Hello, I was wondering if any of you guys is familiar with this issue that I'm having. We recently upgraded to 7.1.5 because of a Call Transfer problem when enabling the Recording Options to "Automatic Call Recording Enabled". However, the upgrade did not fix the problem. So here's what I have tested so far:
- I place a call to another IP phone and ask that person to Transfer me to another person. When the 2nd person press Transfer and dial the 3rd person is fine but when they press Transfer again to complete the transfer then I got a beeping sound. And the call will be terminated.
- It is very much similar to the test above except that I'm pressing ConFrn and try to conference in another person.
These test was perform with the Recording Options: Automatic Call Recording Enabled. If I disabled the Recording Options then it will work but our third party recording software requires that the Recording Options has to be Automatic Call Recording Enabled.
If anyone experienced the same issue and had a fix for, please share your experience and info for this issue. Your help is greatly appreciated.
Re: "Recording Options" on CallManager 7.1.5 problems
So to clarify the topology:
Phone A <---> Phone B. Phone B transfers Phone A to Phone C. When Phone B goes to complete the transfer Phone A hears a tone and the call disconnects.
I am making the assumption that the recording profile is setup for Phone A.
Assuming that this is correct, some things you can look for are:
1. The region relationship (codec used) for the Phone A to Phone B call compared to the region relationship between Phone A and Phone C.
2. After determining the codecs used for the above two calls look at the location setting of the SIP trunk used for recording and make sure that you are not exceeding the bandwidth allocated.
The BIB (Built in Bridge) in the IP phone will send two seperate audio streams to the recording server (one TX and one RX for that phone). The codec selected is based on the codec actually selected for the call. So if Phone A is talking to Phone B successfully and your recordings are working at this point I assume that the location on the SIP trunk is adequate. But when Phone B completes the transfer from Phone A to Phone C this new call checks the region capabilities and uses the appropriat codec. If the region relationship between Phone A and Phone C require a codec that uses more bandwidth than the original call and the location of the SIP trunk restricts the bandwidth to less than the codec offered the call will fail.
I hope that helps you trouble shoot the issue. If you are still unclear and having problems I'm sure TAC would be happy to look at the CUCM traces and be able to tell you why the call is failing.
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