IRC log for #asterisk on 20190303

00:30.21life_of_eHow do dynamic features get enabled after being added to features.conf?  I have DYNAMIC_FEATURES set in the globals section of extensions.conf but they're not working.  Do I have to set DYNAMIC_FEATURES in every context/extension?
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00:37.25Samotlife_of_e: How do you call on them?
00:37.49SamotAnd what do you have as the global variable?
00:38.38life_of_eRight now I put in a test DF in the [applicationmap] section of features.conf:  testdf => #*,peer,Playback,tt-monkeys
00:38.52life_of_ein [globals] I have DYNAMIC_FEATURES=testdf
00:41.02life_of_eit just doesn't seem to be playing
01:13.18life_of_ehmm, well I restarted everything and got it working.  I had already done a reload on features and the dialplan but who knows
01:15.09life_of_eI found an interesting device that can listen for DTMF A,B,C, or D and will flash the PSTN line it's attached to.  This solves my problem of trying to flash over from a SIP phone to the PSTN on an FXO ATA because Asterisk apparently won't send a flash itself.
01:15.30life_of_eAt least not to a SIP FXO
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08:20.59Posterdatihi
08:21.28Posterdatiplease help, I have problem with incoming calls: the call drops after 1 minute! Thanks
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09:47.01Posterdati:(
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10:02.30PosterdatiI disabled the sip alg from the router, but incoming calls always hang up after 1 minute
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10:38.39PosterdatiI've got this during an incoming call --> SIP/2.0 486 Busy Here
10:52.08Posterdatihttps://paste.debian.net/1071192/
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13:23.46Posterdatiseems that the asterisk server tries to authenticate the sip phone during the call the auth is denied
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14:06.05SamotLooks like a phone issue.
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14:47.44Posterdatiwhy the provider wants to authenticate the sip phone which is behind asterisk???
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14:49.00Posterdatihttps://paste.debian.net/1071215/
14:49.02Posterdati???
14:51.29SamotWhat are you talking about?
14:51.33SamotYou're not showing everything.
14:51.38SamotOne packet isn't enough.
14:54.39Posterdatihttps://paste.debian.net/1071216/
14:54.54Posterdatithis is a complete phone call originated from outside
15:02.20Posterdatiany hints?
15:04.25SamotSo you're making the call from the phone through asterisk to the provider?
15:04.45Posterdatino
15:04.57PosterdatiI'm receiving a phone call
15:05.02SamotOK this isn't a full call.
15:05.15SamotYou're missing the first INVITE from the provider if this is an incoming call.
15:05.17Posterdatiprovider --> asterisk --> ip phone
15:05.57SamotAll you are showing is Asterisk calling the phone.
15:06.16SamotProvider -> Asterisk and Asterisk -> Phone
15:06.22SamotThere are TWO legs of this call.
15:06.24SamotNot one.
15:06.39SamotYour provider has no idea Asterisk is sending this call to a phone.
15:08.02Posterdatihttps://paste.debian.net/1071219/
15:08.21Posterdatitwo legs?
15:10.14SamotNo matching peer for '+393293516963' from '5.97.53.10:5060' <-- That's a problem.
15:10.41Samot<--- Reliably Transmitting (NAT) to 5.97.53.10:5060 ---> <-- Providers are generally not behind NAT
15:11.21Posterdati+393293516963 this is the number calling asterisk
15:11.28Posterdati+393293516963 this is the external number calling asterisk
15:12.08SamotReliably Transmitting (NAT) to 10.0.0.3:5069: <-- You're phones should be behind NAT they are on the same lan as the PBX
15:12.39Posterdatiyes
15:12.42Posterdatiso?
15:12.45SamotI get that. You have this setup completely wrong.
15:12.46Posterdatinat=no
15:13.39SamotThat's not what this is showing.
15:13.47Posterdatinow
15:13.54PosterdatiI have
15:14.03Posterdatinat=force_rport,comedia
15:14.12Posterdatiin [general]
15:14.23Posterdatiin [sip-phone]
15:14.28Posterdatiand [telecomitalia]
15:18.15Posterdatishall I place nat=no ?
15:31.42Posterdati:(
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15:51.53SamotDid you try it with nat=no?
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17:22.19Posterdatisame
17:22.49Posterdatiseems that the provide ask the sip phone the auth
17:23.04Posterdatiseems that the provider asks the sip phone the auth
17:23.52Posterdati[Mar  3 18:21:45] NOTICE[-1][C-0000000d]: chan_sip.c:24041 handle_response_invite: Failed to authenticate on INVITE to '<sip:+390624405245@telecomitalia.it;user=phone>;tag=as2908da7f'
17:23.52Posterdati<PROTECTED>
17:26.13fileset the "directmedia" option to "no"
17:26.42filethey aren't trying to authenticate the phone - Asterisk is trying to allow the media to flow directly which won't work
17:27.13Posterdatiexactly, it told me that rtp is established directly among provider and phone
17:27.25Posterdatidirectmedia=no
17:27.29Posterdatiin general?
17:27.48fileor in the individual sections.
17:28.02Posterdatiok
17:28.06Posterdatilet's try
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17:30.53Posterdatiok, now there's no a direct flow, but asterisk continues to tell that the phone is not authenticated
17:30.56Posterdatihow come?
17:31.06Posterdatiit is among the peers list!!!
17:31.23fileyou haven't provided a current trace, so no idea
17:31.35Posterdatiok
17:33.51Posterdati@file: https://paste.debian.net/1071240/
17:35.09fileI see nothing wrong?
17:35.15fileat least in regards to auth
17:36.15Posterdatiyes
17:36.16fileyour NAT settings seem incorrect
17:36.23Posterdatihow?
17:36.25fileyou're giving the VoIP provider a private IP address
17:37.27Posterdatibindport and bindaddress ??
17:37.38filehttps://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L869
17:39.59sibiriaPosterdati: for chan_sip, read up externaddr/externhost
17:40.24sibiriayeah exactly what file pointed to
17:40.29Samot10:10:20 AM <Samot> No matching peer for '+393293516963' from '5.97.53.10:5060' <-- That's a problem.
17:40.41Samottelecomitalia/+3906244052 156.54.82.96                                Yes        Yes            5060     Unmonitored
17:40.56PosterdatiI removed bindaddr and bindport
17:41.33SamotThis call we keep seeing in the debugs is coming from an IP you have no matching Chan_SIP trunk for.
17:41.46SamotSo that pretty much tells me you have this box wide open for anything to send calls to you.
17:42.20SamotReliably Transmitting (NAT) to 5.97.53.10:5060:
17:42.20SamotREGISTER sip:telecomitalia.it SIP/2.0
17:42.43SamotYou seem to be sending and receiving things from an IP that doesn't match your Chan_SIP trunk/peer.
17:47.40Samothttps://www.irccloud.com/pastebin/E7m9sN4T/
17:47.55Samot486 Busy Here really shouldn't be the reply to an OPTIONS request.
17:48.56fileit's not... wrong
17:49.01fileit's technically correct...
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17:50.54Posterdatiseems to be solved
17:51.50PosterdatiI removed bindaddr and bindport lines in [general] now incoming connection is going over 1 minute :)
17:52.06Posterdatistill get busy warning in the log :)
17:52.16Posterdatibut it works
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17:53.34Posterdatithanks people
18:21.50Samot12:49:07 PM <@file> it's technically correct... <- It really shouldn't be the reply. It should be a 404, 408 or 200 but yeah, it did reply...just funky.
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18:22.07fileit's supposed to behave as if it were an INVITE
18:22.33fileif the device in question would respond with a 486 Busy Here, as it's in a call, to an INVITE then the response to the OPTIONS is technically correct
18:25.57SamotYes.
18:26.23SamotBut I can say that this is the first time I've ever seen that.
18:26.41SamotAnd if it's not, it's been so rare and long ago I don't recall it.
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