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3PCC USING FIFO / CALLCENTER
by afshin afzali 26 Aug '10

26 Aug '10

Hi FreeSWITCH, I am thinking to use mod_fifo / mod_callcenter in a 3pcc scenario which the first party of call will be selected dynamically. So I should be able to originate call into these modules. I don't know if it is possible right now or if could be requested as a new feature. BEST, -- afshin

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Module written in C++
by Farhan Husain 26 Aug '10

26 Aug '10

I am trying to write a module having multiple C++ source files. When I run make, I find that only the C++ file having the name same as the module name is being compiled and all others do not produce any object file. As a result when I run FS I get linker error. Can anyone tell me how to get them compiled too? Thanks, Farhan

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Re: [Freeswitch-users] [Freeswitch-dev] Module written in C++
by Farhan Husain 26 Aug '10

26 Aug '10

I just got the solution from Math at the IRC channel. Here is the link to the solution he gave me: http://pastebin.freeswitch.org/13727 On Thu, Aug 26, 2010 at 12:36 AM, Peter Olsson < peter.olsson(a)visionutveckling.se> wrote: > Have you updated the Makefile to build the other files? > > /Peter > ________________________________________ > Från: freeswitch-dev-bounces(a)lists.freeswitch.org [ > freeswitch-dev-bounces(a)lists.freeswitch.org] för Farhan Husain [ > farhan.husain(a)csebuet.org] > Skickat: den 26 augusti 2010 07:27 > Till: FreeSWITCH Users Help; freeswitch-dev(a)lists.freeswitch.org > Ämne: [Freeswitch-dev] Module written in C++ > > I am trying to write a module having multiple C++ source files. When I run > make, I find that only the C++ file having the name same as the module name > is being compiled and all others do not produce any object file. As a result > when I run FS I get linker error. Can anyone tell me how to get them > compiled too? > > Thanks, > Farhan > !DSPAM:4c75fc8232934761421429! > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org >

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php esl compilation PBs
by Jean-Yves F. Barbier 26 Aug '10

26 Aug '10

Debian sid ========== Hi list, I'm trying to figure out a way to compile ESL for PHP on my system. I was obliged to copy ../src/include/* into .../esl/php and modify them a bit, but I still have an error: g++ -I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -I/usr/include/php5/Zend -I/usr/include/php5/ext -I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o esl_wrap.cpp:2583: attention : deprecated conversion from string constant to ‘char*’ g++ esl_wrap.o -lcrypt -lcrypt -lonig -ldb -lqdbm -lrt -lm -ldl -lnsl -lcrypt -lcrypt -lpthread -o ESL.so -L. /usr/bin/ld: cannot find -lonig collect2: ld a retourné 1 code d'état d'exécution make: *** [ESL.so] Error 1 Apparently it is a missing library (onig), but which one? JY -- On a clear disk you can seek forever.

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eavesdrop question
by Seven Du 26 Aug '10

26 Aug '10

Hi, Playing with eavesdrop, it's very nice. And have two questions: 1) A talk to B. C eavesdrop the chanel. C can control whom he want to whisper to by DTMF. Is it possible to use an API/ESL to control ? Would like to make a patch if makes sense. uuid_send_dtmf can send to the remote party, would it make sense to implement a uuid_fake_dtmf to generate dtmf instead of from the remote party? 2) I tested another scenario with: A bridged to B. C eavesdrop the channel. And then D eavesdrop too. C can input DTMF 1/2/3, C worked as a normal eavesdrop. D has exactly the same behavior with C. (e.g. when C talk to A, D also can talk to A, when C talk to B, D can also talk to B). D cannot control anything with DTMF. Is it possible for the eavesdrop also respond to D's DTMF ? Ideally C and D has not effect to each other. Thanks. -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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dialplan variables in XML cdr
by Tim St. Pierre 25 Aug '10

25 Aug '10

Hello, Is there a way to have some custom dialplan variables appear in the XML cdr output? We want to log some things about the call, such as what option was chosen, which carrier route, etc. I want to be able to set a value in the dialplan and then figure out what it was in the CDR so we can do reports on it later. The variables I set aren't in the <variables> section. Is there any way to enable this, or am I missing something? Thanks! -Tim

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Remote-Party-ID manipulation in SIP 200 OK
by Neil Burgess 25 Aug '10

25 Aug '10

Hi, I am bridging a call through FS and have a problem with a Remote-Party-ID header being added to the returned 200 OK SIP message that is sent back on the A-Leg! I need to either re-write or strip it The scenario is that:- 1/ a user dials a number such as 07740777777 2/ we have to rewrite the number in the dial plan to 447740980777777 for routing to the upstream provider on the B leg 3/ the call is then bridged through to our upstream provider who attempts to connect the call successfully, returning a 200 OK. 4/ the a leg then has the Remote-Party-ID field inserted (looks to be done by FS) into the responding 200 OK, looking something like Remote-Party-ID: "447740777777" sip:07740777777@172.30.98.4 Unfortunately the Yealink phones we use then re-write the number from the original dialled to the number provided in the Remote-Party-ID (e.g. 447740980777777), which ends up in the phone's history. This now causes redials to fail! Any ideas on whether we can remove this header/ re-write or indeed control the number inserted into the display name component of the Remote Party ID? Many thanks, Neil

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Lua script help
by jesse 25 Aug '10

25 Aug '10

I have a bridge.lua script: phone1 = argv[1]; phone2 = argv[2]; dialstring1 = "sofia/gateway/xyz.com/" .. phone1; dialstring2 = "sofia/gateway/xyz.com7" .. phone2; session1 = freeswitch.Session(dialstring1); session2 = freeswitch.Session(dialstring2, session1); freeswitch.bridge(session1, session2); freeswitch(a)xyz.com> luarun bridge.lua 16502222222 16503333333 2010-08-24 16:47:55.331681 [NOTICE] switch_channel.c:779 New Channel sofia/external/16502222222 [043e3bb2-afda-11df-b543-e5f35c280c99] +OK freeswitch(a)xyz.com> 2010-08-24 16:47:55.425462 [INFO] sofia.c:662 sofia/external/16502222222 Update Callee ID to "Caller Jesse" <43583> 2010-08-24 16:47:55.427493 [NOTICE] sofia_glue.c:3294 Pre-Answer sofia/external/16502222222! 2010-08-24 16:47:55.428468 [NOTICE] switch_channel.c:779 New Channel sofia/external/16503333333 [044cee96-afda-11df-b544-e5f35c280c99] 2010-08-24 16:47:56.360476 [INFO] sofia.c:662 sofia/external/16503333333 Update Callee ID to "16503333333" <16503333333> 2010-08-24 16:47:56.361475 [NOTICE] sofia_glue.c:3294 Pre-Answer sofia/external/16503333333! 2010-08-24 16:47:56.363477 [ERR] switch_cpp.cpp:1220 Channels not ready 2010-08-24 16:47:56.363477 [NOTICE] switch_cpp.cpp:972 Hangup sofia/external/16503333333 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] 2010-08-24 16:47:56.363477 [NOTICE] switch_cpp.cpp:972 Hangup sofia/external/16502222222 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1220 Session 25 (sofia/external/16503333333) Ended 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1222 Close Channel sofia/external/16503333333 [CS_DESTROY] 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1220 Session 24 (sofia/external/16502222222) Ended 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1222 Close Channel sofia/external/16502222222 [CS_DESTROY] As you can see the call gets dropped immediately after couple rings. however, it works well if i do like this: originate sofia/gateway/xyz.com/16502222222 &bridge(sofia/gateway/xyz.com/16503333333) What is the reason Lua script will fail? any difference between the two approaches? thanks! jesse

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Outbound codecs preference
by Juan Antonio Ibañez Santorum 25 Aug '10

25 Aug '10

I can see when I try to stablish one B-leg call FS gives more precedence to remote codecs order. This way it is difficult to avoid transcoding in some situations where it could be possible. Is there any way to get a behaviour similar inbound-codec-negotiation=greedy for outbounds calls? What would be the best way to avoid transcoding? May be to use inbound-late-negotiation and rewrite codecs strings? Regards

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Suppress DTMF
by Samuel Macedo 25 Aug '10

25 Aug '10

Hi, I want to suppress the dtmf tone in the conference room. When a user sends a DTMF in the conference all others users can hear that DTMF tone. How can I suppress it? My <member-flags> = "waste". Thanks, -- Samuel Macedo

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