April 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Verto
by Gregor Nanger 25 Apr '16

25 Apr '16

I am building my web phone with verto from scratch and using this tutorial: http://evoluxbr.github.io/verto-docs/ BAsically everything is ok, I can register and make calls, but I cannot hear audio. Mic is working, but not speakers. I see error: Cannot read property 'sinkId' of undefined. This is error line: $.verto.dialog.prototype.setAudioPlaybackDevice = function(sinkId, callback, arg) { var dialog = this; var element = dialog.audioStream; * if (typeof element.sinkId !== 'undefined') {* I get dialog object without audioStream property and hence element is null. I am using self signed certificate. Could this be cause? Best regards, Gregor

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Extensions not registered on IPv6 address
by Shisheer Teli 25 Apr '16

25 Apr '16

Dear Team, I installed FreeSWITCH. Directory and dialplan are default ones. My phones are able to register when using just IPv4, and I can call from one to another. When I try the same thing with IPv6, phones not registered. Am I doing something wrong? Maybe I forgot to configure something, but I can't see what it is. -- Regards, Shisheer T

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Programmatic way to get BOTH Channel IDs of a bridged call?
by Chris Mandra 24 Apr '16

24 Apr '16

Hey guys - I have another questions: Is there a programatic way to obtain BOTH the channel ids of a bridged call? I only have the uuid of one call leg. How do i get the other uuid? (similar to what "show calls" or "show channels" shows in cli but I need to start with one call leg uuid and get the other from it) Thanks! chris

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Freeswitch sound installation is failing on CentOS 7
by Varsha Agarwal 23 Apr '16

23 Apr '16

Hi, I have installed FreeSwitch couple of times on CentOS with no problems. I am doing it on a fresh CentOS 7 VM and running into issues installing sounds. yum install sox freeswitch-sounds* This command fails with error 404 not found. I tried yum clean all and yum update but no luck. Can someone help? Thanks, Varsha

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Verto Caller Id and Caller name
by Gregor Nanger 23 Apr '16

23 Apr '16

I can set in verto javascript initialization caller id and caller name. But can I force or override caller id and name in user registration variables or somewhere else, because I do not want to set caller name in javascript, since user can tweak it.. Any experience?

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module not unloading
by Chris Mandra 23 Apr '16

23 Apr '16

Hi Guys - I hope you are well I never really resolved the issue I was having with modules saying they're unloading, but not really unloading (unless I restart freeswitch) I’m having that issue with the module not unloading again when I add in some flex/bison parser code. It seems the parser is causing the problem where module doesn’t unload. What's the best way to diagnose this issue. freeswitch reports module as unloading but lsof shows it is still being retained Thanks! chris

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Verto Status Display
by Jack 23 Apr '16

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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