June 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Help get freeswitch.com back
by Anthony Minessale 03 Jun '10

03 Jun '10

Some squatter has had freeswitch.com for years and it's time we had it back. We need $1,300.00 for a single-party panel and $2,600.00 for a 3-party panel to resolve the dispute. They are using the domain to pose as a VoIP site (if you keep loading http://www.freeswitch.com/ you can see it in action) Don't let creeps like this misuse the internet! Use the paypal button on the site to donate to the cause. Everyone who donates will get their name up on our thank you page on our site. Put "freeswitch.com UDRP and your name as you would like to see it on the thank-you page" in the note on the paypal form. -- 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 <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:+19193869900

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OSTAG - Open Source Telephony Action Group
by Michael Collins 03 Jun '10

03 Jun '10

Good news! The Open Source Telephony Action Group (OSTAG) is almost ready to become an official 501(c)3 non-profit organization. We need to raise $800 in order to file the necessary paperwork. We have set up a PayPal account where we can all send our donations: donations(a)ostag.org Please assist us in moving this important organization forward. OSTAG is dedicated to the advancement of open source telecommuncations software that improves the lives of people all over the world. This is a most worthy cause! Please help now and in the future. -Michael

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Out-Going Call Transfer Question
by Michael Gende 03 Jun '10

03 Jun '10

Hello, I wonder if someone could direct me to some documentation (I'm sure it exists) on how to undertake something. We've been happily using FS for some time now. Our is set-up emulates a pretty standard key system. To the point of my post: I can, for incoming calls, easily and successfully transfer them to conference rooms and other registered extensions. However, if I call out to another party, I can only transfer the call to a conference room, not another extension. I suspect this has to do with FS not knowing - or me not correctly telling it how - to handle the "out going" case. I.e, which "end" of the call to transfer. Sometimes its handy to make a call and transfer the callee to some other extension. Any suggested reading? Thanks, Mike G.

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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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Status of absolute_codec_string
by Kristian Kielhofner 03 Jun '10

03 Jun '10

Hello everyone, Either I am misunderstanding absolute_codec_string or there is something going on with it in trunk. I had a git checkout from a month ago that exhibited this same behavior so I updated about an hour ago. It's still there (or I'm still confused). In short, I have a call that comes in from a remote endpoint that will advertise PCMU and G729 (in that order). When I bridge the call I want to remove G729 capability from the outbound leg in some cases so FS doesn't have to transcode. In other cases (controlled from the dialplan), I want to place G729 as the first codec (again so FS doesn't have to transcode) but I can force G729. I've tried regex on ep_codec_string and I've tried regex on the SDP. Both seemed to work well but what didn't work was setting absolute_codec_string from either of these. Even when I try to set it manually without any fancy conditions it doesn't appear to work. The codec prefs from the SIP profile are presented to the remote endpoint. Full debug here: http://pastebin.freeswitch.org/13088 Any ideas? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

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xml_curl endpoint register
by Gustavo Espeche 03 Jun '10

03 Jun '10

Hello all, i'm new in freeswitch we are migrating our software form yate to freeswitch. I'm writing a registration php script using xml_curl module, the registration was good but some endpoint don't use 5060 as sip port and when i register it, i can't find the endpoint port in curl post, because of then when i dial to an endpoint that isn't listen in sip port 5060 the call fail. Some one know how can i fix it? when i register the endpoint i insert in a mysql db time, ip, and when some one dial to this user i read my db for know the user's ip. but i need insert the endpoint sip's port too for it work. Best Regards. Gustavo Espeche www.easyipcall.com

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I edit a china tones.conf , but it's not useful. does anybody can help me?
by 王洪永 03 Jun '10

03 Jun '10

I edit a china tones.conf ,but it's not useful. does anybody can help me? [cn] generate-dial => v=-7;%(1000,0,475) detect-dial => 475 generate-ring => v=-7;%(1000,4000,475) detect-ring => 475 generate-busy => v=-7;%(350,350,475) detect-busy => 475 generate-attn => v=0;%(100,100,1400,2060,2450,2600) detect-attn => 1400,2060,2450,2600 generate-callwaiting-sas => v=0;%(300,0,440) detect-callwaiting-sas => 440,480 generate-callwaiting-cas => v=0;%(80,0,2750,2130) detect-callwaiting-cas => 2750,2130 detect-fail1 => 913.8 detect-fail2 => 1370.6 detect-fail3 => 1776.7 tones for china China (People’s Republic of) Busy tone - 450 0.35 on 0.35 off Congestion tone - 450 0.7 on 0.7 off Dial tone - 450 continuous Second dial tone - 450 continuous Intrusion tone - 450 0.2 on 0.2 off 0.2 on 0.6 off Number unobtainable tone - 450 3x(0.1 on 0.1 off) 0.4 on 0.4 off Ringing tone - 450 1.0 on 4.0 off Special information tone - I 450 0.4 on 0.04 off Special information tone - II 950 0.4 on 10.0 off Waiting tone - 450 0.4 on 4.0 off 王洪永中企开源-IP通讯部 Thu, 3 Jun 2010 16:45:17 +0800o

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Re: [Freeswitch-users] Set Privacy:id on Outbound Call
by Michael De Lorenzo 02 Jun '10

02 Jun '10

Does this conflict with setting the originator caller id and name? Sorry for replying to the full digest.

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Re: [Freeswitch-users] Using XLite/Softphone with Freeswitch
by Michael De Lorenzo 02 Jun '10

02 Jun '10

Is there a best practice for organizing these dialplans? Or is it more the developer's preference? Apologies for replying to the full digest, it was a mistake. .michael

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No Early Media Ringback
by Praveen Ray 02 Jun '10

02 Jun '10

Hi All I am using Freeswitch 1.0.6 with Vitelity as inbound/outbound provider. My Inbound calls route properly to extension 1000, however, the caller does not hear the extension ringing. The extension at 1000 does ring however and once I pick up the call, the conversation proceeds normally. I just can't figure out why the caller hears silence during phone ringing. I have tried putting instant_ringback in dialplan/public.xml: <extension name="vitel-inbound"> <condition field="destination_number" expression=""> <action application="set" data="instant_ringback=true"/> <action application="transfer" data="1000 XML default"/> </condition> </extension> The log lines during inbound calls show: 2010-06-02 07:55:15.913827 [NOTICE] mod_sofia.c:1992 Pre-Answer sofia/external/7816409134(a)64.2.142.15! 2010-06-02 07:55:15.916855 [DEBUG] switch_core_session.c:703 Send signal sofia/internal/sip:1000@172.168.128.193:5060 [BREAK] 2010-06-02 07:55:15.916855 [DEBUG] sofia.c:4167 Channel sofia/external/ 2123135987(a)64.2.142.15 skipping state [early][183] 2010-06-02 07:55:15.916855 [DEBUG] switch_core_session.c:642 Send signal sofia/external/2123135987(a)64.2.142.15 [BREAK] 2010-06-02 07:55:15.916855 [DEBUG] switch_ivr_originate.c:1124 Raw Codec Activation Success L16@8000hz 1 channel 20ms 2010-06-02 07:55:15.916855 [DEBUG] switch_core_codec.c:122 sofia/external/ 2123135987(a)64.2.142.15 Push codec L16:10 2010-06-02 07:55:15.916855 [DEBUG] switch_ivr_originate.c:1189 Play Ringback Tone [%(2000,4000,440.0,480.0)] 2010-06-02 07:55:15.922793 [DEBUG] sofia.c:4172 Channel sofia/internal/ sip:1000@172.168.128.193:5060 entering state [calling][0] 2010-06-02 07:55:15.981765 [DEBUG] sofia.c:5866 IP 64.2.142.15 Rejected by acl "domains". Falling back to Digest auth. 2010-06-02 07:55:15.984797 [WARNING] sofia_reg.c:1873 Can't find user [ prav_24.90.81.72(a)172.168.128.203] You must define a domain called '172.168.128.203' in your directory and add a user with the id="prav_24.90.81.72" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2010-06-02 07:55:15.984797 [WARNING] sofia_reg.c:1033 SIP auth failure (INVITE) on sofia profile 'internal' for [5087203445(a)24.90.81.72] from ip 64.2.142.15 where 5087203445 is my inbound DID. Can someone please help decipher what the warning means and why there might not be any early media ringback.. thanks -Praveen

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