March 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Using mod_snmp for monitoring and stats
by Julian Pawlowski 04 Mar '13

04 Mar '13

Hi all, I was integrating mod_snmp for monitoring and stats purposes. However it seems snmpwalk does not show any of the mentioned details mentioned on the mod_snmp Wiki page, only the normal system stuff. I don't get any errors in the logs so I guess mod_snmp and agentx are just running fine all in all. Am I missing something? Cheers, Julian

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Unable to Recreate Log Files
by Subhash 04 Mar '13

04 Mar '13

Hi, I used the latest binaries(Windows Installer),version info is given below and observered that when it is not able to rename the log file it is keep on writing the logs in freeswitch.log file which is crossing the limt set in logconf.xml file. The error seen in the freeswitch log file is [CRIT] mod_logfile.c:164 Error renaming log from C:/Program Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program Files/FreeSWITCH/log/freeswitch.log.5 [No error] And the version info is Freswitch version 1.3.13b git 3c7f8f8 2013-02-20 22:44:51z *NOTE* : This issue is critical as we are facing it at production systems. Thanks, Subhash.

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Voicemail and Changing IPv4
by R W 04 Mar '13

04 Mar '13

All, I'm working with FreeSWITCH and I noticed that voicemail seems to be tied to an IP address. when I changed the IPv4 address on my FreeSWITCH 1.2-Stable. All of my greetings and voicemails are no longer associated with their respective extensions. I tried copying all the files from ./freeswitch/storage/voicemail/default folder/x.x.x.x/ to the new address in ./freeswitch/storage/voicemail/default folder/y.y.y.y/ and restarted FreeSWITCH, but none of the voicemails or greetings are associated with any extensions... What's the best to get voicemails and greetings re-associated with each user? Thank you! -Rob

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Freeswitch logging from script
by Kurtis Heimerl 03 Mar '13

03 Mar '13

Hello users, I'm running a few python scripts in my dialplan. I'm making calls to consoleLog (consoleLog('info', "Setting %s=%s\n" % (key, content[key]))) but these are not showing up in my actual FS log. These should be logged, I think, as in vars.xml we have set: <X-PRE-PROCESS cmd="set" data="console_loglevel=info"/> These printouts do appear in the FS console, just not in the stored FS log. So what am I missing? Thanks!

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Re: [Freeswitch-users] Creating a dialer with Event socket
by Sayyed Mohammad Emami Razavi 03 Mar '13

03 Mar '13

Your answer is not helpful. Please describe me more helpful and answer my questions. Assume i use ictdialer! where is my answer? Why does FS tell me that notice and why did not my code work correctly. BTW i can program with LUA. Answer my question for more clarity. Thanks.

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originate caller id number/name not reflected in CDR?
by Jun Sun 03 Mar '13

03 Mar '13

I'm using the following commands to generate outgoing calls and place them in the same conference room: originate {originate_timeout=60,origination_caller_id_number=$myNumber,origination_caller_id_name=\"$myName\"}sofia/internal/$theirNumber@$sipTrunk &conference(\"$myRoom\") While looking at the CDR (Master.csv), I notice the caller ID/name are not correct. See below. "Outbound Call","15102991912","15102991912","default","2013-03-03 01:12:06","2013-03-03 01:12:13","2013-03-03 01:12:25","19","12","NORMAL_CLEARING","5d35ca50-839f-11e2-b8cf-b53f4b8fa294","","","PCMU","PCMU" Caller ID name becomes "Outbound Call", and caller id number is the same as destination number, which is clearly wrong, because we can see the correct caller id number on the phone. Another interesting thing is that if the recipient does not answer, then the cdr enter would reflect the correct caller id name/number. I searched around but could not find a reasonable explanation. Would appreciate any pointers. Cheers. Jun

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Per user voicemail profile
by Muhammad Shahzad 02 Mar '13

02 Mar '13

Hi, I am using xml_curl to get voicemail_ivr module configuration from web server. However, i couldn't find any way to make it on per user basis. The HTTP request does not contain any useful information like caller-id, destination number or unique-id of call etc. All i could get from HTTP request is, hostname=localhost, section=configuration, tag_name=configuration, key_name=name, key_value=voicemail_ivr.conf I tried sending desired data using enable-post-var option, but does not work. Then tried appending the Caller-ID-Number and Unique-ID in binding url, but value is not populated. Please help. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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Creating a dialer with Event socket
by Sayyed Mohammad Emami Razavi 02 Mar '13

02 Mar '13

Dialer means an FS that calls to the number and plays some sounds for him/her and returns back him/her responses(interactive Dialplans). I try to make a dialer with PHP but FS results me: 2013-03-02 22:51:59.882940 [NOTICE] switch_channel.c:968 New Channel sofia/sipinterface_1/223 [7482c392-836e-11e2-8fcd-cdbbeee93600] 2013-03-02 22:52:00.382972 [NOTICE] sofia.c:6381 Hangup sofia/sipinterface_1/223 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2013-03-02 22:52:00.403240 [NOTICE] switch_core_session.c:1517 Session 35 (sofia/sipinterface_1/223) Ended 2013-03-02 22:52:00.403240 [NOTICE] switch_core_session.c:1521 Close Channel sofia/sipinterface_1/223 [CS_DESTROY] My php code is: $this->_fs = new ESLconnection('192.168.54.68', '8021', 'ClueCon'); $eslEvent = $this->_fs->sendRecv('bgapi originate {origination_caller_id_name=Richard-Albuquerque,origination_caller_id_number=1001,ignore_early_media=true}sofia/gateway/trunk_2/223 228') What is the problem? Can i create a dialer with PHP and Event socket?

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execute_on_answer record_session results in twice recording in different speed
by Dmitriy Shumaev 02 Mar '13

02 Mar '13

Hi, all. Next configuration: " <extension name="record_3..16_digits" continue="true"> <condition field="${is_already_recording}" expression="^$"/> <condition field="destination_number" expression="^(\d{3,16})$"> <action application="set" data="RECORD_TITLE=Recording ${caller_id_name} ${caller_id_number} to ${destination_number} at ${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(c) 2013"/> <action application="set" data="RECORD_SOFTWARE=FreeSwitch"/> <action application="set" data="RECORD_ARTIST=FreeSwitch"/> <action application="set" data="RECORD_COMMENT=FreeSwitch"/> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="export" data="ignore_early_media=true"/> <action application="export" data="execute_on_answer=record_session $${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${caller_id_name}_${caller_id_number}_2_${destination_number}.wav"/> <action application="export" data="is_already_recording=true"/> </condition> </extension> " results in some an incomprehensible result. C onversation b eing recorded twice in the same file. The first recording of the conversation played accelerated. The second recorded conversation played at normal speed. The replacement of <action application="export" data="execute_on_answer=record_session $${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${caller_id_name}_${caller_id_number}_2_${destination_number}.wav"/> to the following solved the problem : <action application="set" data="media_bug_answer_req=true"/> <action application="record_session" data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${caller_id_name}_${caller_id_number}_2_${destination_number}.wav"/> Both these examples are from [ http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session ]. So what is the difference between these options? Thanks for help. With best regards, Shumaev DA, KBR Ltd.

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displays the number of messages in aastra phones
by Archana Venugopan 02 Mar '13

02 Mar '13

Hi, If we have 6 voicemail messages, it will show in aastra phone display as '6 messages' and the indicator in phone will blink. If we hear all messages there will not be any displays for messages and the indicator in aastra phone will not blink. But I made changes to my voicemail code. After that even if I listen to all messages I still see the display as '6 messages'. After so many hours or after someone leaves another voicemail message only this is getting refreshed and shows actual message count. Can you please let me know in mod_voicemail.c source, which part of the code actually displays the number of messages in aastra phones? Regards, Archana

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