asterisk users - Jul 2005 - Sip.conf problems

Hi,

I have been trying to configure my Asterisk to use a Sip provider for 
out and incoming calls.
I only have one user and password for connect to my sip provider.

My sip.conf is:

[general]
;disallow=gsm
;allow=ulaw
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
context = default               ; Default for incoming calls
callerid=No CallID
register => user:password@sipprovider.com

[sip_proxy]
type=friend
username=user
fromuser=user
secret=password
host=siprovider
dtmfmode=inband

The problem is:
If i put in the [sip_proxy] section type=friend, incoming calls doesn't 
works. If the type is set to another value (for example peer) incoming 
calls works fine, but outgoing calls doesn't works.

What can I do?

Thanks
David

Does the registration show up?
 try "sip show registry" at the CLI
 also try "sip debug peer sip_proxy" and post the result.
 Might be able to see what's going on there...
 mark


 On 7/1/05, David <asterisk@barnatech.com> wrote: 
> 
> Hi,
> 
> I have been trying to configure my Asterisk to use a Sip provider for
> out and incoming calls.
> I only have one user and password for connect to my sip provider.
> 
> My sip.conf is:
> 
> [general]
> ;disallow=gsm
> ;allow=ulaw
> port = 5060 ; Port to bind to
> bindaddr = 0.0.0.0 <http://0.0.0.0> ; Address to bind to
> context = default ; Default for incoming calls
> callerid=No CallID
> register => user:password@sipprovider.com
> 
> [sip_proxy]
> type=friend
> username=user
> fromuser=user
> secret=password
> host=siprovider
> dtmfmode=inband
> 
> The problem is:
> If i put in the [sip_proxy] section type=friend, incoming calls doesn't
> works. If the type is set to another value (for example peer) incoming
> calls works fine, but outgoing calls doesn't works.
> 
> What can I do?
> 
> Thanks
> David
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 


-- 
regards,

Mark P. Edwards
TEL:+61 408 601 107
SKYPE: mark.p.edwards
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20050701/805baa87/attachment.htm

Try two different entries:

sip.conf:

    [general]
    ;disallow=gsm
    ;allow=ulaw
    port = 5060                     ; Port to bind to
    bindaddr = 0.0.0.0              ; Address to bind to
    context = default               ; Default for incoming calls
    callerid=No CallID
    register => user:password@sipprovider.com/2025551212

    [2025551212]
    type=peer
    realm=sipprovider.com
    fromdomain=sipprovider.com
    username=user
    fromuser=user
    secret=password
    host=sipprovider.com
    dtmfmode=inband

    [sip_provider]
    type=peer
    context=sip_provider-inbound
    host=sipprovider.com


extensions.conf:

    [sip_provider-inbound]

       exten => 2025551212,n,Goto(default,s,1)

       exten => i,1,Goto(default,s,1)

       exten => t,1,Goto(default,s,1)

       exten => h,1,hangup


David wrote:
> Hi,
>
> I have been trying to configure my Asterisk to use a Sip provider for 
> out and incoming calls.
> I only have one user and password for connect to my sip provider.
>
> My sip.conf is:
>
> [general]
> ;disallow=gsm
> ;allow=ulaw
> port = 5060                     ; Port to bind to
> bindaddr = 0.0.0.0              ; Address to bind to
> context = default               ; Default for incoming calls
> callerid=No CallID
> register => user:password@sipprovider.com
>
> [sip_proxy]
> type=friend
> username=user
> fromuser=user
> secret=password
> host=siprovider
> dtmfmode=inband
>
> The problem is:
> If i put in the [sip_proxy] section type=friend, incoming calls 
> doesn't works. If the type is set to another value (for example peer) 
> incoming calls works fine, but outgoing calls doesn't works.
>
> What can I do?
>
> Thanks
> David
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users