[asterisk-users] VoIP interconnection with Acme packet SBC

Kasun Daminda daminda33 at gmail.com
Thu Oct 22 02:20:01 CDT 2009


Dear all,

I fixed the issue  by myself.
I have edited chan_sip.c file to avoid sdp version gettng increment.
I think this is a bug of asterisk. According to RFCs it should increment it
only it there is change on SDP message body. chan_sip.c alway increase it by
one at every SDP message. I have edited the below part

 /* Set RTP Session ID and version */
 if (!p->sessionid) {
  p->sessionid = getpid();
  p->sessionversion = p->sessionid;
 } else
  p->sessionversion*++*;

As......

 /* Set RTP Session ID and version */
 if (!p->sessionid) {
  p->sessionid = getpid();
  p->sessionversion = p->sessionid;
 } else
  p->sessionversion;

I have removed ++. I am not good programmer. But asterisk lover.
I dont know this is the best solution. However I can receive calls from Acme
packet.

And other important thing to tell is THIS IS NOT A CODEC ISSUE.

thanks everybody

kind Rgds
Daminda
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091022/fe6b580b/attachment.htm 


More information about the asterisk-users mailing list