January 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch mod_amqp close connection to RabbitMQ
by Pedro Seabra 30 Jan '17

30 Jan '17

Hello, It would be much appreciated to have some advice on following case. I use mod_amqp to send freeswitch's events to RabbitMQ, from times to times, mod_amqp closes connection to RabbitMQ and this results in loss of that specific event. This issue happens with higher and lower traffic, it's completely random. mod_amqp logs: [CRIT] mod_amqp_producer.c:433 Profile[default] failed to send event on connection[primary]: (unknown error) [CRIT] mod_amqp_utils.c:52 Closing channel: a socket error occurred [CRIT] mod_amqp_utils.c:52 Closing connection: a socket error occurred [DEBUG] mod_amqp_producer.c:501 Send failed with 'socket error' [WARNING] mod_amqp_producer.c:456 Amqp no connection- reconnecting... [DEBUG] mod_amqp_connection.c:119 Profile[default] trying to connect to AMQP broker 10.18.43.80:5672 [DEBUG] mod_amqp_connection.c:136 Profile[default] opened socket connection to AMQP broker 10.18.43.80:5672 INFO] mod_amqp_producer.c:469 Amqp reconnect successful- connected I changed the config of parameter "heartbeat", with higher and lower value, without any success resolving this issue(i was using default configuration). mod_amqp closes connection and reconnect automatically, the next events are sent to RabbitMQ, just the event at that moment is lost. I don't have/detect any network problems in this environment. Cumprimentos | Best Regards Pedro Seabra Research & Development Phone: +351 256 370 980 / Ext. 334 Email: pedro.seabra(a)itcenter.com.pt <http://www.itcenter.com.pt/> <http://www.itcenter.com.pt/> <http://www.itcenter.com.pt/> www.itcenter.com.pt [image: ITCENTER Store] <http://store.itcenter.com.pt/> [image: ITCENTER Helpdesk] <http://support.itcenter.com.pt/otrs/customer.pl> [image: ITCENTER Facebook] <https://www.facebook.com/itcenterpt> [image: ITCENTER Linkedin] <https://www.linkedin.com/company/itcenter> [image: ITCENTER Twitter] <https://twitter.com/ITCenter_PT>

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Re: [Freeswitch-users] G729 Now Royalty Free, now what?
by Ahmed habiba 30 Jan '17

30 Jan '17

Does this mean that we can use the free versions without caring about license issues: i.e. “big729” freeswitch module http://www.linphone.org/technical-corner/bcg729/downloads <http://www.linphone.org/technical-corner/bcg729/downloads> http://romon-bd.blogspot.com/2014/03/freeswitch-g729-module-using-opensourc… <http://romon-bd.blogspot.com/2014/03/freeswitch-g729-module-using-opensourc…> “ bcg729 is dual licensed. It can be licensed and distributed: under GNU GPLv2 <http://www.gnu.org/licenses/gpl-2.0.html> license - for free (open source) under proprietary and commercial license to be used in closed source applications. Contact Belledonne Communications <http://www.belledonne-communications.com/> for costs and other service information. " > From: Brian West <brian(a)freeswitch.org> > Subject: Re: [Freeswitch-users] G729 Now Royalty Free, now what? > Date: January 30, 2017 at 3:56:16 PM GMT+3 > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Doesn't mean the implementation is free, also careful with that assesment there are a few land mines in there you'll have to dodge! > > I still have to file quarterly repoerts with sipro! > > :) > > /b > On Mon, Jan 30, 2017 at 6:41 AM Michael Toop <michaelt(a)mdevt.com <mailto:michaelt@mdevt.com>> wrote: > 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 <http://www.sipro.com/G729.html> > > Cheers, > > Michael > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> > > Official FreeSWITCH Sites > http://www.freeswitch.org <http://www.freeswitch.org/> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/> > http://www.cluecon.com <http://www.cluecon.com/> > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> > http://www.freeswitch.org <http://www.freeswitch.org/> > -- > Brian West > brian(a)freeswitch.org <mailto:brian@freeswitch.org> > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com <http://www.freeswitchbook.com/> > http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/> > https://www.gofundme.com/freeswitch_ubuntu <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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Influencing SIP message fields on outgoing calls through gateway
by Mohit Gupta 30 Jan '17

30 Jan '17

Hello - Can someone explain me how can I configure a gateway so that it sets/unsets full or part of the SIP message fields specifically RURI, TO, FROM, CONTACT and USER AGENT dynamically for each call 'bridged' to this gateway. Also the SIP OPTIONS ping to use some statically configured values? I can play with the params/variables for leg A (incoming) but unable to do the same for leg B (outgoing). Exporting or static config aren't helping much. One example is if I like to use a SIP domain name e.g. sipdev.example.com in SIP messages e.g. RURI of SIP OPTIONS. But sipdev.example.com cannot be used for DNS lookup. It is common across multiple instances and I would like send the calls to specific instance based on called number. If I set this in Gateway param "proxy", sofia tries to perform DNS lookup before sending out SIP OPTIONS. I am unable to find a way so that sofia takes the IP to send the message to but uses the configured domain to format the SIP messages. Is it possible by configuration or does it need some source modification? Regards, Mohit

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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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