[asterisk-users] One Way Audio

[Duncan Turnbull]

Can you do a tcpdump to look at the rtp streams on your box and check they are both generating and aiming at the right places

IAX will have no issue with NAT/firewall but SIP / RTP can. 

tcpdump -nn udp and portrange 10000-20000 
(pick your portrange if its operating on something else)

Should show you mad lines of rtp going backwards and forwards (like below) when there is a conversation in place. If you can see it being sent from the asterisk box but not heard by the client then either try a different client, or something is blocking the return leg to your client

13:00:21.309139 IP 192.168.63.83.16450 > 192.168.63.30.13902: UDP, length 172
13:00:21.328703 IP 192.168.63.83.16450 > 192.168.63.30.13902: UDP, length 172
13:00:21.348572 IP 192.168.63.83.16450 > 192.168.63.30.13902: UDP, length 172
13:00:21.369096 IP 192.168.63.83.16450 > 192.168.63.30.13902: UDP, length 172
13:00:21.388572 IP 192.168.63.83.16450 > 192.168.63.30.13902: UDP, length 172

Cheers Duncan

On 10/03/2011, at 12:26 PM, Tim King wrote:

> Thank you I have also tried those settings. The main thing is coming from my voip provider all I am doing is bridging the calls to two other devices (1 trixbox and 1 digium aa50) via IAX trunks. Both devices are answering with an IVR and when I call in I can not hear the IVR. However if I call directly to a SIP client the person answering the SIP phone can hear me but I can not hear them at all.  Its definately not a NAT issue which is what makes it even more confusing. When the call is in place a sip show channels shows me both lefs of the call and they are both using either alaw or ulaw so it should not be a codec translation issue either.
> 
> On Wed, Mar 9, 2011 at 6:19 PM, Satish Patel <satish_lx at hotmail.com> wrote:
> 

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