- FreeSWITCH-users - FreeSWITCH Mailing Lists

About Verto
by 王聡 09 Jul '18

09 Jul '18

Hey all, I had tried to build a WebRTC video call service via mod_verto and Verto Comunicator. Now I could make calls from (to) an Android device with Chrome, but it didn’t work on iOS with Safari. Is there any point to be modified to run on Safari? Regards.

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collecting dtmf after bridge
by Francesco Facco de Lagarda 09 Jul '18

09 Jul '18

Please help.. I am trying to implement this scenario with js. * Extension dials operator * Operator answers * According to extension's requests, operator dials a digit * Request is executed. My problem is that once the two sessions have been bridged, I cannot collect any dtmf! This is what I am doing: Where AM I going wrong??? var opExt = "1000"; console_log(1,"\n\n\n\n\n\n\n\n\n\n\n\n\n\n\n\n============================= ==========================================="); var dtmf = new Object( ); dtmf.digits = ""; if (session.ready()) { var dtmf = { digits : '' }; var uuid = session.getVariable("uuid"); console_log(1, "Session UUID: " + uuid); console_log(1,"answering session .."); session.answer(); var uuid = session.getVariable("uuid"); console_log(1,"calling operator .."); var opSession = new Session("{UUIDLegA=" + uuid + "}user/" + opExt); if (opSession.ready()) { console_log(1,"Op session ready! ... bridging .."); bridge(session, opSession); console_log(1,"waiting for digit ..."); while (opSession.ready()) { opSession.collectInput( getDigit, dtmf, 0, 0); switch(dtmf.digits) { case "0" : // // etc } } } } console_log(1,"Hanging up sessions"); session.hangup(); opSession.hangup();

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Freeswitch initiate outbound call using SIPs + SRTP error (SRTP unprotect )
by Chhorm Chhatra 09 Jul '18

09 Jul '18

Hello, Currently, I faced a problem regarding SRTP outbound call to user (Leg B). The scenario is like this, - We set up our own root CA to an IP address (e.g 192.168.0.13) - We create a server certificate for freeswitch at 192.168.0.13 - Linphone is used as SIP client and is configured to trust our root CA by default. - Linphone A is configured to register to Freeswitch vis TLS + SRTP. (One leg call to server has both SIPs and SRTP – completely secure) - Linphone B is registered to Freeswitch via TLS + SRTP, and waiting for Linphone A to call to. (One leg call to server, e.g. 9196 (echo test), is completely secure with SRTP + SIPs) - Unfortunately, if A call to B, only A leg has SIPs + SRTP, but Leg B is not encrypted with SRTP and SIPs at all. This causes *SRTP unprotect failed with code 7 (auth check failed)**.* + Dialplan Configuration <action application="set" data="rtp_secure_media=true"/> <action application="export" data="rtp_secure_media=true"/> The dial-string is <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> + Directory Configruation: <param name="dial-string" value="{rtp_secure_media=${regex(${sofia_contact(${dialed_user}@ ${dialed_domain})}|transport=tls)},presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" /> My question is that, is there any configuration left that I have to set up in order to let freeswitch initiate an outbound call to Leg B correctly with SRTP and SIPs (tls)? Any help would be really appreciated. Thank you so much. Best Regard,

1 0

Re: [Freeswitch-users] Failure of Starting Freeswitch Service After A timeout Occurred
by Chhorm Chhatra 09 Jul '18

09 Jul '18

Just for those who encountered the same issue, this is because of the remote server that mod_xml_curl trying to talk is down during the startup of freeswitch. This error will automatically solved when the remote server is up again. On 21 March 2018 at 10:32, Chhorm Chhatra <ch.chhatra(a)gmail.com> wrote: > Dear FreeSwitch, > > Thank you for making freeswitch awesome! > > I have used freeswitch for awhile and there is no problem at all until > there is an issue happened regarding freeswitch daemon service. > > Normally, after enabling any module, I have to restart freeswitch service > by using *systemctl restart freewitch*. > > > > If the starting failed, then if I run *ps aux | grep freeswitch*, I could > see two users (root and freeswitch) are running the same freeswitch unit. > > > > > This issue makes the freeswitch service enters failed state. > > > > > If you do not mind, could you please assist me in fixing this issue? > > I am looking forward to hearing from you. > > Thank you. > > > > Sincerely yours, > > Chhatra Chhorm > >

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Setting Headers in mod_xml_curl and mod_curl
by Chhorm Chhatra 09 Jul '18

09 Jul '18

Thank you for making freeswitch awesome! Currently, i face an issue regarding setting headers in mod_xml_curl as per the security requirements to the remote server. I would like to know how to include custom headers in mod_xml_curl configuration. If you do not mind, please kindly let me know if there is any possibility to do so. Thank you so much. Best Regard, Chhatra Chhorm

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modify codecs in B-leg SDP when bypassing media
by Jeff Pyle 06 Jul '18

06 Jul '18

Hello, I've noticed the normal mechanisms for codec selection (codec_string, absolute_codec_string, etc) don't have any effect when bypass media mode is in use. This makes sense. Still, is there a way to scrub the codec list in the B-leg SDP before bridging? - Jeff

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LOSE_RACE, 3436 Originate Failed. Cause: NONE
by Paul Muaddib 05 Jul '18

05 Jul '18

Hi, sometimes I have the rare case that I lose a call. I can't reproduce it. It seems a bit random. When this happens all group members have a LOSE_RACE and the caller gets disconnected? Why? Thanks for helping Regards, Paul <extension name="Operator"> <condition field="destination_number" expression="^(operator)$|^(90)$" require-nested="false"> <action application="export" data="dialed_extension=operator"/> <action application="set" data="dialed_user=$1@${domain_name}"/> <action application="set" data="call_timeout=90"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bind_digit_action" data="get_digits,~^([1-9][0-9])$,exec:lua,get_digits.lua,peer,self"/> <action application="digit_action_set_realm" data="get_digits"/> <action application="ring_ready"/> <condition field="${office_status}" expression="^(open)$" break="on-true"> <action application="bridge" data="group/buero :_: pickup/global"/> <action application="hangup" data="NO_ANSWER"/> </condition> <condition field="${office_status}" expression="^(closed)$" break="on-true"> <action application="bridge" data="group/buero :_: group/werkstatt,pickup/global"/> <action application="hangup" data="NO_ANSWER"/> </condition> </condition> </extension>

1 0

ignore_early_media
by Sergey Safarov 05 Jul '18

05 Jul '18

Could anybody help me understand difference between "true" and "consume" vales of "ignore_early_media". Sure confluence description is not correct about "consume" value.

1 0

Inbound call one way audio, outbound call works
by Paul Muaddib 04 Jul '18

04 Jul '18

Hi, when I get inbound calls I can here the caller but the caller can't here the callee Other way around, making outbound calls is not a problem. Caller and callee can hear each other (My Setup) Phone -> FS -> NAT -> Gateway <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> What's the problem? Name Type Data State ================================================================================================= 10.0.200.2 alias internal ALIASED external profile sip:mod_sofia@10.0.200.2:5080 RUNNING (0) external::sip-trunk.telekom.de gateway sip:XXXXXXXXXXXX@sip-trunk.telekom.de REGED internal profile sip:mod_sofia@10.0.200.2:5060 RUNNING (0) ================================================================================================= sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT true DBName sofia_reg_internal Pres Hosts 10.0.200.2,10.0.200.2 Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.0.200.2 Ext-RTP-IP 87.157.X.X SIP-IP 10.0.200.2 Ext-SIP-IP 87.157.X.X URL sip:mod_sofia@10.0.200.2:5060 BIND-URL sip:mod_sofia@10.0.200.2:5060;transport=udp,tcp WS-BIND-URL sip:mod_sofia@10.0.200.2:5066;transport=ws WSS-BIND-URL sips:mod_sofia@10.0.200.2:7443;transport=wss HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMA,PCMU CODECS OUT PCMA,PCMU TEL-EVENT 101 DTMF-MODE info CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 6 FAILED-CALLS-IN 1 CALLS-OUT 6 FAILED-CALLS-OUT 4 REGISTRATIONS 7 sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT true DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 10.0.200.2 Ext-RTP-IP 87.157.X.X SIP-IP 10.0.200.2 Ext-SIP-IP 87.157.X.X URL sip:mod_sofia@10.0.200.2:5080 BIND-URL sip:mod_sofia@10.0.200.2:5080;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMA,PCMU CODECS OUT PCMA,PCMU TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 3 FAILED-CALLS-IN 1 CALLS-OUT 2 FAILED-CALLS-OUT 0 REGISTRATIONS 0 nat_map status Nat Type: NAT-PMP, ExtIP: 87.157.68.118 NAT port mapping enabled. I only map RTP ports. SIP is registered via TCP Console debug level 7 2018-07-04 23:12:24.905378 [DEBUG] sofia.c:7084 Channel sofia/external/+ 49XXXXXXXXXX(a)sip-trunk.telekom.de entering state [received][100] 2018-07-04 23:12:24.905378 [DEBUG] sofia.c:7091 Duplicate SDP v=0 o=- 0 2 IN IP4 217.0.15.67 s=on transit c=IN IP4 217.0.132.134 t=0 0 m=audio 19394 RTP/AVP 8 100 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=maxptime:20 a=ptime:20 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4365 Set telephone-event payload to 100@8000 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4708 Set telephone-event payload to 100@8000 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4767 sofia/external/+ 49XXXXXXXXXX(a)sip-trunk.telekom.de Set 2833 dtmf send payload to 100 recv payload to 100 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:6861 Audio params are unchanged for sofia/external/+49XXXXXXXXXX(a)sip-trunk.telekom.de. 2018-07-04 23:12:24.905378 [DEBUG] sofia.c:7999 Processing updated SDP 2018-07-04 23:12:24.925380 [DEBUG] sofia.c:7084 Channel sofia/external/+ 49XXXXXXXXXX(a)sip-trunk.telekom.de entering state [completed][200] 2018-07-04 23:12:24.945382 [DEBUG] sofia.c:7084 Channel sofia/external/+ 49XXXXXXXXXX(a)sip-trunk.telekom.de entering state [ready][200] 2018-07-04 23:12:31.605957 [DEBUG] sofia.c:7084 Channel sofia/external/+ 49XXXXXXXXXX(a)sip-trunk.telekom.de entering state [received][100] 2018-07-04 23:12:31.605957 [DEBUG] sofia.c:7094 Remote SDP: v=0 o=- 0 3 IN IP4 217.0.15.67 s=on transit c=IN IP4 217.0.132.134 t=0 0 m=audio 19394 RTP/AVP 8 100 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=maxptime:20 a=ptime:20 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4365 Set telephone-event payload to 100@8000 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4708 Set telephone-event payload to 100@8000 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4767 sofia/external/+ 49XXXXXXXXXX(a)sip-trunk.telekom.de Set 2833 dtmf send payload to 100 recv payload to 100 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:6861 Audio params are unchanged for sofia/external/+49XXXXXXXXXX(a)sip-trunk.telekom.de. 2018-07-04 23:12:31.605957 [DEBUG] sofia.c:7999 Processing updated SDP 2018-07-04 23:12:31.625959 [DEBUG] sofia.c:7084 Channel sofia/external/+ 49XXXXXXXXXX(a)sip-trunk.telekom.de entering state [completed][200] 2018-07-04 23:12:31.665962 [DEBUG] sofia.c:7084 Channel sofia/external/+ 49XXXXXXXXXX(a)sip-trunk.telekom.de entering state [ready][200] I am trying for days now but google is no real help. Regards, Paul

1 0

Manipulate the sip: string in the INVITE header on an outgoing Bridge
by Joseph Waite 04 Jul '18

04 Jul '18

Hi Guys Were having issues with Asterisk boxes registering to a trunk on our FreeSwitch box. We are sending the DID number called down the trunk in the to: field, however Asterisk doesn’t seem to like this. Is there anyway to manipulate the INVITE to be INVITE sip:DID@ <sip:DID@> instead of what is currently INVITE sip:user@ <sip:user@> Im hoping there is something like sip_invite_to_uri, however google is failing me. Regards

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