Asterisk Issue in Ubuntu server

Asterisk Asterisk Support
1

Hello communitty,

I have installed asterisk 13.28.1 in a Ubuntu Server 18.04.3 (Virtual Machine)

I have also installed 2 softphones (zoiper3), one in my physical windows machine, and the other on the VW.

The Windows machine and the VM are connected to the same network.

When i make the call between the two, I dont have audio.

But when i unplug the WAN wire from the router, in order not to get the external ip, then audio comes back.

I have disabled firewall both on Windows and VM .

Could you please help me ??

Thanks in advance.

Here is a link with the images of the conf. files. : https://imgur.com/a/W921xzj

2

Hello community again,

I’ve solved the issues with the audio.

But i have to turn off the firewall from the VM.

Can someone help me with that ?

Thanks in advance.

3

If you are using pjsip nat parameters is invalid. Use this parameters.

rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes

And should not turn off firewall. Only must give access rtp ports.
By default, SIP uses the UDP port 5060, to open the port run:

sudo ufw allow 5060/udp

If you enabled the Real Time Protocol (RTP) then you also need to open the following port range:

sudo ufw allow 10000:20000/udp

Feel free to adjust the firewall according to your need.

4

You seem to have a confused mixture of chan_sip and chan _pjsip configurations, although your outgoing calls are all chan_sip.

nat= is overused.

If possible, please provide configurations in the posting itself, as text. Failing that, as external plain text files.

5

Hello david,

Thank you for your response.

Here are my configurations :

extensions.conf file :

[from-internal]

exten=>6001,1,Dial(SIP/George,20)
exten=>6001,n,VoiceMail(6001@vm-demo,u)
exten=>6002,1,Dial(SIP/Renia,20)
exten=>6002,n,VoiceMail(6002@vm-demo,u)
exten=>6500,1,Answer(500)
exten=>6500,n,VoiceMailMain(@vm-demo)

pjsip.conf file :

[transport-udp-nat]
type=transport
protocol=udp
bind=0.0.0.0
localnet=192.168.0.0/24
localnet=127.0.0.1/32
external_media_address=x.x.x.x
external_signaling_address=x.x.x.x
qualify=yes
externip=x.x.x.x
nat=force_rport,comedia

endpoint_internal
type=endpoint
context=from-internal
disallow=all
allow=ulaw

auth_userpass
type=auth
auth_type=userpass

aor_dynamic
type=aor
max_contacts=1

George
type=endpoint
context=from-internal
disallow=all
allow=ulaw
transport=transport-udp-nat
direct_media=no
auth=George
aors=George
George
password=1234
username=George
George

Renia
type=endpoint
context=from-internal
disallow=all
allow=ulaw
transport=transport-udp-nat
direct_media=no
auth=Renia
aors=Renia
Renia
password=1234
username=Renia
Renia

sip.conf file :

[general]
transport=udp
nat=force_rport,comedia
directmedia=no
disallow=all
allow=ulaw
;nat=force_rport,media
externip=x.x.x.x
localnet=192.168.0.0/255.255.255.0

friends_internal
type=friend
context=from-internal
host=dynamic
disallow=all
allow=ulaw
externip=x.x.x.x

George
secret=1234

Renia
secret=1234

6

You have both chan_sip and chan_pjsip trying to bind to port 5060. That isn’t going to work.

You have a least one parameter that is only defined for chan_sip (nat=) in your pjsip configuration. I think others may be the same.

7

So what do i have to change to be correct ?

8

your exensions.conf use chan_sip module like "Dial(SIP/…) "
but your pjsip.conf use chan_pjsip module

if i were you, edit extensions.conf ( “SIP” > “PJSIP”) and disable chan_sip module(use only chan_pjsip)

9

You also need to convert what is currently in sip.conf to the equivalent for pjsip.conf.

10

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