After Getting Ring call disconnects on cisco phone 7841

[saleem771]

Hi All,

I am facing issue from and to call SmartPTT Telephone Interconnect Motorola Software. When we call from radio, Cisco Phone 7841 starts ringing and soon after on Radio message appears Phone call failed. In Cisco Phone webpage stream show same receiving and sending codec G.711u.

When we try call from Cisco IP Phone to Radio busy tone appears.

What could be the issue. 

Given below are some logs

INVITE sip:1006@172.17.8.111:5060 SIP/2.0
Via: SIP/2.0/TCP 172.17.7.12:5060;branch=z9hG4bK1159df5a8e37fa
From: <sip:641@172.17.7.12>;tag=14315816~5e790c21-ee5c-4a91-aea4-a5f69ddd48f8-24872262
To: <sip:1006@172.17.8.111>
Date: Mon, 01 Jan 2024 12:05:23 GMT
Call-ID: 94f4900-10001-11044f-93f492a@172.17.7.12
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM12.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:172.17.7.12:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Session-ID: 084c9e9b00105000a0003c410e4b7f2b;remote=00000000000000000000000000000000
Cisco-Guid: 0156190976-0000065536-0000003429-0201789868
Session-Expires: 1800
P-Asserted-Identity: <sip:641@172.17.7.12>
Remote-Party-ID: <sip:641@172.17.7.12>;party=calling;screen=yes;privacy=off
Contact: <sip:641@172.17.7.12:5060;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEP3C410E4B7F2B"
Max-Forwards: 69
Content-Length: 0

Logs attached

[saleem771]

Logs

[Carlo Poggiarelli]

Hi,

I guess you configured a Sip Trunk to Tetra Smart PTT.

You are receiving an "UNAUTHORIZED" message from the SMARTPTT

You can refer to the following guide

https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/118765-technote-cucm-00.html

but the mentioned Normalization Script should be as follows:

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

HTH

Regards

Carlo

[saleem771]

Dear @Carlo Poggiarelli  

Thanks a lot for your reply. Given below is current situation.

From Radio to Phone it is working fine.

But from Phone to Radio it is not working. Getting Forbidden 403 error. logs attached.

I have tried all normalization scripts but still not working.

[Carlo Poggiarelli]

Hi @saleem771 ,

Your SmartPTT system is denying the request from CUCM bebause it requests Digest Authentication.

Try to disable it SmartPTT side.

HTH

Regards

Carlo

[saleem771]

Dear @Carlo Poggiarelli 

Please find attached settings of Radio Server. I was not able to find Digest Authentication.

Can You please indicate what is wrong with settings.

Best Regards,

[Carlo Poggiarelli]

Hi,

Try to follow this guide on ptt server only 

https://support.smartptt.com/hc/en-us/articles/200241318-Setting-up-telephone-interconnection-for-SmartPTT-software

Please let me know 

Regards

Carlo

[Carlo Poggiarelli]

Hi @saleem771 ,

Did you follow the guide I suggested?

I know that if for another call control system but you can find interesting hints.

Please let me know

Thanks

Regards

Carlo

[saleem771]

Dear @Carlo Poggiarelli 

Thanks a lot for your follow up message. We are still struggling to fix the issue. The mentioned guide is of older version and we are using SmartPTT Enterprise 9.7.11.0.

When we dial from phone to radio, busy tone comes. and logs show Invite, 180 ringing and then 403 forbidden message.

When we dial from phone to radioserver voice menu number, then call is automatically attended but no sound we are able to hear. After 30 seconds call gets disconnected.

When we dial from phone to dispatcher number, then same forbidden message is shown in logs.

We have opened ticket with smartptt, they are trying to support but yet not successful.

[Carlo Poggiarelli]

Hi @saleem771 

Thanks for your feedback.

For the no audio issues I would check if MTP Required option is flagged on cucm sip trunk configuration.

For the authentication request I would need a cucm trace log from Cisco Unified Serviceability  menu to dig more.

Can you please post it here?

Thanks a lot

Regards

Carlo