similar to: Transfer Fraud

I worked on some ad data before and I found NYT article very biaised and not far from a fraud... Anyone knows if they got money from a - company - to play 'R'? Check out the title: R U Ready for R? Seems to me this title was stolen from XLSolutions www.xlsolutions-corp.com and they never mentioned XLSolutions in the article! They mentioned commercial R....never mentioned

Hello Everyone, We are looking for solutions where the transcoding is abstracted away from our * box (i.e., to the network layer) using some carrier grade gateway, or router. The reason for my post is to know about solutions people have used in the past, and how it fits into their overall architecture. Our transcoding needs consists mainly of u/alaw <-> g729, and gsm would also be good....

I am trying to setup a couple of virtual pbx's off of my one may asterisk box. So far I have been able to segment most everything via the Dial plan. My only question/problem has to do with the # Transfer function. I had set up # Transfers prior to segmenting the dial plan, and I cannot remember how I was able to specify which context to use when the user presses #. I haven't been able

Hi, Someone reported me that from a PBX on which someone gained fraudulent access, he could observe hundreds of calls to the same destination number. For curiosity's sake, I'm wondering why would this happen (dialing the same number over and over) ? Some special numbers generate here and there revenues for callees (and not for callers). Beside sharing interests with the callee that get

Hi all, I am trying to block all IP addresses from Africa due to a high rate of fraudulent orders coming from them. I have some found some websites that can generate a range of IP addresses. However, you would have to enter the ranges by country. I can use those perhaps but that would mean I would have to get the ranges for each country one by one. Has anyone here a list of addresses from

I have multiple Asterisk instances set up in different locations and would like to modify the callerID of inbound calls to identify which instance the call is coming from.? I knew how to do that with the old sip format, but can't seem to figure it out with PJSip. For example: Currently Location A, extension 10 calls Location B, extension 20.? CallerID on Extension 20 displays

Hello, I tried making a donation via the centos.org website but was blocked by PayPal. I tried today and a few weeks ago. The error I got in both instances was: "The card you entered cannot be used for this payment. Please enter a different credit or debit card number." This strikes me as a strange response. It didn't say the card was declined; it said it "cannot be

When I set CONNECTEDLINE() info for an incoming ISDN call, the calling party sees only CONNECTEDLINE(num) and the name does not get displayed. Some time ago I called a number, where I did get back a name and a number and everything was displayed correctly. So I think the calling site should basically be able to handle all connected line info. Looking at a pcap trace of the D-channel data, I

Beta 1 is now available for testing. One of the problems I''ve had with the Shorewall products is trying to keep them all in sync. There have been two copies of each shell library and four CLI programs. To simplify maintenance, I have collapsed each of the library pairs into a single library and have reduced the number of CLI programs from four to two (one for the

Beta 1 is now available for testing. One of the problems I''ve had with the Shorewall products is trying to keep them all in sync. There have been two copies of each shell library and four CLI programs. To simplify maintenance, I have collapsed each of the library pairs into a single library and have reduced the number of CLI programs from four to two (one for the

Hi, I want to check the status of a blind transfer (only sip endpoint) between various phones. Transfer is working perfectly, using ## from features.conf or using transfer key from phone, here SNOM320. My problem is that if party to transfer to is busy, the transfer fail and the call is ended. What I want to do is to return the call to the party who originate the transfer. I checked

RC 3 is now available for testing. Problems corrected: 1) The Shorewall and Shorewall6 ''load'' and ''reload'' commands previously used the setting of RSH_COMMAND and RCP_COMMAND from /etc/shorewall/shorewall.conf (/etc/shorewall6/shorewall6.conf). These commands now use the .conf file in the current working directory. 2) The new parameterized

RC 3 is now available for testing. Problems corrected: 1) The Shorewall and Shorewall6 ''load'' and ''reload'' commands previously used the setting of RSH_COMMAND and RCP_COMMAND from /etc/shorewall/shorewall.conf (/etc/shorewall6/shorewall6.conf). These commands now use the .conf file in the current working directory. 2) The new parameterized

Is there a recommended way to find out the cause of DIALSTATUS = CONGESTION for PRI/BRI channels? Currently I am evaluating the DIALSTATUS variable and I also count the active ISDN channels for the ISDN trunk in question. Counting the active ISDN channels seems somewhat clumsy as the mapping to a specific trunk must be done by hand (or write even more code). I have a setup where outgoing calls

hello list shorewall I encounter serious problems with this Multi-port Match: Available . my custom kernels do not provide this option that prevents me from compiling shorewall 6 plesase correct this maybe a bug fix --  http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x092164A7  gpg --keyserver pgp.mit.edu --recv-key 092164A7 http://urlshort.eu fakessh @

jg <webaccounts173 at jgoettgens.de> schrieb: > It doesn't really depend on your sip.conf and Asterisk. Your gateway/router > will be the major problem. My summer project will be to look at session Are you sure? Right now I'm using an italian SIP-Provider (Messagenet), configured in my sip.conf and I can receive calls without any problem... So, I don't think, I have to

Hello I can confirm that the variable DIALEDPEERNAME contains the information that I would expect in the variable BRIDGEPEER. But I read nowhere that DIALEDPEERNAME has replaced BRIDGEPEER as of Asterisk version 13 ?! So if this is not the intention, then yes this is probably a bug and should be reported. Kind regards. Jonas. On 18-09-16 19:58, Ludovic Gasc wrote: > Hi, > >

<div>Thanks for the tip. Our goal is to know that call-limit is triggered. And later analyze this info, maybe do some action.</div><div>Yes, we can parse CDRs or execute AGI script but we do not want inmplement this logic on Asterisk because it can affectš<span>performance.</span></div><div>š</div><div>02.07.2015, 15:31, "jg"

I have 2 ISDN BRI boxes, each with 4 spans, where the first one is configured as CPE, the second one as NET(so I don't need real lines for developing and testing). Once in a while I do see the following libpri error messages simultaneously on both boxes: PRI Span: 1 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state 7(Multi-frame established) PRI Span: 2 TEI=0 MDL-ERROR (A): Got

Hello a call goes out and is answered : [Sep 17 11:29:52] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b is making progress passing it to SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b answered SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23522][C-00000051] bridge_channel.c: Channel SIP/myprovider-0000010b joined