May 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

create rpm fail on freeswitch-1.2.rc2
by haloha 15 May '12

15 May '12

hi list i am try to build rpm from source and i get error look like this: /usr/bin/install -c 'scripts/fsxs' '/var/tmp/freeswitch-1.2.rc2-%{BUILD_NUMBER}-root-ha/usr/bin/fsxs' mkdir: cannot create directory `/var/tmp/freeswitch-1.2.rc2-%{BUILD_NUMBER}-root-ha/usr/lib/freeswitch/mod': No such file or directory make[3]: *** [install-exec-local] Error 1 make[2]: *** [install-am] Error 2 Making install in src Making install in mod making install mod_abstraction installing mod_abstraction.so /usr/bin/install: cannot create regular file `/var/tmp/freeswitch-1.2.rc2-%{BUILD_NUMBER}-root-ha/usr/lib/freeswitch/mod': No such file or directory make[5]: *** [/var/tmp/freeswitch-1.2.rc2-%{BUILD_NUMBER}-root-ha/usr/lib/freeswitch/mod/mod_abstraction.so] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_abstraction-install] Error 1 make[2]: *** [install-recursive] Error 1 make[1]: *** [install-recursive] Error 1 make: *** [install] Error 2 error: Bad exit status from /var/tmp/rpm-tmp.46706 (%install) RPM build errors: Bad exit status from /var/tmp/rpm-tmp.46706 (%install) please help thank you

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Transfer Failure in Proxy Mode
by Andrew Paul 15 May '12

15 May '12

Hai , I am using freeswitch in proxy mode . I configured my freeswitch dialplan like this ( http://wiki.freeswitch.org/wiki/Proxy_Media ) <action application="set" data="proxy_media=true"/> <action application="bridge" data="user/(a)192.168.174.23"/> In my sip profile i made inbound-proxy-media equals true. When i am calling B from A and i am putting A side hold ,B side am able to get Music on hold ,then from A side am making one Blind transfer to C . This time freeswitch processing the refer and it is sending BYE to B and it is not furthur porcessing INVITE to C . Is there anything required to add more in configuration . <http://2222@10.20.2.49:5080/>

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mod_gsmopen requires spandsp
by Chris Mylonas 15 May '12

15 May '12

Hi FS List, FYI - as a shortcut to building my freeswitch, I skip spandsp - but it looks like this mod_gsmopen wants it in there. [root@space build]# cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ [root@space mod_gsmopen]# make clean [root@space mod_gsmopen]# make install Compiling gsmopen_protocol.cpp... Compiling /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp... mkdir .libs Compiling /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp ... Creating mod_gsmopen.so... /usr/bin/ld: cannot find -lspandsp collect2: ld returned 1 exit status g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x .libs/mod_gsmopen.o gsmopen_protocol.o -lm /usr/src/freeswitch/.libs/libfreeswitch.so -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl -lcrypto -ldl -lz -lncurses -ljpeg -L/usr/src/freeswitch/libs/spandsp/src -lspandsp -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod make[1]: *** [mod_gsmopen.so] Error 1 make: *** [install] Error 1

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Using start_dtmf on an originated call via ESL.
by Michael Lutz 15 May '12

15 May '12

Hi All, I am having some trouble trying to figure out how to enable the start_dtmf (or equivalant) for DTMF detection on generated calls using the API (ESL) with originate. My inbound leg have set this on in the dialplan based on the provider of the incomming call (1 have one provider which does not support RFC2833) My problem is that my outbound calls use that specific provider. The thing is that I can only detect DTMF from my original inbound leg (these are bridged) but not from the outbound leg. I assume I have to enable the start_dtmf on the outbound leg after the originate. But have no clue how to do that becuase I park the call after originate so my ESL server can pick it up and handle the call. I can not find an API function or equivalant for this, and I have, of course, no session object a I originate the call from ESL. The code I am using is: string cDialString = "{origination_uuid=" + thisDial.CallId_LegB + ",origination_caller_id_number=" + thisDial.CLIP + ",origination_caller_id_name=" + thisDial.CLIP + "}sofia/external/" + thisExtension.Destination + " &park()"; eslEvent = thisFSDial.Connection.Api("originate", cDialString); After the call is succesfully answered I (depending on the script might play some audio to the dialed leg, and) after that I bridge using a uuid_bridge on an inbound uuid and this origination_uuid. (all this works perfectly, it's just the detection of the dtmf's on the b-leg!) I need to fix this by tommorow as I have a live customer who urgently requires this to work, so any help appreciated! Best regards, Mike.

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joining a bridged call to a conference
by Tihomir Culjaga 14 May '12

14 May '12

hello, i got an established call A calling B. A is a number outside FS (any number e.g. 16084191500), B is an extension registered to FS (e.g. 1002). after some time, i would like to join the call A => B to a conference roon e.g. 3001 (of course using ESL) any idea how to do it smooth ? whats the best practice for this ? i tried uuid_transfer but whatever i do it keeps killing one leg... thanks for help, T.

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Voice Activity Detection
by Andrew Cassidy 14 May '12

14 May '12

Hi Guys, What's the default for VAD in the sip profiles if none is specified? I get choppy audio without explicity setting it to none, is there anything I can check/try to improve the situation? Hardware wise it's a rackspace cloud server (but it does have a 1000Hz preemptible kernel, and otherwise seems fine) but I'll be testing again on a proper machine in the coming days and network wise it's over the internet (just in case latency causes issues with VAD). It's not a massive issue in reality, just that rackspace charge for traffic, so thought it might be worth trying to cut it down. -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F<info(a)cassidywebservices.co.uk> *03300 100 961 *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk

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Javascript Outbound Event Socket - Event Locking
by Sara Higfler 14 May '12

14 May '12

Hi, I'm continuing to make progress with the development of some dialplan applications using the Outbound Javascript Event Socket Library. The new challenge I've encountered is (I believe) related to event locking. I'm using the Outbound event socket in async mode. A simplified example of what I want to do is to play an announcement, wait for it to finish, then hangup the channel. Announcements are working fine in other parts of the application, but I'm having trouble forcing the announcement to play before the channel is disconnected. I believe this can be solved by activating event locking per: http://wiki.freeswitch.org/wiki/Event_Socket_Library#setEventLock However, I cannot find an example of this being implemented within javascript... I thought there would be a means of appending it to the execute command, but have been unable to determine how. req.execute('playback', 'conference/conf-enter_conf_number.wav'); Any help would be sincerely appreciated. I'm in the process of writing a guide describing basic examples of using Javascript with outbound event sockets. I'll add this to the list of hints for the newbies like myself once I've got to the bottom of it. Kind regards, Sara.

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FreeSWITCH Weekly News and Notes
by Michael Collins 14 May '12

14 May '12

Greetings from the FreeSWITCH team. We hope your Monday is going well! This week we have an interesting topic in our community conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_05_16?cc12-0514>. We will be joined by Fred Dixon, lead developer of the Big Blue Button<http://www.bigbluebutton.org/>project. Big Blue Button is open software that enables distance learning and remote education. Fred will do a live demonstration of how it works as well as discussing how FreeSWITCH fits into the picture. We invite everyone to join us Wednesday to see this cool project in action. In ClueCon news, we are pleased to announce that we have two more sponsors: NACT <http://www.nact.com?cc12-0514>and HarQen <http://www.harqen.com?cc12-0514>. NACT uses FreeSWITCH in carrier class solutions, such as pre-paid and their commercial Vinci Class 4/5 softswitch. HarQen provides innovative voice-based solutions in their line of Intelligent Voice Services products.Their flagship offering is Voice Advantage, a tool that allows creation of custom recorded interviews. Managers and recruiters can quickly review these interviews, greatly speeding up the recruiting process. We look forward to seeing representatives from NACT and HarQen at ClueCon this August. As a reminder, our ClueCon PowerPlay is still in effect. Register one attendee by the end of May and you'll receive three entries in our great ClueCon prize giveaway. Register two attendees from the same company and they *each *will receive six entries. The more people from your company who register, the more entries each person receives. Visit the ClueCon site<http://www.cluecon.com?cc12-0514>or call us at 877-742-2583 if you have any questions about attending, sponsoring, or speaking at this year's event. See you this August! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0514

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mod_xml_scgi compilation error
by Chris Mylonas 14 May '12

14 May '12

Hi FS Users, I was building FS on a box to test out the mod_gsmopen stuff that has been mentioned on the list recently. Updated CentOS 5 this morning, checked out a new git head. Getting this message that says "warnings being treated as errors" Not sure if that is what is killing the build or if there genuinely is an error. Any help would be great, Cheers Chris making all mod_xml_scgi Compiling ../../../../libs/libscgi/src/scgi.c... cc1: warnings being treated as errors ../../../../libs/libscgi/src/scgi.c: In function ‘scgi_build_message’: ../../../../libs/libscgi/src/scgi.c:94: warning: format ‘%ld’ expects type ‘long int’, but argument 4 has type ‘size_t’ ../../../../libs/libscgi/src/scgi.c:94: warning: format ‘%ld’ expects type ‘long int’, but argument 4 has type ‘size_t’ ../../../../libs/libscgi/src/scgi.c:103: warning: format ‘%ld’ expects type ‘long int’, but argument 4 has type ‘size_t’ ../../../../libs/libscgi/src/scgi.c:103: warning: format ‘%ld’ expects type ‘long int’, but argument 4 has type ‘size_t’ ../../../../libs/libscgi/src/scgi.c:110: warning: format ‘%ld’ expects type ‘long int’, but argument 4 has type ‘size_t’ ../../../../libs/libscgi/src/scgi.c:110: warning: format ‘%ld’ expects type ‘long int’, but argument 4 has type ‘size_t’ make[5]: *** [../../../../libs/libscgi/src/scgi.o] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_xml_scgi-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2

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Fw: AW: configuring gateway for making calls
by Samira Mh 14 May '12

14 May '12

dear erkan, i want to dial my mobile number (9191949637) via freeswitch myfreeswitch server is installed on vm ware with private ip-address(192.168.1.100) and the extension:1000 is register to freeswitch server now i want to call my phone number in order to my mobile phone will ringing; .such example in padf book :page 122 of freeswtch-1.0.6 ebook do you thins that needs worldwide call or internal call? for interal call need to set gateway ? ----- Forwarded Message ----- From: Erkan Ünlü (simalube) <erkan(a)speedingtrade.com> To: 'Samira Mh' <saami_mh(a)ymail.com> Sent: Monday, May 14, 2012 6:52 PM Subject: RE: AW: [Freeswitch-users] configuring gateway for making calls Hi again, yes of course that is right. The dialingplan that I give it to you is only for worldwide calls. You need a second dialing plan section in the XML Look here all sections can be put in the dialing plan xml. Internalcall means (see destination number is 3) that means all 3 long number can be dialed. You can change it to 4 or 5 <extension name="internalcall"> <condition field="destination_number" expression="^(\d{3})$"> <action application="bridge" data="USER/$1@$${domain}" /> </condition> </extension> Here a small sample for echo. Use GSM codec or G711a law codec to call. So you can hear you self. <extension name="echo"> <condition field="destination_number" expression="^11$"> <action application="answer" /> <action application="echo" /> </condition> </extension> These sample gives you a beeeep tone. <extension name="milliwatt"> <condition field="destination_number" expression="^12$"> <action application="answer" /> <action application="playback" data="tone_stream://%(10000,0,1004);loops=-1" /> </condition> </extension> From:Samira Mh [mailto:saami_mh@ymail.com] Sent: Monday, May 14, 2012 4:01 PM To: Erkan Ünlü (simalube) Cc: Free SWITCH Users Help Subject: Re: AW: [Freeswitch-users] configuring gateway for making calls hi, thanks so much for great help but while dialing my umber using extensino:1000 that register on myfreeswitch server the error is as follow: 2012-05-13 20:23:55.098050 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1000(a)192.168.1.100[5c057400-2be8-4080-a401-25b5dc402dd0] 2012-05-13 20:23:55.106289 [INFO] mod_dialplan_xml.c:418 Processing 1000->00989191949637 in context default 2012-05-13 20:23:55.111170 [NOTICE] switch_ivr.c:1447 Transfer sofia/internal/1000(a)192.168.1.100to enum[00989191949637@default] 2012-05-13 20:23:55.248259 [INFO] switch_core_state_machine.c:142 No Route, Aborting 2012-05-13 20:23:55.248259 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/internal/1000(a)192.168.1.100[CS_ROUTING] [NO_ROUTE_DESTINATION] 2012-05-13 20:23:55.257505 [NOTICE] switch_core_session.c:1182 Session 59 (sofia/internal/1000(a)192.168.1.100) Ended 2012-05-13 20:23:55.257505 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/1000(a)192.168.1.100[CS_DESTROY] 1: --------------------------------------------------------- cat /usr/local/freeswitch/conf/dialplan/default/01_shatel.xml <include> <context name="default"> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^00(98/d{10})$"> <action application="bridge" data="sofia/gateway/shatel/$1"/> </condition> </extension> </context> </include> 2:cat /usr/local/freeswitch/conf/directory/default/1000.xml -------------------------------------------------- <include> <user id="1000"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1000"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> ________________________________ From:Erkan Ünlü (simalube) <erkan(a)speedingtrade.com> To: 'Samira Mh' <saami_mh(a)ymail.com> Sent: Monday, May 14, 2012 3:09 PM Subject: AW: [Freeswitch-users] configuring gateway for making calls Hi Samira, you need 3 xml files. 1. Xml file is located at the directory/default Here put this xml sample and try after reloadxml to register your softphone <include> <user id="1000"> <params> <param name="password" value="yourpass" /> <param name="vm-password" value="1000" /> </params> <variables> <variable name="toll_allow" value="domestic,international,local" /> <variable name="accountcode" value="1000" /> <variable name="user_context" value="myWorldWideDialingPlan" /> <variable name="effective_caller_id_name" value="0000" /> <variable name="effective_caller_id_number" value="0000" /> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}" /> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}" /> <variable name="callgroup" value="techsupport" /> </variables> </user> </include> 2. Build your dialingplan The dialingplan is located in the dialingplan folder. Create here a new xml and put this sample inside the file. In the future you can study the default.xml in dialingplan folder to learn more about the dialingplan. In this section you see that the user_context in the directory folder XML file is the same with context_name So the user account is in association with the dialingplan. If you need you can build more different dialingplans for other requirements. <include> <context name="myWorldWideDialingPlan"> <extension name="Worldwide"> <condition field="destination_number" expression="^00 (\d+)$"> <action application="set" data="hangup_after_bridge=true" /> <action application="bridge" data="sofia/gateway/gateway/00$1|sofia/gateway/gateway2/00$1" /> </condition> </extension> </context> 3. The last step is the gateway Your gateway looks good and is reged correctly You see the gateway name you written “custom” You can change it to gateway1 or myfavoriteGW or what you want. But the name of the gatewayname must be the same like this line in section 2 of my description <action application="bridge" data="sofia/gateway/gateway/00$1|sofia/gateway/gateway2/00$1" /> Change the red gateway to the right gatewayname in your sip profiles external folder located XML You see in my sample "sofia/gateway/gateway/00$1|sofia/gateway/gateway2/00$1” I have 2 gateways for failover if one gateway is down or gives errors. | ßthese character is an OR and mean if the first gateway gives an error so try the second gateway You can use also only 1 gateway so delete the second gateway until to “ character Hope that is now clear for you. Kind regards Erkan Von:Samira Mh [mailto:saami_mh@ymail.com] Gesendet: Montag, 14. Mai 2012 12:26 An: FreeSWITCH Users Help Cc: erkan(a)speedingtrade.com Betreff: Re: [Freeswitch-users] configuring gateway for making calls hi erkan; thanks so much for your reply; i think your mean is: i must configure file 01_custom.xml like : <include> <extension name="Dial Out Custom Gateway"> <context name="samira"> <condition field="destination_number" expression="^9(1\d{10})$"> <action application="bridge" data="sofia/gateway/custom/$1"/> </condition> </context> </extension> </include> and because i registered via extension:1000 on freeswitch so : must configure file 1000.xml like : <include> <user id="1000"> <params> <param name="password" value="yourpass" /> <param name="vm-password" value="1000" /> </params> <variables> <variable name="toll_allow" value="domestic,international,local" /> <variable name="accountcode" value="1000" /> <variable name="user_context" value="linktoyourdialplan" /> <variable name="effective_caller_id_name" value="0000" /> <variable name="effective_caller_id_number" value="0000" /> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}" /> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}" /> <variable name="callgroup" value="techsupport" /> </variables> </user> </include> but i on't know what to be set as <variable name="user_context" value="linktoyourdialplan" /> in the 1000.xml? please help, i am new in freeswitch sorry for my english ;; ________________________________ From:Erkan Ünlü (simalube) <erkan(a)speedingtrade.com> To: 'FreeSWITCH Users Help' <freeswitch-users(a)lists.freeswitch.org> Sent: Monday, May 14, 2012 2:01 PM Subject: Re: [Freeswitch-users] configuring gateway for making calls Hi Samira, please build first of all them in the directory/default folder a XML file like this. <include> <user id="1000"> <params> <param name="password" value="yourpass" /> <param name="vm-password" value="1000" /> </params> <variables> <variable name="toll_allow" value="domestic,international,local" /> <variable name="accountcode" value="1000" /> <variable name="user_context" value="linktoyourdialplan" /> <variable name="effective_caller_id_name" value="0000" /> <variable name="effective_caller_id_number" value="0000" /> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}" /> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}" /> <variable name="callgroup" value="techsupport" /> </variables> </user> </include> Now you can register with your softphone to FS. See the user_context area in this section you must link it to your dialing plan The dialing plan is located in dialing plan folder. My simple dialing plan looks like this. <include> <context name="theNameOfDialingplan"> <extension name="Worldwide"> <condition field="destination_number" expression="^00 (\d+)$"> <action application="set" data="hangup_after_bridge=true" /> <action application="bridge" data="sofia/gateway/gateway/00$1|sofia/gateway/gateway2/00$1" /> </condition> </extension> </context> The context name of the dialing plan must be the same in the directory XML user_context I color it in red. I hope that this info are usefully for you. Kind regards. Von:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Samira Mh Gesendet: Montag, 14. Mai 2012 11:11 An: Free SWITCH Users Help Betreff: [Freeswitch-users] configuring gateway for making calls hi guys; i have configured gateway for making calls but it dosn't work; my config files are as follow: first create file that name " iptel.org.xml " in the below path: /usr/local/freeswitch/conf/sip_profiles/external/ i have an account on iptel.org like :(username:aimas;password:mypass) <include> <gateway name="custom"> <param name="username" value="arimas"/> <param name="password" value="mypass"/> <param name="realm" value="iptel.org"/>  <param name="proxy" value="sip.iptel.org"/> </gateway>--> </include> the run : sofia profile external restart reloadxml reloxm and sofia status: Name Type Data State ================================================================================================= external profile sip:mod_sofia@10.0.3.15:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG external::custom gateway sip:arimas@sip.iptel.org REGED internal profile sip:mod_sofia@10.0.3.15:5060 RUNNING (0) 10.0.3.15 alias internal ALIASED ================================================================================================= then create file name "01_custom.xml " in the below path : /usr/local/freeswitch/conf/dialplan/default the content of the files 01_custom.xml are: <include> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^9(1\d{10})$"> <action application="bridge" data="sofia/gateway/custom/$1"/> </condition> </extension> </include> . now when i dial 9, 1-800-555-1212 on eyebeam that register on freeswitch server the call is failed please help; thanks _________________________________________________________________________ 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

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