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| 07:54.03 | linocisco | hi all |
| 07:54.32 | linocisco | i can ring. but Can't hear each other in the internal network |
| 07:54.32 | r00f | hi |
| 07:54.36 | linocisco | it is elastix |
| 07:54.52 | linocisco | elastix channel is not responsive |
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| 08:23.11 | linocisco | no audio in Elastix |
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| 08:29.50 | linocisco | no audio in Elastix |
| 08:53.53 | linocisco | i can ring. but Can't hear each other in the internal network |
| 09:00.00 | ChannelZ | NAT or network |
| 09:07.26 | *** join/#asterisk infobot (ibot@rikers.org) |
| 09:07.26 | *** 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 |
| 09:13.46 | linocisco | ChannelZ, in internal network, calling from one softphone to another |
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| 12:50.35 | [sr] | its cold! |
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| 16:10.24 | toresbe | Does anyone know if asterisk can do voice recognition of individual digits? |
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| 16:10.39 | toresbe | i'd like to be able to just speak an extension into a phone and have it ring. Is that possible? |
| 16:11.51 | [TK]D-Fender | yes, but Asterisk doesn't do the recognition. |
| 16:11.58 | [TK]D-Fender | There are 3rd party modules for that |
| 16:12.46 | [TK]D-Fender | https://www.google.ca/#q=Asterisk+voice+recognition |
| 16:13.12 | [TK]D-Fender | Sphinx, Googles voice API, othe engines, etc |
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| 19:11.00 | nixhr | hello |
| 19:11.26 | nixhr | i would need some advice on using Monitor command |
| 19:11.57 | nixhr | i'm recording inbound audio stream and everything works ok except one detail |
| 19:12.27 | nixhr | let's say I have following dialplan on the destination asterisk: |
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| 19:13.28 | nixhr | http://pastebin.com/USR3mnUj |
| 19:13.41 | nixhr | the problem is |
| 19:14.07 | nixhr | that the pauses between audio are not in the recorded file |
| 19:15.00 | nixhr | i checked the rtp stream in wireshark, and there are no rtp packets while the Wait(5) command is executed |
| 19:16.10 | nixhr | I would like that the recorded file contains silence when there is no rtp |
| 19:16.49 | nixhr | any hint would be appreciated. thanks. |
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| 20:09.33 | [TK]D-Fender | nixhr, I don't see you using Monitor in there at all. |
| 20:09.50 | [TK]D-Fender | nixhr, You need to provide a complete sample with actual call debug. |
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| 20:10.58 | nixhr | [TK]D-Fender: ok, maybe my description wasn't complete. the dialplan is on the destination box |
| 20:11.05 | nixhr | the recording is being made on the originating box |
| 20:11.16 | nixhr | where i create a call file |
| 20:11.55 | nixhr | and when the call is answered i connect it to following extension: |
| 20:12.00 | [TK]D-Fender | I can see you missing the first wait beacuse you haven't actually forcibly initiated RTP at that point |
| 20:12.08 | [TK]D-Fender | a Playback would do that |
| 20:12.18 | [TK]D-Fender | So instead of Wait, Playback(silence/5) |
| 20:12.27 | [TK]D-Fender | Or anything just to lock in the stream |
| 20:12.46 | nixhr | exten => 2,n,Monitor(wav,/var/tmp/recording_incall,o) |
| 20:13.37 | nixhr | I see what you mean, that would result in silence contained in the recording |
| 20:13.55 | nixhr | but i have no access to the destination box |
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| 20:15.01 | [TK]D-Fender | Wel if they are doing something bad... then they are doing something bad. |
| 20:15.10 | nixhr | actually the destination box would be a softswitch from various providers |
| 20:15.21 | [TK]D-Fender | You just gave me dialplan there before |
| 20:15.29 | [TK]D-Fender | What was that then? |
| 20:15.36 | nixhr | yes, that is my testing box |
| 20:16.53 | [TK]D-Fender | Well I've answered for that sample |
| 20:17.04 | [TK]D-Fender | If you have something real to show then let's see it |
| 20:17.58 | nixhr | ok, brb |
| 20:18.02 | nixhr | thanks |
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| 20:44.33 | Andras888 | Happy Holidays! Trying to make Google Voice work with Asterisk 11.15 I get a Unable to add Google ICE candidates as ICE support not available or no candidates available. Plesae help... |
| 20:45.06 | Andras888 | rtp.conf has icesupport=yes |
| 20:46.09 | Andras888 | chan_motif and res_xmpp enabled |
| 20:46.36 | Andras888 | Cannot make an outgoing call. |
| 20:49.03 | Andras888 | Followed instructions of Malcolm Davenport |
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| 21:40.43 | ChannelZ | Hmmm, this is new.. |
| 21:40.44 | ChannelZ | ERROR[1732]: astobj2.c:126 INTERNAL_OBJ: bad magic number for object 0x7f986c009138. Object is likely destroyed. |
| 21:40.53 | ChannelZ | whatnow? |
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| 22:46.58 | goseeped | Hi guys , i dont want to bother you this day but can make a question ? |
| 22:47.40 | WIMPy | ~ask |
| 22:47.40 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 22:49.25 | ChannelZ | Don't bother me, I'm masturbating! |
| 22:54.36 | goseeped | im having problems webrtc VP8 Passthrough support for the video codec , asterisk 13.1 , ubuntu 14.04 , pjsip from asterisk github repo, librtp from cisco git repo , endpoints FF 30 and last chome too |
| 22:55.06 | goseeped | audio is fine ! |
| 22:55.59 | goseeped | debug is fine , asterisk 13 just not accept vp8 offer from web browser . |
| 22:57.44 | goseeped | media_encryption=dtls , dtls_verify=fingerprint, dtls_cert_file=/etc/asterisk/keys/asterisk.pem , dtls_ca_file=/etc/asterisk/keys/ca.crt , dtls_setup=actpass , use_avpf=yes , ice_support=yes , media_use_received_transport=yes |
| 22:58.37 | goseeped | Has anyone tried video with webrtc and pjsip . In my case voice works well but video does not work even when i allow vp8 codec in pjsip.conf |
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| 23:04.47 | goseeped | seems like no body tried video with webrtc and pjsip :s . |
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| 23:09.50 | apb1963 | Holiday greetings & good tidings to all. |
| 23:12.43 | goseeped | thanks apb1963 , Has anyone tried video with webrtc and pjsip . In my case voice works well but video does not work even when i allow vp8 codec in pjsip.conf ? |
| 23:13.27 | Andras888 | Thanks, apb1963, likewise! |
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| 23:23.58 | nixhr | [TK]D-Fender: let me explain |
| 23:24.49 | nixhr | what i'm trying to achieve is to record incoming audio stream to a file, which I would use to check audio quality |
| 23:25.10 | nixhr | so on originating box I have Monitor command in the dialplan |
| 23:25.41 | nixhr | now, the call is being picked up, the stream starts and is being recorded |
| 23:26.35 | nixhr | but if the rtp stream is for some reason interrupted, the caller would hear silence, but that silence does not end in the wav file generated by Monitor command |
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| 23:27.07 | nixhr | the dialplan i pasted is just a test case for such scenario |
| 23:28.29 | nixhr | finally i end up with a wav file from which i cannot recognize the gap in rtp stream. |
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