| 00:01.13 | MycroftMk2 | Ok, I've got the custom userfield enabled now |
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| 00:02.45 | MycroftMk2 | So if I edit extensions.conf and add this to the [from-sip-external] section: |
| 00:02.48 | MycroftMk2 | exten => _X.,1,Set(CDR(userfield)=${CHANNEL(recvip)}) |
| 00:03.09 | MycroftMk2 | and restart, that should start recording the IPs of rejected anonymous calls, yes? |
| 00:06.10 | [TK]D-Fender | no |
| 00:06.20 | [TK]D-Fender | If the hits the dialplan then it is not rejected |
| 00:06.35 | MycroftMk2 | Ok, so this will record all IPs then |
| 00:06.47 | [TK]D-Fender | Perhaps you should be clearer about what intend to DO with this. |
| 00:06.58 | [TK]D-Fender | The records the IP of calls you ACCEPT into the dialplan |
| 00:07.31 | MycroftMk2 | Which is what I want. |
| 00:11.15 | [TK]D-Fender | What do you expect to do with it afterwards? |
| 00:12.05 | MycroftMk2 | Review the ip addresses associated with the calls |
| 00:19.00 | [TK]D-Fender | OR ... you could jue the securinty logging * 10 already offers and just reject the calls outright |
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| 01:31.34 | brodiem | Hi, wondering if anyone has a recommendation on a decent FXO device for a single POTS that I can gateway over SIP? I am currently using a TDM400P and just looking to get rid of needing a dedicated old PC for the PCI card? |
| 01:36.39 | [TK]D-Fender | Linksys SPA-3102 |
| 01:36.51 | [TK]D-Fender | heads out for the night |
| 01:38.25 | brodiem | I figured that would be the popular choice.. might try it out for the price, just never a big fan of the Sipura/Linksys/Cisco adapters in general |
| 01:39.27 | [TK]D-Fender | Obihai Obi110 is also apprently fairly decent |
| 01:39.38 | [TK]D-Fender | (verify which one has the FXO.. I think that was it) |
| 01:39.48 | [TK]D-Fender | actually steps out... |
| 01:39.49 | brodiem | thanks I'll check those out |
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| 01:54.46 | JerJer | meep meep |
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| 02:34.04 | Kattyroo | evening |
| 02:46.33 | Kattyroo | Nugget: jealous. |
| 02:56.11 | Nugget | hi hi :) |
| 03:03.38 | Kattyroo | get all settled in? |
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| 03:23.47 | Nugget | yep |
| 03:23.56 | Kattyroo | yay |
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| 06:40.15 | r00f | is there any way to limit call duartion on sip trunk, instead of putting the limit into all Dial commands? |
| 06:44.50 | [TK]D-Fender | no |
| 06:45.49 | r00f | bad luck |
| 06:53.06 | [TK]D-Fender | heads to bed |
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| 10:24.23 | Marquel | morning |
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| 14:32.00 | puzzled | hi |
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| 20:45.05 | istevenmon | hello guys, in my country there is different movile operators, in old days each one had their unique starting numbers but these days goverment approved number portability so right now i can take my operator number A to opertator number B. So my question is there any asterisk module i could use to identify the number to route it to a specific trunk? |
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