- FreeSWITCH-users - FreeSWITCH Mailing Lists

T.38_gateway & switch_core_media_bug : increase "timeout"
by Weera Suriya 01 Aug '10

01 Aug '10

Hello list, I able to send & receive faxes via SS7 link. but I have a small problem with "switch_core_media_bug". I noticed "Media bug" activate for first 10 seconds to 20 seconds against SIP channels. most of our customers use Tele/FAX option on Fax machines. that means ; they answer the call manually (as a voice call) If there was a fax then they activate Fax mode. the problem is FS can't detect CNG tones after 20 seconds, because "switch_core_media_bug" was removed from the particular channels. If the destination fax machine can generate CNG tones within 10 ~ 20 seconds Then FS can detect CNG tones and change in to T.38 mode. is there any way to increase the window of time (switch_core_media_bug) to listen for the CNG tones on SIP channels? This is my dialplan for G711(from PSTN/SS7 via SIP) to T.38(to SIP ATAs) ------>8-------- <extension name="g711_to_t.38"> <condition field="destination_number" expression="^1(0\d{8})$"> <action application="pre_answer"/> <action application="set" data="ringback=%(1000, 4000, 400.0, 400.0)"/> <action application="t38_gateway" data="peer"/> <action application="bridge" data="sofia/external/$1(a)192.168.11.53"/> <action application="hangup"/> </condition> </extension> This is my dialplan for T.38(from SIP ATAs) to G711(to PSTN/SS7 via SIP) ------>8-------- <extension name="t.38_to_g711"> <condition field="destination_number" expression="^(0\d+)$"> <action application="ring_ready" /> <action application="set" data="ringback=%(1000, 4000, 400.0, 400.0)"/> <action application="start_dtmf_generate" data="true"/> <action application="t38_gateway" data="self"/> <action application="bridge" data="sofia/external/$1@192.168.11.54:5070"/> <action application="hangup"/> </condition> </extension> ------>8-------- Thank you. Weera Suriya

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Asterisk equivalent Bridge() function implementation
by mazilo 01 Aug '10

01 Aug '10

Asterisk v1.6.x uses its new Bridge() function to bridge an existing outbound call with a new incoming call. Using AGI with this http://www.pmarks.net/posted_links/google-voice-dialout.agi Google Voice DialOut Python scripts for Asterisk to simulate a WEB click2call, the Asterisk Bridge() function is a perfect tool to let an Asterisk PBX system to transprently bridge its outbound call with an incoming Google Voice call. I sure would like to use the Python scripts on my FS too, but plainly don't have the knowledge on how to do this. Has anyone done this before and can offer some help to achieve this? Thank you. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Asterisk-equivalent-Bridge-fu… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] nibble_bill and last git
by Madovsky 01 Aug '10

01 Aug '10

lcr difference between SVN and GIT version : the var lcr_rate_1 becomes lcr_rate_fieldl F ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Saturday, July 31, 2010 11:25 PM Subject: Re: nibble_bill and last git my fault I forgot a var, also I need to change my lcr custom.sql myabe ti would be nice to make a coercition with ldc + nibble bill example and nibble bill table Thanks Franck ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Saturday, July 31, 2010 6:49 PM Subject: nibble_bill and last git Nibble bill stopped to debit user accounts since I udpated last git. I didn't change anything in my settings, the console says well nibbleBill Beginning new billing bllbabla, but the cash field iss not debited, even with hearbeat set. Any ? Thanks Franck

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To freeswitch developers
by Madovsky 01 Aug '10

01 Aug '10

Excellent work guys, a lot of optimization done since last SVN version. I tried to reconfigure with my svn xml config from a fresh last git install without problem. for those who scared to reach the step of svn to git I even tried to install the last git version in the same directory of svn version without problem too. But I don't use also a really complex config or script. Happy Clue F

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Re: [Freeswitch-users] nibble_bill and last git
by Madovsky 01 Aug '10

01 Aug '10

my fault I forgot a var, also I need to change my lcr custom.sql myabe ti would be nice to make a coercition with ldc + nibble bill example and nibble bill table Thanks Franck ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Saturday, July 31, 2010 6:49 PM Subject: nibble_bill and last git Nibble bill stopped to debit user accounts since I udpated last git. I didn't change anything in my settings, the console says well nibbleBill Beginning new billing bllbabla, but the cash field iss not debited, even with hearbeat set. Any ? Thanks Franck

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How to multiple delete extensions in Freeswitch?
by Frank Tarczynski 01 Aug '10

01 Aug '10

I'm playing with freeswitch towards replacing asterisk/trixbox. I've managed to create 1000 new extensions by accident. I want to delete all these extensions, but there are so many that it's taking FusionPBX quite a long time to do anything. Is there any way to do a mass deletion of extensions in freeswitch or fusionpbx or using mysql? Since I just starting I don't care if I delete everything. Frank

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Questions on Book
by Nyamul Hassan 01 Aug '10

01 Aug '10

Hi, On page 100, the "terminators" bullet says if "less than <min>". Shouldn't that be <max>? Regards HASSAN

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nibble_bill and last git
by Madovsky 31 Jul '10

31 Jul '10

Nibble bill stopped to debit user accounts since I udpated last git. I didn't change anything in my settings, the console says well nibbleBill Beginning new billing bllbabla, but the cash field iss not debited, even with hearbeat set. Any ? Thanks Franck

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iPad with APNS
by Ivan C Myrvold 31 Jul '10

31 Jul '10

I am using iSip on my iPad, and it registers well with FreeSWITCH. I can make calls, and it receives calls when running. http://pastebin.freeswitch.org/13563 When I have APNS switched on, it should be able to receive calls also when it is offline, because an external server will be registered as the iPad. http://pastebin.freeswitch.org/13564 But unfortunately I do not receive any call on my iPad, and freeswitch eventually times out. Does anyone have any hint to what could be wrong? Here is a SIP trace of the call, from local phone 160(a)imyrvold.dyndns.org: http://pastebin.freeswitch.org/13565 Ivan

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Compile Fail for mod_opal
by Nyamul Hassan 31 Jul '10

31 Jul '10

Hi, I ran the "./build/buildopal.sh" successfully, but still get the following error on a regular "make all". Can someone please help? Regards HASSAN making all mod_opal Compiling /usr/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp... Compiling /usr/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp ... /usr/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In constructor 'FSConnection::FSConnection(OpalCall&, FSEndPoint&, void*, unsigned int, OpalConnection::StringOptions*, switch_caller_profile_t*, switch_core_session_t*, switch_channel_t*)': /usr/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:564: error: no matching function for call to 'OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, void*&, unsigned int&, OpalConnection::StringOptions*&)' /usr/include/opal/opal/localep.h:253: note: candidates are: OpalLocalConnection::OpalLocalConnection(OpalCall&, OpalLocalEndPoint&, void*) /usr/include/opal/opal/localep.h:242: note: OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&) /usr/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)': /usr/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1296: error: 'class OpalMediaPatch' has no member named 'OnStartMediaPatch' /usr/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)': /usr/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1418: error: 'class OpalMediaPatch' has no member named 'OnStartMediaPatch' make[5]: *** [mod_opal.lo] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_opal-all] Error 1 make[2]: *** [all-recursive] Error 1

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