Please check the phone’s volume (a microphone, a headset, or a speaker).
When NAT traversal is in use (when you have communication over mutual networks), please refer “Brekeke SIP Server Administrator’s guide – NAT traversal” for more details.
Please check the network settings at Brekeke SIP Server. Make sure the ports for sound communication are opened by a router or a Firewall.
Do both UAs (phones) codec matches? Check if the codec setting is set correctly.
For Brekeke PBX, check RTP relay setting. If [RTP relay] setting is selected to “on (G.711u only)”, or “off (G711u only)”, please check if the machine supports G.711ulaw. Otherwise, please set it to “on (G711uonly)”. Some ITSPs or machines do not work unless you set it to “on (G711u only). For more details about RTP relay, please refer to ”Brekeke PBX Administrator’s guide (Advanced)”.
When you use Brekeke PBX and neither of your machine or ITSP supports re-INVITE, you need to set [RTP relay] to “on (G711u only)” or “on”. For more details about RTP relay, please refer to ”Brekeke PBX Administrator’s guide (Advanced)”.
On a machine which Brekeke SIP Server is running, capture the UDP packet using a network analyzer, such as Ethereal, and check if the SIP sequences are correct.
Since ITSP tends to use old version of SIP standard, there are some cases that they can not process lr parameters. Please set [B2B-UA mode] on from SIP Server >[Configuration]>[SIP], or set $b2bua = true in the applicable deploy pattern in the dial plan.
