[Asterisk-Users] Is somthing broken?

[Mark Spencer]

Can you clarify any / find me on IRC? (irc.freenode.net/#asterisk/kram)

Mark

On Mon, 29 Sep 2003, WipeOut wrote:

> WipeOut wrote:
>
> > Hi,
> >
> > I updated my live server yesterday(after testing on my Dev server
> > first, all works on the Dev server)..
> >
> > Here is the setup..
> >
> > SIP_UA---[NAT]---Asterisk1---PSTN(chan_capi)
> >
> > The SIP_UA is able to recieve calls from the server with no problems..
> > Initiated from the PSTN or my Dev Asterisk box which is connected to
> > Asterisk1 with IAX..
> >
> > When the SIP_UA tries to make calls out via the PSTN or to Voicemail
> > on Asterisk1 or another extention there is no sound..
> >
> > The definition in sip.conf is fairly standard(included below)..
> >
> > This config has been working fine for months.. the last update was
> > about 1 month ago so sometime between then and now it seems that SIP
> > has changed and so stopped working..
> >
> > Hopefully this can be solved quickly becasue it is a problem..
> >
> > Later..
> >
> >
> > Definition from sip.conf
> > [2014]
> > context=users
> > type=friend
> > secret=magic
> > nat=yes
> > canreinvite=no
> > dtmfmode=info           ; Grandstream
> > host=dynamic
> > mailbox=2014            ; Mailbox for message waiting indicator
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> Looks like there is a problem with SIP, I rolled back to Thurday last
> weeks CVS and the SIP UA behind NAT is now working..
>
> I will be interested to see if anyone else has this problem when
> updating to the current CVS..
>
> Later..
>
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