Hello everybody,
for 2 days I’m testing this phrase in extension.conf:
exten => _0XXX.,1,Set(status=${DEVICE_STATE(DAHDI/G1)})
exten => _0XXX.,n,NoOp(SIP/status 2 has state ${status})
exten => _0XXX.,n,GotoIf("$(status}" = “unknown”]?SIP1:ISDN1)
exten => _0XXX.,n(ISDN1),Set(status=“busy”)
exten => _0XXX.,n,Dial(DAHDI/G1/${EXTEN},r,)
exten => _0XXX.,n,Hangup()
exten => _0XXX.,n(SIP1),SipAddHeader(P-Preferred-Identity:sip:49XXXXXXX@sipconnect.sipgate.de))
exten => _0XXX.,n,Dial(SIP/${EXTEN}@sipconnect.sipgate.de,r,)
exten => _0XXX.,n,Hangup()
This is Asterisk telling:
– Executing [0XXXXXXXXXXX@intern:7] SIPAddHeader(“SIP/50-00000000”, “P-Preferred-Identity:sip:49XXXXXXXXXXX@sipconnect.sipgate.de)”) in new stack
– Executing [0XXXXXXXXXXX@intern:8] Dial(“SIP/50-00000000”, “SIP/0XXXXXXXXXXX@sipconnect.sipgate.de,r,”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/0XXXXXXXXXXX@sipconnect.sipgate.de
– SIP/sipconnect.sipgate.de-00000001 is making progress passing it to SIP/50-00000000
– SIP/sipconnect.sipgate.de-00000001 is ringing
My Asterisk 1.8 is connected via a Sangoma B700 to ISDN and also to a VOIP Trunk.
I succeed to dial out without any problem in a simple DIAL(), also with changed dialout number. But using the gotoif()-function, I’m not able to change the dialout number.
Who knows a solution?
Regards
Waldo