May 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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SCTP on Windows/C++/C#/VB
by Jan Berger 24 May '10

24 May '10

Hi, I minor note in SCTP for those with special interests. The SCTP on Windows is working, thought the current performance is only ca 30,000 messages per sec on my old laptop. The driver performs stable under normal operations, but 100% CPU load over time will expose a snag that the author currently are working to sort out. I have tested on Windows XP Professional with an ordinary laptop. The stack failed to install on Windows Vista 32 bit. For all practical purposes the stack is sufficient for my development work, but I will not recommend using in production yet. I have not tested the alternative which is www.sctp.de , but would like to hear from anyone that has experience of this. If anyone is interested in this work let me know. Jan

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

24 May '10

why choose a library that is hard to work with in the first place? it seems like sofia has many downsides... limited threads per sip-profile and now this On Fri, May 21, 2010 at 3:20 PM, Michael Collins <msc at freeswitch.org> wrote: > > > On Thu, May 20, 2010 at 11:46 PM, Carlos Romero <romerocarlos24 at gmail.com>wrote: > >> what do you mean that you don't have access to it? >> >> you have the code >> > > The SIP stack is supplied by Nokia. Open source or not, it is bad form to > muck with someone else's library without due cause. Like all robust SIP > stacks, Sofia is huge, deep, complex, and otherwise uninviting of those who > would waltz in and do whatever they wish. If you feel that it is even within > the realm of possibility then I recommend that you find the location in the > Sofia library where this takes place, make the appropriate changes, and > thoroughly test for regressions. Submit the patch to the Sofia devs who will > then evaluate it and see if it is worthy of inclusion. > > Or more simply put: "having the code" and "having access" are COMPLETELY > different concepts. > -MC > > >> >> On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale >> <anthony.minessale at gmail.com> wrote: >> > not without intervention from the sofia devs, >> > Sofia sip does that deep down and we don't have access to it. >> > even when we supply it as a one big string it renders it out depending >> on >> > those settings. >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > On Fri, May 21, 2010 at 4:34 AM, Carlos Romero <romerocarlos24(a)gmail.com> wrote: > what do you mean that you don't have access to it? > > you have the code > > On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale > <anthony.minessale at gmail.com> wrote: >> not without intervention from the sofia devs, >> Sofia sip does that deep down and we don't have access to it. >> even when we supply it as a one big string it renders it out depending on >> those settings. >> >

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How do you set outbound from address in SIP header
by Bob Coleman 24 May '10

24 May '10

Hi, I am trying to set the sip from address for outbound calls, but cant seem to find the option. The below is from a sip trace, and shows it set to the defaults eg From: "FreeSWITCH" <sip:0000000000@10.12.78.1>;tag=pQ68N9mKc7rDK I have changed the vars.xml to set the above details but nothing happens, restarted just to make sure. Searched the other conf files to make sure there is not another place. Using an older binary 1.4 , is there any known issues regarding that. Thanks Bob

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