- FreeSWITCH-users - FreeSWITCH Mailing Lists

PRESENCE_IN question
by Adam Kuśmirek 12 Sep '16

12 Sep '16

I need more complicated presence management in my system, so I would like to change blf status in VoIP terminal on my own. I tried with sendevent via event socket: sendevent PRESENCE_IN Answer-State: ringing proto: sip from: 5000(a)ubi2.pl rpid: unknown event_type: presence alt_event_type: dialog presence-call-info-state: progressing presence-call-direction: outbound The problem is that body part of SIP Notify message does not contain status and IPPhone doesnot change BLF LED: <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:5000@ubi2.pl"> </dialog-info> When making call with properly set presence_id, Freeswitch sends SIP Notify message with correct status info: <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="2" state="full" entity="sip:5000@ubiquity.pl"> <dialog id="01ca4d76-3e20-4167-b33c-f134f55b365d" direction="recipient"> <state>ringing</state> </dialog> </dialog-info> It seems that IPPhone needs this state info to change BLF LED button (yealink T21PE2) Is there any way to send NOTIFY message with state info without really making a call ? Regards Adam

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Re: [Freeswitch-users] Hardware requirement for 1000 Calls
by Uday kumar 12 Sep '16

12 Sep '16

Hi, I have developed few future and interface and some modification in old version of freeswitch so my modification is not working in 1.6. This is reason to use old. Its not possible for me to again develop all those changes in new version. I tried in new version but didn't worked so I have only option to use 1.4. Please advise server. Thanks Uday. On Sep 10, 2016 10:55 PM, "Michael Jerris" <mike(a)jerris.com> wrote: Why would you use an old unsupported version? On Saturday, September 10, 2016, Uday kumar <udy786(a)gmail.com> wrote: > Hi All, > > I need your help to setup Freeswitch server for 1000 concurrent call on > CentOS. Freeswitch version 1.4.26 for below modules:- > > *1.* Conference > *2.* CallCenter:- In CallCenter Module will use ring group, queue etc. > *3.* Extension to extension Calls, Extention to Mobile/Landline call > using SIP. > *4.* IVR > *5.* Recording:- Record all Calls (Conference, CallCenter, Extension, IVR > etc) > *6.* To tracking RTP, use tool Voipmonitor. > *7. *For security iptables. > *8.* Database is MySQL. > > To make outgoing and incoming calls will configure multiple Carrier > provider and DID. > > > *Please suggest server configuration for above setup and also how much > Internet speed is required. * > > *Currently I have one dedicated server in cloud with below configuration. * > > *Intel(R) Xeon ** @ 3.30GHz* > *12GB RAM* > *8MB Casche* > *500GB HDD with 7k RPM* > > *Please suggest if this is capable to handle above requirements. * > > > -- > Thanks & Regard > Uday > Site:- www.shareyourknowledge.in > Mobile:- +91-9377579349 > _________________________________________________________________________ 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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Re: [Freeswitch-users] Hardware requirement for 1000 Calls
by Uday kumar 12 Sep '16

12 Sep '16

Thank you everyone for valuable information and suggestions. I will contact consulting(a)freeswitch.org for more information. Thanks Uday. On Sep 12, 2016 10:11 AM, "Ken Rice" <krice(a)freeswitch.org> wrote: correct *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Darren *Sent:* Sunday, September 11, 2016 11:39 PM *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> *Subject:* Re: [Freeswitch-users] Hardware requirement for 1000 Calls Isn’t FreeSWITCH MPL not MIT? *From: *<freeswitch-users-bounces(a)lists.freeswitch.org> on behalf of Sergey Safarov <s.safarov(a)gmail.com> *Reply-To: *FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> *Date: *Sunday, September 11, 2016 at 9:34 PM *To: *FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> *Subject: *Re: [Freeswitch-users] Hardware requirement for 1000 Calls пн, 12 сент. 2016 г. в 3:37, Stanislav Sinyagin <ssinyagin(a)gmail.com>: Actually Ken, the MIT license allows you to do almost anything you want with modified code. There's no obligation to publish the changes. Even the Gnu GPL doesn't obligate to publish the changes, it only allows anyone to publish them for you )) Hello Stanislav Please look at term "Made available". This term requires "Any Modification which You create or to which You contribute must be made available in Source Code ... to anyone to whom you made an Executable version available". http://website-archive.mozilla.org/www.mozilla.org/mpl/MPL/1.0/annotated/ https://www.mozilla.org/en-US/MPL/1.1/annotated/ https://www.mozilla.org/en-US/MPL/2.0/FAQ/ _________________________________________________________________________ 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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test
by Shawn Wheeler 11 Sep '16

11 Sep '16

I am sorry for this test, I have sent two other notes to the listserv that have not been processed today.

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Multiple FreeSWITCH Servers
by Don Hawkins 10 Sep '16

10 Sep '16

In the future I can see our traffic load increasing to the point where I'll want to have more than one FS server. For example, if server 1 is already handling x amount of calls send traffic to server 2. I currently have FS hosted in the cloud so duplicating the entire "box" won't be a problem, so that means authentication won't be a problem. Planning for the future and I'm just curious what are some of the best ways to make this happen?

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Mad Boss Group Call
by Shawn Wheeler 10 Sep '16

10 Sep '16

I have what I think should be an easy question, I am just too ignorant to know where to look. I am using the Madboss to initiate group calls. When the group call is initiated from my handset, I hear a tone. I also hear a tone from my receiving SIP devices. I have been able to change and or mute the tone on the receiving SIP devices. I would like to be able to change the tone on my handset, where the call is initiated. I have looked in the conf\autoload_confs\confernece.conf.xml file. I see enter and exit tones for both wide and ultawide band. However on my system I only hear the entry tone. If you have a thought were I can look, please let me know. Shawn

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Hardware requirement for 1000 Calls
by Uday kumar 10 Sep '16

10 Sep '16

Hi All, I need your help to setup Freeswitch server for 1000 concurrent call on CentOS. Freeswitch version 1.4.26 for below modules:- *1.* Conference *2.* CallCenter:- In CallCenter Module will use ring group, queue etc. *3.* Extension to extension Calls, Extention to Mobile/Landline call using SIP. *4.* IVR *5.* Recording:- Record all Calls (Conference, CallCenter, Extension, IVR etc) *6.* To tracking RTP, use tool Voipmonitor. *7. *For security iptables. *8.* Database is MySQL. To make outgoing and incoming calls will configure multiple Carrier provider and DID. *Please suggest server configuration for above setup and also how much Internet speed is required. * *Currently I have one dedicated server in cloud with below configuration. * *Intel(R) Xeon ** @ 3.30GHz* *12GB RAM* *8MB Casche* *500GB HDD with 7k RPM* *Please suggest if this is capable to handle above requirements. * -- Thanks & Regard Uday Site:- www.shareyourknowledge.in Mobile:- +91-9377579349

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Mad Boss Group Call
by Shawn Wheeler 10 Sep '16

10 Sep '16

I have what I think should be an easy question, I am just too ignorant to know where to look. I am using the Madboss to initiate group calls. When the group call is initiated from my handset, I hear a tone. I also hear a tone from my receiving SIP devices. I have been able to change and or mute the tone on the receiving SIP devices. I would like to be able to change the tone on my handset, where the call is initiated. I have looked in the conf\autoload_confs\confernece.conf.xml file. I see enter and exit tones for both wide and ultawide band. However on my system I only hear the entry tone. If you have a thought were I can look, please let me know. Shawn

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Conference Call drops
by Deepika Yadav 10 Sep '16

10 Sep '16

Hi , I am using conference call service of Freeswitch over ESL with 21- 22 members over GSM for around 1 hour. One problem that I am facing is frequent call drops of the connected callers. My program has a logic of redialing a caller in case of disconnection, if the caller is not reachable due to some reason the redial happens three times. I am maintaining sufficient balance to keep 22 channels in parallel and also for the duration of calls. One thing that I am observing which I am not sure to be the cause is when I increased the redial trial to five times, the frequency of call drops increased. Is the dialing attempt count should be taken care for the balance maintenance or there is some other cause ? The Freeswitch logs are of type - CALL REJECTED, USER BUSY Regards, Deepika

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FreeSWITCH + Verto + Mobile WebRTC
by Joel Serrano 10 Sep '16

10 Sep '16

Hi, Has anyone successfully accessed FS WebRTC from a mobile device (android / iOS) using Verto? We would like to try to access a conference from mobile devices without using SIP at all. Any suggestions? Thanks! Joel.

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