Displaying 3 results from an estimated 3 matches for "sip_reinvit".
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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...
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,