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call pickup vs application="intercept"
by Sean Devoy 18 Apr '14

18 Apr '14

Hi all, I have just installed a new site where the receptionist has a CISCO SPA phone and 32 key extended keypad. The buttons on the keypad are used like an operator console for various extensions. They provide Busy Line Feature (BLF or BLA) and Speed Dial (sd). I will update the CISCO SPA section of the WIKI to document how to do this as soon as I get an email with my username and password back! However, I think I need some clarification on "pickup" verses "intercept". To me these two dialplans are exactly equivalent, please correct me if I missed something. Note, I have restricted the pickup "group" to the dialed extension, eliminating the problem of multiple incoming calls when picking up in a "group". Scenario 1: Main dialplan routes call to extension "10" (simplified version): <extension name="2digit_internal" > <condition field="destination_number" expression="^(10)$" > [SNIP ...] <action application="db" data="insert/call_pickup_uuid/${destination_number}/${uuid}"/> <action application="bridge" data="{sip_invite_domain=[REDECTED]}user/${destination_number}@[REDECTED]" /> <action application="answer"/> <action application="voicemail" data="default voicemail_ESTA $1"/> <action application="hangup"/> </condition> </extension> The application "db" line is the key to link to the events. Now when the receptionist sees extension 10 blinking and unanswered, she can press that key to "pickup" that ringing line. (The Cisco phone dials **10) That dialplan is: <extension name="directed-call-pickup"> <condition field="destination_number" expression="^\*\*(\d\d)$"> <action application="answer"/> <action application="intercept" data="${db(select/call_pickup_uuid/$1)}"/> <action application="sleep" data="2000"/> </condition> </extension> The receptionist "intercepts" the call by uuid stored in the above dialplan. Scenario 2: (same, but different dial plan) <extension name="2digit_internal" > <condition field="destination_number" expression="^(10)$" > [SNIP ...] <action application="bridge" data="{sip_invite_domain=[REDECTED]}user/${destination_number}@[REDECTED], pickup/$1" /> <action application="answer"/> <action application="voicemail" data="default voicemail_ESTA $1"/> <action application="hangup"/> </condition> </extension> No application "db" line, instead added ", pickup/10" to the bridge line. Now when the receptionist sees extension 10 blinking and unanswered, she can press that key to "pickup" that ringing line. (The Cisco phone dials **10) That dialplan is: <extension name="directed-call-pickup"> <condition field="destination_number" expression="^\*\*(\d\d)$"> <action application="answer"/> <action application="pickup" data="$1)}"/> <action application="sleep" data="2000"/> </condition> </extension> So, is this just two different ways to skin a cat or are there subtleties that make one a preferred implementation?

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Couldnt open /etc/freeswitch/freeswitch.xml
by Joli Martinez 17 Apr '14

17 Apr '14

Hello, I have a production Freeswitch box and this morning I noticed FS was not working. I restarted it and got this message. As far as I know no one has touched the box, but not 100% sure. Any ideas why I am getting this. I researched and a lot of the hits I got was due to a bad install, but this box has been working correctly for over 6 months. [ERR] switch_xml.c:1385 Couldn't open /etc/freeswitch/freeswitch.xml (No such file or directory) Thanks,

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The Sipml5 cannot register to freeswitch
by rockxsj 17 Apr '14

17 Apr '14

I found that sometimes the sipml5 client can not register to freeswitch.When I use tcpdump to capture the packet I can see the three-way handshake,but I execute show channels on the freeswitch can not see my channel.Then after several minutes,I re-try call,successful. Why?Could you help me?

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bleg_uuid
by mbo 17 Apr '14

17 Apr '14

I noticed that the variable bleg_uuid is only set when a bridged call gets connected, not if the call end with hangup cause NO_ANSWER or BUSY. Is this by intension? I need to linl the both call legs even when the call get’s bot connected, can I set the bleg_uuid by my self on bridge with the uuid of the a_leg or is this not a good idea? Thanks Markus

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if user_busy then continue_on_fail=false else continue_on_fail=true
by Shahzad Bhatti 16 Apr '14

16 Apr '14

hi, i want to know how to impalement in XML_curl dialplan that *if user busy then **set continue_on_fail=false **else * *continue_on_fail=true* it means when freeswitch get response *USER_BUSY then it stop attempt otherwise continue.* here is my sample dialplan: <document type="freeswitch/xml"> <section name="dialplan" description="RE Dial Plan For FreeSwitch"> <context name="public"> <extension name="Bridge Call to Destination"> <condition field="destination_number" expression="^004917691444955$"> <action application="set" data="process_cdr=b_only"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="nibble_account=1"/> <action application="set" data="nibble_rate=0"/> <action application="export" data="origination_caller_id_number=004917691444955"/> <action application="export" data="nolocal:nibble_account=1"/> <action application="export" data="cust_ip=186.218.49.95"/> <action application="export" data="nolocal:nibble_rate=0.0229"/> <action application="export" data="nolocal:connection_charges=0"/> <action application="export" data="buy_rate=0.0199"/> <action application="export" data="sell_rate=0.0229"/> <action application="export" data="destination_id=470"/> <action application="export" data="destination_name=Germany Mobile O2"/> <action application="export" data="provider_id=9"/> <action application="export" data="domain_ip=194.152.251.110"/> <action application="export" data="nolocal:execute_on_answer=nibblebill adjust -0" /> <action application="export" data="absolute_codec_string=${ep_codec_string},PCMU,G729"/> <action application="phrase" data="sayrate,0:2:29"/> <action application="bridge" data="{absolute_codec_string=PCMA,G729}sofia/external/ 004917691444955(a)194.152.251.110"/> </condition> </extension> </context> </section> </document> Regards Shahzad Bhatti

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Microsoft Lync Integration?
by Jerry Richards 16 Apr '14

16 Apr '14

Does Freeswitch support Microsoft LYNC integration? Or is anyone planning for that? I saw some mailing list references back in 2011/2012, but there are no references in the Freeswitch wiki. Regards, Jerry

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To DEVs and all Forum Responders
by Sean Devoy 16 Apr '14

16 Apr '14

This doesn't get said enough ... THANK YOU. >From my customer with the recent degraded audio problem YOU fixed: Sean & Bobby, Good morning! I don't remember personally thanking you both for fixing my phone! Especially Sean, tirelessly day after day keeping at it, not giving in easily. It feels so good hearing Customers clearly and not complaining about hearing a word I said, or Customers talking over me! Thank you, Thank you! I am so appreciative. Shalin "Sean, tirelessly day after day keeping at it, not giving in easily" translates to "Sean hounding the forum for more help every day." Thanks for a GREAT product and GREAT support.

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ClueCon Weekly! Join us today at 1PM Eastern
by Ken Rice 16 Apr '14

16 Apr '14

Hey guys, don¹t forget to join us today at 1PM EST for ClueCon Weekly on the 888 Bridge. sip:888@conference.freeswitch.org 1PM Eastern 10AM Pacific. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Include gateways via mod_xml_curl
by Daniel Neubert 16 Apr '14

16 Apr '14

I've tried to include custom gateways via mod_xml_curl. The request is correctly issued by mod_xml_curl and received by the backend: "hostname=fire&section=directory&tag_name=domain&key_name=name&key_value=user&Event-Name=REQUEST_PARAMS&Core-UUID=18b2b698-c487-11e3-bcc9-39daad6bb1f3&FreeSWITCH-Hostname=fire&FreeSWITCH-Switchname=fire&FreeSWITCH-IPv4=192.123.321.123&FreeSWITCH-IPv6=%3A%3A1&Event-Date-Local=2014-04-15%2012%3A48%3A57&Event-Date-GMT=Tue,%2015%20Apr%202014%2010%3A48%3A57%20GMT&Event-Date-Timestamp=1397558937493970&Event-Calling-File=sofia.c&Event-Calling-Function=config_sofia&Event-Calling-Line-Number=5174&Event-Sequence=1510&purpose=gateways&profile=user" The backend interprets the request by "purpose=gateways" and "profile=user" to send back all gateways required for profile "user" <document type="freeswitch/xml"> <section name="configuration"> <include> <gateway name="user_gw_1_foobar"> <param name="username" value=""/> <param name="password" value=""/> <param name="proxy" value="foobar.example.de"/> <param name="register" value="false"/> <param name="expire-seconds" value="300"/> <param name="ping" value="300"/> <param name="retry-seconds" value="300"/> <param name="context" value="default"/> <param name="caller-id-in-from" value="true"/> </gateway> </include> </section> </document> I've tried to remove "document", "section", "include" but no effect. Removing all gateway xml files from local filesystem does not change anything, either. Unfortunately the wiki https://wiki.freeswitch.org/wiki/Mod_xml_curl#Example:_purpose.3Dgatewaysdo… not give a proper example Thx and regards Daniel

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live-Accounts in mod_skypopen
by Daniel Neubert 16 Apr '14

16 Apr '14

We are trying to dial-out to a skype account using this dialplan <extension name="skypopen"> <condition field="destination_number" expression="^skype/(.*)$"> <action application="log" data="DESTINATION ${destination_number}"/> <action application="bridge" data="skypopen/RR/$1"/> </condition> </extension> It works with "normal" skype names, but using names like live:foobar does not work because the $(destination_number) is cropped to "skype/live". Any suggestions?

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