[asterisk-users] Linksys SPA 3102 causing me problems

[Alan Chandler]

On Friday 30 March 2007 04:02, Matt Putnam wrote:
> I dont know if you have done this but run a sip show peers and make
> sure that its registered with asterisk. Sounds like it is not
> registering with asterisk which would allow you to call out but when
> it tries to call you it dosent have an ip to contact you at.


ship show peers shows it thus

roo*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
chandler/chandler          192.168.0.10     D          5060     OK (6 
ms)
1 sip peers [1 online , 0 offline]

sip show registry shows nothing.

With sip debug on I get the following as I start to make the call

13 headers, 12 lines
Reliably Transmitting (no NAT) to 192.168.0.10:5060:
INVITE sip:chandler at 192.168.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK1ff726a0;rport
From: "Alan" <sip:11 at 192.168.0.20>;tag=as3e6d82c9
To: <sip:chandler at 192.168.0.10:5060>
Contact: <sip:11 at 192.168.0.20>
Call-ID: 027457326a11cad1430ff9ad20d0c34a at 192.168.0.20
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 30 Mar 2007 06:07:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 29004 29004 IN IP4 192.168.0.20
s=session
c=IN IP4 192.168.0.20
t=0 0
m=audio 19854 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
roo*CLI>
<-- SIP read from 192.168.0.10:5060:
SIP/2.0 100 Trying
To: <sip:chandler at 192.168.0.10:5060>
From: "Alan" <sip:11 at 192.168.0.20>;tag=as3e6d82c9
Call-ID: 027457326a11cad1430ff9ad20d0c34a at 192.168.0.20
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK1ff726a0
Server: roo.home
Content-Length: 0


--- (8 headers 0 lines) ---
roo*CLI>
<-- SIP read from 192.168.0.10:5060:
SIP/2.0 486 Busy Here
To: <sip:chandler at 192.168.0.10:5060>;tag=cf4213264eacc5ei0
From: "Alan" <sip:11 at 192.168.0.20>;tag=as3e6d82c9
Call-ID: 027457326a11cad1430ff9ad20d0c34a at 192.168.0.20
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK1ff726a0
Server: roo.home
Content-Length: 0


--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.0.10:5060:
ACK sip:chandler at 192.168.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK1ff726a0;rport
From: "Alan" <sip:11 at 192.168.0.20>;tag=as3e6d82c9
To: <sip:chandler at 192.168.0.10:5060>;tag=cf4213264eacc5ei0
Contact: <sip:11 at 192.168.0.20>
Call-ID: 027457326a11cad1430ff9ad20d0c34a at 192.168.0.20
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
dial status is BUSY
Destroying call '027457326a11cad1430ff9ad20d0c34a at 192.168.0.20'




>
> On 3/29/07, Alan Chandler <alan at chandlerfamily.org.uk> wrote:
> > I have a linksys SPA 3102 with a DECT phone connected into its
> > Telephone port.
> >
> > It has been working, but something I've done (and I don't know
> > what) means that now everytime asterisk tries to dial it, it says
> > it is busy.
> >
> > I can make calls from it through asterisk
> >
> > I am at a complete loss to know what to try next to fix it.  Any
> > ideas?
> >
> >
> > --
> > Alan Chandler
> > http://www.chandlerfamily.org.uk
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Alan Chandler
http://www.chandlerfamily.org.uk