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coding guidelines, coding guidelines, coding guidelines 

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kevin P. Fleming 2006-11-08 15:28:39 +00:00
parent 5e61077632
commit e108453e02
1 changed files with 18 additions and 15 deletions
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33
channels/chan_sip.c
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@ -2898,24 +2898,27 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
ast_log(LOG_DEBUG, "Outgoing Call for %s\n", p->username);
res = update_call_counter(p, INC_CALL_RINGING);
if ( res != -1 ) {
p->callingpres = ast->cid.cid_pres;
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);
if (res == -1)
return res;
/* Initialize auto-congest time */
p->initid = ast_sched_add(sched, SIP_TRANS_TIMEOUT, auto_congest, p);
}
p->callingpres = ast->cid.cid_pres;
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, SIP_TRANS_TIMEOUT, auto_congest, p);
}
return res;
}