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| 01:09.37 | *** join/#asterisk Apteryx (~maxim@198-48-204-126.cpe.pppoe.ca) |
| 01:14.45 | Apteryx | Hello! I've setup Asterisk (behind a NAT with port forwarding), and it seems to be working for calling out, but calls between devices on the same LAN fail... My device connect to Asterisk using a dynDNS public IP. I have put some log here: http://pastebin.com/VSmEnR5M. |
| 01:15.50 | Apteryx | The external IP is resolved by sip.apteryx.ca, and each devices connect to Asterisk using that IP. The internal address of my router is 192.168.0.1 |
| 01:15.54 | WIMPy | You (generelly) can't connect to your external IP from your LAN. |
| 01:16.14 | [TK]D-Fender | <PROTECTED> |
| 01:16.22 | [TK]D-Fender | Why are you calling yourself? |
| 01:18.53 | Apteryx | Oups, look like a put the wrong log. I've corrected it: http://pastebin.com/AhTz06uz ( -- Executing [100@internal:1] Dial("SIP/maxim-000000bb", "SIP/yuki") in new stack) |
| 01:19.02 | [TK]D-Fender | <--- SIP read from TCP:192.168.0.1:49471 ---> <- source IP |
| 01:19.13 | [TK]D-Fender | Contact: "Maxim Cournoyer" <sip:maxim@192.168.0.101:49471;transport=TCP;ob> <-- contact points to being on the SAME subet ..... |
| 01:19.51 | Apteryx | It is. Why should it fail? |
| 01:20.22 | [TK]D-Fender | [Jul 4 20:46:49] ERROR[20911]: tcptls.c:462 ast_tcptls_client_start: Unable to connect SIP socket to 192.168.0.100:49191: Connection refused |
| 01:20.29 | [TK]D-Fender | Can't connect to that host via TCP. |
| 01:20.41 | [TK]D-Fender | go TELNET to to it from your server (since it's TCP) |
| 01:20.41 | Nugget | telnet is eeeeeeevil! |
| 01:23.54 | Apteryx | I got connection refused as well (doing telnet 192.168.0.100 49191). |
| 01:24.05 | Apteryx | Thanks for pointing me to this. |
| 01:24.59 | Apteryx | I guess the culprit is probably the SIP client (Bria on an iPad) rather than my DIR-815 router? (given that these devices are on the same LAN) |
| 01:25.35 | [TK]D-Fender | "sip show peer yuki" |
| 01:26.01 | [TK]D-Fender | is it registered? "addr->to". pastebin it up |
| 01:26.12 | [TK]D-Fender | has to head out, hopefully someone else will be able to pick this up. |
| 01:27.10 | Apteryx | sip peer show yuki output: http://pastebin.com/nuC1j3Mb |
| 01:27.32 | Apteryx | [TK]D-Fender: thanks for your time & help :) |
| 01:29.03 | Apteryx | WIMPy: I remember I had problems with this before, too. My older router used to have a "NAT loopback" setting that allowed this. |
| 01:29.35 | Apteryx | WIMPy: But I couldn't find that setting with my current router. |
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| 03:16.43 | WIMPy | Apteryx: Do you want to use the external name so you can use the client both on the LAN and from the Internet? |
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| 04:33.13 | Apteryx | WIMPy: yes :) |
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| 09:20.06 | gusto | WIMPy: https://lh5.googleusercontent.com/-l1bewcR-GZM/U7d15Fd_-lI/AAAAAAAANOQ/7yAg_dU1Awo/w593-h563-no/IMG_8781.PNG |
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| 10:06.29 | yottanami | Here is parts of my SIP http://dpaste.com/2TR1VYX I dialed 900 from any other local number it does not ring ( someting like busy tone after some minutes) and here is my console output http://dpaste.com/3S389BK It was working some days ago |
| 10:09.38 | yottanami | I can dial from 900 to other locals |
| 10:10.08 | yottanami | but nobody can not dial 900 from ANYWHERE |
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| 10:36.23 | yottanami | How should I fix this ? |
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| 11:35.47 | WIMPy | gusto: Not realistic. It's more like Alt-SysRq-{S,U,B} |
| 11:43.21 | WIMPy | Hmm. There doesn;t seem to be any AMI event upon a finally failed outbound registration, only for temporary ones. |
| 11:43.45 | WIMPy | But I would swear I have seen thme before. |
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| 12:12.29 | yottanami | How should I find out why I can not dial 900 from anywhere ? http://dpaste.com/3S389BK http://dpaste.com/2TR1VYX |
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| 20:08.19 | Kattyroo | hi lads. |
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| 20:42.06 | FreezingCold | Somewhat stupid general question, does forwarding a call make a line busy? |
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| 21:11.19 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.10.2 (2014/06/13), 1.8.28.2 (2014/06/13); Standard: Asterisk 12.3.2 (2014/06/13); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 21:21.26 | Apteryx | Hello, I discovered I needed to activate STUN to get audio both side when receiving calls or calling to outside my NAT. However, I'm confused why -- I thought Asterisk was taking care of this, especially since I gave it all the information regarding my localnet, externhost, and nat=yes on my devices? |
| 21:22.14 | Apteryx | (STUN was activated on my VoIP client, Bria) |
| 21:42.37 | Apteryx | FreezingCold: Have you tried it? I would guess that no, once the call has been properly transfered. |
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| 21:50.29 | [TK]D-Fender | Shouldn't need STUN. To explain what's happening we'd have to have had your actual configs and proper debug from calls. |
| 21:50.53 | [TK]D-Fender | FreezingCold: What line? |
| 21:59.42 | Apteryx | [TK]D-Fender: Hi again :) |
| 22:14.47 | Apteryx | [TK]D-Fender: Here's my sip.conf: http://pastebin.com/3iVLYyJ1, and my extensions.conf: http://pastebin.com/k1Pdgug8 |
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| 22:16.09 | syadnom | hi all. anyone here hosting asterisk (w/ freepbx) on a VPS? I need some advice on a good provider at a good price. I use ramnode for webhosting, but that's not a latency sensative service |
| 22:21.30 | Apteryx | [TK]D-Fender: The peers: http://pastebin.com/ABLTmXmX |
| 22:26.56 | Apteryx | [TK]D-Fender: And finally, a sip debug log showing the conversation happening, after calling my FreePhoneLine DID using Gmail calls: http://pastebin.com/nMNCjsq7 |
| 22:28.09 | Apteryx | The caller (gmail) audio could not go through, but the called device could send audio back. |
| 22:28.31 | Apteryx | If I enable STUN, problem is solved. |
| 22:30.24 | [TK]D-Fender | X-asterisk-Info: SIP re-invite (External RTP bridge) <---------- |
| 22:30.30 | [TK]D-Fender | You allowed re-invites. |
| 22:30.38 | [TK]D-Fender | This BAD |
| 22:30.56 | [TK]D-Fender | ALL of your peers & general should have "directmedia=no" |
| 22:31.40 | Apteryx | [TK]D-Fender: The line reallowing invites is this: insecure=invite? |
| 22:35.32 | Apteryx | [TK]D-Fender: Thanks for looking into it. I will try to use directmedia=no everywhere and see what it gives. But if I want to use it, STUN becomes a requirement? |
| 22:35.36 | [TK]D-Fender | No, it's you lack of having set the parameterr at all |
| 22:35.55 | [TK]D-Fender | STUN is not magic |
| 22:40.29 | Apteryx | Ok, thanks. What makes the re-invites bad? Is it a security problem? |
| 22:42.04 | [TK]D-Fender | it tells the endpoints to directly connect to one another |
| 22:42.22 | [TK]D-Fender | You're running a double NAT which makes that a suicide attempt |
| 22:42.32 | [TK]D-Fender | because you have those ports forwarded to your server |
| 22:43.04 | [TK]D-Fender | Any NAT'd environment should disable reinvites across an outside border interface |
| 22:47.26 | Apteryx | Ok :) Thanks for the explanation. |
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