This is done with following dialplan "(xx.<:@gw0>|<#9,:>xx.)"
If a call from PSTN is not answered within 20 sec the call should be redirected via voip (see CFNA Solution).
Because my SPA-3102 is behind a firewall i use it as a bridge.
Firewall
modprobe ip_conntrack_sip ports=9060; modprobe ip_nat_sip
iptables -A FORWARD -p udp --dport 9060 -m state --state NEW,ESTABLISHED -j ACCEPT
With this settings i don't need "NAT Keep Alive" and Register every 3600 sec (1 hour) is ok.
Usually udp connectiontracking has a timeout of 180 sec, but ip_conntrack_sip reads the sip content and set the timeout (3600).
My Settings
WAN
Connection Type: DHCP
Enable WAN Web Server: yes
Lan Setup
Networking Service: Bridge
Enable DHCP Server: no
SIP
RFC 2543 Call Hold: no
RTP Port Min: 9000
RTP Port Max: 9059
Provisioning
Provision Enable: no
Upgrade Enable: no
Regional
Ring1 Cadence: 60(1/4)
Ring Waveform: Sinusoid
Ring Frequency: 50
Ring Voltage: 60
CWT Frequency: 420@-20
FXS Port Impedance: 270 + (750 || 150nF)
Caller ID Method: ETSI FSK
Caller ID FSK Standard: v.23
Line 1
Make Call Without Reg: Yes (if no network connected)
Proxy: 1.2.3.4:9060
User ID: sipuid
Password: xxxx
Preferred Codec: G729a
G7* Enable: no
Dial Plan: (xx.<:@gw0>|<#9,:>xx.)
PSTN Line
Make Call Without Reg: Yes
User ID: sipuid
Password: xxxx
Preferred Codec: G729a
G7* Enable: no
PSTN CID For VoIP CID: Yes
medium Min CPC Duration: 0.25
Disconnect Tone: 440@-30,440@-30;2(0.4/0.4/1+2)
FXO Port Impedance: 270 + (750 || 150nF)
On-Hook Speed: 3ms
Line-In-Use Voltage: 25
Current Limiting Enable: Yes
Ring Validation Time: 150 ms
Ring Indication Delay: 0
Ring Timeout: 650 ms
Without this "Ring Timings" after "PSTN Ring Timeout" the call to Line 1 got CANCELED also immediately the following call to VOIP (don't know why - saw it in traces to syslog).
With "PSTN Ring Timeout: 20" i also saw no cancel but that's a dirty workaround for me (better are a correct "Ring Timings").
Enable Traces
System
Syslog Server: 1.2.3.4
Debug Server: 1.2.3.4
Debug Level: 3
Line 1 / PSTN Line
SIP Debug Option: full
CFNA Solution 1:
User 1
Cfwd No Ans Dest: 12345678
Cfwd No Ans Delay: 20
PSTN Line
PSTN Answer Delay: 40 (20 sec Line 1 and 20 sec voip => after that forced hangup)
disable via phone with *93
enable via phone with *92[Number as dialed via dialplan(use #9 for voip)]
Drawback:
* after disabling with *93 the number is deleted
* enable with *92#9NUMBER (without #9 PSTN is used according to Dialplan (xx.<:@gw0>|<#9,:>xx.) it would be NUMBER@gw0)
* only G711 Codec supported (because PSTN calls Line 1 and then only G711 seams to be supported)
CFNA Solution 2:
PSTN Line
Dial Plan 8: (S0<:12345678@1.2.3.4:9060>)
PSTN Caller Default DP: 8
PSTN Answer Delay: 20
Drawback: you cannot dis/en-able via phone
CFNA Solution 3:
Use a local asterisk where you can connect sip clients rinnging at the same time when a call arrives on the SPA-3102
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