LANconfig: VoiP Call Manager / Extended
WEBconfig: LCOS menu tree / Setup / Voice Call Manager / General / ISDN gateway codecs
- Echo canceling from SIP to ISDNActivates the echo canceling of remote echoes. With an echo that is too strong, subscribers can hear their own voices after a short delay. Activating this option reduces the ISDN echo at the SIP > ISDN gateway.
- Prefix from internal to SIP user
This prefix is added to the calling party ID, if available, for an incoming,
internal call if the call is directed to a SIP user.
Note: A call is regarded as external if it comes from a "line". If this line is a SIP PBX line, then the call is only external if the incoming calling party ID is preceded by a "0". All other calls are regarded as internal.For more information about handling the calling party ID, see .
- Prefix from external to SIP user This prefix is added to the calling party ID, if available, for an incoming, external call if the call is directed to a SIP user.
- Prefix from internal to ISDN user This prefix is added to the calling party ID, if available, for an incoming, internal call if the call is directed to an ISDN user. If a line prefix is defined, this is placed in front of the whole of the called number.
- Prefix from external to ISDN user This prefix is added to the calling party ID, if available, for an incoming, external call if the call is directed to an ISDN user. If a line prefix is defined, this is placed in front of the whole of the called number.
- Prefix from internal to analog user This prefix is added to the calling party ID, if available, for an incoming, internal call if the call is directed to a analog user. If a line prefix is defined, this is placed in front of the whole of the called number.
- Prefix from external to analog user This prefix is added to the calling party ID, if available, for an incoming, external call if the call is directed to a analog user. If a line prefix is defined, this is placed in front of the whole of the called number.
- Prefer outgoing packets
Depending on the audio codec that is used for SIP calls, sufficient
bandwidth through the firewall is reserved (provided sufficient bandwidth
is available). To control the firewall, you can configure how the remaining
data packets that do not belong to the SIP data stream are handled.
- PMTU reduction
- Fragmentation
- No change
- Prefer incoming packets
Similar to the outgoing data packets, you configure how non-VoIP data
packets are handled when bandwidth is reserved for SIP data.
- PMTU reduction
- No change
- Reduced packet size This parameter specifies the packet size that should be used for PMTU adjustment or fragmentation while the SIP data have priority.