- FreeSWITCH-users - FreeSWITCH Mailing Lists

Send display update after RTP session is establihed ?
by Melek Oktay 11 Jul '18

11 Jul '18

Hi, I realize very interesting problem of my software stack (FreeSwitch + Kamailio) with Yealink 41P Voip phone.. After 3-4 days research, I realize Yealink phone does not try to start RTP session if it gets UPDATE packet before ACK packet !! In Freeswitch side, do we have a option that send UPDATE packet after RTP session is established? (we have a option in FreeSwitch side: do not send UPDATE packet however this is not good case for me, I update CallerID with this UPDATE packet)

2 1

Incoming RTP with different sequence number and host
by Sharath Kumar 11 Jul '18

11 Jul '18

Hello, I am running FS 1.6.18 and have a provider that is sending me a stream of incoming RTP that changes it's host IP, and RTP sequence number and Timestamps after 10 mins in the call. The SSRC does not change. As a result I am getting either silence or garbled audio. The Sample rate is 20ms as in the original stream. Is there any setting in the FS that can make this work ? Have any of seen this work with inbound stream ? Thank you, Shaks

3 3

mod_verto
by Vishal Pai 11 Jul '18

11 Jul '18

Hi All Can we record the video conferencing using mod_verto. If yes how it is possible and and what would it’s file format. Thanks Vishal Pai

2 6

RECORD_STEREO should not make both channels the same
by David P 11 Jul '18

11 Jul '18

We'd like to record the two legs of our calls separately (both 16kHz). I followed https://freeswitch.org/confluence/display/FREESWITCH/RECORD_STEREO and added <action application="set" data="RECORD_STEREO=true"/> to our dialplan. I now get two channels instead of one in our mp4's but the data in the two channels is identical, containing speech from both legs of the call. This is contrary to what the confluence page says to expect. Btw, I wanted to include our FS version here, and I tried to get it by running the cli and using /help, but it's not shown there.

2 3

Any feature access code of disconnect B-leg
by Ketan Kothari 11 Jul '18

11 Jul '18

Hello All, I have configured one IVR using that IVR customer can i dial destnation number. The flow is like that. Customer Dial DID number ---> Freeswitch --> Its will ask to dial destination number using DTMF --> [USA]Route to Destination A-leg = Customer ----> Freeswitch B leg = Freeswitch ---> Destination[USA] Again generate B- leg B leg = Freeswitch ---> Destination[INDIA] In this case customer are talking with [USA]destination number and now urgently he want to dial one more destination number[INDIA] and want disconnect[USA] So is there any access code or suggestion using that i can achieve above call flow ?

2 1

Re: [Freeswitch-users] mode_distributor in originate command
by Vishal Kakkar 11 Jul '18

11 Jul '18

1st Retry!! On Sat, Jul 7, 2018 at 12:50 AM, Vishal Kakkar <vishal.kakkar(a)gmail.com> wrote: > Hi all, > > How can we use mode_distributor in originate command. > > originate {ignore_early_media=true}sofia/gateway/${distributor(test)}/9822312312 > myExtension xml ext > > Its giving following error on CLI: > > [ERR] mod_sofia.c:4408 Invalid Gateway '%{distributor(test)}' > > > Distributor config: > > <configuration name="distributor.conf" description="Distributor Configuration"> > <lists> > <list name="test" total-weight="10"> > <node name="foo1" weight="1"/> > <node name="foo2" weight="9"/> > </list> > </lists> > </configuration> > > > Thanks > > Vishal. > >

1 0

481 Call does not Exit
by Paul Muaddib 11 Jul '18

11 Jul '18

Hi, if I call someone, who accepts the call and then hangs up, the call does not end on my side Error message: 481 Call does not Exit What is the reason for this? Best regards, Paul

2 2

Re: [Freeswitch-users] 481 Call does not Exit
by kaleem rehman 10 Jul '18

10 Jul '18

Hi Paul, its likely to be firewall blocking ports. looking at the log entry { *m=audio 19394 RTP/AVP 8 100* } other party is using RTP port : 19394 , looking at port, its likely they are using RTP range of 16384 to 32767, is this port allowed on your side? as this is the standard RTP range. regards, Kaleem ---------- Forwarded message ---------- From: Paul Muaddib <paul.muaddib83(a)gmail.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Cc: Bcc: Date: Wed, 4 Jul 2018 18:48:50 +0200 Subject: Re: [Freeswitch-users] 481 Call does not Exit Solved it. 2018-07-03 22:27 GMT+02:00 Paul Muaddib <paul.muaddib83(a)gmail.com>: > Hi, > > if I call someone, who accepts the call and then hangs up, the call does > not end on my side > > Error message: 481 Call does not Exit > > What is the reason for this? > > Best regards, > Paul > ---------- Forwarded message ---------- From: Paul Muaddib <paul.muaddib83(a)gmail.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Cc: Bcc: Date: Thu, 5 Jul 2018 00:10:01 +0200 Subject: [Freeswitch-users] Inbound call one way audio, outbound call works 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 ---------- Forwarded message ---------- From: Paul Muaddib <paul.muaddib83(a)gmail.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Cc: Bcc: Date: Thu, 5 Jul 2018 14:53:46 +0200 Subject: [Freeswitch-users] LOSE_RACE, 3436 Originate Failed. Cause: NONE 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> _______________________________________________ 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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Re: [Freeswitch-users] bridge call back locks fs_cli in js .. the log..
by Francesco Facco de Lagarda 10 Jul '18

10 Jul '18

2018-07-10 10:47:42.628911 [INFO] switch_channel.c:515 RECV DTMF 8:896 2018-07-10 10:47:42.628911 [WARNING] goconf.js:2 Given userData.param of under the rock 2018-07-10 10:47:42.628911 [DEBUG] switch_rtp.c:5261 Send start packet for [8] ts=262880 dur=160/160/896 seq=19828 lw=262880 2018-07-10 10:47:42.648920 [DEBUG] switch_rtp.c:5160 Send middle packet for [8] ts=262880 dur=320/320/896 seq=19829 lw=263040 2018-07-10 10:47:42.668912 [DEBUG] switch_rtp.c:5160 Send middle packet for [8] ts=262880 dur=480/480/896 seq=19830 lw=263200 2018-07-10 10:47:42.688913 [DEBUG] switch_rtp.c:5160 Send middle packet for [8] ts=262880 dur=640/640/896 seq=19831 lw=263360 2018-07-10 10:47:42.708910 [DEBUG] switch_rtp.c:5160 Send middle packet for [8] ts=262880 dur=800/800/896 seq=19832 lw=263520 2018-07-10 10:47:42.728910 [DEBUG] switch_rtp.c:5160 Send end packet for [8] ts=262880 dur=960/960/896 seq=19833 lw=263520 2018-07-10 10:47:42.728910 [DEBUG] switch_rtp.c:5160 Send end packet for [8] ts=262880 dur=960/960/896 seq=19834 lw=263520 2018-07-10 10:47:42.728910 [DEBUG] switch_rtp.c:5160 Send end packet for [8] ts=262880 dur=960/960/896 seq=19835 lw=263520 2018-07-10 10:47:42.728910 [DEBUG] switch_rtp.c:5108 Queue digit delay of 40ms 2018-07-10 10:47:43.548913 [DEBUG] switch_rtp.c:7534 RTP RECV DTMF 9:1152 2018-07-10 10:47:43.548913 [INFO] switch_channel.c:515 RECV DTMF 9:1152 2018-07-10 10:47:43.548913 [WARNING] goconf.js:2 Given userData.param of under the rock 2018-07-10 10:47:43.548913 [DEBUG] switch_rtp.c:5261 Send start packet for [9] ts=270080 dur=160/160/1152 seq=19876 lw=270080 2018-07-10 10:47:43.568915 [DEBUG] switch_rtp.c:5160 Send middle packet for [9] ts=270080 dur=320/320/1152 seq=19877 lw=270240 2018-07-10 10:47:43.588914 [DEBUG] switch_rtp.c:5160 Send middle packet for [9] ts=270080 dur=480/480/1152 seq=19878 lw=270400 2018-07-10 10:47:43.608915 [DEBUG] switch_rtp.c:5160 Send middle packet for [9] ts=270080 dur=640/640/1152 seq=19879 lw=270560 2018-07-10 10:47:43.628914 [DEBUG] switch_rtp.c:5160 Send middle packet for [9] ts=270080 dur=800/800/1152 seq=19880 lw=270720 2018-07-10 10:47:43.648914 [DEBUG] switch_rtp.c:5160 Send middle packet for [9] ts=270080 dur=960/960/1152 seq=19881 lw=270880 2018-07-10 10:47:43.668914 [DEBUG] switch_rtp.c:5160 Send middle packet for [9] ts=270080 dur=1120/1120/1152 seq=19882 lw=271040 2018-07-10 10:47:43.688914 [DEBUG] switch_rtp.c:5160 Send end packet for [9] ts=270080 dur=1280/1280/1152 seq=19883 lw=271040 2018-07-10 10:47:43.688914 [DEBUG] switch_rtp.c:5160 Send end packet for [9] ts=270080 dur=1280/1280/1152 seq=19884 lw=271040 2018-07-10 10:47:43.688914 [DEBUG] switch_rtp.c:5160 Send end packet for [9] ts=270080 dur=1280/1280/1152 seq=19885 lw=271040 2018-07-10 10:47:43.688914 [DEBUG] switch_rtp.c:5108 Queue digit delay of 40ms 2018-07-10 10:47:47.188914 [DEBUG] switch_rtp.c:7534 RTP RECV DTMF 9:1152 2018-07-10 10:47:47.188914 [INFO] switch_channel.c:515 RECV DTMF 9:1152 2018-07-10 10:47:47.188914 [WARNING] goconf.js:2 Given userData.param of under the rock 2018-07-10 10:47:47.188914 [DEBUG] switch_rtp.c:5261 Send start packet for [9] ts=299040 dur=160/160/1152 seq=20060 lw=299040 2018-07-10 10:47:47.208914 [DEBUG] switch_rtp.c:5160 Send middle packet for [9] ts=299040 dur=320/320/1152 seq=20061 lw=299200 2018-07-10 10:47:47.228914 [DEBUG] switch_rtp.c:5160 Send middle packet for [9] ts=299040 dur=480/480/1152 seq=20062 lw=299360 2018-07-10 10:47:47.248914 [DEBUG] switch_rtp.c:5160 Send middle packet for [9] ts=299040 dur=640/640/1152 seq=20063 lw=299520 2018-07-10 10:47:47.268914 [DEBUG] switch_rtp.c:5160 Send middle packet for [9] ts=299040 dur=800/800/1152 seq=20064 lw=299680 2018-07-10 10:47:47.288914 [DEBUG] switch_rtp.c:5160 Send middle packet for [9] ts=299040 dur=960/960/1152 seq=20065 lw=299840 2018-07-10 10:47:47.308918 [DEBUG] switch_rtp.c:5160 Send middle packet for [9] ts=299040 dur=1120/1120/1152 seq=20066 lw=300000 2018-07-10 10:47:47.328913 [DEBUG] switch_rtp.c:5160 Send end packet for [9] ts=299040 dur=1280/1280/1152 seq=20067 lw=300000 2018-07-10 10:47:47.328913 [DEBUG] switch_rtp.c:5160 Send end packet for [9] ts=299040 dur=1280/1280/1152 seq=20068 lw=300000 2018-07-10 10:47:47.328913 [DEBUG] switch_rtp.c:5160 Send end packet for [9] ts=299040 dur=1280/1280/1152 seq=20069 lw=300000 2018-07-10 10:47:47.328913 [DEBUG] switch_rtp.c:5108 Queue digit delay of 40ms 2018-07-10 10:47:50.168914 [DEBUG] switch_rtp.c:7534 RTP RECV DTMF 9:768 2018-07-10 10:47:50.168914 [INFO] switch_channel.c:515 RECV DTMF 9:768 2018-07-10 10:47:50.168914 [WARNING] goconf.js:2 Given userData.param of under the rock 2018-07-10 10:47:50.168914 [DEBUG] switch_rtp.c:5261 Send start packet for [9] ts=322720 dur=160/160/768 seq=20211 lw=322720 2018-07-10 10:47:50.188912 [DEBUG] switch_rtp.c:5160 Send middle packet for [9] ts=322720 dur=320/320/768 seq=20212 lw=322880 2018-07-10 10:47:50.208912 [DEBUG] switch_rtp.c:5160 Send middle packet for [9] ts=322720 dur=480/480/768 seq=20213 lw=323040 2018-07-10 10:47:50.228912 [DEBUG] switch_rtp.c:5160 Send middle packet for [9] ts=322720 dur=640/640/768 seq=20214 lw=323200 2018-07-10 10:47:50.248912 [DEBUG] switch_rtp.c:5160 Send end packet for [9] ts=322720 dur=800/800/768 seq=20215 lw=323200 2018-07-10 10:47:50.248912 [DEBUG] switch_rtp.c:5160 Send end packet for [9] ts=322720 dur=800/800/768 seq=20216 lw=323200 2018-07-10 10:47:50.248912 [DEBUG] switch_rtp.c:5160 Send end packet for [9] ts=322720 dur=800/800/768 seq=20217 lw=323200 2018-07-10 10:47:50.248912 [DEBUG] switch_rtp.c:5108 Queue digit delay of 40ms From: Francesco Facco de Lagarda <francesco(a)delagarda.com> Sent: martedì 10 luglio 2018 10:46 To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: bridge call back locks fs_cli in js Guys please help! Ive been loosing sleep for 4 days on this one. Im trying to : * Answer an internal call * Send to operator (ext 1000) * Wait for dtmf from operator * Do something.. But, the moment op hits a key, script dies and fs_cli disconnects! Ive tried adapting the given script to do bridge with callback, but simply changing the external session (mySession) to dial an extension instead of an external number, and effect is the same! function logAndStopBridge ( session, type, dtmf, userData) { console_log("WARNING","Given userData.param of "+(userData && userData.param)); return false; } session.answer(); while ( session.ready() ) { // var mySession = new Session("sofia/sip/mycall(a)mydomain.com <mailto:sofia/sip/mycall@mydomain.com> "); var mySession = new Session("user/1000"); bridge(session, mySession, logAndStopBridge, {param:"under the rock"}); } I never get the WARNING .. given etc on internal numbers (extension), only works on external lines..

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bridge call back locks fs_cli in js
by Francesco Facco de Lagarda 10 Jul '18

10 Jul '18

Guys please help! I've been loosing sleep for 4 days on this one. I'm trying to : * Answer an internal call * Send to operator (ext 1000) * Wait for dtmf from operator * Do something.. But, the moment op hits a key, script dies and fs_cli disconnects! I've tried adapting the given script to do bridge with callback, but simply changing the external session (mySession) to dial an extension instead of an external number, and effect is the same! function logAndStopBridge ( session, type, dtmf, userData) { console_log("WARNING","Given userData.param of "+(userData && userData.param)); return false; } session.answer(); while ( session.ready() ) { // var mySession = new Session("sofia/sip/mycall(a)mydomain.com <mailto:sofia/sip/mycall@mydomain.com> "); var mySession = new Session("user/1000"); bridge(session, mySession, logAndStopBridge, {param:"under the rock"}); } I never get the "WARNING" .. given .etc on internal numbers (extension), only works on external lines..

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