June 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

No video when calling into conference with Cisco Jabber
by Ari Sonesh 17 Jul '18

17 Jul '18

We are getting audio only but no video when calling into conference with Cisco Jabber. Other SIP video clients (e.g Jabber) calling into the conference are OK. Sip negotiations are fine, SDP is fine and all parties agreeing on H.264 codec. The log (at debug level) doesn't show any relevant errors or warnings. Any ideas what can be wrong and/or how to troubleshoot the problem? Thank you *Ari *

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m=audio 0 RTP/AVP 19
by Victor Bogatyryev 17 Jul '18

17 Jul '18

Hello, I apologize for my bad English. Three years ago there was a message http://lists.freeswitch.org/pipermail/freeswitch-users/2015-June/113829.html and here I am faced with the same problem. A stupid provider sends crypto in RTP/AVP and to ensure processing this request, I inserted in the profile the parameter NDLB-allow-crypto-in-avp=true But Freeswitch answers to INVITE inserting in OK this line: m=audio 0 RTP/AVP 19 and the provider understands this as "no audio" and does not send ACK and does not send rtp. How to deal with this I do not know. Regards. Victor Bogatyryev

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Conference audio issues when Verto user sends video to audio-only conference
by Chad Phillips 13 Jul '18

13 Jul '18

I’ve got an audio-only conference configured, and some Verto clients (iOS to be specific) are also sending video to that conference. I had hoped that the video would simply be discarded, but instead I get this error, and audio doesn’t seem to be sent/received properly: [WARNING] switch_core_media.c:13944 verto.rtc/3521 has no video codec The only solution I’ve found is to configure the conference in question for video as well, but that’s not ideal. Wondering if this is a bug to report to Jira, or possibly there’s some config thing I can do to fix it? Because of iOS limitations, I’m not seeing how to have that client send audio only in my case.

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Freeswitch windows build - libpng
by Gregor Nanger 13 Jul '18

13 Jul '18

Hi! After a year I wanted to build FS on windows and make video tutorial or for FS wiki. I am using latest VS2017 and everything builds except libpng project. Below is error if someone can help me, because I don't have c++ experience: 1>------ Build started: Project: Download zlib, Configuration: Release Win32 ------ 1>Downloading zlib. 2>------ Build started: Project: libpng, Configuration: Release x64 ------ 2>Generating pnglibconf.h 2> 1 file(s) copied. 2>png.c 2>pngerror.c 2>pngget.c 2>pngmem.c 2>pngpread.c 2>pngread.c 2>pngrio.c 2>pngrtran.c 2>pngrutil.c 2>pngset.c 2>pngtrans.c 2>pngwio.c 2>pngwrite.c 2>pngwtran.c 2>pngwutil.c 2>Generating Code... 2>zlib.lib(inflate.obj) : MSIL .netmodule or module compiled with /GL found; restarting link with /LTCG; add /LTCG to the link command line to improve linker performance 2> Creating library D:\Git\freeswitch\x64\Release\libpng16.lib and object D:\Git\freeswitch\x64\Release\libpng16.exp 2>Generating code 2>d:\git\freeswitch\libs\zlib\deflate.c(2097): error C2220: warning treated as error - no 'executable' file generated 2>d:\git\freeswitch\libs\zlib\deflate.c(2097): warning C5045: Compiler will insert Spectre mitigation for memory load if /Qspectre switch specified 2>d:\git\freeswitch\libs\zlib\deflate.c(2097) : note: index 'dist' range checked by comparison on this line 2>d:\git\freeswitch\libs\zlib\deflate.c(2097) : note: feeds memory load on this line 2>d:\git\freeswitch\libs\zlib\deflate.c(1987): warning C5045: Compiler will insert Spectre mitigation for memory load if /Qspectre switch specified 2>d:\git\freeswitch\libs\zlib\deflate.c(1987) : note: index 'dist' range checked by comparison on this line 2>d:\git\freeswitch\libs\zlib\deflate.c(1987) : note: feeds memory load on this line 2>d:\git\freeswitch\libs\zlib\deflate.c(1862): warning C5045: Compiler will insert Spectre mitigation for memory load if /Qspectre switch specified 2>d:\git\freeswitch\libs\zlib\deflate.c(1862) : note: index 'dist' range checked by comparison on this line 2>d:\git\freeswitch\libs\zlib\deflate.c(1862) : note: feeds memory load on this line 2>LINK : fatal error LNK1257: code generation failed 2>Done building project "libpng.vcxproj" -- FAILED. Best regards, Gregor

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Recording calls
by Gregor Nanger 12 Jul '18

12 Jul '18

Hi! One question. Can I record calls with http_api or http_cache modules and post it to webserver? If yes, does anyone one have any example? Best regards, Gregor

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Debian repos
by Igor Olhovskiy 11 Jul '18

11 Jul '18

Hi! Is there a way to set up different version of FreeSWITCH from repos? Means for ex I want to use not 1.6.19, but 1.6.8? Or compile is the only way? Regards, Igor

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Debian 9 and Freeswitch 1.8
by William Colls 11 Jul '18

11 Jul '18

Is there an expected date when Freeswitch 1.8/Debian 9 will become the preferred production configuration?

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[FreeSWITCH v1.6.20] Memory increasing continuously
by Atul Thosar 11 Jul '18

11 Jul '18

Hi All, I am verifying if we can use FreeSWITCH in our Production environment for one of our IVR use case. Our use case involves INVITE containing MIME as a payload, which contains XML and SDP parts. When I ran 10 CPS load with MIME as a payload, I see Memory is continuously growing. Where as same load without MIME (only SDP) payload shows no increase in memory. Is this a known issue? If yes, is this already fixed or patch is available for the same? I am using FreeSWITCH Version 1.6.20+git~20180123T214909Z~987c9b9a2a~64bit (git 987c9b9 2018-01-23 21:49:09Z 64bit) Attached output of top command for both the runs. Btw Following are the changes made in FS configuration - conf/vars.xml + <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.56.101"/> + <X-PRE-PROCESS cmd="set" data="internal_auth_calls=false"/> conf/dialplan/public/00_inbound_did.xml changed as following <include> <extension name="public_did"> <condition field="destination_number" expression="^500$"> <action application="answer"/> <action application="sched_hangup" data="+5 alloted_timeout"/> <action application="park"/> </condition> </extension> </include> conf/autoload_configs/acl.conf.xml + <node type="allow" cidr="192.168.56.102/24"/> Thanks in advance. -- Atul

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Send display update after RTP session is establihed ?
by Melek Oktay 11 Jul '18

11 Jul '18

Hi, I realize very interesting problem of my software stack (FreeSwitch + Kamailio) with Yealink 41P Voip phone.. After 3-4 days research, I realize Yealink phone does not try to start RTP session if it gets UPDATE packet before ACK packet !! In Freeswitch side, do we have a option that send UPDATE packet after RTP session is established? (we have a option in FreeSwitch side: do not send UPDATE packet however this is not good case for me, I update CallerID with this UPDATE packet)

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Incoming RTP with different sequence number and host
by Sharath Kumar 11 Jul '18

11 Jul '18

Hello, I am running FS 1.6.18 and have a provider that is sending me a stream of incoming RTP that changes it's host IP, and RTP sequence number and Timestamps after 10 mins in the call. The SSRC does not change. As a result I am getting either silence or garbled audio. The Sample rate is 20ms as in the original stream. Is there any setting in the FS that can make this work ? Have any of seen this work with inbound stream ? Thank you, Shaks

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