[Asterisk-Users] no audio with sip

[James Jones]

I can make call in to the asterisk server listen to voice mail, and do the echo test. When make a call I get no audio inbound or outbound. When making incoming call I can leave a valid voice message, but when then extentions pick up again no audio inbound or outbound. I am using Xten lite and Broadvoice. Below are the messages from console when call is made and my sip.conf. Any thoughts.

 

 

console info:

 

 

    -- Executing Dial("SIP/xlite2-725e", "SIP/[EMAIL PROTECTED]") in new stack
    -- Called [EMAIL PROTECTED]
    -- SIP/sip99413-6d20 is ringing
    -- SIP/sip99413-6d20 answered SIP/xlite2-725e
    -- Attempting native bridge of SIP/xlite2-725e and SIP/sip99413-6d20
    -- Attempting native bridge of SIP/xlite2-725e and SIP/sip99413-6d20

 

sip.conf:

 

[general]
disallow=all
allow=ulaw
port=5060                       ; Port to bind to
bindaddr=0.0.0.0                ; Address to bind SIP channel to
externip=24.218.94.95
localnet=192.168.2.0
localmask=255.255.255.0
context=default         ; Default context for incoming calls
maxexpirey=180
defaultexpirey=160
canreinvite=no
tos=reliability
srvlookup=yes

register => 4137711401:[EMAIL PROTECTED]/99413

 

[sip99413]
secret=password
username=4137711401
host=sip.broadvoice.com
type=friend
nat=yes
canreinvite=no
dtmfmode=inband
fromuser=4137711401
callerid=4137711401
context=incoming
fromdomain=sip.broadvoice.com
qualify=yes
disallow=all
allow=ulaw

[xlite2]
type=friend
username=xlite2
secret=password
callerid="outcast" <5678>
host=dynamic
nat=yes                       ; X-Lite is behind a NAT router
canreinvite=no                ; Typically set to NO if behind NAT
disallow=all
allow=ulaw

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