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 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="$${external_rtp_ip}"/> Any advice? Thanks. Jimmy
If call has been successfully established and only audio is missing then try playing with http://wiki.freeswitch.org/wiki/Variable_disable_rtp_auto_adjust -- Thanks & Regards, Godson Gera FreeSWITCH Consultant <http://godson.in> On Wed, Mar 27, 2013 at 9:34 AM, Jimmy Chang <chang33.tw@gmail.com> wrote:
Hi,
I could make a call within NAT without any problem. When I made a call from internet, I got no audio. I have read the document
http://wiki.freeswitch.org/wiki/NAT_Traversal
and set the parameters. vars.xml <X-PRE-PROCESS cmd="set" data="bind_server_ip=my.server.ip"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=my.server.ip"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=my.server.ip"/> internal.xml <param name="ext-rtp-ip" value="$${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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
--
<param name="rtp-start-port" value="16384"/> <param name="rtp-end-port" value="32768"/> uncomment this in switch.conf.xml Thanks JT On Wed, Mar 27, 2013 at 12:04 AM, Jimmy Chang <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
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="$${external_rtp_ip}"/>
Any advice? Thanks. Jimmy
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
That will have no effect. That's for setting the port range, and those are already the defaults. You may find opening that port range on your router's firewall may help though. -Steve On 27 March 2013 15:11, johnthan123 <johnthan123@gmail.com> wrote:
<param name="rtp-start-port" value="16384"/>
<param name="rtp-end-port" value="32768"/>
uncomment this in switch.conf.xml
Thanks JT
On Wed, Mar 27, 2013 at 12:04 AM, Jimmy Chang <chang33.tw@gmail.com>wrote:
Hi,
I could make a call within NAT without any problem. When I made a call from internet, I got no audio. I have read the document
http://wiki.freeswitch.org/wiki/NAT_Traversal
and set the parameters. vars.xml <X-PRE-PROCESS cmd="set" data="bind_server_ip=my.server.ip"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=my.server.ip"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=my.server.ip"/> internal.xml <param name="ext-rtp-ip" value="$${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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Yes, problem was the firewall of the softphone pc. I also found another problem. When I used linphone for the softphone client, the agent could't hear voice from linphone. I have set "Direct connection to the Internet" of the linphone but still got the one way audio. Thank you all. JImmy ? 2013/3/28 ?? 12:10, Steven Ayre ??:
That will have no effect. That's for setting the port range, and those are already the defaults.
You may find opening that port range on your router's firewall may help though.
-Steve
On 27 March 2013 15:11, johnthan123 <johnthan123@gmail.com <mailto:johnthan123@gmail.com>> wrote:
<param name="rtp-start-port" value="16384"/>
<param name="rtp-end-port" value="32768"/>
uncomment this in switch.conf.xml
Thanks JT
On Wed, Mar 27, 2013 at 12:04 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
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (5)
-
Godson Gera
-
Jimmy Chang
-
johnthan123
-
royj
-
Steven Ayre