[Asterisk-Users] Sipura 3000 inbound FXO problem

[Steven P. Donegan]

Michael Graves wrote:

>On Thu, 30 Dec 2004 09:04:32 -0800, Steven P. Donegan wrote:
>
>  
>
>>Kristian Kielhofner wrote:
>>
>>    
>>
>>>Steven P. Donegan wrote:
>>>
>>>      
>>>
>>>>I have a Sipura 3000, apparently configured correctly, when incoming 
>>>>calls arrive on the telco port they arrive properly on the Asterisk 
>>>>system - however they don't get routed properly. The Asterisk message:
>>>>
>>>>Dec 30 07:47:16 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed 
>>>>to authenticate user WIRELESS CALLER 
>>>><sip:714XXXXXXX at 1.0.24.5>;tag=7f8072c0c46250f7o1
>>>>
>>>>X's are there to not advertise my phone number :-)
>>>>
>>>>Any idea as to why any kind of authenticate would be done or would 
>>>>fail would be appreciated.
>>>>        
>>>>
>>>Steven,
>>>
>>>    It really seems like you need to setup an entry in sip.conf that 
>>>"PSTN Line" on the sipura can register with.  Do you have an entry in 
>>>sip.conf for it?  How is "PSTN Line" programmed?
>>>
>>>-- 
>>>Kristian Kielhofner
>>>_______________________________________________
>>>Asterisk-Users mailing list
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>>>
>>>
>>>      
>>>
>>Here is sip show peers:
>>
>>www*CLI> sip show peers
>>Name/username    Host            Dyn Nat ACL Mask             Port     
>>Status  
>>1004/1004        1.0.24.223       D          255.255.255.255  5060     
>>Unmonitored
>>1003/1003        1.0.24.223       D          255.255.255.255  5060     
>>Unmonitored
>>1002/1002        1.0.24.222       D          255.255.255.255  5061     
>>Unmonitored
>>1001/1001        1.0.24.222       D          255.255.255.255  5060     
>>Unmonitored
>>1000/1000        (Unspecified)    D          255.255.255.255  0        
>>Unmonitored
>>5 sip peers loaded [4 online , 1 offline]
>>
>>Which seems to say the Sipura is registered...
>>
>>Here is sip.conf:
>>
>>[root at www asterisk]# cat sip.conf
>>[general]
>>port = 5060                     ; Port to bind to
>>bindaddr = 0.0.0.0              ; Address to bind to
>>context = default               ; Default for incoming calls
>>
>>[1000]
>>type=friend
>>username=1000
>>fromuser=1000
>>host=dynamic
>>nat=no
>>canreinvite=yes
>>dtmfmode=rfc2833
>>mailbox=1000 at default
>>disallow=all
>>allow=ulaw
>>
>>[1001]
>>type=friend
>>username=1001
>>fromuser=1001
>>host=dynamic
>>nat=no
>>canreinvite=yes
>>dtmfmode=rfc2833
>>mailbox=1001 at default
>>disallow=all
>>allow=ulaw
>>
>>[1002]
>>type=friend
>>username=1002
>>fromuser=1002
>>host=dynamic
>>nat=no
>>canreinvite=yes
>>dtmfmode=rfc2833
>>mailbox=1002 at default
>>disallow=all
>>allow=ulaw
>>
>>[1003]
>>type=friend
>>username=1003
>>secret=1003
>>canreinvite=no
>>host=dynamic
>>dtmfmode=rfc2833
>>mailbox=1003
>>nat=no
>>disallow=all
>>allow=ulaw
>>
>>[1004]
>>type=friend
>>username=1004
>>secret=1004
>>canreinvite=no
>>host=dynamic
>>dtmfmode=rfc2833
>>mailbox=1004
>>nat=no
>>disallow=all
>>allow=ulaw
>>
>>[root at www asterisk]#
>>
>>Not sure what I'm doing wrong but any suggestions would be welcomed.
>>
>>And BTW - Happy Hollidays!
>>    
>>
>
>When I used the SPA-3000 I had to setup a special context in
>extensions.conf and then use a "hotline" dialplan setup in the SPA.
>This caused all calls incomming on the POTS line to immediately be
>forwarded to the Asterisk context. I essentially bypassed the SPA
>diaplan logic. You can find out more about this at www.voxilla.com
>which hosts a forum for SPA users.
>
>Michael
>
>--
>Michael Graves                           mgraves at pixelpower.com
>Sr. Product Specialist                          www.pixelpower.com
>Pixel Power Inc.                                 mgraves at mstvp.com
>
>o713-861-4005
>o800-905-6412
>c713-201-1262
>
>
>
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>
>
>  
>
Sorry for all the included text - but it is relevant. The problem is not 
the Sipura->Asterisk connection - that is definitely happening - the 
problem is that Asterisk seems to want to authenticate the call in some 
way.  And I have no clue at present as to how to make Asterisk happy 
with the inbound call.