[Asterisk-Users] sip to sip calls thru asterisk

[Karl Brose]

It's not clear how you are making the call.

You should be able to call directly from either phone to the other by 
dialing 5011 or 5012, respectively, if
your context "local"  indeed contains the those extensions, which is not 
clear from your configuration excerpts.

But it seems you are calling another user on carolina.net, who is 
registered with that provider from your asterisk,
so the call will loop back to Asterisk.   SIP does not really have a 
good way to handle such loopbacks, and
therefore you get the error.
If you want to make this work you need to load a second SIP channel 
driver on your asterisk listening on a different port,
The changes are not difficult.

It also seems that both phones are sitting on the same IP address and 
port,  how can that be?
Oh, I see the error message is actually coming from the sip phone, and 
it's because those phones
have the same IP address, and therefore a loop is detected there.  Is 
this just ONE phone with two
proxy-accounts or personalities?



Gary Carr wrote:

> I have a test box setup and I can make outbound calls on the PSTN thru 
> the diguim card, however I can not make a sip user to sip user call by 
> dialing the extensions. I am getting the following error.
>  
> -- Called cisco7960
>     -- Got SIP response 482 "Loop Detected" back from 208.218.14.123
>   == No one is available to answer at this time
>  
>  
>  
> CLI> sip show peers
> Name/username    Host            Dyn Nat ACL Mask             Port     
> Status
>  
> cisco7960/5052   208.218.14.123   D   N      255.255.255.255  5060     
> OK (1 ms)
> garycarr/5011    208.218.14.123   D   N      255.255.255.255  5060     
> OK (1 ms)
>  
>  
> sip.conf statements
>  
> register => garycarr at sip.carolina.net/5011 
> <mailto:garycarr at sip.carolina.net/5011>
> register => cisco7960 at sip.carolina.net/5052 
> <mailto:cisco7960 at sip.carolina.net/5052>
>  
> [cisco7960]
> type=friend
> host=dynamic
> nat=yes
> qualify=200
> dtmfmode=rfc2833
> canreinvite=no
> mailbox=5052
> callerid="Cisco 7960"
> context=local
>  
> [garycarr]
> type=friend
> host=dynamic
> nat=yes
> qualify=200
> dtmfmode=rfc2833
> canreinvite=no
> mailbox=5011
> callerid="Gary Carr"
> context=local
>  
> extensions.conf statements
>  
> exten => 5011,1,dial(SIP/garycarr,20,tr)
> exten => 5052,1,dial(SIP/cisco7960,20,tr)
>  
> Is this a possible nat issue? I can make a good call from behind the 
> firewall doing sip to pstn so it seems 2 way traffic thru the firewall 
> is working.
>  
>  
> I am still sifting thru the sip debug info but anyone has any ideas 
> that would be great.
>  
>  
> Gary
>  
>
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