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play_and_get_digits and macros
by Madovsky 04 Feb '11

04 Feb '11

on speech managment wiki page there's explanation of how to write XML for a macro, but where to put this XML code ? I'd like to use a macro in play_and_get_digits, possible ? if yes what's the syntax ? Thanks in advance

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mod_fsk not detecting call waiting caller id
by Anirudha Shimpi 04 Feb '11

04 Feb '11

I have an analog line connected through an SPA3102 forwarded to an extension on FS. Line has call waiting ID on it (verified). I get the tone when call waiting is sent. A sample dialplan is as follows <action application="answer"/> <action application="sleep" data="1000"/> <action application="fsk_recv"/> <action application="set" data="execute_on_fsk=log 'Got FSK [${fsk_phone_num}] [${fsk_phone_name}]'"/> <action application="transfer" data="1002 XML default"/> When call waiting tone is detected I get the following 2011-02-03 17:25:06.089086 [DEBUG] mod_fsk.c:276 sofia/internal/xxxxx@xxxxx processing execute_on_fsk [log 'Got FSK [] []'] EXECUTE sofia/internal/xxxx@xxxx log('Got FSK [] []') 2011-02-03 17:25:06.089086 [DEBUG] mod_dptools.c:1183 FSK [] []' 2011-02-03 17:25:06.269078 [DEBUG] switch_core_media_bug.c:467 Removing BUG from sofia/internal/xxxxx@xxxx If I do a uuid_dump, the fsk_ (except variable_execute_on_fsk) variables are not present. Any idea what I could be doing wrong here? Never used mod_fsk before, however, need to just log call waiting name and number. Any help would be greatly appreciated.

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Portaudio Improvements
by Mitch Capper 04 Feb '11

04 Feb '11

I have submitted a ticket with a patch for portaudio and my improvements. This includes the improvements I had discussed previously during my call for input on the mailing list and is available at: http://jira.freeswitch.org/browse/FS-3006 It was only tested in windows, however most of the changes should not effect default behavior (but please test if you can). To test you will want to enable some things in the portaudio config including: <param name="live-stream-switch" value="true" /> <param name="no-auto-resume-call" value="true" /> <param name="no-ring-during-call" value="true" /> you will then be able to use the new features fully. Many of these changes were made to add better support for softphone's using freeswitch, I will be releasing my embedded freeswitch phone (FSClient) for the conference call in a week and a half it is .net/c# but WPF so sadly windows only currently. jlink and drk are helping so we should end up with a nice installer also for it. If you would like to test it please let me know. ~Mitch

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voice quality in conference
by Madovsky 04 Feb '11

04 Feb '11

is the audio of outbound call can be normalized when it enters in conference ? it seems that some phones with mic wit low sensitivy level have the start of sentence cut or hashed, some "ssss" and low level voice are not heard in conference. if I make a normal bridge with the same phones the problem not appears.. Thanks

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Enterprise bridge not continuing?
by Tim St. Pierre 04 Feb '11

04 Feb '11

Hello, I'm trying to implement a ring group scenario that supports multiple registrations per user. I have been able to get it to work (mostly) by using the enterprise bridge syntax, but when I do, I can't continue on in the dialplan to go to voicemail, for example. If I change the syntax and use the parallell dial comma, voice mail works fine, but only the first registration of each user entry will ring, and I would really like them all to ring. From what I can tell, dialplan execution stops after the originate, rather than continuing on to process the next steps. Is there some sort of variable I have to set to tell it to continue when using enterprise bridge? Here's the XML debug output of the dialplan I'm sending from xml_curl <?xml version="1.0"?> <document type="freeswitch/xml"> <section name="dialplan"> <context name="internal"> <extension name="Intercom" description="Local Extension" continue="true"> <condition field="destination_number" expression="5240"> <action application="pre_answer" data=""/> <action application="set" data="continue=true"/> <action application="set" data="cdr_destination=Internal"/> <action application="set" data="ringback=$${us-ring}"/> <action application="set" data="sip_callee_id_name=Lab 0"/> <action application="set" data="sip_callee_id_number=5240"/> <action application="set" data="progress_timeout=4"/> <action application="set" data="call_timeout=12"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="cdr_amaflags=INTERNAL"/> <action application="set" data="originate_continue_on_timeout=true"/> <action application="set" data="originate_disposition=NO_ANSWER"/> <action application="set" data="ignore_early_media=true"/> <action application="log" data="INFO Adding user 5242to the group"/> <action application="log" data="INFO Adding user 5244to the group"/> <action application="bridge" data="[call_timeout=12,alert_info=${domain};info=alert-internal]user/5240@communicatefreely.net:_:[ignore_early_media=true,call_timeout=12,origination_caller_id_name=Cordles-${caller_id_name},origination_caller_id_number=${caller_id_number},export_vars=accountcode subdomain,alert_info=${domain};info=alert-group]user/5242@${domain}:_:[ignore_early_media=true,call_timeout=12,origination_caller_id_name=Cordles-${caller_id_name},origination_caller_id_number=${caller_id_number},export_vars=accountcode subdomain,alert_info=${domain};info=alert-group]user/5244@${domain}"/> </condition> </extension> <extension name="Voicemail" description="Leave VM message" continue="true"> <condition field="destination_number" expression="5240"> <action application="set" data="playback_terminators=#0"/> <action application="set" data="cdr_destination=VM Box 5240"/> <action application="answer" data=""/> <action application="set" data="vm_busy_greet=0"/> <action application="set" data="vm_unavail_greet=0"/> <action application="set" data="vm_direct__greet=0"/> <action application="set" data="vm_ivr_enable=false"/> <action application="lua" data="vm_leave.lua 5240 lab new"/> </condition> </extension> </context> </section> </document>

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Session sleep method
by Malay Thakershi 03 Feb '11

03 Feb '11

Hello, I am using .NET module. There is sleep method that has two arguments. What is second argument for? I use it: mObjMainSession.sleep(200, 0); What happens if I pass something other than zero as second argument? I am having inconsistent delays between call paths and that is why I am exploring this argument. Thank you for any help. Malay

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conference audio in last git
by Madovsky 03 Feb '11

03 Feb '11

seems that the audio (audio files and voice) is less stable as before in conference only. some packet loss or like a noise gate with a too high threeshold also without CNG there's noise between ivr voice

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How to record a conference using javascript application
by kapil.rastogi 03 Feb '11

03 Feb '11

Hi, I want to record the chatting b/w all members during the conference. I am using the following statement in my javascript application: session.execute("conference", "conf-room--"+room_id+"@default record /usr/local/freeswitch/abc.wav"); But i am unable to record the conference room chatting. But when i record the same from fs_cli command line, it is working well. Please tell me how to record the conference chatting. Regards, ----- Regards, Kapil Rastogi Telemune Software Solutions P Ltd. kapil.rastogi(a)telemune.net +919013204760 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-record-a-conference-us… Sent from the freeswitch-users mailing list archive at Nabble.com.

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ANNOUNCEMENT CudaTel IP PBX (powered by FreeSWITCH) Reaches Critical Mass
by Anthony Minessale 03 Feb '11

03 Feb '11

I have been splitting my time developing not only FreeSWITCH but the CudaTel Phone system by Barracuda Networks that uses FreeSWITCH as the telephony engine. I'm proud to announce that not only has the product reached a 2.0 status, it's now 100% deployed within Barracuda Networks running the entire company's phone services across multiple locations. The CudaTel is using FreeSWITCH right from GIT with no special modifications at all. The best part is it works great as a PBX you can then feed out to your existing FreeSWITCH setup to route the traffic to custom applications or whatever you can think of. The next step is to develop more exciting and impressive PBX features that can be added for free with an energize update package available at the time of purchase. CudaTel comes with the same great support you receive today on FreeSWITCH. This is also a great opportunity for those of your out there looking to get into the reseller market. Contact me at the consulting link at the top of this page or call 408-588-3633 to learn how you can become a reseller of CudaTel. Use your existing VoIP expertise to support and deploy CudaTel to others. Check it out at http://www.cudatel.com -- 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 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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mod_perl mod_lua
by deniro 03 Feb '11

03 Feb '11

Hi I have compiled version of freeswitch 1.6.x on ubuntu 10.04 I want to install extra freeswitch modules like mod_perl and mod_lua Is there any way to install these modules without re-compileling freeswitch. I looked at freeswitch site and it is talking about editing modules.conf file and compiling from source and I dont even have modules.conf file as I searched. thx deniro--

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