[Asterisk-Dev] I Am Missing Something Somewhere Somehow!

[Whisker, Peter]

I think you need to have "host=dynamic" if your phone is registering with
Asterisk. The "host=address" is only for outgoing connections.

You can limit the incoming connections to one phone with
defaultip=address
permit=address/mask

I think that if you specify "host=address" it will not allow a phone to
register.

Peter

-----Original Message-----
From: Adnan Ahmed [mailto:adnan at xnet.com.pk]
Sent: 21 November 2004 22:45
To: asterisk-dev at lists.digium.com
Subject: [Asterisk-Dev] I Am Missing Something Somewhere Somehow!


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=192.168.10.195
disallow=all
allow=alaw
allow=ulaw

[101]
username=101
type=friend
secret=1234
host=192.168.10.195
context=sip
callerid="101"<101>
defaultip=192.168.10.176


extensions.conf
[globals]
[incoming]
exten => s,1,Dial(Zap/1)

[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,Zap/4(SIP/101)

[sip]
exten => 101,1,Dial(SIP/101,20)

here are the console output : show no errors but also not working 
(running Asterisk in quite mode :-X ).
*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.
I am searching wiki,google but no luck i am tried several configs but 
all in vein please please helping me out :-( .

Mike Dent wrote:

> Dont get caught by the same thing which had me ripping my hair out!
> I had installed Fedora core 2 on a box and forgot that it had 
> installed iptables
> firewall!
> Type iptables -L and see if there are any rules? iptables -F will
> flush them for the time
> being, then try again.
> It worked for me, wow how silly I felt!
> Mike
>  
>
I am using Debian it's not working for me any other thaughts,tips  
suggestions because now i am very exhausted with this error i am looking 
almost everyplace wiki google but no luck kindly helping me out.

el Flynn wrote:

  
Adnan Ahmed wrote:

    

> hi,
> I  am not registered my SIP Phone with Asterisk  i spend almost one
> day  but find no luck my configs are.
>
>       

<snip>

    

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

your "sip show peers" command shows that the phone is indeed connected
to your Asterisk server. If you are having problems doing stuff with
it, may I suggest you changing your dialplan to the following just to
test things out:

[sip]
exten => 1,1,VoicemailMain
exten => 1,2,Hangup

then restart asterisk and dial "1" from your SIP phone. If you can
hear the voicemail application prompts then you're okay.

flynn

Thanks In Advance .
Adnan Ahmed.
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