| 00:00.37 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 00:01.00 | theslob | i know :( |
| 00:01.04 | Kinbote | oh |
| 00:01.22 | theslob | and i cant find any mirrors |
| 00:02.18 | Kinbote | oh well, i was going to spend the afternoon working to compile the dahdi module into my ec2 kernel, but since i can't even get the code, there are zero mirrors, and the stupid distro package managers don't yet have 1.8, i guess i'll go have fun instead of working |
| 00:03.41 | theslob | hmmm gona check something holf on |
| 00:05.35 | theslob | nope, only the dahdi drivers :( |
| 00:05.52 | theslob | oh you need them |
| 00:05.54 | theslob | http://downloads.openvox.cn/pub/drivers/dahdi-linux-complete/openvox_dahdi-linux-complete-2.4.0+2.4.0.tar.gz |
| 00:11.36 | *** join/#asterisk mick_laptop (~mick@clamwin/admin/mickhome) |
| 00:11.56 | mick_laptop | anyone know what happened to the digium servers? |
| 00:12.15 | mick_laptop | svn.digium.com seems to be down |
| 00:12.30 | mick_laptop | along with downloads.asterisk.org? |
| 00:14.23 | theslob | yes, i have the same problem |
| 00:14.59 | thehar | yes |
| 00:15.04 | *** join/#asterisk [cannibalera] (~cannibale@201-35-198-60.fnsce703.dsl.brasiltelecom.net.br) |
| 00:15.05 | thehar | the server is having maintenance done to it |
| 00:15.13 | *** part/#asterisk [cannibalera] (~cannibale@201-35-198-60.fnsce703.dsl.brasiltelecom.net.br) |
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| 00:16.29 | superflit | anyone has success with asterisk and amazon ecs? |
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| 00:33.44 | thehar | zzzzzz |
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| 00:48.04 | superflit | asterisk.org download section is down.. |
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| 01:02.47 | Dovid | seems like downloads.asterisk.org is down |
| 01:02.58 | pabelanger | Yes, they are working on it |
| 01:05.00 | Dovid | ok |
| 01:05.27 | Dovid | do they have an ETA ? |
| 01:05.53 | thehar | They have been upgrading it. |
| 01:06.00 | Dovid | ah ok. |
| 01:06.07 | Dovid | i also do my upgrades on the weekends ;) |
| 01:06.11 | Dovid | thanks for the ingo |
| 01:06.12 | Dovid | info* |
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| 01:49.46 | [TK]D-Fender | pabelanger: You on the dev team? |
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| 03:24.54 | *** join/#asterisk pianoquintet (~pianoquin@user-0cdf9c4.cable.mindspring.com) |
| 03:25.29 | pianoquintet | I am trying to download the asterisk tarball but without success. Is the server down? |
| 03:26.06 | thehar | it's been down all day |
| 03:26.34 | pianoquintet | what's going on? are there any mirrorrs? thank you |
| 03:26.39 | ectospasm | yeah, they're doing a major overhaul of most community-facing systems, including downloads.asterisk.org |
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| 03:27.14 | ectospasm | svn.asterisk.org is down, too |
| 03:27.31 | ectospasm | pianoquintet: they posted a link to the asterisk-users link |
| 03:27.43 | ectospasm | er, |
| 03:27.46 | ectospasm | not links |
| 03:27.51 | ectospasm | a list of sites that are down |
| 03:27.52 | pianoquintet | thanks. sorry, where can I get that? |
| 03:27.59 | ectospasm | lists.digium.com |
| 03:28.40 | pianoquintet | so there is no way one can install asterisk at this point? |
| 03:28.41 | thehar | http://lists.digium.com/pipermail/asterisk-users/2010-November/256076.html |
| 03:29.22 | ectospasm | pianoquintet: which version are you trying to download? |
| 03:29.44 | pianoquintet | i failed with 1.8, so I am now trying to go back to 1.6.2 |
| 03:30.07 | ectospasm | pianoquintet: which *specific* version are you trying to download? 1.6.2 doesn't cut it |
| 03:30.33 | pianoquintet | thought would try the latest: 1.6.2.14 |
| 03:30.58 | pianoquintet | with 1.4.37 addons |
| 03:31.37 | ectospasm | um |
| 03:31.44 | ectospasm | that doesn't sound like a good idea |
| 03:31.51 | thehar | heh |
| 03:31.58 | ectospasm | lemme see if I can get 1.6.2.14 for you |
| 03:32.13 | pianoquintet | oopps, it is suddenly working |
| 03:32.29 | pianoquintet | what does not sound like a good idea? |
| 03:35.03 | ectospasm | 1.4 addons with 1.6.2 |
| 03:35.08 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 03:35.19 | pianoquintet | which addons should I use? |
| 03:35.34 | ectospasm | 1.6.2 specific ones? |
| 03:35.42 | ectospasm | I dunno, I hardly ever deal with addons |
| 03:35.42 | pianoquintet | good point |
| 03:36.00 | pianoquintet | sorry, i am a total newbie, what do i need addons for? |
| 03:54.03 | ChannelZ | valuable cash and prizes |
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| 04:29.57 | chameloid | anyone know how to resolve Starting Asterisk PBX: Unable to setuid to ? attempting to start fresh install of asterisk on a vps - starts fine as root |
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| 05:38.29 | p3nguin | What might cause MP3Player() to not play mp3 files? |
| 05:39.19 | jaytee | tags |
| 05:39.50 | p3nguin | If I stick the file into MoH, it plays okay. |
| 05:40.01 | mick_laptop | hello, would someone tell me why I can't call my "friends"? Here is an example that I took and just adapted it to just get something running: this is my sip.conf -> http://pastebin.com/MF5XxVma |
| 05:40.09 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 05:40.18 | mick_laptop | just the section under the friends |
| 05:40.30 | mick_laptop | i can register just fine, but I can't call another extension |
| 05:40.41 | mick_laptop | i'm using a softphone (QuteCom) |
| 05:41.06 | p3nguin | Extensions go in extensions.conf. Show us what you have or don't have in there. |
| 05:41.36 | [TK]D-Fender | p3nguin: lack of or bad mpg123 |
| 05:41.47 | [TK]D-Fender | p3nguin: its a largely worthless app |
| 05:42.06 | p3nguin | mpg123 appears fine, since it plays everything else. |
| 05:42.18 | p3nguin | It plays mp3 streams and mp3 moh just fine. |
| 05:42.50 | [TK]D-Fender | p3nguin: tags could be it. ID3 = bad |
| 05:43.04 | [TK]D-Fender | p3nguin: there are some bulk tools to strip them. VBR as well <- |
| 05:43.09 | [TK]D-Fender | VBR = bad |
| 05:46.30 | pabelanger | [TK]D-Fender: yes |
| 05:51.08 | mick_laptop | p3nguin: any way to get a clean dump of the config? (without first trying to parse out comments w/ grep -v) |
| 05:52.58 | *** join/#asterisk sshock (~sshock@63.248.133.83) |
| 05:53.05 | mick_laptop | i tried to add: exten => 1005,hint,SIP/tammari&SIP/Somethingstupid and exten => 10052,hint,SIP/tammari2&SIP/Somethingstupid2 but it said that it wasn't in the context |
| 05:53.37 | p3nguin | mick_laptop: Where's the Dial() command? |
| 05:54.15 | sshock | I'm trying to get One Touch Record to work; I uncommented automon => *1 in features.conf, and I added ,,tw to the Dial in my dialplan. |
| 05:54.27 | sshock | What else could I be missing? |
| 05:55.38 | p3nguin | sshock: the reloading of features and dialplan, perhaps. |
| 05:56.09 | [TK]D-Fender | mick_laptop: What said it wasn't in the context? How do we know which one you put those in? Or why anything should CARE about them? |
| 05:56.40 | [TK]D-Fender | sshock: We see mieces, not the whole. There is also a CHANNEL VARIABLE to set for dynamic features. |
| 05:56.41 | p3nguin | hints certainly don't allow the calling of phones. |
| 05:56.43 | sshock | p3nguin: no luck :( |
| 05:57.22 | sshock | [TK]D-Fender: well, the # button is working for transfers... |
| 05:57.48 | p3nguin | because of the t option. |
| 05:57.52 | sshock | but what is this channel variable you speak of? |
| 05:58.07 | sshock | yes, I know, and the "w" is for one touch record, but it's not working |
| 05:58.12 | [TK]D-Fender | sshock: CHANNELVARIABLES.TEX <-- go read |
| 05:58.13 | sshock | I hit * 1 and nothing happens |
| 05:58.30 | p3nguin | Don't waste too much time between the two keys. |
| 05:58.31 | [TK]D-Fender | sshock: Maybe your DTMF isn't even working. Or you aren't typing fast enough |
| 05:58.46 | sshock | then why would # be working for transfers? |
| 05:59.41 | sshock | [TK]D-Fender: is ${TOUCH_MONITOR} what you are referring to? |
| 05:59.49 | [TK]D-Fender | sshock: I'm not seeing any backup worthy of comment... |
| 06:03.00 | [TK]D-Fender | sshock: http://www.voip-info.org/wiki/view/Asterisk+config+features.conf |
| 06:04.13 | sshock | what part of that is supposed to help me? The One Touch Recording example it gives? |
| 06:04.33 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 06:04.34 | sshock | because I'm not trying to do Monitor() in my dial plan; I just want to hit * 1 whenever I'm in a call |
| 06:05.50 | sshock | Monitor() is not required in that case, or is it? |
| 06:06.36 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 06:06.58 | [TK]D-Fender | READ |
| 06:08.36 | sshock | I did |
| 06:08.49 | p3nguin | I removed all ID3 tags using eyeD3 --remove-all myfile.mp3, but MP3Player() still doesn't play it. The CLI says this: [Nov 14 00:07:54] NOTICE[6649]: app_mp3.c:136 timed_read: Poll timed out/errored out with 0 |
| 06:09.25 | sshock | oh there it is |
| 06:09.36 | sshock | Set(DYNAMIC_FEATURES=automon) |
| 06:16.44 | sshock | that's weird because based on everything I see in features.conf and with "show features", it imples that One Touch Monitor is a built-in feature, not a dynamic feature |
| 06:16.59 | sshock | so why would DYNAMIC_FEATURES have anything to do with it? but I'll try it now... |
| 06:21.57 | *** join/#asterisk pabelanger (~pabelange@2607:f2c0:a000:166:218:f3ff:fe51:c71) |
| 06:21.57 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 06:22.49 | sshock | well, that didn't work :( |
| 06:23.17 | sshock | I wonder why One Touch Record is so hard to enable, whereas Transfer was a piece of cake |
| 06:27.38 | sshock | in any case, I'm pretty sure I don't need this DYNAMIC_FEATURES variable; that must be for an old version of * |
| 06:28.49 | atan3 | sshock, did you throw the right flag in Dial() ? |
| 06:28.52 | sshock | Uncommenting "automon => *1" in features.conf and adding ",,w" to my Dial() in extensions.conf should be all I need. Can anyone confirm? |
| 06:29.09 | atan3 | w or W. Do you know which one you need? |
| 06:29.09 | [TK]D-Fender | sshock: Did you restart * after doing that? |
| 06:29.09 | sshock | atan3: yes, the "w" flag, so the callee can record the call |
| 06:29.18 | sshock | yes |
| 06:29.34 | sshock | I've restarted * several times now... |
| 06:29.36 | [TK]D-Fender | sshock: Then maybe if everything else is right. |
| 06:30.31 | atan3 | sshock, and you added the dynamic thinger in there? |
| 06:30.42 | sshock | atan3: I did but that didn't help |
| 06:30.48 | atan3 | sshock, are you pressy *1 really quickly? |
| 06:30.59 | atan3 | Silly question, fine. But if there is much of any delay for me it skips it. |
| 06:31.05 | sshock | no, I tried it slowly |
| 06:31.11 | atan3 | Try it fast. |
| 06:31.16 | sshock | ok |
| 06:31.17 | atan3 | Almost no delay between button presses. |
| 06:31.40 | atan3 | I believe they let the "slow" presses through to the other line so it doesn't totally mess up stuff line telephone banking, and so on |
| 06:31.59 | atan3 | Get into asterisk -rvvvvv to see it as well |
| 06:32.05 | sshock | oh my gosh |
| 06:32.05 | atan3 | It will show up when it picks it up. |
| 06:32.09 | atan3 | Work now? |
| 06:32.11 | sshock | I think it's actually doing something now |
| 06:32.24 | atan3 | Check out /var/spool/asterisk/monitor/ |
| 06:32.34 | atan3 | The files should save in there I believe, by default |
| 06:32.39 | sshock | it worked! |
| 06:32.51 | sshock | I can't believe it was so STPUID!!! |
| 06:33.22 | atan3 | sshock, um. I don't know how 'stupid' it is... but, um. Yeah. Just one of those things you run into. |
| 06:33.24 | sshock | yep, there is a file in there now, finally |
| 06:33.31 | [TK]D-Fender | ;featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 |
| 06:33.35 | [TK]D-Fender | ^^^^ |
| 06:33.42 | sshock | yeah, but how was I supposed to know I have to type these things in so fast? |
| 06:33.53 | *** join/#asterisk dlirit (~lirant@80.74.100.10) |
| 06:33.59 | [TK]D-Fender | sshock: They hid it in the BIG PRINT |
| 06:34.23 | sshock | yeah, they ought to put it in BIG PRINT, right there in features.conf |
| 06:34.32 | atan3 | hides |
| 06:34.52 | [TK]D-Fender | sshock: it IS there. |
| 06:34.59 | sshock | right below [featuremap] should have like a "; YOU MUST TYPE THESE REALLY FAST" |
| 06:35.02 | sshock | where? |
| 06:35.05 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Asterisk+config+features.conf |
| 06:35.19 | [TK]D-Fender | 01:33]<[TK]D-Fender>;featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 |
| 06:35.22 | [TK]D-Fender | [01:33]<[TK]D-Fender>^^^^ |
| 06:35.23 | atan | sshock, I'm curious to know what you're going with it though? |
| 06:35.52 | atan | s/going/doing/ |
| 06:35.53 | sshock | atan: mainly right now I'm just trying to have fun |
| 06:36.13 | sshock | I don't plan on recording any calls, but I think it would be fun to have the ability. |
| 06:36.20 | [TK]D-Fender | sshock: its right there in the sample config |
| 06:36.25 | atan | You can't be serious? infobot picks up on this stuff? :| |
| 06:37.04 | atan | sshock, I have one setup to record calls then save them onto cloud file hosting =) it's pretty sick. I love it. |
| 06:37.05 | sshock | [TK]D-Fender: I still don't see it |
| 06:37.15 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 06:37.22 | sshock | cool |
| 06:37.33 | atan | sshock, what I can't figure out though is MONITOR_EXEC. It just doesn't run for me. I _must_ put an underscore in front of it. =\ |
| 06:38.05 | [TK]D-Fender | sshock: I PASTED the line, right from the WIKI page I sent you to |
| 06:38.30 | sshock | oh, silly IM client; colored your name differently so I didn't see it |
| 06:38.32 | [TK]D-Fender | sshock: http://pastebin.com/iDZ9Sm57 <---- complete full sample config as provided with * |
| 06:38.45 | sshock | ok, featuredigittimeout |
| 06:38.48 | atan | [TK]D-Fender, do you have a reference to what exactly I am doing by making MONITOR_EXEC into _MONITOR_EXEC? Is there like a section on voip-info about these prefixes to functions? |
| 06:39.28 | [TK]D-Fender | atan: I don't see a problem anywhere... |
| 06:40.13 | atan | [TK]D-Fender, I am not on my regular computer here but I ran into an issue on 1.8 where MONITOR_EXEC would not execute. I replaced it with _MONITOR_EXEC= and it runs just fine now. |
| 06:40.23 | atan | I never did figure out why/how things went wrong though. |
| 06:40.47 | [TK]D-Fender | atan: And I am not a subscriber to "Story Time". |
| 06:48.55 | atan | [TK]D-Fender, unless "story time" is a legitimate reference to some mailing list I am on I assume you're trying to belittle me. I may not spend the same amount of time reading config files, change logs, and everything else in between but I fail to see why it's necessary to attack everyone who is just looking to expand their knowledge. |
| 06:49.01 | sshock | atan: does this help? http://www.voip-info.org/wiki/view/Asterisk+variables#InheritanceofChannelVariables |
| 06:49.23 | sshock | I guess the underscores have to do with inheritence |
| 06:49.25 | atan | sshock, hmm... it very well may. Thanks! |
| 06:49.45 | [TK]D-Fender | atan: Had nothing to do with belittling. |
| 06:50.04 | [TK]D-Fender | atan: 2 people with mystery problems SHOWING NOTHING |
| 06:50.04 | sshock | you're welcome, and thank you for helping me earlier |
| 06:50.16 | atan | Well I always sense a personal attack for some reason, when I'm really not trying to cause trouble =\ |
| 06:50.30 | atan | Perhaps it goes to show the limits of emotions on IRC :P :D |
| 06:50.41 | sshock | laptop about to die; later |
| 06:50.42 | *** part/#asterisk sshock (~sshock@63.248.133.83) |
| 06:51.48 | atan | I suppose it is just as bothersome as a patient walking in to my office complaining of a headache, but not explaining why. But the reality of it is they have no clue themselves, which is why they are there in the first place =) |
| 06:52.22 | atan | Anyway, I won't drag this on as I'm sure we both have better places to chew up our time tonight =) You've clearly said nothing personal was intended, I'll accept that =D |
| 06:53.27 | mick_laptop | p3nguin: btw, thanks for pointing out that point about Dial() - that was a part of my problem - it works now :) |
| 06:53.55 | mick_laptop | i can't chat over sip though, is there something that i need to do to enable SIMPLE? |
| 06:56.06 | [TK]D-Fender | mick_laptop: Not supported. Why con't you use SIP exactly? |
| 06:56.12 | [TK]D-Fender | asterisk SIP SIMPLE |
| 06:56.53 | mick_laptop | the option is greyed out in QuteCom (to do chat) - i figured that I might need to enable it |
| 06:57.07 | [TK]D-Fender | mick_laptop: * is not a messaging platform |
| 06:59.16 | atan | [TK]D-Fender, but on the topic of Call Monitor are you aware of any config within Asterisk that would play a simple `ding` to the user who activated the callmon recording? |
| 06:59.53 | [TK]D-Fender | atan: not offhand... |
| 07:01.09 | *** join/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
| 07:01.21 | [TK]D-Fender | atan: as nothing is listing in the sample config there might not be a way currently |
| 07:01.30 | mbrevda | any bug marshals around? I need a un-sanitized log deleted |
| 07:01.31 | [TK]D-Fender | atan: via Automon anyway. |
| 07:07.19 | [TK]D-Fender | checkout time.... |
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| 07:40.07 | *** join/#asterisk xnixan_ (~xnixan@unaffiliated/xnixan) |
| 07:41.25 | xnixan_ | Hi, i will install asterisk for 8 land lines and 60 internal extensions, what would be the hardware requirement for the server that i will install asterisk on, thanks! |
| 07:42.46 | tzanger | xnixan_: any modern cpu |
| 07:43.10 | xnixan_ | icore7 is ok? |
| 07:43.26 | xnixan_ | tzanger, what about RAM? |
| 07:44.08 | *** join/#asterisk pov (~pov@195.8.115.251) |
| 07:44.10 | tzanger | xnixan_: questions like these lead me ot believe that you haven't got any experience with Asterisk and jumping in to set up a 60-extension office is a bad idea. |
| 07:44.57 | xnixan_ | tzanger, what is your advice? |
| 07:45.32 | tzanger | start small. change your home over to a small asterisk box and get it to pass the WAF tests |
| 07:45.46 | tzanger | if you can score high with WAF, you're pretty much ready |
| 07:46.13 | xnixan_ | tzanger, BTW, i had already done that, but years ago! |
| 07:46.24 | tzanger | xnixan_: rules really haven't changed much |
| 07:47.04 | xnixan_ | is there any major deference between 1.4 and 1.6? |
| 07:47.12 | tzanger | I'm still using 1.4 myself |
| 07:47.15 | tzanger | never made the jump |
| 07:47.43 | xnixan_ | tzanger, thanks for your time, but what about my question? |
| 07:48.10 | xnixan_ | icore7 is ok, with about 4GB RAM, or i need server calss hardware? |
| 07:48.16 | tzanger | you've already overspecc'd the box with a corei7, may as well throw 4G of RAM at it |
| 07:48.55 | tzanger | xnixan_: how much failure/downtime can you deal with? |
| 07:49.31 | xnixan_ | let's say arount 0.1% |
| 07:49.46 | xnixan_ | *around |
| 07:50.06 | tzanger | I would think that having 60 people without phones because a motherboard blew up would get you into the stressful liquid poop phase pretty quickly so you'd probably want either a hot failover or replacement hardware (interface boards, memory, power supplies, etc.) on the shelf |
| 07:51.03 | xnixan_ | tzanger, that also is applicable for server class HW |
| 07:52.46 | tzanger | yep. I've always built with COTS stuff and kept spares within arm's reach. redundant supplies etc never really meant much since the box was local |
| 07:54.31 | xnixan_ | tzanger, thanks for your help! |
| 07:54.41 | xnixan_ | tzanger, have a nice day :-) |
| 07:55.05 | tzanger | p |
| 07:55.06 | tzanger | np |
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| 11:07.29 | as001 | Hi, is it possible to be able to login agent via AgentLogin without prompting for password. I want to achive that operator click link on web interface and to log in (AgentLogin) after click without clicking anything on xlite. |
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| 11:34.37 | wikii | <PROTECTED> |
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| 12:23.37 | wikii | <PROTECTED> |
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| 12:56.19 | tzafrir | wikii, what fax do you use? A physical modem? Something that goes through Asterisk? |
| 12:58.30 | Weazel | tzafrir: hey man :D |
| 12:58.44 | tzafrir | Hi |
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| 13:11.01 | wikii | i use iaxmodem |
| 13:11.12 | wikii | soft modem |
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| 13:26.35 | Weazel | Hey guys, I'm having problems with the Conference in the freepbx getting an error msg "that is not a valid conference number" when calling a conference room -- this is the pastbin "http://pastebin.com/g3Z29Twm" i'm a noob in asterisk, sry in advance |
| 13:27.34 | WIMPy | Weazel: next door left #freepbx |
| 13:27.59 | Weazel | thought so, pretty dead there, so i tried the neighbors, oh well |
| 13:28.43 | robl^laptop | Weazel: its early on a sunday morning for most of the ppl in #freepbx, they're probably still asleep |
| 13:28.54 | Weazel | whats the time there now ? |
| 13:29.27 | robl^laptop | in the US? 5:30am on west cost. 8:30am east coast |
| 13:29.50 | Weazel | oh i see, thats logical... oh well i'll wait then thanks |
| 13:31.38 | wikii | <PROTECTED> |
| 13:36.28 | wikii | WImpy please help |
| 13:37.05 | WIMPy | can only think of a dialplan mistake. |
| 13:37.36 | wikii | i use sendfax command |
| 13:39.16 | wikii | my server is in new jersy..it can send faxes on local numbers.. but it cannot send faxes to another state faxnumber..:( |
| 13:41.02 | wikii | Anyone plz help |
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| 13:43.46 | WIMPy | wikii: It's a little hard with the amount of information you give. |
| 13:43.58 | wikii | i have logs |
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| 14:14.51 | wikii | where i will upload my log |
| 14:14.55 | wikii | file |
| 14:15.19 | WIMPy | ~pb |
| 14:15.19 | 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.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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| 14:26.11 | *** join/#asterisk stmaher (~stephen@80.68.89.200) |
| 14:26.13 | stmaher | Hi guys.. |
| 14:26.23 | stmaher | I just compiled 1.8 on etch and iax2 seems to be missing from the cli.. |
| 14:26.31 | stmaher | does anything need to be done at install timet o enable it? |
| 14:26.37 | stmaher | do I need dahdi drivers at all ? |
| 14:27.14 | pabelanger | stmaher: I believe it was fixed for the next 1.8.1 release |
| 14:27.21 | stmaher | DOH |
| 14:27.33 | WIMPy | Generelly no, but you might need it if you want to use IAX trunking. |
| 14:27.35 | stmaher | pabelanger thanks.. do yo uknow of a fix? |
| 14:27.57 | stmaher | everything is SIP or IAX .. no pstn or pri's |
| 14:28.09 | WIMPy | Err, what needs fixing? |
| 14:28.44 | stmaher | WIMPy iax2 support? |
| 14:28.47 | stmaher | for 1.8 |
| 14:29.16 | WIMPy | But what exactely? |
| 14:29.24 | WIMPy | It's working for me. |
| 14:29.46 | stmaher | WIMPy I dont have any cli for iax2 in my new install of 1.8 |
| 14:29.56 | stmaher | sip is there.. but no iax |
| 14:30.41 | WIMPy | I tried 3 1.8.0 installs so far and they are talking iax2 to each other. |
| 14:30.48 | stmaher | ok.. |
| 14:30.53 | pabelanger | stmaher: I might be confusing the fix for something else actually. |
| 14:31.01 | stmaher | WIMPy do you need dahdi installed too? |
| 14:31.08 | stmaher | WIMPy could that be the issue? |
| 14:31.14 | pabelanger | stmaher: what version of gcc are you using? |
| 14:31.16 | WIMPy | I don;t have dahdi installed |
| 14:31.32 | stmaher | gcc version 4.1.2 20061115 (prerelease) (Debian 4.1.1-21) |
| 14:31.58 | stmaher | its an old box that I cant upgrade unfortunately |
| 14:32.03 | WIMPy | Wow. That old, even for debian. |
| 14:32.16 | stmaher | WIMPy yeah i know |
| 14:32.53 | pabelanger | stmaher: *CLI> module load chan_iax2.so |
| 14:33.26 | stmaher | [Nov 14 14:33:09] WARNING[30945]: loader.c:387 load_dynamic_module: Error loading module 'res_crypto': /usr/lib/asterisk/modules/res_crypto.so: cannot open shared object file: No such file or directory |
| 14:33.39 | stmaher | interesting.. it worked with 1.4 |
| 14:34.03 | pabelanger | stmaher: Ya, just what I suspected. Optional API issue |
| 14:35.49 | pabelanger | stmaher: Do you mind creating an issue on the tracker? Upload a copy of your config.log plus include which version of gcc you are using. |
| 14:36.31 | stmaher | pabelanger LOL.. only if I dont have to create an account :-) |
| 14:36.42 | stmaher | ill jump back to 1.6 |
| 14:36.52 | pabelanger | as a work around you should be able to install libssl-dev, ./configure, make, make install |
| 14:38.01 | stmaher | pabelanger thanks for the help.. |
| 14:38.14 | stmaher | Ill give 1.6 a go as the thoughts of 1.8 has just scared me :-) |
| 14:43.29 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 14:43.39 | *** join/#asterisk Andy_S (~Miranda@svoren.cz) |
| 14:43.43 | Andy_S | another problem :( |
| 14:43.53 | *** join/#asterisk v1s (~v1s@203.177.239.40) |
| 14:44.12 | Andy_S | Got SIP response 406 "Not Acceptable" |
| 14:44.17 | Andy_S | http://pastebin.com/3nixQcSP |
| 14:44.38 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:46.01 | Andy_S | only on incoming call |
| 14:46.08 | Andy_S | outgoing call is OK |
| 14:48.33 | *** join/#asterisk Tim_Toady (~moi@77.49.252.191.dsl.dyn.forthnet.gr) |
| 14:49.14 | pabelanger | Andy_S: Are you behind a NAT? |
| 14:49.22 | Andy_S | yes |
| 14:49.38 | Andy_S | i c... |
| 14:49.43 | Andy_S | sip.conf nat=yes |
| 14:49.46 | Andy_S | right? |
| 14:49.49 | pabelanger | ~sipnat |
| 14:49.50 | infobot | i heard sipnat is 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 |
| 14:49.54 | pabelanger | Andy_S: ^^ |
| 14:50.12 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 14:53.25 | Andy_S | asterisk is on the nat server |
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| 14:54.17 | Andy_S | so it have access on both interfaces - public where i receive call from and also private where is private extension |
| 14:54.36 | Andy_S | i thought that in this case NAT doesn't apply |
| 14:58.09 | Andy_S | yep, asterisk is listening on all ifaces ( 0.0.0.0:5060 ) |
| 14:59.04 | *** join/#asterisk v1s (~v1s@203.177.239.40) |
| 15:01.03 | Andy_S | pabelanger is it still valid, or not? |
| 15:02.22 | *** part/#asterisk v1s (~v1s@203.177.239.40) |
| 15:02.56 | *** join/#asterisk reber (~reber@212-198-99-56.rev.numericable.fr) |
| 15:02.58 | reber | hi all |
| 15:03.09 | reber | http://pastebin.ca/1991041 <<-- ideas ? |
| 15:04.39 | Tim_Toady | seems like Everyone is busy/congested at this time |
| 15:05.03 | [TK]D-Fender | I suggest Sudafed |
| 15:05.09 | WIMPy | reber: The peer isn't registered. |
| 15:05.59 | reber | WIMPy, why that. Can i get more logs about it ? Can i have any reasons about why it doesn't register ? |
| 15:06.23 | WIMPy | sip show peers |
| 15:07.08 | reber | WIMPy, when asterisk is started as a daemon, how to start only the console without having to stop asterisk and run asterisk -cvvvvvvvvvvv ? |
| 15:07.23 | Andy_S | -rvvvvvvvvv |
| 15:07.30 | WIMPy | rasterisk |
| 15:07.30 | reber | thanks :) |
| 15:08.11 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 15:09.29 | reber | nat : N means no nat ? ... Weird i have it enabled everywhere i could in sip.conf |
| 15:10.04 | WIMPy | N means NAT |
| 15:10.09 | reber | okay |
| 15:10.26 | WIMPy | off = " " |
| 15:10.51 | Andy_S | well then why the heck my "out1" (trunk) says N ? |
| 15:10.54 | reber | it seems i have a qos problem as if i stop download then asterisk register well ... |
| 15:11.08 | Andy_S | ah, it doesn't now |
| 15:11.11 | Andy_S | weird ! |
| 15:11.59 | WIMPy | Andy_S: If you have external peers as ell (lika an ITSP) you'll need canreinvite and directmedia=no. |
| 15:11.59 | reber | anything to do on asterisk side to don't have a timeout with this qos problem ? |
| 15:12.12 | Andy_S | thx |
| 15:12.31 | WIMPy | reber: Use TC. |
| 15:13.18 | Andy_S | still 406 |
| 15:13.26 | thehar | yawns |
| 15:13.43 | reber | WIMPy, tc ? links as google on 2 letters is not the best |
| 15:13.47 | Andy_S | WIMPy: no external peers, just external SIP provider |
| 15:14.09 | WIMPy | reber: Ask google for the LARTC then. |
| 15:14.12 | reber | kay |
| 15:14.28 | WIMPy | Andy_S: That makes a yes. |
| 15:15.32 | reber | WIMPy, ok then you answer is : nothing to do on asterisk side. OK |
| 15:16.02 | WIMPy | There's nothing to do with data that doesn't get through. |
| 15:16.16 | Andy_S | WIMPy: okay, then i have canreinvite and directmedia both set to NO, still 406 |
| 15:17.53 | WIMPy | Then theres something else going on. Debug the other side. |
| 15:18.00 | Andy_S | http://pastebin.com/s6Rh3P4T |
| 15:18.12 | Andy_S | i cannot debug SIP provider ... |
| 15:21.22 | *** join/#asterisk v1s (~v1s@203.177.239.40) |
| 15:23.36 | v1s | I have 3 outbound sip providers for calling lets say one area and then I have 2 more for calling another area. some times they are down or cant connect so I wrote a macro and call that. |
| 15:24.03 | v1s | in the macro i just have like dial on multiple rows. |
| 15:24.20 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 15:24.43 | v1s | is there a better way to do that? and a way I can put it all into 1 macro instead of having to have one for the group with 3 and one with the group of 2 ? |
| 15:25.47 | WIMPy | Check the DIALSTATUS. |
| 15:27.12 | v1s | WIMPy: so check dialstatus then go to next sip check dialstatus then go to next like that ? is there some way to group the outbound sips? or do I have to put each one into the lloop? |
| 15:29.34 | stmaher | gah make install didnt do /usr/share/asterisk/sounds |
| 15:29.44 | stmaher | is there anyway to make this manually? |
| 15:30.31 | *** join/#asterisk nicoAMG (~nicoAMG@201.237.49.131) |
| 15:30.40 | Andy_S | cp? |
| 15:31.03 | stmaher | Andy_S just bunch all of them intot he one dir? |
| 15:31.25 | Andy_S | didn't make install said something? |
| 15:31.57 | WIMPy | v1s: You can dial them in parallel, off course, and see who connects you fastest :-) |
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| 15:35.43 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 16:02.21 | *** join/#asterisk sque (~sque@77.49.168.115.dsl.dyn.forthnet.gr) |
| 16:03.13 | sque | Hi, I enabled Cel with odbc (mysql) and at the log it shows warnings. |
| 16:03.32 | sque | [Nov 14 18:02:16] WARNING[18960]: cel_odbc.c:712 odbc_log: Column type -8 (field 'asterisk:cel:cidani') is unsupported at this time. |
| 16:03.32 | sque | [Nov 14 18:02:16] WARNING[18960]: cel_odbc.c:712 odbc_log: Column type -8 (field 'asterisk:cel:cidrdnis') is unsupported at this time. |
| 16:03.32 | sque | [Nov 14 18:02:16] WARNING[18960]: cel_odbc.c:712 odbc_log: Column type -9 (field 'asterisk:cel:ciddnid') is unsupported at this time. |
| 16:03.32 | sque | [Nov 14 18:02:16] WARNING[18960]: cel_odbc.c:712 odbc_log: Column type -9 (field 'asterisk:cel:exten') is unsupported at this time. |
| 16:03.36 | sque | and other |
| 16:03.52 | sque | where ever it says -8 I have CHAR(80) declared |
| 16:04.08 | sque | where ever it says -9 I have VARCHAR(80) declared |
| 16:04.27 | sque | cel_odbc does not support CHAR and VARCHAR? What should be the field types then? |
| 16:12.08 | p3nguin | ~pb |
| 16:12.08 | 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.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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| 17:54.33 | jamko | Any ideas how I can strip a "tech-prefix" from the cdr? |
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| 18:46.07 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 18:46.07 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0 (2010/10/21), 1.6.2.14 (2010/11/11), 1.4.37 (2010/11/11), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.4 (2010/09/01) -=- Visit the new official Asterisk wiki: wiki.asterisk.org |
| 18:57.52 | *** join/#asterisk andyoutside (6161c46a@gateway/web/freenode/ip.97.97.196.106) |
| 19:00.51 | andyoutside | I installed asterisk then I installed the digmi card. If I recall I need to run something so the card is setup. What do I need to run? |
| 19:13.12 | pabelanger | andyoutside: You'll need to install DAHDI, then recompile and install asterisk again |
| 19:18.33 | *** join/#asterisk pku (~pete@c-76-120-44-167.hsd1.co.comcast.net) |
| 19:22.36 | andyoutside | is there a yum for that or do I need to compile it? |
| 19:29.17 | p3nguin | What did yum say when you asked it? |
| 19:40.33 | *** join/#asterisk xpot-mobile (~james@155-99-196-66.uconnect.utah.edu) |
| 19:51.38 | andyoutside | Package asterisk16-dahdi-1.6.2.14-1_centos5.i386 already installed and latest version Nothing to do |
| 19:51.38 | luckman212 | if this is the wrong place to ask please forgive. I am wondering if anyone knows a good solution to attaching some kind of "paging system" to an asterisk pbx. e.g. "doctor jones you have a call parked on 704, doctor jones, 704..." which would announce throughout an entire office from some speakerphone type devices, or better yet through the speaker of the phones themselves (polycom). is this possible? |
| 19:51.55 | andyoutside | yes |
| 19:52.58 | andyoutside | this may help you as a starting porint http://www.freepbx.org/freepbx-help-system?freepbx_version=2.8.0.4&freepbx_menuitem=paging |
| 19:53.58 | luckman212 | andyoutside: thanks that looks good. |
| 19:54.02 | luckman212 | starts reading |
| 19:57.13 | *** join/#asterisk alex5771 (~alex@ool-18b92323.dyn.optonline.net) |
| 19:57.18 | alex5771 | hi |
| 19:57.34 | alex5771 | does latest FS supports calls with GV through PSTN? |
| 20:01.44 | luckman212 | are you asking a question about Freeswitch in the #asterisk room? |
| 20:03.26 | alex5771 | no about Asteisk in asterisk room,i know FS supports it |
| 20:03.46 | alex5771 | but Asterisk guys where working on it,so wanted to know the status? |
| 20:24.34 | *** part/#asterisk pku (~pete@c-76-120-44-167.hsd1.co.comcast.net) |
| 20:25.00 | drmessano | That makes no sense |
| 20:33.54 | *** join/#asterisk bigon (bigon@ubuntu/member/bigon) |
| 20:33.58 | bigon | hi |
| 20:34.34 | bigon | is there an easyway to make all my phones ring on incoming calls? or I need to list all the phone by hand? |
| 20:36.02 | thehar | Such as Dial(SIP/100&SIP101)? |
| 20:39.58 | bigon | yep, but that mean that I need to keep a list of all phones right? |
| 20:40.32 | thehar | yup |
| 20:41.19 | *** join/#asterisk guilhermebr (~Guilherme@189.63.48.180) |
| 20:48.45 | atan2 | Hey, err, does SIP somehow have the functionality to force the called telephone to pick up the call? |
| 20:49.12 | atan2 | I don't support that would exist, would it? |
| 20:56.05 | *** join/#asterisk E-bola (~bola@x1-6-00-13-46-83-e5-04.k1098.webspeed.dk) |
| 21:03.39 | *** join/#asterisk meatbun (~wafers4@cpe-98-155-139-88.hawaii.res.rr.com) |
| 21:12.51 | drmessano | atan, yeah.. it does exist and is supported by a lot of phones |
| 21:12.58 | drmessano | Sorry, atan2 |
| 21:13.22 | *** join/#asterisk SaiSoma|AtHome (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 21:18.35 | meatbun | what's a good SIP client /free to test asterisk? |
| 21:18.46 | thehar | x-lite/blink |
| 21:18.46 | meatbun | X-lite is no longer good |
| 21:19.44 | meatbun | for windows? |
| 21:19.49 | E-bola | its fine |
| 21:19.52 | E-bola | u said to test |
| 21:20.48 | meatbun | to call out. x-lite is now rename. a crappy one is now called x lite, while the x lite is call eyebeam or something |
| 21:20.51 | meatbun | which cost $ |
| 21:21.12 | [TK]D-Fender | No, X-Lite is as it always was. |
| 21:21.33 | E-bola | just browse their page more patiently |
| 21:21.39 | E-bola | its a little bit hidden |
| 21:21.56 | *** join/#asterisk tris (~tristan@CAMEL.ETHEREAL.NET) |
| 21:22.08 | meatbun | http://www.counterpath.com/assets/images/13/x-lite-banner.jpg |
| 21:22.13 | meatbun | x lite look like this now |
| 21:22.18 | meatbun | i just installed it yesterday |
| 21:22.54 | meatbun | new naming are like this now: http://www.counterpath.com/assets/images/191/x-lite-banner.jpg |
| 21:22.56 | E-bola | maybe you installed a trial instaid of the free version? |
| 21:23.21 | meatbun | no. 3 options. 1) free, 2) buy 3) buy |
| 21:23.39 | drmessano | Wow |
| 21:23.49 | meatbun | just look at that pic. it explains all |
| 21:23.53 | drmessano | X-lite was upgraded to 4.0, that is ALL |
| 21:24.02 | E-bola | who caresd about how it looks |
| 21:24.08 | E-bola | didnt u just want to test asterisk? |
| 21:24.08 | drmessano | They didn't CHANGE the naming, it's a new version |
| 21:24.13 | [TK]D-Fender | eyebeaNo, not 3 options. |
| 21:24.29 | drmessano | X-Lite is free, Eyebeam is Pay, Bria is Pay |
| 21:24.34 | [TK]D-Fender | Where do you people come up with this stuff... |
| 21:24.38 | drmessano | They are 3 different products |
| 21:25.27 | E-bola | starts to deploy 1.8 on a none virtualized server to see if it deadlocks less |
| 21:25.29 | drmessano | [TK]D-Fender, same people that claim Asterisk 1.8 is now "something different" because some config option changed. Tinfoilists |
| 21:25.47 | E-bola | I hope 1.8.1 is pushed out soon |
| 21:26.05 | drmessano | E-bola, i've been keeping 1.8 updated from SVN and it's gotten better since release |
| 21:26.15 | [TK]D-Fender | drmessano: I like the new Camry name for the old Echo product... |
| 21:26.16 | drmessano | Indeed 1.8.1 will be a good bugfix release |
| 21:26.34 | E-bola | drmessano: yes i've had to patch it with some stuff as well. Hence i would much rather prefer a 1.8.1 so i dont miss stuff etc. |
| 21:26.58 | E-bola | I think its cool that 1.8 final went out of the door with blind transfers not working on most phones lol |
| 21:27.16 | drmessano | E-bola, it happens. I don't miss anything, because I use SVN |
| 21:27.30 | E-bola | drmessano: you use svn in production? |
| 21:27.58 | drmessano | E-bola, of course.. |
| 21:28.09 | E-bola | Doesnt sound like serious production |
| 21:28.29 | drmessano | E-bola, you do realize SVN doesn't mean TRUNK, right? |
| 21:28.36 | drmessano | SVN != trunk |
| 21:29.00 | E-bola | so what then, the 1.8.1 target? Im not familiar with asterisk's svn structure |
| 21:29.01 | drmessano | I use SVN and update the current release branch |
| 21:29.09 | drmessano | 1.8 target |
| 21:29.13 | E-bola | I still wouldnt do that for production |
| 21:29.22 | thehar | then you fail |
| 21:29.23 | thehar | at life |
| 21:29.29 | drmessano | E-bola, why.. using a magic cutoff point is safer? Um no |
| 21:29.40 | E-bola | drmessano: its bound to be tested more |
| 21:29.44 | thehar | I prefer 1.0 |
| 21:29.48 | E-bola | since it has many many more users running it |
| 21:30.20 | drmessano | E-bola, that's insane. So you're telling me the 1.8.0 tarball is safer than the current SVN I grabbed a few hours ago? No |
| 21:30.25 | E-bola | and i hope 1.8 wasnt a magical cut off point, but a well tested release target.... |
| 21:30.26 | drmessano | You |
| 21:30.46 | drmessano | E-bola, every "release" is a "magic cutoff point" There will always be bugs |
| 21:31.28 | E-bola | No need to argue this really :) |
| 21:31.55 | drmessano | E-bola, as always, you had a point to make. There indeed is no need to argue it, because the logic here is flawed |
| 21:32.04 | E-bola | drmessano: I'd rather know if your using both iax and sip, and if yes if you've noticed any stability issues in 1.8? |
| 21:32.24 | drmessano | E-bola, release tarballs and SVN snapshots are no different. It's perception, and the perception is incorrect |
| 21:32.54 | E-bola | drmessano: i dont agree, but as I said, no need to argue about it |
| 21:33.15 | drmessano | E-bola, of course there isn't |
| 21:40.47 | lanning | E-bola: the 1.8 branch in SVN contains bug fix patches, since the 1.8.0 tarball snapshot. |
| 21:41.06 | lanning | the tarball, is just a snapshot in time. |
| 21:41.32 | lanning | it can be tested a lot, but does not contain any fixes for issues found in the testing. |
| 21:43.05 | E-bola | I guess people really like to argue this :) But imho a tarball isnt a random snapshot in time, its a target where a list of bugs/features have been fixed/created and it is then considered to not contain any critical bugs/blockers |
| 21:43.49 | lanning | and all fixes since then is in the 1.8 branch in svn |
| 21:43.52 | E-bola | projects with nightly builds would be like that, but the same cannot be said of numbered final releases |
| 21:44.05 | E-bola | lanning: yes and all the added bugs, cause by minimal testing |
| 21:44.14 | lanning | nope |
| 21:44.26 | lanning | you are thinking of the new dev in trunk on svn |
| 21:44.36 | lanning | not the release brand on svn |
| 21:44.48 | E-bola | Its quite common to break something when you try and fix something else, even in a bugfix only release... |
| 21:45.27 | lanning | not as common as you would think |
| 21:47.02 | lanning | any complaints about 1.8 issues, the answer would be to get the svn 1.8 branch, before complaining. |
| 21:47.15 | lanning | as it might be fixed already. |
| 21:47.41 | lanning | there are no more patches to the 1.8 tar ball, it is fixed in time. |
| 21:47.56 | *** join/#asterisk tris (~tristan@CAMEL.ETHEREAL.NET) |
| 21:48.20 | E-bola | It's always a weigh-off between fixing all the found issues now with svn or waiting for the next stable |
| 21:48.25 | E-bola | I always lean towards the latter |
| 21:48.26 | meatbun | does ekiga offer free SIP service? |
| 21:48.31 | meatbun | over internet |
| 21:48.46 | *** join/#asterisk N3tw0rK (~N3tw0rK@74.197.192.192) |
| 21:49.26 | [TK]D-Fender | yes |
| 21:49.45 | lanning | a lot of people do, because of the misconception. Just depends on how bad the issue you hit is, for you. |
| 21:51.00 | lanning | it is also why you always have a staging environment, to test the next version (no matter if it is svn or tarball), before going into production. |
| 21:51.23 | E-bola | always should have more likely :) |
| 21:58.46 | *** join/#asterisk imox1234 (~imox1234@p4FC5C32F.dip0.t-ipconnect.de) |
| 22:01.49 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 22:02.49 | *** join/#asterisk jayyers (~jayyers@173-15-214-213-BusName-Atlanta.hfc.comcastbusiness.net) |
| 22:03.07 | jayyers | where would asterisk store the ip address of an annonymous inbound sip call? |
| 22:11.25 | [TK]D-Fender | jayyers: What do you mean "store"? |
| 22:12.16 | carrar | look at your sipchaninfo variables |
| 22:14.52 | alex5771 | hi wehats the status of chan_bluetooth and the like to use my cell phone as a trunk to recive calls? |
| 22:15.33 | jayyers | does it keep a logfile of the calls with the originating ip address? |
| 22:16.25 | [TK]D-Fender | jayyers: Nothing more than you see in CDR |
| 22:17.50 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 22:19.37 | jayyers | [TK]D-Fender: the reason i ask is i have annonymous sip enabled and had someone hit my pbx with 50 calls within a minute at 2am and i wanted to prevent that from happening while still having annonymous sip enabled |
| 22:20.09 | [TK]D-Fender | jayyers: Then enable logging and run fail2ban |
| 22:21.18 | carrar | I don't think he's rejecting the calls |
| 22:21.37 | [TK]D-Fender | carrar: And now he can :) |
| 22:21.44 | carrar | heh |
| 22:32.18 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 22:40.01 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 22:41.10 | neurosys | I have my sound files in /usr/lib/asterisk/sounds/en ..... but 1.6.2.13 is still not reading them. Any thoughts? Is there a way from the CLI i can actually see where it thinks the files are? |
| 22:44.27 | pabelanger | neurosys: *CLI> core show settings |
| 22:44.51 | pabelanger | not sure if the paths are listed in 1.6.2 |
| 22:44.51 | pabelanger | maybe 1.8+ only |
| 22:46.44 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 22:48.37 | neurosys | pabelanger, what version are you running? |
| 22:49.20 | pabelanger | neurosys: 1.4 thru 1.8 |
| 22:52.07 | neurosys | Well Im running 1.6.2.13.. and i dont have that CLI command. |
| 23:11.36 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 23:17.25 | *** join/#asterisk Yedidya (~AndChat@nat79.mia.three.co.uk) |
| 23:18.56 | Yedidya | New asterisk 1.6.2.14 rpm not compatible with latest asterisk-addon-mysql rpm! |
| 23:19.40 | Yedidya | That should be asterisk16-addon-mysql |
| 23:20.45 | Yedidya | Who to talk to re asterisk CentOS report? |
| 23:26.24 | Yedidya | russellb |
| 23:28.35 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-103-152-61.cinci.res.rr.com) |
| 23:31.03 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 23:34.36 | Yedidya | Hey titter |
| 23:38.58 | *** join/#asterisk fofware (~Fabian@host186.190-225-12.telecom.net.ar) |
| 23:41.44 | Yedidya | R |
| 23:41.53 | Yedidya | Ru |
| 23:55.23 | *** join/#asterisk mlsmith9999 (~store@adsl-76-240-11-130.dsl.ltrkar.sbcglobal.net) |
| 23:58.11 | *** join/#asterisk corretico (~corretico@201.201.44.82) |
