Hello,
I have running asterisk 1.6.2.16.1 on centos5 working ok, but there is a small problem. On outgoing calls and calls to other extension the ringtone at callersite stop after second ring and you hear nothing until the other end picks up the phone.
Location: Spain, we use national public phone numbers for extensions
extensions.conf
;Nacional
exten => _[6789]XXXXXXXX,1,Dial,(SIP/PROV/0034${EXTEN},30,r)
exten => _[6789]XXXXXXXX,n,Hangup
;Interno
exten => _970123456,1,macro(interno,${EXTEN})
exten => _970123457,1,macro(interno,${EXTEN})
[macro-interno]
exten => s,1,Dial(SIP/${ARG1},30,r)
exten => s,2,GotoIf($["${DIALSTATUS}" = “BUSY”]?100:3)
exten => s,3,Voicemail(${ARG1}@default,u)
exten => s,4,Hangup
exten => s,100,Busy(10)
exten => s,101,Hangup
log:
– Executing [900101010@default:1] Dial(“SIP/970123456-00000492”, “SIP/PROV/0034900101010,30,r”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
– Called PROV/0034900101010
– SIP/PROV-00000493 is making progress passing it to SIP/970123456-00000492
– SIP/PROV-00000493 answered SIP/970123456-00000492
– Packet2Packet bridging SIP/970123456-00000492 and SIP/PROV-00000493
Any help would be apreciated