[Asterisk-Users] Attended transfer problem with Atxfer

[Mickaël Cissé]

Hi everyone,

I'm trying the new atxfer functionality. All seems to work fine at the
beginning, but there is no audio between the party at the end of the
transfer. Plus, after that, even normal calls won't work well (they
can't hangup!).

I'm using the Asterisk CVS from 2005-01-10 with Asterisk-OH323.

Here is my dialplan:
[default]
exten => h,1,NoOp(bye)

exten => _6.,1,Dial(OH323/${EXTEN:1}|20|CrtT)
exten => _6.,2,Hangup

I'm doing the following manipulation:
Phone 41 dial 651.
Phone 51 ring and answer.
I dial *2642 on phone 41
Phone 42 ring and answer.
Phone 41 hang up.
I hear a beep in phone 42.
And now, there is no sound with 42 and 51 :-(
So, I hangup 51, then 42.

Here are the messages from Asterisk:

-- Executing Dial("OH323/R31698", "OH323/51|20|CrtT") in new stack
    -- H.323 call to 51 with codec(s) ulaw 
    -- Called 51
    -- OH323/L29479 is ringing
    -- OH323/L29479 answered OH323/R31698
    -- Started music on hold, class 'default', on OH323/L29479
    -- Playing 'pbx-transfer' (language 'en')
    -- Executing Dial("Local/642 at h323-b9d1,2", "OH323/42|20|CrtT") in
new stack
    -- H.323 call to 42 with codec(s) ulaw 
    -- Called 42
    -- OH323/L29480 is ringing
    -- OH323/L29480 answered Local/642 at h323-b9d1,2
    -- H.323 call 'ip$192.168.254.137:35504/31698' cleared, reason 4
(Cleared by remote user)
    -- Stopped music on hold on OH323/L29479
    -- Playing 'beep' (language 'en')
  == Spawn extension (h323, 651, 1) exited non-zero on 'OH323/R31698'
    -- Executing NoOp("OH323/R31698", "bye") in new stack
    -- Hungup 'OH323/R31698'

Any idea why it doesn't work?

Thanks for reading me.

Mickaël Cissé