[asterisk-users] asterisk sip problem

[MFH]

I don't see anything obvious right away other than have you confirmed 
that the phone is actually working?  Can you get it to ring?  With my 
Sipura adapters that use Linksys software I can view the call status in 
the "Info" section which if you have that panel might tell you if the 
adapter thinks a call is coming in.  I just looked at my Info page with 
a call coming in and I can see the call state as Ringing and a bunch of 
other details.

Call Status
Call 1 State: 	Ringing
Call 1 Tone: 	Ring - Hold
Call 1 Encoder: 	G711u
Call 1 Decoder: 	G711u
Call 1 FAX: 	No
Call 1 Type: 	[L1]Inbound
Call 1 Remote Hold: 	No
Call 1 Callback: 	No
Call 1 Peer Name: 	UNAVAILABLE
Call 1 Peer Phone: 	1XXXXXXXXX
Call 1 Duration: 	
Call 1 Packets Sent: 	0
Call 1 Packets Recv: 	0
Call 1 Bytes Sent: 	0
Call 1 Bytes Recv: 	0
Call 1 Decode Latency: 	0 ms
Call 1 Jitter: 	0 ms
Call 1 Round Trip Delay: 	0 ms
Call 1 Packets Lost: 	0
Call 1 Packet Error: 	0
Call 1 Mapped RTP Port: 	16420 >> 0



manouchk at gmail.com wrote:
> 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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