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| 02:32.10 | tompaw | Erm.. anyone using flowroute here? |
| 02:32.32 | tompaw | WTF is a "dialer surcharge" and why are they nicking $80/day from my credit without any explanation? |
| 02:37.01 | drmessano | Did you ask them? |
| 02:38.40 | tompaw | No, they just charged me that for 3 days and "their offices are now closed". |
| 02:38.56 | drmessano | Are you using a robodialer of some sort? |
| 02:38.57 | tompaw | But damn, even google didn't hear about it. It's nowhere in T&C, legals, nada. |
| 02:39.12 | tompaw | Nope, a call center with live agents, no predictive dialing. |
| 02:40.27 | tompaw | I swear to god, sometimes I wonder if it's me becoming dumber with age or is everyone around just plainly stupid. Why can't they simply take my money and provide the service without all this jerking around. |
| 02:41.15 | drmessano | Perhaps there was some behavior on the line that triggered some process |
| 02:41.57 | tompaw | Perhaps they should inform about it in their pricelist or at least send a fcukin email rather than sneakily rob me of my monies. |
| 02:42.14 | drmessano | oooook |
| 02:42.31 | tompaw | If I didn't check the billing statement I would never spot it. |
| 02:48.28 | ChannelZ | mebbe someone haxx0rs your system and made lots of calls |
| 02:49.10 | tompaw | nope, the cdrs match more or less. |
| 02:49.33 | tompaw | I'll give them a buzz on monday, it just frustrates the hell our of me. |
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| 05:28.16 | Elio19 | does regexten not work with realtime sip peers? |
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| 07:27.43 | doolittlework | hi |
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| 19:06.51 | gruetzkopf | hey. i have a asterisk 11.18 box, with a 4-port Euro-BRI card. link to PBX works fine, but we have a special ISDN phone (on another span) which depends on PBX/telco-performed call forwarding |
| 19:06.55 | gruetzkopf | is that supported? |
| 19:07.24 | gruetzkopf | (because we get no reaction from asterisk to FACILITY messages from said phone) |
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| 19:10.38 | WIMPy | gruetzkopf: Depends on the driver used. It's supposed to work with chan_srx and it can also be done for LCR extensions. |
| 19:10.52 | WIMPy | But no support frm Asterisk itself. |
| 19:13.36 | gruetzkopf | no luck for DAHDI? |
| 19:14.14 | gruetzkopf | its kind of a negative-budget project |
| 19:14.52 | WIMPy | no |
| 19:15.16 | WIMPy | Negative budget? How does that work? |
| 19:15.48 | gruetzkopf | any good way to directly send raw data to a dahdi d-channel, i could just hand-assemble a semi-correct response) |
| 19:16.06 | WIMPy | Nope |
| 19:18.46 | gruetzkopf | last way out: our PBX supports these messages (siemens hicom 150E), but the phone complains about the S0-powersaving. i could wire both the phone and a spare S0 port on the isdn-card to the same port of the PBX. can i get asterisk/dahdi/whatever to keep the l1 powersaving from happening (without occupying a b-channel) |
| 19:20.07 | WIMPy | I'm not sure about dahdi. But LCR can. Why do you need it to be permanently active? |
| 19:22.13 | gruetzkopf | the phone is not exactly Euro-isdn compliant, it is more of a special S0 phone made for a variant of the DMS100 (which never turns the port into idle), so when that happens the phone wants a reboot (which takes 5+ minutes) after every call |
| 19:22.20 | WIMPy | And why do you need real CF? Is it somethign special or just about user experience? |
| 19:22.34 | WIMPy | Ouch! |
| 19:22.55 | gruetzkopf | the phone checks the line by setting up call-forward to itself and dialing |
| 19:23.37 | Ice_Strike | o/ |
| 19:23.43 | gruetzkopf | also, the phone wont self-enable s0 clocking |
| 19:24.20 | WIMPy | Maybe you should get another phone? |
| 19:27.46 | gruetzkopf | the point is to get this specific model working (for museum use |
| 19:28.30 | WIMPy | Then I'd try it with LCR. It might work there without modifications. |
| 19:32.19 | gruetzkopf | okay, thanks for the hints, i'll read some manpages then :) |
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| 20:27.28 | babak | Hi ,Realtime is just for storing configuration files in DB or it is possible to save registration information in it and share it between multiple Asterisk ? |
| 20:31.49 | cyford33 | configuration settings |
| 20:33.37 | cyford33 | but if both servers are using the same database, they can be configuired to authenticate on one server and pass calls to all others,.. especially if you have a sip proxy |
| 20:38.08 | babak | cyford33: thx , I sip clients come in load share on 2 Asterisk server I want to know registered IP to route call |
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| 20:39.20 | babak | 2 Asterisk have access to database so can authenticate |
| 20:40.15 | cyford33 | i have done this before using opensip |
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| 20:42.51 | babak | cyford33: it seems if realtime can have sip registered information with just 2 Asterisk server we can do same |
| 20:49.36 | cyford33 | wouldnt be as reliable |
| 20:50.33 | cyford33 | is mysql on a third server? |
| 20:51.16 | cyford33 | both asterisk servers would be trying to expire sessions? |
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| 20:52.57 | babak | cyford33: I want to run mysql in 2 Asterisk servers Master/Master |
| 20:56.58 | cyford33 | but why.. they both hold the same authentication info.. if one server is out it should be able use the other server without any issues.. just pass reregistration if needed |
| 20:58.02 | cyford33 | are both server sharing a virtual ip , or is there a proxy |
| 20:58.11 | babak | no |
| 20:58.31 | babak | sip client use DNS SRV NAPTr |
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| 20:59.25 | babak | and loadshare and fail over is done with DNS help |
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