[Asterisk-Users] IAX/SIP and openser problem. IAX bug?

[david.castro]

Hello.
I am in a strange situation. I have two asterisk. Asterisk "A" makes a
call for asterisk "B" by IAX. Asterisk "B" recives the call and delivers
it to Openser by SIP. The problem is openser printing this in the screen:

ERROR: parse_to : unexpected char ["] in status 5: <<"David" <sip:>> .
ERROR:parse_from_header: bad from header
ERROR: new_t: no valid From in INVITE
ERROR: t_newtran: new_t failed
ERROR: sl_reply_error used: I'm terribly sorry, server error occurred 
(1/SL)

I got nex sip messages:
U 2006/01/16 12:21:10.968713 10.2.11.35:5062 -> 10.2.11.35:5060
 INVITE sip:204 at 10.2.11.35 SIP/2.0..Via: SIP/2.0/UDP 
10.2.11.35:5062;branch=z9hG4bK31f811a3;rport..From: "Zyxel" 
<sip:Zyxel<sip:204 at 1021150;user=phone at 10.2.11.35:5062>;tag=as56432543..To: 
<sip:204 at 10.2.11.35>..Contact: <sip:Zyxel<sip:204 at 1
 021150;user=phone at 10.2.11.35:5062>..Call-ID: 
7bf16eeb00aca5cd2bd303a93f59bfc4 at 10.2.11.35..CSeq: 102 INVITE..User-Agent:
  Asterisk PBX..Max-Forwards: 70..Date: Mon, 16 Jan 2006 11:21:10 
GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
 SUBSCRIBE, NOTIFY..Content-Type: application/sdp..Content-Length: 
463....v=0..o=root 16102 16102 IN IP4 10.2.11.35..s=s
 ession..c=IN IP4 10.2.11.35..t=0 0..m=audio 14640 RTP/AVP 0 8 4 111 18 
3 97 7 110 5 101..a=rtpmap:0 PCMU/8000..a=rtpmap
 :8 PCMA/8000..a=rtpmap:4 G723/8000..a=rtpmap:111 
G726-32/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:3 G
 SM/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:7 LPC/8000..a=rtpmap:110 
speex/8000..a=rtpmap:5 DVI4/8000..a=rtpmap:101 teleph
 one-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..

U 2006/01/16 12:21:10.969161 10.2.11.35:5060 -> 10.2.11.35:5062
 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 
10.2.11.35:5062;branch=z9hG4bK31f811a3;rport=5062..From: "Zyxel" 
<sip:Zyxel<sip:20
 4 at 1021150;user=phone at 10.2.11.35:5062>;tag=as56432543..To: 
<sip:204 at 10.2.11.35>..Call-ID: 7bf16eeb00aca5cd2bd303a93f59bf
 c4 at 10.2.11.35..CSeq: 102 INVITE..Server: OpenSer (1.0.0-tls 
(i386/linux))..Content-Length: 0..Warning: 392 10.2.11.35:5
 060 "Noisy feedback tells:  pid=15852 req_src_ip=10.2.11.35 
req_src_port=5062 in_uri=sip:204 at 10.2.11.35 out_uri=sip:204
 @10.2.11.35 via_cnt==1"....

U 2006/01/16 12:21:10.969278 10.2.11.35:5060 -> 10.2.11.35:5062
 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 
10.2.11.35:5062;branch=z9hG4bK31f811a3;rport=5062..From: "Zyxel" 
<sip:Zyxel<sip
 :204 at 1021150;user=phone at 10.2.11.35:5062>;tag=as56432543..To: 
<sip:204 at 10.2.11.35>;tag=329cfeaa6ded039da25ff8cbb8668bd2.
 53ff..Call-ID: 7bf16eeb00aca5cd2bd303a93f59bfc4 at 10.2.11.35..CSeq: 102 
INVITE..Server: OpenSer (1.0.0-tls (i386/linux)).
 .Content-Length: 0..Warning: 392 10.2.11.35:5060 "Noisy feedback 
tells:  pid=15852 req_src_ip=10.2.11.35 req_src_port=5
 062 in_uri=sip:204 at 10.2.11.35 out_uri=sip:204 at 10.2.11.35 via_cnt==1"....

U 2006/01/16 12:21:10.969356 10.2.11.35:5062 -> 10.2.11.35:5060
 ACK sip:204 at 10.2.11.35 SIP/2.0..Via: SIP/2.0/UDP 
10.2.11.35:5062;branch=z9hG4bK31f811a3;rport..From: "Zyxel" <sip:Zyxel
 <sip:204 at 1021150;user=phone at 10.2.11.35:5062>;tag=as56432543..To: 
<sip:204 at 10.2.11.35>;tag=329cfeaa6ded039da25ff8cbb8668
 bd2.53ff..Contact: 
<sip:Zyxel<sip:204 at 1021150;user=phone at 10.2.11.35:5062>..Call-ID: 
7bf16eeb00aca5cd2bd303a93f59bfc4 at 10
 .2.11.35..CSeq: 102 ACK..User-Agent: Asterisk PBX..Max-Forwards: 
70..Content-Length: 0....

I think it's all right except the address:
From: "Zyxel" <sip:Zyxel<sip:204 at 1021150;user=phone at 10.2.11.35:5062>

What do you think about all that?

Do you know the reason?
Is this a bug? Which is guilty, asterisk or openser?
I have this problem only in this scenario.