[Spce-user] sipwise:CE 2.5 - peer registration problem

[Michal Biernacki]

Hello Sipwise Community,

I've successfully setup sipwise:ce on dedicated server with static public IP address. For learning purposes I've created several subscribers and I can setup internal calls. Right now I would like to connect to "real" world so I have to setup SIP peers. I have two types of SIP trunks:

1. A SIP trunk which requires registration. In Asterisk I have the following configuration in sip.conf

register => fromuser at fromdomain:secret:authuser at host

[peer1]
disallow=all
type=friend
secret=mysecretpassword
defaultuser=myauthuser
fromuser=224525000		; I've 1000 numbers - this one is the first
host=87.204.129.4
dtmfmode=rfc2833
qualify=no
canreinvite=no
allow=alaw,ulaw
insecure=invite,port
fromdomain=hexatelecom.integralnet.pl
context=custom-get-did-from-sip
sendrpid=yes

2. A SIP trunk which doesn't require any password (IP based authentication)

[peer2]
disallow=all
type=peer
host=195.162.16.200
dtmfmode=rfc2833
qualify=no
canreinvite=no
allow=alaw,ulaw
insecure=invite,port
context=custom-get-did-from-sip
sendrpid=yes

Both examples work under Asterisk. Unfortunately I can replicate this configuration with sipwise:ce. I've created:

1. Peering contract (for billing).
2. Peering group.
3. Two peering servers:
	a. Name = test-gw-1, IP = 87.204.129.4, peer_auth_user = myauthuser, peer_auth_pass = mysecretpassword, peer_auth_realm = hexatelecom.integralnet.pl, peer_auth_register = 1
	b. Name = test-gw-2, IP = 195.162.16.200, peer_auth_user = <empty, peer_auth_pass = <empty>, peer_auth_realm = <empty>, peer_auth_register = 0
4. According to the handbook I've also modified /etc/ngcp-config/templates/etc/sems/etc/reg_agent.conf.tt2 and run ngcpcfg apply. Now I've:

domain=hexatelecom.integralnet.pl
user=myauthuser
pwd=mysecretpassword
proxy=sip:127.0.0.1:5060
contact=sip:my_external_ip:5060

domain2=
user2=
pwd2=
proxy2=sip:127.0.0.1:5060
contact2=sip:my_external_ip:5060

Unfortunately I can't answer incoming call, I can't make outgoing calls :( Any ideas? I really like sipwise:ce solution and I'm thinking about upgrading to :pro but I would like to make some tests :)

Kind regards,
Michal