August 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

No SIP UPDATE will send
by Alexander Haugg 14 Oct '15

14 Oct '15

Hi all, I have the problem, that the freeSwitch don't send a SIP UPDATE to the PBX. Scenario: PBX (Siemens H3K) is connected via SIP trunking to the FreeSwitch. If I have two Legs from the H3K to the Freeswitch I'm using the uuid_bridge command to connect this two legs (works fine), but the call legs will not updated via SIP UPDATE. The PBX have no information to update the displays on the telephones. The target number (toHeader) was the routing number from the H3K to the freeswitch. Now I need a SIP UPDATE from FreeSwitch to PBX to actualize the toHeader with the new partner number (the partner number is the fromHeader of the other leg). In the sofia profile I was try it with the parameter <param name="send-display-update" value="true"/> and without this parameter (in the source code the default is "true") What can I do to force an UPDATE to the call legs? Version of the FreeSwitch FreeSWITCH Version 1.5.15b+git~20150121T080154Z~90ab1d16f5~64bit (git 90ab1d1 2015-01-21 08:01:54Z 64bit) In the INVITE from the PBX the ALLOW header includes the UPDATE Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, INFO, PRACK, UPDATE Many thanks!

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FreeSWITCH Unstable Repo
by Brian West 03 Oct '15

03 Oct '15

FreeSWITCHers, To help speed the testing and release cycle up, Team FreeSWITCH has created a FreeSWITCH Unstable Debian repo, This repo is updated after every commit to master and the debs lag behind about an hour from each commit. WARNING this repo is untested and should NOT be used in production. How to use this repo: 1. New Debian Jessie system 2. Run These Commands: echo "deb http://files.freeswitch.org/repo/deb/debian-unstable/ jessie main" > /etc/apt/sources.list.d/99FreeSWITCH.list echo "deb http://files.freeswitch.org/repo/deb/debian/ jessie main" >> /etc/apt/sources.list.d/99FreeSWITCH.list wget -O - https://files.freeswitch.org/repo/deb/debian/key.gpg |apt-key add - apt-get update apt-get install freeswitch-all 3. To update: apt-get update apt-get dist-upgrade Any commit that goes in while the current build is running will be included in the next build that starts. Questions? Mailing List / IRC / Hipchat Comments? Mailing List / IRC / Hipchat Bugs? https://freeswitch.org/jira/ Thanks, and happy testing! -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> | Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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How to test a Pull Request! aka how to help us out!
by Brian West 02 Oct '15

02 Oct '15

Checkout freeswitch master first! git clone https://freeswitch.org/stash/scm/fs/freeswitch.git freeswitch.git cd freeswitch.git git config --add remote.origin.fetch '+refs/pull-requests/*/from:refs/remotes/origin/pr/*' git config alias.pr '!f() { git checkout pr/$1; git config branch.pr/$1.merge refs/pull-requests/$1/from;} ; f' git pr <PR_NUMBER> build, test, report back. You can view the list of pull requests here: https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests Questions? Comments? Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> | Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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No audio after bridging outbound calls
by Govind Rajan M 25 Sep '15

25 Sep '15

Hi, I was testing the freeswitch with the gateway and when I make two outgoing calls and bridge them together, I am not able to hear the audio. But as soon as I played something on any one of the channel, both the parties can be able to here the audio. Did anyone face this kind of issue? Can you please help me to solve the issue? My machine is running on AWS behind NAT and i have done all the configuration that is specified in the freeswitch wiki page. P.S : when I make one outgoing to sip user and another outgoing call via SIP gateway everything works fine(both the parties able to hear the voice). -- Thanks, Govind

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Compiling under SmartOS
by Stanislav Sinyagin 21 Sep '15

21 Sep '15

I see there are some people on the list, working with SmartOS. The current master fails to compile: https://freeswitch.org/jira/browse/FS-7967 Your input will be appreciated. I just started looking around and getting the feeling what SmartOS is. I worked with Solaris quite a lot, but that was almost 10 years ago.

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systemd won't enable for automatic start of FS
by Bote Man 09 Sep '15

09 Sep '15

I'm struggling to get systemd to start FreeSWITCH at boot time and also leave it running once it starts. Problem #1 - "systemd enable freeswitch" outputs the right steps, but ends with an error message. The error I get is "No such file or directory" but that is less than helpful. Problem #2 - When I start FreeSWITCH with "systemd start freeswitch" it starts alright, but systemd does not detect FS running properly so it kills it after the specified 45 seconds elapses and restarts it. I can see this in top when FS gets a new pid. Details -------- It's a new installation of Debian 8.1 and I chose to use systemd as the init process. I compiled FreeSWITCH from source using the latest Master as of late July. The only section of the unit file that is supposed to be used by the "enable" command is the [Install] section and that simply points to the multi-user.target as expected. I do not know what to do with the 2 other sample systemd files in /usr/src/freeswitch/debian, perhaps they are key. I have read up on systemd and the FS unit file looks good to my untrained eye. I did not use the sysvinit files that are also in that FS debian directory. I successfully used systemd to start, stop, and enable a few other system services with success so there must be a tweak to the FS unit file that I am missing. I can start FS manually and it runs normally. A Google search revealed a number of complaints that systemd would emit an error message with its "enable" command, yet it would successfully enable the service; this was with a few different services on Debian and Ubuntu, so it seems there might still be some weirdness with systemd on Debian. Has anybody gotten systemd to start FS reliably? Has anybody figured out how to make systemd detect FreeSWITCH running so that it doesn't restart it repeatedly? Once I figure out how to tame systemd I will document my findings in Confluence as these are the last steps that allow you to get FS up and running quickly. Thanks. --- Bote

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FreeSwitch - Performance issues
by Shaun Stokes 07 Sep '15

07 Sep '15

Hi, We're experiencing performance issues with FreeSwitch, our target is 500 concurrent sessions, but at the moment this starts to bottleneck around 30. Host system: Windows Server 2012 Hyper-V Host AMD Opteron 4386 (2 processors) - 16 cores total 128GB DDR3 2TB RAID 5 (700MB/s tested read and write throughput) FreeSwitch Virtual Machine: FreeSwitch 1.4.15 Ubuntu 12.04 LTS 16 Virtual cores (high priority) 2GB RAM (would assign more but FreeSwitch never seems to use much) 500GB HD (on VHDX) After around 30 concurrent sessions we begin to see CPU spikes almost every time a new call comes in, as the sessions increase the size and frequency of the CPU spikes also increase. The system seems to be able to sit comfortably with over 100 concurrent sessions and 80% idle CPU, providing we don't have any new calls hitting the platform. The spikes are causing audio (RTP) to stutter or in some cases drop completely for a few seconds. The FreeSwitch spikes are occurring on all 16 cores, we have been monitoring the system using htop and mpstat. This is an example of when we receive an inbound call while we have 34 concurrent sessions: 14:26:04 CPU %usr %nice %sys %iowait %irq %soft %steal %guest %idle 14:26:05 all 1.67 0.00 27.34 0.00 0.25 0.00 0.00 0.00 70.74 14:26:05 0 1.00 0.00 28.00 0.00 0.00 0.00 0.00 0.00 71.00 14:26:05 1 0.99 0.00 26.73 0.00 0.00 0.00 0.00 0.00 72.28 14:26:05 2 0.99 0.00 26.73 0.00 0.00 0.00 0.00 0.00 72.28 14:26:05 3 3.92 0.00 26.47 0.00 0.00 0.00 0.00 0.00 69.61 14:26:05 4 0.99 0.00 25.74 0.00 1.98 0.00 0.00 0.00 71.29 14:26:05 5 0.99 0.00 26.73 0.00 0.99 0.00 0.00 0.00 71.29 14:26:05 6 0.00 0.00 26.47 0.00 0.98 0.00 0.00 0.00 72.55 14:26:05 7 0.00 0.00 26.73 0.00 0.00 0.00 0.00 0.00 73.27 14:26:05 8 3.96 0.00 27.72 0.00 0.00 0.00 0.00 0.00 68.32 14:26:05 9 2.00 0.00 27.00 0.00 0.00 0.00 0.00 0.00 71.00 14:26:05 10 6.00 0.00 32.00 0.00 0.00 0.00 0.00 0.00 62.00 14:26:05 11 1.96 0.00 27.45 0.00 0.00 0.00 0.00 0.00 70.59 14:26:05 12 0.99 0.00 26.73 0.00 0.00 0.00 0.00 0.00 72.28 14:26:05 13 1.00 0.00 28.00 0.00 0.00 0.00 0.00 0.00 71.00 14:26:05 14 0.00 0.00 27.00 0.00 0.00 0.00 0.00 0.00 73.00 This is when the system is not receiving an inbound call but is sitting comfortably at 34 concurrent sessions: 14:25:57 CPU %usr %nice %sys %iowait %irq %soft %steal %guest %idle 14:25:58 all 0.87 0.00 0.62 0.00 0.12 0.00 0.00 0.00 98.39 14:25:58 0 0.99 0.00 0.99 0.00 0.00 0.00 0.00 0.00 98.02 14:25:58 1 0.99 0.00 0.99 0.00 0.00 0.00 0.00 0.00 98.02 14:25:58 2 0.98 0.00 0.98 0.00 0.00 0.00 0.00 0.00 98.04 14:25:58 3 0.00 0.00 0.99 0.00 0.00 0.00 0.00 0.00 99.01 14:25:58 4 0.00 0.00 0.00 0.00 1.98 0.00 0.00 0.00 98.02 14:25:58 5 0.00 0.00 0.00 0.00 0.99 0.00 0.00 0.00 99.01 14:25:58 6 0.00 0.00 0.99 0.00 0.00 0.00 0.00 0.00 99.01 14:25:58 7 0.00 0.00 0.99 0.00 0.00 0.00 0.00 0.00 99.01 14:25:58 8 5.00 0.00 1.00 0.00 0.00 0.00 0.00 0.00 94.00 14:25:58 9 1.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 99.00 14:25:58 10 1.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 99.00 14:25:58 11 0.99 0.00 0.99 0.00 0.00 0.00 0.00 0.00 98.02 14:25:58 12 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 100.00 14:25:58 13 1.98 0.00 0.99 0.00 0.00 0.00 0.00 0.00 97.03 14:25:58 14 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 100.00 14:25:58 15 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 100.00 We're using FreeSwitch in multi-tenant mode, we have tried dedicating a single VM for one tenancy but still experience the issue. My assumption has been that this will be due to the dialplan, I've optimized this slightly by writing LUA scripts to handle some of the inbound calls this appears to take away some of the load but we're still using the internal dialplans for each tenancy (tenancy dialplans have an average of 300 entries). We are using the following arguments when running FreeSwitch: -rp -nc -nonat I've seen another post which says we should operate FreeSwitch in High Performance mode using -hp but can't find anything information about this, is this still a valid argument to use with FreeSwitch? Has anyone experienced similar performance issues before or have any advice? Hope someone may be able help. Thanks, Shaun ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________

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mod_rtmp codec
by Paul Cuttler 07 Sep '15

07 Sep '15

It looks like mod_rtmp has the codec hard-coded to Speex. I was wondering if there was issues with getting other codecs to work. We'd really like to use PCMA/U. I've tried changing a couple of lines of code where the codec is set and it works okay in the incoming (play) direction, but is terribly distorted or just absent in the outgoing (publish) direction. Could someone please advise whether it's possible to use other codecs, and where I'm best to get started. thanks, Paul

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Freeswitch 1.4.21 from source and Speex
by Sharath Kumar 03 Sep '15

03 Sep '15

Hi, I'm trying to cross-compile for an embedded target and building the latest production build from source but it fails trying to build speex. I also noticed in the libs directory there is no speex folder, so I had to copy it from some other place. How do I disable speex from compiling ? is it with --without-speex in the configure script ? thanks, Shaks

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Python based Event Socket Library doesn't connect to remote FreeSWITCH
by Carlos Fererro 02 Sep '15

02 Sep '15

To all Event Socket Library FreeSWITCH gurus! Python implementation of the Event Socket Library doesn't seem to work with FreeSWITCH running on remote host. Only "localhost" or "127.0.0.1" has to be specified as the first parameter to the ESLconnection constructor to let client to successfully connect to the FreeSWITCH. Event if FreeSWITCH is running on the SAME machine (not even remote host) as the client and the "host" is specified with the IP address of the machine in the ESLconnection - no valid connection is getting established. I am pretty sure that such a fundamental thing can't not to work and I am just missing something simple. Any help is greatly appreciated. Thanks Carlos

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