FXS one way media failure on VIC3

[timothy.bailey]

Hi;

I have an issue wherein the media fails on analogue calls in one direction only.

Analogue connected to fxs port dials out over SIP trunk to IP handset on CO switch. After 30 seconds media from analogue fails but still works from  CO switch.

voice-port 0/2/0

cptone GB

station-id 4043271717

If someone could tell me what may cause this to happen so I may fix this this issue. Or some debug commands to run that would be great.

debug voip dialpeer

sh dialplan number 4043271717

Unfortunately I am unable to post the configuration for you so please be patient and thank you for your support in advance.

Tim

[paolo bevilacqua]

You have some issue with the IP addresses used in the trunk, or you may have NAt or firewall hindering communications. You may need to use a bind interface command.

[timothy.bailey]

For information of anyone else experiencing this issue.

After setting the call progress indicator to convert a disconnect into a progress indicator the problem went away

Command : voice call convert-discpi-to-prog always

"The voice call convert-discpi-to-prog command turns an ISDN disconnect message into a progress message. "

Hope this helps someone else.

[paolo bevilacqua]

Very interesting, because there is no logical connection and you do not have ISDN either.

[timothy.bailey]

it would appear that whilst i was running debugs the call stayed up but as soon as it was disabled i went back to one way audio after 30 seconds.

Back to square one again...

Whan I ran debug ccsip events it returned session timed out. Both the router and CO switch have their session timers set to 90

There is no NATing the CO switch is connected to the IP interface of the router that has the FXS card installed. I can ping between devices. I am also binding to the loopback interface.

thanks for the suggestions, please keep them comeing

Tim

[paolo bevilacqua]

Please read post above: your SIP call control device does not recognize the address (eg loopback), or does not receives media packets from router.

[timothy.bailey]

thanks for the feedback.

I have checked that the CO is in the ip trusted list  for sip devices and it is. I can bind to loopback 0 but it makes no  difference and if I bind the control to loopback 0 the call fails to  setup.

The sip trunk is up between the router and CO switch with keepalives set to 20 seconds. What else can I check?

Is there a debug command I can run like debug ccsip messages or something.

Thanks

[paolo bevilacqua]

Router does what is configured to do. You have to check on the device that you call "CO" why it drops the call, you will find that it does not receive vaild media.

[timothy.bailey]

So thanks for the assistance. After several wireshark traces and alot of percerverance I found that what was happening was the following:

CO Switch setup the call CUBE terminated and negotiated the rtp ports.

CUBE sends re-invite advertising its RTP port

CO Switch sent out a new RTP port

CUBE sends re-invite advertising its RTP port

CO Switch sent out a new RTP port

RTP stream is setup but the CUBE was still using the original RTP port.

After speaking with the manufacturer of the CO switch we set it to answer duplicate invites and turned off early invite of SIP messages. This resulted in the CO switch sending the original RTP port in the 200ok with SDP back to the CUBE when the session timer was reached.

Overall result two way audio....Problem solved.