| 00:14.44 | igcewieling1 | does rtptimeout still work if reinvites move the audio off the asterisk server? I'd say no, am not so sure since asterisk sort of supports RTCP as well. |
| 00:14.53 | *** join/#asterisk fulcan (brads@2600:3c00::f03c:91ff:fe70:f0a9) |
| 00:18.16 | file | it doesn't |
| 00:19.26 | *** join/#asterisk war9407 (war@c-71-62-63-105.hsd1.va.comcast.net) |
| 00:21.52 | *** join/#asterisk apardo (~apardo@181.54.141.49) |
| 00:22.16 | apardo | how to can i get the callee id of a call ? |
| 00:22.29 | igcewieling1 | file: thank you. |
| 00:23.47 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:23.47 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:26.14 | *** join/#asterisk navaismo (~navaismo@189.241.118.172) |
| 00:31.10 | [TK]D-Fender | apardo: You get it when it comes in. |
| 00:54.09 | fulcan | I did a 1.4 to 1.6 upgrade and asterisk is failing to start with a permissions error "[2013-02-28 19:50:54] ERROR[6057]: logger.c:1046 init_logger: Unable to create event log: Permission denied" and Unable to access the running directory (Permission denied). on suse |
| 00:55.05 | [TK]D-Fender | fulcan: Nothing 1.6 is supported |
| 00:55.36 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 00:55.49 | fulcan | don't give me that. I had to fight to get 1.4 off |
| 00:56.49 | [TK]D-Fender | You fought for the wrong target... |
| 00:57.09 | [TK]D-Fender | 1.6.0. 1.6.1. 1.6.2 all completely EOL |
| 00:59.21 | *** join/#asterisk BludSuckingFiend (~pi@cpe-24-160-205-248.insight.res.rr.com) |
| 00:59.35 | navaismo | in the other hand you should look at that error check the permissions on that folder and the owner |
| 01:00.05 | fulcan | but which folder, the error is non specific. |
| 01:00.38 | [TK]D-Fender | Well... it says LOGGER. |
| 01:00.44 | navaismo | reight^ |
| 01:00.45 | [TK]D-Fender | my guess is that logger.conf would tell you where |
| 01:04.17 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 01:04.35 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 01:04.43 | *** join/#asterisk g_r_eek (~g_r_eek@173.9.142.122) |
| 01:05.18 | fulcan | logger doesn't have anything and setting the verbose higher doesn't help. |
| 01:06.00 | [TK]D-Fender | hav you looked at /var/log ? |
| 01:06.21 | file | it would probably be in... /var/log/asterisk |
| 01:06.38 | file | (where it is trying to create) |
| 01:06.57 | igcewieling1 | fulcan: are you running freepbx? |
| 01:07.17 | fulcan | vicidial iso |
| 01:07.41 | fulcan | I am chown'ing a bunch of stuff and she is coming alive. |
| 01:07.50 | igcewieling1 | maybe the default user Asterisk runs as changed. |
| 01:09.43 | fulcan | no command line? http://pastie.org/6357602 |
| 01:10.11 | [TK]D-Fender | [2013-02-28 20:05:40] WARNING[7220]: db.c:57 dbinit: Unable to open Asterisk database '/v |
| 01:10.15 | [TK]D-Fender | Not astdb either? |
| 01:10.24 | [TK]D-Fender | Your perms much be royally screwed all over the place |
| 01:12.56 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 01:13.00 | mattwj2002 | hi guys |
| 01:13.10 | mattwj2002 | I need help getting incoming calls working for asterisk |
| 01:13.34 | WIMPy | From where? |
| 01:13.42 | fulcan | where is 'sip show peers'? |
| 01:13.56 | igcewieling1 | fulcan: what is the current user of an asterisk file you did not change the ownership on? |
| 01:14.25 | fulcan | chown -R asterisk:asterisk /var/lib/asterisk brought the system live |
| 01:14.31 | fulcan | igcewieling1 ^^ |
| 01:14.44 | mattwj2002 | what is unmonitored mean? |
| 01:14.54 | igcewieling1 | yeah, but what was the owner before you did that |
| 01:15.20 | WIMPy | mattwj2002: You don;t have qualify enabled. |
| 01:15.21 | fulcan | igcewieling1 not sure, it work now though. |
| 01:15.34 | mattwj2002 | qualify? |
| 01:15.50 | mattwj2002 | where is that/ |
| 01:15.52 | fulcan | where are the sip commands hiding? |
| 01:16.21 | navaismo | mattwj2002, sip.conf |
| 01:16.42 | mattwj2002 | qualify=yes under general? |
| 01:16.54 | file | fulcan, your system could not open the modules directory and thus no modules were loaded |
| 01:16.56 | navaismo | fulcan, probably your asterisk haven't loaded the chan_sip.so |
| 01:17.12 | navaismo | mattwj2002, under your peers |
| 01:17.36 | fulcan | navaismo how do I fix that? |
| 01:17.37 | WIMPy | mattwj2002: You can do so if you want it for everyone. |
| 01:17.48 | [TK]D-Fender | fulcan: you need to fix the permissions on all of *'s folders |
| 01:18.07 | [TK]D-Fender | fulcan: So far you can't load modules, can't touch AstDB, can't log, can't do pretty much anything. |
| 01:18.33 | mattwj2002 | okay! |
| 01:18.34 | mattwj2002 | :D |
| 01:19.42 | mattwj2002 | still not working |
| 01:19.43 | mattwj2002 | :( |
| 01:19.51 | fulcan | [TK]D-Fender where else besides /var/lib/asterisk ? |
| 01:20.11 | [TK]D-Fender | fulcan: modules folder,, /var/log./... |
| 01:20.15 | [TK]D-Fender | fulcan: var/spool, etc |
| 01:20.22 | [TK]D-Fender | fulcan: Go reinstall and do it right |
| 01:20.37 | [TK]D-Fender | fulcan: And set your perms right as well as the runuser |
| 01:20.56 | WIMPy | mattwj2002: What isn;t working? |
| 01:21.42 | [TK]D-Fender | fulcan: You also need to brush up on the CLI changes from 1.4 as we can see the first half-dozen that aren't right for 1.6+ |
| 01:21.48 | fulcan | ls: cannot access /usr/lib/asterisk/: No such file or directory ?? |
| 01:21.50 | navaismo | fulcan, asterisk.conf will show all folders related to asterisk |
| 01:23.17 | *** join/#asterisk serafie (~erin@24.214.158.242) |
| 01:24.31 | fulcan | navaismo correct. why is the module directory missing?astmoddir => /usr/lib/asterisk/modules |
| 01:25.18 | mattwj2002 | not sure |
| 01:25.19 | [TK]D-Fender | first guess : no premissions to even create it. |
| 01:26.01 | [TK]D-Fender | Highly recommend trashing and reinstalling.... |
| 01:26.07 | fulcan | <PROTECTED> |
| 01:26.15 | fulcan | 64 bit directory |
| 01:26.45 | [TK]D-Fender | perhaps you should be pointing there instead.... |
| 01:26.55 | fulcan | [TK]D-Fender I have no clue how vicidial is going to react to asterisk 10 and vicidial is more important that asterisk at this point. |
| 01:27.02 | fulcan | than |
| 01:27.24 | [TK]D-Fender | go check all the patchs for your * install.... |
| 01:27.37 | [TK]D-Fender | Did you just try to COPY your configs from another system entirely as-is? |
| 01:27.45 | [TK]D-Fender | That is what this is beginning to feel like... |
| 01:28.00 | [TK]D-Fender | Where all of the underlying paths are being turned upside down |
| 01:28.53 | navaismo | and I dont know how vicidial install all so better ask in their forum/irc |
| 01:30.46 | fulcan | [TK]D-Fender it's an iso install (out the box) with zero confiig other that a 1.6 upgrade because 1.4 was just st00pid. |
| 01:31.20 | *** join/#asterisk Starstorm (~Starstorm@12.148.212.178) |
| 01:31.31 | fulcan | [TK]D-Fender the vici dial latest release iso comes with 1.4 :( |
| 01:31.46 | Starstorm | Hiya, Anyone willing to help me with a dahdi problem? |
| 01:31.48 | [TK]D-Fender | paths and/or configs got mangled. fix them up and try again |
| 01:32.09 | [TK]D-Fender | ~ask |
| 01:32.09 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 01:32.12 | mattwj2002 | WIMPy incoming calls now |
| 01:32.13 | [TK]D-Fender | Starstorm: ^^ |
| 01:33.39 | WIMPy | fulcan: I'm not sure I'd prefer 1.6 over 1.4. |
| 01:34.04 | fulcan | when I changed to the 64 bit module directory, I start getting this http://pastie.org/6357681 |
| 01:34.10 | WIMPy | mattwj2002: You are not being very specific. |
| 01:34.12 | [TK]D-Fender | BTW ... 1.6 isn't even a BRANCH |
| 01:34.24 | mattwj2002 | sorry WIMPy |
| 01:34.28 | [TK]D-Fender | fulcan: What precise version are you running now anyway? |
| 01:34.44 | mattwj2002 | I call my google voice number using skype and it doesn't ring my phone |
| 01:34.48 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 01:34.49 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 01:34.49 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 01:34.55 | Starstorm | dahdi isn't loading the firmware into the xorcom astribank, it won't get past (lsusb) 1160 when I do xpp_fxloader load. I get fxload command not found fxload failed woth status 127 |
| 01:35.02 | jmetro | Has anyone used the STAT function recently and know what it returns? |
| 01:35.04 | mattwj2002 | I see nothing on the console either |
| 01:35.45 | fulcan | Version: 1.6.2.24-1.2 Installed: 1.6.2.24-1.2 |
| 01:35.57 | fulcan | [TK]D-Fender ^ |
| 01:37.14 | [TK]D-Fender | Well ..... at least you did get to the end of that line... |
| 01:37.20 | BludSuckingFiend | So who here is familiar with the res_phoneprov module? |
| 01:37.37 | jmetro | trying to stat -e for FIle Existence and the documentation |
| 01:37.39 | jmetro | has nothing |
| 01:38.56 | mattwj2002 | is there a debug command I am missing or something? |
| 01:39.20 | fulcan | Asterisk doesn't like those 64 bit modules at all. |
| 01:40.17 | [TK]D-Fender | fulcan: strip it all and redo fresh |
| 01:40.30 | WIMPy | fulcan: Are you mixing different versions? |
| 01:41.08 | fulcan | all I did was take a vicidial iso and upgrade asterisk to 1.6 Thats it. |
| 01:41.28 | fulcan | it is a 64bit iso |
| 01:41.30 | [TK]D-Fender | fulcan: Yes, the upgrade is botched |
| 01:41.32 | [TK]D-Fender | Strip Asterisk |
| 01:41.35 | [TK]D-Fender | Reinstall it |
| 01:42.08 | Starstorm | Asterisk stopped running when it got rebooted. dahdi isn't loading the firmware into the xorcom astribank, it won't get past (lsusb) e4e4-1160 when I do /xpp_fxloader load. I get 'fxload command not found fxload failed woth status 127' Any ideas? |
| 01:43.34 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:44.28 | WIMPy | reinstall dahdi and the xorcom tools? |
| 01:46.23 | [TK]D-Fender | Starstorm: you may have to be patient. Not a lot of Xorcom users around at any given time.... |
| 01:46.57 | pabelanger | Starstorm: you want talk with tzafrir_laptop |
| 01:47.30 | [TK]D-Fender | That would be ideal.... |
| 01:47.45 | Starstorm | ok |
| 01:47.53 | BludSuckingFiend | Xorcom eh? A friend was talking about those the other day. |
| 01:48.01 | BludSuckingFiend | Thinking about using them |
| 01:48.07 | pabelanger | never tested them |
| 01:48.11 | pabelanger | but some cool things |
| 01:48.16 | pabelanger | tdm over USB |
| 01:48.17 | Starstorm | sorry, been at it for 5 hours. downtime sucks |
| 01:48.39 | WIMPy | Nice for debugging if you put them in to a smaller case :-) |
| 01:48.42 | BludSuckingFiend | yeah... I am skeptical about anything time-sensitive over USB |
| 01:48.43 | fulcan | [TK]D-Fender ripping it out and reinstaling and adding the addons worked. I have a normal asterisk command line now. |
| 01:48.58 | BludSuckingFiend | timing sensitive I mean |
| 01:49.01 | [TK]D-Fender | No more logging errors? |
| 01:49.03 | pabelanger | Starstorm: well, 127 is not found |
| 01:49.05 | [TK]D-Fender | AstDB restored? |
| 01:49.07 | [TK]D-Fender | Configs? |
| 01:49.11 | pabelanger | so see if fxload is actually installed |
| 01:49.35 | pabelanger | or just a path issue |
| 01:50.08 | [TK]D-Fender | BludSuckingFiend: Long ago ztdummy used it for just that. And there are USB timing devices specifically for * |
| 01:50.23 | fulcan | [TK]D-Fender how do you check the AstDB again? |
| 01:51.00 | BludSuckingFiend | Yeah, I just have some bias against USB based on years of troubleshooting USB network devices |
| 01:51.08 | BludSuckingFiend | but that was before 2.1/3.0 |
| 01:51.41 | BludSuckingFiend | probably fine now... but I am still reluctant. I like TDMoE to my channel banks... |
| 01:51.52 | *** part/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 01:52.09 | [TK]D-Fender | BludSuckingFiend: I have a bigger isue of business critical hardware being so easily unplugged without so much as a little plastic clip I take for granted on RJ45 :) |
| 01:52.22 | BludSuckingFiend | lol yaeh |
| 01:52.23 | WIMPy | has heard more bad things about the redfones than about the Astribank. |
| 01:52.58 | [TK]D-Fender | WIMPy: Agreed, those can DIAF |
| 01:53.03 | WIMPy | But I wouldn't like to rely on an USB cable, either. |
| 01:53.28 | pabelanger | I got to test a USB controller in a RF chamber. It was pretty cool, but our testing kept causing the kernel driver to panic |
| 01:53.32 | pabelanger | that was not fun |
| 01:57.37 | fulcan | *CLI> reload |
| 01:57.37 | fulcan | <PROTECTED> |
| 01:57.37 | fulcan | <PROTECTED> |
| 01:57.37 | fulcan | <PROTECTED> |
| 01:57.37 | fulcan | <PROTECTED> |
| 01:57.37 | fulcan | <PROTECTED> |
| 01:57.37 | fulcan | <PROTECTED> |
| 01:57.38 | fulcan | <PROTECTED> |
| 01:57.38 | fulcan | <PROTECTED> |
| 01:57.49 | WIMPy | ~pb |
| 01:57.49 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 02:00.34 | [TK]D-Fender | heading out for a while... |
| 02:03.01 | BludSuckingFiend | too late |
| 02:05.20 | *** join/#asterisk fulcan (brads@2600:3c00::f03c:91ff:fe70:f0a9) |
| 02:05.41 | fulcan | huh? -> No such command 'exit' (type 'core show help exit' for other possible commands) |
| 02:06.25 | BludSuckingFiend | you pasted too many lines too fast |
| 02:06.32 | BludSuckingFiend | 20:57 < infobot> A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, |
| 02:06.36 | BludSuckingFiend | <PROTECTED> |
| 02:07.50 | fulcan | BludSuckingFiend thay wasn't me, that was my dirty mouse. |
| 02:07.54 | *** join/#asterisk deo_ (~deo@222.127.13.226) |
| 02:08.06 | WIMPy | Clean it! |
| 02:09.26 | fulcan | winblows clipboard error. http://pastebin.com/c3Tkm8hu <- that's what it was supposed to get. |
| 02:09.44 | fulcan | what happened to my 'exit'? |
| 02:10.04 | WIMPy | Gone. |
| 02:10.15 | WIMPy | No way out any more. |
| 02:10.18 | WIMPy | Too late to leave. |
| 02:10.29 | fulcan | trapped like a rat? |
| 02:10.35 | fulcan | damn |
| 02:10.42 | fulcan | will there be internet? |
| 02:10.54 | WIMPy | I think rat's aren't too bad at diigin their way out. |
| 02:11.25 | WIMPy | Yes, you are trapped in the internet. |
| 02:11.48 | fulcan | as long as I got my porn, i'm good to go! |
| 02:11.53 | fulcan | :) |
| 02:12.44 | WIMPy | Anything you find interesting in that pastebin? |
| 02:13.24 | fulcan | no, it looks clean to me. |
| 02:13.34 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 02:13.41 | fulcan | looks like what you are supposed to see. |
| 02:13.47 | WIMPy | thinks so, too. |
| 02:14.08 | themrrobert | exit doesn't work u want quit |
| 02:14.42 | fulcan | thats the only thing. how to escape the command line. I feel like a retard |
| 02:15.00 | fulcan | No such command 'quit' (type 'core show help quit' for other possible commands) |
| 02:15.50 | themrrobert | what version are u running |
| 02:16.17 | themrrobert | oh i bet you are in the main console |
| 02:16.29 | fulcan | yes, main console |
| 02:16.38 | themrrobert | then to "quit" means shutdown |
| 02:16.46 | fulcan | Version: 1.6.2.24-1.2 Installed: 1.6.2.24-1.2 |
| 02:16.56 | WIMPy | That will stay as long as Asterisk is running. |
| 02:16.59 | fulcan | themrrobert means nothing to me.. :/ |
| 02:17.00 | themrrobert | ^ |
| 02:17.22 | themrrobert | to exit you have to core stop [now,when convenient] |
| 02:17.31 | themrrobert | but that will kill the pbx |
| 02:17.49 | themrrobert | if you're on a "connected" CLI ( via asterisk -r ) then you can exit with quit |
| 02:18.44 | fulcan | oh |
| 02:18.58 | fulcan | how do you exit the cli gracefully? |
| 02:19.30 | WIMPy | Don't start the daemon in the foreground. |
| 02:19.38 | fulcan | oh |
| 02:19.45 | WIMPy | Now it's too late. |
| 02:19.59 | WIMPy | You can stay there or kill it. |
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| 04:32.55 | BludSuckingFiend | I'm not so sure about this new-fangled users.conf... |
| 04:43.05 | ChannelZ | is it? |
| 04:44.47 | BludSuckingFiend | accidently left a parsing error in it and reloaded the dialplan... segfaulted |
| 04:45.57 | *** join/#asterisk chang33 (~jimmy@114-32-7-232.HINET-IP.hinet.net) |
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| 05:02.34 | Yxa | using 11.2.1. help I can't get call parking to work. When I transfer a called to ext 700, it just says it's an invalid extension |
| 05:07.49 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-sepklrgdkhfgagmc) |
| 05:25.06 | themrrobert | ouch BludSuckingFiend that would be disastrous if i did it haha |
| 05:25.37 | BludSuckingFiend | yeah, tell me about it... I dropped over 1 PRI worth of active calls |
| 05:27.37 | BludSuckingFiend | what's worse is I was converting from using sip.conf for my phones to users.conf... now it looks like I'll have to switch back |
| 05:28.04 | BludSuckingFiend | stuff in users.conf seems to be adding some dynamic dialplan entries that are breaking stuff... a gosub to macro-stdexten |
| 05:28.50 | BludSuckingFiend | Why inject "hidden" entries into your dialplan that you can't modify? |
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| 05:32.25 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 05:32.26 | *** mode/#asterisk [+o file] by ChanServ |
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| 06:17.25 | Yxa | repeat: using 11.2.1. help I can't get call parking to work. When I transfer a called to ext 700, it just says it's an invalid extension |
| 06:21.12 | ChannelZ | Does your dial() have the appropriate arguments to allow parking? |
| 06:21.59 | ChannelZ | Is it turned on in features.conf? Are you including the parking lot context in your extensions? |
| 06:23.09 | ChannelZ | or actually you said transfer so the dial options don't matter |
| 06:23.16 | ChannelZ | (except that they can transfer..) |
| 06:27.34 | Yxa | yes everything is in order |
| 06:27.44 | ChannelZ | well something isn't |
| 06:28.50 | ChannelZ | my guess is you don't have the parking lot context included in your dialplan |
| 06:42.04 | Yxa | ChannelZ the include => parkedcalls? it's there |
| 06:42.52 | ChannelZ | under the right context? |
| 06:43.49 | Yxa | yeah |
| 06:45.04 | ChannelZ | so "dialplan show 700@whatever" shows, where 'whatever' is the context your phone is in? |
| 06:45.13 | Yxa | but there isn't a physical [parkedcall] context anywhere |
| 06:46.16 | Yxa | <PROTECTED> |
| 06:47.32 | Yxa | so when i try to transfer the call to 700, it says i'm sorry that's not a valid extension |
| 06:48.28 | *** join/#asterisk chang33 (~jimmy@114-32-7-232.HINET-IP.hinet.net) |
| 06:51.32 | ChannelZ | where is that message coming from? Do you have some other extension pattern that's sucking up 7xx? |
| 06:51.33 | Yxa | anyone? |
| 06:51.54 | ChannelZ | it's sort of tiresome making 500 guesses, pastebin some verbose console output and your extensions.conf |
| 06:57.56 | ChannelZ | wanders off |
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| 07:39.45 | *** join/#asterisk Bradada (~Bradada@59-120-137-34.HINET-IP.hinet.net) |
| 07:39.54 | ChannelZ | guess he really didn't want help |
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| 07:51.36 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.198.19) |
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| 08:01.46 | themrrobert | Yxa? |
| 08:01.53 | ChannelZ | yah |
| 08:02.05 | themrrobert | ya |
| 08:02.34 | themrrobert | probly waws easy fix sounded |
| 08:03.35 | ChannelZ | who knows, I can only make so many guesses without seeing something |
| 08:04.49 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 08:05.11 | niklaswe | hey guys, I have a question.. I trying to user videosupport in my conference, and it working when to ppl are connected, but when third person joining I get "Waiting for remote Video" |
| 08:05.19 | niklaswe | any idé what I have miss? |
| 08:07.28 | *** join/#asterisk jkroon (~jkroon@dsl-244-34-26.telkomadsl.co.za) |
| 08:07.38 | jkroon | how does the qualify in asterisk 11.2.1 work? |
| 08:08.17 | jkroon | afaik it sends OPTIONS and then waits for a SIP response, however, I've got one peer where even though it's not even reachable at an IP level (destination network unreachable ICMP responses) asterik reports the peer as reachable. |
| 08:08.44 | themrrobert | sip set debug peer 333 |
| 08:09.36 | themrrobert | then do a test call, and you will see sip traffic for peer 333, or "sip set debug on" on for all sip traffic to be shown |
| 08:10.00 | themrrobert | can the peer make calls or receive them? |
| 08:10.53 | jkroon | themrrobert, no. |
| 08:11.04 | jkroon | the *IP* level link between the two is non-existent. |
| 08:11.38 | jkroon | but sip show peer 333 shows a status of OK (1 ms) ... even after I issued a "sip qualify peer 333" |
| 08:11.49 | themrrobert | what output are you seeing from asterisk indicating its reachable? |
| 08:14.11 | *** join/#asterisk gerhard7 (~gerhard7@82-169-24-72.ip.telfort.nl) |
| 08:18.33 | themrrobert | it looks like |
| 08:18.40 | themrrobert | its targeting itself |
| 08:18.49 | themrrobert | what is the ip in the register field |
| 08:19.00 | themrrobert | "Reg. Contact" |
| 08:20.36 | jkroon | asterisk -rx "sip show peer 333" | grep Status => Status : OK (1 ms) |
| 08:21.01 | themrrobert | whats result of this: |
| 08:21.10 | themrrobert | asterisk -rx "sip show peer 333" | grep Contact |
| 08:21.21 | jkroon | reg contact is blank, and To Host is 10.0.0.14 |
| 08:24.57 | themrrobert | is traffic possible routed incorrectly, ie traffic with invalid dst possibly being routed? |
| 08:25.23 | themrrobert | what happens when you ping 10.0.0.14 |
| 08:26.30 | Bradada | Hi guys, I just installed asterisk with AsteriskNOW last day, but I can't find the <agent.conf> file in the /etc/asterisk. Should I create one myself? |
| 08:27.07 | themrrobert | what version of asterisk is it |
| 08:27.45 | Bradada | It's Asterisk 11.2.1. |
| 08:27.56 | themrrobert | agent.conf is old |
| 08:29.23 | Bradada | So is there any equivalent setting file to agent.conf? Or newer version has new way to do that? |
| 08:29.26 | themrrobert | I cant completely remember how its been replaced at this time |
| 08:29.33 | themrrobert | new version does it different |
| 08:29.49 | themrrobert | agents are just sip users that join queues |
| 08:29.59 | themrrobert | or however else you want to use them |
| 08:30.56 | themrrobert | jkroon:- after pingtest, try sip unregister 333 and then see if it's still there |
| 08:32.44 | Bradada | Okay, thanks for the help robert. |
| 08:32.51 | jkroon | themrrobert, that's what i'm trying to explain - i explicitly routed traffic incorrectly because the peering is buggered. and it's a host=10.0.0.14 - NOT a dynamic. |
| 08:33.37 | jkroon | so ping at this point responds with destination network not reachable. |
| 08:33.52 | jkroon | my point is that Status *should* go to unreachable and not stay OK |
| 08:35.21 | themrrobert | I see. Well, I know that asterisk does "sip qualify" with OPTIONS, so if traffic were routed thru the asterisk box, and it got a sip reply from port 5060, which it looks like it did, then its going to think its available |
| 08:35.34 | themrrobert | it doesn't do additional authentication checking each qualify |
| 08:35.56 | themrrobert | although if you unregister it, i don't think asterisk should register as the phone.. |
| 08:36.12 | jkroon | themrrobert, it CANNOT receive a reply. |
| 08:36.34 | jkroon | don't worry, REGISTER *to* the server doesn't get a sensible response either, chan_sip seems to be in limbo. |
| 08:37.06 | jkroon | themrrobert, it's a IP<=>IP peering, there is no registering involved between the two ... |
| 08:37.15 | themrrobert | oh gotcha |
| 08:38.54 | themrrobert | But I guess I don't follow what sort of help you're looking for.. |
| 08:39.16 | jkroon | looking to try and figure out why sip qualify can't figure out that the peer is MIA |
| 08:39.29 | themrrobert | I thought you said you purposely rerouted traffic |
| 08:40.19 | jkroon | yes, but asterisk still reports the peer (in spite of qualify=yes) as status OK (1 ms) - which is obviously wrong. |
| 08:40.30 | jkroon | it should become unreachable after a while |
| 08:41.07 | themrrobert | right because you routed traffic to the box, so it sees that (but appearing on another ip)... |
| 08:41.11 | jkroon | with sip set debug on peer foo it shows the outbound OPTIONS frame, but never the returning frame. |
| 08:41.27 | themrrobert | how about with sip set debug on |
| 08:41.32 | themrrobert | turn it on globally |
| 08:41.42 | jkroon | dude, 400 sip peers ... |
| 08:41.50 | themrrobert | lol ok ik how that goes |
| 08:41.56 | themrrobert | its 12:40 am for me tho |
| 08:42.29 | jkroon | lol, i was close to that a few nights back. |
| 08:42.42 | jkroon | ok, with sip set debug on (globally) i still don't see the return traffic. |
| 08:42.59 | jkroon | heck, I don't see *any* SIP traffic for that matter! |
| 08:43.05 | themrrobert | tbh i'm not sure what you're supposed to see |
| 08:43.11 | themrrobert | in response to that |
| 08:43.44 | themrrobert | how exactly did you change the flow of ip traffic |
| 08:44.00 | jkroon | well, with 400 sip peers, of which 350 is currently registered i'd expect to see a bunch of outbound OPTIONS frames to qualify them ... for one. |
| 08:44.20 | jkroon | ip ro ad 10.0.0.14/32 via router.that.doesn't.know.how.to.get.to.10.0.0.14 |
| 08:44.47 | jkroon | normal route is ip ro ad 10.0.0.0/29 via m.p.l.s.route |
| 08:45.11 | themrrobert | do you ever get more than 1ms on your Status? |
| 08:46.09 | jkroon | not for that particular peer no. |
| 08:46.40 | jkroon | but a normal ping to it reports a 0.2ms round-trip anyway ... so i don't expect it to be higher than 1ms. |
| 08:46.53 | jkroon | sporadically, under really massive load i sometimes get like 2ms. |
| 08:49.57 | jkroon | ok, so there is a definite bug in the SIP stack in asterisk 11.2.1 |
| 08:49.59 | jkroon | udp 213056 0 0.0.0.0:5060 0.0.0.0:* 5707/asterisk |
| 08:50.33 | jkroon | that's the recv queue building up, so internally chan_sip stopped responding to the queue ... off to file bug number 3 in two days. |
| 08:52.24 | themrrobert | on my box i have a server plugged in directly via cat 6 cable to my pbx, its supposed to communicate via AMI, but its fubar'd, most events work properly, but some are missed or something, and the client ends up timing out. comparing the two logs was unbelieveably horrible, with no useful results. makes me want to bash something |
| 08:54.03 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:55.38 | jkroon | themrrobert, i hear you! |
| 08:55.55 | themrrobert | have u encountered similar? |
| 08:56.34 | jkroon | had some weird lockups with AMI, eventually found a bug in the code on my side that blocked on stuff it shouldn't have, refactored my code to queue the events instead in a somewhat clever way, problem fixed. |
| 08:57.11 | themrrobert | that sounds like more than ill be able to do =/ |
| 08:57.23 | themrrobert | never got into c as much as i should have probably |
| 08:57.29 | themrrobert | i'm sure i could do it, but i dont have the time |
| 08:59.54 | themrrobert | leifmadsen said he thought it might be locking (my problem) but i haven't more than a vague idea what that could mean |
| 09:00.38 | themrrobert | i assume a handle is locked so its unable to read for a period of time, till it just decides to start running again... |
| 09:00.56 | themrrobert | i need it fixed before i reconfigure everything on 1.8 |
| 09:01.07 | themrrobert | downsgrade 11.2.1 > 1.8 |
| 09:01.27 | jkroon | themrrobert, mail me - got a nifty class for C++ that might give you a few ideas. |
| 09:01.51 | jkroon | oi, you downgrading? |
| 09:34.08 | *** join/#asterisk bramgn (~bram@gw.hybrid-it.nl) |
| 09:48.28 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 09:48.28 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.2.1 (2013/01/22), 10.12.1 (2013/01/22), 1.8.20.1 (2013/01/22), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 09:52.20 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 09:52.20 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.2.1 (2013/01/22), 10.12.1 (2013/01/22), 1.8.20.1 (2013/01/22), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 09:52.25 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
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| 10:19.56 | linocisco | anyone used cisco RV042 with asterisk with Grandstream GXW410? |
| 10:20.17 | linocisco | I can't find how to QoS asterisk UDP traffic on it |
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| 11:17.08 | dipanjan | i have files with the same name but different formats (1.wav, 1.gsm, 1.ogg) in the same folder. Is there a way to make asterisk force choose o play only the gsm files? |
| 11:17.52 | Greenlight | I believe it trys to choose one that matches the codec of the channel it's playing it to |
| 11:18.08 | Greenlight | Althought that does seem somewhat buggy atm |
| 11:19.11 | dipanjan | Greenlight: It plays fine when I call from a SIP phone (twinkle) but when calling from a phone, it tries to play the ogg files and fails. |
| 11:19.36 | Greenlight | Delete the ogg file ? |
| 11:19.59 | dipanjan | I need it for preview under on a web-browser |
| 11:20.21 | Greenlight | On 11.2.1 for me at present for some reason Asterisk tries to play g729 when the channel is alaw. I think there is perhaps a bug in there |
| 11:20.39 | Greenlight | dipanjan: Put the .ogg in a different directory ? |
| 11:20.59 | dipanjan | Greenlight: one solution is to put them in separate folders, but was wondering if I can tell Asterisk to play only gsms. |
| 11:21.04 | dipanjan | Greenlight: yes |
| 11:28.08 | dipanjan | Greenlight: commenting out all other formats under [my-codecs] in /etc/asterisk/sip.conf seems to have done the trick |
| 11:29.15 | Greenlight | Ahh nice one |
| 11:29.31 | Greenlight | I wonder if the order there effects the priority of it playing them then |
| 11:40.57 | *** join/#asterisk kchehab (~david@77.42.241.66) |
| 11:41.56 | kchehab | is there a parameter can be edited to avoid the TLS disconnect when the user have a bad bandwidth ? |
| 12:07.25 | *** join/#asterisk davlefouAMD (~david@197.15.50.19) |
| 12:09.41 | *** join/#asterisk bviktor (~bviktor@unaffiliated/bviktor) |
| 12:09.42 | bviktor | yo |
| 12:09.49 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 12:09.49 | *** mode/#asterisk [+o file] by ChanServ |
| 12:10.19 | bviktor | is it possible to give an immediate busy signal to the caller of a ring group when all members of the ring group are busy? |
| 12:14.56 | *** join/#asterisk FireAndIce (~FireAndIc@175.100.158.250) |
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| 12:17.31 | kchehab | is there a parameter can be edited to avoid the TLS disconnect when the user have a bad bandwidth ? |
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| 12:49.58 | creativx | man |
| 12:50.07 | creativx | It sure is 3 years since last time I had to touch asterisk extensions |
| 12:50.33 | *** join/#asterisk lvlinux (~n1gg@c-50-142-165-230.hsd1.tn.comcast.net) |
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| 13:56.49 | leifmadsen | creativx: extensions.conf is the easy part! :) |
| 14:00.46 | jmetro | Dialplan is the funnerest part. |
| 14:09.37 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:09.37 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 14:20.07 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:25.45 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:44.16 | *** join/#asterisk mihamina (~mihamina@17.216.74.41-ip-dyn.orange.mg) |
| 14:46.37 | glaz | Cote yesterday, https://fbcdn-sphotos-e-a.akamaihd.net/hphotos-ak-prn1/59706_10151465897558416_1664211735_n.jpg |
| 14:46.44 | glaz | he's gonna be a huge 170 !!! |
| 14:47.13 | glaz | wrong window, sorry. |
| 14:47.17 | blitzrage | :) |
| 14:47.57 | glaz | my bad :( |
| 14:48.44 | blitzrage | I like UFC too, so I'm not offended :D |
| 14:48.57 | blitzrage | I am normally 180, but I don't look like that |
| 14:51.05 | glaz | hah, Cote is a friend of mine |
| 14:51.19 | glaz | I train BJJ with him |
| 14:51.36 | glaz | I'm a 205 and he rapes me in BJJ |
| 14:51.52 | *** join/#asterisk Xliff (~clifton@208.58.82.22) |
| 14:52.31 | blitzrage | nice |
| 14:52.44 | glaz | where you from blitzrage ? |
| 14:52.48 | blitzrage | Toronto |
| 14:52.52 | blitzrage | <-- normally leifmadsen btw |
| 14:52.57 | blitzrage | in case you recognize the name |
| 14:52.59 | glaz | Nice, did you go to UFC 157? |
| 14:53.04 | blitzrage | unfortunately I did not :( |
| 14:53.18 | blitzrage | I forget why... I think I had something else that night |
| 14:53.19 | glaz | 158 is in Montreal, I can't wait |
| 14:53.24 | blitzrage | indeed! |
| 14:53.33 | blitzrage | I do like watching on TV though, you get to see so much |
| 14:53.36 | glaz | are you a Leaf fan? :) |
| 14:53.40 | Xliff | Is there a list of Asterisk compatible phones out there? I'm looking on the wiki without much luck. |
| 14:53.41 | blitzrage | unfortunately yes :D |
| 14:53.55 | blitzrage | Xliff: Digium phones are pretty compatible :) |
| 14:53.57 | Xliff | I'm trying to determine if the Altigen IP 720 will work with Asterisk based systems. |
| 14:53.58 | glaz | heh they're doing good this year |
| 14:54.02 | blitzrage | otherwise, just stick to anything SIP |
| 14:54.10 | Xliff | blitz: LOL - Yes. I know. I'm drooling over the top end. |
| 14:54.14 | blitzrage | glaz: yep! lost to Montreal though :\ |
| 14:54.25 | blitzrage | has a D70 sitting in his cabinet that he hasn't setup yet : |
| 14:54.33 | Xliff | blitzrage: Problem - What one vendor advertises as SIP may not really BE SIP. |
| 14:54.38 | blitzrage | I should really use it to replace my Polycom IP335 |
| 14:54.47 | igcewieling1 | We are having a weird SIP issue between two Asterisk servers. Looks like there is an INVITE with a empty SDP, which I may be the problem. Would anyone have a min to look at http://pastebin.ca/2327029 especially line 33, and let me know if you see anything wrong? |
| 14:54.58 | blitzrage | Xliff: true story... I can confirm Panasonic, Polycom, Grandstream, Aastra, and Digium all work fine :D |
| 14:55.17 | blitzrage | igcewieling1: 404 Not Found |
| 14:55.22 | Xliff | Yeah. Those tend to be the more *ahem* "open" of the phones I have worked with. |
| 14:55.28 | glaz | And obviously Cisco phones too :) |
| 14:55.29 | Xliff | Altigen, however...... |
| 14:55.46 | blitzrage | Xliff: if you can just test, then you're likely ok. Asterisk is pretty forgiving to be honest. |
| 14:55.47 | igcewieling1 | blitzrage: maybe you have scripts disabled? works for me when I click on the link |
| 14:55.53 | Xliff | Put this way. This system is less than 5 years old and the phone server is STILL running XP. |
| 14:55.59 | blitzrage | igcewieling1: don't know... pastebin.ca has always worked |
| 14:56.10 | igcewieling1 | http://pastebin.ca/2327029 |
| 14:56.17 | Xliff | blitzrage: Already have a test server from Digium. So far... nothing. |
| 14:56.25 | Xliff | Can't get the bloody thing to run as an IP phone. |
| 14:56.30 | blitzrage | Xliff: oh, do you mean Switchvox? |
| 14:56.35 | Xliff | Yup |
| 14:56.47 | blitzrage | Switchvox doesn't strickly speaking == Asterisk |
| 14:56.55 | blitzrage | strictly* |
| 14:57.20 | *** join/#asterisk eschmidbauer (~chatzilla@cpe-69-204-102-218.buffalo.res.rr.com) |
| 14:57.23 | eschmidbauer | hi |
| 14:57.24 | Xliff | Asterisk-based* |
| 14:57.30 | Xliff | But yeah... I know what you mean. |
| 14:58.05 | Xliff | I think there is some kind of config file that the phones load and there are options there that are screwing us up. |
| 14:58.29 | igcewieling1 | try this one http://pastebin.com/3ZSZELkZ |
| 14:58.43 | Xliff | Switchvox text system saw no IP traffic even tough phone was set on 192.168.0.0 which is automatically NAT to our external IP. |
| 14:59.03 | eschmidbauer | haters gonna hate |
| 14:59.38 | igcewieling1 | Xliff: do you really want to use phones which don't have enough documentation to know if there is a config file? |
| 15:00.31 | Xliff | igcewieling1: You have a point. This is for work though so I have to show due dilligence before I tell 'em they are better off dumping a ~3 phone system and spend the money for a better one. |
| 15:09.12 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
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| 15:33.43 | bitwize | I need help defining an registration string for a SIP-trunk. The ITSP tells me to send "REGISTER sip:a.b.com" with "FROM: a.b.com@b.com" and "TO: a.b.com@b.com" but I cannot resolve this registration pattern. |
| 15:34.24 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
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| 15:39.29 | *** join/#asterisk AkkerKid (~AkkerKid@50.200.18.202) |
| 15:40.17 | AkkerKid | hey all.... what am I doing wrong? this should forward a call to extension 10060 to extension 1060: exten => _10XXX,1,Goto(from-internal,${${EXTEN}-9000},1) |
| 15:40.36 | AkkerKid | but instead it just loops in the cli and doesn't go to the right extension |
| 15:41.16 | WIMPy | That is literal. Use $[] or $MATH(). |
| 15:41.34 | AkkerKid | aha |
| 15:41.48 | AkkerKid | i knew it was going to be something that simple |
| 15:45.27 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:45.28 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:45.33 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
| 15:45.41 | *** join/#asterisk TSM (~the_softw@fw-lon1.wenn.com) |
| 15:45.43 | *** join/#asterisk TechSmurf (~jdaniel@unaffiliated/techsmurf) |
| 15:47.12 | blitzrage | AkkerKid: what WIMPy said; with the way you have it now you basically have requested the contents of a variable ${10060-9000} |
| 15:47.52 | jmetro | What exactly does func STAT() return if a file exists or does not exist? show func STAT doesnt say. |
| 15:48.02 | jmetro | stat -e i mean. |
| 15:48.18 | blitzrage | jmetro: I believe 0 or 1 |
| 15:48.30 | blitzrage | pretty simple test to run though :D |
| 15:49.05 | jmetro | True, but i have no phones hooked up to test dial with =) |
| 15:49.26 | WIMPy | channel originate ... |
| 15:49.32 | blitzrage | that* |
| 15:49.54 | blitzrage | execute a Local channel |
| 15:51.47 | jmetro | have never used originate before, but will look into it |
| 15:53.27 | AkkerKid | Question: when I originate a call in the CLI or AMI, it's doesn't show good callerid info. Am I not going through the right context? unfortunately i'm running elastix. |
| 15:53.47 | blitzrage | see #elastix I guess |
| 15:54.01 | ShoreTel | lol |
| 15:54.08 | AkkerKid | ...ugh |
| 15:54.12 | ShoreTel | define "good callerid |
| 15:54.14 | ShoreTel | " |
| 15:54.27 | Qwell | "ugh, I have to get help where people would actually know how to solve my problem" |
| 15:54.30 | WIMPy | That's normal. If you want something sensible, use a local channel and dialplan functions to set the caller ID. |
| 15:54.54 | blitzrage | ugh, I am using something crazy unique that I don't understand and don't like that people who use vanilla asterisk don't know how to help me |
| 15:54.58 | AkkerKid | when the call hits the receiving extension, it shows only what i defined in the originate |
| 15:55.04 | jmetro | hm. If i originate a channel to an autoattendant, how can I then dial my options? |
| 15:55.31 | blitzrage | originate it to dialplan that will execute dtmf presses on answer |
| 15:55.46 | navaismo | everytime someone say the word elastix a frog in the world dies. |
| 15:55.51 | AkkerKid | don't get your panties in a wad guys... i'm trying to get away from elastix. |
| 15:56.17 | blitzrage | jmetro: see D() option in Dial() |
| 15:56.19 | AkkerKid | I just don't yet have the skills to make it completely with a gi |
| 15:56.31 | blitzrage | AkkerKid: either do we |
| 15:56.55 | AkkerKid | completely without a gui* |
| 15:57.16 | navaismo | wonders which certification it's better dCAP or ECE |
| 15:59.57 | glaz | I don't know what ECE is but I did dCAP, if you already know how to install asterisk and create a dialplan, don't waste your time in a dCAP |
| 16:00.22 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:00.23 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 16:00.24 | jmetro | Blitzrage> I get you, code dialplan to execute testing of other dialplan. |
| 16:00.49 | blitzrage | indeed |
| 16:00.52 | blitzrage | like a boss |
| 16:01.28 | jmetro | Quite Θ ̨Θƪ |
| 16:01.45 | *** join/#asterisk linocisco (~linocisco@203.81.67.114) |
| 16:02.24 | linocisco | hi all |
| 16:02.35 | WIMPy | lo you |
| 16:02.54 | linocisco | anybody who has experience with asterisk + cisco RV042 and Grandstream GXW410 together? |
| 16:03.33 | linocisco | anybody who has experience with asterisk on QNAP + cisco RV042 and Grandstream GXW410 and cisco IP phone 7945G together? |
| 16:03.51 | navaismo | glaz, "Don't waste your time in a dcap"..??? |
| 16:03.56 | navaismo | glaz, why? |
| 16:04.09 | glaz | I thought I was pretty clear.. |
| 16:04.25 | blitzrage | well, if you know how to do that stuff, then getting your dCAP to prove it then is useful :) |
| 16:04.26 | glaz | read before just before that statement |
| 16:04.34 | jmetro | in some ways it can be better to pile on as many certifications as you have time to take tests for |
| 16:04.37 | navaismo | glaz, ECE-->Elastix Certified Engineer |
| 16:04.41 | blitzrage | doing the training is different then writing the certification |
| 16:04.44 | navaismo | blitzrage, my point |
| 16:04.48 | glaz | Elastix, ewww. |
| 16:04.53 | navaismo | vomits |
| 16:05.05 | navaismo | and two frogs died |
| 16:05.20 | glaz | maybe it's just me, but certificates don't prove anything where I work |
| 16:05.25 | blitzrage | if you are comfortable with Asterisk, then writing the dCAP can be useful. If you need the training before taking dCAP, then that can be additionally useful |
| 16:05.34 | blitzrage | glaz: it depends where you work then |
| 16:05.37 | linocisco | navaismo, Elastix does not comply original objective of asterisk. It modified its code to be totally different from asterisk |
| 16:05.43 | blitzrage | that doesn't make the dCAP useless, just useless in your situation |
| 16:05.46 | glaz | I've meet CCIE guys that I could school, my CCNA has expired 8 years ago |
| 16:05.56 | navaismo | so i have experience using asterisk but without a dcap no one take me serious |
| 16:06.08 | glaz | navaismo: really? where do you live? |
| 16:06.10 | blitzrage | write a book, it worked for me |
| 16:06.19 | navaismo | glaz, Mexico |
| 16:06.35 | linocisco | glaz, I am CCNA with 1000/1000 with no school training. |
| 16:06.35 | blitzrage | I may be biased though, I helped write the original training course and dCAP cert |
| 16:06.41 | glaz | funny, all the guys in my dCAP training week were from Mexico |
| 16:06.46 | jmetro | blitzrage =D the definitive book would definitely help credibility |
| 16:06.48 | blitzrage | don't worry, I have the largest penis in here |
| 16:06.57 | blitzrage | that's what we're doing right? |
| 16:06.58 | glaz | blitzrage: :p |
| 16:07.19 | blitzrage | navaismo: if the dCAP could be useful for you, then by all means take it :) |
| 16:07.31 | glaz | how much is the dCAP now? |
| 16:07.43 | linocisco | glaz, 300 USD |
| 16:07.43 | blitzrage | unknown |
| 16:07.50 | linocisco | glaz, dCAA is free |
| 16:07.52 | blitzrage | that's a pretty cheap cert |
| 16:07.56 | navaismo | last time here only the exam was 600USD, so I need to save a lot of money |
| 16:08.07 | navaismo | :'( not for me |
| 16:08.10 | blitzrage | you can also take it at astricon |
| 16:08.11 | glaz | what? why do I remember my job paying like 3500$ for this |
| 16:08.19 | blitzrage | glaz: because you took the training course |
| 16:08.23 | blitzrage | dCAP != training |
| 16:08.26 | navaismo | glaz, maybe you pay the full course & exam |
| 16:08.28 | glaz | aaa |
| 16:08.30 | blitzrage | right |
| 16:08.40 | glaz | yeah probably, I don't remember :\ |
| 16:08.49 | blitzrage | well, the dCAP is not a week long exame |
| 16:08.50 | blitzrage | exam* |
| 16:08.54 | blitzrage | you probably remember that part |
| 16:08.55 | ShoreTel | lol |
| 16:09.00 | glaz | I do :) |
| 16:09.10 | blitzrage | dCAP itself is a 3 hr test |
| 16:09.13 | ShoreTel | my asterisk box outwit's your asterisk box |
| 16:09.19 | glaz | I remember helping the teacher helping others, but nothing much more |
| 16:09.20 | ShoreTel | that's what we're doing here. |
| 16:09.22 | blitzrage | at least from what I remember |
| 16:09.25 | eirirs | my asterisk box are invisible. I win. |
| 16:09.33 | ShoreTel | mine are invincible! |
| 16:09.38 | ShoreTel | xxxoops :/ |
| 16:09.41 | blitzrage | jokes on you, I switched to yate years ago |
| 16:10.10 | glaz | I also remember the guys from Mexico, they were like 6, hangover every morning |
| 16:10.34 | glaz | they really enjoyed Montreal and its hookers |
| 16:10.35 | linocisco | anybody who has experience with asterisk on QNAP + cisco RV042 and Grandstream GXW410 and cisco IP phone 7945G together? |
| 16:11.02 | eirirs | RV042 are not cisco, but linksys :P |
| 16:11.21 | glaz | looks at his phone, 7940, sorry |
| 16:11.38 | linocisco | eirirs, but name is cisco, it was acquired by cisco, now by Belkin or who knows. I dont care whoever acquire it |
| 16:12.03 | jmetro | who here would have a heart attack if you saw "belkin acquires cisco" on tech news |
| 16:12.30 | linocisco | jmetro, it is not amazing news. I have known |
| 16:12.42 | eirirs | linocisco: "cisco SMB" :P |
| 16:12.52 | eirirs | huuuge difference |
| 16:12.57 | igcewieling1 | jmetro: not me, I just shrugged and said to myself "At least people won't think Linksis boxes are real Cisco boxes anymore" |
| 16:13.03 | linocisco | eirirs, yes |
| 16:13.26 | igcewieling1 | "We have a CISCO router!" "No, you have a fscking Linksys consumer POS, now get off my lawn!" |
| 16:13.46 | jmetro | All i know is that Linksys used to be the only alternative to Netgear for home routing, but now I go d-link |
| 16:14.11 | eirirs | d-link??? booo |
| 16:14.13 | igcewieling1 | jmetro: I don't really hate Linksys, what I hate is Cisco calling their Linksys boxes "Cisco" |
| 16:14.21 | glaz | dlink? |
| 16:14.27 | glaz | vomits |
| 16:14.39 | eirirs | igcewieling1: +1 |
| 16:14.43 | linocisco | jmetro, though it is not discussion about products, anyway. linksys, or Belkin or Netgear or D-link aall are SOHO |
| 16:15.06 | linocisco | the worst models are ProLink, TP-Link |
| 16:15.26 | eirirs | TP-link mediaconverter are okay |
| 16:15.27 | eirirs | hehe |
| 16:15.33 | eirirs | cheapo and works |
| 16:15.44 | linocisco | eirirs, fibre media converter? |
| 16:15.47 | eirirs | yep |
| 16:16.26 | linocisco | eirirs, I would not go with crappy fibre media converter, I would use SFP module on switch port |
| 16:16.29 | linocisco | instead |
| 16:16.46 | eirirs | hehe agreed on that, im waiting for a SFP module incoming |
| 16:16.59 | eirirs | though, it works wonder as temporary solution |
| 16:17.41 | linocisco | hi all bros.let's get down to nitty gitty. I have very doubtful questions on asterisk related to be done in limited time |
| 16:18.21 | linocisco | I have asterisk installed on QNAP as my office wont allow me to install on normal PC. |
| 16:19.16 | igcewieling1 | moves slowly away from linocisco |
| 16:20.46 | linocisco | I have Grandstream GXW410 for two CO land lines to be used with asterisk and cisco RV042 and Cisco 7942G IP Phone |
| 16:21.49 | linocisco | cisco RV042 is meant a router or gateway between internet and LAN. or Dual WAN Router if I have another internet line. Calling in and out will be only made through CO Lines |
| 16:22.11 | linocisco | Searching on internet said Cisco RV042 blocks all SIP registration |
| 16:22.37 | linocisco | I have no other routers to be used because it is meant for Dual WAN. |
| 16:22.56 | igcewieling1 | linocisco: buy a new router. |
| 16:23.03 | eirirs | did you disable sip alg and sip whatever functions in that web gui thingy? |
| 16:23.18 | linocisco | eirirs, I didn't find |
| 16:23.26 | igcewieling1 | eirirs: I think this is one of the models where you cannot disable SIP ALG and SPI |
| 16:23.56 | eirirs | igcewieling1: LOL fail |
| 16:24.09 | igcewieling1 | linocisco: you know the RVXXX routers are really Linksys and not actual Cisco routers right? |
| 16:24.38 | igcewieling1 | elguero: I've run across them a couple of times, each time the ONLY solution was to replace the router. |
| 16:24.45 | linocisco | igcewieling1, I am CCNA with 1000/1000. I know that pretty well. I have no choice in my country |
| 16:24.49 | linocisco | in my office |
| 16:25.17 | igcewieling1 | linocisco: I wish you the best of luck, but I doubt anyone here can help you. |
| 16:25.18 | eirirs | I had to replace one RV220 with a 1841, because it somehow decided to run 0.0.0.0 0.0.0.0 gw 255.255.255.0 instead of 255.255.255.248, and fucked up alot, and that route table was NOT editable |
| 16:25.25 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 16:25.44 | eirirs | even that I insert 255.255.255.248 in appropiate WAN input fields |
| 16:25.50 | eirirs | screw it |
| 16:25.55 | linocisco | eirirs, but it is funny it said RVxxx are being run on linux opensource |
| 16:26.29 | eirirs | would love to see a dd-wrt or openwrt or whatever at RV220, which are a piece of neat hw |
| 16:26.50 | eirirs | that linksys-webgui are a joke |
| 16:27.04 | linocisco | eirirs, can we flash RV042 with Openwrt or tomato firmware to still have the same functionality? |
| 16:27.04 | jmetro | thats what i love about my dlink. the webgui is a++ |
| 16:27.17 | eirirs | linocisco: why not? :) |
| 16:27.30 | eirirs | linocisco: you even gets MORE functionality |
| 16:28.00 | eirirs | opensource ppl create sw with love, linksys only "meh, I have to finish that thing up, and get paid" |
| 16:28.00 | linocisco | eirirs, that is cool. As long as I dont lose Dual WAN load balancing functionality+DHCP server+ Routing feature, I am fine with any firmware on That damn RV042 |
| 16:28.23 | eirirs | linux support load balancing +++ |
| 16:30.13 | linocisco | eirirs, I love only opensource. not proprietary |
| 16:30.22 | eirirs | :) |
| 16:30.35 | eirirs | then change fw asap |
| 16:31.08 | linocisco | eirirs, but now feeling like writing WINE code to reverse engineer |
| 16:31.19 | eirirs | haha |
| 16:31.58 | linocisco | eirirs, I got 4 cisco IP Phones 7942G. which comes with SCCP native cisco protocol to be used by costly call manager. |
| 16:32.11 | eirirs | mm |
| 16:32.30 | eirirs | I changed all phones to SIP fw |
| 16:32.36 | linocisco | eirirs, I dont want to flash it with sip firmware because if I can't reverse it to SCCP. I will be fired |
| 16:32.42 | eirirs | you CAN |
| 16:32.43 | eirirs | no problem |
| 16:33.04 | eirirs | just have a tftp server running, and dhcp option 150 pointing to that ip |
| 16:33.07 | linocisco | because i have faced the same problem with Nortel IP phone 1140E which was converted to SIP phone |
| 16:33.18 | eirirs | and have both SCCP fw available there, just in case you need it |
| 16:33.30 | eirirs | well, you won't have problem with cisco phones |
| 16:33.34 | eirirs | I can't say about nortel |
| 16:33.39 | linocisco | eirirs, then i see it is 30 days eval version. I lost my way and can't change it back to Untsim firmware |
| 16:33.49 | eirirs | lol ouch |
| 16:34.04 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:35.14 | linocisco | eirirs, I was nothing in my office. I was blamed and almost had to pay it back for my flashed nortel ph |
| 16:35.47 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 16:36.13 | eirirs | mistakes can make it more costy than having a proprietary cisco phone system hehe |
| 16:36.25 | jmetro | or just get phones that are SIP |
| 16:36.37 | eirirs | hehe |
| 16:39.04 | igcewieling1 | hugs his Polycoms |
| 16:39.35 | linocisco | eirirs,grandstream , yeallink ? what is good enterprise phone? |
| 16:40.09 | linocisco | eirirs, I want license free phone |
| 16:40.30 | jmetro | cisco sip phones are nice. Snoms have some very nice features. Polycom 335's are good for simple desk phones. |
| 16:40.55 | igcewieling1 | ciscos are not license free are they? |
| 16:41.07 | linocisco | jmetro, Cisco produce pure SIP phones?? namely? |
| 16:41.26 | *** join/#asterisk ghost751 (~trechber@dslb-178-010-047-186.pools.arcor-ip.net) |
| 16:41.38 | jmetro | SPA509's |
| 16:41.59 | linocisco | jmetro, it is ATA . not phone, right? |
| 16:43.10 | jmetro | I'm not sure what you mean |
| 16:43.49 | ghost751 | any idea what an "intercept" button on the phone is doing? |
| 16:44.39 | *** join/#asterisk Opperior (~chatzilla@mailhost.lannetwork.com) |
| 16:45.49 | linocisco | eirirs, I looked at openwrt website. but not found Cisco RV042 is supported |
| 16:45.56 | eirirs | linocisco: I use cisco phones |
| 16:46.25 | linocisco | eirirs, how to use? by converting it to SIP ? |
| 16:46.25 | eirirs | no license needed, thats just for cisco's call manager |
| 16:46.39 | eirirs | I just explained you |
| 16:46.48 | eirirs | 173304 < eirirs> just have a tftp server running, and dhcp option 150 pointing to that ip |
| 16:47.59 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:47.59 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:48.17 | linocisco | eirirs, our 4 cisco 7942G phones are sent to us by HQ to be used with Call manager. but as we have only Nortel BCM450. Those are now extra and I want to use them with asterisk. After Cisco CUCM arrive, I have to use them with CUCM. |
| 16:48.58 | eirirs | yes, with tftp and both cisco sip and sccm firmwares available on the tftp server you will have no problem changing back and forth |
| 16:49.01 | eirirs | at those cisco phones |
| 16:49.36 | linocisco | eirirs, do I have to install both fimrware SCCP and SIP at the same time. ? |
| 16:49.54 | ghost751 | can be tricky to change sip<->sccp |
| 16:49.58 | eirirs | ..... no |
| 16:50.11 | eirirs | ghost751: what do you mean? |
| 16:50.15 | linocisco | http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a0080094584.shtml shows it is possible |
| 16:50.33 | ghost751 | dependencies when flashing fw |
| 16:50.59 | linocisco | but I have no confidence as I have Cisco RV042 which most said, will block SIP, and my previous Nortel Phone was destroyed with trial SIP version |
| 16:51.42 | eirirs | linocisco: I bet you can get those nortel back, just google about howtos |
| 16:52.10 | linocisco | eirirs, I waa searching every single where |
| 16:53.01 | linocisco | not worked for me |
| 16:54.35 | linocisco | I have 3 problems. (1. no openwrt firmware for Cisco RV042, 2. cisco IP SCCP <--> SIP ok or not? 3. Nortel Phone's Untsim can't be installed successfully) |
| 16:55.27 | eirirs | I'm 110% sure there are working solutions :) |
| 16:56.06 | igcewieling1 | There are two types of "Cisco" phones. Linksys (SPA series) which are branded by Cisco but do not are not really "Cisco platform", IIRC firmware is free for those phones. There are also Cisco phones based on the Cisco platform and designed for Call Manager, IIRC firmware costs money for these phones. |
| 16:56.35 | *** join/#asterisk vfabi (~fabi@ip-de80.d-net.kiev.ua) |
| 16:57.31 | eirirs | I have SPA501G, 7975G, 7940G |
| 16:57.33 | eirirs | they all works nice |
| 16:58.57 | linocisco | eirirs, what type of Cisco 7942G is ? free or not? |
| 16:59.09 | *** join/#asterisk ipiera (~Paul@ipiera.plus.com) |
| 16:59.29 | igcewieling1 | <PROTECTED> |
| 17:00.27 | linocisco | igcewieling1, so if I install firmware from sip-->SCCP, it is not automatically ok to connect with call manager? need license? |
| 17:01.29 | eirirs | linocisco: phones are free to use. using call manager is not free. |
| 17:02.36 | linocisco | eirirs, we will have license for call manager by HQ by default. but as I am going to flash the phone into sip mode, if I changed it back to SCCP fimrware, can I have problem connecting call manager? |
| 17:02.48 | igcewieling1 | The nice thing about Asterisk, as compared to Call Manager is that you don't need to use scientific notation when quoting the price of an Asterisk install. |
| 17:03.29 | eirirs | linocisco: Nope. |
| 17:03.42 | linocisco | igcewieling1, as I have never done asterisk + cisco business, I dont know the power. but asterisk is free . no TCO as far as I am sure |
| 17:04.05 | eirirs | linocisco: I even flashed one 7975g back to SCCP to experiment with asterisk, then flashed back to SIP again |
| 17:05.01 | linocisco | that is why I am going to die to use asterisk day by day. but my office is not fully aware. HQ is not changing/not recommending their standard from cisco to opensource |
| 17:05.28 | linocisco | eirirs, u use chan-SCCP driver which I heard is no need to flash Cisco Phone to SIP |
| 17:05.39 | linocisco | eirirs, u use chan-SCCP driver ??which I heard is no need to flash Cisco Phone to SIP |
| 17:06.07 | eirirs | I didn't succed, thats why I changed back to SIP |
| 17:10.29 | *** part/#asterisk Opperior (~chatzilla@mailhost.lannetwork.com) |
| 17:15.16 | linocisco | ok bro |
| 17:15.34 | linocisco | eirirs, let me find one on svn.digium.com |
| 17:16.46 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 17:24.05 | eirirs | not now, I've things to do here |
| 17:31.07 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:33.01 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 17:41.00 | linocisco | http://svnview.digium.com/svn/asterisk/trunk/channels/chan_skinny.c?revision=382204&view=markup |
| 17:51.41 | *** join/#asterisk Atreiu (56d510a9@gateway/web/freenode/ip.86.213.16.169) |
| 17:52.22 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 17:52.56 | Atreiu | Hi all, when a user disconnects from his SIP extension, his status in queues stays at Not in use instead of unaivalable. Is this a bug? |
| 17:54.18 | acidfoo | eirirs, what version of asterisk do you use |
| 17:56.06 | acidfoo | anyone is using an SCCP Cisco phone 7911 or 7906 and is using the GListen/GLoff softkey ? |
| 17:56.52 | navaismo | Atreiu, can you show us the cli output for sip show peer & queue show & core show hints? |
| 17:57.08 | navaismo | Atreiu, also which version are you using? |
| 17:57.42 | Atreiu | navaismo: Asterisk 11.2.0 built by root @ pabx on a x86_64 running Linux on 2013-01-17 17:00:41 UTC |
| 17:57.55 | navaismo | nice |
| 17:58.20 | Atreiu | !pb |
| 17:58.23 | Atreiu | damn |
| 17:58.46 | Atreiu | navaismo: I'm pastebining the commands |
| 17:59.39 | navaismo | k |
| 18:01.58 | Atreiu | navaismo: http://pastebin.com/qTd4Mrqs |
| 18:02.10 | Atreiu | Hope it helps... Desesperated. |
| 18:02.58 | Atreiu | As you can see, extensions 110 & 113 are offline, but still marked as available in queues. |
| 18:03.07 | navaismo | seeing |
| 18:04.42 | [TK]D-Fender | Atreiu: show us how you added them as members of your queue, also the queue configs, etc |
| 18:04.46 | navaismo | right, hints and queue marked are not marked correctly. Do you use a GUI or custom dialplan? seems like freepbx |
| 18:04.46 | jmetro | sip show peers like them |
| 18:04.59 | jmetro | queue show |
| 18:05.17 | jmetro | and i thought you had ti add them as SIP/### not local/### |
| 18:05.28 | Atreiu | navaismo: Yes, I use FreePBX |
| 18:06.51 | [TK]D-Fender | jmetro: You can add whatever device you want |
| 18:06.52 | Atreiu | [TK]D-Fender: These are added through FreePBX as static agents. |
| 18:08.20 | Atreiu | Ow, maybe if I tick Generate Device Hints option in queues? |
| 18:08.45 | Atreiu | I don't understand what hints really are... |
| 18:10.04 | igcewieling1 | I imagine someone on #FreePBX would know more about how FreePBX sets up FreePBX queues. |
| 18:10.25 | navaismo | nope no one answer there |
| 18:10.29 | Atreiu | Nobody awnser in this channel... :( |
| 18:11.16 | jeffspeff | Atreiu, we typically use the config files instead of the freepbx gui or any other gui |
| 18:11.55 | [TK]D-Fender | Atreiu: What version of * is this? |
| 18:12.51 | navaismo | 11.2.0 |
| 18:17.08 | navaismo | Atreiu, what is the lie to add your agents: 110,0 or A110,0 or S110,0? |
| 18:17.17 | navaismo | s/lie/line/ |
| 18:21.23 | *** part/#asterisk ipiera (~Paul@ipiera.plus.com) |
| 18:26.09 | acidfoo | HO yeahh ! |
| 18:26.19 | acidfoo | the answer is not in the protocole, it is in the xml conf file ! |
| 18:26.34 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 18:26.37 | acidfoo | i've been banging my head on it for a couple of hours before realizing it |
| 18:26.56 | acidfoo | <enableGroupListen>true</enableGroupListen> |
| 18:27.54 | linocisco | using my Nortel AVaya 1140E SIP phone, after pressing extension, I need to press soft button to ring to another ext. what is the problem? |
| 18:28.10 | linocisco | using my Nortel AVaya 1140E SIP phone, after pressing extension, I need to press soft button"send" to ring to another ext. what is the problem? |
| 18:28.45 | navaismo | phone's dialplan |
| 18:29.00 | navaismo | pattern* |
| 18:29.34 | linocisco | dont know |
| 18:30.08 | ChannelZ | Anyone used the Google Talk app on Android? |
| 18:30.29 | leifmadsen | yep |
| 18:30.31 | leifmadsen | works well |
| 18:30.35 | newtonr | ChannelZ: i do |
| 18:31.06 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:31.16 | igcewieling1 | linocisco: that wasn't a question, it was a statement. The issue is with the dialplan configured on your phone. |
| 18:31.21 | linocisco | leifmadsen, are u real leifmadsen ? |
| 18:31.32 | ChannelZ | oh.. actually I think this is a problem with routing/firewall |
| 18:31.37 | newtonr | linocisco: he's an imposter, don't let him trick you |
| 18:31.45 | leifmadsen | true story |
| 18:31.46 | ChannelZ | nevermind |
| 18:31.56 | leifmadsen | that guy is too awesome to not want to be his online persona |
| 18:32.00 | linocisco | ok |
| 18:32.18 | leifmadsen | (yes, I'm the real leif madsen) |
| 18:32.27 | linocisco | he is hot |
| 18:32.45 | linocisco | on fire |
| 18:32.48 | igcewieling1 | linocisco: I think so too, but the poor thing likes women. |
| 18:32.53 | Atreiu | navaismo 110,0 |
| 18:33.07 | Atreiu | I reconnect through Android |
| 18:33.13 | leifmadsen | o.O |
| 18:33.17 | igcewieling1 | 8-) |
| 18:33.22 | leifmadsen | heh |
| 18:33.44 | linocisco | he failed in fastest dialtone test judged by David Duffett |
| 18:34.05 | leifmadsen | well, I didn't win :) |
| 18:34.07 | leifmadsen | I was also drunk |
| 18:34.26 | leifmadsen | I'm not using that as an excuse though :) |
| 18:34.33 | leifmadsen | some people are pretty clever |
| 18:34.49 | linocisco | who was that guy you are defeated |
| 18:34.56 | leifmadsen | Jared Smith |
| 18:35.38 | linocisco | leifmadsen, since i sent you that youtube link, you quited from my IM list on google talk |
| 18:35.56 | linocisco | leifmadsen, until noow |
| 18:36.04 | *** join/#asterisk Atreiu (~Atreiu@AToulouse-653-1-417-169.w86-213.abo.wanadoo.fr) |
| 18:36.06 | leifmadsen | not sure... I don't always have it online |
| 18:36.22 | Atreiu | Hello back navaismo |
| 18:36.54 | Atreiu | Son, i just put 110,0 not with A or S. |
| 18:37.06 | linocisco | leifmadsen, i like the way you yelled "Ahhh..at keystrokes" in that contest. |
| 18:37.18 | Atreiu | Some of m agents have penalties |
| 18:37.29 | leifmadsen | I do stuff like that when I work at home |
| 18:38.01 | linocisco | leifmadsen, why dont u come to asia to teach asterisk. David Duffett is coming to Malaysia in april |
| 18:38.02 | ChrisInSydney | Hi all. I'm having a challenge compiling Dahdi 2.6.1 against a 3.7.10 Kernel running on a linode VPS |
| 18:38.21 | leifmadsen | because I'm not an asterisk trainer? :) and it's not my job. |
| 18:38.38 | leifmadsen | ChrisInSydney: does the linode VPS allow loading of kernel modules? |
| 18:38.45 | leifmadsen | you can't load dahdi into AWS and other such things |
| 18:39.02 | ChrisInSydney | FATAL: Error inserting dahdi (/lib/modules/3.7.10-x86_64-linode30/dahdi/dahdi.ko): Invalid module format |
| 18:39.03 | ChrisInSydney | <PROTECTED> |
| 18:39.06 | navaismo | Atreiu, have you tried with S+extension,penalty? |
| 18:39.09 | ChrisInSydney | thats the error, |
| 18:39.20 | ChrisInSydney | good question leifmadsen |
| 18:39.58 | leifmadsen | ChrisInSydney: ya that's likely the problem -- you can't load kernel modules |
| 18:40.05 | leifmadsen | because it's a shared kernel |
| 18:40.18 | navaismo | Atreiu, using onli number,penalty fpbx add with this--> 3000 (Local/3000@from-queue/n) (Unavailable) has taken no calls yet |
| 18:40.37 | Atreiu | navaismo: No. What's that? |
| 18:40.43 | ChrisInSydney | bugger. Its a xen platofrm |
| 18:40.47 | navaismo | Atreiu, using S+extens,penalty fpbx adds like this-->SIP/3000 (Unavailable) has taken no calls yet |
| 18:41.01 | *** join/#asterisk pbxbrian (~pbxbrian@unaffiliated/brian98) |
| 18:41.14 | Atreiu | Aaaah okay! I'm trying this right noz |
| 18:41.19 | Atreiu | Now* |
| 18:41.33 | Atreiu | So something like S110,0 ? |
| 18:41.38 | navaismo | right |
| 18:42.08 | Atreiu | If it works, I'll kiss you feet :D |
| 18:43.12 | navaismo | Atreiu, http://02varvara.files.wordpress.com/2011/04/01-how-about-no-bear.jpg?w=800 |
| 18:43.24 | ChrisInSydney | Thanks leifmadsen. I've posted another update to the support ticket. Its 5:30am here so I'm struggling a bit. But got VUC to keep me awake :-) |
| 18:43.41 | ChrisInSydney | 5:45 :-/ |
| 18:45.57 | ChrisInSydney | module load and unload is supported leifmadsen |
| 18:52.57 | *** join/#asterisk areski (~areski@80.174.255.57.dyn.user.ono.com) |
| 18:56.39 | *** join/#asterisk djacob (~IceChat77@pool-96-227-231-204.phlapa.fios.verizon.net) |
| 18:57.57 | linocisco | anybody who has experience with asterisk on QNAP + cisco RV042 and Grandstream GXW410 and cisco IP phone 7945G together? |
| 18:58.10 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:58.10 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:09.03 | [TK]D-Fender | linocisco: You could pick a smaller demographic please? |
| 19:09.39 | linocisco | what do u mean? |
| 19:10.27 | [TK]D-Fender | linocisco: What schmuck is going to have ALL THREE of those things **together** specifically? |
| 19:10.57 | linocisco | [TK]D-Fender, my QNAP has only Asterisk 1.4 not upgradable. no CLI. available. Cisco RV042 is known as blocking SIP regisration. Cisco Phone comes with SCCP |
| 19:11.42 | *** join/#asterisk feeshon (~gaston@ool-45787011.dyn.optonline.net) |
| 19:11.45 | [TK]D-Fender | No CLI? On a dead branch? Talking about a router that blcoks SIP ... and then using a phone spcecifically in SCCP mode? |
| 19:11.52 | [TK]D-Fender | This is an insane combination |
| 19:12.06 | feeshon | Rescheduling destruction for 10000 ms |
| 19:12.07 | igcewieling1 | [TK]D-Fender: He is set up to fail, I'm not wasting any more time on it. |
| 19:12.15 | feeshon | Can anyone help me with this I am getting a Rescheduling destruction for 10000 ms |
| 19:12.27 | [TK]D-Fender | feeshon: How is that a problem? |
| 19:12.30 | feeshon | in my console....looks like a call won't hang up properly |
| 19:12.34 | igcewieling1 | feeshon: that should be a harmless message unless you have an actual problem. |
| 19:12.43 | [TK]D-Fender | feeshon: Perhaps you should show us .. the call. |
| 19:12.45 | [TK]D-Fender | ~pbv |
| 19:12.46 | [TK]D-Fender | ~pb |
| 19:12.47 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:12.49 | [TK]D-Fender | ^^^ |
| 19:13.09 | feeshon | We are using FOP and and it displays that a call is open when it really isn't |
| 19:13.22 | [TK]D-Fender | the FOP has a problem |
| 19:13.35 | [TK]D-Fender | Not Asterisk |
| 19:13.43 | [TK]D-Fender | Chech their support resources |
| 19:14.39 | feeshon | FOP looks fine. I see that issue warning in the asterisk console |
| 19:14.54 | [TK]D-Fender | feeshon: Show us the call. |
| 19:14.58 | feeshon | The call isn't techincally hung up and fop is finding it as still on the line. |
| 19:15.20 | [TK]D-Fender | feeshonWe are using FOP and and it displays that a call is open when it really isn't <-- apparently not it really "is" open |
| 19:15.23 | [TK]D-Fender | now* |
| 19:15.23 | *** join/#asterisk jkroon (~jkroon@105.240.94.113) |
| 19:15.55 | igcewieling1 | feeshon: have you confirmed the call is actually still active with "core show channels" in the CLI? |
| 19:16.00 | jmetro | same => n,hangup then |
| 19:16.06 | *** join/#asterisk ageis (kevin@67.222.146.23) |
| 19:16.24 | [TK]D-Fender | jmetro: .... |
| 19:16.38 | [TK]D-Fender | feeshon: Show us the call |
| 19:16.49 | ChannelZ | and then show me some cookies |
| 19:17.05 | *** join/#asterisk cchhat01 (~cchhat01@207-237-28-173.c3-0.elm-ubr2.qens-elm.ny.cable.rcn.com) |
| 19:18.07 | ageis | I have a Park() implementation in 1.6 that relies on priorityjumping to Park an unanswered call momentarily and then Dial again. I'm interested in upgrading Asterisk but wondering what the alternative to this would be in 1.8, same functionality but without priorityjumping. |
| 19:18.21 | feeshon | yes |
| 19:18.25 | feeshon | It's still active |
| 19:18.30 | [TK]D-Fender | feeshon: Show us the call |
| 19:18.31 | feeshon | give me one sec for a pastie |
| 19:19.22 | ageis | here's the bit: http://pastebin.com/b0FmpXFG |
| 19:20.09 | feeshon | http://pastiebin.com/5130ff233ba75 |
| 19:20.17 | feeshon | the 7276 is still active |
| 19:20.29 | ageis | I have found that without jumping to priority 6, the Dial after Park just hits a busy signal and doesn't work as expected. |
| 19:20.35 | ChrisInSydney | Hey. Anyone help with a DAHDI compile on a Kernel 3.7.10 running on a linode VPS ? |
| 19:20.48 | [TK]D-Fender | feeshon: Show us the actual call. Actually arriving, actually hanging up. And then actually lingering |
| 19:20.54 | ChrisInSydney | FATAL: Error inserting dahdi (/lib/modules/3.7.10-x86_64-linode30/dahdi/dahdi.ko): Invalid module format |
| 19:21.10 | ChrisInSydney | implies that the source is wrong maybe ?? :-/ |
| 19:21.25 | ChrisInSydney | kernel source DAHDI was compiled against |
| 19:21.32 | ChannelZ | ageis: what's really wrong with what you're doing now? |
| 19:21.35 | feeshon | not sure how to get that |
| 19:21.49 | ageis | ChannelZ: it works great. I'm talking about wanting to upgrade Asterisk and I'll lose this functionality, because of priorityjumping. |
| 19:21.50 | jmetro | watch the console while it comes in |
| 19:21.51 | feeshon | This was something that was already there |
| 19:21.51 | ChannelZ | sorry I'm lagging |
| 19:22.06 | feeshon | Wasn't watching the console |
| 19:22.10 | ageis | ChannelZ: So wondering if I can do the same thing without. |
| 19:22.10 | [TK]D-Fender | feeshon: what does that even mean? Look at THEC ALL from beginning to end |
| 19:22.28 | [TK]D-Fender | feeshon: If you're not looking in CLI then you aren't really looking. |
| 19:22.33 | [TK]D-Fender | feeshon: SIP DEBUG <-- |
| 19:22.42 | jmetro | ageis: What are you trying to accomplish, see if it can be done without priority jumping? |
| 19:22.54 | ageis | jmetro: Precisely. |
| 19:23.10 | jmetro | ageis: I know, but i meant, what are you trying to do with that code anyway |
| 19:23.15 | ChannelZ | ageis: not sure what you mean that you will lose this. |
| 19:23.29 | ageis | Because priorityjumping is deprecated in 1.6 (my version) and removed altogether in 1.8. |
| 19:23.31 | ageis | Let me describe |
| 19:23.53 | ChannelZ | Also, is the actual parking lot important? IE is it possible someone who gets parked in this fashion would get picked up by someone while they are in the lot, or are you really just using it as a delay? |
| 19:23.53 | feeshon | I wasn't looking at the CLI when it originally happened....there is an active call that hasn't been removed. |
| 19:23.59 | ageis | It's intended to call a location, ring for 45s, and if no one picks up the call is parked for a while. Then it rings again. If no one picks up then it rings another location. Finally it goes to voicemail. |
| 19:24.17 | ageis | The purpose is so that if someone is busy or on the other line, they have a chance to answer the parked call, because it lights up on the phone. |
| 19:24.55 | jmetro | Have your park timeout to the second location |
| 19:25.06 | ChannelZ | ageis: you're not using priority jumping. It's a function of the Park application as to where it returns to the dialplan. The example you showed seems like it should still work just fine |
| 19:25.07 | ageis | timeout to another context? |
| 19:25.27 | [TK]D-Fender | ageis: that isn't priorityjumping..... |
| 19:25.29 | ageis | ChannelZ: it specifically jumps to priority 6, if you'll notice. I found that putting a Dial() after a Park() without that priority DOES NOT work and results in a busy signal. |
| 19:25.54 | newtonr | feeshon: people will need logs of the call to help https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 19:25.55 | [TK]D-Fender | ageis: ageis What you have shown has nothing to do with that global setting that has been dead for a long time now... |
| 19:26.22 | newtonr | feeshon: this is an Asterisk chat room, they need to see the Asterisk logs, not FOP stuff |
| 19:26.45 | ChannelZ | Yes because Park specifically returns to a different part of the dialplan, it's not a "timeout and continue on" type app. But that's built into the Park app, it's not priority jumping in the old sense of the term (N+101 and such) so what you showed will still work. |
| 19:26.47 | ChannelZ | It's not wrong. |
| 19:27.16 | ageis | cool |
| 19:27.40 | *** join/#asterisk Atreiu (~Atreiu@AToulouse-653-1-417-169.w86-213.abo.wanadoo.fr) |
| 19:27.54 | Atreiu | navaismo !!!!!!! |
| 19:28.08 | Atreiu | I've to kss jour feet |
| 19:28.24 | Atreiu | You solved my problem! |
| 19:28.40 | ageis | ChannelZ, [TK]D-Fender so Park(timeout,return_context,return_exten,return_priority,options,parking_lot_name) this return_priority argument is still a part of the Park() function ? |
| 19:28.50 | Atreiu | navaismo: I'm so happy thank you very very much!! |
| 19:28.51 | ageis | and is not considered jumping |
| 19:29.03 | [TK]D-Fender | ageis: "core show application park" <-- considered looking at the instructions? |
| 19:29.24 | [TK]D-Fender | ageis: Correct it is NOT "priorityjumping" as per that ANCIENT global var |
| 19:29.43 | feeshon | newtonr: I understand and I am collecting that info now |
| 19:29.53 | ageis | excellent, I must have misunderstood. |
| 19:30.00 | *** join/#asterisk youjelly (~youjelly@39.47.224.61) |
| 19:30.38 | youjelly | lol anybody seen this http://www.freeswitch.org/node/437 hilarious webRTC |
| 19:31.39 | ageis | is 1.6 to 1.8 upgrade generally safe in regards to sip.conf and extension.conf syntax or are there any significant changes? |
| 19:32.32 | drmessano | HAHHAHA |
| 19:32.51 | drmessano | JPEG over RTP |
| 19:32.52 | drmessano | lol |
| 19:33.09 | youjelly | xD |
| 19:33.17 | drmessano | GPG! |
| 19:33.37 | youjelly | that guy who made it is a legend, salute |
| 19:33.38 | youjelly | :D |
| 19:37.32 | ageis | s: Silence announcement of the parking space number. -- What's the correct syntax for adding this option? would Park(120000,northampton-open,,6,s) do it? |
| 19:37.34 | igcewieling1 | ageis: read the UPGRADE-*.txt files in the Asterisk source code. This will tell you about such changes. |
| 19:37.40 | ageis | Park([timeout][,return_context[,return_exten[,return_priority[,options]]]]) |
| 19:37.44 | ageis | brackets are confusing me |
| 19:37.48 | ageis | igcewieling1: thanks |
| 19:37.56 | [TK]D-Fender | ageis: Go read the apps instructions. CLI will tell you this... |
| 19:38.08 | ageis | [TK]D-Fender: I just got that from CLI and I'm asking for clarification |
| 19:38.14 | ageis | or interpretation |
| 19:38.21 | igcewieling1 | "core show applications" and "core show functions" to see any new cool stuff |
| 19:38.39 | jmetro | why upgrade from 1.6 to 1.8? go to ast 11 |
| 19:38.52 | ageis | jmetro: will consider it. seems 1.8 is bundled with distro |
| 19:39.02 | navaismo | Atreiu, no problem |
| 19:39.11 | igcewieling1 | jmetro: because Asterisk 11 has not been out long enough |
| 19:39.28 | Atreiu | navaismo: That's cool to learn new things |
| 19:39.37 | navaismo | Atreiu, is recommended to use local channels but your pbx have issues with your hints |
| 19:40.06 | Atreiu | navaismo: So the S before agent means SIP ans what's the A ? |
| 19:40.19 | navaismo | Agent |
| 19:40.53 | Atreiu | What's the purpose ? |
| 19:40.57 | linocisco | [TK]D-Fender, i want to make sure if Cisco RV042 will block Asterisk SIP traffic |
| 19:41.20 | [TK]D-Fender | linocisco: Have you tried? Go something to show us? |
| 19:42.33 | linocisco | [TK]D-Fender, not yet. but found no QoS for udp and port 5060 |
| 19:42.47 | Atreiu | navaismo: What are hints also? |
| 19:42.59 | jmetro | Does asterisk have a build in function to say an extension if you pass it the extension number? |
| 19:43.04 | linocisco | sleepy |
| 19:43.06 | [TK]D-Fender | linoLack of QoS doesn't mean it blocks SIP. |
| 19:43.15 | [TK]D-Fender | linocisco: Lack of QoS doesn't mean it blocks SIP. |
| 19:43.27 | drmessano | I thought the RV042 was the one that had bugs in regard to SIP traffic? |
| 19:43.38 | [TK]D-Fender | linocisco: There is no association between the two. Got something real for us? |
| 19:51.02 | AkkerKid | anyone know of a good reporting interface that'll tie in easily with my asterisk installation? |
| 19:51.26 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 19:51.55 | AkkerKid | i need to see avarage wait times in queues and stuff |
| 19:52.01 | youjelly | cacti |
| 19:52.22 | AkkerKid | dropout rates, avarage hold, calls per extension, |
| 19:52.34 | jmetro | the asterisk cli shows that, doesnt it? with queue show? |
| 19:52.48 | navaismo | AkkerKid, queuemetrics for queues, and maybe you can try the asterisk stats |
| 19:56.07 | AkkerKid | queuemetrics is expensive. |
| 19:58.19 | navaismo | Atreiu, Agents are defined in the agents.conf and are used with queues., hints are used to see the status of device, |
| 19:59.13 | navaismo | Hmm depends on the expensive parameters, but is really useful if you want a good reporting tool for your queues |
| 19:59.31 | AkkerKid | i only have inbound queues... |
| 19:59.34 | Atreiu | navaismo: ans my hints are bugged? |
| 19:59.37 | AkkerKid | does that make a difference? |
| 20:00.18 | navaismo | AkkerKid, not sure if this can give what you are looking http://www.cdr-stats.org/ and this http://www.asternic.net/, but take a look |
| 20:01.04 | navaismo | Atreiu, yes in your pb the hints for your device was idle |
| 20:01.24 | navaismo | instead unavailbale |
| 20:01.56 | acidfoo | AkkerKid, www.xivo.fr |
| 20:02.05 | Atreiu | navaismo: Maybe a freepbx glitch? |
| 20:02.31 | navaismo | its possible |
| 20:15.54 | jmetro | ugh..my mp3 files for voice recording sound so good |
| 20:34.39 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 20:41.28 | *** part/#asterisk feeshon (~gaston@ool-45787011.dyn.optonline.net) |
| 20:43.51 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 20:47.50 | nubbie | any easy way to check a running asterisk process ulimit? |
| 21:15.09 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 21:17.28 | *** join/#asterisk Ice_Strike (Ice_Strike@87.115.83.87) |
| 21:24.11 | Ice_Strike | Hello |
| 21:26.32 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 21:31.31 | apb1963_ | nubbie: ulimit |
| 21:31.41 | apb1963_ | and/or ulimit -a |
| 21:32.02 | nubbie | apb1963_, actually |
| 21:32.20 | nubbie | sudo cat /proc/<asterisk pid>/limits |
| 21:32.48 | nubbie | but, would be nice to see in *CLI> core show settings |
| 21:32.50 | nubbie | or something |
| 21:35.18 | *** join/#asterisk BrokenArrow (~BrokenArr@unaffiliated/brokenarrow) |
| 21:45.01 | leifmadsen | nubbie: just backport it |
| 21:45.09 | leifmadsen | nubbie: exists in asterisk 11 for sure |
| 21:45.16 | leifmadsen | <PROTECTED> |
| 21:45.16 | leifmadsen | <PROTECTED> |
| 21:45.20 | leifmadsen | per 'core show settings'; |
| 21:46.08 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 21:46.23 | nubbie | leifmadsen, Ya, it is there is 1.8 too |
| 21:46.32 | leifmadsen | not sure what else you're looking for then :) |
| 21:46.37 | nubbie | but not the _current_ open file handlers |
| 21:46.39 | leifmadsen | seeing system data is not an asterisk thing |
| 21:46.49 | leifmadsen | ya, that is out of scope afaic |
| 21:46.50 | nubbie | but ya, other tools for that |
| 21:46.56 | nubbie | many setup something in nagios |
| 21:47.03 | leifmadsen | indeed |
| 21:47.04 | nubbie | to monitor |
| 21:47.35 | leifmadsen | nubbie: did you check out that monitoring thing I told you about a couple weeks back? |
| 21:47.43 | leifmadsen | the nagios on steroids? |
| 21:47.57 | nubbie | leifmadsen, Ya, I did. too fancy for me :D |
| 21:48.04 | nubbie | but seemed cool |
| 21:48.27 | leifmadsen | ya very cool |
| 21:49.17 | navaismo | can I ask... what is that tool? |
| 21:51.53 | jmetro | also curious |
| 21:53.01 | *** join/#asterisk teff (~teff@client-80-1-164-21.bsh-bng-011.adsl.virginmedia.net) |
| 21:55.31 | ChannelZ | oh look, another SIP drive-by from OVH. How unexpected. |
| 21:56.12 | navaismo | seems like is top secret and keeps with icinga |
| 21:56.51 | igcewieling1 | OVH? |
| 21:57.25 | leifmadsen | navaismo: Open Monitoring Distribution (OMD) |
| 21:57.34 | navaismo | thanks |
| 21:59.26 | ChannelZ | French internet |
| 21:59.34 | leifmadsen | ah |
| 22:00.11 | ChannelZ | If they're not from China, 9 times out of 10 it's from an IP OVH owns |
| 22:03.20 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-yvzrslgmxakpaowd) |
| 22:03.58 | jmetro | Hm. Can you stick multiple apps in an execif, or should i just use a gotoif. |
| 22:04.15 | jmetro | Im trying to play 3 sound files pretty much |
| 22:05.19 | ChannelZ | yeah, no. |
| 22:06.00 | navaismo | jmetro, not sure if you are asking for that but usually you can playback(file1&file2&file3) or with background |
| 22:06.04 | ChannelZ | You could probably GoSub |
| 22:06.14 | ChannelZ | if you have to do anything wacky |
| 22:06.41 | jmetro | ah i was trying to do sayNumber(1)&Saynumber(2) but i should just do saynumber(1&2&3) you mean |
| 22:07.26 | jmetro | i am making a dynamic DID autoattendant basically, so i have to feed it the extension numbers one by one |
| 22:07.29 | ChannelZ | well no, SayNumber is a different thing |
| 22:07.37 | ChannelZ | but in that case couldn't you use SayDigits? |
| 22:07.51 | jmetro | could i saydigits(127) ? |
| 22:08.14 | jmetro | [i havent used any of the Say applications before so ayDigits sounds perfect if it works like that |
| 22:08.16 | ChannelZ | yeah if you want "one two seven" and not "one hundred twenty seven" |
| 22:08.29 | jmetro | Awesome. Thanks |
| 22:08.52 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-armriwgkptpkpplz) |
| 22:08.52 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:09.28 | jmetro | Oh wait, i forgot I had to do a playback before that. Hm. |
| 22:11.06 | jmetro | http://pastebin.com/Jgjvdw5m |
| 22:11.13 | jmetro | it just seems silly to have to gosub and then return for that |
| 22:13.39 | ChannelZ | well the easy way is just repeat your execif |
| 22:13.48 | jmetro | Oh. |
| 22:13.58 | ChannelZ | do one thing once, something else the next time |
| 22:14.03 | jmetro | And get rid of the Else on the top one. |
| 22:14.12 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 22:14.19 | jmetro | Perfect idea |
| 22:15.26 | ChannelZ | well your syntax on that seems not right but sure |
| 22:16.09 | ChannelZ | you have a [ in there which I think is supposed to be a : |
| 22:16.22 | jmetro | You mean for the Else part? |
| 22:16.26 | jmetro | using a [] rather than :? |
| 22:17.02 | ChrisInSydney | leifmadsen: Got DAHDI to compile and install. Just now trying to work out how to make it compile without having to compile the whole damn kernel :-/ |
| 22:17.13 | leifmadsen | eep :\ |
| 22:17.16 | ChannelZ | well it's not a "rather than" syntax :) it's expression ? doiftrue : doiffalse |
| 22:17.18 | ChrisInSydney | maybe just whip up an RPM |
| 22:17.40 | leifmadsen | ChrisInSydney: ya, hopefully it's not because the kernel is missing something that allows you to load external stuff |
| 22:18.56 | ChannelZ | I think you are misreading the syntax conventions that the help shows. the brackets represent optional arguments |
| 22:18.57 | ChrisInSydney | its the kernel source. Theres some "Stuff in there" that Dahdi wants. I know it doesnt want everything, but without Module.symvers, it wont compile. |
| 22:19.08 | jmetro | Channelz: probably XD |
| 22:19.11 | leifmadsen | :( |
| 22:19.18 | leifmadsen | welp, I think that's it for work today |
| 22:19.27 | leifmadsen | got my RPMs built and such. Time to go play some Forza 4 |
| 22:19.31 | ChannelZ | so it's not literally [:appiffalse(args)], it's :appiffalse(args) 'if you choose to use it' |
| 22:19.41 | leifmadsen | 'bl1tzrage' on Xbox Live for anyone that cares |
| 22:19.49 | jmetro | Right. I understand Channelz. |
| 22:19.52 | ChrisInSydney | leifmadsen: The intergoogletubes recon I should compile the krenel. |
| 22:20.01 | leifmadsen | ChrisInSydney: that is less than ideal |
| 22:20.14 | leifmadsen | runs away |
| 22:20.23 | ChrisInSydney | True, so does the system. i just got a disk usage alert |
| 22:20.27 | jmetro | <PROTECTED> |
| 22:20.27 | jmetro | <PROTECTED> |
| 22:20.35 | jmetro | works |
| 22:20.46 | ChannelZ | yah that looks better |
| 22:22.38 | [TK]D-Fender | <PROTECTED> |
| 22:24.51 | leifmadsen | at that point, why not just use a GoSubIf() ? |
| 22:24.59 | leifmadsen | rather than wrapping all your commands in an ExecIf() ? |
| 22:25.40 | ChannelZ | he didn't want to since it's apparently only 2 things needing doing. |
| 22:25.43 | [TK]D-Fender | 2-3 is worth doing inline.... more does make a subroutine a better idea |
| 22:25.44 | ChannelZ | 5 ways to skin a cat |
| 22:26.13 | leifmadsen | I still find it easier to read and maintain using the subroutines -- you can use priority labels with a GoSubIf() |
| 22:26.30 | leifmadsen | anyways, I'm out |
| 22:26.38 | jzaw | do ppl not use ael2? |
| 22:26.49 | jzaw | i find it way WAY easier to read and write |
| 22:26.49 | ghost75 | no |
| 22:26.56 | leifmadsen | I hate AEL |
| 22:27.01 | jzaw | really? |
| 22:27.07 | leifmadsen | I picked my word carefully too |
| 22:27.08 | jzaw | i never got on with the conf language |
| 22:27.08 | WIMPy | There is an AEL_2_? |
| 22:27.14 | leifmadsen | WIMPy: AEL == AEL2 |
| 22:27.19 | leifmadsen | been v2 forever |
| 22:27.25 | WIMPy | ok |
| 22:27.33 | leifmadsen | since like 1.4 or something |
| 22:27.35 | ghost75 | looks too complicate |
| 22:27.50 | leifmadsen | I just don't like that it simply converts from one parser back into dialplan anyways |
| 22:27.56 | WIMPy | Too restricted. |
| 22:27.58 | leifmadsen | there were too many bugs |
| 22:27.59 | jzaw | interesting you say that ghost75 i found it much simpler to read |
| 22:28.17 | igcewieling1 | UM, you can do anything in AEL which you can do in the regular dialplan |
| 22:28.27 | WIMPy | no |
| 22:28.28 | leifmadsen | I've been looking at dialplan since 0.7 though, so I just look at it and can see missing brackets like woah |
| 22:28.40 | *** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:28.42 | igcewieling1 | WIMPy: what can't you do? |
| 22:28.55 | ChannelZ | skin cats |
| 22:28.59 | ChrisInSydney | leifmadsen. The file was missing from the initial get go, but after a make and clean, its still there. So, tgz it up and thats my source tree :-) Now to test properly |
| 22:29.03 | WIMPy | Combine multiple extensions. |
| 22:29.11 | ChrisInSydney | so you can come back now, its safe ;-) |
| 22:29.18 | WIMPy | And IIRC there's more that doesn't work. |
| 22:29.20 | igcewieling1 | I don't understand. how would you do that in the dialplan? |
| 22:29.21 | jmetro | Yeah i just had one decision to make there |
| 22:29.28 | jmetro | it was either play the voice title or play the extension as a number |
| 22:29.31 | jzaw | pls explain WIMPy |
| 22:29.36 | jmetro | but extension had to be Play extension, play digits. |
| 22:30.16 | jmetro | Full voice prompt will say "To speak to" EXTENSION ONE TWO SEVEN "directly, press 2" |
| 22:30.25 | WIMPy | It's the ugly, but very effective thing of using a pattern, but for certain priorites use more specific lines. |
| 22:30.26 | ChannelZ | you're fine |
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| 22:31.25 | igcewieling1 | WIMPy: ah, I call that "crazy dialplan" |
| 22:31.47 | WIMPy | I like it :-) |
| 22:31.47 | igcewieling1 | I feel it is a great way to write horrible dialplan code. |
| 22:32.15 | WIMPy | Yes. But it's also a great way to write less. |
| 22:33.35 | igcewieling1 | not using spaces, not using newlines and using all one character variables is also a great way to write less, but it is still a bad idea. |
| 22:34.03 | WIMPy | Doesn't work in all languages :-) |
| 22:34.34 | WIMPy | Does anyone remember tokenized code? |
| 22:34.36 | *** join/#asterisk BrokenArrow (~BrokenArr@unaffiliated/brokenarrow) |
| 22:34.45 | ChannelZ | turning on porn is the best way to write less |
| 22:34.49 | jmetro | ^ |
| 22:34.56 | WIMPy | You can write however you want and it will allwys look the same. |
| 22:35.21 | jmetro | I remember using aliases in C++ to write paragraphs or stories that evaluated into code. |
| 22:36.00 | ChrisInSydney | leifmadsen: Perfect :D |
| 22:39.50 | ChrisInSydney | leifmadsen: Thanks for the pointers. Got me looking in the right spot. |
| 22:49.16 | nubbie | +1 to who ever told me to switch bria from UDP to TCP |
| 22:49.22 | nubbie | night and day for my S2 battery |
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| 22:52.01 | *** join/#asterisk billybutts138 (~william@xnat-52.csumb.edu) |
| 22:52.51 | billybutts138 | Can anyone help me with a sip transfer error 403? |
| 22:57.18 | billybutts138 | When a SIP phone gets the transfer section of my dial plan, I get a: |
| 22:57.18 | billybutts138 | [Mar 1 14:34:58] NOTICE[21419]: chan_sip.c:22413 handle_request_notify: Got unknown code '403' in NOTIFY in response to REFER. |
| 22:57.19 | billybutts138 | [Mar 1 14:34:58] NOTICE[21419]: chan_sip.c:22419 handle_request_notify: Transfer failed. Sorry. Nothing further to do with this call |
| 23:00.50 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:03.14 | ChannelZ | forbidden? |
| 23:03.46 | ChannelZ | what are you trying to transfer to? |
| 23:04.08 | WIMPy | ny news on when the Digium phone firmware will be released that was announced yesterday? |
| 23:04.13 | WIMPy | A |
| 23:04.42 | ChannelZ | no, but what do it do? |
| 23:06.13 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:08.15 | WIMPy | Support locales. |
| 23:09.38 | ghost75 | anyone has experience with bluetooth? |
| 23:10.14 | ghost75 | chan_mobile |
| 23:11.17 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:16.31 | [TK]D-Fender | Using Blackberry's causes BlueTooth |
| 23:33.29 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-armriwgkptpkpplz) |
| 23:38.56 | ghost75 | <PROTECTED> |
| 23:38.57 | ghost75 | <PROTECTED> |
| 23:39.36 | ghost75 | i got everything installed but cannot select chan_mobile in menuselect |
| 23:40.07 | WIMPy | Did you run configure again? |
| 23:40.18 | ghost75 | no |
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| 23:44.31 | navaismo | so, i have hundreds of sip channels alive and in some point the server cant handle more calls, complaining about no more rtp ports available, i have increased the ports in rtp but at some point the issue happen again. |
| 23:47.58 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
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