[Asterisk-Users] SIP Problem!

[E. Versaevel]

If you want to Sip REGISTER your phone to asterisk change the
host=192.168.10.193 section of the [101] section to host=dynamic

Currently you are telling asterisk that sip user 101 is on host
192.168.0.193, which is you asterisk box, so when a call goes to 101,
asterisk sends it to itself and then tries to connect the incoming sip call
to 101, hence the loop :)

Kind regards, 

E. Versaevel


-----Oorspronkelijk bericht-----
Van: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] Namens Adnan Ahmed
Verzonden: maandag 22 november 2004 21:34
Aan: asterisk-users at lists.digium.com
Onderwerp: [Asterisk-Users] SIP Problem!

hi,
I  am not registered my SIP Phone with Asterisk  i spend almost one day  
but find no luck.I know very well this is not  kind a problem discussed 
in this group but i try my best and all in vein so finally i am here 
hoping you ppl helping me out.I discussed this problem in 
asterisk's-users group and adding feedback from asterisk-users group my 
configs are


sip.conf

[general]
port=5060
bindaddr=192.168.10.193
allow=all


[101]
username=101
type=friend
secret=12345678
host=192.168.10.193
context=from-sip
callerid="101"<101>
defaultip=192.168.10.176


extensions.conf
[globals]
101=SIP/101

[incoming]
exten => s,1,Dial(Zap/1,20)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(u${announce})
exten => s-NOANSWER,2,Goto(incoming,s,1)
exten => s,3,NoOp,$(CALLERID)
include => outgoing
include => from-sip
callerid=yes   

[outgoing]
exten => _NXXXXXX,1,Dial/Zap/4/${EXTEN:0}
exten => _0NXXXXXXXX,1,Dial,Zap/4/${EXTEN:0}
exten => _0NXXXXXXXXX,1,Dial,Zap/4/${EXTEN:0}
exten => _0NXXXXXXXXXX,1,Dial,Zap/4/${EXTEN:0}
exten => 101,1,Dial(101,20)
include => from-sip
include =>  incoming

[sip]
exten => 101,1,Dial(${101,20})
exten => 101,2,VoicemailMain
exten => 101,3,Hangup
include => outgoing
include => from-sip

here are the console output : :-X ).

*cli>      --Starting simple switch on 'Zap/1-1'
Executing Dial("                   ","                            ") in 
new stack
Called 101
Got SIP Responce 482 "Loop Detected" back from 192.168.10.193
No one is available to answer qt this time
Executing VoiceMailMain("                  ","") in new stack
Playing     'vm-login'       (language   'en' )
Username not entered
Executing Hangup("                              ","") in new stack
Spawn Extension (outgoing ,  101,  3)   exited non-zero on 'Zap/1-1'
Hangup 'Zap/1-1'


*cli>sip show registry
Host                              Username                              
      Refresh State

*cli>sip show users
Username               Secret               Authen                  
Def.Context                  A/C
101                         12345678        md5,plaintext          
sip                                No

*cli>sip show peers
Name/Username            Host                     Mask                  
               Port                  Status
101/101                        192.168.10.195    255.255.255.255      
         5060                Unmonitored

*cli>sip show channels
Peer                User/ANR            Call ID                Seq 
(Tx/Rx)                 Lag                Jitter                Buffer
0 active SIP  channel(s)


Kindly pointout my mistakes/errors and helping me out.
Any Help Is Highly Appreciated.
Thanks in Advance.

Adnan Ahmed.
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