volume levels dropping on calls after being transferred to cell phone

[lwilliams123]

Good morning! I figured someone on this forum would have an answer for me. We have an Avaya IP office Running the latest patched firmware version of V7 36. The owner of our company calls in on his cell phone. He asks to call another employee who is not in the office. The receptionist then transfer's the call to the employee's cell phone. They have to almost yell at one another to be able to speak to one another.

Is there a solution for this other than having the owner call the employee's cell phone directly? I know this also ties up two phone lines. I figure they are losing about -6 dB of volume with the transfer at least if our lines were perfect to the company.

 

[Billx1]

It sounds like they are using Analog CO Lines.
When you connect one Analog CO Line to another in a KSU, you can almost always expect the remote parties to have a hard time hearing each other.
This is because everything they say and hear needs to make 2 trips to the local Central Office, so there is loss in signal.

Since you have a Release 7.?(36) KSU, I would suggest using a SIP Trunk when transferring to the cell phone.
There is no signal loss and it does not tie up any additional CO Lines.
If you have an IPO Basic/Partner, then there are 3 free SIP Trunk Channel Licenses in your KSU already. (you cannot see them , but they are there)
If your KSU is Essential, then you will need to buy a Single SIP Trunk Channel License.

You will have much better results with SIP.

 

[lwilliams123]

Thanks for your response! We have Wireless internet here and we've tried a sip trunk before on our system. We had lots of call quality issues unfortunately. I would like to try it again since we've updated to 7(36). I called around to try to find local companies that could provide the SIP service to us and I did find one. I'm wondering if I should do a direct sip trunk or go with a VOIP device that would plug into our lan then utilize one of the extra analogue trunk ports on our system. I spoke to our wireless provider and they would be able to separate the data and voice coming to us on separate vlans and add qos on the voice side of things.

 

[yankblan]

Easier than that: use the flash feature. Program a button on the phone (you can't create a SC for this, since you can't invoke a SC over an active call). So you press flash, get dial tone, dial second number (without the prefix; you're dialing straight on the trunk at that point), and then you hang up when it rings, or if you press flash a second time, you a have 3-way conference. Your receptionnist can now introduce the parties and hang up.

It uses NO trunks on the system when it's transferred, it all happens at the CO.

YOu need to have the conf feature from your provider

http://interconnexionsld.ca/