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04:25.00 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.34.0 (2020/06/11) 16.11.1 (2020/06/16) Standard: 17.5.1 (2020/06/16); 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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12:45.10 | Zombie | Does anyone here use Asterisk with Hylafax? |
12:45.29 | Zombie | I want to do some troubleshooting of interfacing the two/ |
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14:12.27 | igcewieling | Nah, ReceiveFax/SendFax is enough for me. |
14:12.30 | drmessano | From the looks of it, no one has used Hylafax in years |
14:21.09 | Zombie | I have a Government Agency I communicate with in that fashion. |
14:21.32 | Zombie | (By Faxing things.) |
14:22.17 | drmessano | We know |
14:22.40 | drmessano | Because Coronavirus eats emails |
14:27.47 | Zombie | ... |
14:28.15 | Zombie | They don't allow reporting documentation to be transmitted by that means, |
14:30.36 | igcewieling | Still doesn't require Hylafax |
14:48.52 | Zombie | I would like to move ahead with this anyway. |
14:49.13 | igcewieling | *shrug* you are on your own then. |
14:51.26 | drmessano | Youre bolting on a dead solution |
14:52.04 | Zombie | The part that has me messed up is iaxmodem really. |
14:52.13 | Zombie | Can we focus on that? |
14:52.18 | drmessano | No, not really |
14:52.23 | Zombie | because that has multiple uses. |
14:52.35 | drmessano | The whole thing is a black hole |
14:52.44 | Zombie | I have some legacy hardware that uses Serial. |
14:53.52 | igcewieling | I've not used IAX since...hmm...2004 |
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15:29.38 | shimshon | I need help with FastAGI |
15:35.17 | shimshon | I use FastAGI to run bash script, how can i use the variable that FastAgi send to script? |
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19:53.09 | rhollan | Are there any tips on getting Asterisk working behind NAT? I have asterisk behind NAT and local phones can connect just fine (also behind NAT), but remote phones have to go through NAT to VPN back to the server. |
19:54.23 | rhollan | Basically, I have an IPSEC tunnel exposing my local network (with various parts behind a firewall) on an server with a public IP address. local services are exposed to the world there with appropriate IPtables NAT and forwarding rules. |
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21:06.33 | rhollan | Are there any tips on getting Asterisk working behind NAT? I have asterisk behind NAT and local phones can connect just fine (also behind NAT), but remote phones have to go through NAT to VPN back to the server. |
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22:56.21 | rhollan | Sigh. Getting weird Via headers on an Android client. |
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23:19.22 | Samot | What weird headers? |
23:19.32 | rhollan | Another client works better: can make calls. But NAT is still messing with RTP traffic, not exexpectedly. |
23:19.36 | rhollan | unexpectedly |
23:19.48 | rhollan | Was getting an odd 10.8.x.y Via header. |
23:19.56 | rhollan | I tried a different client and calls completed. |
23:20.01 | Samot | What is that odd? |
23:20.04 | Samot | Why* |
23:20.26 | rhollan | Why should my SIP client go th rough a proxy with a non-routable IP address? |
23:20.36 | rhollan | A different client works better |
23:20.38 | Samot | Via is the path. |
23:20.54 | Samot | All sip messages have Via headers |
23:20.55 | rhollan | Yes, but 10.x.x.x is not publicly routable |
23:20.57 | Samot | Sometimes multiple. |
23:21.01 | Samot | That doesn't matter. |
23:21.04 | Samot | It's the _path_ |
23:21.11 | Samot | It came via this via that to here. |
23:21.24 | rhollan | R ight, b ut I can;'t respond to a non-routable IP address |
23:21.38 | rhollan | In any case, a different client doesn;'t add them and it works. |
23:21.42 | Samot | In what regards? |
23:21.49 | rhollan | Calls complete. |
23:21.52 | Samot | It adds via headers |
23:21.57 | rhollan | But no audio. |
23:22.12 | Samot | This has nothing to do with the via headers. |
23:22.50 | Samot | This has to do with the client and/or router not using public routed IPs in the Contact and/or SDP body. |
23:23.10 | Samot | So if the system is told to send audio to 10.8.x.x over the Internet, then yes you won't get audio because it's not routable. |
23:23.17 | rhollan | I know that, but I was not getting even call competion before. But i still do see Via headers. and this client works, so I suppose they do not refer to publicly addressible proxies |
23:23.35 | rhollan | Right, b ut I still lsee Via headers with 10.8.x.y |
23:23.44 | Samot | Without actually seeing it, I can't say but you're on the wrong road with that thinking. |
23:23.55 | rhollan | And calls now complete. Now sure WHAT those Via headers do if they are not publically routable. |
23:24.03 | rhollan | Not sure what... |
23:24.05 | Samot | The hold the PATH of the message. |
23:24.21 | Samot | What systems the SIP message came through including where it originated. |
23:24.25 | rhollan | But to what end if not publicly routable? |
23:24.37 | Samot | The audio doesn't use the via headers |
23:24.47 | Samot | The audio is based on the SDP body data |
23:24.57 | rhollan | Granted, that is a separate issue, and the only one that remains. I thing NAT is messing that up. |
23:25.07 | Samot | What NAT thing? |
23:26.12 | rhollan | Here's the setup: Asterisk inside a LAN, with local SIP phones. External SIP phones connect to a public server which NATs the connection back to the LAN via IPSEC tunnel |
23:26.20 | rhollan | Network Address Translation. |
23:26.31 | Samot | I know what NAT is rhollan. |
23:26.41 | Samot | What kind of tunnel? |
23:26.50 | Samot | Pure IPSEC or GRE/IPSEC? |
23:27.03 | Samot | A VPN tunnel by passes the NAT |
23:27.09 | Samot | Because it puts you on the _local network_ |
23:27.20 | rhollan | Yes, but I have a public IP address on the other side |
23:27.35 | Samot | What SIP driver are you using? |
23:27.36 | rhollan | pure IPSEC] |
23:27.41 | Samot | So you have proper routes? |
23:27.53 | rhollan | Where? In the androud client? On the server I am running (duh) Asterisk. |
23:28.02 | Samot | On Asterisk. |
23:28.07 | rhollan | routing appears fine, though I am checking audio |
23:28.08 | Samot | Which SIP driver are you using? |
23:28.15 | rhollan | Oh! Hang on |
23:29.59 | rhollan | Wher is that specified again? In extensions.conf I have SIP/100 (local) and SIP/101 (remote) |
23:30.13 | Samot | Then that is Chan_SIP |
23:30.32 | Samot | Same with provider setup? |
23:30.36 | rhollan | If you say so. I am onbiously an Asterisk Noob. |
23:30.44 | rhollan | There is no prov ider. I AM the provider. |
23:30.51 | rhollan | (Well, to me) |
23:31.04 | Samot | So this is just local PBX calling |
23:31.05 | rhollan | Haven't set up a remote trunk for PSTN connections. |
23:31.09 | rhollan | Yes, at this point. |
23:31.22 | Samot | So show your sip.conf |
23:31.50 | Samot | Pastebin it. |
23:32.21 | rhollan | The idea is to have an anonymous cell phone (bought with cash, and air service paid with cash-paid debit card), use a VPN for SIP connections back to a server with a blutooth connection to a cell phone for in/out calls. (or a SIP provider which might be easier). |
23:33.19 | Samot | Show your sip.conf |
23:34.13 | rhollan | [100] |
23:34.14 | rhollan | type=friend |
23:34.14 | rhollan | secret=14Bddoqc% ;NOTE it is important to set up a complex password |
23:34.14 | rhollan | qualify=no ; Will not drop the connection |
23:34.14 | rhollan | nat=no ; This phone wil not be outside the network |
23:34.14 | rhollan | host=dynamic ; This device registers with us |
23:34.16 | rhollan | canreinvite=no ; Asterisk by default tries to redirect |
23:34.18 | rhollan | context=default ; Or whatever context you want to define in Asterisk |
23:34.20 | rhollan | ;;mailbox=103@default ; only if you are configuring voicemail. |
23:34.22 | rhollan | [101] |
23:34.24 | rhollan | externip=72.5.54.188 |
23:34.26 | rhollan | localnet=10.0.0.0/255.0.0.0 |
23:34.28 | rhollan | type=friend |
23:34.30 | rhollan | secret=14Bddoqc% ;NOTE it is important to set up a complex password |
23:34.32 | rhollan | qualify=no ; Will not drop the connection |
23:34.34 | rhollan | nat=yes ; This phone may be outside the network |
23:34.38 | rhollan | host=dynamic ; This device registers with us |
23:34.40 | rhollan | canreinvite=no ; Asterisk by default tries to redirect |
23:34.42 | rhollan | context=default ; Or whatever context you want to define in Asterisk |
23:34.44 | rhollan | ;mailbox=103@default ; only if you are configuring voicemail. |
23:34.46 | rhollan | yeah, will change the passwords. Didn't vet it. |
23:35.33 | Samot | I literally said pastebin it. |
23:35.58 | rhollan | Sorry, missed it. Only two extensions, so I ffigured meh. |
23:36.22 | Samot | externip=72.5.54.188 |
23:36.39 | Samot | localnet=10.0.0.0/255.0.0.0 |
23:36.44 | Samot | Those should be in the general section |
23:36.57 | Samot | I highly doubt you need a full class A for this but OK. |
23:37.01 | rhollan | That's the public facing side and private internal net. Yes, but only one extension is outside the lan |
23:37.18 | Samot | That needs to be in the general section. |
23:37.33 | Samot | You have nothing to tell Chan_SIP that the PBX itself is behind NAT. |
23:37.43 | rhollan | 10,<net class>.<room>.<devcice>. |
23:37.52 | Samot | externip and such should not be in the peer sections. |
23:37.56 | rhollan | But, how do I distinguish extensions inside vs outside? |
23:37.58 | Samot | It should be in the general section. |
23:38.05 | Samot | You tell it the NAT |
23:38.16 | Samot | nat=yes <-- mean peer is behind NAT |
23:38.30 | rhollan | Oh! Much easier, then. |
23:38.40 | Samot | But where did you get these? |
23:38.45 | rhollan | Let me try that. (I still think audio will be messed up, but we will see). |
23:38.57 | Samot | Because canreinvite has been deprecated for over a decade. |
23:38.59 | rhollan | I though I could make them per extension. I was wrong. |
23:39.02 | Samot | It's not in the sample files. |
23:39.30 | Samot | Well first thing here.. |
23:39.32 | rhollan | No, but they don't describe extension inside and outside the NAT only trunks across NAT. |
23:39.34 | Samot | Chan_SIP is dead. |
23:39.49 | Samot | It hasn't been supported or developed by Asterisk in 6 years. |
23:39.50 | rhollan | I read that, but it worked, so... |
23:40.11 | rhollan | I do remember building it with some alternate. |
23:40.14 | Samot | You should be doing thing via Chan_PJSIP which is the primary SIP drvier. |
23:40.16 | Samot | It's supported. |
23:40.21 | Samot | It is what is being developed. |
23:40.34 | rhollan | So, just PJSIP in place of SIP> |
23:40.40 | Samot | No. |
23:40.45 | Samot | Dude, read the wiki. |
23:40.49 | Samot | Read the sample files. |
23:41.01 | Samot | Don't just guess crap. |
23:42.08 | rhollan | Sigh. I tend to do that when the samples and docs don't seem to describe my configuration. |
23:42.28 | Samot | You have rather basic configurations for the SIP stuff. |
23:42.32 | Samot | It's rather normal. |
23:42.38 | Samot | I'm not sure what special things you need. |
23:42.52 | Samot | Now, this whole bluetooth stuff that is a completely different driver and beast. |
23:43.09 | rhollan | I see that there is a separete pjsip.conf |
23:43.15 | Samot | Yes. |
23:43.27 | rhollan | Oooh! And there is a sample |
23:44.58 | rhollan | Hmm pjsip.conf talks about trunks, not extensions. |
23:45.19 | Samot | An endpoint is an endpoint. |
23:45.34 | Samot | It either accepts registrations like for your SIP clients |
23:45.57 | rhollan | But I found an example of converting from one format to the other, |
23:45.59 | Samot | It can send registrations to another system |
23:46.13 | Samot | It can auth based on IPs or various custom headers. |
23:46.28 | Samot | The sample file has examples of all of it |
23:46.37 | Samot | Trunks, SIP endpoints like phones. |
23:46.48 | Samot | It has multiple examples. |
23:46.50 | rhollan | O.K. |
23:47.40 | rhollan | I'll RTFM, then (well, the docs in the sample file -- they look thorough). But until now, I had grief (no call completion) for a bad client on one phone. |
23:47.53 | Samot | You had bad configs |
23:47.59 | Samot | incomplete, not right |
23:48.08 | Samot | One client worked the other didn't. |
23:48.22 | Samot | That could be because one followed the rules more closely |
23:48.29 | Samot | And thus things weren't right so it didn't work |
23:50.21 | rhollan | Hmm, how to distinguish between chan_SIP and res_PJSIP? |
23:50.45 | Samot | It is in the wiki |
23:51.06 | Samot | The is an entire section on configuring the core things. |
23:51.29 | Samot | Pjsip, music on hold, conferences, etc. |
23:51.31 | rhollan | I found it. Just not sure how it picks one: I think I built both drivers. |
23:51.40 | Samot | There are no default configs. |
23:54.23 | Samot | It doesn't pick one |
23:54.44 | Samot | You do. |
23:55.16 | rhollan | Granted, but where do I say which one to use? I guess I can read the docs but I am impatient. I'm tempted to stick to chan_SIP until I get audio working unless pjsip handles NAT better. |
23:55.36 | Samot | It does a lot better not just NAT |
23:55.58 | Samot | You say which to use by configuring them |
23:56.01 | rhollan | granted, but I want things working before I mess with them unless I know they won't. |
23:56.10 | rhollan | I do have this in modules.conf: |
23:56.16 | rhollan | noload => res_hep.so |
23:56.16 | rhollan | noload => res_hep_pjsip.so |
23:56.16 | rhollan | noload => res_hep_rtcp.so |
23:56.16 | rhollan | ; |
23:56.24 | rhollan | so am wondering what exactly that disables. |
23:56.42 | Samot | Those modules |
23:56.55 | Samot | You really need to learn basics |
23:57.02 | rhollan | smartypants. Yes, but what do they DO? I know: RTFM |
23:57.37 | Samot | I'm not explaining the individual modules |
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23:58.29 | rhollan | Wasn't expecting you to. But, moving those stanzas to [general] gave me audio! So I can break for now, and look at m oving toward pjsip as a separate proble,. |
23:58.35 | rhollan | Thanks. |