February 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

fail to build git version of freeswitch
by Lubimov Alexey 01 Mar '12

01 Mar '12

https://bugzilla.redhat.com/show_bug.cgi?id=667529 Patch utlity do not allow to modify files in parent directories anymore. These patch solve this poblem. cat mod_shout.patch --- src/mod/formats/mod_shout/Makefile.orig 2012-02-27 22:47:13.721919787 +0400 +++ src/mod/formats/mod_shout/Makefile 2012-02-27 22:47:42.562919843 +0400 @@ -49,7 +49,7 @@ $(SHOUT_DIR): $(GETLIB) $(SHOUT).tar.gz - patch -p0 -i hack_out_ogg.diff + patch -p 6 -d ../../../../libs/libshout-2.2.2 < hack_out_ogg.diff $(SHOUT_BUILDDIR)/Makefile: $(SHOUT_DIR) mkdir -p $(SHOUT_BUILDDIR)

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Re: [Freeswitch-users] CallerID changing after Bridge using SLA on Polycom Phones
by Weigel, Stefan 29 Feb '12

29 Feb '12

Hi, that's working for me... [..] <action application="export" data="caller_id_number=..."/> <action application="export" data=" effective_caller_id_number=..."/> <action application="set_profile_var" data="caller_id_number=..."/> <action application="set_profile_var" data="effective_caller_id_number=..."/> <action application="set" data="ignore_display_updates=true"/> [..] then do the bridge. Best regards, Stefan Von: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Bob McCarthy Gesendet: Mittwoch, 29. Februar 2012 11:13 An: freeswitch-users(a)lists.freeswitch.org Cc: bill.oneil(a)experient.com Betreff: [Freeswitch-users] CallerID changing after Bridge using SLA on Polycom Phones I am having an issue similar to that experienced by Wellie Chao in his email "Caller ID on inbound calls on Polycom" 2010/4/22. The call is from an Audiocodes gateway with a dial string of *NXXXXXX# CallerID number sent is 101 which corresponds to the FXS port it came from, there is no CallerID Name. I modify the CallerID name and number in the dialplan before the bridge. <action application="export" data="effective_caller_id_name=some name"/> <action application="export" data="effective_caller_id_number=XXX-XXX-XXXX"/> <action application="export" data="ignore_display_updates=false"/> <action application="export" data="import=effective_caller_id_name"/> <action application="export" data="import=effective_caller_id_number"/> <action application="bridge" data="user/CO999x1001.3@${domain_name}"/> <action application="answer"/> On Ring the CallerID name and number are what I have changed it to. On Bridge, 80% of the time the CallerID name and number follow thru, 20% of the time it changes to what Leg A had originally before I used the export application.( name is blank and number is 101). if I put the call on Hold it reverts back to what I expected. I have tried to use set, moved the import after the bridge, ignore_display_updates=true. Any Ideas? On a further note, when phone 2 presses the shared line it shows the same CallerID name and number as phone 1. When you press BargeIn both the CallerID name and number change to 101. i.e. name is 101 number is 101 on both phones. this happens every time, is there a way to stop this behavior as well ? Thanks Bob McCarthy

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attended transfer cdr
by Miha Zoubek 29 Feb '12

29 Feb '12

Hi, in cdr.cvs.xml I have set only for B leg as I need only b. Scenaria: A calls B. Than B makes call FW do C. When this happens I do not get any CDR, so the call can not be charged. Please help me! Thanks Pastebin: http://pastebin.freeswitch.org/18528

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Test a dialplan from fs_cli
by piyush singhai 29 Feb '12

29 Feb '12

Hello, I want to execute a dialplan from fs_cli. Is it possible?? --Piyush

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401 Unauthorized in iptel external gateway
by Denis Gasparin 29 Feb '12

29 Feb '12

Hi to all. Today I installed freeswitch following the guide in Freeswitch book. I made successfully all the tests in the book... well all but one. When I tried to setup an external gateway on my iptel account, the registration process failed with 401 sip message. This is the sip trace: freeswitch@dhcppc7> send 647 bytes to udp/[217.9.36.145]:5060 at 17:59:20.965613: ------------------------------------------------------------------------ REGISTER sip:sip.iptel.org;transport=udp SIP/2.0 Via: SIP/2.0/UDP MY_PUBLIC_IP:5080;rport;branch=z9hG4bKXNc7jeQB7BXrp Max-Forwards: 70 From: <sip:MY_ACCOUNT@iptel.org;transport=udp>;tag=XQ58BvHmj15eD To: <sip:MY_ACCOUNT@iptel.org;transport=udp> Call-ID: 8c2823cd-573d-4cc2-8e01-cba340ae890f CSeq: 24760296 REGISTER Contact: <sip:gw+iptel@MY_PUBLIC_IP:5080;transport=udp;gw=iptel> Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 14-15-32 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 710 bytes from udp/[217.9.36.145]:5060 at 17:59:21.070238: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.9:5080:5080;rport=19198;branch=z9hG4bKXNc7jeQB7BXrp From: <sip: MY_ACCOUNT @iptel.org;transport=udp>;tag=XQ58BvHmj15eD To: <sip: MY_ACCOUNT @iptel.org;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-5904 Call-ID: 8c2823cd-573d-4cc2-8e01-cba340ae890f CSeq: 24760296 REGISTER Expires: 600 Min-Expires: 240 WWW-Authenticate: Digest realm="iptel.org", nonce="T0ki/U9JIsE+F7QdNo/qIktZaDvpS0sn" Server: ser (3.3.0-dev0 (i386/linux)) Content-Length: 0 Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=863 req_src_ip= MY_PUBLIC_IP req_src_port=19198 in_uri=sip:sip.iptel.org;transport=udp out_uri=sip:sip.iptel.org;transport=udp via_cnt==1" ------------------------------------------------------------------------ I'm sure that user account and password are valid. I also tested (successfully) a call from my extension to my account(a)iptel.org. Registration to iptel.org using a sip phone with the same user works just fine. What's wrong? Thank you in advance for your help, Denis

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mod perl prob - ExtUtils/Embed.pm
by spyros papadopoulos 29 Feb '12

29 Feb '12

Hi, I am runnign CentOS 6.2 and am attempting to compile mod_perl. However i am faced with this issue presented below. I have incoorporated the perl-libs & perl-ExtUtils packages including the following to list a few, but still no luck: Any help would be greatly appreciated. cheers, spyros perl-ExtUtils-CBuilder.x86_64 $ sudo make mod_perl making all mod_perl Can't locate ExtUtils/Embed.pm in @INC (@INC contains: /usr/local/lib64/perl5 /usr/local/share/perl5 /usr/lib64/perl5/vendor_perl /usr/share/perl5/vendor_perl /usr/lib64/perl5 /usr/share/perl5 .). BEGIN failed--compilation aborted. Can't locate ExtUtils/Embed.pm in @INC (@INC contains: /usr/local/lib64/perl5 /usr/local/share/perl5 /usr/lib64/perl5/vendor_perl /usr/share/perl5/vendor_perl /usr/lib64/perl5 /usr/share/perl5 .). BEGIN failed--compilation aborted. Compiling freeswitch_perl.cpp... g++ -w -DMULTIPLICITY -DEMBED_PERL -I/home/ec2-user/src/freeswitch/libs/curl/include -I/home/ec2-user/src/freeswitch/src/include -I/home/ec2-user/src/freeswitch/src/include -I/home/ec2-user/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DHAVE_CONFIG_H -c -o freeswitch_perl.o freeswitch_perl.cpp In file included from freeswitch_perl.cpp:2: freeswitch_perl.h:13:20: error: EXTERN.h: No such file or directory freeswitch_perl.h:14:18: error: perl.h: No such file or directory In file included from freeswitch_perl.cpp:2: freeswitch_perl.h:22: error: ISO C++ forbids declaration of âPerlInterpreterâ with no type freeswitch_perl.h:22: error: expected â;â before â*â token freeswitch_perl.h:23: error: ISO C++ forbids declaration of âPerlInterpreterâ with no type freeswitch_perl.h:23: error: expected â;â before â*â token freeswitch_perl.h:26: error: ISO C++ forbids declaration of âSVâ with no type freeswitch_perl.h:26: error: expected â;â before â*â token freeswitch_perl.h:39: error: âSVâ has not been declared freeswitch_perl.h:49: error: âPerlInterpreterâ has not been declared In file included from freeswitch_perl.cpp:3: mod_perl_extra.h:3: error: variable or field âmod_perl_conjure_eventâ declared void mod_perl_extra.h:3: error: âPerlInterpreterâ was not declared in this scope mod_perl_extra.h:3: error: âmy_perlâ was not declared in this scope mod_perl_extra.h:3: error: expected primary-expression before â*â token mod_perl_extra.h:3: error: âeventâ was not declared in this scope mod_perl_extra.h:3: error: expected primary-expression before âconstâ mod_perl_extra.h:4: error: variable or field âmod_perl_conjure_streamâ declared void mod_perl_extra.h:4: error: âPerlInterpreterâ was not declared in this scope mod_perl_extra.h:4: error: âmy_perlâ was not declared in this scope mod_perl_extra.h:4: error: expected primary-expression before â*â token mod_perl_extra.h:4: error: âstreamâ was not declared in this scope mod_perl_extra.h:4: error: expected primary-expression before âconstâ freeswitch_perl.cpp:5: error: âSTRLENâ does not name a type freeswitch_perl.cpp: In constructor âPERL::Session::Session()â: freeswitch_perl.cpp:13: error: âmy_perlâ was not declared in this scope freeswitch_perl.cpp: In constructor âPERL::Session::Session(char*, CoreSession*)â: freeswitch_perl.cpp:18: error: âmy_perlâ was not declared in this scope freeswitch_perl.cpp: In constructor âPERL::Session::Session(switch_core_session_t*)â: freeswitch_perl.cpp:34: error: âmy_perlâ was not declared in this scope freeswitch_perl.cpp: At global scope: freeswitch_perl.cpp:88: error: variable or field âsetPERLâ declared void freeswitch_perl.cpp:88: error: âPerlInterpreterâ was not declared in this scope freeswitch_perl.cpp:88: error: âpiâ was not declared in this scope freeswitch_perl.cpp:94: error: variable or field âsetMEâ declared void freeswitch_perl.cpp:94: error: âSVâ was not declared in this scope freeswitch_perl.cpp:94: error: âpâ was not declared in this scope make[4]: *** [freeswitch_perl.o] Error 1 make[3]: *** [all] Error 1 make[2]: *** [mod_perl-all] Error 1 make[1]: *** [mod_perl] Error 2 make: *** [mod_perl] Error 2

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Subject: ClueCon 2012 - Call For Speaking Proposals
by Michael Collins 29 Feb '12

29 Feb '12

Greetings! The ClueCon team is gearing up for this summer's event, which will be held August 7-9, 2012 at the Wyndham in downtown Chicago. We want to hear your proposals for presentations to be given at ClueCon '12. Here's what we're looking for: - Discussions on technology or software that relate directly or indirectly to open source telephony - Working title or very brief description of the talk - A two or three sentence abstract giving more details - Speaker's name, company, and biographical information - All presentations should be 30 minutes in length, including question and answer time - ClueCon sponsors who wish to have a presentation always get first priority when it comes to scheduling Please send all proposals to marketing(a)cluecon.com. If you spoke at ClueCon last year and your bio has not changed then please make note of this fact. We look forward to hearing from you! The ClueCon Team http://www.cluecon.com

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How to bridge a call to an extension defined in dialplan
by Liu, Xingkun 29 Feb '12

29 Feb '12

Hello, I'd like to bridge an incoming to two endpoints simultaneuously: one is a softphone which is registered to FS (to user 1003), another is just an extension which is defined in conf/dialplan/public.xml It looks like: <extension name="testRecording_bridge"> <condition field="destination_number" expression="^(74990)$"> <action application="bridge" data="sofia/internal/1003%${sip_profile},sofia/internal/74991%${sip_profile}"/> </condition> </extension> <extension name="testRecordingOnly"> <condition field="destination_number" expression="^74991$"> <action application="set" data="record_waste_resources=true"/> <action application="javascript" data="testRecordFile.js"/> </condition> </extension> The incoming call dial to extension 74990, from there I want to bridge the call to the sofphone at 1003 and to the extension 74991. The problem is that FS can not locate user 74991. So my question is what else I need to modify in order to bridge the call also to extension 74991 at the same time? I did lots of Googling, checked the wiki and tried various ways but still without success. Any advices please? Many Thanks! Xing -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012

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FreeSWITCH Community Conference Call Today
by Michael Collins 29 Feb '12

29 Feb '12

Hey all! Don't forget our call in about 15 minutes. Agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_02_29 Talk to you soon! -Michael

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Re: [Freeswitch-users] CallerID changing after Bridge using SLA on Polycom Phones
by Yehavi Bourvine 29 Feb '12

29 Feb '12

Hello, I have MP-124 from Audiocodes and had to add to the dialstring: {origination_privacy=screen} in order for the callerID to be displayed for calls originating from the MP. Regards, __Yehavi: 2012/2/29 Bob McCarthy <bob.mccarthy(a)experient.com> > I am having an issue similar to that experienced by Wellie Chao in his > email "Caller ID on inbound calls on Polycom" 2010/4/22.**** > > ** ** > > The call is from an Audiocodes gateway with a dial string of *NXXXXXX# > CallerID number sent is 101 which corresponds to the FXS port it came from, > there is no CallerID Name. I modify the CallerID name and number in the > dialplan before the bridge.**** > > **** > > <action application="export" data="effective_caller_id_name=some name"/>** > ** > > <action application="export" > data="effective_caller_id_number=XXX-XXX-XXXX"/>**** > > <action application="export" data="ignore_display_updates=false"/>**** > > <action application="export" data="import=effective_caller_id_name"/> **** > > <action application="export" data="import=effective_caller_id_number"/>*** > * > > <action application="bridge" data="user/CO999x1001.3@${domain_name}"/>**** > > <action application="answer"/>**** > > ** ** > > On Ring the CallerID name and number are what I have changed it to.**** > > On Bridge, 80% of the time the CallerID name and number follow thru, 20% > of the time it changes to what Leg A had originally before I used the > export application.( name is blank and number is 101). if I put the call on > Hold it reverts back to what I expected.**** > > ** ** > > I have tried to use set, moved the import after the bridge, > ignore_display_updates=true. Any Ideas?**** > > ** ** > > On a further note, when phone 2 presses the shared line it shows the same > CallerID name and number as phone 1. When you press BargeIn both the > CallerID name and number change to 101. i.e. name is 101 number is 101 on > both phones. this happens every time, is there a way to stop this behavior > as well ?**** > > ** ** > > Thanks**** > > Bob McCarthy**** > > ** ** > > ** ** > > **** > > _________________________________________________________________________ > 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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