June 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Issue with sending fax with t.38
by James Lyons 26 Jun '12

26 Jun '12

I'm attempting to use txfax in mod_spandsp to send a fax over t.38 to a fax machine at a PSTN number. The receiving side hangs up, after appearing to send a number of malformed RTP packets. "Ignoring invalid RTP packet size of 8 bytes." Can anyone explain this? opensips proxy is 1.2.3.4 fs gw is 1.2.3.5 fs originator is 1.2.3.6 Thanks, James

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Re: [Freeswitch-users] softphones and background noise
by Juan Pablo L. 26 Jun '12

26 Jun '12

Hi, thanks, Yes, i have tried many on android, linux and wndows, the same problem with all of them. Yes i have tested with a hardphone, a polycom, and its fine. -----Original Message----- From: Mitch Capper Sent: 25 Jun 2012 16:28:40 GMT To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] softphones and background noise Try Other softphones? Are you using a headset? ~Mitch On Mon, Jun 25, 2012 at 9:19 AM, Juan Pablo L. <jpablolorenzetti(a)hotmail.com> wrote: > > > Good morning, i m having lots of issues with softphones, when a call is initiated/terminated to a softphone > the background noise is too high that you can hardly hear the actual voice. > i have played around with VAD and CNG turning them on and off in the profile and/or the dialplan but i have not had any luck. > it does not matter if the call is internal or goes through a gateway, it is however, worse when it goes through a gateway, those are the > cases you can not even hear the voice for so much noise and when the call is to a next extension the background noise is high > but not as high. > But, if you initiate/terminate the call to a hardphone (like polycom phones) the call is perfectly clear in all cases > (except when the other end is an internal softphone). > can someone please give me a hint. thanks a lot! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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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Bypass Media status and Can't re-establsh media error message
by kawarod 26 Jun '12

26 Jun '12

Hi List, I have the following error message in the console: 2012-06-26 11:29:33.399845 [ERR] switch_ivr.c:1414 Can't re-establsh media on sofia/INSIDE/1000@10.10.10.10:5060 My dialplan is the following: - set bypass_media = true (I don't want to process RTP on freeswitch) - bridge the call - if bridge is unsuccessful, I execute a perl script The Perl script catch the sip_error_code that is returned and I play an annoucement message depending on the SIP error. In the perl script I do the following: - set bypass_media = false - use pre_answer to trigger a SIP 183 cause I don't want the call to be billed - playback error message - hangup with error code Everything is working fine, I have a SIP 183, the message is played and bill_duration = 0; but I have the error message above in console. If I use answer, instead of pre_answer, the message is played and the error message disapear but bill_duration is not null. I tried to use uuid_media in my perl script but this is not working. I got an error message, I assume this is because uuid_media should be used when call is bridged. If somebody as an idea to share :p regards, Rod.

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Call State Diagram
by Guillermo Ruiz Camauer 26 Jun '12

26 Jun '12

Does a Call-State Diagram exist for FreeSwitch? I am looking for something similar to this: http://threebit.net/documentation/dialogic/resource.intel.com/telecom/suppo…. I have found a list of all the different states of a call ( http://wiki.freeswitch.org/wiki/Channel_States) but would like to see what events/actions trigger states changes to other states, etc. Thanks in advance, Guillermo Ruiz Camauer

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Call recording issue
by Balamurugan Mahendran 25 Jun '12

25 Jun '12

All, I was digging for an call recording issue with Skype calls for the past one week, looking here for any clue/help. In a different thread I was advised that its out of mod_skyopen module. Also it seems there is a problem related to file conversion and bit rating. Looks like call recorded with 16k rate but named as 8k. Is there a way to control this parameter? or where should I focus for this. note: I tried different hardware, latest source, and new configuration but could not trace it. It was good till last Friday, and there was no change in any settings as per our knowledge. Thanks for your help!! Thanks, Bala

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Re: [Freeswitch-users] softphones and background noise
by Juan Pablo L. 25 Jun '12

25 Jun '12

Hi, thanks, i m looking in the logs at the moment and playing with codecs too but it gets worse and not better thou. -----Original Message----- From: Peter Olsson Sent: 25 Jun 2012 16:32:51 GMT To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] softphones and background noise I've noticed issues sometimes when I had zrtp enabled, but I guess the reason for that was because of a bad SDP sent in the INVITE (since the endpoint didn't really support zrtp). This is surely a codec issue of some kind, start by checking the logs, and see if you find something suspicious there. /Peter 25 jun 2012 kl. 18:24 skrev "Juan Pablo L." <jpablolorenzetti(a)hotmail.com>: > > > Good morning, i m having lots of issues with softphones, when a call is initiated/terminated to a softphone > the background noise is too high that you can hardly hear the actual voice. > i have played around with VAD and CNG turning them on and off in the profile and/or the dialplan but i have not had any luck. > it does not matter if the call is internal or goes through a gateway, it is however, worse when it goes through a gateway, those are the > cases you can not even hear the voice for so much noise and when the call is to a next extension the background noise is high > but not as high. > But, if you initiate/terminate the call to a hardphone (like polycom phones) the call is perfectly clear in all cases > (except when the other end is an internal softphone). > can someone please give me a hint. thanks a lot! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > 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 > > !DSPAM:4fe88d8932761983755573! > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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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FreeSWITCH Weekly News and Notes
by Michael Collins 25 Jun '12

25 Jun '12

Welcome to the last week of June! Things quieted down a bit last week, but we still had some interesting items for your consideration. Darren Schreiber from the 2600hz<http://www.2600hz.com?cc12-0625>project joined us last week on our conference call <http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12-0625> to present "SIP 101" - a gentle introduction to the SIP protocol for those doing VoIP. It was a nice discussion and we plan to do a follow up, tentatively scheduled for July 18. The audio is available in the usual place<http://torrents.freeswitch.org/conf_call_2012-06-20.torrent?cc12-0625>and the slides are available here <http://wiki.freeswitch.org/files/SIP_101_Presentation.ppt?cc12-0625>. For this week's<http://wiki.freeswitch.org/wiki/FS_weekly_2012_06_27?cc12-0625>conference call we are going to have an open discussion about topics of interest to our community members. Our ClueCon schedule <http://www.cluecon.com/schedule?cc12-0625> is now full! Please visit the schedule to see a list of speakers and talk titles. We have a few presentations whose titles are yet to be announced, so check back often to stay updated. We are pleased to have Brad Pitt joining us this year! (No, not that one <http://en.wikipedia.org/wiki/Brad_pitt> - we have this one <http://www.cluecon.com/speaker/108?cc12-0625>.) Bradley J Pitt is a long-time veteran of the technology and telecom industries and recently joined Barracuda Networks <http://www.cudatel.com?cc12-0625>. His discussion will touch upon both technology and business aspects of the IP PBX world. As a reminder, you still have about one week before the "bits shift" and you go from 16 down to 8 tickets for your ClueCon registration<http://www.cluecon.com/register?cc12-0625>. Register right away so that you have the best chance possible to win one of the 10 Android tablets that will be given away this year! See you in Chicago! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0625

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Sharing: Call Intercept
by Frank Busalacchi Jr 25 Jun '12

25 Jun '12

Over the last 6 months, I have received all kinds of support and help from the freeswitch community (both IRC and this list) in learning to control this incredible piece of software. Im truly grateful, thanks everyone. I run freeswitch at my car dealership. We have approximately 40 sip extensions, an ATT voip connection, and 6 analog outbound fax lines provided by freetdm. Prior to freeswitch, I was running an old Lucent/Avaya Merlin "key" system. One of the features of the Merlin was that it was pretty easy to do "directed callpick". In essence, you would dial *7 + extension you wanted to pickup. Provided the call was either on hold or ringing at that extension, you would pick it up. When I first started using Freeswitch, I was very disappointed in the directed callpick. Before I realized that the call pickup that was built into the sample dialplan was really just an example to show a simple possibility for using intercept. I was convinced that it was a totally useful feature. I couldn't believe that was as good as it got... My problem with the directed pickup that is in the sample dialplan is that it can steal a live call right from another extension, as well as it really only works with the most recent call on that extension. Then I woke up, and rolled my own 8-) . In an effort to provide just another example, I wanted to share the piece of code I've developed. I will warn that this was the first perl script I have ever written, and probably is pretty ugly, but it seems to work perfectly in our environment. Bottom line, do what you want with it...I'm just trying to help other beginner's like myself: To all beginners: You will need to build and enable mod_perl for this to work... First, we need an entry in your XML dialplan to see the user's request to interecept a "held" or "ringing" extension. The only 4 digit extensions at my company are internal ones so I match on ** + 4-digit-extension. In the xML dialplan: <extension name="intercept-ext"> <condition field="destination_number" expression="^\*\*(\d{4})$"> <action application="answer"/> <action application="perl" data="/usr/local/freeswitch/scripts/myintercept.pl $1"/> <action application="sleep" data="2000"/> </condition> </extension> Then, we need the actual perl script: /usr/local/freeswitch/scripts/myintercept.pl: # This is code to allow us to use intercept to grab only calls that are on hold or ringing. # # by Frank Busalacchi 2012-06-13 use strict; use Switch; use Data::Dumper; use XML::Simple; use POSIX qw(strftime); our $session; { my $dest_target = $ARGV[0]; my $api = new freeswitch::API(); my $xml = new XML::Simple; my $chan_xml = $xml->XMLin($api->executeString("show detailed_calls as xml"),KeyAttr => 'row', ForceArray => ['row']); my %candidates; my $result; foreach my $row (@{$chan_xml->{row}}) { if ($row->{callstate} eq "RINGING" && $row->{dest} eq $dest_target) { $candidates{ $row->{created_epoch} - 1000 } = $row->{call_uuid}; #In the case of a tie, we want to pickup ringing calls before held calls... next; } elsif ($row->{callstate} eq "HELD" && $row->{b_sent_callee_num} eq $dest_target ) { $candidates{ $row->{created_epoch} } = $row->{b_uuid}; next; } elsif ($row->{b_callstate} eq "HELD" && $row->{b_callee_num} eq $dest_target ) { $candidates{ $row->{b_created_epoch} } = $row->{uuid}; next; } } if( keys( %candidates ) == 0) { $result = "[Intercept] No suitable channel to intercept on extension $dest_target."; freeswitch::consoleLog("INFO","$result\n"); } else { my $user = $session->getVariable('caller_id_name'); my $cid_num = $session->getVariable('caller_id_number'); for my $winner ( sort keys %candidates ) { my $uuid= $candidates{$winner}; $result = "[Intercept]User:$user" . "[" . "$cid_num" . "] intercepting Extension:$dest_target" . ". UUID:" . "$uuid" . "\n"; freeswitch::consoleLog("INFO","$result\n"); $session->execute("intercept",$uuid); last; } } 1; } Enjoy! -Frank

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show channels, comma vs TAB ?
by Seven Du 25 Jun '12

25 Jun '12

Hi, I use originate user/1000 &bridge({aa=bb,cc=dd}user/1002) show channels gives a comma separated list, however, as some app data contains commas, this will be hard to parse, could we change to TAB? I understand query the db directly would be fine but sometimes it's just easy to parse the cli output. ideas? thanks. uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num d412fdda-5fbc-4754-9803-2db06898a8b0,outbound,2012-06-25 13:47:03,1340603223,sofia/internal/sip:1000@192.168.2.102:26570,CS_EXECUTE,Outbound Call,1000,,1000,bridge,{aa=bb,cc=dd}user/1002,,default,PCMA,8000,64000,PCMA,8000,64000,,seven-macpro.local,1000@192.168.2.102,,ACTIVE,,,RECV,d412fdda-5fbc-4754-9803-2db06898a8b0,, b5ca402e-d72e-4109-a8ee-634e873f3c31,outbound,2012-06-25 13:47:30,1340603250,sofia/internal/sip:1002@192.168.2.102:20676,CS_CONSUME_MEDIA,Extension 1000,1000,192.168.2.102,1002,,,,default,,,,,,,,seven-macpro.local,1002(a)192.168.2.102,,RINGING,Outbound Call,1002,,d412fdda-5fbc-4754-9803-2db06898a8b0,, -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com)

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http_cache not working with latest RPM
by shadowym 25 Jun '12

25 Jun '12

I'm running Whistle or Kazoo or whatever they are changing the name to. The 2600hz cloud thingy. Anways, it uses http_cache for sound files. Using latest Freeswitch RPM it doesn't work. I found out that I needed to create /usr/share/freeswitch/http_cache directory. Without that it won't work because that is where /etc/freeswitch/autoload_configs/http_cache.conf.xml points to by default. Didn't have to do that before so I am assuming the older RPM's created that before. Not sure where I am supposed to post this bug so I am posting it here. The 2600hz guys says it's a problem with the FS RPM SPEC file.

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