Hello,I need to use a SIP account to receive calls on an extension of my Asterisk. I am using the AsterisNow, and set up through the FreePBX.
If I set up my account at any Voip device I can receive calls without problems, but with account configured on Asterik not receive calls. When making calls with SIP same account I have no problems, only to receive is the problem.
Then describe the configuration as I have in the form of FreePBX, the option Trunks:
Outgoing Settings:
Trunk-Name: Voip-Stunt
Peer Details:
username=user
type=friend
secret=pass
qualify=1000
nat=yes
insecure=port,invite
host=sip.voipstunt.com
fromdomain=sip.voipstunt.com
dtmfmode=inband
disallow=all
canreinvite=no
allow=ulaw&alaw&gsm&g722&g726
Incoming Settings:
USER Context: enter
USER Details:
user=user
type=peer
secret=pass
qualify=yes
insecure=port,invite
host=sip.voipstunt.com
context=from-trunk
Register String:
user:pass@sip.voipstunt.com
I clarify that I test with a route incoming calls to take all regardless of the DID to rule out a problem at that stage.
I have set in NAT Settings for SIP, my public IP and the local network.
Thanks for your time.
This incoming and outgoing thing seems to be a FreePBX thing, but you need to make sure that all the options needed for incoming calls are set on the outgoing device entry as Asterisk itself may well match that one, as it will normally do a peer match (use of friend here is almost certain, at best, redundant, and at worst, a security risk) and the order in which the two entries are tried is not necessarily the obvious one.
Also note that canreinvite is deprecated and may no longer have any effect.
If you are able to log onto asterisk CLI(Backend) type, asterisk -r | grep {DDI number}, hit enter and call the DDI. You should see if you are missing a context
hi, thanks for your help.
I made many changes in the parameters, for reference put this:
PEER
username=user
type=friend
secret=pass
qualify=1000
nat=yes
insecure=port,invite
host=sip.voipstunt.com
fromdomain=sip.voipstunt.com
dtmfmode=inband
disallow=all
allow=ulaw&alaw&gsm&g722&g726
USER
user=user
host=sip.voipstunt.com
type=friend
secret=pass
context=from-trunk
insecure=invite
But I still can not make incoming calls.
Look SIP communications that come to my server and nothing appears when I call.
The trunk is recorded in this way:
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
enter/nexttics 77.72.169.134 No No 5060 Unmonitored
entrante/nexttics 77.72.169.129 No No 5060 OK (390 ms)
Any suggestions?
Thanks!!!