CCMv10.5 ->SIP-TRUNK-ASTERISk

[Rafael Jimenez]

I have been trying to setup a sip trunk between ccm and asterisk. I followed all info that I get in other forums.

Currently I just get one way audio (asterisk to ccm). Also the call is dropped approximately one second when the person in the asterisk phone start to speak. I can heard the other end (asterisk side). They can not hear me.

attached a trace, appreciate any help.
thanks.

[Ayodeji Okanlawon]

Rafael,

This log doesn't seem to include the call log.. I cant seen any INVITE request sent to the Asterisk server..

Did you look in this log to see if you can find the calling and called number?

[Rafael Jimenez]

Yes, you are right. I forgot enable the sip traces on the CCM.
after check the trace, I found the cause. Long time ago somebody enabled nat on the asterisk. I found in the trace a unknown (3355@190.0.37.82:5060) ip address talking with the CCM (192.168.110.2).

When I executed a greep command in the asterisk configuration folder I found that ip in the sip.conf file, in a field called Externaddr:.
I disabled the nat and change that address to the internal ip of the asterisk and now is working.

Thanks.

[Dennis Mink]

SIP doesnt like NAT a lot.

http://ciscoshizzle.blogspot.com.au/2014/01/sip-firewall-traversal-and-why-does-sip.html

[Rafael Jimenez]

I dont know much about asterisk. I´m not sure if was the nat, but the fact is the sip trace shows the asterisk talking with the CCM with the internal ip, then when the media exchange started the ip changed to the ip in the externalip field.