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| 00:20.04 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0-rc1 (2018/07/03), Standard: 15.5.0-rc1 (2018/07/03); 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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| 06:46.20 | wyoung | Maliuta_: heh, nah, it looked like you were being that guy (the guy I described before you commented :) ) |
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| 07:56.02 | eqw | I see via sniffer that voip udp traffic goes directly between phones. Is it possible to make it go via server? |
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| 08:32.36 | Maliuta_ | disable reinvite |
| 08:33.40 | Maliuta_ | the version of * you're running may effect what the actual directive is |
| 08:34.45 | Maliuta_ | directmedia=no and directrtpsetup=no |
| 08:43.30 | Maliuta_ | wyoung: well that was the point of that bit of sarcasm :P Notice the yoda like syntax? :D |
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| 09:30.58 | blihtar_ | some one from Bosnia and Hercegovina |
| 09:31.01 | blihtar_ | ? |
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| 10:15.51 | blihtar_ | anyone interestet to test sip trunk between croatia and bosnia and hercegovina |
| 10:27.52 | Samot | No. |
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| 11:10.45 | Maliuta_ | bit too far to walk from the southern hemisphere |
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| 14:28.06 | Chrisbur2 | Good mornin guys |
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| 17:15.15 | n3t | Hi! I'm a total newbie. I try to make Asterisk to handle incoming calls from my SIP service provider. Are there any newbie-friendly resource other than Asterisk wiki about PJSIP? |
| 17:15.29 | n3t | I find it hard to write and understand pjsip.conf. |
| 17:15.58 | file | have you tried wizards? |
| 17:16.52 | n3t | Not yet. I'll check it out, thanks. |
| 17:18.16 | file | https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Wizard |
| 17:19.32 | n3t | That's what I'm reading right now :) |
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| 18:29.16 | forgotmynick | hello what do you call {{ something }} ? |
| 18:31.58 | forgotmynick | i have a trunk, the inbound route changes the cid to "(Trunk) XXXX-XXXX-XXXX". Inbound calls go to an IVR which can go to either a sales or support queue. I want to edit the cid so extensions display "(Department) SALES XXXX-XXXX-XXXX". I know what I have to edit but I don't know which has the data. Is it something like {{ callerid }} ? |
| 18:32.19 | forgotmynick | if i know what the terminology for {{ }} is i can search for it myself. any help would be greatly appreciated |
| 18:33.37 | [TK]D-Fender | SetCallerID <- |
| 18:33.41 | [TK]D-Fender | this is what lets you change the CID |
| 18:35.21 | forgotmynick | i know but i need the {{}} so I can append to the existing CID |
| 18:37.20 | forgotmynick | ah yes it's called variables! https://www.voip-info.org/asterisk-variables/ |
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| 22:30.25 | wyoung | .o/ |
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| 22:39.53 | elvisthedj | Hello - is there anywhere to see output as to why a function is failing to load in func_odbc.conf? I'm referencing a simple function in my dialplan and it's telling me it's not registered. Not sure where to look for what the error is. The function is [TEST] and I'm referencing ODBC_TEST in the dialplan. I see that the dsn is connected with ODBC show (Asterisk 11) |
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| 22:53.15 | elvisthedj | Even the sample custom functions report not registered, so I assume it's just not loading func_odbc.conf |
| 23:23.47 | elvisthedj | ah. odbc was loaded, but the func wasn't. :) |
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