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Busy Lamps Aastra
by list subscriber 10 Sep '10

10 Sep '10

Hi All, Just wanted to pass on some feedback and thanks for improving the busy lamp situation on Aastra phone. Noticed the latest Freeswitch code has improvements which work better on Aastra 5xi devices. However there are some bugs. While the busy lamps work great on the receiving phone (b-leg), the busy lamps do not seem to be activated on the a-leg of calls. Also when a call is placed on hold it appears as "ringing" instead of indicating hold status more accurately. iNeil

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get lcr channel variable from esl
by Nigel Kent 10 Sep '10

10 Sep '10

I have used mod_lcr application in my esl script, digits are passed to mod_lcr and a dial string is found successfully as seen on the debug log. My question is how do I acces the variable ${lcr_auto_route} from esl. Previously in my script I have used this method to get variables which was set earlier in the script @con.api("uuid_getvar", "#{@uuid} myvarible").getBody However this is not working for lcr_auto_route. If I do(a)con.api("uuid_getvar", "#{@uuid} lcr_auto_route").getBody I get a _undef_ error

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macros in customized folder
by ovvenkat 10 Sep '10

10 Sep '10

Hi to all, 1. I have created a folder inside "* /usr/local/freeswitch/conf/lang/en/myfolder*" and placed my macro files. 2. After that, executed "reload" command in *fs_cli*. 3.When I am tying to access those macros from my lua Script, I am getting error like "Can't find macro <macro name>" 4. If I put, same files, inside the "* /usr/local/freeswitch/conf/lang/en/demo*" , Its working fine. Any one tell, where I need to configure so that, I can use macros which is inside the my customized folder. thanks, -- Regards Venkatesan OV.

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freetdm compile error
by Brian May 10 Sep '10

10 Sep '10

making install mod_freetdm libtool: link: gcc -I/home/brian/tree/freeswitch/libs/freetdm/src/include -I/home/brian/tree/freeswitch/libs/freetdm/src/isdn/include -I/home/brian/tree/freeswitch/libs/freetdm/src/include/private -I/home/brian/tree/freeswitch/libs/freetdm/src/ftmod/ftmod_sangoma_boost -DFTDM_CONFIG_DIR=\"/opt/freeswitch/conf\" -DFTDM_MOD_DIR=\"/opt/freeswitch/mod\" -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -O0 -g -ggdb -DPACKAGE_NAME=\"freetdm\" -DPACKAGE_TARNAME=\"freetdm\" -DPACKAGE_VERSION=\"pre-alpha\" "-DPACKAGE_STRING=\"freetdm pre-alpha\"" -DPACKAGE_BUGREPORT=\"bugs(a)freeswitch.org\" -DPACKAGE_URL=\"\" -DPACKAGE=\"libfreetdm\" -DVERSION=\"0.1\" -DSTDC_HEADERS=1 -DHAVE_SYS_TYPES_H=1 -DHAVE_SYS_STAT_H=1 -DHAVE_STDLIB_H=1 -DHAVE_STRING_H=1 -DHAVE_MEMORY_H=1 -DHAVE_STRINGS_H=1 -DHAVE_INTTYPES_H=1 -DHAVE_STDINT_H=1 -DHAVE_UNISTD_H=1 -DHAVE_DLFCN_H=1 -DLT_OBJDIR=\".libs/\" -DHAVE_LIBDL=1 -DHAVE_LIBPTHREAD=1 -DHAVE_LIBM=1 -DSIZEOF_LONG=4 -DHAVE_NETDB_H=1 -DHAVE_SYS_SELECT_H=1 -DHAVE_GETHOSTBYNAME_R=1 "-DDEBUG=/**/" -g -O2 -o .libs/testtones testtones-testtones.o ./.libs/libfreetdm.so -lm -lpthread -ldl -Wl,-rpath -Wl,/opt/freeswitch/lib ./.libs/libfreetdm.so: undefined reference to `ftdm_is_number' collect2: ld returned 1 exit status make[4]: *** [testtones] Error 1 make[3]: *** [local_install] Error 2 make[2]: *** [install] Error 1 make[1]: *** [mod_freetdm-install] Error 1 make: *** [install_mod] Error 2 -- Brian May <brian(a)microcomaustralia.com.au>

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Forcing SIP register time
by Kristian Kielhofner 10 Sep '10

10 Sep '10

Am I missing something or is there no way to force a SIP registration time in a Sofia profile? I know I *could* do it on the device but it would be nice to force random clients to re-register at a specific expiration time specified in the profile. Bounty? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

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how to originate a call on a sip hardphone programatic with ringback tone?
by rentmycoder rentmycoder 09 Sep '10

09 Sep '10

Hi, I'm trying to let a sip phone dial a number via esl. I use the originate api eg. ext 1001 calls ext 1000 from FS console: "originate {originate_timeout=30,origination_caller_id_number=1001,origination_caller_id_name=1001}user/1000 1001" This works, ext 1001 starts ringing, then if I pickup 1001, starts dialing 1000. My problem is, that there is no ringback tone when using this api. When I dial the number manually on the phone, there is a ringback tone indicating the other party to be ringing.... How to turn on the ringback tone using originate api and AFTER I pickup the phone and the other party is ringing??? In this case (correct me if I'm wrong) FS calls the originator, then transfers the b-leg to the destination ext. Any other method ow let the phone start the dialing??? Thanks...

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G729 failover
by Rob Morin 09 Sep '10

09 Sep '10

My experience with G729 has been that if the FS server runs out of licenses, the call fails with a fast busy. Is there a way to check the available licenses and fail to another codec, rather than, effectively, aborting the call? <maybe this should be a dial plan recipe> Thank you, Rob Morin

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quality of google voice
by Larry Matter 09 Sep '10

09 Sep '10

I was very excited to see that freeswitch can make calls through google voice (or is gmail?). I got it working without too much fuss thanks to the wiki. But the sound quality compared to my normal sip trunks is not very good. I'm wondering if I'm doing something wrong or do others have this "get what you pay for" experience with it. When I made a call the experience was a very significant delay in the audio (like a few seconds). I would hear a question, and answer it, and then hear the other person repeat the question as if I didn't hear it. There was also significant "crackle" as if there was a loose wire somewhere. I called back using my regular sip trunk and it was fine. I run freeswitch on an underutilized x86_64 box running Centos 5.5, zippy broadband connection. It is behind a NAT device, and my Cisco 7940 is behind a different NAT device. I'm a novice when it comes to the details of VOIP, but of course love to learn. I would love some suggestions on how to investigate this. Thanks! Larry Matter

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bridge
by babak yakhchali 09 Sep '10

09 Sep '10

Hi is it possible (in mod_managed or something similar) to create a new session and originate a call and if it was successful bridge it as the b leg of a call that moh is playing for it already(a waiting call)? thanx

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mod_opal installation problem
by velusamy Krishnan 09 Sep '10

09 Sep '10

Dear All, When I executed the "make mod_opal" in the source directory, it throws the following error, making all mod_opal Compiling /usr/src/GIT/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp... Compiling /usr/src/GIT/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp ... /usr/src/GIT/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:516:146: error: macro "switch_core_session_request" requires 4 arguments, but only 3 given /usr/src/GIT/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function ‘virtual OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*, unsigned int, OpalConnection::StringOptions*)’: /usr/src/GIT/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:515: error: ‘switch_core_session_request’ was not declared in this scope make[4]: *** [mod_opal.lo] Error 1 make[3]: *** [all] Error 1 make[2]: *** [mod_opal-all] Error 1 make[1]: *** [mod_opal] Error 2 make: *** [mod_opal] Error 2 Kindly anyone give some solution for this.. Regards, Velusamy.

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