Very strange result - someone from the PSTN calls my customer. The call is handed to asterisk via a SIP account provided by the phone company. Asterisk is configured to answer and push the call to the receptionist.
exten => _1902600015,1,Wait(1)
exten => _1902600015,n,Answer()
exten => _1902600015,n,goto(DID_trunk_2,s,1)
Where DID_trunk_2 is:
exten => s,n,Dial(SIP/6600,30,r)
exten => s,n,Dial(SIP/6105&SIP/6109&SIP/6114&SIP/6116&SIP/6106,20,r)
exten => s,n,Background(asterisk-mainmenu)
exten => s,n,Playback(Goodbye)
exten => s,n,Hangup
What happens is when the outside person calls, they hear a ring provided by the phone company because asterisk wait(1), then the call is answered and sent to SIP/6600. Now while the receptionist’s phone DOES ring, the calling party only hears one ring after the the Dial command is processed.
As I understand it, the additional rings are generated by asterisk in the audio stream and delivered back to the PSTN via RTP and since there’s dead air, the calling party hangs up thinking there is something wrong. This is a new install - and running asterisk 220.127.116.11
This is an example of the messages that come over the asterisk cmd console:
-- Executing [1902600014@default:1] Wait("SIP/1902600015-b720a6b0", "1") in new stack -- Executing [1902600014@default:2] Answer("SIP/1902600015-b720a6b0", "") in new stack -- Executing [1902600014@default:3] Goto("SIP/1902600015-b720a6b0", "DID_trunk_1|s|1") in new stack -- Goto (DID_trunk_1,s,1) -- Executing [s@DID_trunk_1:1] Dial("SIP/1902600015-b720a6b0", "SIP/6000|30|m") in new stack -- Called 6000> -- Started music on hold, class 'default', on SIP/1902600015-b720a6b0 -- SIP/6000-097310a0 is ringing
Extension Changed 6000[numberplan-custom-1] new state Ringing for Notify User 6000
This one has me stumped. Any suggestions?