[Asterisk-Users] Asterisk to sip client behind Firewall/NAT - cancall but cannot receive calls ?

Shoval Tomer shoval at softov.co.il
Tue Dec 14 03:30:53 MST 2004


Check your FW-1 tracker and see if any sip packets are dropped during
call initiation.
I had this problem and it went away when I upgraded the BT's firmware to
the latest (16).

Beware, though, that people on the list claim that this firmware breaks
functionality of the message button and autoanswer.
I haven't checked this yet, cause I can't afford to go back a version.
I prefer a phone that can call then a phone that can autoanswer...



> -----Original Message-----
> From: Robert Rozman [mailto:rozman at fri.uni-lj.si]
> Sent: Tuesday, December 14, 2004 11:24 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Asterisk to sip client behind Firewall/NAT -
> cancall but cannot receive calls ?
> 
> Hi,
> 
> I have following setup:
> 
> BT100 ---- Firewall/nat 1 (www.ipcop.org) ---- Internet
----Firewall/nat2
> (Vigor) ---- Asterisk .
> 
> I'd like to use BT100 as local extension to Asterisk. I've done simple
> setup
> and BT100 can call Asterisk and place outgoing calls. However I cannot
set
> him to qualify, cause it is claimed as unreachable.
> 
> I have port redirection at Firewall 1 (to 5060 and rtp 5004 to
> grandstream)
> and 5060 and rtp ports on Firewall2. But I guess I am missing
something.
> I've setup Asterisk to work behing nat and it works OK, on same route
and
> same local network Iax phone is operating also ok in both ways.
> 
> On grandstream I've setup public NAT adress, then keep alive to 10
sec,
> (tried also some other setups but didn't work).
> 
> I'm so close to working state, so would kindly ask for any guidance
(to
> save
> my hair) :-)
> 
> Also I'm missing some understanding of SIP in this story: on what
ports on
> Asterisk machine does Grandstream connect RTP ?  Do I have to transfer
> some
> other ports on Firewall2 because of Grandstream RTP connection ? If
> everything works on outgoing calls there must be some little condition
> that
> gets changed when I try to reach grandstream - what could that be ?
> 
> If anyone has working scenarion, please be so kind to help.
> 
> 
> Regards,
> 
> Rob.
> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> --
> This message has been scanned for viruses and
> dangerous content by MailScanner, and is
> believed to be clean.
> MailScanner thanks transtec Computers for their support.





More information about the asterisk-users mailing list