[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:[EMAIL PROTECTED] SIP/2.0
Call-ID: [EMAIL PROTECTED]
Content-Length: 125
Content-Type: application/sdp
To: sip:[EMAIL PROTECTED]
From: sip:[EMAIL PROTECTED];tag=74763707
Contact: sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED];tag=74763707
To: sip:[EMAIL PROTECTED];tag=as1d9111e1
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED] SIP/2.0
From: sip:[EMAIL PROTECTED];tag=74763707
To: sip:[EMAIL PROTECTED];tag=as1d9111e1
Call-ID: [EMAIL PROTECTED]
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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