mirror of
https://github.com/asterisk/asterisk.git
synced 2025-09-03 11:25:35 +00:00
Setup DSP when SS7 call is connected or early media is available.
Outgoing SS7 calls fail to detect incoming DTMF so any bridged channel that requires out-of-band DTMF will not work. * Added sig_ss7_open_media() calls at appropriate places in sig_ss7.c. The new call converts conditionaled out unconverted code and shows that the code really did something useful. * Improved some chan_dahdi DTMF debug messages to help track DTMF handling. (closes issue ASTERISK-19312) Reported by: Igor Nikolaev ........ Merged revisions 358260 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358261 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
@@ -151,6 +151,22 @@ static void sig_ss7_set_remotelyblocked(struct sig_ss7_chan *p, int is_blocked)
|
||||
}
|
||||
}
|
||||
|
||||
/*!
|
||||
* \internal
|
||||
* \brief Open the SS7 channel media path.
|
||||
* \since 1.8.12
|
||||
*
|
||||
* \param p Channel private control structure.
|
||||
*
|
||||
* \return Nothing
|
||||
*/
|
||||
static void sig_ss7_open_media(struct sig_ss7_chan *p)
|
||||
{
|
||||
if (p->calls->open_media) {
|
||||
p->calls->open_media(p->chan_pvt);
|
||||
}
|
||||
}
|
||||
|
||||
/*!
|
||||
* \internal
|
||||
* \brief Set the caller id information in the parent module.
|
||||
@@ -807,12 +823,7 @@ void *ss7_linkset(void *data)
|
||||
sig_ss7_queue_control(linkset, chanpos, AST_CONTROL_PROGRESS);
|
||||
p->progress = 1;
|
||||
sig_ss7_set_dialing(p, 0);
|
||||
#if 0 /* This code no longer seems to be necessary so I did not convert it. */
|
||||
if (p->dsp && p->dsp_features) {
|
||||
ast_dsp_set_features(p->dsp, p->dsp_features);
|
||||
p->dsp_features = 0;
|
||||
}
|
||||
#endif
|
||||
sig_ss7_open_media(p);
|
||||
}
|
||||
break;
|
||||
default:
|
||||
@@ -1184,12 +1195,8 @@ void *ss7_linkset(void *data)
|
||||
p->call_level = SIG_SS7_CALL_LEVEL_CONNECT;
|
||||
}
|
||||
sig_ss7_queue_control(linkset, chanpos, AST_CONTROL_ANSWER);
|
||||
#if 0 /* This code no longer seems to be necessary so I did not convert it. */
|
||||
if (p->dsp && p->dsp_features) {
|
||||
ast_dsp_set_features(p->dsp, p->dsp_features);
|
||||
p->dsp_features = 0;
|
||||
}
|
||||
#endif
|
||||
sig_ss7_set_dialing(p, 0);
|
||||
sig_ss7_open_media(p);
|
||||
sig_ss7_set_echocanceller(p, 1);
|
||||
sig_ss7_unlock_private(p);
|
||||
}
|
||||
@@ -1622,6 +1629,7 @@ int sig_ss7_answer(struct sig_ss7_chan *p, struct ast_channel *ast)
|
||||
if (p->call_level < SIG_SS7_CALL_LEVEL_CONNECT) {
|
||||
p->call_level = SIG_SS7_CALL_LEVEL_CONNECT;
|
||||
}
|
||||
sig_ss7_open_media(p);
|
||||
res = isup_anm(p->ss7->ss7, p->ss7call);
|
||||
ss7_rel(p->ss7);
|
||||
return res;
|
||||
|
Reference in New Issue
Block a user