00:00.22 | lvlinux | ok, i was under the impression (from how it was worded in the rather short section in the docs) that the SLA app auto-did that and then the phone would be dialing on the PSTN circuit directly. |
00:00.57 | lvlinux | i'll try to mess with that context and see what I can get. |
00:01.09 | [TK]D-Fender | app = dialplan = you have to have it there |
00:01.59 | lvlinux | hehe yep that's the way I'm used to doing it---then I got into this key system nightmare and I'm all confused lol |
00:02.57 | [TK]D-Fender | The nightmare is only beginning... |
00:03.11 | lvlinux | hehe that's what I'm afraid of... |
00:03.37 | lvlinux | thankfully the end result is only a simple three PSTN line key system with three phones. |
00:04.04 | lvlinux | not that there's anything "simple" about it lol |
00:09.23 | lvlinux | ok i'm not getting it or something |
00:10.10 | WIMPy | core set verbose 3 |
00:10.15 | lvlinux | in my sla_stations context, on the extension that the line key speedials, i added a second priority that goes straight to VoicemailMain(101@default) and it doesn't work. |
00:10.18 | lvlinux | ok |
00:10.21 | WIMPy | and look at what's happening. |
00:13.03 | lvlinux | i already had it set verbose---it says that its executing [station1@sla_stations:1] SLAStation("SIP/b101-0000000b", "station1") |
00:13.11 | lvlinux | and then gives Auto fallthrough, channel 'SIP/b101-0000000b' status is 'UNKNOWN' |
00:13.50 | WIMPy | So that application exits and your dialplan doesn;t continue. |
00:14.24 | lvlinux | yes---so obviously that's why it's not going on to the next priority |
00:14.35 | lvlinux | (which is the voicemailmain) |
00:14.50 | WIMPy | No, there is no next priority. |
00:15.02 | lvlinux | so how can i diagnose why it's exiting---there's no "sla set debug on" |
00:15.15 | [TK]D-Fender | you ran out of dialplan |
00:15.23 | WIMPy | NFI |
00:15.27 | [TK]D-Fender | [19:55][TK]D-FenderWhich * throws out if a call is accepted to the dialplan but nothing answers it or passes on any other status |
00:15.37 | lvlinux | what's NFI? |
00:15.45 | [TK]D-Fender | No &^#$ing Idea |
00:15.59 | lvlinux | oh oh |
00:16.00 | [TK]D-Fender | lvlinux: You aren't showing us anything useful. |
00:16.25 | [TK]D-Fender | If you want an autopsy, give us a body |
00:16.30 | lvlinux | ok i'll pastebin some stuff |
00:16.32 | lvlinux | :-) |
00:17.09 | lvlinux | i guess i figured i'd ask questions first to keep from wasting your time---but now i realize that that's wasting your time lol hehe |
00:28.04 | lvlinux | ok here's sla.conf: http://pastebin.com/LuXLnQVR |
00:29.06 | lvlinux | sip.conf: http://pastebin.com/Q591bVek |
00:29.35 | [TK]D-Fender | Still means nothing |
00:29.46 | lvlinux | extension.conf: http://pastebin.com/r86iM4Vx |
00:29.49 | [TK]D-Fender | You don't seem to understand the very basics |
00:29.56 | [TK]D-Fender | You've been told you have a DIALPLAN ERROR |
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00:30.02 | [TK]D-Fender | that is EXTENSIONS.CONF |
00:30.14 | [TK]D-Fender | And you aren't showing us the CALL |
00:30.20 | lvlinux | yes well there's that---i just put the other two just in case you wanted to verify the contexts were correct. |
00:30.36 | [TK]D-Fender | Showing us raw configs also proves precisely nothing about what the phone is actually DIALING |
00:30.49 | lvlinux | k i'll do the * output |
00:31.53 | lvlinux | * output when I press the line 1 button on the phone: http://pastebin.com/60a0XnjZ |
00:32.08 | lvlinux | do you want a sip debug too? |
00:32.31 | [TK]D-Fender | [20:14]lvlinuxyes---so obviously that's why it's not going on to the next priority |
00:32.32 | [TK]D-Fender | [20:14]lvlinux(which is the voicemailmain) |
00:32.52 | [TK]D-Fender | http://pastebin.com/r86iM4Vx <- there is no "voicemailmain" in here at all |
00:33.01 | [TK]D-Fender | Nor is there a 2nd priority on anything |
00:33.03 | lvlinux | well i got rid of that and went down to basics |
00:33.22 | lvlinux | now the extensions.conf i'm using is exactly what you see |
00:33.49 | [TK]D-Fender | You're creating a moving target by the time you are showing us anything. |
00:37.33 | lvlinux | here's where I had the Voicemail line: http://pastebin.com/jdKJttGL |
00:38.53 | lvlinux | and here's the output: http://pastebin.com/Z3f17ynt |
00:40.06 | [TK]D-Fender | voicemail looks a little messed up for aborting like that |
00:40.27 | [TK]D-Fender | And senseless to use there, but is a bit besides the point |
00:41.05 | lvlinux | yes i know it's senseless---just using it for testing |
00:43.10 | [TK]D-Fender | What ver of * are you running? |
00:43.29 | lvlinux | 11.7 |
00:45.13 | [TK]D-Fender | Is dahdi configured and running from CLI? |
00:45.29 | lvlinux | yes |
00:45.30 | [TK]D-Fender | "dahdi show status", "dahdi show channels" |
00:45.49 | lvlinux | status is good |
00:45.57 | lvlinux | channels show the three lines in service |
00:46.08 | lvlinux | and the contexts line1, line2, and line3 |
00:49.00 | [TK]D-Fender | Hrm, the guide I'm seeing say that the dialplan is"generated" |
00:49.49 | lvlinux | that's if you put the autocontext=line1 line at the trunks and stations in sla.conf |
00:50.09 | lvlinux | or autocontext=lineX rather |
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00:50.23 | [TK]D-Fender | exten => station2_line1,1,SLAStation(station2_line1) |
00:50.47 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Basic+SLA+Configuration+Example |
00:51.14 | [TK]D-Fender | Looks like they've changed a thing or two since I saw it in 1.4 |
00:51.40 | lvlinux | :-) |
00:51.56 | [TK]D-Fender | that dialplan line looks like it uses a hybrid name to refer to the sla.conf sections to pick the device |
00:52.18 | lvlinux | yes |
00:52.42 | lvlinux | you mean with the underscore---ie station1_line2 |
00:52.51 | [TK]D-Fender | yes |
00:53.33 | [TK]D-Fender | <PROTECTED> |
00:54.05 | [TK]D-Fender | exten => station1_line1,1,SLAStation(station1_line1) |
00:54.07 | [TK]D-Fender | like this one |
00:54.16 | [TK]D-Fender | So it looks like you made the wrong choiice for your speed-dial |
00:54.37 | [TK]D-Fender | Which should ahve been to "station1_line1" |
00:54.47 | lvlinux | ok i'll give that a shot. |
00:55.16 | lvlinux | BTW when I dial in on the DAHDI lines, the phones don't ring, but the line lights on them do blink correctly for the DAHDI line |
00:56.40 | lvlinux | at least i'm not using polycom phones---these take long enough to reboot but not like the polys lol |
00:59.08 | [TK]D-Fender | You shouldn't have to be rebooting them all the time |
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00:59.30 | [TK]D-Fender | And that's a terrible basis for choosing a phone |
00:59.35 | lvlinux | you have to when you change a SIP parameter |
00:59.43 | [TK]D-Fender | Which, don't worry... will likely manifest later |
00:59.46 | lvlinux | lol i didn't choose them for that---i like the polycoms a lot better |
01:01.46 | lvlinux | ok * gave the same output as before---should i try something besides Voicemail to test this? |
01:02.08 | lvlinux | i should say same effect as before not same output let me PB it. |
01:02.40 | lvlinux | http://pastebin.com/L6ppFgJc |
01:06.26 | [TK]D-Fender | [20:45][TK]D-Fender"dahdi show status", "dahdi show channels" |
01:06.28 | [TK]D-Fender | PB it |
01:06.31 | lvlinux | k |
01:07.13 | lvlinux | http://pastebin.com/UxZBqS1E |
01:07.42 | lvlinux | the DAHDI ports are hooked to an ATA on the other side if that makes any diff. |
01:09.29 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/SLA+Trunk+Configuration <- I'm seeing some more odd context's required for this in this guide page |
01:09.57 | [TK]D-Fender | that handles dialing, etc.... mind you this is with a non-live SIP provider where you have to colelct the # first... |
01:10.05 | [TK]D-Fender | This is really a mess... |
01:11.59 | lvlinux | hehe |
01:12.47 | lvlinux | yeah i tried it with that type of setup w the [lineX_outbound] contexts and it didn't work so that's why i tried to simplify it back down just to get it working first. |
01:13.35 | lvlinux | i don't need SIP dialing though on this system since they only have 3 PSTN lines. |
01:15.02 | [TK]D-Fender | No, I see that.... it looks like it maybe should be working at this point (since you're dialing what it seemed to imply you shave have) |
01:17.41 | lvlinux | if i replace the VoicemailMain call with a call out on DAHDI (same => n,Dial(DAHDI/1/18887779999) I get: |
01:18.01 | lvlinux | http://pastebin.com/edFVXtdz |
01:18.10 | lvlinux | congested/busy |
01:19.08 | [TK]D-Fender | that would imply it is already in use... |
01:19.18 | [TK]D-Fender | "core show channels" |
01:19.29 | [TK]D-Fender | "dahdi show channel 1" |
01:19.32 | lvlinux | it's not0 active channels |
01:19.32 | lvlinux | 0 active calls |
01:20.16 | lvlinux | http://pastebin.com/nyA69vUd |
01:20.25 | lvlinux | thats' dahdi show cahnnel 1 |
01:20.37 | lvlinux | sry "dahdi show channel 1" |
01:22.02 | lvlinux | wait a minute it says it's off hook |
01:22.07 | lvlinux | hmmm |
01:24.35 | lvlinux | when i unplug it it says onhook |
01:24.49 | lvlinux | but when i plug it back in it immediately says offhook |
01:24.59 | lvlinux | same thing for channels 2 and 3 |
01:27.07 | lvlinux | same behavior if i plug it into the actual PSTN line instead of my ATA |
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01:38.17 | [TK]D-Fender | I'd check into that... |
01:38.42 | [TK]D-Fender | Ok, I've drilled about as much into this as I can go.... |
01:38.50 | lvlinux | what do you mean check into what specifically? |
01:38.56 | lvlinux | the TDM card? |
01:39.22 | [TK]D-Fender | For all of the downsides (not being able to use "redial", killing your CDR's, etc... this really seems like a waste to try to fake out SLA |
01:40.01 | lvlinux | you're telling me!!!! you know that saying "the customer is always right" lol well that's what I've gotten myself into here :-( |
01:40.35 | lvlinux | small pizza place---ancient analog PBX switcher key system thing--static, dropped calls, unpredictable. |
01:41.14 | lvlinux | i go and install a nice little * box with 3 IP phones, and they say they can't stand not being able to say "pick up line 3 bob..." |
01:41.47 | [TK]D-Fender | That's what parking is for |
01:41.54 | lvlinux | and they think a blind transfer is "good for an office but not efficient for here..." |
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01:43.11 | lvlinux | hmmm i had thought about parking but didn't look into it too much. |
01:43.38 | lvlinux | can you have the BLF indicators showing the status of parked calls maybe? |
01:43.55 | [TK]D-Fender | Park call. Get lot #. Hang up. Yell "Bob, 702!" |
01:44.15 | [TK]D-Fender | You can get presence on your lots as well |
01:44.23 | lvlinux | nope they won't go for that---- it's the "702" that they'll complain about. |
01:44.32 | lvlinux | they want to press one button on a blinking light |
01:44.48 | [TK]D-Fender | with the lots having BLF they CAN just hit that buttone <- |
01:45.15 | lvlinux | hmm, ok and then I could do a speed dial to the parked lot? |
01:45.30 | [TK]D-Fender | depending on the phone, sure |
01:45.45 | lvlinux | depending on the phone---yeah i was afraid of that. |
01:45.53 | lvlinux | i don't think i can w these phones |
01:46.18 | lvlinux | it only lets me set BLF for shared lines I think :-( |
01:46.51 | [TK]D-Fender | Obviosly not. |
01:47.06 | [TK]D-Fender | Because what the phones consider a "shared line" does not exist in Asterisk |
01:47.20 | [TK]D-Fender | The only thing you were using them for WAS presence + speed-dial |
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01:47.30 | lvlinux | oh? |
01:47.33 | lvlinux | hmmmmmm |
01:47.43 | [TK]D-Fender | your "SLA" WAS a speed-dial. |
01:47.53 | lvlinux | so was i not supposed to set the phones to shared line mode---ooops |
01:47.56 | lvlinux | ? |
01:47.57 | [TK]D-Fender | You missed the big-print somehow... |
01:48.10 | [TK]D-Fender | Oh dear God.... |
01:48.20 | lvlinux | lol i'm going to wet myself |
01:48.21 | [TK]D-Fender | * does NOT support SLA. |
01:48.44 | [TK]D-Fender | those apps = cheap-hack to fake it out. |
01:48.52 | lvlinux | ahhhhhhhhhh |
01:49.03 | [TK]D-Fender | THE CAKE IS A LIE |
01:49.20 | lvlinux | no wonder it wasn't working right |
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01:49.53 | [TK]D-Fender | well I saw it dialing those "text" extensions... I don't really get it at this point.... |
01:50.41 | [TK]D-Fender | that LOOKED like it was just doing a speed-dial across your primary (appearance |
01:51.01 | lvlinux | i think the only way to get BLF on these phones is to set it as DSS/BLF/shared line mode, |
01:51.20 | [TK]D-Fender | that doesn't make sense as a combo. |
01:51.29 | [TK]D-Fender | But... no time to dig further. Got to head out... |
01:51.40 | [TK]D-Fender | Seriously look at parking and save yourself a world of pain. |
01:51.59 | lvlinux | if i can fake it out enough where they think it's like the old system then that's what I'll do. |
01:52.00 | [TK]D-Fender | heads out.... |
01:52.02 | lvlinux | Thanks so much |
01:52.05 | lvlinux | as always |
01:52.18 | lvlinux | beats head on wall |
01:56.30 | lvlinux | well i've got to try to get BLF working on a private line with these phones then. |
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08:02.40 | Sconk | Hi is there any way of logging in the debug log what password a phone is trying to register whit ? |
08:10.04 | jeffspeff | Sconk, are you monitoring the * console? |
08:10.14 | jeffspeff | when the phone tries to regiser? |
08:10.25 | Sconk | yes |
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08:11.49 | Sconk | i keep getting wrong password.. |
08:11.59 | Sconk | and its a sip client |
08:12.35 | jeffspeff | did you verify the password in the sip.conf file and the password in the phone's config? |
08:14.04 | Sconk | the freepbx gui shows right password budt i will try looking in the files |
08:15.18 | Sconk | its the right password |
08:15.36 | Sconk | budt its a old ACN iris 300 i think its playing some tricks with me |
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14:24.17 | YannikS | Hi, when originating calls using AMI no CallerID is sent to the internal phone specified in 'channel'. It should be set to the external number which is being called. How can I fix this? |
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15:53.20 | logical2 | Good morning, I built asterisk from source 11.5.1 back in Sept of last year. I would like to recompile to the latest version.. is there a tutorial anyone is aware of.. my Google searches produced little results.. |
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16:33.39 | file | wobbles in |
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17:01.18 | logical2 | found answer to earlier question, solved. narrative: "wget src, tar, configure, make menuselect, make install" |
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20:32.29 | file | silence! |
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20:54.36 | slav3_kitten | huh i wonder |
20:54.38 | slav3_kitten | i should google to see if i can have inbound numbers give my cisco phones a different ring tone |
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21:32.14 | *** join/#asterisk ChannelZ (channelz@burner.com) |
21:49.22 | bsdice | yo |
21:55.25 | ChannelZ | yoyo to the mofo |
21:59.06 | bsdice | talking to me? |
21:59.09 | bsdice | no mofo :) |
21:59.23 | bsdice | but I have good news... still running asterisk and my phone still works ;) |
22:03.02 | ChannelZ | Well that's a good thing |
22:04.32 | bsdice | yeah, really liking Asterisk |
22:04.38 | bsdice | can't believe it's free |
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22:29.03 | file | feel free to contribute back in any way you can to help the project ;) documentation fixes, helping on here or the -users list, etc |
22:33.09 | bsdice | I will |
22:55.20 | [TK]D-Fender | file: IIRC there is a new option in sip device entries in sip.conf to use that peer's settings for registrations which is needed in the case of having to go out a specific proxy. Do you recall it offhand? I can't see it in the v11 sample config |
22:55.49 | file | I think in the register line instead of specify a host you can specify the name of the peer entry |
22:57.40 | [TK]D-Fender | That would be uniform to how Dial handles "@" |
22:57.54 | [TK]D-Fender | Almost too clean ;) |
22:58.03 | file | it's common code. |
22:59.14 | [TK]D-Fender | Do you know when this got put in? |
22:59.29 | file | oh geez, probably 1.2 or 1.4 days |
23:00.54 | file | there was talk long ago about putting the registration itself into a peer, or having a new type, but I don't think anything happened |
23:02.17 | bsdice | you mean like callbackextension= ? |
23:02.50 | bsdice | visited a friend in Germany and tried to get that going for Deutsche Telekom IP VDSL, failed |
23:03.04 | file | ah yup |
23:03.11 | bsdice | had to go with register= in general section |
23:03.13 | file | the mess that is chan_sip |
23:03.41 | bsdice | hrhrhrhr |
23:03.59 | bsdice | hey, haven't tried 12 yet, because 11 trunk is so damn stable here |
23:04.22 | bsdice | read somewhere calling SIP "extensible hell" and sure it is |
23:04.35 | bsdice | never fell in love with H.323 though |
23:04.54 | bsdice | that always smelled proprietary Cisco, Siemens and whatnot |
23:05.11 | BludSuckingFiend | h.323, yuck... ASN1 and tons of vendor extensions |
23:05.23 | bsdice | sorry... TRIGGER WARNING |
23:05.25 | bsdice | hehehe :D |
23:08.09 | BludSuckingFiend | Get to try putting h.323 through NAT this week and not looking forward to it |
23:08.22 | file | realizes the sound he is hearing is rain |
23:08.39 | BludSuckingFiend | not that it is much better with SIP. Makes me wish IAX caught on across vendors |
23:09.19 | file | the methods of doing it with SIP are understood these days, better than it was |
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23:10.32 | BludSuckingFiend | Yeah, though it is one of the reasons I've always hated the FTP protocol. |
23:10.54 | BludSuckingFiend | Different data/control channels |
23:14.47 | bsdice | enter SFTP ;) |
23:14.57 | bsdice | SIP over NAT is well-controlled these days indeed |
23:15.36 | bsdice | even though the hoops pjsip stack jumps through to get it going is crazy |
23:15.56 | BludSuckingFiend | I end up using TDM trunking anyway though |
23:17.29 | bsdice | how many phones you got under management? |
23:17.44 | BludSuckingFiend | ~200 |
23:18.09 | BludSuckingFiend | on asterisk anyway |
23:18.33 | bsdice | I'm just doing it for friends, with encryption. maybe 10-20 SIP accounts |
23:19.34 | BludSuckingFiend | Yeah, I've found SIP offerings from a lot of enterprise companies lacking. |
23:19.51 | bsdice | ever played with autoframing? |
23:20.11 | BludSuckingFiend | Unfortunately high density TDM is fairly opaque on asterisk |
23:20.16 | bsdice | tried to get big frames going with speex to limit overhead on mobile, not working |
23:20.22 | bsdice | probably too stupid |
23:20.29 | BludSuckingFiend | nope, never |
23:20.51 | bsdice | threw out the last TDM card about half a year ago |
23:20.56 | bsdice | haven't looked back |
23:21.09 | BludSuckingFiend | Just bought a quad span board |
23:21.17 | bsdice | eicon? |
23:21.19 | BludSuckingFiend | and an R515 box to put it in |
23:21.25 | BludSuckingFiend | Sangoma |
23:22.52 | BludSuckingFiend | The company's been running 200 phones and 3 ISDN PRIs off a desktop box for awhile |
23:23.05 | BludSuckingFiend | finally convinced them to actually buy decent hardware for it |
23:23.42 | bsdice | I had been running ISDN BRI for a while, sadly driver was not stable, memleak in newer kernels |
23:23.45 | bsdice | >3.0 |
23:24.13 | bsdice | had worries that fax would no longer work |
23:24.24 | bsdice | but with a-law no problemo |
23:24.46 | bsdice | (via iaxmodem and hylaxfax behind that) |
23:25.06 | BludSuckingFiend | ah, yeah |
23:25.13 | BludSuckingFiend | Using that for a fax server too |
23:25.20 | WIMPy | bsdice: Which driver? |
23:25.38 | bsdice | ugh |
23:25.42 | bsdice | hang on 1 sec |
23:26.18 | bsdice | zaphfc |
23:27.26 | WIMPy | Ah, ok. |
23:28.04 | bsdice | WIMPy if you compare the D-channel hardware irq handler with other HFC-style drivers, it is no wonder. "kaputtgefummelt" |
23:28.30 | bsdice | shame but then, at least in Germany where the box is, ISDN is on its way out |
23:29.59 | WIMPy | I prefer to use LCR over DAHDI. Has worked very well since I finally made the change. |
23:30.36 | BludSuckingFiend | it certainly is old. Hopefully providers will be able to get their act together supporting VoIP properly |
23:30.59 | BludSuckingFiend | for now I just use PRIs for any critical traffic |
23:31.03 | WIMPy | I wouldn't expect that in the near future. |
23:31.10 | bsdice | German Telekom works, if you know what you are doing. Deeply. |
23:31.32 | bsdice | I think I cursed for 10 hrs until I had it going and this is a very big provider in Germany |
23:31.39 | WIMPy | What works? Do they already have full SIP-I? |
23:31.51 | bsdice | what is SIP-I |
23:32.05 | WIMPy | An attempt to make SIP usable. |
23:32.20 | WIMPy | Some horrible mishmash of SIP and H.323. |
23:32.21 | bsdice | well we registered 10 numbers and been using it successfully |
23:33.08 | WIMPy | Or ASN.1 with SIP compatibility or whatever. |
23:33.18 | bsdice | no aware of them supporting that |
23:33.28 | bsdice | at least they appear to have fixed session-timers |
23:33.56 | bsdice | P-Asserted-Identity or whats it called is still a problem |
23:34.08 | WIMPy | In what way? |
23:34.11 | bsdice | need to tweak Asterisk pretty good to get it to do what you want |
23:34.17 | bsdice | hang on looking for snipped |
23:34.48 | bsdice | if you set ;trustrpid=yes then your caller ID will be off |
23:35.09 | bsdice | off in the sense, that asterisk displays RPID instead of From |
23:35.27 | bsdice | which means you get the -0 instead of an extension |
23:35.32 | bsdice | confusing |
23:35.43 | bsdice | Set to yes causes P-Asserted-Identity override |
23:36.22 | bsdice | another real nutbreaker is their login casino |
23:36.47 | bsdice | need to use the email address, not the VDSL dialup login/pass and also not their customer center's login/pass |
23:36.52 | bsdice | which all is different |
23:36.58 | bsdice | nailbiting... |
23:37.26 | bsdice | and finally iptables with stateful block of incoming connections |
23:37.37 | bsdice | i.e. block 5060 from external sources |
23:37.58 | bsdice | which I solved by using qualifyfreq=50 |
23:38.32 | bsdice | to keep the state valid towards tel.t-online.de |
23:38.41 | bsdice | so external calls reach the box still |
23:40.43 | WIMPy | Doesn't Asterisk also support keep-alives? |
23:41.50 | bsdice | I am abusing qualify |
23:42.38 | bsdice | don't know |
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