[Asterisk-Users] sip to sip calls thru asterisk

[Gary Carr]

That was coming from the register statements in the sip.conf file. Once I
removed those and restarted the sip clients everything started to work.




Thanks!



Gary

> 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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