[asterisk-users] Outgoing PJSIP using Kamailio

[Administrator]

Hello,

We have a provider which is using Kamailio as front end. Our asterisk 
13/chan_sip server has no problem to register and pass/receive calls 
form this provider.

Now we want to move to asterisk 16/pjsip and face problem. Registration 
is OK but when we pass a call our INVITE never receive answer from the 
provider. We opened a ticket to their support but in the mean time we 
want to know if someone is using successfully a PJSIP channel against 
Kamailio.

Another one: despite the fact that they use 5061 port, it's not TLS but 
UDP. Our asterisk16 has no TLS configured.

We use wizard which looks like:

[Provider-tootai](!)
;
type = wizard
sends_auth = yes
sends_registrations = yes
accepts_auth = no
accepts_registrations = no
endpoint/call_group = 1
endpoint/pickup_group = 1
endpoint/accountcode = TOOTAi
endpoint/language = fr
endpoint/allow = !all,ulaw,alaw,g729
endpoint/context = incoming-Provider
endpoint/direct_media = no
endpoint/dtmf_mode = inband
registration/retry_interval = 20
registration/max_retries = 0
registration/expiration = 3600
registration/transport = transport-udp
aor/max_contacts = 2
aor/qualify_frequency = 2000

[Provider](Provider-tootai)
;
remote_hosts = sips.provider.eu
endpoint/callerid = "TOOTAi" <00xx xxx xxx xxx>
aor/contact = sip:sips.provider.eu:5061
registration/client_uri = sips:OUR_ID at sips.provider.eu
registration/server_uri = sips:sips.provider.eu:5061
outbound_auth/username = OUR_ID
outbound_auth/password = OUR_PWD
identity/match = PROVIDER_IP

Thanks for any hint

-- 
Daniel