spa8800 restrict IP's from where to place calls to fxo

[sven]

Hello,

We have SPA8800 configured according document: I have 2 questions:

1) (problem): Calls from anyone using a SIP Soft phone on a PC can route calls to the PSTN on the FXO ports: Is there a posibility to restrict only to Ip form Asterisk server? I tried registering, but this results dificult on my Asterisk, as I need then to have host=dynamic in the peer definition in the sip.conf and in my particular situation results imposible...

In the SPA8800 SIP settings on the line tab, I see a YES/NO value with name "Restrict Source IP" what is this and where do I put the IP to restict to?

2) (curiosity): in the DP for the FXO, we use (SO:nnnnnnnn@192.168.x.x:5060>) .... what stands the SO for, and what other options are there?

thanks,

Sven

[Andrey Cassemiro]

Hi Sven,

Please see below the description of the field: Restrict Source IP

If Lines 1 and 2 use the same SIP Port value and the Restrict Source IP feature is enabled, the proxy IP address for Lines 1 and 2 is treated as an acceptable IP address for both lines. To enable the Restrict Source IP feature, select yes. Otherwise, select no. If configured, the PAP2T will drop all packets sent to its SIP Ports originated from an untrusted IP address. A source IP address is untrusted if it does not match any of the IP addresses resolved from the configured (or if is yes).
The default is no.

I think that it fits with your needs.

Regarding the Dial Plan the S means Short Interdigit Timer and it's Value.

Interdigit Timer Master Override
The long and short interdigit timers can be changed in the dial plan (affecting a specific line) by preceding the entire plan with the following syntax: •Long interdigit timer: L : delay-value , •Short interdigit timer: S : delay-value ,
Thus, “L:8,( . . . )” would set the interdigit long timeout to 8 seconds for the line associated with this dial plan. And, “L:8,S:4,( . . . )” would override both the long and the short time-out values.

For more information please visit:

http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf

At page 61.

Regards,

Andrey Cassemiro.

[sven]

Hi Andrey and thank you for your answer...

1 doubt hawever.... I tried to enable the Restrict Source IP feature and I was still able to send calls from anywhere.... the only thing that was different here then "Lines 1 and 2 use the same SIP Port value and the Restrict Source IP feature is enabled" is that each fxo port is using another SIP UDP port (5061, 5062 and 5063) (not for the fxs ports that all 4 use 5060)...

this is because in the asterisk I need to create different trunks and I can't create 3 trunks with the same Ip and port combination.... and haw would the SPA8800 now on an incomming INVITE to which of the fxo ports he has to send the call?

thanks

Sven