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| 00:16.50 | *** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.28.0 (2019/7/25) 16.5.0 (2019/7/25), Security Only: 15.7.3 (2019/7/11); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 14:37.39 | MLC | I have a problem with one-way audio. My phone is behind a NAT firewall and the asterisk server is not. First time I've tried this type of setup. Could that be a NAT issue? |
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| 14:42.42 | Samot | MLC: 99% of the time it is in those setups. |
| 14:44.07 | MLC | I guessed so, but I'm unsure how to correct it. I'm currently trying a PJSIP type=transport section for the phone endpoint. Is that the right way to go? |
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| 14:46.44 | Samot | You need to fix the NAT issue where the phone is. |
| 14:46.53 | Samot | The settings in Asterisk aren't going to fix that. |
| 14:50.03 | MLC | so maybe my firewall is not doing the NAT correctly? |
| 14:50.32 | Samot | Yes, that would be the right track |
| 14:51.55 | MLC | So the phone registers fine, but the problem is the one-way audio. Is there a command in asterisk that will show which IP/port it thinks the media is using so that I can look at those in the firewall? |
| 14:53.09 | file | ensure direct_media=no, rtp_symmetric=yes, and rewrite_contact=yes is set on the endpoint. |
| 14:57.37 | Samot | Turning on the logger will show the SIP messages which will include the SDP |
| 14:57.45 | Samot | Which will have the IP/port being used. |
| 14:57.54 | MLC | @Samot thanks |
| 14:58.01 | MLC | @file - you win!!!! that did it |
| 14:58.04 | MLC | thanks so much |
| 14:58.19 | Samot | You should still fix the NAT issue. |
| 14:58.35 | MLC | agreed |
| 14:59.24 | Samot | If those settings fixed it for now, that means your firewall is sending RFC1918 addresses in the Contact details and probably the SDP. |
| 14:59.46 | Samot | I.e. private non-internet routed IPs. |
| 15:00.24 | MLC | ah, that certainly seems likely |
| 15:01.07 | MLC | I'll have to ask on the #pfsense channel for how to correct that, unless you happen to know |
| 15:01.35 | Samot | My only recommendation for pfsense is to beat it with a bat and get something better. |
| 15:01.54 | MLC | wood batter or aluminum ? |
| 15:02.25 | Samot | Either works better than pfsense. |
| 15:02.33 | MLC | lol |
| 15:12.09 | MLC | In dial plan extension matching, is there something that will match an optional 1 at the beginning of the extension? e.g. a pattern that would match both 2223334444 and 12223334444 |
| 15:12.33 | sibiria | pfsense has a SIP module for doing real-time translations |
| 15:12.41 | sibiria | (if one would ever need to) |
| 15:13.33 | MLC | @sibiria - would that be the siproxd package ? |
| 15:13.51 | sibiria | MLC: yes |
| 15:14.31 | sibiria | it's specifically for this purpose, if your endpoint just can't manage clients behind NAT |
| 15:14.58 | MLC | ok |
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| 21:25.51 | MLC | In dial plan extension matching, is there something that will match an optional 1 at the beginning of the extension? e.g. a pattern that would match both 2223334444 and 12223334444 (sorry for the repeat, I had to leave shortly after asking the first time) |
| 21:27.40 | file | no. |
| 21:27.52 | MLC | bummer, thanks |
| 21:27.58 | file | use a Goto on one. |
| 21:28.45 | MLC | that should work |
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