IRC log for #asterisk on 20210117

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03:06.02grummund_added g722
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06:00.43*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.1.1, 16.15.1 (2020/12/22) Final Bugfix: 13.38.1, 17.9.1 (2020/12/22); 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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11:42.26grummund_i have a problem with one siptrunk (but not another with similar config), that is inbound calls take a long time to connect, if at all.
11:43.54grummund_looking at pjsip history there is no incoming traffic until asterisk attempts a REGISTER to the provider.
11:45.38grummund_the REGISTER gets a 401 reply, asterisk immediately sends another REGISTER which gets a 200 OK
11:47.09grummund_then the provider sends an INVITE, which succeeds, but the caller has already had 10-20s of ringing by this time.
11:51.04grummund_or the caller gets silence and then a connect error, but no traffic in the pjsip history.
11:52.37grummund_is this a NAT issue?  should the router be configured to forward port 5060 to the asterisk box?
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14:25.28grummund_setting 'expiration=60' may have fixed it.
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17:10.02grummund_answering my own question...
17:10.04grummund_it seems the correct way to do this is [..._aor] qualify_frequency=50; qualify_timeout=0;
17:13.41Kobazqualify timeout = 0 ???
17:13.55Kobazi'm jumping in last minute here, but that seems bad on so many levels
17:16.20grummund_ok, well i read that with non-zero qualify_timeout, then upon no reply asterisk will remove the connection, even mid-call.
17:16.44Kobazit's not a connection
17:16.46Kobazit's a contact
17:16.53Kobazit's AOR = address of record
17:17.02Kobazfor when you place a new call to that endpoint that uses the contact
17:17.10Kobazif a qualify timeout fails, you don't drop the call
17:17.19grummund_hmm, i read that it did.
17:17.32Kobazif the ENDPOINT fails, you drop the call.  if your endpoint is failing, all bets are off, and your contact and your call will go away
17:18.04grummund_anyway seeing as all we want is to keep the nat table udp entry alive, does it matter?
17:18.14grummund_qualify is off by default.
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17:18.56Kobazyou might have undefined/adverse behavior for qualify enabled, and qualify_timeout=0
17:19.21Kobazlike, what in your mind would happen, if you're qualifying every, say... 60 seconds, and there's no timeout?
17:19.35Kobazit would just qualify and sit there... forever... doing.... nothing
17:19.55Kobaztimeouts are good, use them for your benefit
17:20.50grummund_ok, in my mind... ;) REGISTER would expire, just as it would without qualify enable.
17:21.34Kobazright, registrations have timeouts
17:21.43grummund_we actually don't care if the OPTIONS packet reaches the server.
17:21.49Kobazthere is an implicit timeout
17:22.04Kobazit's the registration period, becomes the timeout... there's always a timeout in any kind of sane communication system
17:22.30grummund_btw, this is my limited noobie understanding, just piecing thing together.
17:23.12Kobazcurious... why don't you care if options goes out (and comes back?)
17:23.27Kobazit's a great way to test if your stuff's actually working to some effect
17:23.54grummund_it's just to keep the udp entry in the nat table alive (inside the router).
17:24.24grummund_that is my theory as to why incoming calls fail, with this particular sip trunk.
17:24.28Kobazfrom your end.. right?
17:24.43Kobazasterisk is behind a nat?
17:24.46grummund_yes
17:25.02Kobazthat's typically what registration is for
17:25.06grummund_behind a nat and tregistering with a sip provider
17:25.09Kobazregistration from your asterisk -> ITSP
17:25.40Kobaztry lowering the registration period, and if your incoming calls are sometimes failing, your router needs attention
17:25.52grummund_the register expiry is (my guess) longer than the timeout used inside the router for udp entries.
17:26.02Kobazsip options might help, but... sounds like a router problem
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17:26.23grummund_yes, forcing a lower register expiry works too.
17:26.48grummund_<PROTECTED>
17:27.10grummund_http://forums.asterisk.org/viewtopic.php?t=95303
17:27.16Kobazand then you'll know if the service is 'up' or not
17:27.20Kobazif your qualify is failing
17:27.43grummund_true
17:27.55Kobazjust because it's responding to options doesn't mean it will take a call
17:28.00Kobazbut it's a nice early warning system to know something's up
17:28.07Kobazthere's no reason to disable timeouts or muck that up
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17:29.18grummund_with the other siptrunk i have (that works), the response to REGISTER sets an expiry of 60s.
17:29.46grummund_so from the provider's side they effectively fix the issue.
17:32.00grummund_apparently (i read) it's not unusual for a router to delete udp entries from the nat table even down to 30s.
17:33.22Kobazdepends on the router, there's no standard with that
17:33.24Kobazwhat router?
17:33.42grummund_technicolor adsl router
17:34.54grummund_"connection state timeout values vary from product to product, but typical values are 30…180 seconds for UDP and 30…60 minutes for TCP"
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17:35.39Kobazadsl.. i'm sorry
17:44.50grummund_btw, would forwarding udp port 5060 from the router to the asterisk machine also resolve this?
17:45.06grummund_because i have to do that anyway.
17:45.28grummund_(it's currently not forwarded, but will be)
17:48.24Kobazyeah sure would
17:48.33Kobazthen the connection tracking wouldn't expire
17:56.10grummund_so once a call has started the contact is no longer needed?
17:56.23Kobazcorrect
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