[asterisk-users] Call transfer issues

[Kevin Smith]

Hey everyone,

Currently we are running IP601's with 1.6.6 SIP firmware on Asterisk 
1.2.10. It has been reported to me when doing an attended transfer the 
audio drops out. I ran a few different tests and here is what I noticed.

1. Blind transfers work with no problem.
2. Attended transfers were you transfer the call before the person picks 
up works.
3. If the person the call is being transferred to answers and then the 
transfer completes, the audio drops.

I noticed in the CLI the following (I replaced the number with XXXXXXX's)

 -- Attempting native bridge of SIP/989XXXXXXX-b76167c8 and 
SIP/989XXXXXXX-08f956b8
  == Parsing '/etc/asterisk/manager.conf': Found
    -- Stopped music on hold on Zap/2-1
  == Spawn extension (Mercury-Directory-Dialer, 989XXXXXXX, 8) exited 
non-zero on 'SIP/989XXXXXXX-b76167c8<ZOMBIE>'
    -- Executing Hangup("SIP/989XXXXXXX-b76167c8<ZOMBIE>", "") in new stack
  == Spawn extension (Mercury-Directory-Dialer, h, 1) exited non-zero on 
'SIP/989XXXXXXX-b76167c8<ZOMBIE>'
    -- Incoming call: Got SIP response 500 "Internal Server Error" back 
from 64.7.177.103

Now what I noticed is that once the transfer is done, I'm still 
connected the the person that called me to do an attended transfer. 
However, if I hang up the phone, the call drops. If I place the call on 
hold and take them off hold, audio is resumed and everything works 
normally.

Here is the conf information

exten => s,1,SetCallerID(${ARG1})
exten => s,n,Set(DST_EXT_NUM=${ARG2})
exten => s,n,gotoif,$[${ARG2}=989XXXXXX]?TIME:GOON     ;Add test if 
hours is the basis for voice mail

exten => s,n(GOON),AGI(VoiceMail.php)   ;Test for phone status
exten => s,n,Noop(Testing Completed||Code:${ON_PHONE}${IN_QUEUE})
exten => s,n,Dial(SIP/${ARG2},25)

...VoiceMail choice....

exten => h,1,HangUp()

Where I have VoiceMail choice it takes the variables from the AGI script 
and decides which voice message to play. But the problem is happening 
before that occurs so I don't think it has anything to do with the problem.

Any ideas to what could be the cause or how to correct it? SIP version 
or does the new asterisk build have any new features enabled by default 
that the older build would not? Any suggestions or thoughts would be 
greatly helpful.

Kevin