SPA112 can not hear voice on my android phone with CSipSimple

Question

I configure a SPA112 and between lines 1 and 2 the calls and answers is ok (I receive,  attend calls and hear the voice). When my cell phone call for any of the rows, is also okay. The problem is when one of the lines dials to my cell phone. I answer calls and only hear noises in the cell and the line is silent. I've used iptables the,  seep show peers (in asterisk 1.8)  is ok, etc. ... my dial plan is: (* xx | [3469] 11 | 0 | 00 | [2-9] xxxxxx | 1xxx [2-9] xxxxxxS0 | xxxxxxxxxxxx. )
e.g.: I'm using dyndns ... Does the problem is with RTP port?? Or codec?
thank you

Answer 1

Hi Neuberfran,
It is sould like RTP issue part of RTP is codec. If it is codec issue you can go to:
ATA GUI --> select admin mode in top right site,  Select Advanced mode in top right site
select Voice in menu bar,
Line 1:
go to Audio Configuration section
Preferred Codec G711u
Second Preferred Codec G711a
Thired preferred Codec G729a
click Save,
do same setting in your Line 2 because you useing two lines
example: if your ATA IP is 192.168.1.100
go to your router port forrwarding and make rules UDP port 10000 to Port 30000 to your ATA 192.168.1.100
another port forwarding for SIP UDP port 5060 to port 5080 to your ATA 192.168.1.100
Thanks,
Moh