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david.castro

2006-Jan-17 02:10 UTC

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

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@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@1021150;user=phone@10.2.11.35:5062>;tag=as56432543..To:
<sip:204@10.2.11.35>..Contact: <sip:Zyxel<sip:204@1
 021150;user=phone@10.2.11.35:5062>..Call-ID: 
7bf16eeb00aca5cd2bd303a93f59bfc4@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@1021150;user=phone@10.2.11.35:5062>;tag=as56432543..To: 
<sip:204@10.2.11.35>..Call-ID: 7bf16eeb00aca5cd2bd303a93f59bf
 c4@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@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@1021150;user=phone@10.2.11.35:5062>;tag=as56432543..To: 
<sip:204@10.2.11.35>;tag=329cfeaa6ded039da25ff8cbb8668bd2.
 53ff..Call-ID: 7bf16eeb00aca5cd2bd303a93f59bfc4@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@10.2.11.35 out_uri=sip:204@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@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@1021150;user=phone@10.2.11.35:5062>;tag=as56432543..To: 
<sip:204@10.2.11.35>;tag=329cfeaa6ded039da25ff8cbb8668
 bd2.53ff..Contact: 
<sip:Zyxel<sip:204@1021150;user=phone@10.2.11.35:5062>..Call-ID: 
7bf16eeb00aca5cd2bd303a93f59bfc4@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@1021150;user=phone@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.

Moises Silva

2006-Jan-17 08:05 UTC

head link

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

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@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@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@1021150;user=phone@10.2.11.35:5062>;tag=as56432543..To:
> <sip:204@10.2.11.35>..Contact: <sip:Zyxel<sip:204@1
>  021150;user=phone@10.2.11.35:5062>..Call-ID:
> 7bf16eeb00aca5cd2bd303a93f59bfc4@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@1021150;user=phone@10.2.11.35:5062>;tag=as56432543..To:
> <sip:204@10.2.11.35>..Call-ID: 7bf16eeb00aca5cd2bd303a93f59bf
>  c4@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@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@1021150;user=phone@10.2.11.35:5062>;tag=as56432543..To:
> <sip:204@10.2.11.35>;tag=329cfeaa6ded039da25ff8cbb8668bd2.
>  53ff..Call-ID: 7bf16eeb00aca5cd2bd303a93f59bfc4@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@10.2.11.35 out_uri=sip:204@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@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@1021150;user=phone@10.2.11.35:5062>;tag=as56432543..To:
> <sip:204@10.2.11.35>;tag=329cfeaa6ded039da25ff8cbb8668
>  bd2.53ff..Contact:
> <sip:Zyxel<sip:204@1021150;user=phone@10.2.11.35:5062>..Call-ID:
> 7bf16eeb00aca5cd2bd303a93f59bfc4@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@1021150;user=phone@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.
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

--
"Su nombre es GNU/Linux, no solamente Linux, mas info en
http://www.gnu.org"

david.castro

2006-Jan-17 10:01 UTC

head link

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

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@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@10.2.11.50>..Contact: 
<sip:206@10.2.11.50:5061>..Call-ID: 2112bb831f0c32e
  b72118bc703d791bf@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@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@1021150@10.2.11.35:5062>;tag=as2652cbb1..To: 
<sip:206@10.2.11.35>..Contact:
<sip:"David"<sip:205@1021150@
  10.2.11.35:5062>..Call-ID: 
67fbab514449a6b84b2909d404f7ed6b@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?

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