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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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No audio when calling in via SIP phone
by Iqbal Abdullah 30 Jul '14

30 Jul '14

Hello everyone, I have just gotten to know about FreeSWITCH and have tried installing it on a EC2 instance and in the process of trying out some examples. I am facing this issue of not being able to hear any sounds during a call I make to my FreeSWITCH system. My freeswitch system is the latest stable ver.1.4 and I followed the instructions to install from http://clintberry.com/2011/getting-started-with-plivo/ via the installation script show on the site. I changed the installation script to use Slash and also 1.4 instead. One of the exercises is to pass an incoming call to plivo and make plivo return a <say> action, which for some reason is not reading back the text to me (no audio) when I call 1005 from using sflphone on my ubuntu. This is the log on freeswitch: 2014-07-29 18:01:12.581530 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/1000(a)xxxxxxxxxx.amazonaws.com [5650b3dd-1519-4c6f-b720-d4d3a8a65c5c] 2014-07-29 18:01:12.681534 [INFO] mod_dialplan_xml.c:558 Processing 1000 <1000>->1005 in context default 2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000(a)xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established. 2014-07-29 18:01:12.733800 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000(a)xxxxxxxxxx.amazonaws.com! 2014-07-29 18:01:12.733800 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/1000(a)xxxxxxxxxx.amazonaws.com] has been answered 2014-07-29 18:01:14.121516 [NOTICE] mod_dptools.c:1232 Hangup sofia/internal/1000(a)xxxxxxxxxx.amazonaws.com [CS_RESET] [NORMAL_CLEARING] 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1632 Session 1 (sofia/internal/1000(a)xxxxxxxxxx.amazonaws.com) Ended 2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1636 Close Channel sofia/internal/1000(a)xxxxxxxxxx.amazonaws.com [CS_DESTROY] I can see that plivo is correctly calling the XML I have placed, as this is the log I see on my web server: [29/Jul/2014 01:59:13] "GET /answer/?To=1005&Direction=inbound&From=1000&CallerName=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&CallStatus=ringing HTTP/1.1" 200 93 [29/Jul/2014 01:59:14] "GET /answer/?To=1005&Direction=inbound&From=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&HangupCause=NORMAL_CLEARING&CallStatus=completed HTTP/1.1" 200 93 This is the XML I have prepared to test plivo: <?xml version="1.0" encoding="UTF-8" ?> <Response> <Speak>Hello World</Speak> </Response> And this is default.xml on freeswitch which forward any calls to 1005 to plivo: <include> <context name="default">  <extension name="plivo"> <condition field="destination_number" expression="^1005"> <action application="enable_heartbeat" data="60"/> <action application="socket" data="127.0.0.1:8084 async full"/> </condition> </extension> .... .... </include> I have read that NAT issues might be causing a no-audio issue, but I am not sure and do not know how to tell if this is the issue. The only thing which look strange is the logs on freeswitch saying: 2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554 sofia/internal/1000(a)xxxxxxxxxx.amazonaws.com using scheduler due to bypass media or media is not established. but what does that mean?

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Avoiding media timeouts in remote systems
by Richard Brady 30 Jul '14

30 Jul '14

I have FS talking to a carrier, and got on-hold the carrier sends a reINVITE with a=sendonly, so FS stops sending media, but the carrier drops the call due to a media timeout. Appears their SIP and RTP stacks aren't talking to each other very well. Can I compensate for this by continuing to send RTP to keep the their MGW session alive? Regards, Richard

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fax receiving issue (two page compressed into one)
by Marlok Tamas 30 Jul '14

30 Jul '14

Hi, I have the following problem: There are a few fax number from where I all two page faxes are compressed into one page (see the attached pdf). I could not find anything about this issue so far... The log, and the received pdf (generated from the tiff - the problem is not in the tiff-pdf conversion): https://www.dropbox.com/s/vxkjygxyy5zi9d6/log.txt https://www.dropbox.com/s/49udrprgmascc0x/test1.pdf The relevant part of the receiver python script is: the_uuid = session.getVariable("uuid") the_recipient = session.getVariable("recipient") the_caller = session.getVariable("caller_id_number") the_dest = re.sub(r'^(\+?3|0)(6[0-9]*)$', r'3\2',session.getVariable("destination_number")) send_from = the_dest+'@fax.voip123.hu' consoleLog("info", " rxfax receiving from " + the_caller + " for destination " + the_dest + ", uuid is " + the_uuid + "\n") fname = incomingfaxes + "rxfax-" + the_uuid #answer the phone, save the fax file in .tiff format. session.answer() session.execute ("playback", "silence_stream://2000") # session.execute ("set","fax_enable_t38=true") session.execute ("rxfax", fname + ".tiff") Thanks in advance! Cheers, tmarlok

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