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| 01:20.21 | *** 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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| 09:04.09 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 09:05.20 | madduck | Having just upgraded Asterisk from 11.13.1 to 13.14.1, is there something obvious I'm missing regarding the handling of NAT? Phones that are behind NATting gateways can't hear anything, Asterisk sends RTP packets to the RFC1918 addresses. |
| 09:05.27 | madduck | This used to work before the upgrade :/ |
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| 09:23.25 | drmessano | madduck: Still using chan_sip? |
| 09:24.05 | madduck | i suppose so⦠i didn't disable it |
| 09:24.31 | drmessano | Thats a doozy of an answer |
| 09:24.33 | drmessano | Let me try again |
| 09:24.45 | drmessano | Are you using chan_sip or chan_pjsip? |
| 09:24.48 | madduck | i am sorry :( |
| 09:25.06 | madduck | i have never heard of chan_pjsip until now, so I guess chan_sip |
| 09:25.25 | drmessano | Ok, well, nothing has changed |
| 09:26.35 | drmessano | externhost/externip and localnet control that behavior |
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| 09:29.26 | madduck | but asterisk is not behind NAT. just a voip phone that I am connecting to asterisk is. |
| 09:31.58 | drmessano | That would have been an important detail |
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| 09:32.10 | drmessano | Anything else you're leaving out? |
| 09:32.22 | madduck | but I wrote that in the original message⦠"Phones that are behind NATting gateways can't hear anything" |
| 09:32.56 | drmessano | I guess english isn't you're native language. Thats fine. |
| 09:33.28 | madduck | scratches head |
| 09:34.49 | drmessano | Do you have nat=force_rport,comedia set on the device in asterisk? |
| 09:34.56 | drmessano | or nat=yes even? |
| 09:35.20 | madduck | i have auto_force_rport and auto_comedia configured globally, and the SIP peers list says that rport/comedia are correctly detected and both set to yes for the device in question |
| 09:38.33 | madduck | here is the SIP debug output, with the final line being the RTP debug output showing that the server is trying to send RTP to a non-public IP: http://scratch.madduck.net/2018-02-11-223754-vit-L9XgHj.txt |
| 09:38.37 | drmessano | Why are you on 13.14.1? |
| 09:38.49 | madduck | drmessano: Debian stable⦠|
| 09:54.29 | madduck | this is just wrong, asterisk: |
| 09:54.30 | madduck | [Feb 11 10:35:42] Peer audio RTP is at port 192.168.15.112:8388 |
| 10:04.31 | madduck | for one thing, direectrtpsetup changed, but unfortunately disabling that doesn't seem to have any effect |
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| 10:37.06 | madduck | ftr, the problem was that netfilter no longer let nf_conntrack_sip do its job, until an explicit CT --helper rule was used instead |
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| 10:49.17 | madduck | nope, that's apparently not all of it. But at least it has now worked once or twice. |
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| 11:44.38 | Amnesia | question, is it correct that a status change, of a sipclient, is broadcasted through AMI? |
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| 12:03.38 | Samot | N. |
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| 12:18.16 | Samot | madduck: Fix the NAT on the router where the phone is. |
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| 19:01.55 | madduck | Samot: am I right in assuming that it's the router's job to rewrite those SDP data in the SIP dialog to replace the RFC1918 address with the public one? |
| 19:07.16 | [TK]D-Fender | no, that's your job to configure * so it knows its public IP |
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| 19:30.39 | Samot | Yes. |
| 19:30.55 | Samot | Since we're talking about a IP phone _behind_ NAT |
| 19:31.00 | Samot | Not Asterisk. |
| 19:38.24 | drmessano | This is still going on? |
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| 20:00.52 | Primer | When I had this problem it turned out to be my G1100 route fron Fronteir having SIP-SLG enabled and not being able to be turned off |
| 20:01.41 | Primer | Since then I've replaced the router, and it seems this one has SIP-ALG turned off by default...all I know is that it just started working again one day, and the only thing that had changed was that I had replaced the router (due to its wifi having stopped working) |
| 20:01.49 | Primer | s/route/router/ |
| 20:02.04 | Primer | heh, wrong line |
| 20:03.36 | Primer | I can tell you right now that I have no "nat" lines at all in my sip.conf, and it's working. All I have set are externhost=voip.mydomain.com (which is updated in route53 with a script running from ifup) |
| 20:03.51 | Primer | and localnet=10.1.1.0/255.255.255.0 |
| 20:04.02 | Primer | And canreinvite=no |
| 20:04.10 | Primer | as the relevant pieces |
| 20:04.41 | Primer | I also have tcpbinddir=10.1.1.1, but I doubt that makes any real difference in it working through NAT |
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| 20:24.00 | igcewieling | try not using tcpbind, it ususally isn't needed. |
| 20:24.49 | igcewieling | ex. tcpbindaddr=0.0.0.0 or leave it out |
| 20:25.56 | drmessano | Primer: Youre confusing "Working defaults" with "Set these because we dont trust the rest of your config" |
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| 20:31.15 | Samot | Also... |
| 20:31.21 | Samot | I've mentioned this already. |
| 20:31.40 | Samot | This the IP Phone behind a router/NAT remote to the PBX |
| 20:31.49 | Samot | THAT router is screwing up NAT. |
| 20:32.04 | Samot | Why the conversation keeps going back t how to configure Asterisk for NAT is beyond me. |
| 20:32.10 | Samot | It's the _wrong_ side. |
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| 20:50.45 | drmessano | Samot: But to be fair, we really donât know |
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| 22:56.08 | Kobaz | hmmm |
| 22:56.31 | Kobaz | [2018-02-11 17:52:07.909] -- <SIP/1001-0000002f> was taken off hold by its bridged peer: SIP/1010-0000002c... i issue a Redirect, and then [2018-02-11 17:52:08.097] WARNING[10418]: channel.c:6670 __ast_channel_masquerade: Can't setup masquerade. One or both channels is dead. (AsyncGoto/SIP/1001-0000002f <-- SIP/1001-0000002f) |
| 22:56.44 | Kobaz | it's random though |
| 22:56.53 | Kobaz | gotta do some more digging |
| 22:56.59 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 22:58.36 | Kobaz | definitely reproducable |
| 23:00.19 | Kobaz | looks like a timing issue, i wonder if it would be easy to add a wait/retry to masquerade |
| 23:03.34 | Kobaz | gonna try it |
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| 23:15.21 | Kobaz | weeeel, retrying the masquerade didn't work |
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| 23:18.05 | wyoung | Kobaz: I had issues with a cisco router once, I had a crontab entry setup to reload asterisk every 10 minutes as a stop gap measure :) |
| 23:22.06 | *** join/#asterisk fbnts (~fbnts@nat.multiplix.co.uk) |
| 23:23.17 | Samot | Kobaz: How is the Redirect setup? What options are you using? |
| 23:23.37 | fbnts | Hi, I'm looking for a way to create a script that will call multiple SIP phones (grandstream), auto-answer, play a sound file and then hangup. Would this be possible? |
| 23:24.16 | Samot | So you want to Page them? |
| 23:24.22 | Samot | You want them all to answer? |
| 23:25.44 | fbnts | Samot: yes. I tried with a call file into spool but I couldn't figure out a way to set the custom SIP header for auto answer. At the moment I have it hacked together with one phone and a speed dial which does a page |
| 23:26.06 | fbnts | But I want to get away from using the handset and set a button that will execute some sort of bash script |
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| 23:26.59 | [TK]D-Fender | So originate the page on e on side and your announcement on the other |
| 23:27.17 | Samot | How many phones? |
| 23:27.27 | fbnts | 4 phones |
| 23:28.40 | Samot | How are you attempted to set this SIP header? |
| 23:28.45 | Samot | attempting* |
| 23:29.19 | fbnts | I currently use SIPAddHeader(Call-Info: answer-after=0) |
| 23:34.38 | fbnts | I think I have solved it, just found that I can add Set:___SIPADDHEADER1 to the call file |
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| 23:47.33 | lwlvl | can I just ring a phone n times without initiating a call? purpose is to indicate the doorbell... |
| 23:47.58 | lwlvl | something like paging? |
| 23:48.06 | lwlvl | phone is a SIP-phone |
| 23:51.35 | [TK]D-Fender | no |
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| 23:59.48 | Kobaz | Samot: https://pastebin.com/Ddw3mYWU |