Fix sip show channel
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@717 f38db490-d61c-443f-a65b-d21fe96a405b
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@ -2990,8 +2990,8 @@ static int sip_show_channel(int fd, int argc, char *argv[])
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while(cur) {
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if (!strcasecmp(cur->callid, argv[3])) {
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ast_cli(fd, "Call-ID: %s\n", cur->callid);
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ast_cli(fd, "Codec Capability: %s\n", cur->capability);
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ast_cli(fd, "Non-Codec Capability: %s\n", cur->noncodeccapability);
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ast_cli(fd, "Codec Capability: %d\n", cur->capability);
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ast_cli(fd, "Non-Codec Capability: %d\n", cur->noncodeccapability);
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ast_cli(fd, "Theoretical Address: %s:%d\n", inet_ntoa(cur->sa.sin_addr), ntohs(cur->sa.sin_port));
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ast_cli(fd, "Received Address: %s:%d\n", inet_ntoa(cur->recv.sin_addr), ntohs(cur->recv.sin_port));
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ast_cli(fd, "NAT Support: %s\n", cur->nat ? "Yes" : "No");
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