hello all,
i have a problem, and i'm tearing my hair out...any assistance is
appreciated. I am trying to redirect from SER to Asterisk, both on the same
machine. In 1.09 I didnt need to set up a peer for SER, just
autocreatepeer=yes, and rewritehostport from SER as below, and asterisk
accepted the requests without a problem. When I updated to 1.23 requests
from SER to asterisk die quietly, no matter how verbose my asterisk is. It's
as if the requests dont exist at all.
My setup is as follows: asterisk and SER on the same box, SER running on
5060 and asterisk on 5070.
All i want is a simple redirect from SER to asterisk, in ser.cfg thusly:
if (uri == "sip:151@mydomain.com")
{
log(1, "Forwarding to Voicemail\n");
rewritehostport("myIP:5070");
route(1);
break;
}
and in SIP.conf (this is what i have after some hours of trying, but it
doesnt seem to be helping):
bindaddr=myIP
bindport=5070
disallow=all ; Disallow all codecs
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm
dtmfmode=rfc2833
autocreatepeer=yes
insecure=port,invite
[SER]
type=friend
host=myIP
fromdomain=myDomain
context=mycontext
canreinvite=no
insecure=very
if anyone can help i'd me most grateful. I originally thought it would be as
simple as changing "port" to "bindport" in sip.conf. Oh, how
wrong i was.
thanks,
yair
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060130/8af6ea93/attachment.htm
