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Re: [Freeswitch-users] No ACK for 200 OK
by Xenia Obolenskaya 04 Jun '18

04 Jun '18

Hi, All! Thank All! I apologize for my negligence and inadvertence. Because a bare-bones gateway was setup for outbound calls with no username/pass and without registration I omitted an important parameter <param name="register-transport" value="tls"/> Now, with this parameter, the situation has returned to normal. A lot of thanks to All! Xenia Obolenskaya

2 1

How to get working leg timeout
by Sergey Safarov 04 Jun '18

04 Jun '18

Hello i want hangup call if caller not responded for 20 sec. When bleg send RINGING message then leg_timeout works as expected. But if bleg send SESSION_PROGGRESS message then leg_timeout help. Is exist other way to get this working?

6 8

Freeswitch records only 15 seconds of call.
by Ketan Kothari 04 Jun '18

04 Jun '18

Hi All, I'm getting issue on call recordings. Freeswitch can records maximum 15 seconds call but actual call was around 5 minutes. Any suggestion to overcome this issue ?

4 6

No ACK for 200 OK
by Xenia Obolenskaya 02 Jun '18

02 Jun '18

Hi, All! Freeswitch-1 sends 200 OK to Freeswitch-2 via IPv6. Freeswitch-2 received these packets and it can be seen in sip trace. However Freeswitch-2 does not send ACK. recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 13:20:35.903592: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS [2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKZ2359mFaFv98p;rport=55317 From: "Extension 1000" <sip:1000@ [2a02:f680:e0:1210::2123]>;tag=y912pHDgNp3QF To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=pNyNt1y311K1F Call-ID: 6b3983f9-e0f1-1236-0d9d-525400af4b7f CSeq: 123631351 INVITE Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 339 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2] X-FS-Support: update_display,send_info v=0 o=FreeSWITCH 1527902766 1527902767 IN IP6 2a02:7b43:1eb0:eb21::2 s=FreeSWITCH c=IN IP6 2a02:7b43:1eb0:eb21::2 t=0 0 m=audio 32068 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AEAD_AES_256_GCM_8 inline:LshmymiWylaGtFqTrEo3mLmx5sVN0Th32k7FSPcjicE4sDva+ZM4tlP2gck ------------------------------------------------------------------------ recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 13:20:37.903708: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS [2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKZ2359mFaFv98p;rport=55317 From: "Extension 1000" <sip:1000@ [2a02:f680:e0:1210::2123]>;tag=y912pHDgNp3QF To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=pNyNt1y311K1F Call-ID: 6b3983f9-e0f1-1236-0d9d-525400af4b7f CSeq: 123631351 INVITE Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 339 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2] X-FS-Support: update_display,send_info v=0 o=FreeSWITCH 1527902766 1527902767 IN IP6 2a02:7b43:1eb0:eb21::2 s=FreeSWITCH c=IN IP6 2a02:7b43:1eb0:eb21::2 t=0 0 m=audio 32068 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AEAD_AES_256_GCM_8 inline:LshmymiWylaGtFqTrEo3mLmx5sVN0Th32k7FSPcjicE4sDva+ZM4tlP2gck ------------------------------------------------------------------------ recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 13:20:41.903776: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS [2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKZ2359mFaFv98p;rport=55317 From: "Extension 1000" <sip:1000@ [2a02:f680:e0:1210::2123]>;tag=y912pHDgNp3QF To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=pNyNt1y311K1F Call-ID: 6b3983f9-e0f1-1236-0d9d-525400af4b7f CSeq: 123631351 INVITE Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 339 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2] X-FS-Support: update_display,send_info v=0 o=FreeSWITCH 1527902766 1527902767 IN IP6 2a02:7b43:1eb0:eb21::2 s=FreeSWITCH c=IN IP6 2a02:7b43:1eb0:eb21::2 t=0 0 m=audio 32068 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AEAD_AES_256_GCM_8 inline:LshmymiWylaGtFqTrEo3mLmx5sVN0Th32k7FSPcjicE4sDva+ZM4tlP2gck ------------------------------------------------------------------------ recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 13:20:45.904373: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS [2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKZ2359mFaFv98p;rport=55317 From: "Extension 1000" <sip:1000@ [2a02:f680:e0:1210::2123]>;tag=y912pHDgNp3QF To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=pNyNt1y311K1F Call-ID: 6b3983f9-e0f1-1236-0d9d-525400af4b7f CSeq: 123631351 INVITE Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 339 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2] X-FS-Support: update_display,send_info v=0 o=FreeSWITCH 1527902766 1527902767 IN IP6 2a02:7b43:1eb0:eb21::2 s=FreeSWITCH c=IN IP6 2a02:7b43:1eb0:eb21::2 t=0 0 m=audio 32068 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AEAD_AES_256_GCM_8 inline:LshmymiWylaGtFqTrEo3mLmx5sVN0Th32k7FSPcjicE4sDva+ZM4tlP2gck ------------------------------------------------------------------------ recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 13:20:49.904423: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS [2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKZ2359mFaFv98p;rport=55317 From: "Extension 1000" <sip:1000@ [2a02:f680:e0:1210::2123]>;tag=y912pHDgNp3QF To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=pNyNt1y311K1F Call-ID: 6b3983f9-e0f1-1236-0d9d-525400af4b7f CSeq: 123631351 INVITE Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 339 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2] X-FS-Support: update_display,send_info v=0 o=FreeSWITCH 1527902766 1527902767 IN IP6 2a02:7b43:1eb0:eb21::2 s=FreeSWITCH c=IN IP6 2a02:7b43:1eb0:eb21::2 t=0 0 m=audio 32068 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AEAD_AES_256_GCM_8 inline:LshmymiWylaGtFqTrEo3mLmx5sVN0Th32k7FSPcjicE4sDva+ZM4tlP2gck ------------------------------------------------------------------------ recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 13:20:53.905075: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS [2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKZ2359mFaFv98p;rport=55317 From: "Extension 1000" <sip:1000@ [2a02:f680:e0:1210::2123]>;tag=y912pHDgNp3QF To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=pNyNt1y311K1F Call-ID: 6b3983f9-e0f1-1236-0d9d-525400af4b7f CSeq: 123631351 INVITE Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 339 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2] X-FS-Support: update_display,send_info v=0 o=FreeSWITCH 1527902766 1527902767 IN IP6 2a02:7b43:1eb0:eb21::2 s=FreeSWITCH c=IN IP6 2a02:7b43:1eb0:eb21::2 t=0 0 m=audio 32068 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AEAD_AES_256_GCM_8 inline:LshmymiWylaGtFqTrEo3mLmx5sVN0Th32k7FSPcjicE4sDva+ZM4tlP2gck ------------------------------------------------------------------------ 2018-06-02 13:20:55.780247 [DEBUG] sofia_reg.c:2435 Changing expire time to 897 by request of proxy sip:ipvs.sipnet.ru:5061 recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 13:20:57.905810: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS [2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKZ2359mFaFv98p;rport=55317 From: "Extension 1000" <sip:1000@ [2a02:f680:e0:1210::2123]>;tag=y912pHDgNp3QF To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=pNyNt1y311K1F Call-ID: 6b3983f9-e0f1-1236-0d9d-525400af4b7f CSeq: 123631351 INVITE Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 339 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2] X-FS-Support: update_display,send_info v=0 o=FreeSWITCH 1527902766 1527902767 IN IP6 2a02:7b43:1eb0:eb21::2 s=FreeSWITCH c=IN IP6 2a02:7b43:1eb0:eb21::2 t=0 0 m=audio 32068 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AEAD_AES_256_GCM_8 inline:LshmymiWylaGtFqTrEo3mLmx5sVN0Th32k7FSPcjicE4sDva+ZM4tlP2gck ------------------------------------------------------------------------ recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 13:21:01.905406: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS [2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKZ2359mFaFv98p;rport=55317 From: "Extension 1000" <sip:1000@ [2a02:f680:e0:1210::2123]>;tag=y912pHDgNp3QF To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=pNyNt1y311K1F Call-ID: 6b3983f9-e0f1-1236-0d9d-525400af4b7f CSeq: 123631351 INVITE Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 339 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2] X-FS-Support: update_display,send_info v=0 o=FreeSWITCH 1527902766 1527902767 IN IP6 2a02:7b43:1eb0:eb21::2 s=FreeSWITCH c=IN IP6 2a02:7b43:1eb0:eb21::2 t=0 0 m=audio 32068 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AEAD_AES_256_GCM_8 inline:LshmymiWylaGtFqTrEo3mLmx5sVN0Th32k7FSPcjicE4sDva+ZM4tlP2gck ------------------------------------------------------------------------ recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 13:21:05.905437: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS [2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKZ2359mFaFv98p;rport=55317 From: "Extension 1000" <sip:1000@ [2a02:f680:e0:1210::2123]>;tag=y912pHDgNp3QF To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=pNyNt1y311K1F Call-ID: 6b3983f9-e0f1-1236-0d9d-525400af4b7f CSeq: 123631351 INVITE Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 339 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2] X-FS-Support: update_display,send_info v=0 o=FreeSWITCH 1527902766 1527902767 IN IP6 2a02:7b43:1eb0:eb21::2 s=FreeSWITCH c=IN IP6 2a02:7b43:1eb0:eb21::2 t=0 0 m=audio 32068 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AEAD_AES_256_GCM_8 inline:LshmymiWylaGtFqTrEo3mLmx5sVN0Th32k7FSPcjicE4sDva+ZM4tlP2gck So, Freeswitch-1 hangup: 2018-06-02 13:21:06.372001 [DEBUG] sofia.c:7084 Channel sofia/external-ipv6/1000@[2a02:f680:e0:1210::2123] entering state [terminating][0] 2018-06-02 13:21:06.372001 [NOTICE] sofia.c:8273 Hangup sofia/external-ipv6/1000@[2a02:f680:e0:1210::2123] [CS_EXECUTE] [NORMAL_UNSPECIFIED]

1 0

fax T.38
by Hossein Gholizadeh 02 Jun '18

02 Jun '18

Hi all Can any one tell me what is the differences between these two configs? #1 ------------------------------------------------------- *<extension name="t38_gateway">* * <condition field="destination_number" expression="^9108$">* * <action application="set" data="fax_enable_image_resizing=true" />* * <action application="set" data="fax_enable_t38=true"/>* * <action application="answer" />* * <action application="bridge" data="{sip_execute_on_image='t38_gateway self nocng',absolute_codec_string=PCMU}sofia/external/9107@${local_ip_v4}:5080"/>* * </condition>* * </extension>* ------------------------------------------------------- #2 ------------------------------------------------------- * <extension name="fax_detect">* * <condition field="destination_number" expression="^984432250161$">* * <action application="set" data="fax_enable_t38_request=true"/>* * <action application="set" data="fax_enable_t38=true"/>* * <action application="answer" />* * <action application="set" data="transfer_ringback=${us-ring}" />* * <action application="spandsp_start_fax_detect" data="transfer '9978 XML default' 10 ced"/>* * <action application="bridge" data="user/$1000@${domain_name}" />* * </condition>* * </extension>* ------------------------------------------------------- are they both detecting the fax and send it to other extension ?

1 0

Gateway
by Hossein Gholizadeh 01 Jun '18

01 Jun '18

hi ... i have my gateway on 192.168.30.32 and a phone which connected to gateway 984432250161. I configured them like this : 1.created a xml file in \conf\sip_profiles\external with name cheap_tel.xml and the content is this : <include> <gateway name="provider"> <param name="proxy" value="192.168.30.32"/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/>  <param name="username" value="not-used"/> <param name="password" value="not-used"/> </gateway> </include> 2.and in acl.conf added this node : <node type="allow" cidr="192.168.30.32/32"/> is my gateway config right ? because when i want to send a fax file by loopback with this command i have no error but fax is not saved in directory : Command ----> originate sofia/gateway/provider/984432250161 &txfax(C:/test.tiff) And also my dialplan configs for fax is this : <extension name="fax_detect"> <condition field="destination_number" expression="^984432250161$"> <action application="answer" /> <action application="set" data="transfer_ringback=${us-ring}" /> <action application="spandsp_start_fax_detect" data="transfer '9978 XML default' 10 ced"/> <action application="bridge" data="user/$1@${domain_name}" /> </condition> </extension> <extension name="fax_receive"> <condition field="destination_number" expression="^9978$"> <action application="answer" /> <action application="playback" data="silence_stream://2000"/> <action application="rxfax" data="$${base_dir}/faxes/${uuid}.tif"/> <action application="hangup"/> </condition> </extension> Can any one help me?

2 4

Bookmark in Play recording using bind_digit_action on call
by Uday kumar 01 Jun '18

01 Jun '18

Hello All, When any call is coming server, I am just playing a .wav file. Recording file length is 120min. Using *bind_digit_action*, user able to pause, seek, speed volume up and down. But no have time to listen 120min recording at a time. So If user will press a key, example 9 then save that playing location with caller phone number. If same caller calling from same number then start playing file from same where bookmark key was pressed. Please advise. -- Thanks & Regard Uday. Mobile:- +91-9377579349

1 1

Re: [Freeswitch-users] Achieving TLS + SRTP for inbound calls
by David P 01 Jun '18

01 Jun '18

Thanks, Mike, I'll try another distro. About verto, which I already serve under an AWS free non-exportable cert, does it need the gentls-generated CA root cert somewhere? And do I need any more steps to secure the streamed media? On Thu, 24 May 2018, 9:05 am Michael Jerris, <mike(a)jerris.com> wrote: > Try something NOT ubuntu. Ubuntu disables some required stuff in their > openssl making it unusable for webrtc at least in some versions. > > > On May 24, 2018, at 1:27 AM, David P <davidswalkabout(a)gmail.com> wrote: > > While waiting for suggestions, I tried more things. In particular, I > tested whether gentls_cert was present in our FS install (which is at > /opt/freeswitch/ on ubuntu). > > It is present, but the CA root cert step writes to > {prefix}/etc/freeswitch/tls/CA/ instead of {prefix}/conf/ssl/CA/. In > particular, the root CA step generates: > > etc/freeswitch/tls/ > cafile.pem > > etc/freeswitch/tls/CA/ > cacert.pem > cakey.pem > config.tpl > > And the server cert step generates: > > etc/freeswitch/tls/ > agent.pem > > etc/freeswitch/tls/CA/ > cacert.srl > > Reviewing agent.pem shows it's fine: > openssl x509 -noout -inform pem -text -in > /opt/freeswitch/etc/freeswitch/tls/agent.pem > > But it's owned by user root group root, so: > cd /opt/freeswitch/etc/freeswitch/tls/ > chmod 640 agent.pem CA/cacert.pem > chown root.freeswitch agent.pem CA/cacert.pem > > Then I edited /opt/freeswitch/etc/freeswitch/vars.xml to set > internal_ssl_enable and external_ssl_enable to true. > > Then I restarted FS. I checked the CLI and it shows "WS SETUP FAILED" > repeatedly. *Any suggestion?* > > Blazing ahead...then I exposed the public IP of the FS machine under a > subdomain of the CA root cert's domain. (I used a wildcard subdomain for > -org when generating both certs; maybe giving a wildcard this way is > unnecessary or counterproductive.) > > Now, the part that puzzles me: I'm already using a "real CA" cert in order > to serve my verto client files over https from my DNS name, so browsers > won't show security warnings. But I bet I need to have the same CA root > cert installed in FS as I use in the webserver, right? I saw the note that > commercial certs should work, but *it's not clear what steps to follow to > install one*. > > Cheers, > David > > > On Tue, May 22, 2018 at 9:50 PM, David P <davidswalkabout(a)gmail.com> > wrote: > >> We use conferences to allow a verto user to call and connect with an >> Asterisk channel. We would like to secure both signalling and media via TLS >> + SRTP, and I've read >> https://freeswitch.org/confluence/display/FREESWITCH/SIP+TLS a few times >> to understand how to do this. Note that that page has a broken link: >> https://wiki.freeswitch.org/wiki/Secure_RTP >> >> First, is it still true that FS doesn't offer prebuilt installs (for >> Ubuntu) to support this kind of security? >> >> Assuming that it must be compiled, I began to follow >> https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie >> with the additional first step of: apt-get install libssl-dev >> >> I soon ran into "Unable to locate package freeswitch-video-deps-most". >> >> What should I try next? >> >> Cheers, >> David >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org

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Question about blind transfers
by Alex Pierry 31 May '18

31 May '18

Hi all, I'm using the version 1.9.0 of Freeswitch with the following scenario: 1) One leg in park (the call center attendant in nailed up mode) channel uuid A 2) One incoming leg from customer. channel uuid B The call is bridged using api uuid_bridge and customer starts talking with the attendant. At this point everything is ok. But at some time, the attendant needs to transfer the customer to an external number located in another PBX. For this, I'm using the api uuid_broadcast to attendant's channel with att_xfer to dial an external number, as example bellow: uuid_broadcast A att_xfer::sofia/external/5000@192.168.0.24 When the external number answers the call, I set the attendant back to park so the customer can talk to the third-party leg (5000) - Blind transfer. What happens next is that B can hear 5000 but 5000 can't. The curious fact is that if B press any DTMF in his extension, 5000 starts to listen B normally. So, the question is: What am I doing wrong? Thanks in advance. Alex Pierry <http://pt.slideshare.net/marketingteclan> <http://blog.teclan.com.br/>

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Reload mod_commands
by Igor Olhovskiy 31 May '18

31 May '18

Hi! Suddenly API command ‘system’ stop working. Like fs_cli -x 'system /bin/ls' -ERR no reply Is there any way to restart mod_commands or other module to make it work without restarting whole freeswitch? Regards, Igor

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