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| 02:31.05 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0-rc3 (2018/09/20), Standard: 15.6.1 (2018/09/20); 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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| 08:03.22 | jkroon | is anyone aware of a way in which using mod_proxy_wstunnel when proxying connections for /ws on the http module I can get it to (a) report in the auth event to original IP from which apache received the connection, and (b) report that IP in pjsip show aors. I'm assuming if such a mechanism exists it'll be the same method to sort both out. |
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| 08:46.26 | GeneralSpongebob | Hi, I have this in features.conf and expect only the callee to be able to pause recording but the caller can also do it. How can I fix this? pause => 1,peer/callee,GoSub,"pauseMonitor,s,1",default |
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| 10:01.50 | file | jkroon: there is nothing in the websocket transport to support such a thing, and it wouldn't work if the AOR had the IP address behind it. |
| 10:03.33 | jkroon | crap. that sounds like a protocol restriction? |
| 10:03.52 | jkroon | so there is no way, other than to expose the asterisk http port to world? |
| 10:04.08 | file | I have never looked at whether there is anything in a spec to provide such information |
| 10:04.39 | file | if you want access to the websocket yes, otherwise the proxy mechanism would need to forward the traffic as appropriate |
| 10:05.51 | jkroon | as I understand websocket is a http request that then "upgrades", so might it be possible to inject a "X-Original-Source" header from the proxy agent? then it's "just" (this sounds the hard part) a matter of getting asterisk to recognise that. obviously connection-source limited or else world could spoof the source. |
| 10:06.40 | file | PJSIP and everything else has no concept of that, it just knows a packet came from a source |
| 10:07.05 | file | and in the case of a websocket it uses that source to know what connection to use for traffic |
| 10:07.08 | jkroon | ok, so what i'd really need to use is spoof the connection. |
| 10:07.19 | file | it's not easy, and it's uncharted territory |
| 10:07.59 | file | the core websocket API also doesn't provide a mechanism to get headers and such, so that'd have to be extended |
| 10:08.03 | jkroon | cooked some tproxy code for squid before, so thinking using libwebsockets, pick up the connection from apache, and just recreate the connection to asterisk using the original source ip. |
| 10:08.22 | jkroon | then obviously the netfilter rules to fix up the routing. |
| 10:09.07 | jkroon | already have "splice" code that can deal with shoveling the data back and forth (seen benchmarks of 400MB/s+, but that's large packet sizes). |
| 10:11.48 | jkroon | or possibly just hack the apache module ... |
| 10:16.31 | ahmed89 | Hi friends, whats the first asterisk version that support IPv6 ? |
| 10:17.23 | jkroon | ahmed89, all currently supported versions supports it, first use I've had was in 11 but I believe 1.8 was already capable, never tested in 1.6* (was too busy fixing other bugs to notice) |
| 10:27.31 | ahmed89 | jkroon: thanks you , I have 11 |
| 10:30.48 | jkroon | ahmed89, ipv6 for sip? caveat, if you want both ipv4 and ipv6 you have to bind to [::], and you can't bind to specific IP address, eg bind=192.168.x.y and [dead:beaf::1] |
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| 12:08.02 | ahmed89 | I have sip and skinny (cisco) devices |
| 12:09.23 | Samot | Sorry |
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| 15:02.18 | repasp | Hello, how can I make the invalid extension in a context to "do nothing" and continue to work as no key pressed? |
| 15:19.29 | igcewieling | repasp: remove the 'i' extension from the context. |
| 15:20.08 | igcewieling | If you want to provide a better question I can provide a better answer. |
| 15:20.42 | repasp | I want to make a playback unskippable |
| 15:21.09 | repasp | any keypress should be leave out of consideration |
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| 15:30.23 | igcewieling | that was not your question |
| 15:30.41 | igcewieling | If you won't want Playback to be interrupted then use Playback instead of Background. |
| 15:32.42 | repasp | sorry for being equivocal, I'm going to try playback |
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| 23:08.43 | heyjoe | I'm getting a 'chan_sip.c: Failed to authenticate device' <-- where do I start looking for where the problem may be? |
| 23:11.29 | [TK]D-Fender | SIP debug <- |
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| 23:26.19 | heyjoe | where do I start debugging SIP? |
| 23:26.28 | [TK]D-Fender | * CLI |