- FreeSWITCH-users - FreeSWITCH Mailing Lists

G723 codec
by Madovsky 25 May '10

25 May '10

Hi all, is there a chance that in future there will be a mod_g723 transcoder ? Thanks Franck

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Mod_skinny: Call for testing!
by Mathieu Parent 25 May '10

25 May '10

Hello, Skinny Call Control Protocol (SCCP) is a VoIP protocol used on several phones including Cisco IP Phones and Nokia Call Connect 2.0. Contrary to SIP, the phone is controlled by the server. This allow powerfull and funny things (like controlling leds, display, ring type, ...) ; this also allow to use some phones with crappy SIP images, like (guess what?) Cisco ones. Since February, mod_skinny is in freeswitch trunk. Many things happened since. Today the module has reached several milestone (it compiles on windows, call info is shown early, state handling changed to a more FreeSWITCH manner, ...). Most of basic features and more are implemented (from Wiki): * Incoming and outgoing calls * Shared lines and multiple calls per lines * CallInfo set as early as possible * New call made from: OffHook, Softkeys (Redial, NewCall), Stimulus (LastNumberRedial, VoiceMail, SpeedDial) * Answering calls via: OffHook, Softkeys (Answer) * Holding/Unholding calls via Softkeys (Hold, Resume) or Stimulus (Hold), with MOH (music on hold) * Transfer via Softkeys (Transfer) or Stimulus (Transfer) * Ending call via: OnHook, Softkeys (EndCall) * Misc: SoftKeys (Backspace) * Handle firmware version request (VersionReqMessage) per device type or per device * Patterns for dialed numbers to process (patterns-dialplan and patterns-context) Some important missing ones: * Message Waiting indicator (MWI) * Bypass Media (help needed) * Codec Negotiation (help needed) * Forward (transfer already implemented) * Complete list: <http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol#TODO> How can you help: * test if it compiles on your platform * test if it works with your SCCP phone (only 7960 and 7961 have been tested yet) * if you don't have an SCCP phone, test using "perl src/mod/endpoints/mod_skinny/test-skinny.pl" (you need a recent perl, aka 5.10.1, 5.10.0 has buggy thread implementation, at least on Debian Lenny). * write docs about configuration * propose patches (some are easy like API Commands or events ; some are harder ;) * review the code (quality, security, ...) * If you own a Cisco Communication Manager and you know a feature that is not in mod_skinny, make network captures! Some links: * Instructions are at: http://wiki.freeswitch.org/wiki/Mod_skinny * Bugs should go here: http://jira.freeswitch.org/browse/MODSKINNY * Development stuff is at: http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol * Suggested features should go here: http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol#TODO (or bounty or JIRA) Any help is appreciated. happy testing/hacking ;) Mathieu Parent

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did using easyroute
by Meftah Tayeb 25 May '10

25 May '10

hello, i see with easyroute i cabn route just one number and no a range how do i route a range of numbers, like all number starting with 1747XXX to a special gateway, and 1748XXX to another? thanks

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SVN Mirror
by Dave Stevenson 25 May '10

25 May '10

Hi Folks - Mike ? The SVN Mirror looks like it is a bit behind, an SVN update shows the last changes on the 13th May, is there a problem with the Mirror at the moment ? How far behind the Git will the SVN be ? regards Dave

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how can i get the variable "Caller-Channel-Answered-Time" in the API command ?
by daqiang wang 25 May '10

25 May '10

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Question about inbound ACL
by RR 25 May '10

25 May '10

Hello List, I have tried to read as much as I could and try out a bunch of things but it doesn't seem to be working. The scenario is that we have a FS setup as a call distributor to our internal SIP servers. The calls come into the FS when people call any of the 6000 or so DIDs we own. These DIDs are through various providers who's IP addresses we know (obviously). I want to receive / process only calls from these external IP addresses and nothing else. So I did the following in acl.conf.xml file <list name="DIDProviders" default="deny"> <node type="allow" cidr="64.XXX.XXX.XXX/32"/> <node type="allow" cidr="67.XXX.XXX.XXX/30"/> etc... </list> then I went into the $FSHOME/conf/sip_profiles/internal.xml and did the following: <param name="apply-inbound-acl" value="DIDProviders"/> but I still keep seeing calls from other IPs that are not in the "DIDProviders" list getting through. What else do I need to do to prevent this? Should this not be in internal.xml but in external.xml? TIA \RR

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Performing system wide initialization during startup
by David Swardstrom 24 May '10

24 May '10

I am recreating a conferencing application on FreeSwitch to replace a proprietary application originally developed using Dialogic ISDN hardware. I am using mod_conference and a JavaScript application based on the JavaScript samples. As part of this application, I have to track some counters and other data that are global to all of the conference/channel instances. I would like to initialize this code during startup. I could do this multiple ways including using C or C++ code. I would prefer to accomplish this by running a small JavaScript or C based application during Switch Startup to set some Global Variables based on data in an XML file. Is there a way to do this currently? Or is this a case where a new specialized module needs to be developed? Regards, Paul David Swardstrom

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make a remote extension call from a particular gateway
by Wa Halla 24 May '10

24 May '10

I have a branch office that I want to call from a particular gateway. The branch is on extension 1008. With the dialplan below any extension uses gateway4 even extension 1008 <extension name="domestic.uk"> <condition field="${toll_allow}" expression="domestic"/> <condition field="destination_number" expression="^(\d{11})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/gateway4/$1"/> </condition> </extension> <extension name="domestic1008.uk"> <condition field="${toll_allow}" expression="domestic"/> <condition field="sip_auth_username" expression="^(1008)$"/> <condition field="destination_number" expression="^(\d{11})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/gateway1/$1"/> </condition> </extension> Any suggestions of what I am doing wrong. I have tried <condition field="sip_auth_username" expression="^(1008)$"/> with and <condition field="sip_auth_username" expression="^1008$"/> without the brackets so have elimated that. If I switch the gateways in the domestic.uk extension, all calls go from gateway1. So it seems the domestic1008.uk never gets executed as true.

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xml_curl - Binding has no url
by Peder 24 May '10

24 May '10

I am trying to use xml_curl with directory and it is failing with this message: 2010-05-24 08:38:46.333352 [ERR] mod_xml_curl.c:444 Binding has no url! 2010-05-24 08:38:46.333352 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_xml_curl.so **Module load routine returned an error** I am sure I am missing something, but I can't figure out what it is. The module is built: ls -al /usr/local/freeswitch/mod/mod_xml_curl.so -rwxr-xr-x 1 root root 1498152 2010-05-24 08:23 /usr/local/freeswitch/mod/mod_xml_curl.so The config file looks ok from what I can tell: cat /usr/local/freeswitch/conf/autoload_configs/xml_curl.conf.xml <configuration name="xml_curl.conf" description="cURL XML Gateway"> <bindings> <binding name="directory_lookup"> <param name="directory.php" value="http://localhost:80" bindings="directory"/> </binding> </bindings> </configuration> I've stopped and started FS multiple times and still the same error on startup and if I try and manually load the module with "load mod_xml_curl.so". Any ideas? I am sure it is something dumb, but I just can't see it.

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Re: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
by Kristian Kielhofner 24 May '10

24 May '10

Seriously... As the person who started this thread I'm disappointed to see it come to this. Anyways, using mod_lcr and redirect I've been able to get 450 CPS (sustained) from FreeSWITCH on mediocre hardware. I'm more than happy (read: thrilled) with that. Shameless plug: I'll be telling the whole story of how I came to use FS as a SIP redirect/LCR server on my blog in the next couple of days. I'll even include the configs and SIPP scenario I used to test. Should be a good post! -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces(a)lists.freeswitch.org <freeswitch-users-bounces(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org <freeswitch-users(a)lists.freeswitch.org> Sent: Mon May 24 02:22:31 2010 Subject: Re: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices wait until several hundred calls per second isn't enough for my needs and then care about it? This thread officially closed. If anyone has any actual code to contribute to this, they can do so on http://jira.freeswitch.org. Mike On May 24, 2010, at 2:03 AM, Carlos Romero wrote: > Sorry I wasn't aware of this. > > So what do you intend to do to remove the limitations of sofia that > seems to affect performance? > > Carlos Romero > > On Sat, May 22, 2010 at 6:59 PM, Steve Underwood wrote: >> PJSIP as a substitute for Sofia in high volume server applications? Are >> you a jester or a troll? :-) >> >> All stacks have limitations, and Sofia has some. In the early days of FS >> development, every open source SIP stack was tried. All the alternatives >> were found to have far more limitations than Sofia. > _______________________________________________ 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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