[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
>
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