Hello. We are in the process of moving from an Openscape Business X5 System to Asterisk. At the moment there are a few dozend DECT endpoints connected to the OSB. We want to try to keep the OSB system just to handle the DECT phones and move the rest of the setup (SIP-endpoints / FAX / trunking to the SIP-provider to asterisk).
Our goal is to make the OSB connected DECT endpoints reachable from Asterisk endpoints and vice versa.
To integrate the OSB into this scenario, we tried to trunk Asterisk to the OSB.
Asterisk IP: 192.168.104.1
OSB IP: 192.168.100.20
We manged to register Asterisk to a SIP-extension on the OSB PBX (SIP-127). SIP-127 is a licensed SIP-endpoint on the OSB. SIP softphones can connect to it.
asterisk*CLI> pjsip show registration unify-registration
<Registration/ServerURI…> <Auth…> <Status…>
unify-registration/sip:192.168.100.20 unify-auth Registered (exp. 109s)
ParameterName : ParameterValue
auth_rejection_permanent : true
client_uri : sip:127@192.168.100.20
contact_header_params :
contact_user :
endpoint :
expiration : 3600
fatal_retry_interval : 0
forbidden_retry_interval : 0
line : false
max_random_initial_delay : 10
max_retries : 10
outbound_auth : unify-auth
outbound_proxy :
retry_interval : 60
security_negotiation : no
server_uri : sip:192.168.100.20
support_outbound : no
support_path : false
transport :
user_agent :
pjsip.conf
[unify-auth]
type = auth
auth_type = userpass
username = SIP-127
password = pw[unify-aor]
type = aor
contact = sip:192.168.100.20:5060[unify-endpoint]
type = endpoint
context = from-unify
allow = !all,g722,ulaw,alaw
outbound_auth = unify-auth
aors = unify-aor
direct_media = no[unify-registration]
type = registration
outbound_auth = unify-auth
server_uri = sip:192.168.100.20
client_uri = sip:127@192.168.100.20
retry_interval = 60[unify-identify]
type = identify
match = 192.168.100.20
endpoint = unify-endpoint
We created a test extension to call one of the DECT devices (184) on the Openscape Business X5:
; Test Asterisk → OSB
exten => 9993,1,Answer()
same => n,Playback(beep)
same => n,Dial(PJSIP/184@unify-endpoint)
same => n,Hangup()
We always receive a (486 BUSY HERE) from the OSB PBX. The endpoint is not busy and is reachable from other DECT phones attached to the OSB without problems:
2025/08/27 22:13:01.052438 192.168.104.1:5060 → 192.168.100.20:5060
INVITE sip:184@192.168.100.20:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.104.1:5060;rport;branch=z9hG4bKPjdbc100ed-2ef7-49c5-8da1-f5207214ea5b
From: “201” sip:201@192.168.104.1;tag=bf07b32a-83e3-4c6a-b7cd-f90a83fe2576
To: sip:184@192.168.100.20
Contact: sip:asterisk@192.168.104.1:5060
Call-ID: 89fc24fa-6055-4f05-b55f-60cc0f2c5f09
CSeq: 5122 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, INFO
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-master-0086f4a0b8
Content-Type: application/sdp
Content-Length: 287v=0
o=- 1819731907 1819731907 IN IP4 192.168.104.1
s=Asterisk
c=IN IP4 192.168.104.1
t=0 0
m=audio 13860 RTP/AVP 9 8 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv+++++++++++
2025/08/27 22:13:01.111328 192.168.100.20:5060 → 192.168.104.1:5060
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.104.1:5060;rport=5060;branch=z9hG4bKPjdbc100ed-2ef7-49c5-8da1-f5207214ea5b;received=192.168.104.1
From: “201” sip:201@192.168.104.1;tag=bf07b32a-83e3-4c6a-b7cd-f90a83fe2576
To: sip:184@192.168.100.20;tag=2751128079
Call-ID: 89fc24fa-6055-4f05-b55f-60cc0f2c5f09
CSeq: 5122 INVITE
Server: OpenScape Business M5T SIP Stack/4.2.30.338
Content-Length: 0
Could somebody confirm, that our pjsip.conf should work and help to debug this further? Our experience with the OSB is limited and after looking at the sip trace and the Busy for a not busy endpoint, we are out of ideas at the moment.
Thank you very much for any ideas on how to advance here.