| 00:03.42 | pabelanger | patrickod: no |
| 00:07.54 | patrickod | ah that's a shame. |
| 00:19.45 | *** join/#asterisk dandate2 (~dan@180.190.200.167) |
| 00:19.55 | dandate2 | can a remote agent use bluetooth to utilize a softphone? |
| 00:20.38 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
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| 00:22.59 | jdavidow | I don't understand the question? |
| 00:23.15 | jdavidow | are you asking IS there a bluetooth SIP phone? |
| 00:23.41 | jdavidow | err, sip/iax/pots/etc? |
| 00:25.58 | jdavidow | a bluetooth headset will show up as a speaker/mic device, so very likely (you may need to add a bluetooth receiver to the computer though!) |
| 00:26.34 | jdavidow | and keep in mind, bluetooth is only like 2-5meter range. |
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| 01:18.42 | Gaiax | saludos |
| 01:18.47 | Gaiax | alguien por aki? |
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| 01:32.15 | patrickod | pabelanger: do you know if it's possible to use realtime configs and phoneprov ? |
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| 01:38.44 | *** join/#asterisk min3r (1000@173.81.252.114) |
| 01:40.09 | min3r | Im using linphone on my local computer, and connecting to an asterisk server. I kept getting calls from "aa" at 200@mylocalipaddress. Im not running asterisk and ext 200 is not an extension on the asterisk server nor are there any other connected IPs to the server. What is going on ? |
| 01:40.40 | min3r | I ran "ss" and did not see any IPs except the IP for the asterisk server. |
| 01:41.31 | min3r | I have received calls from people scanning and receive like 20 calls at once. but this was one call at a time back to back |
| 01:43.44 | min3r | Would these calls be coming from the asterisk server or my local machine (Even though its not running asterisk) Is someone connecting to the softphone ? |
| 01:44.07 | [TK]D-Fender | ~pb |
| 01:44.07 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 01:44.10 | [TK]D-Fender | show us |
| 01:44.42 | min3r | Show you what exactly |
| 01:45.06 | min3r | When the calls were occuring I did not see them in the asterisk cli |
| 01:46.52 | [TK]D-Fender | Enable SIP DEBUG at * cli to see the requests |
| 01:46.58 | min3r | Also, I do not see anything in the /var/log/asterisk/full. So im assumming maybe it did not come from the asterisk server |
| 01:47.39 | [TK]D-Fender | No, first forget logs. * live CLI with SIP debug. that is what is really happening right now. |
| 01:49.29 | min3r | I ran sip debug on |
| 01:49.35 | min3r | the calls are not occuring at this moment though |
| 01:50.05 | min3r | What Im trying to find out is generally what is occuring when these types of things happen ? |
| 01:52.00 | min3r | Now I just see my softphone from my ip address. and udp read requests from the server ip. |
| 01:54.35 | [TK]D-Fender | Maybe you were being scanned from the outside. But it isn't happening now and there is nothing to see and time travel is kinda stuck on +1s/s so that's not going to get too far |
| 01:55.03 | *** join/#asterisk BuenGenio (~Gene@203.145.92.163) |
| 01:55.17 | [TK]D-Fender | So I'd recoend locking down your peers that you do have, doing the sae to your dialpla, and perhaps setting up fail2ban to look for brute force attempts |
| 01:55.44 | min3r | Well, normally when I see a scan i receive lots of calls at once. this was 1 call at a time, once i hung up. fail2ban is on |
| 01:57.25 | min3r | [TK]D-Fender, what type of scan/scanner is generally used that would make a sip call? |
| 01:58.11 | p3nguin | Wow. That sounds pretty cool... |
| 01:58.40 | p3nguin | Recording of a call when the disk hit 100% usage, the sound is wild. |
| 01:59.11 | p3nguin | Like digital diarrhea or something. |
| 01:59.13 | min3r | p3nguin, almost like a echo techno/feedback noise ? |
| 01:59.34 | p3nguin | You've heard this sound on movies. |
| 01:59.42 | p3nguin | You can just hear the bits. |
| 02:00.00 | min3r | yeah you can simulate the sound sound by calling yourself and setting the record to Master volume rather than microphone |
| 02:00.11 | min3r | Ive ran into that using arecord to record local calls |
| 02:01.31 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 02:02.20 | p3nguin | I was suddenly getting a flood on the console: WARNING[3467]: format_wav_gsm.c:224 update_header: Unable to find our position |
| 02:02.34 | p3nguin | As soon as I cleared some free space, it stopped. |
| 02:04.57 | min3r | [TK]D-Fender, are they making a "skinny" connection or doing something else ? |
| 02:10.57 | [TK]D-Fender | min3r: You haven't shown any debug. How would expect someone to be able to answer that? |
| 02:11.12 | min3r | im speaking in general, nm man |
| 02:11.38 | [TK]D-Fender | You're asking me is it is one specific protocol that is being used. |
| 02:11.44 | [TK]D-Fender | I can't know. We don't see anything |
| 02:11.45 | min3r | I dont have any debug to show, as we mentioned it's not occuring at the moment |
| 02:12.04 | [TK]D-Fender | Statistically it'll be SIP |
| 02:12.06 | min3r | What do people generally use / do when they make such a scan |
| 02:12.14 | [TK]D-Fender | Skinny is a Cisco proprietay protocol. |
| 02:13.04 | [TK]D-Fender | And they'll use soe sort of script to find a responsive server, try to reg as a known device and then start trying to dial numbers to defraud you of LD on your outbound |
| 02:14.01 | min3r | since they were able to call my ext, were they able to register? or can they contact my ext without registeing ? |
| 02:14.13 | min3r | it was coming from ext 200 which is not a valid ext on the server |
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| 02:16.33 | min3r | http://www.voipsa.org/Resources/tools.php -- found that site. im checking it out. |
| 02:17.34 | [TK]D-Fender | Look in your CDR to see what calls actually stuck |
| 02:20.38 | min3r | They didnt make any outgoing calls |
| 02:21.00 | min3r | according to my cdr-csv |
| 02:21.37 | [TK]D-Fender | min3r: well so far we're grabbing at ghosts. |
| 02:21.50 | [TK]D-Fender | So secure what you have now |
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| 03:56.31 | Tech_Travis | Any troubleshooting advice on why Dahdi recognizes my TDM410P in the OS but in the Asterisk CLI the command module load chan_dahdi.so fails? |
| 03:57.10 | [TK]D-Fender | show us the before & after.... |
| 04:02.49 | Tech_Travis | Is this what you meant by before and after? http://pastebin.com/9vzm8RpM |
| 04:06.26 | [TK]D-Fender | Tech_Travis: Good start. start * manually "asterisk -gvvvvc" |
| 04:06.45 | [TK]D-Fender | Tech_Travis: And pastebin. That should probably bomb on load |
| 04:06.57 | [TK]D-Fender | include all of your dahdi configs in the next PB |
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| 04:30.09 | Tech_Travis | [TK]D-Fender: http://pastebin.com/YwLuETa2 |
| 04:31.24 | [TK]D-Fender | That looks like it's actually ok... |
| 04:31.29 | [TK]D-Fender | "dahdi show status" |
| 04:31.36 | [TK]D-Fender | "dahdi show channels" |
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| 04:32.17 | [TK]D-Fender | You have pretty barren configs that you'll need to rework, but it might list the 1 you have if you're lucky |
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| 04:40.43 | Tech_Travis | [TK]D-Fender: I don't think it's my week for good luck. http://pastebin.com/cKF4py1p |
| 04:42.52 | WIMPy | 'core set verbose 9' |
| 04:42.55 | [TK]D-Fender | module unload chan_dahdi.so |
| 04:42.58 | [TK]D-Fender | module load chan_dahdi.so |
| 04:43.00 | WIMPy | 'core set debug 9' |
| 04:43.09 | WIMPy | 'module load chan_dahdi.so' |
| 04:46.20 | Tech_Travis | [TK]D-Fender: http://pastebin.com/BpmtYLxW |
| 04:47.43 | Tech_Travis | WIMPy: http://pastebin.com/wDdKWU1r |
| 04:47.50 | WIMPy | Did you turn up verbose and debug on that one? |
| 04:47.55 | [TK]D-Fender | techPerhaps you've had a kernel update etc and chan_dahdi needs to be recompiled... |
| 04:48.10 | Tech_Travis | WIMPy: yes, to 9, but still no change. |
| 04:48.26 | WIMPy | The second paste only shows an unload. |
| 04:48.39 | Tech_Travis | WIMPy: oops, |
| 04:50.21 | Tech_Travis | WIMPy: http://pastebin.com/9FAkf0Xg |
| 04:51.33 | Tech_Travis | [TK]D-Fender: When I upgraded to * 1.8.9 today I did a recompile on libpri, dahdi complete and the new *, but I can run through it again. |
| 04:51.45 | WIMPy | Hmm. That was not very helpful. |
| 04:52.04 | Tech_Travis | WIMPy: yeah, it's been like that all day. |
| 04:52.09 | WIMPy | The output f Asterisk, that is. |
| 04:52.57 | Tech_Travis | WIMPy: no matter what I've tried in configuring dahdi it's been the same output. |
| 04:54.12 | Tech_Travis | [TK]D-Fender: I did come across some posts mentioning a kernel PAE issue but wasn't understanding what they were referring to. I can try to dig up those posts again. |
| 04:54.50 | WIMPy | cat /proc/dahdi/* |
| 04:56.10 | Tech_Travis | The only other thing I can think of is that this is not in production so the analog line isn't plugged into the card yet. Would that make a difference in not being able to get * to see dahdi? |
| 04:56.27 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 04:56.36 | WIMPy | no |
| 04:56.54 | p3nguin | Oh lookie... 1.8.9.0 dies a terrible death. http://pastebin.com/am0JE5cU |
| 04:57.54 | Tech_Travis | WIMPy: http://pastebin.com/uiuMPZhN |
| 04:58.04 | WIMPy | Didn;t you want to disable jabber in order to get rid of those messages anyway? *eg* |
| 04:58.35 | WIMPy | Ok, so the card seems to be working so far. |
| 04:58.39 | SeRi | p3nguin: you ever got your issue resolved? |
| 04:58.47 | p3nguin | Which one? |
| 04:59.02 | [TK]D-Fender | ok, off for the night... later all |
| 04:59.05 | SeRi | the google voice one |
| 04:59.09 | p3nguin | No and yes. |
| 04:59.27 | Tech_Travis | [TK]D-Fender: thanks for taking a look. |
| 04:59.30 | p3nguin | Ultimately I took care of the problem. |
| 04:59.46 | SeRi | ? |
| 04:59.46 | SeRi | how? |
| 05:00.20 | p3nguin | Log in to gtalk using an IM client. Show all buddies. Delete (not block) all the spammers from your list. Log out of gtalk IM client. |
| 05:00.41 | p3nguin | I did it with Pidgin. Right-click, remove. |
| 05:01.05 | SeRi | ok but it worked even though the otehr user has you in their contact? |
| 05:01.32 | p3nguin | That part doesn't matter. |
| 05:01.50 | p3nguin | The part that matters is that you were somehow subscribed to those users. |
| 05:02.08 | p3nguin | Remove the subscription, done. |
| 05:02.17 | p3nguin | You can block them before removing if you want. |
| 05:02.44 | p3nguin | I'm never signed into chat, so I don't worry about getting messages from them. |
| 05:02.46 | SeRi | well I just removed them. Going to try it out now. |
| 05:02.58 | p3nguin | If they start sending messages, I'll block them. |
| 05:03.19 | Tech_Travis | WIMPy: it was also doing this on Asterisk 1.8.7.2 before I upgraded to 1.8.9.0 |
| 05:03.26 | p3nguin | It was earlier today when I removed them, and I haven't seen one more single PRESENCE PACKET notice. |
| 05:03.51 | p3nguin | 1.8.9.0 did not work out for me. |
| 05:04.14 | p3nguin | I'll have to retest in a few days or weeks. |
| 05:04.41 | SeRi | ok here is hwat flods my logs Got presence packet from |
| 05:04.45 | WIMPy | Tech_Travis: I can only think of it disliking somethign in the config file. But if it won;t tell, it's hard to say. |
| 05:05.18 | SeRi | Got presence packet from user@gmail.com, someone not in our roster!!!! |
| 05:05.46 | SeRi | It's floding my logs right now |
| 05:06.29 | Tech_Travis | WIMPy: any thoughts on which config file to start tweaking? chan_dahdi.conf perhaps? |
| 05:06.38 | WIMPy | yes |
| 05:06.55 | WIMPy | The only other file it reads is users.conf, if you've got one. |
| 05:07.48 | SeRi | so that means they have not add it you to their google chat |
| 05:07.59 | SeRi | cause I removed mine but I am still seen the msges |
| 05:08.17 | Tech_Travis | WIMPy: I started building chan_dahdi.conf from scratch so it is pretty bare. Am not using a users file. What are the must haves in the chan_dahdi.conf file? |
| 05:08.58 | Tech_Travis | WIMPy: all I've got are trunk signaling and channel |
| 05:10.07 | WIMPy | Trunk? Yes, signalling and channel are essential. |
| 05:12.24 | Tech_Travis | WIMPy: I tried it with trunk=DAHDI/1 and without trunk in the file under the [trunkgroups] section. |
| 05:14.19 | WIMPy | trunkgroups are for NFAS only AFAIK, so you shouldn;t have that. |
| 05:14.33 | p3nguin | seri: I removed the users from my IM client and the shit went away. |
| 05:14.43 | Tech_Travis | WIMPy: okay I'll take it back out. |
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| 05:20.45 | p3nguin | seri: I had been getting the flood periodically for several days. After I removed those turds from my list, I haven't seen one more single notice about presence packets from users not on the roster. |
| 05:22.07 | SeRi | yea. I just found that if they have you on their list their client will send you a present packet and since they are not in your list asterisk will complain.... so what I did right now is set buddy=user@gmail.com for each one of them and it supressed the messages... at least so far. |
| 05:22.12 | SeRi | p3nguin: ^^ |
| 05:22.37 | p3nguin | ewww |
| 05:22.58 | p3nguin | Try adding them to your list in IM. Then block them. Then remove them. |
| 05:22.58 | SeRi | lol |
| 05:23.08 | SeRi | good suggestion |
| 05:23.50 | p3nguin | There's no way I would add configuration for turds in my jabber.conf or gtalk.conf. |
| 05:25.52 | SeRi | well so far it has not complaint since they been blocked... |
| 05:26.51 | SeRi | now I need to find out why is not able to dial out |
| 05:27.47 | p3nguin | Bad extension, probably. |
| 05:27.49 | Tech_Travis | WIMPy: I think it might be working now.... |
| 05:28.07 | SeRi | chan_gtalk.c:1864 gtalk_request: Could not find recipient |
| 05:28.11 | WIMPy | "might"? |
| 05:28.17 | SeRi | Unable to create channel of type 'gtalk' (cause 0 - Unknown) |
| 05:29.13 | Tech_Travis | WIMPy: Well, since it never worked to begin with I don't have a frame of reference to know if it is fully functional now or just partly. |
| 05:29.14 | p3nguin | gtalk and jabber not loaded correctly? |
| 05:29.46 | Tech_Travis | WIMPy: but now in the CLI I have some entries for dahdi |
| 05:29.59 | WIMPy | 'dahdi show status' |
| 05:30.04 | WIMPy | or ...channels |
| 05:30.44 | SeRi | p3nguin: which module needs to be reloaded when you make changes? |
| 05:30.50 | SeRi | gtalk or jabber? |
| 05:30.58 | SeRi | I for got which one needs a manual reload |
| 05:31.30 | p3nguin | both |
| 05:31.34 | p3nguin | unload, then load |
| 05:32.38 | SeRi | bingo :) |
| 05:32.41 | SeRi | all works now |
| 05:32.48 | SeRi | just for got to reload them |
| 05:33.11 | Tech_Travis | WIMPy: http://pastebin.com/vfG3zyQd |
| 05:34.11 | WIMPy | Looks like it's working. |
| 05:34.55 | Tech_Travis | WIMPy: that what I like to hear! |
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| 05:43.10 | Tech_Travis | WIMPy: thanks for taking the time to help me out. I appreciate it. |
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| 09:39.28 | krotos | hi all |
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| 10:06.40 | anaxagoras | Good morning, channel |
| 10:08.12 | anaxagoras | I got the time to test asterisk a bit and have an installation running with sipgate-account and several softphones, etc. Everything telephone related works flawlessly. |
| 10:09.25 | anaxagoras | My problem is regarding jabber. I enabled the gtalk module and can send messages, but receiving jabber messages is working. How can i check, if the receive application is installed? |
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| 11:50.08 | sehh | hey people |
| 11:50.18 | sehh | anyone using mISDN? |
| 11:50.36 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 11:52.14 | sehh | when there is an incoming call, the first thing I see in the asterisk console, is this line: |
| 11:52.15 | sehh | <PROTECTED> |
| 11:52.26 | *** join/#asterisk bitwize (bitwize@109.58.96.144.bredband.tre.se) |
| 11:52.28 | sehh | any ides what "failed" is all about? |
| 11:55.07 | kaldemar | you don't get a number in |
| 11:56.04 | kaldemar | so it falls back to exten s since no extension in from-zaptel matches "nothing". |
| 11:56.05 | sehh | i'm sorry i don't understand, you mean Caller ID number? or DID from the trunk? |
| 11:56.22 | kaldemar | DID. the dialed number. |
| 11:56.52 | sehh | why is my DID empty? |
| 11:57.28 | kaldemar | no idea. |
| 11:57.32 | sehh | hmm |
| 11:57.53 | sehh | if DID is empty then I can't route incoming calls based on which line they called |
| 11:58.00 | sehh | this is a problem for me |
| 11:58.36 | sehh | is there a way to debug this? maybe I've broken it by some configuration parameter? |
| 11:59.58 | kaldemar | i don't know about mISDN, with DAHDI you can enable debug on an ISDN. what version of asterisk are you using? |
| 12:00.37 | sehh | i'm still using 1.6 |
| 12:01.03 | sehh | things work fine, incoming and outgoing calls |
| 12:01.23 | sehh | but now I need to route calls based on the DID |
| 12:01.35 | sehh | but it's empty (as you confirmed) |
| 12:03.15 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 12:03.15 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 12:03.40 | cusco | in s extension use DumpChan() to see what you have |
| 12:07.21 | sehh | I don't really know how to do that |
| 12:07.37 | sehh | maybe mISDN is the problem and it is not passing on the DID info |
| 12:15.20 | bitwize | I have a question regarding cancelation of calls originated through AMI, is it possible to cancel the originated call before it is answered? |
| 12:20.55 | kaldemar | sehh: try "pri intense debug span 1" or "pri se debug span 1" or something like that. i'm not 100 % sure on the command. |
| 12:21.30 | kaldemar | sehh: you'll see what you get in from the line in the signalling. |
| 12:33.56 | sehh | kaldemar, thanks man, I'll try it and report back |
| 12:34.24 | cusco | bitwize: you can just return Busy(); instead of Answer(); |
| 12:35.14 | sehh | kaldemar, seems like that only works for PRI lines, while mine is a BRI |
| 12:35.41 | sehh | its an ISDN line and I'm using a Beronet 2-port ISDN |
| 12:35.48 | bitwize | cusco: i see, but this senario is when i originate a call from my asterisk server to the pstn |
| 12:36.53 | cusco | ow... |
| 12:36.59 | cusco | you can define for how long the dial takes |
| 12:38.00 | kaldemar | bitwize: manager has a hangup action, use it for the created channel. |
| 12:38.55 | bitwize | I know, but I need this feature when a user initiates a new call and for some reason want to cancel that call before its answered, and i have to do this through AMI. |
| 12:39.22 | bitwize | kaldemar: how do i know the created channel before I receive the originate response from AMI? |
| 12:40.33 | kaldemar | bitwize: you don't. but you do before an answer. from a new channel event or by listing current channels. |
| 12:41.56 | kaldemar | if that doesn't suit you, maybe you should turn the origination around. |
| 12:44.23 | bitwize | ahh of course, I can see this in the channel-created event i suppose? Can i map these events to the correct originate-action with some key? Maybe the action-id from the originate action? |
| 12:45.22 | bitwize | (sorry for my bad english) |
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| 13:01.05 | kaldemar | bitwize: you know the channel you're dialing don't you? |
| 13:04.51 | bitwize | kaldemar: true, I thought that my provided channel name was renamed with some added value |
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| 15:11.26 | WIMPy | Moin |
| 15:12.20 | krotos | hi :) |
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| 16:58.03 | WIMPy | General warning: If you're using ext4, don't use a kernel >3.1.5. Soewhere between 3.1.5 and 3.2 something nasty happened. |
| 16:59.03 | WIMPy | had a great night with ext4-fs errors. The reproducible kind on both 3.2 and 3.2.2. |
| 17:15.47 | *** join/#asterisk DennisG (~dennisg@541AFD1E.cm-5-3d.dynamic.ziggo.nl) |
| 17:16.55 | DennisG | hi everyone |
| 17:17.13 | DennisG | is it possible to use dynamic/wildcard contexts? |
| 17:17.54 | [TK]D-Fender | Context names are fixed |
| 17:18.30 | DennisG | too bad.. |
| 17:18.40 | DennisG | maybe i have to say my problem in a different way |
| 17:18.51 | WIMPy | wonders what that might be used for. |
| 17:19.25 | DennisG | i wanna put al my ivr's and sip phone routing in my database, because these are very dynamic |
| 17:20.14 | DennisG | but my Asterisk system is a multi tendant system. So multiple companies use this to save rackspace and keep the costs low |
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| 17:21.15 | DennisG | aka i have like 10 customers with each 1 ~ 5 ivrs with multiple steps. In total the ivrs have around 250 ~ 500 contexts |
| 17:21.58 | DennisG | thats why i wanna use some dynamic contexts or something like that to get the realtime switch to work |
| 17:24.03 | [TK]D-Fender | Since that's not how it works you're going to have to use another approach just like the rest do |
| 17:24.38 | DennisG | oke and what is a good approach in best practices terms? |
| 17:25.17 | DennisG | i can use "empty" contexts in the extensions file and point them all to my database. If Asterisk don't care if there are like 500 contexts in the system |
| 17:26.11 | [TK]D-Fender | Doesn't as far as I know |
| 17:27.20 | *** join/#asterisk bbourdage (~BBourdage@ip67-91-166-45.z166-91-67.customer.algx.net) |
| 17:27.50 | DennisG | Oke thanks! then i gonna build a script that generates around 1000 contexts so i can add some new customers haha. I think that Asterisk gonna hate me if I issue a reload command |
| 17:29.10 | cusco | DennisG: what I do, I have a sql table with ddi's and whenever a call comes in I actually query the sql for the context associated with that ddi |
| 17:29.58 | cusco | but there must still be some dialplan in that context, ofcourse |
| 17:30.49 | DennisG | Cusco: oke so you build your extensions.conf via the database? |
| 17:30.59 | DennisG | like a DIY realtime system |
| 17:31.57 | cusco | no, I buil them by hand. If you want some tool to build ivr you should checkout 3rd party solutions such as 'edgebox' |
| 17:33.16 | DennisG | ooh oke, no i build my IVR systems by hand. But my customers wanna have regular changes to there IVRs |
| 17:33.25 | DennisG | thanks anyway for the edgebox solution :) |
| 17:34.06 | DennisG | I thought it was possible to let my customers change the IVRs by themself via a custom GUI (via a website) that speaks with Asterisk realtime |
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| 17:34.25 | cusco | you can have them run agi scripts |
| 17:34.33 | cusco | that would be easier to implement |
| 17:34.42 | cusco | anyway all of it means work |
| 17:34.55 | DennisG | I know but Agi scripts are very heavy (have I heard) |
| 17:35.09 | DennisG | but that was on Asterisk 1.4, maybe it's now different |
| 17:36.40 | DennisG | Are Agi scripts still so heavy? |
| 17:45.37 | cusco | no |
| 17:45.40 | cusco | agi is a script |
| 17:45.49 | cusco | and a scirpt will only be as heavy as any script |
| 17:45.56 | cusco | more lines, the heavier |
| 17:46.01 | cusco | use a profiller to find out |
| 17:47.00 | DennisG | Oke then i gonna use agi hehe |
| 17:47.26 | DennisG | but first have i to find out what's the best approach for this |
| 17:47.30 | DennisG | thank you very much cusco! |
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| 20:54.07 | volga629 | I have asterisk 1.8 and web meet me is not working e: 127.0.0.1 failed to authenticate as 'meet_me' |
| 20:54.43 | WIMPy | Did you set up that user? |
| 20:54.49 | volga629 | I tried change 10 all setting and check permissions, bu no luck |
| 20:55.13 | volga629 | yes user the there I can see from CLI |
| 20:55.27 | volga629 | times |
| 20:57.24 | volga629 | http://fpaste.org/P0Qt/ |
| 20:59.01 | WIMPy | So if the user exists, the remaining parts are the password and the SCL. |
| 20:59.05 | WIMPy | ACL |
| 20:59.35 | volga629 | Error connection to the manager! |
| 21:00.06 | volga629 | I have for right now 0.0.0.0/0.0.0.0 0.0.0.0/0.0.0.0 |
| 21:02.34 | volga629 | might be some thing with webmeetme ? |
| 21:19.48 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 21:26.18 | ChannelZ | why is that listed twice? |
| 21:27.49 | bitwize | volga629: Have you assigned the correct ip-range for that manager in manager.conf? |
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| 21:37.03 | volga629 | what you mean range like access list ? |
| 21:38.30 | volga629 | enabled = yes |
| 21:38.30 | volga629 | port = 5038 |
| 21:38.30 | volga629 | bindaddr = 0.0.0.0 |
| 21:39.04 | volga629 | access-list is 0.0.0.0/0.0.0.0 0.0.0.0/0.0.0.0 |
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| 21:39.46 | WIMPy | The last line doesn't make sense. |
| 21:40.04 | ChannelZ | yeah, it's "permit" |
| 21:40.18 | volga629 | deny all allow all |
| 21:40.31 | ChannelZ | that doesn't make any sense either |
| 21:41.17 | volga629 | <PROTECTED> |
| 21:41.37 | ChannelZ | it's not a range. You're not listening |
| 21:41.52 | ChannelZ | permit=0.0.0.0/0.0.0.0 |
| 21:41.54 | ChannelZ | that's all |
| 21:42.09 | volga629 | permit=127.0.0.1/255.255.255.0 |
| 21:42.33 | volga629 | that in my manager.conf |
| 21:43.49 | volga629 | deny=0.0.0.0/0.0.0.0 |
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| 21:48.20 | volga629 | what is right way add this exten => s,n,MeetMe() ? |
| 21:48.33 | ChannelZ | then it seems like your password is wrong |
| 21:49.40 | volga629 | I checke already, but let me check again and will something simple no special characters |
| 21:49.48 | volga629 | put |
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| 21:52.05 | ChannelZ | As for your extension, not sure what the question really is, that is a right way to add it assuming it's a part of other 's' exten priorities (since you specified 'n') |
| 21:53.37 | volga629 | how I can find where to add this for web meetme |
| 21:54.50 | ChannelZ | well it's an extension, so it goes in extensions.conf - how this relates to "web meetme" I have no idea |
| 21:55.14 | ChannelZ | If I had to guess it probably has to go in some specific context |
| 21:56.29 | volga629 | That allow dynamic room interaction ? |
| 22:00.33 | volga629 | still getting == Connect attempt from '127.0.0.1' unable to authenticate after password change |
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| 22:03.28 | ChannelZ | did you reload manager? |
| 22:03.50 | ChannelZ | and turn on verbose a little and you'll see more notices from manager as to what it possibly doesn't like |
| 22:03.53 | volga629 | yes run reload on asterisk |
| 22:03.54 | ChannelZ | core set verbose 3 |
| 22:10.48 | volga629 | It look like working I removed % from password |
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| 22:41.37 | bitwize | Question: If I have a couple of managers connected through AMI and all of them listening for originate-response events, do all managers get the event when it occurr or just the manager who initiated the originate-action? |
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| 23:05.02 | jamicque | Hi, is there any way to extract extra SIP headers from REFFER in dialplan? Coz SIP_HEAER function only shows the headers from invites... |
| 23:06.10 | pabelanger | jamicque: no, just the INVITE message |
| 23:07.13 | jamicque | pabelanger: I guess that no by SIP_HEADER and any other func. |
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