Conectin a SPA3102 as pstn gateway on FS
Hi Everyone, I'm trying to use a SPA3102 as a PSTN gateway, it is in a remote site with a dinamic IP, I'have it registered in my FS as 100123 user, and my goal is to dial <any pstn number>@<my spa3102> in order to use my SPA3120 to make the call via his pstn line to the number <any pstn number>. The problem is that all the examples I see has a fixed and well know IP address for the SPA3102 gateway in order to use it in the dial plan, but in my scenario the IP address for my SPA3102 gateway is dinamyc and unknow. Please any help in order to understand how can I dial any number to my spa3102 gateway will by apreciated. In this example my FS has the IP 100.10.10.10 and the SPA3102 has the dinamic Adsl IP 210.20.20.20
sofia status profile internal reg Registrations: ================================================================================================= Call-ID: 2378632f-eb1116d@192.168.0.11 User: 100123@100.10.10.10 Contact: "Local Call" <sip:100123@210.20.20.20:5061> Agent: Linksys/SPA3000-3.1.20(GW) Status: Registered(UDP)(unknown) EXP(2015-08-08 04:01:04) EXPSECS(61) Ping-Status: Reachable Host: voip.example.com IP: 210.20.20.20 Port: 5061 Auth-User: 100123 Auth-Realm: voip.example.com MWI-Account: 100123@100.10.10.10 =================================================================================================
for example, if I want to dial to the pstn number 222-1234, then I need to send an INVITE 2221234@210.20.20.20:5061 Thanks for your help. -- Jaime Cruz jaimecm@gmail.com https://www.facebook.com/jaimecruzmendoza
Register gateway as regular user And try dial as originate {sip_invite_to_uri=<sip:${destination_number}@mydomain.org>}user/ reg_user@mydomain.org & park or <action application="bridge" data="{sip_invite_to_uri=<sip:${ destination_number}@mydomain.org>}user/reg_user@mydomain.org"/> On Sat, Aug 8, 2015 at 10:14 AM, Jaime Cruz <jaimecm@gmail.com> wrote:
Hi Everyone,
I'm trying to use a SPA3102 as a PSTN gateway, it is in a remote site with a dinamic IP, I'have it registered in my FS as 100123 user, and my goal is to dial <any pstn number>@<my spa3102> in order to use my SPA3120 to make the call via his pstn line to the number <any pstn number>.
The problem is that all the examples I see has a fixed and well know IP address for the SPA3102 gateway in order to use it in the dial plan, but in my scenario the IP address for my SPA3102 gateway is dinamyc and unknow.
Please any help in order to understand how can I dial any number to my spa3102 gateway will by apreciated.
In this example my FS has the IP 100.10.10.10 and the SPA3102 has the dinamic Adsl IP 210.20.20.20
sofia status profile internal reg Registrations:
================================================================================================= Call-ID: 2378632f-eb1116d@192.168.0.11 User: 100123@100.10.10.10 Contact: "Local Call" <sip:100123@210.20.20.20:5061> Agent: Linksys/SPA3000-3.1.20(GW) Status: Registered(UDP)(unknown) EXP(2015-08-08 04:01:04) EXPSECS(61) Ping-Status: Reachable Host: voip.example.com IP: 210.20.20.20 Port: 5061 Auth-User: 100123 Auth-Realm: voip.example.com MWI-Account: 100123@100.10.10.10
=================================================================================================
for example, if I want to dial to the pstn number 222-1234, then I need to send an INVITE 2221234@210.20.20.20:5061
Thanks for your help.
-- Jaime Cruz jaimecm@gmail.com https://www.facebook.com/jaimecruzmendoza
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Why would you ever need to set the sip_invite_to_uri? On Sun, Aug 9, 2015 at 4:50 AM, Sergey Safarov <s.safarov@gmail.com> wrote:
Register gateway as regular user And try dial as
originate {sip_invite_to_uri=<sip:${destination_number}@mydomain.org
}user/reg_user@mydomain.org & park
or
<action application="bridge" data="{sip_invite_to_uri=<sip:${ destination_number}@mydomain.org>}user/reg_user@mydomain.org"/>
On Sat, Aug 8, 2015 at 10:14 AM, Jaime Cruz <jaimecm@gmail.com> wrote:
Hi Everyone,
I'm trying to use a SPA3102 as a PSTN gateway, it is in a remote site with a dinamic IP, I'have it registered in my FS as 100123 user, and my goal is to dial <any pstn number>@<my spa3102> in order to use my SPA3120 to make the call via his pstn line to the number <any pstn number>.
The problem is that all the examples I see has a fixed and well know IP address for the SPA3102 gateway in order to use it in the dial plan, but in my scenario the IP address for my SPA3102 gateway is dinamyc and unknow.
Please any help in order to understand how can I dial any number to my spa3102 gateway will by apreciated.
In this example my FS has the IP 100.10.10.10 and the SPA3102 has the dinamic Adsl IP 210.20.20.20
sofia status profile internal reg Registrations:
================================================================================================= Call-ID: 2378632f-eb1116d@192.168.0.11 User: 100123@100.10.10.10 Contact: "Local Call" <sip:100123@210.20.20.20:5061> Agent: Linksys/SPA3000-3.1.20(GW) Status: Registered(UDP)(unknown) EXP(2015-08-08 04:01:04) EXPSECS(61) Ping-Status: Reachable Host: voip.example.com IP: 210.20.20.20 Port: 5061 Auth-User: 100123 Auth-Realm: voip.example.com MWI-Account: 100123@100.10.10.10
=================================================================================================
for example, if I want to dial to the pstn number 222-1234, then I need to send an INVITE 2221234@210.20.20.20:5061
Thanks for your help.
-- Jaime Cruz jaimecm@gmail.com https://www.facebook.com/jaimecruzmendoza
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
Also, at least in mine, you have to use port 5061 to dial out. Brian West <brian@freeswitch.org> wrote:
Why would you ever need to set the sip_invite_to_uri?
On Sun, Aug 9, 2015 at 4:50 AM, Sergey Safarov <s.safarov@gmail.com> wrote:
Register gateway as regular user And try dial as
originate {sip_invite_to_uri=<sip:${destination_number}@mydomain.org
}user/reg_user@mydomain.org & park
or
<action application="bridge" data="{sip_invite_to_uri=<sip:${ destination_number}@mydomain.org>}user/reg_user@mydomain.org"/>
On Sat, Aug 8, 2015 at 10:14 AM, Jaime Cruz <jaimecm@gmail.com> wrote:
Hi Everyone,
I'm trying to use a SPA3102 as a PSTN gateway, it is in a remote site with a dinamic IP, I'have it registered in my FS as 100123 user, and my goal is to dial <any pstn number>@<my spa3102> in order to use my SPA3120 to make the call via his pstn line to the number <any pstn number>.
The problem is that all the examples I see has a fixed and well know IP address for the SPA3102 gateway in order to use it in the dial plan, but in my scenario the IP address for my SPA3102 gateway is dinamyc and unknow.
Please any help in order to understand how can I dial any number to my spa3102 gateway will by apreciated.
In this example my FS has the IP 100.10.10.10 and the SPA3102 has the dinamic Adsl IP 210.20.20.20
sofia status profile internal reg Registrations:
================================================================================================= Call-ID: 2378632f-eb1116d@192.168.0.11 User: 100123@100.10.10.10 Contact: "Local Call" <sip:100123@210.20.20.20:5061> Agent: Linksys/SPA3000-3.1.20(GW) Status: Registered(UDP)(unknown) EXP(2015-08-08 04:01:04) EXPSECS(61) Ping-Status: Reachable Host: voip.example.com IP: 210.20.20.20 Port: 5061 Auth-User: 100123 Auth-Realm: voip.example.com MWI-Account: 100123@100.10.10.10
=================================================================================================
for example, if I want to dial to the pstn number 222-1234, then I need to send an INVITE 2221234@210.20.20.20:5061
Thanks for your help.
-- Jaime Cruz jaimecm@gmail.com https://www.facebook.com/jaimecruzmendoza
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
---------------------------------------------------- Alternatives:
---------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it?
John Covici covici@ccs.covici.com
Hi Sergey, Thanks for your help, it' worked to change the Invite "To:" Uri, but as I tested, the linksys SPA3000&3102 is not taken the destination from the To Uri nor from the Invite Uri as "INVITE sip:nnnnnn@200.200.200.200:5061 SIP/2.0" as "nnnnnn" can be any destination that I want to dial via it's PSTN line. Do you think that there's an option to change this in FS, Thanks for your help. JCM ====== originate: {sip_invite_to_uri=<sip:226560600@voip.hablaip.com>}user/ 100123@voip.hablaip.com ====== INVITE sip:100123@200.200.200.200:5061 SIP/2.0 Via: SIP/2.0/UDP 100.100.100.100;rport;branch=z9hG4bK32ScZ5j0X2NQH Max-Forwards: 69 From: "Katty Casa." <sip:Katty Casa@voip.hablaip.com>;tag=XZvcHv1avcZ5c To: <sip:226560600@voip.hablaip.com> Call-ID: 2b0f621f-b9c0-1233-ab8f-000c292ac960 CSeq: 79255480 INVITE Contact: <sip:mod_sofia@100.100.100.100:5060> ====== 2015-08-10 02:04:49.318695 [DEBUG] switch_ivr_originate.c:2101 Parsing global variables 2015-08-10 02:04:49.318695 [DEBUG] switch_event.c:1698 Parsing variable [sip_invite_to_uri]=[<sip:226560600@voip.hablaip.com>] ========== On Sun, Aug 9, 2015 at 11:43 AM, Sergey Safarov <s.safarov@gmail.com> wrote:
Special assignment of used port is not required Using dial-string like "user/reg_user@mydomain.org" automatically do it
Sergey
On Sun, Aug 9, 2015 at 1:34 PM, <covici@ccs.covici.com> wrote:
Also, at least in mine, you have to use port 5061 to dial out.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
-- Jaime Cruz jaimecm@gmail.com https://www.facebook.com/jaimecruzmendoza
Use sip_invite_req_uri https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables On Mon, Aug 10, 2015 at 8:26 AM, Jaime Cruz <jaimecm@gmail.com> wrote:
Hi Sergey, Thanks for your help, it' worked to change the Invite "To:" Uri,
but as I tested, the linksys SPA3000&3102 is not taken the destination from the To Uri nor from the Invite Uri as "INVITE sip:nnnnnn@200.200.200.200:5061 SIP/2.0" as "nnnnnn" can be any destination that I want to dial via it's PSTN line.
Do you think that there's an option to change this in FS,
Thanks for your help. JCM
====== originate: {sip_invite_to_uri=<sip:226560600@voip.hablaip.com>}user/ 100123@voip.hablaip.com ====== INVITE sip:100123@200.200.200.200:5061 SIP/2.0 Via: SIP/2.0/UDP 100.100.100.100;rport;branch=z9hG4bK32ScZ5j0X2NQH Max-Forwards: 69 From: "Katty Casa." <sip:Katty Casa@voip.hablaip.com>;tag=XZvcHv1avcZ5c To: <sip:226560600@voip.hablaip.com> Call-ID: 2b0f621f-b9c0-1233-ab8f-000c292ac960 CSeq: 79255480 INVITE Contact: <sip:mod_sofia@100.100.100.100:5060> ====== 2015-08-10 02:04:49.318695 [DEBUG] switch_ivr_originate.c:2101 Parsing global variables 2015-08-10 02:04:49.318695 [DEBUG] switch_event.c:1698 Parsing variable [sip_invite_to_uri]=[<sip:226560600@voip.hablaip.com>] ==========
On Sun, Aug 9, 2015 at 11:43 AM, Sergey Safarov <s.safarov@gmail.com> wrote:
Special assignment of used port is not required Using dial-string like "user/reg_user@mydomain.org" automatically do it
Sergey
On Sun, Aug 9, 2015 at 1:34 PM, <covici@ccs.covici.com> wrote:
Also, at least in mine, you have to use port 5061 to dial out.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
-- Jaime Cruz jaimecm@gmail.com https://www.facebook.com/jaimecruzmendoza
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Thanks Sergey for the help, I'll use the sip_route_uri & sip_invite_req_uri to manage my SPA3k gateway Thanks JCM On Mon, Aug 10, 2015 at 5:33 AM, Sergey Safarov <s.safarov@gmail.com> wrote:
Use sip_invite_req_uri https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables
On Mon, Aug 10, 2015 at 8:26 AM, Jaime Cruz <jaimecm@gmail.com> wrote:
Hi Sergey, Thanks for your help, it' worked to change the Invite "To:" Uri,
but as I tested, the linksys SPA3000&3102 is not taken the destination from the To Uri nor from the Invite Uri as "INVITE sip:nnnnnn@200.200.200.200:5061 SIP/2.0" as "nnnnnn" can be any destination that I want to dial via it's PSTN line.
Do you think that there's an option to change this in FS,
Thanks for your help. JCM
====== originate: {sip_invite_to_uri=<sip:226560600@voip.hablaip.com>}user/ 100123@voip.hablaip.com ====== INVITE sip:100123@200.200.200.200:5061 SIP/2.0 Via: SIP/2.0/UDP 100.100.100.100;rport;branch=z9hG4bK32ScZ5j0X2NQH Max-Forwards: 69 From: "Katty Casa." <sip:Katty Casa@voip.hablaip.com>;tag=XZvcHv1avcZ5c To: <sip:226560600@voip.hablaip.com> Call-ID: 2b0f621f-b9c0-1233-ab8f-000c292ac960 CSeq: 79255480 INVITE Contact: <sip:mod_sofia@100.100.100.100:5060> ====== 2015-08-10 02:04:49.318695 [DEBUG] switch_ivr_originate.c:2101 Parsing global variables 2015-08-10 02:04:49.318695 [DEBUG] switch_event.c:1698 Parsing variable [sip_invite_to_uri]=[<sip:226560600@voip.hablaip.com>] ==========
On Sun, Aug 9, 2015 at 11:43 AM, Sergey Safarov <s.safarov@gmail.com> wrote:
Special assignment of used port is not required Using dial-string like "user/reg_user@mydomain.org" automatically do it
Sergey
On Sun, Aug 9, 2015 at 1:34 PM, <covici@ccs.covici.com> wrote:
Also, at least in mine, you have to use port 5061 to dial out.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
-- Jaime Cruz jaimecm@gmail.com https://www.facebook.com/jaimecruzmendoza
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
-- Jaime Cruz jaimecm@gmail.com https://www.facebook.com/jaimecruzmendoza
Hi Brian, It's to use a linksys SPA3000&3102 as a gateway to the PSTN line, that this device can connect to. but us I tested this device is taken the phone number to dial from the invite sip uri (and not from the To Uri). Thanks JCM On Sun, Aug 9, 2015 at 7:20 AM, Brian West <brian@freeswitch.org> wrote:
Why would you ever need to set the sip_invite_to_uri?
On Sun, Aug 9, 2015 at 4:50 AM, Sergey Safarov <s.safarov@gmail.com> wrote:
Register gateway as regular user And try dial as
originate {sip_invite_to_uri=<sip:${destination_number}@mydomain.org
}user/reg_user@mydomain.org & park
or
<action application="bridge" data="{sip_invite_to_uri=<sip:${ destination_number}@mydomain.org>}user/reg_user@mydomain.org"/>
On Sat, Aug 8, 2015 at 10:14 AM, Jaime Cruz <jaimecm@gmail.com> wrote:
Hi Everyone,
I'm trying to use a SPA3102 as a PSTN gateway, it is in a remote site with a dinamic IP, I'have it registered in my FS as 100123 user, and my goal is to dial <any pstn number>@<my spa3102> in order to use my SPA3120 to make the call via his pstn line to the number <any pstn number>.
The problem is that all the examples I see has a fixed and well know IP address for the SPA3102 gateway in order to use it in the dial plan, but in my scenario the IP address for my SPA3102 gateway is dinamyc and unknow.
Please any help in order to understand how can I dial any number to my spa3102 gateway will by apreciated.
In this example my FS has the IP 100.10.10.10 and the SPA3102 has the dinamic Adsl IP 210.20.20.20
sofia status profile internal reg Registrations:
================================================================================================= Call-ID: 2378632f-eb1116d@192.168.0.11 User: 100123@100.10.10.10 Contact: "Local Call" <sip:100123@210.20.20.20:5061> Agent: Linksys/SPA3000-3.1.20(GW) Status: Registered(UDP)(unknown) EXP(2015-08-08 04:01:04) EXPSECS(61) Ping-Status: Reachable Host: voip.example.com IP: 210.20.20.20 Port: 5061 Auth-User: 100123 Auth-Realm: voip.example.com MWI-Account: 100123@100.10.10.10
=================================================================================================
for example, if I want to dial to the pstn number 222-1234, then I need to send an INVITE 2221234@210.20.20.20:5061
Thanks for your help.
-- Jaime Cruz jaimecm@gmail.com https://www.facebook.com/jaimecruzmendoza
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
-- Jaime Cruz jaimecm@gmail.com https://www.facebook.com/jaimecruzmendoza
participants (4)
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Brian West
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covici＠ccs.covici.com
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Jaime Cruz
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Sergey Safarov