Please configure your PBX as follows:
- IP Authenticated (if configured) or User Registration
- SIP Server: trunks.okay.network (or the IP that this resolves if your PBX do not support hostnames)
- Domain/REALM: trunks.okay.com.mx (optional if you are using IP Auth)
- Port 5060 UDP or TCP
- PCMU or PCMA codec enable
- SIP CID Type RPID
- Pass the Caller ID
- Caller ID Number in the FROM header
You must send the destination using the country code. For example, if you want to call 6138007370 meaning to call Ottawa, Canada using short North-America notation it won't work. The system will route internationally as 61 international code belongs to Australia. You should send the call to 16138007370. The leading + is optional.
If you are using the legacy Unlimited Termination Trunk, then USA48 and Lower Canada destinations are always enabled by default. International codes by country basis are disabled. Please contact us to enable it. If you are using the prepaid service, all destinations are enabled by default.
After you configure the PBX trunk and a confirmation from us, you can start passing calls.
Usually, when sending a call invite is shown as follows:
INVITE sip:This email address is being protected from spambots. You need JavaScript enabled to view it. SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5080;rport;branch=z9hG4bKN2S2QHegp1y9F
Max-Forwards: 69
From: "Someone" <sip:This email address is being protected from spambots. You need JavaScript enabled to view it.>;;tag=vZ3vQNSNX0rep
To: <sip:This email address is being protected from spambots. You need JavaScript enabled to view it.>;
Call-ID: 95808b34-af87-1238-9394-b2ce997a7de4
CSeq: 14844589 INVITE
Contact: <sip:This email address is being protected from spambots. You need JavaScript enabled to view it.:5080;transport=udp;gw=7377c370-64d9-4971-887f-210a2a5d5495>
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 344
X-Serialnumber: 000413414375
P-Key-Flags: resolution="31x13", keys="4"
X-accountcode: pbx.to-call.me
X-FS-Support: update_display,send_info
Remote-Party-ID: "Someone" <sip:This email address is being protected from spambots. You need JavaScript enabled to view it.>;;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1578767047 1578767048 IN IP4 XXX.XXX.XXX.XXX
s=FreeSWITCH
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 29716 RTP/AVP 18 0 8 3 101 13
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
If you get a SIP code 100, followed by a 183 one, then your trunk is ready.