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| 01:19.55 | \DSAFEW\ | !newbook |
| 01:20.05 | \DSAFEW\ | !book |
| 01:20.28 | \DSAFEW\ | what are those numbers for foreign exchanges again? lost all my bookmarks here |
| 01:20.34 | WIMPy | ~thebook |
| 01:20.35 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 01:20.48 | \DSAFEW\ | ty |
| 01:21.02 | \DSAFEW\ | ~newbook |
| 01:21.02 | infobot | Please see ~thebook for more information about Asterisk: The Definitive Guide |
| 01:21.56 | \DSAFEW\ | that last link is including the comma for me :\ |
| 01:22.20 | *** join/#asterisk cusco (~tralala@a89-152-96-250.cpe.netcabo.pt) |
| 01:22.35 | cusco | hai |
| 01:22.51 | WIMPy | Hmm, yes, that an unfortunate definition. |
| 01:22.52 | cusco | make compiles file, but make install... is not being able to download sound files? |
| 01:23.19 | cusco | http://paste.debian.net/118352/ |
| 01:23.32 | \DSAFEW\ | cusco, which OS/distro? |
| 01:24.22 | cusco | debian |
| 01:24.24 | cusco | why? |
| 01:24.38 | cusco | normally it would wget them files |
| 01:24.45 | cusco | but seems that its not wgetting them |
| 01:24.52 | cusco | and tar gunzip fails... |
| 01:28.46 | cusco | :| |
| 01:36.18 | \DSAFEW\ | IIRC debian uses apt to install things, why are you using the generic tarball? |
| 01:37.13 | cusco | why shouldn't I? |
| 01:37.29 | cusco | \DSAFEW\: tell me |
| 01:37.40 | cusco | and how does that relate to my question? |
| 01:38.42 | deltaray | For some reason, Record(asterisk-recording-test.ulaw||2) isn't working on 1.6. It says file.c:1160 ast_writefile: No such format 'ulaw||2' |
| 01:38.53 | deltaray | Its like its parsing the config wrong. |
| 01:38.57 | \DSAFEW\ | cusco, perhaps you should take this to #debian where they can tell you specifics on installing software for your version with their version of asterisk, not saying you can't make this work, but it seems counterproductive to not use a package manager |
| 01:39.08 | deltaray | If I leave off the ||2 part, it works. |
| 01:39.18 | deltaray | I'm sorry, if I leave off ulaw||2 it works. |
| 01:39.54 | cusco | \DSAFEW\: I don't understand why I should take this to debian. My question is not debian related. |
| 01:40.05 | cusco | its not counterproductive to not use apt in some cases |
| 01:40.08 | WIMPy | deltaray: It's time to get rid of that 1.2 syntax and use , instead of |. |
| 01:40.21 | deltaray | Ahh |
| 01:40.30 | deltaray | The docs don't have examples with , |
| 01:40.31 | cusco | don't get me wrong, debian is my gnu distro of choise due to the DFSG |
| 01:40.37 | deltaray | They use | |
| 01:40.43 | cusco | but there is certain stuff, that you don't install, you compile |
| 01:40.56 | cusco | maybe becuase you don't want unused junk like chan skinny and jingle |
| 01:41.03 | WIMPy | deltaray: It should be , since 1.4. |
| 01:41.16 | cusco | or maybe because you want to compile with snmp |
| 01:41.28 | cusco | \DSAFEW\: my question is not debian related |
| 01:42.00 | deltaray | WIMPy, thanks. That fixed my problem. |
| 01:42.20 | \DSAFEW\ | cusco, well, 1.8 is out, have you tried that? |
| 01:42.25 | cusco | yes I have |
| 01:42.33 | cusco | I'm testing it at home |
| 01:42.50 | cusco | \DSAFEW\: in the call centre I work for, we use asterisk with specific configuration that 1.8 changes |
| 01:43.18 | cusco | for one the realtime mysql configuration, asterisk 1.8 writes the time colum as datetime instead of timestamp |
| 01:43.33 | \DSAFEW\ | ahh, so you need 1.6 then, and debian doesn't do 1.6? |
| 01:43.52 | cusco | \DSAFEW\: why are we talking about debian? I fail to understand that |
| 01:45.24 | cusco | \DSAFEW\: there are compile options that I chose to apply that debian maitainers didn't. Further more, asterisk compiled in my machine with optimize flags, gets optimized to my hardware... |
| 01:45.50 | cusco | This conversation doesn't seem to be productive |
| 01:46.50 | cusco | don't take me wrong tho, for most of stuff I like debian defaults |
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| 03:00.09 | joobie | hey guys.. im setting up a sip provider and my end is not sending a register, just invites |
| 03:00.15 | joobie | any reason why this would be? |
| 03:00.31 | joobie | it is type=friend, though i have others setup this way also (not using peer) |
| 03:00.36 | joobie | .. which i presume work ok |
| 03:00.38 | joobie | this is with 1.4 |
| 03:05.50 | WIMPy | It will only register if you configure a register line in the general section. |
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| 03:17.25 | eyeballer | evening all.. |
| 03:18.10 | eyeballer | im looking for some help installing asterisk 1.8 on an asus rt16 running tomato-usb.. i've got it installed via ipkg.. but i'm lost on how to edit the jabber.conf and gtalk.conf files.. |
| 03:20.00 | maxagaz | hi |
| 03:21.13 | eyeballer | hi maxagaz |
| 03:21.30 | maxagaz | there are two kind of analogic cards, the first is used to connect analogic phones to server, the second is for connecting analogic lines to the server, am I right ? |
| 03:22.10 | WIMPy | correct |
| 03:22.27 | maxagaz | are there cards that can do both ? |
| 03:22.27 | WIMPy | Well, iy's usually different modules on the card. |
| 03:22.42 | WIMPy | Not on the same port. |
| 03:22.49 | WIMPy | Analog is evil. |
| 03:23.12 | maxagaz | how do I know that my card is the first or the second kind ? |
| 03:23.49 | WIMPy | Hmm. Probably by looking at it for clues. |
| 03:24.34 | WIMPy | Some ar color-coded. |
| 03:24.35 | maxagaz | it's a digium card, AEX800 |
| 03:25.03 | WIMPy | The manual schould explain the ports. |
| 03:25.14 | maxagaz | there's no manual |
| 03:25.58 | WIMPy | It's at digium.com. |
| 03:26.59 | maxagaz | is there a standard name for those two kind of cards ? |
| 03:27.16 | maxagaz | I all call them analogic cards |
| 03:27.35 | WIMPy | FXO and FXS |
| 03:35.40 | maxagaz | WIMPy: which one is for what ? |
| 03:35.54 | WIMPy | I always mix that up. |
| 03:38.24 | maxagaz | WIMPy: on digium's website (http://www.digium.com/en/products/analog), I can see that all card are FXO/FXS compliant |
| 03:38.47 | WIMPy | That was the part about the modules. |
| 03:39.17 | maxagaz | WIMPy: what do you mean ? |
| 03:40.04 | joobie | thanks WIMPy |
| 03:40.12 | WIMPy | On many cards you have to install modules per port that will be either FXO or FXS. |
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| 04:56.40 | irroot | Top 'o the morn |
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| 06:03.31 | deltaray | Not sure if anyone is up. |
| 06:03.41 | deltaray | I'm trying to get my first IAX2 connection setup. |
| 06:04.16 | deltaray | Following instructions here: http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers and a few other documents. |
| 06:04.48 | deltaray | But when I place a call, I'm getting: Call rejected by a.b.c.d: No such context/extension |
| 06:04.57 | deltaray | a.b.c.d being the Ip of the server I'm trying to connect with. |
| 06:05.18 | deltaray | Ideas? |
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| 06:09.19 | WIMPy | Wrong context? |
| 06:09.37 | WIMPy | Anyway, it can't find that extension. |
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| 06:16.13 | schmidts | good morning |
| 06:16.27 | kleszcz | morning |
| 06:16.29 | WIMPy | Moinsen |
| 06:17.41 | schmidts | everybody had a nice weekend and is happy that its monday again? |
| 06:20.14 | wdoekes2 | haha, good morning :) |
| 06:20.15 | deltaray | WIMPy, Do the contexts on both servers have to match up? ie, my sip.conf entries at home are in context home, but on the remote side I have the iax user config set to go through a different context. |
| 06:20.32 | WIMPy | No |
| 06:20.34 | deltaray | But that remote context is setup right |
| 06:21.08 | WIMPy | But you need to sepcify the context that holds your extensions. Either in the peer definition or in the dial on the other side. |
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| 06:22.22 | deltaray | Ok, I've done that. |
| 06:22.39 | deltaray | Now I'm getting "__auto_congest: Auto-congesting call due to slow response" on the local side's asterisk console when I try to make a call. |
| 06:22.51 | deltaray | It rings back once before saying that. |
| 06:23.27 | WIMPy | That means your network connection is too lagged. |
| 06:24.32 | deltaray | Well, I kinda doubt its really lagged enough to be a problem, I'm wondering if my DSL modem has some security detection setting that is affecting IAX packets. |
| 06:24.48 | deltaray | I've opened up the ports and forwarded them, so the firewalls should be ready. |
| 06:25.29 | deltaray | If I'm seeing 42ms response on pings to the remote host, is that any indication that it would be fast enough or not? |
| 06:25.47 | WIMPy | That's fast enough. |
| 06:25.52 | WIMPy | Do you have packet loss? |
| 06:25.57 | deltaray | no |
| 06:26.38 | schmidts | question to you folks ;) have anyone has seen this or even used it: mysql-mmm.org |
| 06:26.39 | deltaray | Just sent 100 packets and 0% loss. |
| 06:27.25 | schmidts | its a multi master HA setup for mysql which uses replication and different roles. The setup itself looks very cool with IP takeover for each role and so on but i have a logic problem with this ;) |
| 06:28.12 | robbie` | schmidts: whats that? |
| 06:29.59 | schmidts | robbie` this is a collection of perl scripts which monitors which mysql node (master or slave) is alive and change the ip to a living instance if one of the hosts dies |
| 06:30.38 | schmidts | i really like the idea about this, cause you can have for example 2 masters for write operations and 2 slaves only for reading operations |
| 06:31.59 | deltaray | schmidts, are you on the right channel? This is for asterisk, not mysql. Unless you're talking about integrating the two. |
| 06:32.38 | schmidts | deltaray thx ;) i know this is asterisk i only asked if someone used this allready and yes i want to build a mysql cluster for my 20+ asterisk servers ;) |
| 06:33.14 | robbie` | schmidts: whats your problem with the logic? |
| 06:33.40 | robbie` | i haven't used it, but i am interesting in making asterisk highly available |
| 06:33.55 | deltaray | schmidts, oh sorry. ;-) |
| 06:33.59 | robbie` | interested* |
| 06:34.17 | robbie` | schmidts: to make asterisk highly available are you using heartbeat/pacemaker? |
| 06:34.49 | schmidts | robbie` if you take a look at the installation guide on the page, it says you should enable the master->slave sync on db2,db3 and db4 to the db1 but my problem is if db1 dies only db2 stays available cause db3 and db4 (the slaves) failed syncing |
| 06:35.21 | robbie` | not sure |
| 06:35.51 | schmidts | robbie` not for asterisk, i have build a setup for my sip proxy (kamailio) using corosync if one instance dies the ip switchover to a duplicate server on another host |
| 06:36.30 | WIMPy | There sometime are logic issues with fail over solutions. |
| 06:36.39 | schmidts | my problem with Mysql HA cluster setup is, the normal mysql cluster is around 3 times slower in Reading queries than a normal MyIsam table is, and it gets even slower if more instances queries the same tables |
| 06:36.40 | WIMPy | Or even always? |
| 06:36.53 | schmidts | WIMPy thx that really helped me out :D |
| 06:37.39 | schmidts | this mmm cluster will do nearly the same as mysql-cluster itself does but uses the normal myIsam engine with replication and not the slow NDB engine |
| 06:37.47 | WIMPy | hat that discussion about network block devices. |
| 06:38.03 | deltaray | WIMPy, any other ideas for my problem besides potential packet loss? |
| 06:38.52 | WIMPy | deltaray: I'm not sure if a bad timing source can produce such a thing. |
| 06:38.52 | schmidts | deltaray how do you tested the packet loss? |
| 06:39.30 | deltaray | ping -i0.2 -c 100 serverip |
| 06:39.52 | WIMPy | 0.02 would be realistic. |
| 06:40.23 | schmidts | but a bad timing source can also produce such a problem, atleast with iax |
| 06:40.27 | deltaray | Ok, same result with 0.02 |
| 06:40.40 | deltaray | What do you mean by timing source? The system clock? |
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| 06:41.10 | WIMPy | Asterisks internal timind. |
| 06:41.23 | WIMPy | Do you have any dahdi hardware? |
| 06:41.30 | schmidts | deltaray not really the system clock, but its like this. you need a timing source to synchronize both streams together |
| 06:41.46 | schmidts | deltaray maybe just try it with sip if you have the same problems |
| 06:42.31 | deltaray | schmidts, I was trying to avoid sip due to NAT issues. |
| 06:43.19 | schmidts | deltaray just try it to check if your timing is ok, if you have the same problems then its something on the way between both servers, if not then you possible have a timing problem ;) |
| 06:43.48 | deltaray | WIMPy, I have a Rhino RCB4FXO on the remote side if that's what you mean. |
| 06:43.53 | WIMPy | Try 'timing test'. |
| 06:44.16 | WIMPy | The other box has no hardware timing? |
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| 06:45.33 | WIMPy | Mor importantly, It's not some virtual machine or something, is it? |
| 06:46.06 | deltaray | No |
| 06:46.22 | deltaray | Both are physical. |
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| 06:46.40 | schmidts | deltaray btw which asterisk version do you use? |
| 06:46.43 | deltaray | Does Asterisk 1.4 have the timing test command? |
| 06:47.10 | schmidts | the timing test command comes with dadhi |
| 06:47.11 | WIMPy | I don't think so. |
| 06:47.30 | WIMPy | Do you have dahdi_dummy loaded on the machine without hardware? |
| 06:47.31 | deltaray | local is 1.6.2.7-1ubuntu1.1, remote is 1.4.18 |
| 06:48.53 | deltaray | Is dahdi_dumy a kernel module or an asterisk module? |
| 06:49.04 | WIMPy | kernel |
| 06:49.39 | deltaray | no I don't. |
| 06:50.14 | deltaray | I do have a couple of voip phones connected to this local asterisk server and can make calls between them. |
| 06:50.18 | WIMPy | Wasn't that required <1.8? |
| 06:50.33 | WIMPy | For IAX that is. |
| 06:50.43 | deltaray | Ok, let me look into that. |
| 06:50.44 | kaldemar | only for IAX2 trunking |
| 06:51.17 | deltaray | I'm not doing trunking for what its worth. |
| 06:52.09 | WIMPy | It you're passing multiple calls between those boxes, you might want to. |
| 06:52.36 | WIMPy | But it's worth loading dahdi_dummy anyway. |
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| 06:57.53 | deltaray | WIMPy, perhaps the problem is that I'm missing /dev/dahdi? |
| 06:58.02 | deltaray | the kernel module won't load. Hmmmmmm |
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| 07:11.38 | deltaray | Don't you just love recompiling kernel modules at 3:11am? :-/ |
| 07:14.32 | WIMPy | Only on slow machines :-) |
| 07:15.44 | deltaray | Ok, well that didn't fix the problem. |
| 07:15.51 | deltaray | dahdi_dummy is loaded. |
| 07:16.24 | WIMPy | Is chan_dahdi loaded so that Asterisk can make use of it? |
| 07:16.37 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 07:17.33 | deltaray | yes, but its use count is 0. |
| 07:19.07 | WIMPy | Look at res_timing_dahdi. |
| 07:19.36 | deltaray | yeah, that's loaded too |
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| 07:19.46 | deltaray | but use count is 0 |
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| 07:20.24 | WIMPy | Hmm, I think that one should have a use count. |
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| 07:24.40 | deltaray | Well, I gotta go to sleep. Thanks anyways for the help, maybe I'll be back tomorrow. |
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| 10:23.52 | irroot | $@#$ snom !!! its a mission to move to XML i know im a late adopter and this is why ;) |
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| 10:33.06 | petn-randall | Hi, I'm using asterisk 1.6.2.9 and I'm having trouble with app_fax ("Transmission error"). Is there a possibility to set the debug level selectively for this app? |
| 10:33.59 | petn-randall | I don't want my logs to be swamped, as we have a lot of SIP calls, and occassionally receive a fax every few days. |
| 10:36.29 | coppice | there is a debug option for the app_fax command lines. I can't remember the exact syntax, though. it should tell you if you query the syntax |
| 10:42.28 | petn-randall | from what "core show application ReceiveFAX" tells me from the Asterisk CLI, it only takes a filename and optionally the 'c' argument as parameters. 'c' makes it behave as the calling machine. |
| 10:45.41 | petn-randall | It also sets a few variables, but I can't check them after that in the dialplan because it doesn't continue with the dialplan. |
| 10:45.42 | coppice | it used to take another parameter for the debug option. maybe it was changed to a variable. there should definitely be a debug option that gets you a detailed log of a fax call |
| 10:48.51 | petn-randall | Oh right, app_fax has the 'd'ebug option, but only in asterisk 1.8. I have 1.6 :( |
| 10:50.09 | petn-randall | Is there a possibility to 1) set the debug level within a dialplan? |
| 10:50.13 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 10:50.47 | petn-randall | or 2) let the dialplan continue even if an application bails out? |
| 11:01.01 | *** join/#asterisk zkn (~zkn@195.222.14.202) |
| 11:02.39 | zkn | howdy, how should I go about analysing a problem with 1.8.4 that causes astdb to become "corrupt" from time to time ? |
| 11:04.32 | zkn | it's been 3 times now in 2 weeks that I have had to rename the old astdb so I could start up Asterisk which otherwise just crashes during startup... |
| 11:07.14 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 11:12.39 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 11:22.42 | *** join/#asterisk silkcut (~silkcut@sto93-2-82-228-142-248.fbx.proxad.net) |
| 11:23.56 | schmidts | zkn does your asterisk segfault very often? |
| 11:24.24 | zkn | started only recently after I changed to 1.8.4 |
| 11:24.53 | zkn | and every time it has happened now, the only cure has been renaming the old astdb |
| 11:26.11 | schmidts | znk a possible reason for this could be that asterisk dies unnormal while syncing the astdb to disk, in 1.8 this doesnt happens after every astdb, only every 500ms |
| 11:26.31 | schmidts | s/every astdb/every astdbaction/ |
| 11:31.20 | zkn | yea..for some reason it sometimes happens that Asterisk becomes unresponsive...it is not allowing to call out or accept inbound calls, and when i run whatever command in CLI, i get no output/response, usually I then restart the Asterisk server from init.d after which I can be sure that I will get segmentation fault |
| 11:34.35 | zkn | maybe it is something in Linux that causes this? I'm running OpenSUSE 11.1 32bit since version 1.6 smth |
| 11:34.56 | zkn | since 1.6 of Asterisk, i mean |
| 11:36.53 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 11:36.58 | zkn | and even though I have also ran into similar issues with older versions of Asterisk that have required me to restart Asterisk from init.d, Segmentation Fault has never been the outcome of that |
| 11:37.37 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 11:39.36 | zkn | for some reason I think that 1.8.4.1 wouldn't be any different |
| 11:39.45 | schmidts | zkn this sounds like a possible reason ;) |
| 11:40.10 | zkn | what exactly? |
| 11:40.24 | zkn | that I restart from init.d script? |
| 11:40.48 | schmidts | a normal restart should do anything but if its deadlocked, maybe your init script kills asterisk |
| 11:42.03 | zkn | I guess so.. i understand that I am killing the problem based on the symptom, but how can I find out the root of the problem - what causes Asterisk to become deadlocked? |
| 11:47.24 | zkn | while Asterisk had become unresponsive today, i also ran "lsof | grep asterisk | wc -l" which gave me over a thousand files/sockets being in use by asterisk... maybe that is not the perfect command to use as it's probably lacking some crucial paramters and I also have other stuff on the server also running in group asterisk as user asterisk, so the count is not that exact |
| 11:48.27 | *** part/#asterisk sunfone (~jeff@c-67-184-36-158.hsd1.il.comcast.net) |
| 11:48.46 | zkn | yeah, not that I check the same command, i still get over a thousand occurences |
| 11:48.52 | zkn | not=now |
| 11:50.47 | schmidts | zkn give me a moment there is a document how to trace such problems |
| 11:51.05 | zkn | aah..that'd be great |
| 11:52.10 | schmidts | zkn https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace the part about Getting Information For A Deadlock |
| 12:01.21 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:01.21 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:02.52 | *** join/#asterisk phoenixsampras (~phoenixsa@static-190-181-38-119.acelerate.net) |
| 12:03.23 | *** join/#asterisk Exten (~Exten@mail.gdc.co.il) |
| 12:11.22 | zkn | i don't get BETTER_BACKTRACES option in menuselect |
| 12:13.21 | zkn | 1.8.4 |
| 12:14.42 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 12:19.17 | *** join/#asterisk Exten (~Exten@mail.gdc.co.il) |
| 12:22.33 | leifmadsen | zkn: you need too do ./configure --enable-dev-mode to have that option |
| 12:22.42 | leifmadsen | and the dependencies |
| 12:23.13 | zkn | ok, thanks |
| 12:26.55 | zkn | what exactly is DLADDR ? |
| 12:27.36 | zkn | is there a universal name i can search it by? |
| 12:28.34 | leifmadsen | has no idea |
| 12:28.38 | leifmadsen | never enabled it |
| 12:29.00 | zkn | couldn't find it in yast |
| 12:29.01 | schmidts | zkn its only a guess but maybe this means DownLoadADDress? |
| 12:30.31 | *** join/#asterisk Denial (~Denial@drgi.co.uk) |
| 12:32.23 | zkn | what can DETECT_DEADLOCKS option give me ? |
| 12:34.03 | kaldemar | zkn: see CHANGES |
| 12:36.43 | zkn | ok, got it, so it seems like a usefull option to have enabled |
| 12:42.08 | zkn | damn, this CHANGES doc is a very good source of usefull information :) |
| 12:42.20 | leifmadsen | who'dathunkit! |
| 12:42.36 | zkn | :) |
| 12:43.29 | zkn | often times the noise coming form all the information available is just so intense that I don't bother.. i guess it's my own funeral |
| 12:51.38 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 12:55.42 | *** join/#asterisk stix (~stix@firewall.o4.dk) |
| 12:57.39 | *** join/#asterisk doolittlework (~d@41-134-22-10.dsl.mweb.co.za) |
| 12:57.46 | doolittlework | Hi guys |
| 12:59.09 | doolittlework | If i want to become a sip provider for farms on a wifi network where our telco refuses to lay cable is asterisk the best app for the kob or is there any other app you guys know about that i can read up on? |
| 13:00.11 | leifmadsen | doolittlework: asterisk will likely do what you want, or you could potentially look at a SIP proxy like Kamaillo or OpenSIPS |
| 13:01.12 | leifmadsen | asterisk will integrate with both the PSTN and SIP end points |
| 13:01.24 | doolittlework | thanks |
| 13:01.38 | irroot | indeed having a mix is valid option too have users load balance / fail over to * |
| 13:04.54 | *** join/#asterisk bl0b` (~vsfdfsdfs@host98-89-dynamic.53-79-r.retail.telecomitalia.it) |
| 13:04.55 | bl0b` | hello |
| 13:06.45 | bl0b` | Someone know if is possibile to record a VideoCall ? |
| 13:07.17 | doolittlework | press record on the VCR |
| 13:07.23 | bl0b` | ahahahahah |
| 13:07.25 | bl0b` | :P |
| 13:07.47 | doolittlework | lol sry could not resist, sorry i dont know how |
| 13:07.56 | bl0b` | eheheheh np :P |
| 13:08.27 | leifmadsen | bl0b`: you can record H263 and H264 |
| 13:09.03 | bl0b` | leifmadsen: the question is how... MixMonitor doesnt support it |
| 13:11.12 | leifmadsen | bl0b`: if it gives a warning about unable to support H263 or H264 then MixMonitor() may not have the capability to record video calls. I know Record() can, but that doesn't record an in-progress call |
| 13:11.20 | leifmadsen | it just might not be possible with asterisk |
| 13:11.26 | leifmadsen | (without writing code) |
| 13:11.46 | tzafrir_laptop | mplayer? (RTFM about the command line) |
| 13:11.59 | tzafrir_laptop | has no idea what M exactly |
| 13:13.31 | bl0b` | leifmadsen: I'vnt checked if it give some warning, but for sure MixMonitor (I'm googling from three weeks) record only audio. Record() function do it, but like u'v already told, it didnt work for in-progress call, so it is a lil useless.. (ok it's not useless, I need it for the voicemail :P) |
| 13:14.00 | leifmadsen | ya Record() is more for recording prompts, not calls |
| 13:16.23 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 13:16.47 | bl0b` | tzafrir_laptop: I need to record an inprogress call, not play an already recorded stream :P so, how may it can help me? |
| 13:24.53 | *** join/#asterisk celord (~celord@201.198.102.2) |
| 13:25.01 | *** part/#asterisk dachary (~loic@freenode/sponsor/dachary) |
| 13:33.13 | irroot | been screwing round with TLS want to stop fraud well not spread my drivil openly accross the net |
| 13:33.44 | irroot | dont see any CRL checks in asterisk sip.conf any one know the status of this ?? |
| 13:38.23 | *** join/#asterisk fofware (~fabian@wdctf.siup.gov.ar) |
| 13:41.16 | zkn | core restart when convenient and Segmentation fault, but now i have compiled with all the necessary flags so the next time i should get some core files to trace |
| 13:43.26 | *** join/#asterisk Denial (~Denial@drgi.co.uk) |
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| 14:21.04 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
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| 14:37.35 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 14:39.23 | *** join/#asterisk voiptelecom (~Sebastien@9-voip-telecom.10-cust.tasfrance.com) |
| 14:39.28 | voiptelecom | hello wolrd ! |
| 14:39.54 | voiptelecom | any idea for this message in log : Still have a callno |
| 14:40.21 | voiptelecom | exaclty : NOTICE[24374] chan_iax2.c: Still have a callno... |
| 14:41.15 | voiptelecom | sometimes it's coming, i don't see probleme but i would like ti know what is means |
| 14:42.51 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 14:47.58 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 14:48.48 | paulc | voiptelecom: Have you seen http://www.mail-archive.com/asterisk-users@lists.digium.com/msg209155.html |
| 14:51.23 | voiptelecom | yes i read it thx |
| 14:51.45 | *** join/#asterisk resmo (~msr@office.adfinis.com) |
| 14:51.47 | resmo | hi |
| 14:53.38 | resmo | I have a few phones and wondering if I really have to define every phone (again?) in extension.conf to dial each other ? |
| 14:54.06 | irroot | resmo no that hurts |
| 14:54.18 | irroot | use matching or some other method |
| 14:54.19 | bl0b` | About my issue... (record inprogress videocall) maybe.. Monitor() can record the h263\h264 stream ? |
| 14:54.33 | irroot | resmo exten => _XXXX,1, ... |
| 14:54.39 | irroot | matches all 4 digits |
| 14:55.01 | irroot | resmo exten => _XXXX,1, Dial(SIP/${EXTEN}) |
| 14:55.33 | resmo | irroot, hmm looks much better |
| 14:58.36 | kaldemar | and matches numbers that you don't have |
| 15:01.59 | irroot | kaldemar resmo indeed you can use aditional logic like a db look up more fine look up |
| 15:03.01 | *** join/#asterisk vampi-the-frog (~zyx@mail.cmbtravel.ro) |
| 15:03.54 | irroot | exten => _XXXX,1,Gotoif(${ODBC_XXX(${EXTEN}?:hangup) |
| 15:05.16 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:05.21 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 15:07.27 | *** join/#asterisk Tim_Toady (~moi@188.4.51.59) |
| 15:12.54 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 15:22.36 | bl0b` | ok tryed.. Monitor() too didnt support record h26x streams :( |
| 15:42.32 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 15:45.21 | *** join/#asterisk arossouw (~arossouw@196.36.203.18) |
| 15:45.58 | arossouw | receiving strange isdn code 18 ,when calling from asterisk bri (Junghanns) |
| 15:46.44 | *** part/#asterisk irroot (~irroot@dsl-185-122-118.dynamic.wa.co.za) |
| 15:48.12 | *** join/#asterisk fofware (~fabian@wdctf.siup.gov.ar) |
| 15:48.48 | bl0b` | arossouw http://networking.ringofsaturn.com/Routers/isdncausecodes.php maybe help? |
| 15:49.30 | arossouw | had a look at it, according to that the called party's equipment is at fault, but even when dialing my mobile i get the same error code |
| 15:50.08 | arossouw | zttool shows OK on all spans |
| 15:50.54 | bl0b` | weird, have you tryed to change the dialplan? |
| 15:50.56 | arossouw | also performed bri debug span 1->4, that's a bit greek |
| 15:51.34 | arossouw | yes, i tried changing Zap/g1 to Zap/r1, no luck |
| 15:52.59 | arossouw | RUNK1=Zap/g1 |
| 15:53.00 | arossouw | exten => _X.,1,Dial(${GLOBAL(TRUNK1)}/${EXTEN},85,tT) |
| 15:53.37 | *** join/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 15:53.44 | arossouw | wonder if the pstn provider is at fault |
| 15:54.19 | orn | I'm having the weirdest problem with Asterisk AGI... events aren't updated unless I try to perform some action |
| 15:54.38 | orn | They're buffered up, and then when I try to perform an action (an empty newline will suffice), everything that's been buffered up spews out |
| 15:55.49 | orn | I'm running the same version of asterisk in many other setups on the same operating system and this is the only place where this is a problem |
| 16:06.43 | *** join/#asterisk Poincare (~jefffnode@2001:470:d6b3:4::2) |
| 16:08.45 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 16:09.45 | *** part/#asterisk bl0b` (~vsfdfsdfs@host98-89-dynamic.53-79-r.retail.telecomitalia.it) |
| 16:21.16 | leifmadsen | anyone want to test an automated SIP URI callback website? |
| 16:23.19 | *** join/#asterisk last1 (~dood@65.39.216.2) |
| 16:24.19 | last1 | I am registering some phones through SIP to Asterisk. I see the phones when doing sip show peers and I can ping each phone from the asterisk box |
| 16:24.37 | last1 | however when I call the extensions in between themselves I get: The person at extension 'xxx' is unavailable |
| 16:24.52 | last1 | what can be causing this ? |
| 16:26.00 | paulc | last1: are the phones on the same network? behind a router/NAT? and is your dialplan setup properly as far as you know? |
| 16:26.51 | last1 | there is no nat, there is straight network connection |
| 16:27.07 | last1 | I can ping each phone from their own network + from the asterisk box |
| 16:33.40 | orn | last1: Your dialplan is probably incorrect |
| 16:36.08 | last1 | I'm using Trixbox, it should be fairly standard no ? |
| 16:36.15 | last1 | this is the debug information that I see scrolling on asterisk |
| 16:36.16 | last1 | http://pastebin.com/gTXnDj5v |
| 16:38.01 | orn | last1: You probably need to create a route in your dialplan to send it to an extension. |
| 16:38.44 | orn | '-- Executing [s@macro-dial:4] NoOp("SIP/111-00000000", "Returned from dialparties ____with no extensions to call____ and DIALSTATUS: NOANSWER")' should give you a hint. |
| 16:38.53 | orn | "with no extensions to call" |
| 16:39.57 | orn | actually, if you're calling extension 112, above it says that the extension state is 4 (UNAVAILABLE), which suggests that the extension isn't registered |
| 16:41.09 | last1 | exactly |
| 16:41.15 | last1 | but 112 IS available, it's registered |
| 16:41.30 | last1 | what does 'UNAVAILABLE' mean ? |
| 16:42.14 | orn | AFAIK it means that there's nobody registered to the extension |
| 16:43.03 | last1 | but doesn't this mean the phone registered ? : 112/112 192.168.0.112 D A 5060 OK (170 ms) |
| 16:43.15 | orn | Yes, it does. |
| 16:43.59 | last1 | hence my problem... what is going on ? :) |
| 16:45.00 | last1 | how can I see my current extensionstates ? maybe it's logged as unavail in astdb or something |
| 16:46.24 | *** join/#asterisk irroot (~irroot@197.105.23.194) |
| 16:47.58 | *** join/#asterisk davlefou (~david@41.227.62.16) |
| 16:49.11 | orn | are you sure you reloaded the dialplan and sip users after making changes last time (did you press apply changes?) |
| 16:49.45 | orn | i don't know how to look at extenstates apart from doing it in the dialplan |
| 16:49.59 | orn | or maybe from the AGI |
| 16:50.54 | last1 | yeah, I reloaded all the phones, even the entire server |
| 16:51.35 | orn | is the user in the correct context? |
| 16:58.30 | last1 | I believe so. I'm not sure what that means exactly |
| 17:01.44 | orn | Ok. I suggest to seek further assistance from the Trixbox forums or IRC channel. Trixbox is so far removed from Asterisk that people here will be reluctant to help you. |
| 17:02.39 | leifmadsen | +1 |
| 17:02.52 | leifmadsen | it's nearly impossible to help support unless you're familiar with trixbox inimitely |
| 17:06.30 | *** join/#asterisk phpboy (~shane@blowfish.x86.co.za) |
| 17:09.22 | *** part/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 17:15.24 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 17:15.31 | irroot | trixbox != asterisk like the distro i put together its targeted and not close to core asterisk |
| 17:15.55 | irroot | trixbox is enduser app / wrapper |
| 17:16.03 | last1 | true, but it does use asterisk |
| 17:16.16 | otwieracz | FIGHT |
| 17:16.30 | otwieracz | Let's create dial-plan. |
| 17:16.34 | last1 | it's no big deal |
| 17:16.42 | otwieracz | With voice-menu. |
| 17:16.45 | last1 | I'm sure it's got something to do with the astdb |
| 17:16.59 | last1 | aka, the state being saved there from a previous time when the phones were unavailable or something |
| 17:17.11 | irroot | always keen to help but its hard to seperate UI and core sometimes |
| 17:22.03 | *** join/#asterisk davlefou (~david@41.225.44.156) |
| 17:22.19 | last1 | yeah, it was the astdb |
| 17:22.24 | last1 | it works now |
| 17:26.19 | irroot | ooooh looks like the cert verification is lacking a bit in the TLS code |
| 17:29.38 | *** join/#asterisk caveat- (~false@newshell1.bshellz.net) |
| 17:33.11 | drmessano | I'm not sure if Trixbox's older, forked FreePBX has this available, but there is a FreePBX "command" to rewrite the ASTDB in cases like that |
| 17:33.30 | *** join/#asterisk quidpro (~quid246@CPE00131078b0b5-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 17:34.22 | last1 | what is it ? |
| 17:34.35 | drmessano | Google for "FreePBX" and "resetall" |
| 17:34.58 | drmessano | You'll find forum posts pointing to a URL you drop in that ends in action=resetall |
| 17:35.17 | quidpro | Anybody know why 1.6 spits back "WARNING[21767]: utils.c:1538 __ast_string_field_init: trying to reset empty pool" when an IAX2 call comes in (doesn't seem to happen with SIP). I find it in alot of logs on a Google Search, but no real explanation except to ignore it, seems if it's at WARN level it should be addressed. |
| 17:39.59 | *** join/#asterisk arielb27 (~chatzilla@pdpc/supporter/active/abatista) |
| 17:41.21 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 17:48.52 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 17:50.51 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 17:57.45 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 17:59.30 | last1 | thank you very much everyone |
| 18:01.06 | *** join/#asterisk davlefou (~david@196.203.155.13) |
| 18:02.17 | otwieracz | Where to get Asterisk sounds like âBossâ, âTechnical supportâ etc? |
| 18:11.43 | otwieracz | hmm. |
| 18:11.56 | otwieracz | How to use DTFM in Astrisk? |
| 18:11.59 | ChannelZ | record them? |
| 18:12.08 | otwieracz | (to create voice menu) |
| 18:12.19 | ChannelZ | make extensions |
| 18:12.25 | otwieracz | Simply one extension for one level? |
| 18:12.44 | ChannelZ | if you want to make trees, do them in separate contexts |
| 18:13.01 | otwieracz | Yes, I want tree. |
| 18:13.29 | ChannelZ | "press 1 for more pointless options" -- exten 1 does a Goto context 'pointless'... "press 1 to hear chickens" etc |
| 18:13.29 | otwieracz | But how to do âif 1 then dial foo and then if 2 dial barâ. |
| 18:13.55 | otwieracz | I'll try. |
| 18:15.13 | tzafrir_laptop | otwieracz, look for 'core' and 'extra' sound sets |
| 18:15.21 | otwieracz | Yes, I found some there. |
| 18:15.25 | otwieracz | Maybe enough, |
| 18:15.44 | otwieracz | Ahm, WaitExten() |
| 18:16.31 | ChannelZ | It's basically like a silent Background() of a specified length |
| 18:17.06 | ChannelZ | so use both |
| 18:19.37 | otwieracz | I changed usernames in sip.conf and: |
| 18:19.42 | otwieracz | [May 30 20:19:00] WARNING[23491]: chan_sip.c:13660 check_auth: username mismatch, have <02>, digest has <03> |
| 18:19.56 | otwieracz | Why? |
| 18:20.09 | otwieracz | (previusly it was 03, now 02) |
| 18:28.27 | otwieracz | How to dial more than one numbers? |
| 18:28.41 | otwieracz | (who answers faster) |
| 18:28.48 | WIMPy | & |
| 18:28.56 | leifmadsen | Dial(SIP/foo&SIP/bar) |
| 18:44.15 | *** join/#asterisk chasing`Sol (~cS@smtp.master-zone.net) |
| 18:52.43 | otwieracz | Thanks. |
| 18:52.53 | otwieracz | So awesome menu⦠:) |
| 18:53.51 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 18:59.01 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 19:09.55 | *** join/#asterisk GoatHunter (~a@93-173-50-186.bb.netvision.net.il) |
| 19:23.46 | otwieracz | Hmm. |
| 19:23.57 | otwieracz | I can hear my caller, but he can't hear me. |
| 19:24.18 | beek | otwieracz: Smells like a NAT problem. |
| 19:24.23 | otwieracz | (He's calling from PSTN by Sipgate to me, connected directly to asterisk) |
| 19:24.31 | otwieracz | Yes, I'm behind NAT. |
| 19:24.47 | otwieracz | http://wklej.org/hash/3c3bb767d97/ |
| 19:24.55 | otwieracz | âmainâ is router. |
| 19:25.11 | otwieracz | rtpstart=10000 |
| 19:25.12 | otwieracz | rtpend=10256 |
| 19:26.47 | beek | Do you have "nat=yes" configured for your peer? |
| 19:28.07 | tzanger | I think sip.conf needs a nat=ohhellyes option |
| 19:29.21 | otwieracz | I have nat=yes. |
| 19:39.37 | otwieracz | Hmm. |
| 19:39.55 | otwieracz | I need literals (V, T and S) but without silence before and after. |
| 19:39.59 | otwieracz | (to build âVTSâ) |
| 19:40.05 | otwieracz | (not V--T--S) |
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| 20:00.27 | otwieracz | Hmm⦠How to convert to gsm? |
| 20:00.57 | otwieracz | toast vtssoftware.wav makes a lot of noise only. |
| 20:11.31 | otwieracz | sox |
| 20:11.32 | otwieracz | :) |
| 20:20.33 | phpboy | otwieracz: nat is a serious issue |
| 20:20.34 | phpboy | :T |
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| 20:58.09 | karmst | Hello |
| 20:58.27 | karmst | Which distro of asterisk is the most stable? |
| 20:59.19 | WIMPy | karmst: May the source be with you. |
| 21:00.02 | karmst | I'm looking to build an asterisk / freepbx solution for our company |
| 21:00.12 | karmst | I want something that is rock solid |
| 21:00.26 | karmst | and doesn't take hours to configure |
| 21:00.31 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 21:01.24 | karmst | is asterisknow stable? |
| 21:01.38 | karmst | or should we go with switchvox? |
| 21:01.43 | Freeaqingme | karmst: 1.8.4 supposedly is best |
| 21:02.30 | karmst | what version is in asterisknow 1.7.1? |
| 21:03.34 | WIMPy | Well, stable and stable. It depends on what you do with (to?) it. |
| 21:04.31 | karmst | I want to build a solution for our company |
| 21:04.38 | karmst | I don't want any downtime |
| 21:04.41 | WIMPy | And yes, I'd go for the latest release as well. |
| 21:04.55 | WIMPy | That could mean anything and more. |
| 21:05.15 | *** join/#asterisk PsiTrax (~psi@DSL01.212.114.206.69.ip-pool.NEFkom.net) |
| 21:06.06 | PsiTrax | is there a elseif in ael? |
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| 21:11.05 | otwieracz | I still have one-way-audio. |
| 21:11.05 | otwieracz | In logs I see: |
| 21:11.05 | otwieracz | [May 30 23:08:58] WARNING[25661]: app_dial.c:2041 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
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| 21:11.47 | ChannelZ | well that's not a one-way audio problem, that's a "you're Dial()ing something that doesn't exist or cannot be reached" problem |
| 21:11.51 | WIMPy | otwieracz: That means that some sip device is unreachable. |
| 21:12.24 | otwieracz | Probably it's beucase I'm calling two numbers and only one is online. |
| 21:12.34 | WIMPy | yes |
| 21:16.38 | otwieracz | Ok, so, what can I do with this one-way audio? |
| 21:16.49 | otwieracz | I hear my caller really nice. |
| 21:16.53 | otwieracz | But he don't hear me at all. |
| 21:16.59 | ChannelZ | firewall/NAT issues? |
| 21:17.07 | otwieracz | Probably. |
| 21:17.17 | otwieracz | http://wklej.org/hash/3c3bb767d97/ |
| 21:17.20 | otwieracz | That's from my router. |
| 21:17.46 | otwieracz | rtpstart=10000 |
| 21:17.46 | otwieracz | rtpend=10256 |
| 21:18.07 | ChannelZ | If he can't hear you, either your audio isn't getting out of your firewall, or it isn't getting through his |
| 21:18.39 | ChannelZ | You send your audio to the port the remote end requests, which is a bit out of your control. |
| 21:20.01 | otwieracz | Uhm⦠|
| 21:20.53 | otwieracz | I need to forward anything else? |
| 21:21.22 | ChannelZ | RE: The remote end requests where you should send your audio. That port could be anything. |
| 21:22.23 | ChannelZ | You can look at a SIP debug and see in the call setup where it might be requesting and make a guess about the range to allow |
| 21:22.48 | ChannelZ | or just allow all high-port UDP out |
| 21:23.03 | otwieracz | I was sure that only communication from outside NAT to me could be problem⦠|
| 21:23.16 | ChannelZ | it can be both. |
| 21:23.39 | WIMPy | otwieracz: What kind of character ist that at the end of your lines? |
| 21:24.01 | otwieracz | Character? |
| 21:24.41 | WIMPy | Something I can't see correctly. |
| 21:24.57 | ChannelZ | Hmm. I don't see anything |
| 21:25.26 | WIMPy | <otwieracz> Uhm⦠|
| 21:25.37 | otwieracz | Ahm! |
| 21:25.39 | otwieracz | :) |
| 21:25.42 | ChannelZ | <otwieracz> Uhm⦠|
| 21:25.42 | otwieracz | "..." |
| 21:25.51 | otwieracz | Byt in one character. |
| 21:25.58 | WIMPy | Ah |
| 21:26.06 | ChannelZ | oh. Interesting. |
| 21:26.08 | otwieracz | sryrur |
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| 21:27.46 | ChannelZ | your keyboard must have more keys than mine |
| 21:27.58 | otwieracz | Nope. |
| 21:28.02 | otwieracz | M-, |
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| 21:28.08 | otwieracz | (with my keymap) |
| 21:28.42 | WIMPy | Well, back in the days keyboards used to have a compose key. It's still supported by X, you just have to map it. It's extremely useful. |
| 21:29.19 | ChannelZ | Filthy! There's pictures of people pooping in this font. á¾ |
| 21:29.51 | WIMPy | is purely 8 bit. |
| 21:32.45 | otwieracz | I don't know why it is now working⦠|
| 21:38.46 | otwieracz | I will work on it tomorrow. |
| 21:38.47 | otwieracz | Bye. |
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| 21:59.44 | WiretapSeven | otwieracz, your buddy is behind nat, no? |
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| 22:28.27 | otwieracz | WiretapSeven: He's calling from cellphone, through Sipgate. |
| 22:31.44 | WiretapSeven | ah |
| 22:32.46 | otwieracz | Any ideas? |
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| 22:39.02 | otwieracz | i/aw . |
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| 22:50.36 | spck | hello all |
| 23:04.26 | eyeballer | ayone around to help with a new set up? |
| 23:04.50 | eyeballer | i have asterisk running on my rt16 with tomato usb.. but i dont think jabber/gtalk is connected |
| 23:05.18 | eyeballer | i took the .conf files from http://www.arctangent.net/~superm1/gv_configs/ and modified gthem with my google info |
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| 23:13.57 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 23:15.57 | sawgood | I would like to test a call from one extension to another extension (on the same Asterisk box) to see if the two end points are using G.729 instead of G.711u (they are on the same LAN as each other) ... from the CLI on the Asterisk box, what command would I use to do this? |
| 23:16.14 | sawgood | The * box has no G.729 licenses ... this would be in 'pass-through' mode |
| 23:16.27 | sawgood | If that works, I'll buy some G.729 licenses tomorrow from Digium |
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| 23:17.53 | WIMPy | sip show channel ... |
| 23:18.21 | WIMPy | But paass-through and transcoding are two completely different things. |
| 23:18.37 | sawgood | Hi WIMPy ... ty! |
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| 23:19.06 | sawgood | I have two SIP end points with 'built in' G.729 ... and I want to see if they can call each other using that codec |
| 23:19.37 | WIMPy | Can't those endpoints tell you what they use? |
| 23:20.31 | sawgood | from their web GUI you can choose ...but to see it actually passing it what I wanted to know for confirmation |
| 23:21.03 | WIMPy | No info screen? |
| 23:21.41 | sawgood | no not that I could think of |
| 23:21.59 | sawgood | from sip show channel <channel id> .... which setting is the codec? |
| 23:22.02 | sawgood | Is it Format? |
| 23:22.33 | sawgood | <PROTECTED> |
| 23:22.51 | sawgood | Our Codec Capability: 270 |
| 23:23.08 | WIMPy | Format, yes. |
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| 23:24.51 | sawgood | Is there a list of which Format = which codec? |
| 23:25.11 | ChannelZ | core show codecs - but it should say after |
| 23:25.17 | WIMPy | Current versions show it textual. |
| 23:26.26 | sawgood | there sure are a lot of them |
| 23:26.54 | WIMPy | Then wait for 1.10 :-) |
| 23:27.03 | sawgood | if 0x4 = G.711 u-law ... what is 0x0 then? |
| 23:27.20 | WIMPy | Nothing |
| 23:27.31 | sawgood | Is nothing = PCMU? |
| 23:27.32 | ChannelZ | 270 should be gsm, u-law, a-law, and g729 |
| 23:27.34 | WIMPy | It's a bitmap. |
| 23:31.17 | sawgood | I had G.729 on both phones, but it was the last codec at the bottom of the list, so I moved it to the top on both phones and restarted them |
| 23:31.19 | sawgood | I'll try now |
| 23:32.05 | sawgood | Format = 0x4 (ulaw) |
| 23:33.13 | WIMPy | Do you have any features enabled that require Asterisk to ananlyze the media stream? |
| 23:33.43 | sawgood | Well, not really ... just two SIP phones calling each other through an * 1.6.2.18 configuration |
| 23:34.52 | sawgood | its cool how it went from format = 0x0 (nothing) to format = 0x4 (ulaw) simply by moving the codec order |
| 23:38.06 | sawgood | So, in a nutshell ... why did I have format = 0x0 in the first place (before moving G.729 to the top of the list) on the SIP phones? |
| 23:44.26 | ChannelZ | because if you were looking at 'sip show channels' you were probably looking at a SUBSCRIBE or some other event |
| 23:45.33 | karmst | What is the most stable release of Asterisk? |
| 23:45.48 | karmst | I need to build an asterisk solution for our company |
| 23:45.55 | WIMPy | 2.1.1 |
| 23:45.59 | karmst | it doesn't need to have a bunch of extra features |
| 23:46.03 | ChannelZ | how long is a piece of string? |
| 23:46.05 | sawgood | ChannelZ: got it now ... 0x4 on all my calls no matter what the codec order is on my phones |
| 23:46.09 | WIMPy | goes to the lottery. |
| 23:46.09 | karmst | but it needs to be rock solid |
| 23:46.38 | WIMPy | karmst: Buy some old commercial PBX then. |
| 23:46.55 | sawgood | oh ... that is beautiful ... "how long is a piece of string" |
| 23:47.00 | WIMPy | karmst: You haven't even told us what you want it to do. |
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| 23:47.23 | karmst | provide VoIP SIP trunking |
| 23:48.03 | WIMPy | for ...? |
| 23:48.10 | karmst | a business |
| 23:48.42 | WIMPy | Are you going to use phones? |
| 23:49.36 | karmst | yeah |
| 23:49.46 | WIMPy | Cool. |
| 23:49.51 | WIMPy | What kind of phones? |
| 23:50.02 | ChannelZ | telephones! |
| 23:50.41 | karmst | multiple kinds |
| 23:50.44 | WIMPy | No nearphones? |
| 23:50.56 | WIMPy | karmst: Maybe you should read |
| 23:51.01 | WIMPy | ~thebook |
| 23:51.01 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 23:51.18 | WIMPy | So that you will be able to know what you want. |
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| 23:56.57 | sawgood | wow ... it worked ... 0x100 (g729) .... neat ... |