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 > > [EMAIL PROTECTED] > > 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.. > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
