Heads Up :
The post you are writing will appear in a public forum. Please ensure all content is appropriate for public consumption. Review the employee guidelines for the community here.
Transcoder is regiestered to CUCM 6.1 successfully. When I make a call from G.711 to G.729, the call fail. From the log I can see CUCM allocate the transcoder but then release it without any errors.Any help would be appreciated.
We enable the automatic recording for some phones. After that, we have problems related to conference, transfer. We are using CUCM 6.1. After some troubleshooting, it looks like when the phone is under recording, the codec is locked. Let's say Phone ...
We have a Unity vm server at the data center. It is assign by PRI which has both 1800 number and a local number(416-123-4567). I have configured the SRST gateway and the call forwading for VM. When users dial the 1800 number directly, it works fine. ...
On CCM 6.01, there is a CSS configuration for Voice mail pilot and the actual voice mail port. How does the CSS work? Is it work like Phone and Line CSS?
After looking into the trace file, only the phone A which the softkey Transfer is pressed will disconnect the monitoring session, not phone B.This will cause issue I mentioned above. Just wondering if anyone who implements call recording has any good...
It is not on the unity.Basically, when the gateway is in srst mode, external user call the extension and get forward (no answer) to the 1800 number based on the srst configuration on the gateway.
To enable netflowinterface FastEthernet0/0 ip flow ingressAnd then you can use 'show ip cach verbose flow' to check the netflow static.To using SPAN, here is the link http://www.cisco.com/en/US/products/hw/switches/ps708/products_tech_note09186a008...
The first thing I will try is to setup a SPAN on your switch to sniff what exactly traffic is marked as dscp ef. Or you can enable netflow on the router to look at the traffic.
