- FreeSWITCH-users - FreeSWITCH Mailing Lists

ACL for ESL issue ?
by Pavel 14 Sep '15

14 Sep '15

Hello, I was trying to enable esl connections from outside of fs host. To do so I've followed https://wiki.freeswitch.org/wiki/Mod_event_socket#Configuration and changed default event_socket.conf.xml from: <configuration name="event_socket.conf" description="Socket Client"> <settings> <param name="nat-map" value="false"/> <param name="listen-ip" value="::"/> <param name="listen-port" value="8021"/> <param name="password" value="ClueCon"/> </settings></configuration> to: <configuration name="event_socket.conf" description="Socket Client"> <settings> <param name="nat-map" value="false"/> <param name="listen-ip" value="0.0.0.0"/> <param name="listen-port" value="8021"/> <param name="password" value="ClueCon"/> </settings></configuration> and issued: reload mod_event_socket. Trying to telnet to fs host on port 8021 I observe: Content-Type: text/rude-rejectionContent-Length: 24 Access Denied, go away.Content-Type: text/disconnect-noticeContent-Length: 67 Disconnected, goodbye.See you at ClueCon! http://www.cluecon.com/Connection closed by foreign host. And in fs log i can see the following: mod_event_socket.c:2603 IP "someiphere" Rejected by acl "loopback.auto" But as far as I understand the event_socket.conf.xml doesn't mention any ACL set up against ESL connection ?Would someone please be so kind to point what am I missing ?Thanks.Regards,Pavel.

4 6

Using G729 between apps and SIP trunk
by Michael Nielsen 14 Sep '15

14 Sep '15

I've got FreeSWITCH running and works really well. I would like to use G729 as codec whenever possible. My SIP Trunk does not support G729, only G711. My apps which are connecting to my FreeSWITCH are supporting G729, license has been paid for these. Can I setup FreeSWITCH to use G729 whenever possible, and for SIP Trunks conversion happens, such as: app/G729 <> FS <> app/G729 app/G729 <> FS <> SIP Trunk/G711

4 5

blind transfer behaviour
by huseyin kalyoncu 14 Sep '15

14 Sep '15

Hello, When using blind transfer feature with the following setup: transferee<--->kamailio_proxy<--->freeswitch | | transferor<------------- | If the transferee does not accept the transfer with a 488 reply to REFER, Freeswitch is still ending the call with a BYE message. Is this behaviour correct? Attached is a call flow diagram. Highlighted row is showing the BYE message after 488 reply to REFER. Thanks Huseyin

1 0

Recommendations for TLS certificate?
by Michael Nielsen 14 Sep '15

14 Sep '15

Anyone who can recommend a TLS certificate provider for FS?

4 5

Freeswitch does not upgrade user_agent
by Volodymyr Fedorov 14 Sep '15

14 Sep '15

Greets to all, seems after upgrade to 1.4.21 freeswitch doesn't update field "user_agent" in sqlitedb table "sip_registrations" after phone re-registers. Someone experience the same problems ? Thanks! -- Best regards, Volodymyr

1 0

gsmopen problems
by Pete Kay 12 Sep '15

12 Sep '15

Hi I am using gsmopen to make outbound calls. I am expeiencing some problems. When I have 2 dongles running even they are correctly recognized with "gsm list" , I can only make one calls. The second interface call will get Disconnection Out Of Order error. Also, when the remote end hangs up, the channel is still not getting killed. Therefore, no more call can be made even after remote side hangs up and I need to uuid_kill the channel. Is this an issue or my configuration is wrong? Pete

3 7

Clean installation of FreeSWITCH with basic functionality
by Michael Nielsen 11 Sep '15

11 Sep '15

Dear all, I'm playing around with FreeSWITCH and everything plays very nicely. I'm using the freeswitch-meta-vanilla Debian package on Debian 7. I'm "only" using FreeSWITCH for calls between subscribers, voicemail and outgoing calls to a SIP gateway - so the vanilla installation seems rather "overkill" - especially for a production environment. I've tried to - on a clean Debian - install the freeswitch-meta-default including freeswitch-conf-minimal. It does install some things, but it seems as if some basic functionality is not working - such as fs_cli etc. What is the best way for a very clean FreeSWITCH similar to the vanilla installation, but without any users, dial plans and such...?

6 12

Disable DTLS in Chrome Canary?
by Thomas 11 Sep '15

11 Sep '15

Hello list, Anthony mentioned that DTLS could still be disabled in Canary with a command line option, but I could not find anything. Does anyone know what that option is? Thanks, Thomas

2 2

Replacing leading "+" in outgoing numbers
by Jörn Ott 11 Sep '15

11 Sep '15

Hello all, I am using FusionPBX and freeswitch. Incoming calls are signalled with a leading + and also stored in the snom phones call lists. If someone tries to call back an external caller, I tried to strip off the leading + on the outgoing call with the following condition: <extension name="ext.ddddd" > <condition field="destination_number" expression="^\+(\d\d\d\d\d*)$" > <action application="set" data="sip_h_X-accountcode=${accountcode}" /> <action application="set" data="sip_h_X-Tag=" /> <action application="set" data="call_direction=outbound" /> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="inherit_codec=true" /> <action application="set" data="ignore_display_updates=true" /> <action application="set" data="callee_id_number=$1" /> <action application="set" data="continue_on_fail=true" /> <action application="set" data="outbound_prefix=+" /> <action application="bridge" data="sofia/gateway/781c8f7f-5d96-4e9e-8939-423b883cb675/$1" /> </condition> </extension> Sadly, the condition field already generates compile error. Can you give me a hint, how I can strip the + from the dialled number? Kind regards Joern -- -- Jörn Ott Systemadministrator Chefkoch GmbH Joseph-Schumpeter-Allee 33 53227 Bonn Telefon +49 228 28669-50 Telefax +49 228 28669-511 joern.ott(a)ext.chefkoch.de | http://www.chefkoch.de HRB 18761, Amtsgericht Bonn Geschäftsführer: Martin Meister, Sven Giebler, Arne Wolter Ein Unternehmen von Gruner+Jahr.

5 7

How to capture DTMF
by Dmitriy N. Borisov 11 Sep '15

11 Sep '15

Hi, All! I want to capture any DTMF from answered call to channel variable. How can I do that? -- С уважением, Борисов Дмитрий Отдел облачных технологий 1С-Рарус Тел. +7 (495) 231-20-02 доб.: 15-94 -- with best regards, Dmitriy Borisov 1C-Rarus Cloud Services tel.: +7 (495) 231-20-02 add.: 15-94

2 1