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| 16:16.35 | gtTuna | Can I mix technologies in Dial() ?? Having a weird issue where if I try to do Dial(PJSIP/268&SIP/+16143330000@sbc.domain.com) , it will only dial the 1st endpoint in the list for 268. Dial(PJSIP/268) works fine though |
| 16:17.05 | igcewieling | yes, you can. |
| 16:17.34 | igcewieling | does Dial(SIP/+16143330000@sbc.domain.com) work? |
| 16:18.07 | igcewieling | However, mixing chan_sip with chan_pjsip is just silly. |
| 16:18.16 | gtTuna | It does -- also probably should have had more coffee while I was trying to sanitize that |
| 16:18.39 | gtTuna | Yeah, Im going to try making that SBC trunk PJSIP instead |
| 16:19.12 | igcewieling | Why not setup an endpoint for it instead of dialing my hostname. |
| 16:19.20 | igcewieling | s/my/by/ |
| 16:19.49 | gtTuna | hmmm |
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| 16:36.35 | gtTuna | didn't seem to make a difference |
| 16:38.25 | igcewieling | as long as you are not using FreePBX pastebin the CLI output of a failed call. |
| 17:01.55 | gtTuna | sadly, I am using FreePBX in this scenario, so it's probably something going on there |
| 17:02.29 | gtTuna | but yeah, it's only actually sending INVITEs out to the first listed endpoint under that specific AOR |
| 17:02.39 | gtTuna | so, sorry for wasting your time :) |
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| 18:51.26 | Samot | You are dealing wrong |
| 18:51.33 | Samot | Dialing |
| 18:52.37 | Samot | gtTuna: You need to use PJSIP_DIAL_CONTACTS |
| 18:53.00 | Samot | That will get all the contacts in an AOR and dial them together. |
| 18:53.41 | gtTuna | awesome, thanks Samot |
| 18:53.56 | Samot | ${PJSIP_DIAL_CONTACTS(PJSIP/268)} |
| 18:54.30 | Samot | Sorry no PJSIP/ |
| 18:54.51 | gtTuna | yeah, I'll have to setup some wonky custom dialplan entries to get around what I'm actually trying to accomplish in FreePBX, but that should get me down the right path at least |
| 18:55.10 | Samot | well that is how FreePBX Dialplan does it |
| 18:55.24 | Samot | Ir uses that function |
| 18:55.26 | Samot | It |
| 18:56.18 | gtTuna | well, I was hoping I could just add an additional target with "&" in their "dial" settings on the ext level, but seemingly it must parse it in a fashion that it will only use dial_contacts if "PJSIP/xxx" is the only entry there |
| 18:56.22 | gtTuna | based on my testing |
| 18:56.23 | Samot | And it seems you are doing custom dialplan |
| 18:56.35 | Samot | No |
| 18:57.11 | Samot | Dial(PJSIP/268) will dial them first contact |
| 18:57.13 | Samot | The |
| 18:57.19 | Samot | Fing phone |
| 19:07.46 | Kobaz | Does pjsip sip tls support SNI ? Is that a thing? |
| 19:09.10 | Samot | Does SIP TLS support SIN? |
| 19:09.32 | Samot | Do the RFCs for SIP TLS say this should be a supported thing? |
| 19:10.07 | Kobaz | That's what I'm wondering, is that even a thing |
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| 19:14.29 | Samot | Would have to look |
| 19:14.39 | Kobaz | google-fu's not showing much |
| 19:15.06 | Samot | I have not a need for it, so havent looked lately |
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| 19:47.35 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.2.2, 16.16.2 (2021/03/04) Final Bugfix: 13.38.2, 17.9.3 (2021/03/04); 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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