June 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How create customize CDR in free-Switch .
by Abdullah 21 Jun '13

21 Jun '13

hi all , Please help me for creating CDR in Free-switch SIP Server, i don't want to use XML-CDR or any other build in Module . any idea , thanks for sharing with me your experience . -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-create-customize-CDR-in-f… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Better SBC
by Pepo 21 Jun '13

21 Jun '13

Hi friends, I use a few boxes with FS and now my company (TELCO) will acquire an SBC: What SBC think is best ACME or SONUS?

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A plea from your lead developer: Help my kid play in a Major League Ballpark!
by Anthony Minessale 21 Jun '13

21 Jun '13

Hello, My son is an aspiring baseball player on a select team here in Wisconsin. His team, The Wisconsin Wildcats, has a really special chance to get to play a game inside Miller Park. This is the Major League park where the Milwaukee Brewers play and not very easy for a 13yr old to make it to. The team has to sell as many tickets as possible to 2 games happening in April and May to get the opportunity to play. Everyone on the team is trying hard to sell the tickets and so am I. One problem is most of the people I know live far away =D So, if you do live anywhere near the Milwaukee area and like baseball, the games are: Fri Apr 19th 7:10pm ($40 or $50) V Chicago Cubs. Fri May 3rd 7:10pm ($20 or $25) V St Louis Cardinals. I will include a FREE copy of FreeSWITCH with any ticket purchase or donation! If you live close enough to attend one of these games or will be in the area, email me offline and i can get you the other details. If you live far away and still want to help, send paypal donation to brewers(a)freeswitch.org or to the one on our site with some mention of BASEBALL FUNDRAISER and I'll use the money to buy tickets on your behalf and give them to worthy local baseball fans. Here's a unique chance to thank my son for sharing his dad's time with all of you out there using FreeSWITCH! There is not much time to get all the tickets sold so if you can help, act now! -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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ICTFax Version 2.1.1 released
by Tahir Almas 21 Jun '13

21 Jun '13

ICTFAX v2.1.1 http://www.ictfax.org is released with some minor bug fixes. Following are the updates in the new version. 1. Updated ambiguous create fax descriptions 2. New theme and GUI 3. Bug fix for permission issues in Fax Account News: Now ICTFAX source code is available at github.com. Following is the link to git repository. https://github.com/ictinnovations/ictfax *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT

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Problem connect FreeSWITCH to AudioCodes business Plus via SIP trunk
by Em Nguyễn 21 Jun '13

21 Jun '13

Dear all, My partner company are using an AudioCodes PBX system. But now they are expanding their branch, they want to set up FreeSWITCH PBX in new branch. The requirement is the new FS PBX server can communicate all extension of the current AudioCodes PBX system via internal SIP trunk. I have set up the sip trunk, the result is that two system can call together. But If I make call from FS's extension to AudioCodes's extension, the call is connected and we can talk together, but it will be auto disconect after 10 seconds from AudioCodes side. The phone status at FS side is still connected Otherwise If I make call from AudioCodes's extension to FS's extension, It run well, no time limit. I have checked the console log, it show that the call is still connected at FreeSWICH side, only AudioCodes side is disconnected. If any one understand the problem please help me. Thanks you very much! Em Nguyen

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audio glitches/clicks when using jitterbuffer with GSM
by Christian Tari 21 Jun '13

21 Jun '13

Hi, Im having problems with jitterbuffer when transcoding from GSM to PCMa. On the incoming (GSM) side we have mobile devices using 3G network and generating jitter and packet loss. If a packet arrives late or lost, it gets replaced by a high pitch noise on the output (PCMa) stream. I've tested with other codecs (pcmu,opus) on A leg which worked fine. Settings in dialplan: <action application="set" data="rtp_jitter_buffer_during_bridge=true"/> <action application="set" data="jitterbuffer_msec=200:200:160"/> Log,trace,audio are in ticket #5511<http://jira.freeswitch.org/browse/FS-5511> Could you give me some hints how to make it work? Thanks in advance, Christian

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Weird behavior 3cx with destination 5000 for ivr demo
by Sayyed Mohammad Emami Razavi 20 Jun '13

20 Jun '13

When i call destination 5000 with my registered phone via 3cx for ivr demo, my session hangs up(incompatible destination) suddenly and no sound heard from phone! anybody knows the solution? FS log is here: http://pastebin.freeswitch.org/21084 all configs is default FS configurations. i have just changed these lines in vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>

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Freeswitch and session updates
by Emrah 20 Jun '13

20 Jun '13

Hi all, None of my clients are able to update an audio session with a reinvite to include video. Is there a setting I could use to allow such updates? In other words: If I start an audio call, I cannot add video to it. I have to start a video call in order to have video support during the session. Thanks for any pointers!

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unable to record to .ogg files
by Vincent Xia 20 Jun '13

20 Jun '13

recording to .wav is ok but to .ogg, FS reports: 2013-06-19 20:24:34.993063 [ERR] switch_core_file.c:150 Invalid file format [ogg] for [D:/recordings/1006_11500_20130619202434.ogg]! 2013-06-19 20:24:34.993063 [ERR] switch_ivr_async.c:2004 Error opening D:/recordings/1006_11500_20130619202434.ogg is there any additional mod required for .ogg, just like mod_shout is needed for .mp3?

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Passing custom SIP header through originate command
by Limit 20 Jun '13

20 Jun '13

Hi! I'am trying to create a fake call to extension passing some additional information in custom SIP header "User-to-User". Command syntax: originate {sip_h_X-User-to-User=48656c6c6f;encoding=hex}sofia/internal/1002(a)10.10.104.125 &playback('test.wav') XML test Call originated but according to sofia siptrace my custom header doesn't transmit to 1002 endpoint. SIP trace looks as follows: send 1298 bytes to udp/[10.10.104.125]:5060 at 08:12:48.524061: ------------------------------------------------------------------------ INVITE sip:1002@10.10.104.125 SIP/2.0 Via: SIP/2.0/UDP 10.10.104.125;rport;branch=z9hG4bKU0j6a2D3Zg02g Max-Forwards: 70 From: "" <sip:0000000000@10.10.104.125>;tag=Sa1rSF0gKeHmF To: <sip:1002@10.10.104.125> Call-ID: 093d3725-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE Contact: <sip:mod_sofia@10.10.104.125:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summar y, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 373 X-User-to-User: 48656c6c6f;encoding=hex X-FS-Support: update_display,send_info Remote-Party-ID: <sip:0000000000@10.10.104.125>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1371708570 1371708571 IN IP4 10.10.104.125 s=FreeSWITCH c=IN IP4 10.10.104.125 t=0 0 m=audio 7398 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 7198 RTP/AVP 98 99 34 31 100 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 a=rtpmap:100 H263-2000/90000 ------------------------------------------------------------------------ recv 1298 bytes from udp/[10.10.104.125]:5060 at 08:12:48.525061: ------------------------------------------------------------------------ INVITE sip:1002@10.10.104.125 SIP/2.0 Via: SIP/2.0/UDP 10.10.104.125;rport;branch=z9hG4bKU0j6a2D3Zg02g Max-Forwards: 70 From: "" <sip:0000000000@10.10.104.125>;tag=Sa1rSF0gKeHmF To: <sip:1002@10.10.104.125> Call-ID: 093d3725-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE Contact: <sip:mod_sofia@10.10.104.125:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summar y, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 373 X-User-to-User: 48656c6c6f;encoding=hex X-FS-Support: update_display,send_info Remote-Party-ID: <sip:0000000000@10.10.104.125>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1371708570 1371708571 IN IP4 10.10.104.125 s=FreeSWITCH c=IN IP4 10.10.104.125 t=0 0 m=audio 7398 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 7198 RTP/AVP 98 99 34 31 100 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 a=rtpmap:100 H263-2000/90000 ------------------------------------------------------------------------ send 319 bytes to udp/[10.10.104.125]:5060 at 08:12:48.525061: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.104.125;rport=5060;branch=z9hG4bKU0j6a2D3Zg02g From: "" <sip:0000000000@10.10.104.125>;tag=Sa1rSF0gKeHmF To: <sip:1002@10.10.104.125> Call-ID: 093d3725-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ recv 319 bytes from udp/[10.10.104.125]:5060 at 08:12:48.525061: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.104.125;rport=5060;branch=z9hG4bKU0j6a2D3Zg02g From: "" <sip:0000000000@10.10.104.125>;tag=Sa1rSF0gKeHmF To: <sip:1002@10.10.104.125> Call-ID: 093d3725-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ send 1352 bytes to tcp/[10.10.104.125]:30101 at 08:12:48.551064: ------------------------------------------------------------------------ INVITE sip:1002@10.10.104.125:30101;transport=TCP;ob SIP/2.0 Via: SIP/2.0/TCP 10.10.104.125;branch=z9hG4bKv9BZcXy6vSpNc Max-Forwards: 69 From: "" <sip:0000000000@10.10.104.125>;tag=UvKaX51QD0XSp To: <sip:1002@10.10.104.125:30101;transport=TCP;ob> Call-ID: 09412ee0-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE Contact: <sip:mod_sofia@10.10.104.125:5060;transport=tcp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summar y, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 373 X-User-to-User: 48656c6c6f;encoding=hex X-FS-Support: update_display,send_info Remote-Party-ID: <sip:0000000000@10.10.104.125>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1371708570 1371708571 IN IP4 10.10.104.125 s=FreeSWITCH c=IN IP4 10.10.104.125 t=0 0 m=audio 7398 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 7198 RTP/AVP 98 99 34 31 100 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 a=rtpmap:100 H263-2000/90000 ------------------------------------------------------------------------ recv 284 bytes from tcp/[10.10.104.125]:30101 at 08:12:48.558064: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TCP 10.10.104.125;received=10.10.104.125;branch=z9hG4bKv9BZcXy6vSpNc Call-ID: 09412ee0-5424-1231-178e-171268c989f7 From: <sip:0000000000@10.10.104.125>;tag=UvKaX51QD0XSp To: <sip:1002@10.10.104.125;ob> CSeq: 45521920 INVITE Content-Length: 0 ------------------------------------------------------------------------ recv 479 bytes from tcp/[10.10.104.125]:30101 at 08:12:48.577066: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/TCP 10.10.104.125;received=10.10.104.125;branch=z9hG4bKv9BZcXy6vSpNc Call-ID: 09412ee0-5424-1231-178e-171268c989f7 From: <sip:0000000000@10.10.104.125>;tag=UvKaX51QD0XSp To: <sip:1002@10.10.104.125;ob>;tag=d1fb24be0fba4269bbf3fce36d8bb485 CSeq: 45521920 INVITE Contact: "1002" <sip:1002@10.10.104.125:30101;transport=TCP;ob> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 ------------------------------------------------------------------------ send 930 bytes to udp/[10.10.104.125]:5060 at 08:12:48.580067: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.104.125;rport=5060;branch=z9hG4bKU0j6a2D3Zg02g From: "" <sip:0000000000@10.10.104.125>;tag=Sa1rSF0gKeHmF To: <sip:1002@10.10.104.125>;tag=tKtHUaHmgQ76a Call-ID: 093d3725-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE Contact: <sip:1002@10.10.104.125:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summar y, refer Content-Length: 0 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:1002@10.10.104.125 X-FS-Support: update_display,send_info Remote-Party-ID: "Outbound Call" <sip:1002@10.10.104.125>;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ recv 930 bytes from udp/[10.10.104.125]:5060 at 08:12:48.580067: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.104.125;rport=5060;branch=z9hG4bKU0j6a2D3Zg02g From: "" <sip:0000000000@10.10.104.125>;tag=Sa1rSF0gKeHmF To: <sip:1002@10.10.104.125>;tag=tKtHUaHmgQ76a Call-ID: 093d3725-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE Contact: <sip:1002@10.10.104.125:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summar y, refer Content-Length: 0 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:1002@10.10.104.125 X-FS-Support: update_display,send_info Remote-Party-ID: "Outbound Call" <sip:1002@10.10.104.125>;party=calling;privacy=off;screen=no Maybe there is some originate command option to translate SIP headers, or some sort of variable to do that? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Passing-custom-SIP-header-thr… Sent from the freeswitch-users mailing list archive at Nabble.com.

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