September 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FS community - thank you!
by Cal Leeming [Simplicity Media Ltd] 11 Sep '13

11 Sep '13

Hello all, After having to take a couple of weeks away from telephony to work on another project (not by choice!), I've had a lot of time to reflect. Today I check my emails and see discussions about ClueCon, some new faces contributing on the ML, and even wedding plans! The FS community spirit is probably one of the most genuine and welcoming I've come across in my 16 years on-hook. In the 18 months that I've been part of this community, I've not only been able to learn from other talented engineers but also had the opportunity to hear some amazing stories from some of the old school guys. I'd like to take this opportunity to say thank you to all those who have made this possible, and hope to meet many of you at next years ClueCon (assuming I can get past the border!) Next week I will finally be back at my desk doing my normal day-job, and look forward to being active in the community again after nearly 6 weeks of being off the radar! Cal

2 1

uuid_transfer help
by Usama Zaidi 11 Sep '13

11 Sep '13

Need some help with uuid_transfer api call from esl script on a dtmf event. Following is the issue I'm facing http://pastebin.freeswitch.org/21395 Thanks and regards.

2 1

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

2 1

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

3 4

Debian repository out of date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.

13 28

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

2 1

[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

2 2

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.

3 4

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

1 0

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

4 8