[asterisk-users] callback and bridge problem

[Adam KOSA]

Hi guys,

sorry for the long e-mail, i'm only trying to give as much information 
as i think is relevant to my problem (console log, sip.conf and 
extension.conf parts).

i've been practicing with callback for a while, but i'm at a dead end. 
I hope somebody can help me to move on.

i have troubles getting two calls bridged together.  Scenario is the 
following:

- asterisk calls my cell via a SIP provider called neophone
- my cell rings, i pick up, and i find myself in:

[internal]
; callback is directed here
exten => s,1,WaitExten,50
include => voicemail-context
include => internal_extensions-context
include => dialout_prefix-context


because my call file looks like this:

Channel: SIP/06202222222 at neophonex
Context: internal
Extension: s
Priority: 1

where 06202222222 is my cell.

- after picking up, i dial 95206301111111 where 952 is the dialing 
prefix, 06301111... is another cell.  952 is a prefix for another 
registered account at the same provider (one account is allowed to place 
one call at a time).

After this as you can see, the second number (1111..) is dialed. 
However when i pick up the phone, the call hangs up.

This also happens when i use another prefix (another provider, even 
PSTN) for the second call too.

The relevant part from asterisk console is at the end of this e-mail, i 
don't really understand the warning messages.

----- configs:

In sip.conf, the configuration for the two SIP accounts are:

register => 0621380....:password at sip.neophonex.hu
register => 0621381....:password at sip.neophonex.hu

[neophonex]
type=friend
host=sip.neophonex.hu
context=dialout_prefix-context
username=0621380....
authname=0621380....
fromuser=0621380....
secret=password
callerid=0621380....
fromdomain=sip.neophonex.hu
disallow=all
allow=alaw
allow=g723
dtmfmode=inband
nat=no

[neophonex-out]
type=friend
host=sip.neophonex.hu
context=dialout_prefix-context
username=0621381....
authname=0621381....
fromuser=0621381....
secret=password
callerid=0621381....
fromdomain=sip.neophonex.hu
disallow=all
allow=alaw
allow=g723
dtmfmode=inband
nat=no


extension.conf:

exten => _952.,1,Playback(kapcsolas,noanswer)
exten => _952.,n,Set(CALLERID(name)=0621380....)
exten => _952.,n,Dial(SIP/${EXTEN:3}@neophonex-out)

I have tried every possible setting i know about, but still, when i call 
outside, via 'turning around' in asterisk, both cells hung up when 
answering the call.  I have tried calling a regular landline phone 
number but still hanging up.

Both accounts are valid, registered and have enough credit to dial 
outside its voice network.

The only way the call does not hung up is when i dial extensions within 
asterisk.

The asterisk log:

     -- Called 06301111111 at neophonex-out
     -- Call on SIP/neophonex-out-081a9cc0 left from hold
     -- SIP/neophonex-out-081a9cc0 is making progress passing it to 
SIP/neophonex-081ab240
[Jun 25 16:57:07] WARNING[18232]: chan_sip.c:11839 
handle_response_invite: Re-invite to non-existing call leg on other UA. 
SIP dialog '44c971692552f3245aa7b4e834bdafab at sip.neophonex.hu'. Giving up.
     -- Call on SIP/neophonex-out-081a9cc0 left from hold
     -- SIP/neophonex-out-081a9cc0 answered SIP/neophonex-081ab240
     -- Native bridging SIP/neophonex-081ab240 and 
SIP/neophonex-out-081a9cc0
[Jun 25 16:57:10] WARNING[18232]: chan_sip.c:11839 
handle_response_invite: Re-invite to non-existing call leg on other UA. 
SIP dialog '240370c953d10a75430c0e2e0d4764a6 at sip.neophonex.hu'. Giving up.
   == Spawn extension (internal, 95206301111111, 3) exited non-zero on 
'SIP/neophonex-081ab240'
[Jun 25 16:57:10] NOTICE[18440]: pbx_spool.c:351 attempt_thread: Call 
completed to SIP/06202222222 at neophonex


Please help me to figure out why the calls are hung up.

Thanks
Adam