- Change "rfc2833" to "rtp" in sip.conf. Keeping backwards compatibility.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9294 f38db490-d61c-443f-a65b-d21fe96a405b
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@ -7747,7 +7747,7 @@ static const char *dtmfmode2str(int mode)
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{
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switch (mode) {
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case SIP_DTMF_RFC2833:
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return "rfc2833";
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return "rtp";
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case SIP_DTMF_INFO:
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return "info";
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case SIP_DTMF_INBAND:
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@ -11723,7 +11723,7 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
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ast_clear_flag(flags, SIP_DTMF);
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if (!strcasecmp(v->value, "inband"))
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ast_set_flag(flags, SIP_DTMF_INBAND);
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else if (!strcasecmp(v->value, "rfc2833"))
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else if (!strcasecmp(v->value, "rfc2833") || !strcasecmp(v->value, "rtp"))
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ast_set_flag(flags, SIP_DTMF_RFC2833);
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else if (!strcasecmp(v->value, "info"))
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ast_set_flag(flags, SIP_DTMF_INFO);
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@ -12848,7 +12848,7 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
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}
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static char *synopsis_dtmfmode = "Change the dtmfmode for a SIP call";
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static char *descrip_dtmfmode = "SIPDtmfMode(inband|info|rfc2833): Changes the dtmfmode for a SIP call\n";
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static char *descrip_dtmfmode = "SIPDtmfMode(inband|info|rtp): Changes the dtmfmode for a SIP call\n";
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static char *app_dtmfmode = "SIPDtmfMode";
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static char *app_sipaddheader = "SIPAddHeader";
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@ -12871,7 +12871,7 @@ static int sip_dtmfmode(struct ast_channel *chan, void *data)
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if (data)
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mode = (char *)data;
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else {
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ast_log(LOG_WARNING, "This application requires the argument: info, inband, rfc2833\n");
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ast_log(LOG_WARNING, "This application requires the argument: info, inband, rtp\n");
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return 0;
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}
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ast_mutex_lock(&chan->lock);
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@ -12889,10 +12889,10 @@ static int sip_dtmfmode(struct ast_channel *chan, void *data)
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if (!strcasecmp(mode,"info")) {
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ast_clear_flag(p, SIP_DTMF);
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ast_set_flag(p, SIP_DTMF_INFO);
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} else if (!strcasecmp(mode,"rfc2833")) {
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} else if (!strcasecmp(mode, "rfc2833") || !strcasecmp(mode, "rtp")) {
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ast_clear_flag(p, SIP_DTMF);
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ast_set_flag(p, SIP_DTMF_RFC2833);
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} else if (!strcasecmp(mode,"inband")) {
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} else if (!strcasecmp(mode, "inband")) {
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ast_clear_flag(p, SIP_DTMF);
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ast_set_flag(p, SIP_DTMF_INBAND);
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} else
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@ -96,11 +96,12 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; of performing a "hairpin" call.
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;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
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; a valid phone number
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;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
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;dtmfmode = rtp ; Set default dtmfmode for sending DTMF. Default is rtp (according to RFC 2833)
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; Other options:
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; info : SIP INFO messages
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; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
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; auto : Use rfc2833 if offered, inband otherwise
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; info SIP INFO messages
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; inband Inband audio
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; (requires 64 kbit codec -alaw, ulaw)
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; auto : Use rtp if offered, inband otherwise
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;compactheaders = yes ; send compact sip headers.
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;sipdebug = yes ; Turn on SIP debugging by default, from
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@ -412,7 +413,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
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;language=de ; Use German prompts for this user
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;host=dynamic ; This peer register with us
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;dtmfmode=inband ; Choices are inband, rfc2833, or info
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;dtmfmode=inband ; Choices are inband, rtp, or info
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;defaultip=192.168.0.59 ; IP used until peer registers
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;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
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;vmexten=voicemail ; dialplan extension to reach mailbox
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@ -428,7 +429,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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;context=from-sip ; Context for incoming calls from this user
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;secret=blahpoly
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;host=dynamic ; This peer register with us
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;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
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;dtmfmode=rtp ; Choices are inband, rtp, or info
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;username=polly ; Username to use in INVITE until peer registers
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; Normally you do NOT need to set this parameter
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;disallow=all
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