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Question on switch_core_media_bug_add
by Stephen Wilde 15 Jul '13

15 Jul '13

Hi all, I'm trying to capture early media packets of an originated call without success. I first call the: switch_ivr_originate() with ignore_early_media=false that returns success then I call the: switch_core_media_bug_add() with SMBF_READ_REPLACE that returns also success and also I see a call to my callback with: type=SWITCH_ABC_TYPE_INIT The problem is that during early media phase I don't see any call to my media bug callback. The callback is called only when the call is answered. Any idea on that? What is my error? Stephen

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Wed conf calls
by Ken Rice 15 Jul '13

15 Jul '13

Hey guys, Let us know who you¹d like to see do a presentation on the weekly conf. Are there any good products you would like to see also? -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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Freeswitch stop working error 900
by cjbujold＠accra.ca 15 Jul '13

15 Jul '13

Freeswitch has been working for several months with no problems. This morning we walk in and it will not connect to our gateway. Keep getting Fail_Wait error. When we look at the log we get the following error: [ERR] sophia_reg.c:2236 mountain Registration Failed with status Internal error at nua_client.c:550 [900]. How can we fix this error? Thanks

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FreeSWITCH Weekly News and Notes
by Michael Collins 15 Jul '13

15 Jul '13

Happy Mid-July to Everyone! This week we will continue with the theme of community discussions on the weekly conference call <http://wiki.freeswitch.org/wiki/FS_weekly_2013_07_17>. We will have a follow up from last week's discussion as well as an update from Ken on where we stand with the latest 1.2 version. Last week's discussion<http://wiki.freeswitch.org/wiki/FS_weekly_2013_07_10>covered some topics from the mailing list as well as some assorted questions from the audience. Please join us this week so that we can have a further discussion of subjects that are of interest to you. In ClueCon 2013 news we have two important items. First, we would like make it official: there will be OpenSIPS training<http://www.opensips.org/Community/Summit-2013Chicago>on Monday August 5th! The training will be conducted by Bogdan-Andrei Iancu and Flavio Goncalves. All paid ClueCon attendees should register<http://www.opensips.org/Community/Summit-Registration>with the discount code of "OPENSIPSUMMIT2013". The training schedule can be found here <http://www.opensips.org/Community/Summit-2013Chicago-Schedule>. Second, we would like to remind everyone that you can still increase your chances of winning in our legendary ClueCon gift drawings. Like last year, the "bits will shift" at the end of the week and the number of tickets received will be halved. Register this week and you will receive eight entries into the drawing for the Android tablets and the other great items. Visit our website <http://www.cluecon.com> to register for ClueCon and to book your hotel room at the Hyatt. Only 21 days until ClueCon 2013! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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Re: [Freeswitch-users] Is Cluecon for me?
by Jose Suero 15 Jul '13

15 Jul '13

i'm in your exact same spot. Would love to hear some answers admin <admin(a)tedssupply.com> wrote: >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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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Is Cluecon for me?
by admin 15 Jul '13

15 Jul '13

Hi, Just thought some of you could help me decide about attendance. Just put in first system, learning fast, but many times the hard way! Worked through the books and wiki, so I am past being a beginner. I was looking at the list of presentations and although they seemed very interesting, I need some "intermediate" knowledge. Most conferences I attend, I seem to learn as much around the other attendees as I do in the formal sessions. If you have any thoughts I would greatly appreciate them. I also need to justify the expense... Thanks, James

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Call splitting by audio in/out
by Stanislav Sinyagin 15 Jul '13

15 Jul '13

It looks like it's not available in the dialplan, but at the C library level it seems to be doable: it will be great if a call could be split into two b-legs, one for input audio, and the other one for output. There are a number of usage scenarios when an operator needs to listen and to speak to two different groups of people. This can be partially achieved with a set of conferences bridged together with mute option, but something more straightforward would be great. Another interesting scenario is when the operator speaks two-way with someone, and is also able to listen to some other channel at the same time. regards, stanislav

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xml-curl-mod on call ANSWER
by adahary 15 Jul '13

15 Jul '13

I'm using xml-curl-mod to handle directory/dialplan events and xml-cdr-mod to handle CDRs. I'm missing the Dialplan XML POST that should be triggered when a call is answered. How can I get the an XML POST when call is ANSWERed ? Thanks Assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/xml-curl-mod-on-call-ANSWER-t… Sent from the freeswitch-users mailing list archive at Nabble.com.

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FreeSWITCH adds WebRTC support to new 1.4 BETA.
by Shahrzad A. 15 Jul '13

15 Jul '13

Hi I'm also having the exact problem as Ashley! After upgrading to 'OpenSSL 1.0.1e 11 Feb 2013' I can not even successfully pass the 'make' part and I get "./.libs/libfreeswitch.so: undefined reference to `SSLv2_client_method" Any suggestion ? Sherry

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SIP attack SPA2102
by Miha 15 Jul '13

15 Jul '13

Hi, last night we noticed SIP attack from one of our costumar. Interesting thing is that the call come from Linksys SPA2012 which was behind NAT. The port forward was made for web access(xxx.xxx.xxx.xxx:8099). How it is possible do call from SPA2012 (relay calls)? I am looking at SPA but do not see any of that options. How SIP traffic could be send to SPA from which calls was send as there was no SIP port fwd on router? The device from which calls was send is SPA2012. THank you very much for all explenation!!! Miha

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