March 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Route to the profile and not the gateway
by Anto 05 Mar '12

05 Mar '12

Hello Once I download the XML from the server and see it with an editor, I have resolved many questions of how FreeSWITCH, but even I have several questions I have not found information (I spend all day reading the wiki). I am setting up multiple providers and use a format like this (do not know if it is the best): <profile name="provider1"> <gateways> <X-PRE-PROCESS cmd="include" data="provider1/*.xml"/> </gateways> ......... Is there any way to set the extension when not having to configure all the gateway? (only use the profile). Sample dialplan: <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/> <action application="bridge" data="sofia/gateway/provider1-1/$1"/> <action application="bridge" data="sofia/gateway/provider1-2/$1"/> <action application="bridge" data="sofia/gateway/provider1-3/$1"/> ........... <action application="bridge" data="sofia/PROFILE/$1"/> ???? When set incoming calls, not very well how, not whether to route directly to dialplan or create a profile for this. I have to be routed to multiple internal servers (load balancer, failover, etc), so the above scenario I could serve to cover this. Thanks !! Best regards Anto

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RTMP Codec
by rsaavedra＠ecogizmos.com 05 Mar '12

05 Mar '12

Hello everybody, Using the Flex client of mod_rtmp can I specify the codec to use? Thank you, Ricardo Saavedra

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ph_tor3_e1.c
by Anita Hall 05 Mar '12

05 Mar '12

We are running FreeTDM on a very cheap Atcom card, which used another module ph_tor3_e1 on top of Dahdi. I believe this is derived from Torrenta. http://www.atcom.cn/downloads/TelephonyCard/drivers/AX-4ET/E1/ph_tor3_e1.c On Ubuntu 10.04 this gives problem as the module ph_tor3_e1 (and hence dahdi) does not unload. Sometimes the machine hangs and needs to be rebooted. This module has not been updated for the last 2 years during which the linux kernel has changed (I am told). Is there any other manufacturer of torrent card who would be using the same architecture and keeping his drivers updated ? If not, what steps do I need to take to update this driver to kernel version 2.6.32-37-server ? These are new waters and I feel so helpless :) regards, Anita

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302 redirect variable
by Miha Zoubek 05 Mar '12

05 Mar '12

Hi, in my directory I set variable password (<variable name="password" value="52166"/>) for every user. After I am doing 302 redirect in my public dialplan and transfer call to extension, I can not use varible pasword. How can I get varible password, so that I can authenticate call. public dialplan: <anti-action application="log" data="INFO ################# CALL FORWARD ################\n"/>   <anti-action application="info"/> <anti-action application="set" data="process_cdr=false"/> <anti-action application="execute_extension" data="IZVEDI_PREUSMERITEV XML default"/> default dialplan: <condition field="destination_number" expression="IZVEDI_PREUSMERITEV" /> <condition field="${sip_redirect_contact_user_0}" expression="^0(\d+)$" > <action application="set" data="process_cdr=true"/> <action inline="true" application="set" data="CALLINGNUMBER=${sip_redirect_contact_user_0}"/> <action inline="true" application="set" data="USERNAME=${sip_req_user}"/> <action application="info"/> <action application="set" data="effective_caller_id_name=${sip_req_user}"/> <action application="set" data="origination_caller_id_name=${sip_req_user} "/> <action inline="true" application="set" data="PASSWD=${password}"/> <action application="execute_extension" data="RADIUS_ANI_AUTH XML default"/> <action application="enum" data="386${sip_redirect_contact_user_0:1} enumsbc.softnet.si"/> <action application="bridge" data="{origination_callee_id_name='${effective_caller_id_name}'}${enum_auto_route}"/> <action application="info"/> <action application="bridge" data="sofia/external/386${sip_redirect_contact_user_0:1}@xxx.xxx.xxx.xxx" />

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Voicemail Transcription
by Balamurugan Mahendran 05 Mar '12

05 Mar '12

Hi all, We are looking for a tool to transcribe voicemail, I was trying with Sphinx but the quality is not okay to use. Please suggest me the right tool for Voicemail Transcription. Thanks, Bala

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send_dtmf fails on sla barge in
by Bob McCarthy 05 Mar '12

05 Mar '12

dialplan default.xml has <action application="set" data="dtmf_type=info"/> prior to the bridge internal.xml has no dtmf statements uncommented when sending dtmf from the Polycom dialpad after a barge in using SLA, (also after the barging phone hangs up) I get the following error: 2012-03-01 03:05:53.841090 [ERR] switch_ivr_async.c:547 dmachine overflow error! FreeSWITCH Version 1.0.head (git-23645b6 2012-02-27 16-49-12 -0600) Bob McCarthy

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Re: [Freeswitch-users] Use of ESL and PHP for Call
by Shahzad Bhatti 05 Mar '12

05 Mar '12

Hi, Thanks for the tip but i want to know that what hsndler.php code is and where to put that file i means path of file. second i have an error on this example and log show like that in fs_cli 2012-03-05 22:36:53.293238 [ERR] mod_native_file.c:74 Error opening /usr/local/freeswitch/sounds/en/us/callie/&speak:cepstral|callie|Hi,.PCMU 2012-03-05 22:36:53.293238 [ERR] mod_native_file.c:74 Error opening /usr/local/freeswitch/sounds/en/us/callie/we.PCMU 2012-03-05 22:36:53.293238 [ERR] mod_native_file.c:74 Error opening /usr/local/freeswitch/sounds/en/us/callie/&speak:cepstral|callie|Hi,.PCMU 2012-03-05 22:36:53.293238 [ERR] mod_native_file.c:74 Error opening /usr/local/freeswitch/sounds/en/us/callie/we.PCMU 2012-03-05 22:36:53.293238 [ERR] mod_native_file.c:74 Error opening /usr/local/freeswitch/sounds/en/us/callie/&speak:cepstral|callie|Hi,.PCMU 2012-03-05 22:36:53.293238 [ERR] mod_native_file.c:74 Error opening /usr/local/freeswitch/sounds/en/us/callie/we.PCMU 2012-03-05 22:36:53.293238 [WARNING] switch_ivr_play_say.c:2072 PAGD failure! Transfer to: on / Monday / at Please Guide me Regards Shahzad Bhatti On Sat, Mar 3, 2012 at 5:02 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Use of ESL and PHP for Call (Michael Collins) > 2. Re: a question about freeswitch conference caller-controls > (Michael Collins) > 3. Re: *** No rule to make target `tport/libtport.la', needed by > `libsofia-sip-ua.la'. (Ken Rice) > > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Fri, 2 Mar 2012 15:49:10 -0800 > Subject: Re: [Freeswitch-users] Use of ESL and PHP for Call > Shahzad, > > I think you may have taken an unnecessarily complicated approach to this. > If I were you I would simply use the play_and_get_digits application which > does all the work for you. I would also handle that in the XML dialplan and > then use the api_hangup_hook to call your script to process the result. > (I.e. did the user press 1 or 2 or something else...) > > I recommend you start with something simple like a basic originate command > that calls the user and then drops that call to a dialplan extension: > > originate user/1500 meeting_request_1500 > > Then create that extension: > > <extension name="meeting request"> > <condition field="destination_number" expression="^meeting_request_( > \d+)$"> > <action application="set" data="target_user=$1"/> > <action application="set" data="text=cepstral|callie|Hi, we have a > meeting on Monday at 9, are you joining us, if yes then press, 1. and if > not press, 2."/> > <action application="answer"/> > <action application="sleep" data="1000"/> > <action application="play_and_get_digits" > data="1 1 3 5000 # speak:${text} > ivr/ivr-that_was_an_invalid_entry.wav answer \d"/> > <action application="set" data="api_hangup_hook=system handler.php > ${target_user} ${answer}"/> > <action application="log" data="INFO User ${target_user} responded > with ${answer}"/> > <action application="playback" data="ivr/ivr-Thank_you.wav"/> > <action application="hangup"/> > </condition> > </extension> > > Then you just need to write your handler.php script (or whatever you want > to call it) and have it parse the command line args - the first one is the > user and the second one is the digit pressed. PHP scripts will work for > hangup hooks but they aren't the best choice since there isn't a "mod_php" > like there is a mod_lua or mod_perl. Those modules let you do cool things > like "session_in_hangup_hook." However, what you're doing doesn't look like > it's very difficult so the example I've given should get you going. Please > note that I just did this off the top of my head w/o testing it so be sure > double-check everything if you run into any unusual or unexpected behavior. > > Keep in mind that in this example you will need two different PHP scripts > - one that actually initiates the ESL connection and does the originate and > one that handles the call results. Instead of one semi-complicated script > you will have two very easy scripts. > > Hope this helps... > -MC > > On Fri, Mar 2, 2012 at 4:14 AM, Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com>wrote: > >> >> Hi, >> >> i park the call and then execute the speak command as but execute is not >> working any idea ? >> >> $cmd = "api originate {origination_uuid=$uuid}user/$user &park()"; >> $text = "Hi, we have a meeting on Monday at 9, are you joining us, if >> yes then press, 1. and if not press, 2."; >> >> $e = $esl->sendRecv($cmd); >> $esl->execute('speak',"(cepstral|callie|$text)",$uuid); >> >> >> >> my complete code is >> >> <?php >> >> require_once('ESL.php'); >> >> $host_name = 'localhost'; >> $port = '8021'; >> $password = 'ClueCon'; >> >> $esl = new ESLconnection($host_name,$port,$password); >> >> $e = $esl->sendRecv("api create_uuid"); >> $uuid = $e->getBody(); >> $user = '1500'; >> >> $cmd = "bgapi originate {origination_uuid=$uuid}user/$user &park()"; >> $text = "Hi, we have a meeting on Monday at 9, are you joining us, if >> yes then press, 1. and if not press, 2."; >> >> $e = $esl->sendRecv($cmd); >> $esl->execute('speak',"(cepstral|callie|$text)",$uuid); >> >> >> $e = $esl->sendRecv("events plain all"); >> $e = $esl->filter("Unique-ID",$uuid); >> >> while ($esl->connected()) >> { >> $e = $esl->recvEvent(); >> $result = $e->getType(); >> if($result=='CHANNEL_EXECUTE_COMPLETE') >> { >> // what i do here...! >> $esl->execute("&sleep",10000); >> } >> $digit = $e->getHeader("DTMF-Digit"); >> $state_no = $e->getHeader("Channel-State-Number"); >> $curr_uuid = $e->getHeader("Channel-Call-UUID"); >> >> /* >> $ans_state = $e->getHeader("Answer-State"); >> $state_no = $e->getHeader("Channel-State-Number"); >> $core_id = $e->getHeader("Core-UUID"); >> */ >> if($curr_uuid==$uuid && $digit!=NULL) >> { >> print "input number is " . $digit . "\n"; >> } >> $state = $e->getHeader("Channel-Call-State"); >> if ($state == 'HANGUP') { >> print "\n\nHangup Cause Number: " . $state_no; >> $esl->disconnect(); >> } >> } >> >> ?> >> >> >> Regards >> >> Shahzad Bhatti >> >> >> ---------- Forwarded message ---------- >> From: Peter Olsson <peter.olsson(a)visionutveckling.se> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Fri, 2 Mar 2012 05:31:35 +0000 >> Subject: Re: [Freeswitch-users] Use of ESL and PHP for Call >> You should pass the originate command to &park() instead. >> >> Right now you originate the call and then execute the application speak. >> After this (when speak is finished) FS has no more instructions so it will >> hangup the call. >> >> If you execute &park() instead, the call will be "parked" and wait for >> furter instructions. Then you will also need to execute the speak from ESL >> instead - and afther that you continue to do whatever you want to do with >> the call. >> >> /Peter >> >> _________________________________________________________________________ >> 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 >> >> 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 >> >> > > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Fri, 2 Mar 2012 16:01:44 -0800 > Subject: Re: [Freeswitch-users] a question about freeswitch conference > caller-controls > Try this in your dialplan before sending the call to the conference: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf > -MC > > On Thu, Mar 1, 2012 at 5:30 PM, Erjian Li <eli(a)netspectrum.com> wrote: > >> When I press cellphone's key, although Freeswitch can't see this DTMF, >> but the the other participant of the conference call can hear my DTMF tone. >> Does this situation indicates that the DTMF tone has been forwarded by SIP >> provider's server? >> >> > > ---------- Forwarded message ---------- > From: Ken Rice <krice(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Fri, 02 Mar 2012 18:02:02 -0600 > Subject: Re: [Freeswitch-users] *** No rule to make target `tport/ > libtport.la', needed by `libsofia-sip-ua.la'. > This is because Apple Deprecated SSL on Lion... They don’t want you > using it... > > Configure ‘--without-openssl’ and the problem goes away... > > There is an open jira on this issue I just cant recall the bug number off > the top of my head > > > On 3/2/12 4:57 PM, "Stéphane Lux" <stephane(a)lux.io> wrote: > > I have tried to compile it again: > > > tport_tls.c: In function 'tls_error': > tport_tls.c:665: warning: 'SSL_get_error' is deprecated (declared at > /usr/include/openssl/ssl.h:1501) > tport_tls.c:680: warning: 'SSL_get_shutdown' is deprecated (declared at > /usr/include/openssl/ssl.h:1568) > tport_tls.c: In function 'tls_read': > tport_tls.c:723: warning: 'SSL_read' is deprecated (declared at > /usr/include/openssl/ssl.h:1493) > tport_tls.c: In function 'tls_pending': > tport_tls.c:739: warning: 'SSL_pending' is deprecated (declared at > /usr/include/openssl/ssl.h:1368) > tport_tls.c: In function 'tls_write': > tport_tls.c:807: warning: 'SSL_write' is deprecated (declared at > /usr/include/openssl/ssl.h:1495) > tport_tls.c: In function 'tls_connect': > tport_tls.c:899: warning: 'SSL_accept' is deprecated (declared at > /usr/include/openssl/ssl.h:1491) > tport_tls.c:899: warning: 'SSL_connect' is deprecated (declared at > /usr/include/openssl/ssl.h:1492) > tport_tls.c:900: warning: 'SSL_get_error' is deprecated (declared at > /usr/include/openssl/ssl.h:1501) > tport_tls.c:954: warning: 'ERR_error_string_n' is deprecated (declared at > /usr/include/openssl/err.h:280) > make[9]: *** [tport_tls.lo] Error 1 > make[8]: *** [all] Error 2 > Making all in nta > Making all in nth > Making all in nea > Making all in iptsec > Making all in nua > make[8]: *** No rule to make target `tport/libtport.la', needed by ` > libsofia-sip-ua.la'. Stop. > make[7]: *** [all-recursive] Error 1 > Making all in packages > make[6]: *** [all-recursive] Error 1 > make[5]: *** [all] Error 2 > make[4]: *** [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/ > libsofia-sip-ua.la] Error 2 > make[3]: *** [mod_sofia-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > > On 02.03.2012, at 23:42, Michael Collins wrote: > > We need to see the output that comes prior to this error. Grab the > previous 20+ lines of output prior to this error and let us take a look. > Hopefully the true cause will be apparent. > > -MC > > 2012/3/2 Stéphane Lux <stephane(a)lux.io> > > Hi, > > when compiling FS on Mac OS 10.7.3 I get this error: > > make[8]: *** No rule to make target `tport/libtport.la < > http://libtport.la/> ', needed by `libsofia-sip-ua.la < > http://libsofia-sip-ua.la/> '. Stop. > > How can I solve it? > > Kind Regards, > Stephane > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com <http://www.cudatel.com/> > > Official FreeSWITCH Sites > http://www.freeswitch.org <http://www.freeswitch.org/> > http://wiki.freeswitch.org <http://wiki.freeswitch.org/> > http://www.cluecon.com <http://www.cluecon.com/> > > 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 <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 > > 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 > > 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 > > > _______________________________________________ > 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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What is the best way to predictive dial with FreeSwitch? What is the best way to pull real-time data from FreeSwitch?
by Ast Coder 05 Mar '12

05 Mar '12

Hello everyone, New to FreeSwitch here. Trying to make the change from Asterisk to FreeSwitch but need to plan ahead for our GUI implementation. As I read more of the documentation, I would appreciate some guidance on the following: - What is the best way to pull and display real-time call status from FreeSwitch to a web GUI. E.g. extension status, call connection status, etc... - What is best way to do predictive dialling? I guess I am looking for something like Asterisk spool files or the AMI equivalent. Thanks,

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Re: [Freeswitch-users] how to use " remote_media_ip, remote_media_port" in dialplan
by Peter Olsson 05 Mar '12

05 Mar '12

My guess is that it's set as soon as media is up, so either during early media or when the call has been answered. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För hjz51 Skickat: den 5 mars 2012 07:56 Till: 'freeswitch-users' Ämne: [Freeswitch-users] how to use " remote_media_ip, remote_media_port" in dialplan Hi, I tried to use the variable remote_media_ip, remote_media_port within dialplan, but it is not returning anything. Does anyone know when this variable gets set and how to have this variable to be set as soon as an INVITE hit freeswitch? Thanks, !DSPAM:4f54629c32761872512829!

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uuid in spandsp log
by Anita Hall 05 Mar '12

05 Mar '12

Hi I have verbose enabled in fax which gives me all the lower level fax related message transfers by spandsp. Also, uuid is enabled so all log lines are preceded by uuid. But spandsp related logs do not have a uuid. How can I enable uuid in spandsp log ? Thanks. regards, Anita

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