required support CUCM and TATRA radio Integration

[sagheer.2000]

I get one assignment from my organization to integrate the TATRA Radio system with CUCM, I tried to access information from google, but I didn't.

I need support to understand all the septs which are required from beginning to end, or if anyone has a user manual or guide please share it with me it will help me to understand

Thanks in advance

[Nithin Eluvathingal · Accepted solution]

Create a sip trunk between Tetra and CUCM. MTP is required on the SIP trunk.

On CUCM Create route pattern toward Tetra and choose the above trunk.

For Call between CUCM and tetra to work we need to apply normalization on the TRUNK between CUCM and tetra. There is a guide but normalization script mentioned in the guide is not correct. use the  below this is from production.

M={}
function M.inbound_INVITE(msg)
local invite = msg:getHeader("Via")
local rport=string.gsub(invite,"rport",5060)
msg:modifyHeader("Via",rport)
end
function M.outbound_100_INVITE(msg)
msg:addHeaderValueParameter("Via","rport",5060)
msg:removeHeader("P-Asserted-Identity")
msg:removeHeader("Remote-Party-ID")
end
function M.outbound_180_INVITE(msg)
msg:addHeaderValueParameter("Via","rport",5060)
msg:removeHeader("P-Asserted-Identity")
msg:removeHeader("Remote-Party-ID")
local Sdp = msg:getSdp()
end
function M.outbound_200_INVITE(msg)
msg:addHeaderValueParameter("Via","rport",5060)
msg:removeHeader("P-Asserted-Identity")
msg:removeHeader("Remote-Party-ID")
local Sdp = msg:getSdp()
Sdp = string.gsub(Sdp, "b=TIAS:%d*\r\n", "")
Sdp = string.gsub(Sdp, "b=AS:%d*\r\n", "")
msg:setSdp(Sdp)
end
return M

2.12.0.0

[sagheer.2000]

Thanks, @Nithin Eluvathingal supporting,

Configuration is done, as mentioned above, today I tested the call routing between the telephone and TATRA, it is like calling reaches the TATRA server, I heard the busy tone, but I am not 100% sure, there are any methods to check call is reaching TATRA or not? 

BR

Sagheer Ahmed

[Nithin Eluvathingal · Accepted solution]

Use RTMT and check the call flow. it can show the calls as diagrams. i don't have much knowledge on TETRA as this part was done by TETRA expert.

The below document describes the procedure to review the call flow and signalling for a SIPc (Session initiation protocol) call on Cisco Real Time Monitoring Tool (RTMT), wherein RTMT is a quick and easy tool to analyse the call flow of a SIP call.

https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/213583-procedure-to-analyse-call-flow-of-sip-ca.html

2.12.0.0

[sagheer.2000]

Thanks for the valuable information, after today's workshop, we found the following issue and I need support

• Call from Tetra terminal to Ip telephone. The call is not progressing, according to NMS logs it seems that call is rejected by the other party (pic below). I believe external PBX should allow tetra calls.

2-Call from Ip telephone to Tetra terminal. Looking at to test is done, we are suspecting that they are sending the dialling to NEBULA without previously taking out prefix 456, so NEBULA does not know any ISSI or GSSI 45652001.

In other terms (456) should be used only as prefix.

Calling extension 40000 must dial 45652001 to establish a communication with ISSI 52001. PABX detects prefix 456 and redirects the dialling towards SIP TRUNK connected to NEBULA, but taking out previously such prefix 456, the dialling number that NEBULA receives is just 52001.

Regards

Sagheer Ahmed

[saleem771]

Dear @Nithin Eluvathingal 

Can you please go through the attached logs from Cisco CUCM 12.5 to Motorola SmartPTT SIP Trunk issue. I have added normalization script but still facing 403 forbidden issue.

From Radio to Phone it is working fine. but from Phone to Radio it is not working.