cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
Announcements
Walkthrough Wednesdays
398
Views
0
Helpful
2
Replies
Ejaz Ahmed
Beginner

SIP is registering but No audio

Hi Experts,

I have a Cisco ASA 5510 running 8.4(5) software version and my asterisk server is placed in the DMZ. The asterisk server is NATed with a public IP and forwarded the SIP port 5060(tcp/udp) and RTP 10000 - 20000 (tcp/udp) to the server. I can register the phone and make calls using public IP (of asterisk server) but can't here anything.

Is there any other ports needs to be opened?

 

Regards,

Ejaz

 

2 REPLIES 2

Hi,

can you see on the ASA any blocks / Drops?

Perhaps the RTP Port Range is another, than 10000 - 20000. For instance, Cisco uses UDP 16384 - 32767.

Regards,

DrM

This discussion has been reposted from Hall of Fame to the IP Telephony community.

Content for Community-Ad

Spotlight Awards 2021