May be the proxy-media mode is playing a role. Siptrace client<->FS<->IVVR server would clarify the picture On Wed, 27 Mar 2013 15:58:45 +0800 Jimmy Chang <chang33.tw@gmail.com> wrote:
Hi , I've added <action application="set" data="disable_rtp_auto_adjust=true"/> to my dialplan, but not worked. I need to explain the architecture of our environment. We have a IVVR server. FS, IVVR server and agents are all in the same LAN. The clients are in the internet. Here is the scenario. 1) A client makes call to FS. 2) FS bridges to a IVVR server for playing ivvr. 3) IVVR server bridges the call back to FS queue. 4) Available agent pickups the call.
I know it may be NAT setting problems, and I've tried to create a new profile(template from external) for the ivvr gateway. Now the client can hear the agent, but the agent can't hear client.
Any advice? Thanks. Jimmy
? 2013/3/27 ?? 02:43, Godson Gera ??:
If call has been successfully established and only audio is missing then try playing with http://wiki.freeswitch.org/wiki/Variable_disable_rtp_auto_adjust
-- Thanks & Regards, Godson Gera FreeSWITCH Consultant <http://godson.in>
On Wed, Mar 27, 2013 at 9:34 AM, Jimmy Chang <chang33.tw@gmail.com <mailto:chang33.tw@gmail.com>>wrote:
Hi,
I could make a call within NAT without any problem. When I made a call from internet, I got no audio. I have read the document
http://wiki.freeswitch.org/wiki/NAT_Traversal <http://wiki.freeswitch.org/wiki/NAT_Traversal>
and set the parameters. vars.xml <X-PRE-PROCESS cmd="set" data="bind_server_ip=my.server.ip"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=my.server.ip"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=my.server.ip"/> internal.xml <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> external.xml <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
Any advice? Thanks. Jimmy
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