Hello, who can halp me to setup Asterisk, Linksys Spa 3000 and PSTN?
I don’t know how to setup Asterisk sip trunk to Linksys spa 3000.
I don’t know how to setup incomming and outgoing call rules and dial plans in asterisk and in spa 3000.
I tried different setting but I am new in this area…
I wan’t to setup it at home.
Configure an endpoint in asterisk
Configure the SPA 3000 with Asterisk’s hostname or IP address in the Proxy field
Where in SPA 3000?
LINE 1?
PSTN LINE?
My physical topology:
I want to can call from mobile phone with voip number 6001 from asterisk to Desktop phone.
I want to can call from mobile phone with voip number 6001 from asterisk to any in world over pstn with real pstn number 064928840 ( the country code is +359).
I want to can call from desktop phone to any in world with pstn number 064928840
I want can receive calls from world to pstn number and forward them to desktop phone and to android phone.
I don’t know how to setup sip trunk between astersk and spa 3000.
I don’t know how to setup dial plans and call rules in asterisk, dial plans in spa 3000, in line 1 and in pstn line.
You need to pay for consultancy services. This sort of basic configuration is not an appropriate use of a forum like this. What the forum is good for is answering specific questions when you get stuck.
Although this looks like a technophiles home system, rather than a business configuration, that, if anything gives people less incentive to give you a canned solution, as it will only be helping one person, not all users of the business.
Okay specific question where I stuck.
Why I can do sip trunk between SPA 3000 and Asterisk successfully only with port 5060?
I can’t do it with 5061 or 5062 or 5090?
First I create EXTENTION number 6010 for sip trunk between spa 3000 and asterisk:
Now I create sip trunk between asterisk and spa 3000:
Now I create outgoing calling rules for 112:
Now I create dial plan1 where my rule for 112 is there:
And this is for my Linksys SPA 3000 PSTN LINE:
The result is:
[Jan 18 10:59:21] WARNING[6991]: netsock2.c:216 ast_sockaddr_split_hostport: Port disallowed in 192.168.0.30:5060
When I trying to call 112 from extention number 6001 the log says:
== Using SIP RTP CoS mark 5
> 0x68a93d30 – Strict RTP learning after remote address set to: 192.168.0.22:25116
– Executing [112@DLPN_DialPlan1:1] Macro(“SIP/6001-00000059”, “trunkdial-failover-0.3,SIP/trunk_1/12,trunk_1,”) in new stack
– Executing [s@macro-trunkdial-failover-0.3:1] GotoIf(“SIP/6001-00000059”, “0?1-fmsetcid,1”) in new stack
– Executing [s@macro-trunkdial-failover-0.3:2] GotoIf(“SIP/6001-00000059”, “0?1-setgbobname,1”) in new stack
– Executing [s@macro-trunkdial-failover-0.3:3] Set(“SIP/6001-00000059”, “CALLERID(num)=6001”) in new stack
– Executing [s@macro-trunkdial-failover-0.3:4] Set(“SIP/6001-00000059”, “CALLERID(all)=”) in new stack
– Executing [s@macro-trunkdial-failover-0.3:5] GotoIf(“SIP/6001-00000059”, “0?1-dial,1”) in new stack
– Executing [s@macro-trunkdial-failover-0.3:6] Set(“SIP/6001-00000059”, “CALLERID(all)=”) in new stack
– Executing [s@macro-trunkdial-failover-0.3:7] Set(“SIP/6001-00000059”, “CALLERID(all)=”) in new stack
– Executing [s@macro-trunkdial-failover-0.3:8] Goto(“SIP/6001-00000059”, “1-dial,1”) in new stack
– Goto (macro-trunkdial-failover-0.3,1-dial,1)
– Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial(“SIP/6001-00000059”, “SIP/trunk_1/12”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/trunk_1/12
[Jan 18 12:51:24] WARNING[12792][C-0000024a]: channel.c:6530 ast_channel_make_compatible_helper: No path to translate from SIP/trunk_1-0000005a to SIP/6001-00000059
== Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero on ‘SIP/6001-00000059’ in macro ‘trunkdial-failover-0.3’
== Spawn extension (DLPN_DialPlan1, 112, 1) exited non-zero on ‘SIP/6001-00000059’
You appear to be using a GUI that hasn’t been supported for maybe half a decade.
But it’s work 
I run asterisk in RPI 3 B+.
Internal voip in home lan network works, over vpn and wan too.
If you have better idea you can share it.
In my country I have access to detailed guide about how to move pstn phone line from A to B location over VOIP using Linksys spa 3102 and Linksys PAP2T.
But I can’t find how to do it with Asterisk.
I wan’t to move the pstn line over voip to Asterisk.
Who can help me and how much help will cost for me. I will use system at home.
Thank you.
