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Carlos Medina

2006-Oct-18 19:36 UTC

[asterisk-users] Asterisk + Huawei

Hi everyone,
  
  Im having some troubles getting work Asterisk as SIP Client and a  Huawei
softswitch as SIP server. I already got my asterisk registered  to the Huawei.
Im working with a Sipura SPA 2000 registered to  Asterisk.
  
  When im trying to make an incoming call from the Huawei to asterisk it  rings
but when i answered the call drp down inmediatly. The sip debug  finally show
this message:
  
  Reason: Q.850;cause=100;text="Invalid information element contents"
  
  And when im trying to make an outgoing call i get the following:
  SIP/2.0 503 Service Unavailable
  Reason: Q.850;cause=100;text="Message not compatible with call state or
message type non-existent or not implemented"
  
  Everything is on the same network so i dont have any nat issues.
  
  Thanks a lot for your help.
  
  Carlos Andres Medina
  
 		
---------------------------------
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Ma Zhiyong

2006-Oct-18 19:54 UTC

head link

SPAM Re: [asterisk-users] Asterisk + Huawei

Software zur Erkennung von "Spam" auf dem Rechner

    priamus.teamware-gmbh.de

hat die eingegangene E-mail als m?gliche "Spam"-Nachricht
identifiziert.
Die urspr?ngliche Nachricht wurde an diesen Bericht angeh?ngt, so dass
Sie sie anschauen k?nnen (falls es doch eine legitime E-Mail ist) oder
?hnliche unerw?nschte Nachrichten in Zukunft markieren k?nnen.
Bei Fragen zu diesem Vorgang wenden Sie sich bitte an

    postmaster@teamware-gmbh.de

Vorschau: Can you give more debug information. Usually codec
  incorrupt can cause failure. Can you give more debug information.
  Usually codec incorrupt can cause failure. --Bandwidth and Colocation
  provided by Easynews.com -- [...] 

Inhaltsanalyse im Detail:   (3.7 Punkte, 3.5 ben?tigt)

Pkte Regelname              Beschreibung
---- ---------------------- --------------------------------------------------
 0.1 FORGED_RCVD_HELO       "Received"-Kopfzeilen enthalten gef?lschte
                            HELO-Identifikation
 0.0 DK_POLICY_SIGNSOME     Domain Keys: policy says domain signs some mails
-2.6 BAYES_00               BODY: Spamwahrscheinlichkeit nach Bayes-Test: 0-1%
                            [score: 0.0000]
 0.0 HTML_MESSAGE           BODY: Nachricht enth?lt HTML
 0.2 HTML_FONT_FACE_BAD     BODY: HTML-Schriftart inkorrekt angegeben
 0.2 MIME_BASE64_NO_NAME    RAW: base64-Anhang hat keinen Dateinamen
 1.9 MIME_BASE64_TEXT       RAW: Text getarnt durch base64-Kodierung
 1.6 RCVD_IN_BL_SPAMCOP_NET RBL: Transportiert via Rechner in Liste von
                            www.spamcop.net
              [Blocked - see
<http://www.spamcop.net/bl.shtml?220.181.26.195>]
 1.5 RCVD_IN_SORBS_WEB      RBL: SORBS: Senderechner ist ein ungesicherter
                            WWW-Server
                            [221.221.20.238 listed in dnsbl.sorbs.net]
 0.9 AWL                    AWL: From: address is in the auto white-list

Die urspr?ngliche Nachricht enthielt nicht ausschlie?lich Klartext
(plain text) und kann eventuell eine Gefahr f?r einige E-Mail-Programme
darstellen (falls sie z.B. einen Computervirus enth?lt).
M?chten Sie die Nachricht dennoch ansehen, ist es wahrscheinlich
sicherer, sie zuerst in einer Datei zu speichern und diese Datei danach
mit einem Texteditor zu ?ffnen.

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Subject: Re: [asterisk-users] Asterisk + Huawei
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Carlos Medina

2006-Oct-22 18:59 UTC

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[asterisk-users] Re: Asterisk + Huawei

Thanks for your answer, here is some more debug information, if is a codec
interrupt issue, how can i fix it?

My Sipura uses UID 1234. The huawei softswitch IP address is 10.220.0.2. The
Asterisk IP address  is 10.223.6.98.

The Sipura is registered to the Asterisk box and the Asterisk box is registered
to the Huawei softswitch.

Thanks a lot for your help,

Carlos Andres Medina

------------------- INCOMING ------------------------------------------

    -- Executing Macro("SIP/10.220.0.2-08191e48",
"incoming|SIP/1234") in new stack
    -- Executing Dial("SIP/10.220.0.2-08191e48",
"SIP/1234|30") in new stack
We're at 10.223.6.98 port 19404
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to 10.223.6.99:5150:
INVITE sip:1234@10.223.6.99:5150 SIP/2.0
Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872;rport
From: "Anonymous" <sip:Anonymous@10.223.6.98>;tag=as448023d0
To: <sip:1234@10.223.6.99:5150>
Contact: <sip:Anonymous@10.223.6.98>
Call-ID: 7addcc5162ce1ee57bc58cba40828bbd@10.223.6.98
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 19 Oct 2006 01:56:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 1760 1760 IN IP4 10.223.6.98
s=session
c=IN IP4 10.223.6.98
t=0 0
m=audio 19404 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-- Called 1234

<-- SIP read from 10.223.6.99:5150:
SIP/2.0 100 Trying
To: <sip:1234@10.223.6.99:5150>
From: "Anonymous" <sip:Anonymous@10.223.6.98>;tag=as448023d0
Call-ID: 7addcc5162ce1ee57bc58cba40828bbd@10.223.6.98
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872
Server: Sipura/SPA2000-2.0.10(e)
Content-Length: 0


--- (8 headers 0 lines)---

<-- SIP read from 10.223.6.99:5150:
SIP/2.0 180 Ringing
To: <sip:1234@10.223.6.99:5150>;tag=e2a724add55f408bi0
From: "Anonymous" <sip:Anonymous@10.223.6.98>;tag=as448023d0
Call-ID: 7addcc5162ce1ee57bc58cba40828bbd@10.223.6.98
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872
Server: Sipura/SPA2000-2.0.10(e)
Content-Length: 0


--- (8 headers 0 lines)---
    -- SIP/1234-08197388 is ringing
Transmitting (no NAT) to 10.220.0.2:5061:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.220.0.2:5061;branch=z9hG4bK94161ad88;received=10.220.0.2
From: Anonymous<sip:Anonymous@10.220.0.2>;tag=961d1a68
To: <sip:4875129@10.223.6.98;user=phone>;tag=as40afbad8
Call-ID: 436fedbce988d7eea66f167d06a0558b@10.220.0.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:4875129@10.223.6.98>
Content-Length: 0

<-- SIP read from 10.223.6.99:5150:
SIP/2.0 200 OK
To: <sip:1234@10.223.6.99:5150>;tag=e2a724add55f408bi0
From: "Anonymous" <sip:Anonymous@10.223.6.98>;tag=as448023d0
Call-ID: 7addcc5162ce1ee57bc58cba40828bbd@10.223.6.98
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872
Contact: <sip:1234@10.223.6.99:5150>
Server: Sipura/SPA2000-2.0.10(e)
Content-Length: 229
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 78549 78549 IN IP4 10.223.6.99
s=-
c=IN IP4 10.223.6.99
t=0 0
m=audio 21101 RTP/AVP 8 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (12 headers 12 lines)---
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 10.223.6.99:21101
Found description format PCMA
Found description format NSE
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing),
combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:1234@10.223.6.99:5150>
set_destination: Parsing <sip:1234@10.223.6.99:5150> for address/port to
send to
set_destination: set destination to 10.223.6.99, port 5150
Transmitting (no NAT) to 10.223.6.99:5150:
ACK sip:1234@10.223.6.99:5150 SIP/2.0
Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK403f58ec;rport
From: "Anonymous" <sip:Anonymous@10.223.6.98>;tag=as448023d0
To: <sip:1234@10.223.6.99:5150>;tag=e2a724add55f408bi0
Contact: <sip:Anonymous@10.223.6.98>
Call-ID: 7addcc5162ce1ee57bc58cba40828bbd@10.223.6.98
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/1234-08197388 answered SIP/10.220.0.2-08191e48
We're at 10.223.6.98 port 15322
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.220.0.2:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.220.0.2:5061;branch=z9hG4bK94161ad88;received=10.220.0.2
From: Anonymous<sip:Anonymous@10.220.0.2>;tag=961d1a68
To: <sip:4875129@10.223.6.98;user=phone>;tag=as40afbad8
Call-ID: 436fedbce988d7eea66f167d06a0558b@10.220.0.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:4875129@10.223.6.98>
Content-Type: application/sdp
Content-Length: 233

v=0
o=root 1760 1760 IN IP4 10.223.6.98
s=session
c=IN IP4 10.223.6.98
t=0 0
m=audio 15322 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -
    -- Attempting native bridge of SIP/10.220.0.2-08191e48 and SIP/1234-08197388

<-- SIP read from 10.220.0.2:5061:
ACK sip:4875129@10.223.6.98 SIP/2.0
Via: SIP/2.0/UDP 10.220.0.2:5061;branch=z9hG4bKf4540cd33
Call-ID: 436fedbce988d7eea66f167d06a0558b@10.220.0.2
From: Anonymous<sip:Anonymous@10.220.0.2>;tag=961d1a68
To: <sip:4875129@10.223.6.98;user=phone>;tag=as40afbad8
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0


--- (8 headers 0 lines)---

<-- SIP read from 10.220.0.2:5061:
BYE sip:4875129@10.223.6.98 SIP/2.0
Via: SIP/2.0/UDP 10.220.0.2:5061;branch=z9hG4bK412198644
Call-ID: 436fedbce988d7eea66f167d06a0558b@10.220.0.2
From: Anonymous<sip:Anonymous@10.220.0.2>;tag=961d1a68
To: <sip:4875129@10.223.6.98;user=phone>;tag=as40afbad8
CSeq: 2 BYE
Reason: Q.850;cause=100;text="Invalid information element contents"
Max-Forwards: 70
Content-Length: 0

--- (9 headers 0 lines)---

--------------------- OUTGOING
------------------------------------------------------

<-- SIP read from 10.220.0.2:5060:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK32b32640;rport=5060
Call-ID: 6365afa34dc3ae2318bac62b13945272@10.223.6.98
From: "4875129"<sip:4875129@10.223.6.98>;tag=as1a151f0f
To: <sip:6024042@10.220.0.2>;tag=b1d10bb9
CSeq: 102 INVITE
Reason: Q.850;cause=98;text="Message not compatible with call state or
message type non-existent or not implemented"
Content-Length: 0


--- (8 headers 0 lines)---
-- Got SIP response 503 "Service Unavailable" back from 10.220.0.2
Transmitting (no NAT) to 10.220.0.2:5060:
ACK sip:6024042@10.220.0.2 SIP/2.0
Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK32b32640;rport
From: "4875129" <sip:4875129@10.223.6.98>;tag=as1a151f0f
To: <sip:6024042@10.220.0.2>;tag=b1d10bb9
Contact: <sip:4875129@10.223.6.98>
Call-ID: 6365afa34dc3ae2318bac62b13945272@10.223.6.98
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/epmbogota-08194768 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/1234-0818f228' status is
'CONGESTION'
Transmitting (no NAT) to 10.223.6.99:5150:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.223.6.99:5150;branch=z9hG4bK-8808b7d3;received=10.223.6.99
From: <sip:1234@10.223.6.98>;tag=1fdd8f37d10c2e33o0
To: <sip:6024042@10.223.6.98>;tag=as0d9917db
Call-ID: b4719687-fc4f4f23@10.223.6.99
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6024042@10.223.6.98>
Content-Length: 0
X-Asterisk-HangupCause: Circuit/channel congestion



 		
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Ma Zhiyong

2006-Oct-22 19:30 UTC

head link

[asterisk-users] Re: Asterisk + Huawei

need  debug * and Huawei, not * and client

Carlos Medina

2006-Oct-23 19:10 UTC

head link

[asterisk-users] Re: Asterisk + Huawei

Hi, that was the asterisk debug and the Huawei returns the same errors. The
Huawei debug doesnt say too much, just the answer with the same errors. As you
can see there the INVITE - ACK negotiation between asterisk and Huawei is there
until the call is established. The errors must be in the asterisk side
somewhere. Additionally this is my sip.conf.

[general]
bindport=5060           ; UDP Port to bind to (SIP standard port is 5060)
bindaddr = 0.0.0.0      ; Address to bind to
rtptimeout=120
disallow=all
allow=ulaw
allow=alaw
maxexpirey=500
defaultexpirey=500
rtptimeout=120
;insecure=very
canreinvite=no
context=default                 ; Default context for incoming calls
language=en                     ; Default language setting for all users/peers
dtmfmode=rfc2833
;relaxdtmf=yes                  ; Relax dtmf handling
nat=no                  ; Global NAT settings  (Affects all peers and users)

register=>4875129:cvcol@10.220.0.2/1234

[epmbogota]
type=user
context=default
;username=4875129
;secret=cvcol
deny=0.0.0.0/0.0.0.0
permit=10.220.0.2/255.255.255.255
host=10.220.0.2
nat=no

[epmbogota]
type=peer
;register=yes
username=4875129
secret=cvcol
insecure=invite
host=10.220.0.2
;fromdomain=huawei.com
fromuser=4875129
context=default
;nat=yes
;trustrpid=yes

[1234]
type=friend
username=1234
;secret=1234
;insecure=invite
host=dynamic
context=outgoing
callerid=4875129
;callerid=6055001
qualify=no
dtmfmode=rfc2833
mailbox=4875129
nat=no

Im using G711 codec on both sides.
Thanks a lot for your help.

Ma Zhiyong <mazhiyong@sohu.com> wrote: need  debug * and Huawei, not * and
client

 		
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