***************
/* Make sure it's a SIP URL */
if (strncasecmp(c, "sip:", 4)) {
ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", c);
} else
c += 4;
strncpy(pvt->okcontacturi, c, sizeof(pvt->okcontacturi) - 1);
***************
to:
*************** strncpy(pvt->okcontacturi, c, sizeof(pvt->okcontacturi) - 1);
/* Make sure it's a SIP URL */
if (strncasecmp(c, "sip:", 4)) {
ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", c);
} else
c += 4;
***************
(sorry... I can't provide a unified diff... I back-ported the fix into an older copy of CVS...)
----- Original Message ----- From: "Matt Hess" <[EMAIL PROTECTED]>
To: "Asterisk Developers Mailing List" <[email protected]>
Sent: Monday, December 20, 2004 12:39 AM
Subject: [Asterisk-Dev] stable chan_sip borked
The latest stable version of chan_sip (1.510.2.27) breaks almost all of our stuff.. cisco ata devices no longer work for audio.. most of our voip terminators also spit back a lot of errors.. reverted back to cvs -D2004-12-10 and things were happy again.
just fyi.
--------------------------------------------------------------------------------
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