- FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH Community Conference Call Today
by Michael Collins 27 Mar '13

27 Mar '13

Hello all, Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_27 We have a few quick items to discuss and then we will have Mark Crane give us an update on FusionPBX. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Different ring_back tone if line 2 line
by Miha 27 Mar '13

27 Mar '13

Hi, I need a little help regarding one scenario. A calls B, B answers a call. Then C calls B and it gets ringing, I need for this second calls who is waiting to have different ringback tone as first one how gets answer from B. How to implelment that. Tnx. Miha

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Weekly FreeSWITCH Community Conference Call
by Ken Rice 27 Mar '13

27 Mar '13

Hey Guys, Don¹t forget, FS Weekly Community Conf Call today at 1PM EST (GMT-4). Join us http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_27 K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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It's been almost 3 years and things have changed...
by Dave R. Kompel 27 Mar '13

27 Mar '13

... since I did the demo of writting code for MOD_MANAGED in C#. There is still a recorded session on the WIKI on the MOD_MANAGED page, but it is very out of date. If you are all interested, I can do another one, more up to date, using tools that didn't exist at the time to build a custom FreeSWITCH module to handle Directory, Dialplan, DP Apps, and API, using custom datasources. I would like to do this out of band from a normal Wensday public meeting, so that we can have just the people that are interested, and be more interactive. If there is interest please respond, so I can get a head count, and then get with the people offline to come up with a presentation time. The new presentation will be titled: "Building a custom FreeSWITCH solution with Visual Studio, that can target both Windows and Linux, in less then an hour". Again if you are interested, let me know and we can plan the event. --Dave

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Testing, testing....1..2..3
by Brian Foster 27 Mar '13

27 Mar '13

<knock knock> This thing on..? Thank you, Brian Foster Project Manager/Owner's Representative Davri Investments, Incorporated P: +1-317-787-2686 M: +1-317-600-9753 Indianapolis, Indiana

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skypopen- Insmod ./skypopen.ko
by Anouar Abrik 27 Mar '13

27 Mar '13

Hi Do you suggest I reinstall the server distro completely? Or just remove the desktop on tty1, will that do? P.S sorry for mailing you direclty and on the wrong mailing list and thanks alot for the reply. -- Regards, Anouar Abrik *Graphics Designer* Direct Number: +92-336-229-4557 Skype: donnymaniac * * * *

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Inband DTMF
by Nathan Sullivan 27 Mar '13

27 Mar '13

Hey Guys, I am just dealing with a SIP device that only sends and receives DTMF as inband, and just trying to get the configuration correct. Currently we have it working fine for inbound DTMF or outbound DTMF on a single leg, but not both. For outbound calls (from us), start_dtmf_generate is how we do it now, i believe that only generates inband events to send to the B party when it receives RFC2833/INFO from the A party, correct? For inbound calls (to us), we go start_dtmf to listen for inband DTMF from them on the A party side and send RFC2833/INFO out the B party side, correct? Theres a comment on http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf about not using the two together, as they cause an audio loop inside FS. Is there any other way to get bidirectional DTMF working there...? Regards, Nathan.

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still Waiting for a queue call while outbound
by Jimmy Chang 27 Mar '13

27 Mar '13

Hi, We have a agent login to a callcenter and set available for the queue call. While the agent makes a outbound call, why does the queue call still dispatch to the agent? Does that mean we should set the agent logout the queue while making a call? Thanks. Jimmy

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setting an alternative sip server
by Philippe Le Toquin 27 Mar '13

27 Mar '13

Hello, I have read the wiki and asked google but could not find what I am looking for. I use freephoneline.ca as gateway and recently they have had several outage where either or both in and outgoing stop working Their recommendation is to configure an alternative sip server as apparently one will always be working. On the page below they explain how their box are configure but I can't work out how to transfer it to a freeswitch config http://support.freephoneline.ca/entries/23120323-VoIP-Unlock-Key-Credentials Is it possible? /Philippe

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Error Creating SIP UA for profile: internal on OS X
by Mario M Guzman 26 Mar '13

26 Mar '13

New issue since migrating to Mountain Lion, I verified its port 5060 that is the issue. But it's not as simple as you may think: netstat, lsof, and the stroke utility, all show clear ports and yet freeswitch could not start the internal profile. But then… wait 45-60 seconds before restarting freeswitch and it works. Works on 10.6 and 10.7, only an issue on 10.8. I have a script http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X#Create_the_F… that uses netstat to test for sip on tcp and udp, It tests all things freeswitch uses, it starts freeswitch when all is clear. The script displays the ports (netstat) on terminal (if using terminal) and the console in case of this type of issue. I verified that before freeswitch is started netstat showed the exact same ports as after a reboot but before freeswitch was started. All was clear and yet internal won't start. I tried a bunch of stuff: Anyone with ideas/suggestions? I may have to contact Apple with a bug but would like a workaround first. Driven Nuts Again, Mario G

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