[asterisk-users] asterisk sip problem

[manouchk at gmail.com]

They are on the same lan

the adapter is registered

sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
sippyskypeuser/sippyskype  192.168.2.76                5070     OK (1 ms)
1000/1000                  192.168.2.76     D          5061     OK (1 ms)
freephonie-out/0950607456  212.27.52.5          N      5060     OK (766 ms)
callcentric/17772962667    204.11.192.34        N      5080     OK (206 ms)

the pap2t's IP is 192.168.2.205
and the IP of the asterisk box is 192.168.2.76

sip show registry
Host                            Username       Refresh State
     Reg.Time
freephonie.net:5060             095060xxxx        1785 Registered
     Wed, 09 Jul 2008 10:12:44
callcentric.com:5080            177729xxxxx         46 Registered
     Wed, 09 Jul 2008 10:13:29

I use line2 of my pap2t (line 1 is not enabled). Here is the conf :
http://emmanuelfavrenicolin.free.fr/Public/Divers/Snapshots1/20080709_pap2t.jpg


On 7/9/08, MFH <asterisk-admin at hulber.com> wrote:
> Are asterisk and the phone on the same lan?  I see you have nat=no.  Do
> you see the phone adapter registered?
>
> Emmanuel Favre-Nicolin wrote:
>> Hi,
>>
>> I'm having a problem to receive inbound call from my sip provider. I used
>> to
>> be OK, I may I have change something (for example I switched from asterisk
>>
>> 1.4.20.1 to 1.4.21.1). Could that be a bug? (I doubt that and I guess it a
>>
>> configuration problem on my side!)
>>
>> I have basically a sip account and a linksys voip adapter with a phone on
>> it
>> (sip name 1000), configured in asterisk. Outbound call from the phone just
>>
>> work fine. Inbound call fail to ring my phone. When the inbound call occur
>> I
>> see on the asterisk command line :
>>
>>     -- Executing [17772962667 at from-callcentric:1]
>> Dial("SIP/callcentric.com-081f1ac8", "SIP/1000") in new stack
>>
>>     -- Called 1000
>>
>>     -- SIP/1000-081ed5e0 is ringing
>>
>> but my phone is not ringing
>>
>> in sip.conf:
>>
>> [1000]
>> type=friend
>> secret=blablabla
>> qualify=yes    ; Qualify peer is not more than 2000 mS away
>> nat=no         ; This phone is not natted
>> host=dynamic   ; This device registers with us
>> canreinvite=no ; Asterisk by default tries to redirect
>> context=fromsoftphone
>> port=5061     ; Uncomment this line if Ekiga and Asterisk are on the same
>> host
>>
>>
>> in extensions.conf:
>> [from-callcentric]
>> exten => 17772962667,1,Dial(SIP/1000)
>> exten => 17772962667,n,Hangup()
>>
>>
>> The default extension I got for inbound call is 17772962667 that's why I
>> used
>> that extension. I tu
>>
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>
> _______________________________________________
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