00:02.03 | *** join/#asterisk Pathin (~root@gladsheim.nullbytestudios.net) |
00:08.28 | *** join/#asterisk Pathin (~root@gladsheim.nullbytestudios.net) |
00:10.18 | *** join/#asterisk ExpertOrBust (~ExpertOrB@wsip-24-234-159-70.lv.lv.cox.net) |
00:16.44 | ChannelZ | dang something is missing here... half my 'core' commands are gone |
00:18.25 | WIMPy | thought the idea of the 'core' commands was that they will allways be available. |
00:23.01 | dym | same here. |
00:25.11 | ChannelZ | seems my codec_g729 is hanging the load process which was the problem. hmm. |
00:26.34 | ChannelZ | still can't get gtalk to work |
00:26.34 | dym | Does anyone know of a decent softphone (os x) that can have multiple sip accounts active? |
00:26.44 | dym | or do i actually have to install an asterisk locally? :/ |
00:30.01 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
00:35.16 | p3nguin | dym: Is zoiper or twinkle available for OS X? |
00:35.34 | dym | you tell me, cutiepie |
00:35.36 | dym | :) |
00:35.56 | WIMPy | zoiper is |
00:36.16 | dym | i really like sjphone |
00:36.22 | dym | so slim and decent |
00:37.32 | p3nguin | I'm not sure why it's my responsibility to find software for you. |
00:37.59 | dym | its not :| |
00:38.18 | p3nguin | But the two suggestions I gave you allow multiple active accounts. |
00:38.42 | WIMPy | But zoiper is limited to two in the free version. |
00:38.46 | dym | p3nguin: thanks. im trying one out |
00:46.34 | dym | p3nguin: thanks for the suggestions. clients are too colorful. |
00:47.35 | WIMPy | Set your deskto to greyscale. |
00:47.37 | WIMPy | p |
00:50.42 | dym | :D |
00:50.43 | dym | doubt it |
00:51.29 | p3nguin | Your requirement was to have multiple accounts active. You didn't say anything about the interface must not have colors. |
00:52.26 | dym | p3nguin: why the tenseness? (: |
00:52.32 | dym | all peachy |
01:05.06 | *** join/#asterisk techknowlust (~patrick@li257-211.members.linode.com) |
01:05.36 | techknowlust | I'm having trouble with 486 Busy Here errors when internal SIP calls are answered |
01:05.50 | techknowlust | running asterisk 1.8.5.0 with very basic dialplan |
01:06.07 | *** join/#asterisk Pathin (~root@gladsheim.nullbytestudios.net) |
01:09.41 | *** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45) |
01:30.07 | p3nguin | no supper = room for beer |
01:31.25 | johnnyasterisk | room for beer + lots of beer - supper = hangover |
01:35.07 | p3nguin | I think I saw some pretzels, so I can snack on those if I get hungry later. |
01:49.55 | *** join/#asterisk joker2u (~root@li345-191.members.linode.com) |
01:51.47 | joker2u | what directory are the asterisk modules located? |
01:53.47 | p3nguin | probably /usr/lib/asterisk/modules |
02:02.34 | joker2u | p3nguin thank you |
02:06.03 | techknowlust | I'm getting 486 busy here errors with internal sip calls between softphones. Any ideas as to how I can solve this ? |
02:06.17 | techknowlust | really simple setup. not sure what's causing this |
02:06.32 | p3nguin | You'll need to check the sip debug. |
02:06.44 | techknowlust | ok. |
02:08.10 | techknowlust | http://pastie.org/2224962 that's the sip debug log for the call |
02:11.09 | techknowlust | some 404's and 401's in there which confuse me |
02:13.32 | p3nguin | I don't see any "486 busy here" in there at all. |
02:13.54 | techknowlust | the 486 was shown in the core verbose rather than sip debug |
02:16.28 | techknowlust | basically the call dies as soon as the other end picks up |
02:21.24 | joker2u | which asterisk module effects call forwarding? |
02:22.04 | p3nguin | forwarding of what? |
02:22.10 | p3nguin | a SIP call? |
02:23.50 | joker2u | p3nguin just like an edit in extension.conf. forward to a particular DID. |
02:24.28 | p3nguin | You aren't using the terminology in a manner that makes any sense to me. |
02:24.36 | joker2u | p3nguin app_exec.so??? |
02:24.54 | p3nguin | You don't forward calls to DIDs. Calls are received and extensions run commands. |
02:25.09 | WIMPy | Doesn't make any sense to me, either. What do yu want to do? |
02:25.19 | WIMPy | +o |
02:26.16 | p3nguin | I'm guessing he's just trying to receive a call, but without a sensical expression, it's nothing more than a mere guess. |
02:27.08 | WIMPy | does not feel like being able to make a guess (yet). |
02:27.49 | joker2u | I am working with py-asterisk and the documentation is very limited. Very simple. I have and extension that via the api I want to be able to 'send calls for the extension' to voicemail via agi |
02:28.19 | p3nguin | The extension would execute VoiceMail() to send a call to voice mail. |
02:28.34 | ChannelZ | arf |
02:28.56 | joker2u | I found the voicemail module but forwarding it back to a DiD I can't see what module would be used? |
02:29.17 | dym | 04:24:53 < p3nguin> You don't forward calls to DIDs. Calls are received and extensions run commands. |
02:29.39 | WIMPy | WTF do you mean by "back to a did"??? |
02:30.06 | joker2u | WIMPy because choice is an outside line. not the box. |
02:30.16 | joker2u | choice #2 |
02:30.34 | WIMPy | You want to send it to an external voicemail? |
02:30.42 | p3nguin | It's almost as if he's just regurgitating terms he's heard or read somewhere, without any context for their usages. |
02:31.02 | joker2u | WIMPy voice mail or a pstn number (DiD) those are the options. |
02:31.32 | WIMPy | Where? |
02:31.38 | joker2u | p3nguin DiD = dial in direct. |
02:31.45 | p3nguin | wrong |
02:31.54 | p3nguin | direct inward dialing |
02:31.57 | p3nguin | close, though. |
02:32.18 | p3nguin | I'll give you half a point for a good try. |
02:32.29 | raden | why is it soooo freaking hard to get away from windows |
02:32.30 | joker2u | p3nguin it's a phone number, lordy. |
02:32.36 | raden | I wish this freaking world would embrace linux !!! |
02:32.43 | WIMPy | raden: Is it? |
02:32.45 | p3nguin | raden: I haven't used Windows in 10 years. |
02:32.58 | WIMPy | I have never used Windows. |
02:33.05 | p3nguin | Seems easy enough, then. |
02:33.11 | joker2u | what module would edit the dial plan? |
02:33.15 | dym | p3nguin: lovely :D |
02:33.23 | ChannelZ | Barf. Gtalk just does not want to work. I see the jabber messages on the console with debug on but * just seems to ignore them or something.. no errors, but nothing is happening |
02:33.23 | p3nguin | pbx_config.so |
02:33.29 | WIMPy | But I know people who did, but haven't done so for years. |
02:33.49 | raden | p3nguin, I have a company that insists all cad work be done in alibre cad |
02:33.53 | WIMPy | joker2u: None. Use vi or whatever. |
02:33.58 | raden | p3nguin, try running quickbooks in wine :( |
02:34.06 | raden | p3nguin, I hate windows |
02:34.09 | raden | grrrrr |
02:34.15 | p3nguin | Why would I want to use quickbooks? |
02:34.29 | raden | just the whole business world very hard to get away from windows :( |
02:34.30 | raden | sucks :( |
02:34.31 | joker2u | WIMPy vi? how in the world do you script vi? |
02:34.48 | p3nguin | insert mode |
02:34.51 | ChannelZ | Every time you use vi, another unicorn dies |
02:35.27 | raden | LMA O |
02:35.35 | WIMPy | joker2u: Write a script to generate (part of) your dialplan. |
02:35.51 | WIMPy | I guess most of us do something like that. |
02:35.51 | raden | how stable is 11.4 ? |
02:35.55 | p3nguin | It's not that hard. |
02:35.55 | raden | on 11.1 at moment |
02:35.57 | p3nguin | or just type the entire thing yourself. |
02:36.06 | WIMPy | Or use realtime. |
02:36.07 | p3nguin | Windows 11.4 should be fine. |
02:36.16 | joker2u | I'll stick with python. |
02:37.07 | joker2u | vi isn't 'class' compatible. it's flat, won't work. |
02:37.28 | p3nguin | Asterisk 11.4 should be fine. |
02:37.42 | p3nguin | QuickBooks 11.4 should be fine. |
02:37.55 | p3nguin | VIM 11.4 should be fine. |
02:37.58 | raden | p3nguin, opensuse 11.4 dork |
02:38.10 | dym | (: |
02:38.21 | p3nguin | You expected me to know you were talking about openSUSE, and you call me a dork? |
02:38.21 | dym | raden: why in gods name would you use suse? |
02:38.43 | WIMPy | Luckily I haven't seen suse for many years. But mabe it has become usable in that time. |
02:38.57 | dym | or increasingly homosexual |
02:39.21 | p3nguin | I've installed openSUSE for a few people, and I didn't find much wrong with it. |
02:39.51 | dym | dont like yast |
02:40.09 | p3nguin | use zypper |
02:40.10 | WIMPy | noone like yast |
02:40.14 | dym | aptitude. |
02:40.30 | raden | dym, its alot better than it was in 10.x |
02:40.32 | p3nguin | yast install aptitude? |
02:40.41 | WIMPy | But whe I had contact, they always used very custom kernel, often rather unstable. |
02:40.51 | dym | :D |
02:40.54 | dym | good one p3nguin |
02:41.07 | p3nguin | aptitude install archlinux |
02:41.14 | dym | all within suse, right? |
02:41.15 | dym | :P |
02:41.52 | dym | gawd, just picked up someone from downtime. reminded me how much i hate drunk kids |
02:41.58 | *** join/#asterisk fulcan (~root@li345-191.members.linode.com) |
02:42.02 | dym | downtown* crap - sysadmin kicking in |
02:42.03 | WIMPy | is planning to try lubuntu on his netbook. |
02:42.12 | dym | WIMPy: good choice |
02:42.21 | dym | wait lubuntu? |
02:42.27 | dym | What Window Manager is that? |
02:42.37 | dym | ah, light |
02:42.37 | WIMPy | NFI |
02:42.40 | p3nguin | lxde? |
02:42.44 | WIMPy | I'd expect to get a choice. |
02:42.47 | dym | mhh |
02:42.48 | dym | doubt it |
02:42.56 | raden | I cant believe mac is actually making a comeback in the OS wars |
02:42.57 | p3nguin | You don't get a choice. |
02:42.59 | dym | xfce prolly |
02:43.01 | p3nguin | You get what you get, but then you can change later. |
02:43.03 | WIMPy | But it's likely defaulting to lxde. |
02:43.20 | dym | raden: im a os x user |
02:43.22 | WIMPy | That's choice enough. |
02:43.26 | dym | pretty sarisfies one too |
02:43.32 | dym | looks like gnome |
02:43.41 | dym | (lubuntu) |
02:43.49 | raden | dym, how u like it ? |
02:43.50 | p3nguin | It's still beyond me why anyone would want to use any buntu. |
02:44.00 | dym | raden: i fell in love with it a few years ago |
02:44.10 | dym | i mean come on - it has a native bash |
02:44.14 | dym | whats not to love? |
02:44.16 | raden | p3nguin, why you say that p3nguin |
02:44.23 | raden | dym, that is true |
02:44.33 | dym | p3nguin: why is that? |
02:44.36 | raden | dym, id use it if it wasnt so freaking hardware specific |
02:44.46 | dym | raden: its all about the kernel |
02:44.48 | dym | ;) |
02:45.01 | WIMPy | A standard desktop is absolutely inappropriate for portable devices. |
02:45.07 | p3nguin | Almost everyone I know who has used other distros as well as buntu distros makes fun of buntu distros. |
02:45.20 | dym | With what reason? |
02:45.30 | ChannelZ | Hmm. Jingle is no longer used for gtalk in 1.8 is it? The wiki seems to only talk about the chan_gtalk and jabber. |
02:45.52 | WIMPy | It accesses the harddrive every few seconds, even when doing nothing. |
02:45.59 | raden | p3nguin, there is a lot of stupidity in buntu |
02:46.11 | WIMPy | finds that completely inacceptable. |
02:46.20 | dym | buntu is what it is - linux for monkeys |
02:46.32 | dym | based on a first class distro tho |
02:46.36 | WIMPy | But I don't care to trace the offenders down. |
02:46.58 | dym | WIMPy: rocking the iostat |
02:47.13 | WIMPy | Doesn't show anything. |
02:47.49 | WIMPy | I guess that meas something is contantly forking processes or threads that die immediately again. |
02:48.23 | dym | ps aux / top ? |
02:48.27 | dym | l2debug |
02:48.47 | *** join/#asterisk niekie_ (quasselcor@CAcert/Assurer/niekie) |
02:49.07 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
02:49.08 | WIMPy | top isn't informative, either. |
02:49.16 | p3nguin | How about iotop? |
02:49.18 | dym | WIMPy: sorted right it probably is |
02:49.41 | WIMPy | No, I've been the way up to iotop. |
02:49.58 | dym | Then, dear friend, you shall suffer. |
02:50.32 | WIMPy | That's why I want to try some stripped down distro. |
02:50.34 | p3nguin | iotop -aoP |
02:50.42 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
02:50.47 | dym | Thats a selfdestroy command |
02:50.49 | dym | obv. |
02:52.22 | dym | God im Bored. Time for some prank calls |
02:52.35 | johnnyasterisk | lol |
02:53.03 | johnnyasterisk | sure prank calls.... the ones that cost $2 per min ;-) |
02:53.10 | dym | nah, local |
02:53.15 | dym | <3 call files |
02:53.56 | johnnyasterisk | now thats not really a great use fort asterisk is it |
02:53.58 | johnnyasterisk | lol |
02:54.13 | dym | course it is. |
02:54.42 | johnnyasterisk | we have our own auto dialer which uses asterisk |
02:54.50 | johnnyasterisk | i prefer using originate ;-) |
02:55.07 | dym | thats deprecated since 1.8 |
02:55.14 | dym | isnt it? |
02:55.29 | WIMPy | renamed |
02:55.30 | dym | "core dial" |
02:55.44 | WIMPy | channel originate |
02:55.47 | dym | orly |
02:55.51 | johnnyasterisk | i am still using 1.2.27 |
02:55.54 | johnnyasterisk | ;-) |
02:56.03 | WIMPy | urgs |
02:56.11 | johnnyasterisk | lol |
02:56.13 | dym | oh dear, johnny* |
02:56.47 | johnnyasterisk | have made it to 1.4.40 on another server but took too long to change some of the customizations |
02:56.59 | johnnyasterisk | also found huge problems with reload on 1.4 |
02:57.09 | johnnyasterisk | which prevented moving to it sooner |
02:57.37 | dym | JABBER: Keep alive packet <-- really annoying |
02:57.45 | dym | on what debug level are those msgs supressed? |
03:00.36 | ChannelZ | I think with jabber debug on, you just get everything |
03:00.39 | ChannelZ | that's the point |
03:01.01 | *** part/#asterisk joker2u (~root@li345-191.members.linode.com) |
03:01.16 | ChannelZ | This is driving me crazy. I can call out with gtalk and it works, but if I call in with gtalk nothing happens. |
03:01.33 | ChannelZ | no make-a sense-a |
03:01.54 | dym | OH |
03:02.05 | dym | thanks ChannelZ - forgot about "jabber debug" |
03:02.31 | ChannelZ | Interestingly I don't get these keepalive packets you speak of |
03:04.02 | dym | i get that in debug |
03:04.09 | dym | but im connected to my own jabber server |
03:04.15 | dym | not to gtalk |
03:04.16 | ChannelZ | how often? |
03:04.22 | ChannelZ | oh |
03:04.26 | dym | quite frequently |
03:05.32 | ChannelZ | I must have something set wrong. I get a bunch of jabber messages when I call in but asterisk seems to just do nothing. |
03:05.55 | dym | No idea what you are trying to do. i use jabber to notify on calls. |
03:08.52 | ChannelZ | I'm trying to use Google Talk to call in my Asterisk |
03:09.12 | ChannelZ | not GV via a phone number, but like voice chatting a contact from gmail |
03:10.09 | dym | So Google Voice -> Local Asterisk Number? |
03:12.08 | ChannelZ | mostly |
03:12.17 | ChannelZ | NOT |
03:12.26 | ChannelZ | Google Talk, not Voice |
03:12.36 | ChannelZ | IE not a regular phone number involved anywhere |
03:14.58 | ChannelZ | Maybe it can't work with Talk. It used to. |
03:16.28 | ChannelZ | Wait a minute. It works with the standalone Google Talk client but not the web browser one. |
03:26.31 | ChannelZ | this is annoying |
03:28.36 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
03:42.59 | *** join/#asterisk OldMonk (~raju@59.178.175.121) |
03:43.03 | OldMonk | hi |
03:44.02 | ChannelZ | OH HAI |
03:44.52 | OldMonk | i have some 12 PRIs connected to an asterisk server using redfones. all the channels are in a single group. when dialling, i use something like: Dial(DAHDI/g1/${EXTEN}), which sends the call to the first free channel in the lowest-numbered PRI channel |
03:45.07 | OldMonk | is there any way to load balance the calls across the PRIs? |
03:45.08 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
03:45.53 | OldMonk | so that the first 12 calls, e.g., go to the first channel of each PRI in turn, the next 12 to the second channel of each PRI, etc? |
03:51.34 | ChannelZ | not without manually writing that logic into the dialplan |
03:51.36 | ChannelZ | but who cares? |
03:51.56 | OldMonk | ? |
03:52.19 | ChannelZ | IE what are you really 'balancing'? |
03:52.43 | ChannelZ | It's not like the PRI gets clogged if you use a bunch of channels on one |
03:53.18 | OldMonk | we're facing problems with congestion, and the service provider is asking if we can do this |
03:54.09 | OldMonk | ok, where would you get a situation where the sum total of in-progress + dialling calls exceeds 30 on an E1? |
03:54.14 | OldMonk | we're seeing that regularly |
03:54.33 | ChannelZ | Guess I don't understand the finer points. They're basically guaranteed bandwidth, what good is a PRI where you can only use a few channels |
03:55.16 | OldMonk | to repeat, how the F**K can you have more than 30 calls on an E1? |
03:55.51 | p3nguin | use another E1 |
03:55.52 | *** part/#asterisk fulcan (~root@li345-191.members.linode.com) |
03:56.11 | ChannelZ | I was under the impression you had multiples |
03:56.16 | OldMonk | we do |
03:56.29 | p3nguin | Problem solved. NEXT! |
03:56.38 | OldMonk | but at times we see >30 calls on a single E1 |
03:57.01 | ChannelZ | scratches his head |
03:57.10 | OldMonk | p3nguin: it's asterisk that is assigning those calls to the E1 |
03:57.32 | OldMonk | IOW, problem not solved, next problem may kindly await its turn in the queue |
03:57.55 | ChannelZ | But again, so what? If they're in one group and you're dialing the group, it should go into the next available channel on the next E1... |
03:58.38 | OldMonk | to re-repeat, asterisk is assigning >30 calls to a single E1. under what circumstances would this happen? |
03:59.16 | ChannelZ | it's not possible (sans bug). It sounds like maybe your group is bogus |
03:59.31 | ChannelZ | what do you mean "assigning > 30 calls" |
04:00.02 | OldMonk | sum(in-progress calls + dialling calls) > 30 |
04:00.14 | OldMonk | i don't know how else to state it |
04:01.44 | ChannelZ | The question doesn't make sense. I am getting the feeling this has something to do with these "redfone" devices |
04:02.04 | *** join/#asterisk joker2u (~root@li345-191.members.linode.com) |
04:02.20 | ChannelZ | which I know nothing about what they do or how they work so I'm out |
04:02.32 | p3nguin | I don't know that dahdi can actually put more calls on a "line" than there are channels. |
04:02.42 | OldMonk | p3nguin: precisely |
04:02.56 | p3nguin | So then it's a non-issue and the problem doesn't exist. |
04:03.18 | OldMonk | ...except that it's happening in front of my eyes |
04:03.27 | ChannelZ | It would help to know what your actual problem is. |
04:03.43 | ChannelZ | You get back 'CONGESTION' when trying to place a call? |
04:03.49 | OldMonk | but sure, if you feel that o5strich is a better nick, go right ahead ;) |
04:04.42 | OldMonk | ChannelZ: i don't know what the precise return code is, but here're the symptoms: |
04:04.48 | joker2u | can anyone help me understand astobj2.h? this is the class object for the api for ALL of the asterisk internal objects? this is what py-asterisk connects too? |
04:05.11 | OldMonk | 1. when my callers try to place a call they get a congestion tone (very fast beep for a few seconds, then disconnect) |
04:05.42 | OldMonk | 2. at that time, when i count calls on a per-E1, i see some E1s that have more than 30 calls on them |
04:06.13 | OldMonk | 3. number of in-progress calls is always <30, but sum(in-progress + being dialled) calls >30 |
04:06.16 | OldMonk | . |
04:06.18 | ChannelZ | How do these redfone devices actually connect to your Asterisk such that you're using DAHDI? |
04:06.26 | OldMonk | ChannelZ: TDMoE |
04:06.52 | ChannelZ | it sounds like their driver is broken and is somehow screwing up channel grouping or something |
04:07.59 | OldMonk | i don't think there's a separate driver: it uses whatever passes for the stock DAHDO TDMoE driver |
04:08.03 | ChannelZ | If the entire system is idle, if you dial DAHDI/36 does it go out the second E1? (IE do your DAHDI channels truly correspond 1:1 with your E1 channels) |
04:08.07 | OldMonk | s/DAHDO/DAHDI/ |
04:08.56 | OldMonk | ChannelZ: i didn't know you could specify a channel in a dahdi dial command... i specify channelgroup |
04:09.20 | ChannelZ | Every channel is addressable |
04:09.48 | OldMonk | ok, haven't tried that then |
04:10.09 | ChannelZ | and a channel group is simple a range/collection of channel numbers, which is why we keep saying you can't have "more than 30 calls on an E1"... Asterisk just uses as many channels as you tell it to |
04:10.35 | OldMonk | ChannelZ: then HTF can sum(....) be >30 ? |
04:10.43 | ChannelZ | Which is why I think this virtual DAHDI driver is at fault, not handling all of the actual PRI channels correctly |
04:10.59 | OldMonk | incidentally, the problem is intermittent |
04:11.01 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
04:11.21 | OldMonk | most of the time it just works beautifully |
04:12.29 | *** join/#asterisk Pathin (~root@gladsheim.nullbytestudios.net) |
04:12.35 | ChannelZ | I'd start rattling redfone's cage. Either your DAHDI config is not quite right and your channels aren't grouped properly, or something is going on beyond Asterisk's control |
04:12.53 | OldMonk | holdon, let me paste chan_dahdi.conf |
04:14.07 | ChannelZ | When you dial the group, it should come back and say something like "-- Called DAHDI/g1/5551212" and then " -- DAHDI/X-X making progress" or whatever -- the X-X being the actual channel number it's using |
04:14.20 | OldMonk | http://pastebin.com/uxBzE0Nu |
04:14.24 | ChannelZ | (for me it says 'answered' because I'm using POTS) |
04:15.07 | *** join/#asterisk Mango (~Mango1234@S010620cf30c62cb6.vc.shawcable.net) |
04:16.03 | ChannelZ | Well that looks normal enough.. channels 125 through 619 (mostly) in group 1 |
04:18.48 | ChannelZ | wonders what the maximum number of channels possible even is or where it's defined... |
04:18.55 | OldMonk | 1024 |
04:19.27 | ChannelZ | hmm |
04:19.54 | OldMonk | found out the hard way, by trying to assign more than that to a single dial server |
04:20.10 | ChannelZ | Are you using mixmonitor or chanspy or conferences or anything? |
04:20.21 | OldMonk | chanspy, but on a different server |
04:20.51 | ChannelZ | Doing some random googling I see some references to 512 psuedo-channels |
04:24.41 | *** join/#asterisk nix8n82 (~nate@24.143.28.16) |
04:26.22 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
04:27.01 | OldMonk | guess i have to look into the detailed channel allocation to figure out what's really happening. |
04:27.05 | The_REV | Please Please Please does anyone know a way to export extensions.conf from a box running freepbx so I can rebuild this shitty ox and run is with out freePBX |
04:27.42 | ChannelZ | freePBX uses realtime which means it's in the SQL database |
04:28.04 | The_REV | right... and I want to covert it to a static so I can rebuild... |
04:28.13 | The_REV | stupid SQL fucking things up |
04:28.29 | OldMonk | obviously, if there're >30 calls on an E1 then some of the channels are being reused; i'd presume that asterisk is trying to dial out on a channel that already has a call in progress. the question is, why? |
04:28.56 | ChannelZ | just note that even if you get it exported you're going to have tons of crazy FreePBXisms (macros and stuff) |
04:29.45 | ChannelZ | that said I'm not sure if there's any tools floating around that'll do it |
04:30.06 | The_REV | i can clean up crazies... |
04:30.08 | The_REV | damn... |
04:30.16 | ChannelZ | OldMonk: incorrect status by the TDMoE driver? |
04:30.37 | OldMonk | ChannelZ: c'est possible |
04:30.48 | OldMonk | or maybe incorrect status from the provider |
04:31.11 | ChannelZ | All asterisk knows about is a big pool of channels. |
04:31.46 | OldMonk | ChannelZ: it seems to be either getting the wrong channel status, or misinterpreting |
04:32.07 | ChannelZ | It tries the next available one when you use a group. Maybe one of the channels somewhere in the middle is dead, or returning an error for some reason. You'll really need to look at PRI debug to figure out what precisely is going on |
04:32.09 | OldMonk | if the first, then i need to figure out where that incorrect status is originating |
04:32.34 | OldMonk | if the latter then it's probably a bug |
04:32.58 | OldMonk | and fscked if i can understand pri debug |
04:33.29 | OldMonk | anyhow, the people over at redfone can handle that part |
04:34.39 | OldMonk | tomorrow's a calling day, will investigate |
04:34.41 | OldMonk | thanks folks |
04:35.27 | ChannelZ | That was interesting |
04:37.33 | Mango | I have a phone behind NAT connecting to an Asterisk server outside NAT. I want direct media between the phone and the carrier, so I've set up Handle VIA Received on the phone. This works, but it breaks local calling between phones behind the same NAT. |
04:37.45 | Mango | Is there any way to fix that without port forwarding? |
04:38.34 | ChannelZ | Do you have control over the Asterisk server? |
04:38.40 | Mango | Yes. |
04:39.21 | ChannelZ | Oh.. wait nevermind |
04:39.45 | Mango | I could pretend it's someone else's Asterisk server if that would help :P |
04:40.47 | ChannelZ | no I was just thinking wrong outloud |
04:41.07 | Mango | hehe |
04:42.25 | ChannelZ | I guess if the device doesn't have the concept of an internal and external network in order control when it lies about its IP and when it shouldn't, I'm not sure. |
04:43.58 | Mango | The weird thing is the phones only seem to respect the received= header when it comes from the Asterisk server, not when it comes from each other. |
04:45.07 | Mango | SPA921 |
04:46.29 | ChannelZ | hmm |
04:51.28 | *** join/#asterisk godmachine-x6 (~godmachin@h214.179.90.75.dynamic.ip.windstream.net) |
04:59.36 | ChannelZ | hmmmm. |
05:00.20 | ChannelZ | Soooo the Google Talk in-the-browser is still referencing jingle |
05:01.53 | DrDigital | i need to find these riser cards for this server http://www.weirdstuff.com/cgi-bin/item/62043 mines just a tad slower |
05:02.20 | DrDigital | ahh and i dont have an intel motherboard either |
05:02.30 | DrDigital | but the case is almost identical i think |
05:06.37 | ChannelZ | hmm this is confusing |
05:34.30 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
05:56.30 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
05:57.53 | sawgood | Is Skype for Asterisk worth the $66 dollar per channel license? I was thinking of buying 5 channels before the experiation date ... |
05:58.24 | p3nguin | If you're doing it for the novelty value, you should probably save your money. |
05:58.52 | sawgood | yeah ... that was the case ... just to have it working before they cut it off |
05:59.24 | sawgood | Then after the cut-off allow other Asterisk boxes to 'proxy' through the SFA box with the licenses ... |
05:59.27 | sawgood | If that was possible |
06:04.58 | *** join/#asterisk deltaflyer4747 (5ef24b32@gateway/web/freenode/ip.94.242.75.50) |
06:05.03 | deltaflyer4747 | morning |
06:23.44 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.109) |
06:23.45 | *** join/#asterisk boazb (~b@bzq-82-80-219-90.red.bezeqint.net) |
06:30.38 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
07:00.52 | deltaflyer4747 | now i am screwed, i just upgraded to Asterisk 1.8.5.0-1digium1~squeeze and it crashes on startup :( |
07:02.51 | deltaflyer4747 | lets try fresh install :) |
07:03.38 | deltaflyer4747 | and doesn't work :) |
07:11.23 | deltaflyer4747 | http://pastebin.com/AvFhmrwd - if anyone knows whats wrong, i'd be glad. |
07:18.40 | ChannelZ | well it seems like your database is horked, unavailable, or * is misconfigured to talk to it |
07:19.25 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
07:20.19 | deltaflyer4747 | ChannelZ: this is default installation - just apted the * and there isn't a word about any database prerequisites |
07:20.45 | deltaflyer4747 | i followed this time step by step the https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
07:21.35 | ChannelZ | well it seems to be configured for realtime |
07:22.04 | deltaflyer4747 | meaning? |
07:22.24 | ChannelZ | it's trying to use a MySQL database for some of its configuration |
07:22.38 | ChannelZ | That may or may not be related to your problems, or only a small part of them |
07:22.46 | deltaflyer4747 | okay, i will give it one. |
07:22.52 | deltaflyer4747 | no problem so far |
07:23.03 | ChannelZ | was this a previously working installation? |
07:25.44 | deltaflyer4747 | ChannelZ: i used 1.6, today i upgraded to 1.8, it's not working. I purged old installation, installed new from scratch, same story. |
07:26.20 | ChannelZ | so you're just setting this up for the first time to play? |
07:27.40 | deltaflyer4747 | you could say that. |
07:27.54 | deltaflyer4747 | althrough i ofcourse saved my previous config files :) |
07:28.09 | ChannelZ | well screw these packages, just build the source |
07:28.24 | ChannelZ | it takes 10 minutes and you'll know how it's configured |
07:29.08 | deltaflyer4747 | i thought that digium knows how to compile packages of their own product :) |
07:30.11 | ChannelZ | I'm sure they do but you can too |
07:30.45 | deltaflyer4747 | :) okay, downloading etc :) |
07:30.47 | ChannelZ | If it's just bailing because whatever configs you have are bogus, that's one thing.. but if it's coreing, that's another. |
07:31.51 | deltaflyer4747 | yea, i kept default configs and its the same story :-/ |
07:32.02 | deltaflyer4747 | now... reading through ./configure options... |
07:32.20 | *** join/#asterisk wonderworld (~ww@port-92-201-164-64.dynamic.qsc.de) |
07:32.54 | ChannelZ | just do it |
07:32.56 | ChannelZ | ./configure |
07:32.57 | ChannelZ | make |
07:33.00 | ChannelZ | be happy |
07:33.10 | deltaflyer4747 | straight like that? okay. |
07:33.34 | ChannelZ | you can 'make menuconfig' in between configure and make and tweak some things if you really feel you must |
07:33.49 | ChannelZ | Actually I do just to turn on/off the sound packages as desired |
07:34.06 | deltaflyer4747 | okay |
07:34.09 | deltaflyer4747 | will check |
07:34.41 | ChannelZ | just purge your old package off completely before you install, so you're not tripping over leftover bits possibly in other directory trees getting even more confused |
07:34.44 | deltaflyer4747 | just fixing some prerequsities... |
07:34.51 | deltaflyer4747 | yep, of course :) |
07:35.49 | deltaflyer4747 | well, i hope this will be fix for the chanspy problem i'm facing :) |
07:36.02 | ChannelZ | which was what |
07:36.53 | deltaflyer4747 | yesterday evening we (me + p3nguin )found that chanspy either won't produce neccessary sound route through W option OR kills actual call between spied parties |
07:38.18 | deltaflyer4747 | okay, i have to install dahdi first |
07:38.35 | ChannelZ | hmm. I've never actually used whisper |
07:38.48 | deltaflyer4747 | ChannelZ: http://deltaflyer.cz/ast.txt |
07:38.53 | deltaflyer4747 | thats what i need to do |
07:39.54 | ChannelZ | so you're trying to listen to a security camera |
07:40.04 | deltaflyer4747 | yes |
07:40.10 | deltaflyer4747 | while calling the security camera |
07:40.20 | deltaflyer4747 | that way the security camera will not hear itself |
07:40.50 | deltaflyer4747 | so ... calling 11- 81 and making chanspy for 1001 on 11 with W - putting audio from 1001 to 11 only :) |
07:42.51 | deltaflyer4747 | + i think that WIMPy suggested that long time before we found for ourselves but then i haven't understood him |
07:43.54 | ChannelZ | I'm not sure I understand what extension 81 has to do with it |
07:43.59 | deltaflyer4747 | see |
07:44.04 | deltaflyer4747 | that IPcam has 2way audio |
07:44.14 | deltaflyer4747 | 81 is SIP exten on that camera |
07:44.27 | ChannelZ | it speaks sip? |
07:44.31 | deltaflyer4747 | yep |
07:44.41 | deltaflyer4747 | but only as for incoming audio - ie sound FROM the camera. |
07:44.50 | ChannelZ | So if you dial it, it answers, and sends you audio from it's mic/sends your audio to a speaker? |
07:45.15 | deltaflyer4747 | to get audio out of the camera (ie its MIC) i had to use MOH because its not transferred to SIP channel but only to its video stream |
07:45.32 | deltaflyer4747 | and there goes the 1002 extension :) |
07:45.41 | deltaflyer4747 | 1002 = MOH with audio stream from camera |
07:45.55 | deltaflyer4747 | 81 = sip extension for audio TO camera |
07:46.57 | deltaflyer4747 | its the only way how i can get 2way audio with that camera (btw manufacturer said its impossible) |
07:48.18 | deltaflyer4747 | so i spent whole yesterday trying to get that chanspy working :) |
07:50.36 | deltaflyer4747 | with great help of guys here |
07:54.52 | deltaflyer4747 | ok, compiling. |
07:55.51 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
08:02.06 | deltaflyer4747 | needs faster cpu |
08:02.24 | ChannelZ | don't we all |
08:03.24 | deltaflyer4747 | well, that old singlecore P4 is really slow :) |
08:03.36 | deltaflyer4747 | hooray, compiled |
08:03.45 | deltaflyer4747 | downloading sounds etc :) |
08:04.11 | deltaflyer4747 | luckily (atleast that) i have good network connection :) |
08:08.17 | deltaflyer4747 | okay, lets see. |
08:08.30 | deltaflyer4747 | darn, initscript :) |
08:10.19 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
08:12.57 | deltaflyer4747 | ChannelZ: and guess what :)) Same thing :D |
08:13.27 | *** join/#asterisk Syrex (~syrex@dsl-146-17-03.telkomadsl.co.za) |
08:13.36 | ChannelZ | what exactly is happening? |
08:13.39 | deltaflyer4747 | http://pastebin.com/9zfRm5pE |
08:13.45 | deltaflyer4747 | exactly the same thing :D |
08:13.50 | deltaflyer4747 | asterisk won't start |
08:14.08 | deltaflyer4747 | things it has something to do with druides... |
08:14.14 | deltaflyer4747 | *thinks |
08:14.47 | deltaflyer4747 | so the problem lies elsewhere |
08:15.01 | ChannelZ | yes but is it crashing afterwards or what is going on? |
08:15.02 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
08:15.12 | deltaflyer4747 | it immediatelly crashes |
08:15.23 | ChannelZ | And what configs are you using? it seems to be loading a lot of crap you probably aren't even using |
08:15.24 | deltaflyer4747 | thats everything whats in the logfile |
08:15.31 | deltaflyer4747 | default configs |
08:15.44 | deltaflyer4747 | ie from sources |
08:15.57 | ChannelZ | you have every single one of them installed? |
08:16.13 | ChannelZ | I'm not sure why that would make it crash but it's probably not a good idea regardless |
08:16.19 | deltaflyer4747 | i just made "make samples" |
08:16.29 | deltaflyer4747 | and tried to run the * |
08:16.48 | deltaflyer4747 | ie configure &&make&&make install&&make samples |
08:17.30 | deltaflyer4747 | i can delete all files and put there only asterisk, extensions, sip and musiconhold conf files |
08:20.13 | *** join/#asterisk esperanto (~rusty@pool-71-114-139-216.hrbgpa.dsl-w.verizon.net) |
08:22.50 | ChannelZ | yes |
08:23.25 | ChannelZ | the samples are meant as guides, not a completely functional configuration. There is a ton of crap in there that I'm sure will blow up |
08:25.12 | deltaflyer4747 | ofc |
08:30.24 | ChannelZ | I gotta go to bed |
08:35.37 | deltaflyer4747 | i just copied all neccessary files and ... no go. |
08:35.41 | deltaflyer4747 | still the same story. |
08:36.21 | deltaflyer4747 | 1.8.5.0 won't start for me |
08:39.29 | *** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45) |
08:39.45 | deltaflyer4747 | so i hope someone will be able to help me. . . |
08:45.26 | deltaflyer4747 | hempf, it does work now :) |
08:45.32 | deltaflyer4747 | from packages |
08:46.02 | deltaflyer4747 | that means that the aptitude purge haven't cleaned all previous bits and pieces |
08:46.28 | esperanto | hey fellas, I am trying to generate a sip packet that would take 1.6.2.16-1 down |
08:46.29 | esperanto | vulnerability http://web.nvd.nist.gov/view/vuln/detail?vulnId=CVE-2011-2529 |
08:46.29 | esperanto | does anybody know how exactly sip packet should look like? |
08:46.49 | deltaflyer4747 | lol :D he's back :D |
08:47.02 | deltaflyer4747 | esperanto: okay, i will explain that for loop from yesterday for you. |
08:47.05 | deltaflyer4747 | It was countdown. |
08:47.11 | deltaflyer4747 | meaning - you can guess. |
08:48.13 | deltaflyer4747 | this is not an hacking chan |
08:48.28 | deltaflyer4747 | this is asterisk SUPPORT not DESTRUCTION |
08:48.56 | esperanto | well, I didn't get it, just need to elaborate on this a little bit |
08:49.18 | esperanto | yeah, I understand that, but I need to do poc test, otherwise admins won't change anything |
08:49.28 | *** join/#asterisk batfastad (~benbradle@87-194-37-131.bethere.co.uk) |
08:50.19 | esperanto | I use sipp to generate packets, tried pasting /0 almost everywhere and no luck |
08:52.55 | Bipul | well i have gone through Asterisk architecture i Found IN COMMUNICATING with core PBX we need some protocol like DAHDI Which is DIgum asterisk hardware/ second is SIP which is protocol in communicating peer to peer connection only used for transporting voice data 3 one is IAX2 protocol Inter-Asterisk Exchange Protocol, IAX h.323 old protocol used for gnome and windows one = so my question is can we use RTP protocol ? |
08:53.45 | Bipul | in communicating with PBX core? |
08:55.21 | tzafrir | Bipul, RTP for what, exactly? |
08:56.27 | deltaflyer4747 | bipul: you want to listen to RTP stream, right? :) |
08:56.30 | Bipul | like RTP protocol is used for transporting videos and real time measurement data |
08:56.38 | deltaflyer4747 | oh, ok |
09:02.38 | tzanger | hm, vodafone's got a decent deal |
09:02.48 | tzanger | 4 euro for a microsim with 250MB of data on it |
09:03.45 | tzanger | if I'm not mistaken the recharge is more expensive than just spending another 4 euro for another 250MB |
09:04.29 | batfastad | Hi everyone. I've just checked on our asterisk/freepbx VPS and noticed that /var/log/asterisk/full is getting pretty large... 20MB. Just read this info on how to rotate the Asterisk logs... http://www.voip-info.org/wiki/view/logrotate and just wondering if that info is still current? |
09:10.20 | *** join/#asterisk old_monk (~nightrid3@91.176.160.33) |
09:13.30 | *** join/#asterisk Micc_ (~Micc@c-98-232-41-66.hsd1.wa.comcast.net) |
09:13.47 | Micc_ | why does my 1.8.5.0 keep reloading the dialplan every minute or two? |
09:17.37 | Micc_ | seems like every 60 seconds its reloading the whole dialplan. |
09:24.43 | deltaflyer4747 | check the config |
09:24.48 | deltaflyer4747 | asterisk.conf |
09:26.32 | Micc_ | is there a new setting in there that reloads the dialplan by default? |
09:27.26 | Micc_ | I'm using my 1.6 configs, just changing things that need to be changed along the way. |
09:27.49 | deltaflyer4747 | what does the logs say? |
09:28.00 | Micc_ | nevermind. I think I figured it out. |
09:28.03 | Micc_ | runaway cron job |
09:28.06 | deltaflyer4747 | :) |
09:28.08 | deltaflyer4747 | nice |
09:28.25 | deltaflyer4747 | anyone have experience with chanspy / extenspy ? |
09:29.28 | Micc_ | I've used chansply on 1.6, haven't tested it on 1.8 yet |
09:29.59 | deltaflyer4747 | have you tried "whisper" ? |
09:30.09 | Micc_ | yup, thats what we use it for |
09:30.49 | deltaflyer4747 | so ... the spy can talk to spied person ? |
09:32.35 | deltaflyer4747 | because i can hear the spied extension, but it cannot hear me (spy) |
09:33.44 | deltaflyer4747 | do you have some example i could test please? |
09:34.12 | Micc_ | yeah, the spy should be able to hear the device your spying on plus the call bridged to that device, plus in whisper mode be able to talk to the device without the person bridged to that device hearing anything you say. |
09:34.33 | Micc_ | what version of asterisk are you using? |
09:34.51 | deltaflyer4747 | 1850 |
09:34.54 | deltaflyer4747 | latest |
09:35.02 | deltaflyer4747 | wasn't working in 1.6 either |
09:35.03 | deltaflyer4747 | (for me) |
09:35.16 | Micc_ | have you found 1.8.5.0 to be stable so far? |
09:35.25 | deltaflyer4747 | i have it for ... 1 hour :) |
09:35.34 | deltaflyer4747 | so... so far so good :D |
09:35.49 | Micc_ | ok, I've been testing all day. so far so good. A few minore tweaks and had to fix parking. |
09:36.11 | deltaflyer4747 | well, i need that chanspy and its not working for me :-( |
09:36.16 | Micc_ | I'm about to put it into production I think. I was getting dead locks on 1.6.2.19 |
09:36.25 | Micc_ | let me look up how I'm doing that. |
09:37.12 | deltaflyer4747 | thanks |
09:37.27 | Micc_ | exten => *7701,1,ChanSpy(SIP/sandler1,wq) |
09:37.32 | deltaflyer4747 | as i said - i can hear the spied extension, but that extension cannot hear me. |
09:38.17 | Micc_ | what kind of device are you spying on? |
09:38.42 | deltaflyer4747 | deskphone |
09:38.43 | deltaflyer4747 | sip |
09:39.17 | Micc_ | so you're talking to someone else with the deskphone first, then spying in from another sip device? |
09:39.35 | deltaflyer4747 | yes. |
09:39.38 | Micc_ | what parameters are you calling ChanSpy with? |
09:39.57 | deltaflyer4747 | exten => 3131,n,Chanspy(SIP/11,w) |
09:41.02 | Micc_ | that should work fine. |
09:42.02 | Micc_ | it says don't use it with monitor/mixmonitor/record |
09:42.08 | deltaflyer4747 | doesn't |
09:42.17 | deltaflyer4747 | i'm not using any of those |
09:42.37 | Micc_ | not sure what to tell you. It works for me. |
09:42.49 | Micc_ | although I should probably test it in 1.8.5.0 |
09:42.53 | deltaflyer4747 | i know it SHOULD work. |
09:43.06 | deltaflyer4747 | but it doesn't |
09:43.23 | Micc_ | we do it with aastra phones. |
09:44.23 | deltaflyer4747 | i have linksys deskphone (11), ipcam (81) and siemens wifi sip phone (211). I call 11 -> 81 and then dial from 211 to 3131 |
09:45.07 | Micc_ | try it in a different combination |
09:45.30 | deltaflyer4747 | ok, i can dial 211->81 and 11->3131 :) |
09:45.31 | deltaflyer4747 | sec |
09:45.31 | Micc_ | maybe 81-211 then 11 spy |
09:45.40 | Micc_ | yeah, try that. |
09:47.01 | deltaflyer4747 | same story. |
09:48.56 | deltaflyer4747 | i can hear 211 but 211 cannot hear me |
09:49.34 | *** join/#asterisk bmg505 (~leon@196-209-7-14.dynamic.isadsl.co.za) |
09:49.59 | deltaflyer4747 | like no whisper option was in place at the first time |
09:50.54 | deltaflyer4747 | wouldn't some SIP.conf parameter break that? |
09:51.15 | Micc_ | I don't know. |
09:51.26 | deltaflyer4747 | yea, me neither |
09:51.30 | Micc_ | maybe like reinvite or something. |
09:51.43 | deltaflyer4747 | my thoughts exactly |
09:51.47 | deltaflyer4747 | will enable it |
09:51.51 | deltaflyer4747 | in a few, lunch time :) |
09:51.58 | Micc_ | seems like maybe something could have an affect on it, but I can't think of anything off the top of my head. |
09:52.28 | Micc_ | I don't think that will matter, but you can try it. |
09:52.35 | Micc_ | we have canreinvite=no |
09:53.28 | deltaflyer4747 | okay, that wasn't it. |
09:53.36 | deltaflyer4747 | i had it too... :( |
09:54.30 | deltaflyer4747 | darn |
09:54.50 | deltaflyer4747 | maybe... do you have any zaptel devices? |
09:55.06 | deltaflyer4747 | like isdn cards etc |
09:57.13 | Micc_ | no, I haven't used one in a couple years. |
09:57.17 | deltaflyer4747 | okay |
09:57.21 | Micc_ | used to do a bit with some PRI cards. |
09:58.35 | deltaflyer4747 | okay, but now you don't have any |
09:59.57 | Micc_ | they fixed that cisco phone registration problem in 1.8.5.0 didn't they? |
10:00.57 | deltaflyer4747 | i don't know of any |
10:01.07 | deltaflyer4747 | never had cisco phone |
10:01.30 | Bipul | dude whear is sip.conf in /etc/asterisk ? |
10:01.51 | deltaflyer4747 | exactly there |
10:02.01 | Bipul | but there is nuthing |
10:02.10 | deltaflyer4747 | really? |
10:02.21 | Bipul | bipul@bipul-desktop:/etc/asterisk$ ls |
10:02.21 | Bipul | asterisk.conf manager.d |
10:02.21 | Bipul | bipul@bipul-desktop:/etc/asterisk$ |
10:02.22 | deltaflyer4747 | ls -al /etc/asterisk|grep sip |
10:02.31 | deltaflyer4747 | lol :) |
10:02.39 | deltaflyer4747 | then you are missing a ton of config files :) |
10:02.42 | deltaflyer4747 | what did you do? |
10:03.05 | Bipul | nuthing |
10:03.16 | Bipul | can i reinstall it |
10:03.29 | deltaflyer4747 | then thats exactly what you get : nothing :) |
10:03.42 | deltaflyer4747 | you had that sip.conf yesterday, right? |
10:03.48 | Bipul | yes |
10:03.56 | Bipul | but now there is nuthing |
10:04.36 | Micc_ | *cross fingers* 1.8.5.0 going into production |
10:04.51 | deltaflyer4747 | so you had to delete them. |
10:04.57 | Bipul | deltaflyer4747, now what shud i do now ? |
10:05.00 | deltaflyer4747 | Micc_: GL & hf |
10:05.20 | deltaflyer4747 | Bipul: try to remember what happened. |
10:05.45 | Bipul | <PROTECTED> |
10:05.56 | deltaflyer4747 | those files |
10:06.18 | deltaflyer4747 | there are like 100 files in that directory |
10:06.34 | Bipul | i have reinstalled the asterisk that day |
10:06.45 | deltaflyer4747 | what distro do you use |
10:06.52 | Bipul | ubuntu |
10:06.58 | deltaflyer4747 | great! |
10:07.15 | deltaflyer4747 | then follow this https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages guide |
10:07.45 | deltaflyer4747 | but ubuntu for server... tsk tsk tsk... |
10:08.01 | Bipul | no it's just a desktop |
10:08.13 | deltaflyer4747 | okay then |
10:08.14 | Bipul | will it work on desktop 10.10 ubuntu |
10:08.29 | deltaflyer4747 | look on the webpage i posted |
10:08.50 | deltaflyer4747 | its written there ;) |
10:09.30 | deltaflyer4747 | how did you get previous asterisk? |
10:09.45 | Bipul | well sudo apt-get install asterisk |
10:09.52 | deltaflyer4747 | okay |
10:10.11 | Bipul | i know this link i have follow all the instruction of this page in past |
10:10.19 | deltaflyer4747 | are you 100% sure you want to fully reinstall it and begin from scratch? [y/N] |
10:10.37 | Bipul | but yah |
10:11.04 | deltaflyer4747 | sudo aptitude purge asterisk asterisk-dev asterisk-config |
10:13.10 | deltaflyer4747 | Micc_: will continue in a few minutes |
10:14.03 | Bipul | deltaflyer4747, then after apt-get clean ? |
10:14.08 | deltaflyer4747 | no |
10:14.27 | deltaflyer4747 | aptitude purge erases everything those packages are creating... or SHOULD erase :)) |
10:14.29 | Bipul | sudo aptitude purge asterisk asterisk-dev asterisk-config i have used this command. |
10:14.45 | deltaflyer4747 | did it gave you any errror? |
10:14.57 | Bipul | nops |
10:15.04 | deltaflyer4747 | okay |
10:15.08 | Bipul | let me pastbin you the details |
10:15.13 | deltaflyer4747 | ok. |
10:16.05 | Bipul | http://pastebin.com/CudqAiLz |
10:16.17 | *** join/#asterisk nicola_pav (~chatzilla@mail2.tikalnetworks.com) |
10:16.32 | deltaflyer4747 | good. |
10:16.42 | deltaflyer4747 | now check that you have digium repo in place |
10:16.47 | deltaflyer4747 | cat /etc/apt/sources.list |
10:16.51 | Bipul | so now shud i follow again those instruction |
10:17.09 | deltaflyer4747 | should contain "deb http://packages.asterisk.org/deb " |
10:17.13 | deltaflyer4747 | and deb-src |
10:17.46 | Bipul | yes >>>deb http://packages.asterisk.org/deb maverick-proposed main deb-src http://packages.asterisk.org/deb maverick-proposed main |
10:17.55 | deltaflyer4747 | good. |
10:18.27 | Bipul | now shud i installe it. |
10:18.59 | deltaflyer4747 | aptitude install asterisk-1.8 asterisk-dahdi |
10:19.08 | deltaflyer4747 | sorry, sudo before that |
10:19.28 | Bipul | yes but dahdi is for hardware perpose right ? |
10:19.38 | deltaflyer4747 | then ommit it :) |
10:19.45 | Bipul | but i dont have digium card so it's necessary? to install |
10:20.05 | Bipul | alright ! |
10:20.07 | deltaflyer4747 | if you want to use meetme, then yes |
10:20.26 | deltaflyer4747 | meetme needs zaptel-dummy for timing |
10:20.45 | Bipul | sure why not one day i will come to meet you :p |
10:20.50 | deltaflyer4747 | :D |
10:21.03 | deltaflyer4747 | meetme = conference rooms :) |
10:21.25 | Bipul | oh ic :p |
10:21.34 | deltaflyer4747 | yep |
10:21.59 | Bipul | How much it fetch to perchase a digium card? |
10:22.09 | deltaflyer4747 | what for? |
10:22.13 | deltaflyer4747 | btw... |
10:22.22 | deltaflyer4747 | did you installed those python packages, right? |
10:22.27 | Bipul | connecting betwen PSTN |
10:22.41 | deltaflyer4747 | sudo apt-get install python-software-properties |
10:22.53 | deltaflyer4747 | what pstn |
10:23.38 | Bipul | public switched telephone network |
10:23.39 | deltaflyer4747 | get some SIP provider |
10:23.50 | deltaflyer4747 | thats way cheaper and easier |
10:23.53 | deltaflyer4747 | and more reliable |
10:23.57 | deltaflyer4747 | etc |
10:24.09 | Bipul | yes i am reading the books and i have decided to work on sip |
10:24.33 | deltaflyer4747 | analog nor isdn have no future |
10:24.41 | Bipul | ok |
10:25.40 | Bipul | python is already installed |
10:25.47 | deltaflyer4747 | okay. |
10:25.52 | deltaflyer4747 | just recheck - to be sure |
10:25.59 | nicola_pav | hello. anyone familiar with this kind of error: Channel 0/13, span x got hangup, cause 81? |
10:27.06 | *** join/#asterisk radic (~radic@tmo-096-134.customers.d1-online.com) |
10:30.22 | Micc_ | oh crap, got a major problem. its not parsing sip correctly. |
10:30.33 | Micc_ | my extensions are coming in wrong like this |
10:31.09 | Micc_ | Executing [4252505555;phone-context=+1;npdi=yes@inbound-userfield:1] |
10:32.12 | Micc_ | why would it be doing that? |
10:32.54 | deltaflyer4747 | err... don't get the full story |
10:33.32 | nicola_pav | hello. anyone familiar with this error: !! pri_hangup() line:1431 Called with invalid call ptr (0x9c26158) |
10:36.49 | Micc_ | wtf, its rejecting almost all my incoming calls saying the extension isn't found in the context. but it is there. |
10:37.25 | deltaflyer4747 | include? |
10:38.31 | Micc_ | I have a shitload of includes. |
10:38.41 | Micc_ | but this works fine on 1.6.2.19 |
10:38.56 | Micc_ | I'm gonna have to roll back if I can't figure this out pretty quick. |
10:39.21 | Micc_ | it seems asterisk isn't parsing sip headers the same. |
10:39.58 | Micc_ | is there a way I can remove everything after the ; with some dialplan commands? |
10:43.38 | Micc_ | ok, that works for now, a bit of a hack though. |
10:44.46 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
10:45.31 | Bipul | deltaflyer4747, there is problem in configuring sip.conf for creating SIP account |
10:45.46 | deltaflyer4747 | Micc_: what what ? |
10:45.49 | deltaflyer4747 | Bipul: ? |
10:45.57 | Bipul | https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts << i cant find such matches in my sip.conf |
10:46.42 | deltaflyer4747 | matches to what? those are pure example |
10:47.06 | deltaflyer4747 | http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf |
10:47.12 | deltaflyer4747 | read this |
10:47.20 | Bipul | i mean i want to fix my own password and sip and ip |
10:48.47 | Micc_ | deltaflyer4747, for some reason with 1.8.5.0 my dialplan was matching extra crap in the sip header with the number make it so I could not receive any calls from 360 |
10:49.02 | deltaflyer4747 | weird |
10:49.10 | Micc_ | I almost had a heart attack until I fixed it with a couple lines of dialplan code. |
10:49.21 | deltaflyer4747 | hehe |
10:49.45 | Micc_ | But why its doing that is a big mystery. Have you tested receiving calls from your provider on 1.8.5.0 yet? |
10:49.54 | dym | Hey all. Im having trouble registering my softphone, which is located behind a nat. Here is the logs: http://paste.debian.net/123140/ - nat=yes is enabled, but still things seem to be going wrong. any idea? |
10:49.59 | deltaflyer4747 | will try right away |
10:50.09 | deltaflyer4747 | dym: host=dynamic |
10:50.32 | dym | is |
10:51.00 | deltaflyer4747 | qualify |
10:51.10 | esperanto | enable sip debug and toubleshoot |
10:51.21 | deltaflyer4747 | Micc_: works |
10:51.39 | Micc_ | who is your provider? |
10:51.41 | *** join/#asterisk garymc (~chatzilla@host109-145-110-205.range109-145.btcentralplus.com) |
10:51.45 | Micc_ | I have a couple others I can try. |
10:51.48 | Micc_ | maybe its just 360 |
10:51.53 | deltaflyer4747 | Micc_: some local :) |
10:52.07 | deltaflyer4747 | from my country |
10:53.31 | Micc_ | deltaflyer4747, so you don't have a sip provider? You're using a land line? |
10:53.41 | deltaflyer4747 | its sip |
10:54.34 | Micc_ | vitelity seems to be fine, so maybe its just 360, but it was fine in 1.6.2.19. I guess asterisk is probably trying to be more copmliant with the sip rfc. |
10:55.18 | *** join/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
10:56.17 | deltaflyer4747 | DAMN it |
10:56.28 | deltaflyer4747 | why ccannot i get that whisper working |
10:57.33 | *** join/#asterisk _zoom_ (~Kurmot@196.1.219.122) |
10:57.38 | Micc_ | This doesn't look like proper sip header. |
10:57.50 | deltaflyer4747 | :)) |
10:57.55 | deltaflyer4747 | theres your problem :) |
10:58.12 | Micc_ | I gotta pastebin this so you can see how screwed up this is. 360 is on crack. |
10:59.02 | Micc_ | http://pastebin.com/MJTADYHa |
10:59.32 | _zoom_ | fellas do you have any ideas what is really going on why microsoft killed skype-asterisk license? |
10:59.39 | deltaflyer4747 | nice |
10:59.56 | deltaflyer4747 | zoom: asterisk is free, right? |
11:00.09 | _zoom_ | but skype is not |
11:00.23 | deltaflyer4747 | and did you get ANYTHING for free from m$ ? |
11:00.39 | deltaflyer4747 | except tons of troubles |
11:00.49 | _zoom_ | :) |
11:00.52 | Micc_ | deltaflyer4747, am I right? That isn't correct sip format is it? |
11:01.25 | deltaflyer4747 | Micc_: i'd have to study correcct sip first |
11:01.43 | deltaflyer4747 | never needed to know that |
11:01.49 | _zoom_ | so they are going to bury skype a live |
11:01.51 | _zoom_ | :( |
11:02.07 | *** part/#asterisk batfastad (~benbradle@87-194-37-131.bethere.co.uk) |
11:02.20 | deltaflyer4747 | DAMN I HATE CHANSPY :-( |
11:04.29 | *** part/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
11:04.46 | Micc_ | deltaflyer4747, I wish I could be of more help with that. |
11:05.05 | deltaflyer4747 | i know |
11:05.22 | Micc_ | maybe I should test chanspy on my 1.8.5.0 |
11:06.05 | deltaflyer4747 | http://pastebin.com/KGzESjS5 |
11:06.12 | deltaflyer4747 | this is whole transaction... |
11:08.15 | Micc_ | I thought you had to have the other call going before doing chanspy? Looks like ChanSpy is the first call here. |
11:08.26 | deltaflyer4747 | doesn't matter |
11:08.33 | deltaflyer4747 | i can hear spied party |
11:08.39 | deltaflyer4747 | but not vice versa |
11:09.48 | deltaflyer4747 | http://pastebin.com/XsizAMHU |
11:09.53 | deltaflyer4747 | same story |
11:11.58 | *** join/#asterisk jkroon (~jkroon@197.169.242.104) |
11:13.13 | deltaflyer4747 | i know this SHOULD work, but it doesnt :( |
11:13.41 | Micc_ | deltaflyer4747, works great for me on 1.8.5.0 |
11:13.50 | deltaflyer4747 | :( DAMN |
11:13.58 | deltaflyer4747 | whats wrong with this setup :( |
11:14.11 | deltaflyer4747 | what codec do you use? |
11:14.53 | deltaflyer4747 | ie sip show channels |
11:15.35 | Micc_ | g722 |
11:16.03 | *** join/#asterisk gravin (~gravin@217.71.50.60.brf01-home.tm.net.my) |
11:16.14 | deltaflyer4747 | i c |
11:16.18 | deltaflyer4747 | could be the difference |
11:16.21 | deltaflyer4747 | i use alaw |
11:16.34 | Micc_ | maybe. |
11:16.38 | Micc_ | its worth a shot |
11:16.43 | Micc_ | or try ulaw |
11:16.49 | deltaflyer4747 | or that |
11:16.52 | Micc_ | shouldn't make a difference, but who knows. |
11:17.08 | deltaflyer4747 | but ... i am affraid that the camera knows only alaw ... |
11:17.35 | deltaflyer4747 | lets see :) |
11:21.09 | deltaflyer4747 | my deskphone doesn't support 722 |
11:21.11 | deltaflyer4747 | :-/ |
11:21.13 | deltaflyer4747 | darn |
11:21.24 | Micc_ | deltaflyer4747, were the others all using alaw too? it probably helps if they are all using the same codec. |
11:21.36 | Micc_ | they should all support ulaw I would thing. |
11:21.38 | deltaflyer4747 | they were |
11:21.58 | Micc_ | getting tired. 4:21am here. |
11:22.35 | Micc_ | try a soft phone like zoiper. |
11:22.47 | deltaflyer4747 | will try... |
11:23.00 | deltaflyer4747 | but now... go take a rest, i gotta go shopping anyways |
11:23.03 | deltaflyer4747 | thanks pal |
11:23.39 | Micc_ | good luck and good night. |
11:23.43 | deltaflyer4747 | i hope i will catch someone else in the evening :) |
11:23.51 | deltaflyer4747 | good night to you, 1:23 pm here :) |
11:24.01 | deltaflyer4747 | (gmt + 1 DST) |
11:24.07 | deltaflyer4747 | CET |
11:24.39 | *** join/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
11:24.47 | *** part/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
11:27.57 | joker2u | ? |
11:28.05 | *** join/#asterisk lcb (~Affiliate@unaffiliated/lcb) |
11:31.30 | deltaflyer4747 | * |
11:32.19 | deltaflyer4747 | afk, shopping |
11:32.26 | *** join/#asterisk chasing`Sol (~cS@smtp.master-zone.net) |
11:37.47 | *** join/#asterisk Kamineko (~ender@moya.rainside.sk) |
11:40.20 | *** join/#asterisk fulcan (~root@li345-191.members.linode.com) |
11:40.49 | *** part/#asterisk fulcan (~root@li345-191.members.linode.com) |
11:41.48 | Bipul | permit=192.168.5.0/255.255.255.0 ; replace with your network settings < shud we use my Dynamic IP instead of 192.168.5.0 , as i have define host=dynamic |
11:44.50 | nightrid3r | Bipul u use the address of hosts that are allowed to connect |
11:44.59 | *** join/#asterisk joker2u (~root@li345-191.members.linode.com) |
11:45.16 | joker2u | I am trying uderstand the asterisk ao2 iterator. Is the object I am passing data too? it is my understanding that py-asterisk is the interface to the asterisk api that passes/translates to the ao2 object? astobj2.h to be specific? Am I understanding this correctly? |
11:46.19 | Bipul | nightrid3r, i have used permit=my dynamic IP/255.255.255.0 <-- is it correct formate |
11:48.52 | Bipul | this setting shud be for my other computer |
12:03.52 | *** join/#asterisk _zoom_ (~Kurmot@196.1.219.122) |
12:07.16 | lcb | hi. could you please lead me to directions on how to run a basic interactive answering system for 2 analog phone lines, on a debian based server. Is asterisk too much for it? |
12:08.05 | WIMPy | lcb: That's exactely what Asteris is good at. Try the |
12:08.07 | WIMPy | ~book |
12:08.07 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
12:08.53 | lcb | WIMPy: great. i were trying to overpass the whole manual due to language concerns. |
12:10.04 | lcb | WIMPy: btw, is it possible to have a remote interface /frontend to control it? |
12:10.21 | WIMPy | Control what? |
12:10.47 | lcb | or do i need to run graphically on the server, meaning installing a graphical desktop environment |
12:10.54 | WIMPy | It has several options for remote control. |
12:11.01 | lcb | ohh ok. |
12:11.19 | lcb | WIMPy: thanks so much |
12:11.40 | WIMPy | No. There are web frontends, but they have some limitations and are only recommended for basic installs. |
12:12.15 | lcb | i see. in your opinion a basiv desktop environment would be great, don't you think? |
12:12.24 | lcb | *basic* |
12:12.30 | WIMPy | Asterisk has a shell for control as well as the management interface and the option to call external scripts. |
12:12.49 | WIMPy | Desktop envirronment? |
12:13.26 | lcb | graphical environment (gnome, kde, etc) |
12:13.47 | WIMPy | You don't need that. |
12:14.13 | WIMPy | And it is said to be able to cause timing issues on slow machines. |
12:14.51 | lcb | i see. so a remote solution would be better, i guess. or locally trough CLI |
12:15.06 | lcb | WIMPy: tks. have a good Sunday. |
12:15.40 | WIMPy | I've been using it for years on my desktop without issues. on a Core2Duo 2.4GHz. |
12:16.27 | lcb | WIMPy: that's the same one i'm trying to install asterisk. but is a server, with 4 G ram |
12:17.07 | lcb | an old desktop with no monitor or keyboard (accident) |
12:17.32 | lcb | i mean, laptop, sorry |
12:18.40 | lcb | reading the book. i'll be back in one month ;) |
12:19.20 | WIMPy | thinks it will be earlier :-) |
12:19.56 | *** join/#asterisk joker2u (~root@li345-191.members.linode.com) |
12:20.18 | lcb | :) |
12:20.42 | lcb | paying and downloading the book was fast, though |
12:21.08 | lcb | :) |
12:21.31 | *** join/#asterisk gravin (~gravin@217.71.50.60.brf01-home.tm.net.my) |
12:29.09 | *** join/#asterisk war9407 (war@c-71-62-61-74.hsd1.va.comcast.net) |
12:37.53 | deltaflyer4747 | Bipul: i haven't seen any "permit" line in the example i've sent to you |
12:38.03 | deltaflyer4747 | WIMPy: Hi, have spare minutes? |
12:39.03 | WIMPy | deltaflyer4747: Never. But try to ask anyway :-) |
12:39.09 | deltaflyer4747 | :)) |
12:39.16 | deltaflyer4747 | remember my yesterdays problem? |
12:39.21 | WIMPy | yes |
12:39.26 | deltaflyer4747 | i think i get your idea now |
12:40.08 | deltaflyer4747 | but when i try to chanspy on 11 for 1002, i get no sound through whisper to 11 |
12:40.19 | deltaflyer4747 | 11 is spied-on extension, 1001 is the spy |
12:40.32 | deltaflyer4747 | tested with pure sip devices, but no luck |
12:41.35 | WIMPy | You're saying chanspy doesn't work for you, not even without that camera thing? |
12:41.58 | deltaflyer4747 | yes |
12:42.07 | deltaflyer4747 | http://pastebin.com/XsizAMHU |
12:42.36 | deltaflyer4747 | exten => 3131,1,Chanspy(SIP/11,qw) |
12:42.58 | deltaflyer4747 | i dial from 11 to 81 and then from another sip phone i dial 3131. I can hear 11 but 11 cannot hear me. |
12:44.11 | deltaflyer4747 | when i do W, i ofc. cannot hear 11 yet 11 still cannot hear me. |
12:44.21 | deltaflyer4747 | all extensions use same codec |
12:44.31 | deltaflyer4747 | (alaw) |
12:44.54 | WIMPy | What Version are you using? Have you checkt your version has option w? |
12:44.55 | Bipul | deltaflyer4747, can you past me a link whear i can learn how to setup a trunk to my voip account |
12:45.07 | deltaflyer4747 | it doesn't matter if i create chanspy before or after dialing 11-81 http://pastebin.com/KGzESjS5 |
12:45.11 | deltaflyer4747 | Bipul: http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf |
12:45.15 | deltaflyer4747 | already pasted today ;) |
12:45.28 | deltaflyer4747 | WIMPy: latest available - 1.8.5.0 |
12:45.40 | WIMPy | Hmm. |
12:45.52 | deltaflyer4747 | upgraded today from 1.6 |
12:46.02 | Bipul | have not sleep for 2 days my mind ohh.. |
12:46.19 | deltaflyer4747 | mov bipup,sleep |
12:46.44 | deltaflyer4747 | WIMPy: same setup works for Micc_ |
12:46.48 | deltaflyer4747 | in 1.8.5.0 |
12:46.50 | Bipul | the link which you have given to me is for sip trunking? |
12:47.17 | deltaflyer4747 | Bipul: for sip general config. |
12:47.29 | WIMPy | deltaflyer4747: I haven't actually tried to call ChanSpy with w, only activated it via DTMF. |
12:47.45 | deltaflyer4747 | extension and trunk are not so much different |
12:47.47 | WIMPy | But if it works for others, it starts to become strange. |
12:47.57 | deltaflyer4747 | WIMPy: i tried that as well, wasn't working (option d) |
12:48.31 | deltaflyer4747 | # works for changing volume, 456 does nothing. |
12:50.53 | WIMPy | Ok, just checked. It works for me with qw. |
12:51.22 | WIMPy | So we need to find out why it doesn't work for you. |
12:51.46 | WIMPy | Unfortunatly I don't have an idea what might cause that. |
12:51.58 | deltaflyer4747 | i can give you every log you ask for |
12:52.15 | deltaflyer4747 | but ... i don't have that idea as well. |
12:52.28 | deltaflyer4747 | sec, i will call external line if the problem is in 81... |
12:52.53 | deltaflyer4747 | it is. |
12:52.56 | deltaflyer4747 | DAMN ! |
12:53.01 | WIMPy | Huh? |
12:53.04 | deltaflyer4747 | yes. |
12:53.12 | deltaflyer4747 | i dialed external line and it works! |
12:53.20 | WIMPy | What's happening exactely? |
12:53.30 | deltaflyer4747 | if i dial 81, it doesn't work. |
12:53.41 | deltaflyer4747 | if i dial external line, it works. |
12:53.52 | deltaflyer4747 | gotta go, be back in 0.5h |
12:53.56 | WIMPy | 81 is a phone? |
12:53.58 | deltaflyer4747 | (moving furniture) |
12:54.01 | deltaflyer4747 | 81 is that camera. |
12:54.02 | deltaflyer4747 | afk |
12:54.03 | *** join/#asterisk bmg505 (~leon@196-209-7-14.dynamic.isadsl.co.za) |
12:57.46 | deltaflyer4747 | back |
12:57.51 | deltaflyer4747 | so thats strange. |
12:57.54 | *** join/#asterisk sulex (~sulex@pdpc/supporter/professional/sulex) |
12:58.41 | deltaflyer4747 | let me test something |
12:59.33 | WIMPy | Asterisk might get confused because the cam is not sending RTP. |
13:01.33 | deltaflyer4747 | i don't know really... |
13:01.59 | deltaflyer4747 | i might know the answer |
13:02.06 | deltaflyer4747 | <PROTECTED> |
13:06.41 | *** join/#asterisk batfastad (~benbradle@87-194-37-131.bethere.co.uk) |
13:08.21 | WIMPy | Looks perfectly ok to me. |
13:09.17 | WIMPy | But someone more in to RTP and how ChanSpy works internally might be able to tell you more. |
13:10.17 | deltaflyer4747 | i'm trying to recall how is that core debugging command in cli... |
13:10.56 | WIMPy | core set vebose|debug # |
13:11.08 | deltaflyer4747 | thanks |
13:11.13 | WIMPy | I think there's an RTP debug as well. |
13:11.25 | *** join/#asterisk jkroon (~jkroon@197.171.223.103) |
13:12.23 | deltaflyer4747 | yes |
13:12.40 | deltaflyer4747 | now only how could i see it ... |
13:13.07 | deltaflyer4747 | ok |
13:13.08 | deltaflyer4747 | :) |
13:15.34 | batfastad | Hi everyone. We currently have a hosted Asterisk/FreePBX system running in an OpenVZ container, we're only 15 extensions and it works great. But does anyone have any experience in running under Xen? |
13:16.19 | deltaflyer4747 | okay... i see lot of << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/11-0000000d] |
13:17.46 | batfastad | I'd like to move it under our control on our colocated Xen server. We don't use any special telephony hardware as we have a SIP trunk provider for PSTN conversion |
13:18.55 | deltaflyer4747 | thats apparently the output from the camera itself. |
13:18.56 | WIMPy | deltaflyer4747: I'm off for an hour or so. But that's an area where I won't be able to provide much help anyway. |
13:19.14 | deltaflyer4747 | WIMPy: i c |
13:20.10 | Jasnejac | batfastad: I have several installations under Xen - no problems at all |
13:22.36 | deltaflyer4747 | okay, filtered those null frames with dtmftype=inband |
13:24.51 | batfastad | Jasnejac: That's great news. I figured it should be similar if not better than OpenVZ because they're both paravirt |
13:27.05 | deltaflyer4747 | darn with that routing |
13:35.04 | deltaflyer4747 | i might see an problem here :) |
13:37.06 | *** join/#asterisk esperanto (~rusty@pool-71-114-139-216.hrbgpa.dsl-w.verizon.net) |
14:01.38 | Bipul | dtmfmode= ? what is that is it related to sounds ,tones |
14:01.56 | Bipul | is it codec? |
14:04.06 | *** part/#asterisk batfastad (~benbradle@87-194-37-131.bethere.co.uk) |
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14:14.14 | Jasnejac | Bipul: in call key press sounds |
14:14.36 | deltaflyer4747 | Bipul: have you ever TRIED to read the http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf i've sent you? |
14:14.40 | Bipul | ok thanks Jasnejac |
14:14.48 | deltaflyer4747 | its described there ;) |
14:14.56 | Bipul | yes deltaflyer4747 |
14:15.05 | *** join/#asterisk glam (~glam@183.13.29.107) |
14:17.36 | Bipul | ok after editing the things at sip.conf we have to reload the sip.conf ar CLI> |
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14:24.49 | Bipul | p3nguin, ping |
14:24.56 | p3nguin | yes? |
14:25.11 | Bipul | can i pm you |
14:25.15 | p3nguin | yes. |
14:25.22 | Bipul | thank you. |
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14:44.40 | dym | greetings! |
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14:49.49 | *** mode/#asterisk [+o russellb] by ChanServ |
14:50.17 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
14:50.21 | [sr] | hi |
14:53.25 | WIMPy | Isn't it great if you can start the day with a steak and a cake fom yesterdays BBQ? |
14:54.08 | [sr] | BBQ = ? |
14:54.35 | p3nguin | You don't know what bbq is? |
14:54.37 | WIMPy | Barbecue |
14:54.51 | [sr] | WIMPy: in that case i agree |
14:54.52 | [sr] | :p |
14:55.13 | [sr] | p3nguin: no... i'm portuguese.. :p |
14:55.28 | [sr] | but barbecue I know of course, but now i know what BBQ is :) |
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15:14.31 | deltaflyer4747 | so... anyone able to help me with chanspy problem? Its perhaps something with RTP data |
15:15.58 | p3nguin | I only happens when using the moh through a local channel. When I spy on a phone, it works correctly. |
15:17.04 | p3nguin | s/I/It/ |
15:17.52 | p3nguin | I didn't try spying from my phone using a local channel. I should test that later. |
15:19.41 | *** join/#asterisk pqwild68 (6eae7388@gateway/web/freenode/ip.110.174.115.136) |
15:19.48 | deltaflyer4747 | p3nguin: it happens only when i spy on the 81, if i call regular extension, it works. . . |
15:20.09 | p3nguin | Your incorrect usage of terminology confuses me. |
15:20.18 | deltaflyer4747 | so there is something terribly wrong with that extension |
15:20.36 | deltaflyer4747 | ? |
15:20.39 | p3nguin | Extensions are how you write them. If it's wrong, you wrote it wrong. |
15:20.53 | deltaflyer4747 | ok, to that channell... |
15:22.40 | deltaflyer4747 | afk, brb, going to test motion detection of that camera |
15:22.56 | WIMPy | I suspect the cam might not send any RTP, which is something that caused massive problems in the past. Might still upset ChanSpy. |
15:23.29 | p3nguin | I tested both chanspy and extenspy using two phones and a local channel that calls moh. |
15:23.55 | p3nguin | It connects the local channel, moh starts, but the sound never makes it to the phone. |
15:24.13 | WIMPy | No need to go that far |
15:24.25 | p3nguin | I had to replicate his setup. |
15:24.46 | WIMPy | There's no MOH involved yet. It works on a call between two phones, but not on a call between a phone and the cam. |
15:25.15 | p3nguin | When someone presses the button at him cam, it puts the cam's audio into an moh stream. |
15:25.51 | WIMPy | Unless i misunderstood something. |
15:25.56 | p3nguin | I think you did. |
15:26.20 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
15:26.44 | WIMPy | Yes, but that is not the issue. The connection between the phone and the cam seems to have issues already. |
15:27.02 | p3nguin | The cam has a speaker and a mic. When the button is pressed, it connects a call between the cam and the deskphone. The speaker on the cam can hear the deskphone, but the deskphone cannot hear the mic from the cam. Because of that, he rigged the cam's mic audio into an moh stream. |
15:27.03 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
15:27.27 | WIMPy | Jepp. Been though that setup yesterday. |
15:27.49 | p3nguin | So the deskphone could listen to that moh and hear the cam's audio. |
15:27.50 | pqwild68 | Hello |
15:28.19 | pqwild68 | Is anyone using the alarmreceiver module? |
15:28.59 | WIMPy | I guess it wouldn't exist if noone had a need for it. |
15:28.59 | p3nguin | But the deskphone can only have one call at a time: either connected to the cam so the cam can hear the deskphone, or connected to moh to listen to the cam's audio |
15:29.50 | WIMPy | I know. I suggested to use ChanSpy in whisper mode to inject MOH to only one leg of the call. |
15:30.12 | p3nguin | That's what I tested. It doesn't work. |
15:30.23 | WIMPy | But whisper doesn't work on that call. Not from a 3rd (2nd) phone, either. |
15:30.24 | *** join/#asterisk sourcode (~code@ppp-58-8-161-141.revip2.asianet.co.th) |
15:30.47 | WIMPy | That would make two issues then. |
15:30.53 | p3nguin | As I said, the spy starts, the local channel is connected, the moh starts, but the moh audio is not being put into the call. |
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15:32.05 | p3nguin | Since I use ChanSpy and ExtenSpy regularly and they work correctly when spying from my phone directly, my next test is to spy a local channel which dials my phone. If it fails, then the problem is spying via local channel. |
15:32.46 | WIMPy | It might also be because MOH is alway on hold. |
15:33.20 | WIMPy | has just tested that. There are indeed two issues as it seems. |
15:33.21 | p3nguin | If I call the extension where MusicOnHold() runs, the music plays. |
15:33.25 | WIMPy | Bad. |
15:33.57 | p3nguin | I also need to test calling the moh via local channel, similar to the way I tried to spy from it. |
15:34.08 | p3nguin | So I have two tests to perform. |
15:34.34 | WIMPy | The MOH from an empty confbridge doesn't arrive, either. |
15:34.48 | deltaflyer4747 | guys guys ... |
15:35.03 | deltaflyer4747 | you are both right. |
15:35.16 | deltaflyer4747 | chanspy cannot spy for MOH |
15:35.32 | deltaflyer4747 | and my setup is somewhat faulty as camera doesn't send any RTP (aka "silence detection") |
15:35.58 | p3nguin | I don't understand why moh can't spy via local channel. |
15:36.23 | p3nguin | ~book |
15:36.23 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
15:36.26 | p3nguin | bipul: ^^^ |
15:38.15 | WIMPy | Playback instead of MOH doesn't work, either. |
15:39.34 | WIMPy | Ugh. |
15:39.37 | deltaflyer4747 | so first thing i need to do is somehow produce some RTP data or force asterisk to ignore it |
15:39.58 | WIMPy | After I called ChanSpy from a phone, I got a fragemnt of what I used in the Playback. |
15:41.38 | deltaflyer4747 | is there a way how to force * to ignore "silent detection" on some phones |
15:44.46 | p3nguin | I think Asterisk ignores silence detection from everything, since it doesn't support silence detection. |
15:45.55 | deltaflyer4747 | well... |
15:45.55 | dym | :D |
15:46.31 | deltaflyer4747 | on http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf is important sentence: Asterisk uses the incoming RTP Stream as a timing source for sending its outgoing Stream. If the incoming stream is interrupted due to silence suppression then musiconhold will be choppy. So in conclusion, you cannot use silence suppression. Make sure ALL SIP phones have disabled silence suppression. There is a solution for the silence suppr |
15:47.22 | dym | Sip Phones |
15:47.25 | dym | Not the Asterisk |
15:49.14 | deltaflyer4747 | dym: you don't get the story ;) |
15:49.58 | dym | I guess p3nguin will answer anyways |
15:50.25 | deltaflyer4747 | camera is not sending any RTP data |
15:50.43 | deltaflyer4747 | ie there is "silence detection turned on" on that camera |
15:52.33 | p3nguin | Asterisk does not care. |
15:52.52 | p3nguin | Asterisk does not support silence suppression. |
15:52.58 | WIMPy | It uesed to. |
15:53.02 | WIMPy | -e |
15:53.07 | p3nguin | It never has in my lifetime. |
15:53.33 | deltaflyer4747 | https://issues.asterisk.org/view.php?id=5374 |
15:53.37 | WIMPy | On a phone with silence suppression I couldn't hear anything unless I talked myself. |
15:53.48 | deltaflyer4747 | thats it WIMPy |
15:53.53 | p3nguin | That's the phone's problem and has nothing to do with Asterisk. |
15:53.58 | deltaflyer4747 | Asterisk uses the incoming RTP Stream as a timing source for sending its outgoing Stream. |
15:54.08 | deltaflyer4747 | ie |
15:54.13 | WIMPy | I just tried to put the chanspy on a local channel and use MOH directly. Doesn't work, either. |
15:54.26 | deltaflyer4747 | no incoming RTP stream = no data sent |
15:54.27 | WIMPy | No. That was an Asterisk Problem. |
15:54.54 | WIMPy | Asterisk didn't send RTP unless also receuving. |
15:54.54 | WIMPy | From the same devide, tah is. |
15:54.58 | WIMPy | that |
15:55.39 | WIMPy | Would be logical if it didn't send, what it didn't receive from the other end. |
15:55.47 | pa | i have a local asterisk server with a dahdi channel to connect to the telephone network. how can i configure asterisk to be able to send out faxes through this dahdi channel? (i dont care about receiving faxes, for the moment) |
15:56.49 | dym | pa: http://ofps.oreilly.com/titles/9780596517342/asterisk-Fax.html#Fax_id265396 |
15:57.36 | pa | thanks a lot, i check it out right now! |
15:57.43 | dym | np |
15:59.08 | deltaflyer4747 | so guys... this seems hopeless, right... :( |
15:59.56 | *** join/#asterisk ChUbB (~IceChat7@cpc2-aztw12-0-0-cust229.aztw.cable.virginmedia.com) |
16:01.37 | *** part/#asterisk ChUbB (~IceChat7@cpc2-aztw12-0-0-cust229.aztw.cable.virginmedia.com) |
16:03.01 | WIMPy | Close |
16:03.20 | deltaflyer4747 | ? |
16:03.38 | WIMPy | You might be able to overcome the RTP issue by putting a conference in between. |
16:03.56 | WIMPy | But the injection Issue is also present. |
16:04.36 | WIMPy | But at least we learned something new. |
16:05.20 | deltaflyer4747 | well, as i asked, can i setup such conference, where everybody listens to the manager and manager can choose to whom he listens to? |
16:05.32 | WIMPy | No |
16:05.36 | deltaflyer4747 | :( |
16:06.06 | deltaflyer4747 | or... |
16:06.13 | deltaflyer4747 | might as well do the trick ... |
16:06.26 | deltaflyer4747 | no, won't work |
16:06.34 | WIMPy | I just thought about an additional conference to make the phone actually receive (silent) audio so that ChanSpy *might* work on that channel. |
16:06.43 | deltaflyer4747 | yea i know |
16:06.58 | deltaflyer4747 | but there is still chanspy vs moh |
16:07.15 | WIMPy | Yes. Two issues. |
16:07.18 | deltaflyer4747 | :( |
16:07.21 | deltaflyer4747 | crap |
16:09.22 | WIMPy | Has anyone here tryed using Imcomplete()? |
16:13.34 | WIMPy | Is there any way to manipulate the dialed extension, other thatn Goto()? |
16:15.28 | *** join/#asterisk irroot (~irroot@197.169.159.67) |
16:18.40 | deltaflyer4747 | OMG :) |
16:18.47 | deltaflyer4747 | that camera uses wrt54g hardware :D |
16:19.07 | deltaflyer4747 | line from config: "/etc/wrt54g.large.ico" |
16:20.28 | WIMPy | You might be able to replace the client then. |
16:22.05 | deltaflyer4747 | exactly! |
16:22.11 | deltaflyer4747 | just trying to SSH into it |
16:23.49 | deltaflyer4747 | afk, lift is again stuck. |
16:26.15 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
16:41.55 | [sr] | I'm sad :( |
16:46.04 | deltaflyer4747 | me2 |
16:48.20 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
16:48.43 | p3nguin | metwo! |
16:49.41 | deltaflyer4747 | ? |
16:52.07 | p3nguin | 2 = two |
16:52.12 | p3nguin | me2 = metwo |
16:53.15 | [sr] | p3nguin: :p |
16:55.59 | [sr] | WIMPy: when you invite me to DE to present some friends (girls) to me? :p |
16:56.13 | dym | WIMPy: has friends/girls? |
16:56.15 | dym | :D |
16:56.44 | *** join/#asterisk fabiobik (5d660b57@gateway/web/freenode/ip.93.102.11.87) |
16:56.52 | fabiobik | Hi guys |
16:57.00 | WIMPy | :-) |
16:57.32 | deltaflyer4747 | p3nguin: well then read it aloud... |
16:57.34 | fabiobik | I have an Huawei E220 usb internet modem and i know i can convert it to SMS gateway |
16:57.43 | fabiobik | and i can unlock voice feature too |
16:57.46 | p3nguin | I DID! |
16:58.05 | fabiobik | i want to know if i can make an gsm gateway with asterisk |
16:58.26 | WIMPy | See chan_datacard |
16:58.27 | fabiobik | and this usb modem |
16:58.27 | *** join/#asterisk radic (~radic@tmo-097-61.customers.d1-online.com) |
16:58.36 | WIMPy | Or google for that. |
16:58.49 | fabiobik | p3nguin what you did? |
16:59.01 | dym | fap |
16:59.10 | *** join/#asterisk Nasga (~Nasga@AAmiens-157-1-106-45.w86-208.abo.wanadoo.fr) |
16:59.20 | fabiobik | WIMPy but you know if this possible? |
16:59.28 | deltaflyer4747 | :)) |
16:59.50 | WIMPy | Like I said: Look at chan_datacard. |
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17:00.30 | *** part/#asterisk DennisG_ (~DennisG@ip5454b5b3.adsl-surfen.hetnet.nl) |
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17:01.16 | dym | Anyways - gsm gateway? What exactly do you mean by that |
17:01.18 | dym | ? |
17:01.40 | *** join/#asterisk DennisG_ (~DennisG@ip5454b5b3.adsl-surfen.hetnet.nl) |
17:03.08 | fabiobik | dym gsm gateway to make calls |
17:03.20 | dym | well |
17:03.25 | dym | since its a data device: doubt it. |
17:03.50 | fabiobik | dym my data device can be unlocked to voice too :) |
17:04.08 | nightrid3r | fabiobik there are cards with up to 4 gsm devices on it |
17:04.46 | fabiobik | nightrid3r not understand |
17:04.49 | fabiobik | nightrid3r what u mean |
17:05.10 | WIMPy | The E1550 (which I have) seems to exist with voice support, but that's all I could find out. No idea if is's a matter of unlocking or reflashing or whatever. |
17:05.40 | fabiobik | WIMPy is that Huawei? |
17:05.50 | WIMPy | yes |
17:06.05 | fabiobik | humm |
17:06.11 | nightrid3r | fabiobik http://www.openvox.cn/products/show.php?itemid=150&lang=2 |
17:06.39 | WIMPy | chan_datacard detects it ok, but no calls. |
17:07.58 | fabiobik | nightrid3r yes but that is 300 dollares |
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17:09.33 | Bipul | THanks alot ! to this asterisk community >:D< |
17:09.53 | fabiobik | what im trying to do is this conect my data modem unlocked with asterisk and then make an an record with my voice saying "Dial the number" and then write the number and connect to that number with sip |
17:10.01 | fabiobik | Its possible or im crasy |
17:10.02 | fabiobik | lool |
17:10.30 | fabiobik | ? |
17:10.36 | dym | What does the gsm gw have to do with all that? |
17:11.08 | fabiobik | dym establish the connection between my phone and pc? |
17:12.08 | dym | i dont quite understand the szenario |
17:12.28 | dym | make a voice record and dial the number therefore you need a gsm connection from your mobile to asterisk |
17:12.36 | dym | doesnt get much more odd |
17:12.37 | *** join/#asterisk ExpertOrBust (~ExpertOrB@wsip-24-234-159-70.lv.lv.cox.net) |
17:13.21 | WIMPy | I guess it's a call through thing. But it wasn't clear to me, either. |
17:13.55 | deltaflyer4747 | YAY, i got telnet into camera ! :) |
17:13.56 | Nugget | telnet is eeeeeeevil! |
17:14.08 | dym | telnet into camera? |
17:14.15 | dym | are you all on drugs? |
17:14.29 | nightrid3r | yay ASCI pron :) |
17:14.41 | deltaflyer4747 | dym: ? |
17:14.42 | fabiobik | ya its calling trught that |
17:14.52 | deltaflyer4747 | Nugget: i am glad atleast for that :) |
17:14.57 | fabiobik | my english is not the best |
17:14.57 | fabiobik | lol |
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17:16.15 | fabiobik | dym what i want to doo is make calls trught that ... and for example on connection ear "dial the number" |
17:16.32 | fabiobik | *when the connection was stabilshed |
17:17.13 | *** join/#asterisk Leddy (leddy@89.238.176.88) |
17:18.17 | nightrid3r | fabiobik DISA ? |
17:18.37 | fabiobik | what is that? |
17:18.53 | nightrid3r | i guess thats what your looking for |
17:19.02 | deltaflyer4747 | :D |
17:19.04 | WIMPy | A dialplan application you may want to call for that scenario. |
17:19.33 | fabiobik | phone ----> pc ----> SIP |
17:21.04 | fabiobik | is that? |
17:25.13 | deltaflyer4747 | O M G |
17:25.33 | deltaflyer4747 | do you know what processes that SIP ? |
17:25.55 | WIMPy | I guess you will tell us in a moment. |
17:26.06 | deltaflyer4747 | rtsps... |
17:26.16 | deltaflyer4747 | Streaming Server App Version: 1.1.0.0 RTSPSTREAMING Server Module Version: 1.7.1.4 |
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18:51.08 | tonsofpcs | I'm looking at running an asterisk system as a temporary hold-over between PBXs (replacing an old relay-driven PBX with a modern system -- will eventually use the asterisk setup as a failsafe amongst other things). Is there a nice simple frontend and/or easy to configure pre-made distribution of some kind? |
18:53.29 | deltaflyer4747 | there is |
18:53.42 | deltaflyer4747 | but i don't know if i can mention it there |
18:54.45 | tonsofpcs | can you mention it in a PM? |
18:54.53 | mickecarlsson | is shameless |
18:55.07 | mickecarlsson | FreePBX Distro or AsteriskNOW |
18:55.19 | mickecarlsson | Both run FreePBX |
18:55.26 | deltaflyer4747 | or trixbox for that matter... |
18:55.35 | mickecarlsson | DONT got trixbox, it is dead |
18:55.45 | mickecarlsson | and use a very old freepbx |
18:55.49 | tonsofpcs | thanks :) |
18:56.06 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
18:57.04 | tonsofpcs | asterisknow looks cool.... I wonder if anyone has made a bootCD version |
18:57.26 | *** join/#asterisk nightrid3r (~nightrid3@91.176.182.178) |
18:57.37 | tonsofpcs | (live CD) |
18:57.43 | mickecarlsson | Nope |
18:57.58 | mickecarlsson | AsteriskNOW is made by Digium |
18:58.11 | mickecarlsson | Use FreePBX as the GUI, or their own, chosen at install |
18:58.19 | deltaflyer4747 | mickecarlsson: really? thanks for the info |
18:58.41 | mickecarlsson | Yes |
18:58.54 | deltaflyer4747 | good to know |
18:59.08 | mickecarlsson | FreePBX Distro is a Netinstall, download all setup from Internet |
18:59.27 | mickecarlsson | http://www.asterisk.org/asterisknow/ |
18:59.40 | tonsofpcs | now I just need to find some hardware :) |
18:59.58 | mickecarlsson | You can use VMware or VirtaulBOX to test it out |
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19:00.12 | deltaflyer4747 | :)) |
19:00.15 | tonsofpcs | oh, I mean interface hardware :) |
19:00.24 | deltaflyer4747 | thats expensive. |
19:00.33 | tonsofpcs | (and if someone makes a card that can talk to nitsuko handsets..... |
19:00.33 | mickecarlsson | Well, go by SIP and you wont need it |
19:01.13 | tonsofpcs | mickecarlsson: sure you will, it's just contained in the handset base instead of the PBX 'host'. |
19:01.51 | mickecarlsson | I use DECT phones that talk SIP |
19:02.11 | mickecarlsson | Never used any hardware except for the server |
19:02.23 | mickecarlsson | By hardware, I mean any FXS/FXO card |
19:03.04 | mickecarlsson | gtg, soccer on tv |
19:04.01 | deltaflyer4747 | yea, sip DECT phones are really cheap |
19:05.00 | tonsofpcs | mickecarlsson: it's football and they've been hitting the ball out so much that there's plenty of time to chat |
19:05.05 | tonsofpcs | (go USA!) |
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20:00.31 | AlexForster | Best: 99.967 -- Worst: 96.487 -- Average: 99.186181, Difference: 99.985145 |
20:00.48 | AlexForster | definitely the cause of my choppy audio (asterisk on centos under hyperv), correct? |
20:01.06 | WIMPy | Probably |
20:01.57 | AlexForster | anyone know a solution off the top of their heads, before i go googling? |
20:02.20 | WIMPy | Shared IRQs? |
20:03.02 | AlexForster | that sounds like "install integration services" |
20:03.08 | AlexForster | ? |
20:03.16 | WIMPy | Drivers of other hardware locking up? |
20:03.18 | WIMPy | What? |
20:04.12 | AlexForster | nevermind, misunderstood |
20:09.56 | deltaflyer4747 | is p!$$3d off :-/ |
20:11.55 | p3nguin | Someone once told me that it is better to be pissed off than to be pissed on. |
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20:42.19 | okei | p3nguin: hello, can u help me ? :) |
20:45.17 | okei | guys what is TARGET0 funciton in this string? exten => _20[0-3],1,Set(TARGETNO=${EXTEN}) |
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20:46.04 | p3nguin | TARGETNO is a variable in that line. |
20:46.31 | okei | p3nguin: yes but whats function in reality? |
20:46.44 | okei | it's only formal? |
20:47.18 | p3nguin | Show me the rest of your dial plan and I'll try to figure out its purpose. |
20:47.24 | WIMPy | It's just some variable. |
20:47.47 | p3nguin | I already said that, but apparently that answer wasn't good enough. |
20:47.58 | DrDigital | anyone able to help me figure out what riser cards i need and what kind of ram to buy |
20:48.00 | DrDigital | http://www.alvio.com/xABK_PID56007_S2891G2NR_tyan-computer_tyan-thunder-k8sre-s2891-s2891g2nr-socket-940-eatx-motherboard-oem.html&referrer=froogle |
20:48.10 | DrDigital | i have that in a 1U rackable systems case |
20:48.36 | okei | p3nguin: http://pastebin.com/DBqQMNAA |
20:48.41 | okei | example from tutorial |
20:49.25 | p3nguin | They are using the variable to retain the original number that was called. |
20:51.09 | okei | p3nguin: retain? or determine |
20:51.17 | p3nguin | retain |
20:51.48 | p3nguin | It is being set when the call comes in. It is then being used later when the call goes to voicemail. |
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20:52.36 | WIMPy | ... assuming that extension and VM box are identical. |
20:53.29 | p3nguin | Well, they are doing it regardless of the end result being a failure or not. |
20:56.31 | lcb | someone, pls, could help me resolve this 'make: *** [all] Error 2' while installing on a ubuntu 11.04 server -> |
20:56.45 | lcb | http://pastebin.com/B1S0nGvL |
21:01.10 | lcb | wait... i missed LibPRI install... probably due to that |
21:01.55 | WIMPy | Did you use the latest Asterisk? |
21:02.16 | lcb | WIMPy: i'm following "by the book" :) |
21:02.26 | jmwpc | In my contest for inbound google voice calls, I can't seem to get my extension to ring more than 3 times. On the other end of the call, I can hear 2 more rings before being sent to google voice mail. I'm dialing 2 extesions (it did the same thing with a single extension): Dial(SIP/101&SIP/102,20,D(:1)) ... I have increased the timeout setting up and down with no change, what am I missing? |
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21:02.44 | jmwpc | *context* |
21:03.33 | p3nguin | That's still just one extension, not two. |
21:04.20 | jmwpc | p3nguin: both are ringing :) I even tried a dialgroup... same thing happened. I can't seem to get the timeout to change. |
21:04.37 | p3nguin | Show me your dial plan. |
21:04.40 | lcb | WIMPy: this one 'svn co http://svn.asterisk.org/svn/asterisk/branches/1.8' |
21:05.01 | lcb | WIMPy: isn't that correct? |
21:05.06 | jmwpc | p3nguin: k... let me get it pasted |
21:05.50 | p3nguin | lcb: Don't use the svn unless you know what you're doing. Use the current RELEASEd version. |
21:05.50 | WIMPy | lcb: That's fine. |
21:06.09 | p3nguin | And if you have to ask if it's right, you obviously don't know what you're doing. |
21:06.22 | lcb | p3nguin: is not the first time i'm using it... but if say so. |
21:07.02 | WIMPy | It's 1.8, not trunk. |
21:07.04 | lcb | p3nguin: i asked because as you know might be several sources of it, i don't mean on this software. |
21:07.45 | lcb | btw, i'm doing it all as the book says |
21:07.59 | p3nguin | wimpy: That svn will check out the current releast of 1.8 rather than the current day's build? |
21:08.26 | WIMPy | No, but it's supposed to be stable. |
21:08.57 | WIMPy | Only trunk might not be. In theory that it, off course. |
21:09.05 | WIMPy | is |
21:09.06 | lcb | now i forgot the release * to add it on svn co http://svn.asterisk.org/svn/libpri/tags/1.4. |
21:09.16 | lcb | :( |
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21:09.34 | WIMPy | ??? |
21:10.00 | lcb | WIMPy: to pull it from SVN is necessary that |
21:10.46 | WIMPy | Not sure what you mean. |
21:10.58 | lcb | cd ~/src/asterisk-complete/ | mkdir libpri | cd libpri/ | svn co http://svn.asterisk.org/svn/libpri/tags/1.4.<your version number> |
21:11.04 | jmwpc | p3nguin: http://pastebin.com/EfmvVeWT |
21:11.41 | jmwpc | p3nguin: (2 separate google voice accounts are being handled here) |
21:11.45 | p3nguin | That'll never work with all those spaces in it. |
21:12.36 | jmwpc | p3nguin: the ones by the separator (comma)? |
21:12.53 | WIMPy | uses /branches/1.4 |
21:13.45 | p3nguin | jmwpc: Correct. You can't have those spaces in extensions.conf. |
21:14.08 | jmwpc | p3nguin: ahh... thanks... let me give that a try |
21:14.27 | p3nguin | exten => 123,1,Stuff() |
21:14.46 | p3nguin | provided you have app_stuff.so, of course. :) |
21:15.01 | jmwpc | p3nguin: Coding habit :) |
21:15.41 | p3nguin | lines 26 and 34 |
21:15.52 | p3nguin | Delete those Answer() lines. |
21:15.58 | Kamineko | hello, im looking for someone with draytec ippbox 2820 experience. i have a problem to get sip trunk to my asterisk/frepbx box working. i always get 401/403 eeror messages |
21:16.00 | p3nguin | Don't answer before the dial. |
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21:18.19 | jmwpc | p3nguin: Remove it entirely? or put it somewhere else? (Just for reference... I'm working from this tutorial: http://pcprob.blogspot.com/2011/03/howto-use-google-and-asterisk-for-free.html) |
21:18.37 | p3nguin | Remove the Answer() lines. |
21:19.52 | p3nguin | Either use the SendDTMF() method, or use the Dial(D()) method. There's no need to mix the two. |
21:20.07 | p3nguin | If you use the SendDTMF method, you'll Answer(), then Wait(), then SendDTMF(). |
21:20.27 | p3nguin | If you use the Dial(D()) method, you'd do nothing else before the Dial. |
21:20.52 | p3nguin | nothing that changes the actual call, that is. You can obviously change things like CID and whatnot. |
21:21.09 | jmwpc | p3nguin: so it rings longer now, but never answers. It keeps ringing to VM on the other end. |
21:21.52 | p3nguin | You'll have to pick up the phone, of course. |
21:22.36 | jmwpc | p3nguin: :) Of course... I did, it was dead air, and kept ringing to VM on the calling phone (my cell) |
21:23.08 | p3nguin | I don't know what's going on with that. It's as if the gtalk/jabber stuff isn't actually working right. |
21:23.08 | jmwpc | So the other method would be Answer -> Wait -> SendDTMF -> Dial... correct? |
21:23.33 | p3nguin | right. It's on the blog page you linked me to. |
21:24.10 | jmwpc | p3nguin: I'll give that a try again |
21:24.27 | p3nguin | And when you use the SendDTMF method, you don't need the D(:1) in your Dial(). |
21:24.58 | jmwpc | p3nguin: got it.. . |
21:25.21 | jmwpc | p3nguin: I can just exclude that parameter then? |
21:25.40 | p3nguin | You'll need to take it out of the Dial command. |
21:26.15 | p3nguin | I mean, you could leave it, but then people will hear "1" in their ears when you pick up your phone. |
21:26.26 | p3nguin | It won't technically hurt anything. |
21:26.47 | jmwpc | p3nguin: I'll leave it in there for people I dont' like :) |
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21:28.07 | lcb | successfully installed libpri-1.4.12 . again 'make' of dahdi is not finishing . same pastebin -> http://pastebin.com/B1S0nGvL |
21:28.52 | lcb | (long ago. i was wating for previous issue being done) |
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21:31.13 | WIMPy | lcb: That dahdi version is most probably too old for your kernel version. |
21:32.27 | lcb | so WIMPy, http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/ ? |
21:33.22 | WIMPy | Either that or take the trunk from svn if you feel comfortable with the latest. |
21:33.46 | WIMPy | Should I have said "Trying" the latest? |
21:33.54 | lcb | WIMPy: :) |
21:33.58 | lcb | don't worry |
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21:36.14 | lcb | it seems to be 2.4.1.2+2.4.1/ |
21:39.30 | lcb | no Errors & Warnings, Inc., on make |
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21:40.12 | okei | p3nguin: penguin, when i trying to leave voicemail debug is http://pastebin.com/L075A9he here, when i have writed this string in voicemail.conf 1001 => 123456,Rati Jokhadze, why this message is right? |
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21:41.28 | okei | and what is voicemail number? |
21:41.31 | okei | 1001 this? |
21:41.40 | lcb | WIMPy: no errors at all. however, no hardware found. not worried by now because the modem i have in there for now is a data/fax and i'll buy another one. |
21:42.18 | WIMPy | dahdi (or zaptel for that matter) has never supported modems. |
21:42.22 | lcb | wondering though why that data/fax was not found |
21:42.29 | WIMPy | There used to be another channel for that. |
21:42.29 | lcb | ahh ok |
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21:43.18 | lcb | WIMPy: thanks for the ""Trying" the latest" |
21:44.19 | Mango | I'm using Asterisk 1.8.4.4 and trying to subscribe to MWI from my DID provider. I've added mwi => mango@inbound1/12345 to my sip.conf. It sends the subscribe packet, then inbound1 replies with 401 Unauthorized, but my Asterisk never responds with the challenge. It ignores the 401 Unauthorized and retransmits the original SUBSCRIBE packet a bunch of times. Any ideas? |
21:47.17 | jmwpc | p3nguin: I think I got it. The main issue was probably the spaces. I went back to the D(:1) method since I lose google voice mail if I answer before dialing. I get 4 rings now, just about the same time google kicks to VM. |
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21:57.28 | lcb | should i install samples? 'make samples'? |
21:58.14 | WIMPy | If you want an asterisk that allows a lot more than you want it to: yes. |
21:58.15 | lcb | i still have 'sudo make config' but after make install i got that suggestion |
21:58.28 | lcb | yes. tks |
21:58.40 | WIMPy | Don't run it on the internet with the sample configuration. |
21:58.53 | lcb | ok |
21:59.10 | lcb | WIMPy: keep sudoing, isn't? |
21:59.16 | lcb | sudo make samples |
21:59.16 | WIMPy | The sample configs are good for reference, but you can look at them in the source dir. |
21:59.24 | WIMPy | yes |
21:59.38 | lcb | ahh ok, so might be better to leave the defaults. |
22:00.17 | WIMPy | Default means not supported in many cases. |
22:01.16 | lcb | 'sudo make config' by the book... result: System start/stop links for /etc/init.d/asterisk already exist. |
22:01.19 | WIMPy | Only configure those parts you actually need. And make sure they don't do things you don't want. |
22:01.27 | lcb | so just forget it. is configured |
22:01.40 | WIMPy | So where did they come from? |
22:01.48 | lcb | WIMPy: ok. i'll keep reading the book |
22:02.02 | WIMPy | thought that was a fresh install? |
22:02.23 | lcb | WIMPy: i don't know... probably through any apt-get i did before... :( i don't recall it |
22:03.16 | lcb | WIMPy: :( probably i did it... because i am looking for a good program like this for two weeks |
22:03.28 | WIMPy | Did you remove the packages? |
22:03.36 | lcb | looks like i didn't |
22:03.45 | lcb | i'll go over again |
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22:04.01 | WIMPy | Might be a good idea to do so and the do the 'make install's again. |
22:04.07 | WIMPy | then |
22:04.20 | lcb | i thought i did an install in this machine i'm using now... :( |
22:04.29 | lcb | with apt-get |
22:04.55 | lcb | yes, i will |
22:05.49 | lcb | for now it looks great, smooth installation (besides that dahdi outdated) |
22:06.56 | WIMPy | that's why I usually try to avoid hardware, not supported by the standard kernel. But that isn't always possible here. |
22:10.51 | lcb | i'll install again tomorrow. by now apt-get --purge remove asterisk* cleaned it |
22:11.23 | WIMPy | will try todays version. |
22:11.32 | WIMPy | The one from a few days back seems foul. |
22:11.39 | lcb | WIMPy: thank you very much for your time. i have to go. need to prepare some work for tomorrow. bye |
22:13.05 | WIMPy | definitely needs somethign to detect the deth of an asterisk. |
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22:21.20 | AviMarcus | munin? |
22:21.28 | xbp | hi |
22:22.30 | WIMPy | Maybe it can be done in a channel as well. That wouldn't be too bad. |
22:23.59 | WIMPy | munin doesn't look capable of detecting a dead Asterisk. |
22:24.18 | xbp | i think you can detect by 5060 |
22:24.23 | xbp | telnet to it see if sip is answering |
22:24.32 | p3nguin | SIP is UDP. |
22:24.49 | xbp | hrm? |
22:25.20 | xbp | florida bites today |
22:25.42 | p3nguin | SIP is UDP. The telnet client uses TCP. Ergo, telnet is useless to testing SIP. |
22:25.57 | WIMPy | I have a feeling it locks up when a peer dies during a call. |
22:26.16 | WIMPy | But the last time it was IAX, now it was SIP. |
22:31.30 | p3nguin | Set(__myvar=myvalue) <-- this is correct for indefinite inheritance of the variable, right? |
22:31.58 | WIMPy | yes |
22:32.58 | p3nguin | Is there any reason to only use _ instead of __ for that purpose? I can't see any reason it would hurt to cause it to be indefinite rather than just once. |
22:33.59 | WIMPy | NFI. But I wouldn't see why you couldn't do it to all variables. |
22:34.15 | p3nguin | The only reason I could come up with would be if you might need to test the variable for a null value later, and you don't want to explicitly unset it. |
22:34.32 | WIMPy | yes |
22:39.19 | Mango | So what could cause Asterisk to ignore a 401 Unauthorized from another peer? |
22:41.35 | p3nguin | failure to auth, perhaps? |
22:42.10 | Mango | Please explain more :) |
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22:43.14 | p3nguin | Like, your device has not been preauthorized... so the 401 is the challenge to get your device to authenticate. If it fails authentication, it'll generate a 404. |
22:43.34 | Mango | Well...my Asterisk doesn't exactly fail to authenticate, it doesn't even try. |
22:43.53 | p3nguin | If it failed, it would have received a 404, so I know it didn't fail to auth. |
22:44.02 | Mango | Right. |
22:44.15 | p3nguin | I think it's a 404, anyway. Maybe it's a 403. |
22:44.19 | p3nguin | I'd have to go look up the codes. |
22:45.42 | p3nguin | I don't know what would keep your system from trying to authenticate after getting the 401 from your peer, though. That's the challenge. |
22:45.51 | Mango | scratches his head. |
22:47.13 | Mango | I set up IP authentication. Now I get a different error. |
22:48.05 | Mango | Me: SUBSCRIBE! Remote: 200 OK! Remote: NOTIFY! Messages-Waiting: yes! Me: SIP/2.0 481 What the heck was that! |
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22:48.33 | Mango | Actually I said "Call/Transaction Does Not Exist", not "What the heck was that!" |
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