asterisk users - Jul 2006 - One extension can transfer internal calls, can't transfer incoming external calls

Greetings list,

I've been bashing my head against a brick wall for a couple of weeks now to
try and get this sorted, have been scouring google/the asterisk-users list
archives to no avail.

The problem I am having is that one extension (an off-site iaxy) cannot
transfer incoming calls from our IAX provider, but can transfer internal
calls. We can transfer incoming external calls on site using our cisco
7960's, just not remotely with the iaxy.

I thought I had cracked it this morning when I found out about the
"notransfer=yes" option for the IAX2 peers, to prevent the call from
being
reinvited by the iaxy, and not going through the asterisk server, but
although the call is staying through the asterisk box, it's still not
possible to transfer an incoming call from the iaxy to one of the cisco
phones.

Basically, this is what works and doesn't

Iax provider -> asterisk server -> iaxy   =  iaxy cannot transfer the call
Iax provider -> asterisk server -> cisco 7960 = 7960 can transfer the call
Cisco 7960 <-> asterisk server <-> iaxy = whoever makes the call,
both users
can transfer.

The blind transfer is being done by using the # key, we're using asterisk
1.0.9 (downgraded after trying a higher version (think it was .23ish) that
dropped external calls after 3 minutes).

The (I think) relevant bits from extensions.conf, sip.conf, and iax.conf
(suitably munged for public distribution ;) ) are below. I've tried adding
Tt to the end of every dial string I can, and even tried it on the end of
the GotoIfTime line of the [iaxprovider-in] section of extensions.conf,
which I doubt will make any difference if it's there or not.

The DTMF detection is working fine for both the iaxy and the cisco phone,
both users can use the voicemail application fine, and dtmf tones get passed
through to call centres etc.

Has anybody come across anything like this in the past, where certain
extensions can only sometimes forward calls? I have noticed that in the iaxy
provisioning it's possible to disable call transfer, does this mean that the
iaxy has it's own key combination for call transfer?

Cheers in advance,
Mat


extensions.conf

[default]

 exten => 23,1,dial(SIP/sipuser,12,Tt) 
 exten => 23,2,Voicemail(su23) 
 exten => sipuser,1,goto(23,1)

 exten => 34,1,dial(IAX2/iaxy1@iaxy1,20,Tt) 
 exten => 34,2,Voicemail(su34) 
 

[iaxprovider-in]
  exten => incomingiaxprovidernumber,1,Answer 
  exten => incomingiaxprovidernumber,2,Wait,1 
  exten => incomingiaxprovidernumber,3,NoOp(--- ${CALLERID} calling on
INCOMING IAX PROVIDER (${EXTEN}) ---) 
  exten => incomingiaxprovidernumber,4,Wait,1 
  exten =>
incomingiaxprovidernumber,5,GotoIfTime(9:00-17:00|mon-fri|*|*?office-hours,s
,1,Tt)
  exten => incomingiaxprovidernumber,6,Background(officeclosed) 
  exten => incomingiaxprovidernumber,7,Voicemail(s01) 
  exten => incomingiaxprovidernumber,8,Hangup 
  
[office-hours]  
  exten => s,1,NoOp() 
  exten => s,2,NoOp() 
  exten => s,3,NoOp() 
  exten => s,4,Dial(SIP/sipuser&IAX2/iaxy1@iaxy1,18,Tt) 
  exten => s,5,Answer 
  exten => s,6,Wait,1 
  exten => s,7,Voicemail(su01) 
  exten => s,8,Hangup





iax.conf:

[iaxy1]
type=friend
accountcode=iaxy
host=dynamic
notransfer=yes 
username=iaxy1
secret=secret
context=default
disallow=all
allow=ulaw 
callerid="IAXy 1" <34>
trunk=no


sip.conf

 [sipuser] 
 type=friend 
 host=dynamic
 dtmfmode=inband 
 username=ciscophone
 secret=ciscophone
 qualify=200
 reinvite=no
 canreinvite=no
 disallow=all
 allow=ulaw
 allow=alaw
 nat=yes
 mailbox=23,01
 callgroup=1
 pickupgroup=1
 callerid=Mat <23>