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| 11:39.50 | SQUelcher | hi |
| 11:44.59 | SQUelcher | I'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? |
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| 12:01.45 | SQUelcher | got it, endpoint/from_user was missing :) |
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| 13:06.51 | file | https://www.asterisk.org/community/astricon-live/videos for the AstriCon opening remarks/keynote |
| 13:11.25 | sibiria | is ast16 SRV-aware? |
| 13:11.51 | sibiria | or rather, is chan_cip and pjsip |
| 13:12.08 | sibiria | sip* |
| 13:12.22 | Samot | Chan_SIP doesn't use SRV properly. It only pulls the first record. |
| 13:12.26 | file | PJSIP supports NAPTR and SRV |
| 13:12.33 | Samot | PJSIP handles SRV properly |
| 13:12.49 | sibiria | ok, thanks |
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| 14:00.44 | aoeui | n |
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| 18:00.24 | tehgooch | Is 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. |
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| 18:44.21 | file | From username first and then IP, not configurable |
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| 19:38.02 | MLC | how 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? |
| 20:04.12 | MLC | finally found it: context= in queues.conf |
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