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| 10:32.26 | catphish | does app_voicemail have options to dramatically simplify it? i am ideally looking for voicemail with one folder, one greeting, and no authentication | 
| 10:33.27 | catphish | i realise i could probably hack out the features i don't want, but wondered if there was much that could be done to customize it, i came up a bit blank looking at the documentation | 
| 10:55.01 | jkroon | I use pre-authentication, so that should be viable at least, but yea, the other simplifications you probably need to look at app_minvm | 
| 10:56.42 | jkroon | I've not managed to make that migration. | 
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| 11:55.20 | pa | Hi, if i want to just let a call ring without picking it up, what command should i execute? | 
| 11:55.24 | pa | NoOp? | 
| 12:03.26 | sibiria | Wait | 
| 12:03.49 | pa | ah thanks | 
| 12:03.54 | sibiria | if you actually want it to keep ringing before eventually answering or rejecting | 
| 12:04.12 | sibiria | if you simply don't want to answer the call you can just Hangup(<some suitable cause code>) | 
| 12:05.31 | sibiria | it may also help if you first signal Ringing, in the cases where you want it to keep ringing before doing something | 
| 12:11.43 | pa | sibiria, i want to keep it ringing indefinitely | 
| 12:13.17 | sibiria | Ringing, then Wait(<large number of seconds>) | 
| 12:13.26 | pa | ah i miss the ringing | 
| 12:13.46 | sibiria | it's not needed, it just helps some equipment/pbxes to know what's really going on | 
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| 13:27.56 | nbjoerg | I have a reproducable segmentation fault in 16.16.0. can someone enlighten me on how to avoid it? | 
| 13:40.52 | sibiria | "yeah, stop doing what you're doing" | 
| 13:41.35 | sibiria | best advice would probably come from developer response after a bug report about it | 
| 13:42.10 | nbjoerg | well, I can reproduce it now and I can provide a stack trace. not sure I can do it in 6h :) | 
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| 17:30.09 | *** topic/#asterisk by file -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.2.1, 16.16.1 (2021/02/18) Final Bugfix: 13.38.2, 17.9.2 (2021/02/18); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 | 
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| 19:11.02 | igcewieling | "When re-negotiating for T.38 if the initial remote                             response was delayed just enough Asterisk would send                           both audio and T.38 in the SDP. If this happened, and                          the remote responded with a declined T.38 stream then                          Asterisk would crash." | 
| 19:11.18 | igcewieling | This is a real Asterisk crash, dropping all calls? | 
| 19:11.37 | file | crash crash | 
| 19:11.40 | file | kaboom | 
| 19:11.41 | file | segfault | 
| 19:11.41 | file | no more | 
| 19:12.04 | igcewieling | thanks. | 
| 19:12.59 | igcewieling | My outgoing t38 fax stopped working sometime after 16.11.1 and wondered if this was related.   Looks like it isn't. | 
| 19:13.21 | file | there's Local channel stuff as well? | 
| 19:13.24 | file | un-security | 
| 19:24.26 | nbjoerg | file? | 
| 19:30.19 | igcewieling | out t38 fax is the only thing I use Local channels for, and that is what broke for me post 16.11.1 | 
| 19:30.53 | igcewieling | I'll have to look into it further once I get our server hardware upgraded. | 
| 19:30.54 | file | there is currently a patch up to fix it | 
| 19:30.54 | nbjoerg | any suggestions on how hunt down pjsip ignoring invites? | 
| 19:31.12 | nbjoerg | wants to migrate away from chan_sip | 
| 19:31.25 | file | I'm aiming for release candidates a week from now so it has time to get in for them, in which case it would ideally be in a release 2 weeks from now | 
| 19:31.30 | igcewieling | I'd say you have a matching problem. | 
| 19:31.59 | file | oh actually that change went in, so it will be in the next release candidates | 
| 19:33.36 | Kobaz | nbjoerg: core set debug will show you what matching is going on | 
| 19:33.54 | Kobaz | i don't remember the level needed, core set debug 5 or 6 or something like that should catch them all | 
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| 19:34.32 | file | PJSIP also requires explicit configuration, so if you have no transport configured... then you ain't gettin' SIP traffic unlike chan_sip which has that kind of stuff defaulting on | 
| 19:35.05 | nbjoerg | it worked with one set of snoms here but not with the wireless | 
| 19:40.04 | nbjoerg | and grrr, snom bug report policies | 
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| 20:05.42 | igcewieling | *sigh*  Apparently someone didn't order licenses for our new VPN server.   FFS if I have to do everything, what are we paying the other staff for? | 
| 20:07.48 | file | I got nothin' | 
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| 20:21.29 | nirs | Good afternoon everybody | 
| 20:35.31 | Kobaz | licenses? vpn? fun | 
| 20:35.38 | Kobaz | why no openvpn or wireguard? | 
| 20:41.23 | igcewieling | Kobaz: SonicWall. | 
| 20:42.11 | igcewieling | NetExtender or something like that sounds like a sextoy. | 
| 20:54.06 | Kobaz | haha | 
| 21:01.25 | Kobaz | you can port forward to a vpn server | 
| 21:08.16 | igcewieling | Hmm? | 
| 21:09.30 | Kobaz | run openvpn or wireguard or whatever on a box inside the office | 
| 21:09.34 | Kobaz | port forward from your router | 
| 21:10.22 | Kobaz | vio-la, no license needed | 
| 21:11.28 | igcewieling | Or we could use the device we already purchased. | 
| 21:11.41 | Kobaz | well yeah and buy extra licenses | 
| 21:11.51 | Kobaz | Just suggesting some non-license solutions | 
| 21:11.56 | Kobaz | (If that's needed) | 
| 21:12.12 | igcewieling | if we ran openvpn or wireguard the server would have a public IP anyway, so no port forwarding needed. | 
| 21:13.14 | Kobaz | that's a possibility too | 
| 21:13.45 | igcewieling | Now that we have all the VPN config removed from the Adtran router we were using as a VPN server, it makes things a lot more simple. | 
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