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SIP Source Port differetn from dial-peer one



i have a cisco router 2921 with voice licence and i want to configure SIP trunk with my provider. 

the configuration is next:

voice service voip
 no ip address trusted authenticate
 dtmf-interworking rtp-nte
 allow-connections sip to sip
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g711alaw


dial-peer voice 130 voip
 numbering-type unknown
 preference 2
 destination-pattern .T
 media forking
 rtp payload-type comfort-noise 13
 session protocol sipv2
 session target ipv4:
 session transport udp
 voice-class codec 1
 voice-class sip early-offer forced
 dtmf-relay rtp-nte sip-notify sip-kpml
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
dial-peer voice 160 voip
 description +++To Asterisk+++
 numbering-type unknown
 preference 1
 destination-pattern 4951T
 rtp payload-type comfort-noise 13
 session protocol sipv2
 session target ipv4:
 session transport udp
 voice-class codec 1
 dtmf-relay rtp-nte sip-notify sip-kpml


i can do calls via My provider but when i try to call internall numbers, i fail. i have translation-profile configured correctly, i guess that is not hte problem. 

what i discovered is that i send SIP signaling to, and ISP accests and calls work, but when the Provider initiate SIP session, he initiate with source Port 5062, and my router does not ceate incomming leg:

*May  5 16:06:58.788: //1393/3331D6519C88/SIP/Error/act_recd_invite_send_msg_failure:
 Send Error to for transport UDP
*May  5 16:06:58.788: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQSIG:
 No Inbound Container Created !!!


can somebody help me with this issue? can i change the dialpeer to receive from one source port but send to 5060? 




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