Re: [asterisk-users] Way to start CDR when call is bridged ?
Sorry for the noise, ${ANSWEREDTIME} do the job, CDR's can be forgetted.
Le 10/07/2020 à 16:08, Administrator a écrit :
I forgot to mention that I need billsec and in all those cases I have
2 seconds and not 258 as it should
Le 10/07/2020 à 15:58, Administrator a écrit :
Hi,
in dialplan -Asterisk 16.2 from Debian Buster- we have
same = n,Dial(PJSIP/101/102/103,15,tT)
If thew call is not answered after 20 seconds, we launch a new dial
with same and/or other extensions
same = n,Dial(PJSIP/101/104/110,20,tT)
Looking in CDR we have at the end of the call (here we called 3
extensions which where ringing, let say 110 answered the call)
"2020-07-10 14:18:29","","2020-07-10 14:18:29","2020-07-10
14:22:47","258",""+blabla" ","NO ANSWER","Callee
DID","","DOCUMENTATION"
"2020-07-10 14:18:29","","2020-07-10 14:18:29","2020-07-10
14:22:47","258",""+blabla" ","ANSWERED","Callee
DID","","DOCUMENTATION"
"2020-07-10 14:18:29","","2020-07-10 14:18:29","2020-07-10
14:18:31","2",""+blabla" ","NO ANSWER","Callee
DID","","DOCUMENTATION"
"2020-07-10 14:18:29","","2020-07-10 14:18:29","2020-07-10
14:18:31","2",""+blabla" ","NO ANSWER","Callee
DID","","DOCUMENTATION"
1. How it's possible tha CDR billsec appears in NO ANSWER for an
answered call ?
2. How to set NOCDR for the phones who didn't answer the call ? In
the h extension we set NOCDR for all DIALSTATUS except ANSWER but we
still get those CDRs
Is it possible to start CDR when a call is bridged ? Other solutions ?
Thanks for any hint
--
Daniel
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Re: [asterisk-users] Stir Shaken
On Mon, Jul 13, 2020 at 4:59 PM John Covici wrote: > > But the question is, are his statements correct that we need some > service -- not necessarily his -- to sign the call before sending it > to our normal carrier, or will the normal carrier -- whoever -- sign > the call if they know the number? > Right now the answer is you don't, and it may continue to be that way for you even after this comes into force. How STIR/SHAKEN is going to work in practice is still evolving, and I fully expect it to be differenct between carriers/providers/relationships. Jeff LaCoursiere has gone into this a bit with his response with his own experience and how they're approaching it. If you're concerned I would start a dialogue with your provider(s) to determine the expectations they have and what that looks like. From an Asterisk side we will have the functionality to do the signing and verification, as previously mentioned. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stir Shaken
-Original Message- From: asterisk-users On Behalf Of Matthew Fredrickson Sent: Monday, July 13, 2020 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Stir Shaken On Mon, Jul 13, 2020 at 2:34 PM Saint Michael wrote: >> >> There is a big confusion here about Stir Shaken. It is NOT a provider issue. >> Un fact, all providers are whasing their hands and modifying their swihtches >> to pass-through the Signature. They cannot sign the call because then the >> become the responsible party for the call before the FCC, and liable for any >> illegal call. Every owner of a PBX that sends calls to the network, except >> if you use a trunk for the likes of Vonage, needs to sign their calls. So if >> you send calls with any kind of dialer and use DIDs, real or "borrowed", you >> need to get the signature service urgently or your business will stop >> terminating calls. You cannot self-sign, you cannot get around it, the calls >> will either go to straight to voicemail or fail. Even worse, the carries wil >> play a fake voicemail and charge you a fee, something that some already a >> are doing when they detect robocallig. > > Don't even think about Transnexus, because they use 302 Redirect with a > header, and no version of Asterisk supports it. I am the only game in the > world for Stir-Shaken and Asterisk. I know it sounds arrogant but it is > literally true. If you need to sign your calls to get through, with Asterisk, > you need to connect to my service. I am an approved Service Provider from the > FCC. If you keep thinking this is not happening, it is, and your business > will disappear overnight. > The issue is that Vicidial, for example, does not provide res_odbc and > func_odbc, so you need to solve that first with Vicidial. Then you can apply > the code I provided earlier and your calls with have a legal, binding > signature. The carriers verify each signature and discard the ones that fail > the cryptography test. Sounds like you're trying to sell/direct people towards a service that you've created. Feel free to do so on the -biz list but the -users list isn't the right place for that sort of thing. Matthew Fredirckson He has been told before that this is not the right list. Can't someone delete him from the list? --Don -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stir Shaken
On Mon, 13 Jul 2020, Jeff LaCoursiere wrote:
Some of us may actually be interested in what you have to offer if you
changed the way you were presenting it. Who is going to base their
business on some list guy with a gmail address?
And can't follow directions and honor the mailing list rules. He got
spanked for this back in May.
I don't claim to understand much about this other than it is supposed to
help reduce spam by making providers accountable for sending calls with
CIDs that are not 'theirs.'
I also don't understand how the OP can sprinkle magic fairy dust on a call
and issue a token to any anonymous user for calls to and from CID/DIDs
they don't control as shown below:
mysql\
--batch\
--database=strshk\
--disable-column-names\
--disable-table\
--execute="call
strshk.stir_shaken_signature('7602588003','7602588003');"\
--host=208.73.232.47\
--password=\
--user=anonymous\
| cut --characters=1-30
eyJhbGciOiJFUzI1NiIsInR5cCI6In
I have no business relationship with the OP or 7602588003 so how does this
'token' add any value?
What am I missing?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
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Re: [asterisk-users] Stir Shaken
I used their scam checking service. Below is part of the dialplan I
used. I don't know how their STIR/SHAKEN service works the same.
same = n,GosubIf($[${LEN(${CALLERID(num)})} == 11]?scam-check,${EXTEN},1)
same = n,Goto(from-pstn,${EXTEN},1)
[scam-check]
exten = _XX.,1,Noop
same = n,Set(pres=${CALLERID(pres)})
same = n,ExecIf($["${pres:0:7}" !=
"allowed"]?Set(CALLERID(pres)=allowed_not_screened))
same = n,Dial(SIP/${EXTEN}@clearip,,);
same = n,ExecIf($["${pres:0:7}" != "allowed"]?Set(CALLERID(pres)=${pres}))
same = n,Set(tech=${HANGUPCAUSE(${clearip_chan},tech)})
same = n,ExecIf($["${tech:4:3}" == "603"]?Set(CALLERID(name)=Scam Likely))
same = n,Return
On 7/13/20 3:58 PM, John Covici wrote:
On Mon, 13 Jul 2020 15:44:12 -0400,
Matthew Fredrickson wrote:
On Mon, Jul 13, 2020 at 2:34 PM Saint Michael wrote:
There is a big confusion here about Stir Shaken. It is NOT a provider issue. Un fact, all
providers are whasing their hands and modifying their swihtches to pass-through the
Signature. They cannot sign the call because then the become the responsible party for
the call before the FCC, and liable for any illegal call. Every owner of a PBX that sends
calls to the network, except if you use a trunk for the likes of Vonage, needs to sign
their calls. So if you send calls with any kind of dialer and use DIDs, real or
"borrowed", you need to get the signature service urgently or your business
will stop terminating calls. You cannot self-sign, you cannot get around it, the calls
will either go to straight to voicemail or fail. Even worse, the carries wil play a fake
voicemail and charge you a fee, something that some already a are doing when they detect
robocallig.
Don't even think about Transnexus, because they use 302 Redirect with a
header, and no version of Asterisk supports it. I am the only game in the
world for Stir-Shaken and Asterisk. I know it sounds arrogant but it is
literally true. If you need to sign your calls to get through, with Asterisk,
you need to connect to my service. I am an approved Service Provider from the
FCC. If you keep thinking this is not happening, it is, and your business will
disappear overnight.
The issue is that Vicidial, for example, does not provide res_odbc and
func_odbc, so you need to solve that first with Vicidial. Then you can apply
the code I provided earlier and your calls with have a legal, binding
signature. The carriers verify each signature and discard the ones that fail
the cryptography test.
Sounds like you're trying to sell/direct people towards a service that
you've created. Feel free to do so on the -biz list but the -users
list isn't the right place for that sort of thing.
But the question is, are his statements correct that we need some
service -- not necessarily his -- to sign the call before sending it
to our normal carrier, or will the normal carrier -- whoever -- sign
the call if they know the number?
--
http://help.nyigc.net/
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Re: [asterisk-users] Stir Shaken
On 7/13/20 2:32 PM, Saint Michael wrote: There is a big confusion here about Stir Shaken. It is NOT a provider issue. Un fact, all providers are whasing their hands and modifying their swihtches to pass-through the Signature. They cannot sign the call because then the become the responsible party for the call before the FCC, and liable for any illegal call. I think this, being the basis of your whole argument, is the fallacy. S/S is forcing people to take responsibility, for sure, but carriers won't just let their customers leave because they don't want to sign calls. It will force them to make sure they know who their customers are, and make it impossible for those customers to escape consequences if they misbehave. We supply dialtone to a large number of businesses. We buy DIDs from carriers and resell them. It *may* be up to us to get our direct customers' calls signed, but at the moment we are in talks with our DID providers to do so on our behalf. In the next year I have no doubt if there are niches to be filled in providing CA or outright signing-as-a-service, businesses will be jumping out of the woodwork to provide it. I'm not panicking yet. I am the only game in the world for Stir-Shaken and Asterisk. I know it sounds arrogant but it is literally true. If you need to sign your calls to get through, with Asterisk, you need to connect to my service. I am an approved Service Provider from the FCC. If you keep thinking this is not happening, it is, and your business will disappear overnight. Its not just arrogant, its silly, and you have a serious branding problem. If you really have "The Answer" you should work on getting yourself a domain name at least. Cease the panic-inducing posts and come up with some reasonable fodder you could link to in your signature or something (like when you help with some thread), so you would at least contribute to the list at the same time. Some of us may actually be interested in what you have to offer if you changed the way you were presenting it. Who is going to base their business on some list guy with a gmail address? -- Jeff LaCoursiere StratusTalk, Inc. 703 496 4990 x108 815 546 6599 cell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stir Shaken
On Mon, 13 Jul 2020 15:44:12 -0400, Matthew Fredrickson wrote: > > On Mon, Jul 13, 2020 at 2:34 PM Saint Michael wrote: > >> > >> There is a big confusion here about Stir Shaken. It is NOT a provider > >> issue. Un fact, all providers are whasing their hands and modifying their > >> swihtches to pass-through the Signature. They cannot sign the call because > >> then the become the responsible party for the call before the FCC, and > >> liable for any illegal call. Every owner of a PBX that sends calls to the > >> network, except if you use a trunk for the likes of Vonage, needs to sign > >> their calls. So if you send calls with any kind of dialer and use DIDs, > >> real or "borrowed", you need to get the signature service urgently or your > >> business will stop terminating calls. You cannot self-sign, you cannot get > >> around it, the calls will either go to straight to voicemail or fail. Even > >> worse, the carries wil play a fake voicemail and charge you a fee, > >> something that some already a are doing when they detect robocallig. > > > > Don't even think about Transnexus, because they use 302 Redirect with a > > header, and no version of Asterisk supports it. I am the only game in the > > world for Stir-Shaken and Asterisk. I know it sounds arrogant but it is > > literally true. If you need to sign your calls to get through, with > > Asterisk, you need to connect to my service. I am an approved Service > > Provider from the FCC. If you keep thinking this is not happening, it is, > > and your business will disappear overnight. > > The issue is that Vicidial, for example, does not provide res_odbc and > > func_odbc, so you need to solve that first with Vicidial. Then you can > > apply the code I provided earlier and your calls with have a legal, binding > > signature. The carriers verify each signature and discard the ones that > > fail the cryptography test. > > Sounds like you're trying to sell/direct people towards a service that > you've created. Feel free to do so on the -biz list but the -users > list isn't the right place for that sort of thing. But the question is, are his statements correct that we need some service -- not necessarily his -- to sign the call before sending it to our normal carrier, or will the normal carrier -- whoever -- sign the call if they know the number? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stir Shaken
On Mon, Jul 13, 2020 at 2:34 PM Saint Michael wrote: >> >> There is a big confusion here about Stir Shaken. It is NOT a provider issue. >> Un fact, all providers are whasing their hands and modifying their swihtches >> to pass-through the Signature. They cannot sign the call because then the >> become the responsible party for the call before the FCC, and liable for any >> illegal call. Every owner of a PBX that sends calls to the network, except >> if you use a trunk for the likes of Vonage, needs to sign their calls. So if >> you send calls with any kind of dialer and use DIDs, real or "borrowed", you >> need to get the signature service urgently or your business will stop >> terminating calls. You cannot self-sign, you cannot get around it, the calls >> will either go to straight to voicemail or fail. Even worse, the carries wil >> play a fake voicemail and charge you a fee, something that some already a >> are doing when they detect robocallig. > > Don't even think about Transnexus, because they use 302 Redirect with a > header, and no version of Asterisk supports it. I am the only game in the > world for Stir-Shaken and Asterisk. I know it sounds arrogant but it is > literally true. If you need to sign your calls to get through, with Asterisk, > you need to connect to my service. I am an approved Service Provider from the > FCC. If you keep thinking this is not happening, it is, and your business > will disappear overnight. > The issue is that Vicidial, for example, does not provide res_odbc and > func_odbc, so you need to solve that first with Vicidial. Then you can apply > the code I provided earlier and your calls with have a legal, binding > signature. The carriers verify each signature and discard the ones that fail > the cryptography test. Sounds like you're trying to sell/direct people towards a service that you've created. Feel free to do so on the -biz list but the -users list isn't the right place for that sort of thing. Matthew Fredirckson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Stir Shaken
> > There is a big confusion here about Stir Shaken. It is NOT a provider > issue. Un fact, all providers are whasing their hands and modifying their > swihtches to pass-through the Signature. They cannot sign the call because > then the become the responsible party for the call before the FCC, and > liable for any illegal call. Every owner of a PBX that sends calls to the > network, except if you use a trunk for the likes of Vonage, needs to sign > their calls. So if you send calls with any kind of dialer and use DIDs, > real or "borrowed", you need to get the signature service urgently or your > business will stop terminating calls. You cannot self-sign, you cannot get > around it, the calls will either go to straight to voicemail or fail. Even > worse, the carries wil play a fake voicemail and charge you a fee, > something that some already a are doing when they detect robocallig. Don't even think about Transnexus, because they use 302 Redirect with a header, and no version of Asterisk supports it. I am the only game in the world for Stir-Shaken and Asterisk. I know it sounds arrogant but it is literally true. If you need to sign your calls to get through, with Asterisk, you need to connect to my service. I am an approved Service Provider from the FCC. If you keep thinking this is not happening, it is, and your business will disappear overnight. The issue is that Vicidial, for example, does not provide res_odbc and func_odbc, so you need to solve that first with Vicidial. Then you can apply the code I provided earlier and your calls with have a legal, binding signature. The carriers verify each signature and discard the ones that fail the cryptography test. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stir Shaken is upon us
On Mon, Jul 13, 2020 at 3:59 PM Michael Maier wrote: > On 13.07.20 at 10:54 Joshua C. Colp wrote: > > On Sun, Jul 12, 2020 at 11:37 PM Michael Maier > wrote: > >> One more question, > >> what about the pjsip pcap support? Will it be backported to Asterisk 16, > >> too? Would be absolutely cool! Debugging encrypted SIP is really a pain. > >> > > > > It can't be backported ... because it already is. :D This support is > > actually in the latest releases of 13, 16, and 17. > > This is perfectly good news! How often would I have it already needed in > the past! Thanks! > > Just to be sure: > > pjsip set logger pcap (written to /var/lib/asterisk/) > pjsip set logger on (switches on logging to file and console) > pjsip set logger off (switches off logging to file and console) > > Is it possible to log only to the file and not to the console? > The "pjsip set logger verbose off" CLI command can be used to disable verbose messages to the console. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stir Shaken is upon us
On 13.07.20 at 10:54 Joshua C. Colp wrote: > On Sun, Jul 12, 2020 at 11:37 PM Michael Maier wrote: >> One more question, >> what about the pjsip pcap support? Will it be backported to Asterisk 16, >> too? Would be absolutely cool! Debugging encrypted SIP is really a pain. >> > > It can't be backported ... because it already is. :D This support is > actually in the latest releases of 13, 16, and 17. This is perfectly good news! How often would I have it already needed in the past! Thanks! Just to be sure: pjsip set logger pcap (written to /var/lib/asterisk/) pjsip set logger on (switches on logging to file and console) pjsip set logger off (switches off logging to file and console) Is it possible to log only to the file and not to the console? >> >> BTW: what about the (encrypted) RTP packets? Will they be dumped, too? >> > > Not yet supported but certainly something we'd like to see as well as the > RTCP, ICE, STUN, TURN, and DTLS packets. Would be absolutely necessary to debug broken encrypted packets. Thanks Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stir Shaken is upon us
Thanks Josh for clarifying! I'd assumed it would be backported but didnt want to just assume :) Thanks Matt for doing the video! (hint hint theres a load of good content THIS WEEK over on the commcon youtube channel but that's all I'll say about that here) On Mon, Jul 13, 2020 at 9:55 AM Joshua C. Colp wrote: > On Sun, Jul 12, 2020 at 11:37 PM Michael Maier > wrote: > >> On 13.07.20 at 00:17 Joshua C. Colp wrote: >> > On Sun, Jul 12, 2020 at 7:12 PM Dan Jenkins wrote: >> > >> >> Asterisk 18 will have support based on this asterisk update Matt F did >> for >> >> CommCon's sponsor slots >> >> >> >> https://youtu.be/eas1csaX-wc >> >> >> >> >> > As well support will go into Asterisk 16 and 17 as well. It's just been >> > under active development so we've been waiting for that to finish before >> > bringing it back into those versions. >> >> One more question, >> what about the pjsip pcap support? Will it be backported to Asterisk 16, >> too? Would be absolutely cool! Debugging encrypted SIP is really a pain. >> > > It can't be backported ... because it already is. :D This support is > actually in the latest releases of 13, 16, and 17. > > >> >> BTW: what about the (encrypted) RTP packets? Will they be dumped, too? >> > > Not yet supported but certainly something we'd like to see as well as the > RTCP, ICE, STUN, TURN, and DTLS packets. > > -- > Joshua C. Colp > Asterisk Technical Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stir Shaken is upon us
On Sun, Jul 12, 2020 at 11:37 PM Michael Maier wrote: > On 13.07.20 at 00:17 Joshua C. Colp wrote: > > On Sun, Jul 12, 2020 at 7:12 PM Dan Jenkins wrote: > > > >> Asterisk 18 will have support based on this asterisk update Matt F did > for > >> CommCon's sponsor slots > >> > >> https://youtu.be/eas1csaX-wc > >> > >> > > As well support will go into Asterisk 16 and 17 as well. It's just been > > under active development so we've been waiting for that to finish before > > bringing it back into those versions. > > One more question, > what about the pjsip pcap support? Will it be backported to Asterisk 16, > too? Would be absolutely cool! Debugging encrypted SIP is really a pain. > It can't be backported ... because it already is. :D This support is actually in the latest releases of 13, 16, and 17. > > BTW: what about the (encrypted) RTP packets? Will they be dumped, too? > Not yet supported but certainly something we'd like to see as well as the RTCP, ICE, STUN, TURN, and DTLS packets. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
