Mike Fedyk schrieb:
Elmar Haneke wrote:
The relevant patch should be the change from rev 1.913 to 1.914:
4270c4270
< ast_build_string(a_buf, a_size, "a=fmtp:%d
annexb=no",rtp_code);
---
ast_build_string(a_buf, a_size, "a=fmtp:%d
annexb=no\r\n",rtp_code);
Elmar Haneke wrote:
The relevant patch should be the change from rev 1.913 to 1.914:
4270c4270
< ast_build_string(a_buf, a_size, "a=fmtp:%d annexb=no",rtp_code);
---
ast_build_string(a_buf, a_size, "a=fmtp:%d
annexb=no\r\n",rtp_code);
Please post the patch
Package: asterisk
Version: 1:1.2.0.dfsg-4
Severity: important
Tags: patch
On connecing to an SIP provider for outgoing calls there might be an "Error
400" reported (invalid SIP command).
I was able to fix the problem by replacing chan_sip.c by the current CVS source
(rev 1.917).
The relevant pa
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