'auth-id |user-id' für gw1 - gw4

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28 Okt 2004
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- soeben vom Sipura-Support bekommen...

Well, we have listened to the community requests.
We have planned in the next release to allow you to specify a user-id to
be used in the From and Contact headers of outbound SIP messages.
Hopefully this will greatly increases the call success rate through some
service providers.

Beware that there might still service providers that would not allow you
to call through them despite this additional change. But this is the
best we can do for now. Again we want to emphasize that Gw1-Gw4 are
targetted towards other gateway devices such as SPA-3000.

The firmware release will be posted some time this week.

- Sipura Support

- da bin ich aber mal gespannt...
Die Gebete wurden erhört!
Hoffentlich funktionierts auch,
denn der SPA muss sich wohl beim Provider-Wechsel erstmal registrieren
und ob das so schnell funktioniert, dass ohne Probleme ein abgehender
Ruf zustande kommt... :?:

- schade nur, dass ein abgehender Ruf von FXO nach VoIP1 (7-8-3) scheinbar nicht möglich ist...
- FXO to VoIP2 = ok

- FXO to VoIP1 = nok, -> witch entry in the dialplan pstn-line?

- VoIP2 to FXO = ok

- VoIP1 to FXO = ok

My current pstn-line-dialplan: (<**:>[2-9]|<:>x.)
... der Support hat seine Aussage bestätigt

Quellenangabe: http://www.sipura.com/Documents/rnote/rn3k-2.0.11g.htm
[/quote]Release Notes for 2.0.11(GWg) -- Sipura Phone Adapter

SPA-3000 -- 1 Port FXS, 1 Port FXO, 1 Ethernet Interface

Copyright (C) 2003-2004 Sipura Technology Inc.

* * * * * * * * * * IMPORTANT * * * * * * * * * * * * * * * * *
* Use of Proprietary Information and Copyright Notice: *
* This release note document contains proprietary information *
* that is to be used only by Sipura Technology customers. *
* Any unauthorized disclosure, copying, distribution, or use *
* of this information is prohibited. This restriction includes *
* ALL Internet based discussion forums, e.g. DSLreports. *
* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * *

Bug Fixes
-- since 2.0.11(a)
1. BUG ID=00230
When Forward Line 1 to PSTN gateway, SPA does dialog refresh even
if <VoIP DLG Refresh Intvl> is set to 0

2. bug ID=231
If [PSTN Line]<Line Enable> is "no", the
FXS port relay does not operate properly. If user is calling from
the FXS port phone to the PSTN while the SPA is powering up, the
PSTN call will be disconnected when the SPA boots up.
The problem does not happen if <Line Enable. is "yes" (default)

3. BUGID=?????
SPA will treat CWCID ACK digit from the FXS port as digits detected
at the FXO port and sent AVT tone to the peer for it.

New Features and Enhancements
-- since 2.0.11(a) --
1. Added <PSTN Dial Digit Len> option (under [PSTN Line]) to configure
the on off time of each digit when dialing out to the PSTN line.
The syntax is <on-time>/, where on-time and off-time are
specified in seconds with up to 2 deciaml places.
For example: 0.5/.12
Default is: .1/.1
If value is blank, default is used. On off time are limited to
betwee 0.05s and 3s

2. Support insertion of pause digits. Any digit that is not in the
range 0-9*#abcdABCD are treated as a pause digit. We recommend
to use a ',' (comma) as a pause digit. Each pause digit inserts
a silence period = <on-time>+ of a digit

3. Allow [PSTN Line] to use gw1-gw4 to make VoIP calls.
Hence a PSTN caller can call the SPA-3000, then hop on to
call one of the configured gw1-gw4, just like Line 1 caller.
The dial plan syntax is same as for Line 1

4. Allow mofification of user-id in the From and Contact headers
when making calls via gw1-gw4, to improve chance of success
when the gateway (or service) does check these header fields.
To enable this feature, use the following optional syntax
in the <Gatway 1-4> parameters: [@]hostname
where <userid>@ is optional. If @ is not specified, it
follows the old behavior. Otherwise, the user-id fields in
the From and Contact headers are changed accordingly.
If <user-id> = "$usr" (case-insensitive), then the
corresponding value in <GW1-4 Auth ID> field is used.

[email protected]_company_a.net:5061
fwdnat.pulver.com:5082 (old behavior)

5. If <NSE Process NSE> is set to "no", the NSE payload type will
not be included in outbound SDP

6. Added <DTMF Process AVT> option in [Line 1/2]. If set to "no",
AVT payload type will not be included in outbound SDP

7. When sending CAS signal to the phone during CWCID (call-waiting caller-id)
generation, do not mix in the normal voice audio

8. Extended the parameter <RTP Packet Size> under [SIP tab] to
allow specifying a different size for Line 1 an Line 2.
Syntax: <Line 1 Packet Size>[,]
If <Line 2 Packet Size> is not given, it will assume the same
value of <Line 1 Packet Size>. Unit in seconds. Examples:
.03,.1 ;30 ms for Line 1 and 100 ms for Line 2
.06 ;60 ms for boht Line 1 and Line 2

Default is 0.030 (30 ms both Line 1 and Line 2; same as before)
- If parameter is blank, default value is used
- If planning to use G723, minimum size is 30ms
- All size values are limited to between 10 and 160ms
- All values are rounded to the nearest 10 ms

9. Added UID1 and UID2 as valid provisioning macro variables, which
expand to the contents of User_ID[0] and User_ID[1], respectively.

10. Added two more choices under <DTMF Tx Method>, "InBand+INFO", and
"AVT+INFO". Selecting either of these methods cause the SPA to
send SIP INFO message with DTMF events in addition to
InBand or AVT DTMF signals

11. Added <DTMF Process INFO> option under Line 1/2, to enable/disable
playing back of DTMF tone on SIP INFO Messages. Default is "yes".

12. Relaxed DTMF detection further when waiting for CAS ACK digit from
the FXS port
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