January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Min-SE Header
by DJB 05 Jan '10

05 Jan '10

I am wondering whether it is possible to suppress the Min-SE Header Field in SIP INVITE message. Thank you, Dorn B.

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XML IVR and Javascript menu-exec-app
by Alberto Escudero 05 Jan '10

05 Jan '10

Hi, I am writing several IVRs using Freeswitch XML http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr One of the nodes of the IVR is a Javascript application that records a message. e.g.: <entry action="menu-exec-app" digits="3" param="javascript foo.js"/> The Javascript application starts by issuing a session.answer() [records the voice message] exit(); Once the Javascript exits, the channel is dropped and hence the IVR terminates. Is it possible to write a Javascript application that once is completed, the channel returns back to the top menu of the IVR? I want to emulate the same behavior that "menu-play-sound", that once the file is played, the IVR logic returns to the top menu. -aep -- Stopping junk mailers is good for the environment

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Polycom & TLS - help please...
by Yehavi Bourvine 05 Jan '10

05 Jan '10

Hello, I am writing again because I am quite desparate... I fail to enable TLS on Polycom while on SNOM I can make it work with the same configuration. From TCPDUMP I see the following difference between the two handshakes: - SNOM: The protocol suite used is TLS_RSA_WITH_RC4_128_SHA - Polyco: The protocol suite used is TLS_RSA_WITH_AES_256_CBC_SHA and sessionTicketTLS also appears there. After the key exchange the phone disconnects the connection. Any idea how to debug it? Thanks! __Yehavi:

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How to konw who picks up in group bridge
by François Legal 05 Jan '10

05 Jan '10

Hello, In my diaplan, when a call arrives on some specific channel, it is routed to an extension that tries to bridge it on multiple channels using coma separated list (either openzap, sofia or both). I would like to see in my CDR which channel did pick up which channel did pick up the call. Which variable can I use ? Thanks François

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Unable to start mod_java
by Max Bridgewater 05 Jan '10

05 Jan '10

Hi, I built Freeswitch with mod_java enabled. But now, when Freeswitch starts, I get the following error message: 2010-01-04 22:32:46.574770 [NOTICE] switch_loadable_module.c:271 Adding API Function 'jsrun' 2010-01-04 22:32:46.574811 [NOTICE] switch_loadable_module.c:271 Adding API Function 'jsapi' 2010-01-04 22:32:46.575306 [NOTICE] modjava.c:244 Java Framework Loading... 2010-01-04 22:32:46.575721 [ERR] modjava.c:133 Error loading /usr/local/java/jdk1.6.0_17/jre/lib/i386/client/libjvm.so 2010-01-04 22:32:46.575743 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** 2010-01-04 22:32:46.576742 [CONSOLE] switch_loadable_module.c:890 Successfully Loaded [mod_lua] 2010-01-04 22:32:46.576748 [NOTICE] switch_loadable_module.c:209 Adding Dialplan 'LUA' 2010-01-04 22:32:46.576795 [NOTICE] switch_loadable_module.c:249 Adding Application 'lua' As far as I can tell the /usr/local/java/jdk1.6.0_17/jre/lib/i386/client/libjvm.so is the correct path. Any idea? max.

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No audio after Remote SDP:
by Mark Campbell-Smith 05 Jan '10

05 Jan '10

Hi! I'm sure this is a NAT issue, but I'm not sure what options to use. I have a Linksys SPA3102, NAT'd on the internet (remotely) and connected to my FS on the otherside of the world, which is also natted. A PAP2T is connected on the same subnet as the FS. The 3102 registers successfully and a call can be set up from the PAP2 to the 3102. However, after FS receives the Remote SDP the audio stops (ring tone stops in my case) 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3646 Channel sofia/internal/sip:2001@192.168.1.3:56885 entering state [completing][200] 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3657 Remote SDP: v=0 o=- 18490612 18490612 IN IP4 192.168.1.3 s=- c=IN IP4 192.168.1.3 t=0 0 m=audio 16432 RTP/AVP 2 100 101 a=rtpmap:2 G726-32/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 I notice that the ip address in the o and c fields indicate a local IP address. Should this IP address be an external IP address of the 3102 instead? Thanks

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Force endpoint to use rfc2833 for dtmf
by Peter P GMX 05 Jan '10

05 Jan '10

Hello, in a bigger installation with some thousand endpoints in the field we see, that the endpoint equipment is always using INFO messages (standard setting is auto, so the endpoint decides which method to use). I have 2 questions to that scenario: 1. Is there a way that Freeswitch forces/restricts the endpoint to use rfc2833 or not to send to allow INFO in the invite message? 2. Currently INFO messages do not get forwarded from the caller through freeswitch to called endpoint. How can we enable that FS is fowarding the INFO messages? Best regards Peter

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Re: [Freeswitch-users] Self alarm
by Sharad 05 Jan '10

05 Jan '10

Hi I am also intresting in the same. Is there any script for this functionality. Regards -- View this message in context: http://n2.nabble.com/Self-alarm-tp4235713p4238924.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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time stamp for sched_api
by Sharad 05 Jan '10

05 Jan '10

I am trying the following API. I want this API to run on 3rd Jan 2010 at 15:13 hours. sched_api 010315132010 none originate sofia/external/1006(a)192.168.4.106 &playback(ivr/alarm.wav) But it is not getting activated on the maturity of this time. Can someone let me know the error in the time format. Regards Sharad -- View this message in context: http://n2.nabble.com/time-stamp-for-sched-api-tp4249284p4249284.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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How to control call volume?
by Nicolas Brenner 04 Jan '10

04 Jan '10

Hi, is there a way of controlling the volume of a call? I'm bridging calls with a JS script. Sometimes the people getting the calls complain the volume is too low. I've recorded a few of the calls and most of the times, while playing the recorded wav files, the volume of LegB (second leg of the bridge) is pretty hard to hear, even with the computer and player volume to the max (ok, it's a laptop, but even with headphones). I saw there are some volume control options for conferences, but I couldn't find anything for regular "originate calls" or bridges. I am doing transcoding, so that might help (?). Thank you very much for your help. Best, Nicolas

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