SRTP call between two extensions.
Hello, everyone! I'm having trouble with SRTP calls between two internal extensions. Whenever I configure the phones to require SRTP calls fail and the freeswitch console shows: [INFO] mod_dptools.c:3201 Originate Failed. Cause: INCOMPATIBLE_DESTINATION If I select "allow SRTP" so that encryption is not mandatory I get encryption on a single leg in a call. Here's an example of what's happening: Extension 100 -- SRTP --> FreeSWITCH -- RTP --> 102 (call success) Extension 100 -- RTP --> FreeSWITCH -- SRTP --> 102 (call success) Extension 100 -- SRTP --> FreeSWITCH -- SRTP --> 102 (call fail: INCOMPATIBLE_DESTINATION) I've setup the sip_secure_media=true variable as instructed from https://wiki.freeswitch.org/wiki/Secure_RTP but it's still not working. Any help is greatly appreciated! Kind regards, -- Iskren Hadzhinedev
which FS version is this? we've had this problem with 1.5 tree. On Fri, Jan 10, 2014 at 1:26 PM, Iskren Hadzhinedev < iskren.hadzhinedev@ikiji.com> wrote:
Hello, everyone!
I'm having trouble with SRTP calls between two internal extensions.
Whenever I configure the phones to require SRTP calls fail and the freeswitch console shows:
[INFO] mod_dptools.c:3201 Originate Failed. Cause: INCOMPATIBLE_DESTINATION
If I select "allow SRTP" so that encryption is not mandatory I get encryption on a single leg in a call.
Here's an example of what's happening:
Extension 100 -- SRTP --> FreeSWITCH -- RTP --> 102 (call success)
Extension 100 -- RTP --> FreeSWITCH -- SRTP --> 102 (call success)
Extension 100 -- SRTP --> FreeSWITCH -- SRTP --> 102 (call fail: INCOMPATIBLE_DESTINATION)
I've setup the sip_secure_media=true variable as instructed from
https://wiki.freeswitch.org/wiki/Secure_RTP
but it's still not working. Any help is greatly appreciated!
Kind regards,
--
Iskren Hadzhinedev
_________________________________________________________________________ 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
Hello Daniel, Here's the version according to freeswitch: freeswitch@internal> version FreeSWITCH Version 1.5.8b+git~20131210T191701Z~9ec1e86574~64bit (git 9ec1e86 2013-12-10 19:17:01Z 64bit). I suppose I should build the 1.2 branch then? Thanks, -- Iskren Hadzhinedev System Administrator The Idea Factory | 20 Mearns Street | Aberdeen | AB11 5AT | UK T: 01224 607500 VAT Reg No: 982 4936 74. Company registered in Scotland, SC237116 On Friday 10 January 2014 18:15:30 Daniel Ivanov wrote:
which FS version is this? we've had this problem with 1.5 tree.
On Fri, Jan 10, 2014 at 1:26 PM, Iskren Hadzhinedev <
iskren.hadzhinedev@ikiji.com> wrote:
Hello, everyone!
I'm having trouble with SRTP calls between two internal extensions.
Whenever I configure the phones to require SRTP calls fail and the freeswitch console shows:
[INFO] mod_dptools.c:3201 Originate Failed. Cause: INCOMPATIBLE_DESTINATION
If I select "allow SRTP" so that encryption is not mandatory I get encryption on a single leg in a call.
Here's an example of what's happening:
Extension 100 -- SRTP --> FreeSWITCH -- RTP --> 102 (call success)
Extension 100 -- RTP --> FreeSWITCH -- SRTP --> 102 (call success)
Extension 100 -- SRTP --> FreeSWITCH -- SRTP --> 102 (call fail: INCOMPATIBLE_DESTINATION)
I've setup the sip_secure_media=true variable as instructed from
https://wiki.freeswitch.org/wiki/Secure_RTP
but it's still not working. Any help is greatly appreciated!
Kind regards,
--
Iskren Hadzhinedev
_________________________________________________________________________ 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, Can you pastebin a console log with siptrace enabled when placing a call? Regards, António On Fri, 2014-01-10 at 19:50 +0200, Iskren Hadzhinedev wrote:
Hello Daniel,
Here's the version according to freeswitch:
freeswitch@internal> version
FreeSWITCH Version 1.5.8b+git~20131210T191701Z~9ec1e86574~64bit (git 9ec1e86 2013-12-10 19:17:01Z 64bit).
I suppose I should build the 1.2 branch then?
Thanks,
--
Iskren Hadzhinedev
System Administrator
The Idea Factory | 20 Mearns Street | Aberdeen | AB11 5AT | UK
T: 01224 607500
VAT Reg No: 982 4936 74. Company registered in Scotland, SC237116
On Friday 10 January 2014 18:15:30 Daniel Ivanov wrote:
which FS version is this? we've had this problem with 1.5 tree.
On Fri, Jan 10, 2014 at 1:26 PM, Iskren Hadzhinedev <
iskren.hadzhinedev@ikiji.com> wrote:
Hello, everyone!
I'm having trouble with SRTP calls between two internal extensions.
Whenever I configure the phones to require SRTP calls fail and the
freeswitch console shows:
[INFO] mod_dptools.c:3201 Originate Failed. Cause:
INCOMPATIBLE_DESTINATION
If I select "allow SRTP" so that encryption is not mandatory I get
encryption on a single leg in a call.
Here's an example of what's happening:
Extension 100 -- SRTP --> FreeSWITCH -- RTP --> 102 (call success)
Extension 100 -- RTP --> FreeSWITCH -- SRTP --> 102 (call success)
Extension 100 -- SRTP --> FreeSWITCH -- SRTP --> 102 (call fail:
INCOMPATIBLE_DESTINATION)
I've setup the sip_secure_media=true variable as instructed from
but it's still not working. Any help is greatly appreciated!
Kind regards,
--
Iskren Hadzhinedev
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
Official FreeSWITCH Sites
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
_________________________________________________________________________ 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
You need to export rtp_secure_media=true On Friday, January 10, 2014, Iskren Hadzhinedev wrote:
Hello, everyone!
I'm having trouble with SRTP calls between two internal extensions.
Whenever I configure the phones to require SRTP calls fail and the freeswitch console shows:
[INFO] mod_dptools.c:3201 Originate Failed. Cause: INCOMPATIBLE_DESTINATION
If I select "allow SRTP" so that encryption is not mandatory I get encryption on a single leg in a call.
Here's an example of what's happening:
Extension 100 -- SRTP --> FreeSWITCH -- RTP --> 102 (call success)
Extension 100 -- RTP --> FreeSWITCH -- SRTP --> 102 (call success)
Extension 100 -- SRTP --> FreeSWITCH -- SRTP --> 102 (call fail: INCOMPATIBLE_DESTINATION)
I've setup the sip_secure_media=true variable as instructed from
https://wiki.freeswitch.org/wiki/Secure_RTP
but it's still not working. Any help is greatly appreciated!
Kind regards,
--
Iskren Hadzhinedev
-- Sent from mobile device
Yeah, try kristian's suggestion first. At some point that variable changed. This happened to me last week and turned out my colleagues checked out the 1.5 branch instead of 1.2stable. On Fri, Jan 10, 2014 at 7:14 PM, Kristian Kielhofner <kris@kriskinc.com>wrote:
You need to export rtp_secure_media=true
On Friday, January 10, 2014, Iskren Hadzhinedev wrote:
Hello, everyone!
I'm having trouble with SRTP calls between two internal extensions.
Whenever I configure the phones to require SRTP calls fail and the freeswitch console shows:
[INFO] mod_dptools.c:3201 Originate Failed. Cause: INCOMPATIBLE_DESTINATION
If I select "allow SRTP" so that encryption is not mandatory I get encryption on a single leg in a call.
Here's an example of what's happening:
Extension 100 -- SRTP --> FreeSWITCH -- RTP --> 102 (call success)
Extension 100 -- RTP --> FreeSWITCH -- SRTP --> 102 (call success)
Extension 100 -- SRTP --> FreeSWITCH -- SRTP --> 102 (call fail: INCOMPATIBLE_DESTINATION)
I've setup the sip_secure_media=true variable as instructed from
https://wiki.freeswitch.org/wiki/Secure_RTP
but it's still not working. Any help is greatly appreciated!
Kind regards,
--
Iskren Hadzhinedev
-- Sent from mobile device
_________________________________________________________________________ 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 (4)
-
Antonio
-
Daniel Ivanov
-
Iskren Hadzhinedev
-
Kristian Kielhofner