Configure Freeswitch as an IMS application server
Hello, I installed Freeswitch 1.5.7 and am trying to configure it as an telephony application server in an IMS network. Here is the high-level signaling flow that I am considering. 1. Two SIP users (User-A and User-B) registered with S-CSCF in the IMS network (not with Freeswitch) 2. User-A makes a voice call to User-B. 3. S-CSCF forwards the SIP INVITE to Freeswitch. 4. Freeswitch acting as a B2BUA changes SDP to anchor the media flow and sends SIP INVITE to User-B back to S-CSCF. 5. SIP 200 OK from User-B follows the same path as INVITE in reverse direction. I went through the dial-plan documentation and could not figure out whether such a configuration is possible. Please let me know whether such a configuration is feasible and any broad pointers on how to make it work would be appreciated. Thanks, Thomas
In the dialplan you just need to have FS send the invite back to the S-CSCF for user B... This can be done a couple of ways, 1 is specify the URI for the user@s-cscf or to specify to route it via the s-cscf there are a few ways to choose from on this... See http://wiki.freeswitch.org/wiki/Variable_sip_network_destination for 1 way with links to others On 11/20/13 11:04 PM, "Thomas Titty" <ttitty@kineto.com> wrote:
Hello,
I installed Freeswitch 1.5.7 and am trying to configure it as an telephony application server in an IMS network.
Here is the high-level signaling flow that I am considering.
1. Two SIP users (User-A and User-B) registered with S-CSCF in the IMS network (not with Freeswitch)
2. User-A makes a voice call to User-B.
3. S-CSCF forwards the SIP INVITE to Freeswitch.
4. Freeswitch acting as a B2BUA changes SDP to anchor the media flow and sends SIP INVITE to User-B back to S-CSCF.
5. SIP 200 OK from User-B follows the same path as INVITE in reverse direction.
I went through the dial-plan documentation and could not figure out whether such a configuration is possible.
Please let me know whether such a configuration is feasible and any broad pointers on how to make it work would be appreciated.
Thanks, Thomas
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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I tried all of the variables (sip_network_destination, sip_route_uri, sip_invite_req_uri) that you suggested but the SIP INVITE is not being routed to the p-cscf. I have 2 sip users (say) user-a@example.com<mailto:user-a@example.com> and user-b@example.com<mailto:user-b@example.com>, and P-CSCF at p-cscf.com I would like the SIP INVITE from FS to look like INVITE sip:user-b@example.com SIP/2.0 From: "user-a" <sip:user-a@example.com>;tag=a7c50F74399yc To: <sip:userba@p-cscf.com:5060> Here is what my dial-plan looks like <extension name="pcscf_outbound"> <condition field="destination_number" expression="^(\d{10})$"> <action application="bridge" data="{ sip_network_destination=$1@example.com,sip_route_uri=sip:$1@example.com,sip_invite,sip_invite_req_uri=sip:$1@example.com}sofia/external/$1@p-cscf.com:5060"/> </condition> </extension> With this setting the SIP INVITE looks okay, but is being sent to example.com rather than p-cscf.com. From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, 21 November 2013 10:12 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Configure Freeswitch as an IMS application server In the dialplan you just need to have FS send the invite back to the S-CSCF for user B... This can be done a couple of ways, 1 is specify the URI for the user@s-cscf or to specify to route it via the s-cscf there are a few ways to choose from on this... See http://wiki.freeswitch.org/wiki/Variable_sip_network_destination for 1 way with links to others On 11/20/13 11:04 PM, "Thomas Titty" <ttitty@kineto.com> wrote: Hello, I installed Freeswitch 1.5.7 and am trying to configure it as an telephony application server in an IMS network. Here is the high-level signaling flow that I am considering. 1. Two SIP users (User-A and User-B) registered with S-CSCF in the IMS network (not with Freeswitch) 2. User-A makes a voice call to User-B. 3. S-CSCF forwards the SIP INVITE to Freeswitch. 4. Freeswitch acting as a B2BUA changes SDP to anchor the media flow and sends SIP INVITE to User-B back to S-CSCF. 5. SIP 200 OK from User-B follows the same path as INVITE in reverse direction. I went through the dial-plan documentation and could not figure out whether such a configuration is possible. Please let me know whether such a configuration is feasible and any broad pointers on how to make it work would be appreciated. Thanks, Thomas ________________________________ _________________________________________________________________________ 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch
https://wiki.freeswitch.org/wiki/Sofia-SIP#Specifying_SIP_Proxy_With_fs_path On Nov 21, 2013, at 3:58 PM, Thomas Titty <ttitty@kineto.com> wrote:
I tried all of the variables (sip_network_destination, sip_route_uri, sip_invite_req_uri) that you suggested but the SIP INVITE is not being routed to the p-cscf.
I have 2 sip users (say) user-a@example.com and user-b@example.com, and P-CSCF at p-cscf.com
I would like the SIP INVITE from FS to look like
INVITE sip:user-b@example.com SIP/2.0 From: "user-a" <sip:user-a@example.com>;tag=a7c50F74399yc To: <sip:userba@p-cscf.com:5060>
Here is what my dial-plan looks like
<extension name="pcscf_outbound"> <condition field="destination_number" expression="^(\d{10})$"> <action application="bridge" data="{sip_network_destination=$1@example.com,sip_route_uri=sip:$1@example.com,sip_invite,sip_invite_req_uri=sip:$1@example.com}sofia/external/$1@p-cscf.com:5060"/> </condition> </extension>
With this setting the SIP INVITE looks okay, but is being sent to example.com rather than p-cscf.com.
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf OfKen Rice Sent: Thursday, 21 November 2013 10:12 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Configure Freeswitch as an IMS application server
In the dialplan you just need to have FS send the invite back to the S-CSCF for user B... This can be done a couple of ways, 1 is specify the URI for the user@s-cscf or to specify to route it via the s-cscf there are a few ways to choose from on this... See http://wiki.freeswitch.org/wiki/Variable_sip_network_destination for 1 way with links to others
On 11/20/13 11:04 PM, "Thomas Titty" <ttitty@kineto.com> wrote:
Hello,
I installed Freeswitch 1.5.7 and am trying to configure it as an telephony application server in an IMS network.
Here is the high-level signaling flow that I am considering.
1. Two SIP users (User-A and User-B) registered with S-CSCF in the IMS network (not with Freeswitch)
2. User-A makes a voice call to User-B.
3. S-CSCF forwards the SIP INVITE to Freeswitch.
4. Freeswitch acting as a B2BUA changes SDP to anchor the media flow and sends SIP INVITE to User-B back to S-CSCF.
5. SIP 200 OK from User-B follows the same path as INVITE in reverse direction.
I went through the dial-plan documentation and could not figure out whether such a configuration is possible.
Please let me know whether such a configuration is feasible and any broad pointers on how to make it work would be appreciated.
Thanks, Thomas
_________________________________________________________________________ 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
-- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch _________________________________________________________________________ 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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Ken Rice
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Michael Jerris
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Thomas Titty