December 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Callcenter application codec negotiation problem
by Jimmy Chang 24 Dec '13

24 Dec '13

Hi, I have two clients using different codecs. Client A: PCMA, H264 Client B: G729, H264 Agent: PCMA,G729,H264 FS:PCMA,G729,H264 Now agent could only accept calls of client A but not from client B. When I change the codec order of FS to G729,PCMA,H264, then agent could accept client B but not from client A. I have tried these settings below and all in vain. 1) add <action application="set" data="inherit_codec=true"/> in diaplan (inbound-late-negotiation is true) 2) add <action application="export" data="nolocal:absolute_codec_string=G729,H264"/> in diaplan B. 3) add {absolute_codec_string='G729,H264'} in callcenter application, syntax error Here is the diaplan. <extension name="Test Call Center"> <condition field="destination_number" expression="^555$"> <action application="sleep" data="1000"/> <action application="callcenter" data="555@default"/> </condition> </extension> How can I configure to accept both clients? Any advice would be appreciated. Jimmy --- 由於 avast! 防毒軟體已開啟防護，所以此封電子郵件將不受病毒及惡意軟體的侵害。 http://www.avast.com

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Freeswitch MSI
by Moishe Grunstein 24 Dec '13

24 Dec '13

Looks like there may be an issue with the Windows MSI packages. I just downloaded and installed the 64bit package, the version info is showing FreeSWITCH Version 1.5.7b+git~20131111T224713Z~5456daa010~32bit (git 5456daa 201 3-11-11 22:47:13Z 32bit) Should I open a Jira? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support(a)nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg@01C72F94.9EE45D60]<http://www.nysolutions.com/> Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

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Build ubuntu package
by pablo platt 24 Dec '13

24 Dec '13

Hi, I'm trying to build ubuntu deb package for ubuntu 12.04 and above. I've followed the script in the wiki which is supposed to work for ubuntu 12.04 http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start Unable to locate package freeswitch-meta-vanilla I've tried the debscrbuilder script but it builds a package for debian sid https://github.com/FreeSWITCH/FreeSWITCH/blob/master/scripts/ci/debsrcbuild… Is it possible to fix the script in the wiki or to make the debsrcbuilder script work for ubuntu? Thanks

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MOH crashes freeswitch, incorrect music file directory on Debian Wheezy
by Francis 24 Dec '13

24 Dec '13

Hi all, On a fresh vanilla install from the Debain repositories onto Wheezy (# aptitude install freeswitch-meta-vanilla), MOH fails with the following messages: EXECUTE sofia/internal/1000(a)domain.net playback(local_stream://moh) 2012-12-23 00:16:30.502576 [WARNING] mod_local_stream.c:470 Unknown source moh, trying 'default' 2012-12-23 00:16:30.502576 [ERR] mod_local_stream.c:479 Unknown source default The call is placed on hold, but upon retrieval, freeswitch crashes and needs restarting. Sound files are installed in /usr/share/freeswitch/sounds, and querying fs_cli shows the same directory. mod_local_stream is enabled in modules.conf.xml However- in local_stream.conf.xml, all paths are set as path="$${sounds_dir}/music/{bitrate} but the actual path is /music/default/{bitrate} Copying the soundfiles to the correct directory gets MOH working, but freeswitch still crashes when attempting to retrieve the call.

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SIP 480 on over limit
by Ahmed Sboor 23 Dec '13

23 Dec '13

HI all, when we limit calls on outbound gateway , FS send 480 to A Leg. Which kills client call while client accepts 503 or congestion message if there is a limit on channels . Is it a way to map to release 503 on over limit calls . any help will be highly appreciated. regards Ahmed

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FreeSWITCH-users@lists.freeswitch.org
by Vincent Xia 23 Dec '13

23 Dec '13

Hi Guys, the fs console commmand: originate user/1001 &echo() works fine with chrome client but no echo with doubango webrtc clients as boghe/imdroid which has no DTLS HANDSHAKE after call answered, after capturing DTLS packets with wireshark and i found that the fs recieved "Client Hello" from boghe/imdroid and sent back "Server Hello", but to the wrong address which is the external IP of the client PC, actually both the FS server and the client PC are in the same LAN, so it doesn't make sense sending to an external IP, the fs log says it didn't receive PTCP packet so will use the same IP as RTP (but the media stream is not established, where did FS get RTP). so is it the problem of the client side or server and how to fix? And after all, the most important issue now: Merry Christmas and Happy New Year!

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Re: [Freeswitch-users] Add new headers with APIs
by Sayyed Mohammad Emami Razavi 23 Dec '13

23 Dec '13

Thanks Yossi and Alex, problem was resolved with your useful guides. For other ways in setting sip headers, i will ask other questions, Thanks again, #!/usr/bin/lua require("ESL") command = 'originate {sip_h_X-varA=111,sip_h_X-varB=222,ignore_early_media=true,originate_timeout=60}user/ 1000@192.168.1.5&playback(/usr/local/freeswitch/scripts/ZamanakSoundLogo.wav)'; ... bash# ./makecall.lua and tcpdump shows to me: .... 11:57:55.204690 IP 192.168.1.5.sip > 192.168.1.2.53194: SIP, length: 1126 ......."_<.'..E...Xv..@................n.4INVITE sip:1000@192.168.1.2:53194;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5;rport;branch=z9hG4bKpHZgXQtQgHQKK Max-Forwards: 70 From: "" <sip:0000000000@192.168.1.5>;tag=e2yg1t6gc5t4N To: <sip:1000@192.168.1.2:53194;ob> Call-ID: f67f2b26-e64e-1231-348c-002170ca8788 CSeq: 53557573 INVITE Contact: <sip:mod_sofia@192.168.1.5:5060> User-Agent: FreeSWITCH-mod_sofia/1.2.10 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 205 X-varA: 111 X-varB: 222 X-FS-Support: update_display,send_info Remote-Party-ID: <sip:0000000000@192.168.1.5 >;party=calling;screen=yes;privacy=off ....

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Re: [Freeswitch-users] Add new headers with APIs
by Sayyed Mohammad Emami Razavi 23 Dec '13

23 Dec '13

Hello Alex, I used your guide in my lua script but in tcpdump and fs_cli, i can find no new headers set. This is my lua code: #!/usr/bin/lua require("ESL") local command = 'originate {ignore_early_media=true,originate_timeout=60}user/1000@192.168.1.5&playback(/usr/local/freeswitch/scripts/HelloWorld.wav)'; local args = table.concat({}, " "); local con = ESL.ESLconnection("localhost", "8021", "ClueCon"); con:execute("set", "sip_h_X-Token=abc123"); con:execute("set", "X-Token=abc123"); con:execute("set", "<![CDATA[sip_h_P-Charge-Info=<sip:${caller_id_number}@ ${domain_name}>;npi=0;noa=3]]>"); local e = con:bgapi(command, args); print(e:getBody()); Can you help me more?!

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About Freeswitch capabilities for video (calls, recording, etc...)
by Nick Techa 23 Dec '13

23 Dec '13

Hi, I'm trying to evaluate Freeswitch capabilities and determine if we can use it as an alternative to our current video calls platform (video calls, recording, live calls lawful tapping). The goal is support : 1)-audio/video calls recording support 2)-audio/video calls playback support 3)-audio/video calls lawful tapping (view/listen live calls) Video specific coverage in general seems very limited in books and docs. Assuming we'll use webrtc/sip and/or plain SIP, here is my question: Is Freeswitch currently able to do all the listed items? Any advice / suggestion or links for guides about how (or what might be needed) to support these items will be very helpul. Thanks.

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Dial plan public- HELP
by Andrew 22 Dec '13

22 Dec '13

Hi, I am building a standalone program where I will take an incoming call and do a lookup and then bridge the incoming call with an outbound call. I create a new session using the UUID that the in inbound call then I call the bridge code. What happens is as soon as I call the bridge call the inbound call disconnects. I know I need to edit the dial plan but not sure what to update it to. The other way I tried to do this is pass the bridge info via dial plan variable but the variable hits faster than I can set it. Is there a wait for variable I can put in the dial plan? Andrew

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