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Patch: VMD Configurable MIN_TIME
by Andrew Fritz 25 Nov '09

25 Nov '09

I've created a patch to override the value of MIN_TIME in the vmd modules using a channel variable. In this way, it can be configured on a call by call basis. The channel variable is name "vmd_min_time". I didn't add the other detection parameters, but doing so would be straight forward. So, in our app, we can catch T-Mobile and the other problematic cell carriers beeps. I did this because in our app, we would rather have false positives than miss the start of recording on a voice mail system. This way, anyone using the VMD module can configure the vmd module to be as touchy or hard to trigger as they would like. I not sure how to implement it (at least in the vmd module code), but a way to make mod_vmd more robust to false positives, especially with short beeps would be to have it look for short silence immediately proceeding and/or following the beep. I've noticed that it tends to trigger on noise if there is tone in the noise, if for example I extend a syllable in a word or I have music on in the background. However on a voice mail system there will likely be a short near silence before the tone and an indefinite silence after it. In fact, background noise should be non-existent, except for line noise which should be Gaussian and not look like a structured tone. Looking for a beep + near silence after it for some period should eliminate many false positive where tones are embedded in other sounds (e.g. music or someone holding a vowel for longer than normal). Andrew Index: src/mod/applications/mod_vmd/mod_vmd.c =================================================================== --- src/mod/applications/mod_vmd/mod_vmd.c (revision 15668) +++ src/mod/applications/mod_vmd/mod_vmd.c (working copy) @@ -162,6 +162,8 @@ /*! A count of how long a distinct beep was detected * by the discreet energy separation algorithm. */ switch_size_t timestamp; + /*! The MIN_TIME to use for this call */ + int minTime; } vmd_session_info_t; static switch_bool_t process_data(vmd_session_info_t * vmd_info, switch_frame_t * frame); @@ -312,7 +314,7 @@ if (c < (POINTS - MAX_CHIRP)) { vmd_info->state = BEEP_NOT_DETECTED; - if (vmd_info->timestamp < MIN_TIME) { + if (vmd_info->timestamp < vmd_info->minTime) { break; } @@ -541,6 +543,7 @@ switch_channel_t *channel; vmd_session_info_t *vmd_info; int i; + const char *minTimeString; if (session == NULL) return; @@ -588,6 +591,14 @@ switch_channel_set_private(channel, "_vmd_", bug); + minTimeString = switch_channel_get_variable(channel, "vmd_min_time"); + if (minTimeString != 0) { + sscanf(minTimeString,"%d",&(vmd_info->minTime)); + } else { + vmd_info->minTime = MIN_TIME; + } + + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "MIN_TIME for call: %d\n",vmd_info->minTime); } /*! \brief Called when the module shuts down

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how to enable short recordings
by kokoska rokoska 25 Nov '09

25 Nov '09

Hello all, is there a way how to enable very short recordings (1-3 seconds) in FreeSWITCH other than editing source code and recompiling? Thanks for your time! Best regards, kokoska.rokoska

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How to find whether the destination extension supports encryption
by Yehavi Bourvine 25 Nov '09

25 Nov '09

Hello, We have a mix of phones that support RTP encryption and those that do not. I have to support both types in the meanwhile, and would like to have encryption enabled on the relevant leg, even if the other leg does not support it (why? one of our ATAs either must have it unencrypted or have it encrypted, but cannot have both). How do I find whether the *destination* supports encryption? I do not want to manage an additional table in the database... Thanks! __Yehavi:

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mod_conference kick to abort invitations
by Jan Thiemo Fricke 25 Nov '09

25 Nov '09

Hi members, I'm controlling freeswitch with the conference module via xmlrpc. Is it desired that the kick command can only kick users that are connected to the conference? Is there no chance abort an invitation? The kick command has no effect until the person I invited with the dial command is connected. Thanks in advance! Jan

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modify SDP for 200 OK
by Juan Backson 25 Nov '09

25 Nov '09

Hi, If I am using proxy_media=true, bypass_media=false, is there anyway of modifying o= and c= so that it won't show the IP of the far-end B leg? I am using fs as b2b2a and I want to hide the far-end ip as much as possible. I got to hide the IP for invite by modifying the sdp within C code, but I don't know how to do that for 200 OK. Any idea? Thanks, jb

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Re: [Freeswitch-users] how to enable short recordings
by Jeff Lenk 25 Nov '09

25 Nov '09

Is this for vm? If so set min-record-len on the profile -----Original Message----- From: kokoska rokoska Sent: 11/25/2009 3:11:05 PM To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] how to enable short recordings Hello all, is there a way how to enable very short recordings (1-3 seconds) in FreeSWITCH other than editing source code and recompiling? Thanks for your time! Best regards, kokoska.rokoska _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Call Transfer Help Please
by Dave Stevenson 25 Nov '09

25 Nov '09

Hi, I'm trying to setup call transfer for a phone without a transfer button. I was on IRC last night and got some pointers to how this is setup in dialplan.xml and features.xml and what "bind meta app" does. Once it became clear how the transfer is initiated and that the transfer, in the default config, can only be initiated by the "b" leg of the call, I was able to make this work as configured in the defaults, i.e, to initiate a transfer (for an internal call) from the dialled extension to a new extension. Now the problem . . . I have an incoming PSTN line that rings a group of extensions, what I want to be able to do is to give whoever answers the PSTN call ability to transfer the call on to another extension. There is an ATA (Linksys SPA3101) set up on the PSTN line with a FreeSwitch extension of 1000, it rings the extension phones in the group. I'd hoped that the default transfer setup would handle this without modification - the incoming call on extension 1000 would be the "a" leg, the answering extension would be the "b" leg and a transfer from "b" would work as per the default config. This does not work for me though. I'm struggling a bit with the "bind meta app" options and can't seem to make it do what I want. Could someone please confirm that what I'm trying to do is feasible and perhaps suggest the right parameters to use in dialplan.xml and features.xml please ? Relevant section in the "is_transfer" section in features.xml <action application="transfer" data="-bleg ${digits} XML default"/> And in default.xml from <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> to I've tried posting a call log to the Pastebin (11252/3) but there was an error - it looks like the dump was too big. Not sure what the maximum size on pastebin dumps is ? My understanding (or lack of) of "a" and "b" are in the scenario described is not helping ... Is the "a" leg the call coming in on the PSTN line (on Ext 1000) ? Is the answering extension the "b" leg ? What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ? What is the correct "transfer" data string in features.xml ? Or am I totally on the wrong track here ? If it is possible to do what I want, and changes are required to the dialplan.xml and/or features.xml files, is it possible to have different logic in there such that the actions are different whether it is the "a" leg or "b" leg that's requesting the transfer ? regards Dave FreeSwitch Version 1.0.4 (14460)

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How Register soft sip phones to FreeSWITCH with extension number.
by ovvenkat 25 Nov '09

25 Nov '09

Hi to All, Any one please tell me , How to configure soft sip phone to freeswitch with extension number. -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV.

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How to connect SIP phone to freeswitch
by ovvenkat 25 Nov '09

25 Nov '09

Hi . Could you please tell me, How to connect sip phone (which one is more friendly with freeswitch) to freeswitch. How I can check whether connection is properly established or not? -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV.

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register timeout / cisco 7960
by Phillip Jones 25 Nov '09

25 Nov '09

hi there, I have set up some cisco 7960 up with fs. They work fine - but the only way I can keep them registered is to set the "timer_register_expires" in the Cisco cfg file to something really short like 10s. Does anyone know the default register timeout for fs? And where I might change this in fs? Thanks! Phil

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