[AsteriskBrasil] wi max - freepbx

[Leonardo Gomes Figueira]

Fernando Hallberg - Flex Digital wrote:

> 
> Found RTP audio format 0
> 
> Found RTP audio format 101
> 
> Peer doesn't provide audio. Callid 
> 147b548332e7f5351999f70e05513a9e em 189.xxx.xxx
> 
> Found audio description format PCMU for ID 0
> 
> Found audio description format telephone-event for ID 101
> 
> Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 
> (nothing), combined - 0x4 (ulaw)
> 
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 
> (telephone-event), combined - 0x1 (telephone-event)
> 
> Peer audio RTP is at port 0.0.0.0:31552
> 

O problema está neste ponto: Peer audio RTP is at port 0.0.0.0:31552

Habilite na console um "sip debug peer XXX" e envie os pacotes SIP desde 
o INVITE inicial para analisarmos melhor.

  Leonardo