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intercept in fifo queue
by Vladimir Elizarov 18 Mar '10

18 Mar '10

Hello. How can I intercept a fifo? I tried to use the example of intecept default dialplan, but in no fifo dialed_extension. <extension name="operators"> <condition field="destination_number" expression="^operators$"> <action application="bind_meta_app" data="2 b s record_session::/opt/freeswitch/storage/record/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="answer"/> <action application="playback" data="ivr/ivrvoice/razgovor-mozhet-byt-zapisan.wav"/> <action application="playback" data="ivr/ivr-hold_connect_call.wav"/> <action application="set" data="fifo_music=$${hold_music}"/> <action application="fifo" data="operators@$${domain} in undef"/> </condition> </extension> <fifo name="operators@$${domain}" importance="0"> <member timeout="15" simo="1" lag="20">{fifo_caller_exit_key=9,fifo_consumer_exit_key=9}user/100@$${domain}</member> </fifo> </fifos> -- Best regards, Vladimir Elizarov

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mod_fifo recording with fifo_record_template
by Steven Ward 18 Mar '10

18 Mar '10

I'm looking to use fifo_record_template to specify uniquely named recording files. Here is the situation: I'm using a fifo with on-hook members. I would like each fifo call to be recorded, with each recording-file named uniquely. Thanks to some helpful IRC talk, I have learned that the following works well: <fifo name="Outside_Call@$${domain}" importance="0"> <member timeout="20" simo="1" lag="0">{fifo_record_template=$${base_dir}/recordings/my_recording.wav,fifo_member_wait=nowait}user/35351@ $${domain}</member> </fifo> However, this records each and every fifo call to the file (in my case): /usr/local/freeswitch/recordings/my_recording.wav Every new call overwrites the file. Instead of a statically defined file name, I would like a unique file name for every fifo call. I don't know how to make reference to channel variables in this definition. I tried the following, but it results in an error: <fifo name="Outside_Call@$${domain}" importance="0"> <member timeout="20" simo="1" lag="0">{fifo_record_template=$${base_dir}/recordings/{${uuid}.wav,fifo_member_wait=nowait}user/35351@ $${domain}</member> </fifo> The error I see on the CLI when a call goes into the FIFO is this: 2010-03-18 13:51:11.289448 [CRIT] switch_channel.c:759 Invalid data (${fifo_record_template} contains a variable) It was also suggested to me that I escape the '$' character on the variable. Like this: <fifo name="Outside_Call@$${domain}" importance="0"> <member timeout="20" simo="1" lag="0">{fifo_record_template=$${base_dir}/recordings/{\${uuid}.wav,fifo_member_wait=nowait}user/35351@ $${domain}</member> </fifo> This also resulted in: 2010-03-18 13:52:37.014742 [CRIT] switch_channel.c:759 Invalid data (${fifo_record_template} contains a variable) The recording file doesn't get generated. Any recommendations on using fifo_record_template to generate recording file names that are unique to each call? Is there a way to get channel variables to be evaluated and populate the value of fifo_record_template? Or does fifo_record_template only support static values (meaning I'd have to do a record_session in dial plan?)? Thanks for any feedback.

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Calls
by Jonas Gauffin 18 Mar '10

18 Mar '10

Hello, Is it possible to list calls that are terminated in IVR's, are parked or in early state? "show calls" api just lists established calls, correct? Best regards, Jonas

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Re: [Freeswitch-users] voicemail options [temporarly solved]
by Madovsky 18 Mar '10

18 Mar '10

Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also) it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist. I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification Regards Franck ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 16, 2010 4:35 PM Subject: Re: [Freeswitch-users] voicemail options ok, I tried 50 times to leave a message, it worked only one time. It seems that sendmail needs to be free to send the email.... ----- Original Message ----- From: devel(a)thom.fr.eu.org To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 16, 2010 3:59 PM Subject: Re: [Freeswitch-users] voicemail options Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.…) and cannot fix it. I don't know if it's a priviledge problem, as I got the same results running FS with or without -u. François De : freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Anthony Minessale Envoyé : mardi 16 mars 2010 20:30 À : freeswitch-users(a)lists.freeswitch.org Objet : Re: [Freeswitch-users] voicemail options are you running FS as root or a regular user? maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary? On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos(a)madovsky.org> wrote: I noticed that sendmail makes a segfault when FS is trying to send email and sendmail has some emails locked in folder queue.... Regards Franck _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 ---------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Calculate the Call Duration ( BillSec)
by Aloysius Lloyd 18 Mar '10

18 Mar '10

Hi All, The following lua script for calculate the call duration. But the issue here , the duration is calculating from the outbound initiate time. How to get the call answer time? session:preAnswer(); -- sleep a second session:sleep(1000); -- digits = session:playAndGetDigits(10, 20, 3, 5000, "#", "enter-dest.wav", "invalid-digits.wav", "\\d+|\\*"); -- Initiate an outbound call outSession = freeswitch.Session("sofia/gateway/voipms/"..digits,session) starttime = os.time(); freeswitch.consoleLog("info", "Start Time"..starttime.." \n"); freeswitch.consoleLog("info", "Before Bridge the Call \n"); freeswitch.bridge(session, outSession); freeswitch.consoleLog("info", "After Bridge the Call \n"); endtime = os.time(); freeswitch.consoleLog("info", "End Time"..endtime.." \n"); billsec = os.difftime(endtime,starttime); freeswitch.consoleLog("info", "Blling Sec"..billsec.." \n"); -- hangup session:hangup(); Thanks Lloyd

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Newbie question on domain name
by false 18 Mar '10

18 Mar '10

Hi List i feel strange about default domain in Freswitch: i read the vars.xml and default domain name is IP address: <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/> and i register successfull the 1000, 1002 extension the i copy 1000.xml to 7000.xml then i register 7000 extension but get error Please help Ha` Registrations: ================================================================================================= Call-ID: 56dc9e359865070c(a)192.168.1.6 User: 1002(a)192.168.1.11 Contact: "user" <sip:1002@192.168.1.6:4318;user=phone> Agent: Grandstream BT110 1.0.7.3 Status: Registered(UDP)(unknown) EXP(2010-03-18 04:46:13) Host: fs IP: 192.168.1.6 Port: 4318 Auth-User: 1002 Auth-Realm: 192.168.1.11 MWI-Account: 1002(a)192.168.1.11 Call-ID: 8fa759179877592f(a)192.168.1.12 User: 1009(a)192.168.1.11 Contact: "user" <sip:1009@192.168.1.12:57386;user=phone> Agent: Grandstream BT110 1.0.7.3 Status: Registered(UDP)(unknown) EXP(2010-03-18 04:48:08) Host: fs IP: 192.168.1.12 Port: 57386 Auth-User: 1009 Auth-Realm: 192.168.1.11 MWI-Account: 1009(a)192.168.1.11 ================================================================================================= freeswitch@fs> 2010-03-18 02:52:00.391683 [WARNING] sofia_reg.c:1866 Can't find user [7000(a)192.168.1.11] You must define a domain called '192.168.1.11' in your directory and add a user with the id="7000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2010-03-18 02:52:00.423766 [WARNING] sofia_reg.c:1024 SIP auth failure (REGISTER) on sofia profile 'internal' for [7000(a)192.168.1.11] from ip 192.168.1.2 Thiết kế ngay một Pingbox độc đáo cho riêng bạn! Tạo một nơi để chat trên blog là chuyện nhỏ. http://vn.messenger.yahoo.com/pingbox/

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What is the best way implement the follow me for a extension
by Aloysius Lloyd 18 Mar '10

18 Mar '10

Hi All, I would like to setup follow me dialplan for individual extensions. Some extensions need a follow-me and some not. If the the follow-me not reach any live person the call should go to the FreeSWTICH voicemail.I have the extensions range from 201 -209. Also I am using the following Default dial plan to call between extensions. <extension name="Local_Extension"> <condition field="destination_number" expression="^(20[0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/>  <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/>  <action application="set" data="hangup_after_bridge=true"/>  <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>  <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/${dialed_extension}@ ${domain_name},${followme}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> I use the wiki samples to make a follow me dial plan for extension 201. Here is the dial plan. <extension name="201_followme"> <condition field="destination_number" expression="201"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="ignore_early_media=true"/> <action application="set" data="call_timeout=10"/> <action application="bridge" data="sofia/$${domain}/201"/> <action application="set" data="call_timeout=13"/> <action application="bridge" data="sofia/gateway/voicepulse/14161235555" /> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default $${domain} 201"/> </condition> </extension> Call to extension 201 , there are two dialplan matching the request. Either Local_Extension or follwme_201. Which order freeswtich select? Thank you for your help. Thanks, Lloyd

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Re : how to dial out through a local sip endpoint - mv372 mobile > gateway adapter (rentmycoder rentmycoder)
by Steven Brown 18 Mar '10

18 Mar '10

Hi, You can do it with the Portech registered on FS as a regular endpoint then bridge to it as below or similar <extension name="gsm"> <condition field="destination_number" expression="^9(.*)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/internal/1005^$1@${domain}"> where the Portech registers as 1005 and I prefix the number I wish to dial with 9 or alternatively just bridge directly to it without having to register <action application="bridge" data="sofia/default/$1@192.168.1.3:5060"/> Where 192.168.1.3 is the Portech In both cases you need to configure the Portech Lan to Mobile Table as below Item URL Call Num 0 192.168.1.2 # where 192.168.1.2 is my FS box Steve > Message: 5 > Date: Thu, 18 Mar 2010 13:21:09 +0100 > From: rentmycoder rentmycoder <rentmycoder(a)gmail.com> > Subject: [Freeswitch-users] how to dial out through a local sip > endpoint - mv372 mobile gateway adapter > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: > <50e456911003180521je008a2dme94fcc6901a04b99(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi guys, > Freeswitch rocks! > I've registered an Portech mv-372 mobile adapter to FS as an internal > user as userid/extension=5000. > How to dial out through this endpint? > > In asterisk I've used Dial(sip/1234567@5000) where 1234567 is the > phone no to call... > > In FS: bridge(sophia/internal/1234567@5000) > mod_dptools: normal_temporery_failure > > any ideas how to do the bridge? > or should I setup the adapter as a gateway profile? > > Just to make sure the adapter is ok i've tested it using asterisk and > it worked and was set up like this: > http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk > > > > ------------------------------ > > Message: 6 > Date: Thu, 18 Mar 2010 08:22:10 -0400 > From: "Todd" <phunk0000(a)hotmail.com> > Subject: [Freeswitch-users] FS Best practices > To: "FreeSWITCH-users@lists. freeswitch. org " > <FreeSWITCH-users(a)lists.freeswitch.org> > Message-ID: <BLU110-DS11B84F16B12464595A3B07A72B0(a)phx.gbl> > Content-Type: text/plain; charset="us-ascii" > > Hey List, still kinda new to this whole deal, looking for some advice. We > are building a website that needs to interact with FS. Not looking for > specific instructions just opinions about the best practices for doing > things like click to dial from a web page, calling CDR records to web > applications to display customers stats and general web integration. Does > this all need to be done using scripting languages like Lua or Perl, or can > you directly integrate FS with a website in some other manner. Again, just > getting into all this, but having a blast learning FS, config and power > rock! Anyhow, some places or things I should start looking into would be > greatly appreciated. Thanks List! > >

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how to dial out through a local sip endpoint - mv372 mobile gateway adapter
by rentmycoder rentmycoder 18 Mar '10

18 Mar '10

Hi guys, Freeswitch rocks! I've registered an Portech mv-372 mobile adapter to FS as an internal user as userid/extension=5000. How to dial out through this endpint? In asterisk I've used Dial(sip/1234567@5000) where 1234567 is the phone no to call... In FS: bridge(sophia/internal/1234567@5000) mod_dptools: normal_temporery_failure any ideas how to do the bridge? or should I setup the adapter as a gateway profile? Just to make sure the adapter is ok i've tested it using asterisk and it worked and was set up like this: http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk

1 0

Fscomm unable to detect devices
by Rob Dyck 18 Mar '10

18 Mar '10

16987 on Fedora I realize fscomm is an immature project. I get an error message when it tries to detect my devices ( pci sound card, usb handset, usb camera with microphone ). Is this the expected result? Detection works in Twinkle and Ekiga. Second question - is video support on the roadmap?

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