- FreeSWITCH-users - FreeSWITCH Mailing Lists

Anyone running FS 1.6 + Video in Google Cloud (or AWS)?
by Craig Stevenson 04 Jun '15

04 Jun '15

Has anyone brought up FS 1.6 + video on a Google Cloud instance? Any tips or tricks that I need to know? Alternatively, what about instances on the AWS EC2 cloud?

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Help required regarding billing of call via Dial Httapi command
by Manish Talwar 04 Jun '15

04 Jun '15

Hi, We are successfully integrated FreeSwitch and Nexmo in our IVR application. We have also tested International calls by calling mobile number by "Dial" command of FreeSwitch via Nexmo. Its working fine now but we are unable to get calls details (like call start time, call end time, bill amount, and so on) for the particular request. We need to charge our customer for the calls he make from our IVR. We need to deduct the amount from his account at the same time after disconnecting the call. When user disconnect the call, I got the request back from Nexmo to freeswtich but I am not able to get all required details. Please let me know, how can I get this call details ( via FreeSwitch) at the same time after disconnecting the call. Thanks, Regards, Manish Talwar

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One way Jitter Buffer
by Kamil Nigmatullin 03 Jun '15

03 Jun '15

Dear All, We have developed our own sip application that works with Freeswitch. So, on incoming rtp stream we can ajust jitter buffer on client side, but on outgoing stream I want freeswitch to manage Jitter Buffer. Is there a way to switch Jitter Buffer only for outgoing stream in Freeswitch Thanks -- Kamil Nigmatullin Tel: 77272323748 mob: 7 (707) 2517003 Skype: kamil.nigmatullin

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Eavesdrop to existing UUID
by Allan Kristensen 03 Jun '15

03 Jun '15

Hello All, Can anyone tell me if it's possible to eavesdrop from an existing channel instead of using "originate <dialstr> &eavesdrop(uuid)" method? Let's say I have 20 agents talking to our clients and as I supervisor I want to listen to these calls and perhaps coach agents by activating the "speak with uuid" feature available in eavesdrop. Today I need to make a new call (originate) if I want to start eavesdropping, so supervisor has to disconnect phone and make another call to jump to next agent (using the "*" DTMF is not possible and I also want to control this using ESL). What i'm doing today using the ESL: > originate <agent dial string> &park > originate <client dial string> &park > uuid_bridge <agent_chan_uuid> <client_chan_uuid> ..."Agent and client talks"... Now supervisor/coach wants to give agents some tips: > originate <supervisor dial string> &eavesdrop(<agent_chan_uuid>) "Supervisor can now listen and talk to agent by pressing the "2" DTMF If supervisor needs to help another agent, he/she now needs to disconnect call and do another originate, which is quite tedious. If it was somehow possible, I rather do this: > originate <supervisor dial str> &park > "send api cmd to api that eavesdrop <client_chan_uuid> from <supervisor_chan_uuid>" ...and when done, quickly move to another client by issusing another command without disconnecting the call, perhaps even use the same <supervisor_chan_uuid> to listen to a recorded call. Does it makes sense? :) What approach would you recommend ? Thank you... Best Regards, Allan

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Early media
by Jurijs Ivolga 03 Jun '15

03 Jun '15

Hi! Sorry, issue was on mobile operator side, Early media worked inconsistently and after changing SIM it started to work without any issue. Thank you!

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Early media
by Jurijs Ivolga 03 Jun '15

03 Jun '15

Hi! I'm trying to send early media in Freeswitch, but without success. Here is my code in Lua script: session:preAnswer() session:streamFile("/tmp/prompts/vm-not_available_no_voicemail.wav") When I'm calling from my x-lite, everything works. But when I'm calling from my mobile it is not working. I'm using Voxbeam as DID provider. Same Voxbeam provider works perfectly with early media on Asterisk. Working code from Asterisk: exten => _X.,1,Ringing exten => _X.,n,Progress() exten => _X.,n,Playback(hello-world,noanswer) Any help will be highly appreciated! With kind regards, Jurijs

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FS send notification very often
by Dmitry Mordovin 03 Jun '15

03 Jun '15

Hello, I don't understand why FS send this notify every 4 seconds... Please help me to understanding.. Console: send 892 bytes to udp/[X.X.X.X]:7513 at 12:02:43.219129: ------------------------------------------------------------------------ NOTIFY sip:00294539100@X.X.X.X:7513 SIP/2.0 Via: SIP/2.0/UDP Y.Y.Y.Y;rport;branch=z9hG4bKFK33trF1tDmSr Max-Forwards: 70 From: <sip:00294539100@Y.Y.Y.Y>;tag=3HH1U3FS0360e To: <sip:00294539100@Y.Y.Y.Y> Call-ID: b0e25031-8469-1233-09ae-000255904fb8 CSeq: 76323204 NOTIFY Contact: <sip:mod_sofia@Y.Y.Y.Y:5060> User-Agent: FS Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Event: message-summary Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=noresource Content-Type: application/simple-message-summary Content-Length: 73 Messages-Waiting: no Message-Account: sip:00294539100@Y.Y.Y.Y ------------------------------------------------------------------------

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Originate + Bridge Timeouts
by Joshua Gigg 03 Jun '15

03 Jun '15

Hi, I'm trying to find some documentation on the various timeouts when using originate and bridge. Note: I don't use gateways in my application, rather targeting them directly. What I want to be able to control for each: 1. Time before receiving any SIP message back from my Invite. 2. No Answer Timeout - the amount of time before I get the 200 Ok. I want to be able control them when using both bridge + originate, regardless of early media (For originate, I disable early media, when bridging, I allow it). I've seen originate_timeout, which seems to help with point 1, but when bridging, it gets ignored as soon as I get early media. I've seen call_timeout, but other posts on the mailing list suggest it isn't used anywhere. -- Joshua Gigg

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Are you ClueCon Ready? Register Today! Call 877-74-A-CLUE
by Brian West 02 Jun '15

02 Jun '15

[image: Description: HDD:Users:anthm:Downloads:ccxx.jpg] *June 2nd 2015 Update* August 3rd – August 6th 2015 877-742-2583 • marketing(a)cluecon.com Register NOW! · $899 Staying at the Hotel · $1199 Staying Elsewhere · First 100 Gino’s PIZZA PARTY *IMPORTANT DATES* *Before June 15th* 10 Raffle Tickets Per Day 2 Gigabit Drink Coupons *Before July 1st* 5 Raffle Tickets Per Day 1 Gigabit Drink Coupons *Contact Us* https://cluecon.com marketing(a)cluecon.com *Speaker Submission Deadline* June 15th, 2015 *Suggested Speaking Topics* WebRTC Applications Interesting Open Source Apps Troubleshooting Networks VoIP Database Integration Clustering and H/A Tech and Data Security Standards Billing, Routing and Rating VoIP Infrastructure FreeSWITCH Deployments SIP TDM IVR Open Source Project Status Applied Technology Hosted API Advanced Configurations Your Latest and Greatest *Great Speakers, Hack-a-Thon, WebRTC And MUCH MORE at ClueCon 2015!* Be sure to register as soon as possible for the upcoming ClueCon 2015 Developers Conference. Not only will it give you piece of mind, the sooner you register, the more opportunities you will get to win prizes! You’ll also get more drink coupons for the Gigabit Reception *Tuesday Night*! [image: Description: mac1]The grand prize is a laser engraved commemorative FreeSWITCH 1.6 Edition dual-core 13" Retina MacBook Pro! See the Important Dates Section for Registration details! Why I Think You MUST COME To ClueCon! [image: Description: kk]Hi, I’m Kathleen. I’m the FreeSWITCH and ClueCon Social Media Correspondent. I’ve been working hard all year keeping you all up to date on what’s going on with FreeSWITCH. Today I’m here to let you know more about the upcoming ClueCon 2015 Conference! This year we are adding an optional day on Monday with an all-day Hack-A-Thon with great coding contests, game shows and kick-off fun! If you are interested in WebRTC, Voice over IP or Open Source projects like FreeSWITCH, ClueCon is the greatest opportunity you have to gain exposure to the most knowledge and technology in one place. Also, it’s the most fun you can possibly have while still getting a ton of work done! I really look forward to seeing you all there and enjoying the amazing talks, the Epic Annual Kick-Off Pizza Party, The Gigabit Reception and so much more. Make sure you register today so you can reserve your place among the attendees! Be sure to follow us on Facebook and Twitter to get my latest updates in info! [image: Inline image 1] <https://facebook.com/cluecon> [image: Inline image 2] <https://twitter.com/cluecon> It’s Not Too Late to Sign Up as a Speaker Do you have a great idea or an awesome project you have been working on? We would love to hear about it! Show off what you know! [image: Description: fss]Sign up to speak at ClueCon this year and let us know what you’ve been doing. The call for speakers page for ClueCon 2015 is now open and you can visit https://cluecon.com/speaking-proposals to submit your speaking proposal right now! [image: Inline image 3] <https://www.youtube.com/watch?v=NLaDpGQuZDA> [image: Inline image 4] <https://www.youtube.com/watch?v=9XXgW34t40s> Unsubscribe: http://lists.freeswitch.org/mailman/listinfo/freeswitch-cluecon

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Re - British Telecom (BT) SIP test list before allowing a server on their IP Telephony network
by Andrew Keil 02 Jun '15

02 Jun '15

To FreeSWITCH users, Re - British Telecom (BT) SIP test list before allowing a server on their IP Telephony network. I wondered if someone can take a quick look at this test case spreadsheet from BT in the UK to see if a standard FreeSWITCH installation (current production release) will cover off all these test cases. I assume the answer is yes! However I thought that I would ask in advance. Kind Regards, Andrew Keil

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