[Asterisk-Users] Sipura 3000 inbound FXO problem

[Michael Graves]

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
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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
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