June 2010 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] FXS bridged on FXO ports and DTMF
by devel＠thom.fr.eu.org 08 Jun '10

08 Jun '10

I come back on this topic as I did not find a clean/good solution to this problem. As reminder, the problem occurs when an FXS channel is bridged on an FXO channel (seen with sangoma A400 hardware with HW DTMF detection). The problem is that each leg of the bridge is detecting the inband DTMF, and so freeswitch sends each detected DTMF from one leg to the other, and so on and so forth (as each leg detects the DTMF again and again). I guess what would be needed for these cases is some application that would prevent detected DTMF to be sent to the other leg. Can anybody give me a hint on whether or not such application exists, and if not, a starting point to try to create one. Thanks François

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[Freeswitch-dev] srgs
by Jan Berger 08 Jun '10

08 Jun '10

Hi, I find a way to send ssml directly to the tts, but I can't see a similar function for sending srgs to asr? Does this exist or will I need to add it? Jan

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[Freeswitch-dev] mod_conference - User Classes
by David Swardstrom 07 Jun '10

07 Jun '10

One feature that many conferences have is classes of users. There is the Host/Moderator, speakers, and Guests/participants. In such conferences, the Host may have the ability to Mute all non-speaking Conferees. Currently this requires a separate flag for each user indicating the user class. To toggle conference level mute, the application will need to get a conference list and run through it. For each conferee in the list, the current mute status and user class must be determined. I would like to propose introducing a new member flag: speaker. This indicates that the member is a speaker and should not be automatically muted. This would be provided as a “speaker” flag as part of the status on a list. A new set of Conference APIs would be also be provided: mute & unmute at the conference level. When issued, any member that does not have a moderator or speaker flag set would be muted or unmuted. If the conference is “guest muted”, the first line of the list API response would indicate this a “muted” similar to the locked indicator currently provided. If this seems like a reasonable proposal, I would take a stab at modifying mod_conference and testing it and then submit it via a Jira request. Regards, Paul David Swardstrom

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[Freeswitch-dev] Core dump after 'Auto Changing port'
by vectorno vectorno 07 Jun '10

07 Jun '10

I am calling into freeswith which is on a public ip from a soft phone behind a NAT'ed ip. The call connects and I can hear the playback of the configured audio file but only for a second before a core dump with the message below. 2010-06-05 23:33:59.228651 [INFO] switch_rtp.c:2049 Auto Changing port from 192.168.1.24:8000 to xxx.xxx.xxx.xxx:8000 Segmentation fault (core dumped) Any ideas on what this could be?

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[Freeswitch-dev] Core dump after 'Auto Changing port'
by vectorno vectorno 06 Jun '10

06 Jun '10

I have done a back trace and posted the information on Jira. I hope i did it correctly.

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[Freeswitch-dev] JavaScript - how bad is it?
by Jan Berger 05 Jun '10

05 Jun '10

Hi, I know from a previous postings/discussions that the Javascript we use is considered to be bad - performance and memory wise. Can someone explain a little - who did the Javascript integration? The reason I ask is because I am working on ccxml and vxml - and I need to add ECMAScript due to compatibility with external tools/libraries . The impression I get from wiki is that Lua perform 2 x the speed of JavaScript - does anyone know exactly why? Jan

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[Freeswitch-dev] t38
by Fowler, Peter (Peter) 04 Jun '10

04 Jun '10

What is the current shape of T38 support in mod_fax in the trunk? Is it ready for prime time (real world use) or is it still in development? Peter

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[Freeswitch-dev] 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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[Freeswitch-dev] Status of Bridged Call
by Thirupathi 03 Jun '10

03 Jun '10

Hi, I am receiving the inbound call and i am bridging that call with some other number. But i need to know whether the bridged call (dial out call) has been ringing or answering. I am using the ESL connection method to communicate with the Freeswitch. Here is the snapshot of my code. Receiving inbound call : @sd = s @con = ESL::ESLconnection.new(s.to_i) @con.execute("linger") @con.setEventLock("true") @con.sendRecv("myevents") @con.execute("answer") ... .... After that i am playing and receiving input from the caller then i am bridging the call with some other number. Bridge Call: @con.execute("bridge", "sofia/gateway/junctionnetworks//#{@destination_number}") and waiting for the bridge event WaitForBridgeEvent: def waitForBridge while @con.connected e = @con.recvEvent() next if e == nil printEventData(e) handleHangup if hangupOccurred(e) if e.getHeader("Event-Name") == "CHANNEL_BRIDGE" && e.getHeader("Channel-State") == "CS_EXECUTE" answer_state = e.getHeader("Answer-State") if answer_state == "answered" call_answer_time = e.getHeader("answered_time") end_point_disp = e.getHeader("endpoint_disposition") var_end_point_disp = e.getHeader("var_endpoint_dispoisition") return call_answer_time, end_point_disp, var_end_point_disp end end end end In this code the answer_state is always returning the answered. Also there is always no value for call_answer_time, end_point_disp, var_end_point_disp. Bridging is happening and i have no problem with bridging the call. Please let me know whether the answer-state is returned in this case is for the inbound call or outbound call. (because i am wondering how it is always answered?). Also please help me to get the bridged call status (whether answered, ringing or others) Thanks and Regards, Thirupathi

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[Freeswitch-dev] Freeswitch compilation and openldap
by François Legal 03 Jun '10

03 Jun '10

Hello, I'm wondering whether it would be possible to add a configure option for those (like me) who already have openldap and libldap installed and available on their systems, so that there would not need to have openldap downloaded and compiled during freeswitch compilation, but rather link to the available libldap ? François

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