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| 00:34.48 | tompaw | Hello. |
| 00:35.10 | tompaw | Say, what would be the reason for asterisk to completely ignore my custom astspooldir? |
| 00:37.02 | WIMPy | Do you see it with 'core show settings'? |
| 00:37.45 | [TK]D-Fender | #1 (and just about only) reason : asterisk.conf is wrong |
| 00:37.48 | tompaw | Yes, but it's showing the default location, not the one I set in asterisk.conf |
| 00:37.52 | tompaw | Parsing error? |
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| 00:38.14 | [TK]D-Fender | pretty much is sure he knows already.... |
| 00:38.17 | [TK]D-Fender | so show us |
| 00:38.20 | [TK]D-Fender | PB it |
| 00:39.34 | tompaw | [TK]D-Fender: http://hastebin.com/alaguyizey.coffee |
| 00:39.37 | tompaw | nothing special there |
| 00:39.52 | [TK]D-Fender | (!) = ignorme my custom settings below |
| 00:40.02 | [TK]D-Fender | REMOVE that |
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| 00:41.10 | tompaw | ta, it works. |
| 00:41.20 | tompaw | dare I ask why |
| 00:41.31 | [TK]D-Fender | because that's how they made it work. |
| 00:41.54 | file | that's how templates work, as to why the file is like that - it's because arguments to ./configure actually control the defaults |
| 00:42.17 | file | and noone has made anything to change the asterisk.conf that is copied in as a result of the samples |
| 00:42.41 | tompaw | file: yeah I meant why is it a template by default, what is the point of this. |
| 00:42.49 | tompaw | also, this: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Main+Configuration+File |
| 00:42.52 | file | see above |
| 00:42.52 | tompaw | that's misleading as hell. |
| 00:43.36 | file | how can we make it less misleading? explicitly call out the removal of (!) to make it effective? |
| 00:43.37 | tompaw | since it's mentioned like that everywhere I assumed it's some kind of a magic-low-level-definitely-not-your-usual-template-symbol ! |
| 00:43.40 | [TK]D-Fender | I fail to see a point for having templates in there at all. |
| 00:44.01 | [TK]D-Fender | it's not like sip.conf where you can create shared prfile bits between different sections. |
| 00:44.02 | WIMPy | So do I |
| 00:44.05 | file | [TK]D-Fender, the sample file will break installs if you change the ./configure arguments for where Asterisk is installed |
| 00:44.12 | [TK]D-Fender | that should basically all be the equivalent of [general] |
| 00:44.28 | file | (if it's not a template) |
| 00:44.45 | WIMPy | It would be rather obvious if the lines were just commented out like in the other files instead of diabling the whole block in that somehow obcure way. |
| 00:44.55 | file | yeah, that's an option |
| 00:44.57 | [TK]D-Fender | yup |
| 00:45.00 | tompaw | defo |
| 00:45.02 | [TK]D-Fender | hell fo a lot more sane |
| 00:46.13 | tompaw | Since I got you all smart people here, let me ask you this: why is it impossible to record a channel that belongs to a bridge? |
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| 00:46.23 | file | from? |
| 00:46.30 | tompaw | ARI |
| 00:46.53 | file | because Record isn't "monitor a channel and record to a file" |
| 00:47.01 | file | it's the equivalent of calling Record() in the dialplan |
| 00:47.01 | tompaw | tried it and got a nice error message that said something like "Can't record this channel because it's participating in a bridge" |
| 00:47.51 | file | recording a channel while in a bridge would be done by creating a Snoop channel and calling record on it |
| 00:48.02 | file | which is like doing MixMonitor |
| 00:48.06 | tompaw | So if you have a mixing bridge with multiple channels joining/leaving - is it possible to have separate recordings of that conference from each one's perspective? |
| 00:48.40 | file | yes, by doing the above |
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| 00:50.09 | tompaw | so I snoopChannelWithId on the channel that joins my conf, this will return a "snoop" channel and I record on that? |
| 00:50.19 | file | yes |
| 00:51.33 | file | anything you can do on a normal channel you can do on a Snoop channel, and depending on the properties you created it with - it'll give you a copy of the audio from the channel/to the channel, or whisper into the streams |
| 00:52.20 | tompaw | I want an equivalent of recording a phone line of someone who talks in a conf room, so listening to audio both ways but without interfering. |
| 00:52.48 | file | then configure the Snoop channel that way, and you're good to go |
| 00:52.58 | tompaw | will do now and see what happens |
| 00:53.26 | tompaw | damn, my ari app is growing and growing, I'm not really sure if I'm following the right design path here. |
| 00:53.47 | file | you can find other users in #asterisk-ari and on the asterisk-app-dev mailing list |
| 00:54.16 | WIMPy | Oh, a new cahennel. |
| 00:54.38 | file | it was created by those who use it |
| 00:54.45 | WIMPy | Sounds like a list of all channels might become usefull. |
| 00:57.28 | tompaw | I wonder if there is anyone hat can provide ARI consultation. I am using it in production, since it fits our project OH SO PERFECTLY, but with the lack of guidance online due to new shiny technology, I'm not sure if I'm doing it right. |
| 00:57.44 | tompaw | I mean, it works, but. |
| 00:58.48 | WIMPy | ... and mailing lists. |
| 00:59.39 | file | http://lists.digium.com/mailman/listinfo/ has the list |
| 00:59.58 | tompaw | with snooping channel, should I set it up before or after adding the original channel to a bridge? |
| 01:00.03 | tompaw | (does it matter at all?) |
| 01:00.20 | file | depends on if you want to miss audio potentially |
| 01:00.33 | tompaw | aerosmith, man. |
| 01:01.52 | tompaw | file: so if I don't want to miss anything, then set it up before joining, yes? |
| 01:01.57 | file | yup |
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| 01:02.08 | tompaw | ok, let's see what crashes |
| 01:02.31 | tompaw | will the snooping channel die automatically when I hang up the original one? or do I need to handle it manually? |
| 01:02.42 | file | they will hang up |
| 01:03.05 | file | their life cycle follows that of the snooped channel, although you can explicitly hang 'em up |
| 01:04.53 | tompaw | argh, why do I have to place it in an app rather than simply handle in the original app |
| 01:05.38 | file | I don't understand the statement. An HTTP request to ARI has no indication of what application is invoking it, it has to be told. |
| 01:06.44 | tompaw | file: I mean why do I have to provide an app to snoopChannel, rather than simply grab a reference to it - that was my point |
| 01:06.50 | tompaw | Doesn't matter, I can do it that way. |
| 01:06.56 | file | it has to go somewhere |
| 01:07.09 | tompaw | hm... valid point |
| 01:07.20 | file | and unless it's in your app, you have no guaranteed ownership |
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| 01:38.19 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.4.0 (2015/06/04), 11.18.0 (2015/06/04), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 02:14.47 | *** join/#asterisk ruben23 (~OpenDIAL@203.111.224.65) |
| 02:16.19 | ruben23 | hi guys any help with my asterisk setup for Dahdi channel...when i dial on it the volume is pretty low i cant barely hear any suggestion to increase volume somehow..?? |
| 02:17.56 | WIMPy | If it's POTS check the config fiel for gain options. |
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| 03:39.18 | ruben23 | WIMPy: it ISDN line somehow.. |
| 03:45.21 | ruben23 | hi guys any help with my asterisk setup for Dahdi channel...when i dial on it the volume is pretty low i cant barely hear any suggestion to increase volume somehow..?? ISD line |
| 03:45.24 | ruben23 | ISDN* |
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| 04:06.31 | jmordica | how can PBXWare guarantee this? http://www.bicomsystems.com/pbxware-3-8/t38 |
| 04:10.53 | jeev | probably says t38 is guaranteed? as in t38? |
| 04:12.38 | jmordica | But why does it mention reliable sip trunks if it's just referring to t38? |
| 04:26.17 | [TK]D-Fender | ruben23, So increase the gain on it |
| 04:26.25 | [TK]D-Fender | it's in the sample config..... |
| 04:31.48 | ruben23 | [TK]D-Fender: the normal asterisk config for increasing gain..? |
| 04:31.58 | [TK]D-Fender | DAHDI |
| 04:32.20 | ruben23 | ok got it..thanks a lot |
| 04:34.58 | jmordica | Anyone heard of https://www.4psa.com? |
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| 04:52.36 | janicez | jmordica: no |
| 04:53.45 | jmordica | Distributed asterisk based system. Seems new but they are on version 3.7 so seems the project has been going for sometime. |
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| 08:00.20 | WIMPy | Don't mess with gain on digital interfaces. that will only create distortions. Fix the cause. (i.e. the phone/ATA) |
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| 10:52.01 | babak | Hi, is there a way to save some information with external application like a web server in a temporary variable in Asterisk and use that variable in dialplan and then again re use it ? |
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| 12:12.55 | WIMPy | babak: Yes, you can set variables. Either via the remote shell or via AMI. |
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| 12:30.00 | Ice_Strike | How does dialer test harness system work? |
| 12:30.54 | Ice_Strike | Like if you want to test 500 concurrents call somehow and softphone need to answers it somehow. |
| 12:31.37 | [TK]D-Fender | what "dialer test harness system"? |
| 12:33.49 | Ice_Strike | Well, just a thought the system has two parts consisting of an Agent and a Customer component. The agent component virtually replicates actions agents will do in call handling e.g wrap times. Something like that. |
| 12:34.14 | Ice_Strike | In the way to test the Predictive dialer system |
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| 12:43.41 | [TK]D-Fender | SIPP |
| 12:46.58 | Ice_Strike | Hmm |
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| 13:45.55 | catphish | when using IP based peers, is there any protection at all against 3rd parties initiating a call by spoofing the IP of the peer? |
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| 13:48.30 | [TK]D-Fender | peer = match by IP. That doesn't mean they don't still have to auth |
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| 13:49.38 | catphish | [TK]D-Fender: sure, i mean in a situation where a secret isn't specified |
| 13:50.53 | catphish | i've noticed that some providers use only IP address for authentication, i wondered if there was some kind of 2-way handshake in SIP to do an INVITE, or whether it is totally open to abuse |
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| 13:57.20 | boch | hello, do you know how to log/cdr calls picked up with *8 feature code? |
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| 14:00.17 | babak | WIMPy: thx |
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| 15:30.26 | boch | hello, do you know how to log/cdr calls picked up with *8 feature code? |
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| 15:41.57 | seik0 | boch: feature codes may be customized, no? Why don't you call feature name? |
| 15:42.58 | seik0 | you mean group pickup? |
| 15:43.14 | seik0 | what have you already try log? |
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| 15:48.44 | boch | seik0, you are right, its call pick up feature. ive searched mysql and csv logs but nothing about the pick up |
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| 15:49.42 | seik0 | mysql and csv has no differences for cdr |
| 15:49.54 | seik0 | maybe CEL logging can help |
| 15:50.03 | seik0 | but I am not sure |
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| 15:50.51 | seik0 | and what is the problem with CDR? you see nothing while pickup? or you see wrong data? |
| 15:50.54 | F-G0z | Cuzner: Hi! Last week, you asked me to let you know whether I managed to have two applications (morse and confbridge) on the same time. I've solved this by originating calls with AMI to a macro context. See: http://pastebin.com/Z1z4CWj8 |
| 15:55.32 | Cuzner | F-G0z: neat, thanks. |
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| 15:57.46 | TheSin | hey guys I'm running askterisk form debian testing, and it recently updated to 13.1.0 since then my voicemail app has gone wonky. If I forward a msg by pressing 8 it drops the call due do a missing audio file vm-msgforwarded which I can not find anyplace, Also the notification lights on the phones are very very slow to turn on and off like hours at times |
| 15:58.00 | TheSin | any suggestions? has any of this been brought up? |
| 15:58.54 | [TK]D-Fender | We're at 13.4.0 now, and 13.5.0 pending shortly |
| 15:59.06 | [TK]D-Fender | If any of this is a legit issue it may already have been fixed |
| 15:59.23 | [TK]D-Fender | if you're missing a sound file then go download it direct off of the site |
| 15:59.26 | TheSin | hmm, so talk tot he debian maintainer then? |
| 15:59.43 | [TK]D-Fender | always something to do. |
| 16:02.19 | TheSin | thanks that helepd I just checked the changelog and I see when that happened |
| 16:02.30 | TheSin | appreciate the help |
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| 16:08.51 | TheSin | I see debian only has core-sources 1.4.22 and the vm change needs 1.4.25 I'll report it upstream |
| 16:10.20 | F-G0z | Do some of you happen to know whether it's possible to reduce the volume of one speaker but just for a specific listener? I use confbridge and iax2. :) |
| 16:19.54 | [TK]D-Fender | TheSin: No. |
| 16:19.57 | [TK]D-Fender | 1.4 is LONG dead |
| 16:20.07 | [TK]D-Fender | forget anything other than 11/13 |
| 16:20.23 | TheSin | I dont' mean asterisk I mean core-sounds |
| 16:20.43 | TheSin | asterisk 13.x requires core-sounds 1.4.25+ |
| 16:20.55 | [TK]D-Fender | ok, eyes crossed for a set there |
| 16:20.59 | TheSin | hehe |
| 16:21.00 | [TK]D-Fender | sec |
| 16:21.25 | TheSin | yeah I'm long long done with 1.4 I've already gone through the pains of 1.4 -> 1.6 -> 1.8 -> 1.11 upgrades :) I'm not going back there again ;0 |
| 16:21.45 | mjordan | F-G0z: The VOLUME function may do what you'd like. Although there's currently a bug in it that is getting fixed in 13.5.0-rc1 (11 is fine, sorry about that) |
| 16:26.29 | F-G0z | mjordan: So if I use this function to mute a specific speaker to a specific listener, only the others listeners will hear him. :-) |
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| 18:37.22 | kfife | The https://wiki.asterisk.org/ wiki is a pretty nice layout. Is that a mediaWiki skin/template or some other code base? |
| 18:37.39 | kfife | YAWS (Yet another Wiki Server) |
| 18:39.08 | malcolmd | confluence from atlassian |
| 18:40.03 | kfife | Interesting. |
| 18:40.16 | malcolmd | the web folk here did some work on colors; i'm not sure how much theming they actually did |
| 18:40.51 | kfife | Looks like it's a Cloud/SAS offering. is it open source like MediaWiki? |
| 18:41.08 | malcolmd | i don't believe that confluence is |
| 18:43.47 | kfife | Sounds like Digium uses it for all kinds of stuff, not just documenting Asterisk |
| 18:43.59 | kfife | (at least it DOES all kinds of stuff) |
| 18:44.14 | kfife | Presumably D chose it for more than a doc repo |
| 18:44.49 | kfife | malcolmd : Were you involved in the decisionmaking on that? |
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| 19:12.52 | malcolmd | we were very fond of the old wiki editor; we're less fond of the new one, but it's better than most. it doesn't support markdown, sadly. it has nice integration with the rest of the atlassian products, of which, for asterisk, we make extensive use |
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| 21:01.15 | moe` | atlassian ... seriously? |
| 21:02.11 | file | yes? |
| 21:02.53 | moe` | please tell its not jira |
| 21:03.07 | file | that's what Asterisk uses for an issue tracker, yes |
| 21:03.13 | moe` | wow |
| 21:03.21 | moe` | you poor bastards :) |
| 21:03.35 | moe` | IMHO jira is a steaming turd |
| 21:04.24 | moe` | why is an open source project using that as an issue tracker? |
| 21:04.35 | moe` | there must be some history I'm not aware of |
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| 21:08.24 | binarytempl3 | Anyone know why on-hold music would fail over SIP? Works perfectly well over DAHDI. What should I investigate? |
| 21:10.27 | moe` | I'ma gonna throw a stupid comment because I have no idea how to fix it binarytempl3 : don't use on-hold music :) |
| 21:10.41 | moe` | grins stupidly |
| 21:11.51 | moe` | I probably deserved to be kicked for that, really. |
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| 21:24.26 | malcolmd | i'll bite. what's moe`s issue tracking software package of choice? |
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| 21:28.29 | binarytempl3 | yeah, I just set up an extension that rickrolls |
| 21:28.45 | binarytempl3 | I don't actually have an on hold music setup |
| 21:29.14 | moe` | malcolmd: well, I dunno. Just had bad experiences with Jira. |
| 21:29.16 | binarytempl3 | but from a technical standpoint I'd like to understand why it works over PSTN (DAHDI) but not over SIPO |
| 21:29.26 | binarytempl3 | s |
| 21:29.37 | binarytempl3 | s|SIPO|SIP| |
| 21:30.29 | binarytempl3 | I mean, it's ulaw and all that, and voice works fine from DAHDI to SIP and to conference calls etc |
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| 22:13.52 | moe` | hey, here's a question... what would be a recommeded SIP client for android? |
| 22:14.06 | moe` | free or paid, doesn't matter |
| 22:14.12 | moe` | just one that works well |
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| 22:28.14 | newtonr | moe`, CSIPSimple has worked well for me |
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| 23:01.05 | moe` | hey for shits and giggles, if I enable tls goodies in the [general] section of sip.conf that applies to all subsections unless I override per section correct? |
| 23:01.40 | moe` | I have a proper globalsign cert |
| 23:02.51 | moe` | newtonr : will try that now, thanks |
| 23:04.36 | moe` | will asterisk tolerate multiple SIP clients with the same SIP profile (user/pass)? |
| 23:04.44 | moe` | I guess I'll find out here in a moment |
| 23:05.21 | [TK]D-Fender | moe`, chan_sip = no |
| 23:06.25 | moe` | [TK]D-Fender, the yoda of asterisk, what's that now? |
| 23:07.20 | [TK]D-Fender | chan_sip cannot support that |
| 23:07.27 | moe` | newtonr : I want a SIP client to connect to my own asterisk box, CSIPSimple allows that? |
| 23:07.38 | [TK]D-Fender | every time another device registers it will STEAL the registration from the others |
| 23:07.45 | file | THIEF |
| 23:07.56 | moe` | [TK]D-Fender - what are we talking about here? TLS on asterisk? |
| 23:07.59 | moe` | I lost focus |
| 23:08.08 | [TK]D-Fender | moe`> will asterisk tolerate multiple SIP clients with the same SIP profile (user/pass)? |
| 23:08.30 | moe` | [TK]D-Fender: ah, ok, so one client per SIP profile |
| 23:09.27 | moe` | ok, so [TK]D-Fender .... if I enable tls goodies in the [general] section of sip.conf that applies to all subsections unless I override per section correct? |
| 23:09.50 | [TK]D-Fender | parameters that apply in BOTH will be inherited |
| 23:10.16 | moe` | was only looking for inheritance, other sections have no TLS settings |
| 23:10.33 | moe` | so if I set [general] tls stuff, it should fall down to others |
| 23:10.34 | moe` | ? |
| 23:10.39 | [TK]D-Fender | yes |
| 23:10.43 | moe` | thank you sir |
| 23:10.53 | [TK]D-Fender | that's what I said |
| 23:11.21 | moe` | yeah well forgive my stupidity |
| 23:11.27 | moe` | I've had a few beers |
| 23:14.47 | moe` | any other comments on a SIP client for android? |
| 23:15.12 | moe` | I use Bria on stupid 'doze, it works well but its payware |
| 23:15.24 | moe` | Ekiga on 'BSD |
| 23:15.55 | moe` | I tried Ekiga on 'doze but it annoyed me, so I went payware |
| 23:16.10 | moe` | what's the go for SIP client on android... anyone? |
| 23:18.15 | moe` | I know there are many but almost just as many pains in the ass |
| 23:20.12 | [TK]D-Fender | <newtonr> moe`, CSIPSimple has worked well for me |
| 23:20.21 | [TK]D-Fender | moe`> newtonr : will try that now, thanks |
| 23:20.25 | [TK]D-Fender | Lost focus again? |
| 23:20.36 | [TK]D-Fender | You've already gotten an answer and you said you'd try it... |
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| 23:24.31 | *** join/#asterisk oli-work (~oli-work@230.129.233.220.static.exetel.com.au) |
| 23:24.49 | moe` | I did |
| 23:24.55 | oli-work | hi folks, having a problem that i'm not making any headway with |
| 23:24.57 | moe` | it appeared to want 3rd party SIP servers |
| 23:25.16 | [TK]D-Fender | 3rd from who? |
| 23:25.26 | oli-work | internal > internal > pickup inbound call > return to internal call > no audio or moh |
| 23:25.27 | moe` | various providers |
| 23:26.00 | moe` | ok will install CSIPsimple again, and look closer |
| 23:26.30 | oli-work | two users call each other internally. one of the parties takes an inbound call, which puts the internal party on hold. seemingly randomly, the internal party gets no MoH and no audio when the calling party picks the call back up |
| 23:26.40 | oli-work | http://pastebin.com/hafYvPHB here is a sip debug |
| 23:26.55 | oli-work | voipmonitor also notes that the RTP stream ended before the last SIP packets |
| 23:27.06 | moe` | dude, CSIPsimple appears to want you to use, for example, US GatherCall Mondotalk, OnSIP, ippi, etc |
| 23:27.11 | oli-work | anyone able to please give some guidance? |
| 23:27.20 | moe` | I do not see an immediate way to add my own voip server |
| 23:28.07 | moe` | oh, wait. duh. |
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| 23:30.56 | moe` | bullshit, it shows "registered" but did not ask me for server, etc. it's not hitting my own asterisk instance, but some 3rd party |
| 23:31.24 | moe` | uninstall |
| 23:31.47 | moe` | glad I did that on my test android/S2 |
| 23:32.09 | moe` | sip show peers shows that profile was not connected |
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