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dial plan - event socket question
by jonathan augenstine 19 Mar '08

19 Mar '08

I would like to pass data from the dial plan/mod-java to a Java application on the Event socket interface. Is there anyway to do that through Freeswitch or do I need to pass it via normal Java mechanisms. Jonathan

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Re: [Freeswitch-dev] freeswitch Scenario
by Yogesh Patel 17 Mar '08

17 Mar '08

#1 - Is your test server up and running right now? --> tets server is up and running but its on Private IP. #2 - Are you running with the default configs? --> we are running with default configured, configured wikipbx earlier with freeswitch but removed later when we found it can not help us with what we want. #3 - Have you made any changes or additions to the default configs? --> we have made few changes to make things working and did stuff here and there but nothing removed, and almost everything is same except added some more files. #4 - What revision are you on? --> we use SVN so I guess it will be the latest one. Let's get those nailed down first and then we'll take it from there. Yogesh

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freeswitch Scenario
by Yogesh Patel 17 Mar '08

17 Mar '08

HI Here what we want to do with the switch. Our scenario is like few wholesale carrier will sendus traffic for different destinations, which we will be sending to thedifferent vendors. So we were using 206.222.8.58 as our live switch to testfreeswitch. So we need to understand howfreeswitch is terming following things and where do it makes them so we cankeep adding more customers and vendors. Once it starts working for us. Scenario 1 customer name “Allianz” will be sending call to us from IP 209.62.77.114. presently we can configure this customer as a SIP but we also want to do h323 there. (we don’t authenticate any customer for user/pass, they only come with IP and prefix) they will send a call to us for destinations like India(91) and Egypt(20) so we have to create appropriate dialplan for them. for India(91) calls should go to indiavendor1 on ip address 122.252.236.10 as a 1st preference and once that is fill it sld go to indiavendor2 on IP address 122.252.236.11 calls for Egypt proper (20) and Egypt Mobile(201) should go to Egyptvendor1. Scenario 2 one SIP user created on freeswiotch which we can use to make test calls to the Gateways(vendors) we add to the switch for the testing purpose. that SIP user should be able to call any existing vendors/gateways for different destinations and also new one we will add in future so I think this one will have different dialplan. Scenario 3 1. where do we have to create SIP users which we can give it to retail customersfor them to cone in any ip, means they have to have user/pass and ipprovided by us to let them register in our system and once done they can make acall out NOTE: where it will dump CDR for all above calls ? Thanks Yogesh.

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TEST TO -dev LIST, IGNORE
by Michael Collins 17 Mar '08

17 Mar '08

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SRTP and negotiation and vendor compliance.
by Brian West 15 Mar '08

15 Mar '08

After many talks and various vendors and more reading RFC3711 and RFC4568 below is what we have determined to be correct for SRTP negotiation in the SDP over TLS: Here is how the SDP's should look in all three cases: On but optional/preferred: v=0 o=root 1130561626 1130561626 IN IP4 10.0.1.241 s=call c=IN IP4 10.0.1.241 t=0 0 m=audio 52970 RTP/SAVP 9 0 8 2 3 18 4 101 a=rtpmap:9 g722/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:WvPreyjK82pM0I5vtUY2zkpIKPbRVSKH1QcPrsWP a=ptime:60 m=audio 52970 RTP/AVP 9 0 8 2 3 18 4 101 a=rtpmap:9 g722/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:60 On but Manditory: v=0 o=root 1130561626 1130561626 IN IP4 10.0.1.241 s=call c=IN IP4 10.0.1.241 t=0 0 m=audio 52970 RTP/SAVP 9 0 8 2 3 18 4 101 a=rtpmap:9 g722/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:WvPreyjK82pM0I5vtUY2zkpIKPbRVSKH1QcPrsWP a=ptime:60 Off not offered/Not Supported: v=0 o=root 1130561626 1130561626 IN IP4 10.0.1.241 s=call c=IN IP4 10.0.1.241 t=0 0 m=audio 52970 RTP/AVP 9 0 8 2 3 18 4 101 a=rtpmap:9 g722/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:60 I would like some feedback on this as I'm getting what seems to be some push back from vendors that do this incorrectly. i.e. having crypto lines in an RTP/AVP which is wrong. You should never have crypto lines in an RTP/AVP they belong in the RTP/SAVP profile as per RFC3711. I'm working with as many vendors as I can to ensure that they all behave the same way. Can someone tell me if I'm wrong here? Polycom already does it like this, snom should be following suit and i'm working with Grandstream and Aastra right now on this. In addition a vendor shouldn't turn on the padlock/secure logo if the exchange didn't take place over TLS. Thanks, Brian West FreeSWITCH.org

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Re: [Freeswitch-dev] OZ Question: zap_span_find()
by Anthony Minessale 07 Mar '08

07 Mar '08

correct, Assuming the typo in the example was an accident. (span not my_span_pointer declared on the first line) zap_span_t *my_span_pointer = NULL; uint32_t span_id = 1; zap_status_t result; result = zap_span_find(span_id, &my_span_pointer); if it returns ZAP_SUCCESS, that indicates that the desired span exists, is in use and that the pointer will now point to it. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Michael Collins <mcollins(a)fcnetwork.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Thursday, March 6, 2008 8:27:35 PM Subject: [Freeswitch-dev] OZ Question: zap_span_find()  Does this function do what I think it does? If I give it a span number to look for, it returns ZAP_SUCCESS if it finds the span and it sets the value of the pointer variable that I call it with? Example, what happens if I do this, assuming span 1 is a real span? zap_span_t *span = NULL; uint32_t span_id = 1; result = zap_span_find(span_id, &my_span_pointer); Does *my_span_pointer now point to span 1? Any caveats that I should be aware of? Thanks, MC -----Inline Attachment Follows----- _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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OZ Question: zap_span_find()
by Michael Collins 07 Mar '08

07 Mar '08

Does this function do what I think it does? If I give it a span number to look for, it returns ZAP_SUCCESS if it finds the span and it sets the value of the pointer variable that I call it with? Example, what happens if I do this, assuming span 1 is a real span? zap_span_t *span = NULL; uint32_t span_id = 1; result = zap_span_find(span_id, &my_span_pointer); Does *my_span_pointer now point to span 1? Any caveats that I should be aware of? Thanks, MC

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Re: [Freeswitch-dev] VAD, TALK and NOTALK events
by Anthony Minessale 05 Mar '08

05 Mar '08

The algorithm used is not so great and could stand to be replaced by a better one if we get the chance. basically we decode the audio to signed linear 16bit samples for each packet we receive and iterate the samples and add up the absolute values then compute the average and compare it to the acceptable level considered "talking" then we use a counter to make sure it's in the same state several times in a row before actually changing the status of the channel to talking or not talking. A real vad algorithm would also be able to filter out the voice from the background noise and judge the energy based only on that. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: KMTDBOURSE <kmtdbourse(a)free.fr> To: freeswitch-dev(a)lists.freeswitch.org Sent: Thursday, February 14, 2008 1:51:55 PM Subject: Re: [Freeswitch-dev] VAD, TALK and NOTALK events Hi, I ma interested in the VAD algorithm used where where is the code exactly? maybe a link any document support about the vad alogorithm used therein? thanks, Zelda At 01:47 14/02/2008, Anthony Minessale wrote: They are enabled by the vad params in the sofia profile xml <param name="vad"> Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Cesar Cepeda <cesar(a)auronix.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Wednesday, February 13, 2008 10:17:23 AM Subject: Re: [Freeswitch-dev] VAD, TALK and NOTALK events Please correct me if I’m wrong. The events are indeed fired, but only if the SWITCH_VAD_FLAG_EVENTS_TALK or NOTALK flags are set for an rtp_session. I found no place in the code where those flags are set. That is what I meant by not being “enabled”. I subscribed to the events and received nothing. Then I set the flags on switch_rtp_enable_vad function, and then I started receiving the events. Perhaps there is another way of doing it that I’m missing. Cesar Cepeda. De: freeswitch-dev-bounces(a)lists.freeswitch.org [ mailto:freeswitch-dev-bounces@lists.freeswitch.org] En nombre de Anthony Minessale Enviado el: Miércoles, 13 de Febrero de 2008 08:17 Para: freeswitch-dev(a)lists.freeswitch.org Asunto: Re: [Freeswitch-dev] VAD, TALK and NOTALK events The events are fired. I don't know what you mean by "enabled" You can either bind an event to a callback in the load function of your custom module. or connect to mod_event_socket on tcp and request "events talk notalk" void my_event_handler(switch_event_t *event) { } if (switch_event_bind(modname, SWITCH_EVENT_TALK, SWITCH_EVENT_SUBCLASS_ANY, my_event_handler, NULL) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't bind!\n"); return SWITCH_STATUS_GENERR; } if (switch_event_bind(modname, SWITCH_EVENT_NOTALK, SWITCH_EVENT_SUBCLASS_ANY, my_event_handler, NULL) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't bind!\n"); return SWITCH_STATUS_GENERR; } Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Cesar Cepeda <cesar(a)auronix.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Tuesday, February 12, 2008 6:53:03 PM Subject: [Freeswitch-dev] VAD, TALK and NOTALK events Hi, >From looking at the code of switch_rtp.c I can see that VAD is done if enabled in the sip profile in order to not send packets when there is no noise. There are also two events {TALK, NOTALK} which can be fired when VAD kicks in, nevertheless these two events are enabled nowhere (or at least I didn't find a place in the code where they were enabled), so no matter if you ask for them you don't receive them. My questions are the following: Is there some particular reason for not enabling these two events? Is the event queue easily overflowed if these two events are enabled? Thanks. Cesar Cepeda. _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org Never miss a thing. Make Yahoo your homepage. -----Inline Attachment Follows----- _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org Looking for last minute shopping deals? Find them fast with Yahoo! Search. _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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ATTENTION: Codec Module Changes applies to EVERYONE!
by Brian West 19 Feb '08

19 Feb '08

FreeSWITCHers, Today we removed mod_g711, mod_g722, mod_g726, mod_gsm and mod_l16, g711 and l16 were moved into the core but still use the loadable module interface because without these two you can't do anything. We now have mod_voipcodecs which covers G722, G726, GSM, IMA_ADPCM and LPC10. Changes would need to be made to your config files that are currently installed. 1. modules.conf in your source directory will need these changes: Remove: codecs/mod_l16 codecs/mod_gsm codecs/mod_g722 codecs/mod_g711 codecs/mod_g726 Add: codecs/mod_voipcodecs 2. modules.conf.xml in your prefix/conf/autoload_configs will need to be changed also: Remove: <load module="mod_l16"/> <load module="mod_gsm"/> <load module="mod_g722"/> <load module="mod_g711"/> <load module="mod_g726"/> Add: <load module="mod_voipcodecs"/> With this change we have gained working LPC10 @ 90ms and DVI4 (IMA_ADPCM) at 8k and 16k. Thanks, Brian West FreeSWITCH.org PS: please do not reply ALL to this email ;)

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Re: [Freeswitch-dev] VAD, TALK and NOTALK events
by Anthony Minessale 14 Feb '08

14 Feb '08

They are enabled by the vad params in the sofia profile xml <param name="vad"> Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Cesar Cepeda <cesar(a)auronix.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Wednesday, February 13, 2008 10:17:23 AM Subject: Re: [Freeswitch-dev] VAD, TALK and NOTALK events  Please correct me if I’m wrong. The events are indeed fired, but only if the SWITCH_VAD_FLAG_EVENTS_TALK or NOTALK flags are set for an rtp_session. I found no place in the code where those flags are set. That is what I meant by not being “enabled”. I subscribed to the events and received nothing. Then I set the flags on switch_rtp_enable_vad function, and then I started receiving the events. Perhaps there is another way of doing it that I’m missing. Cesar Cepeda. De: freeswitch-dev-bounces(a)lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] En nombre de Anthony Minessale Enviado el: Miércoles, 13 de Febrero de 2008 08:17 Para: freeswitch-dev(a)lists.freeswitch.org Asunto: Re: [Freeswitch-dev] VAD, TALK and NOTALK events The events are fired. I don't know what you mean by "enabled" You can either bind an event to a callback in the load function of your custom module. or connect to mod_event_socket on tcp and request "events talk notalk" void my_event_handler(switch_event_t *event) { } if (switch_event_bind(modname, SWITCH_EVENT_TALK, SWITCH_EVENT_SUBCLASS_ANY, my_event_handler, NULL) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't bind!\n"); return SWITCH_STATUS_GENERR; } if (switch_event_bind(modname, SWITCH_EVENT_NOTALK, SWITCH_EVENT_SUBCLASS_ANY, my_event_handler, NULL) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't bind!\n"); return SWITCH_STATUS_GENERR; } Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Cesar Cepeda <cesar(a)auronix.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Tuesday, February 12, 2008 6:53:03 PM Subject: [Freeswitch-dev] VAD, TALK and NOTALK events Hi, >From looking at the code of switch_rtp.c I can see that VAD is done if enabled in the sip profile in order to not send packets when there is no noise. There are also two events {TALK, NOTALK} which can be fired when VAD kicks in, nevertheless these two events are enabled nowhere (or at least I didn't find a place in the code where they were enabled), so no matter if you ask for them you don't receive them. My questions are the following: Is there some particular reason for not enabling these two events? Is the event queue easily overflowed if these two events are enabled? Thanks. Cesar Cepeda. _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org Never miss a thing. Make Yahoo your homepage. -----Inline Attachment Follows----- _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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