Passing incoming caller id to sip extension

Hi all,

I have asterisk 1.4 and a sipura-3000 working well at home but I cannot seem to get the caller id from incoming calls to the handsets connected to the FSX port. The caller id is being detected and recorded in my logs ok. It also appears to be being passed in the sip invite:

<--- SIP read from 192.168.2.4:5061 --->
INVITE sip:192.168.2.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.4:5061;branch=z9hG4bK-83988ca7
From: STEVE JONES <sip:519xxx1101@192.168.2.5>;tag=44f64fd32926e7c7o1
To: <sip:192.168.2.5>
Remote-Party-ID: STEVE JONES <sip:519xxx1101@192.168.2.5>;screen=yes;party=calling
Call-ID: 8de389c3-d2d17697@192.168.2.4
CSeq: 101 INVITE
Max-Forwards: 70
Contact: spa3102PSTN <sip:519xxx1101@192.168.2.4:5061>
Expires: 240
User-Agent: Linksys/SPA3102-3.3.6(GW)
Content-Length: 446
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER 
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 198015743 198015743 IN IP4 192.168.2.4
s=-
c=IN IP4 192.168.2.4
t=0 0
m=audio 16410 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

Alas, none of the extensions attached to the FXS port show the name/number of the caller. Any ideas?

I had a similar problem. I downloaded the latest firmware for the Sipura and the problem was resolved. I am sure it caused new problems, but I haven’t noticed any yet.