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mod_cv example?
by Lyle Pratt 22 Sep '20

22 Sep '20

Can anyone provide an example of how to use or experiment with mod_cv? Googling around produces no results. Thanks! - Lyle

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Re: [Freeswitch-users] How to Answer a Call from ESL java
by Ehtasham Ul-Haq 22 Sep '20

22 Sep '20

Thanks, Normando, Actually we are dialing from SIP Phone A to SIP-Phone-B, by directly calling the extension. We are not calling form outside. Malik Ehtasham, Lead ProjDev WWW:[image: domain2.png].expertflow.com <http://www.expertflow.com/> FB: [image: FB-f-Logo__blue_29.png]/Expertflow <https://www.facebook.com/Expertflow> LinkedIn: [image: linkedIn.png] /company/expertflow <https://www.linkedin.com/company/expertflow> Youtube: [image: YouTube-social-square_red_128px.png]/user/expertflow <https://www.youtube.com/user/expertflow> Twitter: [image: twitter.JPG] /Expertflow <https://twitter.com/Expertflow> 361 Model Town Lahore Pakistan ; Mobile +92 3347815664,+92 334 4853664; email, Cisco Spark and Google Talk: ehtasham.malik(a)expertflow.com <andreas.stuber(a)expertflow.com>; Skype: <http://andreas.stuber.expertflow.com/>*shani.awan3* On Tue, Sep 22, 2020 at 9:48 PM Normando Hall via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Normando Hall <nhall(a)unixlan.com.ar> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Tue, 22 Sep 2020 13:08:22 -0300 > Subject: Re: [Freeswitch-users] How to Answer a Call from ESL java > > Of course, after you have handled park call, answer it or bridge or > transfer to destination > El 22/09/2020 a las 01:05 p. m., Normando Hall escribió: > > Hello Malik > > It is strange. Can you chare you dialplan extension where you park the > incoming call? > > I asume you are subscribed for all events and listen for park. > > This is my public dialplan extension: > > <include> <extension name="IVR_FXO"> <condition field="destination_number" expression="^(66[78][0-9])$"> <action application="park"/> </condition> </extension></include> > > The in my js code (snipet) > > conn.on(Events.RECEIVED, event => { const eventName = event.getHeader('Event-Name'); if(eventName != null){ const uuid = event.getHeader('Caller-Unique-ID'); let callerNumber = event.getHeader('Caller-ANI'); if(debug) callerNumber = '4275595' const context = event.getHeader('Caller-Context'); const destination = event.getHeader('Caller-Destination-Number'); const callDirection = event.getHeader('Call-Direction'); const internal = event.getHeader('variable_sip_user_agent'); //logger.info(eventName) switch (eventName) { case ChannelEvents.HANGUP: const hangupCause = event.getHeader('Hangup-Cause'); helpers.delChannel(uuid); //logger.info(hangupCause + ' ' + uuid) break; *case* *ChannelEvents**.**PARK**:* //logger.info("ID: " + uuid + "\nNumero: " + callerNumber + "\nContexto: " + context + "\nDestino: " + destination + "\nTipo: " + callDirection + "\nInterno: " + internal + "\n"); ..... > > > > El 22/09/2020 a las 06:38 a. m., Ehtasham Ul-Haq escribió: > > Hello Normando, > Thanks for your Answer, > However, when we park the callee, the call got disconnected. > > Malik Ehtasham, Lead ProjDev > > WWW:[image: domain2.png].expertflow.com <http://www.expertflow.com/> > FB: [image: FB-f-Logo__blue_29.png]/Expertflow > <https://www.facebook.com/Expertflow> LinkedIn: [image: linkedIn.png] > /company/expertflow <https://www.linkedin.com/company/expertflow> > Youtube: [image: YouTube-social-square_red_128px.png]/user/expertflow > <https://www.youtube.com/user/expertflow> Twitter: [image: twitter.JPG] > /Expertflow <https://twitter.com/Expertflow> > 361 Model Town Lahore Pakistan ; Mobile +92 3347815664,+92 334 4853664; > email, Cisco Spark and Google Talk: ehtasham.malik(a)expertflow.com > <andreas.stuber(a)expertflow.com>; Skype: > <http://andreas.stuber.expertflow.com/>*shani.awan3* > > > On Tue, Sep 22, 2020 at 1:48 AM Normando Hall via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> wrote: > >> >> >> >> ---------- Forwarded message ---------- >> From: Normando Hall <nhall(a)unixlan.com.ar> >> To: freeswitch-users(a)lists.freeswitch.org >> Cc: >> Bcc: >> Date: Mon, 21 Sep 2020 17:14:17 -0300 >> Subject: Re: [Freeswitch-users] How to Answer a Call from ESL java >> Hi. >> >> If you are using inbound ESl, you must park the call in your dialplan, >> and listen for "park" event in your ESL code. Then answer this call >> >> Regards >> >> El 21/09/2020 a las 04:38 p. m., Malik Ehtasham escribió: >> > Hello, >> > I want to answer the call from my web application. For Example, SIP >> Phone 1, >> > dial a call to SIP Phone B, now Phone B is ringing, now I want to >> answer the >> > call on Phone B, via my Web Interface, so there is a button on my >> interface >> > to answer the call. >> > To answer the call I am executing the below command, but this doesn't do >> > anything and the call keeps on ringing on phone B. Any >> > >> > EslMessage response = >> > client.sendSyncApiCommand("uuid_answer",Event_Header.get("Unique-ID")); >> > >> > >> > >> > -- >> > Sent from: http://freeswitch-users.2379917.n2.nabble.com/ >> > >> > >> _________________________________________________________________________ >> > >> > The FreeSWITCH project is sponsored by SignalWire >> https://signalwire.com >> > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> > Build your next product on our scalable cloud platform. >> > >> > Join our online community to chat in real time >> https://signalwire.community >> > >> > Professional FreeSWITCH Services >> > sales(a)freeswitch.com >> > https://freeswitch.com >> > >> > Official FreeSWITCH Sites >> > https://freeswitch.com/oss >> > https://freeswitch.org/confluence >> > https://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 >> > https://freeswitch.com >> >> >> >> >> ---------- Forwarded message ---------- >> From: Normando Hall via FreeSWITCH-users < >> freeswitch-users(a)lists.freeswitch.org> >> To: freeswitch-users(a)lists.freeswitch.org >> Cc: >> Bcc: >> Date: Mon, 21 Sep 2020 13:48:47 -0700 (PDT) >> Subject: Re: [Freeswitch-users] How to Answer a Call from ESL java >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales(a)freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://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 >> https://freeswitch.com > > > > > ---------- Forwarded message ---------- > From: Normando Hall via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Tue, 22 Sep 2020 09:48:12 -0700 (PDT) > Subject: Re: [Freeswitch-users] How to Answer a Call from ESL java > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://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 > https://freeswitch.com

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Re: [Freeswitch-users] How to Answer a Call from ESL java
by Ehtasham Ul-Haq 22 Sep '20

22 Sep '20

Hello Normando, Thanks for your Answer, However, when we park the callee, the call got disconnected. Malik Ehtasham, Lead ProjDev WWW:[image: domain2.png].expertflow.com <http://www.expertflow.com/> FB: [image: FB-f-Logo__blue_29.png]/Expertflow <https://www.facebook.com/Expertflow> LinkedIn: [image: linkedIn.png] /company/expertflow <https://www.linkedin.com/company/expertflow> Youtube: [image: YouTube-social-square_red_128px.png]/user/expertflow <https://www.youtube.com/user/expertflow> Twitter: [image: twitter.JPG] /Expertflow <https://twitter.com/Expertflow> 361 Model Town Lahore Pakistan ; Mobile +92 3347815664,+92 334 4853664; email, Cisco Spark and Google Talk: ehtasham.malik(a)expertflow.com <andreas.stuber(a)expertflow.com>; Skype: <http://andreas.stuber.expertflow.com/>*shani.awan3* On Tue, Sep 22, 2020 at 1:48 AM Normando Hall via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Normando Hall <nhall(a)unixlan.com.ar> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Mon, 21 Sep 2020 17:14:17 -0300 > Subject: Re: [Freeswitch-users] How to Answer a Call from ESL java > Hi. > > If you are using inbound ESl, you must park the call in your dialplan, > and listen for "park" event in your ESL code. Then answer this call > > Regards > > El 21/09/2020 a las 04:38 p. m., Malik Ehtasham escribió: > > Hello, > > I want to answer the call from my web application. For Example, SIP > Phone 1, > > dial a call to SIP Phone B, now Phone B is ringing, now I want to answer > the > > call on Phone B, via my Web Interface, so there is a button on my > interface > > to answer the call. > > To answer the call I am executing the below command, but this doesn't do > > anything and the call keeps on ringing on phone B. Any > > > > EslMessage response = > > client.sendSyncApiCommand("uuid_answer",Event_Header.get("Unique-ID")); > > > > > > > > -- > > Sent from: http://freeswitch-users.2379917.n2.nabble.com/ > > > > _________________________________________________________________________ > > > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > > Build your next product on our scalable cloud platform. > > > > Join our online community to chat in real time > https://signalwire.community > > > > Professional FreeSWITCH Services > > sales(a)freeswitch.com > > https://freeswitch.com > > > > Official FreeSWITCH Sites > > https://freeswitch.com/oss > > https://freeswitch.org/confluence > > https://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 > > https://freeswitch.com > > > > > ---------- Forwarded message ---------- > From: Normando Hall via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Mon, 21 Sep 2020 13:48:47 -0700 (PDT) > Subject: Re: [Freeswitch-users] How to Answer a Call from ESL java > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://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 > https://freeswitch.com

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Is it possible to parse Accept-Contact ?
by Mickael Hubert 22 Sep '20

22 Sep '20

Hi all, I want to be able to parse Accept-Contact header. *I have this configuration just to log header:* <condition field="${sip_h_Accept-Contact}" expression="(.+)" break="never"> <action application="log" data="INFO LOG: parse accept: ${sip_h_Accept-Contact}" /> </condition> *my logs:* Dialplan: sofia/INTERNAL/427842085(a)10.88.83.7 Regex (PASS) [variables] ${sip_h_Accept-Contact}() =~ /(.*)/ break=never Dialplan: sofia/INTERNAL/427842085(a)10.88.83.7 Action log(parse accept: ${sip_h_Accept-Contact}) 2020-09-21 17:34:09.726180 [NOTICE] switch_core_session.c:3017 Execute [depth=2] log(parse accept: ${sip_h_Accept-Contact}) EXECUTE [depth=2] sofia/INTERNAL/427842085(a)10.88.83.7 log(parse accept: ) sip_h_Accept_Contact variable is empty. *header received:* Accept-Contact: *;generic.test_id="test" is it possible to parse this header ? Do you have an idea about that ? thanks in advance

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Verto audio delay with opus
by Gregor Nanger 22 Sep '20

22 Sep '20

Hi! If I put Verto call on hold and unhold it, there is a one way audio delay, from verto client to endpoint. Other way is OK. And it gets more delay with each hold unhold. But it happens only with opus codec. What do you suggests to check it?

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Re: [Freeswitch-users] One-way audio but not video on 10.4
by David P 22 Sep '20

22 Sep '20

Thanks for confirming, Mike. To me, that sounds like FSv10 supports peer-reflexive candidates. In fact, I realize now that I've verified this feature recently. However, we still get one-way media at one of our test sites. A pcap with this display filter... ip.dst == <public IP number of my FS 10.4> ...shows only TLSv1.3 and TCP packets to port 8082, which indicate that the handshake succeeded but verto on Chrome85 never sent any media. When I test from my site with a different Win10 machine but also using Chrome85 and the same FS, my pcap reveals TLSv1.2 and media goes both ways. We're going to do additional tests by varying the client machine and client network, but if something so far suggests a reason for one-way audio, please let me know. ---------- Forwarded message ---------- > From: Mike Jerris <mike(a)freeswitch.org> > > If they start sending to us we will auto -adjust to the right ip after a > few packets. Its a bit slower to set up but should work. > > On Sep 15, 2020, at 7:36 PM, David P <davidswalkabout(a)gmail.com> wrote: > > Although we have STUN+TURN coturn (and FS) deployed in Oregon, a verto > user in San Francisco today made a call in which only (private) host > candidates were in the offer bundle SDP. > > They use IT-managed Ethernet, so any slowness in getting STUN is probably > in their IT management of the connection...and not something we can do > anything about unless our FS can support peer-reflexive candidates. > > They tell me their network has worked fine with Zoom, which is grating to > hear. Is there any prospect of FS supporting peer-reflexive candidates? > > On Tue, Sep 1, 2020 at 12:22 AM < > freeswitch-users-request(a)lists.freeswitch.org> wrote: > >> ---------- Forwarded message ---------- >> From: David P <davidswalkabout(a)gmail.com> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Bcc: >> Date: Mon, 31 Aug 2020 08:36:05 +1200 >> Subject: Re: [Freeswitch-users] One-way audio but not video on 10.4 >> >>> I strongly suspect the problem is due to FS choosing a different port >> than is offered. I've seen this a few times by comparing the offer SDP >> a=candidate entries with the address:port in FS log entries containing >> 'Choose rtp'. This won't work if the user is behind a firewall that allows >> incoming response only through ports opened for outgoing requests. >> >> Another thing I wonder about is when verto provides a BUNDLE offer, >> should it use the same address and port for both audio and video. I think >> I've seen that it doesn't, but maybe BUNDLE doesn't require it. >> >> --------- Forwarded message ---------- >>> From: Nathan Stratton <nathan(a)robotics.net> >>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Bcc: >>> Date: Sat, 29 Aug 2020 11:13:24 -0400 >>> Subject: Re: [Freeswitch-users] One-way audio but not video on 10.4 >>> Looks like your client is not hitting a stun server to get the public IP >>> address, however, I have been wondering why it's not possible to make >>> FreeSWITCH work without using STUN on the client as long as FreeSWITCH is >>> on a public IP. >>> >>> ><> >>> nathan stratton >>> >>> >>> On Sat, Aug 29, 2020 at 3:00 AM David P <davidswalkabout(a)gmail.com> >>> wrote: >>> >>>> The client is current verto running on current Chrome/Win10. >>>> >>>> The client's SDP shows srflx and relay candidates from Twilio at same >>>> network cost as the "private" host candidates, and FSv10.4 chooses the host >>>> candidates. >>>> >>>> ---------- Forwarded message ---------- >>>>> From: Mike Jerris <mike(a)freeswitch.org> >>>>> >>>>> If client still isn’t putting in non rfc1918 candidates in audio you >>>>> need to figure out why that is first. >>>>> >>>>> On Aug 26, 2020, at 7:17 PM, David P <davidswalkabout(a)gmail.com> >>>>> wrote: >>>>> >>>>> I've introduced Twilio STUN and TURN to our iceServers since then. All >>>>> have the same network cost in the SDP and FSv10.4 chooses a host. Verto >>>>> reports the remote stream arrived but getStats shows zero audio bytes in or >>>>> out, and it's silent to our user. I applied a display filter using our >>>>> Freeswitch's public IP ip.src == 52.xx.yy.zz or ip.dst == ip.src == >>>>> 52.xx.yy.zz and in addition to TLS I see some TCP ACKs from our user's >>>>> 10.0.0.189 but I don't know what else to look for. >>>>> >>>>> >>>>> ---------- Forwarded message ---------- >>>>>> From: Mike Jerris <mike(a)freeswitch.org> >>>>>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>>>>> Cc: >>>>>> Bcc: >>>>>> Date: Wed, 26 Aug 2020 16:19:56 -0400 >>>>>> Subject: Re: [Freeswitch-users] One-way audio but not video on 10.4 >>>>>> >>>>> >>>>> >>>>>> Configure stun in your client so you get proper candidates that are >>>>>> reachable. >>>>>> >>>>>> On Aug 19, 2020, at 6:18 PM, David P <davidswalkabout(a)gmail.com> >>>>>> wrote: >>>>>> >>>>>> Yesterday I had a verto call in which the <video> contained a remote >>>>>> echo (which is how we configure it) but there was no audio out nor in >>>>>> (confirmed by getStats showing zero audio bytes out or in). >>>>>> >>>>>> I was using: >>>>>> FreeSWITCH Version >>>>>> 1.10.4-release+git~20200805T110119Z~133fc2c870~64bit (git 133fc2c >>>>>> 2020-08-05 11:01:19Z 64bit) >>>>>> from Chrome84/Win10 >>>>>> >>>>>> I noticed the video candidates included a relay but the audio ones >>>>>> didn't: >>>>>> >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e m=audio 62058 UDP/TLS/RTP/SAVPF >>>>>> 111 103 104 9 0 8 106 105 13 110 112 113 126 >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e c=IN IP4 192.168.208.1 >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=rtcp:9 IN IP4 0.0.0.0 >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=candidate:1252570982 1 udp >>>>>> 2122260223 192.168.208.1 62058 typ host generation 0 network-id 1 >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=candidate:854729925 1 udp >>>>>> 2122194687 172.20.4.169 62059 typ host generation 0 network-id 2 >>>>>> network-cost 10 >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=candidate:69946262 1 tcp >>>>>> 1518280447 192.168.208.1 9 typ host tcptype active generation 0 network-id 1 >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=candidate:2087835701 1 tcp >>>>>> 1518214911 172.20.4.169 9 typ host tcptype active generation 0 network-id 2 >>>>>> network-cost 10 >>>>>> >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e m=video 13143 UDP/TLS/RTP/SAVPF >>>>>> 96 97 98 99 100 101 102 122 127 121 125 107 108 109 124 120 123 119 114 115 >>>>>> 116 >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e c=IN IP4 35.160.47.89 >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=rtcp:9 IN IP4 0.0.0.0 >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=candidate:1252570982 1 udp >>>>>> 2122260223 192.168.208.1 62060 typ host generation 0 network-id 1 >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=candidate:854729925 1 udp >>>>>> 2122194687 172.20.4.169 62061 typ host generation 0 network-id 2 >>>>>> network-cost 10 >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=candidate:69946262 1 tcp >>>>>> 1518280447 192.168.208.1 9 typ host tcptype active generation 0 network-id 1 >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=candidate:2087835701 1 tcp >>>>>> 1518214911 172.20.4.169 9 typ host tcptype active generation 0 network-id 2 >>>>>> network-cost 10 >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e a=candidate:3112167340 1 udp >>>>>> 25042943 35.jj.kk.mm 13143 typ relay raddr 210.55.134.98 rport 53504 >>>>>> generation 0 network-id 2 network-cost 10 >>>>>> >>>>>> The relay was chosen for video, and it appeared that the same >>>>>> candidate was going to be chosen for audio, but then an "AUDIO RTP" >>>>>> suggests it chose a private IPv4. And it seems that private IPv4 wasn't >>>>>> treated as a peer-reflexive candidate, so the media couldn't be routed... >>>>>> >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e 2020-08-19 05:14:33.753568 >>>>>> [DEBUG] switch_core_media.c:4338 Choose rtp candidate, index 2, >>>>>> 35.jj.kk.mm:13143 >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e 2020-08-19 05:14:33.753568 >>>>>> [DEBUG] switch_core_media.c:4104 >>>>>> verto.rtc/1200_a6dd33a4-1def-447c-a62a-ae807bb92da6_v-1*s-4 choosing family >>>>>> v4 >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e 2020-08-19 05:14:33.753568 >>>>>> [DEBUG] switch_core_media.c:4349 Choose same candidate, index 0, for rtcp >>>>>> based on rtcp-mux attribute 35.jj.kk.mm:13143 >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e 2020-08-19 05:14:33.753568 >>>>>> [DEBUG] switch_core_media.c:4401 setting remote video ice addr to index 2 >>>>>> 35.jj.kk.mm:13143 based on candidate >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e 2020-08-19 05:14:33.753568 >>>>>> [DEBUG] switch_core_media.c:4436 Setting remote rtcp video addr to >>>>>> 35.jj.kk.mm:13143 based on candidate >>>>>> f6e23797-4ad2-b761-2a6d-97846c122d7e 2020-08-19 05:14:33.753568 >>>>>> [DEBUG] switch_core_media.c:8663 AUDIO RTP >>>>>> [verto.rtc/1200_a6dd33a4-1def-447c-a62a-ae807bb92da6_v-1*s-4] 10.0.0.100 >>>>>> port 26610 -> 192.168.208.1 port 62058 codec: 111 ms: 20 >>>>>> >>>>>> Am I interpreting the log correctly? >>>>>> >>>>>> Is there a way to avoid this? >>>>>> >>>>>> Note that in /usr/local/freeswitch/conf/autoload_configs/acl.conf.xml >>>>>> we already deny private IPv4 (although I think we should allow them so they >>>>>> could be used peer-reflexively): >>>>>> <list name="disallow-privateIPv4" default="allow"> >>>>>> <node type="deny" host="10.0.0.0" mask="255.0.0.0"/> >>>>>> <node type="deny" host="172.16.0.0" mask="255.240.0.0"/> >>>>>> <node type="deny" host="192.168.0.0" mask="255.255.0.0"/> >>>>>> </list> >>>>>> >>>>>> >

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answer command
by Maxim Tsvetov 21 Sep '20

21 Sep '20

Hello, I found in WIKI list of supported commands http://wiki.freeswitch.org/wiki/Mod_commands. I need command "answer" but it doesn't exist (Freeswitch 1.04, Win32) That's what Freeswitch replies for this command: "answer: Command not found!" Сould you please help? Regards, Maxim Tsvetov

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How to Answer a Call from ESL java
by Malik Ehtasham 21 Sep '20

21 Sep '20

Hello, I want to answer the call from my web application. For Example, SIP Phone 1, dial a call to SIP Phone B, now Phone B is ringing, now I want to answer the call on Phone B, via my Web Interface, so there is a button on my interface to answer the call. To answer the call I am executing the below command, but this doesn't do anything and the call keeps on ringing on phone B. Any EslMessage response = client.sendSyncApiCommand("uuid_answer",Event_Header.get("Unique-ID")); -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

2 1

Experience with FreeSWITCH on capable servers?
by Sam van Herwaarden 19 Sep '20

19 Sep '20

Hi all, I'm working on a FreeSWITCH setup that places phone calls over a SIP line where we have a capacity of 3000 channels. What I'm not sure about yet is if this can be done on a single server or if we will need to run multiple servers in parallel. From what I've read FreeSWITCH parallelizes well. Does anyone here have experience running FreeSWITCH on powerful servers and how far can you get with a single host? I'm thinking about something like a Threadripper setup with 32 cores or so and let's say 64GB of RAM (but if we would need more that's probably possible as well). Any other advice about hardware requirements for FreeSWITCH setups on high capacity systems would be welcome as well! Thank you, Sam

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Freeswitch Fax2Mail App
by Tamer Higazi 19 Sep '20

19 Sep '20

Hi people, As I enjoy Freeswitch very much, I decided to develop an application in python that sends incoming FAX messages directly out to a set of a number of recipients. I published my application here: https://github.com/thigazi/PyFax2Mail The app uses starttls as authentication method only. Further changes will follow. Have fun, for any feedback I am happy to receive any response. best, Tamer

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