Displaying 6 results from an estimated 6 matches for "hangup_cause2sip".
2003 Oct 20
3
Call Waiting on SIP phones
...T_USE);
+ } else {
+ find_user(p, INC_IN_USE);
+ }
} else {
char *res;
if (ast->hangupcause && ((res =
hangup_cause2sip
(ast->hangupcause)))) {
@@ -4708,6 +4719,14 @@
if (p->owner)
ast_queue_control(p->owner,
AST_
CONTROL_BUSY, 0);
break;
+ case 487:
+...
2006 Apr 21
0
HANGUPCAUSE on SIP channels
...lpful to know if anyone actually uses this feature
and if it is working properly for them before we go through with fully
debugging and patching this to work for us.
Here is the our test extension from extensions.conf:
exten => 9218,1,Set(HANGUPCAUSE=1)
exten => 9218,2,Hangup
According to hangup_cause2sip in chan_sip.c a HANGUPCAUSE of 1 should
cause Asterisk to reply to the softswitch with a "404 Not Found" SIP
message. That doesn't seem to be the case, however.
Here is a bit of the verbose console output:
(Please note that I added some extra ast_log calls to the source code to
g...
2009 May 12
1
Wanting to manipulate SIP response headers
My boss has asked me if there is a way to send back a 503 response to a
request at will.
I don't see anything in the documents that would allow for manipulation
in asterisk at that low of a level.
Am I wrong?
Bruce Ferrell
2014 Jul 10
0
PJSIP Transfer not working
...hannel.c:2597
ast_softhangup_nolock: Soft-Hanging (0x10) up channel
'PJSIP/Client.1.1.1.1-00000002'
[Jul 9 21:39:29] DEBUG[47716][C-00000002]: channel.c:2753 ast_hangup:
Hanging up channel 'PJSIP/Client.1.1.1.1-00000002'
[Jul 9 21:39:29] DEBUG[47716][C-00000002]: chan_pjsip.c:1578
hangup_cause2sip: AST hangup cause 0 (no match found in PJSIP)
<--- Transmitting SIP response (369 bytes) to UDP:1.1.1.1:49260 --->
SIP/2.0 603 Decline
v: SIP/2.0/UDP 1.1.1.1:49260;rport;received=1.1.1.1;branch=z9hG4bK-d8754z-22994e127365d474-1---d8754z-
i: MmFjNDM4NDc2NmFhZWNiYTU2MDQ1YmNjNGVmYmMyOTY
f: "...
2010 Jan 28
1
Use of "603 Declined"
...ip it appears that
anytime Asterisk wants to tear down a call before it's brought up, it
sends a 603 Declined:
} else { /* Incoming call, not up */
const char *res;
if (p->hangupcause && (res =
hangup_cause2sip(p->hangupcause)))
transmit_response_reliable(p,
res, &p->initreq);
else
transmit_response_reliable(p,
"603 Declined", &p->initreq);...
2016 Aug 15
2
SIP 603 response when call is not answered
Hi
I have noticed that asterisk returns 'SIP 603' when the called party does
not answer.
My test setup is simple: two SIP phones (extensions: 100 and 111)
registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds.
When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL request to
111 (expected) and a '603 Decline' response to 100 (unexpected &