Sipgate and inbound route
Hello Peeps,
I currently have an outgoing sipgate account configured and working through my trixbox however after hours of looking at configs etc I am still unable to get inbound calls working.
I have created my inbound route using my sipgate account as the DID Number. I was getting the womens voice about the number not being availble so I know its getting to the TB... however after alot of messing about with sip.conf I cant even get that far anymore just seems to cut me off as soon as I dial my 0845 number. I have opened the ports in my firewall router.
Below is the current configuration of sip.conf:
; do not edit this file, this is an auto-generated file by freepbx
; all modifications must be done from the web gui
[general]
;
; enable and force the sip jitterbuffer. If these settings are desired
; they should be set in the sip_general_custom.conf file as this file
; will get overwritten during reloads and upgrades.
;
; jbenable=yes
; jbforce=yes
; These will all be included in the [general] context
;
#include sip_general_additional.conf
#include sip_general_custom.conf
#include sip_nat.conf
#include sip_registrations_custom.conf
#include sip_registrations.conf
; These should all be expected to come after the [general] context
;
#include sip_custom.conf
#include sip_additional.conf
#include sip_custom_post.conf
I hope this is enough to give you an idea.
Look foward to some help.
Kind Regards
Sam
Sam,
It might be your registration string in the FreePBX trunk page, mine is:
1234567:1A1A1A1A@sipgate.co.uk/01212121212
Or the Peer details on the same page:
username=1234567
type=peer
secret=1A1A1A1A
insecure=very
host=sipgate.co.uk
fromuser=1234567
fromdomain=sipgate.co.uk
dtmfmode=rfc2833
disallow=all
context=from-pstn
canreinvite=no
authuser=1234567
allow=alaw&ulaw&gsm
These are mine with the secret stuff changed. They work fine with Sipgate.
I didn't touch sip.conf as far as I remember, most of it gets changed through FreePBX.
Cheers
Steve
My Trunk Incoming Settings
USERCONTEXT: sipgate-in
USER DETAILS:
context=from-trunk
type=friend
host=sipgate.co.uk
fromdomain=sipgate.co.uk
fromuser=SIPGATE-ID
secret=SIPGATE-PASS
insecure=very
qualify=yes
username=SIPGATE-ID
Also I have any DID / any CID to Ring Groups...
Allow Anonymous Inbound SIP Calls? Yes
Simulate Incoming Call works fine... But when I try to call from my home phone or voipcheap account I just get disconnected... Nothing happens... The main problem is that why everything worked fine and now stopped to works... Any help please?


Member Since:
2006-06-29