No audio between two outgoing calls

Hi everyone,

I would like to connect two phone numbers using a web page, asterisk and voipbuster. The web page has two fields: the caller’s extension and the callee’s number. When the caller hits the submit button, the web page calls a java program (asterisk-java) that places the call through asterisk. The caller’s phone rings, he picks up, then the callee’s phone rings and the callee picks up.

When both phones are pstn numbers, there no sound between the two phones.

When the caller’s phone is a sip one (I use X-lite), the sound flows very well between the two phones (exten=>4321,1,DIAL(SIP/mike).

Why there no sound in the first case ?

Thank you in advance

here is my extension.conf


here is my sip.conf

type = friend
username = paul
secret = talvi
host = dynamic
context = outgoing


canreinvite = no
dtmfmode = inband

here is my manager.cong

enabled = yes
port = 5038
bindaddr =


Hi, I’ve had the same problem. I have noticed that with twinkle (a softphone) everything works good, but when connecting two pstn phones the session is estabilished, but no audio goes through the call.

I’ve noticed that it was a firewall problem:

when you estabilish a session with sip, the message’s body contains informations about audio protocol (for instance RTP) and destination and source ip address and ports. Firewalls doesn’t know about RTP ports it has to open so it drops all audio packet incoming in your network. As result: session estabilished but audio doesn’t work.

When I had this problem, i was writing a Sip client, so i solved sending a CN (comfort noise) RTP packet, making the firewall open right ports. It is surely a firewall problem.

Hope this information helps.

N.B. I’ve studied twinkle’s source code and so the mystery is: usually softphones send a cn rtp packet opening firewall’s ports for audio flow. So that is why with softphones it works and with pstn phones does not.

Thanks Nadia,

Is there a way to configure the firewall at the server side (asterisk or its host) rather than sending cn rtp packets from the sip client ?

In addition, when I connect two pstn phones, does the audio automatically pass through my asterisk box ? is there a way to avoid this ?

Sorry i don’t know if there is a way to configure firewall… :frowning:

About your second question, i know that u can configure a canreinvite=yes in your sip.conf. From :

“When SIP initiates the call, the INVITE message contains the information on where to send the media streams. Asterisk uses itself as the end-points of media streams when setting up the call. Once the call has been accepted, Asterisk sends another (re)INVITE message to the clients with the information necessary to have the two clients send the media streams directly to each other”.