- FreeSWITCH-users - FreeSWITCH Mailing Lists

listing moderators in a conference
by Aaron Seelye 11 Sep '13

11 Sep '13

Hi all, I'm trying to find how to list the moderators in a conference. There are plenty of API calls that indicate mod_conference knows who is a mod and who is a user, but I'm not seeing any way to show them on the command line. Is there any easy (or hard) way to do this? TIA, -Aaron

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Re: [Freeswitch-users] Best way to reload modfifo
by Chris B. Ware 11 Sep '13

11 Sep '13

Try "fifo reparse". ------------------------------ Date: Wed, 11 Sep 2013 13:43:38 +0200 From: Antonio Silva <asilva(a)wirelessmundi.com> Subject: [Freeswitch-users] Best way to reload modfifo To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Message-ID: <1378899818.22099.27.camel(a)marces.madrid.commsmundi.com> Content-Type: text/plain; charset="utf-8" Hi, What is the best way to reload mod fifo configuration? Right now i'm using "reload mod_fifo". Thanks, Ant?nio

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Debian repository out of date
by Steven Ayre 11 Sep '13

11 Sep '13

Who maintains the files.freeswitch.org/repo/deb/ Debian repository? It still only has 1.2.10 as the latest release.

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Best way to reload modfifo
by Antonio Silva 11 Sep '13

11 Sep '13

Hi, What is the best way to reload mod fifo configuration? Right now i'm using "reload mod_fifo". Thanks, António

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[ANN] RockFS 0.01 Alpha
by Richard Mace 11 Sep '13

11 Sep '13

Hi all, I am putting together a FS GUI app (currently Windows) which is ready for a feature test. If you have a fresh Debian 7 install, then it should turn it into a FS server for you. Feel free to download each of the .exe files from the location below, put them in the same folder, and then run RockFS.exe Let me know if it works and any thoughts? www.rocksolidvoip.co.uk/downloads Richard

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Options to run FreeSwitch on non-X86 devices?
by Fred-145 11 Sep '13

11 Sep '13

Hello I searched the wiki and Google , but found only a couple of mentions about compiling/running FreeSwitch on ARM processors, including this one: http://lists.freeswitch.org/pipermail/freeswitch-users/2012-May/083996.html Some people run FS on the following non-X86 devices: - Seagate DockStar - Pogoplug - SheevaPlug - Raspberry Pi (SwitchPi) Are there other non-X86 recommended devices to run FS from SOHO use? Thank you.

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Sangoma FreeTDM in NET mode immediatly restarts interface
by kietlak 11 Sep '13

11 Sep '13

I try to make external IVR on E1 Siemens Hicom Trunk using Sangoma A102. E1 port connected to Telco works without any problem. E1 port connected to Siemens doesn't start because is restarted as below. Where I have to look for the reason (and solution) of this problem ? No. Time Source Destination Protocol Info 1 0.000000 Remote User Local Network LAPD U P, func=SABME 2 0.000528 Local Network Remote User LAPD U F, func=UA 3 0.000717 Local Network Remote User Q.931 RESTART 4 0.099999 Remote User Local Network LAPD S, func=RR, N(R)=1 5 0.150849 Remote User Local Network Q.931 RESTART ACKNOWLEDGE 6 0.151372 Local Network Remote User LAPD S, func=RR, N(R)=1 7 2.997662 Local Network Remote User Q.931 RESTART 8 3.091613 Remote User Local Network Q.931 RESTART ACKNOWLEDGE 9 3.092027 Local Network Remote User LAPD S, func=RR, N(R)=2 10 6.997654 Local Network Remote User Q.931 RESTART 11 7.100010 Remote User Local Network LAPD S, func=RR, N(R)=3 12 7.113235 Remote User Local Network Q.931 RESTART ACKNOWLEDGE 13 7.113565 Local Network Remote User LAPD S, func=RR, N(R)=3 14 10.997071 Local Network Remote User Q.931 RESTART

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RTP Over VPN Split Tunnel
by Peter 11 Sep '13

11 Sep '13

This is not a direct FreeSWITCH question, but was hoping someone might have set up a similar configuration. Currently I have a FS server where SIP clients (Linphone) register to via a Cisco AnyConnect VPN. I have inbound-bypass-media set to true so that RTP traffic is peer to peer. If I tunnel all traffic, the SIP transport and RTP media works just fine. If I set up a VPN split tunnel that allows the subnets of the VPN client and the FS server, the SIP signalling is fine, but in the SDP, it specifies a non VPN address, eg: INVITE sips:1004@10.1.1.204;transport=udp SIP/2.0 Via: SIP/2.0 10.1.254.36:53036;branch=z9hG4bK.GRrzYXeCN;rport From: <sip:1001@10.1.1.204>;tag=cN83ZcmHw To: sip:1004@10.1.1.204 CSeq: 20 INVITE Call-ID: e7eEYxoML2 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Content-Length: 426 User-Agent: LinphoneAndroid/2.1.2-177-g015b83c (belle-sip/0.0.1) Contact: <sip:1001@10.1.254.36:53036 ;transport=udp>;+sip.instance="<urn:uuid:83c4b823-9c0d-4e02-b7f7-cdd3ae2f1259>" v=0 o=1001 3921 1865 IN IP4 [ISP address] s=Talk c=IN IP4 [ISP address] b=AS:380 t=0 0 m=audio 1031 RTP/SAVP 0 8 100 101 a=rtpmap:100 iLBC/8000 a=fmtp:100 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:7079 IN IP4 [wireless IP address] I'm not very familiar with how the SDP is constructed, but I'm assuming that it's the SIP client that's responsible for offering it's IP addresses. Are there any FreeSWITCH settings that would affect the IP addresses in the SDP? Thanks Peter

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This week in FreeSWITCH
by Ken Rice 10 Sep '13

10 Sep '13

Hey Guys, Things of note this week in FreeSWITCH Tomorrow Wed Sep 11th, 1PM Eastern Time, join us for ClueCon Weekly. This weeks topic Security! Join us and join in on the conversation! https://wiki.freeswitch.org/wiki/FS_weekly_2013_09_11 If you have a topic you would like to see covered on ClueCon Weekly, Let us Know! What to come on and present something? We have calendar dates open! Email Me! We¹ll continue to video and broadcast to youtube, as well as adding a ton of new features to the weekly call! And for those that havent noticed, our own Mr Brian West is getting married! Brian and Gregg will be tying the knot in NYC on October the 7th. Want to send him a little something to help with the wedding and a thanks for all the help over the years, paypal to brian(a)bkw.org and leave him a note in there. Don¹t forget to follow up on your Jira Tickets! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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mod_gsmopen: call connects, silence, lose /dev/ttyUSB2...
by Jayanth Acharya 10 Sep '13

10 Sep '13

Recently I replaced my Huawei E1550 with a Huawei E173-U2 modem. Both modems are completely unlocked (and voice enabled), with latest firmware files, as per dc-files.com site. While with Huawei E1550, I was having this terrible voice-quality issue, with Huawei E173-U2, I fared much better initially (last night), and my first call went through perfectly, and I got connected to the IVR menu "5000". However, since I was facing signal-quality issues, I used AT^SYSCFG to set the modem to 2G mode, and got excellent signal. Apart from that, I don't remember making other changes, but since this morning, I am getting this 100% repeatable problem, where-in, once I call FS via my mobile, I see the usb-modem attached to FS (gsmopen EP) ringing, and get connected, but I do not hear the IVR menu anymore, but complete silence. After a while the call disconnects, and then I see console log indicating that my /dev/ttyUSB2 (control port) is probably dead, and as a result the /dev/ttyUSB1 and /dev/ttyUSB2 are removed, and /dev/ttyUSB3 and /dev/ttyUSB4 are created. If I reconfigure gsmopen.conf.xml file to use the new audio and control ports, and place another call, the same thing repeats. I've attempted this across several reboots, but is 100% repeatable. What I see in the console log: http://pastebin.com/swnz4sWf The relevant portion of the freeswitch.log file: http://pastebin.com/dNkhaGen Any ideas / clues as to what's up ? I tried going through the log file, and what I figured is that nothing seems to be received by FS on the audio port, and eventually /dev/ttyUSB2 becomes unresponsive, i.e. doesn't respond to AT-commands. Not sure though. - Jay

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