| 00:00.46 | [TK]D-Fender | hangup handlers = pretty awesome |
| 00:00.55 | Kobaz | [TK]D-Fender: thanks :) |
| 00:01.10 | [TK]D-Fender | Kobaz: were you a contributer? |
| 00:01.14 | Kobaz | i wrote it |
| 00:01.20 | [TK]D-Fender | great idea |
| 00:01.23 | Kobaz | yeah |
| 00:01.38 | Kobaz | i didnt do the final revision, it got worked into the asterisk-11 way of doing things |
| 00:02.12 | [TK]D-Fender | we need presence handling next ... or SIP-B. Get to it! |
| 00:02.19 | Kobaz | fine fine |
| 00:02.36 | Kobaz | so wait, what sort of presence handling do you want? |
| 00:04.24 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 00:11.44 | [TK]D-Fender | Kobaz: when identifiable peer posts presence status update, it should trigger dialplan similar to how it not triggers for SIP messages. |
| 00:12.12 | [TK]D-Fender | That would allow to process "away" for dialplan-based DND, etc. |
| 00:13.08 | Kobaz | oooo |
| 00:13.58 | Kobaz | and what's SIP-B? |
| 00:14.09 | Kobaz | like the new pjsua channel driver? |
| 00:14.26 | volga629 | ok found in http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers that sip-->iax-sip not working |
| 00:14.29 | volga629 | :-) |
| 00:14.36 | volga629 | :-( (Note: SIP > IAX > SIP does not currently work for video calls as of Jan 08) |
| 00:14.44 | [TK]D-Fender | no, the spec the involves shared line appearances. |
| 00:15.05 | [TK]D-Fender | That'll help silence the "I want a key-system" people. |
| 00:15.11 | Kobaz | oh like an actual formal sip spec for shared lines |
| 00:15.21 | Kobaz | not the proprietary broadcom stuff |
| 00:15.36 | [TK]D-Fender | that is mostly standard actually... |
| 00:15.42 | [TK]D-Fender | (ish) |
| 00:15.52 | Kobaz | but dont you have to license it from broadcom |
| 00:17.41 | [TK]D-Fender | volga629: 2008 is 5 years ago. Get your ass off that decrpit wiki |
| 00:17.47 | [TK]D-Fender | decrepit |
| 00:18.26 | Kobaz | decrapid |
| 00:20.00 | Mon|A|rch | voip-info just seems like a repository of minsinformation now |
| 00:20.05 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 00:20.08 | Kobaz | well it's really really old |
| 00:20.11 | Kobaz | use the asterisk wiki |
| 00:20.15 | Mon|A|rch | yeah |
| 00:20.36 | Mon|A|rch | there are references on how to use n+101 |
| 00:20.54 | Kobaz | [TK]D-Fender: polycom making a many-button phone in the $100 price range would silence the i-want-a-key-system people as well |
| 00:22.19 | [TK]D-Fender | Kobaz: it's a question about how those "buttons" can be used... |
| 00:22.29 | Kobaz | yeah |
| 00:22.35 | [TK]D-Fender | Kobaz: proper SLA is a demanded thing.... |
| 00:22.36 | Kobaz | at least you could throw blfs on there |
| 00:22.44 | Kobaz | yeah you cant do proper sla |
| 00:22.48 | Kobaz | but it would be closer |
| 00:23.03 | [TK]D-Fender | Kobaz: well IP 600's had 6 keys. many other phone have more... but it isn't about BLF... |
| 00:23.11 | [TK]D-Fender | that's a consolation prize.... |
| 00:23.15 | Kobaz | i get a request for 'bridged lines' once a month |
| 00:23.24 | [TK]D-Fender | Except for receptionist-types |
| 00:23.28 | Kobaz | so i do a simul ring |
| 00:23.30 | Kobaz | and that shuts them up |
| 00:23.59 | [TK]D-Fender | it isn't dual-ring... it's shared-hold-pickup, + party-line |
| 00:24.07 | [TK]D-Fender | that's the golden cow |
| 00:24.53 | [TK]D-Fender | Now sure you can use parking and I accept that... but you've seen just how many "I don't wanna changed, WAHHHHHH!!!!" people come in here... |
| 00:25.11 | Kobaz | yeah |
| 00:25.12 | Kobaz | i know |
| 00:25.20 | Kobaz | i tried to train people on how to use a shared park |
| 00:25.22 | Kobaz | that didn't go well |
| 00:25.23 | [TK]D-Fender | Idiots are a regrettable part of reality, and there are specs to accomodat them :) |
| 00:25.32 | Kobaz | put park+blf on a button |
| 00:25.40 | [TK]D-Fender | yeah I know it works. |
| 00:25.51 | [TK]D-Fender | It's about making it natural for schmuscks |
| 00:25.57 | Kobaz | "look it's easy"... hit transfer, hit the button named park |
| 00:26.06 | Kobaz | the other person hits the park button that's now red |
| 00:26.15 | Kobaz | move the call back, same thing |
| 00:26.17 | [TK]D-Fender | Stop selling that idea to me... I'm the producer :p |
| 00:26.28 | Kobaz | well yeah, im just saying |
| 00:26.32 | Kobaz | people can't wrap their heads around the extra button push |
| 00:26.53 | Kobaz | i mean that's what i say to the customer |
| 00:26.58 | Kobaz | i say to them "hey do this" |
| 00:27.06 | [TK]D-Fender | *I* know... *they* are stuck in their ways and should be tossed on a curb.... except we don't usualy get that luxury... |
| 00:27.17 | Kobaz | and they look at me like i'm speaking ancient Aramaic |
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| 01:01.07 | Mon|A|rch | Kobaz, you probably should just go ahead and speak ancient aramaic |
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| 01:33.32 | Kobaz | hehe |
| 01:34.07 | Kobaz | yay, added voicemail to my configurator |
| 01:34.26 | Kobaz | almost done with my fully-self-service configuration tool for the peoples |
| 01:48.09 | pabelanger | Kobaz, what are you using? |
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| 02:25.02 | floren | that made me LOL: < Kobaz> and they look at me like i'm speaking ancient Aramaic |
| 02:25.05 | floren | heh |
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| 03:37.28 | igcewieling | Polycom bug fixed: 84975/84908 In the locked state, the phone no longer initiates the call to the emergency 911 upon off-hooking the handset twice in quick succession, i.e., 2 seconds |
| 03:37.37 | igcewieling | I'm glad I didn't have to troubleshoot THAT issue |
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| 04:42.45 | blehxor | lol, just came back and saw the polycom bug.. |
| 04:44.27 | blehxor | is it possible to have two MixMonitors on one channel? Looking to have one instance record the entire call and then run several to capture specific segments. |
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| 05:34.58 | wdoekes | blehxor: I think you can.. you might need an extra local channel though |
| 05:35.38 | wdoekes | (so then it wouldn't multiple monitors on the same channel technically, but on the same leg) |
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| 06:10.07 | blehxor | ah got it, thanks wdoekes |
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| 08:22.40 | bombev | :) hello guys |
| 08:22.55 | bombev | what is the AMI command to set CDR destination channel |
| 08:23.18 | bombev | or just destination |
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| 08:39.15 | kaldemar | bombev: CDR(dst) is read-only, you can't set it. |
| 08:42.13 | bombev | :) |
| 08:42.16 | bombev | ho I forgot |
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| 08:43.38 | bombev | kaldemar but CDR(src) is not read only |
| 08:43.39 | bombev | correct? |
| 08:45.29 | kaldemar | not correct. see what "core show function CDR" tells you. |
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| 09:03.20 | domi | hi, is there a simple way to limit incoming calls to an extensions which has more than one device? besides counting curcalls of each device... |
| 09:04.12 | domi | i need something like call-limit but only on incoming calls and over more than one device.... some kind of "busy on busy" |
| 09:08.13 | domi | one way is to use groups, but the promblem with groups is, if i deny the call in the dialplan, i cannot use call completion, for CC you must dial a peer, so the subsystem know which device to monitor |
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| 09:31.43 | Ice_Strike | What the AMI would appear for incoming call? |
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| 09:55.54 | Greenlight | Ice_Strike: NewChannel ? |
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| 09:59.56 | Ice_Strike | Hmm |
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| 10:00.27 | Ice_Strike | Greenlight What about on Asterisk CLI to detect for incoming call? |
| 10:00.53 | Greenlight | There is a similar message indicating a new channel has been created. |
| 10:00.58 | Greenlight | What exactly are you trying to do? |
| 10:02.00 | Ice_Strike | Well just trying out if incoming call be detected |
| 10:02.18 | Moz | Hi all. Does anyone know where I can buy / get the G729.1 codec for asterisk? I tried Digium but it seems it is only G729 (aka G729a) |
| 10:03.03 | Greenlight | Ice_Strike: Yes, by "NewChannel" |
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| 10:03.48 | Greenlight | That event is fired every time a "call" comes in, as a channel is created. |
| 10:04.52 | Greenlight | Moz: Isn't it just the same codec, labelled differently? (G729.1 / G729a) |
| 10:05.11 | domi | no ideas for limiting calls to an extensions with more than one device? |
| 10:05.15 | Ice_Strike | Hmm not getting any NewChannel |
| 10:08.55 | Greenlight | DO you have events enabled? |
| 10:09.43 | Ice_Strike | on the CLI? |
| 10:10.02 | Greenlight | No, the AMI |
| 10:10.13 | Greenlight | [10:31am] <Ice_Strike> What the AMI would appear for incoming call? <-- YOu asked about AMI. |
| 10:10.14 | Ice_Strike | Yes |
| 10:10.39 | Greenlight | Then you should check what permissions are being given in manager.conf |
| 10:11.11 | Moz | Greenlight - not G729.1 is variable and G729a is 8bit/s |
| 10:11.20 | Greenlight | Ahh I see |
| 10:12.11 | coppice | G.729.1 is wideband |
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| 11:59.44 | avb | hey guys, is there is any commercial g729 implementations available for raspberry pi arm board? |
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| 13:51.13 | saliak | in my main menu i have options 1-4 + internal extensions that match _1XXX. For some reason, however, WaitExten returns 1100 when i dial 1000. (http://pastebin.com/Q3Yku2m0). Any ideas? |
| 13:51.54 | igcewieling | sounds like you have relaxdtmf enabled |
| 13:52.50 | igcewieling | also you do not want extensions to overlap with IVR options or you will experience delays when select any options which overlap extensions |
| 13:53.45 | saliak | igcewieling: ok. |
| 13:55.00 | saliak | igcewieling: do i set relaxdtmf for my main trunk then? or do i need to set it for each peer? |
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| 13:56.41 | igcewieling | you remove it if you have it set |
| 13:57.08 | igcewieling | I don't know about peers and trunks, but you should remove it from the sip.conf entries. |
| 13:57.18 | saliak | igcewieling: yeah, just found that. not enabled |
| 13:57.45 | igcewieling | show us the output of a failed call with dtmf debugging enabled in logger.conf |
| 13:58.01 | igcewieling | hopefully someone will want to spend the time to look into it. |
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| 14:41.21 | thansen | how can I prevent this (aside from firewall obviously) http://pastebin.com/Dj494ixe |
| 14:42.07 | igcewieling | you use a firewall (aka iptables) with fail2ban |
| 14:42.13 | thansen | I see no peer/connection from that IP so I'm a bit cloudy as to how that can even be initiated |
| 14:42.35 | thansen | igcewieling: I do have that running but it must be missing that pattern for some reason |
| 14:42.40 | igcewieling | thansen: looks like you have allowguest=yes or something like that |
| 14:42.46 | thansen | but how does it happen in the first place |
| 14:42.53 | igcewieling | thansen: then you should be asking the fail2ban people why it isn't matching |
| 14:43.44 | thansen | igcewieling: relax, I |
| 14:43.53 | igcewieling | thansen: it happens because you don't have allowguest=no |
| 14:43.54 | thansen | 'm not asking why fail2ban isn't working |
| 14:44.02 | thansen | ah, in sip.conf? |
| 14:44.17 | igcewieling | correct. It is not a good idea to allow guest calls. |
| 14:44.37 | thansen | ok, I don't have that at all in sip.conf so the default is yes? |
| 14:44.47 | thansen | meaning, I don't have that option at all |
| 14:44.52 | igcewieling | thansen: depends on the asterisk version. set it to no |
| 14:45.07 | thansen | this goes in 'general' right? |
| 14:45.21 | igcewieling | Before Asterisk 11 (or was it 10 or was it 1.8) it defaulted to yes. |
| 14:45.24 | igcewieling | thansen: correct. |
| 14:45.39 | igcewieling | you REALLY should read the Asteisk book |
| 14:45.41 | igcewieling | ~book |
| 14:45.42 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:45.47 | thansen | igcewieling: excellent, lemme try a reload and see if it shuts it down |
| 14:49.01 | *** join/#asterisk timahvo1 (~rogue@197.237.174.197) |
| 14:51.28 | thansen | igcewieling: now I'm getting a slew of sending fake auth rejection messages..any quick tips before I start hitting google? |
| 14:51.48 | igcewieling | thansen: it is going exactly what you told it. |
| 14:52.04 | igcewieling | someone is trying to hack into your server, this is what you should expect. |
| 14:52.09 | thansen | ok |
| 14:52.25 | igcewieling | now you need to figure out why fail2ban isn't banning them |
| 14:52.28 | thansen | so they're trying to auth as these extensions incrementing by 1 each time |
| 14:52.38 | thansen | nods |
| 14:52.45 | igcewieling | thansen: yup. Expect one of the attacks per week |
| 14:53.20 | igcewieling | We solved the issue using a method most people can't use: block all off-net traffic. |
| 14:53.55 | thansen | understood |
| 14:54.30 | igcewieling | If you are running a GUI you will want to prevent remote access to it or you'll be hacked in no time |
| 14:55.29 | thansen | thankfully no |
| 14:56.00 | thansen | surprised I haven't hit this before as I've had this server running for ~6 years now |
| 14:56.07 | Greenlight | It can also be helpful to limit IP ranges that you allow in. |
| 14:56.24 | Greenlight | For example, we seen a big drop in hacking attempts when we started to only allow UK IP's |
| 14:56.52 | thansen | yeah, pfsense will help with that once I get this behind the hw firewall |
| 14:57.02 | Greenlight | +1 that's what we use |
| 14:58.48 | thansen | this notice message doesn't have the IP of the offender |
| 14:59.30 | igcewieling | thansen: only for some types of rejection messages. One of Asterisk's little annoyances |
| 14:59.54 | Greenlight | I thought they'd fixed that. |
| 15:00.03 | Greenlight | You running latest version yeah? |
| 15:00.04 | *** join/#asterisk vlad_starkov (~vlad_star@85.202.230.46) |
| 15:00.09 | igcewieling | Greenlight: he is running an old version of Asterisk. |
| 15:00.11 | saliak | thansen: i used this guide for failtoban rules http://www.fail2ban.org/wiki/index.php/Asterisk. I had to change the rules a bit to hit the random ports that started getting added in. needed to make <HOST> -> <HOST>.* |
| 15:00.22 | igcewieling | modern Asterisk's default to allowguest=no |
| 15:00.52 | Greenlight | Ahh that explains it then |
| 15:01.31 | Greenlight | I do remembre this being a "gotcha" when using fail2ban with old old asterisk |
| 15:01.53 | thansen | yeah, I haven't updated in a bit |
| 15:02.14 | Greenlight | If your open to public internet, I would really suggest staying updated. |
| 15:02.40 | thansen | hmm |
| 15:02.45 | thansen | actually I'm running 11 |
| 15:03.09 | thansen | thought I was running 1.8 still |
| 15:03.27 | thansen | 11.2.1 |
| 15:03.34 | Greenlight | The rejection messages in 11 should have the IP in them |
| 15:03.51 | thansen | I have an ip, it's my local ip though |
| 15:04.07 | Greenlight | heh not much fail2ban can do with that :) |
| 15:04.20 | Greenlight | I was sure I read somewhere about that issue being fixed |
| 15:04.28 | [TK]D-Fender | sure it can.. your server can start banning itself :) |
| 15:04.29 | thansen | well, the fake auth rejection isn't in the patterns anyway |
| 15:06.28 | thansen | http://bpaste.net/show/123007/ |
| 15:06.36 | igcewieling | Greenlight: I thought so too, but since we don't use fail2ban anymore I never verified. |
| 15:07.09 | igcewieling | thansen: see the security events log, that should have the IP and other needed info. |
| 15:07.16 | Greenlight | http://forums.asterisk.org/viewtopic.php?t=74947#p160291 <-- That suggests you should always get IP address now. |
| 15:07.26 | *** join/#asterisk ChadAragorn (~ChadArago@206.251.40.221) |
| 15:08.06 | igcewieling | Greenlight: that only covers registration. If the attacker tries placing a call without registering, then you get that useless message |
| 15:08.16 | igcewieling | (that useless message == the one without the IP) |
| 15:08.20 | Greenlight | AHh yea I just scrolled that far down now ;) |
| 15:08.36 | igcewieling | Greenlight: heh, I didn't scroll down. |
| 15:08.40 | Greenlight | There's a fix, if you wanna edit source code.... |
| 15:08.41 | thansen | not the message I'm getting |
| 15:08.41 | thansen | http://pastebin.com/KFpqM1SZ |
| 15:09.29 | thansen | I don't see any logs aside from messages, queue_log and event_log |
| 15:09.30 | igcewieling | Greenlight: the fix is to use the security log feature |
| 15:09.43 | Greenlight | http://forums.asterisk.org/viewtopic.php?t=74947#p183312 |
| 15:09.46 | Greenlight | That post |
| 15:09.47 | igcewieling | thansen: see logger.conf.sample, see in that file the security level |
| 15:09.51 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 15:11.33 | thansen | LOL, mofo |
| 15:11.53 | thansen | checks logger.conf |
| 15:12.11 | kaldemar | that "fix" adds a whole new log entry when the existing one could be modified to show the attacker's ip. |
| 15:12.27 | Greenlight | Yea, that did seem "backwards" to me |
| 15:12.34 | kaldemar | anyway, the approach to modify source is poor since security events are generated for those. |
| 15:13.07 | Greenlight | Yea, if security events cover it then all is good. I presumed it wasn't showing in those though since it had already been suggested. |
| 15:13.26 | *** join/#asterisk wasanzy (~wasanzy@41-66-254-58-dedicated.4u.com.gh) |
| 15:13.45 | wasanzy | when a peer is said to be unmonitored, what does that mean? |
| 15:13.50 | thansen | well, I was watching the console which isn't showing those events |
| 15:14.09 | Greenlight | http://www.fail2ban.org/wiki/index.php/Asterisk <-- You didn't configure like this ? |
| 15:14.20 | thansen | added a the security option to logger.conf and now I'm getting something useful here |
| 15:14.22 | Greenlight | wasanzy: Means that it's not monitored. |
| 15:14.31 | igcewieling | wasanzy: it means you do not have qualify= set |
| 15:14.44 | Greenlight | wasanzy: If it's turned off or disconnects ungracefully, you won't know |
| 15:15.38 | wasanzy | igcewieling: in this case what do I have to do? perhaps what is qualify = set |
| 15:15.58 | Greenlight | wasanzy: You don't "have" to do anything. Nothing is "wrong". |
| 15:16.13 | igcewieling | wasanzy: I don't understand the question. Do you want qualify or not? |
| 15:18.57 | thansen | Greenlight: ok, got those tweaked now |
| 15:18.59 | igcewieling | suspects qualify is an alias for randomlydisconnectmypeers |
| 15:19.43 | thansen | time and sending everything to messages, I just need a filter that will check these security messages now for fail2ban |
| 15:19.58 | thansen | anyone have one sitting around? |
| 15:20.14 | Greenlight | Scroll down that page... |
| 15:20.46 | wasanzy | igcewieling: No the problem I have is am trying to monitor the peers with nagios and the plugin is returning ERROR: outgoing: Unmonitored |
| 15:21.11 | Greenlight | So, you *do* want to qualify them. |
| 15:21.18 | Greenlight | Since you want to MONITOR them.... |
| 15:21.30 | thansen | Greenlight: none of those match this.. http://pastebin.com/6RnhWKcE |
| 15:21.47 | wasanzy | Greenlight: yes, but will qualifying them affect something? |
| 15:22.17 | Greenlight | Yes, they'll be monitored, and show as unavaialble it they disconnect |
| 15:22.50 | wasanzy | if that then, what is the directive? |
| 15:23.02 | *** join/#asterisk navaismo (~navaismo@189.241.68.101) |
| 15:23.33 | igcewieling | wasanzy: Perhaps you misunderstand. We are not here to teach you everything. |
| 15:23.49 | igcewieling | Try looking at the sip.conf.sample file for examples of using qualify. |
| 15:24.13 | Greenlight | Or even an educated guess as to what you might do to enable qualify. |
| 15:25.47 | saliak | from what i've read, it looks like WaitExten is setup for 1 digit and Read should be used for more than one. so in my example, since my extensions are 4 digits and my IVR extensions are 1 digit, I need to use Read. does that sound right? |
| 15:26.11 | igcewieling | ~book |
| 15:26.11 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:26.45 | navaismo | no, wiatexten != read |
| 15:27.14 | thansen | Greenlight: http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk |
| 15:27.24 | thansen | that has some regex for the SECURITY entries |
| 15:28.01 | *** join/#asterisk pi3r (~pi3r@86.188.229.227) |
| 15:28.02 | igcewieling | saliak: that in entirely incorrect |
| 15:28.06 | Greenlight | Wow something up-to-date on voipinfo :) |
| 15:28.54 | *** part/#asterisk rrittgarn (~rrittgarn@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 15:29.15 | malcolmd | Greenlight: say what?!?!? ;) |
| 15:30.32 | Greenlight | I even F5'd it three times to be sure. Maybe they were hacked |
| 15:31.11 | igcewieling | up to date? UNPOSSIBLE! |
| 15:34.49 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 15:37.52 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:42.47 | *** join/#asterisk jsjc (~Adium@39.Red-2-136-115.dynamicIP.rima-tde.net) |
| 15:51.20 | thansen | igcewieling, Greenlight: thanks both for the help, I've got the 'mofos' shut down now :) |
| 15:51.40 | Greenlight | Good news :) |
| 15:51.53 | thansen | yeah, time to get it all documented :) |
| 15:52.12 | saliak | igcewieling: hrm. this only happens the first time through WaitExten. If you enter an invalid address and it cycles through WaitExten a 2nd time, then it goes through correctly (8000 = 8000, rather than 8800). weird.. |
| 15:56.35 | *** part/#asterisk domi (~Thunderbi@p578b9696.dip0.t-ipconnect.de) |
| 16:08.59 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:09.00 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:18.11 | Penguin | saliak: Show us. |
| 16:18.43 | igcewieling | (9:57:45 AM) igcewieling: show us the output of a failed call with dtmf debugging enabled in logger.conf |
| 16:18.43 | igcewieling | (9:58:00 AM) igcewieling: hopefully someone will want to spend the time to look into it. |
| 16:18.50 | igcewieling | Penguin: already requested. |
| 16:19.35 | igcewieling | Sounds almost like he did something stupid like set the dtmf mode to auto |
| 16:19.36 | Penguin | saliak: WaitExten doesn't "cycle." You enter an extension while WaitExten waits, and then it goes there. If it is invalid, extension i will be executed. If exten i is missing, it should look for e. If exten e is missing, the call should end. |
| 16:22.08 | *** join/#asterisk bdar77 (~bad53@2.90.75.61) |
| 16:22.45 | saliak | Penguin: here's the CLI output http://pastebin.com/AvUwPjBz. You are correct, my dial plan cycles if the extension is invalid. My diaplan - http://pastebin.com/4q5acNmN |
| 16:23.11 | *** join/#asterisk felipealmeida (~user@querubim.tecgraf.puc-rio.br) |
| 16:23.21 | saliak | Penguin: in this example, i call in and dial "8000". the first time around, it thinks that it's 8800. the 2nd time, it thinks that it's 8000 |
| 16:25.48 | [TK]D-Fender | saliak: You've also spend these past several hours telling us NOTHING about what this call is coming in from. as in everything along the path. tech, protocols, codecs, no configs to speak of, what about the actual phone punching the DTMF? |
| 16:28.20 | Penguin | I'm also curious if the behavior is consistent. Maybe this only happened one time, or maybe only when calling from one particular phone. |
| 16:32.25 | *** join/#asterisk threesome (~threesome@ip-94-113-12-233.net.upcbroadband.cz) |
| 16:33.08 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 16:35.17 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:ac78:d36a:e6dc:5da2) |
| 16:36.44 | saliak | igcewieling: DTMF debug - http://pastebin.com/EVsGhTED |
| 16:36.45 | saliak | Penguin: |
| 16:38.03 | saliak | Penguin: good question! this seems to happen (given my limited test set) from a voyage phone. does not happen with my cell phone. |
| 16:38.17 | saliak | Penguin: the behavior, when it happens, is very consistent |
| 16:38.30 | Penguin | It happens every time... as long as you call from that one phone? |
| 16:38.39 | saliak | Penguin: again, it happens the first time through WaitExten. |
| 16:38.42 | saliak | Penguin: yes |
| 16:38.44 | Penguin | Scrap that phone. |
| 16:40.06 | saliak | doesn't it seem weird that it only happens the firs time through wait extent? also, if i do a single digit extension, it works the first time through |
| 16:42.16 | Penguin | I've had similar happen to me, but it wasn't consistent. |
| 16:42.25 | Penguin | It would happen most of the time, but not every time. |
| 16:42.44 | igcewieling | saliak: yes it is weird. no I don't know what it means |
| 16:44.13 | Penguin | And it wasn't from just one phone. |
| 16:44.39 | *** join/#asterisk flapjacks (~flapjacks@wsip-184-183-148-253.ph.ph.cox.net) |
| 16:45.51 | igcewieling | pastebin the sip.conf [general] and the voip-ms sections removing ONLY the password |
| 16:49.12 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 16:49.19 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 16:49.19 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:50.29 | wasanzy | am using check_asterisk_channels to monitor asterisk channels, but getting this error on nagis: "Unable to connect to remote asterisk(does /var/run/asterisk/asterisk.ctl ) exist" |
| 16:50.29 | wasanzy | <PROTECTED> |
| 16:50.48 | wasanzy | and when I run the command manually in terminal, /usr/local/nagios/libexec/check_asterisk_channels |
| 16:50.48 | wasanzy | <PROTECTED> |
| 16:50.48 | wasanzy | <PROTECTED> |
| 16:50.54 | saliak | igcewieling: http://pastebin.com/X6K6Eaa3 |
| 16:51.09 | igcewieling | saliak: how many g729 licenses do you have? |
| 16:51.09 | WIMPy | Permissions? |
| 16:51.09 | saliak | igcewieling: 2 |
| 16:51.50 | igcewieling | saliak: I don't see anything obviously wrong. Have you tried a different ITSP? |
| 16:52.02 | saliak | igcewieling: ITSP? |
| 16:52.16 | igcewieling | ~itsp |
| 16:52.16 | 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. |
| 16:52.41 | wasanzy | WIMPy: are you refering to me? |
| 16:52.50 | *** join/#asterisk War_Bear (~War_Bear@warbear.co.uk) |
| 16:52.58 | avb | hey guys, is there is any commercial g729 implementations available for raspberry pi arm board? |
| 16:53.01 | *** join/#asterisk Zopieux (zopieux@jerrycraft.tk) |
| 16:53.03 | Zopieux | hello |
| 16:53.22 | avb | i will ask again, as room is more active now :) |
| 16:53.25 | igcewieling | avb: no. |
| 16:53.40 | igcewieling | There is only g729 codec for Asterisk and it is sold by Digium. |
| 16:53.40 | avb | damn, thats bad :) |
| 16:53.51 | saliak | igcewieling: so, someone other than voipms? no. seems like it'd be more consistent across other phones as well if it was the trunk provider, eh? it's strange that it's so consistent on that one phone (again, all that I can test). i updated my plan to read back the extension to the caller, so maybe htye'll just think they mis-keyed it |
| 16:54.02 | Zopieux | I've been struggling to determine what sip channels (queue operators) are "busy", ie. talking to someone |
| 16:54.05 | avb | igcewieling: im sorry, but it is wrong |
| 16:54.08 | Penguin | It is possible that they have one for ARM. Go to the site and check. |
| 16:54.21 | Zopieux | "queue show foo" only shows if operators are online, offline or paused |
| 16:54.24 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 16:54.34 | [TK]D-Fender | zothey show "On Call" as well |
| 16:54.45 | [TK]D-Fender | Zopieux: they show "On Call" as well |
| 16:54.55 | avb | i know at least one more g729 codec sources which are not from digium |
| 16:55.03 | avb | but for x86 :( |
| 16:55.18 | Zopieux | currently I'm parsing the output of sip show channels, as the last column has the name of the operator, but it's very fuzzy (it's truncated to 10 chars for instance) |
| 16:55.42 | Zopieux | [TK]D-Fender, it shows "Not in use" whether he's talking or idling |
| 16:55.45 | avb | Zopieux: wouldnt be easier to parse queue show XXX ? |
| 16:55.53 | [TK]D-Fender | Zopieux: "core show channels concise" |
| 16:55.54 | igcewieling | avb: that is an illegal codec and discussing such things on this channel or the mailing list will get you banned. |
| 16:55.59 | igcewieling | I recommend you do not do so. |
| 16:56.12 | avb | im not distributing |
| 16:56.22 | Penguin | Did you already check Digium's site to see if they have an ARM codec? |
| 16:56.29 | avb | yeh :( |
| 16:56.29 | [TK]D-Fender | Zopieux: and it sounds like you failed to specify a state device for the member |
| 16:56.30 | Zopieux | avb, as I said, it shows "Not in use" if online, but both if they're wanting or answering calls! |
| 16:56.39 | igcewieling | avb: That might be debatable. What is not debatable is talking about it here. |
| 16:56.41 | [TK]D-Fender | Zopieux: and it sounds like you failed to specify a state device for the member <-- |
| 16:56.43 | boom^time | Hey guys, is festival still the go to for voice synth these days? |
| 16:56.44 | Zopieux | maybe [TK]D-Fender |
| 16:56.52 | avb | igcewieling: ok, np |
| 16:56.58 | Zopieux | lemme some time to answer you! I read every message |
| 16:57.04 | [TK]D-Fender | boom^time: Has not been for a long time. Cepstal is significantly better |
| 16:57.09 | *** join/#asterisk SuperNull (~YoMommaSo@24-148-101-238.ip.mhcable.com) |
| 16:57.14 | Zopieux | how could I define such a state device? |
| 16:57.16 | boom^time | [TK]D-Fender, thank you |
| 16:57.49 | SuperNull | hey gents, i have autoload=no in my modules.conf yet, its loading things that have explicit noload => ... (specifically res_adsi.so) any ideas why this happens ? |
| 16:57.54 | Zopieux | in my queue conf, I use member => SIP/foobaragent |
| 16:57.56 | boom^time | [TK]D-Fender, is it not free? |
| 16:58.27 | [TK]D-Fender | boom^time: No, but fairly inexpensive |
| 16:58.48 | WIMPy | wasanzy: Yes, that was for you. |
| 16:58.54 | boom^time | Are there any alternatives? This is just temporary holders for messages |
| 16:59.03 | boom^time | I wouldn't use synth permanently. |
| 16:59.46 | igcewieling | boom^time: then record the files yourself |
| 17:00.07 | boom^time | tedious, but I may. |
| 17:00.16 | Penguin | Google has a free offering. |
| 17:00.40 | boom^time | I enjoyed just using an app in my dialplan: Festival("how are you doing?") |
| 17:00.41 | Penguin | That isn't to say that it is good. |
| 17:00.42 | wasanzy | WIMPy: I have given 777 permission to the file so am not sure that is the problem |
| 17:00.50 | boom^time | quick and simple |
| 17:01.35 | Penguin | wasanzy: ls -dl /var/run/asterisk |
| 17:02.03 | igcewieling | Asterisk recently changed something with the console, causes FreePBX to think it failed to start Asterisk. It is harmless. |
| 17:02.16 | *** join/#asterisk Dovid (~Dovid@ool-457f6ac5.dyn.optonline.net) |
| 17:02.26 | Penguin | I couldn't care less about FreePBX anyway. |
| 17:02.29 | wasanzy | drwxrwxrwx. 2 asteriskpbx asteriskpbx 4096 Aug 9 21:10 /var/run/asterisk |
| 17:02.38 | igcewieling | maybe you didn't to a make config when you installed Asterisk? |
| 17:02.49 | igcewieling | Penguin: exact same symptom |
| 17:03.07 | Penguin | wasanzy: ps -C asterisk u |
| 17:03.47 | wasanzy | root 2336 0.2 0.1 1818408 24656 ? Sl Jul13 114:49 /usr/sbin/asterisk -f -vvvg -c |
| 17:04.01 | Penguin | Why are you running asterisk as root? |
| 17:04.09 | SuperNull | ownage. |
| 17:04.23 | igcewieling | Penguin: have you seen his previous questions? I imagine that is why |
| 17:04.44 | wasanzy | Penguin: could that be the problem? |
| 17:04.50 | Penguin | The only question I saw was why can't the console connect to the running asterisk process. |
| 17:05.14 | Penguin | wasanzy: It is _a_ problem, but not likely _the_ problem. |
| 17:05.29 | wasanzy | Penguin: ok I will change the user |
| 17:06.08 | igcewieling | (11:13:45 AM) wasanzy: when a peer is said to be unmonitored, what does that mean? [discussion of qualify] (11:22:50 AM) wasanzy: if that then, what is the directive? |
| 17:06.21 | igcewieling | that is just the most recent one I found. |
| 17:06.48 | Penguin | I can't read everything. |
| 17:06.48 | boom^time | hah, nm, festival still works great |
| 17:06.54 | boom^time | I just had to remember how to configure |
| 17:07.00 | wasanzy | igcewieling: no that wasn't the recent one |
| 17:07.01 | Penguin | Great? It never worked great. |
| 17:07.11 | igcewieling | questions like that scream "I DIDN'T READ THE BOOK!" to me. |
| 17:07.20 | wasanzy | it goes like this: |
| 17:07.36 | wasanzy | am using check_asterisk_channels to monitor asterisk channels, but getting this error on nagis: "Unable to connect to remote asterisk(does /var/run/asterisk/asterisk.ctl ) exist" |
| 17:07.36 | wasanzy | <wasanzy> when I check the server, /var/run/asterisk/asterisk.ctl actually exist |
| 17:07.47 | wasanzy | and when I run the command manually in terminal, /usr/local/nagios/libexec/check_asterisk_channels |
| 17:07.48 | wasanzy | <wasanzy> result: OK: 479 active channels, Setting active calls | 2013-08-14 16:48:42 479 Setting |
| 17:07.48 | wasanzy | <wasanzy> something seem wrong somewhere |
| 17:08.00 | wasanzy | that is the most recent one |
| 17:08.05 | saliak | Is it possible to have multiple clients login as a single SIP user? So, say, for support, if we have 5 people managing that line, they just need a single sip account. when it's called, they would all ideally ring, and one person gets the call. That seems sketchy, so I'm assuming the answer is no and that i need to setup a simultaneous ring group. If that's the case, how can I make it obvious that the call is coming in from the "Support" |
| 17:08.05 | saliak | line, versus the "Sales" line on each phone (given they have their own extension)? |
| 17:08.11 | igcewieling | wasanzy: I was referring to your previous question. your CURRENT question is not something you could read about. |
| 17:08.14 | Penguin | Oh, maybe I misunderstood. I thought you were trying to asterisk -r to the console. |
| 17:08.15 | boom^time | penguin, are you not a fan of Professor Hawking? |
| 17:08.22 | *** part/#asterisk thansen (~thansen@63-248-113-89.static.ogdl0101.digis.net) |
| 17:08.47 | wasanzy | Penguin: no |
| 17:08.49 | igcewieling | saliak: No! |
| 17:09.08 | Penguin | saliak: No, use queues. |
| 17:09.14 | Zopieux | thanks [TK]D-Fender I forgot the "callcounter=yes" option |
| 17:09.20 | igcewieling | one device = one sip.conf entry |
| 17:09.22 | Zopieux | it's now perfect |
| 17:10.14 | Penguin | saliak: You can display a message on a computer screen, or you could alter the callerid name. |
| 17:10.18 | *** join/#asterisk Mon|A|rch (~sbean@72.29.180.35) |
| 17:10.20 | Mon|A|rch | quick question |
| 17:10.35 | [TK]D-Fender | wasanzy: Go connect to * CLI from CLI yourself |
| 17:10.37 | saliak | Penguin: ok, yeah, I was thinking the callerid name was the way to go |
| 17:10.41 | Mon|A|rch | when I originate a call, i get the "successfully queued" response once the line is picked up |
| 17:11.06 | wasanzy | Penguin: the plugin does "asterisk -rx "core show channels" " |
| 17:11.08 | Mon|A|rch | if I route it to a local context, then call from there, will I get the response once it gets into the context? or when the first leg is made |
| 17:11.09 | igcewieling | saliak: you could also have a 2nd line / account on the phone for Support calls |
| 17:11.30 | saliak | igcewieling: Ah, i see, that's the "agent" account? |
| 17:11.33 | Mon|A|rch | or will I not get that response at all |
| 17:11.55 | [TK]D-Fender | Mon|A|rch: you get it at the same time |
| 17:11.55 | SuperNull | anyone know if asterisk 1.8.23 still supports asterisk.conf: [compat] pbx_realtime=1.4 ? |
| 17:11.56 | saliak | igcewieling: so basically anyone that's a support person gets their own extension, as well as their support call extension |
| 17:12.01 | igcewieling | saliak: call it what you want, it is a plain sip account. For example my phone registers to 5 different SIP accounts |
| 17:12.36 | saliak | igcewieling: yeah. ok, a pain, but makes sense. |
| 17:12.39 | igcewieling | office PBX, one for each of our core servers, and one for my personal line |
| 17:13.18 | Penguin | I don't think you can connect your phone directly to an agent channel. You have to use a sip channel first, then connect the sip account to an agent channel. (If you want to use agent channels at all.) |
| 17:13.30 | wasanzy | I did asterisk -rx "core show channels" in the terminal and got this result: http://paste.scsys.co.uk/265657 |
| 17:14.30 | [TK]D-Fender | wasanzy: if your script is failing then the USER it is running as does not have PERMISSIONS set properly |
| 17:14.37 | Penguin | wasanzy: Nagios probably does not have permission to execute asterisk -r. |
| 17:14.39 | [TK]D-Fender | wasanzy: this is basic UNIX |
| 17:14.46 | [TK]D-Fender | wasanzy: Go fix your permissions & script |
| 17:15.21 | igcewieling | SuperNull: all major changes like not supporting an option should be documented in the UPGRADE*.txt files. |
| 17:15.30 | wasanzy | [TK]D-Fender>: my problem is when I run the plugin manually, it works |
| 17:15.48 | igcewieling | wasanzy: run it as root or as the nagios user? |
| 17:16.07 | igcewieling | This is STILL basic UNIX |
| 17:16.18 | wasanzy | igcewieling: runned it as root man ually |
| 17:16.32 | igcewieling | wasanzy: then it is a useless test |
| 17:16.38 | igcewieling | unless you run nagios as root |
| 17:16.52 | [TK]D-Fender | which is shouldn't be in any sane install |
| 17:16.53 | Penguin | I don't think you're supposed to do that, either. |
| 17:17.02 | [TK]D-Fender | it normally is run as USER "nagios" |
| 17:17.17 | Penguin | Why everyone wants to run all their services as root is beyond me. |
| 17:17.19 | igcewieling | I don't think wasanzy would benefit from security related advice |
| 17:17.25 | igcewieling | Penguin: lazyness |
| 17:17.28 | wasanzy | igcewieling: nagios is running as nagios user |
| 17:17.40 | igcewieling | wasanzy: then run your test as the nagios user |
| 17:17.47 | Penguin | Then any test you do needs to be nagios user, too. |
| 17:17.48 | *** join/#asterisk apb1963_ (~apb1963@174.134.118.16) |
| 17:18.02 | Mon|A|rch | [TK]D-Fender, gotcha |
| 17:18.03 | wasanzy | ok, I guess you are right |
| 17:18.05 | Mon|A|rch | [TK]D-Fender, thanks |
| 17:18.22 | Penguin | We already know the result of the test. |
| 17:18.38 | Penguin | nagios shouldn't have permission to use asterisk -r. |
| 17:19.15 | igcewieling | Penguin: things like this is why http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-13-SECT-4.html |
| 17:19.49 | igcewieling | or this http://www.voip-info.org/wiki/view/Asterisk+non-root |
| 17:20.20 | Penguin | Those should tell you how to NOT run as root. |
| 17:23.01 | igcewieling | Penguin: that is an issue for Digium to resolve. |
| 17:23.19 | Penguin | I can't understand what you're saying. |
| 17:24.17 | igcewieling | Penguin: by default Asterisk runs as root. It takes a lot of work to make it run as non-root. Therefore if you want Asterisk to run as non-root, someone would have to make changes to the Asterisk build system or source code to do so. Who gets to do that? Digium. |
| 17:24.35 | igcewieling | (to run as non-root by default) |
| 17:24.49 | dpeloquin | or, you know, anyone else who wants to contribute... |
| 17:24.50 | Penguin | It doesn't take THAT much work to run as not root. |
| 17:25.02 | Penguin | It's fairly easy, actually. |
| 17:25.03 | asteriskmonkey | yeah its really easy to get running as non-root |
| 17:25.10 | igcewieling | Penguin: looks like it from the voip-info.org page. |
| 17:25.24 | asteriskmonkey | voip-info is like dated as old as dinorausrs in tech |
| 17:25.53 | dpeloquin | so is the 2nd edition of the book, while we are talking about stale documentation |
| 17:26.25 | Penguin | 2nd edition is good if you still use an antique asterisk version. |
| 17:26.27 | igcewieling | dpeloquin: the 2nd edition was the first google result. |
| 17:26.56 | igcewieling | Anyway, you asked why people run Asteirsk as root, you have my view on it. |
| 17:27.24 | Penguin | I guess the primary reason is ignorance. |
| 17:27.35 | Penguin | The secondary reason is probably laziness. |
| 17:27.37 | igcewieling | lazyness |
| 17:27.46 | asteriskmonkey | lazyness for the win |
| 17:31.25 | [TK]D-Fender | Enabling AMI and failing to protect it against attack would be a death-sentence for your server |
| 17:31.59 | igcewieling | [TK]D-Fender: that applies regardless of what user you run it as |
| 17:32.11 | [TK]D-Fender | worse as root itself |
| 17:32.13 | asteriskmonkey | putting a voip pbx on the public internet without some fencing is the same |
| 17:32.24 | [TK]D-Fender | but yes, AMI is a terrifying risk all by itself |
| 17:32.25 | asteriskmonkey | yeah root just = easy win to the login |
| 17:32.42 | asteriskmonkey | AMI though is meant to be proxied |
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| 17:34.24 | jmetro | im trying to setup a phone printer queue |
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| 17:34.33 | [TK]D-Fender | says who? |
| 17:34.48 | Penguin | Print some phones for me, while you're at it. |
| 17:34.53 | jmetro | users will dial in to my call queue and it will enter them in to a print queue, when my agents scan the page into their PC it picks up the call |
| 17:35.19 | jmetro | my question is: do you prefer overhead or underchin paper delivery to the agents? |
| 17:35.48 | jmetro | ive got a large vacuum tube running both ways so either is easy |
| 17:36.12 | [TK]D-Fender | jmetro: https://www.youtube.com/watch?v=NWTvpPg2Wtc |
| 17:36.38 | jmetro | do those fly well in low atmospheres? [vacuum tube] |
| 17:37.16 | [TK]D-Fender | jmetro: increase velocity and angle of attack to compensate |
| 17:37.38 | jmetro | true, thx. i'll have to pick up 3 or 4 of those machines tomorrow to test |
| 17:38.04 | igcewieling | The only issue you should be worried about is not increasing the speed too much or you'll get decapitations. You don't want that -- WAAAYYYY too much paperwork |
| 17:39.13 | [TK]D-Fender | igcewieling: correct ... it would take a lot of paper to work :) |
| 17:39.15 | jmetro | well if i get any HR complains theyll have to wait until the paper can be properly delivered to my agents to sue me. |
| 17:39.31 | [TK]D-Fender | prepares the next 999 papercuts |
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| 17:55.51 | wasanzy | Thank you guys, you have opened my mind. my problem is solved. I added nagios to sudoers and allow it to run nagios plugins as root. working now |
| 17:56.23 | Penguin | You should have allowed it to run the plugin as the user that runs asterisk. |
| 17:56.30 | igcewieling | See what I mean? |
| 17:59.28 | wasanzy | Penguin: you are right but just quick fix, I will change that later |
| 17:59.38 | Penguin | Typical. |
| 17:59.40 | wasanzy | igcewieling: what do you mean? |
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| 18:54.33 | KNERD | What dependencies provide cdr_adaptive_odbc & cdr_odbc? |
| 19:00.04 | SuperNull | KNERd im playing with stuff like that now... |
| 19:00.22 | SuperNull | you might need to preload => res_config_odbc.so in modules.conf |
| 19:00.40 | SuperNull | im not certain .. im playing with realtime so thats more likely my problem than yours but.. no harm in trying. |
| 19:00.41 | KNERD | I NEED the dependencies which provide those |
| 19:01.06 | KNERD | else res_config_odbc.so wil not exist |
| 19:01.39 | SuperNull | you trying to build from source and need the odbc dev files ? |
| 19:01.57 | KNERD | hm...dev files..let me look |
| 19:02.01 | SuperNull | is this debian ? |
| 19:02.12 | SuperNull | if so it can provide all you need as one apt-get line.. |
| 19:02.14 | KNERD | centos |
| 19:02.22 | SuperNull | im literally 'templating' an install for debian |
| 19:02.25 | SuperNull | hmm. |
| 19:02.46 | SuperNull | does centos use yum ? |
| 19:02.52 | KNERD | yes |
| 19:02.59 | SuperNull | one sec. |
| 19:03.33 | SuperNull | KNERD: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Installing_id291070.html read through that .. and a few pages after. |
| 19:03.50 | SuperNull | will provide all you need to do .. by package nicely. |
| 19:04.16 | SuperNull | i used that as my source for advice since were running ancient 1.4 stuff |
| 19:05.36 | igcewieling | KNERD: your question is better asked on asterisk-dev if you want to know what modules depend on those modules |
| 19:06.02 | igcewieling | KNERD: also menuconfig should tell you why it isn't building something |
| 19:06.27 | KNERD | make menuconfig? |
| 19:06.35 | malcolmd | menuconfig does…it points to res_odbc, which points to odbc and ltdl (which should come along with odbc when you install it)... |
| 19:06.47 | igcewieling | yes |
| 19:06.59 | KNERD | yeah I looked that that and it's not telling me anything |
| 19:07.10 | KNERD | and yum whatprovides is turning up nothing |
| 19:07.59 | igcewieling | KNERD: you may have a corrupted install of the source code. menuconfig will tell you SOMETHING. |
| 19:08.12 | igcewieling | evenif it says N/A under Depends |
| 19:08.22 | fullstop | did you re-run configure? |
| 19:08.43 | KNERD | Depends on: res_odbc(M) Conflicts with: N/A Support Level: extended |
| 19:08.54 | KNERD | of course I did |
| 19:08.57 | igcewieling | now what does res_odbc depend on? |
| 19:09.34 | igcewieling | on MY system: Depends on: generic_odbc(E), ltdl(E) |
| 19:09.35 | KNERD | I have no idea |
| 19:10.04 | fullstop | It should tell you in the interface |
| 19:10.11 | igcewieling | KNERD: stop being an idiot res_odbc is 15 lines below res_config_odbc |
| 19:10.20 | KNERD | okay..le tme look |
| 19:10.55 | KNERD | yes yes..sorry generic_odbc(E), ltdl(E) |
| 19:11.26 | igcewieling | now you know what you must install. make sure you include the -dev libraries. Likely you could also have gotten the information from the Asterisk book. |
| 19:11.45 | igcewieling | You've read the Asterisk Book, the section on ODBC? |
| 19:11.55 | KNERD | going to look now |
| 19:12.26 | fullstop | I did this just the other day and found my way. It worked surprisingly well. In the past, I had always jumped to an AGI rather than use ODBC. |
| 19:13.30 | igcewieling | fullstop: I needed to do it for Realtime, which did not turn out well |
| 19:13.50 | fullstop | I've always avoided realtime |
| 19:14.03 | igcewieling | fullstop: It is one of Asterisk's Blue Bells features |
| 19:14.56 | fullstop | I'm sure. I've just never had a need for it. |
| 19:19.25 | asteriskmonkey | If i dont see mp3 in translations, does that mean musiconhold wont play mp3? |
| 19:19.50 | igcewieling | asteriskmonkey: correct |
| 19:20.06 | pabelanger | you are better off converting your audio to something more native to asterisk and your phones |
| 19:20.17 | asteriskmonkey | im running Freebsd and easy way of adding 1 module, or entire recomplile needed? |
| 19:20.21 | igcewieling | or enabling mp3 support |
| 19:20.54 | pabelanger | asteriskmonkey: easier to convert it with sox |
| 19:21.15 | asteriskmonkey | and it will ignore the mp3's if i do that ? |
| 19:21.31 | fullstop | it will choose the least-cost conversion, I believe. |
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| 19:22.17 | igcewieling | fullstop: not for music on hold |
| 19:22.25 | fullstop | Interesting |
| 19:22.42 | igcewieling | asteriskmonkey: don't put mp3 files in the moh directory if your Asterisk does not support moh |
| 19:22.54 | igcewieling | fullstop: I'd have to check the source code to be sure. |
| 19:23.00 | fullstop | So if I convert to ulaw and g729, which will it choose? |
| 19:23.16 | fullstop | If I'm doing pass-through and don't have the codecs to trans-code, I would hope that it would choose the proper format. |
| 19:23.45 | igcewieling | fullstop: for MoH? In my experience MoH plays all files. You are welcome to test it. |
| 19:23.58 | fullstop | I need to check and see what I have down there. |
| 19:24.25 | igcewieling | fullstop: for normal dialplan playback of audio files ]what you are saying is 100% correct. |
| 19:24.26 | fullstop | I have g729 and ulaw |
| 19:24.48 | igcewieling | fullstop: what happens when you wait for the first sound file to finish? |
| 19:24.56 | igcewieling | does it play the second one or replay the first one? |
| 19:25.12 | fullstop | let me check my config |
| 19:25.22 | fullstop | I don't know if I've listened to one the whole way through |
| 19:25.26 | igcewieling | no, I mean as shown in the CLI. that is the only way to know |
| 19:25.51 | fullstop | It lists them without extension. |
| 19:26.27 | igcewieling | fullstop: ah. I guess the only way to be sure is to have two totally different sound files with the same name in different formats. you should hear it immediatly |
| 19:26.28 | fullstop | and they are not duplicated |
| 19:26.54 | fullstop | this is asterisk 1.6 in this example |
| 19:27.26 | [TK]D-Fender | [15:19]asteriskmonkeyIf i dont see mp3 in translations, does that mean musiconhold wont play mp3? <- no, translation shows CODECS. MP3 playback uses a FORMAT module |
| 19:28.06 | igcewieling | [TK]D-Fender: still has to transcode unless his phone supports Mp3 |
| 19:28.40 | [TK]D-Fender | no, his phjone ISN'T running the call in MP3, and Asterisk ENCODES it in the actual format of the call. |
| 19:28.47 | [TK]D-Fender | but it is not a "codec". |
| 19:29.00 | igcewieling | [TK]D-Fender: without mp3 support it can't decode the mp3 to transcode it. |
| 19:29.01 | [TK]D-Fender | * never ENCODES in MP3 as it is a FORMAT, nt a CODEC |
| 19:29.14 | [TK]D-Fender | but that has nothng to do with the "show translation" command |
| 19:29.20 | [TK]D-Fender | Break that association |
| 19:30.03 | [TK]D-Fender | the point is that list shows nothing of value |
| 19:31.59 | asteriskmonkey | thanks TK |
| 19:32.14 | asteriskmonkey | although i did confirm i must be missing mp3 abilities in musicon on hold |
| 19:32.20 | asteriskmonkey | i sox'd one to ulaw and it worked |
| 19:32.35 | igcewieling | asteriskmonkey: make menuconfig, enable format_mp3 |
| 19:32.56 | asteriskmonkey | yeah this is freebsd though it builds from ports |
| 19:33.01 | asteriskmonkey | so its missing in the base build |
| 19:33.02 | igcewieling | is menuconfig deprecated or something? two questions today answered by going into menuconfig 8-) |
| 19:33.12 | asteriskmonkey | its make menuselect |
| 19:33.14 | igcewieling | asteriskmonkey: your broken package is of no concern to me. |
| 19:33.20 | asteriskmonkey | lol |
| 19:33.24 | asteriskmonkey | meh im good |
| 19:33.38 | asteriskmonkey | since this one handles 1000's of calls ill be better of with ulaw anyhow |
| 19:34.02 | igcewieling | make menuconfig also works |
| 19:34.24 | KNERD | Yeah thanks..I was having a brain aneurysm and did nto click to look the other items |
| 19:40.36 | igcewieling | does anyone happen to know industry standard values for sound/silence for detecting a person rather than an answering machine? I have some telemarketers I need to torture. |
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| 21:05.00 | Mon|A|rch | igcewieling, industry secrets :P |
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| 21:11.25 | Mon|A|rch | how do you turn off asterisk debugging? |
| 21:11.55 | Mon|A|rch | or at least turn off the logging |
| 21:22.04 | [TK]D-Fender | logger.conf <--- |
| 21:23.30 | igcewieling | ~book |
| 21:23.31 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:24.22 | igcewieling | or Section 24 of The Book |
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| 23:11.41 | Mon|A|rch | ~buybook |
| 23:11.41 | infobot | You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/0636920025894 so go buy it SERIOUSLY |
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| 23:42.28 | Mon|A|rch | I've heard that adding a short background() before AMD improves success rates |
| 23:42.32 | Mon|A|rch | why is that? is that still a thing? |
| 23:42.40 | Mon|A|rch | was it ever a thing? |
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