March 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Error installing mod com g729
by Dmitry Saratsky 09 Mar '12

09 Mar '12

Hi all, I have an issue similar to this: http://lists.freeswitch.org/pipermail/freeswitch-users/2011-March/070981.htm l Currently I work successfully with: fsg729-167-installer After I run: fsg729-194-installer and then /usr/local/freeswitch/bin/validator - I got: 18:37 fs:/usr/src/g729# /usr/local/freeswitch/bin/validator G.729A licencing tool You will require one or more sales codes to activate G.729A Enter a sales code, or a blank line to end: <my activation code here> Enter a sales code, or a blank line to end: Sales codes to be activated: <my activation code here> OK (Y/N)? y ERROR [(null)] 18:37 fs:/usr/src/g729# And Null size of licence.zip: -rw-r--r-- 1 root root 0 Mar 9 18:37 licences.zip If I run again: fsg729-167-installer - All works fine. Any ideas?

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Destination number
by Miha Zoubek 09 Mar '12

09 Mar '12

Hi, I have user rigstered on FS as exp. 1232456.freeswitch.si. Problem is that I am also getting destination number in CDR as this, but I would like that destiantion number is just number so, in this calse 1232456. IS that possible? Thanks! Miha

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mod_callcenter setting caller name on a per call basis
by Mike 09 Mar '12

09 Mar '12

Hi, I'd like to use mod_callcenter to have the agents answer calls on behalf of multiple clients. The simplest way - and the way that I'm doing it at the moment (without mod_callcenter) is setting the caller_id_name in the dialplan based on the dialled number. So it's a bit of a fudge, (as it should be the called_id_name as it were), but at least it displays on the phone so that the agent can answer properly: 123 calling 555(ACME) displays: From: ACME 123 Agent answers "Hello ACME, how can I help you" 456 calling 666(Bedrock) displays: From: Bedrock 666 Agent answers "Hello Bedrock, how can I help you" With mod_callcenter the wiki says:  <agent name="1001@default" type="callback" contact="[origination_caller_id_name='Queue Caller']user/1001@default <snip> Which works fine with a fixed value, but trying to include a variable set earlier in the dialplan like: <agent name="1001@default" type="callback" contact="[origination_caller_id_name=${the_name_I_set}]user/1001@default <snip> Throws an error: 2012-03-09 10:59:25.810882 [CRIT] switch_channel.c:1183 Invalid data (${sip_from_display} contains a variable) 2012-03-09 10:59:25.810882 [CRIT] switch_channel.c:1183 Invalid data (${sip_full_from} contains a variable) I understand the security reasons for not allowing variables within variables - as explained by Brian here http://jira.freeswitch.org/browse/FS-3015 So - given all that - any suggestions on how to get this to work with mod_callcenter? Mike

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reserve mysql database using odbc in the core
by Yuriy Nasida 09 Mar '12

09 Mar '12

Hello guys! If anybody know how can I use reserve mysql database using odbc in the core? So if my first mysql server was down I would like that FS would use second mysql server for core. http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core Thanks.

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How to Create PHP Application for Freeswicth using ESL
by Shahzad Bhatti 09 Mar '12

09 Mar '12

Hi Everyone, i am a new user and want to learn how to create php application using ESL i just want to play a dynamic message using cepstral and get the customer response using dtmf. but i can't unserstand how to handle call using php as their is no mod_php is available like mod_lua, i park the call and now want to speak in the call but unable to do: please guide me i can't solve that problem with playandgetdigits() example of Mr. Michael Collins that what is the code of [handler.php]. ===================================================================================================================== * * *originate user/1500 meeting_request_1500* Then create that extension: <extension name="meeting request"> <condition field="destination_number" expression="^meeting_request_( \d+)$"> <action application="set" data="target_user=$1"/> <action application="set" data="text=cepstral|callie|Hi, we have a meeting on Monday at 9, are you joining us, if yes then press, 1. and if not press, 2."/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="play_and_get_digits" data="1 1 3 5000 # speak:${text} ivr/ivr-that_was_an_invalid_entry.wav answer \d"/> <action application="set" data="api_hangup_hook=system *handler.php* ${target_user} ${answer}"/> <action application="log" data="INFO User ${target_user} responded with ${answer}"/> <action application="playback" data="ivr/ivr-Thank_you.wav"/> <action application="hangup"/> </condition> </extension> =================================================================================================================== is anyone guide me how to do it php, thanks in advance, Shahzad Bhatti

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UUID variables
by Miha Zoubek 09 Mar '12

09 Mar '12

Hi, is it possible to get variables from different chanel (UUID)? When I make attended transfer different caller_id_name/number appears in my CDR that should. For exp: A calls B, B maks xfer to C. At the end A is talking to C. In my CDR I am getting. A to B. B to C. And B to C. The last should be A to C. Thank you! Miha

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FS_CLI and Log levels -sofia help cant stop unwanted log lines
by Sean Devoy 09 Mar '12

09 Mar '12

Hi, I must be missing some commands to filter down the messages displayed when I run fs_cli. I have tried (with some success): Console loglevel n (n = 1, 2, 3 helps very little) Sofia loglevel all n does not appear to have any affect. Sofia global siptrace on and "off" do what I expected!!!! I still get reams of log lines when any call comes in or goes out. Samples: 2012-03-08 14:34:28.132435 [DEBUG] switch_core_session.c:1180 Send signal sofia/external_noauth/14436046159(a)64.136.174.30 [BREAK] 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:362 (sofia/external_noauth/14436046159(a)64.136.174.30) Running State Change CS_REPORTING 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:662 (sofia/external_noauth/14436046159(a)64.136.174.30) State REPORTING 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:387 (sofia/external/240) State Change CS_REPORTING -> CS_DESTROY 2012-03-08 14:34:28.132435 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/240 [BREAK] 2012-03-08 14:34:28.132435 [DEBUG] switch_core_session.c:1380 Session 403 (sofia/external/240) Locked, Waiting on external entities 2012-03-08 14:34:28.132435 [NOTICE] switch_core_session.c:1398 Session 403 (sofia/external/240) Ended 2012-03-08 14:34:28.132435 [NOTICE] switch_core_session.c:1400 Close Channel sofia/external/240 [CS_DESTROY] 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:79 sofia/external_noauth/14436046159(a)64.136.174.30 Standard REPORTING, cause: NORMAL_CLEARING 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:662 (sofia/external_noauth/14436046159(a)64.136.174.30) State REPORTING going to sleep 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:491 (sofia/external/240) Callstate Change HANGUP -> DOWN 2012-03-08 14:34:28.132435 [DEBUG] switch_core_state_machine.c:387 (sofia/external_noauth/14436046159(a)64.136.174.30) State Change CS_REPORTING -> CS_DESTROY 2012-03-08 14:34:28.132435 [DEBUG] switch_core_session.c:1180 Send signal sofia/external_noauth/14436046159(a)64.136.174.30 [BREAK] What am I missing? Thanks in advance, Sean

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DIALPLAN: Bridge, Hang Up and Voice mail
by Sean Devoy 09 Mar '12

09 Mar '12

Hi, I have a couple of questions regarding XML dialplan functionality. First, using this simple dial plan: <extension name="220" > <condition field="destination_number" expression="^220$"> <action application="set" data="effective_caller_id_number=${internal_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${internal_caller_id_name}"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="call_timeout=20"/> <action application="bridge" data="sofia/external/220%mydomainname.com"/> <action application="answer"/> <action application="voicemail" data="default voicemail_BFIS 220"/> <action application="hangup"/> </condition> </extension> I can call 220, if there is no answer it goes to voicemail. GOOD. However, if that user answers and then hangs up, I still go to voice mail. I thought bridge terminated the call. Question 1: Why does the dial plan continue to voicemail after bridge answers, talks and hangs up? Question 2: If this user's phone is unplugged, the bridge kicks an error and it DOES NOT go to voice mail. How can I detect "extension unavailable" in the dial plan and perhaps forward to a cell phone? Thanks in advance, Sean

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FreeSWITCH Conf Call Tomorrow - Dialplan Basics
by Michael Collins 09 Mar '12

09 Mar '12

Hello all! Here's tomorrow's agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2012_03_06 I will be discussing some dialplan basics along with some regex tricks and basic patterns. Other experts may also be chiming in. The idea is that everyone on the call will come away with a more thorough understanding of what happens when a phone call comes into FreeSWITCH. Talk to you tomorrow! -Michael

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How to get session obj from uuid
by Srini K 09 Mar '12

09 Mar '12

Hi,How to get the Session object if I know the uuid using mod_managed. ThanksSrini

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