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| 10:29.55 | michael_work | i'm back playing with webrtc. Ahve two machines. Ones is debian7 + ast11.11 and other is centos5 + ast 11.11 |
| 10:30.04 | michael_work | self signed certs |
| 10:30.22 | michael_work | same sip.conf and http.conf on both |
| 10:30.56 | michael_work | same extensions with same settings added on both servers |
| 10:31.20 | michael_work | on one it works on other(centos) i still get Rejecting secure audio stream without encryption details: audio 64995 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 |
| 10:31.23 | michael_work | what could i miss? |
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| 13:40.28 | michael_work | file, ping |
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| 14:04.33 | HeN | hey, I'm using asterisk 1.8.23.0 and im trying to install g723 codec but i get an error: WARNING[22712]: loader.c:434 load_dynamic_module: Error loading module 'codec_g723.so': /usr/lib64/asterisk/modules/codec_g723.so: undefined symbol: __ast_verbose |
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| 17:09.02 | P-NuT | Hi guys, How can I force a SIP user to register via TLS only? |
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| 17:16.25 | WIMPy | Configure it as the only transport. |
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| 18:17.29 | TazzNZ | morning all |
| 18:17.52 | file | michael_work, moo? |
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| 21:05.40 | smkelly | le |
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| 21:44.40 | hforjehikiasq6n2 | Question, I want to create a custom announcement so that when users join a conference bridge, instead of it asking for their name, it simply reads off their CID number |
| 21:44.47 | hforjehikiasq6n2 | How do I do that? |
| 21:53.34 | [TK]D-Fender | You originate a call to a local channel passing their original channel name. You then send the oroginal caller into the confbridge without any announcement. that originated channel would wait X seconds, Chanspy to that channel and then read off the CID |
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| 23:05.47 | linuxfool | hello everyone |
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