diff options
author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-10-29 08:45:05 +0000 |
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committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-10-29 08:45:05 +0000 |
commit | 7c6c684d713caf189604448ebed88cd1b32b8e70 (patch) | |
tree | cc823bfce94161d8bb763c58d03f44efe78ca40a /channels | |
parent | 718c960dd41a43afb6dda3aa2e8874cd2bbedd3d (diff) |
Restoring the old logic, since working around it and fixing it seemed too complicated.
- The SIP_OUTGOING flag indicates the direction of the last transaction in the dialog.
- The initreq stores the last request in the dialog, the request that opened the
latest transaction.
Please now retry all the 1.4 bug reports with mixed to/from headers, tags etc
in ACK, BYE, CANCEL. Thanks!
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@46398 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
-rw-r--r-- | channels/chan_sip.c | 12 |
1 files changed, 7 insertions, 5 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 5d3f9f94d..937ecfbc7 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -701,7 +701,7 @@ struct sip_auth { #define SIP_USEREQPHONE (1 << 10) /*!< Add user=phone to numeric URI. Default off */ #define SIP_REALTIME (1 << 11) /*!< Flag for realtime users */ #define SIP_USECLIENTCODE (1 << 12) /*!< Trust X-ClientCode info message */ -#define SIP_OUTGOING (1 << 13) /*!< Is this an outgoing call? */ +#define SIP_OUTGOING (1 << 13) /*!< Direction of the last transaction in this dialog */ #define SIP_CAN_BYE (1 << 14) /*!< Can we send BYE on this dialog? */ #define SIP_DEFER_BYE_ON_TRANSFER (1 << 15) /*!< Do not hangup at first ast_hangup */ #define SIP_DTMF (3 << 16) /*!< DTMF Support: four settings, uses two bits */ @@ -959,7 +959,8 @@ static struct sip_pvt { char lastmsg[256]; /*!< Last Message sent/received */ int amaflags; /*!< AMA Flags */ int pendinginvite; /*!< Any pending invite ? (seqno of this) */ - struct sip_request initreq; /*!< Initial request that opened the SIP dialog */ + struct sip_request initreq; /*!< Request that opened the latest transaction + within this SIP dialog */ int maxtime; /*!< Max time for first response */ int initid; /*!< Auto-congest ID if appropriate (scheduler) */ @@ -6339,6 +6340,7 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p) /* Use this as the basis */ initialize_initreq(p, &req); p->lastinvite = p->ocseq; + ast_set_flag(&p->flags[0], SIP_OUTGOING); /* Change direction of this dialog */ return send_request(p, &req, XMIT_CRITICAL, p->ocseq); } @@ -10406,7 +10408,7 @@ static int sip_show_channel(int fd, int argc, char *argv[]) ast_cli(fd, " * Subscription (type: %s)\n", subscription_type2str(cur->subscribed)); else ast_cli(fd, " * SIP Call\n"); - ast_cli(fd, " Direction: %s\n", ast_test_flag(&cur->flags[0], SIP_OUTGOING)?"Outgoing":"Incoming"); + ast_cli(fd, " Curr. trans. direction: %s\n", ast_test_flag(&cur->flags[0], SIP_OUTGOING) ? "Outgoing" : "Incoming"); ast_cli(fd, " Call-ID: %s\n", cur->callid); ast_cli(fd, " Owner channel ID: %s\n", cur->owner ? cur->owner->name : "<none>"); ast_cli(fd, " Our Codec Capability: %d\n", cur->capability); @@ -12993,14 +12995,14 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int p->pendinginvite = seqno; check_via(p, req); + copy_request(&p->initreq, req); /* Save this INVITE as the transaction basis */ if (!p->owner) { /* Not a re-invite */ - /* Use this as the basis */ - copy_request(&p->initreq, req); if (debug) ast_verbose("Using INVITE request as basis request - %s\n", p->callid); append_history(p, "Invite", "New call: %s", p->callid); parse_ok_contact(p, req); } else { /* Re-invite on existing call */ + ast_clear_flag(&p->flags[0], SIP_OUTGOING); /* This is now an inbound dialog */ /* Handle SDP here if we already have an owner */ if (find_sdp(req)) { if (process_sdp(p, req)) { |