[asterisk-users] No progress tones on transferred call

[CB]

Asterisk 1.4
We are experiencing an issue on transfers where no progress tones are heard
by the caller:
1. Call from 1593 (SPA525G 0026998D2FFFF) to 1595 (SPA922 000B820AFFFFF).
1595 answers
2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4AFFFF). 1595 hears
MoH. 
3. 1597 starts ringing and 1593 presses transfer again. MoH stops but 1595
hears no ringing
When xfer is pressed and the extension is dialled:
U 203.89.001.001:5060 -> 121.98.001.001:1034
INVITE sip:1CDF0F4AFFFF at 192.168.1.72:5060 SIP/2.0..Via: SIP/2.0/UDP
203.89.001.001:5060;branch=z9hG4bK5286810e;rport..From: "C Allerid"
<sip:1593 at 203.89.001.001>;tag=as72616c50..To:
<sip:1CDF0F4AFFFF at 192.168.1.72:5060>..Contact:
<sip:1593 at 203.89.001.001>..Call-ID:
59ba10300b9b8cb5684eba2368c90a77 at 203.89.001.001..CSeq: 102
INVITE..User-Agent: Asterisk PBX..Max-Forwards: 70..Date: Tue, 05 Jun 2012
08:05:02 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO..Supported: replaces..Content-Type:
application/sdp..Content-Length: 262....v=0..o=root 3031 3031 IN IP4
203.89.001.001..s=session..c=IN IP4 203.89.001.001..t=0 0..m=audio 13728
RTP/AVP 0 3 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:101
telephone-event/8000..a=fmtp:1010-16..a=ptime:20..a=sendrecv..
U 121.98.001.001:1034 -> 203.89.001.001:5060
SIP/2.0 100 Trying..To: <sip:1CDF0F4AFFFF at 192.168.1.72:5060>..From: "C
Allerid" <sip:1593 at 203.89.001.001>;tag=as72616c50..Call-ID:
59ba10300b9b8cb5684eba2368c90a77 at 203.89.001.001..CSeq: 102 INVITE..Via:
SIP/2.0/UDP 203.89.001.001:5060;branch=z9hG4bK5286810e..Server:
Cisco/SPA508G-7.4.9c..Content-Length: 0....
U 121.98.001.001:1034 -> 203.89.001.001:5060
SIP/2.0 180 Ringing..To:
<sip:1CDF0F4AFFFF at 192.168.1.72:5060>;tag=53e23c5265d60f06i0..From: "C
Allerid"
<sip:1593 at 203.89.001.001>;tag=as72616c50..Call-ID:59ba10300b9b8cb5684eba2368
c90a77 at 203.89.001.001..CSeq: 102 INVITE..Via: SIP/2.0/UDP
203.89.001.001:5060;branch=z9hG4bK5286810e..Contact: "$USER"
<sip:1CDF0F4AFFFF at 192.168.1.72:5060>..Server:
Cisco/SPA508G-7.4.9c..Content-Length: 0....
After transfer is pressed the second time there is no further SIP messages
with 

Asterisk CLI
-- Executing [s at macro-dial:12] Dial("SIP/000B820AFFFFF-00002d0a", 
"SIP/000E08D6FFFF&SIP/1CDF0F4AFFFF&SIP/000E08D6FFFF1|20|tTwWr") in new 
stack
-- Called 1CDF0F4AFFFF
-- SIP/1CDF0F4AFFFF-00002d0b is ringing
-- Stopped music on hold on SIP/0026998D2FFFF-00002d08

Updated sip.conf
progressinband=yes 

This didn't make any difference

I've tried calls in different directions in case it is to do with the
particular phone firmware but the direction is irrelevant.

Any suggestions appreciated or if you require further information please
ask.