[Asterisk-Users] Is somthing broken?

[WipeOut]

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
>
>
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> Asterisk-Users at lists.digium.com
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>
>
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..