OpenSIPS Load Balancer and FreeSwitch issue
All, Hopefully someone can assist me - I'll gladly give $50 to the person who helps me fix the issue - I have an OpenSIPs server configured as a load balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) that receives the call and forwards it to my Freeswitch Server for an IVR. When I had the call just routing to the Freeswitch server I got the audio working by setting the ext-rtp-ip to the public IP. Now that I route the SIP call through the OpenSIPs server there is no audio - I'm guessing it a NAT issue as always. Both the OpenSIPS and FreeSwitch server have an internal private IP - OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2 Both servers also have a public IP that routes to the internal IP OpensIPS = 47.1.1.1 and Freeswitch = 47.1.1.2 Call gets received by the OpenSIPs via the external IP - routes the call to the internal IP on the Freeswitch server which answers the call - but no audio - I think the FreeSwitch is trying to route the RTP Audio via its internal private IP instead of the public IP of 47.1.1.2. Not sure if this is the real issue or how to configure it route RTP properly ... all help is appreciated. $50 Paypal to whoever fixes this for me! Thanks -Peter
Someone familiar with this might be able to answer right off the bat, but if you pastebin a SIP trace (and an fs_cli log for completeness) , the problem should become apparent. Are you using the internal profile for all the calls? If you use the external.xml profile, you might need to set your ext-rtp-ip in that file, too. Seeing a trace will tell (almost) the whole story. -Avi On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato <peter@spinato.ca> wrote:
All,****
Hopefully someone can assist me - I'll gladly give $50 to the person who helps me fix the issue - I have an OpenSIPs server configured as a load balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) that receives the call and forwards it to my Freeswitch Server for an IVR. When I had the call just routing to the Freeswitch server I got the audio working by setting the ext-rtp-ip to the public IP. Now that I route the SIP call through the OpenSIPs server there is no audio - I'm guessing it a NAT issue as always.****
** **
Both the OpenSIPS and FreeSwitch server have an internal private IP - OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2****
Both servers also have a public IP that routes to the internal IP OpensIPS = 47.1.1.1 and Freeswitch = 47.1.1.2****
** **
Call gets received by the OpenSIPs via the external IP - routes the call to the internal IP on the Freeswitch server which answers the call - but no audio - I think the FreeSwitch is trying to route the RTP Audio via its internal private IP instead of the public IP of 47.1.1.2. Not sure if this is the real issue or how to configure it route RTP properly ... all help is appreciated. $50 Paypal to whoever fixes this for me! Thanks****
** **
-Peter****
_________________________________________________________________________ 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
The calls go to internal profile - I guess case the OpenSIPs server is local lan - is there a way to force that calls to external profile as I find calls that hit that profile load the ext-rtp-ip ip and work. From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Saturday, December 17, 2011 11:07 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue Someone familiar with this might be able to answer right off the bat, but if you pastebin a SIP trace (and an fs_cli log for completeness) , the problem should become apparent. Are you using the internal profile for all the calls? If you use the external.xml profile, you might need to set your ext-rtp-ip in that file, too. Seeing a trace will tell (almost) the whole story. -Avi On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato <peter@spinato.ca> wrote: All, Hopefully someone can assist me - I'll gladly give $50 to the person who helps me fix the issue - I have an OpenSIPs server configured as a load balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) that receives the call and forwards it to my Freeswitch Server for an IVR. When I had the call just routing to the Freeswitch server I got the audio working by setting the ext-rtp-ip to the public IP. Now that I route the SIP call through the OpenSIPs server there is no audio - I'm guessing it a NAT issue as always. Both the OpenSIPS and FreeSwitch server have an internal private IP - OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2 Both servers also have a public IP that routes to the internal IP OpensIPS = 47.1.1.1 and Freeswitch = 47.1.1.2 Call gets received by the OpenSIPs via the external IP - routes the call to the internal IP on the Freeswitch server which answers the call - but no audio - I think the FreeSwitch is trying to route the RTP Audio via its internal private IP instead of the public IP of 47.1.1.2. Not sure if this is the real issue or how to configure it route RTP properly ... all help is appreciated. $50 Paypal to whoever fixes this for me! Thanks -Peter _________________________________________________________________________ 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 _____ No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11
Yes, send it to the freeswitch server port 5080 Or if you don't need the internal profile, remove it and set external to use port 5060. -Avi On Sun, Dec 18, 2011 at 5:07 AM, Peter Spinato <peter@spinato.ca> wrote:
The calls go to internal profile – I guess case the OpenSIPs server is local lan – is there a way to force that calls to external profile as I find calls that hit that profile load the ext-rtp-ip ip and work.****
** **
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Avi Marcus *Sent:* Saturday, December 17, 2011 11:07 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue****
** **
Someone familiar with this might be able to answer right off the bat, but if you pastebin a SIP trace (and an fs_cli log for completeness) , the problem should become apparent.****
** **
Are you using the internal profile for all the calls? If you use the external.xml profile, you might need to set your ext-rtp-ip in that file, too. Seeing a trace will tell (almost) the whole story.****
****
-Avi****
** **
On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato <peter@spinato.ca> wrote:* ***
All,****
Hopefully someone can assist me - I'll gladly give $50 to the person who helps me fix the issue - I have an OpenSIPs server configured as a load balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) that receives the call and forwards it to my Freeswitch Server for an IVR.. When I had the call just routing to the Freeswitch server I got the audio working by setting the ext-rtp-ip to the public IP. Now that I route the SIP call through the OpenSIPs server there is no audio - I'm guessing it a NAT issue as always.****
****
Both the OpenSIPS and FreeSwitch server have an internal private IP - OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2****
Both servers also have a public IP that routes to the internal IP OpensIPS = 47.1.1.1 and Freeswitch = 47.1.1.2****
****
Call gets received by the OpenSIPs via the external IP - routes the call to the internal IP on the Freeswitch server which answers the call - but no audio - I think the FreeSwitch is trying to route the RTP Audio via its internal private IP instead of the public IP of 47.1.1.2. Not sure if this is the real issue or how to configure it route RTP properly ... all help is appreciated. $50 Paypal to whoever fixes this for me! Thanks****
****
-Peter****
_________________________________________________________________________ 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****
** ** ------------------------------
No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11* ***
_________________________________________________________________________ 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
I changed the external profile to port 5060 - I can see it using that profile and it still sends out the local IP at the RTP IP . even though ext-rtp-ip is set to the public one .? From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Saturday, December 17, 2011 10:32 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue Yes, send it to the freeswitch server port 5080 Or if you don't need the internal profile, remove it and set external to use port 5060. -Avi On Sun, Dec 18, 2011 at 5:07 AM, Peter Spinato <peter@spinato.ca> wrote: The calls go to internal profile - I guess case the OpenSIPs server is local lan - is there a way to force that calls to external profile as I find calls that hit that profile load the ext-rtp-ip ip and work. From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Saturday, December 17, 2011 11:07 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue Someone familiar with this might be able to answer right off the bat, but if you pastebin a SIP trace (and an fs_cli log for completeness) , the problem should become apparent. Are you using the internal profile for all the calls? If you use the external.xml profile, you might need to set your ext-rtp-ip in that file, too. Seeing a trace will tell (almost) the whole story. -Avi On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato <peter@spinato.ca> wrote: All, Hopefully someone can assist me - I'll gladly give $50 to the person who helps me fix the issue - I have an OpenSIPs server configured as a load balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) that receives the call and forwards it to my Freeswitch Server for an IVR. When I had the call just routing to the Freeswitch server I got the audio working by setting the ext-rtp-ip to the public IP. Now that I route the SIP call through the OpenSIPs server there is no audio - I'm guessing it a NAT issue as always. Both the OpenSIPS and FreeSwitch server have an internal private IP - OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2 Both servers also have a public IP that routes to the internal IP OpensIPS = 47.1.1.1 and Freeswitch = 47.1.1.2 Call gets received by the OpenSIPs via the external IP - routes the call to the internal IP on the Freeswitch server which answers the call - but no audio - I think the FreeSwitch is trying to route the RTP Audio via its internal private IP instead of the public IP of 47.1.1.2. Not sure if this is the real issue or how to configure it route RTP properly ... all help is appreciated. $50 Paypal to whoever fixes this for me! Thanks -Peter _________________________________________________________________________ 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 _____ No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11 _________________________________________________________________________ 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 _____ No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11
Hey, I think you need to use RTP proxy in bridged mode on OpenSIPS and use the force_rtp_proxy() function with IE and EI flags so that the SDPs IPs change as follows End-USER<===========>[*Public-IP*]OPENSIPS/RTPProxy[*Private-IP* ]<===========>[*Private-IP*]FreeSWITCH Use FreeSWITCH internal or external domain, make sure you've correct properties and contexts set. Regards, Sammy. On Mon, Dec 19, 2011 at 5:28 AM, Peter Spinato <peter@spinato.ca> wrote:
I changed the external profile to port 5060 – I can see it using that profile and it still sends out the local IP at the RTP IP … even though ext-rtp-ip is set to the public one …?****
** **
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Avi Marcus *Sent:* Saturday, December 17, 2011 10:32 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue****
** **
Yes, send it to the freeswitch server port 5080****
Or if you don't need the internal profile, remove it and set external to use port 5060. ****
-Avi****
** **
On Sun, Dec 18, 2011 at 5:07 AM, Peter Spinato <peter@spinato.ca> wrote:** **
The calls go to internal profile – I guess case the OpenSIPs server is local lan – is there a way to force that calls to external profile as I find calls that hit that profile load the ext-rtp-ip ip and work.****
****
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Avi Marcus *Sent:* Saturday, December 17, 2011 11:07 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue****
****
Someone familiar with this might be able to answer right off the bat, but if you pastebin a SIP trace (and an fs_cli log for completeness) , the problem should become apparent.****
****
Are you using the internal profile for all the calls? If you use the external.xml profile, you might need to set your ext-rtp-ip in that file, too. Seeing a trace will tell (almost) the whole story.****
****
-Avi****
****
On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato <peter@spinato.ca> wrote:* ***
All,****
Hopefully someone can assist me - I'll gladly give $50 to the person who helps me fix the issue - I have an OpenSIPs server configured as a load balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) that receives the call and forwards it to my Freeswitch Server for an IVR. When I had the call just routing to the Freeswitch server I got the audio working by setting the ext-rtp-ip to the public IP. Now that I route the SIP call through the OpenSIPs server there is no audio - I'm guessing it a NAT issue as always.****
****
Both the OpenSIPS and FreeSwitch server have an internal private IP - OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2****
Both servers also have a public IP that routes to the internal IP OpensIPS = 47.1.1.1 and Freeswitch = 47.1.1.2****
****
Call gets received by the OpenSIPs via the external IP - routes the call to the internal IP on the Freeswitch server which answers the call - but no audio - I think the FreeSwitch is trying to route the RTP Audio via its internal private IP instead of the public IP of 47.1.1.2. Not sure if this is the real issue or how to configure it route RTP properly ... all help is appreciated. $50 Paypal to whoever fixes this for me! Thanks****
****
-Peter****
_________________________________________________________________________ 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****
**** ------------------------------
No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11* ***
_________________________________________________________________________ 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****
** ** ------------------------------
No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11* ***
_________________________________________________________________________ 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
If that doesn't help, can you provide an actual siptrace of the call so we can SEE what's going on? ngrep can be quite helpful.. -Avi On Mon, Dec 19, 2011 at 6:41 AM, Sammy Govind <govoiper@gmail.com> wrote:
Hey,
I think you need to use RTP proxy in bridged mode on OpenSIPS and use the force_rtp_proxy() function with IE and EI flags so that the SDPs IPs change as follows
End-USER<===========>[*Public-IP*]OPENSIPS/RTPProxy[*Private-IP* ]<===========>[*Private-IP*]FreeSWITCH
Use FreeSWITCH internal or external domain, make sure you've correct properties and contexts set.
Regards, Sammy.
On Mon, Dec 19, 2011 at 5:28 AM, Peter Spinato <peter@spinato.ca> wrote:
I changed the external profile to port 5060 – I can see it using that profile and it still sends out the local IP at the RTP IP … even though ext-rtp-ip is set to the public one …?****
** **
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Avi Marcus *Sent:* Saturday, December 17, 2011 10:32 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue****
** **
Yes, send it to the freeswitch server port 5080****
Or if you don't need the internal profile, remove it and set external to use port 5060. ****
-Avi****
** **
On Sun, Dec 18, 2011 at 5:07 AM, Peter Spinato <peter@spinato.ca> wrote:* ***
The calls go to internal profile – I guess case the OpenSIPs server is local lan – is there a way to force that calls to external profile as I find calls that hit that profile load the ext-rtp-ip ip and work.****
****
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Avi Marcus *Sent:* Saturday, December 17, 2011 11:07 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue****
****
Someone familiar with this might be able to answer right off the bat, but if you pastebin a SIP trace (and an fs_cli log for completeness) , the problem should become apparent.****
****
Are you using the internal profile for all the calls? If you use the external.xml profile, you might need to set your ext-rtp-ip in that file, too. Seeing a trace will tell (almost) the whole story.****
****
-Avi****
****
On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato <peter@spinato.ca> wrote: ****
All,****
Hopefully someone can assist me - I'll gladly give $50 to the person who helps me fix the issue - I have an OpenSIPs server configured as a load balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) that receives the call and forwards it to my Freeswitch Server for an IVR. When I had the call just routing to the Freeswitch server I got the audio working by setting the ext-rtp-ip to the public IP. Now that I route the SIP call through the OpenSIPs server there is no audio - I'm guessing it a NAT issue as always.****
****
Both the OpenSIPS and FreeSwitch server have an internal private IP - OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2****
Both servers also have a public IP that routes to the internal IP OpensIPS = 47.1.1.1 and Freeswitch = 47.1.1.2****
****
Call gets received by the OpenSIPs via the external IP - routes the call to the internal IP on the Freeswitch server which answers the call - but no audio - I think the FreeSwitch is trying to route the RTP Audio via its internal private IP instead of the public IP of 47.1.1.2. Not sure if this is the real issue or how to configure it route RTP properly ... all help is appreciated. $50 Paypal to whoever fixes this for me! Thanks****
****
-Peter****
_________________________________________________________________________ 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****
**** ------------------------------
No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11 ****
_________________________________________________________________________ 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****
** ** ------------------------------
No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11 ****
_________________________________________________________________________ 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
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Monday, December 19, 2011 1:57 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue If that doesn't help, can you provide an actual siptrace of the call so we can SEE what's going on? ngrep can be quite helpful.. -Avi On Mon, Dec 19, 2011 at 6:41 AM, Sammy Govind <govoiper@gmail.com> wrote: Hey, I think you need to use RTP proxy in bridged mode on OpenSIPS and use the force_rtp_proxy() function with IE and EI flags so that the SDPs IPs change as follows End-USER<===========>[Public-IP]OPENSIPS/RTPProxy[Private-IP]<===========>[P rivate-IP]FreeSWITCH Use FreeSWITCH internal or external domain, make sure you've correct properties and contexts set. Regards, Sammy. On Mon, Dec 19, 2011 at 5:28 AM, Peter Spinato <peter@spinato.ca> wrote: I changed the external profile to port 5060 - I can see it using that profile and it still sends out the local IP at the RTP IP . even though ext-rtp-ip is set to the public one .? From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Saturday, December 17, 2011 10:32 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue Yes, send it to the freeswitch server port 5080 Or if you don't need the internal profile, remove it and set external to use port 5060. -Avi On Sun, Dec 18, 2011 at 5:07 AM, Peter Spinato <peter@spinato.ca> wrote: The calls go to internal profile - I guess case the OpenSIPs server is local lan - is there a way to force that calls to external profile as I find calls that hit that profile load the ext-rtp-ip ip and work. From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Saturday, December 17, 2011 11:07 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue Someone familiar with this might be able to answer right off the bat, but if you pastebin a SIP trace (and an fs_cli log for completeness) , the problem should become apparent. Are you using the internal profile for all the calls? If you use the external.xml profile, you might need to set your ext-rtp-ip in that file, too. Seeing a trace will tell (almost) the whole story. -Avi On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato <peter@spinato.ca> wrote: All, Hopefully someone can assist me - I'll gladly give $50 to the person who helps me fix the issue - I have an OpenSIPs server configured as a load balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) that receives the call and forwards it to my Freeswitch Server for an IVR. When I had the call just routing to the Freeswitch server I got the audio working by setting the ext-rtp-ip to the public IP. Now that I route the SIP call through the OpenSIPs server there is no audio - I'm guessing it a NAT issue as always. Both the OpenSIPS and FreeSwitch server have an internal private IP - OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2 Both servers also have a public IP that routes to the internal IP OpensIPS = 47.1.1.1 and Freeswitch = 47.1.1.2 Call gets received by the OpenSIPs via the external IP - routes the call to the internal IP on the Freeswitch server which answers the call - but no audio - I think the FreeSwitch is trying to route the RTP Audio via its internal private IP instead of the public IP of 47.1.1.2. Not sure if this is the real issue or how to configure it route RTP properly ... all help is appreciated. $50 Paypal to whoever fixes this for me! Thanks -Peter _________________________________________________________________________ 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 _____ No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11 _________________________________________________________________________ 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 _____ No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11 _________________________________________________________________________ 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 _____ No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1890 / Virus Database: 2108/4689 - Release Date: 12/18/11
192.168.23.33 is your FS box? So did you edit your new external.xml and set rtp-ip (and ext-rtp-ip?) to the public IP, and then reload FS or just mod_sofia, or just "sofia profile $profile_name restart" ? -Avi On Mon, Dec 19, 2011 at 2:12 PM, Peter Spinato <peter@spinato.ca> wrote:
** **
** **
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Avi Marcus *Sent:* Monday, December 19, 2011 1:57 AM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue****
** **
If that doesn't help, can you provide an actual siptrace of the call so we can SEE what's going on?****
ngrep can be quite helpful..****
****
-Avi****
** **
On Mon, Dec 19, 2011 at 6:41 AM, Sammy Govind <govoiper@gmail.com> wrote:* ***
Hey, ****
** **
I think you need to use RTP proxy in bridged mode on OpenSIPS and use the force_rtp_proxy() function with IE and EI flags so that the SDPs IPs change as follows****
** **
End-USER<===========>[*Public-IP*]OPENSIPS/RTPProxy[*Private-IP* ]<===========>[*Private-IP*]FreeSWITCH****
** **
Use FreeSWITCH internal or external domain, make sure you've correct properties and contexts set.****
** **
Regards,****
Sammy.****
** **
On Mon, Dec 19, 2011 at 5:28 AM, Peter Spinato <peter@spinato.ca> wrote:** **
I changed the external profile to port 5060 – I can see it using that profile and it still sends out the local IP at the RTP IP … even though ext-rtp-ip is set to the public one …?****
****
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Avi Marcus *Sent:* Saturday, December 17, 2011 10:32 PM****
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue****
****
Yes, send it to the freeswitch server port 5080****
Or if you don't need the internal profile, remove it and set external to use port 5060. ****
-Avi****
****
On Sun, Dec 18, 2011 at 5:07 AM, Peter Spinato <peter@spinato.ca> wrote:** **
The calls go to internal profile – I guess case the OpenSIPs server is local lan – is there a way to force that calls to external profile as I find calls that hit that profile load the ext-rtp-ip ip and work.****
****
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Avi Marcus *Sent:* Saturday, December 17, 2011 11:07 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue****
****
Someone familiar with this might be able to answer right off the bat, but if you pastebin a SIP trace (and an fs_cli log for completeness) , the problem should become apparent.****
****
Are you using the internal profile for all the calls? If you use the external.xml profile, you might need to set your ext-rtp-ip in that file, too. Seeing a trace will tell (almost) the whole story.****
****
-Avi****
****
On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato <peter@spinato.ca> wrote:* ***
All,****
Hopefully someone can assist me - I'll gladly give $50 to the person who helps me fix the issue - I have an OpenSIPs server configured as a load balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) that receives the call and forwards it to my Freeswitch Server for an IVR.. When I had the call just routing to the Freeswitch server I got the audio working by setting the ext-rtp-ip to the public IP. Now that I route the SIP call through the OpenSIPs server there is no audio - I'm guessing it a NAT issue as always.****
****
Both the OpenSIPS and FreeSwitch server have an internal private IP - OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2****
Both servers also have a public IP that routes to the internal IP OpensIPS = 47.1.1.1 and Freeswitch = 47.1.1.2****
****
Call gets received by the OpenSIPs via the external IP - routes the call to the internal IP on the Freeswitch server which answers the call - but no audio - I think the FreeSwitch is trying to route the RTP Audio via its internal private IP instead of the public IP of 47.1.1.2. Not sure if this is the real issue or how to configure it route RTP properly ... all help is appreciated. $50 Paypal to whoever fixes this for me! Thanks****
****
-Peter****
_________________________________________________________________________ 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****
**** ------------------------------
No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11* ***
_________________________________________________________________________ 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****
**** ------------------------------
No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11* ***
_________________________________________________________________________ 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****
** ** ------------------------------
No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1890 / Virus Database: 2108/4689 - Release Date: 12/18/11* ***
_________________________________________________________________________ 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'm having hard time understanding which one is FreeSWICTH and which on is openSIPS. Looking at the sip tarces however, I must say you should at-max have a one-way audio because the remote end-point is sending its Public Address for Media-connectivity, you freeswitch in return is sending its private IP to OpenSIPS and openSIPs is relaying the same Private-IP to the remote-end point... U 192.168.23.34:5060 -> 184.150.225.230:5062 SIP/2.0 200 OK. Via: SIP/2.0/UDP 184.150.225.230:5062;received=184.150.225.230;rport=5062;branch=z9hG4bK-4008200999-3775991592-3239888547-3704563422. Record-Route: <sip:192.168.23.34;lr;ftag=917850919-3775991592-3239888547-3704563422;did=5df.895f858>. From: <sip:14167125040@184.150.225.230:5062;user=phone>;tag=917850919-3775991592-3239888547-3704563422. To: <sip:16474272135@66.240.179.221;user=phone>;tag=9mr8cyc2pgjmS. Call-ID: 274bb6c6280f11e1a3c61cc1de26cfdc@184.150.225.230. CSeq: 1 INVITE. Contact: <sip:16474272135@192.168.23.33:5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f4320b5 2011-11-28 08-27-46 -0600. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 249. Remote-Party-ID: "16474272135" <sip:16474272135@66.240.179.221>;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1324008356 1324008357 IN IP4 *192.168.23.33*. s=FreeSWITCH. c=IN IP4 *192.168.23.33*. t=0 0. m=audio 49774 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. And you know the remote end-point is sending the media to provate ip of FS within its LAN...!! So, Like I said use RTPproxy module such that newer INVITE look Like below.. U 192.168.23.34:5060 -> 184.150.225.230:5062 SIP/2.0 200 OK. Via: SIP/2.0/UDP 184.150.225.230:5062;received=184.150.225.230;rport=5062;branch=z9hG4bK-4008200999-3775991592-3239888547-3704563422. Record-Route: <sip:192.168.23.34;lr;ftag=917850919-3775991592-3239888547-3704563422;did=5df.895f858>. From: <sip:14167125040@184.150.225.230:5062;user=phone>;tag=917850919-3775991592-3239888547-3704563422. To: <sip:16474272135@66.240.179.221;user=phone>;tag=9mr8cyc2pgjmS. Call-ID: 274bb6c6280f11e1a3c61cc1de26cfdc@184.150.225.230. CSeq: 1 INVITE. Contact: <sip:16474272135@192.168.23.33:5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f4320b5 2011-11-28 08-27-46 -0600. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 249. Remote-Party-ID: "16474272135" <sip:16474272135@66.240.179.221>;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1324008356 1324008357 IN IP4 *PU.BL.IC.IP of OpenSIPS*. s=FreeSWITCH. c=IN IP4 *PU.BL.IC.IP of OpenSIPS* . t=0 0. m=audio 49774 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. Wish you best of Luck.! Regards, Sammy On Mon, Dec 19, 2011 at 5:12 PM, Peter Spinato <peter@spinato.ca> wrote:
** **
** **
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Avi Marcus *Sent:* Monday, December 19, 2011 1:57 AM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue****
** **
If that doesn't help, can you provide an actual siptrace of the call so we can SEE what's going on?****
ngrep can be quite helpful..****
****
-Avi****
** **
On Mon, Dec 19, 2011 at 6:41 AM, Sammy Govind <govoiper@gmail.com> wrote:* ***
Hey, ****
** **
I think you need to use RTP proxy in bridged mode on OpenSIPS and use the force_rtp_proxy() function with IE and EI flags so that the SDPs IPs change as follows****
** **
End-USER<===========>[*Public-IP*]OPENSIPS/RTPProxy[*Private-IP* ]<===========>[*Private-IP*]FreeSWITCH****
** **
Use FreeSWITCH internal or external domain, make sure you've correct properties and contexts set.****
** **
Regards,****
Sammy.****
** **
On Mon, Dec 19, 2011 at 5:28 AM, Peter Spinato <peter@spinato.ca> wrote:** **
I changed the external profile to port 5060 – I can see it using that profile and it still sends out the local IP at the RTP IP … even though ext-rtp-ip is set to the public one …?****
****
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Avi Marcus *Sent:* Saturday, December 17, 2011 10:32 PM****
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue****
****
Yes, send it to the freeswitch server port 5080****
Or if you don't need the internal profile, remove it and set external to use port 5060. ****
-Avi****
****
On Sun, Dec 18, 2011 at 5:07 AM, Peter Spinato <peter@spinato.ca> wrote:** **
The calls go to internal profile – I guess case the OpenSIPs server is local lan – is there a way to force that calls to external profile as I find calls that hit that profile load the ext-rtp-ip ip and work.****
****
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Avi Marcus *Sent:* Saturday, December 17, 2011 11:07 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue****
****
Someone familiar with this might be able to answer right off the bat, but if you pastebin a SIP trace (and an fs_cli log for completeness) , the problem should become apparent.****
****
Are you using the internal profile for all the calls? If you use the external.xml profile, you might need to set your ext-rtp-ip in that file, too. Seeing a trace will tell (almost) the whole story.****
****
-Avi****
****
On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato <peter@spinato.ca> wrote:* ***
All,****
Hopefully someone can assist me - I'll gladly give $50 to the person who helps me fix the issue - I have an OpenSIPs server configured as a load balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) that receives the call and forwards it to my Freeswitch Server for an IVR. When I had the call just routing to the Freeswitch server I got the audio working by setting the ext-rtp-ip to the public IP. Now that I route the SIP call through the OpenSIPs server there is no audio - I'm guessing it a NAT issue as always.****
****
Both the OpenSIPS and FreeSwitch server have an internal private IP - OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2****
Both servers also have a public IP that routes to the internal IP OpensIPS = 47.1.1.1 and Freeswitch = 47.1.1.2****
****
Call gets received by the OpenSIPs via the external IP - routes the call to the internal IP on the Freeswitch server which answers the call - but no audio - I think the FreeSwitch is trying to route the RTP Audio via its internal private IP instead of the public IP of 47.1.1.2. Not sure if this is the real issue or how to configure it route RTP properly ... all help is appreciated. $50 Paypal to whoever fixes this for me! Thanks****
****
-Peter****
_________________________________________________________________________ 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****
**** ------------------------------
No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11* ***
_________________________________________________________________________ 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****
**** ------------------------------
No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11* ***
_________________________________________________________________________ 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****
** ** ------------------------------
No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1890 / Virus Database: 2108/4689 - Release Date: 12/18/11* ***
_________________________________________________________________________ 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 (3)
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Avi Marcus
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Peter Spinato
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Sammy Govind