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409 Conflict When registering FS with Softswitch
by RR 02 Feb '10

02 Feb '10

Gentlemen, I have a probably a simple problem but I have no idea why it's occurring. I am beyond novice/new to configuring FS and as my first try, tried to configure it to register with our softswitch so I could have the basic registration to happen between FS and it. However, I keep getting the "409 Conflict" message when that username is not actually registered with the softswitch. The profile / config file under sip_profiles/external is configured like so <include> <gateway name="Test-Inbound"> <param name="username" value="user1"/>--> <param name="password" value="user1pwd"/>--> <param name="realm" value="mysoftswitch.net.au"/>--> <param name="extension" value="1000"/> <param name="from-user" value="user1"/>--> <param name="expire-seconds" value="60"/>--> <param name="register" value="true"/>--> </gateway>--> </include> Any ideas why? And how to resolve it or what kind of traces I can send to you guys to help me resolve it? Like I said I am so new to FS that I don't even know what I can turn on in FS to see what's going on Thanks a lot \RR

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Somebody help me understand the 'features' set up please :-)
by mailinglist 02 Feb '10

02 Feb '10

Ok, I've gotten the Freeswitch to register to my VoIP provider. I've gotten my phones to register to Freeswitch, and I can receive and make calls, all very nice. I've gotten a Sipura SPA901 and a Siemens Gigaset S68IP registered to the Freeswitch. When I receive a call, I would like to be able to transfer the call to another phone, or change the call to a conference call with two local phones. So I've been looking at the examples in the wiki, and I can't make them work, not as I understand them anyways. Especially the att_xfer seems to be able to do what I need. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer As I understand Example1, I should answer the call, and then press *3 during the call, and either transfer it or change it to a threeway call. I get the first part, create an extension in the dialplan called att_xfer. But what is meant by the second par 'then bind this feature to DTMF 3', how do I enter that, and where? I hope somebody can help me with this (again)? Best regards Fribse /Running Freeswitch on pfSense running on a VMWare ESXi 4.0

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Hybrid Encryption?
by Nicholas Lee 02 Feb '10

02 Feb '10

The TLS wiki page talks about [1] Freeswitch being able to act as a past though proxy for a SIP phone. Turning an unencrypted SIP+RTP session into a SIPS+SRTP session. Is there howto guide for the above? I'm also wondering if Freeswitch could do a drop-in encryption. ie. Phone <-> Asterisk replaced with Phone <-> FS <==> FS <-> Asterisk, where <==> is encrypted. Nicholas [1] http://wiki.freeswitch.org/wiki/Tls#Hybrid_Encryption

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Re: [Freeswitch-users] Logging question
by paul gore 02 Feb '10

02 Feb '10

I tried "sofia profile internal siptrace on" from console and got SIP trace on the console, but nothing in logs still.

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Limit the extension creation
by Scott Fernandez 02 Feb '10

02 Feb '10

Hi, Is there a way to restrict the number of extension that FS supports/serves?. The idea is to limit the concurrent usage of the system for which we need to restrict the FS to support upto a predefined no. of users/extensions. Can anyone assist please? Thanks, Scott.

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mod_lcr problem
by Mouncif Benniane 01 Feb '10

01 Feb '10

i can't make use of mod_lcr using Intra/Interstate rating, I am using svn: FreeSWITCH Version 1.0.trunk (16517) lcr mysql table structure: CREATE TABLE `lcr` ( `id` INT(11) NOT NULL AUTO_INCREMENT, `digits` VARCHAR(15) DEFAULT NULL, `rate` FLOAT(11,5) DEFAULT NULL, `intrastate_rate` FLOAT(11,5) DEFAULT NULL, `intralata_rate` FLOAT(11,5) DEFAULT NULL, `carrier_id` INT(11) NOT NULL, `lead_strip` INT(11) NOT NULL, `trail_strip` INT(11) NOT NULL, `prefix` VARCHAR(16) NOT NULL, `suffix` VARCHAR(16) NOT NULL, `lcr_profile` VARCHAR(32) DEFAULT NULL, `date_start` DATETIME NOT NULL DEFAULT '1970-01-01 00:00:00', `date_end` DATETIME NOT NULL DEFAULT '2030-12-31 00:00:00', `quality` FLOAT(10,6) NOT NULL, `reliability` FLOAT(10,6) NOT NULL, `cid` VARCHAR(32) NOT NULL DEFAULT '', `enabled` TINYINT(1) NOT NULL DEFAULT '1', PRIMARY KEY (`id`), KEY `carrier_id` (`carrier_id`), KEY `digits` (`digits`), KEY `lcr_profile` (`lcr_profile`), KEY `digits_profile_cid_rate` USING BTREE (`digits`), CONSTRAINT `carrier_id` FOREIGN KEY (`carrier_id`) REFERENCES `carriers` (`id`) ON DELETE CASCADE ON UPDATE CASCADE ) ENGINE=INNODB AUTO_INCREMENT=6 DEFAULT CHARSET=latin1 *lcr_admin show profiles* Name: default custom sql: SELECT l.digits, c.carrier_name, l.${lcr_rate_field}, cg.prefix AS gw_prefix, cg.suffix AS gw_suffix, l.lead_strip, l.trail_strip, l.prefix, l.suffix , cg.codec , l.cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC, ${lcr_rate_field}, quality DESC, reliability DESC, rand(); has %: false has vars: true has intrastate: true has intralata: true has npanxx: true Reorder rate: enabled Info in headers: disabled Quote IN() List: disabled *lc**r 617642 default* returns rate from the rate field table and not intra/inter state fields rates. Any ideas? thanks!

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Logging question
by paul gore 01 Feb '10

01 Feb '10

Hi there, I am running FS 1.0.trunk (14501) (I know it's old but we serve a small community and don't have time to upgrade/test the latest/greatest). I am having troubles understanding how to switch SIP trace in log files, I tried fsctl loglevel debug sofia tracelevel debug but it seem to have no effect, I only get sofia debug messages but no detailed SIP info. What also puzzling me is if I do console loglevel 0 I still get debug information on console. What am I doing wrong?

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PRI Simulator
by Jerry Richards 01 Feb '10

01 Feb '10

Can anyone recommend a good PRI simulator? Sorry this is off topic a bit. Thanks, Jerry

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Presence Change Distribution
by Jerry Richards 01 Feb '10

01 Feb '10

Is there a setting to control how fast FS distributes presence changes to subscribers? Currently, it appears to take several minutes before I see presence changes. I would like to see them almost instantaneously, if possible. Thanks and Best Regards, Jerry

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conference and xmpp
by Frank Carmickle 01 Feb '10

01 Feb '10

Hello I am trying to interact with conferences over xmpp. I want to be able to see status changes and mute/unmute etc... I can only list the members if I ask for a list. I am having trouble understanding the wiki entree on this topic. Profile names are confusing because all the examples show default as a name so it is hard to see what each profile name corresponds to. My config looks like this currently. <profile name="ourchatperms">  <user name="shawn(a)djernes.net" commands="all"/>  <user name="frankiec(a)djernes.net" commands="|deaf|dial|energy|kick|list|lock|mute|norecord|play|record|relate|say|saymember|stop|transfer|undeaf|unlock|unmute|volume_in|volume_out|"/> </profile> </chat-permissions> <profile name="cdquality"> <param name="caller-controls" value="custom-ewr"/> <param name="chat-permissions" value="ourchatperms"/> <param name="domain" value="$${domain}"/> Also we can only talk to the conference by conf+confnum-fqdn@fqdn. Is there a way to make it conf+confnum@fqdn? Any help would be greatly appreciated. Thank you. --FC

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