[asterisk-bugs] Sip request line and To header does not contain the correct called party number

[毛艳焕]

Hi:
I wonder if this is the correct place to post a question on asterisk sip.
The topology of my setup is as follows:
Cisco 7960-----sip----Asterisk-----sip---------SipProxy.

sip.conf
[24057064]
type=friend
host=dynamic                    ; This device needs to register
;nat=yes                        ; X-Lite is behind a NAT router
canreinvite=no                  ; Typically set to NO if behind NAT
context=employee
;disallow=all
;allow=gsm                      ; GSM consumes far less bandwidth than ulaw
allow=ulaw
;
[sh-frodo]
type=peer
host=10.0.20.220
context=incoming_calls
dtmfmode=rfc2833
deny=0.0.0.0/0
permit=10.0.20.220/32
allow=ulaw
canreinvite=yes


extension.conf
[incoming_calls]
exten=>24057063,1,Dial(SIP/24057063)
exten=>24057064,1,Dial(SIP/24057064)

[outbound_local]
exten=>_800XXXXXXX,1,Dial(SIP/sh-frodo)/${EXTEN:1}
exten=>_800XXXXXXX,n,Congestion()
exten=>_800XXXXXXX,n,Hangup()
exten=>_386XXXXXXXX,1,Dial(SIP/sh-frodo)/${EXTEN:0}
exten=>_386XXXXXXXX,n,Congestion()
exten=>_386XXXXXXXX,n,Hangup()

[employee]
include=>outbound_local

exten=>24057063,1,Dial(SIP/24057063)
exten=>24057064,1,Dial(SIP/24057064)

The Invite message from cisco ip phone to asterisk is as follows:
INVITE sip:38624057063 at 10.0.49.44 SIP/2.0
Via: SIP/2.0/UDP 10.74.48.41:5042
;branch=z9hG4bK-d87543-4058735b83308b55-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:24057064 at 10.74.48.41:5042>
To: "38624057063"<sip:38624057063 at 10.0.49.44>
From: "24057064"<sip:24057064 at 10.0.49.44>;tag=0672ee72
Call-ID: M2JmMjIyMjdiZjI1Nzk1NTE0ZGVjODZhMTMzNTE3MzQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 519

v=0
o=- 2 2 IN IP4 10.74.48.41
s=CounterPath X-Lite 3.0
c=IN IP4 10.74.48.41
t=0 0
m=audio 24258 RTP/AVP 107 119 100 106 0 105 98 8 101
a=alt:1 3 : wVyQmDc6 Kv9yxoxC 10.74.48.41 24258
a=alt:2 2 : lrTrQrZy sn3tVDrd 192.168.211.1 24258
a=alt:3 1 : KIqTV8Js uF7GKKXW 192.168.249.1 24258
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv


The invite message send out by Asterisk to Proxy is :
INVITE sip:10.0.20.220 SIP/2.0
Via: SIP/2.0/UDP 10.0.49.44:5060;branch=z9hG4bK2bf1f09f;rport
From: "24057064" <sip:24057064 at 10.0.49.44>;tag=as65ba325a
To: <sip:10.0.20.220>
Contact: <sip:24057064 at 10.0.49.44>
Call-ID: 7590ebc779bf26692460b6ef141f2628 at 10.0.49.44
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 08 Jan 2008 02:58:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 30117 30117 IN IP4 10.0.49.44
s=session
c=IN IP4 10.0.49.44
t=0 0
m=audio 14188 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

As you might have noticed the Request Line is incorrect: INVITE sip:
10.0.20.220 SIP/2.0.

I have searched the web and found one related bug
http://bugs.digium.com/view.php?id=6409, However , That does not helps.

Thank you if someone could give any indication on what is wrong with my
setup.

Henry
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