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
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@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..
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@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.. > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
At 12:33 PM 9/29/2003, you wrote:>Can you clarify any / find me on IRC? (irc.freenode.net/#asterisk/kram)Just FYI: I had similar problems for a while, and then I completely scrapped my CVS directory and did a CVS CHECKOUT (instead of an update). That solved the problem. --Ernest>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@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.. > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users