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What are <> in bridge as opposed to {} []
by Mario 06 Oct '10

06 Oct '10

The wiki explains what {} and [] are used for in the line below, but I was provided <> from this list that fixed a problem and I can't find how <> is different from the others. Anyone? <action application="bridge" data="<originate_timeout=15>group/MGT@${domain_name}:_:sofia/gate...

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ONE WEEK LEFT - Official FreeSWITCH Training in New York City
by Darren Schreiber 06 Oct '10

06 Oct '10

Hi folks, The Official FreeSWITCH Training course in New York is just one week away and is almost full. If you are interested in learning a ton of FreeSWITCH, taught by one of the authors of the FreeSWITCH Book, you should consider signing up. Code FREEVOIP is still valid for $400 off the training. You can register at http://www.voipkb.com/ . There is also a huge discount for registering two or more people together. The course is an in-depth dive into FreeSWITCH with completely different material then is in the book. We will cover: * Understanding configuration files and the default configuration * Call authentication and routing basics * Integration modules (mod_skypiax, mod_dingaling for Skype/GTalk/XMPP integration) * Understanding presence * Load balancing and high availability * FreeSWITCH Internals * How to debug and troubleshoot FreeSWITCH * Building Custom C Modules * Advanced Modules Please let me know if you have any questions. We're excited to continue offering this course and thank the FreeSWITCH team for their continued outstanding work. See you in New York! Thanks, Darren Schreiber Join us in October for FreeSWITCH Training! Visit www.voipkb.com<http://www.voipkb.com/> for more information

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Sending "alert-info:...." to devices from Conference
by Jon Young 06 Oct '10

06 Oct '10

Hi, I'm using a derivative of the "mad_boss_intercom" to set up paging. I have some devices that need an "alert-info:<...url...>; info=alert-autoanswer" in order to auto answer. Can I set or export that from within the paging/intercom XML or can it be included in the "conference_auto_outcall_prefix". Here is what one of the zones looks like: <extension name="paging01"> <condition field="destination_number" expression="^1999$"> <action application="set" data="conference_auto_outcall_caller_id_name=Page1"/> <action application="set" data="conference_auto_outcall_caller_id_number=1999"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/> <action application="set" data="sip_exclude_contact=${network_addr}"/> <action application="conference_set_auto_outcall" data="${group_call(pagezone01)}"/> <action application="conference" data="paging1@default+flags{endconf|deaf}"/> </condition> </extension> This currently works fine with Aastra phones, but not my other phones which need the "alert-info..." Thanks, Jon

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Inbound caller ID lost after return from bridge
by Mario 06 Oct '10

06 Oct '10

THe extension below works fine except that the inbound caller ID is lost between the first and second bridge, it is replaced by 0000000000 on the local extensions. Does the first bridge function cause the inbound caller ID to be lost? Is there a way to keep it so it shows up during the second bridge? Thanks. Mario <extension name="u1phone"> <condition field="destination_number" expression="^2920$">  <action application="set" data="ringback=${us-ring}"/>  <action application="execute_extension" data="2921 XML default"/>  <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/>  <action application="set" data="call_timeout=15"/> <action application="bridge" data="group/MXT@${domain_name}"/>  <action application="execute_extension" data="2922 XML default"/> <action application="playback" data="/usr/local/freeswitch/conf/mvsounds/we_trying_please_hold.wav"/>  <action application="set" data="ringback=$${hold_music}"/>  <action application="set" data="ignore_early_media=true"/>    to ring simoultaneously --> <action application="bridge" data="<originate_timeout=15>group/MXT@${domain_name}:_:sofia/gateway/uid1/19161234567:_:sofia/gateway/uid1/19162223333"/>  <action application="answer"/> <action application="voicemail" data="default ${domain_name} 100"/> </condition> </extension>

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FreeSWITCH CLI description?
by Stas Khirman 06 Oct '10

06 Oct '10

Hi, Sorry for probably dumb questions, but I can't find answer in the docs/wiki: I'm looking for a good list of major CLI commands and their descriptions - any pointers are deeply appreciated. Regards Stas

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compare bill rate of different increment
by Woody Dickson 06 Oct '10

06 Oct '10

Hi, I would like to use mod_lcr but I am wondering how I should due with rate decks from different providers that have different bill increment? If one deck's increment is 6 s and another one is 1 s, is it fair to compare the two deck just purely based on the rate or is there a better way of handling that? Thank you for your input. Thanks, Woody

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Application Play_fsv issues
by Joshua Foshee 06 Oct '10

06 Oct '10

I have a problem that if I play back a FSV file that it plays super-fast. It records fine though when it plays back I see the video and auto go by fast and then it plays a long busy tone. Any ideas? Thanks in advance, Josh

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FreeSWITCH Conference Call Today
by Michael Collins 06 Oct '10

06 Oct '10

Hello all! Our agenda for today is here: http://wiki.freeswitch.org/wiki/FS_weekly_2010_10_06 We have a few things to discuss, such as some additions to the default configs and some Jira tasks that we could use some help with. Also, we are hoping that Darren Schreiber will be calling in to give us an update from ITExpo down in L.A. Talk to you soon, Michael

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FW: Play_fsv application Issue
by Joshua Foshee 06 Oct '10

06 Oct '10

I have something strange happen when I try to play back a fsv file. It will play super-fast. You can hear the audio go fast too then you get a long busy. Any ideas what would cause the speed to increase on playback? Thanks, Josh

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Problem bridging 2 calls with bypass_media=true
by Pete Kelly 06 Oct '10

06 Oct '10

Hi I am having some problems bridging 2 calls and bypassing the media. I am using a lua script executed from the command line to phone out to the first device (leg A), then on answer bridge a call to a second device (leg B). If I set the bypass_media=true, then freeswitch immediately sends out BYEs after the reINVITEs have happened. If I leave bypass_media unset, everything works fine except the media is bridged. The lua script I am using is a very simple one and looks like this: obSession = freeswitch.Session("sofia/lpmedia/91979197@10.15.20.122:5060") obSession:setVariable('bypass_media', 'true'); if obSession:ready() then -- Do something good here obSession:execute("bridge", "sofia/lpmedia/2000@10.15.20.122:5060") end I have tried variations on this including, putting [bypass_media=true] on leg A, and using the [bypass_media_after_bridge=true] on leg B The logs say this: ... 2010-10-06 10:10:05.847278 [NOTICE] sofia.c:5085 Channel [sofia/lpmedia/ 2000@10.15.20.122:5060] has been answered 2010-10-06 10:10:06.650155 [NOTICE] switch_cpp.cpp:976 Hangup sofia/lpmedia/ 91979197@10.15.20.122:5060 [CS_HIBERNATE] [NORMAL_CLEARING] 2010-10-06 10:10:06.650155 [NOTICE] switch_ivr_bridge.c:936 Hangup sofia/lpmedia/2000@10.15.20.122:5060 [CS_HIBERNATE] [NORMAL_CLEARING] 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1228 Session 27 (sofia/lpmedia/91979197@10.15.20.122:5060) Ended 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close Channel sofia/lpmedia/91979197@10.15.20.122:5060 [CS_DESTROY] freeswitch@pete-desktop> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1228 Session 28 (sofia/lpmedia/2000@10.15.20.122:5060) Ended 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close Channel sofia/lpmedia/2000@10.15.20.122:5060 [CS_DESTROY] Does anyone have any ideas as to why freeswitch may be ending the calls? I am using the latest git checkout - the behaviour was slightly different with the 1.06 package. The call would setup but the reINVITEs did not happen as they should. Pete

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