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Qian Lv

2003-Nov-18 07:45 UTC

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

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@155.69.149.13 SIP/2.0
Call-ID: 4700782232023040960@155.69.149.113
Content-Length: 125
Content-Type: application/sdp
To: sip:Raytec@155.69.149.13
From: sip:zhou@155.69.149.113;tag=74763707
Contact: sip:zhou@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@155.69.149.113;tag=74763707
To: sip:Raytec@155.69.149.13;tag=as1d9111e1
Call-ID: 4700782232023040960@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@155.69.149.13 SIP/2.0
From: sip:zhou@155.69.149.113;tag=74763707
To: sip:Raytec@155.69.149.13;tag=as1d9111e1
Call-ID: 4700782232023040960@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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Qian Lv

2003-Nov-18 07:45 UTC

head link

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

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@155.69.149.13 SIP/2.0
Call-ID: 4700782232023040960@155.69.149.113
Content-Length: 125
Content-Type: application/sdp
To: sip:Raytec@155.69.149.13
From: sip:zhou@155.69.149.113;tag=74763707
Contact: sip:zhou@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@155.69.149.113;tag=74763707
To: sip:Raytec@155.69.149.13;tag=as1d9111e1
Call-ID: 4700782232023040960@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@155.69.149.13 SIP/2.0
From: sip:zhou@155.69.149.113;tag=74763707
To: sip:Raytec@155.69.149.13;tag=as1d9111e1
Call-ID: 4700782232023040960@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
================================
---------------------------------
Do you Yahoo!?
Protect your identity with Yahoo! Mail AddressGuard
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Qian Lv

2003-Nov-18 09:12 UTC

head link

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

Hi, I want to correct an error, in my figure, the softphone2's name is
Raytec, not Reltec. As the figure below shows:

Thanks!

Softphone1<-------->Asterisk SIP<------------>Softphone2
(User Agent)                (Proxy)                     (User Agent)
155.69.xx.xx                 155.69.yy.yy              155.69.zz.zz 
  zhou                          mysipproxy.com            Raytec

Thanks!




Regards,

======Lv Qian,
Ph.D Student,
School of Computer Engineering,
Nanyang Technological University,
Singapore 639798
================================
---------------------------------
Do you Yahoo!?
Protect your identity with Yahoo! Mail AddressGuard
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