Shoval Tomer
2004-Dec-14 03:30 UTC
[Asterisk-Users] Asterisk to sip client behind Firewall/NAT - cancall but cannot receive calls ?
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@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 cannotset> 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 missingsomething.> I've setup Asterisk to work behing nat and it works OK, on same routeand> same local network Iax phone is operating also ok in both ways. > > On grandstream I've setup public NAT adress, then keep alive to 10sec,> (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 whatports 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@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.
Robert Rozman
2004-Dec-14 04:53 UTC
[Asterisk-Users] Asterisk to sip client behind Firewall/NAT -cancall but cannot receive calls ?
Hi, I hope I won't bother too much if I ask you to provide some more info about your setup, particularly which ports are open and other things (like how often does Grandstream register, do you use keep alive, etc...). Having that information I could rule out settings and maybe start searching on Firewalls... Thanks in advance, regards, Robert. ----- Original Message ----- From: "Shoval Tomer" <shoval@softov.co.il> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Tuesday, December 14, 2004 11:30 AM Subject: RE: [Asterisk-Users] Asterisk to sip client behind Firewall/NAT -cancall but cannot receive calls ? 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@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 cannotset> 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 missingsomething.> I've setup Asterisk to work behing nat and it works OK, on same routeand> same local network Iax phone is operating also ok in both ways. > > On grandstream I've setup public NAT adress, then keep alive to 10sec,> (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 whatports 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@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._______________________________________________ Asterisk-Users mailing list Asterisk-Users@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
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