September 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Phone that does IPv6 properly?
by Tim St. Pierre 07 Sep '12

07 Sep '12

Hello, I'm excited about moving to IPv6, and not having to deal with NAT ever again. We have IPv6 address space and transit, and FS does IPv6, so now I need some phones. I'm told that there will be IPv6 firmware coming for my 400+ Aastra phones, but that might be a while. Are some IP phones out there today that have good IPv6 support? Specifically, it has to do the usual network protocols (SLAC, router solicitation, etc.) DHCP6 support would be nice. It has to be able to look up an SRV record, and then get AAAA records for the SIP proxies referenced. Everything is native IPv6, so it isn't really important that it can do tunnels or anything special. I'm not even sure that I need IPv4 on it at all, since it will be deployed in an IPv6 only environment. G.722 audio would be nice too. Thanks for any suggestions! -Tim

1 0

att_xfer limitations
by Emrah 06 Sep '12

06 Sep '12

Hi guys, I just briefly tried the att_xfer app and it seems very limited to me. It seems to be either using a gateway or the db, but it never hits the dialplan at any point. I love the extra add-ons you guys built, especially the option to conference before finalizing the transfer, but I think the app should be revised at some point to support a more dialplan oriented approach. Now, when a call comes in, I can att_xfer to a SIP user, but can't call out a cellphone using my dialplan logic with my multiple outbound providers. I encourage you to give it a look and see if it's worth an upgrade. Best as always, Emrah

4 7

Far-end NAT traversal and anti-tromboning with FS
by Phil Quesinberry 06 Sep '12

06 Sep '12

Ah, the joys of NAT. Is FS' NAT functionality supposed to be able to detect and handle far-end NAT traversal? I have the following scenario: Endpoints on a LAN connected via the Internet to FS though NAT FS --> Internet --> NAT firewall --> Endpoints Proxying the media works fine but when attempting to bypass media between endpoints all on the same LAN, I would expect FS to negotiate media between them using their internal IP addresses. So if IP phone A at 10.0.0.101 wants to talk to IP phone B at 10.0.0.102, FS should set up the media streams to go directly between them using those internal addresses. However, that is not the behavior I'm seeing. Instead, FS is trying to set up the media between them using their external IP addresses which results in no audio. Is there a way to get FS to handle this properly? I'm pretty sure that OpenSIPS' can detect and handle this scenario and I'm hoping that FS can as well, without having to put OpenSIPS in front of it. Is this just a configuration issue? Many thanks, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

1 0

1.2 stable x64 installer build issue?
by Lawrence Conroy 06 Sep '12

06 Sep '12

Hi there, I've had issues with 1.2 stable installer build with VS 2010 pro; I'm unsure if this is a bug in the Setup.wixproj (installer build) file or an error in user. Please bear with me -- I've just switched from using an old TiBook with Leopard as that required too many system changes to work => I have little idea what I'm doing. Steps for 1.2 stable x64 build: - Git clone -b v1.2.stable, & fire up VS2010pro+Wix3.5 [note -- not express, and without the SDK7.1] - Select x64 Release; Unload project mod_cepstral, and delete cepstral from Setup references - build -> works fine - select x64 Setup -> fails - change one of Setup's pre-build event commands from: "$(WIX)bin\heat.exe" dir "$(ProjectDir)..\..\Win32\Release\sounds" -cg FreeSWITCHSoundFiles8 -gg -scom -sreg -sfrag -srd -dr SOUNDLOCATION -var var.FreeSWITCHSoundFilesDir -out "$(ProjectDir)Fragments\FreeSWITCHSoundFiles8.wxs" to: "$(WIX)bin\heat.exe" dir "$(ProjectDir)..\..\x64\Release\sounds" -cg FreeSWITCHSoundFiles8 -gg -scom -sreg -sfrag -srd -dr SOUNDLOCATION -var var.FreeSWITCHSoundFilesDir -out "$(ProjectDir)Fragments\FreeSWITCHSoundFiles8.wxs" - build -> works fine => Making this pre-build change in VS2010 seems to have worked for me. [unsurprisingly, with an x64 build, the sounds are placed into the x64/Release path] I *think* that this pre-build command is generated from Setup.wixproj (near the end of that file). - Git clone -b v1.2.stable; - edit freeswitch\w32\Setup\Setup.wixproj to change Win32\Release\sounds -> x64\Release\sounds - select x64 Release; unload/remove cepstral - build -> works fine - select x64 Setup - build -> works fine I have NOT built a w32/x86 version in these steps -- maybe that would have pre-loaded the sound files in the "expected" place? I'm an entire newbie at this Windoz lark, so I'd appreciate some sanity checking. Q: Is this a build file "feature" or have I just not setup VS2010 properly? all the best, Lawrence

2 2

Call duration > 0 while hangup cause is not NORMAL_CLEARING
by Muhammad Naseer Bhatti 06 Sep '12

06 Sep '12

Hi, I am seeing a lot of calls with hangup cause/Enumeration is not NORMAL_CLEARING while the calls still have a valid duration and answered time. For a typical billing scenario, calls are billed if (answered time) billsec>0 and hangup cause = NORMAL_CLEARING. In what cases the enumeration would be other than normal and while the calls would be still a valid answered call? Thanks

6 7

Bat-Phone or Moscow RED PHONE Hotline
by Sean Devoy 05 Sep '12

05 Sep '12

Hi, One of my customers has asked if we can setup a phone on an ATA (or otherwise if necessary) that you just pick up and it is automatically connected to some number? Much like the "Bat-phone" or the US to USSR Emergency Phone in the Situation room out the White House. Thoughts? Thanks, Sean

6 7

Missing libtiff after make current
by Guillermo Ruiz Camauer 05 Sep '12

05 Sep '12

Today, after issuing "make current", span_dsp wouldn't compile. Since I had problems with this before, I issued "make spandsp-reconf" which fixed the problem last time, but not this time. While researching into the matter, I tried "git checkout -- libs/spandsp/INSTALL" - followed by "make current" with no joy. I also tried "git pull && ./bootstrap.sh && ./configure && make" to no avail. All of these where suggestions found in other threads of this list. I also tried "make sure". After these attempts, I am getting the following error upon freeswitch startup: 2012-09-04 15:52:51.286376 [CRIT] switch_loadable_module.c:1310 Error Loading module /usr/local/freeswitch/mod/mod_spandsp.so **libtiff.so.5: cannot open shared object file: No such file or directory**. This is on: CentOS 6.3 Final Linux Freeswitch 2.6.32-279.5.2.el6.x86_64 #1 SMP Fri Aug 24 01:07:11 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux FreeSWITCH Version 1.3.0+git~20120903T183548Z~f012b7abf8 (1.3.0; git at commit f012b7abf8 on Mon, 03 Sep 2012 18:35:48 Z) Started. Any tips will be appreciated.

6 14

FreeSWITCH Conference Call Today
by Michael Collins 05 Sep '12

05 Sep '12

Hello all! Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_05 We have a number of things to discuss as a community. Look forward to seeing you there. -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

1 0

No Dialplan on answer channel
by Khue Nguyen Minh 05 Sep '12

05 Sep '12

Hi all, I want make outbound call to a user with command: *originate sofia/external/1000(a)somewhere.com '&javascript(hello.js /opt/freeswitch/sounds/2.mp3)'* my javascript will play file /opt/freeswitch/sounds/2.mp3. The call make successful but hangup immediately. I see log in fs_cli and see this line "No Dialplan on answer channel". How I can fix this problem? Thanks Khue Nguyen.

3 5

SRTP performance overhead?
by Chris Martineau 05 Sep '12

05 Sep '12

Hi, Does anyone have any information regarding the impact on freeswitch performance when using srtp? i.e. if I can currently run a maximum of say 400 concurrent calls would this drop to 200 or is there no appreciable impact? Many thanks Chris

1 0