April 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Stream audio file/live to multiple SIP endpoints with IP multicast
by Sluschny, Thomas 30 Aug '10

30 Aug '10

Hi, i want to stream audio (from file and live) from FreeSwitch to multiple SIP endpoints (these are proprietary). Because of bandwith limitations we have to use IP multicast for audio data. Now the question: how can i do this smartly in FreeSwitch? I have read some Wiki entries: - mod_esf: seams to use multicast, but i couldn't find usage information - mod_shout: streaming, but from internet ? MP3, but i want wav-files to stream - mod_local_stream: seams to play files in a loop, i want to play a whole file once. - mod_conference: seams not to support IP multicast Im searching for a simple way to send a SIP INVITE to all my endpoints and then stream audio (live from soundcard or from file) over IP multicast to all these endpoints. Thanks in advance, Thomas

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[contrib] a new java ESL inbound/outbound client
by david varnes 30 Aug '10

30 Aug '10

Hi all, I had a basic java inbound ESL client kicking around that I have used in a couple of small projects over the last year. I needed something a little more complete for a new project so I dusted it off and made it less incomplete. It still needs more work and testing, but it certainly is usable right now. I have tested it against FS 1.0.4 and latest trunk. I have put it in my contrib area in svn in hopes that some may find it useful: http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/dvarnes/java/esl-cli… I would be interested in any feedback ... Features * Apache License (ASL) version 2 * Standalone Inbound client * Framework classes to easily create an Outbound socket client * based on Netty [1] nio library version 3.1.5.GA (previously was using Apache MINA, but this is easier) * logging via slf4j * only dependencies are slf4j-api and netty (both Apache licensed) * single jar which is a valid OSGi bundle * built using maven * eclipse projects * reasonable level of java docs Still todo * Docs * Simple example apps * .. more in TODO.txt in project root. There is no binary jar available right now since I don't know how/if I can put files up to file.freeswitch.org. In the meantime to build you need maven [2] installed. If you are unfamiliar with maven usage, I can post a simple howto. davidv [1] http://www.jboss.org/netty/downloads.html [2] http://maven.apache.org

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Diversion header
by Jonas Gauffin 23 Jun '10

23 Jun '10

Hello, I got a incoming call from my provider which goes to a javascript. The javascript transfers the call to an external destination (goes back through my provider). A diversion headers is added by FreeSWITCH on the outbound call. How can I remove it (or if not possible, how do I change it)? javascript: session.setCallerData("caller_id_number", aNumber); session.setVariable("gate_caller_id_number", aNumber); session.setVariable("gate_site_id", SITE_ID); session.execute("transfer", this.dtmfBuffer + " XML internal"); send 1179 bytes to udp/[130.244.Y.YY]:5060 at 16:13:16.752200: ------------------------------------------------------------------------ INVITE sip:024390510@sip-corporate.provider.com<sip%3A024390510(a)sip-corporate.provider.com>SIP/2.0 Via: SIP/2.0/UDP 212.247.XX.XX;rport;branch=z9hG4bKaa59XjFt2N2rS Max-Forwards: 67 From: "0236661XXX" <sip:0243795362@212.247.XX.XX>;tag=BHvytKtv47r2a To: <sip:024390510@sip-corporate.provider.com<sip%3A024390510(a)sip-corporate.provider.com> > Call-ID: 5a022f8f-a0b9-122d-779a-dbdeced2caa5 CSeq: 127654862 INVITE Contact: <sip:gw+dalaconnecttele2@212.247.XX.XX :5060;transport=udp;gw=dalaconnecttele2> User-Agent: Gateon Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 273 Diversion: 0 <sip:0@212.151.Y.Y ;user=phone>;reason=unconditional;counter=1;privacy=off X-FS-Support: update_display Remote-Party-ID: "0236661XXX" <sip:0243795362@212.247.XX.XX >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1267513684 1267513685 IN IP4 212.247.XX.XX s=FreeSWITCH c=IN IP4 212.247.XX.XX t=0 0 m=audio 32712 RTP/AVP 8 3 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20

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Lua script hangup detect
by Chia-Yen Wu 10 Jun '10

10 Jun '10

Hello, i tried to use Lua script to replace xml macro in dialplan, but I found out that Lua wont terminate if client hangup, ,so the session is still on but client is already hangup, is there a way to avoid this ?

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Direct inward dialling
by Peter P GMX 07 Jun '10

07 Jun '10

Dear all, when I get an incoming call from my SIP provider I do receive an invite on the trunk number, e.g. in Germany 0xxxxx9. However I have an extension block of 0xxxxx90 to 0xxxxx999. In the dialplan I checked the condition field="destination_number" but this compares to the number of the trunk 0xxxxx9, so I do not get the direct inward dialling extensions. The SIP to-header contains the direct inward dialling number 0xxxxx90 to 0xxxxx999 but I have no success in accessing it in the dialplan. I tried sip_to_user, sip_to_uri, sip_req_user, sip_req_uri, but no success. They are all empty. Any idea which variable to parse? Best regards Peter

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xml_cdr http post differentiation between legs
by Adam Ford 03 Jun '10

03 Jun '10

Is there any way to tell whether it is the a-leg or b-leg when receiving the POST data from mod_xml_cdr? I wrote a parser in Python which parses the a_*.cdr.xml log files, and only grabs the info in the associated b-leg log files if it needs additional information(for example - in the case of a call that ends with mod_fifo). I want to adapt this to using the POST features of mod_xml_cdr so that it can be real time. Any suggestions are greatly appreciated, -Adam

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Simple IAX2 setup - help with converting from asterisk to freeswitch
by Matthew Law 26 May '10

26 May '10

Hi, I have compiled freeswitch trunk from svn on debian. I am trying to convert the following simple working asterisk config to freeswitch and I would be really grateful if someone could point me in the right direction: pbx:/etc/asterisk# cat iax.conf [general] bindport=4569 bindaddr=my.ip.address jitterbuffer=yes disallow=all allow=ulaw allow=alaw context=deadend [voiptalk] type=peer username=XXXXXX secret=XXXXXX host=iax.voiptalk.org [0843XXXXXX] type=friend username=08433XXXXXX context=incoming requirecalltoken=auto [1000] type=friend host=dynamic mailbox=1000 secret=XXXXXX context=phones requirecalltoken=auto pbx:/etc/asterisk# cat extensions.conf [general] autofallthrough=yes [outgoing] exten => _0[1-9].,1,Dial(IAX2/XXXXXX@voiptalk/44${EXTEN:1}) exten => _00.,1,Dial(IAX2/XXXXXX@voiptalk/${EXTEN:2}) [internal] exten => 901,1,VoiceMailMain() exten => 901,2,Hangup() exten => 902,1,MeetMe(1234,cdM) exten => 902,2,Hangup() [incoming] exten => XXXXXXXX,1,Dial(IAX2/1000,30) exten => XXXXXXXX,2,VoiceMail(1000@internal) [phones] include => internal include => outgoing pbx:/etc/asterisk# cat voicemail.conf [internal] 1000 => XXXX,Matt,me(a)myomain.com format=wav49 maxsilence=0 Sorry to be a bonehead, but I'm struggling with the wiki docs (especially with regard to IAX2) and also concerned about security. I installed the samples when I compiled freeswitch but wondering if that is a security risk as the PBX box is on the public internet? Many thanks, Matt

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Memory leak in mod_nibblebill or in ODBC core?
by Sergey Okhapkin 12 May '10

12 May '10

I want to utilize nibblebill features for prepaid support, but as soon as I enable nibble billing with set nibble_rate=XXX set nibble_account=NNNNNN set enable_heartbeat_events=60 in dialplan, FS process begins to eat more and more memory, RSS grows from 23M to 100M and more after processing few thousands calls and continues to grow. If I comment out those 3 lines from dialplan, then FS RSS grows from initial 23M to 50-60M (depending on the number of concurrent calls) and stays at this value, no memory leaks. Do anybody use mod_nibblebill? I'm experiencing this problem with all svn versions I tried including latest git version.

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2 B Channel Transfer (2BCT) and FreeSWITCH
by Doug Albrechtsen 07 May '10

07 May '10

Is anyone having success using 2B Channel Transfer with FreeSWITCH and Sangoma? If so, what are the driver considerations and what does it take to make the transfer work? Thanks, Doug

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mod_com_g729 DECODER CREATE FAILED
by Peter P GMX 05 May '10

05 May '10

Hello, I tried mod_com_g720 with 2 licenses and ran into a problem: What DOES work with G729: * Calling Mailbox (Phone is Snom 360) * Calling external Numbers through Patton gatway What does NOT work * Calling another Snom Phone with G.729 enabled (both Phones use TLS/SRTP) Here an exempt from the log calling phone is 200, called phone is Snom 320: 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:3632 Audio Codec Compare [G729:18:8000:20]/[G729:18:8000:20] 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:2293 Already using G729 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:3565 Set 2833 dtmf send payload to 101 2010-04-28 16:47:57.022827 [DEBUG] sofia.c:4619 Processing updated SDP 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:2580 Audio params are unchanged for sofia/internal/sip:208@192.168.178.126:5060. 2010-04-28 16:47:57.029474 [INFO] mod_com_g729.c:146 DECODER CREATE - 0x9a09510 0x8e774c8 2010-04-28 16:47:57.128943 [ERR] mod_com_g729.c:142 DECODER CREATE FAILED - 0x8e9f9a8 (nil) 2010-04-28 16:47:57.128943 [ERR] switch_core_io.c:327 Codec G.729 decoder error! 2010-04-28 16:47:57.128943 [DEBUG] switch_ivr_bridge.c:478 sofia/internal/200(a)fs00.telefaks.biz ending bridge by request from read function g729_status Permitted G.729AB channels: 2 Encoders in use: 0 Decoders in use: 0 Here is the dialplan <extension name="Local_Extension_no_Voicemail">. <condition field="destination_number" expression="^\S*$" break="never">. <action application="set" data="transfer_ringback=${us-ring}"/>. <action application="set" data="dialed_ext=208"/>. <action application="set" data="effective_caller_id_number=200"/>. <action application="set" data="effective_caller_id_name=Peter "/>. <action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/>. <action application="db" data="insert/last_dial_ext/${dialed_ext}/${uuid}"/>. <action application="set" data="dialed_ext=208"/>. <action application="export" data="dialed_ext=208"/>. <action application="ring_ready" />. <action application="bridge" data="user/208(a)my.domain"/> </condition>. </extension> Just tested it with an Aastra Phone without TLS: Same Behaviour. Anybody has a clue how to solve this? Best regards Peter

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