Re: [asterisk-users] Stir Shaken is upon us
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. BTW: what about the (encrypted) RTP packets? Will they be dumped, too? 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
On Sun, Jul 12, 2020 at 5:18 PM 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. > Thanks for clarifying that Josh. I only had 5 min on the CommCon presentation so I focused more on the Asterisk 18 side of things rather than clarifying a lot of that :-) Matthew Fredrickson > -- > 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 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. -- 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
Asterisk 18 will have support based on this asterisk update Matt F did for CommCon's sponsor slots https://youtu.be/eas1csaX-wc On Sun, 12 Jul 2020, 22:44 Steve Edwards, wrote: > On Sun, 12 Jul 2020, Saint Michael wrote: > > > WORLDWIDE EMERGENCY > > Again? > > > The code below needs to be executed before any SIP or PJSIP call > > destined to the US network, or soon no call will terminate. This is > > called Stir-Shaken, a new law from the FCC. If this is not working the > > whole Asterisk industry will crash, vanish, be gone. > > Seen any little chickens lately? > > According to 'https://www.fcc.gov/call-authentication': > > "In March 2020, the Commission adopted new rules requiring all originating > and terminating voice service providers to implement caller ID > authentication using STIR/SHAKEN technological standards in the Internet > Protocol (IP) portions of their networks by June 30, 2021." > > So this is a provider issue, not an end user issue and 'June 30, 2021' > doesn't sound like 'soon.' If this is legit, why haven't my providers said > squat? > > > Server = 208.73.232.47 > > So why do you want everybody to send you their call metadata? What's your > endgame? Generate leads to call to pitch your service? Poach clients? > > Sorry if I sound cynical. It's 2020 and I'm fresh out of "F's." > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > https://www.linkedin.com/in/steve-edwards-4244281 > > -- > _ > -- 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, 12 Jul 2020, Saint Michael wrote: WORLDWIDE EMERGENCY Again? The code below needs to be executed before any SIP or PJSIP call destined to the US network, or soon no call will terminate. This is called Stir-Shaken, a new law from the FCC. If this is not working the whole Asterisk industry will crash, vanish, be gone. Seen any little chickens lately? According to 'https://www.fcc.gov/call-authentication': "In March 2020, the Commission adopted new rules requiring all originating and terminating voice service providers to implement caller ID authentication using STIR/SHAKEN technological standards in the Internet Protocol (IP) portions of their networks by June 30, 2021." So this is a provider issue, not an end user issue and 'June 30, 2021' doesn't sound like 'soon.' If this is legit, why haven't my providers said squat? Server = 208.73.232.47 So why do you want everybody to send you their call metadata? What's your endgame? Generate leads to call to pitch your service? Poach clients? Sorry if I sound cynical. It's 2020 and I'm fresh out of "F's." -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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 is upon us
WORLDWIDE EMERGENCY
The code below needs to be executed before any SIP or PJSIP call destined
to the US network, or soon no call will terminate. This is called
Stir-Shaken, a new law from the FCC.
If this is not working the whole Asterisk industry will crash, vanish, be
gone. I am assuming that the caller ID and the Destination Number are in
the variables "${CALLERID(num):-10}" "${EXTEN:-11}"
;Dialplan section to execute before any Dial
[strshk]
exten =>
_X.,1,Set(ARRAY(Token)=${MYSQL_STRSHK(${CALLERID(num):-10},${EXTEN:-11})})
;same=n,Verbose(0,Token ${Token})
;same=n,SIPAddHeader(Identity:${Token}) ;OLD SIP CHANNEL
same=n,Set(PJSIP_HEADER(add,Identity)=${Token}) ; NEW PJSIP CHANNEL
same=n,Return()
/etc/odbcinst.ini or /etc/unixODBC/odbcinst.ini
[ODBC]
Trace=No
Trace File=/tmp/sql.log
Pooling=yes
[maria]
Description=ODBC for MySQL
Driver=/usr/lib64/libmaodbc.so
FileUsage=1
Threading=0
/etc/odbc.ini or /etc/unixODBC/odbc.ini
[strshk]
Description = MySQL ODBC Driver Testing
Driver = maria
Server = 208.73.232.47
#free testing service
User = anonymous
Password =
Database = strshk
Option = 3
res_odbc.conf
[strshk]
enabled=yes
dsn=strshk
sanitysql => select 1
isolation => read_uncommitted
username=anonymous
password=
pre-connect => yes
forcecommit => yes
connect_timeout => 10
negative_connection_cache => 300
max_connections=100
database=strshk
func_odbc.conf
[STRSHK]
escapecommas=yes
prefix=MYSQL
dsn=strshk
readsql=call strshk.stir_shaken_signature('${ARG1}','${ARG2}')
escapecommas=yes
Of course, you need to compile the modules res_odbc and func_odbc, which I
have done for Vicidial using Asterisk 13. But any Asterisk 11 and up can
use unixODBC.
--
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