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| 11:00.06 | grummund | Hi, can anyone point me at info on using a USR5637 usb modem (or similar) as a FXO port? |
| 11:00.26 | grummund | did find a link but since lost it... |
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| 14:06.13 | Kobaz | anyone use SIP TLS? |
| 14:06.53 | Kobaz | and, does it work? ie: I've set it up exactly as noted on the asterisk wiki, and calls have two way audio, sound crystal clear with tls-sip/srtp, and then drop after 60 seconds |
| 14:07.19 | Kobaz | i've had this problem consistently from asterisk 11,13,16 using chan_sip or pjsip |
| 14:08.25 | Kobaz | asterisk sends a BYE all of a sudden, with no indication as to why in core debug 9.. just says connection is terminated |
| 14:08.47 | Kobaz | gonna be building some test cases and pasting up some logs sometime today, but... just curious on some best practices, or if someone had similar issues |
| 14:09.15 | seanbright | before you do that - do you want to open an issue on the bug tracker? |
| 14:09.18 | seanbright | /s |
| 14:09.52 | Kobaz | Yeah, I was collecting data for the bug post actually |
| 14:09.54 | file | I know of people successfully using TLS with no such issue |
| 14:10.15 | Kobaz | file: excellent |
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| 14:19.03 | sibiria | can't recall running into any problem with SIPS/SRTP either. part of my Arm-based test monster is slinging some hundred thousand test calls, some of them several minutes long, between two endpoints with sips/srtp and making sure audio comes across OK |
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| 14:47.47 | Kobaz | k, nice. So I shouldn't be having this issue, assuming configuration/networking is good |
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| 20:16.01 | corretico | hello all |
| 20:16.27 | corretico | It's possible to know the secret using by remote extension when try to connect to asterisk? |
| 20:18.09 | Samot | Pardon? |
| 20:21.05 | seanbright | it is not |
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