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Stacked conditions are not acting like logical AND
by Herman Griffin 18 Jan '12

18 Jan '12

Hello Freeswitchers, I'm using stacked condition that are supposed to behave line logical AND operators. The call trace has been separated in to two section. The first section corresponds to the first set of stacked conditions in my dialplan and the second section corresponds to my second set of stacked conditions. The second condition in the second set of stack condition evaluates to FAIL (false). However the 'action' that are wrapped are still being exacuted. I expect the 'anti-action' to execute when any of the AND operands evaluates to FAIL. Note: I'm using break=never because I don't want a failed condition to break me out of the extension. There are multiple condition blocks in the extension. Regardless of whether break being set to never, it seems that the wrapped actions should not execute unless all conditions evalute to PASS (true) Can someone help point out my mistake? Thank you, Herman Griffin ------------------------------------ This is my call trace: Dialplan: sofia/external/Unknown(a)72.37.252.18 Regex (PASS) [emergency] ${open}(true) =~ /^true$/ break=never Dialplan: sofia/external/Unknown(a)72.37.252.18 Regex (FAIL) [emergency] ${emergency_call}() =~ /^true$/ break=never Dialplan: sofia/external/Unknown(a)72.37.252.18 Regex (PASS) [emergency] ${sip_gateway}(1006_7217) =~ /^1006_7217$/ break=never Dialplan: sofia/external/Unknown(a)72.37.252.18 Regex (FAIL) [emergency] ${emergency_call}() =~ /^true$/ break=never Dialplan: sofia/external/Unknown(a)72.37.252.18 Regex (PASS) [emergency] ${db(select/emergency/autoanswer)}(1) =~ /^1$/ break=never Dialplan: sofia/external/Unknown(a)72.37.252.18 Action set(call_timeout=60) Dialplan: sofia/external/Unknown(a)72.37.252.18 Action set(effective_caller_id_name=${regex(${caller_id_name}|^Emerg(_.*)$|Auto%1)}) Dialplan: sofia/external/Unknown(a)72.37.252.18 Action set(autoanswered=true) Dialplan: sofia/external/Unknown(a)72.37.252.18 Action answer() Dialplan: sofia/external/Unknown(a)72.37.252.18 Action set(group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/emergency/press_to_accept.wav) Dialplan: sofia/external/Unknown(a)72.37.252.18 Action set(group_confirm_key=1) Dialplan: sofia/external/Unknown(a)72.37.252.18 Action bridge(user/1000@${domain_name},sofia/gateway/1006_7217/${mobile_number}) ------------------------------------ This is the corresponding dialplan: <condition field="${open}" expression="^true$" break="never"/> <condition field="${emergency_call}" expression="^true$" break="never"> <action application="set" data="session_in_hangup_hook=true"/> <action application="set" data="api_hangup_hook=python emergency.hangup"/> </condition> <condition field="${sip_gateway}" expression="^${default_provider}$" break="never"/> <condition field="${emergency_call}" expression="^true$" break="never"/> <condition field="${db(select/emergency/autoanswer)}" expression="^1$" break="never"> <action application="set" data="call_timeout=60"/> <action application="set" data="effective_caller_id_name=${regex(${caller_id_name}|^Emerg(_.*)$|Auto%1)}"/> <action application="set" data="autoanswered=true"/> <action application="answer" data=""/> <action application="set" data="group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/emergency/press_to_accept.wav"/> <action application="set" data="group_confirm_key=1"/> <action application="bridge" data="user/1000@${domain_name},sofia/gateway/1006_7217/${mobile_number}"/>  <anti-action application="set" data="effective_caller_id_name=${regex(${caller_id_name}|^Emerg(_.*)$|NotAuto%1)}"/> <anti-action application="set" data="call_timeout=30"/> <anti-action application="set" data="autoanswered=false"/> <anti-action application="bridge" data="user/1000@${domain_name},sofia/gateway/1006_7217/${mobile_number}"/> </condition>

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Dahdi module load error
by Gopalakrishnan N 18 Jan '12

18 Jan '12

Hi, We compiled the mod_dahdi_codec it got compiled when we try to load the module from freswitch command line it gives an error as freeswitch(a)localhost.localdomain> load mod_dahdi_codec -ERR [module load file routine returned an error] 2012-01-18 23:07:56.654211 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_dahdi_codec.so **/usr/local/freeswitch/mod/mod_dahdi_codec.so: invalid ELF header** freeswitch(a)localhost.localdomain> Can you please help in this matter. Regards.

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Building Customized module for Transcoding Card
by Gopalakrishnan N 18 Jan '12

18 Jan '12

Hi, We are developing a module for freeswitch for transcoding card we have some static libraries with extension (*.a) which has to be linked to get the final module for freeswitch. Can any one please help in letting us know where the static libraries are to be placed? And with what option it has to be included in the Makefile. Thanks in advance. Regards.

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sip_auto_answer auto copy to bleg
by Seven Du 18 Jan '12

18 Jan '12

Hi, I use originate {sip_auto_answer}user/1000 &bridge(user/1001) and want 1000 to be auto answered but not 1001, but it seems FS automatically copy to b-leg. I manually comment the following lines and it seems to work. // switch_channel_set_variable(nchannel, "sip_invite_params", "intercom=true"); } switch_ivr_transfer_variable(session, nsession, SOFIA_REPLACES_HEADER); // switch_ivr_transfer_variable(session, nsession, "sip_auto_answer"); would it be good to default to not copy but leave the "export" or equivalent to do that? Or should I patch to add a var to disable that? Thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com)

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announce count mod conference
by Rodney 17 Jan '12

17 Jan '12

the following section of mod conference does not seem to work for me. is this meant to be played for the whole room or just the user entering the room. I would like it only to work for the person entering the room to cut down on the amount of them pressing the dtmf to execute the announce count extension. it seems this is the first thing they do and I want them to not have to do that. I have set this to 0 and 1 and still no caller count announced. how may I trouble shoot this problem? thanks. when i go into an empty room it says my zerocallers.wav but nothing else if i enter a room with 1 or more callers. for announce-count Requires TTS. The system will speak the total number of callers in the conference when a new person joins, but only once the threshold specified in this parameter is reached. 5

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FreeSWITCH Community Conference Call Tomorrow
by Michael Collins 17 Jan '12

17 Jan '12

Hello all! Just a reminder that we have our FreeSWITCH community conference call tomorrow. We have Moc scheduled to discuss some of the new things he's been cooking up. However, he might have to go play with his cable modem for a bit so we'll see how the schedule pans out. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_01_18 Make a note of the upcoming conference calls. Normally I could say that we are booked solid through February, but since 2012 is a leap year we actually have a call scheduled for February 29. I'd love to hear some suggestions for a special leap-year edition of the FS community conference call! Let me know what you think. :) Thanks, Michael

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BLF gets confused with multiple calls
by Tim St. Pierre 17 Jan '12

17 Jan '12

Hello, I'm having all sorts of problems with BLFs not being in the correct state. It works fine in some cases, but others are wrong. Here's what I can see: Single endpoint, single call, everything works fine. Light flashes on ring, goes steady on answer, goes out on hangup. Here's where it gets tricky: If there are two phones registered to the extension, it flashes when they ring, but then goes out when one of them answers. If either phone places an outgoing call, the lamp comes on. If a single phone gets a second call, their lamp flashes again, but then goes out when they answer the second call. I'm constantly getting complaints from users where the lamps are stuck on. It happens more often when they have a lot of phones in ring groups. The lamps work fine for ring groups - they all flash, and whomever picks up the call stays steady while the rest go out. Is there anything I can do to get freeswitch to base the state on whether or not that user has any active calls, rather than just what the last thing that the phone did was? This happens on every model of Aastra phone, and I have all of them. I haven't had a chance to try it yet on Polycom. I'm running FreeSWITCH Version 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500) Any ideas would be more than welcome. -Tim

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record_fsv buffers - Any way to flush incoming video?
by Greg Millam 17 Jan '12

17 Jan '12

Hi folks - I have a freeswitch dialplan + script that first calls play_fsv to play a greeting, then record_fsv to record incoming video to fsv. It works fine, but there's one issue: Apparently, freeswitch is buffering incoming video during play_fsv. When record_fsv is called, that buffer is dumped into the .fsv file, resulting in several seconds of unneeded video. Is there a way to empty that incoming video buffer before record_fsv begins recording? Thank you! - Greg Millam

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eavesdrop problem
by Drazen Blanusa 17 Jan '12

17 Jan '12

Hi, I have one FS instance running localy, and two clients connected. I make one call from A(1019), and one call from B(1010). I want to spy channel A from channel B. I'm using application eavesdrop(uuid) and this is correct. But after application is executed, I hear Input from A mic in A speaker. If I use eavesdrop(all), I hear A and B mic in A speaker. I checked and uuid is correct. >From log: switch_ivr.c:589 sofia/internal/1019(a)XX.XXX.X.XX Command Execute eavesdrop(45b73c25-a1e7-4435-8214-76fecca7c6cd) I'm not using dialplan, just <context name="default"> <extension name="1"> <condition> <action application="park"/> </condition> </extension> Also,I'm using ESL . Any suggestions?Why it is inverted? Tnx in advance! Drazen

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Annoying error while compiling FreeSWITCH
by Muhammad Shahzad 17 Jan '12

17 Jan '12

I am getting build failure repeatedly due to this annoying -Werror. Each time any warning appears in code, the compilation fails treating it as error, e.g. /usr/src/svn-src/freeswitch/src/mod/applications/mod_voicemail_ivr/mod_voicemail_ivr.c:57:6: error: variable 'argc' set but not used [-Werror=unused-but-set-variable] cc1: all warnings being treated as errors Can you guy help me how to get rid of it? I tried finding and replacing -Werror with -Wall but there are so many make files to edit.. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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