August 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

[SOLVED] RE: [ERR] mod_sofia.c:4921 Invalid Gateway 'provider'
by Edwin Reed-Sanchez 15 Aug '15

15 Aug '15

[SOLVED] The issue was in the vars.xml file. <X-PRE-PROCESS cmd="set" data="external_ip=192.168.X.XXX"/> the ip address needed to be set to internet IP address connected to server. > ---------- Forwarded message ---------- > From: Steven Ayre <steveayre(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Fri, 14 Aug 2015 21:32:10 +0100 > Subject: Re: [Freeswitch-users] [ERR] mod_sofia.c:4921 Invalid Gateway > 'provider > >> [ERR] mod_sofia.c:4921 Invalid Gateway 'provider' > > > This means you're using the dialstring sofia/gateway/provider/... and > there's no gateway named 'provider'. In your example you should use > sofia/gateway/callwithus-outbound/... > > Invalid Profile [external] > > > This means the profile hasn't been loaded. Try 'sofia profile external > start' and check for any error messages creating it. > > On 13 August 2015 at 23:48, Edwin Reed-Sanchez <ereedsanchez(a)gmail.com> > wrote: > >> >> I am getting error >> >> [ERR] mod_sofia.c:4921 Invalid Gateway 'provider' >> >> when trying to make a long distance call. >> >> >> My /freeswitch/sip-profiles/external/profile.xml is >> >> ------ >> >> <include> >> >> <gateway name="callwithus-outbound"> >> >> <param name="username" value="05617XXXX"/> >> >> <param name="password" value="05XXXX"/> >> >> <param name="realm" value="callwithus"/> >> >> <param name="proxy" value="sip.callwithus.com"/> >> >> <param name="register" value="true"/> >> >> </gateway> >> >> </include> >> >> ------ >> >> >> In fs_cli I did >> >> sofia profile external rescan >> >> Invalid Profile [external] >> >> >> Thanks, >> -- >> Edwin Reed-Sanchez >> >> >> >

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Fork( ) and Exec ( ) functions
by Кирилл Преслер 15 Aug '15

15 Aug '15

Abdul, have you any success? -- С уважением, Преслер Кирилл Сергеевич Инженер ООО «Мегафит» e-mail: 112(a)megafit.su тел: +7 (383) 311-09-09 (доп.112)

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[ERR] mod_sofia.c:4921 Invalid Gateway 'provider
by Edwin Reed-Sanchez 14 Aug '15

14 Aug '15

I am getting error [ERR] mod_sofia.c:4921 Invalid Gateway 'provider' when trying to make a long distance call. My /freeswitch/sip-profiles/external/profile.xml is ------ <include> <gateway name="callwithus-outbound"> <param name="username" value="05617XXXX"/> <param name="password" value="05XXXX"/> <param name="realm" value="callwithus"/> <param name="proxy" value="sip.callwithus.com"/> <param name="register" value="true"/> </gateway> </include> ------ In fs_cli I did sofia profile external rescan Invalid Profile [external] Thanks, -- Edwin Reed-Sanchez

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Conectin a SPA3102 as pstn gateway on FS
by Jaime Cruz 14 Aug '15

14 Aug '15

Hi Everyone, I'm trying to use a SPA3102 as a PSTN gateway, it is in a remote site with a dinamic IP, I'have it registered in my FS as 100123 user, and my goal is to dial <any pstn number>@<my spa3102> in order to use my SPA3120 to make the call via his pstn line to the number <any pstn number>. The problem is that all the examples I see has a fixed and well know IP address for the SPA3102 gateway in order to use it in the dial plan, but in my scenario the IP address for my SPA3102 gateway is dinamyc and unknow. Please any help in order to understand how can I dial any number to my spa3102 gateway will by apreciated. In this example my FS has the IP 100.10.10.10 and the SPA3102 has the dinamic Adsl IP 210.20.20.20 > sofia status profile internal reg Registrations: ================================================================================================= Call-ID: 2378632f-eb1116d(a)192.168.0.11 User: 100123(a)100.10.10.10 Contact: "Local Call" <sip:100123@210.20.20.20:5061> Agent: Linksys/SPA3000-3.1.20(GW) Status: Registered(UDP)(unknown) EXP(2015-08-08 04:01:04) EXPSECS(61) Ping-Status: Reachable Host: voip.example.com IP: 210.20.20.20 Port: 5061 Auth-User: 100123 Auth-Realm: voip.example.com MWI-Account: 100123(a)100.10.10.10 ================================================================================================= for example, if I want to dial to the pstn number 222-1234, then I need to send an INVITE 2221234@210.20.20.20:5061 Thanks for your help. -- Jaime Cruz jaimecm(a)gmail.com https://www.facebook.com/jaimecruzmendoza

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FreeSWITCH proxy-acl evaluation
by Muhammad Naseer Bhatti 14 Aug '15

14 Aug '15

I have two profiles, out of which 1 ACL is allowing everyone to make calls, because I am forwarding all requests to RADUIS for authentication. That works fine. The second profile, I want to use proxy-acl. The proxy (OpenSIPS) is authenticated but then FreeSWITCH looks for x-auth-ip (Added by proxy) in inbound-acl list. If my cidr is 0.0.0.0 in inbound-acl, then offcourse everyone is allowed for INVITE, and proxy-acl is never evaluated. Is there a way I can check against proxy-acl first, and those calls coming in from there/proxy authenticated (on this profile) I don’t check them against inbound-acl? Thank you, — Muhammad Naseer Bhatti

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 14 Aug '15

14 Aug '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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Virtual DB FreeSWITCH
by Usama Zaidi 14 Aug '15

14 Aug '15

So I found the solution, using HAProxy would be a viable fix. Regards!

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grab RTMP stream into FreeSWITCH
by Marcin Muzylo 14 Aug '15

14 Aug '15

Hi , maybe someone know how to grab some RTMP stream (rtmp://) into FreeSWITCH (to connect it for example to conference room)? there is mod_rtmp - but it seems to be only for web client (tried with flex - works) Anyone can recommend any good RTMP 2 SIP that works ? thanks for any help Marcin

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Loopback SIP invite to originator
by Jurijs Ivolga 14 Aug '15

14 Aug '15

Hi, I need to receive call on Freeswitch and Loopback it to original server from where it came into, without changing any SIP header(to, from, SIP request and etc). Is there easy way to do this? Thank you! With kind regards, Jurijs

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SRTP on outbound leg without TLS
by Jurijs Ivolga 14 Aug '15

14 Aug '15

Hi, I'm struggling with quite simple issue. I need to enable SRTP on outbound leg. Call hits Freeswitch as SRTP but it leaves as regular RTP. I do not use TLS and I don't need it(yes, I know that SRTP keys are sent as plain text in this case). I tried to add following code to my dialplan, but it do not helps: <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never"> <action application="set" data="sip_secure_media=true"/> <action application="export" data="sip_secure_media=true"/> </condition> I tried to add to vars.xml following line too: <X-PRE-PROCESS cmd="set" data="rtp_secure_media_inbound=mandatory"/> But still without success. Maybe somebody can give me a hint? Thank you! With kind regards, Jurijs

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