| 00:08.48 | *** join/#asterisk Galaxor (~Galaxor@208.67.250.157) |
| 00:09.46 | Galaxor | Hi. I am trying to use the built-in android sip client. I need that client to send the number that should appear on caller-id. Does anybody know how to set that up? |
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| 00:56.31 | Lefty | waves |
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| 01:14.23 | ChannelZ | Galaxor: go under |
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| 01:38.34 | Lefty | any of you fine folks ever hooked up an Avaya 96xx model phone to Asterisk via SIP? |
| 01:39.05 | Lefty | I've got mine connected in such a way that it'll make outgoing calls, but gives me a "Not accepting call completion offers from call-forward recipient" errors on inbound |
| 01:39.16 | rfreire | Hi, #asterisk! I'm having a hard time with logging. Can't get any logging, while I do have a very basic logger.conf, I can't get anything to write the log. It's a raspbian-based asterisk. logger show channels is just empty ;-/ Pastebin: http://pastebin.com/jRd4hY25 |
| 01:39.17 | Lefty | but a softphone with the same settings works both ways |
| 01:40.03 | Lefty | it tries to forward to Local@<ip of my asterisk server> |
| 01:41.23 | *** join/#asterisk andrewyager (~andrewyag@114.141.105.250) |
| 01:43.49 | phyrexianslug | @Lefty: came across this: http://blog.michaelfmcnamara.com/2012/01/asterisk-now-with-avaya-ip-phones/ |
| 01:44.04 | phyrexianslug | you got users.dat setup and it's parsed by the phone? |
| 01:46.32 | Lefty | I'm not using users.dat yet, is that something that's specific to the Avayas? |
| 01:46.46 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 01:47.25 | Lefty | I've got a 46xxsettings.txt and a 96xxupgrade.txt set up on my webserver that the phone appears to be using |
| 01:48.12 | phyrexianslug | sorry, I am rather rusty with the Avaya gear (I've been using Cisco stuff for too long!) |
| 01:48.13 | Lefty | it does successfully register with asterisk, as I can make outbound calls on that phone to local extensions and via my sip trunk |
| 01:48.34 | Lefty | I even factory wiped the phone to nuke any saved call forwarding settings |
| 01:48.42 | phyrexianslug | Oh, well if outbound works you should be good with auth |
| 01:49.37 | phyrexianslug | if you send a call to the phone from Asterisk, does the phone report busy when asterisk tries it? (ie at the console?) |
| 01:49.57 | Lefty | the phone's console, or asterisk's? |
| 01:50.02 | phyrexianslug | Asterisk |
| 01:51.09 | Lefty | zoiper gives me "service or option unavailable" when I try to dial the avaya, and asterisk says this: |
| 01:51.19 | Lefty | NOTICE[22256] Ext. 10: Incoming call from "Colin (Zoiper)" <zoiper> to 10 |
| 01:51.19 | Lefty | NOTICE[22256] app_dial.c: Not accepting call completion offers from call-forward recipient Local/10.196.186.70@mycontext-37db;1 |
| 01:51.19 | Lefty | NOTICE[22256] chan_local.c: No such extension/context 10.196.186.70@mycontext while calling Local channel |
| 01:51.19 | Lefty | NOTICE[22256] app_dial.c: Forwarding failed to dial 'Local/10.196.186.70@mycontext' |
| 01:51.48 | Lefty | even though the dialplan says to dial SIP/avaya (which is what I called that phone in sip.conf) |
| 01:52.18 | [TK]D-Fender | PASTEBIN the actual complete call debug |
| 01:52.24 | [TK]D-Fender | Do not flood it in here |
| 01:52.25 | [TK]D-Fender | ~pb |
| 01:52.26 | 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. |
| 01:52.27 | [TK]D-Fender | ^^^ |
| 01:54.30 | phyrexianslug | yeah, something looks wrong there, a copy of what you've setup for that extension in a pastebin would be more helpful. :P |
| 01:55.14 | Lefty | sure, my dialplan and sip.conf are pretty small still, I'll throw them on pastebin |
| 01:55.26 | [TK]D-Fender | Lefty: Show the call..... |
| 01:55.38 | [TK]D-Fender | Lefty: configs are only one part of the story. |
| 01:55.49 | Lefty | the actual sip calls? |
| 01:55.51 | [TK]D-Fender | Lefty: what the other side is sending and matching against is another |
| 01:56.00 | [TK]D-Fender | Lefty: yes, the actual COMPLETE call debug from * CLI |
| 01:56.05 | Lefty | that's all I see in the asterisk log |
| 01:56.07 | Lefty | ok, standby |
| 01:56.08 | [TK]D-Fender | no log |
| 01:56.11 | [TK]D-Fender | * CLI |
| 01:56.14 | [TK]D-Fender | "sip set debug on" |
| 01:56.18 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 01:56.22 | [TK]D-Fender | "core set verbose 10" |
| 01:56.29 | [TK]D-Fender | Start with those |
| 01:56.39 | [TK]D-Fender | then we'll amp up CORE DEBUG after if needed |
| 01:56.47 | Lefty | roger that |
| 01:57.01 | Lefty | I've got sip debug on already, but I'll crank up verbose as well |
| 02:00.37 | *** join/#asterisk dgeary2 (~debian@pa49-187-94-0.pa.nsw.optusnet.com.au) |
| 02:00.52 | Lefty | https://gist.github.com/LeftyBC/6589203 <- that's my relevant configs |
| 02:00.57 | Lefty | I'll grab a debug dump now |
| 02:03.15 | j4jackj | Has anyone here heard of companding? |
| 02:04.28 | *** join/#asterisk iq (~iq@66.182.203.0) |
| 02:05.53 | [TK]D-Fender | j4jackj: Basic audio DAC concept |
| 02:06.08 | [TK]D-Fender | j4jackj: G.711 is "compaded", not "compressed" |
| 02:06.39 | j4jackj | [TK]D-Fender: Heehee |
| 02:07.11 | j4jackj | [TK]D-Fender: Well I use companding to increase SNR for music in IMA ADPCM files. Without it, both in the form of alaw and in the form of sox' own compand, the music would sound horrendous. |
| 02:07.59 | [TK]D-Fender | <PROTECTED> |
| 02:09.31 | j4jackj | Actually, I undo the compression. |
| 02:09.34 | j4jackj | It sounds terrific. |
| 02:10.48 | Lefty | [TK]D-Fender: still working on getting that dump, sorry |
| 02:11.25 | phyrexianslug | No worries, We're here all night. :P |
| 02:15.02 | Lefty | ok, there we go |
| 02:15.06 | Lefty | https://gist.github.com/anonymous/6589288 |
| 02:15.26 | Lefty | that's a complete log (I hope!) of all the sip traffic from when I initiate the call to when it fails |
| 02:18.10 | phyrexianslug | "Got SIP response 305 "Use Proxy" back from 10.196.186.75:5060" |
| 02:19.09 | phyrexianslug | I presume .75 is the handset? |
| 02:19.13 | Lefty | yeah |
| 02:19.31 | Lefty | I'll try changing the sip mode on the handset from proxied to peer-to-peer |
| 02:19.52 | [TK]D-Fender | Contact: <sip:10.196.186.70;transport=udp> |
| 02:20.04 | Lefty | that's the server |
| 02:20.05 | [TK]D-Fender | indeed it looks like it's trying to enforce traffic through a proxy |
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| 02:22.29 | Lefty | hmm, yeah, setting the phone to p2p mode makes it unable to register with asterisk |
| 02:23.27 | [TK]D-Fender | Idea : leave it as proxy, but specify it as * |
| 02:24.09 | Lefty | ok, will do that |
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| 02:28.38 | Lefty | still fails, hmm |
| 02:29.55 | phyrexianslug | want to post the phone config? in case something sticks out? |
| 02:30.24 | Lefty | sure... it's about a million lines long |
| 02:32.18 | Lefty | https://gist.github.com/LeftyBC/6589411 |
| 02:33.03 | Lefty | phone is definitely respecting that file, as I changed a couple of allowed features a few minutes ago and it picked it up |
| 02:34.17 | phyrexianslug | and sorry, what firmware was on the phone? |
| 02:35.00 | Lefty | SIP 2.6.10, which I believe is the latest SIP firmware for the unit |
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| 02:39.08 | phyrexianslug | sorry, just trying to model the enviro, Server is .100 phone 1 is .75, phone 2 is .101 ? |
| 02:39.23 | phyrexianslug | or I guess, phone is .100 and .101? |
| 02:39.47 | phyrexianslug | server would be .75? |
| 02:40.13 | Lefty | server is .70, avaya phone is .75, softphone is on .100 |
| 02:40.54 | phyrexianslug | Oh, that makes more sense! |
| 02:41.11 | phyrexianslug | LOOKS to me like something at .101 is registering as the same extension maybe? |
| 02:41.17 | phyrexianslug | " Via: SIP/2.0/UDP 10.196.186.101:5060;branch=z9hG4bK07936a62 " |
| 02:53.42 | Lefty | I don't think so, but I'll tcpdump the damn thing |
| 03:02.02 | Lefty | I'm stumped. I think I'll look at it again in the AM |
| 03:02.13 | Lefty | been staring at SIP traffic for hours! |
| 03:02.24 | Lefty | thanks for the input! |
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| 05:00.55 | j4jackj | Oy vey, oh weh!: Freenode sucks. Let's move over to one of your choice: www.randvids.tk is my own IRC server, and irc.oftc.net is OFTC's. |
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| 05:22.10 | ChannelZ | why |
| 05:25.46 | Penguin | Oh, you know... |
| 05:26.32 | Penguin | Because moving 182 people to his ircd would make it more popular, or something. |
| 05:26.58 | [TK]D-Fender | 182 is more than 1. |
| 05:27.06 | [TK]D-Fender | But ... that's just math speaking |
| 05:27.06 | *** part/#asterisk g_r_eek (~g_r_eek@ppp-94-68-182-4.home.otenet.gr) |
| 05:27.51 | Penguin | I could not care less about yet another ircd. |
| 05:27.59 | Penguin | I don't use the ones I already knew about. |
| 05:29.07 | ChannelZ | Let's at least move to an _established_ shit-hole like EFnet |
| 05:29.20 | Penguin | I quit connecting there years ago. |
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| 05:40.26 | j4jackj | Stop it. I feel like running '/exec -out yes WEED' and you know what that does. |
| 05:45.46 | kaldemar | it kicks you out of the channel? |
| 05:46.20 | j4jackj | It gets you k lined for spam. |
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| 06:13.23 | ChannelZ | sounds like a personal problem |
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| 07:31.33 | mae_tae | hello ppl, we had 1 line that could call can take long distance locally but how can i configure it in freepbx that I could call inside our ofis |
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| 07:46.33 | niplo | I need some help regarding outbound routing. Can someone help me? |
| 07:46.41 | j4jackj | Oh for screw's sake. Why is water now a commodity? |
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| 07:47.28 | ChannelZ | ~ask |
| 07:47.28 | 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. |
| 07:48.16 | ChannelZ | j4jackj: according to our dumb-ass First Lady water is free. |
| 07:48.29 | ChannelZ | so it must be true |
| 07:48.40 | ChannelZ | if you say otherwise you're a racist |
| 07:48.48 | niplo | I want to disallow certain extension from making mobile calls. Is there anyway toachieve this by using the outbound routing? |
| 07:49.18 | j4jackj | niplo: not without a complete list of mobile numbers in the entire world. |
| 07:49.58 | niplo | j4jackj: I cannot use wild cards? for example 69ZXXXXXXX |
| 07:50.12 | j4jackj | niplo: no. |
| 07:50.21 | j4jackj | niplo: well, yes |
| 07:50.23 | ChannelZ | You put the "extensions" (by which you really mean devices/phones) in different contexts and restrict what extensions (proper extensions) can be dialed in those contexts |
| 07:50.54 | j4jackj | niplo: you see, if the mobiles are in a different area code, 'ts all fine. |
| 07:51.04 | mae_tae | hello ppl, example if i direct the phone and call (02) <- area code, then tel no 4289760, it can call long distance, problem is how will i configure it in freepbx, coz i when i add this pattern 02 - 428XXXX, i only hear number is not assigned |
| 07:51.19 | ChannelZ | #freepbx |
| 07:52.22 | mae_tae | ChannelZ: i mean in asterisk how will i specify i think it has the same configuration |
| 07:52.54 | ChannelZ | I don't really understand your question |
| 07:53.16 | j4jackj | mae_tae: to start with, if you are prepending the area code, you may have to strip it off. |
| 07:53.32 | niplo | j4jackj: Well, our agents are only allowed to call local phones not internationally :) |
| 07:53.43 | niplo | *international |
| 07:55.31 | j4jackj | just dial 024281234, because if you add the wait, it's not going to work. |
| 07:55.45 | j4jackj | *024289760 |
| 07:56.46 | mae_tae | j4jackj: something like this 02 - 428XXXX? |
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| 07:56.59 | j4jackj | mae_tae: you don't do that. |
| 07:57.20 | mae_tae | j4jackj: i mean that's how you specify pattern in inbound route |
| 07:57.27 | mae_tae | so how will i do that? |
| 07:57.28 | j4jackj | it is not. |
| 07:57.35 | kaldemar | mae_tae: ask in #freepbx |
| 07:58.11 | j4jackj | mae_tae: you can cludge around it by creating an extension '02' and then asking Asterisk to wait for another extension in a context called 'dial_ohtwo' |
| 08:00.27 | ChannelZ | I still don't understand the question but don't think that's the answer |
| 08:01.40 | j4jackj | it's an idea |
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| 08:02.55 | mae_tae | ok people thanks |
| 08:04.25 | ChannelZ | I guess I'm missing why, if you want to be able to dial 02 and some numbers, you don't just put that in your extension. |
| 08:05.26 | tm1000 | ChannelZ: He/She doesn't understand dialpatterns…at all |
| 08:05.34 | ChannelZ | OK, I am 10 feet away from my WAP. How can my phone have no bars of wifi? |
| 08:05.34 | tm1000 | they has 02 as a strip prefix |
| 08:05.53 | j4jackj | antenna |
| 08:05.56 | j4jackj | or lack thereof |
| 08:08.06 | ChannelZ | Android is just retarded. |
| 08:09.29 | j4jackj | ChannelZ: nope, the hardware is. |
| 08:10.15 | ChannelZ | No, another wifi app is telling me 90% signal but the indicator at the top has nothing but the dot. It's stuck. |
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| 08:11.30 | j4jackj | Oh... |
| 08:11.57 | j4jackj | Anyway, i have this weird schedule where I delay breakfast and have a small snack 2 hours before breakfast. |
| 08:13.19 | ChannelZ | Someone on a norweigan message forum has linked to my lego phone graphic. |
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| 08:26.32 | j4jackj | ChannelZ: now it's getting colder I'm cycling more often. |
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| 09:23.03 | paule32 | hello |
| 09:23.18 | paule32 | now, i can incall and outcall |
| 09:23.24 | paule32 | but |
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| 09:23.42 | paule32 | if i do that, a greeting message welcomes me |
| 09:24.02 | paule32 | how can i do a call, without these message? |
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| 10:45.07 | vlad_starkov | Question: Is it a known bug: I have a peer in sip.conf with 'sendrpid=no' parameter set explicitly, and this peer context inherited from template, which contain 'sendrpid=yes'. As result I expected to have 'sendrpid=no' for that peer, but got 'sendrpid=yes'. |
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| 10:52.03 | LooserOuting | helo. i got problems with t.38 gateway. sometime it finds a v.21 preamble nad sometimes not. i am using always the same faxsender. the destination is in PSTN. i am calling through a SIP Provider. Can someone please give me some hints how to debug the ? |
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| 13:00.52 | LooserOuting | hello. i got problems with t.38 gateway. sometimes it finds a v.21 preamble and sometimes not. i am using always the same fax sender. the destination is in PSTN. i am calling through a SIP Provider. Can someone please give me some hints how to debug the call? |
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| 13:45.58 | dago100 | Hi I've got a problem with dahdi and sip, after 10 to15min on a ISDN line I get the message in asterisk log [2013-09-17 14:32:41] VERBOSE[31034][C-00000025] sig_pri.c: -- Span 1: Channel 0/1 got hangup, cause 27/[2013-09-17 14:32:41] DEBUG[15732][C-00000025] channel.c: Didn't get a frame from channel: DAHDI/i1/ |
| 13:46.09 | dago100 | during the phone |
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| 15:01.04 | monsterco | Hi everyone - I have a provider that send me SIP signal in IP 20.20.20.20 and then I see that they are trying to establish RTP from IP 20.20.20.25 - or maybe my Asterisk is trying to reach them on the new IP based on SIP packets - I have one way-audio issue - where should I start debugging? |
| 15:02.03 | SuperNull | is nat involved? |
| 15:02.27 | [TK]D-Fender | monsterco: show us the complete call debug masking NOTHING |
| 15:02.40 | monsterco | you want sip debug? |
| 15:03.15 | [TK]D-Fender | yes, + verbose 10 |
| 15:03.20 | [TK]D-Fender | ~pb |
| 15:03.20 | 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. |
| 15:03.21 | [TK]D-Fender | ^^^ |
| 15:04.58 | SuperNull | TK you ever get tired of looking at sip text ? ;-) |
| 15:05.45 | [TK]D-Fender | SuperNull: Yes. Right now I'm at zero. Nowhere to go but dowhill... |
| 15:06.52 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 15:13.18 | monsterco | [TK]D-Fender - before I go about doing that let me ask you a question - Should I EVER see SINGLE:NO_TRAFFIC (on pfsense stats) when dialing echo test for example |
| 15:15.38 | [TK]D-Fender | I don't do PF |
| 15:16.14 | [TK]D-Fender | And echo can have a lack of traffic if done wrong as well |
| 15:16.32 | [TK]D-Fender | The only "consistent" I expect... are mistakes. |
| 15:16.43 | [TK]D-Fender | So time to look at what's actually going on. |
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| 15:32.18 | monsterco | [TK]D-Fender - I am preparing SIP debugs now - funny thing - usually in NAT issues call disconnects after a minute - In this case call keeps staying up for ever but with one way audio |
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| 16:28.44 | Katty | WHO BROKE MY ASTERISK |
| 16:28.46 | Katty | blames Qwell |
| 16:33.05 | drmessano | I heard there were cookies |
| 16:33.51 | drmessano | Something about my browser needing to accept cookies from the Asterisk website. I opened my CD tray several times and one did not materialize |
| 16:33.59 | drmessano | Should I file a bug report? |
| 16:34.09 | Katty | i would recommend a cookie report. |
| 16:34.19 | Katty | please include quantities of all ingredients and appropriate cooking temp/times |
| 16:34.34 | Katty | also! |
| 16:34.38 | Katty | since i've never done this before. |
| 16:34.41 | Katty | pokes at jmetro |
| 16:35.02 | Katty | if i'm register asterisk box b onto asterisk box a |
| 16:35.09 | Katty | the type=friend? |
| 16:35.15 | drmessano | I've heard this rumor you can take a boxed cake mix and a can of pumpkin and make an amazing cake or cupcakes. No oil, eggs, or milk.. Just the pumpkin |
| 16:36.11 | Katty | really? |
| 16:36.16 | Katty | sounds kind of...dry |
| 16:36.24 | *** join/#asterisk aruntomar (~Thunderbi@49.248.152.221) |
| 16:36.57 | drmessano | I wondered that as well, but the Pumpkin maybe has more moisture than we're thinking? |
| 16:37.11 | Katty | i don't cook with cake mix |
| 16:37.14 | Katty | so, idk |
| 16:37.24 | Katty | i know you can substitute pumpkin for eggs, as a binding agent |
| 16:37.35 | Katty | and oil is just flavor/moisture |
| 16:37.51 | drmessano | I don't bake much. I can't make cheesecake anymore, and I hate throwing stuff in a pan and hoping for the best 30 minutes later |
| 16:37.56 | drmessano | So I cook. No bake |
| 16:38.16 | drmessano | Cook = Control. I am all 50 shades of stovetop |
| 16:38.18 | Katty | you do know they have no bake cheesecakes, right? |
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| 16:38.31 | *** part/#asterisk thehar (thehar@diddlebox.thehar.com) |
| 16:38.41 | drmessano | No bake cheesecake is terrible as all hell |
| 16:38.48 | Katty | oh no it's not. |
| 16:39.31 | drmessano | If you've ever had a single bite of MY new york style cheesecake, no bake tastes like pig vomit. Actually, it does anyway. |
| 16:39.42 | Katty | pats drmessano |
| 16:39.45 | Katty | maybe you're just doing it wrong. |
| 16:40.07 | drmessano | How can you "do" a no bake cheesecake wrong? |
| 16:40.10 | drmessano | O.o |
| 16:40.12 | drmessano | o.O |
| 16:40.19 | Katty | like you do any other recipe wrong |
| 16:40.22 | Katty | yo udon't follow instructions |
| 16:40.29 | Katty | or you have a crap recipe |
| 16:40.35 | Katty | decides type=friend |
| 16:40.36 | Maliuta | drmessano: that's a whole other pay-per-view site ;P |
| 16:40.40 | drmessano | lol |
| 16:41.05 | drmessano | type=friend creates a user and a peer. One shot config for bidirectional calling |
| 16:41.29 | *** join/#asterisk nam3l3zz (~quassel@86-46-248-134-dynamic.b-ras1.pgs.portlaoise.eircom.net) |
| 16:41.34 | Maliuta | bidirectional ... that's a big word ;) |
| 16:41.46 | pabelanger | ~backtrace |
| 16:41.46 | infobot | backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt). See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 16:42.56 | Katty | drmessano: yeah, that's what i figured. |
| 16:44.28 | [TK]D-Fender | Last time I did a back-trace she got kinda pissed. I probably shouldn't have used a permanent marker... |
| 16:45.34 | Katty | hey look, it registers. |
| 16:45.35 | Katty | how bout that. |
| 16:45.38 | Katty | i can still follow directions |
| 16:46.01 | Katty | good morning mister fender, how's your back doing? |
| 16:47.10 | [TK]D-Fender | So far today is pretty good... I should start swimming soon. |
| 16:47.21 | [TK]D-Fender | take up a membership for a month or two. |
| 16:47.46 | Katty | sounds like a good idea. |
| 16:51.10 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 16:51.21 | drmessano | You know you don't have to swim to get to the states. That's only for mexico |
| 16:51.49 | drmessano | You CAN swim, but the percentage of land border crossings make it insensible |
| 16:52.23 | boom^time | drmessano, the cool kids use underground tunnels these days |
| 16:53.24 | [TK]D-Fender | drmessano: I thought the SWIMMING was the insensible part... |
| 16:53.24 | boom^time | wait sorry I'm getting my illegal immigration and drug trafficking mixed up. |
| 16:56.42 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 17:01.04 | Katty | well. i have both boxes registered with each other now. that's shining progress. |
| 17:02.35 | paulc | hands Katty a gold star |
| 17:02.47 | paulc | I'm all for rewarding little victories. We don't do that enough. |
| 17:03.20 | paulc | cue Katty beaming and gushing a heartfelt acceptance speech... |
| 17:03.38 | paulc | (it's a slow day in the office today - I'll find something to do soon enough, don't worry...) |
| 17:08.35 | Katty | grins |
| 17:08.49 | Katty | well i'm still struggling with making actual calls, but hopefully here soon, eh? |
| 17:12.02 | *** join/#asterisk rocksfrow (~kyle@50-197-9-21-static.hfc.comcastbusiness.net) |
| 17:12.26 | *** join/#asterisk n8ideas (~joshua@65.112.207.3) |
| 17:12.47 | n8ideas | Can anyone tell me if debug threads or disabling compiler optimization would cause Asterisk to run extremely high CPU usage? |
| 17:12.53 | rocksfrow | hi, is asterisk 11 optimized for multi-core support? |
| 17:13.50 | Katty | paulc: annnd now it does ^_^ |
| 17:13.58 | Katty | THANK YOU UHH. |
| 17:14.03 | Katty | THANK YOU MOM |
| 17:14.03 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 17:14.08 | Katty | for all your support, when i was a kid. |
| 17:14.13 | Katty | blah blah blah. blahblha. |
| 17:14.14 | Katty | kthx. |
| 17:14.44 | Penguin | Two asterisks register to each other? That's weird. |
| 17:16.02 | paulc | Maybe not, if they had dynamic public IPs.. at different locations.. maybe? *clutches at straws* |
| 17:16.17 | Penguin | Yeah, maybe. |
| 17:16.44 | Penguin | But if they have static addresses, there is no reason to register either. |
| 17:20.24 | Penguin | And, if this is the setup we discussed several days ago, I provided an example that would serve as a drop-in config. Just change the settings' values to your own values and away you go. |
| 17:20.28 | *** join/#asterisk rbd_ (~rbd@cpe-076-182-043-018.nc.res.rr.com) |
| 17:20.43 | monsterco | [TK]D-Fender - Attached here full SIP and VERBOSE from Asterisk for one way audio - http://pastebin.com/fdSPCdtH - I have put a line of ****** between the call establishment and disconnect signals |
| 17:21.10 | rbd_ | hey guys....how do I convert a string like "10" to an integer 10 so that I can use it in numeric evaluations (i.e. with the >=, < etc operators)? or does asterisk automatically do this? |
| 17:21.31 | monsterco | Provider is sip signaling us from: 107.12.37.20 and sending us media from 107.12.37.25 based on these packets that I see |
| 17:21.55 | Penguin | rbd_: Don't use quotation marks. |
| 17:22.07 | file | n8ideas, debug threads will kill performance |
| 17:22.18 | file | rocksfrow, it will use multiple cores |
| 17:22.39 | n8ideas | Thanks a bunch. That's what we've been seeing |
| 17:22.42 | rbd_ | Penguin:...actually, I want to do soemthing like: GotoIf($[$[${RECOG_RESULT} : ".*?confidence=\"0\.([1-9]+)\""] >= .10]?RECOGNIZED) ....basically taking the captured match from a regexp (which is numeric) and comparing it against a number |
| 17:23.11 | rbd_ | what I have is wrong (missing using \1) but you get the point hopefully |
| 17:26.03 | rbd_ | is this even possible? |
| 17:27.47 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.69) |
| 17:28.42 | monsterco | [TK]D-Fender - any clues yet? |
| 17:29.06 | monsterco | anyone else has any feedback on my one way audio? http://pastebin.com/fdSPCdtH thanks |
| 17:29.21 | rocksfrow | file, would an old 1.4 box have multi-core support as well? how far back does mutli-core support go? |
| 17:29.45 | file | Asterisk is multithreaded for places where it can be and the operating system will divy threads across cores as it can |
| 17:30.00 | file | so, 1.4 is fine for that |
| 17:30.41 | rocksfrow | file, okay, so there aren't any significant efficient increases in the use of multiple cores with a newer asterisk version over an older? Say asterisk 11 vs 1.4 |
| 17:30.49 | rocksfrow | efficiency* |
| 17:31.05 | file | Asterisk does not explicitly do it |
| 17:31.18 | file | now, that doesn't mean stuff hasn't changed in such a way that performance is better |
| 17:31.25 | rocksfrow | file, ah... so Asterisk leaves it all up to the OS/system |
| 17:31.46 | rocksfrow | file, right... some software is optimized to use multiple cores more efficiently so I was jw |
| 17:31.50 | rocksfrow | thanks for clarifying |
| 17:32.08 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 17:32.10 | rocksfrow | i just had somebody ask about multi-cores not appearing to be used and it's an old system so I wanted to make sure it was as simple as an upgrade |
| 17:32.25 | rocksfrow | (an * upgrade I mean) |
| 17:32.39 | rocksfrow | file, right right... gotcha. thanks |
| 17:37.46 | WIMPy | Well, there's one thing that has changed: You can now use ConfBridge instead of MeetMe. |
| 17:37.46 | monsterco | in world of SIP whose responsibility is it to open the RTP ports and initiate media? the receiver or sender of the call? |
| 17:38.31 | WIMPy | Do you want audio to go both ways? |
| 17:39.54 | monsterco | no - getting audio one way |
| 17:39.58 | j4jackj | has anyone here wiresharked Skype protocol before? |
| 17:40.04 | monsterco | I mean yest - both ways |
| 17:40.29 | monsterco | but I am right now getting one way audio because provider is SIP signaling me from one IP and asking us to receive RTP from another IP |
| 17:40.36 | WIMPy | j4jackj: If you try to do so, they will send someone round to take care of you. |
| 17:40.45 | j4jackj | WIMPy: really? |
| 17:40.48 | file | each side listens on their own UDP ports |
| 17:40.53 | monsterco | full SIP debug and verbose from my inbound call: http://pastebin.com/fdSPCdtH |
| 17:41.23 | j4jackj | WIMPy: prove it. it's not illegal to wiretap something that you have permission to wiretap. |
| 17:41.47 | j4jackj | like: a skype call between you and your mum |
| 17:41.58 | WIMPy | j4jackj: There was a story about someone who wanted to write a diploma or something about skype. That took an interesting twist and never appeared. |
| 17:42.41 | file | monsterco, I see nothing out of the ordinary |
| 17:43.59 | monsterco | On my pfSense firewall I see that Asterisk is trying to reach the ITSPs media server and there is one way traffic there |
| 17:44.20 | file | from a SIP signaling perspective it's all fine |
| 17:46.12 | j4jackj | WIMPy: If my mum agrees, I will publish the raw Wireshark findings. |
| 17:46.14 | Penguin | Asterisk doesn't care if the media comes from a different IP address from what the signaling comes from. |
| 17:47.39 | WIMPy | j4jackj: I think the audio codecs are known. |
| 17:47.46 | Penguin | I'd guess that doing some research about pfSense and asterisk might lead you to a solution. |
| 17:47.46 | file | wireshark won't tell you much re: Skype, it's encrypted |
| 17:47.59 | monsterco | @file - is there any missing information? like info that I should receive back from the media server? I would like to know the process here - correct me if I am wrong. ITSP contacts my Asterisk and tells me the media server IP. Then my asterisk should contact that media server? or should it just wait to receive media from that IP? |
| 17:48.20 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-kwqvfzfacefkegjv) |
| 17:48.21 | Katty | so let's say i dial SIP/2000@BoxB |
| 17:48.25 | file | monsterco, it sends to the IP address and port provided in the SDP, and expects media on the IP address and port it provided |
| 17:48.30 | j4jackj | WIMPy: they are. |
| 17:48.34 | Katty | what context is it looking for on the other side? |
| 17:48.41 | WIMPy | monsterco: Both |
| 17:48.43 | Katty | do you set that in the Dial command? |
| 17:48.48 | Penguin | katty: If BoxB has a peer defined in sip.conf, the format shoudl be SIP/BoxB/2000. |
| 17:48.48 | carrar | Katty, context "cake" |
| 17:48.57 | file | Katty, whatever is configured in the remote sip.conf |
| 17:49.00 | Penguin | And the context is set in sip.conf. |
| 17:49.02 | file | on BoxB |
| 17:49.06 | monsterco | So here is what they sent: c=IN IP4 107.12.37.25 |
| 17:49.10 | WIMPy | monsterco: It's not a TCP stream, You have (at least) one stream per direction. |
| 17:49.13 | monsterco | m=audio 26984 RTP/AVP 18 0 101 |
| 17:49.17 | Katty | file: ty |
| 17:49.22 | Katty | carrar: also, yes. how did you guess?! |
| 17:49.36 | monsterco | so - then whose turn is it to contact 107.12.37.25 who has not been in picture so far? |
| 17:49.47 | file | Asterisk will send RTP to that IP address and port. |
| 17:49.51 | carrar | Katty: It's a top secret algorithm that the NSA wrote |
| 17:49.56 | Galaxor | Hi, I am using the sip client built in to android 2.3. Is there a way to set what it sends for outgoing caller id? |
| 17:50.01 | file | it's connectionless via UDP |
| 17:50.39 | monsterco | would my asterisk initiate RTP UDP or the server 107.12.37.25 in an inbound call to me? |
| 17:51.01 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-ohvcscwdfcqqbfnv) |
| 17:51.01 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 17:51.13 | carrar | Katty: Actually it's a constitutionally out of bounds algorithm that the NSA wrote |
| 17:51.19 | file | there is no "initiation" |
| 17:51.30 | file | each side sends media when it needs to, to the address provided |
| 17:51.49 | Penguin | Any initiation is done in SIP; that's what SIP is for. |
| 17:52.34 | j4jackj | WIMPy: I want to write a basic specification sheet about Skype and I will send it with no return postal address but an email address to Microsoft (with copyrights being the only thing breaking anonymity: (C) 2013- Jack Johnson and Microsoft (protocol implementation)) |
| 17:52.43 | drmessano | carrar, s/AllTheWords/REDACTED/ |
| 17:52.57 | monsterco | @file - I see - but at least both sides should expect to receive media from the other party and do respond (or UDP doesn't require any acknowledgement at all)? |
| 17:53.09 | roderickm | WIMPy: there was an interesting academic paper on the Skype protocol (as of late 2005) published http://www.cs.columbia.edu/~salman/publications/skype1_4.pdf |
| 17:53.10 | file | UDP is connectionless and doesn't receive an acknowledgement |
| 17:53.40 | roderickm | Though it has been overhauled signiificantly since, with Microsoft replacing p2p supernodes with their corporate servers. |
| 17:53.52 | monsterco | @file - ok so in this case if call party can hear me then it means my Asterisk is sending audio just fine but if I am not hearing audio it's because the other party is not sending me audio then |
| 17:54.07 | file | or your firewall (most likely) is blocking it |
| 17:54.10 | Penguin | It does not mean they are not sending it. |
| 17:54.15 | Penguin | It means you didn't receive it. |
| 17:54.57 | monsterco | @file - I am using pfSense and I see absolutely nothing from that IP blocked or any STATS from them. However, I do see my Asterisk UDP connections to that IP go as SINGLE:NO_TRAFFIC |
| 17:56.10 | monsterco | which could be just because there is no UDP acknowledgment or maybe there is nothing from other party at all as they are not expecting me?! same setup works with multiple other providers - I have port 5060 UDP (UDP only not TCP) open to ITSP and then 31000-35000 UDP open to whole world |
| 17:56.40 | Galaxor | I'm also not seeing a way to set the outgoing caller id value in Ekiga. Is that something that is not the responsibility of the sip client? |
| 17:56.42 | *** join/#asterisk zigg (~matt@unaffiliated/zigg) |
| 17:57.07 | monsterco | @file : any thoughts? |
| 17:57.09 | file | your signaling is fine, it's either pfSense or the ITSP... that's all I've got for you |
| 17:57.14 | monsterco | gotit |
| 17:57.46 | monsterco | @file - they asked me for wireshark data - since they have two IPs what wireshark command can I use to filter only traffic to and their servers from my Asterisk? |
| 17:58.33 | zigg | monsterco: you could use a filter like "host 1.2.3.4 or host 5.6.7.8" |
| 17:58.54 | monsterco | zigg - can you please give me a complete command - thanks |
| 17:59.48 | zigg | monsterco: you said wireshark, I assumed you were going to use the wireshark filter box. if you're using tcpdump you could do something like "tcpdump -w capture.pcap -s 0 host 1.2.3.4 or host 5.6.7.8" |
| 17:59.53 | monsterco | zigg - something like this: tcpdump -i eth0 -nqtttt -s 0 -A "host 1.2.3.4 or host 5.6.7.8" |
| 18:00.15 | zigg | monsterco: yeah. though I confess I don't know what half of those flags are ;) -i -n and -s are the only ones I'm familiar with |
| 18:00.20 | monsterco | zigg - would I get the same results as wireshark? |
| 18:00.51 | monsterco | I thought tcpdump is same wireshark |
| 18:01.10 | zigg | monsterco: -w capture.pcap dumps it to a .pcap file which you can load in many capture tools including wireshark |
| 18:01.25 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 18:02.25 | monsterco | zigg - ok, so I don't need to do "yum install wireshark" on CentOS and just use TCPDUMP and send them the result for them to analyze in their own wireshark - is that correct? |
| 18:02.48 | zigg | monsterco: yup. that's what I have people do when I need to see their packet captures |
| 18:02.59 | *** join/#asterisk trend1 (~robert@h254.158.185.173.static.ip.windstream.net) |
| 18:03.19 | monsterco | but wireshark also has the capability to collect data? or just analyzes it? |
| 18:04.21 | zigg | monsterco: yes, wireshark can capture on whatever machine you're running it on. but if you're using it normally it also requires a GUI. tshark is the CLI version of wireshark, but I'm not familiar with that |
| 18:04.33 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-ejeelwksiknkulzo) |
| 18:04.35 | zigg | monsterco: wireshark and tshark have the best analytic apabilities |
| 18:04.41 | Penguin | tshark is a nice tool. |
| 18:05.23 | Penguin | tshark output is not identical to tcpdump output, but you can still look at the same stuff in the packets. |
| 18:05.50 | zigg | Penguin: I did use it once for a very specific use case, but most of the time I'm just having people collect with tcpdump and forward it to me to look at on my system in wireshark. but yeah, it is cool |
| 18:06.58 | monsterco | but either of "tcpdump" or "tshark" commands should give me exact same results right? |
| 18:07.20 | monsterco | Penguin - I typed too fast - ops |
| 18:07.41 | *** join/#asterisk zoid_ (~awainer@181.29.125.155) |
| 18:08.11 | zigg | monsterco: the .pcap file format is standard across all packet capture tools. no matter what tool you use to capture it it will be the right format |
| 18:08.20 | monsterco | Penguin - what would be an equivalent command of tshark to : tcpdump -w capture.pcap -s 0 host 1.2.3.4 or host 5.6.7.8 - ITSP asked for wireshark packets so might as well give them that to leave no room to blame |
| 18:09.04 | Penguin | I would venture to say that you can read the tshark man page just as easily as I can. |
| 18:10.48 | Penguin | Also, if they didn't ask for a pcap file or pcap output, I would have assumed they would the human-readable text output rather than a pcap file. |
| 18:11.37 | Penguin | s/would t/would want t/ |
| 18:12.16 | j4jackj | blobblobb |
| 18:13.22 | trend1 | Afternoon all, I was wondering if someone could point me to some good documentation regarding "Virtual Hold". I googled around some but could not find anything substantial. |
| 18:13.48 | Penguin | What does this virtual hold do? |
| 18:15.13 | zoid_ | is there any difference between the h extension and a hangup handler (when you have only one)? |
| 18:15.42 | *** join/#asterisk serafie (~erin@nat/digium/x-vcrktwzbymtfzorj) |
| 18:15.50 | Penguin | Extension h is executed when the call has ended. I don't know what a hangup handler is. |
| 18:16.02 | *** join/#asterisk luke_ftw (~luke_ftw@unaffiliated/luke-ftw/x-9245329) |
| 18:16.35 | trend1 | Penguin: It's a queue option that allows people to leave a number and when there position in queue is up the pbx will call them and connect them to an agent |
| 18:17.42 | Penguin | Oh, I've heard of that. |
| 18:17.50 | *** join/#asterisk asghar144 (~asghar144@host118-24-dynamic.8-87-r.retail.telecomitalia.it) |
| 18:18.16 | trend1 | yeah, its really cool. I admin a high flow Noc and i would like to implement it |
| 18:19.05 | Penguin | http://www.voip-info.org/wiki/view/Asterisk+Queue+Callback |
| 18:19.23 | Penguin | Beware, this info is probably outdated and could even be wrong. |
| 18:19.48 | trend1 | yeah, iv seen that. It's both wrong and outdated |
| 18:19.57 | trend1 | it was a good starting point though |
| 18:20.16 | zoid_ | Penguin: this https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers |
| 18:21.17 | zoid_ | I'm having a problem with the h extension |
| 18:21.56 | zoid_ | I have an agi call, and the call does not end untill the agi call ends |
| 18:22.18 | zoid_ | I don't think I should Hangup() in h |
| 18:22.18 | Penguin | This must be new in 10 or 11. (I use 1.8.) |
| 18:22.56 | Penguin | I don't know why you would need Hangup() in extension h since h only runs after a hangup has occurred. |
| 18:23.55 | zoid_ | Penguin: this is what I'm seeing: I have an AGI in some extension, that agi sends the hangup command, then, the agi at the h extension runs, but the channel remains open untill the second agi script finishes |
| 18:24.21 | zoid_ | asterisk 1.8.10 |
| 18:26.11 | Penguin | I'm wondering if that's an AGI problem. |
| 18:28.00 | elguero | zoid_: Did you go to the cli and read what "core show application AGI" says? If not, go take a look. |
| 18:28.21 | elguero | zoid_: hopefully that helps :) |
| 18:30.07 | Katty | throws cake at carrar |
| 18:30.27 | *** join/#asterisk serafie (~erin@nat/digium/x-oipknbuhxhfnvxrc) |
| 18:30.32 | zoid_ | elguero: I see |
| 18:30.39 | Katty | hugs serafie |
| 18:30.53 | zoid_ | in that case: Hangup should be the first thing after agi |
| 18:30.57 | zoid_ | I'll try that |
| 18:34.25 | carrar | licks up the cake! |
| 18:38.24 | zoid_ | elguero: nope :/ |
| 18:40.17 | Penguin | Did you see the DeadAGI() applications? |
| 18:41.10 | Penguin | s/ns/n/ |
| 18:42.46 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 18:44.47 | navaismo | Can anyone point me at the error, SIP trace and ports look good but i have no audio on the browser. here is the sipo debug with rtp debug on |
| 18:44.49 | navaismo | http://pastebin.com/CVkrZF0Z |
| 18:44.58 | rbd_ | is it possible to use REPLACE() to remove doulbe quotes from a string?? for the life of me I can't get it to work |
| 18:52.31 | *** join/#asterisk volga629 (~bendersky@host7.pythian.com) |
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| 18:56.25 | *** part/#asterisk Galaxor (~Galaxor@208.67.250.157) |
| 18:59.59 | asghar144 | ~pastebin |
| 19:02.53 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 19:04.45 | navaismo | anyone :'( |
| 19:12.00 | *** join/#asterisk lukerobi (~lukerobi@rrcs-97-79-163-146.sw.biz.rr.com) |
| 19:12.28 | lukerobi | Does anyone know of a free echo test service that I can dial to? |
| 19:15.46 | *** join/#asterisk vlad_sta_ (~vlad_star@77.41.88.195) |
| 19:19.46 | j4jackj | lukerobi: 500@ekiga.net |
| 19:20.51 | *** join/#asterisk vlad_starkov (~vlad_star@77.41.88.195) |
| 19:22.29 | *** join/#asterisk MLNoah (~chatzilla@noc.metalink.net) |
| 19:23.48 | MLNoah | is it expected/documented behavior that if you do exten => foo,1,Set(__FOO=1) and then same =>n,Set(FOO=$[${FOO} + 1]) that FOO will be switched back to a non-inheriting channel variable? |
| 19:27.16 | *** join/#asterisk danjenkins (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginmedia.com) |
| 19:27.59 | Penguin | I would have expected that behavior, but I don't know if it has been documented one way or the other. |
| 19:28.06 | *** join/#asterisk davidbowlby (~textual@99-67-52-166.lightspeed.clmboh.sbcglobal.net) |
| 19:28.23 | MLNoah | i mean, i'll totally allow for me not having read (much/any) of the documentation to figure out that was going to happen. but it surprised me that it was doing that |
| 19:30.45 | Penguin | I expect it to be just like you never set it the first time. It's not like you can "initialize" the variable and then the attributes of it remain intact. |
| 19:36.36 | lukerobi | thanks j4jackj.. I was hoping for a DID to call, but i'll settle for anything so I don't have to keep calling random people. No cell service in the building to call myself either.. |
| 19:39.05 | *** join/#asterisk danjenkins (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginmedia.com) |
| 19:41.13 | Penguin | Do you really need a regular phone number for that? |
| 19:41.44 | *** join/#asterisk tris (tristan@2001:1868:a00a::4) |
| 19:43.31 | *** join/#asterisk vlad_starkov (~vlad_star@77.41.88.195) |
| 19:56.15 | navaismo | giving up for now |
| 20:04.42 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 20:13.50 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:14.16 | *** join/#asterisk felipealmeida (~user@177.40.161.69) |
| 20:18.56 | file | navaismo, ICE support has to be present and enabled. |
| 20:19.32 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.69) |
| 20:19.57 | drmessano | ice, ice, baby.. too cold, too cold |
| 20:26.25 | navaismo | file, in the rtp.conf is enabled |
| 20:26.32 | file | it has to be enabled in sip.conf |
| 20:28.37 | navaismo | ok |
| 20:28.41 | navaismo | let me check that |
| 20:32.45 | navaismo | OMG OMG OMG |
| 20:32.49 | navaismo | are wet now |
| 20:32.56 | Qwell | O.o |
| 20:32.57 | navaismo | thansk file you save the day |
| 20:33.03 | file | uh... huh |
| 20:33.36 | navaismo | all the morning trying to find what is the issue, and you pint me in the right direction thanks |
| 20:36.48 | Wiretap | sounds like someone spilled their beer |
| 20:37.19 | navaismo | now sipml5 & jssip work without issue |
| 20:37.23 | file | ICE is a WebRTC requirement, it's disabled by default |
| 20:38.13 | navaismo | taking note of that |
| 20:54.48 | *** join/#asterisk CeBe (~CeBe@port-92-206-114-40.dynamic.qsc.de) |
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| 21:42.56 | *** part/#asterisk asghar144 (~asghar144@host118-24-dynamic.8-87-r.retail.telecomitalia.it) |
| 22:01.13 | *** join/#asterisk danjenkins_ (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginmedia.com) |
| 22:01.27 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 22:08.02 | navaismo | lukerobi, you are the sip device 105 at 192.168.175.50 |
| 22:09.03 | navaismo | ups wrong channel |
| 22:15.18 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 22:30.08 | ChannelZ | You are the sip device. Be the sip device. |
| 22:31.22 | *** join/#asterisk wdbl (~wdbl@ool-44c284f3.dyn.optonline.net) |
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| 23:16.29 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 23:18.34 | Katty | hi laddies. |
| 23:21.10 | *** join/#asterisk asteriskmonkey (~Tardis@74-51-38-204.telnetcommunications.com) |
| 23:44.04 | *** join/#asterisk rfreire (c951a214@gateway/web/freenode/ip.201.81.162.20) |
| 23:46.54 | rfreire | Hello hello, #asterisk! o/ I'm unable to get *any* logging off my Asterisk, be the /var/log/asterisk/messages or CDR messages. It's a 1.8 raspbian build; pastebin at http://pastebin.com/9Zncuwdv |
| 23:46.58 | rfreire | Help? ;-/ |
| 23:47.32 | rfreire | BTW: permissions are correctly set for asterisk:asterisk to /var/log/asterisk |
| 23:56.56 | navaismo | rfreire, check the logger.conf & the cdr_*.conf for the backends |