January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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!

3 19

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?

2 2

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

4 4

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

3 9

Wrong RTP port submitted?
by Peter P GMX 01 Feb '10

01 Feb '10

I have defined the rtp port range for 12000-12100 in switch.conf.xml. However Freeswitch is offering a port 48320 in the invite message. The result is, that the incoming RTP stream is blocked by the firewall (I can see a reject for UDP 48320). Any hint how to solve this? Best regards Peter See config and invite message:  <param name="rtp-start-port" value="12000"/>--> <param name="rtp-end-port" value="12100"/>--> Invite: INVITE sip:027xxxxxxxx@sip.itsp.de SIP/2.0. Via: SIP/2.0/UDP 217.24.xx.xxx:5080;rport;branch=z9hG4bKjD923NvctXaFm. Max-Forwards: 69. From: "0608xxxxxxx" <sip:4918xxxxxxxxxxxx@sip.itsp.de;transport=udp>;tag=0Kp4tvU44UmXp. To: <sip:027xxxxxxxx@sip.itsp.de>. Call-ID: 30c86b94-85ca-122d-f88e-080027e51f59. CSeq: 126174137 INVITE. Contact: <sip:gw+tgnet3@217.24.xx.xxx:5080;transport=udp;gw=tgnet3>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16032. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 320. P-Key-Flags: keys="3". X-FS-Support: update_display. Remote-Party-ID: "0608xxxxxxx" <sip:0608xxxxxxx@sip.itsp.de>;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1264536651 1264536652 IN IP4 217.24.xx.xxx. s=FreeSWITCH. c=IN IP4 217.24.xx.xxx. t=0 0. m=audio 48320 RTP/AVP 8 0 98 3 101 13. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:98 SPEEX/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20.

3 2

Equivalent to Asterisk's "directrtpsetup=yes"?
by Fred-145 01 Feb '10

01 Feb '10

Hello Thanks to "directrtpsetup=yes", it appears that Asterisk can act as an SIP proxy and a PBX at the same time, ie. RTP packets will flow directly between the two SIP end-points with Asterisk still being able to provide PBX services like call transfer, MoH, etc. The point is to lower latency for packets to reach their final destination and lower CPU load on the Freeswitch server. Does Freeswitch offer the same feature, or must RTP packets always go through the Freeswitch servers? Thank you. -- View this message in context: http://old.nabble.com/Equivalent-to-Asterisk%27s-%22directrtpsetup%3Dyes%22… Sent from the Freeswitch-users mailing list archive at Nabble.com.

9 18

nixevent behavior
by lakshmanan ganapathy 01 Feb '10

01 Feb '10

Dear all I've done the following sample script to experiment the nixevent. I found some difference in behavior because of nixevent. Let me explain my question down the script. require ESL; use IO::Socket::INET; my $ip = "192.168.1.222"; my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8447', Proto => 'tcp', Listen => 1, Reuse => 1 ); die "Could not create socket: $!\n" unless $sock; for(;;) { my $new_sock = $sock->accept(); next if (not defined ($new_sock)); my $pid = fork(); if ($pid) { close($new_sock); next; } print "CHILD PID: $$\n"; my $host = $new_sock->sockhost(); my $fd = fileno($new_sock); my $con = new ESL::ESLconnection($fd); my $info = $con->getInfo(); my $uuid = $info->getHeader("unique-id"); printf "Connected call %s, from %s\n", $uuid, $info->getHeader("caller-caller-id-number"); my $r=$con->execute("answer"); $con->events("plain","all"); ########################## $con->send("nixevent DTMF"); my $val=$con->api("create_uuid"); $val = $val->getBody(); # LINE 1 chomp($val); print "UUID is $val\n"; my $e = $con->recvEvent(); $val = $e->getBody(); # LINE 2 chomp($val); print "UUID is $val\n"; close($new_sock); } # If the line ($con->send("nixevent DTMF");) is commented, then the result of create_uuid is obtained in LINE 1. # else, the result isn't obtained in the LINE 1 and it has nothing. The result is obtained only when I do a recvEvent, # followed by a getBody (LINE 2) Just want to know why the behavior differs when nixevent is present???

2 2

determining the source of receyved call in public context
by Meftah Tayeb 01 Feb '10

01 Feb '10

hi, how do i determine the gateway or the Ip of a receyved call from ITSP's? i am calling my did from my mobile, but i see is processing the the mobile number no the ITSP User or did thanks

2 1

How to populate Voice recording metadata with custom CRM data?
by Roly Maz 01 Feb '10

01 Feb '10

Hi All, I am using Freeswitch 1.0.4 (Latest Windows Installer version) on Windows 7 Ultimate for testing Assuming an outbound call, what would be the most sensible approach to pass custom CRM data into the voice-recording? I would like the voice recording of the call to include the customers social security number in the file title, or even metadata. In other words, is there a way to pass custom info, at time of call, to the dialplan for use in the creating the voice recording. Any pointers would be much appreciated Rgds Roly

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Re: [Freeswitch-users] Freeswitch core dump after upgrade to latest version
by Yehavi Bourvine 01 Feb '10

01 Feb '10

It works ok now (fixed on r16534). Thanks! __Yehavi: 2010/1/25 Yehavi Bourvine <yehavi.bourvine(a)gmail.com> > OK. The sources are under /home/freeswitch. Note that it has Hebrew modules > which we are now developing, but the problem is seen with the vanilla > version as well. > > The execs and all other stuff is at /freeswitch. > > regards, __Yehavi: > > 2010/1/25 Michael Jerris <mike(a)jerris.com> > > I have not had a chance to actually try it yet. I will let you know. >> >> Mike >> >> On Jan 25, 2010, at 2:43 PM, Yehavi Bourvine wrote: >> >> > Hello Mike, >> > >> > I see you are logged-in into our machine. Before I go to sleep: Do you >> need any help? >> > >> > Thanks, __yehavi: >> > >> >>

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