asterisk users - Jul 2003 - Cisco 7960 Transfer Call drop problem

Hi,
 
I'm having problems with transfer from an analog line via a X100p and Cisco
7960's running SIP.
 
With an attended transfer the a call comes in, I transfer it to another
7960, they answer I announce the call, press transfer again, the two parties
talk for 1-2 seconds then the analog line drops, though the Cisco phone is
not aware of this, i.e. nothing on the screen changes. The console output
for this is below. Interestingly enough I seem to have the same problem with
an incoming SIP call, transferring it to another SIP ext, console output
from that below as well.
 
With a blind transfer a call comes in, I transfer it to another extension,
the analog caller hears the hold music, the 7960 that was transferred the
call acts as if it is online with the call but isn't. If the extension that
was transferred the call puts the line on hold and picks it up then the
lines are connected fine. 
 
--------------------Analog to SIP transfer--------------------------
  -- Zap/1-1 answered SIP/206-369e
    -- Started music on hold, class 'default', on Zap/1-1
    -- Executing Macro("SIP/206-bcd1",
"stdexten|SIP/202|202") in new stack
    -- Executing Dial("SIP/206-bcd1", "SIP/202|15") in new
stack
    -- Called 202
    -- SIP/202-7264 is ringing
    -- SIP/202-7264 answered SIP/206-bcd1
    -- Attempting native bridge of SIP/206-bcd1 and SIP/202-7264
    -- Started music on hold, class 'default', on SIP/202-7264
    -- Stopped music on hold on SIP/202-7264
    -- Stopped music on hold on Zap/1-1
  == Spawn extension (intern-ext, 91415XXXXXXX, 1) exited non-zero on
'SIP/206-369e'
  == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'Zap/1-1'
in
macro 'stdexten'
  == Spawn extension (intern-ext, 202, 1) exited non-zero on 'Zap/1-1'
    -- Hungup 'Zap/1-1'
 
--------------------SIP to SIP transfer----------------------------------
   -- Executing Macro("SIP/206-effd",
"stdexten|SIP/255|255") in new stack
    -- Executing Dial("SIP/206-effd", "SIP/255|15") in new
stack
    -- Called 255
    -- SIP/255-8cd8 is ringing
    -- SIP/255-8cd8 answered SIP/206-effd
    -- Attempting native bridge of SIP/206-effd and SIP/255-8cd8
    -- Started music on hold, class 'default', on SIP/255-8cd8
    -- Executing Macro("SIP/206-8437",
"stdexten|SIP/202|202") in new stack
    -- Executing Dial("SIP/206-8437", "SIP/202|15") in new
stack
    -- Called 202
    -- SIP/202-5c6b is ringing
    -- SIP/202-5c6b answered SIP/206-8437
    -- Attempting native bridge of SIP/206-8437 and SIP/202-5c6b
    -- Started music on hold, class 'default', on SIP/202-5c6b
    -- Stopped music on hold on SIP/202-5c6b
    -- Stopped music on hold on SIP/255-8cd8
    -- Attempting native bridge of SIP/206-effd and SIP/255-8cd8
    -- Attempting native bridge of SIP/255-8cd8 and SIP/202-5c6b
    -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back
from
67.xxx.xxx.xxx
  == Spawn extension (macro-stdexten, s, 1) exited non-zero on
'SIP/206-effd' in macro 'stdexten'
  == Spawn extension (intern-ext, s, 1) exited non-zero on
'SIP/206-effd'
  == Spawn extension (macro-stdexten, s, 1) exited non-zero on
'SIP/255-8cd8' in macro 'stdexten'
 
 
Ideas?
 
Thanks,
Justin
 
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20030714/821c766c/attachment.htm