[asterisk-dev] Bounty: SIP conferencing fix

[Brian Jones]

I have a very small patch that was originally written by Mark Spencer
that allows me to use Asterisk as a conference bridge with a Tekelec
T6000.  It works great for ad-hoc conferencing, but fails randomly for
meet me conferencing.  Here is the patch for the 1.2 release versions:

--- chan_sip.c  2005-11-17 11:56:25.000000000 -0600
+++ chan_sip.c.conference       2005-11-17 11:44:46.000000000 -0600
@@ -6520,6 +6520,11 @@
                return -1;
        }
        uri += 4;
+       if (!strncasecmp(uri, "conf=", 5)) {
+               uri+=5;
+               ast_log(LOG_DEBUG, "Detected conference '%s'\n", uri);
+               strncpy(p->context, "conference", sizeof(p->context) -
1);
+       }
        if (!ast_strlen_zero(from)) {
                if (strncmp(from, "sip:", 4)) {
                        ast_log(LOG_WARNING, "Huh?  Not a SIP header
(%s)?\n", from);

In extensions.conf I have:

[conference]

exten => _.,1,answer
exten => _.,2,Set(TIMEOUT(absolute)=10800)
exten => _.,3,MeetMe(${EXTEN},dq)

As you can see from the patch, the invite begins with "conf=" and the a
long oid that asterisk uses to dynamically create the conference (the
"extension"). The T6000 conferences the first two people in with no
problem, but when the thrid call comes in, asterisk responds to the
invite with a "503 Server Error."

I'm offering a $500 bounty to the first person to fix this for me, $750
if I get the patch by this weekend.

Thanks,
----------------------------
Brian Jones
VP of Technology
bjones at kancharla.com
256-705-5012
----------------------------