| 00:05.53 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 00:07.44 | Penguin | It's in the same place as all the other sound files. |
| 00:09.42 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 00:13.25 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 00:14.42 | SupaYoshi | nevermind fixed it all!!! |
| 00:20.17 | *** part/#asterisk nicknam12322 (021d14df@gateway/web/freenode/ip.2.29.20.223) |
| 00:27.05 | *** part/#asterisk navaismo (~Administr@189.191.20.105) |
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| 01:35.05 | asr33 | Good evening folks, what is the best sound to play to deter serial telemarketing calls? busy signal isn't working well. |
| 01:36.11 | zamba | is it possible to get audio input from a sound card directly into a channel= |
| 01:36.27 | zamba | asr33: record the sound of nails against a blackboard |
| 01:36.43 | asr33 | nice |
| 01:36.49 | WIMPy | asr: I just use a cptcha, but you could also give them something to play with. |
| 01:36.55 | WIMPy | zamba: Yes. |
| 01:37.12 | zamba | WIMPy: oh.. sweet.. using the jack application? |
| 01:37.17 | WIMPy | You can use any of the console channels. |
| 01:37.38 | WIMPy | Yes, I think you can even directly use Jack, but I never looked at that. |
| 01:38.09 | zamba | WIMPy: what way have you been using? |
| 01:38.26 | WIMPy | chan_alsa |
| 01:38.41 | zamba | oh |
| 01:39.16 | zamba | any apt repos available for asterisk 11.x? |
| 01:54.07 | asr33 | is going to use Hangup() function |
| 02:15.35 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 02:16.57 | WIMPy | Oh, less than half an hour til the end of time. |
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| 02:29.26 | zopsi | would anyone have any idea how to debug a digium DPMA configuration server that is giving timeout to the digium d70s? |
| 02:44.00 | WIMPy | Broadcast message from maya@earth (Sat Dec 22 03:43 2012): |
| 02:44.05 | WIMPy | The system is going DOWN for system halt in 10 minutes! |
| 02:49.53 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.225) |
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| 03:04.23 | SupaYoshi | Hello |
| 03:04.53 | SupaYoshi | anyone an idea how to setup smtp for voicemails? ive found so far, that it should be in voicemail.conf |
| 03:05.20 | SupaYoshi | i intalled postfix, sendmail doesnt seem to be installed. |
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| 03:29.06 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 03:49.29 | Penguin | wimpy: That was 17 hours ago. |
| 03:50.15 | Penguin | I'm pretty sure the world ended at 04:16 MST. |
| 03:51.14 | WIMPy | Well, looks like the whole thing failed. |
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| 05:02.57 | SeRi | Penguin: I for got how to dial out itads |
| 05:03.12 | SeRi | is it 012*itad |
| 05:03.36 | SeRi | cant remember |
| 05:04.44 | SeRi | ah got it |
| 05:04.46 | SeRi | lol |
| 05:04.51 | SeRi | man is been a while |
| 05:34.38 | SeRi | anybody use freenum isn? |
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| 05:49.22 | *** part/#asterisk rue_mohr (~rue@24-207-103-226.eastlink.ca) |
| 06:15.59 | SeRi | any body uses isn from freenum.org? |
| 06:16.17 | SeRi | people are having issues dialing me |
| 06:16.30 | SeRi | but for the life of me I dont remeber the dialing rules |
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| 10:55.29 | gusto | so |
| 10:55.32 | gusto | here am i again |
| 10:55.37 | gusto | :-D |
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| 11:48.44 | WIMPy | http://i.imgur.com/fXL0h.jpg |
| 11:54.56 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 12:14.28 | [sr] | WIMPy: lol, howdy |
| 12:17.44 | WIMPy | Hi [sr] |
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| 14:40.57 | *** join/#asterisk Ice_Strike (~Ice_Black@94-192-112-241.zone6.bethere.co.uk) |
| 14:45.54 | Ice_Strike | When in im CLI, it show a list of dialplan has been executed... show to show from specfic SIP? |
| 14:46.26 | *** join/#asterisk wonderworld (~w@dsdf-4db5c75d.pool.mediaWays.net) |
| 14:57.09 | [TK]D-Fender | Ice_Strike: No |
| 14:58.32 | Ice_Strike | I meant how to show instead outputting all the sips |
| 14:59.35 | [TK]D-Fender | Ice_Strike: You can't isolate dialplan flow debug |
| 14:59.48 | Ice_Strike | What a shame |
| 15:12.18 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:12.18 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 15:35.26 | *** join/#asterisk the_voice_ (~Ben@109.67.14.81) |
| 15:35.32 | the_voice_ | Hey guys |
| 15:36.01 | the_voice_ | I am pretty new to asterisk, normally I use freeswitch. I just installed asterisk, however, none o fmy sip requests from external sources are being picked up by asterisk |
| 15:36.11 | the_voice_ | I ran wireshark, and they are being received |
| 15:39.24 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 15:40.53 | the_voice_ | I am running on amazon btw |
| 15:46.45 | [TK]D-Fender | the_voice_: show us |
| 15:54.54 | the_voice_ | D-Fender |
| 15:55.03 | the_voice_ | there is nothing to show |
| 15:55.10 | the_voice_ | Asterisk logs show nothing |
| 15:55.18 | the_voice_ | but shark you can see the connection comming in |
| 15:55.24 | the_voice_ | I am guessing it's to do with my config in asterisk |
| 15:55.26 | the_voice_ | but I am not sure |
| 15:55.49 | the_voice_ | freeswitch btw works, so the ports are open(well, this is obvious since I see it in shark) |
| 15:59.52 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.227) |
| 16:04.41 | [TK]D-Fender | Show us * is running, chan_sip is loaded and the port bound. |
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| 16:36.00 | the_voice_ | few minutes D-Fender I am trying a few things first :) |
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| 17:39.43 | SupaYoshi | can i update from 1.8 to 11 without loosing my settings? |
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| 17:53.57 | *** join/#asterisk the_voice_ (~Ben@109.67.14.81) |
| 17:54.02 | the_voice_ | okay I have made some progress :) |
| 17:54.35 | the_voice_ | However, I am trying to get webrtc to work, and I have AVPF set to yes in my sip.conf |
| 17:55.00 | the_voice_ | but I am still getting the error "Received SAVPF profle in audio offer but AVPF is not enabled" |
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| 19:53.34 | dorphalsig | Hello, I'm trying to build debian packages for asterisk-11 |
| 19:53.58 | dorphalsig | however when I try to run dpkg-buildpackage |
| 19:54.23 | dorphalsig | It pukes telling me Makefile:95: *** target pattern contains no `%'. Stop. |
| 19:55.07 | dorphalsig | any ideas why can this be happening? |
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| 20:19.20 | *** join/#asterisk Faheem__ (77492170@gateway/web/freenode/ip.119.73.33.112) |
| 20:19.38 | Faheem__ | Hi |
| 20:19.54 | Faheem__ | I am using Asterisk 11 on Ubuntu 12.04 |
| 20:20.24 | Faheem__ | all works well but no audio while using gtalk chan_motif |
| 20:20.50 | Faheem__ | tried same on asterisk1.8 and that is successful on chan_gtalk |
| 20:20.54 | Faheem__ | help please |
| 20:21.20 | Faheem__ | I want to usne chan_motif for google talk |
| 20:21.31 | Faheem__ | it rings but no audio on both end |
| 20:23.29 | Faheem__ | my asterisk is version 11. it works well but no voice when using chan_motif for gtalk. there is no audio |
| 20:25.47 | *** join/#asterisk smacktalk (~jaspaer@174.34.223.141) |
| 20:28.23 | Faheem__ | help please |
| 20:32.00 | *** join/#asterisk Mikeb_ (~Mikeb@161.185.187.81.in-addr.arpa) |
| 20:34.00 | Mikeb_ | Can someone help me with connecting Asterisk to a RTMP Stream? Im sure it can cant find any docs. |
| 20:34.01 | Faheem__ | my asterisk is version 11. it works well but no voice when using chan_motif for gtalk. there is no audio |
| 20:34.43 | Mikeb_ | Do I need patches for RTMP? I want to connect to a red5 server |
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| 21:10.33 | mathi | hi |
| 21:10.58 | WIMPy | lo |
| 21:14.24 | WIMPy | mathi: What's the news? |
| 21:15.00 | mathi | WIMPy, just found out how epensive these digium cards are:p |
| 21:15.31 | WIMPy | Not if you buy used ;-) |
| 21:17.58 | mathi | WIMPy, it seems that ebfore you could buy analog cards with the number of ports that you needed, 1, 2, ... now it's minimum 4. right? |
| 21:18.33 | [TK]D-Fender | mathi: No. |
| 21:20.18 | the_voice_ | Hi D-Fender |
| 21:20.23 | the_voice_ | so I have made some significant progress |
| 21:20.24 | WIMPy | Analog? Don't know. I wasn't alive in the 19th century. |
| 21:20.47 | the_voice_ | I now have it so I can here sound on my bria - elite soft phone on my iphone |
| 21:21.09 | the_voice_ | but I can't hear sound on the xlite phone on my desktop, or through webrtc |
| 21:21.32 | mathi | :D |
| 21:27.43 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 21:28.47 | SupaYoshi | is a wav sound file 16 bit mono better then a ulaw file? |
| 21:28.53 | SupaYoshi | in sound quality? |
| 21:29.06 | SupaYoshi | my pbx IVR doesn't sound that clear atm |
| 21:29.23 | WIMPy | Yes. |
| 21:29.37 | WIMPy | But with the usual CODESC it won't help. |
| 21:30.00 | [TK]D-Fender | SupaYoshi: What codec is your call using? |
| 21:31.48 | SupaYoshi | ulaw i think :) |
| 21:31.52 | SupaYoshi | oh waitttt |
| 21:31.53 | SupaYoshi | nvm :P |
| 21:32.06 | SupaYoshi | waitt;) i think i know give me 10 minutes :3 its in the trunk settings :P |
| 21:32.16 | SupaYoshi | G... :P something, give me a few sorry >< |
| 21:32.55 | the_voice_ | anyone configured asterisk successfully on ec2 machines? |
| 21:34.52 | mathi | When I dial through my SIP provider an external number, how can I catch the exception where the call doesn't work (e.g. my SIP Provider has some technical issues) ? |
| 21:42.05 | [TK]D-Fender | the_voice_: no different than anything else |
| 21:42.12 | the_voice_ | hmm |
| 21:42.26 | the_voice_ | I still have no sound from my desktop, I think because my x-lite there is behind a firewall itself |
| 21:42.36 | [TK]D-Fender | mathi: ${DIALSTATUS} <- |
| 21:42.37 | the_voice_ | if I connect directly from my cell using 3g I get sound |
| 21:42.55 | mathi | [TK]D-Fender, i'll try that thanks) |
| 21:43.16 | [TK]D-Fender | mathi: Or just dial something else immediately because the only rason you'd hit the next priority is if the call failed anyway |
| 21:43.50 | *** join/#asterisk davlefou (~david@unaffiliated/davlefou) |
| 21:46.28 | SupaYoshi | allow=alaw |
| 21:46.54 | SupaYoshi | =voipbuster trunk =/ |
| 21:49.13 | [TK]D-Fender | SupaYoshi: ulaw and alaw are nearly identicle and transcoding doesn't lose much. going from anything else will certainly be worse. |
| 21:49.24 | [TK]D-Fender | (technically) |
| 21:49.25 | SupaYoshi | ok thanks! |
| 21:49.41 | [TK]D-Fender | You SHOULD be doing your recordings in the codec your calls are in |
| 21:49.52 | SupaYoshi | Yeah, thing is some friend of me recorded a nice IVR voice :) and i putted music behind it, and turned the music down :P |
| 21:50.04 | SupaYoshi | But she spoke so quietly that i had to raise hers, and it was even quiet still |
| 21:50.11 | SupaYoshi | ah i see :P |
| 21:50.35 | SupaYoshi | well i cant record in alaw, i do this: file convert file.wav file.alaw |
| 21:51.39 | SupaYoshi | does anyone here know how to set the smtp settings for voicemail? I installed postfix |
| 21:51.46 | mathi | [TK]D-Fender, you sure? if the person that receives the call hangs up, the caller doesn't continue in the dialplan? |
| 21:52.14 | SupaYoshi | and configged postfix |
| 21:52.24 | SupaYoshi | but i cant find the settings in asterisk for postfix, / freepbx |
| 21:52.46 | [TK]D-Fender | SupaYoshi: ##postfix |
| 21:53.06 | [TK]D-Fender | mathi: If they hung up .... THEN THE CALL WORKED |
| 21:53.16 | [TK]D-Fender | mathiWhen I dial through my SIP provider an external number, how can I catch the exception where the call doesn't work (e.g. my SIP Provider has some technical issues) ? |
| 21:53.52 | [TK]D-Fender | mathi[TK]D-Fender, you sure? if the person that receives the call hangs up, the caller doesn't continue in the dialplan? <- no |
| 21:54.31 | [TK]D-Fender | SupaYoshi: There are no "settings" * calls the sendmail binary (at last the sample configs do) and postfix has its own substitute. |
| 21:54.36 | mathi | [TK]D-Fender, yea but you suggested to Dial something else immediately after, but I was wondering if this won't be called as well as long as the caller doesn't hang up |
| 21:54.48 | mathi | ok |
| 21:54.53 | [TK]D-Fender | mathi: "core show application dial" <- |
| 21:54.59 | mathi | lol ok thanks :) |
| 21:56.15 | SupaYoshi | em, so i dont config anything in asterisk nor freepbx to set it to postfix? |
| 21:56.21 | SupaYoshi | because by default i read sendmail was used? |
| 21:58.26 | [TK]D-Fender | it calls the standard BINARY that postfix provides it's own version of. |
| 21:58.26 | Penguin | its |
| 22:01.25 | SupaYoshi | is CDR on by default? |
| 22:01.35 | SupaYoshi | i can see it in my GUI |
| 22:02.09 | [TK]D-Fender | SupaYoshi: there is no such thing as "default", andd GUI's aren't supportsed here |
| 22:02.18 | SupaYoshi | oops true sorry |
| 22:12.45 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-198-188.red.bezeqint.net) |
| 22:21.40 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 22:35.24 | SupaYoshi | i have the CDR module loaded and installed |
| 22:35.30 | SupaYoshi | though i cannot find info thoruh informatio |
| 22:37.15 | [TK]D-Fender | Where are you looking, and WHICH module? |
| 22:48.10 | SupaYoshi | CDR Reports |
| 22:48.17 | SupaYoshi | I can't let it show my calls somehow :p |
| 22:55.22 | [TK]D-Fender | Again talking in GUI terms |
| 22:56.45 | [TK]D-Fender | Asterissk can log CDR in text files, DB's, etc depending on what you configured. Go prove what was configured and which ASTERISK CDR modules (backends) are actually running, etc. |
| 22:56.52 | [TK]D-Fender | "cdr show". |
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| 23:12.21 | SupaYoshi | cdr show doesnt work :P |
| 23:16.42 | Penguin | "module show like cdr_" |
| 23:18.07 | SupaYoshi | http://pastebin.com/t0ttKZzB |
| 23:18.42 | Penguin | You've got a bunch. |
| 23:19.30 | SupaYoshi | mhm i can see that yeah :3 |
| 23:19.46 | SupaYoshi | though i dont see my calls being stored somewhere yet.. hehe and i cant find some place to config that |
| 23:21.20 | Penguin | Did you configure the cdr config files that are in /etc/asterisk? |
| 23:22.30 | Penguin | cdr.conf, cdr_custom.conf, or cdr_*.conf |
| 23:22.39 | SupaYoshi | no, i haven't cus im in the wrong channel actually, :P since i was looking to do so in freepbx, but from what i understand you cant |
| 23:24.44 | Penguin | The only part of your sentence that I understand is that you are in the wrong channel and won't be configuring the files. |
| 23:25.18 | Penguin | Perhaps you need to visit the right channel and get help on configuring it. |
| 23:25.19 | SupaYoshi | same hehe |
| 23:25.28 | SupaYoshi | i just like your help :p |
| 23:25.44 | SupaYoshi | but so true yes |
| 23:25.44 | Penguin | If you want to configure the files like you're supposed to, I can help. |
| 23:27.14 | SupaYoshi | mhm hold on :) |
| 23:27.49 | Penguin | I don't do GUIs. |
| 23:29.29 | SupaYoshi | ik |
| 23:30.55 | SupaYoshi | yep |
| 23:30.59 | SupaYoshi | lets do it with the files :o |
| 23:31.26 | SupaYoshi | did some research :3 found you couldnt do it in the GUI anyway, and it was beta if you wanted that. so if you might want to help me that would be amazing |
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| 23:34.33 | Penguin | Open up your cdr.conf |
| 23:40.30 | SupaYoshi | done |
| 23:41.01 | Penguin | Is there anything in it already? |
| 23:41.21 | SupaYoshi | ah i seee |
| 23:41.24 | SupaYoshi | yes a example |
| 23:41.38 | SupaYoshi | i could read this and try? |
| 23:41.44 | SupaYoshi | or you can help me? )) |
| 23:41.47 | Penguin | Make sure there is a general section and that you do not define "enable=no" anywhere in it. |
| 23:42.43 | Penguin | Then, you can define the basic csv logging. If that is the sample file, I would have expected it to already be defined, though. |
| 23:43.21 | Penguin | Do you want more than basic csv CDR logging? |
| 23:44.17 | SupaYoshi | i would like to see it in my gui, and be able to search through it |
| 23:44.24 | SupaYoshi | so yes i think so.. |
| 23:44.27 | Penguin | I don't do GUIs. |
| 23:45.06 | Penguin | If you can figure out how the GUI reads CDR, maybe I can help you configure the backend, but I won't do anything more than that. |
| 23:45.28 | SupaYoshi | understood |
| 23:46.37 | Penguin | If I had to guess, I would say that your GUI wants CDR stored in a database. I wouldn't have any idea which database engine, though. |
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