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,
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