[Asterisk-Users] SIP Retransmission Patch

[Andre Bierwirth]

This Helps :)

Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.16
diff -c -r1.16 chan_sip.c
*** chan_sip.c 29 Mar 2003 00:42:16 -0000 1.16
--- chan_sip.c 29 Mar 2003 02:20:46 -0000
***************
*** 3314,3319 ****
--- 3314,3323 ----
       process_sdp(p, req);
      if (p->owner) {
       if (p->owner->_state != AST_STATE_UP) {
+                                if (p->autokillid > -1) {
+                                    ast_sched_del(sched , p->autokillid);
+        p->autokillid = -1;
+       }
        ast_setstate(p->owner, AST_STATE_UP);
        ast_queue_control(p->owner, AST_CONTROL_ANSWER, 0);
       }
***************
*** 3641,3646 ****
--- 3645,3656 ----
       strncpy(p->exten, "s", sizeof(p->exten) - 1);
      /* Initialize tag */
      p->tag = rand();
+
+                                 if (p->autokillid > -1) {
+                                     ast_sched_del(sched , p->autokillid);
+                                     p->autokillid = -1;
+                                 }
+
      /* First invitation */
      c = sip_new(p, AST_STATE_DOWN, strlen(p->username) ? p->username :
NULL);
     }


----- Original Message -----
From: "Luke Howard" <lukeh at PADL.COM>
To: <asterisk-users at lists.digium.com>
Sent: Saturday, March 29, 2003 2:17 AM
Subject: Re: [Asterisk-Users] SIP Retransmission


>
> >I can confirm that.
> >After 10/15 seconds, outgoing sip calls are
> >destroyed.
>
> Interestingly, I'm seeing incoming SIP calls destroyed after 10-15
> seconds, and outgoing SIP calls destroyed almost immediately.
>
> -- Luke
>
> --
> Luke Howard | PADL Software Pty Ltd | www.padl.com
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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