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SIPS scheme
by David Villasmil 15 May '17

15 May '17

Hello guys. I've got this setup, where the client is sending as contact "sip:" coming from kamailio proxy in tls ip until kamailio. Kamailio forwards to FS in tcp, but freeswitch is answering with "sip:" and this breaks the client who answers "SIPS Required"... Is there a way of specifying the contact should be sips ? Has anyone encountered this and solved it? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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300 Multiple Choices - how to force FS to replace From field
by Lyubo Popov 15 May '17

15 May '17

Hello everyone, I would like to express my thanks in advance to anyone who may be able to help me with some insides. I am using a routing software with SIP Redirect to send routes to FS with 300 Multiple Choices and mod_xml_radius to authenticate the SIP users. In the Sip redirect server I am manipulating as well the FROM number and sending back to FS, but FS will not respect this and continue using the SIP account that sent the call in the first place in the FROM field. Here are some SIP packets from both sides to clarify the whole process. 1. Sending call to FS with Zoiper, destination 556230951662 INVITE sip:556230951662@216.x.x.x:5080;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 177.x.x.x:1048;branch=z9hG4bK-d8754z-038f1c7d251308c2-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:551000@177.41.146.98:1048;transport=UDP> To: <sip:556230951662@216.x.x.x:5080;transport=UDP> From: "551000"<sip:551000@216.x.x.x:5080;transport=UDP>;tag=25599d20 Call-ID: NmUzYTAwNmQ1NTZjMDM2ZjVhYTgzMDdiY2RiMmI0ZTc. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Zoiper for Windows 2.43 r24984 Allow-Events: presence, kpml Content-Length: 232 v=0 o=Zoiper_user 0 0 IN IP4 177.x.x.x s=Zoiper_session c=IN IP4 177.x.x.x t=0 0 m=audio 8000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv 2. FS sending INVITE to SIP Redirect server INVITE sip:556230951662@69.x.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP 216.245.218.230;rport;branch=z9hG4bKateZg87rDBpZa Max-Forwards: 69 From: "551000" <sip:551000@177.x.x.x>;tag=FeNXS71300N0c To: <sip:556230951662@69.x.x.x:5060> Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Contact: <sip:mod_sofia@216.x.x.x:5060> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 397 X-FS-Support: update_display,send_info Remote-Party-ID: "551000" <sip:551000@216.x.x.x >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1493809233 1493809234 IN IP4 216.x.x.x 2017-05-03 12:15:09.816119 [ERR] mod_xml_radius.c:911 Didn't match: 69.x.x.x:5060 == ^69\.x\.x\.x s=FreeSWITCH c=IN IP4 216.x.x.x t=0 0 m=audio 22476 RTP/AVP 8 0 18 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=audio 22476 RTP/AVP 4 101 13 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 3. SIP Redirect returns 300 Multiple choices with the termination IP in Contact and with FROM field as instructed ( update 551000 with 1140031556) SIP/2.0 300 Multiple Choices Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKateZg87rDBpZa From: "1140031556" <sip:1140031556@177.x.x.x>;tag=FeNXS71300N0c To: <sip:556230951662@69.x.x.x:5060> Contact: <sip:556230951662@162.x.x.x:5060>;q=1.00 Call-ID: e70ee145-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Max-Forwards: 69 Content-Length: 0 Server: SIP Redirect Server 4. FS will send the call to the Termination IP WITHOUT changing the FROM field INVITE sip:556230951662@162.x.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP 216.x.x.x;rport;branch=z9hG4bKB37Qj3rvamcjp Max-Forwards: 68 From: "551000" <sip:551000@216.x.x.x>;tag=gQepU2j7X9BKr To: <sip:556230951662@162.x.x.x:5060> Call-ID: e7159715-aac6-1235-79ba-002590a0ec9b CSeq: 106579790 INVITE Contact: <sip:mod_sofia@216.x.x.x:5060> 2017-05-03 12:15:09.856127 [ERR] mod_xml_radius.c:914 Result of true match: 162.x.x.x:5060 == ^69\.x\.x\.x User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 397 X-FS-Support: update_display,send_info Remote-Party-ID: "551000" <sip:551000@216.x.x.x >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1493811013 1493811014 IN IP4 216.x.x.x s=FreeSWITCH c=IN IP4 216.x.x.x t=0 0 m=audio 20696 RTP/AVP 8 0 18 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=audio 20696 RTP/AVP 4 101 13 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 This is the dialplan I use to send calls to SIP redirect server <include> <extension name="rejections"> <condition field="${radius_auth_result}" expression="2"> <action application="hangup" data="CALL_REJECTED"/> </condition> </extension> <extension name="timedouts"> <condition field="${radius_auth_result}" expression="1"> <action application="hangup" data="SWITCH_CONGESTION"/> </condition> </extension> <extension name="SIP Redirect Server"> <condition field="${radius_auth_result}" expression="0"/> <condition field="destination_number" expression="^(.+)$">  <action application="export" data="nolocal:h323-call-origin=originate"/> <action application="set" data="sip_h_X-accountcode=${accountcode}" /> <action application="set" data="call_direction=outbound" /> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="inherit_codec=true" /> <action application="set" data="call_timeout=20"/> <action application="set" data="fail_on_single_reject=USER_BUSY" /> <action application="set" data="origination_caller_id_name=${sip_req_user}"/> <action application="set" data="origination_caller_id_number=${sip_from_user}"/> <action application="set" data="execute_on_answer=sched_hangup +${h323-credit-time} alloted_timeout" /> <action application="bridge" data="{sip_invite_from_uri=sip:${sip_from_user}@ ${sip_network_ip}}sofia/internal/${destination_number}@69.x.x.x:5060" /> <action application="hangup" data="${bridge_hangup_cause}"/> </condition> </extension> </include Is there any variable that will force FS to change the FROM field as returned by the SIP Redirect server and send it to the termination provider? Any help on this is really greatly appreciated! Best regards, L. Popov

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Presence Server
by kk-mailinglist＠ednt.de 13 May '17

13 May '17

i try to use the LED´s on the Phone to sgive user a indication of the state of my Phone is redirected or ithis Phone is subscribed in a Queue. For this i use a free extension in sample 1000 and put on the Phone the LED to BLF 1000. This Works until the phone subscribe again after 3600 sec. The my manipulation will be lost in the presence server. So i need a way to take care about this state. Is there a common solution for this issue ? regards

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TLS SIP packet tracing and visualization
by Giovanni Maruzzelli 12 May '17

12 May '17

Hello fellows, after some experimentation with various tools, I come out with a little shell tool that maybe can be useful to you too. It can only work with non-forward secrecy ciphers, obviously, and only if is started before the client do the initial TLS handshake (eg, just restart the client). Forward secrecy cannot be decrypted after fact, so don't waste effort. An example of ciphers that can be decrypted are the "AES256-SHA" openssl cipher group. You can use ssldump to check what cipher is used by serverhello. Enjoy, make it better, and share it :) #!/bin/bash # brought to you by Giovanni Maruzzelli # SERVERIP="192.168.1.150" SERVERPORT="5061" PRIVKEY="/etc/certs/privkey.pem" STDERR2DEVNULL=" 2>/dev/null " REGEX="notyet" if [ -z "$1" ]; then REGEX="\\\.*" else REGEX="$1" fi FILTER="ssl.app_data and sip matches" FILTER2="$FILTER \"$REGEX\"" FILTER3="'$FILTER2'" ARGUMENT="-i 1 -Y $FILTER3 -E header=y -T fields -e frame.number -e frame.time -e frame.time_delta_displayed -e ip.src -e ip.dst -e sip.Status-Line -e sip.Request-Line -e sip.msg_hdr -l -d tcp.port\=\=5061,sip -o \"ssl.keys_list: $SERVERIP,$SERVERPORT,sip,$PRIVKEY\" $STDERR2DEVNULL | sed -u 's/\t/\n/g' | sed -u '/^$/d' | sed -u 's/^[0-9]*$/\n==&==============================/g'" echo "" echo "NB: if it do not works, edit script so that STDERR2DEVNULL=\" \" and try again" echo "" echo "NB: remember to quote and escape match patterns, using triple slash" echo " eg, for matching 1010(a)pbx.example.com, use \"1010(a)pbx.example.com \"" echo " eg, for matching anything, use \"\\\\\\.*\"" echo " eg, for matching *98, use \"\\\\\\*98\"" echo "USAGE: $0 \"\\\\\\*98(a)pbx.example.com\"" echo "" case "$1" in -help|--help|?) exit 0 ;; *) echo "THIS TIME WE'RE DOING:" echo "tshark $ARGUMENT" echo "" bash -c "tshark $ARGUMENT" ;; esac -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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Voicemail not detecting DTMF
by Ketan Kothari 12 May '17

12 May '17

Hello, I'm trying to call voicemail box number 7777 to hear the recorded messages if any, and its asking for password but when enter digit its not showing in freeswitch logs after some seconds its again asking for password. https://pastebin.com/ypZFvXD2 after the above log if i press digit '1' i should get something like this, [DEBUG] switch_rtp.c:6927 RTP RECV DTMF 1:960 But i am getting nothing. However at the time of leaving the voicemail, the DTMF is detected fine. what could be the issue?

1 0

mod_distribution feature
by Antonio Silva 12 May '17

12 May '17

Hi all, Just found mod_distribution and is pretty cool feature to have. Right now it distributes calls to gateways in a weighted round-robin fashion, could it be cool to have /Weighted Random Distribution. /i found this paper explaining an implementation in python: https://www.electricmonk.nl/log/2009/12/23/weighted-random-distribution/ it could be ported to mod_distributor. / /What you guys think about it? -- Saludos / Regards / Cumprimentos, António silva

2 1

STRP
by David Villasmil 12 May '17

12 May '17

Hello Guys, So i've got this scenario where i'm receving in TCP+SRTP(opus) and I need to send to the b-leg in TCP+STRP(PCMA)... i've enabled: <action application="bridge" data="{sip_secure_media=true,sdp_secure_savp_only=true,absolute_codec_string=^^:PCMA:PCMU}sofia/external/${destination_number}@myserver;transport=tcp" /> But the sdp is being went without srtp, what am I missing? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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Changing audio DTLS state from HANDSHAKE to FAIL issue
by sachin 11 May '17

11 May '17

Hello All,I am trying to setup the Webrtc call between 2 SIPML5 client. I am using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN.The clients are getting registered over wss. I have created self signed certificates.  In var.xml I have set the codecs setting as follows  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA">  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA">I am able to establish the call and there is 2 way voice when I call from Sipmpl5 to Linphone and it works. But when I call from Linphone to Firefox (Simpl5) then the call is not getting established and I am getting following error2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake failure 12017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS state from HANDSHAKE to FAIL2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup sofia/internal/sip:1001@df7jal23ls0d.invalid [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER]My setup is as followsSIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> LinphoneI am attaching the logs for the reference.fs-logleve9.txt : Debug trace with loglevel =9fs-sip-trace.txt : Sip tracePlease let me know what could the issue and pointers to resolve the same.Thanks and RegardsSD

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Freeswitch Packet Fragmentation issue
by Charles Bujold 11 May '17

11 May '17

The only codec is PCMU, nothing else listed.

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Why two ilbc(ilbc2) RPMs in CentOS 7 repos
by Richard Chan 11 May '17

11 May '17

Hi, I was wondering what the purpose is of having two RPMs for ilbc, named ilbc and ilbc2. Does the build system check for both? In Fedora (not sure for CentOS) there is an incompatible RPM so perhaps to avoid a package name clash? -- Richard Chan

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