[Asterisk-Users] Redirect SIP calls to the SIP provider sipgate.de

[Johannes van Hulst]

I have an asterisk server and I am trying to set the server up as a redirect
server of all my internet SIP phones.

My Asterisk server as his own internet IP address.

 

At this moment I can make international calls to a IAX provider but I am now
trying to setup a SIP provider as well

And I get the following error

 

 

    -- Executing Dial("SIP/t10002-4666", "SIP/0031201234567 at sipprov|120") in
new stack

    -- Called 0031201234567 at sipprov

Aug 30 15:11:23 NOTICE[159484848]: chan_sip.c:6643 handle_response: Failed
to authenticate on INVITE to '"Han Xten"
<sip:t10001 at 200.179.001.11>;tag=as6a15a27f'

  == Spawn extension (homephone, 0031201234567, 1) exited non-zero on
'SIP/t10002-4666'

Aug 30 15:11:32 WARNING[159484848]: chan_sip.c:675 retrans_pkt: Maximum
retries exceeded on call 537024d80840b3144f54435220e25db8 at 200.179.001.11 for
seqno 104 (Non-critical Request)

 

How can I setup an SIP account, so that a sip phone can connect to the SIP
provider (sipgate.de)

Without traveling trough my asterisk server.

 

For example I have:

 

Asterisk server ip 200.179.001.11

IP phone 200.188.001.12

And a provider on sipgate.de.

 

Now I would like to use the asterisk server as a registration off call
server without handling the SIP packages.

 

 

Is this possible?

 

 

Greetings Han

 

 

 

 

 

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