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

[Moises Silva]

its funny, please tell us where we can see your sip.conf and the
relevant extensions.conf to see how are you registering and dialing.

Regards

On 1/17/06, david.castro <david.castro at adianta.net> wrote:
> 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.
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