Freeswitch routing inbound calls over SIP instead of TLS/SRTP
Debian GNU/Linux 8 (jessie) Linux my 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) x86_64 GNU/Linux FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit) I have a Vtech handset with TLS/SRTP enabled registered with Freeswitch as below: Call-ID: a0000a0a000aa000 User: 2001@my.example.com Contact: "my" <sips:2001@198.51.100.81:58348> Agent: Vtech Vesa VSP736A 2.0.3.2-0 Status: Registered(TLS)(unknown) EXP(2016-11-18 10:56:57) EXPSECS(3646) Ping-Status: Reachable Ping-Time: 0.00 Host: my IP: 198.51.100.81 Port: 58348 Auth-User: 2001 Auth-Realm: my.example.com MWI-Account: 2001@my.example.com sofia_contact is happy : freeswitch@my>sofia_contact internal/2001 sofia/internal/sip:2001@198.51.100.81:58348 I have an inbound dial plan configured as follows: <include> <extension name=“test_inbound"> <condition field="destination_number" expression=“^(15550100)$"> <action application="set" data="domain_name=$${domain}"/> <action application="bridge" data="${sofia_contact(internal/2001)}"/> </condition> </extension> </include> The problem is Freeswitch is sending invites over SIP/RTP and not TLS/SRTP and so the calls never get through : INVITE sip:2001@198.51.100.81:58348 SIP/2.0 Via: SIP/2.0/UDP 203.0.113.4;rport;branch=z9hG4bKvHjgXXpFF77XK From: "Anonymous" <sip:anonymous@203.0.113.4>;tag=rKmXQjZN8SFXp To: <sip:2001@198.51.100.81:58348> m=audio 32190 RTP/AVP 8 98 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:98 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
Looks like a bug to me. Your first snippet shows the contact stored in the database uses the 'sips:' scheme, but sofia_contact is returning 'sip:' In the code it looks like sofia_contact fetches the contact using select_from_profile which invokes contact_callback. In contact_callback it's hardcoded to use sip: plus the result of sofia_glue_strip_proto. That looks to me like it can never return a sips URI even though it's stored in the database. I'd file a jira. Steve On 18 November 2016 at 10:08, Tim Smith <randomdev4@gmail.com> wrote:
Debian GNU/Linux 8 (jessie) Linux my 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) x86_64 GNU/Linux FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit)
I have a Vtech handset with TLS/SRTP enabled registered with Freeswitch as below:
Call-ID: a0000a0a000aa000 User: 2001@my.example.com Contact: "my" <sips:2001@198.51.100.81:58348> Agent: Vtech Vesa VSP736A 2.0.3.2-0 Status: Registered(TLS)(unknown) EXP(2016-11-18 10:56:57) EXPSECS(3646) Ping-Status: Reachable Ping-Time: 0.00 Host: my IP: 198.51.100.81 Port: 58348 Auth-User: 2001 Auth-Realm: my.example.com MWI-Account: 2001@my.example.com
sofia_contact is happy :
freeswitch@my>sofia_contact internal/2001 sofia/internal/sip:2001@198.51.100.81:58348
I have an inbound dial plan configured as follows:
<include> <extension name=“test_inbound"> <condition field="destination_number" expression=“^(15550100)$"> <action application="set" data="domain_name=$${domain}"/> <action application="bridge" data="${sofia_contact( internal/2001)}"/> </condition> </extension> </include>
The problem is Freeswitch is sending invites over SIP/RTP and not TLS/SRTP and so the calls never get through :
INVITE sip:2001@198.51.100.81:58348 SIP/2.0 Via: SIP/2.0/UDP 203.0.113.4;rport;branch=z9hG4bKvHjgXXpFF77XK From: "Anonymous" <sip:anonymous@203.0.113.4>;tag=rKmXQjZN8SFXp To: <sip:2001@198.51.100.81:58348> m=audio 32190 RTP/AVP 8 98 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:98 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Steve, Sorry for the delay ack'ing your mail ... yeah, guess I should maybe look into filing a JIRA. On 20 November 2016 at 00:33, Steven Ayre <steveayre@gmail.com> wrote:
Looks like a bug to me. Your first snippet shows the contact stored in the database uses the 'sips:' scheme, but sofia_contact is returning 'sip:'
In the code it looks like sofia_contact fetches the contact using select_from_profile which invokes contact_callback. In contact_callback it's hardcoded to use sip: plus the result of sofia_glue_strip_proto. That looks to me like it can never return a sips URI even though it's stored in the database.
I'd file a jira.
Steve
On 18 November 2016 at 10:08, Tim Smith <randomdev4@gmail.com> wrote:
Debian GNU/Linux 8 (jessie) Linux my 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) x86_64 GNU/Linux FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit)
I have a Vtech handset with TLS/SRTP enabled registered with Freeswitch as below:
Call-ID: a0000a0a000aa000 User: 2001@my.example.com Contact: "my" <sips:2001@198.51.100.81:58348> Agent: Vtech Vesa VSP736A 2.0.3.2-0 Status: Registered(TLS)(unknown) EXP(2016-11-18 10:56:57) EXPSECS(3646) Ping-Status: Reachable Ping-Time: 0.00 Host: my IP: 198.51.100.81 Port: 58348 Auth-User: 2001 Auth-Realm: my.example.com MWI-Account: 2001@my.example.com
sofia_contact is happy :
freeswitch@my>sofia_contact internal/2001 sofia/internal/sip:2001@198.51.100.81:58348
I have an inbound dial plan configured as follows:
<include> <extension name=“test_inbound"> <condition field="destination_number" expression=“^(15550100)$"> <action application="set" data="domain_name=$${domain}"/> <action application="bridge" data="${sofia_contact(internal/2001)}"/> </condition> </extension> </include>
The problem is Freeswitch is sending invites over SIP/RTP and not TLS/SRTP and so the calls never get through :
INVITE sip:2001@198.51.100.81:58348 SIP/2.0 Via: SIP/2.0/UDP 203.0.113.4;rport;branch=z9hG4bKvHjgXXpFF77XK From: "Anonymous" <sip:anonymous@203.0.113.4>;tag=rKmXQjZN8SFXp To: <sip:2001@198.51.100.81:58348> m=audio 32190 RTP/AVP 8 98 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:98 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
FreeSWITCH doesn't support sips, although I think attempts were made by bkw to get it supported. See if your client can get registered using SIP over TLS as opposed to using SIPS. Are you using SRV entries?
On Nov 21, 2016, at 5:39 PM, Tim Smith <randomdev4@gmail.com> wrote:
Hi Steve,
Sorry for the delay ack'ing your mail ... yeah, guess I should maybe look into filing a JIRA.
On 20 November 2016 at 00:33, Steven Ayre <steveayre@gmail.com> wrote:
Looks like a bug to me. Your first snippet shows the contact stored in the database uses the 'sips:' scheme, but sofia_contact is returning 'sip:'
In the code it looks like sofia_contact fetches the contact using select_from_profile which invokes contact_callback. In contact_callback it's hardcoded to use sip: plus the result of sofia_glue_strip_proto. That looks to me like it can never return a sips URI even though it's stored in the database.
I'd file a jira.
Steve
On 18 November 2016 at 10:08, Tim Smith <randomdev4@gmail.com> wrote:
Debian GNU/Linux 8 (jessie) Linux my 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) x86_64 GNU/Linux FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit)
I have a Vtech handset with TLS/SRTP enabled registered with Freeswitch as below:
Call-ID: a0000a0a000aa000 User: 2001@my.example.com Contact: "my" <sips:2001@198.51.100.81:58348> Agent: Vtech Vesa VSP736A 2.0.3.2-0 Status: Registered(TLS)(unknown) EXP(2016-11-18 10:56:57) EXPSECS(3646) Ping-Status: Reachable Ping-Time: 0.00 Host: my IP: 198.51.100.81 Port: 58348 Auth-User: 2001 Auth-Realm: my.example.com MWI-Account: 2001@my.example.com
sofia_contact is happy :
freeswitch@my>sofia_contact internal/2001 sofia/internal/sip:2001@198.51.100.81:58348
I have an inbound dial plan configured as follows:
<include> <extension name=“test_inbound"> <condition field="destination_number" expression=“^(15550100)$"> <action application="set" data="domain_name=$${domain}"/> <action application="bridge" data="${sofia_contact(internal/2001)}"/> </condition> </extension> </include>
The problem is Freeswitch is sending invites over SIP/RTP and not TLS/SRTP and so the calls never get through :
INVITE sip:2001@198.51.100.81:58348 SIP/2.0 Via: SIP/2.0/UDP 203.0.113.4;rport;branch=z9hG4bKvHjgXXpFF77XK From: "Anonymous" <sip:anonymous@203.0.113.4>;tag=rKmXQjZN8SFXp To: <sip:2001@198.51.100.81:58348> m=audio 32190 RTP/AVP 8 98 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:98 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Unfortunatley I don't think the vtech handsets offer many options in that respect. All the ports in the vtech config are set to 5060. Transport options are only UDP,TCP,TLS And SRTP is a boolean option. On 22 November 2016 at 17:49, Emrah <lists@kavun.ch> wrote:
FreeSWITCH doesn't support sips, although I think attempts were made by bkw to get it supported. See if your client can get registered using SIP over TLS as opposed to using SIPS. Are you using SRV entries?
On Nov 21, 2016, at 5:39 PM, Tim Smith <randomdev4@gmail.com> wrote:
Hi Steve,
Sorry for the delay ack'ing your mail ... yeah, guess I should maybe look into filing a JIRA.
On 20 November 2016 at 00:33, Steven Ayre <steveayre@gmail.com> wrote:
Looks like a bug to me. Your first snippet shows the contact stored in the database uses the 'sips:' scheme, but sofia_contact is returning 'sip:'
In the code it looks like sofia_contact fetches the contact using select_from_profile which invokes contact_callback. In contact_callback it's hardcoded to use sip: plus the result of sofia_glue_strip_proto. That looks to me like it can never return a sips URI even though it's stored in the database.
I'd file a jira.
Steve
On 18 November 2016 at 10:08, Tim Smith <randomdev4@gmail.com> wrote:
Debian GNU/Linux 8 (jessie) Linux my 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) x86_64 GNU/Linux FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit)
I have a Vtech handset with TLS/SRTP enabled registered with Freeswitch as below:
Call-ID: a0000a0a000aa000 User: 2001@my.example.com Contact: "my" <sips:2001@198.51.100.81:58348> Agent: Vtech Vesa VSP736A 2.0.3.2-0 Status: Registered(TLS)(unknown) EXP(2016-11-18 10:56:57) EXPSECS(3646) Ping-Status: Reachable Ping-Time: 0.00 Host: my IP: 198.51.100.81 Port: 58348 Auth-User: 2001 Auth-Realm: my.example.com MWI-Account: 2001@my.example.com
sofia_contact is happy :
freeswitch@my>sofia_contact internal/2001 sofia/internal/sip:2001@198.51.100.81:58348
I have an inbound dial plan configured as follows:
<include> <extension name=“test_inbound"> <condition field="destination_number" expression=“^(15550100)$"> <action application="set" data="domain_name=$${domain}"/> <action application="bridge" data="${sofia_contact(internal/2001)}"/> </condition> </extension> </include>
The problem is Freeswitch is sending invites over SIP/RTP and not TLS/SRTP and so the calls never get through :
INVITE sip:2001@198.51.100.81:58348 SIP/2.0 Via: SIP/2.0/UDP 203.0.113.4;rport;branch=z9hG4bKvHjgXXpFF77XK From: "Anonymous" <sip:anonymous@203.0.113.4>;tag=rKmXQjZN8SFXp To: <sip:2001@198.51.100.81:58348> m=audio 32190 RTP/AVP 8 98 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:98 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Actually, your sofia_contact is not happy.. I'm not seeing the transport param for tls in there. How many profiles are you running? Are you sure sofia_contact isn't giving you the value of another endpoint registered with UDP? Also, I went through your message fast, but I don't think you're securing your rtp...
On Nov 18, 2016, at 11:08 AM, Tim Smith <randomdev4@gmail.com> wrote:
Debian GNU/Linux 8 (jessie) Linux my 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) x86_64 GNU/Linux FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit)
I have a Vtech handset with TLS/SRTP enabled registered with Freeswitch as below:
Call-ID: a0000a0a000aa000 User: 2001@my.example.com Contact: "my" <sips:2001@198.51.100.81:58348> Agent: Vtech Vesa VSP736A 2.0.3.2-0 Status: Registered(TLS)(unknown) EXP(2016-11-18 10:56:57) EXPSECS(3646) Ping-Status: Reachable Ping-Time: 0.00 Host: my IP: 198.51.100.81 Port: 58348 Auth-User: 2001 Auth-Realm: my.example.com MWI-Account: 2001@my.example.com
sofia_contact is happy :
freeswitch@my>sofia_contact internal/2001 sofia/internal/sip:2001@198.51.100.81:58348
I have an inbound dial plan configured as follows:
<include> <extension name=“test_inbound"> <condition field="destination_number" expression=“^(15550100)$"> <action application="set" data="domain_name=$${domain}"/> <action application="bridge" data="${sofia_contact(internal/2001)}"/> </condition> </extension> </include>
The problem is Freeswitch is sending invites over SIP/RTP and not TLS/SRTP and so the calls never get through :
INVITE sip:2001@198.51.100.81:58348 SIP/2.0 Via: SIP/2.0/UDP 203.0.113.4;rport;branch=z9hG4bKvHjgXXpFF77XK From: "Anonymous" <sip:anonymous@203.0.113.4>;tag=rKmXQjZN8SFXp To: <sip:2001@198.51.100.81:58348> m=audio 32190 RTP/AVP 8 98 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:98 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Emrah, Its a test box. So one internal, one external, one endpoint .... no scope for confusion. ;-) On 20 November 2016 at 11:23, Emrah <lists@kavun.ch> wrote:
Actually, your sofia_contact is not happy.. I'm not seeing the transport param for tls in there. How many profiles are you running? Are you sure sofia_contact isn't giving you the value of another endpoint registered with UDP? Also, I went through your message fast, but I don't think you're securing your rtp...
On Nov 18, 2016, at 11:08 AM, Tim Smith <randomdev4@gmail.com> wrote:
Debian GNU/Linux 8 (jessie) Linux my 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) x86_64 GNU/Linux FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit)
I have a Vtech handset with TLS/SRTP enabled registered with Freeswitch as below:
Call-ID: a0000a0a000aa000 User: 2001@my.example.com Contact: "my" <sips:2001@198.51.100.81:58348> Agent: Vtech Vesa VSP736A 2.0.3.2-0 Status: Registered(TLS)(unknown) EXP(2016-11-18 10:56:57) EXPSECS(3646) Ping-Status: Reachable Ping-Time: 0.00 Host: my IP: 198.51.100.81 Port: 58348 Auth-User: 2001 Auth-Realm: my.example.com MWI-Account: 2001@my.example.com
sofia_contact is happy :
freeswitch@my>sofia_contact internal/2001 sofia/internal/sip:2001@198.51.100.81:58348
I have an inbound dial plan configured as follows:
<include> <extension name=“test_inbound"> <condition field="destination_number" expression=“^(15550100)$"> <action application="set" data="domain_name=$${domain}"/> <action application="bridge" data="${sofia_contact(internal/2001)}"/> </condition> </extension> </include>
The problem is Freeswitch is sending invites over SIP/RTP and not TLS/SRTP and so the calls never get through :
INVITE sip:2001@198.51.100.81:58348 SIP/2.0 Via: SIP/2.0/UDP 203.0.113.4;rport;branch=z9hG4bKvHjgXXpFF77XK From: "Anonymous" <sip:anonymous@203.0.113.4>;tag=rKmXQjZN8SFXp To: <sip:2001@198.51.100.81:58348> m=audio 32190 RTP/AVP 8 98 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:98 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (3)
-
Emrah
-
Steven Ayre
-
Tim Smith