| 00:01.47 | TrickkyTyper | lmao |
| 00:01.49 | TrickkyTyper | xD |
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| 00:19.45 | *** 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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| 01:41.37 | jeffspeff | will argparse receive arguments passed to a python AGI script or do I need to use another method to receive the args? |
| 01:42.34 | jeffspeff | I know I could test and try but as this is literally my first agi script, I don't know what I should expect to work or not work. |
| 01:52.12 | drmessano | Geez what a maroon |
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| 03:49.29 | Micc | I need to do some failover testing for a sip peer, how do I have it reject all calls made to a particular peer? I tried setting call-limit=0 and callcounter=0, but it still receives calls. |
| 03:52.55 | Micc | changing the host to something else will do the trick. |
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| 08:37.48 | FarhaadN | hi all , i am using this doc https://wiki.freepbx.org/display/PC/Telephony+Cards+for+Asterisk for for install sangoma card and use with asterisk |
| 08:38.07 | FarhaadN | all is done until wanpipe installation |
| 08:38.41 | FarhaadN | this is error i got : Directory /lib/modules/3.16.0-4-amd64/build/include/linux not found! |
| 08:39.03 | FarhaadN | i am using debian 8 and kernel 3.16.0-4-amd64 |
| 08:40.15 | FarhaadN | can anyone help me to solved problem? |
| 08:42.44 | tuxd00d | FarhaadN: on FreePBX? |
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| 08:45.20 | tuxd00d | Or using a Sangoma card? |
| 08:45.39 | tuxd00d | Your support channel is #freepbx |
| 08:48.26 | FarhaadN | tuxd00d: i want using sangoma card "A104d T1/E1" |
| 08:49.05 | tuxd00d | You will want to ask your question in #freepbx, not #asterisk. |
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| 08:54.27 | tuxd00d | Ø´Ù
ا Ù
Û Ø®ÙاÙÛد سÙاÙات Ø®Ùد را در |
| 08:54.29 | tuxd00d | #freepbx |
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| 09:12.02 | FarhaadN | tuxd00d: ok |
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| 11:34.08 | UncleKiwi | Hi there im thinking about upgrading from asterisk 11 to 13 and i see this PJSIP channel driver is that going to affect my config files ? |
| 11:35.53 | UncleKiwi | regarding the syntax of using sip in version 11 |
| 11:36.31 | file | chan_sip and chan_pjsip are separate |
| 11:36.48 | UncleKiwi | ah ok |
| 11:37.01 | UncleKiwi | so i dont need chan_pjsip ? |
| 11:37.11 | UncleKiwi | if i dont want to use it |
| 11:39.36 | file | correct |
| 11:40.24 | UncleKiwi | what's the intention of PJSIP ? |
| 11:43.04 | sibiria | Newerer(tm), Betterer(tm) |
| 11:43.05 | sibiria | etc. |
| 11:43.29 | UncleKiwi | ok |
| 11:44.10 | file | it's got a design, it's easier to support/extend/optimize |
| 11:44.48 | file | and it uses an existing SIP stack that has seen wide use (you may have even used it without realizing it) |
| 11:44.49 | UncleKiwi | so it would replace chan_sip |
| 11:45.02 | file | it can replace chan_sip, or they can both be used at once |
| 11:45.08 | UncleKiwi | ok |
| 11:47.02 | UncleKiwi | thanks |
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| 13:46.06 | igcewieling | Good morning. I see the assassins have failed. |
| 14:02.48 | Samot | To kill GV projects in Asterisk? Yes, they have. |
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| 19:43.27 | jploh | how much task processors should i have with 274 chan_sip peers? i seem to have encountered a memory leak. but i'm trying to figure out what's causing it. |
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| 22:38.56 | [J]oules | the A20 Digium phones are really good. The ONLY thing so far is the inability to dial on hook without any key pressed. Screen is really nice |
| 23:07.04 | Samot | How does one dial on hook without pressing keys? |
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| 23:16.06 | tuxd00d | thinks about telepathic phones |
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| 23:17.02 | tuxd00d | Perhaps he means auto speakerphone functionally when the keypad is pressed... |
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