March 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

T30 fax analysis from G711 pcap
by Giovanni Maruzzelli 25 Mar '15

25 Mar '15

Hello FreeSWITCHers, there is a way to extract T30 log from a g711 pcap? I must check on various possible incompatibilities and anomalies on a large number of different fax transmission, many of them in pure T30 (faxmachine<->faxmachine). I would need something like http://www.netgencommunications.com/products/faxtapng/... Would it be possible to have a "replay" of the capture to be analysed by spandsp, or some other opensource tool? On T38 I can use wireshark, but on pure T30 it's useless... -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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Polycom headache
by João Mesquita 24 Mar '15

24 Mar '15

Guys, I am sure most of you have been through this but this is my first time. I am having a HOLD problem with Polycom where they refuse to work according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly) to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this is a known thing since I've found several hits on Google about it and they even created an option for it on their provisioning material. Unfortunately, it doesn't work. No matter what I set on the <pre> if ((sdp->sdp_connection && sdp->sdp_connection->c_address && !strcmp(sdp->sdp_connection->c_address, "0.0.0.0"))) { switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, "RFC2543 from March 1999 called; They want their 0.0.0.0 hold method back.....\n"); sendonly = 2; /* global sendonly always wins */ } João Mesquita

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Transfering Attended Call
by Luiz Fernando Softov 24 Mar '15

24 Mar '15

Hi... First of all i make a software with freeswitch and FreeBSD, named Voipr... voipr.brbyte.com It's a softswitch with Billing, Redirect, IVR, Routes, GSM, Gateway and other things. I am not using dialplan, just a outbound socket, with my own ESL Server written in C. I parse the events, and send comands back to make things. Like bridge, play some music, hangup the call.... <extension name="VoiprExt"> <condition> <action application="socket" data="127.0.0.1:3012 async full"/> <action application="hangup"/> </condition> </extension> - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - My question, is how to make a transfer of a call... Like A (1001) call to B (1002). Then i make a BRIDGE between extensions "execute", "set", "dialed_extension=1002" "execute", "export", "dialed_extension=1002" "execute", "set", "call_timeout=30" "execute", "set", "continue_on_fail=true" "execute", "set", "park_after_bridge=true" "execute", "bridge", "user/${dialed_extension}" - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Let's say B (1002) is a secretary who want to transfer the call to C (1003) B (1002), press the key [*], and i play a music, like "enter the desired number and #". B (1002), press the keys 1003# - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - At this point, I have all information, and want to know how 3 things Consider that: the call already bridged between A (1001) and B (1002) - - - - - 1 - Making a call between B and C. Then, when C answer B hangup. And the call return between A and C. - - - - - 2 - Making a call between B and C. Then, when C answer, B talks with C. B press * and B hangup. And the call return between A and C - - - - - 3 - Making a call between A and C. Then, when C answer, A hangup. And the call return to B and C Thank you in advance, who can help me. Sorry for my english. If anyone need help with socket events, gsmopen or xml_curl send me a e-mail. I will be happy to help. Att, Luiz Fernando Softov http://www.softov.com.br fernando(a)softov.com.br

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Freeswitch.org hacked
by tomp＠tomp.uk 24 Mar '15

24 Mar '15

Hi, I think the freeswitch.org site has been hacked. I am seeing adverts for viagra and other drugs top right hand side of the site. Thanks Tom

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internal versus external calls
by Szeto, Steven 24 Mar '15

24 Mar '15

Hi Everyone, When an incoming call comes through via SIP to Freeswitch, is there a way to identify the call as an internal versus external call? An internal call would be from a device hosted by a PBX. An external call would be from the public network (e.g. a cell phone call that has been transited through a PBX to Freeswitch. Regards, Steve -- This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message.

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ClueCon Call For Speakers is Open!
by Ken Rice 24 Mar '15

24 Mar '15

New Post on freeswitch.org from krice387 check it out at http://ift.tt/1GpR0Yu ClueCon Call For Speakers is Open! Call for speakers at ClueCon 2015 is open! Visit http://ift.tt/1FBM7gN to submit your speaking proposal Below! What makes a great ClueCon presentation? The tech savvy crowd that attends ClueCon loves technical presentations. In general, the more technical the presentation, the better. If you are thinking about a presentation then consider these points: * ClueCon talks are 30 minutes in length, including Q&A time with the audience (Lightning Talks are also open! 10 minute talks, 5 minutes Q&A) * ClueCon has a special focus on open source communications, VoIP, and telephony projects like FreeSWITCH, WebRTC OpenSIPS, Asterisk, and Kamailio * Attendees enjoy hearing about projects built with open source tools, especially those in a production environment * Highly technical discussions that show the nuts and bolts are especially well-liked * The audience appreciates seeing and participating in live demonstrations We are especially interested in Security-related talks and demonstrations Please register your proposals at http://ift.tt/1FBM7gN. To complete registration you need the following items: * Working title * Brief description of the talk (abstract) * Name of the presenter(s) * Bio and headshot of presenter(s) * Presenter’s contact information (including mobile phone the presenter will have with them at the conference) Don’t delay! Speaking proposals must be in by July 4, 2015 and scheduling requests are handled on a first come first serve basis. ClueCon 2015: See You There! For more details on ClueCon see https://cluecon.com.

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DTMF missing digits
by Podrigal, Aron 24 Mar '15

24 Mar '15

I am experiencing problems with sonus dtmf. I tried setting different parameters related to the RTP BUGS listed in switch_types.h but it did not seem to fix the problem. I have freeswitch ----> [carrier(sonus), ...] -----> freeswitch *freeswitch sends the following digits to our carrier, and is received by the other end (also a freeswitch) missing the digits marked in black.* *pcap shows the missing digits as being part of the previous packet, so two 1's becomes one 1 with the duration sum of two.* * 1 - 1 - 2 - 1 - 1 - #* *What are the relevant parameters I should set?*

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PDD SIP timer in freeswitch?
by Victor Chukalovskiy 23 Mar '15

23 Mar '15

Good day, Is there a timer in FS that can be set before SIP bridge so that it limits max duration before call gets 180/183/200 response? Something equivalent to timer T310 in PRI terms. If such timer is not implemented, how people are managing to re-route in fail-over / LCR scenarios where a given carrier is taking too long after "100 Trying"? call_timeout does not seem to be the right thing. Thanks in advance! -Victor

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Problem with LUA dbh:query in Freeswitch
by John Carpenter 23 Mar '15

23 Mar '15

I am trying to use Lua to retrieve some currency conversion rates from a postgres database. It works but the numeric field value is only returning the first three decimal places although if I list the data in postgres it has four decimal places. I may be doing something daft as I am quite new to Lua, sample code is below. Any ideas appreciated regards, John local dbh = freeswitch.Dbh("odbc://fs_odbc:fs_user:password") if dbh:connected() == false then freeswitch.consoleLog("notice", "dp.lua cannot connect to database" .. dsn .. "\n") return end assert(dbh:connected()) e_rate = 1.123456789 -- just a test to see decimal places freeswitch.consoleLog("info",string.format(" e_rate = %f", e_rate)) local function eur2usd() local sql = "select eur2usd from currency order by id desc limit 1" dbh:query(sql, function(row) e_rate = row.eur2usd end) end local function eur2gbp() local sql = "select eur2gbp from currency order by id desc limit 1" dbh:query(sql, function(row) e_rate = row.eur2gbp end) end eur2usd() freeswitch.consoleLog("info",string.format(" euro2usd = %f", e_rate)) eur2gbp() freeswitch.consoleLog("info",string.format(" euro2gbp = %f", e_rate))

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FreeSWITCH Week in Review (Master Branch) March 14th-20th
by Ken Rice 23 Mar '15

23 Mar '15

New Post on freeswitch.org from Kathleen check it out at http://ift.tt/1HtsI3F FreeSWITCH Week in Review (Master Branch) March 14th-20th Hello, again. This passed week in the FreeSWITCH master branch we had 6 commits. The features for this week are: the ability to pass -T[path to custom sources path] and -K[custom keyfile path] options to build the FS packages with a custom sources and keyring path when using the debian/utils.sh script to build FS packages and play_and_detect_speech can now detect DTMF if you set the playback_terminators channel variable to any or specific DTMF. Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-7373 When using the debian/utils.sh script to build FreeSWITCH packages, add the ability to pass -T[path to custom sources path] and -K[custom keyfile path] options to build the FS packages with a custom sources and keyring path. FS-7378 Play_and_detect_speech can now detect DTMF if you set the playback_terminators channel variable to any or specific DTMF. The result will be stored in speech_detect_result and in playback_terminator_used. Also added channel variable play_and_detect_speech_close_asr which will release the speech recognition port when the detection is completed. This will prevent speech licenses from being held the entire call. The following bugs were squashed: FS-7297 Fix incomplete patch for previous crash in mod_com_g729.

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