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| 01:53.52 | ssieb | The Ubiquiti APs seem pretty good, but then I run openwrt on them. |
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| 01:57.25 | drmessano | Heh |
| 01:58.26 | drmessano | I dont even remotely see the point |
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| 02:02.15 | Samot | OK, that was new. |
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| 08:36.55 | Cleron | hello anyone awake? |
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| 15:22.45 | drakonan | hey anyond around |
| 15:22.50 | drakonan | anyone* |
| 15:25.05 | drakonan | hey anyone around i was wondering if there is a way to have basically like a hundred numbers you can assign to specific people 1ea and have the number merge to your cell phone with basically like an authentication match so that only the given number will receive from a given number |
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| 15:54.13 | [TK]D-Fender | drakonan, It's your dialplan you can code whatever logic you want into it. |
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| 17:10.21 | igcewieling | Remember: cellphones have voicemail too. |
| 17:14.13 | bluejaypop | hello people, I'm managing call files and some agi scripts for some basic application, is it possible to add a variable in the archive call file after whether is answered or not ? like for example I want to add a variable if the IVR from the survey was answered or not. If this is not possible, I'll go with a database record i think so. |
| 17:15.50 | igcewieling | bluejaypop: I can't think of a way, but others might know of a way. |
| 17:16.03 | bluejaypop | igcewieling, thanks :) |
| 17:17.17 | Samot | It's called editing the actual file. |
| 17:17.45 | Samot | 1) it needs to be archived so it ends up with the outgoing _sent folder. |
| 17:18.00 | Samot | 2) You'll need to edit the file. |
| 17:18.06 | igcewieling | bluejaypop: I set a variable in the original call file, then use that in the dialplan as a key for adding info into a database. |
| 17:18.35 | bluejaypop | Samot, yep, i thought about it too. |
| 17:19.01 | bluejaypop | igcewieling, that's what I was thinking as an "easy" way. |
| 17:19.32 | bluejaypop | thank you both guys, I'll go for the database field. |
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| 20:15.07 | Cleron | Having a problem setting up a trunk via freepbx.... Executing [h@default:1] AGI("SIP/201-00000000", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18-----CHANUNAVAIL---------------)") in new stack |
| 20:15.14 | Cleron | can anyone help? |
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| 23:58.49 | Casper | hi there, not quite asterisk but meh... I have a problem with my voip provider. Outgoing audio drop midcall. Beside a provider issue, what could be the issue on my isp or my end? I'm using TLS UDP... |