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| 02:35.11 | pabelanger | Q: If ast_find_ourip was passed 0.0.0.0, would you expect to resolve an IPv6 address over IPv4? |
| 02:47.57 | pabelanger | Looking at the code, it seems both would be returned. |
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| 02:56.46 | pabelanger | \o/ Fixed the problem with gtalk and IPv6, but not sure if it is the correct solution. I added a family parameter to ast_find_ourip, allow you to only select, ipv4 or ipv6, or the current default both. |
| 02:57.06 | pabelanger | Since chan_gtalk does not support IPv6, there is no need to return ipv6 results. |
| 02:57.16 | pabelanger | will post the patch to reviewboard in the morning |
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| 07:09.32 | schmidts | good morning |
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| 10:00.30 | Chodorenko | Hello All |
| 10:01.15 | Chodorenko | Where i can give Manual by setup SRTP on asterisk 1.8 ? |
| 11:08.13 | leifmadsen | schmidts: morning! :) |
| 11:11.31 | schmidts | leifmadsen nice afternoon from here ;) |
| 11:12.33 | schmidts | how are you? |
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| 11:20.37 | schmidts | welcome back ;) |
| 11:30.56 | leifmadsen | thanks :) |
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| 11:32.35 | schmidts | btw issue 18135 with bindaddress=:: you want to make it for ipv6 or ipv4? |
| 11:32.37 | MuffinMan | [acknowledged] [Asterisk] Channels/chan_sip/General 0018135: chan_sip loses registration and peer information after dnsmgr refresh reported by lmadsen https://issues.asterisk.org/view.php?id=18135 |
| 11:33.51 | leifmadsen | schmidts: ipv4, it may have something to do with the ipv6 stuff, but all my peers and such are ipv4 based |
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| 11:35.58 | schmidts | ok then why you write :: i thought this is only the placeholder for any ipv6 address |
| 11:41.29 | leifmadsen | schmidts: that just means bind to all addresses, both ipv4 and ipv6 |
| 11:41.37 | leifmadsen | schmidts: I'm just saying, that when I use that, I get the error as I described |
| 11:42.10 | schmidts | it was just for my understanding why this could happens here ;) |
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| 11:48.24 | leifmadsen | schmidts: ya I'm not sure why it happens to be honest ;) |
| 11:49.05 | leifmadsen | that just seems to be the way I'm able to reproduce the issues. Because I stopped using IPv6 for a bit I was no longer having the problem, but by enabling IPv6 support it seems to start doing it (even if you're only using IPv4 end points, etc.) |
| 11:50.51 | Chodorenko | Where i can give Manual by setup SRTP on asterisk 1.8 ? |
| 11:52.38 | schmidts | chodorenko http://www.e164.org/wiki/AsteriskSRTP |
| 11:54.34 | Chodorenko | schmidts: its info for http://svn.digium.com/svn/asterisk/team/oej/securertp-trunk not for native SRTP on 1.8 |
| 11:55.30 | russellb | download a 1.8.0-something tarball, open doc/tex/asterisk.pdf |
| 11:55.35 | russellb | there is a chapter on Secure Calls |
| 11:56.25 | russellb | contribute to the documentation by editing doc/tex/secure-calls.tex |
| 11:59.24 | Chodorenko | russellb: variable forceencrypt=yes and encryption=yes work example use Set(CHANNEL(secure_bridge_media)=1) return |
| 11:59.26 | Chodorenko | <PROTECTED> |
| 11:59.26 | Chodorenko | [Oct 14 13:17:29] WARNING[19104]: func_channel.c:491 func_channel_write_real: Unknown or unavailable item requested: 'secure_signaling' |
| 11:59.26 | Chodorenko | [Oct 14 13:17:29] NOTICE[19104]: chan_sip.c:4052 sip_setoption: Unknown option: 9 |
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| 11:59.57 | russellb | ok, open a bug report then |
| 12:00.48 | Chodorenko | but the connection is still established a secure without of its variable |
| 12:07.19 | Chodorenko | russellb: in doc/tex/secure-calls.tex sysntax error CHANNEL(seucre_media) ->CHANNEL(secure_media) |
| 12:08.05 | russellb | thanks |
| 12:10.32 | russellb | fixed |
| 12:12.28 | schmidts | chodorenko sorry i understood you wrong ;) |
| 12:17.59 | Chodorenko | https://issues.asterisk.org/view.php?id=18140 |
| 12:18.14 | leifmadsen | Qwell: ping? |
| 12:19.44 | Chodorenko | schmidts: my final problem is automatic check support SRTP by phone and use it if Phone support |
| 12:19.47 | leifmadsen | M18140 |
| 12:19.49 | MuffinMan | [new] [Asterisk] Resources/res_srtp 0018140: SRTP enable disable from dialplan reported by chodorenko https://issues.asterisk.org/view.php?id=18140 |
| 12:20.00 | leifmadsen | enable disable? |
| 12:20.28 | Chodorenko | SRTP on or OFF on this channel |
| 12:20.56 | Chodorenko | if phone support SRTP enable , if not - disable |
| 12:22.03 | Chodorenko | but communication should always be. those first tries to call for SRTP if it does not, then warn the user and go without encryption |
| 12:23.45 | Chodorenko | in now moment variables secure* not work how they should |
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| 12:30.05 | schmidts | if i understand you right, you want to enable srtp by yourself from the dialplan or do you only want to check if its enabled? |
| 12:34.27 | Chodorenko | check capability of phone and if phone(peer in config) support and if support enable it , if no - disable , as one of way enable force and call to peer , check ${HANGUPCAUSE} variable and if error tried which out encrypt |
| 12:34.49 | Chodorenko | as in sample by doc/tex/secure-calls.tex |
| 12:35.26 | Chodorenko | away 1 hour at this moment |
| 12:42.16 | schmidts | what does your asterisk say if you add a Noop(secure_signaling: ${CHANNEL(secure_signaling)} secure_media: ${CHANNEL(seucre_media)} to your dialplan |
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| 13:48.16 | Chodorenko | Away off :) |
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| 13:53.40 | Chodorenko | schmidts: -- Executing [8888@default:3] NoOp("SIP/test2-0000004d", "secure_signaling: ") in new stack |
| 13:53.40 | Chodorenko | <PROTECTED> |
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| 14:07.47 | Chodorenko | schmidts: interested if i setup CHANNEL(secure_bridge_media)=0 connect not secured to remote party and in any time -- Executing [8888@default:9] NoOp("SIP/test2-0000005f", "secure_media: 1") in new stack |
| 14:08.54 | Chodorenko | and every time (before and after change) [Oct 14 17:04:58] WARNING[19465]: func_channel.c:374 func_channel_read: Unknown or unavailable item requested: 'secure_brige_media' |
| 14:08.54 | Chodorenko | <PROTECTED> |
| 14:09.36 | Chodorenko | <PROTECTED> |
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| 14:20.29 | schmidts | and whats about secure_signaling? |
| 14:22.27 | schmidts | please add this output to the issue too |
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| 14:42.27 | pabelanger | r973 |
| 14:42.31 | MuffinMan | [R973] Add the ability for ast_find_ourip to return IPV4, IPV6 or both. [Pending] Submitted by pabelanger => https://reviewboard.asterisk.org/r/973/ |
| 14:42.37 | pabelanger | feed back appreciated :) |
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| 14:51.39 | leifmadsen | pabelanger: wow! awesome! this was something I ran into just yesterday! |
| 14:52.17 | pabelanger | leifmadsen: Ya, I'm pretty surprised I managed to figure it out :) |
| 14:53.06 | leifmadsen | pabelanger: you rock |
| 14:53.57 | pabelanger | negative... Google Voice and Asterisk rocks |
| 14:54.08 | schmidts | pabelanger looks great |
| 14:54.32 | Chodorenko | schmidts: https://issues.asterisk.org/view.php?id=18140 Updated |
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| 14:55.12 | leifmadsen | pabelanger: +1 |
| 14:55.28 | leifmadsen | pabelanger: so I got google voice working yesterday -- you have to use a real gmail account and not a google apps account :) |
| 14:56.34 | pabelanger | leifmadsen: Ya, same. When I tried my google apps account, it dialled but prompted a message stating cannot complete call. |
| 14:56.46 | leifmadsen | pabelanger: yep same thing here |
| 14:56.55 | leifmadsen | at first I thought it was an IP restriction until I googled |
| 14:57.20 | pabelanger | I still need to try if we can set CallerID |
| 14:58.57 | schmidts | pabelanger is this review for trunk? if yes i think you missed a function call to ast_find_ourip in __oh323_rtp_create |
| 14:59.20 | schmidts | http://www.asterisk.org/doxygen/trunk/chan__h323_8c-source.html#l00957 |
| 14:59.49 | pabelanger | schmidts: 1.8 branch, would need to port to trunk if accepted |
| 15:01.03 | schmidts | pabelanger same in 1.8 ;) |
| 15:01.14 | pabelanger | schmidts: Ya, just seen that :) |
| 15:01.19 | pabelanger | updating |
| 15:05.31 | schmidts | gtg cya all |
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