Hi everyone
I did all steps for external user behind NAT I already confg a sip_nat.conf and sip_custom.conf and opend ports “4569-4569 , 2727-2727 , 5004-5082 , 10000-20000 , 5090-5090” and in external user opend 5060 port , the problem i cant hear him when i call and he can hear me but when he call i can hear him and he can hear me .
i dond know where is the problem and i heared about module for pbx in aflash for external users without configer is that module will help or what o should do ???
I am sorry I do not know how to answer this question. If you have no audio something is blocking the UDP port. You have to trace it out one step at a time.
Usually the debug facility in Asterisk for RTP is sufficient to determine the source of the trouble. If that won’t do it I usually use Ethereal (Wireshark) to examine the data flows on the network.
ok but now you didnt answer me about my case what should i do in that case
“I have repotc router and the remote phone too and i have bigger problem with “netgear” router that i cant call him and he cant call me at all ,i use a xlite softphone before buying any hardware i opened port in repotc router by virtual server like this http://portforward.com/english/routers/port_forwarding/Repotec/RP-IP2401…
and i have sip_nat.conf like that
externhost=myname.endofinternet.org i have that host from www.dyndns.com
externrefresh=120
localnet=192.168.16.0/255.255.255.0
nat=yes
and the same in sip_custom.conf
what should i do ??”
Hamachi will only help for a softphone.
yum install hamachi
setup-hamachi
It is far easier to start your account on a PC then join the network you created on the Linux host.
by the way any one know any thing about “hamachi” for pbxif i think that will help for external users but i dont know how ??
I have repotc router and the remote phone too and i have bigger problem with “netgear” router that i cant call him and he cant call me at all ,i use a xlite softphone before buying any hardware i opened port in repotc router by virtual server like this http://portforward.com/english/routers/port_forwarding/Repotec/RP-IP2401A/Lingo_VoIP.htm
and i have sip_nat.conf like that
externhost=myname.endofinternet.org i have that host from www.dyndns.com
externrefresh=120
localnet=192.168.16.0/255.255.255.0
nat=yes
and the same in sip_custom.conf
what should i do ??
We need far more information to help.
What type of remote phone?
Post your sip_nat.conf file.
You don’t need to have as many ports open as you do. You should also not have to open any ports on the remote phone however for troubleshooting you can hard code UDP 5060 and 10000-20000 on the remote network.
Ensure you have UDP and not TCP opened.
When the call is in progress run an ‘rtp debug’ from the Asterisk CLI and see if you can spot the error. Don’t forget to turn it off with ‘rtp debug off’
Was that TCP or UDP ports? You want UDP. And did you “port forward” or just open the ports?
What type of router are you using? There are some firewall/routers that just won’t work with SIP.