| 18:43.32 | *** join/#asterisk infobot (ibot@rikers.org) |
| 18:43.32 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.1.0 (2014/12/15), 11.15.0 (2014/12/15), 1.8.32.1 (2014/11/20); Standard: 12.8.0 (2014/12/15); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 19:18.16 | *** join/#asterisk infobot (ibot@rikers.org) |
| 19:18.16 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.1.0 (2014/12/15), 11.15.0 (2014/12/15), 1.8.32.1 (2014/11/20); Standard: 12.8.0 (2014/12/15); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 19:18.34 | jeffspeff | hey, is anyone else having problems getting to digium.com or asterisk.org? i can't even resolve those domains through my dns, 8.8.8.8 or through a random nslookup tool i found online. |
| 19:18.57 | [TK]D-Fender | They are having issues and are working on it |
| 19:19.21 | jeffspeff | ok, good to know it's not my network with the problem. Thanks [TK]D-Fender |
| 19:21.57 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 19:41.57 | *** join/#asterisk infobot (ibot@rikers.org) |
| 19:41.57 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.1.0 (2014/12/15), 11.15.0 (2014/12/15), 1.8.32.1 (2014/11/20); Standard: 12.8.0 (2014/12/15); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 19:42.01 | asteriskhelp | i tryd with speech-recog.agi |
| 19:42.10 | asteriskhelp | and no succes |
| 19:44.04 | [TK]D-Fender | You said your audio quality was bad. If that's bad, then you're screwed. |
| 19:44.19 | asteriskhelp | its ok the audio quality now i fixed it |
| 19:44.46 | [TK]D-Fender | Because telephone grade audio is already not good for voice recognition. BAD telephone quality sets you up for extremely high failure rates |
| 19:44.54 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-fgmcfpsggnollvbm) |
| 19:45.16 | asteriskhelp | i want to setup it after the asterisk enter the digits . if the ivr says HELLO . to show me if the answer was hello or no |
| 19:45.17 | [TK]D-Fender | As for "no success", you'll have to show us. |
| 19:46.09 | asteriskhelp | can you help ? |
| 19:46.26 | asteriskhelp | im using /var/lib/asterisk/agi-bin/speech-recog.agi |
| 19:46.27 | [TK]D-Fender | you have to show us what you're doing for us to comment on. |
| 19:46.34 | [TK]D-Fender | Saying what you're using means nothing |
| 19:46.42 | asteriskhelp | ok |
| 19:46.43 | [TK]D-Fender | You have to actually show us your failed call, configs, etc |
| 19:46.44 | asteriskhelp | pastebin ? |
| 19:46.51 | [TK]D-Fender | ALWAYS pastebin |
| 19:47.00 | asteriskhelp | ok wait |
| 19:48.12 | *** join/#asterisk moke (~moke@unaffiliated/moke) |
| 19:48.25 | asteriskhelp | http://pastebin.com/6cuTYGAj |
| 19:49.59 | [TK]D-Fender | exten => s,n,GotoIf($["balance" > "0.8"]?success:retry) |
| 19:50.01 | [TK]D-Fender | this is useless |
| 19:50.08 | [TK]D-Fender | you are not comparing a result there |
| 19:50.30 | [TK]D-Fender | you are comparing the wrod "balance" (in quotes) to "0.8" |
| 19:50.34 | [TK]D-Fender | that is NOT a variable |
| 19:50.40 | [TK]D-Fender | ~book |
| 19:50.40 | 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 |
| 19:50.42 | [TK]D-Fender | ^^^^^^^^ |
| 19:51.07 | [TK]D-Fender | word* |
| 19:51.14 | asteriskhelp | so what i have to put there if the IVR say to me WORD balance . to show if its ok or not ? |
| 19:51.49 | [TK]D-Fender | Go rad the AGI's instructions to see what it is meant to give you back |
| 19:52.53 | asteriskhelp | its hard i cant understand it very well . :( |
| 19:52.55 | asteriskhelp | the agi |
| 19:53.24 | *** join/#asterisk DivideBy0x0 (~DivideBy0@mail.cpubob.net) |
| 19:53.36 | asteriskhelp | http://5.199.165.190/speech-recog.agi |
| 19:53.40 | asteriskhelp | this is the agi that im using |
| 19:54.29 | asteriskhelp | and i dont know how to make the exten line to reconize the word balance. |
| 19:54.34 | [TK]D-Fender | So go check the variables it is supposed to return to you |
| 19:56.08 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 19:56.39 | asteriskhelp | im a beginner :( its hard for me to see witch are the varibles . this is the reason that i need help from you :( |
| 19:57.31 | [TK]D-Fender | It has instructions TELLING YOU which ones it returns |
| 19:57.46 | asteriskhelp | # Detect the sound format used # |
| 19:57.53 | [TK]D-Fender | <PROTECTED> |
| 20:03.23 | WanDang | Hi all, I'm setting up an * server. Currently I have two pots phones plugged into a digium card working (they can call each other and the audio goes both ways). I'm trying to add SIP to the mix. I've got an ekiga softphone registered with the server. the softphone can call the pots phones, but nothing can be heard on the pots phones. Audio from the pots phones can be heard in the ekiga softphone. The softphone is on the LAN with the * server. Everyth |
| 20:03.38 | asteriskhelp | exten => 1234,n,Verbose(1,The text you just said is: ${utterance}) |
| 20:06.13 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 20:06.14 | asteriskhelp | still not working :( |
| 20:06.40 | asteriskhelp | please try to help me [TK]D-Fender |
| 20:06.45 | asteriskhelp | im a beginner :( |
| 20:07.12 | *** join/#asterisk newtonr (~newtonr@173-21-90-203.client.mchsi.com) |
| 20:07.13 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 20:07.28 | asteriskhelp | i want when my asterisk hear the word balance to record the sound using monitor |
| 20:07.41 | asteriskhelp | i dont have any ideea how to make it using speech-recog.agi |
| 20:16.04 | ChannelZ-Wk | You need an API key for Google for one; then it looks like you basically call the script which does the record/submission and returns two channel variables, ${utterance} and ${confidence} |
| 20:17.07 | asteriskhelp | and how i make the exten to show me when the ivr that i call says the word "balance" ? |
| 20:18.55 | ChannelZ-Wk | Well you check ${utterance} by whatever means makes sense.. in a GotoIf perhaps |
| 20:19.44 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 20:19.56 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 20:20.13 | ChannelZ-Wk | GotoIf($["${utterance}"="balance"]?somelabel) perhaps |
| 20:21.11 | [TK]D-Fender | WanDang, go verify that you do not have a firewall in the way on your server. |
| 20:21.15 | [TK]D-Fender | "iptables --list" |
| 20:21.17 | [TK]D-Fender | ~pb |
| 20:21.17 | 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. |
| 20:21.18 | [TK]D-Fender | ^^^ |
| 20:22.58 | ChannelZ-Wk | asteriskhelp, but if you just want to see if it's working, the Verbose (or a NoOp) should suffice. But as I said this whole thing is probably failing because Google is denying access to the speech-to-text API |
| 20:23.04 | asteriskhelp | ChannelZ-Wk and after GotoIf what i have to put if the word is ballance to record the call . and if the word is not ballance to hangup ? |
| 20:23.31 | *** join/#asterisk ghostlines (~ghostline@92.109.90.61) |
| 20:25.02 | [TK]D-Fender | asteriskhelp, Do your Gototif to some dialplan that calls those things |
| 20:25.51 | ChannelZ-Wk | Does the Google Speech API even still exist? I don't see it on their developer site |
| 20:26.37 | asteriskhelp | ChannelZ-Wk tell me something else for voice rec . much simple to use . |
| 20:26.57 | ChannelZ-Wk | You probably won't like the price |
| 20:27.20 | asteriskhelp | tell me . |
| 20:28.37 | ChannelZ-Wk | LumenVox |
| 20:28.56 | asteriskhelp | its easy to use ? |
| 20:30.03 | ChannelZ-Wk | ??? |
| 20:30.46 | [TK]D-Fender | So far Google here is very easy to use |
| 20:32.07 | *** join/#asterisk areski (~areski@80.174.128.116.dyn.user.ono.com) |
| 20:32.35 | ChannelZ-Wk | (I don't actually know if the Google speech API is gone or not, the information seems scattered) |
| 20:33.22 | [TK]D-Fender | Why would you think that it is? |
| 20:34.30 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:34.30 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:34.49 | ChannelZ-Wk | Only because I don't see it listed on their primary developer page - https://developers.google.com/products/ |
| 20:35.59 | [TK]D-Fender | http://www.google.com/intl/en/chrome/demos/speech.html |
| 20:36.26 | Katty | PETER PAN |
| 20:36.44 | [TK]D-Fender | Katty, You know why Peter Pan is always flying? |
| 20:36.46 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.50) |
| 20:37.43 | Katty | [TK]D-Fender: red bull? to work off all the calories? |
| 20:38.01 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 20:38.04 | [TK]D-Fender | Katty, Because he Neverlands :) |
| 20:38.20 | Katty | oh aren't you a clever boy. |
| 20:43.40 | WanDang | [TK]D-Fender: I already have verified. http://paste.debian.net/138635/. Normally, the default policy on the client on the INPUT chain would be DROP, but I've opened things up for testing |
| 20:44.57 | ChannelZ-Wk | Is your softphone on the same LAN as the asterisk box? |
| 20:47.05 | ChannelZ-Wk | And is it running directly on the "client computer" in your paste or is it on another machine behind _that_ firewall? |
| 21:01.01 | WanDang | ChannelZ-Wk: yes the softphone pc and the asterisk server are both on the subnet 192.168.192.0/19. The softphone is running directly on client computer from which the second iptables -L was taken |
| 21:09.49 | WanDang | would my sip.conf and extensions.conf be useful? |
| 21:12.00 | [TK]D-Fender | Show us the call itself |
| 21:12.05 | [TK]D-Fender | "sip set debug on" <- |
| 21:12.32 | [TK]D-Fender | ~pb |
| 21:12.32 | 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. |
| 21:12.33 | [TK]D-Fender | ^^^ |
| 21:13.57 | *** join/#asterisk Johnny5_ (d8cff501@gateway/web/freenode/ip.216.207.245.1) |
| 21:21.05 | *** join/#asterisk retentiveboy (~retentive@2001:558:6011:78:157:3142:bcbc:6bc9) |
| 21:22.48 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 21:22.48 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 21:29.22 | DivideBy0x0 | Hi. When using the Channels ARI to originate a call to an internal extension, I specify a context, but it seems to default to "from-internal" which is the context for that extension from my sip.conf. Is there a way to have it use the context I specify? (same is true without outbound calls, but they are "from-pstn") |
| 21:33.30 | ChannelZ-Wk | not sure I even follow |
| 21:34.04 | DivideBy0x0 | sorry. I want to call an extension using the ARI |
| 21:34.37 | DivideBy0x0 | when I do, it sends me to the from-internal context (freepbx), and plays "goodbye" |
| 21:34.51 | DivideBy0x0 | then I can do what I please with the channel using the ari |
| 21:35.52 | DivideBy0x0 | according to the docs, I can specify a context when I originate the call, but it doesn't seem to work |
| 21:38.32 | file | can you pastebin the exact HTTP request you are doing and the console output? |
| 21:38.46 | ChannelZ-Wk | ^^ |
| 21:38.53 | DivideBy0x0 | http://pastebin.com/h9z28xBx that's the fields I post |
| 21:41.06 | file | and the dialplan |
| 21:41.13 | file | (that context) |
| 21:42.08 | DivideBy0x0 | http://pastebin.com/YZXBU7qD is the cli output |
| 21:43.17 | DivideBy0x0 | dialplan: http://pastebin.com/3ueeyPAK |
| 21:43.22 | DivideBy0x0 | thank you both for your help |
| 21:43.28 | file | yes, that's normal |
| 21:43.54 | ChannelZ-Wk | that's the context of SIP/7001 I assume |
| 21:44.03 | file | at that point in time the channel hasn't been answered, it's only upon answer that it is sent to the provided CEP |
| 21:44.04 | DivideBy0x0 | yes, it is, from sip.conf |
| 21:44.41 | DivideBy0x0 | ok. lemme work on that for a minute. |
| 21:48.09 | DivideBy0x0 | even after the channel is answered, it stays with from-internal. Here's more cli output including after I answer: http://pastebin.com/qZAvK6z7 |
| 21:48.50 | file | aha |
| 21:49.01 | file | you can send an outgoing call to either the dialplan OR Stasis application |
| 21:49.06 | file | the Stasis application takes priority |
| 21:49.14 | file | as you have specified BOTH it went to the Stasis application and not dialplan |
| 21:50.01 | DivideBy0x0 | ok great. That helps! where did you find that out? |
| 21:50.25 | *** join/#asterisk CeBe (~CeBe@brsg-d9beb67c.pool.mediaWays.net) |
| 21:50.28 | file | I know it off the top of my head |
| 21:53.32 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 21:53.41 | WanDang | [TK]D-Fender, ChannelZ-Wk: thanks for helping with this. Sorry for the delay. Took a bit of effort to get the output of the call. Here 'tis http://paste.debian.net/138641/ |
| 21:56.20 | [TK]D-Fender | WanDang, So far everything looks normal. I'd trash the firewall entirely and retest |
| 21:56.21 | DivideBy0x0 | file: when I remove the app, it goes to my dial plan (just like you said). Thanks |
| 21:56.45 | file | you're welcome |
| 22:31.09 | WanDang | firewall trashed (paste.debian.net/138647/) and output from the new attempt (paste.debian.net/138646/) |
| 22:40.33 | *** join/#asterisk retentiveboy (~retentive@2001:558:6011:78:157:3142:bcbc:6bc9) |
| 22:50.48 | [TK]D-Fender | Not sure here... |
| 22:50.54 | [TK]D-Fender | Still looks pretty normal. |
| 23:02.46 | WanDang | could it be that my softphone is not configured correctly? |
| 23:08.12 | *** join/#asterisk retentiveboy (~retentive@2001:558:6011:78:157:3142:bcbc:6bc9) |
| 23:14.53 | ChannelZ-Wk | I forget the original query-- have you done an echo test on the softphone<->asterisk ? |
| 23:15.25 | linetrace | i'm experiencing an issue with AMI on 12.x where dynamic queue members are still shown as logged into the queue (in OrderlyQ & MonAst) after they have successfully logged out (and `queue show <Queue>` confirms this) |
| 23:16.04 | linetrace | what's the best way to troubleshoot this so I can compare w/asterisk 11.x & 12.x to narrow down the issue? |
| 23:16.27 | WanDang | ChannelZ-Wk: no, I haven't. This is probably a naive question, but how do an echo test? |
| 23:16.57 | ChannelZ-Wk | I'm guessing it's missing/not getting the AMI events when they logout, though I don't know why |
| 23:17.41 | ChannelZ-Wk | WanDang, make a test extension that does a Playback() of something (like hello-world) and then Echo() |
| 23:18.03 | ChannelZ-Wk | Call it, see if you hear the hello-world, and then blabber into the phone and see if you get your own audio back. |
| 23:19.19 | linetrace | ChannelZ-Wk: agreed, is there a way to view a log of the AMI events? |
| 23:19.26 | WanDang | oops, I googled after I asked the question and I see it is naive. OK, I'll set up test extension and give it a whirl |
| 23:20.33 | linetrace | ChannelZ-Wk: without using the AMI TCP API, that is |
| 23:20.53 | [TK]D-Fender | WanDang, Answer, then Echo. |
| 23:21.27 | WanDang | K, thanks |
| 23:21.42 | ChannelZ-Wk | linetrace I don't really use manager, but I think we discovered the other day that 'manager debug' doesn't actually say a whole hell of a lot. |
| 23:22.32 | ChannelZ-Wk | It's more debug of the actual manager code, not what is going on in manager connections. |
| 23:22.47 | ChannelZ-Wk | You could always tcpdump/wireshark it :) |
| 23:23.07 | linetrace | still helpful info for me to explore, thanks |
| 23:29.31 | linetrace | very true |
| 23:29.36 | ChannelZ-Wk | WanDang: I'm going to guess it's something silly with your softphone audio setup, like it's not got the right audio device selected for the mic, or it's muted (either in software or on your headset/whatever) as you said the softphones could hear the audio from your analogs |
| 23:33.20 | *** join/#asterisk calum_ (~calum_@cpc70817-harg4-2-0-cust754.7-1.cable.virginm.net) |
| 23:47.26 | WanDang | The echo test is inconclusive. If I call the echo extension from an analog phone hooked up to *, the echo test works fine. On the computer the softphone is on, I get what I am saying back out of the speakers whether I am using the softphone or not, so it is hard to tell if what is coming out of the speakers comes directly from the microphone, or if it is being echo. I can confirm, however, that the hello-world I play before executing the echo() is audib |