[asterisk-users] SIP??

[Florea Igor]

Hi,
I'm new to *,so i apologize for stupid questions.
I'm having problem with this arhitecture:
I'm calling asterisk from behind a NAT(sjphone user) with a low band so I'm 
using GSM codec.
In extensions.conf I have:
exten => 337,1,Dial(SIP/99@<ip_pbx2>)
so when i dial 337 from sjphone Asterisk is colling 99 on ip_pbx2.
RTP stream between sjphone and Asterisk are ok (GSM).
The problem is rtp packets from Asterisk to ip_pbx2 are also GSM although 
ip_pbx2 sip is telling asterisk It only knows "codec 0"
Is this a config problem or a bug?
Igor,