[asterisk-users] SIP peer authentication
I have two asterisk boxes connected via SIP protocol. I want to deploy SIP peer authentication to this connection. What is the needed configuration?I have the following configuration but changing username and secret does not affect the connection at all! sip.conf in box 68:[general] t38pt_udptl=yes,none,maxdatagram=400 [p68] host=192.168.0.68 type=peer username=test secret=testtest context=from-trunk qualify=yes insecure=port,invite canreinvite=no sip.conf in box 67:[general] t38pt_udptl=yes,none,maxdatagram=400 [p67] host=192.168.0.67 type=peer username=test secret=testtest context=from-trunk qualify=yes insecure=port,invite canreinvite=no Thanks in advance -- _ -- 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] sms from softphones
hi. in my asterisk i do have a usb 3g dongle, that i am using it for GSM calls and sms. At the moment all incoming sms is going to email. outgoing sms is through the asterisk console: dongle sms dongle0 mobile_number Hello I would like to ask if it is possible to use my softphones (zoiper) to send sms through the dongle. If yes, how? -- _ -- 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] Asterisk 13.13.1 use_callids = yes Extensions ID as CallerID
Hello, In Master.csv Asterisk is loggin the Company ID set in Extensions.conf, but I configured logger.conf to log the EXT ID. For instance, the SRC in the following line should be my ext. number. Does it make sense? From my extension 4007 I called 78079745, yet in the log below the first number is 2318001800 which is the main company's number set in Extensions.conf. 2318001800 78079745 phones "ITadmin" <2318001800> SIP/4007-00015c0a SIP/voip1-00015c0b Dial SIP/voip1/78079745,80 4/3/2017 15:30 4/3/2017 15:31 2 0 NO ANSWER DOCUMENTATION 1.49E+09 Logger.conf [general] dateformat=%F %T ; ; Customize the display of debug message time stamps ; this example is the ISO 8601 date format (-mm-dd HH:MM:SS) ; ; see strftime(3) Linux manual for format specifiers. Note that there is also ; a fractional second parameter which may be used in this field. Use %1q ; for tenths, %2q for hundredths, etc. ; ;dateformat=%F %T ; ISO 8601 date format ;dateformat=%F %T.%3q ; with milliseconds dateformat = %F %T.%3q ; ISO 8601 date format with milliseconds ; ; ; This makes Asterisk write callids to log messages ; (defaults to yes) use_callids = yes -- _ -- 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] 100% CPU after upgrade.
Those are all rational questions, so here we go: We upgraded from 11.x, though the system was a backup server, so it was never actually used. The system is a 2.4Gh quad-core Xenon with 4G of RAM, so it should have plenty of power for what I'm asking it to do. The system is configured via RT using a local Mysql database. We only use the native SIP channel driver at this time. I honestly don't see any reason for this server to eat 100% of it's cpu, and am hesitant to roll it out to production until I understand why it is. Once again, any suggestions will be welcome. Thanks, Mike Diehl. On Friday, March 31, 2017 01:51:07 PM Matt Fredrickson wrote: > One thing you didn't mention was what version you previously upgraded > from... Also, more information about the system in general would > help. (Endpoints, is it realtime or flat file configured, if > realtime, what type of database, what channel drivers (SIP or PJSIP, > and others). > > Matthew Fredrickson > > On Fri, Mar 31, 2017 at 12:08 PM, Mike Diehl wrote: > > Hi all, > > > > I've upgraded to Asterisk 13.14.0 and now I'm seeing that Asterisk is using 100% CPU. > > > > I have one AMI agent connected that is acting rationally. I've got a hand full of SIP (RT) registrations. There is no other call activity. > > > > I've tried to unload various modules; nothing resolved the issue. > > > > Any suggestions? > > > > -- > > Mike Diehl > > > > > > > > > > -- > > _ > > -- 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 > > > > -- Mike Diehl -- _ -- 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] Define SIP fromuser field in Dial()-command
Hello
how can I set the fromuser field of the SIP INVITE when using the
Dial()-command in the dialplan ?
None of the below Dial() command give the correct result :
exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz)
exten =>
_XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz/${EXTEN})
exten =>
_XX.,n,Dial(SIP/user762:passwdk5j6::user...@myprovider.biz/${EXTEN})
exten => _XX.,n,Dial(SIP/user762:passwdk...@myprovider.biz/${EXTEN})
The From part of the SIP INVITE always has the EXTEN in it in stead of
the user (user762) :
From: "the_extension" ;tag=as224453ac
How can I get :
From: "the_extension" ;tag=as224453ac
??
I know about sip.conf. That is not the question. My question is clear :
how to set this in dialplan ?
Thank you for the feedback.
Kind regards.
--
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