Ricardo
-
this is an Asterisk
issue.
Configure your
SIP.conf in * creating a sip peer with insecure=very fotr the SER peer, like
this:
[ser-peer]
type =
friend
; should you want to receive and make calls to SER
context=from-ser
; the context in dialplan with extensions allowed to be accessed by SER. Here
you must have PSTN extensions capability.
host=200.XXX.XXX.XXX ; the IP address
for SER
fromdomain=domain.com ;the Domain part of uri to be verified by asterisk
on the INVITE received by SER.
qualify=
yes
; just to check the latency between SER and Asterisk (like this, if over 2000ms
Ast will report as unavailable peer).
disallow=all
allow=alaw
allow=g729
insecure=
very
; this line garantees that any username part of Request URI sent by SER in
INVITE to Asterisk will be accepted by Ast and routed to the
dialplan.
So, if SER send an
INVITE to 5531332818847@domain.com.br ,
Asterisk will look for a section of type =user in SIP conf to match the
user part first, it won't find, then it will look for a type=peer. It will find
and try to match the IP address as in host= ...line and will accept any username
part as per insecure=very line. Then, if context=from-ser in the Ast dialplan
allows this dialing string (553132818847), it will proceed from
there.
Hope it
helps.
At.
Walter