Putting peer SIP/XXX into PBX again


#1

I am upgrading our system from 1.4.22 to 1.8.7.1, when I perform an attended transfer:
A Audiocodes Gateway FXS
B Polycom 670 One
C Polycom 670 Two

B attended transfers A to C works flawlessly however

when I follow up with another attended transfer

C attended transfers A to B

although the call transfers, channel C does not hang up and re-enters the dialplan

putting peer SIP/C into PBX again
where the peer calls itself and fails.

putting peer SIP/CO999x1001.1-0000010e into PBX again
– Executing [1001@phones:2] GotoIf(“SIP/CO999x1001.1-0000010e”, “0?6”) in new stack
– Executing [1001@phones:3] GotoIf(“SIP/CO999x1001.1-0000010e”, “1?8”) in new stack
– Goto (phones,1001,8)
– Executing [1001@phones:8] UserEvent(“SIP/CO999x1001.1-0000010e”, “AttendedTransfer,extension: 1001,Conf: 347,Position: 02,Channel: SIP/CO999x1001.1-0000010e,Uniqueid: 1325638299.765,ControllerCallID: 245593053570876347”) in new stack
– Executing [1001@phones:9] Dial(“SIP/CO999x1001.1-0000010e”, “SIP/CO999x1001.1,gM(nomusic)”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/CO999x1001.1
– SIP/CO999x1001.1-0000010e requested special control 17, passing it to SIP/CO999x1001.1-00000111
== Spawn extension (phones, 1001, 9) exited non-zero on ‘SIP/CO999x1001.1-0000010e’

I do not have this issue on 1.4.22
is there a way to prevent the channel from re-entering the dialplan ? or is this a bug ?

update:
I removed the g from the dial command. Now the channel drops into the s extension, which is better but I need the s extension for different dial plan processing. (obtw it starts at priority 2 … both cases)