diff options
| author | Olle Johansson <oej@edvina.net> | 2006-04-07 19:46:50 +0000 |
|---|---|---|
| committer | Olle Johansson <oej@edvina.net> | 2006-04-07 19:46:50 +0000 |
| commit | aefba4ad7ddffb6856078a3b37e3a72d683d4fd9 (patch) | |
| tree | f192a87327d614d008d66d4f0ba6058f43379906 | |
| parent | f235bbe5a5da93a3c1a1c87da1b26f3aaa593bf0 (diff) | |
Add history events for re-invites
(need to nail this issue...)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
| -rw-r--r-- | channels/chan_sip.c | 7 |
1 files changed, 6 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 63b117b0a..85ec6dd03 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -2078,7 +2078,8 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout) res = 0; ast_set_flag(&p->flags[0], SIP_OUTGOING); - ast_log(LOG_DEBUG, "Outgoing Call for %s\n", p->username); + if (option_debug) + ast_log(LOG_DEBUG, "Outgoing Call for %s\n", p->username); res = update_call_counter(p, INC_CALL_LIMIT); if ( res != -1 ) { p->callingpres = ast->cid.cid_pres; @@ -4731,6 +4732,8 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p) add_header(&req, "Allow", ALLOWED_METHODS); if (sipdebug) add_header(&req, "X-asterisk-info", "SIP re-invite (RTP bridge)"); + if (recordhistory) + append_history(p, "%s", "Re-invite sent"); add_sdp(&req, p); /* Use this as the basis */ copy_request(&p->initreq, &req); @@ -10701,6 +10704,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int p->jointcapability = p->capability; ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n"); } + if (recordhistory) /* This is a response, note what it was for */ + append_history(p, "%s", "Re-invite received"); } } else if (debug) ast_verbose("Ignoring this INVITE request\n"); |