February 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Phones losing registration
by Dave Stevenson 02 Feb '10

02 Feb '10

Hi, I have a mixture of phones and am having a problem with some of them periodically losing registration with FreeSwitch. All the phones start off working after provisioning and have registration expiry times configured of 3600 seconds. The Cisco and Thomson phones are OK, staying registered without problems but other phones, including SwissVoice IP10Ss and a cheap wireless VOIP phone lose registration after a random time, sometimes hours, sometimes days. Is this a common issue with known work-arounds or can someone point me in the right direction for how to see if any errors are being generated or if I need to turn on any additional FreeSWITCH logging options to trap the errors please ? regards Dave

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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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