May 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Guidelines for adding a custom SIP INFO package?
by D D 29 May '15

29 May '15

Hi, I would like to add a custom SIP INFO package, controlled by an event socket application. I know I can use sip_h variables to set header fields, but is there a better way to identify to sofiathat I am adding a custom SIP INFO package? Also, do you have any tips on processing INFO events and sending INFO messages? Thanks!David

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Building FreeSWITCH master (> v1.7) on CentOS?
by James Criscuolo 29 May '15

29 May '15

Hi, I've run into a bit of an issue trying to get the latest HEAD compiled after yesterday's merge and update to 1.7. The issue (at least one of them) seems to lie with libyuv. The precompiled package at files.freeswitch.org only has files for debian, and building libyuv from source also does not seem feasible (it looks like Chromium (part of the build process) can only be compiled on Ubuntu for some reason). I am running CentOS 6.2. Has anybody figured out a way around this? It looks like libyuv is a necessary piece of the build, but if there is a way to disable it that would also work (although not preferable). Thanks, James -- James Criscuolo Software Engineer OnSIP www.onsip.com p. 212.933.9190 x7047 SIP/Email: james(a)onsip.com

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Re: [Freeswitch-users] Guidelines for adding a custom SIP INFO package?
by D D 29 May '15

29 May '15

By Package I mean a set of INFO capabilities, as described by RFC 6086(RFC 6086 - Session Initiation Protocol (SIP) INFO Method and Package Framework). | | | | | | | | | RFC 6086 - Session Initiation Protocol (SIP) INFO Method and Package Framework[Docs] [txt|pdf] [draft-ietf-sipcor...] [Diff1] [Diff2] [Errata] PROPOSED STANDARD Errata Exist | | | | View on tools.ietf.org | Preview by Yahoo | | | | | Part of the RFC specifies the setting of header fields like Recv-Info and Supported.I was wondering if I can configure these values somewhere, or if I will need to set themin the esl program. Also, I know I can set these headers in an originate request, but what if I receive an Invite with Recv-Info and I want the 200 OK to respond with the proposed Recv-Info? Thanks!!David > What do you mean by "package"? You can use esl and events to send custom info. > On May 28, 2015, at 11:26 AM, D D <tru083 at yahoo.com> wrote: > > Hi, > > I would like to add a custom SIP INFO package, controlled by an event socket application. > > I know I can use sip_h variables to set header fields, but is there a better way to identify to sofia > that I am adding a custom SIP INFO package? > > Also, do you have any tips on processing INFO events and sending INFO messages? > > Thanks! > David

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Displaying number of incoming calls
by royj＠yandex.ru 29 May '15

29 May '15

Hi, all There is an application 'bridge' with data '[leg_timeout=15]user/100|user/100,sofia/gateway/gateway_name/2000000'. The idea is to call '100' within 15 seconds and then, if no answer, to call '100' and '2000000' simultaneously. The application above works fine, but 100's phone displays two incoming call. This is clear, because after '[leg_timeout=15]user/100' FreeSWITCH sends CANCEL and then again INVITE to 100. Is there any way/practice to perform the idea with one incoming call displayed on 100's phone?

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WEBRTC was working with 1.4, having no luck with fs_video
by Chris Mandra 29 May '15

29 May '15

Hi guys. I had been able to use webrtc via chrome with FS 1.4 for connections and confenerce calls without incident (although the bandwidth and video quality seemed rather low as compared to the apprtc.appspot.com demo.) I have followed the instructions as well I am able on https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video but so far I can't make an actual connection; the certs pass the cert test without issue, but I have no joy using webrtc and fs_video a few questions: 1. Is there any known problem using wildcard certs from godaddy? 2. is there other configuration beyond what's on the web page needed to make webrtc work? (I have the apropos ports open etc) 3. Are there any known issues with fs_video and sip.js? 4. I feel like it's getting hung op on dtls, as it never passes "2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio DTLS state from OFF to HANDSHAKE" (it repeats that twice with: "2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video packet 0 bytes (ice not ready @ line 4241!)" not long after - does that speak to something obvious to you guys? 5. is there a working webrtc / fs_video demo I can look at? I realize this may all be solved by the release of 1.6, but I need this working now. I've been looking forward to this bc I know there have been some issues with webrtc, rtcp, freeswitch and solid BWE. Thanks for all you do, chris

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On call-retry intervals
by Gabriel Peredo 29 May '15

29 May '15

Hello all I have a question regarding call retries. I see that there are two sets of variables relating to this: Those relating to “camping”: campon_retries campon_sleep And two other ones that don’t appear under “variables” in the wiki: retry-seconds expire-seconds (I include this one because I believe that expire-seconds/retry-seconds would be the number of retries?) My question is: if I want to have the first leg on a bridge to be retried if it rejects the call, some minutes apart, which ones should I use? Thanks in advance.

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 29 May '15

29 May '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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Some calls have one-way audio
by Yehavi Bourvine 29 May '15

29 May '15

Hi, Before I dig into this issue and start collecting data in order to open a Jira: Has anyone else noticed it? When I call or get a call sometimes there is one way audio. Hanging up and calling again usually solves it. It happens quite rarely, so I cannot reproduce it at will. However, it annoys the users... I am running FreeSwitch 1.4.18 and use various phones (Polycom, Yealink, SNOM, etc.). Thanks! __Yehavi:

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Install libyuv-dev needed
by jungle Boogie 28 May '15

28 May '15

Hello All, I'm updating master to latest code and my make is failing: checking for libyuv >= 0.0.1280... configure: error: You need to install libyuv-dev. Required library Makefile:936: recipe for target 'config.status' failed make: *** [config.status] Error 1 Where can I get this required library file? I'm running raspberry pi 2 raspbian flavor. Thanks! -- ------- inum: 883510009027723 sip: jungleboogie(a)sip2sip.info xmpp: jungle-boogie(a)jit.si

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change reponse 503 on max-sessions
by Luciano de Oliveira 28 May '15

28 May '15

Hi gentleman, How do i change the SIP/2.0 503 Maximum Calls In Progress, response when max-sessions parameter is reached. I´m having some problems with the provider. Best regards,Luciano.

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