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00:22.37 | *** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.27.1 (2019/7/11) 16.4.1 (2019/7/11), Security Only: 15.7.3 (2019/7/11); 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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14:15.28 | synja | hey guys. does anyone have a working jack & stasis extension config? |
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14:58.02 | acovrig | Iâm tinkering with ARI, Iâd like to add a PJSIP header, looking at https://issues.asterisk.org/jira/browse/ASTERISK-26178 it doesnât seem possible from ARI directly, so Iâm trying to dial LOCAL/*33100 instead of PJSIP/100, the *33 adds the PJSIP header and Dial(PJSIP/100), everything seems to work, but then ARI says the channel isnât in statis |
14:58.18 | acovrig | Is there a preferred way for me to show part of my dialplan? |
14:59.28 | acovrig | and is this an #asterisk or #asterisk-dev question? |
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15:49.39 | synja | combining stasis with other types gets tricky acovrig |
15:56.59 | synja | i've been trying to use jack, agi, eagi, chanspy & stasis, etc. to get a stream of the incoming voice on a channel |
15:57.01 | synja | no luck |
15:57.16 | synja | jack is my last resort before i get "physiCal" |
15:57.47 | synja | have ari controlling the channel, with outbound tts working perfectly |
15:58.41 | synja | the documentation really sucks |
15:58.47 | synja | for being open |
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16:29.46 | Samot | synja: Where is this incoming audio being sourced from? |
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17:47.51 | synja | Samot, the channel |
17:48.03 | synja | although i doubt that's the answer you were looking for |
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18:01.26 | nix8n82 | Hi, I'm using Asterisk: The Definitive Guide 5th ed and I am having problems with pjsip realtime. It will register my remote endpoint and I can send a request to it, but it doesn't save contact information in AoR. Any ideas? |
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18:12.32 | nix8n82 | Also it tells me my device is registered |
18:13.16 | nix8n82 | I'm using digital ocean as my Asterisk server and my soft client on my phone is behind a natted network |
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18:41.03 | Samot | synja: No, how is it getting on the channel? |
18:41.20 | Samot | synja: By source I mean how does Asterisk even know it exists? |
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19:29.14 | synja | exten => 1,s,Stasis(app_name) |
19:29.25 | synja | connects via networked computer to ari to control channel |
19:29.31 | synja | idk if it knows about audio input |
19:29.46 | synja | i assumed so, since it IS a pbx capable of recording |
19:30.09 | synja | can you use Answer() with stasis, Samot? |
19:30.29 | synja | i'd like to do something like this: |
19:30.38 | synja | exten => 1,s,Answer |
19:30.48 | synja | exten => 1,s,Jack(c) |
19:31.01 | synja | exten => 1,s,Stasis(app_name) |
19:31.20 | synja | stasis only works if i use the s extension |
19:31.28 | synja | that code above doesn't work, but i'd like it to |
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19:41.54 | sibiria | you must answer the call before invoking the stasis application? |
19:42.28 | sibiria | your dial plan doesn't work because you have extension and priority confused |
19:42.35 | sibiria | and obviously the prio needs to be sequential |
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19:44.21 | synja | sibiria, i don't think so, but if i just use --- |
19:44.26 | synja | <synja> exten => 1,s,Jack(c) |
19:44.26 | synja | <synja> exten => 1,s,Stasis(app_name) |
19:44.28 | synja | it hangs |
19:44.40 | synja | doesn't pick up |
19:44.57 | sibiria | what if you correct the dial plan then? |
19:45.01 | synja | what would be the correct format for it? |
19:45.04 | synja | to what? |
19:45.05 | sibiria | exten => s,1,Answer |
19:45.11 | sibiria | exten => s,2,Jack |
19:45.14 | sibiria | exten => s,3,Stasis |
19:45.18 | synja | coolio |
19:45.21 | synja | i'll try it |
19:45.22 | synja | thanks |
19:45.29 | sibiria | with the correct parameters you want to use, of course |
19:50.13 | synja | it just disconnects |
19:50.27 | synja | if i have ANYTHING Before stasis, it doesn't work |
19:50.36 | sibiria | so why not do just that? |
19:50.39 | sibiria | s,1,Stasis |
19:50.42 | sibiria | s,2,Answer |
19:50.43 | sibiria | s,3,Jack |
19:51.09 | synja | do i need answer at all? |
19:51.18 | synja | ari picks the call up |
19:51.31 | sibiria | you don't need to answer twice, but _something_ has to answer the call |
19:51.45 | sibiria | also, do check the console (verbosely) to see what goes on |
19:52.58 | synja | [2019-07-14 19:51:57] VERBOSE[6937][C-00000007] sig_analog.c: Starting simple switch on 'DAHDI/1-1' |
19:52.59 | synja | [2019-07-14 19:51:58] VERBOSE[6937][C-00000007] pbx.c: Executing [s@alicia:1] Stasis("DAHDI/1-1", "alicia") in new stack |
19:52.59 | synja | [2019-07-14 19:52:02] VERBOSE[6937][C-00000007] file.c: <DAHDI/1-1> Playing 'All-Systems-Repair-This-is-Alicia-How.slin' (language 'en') |
19:52.59 | synja | [2019-07-14 19:52:11] VERBOSE[6937][C-00000007] sig_analog.c: Hanging up on 'DAHDI/1-1' |
19:52.59 | synja | [2019-07-14 19:52:11] VERBOSE[6937][C-00000007] chan_dahdi.c: Hungup 'DAHDI/1-1' |
19:53.06 | synja | nothing about jack |
19:53.10 | synja | heh |
19:53.53 | synja | it's like, it's frozen at stasis |
19:53.57 | synja | nothing can be before, nor after |
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20:00.36 | synja | okay sibiria |
20:00.42 | synja | i got jack not working now |
20:00.45 | synja | thanks a lot |
20:00.57 | synja | it tries to initialize, so now there's something to debug |
20:01.03 | synja | i really do appreciate your help |
20:01.06 | synja | :) |
20:03.13 | sibiria | not sure if i helped at all - i've never used neither jack nor stasis |
20:03.18 | sibiria | the dial plan did look wrong, though |
20:04.03 | synja | your code didn't work either, but something you said must have helped |
20:04.22 | synja | cheers |
20:04.33 | synja | time to get some work done |
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22:01.30 | nix8n82 | I get the following message after my soft phone registers to my public cloud server and after I try to dial it, run pjsip show enpoint myuser, or pjsip show aors. |
22:02.05 | nix8n82 | <PROTECTED> |
22:02.43 | nix8n82 | I'm using pjsip realtime and connecting to mysql through odbc |
22:04.34 | nix8n82 | the contact shows up in ps_contact table but when I run pjsip show endpoints it tells me it's unavailable. I can place calls to the server and hit the dialplan and if I manually add the contact to ps_aors tables I can then call the device. |
22:05.51 | nix8n82 | If anyone has any insight or a good place to restart my troubleshooting, I will be very grateful |
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