September 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

fifo timeout?
by Tim St. Pierre 29 Sep '11

29 Sep '11

Hello, I'm building a very simple queue system with mod_fifo. For the most part, things are working the way I want, but I can't figure out how to limit the amount of time someone can sit in the queue. I'm trying to implement something like max_wait would do in mod_callcenter. I can use the orbit_exten option, but it wants to transfer to a different extension. I really just want the fifo application to exit so the caller can progress to the next dialplan action. Is there a way to make the caller exit the fifo if they are unanswered after a set amount of time? I did give mod_callcenter a try, but I had a lot of problems with calls getting stuck on hold, with no agents ringing, so I went to mod_fifo as a simpler solution. Thanks for any suggestions. -Tim

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Hardware phones that do 48Khz Celt: Was Re: Is there any license G729?
by Craig Askings 29 Sep '11

29 Sep '11

Are there any hardware phones that support 48 Khz Celt and automated/mass deployment? Craig. 2009/6/30 Jason White <jason(a)jasonjgw.net>: > Brian West <brian(a)freeswitch.org> wrote: >> Everyone has this need for lower bandwidth calls... I tend to march the >> other way. 48kHz baby! (btw you can do 48kHz in the same bandwidth as a >> single ulaw call) > > 48khz Celt (celt@48000 in your codec preferences) sounds wonderful with > FreeSWITCH. To test, run two FreeSWITCH instances, both with mod_portaudio. > This also works well in 48khz conferences. > > I wouldn't use G.729 even if it weren't encumbered by patents - it's G.711, > G.722, G.722.1 and (my current favourite) Celt all the way. > > > _______________________________________________ > 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 > -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig(a)overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365

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FSGUI
by Brad Mina 29 Sep '11

29 Sep '11

I've stumbled across fsgui a few times, ( http://wiki.freeswitch.org/wiki/Fsgui ), and am happy to report the windows release of this program works flawlessly! After toying with the windows version myself and a few other community members wanted to see this application running in linux as well. I downloaded the fsgui.tar.gz and unzipped it to /tmp then installed the following packages on Ubuntu 10.04, which also installed related dependencies: libqt4-dev libqt4-core I tried to chmod+x and execute the application with ./fsgui - however bash reports the following: fsgui: cannot execute binary file Is there something I'm missing here? I'm hoping jmesquita or a fellow community member can respond with insight. Thanks in advance.

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mod_conference.c:6979 Open of conference.conf failed
by Henrik Aagaard Sørensen 29 Sep '11

29 Sep '11

I've installed FS via http://wiki.freeswitch.org/wiki/Linux_Quick_Install_Guide Everything seems to run just fine, but when starting up I get the error: 2011-09-27 14:03:16.336176 [ERR] mod_conference.c:6979 Open of conference.conf failed Can anyone help me out? I guess I don't have a conference.conf(.xml) file. Should I? Should it be installed via the guide or...?

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mod_rtmp and flex client
by David Wafula 28 Sep '11

28 Sep '11

Am finally trying out flex client with mod_rtmp and i just realize how little i know of this module. I have installed mod_rtmp and loaded it (or so i think). On the server, this is what i get: freeswitch@internal>rtmp status -ERR no reply This doesn't look right for starters..what do i expect for a valid reply? To the client. It looks like the basic config on the flex client is to update var flashvars = { xxxxxxxx} to point to my freeswitch server, no ? So, first i try: var flashvars = { rtmp_url: 'rtmp://conference.freeswitch.org/phone' }; And it does connect. Say my freeswitch server is running at 1.2.3.4 So, naturally i end up with var flashvars = { rtmp_url: 'rtmp://1.2.3.4' }; Is that the way to do it? Because this gives me: netStatus: NetConnection.Connect.InvalidApp netStatus: NetConnection.Connect.Closed Any pointers will be appreciated. Thank you -- David Wafula

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SIP client with H264
by Madovsky 28 Sep '11

28 Sep '11

Hi all, is anyone know which free SIP client is the best to test H264 video ? thanks Franck

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Alarm System Central Office over VoIP (FreeSwitch)
by Kubes 28 Sep '11

28 Sep '11

All, I am trying to get an alarm system which calls a central monitoring center to work via FreeSwitch. I am using a Linksys SPA-1001 and a SIP trunk (Vitelity). Even though the call is answered the "modems" never connect, from what I can tell. I see the call being answered in Freeswtich, but the alarm system reports a "comm error". PCMU is being used for the call and it does hang-up with "NORMAL_CLEARING". Does anyone have any ideas how to get "modems" to work over VoIP? Any suggestions would be appreciated. Thanks Kubes

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FreeSWITCH Conference Call Today
by Michael Collins 28 Sep '11

28 Sep '11

Hello folks! We have a few items to discuss today. Here's the agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2011_09_28 We have a few updates to share with everyone and we also would like to hear from you if you have been using mod_sms that just came out last week. Talk to you soon, Michael

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fifo count doesn't indicate the number of agents.
by Yungwei Chen 27 Sep '11

27 Sep '11

Hi, I defined a queue called support in fifo.conf.xml and added an agent to the queue as show below. I am able to send a call to the queue first and then take the call to the agent 1004. However, "fifo count support" always return "support:0:0:0" when an agent is available and nothing in the queue. The same command returns "support:0:1:1" when an agent is available and one call in the queue. What am I missing here? Thanks. <configuration name="fifo.conf" description="FIFO Configuration"> <settings> <param name="delete-all-outbound-member-on-startup" value="false"/> </settings> <fifos> <fifo name="support" importance="0"> <member timeout="60" simo="1" lag="10">{member_wait=nowait}sofia/user/1004(a)192.168.6.8</member> </fifo> </fifos> </configuration>

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Change voicemail, example code available?
by Henrik Aagaard Sørensen 27 Sep '11

27 Sep '11

I would like to change how the voicemail acts in FreeSwitch. But as I've figured out it's hardcoded into Mod_voicemail. So I would like to make my own via an IVR. Does anyone know any sample code for a simple voicemail, that I can use for reference? One thing I would like to change is for example that it shouldn't hang up on wrong ID's, but give one 3 tries to enter a correct ID.

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