diff options
Diffstat (limited to 'net/asterisk16/files/dtmf_debug.diff')
| -rw-r--r-- | net/asterisk16/files/dtmf_debug.diff | 78 |
1 files changed, 39 insertions, 39 deletions
diff --git a/net/asterisk16/files/dtmf_debug.diff b/net/asterisk16/files/dtmf_debug.diff index 81205d40dd46..5179d42225fd 100644 --- a/net/asterisk16/files/dtmf_debug.diff +++ b/net/asterisk16/files/dtmf_debug.diff @@ -1,42 +1,3 @@ ---- channels/chan_sip.c.orig 2008-03-18 16:42:59.000000000 +0200 -+++ channels/chan_sip.c 2008-03-18 17:08:34.000000000 +0200 -@@ -3768,6 +3768,7 @@ - ast_log(LOG_WARNING, "old channel wasn't %p but was %p\n", oldchan, p->owner); - else { - p->owner = newchan; -+ ast_rtp_set_chan_name(p->rtp, newchan->name); - ret = 0; - } - if (option_debug > 2) -@@ -4032,6 +4035,7 @@ - if (i->rtp) { - tmp->fds[0] = ast_rtp_fd(i->rtp); - tmp->fds[1] = ast_rtcp_fd(i->rtp); -+ ast_rtp_set_chan_id(i->rtp, i->callid); - } - if (needvideo && i->vrtp) { - tmp->fds[2] = ast_rtp_fd(i->vrtp); -@@ -4059,6 +4063,8 @@ - if (!ast_strlen_zero(i->language)) - ast_string_field_set(tmp, language, i->language); - i->owner = tmp; -+ ast_rtp_set_chan_name(i->rtp, tmp->name); -+ - ast_module_ref(ast_module_info->self); - ast_copy_string(tmp->context, i->context, sizeof(tmp->context)); - /*Since it is valid to have extensions in the dialplan that have unescaped characters in them -@@ -4479,8 +4485,10 @@ - build_via(p); - if (!callid) - build_callid_pvt(p); -- else -+ else { - ast_string_field_set(p, callid, callid); -+ ast_rtp_set_chan_id(p->rtp, p->callid); -+ } - /* Assign default music on hold class */ - ast_string_field_set(p, mohinterpret, default_mohinterpret); - ast_string_field_set(p, mohsuggest, default_mohsuggest); --- include/asterisk/rtp.h.orig 2008-03-18 13:35:42.000000000 +0200 +++ include/asterisk/rtp.h 2008-03-18 13:35:58.000000000 +0200 @@ -251,6 +251,9 @@ @@ -223,3 +184,42 @@ { { "rtp", "debug", "off", NULL }, rtp_no_debug, "Disable RTP debugging", no_debug_usage, NULL, &cli_rtp_no_debug_deprecated }, +--- channels/chan_sip.c.orig 2008-06-10 00:29:41.000000000 -0700 ++++ channels/chan_sip.c 2008-06-10 00:42:00.000000000 -0700 +@@ -3813,6 +3813,7 @@ + ast_log(LOG_WARNING, "old channel wasn't %p but was %p\n", oldchan, p->owner); + else { + p->owner = newchan; ++ ast_rtp_set_chan_name(p->rtp, newchan->name); + /* Re-invite RTP back to Asterisk. Needed if channel is masqueraded out of a native + RTP bridge (i.e., RTP not going through Asterisk): RTP bridge code might not be + able to do this if the masquerade happens before the bridge breaks (e.g., AMI +@@ -4085,6 +4086,7 @@ + if (i->rtp) { + tmp->fds[0] = ast_rtp_fd(i->rtp); + tmp->fds[1] = ast_rtcp_fd(i->rtp); ++ ast_rtp_set_chan_id(i->rtp, i->callid); + } + if (needvideo && i->vrtp) { + tmp->fds[2] = ast_rtp_fd(i->vrtp); +@@ -4112,6 +4114,8 @@ + if (!ast_strlen_zero(i->language)) + ast_string_field_set(tmp, language, i->language); + i->owner = tmp; ++ ast_rtp_set_chan_name(i->rtp, tmp->name); ++ + ast_module_ref(ast_module_info->self); + ast_copy_string(tmp->context, i->context, sizeof(tmp->context)); + /*Since it is valid to have extensions in the dialplan that have unescaped characters in them +@@ -4531,8 +4535,10 @@ + build_via(p); + if (!callid) + build_callid_pvt(p); +- else ++ else { + ast_string_field_set(p, callid, callid); ++ ast_rtp_set_chan_id(p->rtp, p->callid); ++ } + /* Assign default music on hold class */ + ast_string_field_set(p, mohinterpret, default_mohinterpret); + ast_string_field_set(p, mohsuggest, default_mohsuggest); |