Re: [asterisk-users] Remote UNIX connection / disconnected.
On Sun, 25 Oct 2015, Bryant Zimmerman wrote: Anyone know how to suppress the -- Remote UNIX connection / disconnected messages. I have not tried it, but according to: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/additional_configuration_tasks-asterisk-conf-file.html You can set 'hideconnect = yes' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] repeating TLS error in log file
Greetings, I use TLS and SRTP on all my extensions. I use openssl and distribute my root certificate to my endpoints. Most of the time my calls work just fine. Sometimes I receive a repeating error in my log files however, and I don’t know why this is happening. I’m wondering if this is really from the TLS connection for SIP, or an underlying error with SRTP decoding.. I sometimes get this message in the log when things seems to be working fine. Is there a better way to debug exactly why I’m getting this error? Sometimes I have dozens of these errors in a row. My openssl certificate chain checks out fine with openssl verify command .. [2015-10-26 12:23:42] WARNING[9915] tcptls.c: FILE * open failed! [2015-10-26 12:23:42] VERBOSE[9916] tcptls.c: == Problem setting up ssl connection: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP headers in outofcall messages
Hi, Our custom application sets some SIP headers that we want passed to the called party via asterisk in a simple proxy setup. It works fine for voice calls, but we also use SIP to send outofcall messages. I notice I can't use SIP_HEADER() to get those custom SIP headers in outofcall messages. Is this a bug? I have this in sip.conf: [general] accept_outofcall_message=yes outofcall_message_context=astsms We can send and receive messages between extensions with this without issue. In extensions.conf: [astsms] exten => _X.,1,Set(CHATNOSTORE=${SIP_HEADER(X-Semo-ChatNoStoreForward)}) exten => _X.,n,NoOp(CHATNOSTORE ${CHATNOSTORE}) exten => _X.,n,SIPAddHeader(X-Semo-ChatNoStoreForward: ${CHATNOSTORE}) [snip the parts that actually send on the message] CHATNOSTORE gets set to "", even though I can see the header in a packet trace: 0x0260: 654f 7267 3a20 6865 6c6c 6f0d 0a58 2d53 eOrg:.hello..X-S 0x0270: 656d 6f2d 4368 6174 4e6f 5374 6f72 6546 emo-ChatNoStoreF 0x0280: 6f72 7761 7264 3a20 666f 7277 6172 645f orward:.forward_ 0x0290: 7661 6c75 650d 0a0d 0a54 6573 7420 6d65 valueTest.me 0x02a0: 7373 6167 6520 31 ssage.1 I'm not sure where to take this next... dive into the code for SIP_HEADER? Thanks, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calendar integration : Could not authenticate to server: rejected Basic challenge
try ical url caldav switched to Oauth https://blog.mozilla.org/calendar/2013/09/google-is-changing-the-location-url-of-their-caldav-calendars/ and this looks like you must use Oauth 2.0 https://developers.google.com/google-apps/calendar/caldav/v2/guide Dne 26.10.2015 v 12:17 Jonas Kellens napsal(a): Hello I find very little feedback on the following warning/error when trying to connect to Google calendar : [Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:118 auth_credentials: Invalid username or password for CalDAV calendar 'cal1' [Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:157 caldav_request: Unknown response to CalDAV calendar cal1, request REPORT to /calendar/dav/i...@mydomain.tld/events/: Could not authenticate to server: rejected Basic challenge [cal1] type = caldav url = https://www.google.com/calendar/dav/i...@mydomain.tld/events/ user = i...@mydomain.tld secret = mysecret refresh = 15 timeframe = 60 When I go to the URL https://www.google.com/calendar/dav/i...@mydomain.tld/events/ I can log in with the credentials to the calendar (and get a download window for the calendar file). So it seems not a problem of authentication to me. But what then could be the real issue here ? Thanks Kind regards Jonas. -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calendar integration : Could not authenticate to server: rejected Basic challenge
Hello I find very little feedback on the following warning/error when trying to connect to Google calendar : [Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:118 auth_credentials: Invalid username or password for CalDAV calendar 'cal1' [Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:157 caldav_request: Unknown response to CalDAV calendar cal1, request REPORT to /calendar/dav/i...@mydomain.tld/events/: Could not authenticate to server: rejected Basic challenge [cal1] type = caldav url = https://www.google.com/calendar/dav/i...@mydomain.tld/events/ user = i...@mydomain.tld secret = mysecret refresh = 15 timeframe = 60 When I go to the URL https://www.google.com/calendar/dav/i...@mydomain.tld/events/ I can log in with the credentials to the calendar (and get a download window for the calendar file). So it seems not a problem of authentication to me. But what then could be the real issue here ? Thanks Kind regards Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Function_CHANNEL how to get source ip address in dial plan?
Hi, I using PJSIP as sip driver, I wound like to get source IP on inbound calls from my peers, tried use Function_CHANNEL like ${CHANNEL(pjsip,type,remote_addr)} but it returns only empty string, what I doing wrong? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users