[asterisk-users] asterisk & x-lite

[BERGANZ François]

Try tcpdump to see where RTP go from asterisk.

Configure your x-lite

Use stun server ?

 

 

 

P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

 

De : asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] De la part de zehra yildiz
Envoyé : mardi 22 décembre 2009 10:26
À : asterisk-users at lists.digium.com
Objet : [asterisk-users] asterisk & x-lite

 

Hello All,

I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The
softphone can call the other one but I can' t hear any voice. My
configuration files are below:

[root at localhost asterisk]# cat sip.conf
[general]
canreinvite=yes

[1001]
username=1001
password=1001
type=friend
context=phones
host=dynamic

[1002]
callerid=1002
username=1002
password=1002
type=friend
context=phones
host=dynamic

[root at localhost asterisk]# cat extensions.conf
[globals]

[general]
autofallthrough=yes

[default]

[incoming_calls]

[phones]
exten => _1XXX,1,NoOp()
exten => _1XXX,n,Dial(SIP/${EXTEN},30)
exten => _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => _1XXX,n,Hangup()


PS: My sip server and softphones are in the same network subnet. There are
not any firewall or iptables rules. I tried the "nat=yes" parameter but no
changes.

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