Hi All,
I have Freepbx installed with a TE110P connected to our Siemens Hipath.
The PRI is working fine but I cant figure our how to route the calls from the Hipath to the SIP trunk on the Asterisk server.
Here is the CLI output when dialing from the Siemens box.
This is when I dial 1905 - it only seems to pick our the 1
Connected to Asterisk 1.4.21.2 currently running on pbx (pid = 2988)
Verbosity is at least 6
– Accepting call from ‘0339’ to ‘1’ on channel 0/31, span 1
– Executing [1@from-pstn:1] Set(“Zap/31-1”, “__FROM_DID=1”) in new stack
– Executing [1@from-pstn:2] NoOp(“Zap/31-1”, “Received an unknown call with DID set to 1”) in new stack
– Executing [1@from-pstn:3] Goto(“Zap/31-1”, “s|a2”) in new stack
– Goto (from-pstn,s,2)
– Executing [s@from-pstn:2] Answer(“Zap/31-1”, “”) in new stack
– Executing [s@from-pstn:3] Wait(“Zap/31-1”, “2”) in new stack
– Executing [s@from-pstn:4] Playback(“Zap/31-1”, “ss-noservice”) in new stack
– <Zap/31-1> Playing ‘ss-noservice’ (language ‘en’)
– Executing [s@from-pstn:5] SayAlpha(“Zap/31-1”, “1”) in new stack
– <Zap/31-1> Playing ‘digits/1’ (language ‘en’)
– Executing [s@from-pstn:6] Hangup(“Zap/31-1”, “”) in new stack
== Spawn extension (from-pstn, s, 6) exited non-zero on ‘Zap/31-1’
– Executing [h@from-pstn:1] Hangup(“Zap/31-1”, “”) in new stack
== Spawn extension (from-pstn, h, 1) exited non-zero on ‘Zap/31-1’
– Hungup 'Zap/31-1’
pbx*CLI>
Would anyone know how I can route the calls from Siemens to the SIP trunk.
Thanks, Joe