Incoming Unregistered Calls with no audio

Hello All,

I’m new with Asterisk and I’m integrating it with a SIP Server which forwards specific numbers (phone range) to Asterisk without a registration. Well, the call establishment (SIP signaling) is proceeding ok but there is no audio. I can see RTP packets flowing from both agents but I’m unable to hear anything.
One point that I realized is that during the call setup (Asterisk) is acting as a Proxy and stays in the middle of the signalization during the call. What I’d like to know is why I don’t have audio. Is there anything missing in the configuration?

Please any clues are welcome.

Bellow is my sip.conf and my extension.conf

sip.conf

[general]
bindport => 5070
directrtpsetup=yes

[1000]
type=friend
context=local-calls
host=dynamic
srvlookup=yes

[2000]
type=friend
context=local-calls
host=dynamic
srvlookup=yes

[+551937071000]
type=friend
context=local-calls
host=dynamic
servlookup=yes

[server-trunk]

;Setting the type to peer an incoming call from ims worked fine.
type=friend

host=172.24.21.137
port=6060
domain=open-carrier.net
context=incoming-carrier-calls
deny=0.0.0.0/0
permit=172.24.21.137/32
srvlookup=yes
insecure=very
nat=no
canreinvite=no

=========================

extension.conf

[globals]
TESTE=SIP/+551937071000
SERVER-TRUNK=SIP
SERVER-DOMAIN=open-carrier.net

[general]
autofallthrough=yes

[incoming]
exten => s,1,Answer()
exten => s,n,Background(hello-world)
;Wait for a digit
;exten => s,n,WaitExten()
exten => s,n,Hangup()

[local-calls]

; public numbers (local format)
exten => 1937071000,1,Dial(${TESTE})

; public numbers (global format)
exten => 551937071000,1,Dial(${TESTE})
exten => +551937071000,1,Dial(${TESTE})

include => outbound-carrier-calls

[outbound-carrier-calls]
exten => _193755.,1,Dial(${SERVER-TRUNK}/${EXTEN}@${SERVER-DOMAIN})
exten => _193755.,n,Congestion()
exten => _193755.,n,Hangup()

[incoming-carrier-calls]
exten => +551937071000,1,Answer()
exten => +551937071000,n,Dial(${TESTE})
exten => +551937071000,n,Hangup()

Asterisk is actually a back to back user agent, not a proxy, i.e. the SIP signalling is terminated, on each channel, and signalling between channels is by Asterisk specific means.

However, it can bypass the voice path, but you have forbidden it from doing that.

Your most likely problems are firewall related. There would also be problems if you are really using NAT, although they might compromise the signalling as well.