[asterisk-users] asterisk & x-lite

jonas kellens
Tue, 22 Dec 2009 09:53:58 -0000

Where is your definition of codecs ??

On Tue, 2009-12-22 at 11:26 +0200, zehra yildiz wrote:

> 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.
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...