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.
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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] Namens Adnan Ahmed
Verzonden: maandag 22 november 2004 21:34
Aan: asterisk-users@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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