One way audio problem with SIP Provider

Asterisk Asterisk Support
1

Hi everyone,

I’m Asterisk newbie and I have Linux/Debian box with two ethernet cards, one for local network and the other for sip provider.

Here my configuration:
sip.conf

[general]

; directmedia=no i’m wondering if this is necesary
careinvite=no

[siprovider]
type=friend
context=incoming
insecure=port,invite
allowguest=yes
host=x.x.x.x ;(ip service provider)
careinvite=no
disallow=all
allow=alaw

[25928150]
type=friend
host=dynamic ;
careinvite=no;
context=phones
disallow=all
allow=alaw

[25928151]
type=friend
host=dynamic ;
careinvite=no;
context=phones ;
disallow=all
allow=alaw

extension.conf

[globals]

[general]
autofallthrough=yes

[default]

[incoming_calls]

[phones]
include => internal
include => remote

[internal]
exten => _25928150,1,Dial(SIP/${EXTEN})
exten => _25928151,1,Dial(SIP/${EXTEN})

[remote]
exten => _X.,1,NoOp() ;
exten => _X.,2,Dial(SIP/${EXTEN}@siprovider)
exten => _X.,n,Hangup()

[incoming]
include => internal

I have audio in service provider site, but from provider to asterisk there is none.

(By the way this is test enviroment for now, not in production yet)

Please help me if someone have a clue how to fix this. (Sorry for my english :smile: )

2

The normal reasons are firewalls or NAT. You are likely in a NAT environment but I see no specification of a public address.

Note that allowguest is both undesirable and only supported in the general section.

3

I solved the issue.
I connect my laptop with my own softphone in local network and works! Seems clients are using antivirus and firewall that was bocking rtp ports.
Now I testing redirect and call fowarding between service provider, hope works!
Thanks david for help!