[Asterisk-Users] SIP Peers unreachable

[Andrew Gillham]

On Sat, May 03, 2003 at 11:00:16AM -0400, Uriel Carrasquilla wrote:
> I have exactly the same problem using Xten.  I have tried with different
> codecs such as ulaw, 711 and gsm.  My extension does ring and after two
> rings it hangs up.


Try using Dial(SIP/1234 at sipset) if the phone is configured for extension 1234.
Otherwise just try using sipset at sipset.

I have had some troubles with some devices wanting to know what extension
or line the call is for, otherwise they return a busy / not available.


-Andrew

> 
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Chris
> Sent: Friday, May 02, 2003 12:42 PM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] SIP Peers unreachable
> 
> 
> Hi Everyone,
> 
> I'm new to * and I'm trying to setup a small configuration of SIP clients.
> Eventually when I get this working I plan on expanding with a Digium
> developers kit to add analog phones and PSTN access.
> 
> My two end points are an Xten softphone and a Mitel 5055 SIP phone. Both
> peers seem to register with * but I cannot call to one another. When I dial
> the associated extension, the call goes to the programmed voicemail
> extension (busy) yet if I create an extension to call out through the proxy
> (IX66), I can still reach my destination. It's just calling within * there
> is a problem. I suspect it's because the status is unreachable but I'm not
> sure how to fix it.
> 
> Here is the sip show peers output.
> 
> Name/username    Host                 Mask             Port     Status
> sipset/sipset    192.200.14.31    (D)  255.255.255.255  5060     UNREACHABLE
> sippc/sippc      192.200.14.33    (D)  255.255.255.255  5060     UNREACHABLE