FreeSwith does not setup secure call with leg B.
Hi All, I have setup FreeSwith PBX. I am facing an issue of not having end to end secure call. Caller sends INVITE request with SIPS in request URI and RTP/SAVP in SDP. But when FreeSwith forwards the request to caller, it is not using RTP/SAVP in SDP. I have followed the steps mentioned in WIKI ( https://wiki.freeswitch.org/wiki/SIP_TLS#Configuration) for TLS configuration and marked "exports sip_secure_media" true in "conf/dialplan/default.xml" file (as per https://wiki.freeswitch.org/wiki/SRTP). But still FreeSwitch does not use RTP/SAVP for leg-B. Also, I am bit confused with following condition in "default.xml" when wiki page suggests that late coded negotiation must NOT to be turned on. <!-- Since we have inbound-late-negotation on by default now the above behavior isn't the same so you have to do one extra step. --> <condition field="${endpoint_disposition}" expression="^(DELAYED NEGOTIATION)"/> <condition field="${switch_r_sdp}" expression="(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)" break="never"> <action application="set" data="sip_secure_media=true"/> <!-- Offer SRTP on outbound legs if we have it on inbound. --> <action application="export" data="sip_secure_media=true"/> </condition> By commenting/un-commenting "sip_secure_media=true" above, it did not worked. Any help with this would be greatly appreciated. -- Thanks CWM
You didn't mention what rev you're using, If you're in 1.4 then its rtp_secure_media, please see latest configs and extensive docs in vars.xml about srtp and all the nice knobs you can use to tweak it. On Tue, Sep 9, 2014 at 12:52 AM, Chandrakant Marathe <cwmarathe@gmail.com> wrote:
Hi All, I have setup FreeSwith PBX. I am facing an issue of not having end to end secure call. Caller sends INVITE request with SIPS in request URI and RTP/SAVP in SDP. But when FreeSwith forwards the request to caller, it is not using RTP/SAVP in SDP.
I have followed the steps mentioned in WIKI ( https://wiki.freeswitch.org/wiki/SIP_TLS#Configuration) for TLS configuration and marked "exports sip_secure_media" true in "conf/dialplan/default.xml" file (as per https://wiki.freeswitch.org/wiki/SRTP). But still FreeSwitch does not use RTP/SAVP for leg-B.
Also, I am bit confused with following condition in "default.xml" when wiki page suggests that late coded negotiation must NOT to be turned on.
<!-- Since we have inbound-late-negotation on by default now the above behavior isn't the same so you have to do one extra step. -->
<condition field="${endpoint_disposition}" expression="^(DELAYED NEGOTIATION)"/> <condition field="${switch_r_sdp}" expression="(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)" break="never"> <action application="set" data="sip_secure_media=true"/> <!-- Offer SRTP on outbound legs if we have it on inbound. --> <action application="export" data="sip_secure_media=true"/> </condition>
By commenting/un-commenting "sip_secure_media=true" above, it did not worked.
Any help with this would be greatly appreciated.
-- Thanks CWM
_________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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 *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
Sorry Brian, I should have told the version. My bad. I am using 1.2 stable release. When I hit "git branch -av", it outputs - * v1.2.stable 2b62885 fs_cli: fix compiler error on CentOS 6 caused by recent short uuid logging change And from FreeSwitch console, when I hit "version" command, it shows - FreeSWITCH Version 1.2.24+git~20140630T213113Z~2b62885f21~32bit (git 2b62885 2014-06-30 21:31:13Z 32bit) I went through "vars.xml" and found only one config related with "rtp_secure_media" and that to related with zrtp <X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/> Is there any more configuration to do in v1.2 so as to enable end-to-end secure call or shall I move to v1.4 beta? Thanks. -- CWM On Tue, Sep 9, 2014 at 9:04 PM, Brian West <brian@freeswitch.org> wrote:
You didn't mention what rev you're using, If you're in 1.4 then its rtp_secure_media, please see latest configs and extensive docs in vars.xml about srtp and all the nice knobs you can use to tweak it.
On Tue, Sep 9, 2014 at 12:52 AM, Chandrakant Marathe <cwmarathe@gmail.com> wrote:
Hi All, I have setup FreeSwith PBX. I am facing an issue of not having end to end secure call. Caller sends INVITE request with SIPS in request URI and RTP/SAVP in SDP. But when FreeSwith forwards the request to caller, it is not using RTP/SAVP in SDP.
I have followed the steps mentioned in WIKI ( https://wiki.freeswitch.org/wiki/SIP_TLS#Configuration) for TLS configuration and marked "exports sip_secure_media" true in "conf/dialplan/default.xml" file (as per https://wiki.freeswitch.org/wiki/SRTP). But still FreeSwitch does not use RTP/SAVP for leg-B.
Also, I am bit confused with following condition in "default.xml" when wiki page suggests that late coded negotiation must NOT to be turned on.
<!-- Since we have inbound-late-negotation on by default now the above behavior isn't the same so you have to do one extra step. -->
<condition field="${endpoint_disposition}" expression="^(DELAYED NEGOTIATION)"/> <condition field="${switch_r_sdp}" expression="(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)" break="never"> <action application="set" data="sip_secure_media=true"/> <!-- Offer SRTP on outbound legs if we have it on inbound. --> <action application="export" data="sip_secure_media=true"/> </condition>
By commenting/un-commenting "sip_secure_media=true" above, it did not worked.
Any help with this would be greatly appreciated.
-- Thanks CWM
_________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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
*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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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
Does it work on the latest 1.4 release with the config change he mentioned? On Sep 10, 2014, at 1:48 AM, Chandrakant Marathe <cwmarathe@gmail.com> wrote:
Sorry Brian, I should have told the version. My bad. I am using 1.2 stable release. When I hit "git branch -av", it outputs -
* v1.2.stable 2b62885 fs_cli: fix compiler error on CentOS 6 caused by recent short uuid logging change
And from FreeSwitch console, when I hit "version" command, it shows -
FreeSWITCH Version 1.2.24+git~20140630T213113Z~2b62885f21~32bit (git 2b62885 2014-06-30 21:31:13Z 32bit)
I went through "vars.xml" and found only one config related with "rtp_secure_media" and that to related with zrtp
<X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/>
Is there any more configuration to do in v1.2 so as to enable end-to-end secure call or shall I move to v1.4 beta?
Thanks. -- CWM
On Tue, Sep 9, 2014 at 9:04 PM, Brian West <brian@freeswitch.org> wrote: You didn't mention what rev you're using, If you're in 1.4 then its rtp_secure_media, please see latest configs and extensive docs in vars.xml about srtp and all the nice knobs you can use to tweak it.
On Tue, Sep 9, 2014 at 12:52 AM, Chandrakant Marathe <cwmarathe@gmail.com> wrote: Hi All, I have setup FreeSwith PBX. I am facing an issue of not having end to end secure call. Caller sends INVITE request with SIPS in request URI and RTP/SAVP in SDP. But when FreeSwith forwards the request to caller, it is not using RTP/SAVP in SDP.
I have followed the steps mentioned in WIKI (https://wiki.freeswitch.org/wiki/SIP_TLS#Configuration) for TLS configuration and marked "exports sip_secure_media" true in "conf/dialplan/default.xml" file (as per https://wiki.freeswitch.org/wiki/SRTP). But still FreeSwitch does not use RTP/SAVP for leg-B.
Also, I am bit confused with following condition in "default.xml" when wiki page suggests that late coded negotiation must NOT to be turned on.
<!-- Since we have inbound-late-negotation on by default now the above behavior isn't the same so you have to do one extra step. -->
<condition field="${endpoint_disposition}" expression="^(DELAYED NEGOTIATION)"/> <condition field="${switch_r_sdp}" expression="(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)" break="never"> <action application="set" data="sip_secure_media=true"/> <!-- Offer SRTP on outbound legs if we have it on inbound. --> <action application="export" data="sip_secure_media=true"/> </condition>
By commenting/un-commenting "sip_secure_media=true" above, it did not worked.
Any help with this would be greatly appreciated.
-- Thanks CWM
_________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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
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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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
First of all, srtp is not end-to-end, if you are looking for end to end, you should look over the zrtp_* varset. And also look over whether you clients are configured to use SRTP or ZRTP. 10 сент. 2014 г. 15:50 пользователь "Michael Jerris" <mike@jerris.com> написал:
Does it work on the latest 1.4 release with the config change he mentioned?
On Sep 10, 2014, at 1:48 AM, Chandrakant Marathe <cwmarathe@gmail.com> wrote:
Sorry Brian, I should have told the version. My bad. I am using 1.2 stable release. When I hit "git branch -av", it outputs -
* v1.2.stable 2b62885 fs_cli: fix compiler error on CentOS 6 caused by recent short uuid logging change
And from FreeSwitch console, when I hit "version" command, it shows -
FreeSWITCH Version 1.2.24+git~20140630T213113Z~2b62885f21~32bit (git 2b62885 2014-06-30 21:31:13Z 32bit)
I went through "vars.xml" and found only one config related with "rtp_secure_media" and that to related with zrtp
<X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/>
Is there any more configuration to do in v1.2 so as to enable end-to-end secure call or shall I move to v1.4 beta?
Thanks. -- CWM
On Tue, Sep 9, 2014 at 9:04 PM, Brian West <brian@freeswitch.org> wrote:
You didn't mention what rev you're using, If you're in 1.4 then its rtp_secure_media, please see latest configs and extensive docs in vars.xml about srtp and all the nice knobs you can use to tweak it.
On Tue, Sep 9, 2014 at 12:52 AM, Chandrakant Marathe <cwmarathe@gmail.com
wrote:
Hi All, I have setup FreeSwith PBX. I am facing an issue of not having end to end secure call. Caller sends INVITE request with SIPS in request URI and RTP/SAVP in SDP. But when FreeSwith forwards the request to caller, it is not using RTP/SAVP in SDP.
I have followed the steps mentioned in WIKI ( https://wiki.freeswitch.org/wiki/SIP_TLS#Configuration) for TLS configuration and marked "exports sip_secure_media" true in "conf/dialplan/default.xml" file (as per https://wiki.freeswitch.org/wiki/SRTP). But still FreeSwitch does not use RTP/SAVP for leg-B.
Also, I am bit confused with following condition in "default.xml" when wiki page suggests that late coded negotiation must NOT to be turned on.
<!-- Since we have inbound-late-negotation on by default now the above behavior isn't the same so you have to do one extra step. -->
<condition field="${endpoint_disposition}" expression="^(DELAYED NEGOTIATION)"/> <condition field="${switch_r_sdp}" expression="(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)" break="never"> <action application="set" data="sip_secure_media=true"/> <!-- Offer SRTP on outbound legs if we have it on inbound. --> <action application="export" data="sip_secure_media=true"/> </condition>
By commenting/un-commenting "sip_secure_media=true" above, it did not worked.
Any help with this would be greatly appreciated.
-- Thanks CWM
_________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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
*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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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 you can't trust your PBX then WHO can you trust? :P ZRTP won't be end to end out of the box unles you have clients that add the zrtp hash to the sdp's in the invite and 200ok coming back from the far end. On Wed, Sep 10, 2014 at 8:55 AM, Daniel Ivanov <sertys@gmail.com> wrote:
First of all, srtp is not end-to-end, if you are looking for end to end, you should look over the zrtp_* varset. And also look over whether you clients are configured to use SRTP or ZRTP. 10 сент. 2014 г. 15:50 пользователь "Michael Jerris" <mike@jerris.com> написал:
Does it work on the latest 1.4 release with the config change he mentioned?
On Sep 10, 2014, at 1:48 AM, Chandrakant Marathe <cwmarathe@gmail.com> wrote:
Sorry Brian, I should have told the version. My bad. I am using 1.2 stable release. When I hit "git branch -av", it outputs -
* v1.2.stable 2b62885 fs_cli: fix compiler error on CentOS 6 caused by recent short uuid logging change
And from FreeSwitch console, when I hit "version" command, it shows -
FreeSWITCH Version 1.2.24+git~20140630T213113Z~2b62885f21~32bit (git 2b62885 2014-06-30 21:31:13Z 32bit)
I went through "vars.xml" and found only one config related with "rtp_secure_media" and that to related with zrtp
<X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/>
Is there any more configuration to do in v1.2 so as to enable end-to-end secure call or shall I move to v1.4 beta?
Thanks. -- CWM
On Tue, Sep 9, 2014 at 9:04 PM, Brian West <brian@freeswitch.org> wrote:
You didn't mention what rev you're using, If you're in 1.4 then its rtp_secure_media, please see latest configs and extensive docs in vars.xml about srtp and all the nice knobs you can use to tweak it.
On Tue, Sep 9, 2014 at 12:52 AM, Chandrakant Marathe < cwmarathe@gmail.com> wrote:
Hi All, I have setup FreeSwith PBX. I am facing an issue of not having end to end secure call. Caller sends INVITE request with SIPS in request URI and RTP/SAVP in SDP. But when FreeSwith forwards the request to caller, it is not using RTP/SAVP in SDP.
I have followed the steps mentioned in WIKI ( https://wiki.freeswitch.org/wiki/SIP_TLS#Configuration) for TLS configuration and marked "exports sip_secure_media" true in "conf/dialplan/default.xml" file (as per https://wiki.freeswitch.org/wiki/SRTP). But still FreeSwitch does not use RTP/SAVP for leg-B.
Also, I am bit confused with following condition in "default.xml" when wiki page suggests that late coded negotiation must NOT to be turned on.
<!-- Since we have inbound-late-negotation on by default now the above behavior isn't the same so you have to do one extra step. -->
<condition field="${endpoint_disposition}" expression="^(DELAYED NEGOTIATION)"/> <condition field="${switch_r_sdp}" expression="(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)" break="never"> <action application="set" data="sip_secure_media=true"/> <!-- Offer SRTP on outbound legs if we have it on inbound. --> <action application="export" data="sip_secure_media=true"/> </condition>
By commenting/un-commenting "sip_secure_media=true" above, it did not worked.
Any help with this would be greatly appreciated.
-- Thanks CWM
_________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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
*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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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 *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
No, I have not yer check with 1.4. I'll check and let you know. -- CWM On Wed, Sep 10, 2014 at 7:52 PM, Brian West <brian@freeswitch.org> wrote:
If you can't trust your PBX then WHO can you trust? :P ZRTP won't be end to end out of the box unles you have clients that add the zrtp hash to the sdp's in the invite and 200ok coming back from the far end.
On Wed, Sep 10, 2014 at 8:55 AM, Daniel Ivanov <sertys@gmail.com> wrote:
First of all, srtp is not end-to-end, if you are looking for end to end, you should look over the zrtp_* varset. And also look over whether you clients are configured to use SRTP or ZRTP. 10 сент. 2014 г. 15:50 пользователь "Michael Jerris" <mike@jerris.com> написал:
Does it work on the latest 1.4 release with the config change he
mentioned?
On Sep 10, 2014, at 1:48 AM, Chandrakant Marathe <cwmarathe@gmail.com> wrote:
Sorry Brian, I should have told the version. My bad. I am using 1.2 stable release. When I hit "git branch -av", it outputs -
* v1.2.stable 2b62885 fs_cli: fix compiler error on CentOS 6 caused by recent short uuid logging change
And from FreeSwitch console, when I hit "version" command, it shows -
FreeSWITCH Version 1.2.24+git~20140630T213113Z~2b62885f21~32bit (git 2b62885 2014-06-30 21:31:13Z 32bit)
I went through "vars.xml" and found only one config related with "rtp_secure_media" and that to related with zrtp
<X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/>
Is there any more configuration to do in v1.2 so as to enable end-to-end secure call or shall I move to v1.4 beta?
Thanks. -- CWM
On Tue, Sep 9, 2014 at 9:04 PM, Brian West <brian@freeswitch.org> wrote:
You didn't mention what rev you're using, If you're in 1.4 then its rtp_secure_media, please see latest configs and extensive docs in vars.xml about srtp and all the nice knobs you can use to tweak it.
On Tue, Sep 9, 2014 at 12:52 AM, Chandrakant Marathe < cwmarathe@gmail.com> wrote:
Hi All, I have setup FreeSwith PBX. I am facing an issue of not having end to end secure call. Caller sends INVITE request with SIPS in request URI and RTP/SAVP in SDP. But when FreeSwith forwards the request to caller, it is not using RTP/SAVP in SDP.
I have followed the steps mentioned in WIKI ( https://wiki.freeswitch.org/wiki/SIP_TLS#Configuration) for TLS configuration and marked "exports sip_secure_media" true in "conf/dialplan/default.xml" file (as per https://wiki.freeswitch.org/wiki/SRTP). But still FreeSwitch does not use RTP/SAVP for leg-B.
Also, I am bit confused with following condition in "default.xml" when wiki page suggests that late coded negotiation must NOT to be turned on.
<!-- Since we have inbound-late-negotation on by default now the above behavior isn't the same so you have to do one extra step. -->
<condition field="${endpoint_disposition}" expression="^(DELAYED NEGOTIATION)"/> <condition field="${switch_r_sdp}" expression="(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)" break="never"> <action application="set" data="sip_secure_media=true"/> <!-- Offer SRTP on outbound legs if we have it on inbound. --> <action application="export" data="sip_secure_media=true"/> </condition>
By commenting/un-commenting "sip_secure_media=true" above, it did not worked.
Any help with this would be greatly appreciated.
-- Thanks CWM
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*Brian West* brian@freeswitch.org
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_________________________________________________________________________ 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
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_________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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Hi All, I tried with 1.4, but it is still not working. The issue is FS is not setting call with Leg-B. I have made following config changes 1. In var.xml, set following <X-PRE-PROCESS cmd="set" data="rtp_secure_media=optional"/> 2. In default.xml set following <action application="export" data="rtp_secure_media=true"/> In logs, I can see following error (while calling 1000 from 1001) - 2014-10-05 23:09:36.007270 [DEBUG] switch_ivr_originate.c:3695 Originate Resulted in Error Cause: 19 [NO_ANSWER] But 1000 does not receive any INVITE message Any pointer on this? Please find below logs from FS console - freeswitch@atul-amd> freeswitch@atul-amd> freeswitch@atul-amd> show registrations reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata 1001,192.168.0.103,2893d7a0-0605-4030-8094-e28be54c076f,sofia/internal/sip:1001@192.168.0.103:7001 ;transport=TLS;ob,1412531006,192.168.0.103,48732,tls,atul-amd, 1000,192.168.0.103,8d115512-6ee5-47bc-8dfe-1ff980e66166,sofia/internal/sip:1000@192.168.0.103:6001 ;transport=TLS;ob,1412531010,192.168.0.103,49206,tls,atul-amd, 2 total. freeswitch@atul-amd> 2014-10-05 23:08:55.967291 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/1001@pbx.myfreeswitch.org [24cb8d75-8119-4814-8530-d6b262194dfe] 2014-10-05 23:08:55.967291 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:08:55.967291 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1001@pbx.myfreeswitch.org) Running State Change CS_NEW 2014-10-05 23:08:55.967291 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:08:55.967291 [DEBUG] sofia.c:8570 sofia/internal/ 1001@pbx.myfreeswitch.org receiving invite from 192.168.0.103:48732 version: 1.4.8 git 498f586 2014-08-25 16:34:46Z 32bit 2014-10-05 23:08:55.967291 [DEBUG] sofia.c:8737 IP 192.168.0.103 Rejected by acl "domains". Falling back to Digest auth. 2014-10-05 23:08:55.967291 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1001@pbx.myfreeswitch.org) State NEW 2014-10-05 23:08:55.967291 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:08:55.967291 [DEBUG] sofia.c:2062 detaching session 24cb8d75-8119-4814-8530-d6b262194dfe 2014-10-05 23:08:56.007269 [DEBUG] sofia.c:2169 Re-attaching to session 24cb8d75-8119-4814-8530-d6b262194dfe 2014-10-05 23:08:56.007269 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:08:56.007269 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:08:56.027269 [DEBUG] sofia.c:8570 sofia/internal/ 1001@pbx.myfreeswitch.org receiving invite from 192.168.0.103:48732 version: 1.4.8 git 498f586 2014-08-25 16:34:46Z 32bit 2014-10-05 23:08:56.027269 [DEBUG] sofia.c:8737 IP 192.168.0.103 Rejected by acl "domains". Falling back to Digest auth. 2014-10-05 23:08:56.027269 [DEBUG] sofia.c:6408 Channel sofia/internal/ 1001@pbx.myfreeswitch.org entering state [received][100] 2014-10-05 23:08:56.027269 [DEBUG] sofia.c:6418 Remote SDP: v=0 o=- 3621519535 3621519535 IN IP4 192.168.0.103 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 9000 RTP/SAVP 98 97 99 104 3 0 8 9 96 c=IN IP4 192.168.0.103 b=AS:64000 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=rtcp:9001 IN IP4 192.168.0.103 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:QckTgP/H74P0S6u9sZm5ySmQ4i+zeZ/sw56endR9 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:/RZABbN0vRSybRtruRATkMrKo/P6wezyNcSkp0AX 2014-10-05 23:08:56.027269 [DEBUG] sofia.c:6673 (sofia/internal/ 1001@pbx.myfreeswitch.org) State Change CS_NEW -> CS_INIT 2014-10-05 23:08:56.027269 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:08:56.027269 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1001@pbx.myfreeswitch.org) Running State Change CS_INIT 2014-10-05 23:08:56.027269 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/1001@pbx.myfreeswitch.org) State INIT 2014-10-05 23:08:56.027269 [DEBUG] mod_sofia.c:87 sofia/internal/ 1001@pbx.myfreeswitch.org SOFIA INIT 2014-10-05 23:08:56.027269 [DEBUG] switch_core_state_machine.c:40 sofia/internal/1001@pbx.myfreeswitch.org Standard INIT 2014-10-05 23:08:56.027269 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/1001@pbx.myfreeswitch.org) State Change CS_INIT -> CS_ROUTING 2014-10-05 23:08:56.027269 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:08:56.027269 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/1001@pbx.myfreeswitch.org) State INIT going to sleep 2014-10-05 23:08:56.047282 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1001@pbx.myfreeswitch.org) Running State Change CS_ROUTING 2014-10-05 23:08:56.047282 [DEBUG] switch_channel.c:2184 (sofia/internal/ 1001@pbx.myfreeswitch.org) Callstate Change DOWN -> RINGING 2014-10-05 23:08:56.047282 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/1001@pbx.myfreeswitch.org) State ROUTING 2014-10-05 23:08:56.047282 [DEBUG] mod_sofia.c:123 sofia/internal/ 1001@pbx.myfreeswitch.org SOFIA ROUTING 2014-10-05 23:08:56.047282 [DEBUG] switch_core_state_machine.c:166 sofia/internal/1001@pbx.myfreeswitch.org Standard ROUTING 2014-10-05 23:08:56.047282 [INFO] mod_dialplan_xml.c:558 Processing 1001 <1001>->1000 in context default Dialplan: sofia/internal/1001@pbx.myfreeswitch.org parsing [default->unloop] continue=false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org parsing [default->tod_example] continue=true Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Date/TimeMatch (FAIL) [tod_example] break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org parsing [default->holiday_example] continue=true Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Date/TimeMatch (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org parsing [default->global-intercept] continue=false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (FAIL) [global-intercept] destination_number(1000) =~ /^886$/ break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org parsing [default->group-intercept] continue=false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (FAIL) [group-intercept] destination_number(1000) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (FAIL) [intercept-ext] destination_number(1000) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org parsing [default->redial] continue=false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (FAIL) [redial] destination_number(1000) =~ /^(redial|870)$/ break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org parsing [default->global] continue=true Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (PASS) [global] ${default_password}(1234) =~ /^1234$/ break=never Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action log(CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING ) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action log(CRIT Open ${base_dir}/conf/vars.xml and change the default_password.) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action log(CRIT Once changed type 'reloadxml' at the console.) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action log(CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING ) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action sleep(10000) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (FAIL) [global] ${rtp_has_crypto}() =~ /^(AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH)$/ break=never Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (PASS) [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED NEGOTIATION)/ break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (PASS) [global] ${switch_r_sdp}(v=0 o=- 3621519535 3621519535 IN IP4 192.168.0.103 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 9000 RTP/SAVP 98 97 99 104 3 0 8 9 96 c=IN IP4 192.168.0.103 b=AS:64000 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=rtcp:9001 IN IP4 192.168.0.103 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:QckTgP/H74P0S6u9sZm5ySmQ4i+zeZ/sw56endR9 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:/RZABbN0vRSybRtruRATkMrKo/P6wezyNcSkp0AX ) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action set(rtp_secure_media=true) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Absolute Condition [global] Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (FAIL) [snom-demo-2] destination_number(1000) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (FAIL) [snom-demo-1] destination_number(1000) =~ /^9000$/ break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (FAIL) [eavesdrop] destination_number(1000) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (FAIL) [eavesdrop] destination_number(1000) =~ /^779$/ break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org parsing [default->call_return] continue=false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (FAIL) [call_return] destination_number(1000) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org parsing [default->del-group] continue=false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (FAIL) [del-group] destination_number(1000) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org parsing [default->add-group] continue=false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (FAIL) [add-group] destination_number(1000) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (FAIL) [call-group-simo] destination_number(1000) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org parsing [default->call-group-order] continue=false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (FAIL) [call-group-order] destination_number(1000) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (FAIL) [extension-intercom] destination_number(1000) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Regex (PASS) [Local_Extension] destination_number(1000) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action export(dialed_extension=1000) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action set(ringback=${us-ring}) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action set(call_timeout=30) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action set(hangup_after_bridge=true) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action set(continue_on_fail=true) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action answer() Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action sleep(1000) Dialplan: sofia/internal/1001@pbx.myfreeswitch.org Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 2014-10-05 23:08:56.047282 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/1001@pbx.myfreeswitch.org) State Change CS_ROUTING -> CS_EXECUTE 2014-10-05 23:08:56.047282 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:08:56.047282 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/1001@pbx.myfreeswitch.org) State ROUTING going to sleep 2014-10-05 23:08:56.047282 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1001@pbx.myfreeswitch.org) Running State Change CS_EXECUTE 2014-10-05 23:08:56.047282 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/1001@pbx.myfreeswitch.org) State EXECUTE 2014-10-05 23:08:56.047282 [DEBUG] mod_sofia.c:178 sofia/internal/ 1001@pbx.myfreeswitch.org SOFIA EXECUTE 2014-10-05 23:08:56.047282 [DEBUG] switch_core_state_machine.c:258 sofia/internal/1001@pbx.myfreeswitch.org Standard EXECUTE EXECUTE sofia/internal/1001@pbx.myfreeswitch.org log(CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING ) 2014-10-05 23:08:56.047282 [CRIT] mod_dptools.c:1628 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING EXECUTE sofia/internal/1001@pbx.myfreeswitch.org log(CRIT Open /usr/local/freeswitch/conf/vars.xml and change the default_password.) 2014-10-05 23:08:56.047282 [CRIT] mod_dptools.c:1628 Open /usr/local/freeswitch/conf/vars.xml and change the default_password. EXECUTE sofia/internal/1001@pbx.myfreeswitch.org log(CRIT Once changed type 'reloadxml' at the console.) 2014-10-05 23:08:56.047282 [CRIT] mod_dptools.c:1628 Once changed type 'reloadxml' at the console. EXECUTE sofia/internal/1001@pbx.myfreeswitch.org log(CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING ) 2014-10-05 23:08:56.047282 [CRIT] mod_dptools.c:1628 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING EXECUTE sofia/internal/1001@pbx.myfreeswitch.org sleep(10000) EXECUTE sofia/internal/1001@pbx.myfreeswitch.org set(rtp_secure_media=true) 2014-10-05 23:09:06.067267 [DEBUG] mod_dptools.c:1435 sofia/internal/ 1001@pbx.myfreeswitch.org SET [rtp_secure_media]=[true] EXECUTE sofia/internal/1001@pbx.myfreeswitch.org hash(insert/192.168.0.103-spymap/1001/24cb8d75-8119-4814-8530-d6b262194dfe) EXECUTE sofia/internal/1001@pbx.myfreeswitch.org hash(insert/192.168.0.103-last_dial/1001/1000) EXECUTE sofia/internal/1001@pbx.myfreeswitch.org hash(insert/192.168.0.103-last_dial/global/24cb8d75-8119-4814-8530-d6b262194dfe) EXECUTE sofia/internal/1001@pbx.myfreeswitch.org export(RFC2822_DATE=Sun, 05 Oct 2014 23:09:06 +0530) 2014-10-05 23:09:06.067267 [DEBUG] switch_channel.c:1247 EXPORT (export_vars) [RFC2822_DATE]=[Sun, 05 Oct 2014 23:09:06 +0530] EXECUTE sofia/internal/1001@pbx.myfreeswitch.org export(dialed_extension=1000) 2014-10-05 23:09:06.067267 [DEBUG] switch_channel.c:1247 EXPORT (export_vars) [dialed_extension]=[1000] EXECUTE sofia/internal/1001@pbx.myfreeswitch.org bind_meta_app(1 b s execute_extension::dx XML features) 2014-10-05 23:09:06.067267 [INFO] switch_ivr_async.c:3661 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE sofia/internal/1001@pbx.myfreeswitch.org bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/1001.2014-10-05-23-09-06.wav) 2014-10-05 23:09:06.067267 [INFO] switch_ivr_async.c:3661 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1001.2014-10-05-23-09-06.wav EXECUTE sofia/internal/1001@pbx.myfreeswitch.org bind_meta_app(3 b s execute_extension::cf XML features) 2014-10-05 23:09:06.067267 [INFO] switch_ivr_async.c:3661 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE sofia/internal/1001@pbx.myfreeswitch.org bind_meta_app(4 b s execute_extension::att_xfer XML features) 2014-10-05 23:09:06.067267 [INFO] switch_ivr_async.c:3661 Bound B-Leg: *4 execute_extension::att_xfer XML features EXECUTE sofia/internal/1001@pbx.myfreeswitch.org set(ringback=%(2000,4000,440,480)) 2014-10-05 23:09:06.067267 [DEBUG] mod_dptools.c:1435 sofia/internal/ 1001@pbx.myfreeswitch.org SET [ringback]=[%(2000,4000,440,480)] EXECUTE sofia/internal/1001@pbx.myfreeswitch.org set(transfer_ringback=local_stream://moh) 2014-10-05 23:09:06.067267 [DEBUG] mod_dptools.c:1435 sofia/internal/ 1001@pbx.myfreeswitch.org SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/1001@pbx.myfreeswitch.org set(call_timeout=30) 2014-10-05 23:09:06.067267 [DEBUG] mod_dptools.c:1435 sofia/internal/ 1001@pbx.myfreeswitch.org SET [call_timeout]=[30] EXECUTE sofia/internal/1001@pbx.myfreeswitch.org set(hangup_after_bridge=true) 2014-10-05 23:09:06.067267 [DEBUG] mod_dptools.c:1435 sofia/internal/ 1001@pbx.myfreeswitch.org SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/1001@pbx.myfreeswitch.org set(continue_on_fail=true) 2014-10-05 23:09:06.067267 [DEBUG] mod_dptools.c:1435 sofia/internal/ 1001@pbx.myfreeswitch.org SET [continue_on_fail]=[true] EXECUTE sofia/internal/1001@pbx.myfreeswitch.org hash(insert/192.168.0.103-call_return/1000/1001) EXECUTE sofia/internal/1001@pbx.myfreeswitch.org hash(insert/192.168.0.103-last_dial_ext/1000/24cb8d75-8119-4814-8530-d6b262194dfe) EXECUTE sofia/internal/1001@pbx.myfreeswitch.org set(called_party_callgroup=techsupport) 2014-10-05 23:09:06.087251 [DEBUG] mod_dptools.c:1435 sofia/internal/ 1001@pbx.myfreeswitch.org SET [called_party_callgroup]=[techsupport] EXECUTE sofia/internal/1001@pbx.myfreeswitch.org hash(insert/192.168.0.103-last_dial_ext/techsupport/24cb8d75-8119-4814-8530-d6b262194dfe) EXECUTE sofia/internal/1001@pbx.myfreeswitch.org hash(insert/192.168.0.103-last_dial_ext/global/24cb8d75-8119-4814-8530-d6b262194dfe) EXECUTE sofia/internal/1001@pbx.myfreeswitch.org hash(insert/192.168.0.103-last_dial/techsupport/24cb8d75-8119-4814-8530-d6b262194dfe) EXECUTE sofia/internal/1001@pbx.myfreeswitch.org bridge(user/ 1000@192.168.0.103) 2014-10-05 23:09:06.087251 [DEBUG] switch_channel.c:1201 sofia/internal/ 1001@pbx.myfreeswitch.org EXPORTING[export_vars] [RFC2822_DATE]=[Sun, 05 Oct 2014 23:09:06 +0530] to event 2014-10-05 23:09:06.087251 [DEBUG] switch_channel.c:1201 sofia/internal/ 1001@pbx.myfreeswitch.org EXPORTING[export_vars] [dialed_extension]=[1000] to event 2014-10-05 23:09:06.087251 [DEBUG] switch_ivr_originate.c:2079 Parsing global variables 2014-10-05 23:09:06.087251 [DEBUG] switch_channel.c:1201 sofia/internal/ 1001@pbx.myfreeswitch.org EXPORTING[export_vars] [RFC2822_DATE]=[Sun, 05 Oct 2014 23:09:06 +0530] to event 2014-10-05 23:09:06.087251 [DEBUG] switch_channel.c:1201 sofia/internal/ 1001@pbx.myfreeswitch.org EXPORTING[export_vars] [dialed_extension]=[1000] to event 2014-10-05 23:09:06.087251 [DEBUG] switch_ivr_originate.c:2079 Parsing global variables 2014-10-05 23:09:06.087251 [DEBUG] switch_event.c:1688 Parsing variable [sip_invite_domain]=[192.168.0.103] 2014-10-05 23:09:06.087251 [DEBUG] switch_event.c:1688 Parsing variable [presence_id]=[1000@192.168.0.103] 2014-10-05 23:09:06.087251 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:1000@192.168.0.103:6001 [4c71ac74-e1bb-4053-b7e2-c306cc552496] 2014-10-05 23:09:06.087251 [DEBUG] mod_sofia.c:4577 (sofia/internal/ sip:1000@192.168.0.103:6001) State Change CS_NEW -> CS_INIT 2014-10-05 23:09:06.087251 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/sip:1000@192.168.0.103:6001 [BREAK] 2014-10-05 23:09:06.087251 [DEBUG] mod_sofia.c:4647 [zrtp_passthru] Setting a-leg inherit_codec=true 2014-10-05 23:09:06.087251 [DEBUG] mod_sofia.c:4650 [zrtp_passthru] Setting b-leg absolute_codec_string='GSM@8000h@20i@13200b,PCMU@8000h@20i@64000b ,PCMA@8000h@20i@64000b,G722@8000h@20i@64000b' 2014-10-05 23:09:06.087251 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/sip:1000@192.168.0.103:6001) Running State Change CS_INIT 2014-10-05 23:09:06.087251 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/sip:1000@192.168.0.103:6001) State INIT 2014-10-05 23:09:06.087251 [DEBUG] mod_sofia.c:87 sofia/internal/ sip:1000@192.168.0.103:6001 SOFIA INIT 2014-10-05 23:09:06.087251 [DEBUG] switch_core_media.c:922 Set Local audio crypto Key [1 AEAD_AES_256_GCM_8 inline:KVEHHfI0WMn8dY3j+9inOSCAL6i7rGi/RYP00gK1IY0aLfbL/wByWNUe/MY] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_media.c:922 Set Local video crypto Key [1 AEAD_AES_256_GCM_8 inline:BoZ1mTg+ci5qrLY8Vme3tWPoF4AINFglxK+8QdvZqPdS55I4E5Nk9IThbr0] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_media.c:922 Set Local audio crypto Key [2 AEAD_AES_128_GCM_8 inline:u9irqfqfMzxPXAWhC47VAXn2/gZNOBuPyM16xw] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_media.c:922 Set Local video crypto Key [2 AEAD_AES_128_GCM_8 inline:Q+VIFKfLIsUIjn6FHl1IaEuKhoYlhUx+vtQIfw] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_media.c:922 Set Local audio crypto Key [3 AES_CM_256_HMAC_SHA1_80 inline:QZblg/vbhhKyJ5WgMe4JSrnDFsXJDFYRwu+W3FzW0JCdyihzTEdyef0QAqf3BA] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_media.c:922 Set Local video crypto Key [3 AES_CM_256_HMAC_SHA1_80 inline:+LVeoi/jd0DTZJe0+cDxteF+kt0HRRCtCCTK6m/OS4/mVMP8WI5nJXK6CybcKw] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_media.c:922 Set Local audio crypto Key [4 AES_CM_192_HMAC_SHA1_80 inline:IfPHDgSIgeEqMQCPXxOmsyYSwizeMCQGoW3JH5dGTCKLPT9SqVk] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_media.c:922 Set Local video crypto Key [4 AES_CM_192_HMAC_SHA1_80 inline:5uy2A3XMR0RJ7fNtZF8rwJVFhzPn90irVu0s/xayqUZHPDQjbJA] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_media.c:922 Set Local audio crypto Key [5 AES_CM_128_HMAC_SHA1_80 inline:XQ1YW+zu79OD/F/nbweFbsjF1vcLSqgY3uyn+kYS] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_media.c:922 Set Local video crypto Key [5 AES_CM_128_HMAC_SHA1_80 inline:wzbsbwWbv63LqjR5MN87TsHNleDIqkCyeACd4UyA] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_media.c:922 Set Local audio crypto Key [6 AES_CM_256_HMAC_SHA1_32 inline:X2nN2U5g05vNNT4RlVMY1rcgqv0XQe9rMTg1fzURJqmQoQbwdvQv0RDsDNLKnQ] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_media.c:922 Set Local video crypto Key [6 AES_CM_256_HMAC_SHA1_32 inline:FFyXNGu0vcyDhYX++uGDxCmFKDnDX6q4sSsVxWMaARWTHF2d7wyxcknZA3aI1Q] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_media.c:922 Set Local audio crypto Key [7 AES_CM_192_HMAC_SHA1_32 inline:xiyNA3TiFXRDvE+jN5EMOvx/txeAVrNpiUI+9DG0TlrHU3ETiEo] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_media.c:922 Set Local video crypto Key [7 AES_CM_192_HMAC_SHA1_32 inline:bLr4agNy60XVpWV+hwvrsWNUYkTMos651WHAUcIJEYsHJaOtxH0] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_media.c:922 Set Local audio crypto Key [8 AES_CM_128_HMAC_SHA1_32 inline:ca5kOm9+odVtf2GbD5w/VfF9IVuNHbevuh8NRkfO] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_media.c:922 Set Local video crypto Key [8 AES_CM_128_HMAC_SHA1_32 inline:wq/MWC9DPKFJBXOarj8CUx7ULmDFO9LVM+FV70GM] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_media.c:922 Set Local audio crypto Key [9 AES_CM_128_NULL_AUTH inline:7PAnhG9Qw1yJUug8nqkxGpdapmls7m6wVoR5+OZU] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_media.c:922 Set Local video crypto Key [9 AES_CM_128_NULL_AUTH inline:bz7r0zGWjmDg7RLzF9vLkIBzoxM3Esswn6GlLt66] 2014-10-05 23:09:06.087251 [DEBUG] sofia_glue.c:1226 sofia/internal/ sip:1000@192.168.0.103:6001 sending invite version: 1.4.8 git 498f586 2014-08-25 16:34:46Z 32bit Local SDP: v=0 o=FreeSWITCH 1412505862 1412505863 IN IP4 192.168.0.103 s=FreeSWITCH c=IN IP4 192.168.0.103 t=0 0 m=audio 24884 RTP/SAVP 3 0 8 9 101 13 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AEAD_AES_256_GCM_8 inline:KVEHHfI0WMn8dY3j+9inOSCAL6i7rGi/RYP00gK1IY0aLfbL/wByWNUe/MY a=crypto:2 AEAD_AES_128_GCM_8 inline:u9irqfqfMzxPXAWhC47VAXn2/gZNOBuPyM16xw a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:QZblg/vbhhKyJ5WgMe4JSrnDFsXJDFYRwu+W3FzW0JCdyihzTEdyef0QAqf3BA a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:IfPHDgSIgeEqMQCPXxOmsyYSwizeMCQGoW3JH5dGTCKLPT9SqVk a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:XQ1YW+zu79OD/F/nbweFbsjF1vcLSqgY3uyn+kYS a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:X2nN2U5g05vNNT4RlVMY1rcgqv0XQe9rMTg1fzURJqmQoQbwdvQv0RDsDNLKnQ a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:xiyNA3TiFXRDvE+jN5EMOvx/txeAVrNpiUI+9DG0TlrHU3ETiEo a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:ca5kOm9+odVtf2GbD5w/VfF9IVuNHbevuh8NRkfO a=crypto:9 AES_CM_128_NULL_AUTH inline:7PAnhG9Qw1yJUug8nqkxGpdapmls7m6wVoR5+OZU a=ptime:20 a=sendrecv m=audio 24884 RTP/AVP 3 0 8 9 101 13 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2014-10-05 23:09:06.087251 [DEBUG] switch_core_state_machine.c:40 sofia/internal/sip:1000@192.168.0.103:6001 Standard INIT 2014-10-05 23:09:06.087251 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/sip:1000@192.168.0.103:6001) State Change CS_INIT -> CS_ROUTING 2014-10-05 23:09:06.087251 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/sip:1000@192.168.0.103:6001 [BREAK] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/sip:1000@192.168.0.103:6001) State INIT going to sleep 2014-10-05 23:09:06.087251 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1000@192.168.0.103:6001 [BREAK] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/sip:1000@192.168.0.103:6001) Running State Change CS_ROUTING 2014-10-05 23:09:06.087251 [DEBUG] sofia.c:6408 Channel sofia/internal/ sip:1000@192.168.0.103:6001 entering state [calling][0] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/sip:1000@192.168.0.103:6001) State ROUTING 2014-10-05 23:09:06.087251 [DEBUG] mod_sofia.c:123 sofia/internal/ sip:1000@192.168.0.103:6001 SOFIA ROUTING 2014-10-05 23:09:06.087251 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/sip:1000@192.168.0.103:6001) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2014-10-05 23:09:06.087251 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/sip:1000@192.168.0.103:6001 [BREAK] 2014-10-05 23:09:06.087251 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/sip:1000@192.168.0.103:6001) State ROUTING going to sleep 2014-10-05 23:09:06.087251 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/sip:1000@192.168.0.103:6001) Running State Change CS_CONSUME_MEDIA 2014-10-05 23:09:06.087251 [DEBUG] switch_core_state_machine.c:547 (sofia/internal/sip:1000@192.168.0.103:6001) State CONSUME_MEDIA 2014-10-05 23:09:06.087251 [DEBUG] switch_core_state_machine.c:547 (sofia/internal/sip:1000@192.168.0.103:6001) State CONSUME_MEDIA going to sleep 2014-10-05 23:09:36.007270 [NOTICE] switch_ivr_originate.c:3467 Hangup sofia/internal/sip:1000@192.168.0.103:6001 [CS_CONSUME_MEDIA] [NO_ANSWER] 2014-10-05 23:09:36.007270 [DEBUG] switch_channel.c:3222 Send signal sofia/internal/sip:1000@192.168.0.103:6001 [KILL] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/sip:1000@192.168.0.103:6001 [BREAK] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/sip:1000@192.168.0.103:6001) Running State Change CS_HANGUP 2014-10-05 23:09:36.007270 [DEBUG] switch_core_state_machine.c:735 (sofia/internal/sip:1000@192.168.0.103:6001) Callstate Change DOWN -> HANGUP 2014-10-05 23:09:36.007270 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/sip:1000@192.168.0.103:6001) State HANGUP 2014-10-05 23:09:36.007270 [DEBUG] mod_sofia.c:413 Channel sofia/internal/ sip:1000@192.168.0.103:6001 hanging up, cause: NO_ANSWER 2014-10-05 23:09:36.007270 [NOTICE] switch_ivr_originate.c:2708 Cannot create outgoing channel of type [user] cause: [NO_ANSWER] 2014-10-05 23:09:36.007270 [DEBUG] switch_ivr_originate.c:3695 Originate Resulted in Error Cause: 19 [NO_ANSWER] 2014-10-05 23:09:36.007270 [DEBUG] mod_sofia.c:475 Sending CANCEL to sofia/internal/sip:1000@192.168.0.103:6001 2014-10-05 23:09:36.007270 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:1000@192.168.0.103:6001 Standard HANGUP, cause: NO_ANSWER 2014-10-05 23:09:36.007270 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/sip:1000@192.168.0.103:6001) State HANGUP going to sleep 2014-10-05 23:09:36.007270 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:1000@192.168.0.103:6001) State Change CS_HANGUP -> CS_REPORTING 2014-10-05 23:09:36.007270 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/sip:1000@192.168.0.103:6001 [BREAK] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/sip:1000@192.168.0.103:6001) Running State Change CS_REPORTING 2014-10-05 23:09:36.007270 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/sip:1000@192.168.0.103:6001) State REPORTING 2014-10-05 23:09:36.007270 [DEBUG] switch_core_state_machine.c:104 sofia/internal/sip:1000@192.168.0.103:6001 Standard REPORTING, cause: NO_ANSWER 2014-10-05 23:09:36.007270 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/sip:1000@192.168.0.103:6001) State REPORTING going to sleep 2014-10-05 23:09:36.007270 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2014-10-05 23:09:36.007270 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/sip:1000@192.168.0.103:6001) State Change CS_REPORTING -> CS_DESTROY 2014-10-05 23:09:36.007270 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/sip:1000@192.168.0.103:6001 [BREAK] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_session.c:1615 Session 2 (sofia/internal/sip:1000@192.168.0.103:6001) Locked, Waiting on external entities 2014-10-05 23:09:36.007270 [NOTICE] switch_core_session.c:1633 Session 2 (sofia/internal/sip:1000@192.168.0.103:6001) Ended 2014-10-05 23:09:36.007270 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:1000@192.168.0.103:6001 [CS_DESTROY] EXECUTE sofia/internal/1001@pbx.myfreeswitch.org answer() 2014-10-05 23:09:36.007270 [DEBUG] switch_core_state_machine.c:626 (sofia/internal/sip:1000@192.168.0.103:6001) Running State Change CS_DESTROY 2014-10-05 23:09:36.007270 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/sip:1000@192.168.0.103:6001) State DESTROY 2014-10-05 23:09:36.007270 [DEBUG] mod_sofia.c:323 sofia/internal/ sip:1000@192.168.0.103:6001 SOFIA DESTROY 2014-10-05 23:09:36.007270 [DEBUG] switch_core_state_machine.c:111 sofia/internal/sip:1000@192.168.0.103:6001 Standard DESTROY 2014-10-05 23:09:36.007270 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/sip:1000@192.168.0.103:6001) State DESTROY going to sleep 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:1185 looking for crypto suite [AEAD_AES_256_GCM_8] in [1 AES_CM_128_HMAC_SHA1_80 inline:QckTgP/H74P0S6u9sZm5ySmQ4i+zeZ/sw56endR9] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:1185 looking for crypto suite [AEAD_AES_128_GCM_8] in [1 AES_CM_128_HMAC_SHA1_80 inline:QckTgP/H74P0S6u9sZm5ySmQ4i+zeZ/sw56endR9] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:1185 looking for crypto suite [AES_CM_256_HMAC_SHA1_80] in [1 AES_CM_128_HMAC_SHA1_80 inline:QckTgP/H74P0S6u9sZm5ySmQ4i+zeZ/sw56endR9] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:1185 looking for crypto suite [AES_CM_192_HMAC_SHA1_80] in [1 AES_CM_128_HMAC_SHA1_80 inline:QckTgP/H74P0S6u9sZm5ySmQ4i+zeZ/sw56endR9] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:1185 looking for crypto suite [AES_CM_128_HMAC_SHA1_80] in [1 AES_CM_128_HMAC_SHA1_80 inline:QckTgP/H74P0S6u9sZm5ySmQ4i+zeZ/sw56endR9] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:1190 Found suite AES_CM_128_HMAC_SHA1_80 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:1256 Set Remote Key [1 AES_CM_128_HMAC_SHA1_80 inline:QckTgP/H74P0S6u9sZm5ySmQ4i+zeZ/sw56endR9] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:922 Set Local audio crypto Key [1 AES_CM_128_HMAC_SHA1_80 inline:erDhB7sHda1RuOPnM6kFz2wq8cmBUUvJc7GRIY4m] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [speex:98:16000:20:0:1]/[opus:116:48000:20:0:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [speex:98:16000:20:0:1]/[G722:9:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [speex:98:16000:20:0:1]/[PCMU:0:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [speex:98:16000:20:0:1]/[PCMA:8:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [speex:98:16000:20:0:1]/[GSM:3:8000:20:13200:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [speex:97:8000:20:0:1]/[opus:116:48000:20:0:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [speex:97:8000:20:0:1]/[G722:9:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [speex:97:8000:20:0:1]/[PCMU:0:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [speex:97:8000:20:0:1]/[PCMA:8:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [speex:97:8000:20:0:1]/[GSM:3:8000:20:13200:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [speex:99:32000:20:0:1]/[opus:116:48000:20:0:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [speex:99:32000:20:0:1]/[G722:9:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [speex:99:32000:20:0:1]/[PCMU:0:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [speex:99:32000:20:0:1]/[PCMA:8:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [speex:99:32000:20:0:1]/[GSM:3:8000:20:13200:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [iLBC:104:8000:30:0:1]/[opus:116:48000:20:0:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [iLBC:104:8000:30:0:1]/[G722:9:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [iLBC:104:8000:30:0:1]/[PCMU:0:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [iLBC:104:8000:30:0:1]/[PCMA:8:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [iLBC:104:8000:30:0:1]/[GSM:3:8000:20:13200:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [GSM:3:8000:20:13200:1]/[opus:116:48000:20:0:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [GSM:3:8000:20:13200:1]/[G722:9:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMU:0:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMA:8:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [GSM:3:8000:20:13200:1]/[GSM:3:8000:20:13200:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3530 Audio Codec Compare [GSM:3:8000:20:13200:1] ++++ is saved as a match 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3530 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3530 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [G722:9:8000:20:64000:1]/[opus:116:48000:20:0:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3530 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [G722:9:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 2014-10-05 23:09:36.007270 [DEBUG] switch_core_media.c:3396 Set telephone-event payload to 96 2014-10-05 23:09:36.027261 [DEBUG] switch_core_media.c:2352 Set Codec sofia/internal/1001@pbx.myfreeswitch.org GSM/8000 20 ms 160 samples 13200 bits 1 channels 2014-10-05 23:09:36.027261 [DEBUG] switch_core_codec.c:111 sofia/internal/ 1001@pbx.myfreeswitch.org Original read codec set to GSM:3 2014-10-05 23:09:36.027261 [DEBUG] switch_core_media.c:3721 Set 2833 dtmf send/recv payload to 96 2014-10-05 23:09:36.027261 [DEBUG] switch_core_media.c:4941 AUDIO RTP [sofia/internal/1001@pbx.myfreeswitch.org] 192.168.0.103 port 22482 -> 192.168.0.103 port 9000 codec: 3 ms: 20 2014-10-05 23:09:36.027261 [DEBUG] switch_rtp.c:3386 Starting timer [soft] 160 bytes per 20ms 2014-10-05 23:09:36.027261 [INFO] switch_core_media.c:5158 Activating RTCP PORT 9001 2014-10-05 23:09:36.027261 [DEBUG] switch_rtp.c:3732 RTCP send rate is: 5000 and packet rate is: 20000 Remote Port: 9001 2014-10-05 23:09:36.027261 [DEBUG] switch_rtp.c:2193 Setting RTCP remote addr to 192.168.0.103:9001 2014-10-05 23:09:36.027261 [DEBUG] switch_core_media.c:5293 Set 2833 dtmf send payload to 96 2014-10-05 23:09:36.027261 [DEBUG] switch_core_media.c:5299 Set 2833 dtmf receive payload to 96 2014-10-05 23:09:36.027261 [INFO] switch_rtp.c:3203 Activating Audio Secure RTP SEND 2014-10-05 23:09:36.027261 [INFO] switch_rtp.c:3181 Activating Audio Secure RTP RECV 2014-10-05 23:09:36.027261 [DEBUG] switch_core_sqldb.c:2568 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 2014-10-05 23:09:36.027261 [DEBUG] switch_core_sqldb.c:2568 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 2014-10-05 23:09:36.027261 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1001@pbx.myfreeswitch.org! 2014-10-05 23:09:36.027261 [DEBUG] switch_channel.c:3399 (sofia/internal/ 1001@pbx.myfreeswitch.org) Callstate Change RINGING -> EARLY 2014-10-05 23:09:36.027261 [DEBUG] mod_sofia.c:778 Local SDP sofia/internal/ 1001@pbx.myfreeswitch.org: v=0 o=FreeSWITCH 1412508294 1412508295 IN IP4 192.168.0.103 s=FreeSWITCH c=IN IP4 192.168.0.103 t=0 0 m=audio 22482 RTP/SAVP 3 96 a=rtpmap:3 GSM/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ptime:20 a=sendrecv a=rtcp:22483 IN IP4 192.168.0.103 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:erDhB7sHda1RuOPnM6kFz2wq8cmBUUvJc7GRIY4m 2014-10-05 23:09:36.027261 [DEBUG] switch_core_session.c:908 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:09:36.027261 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/1001@pbx.myfreeswitch.org] has been answered 2014-10-05 23:09:36.027261 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:09:36.027261 [DEBUG] switch_channel.c:3689 (sofia/internal/ 1001@pbx.myfreeswitch.org) Callstate Change EARLY -> ACTIVE 2014-10-05 23:09:36.027261 [DEBUG] sofia.c:6408 Channel sofia/internal/ 1001@pbx.myfreeswitch.org entering state [completed][200] EXECUTE sofia/internal/1001@pbx.myfreeswitch.org sleep(1000) 2014-10-05 23:09:36.027261 [DEBUG] switch_ivr.c:191 Codec Activated L16@8000hz 1 channels 20ms 2014-10-05 23:09:36.027261 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:09:36.027261 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:09:36.027261 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:09:36.047255 [DEBUG] sofia.c:6408 Channel sofia/internal/ 1001@pbx.myfreeswitch.org entering state [ready][200] 2014-10-05 23:09:36.067268 [DEBUG] switch_rtp.c:5644 Correct ip/port confirmed. EXECUTE sofia/internal/1001@pbx.myfreeswitch.org bridge(loopback/app=voicemail:default 192.168.0.103 1000) 2014-10-05 23:09:37.027269 [DEBUG] switch_channel.c:1766 (sofia/internal/ 1001@pbx.myfreeswitch.org) Callstate Change ACTIVE -> RING_WAIT 2014-10-05 23:09:37.027269 [DEBUG] switch_channel.c:1201 sofia/internal/ 1001@pbx.myfreeswitch.org EXPORTING[export_vars] [RFC2822_DATE]=[Sun, 05 Oct 2014 23:09:06 +0530] to event 2014-10-05 23:09:37.027269 [DEBUG] switch_channel.c:1201 sofia/internal/ 1001@pbx.myfreeswitch.org EXPORTING[export_vars] [dialed_extension]=[1000] to event 2014-10-05 23:09:37.027269 [DEBUG] switch_ivr_originate.c:2079 Parsing global variables 2014-10-05 23:09:37.027269 [NOTICE] switch_channel.c:1055 New Channel loopback/app=voicemail:default 192.168.0.103 1000-a [949e0eba-57d7-4b3b-b737-d7355bf0dc44] 2014-10-05 23:09:37.027269 [DEBUG] mod_loopback.c:157 loopback/app=voicemail:default 192.168.0.103 1000-a setup codec GSM/8000/20 2014-10-05 23:09:37.027269 [NOTICE] switch_channel.c:1053 Rename Channel loopback/app=voicemail:default 192.168.0.103 1000-a->loopback/voicemail-a [949e0eba-57d7-4b3b-b737-d7355bf0dc44] 2014-10-05 23:09:37.027269 [DEBUG] mod_loopback.c:1160 (loopback/voicemail-a) State Change CS_NEW -> CS_INIT 2014-10-05 23:09:37.027269 [DEBUG] switch_core_session.c:1388 Send signal loopback/voicemail-a [BREAK] 2014-10-05 23:09:37.027269 [DEBUG] mod_loopback.c:590 loopback/voicemail-a CHANNEL KILL 2014-10-05 23:09:37.027269 [DEBUG] switch_core_state_machine.c:472 (loopback/voicemail-a) Running State Change CS_INIT 2014-10-05 23:09:37.027269 [DEBUG] switch_core_state_machine.c:512 (loopback/voicemail-a) State INIT 2014-10-05 23:09:37.027269 [NOTICE] switch_channel.c:1055 New Channel loopback/voicemail-b [b26df172-6b38-46a9-8649-d1294f674be8] 2014-10-05 23:09:37.027269 [DEBUG] mod_loopback.c:157 loopback/voicemail-b setup codec GSM/8000/20 2014-10-05 23:09:37.027269 [DEBUG] mod_loopback.c:273 (loopback/voicemail-b) State Change CS_NEW -> CS_INIT 2014-10-05 23:09:37.027269 [DEBUG] switch_core_session.c:1388 Send signal loopback/voicemail-b [BREAK] 2014-10-05 23:09:37.027269 [DEBUG] mod_loopback.c:590 loopback/voicemail-b CHANNEL KILL 2014-10-05 23:09:37.027269 [DEBUG] mod_loopback.c:359 (loopback/voicemail-a) State Change CS_INIT -> CS_ROUTING 2014-10-05 23:09:37.027269 [DEBUG] switch_core_session.c:1388 Send signal loopback/voicemail-a [BREAK] 2014-10-05 23:09:37.027269 [DEBUG] mod_loopback.c:590 loopback/voicemail-a CHANNEL KILL 2014-10-05 23:09:37.027269 [DEBUG] switch_core_state_machine.c:512 (loopback/voicemail-a) State INIT going to sleep 2014-10-05 23:09:37.027269 [DEBUG] switch_core_state_machine.c:472 (loopback/voicemail-b) Running State Change CS_INIT 2014-10-05 23:09:37.027269 [DEBUG] switch_core_state_machine.c:512 (loopback/voicemail-b) State INIT 2014-10-05 23:09:37.027269 [DEBUG] mod_loopback.c:359 (loopback/voicemail-b) State Change CS_INIT -> CS_ROUTING 2014-10-05 23:09:37.027269 [DEBUG] switch_core_session.c:1388 Send signal loopback/voicemail-b [BREAK] 2014-10-05 23:09:37.027269 [DEBUG] mod_loopback.c:590 loopback/voicemail-b CHANNEL KILL 2014-10-05 23:09:37.027269 [DEBUG] switch_core_state_machine.c:512 (loopback/voicemail-b) State INIT going to sleep 2014-10-05 23:09:37.027269 [DEBUG] switch_core_state_machine.c:472 (loopback/voicemail-b) Running State Change CS_ROUTING 2014-10-05 23:09:37.027269 [DEBUG] switch_channel.c:2184 (loopback/voicemail-b) Callstate Change DOWN -> RINGING 2014-10-05 23:09:37.027269 [DEBUG] switch_core_state_machine.c:472 (loopback/voicemail-a) Running State Change CS_ROUTING 2014-10-05 23:09:37.027269 [DEBUG] switch_core_state_machine.c:528 (loopback/voicemail-a) State ROUTING 2014-10-05 23:09:37.027269 [DEBUG] switch_core_state_machine.c:528 (loopback/voicemail-b) State ROUTING 2014-10-05 23:09:37.027269 [DEBUG] mod_loopback.c:391 loopback/voicemail-a CHANNEL ROUTING 2014-10-05 23:09:37.027269 [DEBUG] mod_loopback.c:391 loopback/voicemail-b CHANNEL ROUTING 2014-10-05 23:09:37.027269 [DEBUG] switch_ivr_originate.c:67 (loopback/voicemail-a) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2014-10-05 23:09:37.027269 [DEBUG] mod_loopback.c:410 (loopback/voicemail-b) State Change CS_ROUTING -> CS_EXECUTE 2014-10-05 23:09:37.027269 [DEBUG] switch_core_session.c:1388 Send signal loopback/voicemail-a [BREAK] 2014-10-05 23:09:37.027269 [DEBUG] switch_core_session.c:1388 Send signal loopback/voicemail-b [BREAK] 2014-10-05 23:09:37.027269 [DEBUG] mod_loopback.c:590 loopback/voicemail-a CHANNEL KILL 2014-10-05 23:09:37.027269 [DEBUG] mod_loopback.c:590 loopback/voicemail-b CHANNEL KILL 2014-10-05 23:09:37.027269 [DEBUG] switch_core_state_machine.c:528 (loopback/voicemail-b) State ROUTING going to sleep 2014-10-05 23:09:37.027269 [DEBUG] switch_core_state_machine.c:528 (loopback/voicemail-a) State ROUTING going to sleep 2014-10-05 23:09:37.027269 [DEBUG] switch_core_state_machine.c:472 (loopback/voicemail-b) Running State Change CS_EXECUTE 2014-10-05 23:09:37.027269 [DEBUG] switch_core_state_machine.c:472 (loopback/voicemail-a) Running State Change CS_CONSUME_MEDIA 2014-10-05 23:09:37.027269 [DEBUG] switch_core_state_machine.c:535 (loopback/voicemail-b) State EXECUTE 2014-10-05 23:09:37.027269 [DEBUG] mod_loopback.c:433 loopback/voicemail-b CHANNEL EXECUTE 2014-10-05 23:09:37.027269 [DEBUG] switch_core_state_machine.c:547 (loopback/voicemail-a) State CONSUME_MEDIA 2014-10-05 23:09:37.027269 [DEBUG] switch_core_state_machine.c:258 loopback/voicemail-b Standard EXECUTE 2014-10-05 23:09:37.027269 [DEBUG] mod_loopback.c:650 CHANNEL CONSUME_MEDIA 2014-10-05 23:09:37.027269 [DEBUG] switch_core_state_machine.c:547 (loopback/voicemail-a) State CONSUME_MEDIA going to sleep EXECUTE loopback/voicemail-b pre_answer() 2014-10-05 23:09:37.027269 [NOTICE] mod_loopback.c:947 Pre-Answer loopback/voicemail-a! 2014-10-05 23:09:37.027269 [DEBUG] switch_channel.c:3395 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:09:37.027269 [DEBUG] switch_channel.c:3399 (loopback/voicemail-a) Callstate Change DOWN -> EARLY 2014-10-05 23:09:37.027269 [DEBUG] switch_core_session.c:908 Send signal loopback/voicemail-b [BREAK] 2014-10-05 23:09:37.027269 [DEBUG] mod_loopback.c:590 loopback/voicemail-b CHANNEL KILL 2014-10-05 23:09:37.027269 [NOTICE] mod_dptools.c:1293 Pre-Answer loopback/voicemail-b! 2014-10-05 23:09:37.027269 [DEBUG] switch_channel.c:3399 (loopback/voicemail-b) Callstate Change RINGING -> EARLY EXECUTE loopback/voicemail-b voicemail(default 192.168.0.103 1000) 2014-10-05 23:09:37.047293 [DEBUG] switch_ivr_originate.c:3552 Originate Resulted in Success: [loopback/voicemail-a] 2014-10-05 23:09:37.047293 [DEBUG] switch_channel.c:1979 (sofia/internal/ 1001@pbx.myfreeswitch.org) Callstate Change RING_WAIT -> ACTIVE 2014-10-05 23:09:37.047293 [DEBUG] switch_core_session.c:908 Send signal loopback/voicemail-a [BREAK] 2014-10-05 23:09:37.047293 [DEBUG] mod_loopback.c:590 loopback/voicemail-a CHANNEL KILL 2014-10-05 23:09:37.047293 [DEBUG] switch_core_session.c:908 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:09:37.047293 [DEBUG] switch_ivr_bridge.c:1465 (loopback/voicemail-a) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2014-10-05 23:09:37.047293 [DEBUG] switch_core_session.c:1388 Send signal loopback/voicemail-a [BREAK] 2014-10-05 23:09:37.047293 [DEBUG] mod_loopback.c:590 loopback/voicemail-a CHANNEL KILL 2014-10-05 23:09:37.047293 [DEBUG] switch_core_state_machine.c:472 (loopback/voicemail-a) Running State Change CS_EXCHANGE_MEDIA 2014-10-05 23:09:37.047293 [DEBUG] switch_core_state_machine.c:538 (loopback/voicemail-a) State EXCHANGE_MEDIA 2014-10-05 23:09:37.047293 [DEBUG] mod_loopback.c:612 CHANNEL LOOPBACK 2014-10-05 23:09:37.127282 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en] 2014-10-05 23:09:37.147268 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-person.wav] (en:en) 2014-10-05 23:09:37.167266 [DEBUG] switch_ivr_play_say.c:1305 Codec Activated L16@8000hz 1 channels 20ms 2014-10-05 23:09:38.487267 [DEBUG] switch_ivr_play_say.c:1719 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-person.wav 2014-10-05 23:09:38.587267 [DEBUG] switch_ivr_play_say.c:250 Handle say:[1000] (en:en) 2014-10-05 23:09:38.587267 [DEBUG] switch_ivr_play_say.c:1305 Codec Activated L16@8000hz 1 channels 20ms 2014-10-05 23:09:40.887267 [DEBUG] switch_ivr_play_say.c:1719 done playing file file_string://digits/1.wav!digits/0.wav!digits/0.wav!digits/0.wav 2014-10-05 23:09:40.987268 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-not_available.wav] (en:en) 2014-10-05 23:09:40.987268 [DEBUG] switch_ivr_play_say.c:1305 Codec Activated L16@8000hz 1 channels 20ms 2014-10-05 23:09:41.967269 [DEBUG] switch_ivr_play_say.c:1719 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-not_available.wav 2014-10-05 23:09:42.087269 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en] 2014-10-05 23:09:42.127281 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-record_message.wav] (en:en) 2014-10-05 23:09:42.127281 [DEBUG] switch_ivr_play_say.c:1305 Codec Activated L16@8000hz 1 channels 20ms 2014-10-05 23:09:46.687267 [DEBUG] switch_ivr_play_say.c:1719 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_message.wav 2014-10-05 23:09:46.807267 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:09:46.827270 [NOTICE] sofia.c:947 Hangup sofia/internal/ 1001@pbx.myfreeswitch.org [CS_EXECUTE] [NORMAL_CLEARING] 2014-10-05 23:09:46.827270 [DEBUG] switch_channel.c:3222 Send signal sofia/internal/1001@pbx.myfreeswitch.org [KILL] 2014-10-05 23:09:46.827270 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:09:46.827270 [DEBUG] switch_ivr_bridge.c:579 sofia/internal/ 1001@pbx.myfreeswitch.org ending bridge by request from write function 2014-10-05 23:09:46.827270 [DEBUG] switch_ivr_bridge.c:660 BRIDGE THREAD DONE [loopback/voicemail-a] 2014-10-05 23:09:46.827270 [DEBUG] switch_ivr_bridge.c:690 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:09:46.827270 [NOTICE] switch_ivr_bridge.c:752 Hangup loopback/voicemail-a [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL] 2014-10-05 23:09:46.827270 [DEBUG] switch_channel.c:3222 Send signal loopback/voicemail-a [KILL] 2014-10-05 23:09:46.827270 [DEBUG] mod_loopback.c:590 loopback/voicemail-a CHANNEL KILL 2014-10-05 23:09:46.827270 [DEBUG] switch_core_session.c:1388 Send signal loopback/voicemail-a [BREAK] 2014-10-05 23:09:46.827270 [DEBUG] mod_loopback.c:590 loopback/voicemail-a CHANNEL KILL 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:538 (loopback/voicemail-a) State EXCHANGE_MEDIA going to sleep 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:472 (loopback/voicemail-a) Running State Change CS_HANGUP 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:735 (loopback/voicemail-a) Callstate Change EARLY -> HANGUP 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:737 (loopback/voicemail-a) State HANGUP 2014-10-05 23:09:46.827270 [DEBUG] mod_loopback.c:537 loopback/voicemail-a CHANNEL HANGUP 2014-10-05 23:09:46.827270 [NOTICE] mod_loopback.c:553 Hangup loopback/voicemail-b [CS_EXECUTE] [ORIGINATOR_CANCEL] 2014-10-05 23:09:46.827270 [DEBUG] switch_channel.c:3222 Send signal loopback/voicemail-b [KILL] 2014-10-05 23:09:46.827270 [DEBUG] mod_loopback.c:590 loopback/voicemail-b CHANNEL KILL 2014-10-05 23:09:46.827270 [DEBUG] switch_core_session.c:1388 Send signal loopback/voicemail-b [BREAK] 2014-10-05 23:09:46.827270 [DEBUG] mod_loopback.c:590 loopback/voicemail-b CHANNEL KILL 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:60 loopback/voicemail-a Standard HANGUP, cause: ORIGINATOR_CANCEL 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:737 (loopback/voicemail-a) State HANGUP going to sleep 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:504 (loopback/voicemail-a) State Change CS_HANGUP -> CS_REPORTING 2014-10-05 23:09:46.827270 [DEBUG] switch_core_session.c:1388 Send signal loopback/voicemail-a [BREAK] 2014-10-05 23:09:46.827270 [DEBUG] mod_loopback.c:590 loopback/voicemail-a CHANNEL KILL 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:472 (loopback/voicemail-a) Running State Change CS_REPORTING 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:823 (loopback/voicemail-a) State REPORTING 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:104 loopback/voicemail-a Standard REPORTING, cause: ORIGINATOR_CANCEL 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:823 (loopback/voicemail-a) State REPORTING going to sleep 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:498 (loopback/voicemail-a) State Change CS_REPORTING -> CS_DESTROY 2014-10-05 23:09:46.827270 [DEBUG] switch_core_session.c:1388 Send signal loopback/voicemail-a [BREAK] 2014-10-05 23:09:46.827270 [DEBUG] mod_loopback.c:590 loopback/voicemail-a CHANNEL KILL 2014-10-05 23:09:46.827270 [DEBUG] switch_core_session.c:1615 Session 3 (loopback/voicemail-a) Locked, Waiting on external entities 2014-10-05 23:09:46.827270 [DEBUG] switch_ivr_bridge.c:660 BRIDGE THREAD DONE [sofia/internal/1001@pbx.myfreeswitch.org] 2014-10-05 23:09:46.827270 [DEBUG] switch_ivr_bridge.c:690 Send signal loopback/voicemail-a [BREAK] 2014-10-05 23:09:46.827270 [DEBUG] mod_loopback.c:590 loopback/voicemail-a CHANNEL KILL 2014-10-05 23:09:46.827270 [WARNING] mod_voicemail.c:1238 Failed to delete file [/usr/local/freeswitch/storage/voicemail/default/ 192.168.0.103/1000/msg_2e38d3d7-1ca5-4cfe-a682-ffa8908d245c.wav] 2014-10-05 23:09:46.827270 [DEBUG] mod_voicemail.c:1245 Message is less than minimum record length: 3, discarding it. 2014-10-05 23:09:46.827270 [DEBUG] switch_core_session.c:2888 loopback/voicemail-b skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:535 (loopback/voicemail-b) State EXECUTE going to sleep 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:472 (loopback/voicemail-b) Running State Change CS_HANGUP 2014-10-05 23:09:46.827270 [DEBUG] switch_ivr_bridge.c:1566 sofia/internal/ 1001@pbx.myfreeswitch.org skip receive message [UNBRIDGE] (channel is hungup already) 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:735 (loopback/voicemail-b) Callstate Change EARLY -> HANGUP 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:737 (loopback/voicemail-b) State HANGUP 2014-10-05 23:09:46.827270 [DEBUG] mod_loopback.c:537 loopback/voicemail-b CHANNEL HANGUP 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:60 loopback/voicemail-b Standard HANGUP, cause: ORIGINATOR_CANCEL 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:737 (loopback/voicemail-b) State HANGUP going to sleep 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:504 (loopback/voicemail-b) State Change CS_HANGUP -> CS_REPORTING 2014-10-05 23:09:46.827270 [DEBUG] switch_core_session.c:1388 Send signal loopback/voicemail-b [BREAK] 2014-10-05 23:09:46.827270 [NOTICE] switch_core_session.c:1633 Session 3 (loopback/voicemail-a) Ended 2014-10-05 23:09:46.827270 [NOTICE] switch_core_session.c:1637 Close Channel loopback/voicemail-a [CS_DESTROY] 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:626 (loopback/voicemail-a) Running State Change CS_DESTROY 2014-10-05 23:09:46.827270 [DEBUG] mod_loopback.c:590 loopback/voicemail-b CHANNEL KILL 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:472 (loopback/voicemail-b) Running State Change CS_REPORTING 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:823 (loopback/voicemail-b) State REPORTING 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:104 loopback/voicemail-b Standard REPORTING, cause: ORIGINATOR_CANCEL 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:823 (loopback/voicemail-b) State REPORTING going to sleep 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:636 (loopback/voicemail-a) State DESTROY 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:111 loopback/voicemail-a Standard DESTROY 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:636 (loopback/voicemail-a) State DESTROY going to sleep 2014-10-05 23:09:46.827270 [DEBUG] switch_core_session.c:2888 sofia/internal/1001@pbx.myfreeswitch.org skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/1001@pbx.myfreeswitch.org) State EXECUTE going to sleep 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1001@pbx.myfreeswitch.org) Running State Change CS_HANGUP 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:498 (loopback/voicemail-b) State Change CS_REPORTING -> CS_DESTROY 2014-10-05 23:09:46.827270 [DEBUG] switch_core_session.c:1388 Send signal loopback/voicemail-b [BREAK] 2014-10-05 23:09:46.827270 [DEBUG] mod_loopback.c:590 loopback/voicemail-b CHANNEL KILL 2014-10-05 23:09:46.827270 [DEBUG] switch_core_session.c:1615 Session 4 (loopback/voicemail-b) Locked, Waiting on external entities 2014-10-05 23:09:46.827270 [NOTICE] switch_core_session.c:1633 Session 4 (loopback/voicemail-b) Ended 2014-10-05 23:09:46.827270 [NOTICE] switch_core_session.c:1637 Close Channel loopback/voicemail-b [CS_DESTROY] 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:626 (loopback/voicemail-b) Running State Change CS_DESTROY 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:735 (sofia/internal/1001@pbx.myfreeswitch.org) Callstate Change ACTIVE -> HANGUP 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:636 (loopback/voicemail-b) State DESTROY 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/1001@pbx.myfreeswitch.org) State HANGUP 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:111 loopback/voicemail-b Standard DESTROY 2014-10-05 23:09:46.827270 [DEBUG] mod_sofia.c:413 Channel sofia/internal/ 1001@pbx.myfreeswitch.org hanging up, cause: NORMAL_CLEARING 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:636 (loopback/voicemail-b) State DESTROY going to sleep 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1001@pbx.myfreeswitch.org Standard HANGUP, cause: NORMAL_CLEARING 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/1001@pbx.myfreeswitch.org) State HANGUP going to sleep 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1001@pbx.myfreeswitch.org) State Change CS_HANGUP -> CS_REPORTING 2014-10-05 23:09:46.827270 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1001@pbx.myfreeswitch.org) Running State Change CS_REPORTING 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/1001@pbx.myfreeswitch.org) State REPORTING 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1001@pbx.myfreeswitch.org Standard REPORTING, cause: NORMAL_CLEARING 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/1001@pbx.myfreeswitch.org) State REPORTING going to sleep 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1001@pbx.myfreeswitch.org) State Change CS_REPORTING -> CS_DESTROY 2014-10-05 23:09:46.827270 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/1001@pbx.myfreeswitch.org [BREAK] 2014-10-05 23:09:46.827270 [DEBUG] switch_core_session.c:1615 Session 1 (sofia/internal/1001@pbx.myfreeswitch.org) Locked, Waiting on external entities 2014-10-05 23:09:46.827270 [NOTICE] switch_core_session.c:1633 Session 1 (sofia/internal/1001@pbx.myfreeswitch.org) Ended 2014-10-05 23:09:46.827270 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/1001@pbx.myfreeswitch.org [CS_DESTROY] 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:626 (sofia/internal/1001@pbx.myfreeswitch.org) Running State Change CS_DESTROY 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/1001@pbx.myfreeswitch.org) State DESTROY 2014-10-05 23:09:46.827270 [DEBUG] mod_sofia.c:323 sofia/internal/ 1001@pbx.myfreeswitch.org SOFIA DESTROY 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:111 sofia/internal/1001@pbx.myfreeswitch.org Standard DESTROY 2014-10-05 23:09:46.827270 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/1001@pbx.myfreeswitch.org) State DESTROY going to sleep freeswitch@atul-amd> freeswitch@atul-amd> freeswitch@atul-amd> freeswitch@atul-amd> freeswitch@atul-amd> -- Thanks CWM On Thu, Sep 11, 2014 at 10:50 AM, Chandrakant Marathe <cwmarathe@gmail.com> wrote:
No, I have not yer check with 1.4. I'll check and let you know.
-- CWM
On Wed, Sep 10, 2014 at 7:52 PM, Brian West <brian@freeswitch.org> wrote:
If you can't trust your PBX then WHO can you trust? :P ZRTP won't be end to end out of the box unles you have clients that add the zrtp hash to the sdp's in the invite and 200ok coming back from the far end.
On Wed, Sep 10, 2014 at 8:55 AM, Daniel Ivanov <sertys@gmail.com> wrote:
First of all, srtp is not end-to-end, if you are looking for end to end, you should look over the zrtp_* varset. And also look over whether you clients are configured to use SRTP or ZRTP. 10 сент. 2014 г. 15:50 пользователь "Michael Jerris" <mike@jerris.com> написал:
Does it work on the latest 1.4 release with the config change he
mentioned?
On Sep 10, 2014, at 1:48 AM, Chandrakant Marathe <cwmarathe@gmail.com> wrote:
Sorry Brian, I should have told the version. My bad. I am using 1.2 stable release. When I hit "git branch -av", it outputs -
* v1.2.stable 2b62885 fs_cli: fix compiler error on CentOS 6 caused by recent short uuid logging change
And from FreeSwitch console, when I hit "version" command, it shows -
FreeSWITCH Version 1.2.24+git~20140630T213113Z~2b62885f21~32bit (git 2b62885 2014-06-30 21:31:13Z 32bit)
I went through "vars.xml" and found only one config related with "rtp_secure_media" and that to related with zrtp
<X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/>
Is there any more configuration to do in v1.2 so as to enable end-to-end secure call or shall I move to v1.4 beta?
Thanks. -- CWM
On Tue, Sep 9, 2014 at 9:04 PM, Brian West <brian@freeswitch.org> wrote:
You didn't mention what rev you're using, If you're in 1.4 then its rtp_secure_media, please see latest configs and extensive docs in vars.xml about srtp and all the nice knobs you can use to tweak it.
On Tue, Sep 9, 2014 at 12:52 AM, Chandrakant Marathe < cwmarathe@gmail.com> wrote:
Hi All, I have setup FreeSwith PBX. I am facing an issue of not having end to end secure call. Caller sends INVITE request with SIPS in request URI and RTP/SAVP in SDP. But when FreeSwith forwards the request to caller, it is not using RTP/SAVP in SDP.
I have followed the steps mentioned in WIKI ( https://wiki.freeswitch.org/wiki/SIP_TLS#Configuration) for TLS configuration and marked "exports sip_secure_media" true in "conf/dialplan/default.xml" file (as per https://wiki.freeswitch.org/wiki/SRTP). But still FreeSwitch does not use RTP/SAVP for leg-B.
Also, I am bit confused with following condition in "default.xml" when wiki page suggests that late coded negotiation must NOT to be turned on.
<!-- Since we have inbound-late-negotation on by default now the above behavior isn't the same so you have to do one extra step. -->
<condition field="${endpoint_disposition}" expression="^(DELAYED NEGOTIATION)"/> <condition field="${switch_r_sdp}" expression="(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)" break="never"> <action application="set" data="sip_secure_media=true"/> <!-- Offer SRTP on outbound legs if we have it on inbound. --> <action application="export" data="sip_secure_media=true"/> </condition>
By commenting/un-commenting "sip_secure_media=true" above, it did not worked.
Any help with this would be greatly appreciated.
-- Thanks CWM
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_________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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
*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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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)
-
Brian West
-
Chandrakant Marathe
-
Daniel Ivanov
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Michael Jerris