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Freeswitch and luarocks
by Alex Lake 13 Jun '13

13 Jun '13

I was wondering about how one might use luarocks with the lua used by Freeswitch - is this possible? Can't see much of help in the wiki, but if I find out, will pay the tax! ;-)

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fs_ivrd exits/crashes after each call
by Ali Pey 13 Jun '13

13 Jun '13

Hello, I have installed freeswitch 1.2.9 and am using fs_ivrd and perl to control the call. When the call/script ends, fs_ivrd also ends. Any ideas? How can I debug this? Is there any logs for fs_ivrd that I can look at? Any help would be very much appreciated. Regards, Ali

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G.729 transcoder licensing
by Alex Lake 13 Jun '13

13 Jun '13

If we want to just have a play with G.729, can we buy just a couple of channels and then add to them later? And how does it cope with some channels? Presumably it stops offering G.729 when they're all in use? The other question is how does the G.729 licence bind itself to a system? Is this a hardware ID thing, or a Freeswitch ID? Does it work on VMs?

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fs_ivrd vs. outbound event socket
by Ali Pey 13 Jun '13

13 Jun '13

Hello, What are the differences between fs_ivrd and outbound event socket? Why would someone pick one vs. the other? Thanks, Ali Pey

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Mid Call Transfer
by Thiruvallar Raveendran 13 Jun '13

13 Jun '13

Hi everyone, What is the best way to implement below scenario in freeswitch? I have used att_xfer to transfer the calls from B to C. I couldn't implement the below numeric key action in receiver side with att_xref. You help on this issue will be much appreciated. Midcall Transfer Party A - The Caller Party B - The First Receiver Party C - The Second Receiver In a typical scenario, Party A dials a DDI and Party B receives the call. The Midcall Transfer control is now available to Party B. Options: If Party B presses the hash key (#) Party A will be placed on hold and holding music will be played. The hold will be cancelled if Party B presses the star key (*). While Party A is on hold, Party B may dial the number of Party C. After Party C answers the call, Party B can use the numerical keys for the following options: . 1 Key - disconnects Party B and connects Parties A and C . 2 Key - disconnects Party C allowing Party B to dial a new destination (to dial an alternative Party C) . 3 Key - disconnects Party C and reconnects with Party A . 4 Key - toggles between Parties A and C Thanks, Cheers, Thiru

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LUA XML libraries
by Alex Lake 13 Jun '13

13 Jun '13

What LUA XML libraries are people currently using here? I'm interested in simply parsing and manipulating XML documents. Don't really need xpath or suchlike.

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Event Socket Library (ESL) Perl and Yum (Centos)
by Ali Pey 13 Jun '13

13 Jun '13

Hello, Can I install freeswitch on Centos with yum and still get the perl ESL libraries? If not, how can I add ESL perl libraries to an already yum installed freeswitch version 1.2.9? Thanks, Ali Pey

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Is ODBC compiled in or not?
by Alex Lake 13 Jun '13

13 Jun '13

Is there a way to tell if ODBC has been compiled into the FS core? Maybe there's a log file left over from the make somewhere?

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Unable to secure b-leg to user
by Peter Waldheim 13 Jun '13

13 Jun '13

Dear list, I am trying to configure SRTP. This seems to work fine for calls from a local sip user. Clients are registered via TLS and outbound-call a-legs use SRTP. The problem is to secure the call to local sip users, coming in via openzap in this case. On this call's b-leg, I am unable to make freeswitch offer SAVP (in this case then leading to bria refusing the call due to being configured to accept secure calls only). The local sdp: 2013-06-13 13:13:50.040774 [DEBUG] sofia_glue.c:1219 Local SDP: v=0 o=FreeSWITCH 1371101270 1371101271 IN IP4 out.ip.address s=FreeSWITCH c=IN IP4 our.ip.address t=0 0 m=audio 20760 RTP/AVP 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 19462 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=rtpmap:98 H264/90000 In the dialplan I tried combinations of these: <action application="set" data="sip_secure_media=true" /> <action application="set" data="sdp_secure_savp_only=true" /> <action application="export" data="sip_secure_media=true" /> <action application="export" data="sdp_secure_savp_only=true" /> <action application="bridge" data="{local_var_clobber=true,sip_secure_media=true}user/$1(a)our.profile"/> Also I tried setting the sip_secure_media variable in the user's profile. Any hints on how to make this work (or further debug) would be greatly appreciated. Best regards, Peter

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G.719
by Brian West 13 Jun '13

13 Jun '13

Anyone have any endpoints to test this codec with? email me off list please. FYI its a 48kHz full band codec. -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest

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