DTMF rfc2833 hanging up the calls
Hi, if I use SIP INFO on leg A, FS detects everything fine, but in case I use rfc2833, after pressing number, it instantly hangs up the call. n2013-06-06 17:13:16.319289 [NOTICE] mod_sofia.c:1137 Hangup sofia/myprofile/sip:200@x.x.x.x:45703 [CS_EXCHANGE_MEDIA] [MEDIA_TIMEOUT] 2013-06-06 17:13:16.319289 [DEBUG] switch_channel.c:3096 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [KILL] 2013-06-06 17:13:16.319289 [DEBUG] switch_core_session.c:1333 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [BREAK] 2013-06-06 17:13:16.319289 [DEBUG] switch_ivr_bridge.c:533 sofia/myprofile/sip:200@x.x.x.x:45703 ending bridge by request from read function I do not use any ivr module, or start_dtmf, or whatsoever setting related to dtmf. Any hint? Thanks, Mino
FS hangs up the call when it receives RTP DTMF Event. FreeSwitch 1.2.10 profile: <profile name="myprofile"> <settings> <param name="context" value="mycontext"/> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="dtmf-type" value="rfc2833"/> <param name="liberal-dtmf" value="true"/> <param name="codec-ms" value="20"/>--> <param name="use-rtp-timer" value="true"/> <param name="sip-ip" value="192.168.x.x"/> <param name="rtp-ip" value="192.168.x.x"/> <param name="rtp-timeout-sec" value="3000"/> <param name="manage-presence" value="false"/> <param name="rtp-timeout-sec" value="5"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> </settings> </profile> dialplan: <extension name="Local_Extension_3"> <condition field="destination_number" expression="^300$"> <action application="bridge" data="sofia/myprofile/300@192.168.x.x :5060"/> </condition> </extension> modules: <load module="mod_console"/> <load module="mod_logfile"/> <load module="mod_enum"/> <load module="mod_event_socket"/> <load module="mod_sofia"/> <load module="mod_loopback"/> <load module="mod_commands"/> <load module="mod_dptools"/> <load module="mod_expr"/> <load module="mod_fifo"/> <load module="mod_hash"/> <load module="mod_amr"/> <load module="mod_speex"/> <load module="mod_spandsp"/> <load module="mod_com_g729"/> <load module="mod_cluechoo"/>--> <load module="mod_dialplan_xml"/> <load module="mod_g723_1"/> <load module="mod_native_file"/> <load module="mod_local_stream"/> <load module="mod_tone_stream"/> <load module="mod_xml_cdr"/> On Thu, Jun 6, 2013 at 5:20 PM, Mino Haluz <mino.haluz@gmail.com> wrote:
Hi,
if I use SIP INFO on leg A, FS detects everything fine, but in case I use rfc2833, after pressing number, it instantly hangs up the call.
n2013-06-06 17:13:16.319289 [NOTICE] mod_sofia.c:1137 Hangup sofia/myprofile/sip:200@x.x.x.x:45703 [CS_EXCHANGE_MEDIA] [MEDIA_TIMEOUT] 2013-06-06 17:13:16.319289 [DEBUG] switch_channel.c:3096 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [KILL] 2013-06-06 17:13:16.319289 [DEBUG] switch_core_session.c:1333 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [BREAK] 2013-06-06 17:13:16.319289 [DEBUG] switch_ivr_bridge.c:533 sofia/myprofile/sip:200@x.x.x.x:45703 ending bridge by request from read function
I do not use any ivr module, or start_dtmf, or whatsoever setting related to dtmf. Any hint?
Thanks, Mino
Can you share a complete debug-level log of the call? Are any other extensions being executed first? There are api calls that could bind a dtmf key to hanging up the bridge, perhaps that's being done here earlier in the dialplan. -Steve On Thursday, June 6, 2013, Mino Haluz wrote:
FS hangs up the call when it receives RTP DTMF Event.
FreeSwitch 1.2.10
profile: <profile name="myprofile"> <settings> <param name="context" value="mycontext"/> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="dtmf-type" value="rfc2833"/> <param name="liberal-dtmf" value="true"/> <param name="codec-ms" value="20"/>--> <param name="use-rtp-timer" value="true"/> <param name="sip-ip" value="192.168.x.x"/> <param name="rtp-ip" value="192.168.x.x"/> <param name="rtp-timeout-sec" value="3000"/> <param name="manage-presence" value="false"/> <param name="rtp-timeout-sec" value="5"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> </settings> </profile>
dialplan: <extension name="Local_Extension_3"> <condition field="destination_number" expression="^300$"> <action application="bridge" data="sofia/myprofile/300@192.168.x.x :5060"/> </condition> </extension>
modules: <load module="mod_console"/> <load module="mod_logfile"/> <load module="mod_enum"/> <load module="mod_event_socket"/> <load module="mod_sofia"/> <load module="mod_loopback"/> <load module="mod_commands"/> <load module="mod_dptools"/> <load module="mod_expr"/> <load module="mod_fifo"/> <load module="mod_hash"/> <load module="mod_amr"/> <load module="mod_speex"/> <load module="mod_spandsp"/> <load module="mod_com_g729"/> <load module="mod_cluechoo"/>--> <load module="mod_dialplan_xml"/> <load module="mod_g723_1"/> <load module="mod_native_file"/> <load module="mod_local_stream"/> <load module="mod_tone_stream"/> <load module="mod_xml_cdr"/>
On Thu, Jun 6, 2013 at 5:20 PM, Mino Haluz <mino.haluz@gmail.com<javascript:_e({}, 'cvml', 'mino.haluz@gmail.com');>
wrote:
Hi,
if I use SIP INFO on leg A, FS detects everything fine, but in case I use rfc2833, after pressing number, it instantly hangs up the call.
n2013-06-06 17:13:16.319289 [NOTICE] mod_sofia.c:1137 Hangup sofia/myprofile/sip:200@x.x.x.x:45703 [CS_EXCHANGE_MEDIA] [MEDIA_TIMEOUT] 2013-06-06 17:13:16.319289 [DEBUG] switch_channel.c:3096 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [KILL] 2013-06-06 17:13:16.319289 [DEBUG] switch_core_session.c:1333 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [BREAK] 2013-06-06 17:13:16.319289 [DEBUG] switch_ivr_bridge.c:533 sofia/myprofile/sip:200@x.x.x.x:45703 ending bridge by request from read function
I do not use any ivr module, or start_dtmf, or whatsoever setting related to dtmf. Any hint?
Thanks, Mino
Hi, so this is the call log: http://pastebin.freeswitch.org/21035 the only dialplan I have: http://pastebin.freeswitch.org/21036 my sip profile: http://pastebin.freeswitch.org/21038 anything else? Mino On Thu, Jun 6, 2013 at 8:12 PM, Steven Ayre <steveayre@gmail.com> wrote:
Can you share a complete debug-level log of the call? Are any other extensions being executed first?
There are api calls that could bind a dtmf key to hanging up the bridge, perhaps that's being done here earlier in the dialplan.
-Steve
On Thursday, June 6, 2013, Mino Haluz wrote:
FS hangs up the call when it receives RTP DTMF Event.
FreeSwitch 1.2.10
profile: <profile name="myprofile"> <settings> <param name="context" value="mycontext"/> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="dtmf-type" value="rfc2833"/> <param name="liberal-dtmf" value="true"/> <param name="codec-ms" value="20"/>--> <param name="use-rtp-timer" value="true"/> <param name="sip-ip" value="192.168.x.x"/> <param name="rtp-ip" value="192.168.x.x"/> <param name="rtp-timeout-sec" value="3000"/> <param name="manage-presence" value="false"/> <param name="rtp-timeout-sec" value="5"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> </settings> </profile>
dialplan: <extension name="Local_Extension_3"> <condition field="destination_number" expression="^300$"> <action application="bridge" data="sofia/myprofile/300@192.168.x.x :5060"/> </condition> </extension>
modules: <load module="mod_console"/> <load module="mod_logfile"/> <load module="mod_enum"/> <load module="mod_event_socket"/> <load module="mod_sofia"/> <load module="mod_loopback"/> <load module="mod_commands"/> <load module="mod_dptools"/> <load module="mod_expr"/> <load module="mod_fifo"/> <load module="mod_hash"/> <load module="mod_amr"/> <load module="mod_speex"/> <load module="mod_spandsp"/> <load module="mod_com_g729"/> <load module="mod_cluechoo"/>--> <load module="mod_dialplan_xml"/> <load module="mod_g723_1"/> <load module="mod_native_file"/> <load module="mod_local_stream"/> <load module="mod_tone_stream"/> <load module="mod_xml_cdr"/>
On Thu, Jun 6, 2013 at 5:20 PM, Mino Haluz <mino.haluz@gmail.com> wrote:
Hi,
if I use SIP INFO on leg A, FS detects everything fine, but in case I use rfc2833, after pressing number, it instantly hangs up the call.
n2013-06-06 17:13:16.319289 [NOTICE] mod_sofia.c:1137 Hangup sofia/myprofile/sip:200@x.x.x.x:45703 [CS_EXCHANGE_MEDIA] [MEDIA_TIMEOUT] 2013-06-06 17:13:16.319289 [DEBUG] switch_channel.c:3096 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [KILL] 2013-06-06 17:13:16.319289 [DEBUG] switch_core_session.c:1333 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [BREAK] 2013-06-06 17:13:16.319289 [DEBUG] switch_ivr_bridge.c:533 sofia/myprofile/sip:200@x.x.x.x:45703 ending bridge by request from read function
I do not use any ivr module, or start_dtmf, or whatsoever setting related to dtmf. Any hint?
Thanks, Mino
_________________________________________________________________________ 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
I forgot to delete all sofia_reg lines (registrations made a bit mess in call log). So for example this line should not be there: 2013-06-07 10:47:31.919290 [DEBUG] sofia_reg.c:1520 Send challenge for [300@ 192.168.80.55] On Fri, Jun 7, 2013 at 10:57 AM, Mino Haluz <mino.haluz@gmail.com> wrote:
Hi,
so this is the call log: http://pastebin.freeswitch.org/21035
the only dialplan I have: http://pastebin.freeswitch.org/21036
my sip profile: http://pastebin.freeswitch.org/21038
anything else?
Mino
On Thu, Jun 6, 2013 at 8:12 PM, Steven Ayre <steveayre@gmail.com> wrote:
Can you share a complete debug-level log of the call? Are any other extensions being executed first?
There are api calls that could bind a dtmf key to hanging up the bridge, perhaps that's being done here earlier in the dialplan.
-Steve
On Thursday, June 6, 2013, Mino Haluz wrote:
FS hangs up the call when it receives RTP DTMF Event.
FreeSwitch 1.2.10
profile: <profile name="myprofile"> <settings> <param name="context" value="mycontext"/> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="dtmf-type" value="rfc2833"/> <param name="liberal-dtmf" value="true"/> <param name="codec-ms" value="20"/>--> <param name="use-rtp-timer" value="true"/> <param name="sip-ip" value="192.168.x.x"/> <param name="rtp-ip" value="192.168.x.x"/> <param name="rtp-timeout-sec" value="3000"/> <param name="manage-presence" value="false"/> <param name="rtp-timeout-sec" value="5"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> </settings> </profile>
dialplan: <extension name="Local_Extension_3"> <condition field="destination_number" expression="^300$"> <action application="bridge" data="sofia/myprofile/300@192.168.x.x :5060"/> </condition> </extension>
modules: <load module="mod_console"/> <load module="mod_logfile"/> <load module="mod_enum"/> <load module="mod_event_socket"/> <load module="mod_sofia"/> <load module="mod_loopback"/> <load module="mod_commands"/> <load module="mod_dptools"/> <load module="mod_expr"/> <load module="mod_fifo"/> <load module="mod_hash"/> <load module="mod_amr"/> <load module="mod_speex"/> <load module="mod_spandsp"/> <load module="mod_com_g729"/> <load module="mod_cluechoo"/>--> <load module="mod_dialplan_xml"/> <load module="mod_g723_1"/> <load module="mod_native_file"/> <load module="mod_local_stream"/> <load module="mod_tone_stream"/> <load module="mod_xml_cdr"/>
On Thu, Jun 6, 2013 at 5:20 PM, Mino Haluz <mino.haluz@gmail.com> wrote:
Hi,
if I use SIP INFO on leg A, FS detects everything fine, but in case I use rfc2833, after pressing number, it instantly hangs up the call.
n2013-06-06 17:13:16.319289 [NOTICE] mod_sofia.c:1137 Hangup sofia/myprofile/sip:200@x.x.x.x:45703 [CS_EXCHANGE_MEDIA] [MEDIA_TIMEOUT] 2013-06-06 17:13:16.319289 [DEBUG] switch_channel.c:3096 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [KILL] 2013-06-06 17:13:16.319289 [DEBUG] switch_core_session.c:1333 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [BREAK] 2013-06-06 17:13:16.319289 [DEBUG] switch_ivr_bridge.c:533 sofia/myprofile/sip:200@x.x.x.x:45703 ending bridge by request from read function
I do not use any ivr module, or start_dtmf, or whatsoever setting related to dtmf. Any hint?
Thanks, Mino
_________________________________________________________________________ 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
Need a siptrace, please run 'sofia global siptrace on' before you take the console log and post it on pastebin.freeswitch.org. Reply back with the link. Hopefully we can glean some more info. - BDF On Jun 7, 2013 5:09 AM, "Mino Haluz" <mino.haluz@gmail.com> wrote:
I forgot to delete all sofia_reg lines (registrations made a bit mess in call log). So for example this line should not be there:
2013-06-07 10:47:31.919290 [DEBUG] sofia_reg.c:1520 Send challenge for [ 300@192.168.80.55]
On Fri, Jun 7, 2013 at 10:57 AM, Mino Haluz <mino.haluz@gmail.com> wrote:
Hi,
so this is the call log: http://pastebin.freeswitch.org/21035
the only dialplan I have: http://pastebin.freeswitch.org/21036
my sip profile: http://pastebin.freeswitch.org/21038
anything else?
Mino
On Thu, Jun 6, 2013 at 8:12 PM, Steven Ayre <steveayre@gmail.com> wrote:
Can you share a complete debug-level log of the call? Are any other extensions being executed first?
There are api calls that could bind a dtmf key to hanging up the bridge, perhaps that's being done here earlier in the dialplan.
-Steve
On Thursday, June 6, 2013, Mino Haluz wrote:
FS hangs up the call when it receives RTP DTMF Event.
FreeSwitch 1.2.10
profile: <profile name="myprofile"> <settings> <param name="context" value="mycontext"/> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="dtmf-type" value="rfc2833"/> <param name="liberal-dtmf" value="true"/> <param name="codec-ms" value="20"/>--> <param name="use-rtp-timer" value="true"/> <param name="sip-ip" value="192.168.x.x"/> <param name="rtp-ip" value="192.168.x.x"/> <param name="rtp-timeout-sec" value="3000"/> <param name="manage-presence" value="false"/> <param name="rtp-timeout-sec" value="5"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> </settings> </profile>
dialplan: <extension name="Local_Extension_3"> <condition field="destination_number" expression="^300$"> <action application="bridge" data="sofia/myprofile/300@192.168.x.x :5060"/> </condition> </extension>
modules: <load module="mod_console"/> <load module="mod_logfile"/> <load module="mod_enum"/> <load module="mod_event_socket"/> <load module="mod_sofia"/> <load module="mod_loopback"/> <load module="mod_commands"/> <load module="mod_dptools"/> <load module="mod_expr"/> <load module="mod_fifo"/> <load module="mod_hash"/> <load module="mod_amr"/> <load module="mod_speex"/> <load module="mod_spandsp"/> <load module="mod_com_g729"/> <load module="mod_cluechoo"/>--> <load module="mod_dialplan_xml"/> <load module="mod_g723_1"/> <load module="mod_native_file"/> <load module="mod_local_stream"/> <load module="mod_tone_stream"/> <load module="mod_xml_cdr"/>
On Thu, Jun 6, 2013 at 5:20 PM, Mino Haluz <mino.haluz@gmail.com>wrote:
Hi,
if I use SIP INFO on leg A, FS detects everything fine, but in case I use rfc2833, after pressing number, it instantly hangs up the call.
n2013-06-06 17:13:16.319289 [NOTICE] mod_sofia.c:1137 Hangup sofia/myprofile/sip:200@x.x.x.x:45703 [CS_EXCHANGE_MEDIA] [MEDIA_TIMEOUT] 2013-06-06 17:13:16.319289 [DEBUG] switch_channel.c:3096 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [KILL] 2013-06-06 17:13:16.319289 [DEBUG] switch_core_session.c:1333 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [BREAK] 2013-06-06 17:13:16.319289 [DEBUG] switch_ivr_bridge.c:533 sofia/myprofile/sip:200@x.x.x.x:45703 ending bridge by request from read function
I do not use any ivr module, or start_dtmf, or whatsoever setting related to dtmf. Any hint?
Thanks, Mino
_________________________________________________________________________ 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
Complete log with siptrace on: http://pastebin.freeswitch.org/21046 On Mon, Jun 10, 2013 at 6:51 AM, Brian Foster <bdfoster@davri.com> wrote:
Need a siptrace, please run 'sofia global siptrace on' before you take the console log and post it on pastebin.freeswitch.org. Reply back with the link. Hopefully we can glean some more info.
- BDF On Jun 7, 2013 5:09 AM, "Mino Haluz" <mino.haluz@gmail.com> wrote:
I forgot to delete all sofia_reg lines (registrations made a bit mess in call log). So for example this line should not be there:
2013-06-07 10:47:31.919290 [DEBUG] sofia_reg.c:1520 Send challenge for [ 300@192.168.80.55]
On Fri, Jun 7, 2013 at 10:57 AM, Mino Haluz <mino.haluz@gmail.com> wrote:
Hi,
so this is the call log: http://pastebin.freeswitch.org/21035
the only dialplan I have: http://pastebin.freeswitch.org/21036
my sip profile: http://pastebin.freeswitch.org/21038
anything else?
Mino
On Thu, Jun 6, 2013 at 8:12 PM, Steven Ayre <steveayre@gmail.com> wrote:
Can you share a complete debug-level log of the call? Are any other extensions being executed first?
There are api calls that could bind a dtmf key to hanging up the bridge, perhaps that's being done here earlier in the dialplan.
-Steve
On Thursday, June 6, 2013, Mino Haluz wrote:
FS hangs up the call when it receives RTP DTMF Event.
FreeSwitch 1.2.10
profile: <profile name="myprofile"> <settings> <param name="context" value="mycontext"/> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="dtmf-type" value="rfc2833"/> <param name="liberal-dtmf" value="true"/> <param name="codec-ms" value="20"/>--> <param name="use-rtp-timer" value="true"/> <param name="sip-ip" value="192.168.x.x"/> <param name="rtp-ip" value="192.168.x.x"/> <param name="rtp-timeout-sec" value="3000"/> <param name="manage-presence" value="false"/> <param name="rtp-timeout-sec" value="5"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> </settings> </profile>
dialplan: <extension name="Local_Extension_3"> <condition field="destination_number" expression="^300$"> <action application="bridge" data="sofia/myprofile/300@192.168.x.x :5060"/> </condition> </extension>
modules: <load module="mod_console"/> <load module="mod_logfile"/> <load module="mod_enum"/> <load module="mod_event_socket"/> <load module="mod_sofia"/> <load module="mod_loopback"/> <load module="mod_commands"/> <load module="mod_dptools"/> <load module="mod_expr"/> <load module="mod_fifo"/> <load module="mod_hash"/> <load module="mod_amr"/> <load module="mod_speex"/> <load module="mod_spandsp"/> <load module="mod_com_g729"/> <load module="mod_cluechoo"/>--> <load module="mod_dialplan_xml"/> <load module="mod_g723_1"/> <load module="mod_native_file"/> <load module="mod_local_stream"/> <load module="mod_tone_stream"/> <load module="mod_xml_cdr"/>
On Thu, Jun 6, 2013 at 5:20 PM, Mino Haluz <mino.haluz@gmail.com>wrote:
Hi,
if I use SIP INFO on leg A, FS detects everything fine, but in case I use rfc2833, after pressing number, it instantly hangs up the call.
n2013-06-06 17:13:16.319289 [NOTICE] mod_sofia.c:1137 Hangup sofia/myprofile/sip:200@x.x.x.x:45703 [CS_EXCHANGE_MEDIA] [MEDIA_TIMEOUT] 2013-06-06 17:13:16.319289 [DEBUG] switch_channel.c:3096 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [KILL] 2013-06-06 17:13:16.319289 [DEBUG] switch_core_session.c:1333 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [BREAK] 2013-06-06 17:13:16.319289 [DEBUG] switch_ivr_bridge.c:533 sofia/myprofile/sip:200@x.x.x.x:45703 ending bridge by request from read function
I do not use any ivr module, or start_dtmf, or whatsoever setting related to dtmf. Any hint?
Thanks, Mino
_________________________________________________________________________ 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
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Ok timer_name = soft did it and now it works. Can someone elaborate what this value means in fact? On Mon, Jun 10, 2013 at 3:11 PM, Mino Haluz <mino.haluz@gmail.com> wrote:
Complete log with siptrace on: http://pastebin.freeswitch.org/21046
On Mon, Jun 10, 2013 at 6:51 AM, Brian Foster <bdfoster@davri.com> wrote:
Need a siptrace, please run 'sofia global siptrace on' before you take the console log and post it on pastebin.freeswitch.org. Reply back with the link. Hopefully we can glean some more info.
- BDF On Jun 7, 2013 5:09 AM, "Mino Haluz" <mino.haluz@gmail.com> wrote:
I forgot to delete all sofia_reg lines (registrations made a bit mess in call log). So for example this line should not be there:
2013-06-07 10:47:31.919290 [DEBUG] sofia_reg.c:1520 Send challenge for [ 300@192.168.80.55]
On Fri, Jun 7, 2013 at 10:57 AM, Mino Haluz <mino.haluz@gmail.com>wrote:
Hi,
so this is the call log: http://pastebin.freeswitch.org/21035
the only dialplan I have: http://pastebin.freeswitch.org/21036
my sip profile: http://pastebin.freeswitch.org/21038
anything else?
Mino
On Thu, Jun 6, 2013 at 8:12 PM, Steven Ayre <steveayre@gmail.com>wrote:
Can you share a complete debug-level log of the call? Are any other extensions being executed first?
There are api calls that could bind a dtmf key to hanging up the bridge, perhaps that's being done here earlier in the dialplan.
-Steve
On Thursday, June 6, 2013, Mino Haluz wrote:
FS hangs up the call when it receives RTP DTMF Event.
FreeSwitch 1.2.10
profile: <profile name="myprofile"> <settings> <param name="context" value="mycontext"/> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="dtmf-type" value="rfc2833"/> <param name="liberal-dtmf" value="true"/> <param name="codec-ms" value="20"/>--> <param name="use-rtp-timer" value="true"/> <param name="sip-ip" value="192.168.x.x"/> <param name="rtp-ip" value="192.168.x.x"/> <param name="rtp-timeout-sec" value="3000"/> <param name="manage-presence" value="false"/> <param name="rtp-timeout-sec" value="5"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> </settings> </profile>
dialplan: <extension name="Local_Extension_3"> <condition field="destination_number" expression="^300$"> <action application="bridge" data="sofia/myprofile/300@192.168.x.x:5060"/> </condition> </extension>
modules: <load module="mod_console"/> <load module="mod_logfile"/> <load module="mod_enum"/> <load module="mod_event_socket"/> <load module="mod_sofia"/> <load module="mod_loopback"/> <load module="mod_commands"/> <load module="mod_dptools"/> <load module="mod_expr"/> <load module="mod_fifo"/> <load module="mod_hash"/> <load module="mod_amr"/> <load module="mod_speex"/> <load module="mod_spandsp"/> <load module="mod_com_g729"/> <load module="mod_cluechoo"/>--> <load module="mod_dialplan_xml"/> <load module="mod_g723_1"/> <load module="mod_native_file"/> <load module="mod_local_stream"/> <load module="mod_tone_stream"/> <load module="mod_xml_cdr"/>
On Thu, Jun 6, 2013 at 5:20 PM, Mino Haluz <mino.haluz@gmail.com>wrote:
> Hi, > > if I use SIP INFO on leg A, FS detects everything fine, but in case > I use rfc2833, after pressing number, it instantly hangs up the call. > > n2013-06-06 17:13:16.319289 [NOTICE] mod_sofia.c:1137 Hangup > sofia/myprofile/sip:200@x.x.x.x:45703 [CS_EXCHANGE_MEDIA] > [MEDIA_TIMEOUT] > 2013-06-06 17:13:16.319289 [DEBUG] switch_channel.c:3096 Send signal > sofia/myprofile/sip:200@x.x.x.x:45703 [KILL] > 2013-06-06 17:13:16.319289 [DEBUG] switch_core_session.c:1333 Send > signal sofia/myprofile/sip:200@x.x.x.x:45703 [BREAK] > 2013-06-06 17:13:16.319289 [DEBUG] switch_ivr_bridge.c:533 > sofia/myprofile/sip:200@x.x.x.x:45703 ending bridge by request from > read function > > I do not use any ivr module, or start_dtmf, or whatsoever setting > related to dtmf. Any hint? > > Thanks, > Mino >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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You shouldn't need to set that. Are you sure it's not caused by the low RTP timeout of 5 seconds on the profile? /Peter 2013/6/10 Mino Haluz <mino.haluz@gmail.com>
Ok timer_name = soft did it and now it works. Can someone elaborate what this value means in fact?
On Mon, Jun 10, 2013 at 3:11 PM, Mino Haluz <mino.haluz@gmail.com> wrote:
Complete log with siptrace on: http://pastebin.freeswitch.org/21046
On Mon, Jun 10, 2013 at 6:51 AM, Brian Foster <bdfoster@davri.com> wrote:
Need a siptrace, please run 'sofia global siptrace on' before you take the console log and post it on pastebin.freeswitch.org. Reply back with the link. Hopefully we can glean some more info.
- BDF On Jun 7, 2013 5:09 AM, "Mino Haluz" <mino.haluz@gmail.com> wrote:
I forgot to delete all sofia_reg lines (registrations made a bit mess in call log). So for example this line should not be there:
2013-06-07 10:47:31.919290 [DEBUG] sofia_reg.c:1520 Send challenge for [300@192.168.80.55]
On Fri, Jun 7, 2013 at 10:57 AM, Mino Haluz <mino.haluz@gmail.com>wrote:
Hi,
so this is the call log: http://pastebin.freeswitch.org/21035
the only dialplan I have: http://pastebin.freeswitch.org/21036
my sip profile: http://pastebin.freeswitch.org/21038
anything else?
Mino
On Thu, Jun 6, 2013 at 8:12 PM, Steven Ayre <steveayre@gmail.com>wrote:
Can you share a complete debug-level log of the call? Are any other extensions being executed first?
There are api calls that could bind a dtmf key to hanging up the bridge, perhaps that's being done here earlier in the dialplan.
-Steve
On Thursday, June 6, 2013, Mino Haluz wrote:
> FS hangs up the call when it receives RTP DTMF Event. > > FreeSwitch 1.2.10 > > profile: > <profile name="myprofile"> > <settings> > <param name="context" value="mycontext"/> > <param name="debug" value="1"/> > <param name="rfc2833-pt" value="101"/> > <param name="sip-port" value="5060"/> > <param name="dialplan" value="XML"/> > <param name="dtmf-duration" value="100"/> > <param name="dtmf-type" value="rfc2833"/> > <param name="liberal-dtmf" value="true"/> > <param name="codec-ms" value="20"/>--> > <param name="use-rtp-timer" value="true"/> > <param name="sip-ip" value="192.168.x.x"/> > <param name="rtp-ip" value="192.168.x.x"/> > <param name="rtp-timeout-sec" value="3000"/> > <param name="manage-presence" value="false"/> > <param name="rtp-timeout-sec" value="5"/> > <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> > <param name="outbound-codec-prefs" > value="$${global_codec_prefs}"/> > </settings> > </profile> > > dialplan: > <extension name="Local_Extension_3"> > <condition field="destination_number" expression="^300$"> > <action application="bridge" > data="sofia/myprofile/300@192.168.x.x:5060"/> > </condition> > </extension> > > modules: > <load module="mod_console"/> > <load module="mod_logfile"/> > <load module="mod_enum"/> > <load module="mod_event_socket"/> > <load module="mod_sofia"/> > <load module="mod_loopback"/> > <load module="mod_commands"/> > <load module="mod_dptools"/> > <load module="mod_expr"/> > <load module="mod_fifo"/> > <load module="mod_hash"/> > <load module="mod_amr"/> > <load module="mod_speex"/> > <load module="mod_spandsp"/> > <load module="mod_com_g729"/> > <load module="mod_cluechoo"/>--> > <load module="mod_dialplan_xml"/> > <load module="mod_g723_1"/> > <load module="mod_native_file"/> > <load module="mod_local_stream"/> > <load module="mod_tone_stream"/> > <load module="mod_xml_cdr"/> > > > > On Thu, Jun 6, 2013 at 5:20 PM, Mino Haluz <mino.haluz@gmail.com>wrote: > >> Hi, >> >> if I use SIP INFO on leg A, FS detects everything fine, but in case >> I use rfc2833, after pressing number, it instantly hangs up the call. >> >> n2013-06-06 17:13:16.319289 [NOTICE] mod_sofia.c:1137 Hangup >> sofia/myprofile/sip:200@x.x.x.x:45703 [CS_EXCHANGE_MEDIA] >> [MEDIA_TIMEOUT] >> 2013-06-06 17:13:16.319289 [DEBUG] switch_channel.c:3096 Send >> signal sofia/myprofile/sip:200@x.x.x.x:45703 [KILL] >> 2013-06-06 17:13:16.319289 [DEBUG] switch_core_session.c:1333 Send >> signal sofia/myprofile/sip:200@x.x.x.x:45703 [BREAK] >> 2013-06-06 17:13:16.319289 [DEBUG] switch_ivr_bridge.c:533 >> sofia/myprofile/sip:200@x.x.x.x:45703 ending bridge by request >> from read function >> >> I do not use any ivr module, or start_dtmf, or whatsoever setting >> related to dtmf. Any hint? >> >> Thanks, >> Mino >> > >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ 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
DTMF tone lasted 100ms, so I am sure this timer could not be hit.. When I delete the soft timer, it does not work. Maybe it's something with - http://wiki.freeswitch.org/wiki/RTP_Issues On Mon, Jun 10, 2013 at 4:19 PM, Peter Olsson <peter@olssononline.se> wrote:
You shouldn't need to set that. Are you sure it's not caused by the low RTP timeout of 5 seconds on the profile?
/Peter
2013/6/10 Mino Haluz <mino.haluz@gmail.com>
Ok timer_name = soft did it and now it works. Can someone elaborate what this value means in fact?
On Mon, Jun 10, 2013 at 3:11 PM, Mino Haluz <mino.haluz@gmail.com> wrote:
Complete log with siptrace on: http://pastebin.freeswitch.org/21046
On Mon, Jun 10, 2013 at 6:51 AM, Brian Foster <bdfoster@davri.com>wrote:
Need a siptrace, please run 'sofia global siptrace on' before you take the console log and post it on pastebin.freeswitch.org. Reply back with the link. Hopefully we can glean some more info.
- BDF On Jun 7, 2013 5:09 AM, "Mino Haluz" <mino.haluz@gmail.com> wrote:
I forgot to delete all sofia_reg lines (registrations made a bit mess in call log). So for example this line should not be there:
2013-06-07 10:47:31.919290 [DEBUG] sofia_reg.c:1520 Send challenge for [300@192.168.80.55]
On Fri, Jun 7, 2013 at 10:57 AM, Mino Haluz <mino.haluz@gmail.com>wrote:
Hi,
so this is the call log: http://pastebin.freeswitch.org/21035
the only dialplan I have: http://pastebin.freeswitch.org/21036
my sip profile: http://pastebin.freeswitch.org/21038
anything else?
Mino
On Thu, Jun 6, 2013 at 8:12 PM, Steven Ayre <steveayre@gmail.com>wrote:
> Can you share a complete debug-level log of the call? Are any other > extensions being executed first? > > There are api calls that could bind a dtmf key to hanging up the > bridge, perhaps that's being done here earlier in the dialplan. > > -Steve > > > > On Thursday, June 6, 2013, Mino Haluz wrote: > >> FS hangs up the call when it receives RTP DTMF Event. >> >> FreeSwitch 1.2.10 >> >> profile: >> <profile name="myprofile"> >> <settings> >> <param name="context" value="mycontext"/> >> <param name="debug" value="1"/> >> <param name="rfc2833-pt" value="101"/> >> <param name="sip-port" value="5060"/> >> <param name="dialplan" value="XML"/> >> <param name="dtmf-duration" value="100"/> >> <param name="dtmf-type" value="rfc2833"/> >> <param name="liberal-dtmf" value="true"/> >> <param name="codec-ms" value="20"/>--> >> <param name="use-rtp-timer" value="true"/> >> <param name="sip-ip" value="192.168.x.x"/> >> <param name="rtp-ip" value="192.168.x.x"/> >> <param name="rtp-timeout-sec" value="3000"/> >> <param name="manage-presence" value="false"/> >> <param name="rtp-timeout-sec" value="5"/> >> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> >> <param name="outbound-codec-prefs" >> value="$${global_codec_prefs}"/> >> </settings> >> </profile> >> >> dialplan: >> <extension name="Local_Extension_3"> >> <condition field="destination_number" expression="^300$"> >> <action application="bridge" >> data="sofia/myprofile/300@192.168.x.x:5060"/> >> </condition> >> </extension> >> >> modules: >> <load module="mod_console"/> >> <load module="mod_logfile"/> >> <load module="mod_enum"/> >> <load module="mod_event_socket"/> >> <load module="mod_sofia"/> >> <load module="mod_loopback"/> >> <load module="mod_commands"/> >> <load module="mod_dptools"/> >> <load module="mod_expr"/> >> <load module="mod_fifo"/> >> <load module="mod_hash"/> >> <load module="mod_amr"/> >> <load module="mod_speex"/> >> <load module="mod_spandsp"/> >> <load module="mod_com_g729"/> >> <load module="mod_cluechoo"/>--> >> <load module="mod_dialplan_xml"/> >> <load module="mod_g723_1"/> >> <load module="mod_native_file"/> >> <load module="mod_local_stream"/> >> <load module="mod_tone_stream"/> >> <load module="mod_xml_cdr"/> >> >> >> >> On Thu, Jun 6, 2013 at 5:20 PM, Mino Haluz <mino.haluz@gmail.com>wrote: >> >>> Hi, >>> >>> if I use SIP INFO on leg A, FS detects everything fine, but in >>> case I use rfc2833, after pressing number, it instantly hangs up the call. >>> >>> n2013-06-06 17:13:16.319289 [NOTICE] mod_sofia.c:1137 Hangup >>> sofia/myprofile/sip:200@x.x.x.x:45703 [CS_EXCHANGE_MEDIA] >>> [MEDIA_TIMEOUT] >>> 2013-06-06 17:13:16.319289 [DEBUG] switch_channel.c:3096 Send >>> signal sofia/myprofile/sip:200@x.x.x.x:45703 [KILL] >>> 2013-06-06 17:13:16.319289 [DEBUG] switch_core_session.c:1333 Send >>> signal sofia/myprofile/sip:200@x.x.x.x:45703 [BREAK] >>> 2013-06-06 17:13:16.319289 [DEBUG] switch_ivr_bridge.c:533 >>> sofia/myprofile/sip:200@x.x.x.x:45703 ending bridge by request >>> from read function >>> >>> I do not use any ivr module, or start_dtmf, or whatsoever setting >>> related to dtmf. Any hint? >>> >>> Thanks, >>> Mino >>> >> >> > > _________________________________________________________________________ > 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
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Are you pressing * by any chance? In any case, get a complete call log from start to finish and put on pastebin.freeswitch.org with "FreeSWITCH Log" as syntax highlighting. -MC On Thu, Jun 6, 2013 at 8:20 AM, Mino Haluz <mino.haluz@gmail.com> wrote:
Hi,
if I use SIP INFO on leg A, FS detects everything fine, but in case I use rfc2833, after pressing number, it instantly hangs up the call.
n2013-06-06 17:13:16.319289 [NOTICE] mod_sofia.c:1137 Hangup sofia/myprofile/sip:200@x.x.x.x:45703 [CS_EXCHANGE_MEDIA] [MEDIA_TIMEOUT] 2013-06-06 17:13:16.319289 [DEBUG] switch_channel.c:3096 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [KILL] 2013-06-06 17:13:16.319289 [DEBUG] switch_core_session.c:1333 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [BREAK] 2013-06-06 17:13:16.319289 [DEBUG] switch_ivr_bridge.c:533 sofia/myprofile/sip:200@x.x.x.x:45703 ending bridge by request from read function
I do not use any ivr module, or start_dtmf, or whatsoever setting related to dtmf. Any hint?
Thanks, Mino
_________________________________________________________________________ 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
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
participants (5)
-
Brian Foster
-
Michael Collins
-
Mino Haluz
-
Peter Olsson
-
Steven Ayre