Fix SIP_CODEC support
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1729 f38db490-d61c-443f-a65b-d21fe96a405b
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@ -1138,7 +1138,7 @@ static int sip_answer(struct ast_channel *ast)
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fmt=ast_getformatbyname(codec);
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if (fmt) {
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ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n",codec);
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p->capability=fmt;
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p->jointcapability=fmt;
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} else ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n",codec);
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}
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