SERVICE MANUAL AS55X
55Xseries_manual_V3.22 (18-08-15)
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number will be transmitted. Otherwise you will see the GSM CLIP link in “always pass
GSM CLIP” mode. Alternative you can select that the GSM CLIP is passed only if there
is no match in the GSM PBX link list.
Default value:
Never pass GSM CLIP
Request SIP PRACK sequence
In cases of missing early media or in cases of one way audio it is possible to request
or suppress this sequence unconditionally.
If the AS55X is working with the value “never” there are no PRACK requests at any
time. If you use “always” every SESSION PROGRESS requests a PRACK. With the
setting “if supported” the AS55X evaluate the received INVITE message. If there is a
“100REL” offer within a supported header, the AS55X uses PRACK requests,
otherwise the requests are not longer required.
Default value:
If supported
Allow direct GSM number dialling in client mode
Normally in SIP client mode only DTMF suffix dialling is possible. But sometimes it´s
also possible, that there are requirements for complete number dialling in client
mode. Therefore you can check this box.
The AS55X will work like SIP client and SIP peer together. All received numbers,
except the configured local username and the SIP numbers of the GSM PBX link list,
will cause a direct GSM call. These numbers could be received from anywhere.
Therefore also check the next configuration box (Require proxy) to avoid illegal
usage.
Require proxy / Only accept call-setup from peer(s)
This checkbox has different names in SIP peer and in SIP client mode.
In SIP client mode you can configure direct GSM number dialling if it is possible (see
configuration checkbox above). Normally everybody could send a call-setup. To avoid
unauthorised usage, the AS55X uses the SIP contact header for restricting this access.
Only the SIP Server is legitimate to sent messages, because on SIP client registration
the CONTACT header will be forwarded. So it is only possible, that the SIP Server
knows the correct header. If your SIP Server does not support this feature, you can
restrict this access only to the IP address of the SIP registrar. Therefore you have to
set bit 52 in special settings (see 0).
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