August 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

CHANNEL_HANGUP_COMPLETE not firing‏
by Gaurav Khanna 01 Sep '15

01 Sep '15

Hi, I am using ESL Inbound connection to get CDR information. I have added "ALL" as filter so that I get all the events which are raised by freeswitch. But for some reason "Channel_hangup_complete" event is not raised. "Channel_Hangup" and "Channel_Execute_complete" works correctly. Also tried "Plain event Channel_hangup_complete" command but that also doesn't work. Also tried linger() function but that also doesn't work. Is there any settings in config folder or in var file which I need to set. Also tried following settings in dialplan Thanks, Gaurav

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call disconnect in 60 sec
by Bharathkumar J 01 Sep '15

01 Sep '15

Hi, we are ususing FS 1.7 with sipml5 ,when make outgoing video calls through sipml5 it disconnect exactly 60 sec *Regards,* *J.Bharathkumar*

2 1

Loop play while wait DTMF digit
by Dmitry Mordovin 01 Sep '15

01 Sep '15

Hello This example play conf-pin.wav and wait DTMF. <extension name="play_and_get_digits example"> <condition field="destination_number" expression="^(1888)$"> <action application="play_and_get_digits" data="2 5 3 7000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav foobar \d+"/> <action application="log" data="CRIT ${foobar}"/> </condition> </extension> Is it possible to play WAV file in infinity loop and wait user DTMF? And how can I check DTMF input after user entered DTMF? In dialpeer, like. If ${DTMF} = 1 then bridge to XXX If ${DTMF} = 2 then play file and finish session Thank you. Dmitry

2 1

Broken silence with webrtc
by Gary Foreman 31 Aug '15

31 Aug '15

Hi all, Hoping someone can point me in the right direction because after several hours I'm out of ideas. I'm having an issue where the 2nd leg of a call that is bridged to a verto endpoint has broken silence. When there is no sound on the line the audio goes completely silent (all background noise is dropped) but around every third of a second it repeatedly cuts back in for a fraction, then goes completely silent again. This only happens when the call is created using the originate command. Can anyone give me an idea of where to look next? I have a wireshark trace that during playback shows the audio cutting out periodically on the 2nd leg during periods of silence. The 1st leg is using webrtc encryption and I cant decode the stream. Thanks in advance!

2 1

How send in To header anonymous and in P-Asserted-Identity caller-id-number
by Alex Polischuk 31 Aug '15

31 Aug '15

Hi all, How I can define different users and domains in To and P-Asserted-Identity headers? Thanks, Alex

1 0

problem with "play_and_get_digits" command (DTMF delay time)
by Manish Talwar 31 Aug '15

31 Aug '15

Hi, I have found one small problem with "play_and_get_digits" command. We have a IVR application and I am trying to get either "16 digit" card number or "0" for move to customer support from the user. Please find my dialplan as follows: 2015-08-25 18:05:37.535806 [CRIT] mod_httapi.c:1148 Debugging Return Data: <document type="xml/freeswitch-httapi"><params></params><work><execute applicati on="log" data="card number inputed: 2"/><execute application="flush_dtmf"/><exec ute application="sleep" data="2000"/><execute application="clear_digit_action" data="my_digit" /><execute application="bind_digit_action" data="my_digit,~\d{1 6}|^[0]{1},exec:transfer,dummy_transfer" /><execute action="http:/ /localhost:8080/ivr/fsentercardnumber/response/" application="play_and_get_digi ts" data="0 16 1 1000 # en-US/enter_card_number.mp3 '' Digits '' 1000 '' " /><ge tVariable name="Digits"/><execute action="http://localhost:8080/ivr/fsentercardnumber/response/" a pplication="play_and_get_digits" data="0 16 1 20000 # en-US/menu_speak_csr.mp3 ' ' Digits '' 20000 '' " /><getVariable name="Digits"/><execute application="clea r_digit_action" data="my_digit" /></work></document> It is working fine as expected if user input 16 digit number continuously without any delay. After getting full 16 digit number it will match the "bind_digit_action" and move to other dialplan. But my problem is if a user enter card number slowly then its trying to match already inputed DTMF values with matching binding and received next inputed value as different DTMF values. As a example, if I tried to input 55 and then wait for 1-2 seconds then Freeswitch tried to match 55 with binding. 2015-08-25 18:05:39.535920 [DEBUG] switch_ivr_play_say.c:1305 Codec Activated L1 6@8000hz 1 channels 20ms 2015-08-25 18:05:39.755933 [DEBUG] switch_rtp.c:5819 RTP RECV DTMF 5:2080 2015-08-25 18:05:40.015948 [DEBUG] switch_rtp.c:5819 RTP RECV DTMF 5:2080 2015-08-25 18:05:41.536035 [DEBUG] mod_dptools.c:132 sofia/internal/18188535351@ 192.168.1.112 Digit NOT match binding [55] 2015-08-25 18:05:41.536035 [DEBUG] switch_channel.c:486 RECV DTMF 5:2000 2015-08-25 18:05:41.536035 [DEBUG] switch_channel.c:582 sofia/internal/181885353 51(a)192.168.1.112 Queue dtmf digit=5 ms=250 samples=2000 2015-08-25 18:05:41.536035 [DEBUG] switch_channel.c:486 RECV DTMF 5:2000 2015-08-25 18:05:41.536035 [DEBUG] switch_channel.c:582 sofia/internal/181885353 51(a)192.168.1.112 Queue dtmf digit=5 ms=250 samples=2000 I don't want fresswtich tried to match the binding if there is any pause time while inputed the number as user can type slowly. I want to reset this binding delay time so that user can type slowly and we will get full 16 digits number for matching inputed value in certain time interval. Please suggest me how can I resolve this problem and increase the DTMF delay time. Thanks, Regards, Manish Talwar

3 5

XML RPC
by Abdul Hakeem 31 Aug '15

31 Aug '15

Hello Experts, Is there any possibility of using a different XML-RPC engine that does not involve HTTP ? The reason being, there are a few XML engines out there that avoids the HTTP overhead, TCP setup-teardown, & also uses very minimal resources Best regards, Abdul Hakeem

3 2

Python implementation of Event Socket Library doesn't work on Mac
by Carlos Fererro 30 Aug '15

30 Aug '15

All, to be more specific (if needed) - I am running on OS X 10.10.5 Yosemite. Not only it was a challenge to build the library on OS X (had to remove some flags some invalid flag(s) to avoid obscure "atom not found in symbolIndex ..." linker error) ... But!!! No client connection to the FreeSWITCH is getting established - even if client program and FreeSWITCH are running on the same machine and "host" parameter to the ESLconnection is "localhost" (which seems to be working fine on all flavors of Linux). Will be more than happy to provide any additional information is needed. Any help/advise is greatly appreciated. Thanks Carlos

1 0

Using hunt, original extension is busy while other one answered
by Yehavi Bourvine 30 Aug '15

30 Aug '15

Hi, We are using extension hunting (multiple destinations separated by | ) in order to allow the call to jump to the next extension if the first one is busy or does not answer. We are also using the limit() function in order to know whether an extension is in a call or not (people here do not like waiting calls). I also set limit_ignore_transfer=false. The problem we have is: If the destination is A|B, A does not answer and the call jumps to B and answered by him. At this state, limit shows that A has one call and B has 0 calls; That way, B cannot transfer the call back to A. Is this something wrong I am doing, an intended behavior, or a bug (for which I have to open a JIRA)? Thanks! __Yehavi:

2 4

Freeswitch dialing Via Broadband phone
by Prashant Choudhary 30 Aug '15

30 Aug '15

Hello, I have my FreeSwitch configured in my local machine. I am using a broadband connection which comes with additional phone line connected via ADSL splitter. Which can be used to make calls, The router has the details of the phone connection. Now the thing is, can I use Freeswitch configured under same router to make outbound calls. Thanks

2 1