Attended transfer.

1. Brekeke Product Name and version:

2. Java version:

3. OS type and the version:

4. UA (phone), gateway or other hardware/software involved:

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :

6. Your problem:

Hi everybody. I have some problems with making an Attended transfer, because when i press #9, the pbx doesn't give me a tone in order to dial the extension that i want to transfer the call to. I suppose it is happening bacause all my ip phones are working sending dtmf via rtp. I don't know what to do, because my gateways must work with that. The gateways can't work sending dtmf via sip info.

What should i do??

Hi,

Well first think is to make sure rtp relay is on in both the sip server (config screen) and the pbx config screen. Then restart both to activate. The best results are if you set the ip phones to RTP in-band if you have that option. While the call is in progress, go to the sip server and display active sessions to make sure there are 4 processes going on during a call. That insures you have rtp relay on.