OK, here is a pastebin minus the sip tracing with the highlighting: http://pastebin.freeswitch.org/19788 On 2012-08-30 19:39:02 +0000, Michael Collins said:
Dave, Can you turn off the sofia debugging? It's almost completely useless in these scenarios. (only a few devs really understand it and they'll let you know if you need to turn it on.) Also, please use "FreeSWITCH Log" as syntax highlighting - that makes life much easier for those of us whose eyes have higher mileage... -MC
On Thu, Aug 30, 2012 at 12:31 PM, Dave Horton <daveh@beachdognet.com> wrote: On 2012-08-30 18:59:34 +0000, Dave Horton said:
Next I will try the conferencing version and paste those results
Below is the configuration I used with the bridged conferencing. Something must be wrong because there is no audio path between the a and b legs; also, when b hangs up a stays connected.
Pastebin is at http://pastebin.freeswitch.org/19786 and configuration is below. I'd welcome any thoughts on what is wrong...
<extension name="6174973896" continue="false"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="6174973896"> <action application="set" data="call_limit=1"/> <action application="set" data="callFactor=1"/> <action application="set" data="customer=cps_play"/> <action application="ring_ready"/> <action application="pre_answer"/> <action application="start_dtmf"/> <action application="set" data="call_id=${strftime(%Y%m%d_%H%M%S)}_${sip_from_tag}"/> <action application="set" data="outfile=$${base_dir}/recordings/${call_id}.wav"/> <action application="export" data="nolocal:jitterbuffer_msec=100"/> <action application="set" data="RECORD_STEREO=true"/> <action application="set" data="aleg_uuid=${uuid}"/> <action application="set" data="call_id=${strftime(%Y%m%d_%H%M%S)}_${sip_from_tag}"/> <action application="set" data="outfile=$${base_dir}/recordings/${call_id}.wav"/> <action application="record_session" data="${outfile}"/> <action application="set" data="progress_timeout=60"/> <action application="set" data="call_timeout=120"/> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE"/> <action application="set" data="hangup_after_bridge=true"/> <action application="conference" data="bridge:_uuid_@simple:sofia/normal_customer/15083084809@${egress_gateway}"/>
<action application="set" data="conference_enter_sound=misc/this-call-is-being-recorded"/>
</condition> </extension>
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