July 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to get conference call status?
by jobin dcruz 01 Aug '14

01 Aug '14

Hi All, I initiate call conference from this command "conference conference_name bgdial {originate_timeout=60}user/1000". But how to get live call status? if a call is answered,how to get call status and triggered to a url from lua script?. How to find conference member id from uuid?. Thanks -- Jobin D'cruz (LAMP Developer) Email: jobindcruz(a)gmail.com | lampdeveloper.jobindcruz(a)gmail.com Php MySql Linux Apache Ajax FFmpeg FFmpeg-php Mplayer Zend Framework Codeigniter CakePhp Wordpress Drupal Joomla Html5 Css3 jQuery YUI MooTools Smarty Firefox Extension

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Codec Negotiation
by Stephen Wilde 01 Aug '14

01 Aug '14

I have a "no audio" issue with FreeSWITCH with a specific sip supplier, no issue with all others. What I see is a strange codec negotiation: FreeSWITCH sends the SDP in the INVITE with: m=audio 22026 RTP/AVP 18 8 101 The Supplier sends its SDP in the 200 OK with: m=audio 34020 RTP/AVP 8 0 18 4 101 After that, FreeSWITCH uses the G729 and the supplier uses the PCMA so something is wrong in the codec negotiation. Who is in error FreeSWITCH or the supplier and why? Thanks in advance. Stephen

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mod_blacklist
by Terry Barnum 01 Aug '14

01 Aug '14

I started playing around this weekend with blacklisting certain telemarketers that seem to call like clockwork. I got it working using <https://confluence.freeswitch.org/display/FREESWITCH/mod_blacklist> as a starting point. For now I'm just using the console to add numbers, not a dialplan extension. If I have two numbers in the blocked list and delete one of them using the console, then save the list, it still saves both numbers to the blocked file. I have to manually edit blocked.list to remove the number. FreeSWITCH Version 1.4.6+git~20140603T193516Z~9479729340~64bit (git 9479729 2014-06-03 19:35:16Z 64bit) on a 2.3GHz i7 MacMini. From the console: freeswitch(a)mini.local> blacklist add blocked 1234567890 +OK 2014-07-28 18:09:26.556327 [INFO] mod_blacklist.c:228 Added [1234567890] to list [blocked] freeswitch(a)mini.local> blacklist check blocked 1234567890 true freeswitch(a)mini.local> blacklist del blocked 1234567890 +OK 2014-07-28 18:09:50.776324 [INFO] mod_blacklist.c:250 Removed [1234567890] from list [blocked] freeswitch(a)mini.local> blacklist check blocked 1234567890 false freeswitch(a)mini.local> blacklist save blocked 2014-07-28 18:10:11.056373 [INFO] mod_blacklist.c:280 Saving blocked to /usr/local/freeswitch/conf/blacklists/blocked.list 2014-07-28 18:10:11.056373 [DEBUG] mod_blacklist.c:288 adding 15129160420 to the dump file +OK 2014-07-28 18:10:11.056373 [DEBUG] mod_blacklist.c:288 adding 1234567890 to the dump file The last line shows fs saving the deleted number to the blocked list. Are the docs incorrect? Is there another way to do this? Thanks, -Terry

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Server having two Sangoma cards and can't handle full capacity
by Sambit Jaysingh 01 Aug '14

01 Aug '14

We are having two (Sangoma) Voice Cards, one with 8-ports and other with 16-ports. I have integrated ESL library with java in INBOUND MODE and making outbound calls. The issue what we are facing are 1. When I use all 720 channels or more than 450 channels, we see the events processed by ESL is very slow and we are not getting the events properly from ESL after placing the calls, even after so many minutes, so we have to forcefully close the connection. 2. And I see error mod_event_socket.c:375 Lost 502 events! And the java app also crashes with error ESL connection timeout. 3. And I see segmentation_fault in Freeswitch and big dump file is written and which would crash the system, because of high CPU\IO. But if we reduce the channels say, 450 we don't see any issue at all. What could be the reason for this issue. Appreciate any Help... -- Regards Sambit +91-9916211211

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How to get real-time rtp audio and video ports
by takashi higashi 31 Jul '14

31 Jul '14

Hey guys, I sent e-mail to freeswitch developer about how to get rtp packet with ffmpeg. and I asked to him below question. How can I get rtp packet with ffmpeg? I want to trancoding real time rtp packet to motion jpeg and transfer node.js. and then, he answered like this, The following works for me: ffmpeg -re -f lavfi -i aevalsrc="sin(400*2*PI*t)" -ar 8000 -f mulaw -f rtp rtp://127.0.0.1:6666 ffmpeg -i pcmu.sdp -acodec copy /tmp/a.wav or ffmpeg -i bingbing.mp4 -an -vcodec h264 -filter:v scale=1280:720 -f rtp rtp://127.0.0.1:6668 looks like pcmu is not a valid codec for mp4, so I use aac ffmpeg -i pcmu-264.sdp -acodec libfaac -vcodec copy /tmp/a.mp4 So I asked about what is 6666 ports? How can I get rtp real-time audio and video ports? but I couldn't get answer... Somebody help me, how to get real-time rtp audio and video ports? please.

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Invite SDP
by Andrew 31 Jul '14

31 Jul '14

Hi, Not sure where to turn to. Here is my issue. I am running FS and everything is good. After a few minutes FreeSWITCH seems to stop responding. Per Wireshark I see a lot of re invites. If I stop traffic then send more calls Everything seems fine. I am using windows FreeSwitch embedded on with 128 gb of ram running in a ram drive with no sql and dual intel xeon cpu e5-2690 at 10% cpu and 40% memory in use. I also have no errors in the log.

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How to get conference live call status?
by jobin dcruz 31 Jul '14

31 Jul '14

Hi All, I initiate call conference from this command "conference conference_name bgdial {originate_timeout=60}user/1000". But how to get live call status? if a call is answered,how to get call status and triggered to a url from lua script?. How to find conference member id from uuid? Thanks -- Jobin D'cruz (LAMP Developer) Email: jobindcruz(a)gmail.com | lampdeveloper.jobindcruz(a)gmail.com Php MySql Linux Apache Ajax FFmpeg FFmpeg-php Mplayer Zend Framework Codeigniter CakePhp Wordpress Drupal Joomla Html5 Css3 jQuery YUI MooTools Smarty Firefox Extension

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Upgrading FreeSWITCH v1.2.x from Debian repos
by Richard Brady 31 Jul '14

31 Jul '14

Hi Ken and team Is 1.2 still maintained? I have a few FS boxes running 1.2 installed from Debian repos. Can't upgrade to 1.4 yet but will do so as soon as I can. In the meantime is it possible to do something like apt-get install freeswitch=1.2.23 to upgrade from repos without going to 1.4? Richard

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Auto answer for inbound calls?
by Bhavesh Soni 31 Jul '14

31 Jul '14

Hey guys, Is it possible to set auto-answer true for an inbound call? I know there must be a way using sip headers, But I need full list of options to try the optimum one. -- Regards Bhavesh GS LAB (O) +91 20 4671 1256 (M) +91 86000 64681

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IVR
by Roman Gelfand 30 Jul '14

30 Jul '14

I have coded a voicexml 2.1 dtmf application. Does freeswitch have ivr gateway to host it? If yes, do I need to install anything other than base freeswitch server? Could you point me to documentation describing how to deploy ivr applications and confgure ivr gateway. Thanks in advance

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