Hi.. We have firepower FW that is used to filter the VOIP traffic to our telephony platform.When we are troubleshooting voip issues it is good if we can dump the SIP and RTP on the firewall.We can do this from the FMC but the buffer is limited to 32 ...

Hi...   We have made an Sip normalisation script that add a Privacy : id header if from header is anonymous, was a demand from the telephony provider. That works as intended , but after the call has been established for 10-15 minutes a new INVITE is ...

Hi... We are using CIsco RTMT (version 11.5) when troublehooting various telephony problems. When we us the "Real Time Data" the calls are shown and we trace the call, but when we press for instance the INVITE it takes up to 30 minutes before the SI...

Hi... Is it possible to route calls on the number in the sip from header (calling number) in cucm 8.5. I would like to route the calls that comes from a specific phone number from pstn to one of our linemachines that is behind cisco ccm and has its ...

ASA version 8.6 Hi ... We have a remote VPN  that works as it should when it comes to access the reourses on the LAN, we also have a site-to site VPN.on the same ASA firewall.The problem is that I can't access the site-to-site VPN when we are using t...