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
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