asterisk users - May 2009 - connection fail between Service provider's proxy server and my asterisk server

I wanna connect proxy server.

 

my IP Phone -> my asterisk -> service provider's proxy server ->
extern PSTN phone

 

but asterisk server can't register to proxy server.

 

I think that configuration is right.

 

When asterisk send to register request, proxy server don't response.

 

I did capture packet. but no response.

 

 

MY setting

 

sip.conf

 

[kms]

username=kms

type=friend

secret=1111

host=dynamic

nat=yes

qualify=yes

callerid=0134

 

register => 070XXXX0134:password at proxy.SP.co.kr:5060/0134

 

[my-out]

type=peer

host=SP's proxy IP

username=070XXXX0134

secret=password

fromuser=070XXXX0134

fromdomain=proxy.SP.co.kr

 

extensions.conf

 

[default]

exten => _X.,1,Dial(SIP/${EXTEN}@my-out)

 

 

 

If lines provided is not a form of trunk, can't my asterisk server connect
to proxy?

I could connect my IPPhone to proxy directly.

but asterisk not.

 

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2009/5/29 ??? <kimms at infosec.co.kr>
>  I wanna connect proxy server.
>
>
>
> my IP Phone -> my asterisk -> service provider's proxy server
-> extern
> PSTN phone
>
>
>
> but asterisk server can't register to proxy server.
>
>
>
> I think that configuration is right.
>
>
>
> When asterisk send to register request, proxy server don't response.
>
>
>
> I did capture packet. but no response.
>
>
>
>
>
> MY setting
>
>
>
> sip.conf
>
>
>
> [kms]
>
> username=kms
>
> type=friend
>
> secret=1111
>
> host=dynamic
>
> nat=yes
>
> qualify=yes
>
> callerid=0134
>
>
>
> register => 070XXXX0134:password at proxy.SP.co.kr:5060/0134
>
>
>
> [my-out]
>
> type=peer
>
> host=SP's proxy IP
>
> username=070XXXX0134
>
> secret=password
>
> fromuser=070XXXX0134
>
> fromdomain=proxy.SP.co.kr
>
>
>
> extensions.conf
>
>
>
> [default]
>
> exten => _X.,1,Dial(SIP/${EXTEN}@my-out)
>
>
>
>
>
>
>
> If lines provided is not a form of trunk, can't my asterisk server
connect
> to proxy?
>
> I could connect my IPPhone to proxy directly.
>
> but asterisk not.
>
>
>
We need the sip trace for the call.
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