Hi,
I’m currently fighting with sip configuration in order to make outbound calls via Huawei SoftX3000.
I have tried several configurations without success.
I always get: Got SIP response 503 “Service Unavailable”
I can register and make calls via Xlite and Kphone using Asterisk credentials on Softswitch…
I have done several searchs on the forums\lists and find lots of people with same problems… tried several combinations but no one worked for me. 
Here are my currently sip.conf
[general]
rtcachefriends = yes
maxexpirey = 99999999999999999
videosupport = yes
tos = 0x28
allowguest = yes
defaultexpirey = 1000
register = 210127307@195.23.145.1/210127307
[210127307]
fromuser = 210127307
nat = never
insecure=very
fromdomain = 195.23.145.1
call_limit = 10
host = 195.23.145.1
username = 210127307
canreinvite=no
disallow = all
context = incoming
allow = ulaw
allow = gsm
type = friend
Extensions.conf
exten = _X.,1,Set(CALLERID(name)=210127307)
exten = _X.,2,Dial(SIP/${CALL_NUM}@210127307)
exten = _X.,3,Hangup()
I can register correctly but I can’t make calls…
ebox*CLI> sip show registry
Host Username Refresh State
195.23.145.1:5060 210127307 985 Registered
Here my SIP packets:
NVITE sip:934484746@195.23.145.1 SIP/2.0
Via: SIP/2.0/UDP 195.23.145.59:5060;branch=z9hG4bK113c0bba
From: “210127307” sip:210127307@195.23.145.1;tag=as213fcebd
To: sip:934484746@195.23.145.1
Contact: sip:210127307@195.23.145.59
Call-ID: 502e569c641210430a02c48738d7384e@195.23.145.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 07 Apr 2006 15:33:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 183
v=0
o=root 4671 4671 IN IP4 195.23.145.59
s=session
c=IN IP4 195.23.145.59
t=0 0
m=audio 19078 RTP/AVP 0 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
-- Called 934484746@210127307
ebox*CLI>
<-- SIP read from 195.23.145.1:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 195.23.145.59:5060;branch=z9hG4bK113c0bba
Call-ID: 502e569c641210430a02c48738d7384e@195.23.145.1
From: “210127307” sip:210127307@195.23.145.1;tag=as213fcebd
To: sip:934484746@195.23.145.1
CSeq: 102 INVITE
Content-Length: 0
— (7 headers 0 lines)—
ebox*CLI>
<-- SIP read from 195.23.145.1:5060:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 195.23.145.59:5060;branch=z9hG4bK113c0bba
Call-ID: 502e569c641210430a02c48738d7384e@195.23.145.1
From: “210127307” sip:210127307@195.23.145.1;tag=as213fcebd
To: sip:934484746@195.23.145.1;tag=553e4e9c
CSeq: 102 INVITE
Reason: Q.850;cause=98;text="unknown"
Content-Length: 0
I know that softswitch are not on the same network of Asterisk, but I don’t have NAT between then.
I also tried NAT configs (just for confirmation) and got the same result…
oh… here are my invite made with kphone which work fine…
INVITE sip:934484746@195.23.145.1 SIP/2.0
Via: SIP/2.0/UDP 195.23.145.59;branch=z9hG4bK4CA160C9
CSeq: 2862 INVITE
To: sip:934484746@195.23.145.1
Content-Type: application/sdp
From: “210127307” sip:210127307@195.23.145.1;tag=FF5358E
Call-ID: 1462737622@195.23.145.59
Subject: sip:210127307@195.23.145.1
Content-Length: 230
User-Agent: kphone/4.2
Contact: “210127307” sip:210127307@195.23.145.59;transport=udp
v=0
o=username 0 0 IN IP4 195.23.145.59
s=The Funky Flow
c=IN IP4 195.23.145.59
t=0 0
m=audio 32818 RTP/AVP 0 97 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
