Hello, Setup is as following: X-Lite (Test phone) ------> SIP Proxy ------> FS (as SBC) ------> Carrier ------> End-user phone Whenever I place a call using X-Lite to the End-user phone, in this case my own mobile phone or office phone, and start sending DTMF tones from the remote telephone, the DMTF tones arrive with a crackling sound at the end of the tone. Console debug: 2013-08-23 09:00:13.813207 [DEBUG] switch_rtp.c:3829 Send start packet for [2] ts=3778476449 dur=160/160/1280 seq=5098 lw=-516490847 2013-08-23 09:00:13.833208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=320/320/1280 seq=5099 lw=161 2013-08-23 09:00:13.853208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=480/480/1280 seq=5100 lw=161 2013-08-23 09:00:13.873208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=640/640/1280 seq=5101 lw=161 2013-08-23 09:00:13.893208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=800/800/1280 seq=5102 lw=161 2013-08-23 09:00:13.913208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=960/960/1280 seq=5103 lw=161 2013-08-23 09:00:13.933208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=1120/1120/1280 seq=5104 lw=161 2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5105 lw=1 2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5106 lw=1 2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5107 lw=1 2013-08-23 09:00:14.813206 [DEBUG] switch_rtp.c:5117 RTP RECV DTMF 1:1280 2013-08-23 09:00:14.813206 [DEBUG] switch_channel.c:471 RECV DTMF 1:1280 Both internal and external sip profiles have the following configuration for DTMF: <param name="liberal-dtmf" value="true"/> <param name="rfc2833-pt" value="101"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="inbound-late-negotiation" value="true"/> Internal.xml (Dialplan) <extension name="internal_to_external"> <condition field="${sip_req_host}" expression="^10\.0\.0\.1$"> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="sofia/external/${sip_req_uri}"/> </condition> </extension> Is there a way to tune the DTMF tones? If so, which parameters do I need to add or edit? Thanks! Grant
Have you tried using Xlite on an IVR or something? Call your bank or your cable company and see if you can get through the menus fine. I'd bet that you can. Post back and let us know. They sound short but that should be ok. Thank you, Brian Foster Project Manager/Owner's Rep. Davri Investments, Inc. O: 317-787-2686 x2102 M: 317-600-9753 E: bdfoster@davri.com Indianapolis, Indiana Sent from a mobile device. On Aug 23, 2013 3:21 AM, "Grant Bagdasarian" <GB@cm.nl> wrote:
Hello,****
** **
Setup is as following:****
** **
X-Lite (Test phone) ------> SIP Proxy ------> FS (as SBC) ------> Carrier ------> End-user phone****
** **
Whenever I place a call using X-Lite to the End-user phone, in this case my own mobile phone or office phone, and start sending DTMF tones from the remote telephone, the DMTF tones arrive with a crackling sound at the end of the tone.****
** **
Console debug:****
2013-08-23 09:00:13.813207 [DEBUG] switch_rtp.c:3829 Send start packet for [2] ts=3778476449 dur=160/160/1280 seq=5098 lw=-516490847****
2013-08-23 09:00:13.833208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=320/320/1280 seq=5099 lw=161****
2013-08-23 09:00:13.853208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=480/480/1280 seq=5100 lw=161****
2013-08-23 09:00:13.873208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=640/640/1280 seq=5101 lw=161****
2013-08-23 09:00:13.893208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=800/800/1280 seq=5102 lw=161****
2013-08-23 09:00:13.913208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=960/960/1280 seq=5103 lw=161****
2013-08-23 09:00:13.933208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=1120/1120/1280 seq=5104 lw=161****
2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5105 lw=1****
2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5106 lw=1****
2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5107 lw=1****
2013-08-23 09:00:14.813206 [DEBUG] switch_rtp.c:5117 RTP RECV DTMF 1:1280* ***
2013-08-23 09:00:14.813206 [DEBUG] switch_channel.c:471 RECV DTMF 1:1280** **
** **
Both internal and external sip profiles have the following configuration for DTMF:****
<param name="liberal-dtmf" value="true"/>****
<param name="rfc2833-pt" value="101"/>****
<param name="dtmf-duration" value="2000"/>****
<param name="inbound-codec-negotiation" value="generous"/>****
<param name="inbound-late-negotiation" value="true"/>****
** **
Internal.xml (Dialplan)****
<extension name="internal_to_external">****
<condition field="${sip_req_host}" expression="^10\.0\.0\.1$">****
<action application="set" data="hangup_after_bridge=false"/>****
<action application="set" data="continue_on_fail=true"/>****
<action application="bridge" data="sofia/external/${sip_req_uri}"/> ****
</condition>****
</extension>****
** **
Is there a way to tune the DTMF tones? If so, which parameters do I need to add or edit?****
** **
Thanks!****
** **
Grant****
** **
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Yes, that works fine. The cracking DTMF tones only happen when the remote party is sending DTMF tones. So if I call my cellphone with X-Lite through FS and my cellphones starts sending DTMF tones, that's when the tones come in with some sort of cracking sound. From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian Foster Sent: Saturday, August 24, 2013 3:13 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Tuning DTMF Have you tried using Xlite on an IVR or something? Call your bank or your cable company and see if you can get through the menus fine. I'd bet that you can. Post back and let us know. They sound short but that should be ok. Thank you, Brian Foster Project Manager/Owner's Rep. Davri Investments, Inc. O: 317-787-2686 x2102 M: 317-600-9753 E: bdfoster@davri.com<mailto:bdfoster@davri.com> Indianapolis, Indiana Sent from a mobile device. On Aug 23, 2013 3:21 AM, "Grant Bagdasarian" <GB@cm.nl<mailto:GB@cm.nl>> wrote: Hello, Setup is as following: X-Lite (Test phone) ------> SIP Proxy ------> FS (as SBC) ------> Carrier ------> End-user phone Whenever I place a call using X-Lite to the End-user phone, in this case my own mobile phone or office phone, and start sending DTMF tones from the remote telephone, the DMTF tones arrive with a crackling sound at the end of the tone. Console debug: 2013-08-23 09:00:13.813207 [DEBUG] switch_rtp.c:3829 Send start packet for [2] ts=3778476449 dur=160/160/1280 seq=5098 lw=-516490847 2013-08-23 09:00:13.833208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=320/320/1280<tel:320%2F320%2F1280> seq=5099 lw=161 2013-08-23 09:00:13.853208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=480/480/1280<tel:480%2F480%2F1280> seq=5100 lw=161 2013-08-23 09:00:13.873208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=640/640/1280 seq=5101 lw=161 2013-08-23 09:00:13.893208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=800/800/1280<tel:800%2F800%2F1280> seq=5102 lw=161 2013-08-23 09:00:13.913208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=960/960/1280 seq=5103 lw=161 2013-08-23 09:00:13.933208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=1120/1120/1280 seq=5104 lw=161 2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5105 lw=1 2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5106 lw=1 2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5107 lw=1 2013-08-23 09:00:14.813206 [DEBUG] switch_rtp.c:5117 RTP RECV DTMF 1:1280 2013-08-23 09:00:14.813206 [DEBUG] switch_channel.c:471 RECV DTMF 1:1280 Both internal and external sip profiles have the following configuration for DTMF: <param name="liberal-dtmf" value="true"/> <param name="rfc2833-pt" value="101"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="inbound-late-negotiation" value="true"/> Internal.xml (Dialplan) <extension name="internal_to_external"> <condition field="${sip_req_host}" expression="^10\.0\.0\.1$"> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="sofia/external/${sip_req_uri}"/> </condition> </extension> Is there a way to tune the DTMF tones? If so, which parameters do I need to add or edit? Thanks! Grant _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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
Yep that's perfectly normal. As long as FS is interpreting the DTMF correctly you are good to go. As a quick test if you are still unsure call into FS and transfer to the demo ivr and see if it does what you expect. Thank you, Brian Foster Project Manager/Owner's Rep. Davri Investments, Inc. O: 317-787-2686 x2102 M: 317-600-9753 E: bdfoster@davri.com Indianapolis, Indiana Sent from a mobile device. On Aug 26, 2013 2:32 AM, "Grant Bagdasarian" <GB@cm.nl> wrote:
Yes, that works fine. The cracking DTMF tones only happen when the remote party is sending DTMF tones. So if I call my cellphone with X-Lite through FS and my cellphones starts sending DTMF tones, that’s when the tones come in with some sort of cracking sound.****
** **
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Brian Foster *Sent:* Saturday, August 24, 2013 3:13 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Tuning DTMF****
** **
Have you tried using Xlite on an IVR or something? Call your bank or your cable company and see if you can get through the menus fine. I'd bet that you can. Post back and let us know. They sound short but that should be ok. ****
Thank you,****
Brian Foster Project Manager/Owner's Rep. Davri Investments, Inc. O: 317-787-2686 x2102 M: 317-600-9753 E: bdfoster@davri.com Indianapolis, Indiana****
Sent from a mobile device.****
On Aug 23, 2013 3:21 AM, "Grant Bagdasarian" <GB@cm.nl> wrote:****
Hello,****
****
Setup is as following:****
****
X-Lite (Test phone) ------> SIP Proxy ------> FS (as SBC) ------> Carrier ------> End-user phone****
****
Whenever I place a call using X-Lite to the End-user phone, in this case my own mobile phone or office phone, and start sending DTMF tones from the remote telephone, the DMTF tones arrive with a crackling sound at the end of the tone.****
****
Console debug:****
2013-08-23 09:00:13.813207 [DEBUG] switch_rtp.c:3829 Send start packet for [2] ts=3778476449 dur=160/160/1280 seq=5098 lw=-516490847****
2013-08-23 09:00:13.833208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=320/320/1280 seq=5099 lw=161****
2013-08-23 09:00:13.853208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=480/480/1280 seq=5100 lw=161****
2013-08-23 09:00:13.873208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=640/640/1280 seq=5101 lw=161****
2013-08-23 09:00:13.893208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=800/800/1280 seq=5102 lw=161****
2013-08-23 09:00:13.913208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=960/960/1280 seq=5103 lw=161****
2013-08-23 09:00:13.933208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=1120/1120/1280 seq=5104 lw=161****
2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5105 lw=1****
2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5106 lw=1****
2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5107 lw=1****
2013-08-23 09:00:14.813206 [DEBUG] switch_rtp.c:5117 RTP RECV DTMF 1:1280* ***
2013-08-23 09:00:14.813206 [DEBUG] switch_channel.c:471 RECV DTMF 1:1280** **
****
Both internal and external sip profiles have the following configuration for DTMF:****
<param name="liberal-dtmf" value="true"/>****
<param name="rfc2833-pt" value="101"/>****
<param name="dtmf-duration" value="2000"/>****
<param name="inbound-codec-negotiation" value="generous"/>****
<param name="inbound-late-negotiation" value="true"/>****
****
Internal.xml (Dialplan)****
<extension name="internal_to_external">****
<condition field="${sip_req_host}" expression="^10\.0\.0\.1$">****
<action application="set" data="hangup_after_bridge=false"/>****
<action application="set" data="continue_on_fail=true"/>****
<action application="bridge" data="sofia/external/${sip_req_uri}"/> ****
</condition>****
</extension>****
****
Is there a way to tune the DTMF tones? If so, which parameters do I need to add or edit?****
****
Thanks!****
****
Grant****
****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Everything works perfectly! Thanks From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian Foster Sent: Monday, August 26, 2013 11:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Tuning DTMF Yep that's perfectly normal. As long as FS is interpreting the DTMF correctly you are good to go. As a quick test if you are still unsure call into FS and transfer to the demo ivr and see if it does what you expect. Thank you, Brian Foster Project Manager/Owner's Rep. Davri Investments, Inc. O: 317-787-2686 x2102 M: 317-600-9753 E: bdfoster@davri.com<mailto:bdfoster@davri.com> Indianapolis, Indiana Sent from a mobile device. On Aug 26, 2013 2:32 AM, "Grant Bagdasarian" <GB@cm.nl<mailto:GB@cm.nl>> wrote: Yes, that works fine. The cracking DTMF tones only happen when the remote party is sending DTMF tones. So if I call my cellphone with X-Lite through FS and my cellphones starts sending DTMF tones, that's when the tones come in with some sort of cracking sound. From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Brian Foster Sent: Saturday, August 24, 2013 3:13 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Tuning DTMF Have you tried using Xlite on an IVR or something? Call your bank or your cable company and see if you can get through the menus fine. I'd bet that you can. Post back and let us know. They sound short but that should be ok. Thank you, Brian Foster Project Manager/Owner's Rep. Davri Investments, Inc. O: 317-787-2686 x2102<tel:317-787-2686%20x2102> M: 317-600-9753<tel:317-600-9753> E: bdfoster@davri.com<mailto:bdfoster@davri.com> Indianapolis, Indiana Sent from a mobile device. On Aug 23, 2013 3:21 AM, "Grant Bagdasarian" <GB@cm.nl<mailto:GB@cm.nl>> wrote: Hello, Setup is as following: X-Lite (Test phone) ------> SIP Proxy ------> FS (as SBC) ------> Carrier ------> End-user phone Whenever I place a call using X-Lite to the End-user phone, in this case my own mobile phone or office phone, and start sending DTMF tones from the remote telephone, the DMTF tones arrive with a crackling sound at the end of the tone. Console debug: 2013-08-23 09:00:13.813207 [DEBUG] switch_rtp.c:3829 Send start packet for [2] ts=3778476449 dur=160/160/1280 seq=5098 lw=-516490847 2013-08-23 09:00:13.833208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=320/320/1280<tel:320%2F320%2F1280> seq=5099 lw=161 2013-08-23 09:00:13.853208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=480/480/1280<tel:480%2F480%2F1280> seq=5100 lw=161 2013-08-23 09:00:13.873208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=640/640/1280 seq=5101 lw=161 2013-08-23 09:00:13.893208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=800/800/1280<tel:800%2F800%2F1280> seq=5102 lw=161 2013-08-23 09:00:13.913208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=960/960/1280 seq=5103 lw=161 2013-08-23 09:00:13.933208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=1120/1120/1280 seq=5104 lw=161 2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5105 lw=1 2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5106 lw=1 2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5107 lw=1 2013-08-23 09:00:14.813206 [DEBUG] switch_rtp.c:5117 RTP RECV DTMF 1:1280 2013-08-23 09:00:14.813206 [DEBUG] switch_channel.c:471 RECV DTMF 1:1280 Both internal and external sip profiles have the following configuration for DTMF: <param name="liberal-dtmf" value="true"/> <param name="rfc2833-pt" value="101"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="inbound-late-negotiation" value="true"/> Internal.xml (Dialplan) <extension name="internal_to_external"> <condition field="${sip_req_host}" expression="^10\.0\.0\.1$"> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="sofia/external/${sip_req_uri}"/> </condition> </extension> Is there a way to tune the DTMF tones? If so, which parameters do I need to add or edit? Thanks! Grant _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 (2)
-
Brian Foster
-
Grant Bagdasarian