January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 43, Issue 187
by Jerry Richards 22 Jan '10

22 Jan '10

I tried sending this Email earlier in the week but it got kicked back by our Mail Server, so if this is a duplicate, my apologies... Are you still working toward a release version 1.0.5? Best Regards, Jerry

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Choppy conference audio
by Tim St. Pierre 22 Jan '10

22 Jan '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello all, I'm having a tough time sorting out the audio in mod_conference, I'm hoping someone can point me in the right direction. I took a good look around the wiki, but couldn't really find what I was looking for. I have a test system set up, with three Aastra 9133i, a 6731i, and a 57i. All register, call fine, etc. Music on hold audio is perfect. Calls between phones are perfect. CPU load on the lab machine is about 8%, so not really busy. Conference audio is choppy, in the way that things get choppy when there is some sort of timing issue. Even with one member in the conference, the prompts and moh sound choppy. I played around with the rate and interval values, there wasn't any change. I tried disabling the monotonic timer, and that didn't change anything either. I'm running FreeBSD 7, with the latest release of FS downloaded last week. Thanks for any help. Let me know if I need to post any other information. - -Tim -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.4 (FreeBSD) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQCVAwUBS1oK2YqVcvNCnHOrAQLriAP/csjIfD/VP0CA3ePyRBKXbDPfxEiqx/cP iWILZ65F0bxaryKTYT2ZV8W7u+wH6YxMCaoej+H+yd1XG08hZr4kLUcewUaCTMna S8Zd24ReknEmL8d0AXCMgospf2wmwZZx2pJMFmbPN3IlXup20dKWOESBp/Dru3vL 63GE8s6AdyA= =ndAy -----END PGP SIGNATURE-----

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skypiax and xml-curl
by Peter P GMX 22 Jan '10

22 Jan '10

Hello, is there a way to manage skypiax via XML-curl besides the dialplan? Best regards Peter

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Wanpipe Driver Install Before FS Install
by Jerry Richards 22 Jan '10

22 Jan '10

Is it true that the Sangoma Wanpipe Driver should be installed before Freeswitch, because the Freeswitch build will autodetect the wanpipe drivers? Thanks, Jerry

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Script ends when originate receives INVALID_NUMBER_FORMAT
by Oscav 22 Jan '10

22 Jan '10

Hi, My script ends when I received a 484 INVALID_NUMBER_FORMAT, and doesn't continue even the script or a hangup Hook. Here is the script : session.setVariable("hangup_after_bridge",false); ... session.setVariable("continue_on_fail","true"); ... session.preAnswer(); ... new_session = new Session(route,session); new_session.waitForAnswer(15000); if (new_session.ready()) { bridge(session, new_session); } ... console_log("info","call ended") ... Is there something missing on my script ?? Thanks. -- View this message in context: http://old.nabble.com/Script-ends-when-originate-receives-INVALID_NUMBER_FO… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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No external sangoma calls running FS as daemon
by Robert Hadley 22 Jan '10

22 Jan '10

I have freeswitch running as daemon as user freeswitch. Internal calls work. I get 503 service unavailable when I make external calls through sangoma. If I change the FS daemon to run as root then external sangoma calls work. Does anybody know what permissions I need to fix? Thanks, Robert

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FreeSWITCH Weekly Call Starting in 30 min
by Michael Collins 22 Jan '10

22 Jan '10

Hello all, The weekly conference call will be starting in a bit. Here's the agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_22 I'll be on just after 9am my time as I have to get my kids off to school. Talk to you all soon. -Michael

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Re: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
by Anthony Minessale 22 Jan '10

22 Jan '10

You don't need loopback if you were calling over sofia back to your own box, you just need to use a dest that could be regexed to right to vm when it comes back around. The loopback to an app should work you probably just have a syntax err On Jan 21, 2010 11:03 PM, "Troy Anderson" <troy(a)tlainvestments.com> wrote: I get the idea, but can't seem to get it to work. I tried doing a bridge to "loopback/app=bridge ${sofia_contact(${dialed_user}@${dialed_domain}", but fs complained => Cannot create outgoing channel of type [loopback=app:sofia] Also, I tried "loopback/app=voicemail:default ${domain_name} ${dialed_extension}" and it did get to voicemail, but it didn't prompt me for any info - just immediately complained that the recording was too short. I tried the suggestions about modifying the sip.cfg for the phones and that does work (thanks!), but that forces a Consultative Transfer. It would be nice to get this method working, which would result in the transfer button doing a Consultive Transfer unless you hang up, then it would be like a blind transfer. Thanks, Troy On Jan 21, 2010, at 5:48 PM, Anthony Minessale wrote: > if you used the loopback endpoint to loop... _______________________________________________ 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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Server Disconnected when SIGINT occured
by lakshmanan ganapathy 22 Jan '10

22 Jan '10

Hi all, I've done a sample program (In perl ESL) , which play a file to the caller and then it will call recvEvent() and print the event name. I've handled signals also. When I send SIGINT to my program (Perl), the signal handler is called and I can see the print output. But in the same time, I received SERVER_DISCONNECTED from freeswitch as event. I don't know why I received SERVER_DISCONNECTED from freeswitch. Is it because, the recvEvent() from perl internally calls the recvevent function in the Esl.c and when it waits to receive the information from socket, the signal occurred??? Please clarify me!! Here is my program require ESL; use IO::Socket::INET; use Data::Dumper; 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; &register_Signals(); for(;;) { my $new_sock = $sock->accept(); next if (not defined ($new_sock)); my $pid = fork(); if ($pid) { close($new_sock); next; } print "CHILD PID: $$\n"; my $host = $new_sock->sockhost(); my $fd = fileno($new_sock); my $con = new ESL::ESLconnection($fd); my $info = $con->getInfo(); my $uuid = $info->getHeader("unique-id"); printf "Connected call %s, from %s\n", $uuid, $info->getHeader("caller-caller-id-number"); my $r=$con->execute("answer"); print Dumper $r; $con->events("plain","all"); my $re=$con->execute("playback","/usr/local/freeswitch1/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav"); while($con->connected()) { my $e = $con->recvEvent(); if ($e) { my $name = $e->getHeader("event-name"); print "EVENT [$name]\n"; if ($name eq "DTMF") { my $digit = $e->getHeader("dtmf-digit"); print "$digit\n"; } } } close($new_sock); } sub register_Signals() { foreach ( keys %SIG ) { $SIG{$_} = 'sig_Handler'; } } sub sig_Handler() { my $handle=$_[0]; if($handle eq "INT") { print "$$: SIGNAL SIG$handle is generated\n"; } }

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Can't register Polycom
by Lars Zeb 21 Jan '10

21 Jan '10

I am having trouble registering a Polycom 550. From the siptrace it looks like there is no username coming from the Polycom. I configured the Polycom via the web interface. I have inserted "1008" in the "Display Name", "Address" and "Auth User ID" fields. In conf/dialplan/default/1008.xml the first line is <user id="1008" cidr="192.168.10.105/24">. What am I missing? Thanks, Lars http://pastebin.freeswitch.org/11875

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