00:30.21 | life_of_e | How 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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00:37.25 | Samot | life_of_e: How do you call on them? |
00:37.49 | Samot | And what do you have as the global variable? |
00:38.38 | life_of_e | Right now I put in a test DF in the [applicationmap] section of features.conf: testdf => #*,peer,Playback,tt-monkeys |
00:38.52 | life_of_e | in [globals] I have DYNAMIC_FEATURES=testdf |
00:41.02 | life_of_e | it just doesn't seem to be playing |
01:13.18 | life_of_e | hmm, well I restarted everything and got it working. I had already done a reload on features and the dialplan but who knows |
01:15.09 | life_of_e | I 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. |
01:15.30 | life_of_e | At least not to a SIP FXO |
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08:20.59 | Posterdati | hi |
08:21.28 | Posterdati | please help, I have problem with incoming calls: the call drops after 1 minute! Thanks |
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09:47.01 | Posterdati | :( |
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10:02.30 | Posterdati | I disabled the sip alg from the router, but incoming calls always hang up after 1 minute |
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10:38.39 | Posterdati | I've got this during an incoming call --> SIP/2.0 486 Busy Here |
10:52.08 | Posterdati | https://paste.debian.net/1071192/ |
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13:23.46 | Posterdati | seems that the asterisk server tries to authenticate the sip phone during the call the auth is denied |
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14:06.05 | Samot | Looks like a phone issue. |
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14:47.44 | Posterdati | why the provider wants to authenticate the sip phone which is behind asterisk??? |
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14:49.00 | Posterdati | https://paste.debian.net/1071215/ |
14:49.02 | Posterdati | ??? |
14:51.29 | Samot | What are you talking about? |
14:51.33 | Samot | You're not showing everything. |
14:51.38 | Samot | One packet isn't enough. |
14:54.39 | Posterdati | https://paste.debian.net/1071216/ |
14:54.54 | Posterdati | this is a complete phone call originated from outside |
15:02.20 | Posterdati | any hints? |
15:04.25 | Samot | So you're making the call from the phone through asterisk to the provider? |
15:04.45 | Posterdati | no |
15:04.57 | Posterdati | I'm receiving a phone call |
15:05.02 | Samot | OK this isn't a full call. |
15:05.15 | Samot | You're missing the first INVITE from the provider if this is an incoming call. |
15:05.17 | Posterdati | provider --> asterisk --> ip phone |
15:05.57 | Samot | All you are showing is Asterisk calling the phone. |
15:06.16 | Samot | Provider -> Asterisk and Asterisk -> Phone |
15:06.22 | Samot | There are TWO legs of this call. |
15:06.24 | Samot | Not one. |
15:06.39 | Samot | Your provider has no idea Asterisk is sending this call to a phone. |
15:08.02 | Posterdati | https://paste.debian.net/1071219/ |
15:08.21 | Posterdati | two legs? |
15:10.14 | Samot | No matching peer for '+393293516963' from '5.97.53.10:5060' <-- That's a problem. |
15:10.41 | Samot | <--- Reliably Transmitting (NAT) to 5.97.53.10:5060 ---> <-- Providers are generally not behind NAT |
15:11.21 | Posterdati | +393293516963 this is the number calling asterisk |
15:11.28 | Posterdati | +393293516963 this is the external number calling asterisk |
15:12.08 | Samot | Reliably Transmitting (NAT) to 10.0.0.3:5069: <-- You're phones should be behind NAT they are on the same lan as the PBX |
15:12.39 | Posterdati | yes |
15:12.42 | Posterdati | so? |
15:12.45 | Samot | I get that. You have this setup completely wrong. |
15:12.46 | Posterdati | nat=no |
15:13.39 | Samot | That's not what this is showing. |
15:13.47 | Posterdati | now |
15:13.54 | Posterdati | I have |
15:14.03 | Posterdati | nat=force_rport,comedia |
15:14.12 | Posterdati | in [general] |
15:14.23 | Posterdati | in [sip-phone] |
15:14.28 | Posterdati | and [telecomitalia] |
15:18.15 | Posterdati | shall I place nat=no ? |
15:31.42 | Posterdati | :( |
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15:51.53 | Samot | Did you try it with nat=no? |
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17:22.19 | Posterdati | same |
17:22.49 | Posterdati | seems that the provide ask the sip phone the auth |
17:23.04 | Posterdati | seems that the provider asks the sip phone the auth |
17:23.52 | Posterdati | [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' |
17:23.52 | Posterdati | <PROTECTED> |
17:26.13 | file | set the "directmedia" option to "no" |
17:26.42 | file | they aren't trying to authenticate the phone - Asterisk is trying to allow the media to flow directly which won't work |
17:27.13 | Posterdati | exactly, it told me that rtp is established directly among provider and phone |
17:27.25 | Posterdati | directmedia=no |
17:27.29 | Posterdati | in general? |
17:27.48 | file | or in the individual sections. |
17:28.02 | Posterdati | ok |
17:28.06 | Posterdati | let's try |
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17:30.53 | Posterdati | ok, now there's no a direct flow, but asterisk continues to tell that the phone is not authenticated |
17:30.56 | Posterdati | how come? |
17:31.06 | Posterdati | it is among the peers list!!! |
17:31.23 | file | you haven't provided a current trace, so no idea |
17:31.35 | Posterdati | ok |
17:33.51 | Posterdati | @file: https://paste.debian.net/1071240/ |
17:35.09 | file | I see nothing wrong? |
17:35.15 | file | at least in regards to auth |
17:36.15 | Posterdati | yes |
17:36.16 | file | your NAT settings seem incorrect |
17:36.23 | Posterdati | how? |
17:36.25 | file | you're giving the VoIP provider a private IP address |
17:37.27 | Posterdati | bindport and bindaddress ?? |
17:37.38 | file | https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L869 |
17:39.59 | sibiria | Posterdati: for chan_sip, read up externaddr/externhost |
17:40.24 | sibiria | yeah exactly what file pointed to |
17:40.29 | Samot | 10:10:20 AM <Samot> No matching peer for '+393293516963' from '5.97.53.10:5060' <-- That's a problem. |
17:40.41 | Samot | telecomitalia/+3906244052 156.54.82.96 Yes Yes 5060 Unmonitored |
17:40.56 | Posterdati | I removed bindaddr and bindport |
17:41.33 | Samot | This call we keep seeing in the debugs is coming from an IP you have no matching Chan_SIP trunk for. |
17:41.46 | Samot | So that pretty much tells me you have this box wide open for anything to send calls to you. |
17:42.20 | Samot | Reliably Transmitting (NAT) to 5.97.53.10:5060: |
17:42.20 | Samot | REGISTER sip:telecomitalia.it SIP/2.0 |
17:42.43 | Samot | You seem to be sending and receiving things from an IP that doesn't match your Chan_SIP trunk/peer. |
17:47.40 | Samot | https://www.irccloud.com/pastebin/E7m9sN4T/ |
17:47.55 | Samot | 486 Busy Here really shouldn't be the reply to an OPTIONS request. |
17:48.56 | file | it's not... wrong |
17:49.01 | file | it's technically correct... |
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17:50.54 | Posterdati | seems to be solved |
17:51.50 | Posterdati | I removed bindaddr and bindport lines in [general] now incoming connection is going over 1 minute :) |
17:52.06 | Posterdati | still get busy warning in the log :) |
17:52.16 | Posterdati | but it works |
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17:53.34 | Posterdati | thanks people |
18:21.50 | Samot | 12:49:07 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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18:22.07 | file | it's supposed to behave as if it were an INVITE |
18:22.33 | file | if 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 |
18:25.57 | Samot | Yes. |
18:26.23 | Samot | But I can say that this is the first time I've ever seen that. |
18:26.41 | Samot | And if it's not, it's been so rare and long ago I don't recall it. |
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