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| 16:15.04 | igcewieling | "According to the Verizon portal, FTTI is not available on the 1st floor, it is only available in the Basement and 2nd floor." |
| 16:30.06 | Samot | My take is either A) Those are the two spots you can draw from or B) No one on the first floor has service so nothing is built out there. |
| 17:20.46 | igcewieling | FYI, Google TTS supports generating SLN16 audio with a WAV/RIFF header. If the file extension is .wav, Asterisk complain about an unsupported bitrate. If the file extension is .sln16 there is a little blip at the beginning of audio playback. It would be nice if Asterisk detected the 44 byte wav header and remove it when playing back. |
| 17:21.24 | igcewieling | Currently I remove the extra bytes with the script I use to download the audio before saving the data. |
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| 20:04.20 | ghenry | Anyone in the UK looking for a change? I have an opening at SureVoIP - https://www.surevoip.co.uk/about/careers/customer-support-voip-engineer |
| 20:04.26 | ghenry | Thanks |
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| 23:17.46 | catphish | i have another question around asterisk/kamailio setups: if one uses kamailio as a proxy between UAs and aserisk, and doing registrations, how does one usually identify which UA is sending an INVITE once it reaches asterisk? would kamilio rewrite something, or add a header when doing auth? |
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| 23:37.05 | Samot | All I do is just rewrite the domain:port part. |
| 23:38.23 | Samot | INVITE comes in, I do all my checks and at the end the $du is updated to route to the active PBX. |
| 23:38.43 | Samot | And I rewrite the ruri as well. |
| 23:40.09 | Samot | catphish: For the most part, just like Asterisk, Kamailio is a blank slate. It does what it is told to do. |
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| 23:41.24 | catphish | Samot: i think what i'm probably interested in is the from address (ie callerid), so furi and possibly friends like PAID |
| 23:41.51 | Samot | OK so you just do something like: |
| 23:41.56 | catphish | as a simple example, i'd like a UA to be able to dial a number to retrieve voicemail without authentication |
| 23:42.10 | catphish | so asterisk needs to know reliably which authenticated user is dialing |
| 23:42.13 | Samot | $fu = $_s($var(username)@$var(domain):$var(port)); |
| 23:42.20 | Samot | I do that too. |
| 23:42.44 | Samot | Well there's more to that just the user |
| 23:42.51 | catphish | thanks, that looks a lot like i was expecting <3 |
| 23:43.04 | Samot | The INVITE will be authed by Asterisk, but you need to deal with the rest in dialplan. |
| 23:43.18 | catphish | auth'd in what way? |
| 23:43.31 | catphish | i was expecting asterisk not to authenticate anything |
| 23:43.41 | Samot | Well it needs to associate to an endpoint. |
| 23:43.52 | Samot | Be it by user/domain, IP or another header. |
| 23:44.33 | Samot | So on Asterisk I have an endpoint called [yxzendpoint] |
| 23:44.37 | catphish | interesting, i was mistakenly under the impressing it would always associate with a "kamailio" endpoint rather than being able to match per-user |
| 23:44.46 | Samot | If I auth by user/domain, I need the from user to be xyzendpoint |
| 23:45.05 | Samot | Well if you want to do things over a standard endpoint you can. |
| 23:45.25 | catphish | i'm not sure which i want yet, but good to know it's possible both ways |
| 23:45.46 | Samot | I do it per user/endpoint because I can then add/control things in asterisk based on the endpoint. |
| 23:45.50 | catphish | it might be useful to have kamailio set the from header according to its password auth check, then have asterisk trust that |
| 23:46.03 | catphish | yeah, that may be much better |
| 23:46.21 | catphish | i'll look more closely at pjsip's matching options |