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send_dtmf of Hook Flash didn't work at all. [ With Analog Card Diguim TDM-410P ]
by jack 20 Apr '15

20 Apr '15

To whom it may concern, Hi, there! I got a question about send_dtmf APP. Detail config files and platform infos note as below. =============================================== OS : Ubuntu 13.10 ( Linux kernel 3.11.0 ) Add-on Telephony Card : Diguim TDM-410P with driver wctdm24xx FreeSWITCH version: 1.4.18~32 bits =============================================== Configuration Files: Here's my freetdm.conf: [span zt myFXS] number => 4001 fxs-channel => 1 number => 4002 fxs-channel => 2 [span zt myFXO] number => 382 fxo-channel => 3 number =>383 fxs-channel =>4 Here's my freetdm.conf.xml: <configuration name="freetdm.conf" description="FreeTDM Configuration"> <settings> <param name="debug" value="0"/> <param name="hold-music" value="$${moh_uri}"/> </settings> <analog_spans> <span name="myFXS"> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="enable-callerid" value="true"/> <param name="dialplan" value="XML"/> <param name="context" value="test"/> </span> <span name="myFXO"> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="enable-callerid" value="true"/> <param name="dialplan" value="XML"/> <param name="context" value="test"/> </span> </analog_spans> </configuration> And here's my very simple dialplan in test.xml: <extension name="send_dtmf_test"> <condition field="destination_number" expression="^4001$"> <action application="send_dtmf" data="0123456789ABCDF*#"/> </condition> </extension> ======================================================================== ISSUE: When I use my analog phone and dial 4001, the send_dtmf APP works just fine except the "F"(Hook Flash) part. I heard all DTMF tones from analog phone speaker but got a Flash Failed showed in the CLI and I paste it as follow. 2015-04-20 10:03:05.730643 [DEBUG] ftdm_io.c:3991 [s1c1][1:1] Generating DTMF [0123456789ABCDF*#] 2015-04-20 10:03:05.730643 [ERR] ftmod_zt.c:773 [s1c1][1:1] FLASH Failed Is there any detailed explanation about this phenomenon? I searched mailing list and got nothing like this. Thanks in advance! Kind Regard, Jack Huang

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How to reset "read_terminator_used" to None
by Aqs Younas 20 Apr '15

20 Apr '15

Hi, Users. I am playing different ivr for different dtmf with read command in my python script. Like. for * play previous file. for # play next file for any other digit play main massage. for no digit continue in script. I see read_terminator_used variable always set if it is previously used. This is making difficult for me to find out have user pressed the terminator(#,*) and entered nothing. Is there any way to reset this variable? Thanks.

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read command playing wav file but not mp3.
by Aqs Younas 19 Apr '15

19 Apr '15

Hi, Users I just upgraded my freeeswitch from 1.4.14 to 1.4.18 and saw read command does not play mp3 file but just wav files. While playing mp3 file I see this. switch_core_file.c:149 Invalid file format [mp3] for [/ bible.didradio.com/bible/languages/english/012054.mp3] and wav file plays smoothly. Is there any change in new freeswitch read command. Because in 1.4.14 both mp3 and wav files were playing. this is my dialplan. <action application="answer"/> <action application="log" data="Online Bible System"/> <action application="read" data="0 10 / bible.didradio.com/bible/languages/english/012054.mp3 res 10000 #"/> Thanks for any pointer.

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Alerts for missed calls
by Podrigal, Aron 19 Apr '15

19 Apr '15

Hi, I would like to know, what would be the best way to handle email alerts for missed calls. 1) Only send the notification if the call was not answered at all. for example when it was ringing a group of extensions. Maybe use a special status code or custom event header? 2) When the call was answered by voicemail and only if the caller did not leave a voicemail, it should be considered missed. 3) If a caller hangs up a call within a IVR. Thanks. -- Aron Podrigal - //Be happy :-)

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playback question
by Dmitry Saratsky 18 Apr '15

18 Apr '15

Hi all, I have .wav file which was recorded by freeswitch in stereo mode a-leg to left channel and b-leg to right channel. now I want to play only one channel to caller using lua. sometimes left one and sometimes right one. I can't find any documentation on playback or any other apps how to play only one channel... Any ideas? Thanks a lot! Dmitry.

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mod_xml_cdr note stripping <'s and >'s as needed
by Kurtis Heimerl 17 Apr '15

17 Apr '15

Hey Users, I seem to have found a bug. mod_xml_cdr is sending along invalid XML as it is not stripping the <'s and >'s off of important fields. For instance, we're sending along a P-Preferred-Identity header: P-Preferred-Identity: <sip:test@127.0.0.1:5060> and it's getting sent as the following in the xml: <sip_P-Preferred-Identity><sip:test@127.0.0.1></sip_P-Preferred-Identity> However, this is invalid xml (for obvious reasons) and won't parse. I'm currently in the process of double checking, but this seems to be new behavior. I tried to strip some of these headers, but they are important stuff like sip_full_from which I don't think I can. <sip_full_from><sip:test@127.0.0.1:5062 >;tag=OBTSiqkuupshsxlztuyb</sip_full_from> Ideas on what's going on here? Thanks!

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cURL response code from API
by Oleg Stolyar 17 Apr '15

17 Apr '15

Hello, if I use in Lua resp = api:execute("curl", some_url) where can I get the curl response code? If I use session:execute("curl", some_url), the code will be in the curl_response_code session variable, but where can I get it if I use api:execute?

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SIP Header Via - IP Wrong
by T Fred Farmington 17 Apr '15

17 Apr '15

My FreeSWITCH server has 2 IP addresses An Internal 192.168,nnn.nn & an External 107,1.nnn.nn This is accomplished via network 'magic' (I'm not the network guy) using a single network card And, in general, it works. I can Receive Outside calls from our SIP Trunk vendor But I can Not Send calls Out through the SIP Trunk vendor. The calls are being blocked (403 'Forbidden") by the vendor due to the contents of our SIP Header packet. Within the Via: part of the packet it shows: Via: SIP/2.0/UDP 192.168.nnn.nn;rport;branch=z9hG4bKQKa7126jtX4mm They need it to show the External IP here. I have even tried setting sip_via_host = 107.1.nnn.nn but that did not change what it being sent. How/Where can I get this to change?

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Disabling REFER on a single gateway
by Tomáš Boros 17 Apr '15

17 Apr '15

Hello freeswitchers, I am trying to disable transfer on a single gateway I am calling bridge using {origination_uuid=${out_uuid},sip_cid_type=pid,disable-transfer=true}sofia/gateway/telekom-ip/${destination_number} and I have also defined param <param name="disable-transfer" value="true"/> for the gateway. When I make calls I can clearly see, that the variable is set "[disable-transfer]=[true]" but in the siptrace I still can see "Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY" and "Allow-Events: talk, hold, conference, refer". REFER should not be there as transfers are disabled. Although the REFER message is there, REFER requests are denied with a 403 forbidden message. Do you think it is a bug? Are there any other purposes of the REFER message than transferring calls? Thank you, -- Tomáš Boros System & Network Administrator VM Telecom, s.r.o. Panenská 8 811 03 Bratislava Slovakia www.vmtele.com <http://www.vmtele.com> | boros(a)vmtele.com <mailto:boros@vmtele.com> Tel: +421 2 20 28 99 79 | Fax: +421 2 20 28 20 30 <http://www.vmtele.com/>

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OpenSIPS Summit May 12 & 13 in Amsterdam!
by Alex Goulis 17 Apr '15

17 Apr '15

The OpenSIPS Project is excited to announce our first European summit on May 12 & 13 2015 at the Westcord Fashion Hotel in Amsterdam! With the release of OpenSIPS 2.1 we'll have so much to cover that we've decided to make this a 2 day event. You can get more information about the summit at: http://www.opensips.org/Community/Summit-2015Amsterdam-Schedule For beginners, get answers to questions like: * why OpenSIPS is the right choice for you network * what are the OpenSIPS capabilities, features and limits * how to extend, expand and enhance your SIP service with OpenSIPS Seasoned professionals can: * Learn about new modules and features * Discuss advanced topics * Discuss the development roadmap for OpenSIPS Speakers include: * Alexandr Dubovikov - Founder of Homer Sipcapture project * Pete Kelly - Sourcevox * Dan Christian Bogos - ITsysCOM * Tijmen de Mes - AG Projects Hotel information: http://www.westcordhotels.com/sea/westcord-fashion-hotel-amsterdam Conference schedule: http://www.opensips.org/Community/Summit-2015Amsterdam-Schedule#toc1 Information on previous summits: http://www.opensips.org/Community/Summits We look forward to seeing you in Amsterdam! -The OpenSIPS Project Team

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