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

[david.castro]

Hello.
Asterisk "A" is version 1.2.1.
Asterisk "B" is version 1.0.9.

If I call by IAX from Asterisk "A" to B, and after that, Asterisk "B" 
call by SIP to Openser, the call works.
The invite message from Asterisk to openser by Sip is:

U 2006/01/17 17:50:49.261265 10.2.11.50:5061 -> 10.2.11.50:5060
  INVITE sip:205 at 10.2.11.50 SIP/2.0..Via: SIP/2.0/UDP 
10.2.11.50:5061;branch=z9hG4bK722ced70..From: "Analogico" <sip:206@
  10.2.11.50:5061>;tag=as4cdf4533..To: <sip:205 at 10.2.11.50>..Contact: 
<sip:206 at 10.2.11.50:5061>..Call-ID: 2112bb831f0c32e
  b72118bc703d791bf at 10.2.11.50..CSeq: 102 INVITE..User-Agent: Asterisk 
PBX..Date: Tue, 17 Jan 2006 16:50:49 GMT..Allow: I
  NVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Content-Type: 
application/sdp..Content-Length: 442....v=0..o=root 17529 17529
  IN IP4 10.2.11.50..s=session..c=IN IP4 10.2.11.50..t=0 0..m=audio 
12064 RTP/AVP 0 4 3 8 111 5 7 18 110 97 101..a=rtpmap
  :0 PCMU/8000..a=rtpmap:4 G723/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8 
PCMA/8000..a=rtpmap:111 G726-32/8000..a=rtpmap:5 DV
  I4/8000..a=rtpmap:7 LPC/8000..a=rtpmap:18 G729/8000..a=rtpmap:110 
speex/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:101 telep
  hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..



If I call by IAX from Asterisk "B" to Asterisk "A", and afterwords, 
Asterisk "A" cal by SIP to openser, it fails.
The invite message from Asterisk to ser is:

U 2006/01/17 17:57:16.896269 10.2.11.35:5062 -> 10.2.11.35:5060
  INVITE sip:206 at 10.2.11.35 SIP/2.0..Via: SIP/2.0/UDP 
10.2.11.35:5062;branch=z9hG4bK256111a9;rport..From: "David" <sip:"D
  avid"<sip:205 at 1021150@10.2.11.35:5062>;tag=as2652cbb1..To: 
<sip:206 at 10.2.11.35>..Contact: <sip:"David"<sip:205 at 1021150@
  10.2.11.35:5062>..Call-ID: 
67fbab514449a6b84b2909d404f7ed6b at 10.2.11.35..CSeq: 102 
INVITE..User-Agent: Asterisk PBX..Max
  -Forwards: 70..Date: Tue, 17 Jan 2006 16:57:16 GMT..Allow: INVITE, 
ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
  .Content-Type: application/sdp..Content-Length: 463....v=0..o=root 
24590 24590 IN IP4 10.2.11.35..s=session..c=IN IP4 1
  0.2.11.35..t=0 0..m=audio 16578 RTP/AVP 4 0 8 111 18 3 97 7 110 5 
101..a=rtpmap:4 G723/8000..a=rtpmap:0 PCMU/8000..a=rt
  pmap:8 PCMA/8000..a=rtpmap:111 G726-32/8000..a=rtpmap:18 
G729/8000..a=fmtp:18 annexb=no..a=rtpmap:3 GSM/8000..a=rtpmap:
  97 iLBC/8000..a=rtpmap:7 LPC/8000..a=rtpmap:110 speex/8000..a=rtpmap:5 
DVI4/8000..a=rtpmap:101 telephone-event/8000..a=
  fmtp:101 0-16..a=silenceSupp:off - - - -..



This message is wrong. Which Asterisk works bad, 1.0.9 or 1.2.1?