Fix typo

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1301 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b847ef9..a1b9adc 100755 (executable)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1820,7 +1820,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 static int add_header(struct sip_request *req, char *var, char *value)
 {
        if (req->len >= sizeof(req->data) - 4) {
-               ast_log(LOG_WARNING, "Out of space, can't add anymore\n");
+               ast_log(LOG_WARNING, "Out of space, can't add anymore (%s:%s)\n", var, value);
                return -1;
        }
        if (req->lines) {
@@ -2087,6 +2087,7 @@ static int respprep(struct sip_request *resp, struct sip_pvt *p, char *msg, stru
        copy_header(resp, req, "Call-ID");
        copy_header(resp, req, "CSeq");
        add_header(resp, "User-Agent", "Asterisk PBX");
+       add_header(resp, "Allow", ALLOWED_METHODS);
        if (p->expiry) {
                /* For registration responses, we also need expiry and
                   contact info */
@@ -2225,7 +2226,6 @@ static int transmit_response_with_allow(struct sip_pvt *p, char *msg, struct sip
 {
        struct sip_request resp;
        respprep(&resp, p, msg, req);
-       add_header(&resp, "Allow", ALLOWED_METHODS);
        add_header(&resp, "Accept", "application/sdp");
        add_header(&resp, "Content-Length", "0");
        add_blank_header(&resp);
@@ -2517,6 +2517,7 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p, struct ast_rtp *rtp, st
                reqprep(&req, p, "UPDATE", 0);
        else
                reqprep(&req, p, "INVITE", 0);
+       add_header(&req, "Allow", ALLOWED_METHODS);
        add_sdp(&req, p, rtp, vrtp);
        /* Use this as the basis */
        copy_request(&p->initreq, &req);