Apr 18 2017 08:03 AM
hi,
I had installed CCE 1.4.1 previously and it is working fine with SIP Trunking to PSTN, except for the below issue. Hope someone can help or give suggestions to resolve it:
1. After 3 hours of no call activity, I make a call from PSTN to SFB client. The first call will fail, and subsequent calls are OK.
2. If the first call is from SFB client to PSTN, the call is OK.
This weird behaviour is most puzzling.
Thank you.
Apr 20 2017 12:25 PM - edited Apr 24 2017 11:51 PM
Apr 20 2017 12:25 PM - edited Apr 24 2017 11:51 PM
That is an interesting issue. I don't see that in my environment running on 1.4.1.
Could you try to test the scenario running IncomingAndOutgoingCall CLS scenario on all servers??
Start-CsCLSlogging -Scenario IncomingAndOutgoingCall
Reproduce the issue.
Stop-CsCLSLogging -Scenario IncomingAndOutgoingCall
Search-CsCLSLogging -FileName logfile.txt
Have a look at the logfile or post it in this thread.
Hope this helps you
/Kenneth ML
Apr 26 2017 02:48 PM
Please shaare the logs. How many Cloud COnnector instances do you have?
Apr 28 2017 01:15 AM
May 01 2017 12:39 AM
Hi.
<<<<<<<<<<<<Incoming SipMessage c=[<SipTcpConnection_10FC87D>], 192.168.1.127:5068<-192.168.1.126:35715
CANCEL sip:+6568130463@gateway1.axiomdemosite.com SIP/2.0
FROM: "96794134" <sip:+6596794134@202.163.63.213>;tag=1c1646615811
TO: <sip:+6568130463@gateway1.axiomdemosite.com>
CSEQ: 1 CANCEL
CALL-ID: 1270468232254201711120@192.168.1.126
MAX-FORWARDS: 10
VIA: SIP/2.0/TCP 192.168.1.126:5060;alias;branch=z9hG4bKac96743477
CONTENT-LENGTH: 0
USER-AGENT: Mediant SW/v.7.20A.001.501
According to this SIP message, the Audiocodes device (IP 192.168.1.126) sends the cancel to the mediation server (IP 192.168.1.127). We cannot tell from this why it does so, but maybe you can check the log in the Audicodes device or include more of the SIP logs from the beginning of the sequence starting with the invite.
/Kenneth ML