SIP No Audio on Both Side

Asterisk Asterisk Support
1

Hi

I have SIP incoming number and my asterisk server also on same network

SIP host = ..12.12
Router’s IP = .***.10.
Asterisk Server IP = 192.168.1.1

extensions.conf =>
[general]

static=yes

writeprotect=no

[incoming]
exten => s,1,Ringing
exten => s,2,Wait,5
exten => s,3,Answer
exten => s,4,Backgroung(transfer)
exten => s,5,Dial,Sip/100
…
sip.conf =>

[general]

[general]
context=incoming
bindport=5060
bindaddr=0.0.0.0
insecure=very
disallow=all

allow=iLbc

nat=no
register => mynumber:mypass@myprovider.com:4545(not using 5060)

When call comes in caller listening ringing for 5 seconds but cannt hear Background(transfer).
But it works fine when it got calls from PSTN from X100P :smile:

Any Idea

Regards,

2

If you’re firewalled make sure you’re allowing RTP traffic on UDP ports (10k-20k typically)

3

I think this might be a network issue. The following thread might be of help; forums.digium.com/viewtopic.php?t=7854

4

I already tried everything but :frowning:

5

your router would appear to be on a different subnet to your Asterisk server (depends on your mask)

if your Asterisk server is behind a NAT interface you need nat=yes and externip=xxx.xxx.xxx.xxx in [global]. you’ll also need a signalling port and the RTP ports forwarded to your Asterisk box.

to get appropriate help you need to post log fragments for a call, and make sure you’ve searched the wiki … you’re not the first person to have problems and thankfully lots of people have contributed their experience to the wiki.