Voice is not coming in SIP to SIP call


#1

Hi,

I am using the Asterisk 1.6.2 and I setup the two SIP extensions let say 2001 and 2002. When I am calling from 2002 to 2001 then 2001 getting the incoming call ring. But after the answered the call there is no voice.

And one more issue is that when I am calling from 2001 to 2002, then i get the Busy message. But the 2001 phone is free.

I have done the following settings in sip.conf

[2001]
type = friend
user = 2001
secret = 2001
fromuser = 2001
context = default
nat = yes
qualify = yes
host=dynamic
disallow = all
allow = all

[2002]
type = friend
user = 2002
secret = 2002
fromuser = 2002
context = default
nat = yes
qualify = yes
host=dynamic
disallow = all
allow = all

Can you please let me know what I am missing in configuration.

Thanks in advance.

Thanks & Regards,

Ketan


#2

[quote=“ketan.jadhav”]
Can you please let me know what I am missing in configuration.[/quote]

Not until you provide the network topology and diagnostic output.

At a guess, though, many people think they need nat=yes when they actually need externip.


#3

Hi,

yes, I set the externip and then voice is coming now.

But other issue is voice quality is not good some time voice is cracking. I am using the G711 ulaw codec in iphone.

But when I used the xlite to xlite call from pc there is no voice. can you suggest me for better voice quality which settings required.

Regards,

Ketan


#4

Voice quality problems are due to network or CPU overloads, or due to high scheduling latencies as the result of trying to run on a virtual machine without very careful setup and control of conflicting loads.