IRC log for #asterisk on 20200621

04:25.00*** join/#asterisk infobot (ibot@96-86-209-99-static.hfc.comcastbusiness.net)
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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12:45.10ZombieDoes anyone here use Asterisk with Hylafax?
12:45.29ZombieI want to do some troubleshooting of interfacing the two/
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14:12.27igcewielingNah, ReceiveFax/SendFax is enough for me.
14:12.30drmessanoFrom the looks of it, no one has used Hylafax in years
14:21.09ZombieI have a Government Agency I communicate with in that fashion.
14:21.32Zombie(By Faxing things.)
14:22.17drmessanoWe know
14:22.40drmessanoBecause Coronavirus eats emails
14:27.47Zombie...
14:28.15ZombieThey don't allow reporting documentation to be transmitted by that means,
14:30.36igcewielingStill doesn't require Hylafax
14:48.52ZombieI would like to move ahead with this anyway.
14:49.13igcewieling*shrug*  you are on your own then.
14:51.26drmessanoYoure bolting on a dead solution
14:52.04ZombieThe part that has me messed up is iaxmodem really.
14:52.13ZombieCan we focus on that?
14:52.18drmessanoNo, not really
14:52.23Zombiebecause that has multiple uses.
14:52.35drmessanoThe whole thing is a black hole
14:52.44ZombieI have some legacy hardware that uses Serial.
14:53.52igcewielingI've not used IAX since...hmm...2004
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15:29.38shimshonI need help with FastAGI
15:35.17shimshonI use FastAGI to run bash script, how can i use the variable that FastAgi send to script?
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19:51.56*** join/#asterisk rhollan (~rhollan@v-72-5-54-188.unman-vds.premium-seattle.nfoservers.com)
19:53.09rhollanAre 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.
19:54.23rhollanBasically, 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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21:06.27*** join/#asterisk rhollan (~rhollan@199.229.250.180)
21:06.33rhollanAre 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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22:55.34*** join/#asterisk rhollan (~rhollan@v-72-5-54-188.unman-vds.premium-seattle.nfoservers.com)
22:56.21rhollanSigh. Getting weird Via headers on an Android client.
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23:19.22SamotWhat weird headers?
23:19.32rhollanAnother client works better: can make calls. But NAT is still messing with RTP traffic, not exexpectedly.
23:19.36rhollanunexpectedly
23:19.48rhollanWas getting an odd 10.8.x.y Via header.
23:19.56rhollanI tried a different client and calls completed.
23:20.01SamotWhat is that odd?
23:20.04SamotWhy*
23:20.26rhollanWhy should my SIP client go th rough a proxy with a non-routable IP address?
23:20.36rhollanA different client works better
23:20.38SamotVia is the path.
23:20.54SamotAll sip messages have Via headers
23:20.55rhollanYes, but 10.x.x.x is not publicly routable
23:20.57SamotSometimes multiple.
23:21.01SamotThat doesn't matter.
23:21.04SamotIt's the _path_
23:21.11SamotIt came via this via that to here.
23:21.24rhollanR ight,  b ut I can;'t respond to a non-routable IP address
23:21.38rhollanIn any case, a different client doesn;'t add them and it works.
23:21.42SamotIn what regards?
23:21.49rhollanCalls complete.
23:21.52SamotIt adds via headers
23:21.57rhollanBut no audio.
23:22.12SamotThis has nothing to do with the via headers.
23:22.50SamotThis has to do with the client and/or router not using public routed IPs in the Contact and/or SDP body.
23:23.10SamotSo 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.
23:23.17rhollanI 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
23:23.35rhollanRight, b ut I still lsee Via headers with 10.8.x.y
23:23.44SamotWithout actually seeing it, I can't say but you're on the wrong road with that thinking.
23:23.55rhollanAnd calls now complete. Now sure WHAT those Via headers do if they are not publically routable.
23:24.03rhollanNot sure what...
23:24.05SamotThe hold the PATH of the message.
23:24.21SamotWhat systems the SIP message came through including where it originated.
23:24.25rhollanBut to what end if not publicly routable?
23:24.37SamotThe audio doesn't use the via headers
23:24.47SamotThe audio is based on the SDP body data
23:24.57rhollanGranted, that is a separate issue, and the only one that remains. I thing NAT is messing that up.
23:25.07SamotWhat NAT thing?
23:26.12rhollanHere'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
23:26.20rhollanNetwork Address Translation.
23:26.31SamotI know what NAT is rhollan.
23:26.41SamotWhat kind of tunnel?
23:26.50SamotPure IPSEC or GRE/IPSEC?
23:27.03SamotA VPN tunnel by passes the NAT
23:27.09SamotBecause it puts you on the _local network_
23:27.20rhollanYes, but I have a public IP address on the other side
23:27.35SamotWhat SIP driver are you using?
23:27.36rhollanpure IPSEC]
23:27.41SamotSo you have proper routes?
23:27.53rhollanWhere? In the androud client? On the server I am running (duh) Asterisk.
23:28.02SamotOn Asterisk.
23:28.07rhollanrouting appears fine, though I am checking audio
23:28.08SamotWhich SIP driver are you using?
23:28.15rhollanOh! Hang on
23:29.59rhollanWher is that specified again? In extensions.conf I have SIP/100 (local) and SIP/101 (remote)
23:30.13SamotThen that is Chan_SIP
23:30.32SamotSame with provider setup?
23:30.36rhollanIf you say so. I am onbiously an Asterisk Noob.
23:30.44rhollanThere is no prov ider. I AM the provider.
23:30.51rhollan(Well, to me)
23:31.04SamotSo this is just local PBX calling
23:31.05rhollanHaven't set up a remote trunk for PSTN connections.
23:31.09rhollanYes, at this point.
23:31.22SamotSo show your sip.conf
23:31.50SamotPastebin it.
23:32.21rhollanThe 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).
23:33.19SamotShow your sip.conf
23:34.13rhollan[100]
23:34.14rhollantype=friend
23:34.14rhollansecret=14Bddoqc% ;NOTE it is important to set up a complex password
23:34.14rhollanqualify=no ; Will not drop the connection
23:34.14rhollannat=no ; This phone wil not be outside the network
23:34.14rhollanhost=dynamic ; This device registers with us
23:34.16rhollancanreinvite=no ; Asterisk by default tries to redirect
23:34.18rhollancontext=default ; Or whatever context you want to define in Asterisk
23:34.20rhollan;;mailbox=103@default ; only if you are configuring voicemail.
23:34.22rhollan[101]
23:34.24rhollanexternip=72.5.54.188
23:34.26rhollanlocalnet=10.0.0.0/255.0.0.0
23:34.28rhollantype=friend
23:34.30rhollansecret=14Bddoqc% ;NOTE it is important to set up a complex password
23:34.32rhollanqualify=no ; Will not drop the connection
23:34.34rhollannat=yes ; This phone may be outside the network
23:34.38rhollanhost=dynamic ; This device registers with us
23:34.40rhollancanreinvite=no ; Asterisk by default tries to redirect
23:34.42rhollancontext=default ; Or whatever context you want to define in Asterisk
23:34.44rhollan;mailbox=103@default ; only if you are configuring voicemail.
23:34.46rhollanyeah, will change the passwords. Didn't vet it.
23:35.33SamotI literally said pastebin it.
23:35.58rhollanSorry, missed it. Only two extensions, so I ffigured meh.
23:36.22Samotexternip=72.5.54.188
23:36.39Samotlocalnet=10.0.0.0/255.0.0.0
23:36.44SamotThose should be in the general section
23:36.57SamotI highly doubt you need a full class A for this but OK.
23:37.01rhollanThat's the public facing side and private internal net. Yes, but only one extension is outside the lan
23:37.18SamotThat needs to be in the general section.
23:37.33SamotYou have nothing to tell Chan_SIP that the PBX itself is behind NAT.
23:37.43rhollan10,<net class>.<room>.<devcice>.
23:37.52Samotexternip and such should not be in the peer sections.
23:37.56rhollanBut, how do I distinguish extensions inside vs outside?
23:37.58SamotIt should be in the general section.
23:38.05SamotYou tell it the NAT
23:38.16Samotnat=yes <-- mean peer is behind NAT
23:38.30rhollanOh! Much easier, then.
23:38.40SamotBut where did you get these?
23:38.45rhollanLet me try that. (I still think audio will be messed up, but we will see).
23:38.57SamotBecause canreinvite has been deprecated for over a decade.
23:38.59rhollanI though I could make them per extension. I was wrong.
23:39.02SamotIt's not in the sample files.
23:39.30SamotWell first thing here..
23:39.32rhollanNo, but they don't describe extension inside and outside the NAT only trunks across NAT.
23:39.34SamotChan_SIP is dead.
23:39.49SamotIt hasn't been supported or developed by Asterisk in 6 years.
23:39.50rhollanI read that, but it worked, so...
23:40.11rhollanI do remember building it with some alternate.
23:40.14SamotYou should be doing thing via Chan_PJSIP which is the primary SIP drvier.
23:40.16SamotIt's supported.
23:40.21SamotIt is what is being developed.
23:40.34rhollanSo, just PJSIP in place of SIP>
23:40.40SamotNo.
23:40.45SamotDude, read the wiki.
23:40.49SamotRead the sample files.
23:41.01SamotDon't just guess crap.
23:42.08rhollanSigh. I tend to do that when the samples and docs don't seem to describe my configuration.
23:42.28SamotYou have rather basic configurations for the SIP stuff.
23:42.32SamotIt's rather normal.
23:42.38SamotI'm not sure what special things you need.
23:42.52SamotNow, this whole bluetooth stuff that is a completely different driver and beast.
23:43.09rhollanI see  that there is a separete pjsip.conf
23:43.15SamotYes.
23:43.27rhollanOooh! And there is a sample
23:44.58rhollanHmm pjsip.conf talks about trunks, not extensions.
23:45.19SamotAn endpoint is an endpoint.
23:45.34SamotIt either accepts registrations like for your SIP clients
23:45.57rhollanBut I found an example of converting from one format to the other,
23:45.59SamotIt can send registrations to another system
23:46.13SamotIt can auth based on IPs or various custom headers.
23:46.28SamotThe sample file has examples of all of it
23:46.37SamotTrunks, SIP endpoints like phones.
23:46.48SamotIt has multiple examples.
23:46.50rhollanO.K.
23:47.40rhollanI'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.
23:47.53SamotYou had bad configs
23:47.59Samotincomplete, not right
23:48.08SamotOne client worked the other didn't.
23:48.22SamotThat could be because one followed the rules more closely
23:48.29SamotAnd thus things weren't right so it didn't work
23:50.21rhollanHmm, how to distinguish between chan_SIP and res_PJSIP?
23:50.45SamotIt is in the wiki
23:51.06SamotThe is an entire section on configuring the core things.
23:51.29SamotPjsip, music on hold, conferences, etc.
23:51.31rhollanI found it. Just not sure how it picks one: I think I built both drivers.
23:51.40SamotThere are no default configs.
23:54.23SamotIt doesn't pick one
23:54.44SamotYou do.
23:55.16rhollanGranted, 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.
23:55.36SamotIt does a lot better not just NAT
23:55.58SamotYou say which to use by configuring them
23:56.01rhollangranted, but I want things working before I mess with them unless I know they won't.
23:56.10rhollanI do have this in modules.conf:
23:56.16rhollannoload => res_hep.so
23:56.16rhollannoload => res_hep_pjsip.so
23:56.16rhollannoload => res_hep_rtcp.so
23:56.16rhollan;
23:56.24rhollanso am wondering what exactly that disables.
23:56.42SamotThose modules
23:56.55SamotYou really need to learn basics
23:57.02rhollansmartypants. Yes, but what do they DO? I know: RTFM
23:57.37SamotI'm not explaining the individual modules
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23:58.29rhollanWasn'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,.
23:58.35rhollanThanks.

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