Grandstream 813 FXO port

in can change the sip server to the internal address and makes no change

In Asterisk SIP settings for chan_sip, what is the value of Bind Port? The default is 5160.

In the HT, the value for Primary SIP server should be
192.168.1.100:5160
where 192.168.1.100 is the LAN address of the PBX and 5160 is whatever you have in Bind Port.

Please don’t use a domain name; that introduces DNS lookup as another possible failure point.

If you still can’t register, report what, if anything, appears in the Asterisk log for the attempt.

My settings are: (I’ve changed a few so i’m not posting my numbers/secrets)

FreePBX Trunk sip settings:
Trunk name on the sipsettings, outgoing tab must be the same as username below.
username=012345678910
type=peer
secret=ht813
qualify=yes
port=5062 (this is the same as set on HT813)
host=dynamic
dtmfmode=rfc2833
disallow=all
context=from-trunk
allow=ulaw

On HT813:
Basic settings:
Unconditional call forward to voip:
User ID: 012345678910
SIP Server: 192.168.1.100
Port: 5160 (Chan Sip Port in freepbx)

FXO port settings:
Account active: yes
Primary sip server: 192.168.1.100:5160
Outbound proxy: 192.168.1.100:5160
SIP User id: 012345678910
Authenicate id: 012345678910
password: ht813 (whatever you set above in freepbx)
Sip registration: yes
outgoing call without registration: yes
local sip port: 5062 (just make sure its the same as in freepbx trunk settings)
caller id scheme: sin227-bt

Those are my settings and I have it working, apart from disconnect detection. The trunk name in sip settings must be the same as username/user id. I had a few issues with call quality on the newer firmware so i rolled back to 1.0.0.8 and its been fine since.

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