[asterisk-users] port 5060 is blocked by ISP

[SamyGo]

actually its a one-way audio issue due to NAT !

alok , please explain your network flow for end to end client-server-client.

You may need to set nat=yes for your sip peer behind NAT. If the server is
behind NAT router/firewall use externip=<public.ip.of.server> field.
Also provide sip traces of this call.

Another thing to do for your learning. Execute wireshark on both softphone
systems and set "sip | rtp" as filter and see where are the RTP streams
going on each end !

Take a complete capture on Asterisk server by executing the command "sip
set debug on" and make a call.

BR
Sammy


On Mon, Jul 2, 2012 at 4:39 PM, Thomas Kenyon <digium at sanguinarius.co.uk>wrote:

> alok srivastava wrote:
>
>> dear
>> i have configured properly asterisk. At the one end i am using x-lite
>> soft ph and another end twinkle. call is going properly from both end but
>> after picking the phone not able to listen other one.
>> when i checked the port 5060 on the asterisk server it is always showing
>> closed while i have flushed all the rules from iptables (iptables -F)
>>
>> PORT     STATE  SERVICE VERSION
>> 5060/tcp closed sip
>>
>>  telnet localhost 5060 (could not connect)
>>
>> regards
>> alok
>>
>>
>>  SIP is only used to setup (and stop etc.) the call. The actual audio is
> sent via rtp.
>
> /etc/asterisk/rtp.conf
>
> Should tell which ports asterisk is using for rtp, you will need to make
> sure that the remote host can connect to these ports.
>
> There are lots of articles around on how to resolve this.
>
>
>
>
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