[Asterisk-Users] NAT and sip issues
I have an asterisk server behind NAT - no audio on the test external calls I have tried making so far. Read http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions - No solution evident from there, sounds like I have case 9. I would have thought that all I would have to do is port foward and have the external IP on the asterisk server, which I have done I have fowared 5060UDP, 8000UDP, and 35000 to 37000 UDP to the internal IP (192.168.1.115) I have put 35000 and 37000 into the rtp.conf as the start/end ports extracts of sip.conf: externip = 60.234.129.154 localnet = 192.168.1.115 localmask = 255.255.255.0 [88] type=friend secret=** dtmfmode=rfc2833 nat=yes host=dynamic canreinvite=no Trying with xlite at the other end Registered ok, can dial both ways, just no audio at all. In the log of xlite (cant see it at the moment as im not vnc'd in at the moment) it showed the xlite machines private IP address on some of the transactions that were logged. The client has a dynamic IP address so cant really be specified anywhere in the xlite configuration, I am also not sure on all the different firewall types. I was under the impression that there was no need to configure any portfowards at the sip softphone end. I will hopefully be using xlite or similar from a location with a very locked down firewall environment. I want to check all works on a normal nat router before trying it behind the nasty nat/firewall at this location. smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] NAT and sip issues
The rtp audio is going phone to phone, not via asterisk. This is one of the reasons I am trying to set up SER with Asterisk. I have an asterisk server behind NAT - no audio on the test external calls I have tried making so far. Read http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions - No solution evident from there, sounds like I have case 9. I would have thought that all I would have to do is port foward and have the external IP on the asterisk server, which I have done I have fowared 5060UDP, 8000UDP, and 35000 to 37000 UDP to the internal IP (192.168.1.115) I have put 35000 and 37000 into the rtp.conf as the start/end ports extracts of sip.conf: externip = 60.234.129.154 localnet = 192.168.1.115 localmask = 255.255.255.0 [88] type=friend secret=** dtmfmode=rfc2833 nat=yes host=dynamic canreinvite=no Trying with xlite at the other end Registered ok, can dial both ways, just no audio at all. In the log of xlite (cant see it at the moment as im not vnc'd in at the moment) it showed the xlite machines private IP address on some of the transactions that were logged. The client has a dynamic IP address so cant really be specified anywhere in the xlite configuration, I am also not sure on all the different firewall types. I was under the impression that there was no need to configure any portfowards at the sip softphone end. I will hopefully be using xlite or similar from a location with a very locked down firewall environment. I want to check all works on a normal nat router before trying it behind the nasty nat/firewall at this location. ___ 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 ___ 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
Re: [Asterisk-Users] NAT and sip issues
G.Marshall wrote: The rtp audio is going phone to phone, not via asterisk. This is one of the reasons I am trying to set up SER with Asterisk. I thought that canreinvite=no was supposed to force the audio to go via asterisk? smime.p7s Description: S/MIME Cryptographic Signature ___ 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
