April 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Outbound ESL, typical load average
by royj＠yandex.ru 07 May '15

07 May '15

Hi all. Is there anyone here, who uses outbound ESL and have 200 and more total channels on FreeSWITCH instance. There is https://freeswitch.org/jira/browse/FS-7400 . Seems like FreeSWITCH is OK, at least developers did not found something abnormal with it. If somebody here, who uses similar configuration, what is your typical load average. Any comments are appreciated.

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Polycom VVX Shared Call Appearance
by Bob McCarthy 06 May '15

06 May '15

Trying to upgrade our Polycom Phones from SoundPoint 670's to VVX 600's. We have had Shared Call appearance working for the last couple of years on the 670's. We have it mostly working on the VVX except for the on hold pick-off. The pick-off enters the dial-plan which then fails. We never needed to dial plan this for the 670's. Using the same registrations / IP addresses i.e. one for one swap out. Anyone have VVX's working with SCA ? Bob

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Receive chat
by Sameer Khan 05 May '15

05 May '15

Hi, Is there any command which can receive chat from sip channel so I can put that received message in to a variable.

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407 Proxy authentication
by bhavik patel 05 May '15

05 May '15

Hi, We need 407 Proxy authentication require sip dialog while calling between sip to sip extension. Currently when we are calling, we never got that sip dialog. I am sure there must be some configuration to achieve this. Can anyone please help us to make it working? -- Thanks, Bhavik Patel

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Help with FSV record.
by dex_Cz 04 May '15

04 May '15

Hi guys! I have a question. Somebody tried to record in the FSV file when i bridge call? I hame a simple context from example. But it is not working. <include> <context name="centos"> <extension name="from-centos"> <condition field="destination_number" expression="^(onops)$"> <action application="record_fsv" data="/usr/local/freeswitch/recordings/testrecord.fsv"/> <action application="sleep" data="100"/> <action application="bridge" data="user/1004(a)172.16.0.2"/> <action application="hangup"/> </condition> </extension> </context> </include> I tried to change the context <include> <context name="centos"> <extension name="from-centos"> <condition field="destination_number" expression="^(onops)$"> <action application="export" data="execute_on_answer=uuid_record ${uuid} start /usr/local/freeswitch/recordings/testrecord.fsv"/> <action application="bridge" data="{ignore_early_media=true}user/1004(a)172.16.0.2"/> <action application="hangup"/> </condition> </extension> </context> </include> It also did not help. In the end I try to enter these commands from the console uuid_setvar <uuid> enable_file_write_buffering false (its says ok) and uuid_record <uuid> start /usr/local/freeswitch/recordings/testrecord.fsv But i have a error 2015-04-30 12:27:16.635463 [ERR] mod_fsv.c:702 You are asking to write 342 bytes of data which is not supported. Please set enable_file_write_buffering=false to use .fsv format 5e9ce2b0-ef23-11e4-b181-d1f037dabce9 2015-04-30 12:27:16.635463 [ERR] switch_ivr_async.c:1155 Error writing /usr/local/freeswitch/recordings/testrecord.fsv Anyone can help me, What I'm doing wrong? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Help-with-FSV-record-tp759615… Sent from the freeswitch-users mailing list archive at Nabble.com.

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voicemail + S3
by Alejandro 01 May '15

01 May '15

Hi all, I read that if I combine, mod_voicemail with mod_http_cache I can save my voicemail message directly into S3 storage. Any are using this implementation? can share with me some config file to understand how i need setup this? Thanks in advance Alejandro

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Custom Ringback
by ankitdoshi 30 Apr '15

30 Apr '15

Hello friends, I am newbie for ringback http://https://wiki.freeswitch.org/wiki/Variable_ringback <http://> i need something like whenever i will make call until call answer ring must be replace with my custom ring i have tried different solutions for that but didn't get exact solution as i need My current dialplan show me like <action application="bridge" data={ring_back=file_string:///usr/local/freeswitch/sounds/en/us/callie/promofiles/sound_file.wav,ignore_early_media=true}[leg_timeout=30]sofia/gateway/gateway_name/XXXXX"/> but its didn't work is it right way to make a ringback as dialplan shows ? Please let me know if i am doing anything wrong or need to change something to make it work Thanks in advance for suggestions and help -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Custom-Ringback-tp7596156.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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adding policies
by Juan Pablo L. 30 Apr '15

30 Apr '15

Hi, i m trying to enabled policies in my AAA, but i m getting the following error: Thu Apr 30 13:02:22 2015 : Error: ~/etc/raddb/policy.d/filter[66]: Failed to find "reject" as a module or policy. Thu Apr 30 13:02:22 2015 : Error: ~/etc/raddb/policy.d/filter[66]: Please verify that the configuration exists in ~/etc/raddb/mods-enabled/reject. Thu Apr 30 13:02:22 2015 : Error: ~/etc/raddb/policy.d/filter[66]: Failed to parse "reject" entry. Thu Apr 30 13:02:22 2015 : Error: ~/etc/raddb/policy.d/filter[65]: Failed to parse "if" subsection. Thu Apr 30 13:02:22 2015 : Error: ~/etc/raddb/sites-enabled/default[220]: Errors parsing authorize section. i have enabled policies in radius.conf: policy { $INCLUDE policy.d/ } i have defined the policy in the ~/etc/raddb/policy.d/filter filter_username { if (User-Name != "%{tolower:%{User-Name}}") { reject } } and i configured sites-available/default to call the policy in the authorize section by simply calling the name of the filter: filter_username but i get the above error, i would appreciate where to look for more info about this. thanks!!

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Remote Control for SIP Clients
by Paul Atreides 30 Apr '15

30 Apr '15

Hi, is it possible to remotely control a sip client. Like dialing numbers over an computer program or picking up the phone?

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FreeBSD installation error
by Brad Knight 30 Apr '15

30 Apr '15

Hello, I'm having trouble downloading the source to a FreeBSD box (pfSense2.2.2). /usr/local/src: git clone https://stash.freeswitch.org/scm/fs/freeswitch.git fatal: could not create work tree dir 'freeswitch'.: No such file or directory I was following this guide. https://freeswitch.org/confluence/display/FREESWITCH/FreeBSD Any help/direction would great to what I may be doing wrong. Thanks, Brad

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