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| 04:14.37 | drmessano | freebs: Thats nice |
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| 06:19.43 | wyoung | hey drmessano! |
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| 07:58.58 | FNTM-U3 | hello |
| 08:00.03 | FNTM-U3 | I I have a gentoo server and I want to start a Voip call on it. sending/receive files text chats and video calls voice calls |
| 08:00.08 | FNTM-U3 | Need help. |
| 08:00.15 | *** join/#asterisk ntt (~smuxi@195.60.190.156) |
| 08:00.50 | FNTM-U3 | and just that no need for telephone switches and actual phone numbers |
| 08:00.55 | FNTM-U3 | just SIP calls |
| 08:01.04 | ntt | Hi, I have a freepbx (with asterisk 13.10.0) server and when I try to call non existent numbers I have no audio. My provider told me that it sends an 183 Session In Progress response. So, 183 contains sdp but I have no audio on the phone. Someone can help me please? |
| 08:16.45 | wyoung | FNTM-U3: werd up? |
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| 09:38.11 | Dovid | Us there any way of creating a custom Language in Asterisk? |
| 09:38.11 | *** join/#asterisk miralin1 (~Thunderbi@195.19.212.23) |
| 09:38.46 | Dovid | I have my own files that I want to use instead of the defaults but I want when I tell Asterisk to say play million that it goes to my_language/digits/million |
| 09:40.19 | *** join/#asterisk clopez (~tau@neutrino.es) |
| 09:42.21 | Dovid | nm. figured it out |
| 09:47.40 | ruied | how can I include another file in pjsip.conf like "#include sipAnotherFile.conf " in sip.conf ? |
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| 09:59.34 | FNTM-U3 | wyoung: what? I didn't catch you.! |
| 09:59.58 | wyoung | FNTM-U3: sup? |
| 10:00.28 | FNTM-U3 | I am trying to make a SIP server. wyoung |
| 10:03.04 | *** join/#asterisk mirela666 (~mirkob@78.24.109.13) |
| 10:03.07 | FNTM-U3 | and I need help |
| 10:03.12 | FNTM-U3 | and just that no need for telephone switches and actual phone numbers |
| 10:03.45 | ruied | ah, it's the same, I had an error in the file.... |
| 10:08.06 | *** join/#asterisk samneo (~samneo@123.63.166.181) |
| 10:08.14 | samneo | hi |
| 10:09.16 | samneo | can anyone tell if dahdi pri CRC and BPVIOL |
| 10:09.21 | samneo | keep increasing |
| 10:09.29 | samneo | so calls are not connecting |
| 10:09.30 | samneo | ? |
| 10:28.32 | RudyValencia | Hi, so I have my Asterisk SIP server up, but I want to add a service like Skype has where it records 10 seconds of audio and then replays it, how do I put something like that into a dialplan? |
| 10:29.01 | wyoung | FNTM-U3: asterisk does that for you though |
| 10:29.07 | wyoung | you don';t need to make one |
| 10:29.49 | wyoung | RudyValencia: Record(), and Playback() |
| 10:29.52 | samneo | has anyone faced issue with dahdi pri CRC keep increasing |
| 10:29.53 | samneo | ? |
| 10:29.59 | wyoung | samneo: I don't know what that means |
| 10:30.12 | wyoung | I have had issues with DAHDI which is why I use a linksys SPA now |
| 10:30.32 | wyoung | but I have switched over most of my analouge devices to snom sip phones now |
| 10:31.12 | FNTM-U3 | wyoung: what about the part that relates to PSTN or phone line numbers. I don't have those. (haedware-wise) and I don't need them.?! |
| 10:31.42 | wyoung | FNTM-U3: so don't use them |
| 10:31.57 | wyoung | FNTM-U3: you don't have to use just numbers in SIP |
| 10:32.01 | wyoung | you can use any character |
| 10:32.17 | wyoung | wyoung@sip.myhostname.tld |
| 10:32.26 | wyoung | is a valid SIP number |
| 10:32.40 | wyoung | it accepts a URI |
| 10:35.11 | samneo | i have PRI card with dahdi |
| 10:35.36 | samneo | and it keep on showing |
| 10:35.47 | samneo | CRC values |
| 10:35.58 | samneo | that keep on increasing |
| 10:36.16 | samneo | and i see Alarms being set |
| 10:36.21 | samneo | every minute |
| 10:36.37 | wyoung | oh |
| 10:36.52 | wyoung | have to set a DEBUG option in /etc/dahdi/modules? |
| 10:43.31 | samneo | ok |
| 10:43.36 | samneo | for a module |
| 10:43.38 | samneo | ? |
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| 11:08.47 | *** join/#asterisk idrisavc_ (~idrisavci@195.142.112.66) |
| 11:09.26 | idrisavc_ | Hi |
| 11:10.39 | *** join/#asterisk mirela666 (~mirkob@78.24.109.11) |
| 11:11.25 | idrisavc_ | I'm trying to build a webrtc solution using asterisk 11.23.0, OverSIP and sip.js |
| 11:11.53 | idrisavc_ | and I'm stuck at DTLS verification |
| 11:12.21 | idrisavc_ | I get "res_rtp_asterisk.c:2065 dtls_srtp_setup: Could not set policies when setting up DTLS-SRTP on" error and I couldn't find useful resources on this error |
| 11:12.45 | idrisavc_ | Does anybody has experience on this error ? |
| 11:13.12 | Samot | Why are you using DTLS? |
| 11:13.54 | Samot | There's nothing in the WebRTC wiki about using DTLS. |
| 11:15.05 | idrisavc_ | As far as I know DTLS is mandatory in recent browsers |
| 11:16.02 | Samot | I don't use DTLS and it works fine. |
| 11:16.40 | idrisavc_ | Which client do you use ? Sip.JS ? |
| 11:18.56 | Samot | DTLS is for UDP based traffic, WebRTC connects on TCP/TLS |
| 11:19.17 | file | DTLS-SRTP is mandatory for WebRTC |
| 11:21.22 | Samot | Then the WIki should be updated because it nevers mention DTLS-SRTP. |
| 11:22.00 | file | it's part of the configuration in the wiki? |
| 11:22.29 | Samot | Not the WebRTC part. It just says WebRTC needs SRTP and thus TLS. |
| 11:22.40 | file | what wiki page? |
| 11:22.42 | Samot | Which points to the Secure Calling page. |
| 11:22.55 | Samot | https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support |
| 11:23.28 | file | ah, the olddddddddddddd page |
| 11:23.41 | Samot | Well the only page. |
| 11:23.46 | file | nope |
| 11:23.56 | file | thereis also https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 |
| 11:24.59 | file | I've made a note to update the other one |
| 11:27.18 | Samot | So the SIPML5 says that DTLS-SRTP is required and also points to the Secure Calling page on the wiki to do this. Which does not show any DTLS settings. |
| 11:27.44 | file | the DTLS settings are in the WebRTC wiki page |
| 11:27.56 | file | the secure calling page has instructions for generating certificates, which is needed for DTLS |
| 11:28.14 | file | if you can suggest a clearer way add a comment and newtonr will sort it |
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| 11:42.50 | idrisavc_ | I have several asterisk servers and I want to deploy a proxy like OverSIP or Kamailio to handle webrtc sessions |
| 11:43.04 | idrisavc_ | Does anybody have a similar config ? |
| 11:51.33 | Samot | For WebRTC proxy? No. |
| 11:51.56 | Samot | #kamailio might be the place to ask about Kamailio stuff |
| 12:07.01 | lee | in asterisk 13, I am trying to implement a "press X to pause recording, press Y to resume" with MixMonitor into the same file (can't record credit card info for PCIDSS reasons). the "into the same file" bit seems to be the problem - pressing X (#1) & Y (#3) works, but the recording ends after #1 and never seems to be resumed, despite the console saying it is. how can I debug this further? |
| 12:10.41 | lee | http://pastebin.com/cWMe3WX7 |
| 12:13.36 | *** join/#asterisk [TK]D-Fender (~joe@216-191-106-165.dedicated.allstream.net) |
| 12:20.48 | RudyValencia | I'm trying to set up voicemail in Asterisk 13.10, but it skips recording voicemail greetings, going directly to the next step, why is this happening? |
| 12:20.49 | *** join/#asterisk acidfu_ (~acidfoo@modemcable002.114-70-69.static.videotron.ca) |
| 12:22.33 | [TK]D-Fender | RudyValencia, Show us the complete call and maybe we'll have an idea |
| 12:22.34 | [TK]D-Fender | ~pb |
| 12:22.34 | infobot | i heard 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 12:22.36 | [TK]D-Fender | ^^^ |
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| 12:24.39 | RudyValencia | oh, I see |
| 12:24.52 | RudyValencia | it's the permissions on the spool directory |
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| 12:39.29 | somepoortech | anyone know how I can show digits dialed in meetme for conference bridge numbers? I have verbosity and debug up to 10 and dont see anything |
| 13:00.34 | [TK]D-Fender | If you have verbose and core debug maxxed out then that is all you get. |
| 13:03.45 | [TK]D-Fender | somepoortech, What are you actually trying to accomplish? |
| 13:03.45 | somepoortech | I also have dtmf in console log ... I appear to have something else wrong as a different system shows it |
| 13:04.17 | somepoortech | trying to find out why a conference system isn't working... looks like it's getting nothing on dtmf over sip |
| 13:04.31 | *** join/#asterisk mirela666 (~mirkob@89.184.168.160) |
| 13:04.54 | somepoortech | unfortunately it matches another system that's working fine |
| 13:05.08 | *** join/#asterisk jetlag (~jetlag@c-71-226-222-56.hsd1.nj.comcast.net) |
| 13:05.11 | Samot | What does a debug show? |
| 13:05.29 | Samot | When you up the verbosity on the logs and make the call? |
| 13:05.42 | somepoortech | just call info, pickup announce hangup |
| 13:05.57 | somepoortech | I put sip debuging on for fun too |
| 13:06.00 | Samot | What is your DTMF set to? |
| 13:07.10 | somepoortech | dtmfmode=rfc2833 |
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| 14:34.49 | voipmonk | anyone have something better than https://speech-to-text-demo.mybluemix.net/ for speech to text ? :) |
| 14:41.59 | *** join/#asterisk Tapps (~mosswald@b2b-92-50-68-50.unitymedia.biz) |
| 14:42.15 | Tapps | Hello, someone here with knowledge of snom-phones? |
| 14:44.17 | stefan27 | in IRC it's usually more efficient to ask the question directly, rather than inquiring if someone might know something about the topic |
| 14:44.39 | stefan27 | I have some limited experience of snom 720 phones |
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| 14:45.45 | Tapps | I want my Phones (Snom 870) virtual Buttons Text (Or color) to change, after calling an URL. I can return any XML, since i got full control about the server. |
| 14:46.26 | stefan27 | Can't help you with that issue :) |
| 14:46.34 | Tapps | Damn :( |
| 14:47.06 | stefan27 | but there's 215 others, you might get lucky, but you question is specific, can't you consult manuals? |
| 14:55.12 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
| 15:00.26 | Tapps | Manuals are useless.. i like snom-phones, but theire documentation is bad. |
| 15:12.56 | WIMPy | Show me a phone with better documentation. |
| 15:17.31 | Samot | Better in what way? |
| 15:17.37 | Samot | Presentation? |
| 15:18.19 | *** join/#asterisk ruied (~ruied@a81-84-110-3.cpe.netcabo.pt) |
| 15:18.34 | Samot | The Snom docs are just fine. I went from never have touched an Snom to getting one to do pretty much the stuff I wanted via the Snom Wiki. |
| 15:19.30 | *** join/#asterisk jeffspeff (~Jeff@12.49.160.131) |
| 15:19.46 | WIMPy | Their wiki is not complete, but it's the best documentation I've seen so far. |
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| 15:40.55 | ruied | I have already pjsip migrated in one * box I'm now with some motivation to change... I configure my dialplans with extensions.conf file. Shall I switch to lua or ael? |
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| 16:19.07 | [TK]D-Fender | ruied, AEL is a lie. LUA ... just means more pain for anyone else broughtt in to maintain your system. |
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| 16:21.37 | jrun | on gentoo ncurses has libtinfo built seperately, trying to tell make to add that lib to ldflags at build time i pass ASTLDFLAGS env var to make but i doesn't seem to have any effect. |
| 16:24.22 | jrun | https://gist.github.com/62a905fa1886ca130da3f462817b7e87 |
| 16:24.28 | jrun | that's the ncurses error msg |
| 16:25.08 | jrun | and this is tracker ticket for the but in gentoo; |
| 16:25.10 | jrun | https://bugs.gentoo.org/show_bug.cgi?id=457530 |
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| 16:28.37 | jrun | export LDFLAGS+="-ltinfo" in menuselect/Makefile (_not_ in ./Makefile) will fix it |
| 16:28.47 | jrun | i mean it _does_ fix it here |
| 16:29.29 | BLAH_BLAH_BLAH | I was wondering if anyone knew (if any) of any voice biometric open source software to pair up with asterisk? |
| 16:31.17 | [TK]D-Fender | BLAH_BLAH_BLAH, No more than the last time you asked for this |
| 16:31.27 | [TK]D-Fender | There is no "multiple speaker detection" |
| 16:31.38 | [TK]D-Fender | or "conference call detection" |
| 16:32.06 | BLAH_BLAH_BLAH | i am speaking of voice detection software |
| 16:32.48 | BLAH_BLAH_BLAH | you have your name tied to a pin number, you put your pin number, say your name and it matches your voice to the one on record |
| 16:32.53 | WIMPy | Speaker detection is built in to ConfBridge. |
| 16:33.20 | WIMPy | Well, we don't have that. |
| 16:34.17 | Samot | Sometimes there is no OSS option and you have to bite the bullet and pay for something. |
| 16:34.30 | Samot | Biometric options seem to be one of those. |
| 16:34.43 | BLAH_BLAH_BLAH | that is fine |
| 16:34.53 | WIMPy | Or find the API to the NSA. |
| 16:34.56 | BLAH_BLAH_BLAH | do you happen to know any top software companies for this |
| 16:35.36 | Samot | No, I don't. I've never had a need or request for a biometric option. |
| 16:35.45 | Samot | So I've never looked into one. |
| 16:35.56 | BLAH_BLAH_BLAH | Nuance seems to be pretty good |
| 16:37.00 | Samot | You have probably looked into biometric integration the most in the room right now. |
| 16:37.27 | WIMPy | Possibly ever :-) |
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| 16:39.03 | BLAH_BLAH_BLAH | LOL |
| 16:39.22 | ruied | [TK]D-Fender, Thanks |
| 16:39.59 | BLAH_BLAH_BLAH | I've looked before and tested some but there were lots of false positive to the point where it wasn't worth implementing |
| 16:40.50 | BLAH_BLAH_BLAH | seems that pairing with a pin and testing the voice to that pin will help with that |
| 16:41.21 | BLAH_BLAH_BLAH | so that it doesn't have to test against all audio files, just the one tied to that pin |
| 16:42.25 | WIMPy | Some compnies claim Extremely high success rates after just a few seconds. |
| 16:42.46 | WIMPy | You should look in the intelligence supplies catalogues. |
| 16:42.57 | lee | it's been a few hours and no response, so once again: http://pastebin.com/cWMe3WX7 |
| 16:44.27 | WIMPy | lee: I'm not sure how many would click a links without a short summary of the question. |
| 16:45.30 | lee | the question is in the paste, and it's not short |
| 16:45.44 | jeffspeff | I'm getting this error "Error when verifying peer's chain of certificates: unable to get local issuer certificate, depth=0" on Bria when it tries to verify the cert on my server. is that due to the CA not being trusted on my client or misconfiguration on my server? |
| 16:50.34 | lee | but if it gets a response... |
| 16:50.41 | lee | in asterisk 13, I am trying to implement a "press X to pause recording, press Y to resume" with MixMonitor into the same file (can't record credit card info for PCIDSS reasons). the "into the same file" bit seems to be the problem - pressing X (#1) & Y (#3) works, but the recording ends after #1 and never seems to be resumed, despite the console saying it is. how can I debug this further? |
| 16:52.34 | *** join/#asterisk bhans (~bhans@unaffiliated/bhans) |
| 16:56.07 | Samot | lee: As far as I know, there's no pause in call recording. |
| 16:56.12 | Samot | Like there's no crying in baseball. |
| 16:56.27 | Samot | You are either starting a recording or stopping the recording. |
| 16:56.32 | lee | right |
| 16:56.48 | Samot | So then you would need to combine those two files. |
| 16:57.00 | Samot | Because when they start the recording again, it's a new file. |
| 16:57.04 | lee | why? |
| 16:57.12 | Samot | As far as I know. |
| 16:57.21 | Samot | WIMPy, any corrections on this? |
| 16:57.36 | lee | am I just screwing up the syntax for MixMonitor or what? |
| 16:57.48 | Samot | That's a total possibility. |
| 16:58.51 | WIMPy | No, it is possible to pause recording. At least via AMI. |
| 16:59.49 | jeffspeff | can you not use some dialplan to set the filename to something you can anticipate then use #1 record to file-1 #2 stop recording #1 record to file-2 then set a hangup exten to merge the two files? |
| 17:00.25 | WIMPy | Just append a counter. |
| 17:01.04 | lee | why can't I just append to the original filename? |
| 17:01.51 | WIMPy | You can always add an option to app_mixmonitor to append to an existing file. |
| 17:02.27 | lee | so am I missing something and getting it wrong? |
| 17:02.49 | WIMPy | You've got three ways to do it so far. |
| 17:02.57 | lee | as far as I can tell that is exactly what I am doing |
| 17:03.01 | lee | it just doesn't work |
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| 17:04.01 | WIMPy | Maybe it's just a bug. |
| 17:04.34 | lee | how do I tell? |
| 17:05.08 | Samot | See, I didnt know. |
| 17:05.17 | WIMPy | Try a really simple test case. |
| 17:05.25 | Samot | And now i do. |
| 17:05.32 | WIMPy | If it doesn't work then, it's likely a bug. |
| 17:05.45 | lee | ok, well can you tell me if the syntax looks good? |
| 17:05.51 | WIMPy | Or maybe someone else can confirm your findingd. |
| 17:06.20 | lee | IMO it's more likely that I'm just getting it wrong somehow, but I don't see how |
| 17:06.30 | WIMPy | In a few minutes. |
| 17:17.15 | WIMPy | lee: I don't see you recording before you try to continue. |
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| 17:24.51 | lee | WIMPy: queues.conf has monitor-type=MixMonitor and monitor-format=wav, for the template for that queue |
| 17:26.44 | lee | recording the first part is working fine, and it's ending when I hit #1, as expected |
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| 19:54.10 | marlinc | Is there a current replacement for libmyodbc on Ubuntu? |
| 19:54.13 | marlinc | Ubuntu 16.04* |
| 19:54.16 | marlinc | Or another solution |
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