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| 03:35.16 | phix | ok I am upgrading from 1.4 to 1.6 now |
| 03:35.21 | phix | Wish me luck :D |
| 03:39.43 | ChannelZ | Don't need luck, just 'make'. |
| 03:39.57 | WIMPy | make luck? |
| 03:40.08 | ChannelZ | but if you're going to bother at all, why not 1.8? |
| 03:42.25 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 03:42.37 | phix | umm because 1.8 is not in squeeze repo |
| 03:42.46 | phix | or is it |
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| 03:43.18 | shmaltz | hi everyone |
| 03:43.32 | phix | hi pal! |
| 03:43.48 | shmaltz | what up? |
| 03:44.17 | phix | Updating Debian from Lenny to Squeeze |
| 03:44.22 | phix | so from asterisk 1.4 to 1.6 |
| 03:45.58 | phix | Cryptoloop has been dropped from kernel :( |
| 03:46.22 | p3nguin | phix: Who cares if it's in the repo or not? Get the source, run ./configure --prefix=/usr --sysconfdir=/etc , then make, then checkinstall -D. Now you'll have 1.8 and it'll be registered in your package manager. |
| 03:49.08 | phix | It support debian? |
| 03:51.08 | p3nguin | Your question does not parse for me. |
| 03:51.49 | p3nguin | If you're trying to ask if checkinstall makes debs, yes it does... the -D makes a deb and registers it using dpkg. |
| 03:51.52 | phix | Asterisk has support for a particular Debian distro? |
| 03:52.01 | phix | linix distro even |
| 03:52.40 | p3nguin | No, that's not how Asterisk works. It's simply software. |
| 03:53.32 | phix | that is what I thought |
| 03:53.37 | p3nguin | If you never compiled your own software before, you might be required to install a few things before you can build Asterisk from source. |
| 03:53.38 | phix | so what is checkinstall -D then? |
| 03:54.26 | p3nguin | checkinstall is an app that builds a package out of a source build. The -D option, as I already said, makes a .deb packge and registers it using dpkg. |
| 03:54.48 | phix | ah ok |
| 03:54.55 | phix | I did not know that |
| 03:54.56 | p3nguin | checkinstall also makes rpms and probably a few other package formats, but since you're using Debian, you only need to worry about using -D. |
| 03:55.21 | phix | is that part of LSB or something? |
| 03:55.33 | p3nguin | not that I know of |
| 03:56.02 | phix | ok |
| 03:56.25 | p3nguin | You seem to understand the importance of using your package manager on your Debian system, so I'm offering you a safe way to install Asterisk from source and still use the package manager. |
| 03:56.35 | phix | yes it is important |
| 03:57.00 | p3nguin | If you don't like it or want to remove it later, you can use dpkg to remove the asterisk package that you're creating with checkinstall. |
| 03:57.37 | phix | nice |
| 03:57.55 | phix | I will read up on that |
| 03:58.13 | p3nguin | I rarely ever recommend to someone to just go install something from source blindly. Always use the package manager. But sometimes the software or specific version of the software is not available from the repos and you can't get it in package form from the distributor. |
| 03:58.32 | p3nguin | checkinstall takes care of that problem by making your source build into a package. |
| 03:59.21 | p3nguin | There's really nothing to read... Instead of running "make install" after the configure and make, you just use "checkinstall -D" which is the same as "checkinstall -D make install". |
| 03:59.46 | phix | so it guesses which files re being instlaled |
| 03:59.56 | p3nguin | No. |
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| 04:00.29 | phix | so then I should read up on it then :) |
| 04:00.52 | *** part/#asterisk Eldonn (~Eldonn@S0106e0cb4e3ba016.cg.shawcable.net) |
| 04:01.00 | phix | I don't like thinking there is voodoo happening |
| 04:01.02 | p3nguin | The Makefile determines what files are installed. If you ran "make install", it would throw the files all over the place where they belong, right? checkinstall installs the files exactly the same way, but by package so you can keep it in your package manager. |
| 04:01.11 | p3nguin | It's not voodoo. |
| 04:01.26 | p3nguin | Have you never installed anything from source, ever, on any system? |
| 04:01.36 | phix | IF the makefile calls scripts then how does itt know? |
| 04:01.56 | p3nguin | The Makefile doesn't call scripts. |
| 04:02.13 | phix | A Makefile can though |
| 04:02.56 | p3nguin | checkinstall simply makes a package out of the files that are installed using make install. That's all; no magic involved. |
| 04:04.46 | phix | so the Makefile has to be in a particular formta? |
| 04:05.16 | phix | any way I will just read up on it :) |
| 04:08.09 | phix | I do know zaptel has been uninstalled |
| 04:08.10 | p3nguin | I've never heard of anyone saying, "I need to run make and then make install... so I need to be sure the Makefile is in a particular format." It just doesn't happen. |
| 04:08.34 | p3nguin | Zap stuff has been deprecated in favor of Dahdi. |
| 04:08.57 | phix | p3nguin: A Makefile can do what ever it wants in it, it can call other Makefile's, run scripts which don't nessecary specifiy all output files |
| 04:09.24 | p3nguin | And running make install does all that stuff automatically. |
| 04:09.41 | phix | so the only way I see checkinstall working is chrooting it somewhere, then packing it |
| 04:09.49 | p3nguin | checkinstall just spies on the underlying make install, recording where all the files go. |
| 04:10.07 | phix | ok |
| 04:11.04 | p3nguin | It doesn't care what actually happens, it just records the files and where they go. That way you can easily remove them with a single dpkg command or from a graphical front-end such as synaptics. |
| 04:11.14 | phix | yay |
| 04:11.33 | phix | although what happens if it owns a conflicting file with another package? |
| 04:12.13 | p3nguin | Not very many Makefiles actually support make deinstall or make uninstall. If Asterisk does, that's another topic. checkinstall was created because of the lack of uninstall targets in typical Makefiles. |
| 04:12.45 | p3nguin | If there is a file conflict, it'll be shown just like every other package shows a conflict, and the package will be created but not installed. |
| 04:12.58 | p3nguin | That's the purpose of a package manager. |
| 04:14.17 | p3nguin | Then, just like every other .deb package, you can use dpkg -c asterisk-whatever.i686.deb to check the contents of it. |
| 04:19.12 | phix | <3 |
| 04:20.44 | phix | Voicetronix? wtf? |
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| 05:40.06 | *** join/#asterisk Jouva (Jouva@fluffy.moufette.com) |
| 05:44.26 | Jouva | Hey folks. I'm still a bit new to Asterisk and SIP and such. I know enough to atleast connect a softphone and connect Asterisk to a service such as Gizmo5 and setup extensions, but not much else. A small company I work for recently stopped using a VoIP provider due to a billing issue and we want to use the phones on our own and roll our own solution. But I've got a few questions |
| 05:45.36 | Jouva | First I know the phone supports transfer. And I see Asterisk does as well through keypad entry and that can be changed. But do these phones usually generate DTMF tones that can be heard by the other party? Or are they often sent through other means (SIP INFO I believe is what one of these methods is called)? |
| 05:45.52 | Jouva | Cause there's a built in transfer button |
| 05:47.14 | *** join/#asterisk smash- (smash@218.sub-75-196-9.myvzw.com) |
| 05:47.42 | smash- | Hey, im asking if asterisk kernel can handle two nic or more for multiple diff subnets? |
| 05:47.53 | smash- | or do i need to make a static route |
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| 05:49.36 | Kobaz | anyone know what up.screenCapture.enabled does on polycom? |
| 05:49.54 | Kobaz | smash-: i wouldn't see why not |
| 05:50.16 | Kobaz | smash-: i routinely have two nics on two different subnets... works the same as any other os |
| 05:50.16 | IsUp | Jouva: they are not generating DTMF on transfer, i think |
| 05:50.44 | Jouva | IsUp: So it's generally a SIP request with the proper keypad entries? |
| 05:51.40 | IsUp | yes, as far i know |
| 05:51.50 | IsUp | you can enable SIP debugging and see whats going on background |
| 05:54.50 | p3nguin | Asterisk does support DTMF transfer as a feature, but real transfers are preferred if they are available. |
| 05:55.29 | p3nguin | When I say feature, I mean check features.conf. |
| 05:55.48 | Jouva | Right I saw features.conf (which is why I mentioned the DTMF stuff) |
| 05:55.54 | Jouva | Is a "real transfer" part of how SIP works? |
| 05:56.15 | Jouva | Like are there actual transfer requests as part of the protocol? |
| 05:57.18 | p3nguin | yes |
| 05:57.33 | Jouva | Ahh ok then |
| 05:57.38 | Jouva | That's probably how this phone does it |
| 05:57.49 | p3nguin | If your phone supports a real transfer, there is no DTMF involved. |
| 05:57.52 | Jouva | I'm also trying to find info on specific features related to the Aastra 6755i and how they relate to asterisk configuration (i.e.: parking, intercom, etc) |
| 05:58.17 | p3nguin | DTMF transfer is used for phones that don't have a tranfer key. |
| 05:58.39 | Jouva | It probably does support that, as there's configuration in the web admin panel to set a soft button on the phone to do transferring |
| 06:00.41 | Jouva | but yeah I've been looking around on google for info about using this series of Aastra phones on Asterisk and coming up empty. Just wanting to make sure I don't miss some piece of configuration as well as get as much functionality as I can |
| 06:02.05 | p3nguin | If the phone is a SIP phone, I'm sure it'll work with Asterisk. |
| 06:03.32 | Jouva | Right. That much I figured ;) I just mean getting the most out of the phone since there's a lot more configuration options than I thought on here. And setting up Asterisk to support everything the phone can do |
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| 06:22.38 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 06:25.40 | dieno | hello everyone |
| 06:25.58 | dieno | have anyone tried ROHC on asterisk before ? |
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| 06:33.34 | smash- | =; |
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| 06:39.47 | phix | ok updated to squeeze |
| 06:40.01 | phix | fixed token required issue in iax |
| 06:40.16 | phix | now I need to figure out how to change from zaptel to whatever it is called now |
| 06:42.44 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 06:43.40 | phix | What is it called in asterisk 1.6? |
| 06:43.51 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 06:45.46 | phix | ok dahdi |
| 06:54.52 | ChannelZ | mahmi |
| 06:56.12 | domi | mah, da wacht man auf, will mal eben wo anrufen, geht ned. dann braucht man 2min um sich auf den server im LAN! einzuloggen um festzustellen, dass der asterisk massiv geDOSed wird... |
| 06:56.23 | domi | toller start in den tag |
| 06:57.26 | kaldemar | domi: auf englisch bitte. |
| 06:57.42 | domi | oops, sorry |
| 06:59.05 | domi | wrong window ;) |
| 06:59.51 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 07:01.22 | Jouva | Can somebody explain to me what might be causing this: http://pastebin.com/vhSVVjxm |
| 07:02.47 | Jouva | I'm wondering if this is more of an issue on the hardphone's end or Asterisk's end. It was able to successfuly REGISTER and even receive a call that came through a GV call that then dialed extension 120 |
| 07:04.15 | smash- | Kobaz, what are the uptimes ever have any failures? |
| 07:04.44 | kaldemar | Jouva: is 444@x.x.x.x a phone or a gateway? |
| 07:04.51 | SiNGLer | Jouva: in a nutshell problem probably is that peer 120 is also defined at server to which you try to dial. and because of that auth fails |
| 07:05.58 | Jouva | But it's the same machine |
| 07:06.21 | Jouva | Lemme try something much simpler |
| 07:08.16 | kaldemar | then dial using SIP/444 instead of SIP/444@x.x.x.x |
| 07:08.59 | Jouva | It's a hardphone |
| 07:10.44 | Jouva | ok so 444 was just some extension I played around with that was more well... just an audio test thing. Pretty pointless. So I got 101 as well which is a softphone |
| 07:10.52 | Jouva | And 101 can't call 120 or vice versa |
| 07:11.01 | Jouva | And they register to the same server |
| 07:12.01 | kaldemar | pastebin your dialplan and a failed attempt to call. |
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| 07:25.26 | phix | No such command 'dahdi' (type 'core show help dahdi' for other possible commands) |
| 07:27.42 | *** join/#asterisk kleszcz (tick@80.54.23.253) |
| 07:29.49 | f2Knight | Q: This is a wild shot, but does anyone have a deb-build script to make asterisk debs? |
| 07:31.59 | phix | hi |
| 07:32.10 | phix | What's the command to make dahdi work? |
| 07:32.27 | SiNGLer | phix: you need to install dahdi |
| 07:33.10 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:33.12 | schmidts | good morning |
| 07:33.21 | SiNGLer | f2Knight: you can try checkinstall |
| 07:34.43 | *** join/#asterisk jkroon (~jkroon@dsl-241-233-144.telkomadsl.co.za) |
| 07:35.45 | jkroon | hi all, i had a client that requested me to automatically remove voicemails older than 7 days, so a simple find ... -exec rm {} \; gets that done simply enough but now this leaves gaps in the numbers, which I can fix, but since I started with this asterisk gets very confused with respect to voicemail. |
| 07:36.00 | f2Knight | SiNGLer, does not build to install on different system. I have a clean system and i would like to keep from in stalling un needed software on it. and Ubuntu only has 1.6 asterisk .. want to use 1.8 with my own config files. |
| 07:36.05 | jkroon | when modifying the files on disk - is there any special action I need to take to let asterisk know things have changes? |
| 07:38.58 | *** join/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
| 07:39.04 | SiNGLer | f2Knight: you can with checkinstall create deb on dev environment, then move generated deb to clean system, installing it and runtime deps, if this does not suit you, I'm out oš suggestions |
| 07:40.27 | *** join/#asterisk shamelessn00b (~chatzilla@203.175.64.42) |
| 07:40.45 | shamelessn00b | Hi, how do I change the default directory where logs and cdrs are creeated |
| 07:40.58 | shamelessn00b | I've already changed the path in /etc/Asterisk.conf |
| 07:41.04 | shamelessn00b | but its not taking effect |
| 07:41.36 | shamelessn00b | astlogdir => /mnt/log/asterisk |
| 07:42.03 | shamelessn00b | I realoaded, but even then its logging in /var/log/asterisk/messages |
| 07:42.12 | f2Knight | SiNGLer, thanks for the help but I am looking for something more along the lines of building the deb file like that used by the repositories |
| 07:42.45 | shamelessn00b | 1.8.2.3 |
| 07:45.28 | kaldemar | shamelessn00b: you may need a restart for those. |
| 07:45.37 | shamelessn00b | I restarted as well |
| 07:45.57 | Jouva | n/m I got it sorted out |
| 07:46.05 | kaldemar | is /etc/Asterisk.conf what your asterisk uses? |
| 07:46.05 | Jouva | I was in the wrong context |
| 07:46.19 | shamelessn00b | <PROTECTED> |
| 07:46.38 | shamelessn00b | I changed a couple of other settings in that file, which took effect |
| 07:46.43 | shamelessn00b | like the light background option |
| 07:46.45 | kaldemar | shamelessn00b: did you remove (!) in [directories](!)? |
| 07:46.51 | shamelessn00b | no |
| 07:47.16 | shamelessn00b | do I have to? |
| 07:47.21 | kaldemar | yes. |
| 07:47.59 | shamelessn00b | astlogdir => /mnt/log/asterisk, does that property control the location for where the CDRs are created as well? |
| 07:48.07 | kaldemar | "[directories](!) ; remove the (!) to enable this" |
| 07:48.47 | kaldemar | shamelessn00b: yes. |
| 07:49.01 | *** join/#asterisk lftsy (~lftsy@194.38.160.117) |
| 07:49.48 | shamelessn00b | ok thanks kaldemar, one more thing, in the same file under the [options] context, is there a default value for those settings, as they are all commented out |
| 07:50.13 | kaldemar | yes. |
| 07:50.19 | shamelessn00b | If so, how can I find out what that is? |
| 07:51.27 | shamelessn00b | maxcalls = 10, huh what |
| 07:52.31 | kaldemar | look at the source? |
| 07:53.16 | shamelessn00b | ok, let me |
| 07:53.59 | kaldemar | most (if not all) seem to be in main/asterisk.c |
| 07:58.52 | shamelessn00b | ok thanks |
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| 08:55.30 | whtsup | hello |
| 08:55.35 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 08:55.42 | whtsup | anyone help me in making asterisk dialplan |
| 08:55.47 | whtsup | i m newbie |
| 08:56.13 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 08:56.52 | kaldemar | whtsup: do you have a concrete issue you want someone to solve? |
| 08:57.10 | whtsup | yah i want to make custom dialplan |
| 08:57.34 | whtsup | can i mess u in pvt |
| 08:57.46 | kaldemar | no. ask in here. |
| 08:58.22 | whtsup | well scenario is i have 2 sip outbound route each have 8 channels |
| 08:59.06 | whtsup | if 8 channels of 1st sip is inuse then call shud go to another sip account |
| 08:59.17 | kaldemar | are you using freepbx to configure your asterisk? |
| 08:59.32 | whtsup | no i m not using freepbx |
| 08:59.43 | whtsup | i m using in climode |
| 09:01.13 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:d9c9:4376:f825:a237) |
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| 09:01.55 | kaldemar | such dialplan functions as GROUP and GROUP_COUNT and app GotoIf will help you with that. you can read function and application documentation in CLI with "core show function GROUP" and "core show application GotoIf" for example. |
| 09:02.04 | kaldemar | ~newbook |
| 09:02.04 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 09:02.23 | kaldemar | ^ this is also good for learning dialplan basics and structure. |
| 09:03.58 | whtsup | ok |
| 09:04.16 | E-bola | Do anybody know a way I can monitor for SIP connectivity? I would like a util/script that calls a number via my asterisk box and verifies it works. Like it shuld call some external service i could setup and verify audio works both ways |
| 09:09.29 | kaldemar | E-bola: http://sipp.sourceforge.net/ |
| 09:11.51 | E-bola | kaldemar: wuld that be able to evaluate audio functionality? |
| 09:12.24 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 09:14.02 | E-bola | Seems very sip protocol oriented |
| 09:14.47 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 09:15.59 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
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| 09:33.01 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 09:41.48 | flashdeluxe | hi! I have a soundfile and i want my voicemail with the number "40" to playback this file, how can i assign this sound file to my voicemailbox "40"? |
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| 09:46.15 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 09:48.59 | E-bola | kaldemar: this seems interesting http://www.opennms.org/wiki/AsteriskAudioMonitor_Specification |
| 09:52.48 | kaldemar | E-bola: that just about listening to events to see what is happening. not making calls and certainly not verifying any media. |
| 09:56.35 | E-bola | Mmm its not finnished in any case |
| 09:57.03 | E-bola | I find it odd this apears to be such an odd thing to want to monitor |
| 09:57.19 | E-bola | i basically wanna make sure my VOIP infrastruxture works from one endpoint to another |
| 09:57.32 | E-bola | weird its not more common to monitor |
| 10:01.10 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 10:02.52 | kaldemar | E-bola: what's wrong with sipp? |
| 10:07.26 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 10:11.39 | E-bola | kaldemar: after reading some of theri examples and some docs, it didnt apear to me that it could evaluate if 2way audio was functioning |
| 10:17.58 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 10:18.26 | benngard | any knows if there is a way to check the status of a cell phone? |
| 10:18.43 | E-bola | benngard: there is here in denmark |
| 10:18.54 | benngard | how? |
| 10:18.54 | E-bola | but its not free na d ofcourse requires the support of ur provider |
| 10:19.29 | benngard | i am living in sweden, havnt heard of any supplier that supprts it |
| 10:19.48 | benngard | i was thinking of doing a short call and check status |
| 10:20.39 | WIMPy | Or send a text message. |
| 10:20.50 | BlackBishop | backup-mirror asterisk # find / -iname "libmyodbc.so" |
| 10:20.51 | BlackBishop | backup-mirror asterisk # find / -iname "libodbcmyS.so" |
| 10:20.52 | BlackBishop | hmmm |
| 10:21.02 | BlackBishop | what whould I need on a gentoo box for those ? |
| 10:21.19 | BlackBishop | trying to set up realtime dynamic |
| 10:21.48 | E-bola | in dk all providers support it i think |
| 10:21.52 | E-bola | it was invented by a dk company |
| 10:21.57 | E-bola | blue position i think its called |
| 10:22.12 | E-bola | We have 1 client that uses it |
| 10:22.28 | E-bola | so the receptionist knows if the mobil sales ppl are busy ont he phone or not when transfering calls to them |
| 10:23.11 | benngard | that excactly what i am looking for |
| 10:24.30 | E-bola | they have great success |
| 10:24.34 | E-bola | so u mgiht wanna look into that |
| 10:24.48 | E-bola | mm they got a uk, no and ofcourse .dk page |
| 10:24.51 | E-bola | but no .se apparently |
| 10:25.02 | WIMPy | Depending on th network operator it might work to send a data call to the phone. |
| 10:25.08 | E-bola | http://www.blueposition.com/en/frontpage/ |
| 10:25.32 | E-bola | WIMPy: how would sending an sms help anything? |
| 10:26.08 | WIMPy | Delivery status |
| 10:26.34 | E-bola | You mean to determine if the phone is turned on an uner coverage? |
| 10:26.40 | E-bola | and under even |
| 10:26.52 | WIMPy | yes |
| 10:26.52 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 10:27.11 | E-bola | ya ok can see how that would do that, i think however benngard wanted the thing i described |
| 10:45.00 | *** part/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 10:46.47 | benngard | what i am looking for is call it in_use |
| 10:48.18 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 10:50.19 | benngard | maybe i can use Originate from ami and do a short call |
| 10:52.33 | alvdav | tach leute |
| 10:52.47 | alvdav | man man man seit gestern uerbtreib ichs voll mit pause machen xD |
| 10:52.55 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 10:53.03 | alvdav | fc... |
| 10:53.12 | alvdav | im sry |
| 10:58.33 | *** join/#asterisk JoTo (~JoTo@p5DE7B442.dip.t-dialin.net) |
| 10:58.54 | JoTo | Hi all |
| 11:00.23 | *** join/#asterisk Diffen (~diffen@c-2472e555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 11:03.15 | BlackBishop | any way I can make asterisk do something upon a call is answered ? |
| 11:03.36 | BlackBishop | I'm currently calling a macro before the dial .. but I'd only want that to happen if the dial is answered |
| 11:05.11 | saxa | hi, can somebody tell me how is transfer of a call working ? I mean , when I pick up at one SIP phone, then how do I transfer it to another extension ? |
| 11:07.15 | plundra | Isn't it possible to have patterns on the callermatching part of an extension? |
| 11:07.53 | BlackBishop | googles chan_sip messages .. |
| 11:08.56 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 11:10.19 | BlackBishop | [2011-02-24 13:02:21] WARNING[31269]: chan_sip.c:15271 receive_message: Received message to <sip:503@192.168.1.101> from <sip:501@192.168.1.105>;tag=92WDG.9.0Vsi5-BrxGl5UcS0PMcowk6y, dropped it... |
| 11:10.32 | *** join/#asterisk qjb (~qjb@a83-163-158-168.adsl.xs4all.nl) |
| 11:10.32 | BlackBishop | why did it drop it .. how can I use that to do stuff with that message ? |
| 11:11.20 | BlackBishop | damn |
| 11:11.24 | BlackBishop | https://issues.asterisk.org/view.php?id=6318 |
| 11:11.25 | BlackBishop | :| |
| 11:12.18 | plundra | Hmm, it is supposed to work. Althoug my ext/_XXXfoo did not. Using ext/322foo is fine. |
| 11:14.21 | kaldemar | BlackBishop: core show application dial |
| 11:15.40 | *** join/#asterisk mechbangirc (~mechbangi@mbl-82-58-226.dsl.net.pk) |
| 11:16.11 | plundra | Oh! [f][o][o] did work, but man that's horrible-looking :-) |
| 11:16.17 | mechbangirc | is it possible to have roaming from one AP to another without disconnect in wifi environment? |
| 11:16.52 | WIMPy | mechbangirc: Sure. But usually not seemless. |
| 11:17.37 | BlackBishop | reads what kaldemar said |
| 11:17.39 | mechbangirc | WIMPy: what exactly means seamless? will i get disconnect? |
| 11:18.13 | WIMPy | No, but a short time without traffic passing. |
| 11:18.33 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 11:18.37 | mechbangirc | k even on DHCP right? |
| 11:18.53 | WIMPy | That doesn't matter. |
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| 11:19.51 | *** join/#asterisk nicola_pav (~nicola@mail2.tikalnetworks.com) |
| 11:19.52 | mechbangirc | do you happen to have such setup. also can you tell me how long it usually takes to get traffic back |
| 11:20.30 | BlackBishop | kaldemar: I don't get it :) |
| 11:20.48 | *** join/#asterisk nicola_pav (~nicola@mail2.tikalnetworks.com) |
| 11:20.54 | WIMPy | Yes, I use 2 APs. In my Netbook I don't notice during normal operation. Havent tried to stream something in that situation. |
| 11:20.58 | kaldemar | BlackBishop: you'll find an option in the application documentation that executes a macro upon answer. |
| 11:21.10 | BlackBishop | reads again |
| 11:21.16 | WIMPy | With the wifi phones I get about 2 seconds of silence. |
| 11:21.21 | nicola_pav | hello. i downloaded a free fax license from digium to test sending receiving fax |
| 11:21.32 | nicola_pav | i followed instructions and i registered. all good |
| 11:21.33 | BlackBishop | M ! |
| 11:21.35 | BlackBishop | yeeey |
| 11:21.40 | nicola_pav | now what? |
| 11:21.50 | nicola_pav | i need to test to send and receive fax |
| 11:21.56 | nicola_pav | what r the steps? |
| 11:22.28 | mechbangirc | have you ever tried DECT |
| 11:22.47 | WIMPy | yes |
| 11:23.13 | WIMPy | But the DECT system I've got only supports manual handover. |
| 11:23.46 | mechbangirc | manual handoever? |
| 11:24.14 | mechbangirc | i have read DECT is better in roaming from bs to bs |
| 11:24.25 | WIMPy | If you walk out of reach, you have to press a soft key and reselect base. |
| 11:24.28 | JoTo | can anyone point me in the right direction pls. : I want to setup a multilang, multilevel IVR on my asterisk 1.6.2 (Elastix) |
| 11:24.32 | JoTo | my problem is: |
| 11:24.40 | JoTo | i can select only one action in the ivr |
| 11:24.58 | JoTo | e.g. i have an ivr with the announcement: for english press 1, for german press 2 |
| 11:25.04 | *** join/#asterisk Amnesia (~Amnesia@unaffiliated/amnesia) |
| 11:25.14 | JoTo | now i want to change language as the user press 1 or 2 |
| 11:25.16 | Amnesia | hm, when I've set a timeout |
| 11:25.26 | Amnesia | how can I get an action to happen next? |
| 11:25.31 | JoTo | but i want also to go ahead to the next ivr with the correct language set |
| 11:25.54 | Amnesia | http://pastie.org/1601802 |
| 11:26.03 | JoTo | is that possible to setup through freepbx/elastix? if not, what do i need to do where? |
| 11:26.45 | JoTo | TYIA |
| 11:27.01 | Amnesia | exten => t,1,Dial(${kantoor}) should be fine right? |
| 11:27.15 | BlackBishop | kaldemar: thanks :D solved. |
| 11:27.24 | mechbangirc | may i know which wifi phone you use. or can you recommend me compatible with asterisk ofcourse |
| 11:27.59 | *** join/#asterisk asilva (~andre@2801:88:1000:2:240:a7ff:fe10:28f) |
| 11:28.10 | Amnesia | anyone? |
| 11:31.19 | asilva | Hello i'm implementing DUNDi using DUNDi functions(DUNDIRESULT,QUERY,LOOKUP) but i have more than one mapping, anyone has done this with multiple mappings ? |
| 11:31.39 | asilva | has create a dialplan to check more than one mapping? |
| 11:31.55 | WIMPy | mechbangirc: I can't really recommend using wifi phones. It's a nice gadget when you're in some other place, but they're not the quality you want to use regularly. |
| 11:32.29 | Amnesia | WIMPy: every used the timeout function? |
| 11:33.14 | WIMPy | Amnesia: Only with (absolute) |
| 11:33.21 | Amnesia | hm |
| 11:33.30 | Amnesia | guess it should work the same |
| 11:33.32 | mechbangirc | and how about DECT in comparision with wifi. i am really confused b/w the two |
| 11:33.48 | Amnesia | http://pastebin.com/psS3X5LR |
| 11:33.57 | Amnesia | it hangs up after the timeout |
| 11:34.04 | WIMPy | But I have noticed that timeouts differ depending on the situation. |
| 11:34.39 | Amnesia | o0 |
| 11:34.50 | Amnesia | I arent getting in the t extenstion at all |
| 11:35.16 | WIMPy | mechbangirc: DECT works well, but if you want seamless handover, you have to pick some professional system. |
| 11:36.45 | WIMPy | Amnesia: Are you sure, you want to use Disa? WaitExten might be more appropriate. |
| 11:37.18 | WIMPy | I don't think Disa is supposed to jump to the special extensions. |
| 11:37.35 | Amnesia | WIMPy: no:P |
| 11:37.55 | Amnesia | I just want something like press the extension you want to dial, and then after the 3 sec timeout all phones should ring |
| 11:38.33 | WIMPy | Use WaitExten or even Read. |
| 11:39.02 | Amnesia | hm let's google |
| 11:39.40 | BlackBishop | kaldemar: what if I want to call an ODBC_FUNCTION ? |
| 11:39.40 | BlackBishop | :/ |
| 11:43.39 | nicola_pav | any guide on how to test send/receive fax with fax for asterisk? |
| 11:44.01 | nicola_pav | i already installed a free key license and all is good |
| 11:44.16 | kaldemar | BlackBishop: do it in a macro/gosub. |
| 11:46.19 | asilva | Hello i'm implementing DUNDi using DUNDi functions(DUNDIRESULT,QUERY,LOOKUP) but i have more than one mapping, anyone has implemented this with multiple mappings ? |
| 11:47.50 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 11:48.37 | kaldemar | asilva: most likely yes. what is the problem? |
| 11:48.58 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 11:53.00 | asilva | kaldemar, just need a hint to create the ael structure to search the EXTEN in three 3 mappings and if finds a result in one of the 3 stop searching. and continues the rest of dial plan |
| 11:57.34 | Amnesia | WIMPy: is it correct waitexten can only read one digit? |
| 11:59.10 | kaldemar | asilva: make 3 queries in different priorities in gotoif's for example. |
| 12:01.15 | Amnesia | ugh has anyone used waitexten her? |
| 12:01.58 | benngard | any recomandations for a sip softphone for iphone? |
| 12:04.28 | asilva | kaldemar, could do it... just a doubt is it possivel to use var arrays in AEL ? like VAR[0] VAR[1] ? |
| 12:09.47 | asilva | benngard, yealink phones are good. |
| 12:09.55 | BlackBishop | kaldemar: I'm trying to get rid of the macros because all I do in 'em are some sql queries |
| 12:10.11 | asilva | benngard, I'm having good experiences with T22 model. |
| 12:10.32 | BlackBishop | so .. I'm trying to use the odbc + mysql I just set up and get rid of the macros .. |
| 12:11.00 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 12:12.19 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 12:14.11 | WIMPy | Amnesia: That depends on your dialplan. |
| 12:14.15 | kaldemar | asilva: why would you need to use arrays? |
| 12:14.32 | WIMPy | It will continue until it finds a match or runs into timeout. |
| 12:16.07 | asilva | kaldemar, to do the mapping search under for |
| 12:16.11 | *** join/#asterisk skrusty (~ben@93-97-20-22.zone5.bethere.co.uk) |
| 12:17.33 | kaldemar | BlackBishop: you need an extension to execute a function anyway, so what harm does a macro do? |
| 12:18.29 | BlackBishop | well .. the macro has the username and password in it |
| 12:18.42 | BlackBishop | I don't want the others to see when they connect to the asterisk console |
| 12:35.29 | BlackBishop | kaldemar: I guess I'll use the gosub :) |
| 12:35.59 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 12:45.33 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 12:47.12 | BlackBishop | func_odbc.c Section 'MYSQLSTART' was found, but there was no SQL to execute. Ignoring. |
| 12:47.15 | BlackBishop | [MYSQLSTART] |
| 12:47.16 | BlackBishop | what's wrong with that ? :/ |
| 12:47.18 | BlackBishop | dsn=asterisk |
| 12:47.21 | BlackBishop | insertsql=INSERT INTO `logs` (`username`,`numar`,`uniqueid`) VALUES ('${ARG1}','${ARG2}', '${ARG3}') |
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| 12:54.22 | Amnesia | WIMPy: hm I put the extensions into the context and it works now:) |
| 12:59.26 | plundra | Does JABBER_STATUS trigger any "traffic" or whatever, or does it just check some status internally? |
| 13:00.44 | plundra | I mean, can I use it a lot or should I avoid it? Currently I trigger those parts of my dialplan via crontab. Checking peoples status and chaning stuff accordingly. |
| 13:01.18 | plundra | But if it just does a local lookup, I could do it on a per-call basis instead. To remove the lag. |
| 13:01.43 | saxa | hi, how do I set up in extensions.conf the outgoing lines , to be able to automatically use DAHDI/2 when DAHDI/1 is in use ? |
| 13:02.40 | *** join/#asterisk Akiraa (~Akira@82.77.116.39) |
| 13:04.31 | *** join/#asterisk E-bola (~bola@x1-6-00-13-46-83-e5-04.k892.webspeed.dk) |
| 13:05.54 | WIMPy | plundra: Not sure, what you're on about, but maybe you just want to enable qualify and devicestates? |
| 13:06.16 | *** join/#asterisk dr_ (~duckz@95.76.24.119) |
| 13:06.54 | plundra | WIMPy: What? Did you want to tell saxa that perhaps? |
| 13:07.17 | plundra | If not, I don't know how that would be related :-) |
| 13:07.23 | WIMPy | No |
| 13:07.44 | WIMPy | Well, what do you want to use jabber status for? |
| 13:07.49 | plundra | I do some lookups on peoples jabberstatus, if they are Free for chat or not. And depending on that, add them or remove them to some queues. |
| 13:08.32 | WIMPy | So external jabber status, not phone/extension related? |
| 13:08.49 | plundra | Yes, physically related but nothing else. |
| 13:09.02 | WIMPy | ok |
| 13:09.39 | WIMPy | saxa: Use Groups. |
| 13:09.48 | plundra | Check john.doe@jabber.foo, then add local/john.doe to the queues. |
| 13:09.52 | saxa | WIMPy: thx, I was just reading about it |
| 13:10.38 | saxa | it should be just DAHDI/G0 in case in extensions.conf correct ? |
| 13:10.54 | bullium | I'd like to ad an extension to my dialplan that prompts the user for the voice mailbox number they wish to check, then asks for that vmb password? any ideas? |
| 13:11.06 | WIMPy | saxa@ Fot example, yes. |
| 13:11.31 | plundra | Judging from the code, it looks like a local lookup only. Which makes sense, it'd be hard to have polling for getting the status etc. of people :) |
| 13:11.36 | WIMPy | bullium: Just call VoiceMailMain. If you don't supply the mailbox as parameter, it will ask. |
| 13:12.18 | saxa | WIMPy: thx |
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| 13:20.26 | bullium | WIMPy: awesome...exten => s,1,VoiceMailMain() |
| 13:20.34 | bullium | WIMPy: does that work? |
| 13:20.39 | WIMPy | yes |
| 13:20.53 | bullium | WIMPy: ok...I'll add that and give it a shot, thanks for the help |
| 13:25.09 | bullium | WIMPy: that did it, thanks |
| 13:27.34 | *** join/#asterisk hurdman (~ygcheny@88.191.131.168) |
| 13:27.37 | hurdman | hi |
| 13:29.23 | leifmadsen | ohai |
| 13:29.40 | E-bola | hey |
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| 13:30.39 | leifmadsen | E-bola: to answer your question from yesterday, no, there is no 1.8.3-rc4. The 1.8.3-rc3 is being released as 1.8.3 and 1.8.4-rc1 will be tagged |
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| 13:32.15 | *** part/#asterisk JoTo (~JoTo@p5DE7B442.dip.t-dialin.net) |
| 13:34.14 | E-bola | yep was what i was hoping for/counting on |
| 13:34.30 | leifmadsen | released should be out this afternoon. I just have to write up some release announcments |
| 13:34.47 | E-bola | i deployed rc3 last night, we still have our IAX channel based crashes (i believe its caused by iax anyway) although its less frequent now |
| 13:35.48 | E-bola | Im going to install 1.8 on a new xen vm soon where i will try to use the dahdi timer instaid and see if that helps any |
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| 13:41.39 | *** join/#asterisk shuki (~chatzilla@bzq-79-178-196-233.red.bezeqint.net) |
| 13:42.18 | shuki | Hello. I have a simple question: Asterisk can do audio web conferencing? |
| 13:42.25 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 13:42.39 | shuki | Or is it over the telephone only? |
| 13:43.56 | *** join/#asterisk fauxalliance (~fauxallia@142.162.197.28) |
| 13:44.23 | kaldemar | it does conferencing over multiple protocols and doesn't care if the client is a telephone, or web or other application. |
| 13:44.56 | leifmadsen | however you need to develop (or find) a web application that can communicate with asterisk |
| 13:45.01 | leifmadsen | there is no "make web conference" |
| 13:45.32 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 13:45.37 | shuki | LOL. I am sure there is no push button for that. |
| 13:45.59 | shuki | Thank you very much for clearing this for me. :-) |
| 13:46.03 | leifmadsen | but ya, there is conferencing in asterisk, and as long as your web app can talk to asterisk somehow, then asterisk won't care |
| 13:46.20 | *** join/#asterisk Khratos (~Khratos@190.166.129.120) |
| 13:46.26 | shuki | Great. It really looks like a great project. |
| 13:46.40 | leifmadsen | zoiper has a webphone fyi |
| 13:47.49 | shuki | Oh, that looks nice. |
| 13:48.11 | shuki | I actually try to find something for PHP... I wanna embed it on my PHP project... |
| 13:48.57 | Chainsaw | Well, you can control Asterisk using the manager interface. It'll probably end up being a flash applet of some description that gets the audio to you. Something that does SIP telephony. |
| 13:48.58 | BlackBishop | [2011-02-24 15:29:42] WARNING[1562]: chan_sip.c:12919 __set_address_from_contact: Invalid contact uri sp518.3105:63;rnpr=d (missing sip: or sips:), attempting to use anyway |
| 13:49.01 | BlackBishop | interesting |
| 13:49.04 | shuki | Oh, I see they have a web interface. |
| 13:51.01 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 13:52.28 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f396812d.cpe.net.cable.rogers.com) |
| 13:54.46 | leifmadsen | BlackBishop: what version? |
| 13:54.52 | leifmadsen | BlackBishop: pretty sure that's been resolved |
| 13:57.56 | BlackBishop | 1.8.2.3 |
| 14:00.38 | E-bola | Ancient, there 1.8.2.4 dude :P |
| 14:00.44 | E-bola | +s |
| 14:00.51 | leifmadsen | BlackBishop: try the 1.8 branch or 1.8.3-rc3 as I'm pretty sure that was already resolved |
| 14:00.57 | leifmadsen | E-bola: there are no bug fixes in 1.8.2.4 |
| 14:01.03 | leifmadsen | (other than a security release) |
| 14:01.09 | leifmadsen | 1.8.2.4 would get him nothing over 1.8.2.3 |
| 14:01.17 | E-bola | sends leifmadsen some humour |
| 14:01.37 | leifmadsen | except that type of humour is misleading to people trying to resolve an issue on their box |
| 14:02.36 | E-bola | .... if they dont get it, you sure are correct. |
| 14:02.59 | BlackBishop | lemme see what the gentoo repo has |
| 14:03.03 | leifmadsen | yuck |
| 14:03.07 | leifmadsen | they won't have anything newer |
| 14:03.09 | BlackBishop | emerge --sync |
| 14:03.14 | BlackBishop | dang |
| 14:03.15 | BlackBishop | :) |
| 14:03.19 | leifmadsen | you'll have to actually compile it yourself |
| 14:03.21 | leifmadsen | *gasp* |
| 14:03.23 | BlackBishop | well, I'll wait 'till the next release .. |
| 14:03.30 | leifmadsen | enjoy! |
| 14:03.44 | BlackBishop | it doesn't have any problems .. but the warning was weird |
| 14:03.46 | BlackBishop | altough |
| 14:03.50 | BlackBishop | come to think about it |
| 14:03.57 | BlackBishop | this is the first time I see this |
| 14:04.03 | BlackBishop | only changes since last time is the odbc+mysql |
| 14:04.06 | leifmadsen | it's just the fullcontact field is going to be populated with junk |
| 14:04.09 | BlackBishop | and the peers are in sql now |
| 14:04.12 | leifmadsen | yes i know what the issue is |
| 14:04.31 | leifmadsen | it's a corruption with the fullcontact field in the SIP table via ODBC realtime |
| 14:04.38 | BlackBishop | ahuh .. |
| 14:04.55 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:05.06 | leifmadsen | https://issues.asterisk.org/view.php?id=18856 |
| 14:05.28 | leifmadsen | fixed here actually: https://issues.asterisk.org/view.php?id=18251 |
| 14:06.11 | leifmadsen | fixed in this revision: Revision: 298482 |
| 14:07.27 | BlackBishop | interesting |
| 14:08.54 | *** join/#asterisk fauxalliance (~fauxallia@142.162.197.28) |
| 14:10.25 | BlackBishop | so why isn't that in 1.8.2.3 or 4 ? :) |
| 14:12.20 | leifmadsen | because those are not bug fix releases |
| 14:12.23 | leifmadsen | they are security releases |
| 14:12.29 | leifmadsen | look at the changelogs |
| 14:12.46 | leifmadsen | 1.8.X.Y releases are either security or regression fixes |
| 14:12.56 | leifmadsen | 1.8.X releases are bug fixes based on issues merged to the branch |
| 14:13.14 | Katty | gooooooooooooooooood morning asterisk!!!! |
| 14:13.19 | leifmadsen | Katty: zup yo?! |
| 14:13.38 | leifmadsen | thinks he is going to have to write another blog post about how Asterisk releases are done |
| 14:14.11 | Tozz_ | write a book about it! |
| 14:14.17 | leifmadsen | I already have ;) |
| 14:14.37 | E-bola | leifmadsen: well it isnt particularly easy to understand, atleast it wasnt for me :) |
| 14:15.12 | leifmadsen | http://ofps.oreilly.com/titles/9780596517342/ch02.html#asterisk-versioning |
| 14:15.31 | BlackBishop | so I'll have to wait for 1.8.3 .. |
| 14:15.39 | leifmadsen | BlackBishop: yes |
| 14:15.49 | BlackBishop | which .. will be quite some time I guess |
| 14:15.56 | BlackBishop | thanks for the info :) |
| 14:16.30 | leifmadsen | 1.8.3 is available as a tag right now |
| 14:16.36 | *** join/#asterisk moy (~moy@CPE0026f3a17020-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 14:16.42 | leifmadsen | but if you're dependent on gentoo emerge, then ya, I have no idea how often they update that |
| 14:18.29 | Katty | hugs leifmadsen |
| 14:18.37 | Katty | leifmadsen: caffinating so far. |
| 14:19.05 | beek | waves to the gang. |
| 14:22.26 | saxa | how do I create templates in extensions.conf ? it is different than in sip.conf correct ? |
| 14:22.33 | BlackBishop | leifmadsen: I'm not dependent .. but .. I'd like to keep using it for updates and stuff |
| 14:22.47 | saxa | i see its [template-name] |
| 14:23.03 | saxa | but then how do i reffer to it ? |
| 14:23.10 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 14:23.45 | saxa | needs to reopen the aterisk book :) |
| 14:23.59 | *** join/#asterisk Tim_Toady (~moi@77.49.3.6) |
| 14:24.29 | leifmadsen | saxa: templates are not dependent on file |
| 14:24.37 | leifmadsen | [my_template](!) |
| 14:24.44 | leifmadsen | [context](my_template) |
| 14:24.57 | leifmadsen | although why don't you just use an include? |
| 14:25.07 | leifmadsen | I've never used a template in extensions.conf.... |
| 14:25.27 | leifmadsen | actually I guess I can think of a reason why that might be useful.... |
| 14:25.39 | leifmadsen | anyways ya, you define the template, then you refer to the template |
| 14:25.47 | leifmadsen | define with (!), then refer with (template_name) |
| 14:26.22 | saxa | leifmadsen: i think i want to use a macro |
| 14:26.34 | leifmadsen | probably :) |
| 14:26.36 | leifmadsen | or GoSub() |
| 14:26.43 | saxa | leifmadsen: thx for confirming the template thing |
| 14:26.59 | saxa | is a little bit confused |
| 14:27.29 | saxa | havent worked a long time already with asterisk |
| 14:27.40 | leifmadsen | perhaps you should just say what you're trying to do :) |
| 14:27.47 | saxa | so im a bit confusing things. |
| 14:28.25 | leifmadsen | well I can tell you how things work, but without knowing what you're trying to do it's hard to tell you what to use :) |
| 14:28.39 | saxa | just wanted to create a template like for a extension where if busy or unavailable it can be used for any extension i have |
| 14:28.50 | leifmadsen | ya you want a GoSub() or Macro() |
| 14:28.57 | saxa | ok |
| 14:28.58 | leifmadsen | (GoSub()'s are preferred for new dialplans) |
| 14:29.04 | leifmadsen | ~newbook |
| 14:29.04 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 14:29.15 | leifmadsen | consider checking out that documentation -- read the dialplan chapters |
| 14:30.37 | saxa | since i have that code http://pastebin.com/0cXhpNbQ |
| 14:30.42 | *** join/#asterisk JamesHarrison (~jharr@hometree.mmmetrics.co.uk) |
| 14:30.58 | saxa | and I want it to use with all extensions i have , not only 1001 |
| 14:31.19 | saxa | but is there a way to not repeat all that for each exten ? |
| 14:31.25 | leifmadsen | saxa: ya, I would turn that into a GoSub() and then you can perform a lookup in the AstDB (or wherever) to determine what extension 1001 (or whatever extension was dialed) maps to device wise |
| 14:31.31 | leifmadsen | yes that's exactly what a GoSub() is for |
| 14:31.45 | *** join/#asterisk nzw (~rychu@85.11.67.125) |
| 14:31.51 | saxa | great, thx leifmadsen |
| 14:31.59 | nzw | hi |
| 14:32.07 | leifmadsen | saxa: try starting around here: http://ofps.oreilly.com/titles/9780596517342/ch10.html#GoSub |
| 14:32.14 | nzw | do anyone use MGCP NCS version at asterisk 1.8? |
| 14:32.15 | BlackBishop | gentoo seem to have 1.8.2.4 |
| 14:32.15 | *** join/#asterisk manji (~manjiki@penicuik.physics.upatras.gr) |
| 14:32.26 | BlackBishop | so I might not have to wait that long for 1.8.3 when it gets oout |
| 14:32.43 | nzw | i'm trying to make my Arris cable modem work, but no luck |
| 14:32.51 | nzw | any advices? |
| 14:32.57 | leifmadsen | has never used MGCP |
| 14:33.22 | manji | the fact that asterisk will open video ports and let eg the caller send video, even though the callee does not support video |
| 14:33.22 | *** join/#asterisk fauxalliance (~fauxallia@142.162.197.28) |
| 14:33.30 | manji | is a known bug? |
| 14:33.58 | saxa | another thing i would like to set is , when I get a call on DAHDI/2 it rings the phone on DAHDI/3 |
| 14:34.29 | saxa | so DAHDI/2 is the analog line I use usually for fax |
| 14:34.29 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:34.45 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 14:34.45 | saxa | and on DAHDI/3 I have an old fax machine |
| 14:35.19 | saxa | so when a call comes over to DAHDI/2 I would like it rings the phone on DAHDI/3 |
| 14:35.44 | saxa | but I use both lines DAHDI/1 and DAHDI/2 from my SIP phone for calling out |
| 14:36.20 | saxa | now I set up groups |
| 14:36.43 | saxa | so when calling from sip it calls out on DAHDI/G0 |
| 14:37.08 | saxa | which is firstly on DAHDI/2 and if that is busy it falls on DAHDI/1 |
| 14:37.48 | saxa | but have no idea how to answer only DAHDI/2 and route it to the fax machine ? |
| 14:40.36 | Tim_Toady | saxa you define a context for dahdi 2 and in that context you put something like exten => _X.,1,Dial(DAHDI/3) |
| 14:41.07 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:41.24 | Tim_Toady | so every call that comes to channel 2 is routed to dahdi channel 3 |
| 14:41.26 | leifmadsen | saxa: I highly suggest you read some documentation as all of your questions could be answered there |
| 14:42.26 | saxa | ah, I have already the following in extensions.conf for [incoming] context http://pastebin.com/FxyX8atL |
| 14:42.34 | wdoekes2 | leifmadsen: "slightly than that for Macro() because" <-- Macro is not in <code> style |
| 14:42.57 | leifmadsen | wdoekes2: I have a feeling that was resolved already by our production editor :) |
| 14:43.06 | leifmadsen | that website does not have the latest changes after the production review |
| 14:43.14 | saxa | leifmadsen: yes, i will for sure, thx for the help with gosub |
| 14:43.21 | leifmadsen | she changed nearly every line in the book :) |
| 14:43.28 | wdoekes2 | heh |
| 14:43.28 | leifmadsen | wdoekes2: thanks though |
| 14:43.47 | wdoekes2 | can't help it.. *must* *click* *every* *link* |
| 14:43.53 | leifmadsen | lol |
| 14:44.12 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 14:44.32 | BlackBishop | shouldn't exten => _XXXXXXXXXX,n,Dial(Datacard/${digito},,U(mysqlstart)) go to exten => _XXXXXXXXXX,n(mysqlstart),ODBC_MYSQLSTART(${CALLERID(num)},${digito},${UNIQUEID}) ?! ( which is further down the road .. ) |
| 14:44.50 | BlackBishop | it should go there upon answer of the Dial .. |
| 14:46.22 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
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| 14:49.48 | *** join/#asterisk esterpader (~esterpade@58.69.18.6) |
| 14:49.49 | BlackBishop | because I get gosub_exec: Attempt to reach a non-existent destination for gosub: (Context:mysqlstart, Extension:s, Priority:1) |
| 14:49.50 | kaldemar | BlackBishop: it used to behave just like the macro, in your case going to mysqlstart,s,1. |
| 14:49.52 | BlackBishop | :| |
| 14:50.22 | BlackBishop | ahuh, so I should make it go to ${EXTEN},?! |
| 14:51.01 | esterpader | Good day guys, Please bare with me if this has been asked for so many times, does asterisk queues can handle at least 80 members in queue? |
| 14:51.45 | kaldemar | try ${CONTEXT},${EXTEN},mysqlstart or something. anyway, it doesn't work like the Gosub app. |
| 14:52.15 | leifmadsen | esterpader: asterisk does not care about the number of agents or members in a queue -- just your CPU power does |
| 14:52.26 | leifmadsen | esterpader: asterisk will handle it -- will your server though? |
| 14:52.29 | leifmadsen | that's the question you need to ask |
| 14:52.46 | leifmadsen | before you ask if your X.Y server will handle it -- we don't know -- you'll have to test it |
| 14:54.02 | kaldemar | BlackBishop: you're not the first one to expect it to work like Gosub would... |
| 14:54.10 | BlackBishop | that's good to know |
| 14:54.11 | BlackBishop | :) |
| 14:54.24 | BlackBishop | all I want it to do is call a function .. |
| 14:54.25 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 14:54.36 | BlackBishop | with some params |
| 14:54.37 | esterpader | leifmadsen, our server is xeon with 4 gb of ram |
| 14:56.42 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
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| 14:58.04 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:58.22 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 15:00.05 | manji | ah great my turn |
| 15:00.05 | BlackBishop | kaldemar: same thing .. non-existant destination |
| 15:00.18 | BlackBishop | well, I'll just use a macro then |
| 15:00.23 | *** join/#asterisk fofware (~Fabian@host132.186-108-156.telecom.net.ar) |
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| 15:00.36 | manji | when having videosupport=yes |
| 15:01.10 | manji | and canreinvite=no |
| 15:01.37 | wdoekes2 | BlackBishop: U(x) should go to context [x] |
| 15:01.52 | manji | is it normal that asterisk will let on client send video even though the other end does not support video calls? |
| 15:02.10 | kaldemar | BlackBishop: you don't need macro, you can do a context by the name mysqlstart with exten s. |
| 15:05.11 | BlackBishop | kaldemar: yeah, but I should __ some vars |
| 15:05.23 | BlackBishop | and it seems to get even more complicated .. I want to simplify things |
| 15:06.02 | BlackBishop | the main idea is that upon the answer of a call I want ODBC_MYSQLSTART(param1,param2,param3) should get called |
| 15:06.17 | BlackBishop | ODBC_MYSQLSTART being a [MYSQLSTART] in func_odbc.conf |
| 15:07.39 | leifmadsen | esterpader: should be fine |
| 15:08.33 | *** join/#asterisk Kedare (~Kedare@onyxia.netyxia.net) |
| 15:09.23 | esterpader | leifmadsen, well thanks that's good to know, the client is a major bank in our country and they are gonna use asterisk business edition for this matter and it's very critical for them not to reboot the server from time to time :D |
| 15:10.00 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 15:12.05 | leifmadsen | esterpader: test test test :) |
| 15:12.10 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
| 15:12.15 | leifmadsen | then when you think you're done, add a couple new tests |
| 15:12.28 | leifmadsen | whatever you're projecting the project to take, at least double it |
| 15:13.12 | *** join/#asterisk Mhaddog__ (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 15:13.30 | iprouteth0 | Could anyone give some assistance with trouble building dahdi? |
| 15:13.46 | BlackBishop | <PROTECTED> |
| 15:13.52 | BlackBishop | so I can't just call that function like this |
| 15:13.55 | esterpader | leifmadsen, ok, thanks for your advice |
| 15:14.12 | leifmadsen | BlackBishop: sounds like you're using it wronge because that's a function, not an application |
| 15:14.23 | BlackBishop | googles how to call a function ! |
| 15:14.37 | leifmadsen | documentation does exist for this eh? |
| 15:14.46 | leifmadsen | infobot: tell BlackBishop about newbook |
| 15:14.53 | BlackBishop | I'm on it |
| 15:15.06 | BlackBishop | ofps.oreilly.com/titles/9780596517342 |
| 15:15.07 | BlackBishop | :) |
| 15:15.47 | iprouteth0 | http://pastebin.com/fepj3DCa <= build error under gentoo |
| 15:15.50 | titter | Hrmm 1.8.2.4 went back in time ... 2010/02/21, silly topic. |
| 15:15.52 | *** join/#asterisk [netman] (~netman@133.Red-80-39-48.staticIP.rima-tde.net) |
| 15:16.15 | kaldemar | BlackBishop: an extension needs an application, be it NoOp or whatever else that actually does something. |
| 15:16.35 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.2.4 (2011/02/21), 1.6.2.16.2 (2011/02/21), 1.4.39.2 (2011/02/21), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 15:16.44 | leifmadsen | I have no idea what you're talking about it says 2011 fo rme |
| 15:17.56 | BlackBishop | kaldemar: exten => s,1,NoOp(ODBC_MYSQLSTART(${ARG1},${ARG2},${ARG3})) |
| 15:17.59 | BlackBishop | like that ? |
| 15:18.19 | BlackBishop | which was called by ,M(mysqlstart^500^0760905294^1298560079.0)") |
| 15:18.19 | titter | ;) |
| 15:18.29 | kaldemar | BlackBishop: yep. |
| 15:18.53 | BlackBishop | now to solve the MYSQLSTART thing |
| 15:19.51 | BlackBishop | ow heck .. load_dynamic_module: Error loading module 'res_config_odbc.so': /usr/lib64/asterisk/modules/res_config_odbc.so: undefined symbol: ast_odbc_clear_cache |
| 15:20.12 | BlackBishop | init_acf_query: Section 'MYSQLSTART' was found, but there was no SQL to execute. Ignoring. |
| 15:20.16 | BlackBishop | wth ? |
| 15:20.24 | BlackBishop | [MYSQLSTART] |
| 15:20.24 | BlackBishop | dsn=asterisk |
| 15:20.24 | BlackBishop | insertsql=INSERT INTO `logs` (`username`,`numar`,`uniqueid`) VALUES ('${ARG1}','${ARG2}', '${ARG3}') |
| 15:20.29 | BlackBishop | what doesn't it like about that ? |
| 15:20.57 | leifmadsen | BlackBishop: actually no that is not right |
| 15:21.02 | BlackBishop | :/ |
| 15:21.10 | BlackBishop | looks more closely |
| 15:21.14 | leifmadsen | Set(ODBC_MYSQLSTART(ARG1)=VAL1) |
| 15:21.20 | leifmadsen | you're trying to do an insert |
| 15:22.04 | BlackBishop | so I should set ARG1,2 and 3 in the macro |
| 15:22.08 | BlackBishop | then call it ? |
| 15:22.22 | leifmadsen | your question doesn't compute |
| 15:22.32 | BlackBishop | but the error looks like there's a problem in MYSQLSTART in func_odbc... |
| 15:22.54 | leifmadsen | there error is in how you're using it |
| 15:23.36 | BlackBishop | ok, so step1) add Set(ODBC_MYSQLSTART(ARG1)={$ARG1}) in the [macro-mysqlstart] |
| 15:23.50 | BlackBishop | same for arg2 and 3 |
| 15:23.52 | BlackBishop | right ? |
| 15:23.55 | leifmadsen | not quite |
| 15:24.06 | BlackBishop | is lost again |
| 15:24.27 | leifmadsen | Set(ODBC_MYSQLSTART()=${ARG1},${ARG2},${ARG3}) |
| 15:24.45 | leifmadsen | then the args passed to the macro are available as ${VAL1} etc |
| 15:25.31 | BlackBishop | exten => _XXXXXXXXXX,n(setcosmote),Dial(Datacard/${digito},,M(mysqlstart^${CALLERID(num)}^${digito}^${UNIQUEID})) |
| 15:25.42 | BlackBishop | so the vars passed to the macro are available as ${ARG1} etc... |
| 15:25.49 | BlackBishop | right ? |
| 15:25.58 | leifmadsen | when you do the Set(ODBC_MYSQLSTART()=...) everything in the ODBC_MYSQLSTART(arg1,arg2,arg3)=val1,val2,val3) |
| 15:26.10 | leifmadsen | yes the values are passed as ARG1 etc |
| 15:26.25 | leifmadsen | but then the ARG1 and VAL1 in the func_odbc.conf file are entirely separate of that |
| 15:26.38 | leifmadsen | ${ARG1} in func_odbc.conf is NOT the same ${ARG1} you passed from the macro |
| 15:26.57 | leifmadsen | please look at my example above |
| 15:26.57 | BlackBishop | well, the macro now does this: |
| 15:26.59 | BlackBishop | exten => s,1,NoOp(ODBC_MYSQLSTART(${ARG1},${ARG2},${ARG3})) |
| 15:27.03 | leifmadsen | NO!!!!!! |
| 15:27.06 | leifmadsen | you're not looking |
| 15:27.06 | BlackBishop | I am looking at it but I don't get it |
| 15:27.12 | leifmadsen | you're not setting it in the NoOp() |
| 15:27.16 | leifmadsen | Set()!!! |
| 15:27.18 | leifmadsen | Set()! |
| 15:27.34 | leifmadsen | Set(ODBC_MYSQLSTART()=${ARG1},${ARG2},${ARG3}) |
| 15:27.45 | esterpader | BlackBishop, use Set instead of noop, noop means no operation |
| 15:27.49 | leifmadsen | the args are then available as ${VAL1},${VAL2},${VAL3} in the func_odbc |
| 15:28.03 | BlackBishop | <PROTECTED> |
| 15:28.10 | BlackBishop | same thing |
| 15:28.20 | leifmadsen | sorry, I can't help you it seems |
| 15:28.38 | leifmadsen | did you do Set(ODBC_MYSQLSTART and not Set(MYSQLSTART) ? |
| 15:28.48 | BlackBishop | [macro-mysqlstart] |
| 15:28.48 | BlackBishop | exten => s,1,Set(ODBC_MYSQLSTART()=${ARG1},${ARG2},${ARG3}) |
| 15:28.50 | leifmadsen | because it sounds like you did the latter |
| 15:29.24 | leifmadsen | sorry I have to run and finish some release annoncements |
| 15:29.31 | titter | ${VAL1} |
| 15:30.21 | leifmadsen | ok I'm just going to write the dialplan for you because this is bugging me |
| 15:30.32 | BlackBishop | [2011-02-24 17:21:21] WARNING[11518]: func_odbc.c:1005 init_acf_query: Section 'MYSQLSTART' was found, but there was no SQL to execute. Ignoring. |
| 15:30.35 | BlackBishop | [2011-02-24 17:21:21] ERROR[11518]: func_odbc.c:1425 load_module: Invalid parameters for category MYSQLSTART |
| 15:30.39 | BlackBishop | wth :/ |
| 15:30.54 | BlackBishop | this error seems like a thing in func_odbc.conf :\ |
| 15:31.02 | titter | So what's that look like now? |
| 15:31.13 | BlackBishop | [MYSQLSTART] |
| 15:31.13 | BlackBishop | dsn=asterisk |
| 15:31.13 | BlackBishop | insertsql=INSERT INTO `logs` (`username`,`numar`,`uniqueid`) VALUES ('${VAL1}','${VAL2}', '${VAL3}') |
| 15:32.04 | pabelanger | BlackBishop: you are using ${ARG#} in your dialplan, but ${VAL#} in func_odbc. they must be the same |
| 15:32.30 | leifmadsen | wrong |
| 15:32.36 | BlackBishop | set to insertsql=INSERT INTO `logs` (`username`,`numar`,`uniqueid`) VALUES ('${ARG1}','${ARG2}', '${ARG3}') |
| 15:32.47 | BlackBishop | same thing |
| 15:33.30 | pabelanger | s/insertsql/writesql |
| 15:33.34 | pabelanger | BlackBishop: ^ |
| 15:34.08 | ssureshot | dahdi_hardware shows my device but when I run genconf it tells me there are no spans.. dahdi_tool does not report any cards???? |
| 15:34.25 | leifmadsen | pabelanger: insertsql is value |
| 15:34.28 | leifmadsen | valid* |
| 15:34.34 | BlackBishop | ; writesql The statement to execute when writing to the function class. |
| 15:34.34 | BlackBishop | ; insertsql The statement to execute when writing to the function class |
| 15:34.41 | WIMPy | ssureshot: You need to load the driver. |
| 15:35.06 | BlackBishop | leifmadsen: I don't get that error with writesql though |
| 15:35.12 | leifmadsen | BlackBishop: core show function ODBC_MYSQLSTART |
| 15:35.13 | titter | [MYSQLSTART] looks to be the issue to me |
| 15:35.21 | BlackBishop | titter: that's what I said |
| 15:35.25 | leifmadsen | titter: how so? |
| 15:35.42 | ssureshot | WIMPy: it looks as though it is loaded.. dahdi 198225 12 xpd_pri,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp |
| 15:35.53 | BlackBishop | [Syntax] |
| 15:35.53 | BlackBishop | ODBC_MYSQLSTART(<arg1>[...[,<argN>]]) |
| 15:36.10 | WIMPy | ssureshot: Much more than just "it". |
| 15:36.42 | pabelanger | leifmadsen: true, but I could only get writesql working when I did some local testing |
| 15:36.46 | WIMPy | ssureshot: What card? |
| 15:37.15 | ssureshot | it's a Xorcom Astribank2 |
| 15:37.24 | ssureshot | usb |
| 15:37.35 | BlackBishop | leifmadsen: with writesql it seems to do what's it supposed to do |
| 15:38.18 | BlackBishop | anybody want to help make it work with insertsql or should I leave it like this ? |
| 15:38.28 | leifmadsen | possibly a bug with insertsql |
| 15:38.30 | leifmadsen | I'd file an issue |
| 15:38.32 | titter | leifmadsen: Maybe a typo between the dp and func ... but now it seems like that's not the case |
| 15:38.49 | leifmadsen | show what happens with writesql and what happens with insertsql |
| 15:38.59 | leifmadsen | change NOTHING but that one line |
| 15:39.14 | pabelanger | also 'core set debug 15' |
| 15:39.20 | BlackBishop | <PROTECTED> |
| 15:39.32 | BlackBishop | with writesql |
| 15:39.35 | BlackBishop | and with insertsql |
| 15:39.37 | BlackBishop | I get this: |
| 15:39.37 | BlackBishop | [2011-02-24 17:21:21] WARNING[11518]: func_odbc.c:1005 init_acf_query: Section 'MYSQLSTART' was found, but there was no SQL to execute. Ignoring. |
| 15:39.41 | BlackBishop | [2011-02-24 17:21:21] ERROR[11518]: func_odbc.c:1425 load_module: Invalid parameters for category MYSQLSTART |
| 15:40.24 | pabelanger | what version of asterisk? |
| 15:40.28 | BlackBishop | 1.8.2.4 |
| 15:40.36 | BlackBishop | that error appears at start-up |
| 15:40.42 | BlackBishop | when it's trying to register that function |
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| 15:41.07 | *** mode/#asterisk [+o mnicholson] by ChanServ |
| 15:43.12 | leifmadsen | insertsql might only be valid when you already have writesql in use |
| 15:43.23 | leifmadsen | writesql is more for an UPDATE and insertsql for INSERT I believe |
| 15:43.29 | leifmadsen | it's something like that |
| 15:44.03 | BlackBishop | well, that's what I thought |
| 15:44.08 | BlackBishop | that's why I tried using insertsql |
| 15:44.13 | BlackBishop | because I am using an INSERT !@ |
| 15:45.24 | leifmadsen | shrugs |
| 15:45.28 | leifmadsen | Corydon76-home: ^^^ |
| 15:46.48 | pabelanger | Ya, it looks like you need to have writesql and readsql to use insertsql. Or, just writesql. |
| 15:47.12 | BlackBishop | I'll stick to writesql then .. |
| 15:47.30 | pabelanger | just looked at func_odbc, if that is the case we should update the documentation |
| 15:47.39 | pabelanger | but Corydon76-home will know more |
| 15:47.54 | BlackBishop | now ... how do I call a damn ODBC_FUNCTION without Set ? |
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| 15:48.11 | BlackBishop | I don't care about it's return value |
| 15:49.27 | pabelanger | Return(${ODBC_BLAH(${ARG1},${ARG2})}) ? |
| 15:49.55 | pabelanger | I wrapped mine in a Gosub |
| 15:50.23 | nathan7 | boom |
| 15:52.24 | leifmadsen | pabelanger: make sure you tell him that when you use that method you need to put the INSERT into the readsql= line |
| 15:53.07 | leifmadsen | NoOp(${SET(ODBC_MYSQLSTART()=one,two,three)}) |
| 15:53.45 | leifmadsen | although I don't get why the use of Set() is so bothersome. You're trying to write values, thus you have to use the Set() method. |
| 15:53.50 | leifmadsen | you're not saving any return values either |
| 15:54.28 | leifmadsen | using a Set() causes the writesql= line to be used. Otherwise, when you use the reading method (like pabelanger showed) you're using the readsql= line |
| 15:55.04 | pabelanger | yar! |
| 15:56.10 | pabelanger | I plan to do some testing with the transaction layer soon(tm). Looks to be interesting, and powerful :) |
| 15:56.12 | *** part/#asterisk erinspice (~erin@nat/digium/x-sssddinyeuaezhgm) |
| 15:56.47 | leifmadsen | pabelanger: just like me! |
| 15:57.09 | pabelanger | HIO! |
| 15:57.27 | leifmadsen | we should plan a trip soon |
| 15:58.51 | pabelanger | leifmadsen: End of March! |
| 15:59.00 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 15:59.03 | leifmadsen | pabelanger: I like it! |
| 15:59.10 | leifmadsen | THE IDES OF MARCH! |
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| 16:05.14 | BlackBishop | So I should use SET(ODBC_MYSQLSTART()=one,two,three) + writesql |
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| 16:08.40 | BlackBishop | is writesql ok for an UPDATE too ? |
| 16:09.05 | BlackBishop | I guess so |
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| 16:18.16 | esterpader | anyone care to share, is there any dialplan function that can get the position of the occurence of certain character in the string, i know it can be done via AGI but there were some situation agi is to complex for a simple job |
| 16:18.40 | esterpader | it's just like strchr in C function? |
| 16:19.43 | wdoekes2 | LENGTH(CUT(data,char)) ? |
| 16:20.36 | wdoekes2 | or func_odbc and sql functions ;) |
| 16:22.42 | esterpader | wdoekes2, let me try that !! thanks |
| 16:23.03 | leifmadsen | LEN() is the function I think |
| 16:23.23 | leifmadsen | and could have sworn there was a function that did that.... |
| 16:23.27 | esterpader | wdoekes2, by the way i'm yours on your STRREPLACE function, though it might be merge to trunk |
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| 16:28.06 | esterpader | wdoekes2, i'll give you an example exten => _111,1,Set(NUMBER=150.00) the question is how can i get the position of the . in this case the answer should be 3 ? |
| 16:28.33 | p3nguin | If the extension is literally 111, why do you have an underscore on the front? |
| 16:29.22 | esterpader | p3nguin, yeah my bad |
| 16:29.33 | p3nguin | And the position of the . is 4, not 3. |
| 16:29.39 | esterpader | p3nguin, it should be 111 I stand corrected |
| 16:29.45 | esterpader | yep it's 4 |
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| 17:44.08 | ssureshot | would dahdi_hardware see my hardware if it didn't load the driver? |
| 17:46.01 | Qwell | no |
| 17:46.05 | Qwell | well |
| 17:46.17 | Qwell | well |
| 17:46.22 | Qwell | yes |
| 17:46.42 | p3nguin | ssureshot: I can tell you one thing... |
| 17:46.47 | p3nguin | ssureshot: The answer is either yes or no. |
| 17:47.05 | chazzam | dahdi_hardware does I think |
| 17:47.06 | Qwell | it's basically a wrapper for lspci, so yes |
| 17:47.15 | Qwell | I just tested it |
| 17:47.24 | chazzam | nothing else dahdi_* does though |
| 17:47.32 | ssureshot | yeah, I unloaded dahdi and used it.. it still sees it... dummy me.. |
| 17:47.53 | Qwell | sigh, reboot time. |
| 17:48.17 | p3nguin | Why? Still running Windows? |
| 17:48.27 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 17:48.29 | Qwell | replacing hardware |
| 17:48.47 | ssureshot | yeah.. my first install with ubuntu saw the card with no problems,, I'm trying now to compile them from source and am having issues with dahdi recognizing the astribank |
| 17:48.48 | p3nguin | No hot plug? |
| 17:48.52 | Qwell | p3nguin: if only |
| 17:49.01 | p3nguin | It would be more convenient. |
| 17:49.05 | Qwell | indeed |
| 17:49.17 | Qwell | <detached> |
| 17:49.29 | tzafrir_laptop | ssureshot, what problem? |
| 17:49.37 | chazzam | you should slap the pci bus and tell it to get with the program. Then just shove the cards in anyway, or rip them out. whichever |
| 17:49.50 | padwasabi | Greetings. Anybody here know if there is a common cause for: Got SIP response 503 "No Circuit Available"? |
| 17:50.03 | padwasabi | I've googled and found nothing useful |
| 17:50.05 | ssureshot | with the default debian packages and astrisk 1.6 the card worked... when I compile from source I can't get dahdi to see the card /spans |
| 17:50.05 | tzafrir_laptop | ssureshot, please pastebin the output of: dahdi_hardware -v; lsdahdi |
| 17:50.30 | ssureshot | will do 1 sec let me reboot it so it's fresh |
| 17:51.06 | *** join/#asterisk ariel_ (~chatzilla@99-1-236-49.lightspeed.miamfl.sbcglobal.net) |
| 17:51.51 | padwasabi | The trouble is that though I CAN make calls from my SIP soft phone, I CAN'T originate the same call over the AMI |
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| 17:52.37 | ssureshot | tzafrir_laptop: http://pastebin.com/Rnkf5njp |
| 17:53.00 | padwasabi | When watch the CLI I see the same msgs in both cases, but after trying to call the outbound number I get the SIP 503 from the AMI |
| 17:54.03 | tzafrir_laptop | ssureshot, what version of DAHDI? E1 or T1? |
| 17:54.06 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:54.45 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
| 17:55.32 | ssureshot | T1 |
| 17:56.16 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 18:00.08 | ssureshot | tzafrir_laptop: T1,, (again just in case you were waiting on your client teling you there was a message.) |
| 18:02.15 | tzafrir_laptop | ssureshot, yeah. I was elsewhere, indeed. What's the output of grep ^pri_protocol /etc/dahdi/xpp.conf |
| 18:03.57 | tzafrir_laptop | oh, and looking at the backlog: some versions, including 2.4.0 you seem to have, |
| 18:04.14 | tzafrir_laptop | don't properly show if xpp_usb is loaded without -v |
| 18:04.29 | tzafrir_laptop | This hopefully got fixed for most systems |
| 18:04.35 | ssureshot | tzafrir_laptop: hmm that file does not exist |
| 18:04.57 | tzafrir_laptop | You have to set that if you want to set the port as T1, rather than E1 |
| 18:05.28 | tzafrir_laptop | (The fact that you get an error is a bug, already fixed in SVN, but that happens to be a workaround for it as well) |
| 18:05.42 | ssureshot | no auto setup for that one I take it? |
| 18:06.07 | tzafrir_laptop | ssureshot, how can an auto setup tell if you want T1 or E1? |
| 18:06.09 | jkroon | when using SIP, is it possible to see multiple m=audio lines in an SDP description? |
| 18:06.39 | tzafrir_laptop | You don't have that file? If you installed from source: 'make samples' in dahdi-tools |
| 18:07.01 | ssureshot | tzafrir_laptop: right I'm not sure what the syntax is supposed to be I guess is where I was going with that ... oh got ya.. didn't run make samples |
| 18:07.35 | tzafrir_laptop | ssureshot, or rather, just: echo pri_protocol T1 >/etc/dahdi/xpp.conf |
| 18:07.43 | tzafrir_laptop | Would also do |
| 18:09.02 | ssureshot | <tzafrir_laptop>: have to go with the latter no rule to make samples |
| 18:09.48 | tzafrir_laptop | gah. it's 'config' . 'samples' is for asterisk |
| 18:10.04 | tzafrir_laptop | nm |
| 18:10.05 | ssureshot | lol k |
| 18:15.18 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 18:15.35 | ssureshot | <tzafrir_laptop>: ah now it sees it,, |
| 18:19.41 | ssureshot | <tzafrir_laptop>: thank you very much,, now I'm getting somewhere |
| 18:33.21 | p3nguin | Hmm, when did tzafrir_laptop say those things? |
| 18:35.52 | WIMPy | Some time between 18:02 and 19:15 |
| 18:37.00 | WIMPy | Some time between 18:02 and 18:15 |
| 18:37.11 | p3nguin | I don't find any record of tzafrir_laptop saying "ah now it sees it,," nor "thank you very much,, now I'm getting somewhere" but ssureshot was able to quote him somehow. |
| 18:37.51 | WIMPy | Maybe you should filter the <>? |
| 18:38.04 | p3nguin | I should filter it? Why should I filter it? |
| 18:38.27 | WIMPy | Because you don't think it's a quote then. |
| 18:38.56 | p3nguin | If I filter those symbols, then when someone quotes someone else, it'll look to me like they are speaking to the person rather than quoting him. |
| 18:39.18 | p3nguin | That'll get really confusing and bothersome. |
| 18:39.25 | WIMPy | That's what I think is the case. |
| 18:41.56 | *** join/#asterisk LemensTS (~matthew@70.238.156.107) |
| 18:42.19 | LemensTS | Is there anyway to see just the SIP trunks and not the SIP extensions in the cli? |
| 18:42.37 | WIMPy | ~siptrunk |
| 18:42.37 | infobot | extra, extra, read all about it, siptrunk is something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 18:42.40 | p3nguin | double fail |
| 18:42.49 | LemensTS | woot |
| 18:43.19 | p3nguin | SIP also has no concept of extensions. SIP only has users, peers, and friends. |
| 18:45.59 | *** part/#asterisk LemensTS (~matthew@70.238.156.107) |
| 18:46.03 | p3nguin | So... what do you really want to do? |
| 18:46.10 | p3nguin | Oh, nevermind. |
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| 18:51.00 | drmessano-lt | type=notmyfriend |
| 18:55.28 | p3nguin | :( |
| 19:05.40 | ssureshot | p3nguin: haha sorry I guess < > means quote |
| 19:05.40 | ssureshot | so let me reiterate that |
| 19:06.49 | ssureshot | tzafrir_laptop: thank you for your help,,, greatly appreciated. the T1 is up and running with asterisk!! now I just have to work on my phones booting |
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| 19:11.30 | carrar | moof |
| 19:12.35 | Kobaz | oh man |
| 19:12.39 | Kobaz | i keep finding bugs in pjsua |
| 19:13.02 | carrar | In 1.8, is there a easy CLI command to see if a sip channel has re-invited to T.38 when passing through asterisk? |
| 19:13.20 | carrar | that would be super if there was |
| 19:15.31 | leifmadsen | carrar: I doubt it -- not really a CLI type of thing |
| 19:16.15 | carrar | Would be nice to be able to see that a channel has switched to T.38 |
| 19:16.25 | carrar | without having to do debug or packet trace |
| 19:16.31 | *** join/#asterisk kfife (~Miranda@home.chicagoventure.com) |
| 19:17.37 | leifmadsen | that is pretty specific functionality. Someone would have to write the code to do it. |
| 19:17.51 | carrar | Let me know when you've completed coding that ;) |
| 19:17.59 | leifmadsen | I don't code |
| 19:18.20 | carrar | so next week sometime then! |
| 19:18.22 | carrar | GREAT |
| 19:18.25 | leifmadsen | heh |
| 19:18.29 | leifmadsen | yep check back then |
| 19:18.33 | carrar | heh |
| 19:18.55 | carrar | Maybe I can get Katty to code that up |
| 19:20.50 | Katty | i beeped. |
| 19:21.00 | Katty | sure! i'll get right on it |
| 19:21.04 | Katty | hugskwishes carrar |
| 19:21.10 | carrar | w00 h00!! |
| 19:21.19 | carrar | hugcodewishes Katty!! |
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| 19:24.16 | *** join/#asterisk SexyKen (~ken@c-98-207-5-125.hsd1.ca.comcast.net) |
| 19:24.36 | SexyKen | Hey guys - what phones do you recommend for home use? Prefer 2 lines - and cordless... |
| 19:25.00 | thehar | hugs Katty |
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| 19:25.51 | leifmadsen | SexyKen: I prefer Polycom, but for cordless I recommend an ATA and a 6GHz DECT phone (best buy, walmart, wherever) |
| 19:26.48 | SexyKen | leifmadsen: Yea - I use Polycom IP 600 for the offices, but I think I'm gonna upgrade them, hopefully soon. Great phone - but in a dimmly lit environment, need one with the backlights. |
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| 19:27.37 | SexyKen | leifmadsen: ATA...that's a unit that will handle the SIP registration, and then send a dialtone through my home phone lines? |
| 19:29.57 | *** join/#asterisk zingbats (~androirc@cpc4-sals3-2-0-cust448.15-1.cable.virginmedia.com) |
| 19:30.31 | titter | SexyKen: The 551's are a nice model if you are looking for back lighting. |
| 19:30.43 | SexyKen | I wonder why stores like Best Buy, Frys, Microcenter, RadioShack, etc do not sell freakin' VOIP ATA's or even VOIP phones that aren't locked into a provider. |
| 19:31.30 | carrar | SexyKen, whats your budget? |
| 19:31.39 | titter | err 550* |
| 19:31.54 | carrar | You can get a nice DECT solution by Aastra or Polycom |
| 19:31.55 | carrar | :) |
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| 19:32.29 | JustinCampbell | anyone know if its possible to specify transport per peer |
| 19:32.53 | JustinCampbell | apparently it works in trixbox but not asterisk http://www.fonality.com/trixbox/forums/trixbox-forums/help/tcp-extension |
| 19:33.04 | zingbats | hey guys, i have a spa3102 and its registered to an asterisk (trixbox) extension. this works great incoming, but i guess an extension cant be used as an outbound route per se? |
| 19:33.06 | SexyKen | carrar: I dont mind spending a few hundred for something I can use for a few years. |
| 19:33.07 | JustinCampbell | sorry, freepbx |
| 19:34.08 | pabelanger | JustinCampbell: yes |
| 19:34.22 | JustinCampbell | i have it configured but its not working |
| 19:34.29 | JustinCampbell | tcpenable=yes |
| 19:34.39 | JustinCampbell | transport=udp,tcp in sip.conf |
| 19:34.45 | JustinCampbell | and transport=tcp for my peer |
| 19:34.51 | carrar | SexyKen, those are over a few hundreded |
| 19:34.55 | carrar | WAY over |
| 19:35.38 | SexyKen | carrar: What are you looking at? |
| 19:35.47 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 19:35.53 | carrar | Buy a old cell tranmistter off Ebay and hook it into asterisk :) |
| 19:36.25 | carrar | SexyKen, Aastra & Polycom make nice DECT to SIP solutions |
| 19:36.44 | carrar | but probably not pratical for home unless you have gobs of cash |
| 19:36.55 | carrar | GOBS!! |
| 19:36.59 | pabelanger | ~collectdebug |
| 19:37.00 | infobot | hmm... collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 19:37.13 | pabelanger | JustinCampbell: ^ pb a debuglog showing the issue |
| 19:37.27 | JustinCampbell | pabelanger: ok ill take a look |
| 19:37.32 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 19:37.42 | SexyKen | Where the heck can one buy a ATA locally |
| 19:37.57 | carrar | At your local VOIP store |
| 19:38.21 | SexyKen | lol |
| 19:39.36 | titter | SexyKen: Aastra has a DECT that starts at $212 ... with one handset I believe |
| 19:40.48 | *** join/#asterisk Subdolus (dexterity@creep.bur.st) |
| 19:42.03 | SexyKen | Panasonic makes one: http://www.voipsupply.com/panasonic-kx-tgp550t04 |
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| 19:43.47 | *** mode/#asterisk [+o mnicholson] by ChanServ |
| 19:46.01 | p3nguin | sexyken: Amazon.com or ebay.com ... those are probably as local as you're going to get. |
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| 19:53.44 | JustinCampbell | pabelanger: thanks for your help, it was working, previous test was fubared |
| 19:53.45 | JustinCampbell | thanks again |
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| 20:01.56 | *** join/#asterisk mac-mini (~mac-mini@unaffiliated/macmini/x-648924) |
| 20:15.43 | *** join/#asterisk jero (~jero@2607:fad8:4:0:5eff:35ff:fe08:c694) |
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| 20:21.29 | *** join/#asterisk alecdavis (~alec@202.36.76.21) |
| 20:23.17 | alecdavis | anyone having troble with 1.8.x transferring SIP calls within the same asterisk box, if so have a look at bug 18837 |
| 20:23.28 | *** join/#asterisk bjhaid (~abejide@41.206.11.4.vgccl.net) |
| 20:25.23 | bjhaid | I converted a .ogg file to .wav with and played it through asterisk, the quality/volume of the sound file reduced, please can anyone suggest how i can convert the file to maintain quality and still have a limited impact on bandwidth |
| 20:25.34 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 20:27.20 | alecdavis | Also of note, is still with 1.8.x, asterisk may segfault after an internal SIP transfer of an IAX call, bug 18781 hopefully addresses it, or seems to through 'Bad magic Number' when transferring a DAHDI call. |
| 20:28.23 | *** join/#asterisk LemensTS (~matthew@70.238.156.107) |
| 20:28.28 | ChannelZ | bjhaid: either are getting transcoded to something else even by asterisk |
| 20:29.02 | LemensTS | Can I connect to asterisk -vr as a different user besides the owner of Asterisk in Debian? |
| 20:29.22 | p3nguin | You can if you allow it. |
| 20:29.32 | bjhaid | ChannelZ: i dont understand you |
| 20:30.55 | ChannelZ | bjhaid: what did you convert the ogg to wav with, and with what settings? |
| 20:31.13 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
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| 20:31.43 | ChannelZ | bjhaid: and you said the quality was reduced - compared to what? the original ogg played back on your computer? |
| 20:32.38 | *** join/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com) |
| 20:32.57 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 20:33.08 | bjhaid | i converted it with sox and the settings are: sox a.ogg -r 8000 -c 1 -s 2.wav resample -ql |
| 20:33.28 | bjhaid | ChannelZ: yes compared to the ogg played on my computer |
| 20:33.36 | *** part/#asterisk LemensTS (~matthew@70.238.156.107) |
| 20:33.44 | p3nguin | Actually, I take that back in order to be more clear. You can certainly use another user to connect to the asterisk console by using the sudo mechanism. Connecting to the console as another user is a bit more difficult, though. |
| 20:33.48 | ChannelZ | well welcome to telephony where the bandwidths are low |
| 20:33.50 | p3nguin | wtf |
| 20:34.07 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 20:34.09 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-hjvpfrvjpfetcuuj) |
| 20:34.09 | *** mode/#asterisk [+o mnicholson] by ChanServ |
| 20:34.32 | p3nguin | Everytime he comes here, he asks a question, hangs around for about 2 or 3 minutes, then leaves. |
| 20:35.24 | *** join/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
| 20:35.25 | leifmadsen | I hate when people do that |
| 20:35.46 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 20:36.00 | ChannelZ | You answer! You answer now! |
| 20:36.02 | *** join/#asterisk Woody4286 (~Woody2143@machine76.Level3.com) |
| 20:36.05 | p3nguin | I guess it will be easier to not bother trying to answer questions from people who have that tendency. |
| 20:36.21 | bjhaid | ChannelZ: this is just a test run, I am doing this on a 10/100mbps switch in my room as a test |
| 20:36.37 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 20:36.40 | ChannelZ | still |
| 20:36.44 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-bptfjgataugdsrbo) |
| 20:36.46 | *** part/#asterisk JustinCampbell (~JustinCam@74-94-59-227-Philadelphia.hfc.comcastbusiness.net) |
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| 20:40.21 | bjhaid | ChannelZ: anything I can do to ensure I get the best quality from my call, not too concerned on bandwidth now, call quality is my interest |
| 20:43.59 | pabelanger | heh, this is odd. I'm testing cdr_adaptive_odbc.so but cannot get calldate to populate properly. I have it setup as a calldate in MySQL |
| 20:44.44 | *** join/#asterisk marienz (~marienz@freenode/staff/marienz) |
| 20:45.18 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 20:45.19 | ChannelZ | the public telephone system (digitally speaking) is 8-bit 8kHz. |
| 20:46.17 | ChannelZ | voip-to-voip there are some wideband codecs that sound better but your endpoints have to support them |
| 20:46.52 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 20:47.34 | ChannelZ | one thing to consider in your current tests too is whatever phone/softphone you are using generally has a list of preferred codecs, so it's entirely possible your device chose something like gsm which is a bit crap |
| 20:52.48 | bjhaid | ChannelZ: thanks |
| 20:53.55 | bjhaid | ChannelZ: I think i realize where the problem comes from gsm consumes lesser bandwidth but comes with poor voice quality |
| 20:57.26 | *** join/#asterisk cd101 (d07cd9d5@gateway/web/freenode/ip.208.124.217.213) |
| 21:00.47 | ChannelZ | right |
| 21:01.20 | ChannelZ | ulaw/alaw is sort-of 'best case' but 64kbps |
| 21:02.01 | bjhaid | ulaw runs at 64kbps? that's huge |
| 21:03.37 | ChannelZ | 8000 samples/sec * 8bit = 64000 bits/sec |
| 21:04.11 | *** join/#asterisk espiceland (~erin@207.98.195.107) |
| 21:04.34 | ChannelZ | 8k bytes/sec |
| 21:05.06 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 21:05.27 | bjhaid | ChannelZ: ulaw runs at 8kbps or 64kbps? |
| 21:06.10 | ChannelZ | 64kbps - little 'b', bits |
| 21:06.34 | ChannelZ | which is 8kB/sec - big 'B', bytes |
| 21:06.41 | *** join/#asterisk espiceland (~erin@207.98.195.107) |
| 21:06.48 | bjhaid | ok, how abt g729? |
| 21:07.00 | cd101 | If I have 2 servers connected via IAX2, can a call originating from server1 return to server1 if server2 says the device is busy or not available? |
| 21:07.59 | ChannelZ | I don't remember, there's a few different types, but I think it maxes out at around 12kpbs or something? |
| 21:08.15 | ChannelZ | cd101: yes |
| 21:08.17 | Kobaz | is there a way to turn down console verbosity but still log to the log files |
| 21:08.28 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-samvlcoacwqqdkhv) |
| 21:08.28 | *** mode/#asterisk [+o mnicholson] by ChanServ |
| 21:09.00 | p3nguin | Console verbosity affects logger? |
| 21:09.06 | Kobaz | yeah |
| 21:09.26 | p3nguin | Wow. All this time I've been using Asterisk and I never knew that. |
| 21:09.41 | Kobaz | yeah if you core set verbose 0 |
| 21:09.45 | cd101 | ChannelZ: is the only way to to another Dial command from server2? or is there something I can do in the dialplan on server1? |
| 21:09.52 | Kobaz | you will no longer be logging verbose messages either |
| 21:10.19 | ChannelZ | cd101: If the channel on server 2 returns 'congestion' or whatnot, you just handle that in your server 1 dialplan (it continues on) |
| 21:10.28 | Kobaz | that's killed me a few times until i realized what was going on |
| 21:10.36 | Kobaz | i'm like WHERE'S MY LOGS |
| 21:10.56 | p3nguin | I always run at verbose 0 unless I need to turn it up to see something. |
| 21:11.01 | Kobaz | you're insane |
| 21:11.06 | Kobaz | but if that works for you |
| 21:12.29 | cd101 | thank you ChannelZ, I will investigate further. |
| 21:12.40 | *** join/#asterisk Woody4286 (~Woody2143@machine76.Level3.com) |
| 21:13.07 | ChannelZ | BBL lunch time |
| 21:13.34 | Kobaz | if you manage many servers with thousands of calls per day... I don't know about you, but I need logging |
| 21:14.57 | Kobaz | if you have a home box with 5 calls a day, well.. i guess you can get away without logging |
| 21:15.05 | WIMPy | I never look at the log. Only CDRs. |
| 21:15.54 | Kobaz | i've went through this discussion before... if you can get away without logging, that's great... heh |
| 21:15.56 | p3nguin | I guess since I don't have issues/problems with Asterisk, I don't really bother reading logs. |
| 21:16.14 | Kobaz | as usual, it depends how you use asterisk |
| 21:16.22 | WIMPy | Well, I also read core dumps. |
| 21:16.30 | p3nguin | I see it differently |
| 21:16.38 | Kobaz | i used to routinely crash asterisk before i started maintaining my own branch |
| 21:16.50 | Kobaz | core dumps are great... but to see the leadup to the crash is priceless |
| 21:16.56 | p3nguin | If my Asterisk processes millions of calls and I don't have any problems, why would I care about logs? |
| 21:17.15 | Kobaz | if you're doing simple switching, i can see that generally works |
| 21:17.45 | Kobaz | if you have thousands of lines of dialplan, then logging helps a boatload in diagnosing issues |
| 21:18.28 | Kobaz | well, we've moved all the complex stuff to perl pretty much, but logging the asterisk side is still really critical for us |
| 21:19.13 | Kobaz | p3nguin: i guess it's that style thinking that gets people into trouble |
| 21:19.15 | Kobaz | 'if it processed a million calls, it's going to process a million more just fine' |
| 21:19.41 | p3nguin | If I run into problems, I'll start checking what's going on. |
| 21:19.44 | Kobaz | which is completely unguaranteed |
| 21:20.16 | _Corey_ | Disks are cheap, information is priceless :) |
| 21:20.26 | Kobaz | heh, exactly |
| 21:20.33 | Kobaz | you can't create data that was never stored |
| 21:20.43 | Kobaz | you can always filter or delete it later |
| 21:20.47 | _Corey_ | I can't tell you how many times a verbose thread trace has saved my butt |
| 21:21.07 | Kobaz | if you're routing a call from a to b... i can see where you wouldn't need logging |
| 21:21.15 | WIMPy | Except for that machine that stalled after about a minute, every time a call went on, because the disk couldn't handle the debug log. |
| 21:21.21 | Kobaz | but any kind of involved decision making based on doezens or more factors... |
| 21:21.32 | Kobaz | WIMPy: ramdisk |
| 21:21.48 | WIMPy | That happens automatically. |
| 21:22.05 | WIMPy | That's why it ran for about a minute. Then the RAM was full. |
| 21:22.17 | Kobaz | what happens automatically? |
| 21:22.21 | Kobaz | that's just disk io buffering |
| 21:22.33 | WIMPy | Then after a few minutes when it was all transferred to disk, you could cal again. |
| 21:22.34 | *** join/#asterisk bjhaid (~IceChat7@41.220.69.22) |
| 21:22.42 | WIMPy | Yes. |
| 21:23.57 | _Corey_ | Just curious... how many have experimented with CEL in 1.8? |
| 21:24.05 | Kobaz | i've looked at the code |
| 21:24.15 | Kobaz | and the data it stores... it looks cool |
| 21:24.16 | *** part/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 21:24.33 | Kobaz | but i have a large framework that i wrote for 1.6.0 that basically is CEL on crack |
| 21:24.47 | Kobaz | and we're still on 1.6.0.. so i haven't really used it yet |
| 21:25.56 | _Corey_ | I've just scratched the surface with it but like what I've seen |
| 21:27.27 | titter | Asterisk supports G726 @ 32kbps only correct? |
| 21:32.07 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:32.07 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:33.41 | pabelanger | titter: Trunk should support higher |
| 21:34.51 | titter | pabelanger: Thanks. Trying to decided between purchasing 729 or utilizing 726. |
| 21:35.33 | bjhaid | how much does it cost to own g729? |
| 21:35.37 | pabelanger | titter: With 1.10, you have greater control over codecs within Asterisk :) |
| 21:35.46 | pabelanger | $10? |
| 21:36.22 | bjhaid | is that per box or? and how is it installed? |
| 21:36.42 | titter | pabelanger: Looking forward to 1.10 ... still have 1.6.2 deployed mostly all over, with a few older 1.4 boxes around. Will probably go to 1.8.2 before 1.10 ... unless of course it's for my personal stuff :) |
| 21:36.53 | pabelanger | bjhaid: per channel |
| 21:36.57 | pabelanger | http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC |
| 21:37.13 | pabelanger | s/per channel/concurrent call |
| 21:37.46 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 21:37.49 | titter | bjhaid: You can find "test" versions of G729 via a Google search ... however, while they work, I did run into some problems with their quality. |
| 21:38.22 | bjhaid | ok |
| 21:45.38 | bjhaid | @pabelanger: g729 codec has to be supported @ my end and my ITSP's end for it to work right? |
| 21:58.53 | *** join/#asterisk Scorpio2007 (~Scorpio20@jose-tc.ctc.biz) |
| 21:59.15 | Scorpio2007 | anyone know of a good open srouce billing solution for asterisk? i have all my call records in a cdr psql database .. |
| 22:03.21 | carrar | woot shuttle now in space! |
| 22:08.01 | titter | carrar: Damn clouds ... I couldn't see it :( |
| 22:08.08 | WIMPy | Can someone enlighten me about the concept of ANI in Asterisk? Is it supposed to always be set or only if different from the Caller ID? |
| 22:09.03 | carrar | it's ZOOMin |
| 22:11.14 | p3nguin | 13,000 MPH |
| 22:12.06 | titter | Yup. I live on the opposite coast of Florida ... so you can actually see it when it launches. |
| 22:12.08 | esterpader | Hi! can i ask something what is the dependencies of BETTER_BACKTRACES, i can't find it on the selection, binutils-dev is already installed |
| 22:12.20 | esterpader | SVN 1.6 branches |
| 22:13.28 | titter | When landing in Cape Canaveral depending on the flight path, it ends up going right over my city while still breaking the sound barrier. Sonic booms are cool. |
| 22:17.35 | p3nguin | Discovery has 143 million miles on its odometer. |
| 22:17.36 | lordvadr | WIMPy: In theory, ANI is the "billing phone number", whereas callerid is DID the call came from. |
| 22:18.15 | lordvadr | So if you were to pick up a phone and call your phone company, they'd care about the ANI, not Jim in Accounting's phone number. |
| 22:18.33 | lordvadr | none of that works in sip, but it can in the pstn world |
| 22:18.56 | p3nguin | Calls don't come from DIDs. |
| 22:19.20 | WIMPy | Many things don't work in SIP. I'm just interested in what it should be like in Asterisk. |
| 22:19.52 | lordvadr | p3nguin: sure they do, or at least the "from" indication for whatever technology you're using (caller id in pstn, sip from header, etc). |
| 22:20.18 | p3nguin | DID = direct INWARD dialing |
| 22:20.48 | lordvadr | penguin: and when you call out, you put that DID in the "from" of whatever technology so that they can call you back |
| 22:21.05 | p3nguin | No, I put a phone number in it. |
| 22:21.31 | lordvadr | agreed, but that's symantics. When I order DID's from my carrier, I get phone numbers. |
| 22:22.21 | lordvadr | WIMPy: It's not something to worry about unless you're getting PRI or SS7 service, and my PRI's done even deliver ANI anyway so I can't tell you for sure. Are you in the US? |
| 22:22.27 | WIMPy | Ok, apart from the terminology thing, do you know it it should always be set if possible or not? |
| 22:22.49 | lordvadr | WIMPy: As a function of completness, yes, will it do you any damn good, no. |
| 22:23.03 | WIMPy | I want to make sure it is set sorrectly for local phones. |
| 22:23.14 | lordvadr | what do you mean local phones? |
| 22:23.36 | WIMPy | I'm hacking round a channel driver. |
| 22:24.37 | lordvadr | Not sure what ANI and CDR has to do with a channel driver, unless you're working on a pstn driver. |
| 22:24.58 | *** part/#asterisk jero (~jero@2607:fad8:4:0:5eff:35ff:fe08:c694) |
| 22:25.03 | WIMPy | It sets the information. |
| 22:25.18 | lordvadr | then yes, if you have it, set it |
| 22:25.57 | WIMPy | Ok, so set both and in the other direction filter one if both are set to the same value. |
| 22:26.28 | lordvadr | but that's really a carrier thing. Call mci, 18004444444, and you'll get your ANI. We don't set it on the outbound, but our carrier sets it. Sort of how you can't *69 911 |
| 22:27.04 | *** join/#asterisk jero (~jero@2607:fad8:4:0:5eff:35ff:fe08:c694) |
| 22:27.06 | lordvadr | WIMPy: might as well leave them both on in the "other direction" if the technology supports it |
| 22:27.11 | jero | hi |
| 22:27.16 | WIMPy | Well, ideally Asterisk should behave just like any other switch. |
| 22:27.44 | lordvadr | define "switch"? I have to get into that argument all the time with my phone guys. |
| 22:27.48 | WIMPy | That only makes sense if they are different. |
| 22:28.03 | lordvadr | Carrier switch is one thing, office PBX is another |
| 22:28.05 | WIMPy | Anything that switches calls. |
| 22:28.27 | WIMPy | Shouldn't be much of a difference, hopefully. |
| 22:28.41 | lordvadr | well, their behavior is all over the map |
| 22:28.48 | jero | is it possible to have several phones, some with g722, some without (running 1.6.2.6) ? If one of my phones with allow=g722,ulaw calls another with allow=ulaw only, i get no audio. |
| 22:30.14 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 22:30.14 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 22:30.29 | p3nguin | jero: Configure it to use only the codec you want the phone to use. Asterisk will transcode to any phone which is not using the same codec. |
| 22:31.06 | jero | p3nguin: so only g722 for capable ones, and only ulaw for others? thanks |
| 22:31.43 | p3nguin | jero: If you want a phone to only use ulaw, disallow all and then allow only ulaw. Same for every codec. |
| 22:32.06 | jero | why is the negotiation not working in my case ? |
| 22:32.25 | p3nguin | Show me the sip debug and maybe I'll be able to figure out why. |
| 22:34.44 | WIMPy | Hmm. Why does chan_sip set the caller ID as unscreened, even if it came from sip.conf? |
| 22:35.37 | jero | p3nguin: here is my sip debug: http://pastebin.com/9iqespmk |
| 22:36.02 | jero | p3nguin: in the log, 125(g722,ulaw) calls 115(ulaw) |
| 22:36.30 | WIMPy | thinks that should be either network provided or verified and passed. |
| 22:38.18 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 22:38.18 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 22:38.45 | p3nguin | I only see one codec negotiation in that debug, and it says us - 0x1804 (ulaw|g726|g722), peer - audio=0x1000 (g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x1000 (g722) |
| 22:38.58 | p3nguin | g722 was the agreed-upon codec to use for that call. |
| 22:41.12 | *** join/#asterisk thews (~thews@24-119-172-234.cpe.cableone.net) |
| 22:45.01 | p3nguin | I'm not seeing a problem yet. Is there more? |
| 22:45.24 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 22:45.24 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 22:46.02 | jero | p3nguin: here is the two-way debug log: http://pastebin.com/2DLyiB63 |
| 22:46.44 | p3nguin | That one looks like ulaw was used in all cases. |
| 22:47.10 | jero | yeah, but I get no audio at all : ) this is strange to me. |
| 22:47.24 | jero | anyway, i'll try to dig deeper |
| 22:47.45 | p3nguin | Are any of these phones having to pass through a NAT to make calls? |
| 22:49.12 | jero | p3nguin: not at all, all sit on the same LAN |
| 22:49.22 | jero | p3nguin: and all are set with directmedia=nonat |
| 22:49.53 | jero | p3nguin: localnet configuration is properly set |
| 22:50.14 | p3nguin | If there is no NAT involved, localnet shouldn't be configured to anything. |
| 22:50.27 | p3nguin | Is Asterisk behind a NAT? |
| 22:50.53 | jero | p3nguin: yes, there are some other phones outside, accessing asterisk through a NAT |
| 22:51.02 | jero | they work fine though |
| 22:51.04 | p3nguin | Okay, so localnet is needed for Asterisk, but it's not necessarily being used for these calls because the phones are on the LAN with Asterisk. |
| 22:51.13 | jero | p3nguin: exactly |
| 22:51.20 | p3nguin | I'm on the same page as you, now, I promise. |
| 22:51.44 | jero | i'm clueless :) |
| 22:52.00 | p3nguin | Are you keeping Asterisk in the media path, or do you allow reinvites between these phones? |
| 22:52.17 | jero | p3nguin: i'm allowing it, directmedia=nonat |
| 22:52.28 | p3nguin | Do you have any Dial() options? |
| 22:54.18 | p3nguin | Get Asterisk in the path between these phones. See if you have audio while it's in the path. |
| 22:55.11 | jero | p3nguin: yep, when asterisk is in the path, it works |
| 22:55.12 | WIMPy | Maybe you need to disable some security setting on the phone to allow directmedia? |
| 22:55.29 | jero | WIMPy: well, it works fine when setting both phones to ulaw |
| 22:55.48 | jero | i think i'll juste leave asterisk in the path for my g722 template. |
| 22:56.00 | WIMPy | Oh, thought it was the other way round. |
| 22:56.59 | jero | i'll also try with a more recent version of both the phone's firmware and asterisk |
| 23:07.42 | jero | thanks for the help ! |
| 23:11.14 | *** join/#asterisk arthurh (~arthurh@70.56.139.60) |
| 23:19.07 | *** join/#asterisk geoffmcc (~geoffmcc@cpe-76-180-197-68.buffalo.res.rr.com) |
| 23:19.43 | *** join/#asterisk luckman212 (~irc@pool-173-77-253-145.nycmny.fios.verizon.net) |
| 23:21.34 | geoffmcc | anyone out there using ipkall and know if its down. I have set this up before and i cant seem to get it to work now |
| 23:25.07 | WIMPy | lordvadr: I think I should try what Asterisk makes of it, but acutally I have one ITSP that provides ANI. |
| 23:29.32 | WIMPy | Asterisk ignores it and puts the CallerID in to ANI as well. |
| 23:39.48 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 23:40.07 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 23:44.57 | *** join/#asterisk leroybuckingham (637c543e@gateway/web/freenode/ip.99.124.84.62) |
| 23:46.28 | leroybuckingham | Hi fellas, I'm having an issue with digium cards on several installations running on asterisk 1.6.2.13. They're making rather loud popping and clicking sounds on the remote end. |
| 23:47.06 | leroybuckingham | I noticed this patch pertaning to asterisk 1.4, I'm wondering if a similiar bug could be affecting my version https://issues.asterisk.org/view.php?id=17529 |
| 23:47.58 | tzanger | what kind of cards? |
| 23:50.07 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 23:50.31 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 23:50.46 | *** part/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 23:51.56 | leroybuckingham | Sorry I'm trying to get in to find out, unfortunately i'm not on site right now |
| 23:52.46 | tzanger | is it a digital card or an analog card |
| 23:52.57 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 23:56.56 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 23:59.31 | leroybuckingham | Well, I'll have to check back tomorrow, my contact must have left for the day. Thank you though. |