| 00:09.49 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 00:16.01 | *** join/#asterisk bsdice (~bsdice@meran.embinet.com) |
| 00:16.29 | bsdice | Good evening |
| 00:18.59 | bsdice | !seen mjordan |
| 00:20.07 | bsdice | file Just a heads up, this https://trac.pjsip.org/repos/changeset/4899 and https://trac.pjsip.org/repos/changeset/4900 breaks TLS with Asterisk 11-trunk |
| 00:20.40 | bsdice | i.e. latest CSipSimple trunk which includes this fix somehow mis-parses the Contact: line of the server |
| 00:20.53 | bsdice | it should notice the ;transport=TLS but ... doesn't |
| 00:22.10 | bsdice | I downgraded to an earlier build and all works again. |
| 00:24.03 | newtonr | hmm |
| 00:27.15 | bsdice | on invite server sends this |
| 00:27.17 | bsdice | Contact: <sip:9990@192.168.129.1:5061;transport=TLS> |
| 00:27.27 | bsdice | I think what pjsip expects is this |
| 00:27.32 | bsdice | Contact: <sips:9990@192.168.129.1:5061;transport=TLS> |
| 00:28.30 | bsdice | The packet from CSipSimple will contain: Warning: 381 localhost "SIPS Required" |
| 00:29.14 | newtonr | interesting. Beyond me at the moment. I recommend filing a bug on issues.asterisk.org with an example pcap and log and I'll get a SIP expert to take a look when available. |
| 00:30.26 | bsdice | I'll test some more to not waste your time |
| 00:31.34 | bsdice | I figure only people running in very high secure modes will get bit. ;-) as usual |
| 00:32.17 | newtonr | :D Yeah I've messed with TLS a bit, but not much |
| 00:32.30 | bsdice | whats your job at Asterisk? |
| 00:32.34 | newtonr | I'm an Asterisk generalist and not a SIP expert either |
| 00:33.00 | newtonr | Community Support Manager (aka, do whatever needs to be done so people don't explode) |
| 00:33.07 | bsdice | hahaha |
| 00:33.14 | bsdice | aka shitstorm catcher |
| 00:33.29 | newtonr | i do bug triage on the tracker, and work on documentation and user experience for Asterisk |
| 00:33.51 | newtonr | i share the shitstorms with many others, so it isn't that bad |
| 00:33.54 | newtonr | and our community is pretty awesome |
| 00:34.06 | bsdice | yeah, not many strange/angry people |
| 00:34.29 | bsdice | SIP keeps the nutjobs away, too complicated and convoluted :D |
| 00:34.34 | newtonr | lol |
| 00:35.03 | newtonr | SIP is nuts. I feel like I get to read a new SIP RFC every week |
| 00:35.12 | newtonr | maybe not new, but new to me |
| 00:35.49 | bsdice | I got it running very well, although pretty complicated setup |
| 00:35.54 | newtonr | I forget most of it though since I read something to work on one specific thing for a day and then never touch that part for a few months |
| 00:36.20 | newtonr | bsdice, what do you guys do with Asterisk? |
| 00:36.42 | bsdice | I set one up for friends and to connect to work |
| 00:37.22 | bsdice | two SIP upstream peers (Deutsche Telekom, Sipgate), some Snom 821 phones, CSipSimple on Android clients, MicroSIP windows clients, uhhh |
| 00:37.52 | bsdice | yesterday I patched chan_dongle and put a Huawei E169 to work |
| 00:38.13 | bsdice | mostly encryption trials though |
| 00:38.33 | bsdice | stuff like tlscipher=AES256+kEECDH+aRSA:AES256+kEDH+aRSA |
| 00:38.53 | bsdice | patched in the Bitcoin elliptic curve |
| 00:39.08 | bsdice | oh yeah I use it for phone calls :) |
| 00:39.14 | newtonr | wow, all over the place |
| 00:39.21 | newtonr | That is cool, exactly what Asterisk is for |
| 00:39.25 | newtonr | connect all the things |
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| 00:39.44 | bsdice | works, even fax (via iaxmodem and hylafax) |
| 00:39.49 | newtonr | I have two bitcoin! |
| 00:39.56 | bsdice | keep them :P |
| 00:40.23 | newtonr | I bought them early and thats what I did. I have the keys in offline safekeeping |
| 00:40.48 | newtonr | Don't know what I would do with them otherwise. I don't keep up with that scene much. |
| 00:40.58 | newtonr | It was just a geeky urge |
| 00:41.29 | bsdice | you could have ordered some drugs when Silkroad was still up |
| 00:41.39 | newtonr | LOL |
| 00:42.09 | newtonr | I'm always sick so I get enough drugs from the pharmacy. I prefer some hot tea. |
| 00:42.24 | bsdice | sick as in flu |
| 00:43.21 | newtonr | Usually minor upper respiratory stuff |
| 00:43.37 | newtonr | I'm allergic to everything, combined with a deviated septum and lots of mucus anyway. I'm just always yucky |
| 00:43.37 | bsdice | oh yeah I used to get that |
| 00:43.54 | newtonr | lately my cure is ginger tea with cayenne pepper :D works magic |
| 00:44.20 | newtonr | both are spicy in different ways and it unclogs superbly |
| 00:44.48 | bsdice | I'm a mountain biker and right now I am back in Germany and did not want to stop cycling so cycling at 0°C (32°F) |
| 00:45.11 | newtonr | Alright well the weekend is here. Nice to meet you bsdice - don't forget to find me on Linkedin and be social and what not. |
| 00:45.27 | bsdice | same here, take care |
| 00:45.27 | newtonr | biking is fun!, at that temperature... ouch |
| 00:45.31 | newtonr | later! |
| 00:45.34 | bsdice | later |
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| 01:14.05 | bsdice | I see there was a discussion about this in September 2013 http://lists.digium.com/pipermail/asterisk-dev/2013-September/062531.html |
| 01:14.11 | bsdice | Oh boy my head hurts |
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| 04:37.53 | warewolf | woo, lots of folks in here. |
| 04:38.15 | warewolf | Alright - I'm trying to get sip-tls and srtp working on outgoing sip uri calls |
| 04:39.06 | warewolf | I'm pretty sure the connection between my ip phone and my asterisk box is tps and srtp, but for some reason the outgoing call from asterisk appears /not/ to be tls or srtp |
| 04:40.34 | warewolf | I've tried a mix mash of different variables in the extensions.conf; _SIP_SRTP_SDES=1, _SIPSRTP_CRYPTO=enable, _SIPSRTP=1, CHANNEL(secure_bridge_signaling)=1, CHANNEL(secure_bridge_media)=1) |
| 04:41.12 | warewolf | and my this is my Dial command: Dial(SIP/user@host.com,20,tr) |
| 04:42.26 | warewolf | when I have debugging on SIP, I see "encrypted signaling is required" .. which side of the call is that referring to? ip phon <--> asteirsk, or asterisk <--> dialed URI? |
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| 06:08.38 | qwaz | Hello, I have a few grandstreams connecting to my asterisk server but am having trouble with a mobile phone connecting using app CsipSimple v1.02.03 (newest). I'm getting the following error: https://bpaste.net/show/aff7e9910106 I have set tlsclientmethod=tlsv1,SSLv2,SSLv3 in sip.conf. Any ideas? |
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| 06:09.06 | qwaz | tried tlsclientmethod=ALL and tlsclientmethod=tlsv1 as well |
| 06:25.32 | lvlinux | warewolf and qwaz: I don't have experience with Asterisk and TLS/SRTP yet, but hang around and someone that knows will likely help you during more normal business hours :-) |
| 06:26.15 | qwaz | ok thank. I will idle |
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| 12:57.46 | wonderworld | which STUN server should i install if i want to host my own? is there a "standard" daemon everyone uses? |
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