[asterisk-users] The call is established but without exchanged voice packets

Asmaa Ahmed asabatgirl at hotmail.com
Fri Sep 20 10:26:09 CDT 2013


Hi Matthew,
Indeed I missed your previous message!After changing the externip, it worked successfully... The sip session is established with the complete  three-way handshake, and the voice packet is exchanged with no problem!
Many thanks.   
> Date: Fri, 20 Sep 2013 10:01:52 -0500
> From: mroth at imminc.com
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] The call is established but without exchanged voice packets
> 
> Asmaa, 
> 
> You're getting ahead of yourself.  How do you expect audio to work if
> your firewall/NAT settings aren't even configured correctly to
> establish SIP sessions?
> 
> Go back and read the message that I sent yesterday.  Fix the SIP 
> three-way handshake problem.  That is step 1 and you'll know you have
> it right when you stop seeing 'Retransmission timeout reached on
> transmission' errors.
> 
> You still won't have audio but that's step 2.  It requires properly
> configuring Asterisk's NAT settings and the firewall(s) between the
> phones and the server to allow RTP traffic to flow, but don't worry
> about it until step 1 is complete.
> 
> Regards,
> 
> Matthew Roth
> InterMedia Marketing Solutions
> Software Engineer and Systems Developer
> 
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
> 
> 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...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130920/bc81499b/attachment.htm>


More information about the asterisk-users mailing list