SIP w/NAP remote * problems - no sound - please help

Asterisk Asterisk Support
1

I have worked on this for a week now and have looked everywhere. I know a lot about NAT and stuff now. Have tried everything from every post. Can someone look at this and walk me through?

First of all, the very first time I tried a softphone(X-Lite) to my * box and dialed the demo, I got the voice and sound perfectly. Then I changed something in the conf’s and now no sound. The softphone registers and dials. The CLI shows that the file is playing, but no sound from my speakers.

The * box is a public server not behind a nat. It is remote (a RackSpace linux hosted server). I’m using a softphone from my desktop to connect to * via SIP. My PC is connected to a wireless router(nat I think) connected to the cable modem. Not using DMZ, although I have tried that too.

sip.conf
[general]
port = 5060
context = sip
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
externip = xx.x.xxx.xx
localnet=xx.x.xxx.xx/255.255.255.0
nat=no
[042100]
type=friend
secret=ready
username=042100
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=gsm
context=demo
nat=no ;if set to yes, phone gets ring tone and no answer.
canreinvite=no
dtmfmode=auto
qualify=yes

extensions.conf(relevant piece)
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
;
[globals]
;
[demo]
exten => s,1,Wait,1
exten => s,n,Answer
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n(restart),BackGround(demo-congrats)
exten => s,n(instruct),BackGround(demo-instruct)
exten => s,n,WaitExten

exten => 2,1,BackGround(demo-moreinfo)
exten => 2,n,Goto(s,instruct)

exten => 3,1,Set(LANGUAGE()=fr)
exten => 3,n,Goto(s,restart)

exten => 1000,1,Goto(default,s,1)

dialing 1000, CLI shows this:
– Executing Goto(“SIP/042100-0840c8b0”, “default|s|1”) in new stack
– Goto (default,s,1)
– Executing Wait(“SIP/042100-0840c8b0”, “1”) in new stack
– Executing Answer(“SIP/042100-0840c8b0”, “”) in new stack
– Executing Set(“SIP/042100-0840c8b0”, “TIMEOUT(digit)=5”) in new stack
– Digit timeout set to 5
– Executing Set(“SIP/042100-0840c8b0”, “TIMEOUT(response)=10”) in new stack
– Response timeout set to 10
– Executing BackGround(“SIP/042100-0840c8b0”, “demo-congrats”) in new stack
– Playing ‘demo-congrats’ (language ‘en’)

No sound.

2

There are a couple of things you should change around to see if it will start working. Firstly, when you define your extension in sip.conf “[042100]”, you should try changing the following:

nat = no should be nat=yes
context= (whatever context you have set up for dialing. If you are still using the demo, then it will be demo)
dtmf=rfc2833 (instead of auto)

Also, I would remove many of the lines that may not be necessary for your setup. Lines like “externip” and “localnet”. I would also remove the nat=no from the [general] context.

See if that does anything and let us know.