Add support for changing the outbound codec on a SIP call using

a dialplan variable.

This adds a dialplan variable (SIP_CODEC_OUTBOUND) which controls
the codec offered for an outgoing SIP call. This is much like the
SIP_CODEC dialplan variable and has the same restrictions. The codec
set must be one that is configured for the call.

(closes issue #13243)
Reported by: samdell3
Patches:
      13243.diff uploaded by file (license 11)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186624 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/CHANGES b/CHANGES
index a9205024fa6a6b43830bec8f7bb2bf41a6ef328c..9397ae1047514003ccb47210a0aa7a04eeb0d819 100644 (file)
--- a/CHANGES
+++ b/CHANGES
@@ -15,6 +15,9 @@ SIP Changes
 -----------
  * Added preferred_codec_only option in sip.conf. This feature limits the joint
    codecs sent in response to an INVITE to the single most preferred codec.
+ * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
+   to be used for the outgoing call. It must be one of the codecs configured
+   for the device.
 
 Applications
 ------------

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 10883c5f32c3ed05208655f4512ff36a05dad6c2..17b7c8cbba41b58a6725f3bff89fdcfb969402bf 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5836,7 +5836,12 @@ static void try_suggested_sip_codec(struct sip_pvt *p)
        int fmt;
        const char *codec;
 
-       codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC");
+       if (p->outgoing_call) {
+               codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC_OUTBOUND");
+       } else if (!(codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC_INBOUND"))) {
+               codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC");
+       }
+
        if (!codec) 
                return;
 
@@ -9838,6 +9843,7 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version, int old
 
        if (p->do_history)
                append_history(p, "ReInv", "Re-invite sent");
+       try_suggested_sip_codec(p);
        if (t38version)
                add_sdp(&req, p, oldsdp, FALSE, TRUE);
        else
@@ -10199,8 +10205,10 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
                        ast_udptl_offered_from_local(p->udptl, 1);
                        ast_debug(1, "T38 is in state %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
                        add_sdp(&req, p, FALSE, FALSE, TRUE);
-               } else if (p->rtp) 
+               } else if (p->rtp) {
+                       try_suggested_sip_codec(p);
                        add_sdp(&req, p, FALSE, TRUE, FALSE);
+               }
        } else {
                if (!p->notify_headers) {
                        add_header_contentLength(&req, 0);

diff --git a/doc/tex/channelvariables.tex b/doc/tex/channelvariables.tex
index 4c3195246d867e397d38b3fa9c6db5368e057e50..4c7b4d5ff7cf9bb0079424a24e19df024bbec31d 100644 (file)
--- a/doc/tex/channelvariables.tex
+++ b/doc/tex/channelvariables.tex
@@ -925,7 +925,9 @@ ${SIPDOMAIN}         * SIP destination domain of an inbound call (if appropriate
 ${SIPFROMDOMAIN}       Set SIP domain on outbound calls
 ${SIPUSERAGENT}      * SIP user agent (deprecated)
 ${SIPURI}            * SIP uri
-${SIP_CODEC}           Set the SIP codec for a call
+${SIP_CODEC}           Set the SIP codec for an inbound call
+${SIP_CODEC_INBOUND}   Set the SIP codec for an inbound call
+${SIP_CODEC_OUTBOUND}  Set the SIP codec for an outbound call
 ${SIP_URI_OPTIONS}   * additional options to add to the URI for an outgoing call
 ${RTPAUDIOQOS}         RTCP QoS report for the audio of this call
 ${RTPVIDEOQOS}         RTCP QoS report for the video of this call