phonetrovert

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phonetrovert last won the day on December 3 2015

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About phonetrovert

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    The phorce is with me!

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  1. On that DMS-100 DISA extension 02600 is a pretty messed up NIS recording.
  2. Any calls that connected were all local, and it was an outsourced operator from a COCOT, so in hindsight I think the redbox tones and/or the piss poor line quality allowed me to trick the op personally. That same afternoon a different operator wouldn't do it for me anymore at all. Edit: Are there verizon payphones anywhere anymore? For certain not in the Metro NY area that I have seen.
  3. 800-523-1231 Credit card gateway(?). Answers with a carrier.
  4. If you call to toll-free numbers the person your calling has to pay for the call, so it doesn't cost the operator any money to just patch you through. But on the other hand, if you call a non toll-free number, You'd be responsible for payment, which is why that happened I think.. Does it work when you dial local non toll-frees?
  5. Without *67: http://imgur.com/MXMwPsZ With *67: http://imgur.com/0smhTFO Kc9pke posted these, It looks like p-charge-info and ani2 gets passed to asterisk. These fields would be SIP-T, right? There's alot of other unrelated changes that have happened to asterisk (i.e, scrapping meetme for confbridge, and changing some conf file syntax), so it wouldnt surprise me if they updated these capabilities also.
  6. I was wondering what happens when you call forward to your 800 line, Is there any information about the originator of the call, as opposed to the number doing the forwarding? I think I read somewhere that's how alot of those anonymous call unmasking services work.
  7. Wait, Im sorry.. Im slow lol.. I see the charge-info is there, that's the billing number, so that's cool. And the asserted identity get's passed as well. Sweet! it works. Apparently asterisk is doing as I said, and placing the CPN in the ANI field because your doing this over SIP-T and not PRI (the CALLERID(ani) var is PRI specific). But the other SIP-T fields are there with the same info. Edit: Reading stuff like this https://tools.ietf.org/html/draft-york-dispatch-p-charge-info-02to try and learn more about everything, Im realizing I shouldn't call it a BTN or billing number, but a charge number seems to be the more common lingo.. Edit: Im pretty sure theyd be violating FCC regulation by not strictly enforcing *67 on a non toll free number..Thats why I was asking, would have been pretty surprised. Sorry I did give alot of misleading tips/wrong info, My flowroute account is out of credits atm, so alot of things I couldnt test myself, and I was reading through like 1,000,000 acronyms that all describe similar things (i.e, I've seen ANI2 digit referred to as ISUP/OLI, CPC, and ANI-II digit). I learned alot though..
  8. I just did a tcpdump on the Asterisk box itself and called with and without *67. Without *67 the From:, Contact:, and P-Asserted-Identity fields do populate with ANI, but ANI II is nowhere to be found in the INVITE header at all. With *67, the three aforementioned headers just populate with Anonymous. EDIT: It's apparently with toll frees only. An extra header named P-Charge-Info has the ANI in the format of tel:NPANXXXXXX Without *67: http://imgur.com/MXMwPsZ With *67: http://imgur.com/0smhTFO Yeah the CPN from P-asserted-identity on a private inbound call trick is with toll-free only I think.. EDIT: If the ANI is anonymous then your not getting ANI, its giving you CPN, try it with a toll-free..And before you do, I'd just ask them outright if they'll give you a billing number anywhere, because that's what your looking for I think..(you already have ANI2 digits, and CPN, thats the missing piece) EDIT: P-Asserted-Identity gets passed even with *67? On a non-toll free number?
  9. OK so forget that idea that they might be populating the ANI fields in the dial plan (I was just hoping life might be easier on this one..). is the number toll-free? BTW the "0" ANI-2 digit is for POTS, a cellphone should be 23, 61, 62 or 63. So it's definitely just giving you callerid Keep in mind ANI should never be anonymous unless someone did some tricky, deliberate stuff to get it that way. http://www.nanpa.com/number_resource_info/ani_ii_assignments.html Try n,Set(RPID=${SIP_HEADER(P-Asserted-Identity)})or n,Set(RPID=${SIP_HEADER(Remote-Party-Id)})in your dial plan, do either of these values yield anything when you press *67? ( At least we'd know you could unmask a blocked callerid for starters.. ) As instructed in that flowroute link, ngrep for the "INVITE" header and see if the ANI data is in there..Should be. If poking around like this yields no success. I would contact them for further details, i.e, where specifically to find the extra SIP-T ANI stuff. Edit: I see you say it's not a toll-free number, I'm not sure if that makes a difference or not. I would have imagined it did, but I don't see anything about it in that link. I think SIP-T is an addition to regular SIP at the protocol level. Use some kind of network debugging software at the protocol level to figure out the exact headers you need, and you can then add the headers in the asterisk dial plan with SIPAddHeader and process the extra information from flowroute I would think. (That's wrong SIPAddHeader() adds a header to an INVITE message on an outbound call, that's what I get for skimming documentation and forums..) I have an account from them too,If I get some time I will try experimenting with this as well. I feel like it should work since they're saying it would..
  10. I'm thinking since that link I posted from flowroute said they support all of this, some of these standard channel variables for asterisk 1.8 should work, Even though the documentation says some of the fields are for PRI channels. ${CALLERID(ani)} ${CALLERID(ani2)} Check out the asterisk documentation here: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables Also be aware that I read in earlier documentation (from asterisk 1.4) that from (Im assuming) then on all channel drivers would pass an ANI, and it would pass callerid through the ANI field if there was no ANI (in situations like VOIP to VOIP calls). So you should check the results against the ANAC that gives ANI at 800-223-1104, to make sure your getting the proper information in asterisk. Edit: That would be my first guess (They just fill the relevant PRI specific fields) but Im noticing their article says something about going through Sip-Invite and NPAC metadata.. Which is definitely something I'd have to read up on, because I dont know what the hell that means at this point lol. If the standard channel variables dont work I'd ask flowroute support how it's done. I can't see them not giving out all that extra info (on top of unblocking CPN), since they specifically said they do. Let us know how everything goes P.S, Alternatively, if you get incoming calls to your asterisk box working, Do the following in the terminal, And try to go through the asterisk CLI output to see where some of the headers of interest are possibly.. /path/to/asterisk -crvvvvvv > debug.txt and grep debug as required .. Of course that's a last resort anyway, probably just go to support worst case. (Everything lined out is wrong.. Im shot..) To debug SIP, use something like ngrep, or setup logger.conf in asterisk specifically to log at the protocol level (if possible..otherwise use ngrep). http://www.voip-info.org/wiki/view/Asterisk+config+logger.conf
  11. 714-505-0005, 0099, 0101 etc. Old Disconnected message from Pat Fleet 714-505-0019 carrier after you wait about :50 - 1 minute 714-505-0016 VMB 714-201-00xx,01,05, 69 Beepers
  12. Check out this presentation from Kevin Mitnick. Basically, run a toll-free number to your asterisk box (with SIP), since your paying for the call, You have a right to know the billing number, even if the CPN is blocked (at least this is true with some providers). I did this back in 2009 or so, I think I was using flowroute. EDIT: To be honest, now that I got to re-watch the video (no sound..wtf), I might have given you mixed up information. They're (flowroute) allowing you to see the p-asserted-identity with a toll-free number. It still unblocks the caller ID, and I have read in various other places that owners of toll-free numbers are entitled to see your ANI, but I might be wrong in saying p-asserted-identity is equivalent to ANI (? I think its just plain caller id) Although my memory on all this stuff is hazy,I feel like I was able to filter out payphone origination to my toll-free when I was messing around with it, because I remember wanting to avoid the surcharges. How else could I have done that without ANI? Unfortunately I really don't remember since it was a good while ago. The more I read, the more Im convinced I remember correctly. Pretty sure flowroute passes you everything with a toll-free. EDIT: I can say I'm certain you get the Calling Party Category or ANI2 digit, and unblock caller ID with a toll-free number, but I never really tried to get the billing number/ANI itself in SIP, so I can't say for sure whether it's possible. I'm trying to find out more information on that specifically. Check this out apparently flowroute definitely does what you want it to, and I'll bet plenty of other SIP providers do as well (Wonder if its possible for any providers to do this without toll-free?): https://blog.flowroute.com/2015/08/27/using-advanced-signaling-to-detect-call-center-fraud/Apparently you can unmask alot of caller information including ANI over SIP, including the ANI2/CPC digit, amongst other things, so this thread is still a valid answer to your question.. Just wanted to verify I wasn't recalling anything incorectly.
  13. I just tried calling it from work yesterday, and every second call it appeared to put me on some kind of room monitor. No one spoke english, and one person greeted me with asalamaleichum and started speaking a language I dont understand..Didn't sound like arabic, but outside of that not sure. Not even sure about that.. I do know that's a wierd thing going on with my work VOIP provider. It works from my house now (not the wierd room monitor thing at work, that must have something to do with VOIP, but I now get the same error message I get on other service providers! For the longest time i wasn't getting that error message (the one I uploaded before) when I called from my home fios line, just a standard CBCAD message from my local switch, but it seems like they updated things based on my repeated attempts to connect with this exchange,, haha! or it could be that I recently moved one floor up and am somehow connecting differently. No idea.. (more likely it's the latter case come to think of it)
  14. xhausted110, on 01 Dec 2015 - 11:27 PM, said: I called (253) 561-0100, pressing 0 transferred me to some school's IVR. That's wierd.. if I call private from 718 area code I get a reorder after pressing 0.. I'm gonna try it on google voice. Yeah It hangs up on me on google voice too.
  15. numbers apparently on a Siemens DE4 EWSD, but goes to a 4ess long distance tandem "All Ciruits Are Busy" message on 089T I used to have the list of long distance tandems.. Can anyone remind where 089T is again? 206 408 91xx "You have reached your districts voice messaging service, please press 0..etc." no matter what I do it disconnects. sounds like an oldschool speech synthesis device. 253-561-0100