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| 01:19.06 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0 (2018/10/09), Security Only: 15.6.1 (2018/09/20); 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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| 14:02.19 | jsmith | Morning all... |
| 14:04.01 | sibiria | top o' the mawnin' to ya, guv'nah |
| 14:04.14 | file | moo |
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| 15:49.28 | darkunderlord | All, I keep getting early hangups from fax machines on incoming faxes. I've had this same setup on Asterisk 11 with Digium Fax for Asterisk for years and the last week or two it's been having issues. I can recreate with several fax machines we have. I tried an online test and it came through fine. It appears to be and early hangup by the remote party (even my own faxes for testing) with a code 16 (Span 2: Channel 0/3 |
| 15:49.28 | darkunderlord | got hangup request, cause 16) . Also I get the error "generic_fax_exec: channel 'DAHDI/i2/7657425815-a' FAX session '74' failure, reason: 'fax session timed-out' (TIMEOUT)" |
| 15:49.46 | darkunderlord | Any ideas? I tried increasing fax buffers, etc but seems to do the same thing. |
| 15:50.23 | darkunderlord | This is being done over an ATT Flex Line, handed to me as a PRI. GYI |
| 15:50.25 | darkunderlord | FYI |
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| 18:13.11 | trmg | Using the Asterisk basekd 3-way calling via feature code(s) set in features.conf, does anyone know if it's possible to separate a call once it's bridged. |
| 18:13.18 | trmg | s/basekd/based |
| 18:23.35 | darkunderlord | not me :) |
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| 19:09.52 | darkunderlord | anything? I'm at my wits end trying to figure out why the calls are dropping just before it's done. file do you have any ideas or ways you can point me? |
| 19:14.11 | Samot | So Asterisk is what the fax machines are connect to? Or ATAs to Asterisk? |
| 19:15.54 | darkunderlord | The fax machines aren't even on Asterisk. For my own testing, I'm sending from our old analog lines that have no PBX and then receive the call on our IPFlex line. Other examples would be whatever our customers, use, etc. but in this case it's straight POTS. |
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| 19:16.40 | darkunderlord | I'm just trying to receive a fax like I always have. Literally been working for probably 5 years or more. So odd. |
| 19:20.04 | mjflick | are there any decent guides for configuring sip in aws? i keep hitting retransmission timeouts, but I'm figuring someone else surely has hit this scenario before... |
| 19:24.36 | salviadud | mjflick, I haven't done that, I wonder if you are just having regular firewall trouble |
| 19:30.06 | salviadud | mjflick, have you set your rtp.conf correctly? |
| 19:31.39 | mjflick | probably not, i'm still rather new at this... |
| 19:32.16 | salviadud | ok, lets go through the basics, I'm guessing you have an asterisk server in aws |
| 19:32.28 | salviadud | are you using sip or pjsip? |
| 19:32.28 | mjflick | yep |
| 19:32.42 | mjflick | sip |
| 19:32.56 | salviadud | Well, you are going to need a few ports |
| 19:33.01 | salviadud | 5060 for registration |
| 19:33.21 | salviadud | You could use another one, but that's usually what sip uses. |
| 19:33.35 | salviadud | For your voice packets, which are done in rtp |
| 19:33.54 | salviadud | You will need to open up a range of ports. |
| 19:34.07 | salviadud | I usually do 10,000 - 20,000 |
| 19:34.13 | salviadud | That is configured in rtp.conf |
| 19:34.21 | mjflick | ok, so 5060/udp seems to be working fine for registration (and everything's registered) |
| 19:34.44 | salviadud | Well, you need to look into rtp.conf |
| 19:35.00 | salviadud | That's where the voice happens. |
| 19:35.13 | salviadud | Set your outbound rules accordingly |
| 19:35.22 | mjflick | right now, i just have rtpstart=10000 / rtpend=20000 (guessing that's UDP) |
| 19:35.36 | salviadud | that's the default |
| 19:35.44 | salviadud | Have you setup your firewall rules for that? |
| 19:36.43 | mjflick | yeah, those are clear on inbound |
| 19:37.03 | salviadud | Well, what does sip debug on show? |
| 19:37.32 | Samot | darkunderlord: So you have FAX machines connected to POTS lines and having fax issues? |
| 19:37.55 | salviadud | sorry it is 'sip set debug on' |
| 19:38.15 | salviadud | mjflick, you might want to try it on a single peer |
| 19:42.33 | mjflick | yeah, stepping through that now |
| 19:44.23 | darkunderlord | Samot: yeah, but the issues is receiving faxes on my asterisk server. I WAS able to get a free fax test from the internet fine, but alot of our customers are not able to send us faxes properly, and I can recreate it with more than one of our faxes on POTS lines. |
| 19:45.22 | darkunderlord | it's like the other side sending teh fax hangs up just a hair too early. It starts creating the TIF file, etc but then it hangs up before my side can complete the fax receipt. |
| 19:47.43 | mjflick | salviadud: ok, i tried it for a single peer.... https://pastebin.com/0WsM4WKR (if that helps) |
| 19:48.19 | mjflick | one thing i've noticed is that the From: packet entries seem to be from an internal IP, rather than the public facing one |
| 19:48.27 | mjflick | i'm not sure whether that would play into this |
| 19:51.25 | salviadud | post your sip.conf without compromising passwords please. |
| 19:53.26 | darkunderlord | Is there any way to have fax detect on an IAX trunk? I've not found any way other than SIP or DAHDI |
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| 19:59.05 | mjflick | salviadud: https://pastebin.com/BB8vLARJ |
| 20:03.16 | mjflick | i have no firewall rules in place to the server, so i'm guessing it's probably not firewall related |
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| 20:08.14 | salviadud | mjflick, do you have an rtp.conf file? |
| 20:13.49 | mjflick | salviadud: yeah, [general]\nrtpstart=10000\nrtpend=20000\nrtpchecksums=no\nstrictrtp=no |
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| 21:15.21 | TheGallopingFox | i cannot get a sip client to register over openvpn? |
| 21:15.37 | TheGallopingFox | everything else works through openvpn, strange |
| 21:16.11 | darkunderlord | yeah that is weird. and you're sure you can ping the server, etc? Any errors? |
| 21:17.02 | TheGallopingFox | im using pfsense opevpn |
| 21:17.06 | TheGallopingFox | openvpn |
| 21:28.00 | TheGallopingFox | yes can ping the server fine |
| 21:28.06 | TheGallopingFox | also access other services on the server |
| 21:28.13 | TheGallopingFox | but sip does not register |
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| 21:41.18 | Samot | TheGallopingFox: Does the request hit the server? |
| 21:45.47 | TheGallopingFox | if im on LAN > OPENVPN > ASTERISK it works, if im on WAN > OPENVPN > ASTERISK it doesnt work |
| 21:45.53 | TheGallopingFox | so maybe this is a pfsense NAT problem |
| 21:47.01 | TheGallopingFox | maybe 5060 does strange things with packets |
| 21:48.53 | TheGallopingFox | or maybe Linphone just sucks |
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| 22:03.21 | Samot | So then the answer is no, they do not hit the server. |
| 22:03.40 | Samot | Well pfsense is god awful. Linphone is right there with it. |
| 22:07.15 | sibiria | it's not really a pfsense problem. it's nat/binat problem - yes, you obviously need to redirect and translate traffic from WAN into your VPN |
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| 23:33.08 | ih8wndz | shorewall is nice |
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