diff options
| -rw-r--r-- | channels/chan_sip.c | 39 |
1 files changed, 19 insertions, 20 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index a255c0d4c..99f9e87aa 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -1269,7 +1269,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate, char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, int debug); -static int add_sdp(struct sip_request *resp, struct sip_pvt *p); +static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p); /*--- Authentication stuff */ static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len); @@ -2783,16 +2783,21 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout) res = update_call_counter(p, INC_CALL_RINGING); if ( res != -1 ) { p->callingpres = ast->cid.cid_pres; - p->jointcapability = p->capability; - p->t38.jointcapability = p->t38.capability; - if (option_debug) - ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability); - transmit_invite(p, SIP_INVITE, 1, 2); - if (p->maxtime) + p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec); + + /* If there are no audio formats left to offer, punt */ + if (!(p->jointcapability & AST_FORMAT_AUDIO_MASK)) { + ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username); + res = -1; + } else { + p->t38.jointcapability = p->t38.capability; + if (option_debug > 1) + ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability); + transmit_invite(p, SIP_INVITE, 1, 2); + /* Initialize auto-congest time */ - p->initid = ast_sched_add(sched, p->maxtime * 4, auto_congest, p); - else - p->initid = ast_sched_add(sched, SIP_TRANS_TIMEOUT, auto_congest, p); + p->initid = ast_sched_add(sched, p->maxtime ? (p->maxtime * 4) : SIP_TRANS_TIMEOUT, auto_congest, p); + } } return res; } @@ -5966,7 +5971,7 @@ static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_ } /*! \brief Add Session Description Protocol message */ -static int add_sdp(struct sip_request *resp, struct sip_pvt *p) +static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) { int len = 0; int alreadysent = 0; @@ -6008,7 +6013,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p) if (!p->rtp) { ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n"); - return -1; + return AST_FAILURE; } /* Set RTP Session ID and version */ @@ -6032,14 +6037,8 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p) dest.sin_port = sin.sin_port; } - /* Ok, let's start working with codec selection here */ - capability = ast_translate_available_formats(p->jointcapability, p->prefcodec); + capability = p->jointcapability; - /* If there are no audio formats left to offer, punt */ - if (!(capability & AST_FORMAT_AUDIO_MASK)) { - ast_log(LOG_WARNING, "No audio format found to offer.\n"); - return -1; - } if (option_debug > 1) { char codecbuf[BUFSIZ]; @@ -6221,7 +6220,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p) ast_log(LOG_DEBUG, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, BUFSIZ, capability)); } - return 0; + return AST_SUCCESS; } /*! \brief Used for 200 OK and 183 early media */ |