 13:36.17 | leifmadsen | hasn't been for years |
| 13:36.20 | leifmadsen | it's just called a release |
| 13:36.22 | gavimobile | could you post me a link or possible thave the chan bot post me |
| 13:36.40 | leifmadsen | you can search the wiki like I would do |
| 13:36.42 | gavimobile | possibly have the* |
| 13:36.57 | gavimobile | search terim "asterisk wiki versions page"? |
| 13:37.18 | leifmadsen | I suggest trying |
| 13:37.24 | leifmadsen | I teach how to fish, not provide the fish |
| 13:37.31 | gavimobile | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 13:37.47 | gavimobile | leifmadsen: lol great analogy |
| 13:38.04 | gavimobile | 11.x LTS 2012-10-xx (tentative) 2016-10 (tentative) 2017-10 (tentative) |
| 13:38.16 | gavimobile | Release Series Release Type Release Date Security Fix Only EOL |
| 13:38.47 | gavimobile | so were expecting in october! |
| 13:39.18 | gavimobile | im getting married on oct :-) |
| 13:39.53 | gavimobile | if I apply the beta now, will it be as easy a yum update to update my beta to release? |
| 13:58.02 | *** join/#asterisk j4m3s_ (~j4m3s_@adsl-98-77-104-115.mia.bellsouth.net) |
| 13:58.07 | zamba | i want to be able to dial to a certain extension and then the user just gets a dial tone and can from there use dtmf to dial the number he wants to.. can this be accomplished? |
| 13:58.38 | [TK]D-Fender | zamba, "core show application disa" |
| 13:58.51 | [TK]D-Fender | zamba, or jsut make an IVR/Read with a recording of dialtone. |
| 13:59.52 | zamba | i see this is from the outside going in, but i guess there's nothing stopping this from working the other way? |
| 14:00.13 | *** join/#asterisk youjelly (~bwahahaha@39.47.155.90) |
| 14:00.53 | zamba | basically i just want to dial 1000 and then get a dialtone and from there just dial the full number |
| 14:01.01 | zamba | disa is still the way to go? |
| 14:01.09 | zamba | it seems like a bit of an overkill, to be frank |
| 14:02.33 | [TK]D-Fender | what is the point of an inside dialing 1 number and waiting AGAIN to dial another? |
| 14:03.43 | zamba | the problem is our video conferencing unit.. it's set up with h323 as it's preferred protocol for dialing.. so instructing our users to prefix each call with "sip:<number>" is not going to work, so i thought about just adding one number "sip:1000" to the directory and then they can just call that and get a dial tone and from there make the outgoing call |
| 14:04.11 | zamba | and please don't say that asterisk can handle h323 as well :) |
| 14:04.35 | zamba | we don't want to run h323 through asterisk |
| 14:05.05 | zamba | anyway.. i was able to play back the dial tone now.. so now i just have to read in some digits |
| 14:05.32 | zamba | use Read()? |
| 14:05.57 | [TK]D-Fender | Or an IVR. Or DISA. |
| 14:05.59 | [TK]D-Fender | Or whatever |
| 14:08.10 | zamba | oh, it works |
| 14:09.02 | zamba | http://pastie.org/4720064 |
| 14:09.19 | zamba | how can i stop the dial tone after i've entered the first digit? |
| 14:10.04 | zamba | (this is a very simple example, but i'll just present it as a proof of concept) |
| 14:10.50 | zamba | ah, damn.. nevermind.. this won't work.. the dial history will be all messed up |
| 14:11.05 | zamba | instead of the actual number the users dialled, i will have lots of sip:1000 :p |
| 14:11.12 | [TK]D-Fender | userfield <- |
| 14:11.21 | zamba | hm? |
| 14:11.28 | [TK]D-Fender | for your CDR |
| 14:11.35 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-117-244.user.veloxzone.com.br) |
| 14:11.46 | zamba | yeah, but it has to be presented back to the video conferencing unit |
| 14:11.53 | zamba | that's where i want the call history |
| 14:12.11 | [TK]D-Fender | Tell them to stop being lazy then. |
| 14:14.48 | *** join/#asterisk Jinxed- (~tmccurdy@147.177.63.192) |
| 14:15.15 | Jinxed- | Anyone have a good link on how to setup a trunk between asterisk and call manager express using type=peer as opposed to type=friend |
| 14:15.48 | [TK]D-Fender | I fail to see much of a difference. There is no other complimentary setting to care about... |
| 14:16.07 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 14:19.37 | Jinxed- | [TK]D-Fender, the links I have found so far kind of do it what I thought was the hacky incorrect way: ie http://mccltd.net/blog/?p=916 |
| 14:19.44 | Jinxed- | where they don't setup a trunk as a peer |
| 14:19.54 | Jinxed- | and basically trunk only using extensions.conf |
| 14:21.29 | Jinxed- | right now I can call from cme to asterisk but not the other way |
| 14:22.00 | bchia | Jinxed, the peers and friends can both be used for trunks. Friend will match on username in the SIP From: header, Peer will match on IP address in the SIP From: header http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones/configs/sip.conf.sample (ctrl + F for "Naming Devices") |
| 14:22.59 | bchia | Has cme registered to Asterisk? (When you execute "sip show peers" on the CLI is there an IP where Asterisk can route the call to?) |
| 14:24.28 | Jinxed- | siptrunk 192.168.1.10 5060 Unmonitored |
| 14:25.27 | Jinxed- | [siptrunk] |
| 14:25.28 | Jinxed- | <PROTECTED> |
| 14:25.28 | Jinxed- | <PROTECTED> |
| 14:25.28 | Jinxed- | <PROTECTED> |
| 14:25.28 | Jinxed- | <PROTECTED> |
| 14:25.28 | Jinxed- | <PROTECTED> |
| 14:25.34 | Jinxed- | sorry I thought that was on one line |
| 14:25.36 | [TK]D-Fender | There is no such thing as a "trunk with extensions.conf" |
| 14:25.37 | Jinxed- | new irc client |
| 14:25.50 | bchia | try pasting your stuff into pastebin.com and then post the link :) |
| 14:25.51 | [TK]D-Fender | Extensions.conf = DIALPLAN as has nothign to do with how you auth SIP calls |
| 14:26.45 | Jinxed- | my sip.conf |
| 14:26.48 | Jinxed- | http://pastie.org/private/vgc8amk4iqjn6h3zu9hqwq |
| 14:26.52 | [TK]D-Fender | Jinxed-, You should be looking at your actual call attempts |
| 14:27.03 | [TK]D-Fender | Jinxed-, Don't jsut show your configs.. show what HAPPENS with them |
| 14:27.11 | bchia | Yeah - make a call and paste the verbose CLI output |
| 14:27.20 | [TK]D-Fender | jinkA catalog picture of your car won't help us determine the cause for your crash. |
| 14:27.26 | [TK]D-Fender | ~pb |
| 14:27.27 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:27.29 | [TK]D-Fender | ^^^^ |
| 14:27.34 | Dovid | Morning TK |
| 14:29.53 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:31.07 | Jinxed- | http://pastie.org/private/uwnjqeeojk4xfhgn9v6q0a |
| 14:31.16 | Jinxed- | call flow with sip debugging enabled |
| 14:32.17 | *** join/#asterisk j4m3s_ (~j4m3s_@adsl-98-77-104-115.mia.bellsouth.net) |
| 14:33.39 | [TK]D-Fender | Cause No. 38 - network out of order [Q.850 |
| 14:33.45 | Jinxed- | that is from my phone on asterisk to my softphone registered with cme |
| 14:33.50 | Jinxed- | doesn't ring at all |
| 14:34.07 | Jinxed- | the other way rings the phone and it can be accepted, but then no audio works |
| 14:34.10 | Jinxed- | http://pastie.org/private/ojq0ijnw2hbnvkoaezhg |
| 14:34.16 | bchia | Well, Asterisk is getting the call out, and it's not getting 404'd so it seems like cme is not accepting the call |
| 14:35.11 | Jinxed- | hmm |
| 14:35.49 | [TK]D-Fender | What is this "UNIX" channel type? |
| 14:37.35 | [TK]D-Fender | "UNIX/0/6001" <----------- |
| 14:37.58 | [TK]D-Fender | I see no reference to it anywhere, or that other CLI bit with output of "unixnew" etc |
| 14:38.42 | Jinxed- | just the type of phone connected |
| 14:39.03 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 14:39.58 | [TK]D-Fender | what is that CHANNEL type. I do not see reference to it's existance anywhere |
| 14:40.16 | [TK]D-Fender | SIP is SIP, IAX2 is IAX2... wtf is "UNIX" as an Asterisk Channel Type? |
| 14:44.34 | bchia | The UNIX channeltype definitely seems weird. I don't know if that'd cause a 503 from cme, but it's worth configuring the phone properly - is that a SIP phone registered to Asterisk? |
| 14:50.07 | Jinxed- | believe it is used with unix sockets |
| 14:50.21 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-117-244.user.veloxzone.com.br) |
| 14:50.42 | Jinxed- | not sure exactly, but I know I had this all workign in the past at some point |
| 14:50.59 | Jinxed- | the cme config is pretty vanilla |
| 14:51.05 | Jinxed- | i don't know why it would reject a call |
| 14:51.44 | [TK]D-Fender | You seem to be dodging a pretty direct question of where that channel driver even came from. |
| 14:51.52 | [TK]D-Fender | and I gave you the CME reject reason. |
| 14:52.00 | [TK]D-Fender | <[TK]D-Fender> Cause No. 38 - network out of order [Q.850 |
| 14:52.04 | *** join/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 14:52.50 | [TK]D-Fender | You should be able to look up the reasons the CME would respond that. is IT expecting some sort of auth? user? secret? Is it set to accept that kind of number based on how you send it? |
| 14:56.13 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 15:01.57 | Jinxed- | My call from CME: http://pastie.org/private/8qarlwkd6meftqyjxarb8w |
| 15:02.00 | Jinxed- | reading it now |
| 15:05.17 | *** join/#asterisk k610 (~Instantbi@cred.epid.ucl.ac.be) |
| 15:06.34 | Unchained | hi |
| 15:09.14 | *** join/#asterisk bchia (~Adium@nat/digium/x-cgnmipctajnfoxlo) |
| 15:10.45 | *** join/#asterisk k610 (~Instantbi@cred.epid.ucl.ac.be) |
| 15:13.18 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 15:13.20 | eXcAliBuR | yay i'm on |
| 15:13.24 | eXcAliBuR | it only took 2 hours |
| 15:13.26 | eXcAliBuR | :P |
| 15:13.30 | eXcAliBuR | damn firewalls |
| 15:14.55 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 15:15.05 | eXcAliBuR | ok i have lovely digium d50 phones - latest firmware. Using DPMA to provision, when all is said and done, the phone shows Digium at line1 instead of my extension. My user accounts are in users.conf and not sip.conf because I also use the web gui. I think that is where the problems at. how might I correct this? |
| 15:15.42 | Qwell | eXcAliBuR: I've seen that happen when the XML config that the phone gets is malformed. Can you pastebin your res_digium_phone.conf? |
| 15:15.46 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:15.46 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:17.08 | eXcAliBuR | I need to be reminded how to copy the text of a file, my little putty window isn't big enough for all the text |
| 15:17.30 | Qwell | You can scroll in PuTTY |
| 15:17.39 | eXcAliBuR | oh ya |
| 15:17.40 | pabelanger | or log |
| 15:17.41 | eXcAliBuR | :] |
| 15:19.56 | *** join/#asterisk Galen (~Galen@rrcs-24-43-20-118.west.biz.rr.com) |
| 15:20.48 | eXcAliBuR | http://asterisk.pastebin.ca/2204241 |
| 15:20.51 | eXcAliBuR | there we go |
| 15:21.27 | eXcAliBuR | i tried config_auth=mac but it didn't work either so i put it back to disable |
| 15:21.57 | eXcAliBuR | errr wait thats all wrong |
| 15:22.00 | eXcAliBuR | it's not showing my lines |
| 15:22.49 | Qwell | ugh, you enabled the web UI |
| 15:22.51 | Qwell | don't do that |
| 15:22.52 | eXcAliBuR | damn it... i'm having a horrible time trying to copy the whole thing |
| 15:22.57 | Qwell | Disable that, and factory reset. |
| 15:23.04 | eXcAliBuR | I enabled it to do firmware update |
| 15:23.08 | Qwell | why? |
| 15:23.15 | Qwell | DPMA does firmware updates... |
| 15:23.27 | eXcAliBuR | because i'm not sure how to do it yet with dpma |
| 15:23.42 | eXcAliBuR | I have the files loaded but for now i want the web gui |
| 15:23.46 | Qwell | Well, web UI is what's causing stuff to be broken |
| 15:26.03 | eXcAliBuR | my txt gets cut off when i scroll |
| 15:27.13 | eXcAliBuR | i updated my post |
| 15:27.53 | eXcAliBuR | or rather i didn't |
| 15:28.00 | eXcAliBuR | wtf is going on with this |
| 15:28.04 | k610 | right click top left -> copy all to clipboard (Putty) |
| 15:28.25 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 15:28.26 | eXcAliBuR | but it cuts off |
| 15:28.47 | eXcAliBuR | for example i do the command vi res_..... then it will only show a bit of my config |
| 15:28.59 | k610 | do cat res_ |
| 15:29.02 | *** part/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 15:29.27 | eXcAliBuR | that works |
| 15:30.01 | eXcAliBuR | ok updated the paste |
| 15:30.03 | eXcAliBuR | http://asterisk.pastebin.ca/2204245 |
| 15:32.43 | Qwell | Let me know after you've reproduced it after disabling the web UI and factory resetting the phones. |
| 15:32.51 | eXcAliBuR | same issue |
| 15:33.05 | eXcAliBuR | i didn't reload] |
| 15:33.07 | eXcAliBuR | wait |
| 15:33.51 | *** join/#asterisk dddh (~dddh@pdpc/supporter/active/dddh) |
| 15:34.11 | eXcAliBuR | horrible friday, faulty fire alarm system in the building :[ |
| 15:34.50 | eXcAliBuR | ok, reloaded reset same issue |
| 15:34.55 | eXcAliBuR | shows Digium for line 1 |
| 15:36.58 | *** join/#asterisk kriger (~norge@68-190-90-80.dhcp.mdsn.wi.charter.com) |
| 15:41.41 | eXcAliBuR | it's getting the right timezone at least... just not registering the users |
| 15:45.57 | eXcAliBuR | anyone... |
| 15:45.58 | eXcAliBuR | :[ |
| 15:51.05 | eXcAliBuR | is there a place that tells it to check my sip.conf for users that I could change the path to users.conf ? |
| 15:52.43 | eXcAliBuR | i don't know if this is helpful http://asterisk.pastebin.ca/2204260 |
| 15:52.54 | eXcAliBuR | my asterisk cli when the phones are doing there thing |
| 15:53.16 | pabelanger | eXcAliBuR, just contact Digium support |
| 15:53.24 | pabelanger | we don't have access to the source to tell you what is going on |
| 15:54.25 | eXcAliBuR | I called them yesterday and the guy wasn't very nice to me |
| 15:54.27 | eXcAliBuR | :[ |
| 15:54.39 | pabelanger | I find that hard to believe |
| 15:54.42 | eXcAliBuR | He told me to reinstall a certified version of asterisk |
| 15:54.47 | pabelanger | right |
| 15:54.53 | pabelanger | So do it |
| 15:54.54 | eXcAliBuR | because 10.8 isn't certified |
| 15:55.18 | pabelanger | Do you understand, we cannot support Digium phones here |
| 15:55.38 | eXcAliBuR | but thats not what i wanted to hear :[ |
| 15:55.45 | pabelanger | so you need to contact Digium |
| 15:56.13 | eXcAliBuR | after lunch |
| 15:56.30 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 16:01.04 | *** join/#asterisk navaismo (~navaismo@189.144.249.74) |
| 16:03.18 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
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| 16:06.15 | *** join/#asterisk GameGamer43 (uid5533@gateway/web/irccloud.com/x-aqswbescjsgpssuk) |
| 16:14.57 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 16:17.43 | *** join/#asterisk mathi (~Matthew@78.129.54.191) |
| 16:17.44 | mathi | hi |
| 16:18.16 | mathi | the asterisk server gets the wrong file, on the command line I see: "Playing 'select-language.slin'", but I have only a wav file |
| 16:18.31 | mathi | why is trying to play a file that it cannot find? |
| 16:20.41 | *** join/#asterisk MLNoah (~MLNoah@noc.metalink.net) |
| 16:21.56 | *** join/#asterisk viLeR (~viLeR___@190.145.50.28) |
| 16:23.28 | Kobaz | mathi: it's an output oddity, it's converting it internally to slin and playing it |
| 16:24.21 | mathi | Kobaz, how can I stop it converting it? I recorded wav files with audacity, 8000Hz, 16 bits PCM, mono |
| 16:24.47 | Kobaz | if you dont want conversions then have files in the native codec of the call |
| 16:25.32 | mathi | Kobaz, it's okif asterisk converts them, but I cannot hear anything |
| 16:26.35 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 16:26.54 | Kobaz | it may be a different problem you're having |
| 16:27.02 | Kobaz | like one way audio due to nat or something like that |
| 16:30.08 | mathi | Kobaz, right... because lcoally it works just fine |
| 16:30.19 | mathi | Kobaz, how can I fix this problem? I opened the ports |
| 16:30.37 | mathi | well just the port 5064 (I use custom port 5064 instead of 5060) |
| 16:32.13 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 16:32.51 | rrittgarn | What's the easiest way to run two Sub/Macro's at the same time in the dialplan? |
| 16:35.02 | Kobaz | rrittgarn: same time on one channel? you cant |
| 16:35.23 | Kobaz | mathi: you'll need some sip debugs, and a paste of your setup |
| 16:36.17 | rrittgarn | Kobaz: I have a sub that checks for a user's status, and I'm trying to dial multiple users at the same time without re-writing all of my logic for ring groups |
| 16:38.28 | Kobaz | Dial(user1&user2&user3...) |
| 16:39.40 | rrittgarn | Kobaz: I should add that I don't utilize the standard users.conf which makes my dialing a bit trickyier than that |
| 16:40.28 | MLNoah | rrittgarn > if the user list is constant, you might look into a queue, with static members |
| 16:41.06 | MLNoah | using the ringall strategy and ringinuse=no you can get Asterisk to only dial not-in-use interfaces, without any gymnastics (assuming you spell callcounters=yes correctly in sip.conf) |
| 16:41.19 | navaismo | mathi, do you open the rtp ports too? |
| 16:41.38 | MLNoah | though from the queues documentation, I don't think that strategy is guaranteed to work for non-SIP interfaces. |
| 16:43.06 | mathi | navaismo, I will do that, is it UDP also ? |
| 16:43.38 | rrittgarn | actually might have just solved my own problem... Dial(Local/101@internal&102@internal) |
| 16:44.07 | rrittgarn | where internal is the context that calls the sub based on the exten. Looks like it worked just now... |
| 16:44.30 | navaismo | mathi, yes are UDP |
| 16:44.37 | rrittgarn | sitll checked my user statuses, and dialed the endpoints simultaneously... |
| 16:45.18 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 16:45.20 | rrittgarn | although... i found a problem, and i think the queueing strategy might be a better idea |
| 16:46.42 | mathi | navaismo, yea just opened them, didn't work |
| 16:48.03 | mathi | Kobaz, what did you mean by some sip debugs ? |
| 16:49.37 | navaismo | and are the same on the rtp.conf file? |
| 16:49.51 | navaismo | you need to enable the sip debug in the asterisk cli with: sip set debug on |
| 16:51.54 | MLNoah | rrittgarn: yeah, Local channels are a pretty viable way to do it as well |
| 16:56.37 | Tom_123 | Any idea why I'm able to call my device but the audio isn't working once the call is setup? http://pastie.org/private/bfkur6z9fm7dmwc6txsqya |
| 16:56.54 | *** join/#asterisk jrgill (~jrgill@unaffiliated/jrgill) |
| 16:59.44 | cusco | Tom_123: often that is due to NAT |
| 17:00.16 | *** join/#asterisk italorossi (~textual@187.61.199.186) |
| 17:00.17 | Tom_123 | cusco, No nat in this system |
| 17:00.22 | Tom_123 | both call managers are on the same subnet |
| 17:00.23 | Tom_123 | even |
| 17:00.49 | cusco | does the peer configuration has nat=no ? |
| 17:01.48 | cusco | nvm |
| 17:02.12 | cusco | so this talk is between 192.168.1.10 and 192.168.1.10 |
| 17:05.00 | [TK]D-Fender | Tom_123, what's this about "both call managers"? Before yuo only mentioned ONE end was CME. What's this about BOTH? And again, What is this UNIX Channel Type for the calling end? |
| 17:05.57 | mathi | navaismo, trying to see something interesting in the debug logs... but one question: the SIP thing is temorary just for testing, in prod there will be calls from PSTN, not SIP. Will I encoutner the same problems because of NAT etc. ? |
| 17:06.12 | *** join/#asterisk bchia (~Adium@nat/digium/x-byyoljxcbnyxrwae) |
| 17:07.58 | ChannelZ | If by PSTN you mean an analog ATA or T1/E1/PRI then no |
| 17:08.26 | mathi | thanks god) |
| 17:08.40 | ChannelZ | you'll just have other problems depending on which |
| 17:08.54 | mathi | what kind of problems for isntance? |
| 17:08.58 | [TK]D-Fender | There is a lot more to set besides just opening the SIP port |
| 17:09.10 | [TK]D-Fender | So stop guessing at problems |
| 17:09.53 | [TK]D-Fender | Your current setup is vague. Your prospective setup is equally vague. |
| 17:10.08 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:10.08 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:10.09 | eXcAliBuR | oh wow digium has a toll free number |
| 17:10.11 | eXcAliBuR | who knew |
| 17:10.17 | ChannelZ | digium |
| 17:10.20 | [TK]D-Fender | eXcAliBuR, www.digium.com does I'm sure |
| 17:10.20 | eXcAliBuR | :P |
| 17:10.30 | eXcAliBuR | i never saw it before |
| 17:10.39 | [TK]D-Fender | http://www.digium.com/en/company/contact/ |
| 17:10.43 | eXcAliBuR | always did the 256 one |
| 17:10.47 | [TK]D-Fender | eXcAliBuR, I'm sure you looked REAL hard |
| 17:11.03 | [TK]D-Fender | eXcAliBuR, like that big "toll free" written on their contact page... |
| 17:11.14 | mathi | [TK]D-Fender, trying to figure what to do... I' googling beside asking questions here. |
| 17:11.33 | pabelanger | They also have SIP URI dialling to |
| 17:11.34 | pabelanger | too* |
| 17:11.44 | pabelanger | 6060@digium.com |
| 17:11.45 | pabelanger | I think |
| 17:11.48 | pabelanger | sorry |
| 17:11.58 | pabelanger | sip:6000@digium.com |
| 17:11.59 | eXcAliBuR | i never looked over to the right i just saw tech support n the number under it. |
| 17:12.01 | [TK]D-Fender | mathi, First youprovide an actual description of where * is and where the other end is and what's in between. |
| 17:12.13 | eXcAliBuR | how cool is that |
| 17:12.32 | [TK]D-Fender | cool[-1] |
| 17:13.33 | Tom_123 | [TK]D-Fender, Both Call Managers (1. Asterisk 2. CME) |
| 17:13.59 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 17:14.03 | Tom_123 | UNIX is the device type, I don't know more than that, but I have used it with SIP without issue in the past |
| 17:14.11 | [TK]D-Fender | Tom_123, Asterisk is Asterisk. Do not call it "call manager" |
| 17:14.20 | [TK]D-Fender | CME = Call Manager |
| 17:14.26 | mathi | I have an Asterisk server inside a NAT, and I coded a small IVR system. Inside the NAT, I can hear audio, I can give DMTF input, no problems. Outside the NAT I neither can hear audio, nor can I send input. I forwarded 5064 port (I am using 5064 instead of default 5060). I also forwarded port 10002 to 19999, which I specified in the rtp.conf. |
| 17:14.53 | ChannelZ | mathi: audio (RTP) goes out a different port than SIP. Audio leaving Asterisk goes to whatever port the client requests which is out of your control, but your firewall must allow UDP out on that port. Audio entering Asterisk comes in on whatever port Asterisk selects based on the range in rtp.conf and those incoming ports must be allowed/forwarded by your firewall. |
| 17:15.06 | pabelanger | ~sipnat |
| 17:15.06 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 17:15.11 | pabelanger | mathi: ^ |
| 17:15.38 | [TK]D-Fender | Tom_123, that is not a native Asterisk channel device driver. I cannot see us supporting it. |
| 17:15.50 | pabelanger | who ever own this sipnat blog must be getting some nice ad $$$ |
| 17:16.08 | mathi | ChannelZ, as said before, I opened the ports in rtp.conf. But I just have to create the file rtp.conf and asterisk will interpret this file? Don't I have to tell Asterisk somehow that there is an rpt.conf file ? |
| 17:16.10 | mathi | *rtp |
| 17:16.18 | [TK]D-Fender | pabelanger, That's my site, and I have (had) no ads on it |
| 17:16.35 | [TK]D-Fender | (it's currently taken down) |
| 17:16.51 | [TK]D-Fender | well.. the good one, not voip-info |
| 17:16.58 | [TK]D-Fender | that one IS riddled with ads |
| 17:17.52 | ChannelZ | mathi: it's read by some core module, I forget which, but you should see it on a reload. You also need to properly configure sip.conf with your externaddr and localnet. |
| 17:18.14 | mathi | exeternaddr and localnet has been done |
| 17:18.23 | mathi | I will check for the module now |
| 17:18.41 | ChannelZ | And beyond that we have no idea how else to help having seen no config or debug. We also can't diagnose firewall problems for your device outside your network, that's up to you. |
| 17:18.54 | ChannelZ | Or if your Asterisk is behind one for that matter (which it is) |
| 17:19.10 | mathi | I know, I don't want to bother ou with all these files, just some ideas to investigate myself is enough, thank you |
| 17:19.20 | Tom_123 | [TK]D-Fender, Ignoring the UNIX part of it, from the debug, do you know why audio isn't going? |
| 17:19.25 | Tom_123 | negotiated codec? |
| 17:19.29 | ChannelZ | The firewall * is behind has to have open communication, as does the device. The traffic could be being blocked at either end. |
| 17:19.30 | Tom_123 | anything |
| 17:19.42 | [TK]D-Fender | tomCannot ignore it. that is HALF of the call. |
| 17:20.04 | [TK]D-Fender | Tom_123, Cannot ignore it. that is HALF of the call. |
| 17:20.17 | Tom_123 | So if I used a sip phone off of asterisk |
| 17:20.24 | Tom_123 | you would be able to debug the audio? |
| 17:20.33 | Tom_123 | or could at least try? |
| 17:20.35 | [TK]D-Fender | Tom_123, Maybe you should test that. |
| 17:20.39 | Tom_123 | alright |
| 17:20.41 | Tom_123 | fair ennough |
| 17:21.28 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:23.15 | *** join/#asterisk frawd (~francois@221.red-80-28-139.adsl.static.ccgg.telefonica.net) |
| 17:25.22 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:26.17 | *** join/#asterisk sideffect (sideffect@gateway/shell/bshellz.net/x-jixdnwnohaeyjnll) |
| 17:32.23 | mathi | I found the following while checking debug logs of SIP: Transmitting (NAT) to 78.129.54.191:58034 |
| 17:32.24 | mathi | SIP/2.0 404 Not Found |
| 17:32.49 | mathi | (78.129.54.191 being the client IP, not that of the asterisk server) |
| 17:33.33 | [TK]D-Fender | mathi, PASTEBIN <--- |
| 17:33.35 | [TK]D-Fender | ~pb |
| 17:33.35 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:35.17 | ChannelZ | These little fragments don't tell us anything |
| 17:35.40 | mathi | http://pastebin.com/raw.php?i=YNEhGpTM |
| 17:35.45 | mathi | at the end it's written 404 |
| 17:38.10 | [TK]D-Fender | Looking for laptop-mett-asus-f3sc in LocalSets (domain 109.133.203.166) |
| 17:38.22 | [TK]D-Fender | Clearly no match for what it's looking for in your diallpan |
| 17:38.26 | [TK]D-Fender | dialplan* |
| 17:38.27 | atan | So this Cisco 7960 shows "protocol application invalid" when it boots and resets don't resolve it O_O way to finish the week off :D |
| 17:38.48 | [TK]D-Fender | There is no hint for that |
| 17:39.37 | atan | :-) I can only assume a file on the device has gone bad :D now I wonder what tftp it's looking to connect to... hmm... |
| 17:41.17 | ChannelZ | Also you some some issue or another with 78.129.54.191 with all these retransmissions |
| 17:41.26 | ChannelZ | s/some some/have some/ |
| 17:41.53 | eXcAliBuR | the poor digium guy has been working in my box for 30 mins :( I think I made a real mess of things |
| 17:43.04 | eXcAliBuR | is there any money in knowing asterisk really well? |
| 17:43.18 | eXcAliBuR | like are people bundling this up as a voip solution and selling it? |
| 17:43.34 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.38) |
| 17:44.24 | newtonr | eXcAliBuR: yes, lots |
| 17:44.44 | Qwell | newtonr: You mean it's not just a fad? |
| 17:44.50 | leifmadsen | eXcAliBuR: ya, I live on a yacht and have a condo in Toronto, and a villa in France |
| 17:44.53 | leifmadsen | is ballin' |
| 17:44.54 | eXcAliBuR | i suppose you have to deliver a working product tho? |
| 17:44.56 | mjordan | Qwell: phones are a fad |
| 17:44.59 | mjordan | Qwell: as are computers |
| 17:45.00 | Qwell | eXcAliBuR: That usually helps. |
| 17:45.05 | drmessano | Wait, people make money off Asterisk? |
| 17:45.10 | eXcAliBuR | hmmm |
| 17:45.11 | leifmadsen | eXcAliBuR: working products are useful in most industries |
| 17:45.11 | drmessano | I was giving it away for free :( |
| 17:45.16 | mjordan | leifmadsen: I'm going to use my jetpack and come visit |
| 17:45.18 | leifmadsen | drmessano: rookie |
| 17:45.22 | drmessano | lol |
| 17:45.26 | leifmadsen | mjordan: water or fire pack? |
| 17:45.30 | mjordan | both |
| 17:45.35 | mjordan | water out of the left, fire out of the right |
| 17:45.37 | leifmadsen | oh you upgraded finally |
| 17:45.41 | mjordan | the vapor propels me |
| 17:45.53 | Qwell | mjordan: People once made a roadtrip to leifmadsen's house for the weekend. |
| 17:45.57 | Qwell | mjordan: just throwing that out there :p |
| 17:46.10 | leifmadsen | Qwell: it totally happened and was awesome |
| 17:46.10 | mathi | ChannelZ, [TK]D-Fender, got it working!! sorry it was just an externip problem.... I know my IP is dynamic and I always made sure to correct it in the config file, but for I didn't see it change lately :| heh |
| 17:46.17 | mjordan | heh. Katie always has wanted to visit Canada. She doesn't believe me when I tell her I've been there. |
| 17:46.19 | Qwell | leifmadsen: I was totally 2 seconds from joining them. |
| 17:46.27 | leifmadsen | Qwell: WHAT?! you should have |
| 17:46.30 | leifmadsen | it was a great time |
| 17:46.32 | Qwell | well, 2 seconds and a 4 year old |
| 17:46.36 | Qwell | BUT STILL |
| 17:46.39 | leifmadsen | ah ya |
| 17:46.40 | newtonr | eXcAliBuR: yeah its like anything else in business, you have to provide something that works, is useful, and provides value over competing solutions |
| 17:46.41 | mathi | ChannelZ, [TK]D-Fender, and also the RTP ports made it working, so thanks! |
| 17:46.43 | leifmadsen | STILL! :) |
| 17:46.48 | leifmadsen | Qwell: you should go with mjordan |
| 17:46.56 | Qwell | for sure |
| 17:46.56 | leifmadsen | I would allow it |
| 17:47.15 | Qwell | as if you had a say in the matter :p |
| 17:47.19 | leifmadsen | :D |
| 17:50.35 | navaismo | mathi, intead using externip create a domain like dyndns and use exterhost and externrefresh |
| 17:53.34 | eXcAliBuR | my case is being escalated |
| 17:53.37 | eXcAliBuR | :( |
| 17:55.04 | eXcAliBuR | I use the asterisk-gui, and my digium phone isn't getting user info from dpma - shows digium near line 1 - not registering to sip |
| 17:59.16 | eXcAliBuR | i'm getting a headache watching my cli |
| 17:59.29 | eXcAliBuR | i have no idea what the digium crew is doing |
| 18:00.35 | *** join/#asterisk bchia (~Adium@nat/digium/x-gvpnjqizjdzgjaqs) |
| 18:01.13 | eXcAliBuR | they fixed it |
| 18:01.14 | eXcAliBuR | :] |
| 18:01.21 | eXcAliBuR | i was missing this line - |
| 18:01.21 | eXcAliBuR | cidr=10.123.4.0/24 |
| 18:01.59 | eXcAliBuR | yay |
| 18:02.07 | eXcAliBuR | all morning and I got 1 thing done |
| 18:04.14 | eXcAliBuR | now the really fun stuff begins... I try line sharing |
| 18:04.22 | eXcAliBuR | and creating a contact.xml |
| 18:04.23 | eXcAliBuR | ;D |
| 18:06.23 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 18:06.52 | ChannelZ | cidr= xxx missing from where? |
| 18:07.17 | ChannelZ | oh, database thing maybe? |
| 18:07.23 | eXcAliBuR | my res_digium_phone.conf |
| 18:07.35 | ChannelZ | Ah. |
| 18:07.48 | ChannelZ | Knew I didn't recognize it for some reason |
| 18:08.08 | eXcAliBuR | in the wiki it was confusing me, because the network listed for there cidr wasn't the same as anything else |
| 18:08.34 | eXcAliBuR | there asterisk was 10.10.10.10 and there cidr was 192.168.1.0 |
| 18:08.40 | eXcAliBuR | so i didn't understand what to do |
| 18:08.41 | eXcAliBuR | :[ |
| 18:11.13 | ChannelZ | The wiki tends to be unclear on things |
| 18:15.38 | *** join/#asterisk jsjc (~Adium@169.154.78.188.dynamic.jazztel.es) |
| 18:22.20 | leifmadsen | eXcAliBuR: what wiki / page? |
| 18:22.51 | leifmadsen | also, it helps when you understand what CIDR is and how it works, as that is a pretty fundamental networking principle |
| 18:24.15 | eXcAliBuR | the contact.xml is unclear to me |
| 18:24.17 | eXcAliBuR | :[ |
| 18:24.25 | eXcAliBuR | they made such a mess of the wiki |
| 18:24.25 | leifmadsen | O.O |
| 18:24.29 | leifmadsen | which one? |
| 18:24.38 | eXcAliBuR | why doesn't your book have dpma in it? |
| 18:24.42 | eXcAliBuR | it would be easy then |
| 18:24.49 | eXcAliBuR | https://wiki.asterisk.org/wiki/display/DIGIUM/Contacts |
| 18:24.50 | leifmadsen | that's not even a valid question |
| 18:25.12 | leifmadsen | contacts.xml looks perfect clear to me |
| 18:25.19 | leifmadsen | I've made it work before |
| 18:27.22 | mjordan | eXcAliBuR: if you have specific criticisms of a wiki page with content that you feel needs more explanation or could be clearer, I'm sure malcolmd would be interested to hear of the content suggestions. |
| 18:29.22 | eXcAliBuR | i'm going to try from what I understand and see if it works. If it doesn't, I'm going to jump up and down, and then sit in the dark for a while |
| 18:30.51 | navaismo | the question here is are familiar with XML docs |
| 18:32.28 | eXcAliBuR | i did learn tho from watching the digium guy how to reconfigure the phones from the cli |
| 18:32.29 | eXcAliBuR | :] |
| 18:32.37 | eXcAliBuR | no more unplugging - replugging for me |
| 18:33.36 | *** join/#asterisk YoMomma (~YoMomma@71-95-164-90.static.mtpk.ca.charter.com) |
| 18:34.31 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-117-244.user.veloxzone.com.br) |
| 18:45.43 | eXcAliBuR | how do i proceed with getting rid of this : Found a swap file by the name ".res_digium_phone.conf.swp" |
| 18:45.51 | eXcAliBuR | :/ |
| 18:46.21 | eXcAliBuR | oh i can just delete it |
| 18:46.22 | eXcAliBuR | :} |
| 18:50.04 | jpsharp | rm * |
| 18:50.21 | navaismo | O_O |
| 18:50.38 | jpsharp | is ever helpful |
| 18:51.42 | navaismo | \/ ]# rm * ??? |
| 18:51.52 | sruffell | jpsharp: that's dangerous typin' in these here parts. |
| 18:52.01 | eXcAliBuR | this is what my contacts.xml looks like, needless to say it doesn't work. http://asterisk.pastebin.ca/2204293 |
| 18:52.49 | *** join/#asterisk YoMomma (~YoMomma@71-95-164-90.static.mtpk.ca.charter.com) |
| 18:56.39 | eXcAliBuR | anyone |
| 18:56.51 | ectospasm | eXcAliBuR: none of your contacts have <number> or <email> tags... |
| 18:56.59 | eXcAliBuR | I thought that was optional |
| 18:57.49 | eXcAliBuR | I want it on my speed dial keys |
| 18:57.59 | eXcAliBuR | i have 10 speed dial keys |
| 18:58.26 | ectospasm | ...Definition of a primary number for a contact... is mandatory if that contact is attached to a Rapid Dial key. This number is the dialed number when the Rapid Dial key is pressed. |
| 18:59.59 | *** join/#asterisk MLNoah (~MLNoah@noc.metalink.net) |
| 19:03.21 | eXcAliBuR | ok i found some errors in what i did |
| 19:03.28 | eXcAliBuR | so i re-did the config lets see |
| 19:04.08 | ectospasm | are you using DPMA? |
| 19:04.13 | eXcAliBuR | yes |
| 19:09.12 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 19:11.20 | chris_n | jpsharp: no, no... its 'sudo rm -fr /' |
| 19:11.22 | eXcAliBuR | yay it's working |
| 19:11.23 | eXcAliBuR | partly |
| 19:11.25 | eXcAliBuR | :] |
| 19:13.34 | navaismo | sudo eeew |
| 19:18.02 | eXcAliBuR | trying now to get blf_contact_group working |
| 19:19.40 | navaismo | rtm |
| 19:22.00 | eXcAliBuR | the manual isn't working very well for that but google is |
| 19:22.01 | eXcAliBuR | :] |
| 19:23.15 | ectospasm | make sure you put the directives in the proper section... afaik there are no directives that work in both [general] and [<phone>], [<line>], etc... |
| 19:25.15 | eXcAliBuR | it's working |
| 19:25.16 | eXcAliBuR | yay |
| 19:25.23 | eXcAliBuR | i have lights lightin up it's great |
| 19:25.24 | eXcAliBuR | :] |
| 19:25.36 | eXcAliBuR | now monday i can connect it to a live phone line and do line sharing |
| 19:25.37 | eXcAliBuR | yay |
| 19:27.24 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.38) |
| 19:27.58 | ectospasm | eXcAliBuR: do you mean SLA or something else? |
| 19:28.06 | eXcAliBuR | SLA |
| 19:28.08 | eXcAliBuR | yaps thats it |
| 19:28.18 | ectospasm | mjordan can probably tell you SLA is voodoo magic |
| 19:39.14 | *** join/#asterisk _Corey_ (~chatzilla@pool-72-78-178-17.phlapa.fios.verizon.net) |
| 19:43.10 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.38) |
| 19:49.36 | [TK]D-Fender | Yes.... it requires a sacrifice. Usually several, and the results like all Black Magic are never exactly what yuo asked for.... |
| 19:50.50 | TechSmurf | Any idea what the status is on the digium phone recall? |
| 19:53.43 | ectospasm | TechSmurf: what recall? |
| 19:54.03 | TechSmurf | my understanding is there was a batch of bad boards |
| 19:55.04 | carrar | Aren't they all bad? |
| 19:55.06 | carrar | heh |
| 19:55.45 | ectospasm | I haven't heard that. |
| 19:56.06 | mjordan | TechSmurf: I have no idea what you're referring to. |
| 19:56.20 | TechSmurf | that's frightening |
| 19:56.29 | TechSmurf | now I wonder what bs my suppliers were tossing at me |
| 19:57.14 | TechSmurf | I had two separate suppliers cancel orders on me citing a recall |
| 19:57.42 | ectospasm | news to me |
| 19:59.02 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:59.04 | ectospasm | ...and I'm inclined to think that your suppliers couldn't get their hands on enough of them to fulfill your orders |
| 19:59.08 | TechSmurf | Diguim recalled this current the remain stock and all these items are backordered. |
| 19:59.08 | ectospasm | ...but that's just a guess |
| 19:59.11 | TechSmurf | Telecom Nation USA |
| 19:59.59 | TechSmurf | "it was however digium pulled all of their stock from distribution because they had a bad circuit board in the remaining units" |
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| 20:01.08 | TechSmurf | shrugs helplessly |
| 20:01.46 | ectospasm | like I said, I don't think that's true, or I would have heard about it |
| 20:01.52 | ectospasm | is a Digiumite. |
| 20:02.20 | jpsharp | mmmm digiumite on toast. |
| 20:02.32 | ectospasm | heh |
| 20:05.19 | sruffell | yeah..that's all news to me as well. |
| 20:06.56 | mjordan | TechSmurf: just walked down the hall and bugged people. There is no recall, and no one had any clue what the supplier was referring to when they stated there was a bad circuit board. |
| 20:10.03 | newtonr | TechSmurf: we've had an order backlog for a while just due to sheer popularity and number of orders |
| 20:12.13 | jpsharp | An IP phone with an actual gigabit port on it? |
| 20:12.35 | WIMPy | Not that new. |
| 20:13.26 | ectospasm | Gigabit would be a waste, even if you loaded a superheavyweight codec onto it |
| 20:14.02 | jpsharp | If it was just a single gigport on the phone. Gigabit passthrough is a good thing, though. |
| 20:14.06 | WIMPy | 3D HD video with 7.1 audio. |
| 20:14.20 | ectospasm | ...unless you have more than one person using the phone at once, but I don't see how that's practical. |
| 20:15.04 | jpsharp | I hate having to have two ethernet runs to my desk. One for gigabit to my desktop, one for my ip phone which only runs at 100. |
| 20:15.26 | ectospasm | I dunno, would passthrough depend on the actual network bandwidth of the phone port? |
| 20:15.35 | ectospasm | I'm more thinking out loud here. |
| 20:17.28 | jpsharp | Shouldn't, not as along as you use a wire-speed capable 3-port switch on the guts of the phone. |
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| 20:21.46 | chris_n | is the bug mentioned here still applicable in * 10? http://www.voip-info.org/wiki/view/Asterisk+func+regex |
| 20:24.28 | [TK]D-Fender | What "bug"? |
| 20:24.50 | [TK]D-Fender | That page refers to * 1.2, to which I would normally ask "You're not &$ing serious are you?" |
| 20:25.04 | [TK]D-Fender | AEl1? |
| 20:25.05 | [TK]D-Fender | DEAD |
| 20:25.07 | Kobaz | heh |
| 20:25.12 | [TK]D-Fender | AEl at all = bleh. |
| 20:25.19 | [TK]D-Fender | AEL* |
| 20:25.25 | Kobaz | ael is better than dialplan |
| 20:25.25 | [TK]D-Fender | darn shift-release |
| 20:25.35 | Kobaz | perl is even better |
| 20:25.49 | [TK]D-Fender | AEL is no better than straight dialplan. It can only do LESS |
| 20:25.56 | Kobaz | less? |
| 20:25.58 | Kobaz | no way |
| 20:26.00 | [TK]D-Fender | Correct |
| 20:26.04 | Kobaz | you can do more with less code |
| 20:26.05 | [TK]D-Fender | YES way |
| 20:26.14 | chris_n | I'm serious about not busting my dial plan :) |
| 20:26.19 | Kobaz | and do anything in ael that you can do in straight dialplan |
| 20:26.24 | [TK]D-Fender | You may be able to do something in a nicer looking way, but can cannot do as much |
| 20:26.47 | [TK]D-Fender | <Kobaz> and do anything in ael that you can do in straight dialplan <- Absolutely the REVERSE |
| 20:26.48 | Kobaz | what can straight dialplan do that ael2 cannot |
| 20:26.59 | [TK]D-Fender | AEl gets parsed BACk to extensions.conf logic. Yuo can only LOSE with that in mind |
| 20:27.13 | Kobaz | but you can use any function/application in ael |
| 20:27.21 | [TK]D-Fender | Simple math. AEL didn't have presence support, and a few other oddities. |
| 20:27.38 | [TK]D-Fender | you don't have the fine-grained priority controls, etc... |
| 20:27.38 | Kobaz | presece, like hints? |
| 20:27.44 | [TK]D-Fender | yes |
| 20:27.49 | Kobaz | you can do hints in ael |
| 20:27.52 | Kobaz | i do that all the time |
| 20:28.02 | [TK]D-Fender | Perhaps fixed in AEL2, but there were other things. |
| 20:28.12 | Kobaz | you can use labels for fine grained priority control |
| 20:28.13 | WIMPy | And you can't share only some priorites of different extensions. |
| 20:28.35 | Kobaz | why would you want to go back to priorities, when you can use labels instead |
| 20:29.09 | Kobaz | why would you want to share priorities, just write stuff using gosub |
| 20:29.09 | [TK]D-Fender | Kobaz, its also one more level that can break. |
| 20:29.22 | Kobaz | okay, so i rephrase my statement |
| 20:29.25 | WIMPy | It's about the dialplanThat's how the dialplan works. |
| 20:29.38 | WIMPy | It doesn;t matter if you add some abstraction layer. |
| 20:29.40 | Kobaz | there's nothing worth caring about in ael that it cannot do, that you can do in extensions.conf |
| 20:30.08 | Kobaz | [TK]D-Fender: since ael is parsed into dialplan, once you test it you're good to go, what's to break? |
| 20:30.31 | [TK]D-Fender | AEL also becomes one more thing to learn and who wants to look at the crap it generates by the time it comes to having to DEBUG any of it. |
| 20:30.46 | Kobaz | i guess if you upgrade asterisk, the ael behavior might change, but then again dialplan behavior might change as well and you need to test either way |
| 20:30.57 | [TK]D-Fender | Why add more layered crap? |
| 20:31.09 | Kobaz | it takes about 5 minutes to learn ael and how to debug it |
| 20:31.25 | [TK]D-Fender | it gives me nothing that I could care about. The only things complex enough that I might care are also better off in AGI, etc. |
| 20:31.28 | Kobaz | and then your dialplan becomes much less like BASIC and more like an actual structured language |
| 20:31.53 | [TK]D-Fender | "n" + same already kept heading down that road. |
| 20:31.58 | Kobaz | meh |
| 20:32.08 | [TK]D-Fender | And its the sad illusion of being a high-level language. |
| 20:32.13 | Kobaz | that's still using lines and a flat model |
| 20:32.42 | [TK]D-Fender | but... it's checkout time.... |
| 20:32.43 | Kobaz | it's al illusion but it's a pretty decent illusion |
| 20:32.54 | [TK]D-Fender | I'm off... BBl; |
| 20:32.59 | Kobaz | if it helps make development faster, than who cares |
| 20:33.04 | Kobaz | do de do |
| 20:33.43 | ectospasm | [TK]D-Fender is... opinionated |
| 20:33.54 | ectospasm | I happen to like AEL, but I hardly ever use it |
| 20:33.56 | ectospasm | just at home |
| 20:34.04 | ectospasm | written once, I don't touch it |
| 20:34.09 | Kobaz | everyone is opinionated in some fashion |
| 20:34.19 | Kobaz | if people weren't, then we would be all the same |
| 20:40.48 | ectospasm | yes, but sometimes we have no opinions |
| 20:41.06 | Kobaz | no overt ones |
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| 21:27.19 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 21:27.22 | [sr] | hellou |
| 21:27.35 | WIMPy | Hi [sr] |
| 21:27.43 | [sr] | guys, where's the monitor path ? i'm with amnesia :( cant find it |
| 21:27.49 | [sr] | hi WIMPy, how r u |
| 21:28.42 | WIMPy | /var/spool/asterisk/monitor |
| 21:28.46 | [sr] | ya |
| 21:28.49 | [sr] | spool |
| 21:28.51 | [sr] | than |
| 21:28.52 | WIMPy | Doing to many different things. |
| 21:28.59 | [sr] | than=damn |
| 21:29.14 | [sr] | WIMPy: u have no idea :| and still no vacation this year |
| 21:29.33 | WIMPy | va-what? |
| 21:30.38 | [sr] | va = ? |
| 21:31.04 | WIMPy | That word starting with va... |
| 21:31.43 | [sr] | vacation ? |
| 21:32.02 | WIMPy | What's that? |
| 21:32.19 | [sr] | i'd love to know and try that this year... :D |
| 21:32.26 | [sr] | but i guess wont be possible.. |
| 21:36.11 | Deeewayne | anyone know where I can find the digium 'register' utility on the downloads site? |
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| 21:42.06 | ectospasm | Deeewayne: http://downloads.digium.com/pub/register |
| 21:42.35 | Deeewayne | ectospasm, thanks |
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| 21:44.13 | ectospasm | no problem |
| 21:48.18 | [sr] | WIMPy: continuing that discuss, it's not the fact of having vacation, but the need to rest some time |
| 22:19.29 | [sr] | ei gotta sleep |
| 22:19.30 | [sr] | see ya |
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| 22:30.18 | *** join/#asterisk robmv (~robert@201.243.40.249) |
| 22:31.31 | robmv | Hi, just a quick question before calling digium support for anyone with a 410 series card |
| 22:32.02 | robmv | doc says "Illumination of ... LED indicates that a module is installed for the port located directly" |
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| 22:32.48 | robmv | just installed the card, no software installed yet, no led turns on, it is normal, or the leds turns on later when drivers are loaded? |
| 22:33.17 | sruffell | robmv: After the drivers are loaded and detect a module in the slot. |
| 22:33.57 | robmv | sruffell, thanks, I was worried about the card being faulty, I will continue with the software then |
| 22:34.38 | sruffell | np |
| 22:40.07 | ectospasm | robmv: if you need to, make sure the card is registered (http://digium.com/register), and call Digium Technical Support (+1.256.428.6161) |
| 22:40.48 | robmv | ectospasm: sure, I gave the data here to the account owner to register it |
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