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23 Apr '16

I am trying to show the shorter display for STATUS in the index.html in the video_demo. So it shows like the Freeswitch conference status. It seems like the CFLAG_JSON_STATUS is set to true to display the detailed status. Can someone tell me how to set this flag to false or where it is being set to true? Thanks, Jack Loranger

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JitterBuffer possible problem when configured with number of packets.
by Joel Serrano 22 Apr '16

22 Apr '16

Hi all, I'm doing several tests jitterbuffer related. I've read in different posts to the list that you can specify packet size ("20"), or the number of packets ("1p"), why is it not being enabled if I use 1p instead of 20? (Example: http://lists.freeswitch.org/pipermail/freeswitch-docs/2015-November/000532.… ) Scenario: ** Add in the sofia profile: <param name="auto-jitterbuffer-msec" value="20"/> --- Make a call ** Output in debug log: [...] 4557180d-579a-4c3c-98a3-287cb078e913 2016-04-20 00:57:03.570889 [DEBUG] switch_core_media.c:1964 Setting Jitterbuffer to 20ms (1 frames) (50 max frames) [...] Scenario 2: ** Add in the sofia profile: <param name="auto-jitterbuffer-msec" value="1p"/> --- Make a call Log doesn't say that jitterbuffer has been enabled. Can it be that 1p doesn't output the log but is working anyway? I have removed the auto-jitterbuffer-msec from the sofia profile, and done both tests setting jitterbuffer_msec directly in the dialplan with "20" and with "1p", the result is the same: 20 works, 1p doesn't. If I use 1p, I don't have to worry about the size as if its 20ms, 40ms or whatever it will work, using "20", the setting would not be valid for 40ms.. (please correct me if I'm wrong). I know this can be a bug and thus should be handled in JIRA, I will be happy to create the ticket etc etc, I first want to make sure that the problem is not my config. Tests are with version 1.6.7 -14-d38d065 64bit Thanks! Joel.

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Recommendations for a 'demo' PBX setup?
by Piotr Starzyk 22 Apr '16

22 Apr '16

I’m really liking FreeSWITCH so far, and am seriously considering replacing our current Toshiba Strata system, with a FreeSWITCH implementation. To that end, I’m trying to put together a ‘proof of concept’ system to demo to my supervisor. I’ve done a bit of googling, but couldn’t find a satisfactory answer. I basically plan on purchasing two SIP phones, and connecting them via FreeSWITCH to an analog phone line. I will then demonstrate different features, like Voicemail, IVR, Presence, Conferencing (basically a lot of what FreeSWITCH demo dialplan has to offer). My question is, what are some relatively cheap components that are compatible with FreeSWITCH? From my online searches, it looks like Sangoma is the right choice, but the prices on their FXS/FXO cards start at a couple hundred dollars, and while it would be fine for final install, I can’t justify that for a ‘proof of concept’ setup. I’m hoping to spend less than $100 per each component below (so about $400 total). The less the better, but it should obviously be usable. What I’d like is recommendations for the following, FreeSWITCH compatible hardware: - 2 SIP Phones - FXS/FXO card (PCI or USB, or gateway or whatever is most economical) - Linux (or Windows) PC capable of running FreeSWITCH and interfacing with the above card I’d appreciate any suggestions. Thanks.

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Re: [Freeswitch-users] Freeswitch and Ping
by jungle Boogie 22 Apr '16

22 Apr '16

On 20 April 2016 at 18:10, Clive Lansink <clive(a)lansink.co.nz> wrote: > > Is there something about pinging that might impact on Freeswitch or VOIP? What would cause this? Were you connecting to an ITSP or using freeswitch for sip calling directly? If it's the former, have your soft client register directly to the provider and see what happens while running ping? What's your soft client anyway? -- ------- inum: 883510009027723 sip: jungleboogie(a)sip2sip.info xmpp: jungle-boogie(a)jit.si

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Re: [Freeswitch-users] Freeswitch and Ping
by Clive Lansink 22 Apr '16

22 Apr '16

I did wonder about that but then I tried it on the LAN and got the same results. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 -----Original message----- From: Michael Jerris <mike(a)jerris.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Freeswitch and Ping Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Date: Wed, 20 Apr 2016 22:52:30 -0400 sounds like a broken router to me On Wednesday, April 20, 2016, Clive Lansink <clive(a)lansink.co.nz> wrote: > > Hi everyone. I often use Freeswitch set up as a client to provide a soft > phone on my laptop. > > But recently I noticed a significant problem when at the same time I was > running a standard Windows "ping" in the background. The speech I heard > from the other end would often stutter, and the remote party complained > that my speech sometimes sounded gittery. > > I reasoned that ping doesn't take up much network bandwidth so while in > conversation I did something like copy a large file from my laptop to > another PC on the network and there was no disturbance to the conversation. > It only seems to be ping that causes this. > > Is there something about pinging that might impact on Freeswitch or VOIP? > What would cause this? > > Further information. > > the above refers to the standard Windows ping command. I also have a > Python script that pings a specified host, and running it also caused the > call quality to degrade. I tried various changes to the source code to no > effect, including: > * only creating the socket once and then reusing it instead of it being > created for each ping. > * Only creating the data packet once and reusing it instead of it being > created for each ping. > * Using a very short packet size. > * Eliminating the select statement and just sleeping for a period of time > before reading the response in case the select itself was at fault. > * Eliminating the step of dissecting and decoding the received packet. > None of these made any difference so it seems to me that whatever it is > must be something quite fundamental to something at the lower networking > level. > > Thank you. > > > Clive Lansink > Email: Clive(a)Lansink.Co.NZ <javascript:;> > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <javascript:;> > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <javascript:;> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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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Verto latency
by Gregor Nanger 22 Apr '16

22 Apr '16

If I make call from SIP client everything is ok. But if I make call from verto demo client, there is 2 second delay of audio in direction from browser (latest chrome) to callee. Other direction is ok. Is it posible to tweak lower this latency?

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Nat issue
by Miha 22 Apr '16

22 Apr '16

Hi our FS is on public ip but the voip server to which trunk is made (SFB) is behind NAT. So the issue is that in SDP is private ip and FS sends RTP to this private ip, it does not change to src ip from which sip request was received. Remote SDP: 2016-04-21 08:04:00.954752 [DEBUG] sofia.c:5614 Remote SDP: v=0 o=- 46176 1 IN IP4 172.16.2.3 s=session c=IN IP4 172.16.2.3 b=CT:1000 t=0 0 m=audio 31200 RTP/AVP 8 101 13 c=IN IP4 172.16.2.3 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=rtcp:31201 a=label:Audio And then: xxx.xxx.xxx.xxx (FS public ip) port 17094 -> 172.16.2.3 port 31200 codec: 8 ms: 30 I tried with: <param name="NDLB-force-rport" value="true"/> <param name="aggressive-nat-detection" value="true"/> on internal profile and on trunk profile but the same. tnx for help! Miha

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Codec negotiation issue
by Mickael Hubert 22 Apr '16

22 Apr '16

Hi list, I have an issue with the negociation codec in my Freeswitch. I followed this doc: https://wiki.freeswitch.org/wiki/Codec_Negotiation The call flow: 1) INVITE: UAC -- (G729, PCMA) --> FS -- (G729, PCMA) --> SVI Asterisk 2) 200OK UAC <-- (G729) -- FS <-- (PCMA) -- SVI Asterisk (HOMER) I have inherit_codec=true in my dialplan and inbound-late-negotiation true in sip-profile. But freeswitch do not force codec learned from leg B to leg A. *LEG B (200OK):* *2016-04-01 11:28:16.453784 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMA:8:8000:150:64000]/[G729:18:8000:20:8000]2016-04-01 11:28:16.453784 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMA:8:8000:150:64000]/[PCMA:8:8000:20:64000]2016-04-01 11:28:16.453784 [DEBUG] switch_core_media.c:3248 Audio Codec Compare [PCMA:8:8000:20:64000] ++++ is saved as a match2016-04-01 11:28:16.453784 [DEBUG] switch_core_codec.c:111 sofia/internal/06********@HOMER Original read codec set to PCMA:8* LEG B is in G711A, OK, next: 2016-04-01 11:28:16.453784 [NOTICE] sofia.c:6727 Channel [sofia/internal/06*******@HOMER] has been answered 2016-04-01 11:28:16.453784 [DEBUG] switch_channel.c:3686 (sofia/internal/06*********@HOMER) Callstate Change DOWN -> ACTIVE 2016-04-01 11:28:16.473813 [DEBUG] switch_ivr_originate.c:412 *Setting codec string on sofia/external/0326793005(a)1.1.1.1 <0326793005(a)1.1.1.1> to PCMA@8000h@20i* We can see, FS set the correct codec to LEG A; good ! , next... *2016-04-01 11:28:16.473813 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [G729:18:8000:20:8000]/[G729:18:8000:20:8000]2016-04-01 11:28:16.473813 [DEBUG] switch_core_media.c:3248 Audio Codec Compare [G729:18:8000:20:8000] ++++ is saved as a match2016-04-01 11:28:16.473813 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]2016-04-01 11:28:16.473813 [DEBUG] switch_core_media.c:3248 Audio Codec Compare [PCMA:8:8000:20:64000] ++++ is saved as a match2016-04-01 11:28:16.473813 [DEBUG] switch_core_media.c:2139 Set Codec sofia/external/0326793005(a)1.1.1.1 <0326793005(a)1.1.1.1> G729/8000 20 ms 160 samples 8000 bits* Why FS compare and re set a new codec (G729) ? Next: *2016-04-01 11:28:16.613776 [ERR] mod_g729.c:145 This codec is only usable in passthrough mode!2016-04-01 11:28:16.613776 [ERR] switch_core_io.c:1245 Codec G.729 decoder error!* normal .... *For informations:* - FreeSWITCH Version 1.5.8b+git~20140214T000311Z~fe2a4d6d47~64bit (git fe2a4d6 2014-02-14 00:03:11Z 64bit) Is it a bug ? an configuration error ? Thanks in advance PS: *My dialplan:* *<condition field="${destination_number}" expression="^0([1-79][0-9]{8})$" break="on-true"><action application="export" data="codec_string=${ep_codec_string}"/><action application="set" data="inherit_codec=true"/><action application="bridge" data="sofia/internal/${destination_number}@HOMER"/></condition>* -- Cordialement HUBERT Mickaël Ingénieur VOIP - Hexanet --

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London Marathon 2016
by Andrew Cassidy 22 Apr '16

22 Apr '16

Hi Guys, Not like me to spam this sort of stuff to the list, but it's worth a go! Sunday is the London Marathon. My other half is running this year for the charity WhizzKidz. They're a UK based charity who supply equipment to disabled children, more information can be found here: http://www.whizz-kidz.org.uk/ Any contributions would be much appreciated. Here's the link to her justgiving page: https://www.justgiving.com/Rachel-Wilson39/ Thanks for your time! Kind regards, -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director 03303 880 960 andrew(a)cassidyweb.co.uk <andrew(a)cassidywebservices.co.uk> www.cassidyweb.co.uk <http://www.cassidywebservices.co.uk>

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How to send Re_invite
by amani mansour 22 Apr '16

22 Apr '16

Hi all, I need to configure FS to send 491 request pending message after receiving Re invite from the caller , The problem that i don't know how to send the re invite request ,which xml file i whould modifie it ? please help me ,i have done many research but i didn't find anything . Thank you , Best Regards Amani

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