October 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] How to passthrough 180 after 183
by Ержан Туле пов 02 Oct '12

02 Oct '12

Acctually, now I'm using what you've preposed. But that not what I need. ----- Original Message ----- From: Paul Cupis [mailto:paul@cupis.co.uk] To: FreeSWITCH Users Help [mailto:freeswitch-users@lists.freeswitch.org] Sent: Tue, 02 Oct 2012 15:51:00 +0600 Subject: Re: [Freeswitch-users] How to passthrough 180 after 183 > On Tue, Oct 02, 2012 at 03:00:49PM +0600, ?????????? ?????????????? wrote: > > FS receives Invite, then replies with 100. Then I do Phrase, which, in > turn, changes leg A's status to early media and send 183 to A. Then I try to > bridge A to B. From B I get 180 (ringing), but FS doesn't forward this to A. > How can I force FS to pass it through? > > Not sure if you can. > > I'd suggest trying something like: > > <action application="set" data="ringback=$${uk-ring}"/> > > before the bridge, and see if this behaves as you need it to? > > Regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 >

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snom call pickup blf
by georg＠riseup.net 02 Oct '12

02 Oct '12

Hi, Its been quite a long time, since you started the thread [1]. I've got the same problem you've described. Did you solve this? [1] http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/049500…

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How to passthrough 180 after 183
by Ержан Туле пов 02 Oct '12

02 Oct '12

Hi, FS receives Invite, then replies with 100. Then I do Phrase, which, in turn, changes leg A's status to early media and send 183 to A. Then I try to bridge A to B. From B I get 180 (ringing), but FS doesn't forward this to A. How can I force FS to pass it through? Regards, Y.

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Using FS like SBC with several Operators
by Paul Mancheno H. 02 Oct '12

02 Oct '12

Please help me, I am currently trying to use freeswtich as SBC, the idea is that the internal area has multiple interconnections with several operators, I want to do is for each interconnection: * Having a trunk from the internal to the IP of a virtual interface eth0:1 in FreeSWITCH server, then from FreeSWITCH using interface eth1:1 to the operator XX. * Having a trunk from the internal to the IP of a virtual interface eth0:2 in FreeSWITCH server, then from FreeSWITCH using interface eth1:2 to the operator YY. * And so on. Please can you give me a guide to how I configure SIP profiles to achieve the above?. A lot of thanks, Pepo.

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FreeSWITCH Weekly News and Notes
by Michael Collins 01 Oct '12

01 Oct '12

Welcome to October! I hope the weather is nice where you are. Here it's still above 100F. :) Last week was a little bit quieter than the previous few weeks. I had a chance to work on the FreeSWITCH change log and I made a list<http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_26#News.2C_Notes.2C_.26_M…>of some of the APIs, dialplan tools, and channel variables. These have all been added since 1.2.0 was initially released in early August. All of them have wiki entries - thanks to those who took the initiative to do add them. Feel free to add your knowledge and experience to the mix. Last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_26>was an object lesson in the challenges of getting a SIP proxy working with TLS and FreeSWITCH. This week<http://wiki.freeswitch.org/wiki/FS_weekly_2012_10_03>we are going to change direction and look at something that has been slowly (and painfully) advancing the past few years: mobile VoIP. We will be having Daniel Pocock share with us some information about Lumicall<http://www.lumicall.org>, an open source mobile VoIP client for Android devices. There is also a service component to Lumicall and we'll be learning about that as well. Come join us to see the state of mobile VoIP<http://en.wikipedia.org/wiki/Mobile_VoIP>. We are working on some fresh presentations for later this month. We hope to have an update on e164.org and how we can all get involved. We are also preparing a presentation on how to perform some of the data-gathering techniques <http://wiki.freeswitch.org/wiki/Reporting_Bugs> that are needed for basic and advanced troubleshooting. If you have some input on these or other presentations please let me or Ken Rice know. Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Possible CPU issue on Ubuntu
by Michael Giagnocavo 01 Oct '12

01 Oct '12

This may be related to the CentOS CPU issue; perhaps it's a problem in CFS itself, or Ubuntu has different policies so it's not always triggered? -Michael ** Changed in: linux (Ubuntu) Status: New => Confirmed -- You received this bug notification because you are subscribed to the bug report. https://bugs.launchpad.net/bugs/1058854 Title: High CPU usage of CFS Status in "linux" package in Ubuntu: Confirmed Bug description: It appears that there is an issue with the Completely Fair Scheduler (CFS) not intereacting well with sleep commands. I have written up a test program to help demonstrate the issue and it can be found at: www.github.com/daveisfera/test_sleep/<http://www.github.com/daveisfera/test_sleep/> The test outputs wall time (CPU clock), user time, and system time, but usually only the wall time and user time outputs are meaningful because the system time is so small and usually within the standard deviation for all tests. I have also attached the results from running this test on an HP Pavilion dm4 and I believe that the gap at the low number of threads is caused by either the processor going into a power saving state because of the sleeping and/or a poor interaction with HyperThreading (scheduling the 2 threads on logical cores rather than separate physical cores). Because of this, I then ran 4 instances of: schedtool -a <CPU#> -e nice -19 yes > /dev/null so that all of the cores would be kept busy during the sleeping and this had an unexpected effect of causing the wall time to increase significantly. I am not sure what the cause of the wall time increase is, but it does help the user time plot display the gap between the sleep methods and the none sleep methods that I believe is a symptom/display of the unexpectedly high CPU usage when the process should be sleeping. ProblemType: Bug DistroRelease: Ubuntu 12.04 Package: linux-image-3.2.0-31-generic 3.2.0-31.50 ProcVersionSignature: Ubuntu 3.2.0-31.50-generic 3.2.28 Uname: Linux 3.2.0-31-generic x86_64 AlsaVersion: Advanced Linux Sound Architecture Driver Version 1.0.24. ApportVersion: 2.0.1-0ubuntu13 Architecture: amd64 ArecordDevices: **** List of CAPTURE Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 AudioDevicesInUse: USER PID ACCESS COMMAND /dev/snd/controlC0: dlj 1778 F.... pulseaudio Card0.Amixer.info<http://Card0.Amixer.info>: Card hw:0 'PCH'/'HDA Intel PCH at 0xc0800000 irq 50' Mixer name : 'Intel CougarPoint HDMI' Components : 'HDA:111d76e0,103c1793,00100102 HDA:80862805,103c1793,00100000' Controls : 22 Simple ctrls : 10 Date: Fri Sep 28 20:57:35 2012 InstallationMedia: Ubuntu 12.04.1 LTS "Precise Pangolin" - Release amd64 (20120823.1) MachineType: Hewlett-Packard HP Pavilion dm4 Notebook PC ProcEnviron: TERM=xterm PATH=(custom, no user) LANG=en_US.UTF-8 SHELL=/bin/bash ProcFB: 0 inteldrmfb ProcKernelCmdLine: BOOT_IMAGE=/boot/vmlinuz-3.2.0-31-generic root=UUID=162256F02256D479 loop=/hostname/disks/root.disk ro quiet splash vt.handoff=7 RelatedPackageVersions: linux-restricted-modules-3.2.0-31-generic N/A linux-backports-modules-3.2.0-31-generic N/A linux-firmware 1.79.1 SourcePackage: linux StagingDrivers: rts_pstor mei UpgradeStatus: No upgrade log present (probably fresh install) dmi.bios.date: 01/17/2012 dmi.bios.vendor: Hewlett-Packard dmi.bios.version: F.08 dmi.board.asset.tag: Base Board Asset Tag dmi.board.name<http://dmi.board.name>: 1793 dmi.board.vendor: Hewlett-Packard dmi.board.version: 41.1C dmi.chassis.asset.tag: Chassis Asset Tag dmi.chassis.type: 10 dmi.chassis.vendor: Hewlett-Packard dmi.chassis.version: Chassis Version dmi.modalias: dmi:bvnHewlett-Packard:bvrF.08:bd01/17/2012:svnHewlett-Packard:pnHPPaviliondm4NotebookPC:pvr068E100002204710000622100:rvnHewlett-Packard:rn1793:rvr41.1C:cvnHewlett-Packard:ct10:cvrChassisVersion: dmi.product.name<http://dmi.product.name>: HP Pavilion dm4 Notebook PC dmi.product.version: 068E100002204710000622100 dmi.sys.vendor: Hewlett-Packard To manage notifications about this bug go to: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1058854/+subscriptions

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Re: [Freeswitch-users] Empty wiki and my head
by Michael Collins 01 Oct '12

01 Oct '12

On Sat, Sep 29, 2012 at 3:33 AM, João Mesquita <jmesquita(a)freeswitch.org>wrote: > Just so you know,i am documenting all modules on their configuration files > but that's far from done since I've been flying around the world for the > past 3 weeks. I will get there eventually. > Glad to hear it. Please keep me posted on your progress and also if you need any help. No one person should have to do it all by himself, although it's great to have one person take the lead. -MC

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Migration to freeswitch
by Marco Colombo 01 Oct '12

01 Oct '12

Hi All, I'm using an Asterisk system, but I'm going to migrate it on freeswitch. I have a couple of questions Can freeswitch send update/re-invite package when i resume call from hold? Freeswitch support History-info(RFC4244)? Thanks Best Regards Marco [cid:ca90184ce3384c9ab9fb9ee6c127699d]<http://www.cloudup.it/>

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 75, Issue 223
by William Alianto 01 Oct '12

01 Oct '12

Hi, I just installed 2 spans of OpenVox's A800E analog card to my server. Here is my freetdm.conf : [general] global_parameter => value [span zt FXO] fxo-channel => 1:1 fxo-channel => 1:2 fxo-channel => 1:3 fxo-channel => 1:4 fxo-channel => 1:5 fxo-channel => 1:6 fxo-channel => 1:7 fxo-channel => 1:8 fxo-channel => 2:1 fxo-channel => 2:2 fxo-channel => 2:3 fxo-channel => 2:4 [span zt FXS] fxs-channel => 2:5 fxs-channel => 2:6 fxs-channel => 2:7 fxs-channel => 2:8 and my freetdm.conf.xml <configuration name="freetdm.conf" description="FreeTDM Configuration"> <settings> <param name="debug" value="0"/> <param name="hold-music" value="$${moh_uri}"/> </settings>  <analog_spans> <span name="FXS">  <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable_callerid" value="true"/>  </span> <span name="FXO"> <param name="dialplan" value="XML"/> <param name="context" value="default"/>  <param name="tonegroup" value="us"/> <param name="enable_callerid" value="true"/> </span> </analog_spans> </configuration> Also here is my dialplan configuration : <extension name="inbound" > <condition field="context" expression="public" /> <condition field="destination_number" expression="^(\d+)$" > <action application="transfer" data="101 XML default" /> </condition> </extension> <extension name="freetdm.9d" > <condition field="destination_number" expression="^9(\d+)$" > <action application="set" data="sip_h_X-accountcode=${accountcode}" /> <action application="set" data="sip_h_X-Tag=" /> <action application="set" data="call_direction=outbound" /> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}" /> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}" /> <action application="set" data="inherit_codec=true" /> <action application="bridge" data="freetdm/1/a/$1" /> </condition> </extension> But i still cannot make outbound and inbound call using the card. Can anyone help me? Regards

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Pause in bridge
by Systeembeheer 01 Oct '12

01 Oct '12

Hi, I'm trying to configure the following: When a call comes in, it is first bridged to one extension and after 10 seconds to a second extension. It is possible to issue the bridge command with a timeout and continue settings to one extension and a second bridge command to both extensions. There's some problems with that: - The first extension can answer and 'just miss' the first bridge attempt. - If the first call is cancelled by the user, it rings again 10 sec. later. Is there a way to bridge to an extension and bridge to a second extension 10 sec. later without breaking the first bridge? Or bridge to extensions with a pause of 10 sec. for one of them? Thanks! steveN

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