October 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

is there any variable to set in directory user variables to execute an application on every channel
by Babak Yakhchali 15 Oct '12

15 Oct '12

hi is there any variable to set in directory user variables so that an application is called on every created channel for that user (or on every channel answered)? <include> <user id="3003"> <params> ... </params> <variables> ... <variable name="execute_on_any_channel" value="my_app"/> </variables> </user> </include> thanx

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Linux Kernel Options
by Spencer Thomason 15 Oct '12

15 Oct '12

Hello, I'm working on recompiling a custom kernel for our FreeSWITH installations. Are there any recommendations in regard to the options that should or should not be set for optimum performance. This is based on Linux Kernel 2.6.32. Is disabling the tickless feature and building a 1000hz timer still a recommendation? Is there any recommendation as to the preemption model for the CFS scheduler in this version? Thanks, Spencer

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Java.conf.xml question
by junkmail 15 Oct '12

15 Oct '12

Hello. I have a question about the java.conf.xml and how it is to be used. specifically the <startup> tag I thought it would operate much like the lua.conf.xml, but maybe it doesn't. I created a class and had it run like this. <startup class="listener" method="startup" args=""/> But unless the class returns(exits) it will hang freeswitch till freeswitch kills itself or kill the mod_java module. I was looking in how to get a java process to "back ground itself" like forking in other languages it seems that this is not commonly done or at least in the same sense. So here I am just wondering if I am trying to make this do something that it was not meant to do? Also if you would be able to explain what the purpose is. ;) thanks

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H.323
by Tamas Jalsovszky 15 Oct '12

15 Oct '12

Hello, Is mod_h323 or mod_opal stable enough to use it in production? (just voice calls, h323v2, no fax) If yes, which one is proven to work well? Maybe is FS(sip)+yate(sip-h323) a better way to go? Thanks in advance, Jalsot

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sip_presence Table
by Gerald Weber 15 Oct '12

15 Oct '12

Hi all, playing around with the sip presence feature, i noticed that the sip_presence table in share_presence.db is always empty. I've set the following parameters in my internal.xml : <param name="manage-presence" value="true"/> <param name="send-presence-on-register" value="true"/> <param name="presence-probe-on-register" value="true"/> <param name="presence-proto-lookup" value="true"/> <param name="manage-shared-appearance" value="true"/> <param name="dbname" value="share_presence"/> <param name="presence-hosts" value="$${domain}"/> <param name="force-register-domain" value="$${domain}"/> <param name="force-register-db-domain" value="$${domain}"/> My dialplan contains: <extension name="dw2xxx"> <condition field="destination_number" expression="^(200[0-9])$"> <action application="set" data="presence_id=$1@$${domain}"/> <action application="export" data="dialed_extension=$1"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> </condition> </extension> When i make a call from 2001 to 2000, i see 2 rows in sip_dialogs. So i think presence is basically working. But whats the purpose of sip_presence ? Where, for example, does a "presence in 2000(a)192.168.20.73 away Away" go ? would i see this in any table ? thanks&regards, gw

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Local vs external IP
by Malay Thakershi 15 Oct '12

15 Oct '12

Hello, How do I make FreeSwitch treat certain IP as local network IP address? The freeswitch server seems to not understand that 10.25.20.x is an internal network. >From what I read, FreeSwitch has settings for local addresses and external addresses. When it talks to non-local hosts it should tell them to reply to the external addresses. In my configuration, it does this just fine. When it talks to hosts on internal networks it should use its internal address. Instead it is telling internal hosts to communicate with it via its external address. This is causing a call setup failure. My network IP config: Ethernet adapter Local Area Connection 2: Media State . . . . . . . . . . . : Media disconnected Connection-specific DNS Suffix . : Ethernet adapter Local Area Connection: Connection-specific DNS Suffix . : IPv4 Address. . . . . . . . . . . : 10.25.20.202 Subnet Mask . . . . . . . . . . . : 255.255.255.0 Default Gateway . . . . . . . . . : 10.25.20.1 Ethernet adapter VirtualBox Host-Only Network: Connection-specific DNS Suffix . : Link-local IPv6 Address . . . . . : xxxxxxxxxxxxxxxxxxxx IPv4 Address. . . . . . . . . . . : 192.168.56.1 Subnet Mask . . . . . . . . . . . : 255.255.255.0 Default Gateway . . . . . . . . . : Thanks for help. Malay Thakershi

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Re: [Freeswitch-users] Compiling mod_h323 on Windows
by Peter Olsson 15 Oct '12

15 Oct '12

First of all, download h323plus and ptlib from here: http://www.h323plus.org/source/ Extract the archives on the same level as the freeswitch main source directory, my structure looks like this; - C:\dev\ - freeswitch - h323plus - ptlib Then you will need to build ptlib, and then h323plus. You will only need the static libraries. It might be tricky to get these things built, but you might find some more info on Google if needed. You might need to change the paths to make sure the h323plus finds ptlib. Also, there is no 64-bit build in the project files, so if you need 64-bit you will need to update the project files (I have a modified version, which might help if you need it). After this, build FS an you usually do, when the build is done, build mod_h323 manually. The mod_h323 project will look in the folders specified above, so it's important that you put ptlib and h323plus directories on the same directory level as the main freeswitch source directory (as my example above). Good luck! /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Shishko Skickat: den 12 oktober 2012 16:12 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] Compiling mod_h323 on Windows Hi, could anybody give me basic instructions on how to build mod_h323 on Windows using Visual Studio 2010? Thanks _________________________________________________________________________ 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 !DSPAM:5078465332761226277865!

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FreeTDM DTMF generation not stable.
by William Alianto 15 Oct '12

15 Oct '12

Hi, I have setup a FS box and added OpenVox A800E cards for FXO ports. The freetdm has been configured properly. But when I tried to dial out from the FXO ports, the call always fail. It seems like the system didn't send out the DTMF properly. Sometimes it's not even generating any DTMF. http://pastebin.com/ZSzyX7id Have anybody experienced this before? Regards, William

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Freeswitch + Python - request for paid advice
by Cal Leeming [Simplicity Media Ltd] 14 Oct '12

14 Oct '12

Hello all, I'm currently looking for someone who can offer a few hours (4-6?) of paid advice over Skype relating to architectural design of using Freeswitch and Python. I felt it would be unfair to expect the community to answer so many questions such detail, hence why I'm more than happy to pay for the advice. I have around 8 years solid Python development experience, and have spent a week or so reading through the masses of wikis/documentation/mailing lists relating to Freeswitch. Prior to this I spent a considerable amount of time comparing other platforms to FS, and reading various articles about the technical differences between them as well, before coming to the conclusion that FS was the way forward. Some of the questions I would be asking are; * Are there any surprising 'gotchas' when using Python with FS? * Is there a preferred way of integrating Python into FS? - For example, a call queue system, should I have a bunch of external scripts generating the XML config on the fly, or should I have FS call a Python method which calls a database and then returns a chunk of XML on the fly (xml_fetch?), or is there other ways? * There are various projects relating to integrating Event Socket with Python/Twisted.. what are the pros/cons of the most popular ones (such as fiorix/eventsocket on gh). * Some real world examples of why you would use Event Socket, and the correct way of doing so.. Do you really have to use Twisted, or can you use eventlet instead? * Any unexpected 'gotchas' when using Django with python+fs (I saw the bugs relating to sockets not being closed, which appears to now be fixed?) * The correct way to provide dual redundancy (i.e. I'm assuming you can't run two freeswitch instances in active<>active mode, so you'd have them in active<>passive mode and then have your load balancer fail over in the event of a problem? But this raises questions about how you keep both instances up to date with the same config, what would happen to the existing calls, etc etc). There are only a few caveats to this; * The person should have a lot of prior experience with integrating Python with Freeswitch, and be able to demonstrate this in some way. * Any information given would be summarized and sent onto this mailing list (and my own blog) so that it may benefit others in the future - you can chose whether or not to have your credits included. * The person should ideally speak reasonably good English If you (or anyone you know) would be interested in something like this, please feel free to hit me up directly. Thanks in advance to everyone Cal

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att_xfer not working properly (One Way Sound)
by Nitin India 14 Oct '12

14 Oct '12

Hi, 101 Calls on 102 then 102 Do a att_xfer by pressing 1 while conversation on which bind_digit_action executes the att_xfer to 103 if 103 Answers the Call 101 Gets the Hold Message and 102 Start getting sound of 103 But, 103 is unable to hear the sound of 102 who done the att_xfer and if 102 Disconnects the Call then 103 is Bridged with 101 properly and both users can talk and hear each other. So, the problem is 103 is unable to hear 102 and there are no sip problems etc.. as we are using PRI/ISDN Lines and have also tried with sip both have same issue. See my Dial Plan. <extension name="Main IVR"> <condition break="on-false" expression="^101" field="destination_number"> <action application="answer"/> <action application="bridge" data="{execute_on_answer='bind_digit_action bridge_1,1,exec:att_xfer,freetdm/1/a/103'}[leg_timeout=60,origination_caller_id_number=123456]freetdm/1/a/102"/> <action application="hangup"/> </condition> </extension> -- ------------------------------------------------ Thanks & Regards Nitin

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