IRC log for #asterisk on 20170808

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03:23.29mbit_brendanIs there a way to execute dialplan when a device subscribes to a hint? I would like to have a hint based on a database key when a device subscibes it checks that key and sets the hint based on the key can that be done?
03:45.05wyoung.o/
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04:03.40ruben23hi there guys any idea i got a voicemail but when i tried to record one, the output is empty no audio is being recorded at all, any suggestion
04:09.43mbit_brendanif it's not possible to run dialplan on subscribe how would you set the state of a custom hint on reboot/asterisk restart?
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04:53.12volga629_Hello Everyone,  is possible configure pjsip transport to reply 200 OK  for OPTIONS from upstream proxy ?
04:54.27[TK]D-Fendervolga629_, IIRC it should if you have the appropriate extension in the context that should get hit by the subscriber
04:54.34[TK]D-FenderWatch your CLI.
04:55.02volga629_[Aug  7 23:54:51] NOTICE[26944]: res_pjsip/pjsip_distributor.c:368 log_unidentified_request: Request from '<sip:vprx01@10.30.100.42>' failed for '10.30.100.42:5060' (callid: 06c84adb04aef23a-19480@10.30.100.42) - No matching endpoint found
04:55.29volga629_not sure why endpoint
04:55.55Samotvprx01 <-- Does this exist?
04:56.12volga629_yes this kamailio
04:56.18SamotNo.
04:56.23SamotOn Asterisk.
04:56.37SamotThat is Asterisk telling you there is no PJSIP endpoint found for that request
04:56.52SamotDoes vprx01 exist as a PJSIP endpoint on Asterisk?
04:56.56volga629_no
04:57.01SamotThen there you go
04:57.35volga629_or I see I thought kamailio dispatcher can send OPTIONS
04:57.42SamotIt can
04:57.50volga629_that the reply
04:58.10SamotYou're sending an OPTIONS to an endpoint/user/device that doesn't exist.
04:58.18SamotOf course Asterisk should not send back a 200 OK
04:58.22SamotBecause it's not OK.
04:59.43volga629_so this modparam("dispatcher", "ds_ping_from", "sip:vprx01@10.30.100.42") will not work
05:00.12SamotHow about you actually show the OPTIONS in a SIP debug
05:00.29SamotBecause now you're just throwing out settings from something that's not even Asterisk.
05:01.09volga629_here are some trace
05:01.10volga629_https://paste.fedoraproject.org/paste/iZXTbXTb0ZEHb3c92I~QJg
05:01.16volga629_asterisk reply 401
05:02.34SamotWell that's because Asterisk wants you to auth the request
05:02.45SamotWhich is not something that should happen with an OPTIONS
05:02.53volga629_yes based on 401
05:02.54SamotHow is Asterisk connected to Kamailio?
05:03.28volga629_just dispatcher from kamailio and endoints on pjsip use outbound_proxy
05:03.39SamotUhm.
05:03.54SamotThat's not how that works.
05:03.55SamotAt all.
05:04.05volga629_why ?
05:04.07SamotDispatcher is going to send OPTIONS to the server.
05:04.13SamotNot the endpoints.
05:04.17SamotYou need a TRUNK
05:04.29SamotBetween Asterisk and Kamailio
05:05.04volga629_no if I comment out ping staff in kamailio everything works no problem
05:05.13SamotJFC.
05:05.18SamotOf course.
05:05.26SamotBecause it stops sending the OPTIONS
05:05.30volga629_yes
05:05.36SamotIf you stop sending the OPTIONS you stop the error.
05:05.40SamotNow listen to me one more time.
05:05.49SamotYou need a trunk, on Asterisk, between Asterisk and Kamailio
05:05.52volga629_but you don't need any trunks
05:05.55SamotYES
05:06.01volga629_why ?
05:06.30SamotSo Asterisk handles the requests from Kamailio properly.
05:06.34volga629_it just a proxy it not handle anything except endpoints
05:06.52SamotDispatcher is going to send the OPTIONS to the SERVER
05:06.54SamotSERVER
05:07.17SamotIf you put the Asterisk server in the Dispatcher table, it will send the OPTIONS to that IP
05:07.50SamotIf Asterisk doesn't have a trunk for Kamailio's IP, it's going to treat it like an outside request
05:08.05SamotHaving the trunk stops it from trying to AUTH the OPTIONS
05:09.00volga629_so pjsip_acl might be helpful
05:09.11SamotSure.
05:09.17volga629_to allow kamailio ip
05:10.53SamotActually, no that won't
05:11.05SamotBecause ACLs are just what can REGISTER
05:13.23volga629_yes just red you right it just REGISTER
05:19.33volga629_that similar https://marc.info/?l=asterisk-users&m=139710943715460&w=4
05:21.50volga629_thanks Samot
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10:32.45Demon_VoIPPJSIP transport option async_operations. Tell me what it influences and what it depends on? How to determine the value?
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14:37.11nibbierI have a callgroup that rings several phones. when no one is picking up i want anyone else to actively take this call (*8 or such) is that possible, while its ringing?
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15:08.37voipmonknods , "You can manifest that experience."
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15:44.00wabbitsdoes asterisk support multiple registrations for the same user?
15:45.04eric_hillwabbits, no
15:45.16SamotIncorrect.
15:45.22SamotChan_PJSIP allows for this.
15:45.45eric_hillThat's handy.
15:45.58wabbitsThanks Samot
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16:45.08seanbrightis there a way to determine which codecs a chan_sip channel are using?
16:53.15[TK]D-Fender"sip show channel......"
16:53.27[TK]D-FenderYou should .... look at the channel...
16:53.38seanbrightsorry, i wasn't clear. from AMI/dialplan
16:54.44[TK]D-FenderAMI has a command reference
16:54.57seanbrightyes it does
16:54.59[TK]D-FenderIf you don't see it, it isn't there
16:55.07[TK]D-FenderExcept for the obvious use of COMMAND
16:55.08seanbrightok, thanks for your help
16:55.21[TK]D-FenderSo yes there is a clear way via AMI
16:55.33[TK]D-FenderNow if they have a nicer way.... well.. hit up the command list
16:55.36seanbrightis there a way to determine which codecs a chan_sip channel is using via AMI?
16:55.52[TK]D-Fender<[TK]D-Fender> Except for the obvious use of COMMAND
16:55.57[TK]D-Fender<[TK]D-Fender> So yes there is a clear way via AMI
16:56.03seanbrightugh, i keep fucking up
16:56.09seanbrightanyone else? is there a way to determine which codecs a chan_sip channel is using via AMI?
16:56.46[TK]D-FenderGot an issue with the solution I just handed you?
16:56.53seanbrighti don't, no
16:56.56seanbrightwhy?
16:57.02[TK]D-FenderYou're still asking...
16:57.08seanbrightcorrect
16:57.17[TK]D-FenderYou ask if there's a qay.  I've already given a "yes"
16:57.18seanbrightthere may be other more helpful answers
16:57.42SamotWell it is a channel variable.
16:57.54SamotThat can either be assigned before the call happens
16:58.00SamotOr grabbed after the channel is up
16:58.11[TK]D-FenderThat sets a preference, and you'd have to actually do it
16:58.12SamotBut you need to know the channel you're getting the details from.
16:58.13seanbrightSIP_CODEC? i didn't know that was set
16:58.17[TK]D-Fenderand ony pertains to an outbound leg.
16:58.43[TK]D-FenderIf you're setting it that way you shouldn't need to look it up anyway.. you're setting it explicitly
16:58.57[TK]D-FenderLooking up stuff you forcibly set is rhetorical
16:59.12seanbrightSIP_CODEC is never written to by chan_sip, so that's a no-go
17:00.33[TK]D-Fenderhttps://wiki.asterisk.org/wiki/display/AST/Asterisk+14+AMI+Actions
17:00.44[TK]D-FenderClearly nothing saying "sip channels" explicitly.
17:01.01[TK]D-FenderSo that's a "no".
17:01.05[TK]D-FenderWhich leaves my solution
17:04.07filethe CHANNEL dialplan function allows you to query the underlying format information for the channel
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17:04.14seanbrightaudioreadformat
17:04.17seanbrightyeah, i was just looking at that
17:04.38seanbrightthat might be a winner
17:04.40seanbrightthanks file!
17:04.43filenative is on the wire, read/write is what the application/whatever wants/provides
17:16.39volga629_Hello Everyone, pjsip give me error for  outobund_proxy
17:16.41volga629_[Aug  8 12:15:26] ERROR[5231]: res_pjsip.c:2959 create_out_of_dialog_request: Unable to apply outbound proxy on request NOTIFY to endpoint 8422-10 as outbound proxy URI 'lr' is not valid
17:17.04volga629_outbound_proxy = 'sip:10.30.100.28:5060\;transport=udp\;lr'
17:19.12seanbrightfile: on a 'typical' call - when would the native formats be accurate? after answer?
17:19.57fileyes
17:20.02seanbrightthanks
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18:02.40volga629_pjsip realtime mysql outobund_proxy parameter is not parsed correctly
18:02.42volga629_<PROTECTED>
18:03.03volga629_in database  outbound_proxy: "sip:10.30.100.28:5060\;transport=udp\;lr"
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18:04.30volga629_look like a  bug
18:07.10Samotsip:10.30.100.28:5060\;lr <-- Have you tried that?
18:07.30volga629_yes
18:07.37volga629_I tried add ""
18:07.42volga629_and without
18:07.58SamotYou saw that I removed something?
18:08.00volga629_same :lr or :lr"
18:08.06SamotNot just the "" which shouldn't be there anyways
18:08.16volga629_I so you remove transport
18:08.21volga629_removed
18:08.30SamotAnd I asked if you tried it with just that?
18:08.33SamotYou said yes
18:08.41volga629_yes
18:08.41SamotBut then clearly shown you have not.
18:08.51volga629_I tried all variation
18:09.08SamotSo where is the issue?
18:09.11RovingWriterevery possible variation
18:09.14volga629_and asterisk cli pjsip show endpoint is set :lr
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18:14.01volga629_going open bug report
18:17.58fileit's a limitation of realtime itself, it uses ; to separate multiple entries for an option so the hex has to be used - sip:10.30.100.28:5060%3Blr
18:18.00fileshould do it
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18:46.58volga629_file thanks for answer
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18:47.06volga629_but didn't helped
18:47.07volga629_<PROTECTED>
18:47.07volga629_<PROTECTED>
18:47.07volga629_[Aug  8 13:40:35] ERROR[10067]: res_pjsip.c:2959 create_out_of_dialog_request: Unable to apply outbound proxy on request NOTIFY to endpoint 8422-10 as outbound proxy URI 'sip:10.30.100.28:5060\%3Btransport=udp\%3Blr' is not valid
18:47.08volga629_[Aug  8 13:40:35] WARNING[10067]: res_pjsip_mwi.c:379 send_unsolicited_mwi_notify_to_contact: Unable to create unsolicited NOTIFY request to endpoint 8422-10 URI sip:8422-10@10.30.100.42:5060;rinstance=8B7EFC83
18:47.53volga629_Also I tried remove \ same result
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18:48.06filedon't put \ in front
18:48.30volga629_[Aug  8 13:45:24] ERROR[10230]: res_pjsip.c:2959 create_out_of_dialog_request: Unable to apply outbound proxy on request NOTIFY to endpoint 8422-10 as outbound proxy URI 'sip:10.30.100.28:5060%3Btransport=udp%3Blr' is not valid
18:50.39volga629_file I reported https://issues.asterisk.org/jira/browse/ASTERISK-27188
18:50.42filelooks into legacy code
18:50.43fileI Know
18:51.04volga629_ok thank you
18:51.23fileoh I know why, it's ^3B
18:51.28filenot %3B
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18:53.34volga629_let me try
18:54.44volga629_yes that correct under endpoint right now
18:54.44volga629_outbound_proxy                     : sip:10.30.100.28:5060;transport=udp;lr
18:58.47volga629_file I though put note in part of docs https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime
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19:00.30filek
19:02.25volga629_I log in, but not sure if I have edit permissions
19:03.22fileyou don't, we don't give out edit permissions to everyone - if someone provides useful feedback a few times then we'll grant it
19:03.29fileit helps the wiki from becoming voip-info.org
19:16.56Samot*snicker*
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19:39.35volga629file again thank for help
19:42.01volga629Do  I need request edit permission to add this note to wiki ?
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19:47.19fileyou can leave a comment and it will get looked at
19:48.41volga629ok
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20:56.33*** join/#asterisk polysics (~polysics@104.244.241.142)
20:56.50polysicshello! question for you all: is there a way to set a global timeout on an originated call?
20:56.59polysicssomething like "no matter what, end this call after 2h"
21:00.30polysicsDial() has TIMEOUT(absolute), not sure AMI originate has something similar
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22:12.30rrittgarnlittle late, but polysics, you could always originate to a local channel that has that in the Dial argument
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22:20.40[TK]D-FenderIndeed too late.
22:20.43[TK]D-Fenderhe left a while ago
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