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03:06.02 | grummund_ | 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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11:42.26 | grummund_ | 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. |
11:43.54 | grummund_ | looking at pjsip history there is no incoming traffic until asterisk attempts a REGISTER to the provider. |
11:45.38 | grummund_ | the REGISTER gets a 401 reply, asterisk immediately sends another REGISTER which gets a 200 OK |
11:47.09 | grummund_ | then the provider sends an INVITE, which succeeds, but the caller has already had 10-20s of ringing by this time. |
11:51.04 | grummund_ | or the caller gets silence and then a connect error, but no traffic in the pjsip history. |
11:52.37 | grummund_ | is this a NAT issue? should the router be configured to forward port 5060 to the asterisk box? |
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14:25.28 | grummund_ | setting 'expiration=60' may have fixed it. |
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17:10.02 | grummund_ | answering my own question... |
17:10.04 | grummund_ | it seems the correct way to do this is [..._aor] qualify_frequency=50; qualify_timeout=0; |
17:13.41 | Kobaz | qualify timeout = 0 ??? |
17:13.55 | Kobaz | i'm jumping in last minute here, but that seems bad on so many levels |
17:16.20 | grummund_ | ok, well i read that with non-zero qualify_timeout, then upon no reply asterisk will remove the connection, even mid-call. |
17:16.44 | Kobaz | it's not a connection |
17:16.46 | Kobaz | it's a contact |
17:16.53 | Kobaz | it's AOR = address of record |
17:17.02 | Kobaz | for when you place a new call to that endpoint that uses the contact |
17:17.10 | Kobaz | if a qualify timeout fails, you don't drop the call |
17:17.19 | grummund_ | hmm, i read that it did. |
17:17.32 | Kobaz | if 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 |
17:18.04 | grummund_ | anyway seeing as all we want is to keep the nat table udp entry alive, does it matter? |
17:18.14 | grummund_ | qualify is off by default. |
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17:18.56 | Kobaz | you might have undefined/adverse behavior for qualify enabled, and qualify_timeout=0 |
17:19.21 | Kobaz | like, what in your mind would happen, if you're qualifying every, say... 60 seconds, and there's no timeout? |
17:19.35 | Kobaz | it would just qualify and sit there... forever... doing.... nothing |
17:19.55 | Kobaz | timeouts are good, use them for your benefit |
17:20.50 | grummund_ | ok, in my mind... ;) REGISTER would expire, just as it would without qualify enable. |
17:21.34 | Kobaz | right, registrations have timeouts |
17:21.43 | grummund_ | we actually don't care if the OPTIONS packet reaches the server. |
17:21.49 | Kobaz | there is an implicit timeout |
17:22.04 | Kobaz | it's the registration period, becomes the timeout... there's always a timeout in any kind of sane communication system |
17:22.30 | grummund_ | btw, this is my limited noobie understanding, just piecing thing together. |
17:23.12 | Kobaz | curious... why don't you care if options goes out (and comes back?) |
17:23.27 | Kobaz | it's a great way to test if your stuff's actually working to some effect |
17:23.54 | grummund_ | it's just to keep the udp entry in the nat table alive (inside the router). |
17:24.24 | grummund_ | that is my theory as to why incoming calls fail, with this particular sip trunk. |
17:24.28 | Kobaz | from your end.. right? |
17:24.43 | Kobaz | asterisk is behind a nat? |
17:24.46 | grummund_ | yes |
17:25.02 | Kobaz | that's typically what registration is for |
17:25.06 | grummund_ | behind a nat and tregistering with a sip provider |
17:25.09 | Kobaz | registration from your asterisk -> ITSP |
17:25.40 | Kobaz | try lowering the registration period, and if your incoming calls are sometimes failing, your router needs attention |
17:25.52 | grummund_ | the register expiry is (my guess) longer than the timeout used inside the router for udp entries. |
17:26.02 | Kobaz | sip options might help, but... sounds like a router problem |
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17:26.23 | grummund_ | yes, forcing a lower register expiry works too. |
17:26.48 | grummund_ | <PROTECTED> |
17:27.10 | grummund_ | http://forums.asterisk.org/viewtopic.php?t=95303 |
17:27.16 | Kobaz | and then you'll know if the service is 'up' or not |
17:27.20 | Kobaz | if your qualify is failing |
17:27.43 | grummund_ | true |
17:27.55 | Kobaz | just because it's responding to options doesn't mean it will take a call |
17:28.00 | Kobaz | but it's a nice early warning system to know something's up |
17:28.07 | Kobaz | there's no reason to disable timeouts or muck that up |
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17:29.18 | grummund_ | with the other siptrunk i have (that works), the response to REGISTER sets an expiry of 60s. |
17:29.46 | grummund_ | so from the provider's side they effectively fix the issue. |
17:32.00 | grummund_ | apparently (i read) it's not unusual for a router to delete udp entries from the nat table even down to 30s. |
17:33.22 | Kobaz | depends on the router, there's no standard with that |
17:33.24 | Kobaz | what router? |
17:33.42 | grummund_ | technicolor adsl router |
17:34.54 | grummund_ | "connection state timeout values vary from product to product, but typical values are 30â¦180 seconds for UDP and 30â¦60 minutes for TCP" |
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17:35.39 | Kobaz | adsl.. i'm sorry |
17:44.50 | grummund_ | btw, would forwarding udp port 5060 from the router to the asterisk machine also resolve this? |
17:45.06 | grummund_ | because i have to do that anyway. |
17:45.28 | grummund_ | (it's currently not forwarded, but will be) |
17:48.24 | Kobaz | yeah sure would |
17:48.33 | Kobaz | then the connection tracking wouldn't expire |
17:56.10 | grummund_ | so once a call has started the contact is no longer needed? |
17:56.23 | Kobaz | correct |
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