| 00:06.29 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:06.29 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:12.44 | *** part/#asterisk mackhendricks (~mackhendr@99.71.235.30) |
| 00:13.56 | *** join/#asterisk Bullmoose (~Bullmoose@12.50.16.66) |
| 00:14.55 | dijib | suedoh: Background is definitly the way to go. |
| 00:15.07 | dijib | same => n,BackGround(IVR); |
| 00:18.29 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 00:19.53 | suedoh | dijib, Thanks |
| 00:20.14 | suedoh | Is there any way to listen in on a Queued caller without picking up the Queued call? |
| 00:21.08 | dijib | im not that pro here, i just use monitor for everything |
| 00:21.21 | dijib | i'll look into channel commands |
| 00:21.23 | *** join/#asterisk jsjc (~Adium@103.Red-2-136-100.dynamicIP.rima-tde.net) |
| 00:26.10 | *** join/#asterisk retentiveboy (~retentive@72.54.144.26) |
| 00:30.18 | dijib | suedoh: ok i have figured out that you can enable monitor when you recieve the call and then listen in realtime to that sound file. |
| 00:30.41 | suedoh | dijib, Nice, I'm currently experimenting with ChanSpy |
| 00:30.51 | suedoh | Which method have you discovered? |
| 00:30.59 | dijib | ahh that would do it |
| 00:31.08 | dijib | see i dont use that, but same idea i guess |
| 00:31.22 | suedoh | Which idea do you have? |
| 00:31.45 | suedoh | I'm actually not sure if ChanSpy would work now, since I don't have the call fully bridged |
| 00:32.06 | dijib | wait this is while at the background ? |
| 00:32.12 | dijib | that works with monitor |
| 00:32.37 | suedoh | What I'm trying to do is tricky I guess for me at the very least |
| 00:32.43 | dijib | why is the call not connected? |
| 00:32.45 | dijib | bridged? |
| 00:32.58 | suedoh | I have a caller in Queue, I want to have an agent press a number to listen in on the user |
| 00:33.26 | suedoh | While the caller is still in the Queue |
| 00:33.56 | suedoh | With ChanSpy it seems like the call gets picked up, or something bad happens that the MOH stops playing |
| 00:34.09 | dijib | ahh ok. i dont have any queues running to pay with, but when im listening to one of my monitor files becuase its set before the Background() it records the IVR & input. not positive about voices. |
| 00:34.22 | dijib | do they leave queue? |
| 00:34.30 | suedoh | no, they are still sitting in queue |
| 00:35.21 | dijib | but the moh does stop? |
| 00:35.36 | suedoh | yea, it stopped with ChanSpy |
| 00:44.45 | dijib | found this dont know if there is anything helpfull, still reading. http://www.jonathanmanning.com/2009/10/29/monitoring-agents-in-asterisk-with-chanspy/ |
| 00:46.58 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 00:49.19 | *** join/#asterisk netman (netman@83.56.78.188.dynamic.jazztel.es) |
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| 01:26.32 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 01:32.43 | suedoh | If I'd like to include an entire context inside of an extensions.conf file, would that be possible? (Just for tidying up purposes) |
| 01:33.14 | suedoh | I've tried just a standard include => filename, and placing the entire context inside of that file, but that doesn't appear to work. |
| 01:33.31 | suedoh | I've also tried to place the [context] then the include=> which contains the priorities for that context, and that doesn't work either. |
| 01:35.35 | suedoh | So it appears that include => is just to include an entire context into another context, is that correct? |
| 15:09.32 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 15:09.32 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.3.0 (2012/03/29), 1.8.11.0 (2012/03/29), dahdi-linux 2.6.0 (2012/01/04), dahdi-tools 2.6.0 (2012/01/04), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 15:11.11 | *** join/#asterisk TheMan (~theman@66.237.29.132.ptr.us.xo.net) |
| 15:14.01 | TheMan | good morning all |
| 15:26.22 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 15:36.27 | jeffspeff | ok, so after I compile 10.3.0 and go into the cli via asterisk -crv I don't have any sip commands. it's like sip isn't there. i went back and double checked that the channel driver was selected in menuselect (it was) and i continued to recompile, still not getting sip |
| 15:37.04 | Qwell | jeffspeff: pastebin your sip.conf |
| 15:37.29 | jeffspeff | when i rebuilt, i removed all docs and they're back to the defaults created with make samples |
| 15:38.53 | [TK]D-Fender | "docs"? |
| 15:39.58 | beek | waves to Katty |
| 15:40.09 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v010-227.mobile.uci.edu) |
| 15:40.51 | jeffspeff | Qwell, http://pastebin.com/e632xxsu |
| 15:41.34 | Qwell | module reload chan_sip.so |
| 15:41.35 | Qwell | pastebin it |
| 15:44.07 | jeffspeff | Qwell, it wasn't listed under reload, but found it under load. it loaded and sip commands are there now here's the cli output http://pastebin.com/u0ANNuuD |
| 15:46.03 | *** join/#asterisk Johnny- (~John@23-24-48-118-static.hfc.comcastbusiness.net) |
| 15:46.06 | *** join/#asterisk timahvo1 (~rogue@41.81.137.211) |
| 15:46.54 | Johnny- | heya |
| 15:46.59 | neurosys | What part of the SIP shake does asterisk sent the session timers? |
| 15:48.11 | [TK]D-Fender | jeffspeff, "ls -la /etc/asterisk" |
| 15:48.20 | Johnny- | does anyone have the 2.6 firmware for an aastra 6757i |
| 15:48.45 | TheMan | you can download all previous versions from the Aastra website\ |
| 15:49.02 | TheMan | I believe there's an option for previous firmwares |
| 15:49.19 | neurosys | TheMan, yes. hit the little PLUS icon |
| 15:49.33 | jeffspeff | [TK]D-Fender, http://pastebin.com/P2tYgmr9 |
| 15:50.28 | [TK]D-Fender | jeffspeff, Running * as root is not recommended... |
| 15:50.34 | TheMan | neurosys, yup, thats what I was thinking |
| 15:51.05 | jeffspeff | I know, this is just a test system at the moment, just to get the dialplan logic working... not in a production environment. |
| 15:53.34 | [TK]D-Fender | jeffspeff, I'd like to see what a raw load of * not as a daemon looks like.... |
| 15:53.52 | dijib | anybody using googe voice recognition with * here? |
| 15:54.35 | jeffspeff | [TK]D-Fender, just a sec. After I loaded the sip module, i did a 'service asterisk resetart' to see if it'd load sip again automatically, and i haven't been able to get asterisk started back up. |
| 15:56.53 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 15:57.36 | jeffspeff | [TK]D-Fender, every time that i start *, and then check the status of it (service asterisk status) i get a message saying "asterisk deatd but subsys locked" |
| 15:57.56 | [TK]D-Fender | "not as a daemo" <----------------- |
| 15:58.03 | [TK]D-Fender | +n |
| 15:59.41 | jeffspeff | [TK]D-Fender, i've always ran * as a daemon, how do you start it w/o? |
| 15:59.54 | [TK]D-Fender | asterisk -gvvvvvvvvvc |
| 16:01.01 | *** join/#asterisk paolosupino (~paolo-sup@net-2-38-113-188.cust.dsl.vodafone.it) |
| 16:01.05 | paolosupino | hi |
| 16:01.33 | paolosupino | anyone here has experience with Eutelia VOIP (Italian VOIP provider)? |
| 16:02.19 | *** part/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 16:02.30 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 16:02.33 | [TK]D-Fender | hrm |
| 16:02.47 | jeffspeff | [TK]D-Fender, here it is http://pastebin.com/xDpWSRbv the office just surprised me and wants to take me out to lunch for my bday... i'll be back. thanks for your help |
| 16:04.12 | [TK]D-Fender | [Apr 12 04:58:19] WARNING[2948]: chan_sip.c:29589 reload_config: No valid transports available, falling back to 'udp'. |
| 16:04.12 | [TK]D-Fender | [Apr 12 04:58:19] ERROR[2948]: netsock2.c:263 ast_sockaddr_resolve: getaddrinfo("ast.corporate.wellsco.net", "(null)", ...): Name or service not known |
| 16:04.28 | [TK]D-Fender | jeffspeff, Seemed to load chan_sip, but you look to have DNS issues, etc |
| 16:04.39 | [TK]D-Fender | jeffspeff, and some syntax issues with regards to transort, etc |
| 16:09.52 | ectospasm | ~thebook |
| 16:09.52 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:12.08 | *** join/#asterisk TheMan (~theman@66.237.29.132.ptr.us.xo.net) |
| 16:17.01 | *** join/#asterisk vfabi (~fabi@178.76.83.73) |
| 16:18.18 | dijib | [TK]D-Fender: that looks like a dns to me |
| 16:24.46 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 16:33.33 | *** join/#asterisk radic (~radic@dslb-094-216-247-129.pools.arcor-ip.net) |
| 16:42.42 | *** join/#asterisk Johnny- (~John@23-24-48-118-static.hfc.comcastbusiness.net) |
| 16:43.44 | Johnny- | thanks, TheMan. Didn't think to click on that |
| 16:44.15 | *** join/#asterisk timahvo1 (~rogue@41.81.137.211) |
| 16:48.10 | TheMan | np Johnny, It's caught me a few times |
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| 17:24.56 | *** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net) |
| 17:26.09 | aberrios_ | quick question about Yealink and Auto-Answer. In Firmware v61 manual point 25 mentions "Auto answer when receiving the special NOTIFY message from Asterisk server"...but there hardly any documentation on what notify message is needed or where to set this up on the phone. ANyone used this? |
| 17:31.05 | paolosupino | what does "Got SIP response 503 "Service not available - No gateways"" means? |
| 17:31.07 | *** join/#asterisk _Raptor_ (raptorblue@andariel.informatik.uni-erlangen.de) |
| 17:32.02 | pabelanger | how bad is flowroute's support? Waiting for them to reactivate my account at it has already been ~24 hours |
| 17:32.13 | pabelanger | not sure why it was deactivated in the first place |
| 17:32.46 | din3sh | res_rtp_asterisk.c: No remote address on RTP instance '0x7ffd4c0ebed8' so dropping frame |
| 17:33.09 | din3sh | could this cause one way audio on attended xfer? |
| 17:34.05 | *** join/#asterisk thegoat (~thegoat@c-71-224-170-221.hsd1.pa.comcast.net) |
| 17:37.50 | *** join/#asterisk jmpiloq (~jmpiloq@static-87-79-94-28.netcologne.de) |
| 17:38.50 | pabelanger | din3sh: sounds about right |
| 17:43.38 | michael-i | Hi all, I'm trying to track down a jittery moh bug in 10.2.1. When I dial into the MOH application directly it sounds clean. When MOH is piped into confbridge it's garbage. The MOH source file is g722, all phones are g722 only. Timing is timerfd |
| 17:44.30 | michael-i | …and all of this loveliness is on a mips64 platform w/enough horsepower to drive things |
| 17:45.54 | *** join/#asterisk Foxi352 (~Foxi352@v-172-4.access.restena.lu) |
| 17:46.56 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 17:46.56 | coppice | you don't see a mips64 box every day. so exclusive :-) |
| 17:48.32 | michael-i | only exclusively frustrating at times ;) |
| 17:48.39 | din3sh | @pabelanger: calls through a BRI gateway come through to SIP extensions corectly but when get attended-xfer to other sip extensions, caller cannot hear SIP extension |
| 17:49.00 | din3sh | sip extension can hear the BRI caller |
| 17:49.08 | coppice | michael-i: what box are you using? |
| 17:50.08 | michael-i | coppice: it's a development board, seeing what's possible on it |
| 17:50.53 | pabelanger | michael-i: only g722 affected? Could be a bug, the code is pretty new |
| 17:52.11 | michael-i | pabelanger: I need to try other music formats still |
| 17:52.32 | coppice | yeah, the G.722 only went in about 6 years ago |
| 17:52.38 | pabelanger | You could try adding JITTERBUFFER to your dialplan, see if that helps |
| 17:53.08 | pabelanger | coppice: no, asterisk 10 media format was rewritten, same with app_confbridge |
| 17:53.31 | *** join/#asterisk timahvo1 (~rogue@41.90.76.226) |
| 17:54.02 | michael-i | It feels like asterisk is killing itself syncing somewhere… sometimes the music sounds good and cpu sits at <10%, then things fall apart and it jumps to 100%, gets choppy and sometimes crashes. |
| 17:54.07 | coppice | MIPs seemed to be doing well with at higher end of the embedded market until about 5 years ago. now they seem to have been nearly wiped out |
| 17:54.40 | *** join/#asterisk andygraybeal (~andy.gray@obsidian.casanueva.com) |
| 17:58.32 | *** join/#asterisk FinboySlick (~shark@74.117.40.10) |
| 17:59.32 | FinboySlick | After some more testing, it seems my interoperability problem with Metaswitch relates to the fact that it doesn't like asterisk putting "" in the From: sip header. Anyone knows how to work around that? |
| 18:05.59 | *** join/#asterisk eicto (~eicto@144-71.dsl.aichyna.com) |
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| 18:10.41 | jeffspeff | [TK]D-Fender, thanks for the help... i found the root cause of the problems though, i still had SELinux enabled... i disabled that, recompiled, rebooted, now all is well. |
| 18:23.59 | *** join/#asterisk bzort (~bzort@gateway/tor-sasl/bzort) |
| 18:29.32 | *** join/#asterisk timahvo1 (~rogue@197.178.97.88) |
| 18:38.58 | krotos | hi all :) |
| 18:39.34 | krotos | can anyone help me to understand how codec negotiation work? i make a scheme of what i am studing, and where "transcondig happen" |
| 18:40.43 | krotos | i want to understand how SIP/183 negotiation happen trouhght 2 phone and a simple asterisk configuration |
| 18:42.04 | krotos | this is my scheme http://imageshack.us/photo/my-images/26/img20120412203013.jpg/ |
| 18:42.43 | [TK]D-Fender | Transcoding happens once a channel with * with one codec is bridged to one that isn't. Or when a file that you don't have in the current call's format needs to be played back. Same applies to ringing toens, etc, which are all generated |
| 18:44.24 | dijib | how do i debu an agi? |
| 18:45.13 | krotos | [TK]D-Fender: is, but for example if as in example (uploaded on imageshack) i force the codec all on g729 or alaw |
| 18:46.19 | p3nguin | Is there any way to apply an amplitude adjustment to an existing channel? |
| 18:46.30 | p3nguin | The channel is bridged in a call. |
| 18:47.00 | [TK]D-Fender | p3nguin, func_volume |
| 18:47.01 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:47.24 | [TK]D-Fender | krotos, You seem to be allowing multiple codecs. allow only one |
| 18:47.44 | p3nguin | I'm familiar with the function, but I don't see how I can use it to adjust the existing bridged channel. |
| 18:48.20 | [TK]D-Fender | You set it like any other... |
| 18:48.28 | krotos | [TK]D-Fender: ok, but if i allow multiple codecs on asterisk boxes |
| 18:48.44 | [TK]D-Fender | krotos, Multiple != forced. |
| 18:48.49 | krotos | and i act only on phone |
| 18:48.56 | krotos | asterisk does not transcode, right? |
| 18:49.04 | p3nguin | It will. |
| 18:49.26 | *** join/#asterisk donnib (~donnib@0x555281d0.adsl.cybercity.dk) |
| 18:50.03 | krotos | ..why? if i force only on phone 1A (in scheme) and the asterisk boxes allow g729 and alaw, and also the phone b accept g729 and alaw |
| 18:50.39 | jeffspeff | krotos, do you have licenses for g729? |
| 18:50.44 | krotos | yes |
| 18:51.00 | jeffspeff | so why not just use it instead of alaw? |
| 18:51.07 | krotos | i've got it |
| 18:51.58 | krotos | because some customers say that the audio on g729 is not good..(fine ear) |
| 18:52.19 | krotos | and i change only the codec and the configuration on asterisk for these users |
| 18:52.22 | *** join/#asterisk wonderworld (~ww@dsdf-4db5f99a.pool.mediaWays.net) |
| 18:52.49 | krotos | but when i watch with core show channel <namechannel> i see that on one side transcode happens |
| 18:53.18 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-prdsupivlfwiigrf) |
| 18:53.26 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ovfqspkzvnjkfwzs) |
| 18:53.26 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 18:54.27 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 18:55.33 | p3nguin | So there's no way to change the amplitude of an existing bridged channel, I guess. |
| 19:00.14 | [TK]D-Fender | p3nguin, I don't see why you can't use the function like normal.... |
| 19:02.49 | p3nguin | Okay... tell me how I'm going to apply a function to an EXISTING BRIDGED CHANNEL. |
| 19:04.13 | [TK]D-Fender | You know AMI has a function to set a variable..... |
| 19:04.21 | [TK]D-Fender | that can be used with FUNCTIONS. |
| 19:04.54 | p3nguin | No, I don't know, because I know nada about AMI. |
| 19:05.09 | [TK]D-Fender | You should do something about that. |
| 19:05.24 | [TK]D-Fender | it is a treasure-trove of "how can I?" |
| 19:05.32 | p3nguin | Setting it "like normal" would involve dialplan and Set(), so I couldn't see any way to apply it. |
| 19:05.57 | [TK]D-Fender | normal = 0. you set it by db |
| 19:06.09 | p3nguin | I also don't know the valid range of values for VOLUME(). |
| 19:06.15 | [TK]D-Fender | db <- |
| 19:06.19 | p3nguin | -12 doesn't seem to make it any quieter. |
| 19:06.30 | [TK]D-Fender | should. show me the attempt |
| 19:06.37 | p3nguin | I've applied both tx and rx. |
| 19:07.02 | [TK]D-Fender | p3nguin, Also depends if you're setting TX vs RX, if the audio is in fact passing through * (kinda critical) |
| 19:07.09 | [TK]D-Fender | and you've got the right channel, etc. |
| 19:07.12 | [TK]D-Fender | so PB it up |
| 19:08.34 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 19:08.49 | p3nguin | -12 should have made both legs very quiet, I would have thought. |
| 19:10.15 | dijib | debugging of agi, how do i do it? |
| 19:10.33 | [TK]D-Fender | dijib, "help agi" <- |
| 19:13.01 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 19:13.17 | dijib | http://pastebin.com/yEciFNdz |
| 19:13.36 | dijib | i have no clue... just dies after agi_arg_1 |
| 19:13.43 | dijib | says it completes |
| 19:14.19 | [TK]D-Fender | So it's doing nothing |
| 19:16.42 | *** join/#asterisk SaRSAeOL (~sarsaeol@66-113-78-49.rev.ibsinc.com) |
| 19:18.21 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 19:19.23 | p3nguin | I figured out the problem. |
| 19:19.44 | *** join/#asterisk jsjc (~Adium@103.Red-2-136-100.dynamicIP.rima-tde.net) |
| 19:20.04 | p3nguin | I was spying on the phone's channel rather than the channel that the call was bridged with, so I was hearing full volume rather than the adjusted volume level. :/ |
| 19:20.39 | [TK]D-Fender | p3nguin, \o/ |
| 19:23.18 | p3nguin | ChanSpy outputs three lines and it is very confusing as to what I was actually attached to. |
| 19:25.48 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 19:38.21 | dijib | oh right im missing format_sln format |
| 19:44.48 | p3nguin | As advised, I am learning about basic management via AMI. |
| 19:48.05 | *** join/#asterisk timahvo1 (~rogue@197.177.235.20) |
| 19:48.20 | *** part/#asterisk thafreak (~thafreak@unaffiliated/thafreak) |
| 19:52.45 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 19:59.15 | p3nguin | [tk]d-fender: That. Is. Awesome. |
| 19:59.36 | p3nguin | [tk]d-fender: It does permit me to set variables and make changes on the fly using functions. |
| 19:59.37 | [TK]D-Fender | p3nguin, Found a few tricks you can get away with in there? |
| 19:59.47 | p3nguin | Action: Setvar |
| 19:59.52 | p3nguin | Channel: mychannel |
| 19:59.58 | p3nguin | Variable: VOLUME(rx) |
| 20:00.06 | p3nguin | Value: -3 |
| 20:00.13 | [TK]D-Fender | p3nguin, AMI & Local channels (often originated) are great ways to mangle the shit out of your calls) |
| 20:00.18 | p3nguin | Worked perfectly. |
| 20:00.47 | p3nguin | Why didn't you convince me to look into this sooner? |
| 20:01.21 | p3nguin | It isn't convenient to manage it this way (using nc on a terminal), but it sure is effective. |
| 20:02.25 | [TK]D-Fender | p3nguin, As your need grows so will your desire to make the task more convenient. You'll get to making smarter scripts when it becomes worthwhile to do so.... |
| 20:02.56 | p3nguin | I'll be working on shell scripts to control this very soon. This is bothersome to do a lot of stuff via nc. |
| 20:03.14 | *** join/#asterisk wooster (~drewp@cusse.org) |
| 20:03.22 | wooster | hello |
| 20:03.41 | wooster | if i have multiple SIP devices beind NAT with IPtables, how do I make IPtables do symmetric NAT? |
| 20:04.07 | wooster | i just want stuff to work, but i can see devices sending packets to the server on udp 5060, and the response coming back to the router, but not making it back from the router to the device |
| 20:04.54 | p3nguin | What I will probably end up doing now that I see how it operates is write my commands into a temporary text doc and them just paste into AMI. |
| 20:05.22 | p3nguin | I still have to figure out how to make it less noisy. |
| 20:05.28 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 20:06.19 | michael-i | This might be a lightweight wrapper which could save you some time: http://marcelog.github.com/articles/bash_asterisk_manager_interface_client_shell_script.html |
| 20:06.55 | [TK]D-Fender | p3nguin, PHP-AGI also has an AMI class. Probably worth you're using... |
| 20:07.00 | [TK]D-Fender | your* |
| 20:07.29 | p3nguin | Now you make it sound like it involves programming again. |
| 20:09.14 | [TK]D-Fender | It involves programming. |
| 20:09.35 | [TK]D-Fender | There, now it's an overt statement instead of "alluding" or "implying" :) |
| 20:10.38 | p3nguin | I put off looking at AMI all this time because I thought it involved programming, which I don't do. |
| 20:13.48 | [TK]D-Fender | First rule of procrastination : Never put off till tomorrow what you can put off to the day after that just as easily... |
| 20:14.06 | p3nguin | I will have to see if this bami thing does any good for me. |
| 20:14.20 | p3nguin | It could save me some time writing a bunch of shell scripts. |
| 20:15.11 | p3nguin | For basic operations, I'll probably suck it up and use nc in raw form. |
| 20:15.38 | p3nguin | If I can make the AMI less noisy, that will help a lot. |
| 20:18.27 | *** join/#asterisk wonderworld (~ww@dsdf-4db5f99a.pool.mediaWays.net) |
| 20:19.35 | p3nguin | In your opinion, would it be best to start out with read perms of nothing and add them as I need to see things? |
| 20:19.40 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 20:19.46 | p3nguin | That would surely make it quiet. |
| 20:24.07 | [TK]D-Fender | not sure what you are calling "noise". |
| 20:24.15 | [TK]D-Fender | But.. it's checkout time... perhaps later |
| 20:24.16 | [TK]D-Fender | BBIAB |
| 20:25.56 | citywok | gah i'm using func_odbc and trying to do a write, and i can't figure out why it isn't working. debug doesn't show any read/write queries when you use func_odbc, even if the query is succesful (i can do writes) |
| 20:26.39 | *** join/#asterisk oej_ (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 20:26.44 | citywok | is there any way to get debugging output from func_odbc? |
| 20:28.43 | Kobaz | core set debug x maybe |
| 20:28.47 | Kobaz | try like maybe 3 |
| 20:28.54 | Kobaz | if that's not enough info maybe try 5 |
| 20:29.00 | Kobaz | if not, then it's time to hack some source |
| 20:29.24 | citywok | i set it to 10 lol |
| 20:29.41 | Kobaz | past a certain number (offhand I don't know the number), there's no more debug levels |
| 20:30.23 | p3nguin | Just set it to 2147483647 to ensure you cover them all. |
| 20:30.40 | Kobaz | is debug level signed or unsigned? |
| 20:31.03 | Kobaz | if it's unsigned you better use 4294967296 |
| 20:31.20 | pabelanger | no reason to go above 15 |
| 20:31.47 | Kobaz | pabelanger: i very badly want to convert everything to be module-level debugging |
| 20:31.58 | Kobaz | moh set debug 5 |
| 20:31.58 | Kobaz | etc |
| 20:32.41 | Kobaz | channel debug would pass debug levels through everything per-channel |
| 20:32.48 | p3nguin | It should already be that way, and I don't know why someone didn't speak up when it was being made the way it currently is. |
| 20:32.51 | Kobaz | you could set it globally if you like |
| 20:33.07 | Kobaz | p3nguin: because code gets written piecewise, and a global debug level was the easiest thing to build |
| 20:33.13 | *** join/#asterisk zaf (~zaf@76.72.88.14) |
| 20:34.16 | Kobaz | and when the developer needs outgrow what the framework can provide, then the framework gets updated |
| 20:34.16 | citywok | does anybody see anything wrong with how i'm doing it maybe? http://pastebin.com/Wxvgk08J |
| 20:35.14 | pabelanger | Kobaz: well, setting debug level per console connection was recently added |
| 20:35.20 | Kobaz | yeah |
| 20:35.25 | Kobaz | the next thing is per-channel |
| 20:35.36 | Kobaz | and then per-modulke |
| 20:36.06 | p3nguin | There's some goofy module debug thing in there now that doesn't work. |
| 20:37.12 | p3nguin | You can enable debug (which does not work correctly) on a module, but you can never disable it again. |
| 20:38.23 | pabelanger | Kobaz: talk with jrose_atDigium. He is adding per channel logging identifiers right now. You should see what would be needed to add it to his stuff |
| 20:38.33 | pabelanger | https://reviewboard.asterisk.org/r/1823/ |
| 20:38.33 | Kobaz | ah nice |
| 20:38.58 | pabelanger | added actually |
| 20:39.14 | Kobaz | i have a number of logging improvements that would go well with this |
| 20:43.05 | dijib | how do i get format_sln working? |
| 20:43.22 | dijib | its in /usr/lib/asterisk/modules/format_sln.so |
| 20:43.33 | mjordan | Kobaz: https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging |
| 20:43.44 | dijib | but it wont run. and i have in modules.conf load=> format_sln.so |
| 20:43.47 | pabelanger | dijib: *CLI> module load format_sln |
| 20:43.59 | dijib | pabelanger: didnt work. |
| 20:44.10 | pabelanger | show us the problem |
| 20:44.11 | dijib | i have also chown chmod to no avail. |
| 20:44.12 | pabelanger | ~debuglog |
| 20:44.20 | Kobaz | nifty |
| 20:44.20 | dijib | swissarms*CLI> module load format_sln |
| 20:44.21 | dijib | Unable to load module format_sln |
| 20:44.22 | Kobaz | speaking of coding |
| 20:44.31 | pabelanger | ~debug |
| 20:44.31 | infobot | ACTION DeBuggers $1 |
| 20:44.31 | Kobaz | anyone want to tackle reviewing predial and hangup handlers |
| 20:44.50 | dijib | 160 -rwxr-xr-x. 1 root root 162421 Mar 21 15:39 /usr/lib/asterisk/modules/format_sln.so |
| 20:44.54 | Kobaz | maybe i should offer cookies and beer to the first person who gives a ship it |
| 20:44.59 | dijib | oh theres my problem.. root:root |
| 20:45.02 | pabelanger | dijib: show us a debug log |
| 20:45.07 | pabelanger | ~collectdebug |
| 20:45.08 | infobot | collectdebug is, like, a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 20:45.15 | pabelanger | dijib: ^ |
| 20:45.21 | mjordan | Kobaz: I know there's some feedback waiting for you on pre-dials |
| 20:45.27 | Kobaz | hmm |
| 20:45.28 | Kobaz | lemme check |
| 20:45.35 | Kobaz | i think i addressed at least one review |
| 20:46.05 | Kobaz | yeap |
| 20:46.09 | Kobaz | already updated, waiting for you guys |
| 20:46.13 | mjordan | ah |
| 20:46.29 | mjordan | I think we missed it due to no comments on the findings |
| 20:46.33 | mjordan | I'll bug Richard about it |
| 20:46.40 | Kobaz | oh wait |
| 20:46.44 | Kobaz | the review is loading more pages |
| 20:47.17 | Kobaz | oh okay |
| 20:47.20 | Kobaz | i have some stuff to do |
| 20:47.23 | mjordan | :-D |
| 20:47.59 | dijib | ok nevermind format_sln is working now... just not googletts.agi or speechrecg.agi |
| 20:50.52 | krotos | guy, i 've got a problem.. |
| 20:51.00 | krotos | it's possible |
| 20:51.13 | krotos | to specify what interface use to exit with asterisk? i forced with iptables |
| 20:51.22 | krotos | but i've got a mono-directional audio |
| 20:51.28 | Qwell | krotos: nope |
| 20:51.37 | Qwell | it's up to the kernel to look at your routing table to handle that |
| 20:52.44 | krotos | i've got a iptables route (static route) for forcing outgoing traf. on specific ip |
| 20:52.45 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:53.07 | krotos | but on the SDP trace, there are the other ip |
| 20:53.07 | *** part/#asterisk paolosupino (~paolo-sup@net-2-38-113-188.cust.dsl.vodafone.it) |
| 20:53.14 | Qwell | iptables is not the correct way to do that |
| 20:53.19 | Qwell | and you need to specify an externip |
| 20:54.12 | krotos | my situation is: NO NAT, direct ip on machine. First ip config on eth0, and the second ip in alias mode |
| 20:54.15 | krotos | on the same interface |
| 20:55.14 | krotos | and what is the right way to do this? |
| 20:55.19 | Qwell | <Qwell> it's up to the kernel to look at your routing table to handle that |
| 20:57.31 | dijib | i have enabled debug on the script yet its still failing as complete with no errors. http://pastebin.com/g315GyuP |
| 20:57.39 | dijib | googletts.agi is the script |
| 20:58.46 | p3nguin | (1544.59) <dijib> oh theres my problem.. root:root <---- no |
| 20:59.19 | dijib | that was my problem with format_sln.so but ive resolved that and moved on to the next |
| 20:59.26 | p3nguin | No, that was not your problem. |
| 20:59.45 | dijib | how wasnt it? |
| 20:59.49 | p3nguin | Every asterisk module I have is owned by root, like they should be. |
| 20:59.56 | dijib | oh |
| 21:00.03 | dijib | well mine are all asterisk:asterisk now |
| 21:00.09 | p3nguin | Now they're all wrong. |
| 21:00.35 | dijib | yeh but everythings working so... im ok with it |
| 21:00.47 | p3nguin | Of course you are. |
| 21:01.10 | krotos | Qwell: i don't understand too much. Can i set externip for a single peer |
| 21:01.12 | krotos | ? |
| 21:01.14 | dijib | well dude im trying here. format_sln was not working prior to the asterisk account owning it |
| 21:01.24 | *** join/#asterisk Skorski (~Skorski@72.12.218.163) |
| 21:02.17 | p3nguin | [tk]d-fender: By noise, I was talking about all the chatter given by all the events. I really don't need to see all the events all the time I am connected to the AMI. |
| 21:02.53 | [TK]D-Fender | p3nguin: Events: Off |
| 21:03.08 | p3nguin | Is it permanent, or do I have to do it every time? |
| 21:03.13 | [TK]D-Fender | Per connection |
| 21:03.25 | p3nguin | Is there a setting to make it always off? |
| 21:04.35 | p3nguin | I see there's an eventfilter setting. |
| 21:04.35 | [TK]D-Fender | nope |
| 21:05.27 | generalhan | i am trying to plan my new setup ... currently i have 2 sites each with their own PRI and stupid Avaya system... when i give that thing the boot and move everything over to Asterisk, i am trying to decide if i should keep a PRI at each site, or move them both to one site, and have all the SIP phones at the remote site go over the point-to-point pipe (5M)... or if i should have 2 different |
| 21:05.27 | generalhan | hardware setups and connect the systems via IAX, or similar. |
| 21:05.34 | p3nguin | I guess I could use that to tune out all the noise. |
| 21:05.48 | generalhan | i know YMMV but i figured it would still be a good idea to get the experts' opinions |
| 21:05.51 | Kobaz | i wrote FilterAdd which is in 1.10, but there's no FilterRemove since I haven't figured out the security issues |
| 21:08.01 | Kobaz | so you can connect to ami and do a bunch of filteradds for what you dont want to see |
| 21:08.29 | p3nguin | Since this is day 0 for me on AMI, I'll get into that pretty soon. |
| 21:10.40 | *** join/#asterisk X-Rob (~Rob@thomas232.lnk.telstra.net) |
| 21:11.17 | Kobaz | dijib: in general, you do not want resource files that do not need to be modified to be owned by the user running them |
| 21:11.37 | p3nguin | Good luck getting anything like that through to him. |
| 21:12.00 | p3nguin | I told him, and I guess you saw the response. |
| 21:12.04 | Kobaz | dijib: if someone gets access to your asterisk, and asterisk has write access to your .so files (and other things it shouldn't) then an attacker can further compromise your system in very hard to detect ways |
| 21:12.29 | p3nguin | He doesn't care about things like that... he IRCs as root. |
| 21:12.33 | Kobaz | haha |
| 21:12.47 | p3nguin | Well, did, until he was forced to change it if he wanted to remain in the ##linux channel. |
| 21:13.23 | p3nguin | He doesn't care about anything other than "make it go." Doesn't matter how much he mangles it, as long as the end result is what he wants. |
| 21:13.31 | Kobaz | sounds fun |
| 21:13.40 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 21:14.02 | p3nguin | (1600.09) <p3nguin> Now they're all wrong. |
| 21:14.03 | p3nguin | (1600.35) <dijib> yeh but everythings working so... im ok with it |
| 21:14.25 | Kobaz | ftw |
| 21:14.41 | p3nguin | That is the primary reason I quit helping him with most things. |
| 21:17.46 | p3nguin | So... |
| 21:17.54 | *** join/#asterisk CyBeRxIxO (~efernande@190.41.182.228) |
| 21:18.06 | p3nguin | AMI seems very effective, and I wish someone would have forced me to look at it a long time ago. |
| 21:18.14 | Kobaz | hah |
| 21:18.14 | Kobaz | yeah |
| 21:18.20 | Kobaz | that's the same thing i thought |
| 21:18.44 | CyBeRxIxO | wats sup ppl, im having noise between sip stations on the LAN, any ideas for solution? |
| 21:18.46 | p3nguin | I always put it off because I got the impression that I needed to code to use it. |
| 21:19.00 | Kobaz | well you still need to code a bit |
| 21:19.04 | Kobaz | but it's not hard |
| 21:19.08 | p3nguin | cyberxixo: What kind of noise? |
| 21:19.23 | p3nguin | I didn't have to code anything to use it earlier. |
| 21:19.27 | Kobaz | oh |
| 21:19.29 | Kobaz | well i mean |
| 21:19.35 | Kobaz | you can telnet to it and log in and look at events |
| 21:19.39 | p3nguin | Basic TCP connection with nc, run some commands, get results. |
| 21:19.43 | Kobaz | if you want to do something with those events, you should write some code |
| 21:19.44 | Kobaz | yeah |
| 21:19.51 | CyBeRxIxO | hum |
| 21:20.14 | Kobaz | like count how many channels got made per day |
| 21:20.16 | Kobaz | or whatever |
| 21:20.20 | Kobaz | you can write a little script |
| 21:20.20 | p3nguin | I thought I needed to know C to be able to use it for anything useful. |
| 21:20.26 | CyBeRxIxO | when press speaker it becomes normal, and happen again after seconds |
| 21:20.30 | Kobaz | you don't need c, you can use any languag |
| 21:20.40 | p3nguin | That was the impression I got from people who talk about AMI here. |
| 21:20.46 | Kobaz | any language that can connect to a tcp socket |
| 21:21.04 | Kobaz | well |
| 21:21.09 | _Corey_ | p3nguin: I do my AMI stuff in PHP or Perl |
| 21:21.14 | p3nguin | Rarely did anyone say, "Just type some stuff in a netcat session." |
| 21:21.18 | Kobaz | when we talk about ami, we're generally talking about writing ami commands, or fixing them |
| 21:21.32 | p3nguin | I don't program. At all. |
| 21:21.48 | p3nguin | Therefore, I thought I could not use AMI. |
| 21:21.49 | Kobaz | well, if you have a certain level of sysadmin background it's obvious you can netcat to it |
| 21:22.00 | p3nguin | I didn't know that. |
| 21:22.03 | Kobaz | yeah |
| 21:22.06 | Kobaz | now you know :) |
| 21:22.07 | p3nguin | (until today) |
| 21:22.09 | _Corey_ | AMI is just plain text in/out of TCP |
| 21:22.10 | p3nguin | Now I know! |
| 21:22.23 | p3nguin | I've been missing out for ages. |
| 21:22.30 | Kobaz | ami is YALBD |
| 21:22.38 | Kobaz | (Yet Another Line Based Daemon) |
| 21:22.58 | Kobaz | like smtp, pop3, imap, etc, etc etc |
| 21:23.06 | p3nguin | No one told me. |
| 21:23.11 | Kobaz | with any of those you can just telnet/nc to the port and send it stuff |
| 21:23.14 | Kobaz | and get stuff back |
| 21:23.47 | p3nguin | And since I started using Asterisk prior to having The Book, I never thought to look it up to see what was said about using AMI. |
| 21:23.57 | Kobaz | hell you can telnet to port 80 and type GET / |
| 21:24.21 | p3nguin | Had I read the book prior to using asterisk, it may not have been a mystery. |
| 21:24.27 | Kobaz | hehe |
| 21:24.34 | citywok | has anybody ever tried executing a stored procedure from func_odbc? |
| 21:24.35 | _Corey_ | yeah, I wish there was a book when I started using Asterisk... lol |
| 21:24.51 | Kobaz | citywok: should work just the same as any other sql query |
| 21:25.02 | Kobaz | select * from func() |
| 21:25.33 | Qwell | _Corey_: How do I get a cool title like "Managing Partner"? |
| 21:25.37 | citywok | Kobaz: execute storedProcedureName 'var1','var2' |
| 21:25.43 | CyBeRxIxO | i got buzzing noise between sip stations on LAN, to become call normal user have to press the speaker button on the phone and then press it again to use the hardset, and call is normal again for a while |
| 21:25.45 | citywok | is what i would run in the console, not a select statement |
| 21:26.02 | Kobaz | citywok: you need to get the select statement |
| 21:26.08 | _Corey_ | Qwell: Titles are pretty arbitrary... ;) I actually haven't decided what I'm going to call myself at the new company |
| 21:26.18 | Kobaz | CyBeRxIxO: sounds like a buggy phone |
| 21:26.51 | p3nguin | echo -e "GET / HTTP/1.0\n"|nc www.digium.com 80 |
| 21:27.14 | CyBeRxIxO | it only happens on some phones lets say 5 of 15 phone, them are yealink t9 |
| 21:27.19 | p3nguin | The server doesn't appear to like this. |
| 21:27.20 | X-Rob | _Corey_, I've always liked 'Chief Button Pusher' |
| 21:27.23 | Kobaz | _Corey_: quite arbitrary... some days i'm CTO, some days I'm Managing Partner, and some days I'm Slacker Extraordinare |
| 21:27.41 | krotos | Qwell: is not a right choiche to use media_address |
| 21:27.42 | krotos | in sip.conf? |
| 21:27.46 | krotos | or does not work? |
| 21:28.01 | p3nguin | echo -e "GET /en/ HTTP/1.0\n"|nc www.digium.com 80 |
| 21:28.09 | Kobaz | GET / |
| 21:28.09 | Kobaz | Connection closed by foreign host. |
| 21:28.10 | Kobaz | does not like |
| 21:28.22 | p3nguin | ^^ that works fine. |
| 21:28.41 | Kobaz | i see p3nguin found a new toy |
| 21:28.43 | X-Rob | you need the protocol name. |
| 21:28.45 | citywok | Kobaz: ah hah: select * from openquery(loopback, 'exec getCallbackTimes') |
| 21:28.52 | p3nguin | only AMI |
| 21:29.03 | p3nguin | That's the only new thing I've discovered today. |
| 21:29.31 | Kobaz | okay |
| 21:29.32 | Kobaz | gym tiome |
| 21:29.35 | Kobaz | gym time |
| 21:29.43 | Kobaz | this body doesn't get fit from programming 16 hours a day |
| 21:31.34 | Kobaz | time for some yoga and indoor rock climbing |
| 21:33.00 | Kobaz | real rocks are much better, but alas |
| 21:33.06 | Kobaz | anyway... /me is gone |
| 21:34.15 | jrose_atDigium | peeks |
| 21:36.15 | jrose_atDigium | I don't believe Kobaz's stuff is too likely to conflict with mine in any way. Call identifiers are pretty isolated from most things as they are in trunk right now. |
| 21:39.16 | CyBeRxIxO | yealink sip phone have any special configuration for best working, default conf is enough? |
| 21:54.32 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:54.59 | p3nguin | cyberxixo: Your noise issue doesn't sound like a configuration problem by any means. It sounds to me like a hardware problem. |
| 21:56.41 | WIMPy | Kobaz: What's that prdial thing? Do you have a link for me? |
| 22:00.35 | *** join/#asterisk hfb (~hfb@pool-98-112-209-179.lsanca.dsl-w.verizon.net) |
| 22:01.08 | *** join/#asterisk hb_sip (hb@smooth.operator.tc) |
| 22:09.46 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ovfqspkzvnjkfwzs) |
| 22:14.39 | CyBeRxIxO | p3ng, but then why it happens on many phones? |
| 22:15.44 | CyBeRxIxO | anything i could do to make it to do best qualify sound? remember its a LAN |
| 22:22.20 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-ddsitoclfeykvtqz) |
| 22:27.41 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:29.11 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 22:31.57 | fprior | Hi all, here again with the case of the spa400 and * 1.8 . http://pastebin.com/fVC9JtV2 |
| 22:36.12 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 23:03.31 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 23:05.36 | citywok | Kobaz: i was unable to make it work. i ended up using a system call to isql to do it :( System(echo "EXECUTE master.dbo.getCallbackTimes '${CDR(UNIQUEID)}','100468','${TZ}'" | /usr/bin/isql ARSystem username password -w) |
| 23:11.13 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 23:31.23 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
