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03:23.29 | mbit_brendan | Is 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? |
03:45.05 | wyoung | .o/ |
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04:03.40 | ruben23 | hi 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 |
04:09.43 | mbit_brendan | if 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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04:53.12 | volga629_ | Hello Everyone, is possible configure pjsip transport to reply 200 OK for OPTIONS from upstream proxy ? |
04:54.27 | [TK]D-Fender | volga629_, IIRC it should if you have the appropriate extension in the context that should get hit by the subscriber |
04:54.34 | [TK]D-Fender | Watch your CLI. |
04:55.02 | volga629_ | [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 |
04:55.29 | volga629_ | not sure why endpoint |
04:55.55 | Samot | vprx01 <-- Does this exist? |
04:56.12 | volga629_ | yes this kamailio |
04:56.18 | Samot | No. |
04:56.23 | Samot | On Asterisk. |
04:56.37 | Samot | That is Asterisk telling you there is no PJSIP endpoint found for that request |
04:56.52 | Samot | Does vprx01 exist as a PJSIP endpoint on Asterisk? |
04:56.56 | volga629_ | no |
04:57.01 | Samot | Then there you go |
04:57.35 | volga629_ | or I see I thought kamailio dispatcher can send OPTIONS |
04:57.42 | Samot | It can |
04:57.50 | volga629_ | that the reply |
04:58.10 | Samot | You're sending an OPTIONS to an endpoint/user/device that doesn't exist. |
04:58.18 | Samot | Of course Asterisk should not send back a 200 OK |
04:58.22 | Samot | Because it's not OK. |
04:59.43 | volga629_ | so this modparam("dispatcher", "ds_ping_from", "sip:vprx01@10.30.100.42") will not work |
05:00.12 | Samot | How about you actually show the OPTIONS in a SIP debug |
05:00.29 | Samot | Because now you're just throwing out settings from something that's not even Asterisk. |
05:01.09 | volga629_ | here are some trace |
05:01.10 | volga629_ | https://paste.fedoraproject.org/paste/iZXTbXTb0ZEHb3c92I~QJg |
05:01.16 | volga629_ | asterisk reply 401 |
05:02.34 | Samot | Well that's because Asterisk wants you to auth the request |
05:02.45 | Samot | Which is not something that should happen with an OPTIONS |
05:02.53 | volga629_ | yes based on 401 |
05:02.54 | Samot | How is Asterisk connected to Kamailio? |
05:03.28 | volga629_ | just dispatcher from kamailio and endoints on pjsip use outbound_proxy |
05:03.39 | Samot | Uhm. |
05:03.54 | Samot | That's not how that works. |
05:03.55 | Samot | At all. |
05:04.05 | volga629_ | why ? |
05:04.07 | Samot | Dispatcher is going to send OPTIONS to the server. |
05:04.13 | Samot | Not the endpoints. |
05:04.17 | Samot | You need a TRUNK |
05:04.29 | Samot | Between Asterisk and Kamailio |
05:05.04 | volga629_ | no if I comment out ping staff in kamailio everything works no problem |
05:05.13 | Samot | JFC. |
05:05.18 | Samot | Of course. |
05:05.26 | Samot | Because it stops sending the OPTIONS |
05:05.30 | volga629_ | yes |
05:05.36 | Samot | If you stop sending the OPTIONS you stop the error. |
05:05.40 | Samot | Now listen to me one more time. |
05:05.49 | Samot | You need a trunk, on Asterisk, between Asterisk and Kamailio |
05:05.52 | volga629_ | but you don't need any trunks |
05:05.55 | Samot | YES |
05:06.01 | volga629_ | why ? |
05:06.30 | Samot | So Asterisk handles the requests from Kamailio properly. |
05:06.34 | volga629_ | it just a proxy it not handle anything except endpoints |
05:06.52 | Samot | Dispatcher is going to send the OPTIONS to the SERVER |
05:06.54 | Samot | SERVER |
05:07.17 | Samot | If you put the Asterisk server in the Dispatcher table, it will send the OPTIONS to that IP |
05:07.50 | Samot | If Asterisk doesn't have a trunk for Kamailio's IP, it's going to treat it like an outside request |
05:08.05 | Samot | Having the trunk stops it from trying to AUTH the OPTIONS |
05:09.00 | volga629_ | so pjsip_acl might be helpful |
05:09.11 | Samot | Sure. |
05:09.17 | volga629_ | to allow kamailio ip |
05:10.53 | Samot | Actually, no that won't |
05:11.05 | Samot | Because ACLs are just what can REGISTER |
05:13.23 | volga629_ | yes just red you right it just REGISTER |
05:19.33 | volga629_ | that similar https://marc.info/?l=asterisk-users&m=139710943715460&w=4 |
05:21.50 | volga629_ | thanks Samot |
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10:32.45 | Demon_VoIP | PJSIP transport option async_operations. Tell me what it influences and what it depends on? How to determine the value? |
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14:37.11 | nibbier | I 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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15:08.37 | voipmonk | nods , "You can manifest that experience." |
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15:44.00 | wabbits | does asterisk support multiple registrations for the same user? |
15:45.04 | eric_hill | wabbits, no |
15:45.16 | Samot | Incorrect. |
15:45.22 | Samot | Chan_PJSIP allows for this. |
15:45.45 | eric_hill | That's handy. |
15:45.58 | wabbits | Thanks Samot |
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16:45.08 | seanbright | is 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-Fender | You should .... look at the channel... |
16:53.38 | seanbright | sorry, i wasn't clear. from AMI/dialplan |
16:54.44 | [TK]D-Fender | AMI has a command reference |
16:54.57 | seanbright | yes it does |
16:54.59 | [TK]D-Fender | If you don't see it, it isn't there |
16:55.07 | [TK]D-Fender | Except for the obvious use of COMMAND |
16:55.08 | seanbright | ok, thanks for your help |
16:55.21 | [TK]D-Fender | So yes there is a clear way via AMI |
16:55.33 | [TK]D-Fender | Now if they have a nicer way.... well.. hit up the command list |
16:55.36 | seanbright | is 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 |
16:56.03 | seanbright | ugh, i keep fucking up |
16:56.09 | seanbright | anyone else? is there a way to determine which codecs a chan_sip channel is using via AMI? |
16:56.46 | [TK]D-Fender | Got an issue with the solution I just handed you? |
16:56.53 | seanbright | i don't, no |
16:56.56 | seanbright | why? |
16:57.02 | [TK]D-Fender | You're still asking... |
16:57.08 | seanbright | correct |
16:57.17 | [TK]D-Fender | You ask if there's a qay. I've already given a "yes" |
16:57.18 | seanbright | there may be other more helpful answers |
16:57.42 | Samot | Well it is a channel variable. |
16:57.54 | Samot | That can either be assigned before the call happens |
16:58.00 | Samot | Or grabbed after the channel is up |
16:58.11 | [TK]D-Fender | That sets a preference, and you'd have to actually do it |
16:58.12 | Samot | But you need to know the channel you're getting the details from. |
16:58.13 | seanbright | SIP_CODEC? i didn't know that was set |
16:58.17 | [TK]D-Fender | and ony pertains to an outbound leg. |
16:58.43 | [TK]D-Fender | If 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-Fender | Looking up stuff you forcibly set is rhetorical |
16:59.12 | seanbright | SIP_CODEC is never written to by chan_sip, so that's a no-go |
17:00.33 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+AMI+Actions |
17:00.44 | [TK]D-Fender | Clearly nothing saying "sip channels" explicitly. |
17:01.01 | [TK]D-Fender | So that's a "no". |
17:01.05 | [TK]D-Fender | Which leaves my solution |
17:04.07 | file | the CHANNEL dialplan function allows you to query the underlying format information for the channel |
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17:04.14 | seanbright | audioreadformat |
17:04.17 | seanbright | yeah, i was just looking at that |
17:04.38 | seanbright | that might be a winner |
17:04.40 | seanbright | thanks file! |
17:04.43 | file | native is on the wire, read/write is what the application/whatever wants/provides |
17:16.39 | volga629_ | Hello Everyone, pjsip give me error for outobund_proxy |
17:16.41 | volga629_ | [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 |
17:17.04 | volga629_ | outbound_proxy = 'sip:10.30.100.28:5060\;transport=udp\;lr' |
17:19.12 | seanbright | file: on a 'typical' call - when would the native formats be accurate? after answer? |
17:19.57 | file | yes |
17:20.02 | seanbright | thanks |
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18:02.40 | volga629_ | pjsip realtime mysql outobund_proxy parameter is not parsed correctly |
18:02.42 | volga629_ | <PROTECTED> |
18:03.03 | volga629_ | in database outbound_proxy: "sip:10.30.100.28:5060\;transport=udp\;lr" |
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18:04.30 | volga629_ | look like a bug |
18:07.10 | Samot | sip:10.30.100.28:5060\;lr <-- Have you tried that? |
18:07.30 | volga629_ | yes |
18:07.37 | volga629_ | I tried add "" |
18:07.42 | volga629_ | and without |
18:07.58 | Samot | You saw that I removed something? |
18:08.00 | volga629_ | same :lr or :lr" |
18:08.06 | Samot | Not just the "" which shouldn't be there anyways |
18:08.16 | volga629_ | I so you remove transport |
18:08.21 | volga629_ | removed |
18:08.30 | Samot | And I asked if you tried it with just that? |
18:08.33 | Samot | You said yes |
18:08.41 | volga629_ | yes |
18:08.41 | Samot | But then clearly shown you have not. |
18:08.51 | volga629_ | I tried all variation |
18:09.08 | Samot | So where is the issue? |
18:09.11 | RovingWriter | every possible variation |
18:09.14 | volga629_ | and asterisk cli pjsip show endpoint is set :lr |
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18:14.01 | volga629_ | going open bug report |
18:17.58 | file | it'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 |
18:18.00 | file | should do it |
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18:46.58 | volga629_ | file thanks for answer |
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18:47.06 | volga629_ | but didn't helped |
18:47.07 | volga629_ | <PROTECTED> |
18:47.07 | volga629_ | <PROTECTED> |
18:47.07 | volga629_ | [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 |
18:47.08 | volga629_ | [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 |
18:47.53 | volga629_ | Also I tried remove \ same result |
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18:48.06 | file | don't put \ in front |
18:48.30 | volga629_ | [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 |
18:50.39 | volga629_ | file I reported https://issues.asterisk.org/jira/browse/ASTERISK-27188 |
18:50.42 | file | looks into legacy code |
18:50.43 | file | I Know |
18:51.04 | volga629_ | ok thank you |
18:51.23 | file | oh I know why, it's ^3B |
18:51.28 | file | not %3B |
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18:53.34 | volga629_ | let me try |
18:54.44 | volga629_ | yes that correct under endpoint right now |
18:54.44 | volga629_ | outbound_proxy : sip:10.30.100.28:5060;transport=udp;lr |
18:58.47 | volga629_ | file I though put note in part of docs https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime |
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19:00.30 | file | k |
19:02.25 | volga629_ | I log in, but not sure if I have edit permissions |
19:03.22 | file | you don't, we don't give out edit permissions to everyone - if someone provides useful feedback a few times then we'll grant it |
19:03.29 | file | it helps the wiki from becoming voip-info.org |
19:16.56 | Samot | *snicker* |
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19:39.35 | volga629 | file again thank for help |
19:42.01 | volga629 | Do I need request edit permission to add this note to wiki ? |
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19:47.19 | file | you can leave a comment and it will get looked at |
19:48.41 | volga629 | ok |
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20:56.50 | polysics | hello! question for you all: is there a way to set a global timeout on an originated call? |
20:56.59 | polysics | something like "no matter what, end this call after 2h" |
21:00.30 | polysics | Dial() has TIMEOUT(absolute), not sure AMI originate has something similar |
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22:12.30 | rrittgarn | little 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-Fender | Indeed too late. |
22:20.43 | [TK]D-Fender | he left a while ago |
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