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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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G729 coded issues
by Ahmed Naji 21 Jan '10

21 Jan '10

Hi everyone, I have the following scenario and a major customer-affecting issue thereof. Here is the scenario: customer traffic encoded as G.729 from a cisco gateway -> our FS (G729 passthrough) -> remote end gw (G729) Calls were failing at an alarming rate, so I looked at the debug logs. It transpired that the Cisco is offering G729 annex b, while the remote end can only do G729a. Besides changing source or destination preferences, is there a way to ensure that G729a is used end-end ? Thanks in advance. -- Ahmed Naji

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All channels are frozen while receiving DTMF
by Oscav 21 Jan '10

21 Jan '10

Hi, I'm running a script that gets some DTMF from caller. I found that when a caller is entering DTMF , all the others channels are frozen until all the DTMF are received. In the logs I see that each DTMF takes 1 second. It means that if the caller enters 10 digits then all the other running scripts are paused for 10 seconds. The problem is exponential with traffic load. Anyone have an idea ?? Thanks -- View this message in context: http://old.nabble.com/All-channels-are-frozen-while-receiving-DTMF-tp272551… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Additional endpoints
by Lars Zeb 21 Jan '10

21 Jan '10

I know this has been answered before, but I cannot find it. How do I setup more than the default 20 endpoints (1000-1019)? Do I extend the definition in dialplan/public.xml (public_extensions) and add the extra in directory/default? Thanks, Lars

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