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| 01:19.37 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 01:35.24 | dl12 | hey guys some of my phones are registered on port 1107 ( sip show peers ) is it normal? |
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| 01:35.30 | dl12 | some of them are on 5060 |
| 01:35.35 | dl12 | chan-sip for all of them |
| 01:36.33 | nafg | Does AGI STREAM FILE accept an http URI? |
| 01:37.57 | nafg | and if so, does it want the extension or how does it determine the format? |
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| 03:01.28 | nafg | Anyone? |
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| 06:21.23 | nafg | Anyone? |
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| 07:41.01 | Samot | Ask |
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| 08:51.58 | ChannelZ | nafg: no |
| 08:55.26 | ChannelZ | As for the extension on files you always leave it off as asterisk will try to play back a file in the native format of the channel to avoid transcoding. |
| 08:56.46 | ChannelZ | So you could have files myprompt.ulaw, myprompt.gsm, myprompt.sln16 etc all in the same directory and you would just request playback of "myprompt" |
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| 13:44.18 | [sr] | howdy |
| 13:44.47 | [sr] | i have one extension that is ACL'ed to ip 192.168.1.26, and somehow it was possible to register from another ip |
| 13:45.22 | [sr] | i have: permit=192.168.1.26 |
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| 13:47.05 | [TK]D-Fender | PB the whole peer |
| 13:49.55 | Samot | Well that's already wrong. |
| 13:50.18 | Samot | Where's the subnet? |
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| 15:00.31 | Harm133 | Hey guys, what is the content of the kvstore table? I´m getting issues with it on a few machines |
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| 15:04.31 | file | kvstore table? do you mean the astdb? |
| 15:05.29 | Harm133 | Hmm its on a freepbx machine |
| 15:05.31 | Harm133 | maybe I should ask there |
| 15:05.48 | Harm133 | but I have an asterisk database, with an kvstore table, which crashes a lot. |
| 15:06.27 | file | you'll need to provide more information such as what you're seeing and how you are determining it is this kvstore table... |
| 15:08.20 | Harm133 | when I tried to open the freepbx webinterface I get the error KVstore has crashed |
| 15:08.29 | file | then that would be FreePBX. |
| 15:08.43 | Harm133 | Kvstore table |
| 15:08.47 | Harm133 | from the asterisk database |
| 15:08.58 | Harm133 | I repaired it with mysqlcheck |
| 15:09.15 | file | yup, all FreePBX stuff |
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| 16:05.23 | [sr] | Samot: well someone told me that just the IP was fine, but if the subnet is required, good then |
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| 16:05.55 | Samot | What is the deny set to? |
| 16:06.55 | [sr] | 0.0.0.0/0.0.0.0 |
| 16:08.13 | Samot | OK. |
| 16:08.21 | Samot | So if you want a single IP, add the subnet |
| 16:08.26 | Samot | Or CIDR notation. |
| 16:08.33 | Samot | Those are part of the setting. |
| 16:09.19 | [sr] | 192.168.1.26/32 |
| 16:09.45 | [sr] | thanks for the clarification, that the mask is required |
| 16:10.12 | [sr] | what is better to use? the netmask or CIDR? |
| 16:10.21 | [sr] | 255.255.255.255 or /32 ? |
| 16:10.26 | Samot | Either is fine. |
| 16:11.10 | [sr] | ok |
| 16:11.24 | [sr] | its still valid, for multiple ip's use the ampersand right? |
| 16:11.35 | [sr] | 192.168.1.26/32&192.168.1.111/32 ? |
| 16:11.48 | [sr] | i can use as many i need separated with the ampersand, correct? |
| 16:12.34 | Samot | https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample |
| 16:13.52 | [sr] | cool |
| 16:13.57 | [sr] | perfect then |
| 16:14.09 | [sr] | Samot: to finish, i can use a dns name instead of ip? |
| 16:14.51 | Samot | I don't know. |
| 16:15.23 | drmessano | Why even do that |
| 16:15.31 | drmessano | Why not just leave it at 0.0.0.0 |
| 16:15.49 | drmessano | The whole point is creating access control |
| 16:15.58 | drmessano | Using a hostname is insecure, even if it was supported |
| 16:17.02 | drmessano | If these are LAN IPs, set DHCP reservations |
| 16:17.13 | drmessano | Then ACL them properly |
| 16:19.17 | [sr] | drmessano: the question for the dnsname is not for LAN extension |
| 16:19.30 | drmessano | Even more of a reason not to do it |
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| 16:19.52 | [sr] | it's a situation where it uses dyndns, and a DNS name like: myremoteext.domain.tld/32 |
| 16:19.57 | [sr] | would this work? |
| 16:20.01 | drmessano | No |
| 16:20.47 | [sr] | quesiton is asterisk would have to refresh the dns name ip for some time, like it does for the external ip setting when a dns name is inserted |
| 16:21.01 | [sr] | ok then, not problematic, just to know if it would work |
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| 16:29.03 | [sr] | thanks drmessano and Samot |
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| 21:19.30 | SupaYoshi | Anyone here that can help writing some code for /etc/asterisk/extensions_custom.conf ? |
| 21:20.00 | SupaYoshi | I need some support. or documentation on how to write the code. |
| 21:27.43 | [TK]D-Fender | ? book |
| 21:27.47 | [TK]D-Fender | ~book |
| 21:27.47 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:31.18 | SupaYoshi | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#asterisk-DP-Basics-SECT-1 |
| 21:31.19 | SupaYoshi | ty |
| 21:39.48 | *** join/#asterisk tpdmoore (4b96ddc7@gateway/web/freenode/ip.75.150.221.199) |
| 21:41.01 | tpdmoore | Hi, my SIP trunking provider says that some calls are being dropped because my asterisk instance is sending a SIP UPDATE right after a SIP INVITE. I have canreinvite=no configured for this peer, but it's not clear to me whether setting that to 'no' is intended to suppress SIP UPDATE or not. My goal is to prevent SIP UPDATE, any hints? |
| 21:46.42 | tpdmoore | it looks like this UPDATE includes the custom SIP header 'X-Asterisk-rpid-update: Yes' so I guess I need to disable sendrpid? |
| 21:48.16 | igcewieling | that would a reasonable thing to try |
| 21:49.05 | igcewieling | If that doesn't help, try disabling session-timers, but I doubt that is the isue. |
| 21:50.13 | tpdmoore | igcewieling: i didn't think of that, brilliant, thanks for the tip |
| 21:50.45 | tpdmoore | i'm going to try rpid first, will not be able to change for some hours yet, unforunately. thanks again |
| 22:01.00 | drmessano | tpdmoore: Why cant you change it now? |
| 22:01.07 | drmessano | Change, reload |
| 22:02.19 | tpdmoore | drmessano: for reasons unrelated to the capabilities of asterisk |
| 22:02.29 | drmessano | lol ok |
| 22:03.10 | *** join/#asterisk mohadib (~jdavis@unaffiliated/mohadib) |
| 22:05.28 | mohadib | Hello, is it possible to get the wrapuptime setting for a queue with AMI? |
| 22:05.59 | tpdmoore | ALSO, I just found rpid_update=no in sip.conf.sample, which i overlooked before, i think that's going to be the ticket |
| 22:06.20 | igcewieling | tpdmoore: Layer 8 issue? |
| 22:06.45 | tpdmoore | on the nose |
| 23:11.03 | *** join/#asterisk infobot (ibot@rikers.org) |
| 23:11.03 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 23:12.47 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 23:27.53 | *** join/#asterisk pruonckk (~pruonckk@135-143-11-177.raimax.com.br) |
| 23:40.44 | *** join/#asterisk jeffspeff (~me@209.141.208.197) |
| 23:43.58 | *** join/#asterisk qxork (~qxork@unaffiliated/qxork) |