[Asterisk-Users] Asterisk v1.0 sends incorrect invite to Sipura SPA-3000?

[Mike Benoit]

I recently upgraded from a few month old CVS version of Asterisk to
v1.0.1, and dialing out through my SPA-3000 stopped working. 

Notice right after INVITE, in the old CVS version, it includes the
number I'm trying to dial (8019596) which works fine, however in v1.0.1,
it doesn't include the number and of course the dial fails. 

Did a config option change out from underneath me or something?

Old CVS version of Asterisk: (works fine)
--------------------------------
Oct  4 23:18:07 192.168.1.190 INVITE sip:8019596 at 192.168.1.190:5061
SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.3:5060
;branch=z9hG4bK522738f1^M From: "asterisk"
<sip:asterisk at 192.168.1.3>;tag=as52daeb2d^M To: <sip:8019596 at 192.168
.1.190:5061>^M Contact: <sip:asterisk at 192.168.1.3>^M Call-ID:
2d02c0cc392a99264f5f09666c3ff875 at 192.168.1.3^M CS
eq: 102 INVITE^M User-Agent: Asterisk PBX^M Date: Tue, 05 Oct 2004
06:18:07 GMT^M Allow: INVITE, ACK, CANCEL, O
PTIONS, BYE, REFER^M Content-Type: application/sdp^M Content-Length:
214^M ^M v=0^M o=root 27838 27838 IN IP4 1
92.168.1.3^M s=session^M c=IN IP4 192.168.1.3^M t=0 0^M m=audio 13232
RTP/AVP 0 101^M a=rtpmap:0 PCMU/8000^M a=
rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=silenceSupp:off -
- - -^M


Asterisk v1.0.1: (doesn't work)
---------------------------------
Sep 30 20:38:45 192.168.1.190 INVITE sip:500 at 192.168.1.190:5061
SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK36cac9c5^M
From: "2508019596" <sip:2508019596 at 192.168.1.3>;tag=as7f1bd067^M To:
<sip:500 at 192.168.1.190:5061>^M Contact: <sip:2508019596 at 192.168.1.3>^M
Call-ID: 0701aef72bda06da6e3fb4593dc78e31 at 192.168.1.3^M CSeq: 102
INVITE^M User-Agent: Asterisk PBX^M Date: Fri, 01 Oct 2004 03:38:45
GMT^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER^M Content-Type:
application/sdp^M Content-Length: 214^M ^M v=0^M o=root 22051 22051 IN
IP4 192.168.1.3^M s=session^M c=IN IP4 192.168.1.3^M t=0 0^M m=audio
14996 RTP/AVP 0 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:101 telephone-
event/8000^M a=fmtp:101 0-16^M a=silenceSupp:off - - - -^M


-- 
Mike Benoit <ipso at snappymail.ca>