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| 01:41.21 | pabelanger | Man, just burned through 9 hours of toll-free minutes because a meetme channel got stuck |
| 01:41.22 | pabelanger | looks like all meetings last 30mins now |
| 01:41.22 | pabelanger | then, you are kicked |
| 01:41.29 | pabelanger | also surprised we don't have a default conference limit prompt |
| 01:41.44 | dijib | wicked |
| 01:41.44 | dijib | ive done that b4 |
| 01:41.45 | dijib | not the limit |
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| 07:31.51 | Rhomber | is there more of a real time way to check the SIP channel status? |
| 07:32.04 | Rhomber | ChanIsAvail always returns 0 |
| 07:32.05 | ectospasm | than... what? |
| 07:32.24 | Rhomber | than dialing and waiting for a CHANUNAVAIL result |
| 07:32.30 | ectospasm | you sure you're running it on the right channel? |
| 07:32.52 | ectospasm | s/g it/g ChanIsAvail/ |
| 07:32.56 | Rhomber | i can check the sip peers, but they are only updated with the qualify |
| 07:33.03 | Rhomber | lol, of course. |
| 07:34.10 | Rhomber | ChanIsAvail(SIP/david-mobile) |
| 07:34.33 | ectospasm | that's not the full channel name. |
| 07:34.48 | Rhomber | this is just before a Dial, so I can determine if I need to use a fallback delivery means (like a traditional DID via trunk) |
| 07:34.57 | ectospasm | it'd be something like ChanIsAvail(SIP/david-mobile-00000003b) |
| 07:35.11 | Rhomber | i see |
| 07:35.25 | Rhomber | so i have to resolve what the full channel id for the SIP peer is first? |
| 07:35.41 | ectospasm | Before you dial, you can write the channel name to a variable right before you Dial() |
| 07:35.53 | ectospasm | you may need to play around whether you need a global variable, though probably not. |
| 07:36.53 | Rhomber | so I guess, if multiple devices are logged in with that sip account there will be multiple channel id's? |
| 07:37.13 | Rhomber | not that i expect that in my scenario, but just to clarify |
| 07:37.31 | ectospasm | yes. Use the CHANNEL dialplan function to extract the name of the channel for this particular call. |
| 07:38.01 | ectospasm | specifically, CHANNEL(name) |
| 07:39.05 | ectospasm | use something like: Set(MYCHAN=${CHANNEL(name)} |
| 07:39.44 | ectospasm | ...or you may want to set an ASTDB key/value pair to that, so other threads can find that call |
| 07:40.16 | Rhomber | im using a java environment, so it will be different.. more need to understand the underlying concepts and i'll be fine |
| 07:40.49 | Rhomber | i'll have a look into CHANNEL, what you say confuses me a bit.. but looking into it might clear up my confusion :) thanks |
| 07:41.12 | ectospasm | Rhomber: see "core show function CHANNEL" |
| 07:41.18 | ectospasm | ...in the Asterisk CLI |
| 07:41.34 | ectospasm | ...that's meant to be used in dialplan, I dunno how to use it in Java/AGI |
| 07:42.19 | Rhomber | Java/AGI can exec any dialplan command/function so it's not a problem :) |
| 07:42.41 | ectospasm | there ya go, then. |
| 07:43.28 | Rhomber | what confuses me is that there won't be a channel related to the SIP peer in the current context.. |
| 07:43.37 | Rhomber | until i bridge it with dial |
| 07:43.50 | Rhomber | so not sure how CHANNEL helps me, though it does look interesting |
| 07:44.17 | Rhomber | just to clarify, is the CHANNEL function relative to the current channel? or other channels active within asterisk too? |
| 07:45.38 | ectospasm | current channel |
| 07:45.57 | ectospasm | it executes within the context of the current channel. |
| 07:46.01 | Rhomber | okies, well you've given me a place to look :) |
| 07:46.12 | Rhomber | i think i can find the full channel name |
| 07:46.34 | Rhomber | probably contained within the PeerEntryEvent :) |
| 07:46.39 | Rhomber | just inspecting it now |
| 07:52.39 | Rhomber | wait, will a channel even exist before it's dialed or dials in? |
| 07:54.06 | ectospasm | there will be the half that reaches the Dial command |
| 07:54.11 | ectospasm | other than that there's no channel |
| 07:56.18 | Rhomber | so there's no way to see if a SIP channel is available without dialing it? |
| 07:56.21 | Rhomber | that's pretty lame |
| 07:56.40 | Rhomber | surely there is some means of SIP ping or something? |
| 07:56.57 | ectospasm | what are you trying to achieve? Asterisk will allocate another channel data structure as needed. |
| 07:57.36 | Rhomber | i already said.. i want to see if the SIP channel is available.. if it is.. use it.. otherwise connect the call using the trunk and a regular DID |
| 07:57.42 | Rhomber | pretty standard stuff really |
| 07:57.54 | Rhomber | perhaps my terminology is wrong |
| 07:57.58 | ectospasm | yes |
| 07:58.30 | ectospasm | if the trunk/whatever is registered, and qualify says it's OK, then you should be able to send a call, or receive a call |
| 07:58.48 | ectospasm | no need to test the waters further, you'll just go mad |
| 07:58.55 | kaldemar | there is no channel until there is a call. |
| 07:59.14 | *** join/#asterisk nix8n82 (~AndChat27@66.110.182.182) |
| 07:59.20 | Rhomber | so, you have a qualify on the SIP which happens every so often.. say 60 seconds.. a call about to go out.. so if the sip peer is registered.. but perhaps this is within the 60 seconds of qualify.. i want to ping the SIP peer to make sure.. |
| 07:59.32 | Rhomber | that's not the case |
| 07:59.51 | Rhomber | and it's kind of a silly thing to have not considered IMHO |
| 08:00.13 | Rhomber | if it's to save bandwidth, it makes no sense.. a simple PING .. ACK.. wouldn't be much overhead |
| 08:00.48 | Rhomber | so i guess i'm left with just making a Dial call and then having it failing, and then having to start over again after temporarily banning the interface in my system |
| 08:00.52 | Rhomber | which is pretty lame |
| 08:01.19 | kaldemar | you're thinking of SIP like it was a bundle of circuits. |
| 08:01.23 | Rhomber | it would be much better and logical to be able to test the interface before the dial command is invoked |
| 08:01.37 | Rhomber | no, im thinking that it's a protocol |
| 08:01.43 | Rhomber | that can send and receive |
| 08:01.46 | Rhomber | is it not? |
| 08:02.25 | Rhomber | fine sure, if you guys don't see a problem.. i really couldn't care. but it's pretty obvious to me :) |
| 08:02.28 | kaldemar | and you'll know whether it can when you dial. |
| 08:02.48 | Rhomber | sure, but that's not how i want to handle it |
| 08:02.49 | ectospasm | You can send an OPTIONS message, and you should get an appropriate response |
| 08:02.55 | Rhomber | though, im going to have to do that |
| 08:02.57 | kaldemar | that's the "test". and you'll define what happens after the dial if it were to fail. |
| 08:03.31 | Rhomber | hmm, interesting |
| 08:03.36 | Rhomber | i'll have a look at the options |
| 08:03.42 | kaldemar | if you know that a peer only handles for example a fixed amount of calls at a time, you can use device state or group functions. |
| 08:05.44 | Rhomber | that's not the problem really |
| 08:05.53 | Rhomber | it's that the SIP client will be running on the phone on 3G |
| 08:06.06 | Rhomber | and will for sure at times be unreliable |
| 08:06.18 | Rhomber | and also wanting to detect that before Dial occurs |
| 08:06.45 | Rhomber | but as discussed, i might just have to contend with using dial and going again with a fallback configuration |
| 08:09.16 | kaldemar | "yes" is not the only options for qualify though... there's also qualifyfreq and qualifygap. |
| 08:09.31 | Rhomber | yeah, i fine tuned those a bit |
| 08:09.55 | Rhomber | but ultimately I don't want to set the freq to like 2 seconds either.. that would burden multiple things |
| 08:10.26 | Rhomber | looks like i can't easily send an OPTIONS request from asterisk |
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| 08:13.32 | ectospasm | Rhomber: OPTIONS is usually how qualify works |
| 08:14.17 | ectospasm | ...the side receiving the OPTIONS message will usually send a 4xx message, when any message would do. |
| 08:14.42 | Rhomber | yeah i saw that |
| 08:14.52 | Rhomber | but can't see how to instigate it myself |
| 08:15.16 | Rhomber | a way to force qualify would be good |
| 08:15.34 | ectospasm | you'd have to construct your own SIP request. I can't say I know how to do that from within Asterisk. Not sure if an AGI script could do it |
| 08:16.20 | Rhomber | the java engine has direct access to the manager too |
| 08:16.29 | Rhomber | but i'd still have no idea either |
| 08:17.24 | kaldemar | http://pastebin.com/j6y5NBYE |
| 08:17.44 | kaldemar | feel free to abuse/make it work/whatever. |
| 08:21.40 | Rhomber | thanks :) |
| 08:22.17 | Rhomber | I just saw in asterisk-java SipNotifyAction .. that might be an easy alternative? |
| 08:22.44 | Rhomber | unless it's send and forget i guess |
| 08:23.14 | *** join/#asterisk markwaters (~markwater@unaffiliated/markwaters) |
| 08:24.31 | kaldemar | it just sends a notify message, you won't get a response to it. |
| 08:26.20 | Rhomber | hmm :( |
| 08:26.21 | Rhomber | thanks |
| 08:26.53 | markwaters | hi guys , does confbridge depend on dahdi ? |
| 08:27.32 | kaldemar | markwaters: no. |
| 08:27.57 | markwaters | its enabled in menuselect but the core show applications in the console doesnt show it and my dialplan fails when it hits it |
| 08:28.11 | Rhomber | actually, it seems to work :) |
| 08:28.19 | markwaters | kaldemar: thanks |
| 08:28.20 | Rhomber | message='Notify Sent'; response='Success'; (when the channel is up) |
| 08:28.45 | Rhomber | message='Could not create address'; response='Error'; (when the channel is down) |
| 08:29.02 | Rhomber | i'll test a few times and see |
| 08:29.16 | Rhomber | but that might be a relatively painless alternative |
| 08:33.31 | Rhomber | nah, it was just a fluke |
| 08:33.40 | Rhomber | the qualify timed out during the test case haha |
| 08:34.10 | Rhomber | i guess i'll just have to rely on the Dial state, seems safest.. but it would be nice if there was an alternate |
| 08:34.21 | Rhomber | thanks for the code though kaldemar :) |
| 08:39.30 | kaldemar | markwaters: what is the failure message you see in CLI when making a call? |
| 08:42.27 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 08:43.51 | markwaters | kaldemar: [2013-04-06 10:03:39] WARNING[9712][C-00000000]: pbx.c:4621 pbx_extension_helper: No application 'ConfBridge' for extension (conference, s, 4) |
| 08:44.22 | markwaters | i am rebuilding from source again with every conceivable depency installed |
| 08:45.49 | markwaters | lol , staring at the gcc screensaver |
| 08:45.55 | ectospasm | markwaters: what version of Asterisk? |
| 08:45.59 | kaldemar | check that it (app_confbridge.so) actually gets built and installed in /usr/lib/asterisk/modules. if it does and still no go, try to unload/load it manually with "module unload" and "module load" in CLI. |
| 08:46.31 | markwaters | ectospasm: its the latest 11.3 , downloaded from the asterisk.org site this morning |
| 08:46.39 | markwaters | kaldemar: ok , let me try that |
| 08:47.28 | Rhomber | actually i can't rely on Dial, as when using multiple interfaces (&'ed) and only one is unavailable, it doens't return CHANUNAVAIL |
| 08:48.51 | markwaters | kaldemar: ok , i am getting somewhere now |
| 08:49.17 | markwaters | i can see the module is failing to load because of problems in the confbridge.conf file |
| 08:49.21 | kaldemar | Rhomber: see app ChanIsAvail, maybe it can do you some good. |
| 08:49.22 | markwaters | kaldemar: thanks! |
| 09:06.47 | Rhomber | kaldemar: that's what i started asking questions about. |
| 09:07.18 | kaldemar | roger. didn't read the conversation from the beginning. |
| 09:07.39 | Rhomber | It always returns 0, and ectospasm said I was passing the wrong channel through.. I gave it SIP/david-mobile .. he said i needed SIP/david-mobile-3432f8b or something.. |
| 09:07.59 | Rhomber | and that 'active channel id' doesn't exist before a Dial to or from the SIP device |
| 09:08.51 | Rhomber | trying to get the Sip options code to work |
| 09:08.58 | Rhomber | not sure how it's meant to work with NAT :( |
| 09:12.45 | ectospasm | heh, SIP takes work to traverse NAT. Usually we let Asterisk do that |
| 09:12.48 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 09:12.52 | ectospasm | gotta mangle headers |
| 09:14.16 | Rhomber | damn :( |
| 09:14.33 | Rhomber | the SipPeerEntry has |
| 09:14.42 | Rhomber | IPaddress: 60.240.245.17 .. IPport: 1024 |
| 09:14.55 | Rhomber | but I guess that's only part of the solution |
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| 09:18.24 | Rhomber | sounds like it might be easier to write a c module to force qualify |
| 09:18.25 | Rhomber | lol |
| 09:31.56 | *** join/#asterisk shtoom (~shtoom@183.82.27.135) |
| 09:35.36 | Rhomber | looks like there is an undocumented option in Dial |
| 09:35.57 | Rhomber | b and B for predial, where the channel to be used for the Dial is constructed |
| 09:36.07 | Rhomber | ChanIsAvail may work there :) |
| 09:36.19 | *** join/#asterisk vlad_starkov (~vlad_star@109.95.84.116) |
| 09:36.23 | Rhomber | not ideal, but perhaps a work around |
| 09:37.41 | WIMPy | What are you trying to do? |
| 09:38.33 | Rhomber | https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers |
| 09:38.49 | Rhomber | detect if a SIP channel is available between qualifies |
| 09:39.12 | WIMPy | o.O |
| 09:39.14 | Rhomber | i.e. if the sip channel is down and hasn't timed out within a qualify period.. use the backup interface |
| 09:39.44 | Rhomber | ideally it would be nice if ChanIsAvail worked, but apparently that's only for established channels.. not SIP peer channels |
| 09:39.59 | Rhomber | or it would be nice if i could force a qualify.. which i can't |
| 09:40.07 | WIMPy | If it's not known yet, you can only try and see that it failed, then retry. |
| 09:40.34 | Rhomber | relying on Dial to return CHANUNAVIL isn't good enough as when i use multiple interfaces, it just returns CANCEL as one keeps ringing and times out and the other is not available |
| 09:40.44 | Rhomber | as i said, not good enough |
| 09:40.55 | WIMPy | What good would that be? You only delay the connection attempt. And what if it fails at exactely that time? |
| 09:41.30 | Rhomber | it would do a lot of good, because the qualify range is 1 minute.. that's a lot of time for failure |
| 09:41.41 | WIMPy | That is you dial a group of peers and want backups for single members of that group? |
| 09:41.43 | Rhomber | where by the 10-100ms period is significantly less |
| 09:42.16 | Rhomber | well the specific scenario is this |
| 09:42.58 | Rhomber | Dial could be for either SIP/david-snom&SIP/david-mobile ... or SIP/david-snom&SIP/trunk-outgoing/1323232 as the fallback for the mobile SIP being down to call the mobile via regular DID on the trunk |
| 09:43.17 | Rhomber | i've already implemented it by detecting the sip peer status |
| 09:43.26 | WIMPy | That's easy. |
| 09:43.29 | Rhomber | but it's unreliable as there is a 60second period of failure |
| 09:43.34 | WIMPy | Just use local channels. |
| 09:43.52 | Rhomber | it's already got to this stage with a local channel :) |
| 09:44.02 | Rhomber | are you suggesting i go another level deep? |
| 09:44.04 | Rhomber | LOL |
| 09:44.08 | WIMPy | yes |
| 09:44.47 | Rhomber | man im going to have to rewrite half the code to do it like that |
| 09:44.51 | Rhomber | *sigh* |
| 09:45.01 | Rhomber | why can't i just have asterisk ping a SIP peer? |
| 09:45.12 | WIMPy | Dial(sip/david-snom&local/david-mobile) then call sip/david-mobile and thereafter trunk/number in exten => david-mobile |
| 09:45.14 | Rhomber | seems more than lame that it can't.. it has the code.. why no interface/app? |
| 09:45.16 | Rhomber | can i write an app? |
| 09:45.36 | WIMPy | interface/app? |
| 09:46.12 | Rhomber | yeah that's not how the code works.. sure it's simple if you just use dialplans.. and have a huge convoluted extensions.conf.. but that's not how it's happening |
| 09:46.15 | Rhomber | as in.. |
| 09:46.24 | Rhomber | write an app to perform the SIP OPTIONS PING |
| 09:46.34 | Rhomber | or an app to force a qualify |
| 09:47.14 | WIMPy | No idea if that works, but you could try a dial with a 0s timeout. |
| 09:47.31 | Rhomber | i had thought of that |
| 09:47.37 | Rhomber | i guess it's worth a try |
| 09:48.01 | Rhomber | network lag may cause it to fail though? |
| 09:48.09 | WIMPy | You should really modify your configuration for the local thing. |
| 09:48.40 | Rhomber | the config i have is in java .. to make the asterisk stuff more elegant.. with quite a few Agi hooks |
| 09:48.55 | Rhomber | and java assembles.. members.. devices.. and interfaces |
| 09:49.10 | Rhomber | which are concepts i've defined :) |
| 09:49.17 | Rhomber | i..e my mobile is a device.. with 2 interfaces.. either SIP or trunk DID |
| 09:49.31 | Rhomber | so.. i can do what your suggesting.. but yeah, lots of rewriting |
| 09:49.36 | Rhomber | might not have a choice though |
| 09:50.00 | WIMPy | It's the only fail safe option anyway. |
| 09:51.28 | Rhomber | so, Local/9010@ctx from the queue.. then... Local/9011@otherctx with a Macro for member acknowledgement... and then 9011 => to deliver a real Dial |
| 09:51.36 | Rhomber | will the macro in the middle work still? |
| 09:51.52 | WIMPy | macros are deprecated. |
| 09:52.38 | Rhomber | but they are the only way to do a queue member acknowledgement? |
| 09:52.52 | Rhomber | everything else will result in the queue caller being dropped or bridged? |
| 09:53.56 | WIMPy | Not being a queue user, I'm not sure what that's about. |
| 09:54.07 | Rhomber | from my experience that's the case |
| 09:54.14 | Rhomber | i tried to use a Gosub, doesn't work |
| 09:54.52 | Rhomber | as for handing the member ack in the 2nd Local dial, that wouldn't work would it? |
| 09:55.03 | Rhomber | once the real Dial is performed, the queue call will be connected.. |
| 09:56.26 | *** join/#asterisk vlad_starkov (~vlad_star@109.95.84.116) |
| 10:03.57 | Rhomber | i think i can make it work |
| 10:04.16 | Rhomber | but will be interesting to see how it goes |
| 10:11.32 | *** join/#asterisk morfin (~morfin@morfin.telenet.ru) |
| 10:11.33 | morfin | hello |
| 10:11.52 | morfin | can anyone tell me can asterisk use UNIX socket for AMI? |
| 10:12.06 | WIMPy | no |
| 10:12.23 | morfin | good for me |
| 10:12.30 | morfin | and bad |
| 10:12.54 | WIMPy | The remote console uses a unix socket. |
| 10:13.25 | morfin | remote console? |
| 10:13.29 | morfin | ow |
| 10:13.32 | WIMPy | asterisk -r |
| 10:13.35 | morfin | i know |
| 10:14.35 | WIMPy | So it's only for extremely limited amounts of "remote". |
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| 12:33.34 | pzn | I need to make a report of incoming unanswered calls of a given period of time... any hints? which log files can I look? |
| 12:34.05 | kaldemar | pzn: cdr |
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| 12:36.15 | BeeBuu | hello,all. |
| 12:36.45 | BeeBuu | is there anyone know about h.248? is it supported by asterisk? |
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| 12:39.36 | pzn | kaldemar, thanks, problem solved :-) |
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| 12:49.19 | BeeBuu | hi,blitzrage |
| 12:49.29 | blitzrage | hi |
| 12:49.51 | BeeBuu | did you know asterisk support H.248 or not? |
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| 12:50.48 | BeeBuu | leifmadsen? Is asterisk support H.248 or not? |
| 12:51.19 | leifmadsen | depends what you mean by "support". I believe it can pass it through between devices that handle it. I just use H.264 |
| 12:51.37 | leifmadsen | Asterisk can't {en,de}code video codecs though |
| 12:52.03 | kaldemar | h.248 is mgcp |
| 12:52.10 | leifmadsen | oooooh |
| 12:52.17 | leifmadsen | it's early... and I have no had coffee lolz |
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| 12:52.31 | leifmadsen | wtf was I thinking then.... |
| 12:52.41 | leifmadsen | ok, gotta go |
| 12:52.43 | leifmadsen | runs away |
| 12:52.52 | leifmadsen | BeeBuu: sorry, I'm sure my answer was incredibly confusing, and not applicable :) |
| 12:53.08 | leifmadsen | Asterisk does support MGCP, sort of. Not well supported I don't think, but chan_mgcp does exist. |
| 12:53.56 | BeeBuu | oh,thanks. is it same as megaco? |
| 12:55.24 | kaldemar | yes |
| 12:59.07 | BeeBuu | kaldemar: thanks.is there any docs about how to config mgcp support h.248? |
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| 15:43.11 | Rhomber | WIMPy: Finished implementing the Local -> Local solution, works a treat.. :) |
| 15:45.03 | Rhomber | (^ Thanks) |
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| 15:51.57 | slav3_kitten | fucking awesome. i'm getting congestion when i try to dial out |
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| 15:59.38 | slav3_kitten | oh hey... |
| 16:01.06 | slav3_kitten | my static IP i pay 15 dollars a month to have... changed |
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| 16:25.38 | carrar | hahha |
| 16:25.48 | carrar | slav3_kitten, What provider? |
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| 18:07.00 | slav3_kitten | carrar, a rural wisp that's changed it's name twice |
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| 18:30.02 | carrar | nicr |
| 18:30.03 | carrar | nice |
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| 19:19.16 | *** join/#asterisk cusco (~tralala@2001:41d0:1:6caf::1) |
| 19:19.19 | cusco | hi |
| 19:19.40 | cusco | while seting a var using Set() how can I set it to a string with new line feeds ? |
| 19:19.47 | cusco | \n goes unescaped |
| 19:23.41 | navaismo | dont know the answer but wondering why you need multiline in a VAR |
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| 19:34.46 | cusco | navaismo: because I'm sending a jabber message and I wouldlike to multi-line it |
| 19:37.39 | navaismo | i see |
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| 19:38.39 | ccherrett | is there any opensource asterisk web based switchboard programs? |
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| 19:42.03 | cusco | fop |
| 19:45.55 | ccherrett | thank you |
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| 20:00.37 | cusco | in function CUT, can the delimiter be a " |
| 20:00.39 | cusco | ? |
| 20:01.13 | cusco | errors out: missing argument |
| 20:01.17 | cusco | dialplan has: Set(smsDATE=${CUT(CMGR,",6)}); |
| 20:02.26 | cusco | ok escaped quote \" works |
| 20:04.07 | [TK]D-Fender | FOP1 was OSS and is long-since dead |
| 20:05.25 | [TK]D-Fender | FOP2 is closed-source |
| 20:07.04 | ChannelZ | and it sounds too much like 'fap' |
| 20:07.12 | navaismo | cusco, yeah can't add a \n in the dialplan using jabbersend |
| 20:07.41 | navaismo | so far using an AGI to set the message and add \r does the job but overkill |
| 20:13.07 | cusco | okok no worries, thanks |
| 20:28.20 | cusco | whatmodule provides function LEN() ? |
| 20:30.14 | igcewieling | navaismo: shells use ctrl-V to escape the next character, maybe you can do that with a variable? I doubt it, but can't hurt to try |
| 20:30.27 | [TK]D-Fender | cusco: func_strings.c |
| 20:30.40 | cusco | thank you [TK]D-Fender |
| 20:30.48 | igcewieling | navaismo: are you sure you just need a \n and not a \r\n |
| 20:32.08 | cusco | that was me, yes I was sure, and I need it no longer |
| 20:32.49 | cusco | less overwkill would be using ${SHELL echo -e ......} still overkill.. |
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| 21:28.37 | ChannelZ | nothing serious but weird; why does 'make menuconfig' on one machine have a blue background (libnewt version) but on my other machine it's purple? |
| 21:41.56 | slav3_kitten | because purple is awesome? |
| 21:42.29 | brabo | 42 |
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| 21:48.27 | ChannelZ | and I can't remember, does 'make config' install just the init scripts or it stomps on the /etc/asterisk configs as well? |
| 21:48.40 | ChannelZ | oh.. no that's 'make samples' right |
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| 23:55.57 | igcewieling | make samples overwrites the configs. make config setup the init scripts |
| 23:56.11 | igcewieling | totally logical, I can't see how you didn't understand that! 8-) |