| 00:00.02 | WIMPy | Don't do that. |
| 00:00.14 | spengler1 | okay I will get rid of it |
| 00:00.16 | Penguin | You mean you copied and pasted without knowing what it means. |
| 00:00.58 | spengler1 | i forget why i did it ; i have been working on this on and off since march |
| 00:01.24 | WIMPy | Then you should have had enough time to find out. |
| 00:01.53 | navaismo | destroyed many ~books today |
| 00:02.03 | navaismo | ~book |
| 00:02.03 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 00:02.08 | navaismo | spengler1, ^ |
| 00:02.45 | spengler1 | I took that out |
| 00:02.56 | spengler1 | looks better now |
| 00:03.12 | Penguin | And how about those numbered priorities? |
| 00:03.39 | WIMPy | And Queue() doesn't auto-answer, does it? |
| 00:03.41 | spengler1 | same => n??? |
| 00:03.47 | Penguin | If you had used n instead of numbers, you wouldn't have had to reorder the ones you didn't remove. |
| 00:03.59 | Penguin | Queue() never answered in the past. |
| 00:04.07 | spengler1 | WIMPy ; it answers if someone is in queue |
| 00:04.13 | Penguin | Since when? |
| 00:04.28 | Penguin | Shall we take a look at app_queue? |
| 00:04.43 | navaismo | or redesign asterisk |
| 00:05.12 | navaismo | or create freeswitch v2 |
| 00:05.25 | [TK]D-Fender | Penguin: app_queue always ansers the call uless you set the option for it not to |
| 00:05.31 | WIMPy | Something new whould be really nice. |
| 00:05.48 | Penguin | That must be new. |
| 00:05.51 | Penguin | It never did that before. |
| 00:06.10 | spengler1 | i mean it answers if an agent is in the queue |
| 00:10.19 | Penguin | That used to be a big problem for people. If that has been changed, that would be nice. |
| 00:11.00 | *** join/#asterisk pensmit (~pensmit@unaffiliated/pensmit) |
| 00:11.02 | WIMPy | Why? It's nice to be able to queue calls without answering them. |
| 00:11.17 | WIMPy | Or might even be a legal requirement. |
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| 00:34.06 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 01:19.36 | spengler1 | well WIMPy ; you aren't really queuing calls when joinwhenempty is set to no |
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| 01:46.14 | carbinemonoxide | I am having issues with asterisk connecting calls using Google Voice. When I call someone, if they pick up my SIP phone continues to ring. Asterisk sees that they pick up but never connects us. http://pastie.org/private/mo6xjd91rrfvo5aj6oytg |
| 01:46.18 | carbinemonoxide | It was working fine then one day, without me changing anything it started doing this. |
| 01:47.01 | [TK]D-Fender | just seeing one pack back from Google is not enough debug to do anything meaningful with |
| 01:47.18 | [TK]D-Fender | We'd need to see the entire actual comms, including those of your SIP phone |
| 01:47.45 | [TK]D-Fender | Along with all the other pertinent details of your environment |
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| 07:28.57 | jeffspeff | what's up everybody? |
| 07:29.08 | ChannelZ | The moon |
| 07:29.15 | jeffspeff | true |
| 07:30.32 | jeffspeff | this channel is pretty dead at this hour |
| 07:30.47 | ChannelZ | on weekends |
| 07:31.21 | jeffspeff | i'm trying to wrap up a few things for work before i quit for the weekend |
| 07:31.37 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 07:31.59 | jeffspeff | having some issues witha curl command... the #linux and #curl channels are dead too |
| 07:33.15 | ChannelZ | do tell |
| 07:33.31 | jeffspeff | i'll copy paste. just a sec |
| 07:33.44 | jeffspeff | what's wrong with this command? curl -w %{http_code} --silent https://api.url.com/v2/phone/$1?format=pbx&account_sid=123&auth_token=1234565&cache=false |
| 07:33.53 | jeffspeff | if i do the same url in a browser, it works. this curl always responds with a http 404 code |
| 07:34.17 | jeffspeff | this is running in a bash srcipt on centos |
| 07:36.17 | jeffspeff | ChannelZ, any ideas? |
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| 07:37.56 | ChannelZ | I imagine it's the & |
| 07:38.00 | ChannelZ | escape them with \ |
| 07:39.02 | jeffspeff | the output of that command is set as a variable |
| 07:39.12 | ChannelZ | huh? |
| 07:39.37 | jeffspeff | it's ran as resul=`curl command` |
| 07:39.49 | jeffspeff | s/resul/result |
| 07:40.09 | ChannelZ | ok.. still & is a shell character to background the task |
| 07:40.33 | jeffspeff | ok. let me give that a shot |
| 07:40.49 | ChannelZ | what is the $1 bit |
| 07:42.27 | jeffspeff | it's a variable set in the script |
| 07:44.19 | ChannelZ | ah |
| 07:44.41 | jeffspeff | holy shit, the back-slashes worked! |
| 07:44.59 | jeffspeff | thank you sir |
| 07:45.06 | ChannelZ | sure |
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| 08:38.00 | j4jackj | I made a file that you can play to anyone who has done something so as not to deserve this hour's meal. |
| 08:38.03 | j4jackj | http://www.randvids.tk:8080/nofood4u.wav |
| 08:49.16 | ChannelZ | .... ..- .... |
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| 10:51.37 | [sr] | ai |
| 10:57.54 | WIMPy | oi |
| 10:58.26 | [sr] | WIMPy: :) |
| 10:58.29 | [sr] | how are you? |
| 10:59.04 | WIMPy | Ok, just trying to do more than I can ;-) |
| 10:59.37 | [sr] | im kind sleepy also.. |
| 10:59.58 | [sr] | just trying to rest and not thinking that for the 4th time in my life someone i care is going to leave country |
| 11:00.02 | [sr] | ***k |
| 11:02.20 | WIMPy | Here most people leave, at lesast to the next federal land. |
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| 11:03.35 | [sr] | ya but that's near |
| 11:03.57 | WIMPy | Sometimes |
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| 11:18.13 | [sr] | WIMPy: i unserstand... the far concept is relative sometimes... but hey 4 times? damn |
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| 11:21.06 | WIMPy | I can't count the number of friends who left. |
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| 12:43.44 | paule32 | hello |
| 12:44.16 | paule32 | which ports must be open for asterisk? |
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| 12:59.04 | paule32 | ok the doc says 10000-20000 |
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| 13:15.22 | paule32 | ok, i can now speak to the world outside my pbx box |
| 13:15.34 | paule32 | but incoming calls doesn't reach the asterisk server |
| 13:15.50 | paule32 | what can i do? |
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| 13:25.56 | carbinemonoxide | Anyone know how to get xmpp debugging to show up in my debug logs? |
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| 14:18.22 | paule32 | what does this mean: |
| 14:18.26 | paule32 | [Sep 14 16:16:49] NOTICE[10029]: chan_sip.c:22759 handle_request_invite: Sending fake auth rejection for device "jkallup" <sip:001001@fritz.fonwlan.box>;tag=7F1CBD54B31B27BF |
| 14:18.28 | paule32 | ? |
| 14:22.00 | [TK]D-Fender | it's just an auth challenge |
| 14:23.59 | paule32 | in sip.conf? |
| 14:25.29 | [TK]D-Fender | alwaysauthreject=yes <- |
| 14:25.32 | [TK]D-Fender | this causes it |
| 14:25.44 | [TK]D-Fender | and is normally a GOOD thing |
| 14:26.04 | [TK]D-Fender | this is just a notice, not a problem. |
| 14:38.31 | paule32 | http://codepad.org/H6bvWns2 |
| 14:38.50 | paule32 | this is my actual config |
| 14:41.32 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 14:52.33 | paule32 | i have a context "default" and an other context "deutschland" |
| 14:52.56 | paule32 | in default: i do include => deutschland |
| 14:54.03 | paule32 | but asterisk tells me, that the included extension in deutschalnd is not in default |
| 14:54.40 | paule32 | [Sep 14 16:50:53] NOTICE[11912]: chan_sip.c:22866 handle_request_invite: Call from '1001' (192.168.178.79:5061) to extension '03xxxxxxxxx' rejected because extension not found in context 'default'. |
| 15:03.42 | [TK]D-Fender | paule32: "dialplan show" <- pastebin from CLI |
| 15:09.55 | paule32 | http://codepad.org/8Rnjkjrh |
| 15:11.01 | [TK]D-Fender | looks like LUA has taken over [default] |
| 15:11.30 | [TK]D-Fender | you also have a lor of AEL in there you should disable if you're not using |
| 15:11.48 | [TK]D-Fender | "noload => pbx_ael.so" <- add to modules.conf |
| 15:12.26 | [TK]D-Fender | Actually |
| 15:12.32 | [TK]D-Fender | ... that doesn't match AT ALL |
| 15:12.37 | [TK]D-Fender | exten => 1001,2,Playback(hello.world) |
| 15:12.41 | [TK]D-Fender | your "config" |
| 15:12.50 | [TK]D-Fender | <PROTECTED> |
| 15:13.04 | [TK]D-Fender | What CLI shows does NOT have the "." and has a "-" |
| 15:13.12 | [TK]D-Fender | Looks like you didn't reload after making changes |
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| 15:27.37 | paule32 | re |
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| 15:52.46 | paule32 | [Sep 14 17:52:23] NOTICE[3848]: chan_sip.c:22866 handle_request_invite: Call from '620' (192.168.178.1:5060) to extension 's' rejected because extension not found in context 'deutschland'. |
| 15:54.47 | paule32 | http://codepad.org/QF4BxQaB |
| 15:55.58 | [TK]D-Fender | paule32: this new dialplan looks nothing like the first dialplan you showed me today ... and nothing like the "dialplan show" I asked for after. None of your code is ever matching. Also you should not be dialing a raw IP like that. Set up a peer again for it. |
| 15:56.39 | [TK]D-Fender | paule32: as for this message... ti is telling you exactly what it is looking for a match for, and where. You don't have a match there.... so you might want to consider making one so it can process that call. You DO want to process it, don't you? |
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| 17:23.30 | carbinemonoxide | I'm trying to debug my asterisk server, but I can't seem to get my xmpp debug info to show up in the debug log. I can see it in the console but it's a pain to read that way |
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| 18:12.10 | *** join/#asterisk Geek-Linux (~mubbashir@182.185.102.158) |
| 18:14.30 | Geek-Linux | hello |
| 18:15.31 | Geek-Linux | How r u all, i am facing a problem with my asterisk server of contineous crash due to segfault, can any one help me please |
| 18:18.15 | Geek-Linux | i have searched, but only way left was to check on irc, if some one knows. |
| 18:24.49 | *** join/#asterisk navaismo (~navaismo@189.241.77.253) |
| 18:27.16 | Geek-Linux | How r u all, i am facing a problem with my asterisk server of contineous crash due to segfault, can any one help me please |
| 18:36.39 | Geek-Linux | How r u all, i am facing a problem with my asterisk server of contineous crash due to segfault, can any one help me please |
| 18:36.42 | *** join/#asterisk xpheres (~root@91-65-34-26-dynip.superkabel.de) |
| 18:36.51 | xpheres | hello |
| 18:36.59 | carbinemonoxide | Hello xpheres |
| 18:37.18 | xpheres | anyone has been successfull configuring a cisco 7940 hardphone for sip? |
| 18:37.42 | carbinemonoxide | http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a0080094584.shtml |
| 18:38.24 | xpheres | thank you |
| 18:38.29 | navaismo | Geek-Linux, you need to collect the backtrace https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 18:38.44 | xpheres | I already saw that website carbinemonoxide |
| 18:38.59 | xpheres | but the tftp server doesnt seem to work |
| 18:39.07 | xpheres | I dont know what am I doing wrong |
| 18:39.18 | Geek-Linux | Thanks navaismo, ya i did and i got segment fault in it. |
| 18:39.47 | Geek-Linux | Navaismo. Now the reason behind is not known |
| 18:39.53 | *** join/#asterisk spengler1 (~spengler1@pool-98-117-213-86.bltmmd.fios.verizon.net) |
| 18:40.33 | carbinemonoxide | Anyone know how to get my xmpp debugging info into my debug logs? Right now it just shows up in my CLI and it's impossible to read that way |
| 18:46.27 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 18:48.04 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 18:54.57 | paule32 | i can't get working |
| 18:55.00 | paule32 | [Sep 14 20:54:09] NOTICE[13446]: chan_sip.c:22866 handle_request_invite: Call from '620' (94.222.34.237:5060) to extension '36917034687' rejected because extension not found in context 'myprovider'. |
| 18:55.16 | Penguin | Do you understand what that says? |
| 18:55.31 | paule32 | [myprovider] |
| 18:55.31 | paule32 | exten => _0X.,1,Dial(SIP/192.168.178.1/${EXTEN},20) |
| 18:55.47 | Penguin | It says extension 36917034687. |
| 18:55.48 | paule32 | the number is not in context Penguin ? |
| 18:56.01 | Penguin | 36917034687 does not match _0X. |
| 18:56.17 | Penguin | And you still haven't fixed that dial string. |
| 18:56.28 | Penguin | Dial(SIP/192.168.178.1 <---- still wrong. |
| 18:56.50 | navaismo | Geek-Linux, you need to publish your backtrace to the IRC via pastebin or in the JIRA and someone will help you |
| 18:57.08 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 18:57.21 | navaismo | seriously that guy need a print copy of the book |
| 18:57.25 | navaismo | ~book |
| 18:57.25 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:57.32 | Penguin | paule32 does, yes. |
| 18:57.47 | Penguin | I'm going to start billing him for every time I have to repeat the same thing. |
| 18:58.05 | navaismo | IGNORE *!*@dslb-094-222-034-237.pools.arcor-ip.net |
| 18:58.12 | navaismo | fuuu misisng slash |
| 18:58.14 | paule32 | oops |
| 19:00.02 | spengler1 | do u guys work as asterisk consultants? |
| 19:01.30 | paule32 | [Sep 14 21:00:45] NOTICE[13896]: chan_sip.c:22866 handle_request_invite: Call from '620' (94.222.34.237:5060) to extension '123456789' rejected because extension not found in context 'myprovider'. |
| 19:02.04 | paule32 | [myprovider] |
| 19:02.05 | paule32 | exten => 123456789.,1,Dial(SIP/620/49${EXTEN:1},20) |
| 19:02.39 | [TK]D-Fender | paule32: and what does "." mean in that pattern? |
| 19:03.00 | [TK]D-Fender | primes up the double-whammy |
| 19:03.08 | paule32 | oops, type mistake |
| 19:03.14 | [TK]D-Fender | Both of them |
| 19:04.04 | pabelanger | paule32, *CLI> dialplan show 123456789@myprovider |
| 19:04.08 | pabelanger | is a great debug tool |
| 19:04.37 | [TK]D-Fender | psSince he has virtually nothing in his contexts... there isn't much to mix up in there |
| 19:04.41 | [TK]D-Fender | pabelanger: * |
| 19:04.59 | [TK]D-Fender | pabelanger: He has consistently failed to read up on the very basics of dialplan patterns |
| 19:05.19 | spengler1 | anyone ever get termination errors between att -> vitelity? |
| 19:05.54 | [TK]D-Fender | spengler1: Could you elaborate on "errors". That is impractically vague. |
| 19:05.56 | paule32 | http://codepad.org/AdhdvmuJ |
| 19:06.39 | spengler1 | [TK]D-Fender ; calls made from the ATT network to Vitelity fail ; it's very random but it happens ; maybe 1 call out of 50 |
| 19:07.09 | spengler1 | it has nothing to do with Asterisk because it does it with a standalone ATA as weell |
| 19:07.13 | [TK]D-Fender | spengler1: What kind fo failure? Where's the debug? |
| 19:07.14 | spengler1 | just wondering |
| 19:07.59 | spengler1 | the att girl jumps on and says the call cannot be completed as dialed |
| 19:08.36 | spengler1 | just wondering if it is a widespread issue |
| 19:12.06 | paule32 | debian*CLI> dialplan show myprovider |
| 19:12.06 | paule32 | [ Context 'myprovider' created by 'pbx_config' ] |
| 19:12.06 | paule32 | <PROTECTED> |
| 19:12.06 | paule32 | -= 1 extension (1 priority) in 1 context. = |
| 19:14.31 | [TK]D-Fender | paule32: You keep just pasting random bits of code without communicating a QUESTION. |
| 19:17.01 | ChannelZ | "Do you find my extensions sexy?" |
| 19:17.55 | [TK]D-Fender | C'mon babay tell me so! </stewart> |
| 19:18.08 | paule32 | sorry for my showings, im testing and testing, reading and reading, but nothing will work |
| 19:18.55 | [TK]D-Fender | paule32: What isn't working is your ability to show us something complete and ask us a proper question about it |
| 19:19.14 | [TK]D-Fender | [15:11]paule32'036917034688' => 1. Dial(SIP/620/49${EXTEN},20) [pbx_config] <- I';m sure this line will do exactly what it says it will |
| 19:19.45 | [TK]D-Fender | (which mind you... seems to be pointless.. but hey, that's another matter) |
| 19:20.11 | paule32 | im not sure |
| 19:20.18 | [TK]D-Fender | Not sure of what? |
| 19:20.43 | paule32 | i have a router, config it with user phone 620 |
| 19:20.51 | paule32 | give it a password |
| 19:21.10 | paule32 | then i register the pbx at the router |
| 19:21.12 | [TK]D-Fender | That backstory does help... |
| 19:21.27 | [TK]D-Fender | What helps is looking at what is actually happening |
| 19:21.31 | spengler1 | lmao |
| 19:21.38 | [TK]D-Fender | Which you aren't doing |
| 19:21.41 | [TK]D-Fender | does not* |
| 19:21.44 | paule32 | register => 630:<password>@192.168.178.1 |
| 19:21.54 | spengler1 | this is a story about a boy, his router and the user phone 620 |
| 19:22.01 | [TK]D-Fender | paule32: Stop and look at what is HAPPENING. |
| 19:22.21 | paule32 | oops 620 sorry |
| 19:22.37 | Penguin | Who are you registering to? |
| 19:22.38 | ChannelZ | you don't register asterisk to its self |
| 19:23.11 | paule32 | sip show registry display me the sip router |
| 19:23.18 | Penguin | Who are you registering to? |
| 19:23.22 | *** join/#asterisk joesmoe (~joesmoe@admins.phreefilez.com) |
| 19:23.33 | Penguin | 192.168.178.1 is whose address? |
| 19:23.37 | paule32 | asterisk runs in vm box 192.168.178.200 |
| 19:23.44 | Penguin | 192.168.178.1 is whose address? |
| 19:23.46 | Penguin | Who are you registering to? |
| 19:23.49 | [TK]D-Fender | paule32: I can stare at my bank balance of $530 for HOURS and ask the world "Why can't I complete my purchase?!?! I have the $530 you see right there!". This is the part where one should notice that the item I'm trying to buy is $1000 and the cashier is telling me to "fuck off" |
| 19:24.03 | xpheres | in order to make a tftp connection to a cisco hardphone should I set up the tftp server 1 option to the local ip address where I run the tftp server? |
| 19:24.15 | [TK]D-Fender | paule32: Stop wasting time with the useless story. LOOK AT THE CALL. |
| 19:24.30 | paule32 | and the pbx is started and bind on vm box 192.168.178.200 |
| 19:24.31 | navaismo | maybe you guys need to start asking if he understand the concept of creating peers registering etc etc |
| 19:24.33 | Penguin | Who are you registering to? |
| 19:24.34 | Penguin | 192.168.178.1 is whose address? |
| 19:24.41 | [TK]D-Fender | Penguin: not the problem |
| 19:24.47 | Penguin | paule32: Who are you registering to? |
| 19:24.49 | navaismo | concept problems |
| 19:24.52 | Penguin | paule32: 192.168.178.1 is whose address? |
| 19:25.05 | [TK]D-Fender | Ok, I guess everyone else is blind too. |
| 19:25.10 | [TK]D-Fender | Welcome to "the weekend" |
| 19:25.10 | paule32 | of the router at home Penguin |
| 19:25.15 | navaismo | xpheres, usually yes |
| 19:25.16 | ChannelZ | He can't call anything by the sounds of it.. he's defined a peer, and is trying to register the same asterisk to that peer. |
| 19:25.21 | xpheres | ok |
| 19:25.37 | [TK]D-Fender | ChannelZ: Not the problem. |
| 19:25.40 | navaismo | see concept issues |
| 19:25.40 | Penguin | paule32: Is there an asterisk on that router at home? |
| 19:25.43 | ChannelZ | I'd say it's a huge problem. |
| 19:25.44 | [TK]D-Fender | None of you are getting it |
| 19:25.50 | ChannelZ | He has no idea what he's doing. |
| 19:26.06 | navaismo | i would expect osmosis work one day and the book enter in our minds magically |
| 19:26.11 | xpheres | I dont know what happened, I installed the tftp server, I copied the files to the tftboot folder, I changed the tftp ip address on the phone to the machine where I run the server |
| 19:26.12 | [TK]D-Fender | Neither does anyone else here because none of you were paying attention. |
| 19:26.32 | xpheres | and nothing happens, the telephone keeps showing all the time configuring cm list, etc |
| 19:26.33 | paule32 | on the router, no, but on the vm box, the router has sip features call through, so you can do calls to pc |
| 19:26.42 | navaismo | xpheres, enable the tftp verbose and see if the files are requested |
| 19:26.56 | xpheres | what is tftp verbose? |
| 19:27.00 | Penguin | paule32: Does the router at 192.168.178.1 have a SIP server? Why are you registering your asterisk to it? |
| 19:27.17 | navaismo | xpheres, the verbose of the TFTP LOL! |
| 19:27.37 | xpheres | I dont know how to do it |
| 19:27.45 | [TK]D-Fender | paule32: Stop. Now. You are wasting your time on this. You do not have a registration problem. You do not have a peer problem. You have a problem of being distracted and not LOOKING AT THE CALL. |
| 19:27.46 | navaismo | xpheres, it show when some device is trying to pull files |
| 19:27.51 | paule32 | Penguin, not directly a SIP server, oppurnuties |
| 19:28.01 | paule32 | so you can make voip calls |
| 19:28.15 | paule32 | it is a fritz box |
| 19:28.15 | navaismo | xpheres, are you using centos and the tftp on xinet? |
| 19:28.32 | Penguin | I guess I don't understand the topology. I can't see why you would want to register your asterisk to your router. |
| 19:28.33 | xpheres | I have asterisk in a virtual machine with centos |
| 19:28.43 | xpheres | I installed based on freepbx distro |
| 19:28.58 | xpheres | but the tftp server I have in my machine, out of the virtual machine |
| 19:29.20 | navaismo | like pumpking server? |
| 19:29.35 | navaismo | well the verbose/logging depends on each software |
| 19:30.17 | paule32 | [TK]D-Fender, so how i interpret the message, is, that i call the internal phone number 620 on the router |
| 19:30.29 | paule32 | this call goes to pbx |
| 19:30.33 | [TK]D-Fender | paullLOOK at the message. Paste it HERE |
| 19:30.40 | paule32 | and can't find extension in myphones |
| 19:30.45 | navaismo | xpheres, which CentOS 6 or 5? |
| 19:30.51 | xpheres | 6 I think |
| 19:31.02 | [TK]D-Fender | paulShow us the message and what you expect it to match. |
| 19:31.03 | xpheres | but as I told you the ftpt is installed in another machine |
| 19:31.05 | xpheres | ubuntu |
| 19:31.07 | navaismo | xpheres http://n40lab.wordpress.com/2013/01/29/centos-6-3-installing-a-tftpd-server-for-uploading-configuration-files/ |
| 19:31.17 | paule32 | [Sep 14 21:18:42] NOTICE[14859]: chan_sip.c:22866 handle_request_invite: Call from '620' (94.222.34.237:5060) to extension '4936917034688' rejected because extension not found in context 'myprovider'. |
| 19:31.32 | [TK]D-Fender | paule32: because there is NO MATCH |
| 19:31.32 | Penguin | Did you ever learn what that message means? |
| 19:31.50 | navaismo | xpheres, ok, you need to enable the logging/verbose and see if your tftp is wroking and receiving connections |
| 19:31.55 | Penguin | It means exactly what it says. '4936917034688' is not in context 'myprovider'. |
| 19:32.33 | paule32 | this stands in my extensions.conf: |
| 19:32.35 | paule32 | [myprovider] |
| 19:32.36 | paule32 | exten => 4936917034688,1,Dial(SIP/620/${EXTEN}) |
| 19:32.55 | [TK]D-Fender | paule32: "dialplan show" <- |
| 19:32.57 | ChannelZ | is that dialplan loaded? |
| 19:33.51 | xpheres | when I try to change the /etc/xinetd.d/tftp file, the file is blank, there is nothing, I think the file is not created |
| 19:34.05 | xpheres | when I try to restart xinetd I have the following message |
| 19:34.32 | xpheres | stop: Rejected send message, 1 matched rules; type="method_call", sender=":1.56" (uid=1000 pid=17317 comm="stop xinetd ") interface="com.ubuntu.Upstart0_6.Job" member="Stop" error name="(unset)" requested_reply="0" destination="com.ubuntu.Upstart" (uid=0 pid=1 comm="/sbin/init") |
| 19:34.38 | paule32 | this is the dialplan: http://codepad.org/vu86rgZh |
| 19:35.11 | navaismo | so it begins... |
| 19:35.17 | [TK]D-Fender | paule32: [ Context 'myprovider' created by 'pbx_config' ] '36917034688' => 1. Dial(SIP/620/49${EXTEN}) [pbx_config] |
| 19:35.23 | [TK]D-Fender | paule32: NOT THE SAME NUMBER |
| 19:35.34 | navaismo | grab the popcorns |
| 19:35.38 | ChannelZ | 36917034688 != 4936917034688 |
| 19:35.39 | [TK]D-Fender | paule32: You are not applying your changes when updating your configs |
| 19:35.52 | paule32 | reload ? |
| 19:35.55 | [TK]D-Fender | YES |
| 19:35.59 | navaismo | xpheres, http://mohammadthalif.wordpress.com/2010/03/05/installing-and-testing-tftpd-in-ubuntudebian/ |
| 19:36.02 | Penguin | Every single time when you are done editing extensions.conf, you have to run dialplan reload. |
| 19:36.05 | Penguin | dialplan reload |
| 19:36.06 | Penguin | every time |
| 19:36.17 | ChannelZ | do it randomly even! |
| 19:36.18 | Penguin | Change extensions.conf, dialplan reload. |
| 19:36.20 | Penguin | Got it? |
| 19:36.22 | Penguin | dialplan reload? |
| 19:36.24 | Penguin | When? |
| 19:36.25 | [TK]D-Fender | paule32: You've done this several times already and I've warned you about it. You don't seem to be learning. |
| 19:36.45 | navaismo | imagine Penguin slapping the book in his table after those questions |
| 19:37.03 | Penguin | I'm about to go around running dialplan reload on every asterisk I can find... just for the hell of it. |
| 19:37.34 | paule32 | please don't worry, im a beginner, i have disadvanrage, and i can't very well english |
| 19:37.43 | Penguin | dialplan reload |
| 19:37.55 | Penguin | This is the same in each language, since it is the asterisk command. |
| 19:38.16 | paule32 | yes Penguin , i mean [TK]D-Fender |
| 19:38.56 | [TK]D-Fender | paule32: learn the commands to RELOAD yoru configs. Same applies to sip.conf and all the others |
| 19:39.09 | [TK]D-Fender | paule32: Asterisk does not read them live constantly |
| 19:39.34 | Penguin | If you edit a config, you must reload the component for that config. |
| 19:39.43 | [TK]D-Fender | paule32: when I ask you to look at "dialplan show" it's because no-one should trust that your config file was even ever read |
| 19:40.03 | [TK]D-Fender | paule32: Pay attention to what * is actually doing. |
| 19:41.11 | navaismo | its hard to understand a foreign language you know |
| 19:41.58 | paule32 | yes, ok, you right! but the other peoples here post messages any time and the text scrolls up, so it is not simple to follow one of you, but i know, im not alone here |
| 19:44.35 | Geek-Linux | Navaismo, what is pastebin or JIRA i am new to IRC so dont know them ? |
| 19:44.43 | Penguin | ~pb |
| 19:44.43 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:44.46 | Penguin | ^ |
| 19:45.26 | navaismo | xpheres, if you are using freepbx maybe you may take a look on the OSS ENDPOINT MANAGER |
| 19:45.28 | [TK]D-Fender | paule32: You can stare at configs all day long and wonder why things don't work. if you aren't looking at the CALL to prove what is coming in, you won't see that it doesn't amtch what you think it should. |
| 19:47.17 | [TK]D-Fender | "Why can't I drive to work? My car is fine!" - Answer : The roads were all blown up by a massive bombing strike, so who cares about your car being alright? No roads = GAME OVER |
| 19:47.47 | ChannelZ | Amateur. I have a flying car. |
| 19:47.47 | navaismo | jeep 4x4 |
| 19:47.51 | Penguin | |
| 19:48.22 | raidghost | things you put in sip.conf like example #include file.conf (doesnt make asterisk put info that the file.conf asks asterisk to work with) |
| 19:48.44 | raidghost | Any reason why it seems like asterisk is ignoring things put in sip.conf that starts with #include ? |
| 19:48.47 | paule32 | ok, the call will be answer, 1 ring, and then hangup |
| 19:49.23 | [TK]D-Fender | raidghost: No. |
| 19:49.53 | paule32 | http://codepad.org/tVsQtIOT this is now the output |
| 19:50.43 | Penguin | lol |
| 19:51.02 | Penguin | One thing after another. |
| 19:51.28 | [TK]D-Fender | paule32: "Too many open files" <- massive error |
| 19:51.29 | ChannelZ | raidghost: your syntax is wrong? the file doesn't exist? |
| 19:51.38 | [TK]D-Fender | paule32: with your entire filesystem |
| 19:52.15 | [TK]D-Fender | [Sep 14 21:43:36] ERROR[16169]: cdr_sqlite3_custom.c:278 write_cdr: disk I/O error. SQL: INSERT INTO cdr (calldate,clid,dcontext,channel,d |
| 19:52.29 | [TK]D-Fender | Can't write to AstDB, can't write to CDR |
| 19:52.31 | ChannelZ | sounds like time for a reboot |
| 19:52.35 | [TK]D-Fender | your system is SCREWED UP right now |
| 19:52.40 | paule32 | [myphones] |
| 19:52.41 | paule32 | ;exten => 1001,1,NoCDR() |
| 19:52.41 | paule32 | exten => 1001,1,Dial(SIP/1001,20) |
| 19:52.51 | paule32 | i comment out nodr line |
| 19:52.51 | [TK]D-Fender | paule32: Stop pastebing dialplan |
| 19:52.54 | Penguin | My system is screw from running that command! |
| 19:53.03 | ChannelZ | Yes! |
| 19:53.03 | *** part/#asterisk xpheres (~root@91-65-34-26-dynip.superkabel.de) |
| 19:53.07 | ChannelZ | I forgot about my system is screw |
| 19:53.08 | paule32 | then the messages were away |
| 19:53.09 | [TK]D-Fender | paule32: Your entire opperating environment is SCREWED right now |
| 19:53.20 | paule32 | reboot? |
| 19:53.27 | [TK]D-Fender | paule32: at a minimum. |
| 19:53.47 | [TK]D-Fender | [15:51][TK]D-Fenderpaule32: "Too many open files" <- massive error |
| 19:53.59 | [TK]D-Fender | This is preventing all processes from continuing |
| 19:54.21 | paule32 | ok will back in one minute reboot |
| 19:54.47 | ChannelZ | can't wait |
| 19:56.34 | navaismo | I just hope has the pbx and the im cleint in the same machine and didnt reboot his PC instead the pbx |
| 19:56.57 | Penguin | hahaha |
| 19:57.20 | Penguin | His asterisk is in a virtual machine. |
| 19:57.36 | raidghost | Fun when you are like % done with the billing thingy. and the stupiid gui does not save the voip info to file. |
| 19:57.44 | raidghost | 98% done |
| 19:58.25 | [TK]D-Fender | What does "save the voip info" mean? |
| 19:59.51 | raidghost | Well. a2billing is supposed to save the voip info direct to /etc/asterisk/additional_a2billing.sip.conf (since its defined in a2billing to do that) |
| 20:00.29 | raidghost | and i have setin sip.conf #include /etc/asterisk/additional_a2billing_sip.conf and the iax.conf |
| 20:00.51 | [TK]D-Fender | Ah, an a2billing problem! |
| 20:00.54 | raidghost | Tried without the location, and doesnt happend a thing. |
| 20:00.55 | [TK]D-Fender | </caring> |
| 20:01.11 | [TK]D-Fender | As long as Asterisk works! |
| 20:01.45 | Penguin | When you run sip reload, the cli would output some information about loading sip.conf and included files. That information should provide a clue. |
| 20:01.55 | raidghost | I have set debug mode on in asterisk. Trying to get the info to the file: asterisk doesnt say a thing aboute something "maybe been done" |
| 20:02.11 | ChannelZ | like maybe it has no permission to write to /etc/asterisk and never made a file. |
| 20:02.22 | raidghost | the file is owned by www-data |
| 20:02.32 | [TK]D-Fender | maybe it did make a file and Asterisk can't open it. |
| 20:02.44 | raidghost | i did make the file manualy |
| 20:02.46 | [TK]D-Fender | Maybe it did make a file and the include is in a bad place |
| 20:02.52 | raidghost | following the install.srt file |
| 20:03.21 | Penguin | Did they send a dodge.srt4 file, too? |
| 20:04.00 | raidghost | you are not funny |
| 20:04.34 | Penguin | At this point, I only care what the cli said when you ran sip reload. |
| 20:05.37 | raidghost | *reading some more dry manual* |
| 20:06.48 | [TK]D-Fender | I also don't trust that Asterisk belongs to the right groups, that the permissions on the file are sane, that it's even in the right place, that the #include itself is in a sane place, or that the contents of this hypothetical file are meaningful to be merged where it is supposedly happening. |
| 20:07.14 | Penguin | Skeptic. |
| 20:07.21 | [TK]D-Fender | And wisely so |
| 20:08.27 | raidghost | so [TK]D-Fender you saying its a file permission issue Mostly? |
| 20:09.03 | [TK]D-Fender | raidghost: No, I'm saying that is one of the half-dozen-plu point I just made. |
| 20:09.07 | [TK]D-Fender | raidghost: NONE of which I trust |
| 20:09.22 | [TK]D-Fender | raidghost: YOU are showing nothing so I'm not going to waste my time playing psychic |
| 20:09.49 | raidghost | [TK]D-Fender: Why you beeing so Loud with your voice,. I know that debug asterisk should give answers |
| 20:09.52 | raidghost | But it doesnt |
| 20:09.56 | Geek-Linux | navaismo, i have pasted it in pastebin, can you have a look at the backtrace log. |
| 20:10.00 | *** join/#asterisk paule32 (~paul@dslb-094-222-034-237.pools.arcor-ip.net) |
| 20:10.02 | paule32 | re |
| 20:10.08 | [TK]D-Fender | raidghost: to you. Who knows what your eyes see or just how far you're looking |
| 20:10.17 | paule32 | so i increase the ulimit -n |
| 20:11.26 | paule32 | but when i try a call to the world, it will ring 1 times, and then the console/apbx will be disconnected |
| 20:11.39 | [TK]D-Fender | paule32: OK |
| 20:14.58 | paule32 | debian*CLI> |
| 20:14.58 | paule32 | Disconnected from Asterisk server |
| 20:16.24 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:16.25 | Penguin | exten => _.,1,System(asterisk -rx 'core stop now') |
| 20:16.58 | ChannelZ | My system is screw after running that command |
| 20:17.06 | Penguin | hahaha! |
| 20:17.43 | ChannelZ | I think someone else's system is screw if it cores on a Dial |
| 20:18.01 | Penguin | <Trupsalms> my system is screw from running that command |
| 20:18.08 | navaismo | Geek-Linux, where is the link |
| 20:18.18 | navaismo | i cant take a look but i dont understand anything haha |
| 20:19.51 | navaismo | s/cant/can/ |
| 20:23.32 | Geek-Linux | http://pastebin.com/j1FiawQT |
| 20:24.24 | [TK]D-Fender | Geek-Linux: You haven't told us what you're running yet.... |
| 20:24.48 | Geek-Linux | Means. I am running asterisk |
| 20:24.58 | [TK]D-Fender | Geek-Linux: You haven't told us what you're running yet.... |
| 20:26.32 | Geek-Linux | i am running asterisk boxs and all are configured with trunk to transfer calls, every thing was ok but now they are crashing in specific time from 9 AM till 9 PM and rest they are ok |
| 20:26.47 | [TK]D-Fender | Geek-Linux: .... VERSION <----------------------------- |
| 20:26.49 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 20:27.24 | navaismo | Geek-Linux, also collect the backtrace as described in the wiki |
| 20:28.00 | Geek-Linux | asterisk version is 1.6 |
| 20:28.11 | Penguin | That isn't a version. |
| 20:28.19 | [TK]D-Fender | Geek-Linux: Or a branch |
| 20:28.26 | [TK]D-Fender | Geek-Linux: Or supported at all. |
| 20:28.48 | [TK]D-Fender | Geek-Linux: 1.6.0, 1.6.1, 1.6.2 branches are all EOL. NO support at all. |
| 20:28.53 | [TK]D-Fender | Geek-Linux: Upgrade |
| 20:28.58 | Penguin | ~upgrade |
| 20:28.58 | infobot | Upgrading is easy! Go that way, really fast. If something gets in your way, turn. |
| 20:29.16 | paule32 | this is my actual config: http://codepad.org/kD4QM3E6 |
| 20:29.45 | Penguin | You don't need to Answer() before a Dial(). |
| 20:30.15 | [TK]D-Fender | paule32: Config is HALF of the picture. The other is what is happening. |
| 20:30.49 | [TK]D-Fender | paule32: If your Asterisk instance CRASHES, it isn't because of your configs |
| 20:31.14 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 20:31.35 | paule32 | i do 2 reboots, with same error |
| 20:31.37 | Penguin | The 620 peer is well-misconfigured, too. |
| 20:32.00 | [TK]D-Fender | [16:31]paule32i do 2 reboots, with same error <- your overall system is screwed up and unstable then |
| 20:32.15 | [TK]D-Fender | paule32: what version of * are you running on it? |
| 20:32.29 | paule32 | 10.0 |
| 20:32.47 | [TK]D-Fender | paule32: Not supported either |
| 20:33.13 | [TK]D-Fender | paule32: And I told you this last time and you said you didn't ahve time to upgrade.... even though this is not a production machine and you had nmothing to lose |
| 20:33.30 | [TK]D-Fender | paule32: and you were running an RC, and not even the FULL first release of that branch. |
| 20:33.37 | [TK]D-Fender | paule32: UPGRADE |
| 20:34.00 | paule32 | have you a wget link? |
| 20:34.10 | [TK]D-Fender | paule32: www.digium.com <- |
| 20:34.23 | paule32 | thx |
| 20:35.27 | *** join/#asterisk Cubber (~ronny@192.69.4.96) |
| 20:37.11 | *** join/#asterisk Cubber (~ronny@192.69.4.96) |
| 20:40.48 | WIMPy | Yes, pretty screwed. |
| 20:41.05 | WIMPy | Don't try to connect to external IPs on your LAN. |
| 20:42.41 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 20:51.23 | paule32 | cool WIMPy |
| 20:51.27 | paule32 | thank you |
| 20:51.32 | paule32 | for the tip |
| 20:51.51 | WIMPy | That also means no NAT. |
| 20:52.01 | paule32 | now, i can call to the world out of the box |
| 20:52.33 | paule32 | but incoming calls does not work |
| 20:53.56 | WIMPy | You haven't configured an extension, so that's rather obvious. |
| 20:54.40 | [TK]D-Fender | "not work" is vague |
| 20:57.02 | paule32 | i add following extension: |
| 20:57.19 | paule32 | exten => 620,1,Dial(SIP/1001) |
| 20:58.08 | WIMPy | Sounds sensible. |
| 20:58.26 | paule32 | but it comes no call |
| 20:58.40 | WIMPy | Show us |
| 21:00.37 | [TK]D-Fender | paule32: And you are failing to learn the same lesson over and over |
| 21:00.57 | [TK]D-Fender | paule32: You are not looking at THE CALL |
| 21:01.23 | paule32 | http://codepad.org/RZpUO0rm |
| 21:02.07 | WIMPy | No call there. |
| 21:03.57 | paule32 | you mean 's' ? |
| 21:04.10 | Penguin | He means there is no "CALL" in that paste. |
| 21:04.24 | Penguin | All you did was paste dialplan show and some extensions. |
| 21:04.28 | WIMPy | As in nothing happening. |
| 21:04.39 | Penguin | Nothing happening means everything is working OK. |
| 21:04.42 | paule32 | the cli produces no output |
| 21:04.49 | Penguin | core set verbose 3 |
| 21:04.55 | [TK]D-Fender | paule32: "sip set debug on" <- NOW try looking |
| 21:04.56 | WIMPy | Turn up verbose and debug. |
| 21:04.57 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.251) |
| 21:05.00 | [TK]D-Fender | SIP DEBUG |
| 21:05.06 | Penguin | Now it will. |
| 21:05.07 | [TK]D-Fender | because without it, you're jsut kidding yourself |
| 21:05.22 | [TK]D-Fender | Look for the magic packets of your call |
| 21:05.23 | paule32 | haha |
| 21:05.25 | WIMPy | And yes, if you still see nothing turn on sip debug. |
| 21:05.49 | *** join/#asterisk Geek-Linux (~mubbashir@182.185.102.158) |
| 21:10.42 | paule32 | http://codepad.org/QxzU2BmJ |
| 21:10.50 | paule32 | also NAT |
| 21:11.09 | paule32 | this means in sip.conf nat=yes ? |
| 21:11.09 | Penguin | Where's the rest of it? |
| 21:11.22 | [TK]D-Fender | paule32: Which you should probably have set for your peer... also, that is not the ENTIRE call. That is just Asterisk's ANSWER to a previous packet |
| 21:11.24 | Penguin | Yes, it means you have nat=yes set. |
| 21:12.13 | WIMPy | >>That also means no NAT. |
| 21:12.22 | [TK]D-Fender | paule32: Which BTW ... was a REGISTER ATTEMPT, and not a CALL |
| 21:13.10 | paule32 | the console is full with text, i can't copy all of them |
| 21:14.39 | [TK]D-Fender | paule32: use PuTTY SSH from your windows PC and NOT the direct OS console on your VM |
| 21:14.49 | [TK]D-Fender | And set a BIG scroll-back buffer |
| 21:14.57 | WIMPy | Or use tee |
| 21:15.12 | Penguin | Or any of the other various ways available. |
| 21:16.08 | Penguin | Heck, with PuTTY you don't even need to scroll in the terminal. Just log the output to a file. |
| 21:19.49 | paule32 | the call will come to the asterisk server, but on the phone no voice, on the softphone no ring |
| 21:20.05 | paule32 | one is clear the server is respond |
| 21:21.04 | [TK]D-Fender | paule32: We don't see a call yet |
| 21:21.35 | *** join/#asterisk SGjunior (~sgjunior@out-pq-205.wireless.telus.com) |
| 21:21.48 | paule32 | it takes 20 seconds, and the call is dropped |
| 21:22.51 | WIMPy | You can publish that on twitter. If you want to get on, show some facts. |
| 21:23.12 | paule32 | mom please i get startet putty |
| 21:24.50 | *** join/#asterisk Cubber (~ronny@192.69.4.96) |
| 21:29.54 | navaismo | see you |
| 21:33.56 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 21:34.29 | *** join/#asterisk paule32 (~paule32@dslb-094-222-034-237.pools.arcor-ip.net) |
| 21:34.49 | paule32 | http://codepad.org/hpW3m9qP |
| 21:37.28 | [TK]D-Fender | paule32: All we see is your not having set up Zoiper properly and it constantly getting regected while trying to register and subscribe. |
| 21:37.45 | [TK]D-Fender | paule32: Nowhere do we see any kind of call come in. |
| 21:38.09 | [TK]D-Fender | rejected* |
| 21:38.11 | [TK]D-Fender | gah |
| 21:40.01 | WIMPy | Did you remove all references to the external IP? |
| 21:41.26 | paule32 | yes |
| 21:41.34 | WIMPy | Or more general fix your NAT settings. |
| 21:42.58 | WIMPy | You have an externhost set, but no localnets. |
| 21:43.51 | *** join/#asterisk mitchrodrigues (~mitchrodr@71-219-60-97.slkc.qwest.net) |
| 21:45.43 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 21:59.17 | j4jackj | ChannelZ: what do you think of my nfood4u file? |
| 22:24.20 | ChannelZ | huh? |
| 22:36.39 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.251) |
| 22:57.45 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 23:05.21 | *** join/#asterisk j4jackj (~j4jackj@99.199.11.127) |
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