I just bought a mediatrix 1204 as a home gateway. I have it hooked up to a raspberry pbx and everything is working fine on the SIP end. It is connected to a POTS port supporting BT infinity. I am struggling with the following however:
CLID not being passed from PSTN to SIP extension
Call hangup not working from PSTN to SIP extension
Dial tone not detected by mediatrix for outbound calling from SIP extension to PSTN
I set the country parameter to uk-etsi-fsk but not sure what I am doing in terms of matching the tones etc. I set DTMF to inband (also tried info and 2833) but none of the options fixed any of the problems.
Can anyone advise?
This is a Mediatrix question, not an Asterisk one!
BT have a special caller ID protocol, which involves a line state handshake before the first ring.
BT use various ways of signaling disconnect supervision, depending on the type of exchange to which you are connected. Generally, if disconnect supervision is important to you, use ISDN, as it is always a fudge on analogue.
Details of BT protocols can be found in their Supplier Information Notes: https://www.btplc.com/SINet/SINs/index.htm
It wont work !
Is your POTS line on a System X, AXE or DMS 100 ?
CLID on POTS is controlled on the DMS100 by the PDTC’s CMR card… it uses a BT proprietary FSK based inband protocol to burst the CLID and NOA taken from the IUP IAM C7 calling info element.
There are no call markings for NOA (C7 nature of address) using SiPv2 so there can be no mapping of BT’s callerID without country specific software. The UK fixed network is NOT ISUP based it is IUP/BTNR167 based so country specific IUP/ISUP call markings differ.
Also there is no caller IE screening controls in SIPV2 as they are just RFC’s and mean nothing, the number is either there of withheld, again this does not map to the UK caller ID C7 call markings.
Thanks for the responses.
I am not overly bothered about CLID. The wobbly call teardown is more annoying however.
If I call my asterisk voicemail and hangup the teardown works better (I get a short (5 second) disconnected tone at the end of the voice message). But if I call my SIP extension and hangup before answering, the SIP channel remains open and continues to call the extension. I have to pickup the call and drop it manually. Any ideas anyone?
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