asterisk users - May 2006 - No sound when the call is diverted

Hi Guys,

I'm having sound problems when diverting a call using asterisk@home 1.5. I 
am using the following configuration in extensions_custom.conf, 
extensions_additional.conf and extensions.conf

[custom-Sales]
exten => s,1,SetVar(DivertNumber=02XXXXXXXX)
exten => s,2,Dial(SIP/116, 15)
exten => s,3,Goto(outrt-010-outside3,9${DivertNumber},1)

(i have replaced the diverted phone number with XXXXXXXX above)


[outrt-010-outside3] it's the context to make outbound calls via SIP trunk

The custom-Sales context is used in the following ext-did context for 
incoming calls,

[ext-did]

exten => 02YYYYYYYY,1,SetVar(FROM_DID=02YYYYYYYY)	;
exten => 02YYYYYYYY,2,Goto(custom-Sales,s,1)	;

(i have replaced the called DID number with YYYYYYYY above)


So when ringing 02YYYYYYYY, after 15 seconds the call is successfully 
diverted to 02XXXXXXXX
however when the call is answered there is not any sound on any end. Can any 
one that has this
working please point me on the right direction I will appreciate it. I'm not
too sure what
would be affecting the sound on the call as it is diverted.

See below for relevant debug output from the console.

-- Executing SetVar("SIP/02YYYYYYYY-a1a7",
"FROM_DID=02YYYYYYYY") in new
stack
    -- Executing Goto("SIP/02YYYYYYYY-a1a7",
"custom-Sales|s|1") in new
stack
    -- Goto (custom-Sales,s,1)
    -- Executing SetVar("SIP/YYYYYYYY-a1a7",
"DivertNumber=02XXXXXXXX") in
new stack
    -- Executing Dial("SIP/02YYYYYYYY-a1a7", "SIP/116| 15")
in new stack
    -- Called 116
    -- SIP/116-ca11 is ringing
    .
    .
    .

    -- Executing SetVar("SIP/02YYYYYYYY-e487",
"DIAL_NUMBER=02XXXXXXXX") in
new stack
    -- Executing SetVar("SIP/02YYYYYYYY-e487",
"DIAL_TRUNK=11") in new stack
    -- Executing AGI("SIP/02YYYYYYYY-e487",
"fixlocalprefix") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
  fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
    -- AGI Script fixlocalprefix completed, returning 0
    -- Executing SetVar("SIP/02YYYYYYYY-e487",
"OUTNUM=02XXXXXXXX") in new
stack
    -- Executing Cut("SIP/02YYYYYYYY-e487",
"custom=OUT_11|:|1") in new
stack
    -- Executing GotoIf("SIP/02YYYYYYYY-e487", "0?20") in
new stack
    -- Executing NoOp("SIP/02YYYYYYYY-e487", "02XXXXXXXX")
in new stack
    -- Executing Dial("SIP/02YYYYYYYY-e487",
"SIP/sales/02XXXXXXXX") in new
stack
    -- Called sales/02XXXXXXXX
    -- SIP/sales-7d0b is making progress passing it to SIP/02YYYYYYYY-e487
    -- SIP/sales-7d0b answered SIP/02YYYYYYYY-e487
    -- Attempting native bridge of SIP/0282058347-e487 and SIP/sales-7d0b

asterisk*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)   Format

202.177.222.24   02XXXXXXXX  01f672b7696  00103/00000   g729
202.177.222.24   02YYYYYYYY  447542a4000  00101/31350   g729
4 active SIP channel(s)

(I changed the numbers to XXXXXXXX and YYYYYYYY in the debug output as well)

Thanks in advance,

Paul

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