- FreeSWITCH-users - FreeSWITCH Mailing Lists

old time of day syntax request
by Sean Devoy 30 Jan '17

30 Jan '17

Hi all, The new syntax is date function is EXCELLENT. However, I have a customer who will not allow a FreeSwicth upgrade since "it does everything we and we have experienced no problems." So, I am stuck at v1.4 :( Can someone send me the OLD syntax for time of day dialing? Hours? Day of the week? Thanks, Sean

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G729 Now Royalty Free, now what?
by Michael Toop 30 Jan '17

30 Jan '17

Hi, I see G729 is now royalty free. Now what, has someone implemented this in a free way for FS or has any suggestions how to get his working? I would rather sponsor beer & lunch than pay for licenses :) http://www.sipro.com/G729.html Cheers, Michael

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FreeSWITCH 1.8 and you!
by Brian West 30 Jan '17

30 Jan '17

Dear FreeSWITCHers, With the release date of 1.8 fast approaching and our annual developer meeting is all next week. I have to ask that you the FreeSWITCH community to step up and help us next week. I thank everyone that has already contributed, but we've fallen short of our goal. I would like to remind everyone how much time and effort it takes to mange and maintain this project. We the core team have spent countless hours to bringing you FreeSWITCH at the low low price of $0.00 and we are working diligently to keep it that way. Unfortunately the power company and grocery stores look at me funny when I try to pay with a diff file! Comedy aside this is your chance to show some appreciation to the team, I personally have to thank Anthony for giving me a chance to work with him way back when my age still started with the number 2... I've grown and learned more than I could have ever dreamed in this field and managing an open source some can a very thankless job and at times grinds you down! Then Anthony commits the latest feature or tweak and the passion starts all over! This is how you can donate to our engineering meeting next week: http://www.gofundme.com/freeswitch-developer-meeting-2017 We have hotels, plane tickets and lunch and dinners to pay for! The result of all of this you ask? FreeSWITCH 1.8, and after that just think we'll be at FreeSWITCH 2.0! I was here when the folder was empty! We have more in store for our users so stay tuned. And remeber 🐜 go in JIRA! Thanks, /b -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest

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lua EventConsumer
by royj＠yandex.ru 30 Jan '17

30 Jan '17

Hi Do I understand correctly that to EventConsumer() one can set only one event name. It could be "all", "CHANNEL_HANGUP" and others. But not, for example, "CHANNEL_HANGUP" and "CHANNEL_STATE" with some separator. And if you want two events you should set "all" and filter them in the script. Right?

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send_info is not supported
by royj＠yandex.ru 30 Jan '17

30 Jan '17

Hi I am trying to utilize uuid_send_info, but face some unclear behaviour for me. I run in fs_cli like "uuid_send_info <uuid> <message>" and get "mod_sofia.c:1709 <channel name here> send_info is not supported". There is channel var fs_send_unsupported_info true in that channel. Is anybody used this functionality, am I missing something?

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Re: [Freeswitch-users] FreeSWITCH on VirtualBox - no sound using Chromium + XML RPC
by Rövşən Bağırov 29 Jan '17

29 Jan '17

Just tried with jssip and it also did not work. Tried with sipML5 and it works! 29.01.2017, 08:52, "Rövşən Bağırov" <rovshanb(a)yandex.ru>: > Chromium + XML RPC => Chromium + Web RTC > > 29.01.2017, 08:45, "Rövşən Bağırov" <rovshanb(a)yandex.ru>: >> Hi, >> >> I've installed FreeSWITCH inside VirtualBox. Below I list versions of software that I use: >> >> Host OS: Linux Mint 18 Sarah >> Host IP: 192.168.8.100 >> Guest OS: Debian GNU/Linux 8 (jessie) >> Guest IP: 192.168.8.101 >> FreeSWITCH: FreeSWITCH (Version 1.6.14 -23-e460bf8 64bit) >> Chromium: Version 55.0.2883.87 >> in /etc/hosts: 192.168.8.101 c1.flexy.az >> >> I configured FreeSWITCH to be multi-tenant and serve configuration through xml curl module. >> When using Linphone to connect to FreeSWITCH everything works perfectly. Now I try to use Chromium + XML RPC using SIP JS javascript library but there is no sound. On the FreeSWITCH side call procedes normally but on the client side there is no sound. I have searched online but could not find any solution that resolves the problem. >> >> FreeSWITCH log is here (top parts of the console got out of buffer, but it is understandable what is going on): http://pastebin.com/HK70AEnX >> Chromium log: http://pastebin.com/7n763mEd >> Wireshark: http://filebin.ca/3AW3of2XdBLq >> >> Any help is very highly appreciated. >> Best Regards, >> Rovshan

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FreeSWITCH on VirtualBox - no sound using Chromium + XML RPC
by Rövşən Bağırov 29 Jan '17

29 Jan '17

Hi, I've installed FreeSWITCH inside VirtualBox. Below I list versions of software that I use: Host OS: Linux Mint 18 Sarah Host IP: 192.168.8.100 Guest OS: Debian GNU/Linux 8 (jessie) Guest IP: 192.168.8.101 FreeSWITCH: FreeSWITCH (Version 1.6.14 -23-e460bf8 64bit) Chromium: Version 55.0.2883.87 in /etc/hosts: 192.168.8.101 c1.flexy.az I configured FreeSWITCH to be multi-tenant and serve configuration through xml curl module. When using Linphone to connect to FreeSWITCH everything works perfectly. Now I try to use Chromium + XML RPC using SIP JS javascript library but there is no sound. On the FreeSWITCH side call procedes normally but on the client side there is no sound. I have searched online but could not find any solution that resolves the problem. FreeSWITCH log is here (top parts of the console got out of buffer, but it is understandable what is going on): http://pastebin.com/HK70AEnX Chromium log: http://pastebin.com/7n763mEd Wireshark: http://filebin.ca/3AW3of2XdBLq Any help is very highly appreciated. Best Regards, Rovshan

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ICTFax Version 3.7.4 released
by Tahir Almas 28 Jan '17

28 Jan '17

Pleased to update about the release of ICTFax Version 3.7.4 using freeswitch based ICTCore communication framework http:/ictcore.org Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT

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Dial Plan tidbit for a HOTLINE automatic dial
by Sean Devoy 27 Jan '17

27 Jan '17

I have never posted here with information, only enquiries. Based on a posting I made a few years ago, no one had a solution "A HOTLINE" that dials automatically. Essentially, I learned FS gets no events for a phone going off hook so it is really a phone function not a switch function (on VOIP devices). I have recently discovered a dial plan to achieve it. It may be obvious to many or have been posted here before, but I don't see it in any quick searches. Cisco/Linksys phones and ATAs conform to the standards specified by MGCP and MEGACO. I suspect most venders do, so, this is likely to work for most everyone. There is a dial plan feature "<: xxx >" that allows you to insert additional digits when sent to the switch but not be required in the dial plan match. Seven digit dialing in the U.S. is a good example. To insert 1 + area code (410 for example), use "<:1410>[2-9]xxxxxx S0" Note also the S0 which will send to the switch immediately if matched. Extending that concept to somewhat of an extreme "S0 <:18005551212>" will dial that inserted number as soon as the phone is taken off hook, Another option is to extend the timeout to allow the user to dial, but dial the hot line if the user does nothing. "S3 <:18005551212>" I found it interesting and helpful. Hope it helps someone else to0. If it is old news, sorry for wasting your time, Sean

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should FS put/remove the call on hold on update request
by Antonio Silva 27 Jan '17

27 Jan '17

Hi all, i've seem some dialog with avaya device and i'm not sure if the behaviour is the correct from fs on update request. I have: FS <--------> remote end invite-----> <---- trying <----- 200ok <----- invite (a=sendonly) 200ok (a=recvonly) ----- > <----- update (a=sendrecv) (1)200 ok ----> <---- bye ack ----> in (1) FS do not change the call flow and it stays in hold. This is the correct behaviour? should it process the update and remove the call from hold, responding wih 200 ok (a=sendrecv) ? I could file a jira if you think is a bad behaviour.. Normally, instead of the update would expect an invite (a=sendrecv)... so i'm pretty sure that it could be a bug in the avaya device. Thanks, António

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