Hello everyone, I have just gotten to know about FreeSWITCH and have tried installing it on a EC2 instance and in the process of trying out some examples. I am facing this issue of not being able to hear any sounds during a call I make to my FreeSWITCH system. My freeswitch system is the latest stable ver.1.4 and I followed the instructions to install from http://clintberry.com/2011/getting-started-with-plivo/ via the installation script show on the site. I changed the installation script to use Slash and also 1.4 instead. One of the exercises is to pass an incoming call to plivo and make plivo return a <say> action, which for some reason is not reading back the text to me (no audio) when I call 1005 from using sflphone on my ubuntu. This is the log on freeswitch: 2014-07-29 18:01:12.581530 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/1000@xxxxxxxxxx.amazonaws.com [5650b3dd-1519-4c6f-b720-d4d3a8a65c5c] 2014-07-29 18:01:12.681534 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->1005 in context default 2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established. 2014-07-29 18:01:12.733800 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@xxxxxxxxxx.amazonaws.com! 2014-07-29 18:01:12.733800 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/1000@xxxxxxxxxx.amazonaws.com] has been answered 2014-07-29 18:01:14.121516 [NOTICE] mod_dptools.c:1232 Hangup sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_RESET] [NORMAL_CLEARING] 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1632 Session 1 (sofia/internal/1000@xxxxxxxxxx.amazonaws.com) Ended 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1636 Close Channel sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_DESTROY] I can see that plivo is correctly calling the XML I have placed, as this is the log I see on my web server: [29/Jul/2014 01:59:13] "GET /answer/?To=1005&Direction=inbound&From=1000&CallerName=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&CallStatus=ringing HTTP/1.1" 200 93 [29/Jul/2014 01:59:14] "GET /answer/?To=1005&Direction=inbound&From=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&HangupCause=NORMAL_CLEARING&CallStatus=completed HTTP/1.1" 200 93 This is the XML I have prepared to test plivo: <?xml version="1.0" encoding="UTF-8" ?> <Response> <Speak>Hello World</Speak> </Response> And this is default.xml on freeswitch which forward any calls to 1005 to plivo: <include> <context name="default"> <!-- This extension allows calling any digits of number freeswitch will call plivo outbound server on every incoming call --> <extension name="plivo"> <condition field="destination_number" expression="^1005"> <action application="enable_heartbeat" data="60"/> <action application="socket" data="127.0.0.1:8084 async full"/> </condition> </extension> .... .... </include> I have read that NAT issues might be causing a no-audio issue, but I am not sure and do not know how to tell if this is the issue. The only thing which look strange is the logs on freeswitch saying: 2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established. but what does that mean?
Try explicitly answering the call before your <Speak/> action. On 29 July 2014 07:04, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> wrote:
Hello everyone,
I have just gotten to know about FreeSWITCH and have tried installing it on a EC2 instance and in the process of trying out some examples. I am facing this issue of not being able to hear any sounds during a call I make to my FreeSWITCH system.
My freeswitch system is the latest stable ver.1.4 and I followed the instructions to install from http://clintberry.com/2011/getting-started-with-plivo/
via the installation script show on the site. I changed the installation script to use Slash and also 1.4 instead.
One of the exercises is to pass an incoming call to plivo and make plivo return a <say> action, which for some reason is not reading back the text to me (no audio) when I call 1005 from using sflphone on my ubuntu.
This is the log on freeswitch:
2014-07-29 18:01:12.581530 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/1000@xxxxxxxxxx.amazonaws.com [5650b3dd-1519-4c6f-b720-d4d3a8a65c5c] 2014-07-29 18:01:12.681534 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->1005 in context default 2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established. 2014-07-29 18:01:12.733800 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@xxxxxxxxxx.amazonaws.com! 2014-07-29 18:01:12.733800 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/1000@xxxxxxxxxx.amazonaws.com] has been answered 2014-07-29 18:01:14.121516 [NOTICE] mod_dptools.c:1232 Hangup sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_RESET] [NORMAL_CLEARING] 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1632 Session 1 (sofia/internal/1000@xxxxxxxxxx.amazonaws.com) Ended 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1636 Close Channel sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_DESTROY]
I can see that plivo is correctly calling the XML I have placed, as this is the log I see on my web server:
[29/Jul/2014 01:59:13] "GET
/answer/?To=1005&Direction=inbound&From=1000&CallerName=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&CallStatus=ringing HTTP/1.1" 200 93 [29/Jul/2014 01:59:14] "GET
/answer/?To=1005&Direction=inbound&From=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&HangupCause=NORMAL_CLEARING&CallStatus=completed HTTP/1.1" 200 93
This is the XML I have prepared to test plivo:
<?xml version="1.0" encoding="UTF-8" ?> <Response> <Speak>Hello World</Speak> </Response>
And this is default.xml on freeswitch which forward any calls to 1005 to plivo:
<include> <context name="default">
<!-- This extension allows calling any digits of number freeswitch will call plivo outbound server on every incoming call --> <extension name="plivo"> <condition field="destination_number" expression="^1005"> <action application="enable_heartbeat" data="60"/> <action application="socket" data="127.0.0.1:8084 async full"/> </condition> </extension> .... .... </include>
I have read that NAT issues might be causing a no-audio issue, but I am not sure and do not know how to tell if this is the issue. The only thing which look strange is the logs on freeswitch saying:
2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established.
but what does that mean?
_________________________________________________________________________ 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
On 29 July 2014 21:54, Ben Langfeld <ben@langfeld.co.uk> wrote:
Try explicitly answering the call before your <Speak/> action.
Thanks Ben, I'm sorry but I do not see any explicit answer XML tag which I can use for Plivo. What do you mean by explicitly answering the call before the Speak action?
On 29 July 2014 07:04, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> wrote:
Hello everyone,
I have just gotten to know about FreeSWITCH and have tried installing it on a EC2 instance and in the process of trying out some examples. I am facing this issue of not being able to hear any sounds during a call I make to my FreeSWITCH system.
My freeswitch system is the latest stable ver.1.4 and I followed the instructions to install from http://clintberry.com/2011/getting-started-with-plivo/
via the installation script show on the site. I changed the installation script to use Slash and also 1.4 instead.
One of the exercises is to pass an incoming call to plivo and make plivo return a <say> action, which for some reason is not reading back the text to me (no audio) when I call 1005 from using sflphone on my ubuntu.
This is the log on freeswitch:
2014-07-29 18:01:12.581530 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/1000@xxxxxxxxxx.amazonaws.com [5650b3dd-1519-4c6f-b720-d4d3a8a65c5c] 2014-07-29 18:01:12.681534 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->1005 in context default 2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established. 2014-07-29 18:01:12.733800 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@xxxxxxxxxx.amazonaws.com! 2014-07-29 18:01:12.733800 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/1000@xxxxxxxxxx.amazonaws.com] has been answered 2014-07-29 18:01:14.121516 [NOTICE] mod_dptools.c:1232 Hangup sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_RESET] [NORMAL_CLEARING] 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1632 Session 1 (sofia/internal/1000@xxxxxxxxxx.amazonaws.com) Ended 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1636 Close Channel sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_DESTROY]
I can see that plivo is correctly calling the XML I have placed, as this is the log I see on my web server:
[29/Jul/2014 01:59:13] "GET
/answer/?To=1005&Direction=inbound&From=1000&CallerName=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&CallStatus=ringing HTTP/1.1" 200 93 [29/Jul/2014 01:59:14] "GET
/answer/?To=1005&Direction=inbound&From=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&HangupCause=NORMAL_CLEARING&CallStatus=completed HTTP/1.1" 200 93
This is the XML I have prepared to test plivo:
<?xml version="1.0" encoding="UTF-8" ?> <Response> <Speak>Hello World</Speak> </Response>
And this is default.xml on freeswitch which forward any calls to 1005 to plivo:
<include> <context name="default">
<!-- This extension allows calling any digits of number freeswitch will call plivo outbound server on every incoming call --> <extension name="plivo"> <condition field="destination_number" expression="^1005"> <action application="enable_heartbeat" data="60"/> <action application="socket" data="127.0.0.1:8084 async full"/> </condition> </extension> .... .... </include>
I have read that NAT issues might be causing a no-audio issue, but I am not sure and do not know how to tell if this is the issue. The only thing which look strange is the logs on freeswitch saying:
2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established.
but what does that mean?
_________________________________________________________________________ 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
On a hunch, I launched an instance, set the firewall up properly. Checked out, built and installed FS, No issues at all. ;) Just making sure someone else was reporting issues so I thought I would spin it up to double check a few things. On Tue, Jul 29, 2014 at 9:12 AM, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> wrote:
On 29 July 2014 21:54, Ben Langfeld <ben@langfeld.co.uk> wrote:
Try explicitly answering the call before your <Speak/> action.
Thanks Ben,
I'm sorry but I do not see any explicit answer XML tag which I can use for Plivo. What do you mean by explicitly answering the call before the Speak action?
On 29 July 2014 07:04, Iqbal Abdullah <iqbal.b.abdullah@gmail.com>
wrote:
Hello everyone,
I have just gotten to know about FreeSWITCH and have tried installing it on a EC2 instance and in the process of trying out some examples. I am facing this issue of not being able to hear any sounds during a call I make to my FreeSWITCH system.
My freeswitch system is the latest stable ver.1.4 and I followed the instructions to install from http://clintberry.com/2011/getting-started-with-plivo/
via the installation script show on the site. I changed the installation script to use Slash and also 1.4 instead.
One of the exercises is to pass an incoming call to plivo and make plivo return a <say> action, which for some reason is not reading back the text to me (no audio) when I call 1005 from using sflphone on my ubuntu.
This is the log on freeswitch:
2014-07-29 18:01:12.581530 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/1000@xxxxxxxxxx.amazonaws.com [5650b3dd-1519-4c6f-b720-d4d3a8a65c5c] 2014-07-29 18:01:12.681534 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->1005 in context default 2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established. 2014-07-29 18:01:12.733800 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@xxxxxxxxxx.amazonaws.com! 2014-07-29 18:01:12.733800 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/1000@xxxxxxxxxx.amazonaws.com] has been answered 2014-07-29 18:01:14.121516 [NOTICE] mod_dptools.c:1232 Hangup sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_RESET] [NORMAL_CLEARING] 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1632 Session 1 (sofia/internal/1000@xxxxxxxxxx.amazonaws.com) Ended 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1636 Close Channel sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_DESTROY]
I can see that plivo is correctly calling the XML I have placed, as this is the log I see on my web server:
[29/Jul/2014 01:59:13] "GET
/answer/?To=1005&Direction=inbound&From=1000&CallerName=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&CallStatus=ringing
HTTP/1.1" 200 93 [29/Jul/2014 01:59:14] "GET
/answer/?To=1005&Direction=inbound&From=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&HangupCause=NORMAL_CLEARING&CallStatus=completed
HTTP/1.1" 200 93
This is the XML I have prepared to test plivo:
<?xml version="1.0" encoding="UTF-8" ?> <Response> <Speak>Hello World</Speak> </Response>
And this is default.xml on freeswitch which forward any calls to 1005 to plivo:
<include> <context name="default">
<!-- This extension allows calling any digits of number freeswitch will call plivo outbound server on every incoming call --> <extension name="plivo"> <condition field="destination_number" expression="^1005"> <action application="enable_heartbeat" data="60"/> <action application="socket" data="127.0.0.1:8084 async full"/> </condition> </extension> .... .... </include>
I have read that NAT issues might be causing a no-audio issue, but I am not sure and do not know how to tell if this is the issue. The only thing which look strange is the logs on freeswitch saying:
2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established.
but what does that mean?
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
_________________________________________________________________________ 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
-- *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
Brian, On 30 July 2014 00:06, Brian West <brian@freeswitch.org> wrote:
On a hunch, I launched an instance, set the firewall up properly. Checked out, built and installed FS, No issues at all. ;) Just making sure someone else was reporting issues so I thought I would spin it up to double check a few things.
Thanks for looking it up. Can you tell me what ports of the firewall did you open up? Maybe that's the issue. I only have 5060 (and ssh) open by the way.
On Tue, Jul 29, 2014 at 9:12 AM, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> wrote:
On 29 July 2014 21:54, Ben Langfeld <ben@langfeld.co.uk> wrote:
Try explicitly answering the call before your <Speak/> action.
Thanks Ben,
I'm sorry but I do not see any explicit answer XML tag which I can use for Plivo. What do you mean by explicitly answering the call before the Speak action?
On 29 July 2014 07:04, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> wrote:
Hello everyone,
I have just gotten to know about FreeSWITCH and have tried installing it on a EC2 instance and in the process of trying out some examples. I am facing this issue of not being able to hear any sounds during a call I make to my FreeSWITCH system.
My freeswitch system is the latest stable ver.1.4 and I followed the instructions to install from http://clintberry.com/2011/getting-started-with-plivo/
via the installation script show on the site. I changed the installation script to use Slash and also 1.4 instead.
One of the exercises is to pass an incoming call to plivo and make plivo return a <say> action, which for some reason is not reading back the text to me (no audio) when I call 1005 from using sflphone on my ubuntu.
This is the log on freeswitch:
2014-07-29 18:01:12.581530 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/1000@xxxxxxxxxx.amazonaws.com [5650b3dd-1519-4c6f-b720-d4d3a8a65c5c] 2014-07-29 18:01:12.681534 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->1005 in context default 2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established. 2014-07-29 18:01:12.733800 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@xxxxxxxxxx.amazonaws.com! 2014-07-29 18:01:12.733800 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/1000@xxxxxxxxxx.amazonaws.com] has been answered 2014-07-29 18:01:14.121516 [NOTICE] mod_dptools.c:1232 Hangup sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_RESET] [NORMAL_CLEARING] 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1632 Session 1 (sofia/internal/1000@xxxxxxxxxx.amazonaws.com) Ended 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1636 Close Channel sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_DESTROY]
I can see that plivo is correctly calling the XML I have placed, as this is the log I see on my web server:
[29/Jul/2014 01:59:13] "GET
/answer/?To=1005&Direction=inbound&From=1000&CallerName=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&CallStatus=ringing HTTP/1.1" 200 93 [29/Jul/2014 01:59:14] "GET
/answer/?To=1005&Direction=inbound&From=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&HangupCause=NORMAL_CLEARING&CallStatus=completed HTTP/1.1" 200 93
This is the XML I have prepared to test plivo:
<?xml version="1.0" encoding="UTF-8" ?> <Response> <Speak>Hello World</Speak> </Response>
And this is default.xml on freeswitch which forward any calls to 1005 to plivo:
<include> <context name="default">
<!-- This extension allows calling any digits of number freeswitch will call plivo outbound server on every incoming call --> <extension name="plivo"> <condition field="destination_number" expression="^1005"> <action application="enable_heartbeat" data="60"/> <action application="socket" data="127.0.0.1:8084 async full"/> </condition> </extension> .... .... </include>
I have read that NAT issues might be causing a no-audio issue, but I am not sure and do not know how to tell if this is the issue. The only thing which look strange is the logs on freeswitch saying:
2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established.
but what does that mean?
_________________________________________________________________________ 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
_________________________________________________________________________ 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
--
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
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
Well that's probably your problem. You need to open up ports for media (RTP) to flow. If your setup is still default, then you'll need to open up ports 16384-32768 See https://wiki.freeswitch.org/wiki/Firewall On Tue, Jul 29, 2014 at 6:33 PM, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> wrote:
Brian,
On 30 July 2014 00:06, Brian West <brian@freeswitch.org> wrote:
On a hunch, I launched an instance, set the firewall up properly.
Checked out, built and installed FS, No issues at all. ;) Just making sure someone else was reporting issues so I thought I would spin it up to double check a few things.
Thanks for looking it up. Can you tell me what ports of the firewall did you open up? Maybe that's the issue. I only have 5060 (and ssh) open by the way.
On Tue, Jul 29, 2014 at 9:12 AM, Iqbal Abdullah <
iqbal.b.abdullah@gmail.com> wrote:
On 29 July 2014 21:54, Ben Langfeld <ben@langfeld.co.uk> wrote:
Try explicitly answering the call before your <Speak/> action.
Thanks Ben,
I'm sorry but I do not see any explicit answer XML tag which I can use for Plivo. What do you mean by explicitly answering the call before the Speak
action?
On 29 July 2014 07:04, Iqbal Abdullah <iqbal.b.abdullah@gmail.com>
wrote:
Hello everyone,
I have just gotten to know about FreeSWITCH and have tried installing it on a EC2 instance and in the process of trying out some examples.
I
am facing this issue of not being able to hear any sounds during a call I make to my FreeSWITCH system.
My freeswitch system is the latest stable ver.1.4 and I followed the instructions to install from http://clintberry.com/2011/getting-started-with-plivo/
via the installation script show on the site. I changed the installation script to use Slash and also 1.4 instead.
One of the exercises is to pass an incoming call to plivo and make plivo return a <say> action, which for some reason is not reading back the text to me (no audio) when I call 1005 from using sflphone on my ubuntu.
This is the log on freeswitch:
2014-07-29 18:01:12.581530 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/1000@xxxxxxxxxx.amazonaws.com [5650b3dd-1519-4c6f-b720-d4d3a8a65c5c] 2014-07-29 18:01:12.681534 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->1005 in context default 2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established. 2014-07-29 18:01:12.733800 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@xxxxxxxxxx.amazonaws.com! 2014-07-29 18:01:12.733800 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/1000@xxxxxxxxxx.amazonaws.com] has been answered 2014-07-29 18:01:14.121516 [NOTICE] mod_dptools.c:1232 Hangup sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_RESET] [NORMAL_CLEARING] 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1632 Session 1 (sofia/internal/1000@xxxxxxxxxx.amazonaws.com) Ended 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1636 Close Channel sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_DESTROY]
I can see that plivo is correctly calling the XML I have placed, as this is the log I see on my web server:
[29/Jul/2014 01:59:13] "GET
/answer/?To=1005&Direction=inbound&From=1000&CallerName=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&CallStatus=ringing
HTTP/1.1" 200 93 [29/Jul/2014 01:59:14] "GET
/answer/?To=1005&Direction=inbound&From=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&HangupCause=NORMAL_CLEARING&CallStatus=completed
HTTP/1.1" 200 93
This is the XML I have prepared to test plivo:
<?xml version="1.0" encoding="UTF-8" ?> <Response> <Speak>Hello World</Speak> </Response>
And this is default.xml on freeswitch which forward any calls to 1005 to plivo:
<include> <context name="default">
<!-- This extension allows calling any digits of number freeswitch will call plivo outbound server on every incoming call --> <extension name="plivo"> <condition field="destination_number" expression="^1005"> <action application="enable_heartbeat" data="60"/> <action application="socket" data="127.0.0.1:8084 async full"/> </condition> </extension> .... .... </include>
I have read that NAT issues might be causing a no-audio issue, but I am not sure and do not know how to tell if this is the issue. The only thing which look strange is the logs on freeswitch saying:
2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established.
but what does that mean?
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
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
--
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
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
Thank you Peter and Robert, I opened up the suggested ports on the wiki page but that didn't work, so I ended up opening all TCP and UDP ports but the audio doesn't come either. Trying to call a different number (1001, which doesn't have a user) gave me this logs: 2014-07-30 09:42:54.121512 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/1000@xxxxxx.amazonaws.com [34c19686-9e28-459d-813c-fbf57d42c89a] 2014-07-30 09:42:54.261504 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->1001 in context default 2014-07-30 09:42:54.261504 [INFO] switch_ivr_async.c:3653 Bound B-Leg: *1 execute_extension::dx XML features 2014-07-30 09:42:54.261504 [INFO] switch_ivr_async.c:3653 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1000.2014-07-30-09-42-54.wav 2014-07-30 09:42:54.261504 [INFO] switch_ivr_async.c:3653 Bound B-Leg: *3 execute_extension::cf XML features 2014-07-30 09:42:54.261504 [INFO] switch_ivr_async.c:3653 Bound B-Leg: *4 execute_extension::att_xfer XML features 2014-07-30 09:42:54.281497 [NOTICE] switch_ivr_originate.c:2708 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2014-07-30 09:42:54.281497 [NOTICE] switch_ivr_originate.c:2708 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2014-07-30 09:42:54.281497 [INFO] mod_dptools.c:3234 Originate Failed. Cause: USER_NOT_REGISTERED 2014-07-30 09:42:54.281497 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@xxxxxx.amazonaws.com! 2014-07-30 09:42:54.281497 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/1000@xxxxxx.amazonaws.com] has been answered 2014-07-30 09:42:55.281513 [NOTICE] switch_channel.c:1054 New Channel loopback/app=voicemail:default 10.142.199.62 1001-a [e83b067a-9687-4fe2-8691-ea0f4c8a04a3] 2014-07-30 09:42:55.281513 [NOTICE] switch_channel.c:1052 Rename Channel loopback/app=voicemail:default 10.142.199.62 1001-a->loopback/voicemail-a [e83b067a-9687-4fe2-8691-ea0f4c8a04a3] 2014-07-30 09:42:55.281513 [NOTICE] switch_channel.c:1054 New Channel loopback/voicemail-b [cba53c4a-07c6-4c5a-8bf1-4d4029b6cb66] 2014-07-30 09:42:55.281513 [NOTICE] mod_loopback.c:947 Pre-Answer loopback/voicemail-a! 2014-07-30 09:42:55.281513 [NOTICE] mod_dptools.c:1293 Pre-Answer loopback/voicemail-b! 2014-07-30 09:43:26.301488 [NOTICE] sofia.c:7256 Hangup sofia/internal/1000@xxxxxx.amazonaws.com [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2014-07-30 09:43:26.301488 [NOTICE] switch_ivr_bridge.c:752 Hangup loopback/voicemail-a [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL] 2014-07-30 09:43:26.301488 [NOTICE] mod_loopback.c:553 Hangup loopback/voicemail-b [CS_EXECUTE] [ORIGINATOR_CANCEL] 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1632 Session 10 (sofia/internal/1000@xxxxxx.amazonaws.com) Ended 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1636 Close Channel sofia/internal/1000@xxxxxx.amazonaws.com [CS_DESTROY] 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1632 Session 12 (loopback/voicemail-b) Ended 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1636 Close Channel loopback/voicemail-b [CS_DESTROY] 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1632 Session 11 (loopback/voicemail-a) Ended 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1636 Close Channel loopback/voicemail-a [CS_DESTROY] Doesn't seem to have any issues with it at all. On 30 July 2014 02:55, Peter Villeneuve <petervnv1@gmail.com> wrote:
Well that's probably your problem. You need to open up ports for media (RTP) to flow.
If your setup is still default, then you'll need to open up ports 16384-32768 See https://wiki.freeswitch.org/wiki/Firewall
On Tue, Jul 29, 2014 at 6:33 PM, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> wrote:
Brian,
On 30 July 2014 00:06, Brian West <brian@freeswitch.org> wrote:
On a hunch, I launched an instance, set the firewall up properly. Checked out, built and installed FS, No issues at all. ;) Just making sure someone else was reporting issues so I thought I would spin it up to double check a few things.
Thanks for looking it up. Can you tell me what ports of the firewall did you open up? Maybe that's the issue. I only have 5060 (and ssh) open by the way.
On Tue, Jul 29, 2014 at 9:12 AM, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> wrote:
On 29 July 2014 21:54, Ben Langfeld <ben@langfeld.co.uk> wrote:
Try explicitly answering the call before your <Speak/> action.
Thanks Ben,
I'm sorry but I do not see any explicit answer XML tag which I can use for Plivo. What do you mean by explicitly answering the call before the Speak action?
On 29 July 2014 07:04, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> wrote:
Hello everyone,
I have just gotten to know about FreeSWITCH and have tried installing it on a EC2 instance and in the process of trying out some examples. I am facing this issue of not being able to hear any sounds during a call I make to my FreeSWITCH system.
My freeswitch system is the latest stable ver.1.4 and I followed the instructions to install from http://clintberry.com/2011/getting-started-with-plivo/
via the installation script show on the site. I changed the installation script to use Slash and also 1.4 instead.
One of the exercises is to pass an incoming call to plivo and make plivo return a <say> action, which for some reason is not reading back the text to me (no audio) when I call 1005 from using sflphone on my ubuntu.
This is the log on freeswitch:
2014-07-29 18:01:12.581530 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/1000@xxxxxxxxxx.amazonaws.com [5650b3dd-1519-4c6f-b720-d4d3a8a65c5c] 2014-07-29 18:01:12.681534 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->1005 in context default 2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established. 2014-07-29 18:01:12.733800 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@xxxxxxxxxx.amazonaws.com! 2014-07-29 18:01:12.733800 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/1000@xxxxxxxxxx.amazonaws.com] has been answered 2014-07-29 18:01:14.121516 [NOTICE] mod_dptools.c:1232 Hangup sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_RESET] [NORMAL_CLEARING] 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1632 Session 1 (sofia/internal/1000@xxxxxxxxxx.amazonaws.com) Ended 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1636 Close Channel sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_DESTROY]
I can see that plivo is correctly calling the XML I have placed, as this is the log I see on my web server:
[29/Jul/2014 01:59:13] "GET
/answer/?To=1005&Direction=inbound&From=1000&CallerName=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&CallStatus=ringing HTTP/1.1" 200 93 [29/Jul/2014 01:59:14] "GET
/answer/?To=1005&Direction=inbound&From=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&HangupCause=NORMAL_CLEARING&CallStatus=completed HTTP/1.1" 200 93
This is the XML I have prepared to test plivo:
<?xml version="1.0" encoding="UTF-8" ?> <Response> <Speak>Hello World</Speak> </Response>
And this is default.xml on freeswitch which forward any calls to 1005 to plivo:
<include> <context name="default">
<!-- This extension allows calling any digits of number freeswitch will call plivo outbound server on every incoming call --> <extension name="plivo"> <condition field="destination_number" expression="^1005"> <action application="enable_heartbeat" data="60"/> <action application="socket" data="127.0.0.1:8084 async full"/> </condition> </extension> .... .... </include>
I have read that NAT issues might be causing a no-audio issue, but I am not sure and do not know how to tell if this is the issue. The only thing which look strange is the logs on freeswitch saying:
2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established.
but what does that mean?
_________________________________________________________________________ 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
_________________________________________________________________________ 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
--
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
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
_________________________________________________________________________ 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
Take a look in the ./conf/sip_profiles/external.xml file an in there you'll find variables for the external rtp ip. I'm betting you haven't set the external ip in the ./conf/vars.xml file. See here for more info: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#ext-rtp-ip William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king@quentustech.com On 07/29/2014 05:49 PM, Iqbal Abdullah wrote:
Thank you Peter and Robert,
I opened up the suggested ports on the wiki page but that didn't work, so I ended up opening all TCP and UDP ports but the audio doesn't come either.
Trying to call a different number (1001, which doesn't have a user) gave me this logs:
2014-07-30 09:42:54.121512 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/1000@xxxxxx.amazonaws.com [34c19686-9e28-459d-813c-fbf57d42c89a] 2014-07-30 09:42:54.261504 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->1001 in context default 2014-07-30 09:42:54.261504 [INFO] switch_ivr_async.c:3653 Bound B-Leg: *1 execute_extension::dx XML features 2014-07-30 09:42:54.261504 [INFO] switch_ivr_async.c:3653 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1000.2014-07-30-09-42-54.wav 2014-07-30 09:42:54.261504 [INFO] switch_ivr_async.c:3653 Bound B-Leg: *3 execute_extension::cf XML features 2014-07-30 09:42:54.261504 [INFO] switch_ivr_async.c:3653 Bound B-Leg: *4 execute_extension::att_xfer XML features 2014-07-30 09:42:54.281497 [NOTICE] switch_ivr_originate.c:2708 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2014-07-30 09:42:54.281497 [NOTICE] switch_ivr_originate.c:2708 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2014-07-30 09:42:54.281497 [INFO] mod_dptools.c:3234 Originate Failed. Cause: USER_NOT_REGISTERED 2014-07-30 09:42:54.281497 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@xxxxxx.amazonaws.com! 2014-07-30 09:42:54.281497 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/1000@xxxxxx.amazonaws.com] has been answered 2014-07-30 09:42:55.281513 [NOTICE] switch_channel.c:1054 New Channel loopback/app=voicemail:default 10.142.199.62 1001-a [e83b067a-9687-4fe2-8691-ea0f4c8a04a3] 2014-07-30 09:42:55.281513 [NOTICE] switch_channel.c:1052 Rename Channel loopback/app=voicemail:default 10.142.199.62 1001-a->loopback/voicemail-a [e83b067a-9687-4fe2-8691-ea0f4c8a04a3] 2014-07-30 09:42:55.281513 [NOTICE] switch_channel.c:1054 New Channel loopback/voicemail-b [cba53c4a-07c6-4c5a-8bf1-4d4029b6cb66] 2014-07-30 09:42:55.281513 [NOTICE] mod_loopback.c:947 Pre-Answer loopback/voicemail-a! 2014-07-30 09:42:55.281513 [NOTICE] mod_dptools.c:1293 Pre-Answer loopback/voicemail-b! 2014-07-30 09:43:26.301488 [NOTICE] sofia.c:7256 Hangup sofia/internal/1000@xxxxxx.amazonaws.com [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2014-07-30 09:43:26.301488 [NOTICE] switch_ivr_bridge.c:752 Hangup loopback/voicemail-a [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL] 2014-07-30 09:43:26.301488 [NOTICE] mod_loopback.c:553 Hangup loopback/voicemail-b [CS_EXECUTE] [ORIGINATOR_CANCEL] 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1632 Session 10 (sofia/internal/1000@xxxxxx.amazonaws.com) Ended 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1636 Close Channel sofia/internal/1000@xxxxxx.amazonaws.com [CS_DESTROY] 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1632 Session 12 (loopback/voicemail-b) Ended 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1636 Close Channel loopback/voicemail-b [CS_DESTROY] 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1632 Session 11 (loopback/voicemail-a) Ended 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1636 Close Channel loopback/voicemail-a [CS_DESTROY]
Doesn't seem to have any issues with it at all.
On 30 July 2014 02:55, Peter Villeneuve <petervnv1@gmail.com> wrote:
Well that's probably your problem. You need to open up ports for media (RTP) to flow.
If your setup is still default, then you'll need to open up ports 16384-32768 See https://wiki.freeswitch.org/wiki/Firewall
On Tue, Jul 29, 2014 at 6:33 PM, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> wrote:
Brian,
On 30 July 2014 00:06, Brian West <brian@freeswitch.org> wrote:
On a hunch, I launched an instance, set the firewall up properly. Checked out, built and installed FS, No issues at all. ;) Just making sure someone else was reporting issues so I thought I would spin it up to double check a few things.
Thanks for looking it up. Can you tell me what ports of the firewall did you open up? Maybe that's the issue. I only have 5060 (and ssh) open by the way.
On Tue, Jul 29, 2014 at 9:12 AM, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> wrote:
On 29 July 2014 21:54, Ben Langfeld <ben@langfeld.co.uk> wrote:
Try explicitly answering the call before your <Speak/> action.
Thanks Ben,
I'm sorry but I do not see any explicit answer XML tag which I can use for Plivo. What do you mean by explicitly answering the call before the Speak action?
On 29 July 2014 07:04, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> wrote: > > Hello everyone, > > I have just gotten to know about FreeSWITCH and have tried > installing > it on a EC2 instance and in the process of trying out some examples. > I > am facing this issue of not being able to hear any sounds during a > call I make to my FreeSWITCH system. > > My freeswitch system is the latest stable ver.1.4 and I followed the > instructions to install from > http://clintberry.com/2011/getting-started-with-plivo/ > > via the installation script show on the site. I changed the > installation script to use Slash and also 1.4 instead. > > One of the exercises is to pass an incoming call to plivo and make > plivo return a <say> action, which for some reason is not reading > back > the text to me (no audio) when I call 1005 from using sflphone on my > ubuntu. > > This is the log on freeswitch: > > 2014-07-29 18:01:12.581530 [NOTICE] switch_channel.c:1054 New > Channel > sofia/internal/1000@xxxxxxxxxx.amazonaws.com > [5650b3dd-1519-4c6f-b720-d4d3a8a65c5c] > 2014-07-29 18:01:12.681534 [INFO] mod_dialplan_xml.c:558 Processing > 1000 <1000>->1005 in context default > 2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 > sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to > bypass media or media is not established. > 2014-07-29 18:01:12.733800 [NOTICE] sofia_media.c:92 Pre-Answer > sofia/internal/1000@xxxxxxxxxx.amazonaws.com! > 2014-07-29 18:01:12.733800 [NOTICE] mod_dptools.c:1258 Channel > [sofia/internal/1000@xxxxxxxxxx.amazonaws.com] has been answered > 2014-07-29 18:01:14.121516 [NOTICE] mod_dptools.c:1232 Hangup > sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_RESET] > [NORMAL_CLEARING] > 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1632 > Session > 1 (sofia/internal/1000@xxxxxxxxxx.amazonaws.com) Ended > 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1636 Close > Channel sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_DESTROY] > > I can see that plivo is correctly calling the XML I have placed, as > this is the log I see on my web server: > > [29/Jul/2014 01:59:13] "GET > > > /answer/?To=1005&Direction=inbound&From=1000&CallerName=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&CallStatus=ringing > HTTP/1.1" 200 93 > [29/Jul/2014 01:59:14] "GET > > > /answer/?To=1005&Direction=inbound&From=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&HangupCause=NORMAL_CLEARING&CallStatus=completed > HTTP/1.1" 200 93 > > This is the XML I have prepared to test plivo: > > <?xml version="1.0" encoding="UTF-8" ?> > <Response> > <Speak>Hello World</Speak> > </Response> > > > And this is default.xml on freeswitch which forward any calls to > 1005 to > plivo: > > <include> > <context name="default"> > > <!-- > This extension allows calling any digits of number > freeswitch will call plivo outbound server on every incoming > call > --> > <extension name="plivo"> > <condition field="destination_number" expression="^1005"> > <action application="enable_heartbeat" data="60"/> > <action application="socket" data="127.0.0.1:8084 async > full"/> > </condition> > </extension> > .... > .... > </include> > > I have read that NAT issues might be causing a no-audio issue, but I > am not sure and do not know how to tell if this is the issue. The > only > thing which look strange is the logs on freeswitch saying: > > 2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 > sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to > bypass media or media is not established. > > but what does that mean? > > > _________________________________________________________________________ > 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
_________________________________________________________________________ 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
--
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
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
_________________________________________________________________________ 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
Thanks William, As you pointed out, the IPs were not changed in vars.xml I have then changed 3 settings to use the EC2's instance public IP: <X-PRE-PROCESS cmd="set" data="external_sip_ip=xxx.xxx.xxx.xxx"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=xxx.xxx.xxx.xxx"/> <X-PRE-PROCESS cmd="set" data="bind_server_ip=xxx.xxx.xxx.xxx"/> After shutting down and restarting FS, I tried calling again but still cannot get audio. If it helps, these are the status of sofia's external and internal profiles: freeswitch@ip-10-142-199-62> sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 10.142.199.62 SIP-IP 10.142.199.62 URL sip:mod_sofia@10.142.199.62:5080 BIND-URL sip:mod_sofia@10.142.199.62:5080;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN OPUS,G722,PCMU,PCMA,GSM CODECS OUT PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 0 freeswitch@ip-10-142-199-62> sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts 10.142.199.62,10.142.199.62 Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.142.199.62 SIP-IP 10.142.199.62 URL sip:mod_sofia@10.142.199.62:5060 BIND-URL sip:mod_sofia@10.142.199.62:5060;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN OPUS,G722,PCMU,PCMA,GSM CODECS OUT OPUS,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 2 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 1 On 30 July 2014 14:56, William King <william.king@quentustech.com> wrote:
Take a look in the ./conf/sip_profiles/external.xml file an in there you'll find variables for the external rtp ip. I'm betting you haven't set the external ip in the ./conf/vars.xml file. See here for more info: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#ext-rtp-ip
William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king@quentustech.com
On 07/29/2014 05:49 PM, Iqbal Abdullah wrote:
Thank you Peter and Robert,
I opened up the suggested ports on the wiki page but that didn't work, so I ended up opening all TCP and UDP ports but the audio doesn't come either.
Trying to call a different number (1001, which doesn't have a user) gave me this logs:
2014-07-30 09:42:54.121512 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/1000@xxxxxx.amazonaws.com [34c19686-9e28-459d-813c-fbf57d42c89a] 2014-07-30 09:42:54.261504 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->1001 in context default 2014-07-30 09:42:54.261504 [INFO] switch_ivr_async.c:3653 Bound B-Leg: *1 execute_extension::dx XML features 2014-07-30 09:42:54.261504 [INFO] switch_ivr_async.c:3653 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1000.2014-07-30-09-42-54.wav 2014-07-30 09:42:54.261504 [INFO] switch_ivr_async.c:3653 Bound B-Leg: *3 execute_extension::cf XML features 2014-07-30 09:42:54.261504 [INFO] switch_ivr_async.c:3653 Bound B-Leg: *4 execute_extension::att_xfer XML features 2014-07-30 09:42:54.281497 [NOTICE] switch_ivr_originate.c:2708 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2014-07-30 09:42:54.281497 [NOTICE] switch_ivr_originate.c:2708 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2014-07-30 09:42:54.281497 [INFO] mod_dptools.c:3234 Originate Failed. Cause: USER_NOT_REGISTERED 2014-07-30 09:42:54.281497 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@xxxxxx.amazonaws.com! 2014-07-30 09:42:54.281497 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/1000@xxxxxx.amazonaws.com] has been answered 2014-07-30 09:42:55.281513 [NOTICE] switch_channel.c:1054 New Channel loopback/app=voicemail:default 10.142.199.62 1001-a [e83b067a-9687-4fe2-8691-ea0f4c8a04a3] 2014-07-30 09:42:55.281513 [NOTICE] switch_channel.c:1052 Rename Channel loopback/app=voicemail:default 10.142.199.62 1001-a->loopback/voicemail-a [e83b067a-9687-4fe2-8691-ea0f4c8a04a3] 2014-07-30 09:42:55.281513 [NOTICE] switch_channel.c:1054 New Channel loopback/voicemail-b [cba53c4a-07c6-4c5a-8bf1-4d4029b6cb66] 2014-07-30 09:42:55.281513 [NOTICE] mod_loopback.c:947 Pre-Answer loopback/voicemail-a! 2014-07-30 09:42:55.281513 [NOTICE] mod_dptools.c:1293 Pre-Answer loopback/voicemail-b! 2014-07-30 09:43:26.301488 [NOTICE] sofia.c:7256 Hangup sofia/internal/1000@xxxxxx.amazonaws.com [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2014-07-30 09:43:26.301488 [NOTICE] switch_ivr_bridge.c:752 Hangup loopback/voicemail-a [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL] 2014-07-30 09:43:26.301488 [NOTICE] mod_loopback.c:553 Hangup loopback/voicemail-b [CS_EXECUTE] [ORIGINATOR_CANCEL] 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1632 Session 10 (sofia/internal/1000@xxxxxx.amazonaws.com) Ended 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1636 Close Channel sofia/internal/1000@xxxxxx.amazonaws.com [CS_DESTROY] 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1632 Session 12 (loopback/voicemail-b) Ended 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1636 Close Channel loopback/voicemail-b [CS_DESTROY] 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1632 Session 11 (loopback/voicemail-a) Ended 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1636 Close Channel loopback/voicemail-a [CS_DESTROY]
Doesn't seem to have any issues with it at all.
On 30 July 2014 02:55, Peter Villeneuve <petervnv1@gmail.com> wrote:
Well that's probably your problem. You need to open up ports for media (RTP) to flow.
If your setup is still default, then you'll need to open up ports 16384-32768 See https://wiki.freeswitch.org/wiki/Firewall
On Tue, Jul 29, 2014 at 6:33 PM, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> wrote:
Brian,
On 30 July 2014 00:06, Brian West <brian@freeswitch.org> wrote:
On a hunch, I launched an instance, set the firewall up properly. Checked out, built and installed FS, No issues at all. ;) Just making sure someone else was reporting issues so I thought I would spin it up to double check a few things.
Thanks for looking it up. Can you tell me what ports of the firewall did you open up? Maybe that's the issue. I only have 5060 (and ssh) open by the way.
On Tue, Jul 29, 2014 at 9:12 AM, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> wrote:
On 29 July 2014 21:54, Ben Langfeld <ben@langfeld.co.uk> wrote: > Try explicitly answering the call before your <Speak/> action.
Thanks Ben,
I'm sorry but I do not see any explicit answer XML tag which I can use for Plivo. What do you mean by explicitly answering the call before the Speak action?
> > > On 29 July 2014 07:04, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> > wrote: >> >> Hello everyone, >> >> I have just gotten to know about FreeSWITCH and have tried >> installing >> it on a EC2 instance and in the process of trying out some examples. >> I >> am facing this issue of not being able to hear any sounds during a >> call I make to my FreeSWITCH system. >> >> My freeswitch system is the latest stable ver.1.4 and I followed the >> instructions to install from >> http://clintberry.com/2011/getting-started-with-plivo/ >> >> via the installation script show on the site. I changed the >> installation script to use Slash and also 1.4 instead. >> >> One of the exercises is to pass an incoming call to plivo and make >> plivo return a <say> action, which for some reason is not reading >> back >> the text to me (no audio) when I call 1005 from using sflphone on my >> ubuntu. >> >> This is the log on freeswitch: >> >> 2014-07-29 18:01:12.581530 [NOTICE] switch_channel.c:1054 New >> Channel >> sofia/internal/1000@xxxxxxxxxx.amazonaws.com >> [5650b3dd-1519-4c6f-b720-d4d3a8a65c5c] >> 2014-07-29 18:01:12.681534 [INFO] mod_dialplan_xml.c:558 Processing >> 1000 <1000>->1005 in context default >> 2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 >> sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to >> bypass media or media is not established. >> 2014-07-29 18:01:12.733800 [NOTICE] sofia_media.c:92 Pre-Answer >> sofia/internal/1000@xxxxxxxxxx.amazonaws.com! >> 2014-07-29 18:01:12.733800 [NOTICE] mod_dptools.c:1258 Channel >> [sofia/internal/1000@xxxxxxxxxx.amazonaws.com] has been answered >> 2014-07-29 18:01:14.121516 [NOTICE] mod_dptools.c:1232 Hangup >> sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_RESET] >> [NORMAL_CLEARING] >> 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1632 >> Session >> 1 (sofia/internal/1000@xxxxxxxxxx.amazonaws.com) Ended >> 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1636 Close >> Channel sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_DESTROY] >> >> I can see that plivo is correctly calling the XML I have placed, as >> this is the log I see on my web server: >> >> [29/Jul/2014 01:59:13] "GET >> >> >> /answer/?To=1005&Direction=inbound&From=1000&CallerName=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&CallStatus=ringing >> HTTP/1.1" 200 93 >> [29/Jul/2014 01:59:14] "GET >> >> >> /answer/?To=1005&Direction=inbound&From=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&HangupCause=NORMAL_CLEARING&CallStatus=completed >> HTTP/1.1" 200 93 >> >> This is the XML I have prepared to test plivo: >> >> <?xml version="1.0" encoding="UTF-8" ?> >> <Response> >> <Speak>Hello World</Speak> >> </Response> >> >> >> And this is default.xml on freeswitch which forward any calls to >> 1005 to >> plivo: >> >> <include> >> <context name="default"> >> >> <!-- >> This extension allows calling any digits of number >> freeswitch will call plivo outbound server on every incoming >> call >> --> >> <extension name="plivo"> >> <condition field="destination_number" expression="^1005"> >> <action application="enable_heartbeat" data="60"/> >> <action application="socket" data="127.0.0.1:8084 async >> full"/> >> </condition> >> </extension> >> .... >> .... >> </include> >> >> I have read that NAT issues might be causing a no-audio issue, but I >> am not sure and do not know how to tell if this is the issue. The >> only >> thing which look strange is the logs on freeswitch saying: >> >> 2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 >> sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to >> bypass media or media is not established. >> >> but what does that mean? >> >> >> _________________________________________________________________________ >> 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 >
_________________________________________________________________________ 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
--
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
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
_________________________________________________________________________ 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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hello everyone, I finally got the audio working. I had a hunch that the issue is some sort of networking (i.e NAT) issue specific to EC2 and I found what I was looking for here: https://wiki.freeswitch.org/wiki/Amazon_EC2 I did not attach an Elastic IP and just used the allocated public IP. Since all my needed ports are already open, I made changes to the files below as shown: conf/sip_profiles/internal.xml conf/sip_profiles/external.xml conf/autoload/switch.conf.xml Now when I call, I can see that the ports are being automatically changed, where bbb.bbbb.bbb.bbb is my laptop's internal IP and ccc.ccc.ccc.ccc is my laptop's public IP from the router: 2014-07-30 22:44:37.021489 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/1000@xxxxxx.amazonaws.com [f08fd128-917a-4325-9d22-2b1f1ac367bb] 2014-07-30 22:44:37.101493 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->1005 in context default 2014-07-30 22:44:37.101493 [WARNING] switch_core_session.c:1554 sofia/internal/1000@xxxxxx.amazonaws.com using scheduler due to bypass media or media is not established. 2014-07-30 22:44:37.178308 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@xxxxxx.amazonaws.com! 2014-07-30 22:44:37.178308 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/1000@xxxxxx.amazonaws.com] has been answered 2014-07-30 22:44:37.861504 [INFO] switch_rtp.c:5597 Auto Changing port from bbb.bbbb.bbb.bbb:37160 to ccc.ccc.ccc.ccc:37160 2014-07-30 22:44:42.201501 [NOTICE] mod_dptools.c:1232 Hangup sofia/internal/1000@xxxxxx.amazonaws.com [CS_RESET] [NORMAL_CLEARING] 2014-07-30 22:44:42.201501 [NOTICE] switch_core_session.c:1632 Session 9 (sofia/internal/1000@xxxxxx.amazonaws.com) Ended 2014-07-30 22:44:42.201501 [NOTICE] switch_core_session.c:1636 Close Channel sofia/internal/1000@xxxxxx.amazonaws.com [CS_DESTROY] And then if I check the status of my internal and external profile, it is showing the public ip in Ext-SIP-IP and Ext-RTP-IP: freeswitch@ip-10-142-199-62> sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts 10.142.199.62,10.142.199.62 Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.142.199.62 Ext-RTP-IP xxxx.xxxx.xxxx SIP-IP 10.142.199.62 Ext-SIP-IP xxxx.xxxx.xxxx URL sip:mod_sofia@xxxxx.xxxx.xxx:5060 BIND-URL sip:mod_sofia@xxxx.xxxx.xxx:5060;maddr=10.142.199.62;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN OPUS,G722,PCMU,PCMA,GSM CODECS OUT OPUS,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT true CALLS-IN 8 FAILED-CALLS-IN 0 CALLS-OUT 1 FAILED-CALLS-OUT 1 REGISTRATIONS 1 freeswitch@ip-10-142-199-62> sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 10.142.199.62 Ext-RTP-IP xxx.xxxx.xxx.xxx SIP-IP 10.142.199.62 Ext-SIP-IP xxx.xxxx.xxx.xxx URL sip:mod_sofia@ xxx.xxxx.xxx.xxx:5080 BIND-URL sip:mod_sofia@ xxx.xxxx.xxx.xxx:5080;maddr=10.142.199.62;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN OPUS,G722,PCMU,PCMA,GSM CODECS OUT PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT true CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 0 I hope this will help anyone else having similar issues in the future. On 30 July 2014 17:17, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> wrote:
Thanks William,
As you pointed out, the IPs were not changed in vars.xml I have then changed 3 settings to use the EC2's instance public IP:
<X-PRE-PROCESS cmd="set" data="external_sip_ip=xxx.xxx.xxx.xxx"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=xxx.xxx.xxx.xxx"/> <X-PRE-PROCESS cmd="set" data="bind_server_ip=xxx.xxx.xxx.xxx"/>
After shutting down and restarting FS, I tried calling again but still cannot get audio.
If it helps, these are the status of sofia's external and internal profiles:
freeswitch@ip-10-142-199-62> sofia status profile external
================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 10.142.199.62 SIP-IP 10.142.199.62 URL sip:mod_sofia@10.142.199.62:5080 BIND-URL sip:mod_sofia@10.142.199.62:5080;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN OPUS,G722,PCMU,PCMA,GSM CODECS OUT PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 0
freeswitch@ip-10-142-199-62> sofia status profile internal
================================================================================================= Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts 10.142.199.62,10.142.199.62 Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.142.199.62 SIP-IP 10.142.199.62 URL sip:mod_sofia@10.142.199.62:5060 BIND-URL sip:mod_sofia@10.142.199.62:5060;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN OPUS,G722,PCMU,PCMA,GSM CODECS OUT OPUS,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 2 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 1
On 30 July 2014 14:56, William King <william.king@quentustech.com> wrote:
Take a look in the ./conf/sip_profiles/external.xml file an in there you'll find variables for the external rtp ip. I'm betting you haven't set the external ip in the ./conf/vars.xml file. See here for more info: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#ext-rtp-ip
William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king@quentustech.com
On 07/29/2014 05:49 PM, Iqbal Abdullah wrote:
Thank you Peter and Robert,
I opened up the suggested ports on the wiki page but that didn't work, so I ended up opening all TCP and UDP ports but the audio doesn't come either.
Trying to call a different number (1001, which doesn't have a user) gave me this logs:
2014-07-30 09:42:54.121512 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/1000@xxxxxx.amazonaws.com [34c19686-9e28-459d-813c-fbf57d42c89a] 2014-07-30 09:42:54.261504 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->1001 in context default 2014-07-30 09:42:54.261504 [INFO] switch_ivr_async.c:3653 Bound B-Leg: *1 execute_extension::dx XML features 2014-07-30 09:42:54.261504 [INFO] switch_ivr_async.c:3653 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1000.2014-07-30-09-42-54.wav 2014-07-30 09:42:54.261504 [INFO] switch_ivr_async.c:3653 Bound B-Leg: *3 execute_extension::cf XML features 2014-07-30 09:42:54.261504 [INFO] switch_ivr_async.c:3653 Bound B-Leg: *4 execute_extension::att_xfer XML features 2014-07-30 09:42:54.281497 [NOTICE] switch_ivr_originate.c:2708 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2014-07-30 09:42:54.281497 [NOTICE] switch_ivr_originate.c:2708 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2014-07-30 09:42:54.281497 [INFO] mod_dptools.c:3234 Originate Failed. Cause: USER_NOT_REGISTERED 2014-07-30 09:42:54.281497 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@xxxxxx.amazonaws.com! 2014-07-30 09:42:54.281497 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/1000@xxxxxx.amazonaws.com] has been answered 2014-07-30 09:42:55.281513 [NOTICE] switch_channel.c:1054 New Channel loopback/app=voicemail:default 10.142.199.62 1001-a [e83b067a-9687-4fe2-8691-ea0f4c8a04a3] 2014-07-30 09:42:55.281513 [NOTICE] switch_channel.c:1052 Rename Channel loopback/app=voicemail:default 10.142.199.62 1001-a->loopback/voicemail-a [e83b067a-9687-4fe2-8691-ea0f4c8a04a3] 2014-07-30 09:42:55.281513 [NOTICE] switch_channel.c:1054 New Channel loopback/voicemail-b [cba53c4a-07c6-4c5a-8bf1-4d4029b6cb66] 2014-07-30 09:42:55.281513 [NOTICE] mod_loopback.c:947 Pre-Answer loopback/voicemail-a! 2014-07-30 09:42:55.281513 [NOTICE] mod_dptools.c:1293 Pre-Answer loopback/voicemail-b! 2014-07-30 09:43:26.301488 [NOTICE] sofia.c:7256 Hangup sofia/internal/1000@xxxxxx.amazonaws.com [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2014-07-30 09:43:26.301488 [NOTICE] switch_ivr_bridge.c:752 Hangup loopback/voicemail-a [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL] 2014-07-30 09:43:26.301488 [NOTICE] mod_loopback.c:553 Hangup loopback/voicemail-b [CS_EXECUTE] [ORIGINATOR_CANCEL] 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1632 Session 10 (sofia/internal/1000@xxxxxx.amazonaws.com) Ended 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1636 Close Channel sofia/internal/1000@xxxxxx.amazonaws.com [CS_DESTROY] 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1632 Session 12 (loopback/voicemail-b) Ended 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1636 Close Channel loopback/voicemail-b [CS_DESTROY] 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1632 Session 11 (loopback/voicemail-a) Ended 2014-07-30 09:43:26.301488 [NOTICE] switch_core_session.c:1636 Close Channel loopback/voicemail-a [CS_DESTROY]
Doesn't seem to have any issues with it at all.
On 30 July 2014 02:55, Peter Villeneuve <petervnv1@gmail.com> wrote:
Well that's probably your problem. You need to open up ports for media (RTP) to flow.
If your setup is still default, then you'll need to open up ports 16384-32768 See https://wiki.freeswitch.org/wiki/Firewall
On Tue, Jul 29, 2014 at 6:33 PM, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> wrote:
Brian,
On 30 July 2014 00:06, Brian West <brian@freeswitch.org> wrote:
On a hunch, I launched an instance, set the firewall up properly. Checked out, built and installed FS, No issues at all. ;) Just making sure someone else was reporting issues so I thought I would spin it up to double check a few things.
Thanks for looking it up. Can you tell me what ports of the firewall did you open up? Maybe that's the issue. I only have 5060 (and ssh) open by the way.
On Tue, Jul 29, 2014 at 9:12 AM, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> wrote: > > On 29 July 2014 21:54, Ben Langfeld <ben@langfeld.co.uk> wrote: >> Try explicitly answering the call before your <Speak/> action. > > Thanks Ben, > > I'm sorry but I do not see any explicit answer XML tag which I can use > for Plivo. > What do you mean by explicitly answering the call before the Speak > action? > >> >> >> On 29 July 2014 07:04, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> >> wrote: >>> >>> Hello everyone, >>> >>> I have just gotten to know about FreeSWITCH and have tried >>> installing >>> it on a EC2 instance and in the process of trying out some examples. >>> I >>> am facing this issue of not being able to hear any sounds during a >>> call I make to my FreeSWITCH system. >>> >>> My freeswitch system is the latest stable ver.1.4 and I followed the >>> instructions to install from >>> http://clintberry.com/2011/getting-started-with-plivo/ >>> >>> via the installation script show on the site. I changed the >>> installation script to use Slash and also 1.4 instead. >>> >>> One of the exercises is to pass an incoming call to plivo and make >>> plivo return a <say> action, which for some reason is not reading >>> back >>> the text to me (no audio) when I call 1005 from using sflphone on my >>> ubuntu. >>> >>> This is the log on freeswitch: >>> >>> 2014-07-29 18:01:12.581530 [NOTICE] switch_channel.c:1054 New >>> Channel >>> sofia/internal/1000@xxxxxxxxxx.amazonaws.com >>> [5650b3dd-1519-4c6f-b720-d4d3a8a65c5c] >>> 2014-07-29 18:01:12.681534 [INFO] mod_dialplan_xml.c:558 Processing >>> 1000 <1000>->1005 in context default >>> 2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 >>> sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to >>> bypass media or media is not established. >>> 2014-07-29 18:01:12.733800 [NOTICE] sofia_media.c:92 Pre-Answer >>> sofia/internal/1000@xxxxxxxxxx.amazonaws.com! >>> 2014-07-29 18:01:12.733800 [NOTICE] mod_dptools.c:1258 Channel >>> [sofia/internal/1000@xxxxxxxxxx.amazonaws.com] has been answered >>> 2014-07-29 18:01:14.121516 [NOTICE] mod_dptools.c:1232 Hangup >>> sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_RESET] >>> [NORMAL_CLEARING] >>> 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1632 >>> Session >>> 1 (sofia/internal/1000@xxxxxxxxxx.amazonaws.com) Ended >>> 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1636 Close >>> Channel sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_DESTROY] >>> >>> I can see that plivo is correctly calling the XML I have placed, as >>> this is the log I see on my web server: >>> >>> [29/Jul/2014 01:59:13] "GET >>> >>> >>> /answer/?To=1005&Direction=inbound&From=1000&CallerName=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&CallStatus=ringing >>> HTTP/1.1" 200 93 >>> [29/Jul/2014 01:59:14] "GET >>> >>> >>> /answer/?To=1005&Direction=inbound&From=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&HangupCause=NORMAL_CLEARING&CallStatus=completed >>> HTTP/1.1" 200 93 >>> >>> This is the XML I have prepared to test plivo: >>> >>> <?xml version="1.0" encoding="UTF-8" ?> >>> <Response> >>> <Speak>Hello World</Speak> >>> </Response> >>> >>> >>> And this is default.xml on freeswitch which forward any calls to >>> 1005 to >>> plivo: >>> >>> <include> >>> <context name="default"> >>> >>> <!-- >>> This extension allows calling any digits of number >>> freeswitch will call plivo outbound server on every incoming >>> call >>> --> >>> <extension name="plivo"> >>> <condition field="destination_number" expression="^1005"> >>> <action application="enable_heartbeat" data="60"/> >>> <action application="socket" data="127.0.0.1:8084 async >>> full"/> >>> </condition> >>> </extension> >>> .... >>> .... >>> </include> >>> >>> I have read that NAT issues might be causing a no-audio issue, but I >>> am not sure and do not know how to tell if this is the issue. The >>> only >>> thing which look strange is the logs on freeswitch saying: >>> >>> 2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 >>> sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to >>> bypass media or media is not established. >>> >>> but what does that mean? >>> >>> >>> _________________________________________________________________________ >>> 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 >> > > > _________________________________________________________________________ > 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
--
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
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
_________________________________________________________________________ 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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi, Refer to this wiki page: https://wiki.freeswitch.org/wiki/Firewall Good luck, Robert -----Original Message----- From: Iqbal Abdullah [mailto:iqbal.b.abdullah@gmail.com] Sent: Tuesday, July 29, 2014 10:34 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] No audio when calling in via SIP phone Brian, On 30 July 2014 00:06, Brian West <brian@freeswitch.org> wrote:
On a hunch, I launched an instance, set the firewall up properly. Checked out, built and installed FS, No issues at all. ;) Just making sure someone else was reporting issues so I thought I would spin it up to double check a few things.
Thanks for looking it up. Can you tell me what ports of the firewall did you open up? Maybe that's the issue. I only have 5060 (and ssh) open by the way.
On Tue, Jul 29, 2014 at 9:12 AM, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> wrote:
On 29 July 2014 21:54, Ben Langfeld <ben@langfeld.co.uk> wrote:
Try explicitly answering the call before your <Speak/> action.
Thanks Ben,
I'm sorry but I do not see any explicit answer XML tag which I can use for Plivo. What do you mean by explicitly answering the call before the Speak action?
On 29 July 2014 07:04, Iqbal Abdullah <iqbal.b.abdullah@gmail.com> wrote:
Hello everyone,
I have just gotten to know about FreeSWITCH and have tried installing it on a EC2 instance and in the process of trying out some examples. I am facing this issue of not being able to hear any sounds during a call I make to my FreeSWITCH system.
My freeswitch system is the latest stable ver.1.4 and I followed the instructions to install from http://clintberry.com/2011/getting-started-with-plivo/
via the installation script show on the site. I changed the installation script to use Slash and also 1.4 instead.
One of the exercises is to pass an incoming call to plivo and make plivo return a <say> action, which for some reason is not reading back the text to me (no audio) when I call 1005 from using sflphone on my ubuntu.
This is the log on freeswitch:
2014-07-29 18:01:12.581530 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/1000@xxxxxxxxxx.amazonaws.com [5650b3dd-1519-4c6f-b720-d4d3a8a65c5c] 2014-07-29 18:01:12.681534 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->1005 in context default 2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established. 2014-07-29 18:01:12.733800 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@xxxxxxxxxx.amazonaws.com! 2014-07-29 18:01:12.733800 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/1000@xxxxxxxxxx.amazonaws.com] has been answered 2014-07-29 18:01:14.121516 [NOTICE] mod_dptools.c:1232 Hangup sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_RESET] [NORMAL_CLEARING] 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1632 Session 1 (sofia/internal/1000@xxxxxxxxxx.amazonaws.com) Ended 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1636 Close Channel sofia/internal/1000@xxxxxxxxxx.amazonaws.com [CS_DESTROY]
I can see that plivo is correctly calling the XML I have placed, as this is the log I see on my web server:
[29/Jul/2014 01:59:13] "GET
/answer/?To=1005&Direction=inbound&From=1000&CallerName=1000&CallU UID=b74c06be-f4e9-4830-af68-e424b3b31e1c&CallStatus=ringing HTTP/1.1" 200 93 [29/Jul/2014 01:59:14] "GET
/answer/?To=1005&Direction=inbound&From=1000&CallUUID=b74c06be-f4e 9-4830-af68-e424b3b31e1c&HangupCause=NORMAL_CLEARING&CallStatus=co mpleted HTTP/1.1" 200 93
This is the XML I have prepared to test plivo:
<?xml version="1.0" encoding="UTF-8" ?> <Response> <Speak>Hello World</Speak> </Response>
And this is default.xml on freeswitch which forward any calls to 1005 to plivo:
<include> <context name="default">
<!-- This extension allows calling any digits of number freeswitch will call plivo outbound server on every incoming call --> <extension name="plivo"> <condition field="destination_number" expression="^1005"> <action application="enable_heartbeat" data="60"/> <action application="socket" data="127.0.0.1:8084 async full"/> </condition> </extension> .... .... </include>
I have read that NAT issues might be causing a no-audio issue, but I am not sure and do not know how to tell if this is the issue. The only thing which look strange is the logs on freeswitch saying:
2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000@xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established.
but what does that mean?
_________________________________________________________________________ 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
_________________________________________________________________________ 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
--
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
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (6)
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Ben Langfeld
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Brian West
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Iqbal Abdullah
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Peter Villeneuve
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Robert Hadley
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William King