Hi
I have a problem with iax extension calls.

If an extension a <101> call an extension b <100> after a few second from the cli and from the voixmanager the call hang up, But both user are connected and audio is passing through.

Where is the problem?
I attach an example

-- Accepting AUTHENTICATED call from 192.168.1.5:
> requested format = gsm,
> requested prefs = (),
> actual format = gsm,
> host prefs = (gsm|alaw),
> priority = mine
-- Executing [100@default:1] Macro("IAX2/101-958", "stdexten|100|IAX2/100") in new stack
-- Executing [s@macro-stdexten:1] Dial("IAX2/101-958", "IAX2/100|20") in new stack
-- Called 100
-- Call accepted by 127.0.0.1 (format gsm)
-- Format for call is gsm
-- IAX2/100-802 is ringing
-- IAX2/100-802 answered IAX2/101-958
-- Channel 'IAX2/100-802' ready to transfer
-- Channel 'IAX2/101-958' ready to transfer
-- Releasing IAX2/101-958 and IAX2/100-802
-- Hungup 'IAX2/100-802'
== Spawn extension (macro-stdexten, s, 1) exited non-zero on 'IAX2/101-958' in macro 'stdexten'
== Spawn extension (macro-stdexten, s, 1) exited non-zero on 'IAX2/101-958'
-- Hungup 'IAX2/101-958'

thanks

You have to disable IAX native transfer.

Which version of Asterisk are you using?

I'm using Asterisk 1.4.21.2

place "transfer = no"

in the general contest of the iax.conf, and if you are using the asterisk gui in the general contest of users.conf

for older asterisk (1.2.xx) version use:
"notransfer = yes"

this should work.

Yes work

thank you