[Asterisk-Users] ask problem about softphone--asterisk--softphone, Urgent!!!

[Qian Lv]

Hi, all,
I want to use asterisk SIP as a proxy, and two softphone (Ubiquity SIP Phone) as user agent, like below:
 
Softphone1<-------------->Asterisk SIP<------------>Softphone2
(User Agent)               (Proxy)                        (User Agent)
155.69.xx.xx             155.69.yy.yy                 155.69.zz.zz            
  zhou                       mysipproxy.com              Reltec
 
If I use softphone1(zhou) to dial softphone2(Reltec) directly, not accroess Asterisk SIP (proxy), it can work. But when I use asterisk SIP as a proxy, then Softphone1(zhou) shows "Not Found", and it seems it can not find softphone2's address.
 
It seems an easy problem, but it waste me about one week's time. The main content in my [sip.conf] file is:
  ...  
   [general]
     port=5060
     bindaddr=0.0.0.0
     context=bogon-calls
     allow=all
    
  [mysipproxy.com]
     type=friend
     host=155.69.yy.yy
     fromuser=lq
 
 [zhou]
     type=friend
     host=dynamic
     defaultip=155.69.xx.xx
     context=from-sip
     fromdomain=mysipproxy.com
 
 [Raytec]
     type=friend
     host=dynamic
     defaultip=155.69.zz.zz
     context=from-sip
     fromdomain=mysipproxy.com

 
The main content in my [extensions.conf] is:
  ...
 [bogon-calls]
    exten=>_.,1,Congestion
 [from-sip]
    exten => 1, 1, Dial(SIP/zhou,20)
    exten => 1, 102, Hangup
    exten => 2, 1, Dial(SIP/Raytec,20)
    exten => 2, 102, Hangup
     
The result in asterisk SIP is listed below:
 
*CLI> sip debug
SIP Debugging Enabled
*CLI> Sip read:
INVITE sip:Raytec at 155.69.149.13 SIP/2.0
Call-ID: 4700782232023040960 at 155.69.149.113
Content-Length: 125
Content-Type: application/sdp
To: sip:Raytec at 155.69.149.13
From: sip:zhou at 155.69.149.113;tag=74763707
Contact: sip:zhou at 155.69.149.113:5060
CSeq: 1 INVITE
Via: SIP/2.0/UDP 155.69.149.113:5060;branch=9B45957113C4000000F8EF064B14-33*0
v=0
o=- 1069165096343 1069165096343 IN IP4 155.69.149.113
s=-
c=IN IP4 155.69.149.113
t=0 0
m=audio 5006 RTP/AVP 3 0 8
9 headers, 6 lines
Using latest request as basis request
Sending to 155.69.149.113 : 5060 (non-NAT)
Found audio format UNKN
Found audio format UNKN
Found audio format ALAW
Capabilities: us - 2147483647, them - 14/0, combined - 14
Non-codec capabilities: us - 1, them - 0, combined - 0
Looking for Raytec in from-sip
Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 155.69.149.113:5060;branch=9B45957113C4000000F8EF064B14-33*0
From: sip:zhou at 155.69.149.113;tag=74763707
To: sip:Raytec at 155.69.149.13;tag=as1d9111e1
Call-ID: 4700782232023040960 at 155.69.149.113
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:@155.69.149.112>
Content-Length: 0

 to 155.69.149.113:5060
Sip read:
ACK sip:Raytec at 155.69.149.13 SIP/2.0
From: sip:zhou at 155.69.149.113;tag=74763707
To: sip:Raytec at 155.69.149.13;tag=as1d9111e1
Call-ID: 4700782232023040960 at 155.69.149.113
CSeq: 1 ACK
Via: SIP/2.0/UDP 155.69.149.113:5060;branch=9B45957113C4000000F8EF064B14-33*0
Content-Length: 0

7 headers, 0 lines
 
 
 
 
 
 
 
 
 
 
 
 
It seems the extensions.conf has some problem, but I don't know how to write the correct dialplan. Any suggestions will be appreciated.

Thanks.
 
 
 



Regards,

=======
Lv Qian,
Ph.D Student,
School of Computer Engineering,
Nanyang Technological University,
Singapore 639798
=================================

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