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| 09:26.38 | muchwakar | I have been using a django api for realtime peers storage successfully. I intend to secure the api using Token Authorization but I can't seem to find documentation on configuring asterisk accordingly. I am aware of the curl_opts function but that seems only useful in the dialplan and not in extconfig |
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| 12:13.46 | LunaLovegood | Is there a way to do a "sip set debug on" but only for a specific console (asterisk -r) ? |
| 12:14.02 | LunaLovegood | Cause right now it dumps SIP packets on all consoles. |
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| 16:23.02 | Kobaz | LunaLovegood: don't believe you had an option for that |
| 16:23.17 | Kobaz | anyone have any recommendations for a pc/bluetooth headset |
| 16:23.38 | Kobaz | never really got into these... do they have like an 'answer call' button on them? |
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| 16:51.31 | igcewieling | Kobaz: the "digital" (Electronic Switchhook) ones can have an answer call button. |
| 16:57.25 | *** join/#asterisk zuwuko2 (be8ddee9@gateway/web/freenode/ip.190.141.222.233) |
| 16:57.56 | zuwuko2 | Can I transfer incoming calls? I have only been able to transfer calls I place. |
| 16:59.54 | [TK]D-Fender | Every call is just a call |
| 17:00.29 | zuwuko2 | exten => 101,1,Dial(PJSIP/zuwuko1,30,T) |
| 17:00.40 | zuwuko2 | exten => 102,1,Dial(PJSIP/zuwuko2,30,T) |
| 17:01.00 | zuwuko2 | the *2 transfer works only on the person that initiated the call.. |
| 17:02.55 | [TK]D-Fender | What are you using for phones? |
| 17:03.14 | zuwuko2 | testing with android sip phones |
| 17:03.35 | [TK]D-Fender | Which I'm now assuming don't have basic SIP transfers as a feature... |
| 17:03.42 | [TK]D-Fender | "core show application dial" <----- |
| 17:04.00 | zuwuko2 | Both work if they initiate the call. |
| 17:05.20 | [TK]D-Fender | "core show application dial" <----- |
| 17:05.27 | zuwuko2 | reading through it now |
| 17:06.28 | zuwuko2 | ahh, theres a difference etween T and t |
| 17:06.37 | zuwuko2 | let me try, thank you for pointing me in the right direction |
| 17:08.03 | [TK]D-Fender | You're welcome... |
| 17:08.45 | zuwuko2 | Works like charm |
| 17:10.26 | zuwuko2 | when dialing from an Analog phone (connected to a linksys PAP2) it waits like 10 seconds before dialing. any idea? |
| 17:11.14 | [TK]D-Fender | taht device has it's own timeout rules before deciding when to actually send a call to your server |
| 17:11.14 | Penguin | [TK]D-Fender: its |
| 17:11.28 | zuwuko2 | the PAP2? |
| 17:12.28 | [TK]D-Fender | yes |
| 17:14.10 | zuwuko2 | "Interdigit Long Timer:"? |
| 17:14.21 | zuwuko2 | its 10 now. i'll test with lower and see. |
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| 17:29.18 | zuwuko2 | Thanks that worked. |
| 17:29.36 | zuwuko2 | Anyone using a GSM gateway? |
| 17:30.11 | [TK]D-Fender | heads to a meeting |
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| 18:54.21 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0 (2018/10/09), Security Only: 15.6.1 (2018/09/20); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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