- FreeSWITCH-users - FreeSWITCH Mailing Lists

issue with Acct-Status-Type
by Miha 16 Jan '17

16 Jan '17

Hello i need a little help. I am getting in Acct-Status-Type = 0 and due to this "0" i get this error: softswitch_rs: EXPAND %{tolower:type.%{Acct-Status-Type}.query} (39) softswitch_rs: --> type.0.query No such configuration item .type.0.query Reference is define like this: reference = "%{tolower:type.%{Acct-Status-Type}.query} How can i set in default (sites-available) that ifAcct-Status-Type = 0 i set it to Stop so that the same confg will be like for "Stop". Tnx Miha

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Configuring FreeSWITCH for federation with other platforms
by Shaun Stokes 13 Jan '17

13 Jan '17

Hi All, Has anyone tried or have any guidance on configuring FreeSWITCH for federation with other platforms (i.e. via XMPP using DNS)? I've looked at SylkServer but as far as I can tell it's a closed platform on SIP2SIP\SIP Thor. Perhaps this might be possible using Openfire integration with FreeSWITCH? Thanks, Shaun [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes(a)itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________

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FreeSwitch as proxy for "407 Proxy Authentication Required"
by vivek＠advaitamtech.com 13 Jan '17

13 Jan '17

Hi All, Am trying to proxy the "407 Proxy Authentication Required" through freeswitch, My Objective is to proxy the "407 Proxy Authentication Required" received from the third party server to another server which is connected to freeswitch. Is it possible to do this?. If possible how do I do it. Please help me out. Thanks, Vivek.

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Profile misunderstanding
by Igor Olhovskiy 13 Jan '17

13 Jan '17

Hi! I have profile named default: sofia status profile default ================================================================================================= Name default Domain Name N/A Auto-NAT false DBName sofia_reg_default Pres Hosts 172.31.1.100,172.31.1.100 Dialplan XML Context default Challenge Realm auto_from RTP-IP 172.31.1.100 Ext-RTP-IP <EXT_ADDR> SIP-IP 172.31.1.100 Ext-SIP-IP <EXT_ADDR> URL sip:mod_sofia@<EXT_ADDR>:5060 BIND-URL sip:mod_sofia@<EXT_ADDR>:5060;maddr=172.31.1.100;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMA,PCMU,G729 CODECS OUT PCMA,PCMU,G729 TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false ZRTP-PASSTHRU false AGGRESSIVENAT true Also: sip-ip [172.31.1.100] sip-port [5060] rtp-ip [172.31.1.100] dialplan [XML] user-agent-string [ASTPP] debug [0] sip-trace [no] tls [false] inbound-reg-force-matching-username [true] disable-transcoding [true] all-reg-options-ping [false] unregister-on-options-fail [true] log-auth-failures [true] status [0] inbound-bypass-media [false] inbound-proxy-media [false] disable-transfer [true] enable-100rel [false] rtp-timeout-sec [400] dtmf-duration [2000] manual-redirect [true] aggressive-nat-detection [true] enable-timer [false] minimum-session-expires [120] session-timeout-pt [1800] auth-calls [true] apply-inbound-acl [default] inbound-codec-prefs [PCMA,PCMU,G729] outbound-codec-prefs [PCMA,PCMU,G729] inbound-late-negotiation [false] sip-capture [no] forward-unsolicited-mwi-notify [false] context [default] rfc2833-pt [101] rtp-timer-name [soft] hold-music [local_stream://moh] manage-presence [true] presence-hosts [172.31.1.100,172.31.1.100] presence-privacy [false] inbound-codec-negotiation [generous] auth-all-packets [false] ext-rtp-ip [<EXT_ADDR>] ext-sip-ip [<EXT_ADDR>] rtp-hold-timeout-sec [1800] force-register-domain [172.31.1.100] force-subscription-domain [172.31.1.100] force-register-db-domain [172.31.1.100] challenge-realm [auto_from] nonce-ttl [60] pass-callee-id [false] rtcp-audio-interval-msec [5000] local-network-acl [localnet.auto] NDLB-force-rport [true] But problem is, when client tries to connect to FS, it still answers to port 5060, not looking on NDLB-force-rport option. As well, as private addresses shown across packets, not using ext-sip and ext-rtp options. ------------------------------------------------------------------------ recv 378 bytes from udp/[136.169.20.219]:3589 at 19:54:34.619718: ------------------------------------------------------------------------ REGISTER sip:172.31.1.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK322184305 From: <sip:6399081327@172.31.1.100:5060>;tag=1569039960 To: <sip:6399081327@172.31.1.100:5060> Call-ID: 1500202051(a)192.168.0.101 CSeq: 197 REGISTER Contact: <sip:6399081327@192.168.0.101:5060>;expires=5 Max-Forwards: 30 User-Agent: dble Expires: 5 Content-Length: 0 ------------------------------------------------------------------------ send 571 bytes to udp/[136.169.20.219]:5060 at 19:54:34.619864: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK322184305;received=136.169.20.219 From: <sip:6399081327@172.31.1.100:5060>;tag=1569039960 To: <sip:6399081327@172.31.1.100:5060>;tag=jD7SQFyXcK1gr Call-ID: 1500202051(a)192.168.0.101 CSeq: 197 REGISTER User-Agent: ASTPP Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces WWW-Authenticate: Digest realm="172.31.1.100", nonce="7d83af94-2a69-40cc-a9c3-1659945a34b2", algorithm=MD5, qop="auth" Content-Length: 0 Can you please point what I’m missing? Thanks! Regards, Igor

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issue with disposition cause overriding
by devang nathwani 13 Jan '17

13 Jan '17

i am trying to override specific disposition cause coming from leg B and sending the modified cause to leg A using, https://freeswitch.org/confluence/display/FREESWITCH/Cause+Code+Substitutio… the lua file returns sip_invite_failure_status variable as '-ERR No such channel!' and not the code 480 However, from sip trace i can see '480 temporary unavailable' from leg B and because of that my lua script which i am calling before bridge application is not able to override the cause from 480 to 503 What i am missing?

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rtp port update after reinvite when initiator is behind nat
by lionelfont＠free.fr 13 Jan '17

13 Jan '17

Hi, i'm an user from freeswitch because my trunk's provider is using a freeswitch. i can't give you the configuration using by my provider and i can't give you any trace from freeswitch but i'd like explain my problem to know if anyone have an idea from what can cause my issue. My situation: Freeswitch (public ip A.A.A.A) <----> WAN <---> (my public ip B.B.B.B) FIREWALL <----> (my private ip C.C.C.C) IPBX ALCATEL During a call, after invite or reinvite, Freeswitch send me RTP DATA on wrong RTP port... For sample: I receive a call, after invite from FS and OK from my IPBX, we have a communication who is good. We hear on the two sides. FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB1 ---------------- C.C.C.C:RTPC1 <> IPBX If i need to hold the call, all is ok. My holding music is hearing. FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB1 ---------------- C.C.C.C:RTPC1 <> IPBX FS<-- A.A.A.A:SIPA1 -------- B.B.B.B:SIPB1 -------------- C.C.C.C:SIPC1 <-- IPBX (INVITE) (OK) FS --> A.A.A.A:SIPA1 ------- B.B.B.B:SIPB1 -------------- C.C.C.C:SIPC1 --> IPBX FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB2 ---------------- C.C.C.C:RTPC2 <> IPBX But when i unhold the call, "randomly", FS send me RTP DATA on the RTP port used before the unhold.. And we have no sound on the two sides... Case good: FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB2 ---------------- C.C.C.C:RTPC2 <> IPBX FS<-- A.A.A.A:SIPA1 -------- B.B.B.B:SIPB1 -------------- C.C.C.C:SIPC1 <-- IPBX (INVITE) (OK) FS --> A.A.A.A:SIPA1 ------- B.B.B.B:SIPB1 -------------- C.C.C.C:SIPC1 --> IPBX FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB3 ---------------- C.C.C.C:RTPC3 <> IPBX Case bad: FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB2 ---------------- C.C.C.C:RTPC2 <> IPBX FS<-- A.A.A.A:SIPA1 -------- B.B.B.B:SIPB1 -------------- C.C.C.C:SIPC1 <-- IPBX (INVITE) (OK) FS --> A.A.A.A:SIPA1 ------- B.B.B.B:SIPB1 -------------- C.C.C.C:SIPC1 --> IPBX FS<-- A.A.A.A:RTPA1 -------- B.B.B.B:RTPB3 ---------------- C.C.C.C:RTPC3 <-- IPBX FS--> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB2 ---------------- C.C.C.C:RTPC2 --> IPBX I don't understand why the rtp port is not good updating by freeswitch randomly. I know freeswitch don't use the port that my IPBX give in INVITE packet because it's not the good with the nat from my firewall. I suppose that's the "NDLB-connectile-dysfunction" who is used by freeswitch to find the good rtp port. If with a little congestion, Freeswitch receive after my reinvite, one or two RTP packet from previous RTP DATA, Freeswitch can use the wrong RTP port? Is there a parameter to ignore after reinvite the five first RTP packet to autoadjust the RTP port? Thanks to all helping and sorry for my poor english. Best regards

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Re: [Freeswitch-users] multiple profiles and user not registered (SOLVED)
by David Villasmil 13 Jan '17

13 Jan '17

John, <action application="bridge" data="${sofia_contact(*/1001(a)1.2.3.4)}"/> note: 1.2.3.4 being my domain (or IP1) worked perfectly! Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 On Thu, Jan 12, 2017 at 6:52 PM, David Villasmil < david.villasmil.work(a)gmail.com> wrote: > I will try this as soon as i can > > Thanks! > > On Thu, Jan 12, 2017 at 2:59 PM John Covici <covici(a)ccs.covici.com> wrote: > >> OK, the form of this is on the bridge itself use <action >> application="bridge" >> data="${sofia_contact(*/${dialed_extension}@${domain})} and what >> ever else. >> >> Hope this helps. >> >> On Thu, 12 Jan 2017 14:29:24 -0500, >> David Villasmil wrote: >> > >> > [1 <multipart/alternative (7bit)>] >> > [1.1 <text/plain; UTF-8 (quoted-printable)>] >> > [1.2 <text/html; UTF-8 (quoted-printable)>] >> > Hello >> > >> > That didn't worked for me, and i can't find anything similar to that >> > >> > David >> > >> > On Thu, Jan 12, 2017 at 12:12 PM John Covici <covici(a)ccs.covici.com> >> wrote: >> > >> > There is a form of sofia-contact which will search all profiles, if I >> > remember its */ followed by user name. >> > >> > On Thu, 12 Jan 2017 11:46:06 -0500, >> > David Villasmil wrote: >> > > >> > > [1 <multipart/alternative (7bit)>] >> > > [1.1 <text/plain; UTF-8 (quoted-printable)>] >> > > [1.2 <text/html; UTF-8 (quoted-printable)>] >> > > this does work: >> > > >> > > <action application="bridge" data="${sofia_contact( >> internal-second/1001)}"/> >> > > >> > > but i have 2 profiles through which the user can be called, >> "internal" and "internal-second", and doing it like this kind of kills the >> purpose, no? >> > > >> > > I can think of maybe doing: >> > > >> > > <extension name="call_internal" continue="true"> >> > > <condition field="context" expression="public" /> >> > > <condition field="destination_number" expression="1001"> >> > > <action application="set" data="sofia_result=${sofia_ >> contact(internal/${destination_number})}"/> >> > > <action application="log" data="INFO sofia contact result: >> ${sofia_result}"/> >> > > <condition field="${result}" data="sofia.*"> >> > > >> > > <action application="bridge" data="sofia/internal/${ >> destination_number}"/> >> > > <anti-action application="bridge" data="sofia/internal-second/${ >> destination_number}"/> >> > > >> > > </condition> >> > > </condition> >> > > </extension> >> > > >> > > i will try this >> > > * >> > > ᐧ >> > > >> > > Regards, >> > > >> > > David Villasmil >> > > email: david.villasmil.work(a)gmail.com >> > > phone: +34669448337 <+34%20669%2044%2083%2037> >> > > >> > > On Thu, Jan 12, 2017 at 11:20 AM, David Villasmil < >> david.villasmil.work(a)gmail.com> wrote: >> > > >> > > In the form of: >> > > >> > > <action application="bridge" data="user/${sofia_contact( >> 361110000)}"/> >> > > >> > > didn't work... was that the correct way of using it? >> > > >> > > thanks again >> > > * >> > > ᐧ >> > > >> > > Regards, >> > > >> > > David Villasmil >> > > email: david.villasmil.work(a)gmail.com >> > > phone: +34669448337 <+34%20669%2044%2083%2037> >> > > >> > > On Thu, Jan 12, 2017 at 10:45 AM, Yehavi Bourvine < >> yehavi.bourvine(a)gmail.com> wrote: >> > > >> > > try bridging to sofia_contact(1001) and see whether it works. >> > > >> > > __Yehavi: >> > > >> > > 2017-01-12 16:24 GMT+02:00 David Villasmil < >> david.villasmil.work(a)gmail.com>: >> > > >> > > Hello, >> > > >> > > Thanks for replying! I'm simply bridging to the user like >> > > >> > > user/1001 >> > > >> > > This fails, of course. >> > > >> > > David >> > > >> > > On Wed, Jan 11, 2017 at 11:45 PM Yehavi Bourvine < >> yehavi.bourvine(a)gmail.com> wrote: >> > > >> > > Can you post your bridge command call? Do you use sofia_contact() to >> build the dial string? >> > > >> > > Regards, __Yehavi: >> > > >> > > 2017-01-11 4:45 GMT+02:00 David Villasmil < >> david.villasmil.work(a)gmail.com>: >> > > >> > > hello guys, >> > > >> > > i have 2 profiles for two different ip addresses (IP1 and IP2), both >> profiles have force-register-domain and force-register-db-domain to the >> IP1, i can register on IP2 but when calling this user like user@IP1, fs >> > > returns USER_NOT_REGISTERED... i thought by forcing the register >> domain i could call the user to any of the 2 ips... any ideas? >> > > >> > > Regards, >> > > >> > > David Villasmil >> > > email: david.villasmil.work(a)gmail.com >> > > phone: +34669448337 <+34%20669%2044%2083%2037> >> > > * >> > > ᐧ >> > > >> > > ____________________________________________________________ >> _____________ >> > > 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 >> > > >> > > ____________________________________________________________ >> _____________ >> > > 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 >> > > >> > > ____________________________________________________________ >> _____________ >> > > 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 >> > > >> > > ____________________________________________________________ >> _____________ >> > > 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 >> > > >> > > [2 <text/plain; us-ascii (7bit)>] >> > > ____________________________________________________________ >> _____________ >> > > 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 >> > >> > -- >> > Your life is like a penny. You're going to lose it. The question is: >> > How do >> > you spend it? >> > >> > John Covici >> > covici(a)ccs.covici.com >> > >> > ____________________________________________________________ >> _____________ >> > 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 >> > >> > [2 <text/plain; us-ascii (7bit)>] >> > ____________________________________________________________ >> _____________ >> > 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 >> >> -- >> Your life is like a penny. You're going to lose it. The question is: >> How do >> you spend it? >> >> John Covici >> covici(a)ccs.covici.com >> >> _________________________________________________________________________ >> 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 > > ᐧ

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Quick Question
by Magnus Kelly 13 Jan '17

13 Jan '17

Hello All, would be very helpful if someone could help me with quick reminder on how to change SIP header on bridge as in - incoming call on LegA Bridged to Leg B while changing the "ip:FreeSWITCH" to sip: +44xxxxxxxxxx now this format appears as default - From: "+44xxxxxxxxxx" <sip:FreeSWITCH@10.11X.X.XX:5068;transport=tcp>;tag= UB7tNX0mmBt7j And I would like help to change it to be in this format - From: "+44xxxxxxxxxx " <sip: +44xxxxxxxxxx@10.11X.X.X:5068; transport=tcp>;tag=UB7tNX0mmBt7j Thanks

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multiple profiles and user not registered
by David Villasmil 12 Jan '17

12 Jan '17

hello guys, i have 2 profiles for two different ip addresses (IP1 and IP2), both profiles have force-register-domain and force-register-db-domain to the IP1, i can register on IP2 but when calling this user like user@IP1, fs returns USER_NOT_REGISTERED... i thought by forcing the register domain i could call the user to any of the 2 ips... any ideas? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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B-leg for each user
by Murugan Pandian 12 Jan '17

12 Jan '17

I try get each user cdr from goup call.its work fine if user are sip when i make call user pstn using loopbackm i am getting only one leg event all user busy

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