IRC log for #asterisk on 20191029

00:11.14*** join/#asterisk Typhon (~Typhon@ipservice-092-218-218-131.092.218.pools.vodafone-ip.de)
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11:39.22*** join/#asterisk SQUelcher (~squ@irn-bru.nerdsurf.de)
11:39.50SQUelcherhi
11:44.59SQUelcherI've noticed that when using pjsip_wizard, asterisk sends out the [mytrunk] endpoint name in the sip url. siproxd on another hosts logs something like "comparing urls: sip:mytrunk@asterisk.ip -> sip:real_username@myisp.ip: username mismatch" - how can I fix that?
11:52.57*** join/#asterisk derPlexus (~plexus@81.173.204.226)
12:01.45SQUelchergot it, endpoint/from_user was missing :)
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12:23.14*** mode/#asterisk [+o gtjoseph] by ChanServ
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13:06.51filehttps://www.asterisk.org/community/astricon-live/videos for the AstriCon opening remarks/keynote
13:11.25sibiriais ast16 SRV-aware?
13:11.51sibiriaor rather, is chan_cip and pjsip
13:12.08sibiriasip*
13:12.22SamotChan_SIP doesn't use SRV properly. It only pulls the first record.
13:12.26filePJSIP supports NAPTR and SRV
13:12.33SamotPJSIP handles SRV properly
13:12.49sibiriaok, thanks
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14:00.44aoeuin
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14:28.47*** mode/#asterisk [+o bford] by ChanServ
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18:00.24tehgoochIs there any description of how chan_sip matches an incoming request (e.g. INVITE) to a peer? I am experiencing an issue where there is a trunk with dynamic host and username as a phone number. Any incoming calls with a 'From' header as this phone number seem to be matched to this trunk.
18:27.40*** join/#asterisk dandann00dle (~dandann00@50.236.95.125)
18:44.21fileFrom username first and then IP, not configurable
18:47.52*** join/#asterisk ircarcs (~quassel@169.9.159.77.rev.sfr.net)
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19:36.42*** join/#asterisk MLC (~MLC@63.249.40.11)
19:38.02MLChow do I configure a queue so that the caller can break out of the music on hold with a key to get to a voicemail box?
20:04.12MLCfinally found it: context= in queues.conf
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