search for: sip_reinvite

Displaying 3 results from an estimated 3 matches for "sip_reinvite".

2014 Jul 09
1
switching from simple_bridge technology to native_rtp issue
Hi, with canreinvite=no and directmedia=no I and getting the message in the logs for all calls "switching from simple_bridge technology to native_rtp" -- Executing [102 at mkg:1] Dial("SIP/101-00000017", "SIP/102") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/102 -- SIP/102-00000018 is ringing -- SIP/102-00000018 answered SIP/101-00000017
2014 Dec 15
1
T.38 not working - help needed with log interpretation
...below major version 13 any more. Just use 'directmedia'. They are the same setting (snippet from chan_sip's configuration parsing): } else if (!strcasecmp(v->name, "directmedia") || !strcasecmp(v->name, "canreinvite")) { ast_set_flag(&mask[0], SIP_REINVITE); ... Note that these settings and their behaviour is the same from 1.8 through 13. While I'm glad to see anyone using the latest and greatest - yay Asterisk 13! - this isn't a reason to go to Asterisk 13. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis D...
2014 Dec 11
6
T.38 not working - help needed with log interpretation
Hello, at first, thanks for helping! In the meantime, I have done a lot of research and trial and error, and I could solve that specific problem. Obviously, the dialplan application "Answer" was playing a key role here. My original dialplan snippet (which produced that problem) was: exten => _00., 1, NoOp() same => n, Set(FAXOPT(gateway)=yes) same => n,